| 00:03.20 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 00:04.21 | CrashHD | where can a sangnoma a104d be purchased? |
| 00:05.30 | smackus | the message I get on the cli side of this when i dial 112 is "got sip respons 302 "moved temporarily" back from 10.0.0.203" is that consistant with a phone that has been forwarded? |
| 00:07.05 | smackus | must be... thanks |
| 00:08.01 | djPepse | Anyone have stanaphone configured for incoming calls? |
| 00:09.07 | justinu | smackus: you might be able to reset the phone if you can modify the config file and send it a notify check-cfg |
| 00:09.15 | djPepse | i want to know how to set it in extensions.conf, since _. is not a good idea |
| 00:14.24 | anthm | CrashHD try pbxeq.com |
| 00:17.12 | CrashHD | ok I'll take a look |
| 00:18.57 | CrashHD | thanks |
| 00:18.59 | CrashHD | they any good? |
| 00:20.10 | anthm | ya |
| 00:22.38 | djPepse | hm. this is weird. i can finally connect into asterisk over iax with idefisk, and I can dial out on the fxo, but I can't dial out over fwd |
| 00:28.16 | *** join/#asterisk Carp1 (i=Carp1@ip-204-97-151-235.modem.logical.net) |
| 00:28.20 | Carp1 | Hey all! |
| 00:28.25 | *** join/#asterisk dant (n=dan@2001:618:400:3f8d:204:76ff:fe1e:585e) |
| 00:28.46 | Carp1 | I am connecting to NuFone using IAX. Is there a CLI command that will tell me if I'm connected? |
| 00:29.37 | *** join/#asterisk pdthome (n=pdthome@c-68-53-40-50.hsd1.tn.comcast.net) |
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| 00:32.05 | *** part/#asterisk diclophis (n=diclophi@65.203.37.58) |
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| 00:36.00 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
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| 00:43.26 | Carp1 | I am connecting to NuFone using IAX. Is there a CLI command that will tell me if I'm connected? |
| 00:43.42 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:47.35 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 00:48.25 | drray | iax2 show peers? |
| 00:48.51 | *** join/#asterisk Jaxxan (n=jaxxan@202.70.125.60) |
| 00:48.52 | *** join/#asterisk phalacee (n=Sunforge@202.3.110.65) |
| 00:49.51 | Carp1 | pbx*CLI> iax2 show peers |
| 00:49.52 | Carp1 | Name/Username Host Mask Port Status |
| 00:49.52 | Carp1 | NuFone 66.225