| 00:00.02 | robl^ | dlynes_office: but are the cards designed for printing using a laser printer? |
| 00:00.04 | agboris | <file> actually i want say is that i have made all settings that i had made before...even i am using same ip...and calls should come that particular ip...and should be recieved without changing any settings |
| 00:00.18 | dlynes_office | robl^: not afaik |
| 00:00.23 | robl^ | dlynes_office: http://www.desi.com/ |
| 00:00.36 | dlynes_office | robl^: however, if you talk to david sayson at Sayson Technologies |
| 00:00.44 | file | agboris: but they aren't, despite everything being the same - so talk to your provider if everything is the same on your side and it shows you registered fine |
| 00:00.48 | dlynes_office | robl^: he can set you up in the partner program for aastra |
| 00:00.56 | file | agboris: work it out with them and see if they can help you figure out what is up |
| 00:00.57 | dlynes_office | erm ....sorry...it's not called Sayson Technologies anymore |
| 00:01.06 | dlynes_office | they're a division of Aastra now |
| 00:01.13 | agboris | ok |
| 00:01.36 | dlynes_office | robl^: and then you can get in on cool stuff like rebranding aastra phones with your own branding and you can get extra cards, and that kinda thing |
| 00:01.59 | robl^ | dlynes_office: is there a minimum requirement for joinging? |
| 00:02.11 | dlynes_office | robl^: well, not for the rebranding, i don't think |
| 00:02.14 | *** part/#asterisk jbalcomb (n=JimBalco@m495e36d0.tmodns.net) |
| 00:02.25 | dlynes_office | robl^: but the other partner-related stuff there are minimum volumes |
| 00:03.13 | robl^ | dlynes_office: ahh.. I am not going to have any sort of large volume in the near future. maybe 15 phones a year |
| 00:03.17 | agboris | how to change the sip port in asterisk |
| 00:03.40 | dlynes_office | robl^: his number is 604-730-1842, option '1' - their business hours are 10am to 3pm, PDT |
| 00:03.42 | file | sip.conf, bindport option in the general context |
| 00:04.02 | dlynes_office | robl^: ah...why only 15 phones a year? |
| 00:04.20 | *** part/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
| 00:04.54 | dlynes_office | voip systems are pretty easy to sell |
| 00:04.58 | dlynes_office | they almost sell themselves |
| 00:05.00 | *** join/#asterisk iq (n=iq@unaffiliated/iq) |
| 00:05.24 | dlynes_office | especially when they're asterisk-based |
| 00:05.30 | robl^ | dlynes_office: I am not a telecom person by trade.. only hobby at the moment. I have to asterisk systems I support. 10 phone and 4 phones respectively. ;-) |
| 00:05.47 | dlynes_office | robl^: i'm not a telecom person either |
| 00:05.52 | Corydon-w | mog: we should both get an Ethernet card for our Apple IIe's, then put up a VPN, so we can communicate back and forth between them |
| 00:05.54 | iq | Hi |
| 00:05.56 | dlynes_office | robl^: i'm a programmer/system administration geek |
| 00:06.25 | Qwell[] | Corydon-w: You're just sick, you know that, right? |
| 00:06.36 | robl^ | dlynes_office: same here. I provide sysadmin / tech support for a large law firm. |
| 00:06.48 | Corydon-w | This from the guy who spooned me in Las Vegas |
| 00:06.53 | dlynes_office | robl^: ah...so you're not self-employed then...you're an employee? |
| 00:06.55 | Qwell[] | You wish it wa Vegas |
| 00:07.04 | Corydon-w | Oh, right, San Jose |
| 00:07.04 | dlynes_office | yeah...it was really reno |
| 00:07.14 | Qwell[] | (I wish it was Vegas too...it would've stayed there. :P) |
| 00:07.26 | Corydon-w | Well, you might spoon me in Vegas yet. ;-) |
| 00:07.36 | Qwell[] | note to self: Don't go to defcon |
| 00:07.46 | Qwell[] | ...ever |
| 00:08.00 | robl^ | dlynes_office: currently. I did the slef employed thing on and off. but things really dried up for a while and had too many bills so I re-joined the ramnks of abused IT staff |
| 00:08.25 | dlynes_office | ah...that would explain why you're not planning on selling many units then |
| 00:08.28 | file | Qwell[]: also don't ever go to Nashville since Corydon is near there... he might sense your presence |
| 00:08.43 | robl^ | dlynes_office: but things can change. ;-) |
| 00:08.44 | dlynes_office | asterisk systems are pretty easy to sell...just don't tell your customers it's asterisk |
| 00:08.47 | Qwell[] | note to self: Don't pick file up from airport |
| 00:08.54 | dlynes_office | tell them it's a black box with phone lines and voicemail |
| 00:09.01 | file | Qwell[]: don't travel with me either |
| 00:09.06 | file | mucho grande bad luck |
| 00:09.21 | Corydon-w | Yeah, file might be feeling amorous at the Toronto airport |
| 00:09.30 | Qwell[] | :D |
| 00:09.40 | dlynes_office | Corydon-w: there's lots of queers in Toronto...I'm sure file's not the only one |
| 00:09.50 | file | I don't live in Toronto |
| 00:10.02 | Corydon-w | I prefer Texas for their queers and steers |
| 00:10.15 | robl^ | dlynes_office: yeah. its just a matter of getting the right contacts to get started. after things calm down at the office, I nay get things in line to try to do some more telecom stuff |
| 00:10.26 | dlynes_office | robl^: ah |
| 00:10.30 | robl^ | *waves Texas flag* |
| 00:10.32 | file | I just travel through Toronto usually for flights... |
| 00:10.44 | dlynes_office | file: ummm...hmmmm....sure, sure |
| 00:10.53 | Corydon-w | Sure, sure... is that the reason for the 8 hour layover? |
| 00:11.00 | dlynes_office | file: we know you go check out Wellesley Street every time you're there |
| 00:11.00 | file | that was in Newark |
| 00:11.11 | file | dlynes_office: I have never actually been around Toronto |
| 00:11.16 | file | should go sometime |
| 00:12.12 | *** join/#asterisk Trazz (i=Trazz@c-67-163-92-37.hsd1.il.comcast.net) |
| 00:12.28 | *** join/#asterisk Trazzz (i=Trazz@c-67-163-92-37.hsd1.il.comcast.net) |
| 00:14.30 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 00:15.10 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 00:15.32 | Nugget | gonna go see dollywood and twitty city? |
| 00:15.49 | file | Corydon-w: I bet you would... |
| 00:16.04 | Corydon-w | Dollywood is 3 hours east of here |
| 00:16.48 | Corydon-w | Well, probably about time to head home |
| 00:17.01 | file | yes.. go |
| 00:17.11 | Qwell[] | Corydon-w: You scared him |
| 00:17.12 | Corydon-w | Bitch. :-P |
| 00:17.24 | file | be gone! |
| 00:17.55 | dlynes_office | I bet file just loves a nice, strong man |
| 00:17.57 | Corydon-w | I'll be back on to pester^H^H^H^H^H^Hentertain you in an hour or so |
| 00:18.15 | agboris | file: i have talked to my provider they are saying that there is no issue at their end |
| 00:18.39 | agboris | and even the said they dont have support for asterisk |
| 00:18.52 | dlynes_office | hehe |
| 00:19.00 | dlynes_office | agboris: that was your first mistake |
| 00:19.14 | dlynes_office | agboris: don't tell them you're using asterisk, or it's automatically your fault that it's not working |
| 00:19.29 | agboris | Hmmmmmmm |
| 00:19.32 | *** join/#asterisk bkw_ (n=brian@asterisk/friend-and-developer/bkw) |
| 00:19.45 | dlynes_office | agboris: mostly because those twits usually don't even know what asterisk is |
| 00:19.55 | agboris | But dear, I know my other asterisk with same configurations is workign well |
| 00:20.17 | dlynes_office | agboris: probably not the same configuration, or it would be working |
| 00:20.35 | dlynes_office | agboris: rm -rf /etc/asterisk ; scp -r user@host:/etc/asterisk . |
| 00:20.37 | file | "Can you confirm you are sending an INVITE to me, and if so - what IP address and port are they being sent to?" |
| 00:20.43 | dlynes_office | that'll confirm that it's the same |
| 00:21.18 | Qwell[] | O M G |
| 00:22.16 | agboris | it will just wipp out my all settings dlyne |
| 00:22.40 | dlynes_office | agboris: you insisted they were the same |
| 00:22.46 | dlynes_office | agboris: that'll ensure they're the same |
| 00:23.09 | file | something is rotten in the state of Denmark |
| 00:23.38 | tessier_ | Sorry, I farted. |
| 00:23.39 | agboris | yes...he is quite busy that why he is .......... |
| 00:23.50 | dlynes_office | agboris: you can also try tar jcvf /home/username/asterisk-etc.tar.bz2 /etc/asterisk ; rm -rf /etc/asterisk ; scp -r username@hostname:/etc/asterisk /etc |
| 00:24.05 | dlynes_office | agboris: if you want to make a backup first |
| 00:24.49 | agboris | so deploying backup can help me out form this issue |
| 00:24.58 | agboris | i have backup of my old pbx |
| 00:25.11 | dlynes_office | agboris: no, but you said you had another machine where everything was working just peachy keen |
| 00:25.14 | file | dlynes_office: you're taking over |
| 00:25.35 | fuser | i want to work at IBM... in something like, the spyware spreading department or something |
| 00:25.47 | Qwell[] | fuser: sales? why? |
| 00:25.54 | dlynes_office | file: ? |
| 00:25.58 | agboris | so dlynes_office this not the way u are thinking .......... |
| 00:26.11 | agboris | file ? |
| 00:26.21 | fuser | becuase i have issue Qwell[] |
| 00:26.37 | robl^ | yo bkw_! |
| 00:26.39 | Qwell[] | ast_mutex_lock(bkw_->lock) |
| 00:26.39 | fuser | i threw a wifi sip phone from the 12th floor of an office today |
| 00:27.42 | robl^ | fuser: that was you?!!?!? that phone smashed the window in my car! |
| 00:27.57 | agboris | file] i need to ask few things i u can help.... |
| 00:28.01 | fuser | robl^: dont worry, the phone was broken anyway |
| 00:28.03 | fuser | no harm done |
| 00:28.51 | file | agboris: ask questions and they may get answered |
| 00:29.00 | agboris | ok |
| 00:31.21 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:33.24 | andymul | Anyone interested in some PHP/Asterisk work please PM me |
| 00:38.39 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.88) |
| 00:38.52 | fuser | pee ache pee |
| 00:39.12 | fuser | how bout asterisk on rails |
| 00:39.18 | fuser | or, asterisk on roids |
| 00:40.56 | *** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie) |
| 00:40.58 | *** join/#asterisk trbldwine (i=troubled@71.194.161.170) |
| 00:41.26 | fuser | habanero flavored doritos rock. they make it hard to write a dial plan quickly. |
| 00:42.06 | file | blame the Doritos, uh huh |
| 00:42.15 | *** part/#asterisk dudes (n=dudes@71-87-34-39.dhcp.stcd.mn.charter.com) |
| 00:42.20 | fuser | haha, i swear it was the chips! |
| 00:42.35 | fuser | had nothing to do with the budweiser... honest... |
| 00:42.57 | file | I totally believe you |
| 00:45.45 | watchy2 | im fat |
| 00:45.46 | ariel_ | or the Cigar? |
| 00:45.56 | ariel_ | evening everyone |
| 00:46.40 | fuser | woohoo! i get to spend tomorrow building out a giganitic coast to coast vpn instead of writing xml into polycom config files |
| 00:46.58 | ariel_ | andymul, I would guess there are a few people interested in your project. I would also think that the asterisk-biz would be the better place to ask. |
| 00:47.34 | watchy2 | fuser: can i come help |
| 00:47.43 | fuser | i dont know, you are fat |
| 00:47.54 | watchy2 | i promise ill try to be less fat around you |
| 00:48.00 | fuser | might not be able to afford the meals |
| 00:48.11 | watchy2 | ill pay my own food and labor |
| 00:48.15 | fuser | watchy2: if i need help i'll call you |
| 00:48.27 | watchy2 | sweet. |
| 00:48.38 | watchy2 | where wuold i have to fly to |
| 00:48.58 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-42-234.losaca.adelphia.net) |
| 00:49.01 | fuser | my pbx is going to be a hub for 28 locations from florida to california inside of some of at&t's non-routable 'public' address space on the 12. |
| 00:49.28 | fuser | it will be the gateway, firewall, vpn server and a whole host of other shit |
| 00:49.41 | fuser | not to mention transcoding 6 T1's worth of calls all day |
| 00:49.46 | watchy2 | is it a super awesome pbx |
| 00:49.55 | vader-- | it's the gibson |
| 00:50.01 | vader-- | hack the gibson |
| 00:50.13 | watchy2 | i need to get a entry level job in asterisk shit |
| 00:50.29 | vader-- | watchy there isn't exactly alot of places looking for asterisk administrators |
| 00:50.29 | fuser | it has 2 quad port t1 cards and 2 of those wcte24xp amphenol connections, the 24 port fxs |
| 00:50.30 | watchy2 | i put in 1 asterisk phone system here but in my small town aint much work in it |
| 00:50.35 | ariel_ | fuser, are you going to create your own MPLS network? |
| 00:50.43 | fuser | ariel_: you got it |
| 00:50.50 | vader-- | watchy how long did it take you to set it up? |
| 00:51.02 | vader-- | im setting up my first one now and it's taken me a couple of months |
| 00:51.08 | vader-- | but i haven't been working on it every day |
| 00:51.09 | watchy2 | dont make me beg vader |
| 00:51.30 | watchy2 | vader: weeks but im still working on it for the company u know? bugs, features etc |
| 00:51.37 | vader-- | ya |
| 00:51.45 | watchy2 | i probably got lots more time to do it |
| 00:51.54 | vader-- | i spent alot of stupid shit |
| 00:51.56 | watchy2 | whats sad is i could do a phone system the way they got it now in afew hours |
| 00:52.02 | vader-- | going back and retweaking shit |
| 00:52.06 | watchy2 | yea |
| 00:52.12 | watchy2 | you using analog/ |
| 00:52.17 | vader-- | sip and some analog |
| 00:52.23 | vader-- | im using 60 cisco 7940G phones |
| 00:52.24 | fuser | ariel_: how is that system going? |
| 00:52.29 | watchy2 | what kinda cards you connecting to pstn? |
| 00:52.30 | ariel_ | well wed. I move off an Avaya an start a full strata-dailer to an Asterisk setup...All over MPLS |
| 00:52.39 | vader-- | and like 20 analog channels provided by a wtc2400p board |
| 00:52.46 | fuser | im a little anxious about this. its taken at&t forever to get their outsourcing straight.... |
| 00:52.47 | ariel_ | fuser, it works |
| 00:53.06 | fuser | good enough! |
| 00:53.12 | watchy2 | vader: i got poly 501s and 601s, about 30, adding 30 sipura ATAs |
| 00:53.31 | vader-- | nice |
| 00:53.31 | fuser | my main box is idle atm but we are gonna knock over the main network this weekend |
| 00:53.31 | ariel_ | fuser, at least your doing it with just one vendor. |
| 00:53.33 | vader-- | im hinde sight i would of probably went with poly's |
| 00:53.36 | vader-- | instead of cisco |
| 00:53.37 | watchy2 | the ATAs are going in bunkers where they build explosives so they wanted cheap walmart phones haha |
| 00:53.42 | fuser | haha good point |
| 00:53.43 | ariel_ | we have it via Telcove, XO, and Paetec |
| 00:53.45 | vader-- | or atleast go with cisco's newer line of phones |
| 00:53.52 | watchy2 | u know i bought 7960 for testing 2 of em and played with them, but i didnt like them |
| 00:54.07 | watchy2 | i ordered polys for this co without playing with them and havent regreted it one bit |
| 00:54.09 | fuser | watchy2: you can do alot with those phones |
| 00:54.09 | vader-- | they are nice |
| 00:54.17 | fuser | watchy2: theres a project out to check your pop3 email on one |
| 00:54.26 | fuser | services, services, services |
| 00:54.27 | watchy2 | on polys or cisco? |
| 00:54.30 | fuser | cisco |
| 00:54.36 | vader-- | cisco phones are powerful phones |
| 00:54.36 | fuser | im sure you could hack up the same thing for polycom |
| 00:54.39 | *** join/#asterisk javar (n=javar@200.118.174.253) |
| 00:54.39 | watchy2 | yea you can do some insane stuff on the cisco |
| 00:54.44 | fuser | the api is so much more readily available |
| 00:55.09 | fuser | we change the soft keys on the polycoms all the time but havent had the time to do anything with services |
| 00:55.26 | watchy2 | these people i set this system up for love the polys |
| 00:55.40 | fuser | who really wants weather on their polycom which is next to a 2000 dollar dell with a 24 inch LCD |
| 00:55.49 | watchy2 | yea |
| 00:55.57 | watchy2 | id rather have porn on my phone |
| 00:56.28 | *** join/#asterisk Defraz (n=t0tal@fw.centrisys.com) |
| 00:56.33 | fuser | i think some of the women customers might disagree with that feature set |
| 00:56.36 | fuser | /some/ |
| 00:56.51 | watchy2 | well we could put dilbert comics on them |
| 00:57.18 | vader-- | dilbert comics would be cool |
| 00:57.43 | watchy2 | im on a diet so im gonna go eat fajitas in a minute with no rice and beans :( |
| 00:57.52 | fuser | unless there is no pc near a phone like that i cant really see any marketability for services |
| 00:58.13 | watchy2 | yea me either |
| 00:58.19 | vader-- | im actually using the services button on my cisco phones to display a list of extensions for secretaries and the extensions' availability |
| 00:58.27 | vader-- | like if the person is on the phones, etc |
| 00:58.27 | ariel_ | fuser, are you in Florida? |
| 00:58.37 | fuser | vader--: that stuff is native to polycom phones |
| 00:58.38 | vader-- | ya |
| 00:58.39 | fuser | ariel_: houston |
| 00:58.42 | ariel_ | ahh |
| 00:58.44 | vader-- | trust me i know |
| 00:59.08 | robl^ | fuser: you must be one of my neighbors ;-) |
| 00:59.35 | fuser | robl^: oh yeah? what part of town you in? |
| 00:59.42 | fuser | im off the beltway and westheimer |
| 00:59.45 | fuser | ;P |
| 00:59.54 | robl^ | fuser: west die. almost to Katy |
| 00:59.57 | vader-- | ok time to leave work |
| 01:00.14 | fuser | robl^: maybe you can wear my clothes and slackware hat to work tomorrow posing as me |
| 01:01.09 | fuser | just go around saying 'yeah mpls is cool we 0wn0r at&t f00! iptables MANG!!!" |
| 01:01.12 | robl^ | fuser: very close. westhiemer.. jsut on the east side of bw8 |
| 01:01.15 | fuser | then i can go fishing |
| 01:01.47 | fuser | robl^: you might need to bust out some serious sed on the polycom files though |
| 01:02.17 | robl^ | I like my Aastra... ;-) |
| 01:02.22 | fuser | robl^: what do you do? know about asterisk and linux and networking and iptables? wanna job? |
| 01:03.25 | robl^ | fuser: I'm a sysadmin for a law firm.. I know Linx and Asterisk.. and basic iptables. ;-) |
| 01:03.43 | ariel_ | wow asterisk, linux and networking IPtables. all easy stuff to know. |
| 01:04.06 | fuser | yep |
| 01:04.20 | fuser | you'd be suprised how hard it is to find help though |
| 01:04.34 | fuser | i master the image AND pull cable |
| 01:04.48 | robl^ | houston is filled with "techs" that can barely change a screen saver on XP |
| 01:05.07 | fuser | robl^: you got that right. you should see FEMA's IT department here |
| 01:05.20 | fuser | these guys quick foot locker to work for FEMA so they wouldnt have to talk to people |
| 01:06.04 | robl^ | I really need a spel checker for IRC.. ;-) |
| 01:06.36 | fuser | i type too fast to care |
| 01:07.23 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 01:07.27 | fuser | Governmen Job Posting: FEMA is in need of a Senior Sytem Administrator for their call center. Minimum 1 yr. high school preferred. |
| 01:07.38 | robl^ | lol |
| 01:07.49 | Snake-Eyes | lol |
| 01:07.50 | fuser | Must know how to find the 'Control Panel' and 'My Computer' |
| 01:08.13 | fuser | Pay DoE starting at 7.50/hr. |
| 01:08.28 | ariel_ | wow we start people here in the call center at 10. |
| 01:08.28 | Faithful | Can you adapt a plantronics headset with an RJ type plug to work on a Linksys phone? |
| 01:08.32 | fuser | you guys think im joking. this guy calls me when he cant print. |
| 01:08.49 | litage | Faithful: it's a matter of finding the appropriate adapter |
| 01:08.50 | ariel_ | fuser, I belive |
| 01:08.59 | Faithful | with a name like fuser it is no wonder |
| 01:09.18 | ariel_ | Faithful, plantronics head sets work fine. on Polycoms. But I have no Linksys to test with. |
| 01:09.32 | fuser | ~fuser |
| 01:09.34 | jbot | [fuser] a handy command to 'identify processes using files or sockets' or if you find a port number that you want more info on, do: fuser -n tcp <portnum> ., or some people think it's is short for 'what the f**k is using that damn file!!!!! |
| 01:09.37 | Faithful | I guess I might be able to make up an adapter |
| 01:09.52 | fuser | ahaha... love it... |
| 01:10.36 | fuser | well, time to kill people in the ae_sniper_challenge |
| 01:13.09 | *** join/#asterisk |dennis| (n=dennis@vsat-148-64-30-39.c050.t7.mrt.starband.net) |
| 01:14.24 | *** join/#asterisk Ciber311 (n=Ciber@user-1087e94.cable.mindspring.com) |
| 01:14.37 | *** join/#asterisk yxa (n=diablo@58.185.90.101) |
| 01:17.36 | *** join/#asterisk Trazz (n=traderz@c-67-163-92-37.hsd1.il.comcast.net) |
| 01:24.08 | Ciber311 | anyone in here compile asterisk on osx? |
| 01:43.23 | *** join/#asterisk _GiGi_ (i=gigi@disc.more.pl) |
| 01:43.33 | _GiGi_ | hello |
| 01:43.58 | *** join/#asterisk livinded (n=livinded@cpe-24-24-186-88.socal.res.rr.com) |
| 01:45.08 | _GiGi_ | i have some trouble in AGI script when i run record file. Afrer record my script die.. in older version of asterisk its work fine... |
| 01:45.26 | livinded | would there a reason why 1 of my ipkall numbers goes directly to my asterisk box and another wouldn't? They both have the same sip proxy but different numbers. The one that isn't working, doesn't even hit the box. |
| 01:48.09 | livinded | oh wrong context :D |
| 01:50.05 | *** join/#asterisk foo (n=foo@unaffiliated/foo) |
| 01:54.16 | russellb | Ciber311: yeah, i do some development on my powerbook |
| 01:57.58 | *** join/#asterisk niteowldave (n=dave@203.82.162.41) |
| 01:58.11 | niteowldave | file: do you have a minute |
| 01:58.12 | russellb | many of the developers do ... |
| 01:58.20 | _GiGi_ | i have some trouble in AGI script when i run record file. Afrer record my script die.. in older version of asterisk its work fine... |
| 02:06.12 | niteowldave | I have a problem with t.38 passthrough in trunk, anybody here got any experience getting this working? |
| 02:11.29 | Ciber311 | russellb: will i run into any issues trying to compile it? 10.4.7 here |
| 02:11.35 | *** join/#asterisk ceL_ (i=cel@69-166-132-70.clvdoh.adelphia.net) |
| 02:12.21 | russellb | Ciber311: you shouldn't ... you'll need the xcode tools (so you have gcc and such) |
| 02:12.25 | *** join/#asterisk benjk_ (n=benjamin@f8a01-0357.din.or.jp) |
| 02:13.05 | *** part/#asterisk ceL_ (i=cel@69-166-132-70.clvdoh.adelphia.net) |
| 02:20.05 | *** join/#asterisk trelane (i=trelane@unaffiliated/trelane) |
| 02:24.35 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 02:28.00 | *** join/#asterisk RF_MIA (n=redfone@68-235-157-35.miamfl.adelphia.net) |
| 02:33.18 | *** part/#asterisk RF_MIA (n=redfone@68-235-157-35.miamfl.adelphia.net) |
| 02:35.47 | *** join/#asterisk rajiv_ (n=irc@gentoo/developer/rajiv) |
| 02:37.19 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 02:43.38 | *** join/#asterisk pengyong (n=lala@218.93.158.79) |
| 02:55.52 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 02:57.52 | shmaltz | hi everyone |
| 02:58.04 | livinded | hey |
| 02:58.21 | shmaltz | hi |
| 02:58.42 | russellb | hi |
| 02:58.55 | livinded | why is it that support channels are always so quiet? |
| 02:58.57 | shmaltz | hi russellb |
| 02:59.07 | file | 42 |
| 02:59.10 | shmaltz | livinded, it's not |
| 02:59.24 | shmaltz | don't you see the chatter thats going on here for the last 30 seconds or so |
| 02:59.29 | shmaltz | :P |
| 02:59.34 | livinded | shmaltz: sure it is, there are hundreds of people in them and nobody ever talks |
| 02:59.48 | Juggie | because its evening in most of the usa |
| 02:59.51 | russellb | livinded: it's usually quite lively |
| 02:59.54 | shmaltz | livinded, yes they do, on the phones :P |
| 02:59.55 | russellb | tonight has been creepy |
| 02:59.59 | JT | most people want support, not to give it |
| 03:00.04 | russellb | pfft... i hate using phones |
| 03:00.08 | Juggie | most of the convo happens during north amercain business hours |
| 03:00.14 | JT | and computer types don't like logging off |
| 03:00.18 | shmaltz | russellb, then why work with a PBX system????? |
| 03:00.28 | livinded | i guess i'm not in here enough to notice, i usually only come when i need help |
| 03:00.32 | file | phones are evil |
| 03:00.37 | file | you pick up and then have to talk to another person |
| 03:00.50 | russellb | shmaltz: i don't know ... because there are a lot of interesting problems to solve, i guess |
| 03:00.57 | Faithful | Does anyone know how to take advantage of a VSP's SMS services from asterisk or such? |
| 03:00.58 | russellb | it's fun to work on ... not use :) |
| 03:01.00 | livinded | file: thats what call rejection is for |
| 03:01.08 | shmaltz | file, not only that it take away one hand and one ear, and the other person notices imediatly if you are doing something else :P |
| 03:01.18 | Juggie | i only answer when i feel like being helpful |
| 03:01.27 | Juggie | alof of times i see questions and i dont want to help |
| 03:01.37 | Juggie | because its obvious the person has made no attempt to solve it on his own |
| 03:01.39 | shmaltz | Faithfuk, whats VSP? |
| 03:01.45 | shmaltz | ~VSP |
| 03:01.48 | JT | voice service provider |
| 03:02.07 | shmaltz | Faithful, you in the UK? |
| 03:02.16 | Juggie | * can deliver SMS over isdn in some situations |
| 03:02.39 | livinded | Juggie: or you can route sms through free online services :D |
| 03:02.46 | shmaltz | Juggie, I think over POTS as well if it's supported as is the case in the uk under BT |
| 03:02.48 | livinded | why pay for your own line to do it |
| 03:03.04 | Juggie | http://www.voip-info.org/wiki/view/SMS |
| 03:04.14 | shmaltz | http://www.voip-info.org/wiki/view/Asterisk+cmd+Sms |
| 03:04.35 | carrar | hahah |
| 03:04.40 | carrar | get rich quick |
| 03:04.43 | carrar | thats funny |
| 03:04.45 | shmaltz | livinded, no such thing. |
| 03:05.50 | livinded | shmaltz: ya but I can wish can't I? |
| 03:06.04 | shmaltz | livinded, sure you can wish |
| 03:06.12 | shmaltz | but the wish can go without asterisk in it |
| 03:06.23 | shmaltz | I think with the lottery you actualy have better chances |
| 03:07.40 | livinded | i'm not old enough to play the lottery |
| 03:08.34 | shmaltz | livinded, how young are you? |
| 03:08.42 | livinded | 17 |
| 03:09.03 | shmaltz | cool, asterisk has got some teenage fans |
| 03:10.10 | livinded | i wish my parents were fans, i've been trying to convinve them to let me connect it to our punchdown block and run the phones through it |
| 03:10.22 | livinded | every house should have an ivr and zapateller :P |
| 03:11.02 | shmaltz | livinded, you realy are young, it's overdone for homes |
| 03:12.00 | *** join/#asterisk andymul (n=andymul@cpe-69-203-217-237.nyc.res.rr.com) |
| 03:12.04 | andymul | Anyone interested in some PHP/Asterisk work please PM me |
| 03:12.18 | shmaltz | andymul, for what purpose? |
| 03:12.56 | livinded | shmaltz: why is it overdone for a house? |
| 03:13.00 | shmaltz | anybody seen this? its realy funny: |
| 03:13.02 | andymul | Some AGI scripting and some PHP reports, need someone to finish an app that is about 90% done |
| 03:13.02 | shmaltz | http://video.google.com/videoplay?docid=-3412452712894373669 |
| 03:18.51 | Faithful | shmaltz: No I am in AU |
| 03:19.20 | shmaltz | Faithful, that link I gave should have it all |
| 03:19.30 | shmaltz | http://www.voip-info.org/wiki/view/Asterisk+cmd+Sms |
| 03:19.50 | Faithful | Oh thanks shmaltz |
| 03:24.58 | shmaltz | not bad: |
| 03:25.00 | shmaltz | http://video.google.com/videoplay?docid=2402668272873798812 |
| 03:25.42 | *** part/#asterisk foo (n=foo@unaffiliated/foo) |
| 03:26.36 | shmaltz | http://video.google.com/videoplay?docid=7780916420567729697 |
| 03:29.07 | shmaltz | very funny: |
| 03:29.09 | shmaltz | http://video.google.com/videoplay?docid=2393895885958259815 |
| 03:30.24 | *** join/#asterisk n838901_2 (n=n8@24-117-16-198.cpe.cableone.net) |
| 03:30.32 | n838901_2 | hello |
| 03:31.01 | shmaltz | n838901_2, helo |
| 03:31.30 | n838901_2 | can asterisk be used as a voip gateway for sip software clients? |
| 03:31.40 | shmaltz | n838901_2, yes |
| 03:31.45 | n838901_2 | hmm |
| 03:31.46 | shmaltz | ~docs |
| 03:31.51 | jbot | rumour has it, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 03:31.57 | n838901_2 | lol |
| 03:31.59 | n838901_2 | true |
| 03:32.28 | n838901_2 | heres my situation...two physical networks connected by vpn |
| 03:32.41 | n838901_2 | dont want to use skype..would rather use something within the lan |
| 03:33.03 | n838901_2 | any suggestions? |
| 03:33.16 | n838901_2 | software only |
| 03:33.25 | shmaltz | n838901_2, yes use asterisk |
| 03:33.28 | shmaltz | ~asterisk |
| 03:33.30 | jbot | asterisk is the best free PBX in the world. |
| 03:33.58 | shmaltz | ~wiki-asterisk |
| 03:34.05 | n838901_2 | do you recommend a good *free* sip client? |
| 03:34.20 | shmaltz | xlite |
| 03:34.26 | n838901_2 | ..or any protocol for that matter |
| 03:34.33 | shmaltz | xlite, will do |
| 03:34.38 | n838901_2 | thanx a bunch |
| 03:34.46 | shmaltz | and sip is an excellent protocol |
| 03:35.12 | n838901_2 | i will trudge through the wiki and get it rolling |
| 03:46.09 | Ciber311 | lol |
| 03:46.36 | Ciber311 | i'm compiling asterisk on a 500 mhz G4 |
| 03:46.39 | Ciber311 | this is gonna be a while |
| 03:48.28 | *** join/#asterisk rikstah (n=rick@c-24-17-81-231.hsd1.or.comcast.net) |
| 03:49.00 | rikstah | Hey all, I'm looking for the name of a VOIP provider that can provide me with a 900 number, do you know of any? |
| 03:50.06 | watchy2 | ill do it if you hug me |
| 03:50.12 | watchy2 | we can sell porn |
| 03:53.40 | rikstah | it's got nothing to do with porn :) |
| 03:55.07 | *** join/#asterisk SwK (n=Silik0nJ@204.250.115.179) |
| 03:56.36 | watchy2 | echo ratio = 0.0100 (111.0 / 11145.0) |
| 03:56.40 | watchy2 | is that good for fxotune? |
| 04:02.48 | *** join/#asterisk pengyong (n=lala@222.188.131.126) |
| 04:03.21 | watchy2 | any fxotune users her? |
| 04:03.44 | *** part/#asterisk javar (n=javar@200.118.174.253) |
| 04:10.13 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 04:12.59 | *** join/#asterisk BugKham (i=BugKham@202.8.86.164) |
| 04:13.09 | BugKham | hi, there |
| 04:13.26 | *** join/#asterisk daysmen3 (n=primus@host86-139-114-24.range86-139.btcentralplus.com) |
| 04:17.33 | *** part/#asterisk BugKham (i=BugKham@202.8.86.164) |
| 04:18.49 | *** join/#asterisk tempest1 (n=asf@adsl-153-43-12.chs.bellsouth.net) |
| 04:19.30 | *** join/#asterisk mpls-eric (n=ejo1@204.250.115.155) |
| 04:20.53 | mpls-eric | Anyone at Cluecon? |
| 04:21.09 | russellb | nope |
| 04:21.45 | mpls-eric | Not much of a lobby here to meet up in... |
| 04:22.07 | docelmo | I will be tomorrow |
| 04:22.50 | mpls-eric | Ah, what time you arriving? I haven't looked over the schedule even, but I think things are underway in the AM |
| 04:23.05 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
| 04:23.25 | docelmo | yep.. I will be there right at 9am |
| 04:23.56 | *** join/#asterisk Lyfe (n=lyfe@69.8.146.78) |
| 04:23.59 | CrashHD | Hello |
| 04:24.43 | Lyfe | anyone know if the functionality from ODBCget has been integrated into asterisk somehow, specifically if it's still done this way? ( http://www.voip-info.org/wiki/view/Asterisk+app_dbodbc is the webpage i'm reading this from) |
| 04:25.08 | russellb | there is func_odbc |
| 04:25.13 | russellb | an ODBC dialplan function |
| 04:25.29 | russellb | in the trunk, but there is a 1.2 backport available ... www.asterisk-backports.org i think |
| 04:26.08 | Lyfe | so, somewhere, i should be able to find a doc on func_odbc? |
| 04:27.15 | Ciber311 | russellb: compiled just fine, thanks! :) |
| 04:28.35 | russellb | Ciber311: awesome, you're welcome |
| 04:28.55 | russellb | Lyfe: perhaps ...... i don't know :) |
| 04:29.14 | Ciber311 | umm |
| 04:29.18 | fuser | ooh, race wars on /. !!! |
| 04:29.19 | Ciber311 | where did it put the configs? lol |
| 04:29.37 | russellb | Ciber311: nowhere, unless you did "make samples" |
| 04:29.42 | Ciber311 | i did |
| 04:29.44 | Lyfe | russellb: heh.. fair 'nuff.. i seem to have stumbled upon a slightly more advanced func_odbc example to do some trickery with voicemail, i think i can manage from there. |
| 04:30.05 | Lyfe | russellb: thanks though.. this should keep me from getting too crazy. |
| 04:30.11 | russellb | Lyfe: alright. it's fairly new, so docs are probably slim. Also see the "show function ODBC" output |
| 04:30.21 | russellb | Ciber311: ah, then /etc/asterisk |
| 04:30.40 | Lyfe | *CLI> show function ODBC |
| 04:30.40 | Lyfe | No function by that name registered. |
| 04:30.42 | Ciber311 | russellb: what the heck are these? Aug 1 00:30:26 NOTICE[12738]: res_musiconhold.c:511 monmp3thread: Request to schedule in the past?!?! |
| 04:30.47 | Ciber311 | spamming the console |
| 04:31.01 | Lyfe | maybe part of asterisk-addons? |
| 04:31.14 | russellb | Ciber311: yeah, that's because you don't have zaptel ... which isn't available for mac |
| 04:31.19 | Ciber311 | k |
| 04:31.29 | Ciber311 | sigh |
| 04:31.30 | russellb | Lyfe: no. if you want it for 1.2, it's in svncommunity.digium.com |
| 04:31.37 | Ciber311 | now to setup voxbone and axvoice |
| 04:31.43 | Lyfe | so it's not in the 1.2.10 release then? |
| 04:31.47 | Ciber311 | this should be fun |
| 04:31.49 | russellb | Lyfe: svn co http://svncommunity.digium.com/svn/func_odbc/1.2 func_odbc-1.2 |
| 04:31.57 | russellb | Lyfe: no |
| 04:31.59 | Lyfe | ahh, i follow. |
| 04:32.15 | russellb | the 1.2 version is a backport, it's a new feature for the upcoming 1.4 release |
| 04:32.24 | Lyfe | interesting. |
| 04:33.15 | *** join/#asterisk SwK (n=Silik0nJ@204.250.115.130) |
| 04:33.16 | Lyfe | is 1.4 looking very promising? |
| 04:33.28 | russellb | i think so, but I'm biased :) |
| 04:33.42 | Lyfe | of course. |
| 04:36.14 | russellb | but lots of cool stuff has been done ... new features and architecture improvements |
| 04:36.24 | russellb | we'll have a nice marketing-style list at some point |
| 04:36.41 | *** join/#asterisk oej (n=oej@65.197.203.67) |
| 04:37.00 | Juggie | hmmmmmm, rus, i'm having a really weird bug w/ ex-girlfriend logic. |
| 04:37.37 | Juggie | if i do did/1234567890,1,Noop then ,2,Dial(...) it works |
| 04:38.30 | Juggie | but if i do did/1234567890,1,Set(CALLERID(num)=1234567890) then ,2,Dial(...) it does the first exten but then skips out and doesnt do the 2nd priority. |
| 04:39.07 | russellb | huh? |
| 04:39.22 | russellb | why would you set the cid num to be the same as what it already is ... |
| 04:39.42 | Juggie | that was just an example |
| 04:39.48 | Juggie | its being set to something different |
| 04:39.49 | file | I'm confuzzled |
| 04:39.51 | russellb | ok |
| 04:40.02 | russellb | now, does priority 2 have a cid num match, too? |
| 04:40.15 | Juggie | yes |
| 04:40.20 | russellb | which is? |
| 04:40.29 | russellb | the original, or what you just set it to be |
| 04:41.03 | Juggie | exten => _3354/6131234567,1,Noop() |
| 04:41.03 | Juggie | exten => _3354/6131234567,n,Dial(Zap/g3/6131111111) |
| 04:41.04 | Juggie | that works |
| 04:41.14 | russellb | ok |
| 04:41.21 | Juggie | but if i change the Noop -> Set(CALLERID(all)="Me <111>") |
| 04:41.27 | russellb | Well yeah |
| 04:41.30 | Juggie | it skips out doesnt do the next prority |
| 04:41.34 | russellb | because the cid isn't that number anymore! |
| 04:41.41 | russellb | why would it continue to match? |
| 04:41.43 | Juggie | ahhhhhhh |
| 04:41.48 | file | :D |
| 04:41.49 | russellb | :) |
| 04:41.54 | file | russellb: you so smart |
| 04:42.01 | Juggie | jesus thats a good one |
| 04:42.02 | Juggie | hehe |
| 04:42.31 | Juggie | btw, i looked @ the app_voicemail code in turnk |
| 04:42.34 | Juggie | *trunk |
| 04:42.41 | Juggie | theres still really nothing thre for language abstraction that i could see. |
| 04:42.42 | russellb | Juggie: so, just remove the cid match on the Dial ... |
| 04:42.55 | russellb | it's the say.c code that has been abstracted |
| 04:42.59 | russellb | and, it's currently optional |
| 04:43.02 | russellb | the old stuff is still there |
| 04:43.09 | russellb | look at configs/say.conf.sample |
| 04:43.09 | Juggie | ok thats probally why i didnt see it |
| 04:43.12 | Juggie | i just skimed it. |
| 04:43.23 | Juggie | but in the future all talk through voicemail will be pushed through say? |
| 04:43.36 | Juggie | so i can just say (psudo code) play(welcome,dk) |
| 04:43.38 | russellb | most of it probably, i don't know |
| 04:43.43 | russellb | it's pretty far down the priority list |
| 04:43.53 | file | hardy har har... priority list |
| 04:44.14 | russellb | and in my extension, I'm pretty sure there is a Hangup() before I get to that project |
| 04:44.41 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 04:44.41 | *** join/#asterisk mpls-eric (n=ejo1@204.250.115.155) |
| 04:45.13 | Juggie | idealy, it would be nice to abstract the language out of the code |
| 04:45.24 | Juggie | totally, maybe someday |
| 04:45.36 | russellb | yup |
| 04:45.50 | file | not tomorrow... |
| 04:45.54 | file | not the day after... |
| 04:45.59 | Juggie | so all that voicemail does is say play(welcome) or play(you have x new messages) |
| 04:46.11 | Juggie | and some other nice part takes care of playing that in the proper language |
| 04:46.20 | russellb | well, app_voicemail should be written in AEL2, IMO |
| 04:46.32 | Juggie | i dunno about that :) |
| 04:46.38 | Juggie | i thought about writing it in php one day |
| 04:46.41 | Juggie | when it really pissed me off. |
| 04:46.51 | Juggie | but then i realized i dont use it |
| 04:46.55 | Juggie | so i said screw it |
| 04:47.16 | russellb | but you know ... app_voicemail is one of the highest level pieces of code we have |
| 04:47.21 | russellb | it's really a user interface |
| 04:47.37 | CrashHD | isn't dlynes or someone working on rewriting app_voicemail? |
| 04:47.46 | Juggie | its just really un-maintanable due to the language stuff. |
| 04:47.53 | Juggie | if you make a change in one of the menus or whatever |
| 04:47.56 | Juggie | you have to make it in like 10 places |
| 04:47.57 | russellb | I mean, lots of the stuff we do is network handling, hardware interfacing, audio processing, etc. that's all great in C |
| 04:48.01 | russellb | user interfaces in C, not so much |
| 04:48.02 | Juggie | its retarded. |
| 04:48.24 | CrashHD | could always opt for a xml type interface |
| 04:48.35 | russellb | CrashHD: yep, i have considered that. |
| 04:48.43 | russellb | However, a difficult part of that is in the prompts |
| 04:48.47 | Juggie | it would be only about 20% of the size it is now if it were language free. |
| 04:48.58 | CrashHD | let C do what it does best and offload the other stuff |
| 04:49.04 | russellb | the prompts would have to be seriously cut up ... and then still sound smooth when pieced together in whatever order your config needs |
| 04:49.15 | CrashHD | ya true |
| 04:49.45 | russellb | it's not really an easy thing to do ... and it's hard to justify spending a ton of time on something that works just fine |
| 04:50.04 | CrashHD | priorities |
| 04:50.10 | russellb | yup. |
| 04:50.19 | CrashHD | it's understandable |
| 04:50.23 | russellb | not that I would discourage anyone interested in working on it |
| 04:50.32 | Juggie | if i had the time i would |
| 04:50.35 | CrashHD | heh |
| 04:50.40 | Juggie | but i dont, i work two jobs as it is |
| 04:50.40 | CrashHD | story of my life |
| 04:50.40 | russellb | but I also think there are other things that could use attention |
| 04:50.42 | Juggie | i really wish i did. |
| 04:50.54 | CrashHD | wish in one hand... |
| 04:51.01 | *** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net) |
| 04:51.40 | CrashHD | there is deffinite refinement needs |
| 04:51.48 | CrashHD | I think dlynes is already working on app_voicemail |
| 04:51.53 | CrashHD | suggest the xml though |
| 04:51.55 | CrashHD | *t |
| 04:52.10 | Juggie | rus, does the say code in voicemail support just numbers? |
| 04:52.27 | Juggie | i should look at the example config |
| 04:52.34 | *** join/#asterisk SwK (n=Silik0nJ@204.250.115.141) |
| 04:53.50 | Juggie | ahh i see |
| 04:53.55 | Juggie | its only setup for numbers now really and dates. |
| 04:54.15 | CrashHD | rus could you tell me off hand where the dtmf digits are buffered for feature codes, such as blindtransfer? |
| 04:54.17 | *** join/#asterisk angom_h (n=Angel@red-corp-200.76.226.132.telnor.net) |
| 04:54.37 | russellb | CrashHD: that's does in res_features.c |
| 04:54.57 | CrashHD | that's where I've been looking. I'll just look harder |
| 04:54.58 | russellb | done* |
| 04:55.03 | russellb | it's not trivial code ... |
| 04:55.12 | Juggie | russellb, i keep waiting patientally for dialogic support :) i think i'm in week 23 |
| 04:55.13 | CrashHD | trivial as in simple? |
| 04:55.23 | russellb | Juggie: heh |
| 04:55.42 | Juggie | you have to want my business, i get 85k calls a day :) |
| 04:55.43 | Juggie | hurry up |
| 04:55.49 | Juggie | :P |
| 04:56.45 | russellb | don't look at me |
| 04:56.52 | file | we just write stuff |
| 04:56.52 | russellb | CrashHD: right, it is not simple |
| 04:56.57 | file | we don't think about who uses it! |
| 04:56.57 | Juggie | i know :) |
| 04:57.00 | russellb | CrashHD: specifically, look at ast_bridge_call() |
| 04:57.20 | CrashHD | *nods* saw that code |
| 04:57.33 | russellb | CrashHD: a ways down you'll see something like ...featurecode[strlen(featurecode)] = f->subclass; |
| 04:57.39 | russellb | that is literally the line where it gets queued |
| 04:57.40 | CrashHD | ahhh |
| 04:57.54 | CrashHD | needs a patch to dump the queue after feature timeout |
| 04:58.43 | russellb | it's sad that I could point out that line so quickly ... :/ |
| 04:58.43 | CrashHD | heh |
| 04:58.43 | CrashHD | not sad |
| 04:58.51 | CrashHD | *searching for a better word* |
| 04:59.05 | russellb | funny, i don't know |
| 04:59.15 | CrashHD | admirable |
| 04:59.19 | file | russellb: homework young man! |
| 04:59.22 | russellb | file: yes sir |
| 04:59.25 | CrashHD | lol |
| 05:08.11 | linlin | what might be causing a "registration refused" error when trying to register my aix2 softphone |
| 05:08.39 | linlin | iax2* |
| 05:10.23 | russellb | incorrect username, incorrect password, or both |
| 05:10.45 | jbroome | god, i hope there isn't an aix phone. :) |
| 05:11.16 | JT | quick question for all the north americans out there |
| 05:11.32 | livinded | jbroome: why? whats wrong with iax? |
| 05:11.34 | russellb | they're all in bed ... or should be, at least |
| 05:11.41 | JT | is there an easy way to determine if a +1 number is a mobile number or landline? |
| 05:11.43 | russellb | livinded: "aix", not iax |
| 05:11.48 | livinded | oh |
| 05:12.16 | file | JT: nope |
| 05:12.21 | russellb | not that i know of ... |
| 05:12.23 | JT | really? |
| 05:12.24 | file | russellb: schooooooool |
| 05:12.27 | CrashHD | I see the code for feature timer reset |
| 05:12.29 | russellb | file: I AM |
| 05:12.34 | CrashHD | but it is not working as expected |
| 05:12.35 | JT | so it's just random numbers, no groups of prefixes? |
| 05:12.42 | russellb | yup |
| 05:12.43 | livinded | JT: use a reverse lookup |
| 05:12.53 | livinded | or if you have access ANI II |
| 05:13.04 | file | reverse lookup won't always be accurate |
| 05:13.10 | JT | isn't that annoying, not knowing if a number you are calling will be charged at mobile or landline rates? |
| 05:13.12 | livinded | file: but its better than nothing |
| 05:13.28 | file | JT: we don't have mobile/landline rates |
| 05:13.33 | livinded | its all the same here |
| 05:13.40 | JT | is everything timecharged? |
| 05:13.49 | livinded | time charged or unlimited |
| 05:14.01 | JT | does the call receiver have to pay too? |
| 05:14.11 | file | depends on their plan |
| 05:14.16 | JT | heh |
| 05:14.29 | JT | i can see why voip is taking off so well there :P |
| 05:14.32 | file | technically they'd pay per month even if they had unlimited, so yes |
| 05:14.36 | *** join/#asterisk JohnJacob (n=JohnJaco@pool-71-246-132-221.aubnin.fios.verizon.net) |
| 05:14.43 | livinded | on cell phones its all minutes so both parties pay unless there is free m2m, ot the in stuff , or familyplan |
| 05:16.35 | CrashHD | what is the purpose of ast_strlen_zero()? |
| 05:17.22 | Qwell | to check if the len is zero? |
| 05:17.56 | CrashHD | doesn't return true or false? but rather end of string? |
| 05:19.15 | CrashHD | I don't understand what this: return (!s || (*s == '\0')); |
| 05:19.17 | CrashHD | is doing |
| 05:19.42 | CrashHD | can someone explain? |
| 05:20.08 | Qwell | checks that it isn't null or that the first char isn't \0 |
| 05:20.20 | brookshire | damn.. beat me too it |
| 05:20.24 | livinded | me too |
| 05:20.39 | brookshire | to also.. ugh |
| 05:20.49 | CrashHD | what would it return if it was null? |
| 05:20.49 | brookshire | my typing has become sooo horrible lately |
| 05:20.57 | *** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net) |
| 05:20.58 | Qwell | brookshire: You and me both |
| 05:21.03 | livinded | CrashHD: whatever the program tells it to |
| 05:21.12 | brookshire | crash: strippers! |
| 05:21.17 | CrashHD | lol |
| 05:21.19 | russellb | CrashHD: 1 |
| 05:21.31 | CrashHD | so |
| 05:21.42 | livinded | oh :D i didn't see the return |
| 05:21.45 | russellb | CrashHD: that will return non-zero if it is NULL, or if the string is empty (first char is the NULL character) |
| 05:22.07 | brookshire | c is so annoyingly fun |
| 05:22.12 | russellb | C is hot |
| 05:22.30 | brookshire | speaking of hot.. i just say your girlfriend's myspace page |
| 05:22.31 | brookshire | heh |
| 05:22.34 | brookshire | swa |
| 05:22.38 | *** part/#asterisk Wi_Fi (n=OUT@c-24-127-12-85.hsd1.ca.comcast.net) |
| 05:22.40 | brookshire | SAW damnit! |
| 05:22.43 | CrashHD | so hasfeatures = |
| 05:22.43 | russellb | :D |
| 05:22.50 | CrashHD | is setting a bool. value? |
| 05:22.56 | file | brookshire: you CAN spell and type... right? |
| 05:22.59 | CrashHD | based on cha_featurecode being non null or not? |
| 05:23.03 | russellb | brookshire: she r0x0rz |
| 05:23.12 | brookshire | file: i'm losing it |
| 05:23.24 | *** join/#asterisk oadaeh (n=jason@wsip-24-234-160-51.lv.lv.cox.net) |
| 05:23.27 | brookshire | lately i have been just leaving out whole words |
| 05:23.32 | file | brookshire: I can sell you it... for a price |
| 05:23.44 | brookshire | sell me what? |
| 05:23.48 | file | it |
| 05:24.22 | brookshire | only if you throw in the 'sh' |
| 05:25.14 | file | nope! and now I sleep |
| 05:25.22 | brookshire | lame! |
| 05:25.26 | brookshire | i just woke up |
| 05:25.50 | brookshire | i need to sleep... |
| 05:26.25 | CrashHD | so |
| 05:26.33 | CrashHD | bug report time |
| 05:26.36 | russellb | nooooooooooooooo |
| 05:26.38 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 05:26.44 | russellb | there are no bugs. |
| 05:26.49 | brookshire | only features |
| 05:26.56 | CrashHD | "special" features |
| 05:26.57 | russellb | brookshire: word |
| 05:27.26 | CrashHD | russellb: for whatever reason the dtmf flush for feature timeout is not working |
| 05:27.44 | russellb | brookshire: sure :D |
| 05:27.45 | CrashHD | I didn't think the code to flush would be there |
| 05:27.47 | CrashHD | but it is |
| 05:27.51 | russellb | CrashHD: 1.2 or trunk |
| 05:27.57 | CrashHD | 1.2.10 |
| 05:28.00 | brookshire | i just need to firmware file first, lol |
| 05:28.01 | CrashHD | well |
| 05:28.06 | brookshire | s/to/the |
| 05:28.08 | CrashHD | whatever I see in the doxygen |
| 05:28.13 | CrashHD | haven't confirmed in my source |
| 05:28.13 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:28.23 | russellb | CrashHD: that's probably trunk, then. |
| 05:28.27 | russellb | 1.2 doxygen is available, though |
| 05:28.32 | CrashHD | oh |
| 05:28.34 | russellb | http://www.asterisk.org/doxygen/1.2 |
| 05:28.40 | CrashHD | I'll go through that |
| 05:28.49 | CrashHD | thanks |
| 05:29.43 | CrashHD | code is there too |
| 05:33.39 | Juggie | it sure is |
| 05:33.55 | CrashHD | I meant the code for the feature digit timeout |
| 05:34.00 | *** join/#asterisk clyrrad (n=ddd@CPE001195f553c7-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 05:34.37 | russellb | so, what about it isn't working |
| 05:34.54 | CrashHD | When I use ## for blind transfer |
| 05:35.26 | CrashHD | a single # is not sent |
| 05:35.28 | CrashHD | if I sent |
| 05:35.36 | CrashHD | until the call is hungup |
| 05:35.56 | CrashHD | so it's as if the system is buffering waiting for the next digit |
| 05:36.04 | CrashHD | but the first digit doesn't get flushed after timeout |
| 05:36.05 | russellb | and never clears it on timeout |
| 05:36.06 | russellb | i gotcha |
| 05:36.11 | russellb | lemme test it ... |
| 05:36.41 | CrashHD | it drove me nuts for the longest time |
| 05:36.47 | CrashHD | thought it was my carrier |
| 05:36.58 | CrashHD | then thought it was the combination of how I had things routed |
| 05:37.15 | CrashHD | then I saw the dtmf wasn't even leaving my first switch, until channel hungup or additional digits were pressed |
| 05:37.54 | russellb | i don't know my brain can fix this bug at this hour |
| 05:37.56 | russellb | but i'll try |
| 05:38.02 | CrashHD | lol |
| 05:38.14 | CrashHD | wouldn't blame ya |
| 05:38.27 | CrashHD | although I'm on nights at the moment so this is early for me |
| 05:39.10 | brookshire | crash: bribe him with redbull |
| 05:39.12 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 05:39.17 | CrashHD | lol |
| 05:39.18 | russellb | yep, that will do it. |
| 05:39.42 | kmilitzer | Morning everyone ... |
| 05:39.42 | brookshire | haha.. that's worth at least 6 bugs |
| 05:39.50 | russellb | paypal: russelb@clemson.edu ... they accept redbull transfers, now |
| 05:40.00 | Juggie | crash |
| 05:40.08 | Juggie | paste the link to the code in question |
| 05:40.09 | CrashHD | Juggie |
| 05:40.12 | Juggie | in doxygen |
| 05:40.39 | CrashHD | it's 01325-01382 of res_features |
| 05:40.47 | Juggie | common i'm lazy |
| 05:40.49 | Juggie | give me a link |
| 05:40.57 | jhamlyn | :-) |
| 05:40.59 | CrashHD | http://www.asterisk.org/doxygen/1.2/res__features_8c.html#a72 |
| 05:41.21 | russellb | Juggie: if you're that lazy, i doubt you'll be helpful :-p |
| 05:41.26 | CrashHD | hah |
| 05:41.32 | CrashHD | I see the code |
| 05:41.34 | CrashHD | for the flush |
| 05:41.37 | CrashHD | logic makes sense |
| 05:41.40 | Juggie | and whats the problem? |
| 05:41.40 | CrashHD | not sure why it isn't working |
| 05:42.14 | CrashHD | Juggie: read up |
| 05:42.18 | CrashHD | I'm lazy |
| 05:42.22 | CrashHD | *rolls his eyes* |
| 05:42.23 | CrashHD | :) |
| 05:45.51 | CrashHD | although |
| 05:45.54 | CrashHD | let me run by a thought |
| 05:46.02 | CrashHD | the else |
| 05:46.05 | CrashHD | on 01371 |
| 05:46.20 | CrashHD | if feature_timer was not being set |
| 05:46.21 | linlin | gah, why wont this phone connect |
| 05:46.42 | linlin | what are some decent iax2 sofphones |
| 05:46.44 | CrashHD | on line 01475 |
| 05:46.47 | CrashHD | for whatever reason |
| 05:47.01 | CrashHD | although |
| 05:47.10 | CrashHD | then the feature would never go through |
| 05:47.12 | CrashHD | nevermind |
| 05:47.31 | Juggie | i dont think i have the patience to figure this out right now :) |
| 05:48.10 | russellb | Juggie: told you |
| 05:48.13 | CrashHD | lol |
| 05:48.41 | CrashHD | where are multiple digits concat'd at? |
| 05:50.01 | russellb | CrashHD: the same line i showed you earlier ... |
| 05:50.07 | russellb | but you are correct, this is not working |
| 05:50.19 | CrashHD | :) |
| 05:50.31 | Qwell | #*-bugs :p |
| 05:51.25 | brookshire | you mean #asterisk-bugs-2,000 |
| 05:51.29 | CrashHD | ohhh |
| 05:51.31 | CrashHD | I see |
| 05:51.35 | CrashHD | nifty piece of code |
| 05:51.38 | CrashHD | if featurecode is 0 |
| 05:51.45 | CrashHD | add's the the 0 diminsion |
| 05:51.59 | CrashHD | if it already had a digit adds to the 1 diminsion |
| 05:52.01 | CrashHD | hmm |
| 05:52.05 | CrashHD | there is an e in that word |
| 05:52.06 | CrashHD | but ya |
| 05:52.08 | CrashHD | ok |
| 05:53.18 | russellb | i think i found the bug ... |
| 05:53.43 | Qwell | russellb: Don't hate me. http://pastebin.ca/109595 |
| 05:54.02 | Juggie | russellb, where russ. |
| 05:54.40 | russellb | Qwell: looks like an old checkout, since it's also talking about g723 ... rm -f menuselect.makeopts |
| 05:54.49 | russellb | then if you still have a problem, let me know |
| 05:55.01 | Qwell | nah, that always fixes it |
| 05:55.13 | russellb | something that happens a lot? |
| 05:55.18 | Qwell | often enough |
| 05:55.41 | russellb | are you trying to select pbx_kdeconsole? |
| 05:55.49 | Qwell | nah, did an svn up, then make install |
| 05:56.11 | russellb | well i don't know ... |
| 05:56.21 | russellb | if you can find a way to recreate it, let me know, heh |
| 05:56.24 | jhamlyn | Can anyone tell me how to set a peer with a short register time and a local extension with a longer register timeout |
| 05:56.27 | *** join/#asterisk MikeJ (n=vircuser@204.250.115.180) |
| 05:56.42 | Qwell | will do |
| 05:57.27 | CrashHD | russellb: line: 01339 |
| 05:57.35 | CrashHD | checking the backup feature timer |
| 05:57.44 | CrashHD | wouldn't that do it |
| 05:58.11 | *** join/#asterisk niteowldave (n=dave@203.82.162.41) |
| 05:58.27 | CrashHD | hmm |
| 05:58.28 | CrashHD | nm |
| 06:01.33 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 06:01.37 | angom_h | Hi, anyone sending & receiving SMS messages over a GSM link ? |
| 06:01.46 | russellb | CrashHD: still looking ... |
| 06:02.00 | CrashHD | :) |
| 06:02.04 | CrashHD | it's odd |
| 06:02.06 | CrashHD | it seems it would work |
| 06:02.10 | CrashHD | from the code |
| 06:02.29 | *** join/#asterisk CyberMad (n=cybermad@202.73.117.106) |
| 06:03.02 | russellb | CrashHD: well, ast_bridge_call() correctly calls ast_channel_bridge() with a feature_timeout of 500 ms |
| 06:03.20 | russellb | CrashHD: so, if nothing happens, ast_channel_bridge() should break back out after 500 ms, but it does not |
| 06:03.25 | russellb | so, onward to channel.c |
| 06:03.31 | CrashHD | ahh |
| 06:03.48 | CrashHD | I didn't look up the tree that far |
| 06:03.51 | Qwell | fyi: tool <3 |
| 06:04.00 | CrashHD | I'm not familiar with the structure of * at all |
| 06:04.20 | russellb | yeah, like i told you when we started this conversation, this is not trivial code :) |
| 06:04.41 | Juggie | i coudnt find anything obvious |
| 06:04.50 | Juggie | and i'm also having a hard time wrapping my head around it :) |
| 06:05.37 | brookshire | i wrote my first res and i still don't understand it, lol |
| 06:05.49 | CrashHD | lol, not trivial, understatement of the night |
| 06:05.59 | CrashHD | I gotta grab some dinner |
| 06:06.02 | CrashHD | before I pass out |
| 06:06.08 | CrashHD | I'll be back in a little bit |
| 06:06.18 | russellb | alright |
| 06:06.26 | russellb | i feel close to fixing it |
| 06:06.40 | CrashHD | let me know if you come across anything you need my assistance with, testing or whatever |
| 06:07.11 | russellb | will do |
| 06:07.57 | Juggie | russ, something is blocking that shoudnt be? |
| 06:08.08 | russellb | Juggie: well, sort of. |
| 06:08.26 | russellb | it's the fact that channel.c doesn't give a crap about the feature_timeout at all |
| 06:08.40 | russellb | it's not referenced at all |
| 06:08.47 | russellb | So ... yeah. |
| 06:08.51 | russellb | it doesn't do anything. |
| 06:08.52 | Juggie | its passed in but not referenced? |
| 06:08.56 | russellb | correct |
| 06:09.04 | Juggie | hah. |
| 06:09.05 | Juggie | nice :) |
| 06:09.07 | russellb | it's a part of the bridge config structure that is passed ... |
| 06:09.39 | *** join/#asterisk gursikh (n=guriskh1@adsl-68-93-88-157.dsl.hstntx.swbell.net) |
| 06:09.42 | Juggie | right |
| 06:09.44 | Juggie | i see that |
| 06:13.48 | Juggie | its checking for config->timelimit but not for the feature timeout |
| 06:13.59 | russellb | yup |
| 06:14.02 | Juggie | i guess this will require a lil extra logic |
| 06:14.57 | Juggie | well that was fun i wish i had more time to learn this. |
| 06:21.35 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 06:23.20 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 06:23.31 | blitzrage | yo |
| 06:23.40 | russellb | greetings |
| 06:23.51 | blitzrage | russellb: !!! |
| 06:23.55 | russellb | :D |
| 06:24.04 | russellb | nothing like working on bugs at 2:30 AM |
| 06:24.13 | blitzrage | well that was a long day of accounting/programming... |
| 06:24.18 | blitzrage | russellb: aye -- good times |
| 06:24.25 | blitzrage | no school to go to tomorrow... why not? :) |
| 06:25.03 | blitzrage | I just walked in the door from driving an hour from Sarnia to my parents house... long day, but bought a new toy :) |
| 06:25.04 | russellb | heh, yeah |
| 06:25.10 | russellb | ooh, what'd you get |
| 06:26.28 | *** join/#asterisk Pazzo (n=thomas@dialin-225136.rol.raiffeisen.net) |
| 06:26.41 | blitzrage | broke down and bought a blackberry... |
| 06:26.47 | brookshire | crackberry! |
| 06:27.15 | blitzrage | 8700r with unlimited* data plan... so when I'm on the road I can still ssh into an asterisk server if I really need to fix an emergency |
| 06:27.52 | blitzrage | brookshire: indeed |
| 06:28.07 | brookshire | i hope you do not want to use it for a phone |
| 06:28.17 | blitzrage | still charging... but I haven't had a new phone for like... 5 years now -- the little nokia has held up surprisingly well! |
| 06:28.17 | *** part/#asterisk gursikh (n=guriskh1@adsl-68-93-88-157.dsl.hstntx.swbell.net) |
| 06:28.35 | blitzrage | brookshire: yah, its a phone -- my room mate has one too |
| 06:28.46 | brookshire | hah.. |
| 06:28.52 | brookshire | better get a headset |
| 06:28.57 | blitzrage | brookshire: problem is what I envision for a phone doesn't exist |
| 06:29.04 | blitzrage | why? I don't understand what the issue is |
| 06:29.15 | Juggie | i havnt found a phone that improves on my t616 |
| 06:29.34 | Juggie | eg, menu intuitivenes, sound quality, rf.. etc. |
| 06:29.44 | blitzrage | Juggie: no keyboard |
| 06:29.50 | Juggie | i dont want a keyboard |
| 06:29.56 | Juggie | i just want a slim functional candy bar phone |
| 06:30.19 | blitzrage | yah, I have a nokia which is even smaller than what you have there -- it's actually smaller than a motorola razr |
| 06:30.37 | Juggie | the t616 is pretty damn small |
| 06:30.39 | blitzrage | awesome phone, but decided to get something that allows me to use data and ssh from |
| 06:30.47 | blitzrage | Juggie: I've seen that phone -- mine is smaller |
| 06:30.49 | Juggie | but there are better now that phone is 3-4 years old |
| 06:31.04 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 06:31.14 | Juggie | i can use data, but no ssh. |
| 06:31.22 | brookshire | the razr isn't particularly small.. it is just thin.. so it feels smaller in your pocket |
| 06:31.29 | Juggie | the java implementation on this phone doesnt support sockets |
| 06:31.56 | Juggie | if i could get a tiny phone with like a lil addon keyboard |
| 06:31.57 | Juggie | that would be hot |
| 06:32.01 | Juggie | that i could use if i needed to |
| 06:32.43 | *** join/#asterisk lorinc (n=ang@caracas-3838.adsl.interware.hu) |
| 06:33.31 | brookshire | one day we will look back on the blackberry and laugh! |
| 06:33.31 | blitzrage | brookshire: agreed -- but a larger phone doesn't bother me because I just buy cargo shorts with the big pockets on the side specifically to hold things since regular pockets aren't big enough to carry all the equipment I usually have on me. And I wear shorts all year round (even in the winter). I have 3 pairs of jeans, all of which have the cargo pockets on the sides |
| 06:33.57 | blitzrage | I already laugh at it -- but no other phones really do what I was looking for... the HP iPaq phone was close... |
| 06:34.29 | brookshire | blitz: trust me on this though, i do not know many people without a headset for those things |
| 06:34.44 | blitzrage | brookshire: thats fine... it's bluetooth, so I get a headset... |
| 06:34.47 | *** join/#asterisk bmg505 (n=leon@c1-57-5.rndf.isadsl.co.za) |
| 06:34.52 | blitzrage | not sure what the issue is :) |
| 06:34.59 | blitzrage | heck, it comes with a plugin headset :) |
| 06:35.00 | brookshire | the phone is not high quality |
| 06:35.04 | blitzrage | (earbud) |
| 06:35.21 | Juggie | blitzrage, you getting much thunder and lightning in toronto? |
| 06:35.32 | Juggie | its been going crazy here for 2hrs now |
| 06:36.22 | blitzrage | Juggie: I'm in Glencoe right now, but nope, perfectly clear sky here (London) |
| 06:36.27 | Juggie | cool |
| 06:36.29 | blitzrage | stars look awesome right now |
| 06:36.37 | Juggie | its been pouring off and on here and lightning all night |
| 06:36.37 | blitzrage | stormed here about 2 days ago |
| 06:36.43 | Juggie | i think i'm just outside the storm too |
| 06:36.52 | Juggie | its worse a little further east by the looks of the radar |
| 06:37.08 | Juggie | on the othre side of ottawa |
| 06:37.30 | russellb | CrashHD: I'm getting close to fixing that bug ... but I've got to sleep for now. I'll fix it tomorrow, get it touch with me on IRC then sometime |
| 06:37.39 | blitzrage | this phone http://tinyurl.com/nqoco would be perfect if it had 802.11x wireless built in |
| 06:37.58 | blitzrage | russellb: night! |
| 06:38.07 | russellb | g'night |
| 06:38.24 | Juggie | blitzrage, you could probally add wifi to that |
| 06:38.49 | *** join/#asterisk Dico_ (n=niko@60.51.217.61) |
| 06:39.04 | blitzrage | yah... with an external SD I/O card... but ugh... probably would get broke |
| 06:39.24 | Juggie | did you look for another model of that phone rogers doesnt carry? |
| 06:39.36 | *** join/#asterisk niteowldave (n=dave@203.82.162.41) |
| 06:39.56 | niteowldave | file: are you alive |
| 06:40.08 | blitzrage | Juggie: didn't notice any with the wifi |
| 06:40.43 | Juggie | i found some on google |
| 06:40.47 | Juggie | dif models of the hp |
| 06:41.11 | blitzrage | I see |
| 06:41.16 | blitzrage | anyways, I'm off to read then sleep, night |
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| 07:16.59 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:18.05 | *** join/#asterisk dlynes_ (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 07:18.27 | *** join/#asterisk adorah (n=Administ@84.94.122.203.cable.012.net.il) |
| 07:23.38 | *** join/#asterisk dlynes_ (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 07:24.34 | *** join/#asterisk [pyro] (i=pyro@tor/regular/bracketed-pyro) |
| 07:24.38 | *** join/#asterisk qdk (n=qdk@213.237.44.34) |
| 07:27.17 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 07:32.32 | *** join/#asterisk r0d3nt|m (i=anonymou@tinfoilhat.net) |
| 07:34.29 | *** join/#asterisk vlt (n=dm@p54B3432B.dip0.t-ipconnect.de) |
| 07:35.04 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
| 07:38.51 | vlt | Hello. Is it possible to send SMS messages with an asterisk server (connected via SIP to my voip provider)? What client do I need (asterisk is running on ubuntu/linux)? Does the provider have to provide SMS sending? Thanks. |
| 07:40.04 | *** join/#asterisk dlynes__ (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 07:40.35 | *** join/#asterisk proxyfrog1 (n=bluewave@adsl-75-4-195-66.dsl.irvnca.sbcglobal.net) |
| 07:40.55 | proxyfrog1 | hello |
| 07:42.06 | proxyfrog1 | anyone? |
| 07:42.18 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 07:42.28 | proxyfrog1 | hi |
| 07:42.32 | proxyfrog1 | uwe |
| 07:42.40 | uwe | hi |
| 07:42.43 | uwe | proxyfrog1, |
| 07:42.59 | proxyfrog1 | how r u |
| 07:43.21 | uwe | um ... fine |
| 07:43.36 | proxyfrog1 | hows ur asterisk box running? |
| 07:43.44 | uwe | fine |
| 07:44.21 | proxyfrog1 | i'm new to this chatroom......how about u |
| 07:44.48 | uwe | i pass by every now and then |
| 07:45.31 | proxyfrog1 | kool....... |
| 07:45.47 | proxyfrog1 | can i ask u a question about asterisk email? |
| 07:46.11 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 07:46.20 | uwe | um, you can ask, in general, but i dont promise to be able to answer |
| 07:46.28 | proxyfrog1 | kool..... |
| 07:46.28 | Snake-Eyes | is softhangup() ment to hang up a call right away ? |
| 07:46.51 | proxyfrog1 | i got webmin installed on my box........u know wat webmin is? |
| 07:47.04 | uwe | yes, but i never used it |
| 07:47.05 | proxyfrog1 | its just a webtool to interface with linux....... |
| 07:47.09 | proxyfrog1 | kool........ |
| 07:47.27 | Snake-Eyes | proxyfrog1, its a evil spider that taps into everything :P |
| 07:47.29 | proxyfrog1 | if i use webmin to compose and send email out, the email goes out |
| 07:47.38 | proxyfrog1 | haha........yea |
| 07:48.04 | proxyfrog1 | when i got a voicemail, the outgoing email gets user authentication error |
| 07:48.42 | proxyfrog1 | its odd..........i cant send out my voicemails as emails from asterisk, yet i can send a manually created email out of linux....... |
| 07:49.01 | Snake-Eyes | is this asterisk voicemail module trying to send a email out i assume |
| 07:49.02 | proxyfrog1 | what places should i be checking? |
| 07:49.34 | proxyfrog1 | yes, asterisk does send out the email.......but gets rejected by my isp's mail server....... |
| 07:49.53 | proxyfrog1 | my isp's mail server gives a user authentication error in the bounced email mesg...... |
| 07:50.03 | Snake-Eyes | i would start by looking in your voicemail.conf file |
| 07:50.38 | proxyfrog1 | i believe my voicemail.conf has only a few lines........ |
| 07:50.50 | *** join/#asterisk Gunnar (n=gunnar@62.97.242.6) |
| 07:51.07 | uwe | well, i dont know really, but i suppose that a user authentication error is a user authentication error, although im not even sure what you are talking about , but you should check what username/password/protocol you are using to sent these emails |
| 07:51.24 | uwe | s/you/asterisk |
| 07:51.50 | uwe | um, well this should apply only for the last "you" |
| 07:52.13 | proxyfrog1 | my voicemail.conf has a [general] section which just says to include vm_general.inc and vm_email.inc |
| 07:52.31 | Snake-Eyes | proxyfrog1, also look at your program that sends the emails out, we use nullmailer really simple |
| 07:53.07 | proxyfrog1 | bro, i would rather use a simpler mta than sendmail........sendmail is too overkill |
| 07:53.25 | Snake-Eyes | ? |
| 07:53.30 | *** join/#asterisk tengulre11 (n=tengulre@222.90.66.156) |
| 07:53.53 | proxyfrog1 | i got trixbox, which uses sendmail to send out emails |
| 07:54.29 | Snake-Eyes | ah, well I dont use trixbox thus no sendmail :) |
| 07:54.40 | proxyfrog1 | i'd be happy to use anything thats simpler than sendmail..........i've heard of postfix.......havent heard of nullmailer |
| 07:54.57 | tengulre11 | hi,all ! when I using IAX2 to connect other asterisk , I got .. http://rafb.net/paste/results/fsXJqC94.html |
| 07:55.16 | tengulre11 | anybody know why?? |
| 07:55.28 | Snake-Eyes | nullmailer just passes the email off to mta, its not a proper mta itself, but seeing how your ISP is providing a mta any way .... |
| 07:56.12 | Snake-Eyes | eg nullmailer sends to postfix |
| 07:56.22 | proxyfrog1 | how did u connect nullmailer to asterisk? |
| 07:57.27 | proxyfrog1 | hey is it normal that we got over 50 people in this room and only 2 people talking? |
| 07:57.41 | Snake-Eyes | at this time of day yes |
| 07:57.51 | proxyfrog1 | more like over 80 people....... |
| 07:57.58 | proxyfrog1 | i c......kool.... |
| 07:58.34 | proxyfrog1 | dude.........this email thing is killing me..........i can send email but not thru asterisk........ |
| 07:58.55 | Snake-Eyes | astrisk just users what ever mail program you have installed, so you shouldnt have to change asterisk for nullmailer (not absolutly sure on this) |
| 07:59.38 | proxyfrog1 | so wat if i installed nullmailer.........asterisk somehow just knows to use it? |
| 07:59.51 | proxyfrog1 | do i need to shut off sendmail? |
| 08:00.49 | Snake-Eyes | depends on your OS, but yes you wil have to stop sendmail |
| 08:01.12 | uwe | hmm, do you need an MTA to send emails? |
| 08:01.47 | tengulre11 | http://rafb.net/paste/results/fsXJqC94.html |
| 08:01.51 | proxyfrog1 | uwe.....i'm using sbcglobal to send off my email |
| 08:02.28 | proxyfrog1 | i just need sbcglobal's smtp server to accept my email.........right now its rejecting it cuz of user authentication error.... |
| 08:02.28 | Snake-Eyes | just stop sendmail install something like nullmailer and put your isp mail details into nullmailer and see what happens :) |
| 08:02.31 | tengulre11 | when I connect other asterisk with IAX2, I can not registry! |
| 08:03.30 | proxyfrog1 | sounds like a good alternative........ |
| 08:03.55 | proxyfrog1 | hey i'll be right back...... |
| 08:04.06 | Snake-Eyes | proxyfrog1, worst case you uninstall nullmailer and restart sendmail :) |
| 08:05.11 | Snake-Eyes | tengulre11, are using zap stuff to connect asterisk ? |
| 08:05.32 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 08:05.58 | tengulre11 | Sanke-Eyes: nope! it is IAX2 protocol. |
| 08:09.15 | tengulre11 | I have two asterisk servers in different two citys, server A can registered to server B, but server B can registered server A, they are static ip address. |
| 08:10.58 | tengulre11 | when I using 'netstat -na| grep 4569' the port is listened. |
| 08:11.04 | Snake-Eyes | tengulre11, err do you mean 'but server B can't registered servered server A' ? |
| 08:11.37 | tengulre11 | sorry! server B can not registered server A. |
| 08:12.33 | tengulre11 | how to config the iax.conf if it cross firewall? |
| 08:14.34 | Snake-Eyes | you shouldnt have to do anything only NAT can cause problems (besides opening a port up in the firewall) |
| 08:14.42 | *** join/#asterisk ccherrett (n=chris@s142-59-14-94.ab.hsia.telus.net) |
| 08:14.48 | tengulre11 | in server a: iax.conf, [general]... register => serverb:serverbpwd@xxx.xxx.xxx.xxx [servera] type=friend username=servera secret=servera host=dynamic |
| 08:15.19 | ccherrett | is there a way to send a prerecorded message over a simple modem to a phone or do I need a telephony system? |
| 08:15.32 | Snake-Eyes | tengulre11, I would just compare configs between the two |
| 08:15.57 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 08:16.05 | waba | I have a SNOM 300 that works fine for SUBSCRIBE. Same config for a 360, but this one ends up with Subbscription-status: terminated;cause=timeout. Any idea why? |
| 08:16.26 | tengulre11 | in server a: iax.conf, [general]... register => servera:servera@xxx.xxx.xxx.xxx [serverb] type=friend username=serverb secret=serverbpwd host=dynamic |
| 08:16.36 | waba | (and so the LEDs don't get updated after registration) |
| 08:16.42 | tengulre11 | sorry this is server b |
| 08:16.59 | ccherrett | I am trying to determine if I need to install asterisk or not |
| 08:17.38 | Snake-Eyes | ccherrett, far as i know a normal modem cant send voice |
| 08:18.29 | tengulre11 | pls!!! |
| 08:18.31 | Snake-Eyes | tengulre11, are the account details correct for server b |
| 08:18.33 | ccherrett | Snake-Eyes: too bad |
| 08:18.52 | tengulre11 | Snake-Eyes: yes! |
| 08:19.09 | Snake-Eyes | tengulre11, I dont work with iax2 much, mostly sip ;( |
| 08:19.35 | tengulre11 | Sanke-Eyes: but the sip can not cross the firwall? |
| 08:19.50 | tengulre11 | s/firwall/firewall |
| 08:19.59 | ccherrett | Snake-Eyes: I need to allert someone if an event occurs and I thought that a phone call would work best |
| 08:20.20 | ccherrett | Snake-Eyes: Is it complicated to set up in Asterisk? |
| 08:20.30 | tengulre11 | most of people suggest me to using IAX2. but I failed! |
| 08:20.38 | *** join/#asterisk potsboy (n=chrisg@196.211.16.202) |
| 08:21.09 | Snake-Eyes | tengulre11, you getting things confused, anything can cross firewall, the main difference is NAT http://www.voip-info.org/wiki-IAX+versus+SIP |
| 08:21.47 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 08:21.48 | Snake-Eyes | ccherrett, hehe, depends if you have used asterisk before, and how you plan to do it |
| 08:21.51 | potsboy | hey all, doesnt any know if astericon dallas will be doing dcap certification? |
| 08:21.59 | tengulre11 | Snake-Eyes: thank you very much!! I m reading now. |
| 08:22.25 | ccherrett | Snake-Eyes: thanks I will continue to investigate |
| 08:22.44 | *** join/#asterisk kernelbee (n=Naveed@202.63.226.41) |
| 08:23.21 | Snake-Eyes | ccherrett, also depends on if you use hardware or voip provider to termintate pstn calls, and if you use a frontend/gui to asterisk |
| 08:23.27 | tengulre11 | Snake-Eyes: what's NAT function? |
| 08:24.14 | Snake-Eyes | tengulre11, Network Address Translation - it takes private ip's and translates them to public ip |
| 08:24.22 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 08:24.34 | Snake-Eyes | tengulre11, in your case you shouldnt need to worry about NAT |
| 08:24.55 | tengulre11 | Snake-Eyes: thank u! :) |
| 08:25.30 | Snake-Eyes | tengulre11, if you using public ip address on both machines all you need to do is open up the ports on your firewall for iax2 |
| 08:25.59 | tengulre11 | Snake-Eyes: all ports is opened! |
| 08:26.05 | kernelbee | yes, you need to open up the ports |
| 08:26.24 | tengulre11 | the firewall only is a simple router |
| 08:26.41 | kernelbee | its better to set ranage ports in configuration and open only those udp ports... |
| 08:29.29 | kernelbee | Snake-Eyes: in NAT packets directly goes to sender of traffic, its not caring about the packet define sender ip address :) |
| 08:30.16 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 08:31.03 | kernelbee | Snake-Eyes: for the security its necessary to open only those ports which you use for rtp and for IAX/SIP protocol. |
| 08:32.51 | Snake-Eyes | kernelbee, huh? |
| 08:33.38 | Snake-Eyes | kernelbee, you want me tell you how SIP and NAT works as well :P |
| 08:40.13 | kernelbee | Snake-Eyes: I think, knowledge sharing is also a way to incerase knowdlege.. I hope you will be part of those who helps me to increase my knowledge..:) |
| 08:41.08 | *** join/#asterisk Astinus- (n=aa@85.19.143.16) |
| 08:42.04 | Snake-Eyes | hehe |
| 08:43.31 | Snake-Eyes | now to ask my question again: is softhangup() ment to hang up a call right away ? |
| 08:44.13 | *** join/#asterisk MikeJ (n=vircuser@204.250.115.180) |
| 08:47.10 | kernelbee | Snake-Eyes: yes |
| 08:49.07 | *** part/#asterisk proxyfrog1 (n=bluewave@adsl-75-4-195-66.dsl.irvnca.sbcglobal.net) |
| 08:49.28 | kernelbee | <PROTECTED> |
| 09:02.22 | *** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 09:03.06 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 09:04.54 | jhamlyn | Can some one help me set the peer register timeout so it is different to my user register timeout - Please :-) |
| 09:05.42 | jhamlyn | is it possible....? |
| 09:06.21 | jhamlyn | I have a sip provider who requires the timeout to be above 600 and local nat user who have router timout of less than 180 seconds |
| 09:11.50 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
| 09:12.23 | *** join/#asterisk eDIsonxl (i=eDIsonxl@59-124-181-42.HINET-IP.hinet.net) |
| 09:13.26 | *** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net) |
| 09:25.39 | *** join/#asterisk Bert- (n=bert@bas33-1-82-66-4-198.fbx.proxad.net) |
| 09:25.44 | Bert- | hello there |
| 09:26.28 | Bert- | I've a little question please : when a users call my IVR, and dials some numbers, in which variables these digits are stored please ? |
| 09:26.49 | *** join/#asterisk ghenry (n=ghenry@82-69-192-46.dsl.in-addr.zen.co.uk) |
| 09:33.14 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 09:34.10 | docelmo | Aparently you havent done your home work. You can check the basics on www.voip-info.org |
| 09:34.11 | uwe | $arg2 |
| 09:34.25 | docelmo | but to answer your question.. |
| 09:34.28 | docelmo | ${EXTEN} |
| 09:34.43 | *** join/#asterisk linlin (i=linlin@c-67-184-230-25.hsd1.il.comcast.net) |
| 09:34.43 | uwe | :) |
| 09:34.44 | docelmo | arg2 is only valid if you do it in a macro |
| 09:35.14 | uwe | i just did grep number * , and got ; arg1 = trunk number, arg2 = number, arg3 = route password |
| 09:36.50 | uwe | i c |
| 09:37.04 | waba | could my SUSCRIBE-tion getting timed out (as exposed above, @now-20mins) be related to a "sched_settime: request to schedule in the past" message? |
| 09:37.25 | *** join/#asterisk fourcheeze (n=rich@office.callmaster.co.uk) |
| 09:38.21 | waba | also it only seems to happen when the phone dials the SUSCRIBEd for extension, if I change the state from another phone it stays SUSCRIBEd |
| 09:40.10 | docelmo | your not asking the right questions cause that makes no sense |
| 09:43.07 | *** join/#asterisk linlin (i=linlin@c-67-184-230-25.hsd1.il.comcast.net) |
| 09:43.16 | waba | me? how does that make no sense? |
| 09:52.45 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 09:55.29 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 09:55.38 | *** join/#asterisk key2 (n=key2@sd-420.dedibox.fr) |
| 09:58.26 | *** join/#asterisk MatsK (i=MatsK@83.233.97.229) |
| 10:07.29 | *** join/#asterisk niteowldave (n=dave@CPE-203-51-61-57.nsw.bigpond.net.au) |
| 10:10.28 | *** join/#asterisk kiddy (n=achu@124.125.39.182) |
| 10:11.34 | kiddy | anybody used festival ? |
| 10:12.16 | kiddy | when I tried festival it is not working means : it is connecting and disconnecting at the same time |
| 10:12.36 | kiddy | no sounds hearing |
| 10:12.46 | *** join/#asterisk Blafasel (n=bpodszun@relay3.vistream.de) |
| 10:12.54 | kiddy | any help ? |
| 10:13.34 | Blafasel | G'Day.. I just have to try it again: Is anyone able to lend me a helping hand with a SIP <-> SS7 bridge where connections can listen to MOH, but otherwise are muted on both sides? |
| 10:20.13 | *** join/#asterisk Zombie (n=masterz@adsl-78-71.lex.bluegrass.net) |
| 10:20.19 | Zombie | Hello. |
| 10:20.39 | kiddy | when I tried festival it is not working means : it is connecting and disconnecting at the same time |
| 10:20.47 | kiddy | any help to slove this ? |
| 10:26.03 | *** join/#asterisk Mr-packet (n=a@222-154-239-122.adsl.xtra.co.nz) |
| 10:28.43 | *** join/#asterisk [pyro] (i=pyro@tor/regular/bracketed-pyro) |
| 10:29.13 | *** join/#asterisk Chai_Sangeen (n=Chai_San@access.bahrainedb.com) |
| 10:29.19 | Chai_Sangeen | hello everybody |
| 10:30.16 | Zombie | Hello. |
| 10:30.36 | Zombie | I'm technically looking for non-Pompous PHP Coders. |
| 10:30.44 | creativx | they are non-existant |
| 10:30.57 | *** join/#asterisk s0lid (n=jlq@210.213.241.226) |
| 10:30.58 | Zombie | I find that hard to believe. |
| 10:32.32 | *** part/#asterisk Chai_Sangeen (n=Chai_San@access.bahrainedb.com) |
| 10:34.08 | *** join/#asterisk gaspiz (n=gaspiz@86.35.34.63) |
| 10:35.05 | gaspiz | hi, I am running asterisk 1.2.1 with realtime voicemail, changing the pswd from VoicemailMain doesn't change the value |
| 10:35.19 | gaspiz | does anyone know something about this? |
| 10:50.48 | *** join/#asterisk zaffa (n=x@62.236.135.12) |
| 10:53.29 | *** part/#asterisk Samoied (n=Samoied@201.21.216.149) |
| 10:54.16 | zaffa | Are there a way to add background music to ZAP<->SIP call (or playback a file to both ends)? Like chanspy feature but playback (not recording). |
| 10:56.00 | *** join/#asterisk Hoondie (n=h@59.167.25.7) |
| 10:57.23 | *** join/#asterisk [Airwolf] (n=airwolf@dsl5402DE03.pool.t-online.hu) |
| 10:57.33 | *** join/#asterisk champster (n=asterisk@AH.tescogroup.com) |
| 10:57.37 | key2 | kiddy: what u trying to do |
| 10:57.45 | *** part/#asterisk Zombie (n=masterz@adsl-78-71.lex.bluegrass.net) |
| 10:59.20 | Hoondie | anyone know what the problem might be with this.. basically when i call out (from my voip phonv via my ISP using SIP) to my cell, it sounds like it's ringing, but my cell phone never rings.. then i get this on the console "-- Nobody picked up in 30000 ms -- Got SIP response 408 "Timeout after cancelling request" back from 203.2.134.1" |
| 11:03.21 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 11:06.02 | kernelbee | Hoondie: your isp supporting sip to Mobile network translation ? |
| 11:07.16 | kernelbee | Hoondie: or you have ISDN card? |
| 11:08.32 | key2 | kernelbee: he said his ISP |
| 11:09.49 | *** join/#asterisk _4d4m_ (n=adam@62.69.102.99) |
| 11:10.17 | key2 | hoondie: internode.on.net |
| 11:10.21 | key2 | that's ur isp right / |
| 11:10.22 | key2 | ? |
| 11:11.18 | *** join/#asterisk shadebob (n=chatzill@adsl-159-135-192-81.adsl2.iam.net.ma) |
| 11:11.37 | *** join/#asterisk daysmen3 (n=primus@host86-139-114-24.range86-139.btcentralplus.com) |
| 11:12.13 | shadebob | Hi, I have a little problem with a server with 2 tdm and 1 te110p. In zap show status, what is the signification of IRQ columns? |
| 11:13.08 | Hoondie | key2: yep.. |
| 11:14.29 | Hoondie | kernelbee: it works if i use xlite to dial.. just asterisk's thinks it's dialing, but it's actually not |
| 11:15.52 | key2 | Hoondie: lol :) |
| 11:16.36 | key2 | Hoondie: inc the ptime |
| 11:16.37 | Hoondie | it used to work fine.. now it's not for some reason |
| 11:17.04 | key2 | defaultexpirey=1800 |
| 11:17.04 | key2 | dtmfmode=auto |
| 11:17.04 | key2 | qualify=yes |
| 11:17.32 | Hoondie | should i add this in the [general] part of sip.conf? |
| 11:18.38 | Sonderblade | how can you stop asterisk's infinite loop detection? it screws up for my recursive calls |
| 11:18.43 | key2 | y |
| 11:18.44 | kernelbee | Hoondie, which version of asterisk you are using |
| 11:18.55 | Hoondie | 1.2.9.1 |
| 11:22.10 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 11:22.32 | kiddy | can anybody help me to configure festival for asterisk ? |
| 11:22.50 | kernelbee | Hoondi, which codec you are using to connect with your isp? |
| 11:23.01 | kernelbee | my mean for rtp |
| 11:23.22 | *** join/#asterisk Arnar_ (n=arnarb@landi.oddi.is) |
| 11:23.23 | Hoondie | is this what you mean: |
| 11:23.24 | Hoondie | allow=ulaw |
| 11:23.25 | Hoondie | allow=ilbc |
| 11:23.32 | kernelbee | yes, |
| 11:24.00 | Hoondie | thats all i got in there |
| 11:24.11 | kernelbee | Hoondie, do you know which codec your isp support? |
| 11:25.20 | Hoondie | G.711, G.729a, GSM |
| 11:27.19 | kernelbee | Hoondie: ok, remove ilbc, and write " disallow=all \r\n allow=alaw \r\n allow=ulaw" |
| 11:27.36 | kernelbee | Hoondie: \r\n -> new line |
| 11:27.43 | Hoondie | yep |
| 11:28.13 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 11:29.43 | Hoondie | nope.. still the same |
| 11:30.36 | Hoondie | even if rtp is not working, should the phone ring at all? |
| 11:30.45 | *** part/#asterisk Arnar_ (n=arnarb@landi.oddi.is) |
| 11:32.31 | kernelbee | Hoondie: do you have firewall on your system? |
| 11:34.30 | Hoondie | on the linux system? nope, there's nothing in iptables.. i am behind a NAT router though, port forwarded 5060 and 10000-20000 to my asterisks box |
| 11:36.33 | Hoondie | when i type sip show registry in the console, i get this: sip.internode.on.net:5060 029043xxxx 1785 Registered so it looks like it's registered.. |
| 11:38.00 | kernelbee | Hoondie: in configuration do you have nat=yes ? |
| 11:38.08 | Hoondie | yep |
| 11:41.33 | Blafasel | Apart from port 5060, what do I need for SIP? |
| 11:41.54 | shadebob | someone use a sagem dcn shdsl here? |
| 11:41.56 | Hoondie | ports 10000-20000 UDP |
| 11:42.36 | kernelbee | Hoondie, monitor system activity through tcpdump, is traffic bidirectional or unidirectional |
| 11:42.50 | kernelbee | Hoondie: b/w your system and isp |
| 11:44.34 | kernelbee | Hoondie: also on sip debugging in asterisk |
| 11:46.04 | shadebob | anyone use on a same server tdm ad te card? I m lost with span definition.... |
| 11:46.30 | jhamlyn | Is anyone working on the b410p bri card.. |
| 11:46.51 | jhamlyn | I have installed drivers and have card running on kernel .26 |
| 11:47.29 | jhamlyn | looking for approach to feed from one channel straight back to the other... |
| 11:47.36 | Hoondie | kernelbee: got it working, not sure what the problem.. copied config from someone else on the same ISP from a forum.. |
| 11:51.51 | *** join/#asterisk LakeSolon (n=blake@12-227-169-99.client.mchsi.com) |
| 11:54.32 | Hoondie | found it, needed this in the config: fromdomain=sip.internode.on.net |
| 11:54.45 | Blafasel | I'm having a SIP v. Network issue here, I guess. Layout: * server <-VPN-> company network. IAX calls from the company network succeed, SIP calls stay silent. Any ideas which prerequisite I might have missed here? I'm looking at tcpdump, but don't see lost/blocked packets so far.. |
| 11:55.01 | *** join/#asterisk antony_ (n=chazapis@bobble.cslab.ece.ntua.gr) |
| 11:55.51 | antony_ | can someone plz help? When using attended transfer the caller is able to hit # and transfer the callee! |
| 11:57.29 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:03.39 | antony_ | hi all. can someone plz help with an extensions.conf prob? |
| 12:04.12 | Hoondie | whats the problem |
| 12:06.05 | antony_ | when I transfer calls around using blind transfer, only the one receiving the call can transfer again (I use "t"). When I use attended transfer, it seems that everyone can transfer - which is problem when the caller is from my ISDN line |
| 12:06.52 | Hoondie | what does this particular line look like in extensions.conf? |
| 12:07.49 | antony_ | i.e. exten => 403,1,Dial(SIP/403,,t) |
| 12:08.01 | antony_ | is an internal SIP line (through a PAP2) |
| 12:08.47 | Hoondie | and who is able to transfer the call? |
| 12:09.24 | antony_ | ok. let's say someone calls from my ISDN line. He gets dropped into the "incoming" context: exten => s,1,Dial(SIP/401&SIP/402&SIP/403&SIP/404,,t) |
| 12:10.02 | *** join/#asterisk bkidney (n=bkidney@nat01-128.stratos.ca) |
| 12:10.04 | antony_ | I pickup the phone, hit # and can send him around any other SIP line and so on. Only the one receiving the call can transfer again |
| 12:10.15 | antony_ | But with attended transfer this does not work |
| 12:10.57 | Hoondie | i'm not sure.. |
| 12:10.59 | shadebob | I have 2 tdm cards and 1 te110p on my *box. Someone can help me with span definition in zaptel.conf |
| 12:11.15 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 12:11.16 | bkidney | Has anyone seen this problem? When I dial from my SIP phone to an external line, I still hear ringing in the SIP earpiece after the other extension has picked up (we can talk and he does not here the ringing)? |
| 12:11.16 | antony_ | say I receive the call from 402 and hit *2 and transfer to 403. Now the caller from the outside line can hit # and transfer me! |
| 12:12.03 | antony_ | bkidney: are you using "r" in the Dial statement? |
| 12:12.24 | bkidney | Antony: No. Is that my problem? |
| 12:12.59 | antony_ | bkidney: I guess it could be if you were using it :) |
| 12:13.34 | antony_ | bkidney: I had the same problem with an ISDN line. I was using "b" and stopped using it |
| 12:13.50 | bkidney | antony_: Oh, I thought the problem might be I wasn't using it. |
| 12:14.16 | antony_ | bkidney: You can try it anyway... |
| 12:16.21 | bkidney | antony_: The exact dail statement for the call is: exten => _9XXX,1,Dail(Zap/4/${EXTEN:1}) |
| 12:16.45 | kiddy | when I connected to festival server it disconnects immediately any help for solving this ? |
| 12:24.10 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:26.50 | *** join/#asterisk ree (n=ree@3e44abf5.adsl.enternet.hu) |
| 12:27.34 | ree | hi I have a setup problem that I can't figure out |
| 12:28.03 | ree | I "register" an account on an external SIP server to have an incoming connection |
| 12:28.32 | ree | when the call comes in, it resolves to the correct local extension, but then I get "Authentication failed" |
| 12:28.57 | ree | what does it try to authenticate against, at this point? And how could I disable that? Any ideas? |
| 12:29.53 | ree | helping url is appreciated too... |
| 12:37.17 | *** join/#asterisk bigf1sh (n=mat@admin.iseek.com.au) |
| 12:39.59 | *** part/#asterisk bkidney (n=bkidney@nat01-128.stratos.ca) |
| 12:47.49 | [TK]D-Fender | antony_ : What kind of phones do you have? |
| 12:48.59 | *** join/#asterisk s0lid (n=jlq@gr-153-4.eglobalreach.net) |
| 12:50.17 | antony_ | SIP phones connected to PAP2 ATAs |
| 12:50.46 | *** join/#asterisk dacleric (n=dacleric@p5482300E.dip0.t-ipconnect.de) |
| 12:50.51 | *** join/#asterisk Dandre (n=Dandre@was59-3-82-236-48-30.fbx.proxad.net) |
| 12:50.58 | Dandre | Hello all, |
| 12:52.25 | [TK]D-Fender | antony_ : Then stop using *'s DTMF transfers and start using the ATA's native capabilities. |
| 12:52.38 | Dandre | I have an asterisk box with one diva isdn card. I use chan_capi and I can dialout but I can't receive calls on thi interface. I don't see anything in asterisk log |
| 12:52.49 | [TK]D-Fender | antony_ : that way you won't find yourself giveing callers abilities they shouldn't. |
| 12:53.15 | antony_ | any pointers on how to do this? |
| 12:53.16 | *** join/#asterisk [Airwolf] (n=airwolf@dsl5402DE03.pool.t-online.hu) |
| 12:54.53 | [TK]D-Fender | antony_ : Your PAP2 manual. it tells you how to use it to transfer calls, do 3-way, etc... |
| 12:59.22 | *** join/#asterisk eric-xx (i=Eric@cm83.epsilon192.maxonline.com.sg) |
| 12:59.51 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 13:01.08 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 13:01.10 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.88) |
| 13:01.25 | Hoondie | is there anyway i can answer a call to another extension? |
| 13:01.54 | [TK]D-Fender | Hoondie : Look for "call pickup" on the wiki. |
| 13:02.22 | *** join/#asterisk ESCulapio__ (n=ESCulapi@200.88.44.66) |
| 13:02.42 | nortex | If I use the ringall strategy on a queue will it only ring members who are not on the phone? or will it ring all members? |
| 13:02.55 | antony_ | or you can put all relevant extensions into the same pickupgroup and define in fetaures.conf a sequence to pickup the phone |
| 13:05.17 | *** join/#asterisk Mercestes (n=Merceste@216.54.143.2) |
| 13:05.17 | *** join/#asterisk GyrosGeier (n=richter@p54995093.dip.t-dialin.net) |
| 13:05.47 | *** part/#asterisk GyrosGeier (n=richter@p54995093.dip.t-dialin.net) |
| 13:06.09 | *** join/#asterisk Egonis (n=chultay@207.245.14.10) |
| 13:06.37 | [TK]D-Fender | nortex : Depends on what kind of members, what kind of calls, etc. |
| 13:07.12 | Egonis | I have FXS channels setup with the context 'out', however as soon as I pick up the line/channel, I get Channel Zap/12 is sent into invalid extension 's' in context 'default' but no invalid handler |
| 13:07.14 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:08.14 | Blafasel | What's the right syntax in a Dial() to reach someone via IAX on another server? IAX/<What comes here>? |
| 13:08.39 | nortex | [TK]D-Fender, Can you explain what you mean by the types of calls and members? I have 6 static members 3 with a penalty of 1 and 3 with a penalty of 2. But the calls only ring on the first group even if all 3 are on the phone. |
| 13:08.52 | [TK]D-Fender | Egonis : Get rid of "immediate=yes" |
| 13:09.02 | Egonis | [TK]D-Fender: Ah! Thank you |
| 13:09.10 | [TK]D-Fender | nortex : pastebin the config. |
| 13:12.39 | *** join/#asterisk beyond (n=beyond@200.192.160.100) |
| 13:12.50 | shadebob | between te110p and ISDN CPE I have to use cross cable or streight cable? |
| 13:14.09 | nortex | [TK]D-Fender, www.pastebin.ca/110125 |
| 13:15.30 | shadebob | because I have a blue alarm and I don't known if it's a wiring issue |
| 13:17.53 | [TK]D-Fender | nortex : Should ring all members not on a queue call. |
| 13:18.10 | nortex | shadebob, In my own limited experince the wrong cable would result in a red alarm. But I have never seen or heard of a blue alarm. Have you tried a different cable? |
| 13:19.00 | shadebob | nortex : I have a cable with 1-4 2-5 pineout |
| 13:19.42 | nortex | [TK]D-Fender, No matter the penalty? |
| 13:20.09 | *** join/#asterisk TheCompWiz (n=TheCompW@wsip-68-109-200-102.mc.at.cox.net) |
| 13:20.34 | TheCompWiz | can anyone recomend a good (cheap is also desireable) fxs adapter? |
| 13:22.10 | *** join/#asterisk lunk (n=lunk@negative-influence.com) |
| 13:23.17 | nortex | [TK]D-Fender, I guess I will try to build cascading queues since the penalty is not doing what I had hoped. |
| 13:23.24 | [TK]D-Fender | nortex : Could be on equl penalty basis... not sure... |
| 13:23.26 | *** join/#asterisk trnygaar (i=AQrmZf8t@antapex.odalen.com) |
| 13:23.36 | [TK]D-Fender | TheCompWiz : SPA-2002 |
| 13:24.19 | *** join/#asterisk champster (n=asterisk@AH.tescogroup.com) |
| 13:24.32 | trnygaar | I need to use different trunk for each extension, can make 1 context for each extension, that calls the different trunks, but can this be done with a macro. Like extension "2600" should use trunk "trunk2600" |
| 13:25.56 | champster | Are there any problems with always having nat=yes and qualify=yes? |
| 13:26.51 | trnygaar | if nat=yes, it will never bridge connections i think? |
| 13:27.28 | trnygaar | I am still trying to learn, so don't take it as an definitive answer :P |
| 13:27.56 | nortex | trnygaar, you could try something like trunk${callerid(num)} to add the extensions number to trunk in a dial command. |
| 13:30.32 | *** join/#asterisk forensics (n=f@adsl-75-21-9-22.dsl.irvnca.sbcglobal.net) |
| 13:30.51 | trnygaar | was something like that i thought of, just need to figure out the right place to do that, i "cheat" with trixbox :) |
| 13:31.17 | trnygaar | it seems to be set up with a macro that uses numbers |
| 13:32.04 | *** join/#asterisk userdefined (n=jross@cpe-24-169-142-23.rochester.res.rr.com) |
| 13:32.12 | shadebob | crc4 parameter depend of the telco or of the cpe? |
| 13:32.39 | TheCompWiz | anyone? opinions about fxs adapters? ??? |
| 13:33.15 | Blafasel | I'm trying to set a IAX2 <-> IAX2 link up. Can someone correct my terminology here? Those are friends (since the relation is bidrectional), right? Do they have a username and a host entry? Do they exist on both hosts (on one host to configure the link with a password etc., on the other to set an account up to log in to?) |
| 13:34.59 | nortex | champster, I use qualify=yes on almost all sip devices even though they are on the local net |
| 13:35.41 | champster | thanks. |
| 13:35.49 | champster | The NAT issue has me concerned. |
| 13:36.06 | champster | I used to require a VPN and had them all set to nat=never |
| 13:36.18 | nortex | TheCompWiz, I have used an IAXy and a Linksys ATA with good results, but they are not in the cheap end. |
| 13:36.25 | CrashHD | phones should have built in vpn clients |
| 13:36.29 | champster | I no longer require a VPN, and had to set nat=yes to get the phone to work from home. |
| 13:36.30 | hmmhesays | haha that was a blast from the past |
| 13:36.43 | *** join/#asterisk coppice (n=chatzill@127.166.17.210.dyn.pacific.net.hk) |
| 13:37.03 | champster | phones should have built in vpn clients |
| 13:37.26 | *** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com) |
| 13:37.29 | coppice | and send e-mail |
| 13:37.40 | *** join/#asterisk [g2] (n=g2@nslu2-linux/g2) |
| 13:38.04 | *** join/#asterisk c4t3l (n=c4t3l@69.15.174.114) |
| 13:38.40 | nortex | Based on the books explanation of the nat=yes I would say that it would not be a problem. |
| 13:38.41 | *** join/#asterisk jbalcomb (n=jbalcomb@216.28.180.158) |
| 13:38.52 | champster | trnygaar: By never bridge, you mean the calls can't do canreinvite of something else. (I have only dealt with zap bridging) |
| 13:39.56 | nortex | The nat setting should not effect reinvites since it only tells asterisk to look at the IP address and port in the IP header instaed of the SIP header |
| 13:40.25 | noname32 | i got a question i was to hand code some speed dials wich seem to only work uner ext-local but that ts in my ext_addtional file freepbx .. and it over rights my addions how do i make the addition to extentions.conf or make another conf with my speed dials apart from freepbx? |
| 13:40.35 | jbalcomb | Is there a GSM codec for Windows Media Player or another player good for gsm files? |
| 13:40.40 | hmmhesays | champster: my wrt has a vpn client |
| 13:41.08 | hmmhesays | noname32: it is tricky |
| 13:41.21 | champster | QuickTime plays GSMs. I assume that VLC does too. |
| 13:41.23 | hmmhesays | i could tell you how |
| 13:41.27 | nortex | noname32, Look to the extensions_custom.conf to avoid being overwritten. |
| 13:42.21 | hmmhesays | nortex: thats not entirely true |
| 13:43.00 | hmmhesays | one well placed include in extensions_additional.conf makes all the difference in the world |
| 13:43.16 | *** join/#asterisk devel (n=devel@wiggum.digitalcoven.com) |
| 13:43.25 | nortex | hmmhesays, You mean include your own files? |
| 13:43.32 | hmmhesays | on |
| 13:43.33 | hmmhesays | *no |
| 13:44.18 | *** join/#asterisk [Airwolf] (n=airwolf@dsl5402DE03.pool.t-online.hu) |
| 13:44.40 | Blafasel | Any hints for "Call rejected by 172.31.45.41: No authority found" when I try to call via Dial(IAX2/anothernode/1234)? |
| 13:44.42 | nortex | ok, youv'e peaked my interest, even though I have not used it in months, what is this golden nugget you know of. |
| 13:44.49 | hmmhesays | my mistake, one well placed include in extensions.conf makes all the difference in the world |
| 13:45.05 | hmmhesays | i forgot which file held context from-internal |
| 13:45.38 | hmmhesays | [from-internal] include => custom-speed-dial |
| 13:46.04 | hmmhesays | will not get overwritten, and it gives your extensions access to custom-speed-dial if you use the default context when you add an extension |
| 13:46.08 | *** join/#asterisk MikeJ (n=vircuser@204.250.115.180) |
| 13:47.27 | *** join/#asterisk jbsolutios (n=jbenson@217.169.50.74) |
| 13:48.48 | hmmhesays | you smell what i'm cooking nortex? |
| 13:48.51 | jbsolutios | Hi All - I am running 1.2.10 with Snom360 handsets. I have them set up with hints, so that when someone calls you can see, but does anyone know how to set it up so that when press the line button which is flashing, it picks up the call please? |
| 13:48.52 | Blafasel | The other side says "Rejected connect attempt from 192.168.49.5, who was trying to reach 'somenumber@'".. |
| 13:49.14 | hmmhesays | Auth error man |
| 13:49.28 | Dandre | I have an asterisk box with one diva isdn card. I use chan_capi and I can dialout but I can't receive calls on thi interface. I don't see anything in asterisk log |
| 13:49.50 | *** join/#asterisk Shark_y (n=paoloc@adsl-ull-206-38.46-151.net24.it) |
| 13:49.59 | hmmhesays | Blafasel: my guess is you have a username auth mismatch |
| 13:50.01 | Blafasel | Well - any more details? I'm calling IAX2/user:pass@otherhost - where the otherhost has a iax.conf section with a peer [user] and the same pass |
| 13:50.15 | Blafasel | Can I get a more verbose output? |
| 13:50.22 | hmmhesays | iax2 debug |
| 13:50.24 | hmmhesays | send a call |
| 13:50.29 | hmmhesays | paste your iax.conf |
| 13:50.36 | hmmhesays | and the output from the debug in pastebin |
| 13:50.37 | *** join/#asterisk kpettit (n=keith@69.15.174.114) |
| 13:50.46 | hmmhesays | send me a bottle of vodka when i fix your problem, not the cheap shit either |
| 13:51.00 | Blafasel | I'm from germany. We're better at making beer.. |
| 13:51.10 | kpettit | having a queue problem. From asterisk console I keep seeing calls to a Sip/7777 but there is no sip/7777 anywhere |
| 13:51.19 | nortex | Got it, I had not realized the context was protected, that is much easier. |
| 13:51.48 | noname32 | got it thanks nortex |
| 13:51.49 | hmmhesays | Blafasel: no russian friends? |
| 13:51.50 | kpettit | phones can't foward to a sip peer, and I've grep'd /etc/asterisk and there isn't a 7777 anywhere |
| 13:51.53 | Shark_y | I really need help!! with a tdm400p card with 2 fxo I always get: == Everyone is busy/congested at this time (1:0/0/1) |
| 13:51.59 | hmmhesays | lol, noname32 what the crap |
| 13:52.04 | hmmhesays | i get no love |
| 13:52.10 | Blafasel | hmmhesays: I fear no.. Both iax.confs? Or from which host? |
| 13:52.25 | Shark_y | I really appreciate any help |
| 13:52.41 | hmmhesays | origination and termination side iax.conf's and the iax debug from the terminating box |
| 13:53.02 | nortex | Shark_y, Can you pastebin the cli output and zapata.conf |
| 13:53.04 | hmmhesays | that's ok Blafasel you can paypal me 40 bucks and I'll just go to the liqour store |
| 13:53.46 | nortex | hmmhesays, I appreciate the tip if no one else does ;) |
| 13:53.49 | Blafasel | hrhr.. You're either from scandinavia or drink 40y old vodka from barrels ;) |
| 13:53.58 | noname32 | hmmhesays, lol :) i do thank u for responding but i needed to slaped to actualy read the info in ext_custom lol |
| 13:54.06 | [TK]D-Fender | Shark_y : pastebin CLI output of the failed call (ALL OF IT). as well as CLI output of "show channels". |
| 13:54.07 | [TK]D-Fender | ~pb |
| 13:54.08 | jbot | i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 13:54.16 | coppice | Everyone is busy/congested at this time == flu season |
| 13:54.30 | nortex | lol |
| 13:54.43 | hmmhesays | cold/flu season I had a wicked one earlier |
| 13:54.53 | hmmhesays | awww nortex |
| 13:55.08 | hmmhesays | Blafasel: 40y old vodka from barrels? lol |
| 13:55.19 | noname32 | ohh crap hmmhesays i did see the rest of what u wrote haha sorry thanks mate |
| 13:55.26 | Blafasel | For 40 bucks (as in Euro) you get 4 good bottles here ;) |
| 13:55.26 | hmmhesays | haha |
| 13:55.39 | hmmhesays | Blafasel: I have premium vodka in my home state |
| 13:55.55 | hmmhesays | * made in my home state |
| 13:57.05 | nortex | hmmhesays, You a Crimson Tide fan |
| 13:57.08 | Shark_y | D-Fender pastebin is not working at the moment |
| 13:57.19 | nortex | try pastebin.ca |
| 13:57.23 | *** join/#asterisk hohum (n=dcorbe@12.195.58.235) |
| 13:57.38 | [TK]D-Fender | Shark_y : use .ca |
| 13:57.41 | hmmhesays | hmm, not that I know of |
| 13:58.17 | *** join/#asterisk n9urk (n=leonard@user-0ce2dhc.cable.mindspring.com) |
| 13:58.50 | n9urk | hi all. In the mysql cdr is "duration" in seconds? so 102 = 102 seconds and not 1 minute 2 seconds? |
| 13:58.56 | coppice | why would anyone be a fan of crimson tides? they kill all the sea life |
| 13:59.10 | *** join/#asterisk jero (n=jero@savoirfairelinux.net) |
| 14:00.00 | nortex | coppice, different Crimson Tide, I was actual refering to the University of Alabama and their football program. |
| 14:00.13 | n9urk | Fold Tide Fold |
| 14:00.23 | *** join/#asterisk s0lid (n=jlq@gr-153-4.eglobalreach.net) |
| 14:00.43 | coppice | does the football team kill a lot of sea life? |
| 14:01.12 | nortex | Only if it comes on the field :) |
| 14:01.23 | Bert- | * is really the coolest soft I've ever seen/tried/played with ! ;) |
| 14:01.24 | n9urk | I remember the madess in college when we finally beat the bloody tide |
| 14:01.30 | coppice | its been a while since we had a crimson tide around here. |
| 14:01.53 | n9urk | we did it once in something like 20 years |
| 14:01.57 | n9urk | our football program sucks |
| 14:02.04 | nortex | n9urk, Where did you go? |
| 14:02.11 | hmmhesays | i play nfl xtreme 2 on psx |
| 14:02.13 | n9urk | University of Kentucky |
| 14:02.23 | Shark_y | D-Fender & nortex http://pastebin.ca/110240 I'm using loops... instead of ks only as last tests. THX |
| 14:02.50 | coppice | why is a football program named after a toxic red algae? |
| 14:02.57 | Dr-Linux|work | [TK]D-Fender, i have a question maybe you can help, problem is: |
| 14:03.05 | nortex | Kentucky is legendary in Baskettball though :) |
| 14:03.09 | lunk | good morning |
| 14:03.47 | lunk | is it possible to create a .call file that will use generic Dial, so it can use my already configured trunk logic? |
| 14:03.53 | Shark_y | D-Fender & nortex I'm in italy ... don't know if it is important |
| 14:03.55 | n9urk | nortex: we put all our resources into Basket Ball. But from the looks of it since spring 1998 it would appear that we are broke ;) |
| 14:04.10 | Dr-Linux|work | [TK]D-Fender, Cell phone >> Asterisk 1 >>> asterisk 2 >> cell phone = hears very low .. almost can't hear |
| 14:04.27 | n9urk | in the cdr logs is the duration in seconds? so 102 = 102 seconds and not 1 minute 2 seconds? |
| 14:04.48 | Dr-Linux|work | [TK]D-Fender, Asterisk 1 internal softphone/hardphone >> asterisk 2 >> cellphone = just fine |
| 14:05.41 | *** join/#asterisk [Airwolf] (n=airwolf@dsl5402DE03.pool.t-online.hu) |
| 14:06.10 | *** join/#asterisk expat_iain (n=expat_ia@194.204.99.166) |
| 14:06.55 | Shark_y | D-Fender & nortex ztcfg and zttool is reporting that's all right, no alarms |
| 14:06.55 | macTijn | ~pb = http://www.paste-it.net/ |
| 14:07.02 | [TK]D-Fender | Shark_y : on line 46 : ZAP/2-1 is not a valid tech format, and its only picking 1 channel, not a GROUP. Also you are running AMP, please read the channel topic |
| 14:07.02 | macTijn | ~pb is http://www.paste-it.net/ |
| 14:07.04 | jbot | ...but pb is already something else... |
| 14:07.07 | *** join/#asterisk iq (n=iq@unaffiliated/iq) |
| 14:07.11 | macTijn | ~pb is also http://www.paste-it.net/ |
| 14:07.12 | jbot | okay, macTijn |
| 14:07.21 | expat_iain | Anyone seen negative values using zttest? Like this: Best: 100.000000 -- Worst: -51.184082 -- Average: 95.121913 |
| 14:07.21 | [TK]D-Fender | Dr-Linux|work : Translation : cell phones suck |
| 14:07.36 | macTijn | too bad it's down now :( |
| 14:07.53 | Dr-Linux|work | [TK]D-Fender, which cell phone? the one is calling? or far end? |
| 14:08.07 | [TK]D-Fender | Dr-Linux|work : ALL |
| 14:08.07 | Sonderblade | anyone know of a player that can play .gsm files? |
| 14:08.15 | [TK]D-Fender | Sonderblade : SOX, Winamp |
| 14:08.27 | hmmhesays | vlc can probably do it |
| 14:08.30 | nortex | [TK]D-Fender, Is "alaw => 1-2" in Shark_y's Zapata.conf file a valid entry? |
| 14:08.39 | Dr-Linux|work | [TK]D-Fender, cellphone was just an example even on analog phone as well the same problem |
| 14:08.52 | Sonderblade | [TK]D-Fender: sox can't play, it only converts |
| 14:09.02 | Shark_y | D-Fender ZAP/g0 is a valid entry? |
| 14:09.14 | coppice | sox can play |
| 14:10.31 | *** join/#asterisk N0S3 (n=terminal@host184.201-252-200.telecom.net.ar) |
| 14:10.46 | nortex | Shark_y, ZAP/g0 is valid, but ZAP/2-1 is not. |
| 14:10.50 | Shark_y | D-Fender changing to g0 and with thisExecuting Dial("SIP/301-9c39", "ZAP/g0/338000000|120|r") I got the same result |
| 14:10.52 | hmmhesays | in mother russia sox plays you |
| 14:11.26 | coppice | in mother russia you wear a lot of sox |
| 14:13.16 | *** join/#asterisk SwK (n=Silik0nJ@65.169.134.2) |
| 14:13.52 | hmmhesays | god I can't get flash working in fc5 for the life of me |
| 14:14.14 | Shark_y | nortex but I've got the same result with ZAP/g0 |
| 14:14.18 | brookshire | use debian :) |
| 14:14.43 | hmmhesays | whats the use, same gui |
| 14:14.59 | nortex | Shark_y, Just on a whim try commenting out "alaw => 1-2" in Zapata.conf. |
| 14:15.00 | hmmhesays | this isn't a fc5 v. debian problem |
| 14:15.52 | *** join/#asterisk fourcheeze (n=rich@office.callmaster.co.uk) |
| 14:15.57 | Shark_y | nortex ok, I'm trying, I just added that because I want to force alaw |
| 14:17.18 | nortex | Shark_y, I have only seen that setting in voip channels. |
| 14:18.05 | Shark_y | nortex done and I still got :Everyone is busy/congested at this time (1:0/0/1) |
| 14:18.59 | Shark_y | nortex zap show channel 1 tells me: Default law: ulaw |
| 14:19.18 | Shark_y | and surely is not good |
| 14:20.15 | *** join/#asterisk kiddy (n=achu@124.125.39.182) |
| 14:20.29 | kiddy | what this error means : == Spawn extension (from-internal, 1022, 2) exited non-zero on 'SIP/2004-09ee4980' |
| 14:21.33 | kiddy | any idea? |
| 14:22.27 | n9urk | where is the best place to get an ATA? |
| 14:22.53 | hmmhesays | me |
| 14:23.02 | n9urk | got url? |
| 14:23.03 | hmmhesays | whaht |
| 14:23.07 | hmmhesays | kidding |
| 14:23.10 | hmmhesays | what are you looking for/ |
| 14:23.37 | n9urk | I am not exactly sure what the best value is right now. 1 or 2 line ATA |
| 14:23.38 | hmmhesays | i like my mediatrix 2102's a bitch to configure, but so many features |
| 14:25.13 | *** join/#asterisk [Airwolf] (n=airwolf@dsl5402DE03.pool.t-online.hu) |
| 14:27.38 | tzanger | what's the "name" of the spec that allows ethernet interfaces to autonegotiate 10/100 FDX/HDX? |
| 14:28.10 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 14:28.40 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:30.15 | kiddy | Please help me to solve the error : == Spawn extension (from-internal, 555, 2) exited non-zero on 'SIP/2004-098f9ce0' |
| 14:30.44 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 14:31.30 | Shark_y | nortex any suggestion??? Please, I'm desperate! |
| 14:31.32 | nortex | kiddy, That does not look like an error. |
| 14:32.11 | *** join/#asterisk postel_ (n=jp@unaffiliated/postel) |
| 14:32.15 | nortex | Shark_y, I'm looking over the zapata.conf info. I have not done zaptel stuff in an international setting before. |
| 14:32.29 | [TK]D-Fender | hmmhesays : Yeah the 2102's transparent proxy is wicked cool... |
| 14:32.48 | [TK]D-Fender | hmmhesays : And while I wouldn't call it "a bitch" its not an SPA, thats for sure |
| 14:33.05 | kiddy | nortex : but my connection is ending after this log |
| 14:33.07 | hmmhesays | [TK]D-Fender: for the average user it sucks to configure, but I do everything through tftp anyway |
| 14:33.20 | [TK]D-Fender | hmmhesays : Yeah Joe Blow would be lost in space on it... |
| 14:33.21 | hmmhesays | [TK]D-Fender: i really like mediatrix ata's though |
| 14:33.32 | kiddy | nortex : I mean it is going to handup mode after it |
| 14:33.33 | [TK]D-Fender | hmmhesays : Me too, too bad they're so pricey |
| 14:33.45 | [TK]D-Fender | n9urk : depends where you are. |
| 14:33.46 | hmmhesays | pricey? what are you paying? |
| 14:33.53 | *** join/#asterisk murf (n=steve_mu@216.166.159.235) |
| 14:34.06 | nortex | kiddy, that is because Asterisk processed the hangup, if it was not supposed to then there should be something before that entry that says why it is hanging up. |
| 14:34.13 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 14:34.34 | noname32 | any one here use snom 360? |
| 14:35.17 | noname32 | was woundering if there is a way to do attn trans with a button with out the phone hanging up |
| 14:36.15 | kiddy | nortex : http://pastebin.ca/110273 please look at it |
| 14:36.47 | *** join/#asterisk javar (n=javar@200.118.174.253) |
| 14:37.08 | [TK]D-Fender | hmmhesays : Most plasces list it over $100 |
| 14:37.40 | hmmhesays | $105 is our base price for 1 |
| 14:37.48 | [TK]D-Fender | hmmhesays : 140$ at voipsupply |
| 14:38.01 | [TK]D-Fender | hmmhesays : Yeah wholesale is decent, its retail that sucks |
| 14:38.54 | hmmhesays | if you need any drop me a line |
| 14:39.08 | [TK]D-Fender | hmmhesays : I might want 1 just to say "yeah I've got one" |
| 14:39.10 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 14:40.01 | hmmhesays | [TK]D-Fender: yeah we stock them here, i dunno how much shipping would be to you though |
| 14:40.32 | [TK]D-Fender | hmmhesays : the fact it can do dual G729 and transparent proxying really pays in certain applications. |
| 14:40.41 | [TK]D-Fender | hmmhesays : I'd pay shipping, my rates are great |
| 14:41.01 | tzanger | wtf |
| 14:41.03 | [TK]D-Fender | hmmhesays : Will think about... I jsut finished buying my 3rd Polycom for home :) |
| 14:41.15 | tzanger | voipsupply is saying how great a deal $140 for a ip430 is and your price to me is $105? |
| 14:41.20 | l-fy | [TK]D-Fender > you should see the new linksys |
| 14:41.22 | [TK]D-Fender | hmmhesays : So I should cool it on the telephony expense :) |
| 14:41.25 | l-fy | is a beauty |
| 14:41.26 | [TK]D-Fender | l-fy : Which? |
| 14:41.30 | l-fy | spa921 |
| 14:41.37 | l-fy | works like a charm |
| 14:41.53 | [TK]D-Fender | tzanger We're talking about the Mediatrix 2102. |
| 14:42.00 | tzanger | oh |
| 14:42.01 | tzanger | :-) |
| 14:42.14 | [TK]D-Fender | l-fy : I owned an SPA-941. Never again. |
| 14:42.28 | [TK]D-Fender | l-fy : Works = yes, inferior = yes. |
| 14:42.29 | l-fy | [TK]D-Fender > why? |
| 14:42.34 | l-fy | inferior = ???? |
| 14:42.35 | l-fy | why? |
| 14:42.59 | hmmhesays | yeah I have 2102's deployed in many places |
| 14:43.05 | [TK]D-Fender | l-fy : Polycom offers me much better control, audio & manufacturing quality, presence, etc. |
| 14:43.13 | l-fy | kidding right? |
| 14:43.21 | l-fy | it's working perfect |
| 14:43.24 | kiddy | nortex : any idea about it ? |
| 14:43.25 | l-fy | never had a problem |
| 14:43.55 | [TK]D-Fender | l-fy : Not at all... I had them side by side for a long while. Not saying it was a "problem", but that it their price point Linksyst does not factor into my valuation chart in North America. |
| 14:44.17 | [TK]D-Fender | l-fy : Now if you're talking about budgets overseas, THEN they come into play where Polycom is noticably more expensive. |
| 14:44.48 | [TK]D-Fender | l-fy : But an IP 501 outclasses the SPA's. |
| 14:44.56 | tzanger | I really like Lily and Parrots |
| 14:45.06 | tzanger | (sung by Sun Kil Moon) |
| 14:45.11 | [TK]D-Fender | l-fy : No paying extra for more than a 2nd line appearance, not limited to 1 call per line key, etc |
| 14:45.12 | tzanger | kind of indie/gritty sound |
| 14:45.49 | [TK]D-Fender | l-fy : Polycom also makes considerably better use of its also larger LCD. only thing Linksys has going for it is the backlight on the SPA-942. |
| 14:46.04 | [TK]D-Fender | l-fy : and I for one don't particularly care. |
| 14:46.14 | nortex | kiddy, So you are calling 102 when you get the hangup? |
| 14:46.19 | [TK]D-Fender | l-fy : For some it is a make or break point, but you can't make EVERYONE happy. |
| 14:46.50 | kiddy | nortex : yes when I dia 102 it going straing to hangup |
| 14:47.23 | kiddy | nortex : sorry , when I dial 102 it going strait to hangup |
| 14:49.02 | nortex | kiddy, The festival command executed is not part of 102, is exten 102 in the dailplan anywhere else? |
| 14:49.25 | kiddy | no its only in extension.conf |
| 14:49.27 | Sonderblade | when an agent picks up a call from a queue, it gets english voice prompts. but in all other places in asterisk i get localized prompts, is there a special language setting for agens or something? |
| 14:49.58 | hmmhesays | agents.conf maybe? I dunno |
| 14:50.10 | kiddy | nortex : I created it for testing purpose |
| 14:51.43 | kiddy | nortex , also the festival server is showing this error when I am dialing 102 "accepted from server" "disconnected" |
| 14:51.49 | Sonderblade | hmmhesays: no |
| 14:52.06 | hmmhesays | Sonderblade: was just guessing |
| 14:52.08 | [TK]D-Fender | Sonderblade : How does the queue call the agents? |
| 14:52.09 | kiddy | nortex , This is happening immediately |
| 14:52.17 | nortex | kiddy, I would look for the command Festival("mary had a little lamb") in the dial plan since you are going there instead of the posted section. |
| 14:52.29 | Sonderblade | [TK]D-Fender: im adding the extensions as members of the queue |
| 14:53.19 | kiddy | nortex , I have created it also in extensions.conf . It also not working |
| 14:54.23 | *** join/#asterisk iq (n=iq@unaffiliated/iq) |
| 14:54.40 | [TK]D-Fender | Sonderblade : Show me. |
| 14:55.01 | Sonderblade | [TK]D-Fender: in the queues section: member => SIP/403 |
| 14:55.08 | *** join/#asterisk trbldwine (i=troubled@adam.ur.northwestern.edu) |
| 14:57.23 | [TK]D-Fender | Sonderblade : what language is the call in until that point? |
| 14:57.58 | hmmhesays | so anyone have experience with vision servers? |
| 14:58.16 | Sonderblade | [TK]D-Fender: the caller is in the right language, but i have reportholdtime = yes and the person answering the queued call gets english voice prompts |
| 14:58.45 | coppice | vision servers? is this some kind of hallucinations over IP? |
| 14:59.19 | hmmhesays | nope vision computers |
| 14:59.24 | hmmhesays | tiger directs flagship model |
| 15:01.02 | [TK]D-Fender | Sonderblade : I believe the queue has a language setting of its own. what is it set to? |
| 15:01.14 | [TK]D-Fender | hmmhesays : Tigerdirect sucks ass.. avoid |
| 15:01.35 | Sonderblade | [TK]D-Fender: i didn't know that.. and it isn't documented |
| 15:02.05 | *** join/#asterisk trbldwine (i=troubled@adam.ur.northwestern.edu) |
| 15:03.02 | Sonderblade | [TK]D-Fender: and setting language in the queue has no effect |
| 15:03.20 | [TK]D-Fender | Sonderblade : did you check SIP/403's language setting? |
| 15:03.55 | hmmhesays | [TK]D-Fender: i buy stuff from there without problems |
| 15:04.34 | Sonderblade | [TK]D-Fender: yes, [general] in sip.conf contain the correct language setting |
| 15:04.47 | [TK]D-Fender | hmmhesays : The screwed up a pile of my orders, promised delivery failures, etc. And Vision isn't a really known brand. Depends what you want I guess... I hat e to say I'd sooner go Dell. Nobody ever got fired for buying IBM... |
| 15:05.09 | [TK]D-Fender | Sonderblade : I asked about the PHONE. set it in there and tell me how it works out. |
| 15:05.29 | Sonderblade | [TK]D-Fender: tried that too, makes no difference |
| 15:05.40 | [TK]D-Fender | Sonderblade : hrm |
| 15:05.40 | hmmhesays | [TK]D-Fender: vision has a fairly good reputation, they are just smaller |
| 15:05.57 | [TK]D-Fender | hmmhesays : If you say so... I did like the price point personally, but they make me nervous. |
| 15:05.59 | Sonderblade | it must be a bug |
| 15:06.14 | *** join/#asterisk SwK (n=Silik0nJ@65.169.134.2) |
| 15:06.35 | hmmhesays | if you are going to go with dell you might as well go straight to compaq, easier warranty info |
| 15:07.35 | hmmhesays | [TK]D-Fender: dell has so many config options it gets confusing comparing among models |
| 15:07.35 | nortex | hmmhesays, How do you figure, Dell has never fixed my stuff when under warranty. |
| 15:07.55 | hmmhesays | compaq's no questions asked replacement policy is good |
| 15:09.36 | hmmhesays | anyone know a good windows installer builder? |
| 15:10.05 | [TK]D-Fender | hmmhesays : BartPE |
| 15:10.12 | *** join/#asterisk acrg (n=aragon@decoder.geek.sh) |
| 15:10.19 | acrg | hiya |
| 15:10.19 | [TK]D-Fender | hmmhesays : Or are you referring to software installer front-end? |
| 15:10.30 | *** join/#asterisk trbldwine (i=troubled@adam.ur.northwestern.edu) |
| 15:10.41 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 15:11.33 | hmmhesays | [TK]D-Fender: the latter |
| 15:12.23 | acrg | if I have a setup whereby I have a PRI for connection to a telco and all SIP phones for office extensions, how do I prevent an office phone from transferring a call that came in via the PRI to another call going out the PRI? |
| 15:12.29 | [TK]D-Fender | hmmhesays : Oh.. nvm :) |
| 15:12.56 | [TK]D-Fender | acrg : You can't really, without cutting off their ability to call out period |
| 15:13.17 | hmmhesays | i can't get idefisk to run in linux either argh, what a day |
| 15:13.23 | [TK]D-Fender | acrg : Or attempting to cancel their ability to transfer ANY call at all. |
| 15:13.39 | acrg | hrm |
| 15:13.56 | nortex | hmmhesays, I have used one from Nullsoft and one from Caphyon, the free one :) |
| 15:13.57 | acrg | that's quite serious :) |
| 15:15.42 | TheCompWiz | anyone know why the voice mail system wouldn't work? |
| 15:15.43 | [TK]D-Fender | acrg : The caller can do whatever the caller can do..... thats their right |
| 15:15.51 | acrg | do you know if this is regarded as a bug/flaw ? |
| 15:15.53 | hmmhesays | ./idefisk: error while loading shared libraries: /usr/lib/libiaxclient.so: cannot restore segment prot after reloc: Permission denied |
| 15:16.04 | [TK]D-Fender | TheCompWiz : You could try elaborating on your exact problem you know... |
| 15:16.28 | [TK]D-Fender | acrg : Neither. Its what is EXPECTED. |
| 15:16.33 | TheCompWiz | I dial *97 .... and something answers... but there is no voicemail greeting & such... |
| 15:17.13 | nortex | TheCompWiz, Does the CLI tell you anything about missing files? |
| 15:17.17 | [TK]D-Fender | TheCompWiz : Do I know your extensions.conf by heart? Maybe you could show us as well as CLI output of the call thats not working as expected... |
| 15:17.27 | [TK]D-Fender | *97 = who know what... |
| 15:17.28 | TheCompWiz | nortex... not that I've seen yet... still looking. |
| 15:17.57 | file | hmmhesays: selinux is running |
| 15:18.03 | acrg | yea, the problem is the caller's right shouldn't be allowed to pass on to an outsider |
| 15:18.43 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:18.45 | acrg | if a call is transferred as in my example, an outsider is given the ability to keep a toll call open as long as they like :/ |
| 15:19.02 | hmmhesays | file: yeah caught that |
| 15:19.05 | hmmhesays | just now |
| 15:19.15 | hmmhesays | now I set SELINUX=disabled |
| 15:19.19 | *** join/#asterisk smackus (n=ckwall@63.149.122.93) |
| 15:19.41 | hmmhesays | now how do I reload it |
| 15:20.03 | *** join/#asterisk Trakkasure (n=Nfebvib@24-50-26-239.atlsfl.adelphia.net) |
| 15:20.08 | AndyCap | hmmhesays: don't, use getenforce/setenforce instead. |
| 15:20.18 | [TK]D-Fender | hmmhesays : "shutdown -r now"? :) |
| 15:20.25 | *** join/#asterisk kernelbee (n=Naveed@80.77.12.2) |
| 15:21.50 | smackus | hey, all... looking for simple syntax help. I have the Polycom 301 and 501 phones. I notice sometimes that when I dial a number, the phone will automatically send the call. For example if I use all 10 digits of a number. Anything less than that, I have to hit the send button on the phone. How do I make that work for all calls less than 10 digits? here is what I have for dialing. http://pastebin.ca/110318 |
| 15:23.39 | smackus | also, rather than specifying what can be dialed, is it possible to restrict what can be dialed? for example, 900 numbers? |
| 15:24.00 | acrg | [TK]D-Fender In the dialplan I think I could check ${BLINDTRANSFER} to block this cases, but do you know how attended transfers are indicated in the dialplan ? |
| 15:24.26 | eKo1 | smackus: You should change the settings on your Polycom so that it dials immediately after dialing whatever x amount of digits you want. |
| 15:24.41 | smackus | is that done in the sip.conf? |
| 15:24.43 | *** join/#asterisk c4t3l (n=c4t3l@69.15.174.114) |
| 15:24.44 | Trakkasure | smackus: It's in the phone's dialplan in the sip.cfg file (if provisioning) or directly on the phone's web console |
| 15:25.02 | eKo1 | smackus: to answer your second question, yes |
| 15:25.17 | Trakkasure | eko1: I wouldn't do that.. because then 10 digit numbers would have a problem... |
| 15:25.26 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 15:25.37 | Trakkasure | typically, you'd prefix with a 9 or 8 for long distance... or external numbers that aren't extensions. |
| 15:25.52 | eKo1 | e.g. to restrict calling 900 numbers, you would have the following in your dialplan: exten => _900X.,1,hangup |
| 15:27.06 | smackus | eKo1: can I specify per sip.cfg a dial plan rather than in the extensions.conf? |
| 15:27.47 | syzygyBSD | so am I the only one that doesn't trust SELINUX cuz it came from the NSA? |
| 15:28.24 | eKo1 | smackus: sip.cfg? |
| 15:28.40 | eKo1 | syzygyBSD: I don't use selinux and currently have it disabled. |
| 15:28.50 | syzygyBSD | ya, I do on all my boxes |
| 15:29.18 | syzygyBSD | on "disable" hmm... it was still compiled into your kernel though |
| 15:29.19 | smackus | eKo1: yeah, is it possible to set the dial plan there instead? rather than doing it in my extensions.conf? |
| 15:29.29 | syzygyBSD | smackus: no |
| 15:29.36 | smackus | damn |
| 15:29.40 | smackus | ok... thanks |
| 15:29.49 | syzygyBSD | you can pick what context each extension goes into |
| 15:30.09 | syzygyBSD | what are you trying to do |
| 15:30.34 | hmmhesays | ahh it works now |
| 15:30.45 | hmmhesays | is there any better iax softphone than idefisk? |
| 15:31.07 | [TK]D-Fender | acrg : not indicated IIRC |
| 15:31.19 | eKo1 | hmmhesays: i use iaxcomm |
| 15:31.33 | eKo1 | it serves my testing purposes.... |
| 15:31.59 | hmmhesays | i'm looking end user wise |
| 15:32.38 | [TK]D-Fender | syzygyBSD : They use it to protect themselves. Also its open source so all eyes are on it. |
| 15:32.40 | eKo1 | You mean dumb-customer wise |
| 15:33.25 | syzygyBSD | oh, i read their page on it, and I know i can see the source if I want. I still dont' trust it |
| 15:34.13 | syzygyBSD | maybe it is some of their policies on other privacy issues that has me worried... |
| 15:38.39 | syzygyBSD | I just don't trust anyone with a bunch of secrets to keep mi........NO CARRIER |
| 15:40.29 | hmmhesays | eKo1: exactly |
| 15:40.29 | *** join/#asterisk xnon (i=xnon@200.8.4.227) |
| 15:40.38 | xnon | hello friends |
| 15:40.41 | hmmhesays | haha |
| 15:40.46 | Shark_y | [TK]D-Fender solved or getting close, the problems is the SPA941 phone because the X-lite works, any suggestion on ho configure that phone? |
| 15:40.55 | xnon | i have any probs with my phone central |
| 15:41.14 | xnon | i have a handytone-286 |
| 15:41.39 | xnon | but i cant enter in a mailbox with him |
| 15:41.46 | xnon | anybody can helpme |
| 15:41.48 | xnon | ? |
| 15:41.59 | *** join/#asterisk tRSS (n=tRSS@193.220.221.2) |
| 15:42.01 | xnon | voice box sorry |
| 15:42.05 | Shark_y | [TK]D-Fender maybe the 941 does'nt accept the delay that the tdm400 gives |
| 15:42.24 | [TK]D-Fender | Shark_y : what dealy are you talking about? |
| 15:42.30 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
| 15:42.44 | eKo1 | xnon: phone central voice box? |
| 15:42.50 | xnon | yeap |
| 15:43.00 | eKo1 | no clue |
| 15:43.01 | xnon | i push 8500 numer |
| 15:43.07 | xnon | number |
| 15:43.17 | tRSS | quick question: I have setup an extension in sip.conf and i want the user to be able to have his softphones registered from two different places at the same time. how can I do that? |
| 15:43.25 | xnon | and the operator say that enter mailbox |
| 15:43.33 | tRSS | and I also want both softphones to ring if a call comes? |
| 15:43.34 | xnon | and later password |
| 15:43.44 | *** join/#asterisk ComputerWarm (n=donc@209.29.156.149) |
| 15:43.57 | ComputerWarm | hello |
| 15:44.00 | xnon | i try enter dis data but she says that is incorrect |
| 15:44.02 | TheCompWiz | is anyone having problems with the "Online Module Repository"? |
| 15:44.07 | *** join/#asterisk sp0n9e (n=sp0n9e@69.12.216.48) |
| 15:44.13 | xnon | eKol can u helpme |
| 15:44.17 | *** join/#asterisk SplasPood (n=jwb@gate.lga2.us.voxel.net) |
| 15:44.19 | eKo1 | tRSS: set up an entry for each phone a sip.conf and an appropriate dialplan in extensions.conf to have both phones ring. |
| 15:44.23 | ComputerWarm | how does one go about attaching a xml to the dial plan |
| 15:44.32 | sp0n9e | is there any reason festival cuts off the end of a string when speaking? |
| 15:44.35 | eKo1 | xnon: I don't understand you. Sorry. |
| 15:44.51 | tRSS | eKo1: i think there should be an easier way then doing what you just said |
| 15:45.09 | eKo1 | tRSS: That is the easiest. |
| 15:45.19 | xnon | ok my english is not so good |
| 15:45.25 | xnon | :(sorry |
| 15:45.32 | Shark_y | [TK]D-Fender between the moment that the call is placed and when the called phone is actually ringing |
| 15:45.34 | *** part/#asterisk xnon (i=xnon@200.8.4.227) |
| 15:45.37 | *** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.mn.comcast.net) |
| 15:46.18 | *** join/#asterisk ree (n=ree@3e70ccca.adsl.enternet.hu) |
| 15:46.35 | Shark_y | [TK]D-Fender the same delay as when I call from external and the moment that * calls the internal extension |
| 15:46.44 | hmmhesays | is there any reverse arp commands in linux to find an ip address? |
| 15:47.32 | *** part/#asterisk acrg (n=aragon@decoder.geek.sh) |
| 15:48.14 | Shark_y | [TK]D-Fender for example I hear already a ringing when * still has not started to forward the call |
| 15:48.23 | eKo1 | hmmhesays: you mean a command that takes a mac address and returns an IP address? |
| 15:48.23 | fourcheeze | hmmhesays: arp -n |
| 15:48.33 | hmmhesays | eKo1: yes, reverse arp |
| 15:48.54 | fourcheeze | hmmhesays: arp -n | grep MA:CA:DR:ESS |
| 15:49.08 | hmmhesays | that is only if the equipment is in your arp cache |
| 15:49.23 | fourcheeze | ahh you want to send out a "who has" ? |
| 15:49.28 | *** join/#asterisk KDan (i=nobody@sleek.sleektech.nl) |
| 15:49.33 | hmmhesays | yes a reverse arp request |
| 15:49.39 | fourcheeze | hmm |
| 15:50.49 | *** join/#asterisk beyond (n=beyond@200.192.160.100) |
| 15:51.18 | KDan | Hello. Am trying to set up eSMS.com to point to my asterisk server... my server sits behind a NAT and I've applied the "tips" at http://www.voip-info.org/wiki/view/tips .. When I make a call using Xlite from my laptop on the LAN, Asterisk responds. When I call the 0845 with my phone, Asterisk says that the "default" context isn't defined. If I define [default], then all that happens is.. nothing at all - no asterisk notices of incoming calls, nada |
| 15:51.30 | ComputerWarm | any ideas please?? |
| 15:51.33 | [TK]D-Fender | Shark_y : * considers the line ringing as soon as Zap starts accepting the #. Thats jsut the ways it is |
| 15:51.49 | Dr-Linux|work | question, i just made some changes in zapata.conf, does it take effect with only "reload" or i need to restart the asterisk? |
| 15:52.00 | fourcheeze | hmm, you could do a broadcast ping of the network |
| 15:52.06 | fourcheeze | and then the answer would be in your arp cache |
| 15:52.18 | KDan | am kind of at a loss as to what to do next. is the problem a networking/NAT problem or something else? |
| 15:52.38 | eKo1 | KDan: maybe your default context is wrong |
| 15:52.49 | eKo1 | Dr-Linux|work: depends on the changes |
| 15:52.52 | KDan | eKo1: when I call my default context from xlite it works |
| 15:53.01 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
| 15:53.01 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:53.18 | fourcheeze | s/hmm/ hmmhesays |
| 15:53.18 | eKo1 | KDan: and when does it not work? |
| 15:53.27 | KDan | when calling using the PSTN |
| 15:54.01 | KDan | if the context isn't defined, asterisk complains that the context isn't defined - so the 0844 number is reaching my server, clearly... but when the context is defined, nothing happens and the line hangs up after a while |
| 15:54.23 | Dr-Linux|work | eKo1, i just changed the txgain |
| 15:54.53 | eKo1 | Dr-Linux|work: check the docs. I'm pretty sure that needs a restart (or unloading/reloading chan_zap.so). |
| 15:55.17 | hmmhesays | fourcheeze maybe |
| 15:55.31 | Dr-Linux|work | eKo1, unloading/reloading chan_zap.so is same like reload |
| 15:55.44 | eKo1 | Dr-Linux|work: No it isn't. |
| 15:55.46 | Shark_y | [TK]D-Fender but if I call from external, and I hook as soon as I see "Starting simple switch on 'Zap/1-1'" asterisk contiunes to call the hunt group for at least 5 seconds so the phone are ringing when the call is not anymore active |
| 15:56.22 | Shark_y | [TK]D-Fender the tdm400p seems delayed |
| 15:56.28 | Dr-Linux|work | eKo1, well, i done "reload chan_zap.so" |
| 15:56.44 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.41) |
| 15:57.30 | *** join/#asterisk watchy (n=gweg@office2.gwhsi.com) |
| 15:57.52 | watchy | anyone know digiums address so i can fly there and shove these mother fucking tdm400ps up everyones ass |
| 15:58.07 | mog | oucha |
| 15:58.15 | sp0n9e | those cards can be sharp too |
| 15:58.20 | mog | 150 west park loop suite 100 huntsville al usa 35806 |
| 15:58.23 | mog | is our address |
| 15:58.31 | watchy | wow al is pretty close i can drive |
| 15:58.33 | mog | anything i can assist you with waba |
| 15:58.34 | KDan | that's not a very nice thing to do, watchy |
| 15:58.39 | mog | er watchy |
| 15:58.41 | eKo1 | Dr-Linux|work: it is not the same I think. Check the source code. |
| 15:58.47 | watchy | kdan: niether is echo that cant be fixed |
| 15:59.00 | watchy | i;m about to shoot myself live on webcam and have it emailed to digium |
| 15:59.11 | mog | thats pretty graphic |
| 15:59.12 | Shark_y | should rely this on the fact that when I compiled zaptel it has warned that something with rtc is not ok? which are the correct kernel settings? |
| 15:59.16 | KDan | watchy: I'd rather have echo than a sharp-cornered pci card up my ass |
| 15:59.18 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 15:59.19 | mog | have you called support watchy ? |
| 15:59.23 | kpettit | death by asterisk. that's a new one |
| 15:59.30 | watchy | yes he told me try mark2 and that was about it |
| 15:59.34 | watchy | he was as helpful as a dead dog |
| 15:59.40 | mog | well let me see what i can do for you |
| 15:59.46 | mog | do you have a ticket number or anything? |
| 16:00.02 | watchy | i'll give you root to the box if you want |
| 16:00.04 | watchy | i'm so tired of it |
| 16:00.09 | watchy | yea let me check my mail |
| 16:00.10 | mog | i understand |
| 16:00.15 | mog | im gonna get you some help |
| 16:00.17 | Qwell | mog++ |
| 16:00.25 | mog | just more information you can give me the better |
| 16:00.32 | [TK]D-Fender | Shark_y : You referring to a cally coming IN on the TDM or out TO the TDM? |
| 16:00.37 | Qwell | mog: You're like...level 9 support. :) |
| 16:00.47 | watchy | i got 2 TDM400ps with 8 fxos |
| 16:01.05 | watchy | could the actually fucking phone lines suck so bad its impossible to do echo cancel in software? |
| 16:01.09 | *** part/#asterisk [g2] (n=g2@nslu2-linux/g2) |
| 16:01.10 | KDan | anyone got a clue about my asterisk problem? :-( Repasting: 17:40 < KDan> Hello. Am trying to set up eSMS.com to point to my asterisk server... my server sits behind a NAT and I've applied the "tips" at http://www.voip-info.org/wiki/view/tips .. When I make a call using Xlite from my laptop on the LAN, Asterisk responds. When I call the 0845 with my phone, Asterisk says that the "default" context isn't defined. If I define [default], then all that happe |
| 16:01.15 | mog | its very rare |
| 16:01.18 | mog | watchy, |
| 16:01.25 | watchy | i;m lookin for my ticket |
| 16:01.32 | mog | our software echo can and other stuff rock pretty hard |
| 16:01.37 | Qwell | bbl |
| 16:01.40 | *** part/#asterisk fourcheeze (n=rich@office.callmaster.co.uk) |
| 16:01.49 | watchy | i belieeve you mog |
| 16:02.02 | E-bola | IF you got a pure sip based setup and use a provider to connect to PSTN |
| 16:02.06 | *** join/#asterisk Astinus- (n=a@213.167.111.138) |
| 16:02.17 | E-bola | is there anything u can do to do echo cancellation? |
| 16:02.19 | mog | just a bit tricky to get going at first |
| 16:02.37 | mog | for each individual place that is |
| 16:02.43 | sp0n9e | E-bola: there is both hardware and software echo cancellation |
| 16:02.51 | watchy | we had crackles and pops etc but i replaced the server |
| 16:02.56 | watchy | that fixed that i think %100 |
| 16:03.03 | mog | good stuff |
| 16:03.12 | E-bola | sp0n9e: so even if the echo is introduced at my providers connect i can still cancel it out? |
| 16:03.13 | watchy | but i have never beenable to get echo gone |
| 16:03.33 | Astinus- | Is T1/E1 a physical medium? |
| 16:03.37 | watchy | mog: you aint vlad are you? |
| 16:03.45 | Juggie | Astinus-, no. |
| 16:03.49 | mog | nope |
| 16:03.51 | mog | im mog |
| 16:03.54 | watchy | cuz hes a bitch that never answered an email i replied to him |
| 16:03.56 | mog | aka matt ogorman |
| 16:04.01 | watchy | so find him in your off and punch him |
| 16:04.01 | sp0n9e | E-bola: the echo afaik is caused by the line and the left hand rule causing "crosstalk" |
| 16:04.04 | Astinus- | Juggie: more like, a link layer thingie? |
| 16:04.04 | mog | i am sorry for that |
| 16:04.09 | watchy | PUNCH |
| 16:04.18 | watchy | 60591 |
| 16:04.21 | *** join/#asterisk jbalcomb (n=jbalcomb@216.28.180.158) |
| 16:04.22 | Juggie | Astinus-, i'm not sure of the exact terminology but something like that yes |
| 16:04.23 | E-bola | sp0n9e: i was under the impression it was also a combination of delay |
| 16:04.31 | Juggie | t1 can run over coper/fiber/ethernet/coax etc... |
| 16:04.34 | Juggie | hence its not physical |
| 16:04.34 | E-bola | something about normal phones having such a good responsetime u cant hear the echo |
| 16:04.39 | E-bola | but the more delay u have the worse? |
| 16:04.45 | *** join/#asterisk af_ (n=af@ip-164-6.sn2.eutelia.it) |
| 16:04.49 | *** join/#asterisk bkw_ (n=brian@asterisk/friend-and-developer/bkw) |
| 16:04.58 | sp0n9e | i'm new to all this |
| 16:04.58 | Astinus- | Juggie: so, i can have a adsl2 line and have a t1 on it? |
| 16:05.02 | E-bola | me too hehe |
| 16:05.14 | E-bola | Whats anoyign though, is the conversations are always 100% perfect on my end |
| 16:05.15 | sp0n9e | i'm not even finished my first major dialplan, lol |
| 16:05.20 | Shark_y | [TK]D-Fender both but the delay is more evident in incoming calls |
| 16:05.21 | E-bola | its the ppl im calling who complain im "far away" |
| 16:05.22 | Juggie | i'm not sure.... probally high latency? |
| 16:05.27 | E-bola | and that they can hear themselfs sometimes |
| 16:05.38 | Juggie | like i said, i'm not really too aware. |
| 16:05.54 | E-bola | well an E1 line is 2mbit |
| 16:06.00 | E-bola | so u cant have that on an adsl2 line |
| 16:06.08 | E-bola | atleast the adsl2 lines we have here are only 1mbit up |
| 16:06.31 | eKo1 | adsl2 ? |
| 16:06.35 | E-bola | i dont know how it related to phones though, but data communication wise an adsl2 woudl be too slow upstream |
| 16:06.42 | jbalcomb | this conversation doesn't make any sense. |
| 16:06.43 | Astinus- | E-bola: they are 20 / 12 her |
| 16:06.53 | E-bola | astinus: u sure thats not vdsl? |
| 16:07.01 | Astinus- | THEY CAll it adsl2+ here |
| 16:07.05 | E-bola | there is like a million dsl technologies hehe |
| 16:07.21 | sp0n9e | i live in the sticks and i get cable at 3/.25 :( |
| 16:07.26 | sp0n9e | where i used to live i had 10/1 |
| 16:07.39 | mut | least you can get cable |
| 16:07.44 | *** join/#asterisk SwK (n=Silik0nJ@65.169.134.2) |
| 16:07.47 | *** join/#asterisk DarKnesS_WolF (n=wolf@81.10.111.8) |
| 16:07.53 | mut | heh we're the only company with dsl around here and we just rolled that out this year |
| 16:08.01 | sp0n9e | lol |
| 16:08.05 | E-bola | 3rd world country? |
| 16:08.06 | E-bola | :) |
| 16:08.10 | mut | northern michigan |
| 16:08.14 | eKo1 | lol |
| 16:08.15 | E-bola | indeed |
| 16:08.16 | E-bola | :P |
| 16:08.23 | *** join/#asterisk bkw__ (n=brian@asterisk/friend-and-developer/bkw) |
| 16:08.25 | sp0n9e | the isp i was at before rolled out fiber everywhere...sometimes i regret moving |
| 16:08.29 | eKo1 | mut: adsl or sdsl? |
| 16:08.30 | E-bola | got a friend in maine, all he can get is 4mbit dsl |
| 16:08.32 | *** part/#asterisk SwK (n=Silik0nJ@65.169.134.2) |
| 16:08.37 | *** join/#asterisk SwK (n=Silik0nJ@65.169.134.2) |
| 16:08.38 | *** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie) |
| 16:08.51 | *** join/#asterisk evisu (i=hIRC@bzq-88-155-80-250.red.bezeqint.net) |
| 16:09.15 | mut | adsl/adsl2+ and sdsl |
| 16:09.20 | watchy | y un |
| 16:09.25 | *** join/#asterisk gandhijee (n=gandhije@65.169.134.2) |
| 16:09.26 | *** join/#asterisk s0lid (n=jlq@203.192.160.234) |
| 16:09.44 | eKo1 | mut: does that include phone service? |
| 16:09.56 | Juggie | i wish i could get fiber to the home |
| 16:09.59 | Juggie | that would be sweet |
| 16:10.05 | mut | yea, we also provide phone service if they want |
| 16:10.07 | mut | or voip |
| 16:10.11 | Juggie | the best available around my area is fiber to the node |
| 16:10.18 | sp0n9e | Juggie: it's not as cool as you think if your isp doesn't buy enough peering |
| 16:10.34 | *** join/#asterisk LenOK (n=ln@office-181.telengy.net) |
| 16:11.12 | eKo1 | mut: you mean you have dsl modems that do voip as well? |
| 16:11.20 | mut | yea |
| 16:11.20 | *** join/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
| 16:11.27 | mut | or use an ata |
| 16:11.45 | eKo1 | What modems? |
| 16:11.53 | mut | zooms |
| 16:12.04 | Shark_y | [TK]D-Fender I'm recompiling the kernel with RTC enabled, and then try to recompile zaptel, what else can I do to avoid such delays? |
| 16:12.54 | *** join/#asterisk coppice (n=chatzill@127.166.17.210.dyn.pacific.net.hk) |
| 16:13.09 | KDan | if i have a user named "sipstream" in my sip config ([sipstream] ... username=sipstream) then the SIP url for this would be "sipstream@84.x.y.z" right? |
| 16:13.17 | mut | i personally don't recommend them |
| 16:13.18 | eKo1 | mut: Interesting. Thanks. |
| 16:13.24 | eKo1 | Why not? |
| 16:13.38 | mut | if you run ata's over the modems w/o voip they CRASH like crazy |
| 16:13.43 | mut | if you run sip of any kind |
| 16:13.45 | mut | they die |
| 16:13.58 | eKo1 | That's not good. |
| 16:13.59 | mut | the adsl/voip ones seem to do fine tho |
| 16:14.20 | mut | processing power for the quick packet stream maybe i dunno |
| 16:14.27 | mut | but they can't handle it |
| 16:15.34 | tzanger | 23 IP501s for $3800 + shipping, not bad... |
| 16:20.19 | jbalcomb | tzanger: where'd you find tha deal? |
| 16:20.35 | *** part/#asterisk javar (n=javar@200.118.174.253) |
| 16:21.13 | smackus | I am trying to figure out the dialplan portion of the sip.cfg on my polycom 301 phones. I have in my extensions.conf an extension '#1' for voice mail access... but I suspect that it is not working because of the sip.cfg dial plan. So i added the # to the dialplan. I don't know if i am understanding this portion of the file. Could someone please shed some light? http://pastebin.ca/110367 |
| 16:21.23 | tzanger | jbalcomb: -biz |
| 16:21.55 | jbalcomb | tzanger: eh? |
| 16:22.01 | tzanger | the -biz list |
| 16:24.49 | mtaht4 | is anyone hacking on T.38 out there? |
| 16:26.03 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 16:26.03 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 16:26.59 | mog | mtaht4, what you mean? |
| 16:27.30 | mtaht4 | Well, I have just setup a couple asterisk boxes straight from svn |
| 16:27.34 | *** join/#asterisk Delta239 (n=blablabl@200.124.18.171) |
| 16:27.41 | mog | k |
| 16:27.46 | *** join/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br) |
| 16:28.03 | mtaht4 | was going to give T.38 a try with the ATA I have but there seems to be a gap between documentation and practice |
| 16:28.03 | mog | then you should have latest t38 pass through stuff we brought in |
| 16:28.08 | mog | its documented in sip.conf |
| 16:28.09 | Delta239 | hey is this config that i have on my extensions.conf supposed to do a roll over? |
| 16:28.10 | Delta239 | http://pastebin.ca/110369 |
| 16:28.35 | Delta239 | all my sip accounts are registered propperly on sip.conf and on the cli it shows registered |
| 16:29.53 | mtaht4 | mog: thx |
| 16:30.10 | *** join/#asterisk burnproof (n=burnproo@host-202-163-241-94.dhcp.infocom.ph) |
| 16:30.37 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 16:31.45 | *** part/#asterisk smackus (n=ckwall@63.149.122.93) |
| 16:32.51 | Dr-Linux|work | question, what we can tx/rx gain value? is it frequency or what? |
| 16:32.58 | mog | db |
| 16:33.07 | mog | you can go between -5 to 5 reliably |
| 16:33.13 | mog | any more than that is crazy |
| 16:33.26 | nortex | It is decibles. |
| 16:33.35 | E-bola | my linksys phone lets me chose between -/+ 6 |
| 16:33.51 | mog | well different devices can handle it better |
| 16:34.00 | mog | but i think going between 5 and -5 in software is safe |
| 16:34.01 | Qwell[] | E-bola: Some devices let you go -/+ 100, but that doesn't mean it's a good idea :) |
| 16:34.19 | mog | you can go between -255 and 255 i believe in software |
| 16:34.22 | mog | it just isnt good idea |
| 16:34.27 | *** join/#asterisk hmmhesays (n=ohyeah@66.173.103.110) |
| 16:34.46 | *** join/#asterisk willy_1234 (n=icechat5@62.231.36.101) |
| 16:35.00 | E-bola | I got a stupid question: Adjusting the gain is simply to get a higher/lower volume right? |
| 16:35.06 | willy_1234 | im using freepbx and i cant record incomming calls |
| 16:35.15 | mog | yeah it ups and lowers volume |
| 16:35.21 | Qwell[] | E-bola: pretty much - but it helps (A LOT) with echo |
| 16:35.23 | willy_1234 | any help please? |
| 16:35.31 | mog | but to much can cause echo and top outs |
| 16:35.35 | Qwell[] | indeed |
| 16:35.41 | E-bola | Qwell: to cancel echo should u increase or decrease? |
| 16:35.48 | mog | FreePBX/AMP/Asterisk@Home/Trixbox support in #freepbx |
| 16:35.50 | Qwell[] | So, what's the rule of thumb, if you have echo, lower it? |
| 16:35.51 | mog | in topic willy_1234 |
| 16:36.01 | mog | dont do that Quintana |
| 16:36.05 | mog | er Qwell |
| 16:36.11 | mog | rule of thumb is dont touch gain |
| 16:36.14 | Qwell[] | heh |
| 16:36.15 | mog | its the last thing you try |
| 16:36.29 | Qwell[] | ok |
| 16:36.36 | E-bola | Hmm i increased my spa922's gain to +6 |
| 16:36.44 | E-bola | maybe thats why ppl started telling me they could hear themselfs hehe |
| 16:36.47 | Qwell[] | I know about >< much about tdm |
| 16:36.56 | Dr-Linux|work | Qwell[], Red Hat Inc is using strong arm with me for my domain :( |
| 16:36.57 | E-bola | But before that they complained i was "very far away" and could barely hear me |
| 16:36.57 | E-bola | :/ |
| 16:37.07 | mog | what domain do you have Dr-Linux|work |
| 16:37.14 | Qwell[] | Dr-Linux|work: Sorry to hear that... Let me talk to my friend tonight |
| 16:37.32 | willy_1234 | so how would u setup call recourding in a queue manually using the commandline and text files |
| 16:37.33 | Qwell[] | Dr-Linux|work: He may be able to, at the very least, get you a few bucks (ie; what you paid for it) |
| 16:37.37 | Qwell[] | DON'T SAY ANYTHING TO THEM YET |
| 16:37.43 | Dr-Linux|work | mog: i have alot of .. about 25 |
| 16:37.58 | Dr-Linux|work | mog, but this one is redhat.pk |
| 16:38.01 | *** join/#asterisk [Airwolf] (n=airwolf@dsl5402DE03.pool.t-online.hu) |
| 16:38.03 | mog | which one are they strongarming you about is my question |
| 16:38.04 | mog | ahh |
| 16:38.11 | mog | what are you using it for? |
| 16:38.12 | *** join/#asterisk ronaldl79 (n=chatzill@d198-53-139-22.abhsia.telus.net) |
| 16:38.19 | Qwell[] | Dr-Linux|work: Don't answer that question :) |
| 16:38.25 | watchy | anyone know what 1.6.7 of polycom firmware does 1.6.6 dont? |
| 16:38.26 | E-bola | lol |
| 16:38.29 | watchy | should i upgrade? |
| 16:38.34 | mog | weird |
| 16:38.40 | E-bola | if ur happy with 1.6.6 why upgrade? |
| 16:38.45 | E-bola | dont fix it if it isnt broken :) |
| 16:38.46 | mog | so you have your own red _ hat linux? |
| 16:38.54 | watchy | true |
| 16:38.55 | Dr-Linux|work | Qwell[], i'm using it for passion .. it's not a commercial :( |
| 16:39.11 | Qwell[] | Dr-Linux|work: seriously though, if you haven't already responded to their email...please don't |
| 16:39.11 | Dr-Linux|work | mog, i just wanna know my rights |
| 16:39.19 | Dr-Linux|work | i don't know the stuff |
| 16:39.19 | Qwell[] | and if you would, forward their email to me |
| 16:39.24 | mog | eh /me is not a lawyer either |
| 16:39.32 | Qwell[] | I'll ask my friend tonight about what you can do...he's well trained in this regard |
| 16:39.40 | mog | personaly i think your gonna have to give it up |
| 16:39.46 | Qwell[] | mog: definitely |
| 16:39.48 | Dr-Linux|work | Qwell[], where you were before :( :( :( i respended them once email reply |
| 16:39.58 | mog | unless you live in pk |
| 16:40.00 | Qwell[] | Dr-Linux|work: it's still okay...send me your reply also |
| 16:40.04 | E-bola | put a picture of a redhat on it |
| 16:40.04 | mog | and pk doesnt observer us law |
| 16:40.08 | mog | in any way |
| 16:40.09 | E-bola | and claim u are a fan of red hats |
| 16:40.13 | E-bola | its not a name its a thing |
| 16:40.19 | E-bola | so i think uir chances are pretty good |
| 16:40.32 | E-bola | like apple cant claim rights for apple.something |
| 16:40.42 | E-bola | because their name is a common object |
| 16:40.49 | E-bola | atleast that how it works here in denmark |
| 16:40.52 | Qwell[] | Dr-Linux|work: I /msg'd you my email address...forward the stuff there, if you would |
| 16:41.00 | Dr-Linux|work | aww |
| 16:41.17 | Dr-Linux|work | should i remove all data related red hat linux? |
| 16:41.27 | Qwell[] | Dr-Linux|work: Just leave the site as it is, for now |
| 16:41.50 | Qwell[] | they've already got snapshots of the site, and everything... changing it now is an admission of guilt - which is why I said, don't do anything yet |
| 16:41.51 | Dr-Linux|work | Qwell[], thanks dude, hopefully you will help me for other doamins as well |
| 16:41.52 | Dr-Linux|work | like |
| 16:41.53 | mog | redhat isnt as common |
| 16:41.57 | Dr-Linux|work | compaq.pk |
| 16:41.58 | mog | its combo word |
| 16:42.04 | Dr-Linux|work | toyota.pk |
| 16:42.11 | Dr-Linux|work | sony.pk |
| 16:42.12 | mog | man |
| 16:42.17 | Qwell[] | toyota you could argue, heh |
| 16:42.20 | mog | you are just trying to piss off the world |
| 16:42.23 | Dr-Linux|work | and many more |
| 16:42.25 | mog | why toyota? |
| 16:42.30 | Qwell[] | mog: last name.. |
| 16:42.31 | E-bola | heh u just registered random company names? |
| 16:42.35 | E-bola | i hope they take it form you |
| 16:42.36 | Qwell[] | mog: ala nissan.com |
| 16:42.39 | E-bola | without you getting a cent |
| 16:42.40 | Dr-Linux|work | hehe |
| 16:42.44 | Dr-Linux|work | noooooooot only companies |
| 16:42.49 | Dr-Linux|work | the safe one as well |
| 16:42.52 | Dr-Linux|work | tele.pk |
| 16:42.55 | Dr-Linux|work | networks.pk |
| 16:43.00 | E-bola | domain pirating is lame |
| 16:43.06 | burnproof | :) |
| 16:43.17 | mog | yeah seriously |
| 16:43.25 | mog | why did you do this Dr-Linux|work |
| 16:43.35 | mog | buying tele or networks is fine |
| 16:43.37 | Qwell[] | Don't answer that question either :P |
| 16:43.55 | mog | but all the others |
| 16:43.56 | mog | dizamn |
| 16:43.56 | *** join/#asterisk champster (n=asterisk@AH.tescogroup.com) |
| 16:43.56 | *** join/#asterisk smackus (n=ckwall@63.149.122.93) |
| 16:43.59 | burnproof | hehehe interesting :p |
| 16:44.04 | Dr-Linux|work | mog: bcoz it's valuable domains, and why the hell they didn't registerd their domains by themself |
| 16:44.12 | Dr-Linux|work | well, i damn care about all |
| 16:44.17 | Dr-Linux|work | but care about redhat.pk |
| 16:44.50 | Qwell[] | Dr-Linux|work: let me know when you've forwarded those emails, so I can pass them along |
| 16:45.01 | willy_1234 | the sound recordings are 3.2K files any ideas why it wont record as it creates the files |
| 16:45.19 | Dr-Linux|work | Qwell[], what's your email address? |
| 16:45.21 | ronaldl79 | Any recommendations for unlimited incoming DIDs? |
| 16:45.35 | nortex | willy_1234, Did you check in #freepbx ? |
| 16:45.37 | Qwell[] | Dr-Linux|work: I /msg'd you :) |
| 16:45.46 | Dr-Linux|work | aww on wait |
| 16:45.55 | Dr-Linux|work | s/on/ok |
| 16:46.21 | smackus | I am trying to to dial #1. cannot get it to work. |
| 16:46.28 | smackus | i am using the polycom 301 |
| 16:46.55 | willy_1234 | they dont seem to know |
| 16:47.36 | E-bola | Do anybody know how u get a SPA922 to dial as soon as u have entered 8 digits? |
| 16:47.49 | E-bola | all phone numbers in .dk are 8 digits, and its anoying to have to press dial everytime |
| 16:47.51 | E-bola | like it was a cellphone |
| 16:48.26 | burnproof | E-bola: sorry, i don't know about spa222 but you could also check if spa9222 does have a dial plan |
| 16:48.32 | watchy | ok to clarify i dont want to kill digium anymore |
| 16:48.41 | watchy | i will not be shaving tdms up anyones ass today |
| 16:48.44 | watchy | shoving |
| 16:48.45 | burnproof | E-bola: like my PAP-N2 linksys ATA |
| 16:48.48 | Dr-Linux|work | Qwell[], i sent you the email |
| 16:48.50 | *** join/#asterisk Juggie (n=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 16:48.53 | Qwell[] | Dr-Linux|work: ok |
| 16:48.54 | E-bola | burnproof: it doesnt as far as i know |
| 16:49.05 | burnproof | E-bola: ouch :p great |
| 16:49.12 | E-bola | it has something called assisted dialing |
| 16:49.16 | E-bola | but dunno what that is really |
| 16:49.23 | watchy | thanks digium and mog |
| 16:49.34 | mog | ^_^ |
| 16:50.17 | Qwell[] | Dr-Linux|work: Is the server in the us or pk? |
| 16:50.23 | Qwell[] | the server that hosts that domain |
| 16:51.07 | burnproof | mog: i'll really like your jingle stuff really thanks man, my boss was pretty amaze how asterisk + jingle works together |
| 16:51.38 | mog | man i need a gif of the mog dancing from final fantasy 6 |
| 16:51.49 | burnproof | =)) |
| 16:51.51 | E-bola | whats jingle? |
| 16:52.07 | mog | google talk protocol |
| 16:52.12 | mog | but its open |
| 16:52.15 | mog | similar to sip |
| 16:52.18 | mog | but over jabber |
| 16:52.31 | Blafasel | No.jingle is client to client. |
| 16:52.33 | Dr-Linux|work | Qwell[], what server? |
| 16:52.35 | Blafasel | jabber is dead. |
| 16:52.40 | Qwell[] | Dr-Linux|work: redhat.pk server |
| 16:52.42 | Blafasel | xmpp is the control line more or less.. |
| 16:52.43 | mog | are you kidding Blafasel |
| 16:52.46 | Blafasel | No |
| 16:52.48 | burnproof | Blafasel: i beg your pardon |
| 16:52.49 | mog | you have no idea what you are talking about |
| 16:52.51 | Blafasel | Just nitpicking ;) |
| 16:52.53 | mog | jingle runs over jabber |
| 16:52.56 | Blafasel | No |
| 16:52.57 | mog | its an iq message |
| 16:53.01 | mog | its not peer to peer |
| 16:53.02 | Blafasel | It's client-side only |
| 16:53.04 | mog | just the media is |
| 16:53.11 | mog | iq messages run up to server down to client |
| 16:53.16 | mog | believe me i know what im talking about |
| 16:53.17 | Dr-Linux|work | Qwell[], the hosting server or domain server? |
| 16:53.18 | Blafasel | Right, the routing is xmpp. |
| 16:53.23 | Blafasel | The server doesn't know jingle though |
| 16:53.29 | ComputerWarm | a C api is it compiled ? |
| 16:53.30 | Qwell[] | Dr-Linux|work: hosting server |
| 16:53.30 | mog | it passes the message |
| 16:53.41 | Blafasel | Right - it passes the control messages |
| 16:53.44 | Blafasel | Not the voice data |
| 16:53.49 | mog | thats like saying sip is dead long live rtp |
| 16:54.02 | mog | you need signalling protocol to get to the point you can transfer media |
| 16:54.10 | mog | jingle is just subset of jabber protocol |
| 16:54.10 | Blafasel | No - jabber = dead was related to the old name.. It's xmpp (im) for a long time ;) |
| 16:54.12 | Dr-Linux|work | Qwell[], my hosting is different and domain is different |
| 16:54.21 | mog | xmpp is stupid name |
| 16:54.22 | Dr-Linux|work | Qwell[], web hosting is from US servre |
| 16:54.23 | Qwell[] | Dr-Linux|work: right, but where is the actual server? |
| 16:54.26 | Qwell[] | ok |
| 16:54.27 | mog | jabber rolls of tongue much easier |
| 16:54.51 | mog | its like refering to mgcp as rfc 2XXX < me forgot it |
| 16:54.51 | Dr-Linux|work | Qwell[], that's no problem for me, i can just change the DNS in 2 minutes |
| 16:54.55 | mog | mgcp just sounds better |
| 16:54.56 | burnproof | mog: any idea google talk will support dtmf in the neat future? |
| 16:55.00 | burnproof | mog: :) |
| 16:55.02 | Blafasel | That's why I said I was just nitpicking. |
| 16:55.10 | mog | we will support it today |
| 16:55.16 | mog | i dont know when clients will support it |
| 16:55.20 | Qwell[] | Dr-Linux|work: okay, I sent him an email..hopefully he'll get back to me today, if not, I'll talk to him tonight |
| 16:55.21 | Blafasel | But jingle is related to jabber like sip is to udp.. |
| 16:55.23 | mog | jabbin said they will have some test code soon |
| 16:55.28 | mog | no its not |
| 16:55.34 | mog | your just nubbing all over the place |
| 16:55.43 | Dr-Linux|work | Qwell[], can you tell me what you have sent to him? :) |
| 16:55.51 | Qwell[] | Dr-Linux|work: I just asked what you can do |
| 16:55.52 | mog | ugh lets just stop arguing Blafasel |
| 16:55.55 | KDan | when a call is coming in from the PSTN onto SIP, does it automatically have an extension set already, or does it start on s? |
| 16:55.55 | Blafasel | Aye |
| 16:55.58 | burnproof | mog: the code is already there in trunk? |
| 16:56.02 | Qwell[] | and, how you can get the money back that you paid for the domain, if possible |
| 16:56.06 | Dr-Linux|work | Qwell[], i mean to your friend? |
| 16:56.12 | mog | not dtmf code |
| 16:56.17 | Dr-Linux|work | i thought you send an email to red hat :P |
| 16:56.18 | mog | i have been meening to put it in |
| 16:56.20 | Qwell[] | no, heh |
| 16:56.21 | burnproof | oic |
| 16:56.22 | Qwell[] | to my friend |
| 16:56.24 | mog | but there is a big patch i have been working on |
| 16:56.28 | mog | and i want to finish it |
| 16:56.41 | Qwell[] | brb |
| 16:56.50 | *** join/#asterisk expat_iain (n=expat_ia@194.204.99.166) |
| 16:57.04 | burnproof | mog: great i'll be glad to hear from it soon :p |
| 16:57.10 | mog | and i have been working on other secret things that have eaten time |
| 16:57.20 | burnproof | heheheheh |
| 16:57.22 | mog | but i am finishing it today, so that we can test interop with jabbin tommorrow |
| 16:57.23 | KDan | Question: when a call is coming in from the PSTN onto SIP, does it automatically have an extension set already, or does it start on s? |
| 16:57.44 | mog | it does what ever you tell it to KDan |
| 16:57.52 | mog | you have dial plan in between the two |
| 16:57.57 | mog | it cant just bridge to sip |
| 16:58.06 | *** join/#asterisk bkw_ (n=brian@asterisk/friend-and-developer/bkw) |
| 16:58.06 | KDan | mog: I'm not telling it, an external SIP provider is forwarding the call to my asterisk SIP server |
| 16:58.31 | KDan | i can't seem to be able to make it pick up anything on the context i've forwarded it to |
| 16:58.45 | mog | you can do a sip debug |
| 16:58.50 | burnproof | KDan: pastebin is your friend :p |
| 16:58.54 | mog | and figure out what its doing real fast |
| 16:59.55 | burnproof | mog: any idea when will be the 1.4-beta will be release ? or any from you guys? |
| 17:00.05 | KDan | mog: i set debug level to 10 and also typed "debug channel sip" but, a) the CLI complained that that channel didn't exist and said it would display all debug channels, and b) there's no more debug messages than there was before |
| 17:00.18 | mog | sip debug |
| 17:00.20 | hmmhesays | paranoia paranoia everybody's coming to get me |
| 17:00.21 | mog | is the command |
| 17:00.24 | KDan | aha |
| 17:00.29 | mog | dont need debug level 10 |
| 17:00.32 | mog | just sip debug |
| 17:00.32 | KDan | cheers |
| 17:00.51 | mog | svn co http://svn.digium.com/svn/asterisk/trunk asterisk-1.4-beta |
| 17:00.53 | mog | ^_^ |
| 17:01.01 | file | naughty mog |
| 17:01.03 | KDan | excelltn, am getting some debug info now. thanks mog :-) |
| 17:01.08 | burnproof | :) |
| 17:01.09 | mog | no problem KDan |
| 17:01.35 | mog | there are a few things that need to get settled before we go beta |
| 17:01.46 | mog | one of them being imap support ,yet another thing i need to finish |
| 17:01.52 | mog | some jingle stuff |
| 17:02.04 | E-bola | imap???? |
| 17:02.08 | burnproof | imap support on voicemail? |
| 17:02.08 | mog | and im sure there are one or two other projects people need to clean a little before we finsih |
| 17:02.12 | mog | yes burnproof |
| 17:02.14 | mog | its in trunk |
| 17:02.19 | mog | it just has a few kinks |
| 17:02.20 | E-bola | whats the purpose of that lol |
| 17:02.24 | mog | er not in trunk |
| 17:02.27 | mog | in a trunk branch |
| 17:02.38 | mog | so that you check your voicemail from email it gets deleted in voicemail |
| 17:02.39 | mog | for one |
| 17:02.47 | E-bola | check it how? |
| 17:02.52 | mog | in your email |
| 17:02.52 | E-bola | each message is an email with attachment? |
| 17:02.53 | *** join/#asterisk tzanger (n=tzanger@mixdown.ca) |
| 17:03.00 | mog | yeah we already do that |
| 17:03.07 | mog | but if you listen to them in your mail |
| 17:03.13 | E-bola | i know i get a mail everytime i get a msg |
| 17:03.16 | mog | you still have to delete them from the phone |
| 17:03.17 | E-bola | with ther msg attached |
| 17:03.24 | mog | now you wont |
| 17:03.24 | E-bola | hmmm |
| 17:03.25 | *** part/#asterisk SpaceBass (n=sp@static-71-251-230-6.rcmdva.fios.verizon.net) |
| 17:03.26 | mog | if you use it |
| 17:03.29 | mog | its spiffy |
| 17:03.34 | E-bola | but u would have to use asterisk as ur imap server then or? |
| 17:03.49 | mog | no |
| 17:03.49 | E-bola | i dont think i understand the architecture |
| 17:03.49 | mog | it connects as a client |
| 17:03.50 | mog | or administarative cleint |
| 17:04.05 | mog | byes |
| 17:04.10 | burnproof | ok :) |
| 17:04.10 | E-bola | np :) Enjoy ur dinner |
| 17:04.18 | burnproof | hehehe |
| 17:07.49 | hmmhesays | want to put my tender heart in a blender, watch it spin around to a beautiful oblivion |
| 17:08.23 | tzanger | that is an awesome song |
| 17:08.37 | coppice | bloody windows users. masochistic to the last |
| 17:08.44 | [TK]D-Fender | hmmhesays : excellent example of off-time lyrics |
| 17:08.46 | tzanger | although that guy must have a lung capacity rivaling that of turtles |
| 17:08.53 | hmmhesays | [TK]D-Fender: heh yeah |
| 17:08.56 | [TK]D-Fender | hmmhesays : performed it a few times |
| 17:09.08 | hmmhesays | eve 6 was one of the few bands to be signed while still in highschool |
| 17:09.29 | hmmhesays | [TK]D-Fender: yeah not too difficult |
| 17:10.02 | tzanger | not too difficult? When the hell do you breathe in that song? |
| 17:10.12 | *** join/#asterisk crich1999 (n=crich@port-212-202-210-134.dynamic.qsc.de) |
| 17:11.10 | Dr-Linux|work | [TK]D-Fender, i called a context, digit timeout is defined within this context, so i move the the other context using Goto(other-context), here is no timeout defined. so how it will work? |
| 17:11.27 | [TK]D-Fender | tzanger : It comes to you... not that hard really.. but it is "busy" lyrically. |
| 17:11.31 | Dr-Linux|work | [TK]D-Fender, will it consider the previous context digit timeout? |
| 17:11.44 | [TK]D-Fender | Dr-Linux|work : no idea. |
| 17:12.00 | [TK]D-Fender | ~8ball Will the new context inherit the previously defined timeout? |
| 17:12.04 | jbot | No. |
| 17:12.10 | hmmhesays | we're talking about eve 6 right? |
| 17:12.12 | hmmhesays | where do you breath? |
| 17:12.22 | [TK]D-Fender | JBOT HAS SPOKEN! |
| 17:12.33 | Dr-Linux|work | tzanger, any clue about my question? |
| 17:13.12 | tzanger | Dr-Linux|work: test it |
| 17:13.39 | [TK]D-Fender | hmmhesays : "so cal is where my mind states (breath) I drink sick like ginger-ale (breath) my stomach turns and I exhale (long -ale then breath) |
| 17:14.03 | [TK]D-Fender | Breath in on the "I"'s |
| 17:14.08 | Dr-Linux|work | tzanger, i tested, it's still considering previous context digit timeout |
| 17:14.18 | [TK]D-Fender | hmmhesays : Correct |
| 17:14.39 | Dr-Linux|work | tzanger, but i don't want that previous context time in new context |
| 17:15.14 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-189-246.lmdaca.adelphia.net) |
| 17:15.31 | Dr-Linux|work | tzanger, but i'm not sure, what can i do, .... asking for suggestions, should i define digit timeout in the new context as well? |
| 17:15.53 | tzanger | Dr-Linux|work: precisely. that's what I'd do |
| 17:15.54 | [TK]D-Fender | Dr-Linux|work : Yes. think "implicit" and you won't wonder why things don't work all the time |
| 17:15.56 | *** join/#asterisk sky1234 (n=sky1234@12.44.122.130) |
| 17:16.27 | sky1234 | Hello. Does anyone have experience with Asterisk working successfully behind Astaro firewalls?? |
| 17:16.31 | Dr-Linux|work | :S |
| 17:17.29 | Dr-Linux|work | [TK]D-Fender, asking expert suggestions is very good thing i believe |
| 17:19.26 | *** join/#asterisk Blafasel (n=bpodszun@relay3.vistream.de) |
| 17:19.41 | [TK]D-Fender | Dr-Linux|work : You are trying to find shortcuts that will betray you when you jump from a context with different rules... never assume anything if you don't have to. |
| 17:20.38 | tzanger | heh someone else wanting to bind multiple DSL interfaces |
| 17:20.44 | tzanger | and load balance |
| 17:21.47 | KDan | ok... this is what I'm getting in my sip debug... http://textpaste.net/3ebn17 ... still doesn't make any sense though... my asterisk receives an INVITE, responds with NAT details, then an ACK comes in, then for some reason there's a new INVITE, then NAT details again and ACK again... then the line is apparently connected except i can't hear anything on my phone... the call is scheduled to be destroyed, and then it's destroyed... |
| 17:22.55 | KDan | unfortunately it doesn't tell me what context it tries to call and why that fails to do anything (I suspect that's where the problem is) |
| 17:23.01 | KDan | is there a way i can find that out? |
| 17:23.22 | *** join/#asterisk gandhijee (n=gandhije@65.169.134.2) |
| 17:23.26 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 17:25.34 | *** join/#asterisk bkw_ (n=brian@asterisk/friend-and-developer/bkw) |
| 17:26.32 | Dr-Linux|work | [TK]D-Fender, yeah, but it didn't Set new digit timeout for me |
| 17:26.34 | Dr-Linux|work | <PROTECTED> |
| 17:26.35 | Dr-Linux|work | <PROTECTED> |
| 17:27.28 | KDan | is there any way to enable a debug mode that tells me why this particular call is not being picked up by any extension in the context that I've assigned to it and that it is apparently going to? |
| 17:27.34 | Dr-Linux|work | i defined 6 in next context, but it still coming with previous context digit timeout |
| 17:28.02 | *** join/#asterisk Trakkasure (n=Nfebvib@24-50-26-239.atlsfl.adelphia.net) |
| 17:30.00 | *** join/#asterisk svemuri (n=svemuri@c-24-98-122-69.hsd1.ga.comcast.net) |
| 17:30.50 | *** join/#asterisk s0lid (n=jlq@124.6.176.100) |
| 17:34.34 | *** join/#asterisk signuts (n=signuts@sig.triton.net) |
| 17:34.53 | signuts | Ok, I am having a horiffic time detecting a valid dial comand from AGI. Why in the world would Dial return -1 on success and 0 on failure!? |
| 17:35.28 | signuts | There has to be some logic ot this nature, i'm flabbergasted |
| 17:36.25 | daysmen3 | this might be a silly question but im fairly new to asterisks and wanted to know whether anyone has multidialing written into their dial plan - |
| 17:36.33 | sky1234 | For major production...running asterisk behind a firewall...what firewall do folks recommend? |
| 17:37.48 | [TK]D-Fender | sky1234 : PF or iptables :) |
| 17:37.51 | [hC] | any of you guys had problems with echo/crackling/etc on polycom phones? The few unique things about mine are that i do use it as a computer passthru (using the switch in the phone), firmware 1.6.6, and i have a headset plugged in (but not on all of em that do it) |
| 17:38.17 | sky1234 | [TK]D-Fender: When you refer to PF whats that?? |
| 17:38.35 | [TK]D-Fender | [hC] : nope. I've got my 4 CSR's using IP600's with headsets daisy-chained into each other. |
| 17:38.48 | [TK]D-Fender | sky1234 : BSD's default firewall :) |
| 17:38.55 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:38.56 | sky1234 | [TK]D-Fender: specifically any good hardware based one out there? Im pulling my hair out trying to use the Astaro.. |
| 17:39.11 | [hC] | [TK]D-Fender: hmm..... Interesting. |
| 17:39.12 | sky1234 | or is it worth going hardware?? |
| 17:40.03 | KDan | anyone got a clue what could be going wrong here: http://textpaste.net/3ebn17 ?? or does this sip debug log look fine? |
| 17:40.17 | [TK]D-Fender | sky1234 : Not IMO.. then again my * box IS my gateway |
| 17:41.46 | [TK]D-Fender | daysmen3 : Care to clarify "multi-dialing"? |
| 17:43.05 | *** join/#asterisk charles___ (n=charles@fw.invosat.com) |
| 17:43.35 | *** join/#asterisk Pazzo (n=thomas@dialin-225136.rol.raiffeisen.net) |
| 17:43.51 | h3x | sky1234: many hardware based firewalls are actually freebsd, openbsd, or linux based |
| 17:44.00 | svemuri | Is any one running 1.2.10 in production. We were running 1.2.9.1 with out a lot of trouble except for the bug where * thinks a PRI channel is in use when it is really not and that causes inbound calls to return busy. I've tried to go to 1.2.10 see if that bug is fixed. But user complained that calls longer than 4-5min are being dropped randomly and on top that the old bug is still there. |
| 17:44.08 | Dr-Linux|work | it never detects never context digit timeout :S |
| 17:44.40 | *** join/#asterisk nDuff (n=ccd@64.128.31.220) |
| 17:45.34 | nDuff | Can I disable musiconhold for a queue, such that folks in-queue hear ringing? |
| 17:45.44 | nDuff | (It's a very low-wait queue) |
| 17:45.50 | [TK]D-Fender | nDuff : Yes |
| 17:46.07 | Alric | Queue option r I believe. |
| 17:46.27 | nDuff | ahh. Thanks! |
| 17:46.29 | Alric | "show application queue" from the CLI just to be sure. |
| 17:47.48 | *** join/#asterisk mountainm2k (n=mountain@cbit-98.bullseye9.com) |
| 17:48.01 | sp0n9e | i'm getting a notice "musiconhold.c:511 monmp3thread: Request type schedule in the past?!?!" |
| 17:48.16 | sp0n9e | is this why i'm not hearing music on hold? |
| 17:48.19 | KDan | anyone got a clue what could be going wrong here: http://textpaste.net/3ebn17 ?? or does this sip debug log look fine and the problem is somewhere else? |
| 17:48.53 | [TK]D-Fender | KDan : without the SIP debug what is the problem? |
| 17:49.21 | KDan | when i dial in with my phone, nothing happens - the line dies after about ten seconds |
| 17:49.37 | KDan | no dialplan is ever executed, even though sip debug clearly shows that something is going on somewhere |
| 17:50.14 | KDan | http://textpaste.net/jbpvq7 << dialplan |
| 17:50.18 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 17:50.45 | KDan | i first tried using extension s, then using the number that i'm dialing on my phone... then using i as well... |
| 17:50.55 | mountainm2k | [TK]D-Fender: Got a sec to coach me through adding a ring-tone? Or is there a simple doc for it? |
| 17:51.00 | KDan | and even on timeout there's supposed to be something happening - but nothing ever happens |
| 17:51.47 | [TK]D-Fender | KDan : What kind of phone, and what # are you dialing? |
| 17:52.22 | [TK]D-Fender | KDan : An describe the networking between you * box and the point of origin of the call. |
| 17:52.32 | KDan | normal PSTN landline phone, dialing 08444847235, which is a number provided by esms.com, who provide free numbers that you can then point to sip addresses |
| 17:52.49 | [TK]D-Fender | KDan : connected how? |
| 17:52.57 | KDan | my asterisk box is on my LAN, which is behidn a router. port 5060 is forwarded to the * box |
| 17:53.28 | KDan | i gave, to esms.com, the address S:sipstream@84.x.y.z (where the latter part is the ip of my router) |
| 17:53.31 | [TK]D-Fender | KDan : How is this phone connected? |
| 17:53.42 | *** join/#asterisk SwK (n=Silik0nJ@65.169.134.2) |
| 17:53.54 | KDan | through a normal phone socket - it goes through an adsl filter too |
| 17:53.55 | [TK]D-Fender | KDan : So esms sends a SIP call to * basically? |
| 17:53.59 | KDan | yes |
| 17:54.23 | [TK]D-Fender | KDan : I'm confued as to how this analog phone is connected to *.... |
| 17:54.26 | KDan | and * clearly receives the call, but then i can't get it to do anything with it other than kill it after about 10 seconds without having played any sounds |
| 17:54.43 | KDan | the phone is a normal phone. I dial a phone number on it. the phone number is one owned by esms.com |
| 17:54.56 | KDan | esms.com convert this to a bunch of SIP packets |
| 17:55.02 | [TK]D-Fender | KDan : Then the phone has NOTHING to do with *... its calling somewhere ELSE then? |
| 17:55.03 | KDan | and forward this to the address i gave them |
| 17:55.07 | mountainm2k | I think he's saying the phone is _not_ connected to do with * |
| 17:55.35 | [TK]D-Fender | mountainm2k : And driving me crazy with the irrelevent bits if so... |
| 17:55.46 | KDan | i don't know yet which bits might be relevant |
| 17:55.51 | daysmen3 | [TK]D-Fender: "hunt groups" i think thats the right wording - what i wanted to know is if its possible to include in dial plan with call forwarding since the numbers would have to be recursively looked at for CF enabled |
| 17:55.52 | KDan | sorry if i'm giving tmi |
| 17:55.56 | mountainm2k | sounds like he's calling from a normal PSTN phone to his DID provider, and wishing his * box would answer... |
| 17:56.02 | KDan | yes |
| 17:56.04 | KDan | that is right |
| 17:56.28 | [TK]D-Fender | KDan : keep the irrelevent stuff out. you should also forward 10000-20000 to your router for RTP and make sure your SIP.conf is properly set up to forge your headers. |
| 17:56.56 | mountainm2k | Or use IAX insted... :-P |
| 17:57.01 | KDan | sip.conf contains the nat=yes and the externip settings |
| 17:57.19 | KDan | i need to forward 10000-20000 too? |
| 17:57.24 | Astinus- | what does it mean to buy call termination minutes? |
| 17:57.26 | [TK]D-Fender | yes |
| 17:57.29 | charles___ | Hey, did anyone have had problems terminating calls to 1800 numbers ? |
| 17:57.48 | KDan | can they be forwarded to a single port since i only ever expect one call at a time? |
| 17:57.48 | mountainm2k | KDan: those ports need to forward from your outside IP _TO_ the * box |
| 17:57.57 | KDan | ok, on the router then |
| 17:58.02 | charles___ | I'm having weird problems over my provider (vonage) can call local and long distance but some 1800 doesn't work |
| 17:58.02 | mountainm2k | KDan You need to forward the entire range |
| 17:58.16 | mountainm2k | any connection to your outside IP on those ports must come through to * |
| 17:58.26 | [TK]D-Fender | daysmen3 : You mean for incoming calls? So that if someone phones you on your primary line and its busy it'll ring on the next, and so on? |
| 17:58.33 | Qwell[] | charles___: Are you/they setting your CID number to a tollfree DID? |
| 17:58.47 | Qwell[] | charles___: because, I've seen tollfree providers straight up reject a call when I do that |
| 17:58.50 | robl^ | charles___: that's a Vonage issue. they have something not set up correctly on their side. |
| 17:59.02 | charles___ | Qwell: yes, I've get caller id when I call my cell |
| 17:59.14 | Qwell[] | charles___: yes, but is the callerid that of a tollfree DID? |
| 17:59.20 | charles___ | robl^: are you having the same issue ? |
| 17:59.37 | Qwell[] | I had to put a little part in my dialplan, where when I call tollfree numbers, I set it to a "valid" non-tollfree DID |
| 17:59.55 | Qwell[] | (literally, the Simpsons phone number...which is in the Philippines) |
| 18:00.00 | robl^ | charles___: no. I don't us Vonage. If some calls gor throubh, but not others.. it's 99% likely it is with the provider |
| 18:00.04 | charles___ | Qwell: all my calls go out with a 954 caller id |
| 18:00.11 | Qwell[] | charles___: different issue then :) |
| 18:00.15 | Qwell[] | call Vonage |
| 18:00.23 | charles___ | Qwell: I did |
| 18:00.26 | Qwell[] | and? |
| 18:00.38 | KDan | mountainm2k & [TK]D-Fender: unfortunately my router only allows me to forward port ranges to a single mapped port at a time... apart from typing 10'000 port forward rules, does that mean I can't get SIP to connect successfully? |
| 18:00.57 | KDan | i.e. i can do 10k-20k -> 10k, but not 10k-20k -> 10k-20k |
| 18:01.15 | Dr-Linux|work | Qwell[], do you have any idea, how can i get new digit timeout after Goto(new-context) ? |
| 18:01.20 | charles___ | Qwell: but the problem is, when I call from my BudgeTone-100 from the same vonage Account, it does get complete. |
| 18:01.23 | KDan | or is there a way i can tell asterisk to use only a few specific ports? |
| 18:01.25 | Qwell[] | Dr-Linux|work: dunno |
| 18:01.31 | robl^ | KDan: save yourself a hassle and buy a new $50 router |
| 18:01.34 | Qwell[] | charles___: okay, that changes things a bit, eh? |
| 18:01.40 | charles___ | Qwell: exactly |
| 18:01.42 | daysmen3 | [TK]D-Fender: yea thats right - also would apply to calls called internally |
| 18:01.53 | Qwell[] | so...hmm |
| 18:02.05 | Qwell[] | charles___: That doesn't make a whole lot of sense though. Asterisk doesn't care |
| 18:02.09 | charles___ | Qwell: I believe that some calls to some 1800 go to different gateway inside vonage, and that gateway is using a different SIP version that my asterisk can't handle |
| 18:02.10 | Qwell[] | is it ALL 800 numbers? |
| 18:02.22 | charles___ | but running Asterisk 1.2.9.1 |
| 18:02.31 | hmmhesays | i know who I want to take me hoe |
| 18:02.33 | hmmhesays | *home |
| 18:02.37 | charles___ | Qwell: yes it is. |
| 18:02.40 | hmmhesays | I know who I want to take me home |
| 18:02.43 | KDan | robl^: router cost about 60 pounds from PC World actually, a month or so ago :-) |
| 18:03.24 | Qwell[] | charles___: check your dialplan in * |
| 18:03.28 | robl^ | KDan: you really don't want to set up rules for each and every port 1 by 1. |
| 18:03.38 | charles___ | Qwell: all calls going thru the same gateway |
| 18:03.45 | Qwell[] | make sure |
| 18:03.46 | charles___ | Qwell: I also tcpdumped it |
| 18:03.52 | KDan | robl^: that's for sure! :-) |
| 18:04.10 | KDan | guess i'll use AIX for my testing isntead then *sigh* |
| 18:04.21 | KDan | sorry IAX |
| 18:04.34 | daysmen3 | ive rewritten (almost finished) dialparties.pl to asterisk native extensions.conf language |
| 18:04.41 | robl^ | KDan: I use a US $50 Netgrear router.. works like a charm |
| 18:04.44 | *** join/#asterisk alexrch (n=alex_rch@cpe-212-18-59-51.dynamic.amis.net) |
| 18:05.14 | alexrch | hi guys, I have a question regarding queues, is anyone here who has any experiences with Queues and Asterisk? |
| 18:05.15 | KDan | robl^: tip: don't buy D-link :-) |
| 18:05.23 | *** join/#asterisk Waverly360 (n=9893acdf@65.169.134.2) |
| 18:05.37 | robl^ | Dlink = EVIL |
| 18:05.44 | h3x | d-unlinked |
| 18:05.54 | h3x | works almost as good as an airgap(tm) firewall |
| 18:06.07 | Waverly360 | hola |
| 18:06.34 | Astinus- | d-stinks |
| 18:06.41 | alexrch | ....anyone ready to answer a newby question :) ??? |
| 18:06.48 | Astinus- | alexrch: shoot |
| 18:06.56 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 18:06.58 | robl^ | for small / cheap installs, I always use Netgear.. sometimes Linksys. For larger, I'd suggest maybe Cisco routers.. or a dedicated Linux router |
| 18:07.11 | daysmen3 | [TK]D-Fender: theirs no array functionality that allows me to recursively check each number so i wanted to know whether its something i could leave out (for now). Wanted to knwo how important it was as a selling option? |
| 18:07.53 | paolob | Hi guys! I have a fax connected to asterisk via a linksys pap2, but when I receive a fax call and I pass it to the fax, it can't receive it. Anyone can tell me why? thank you! |
| 18:08.21 | alexrch | okie dokie: I'm trying to use the dynamic queue agents, using AddQueueMember, but by adding a new dynamic member (for example: AddQueueMember(testQ,Local/123@context) ) I get queue member of "unknown" state...meaning that calls are never distributed to him/her |
| 18:08.22 | KDan | ok, what i've actually done is change the port range in rtp.conf to 10000-10002, and forwarded those ports by hand |
| 18:08.40 | KDan | that has, however, made no difference :-( |
| 18:08.48 | charles___ | Qwell: always sending the call regularly to Vonage |
| 18:09.06 | alexrch | how can I use AddQueueMember with Local channel, so that queue member would not be in "unknown" state, but in "not in use" (alias ready to accept calls) |
| 18:09.07 | alexrch | ? |
| 18:09.08 | charles___ | Qwell: invite 18008828880@sphone.vopr.vonage.net |
| 18:09.31 | charles___ | Qwell: I have a answer from Vonage : TRYING |
| 18:10.02 | charles___ | Qwell: then I start receiving RTP's from another IP , I believe those RTP's are the RINGING |
| 18:10.04 | KDan | so the problem is not with the port forwarding now... what else could cause the symptoms I've been seeing? |
| 18:10.22 | nortex | paolob, Voip and faxing tend to be very problematic. What codec are you using on the sip end |
| 18:10.37 | KDan | i.e.: sip call comes in from DID provider, asterisk does some sip stuff, call dies - nothing else happens |
| 18:10.51 | alexrch | ....hmm, was my question clear? :) |
| 18:11.36 | paolob | nortex, do you mean on the sipura where the fax is connected? g711u |
| 18:12.10 | nortex | okay and then you have ulaw in sip.conf for that sipura? |
| 18:12.16 | hmmhesays | anyone in here using junction networks? |
| 18:13.16 | h3x | i have |
| 18:13.20 | nDuff | (Might try playing with a local iaxfax+hylafax setup at some point) |
| 18:13.24 | h3x | i had 600 lines of it working here |
| 18:13.28 | h3x | over g711 |
| 18:13.29 | h3x | heh |
| 18:13.41 | nDuff | h3x: My LAN's never been particularly reliable -- I could believe it as a jitter issue... |
| 18:13.43 | [TK]D-Fender | daysmen3 : Sorry, that just made no sense at all to me.. try again... |
| 18:13.44 | alexrch | did anyone here ever used AddQueueMember with local agents? |
| 18:14.09 | h3x | Jitterbuffers are broken in asterisk |
| 18:14.10 | nDuff | h3x: ...but we lost a *lot* of faxes 'till we gave up and put in a channel bank. |
| 18:14.14 | h3x | turn them off |
| 18:14.36 | nDuff | h3x: Did. I'm talking about in the network (due to collisions and assorted weirdness). |
| 18:14.37 | [TK]D-Fender | daysmen3 : Lets see if I can guess this out : You want to "just pick any free line going out", and have calls to the primary number just take up any free incoming line (on analog)? |
| 18:14.53 | h3x | perhaps you should check your duplex / media type settings |
| 18:15.13 | h3x | i screwed that one up on my catalyst a couple months ago |
| 18:15.31 | *** join/#asterisk SkramX (n=MarkS@70.86.176.56) |
| 18:15.35 | SkramX | Aug 1 13:14:48 NOTICE[15855]: channel.c:2437 __ast_request_and_dial: Don't know what to do with control frame 15 |
| 18:15.40 | SkramX | what exactly does that mean? |
| 18:16.45 | charles___ | Qwell: going to get a sip debug on it |
| 18:16.58 | hmmhesays | i need a new decent itsp for business purposes |
| 18:17.12 | charles___ | Qwell: with tcpdump I can't differenciate |
| 18:17.26 | paolob | nortex, I have disallow=all, allow=gsm, allow=ulaw, allow=alaw, allow=g729 |
| 18:17.43 | ComputerWarm | can anyone please answer my question about C API scripts... do you compile them or leave them open? |
| 18:17.49 | nortex | set it to disallow=all and allow=ulaw only. |
| 18:17.55 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.18) |
| 18:18.15 | paolob | nortex, let me see |
| 18:21.06 | svemuri | Any one running 1.2.10 with out problems? |
| 18:21.10 | *** join/#asterisk gursikh (n=guriskh1@dsl254-123-245.nyc1.dsl.speakeasy.net) |
| 18:21.15 | nortex | Yup |
| 18:21.20 | *** join/#asterisk doughecka (n=Tad@unaffiliated/doughecka) |
| 18:21.22 | hmmhesays | yes yes it can |
| 18:21.32 | hmmhesays | I'm not having any issues |
| 18:21.36 | doughecka | Any use the new 8.3 firmware on the cisco phones? |
| 18:21.47 | nortex | But I do not use a lot of Queues |
| 18:21.53 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 18:22.23 | svemuri | Just upgraded to it this morning (about 60 users) and immediately got complaints about outgoing calls that are more than 4 mins being randomly dropped |
| 18:22.35 | paolob | nortex, I set disallow=all and allow=ulaw in sip.conf for the fax, i reloaded asterisk, but lamentably it doesn't work yet |
| 18:23.08 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 18:23.25 | svemuri | I've experienced that myself. No CDR when that happens either. Had to roll back to 1.2.9.1 |
| 18:24.00 | *** join/#asterisk bkw_ (n=brian@asterisk/friend-and-developer/bkw) |
| 18:24.24 | nortex | paolob, You might do a search on wiki to find out how to turn the jitterbuffer off and see if that helps. |
| 18:24.39 | Un1x | anyuone know how much inflation effects the economy like how much things rise up in price, etc |
| 18:24.51 | paolob | nortex, wait, the voip trasmission between the two fax is made throug iax, and it uses gsm! |
| 18:25.00 | jbalcomb | Un1x: yes, somewhat. |
| 18:25.47 | jbalcomb | paolob: have you tried an analog phone on that ata just to make sure it works well enough? |
| 18:26.03 | nortex | paolob, I thought you said the fax was on a sipura. |
| 18:26.16 | paolob | jbalcomb, it works quite well |
| 18:26.39 | jbalcomb | paolob: does your fax pick up the call? |
| 18:26.41 | paolob | nortex, I have two fax connected each one with sipura to a asterisk server, |
| 18:26.50 | paolob | jbalcomb, yes, it pick up |
| 18:27.23 | paolob | nortex, on the receiving one I get !rtp.c:564 ast_rtp_read: Unknown RTP codec 100 received! |
| 18:28.10 | nortex | paolob, So is the IAX between the servers? |
| 18:28.16 | jbalcomb | paolob: if it works fine with the phone you shouldn't need to be monkeying with codecs. |
| 18:28.32 | paolob | nortex, yes, and iax uses gsm |
| 18:29.11 | watchy | what would cause a transfered call to get droped but a blindxfr to work? |
| 18:29.52 | paolob | now I'm trying to use ulaw with iax |
| 18:29.53 | nortex | paolob, Ok, the compression of the audio could be the problem with the fax. |
| 18:30.39 | watchy | -- Incoming call: Got SIP response 500 "Internal Server Error" back from 192.168.0.206 |
| 18:30.42 | watchy | what would cause that |
| 18:31.16 | paolob | nortex, with ulaw it works! thank you!!! |
| 18:31.30 | paolob | jbalcomb, thank you very much for your attention! |
| 18:31.36 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 18:32.41 | jbalcomb | nortex: nice work. :) have you heard anything about alaw working better for faxing than ulaw? Any credit to the idea of it? |
| 18:32.45 | *** part/#asterisk LenOK (n=ln@office-181.telengy.net) |
| 18:33.44 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 18:37.02 | KDan | Oh well... switching to IAX instead... thanks for your help everyone who helped though |
| 18:37.40 | nortex | jbalcomb, I just remeber that the fax tones could not be compressed and ulaw/alaw are both the codecs used in T1/E1 circuits. Things like jitterbuffers and echo cancelation can also alter the tones/frequency and break faxing. Which is why T.38 is so exciting, if it works. :) |
| 18:39.07 | jbalcomb | nortex: yeah, i keep hearing about T.38 but haven't looked into it. It'd be nice cause I have faxes ib an analog PBX I'd like to clobber. |
| 18:39.19 | jbalcomb | s/ib/on |
| 18:39.58 | jbalcomb | nortex: is there any way to have jitterbuffers and EC off for certain calls? |
| 18:40.26 | nortex | jbalcomb, Personally my fax modems and machines on a channel bank connected to a Quad t-1 card with the PRI. Someday I would like to fax over VoIP to remote locations though. |
| 18:41.21 | *** join/#asterisk xnon (i=xnon@200.8.4.227) |
| 18:41.25 | xnon | hello |
| 18:41.37 | Seba_soy | I am receiving faxos on my Fax phone connected to asterisk with 711, then this asterisk is connected to another asterisk witj 711 and that second asterisk have a zaptel 1E1 |
| 18:41.40 | Seba_soy | all works ok |
| 18:41.41 | xnon | i have a problem with a sip extensions and voicemail |
| 18:41.47 | Seba_soy | I can receive fax very good |
| 18:41.47 | jbalcomb | nortex: pretty much the same setup. i'm thinking i could move the fax lines to another PRI and run it off a single PRI card so i can set up zaptel and zapata the way faxes prefer. |
| 18:41.54 | xnon | Seba_soy do u speak spanish? |
| 18:41.59 | Seba_soy | eys |
| 18:42.00 | Seba_soy | yes |
| 18:42.04 | Seba_soy | i am from argentina :) |
| 18:42.07 | xnon | como estas |
| 18:42.11 | Seba_soy | bien |
| 18:42.12 | xnon | soy de Venezuela |
| 18:42.14 | Seba_soy | q tal |
| 18:42.18 | xnon | incursionando en el mundo del asterisk |
| 18:42.28 | xnon | tengo un problema y quizas sabras algo al respecto |
| 18:42.32 | xnon | tienes un par de minutos? |
| 18:42.39 | Seba_soy | si, por privado |
| 18:42.46 | Seba_soy | dale? |
| 18:42.51 | Seba_soy | aca hablan ingles |
| 18:43.11 | jbalcomb | Seba_soy: how many faxes do you get a day? |
| 18:43.11 | nortex | jbalcomb, The jitterbuffer might be peer device and the EC can be turned off per channel if I remeber right. |
| 18:43.30 | nortex | jbalcomb, EC per Zap channel that is. |
| 18:44.07 | jbalcomb | nortex: i think i'm creeping towards my next big asterisk project. (my boss fears the light in my eyes) |
| 18:44.12 | *** join/#asterisk NewSole (n=dave@d38-53-48.commercial1.cgocable.net) |
| 18:44.50 | jbalcomb | cacajuates? |
| 18:45.49 | *** join/#asterisk oadaeh (n=jason@las-static-208.57.199.83.mpowercom.net) |
| 18:46.06 | watchy | - Incoming call: Got SIP response 500 "Internal Server Error" back from 192.168.2.105 |
| 18:46.13 | watchy | im getting that a hell of alot of times |
| 18:46.18 | watchy | anyone know what it is |
| 18:46.33 | nortex | watchy, Do you have Polycoms and hints configured? |
| 18:46.48 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 18:46.53 | watchy | nortex: yes |
| 18:46.59 | watchy | do i need to have the phones reboot? |
| 18:47.11 | jbalcomb | watchy: 24/7 |
| 18:47.28 | watchy | well i wasnt getting this till i restarted * like 50 times trying to fix echo |
| 18:47.57 | nortex | watchy, It is not really a problem, but likely the Polycoms are not responding to buddy list watches and nightly rebooting the phones will fix it. |
| 18:48.12 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.18) |
| 18:48.19 | *** join/#asterisk zamsler (n=zamsler@65.169.134.2) |
| 18:48.58 | nortex | watchy, My *opinion* is that the hint traffic is more fequent then the phone likes it to be and it quits responding. |
| 18:48.59 | xnon | Seba_soy in a private is all |
| 18:49.41 | *** join/#asterisk iq (n=iq@unaffiliated/iq) |
| 18:51.08 | watchy | nortex: apparently the phones are having problems with transfers |
| 18:51.43 | E-bola | How do you go about chosing a prefered codec to use on ur phones? |
| 18:51.58 | jbalcomb | E-bola: google |
| 18:52.02 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:52.11 | nortex | watchy, Transfering with the soft keys or the pbxtransfer feature |
| 18:52.15 | watchy | whats the default l/p for polys web interface? |
| 18:52.15 | E-bola | jbalcomb: isnt there simply one codec ppl chose? |
| 18:52.16 | jbalcomb | E-bola: LAN or WAN? |
| 18:52.22 | E-bola | i dont feel like listening to tons of samples |
| 18:52.23 | watchy | softkeys nortex |
| 18:52.29 | *** join/#asterisk Trionnis (i=lordkuri@12.206.2.116) |
| 18:52.33 | E-bola | jbalcomb: well ile be calling both wan and lan if thats what u mean? |
| 18:52.39 | E-bola | the asterisk server is on the lan though |
| 18:52.40 | nortex | Polycom/456 |
| 18:52.44 | nortex | I think |
| 18:52.52 | Trionnis | would anyone be able to point me towards an explanation for this: |
| 18:52.52 | Trionnis | <PROTECTED> |
| 18:52.53 | Trionnis | <PROTECTED> |
| 18:52.59 | jbalcomb | E-bola: LAN + North America = uLaw; LAN + Europe = aLaw; |
| 18:53.01 | Trionnis | over and over and over in my console? |
| 18:53.08 | watchy | thats it thanks |
| 18:53.13 | watchy | im gonna reboot it |
| 18:53.26 | *** join/#asterisk bpiper (n=bpiper@70.159.49.40) |
| 18:53.32 | E-bola | jbalcom: arent those rather uncompressed? |
| 18:53.33 | jbalcomb | Trionnis: http://www.google.com/search?hl=en&q=Remote+UNIX+connection+disconnected&btnG=Google+Search |
| 18:53.50 | jbalcomb | E-bola: WAN = GSM ($10.00 USD) |
| 18:54.17 | jbalcomb | E-bola: past that you can track down charts, check with providers, and pick your own. |
| 18:54.25 | nortex | watchy, You can also do a sip notify polycom-check-cfg "devicename" from the CLI |
| 18:54.30 | Trionnis | I found that a few days ago |
| 18:54.33 | Trionnis | isn't very clear |
| 18:54.58 | Trionnis | I have nothing in crontab related to * |
| 18:54.58 | jbalcomb | Trionnis: you found google a few days ago? |
| 18:54.58 | E-bola | if my asterisk server have to transcode or my providers asterisk server has to transcode |
| 18:55.00 | Trionnis | ... |
| 18:55.02 | E-bola | doesnt that introduce delay? |
| 18:55.02 | *** part/#asterisk smackus (n=ckwall@63.149.122.93) |
| 18:55.15 | Trionnis | I found that particular result when I searched google a few days ago for the same issue |
| 18:55.17 | Trionnis | that better? |
| 18:55.18 | nortex | lol |
| 18:55.19 | Trionnis | :) |
| 18:55.20 | watchy | whats that do nortex? makes the polys reload conf? |
| 18:55.44 | jbalcomb | Trionnis: there are "Results 1 - 10 of about 1,040,000 for Remote UNIX connection disconnected." |
| 18:56.10 | Trionnis | ok, I get the point... does anyone else know of the issue besides smartass over there? |
| 18:56.31 | jbalcomb | Trionnis: you might check more than just that one. I think .0000001 amount of effort is a bit weak. |
| 18:56.34 | nortex | waba, it sends a sip packet to the phone telling it to check its config. if the provisioning file has changed the phone will reboot. There is a option in the polycom sip.cfg to always reboot. Very handy in scheduled reboots. |
| 18:56.36 | Trionnis | right |
| 18:56.37 | *** join/#asterisk tempest1 (n=asf@adsl-153-43-12.chs.bellsouth.net) |
| 18:56.46 | *** join/#asterisk SwK (n=Silik0nJ@65.169.134.2) |
| 18:56.54 | Trionnis | 'cause you know for a fact that I didn't spend about 4 hours digging through those results, right? |
| 18:57.04 | *** join/#asterisk cybertrickle_ (n=cybertri@wsip-70-167-111-3.ph.ph.cox.net) |
| 18:57.08 | Trionnis | I'm just some st00pid n00b that doesn't google things before asking, right? |
| 18:57.26 | watchy | ah so if it checks its conf it'll just reboot? |
| 18:57.39 | bpiper | Trionnis: I just joined, what's your question about the Unix connection disconnected? |
| 18:57.40 | Trionnis | here's a hint: I don't come in here and ask about something until I've spent at least a few hours looking for the solution |
| 18:57.54 | cybertrickle_ | I am trying to originate a call via the asterisk api. It is setting my context to default even though I tell it to set it to something else in the command. Any ideas ? |
| 18:58.08 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 18:58.12 | Trionnis | bpiper: I'm getting that repeatedly in my console, and searching hasn't come up with much along the lines of a reason |
| 18:58.21 | cybertrickle_ | ast_freak, fancy seeing you here |
| 18:58.25 | Trionnis | just wondering if anyone else has seen it and can point me in a different direction |
| 18:58.26 | *** join/#asterisk derekS (n=dereks@unaffiliated/dereks) |
| 18:58.33 | jbalcomb | haha.. i just couldn't see how you were relating it to Asterisk is all. |
| 18:58.44 | *** join/#asterisk bkw_ (n=brian@asterisk/friend-and-developer/bkw) |
| 18:58.52 | bpiper | Trionnis: are you running a GUI? Sounds like the manager API maybe showing that |
| 18:58.57 | Trionnis | nope, no gui |
| 18:59.02 | watchy | anyone know what the new polycom firmware of 1.6.7 does? |
| 18:59.10 | Trionnis | it just started randomly about a week ago |
| 18:59.11 | derekS | hi. I am looking for a preconfigured vmware image (or an iso) that includes a working copy of asterisk (so no/minimal install/config needed if i want to use a software client) |
| 18:59.40 | Trionnis | only thing I have remotely close to a gui is the web cdr scripts, but those aren't being accessed |
| 19:00.12 | nortex | Trionnis, And no cron jobs you said right. |
| 19:00.15 | NetIQSystems | Trionnis, what do you have??? |
| 19:00.16 | Trionnis | correct |
| 19:00.26 | Trionnis | net: this: |
| 19:00.27 | NetIQSystems | are you using trixbox? |
| 19:00.31 | Trionnis | nope |
| 19:00.38 | NetIQSystems | FOP? |
| 19:00.41 | NetIQSystems | FREEPBX? |
| 19:00.42 | Trionnis | <PROTECTED> |
| 19:00.43 | Trionnis | <PROTECTED> |
| 19:00.54 | Trionnis | sl-host01*CLI> show version |
| 19:00.54 | Trionnis | Asterisk 1.2.9.1 built by root @ sl-host01.firestormnetworks.net on a i686 running Linux on 2006-07-05 04:47:03 UTC |
| 19:01.02 | Trionnis | cvs build |
| 19:01.09 | *** join/#asterisk wunderkin (n=wunderki@216-19-202-13.getnet.net) |
| 19:01.09 | hmmhesays | thats odd my fax came out sideways |
| 19:01.14 | NetIQSystems | well.. |
| 19:01.24 | nortex | hmmhesays, Special just for you :) |
| 19:01.26 | NetIQSystems | it means that something is connecting to your asterisk. |
| 19:01.30 | *** part/#asterisk xnon (i=xnon@200.8.4.227) |
| 19:01.41 | Trionnis | I've determined that much, I'm just trying to figure out what it could be :) |
| 19:01.48 | jbalcomb | NetIQSystems: asterisk -r amounts to something connection to your asterisk |
| 19:01.49 | hmmhesays | can someone send me a fax please |
| 19:01.56 | NetIQSystems | yes. |
| 19:02.12 | hmmhesays | as in yes you'll send me a fax? |
| 19:02.15 | Trionnis | yes, but it wouldn't show a connect/disconnect cycle |
| 19:02.20 | Trionnis | over and over and over |
| 19:02.25 | NetIQSystems | say yes it would. |
| 19:02.31 | jbalcomb | Trionnis: You did try turning off the the manager? |
| 19:02.42 | NetIQSystems | it really sounds like you are usinf freepbx.. |
| 19:02.49 | Trionnis | I'm not, trust me |
| 19:02.57 | Trionnis | I'm not an asterisk noob ;) |
| 19:02.59 | NetIQSystems | any config managers? |
| 19:03.01 | Trionnis | nope |
| 19:03.16 | Trionnis | it's just a raw CLI cvs build |
| 19:03.21 | NetIQSystems | well then... |
| 19:03.22 | hmmhesays | [TK]D-Fender: you around? |
| 19:03.24 | NetIQSystems | do... |
| 19:03.26 | NetIQSystems | set verbose 0 |
| 19:03.32 | NetIQSystems | and forget it. |
| 19:03.34 | Trionnis | haha |
| 19:03.36 | NetIQSystems | ;) |
| 19:03.40 | *** join/#asterisk SwK (n=Silik0nJ@65.169.134.2) |
| 19:03.42 | [TK]D-Fender | hmmhesays : Yes, good timing |
| 19:03.42 | ccherrett | I would like to have a machine send a prerecorded message to a person on a regular phone line if a certain event occurs. What equipment do I need to make this work. Can it be done with a regular voice modem? |
| 19:03.51 | [TK]D-Fender | Just got SIP1.6.7 from my Polycom vendor |
| 19:03.54 | hmmhesays | [TK]D-Fender: can you send me a test fax? |
| 19:03.54 | Trionnis | I'm one that prefers to treat the cause, not the symptom ;) |
| 19:04.07 | NetIQSystems | then you have a problem. |
| 19:04.12 | Trionnis | it would seem so :) |
| 19:04.18 | NetIQSystems | because we have been over everything it might. |
| 19:04.20 | NetIQSystems | be. |
| 19:04.25 | E-bola | ccherrett: yes |
| 19:04.27 | NetIQSystems | so you are actually a noob... |
| 19:04.28 | nortex | Trionnis, Anyone esle have access to the system? |
| 19:04.34 | NetIQSystems | or you can't build asterisk. |
| 19:04.36 | NetIQSystems | ;) |
| 19:04.47 | watchy | tk: is it worth upgrading? |
| 19:04.49 | jbalcomb | ccherrett: you can do that with asterisk if it has access to a phone line via FXO (<-right?) or SIP provider |
| 19:04.55 | charles___ | Strange, I can call 1800 PETMEDS but can't call 1 800 8828880 |
| 19:05.04 | charles___ | Over the same termination |
| 19:05.08 | nortex | jbalcomb, right on the FXO |
| 19:05.19 | Trionnis | nortex: no |
| 19:05.20 | charles___ | But with my budgetone I can call both |
| 19:05.32 | Trionnis | NetIQSystems: haha :P |
| 19:05.40 | jbalcomb | nortex: excellent. only took me six months to stop mixing them up. |
| 19:05.49 | NetIQSystems | I have 60 installs running. |
| 19:05.51 | ccherrett | E-bola: Yes it can be done with a modem? |
| 19:05.53 | nortex | Trionnis, Well I'm out of ideas :) |
| 19:05.54 | NetIQSystems | ther is always a reason.. |
| 19:05.56 | E-bola | ccherrett: yes |
| 19:06.04 | Trionnis | ok, thanks for at least thinking about it :) |
| 19:06.15 | NetIQSystems | Trionnis, what OS? |
| 19:06.16 | ccherrett | E-bola: vgetty? |
| 19:06.27 | E-bola | no idea how its done |
| 19:06.30 | Trionnis | RHEL 4 |
| 19:06.33 | E-bola | but its rather simply os ofcourse it can be done |
| 19:06.36 | NetIQSystems | that explains it all. |
| 19:06.37 | TheCompWiz | can someone help me diagnose what's wrong with my voice mail? |
| 19:06.38 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.60) |
| 19:06.45 | Trionnis | hmm |
| 19:06.53 | Trionnis | unless you're being facetious, I'm not sure what you mean |
| 19:07.02 | NetIQSystems | redhat sux |
| 19:07.05 | NetIQSystems | ;) |
| 19:07.09 | nortex | Here comes a distro rant |
| 19:07.11 | Trionnis | I didn't have this issue until I installed it back on 07/05 |
| 19:07.12 | NetIQSystems | lol |
| 19:07.17 | Trionnis | yeah, no kidding |
| 19:07.19 | Trionnis | haha |
| 19:07.20 | NetIQSystems | then quit playing with it.. |
| 19:07.24 | TheCompWiz | anyone? |
| 19:07.32 | nortex | TheCompWiz, shoot |
| 19:07.32 | Trionnis | oh, 1.2.10 is out |
| 19:07.34 | bpiper | NetIQ: although I agree, that definately isn't the root problem |
| 19:07.37 | Trionnis | maybe I'll upgrade to that too |
| 19:07.42 | NetIQSystems | lol yeah |
| 19:07.43 | Trionnis | add more problems ;) |
| 19:07.54 | NetIQSystems | REDHAT != LINUX |
| 19:08.04 | bpiper | Troinnis: sounds like you have a Rogue php or perl script that is being accessed |
| 19:08.08 | hmmhesays | heh |
| 19:08.19 | Trionnis | zomg u sux0r! r3dh4t r0x!!11 |
| 19:08.22 | Trionnis | ;) |
| 19:08.29 | bpiper | wtf? |
| 19:08.32 | Trionnis | bpiper: I'll look into that |
| 19:08.39 | Trionnis | mocking NetIQSystems, just ignore it |
| 19:08.40 | NetIQSystems | hmm anyone from cluecon here? |
| 19:08.41 | Trionnis | ;) |
| 19:09.04 | TheCompWiz | um.... help? |
| 19:09.12 | bpiper | Get CentOS & forget RH :-) |
| 19:09.18 | E-bola | How can i make my spa922 transmit silence? |
| 19:09.21 | E-bola | cant find an option for it |
| 19:09.24 | NetIQSystems | TheCompWiz, you need professinoal help. |
| 19:09.44 | NetIQSystems | I mean.. how can we help someone who claims to be TheCompWiz |
| 19:09.46 | NetIQSystems | ? |
| 19:09.48 | Trionnis | I have centos on another box |
| 19:09.52 | nortex | TheCompWiz, You got a patsebin or more details somehwere. |
| 19:09.53 | TheCompWiz | :P |
| 19:09.57 | Trionnis | but this one only came with RHEL or Win2k3 |
| 19:10.00 | Trionnis | which would you choose? |
| 19:10.02 | Trionnis | =) |
| 19:10.06 | [TK]D-Fender | bpiper : .... CentOS = RHEL! |
| 19:10.08 | NetIQSystems | Trionnis, debian. |
| 19:10.20 | NetIQSystems | I would find a host provider that installed what I wanted. ;) |
| 19:10.22 | nortex | RHEL the format it :) |
| 19:10.23 | bpiper | Duhh |
| 19:10.26 | Trionnis | haha |
| 19:10.42 | watchy | hey nortex |
| 19:10.43 | watchy | <PROTECTED> |
| 19:10.44 | Trionnis | well this is a cpanel box also, so debian is out of the question unfortunately |
| 19:10.49 | watchy | is that what i want to set 1 on? |
| 19:10.50 | NetIQSystems | paying $500/mo for a server and I can't pick the OS.. BS>>.... |
| 19:10.53 | nortex | watchy, bingo :) |
| 19:11.00 | Trionnis | then you're paying too much :) |
| 19:11.10 | Trionnis | maybe I should give you a quote :) |
| 19:11.11 | Trionnis | haha |
| 19:11.12 | NetIQSystems | no.. I just need bandwidth ;) |
| 19:11.24 | Trionnis | I can get you unmetered 100mbit for about 250/mo |
| 19:11.25 | nortex | TheCompWiz, Hello |
| 19:11.27 | Trionnis | beat that |
| 19:11.28 | Trionnis | :) |
| 19:11.29 | TheCompWiz | so, NetIQSystems... any chance you can help me? |
| 19:11.31 | NetIQSystems | cogent crap. |
| 19:11.34 | Trionnis | nope |
| 19:11.37 | Trionnis | no cogent |
| 19:11.43 | TheCompWiz | hey nortex.. |
| 19:11.49 | NetIQSystems | then quit teasing me.. |
| 19:11.57 | NetIQSystems | who is it? |
| 19:11.58 | watchy | nortex: you know how to make a polycom 501 ring when a 2nd call is incoming instead of callwaiting beep |
| 19:12.02 | Trionnis | I could say the same :) |
| 19:12.03 | nortex | TheCompWiz, Waht is wrong with your voice mail |
| 19:12.17 | Trionnis | abovenet, telia, savvis, and at&T |
| 19:12.23 | NetIQSystems | sweet. |
| 19:12.25 | TheCompWiz | voice mail answers... but there is no "voice" ... |
| 19:12.34 | NetIQSystems | TheCompWiz, codec issue. |
| 19:12.39 | nortex | watchy, never tried, But I know that [TK]D-Fender is a Polycom God :) |
| 19:12.40 | NetIQSystems | now go RTFM |
| 19:12.56 | watchy | he says he dunno if it can be don |
| 19:12.57 | watchy | e |
| 19:13.03 | TheCompWiz | NetIQSystems... ok... that I believe. and which M would you like me to be reading? |
| 19:13.10 | nortex | TheCompWiz, do you have a CLI output you can pastbin ? |
| 19:13.11 | NetIQSystems | all of them. |
| 19:13.30 | hmmhesays | wow my tif's are coming out all funneh |
| 19:13.37 | charles___ | Anyone have had issues terminating with Vonage ? |
| 19:13.39 | TheCompWiz | nortex... I don't even know what to paste. (not too familiar with CLI) |
| 19:13.43 | NetIQSystems | Trionnis, I have a pretty good provider right now, but there is always a catch.. |
| 19:13.51 | Trionnis | yes, there usually is |
| 19:14.26 | Trionnis | brb |
| 19:14.54 | [TK]D-Fender | I'm f'n busy... got something else to add?! ;) |
| 19:15.44 | [TK]D-Fender | watchy : There is no current way to have it force-ring through the speaker spcifically. It will always default to the current audio device, but you CAN substitute the audio BEEP with another sound. |
| 19:16.33 | ccherrett | E-bola: vgetty can do the phone calls for me. Thanks |
| 19:16.47 | E-bola | np |
| 19:17.42 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.18) |
| 19:19.41 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 19:20.44 | *** part/#asterisk derekS (n=dereks@unaffiliated/dereks) |
| 19:21.51 | watchy | tk: well the guy at poly told me this but u know hes probably a idiot like most people who work at thier company |
| 19:22.06 | watchy | you probably work with the phones more then him but i havent had time to test his theory |
| 19:23.00 | watchy | he says set |
| 19:23.01 | watchy | reg.1.callsperlinekey=2 |
| 19:23.01 | watchy | but i havent had time to test it |
| 19:23.35 | *** join/#asterisk c4t3l (n=c4t3l@69.15.174.114) |
| 19:23.59 | nortex | watchy, I use that command. Let me see if it rings. |
| 19:24.17 | watchy | and i gave my asterisk dev box to a company because their server sucked so bad that they needed it yesterday |
| 19:24.39 | watchy | im gonna have to work late to get another dev box up it sucks |
| 19:24.56 | watchy | if it rings these bastards will be happy if not back to square one |
| 19:25.07 | watchy | tk already said i should send the woman to the ear doctor |
| 19:25.21 | [TK]D-Fender | watchy : No that will simply not cascade the call to the next line key (which people find a NATURAL thing) |
| 19:25.48 | [TK]D-Fender | watchy : That will STILL cause a CW bbep. ANY incoming call while you're on the phone will merely beep. |
| 19:25.58 | watchy | yea thats what i thought |
| 19:26.19 | watchy | im gonna change to beep to FINANSWER.gsm i guess |
| 19:26.24 | watchy | and have it scream ANSWER |
| 19:26.29 | [TK]D-Fender | watchy : Oh yeah now I remember... I TOLD YOU ALL THIS BEFORE! I mean sure I like the sound of my own voice, but not THAT much! |
| 19:26.38 | *** join/#asterisk luke-jr_ (n=luke-jr@user-0c93tin.cable.mindspring.com) |
| 19:26.46 | [hC] | have you guys heard about this USF voip tax stuff? The "Universal Service Fund?" |
| 19:26.48 | watchy | this is irc tk there is no voices :( |
| 19:26.51 | luke-jr_ | Is landline latency any better than VoIP? |
| 19:27.10 | [TK]D-Fender | watchy : They don't like it when you talk about them like that!~ |
| 19:29.10 | watchy | dont make me cry |
| 19:29.36 | syzygyBSD | latency... hmmm, do they really use that term in landlines? |
| 19:31.20 | nortex | watchy, no luck here my pretty 501 just beeps about the second call an flashes me a pretty red light. [TK]D-Fender was right again. |
| 19:31.22 | Trionnis | hmm, wonder if NuFone will actually refund my balance without me having to take them to small claims now that I've caught them trying to scam me for $60 to "force the port" of a number that was released into the general pool a month ago |
| 19:31.26 | watchy | would mp3123 missing for on hold music |
| 19:31.29 | watchy | fuck up transfers |
| 19:31.34 | watchy | if you moved everything |
| 19:31.36 | Trionnis | anyone wanna lay odds on it? :) |
| 19:31.41 | watchy | and forgot to install mp3123 |
| 19:32.22 | charles___ | watchy: mpg123 |
| 19:32.32 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 19:34.31 | watchy | yea thats what i meant |
| 19:34.36 | watchy | i think its been breaking my transfers |
| 19:35.03 | watchy | WELL I GOT ON HOLD MUSIC AGAIN |
| 19:35.07 | watchy | MAYBE TRANSFERS WILL FUCKING WORK |
| 19:35.17 | watchy | god ive been fighting this shit all day |
| 19:35.29 | watchy | i was about to shoot myself |
| 19:35.34 | Seba_soy | hello experts! |
| 19:35.42 | watchy | dont look at me |
| 19:36.06 | Seba_soy | :) |
| 19:36.12 | watchy | im as useful with asterisk as a sped is with ghost recon on xbox 360 |
| 19:36.20 | hmmhesays | anyone really familiar with imagemagick's convert tooL? |
| 19:36.53 | Seba_soy | guys, someone can tell me if asterisk or similar can handle 1.000 concurrent calls? |
| 19:36.55 | watchy | god whys my life suck |
| 19:36.58 | Seba_soy | maybe sipx |
| 19:37.05 | Trionnis | 1.000 calls? |
| 19:37.06 | Seba_soy | I want a softswitch solution! |
| 19:37.09 | Trionnis | I'm pretty sure it could |
| 19:37.19 | [TK]D-Fender | Seba_soy : "depends" |
| 19:37.24 | Trionnis | oh wait, EU doesn't use commas do they? |
| 19:37.35 | eKo1 | Trionnis: commas? |
| 19:37.37 | Seba_soy | [TK]D-Fender: of what? |
| 19:37.49 | Trionnis | 1,000 != 1.0 in US notation |
| 19:37.58 | Trionnis | 1.0 = one |
| 19:38.06 | Seba_soy | one thousand |
| 19:38.07 | [TK]D-Fender | Seba_soy : Are you talking 1000 ports of PRI direct by PCI into 1 * box..... |
| 19:38.19 | Trionnis | what format? |
| 19:38.24 | Seba_soy | no, 1000 sip calls |
| 19:38.24 | eKo1 | The comma and the dot are interchanged in some countries. |
| 19:38.26 | Trionnis | pri, sip, iax? |
| 19:38.29 | Seba_soy | all voip |
| 19:38.34 | [TK]D-Fender | Seba_soy : If you mean as a pure SIP softswitch, sure... without any transcoding and gigabit. |
| 19:38.34 | Trionnis | sip, iax? |
| 19:39.01 | Trionnis | SER might be a bit better if you're using SIP |
| 19:39.06 | Trionnis | with that kind of volume |
| 19:39.11 | Seba_soy | I wanmt a softswitch who accepto 1000,1500 concurrent call and route this to cisco connectedo to pstn |
| 19:39.33 | [TK]D-Fender | Seba_soy : You might be better off with SER and soon FreeSWITCH |
| 19:39.37 | Trionnis | omg |
| 19:39.41 | Trionnis | you said the "C" word |
| 19:40.09 | Seba_soy | maybe freeswitch |
| 19:40.28 | *** join/#asterisk bcnl (n=mike@S010600131078957c.vc.shawcable.net) |
| 19:40.29 | Seba_soy | when it have cdr :) |
| 19:41.02 | bcnl | has anyone ever seen error messages like this when people are parked and listening to MoH? |
| 19:41.05 | bcnl | WARNING[7526]: chan_sip.c:2552 sip_write: Asked to transmit frame type 64, while native formats is 256 (read/write = 256/256) |
| 19:41.14 | Seba_soy | transcoding |
| 19:41.19 | Trionnis | ack, anyone remember what the default audio file is for the ringing sound? |
| 19:41.34 | Trionnis | it's not "ring" or "ringing", but I can't find the damn thing |
| 19:41.53 | bcnl | Seba_soy: sorry? are you saying it has to do with the format my MoH is in? |
| 19:42.05 | Seba_soy | it is transcoding problem, sure |
| 19:42.09 | [TK]D-Fender | bcnl : Yup.. you're trying to transcode to G729 in passthrough mode w/o licenses or native recordings for it. |
| 19:42.24 | bcnl | [TK]D-Fender: I have 8 licenses for g729 though |
| 19:42.26 | eKo1 | Trionnis: I didn't know there was an audio file for ringing. |
| 19:42.32 | bcnl | 0/0 encoders/decoders of 8 licensed channels are currently in use |
| 19:42.33 | Trionnis | I was thinking there was |
| 19:42.38 | Trionnis | hrm |
| 19:42.49 | file | bcnl: what version of Asterisk? |
| 19:42.55 | bcnl | 1.2.10 |
| 19:42.59 | [TK]D-Fender | bcnl : Hrm |
| 19:42.59 | eKo1 | The ringing sound usually comes from the phone. |
| 19:43.16 | Trionnis | well yes, but this is for an IVR menu |
| 19:43.26 | file | Asterisk generates it |
| 19:43.29 | Trionnis | I need to put a single "ring" at the start of the menu |
| 19:43.38 | hmmhesays | convert seems to not like to resample my tiff images |
| 19:43.39 | Trionnis | hi file! |
| 19:43.42 | Trionnis | long time no see |
| 19:43.42 | file | helllllo |
| 19:43.47 | Trionnis | how goes it :) |
| 19:43.49 | eKo1 | Trionnis: a single ring? what for? |
| 19:43.55 | bcnl | if I were to take the time to use sox to transcode my MoH to all the formats and put them in a directory with the right file extensions, would that help? |
| 19:43.56 | file | Trionnis: fine... do I know you? |
| 19:43.56 | Seba_soy | Trionnis try to generate the tone |
| 19:44.02 | Trionnis | 'cause right now it drops right into the menu audio |
| 19:44.15 | Seba_soy | what format have moh now, bcnl |
| 19:44.28 | file | there's an application called Ringing, then you do a Wait |
| 19:44.30 | eKo1 | Trionnis: Just put a wait command |
| 19:44.31 | bcnl | Trionnis: http://voip-info.org/wiki/index.php?page=Asterisk+cmd+Ringing |
| 19:44.31 | *** part/#asterisk nDuff (n=ccd@64.128.31.220) |
| 19:44.34 | Trionnis | file: I would hope so, I'm the one that was in here a while back and gave you some kind of "good idea" |
| 19:44.44 | Trionnis | although you never would tell me what it was :) |
| 19:44.50 | file | Trionnis: ah |
| 19:44.54 | Trionnis | ack, that's right |
| 19:44.55 | Trionnis | Ringing |
| 19:44.58 | Trionnis | der |
| 19:44.59 | Trionnis | thanks :) |
| 19:45.05 | bcnl | now back to me :P |
| 19:45.25 | Seba_soy | so, do you think i can handle 1000 concurrent sip calls with asterisk? |
| 19:45.34 | Trionnis | SER |
| 19:45.36 | Seba_soy | and some 100 calls per second? |
| 19:45.42 | Trionnis | you want SER |
| 19:45.52 | file | bcnl: Asterisk will automatically pick the best format when reading stuff like this, so if you were to put your MOH as G729... it should read it in as G729 and just feed it to the channels, no transcoding |
| 19:45.53 | Trionnis | Asterisk can probably do it, but it would be taxing |
| 19:45.57 | Seba_soy | I dont like SER |
| 19:46.08 | Seba_soy | I will wait to freeswitch |
| 19:46.12 | Seba_soy | instead |
| 19:46.18 | Trionnis | well, if you want to throw enough hardware at it, * could probably do it |
| 19:46.34 | bcnl | file: ok and if a call is SIP<->SIP then I'll have to use one of my transcoding licenses ? |
| 19:46.39 | bcnl | which should be OK since I have 8 |
| 19:46.41 | Seba_soy | Trionnis: and what hardware is it? |
| 19:46.50 | *** join/#asterisk anthm (n=anthm@65.169.134.2) |
| 19:46.50 | *** mode/#asterisk [+o anthm] by ChanServ |
| 19:46.59 | file | bcnl: if transcoding has to take place, then it'll use a license - ie: one side is ULAW, and the other is G729 |
| 19:47.13 | watchy | i guess if you dont have mpg123 installed and you transfer folks it dont work |
| 19:47.29 | eKo1 | it seems a lot of people are using * beyond its pbx capabilities |
| 19:47.30 | bcnl | but those errors I'm seeing are related to transcoding |
| 19:47.34 | Trionnis | well, as was mentioned, you'd probably need at least gigabit |
| 19:47.40 | Trionnis | network* |
| 19:47.44 | bcnl | I seem to remember going through the hassle of making all my MoH in raw format |
| 19:47.47 | bcnl | and using rawplayer |
| 19:47.58 | Trionnis | I'd hazard a guess at a dual xeon with about 4g of ram would be minimum |
| 19:48.08 | Trionnis | that might not even be enough though |
| 19:48.24 | Trionnis | file would be the guy to answer that question with more certainty |
| 19:48.48 | file | me? hehe |
| 19:48.54 | Trionnis | I do know that any transcoding would kill you |
| 19:48.57 | Trionnis | haha :) |
| 19:49.05 | Seba_soy | 2 mb bandwidth are 30 calls |
| 19:49.07 | Trionnis | you know more about the hardware req's than I do |
| 19:49.15 | Seba_soy | so 300 calls are 20 mb bw |
| 19:49.20 | anthm | call for a live feed of cluecon .... IAX2/66.250.68.194/888 | SIP/888@66.250.68.194 | PSTN: 712-432-7800 |
| 19:49.24 | Seba_soy | 1000 calls = 70mb bw |
| 19:49.29 | Seba_soy | I am wrong? |
| 19:49.44 | eKo1 | that depends on the codec |
| 19:49.45 | Trionnis | well, 100mbit might be enough |
| 19:49.52 | Trionnis | if you're using 729 or something |
| 19:50.06 | eKo1 | and you're going to have to take into account packet header overhead |
| 19:50.08 | Seba_soy | if 729 then it is half |
| 19:50.16 | Seba_soy | 1000 calls = 40mb bw |
| 19:50.27 | hmmhesays | bah, convert won't let me change the aspect ratio damnit |
| 19:50.32 | Trionnis | and 729 will add a buttload of overhead unless you're going native 729 end to end |
| 19:50.48 | Seba_soy | sure, I will let transcoding on end side |
| 19:50.59 | Trionnis | hey anthm: you wanna shoutcast that? I'll offer you a server temporarily |
| 19:51.05 | Seba_soy | I just want softswitch for billing and call control |
| 19:51.41 | anthm | if we can work out to logistics perhaps |
| 19:51.43 | eKo1 | haha, me too |
| 19:51.50 | watchy | i just want someone to love' |
| 19:51.54 | Trionnis | check pm |
| 19:52.24 | bcnl | can sox transcode mp3's into g729 ? |
| 19:53.01 | bcnl | and for MoH if I were to have two copies of the file, one as g729 and one as ulaw, is there a chance that asterisk would pick the "correct" one when it needed a audio file? |
| 19:53.01 | Seba_soy | without license? |
| 19:53.01 | watchy | today has been the worst day of my 3month old career in computerz |
| 19:53.03 | Seba_soy | it cant |
| 19:53.07 | watchy | i just graduated from itt tech bitchs |
| 19:53.09 | watchy | and it sucks |
| 19:53.17 | Corydon-w | No, not until the patent expires |
| 19:53.22 | bcnl | Seba_soy: well I have a license for it, I just want to reformat my Moh |
| 19:53.33 | nortex | [TK]D-Fender, Is Polycom 1.6.7 what they talked about being 2.0? or is that still on the radar? |
| 19:53.40 | Seba_soy | there is a free tool on the web |
| 19:53.47 | Seba_soy | on asteriskguru |
| 19:53.50 | Seba_soy | I think |
| 19:54.00 | watchy | whats 1.6.7 add? |
| 19:54.08 | bcnl | Seba_soy: thanks |
| 19:54.24 | Trionnis | itt tech? |
| 19:54.31 | bcnl | http://www.asteriskguru.com/tools/audio_conversion.php |
| 19:54.38 | Trionnis | isn't that kinda like getting a GED instead of going to high school? |
| 19:54.43 | Trionnis | (yes, I'm kidding) |
| 19:54.44 | nortex | ouch |
| 19:54.59 | file | be nice to the general #asterisk population |
| 19:55.06 | Trionnis | hey, it was a joke |
| 19:55.14 | Trionnis | I am being nice :) |
| 19:55.20 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
| 19:55.21 | bcnl | heh |
| 19:55.40 | Trionnis | unlike some people that like to bash my choice of distro |
| 19:55.57 | *** join/#asterisk terje (n=joem@67.41.208.129) |
| 19:56.20 | terje | How can I get my asterisk server to dial a number and play a .wav file? |
| 19:56.21 | watchy | i was lying i didnt goto itt tech haha |
| 19:56.23 | bcnl | eKo1: blue from holding it's breath between release cycles |
| 19:56.31 | watchy | ive been in it for 10 years and im tired of it |
| 19:56.32 | eKo1 | bcnl: lol |
| 19:56.49 | eKo1 | terje: with the proper dialplan |
| 19:56.49 | Trionnis | hahaha |
| 19:56.52 | Trionnis | that was a good one |
| 19:57.05 | bcnl | FTR, I've run debian since it was debbie and ian |
| 19:57.10 | bcnl | thanks |
| 20:00.18 | terje | thanks eKo1 |
| 20:00.27 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 20:00.56 | Trionnis | hrm |
| 20:01.02 | Trionnis | Ringing doesn't seem to be doing anything |
| 20:01.05 | Trionnis | =/ |
| 20:01.48 | file | depending on the state of the channel it can either signal back to indicate ringing, or generate ringing as audio |
| 20:01.56 | *** join/#asterisk Pazzo (n=thomas@host130-250-static.72-81-b.business.telecomitalia.it) |
| 20:01.57 | Trionnis | hrm |
| 20:02.15 | Trionnis | well, my dialplan kicks the incoming call right into the menu |
| 20:02.20 | eKo1 | Trionnis: try doing a wait() first, then an answer() |
| 20:02.24 | Trionnis | ok |
| 20:02.28 | Trionnis | will do, thanks :) |
| 20:02.31 | *** join/#asterisk riddlebox (n=blah@24-207-167-238.dhcp.stls.mo.charter.com) |
| 20:02.35 | riddlebox | hello |
| 20:02.49 | watchy | i want someone kick me in the face with a loop |
| 20:03.59 | riddlebox | if I want to use a modem connected my sipura box, through asterisk, do I need to do anything special in asterisk? |
| 20:04.08 | bcnl | watchy: sounds sadistic |
| 20:04.28 | watchy | tis been a bad day |
| 20:04.36 | bcnl | riddlebox: you want to use the modem for data connections? |
| 20:04.44 | watchy | first echo out the fucking ass but the nice people at digium worked magic and fixed it |
| 20:04.46 | *** join/#asterisk pointer (i=pointer@aj.catt.com) |
| 20:04.52 | eKo1 | riddlebox: that is a bad idea |
| 20:05.04 | riddlebox | bcnl, I want to dial into some pbx's and voicemails we have |
| 20:05.08 | eKo1 | riddlebox: but feel free to try it out. make sure the codec is ulaw or alaw though |
| 20:05.10 | watchy | then transfers didnt work because if you move all your confs to a new box and use MOH and dont have a player well it breaks transfers |
| 20:05.11 | alexrch | why does "show queue ..." return "unknown" status for queue member that has just been added to the queue using AddQueueMember: Local/1@test with penalty 3 (dynamic) (Unknown) has taken no calls yet |
| 20:05.19 | *** join/#asterisk arkonadev (n=chatzill@65.203.186.131) |
| 20:05.25 | arkonadev | we got any dial plan gurus? |
| 20:05.53 | eKo1 | arkonadev: no gurus, just experts |
| 20:05.58 | riddlebox | eko1, I have dissallowed all then allowed ulaw, and alaw but it still doesnt connect |
| 20:06.13 | watchy | now theh yare bitching about it not dialing other people |
| 20:06.17 | nestar | lower your baud rate |
| 20:06.17 | eKo1 | riddlebox: then it doesn't work. move along. |
| 20:06.27 | n9urk | I have installed asterisk addons. Do I need to do anything other that to get cdr_mysql? |
| 20:06.28 | watchy | so i have the sec take a call if in 15secs she dont answer it dials like 5 otherp hones |
| 20:06.30 | nestar | 14,400 or 9600 |
| 20:06.32 | watchy | then theycomplain it dont work |
| 20:06.42 | watchy | cuz some fuck answers it on ring #1 |
| 20:06.48 | watchy | so they htink it hangs up after 1 ring |
| 20:06.53 | n9urk | I don't see it in /usr/lib/asterisk/modules |
| 20:07.18 | arkonadev | well im sure that will do....i am working with an asterisk sold by a company www.fonality.com anyways there dialplan is really different from tutorials and other stuff i have seen basically their extensions.conf consists of only one include and in the included files there is just another list of files so basically i need to add in a call to an AGI script everytime an inbound call comes in or when the call is transfered |
| 20:07.32 | riddlebox | eko1, is there anyway other way to do this or am I out of luck all together? |
| 20:07.43 | riddlebox | btw I have broadvoice as well |
| 20:07.49 | nestar | riddlebox: try lowering your baud rate |
| 20:07.52 | arkonadev | any ideas |
| 20:07.54 | eKo1 | arkonadev: fonality? how did that work out? |
| 20:08.09 | eKo1 | riddlebox: get a regular land line |
| 20:08.38 | arkonadev | wel im doing an internship for a car dealership management company and they had licensed at pbxtra to resell so thats what im stuck with using |
| 20:09.18 | arkonadev | at=out |
| 20:10.10 | Trionnis | eKo1: no love, still not generating it |
| 20:10.49 | eKo1 | Trionnis: what happens? |
| 20:10.55 | Trionnis | nothing, just dead air |
| 20:11.03 | Trionnis | then it goes into the IVR audio |
| 20:11.24 | eKo1 | arkonadev: I suggest dumping their entire setup and making your own. |
| 20:11.36 | eKo1 | Unless you want to reverse engineer what they're doing. |
| 20:12.01 | *** join/#asterisk Wazb^ (n=wazb@199.243.74.220) |
| 20:12.04 | Wazb^ | hi to all |
| 20:12.08 | hmmhesays | hmm this is working pretty well now |
| 20:12.22 | eKo1 | Trionnis: do this then: answer(), ringing(), wait() |
| 20:12.32 | Trionnis | that's what I tried the first time |
| 20:12.33 | Trionnis | heh |
| 20:12.36 | arkonadev | well the problem is were just trying to make a simple add on app that has an agi script that should be called whenever a call is coming in or transfered......rebuildling the whole system is kind of out of the scope of my project |
| 20:12.39 | Trionnis | oh, hang on |
| 20:12.48 | Trionnis | I might have just caught a typo |
| 20:12.53 | arkonadev | we can get the AGI script to work but then there stuff doesnt or viceversa |
| 20:13.35 | eKo1 | arkonadev: you're going to have to figure out how fonality does things |
| 20:13.50 | Trionnis | nope |
| 20:13.53 | Trionnis | still doesn't work |
| 20:13.56 | Wazb^ | i just installed Trixbox with its all updation and i installed 2 lincenses of G729 with it. I want to use this box for Calling Card application. But when DID hit this box i cannot hear nothing , i can see on console say playing 'prepaid-enter-pin-number' |
| 20:14.03 | arkonadev | yeah and fonality really sucks...we called the support lines asking them what we should edit in the conf files and they said it edit them through the web interface |
| 20:14.05 | *** join/#asterisk IvyUK (n=mark@194.201.148.122) |
| 20:14.16 | [TK]D-Fender | watchy, nortex : 1.6.7 adds a bunch of important bug fixes, includes 1.6.6b/c support for the IP430, inproved SIP response time, etc |
| 20:14.20 | Wazb^ | do i need all prompts in G729 codec? |
| 20:14.28 | Trionnis | Wazb^: read topic :) |
| 20:14.39 | IvyUK | hi all, can anyone confirm if you need to patch asterisk with the uk CID patch when using PRI |
| 20:14.43 | eKo1 | Trionnis: how about ringing(), wait(), answer() |
| 20:14.44 | *** part/#asterisk terje (n=joem@67.41.208.129) |
| 20:14.45 | Wazb^ | oh ok |
| 20:14.48 | Wazb^ | thanks |
| 20:14.52 | Trionnis | sure, np :) |
| 20:14.59 | Trionnis | eKo1: that doesn't make sense to me though |
| 20:15.11 | Trionnis | how could it play ringing tones before it even answers the channel? |
| 20:15.31 | eKo1 | Trionnis: we're running out of options here so might as well |
| 20:15.44 | Trionnis | I'll give it a shot, 1 sec |
| 20:16.27 | Trionnis | holy crap |
| 20:16.37 | Trionnis | that makes no sense whatsoever |
| 20:16.39 | Trionnis | but it works |
| 20:16.41 | Trionnis | ... |
| 20:16.44 | Trionnis | haha, thanks! |
| 20:16.45 | eKo1 | hehehe |
| 20:16.49 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 20:18.06 | jero | anyone has issues with grandstreams loosing registration after some random time and unable to receive any call ? |
| 20:18.13 | jero | (grandstream budgetones |
| 20:18.33 | eKo1 | I've had that and other issues. |
| 20:18.44 | eKo1 | Just reboot it. |
| 20:20.02 | Wazb^ | ok could anyone tell me is there any utility to convert all gsm files into g729 files ? |
| 20:20.20 | bcnl | Wazb^: http://www.asteriskguru.com/tools/audio_conversion.php |
| 20:20.35 | bcnl | since I just asked that about 15 minutes ago :> |
| 20:20.36 | jero | eKo1, the problem is, one cannot know when one lost registration |
| 20:20.56 | eKo1 | Sure you can because the phone won't work. |
| 20:21.03 | eKo1 | So if it doesn't work, reboot it. |
| 20:21.05 | jero | it does, i can place calls |
| 20:21.22 | Trionnis | Wazb^: if you have a bunch of them, you could probably do something in perl to batch convert them |
| 20:21.23 | Wazb^ | thanks |
| 20:21.51 | E-bola | lol |
| 20:21.55 | E-bola | try explaining a user that |
| 20:22.01 | eKo1 | jero: so what is the problem then? |
| 20:22.02 | E-bola | "u have to reboot your phone..." |
| 20:22.06 | Trionnis | who said anything about explaining? |
| 20:22.16 | Trionnis | I just told him of a possibility ^.^ |
| 20:22.51 | eKo1 | E-bola: I just say: "Unplug the power cord of your phone, wait 10 secs., then plug it back in." |
| 20:23.05 | Trionnis | oh, misread |
| 20:23.21 | bcnl | Trionnis: you can transcode g729 in perl? |
| 20:23.21 | jero | eKo1, incoming calls go to voicemail without ringing the phone |
| 20:23.26 | E-bola | well its a phone |
| 20:23.26 | E-bola | not a windows pc |
| 20:23.26 | E-bola | its not accesible for it to crash |
| 20:23.28 | bcnl | cause that'd kickass on using a web interface |
| 20:23.32 | jero | eKo1, which is solved when rebooting the phone |
| 20:23.46 | Trionnis | well, if you have a shell program to do it, why couldn't you? |
| 20:23.55 | eKo1 | bcnl: I think mplayer can transcode. |
| 20:23.56 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 20:24.00 | Trionnis | just awk the file list, and loop it through |
| 20:24.07 | [TK]D-Fender | E-bola : Clearly you've never owned a Snom phone before :) |
| 20:24.15 | *** join/#asterisk pardove (n=pardove@217.219.250.25) |
| 20:24.22 | E-bola | We are os far basing our clients on Linksys phones |
| 20:24.30 | E-bola | Spa922 and similar |
| 20:24.35 | E-bola | they seem pretty stable so far |
| 20:24.35 | [TK]D-Fender | E-bola : Where are you located? |
| 20:24.45 | eKo1 | jero: ah, so that is a different problem |
| 20:25.00 | jero | eKo1, thats evil |
| 20:25.11 | pardove | rx_fax fails to get fax on a bit noisy lines but real fax devices can do that on the same line! what's the problem? |
| 20:25.30 | E-bola | Denmark |
| 20:27.04 | [TK]D-Fender | E-bola : Yeah I guess where you are they're a noticably cheaper option than Cisco/Polycom... |
| 20:27.19 | [TK]D-Fender | E-bola : Not my preference, but budget does come to mind... |
| 20:27.38 | [TK]D-Fender | E-bola : The SPA's are very stable in my esxperience |
| 20:28.04 | E-bola | It is budget phones |
| 20:28.10 | E-bola | all our clients are in the smb segment |
| 20:28.27 | *** join/#asterisk caloi (n=caloi@65.169.134.2) |
| 20:29.13 | [TK]D-Fender | E-bola : Yeah, they're the bottom end (them or Aastra) on my "suggest" list. |
| 20:29.14 | arkonadev | does anyone know an excellent reference for diaplan creation? |
| 20:29.15 | E-bola | heh sucks to be on the internet when there s thunderstorm outside :( |
| 20:29.23 | [TK]D-Fender | arkonadev : ... |
| 20:29.24 | [TK]D-Fender | ~docs |
| 20:29.26 | jbot | well, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 20:29.26 | [TK]D-Fender | ~book |
| 20:29.27 | jbot | it has been said that book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 20:29.34 | E-bola | [TK]D-Fender: whats a little better than the linksys spa's? |
| 20:29.38 | E-bola | but not alot more expensive |
| 20:29.42 | arkonadev | thanks |
| 20:29.43 | pardove | rx_fax fails to get fax on a bit noisy lines but real fax devices can do that on the same line! what's the problem? |
| 20:29.47 | [TK]D-Fender | E-bola : Polycom beats them in just about every way. |
| 20:29.59 | E-bola | i saw the ip501 i think it was called |
| 20:30.04 | E-bola | that sure doesnt beat them designwise |
| 20:30.10 | [TK]D-Fender | E-bola : Except the backlight on the SPA-942, but frankly I wouldn't make that the deciding factor :) |
| 20:30.52 | [TK]D-Fender | E-bola : I own every model they produce. Yeah the linksys is a nice deisn, but poor LCD usability, and cheaper feel. |
| 20:31.07 | E-bola | ? the LCD on my spa922 is incredible |
| 20:31.58 | [TK]D-Fender | E-bola : Try comparing to the res & text readability of the 601, and then compare the featureset. SPA's come up pretty short. No presence support, 2 lines (w/o paying more), poor level of call volume handling. |
| 20:32.01 | nortex | Ebola, It is no the same, I have one at home and my Polycoms at work are nicer. |
| 20:32.18 | Ebola | Hi |
| 20:32.20 | [TK]D-Fender | E-bola : Clear yes, but the fonts are god-aweful small ajust like the screen.. |
| 20:32.26 | Ebola | You want that guy over there, not me. |
| 20:32.26 | nortex | Although I like my linksys for at home :) |
| 20:32.31 | E-bola | I guess i dont got alot to compare with |
| 20:32.47 | E-bola | But it on first looks it scores very high on the coolness factor hehe |
| 20:33.02 | E-bola | i did indeed have problems with call volume though :/ |
| 20:33.05 | [TK]D-Fender | E-bola : I had an IP 600 next to my SPA-941 for a while... I had plenty of time to regret my SPA purchase before getting rid of it :) |
| 20:33.18 | E-bola | the IP line is the polycoms? |
| 20:33.19 | nortex | Thats what one of my bosses said about the Cisco, till price came around :) |
| 20:33.34 | [TK]D-Fender | E-bola : Now keep in mind where you are Polycom IS actually noticably more expensive, so that does factor in. |
| 20:33.49 | E-bola | i havent seen any distributors |
| 20:33.53 | [TK]D-Fender | E-bola : Here however, they are close enough that SPA's don't deserve to be on the list. |
| 20:33.54 | E-bola | so i dont even know if i can get them if i wanted to |
| 20:34.00 | [TK]D-Fender | E-bola : Correct. |
| 20:34.05 | E-bola | i mainly see cisco avaya snom and linksys |
| 20:34.08 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 20:34.33 | [TK]D-Fender | E-bola : Cisco's are top-end for sure. SIP is still a little flakey, but you'll get no arument from me they are a great phone. |
| 20:34.44 | [TK]D-Fender | E-bola : However no real presence support and a few other things. |
| 20:35.13 | Trionnis | ok, gotta run, thanks for the help everyone |
| 20:35.27 | E-bola | cant pressence support be handle on the server? |
| 20:35.37 | E-bola | i mean what does a phone need to support? |
| 20:35.37 | *** join/#asterisk TrevorSHarrison (n=trevorsh@24-49-36-218-st.chvlva.adelphia.net) |
| 20:35.44 | [TK]D-Fender | E-bola : the phone has no way of telling you however. |
| 20:35.55 | E-bola | true u'd have to remember |
| 20:36.18 | [TK]D-Fender | E-bola : You want a light on your phone when a resource is in use right? Like for receptionists to know who's on the phone? Forget it with Cisco, Linksys. |
| 20:36.29 | [TK]D-Fender | E-bola : Remember? How would you even KNOW? |
| 20:36.30 | watchy | get a poly |
| 20:36.31 | E-bola | Ive been considering that actualy |
| 20:36.39 | E-bola | so far it havent been necesary but i bet somebody will want it |
| 20:36.48 | [TK]D-Fender | E-bola : Thats the point... Polycom IP601 + Attendant modules = godly |
| 20:36.58 | E-bola | and that works flawlessly with asterisk? |
| 20:36.59 | [TK]D-Fender | E-bola : OH YEAH they're gonna.... |
| 20:36.59 | watchy | tk:agreed |
| 20:37.04 | [TK]D-Fender | E-bola : WOrks great |
| 20:37.14 | *** part/#asterisk Trionnis (i=lordkuri@12.206.2.116) |
| 20:37.14 | E-bola | also in a purely IP based setup? |
| 20:37.15 | watchy | yea they do bola, i got about 30 deployed |
| 20:37.21 | E-bola | no fxo ports etc |
| 20:37.23 | [TK]D-Fender | E-bola : thats all they are... SIP phones. |
| 20:37.30 | E-bola | Cool |
| 20:37.53 | [TK]D-Fender | I run a 30 seat Ploycom setup here, (60x, 301), and own a 501, 430, and 301 at home |
| 20:38.20 | [TK]D-Fender | E-bola : EU pricing is NOT so kind unfortunately. |
| 20:38.30 | E-bola | it rarely is :/ |
| 20:38.39 | [TK]D-Fender | E-bola : which is why I'm not suggesting you just dump yoru SPA's outright :) |
| 20:38.43 | watchy | i love my polys i hate my 2 ciscos i bought |
| 20:38.52 | [TK]D-Fender | I know Linksys is pretty cheap world-wide |
| 20:39.09 | [hC] | [TK]D-Fender: how do you like the 430? |
| 20:39.10 | nortex | I have over 70 Polys deployed here and 8 sites to go :) |
| 20:39.11 | [TK]D-Fender | watchy : Yeah so far I have not seen anything I'd suggest over Polycom. |
| 20:39.36 | E-bola | lol damm |
| 20:39.45 | [hC] | I have the occasional complaing about polycom audio quality being echo-ey or crackly, but im going to check into my gains in sip.cfg, as someone suggested |
| 20:39.48 | E-bola | a ip501 is more than twice the price of my spa922 |
| 20:39.49 | E-bola | :( |
| 20:40.06 | [TK]D-Fender | [hC] : LOVE it. great hybrid phone. great LCD usability for its size, lighted line-key indicators, small frame. #1 SMB general phone (for PoE). If you don't need PoE I'd still suggest the 501 in many cases. |
| 20:40.17 | [TK]D-Fender | E-bola : I *sis* tell you about that :) |
| 20:40.20 | [TK]D-Fender | DID * |
| 20:40.27 | gursikh | I purchased the ip 501's based on recomdations here and in #freepbx, Have not been disapointed |
| 20:40.30 | [TK]D-Fender | E-bola : You should call up to find a beter rate |
| 20:40.31 | E-bola | are cisco's even more expensive? |
| 20:40.47 | E-bola | [TK]D-Fender: true this was just a random shop online in denmark |
| 20:40.50 | [TK]D-Fender | E-bola : Dunno in your area. You need to really shop around. |
| 20:40.52 | E-bola | but its still gonna be alot more expensive |
| 20:41.01 | [TK]D-Fender | E-bola : I'd say maybe 50% |
| 20:41.18 | [TK]D-Fender | E-bola : Here a 922 costs about the same as an IP 501. |
| 20:41.22 | [hC] | [TK]D-Fender: why would you suggest the 501 over the 430 for non poe installs? |
| 20:42.17 | [TK]D-Fender | [hC] : the 501 has a bigger LCD which is easier to browsethe menus and deal with calls, 3 line keys (more appearances). |
| 20:42.40 | [hC] | [TK]D-Fender: nod. |
| 20:42.40 | [TK]D-Fender | [hC] : Owning both the 501 is still higher end. the 430 is a great GENERAL phone however. |
| 20:42.56 | [hC] | Gotcha. |
| 20:43.07 | [TK]D-Fender | [hC] : normal people don't NEED more, but I for instance use each line appearance seperate for my customers. |
| 20:43.26 | [hC] | instead of having htem take multiple calls on one line key? |
| 20:43.32 | [TK]D-Fender | IP 430 is the "quicky" PoE phone of choice with a 601+Modules for receptionist for SMB. |
| 20:44.03 | [TK]D-Fender | [hC] : Each line key is its own reg with its own link to my customers PBX's. Each one supports multiple calls each. |
| 20:44.09 | watchy | i only sell 501s and 601s but 501s are prob overkill for my users |
| 20:44.23 | watchy | they could probably use 430s |
| 20:44.25 | [hC] | [TK]D-Fender: oh i see. so you have an appearance on their pbx's. |
| 20:44.27 | [TK]D-Fender | [hC] : this is instead of just dumping all calls on 1 reg and casccading multiple calls. |
| 20:44.37 | watchy | or whatyevers below a 501 |
| 20:45.01 | [TK]D-Fender | [hC] : Exactly. it means I don't have to mangle my dialplan to choose to select their server for testing. it means I can just dial like they do. |
| 20:45.16 | [TK]D-Fender | watchy : IP 301. |
| 20:45.49 | watchy | i think my clients finnaly happy with theri setup as of today |
| 20:45.51 | [TK]D-Fender | [hC] : the 430 is a 301 with better LCD, but the same # of text lines for scrolling. Just with underline & font support that much nicer in the same footprint. Full pixel |
| 20:47.02 | E-bola | what about the polycom ip300? |
| 20:47.14 | E-bola | thats pretty cheap |
| 20:47.17 | [hC] | [TK]D-Fender: and two way speaker phone. |
| 20:47.18 | [hC] | :) |
| 20:47.21 | E-bola | only a little bit more expensive than the spa922 |
| 20:48.32 | *** join/#asterisk rogier (n=rogier@83-67-dsl.ipact.nl) |
| 20:48.36 | [TK]D-Fender | E-bola : Great phone, lacks speakerphone, but thats about it. |
| 20:48.46 | E-bola | so its a 1-way speaker? |
| 20:48.48 | [TK]D-Fender | [hC] : IP 430 = full duplex |
| 20:48.51 | blitzrage | I really like the spa942 |
| 20:49.19 | champster | anyone used one of those with the metermaid patch? |
| 20:49.31 | champster | With hints for the parking pos.s |
| 20:49.39 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
| 20:51.27 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 20:51.31 | *** part/#asterisk pointer (i=pointer@aj.catt.com) |
| 20:51.55 | devel | hey all. just a quick question regarding realtime. i have it working but 'sip show peer(s)' is a pretty important debugging tool for me.... any alternatives, once using sip in realtime? |
| 20:52.08 | *** join/#asterisk Waverly360 (n=9893acdf@65.169.134.2) |
| 20:54.24 | Alric | Caching? |
| 20:54.36 | *** join/#asterisk anthm (n=anthm@65.169.134.2) |
| 20:54.36 | *** mode/#asterisk [+o anthm] by ChanServ |
| 20:55.01 | *** join/#asterisk ronaldl79 (n=chatzill@d198-53-139-22.abhsia.telus.net) |
| 20:55.20 | ronaldl79 | Anyone on trunk noticing excessive timeouts with SIP? |
| 20:55.41 | Qwell[] | devel: sip show peer <peer> load |
| 20:55.44 | Qwell[] | or some such |
| 20:56.41 | *** join/#asterisk arkonadev (n=chatzill@65.203.186.131) |
| 20:57.20 | arkonadev | whats the difference between fxo_ls, fxs_ks,fxo_ks? |
| 20:57.55 | Qwell[] | arkonadev: loopstart vs kewlstart, and fxo vs fxs |
| 20:57.57 | Qwell[] | ~fxofxs |
| 20:58.00 | jbot | from memory, fxofxs is An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
| 20:58.08 | Qwell[] | signalling is..."reversed" |
| 20:58.17 | Qwell[] | so, with an fxo, you'll want fxs signalling |
| 20:58.21 | convey | what do you guys think of the UTStarcom F3000? |
| 20:58.24 | Qwell[] | ~docs |
| 20:58.25 | jbot | extra, extra, read all about it, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 20:58.38 | Qwell[] | convey: I hear they suck, but I've not tried one |
| 20:58.41 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 20:59.10 | convey | @Quell: I just got one and I think they may suck.. |
| 20:59.21 | arkonadev | so in the zapatel.conf they have have it broken into 3 parts each part has the signalling set to one of those values |
| 20:59.26 | arkonadev | so what exactly is that doing |
| 21:00.06 | convey | @Quell: I would not advise that you buy one :) |
| 21:00.20 | *** join/#asterisk Tall-guy (i=tall-guy@207-195-103-110.regn.static.sasknet.sk.ca) |
| 21:01.14 | arkonadev | i guess what im really asking is under what condition would the fxo_ls would be used and when would fxo_ks be used and when would fxs_ks be used |
| 21:01.58 | arkonadev | becuase it looks like based on what signalling is used they go to different contexts in the dialplan |
| 21:03.33 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 21:04.20 | gursikh | Anyone, developer or advanced user, provide support services to Local New York City area? |
| 21:05.59 | charles___ | gursikh: full time ? |
| 21:06.10 | *** join/#asterisk iq (n=IQ@unaffiliated/iq) |
| 21:07.04 | Waverly360 | arkonadev: we just use the _ks stuff...but as far as fxo/fxs devices go, you use fxo_ks when you're connecting to an fxs device, and fxs_ks when you're connecting to an fxo device |
| 21:07.14 | gursikh | Well, One time now, Full time possibility |
| 21:07.32 | arkonadev | so which device handles incoming calls and which device handles outgoing calls |
| 21:07.34 | Waverly360 | an fxo device is a line that connects your pbx to an analog phone line |
| 21:07.44 | Waverly360 | well |
| 21:08.07 | Waverly360 | the fxo device (fxs_ks) is the one that deals with incoming and outgoing calls into and out of the pbx. |
| 21:08.23 | Waverly360 | an fxs device is a phone plugged directly into the pbx |
| 21:08.52 | charles___ | gursikh: I'm interested in offering but I'm in florida |
| 21:08.56 | arkonadev | fxs would handle all the extensions within a company and the fxo would handle all inbound and outbound traffic |
| 21:09.14 | Waverly360 | IF all of your internal lines are analog phones |
| 21:09.19 | Waverly360 | yes |
| 21:09.23 | arkonadev | k |
| 21:09.26 | Waverly360 | god I hope I'm explaining that right.. |
| 21:09.31 | arkonadev | lol |
| 21:09.35 | Waverly360 | someone tell me if I'm explaining it to him wrong |
| 21:09.51 | gursikh | Well, what I REALLY need now is someone extremely knowledgable in ASterisk and all the networking and IT stuff that goes along with it to come in (remotly) and Fix my installation. |
| 21:10.00 | Waverly360 | lol |
| 21:10.00 | Tall-guy | ZTOOL question: When I use "zttool", My 4 PSTN channels all show "Active"...(no calls present at the time tho), whats up with that?? |
| 21:10.09 | Tall-guy | (TDM400P) |
| 21:10.11 | Waverly360 | what's wrong with your gursikh? |
| 21:10.16 | arkonadev | so then one more thing is when does an loopstart get used and when does a kewlstart get used |
| 21:10.40 | Waverly360 | that I'm not sure what to tell you. We've always used kewlstart..and they just work |
| 21:10.51 | Waverly360 | I'm somewhat of a n00b myself...my apologies |
| 21:10.59 | gursikh | I had hired some guy (off of here or #freepbx I dont recall) and he had done the installation "ok" but many minor issues still exist, and I dont feel confident that he can fix them. |
| 21:10.59 | arkonadev | well im a super noob :P |
| 21:11.50 | Waverly360 | gursikh:can you explain some of the issues? |
| 21:11.54 | *** join/#asterisk Ark_Molt (n=chatzill@65.203.186.131) |
| 21:11.57 | Tall-guy | Kewlstart is loopstart with far end disconnect supervision |
| 21:12.15 | arkonadev | so when would a loopstart get used and when would the kewlstart be used |
| 21:12.42 | Waverly360 | well...couldn't you just use kewlstart and it just work? |
| 21:13.03 | arkonadev | thats the thing im working with a system already in place that i have to customize |
| 21:13.14 | *** join/#asterisk Bobcat_1966 (n=chatzill@cpe-069-132-139-254.carolina.res.rr.com) |
| 21:13.47 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 21:13.52 | devel | Qwell[], that's the stuff, thanks a lot. |
| 21:14.03 | Tall-guy | arkonadev: use kewlstart, there are VERY few instances where it won't work for you. |
| 21:14.30 | arkonadev | so let me give you a rundown of what is going on when signalling=fxo_ls they go the default context when the signalling=fxs_ks the context is incoming and when the singalling=fxo_ks the context internal so im just trying to figure out a reason for them to do this |
| 21:14.41 | gursikh | People cannot Get though to us (fast busy, or nothing) Dialing out gives error messages (it. "dial '1' to reach, when you dont have to, and when you do dial one it doesn't work) CallerID (outbound) is not working 80% of the time) Certain numbers (especially 1-800's dont work period (outbound). Voicemail drops the call when checking messages. Button's for phone for message and voicemail are not ocnfigured. |
| 21:14.50 | Waverly360 | well...you setup the context in zapata.conf |
| 21:14.50 | gursikh | and so on :-( |
| 21:14.51 | arkonadev | why are they going to different contexts based on the signalling |
| 21:15.11 | Waverly360 | those devices will go wherever you want them to...look in zapata.conf |
| 21:15.12 | gursikh | And he couldn't get AstaTapi to work . |
| 21:15.15 | arkonadev | no i didnt set it up the company i work for bought a pre setup asterisk server from www.fonality.com now i have to figure out what the hell they are doing |
| 21:17.13 | Waverly360 | gursikh: hmm. Can you give me details on the hardware config? PRI? Analog lines? IAX2? |
| 21:18.47 | arkonadev | on quick question thats off the topic of the first couple is if you include a context in another is that context called before the context your in |
| 21:19.16 | Waverly360 | well..I think it's inserted where your actual include line is |
| 21:20.08 | gursikh | Waverly360: standby please |
| 21:20.12 | Waverly360 | gursikh: I got 1.5 hours of battery life left :P |
| 21:20.26 | Un1x | anyone know of some small adapter or something |
| 21:20.30 | Un1x | i can stick onto a PSTN line |
| 21:20.37 | Un1x | for it to broadcast the shit over Antenna |
| 21:20.54 | gursikh | May I PM you, or shall I leave it here? |
| 21:21.07 | c4t3l | if you have a long enough wire I suppose you could use an ATA |
| 21:21.09 | Waverly360 | you can leave it here..unless it's HUGE :) |
| 21:21.16 | gursikh | no, I'll just type it as I go |
| 21:21.41 | c4t3l | or you could make a massive tone amp |
| 21:21.44 | eKo1 | Un1x: a cordless phone? |
| 21:21.54 | Un1x | no |
| 21:22.00 | Un1x | i mean some tiny small adapter |
| 21:22.04 | Un1x | comes with a phone or a receiver |
| 21:22.23 | Un1x | so i could like plug in the adapter into my freinds house :P and bring the reciever to mien |
| 21:22.28 | Un1x | and plug my phone into the receiver :D |
| 21:22.37 | Bobcat_1966 | hello All, anybody know what this module does?" format_au.so". When using asterisk 1.2 it asterisk starts fine but after updating to Asterisk SVN Trunk Asterisks will not start unless I commint the module out in the modules.conf file. |
| 21:22.46 | eKo1 | Un1x: you've just described a cordless phone |
| 21:22.49 | file | Bobcat_1966: wipe your modules directory |
| 21:22.57 | file | Bobcat_1966: and do a make install again from trunk |
| 21:23.33 | Bobcat_1966 | yep I did but format_au.so is in the modules.conf file....The system seems to work fine when I comment it out but I was just wondering what it did. |
| 21:23.52 | file | it was consolidated into another format file since they did the same thing basically |
| 21:24.00 | file | so it doesn't exist as a seperate module anymore |
| 21:24.03 | Un1x | ek01 no i have not |
| 21:24.10 | Un1x | i cannot attach a cordless phone onto a naked pstn |
| 21:24.46 | Bobcat_1966 | cool, everytime I do and svn update it uncomments this file and I have to dissable it. Is this a bug? |
| 21:25.06 | gursikh | ok Running on DSL line, 6mb each way coming in. I have this card: http://www.voipsupply.com/product_info.php?manufacturers_id=13&products_id=295&osCsid=109988384f32e43ec765d4ddd921b437 I have 2 boxes running A@H, one primary with the digium card, the other secondary asterisk (failsafe) and runs Iptables. I have a dell 2724 managed switch that's supposed to be doing QOS. |
| 21:25.11 | c4t3l | Un1x: i think that the old russians used to use some-such device |
| 21:25.16 | gursikh | I have 6 Polycom Ip 501 Phones |
| 21:25.34 | file | Bobcat_1966: Asterisk won't modify your modules.conf unless you tell it to |
| 21:25.52 | eKo1 | Un1x: not the phone, the base station |
| 21:25.53 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 21:25.56 | eKo1 | the phone is the receiver |
| 21:26.07 | gursikh | I have one fax that uses the pstn and the pstn also is supposed to be inbound and ring ALL the phones. |
| 21:26.08 | Bobcat_1966 | Interesting maybe its Freepbx doing it..thanks |
| 21:26.22 | gursikh | VOip service from ThinkBright and VoipJet |
| 21:26.51 | gursikh | (thinkbright primary single outbound line) (Voipjet ALL else outbound.) (Pstn only for fax and inbound collect calls) |
| 21:27.07 | eKo1 | gursikh: what is your point? |
| 21:27.21 | Un1x | where can i get oen :p |
| 21:27.44 | gursikh | Sorry eKo1: This is for Waverly360. He was wanting my config, asked that I state it publicly (please scroll up and see problems and questions) |
| 21:28.13 | *** join/#asterisk queuetue (n=scott@toronto-HSE-ppp4122670.sympatico.ca) |
| 21:29.05 | Waverly360 | hmm |
| 21:29.37 | queuetue | Hi. How much can I do from the console? Can I ring an extension and connect it to voicemail? I'm not sure if it's a real shell, or just for monitoring and configuration. |
| 21:30.04 | Waverly360 | gursikh: chances are I might be able to help you some later on...definitely not right at this moment. I'm not promising that I can fix everything though. |
| 21:30.15 | eKo1 | queuetue: just for monitoring mostly |
| 21:30.30 | Waverly360 | gursikh: my areas of expertise are..well..nada in asterisk. I've done it..and I'm getting better at troubleshooting dial plans and the like |
| 21:30.32 | eKo1 | queuetue: you could make an app to do that though |
| 21:30.40 | queuetue | eKo1, Ok. |
| 21:31.11 | _4d4m_ | queuetue: u can dial from the CLI |
| 21:31.11 | Waverly360 | gursikh: If you want, email your configs to me at waverly@datder.net and I'll go through them and see if I can't help solve a few probs. |
| 21:31.36 | _4d4m_ | queuetue: in the formal dial extension@context |
| 21:32.35 | gursikh | Waverly360: I will seriously consider your offfer, especially if I cannot find an "asterisk guru" to do this by tomorrow |
| 21:32.38 | gursikh | thank you |
| 21:32.41 | eKo1 | _4d4m_: do tell |
| 21:32.56 | Waverly360 | gursikh: No charge btw...I need the practice. |
| 21:33.09 | gursikh | Oh, I c . Thanks. |
| 21:33.42 | gursikh | But I really only have a very short time to get this fixed (This office is not in the state where I live, here for only a few days) |
| 21:33.59 | *** join/#asterisk wzlwzl (n=wzlwzl@wsip-70-183-60-181.oc.oc.cox.net) |
| 21:34.04 | wzlwzl | anyone know if its possible to use a polycom soundstation conference phone that is not VOIP-capable with a cisco ATA? |
| 21:34.29 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 21:35.27 | Waverly360 | understood gursikh. I'm actually at a conference..and was checking out the forum while listening to an uninteresting topic :P |
| 21:35.37 | gursikh | hahaha |
| 21:36.14 | [TK]D-Fender | wzlwzl : Sure |
| 21:36.34 | wzlwzl | we'd just lose the functionality like transferring, right? |
| 21:36.50 | [TK]D-Fender | wzlwzl : Nope. ATA should offer all the normal features you'd expect |
| 21:36.57 | wzlwzl | oh |
| 21:36.59 | wzlwzl | neat |
| 21:37.06 | [TK]D-Fender | wzlwzl : All throught hookflash & DTMF signalling. |
| 21:37.11 | devel | ok, so with extensions.conf info in realtime, how can i look at a dialplan? |
| 21:37.21 | wzlwzl | anyone know how good the polycom soundstations are? |
| 21:37.26 | [TK]D-Fender | wzlwzl : I run a SoundStation 2W (wireless) on a Sipura ATA. Works great |
| 21:37.34 | Waverly360 | devel: you mean from the CLI? |
| 21:37.39 | wzlwzl | wireless? |
| 21:37.41 | wzlwzl | sounds sexy. |
| 21:37.42 | [TK]D-Fender | wzlwzl : Yup |
| 21:37.45 | wzlwzl | and useful, too |
| 21:37.51 | devel | aye, Waverly360 |
| 21:37.56 | [TK]D-Fender | wzlwzl : indeed. I specialize in Polycom IP/analog. |
| 21:38.07 | Waverly360 | devel: why is your extensions.conf different from what would be in memory? |
| 21:38.14 | Waverly360 | devel: or is it? |
| 21:38.33 | devel | there is no extensions.conf, because it's in realtime. unless that's wrong, which would explain a lot :) |
| 21:38.53 | Waverly360 | devel: Well...when asterisk is first started..it has to read the info from somewhere |
| 21:39.01 | Waverly360 | devel: either extensions.conf, or a database |
| 21:39.09 | devel | right, realtime == database |
| 21:39.36 | Waverly360 | devel: well..you could connect to the database..I've not dealt with the database backend of asterisk |
| 21:39.43 | Waverly360 | devel: isn't it just mysql? |
| 21:39.50 | devel | aye, it is |
| 21:40.05 | wzlwzl | [TK]D-Fender: nice |
| 21:40.07 | Waverly360 | so #mysql asterisk |
| 21:40.08 | wzlwzl | thx for the info |
| 21:40.09 | Waverly360 | ? |
| 21:40.47 | devel | yeah, if you've not worked with realtime, then you'll be as baffled as i am, trust me. |
| 21:40.49 | gandhijee | there is some stuff about using * with a DB on voip-info |
| 21:41.36 | Waverly360 | devel: well..maybe I'm wrong..but I would assume you connect to the database with the mysql commandline client and just select * from extensions; |
| 21:42.09 | devel | yeah, you're "wrong" as the case were. the records are there, so i need to verify that asterisk is seeing those records. |
| 21:43.55 | devel | Qwell[] just taught me that to see the sip entires, instead of doing "sip show peer 100" you have to do "sip show peer 100 load". so there must (should) be an equivalent for the dialplan. |
| 21:44.10 | devel | ^entires^entries |
| 21:45.52 | devel | i'm looking over the wiki (voip-info.org) again, in case i missed something |
| 21:45.56 | Waverly360 | show dialplan |
| 21:45.58 | Waverly360 | ? |
| 21:46.07 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 21:46.20 | devel | that doesn't work if it's in realtime |
| 21:46.20 | Waverly360 | also |
| 21:46.23 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 21:46.24 | Waverly360 | database show |
| 21:46.52 | devel | that is the on disk dbm database i believe |
| 21:47.14 | Waverly360 | those are the only cli commands I see that might be useful |
| 21:47.20 | wzlwzl | any suggestions on where to get a soundstation? |
| 21:47.38 | Waverly360 | I find it odd that 'show dialplan' won't show the dialplan though..even if it is realtime |
| 21:48.31 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 21:48.55 | Waverly360 | This looks like it might be useful: http://www.voip-info.org/tiki-index.php?page=Asterisk+tips+managing+CID+names |
| 21:49.32 | Waverly360 | hrm...maybe not as helpful as I thought..n/m |
| 21:49.32 | devel | wait, i see what i'm doing wrong. i missed the very first part in the wiki. i _do_ still need extensions.conf, just a shell that points to realtime stuff. |
| 21:49.49 | Waverly360 | even with realtime? |
| 21:49.58 | queuetue | _4d4m_, Like "dial SIP/<extension>@<context>" ? |
| 21:50.16 | queuetue | That doesn't work ... |
| 21:50.18 | *** join/#asterisk IvyUK (n=mark@194.201.148.137) |
| 21:50.26 | devel | switch => Realtime/mycontext@realtime_ext |
| 21:51.32 | _4d4m_ | queuetue: no 'dial exten@context'. it's not the full Dial() you are used to working with from within the dialplan, but it does work. |
| 21:52.10 | *** join/#asterisk adfsadfae (n=chatzill@cpe-66-27-167-15.socal.res.rr.com) |
| 21:52.47 | adfsadfae | can someone help me with mysql/php server not giving permissions |
| 21:53.02 | adfsadfae | to my asterisk server |
| 21:53.03 | Waverly360 | devel: Are you sure about that? I can't seem to find where you're seeing that. |
| 21:53.28 | devel | Waverly360, http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions |
| 21:53.54 | devel | once i added that line in to the context, it worked. however, that still leaves me with my original problem (i.e. i can't see what all extensions are defined in the context) |
| 21:53.55 | Waverly360 | oh..hah...I just saw that...my bad |
| 21:54.18 | Waverly360 | what happens if you do a show dialplan now? |
| 21:54.28 | devel | i see the switch statement, that is all |
| 21:54.51 | Waverly360 | show dialplan simply outputs the config to the cli? |
| 21:54.59 | Waverly360 | the config file I mean |
| 21:55.13 | devel | more or less |
| 21:55.20 | Waverly360 | that's unintuitive |
| 21:55.34 | *** join/#asterisk redondos (n=redondos@190.48.6.244) |
| 21:55.40 | devel | which is why i'm here begging for help, i'm a pretty intuitive guy. |
| 21:55.43 | devel | :) |
| 21:55.52 | Waverly360 | hah..I meant no offense :) |
| 21:55.53 | adfsadfae | help me |
| 21:56.08 | adfsadfae | Ive been working on this for days |
| 21:56.10 | *** join/#asterisk Tall-guy (i=tall-guy@207-195-103-110.regn.static.sasknet.sk.ca) |
| 21:56.20 | adfsadfae | PLZ |
| 21:56.39 | *** join/#asterisk oadaeh (n=jason@las-static-208.57.199.83.mpowercom.net) |
| 21:56.48 | adfsadfae | problem connecting to "67.52.187.22", port 5038: Connection refused at /home/cron/AST_conf_update.pl line 144 |
| 21:56.55 | devel | so now i know it's working, and can "trust" it, but for example if i typo'd a context or something, it may take much longer to catch, where 'show dialplan realtime' (or so) i'd see "oh, that exten doesn't show in that context, i'd better examine the record very carefully" |
| 21:57.23 | Waverly360 | I see. |
| 21:57.50 | adfsadfae | can ne body help |
| 21:57.55 | adfsadfae | ???\\ |
| 21:58.28 | Waverly360 | adfsadfae: You've done all the normal stuff like make sure your firewall/iptables isn't blocking that port on that box? |
| 21:58.50 | adfsadfae | its on the outside |
| 21:59.00 | Waverly360 | on the outside...of what? |
| 21:59.12 | adfsadfae | of the router |
| 21:59.58 | Waverly360 | well..do you have access to the box? linux I assume? |
| 22:00.09 | adfsadfae | yeah |
| 22:00.24 | adfsadfae | what do you mean by access |
| 22:01.16 | adfsadfae | Im on a public IP |
| 22:02.03 | adfsadfae | anybody still here? |
| 22:02.03 | eKo1 | adfsadfae: First question: Is your firewall turned off? |
| 22:02.05 | Waverly360 | can you describe your setup in a bit more detail? where are you? what's between you and the box? |
| 22:02.28 | adfsadfae | its a modem/router with nat turned off |
| 22:02.57 | *** join/#asterisk viler (i=1000@200.114.70.228) |
| 22:02.59 | adfsadfae | im accessing it from within a router |
| 22:03.10 | queuetue | How far behind (time and featurewise) is 1.0.9? |
| 22:03.19 | adfsadfae | but the 2 servers are outside |
| 22:03.38 | *** join/#asterisk adorah (n=Administ@87.68.169.196.cable.012.net.il) |
| 22:03.45 | adfsadfae | on public IP |
| 22:03.47 | Waverly360 | so you're trying to get from one outside server to the other server on the public internet? |
| 22:04.06 | adfsadfae | im trying to make the 2 servers communicate |
| 22:04.17 | adfsadfae | 1 is a mysql/php server |
| 22:04.41 | adfsadfae | the other one is asterisks/vicidial server |
| 22:04.56 | Waverly360 | does the mysql/php server have iptables turned on? |
| 22:05.22 | adfsadfae | how do i do that |
| 22:05.40 | adfsadfae | sorry THis is my first time working with linux |
| 22:05.57 | Waverly360 | well..here's the thing...iptables is your firewall..if you turn off iptables..you leave the server wide open to attacks |
| 22:06.07 | *** join/#asterisk Hughes (n=Hughes@209-221-212-016.qnet.com) |
| 22:06.07 | redondos | http://pastebin.lugmen.org.ar/169 |
| 22:06.12 | redondos | Can you please look at that? |
| 22:06.22 | adfsadfae | im using slackware 10.2 |
| 22:06.25 | *** part/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
| 22:06.34 | Waverly360 | since that server is on the internet..you need to keep iptables on..but that particular port needs to be opened |
| 22:06.35 | adfsadfae | how would i access iptables |
| 22:06.41 | Waverly360 | I can't walk you through it |
| 22:06.45 | redondos | I have an E200p card and exten => 1120,1,Dial(Zap/g1/4397070,60,m) |
| 22:06.51 | Waverly360 | you really need to google iptables...and the commands that go with it |
| 22:06.58 | Waverly360 | or find an iptables guru on here. |
| 22:06.58 | adfsadfae | just let me know the name of the file |
| 22:07.07 | Waverly360 | well...it depends on how it's setup |
| 22:07.11 | Waverly360 | man iptables |
| 22:07.32 | Waverly360 | I think it might be /etc/sysconfig/iptables/ |
| 22:07.38 | Waverly360 | but I'm not positive..it's been awhile |
| 22:08.09 | Tall-guy | can someone tell me what "ztspeed" does? |
| 22:08.26 | *** join/#asterisk mitcheloc (n=mitchelo@69-167-145-62.lmdaca.adelphia.net) |
| 22:08.34 | Hughes | Hey all. Any thoughts on why I'd get no audio between two local SIP phones when NAT's not involved? |
| 22:08.37 | Waverly360 | adfsadfae: if you're a linux beginner, be careful with your iptables config..you don't wanna leave it open to attacks from hackers. |
| 22:08.57 | adfsadfae | lemme take a look at it real quick |
| 22:08.59 | Tall-guy | hughes: are you sipping thru asterisk, or sipping via ip directly to phones? |
| 22:09.27 | Hughes | I'm registering with asterisk and using an extension to call the other phone. |
| 22:09.36 | Tall-guy | hughes: codec? |
| 22:09.39 | adfsadfae | is it the sbin/iptables |
| 22:09.45 | Hughes | None defined in sip.conf |
| 22:10.08 | Hughes | I believe both devices are set to u-law first. |
| 22:10.09 | *** join/#asterisk mitcheloc (n=mitchelo@69-167-145-62.lmdaca.adelphia.net) |
| 22:10.58 | Tall-guy | what kinda sip devices? |
| 22:11.18 | Hughes | a grandstream 100 and a sipura spa2000 |
| 22:12.26 | Tall-guy | hughes: try doing something like calling asterisk voicemail, and leaving a message (and listening to it), from each device in turn...... |
| 22:12.43 | *** join/#asterisk Shark_y (n=paoloc@adsl-ull-206-38.46-151.net24.it) |
| 22:12.58 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 22:13.00 | Hughes | k |
| 22:13.34 | redondos | Question: I have an E1 line and an E200P card. All of a sudden I can only dial toll-free numbers. The console says "== Everyone is busy/congested at this time (1:0/0/1)". Is it safe to assume it is my provider's fault and not Asterisk's? |
| 22:13.41 | Dr-Linux | Qwell[]: Qwell around? |
| 22:13.46 | Qwell[] | nope |
| 22:13.50 | redondos | Well, not toll-free but the phone company's numbers. |
| 22:14.14 | Dr-Linux | Qwell[]: i got a reply from Red Hat |
| 22:14.25 | Qwell[] | Dr-Linux: good? |
| 22:14.28 | Qwell[] | bad? |
| 22:14.28 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 22:14.44 | Dr-Linux | Qwell[]: you tell me after seeing |
| 22:14.50 | Qwell[] | ok, fwd it |
| 22:14.55 | Dr-Linux | lemme forward you |
| 22:15.08 | Dr-Linux | Qwell[]: email? again |
| 22:15.20 | Qwell[] | msg |
| 22:15.24 | Dr-Linux | ko |
| 22:16.16 | Dr-Linux | Qwell[]: sent |
| 22:16.17 | puzzled | hi |
| 22:16.19 | Dr-Linux | along my reply |
| 22:16.24 | Qwell[] | ok |
| 22:16.37 | redondos | Anyone? |
| 22:17.04 | Qwell[] | Dr-Linux: slow mail server :p |
| 22:17.18 | Dr-Linux | Qwell[]: mine? |
| 22:17.26 | Qwell[] | yours or mine, heh |
| 22:17.26 | Shark_y | guys, a little help, why is * delaying of at least one ring a call that arrives on a tdm400p ? |
| 22:17.46 | Dr-Linux | mine looks fine |
| 22:18.11 | Qwell[] | Your last one came really fast |
| 22:18.15 | Tall-guy | sharky_ look for a "wait" in your dialplan |
| 22:18.29 | Dr-Linux | Qwell[]: that i sent from gmail account |
| 22:18.35 | Qwell[] | yeah |
| 22:18.37 | Dr-Linux | this one is from admin@redhat.pk :P |
| 22:18.39 | Qwell[] | ahh |
| 22:19.06 | Dr-Linux | Qwell[]: do you know how they found my email address? |
| 22:19.08 | Qwell[] | Dr-Linux: It has to stop through US customs |
| 22:19.15 | Dr-Linux | Qwell[]: they use feedback on my site |
| 22:19.18 | *** join/#asterisk oadaeh (n=jason@las-static-208.57.199.83.mpowercom.net) |
| 22:19.20 | Qwell[] | haha |
| 22:19.20 | *** join/#asterisk Vulture- (n=Vulture@223.176.119.70.cfl.res.rr.com) |
| 22:19.30 | Qwell[] | usually they'll just get the records from the registrar |
| 22:19.44 | Dr-Linux | and feedback comes to gmail, but i reply them from redhat.pk |
| 22:20.03 | Dr-Linux | Qwell[]: no the records is something else :P |
| 22:20.40 | Qwell[] | Dr-Linux: I might have your server blacklisted :P |
| 22:20.42 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.173) |
| 22:20.44 | Qwell[] | try sending it from gmail? |
| 22:21.10 | Shark_y | Tall-guy could be a wait inserted to try to get a CID? |
| 22:21.26 | Dr-Linux | Qwell[]: okey lemme do |
| 22:21.51 | *** join/#asterisk Sedorox (i=sedorox@smartserv/cna/Sedorox) |
| 22:22.03 | *** join/#asterisk svemuri (n=svemuri@c-24-98-122-69.hsd1.ga.comcast.net) |
| 22:22.49 | Waverly360 | battery dying..later |
| 22:24.24 | Qwell[] | Dr-Linux: slow slow slow :P |
| 22:25.30 | Dr-Linux | Qwell[]: gmail is not opening .. trying ...... |
| 22:26.54 | *** join/#asterisk oadaeh (n=jason@las-static-208.57.199.83.mpowercom.net) |
| 22:27.15 | *** join/#asterisk KranZ (n=user@sme.bestline.net) |
| 22:28.32 | Dr-Linux | Qwell[]: sent from gmail |
| 22:28.37 | Qwell[] | there it is |
| 22:28.38 | *** part/#asterisk viler (i=1000@200.114.70.228) |
| 22:29.24 | *** join/#asterisk mivck (i=1000@200.114.70.228) |
| 22:29.27 | Qwell[] | Dr-Linux: want my honest opinion? |
| 22:29.38 | Dr-Linux | Qwell[]: sure |
| 22:30.00 | Qwell[] | give it to them, and let them reimburse you |
| 22:30.07 | Qwell[] | (not in that order though) |
| 22:30.34 | Dr-Linux | in the order? what do you mean? |
| 22:30.57 | Qwell[] | say you'll give it to them, and after you get the check, let them have it |
| 22:31.08 | *** part/#asterisk bpiper (n=bpiper@70.159.49.40) |
| 22:31.17 | Qwell[] | just say "it cost me exactly this much to register it, and blah, blah, blah" |
| 22:31.27 | Qwell[] | "Here is my mailing address, etc" |
| 22:31.37 | Tall-guy | guys, if I have a TDM400P and a X100P on the same interrupt, should I look to that as the cause of zaptel voice quality issues, even if the cards show no IRQ issues with "zttool" |
| 22:31.47 | Dr-Linux | Qwell[]: you mean only $14 ? |
| 22:31.56 | Qwell[] | Dr-Linux: pretty much ;/ |
| 22:32.02 | Dr-Linux | what about this time i spent and hosting .. and blah blah? |
| 22:32.45 | Qwell[] | Dr-Linux: I honestly don't think they'll pay for time you spent on it |
| 22:32.52 | Qwell[] | BUT! |
| 22:33.06 | Qwell[] | tell them to send you some goodies :P |
| 22:33.15 | Dr-Linux | Qwell[]: goodies? :S |
| 22:33.20 | Qwell[] | pens, shirts, etc :p |
| 22:33.24 | Dr-Linux | Qwell[]: can i have some bucks from them? |
| 22:33.34 | Qwell[] | I doubt they'll give you more than what you actually paid |
| 22:33.45 | Qwell[] | Dr-Linux: How long have you had the domain? |
| 22:34.03 | Dr-Linux | Qwell[]: 2 months |
| 22:34.10 | Qwell[] | yeah, you won't get much |
| 22:34.53 | Tall-guy | ...anyone????? |
| 22:34.56 | Dr-Linux | hhmm.. :S |
| 22:35.03 | Dr-Linux | Qwell[]: so no way? |
| 22:35.13 | Qwell[] | Dr-Linux: well, you can always ask, but... |
| 22:37.02 | Dr-Linux | Qwell[]: i didnt paid myself for this domain |
| 22:37.11 | Dr-Linux | someone else paid and i got here |
| 22:37.17 | Dr-Linux | maybe i'll give to someone else |
| 22:37.24 | Dr-Linux | and he/she will handle |
| 22:37.53 | Hughes | Tall-Guy: Both phones can leave voicemail just fine. They appear to talk to asterisk okay but when they call each other, no audio... |
| 22:38.00 | Dr-Linux | Qwell[]: i will leave the domain, but i'll never give them with $14 , that's bad for my efforts |
| 22:38.16 | Tall-guy | hughes: any errors on asterisk console when calls are placed? (between phones?) |
| 22:38.58 | Hughes | Nope. |
| 22:39.33 | Tall-guy | hughes: got the asterisk started with -vvvvvgc? |
| 22:39.59 | Hughes | Just the vvvv hang on. |
| 22:40.15 | *** join/#asterisk oadaeh (n=jason@las-static-208.57.199.83.mpowercom.net) |
| 22:40.28 | Hughes | still no errors logging. |
| 22:41.13 | Tall-guy | hughes: if you do a "sip show peers" on the console....are they at the "Expected" ports? |
| 22:41.48 | Hughes | yup. all 5060 |
| 22:42.05 | Hughes | with no NAT |
| 22:42.18 | Tall-guy | beats the hell outta me :) |
| 22:42.31 | Hughes | Isn't there a way to force asterisk to handle/convert all audio? |
| 22:42.33 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.175) |
| 22:42.46 | Hughes | Seems like I read it yeaterday but can't remember where. |
| 22:44.04 | Tall-guy | I want to pull out a bad Zaptel card that I don't happen to be using...but I don't want to adjust my dialplan with Zap channel numbering, is there a way to "reserve" this channel so I can keep my channel numbering scheme the same? |
| 22:46.27 | Zodiacal | anyone know why features.conf xfersound = beep doesn't beep? |
| 22:49.11 | *** join/#asterisk Blafasel (n=bpodszun@pd95b71ae.dip0.t-ipconnect.de) |
| 22:49.29 | Qwell[] | Zodiacal: just a guess, but do you have a beep sound? |
| 22:49.52 | Tall-guy | What is the weight of an unladen swallow? |
| 22:50.08 | Blafasel | Hi.. I'm struggling to understand if I need the zap_ata.conf for a chan_ss7 setup - and if not: How can I try to fix issues with lots of echos? |
| 22:50.08 | Zodiacal | qwell yeah beep.gsm |
| 22:50.51 | *** join/#asterisk caloi (n=caloi@204.250.115.224) |
| 22:51.08 | hads | African or European? |
| 22:51.16 | *** part/#asterisk caloi (n=caloi@204.250.115.224) |
| 22:51.27 | redondos | I think my lines don't work because of my provider. Can anyone please help me make sure? |
| 22:51.37 | Qwell[] | redondos: pick up a phone, dial a number |
| 22:51.43 | Qwell[] | if it fails, it's the provider |
| 22:51.51 | redondos | It's not that simple. |
| 22:51.55 | Qwell[] | the phone, obviously, would be plugged directly into the line |
| 22:51.56 | [hC] | Qwell: but.. what if it works? |
| 22:51.57 | [hC] | :) |
| 22:52.02 | Qwell[] | [hC]: Then * is hosed :P |
| 22:52.31 | redondos | I can call my server, but the server cannot call anything other than some free numbers. |
| 22:52.38 | redondos | Such as '110' (phone directory) |
| 22:53.13 | redondos | It says: == Everyone is busy/congested at this time (1:0/0/1) |
| 22:53.32 | redondos | What's this about? -> -- Requested transfer capability: 0x00 - SPEECH |
| 22:54.23 | *** join/#asterisk caloi (n=caloi@204.250.115.224) |
| 22:54.44 | redondos | Hello? |
| 22:54.52 | redondos | Nobody knows about that 'SPEECH' thing? |
| 22:55.22 | Qwell[] | 'speech' is what it's called when a person makes comprehendable noises with their vocal chords |
| 22:55.30 | Tall-guy | hads :) |
| 22:55.42 | hads | :) |
| 22:55.50 | Qwell[] | oftentimes, another party to the "conversation" will also respond, with 'speech' |
| 22:56.19 | Qwell[] | occasionally, party A will respond to his own inquiries |
| 22:56.37 | Qwell[] | and most of the time, said party will be prescribed 'drugs' to 'fix' the situation |
| 22:57.41 | redondos | 'ok', can you try to help me now? |
| 22:57.50 | *** join/#asterisk niZon (n=ilt@S010600080db4ab60.wp.shawcable.net) |
| 22:57.54 | redondos | Please. I just want to make sure it's them. |
| 22:58.06 | Qwell[] | redondos: Is this analog? |
| 22:58.10 | redondos | No. |
| 22:58.15 | redondos | euroisdn |
| 22:58.16 | Qwell[] | then it's them |
| 22:58.51 | Qwell[] | Have you called them? |
| 22:59.07 | redondos | They claim everything is fine, but it's not the first time they screw up. |
| 22:59.15 | Qwell[] | get a new provider.. |
| 22:59.21 | redondos | Yes. |
| 22:59.26 | redondos | There's only 2 providers here. |
| 22:59.33 | redondos | And the second one is really expensive -> I now see why. |
| 23:01.17 | redondos | Qwell[]: You think it can't be a problem with zaptel or zapata.conf? |
| 23:01.31 | Qwell[] | it works for some numbers? I highly doubt it |
| 23:02.36 | redondos | Works for just one number AFAItried. |
| 23:03.22 | redondos | Two numbers: 110 and 112. |
| 23:03.33 | redondos | Basically free numbers answered by the providers PBX. |
| 23:04.16 | Qwell[] | yeah, it's them... |
| 23:05.04 | redondos | Ok, calling them tomorrow. They don't even have 24x7 support. |
| 23:06.04 | *** join/#asterisk hohum (n=dcorbe@69-175-203-11.chvlva.adelphia.net) |
| 23:09.52 | *** join/#asterisk Delta239 (n=adfadsf@200.124.18.171) |
| 23:10.20 | Delta239 | hey where is the place in extensions.conf where i can put the caller id to show to people? |
| 23:14.53 | Blafasel | Delta239: You can Set(CALLERID()) in there.. |
| 23:18.04 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 23:20.00 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 23:21.14 | *** join/#asterisk MikeJ (n=vircuser@204.250.115.210) |
| 23:21.57 | *** join/#asterisk rene- (n=rene-@gea-gye-internet.telconet.net) |
| 23:22.06 | rene- | hey |
| 23:23.42 | *** join/#asterisk ManxPower (i=ewieling@79.sub-70-210-152.myvzw.com) |
| 23:24.38 | rene- | i ran into a problem today, it also happened yesterday and once one time ago,, while using asterisk with softphones namely SJPhone, in an ACD agents/queues setup asterisk would crash while a very loud static noise would be heard at one of the softphones, this is while using the manager interfase .. somewhat heavily... it has happened to me within an hour or so... what could it be?? |
| 23:25.09 | rene- | i wonder if the softphone is what is causing it? |
| 23:25.37 | ManxPower | ~ecfo |
| 23:25.43 | jbot | Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as "screeching", "feedback", "static", or other useless terms. If users report "static" on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. |
| 23:26.05 | ManxPower | rene-, run a backtrace on the .core file and file it on bugs.digium.com |
| 23:26.17 | rene- | that would be pretty much my system: pri + sip |
| 23:26.29 | ManxPower | See /path/to/src/asterisk/docs for the README.backtrace (I think it is called that) |
| 23:26.43 | rene- | thanks you read my mind, i do need instructions for doing a backtrace |
| 23:27.23 | rene- | cool |
| 23:29.19 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 23:32.26 | rene- | Manx, would just disabling echo cancelling on my zapata.conf do any good? |
| 23:33.44 | *** join/#asterisk beyond (n=beyond@200.192.160.100) |
| 23:37.30 | *** join/#asterisk fgwaller (n=frank@65.105.5.40.ptr.us.xo.net) |
| 23:41.31 | *** join/#asterisk iq (n=IQ@unaffiliated/iq) |
| 23:43.31 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 23:44.09 | znoG | OT: anyone use DIDX? |
| 23:44.44 | *** join/#asterisk bkw_ (n=brian@asterisk/friend-and-developer/bkw) |
| 23:45.02 | robl^ | NEXT!!! |
| 23:45.06 | *** join/#asterisk glm2k (n=glm@rrcs-24-199-11-46.west.biz.rr.com) |
| 23:45.30 | *** join/#asterisk rene- (n=rene-@gea-gye-internet.telconet.net) |
| 23:46.32 | *** join/#asterisk guillecabeza (n=elguille@r200-40-68-22.ae-static.anteldata.net.uy) |
| 23:46.54 | *** part/#asterisk guillecabeza (n=elguille@r200-40-68-22.ae-static.anteldata.net.uy) |
| 23:48.50 | *** part/#asterisk mountainm2k (n=mountain@cbit-98.bullseye9.com) |
| 23:48.52 | *** join/#asterisk watchy2 (n=wiit@h236.176.255.206.cable.cmdn.cablelynx.com) |
| 23:49.07 | watchy2 | i tried making my tivo call from a sipura ata but it dont work :( |
| 23:51.23 | FaithX | I think IVR is failing to recognize when people hangup... consequently the line is help open and engaged 1. How can I hang up the line withouth restarting 2. Is this a know bug I can fix |
| 23:51.26 | redondos | Noone will help unless you provide all the necessary technical information. |
| 23:51.45 | redondos | FaithX: Is it analog? |
| 23:51.50 | FaithX | yes |
| 23:51.56 | redondos | Well, uhm, there you go. |
| 23:52.01 | FaithX | ? |
| 23:52.10 | redondos | I haven got much experience, but I always had that problem with x100p |
| 23:52.14 | FaithX | It is a TDM4000 |
| 23:52.23 | redondos | Whatever. Analog sucks., |
| 23:52.33 | FaithX | with FXS ports |
| 23:52.36 | redondos | But hey, stay around, someone might be able to help, I'm sure,. |
| 23:55.00 | znoG | FaithX: have you got busydetect on? |
| 23:55.13 | znoG | FaithX: it happens from time to time that my zaptel card doesn't detect a hangup |
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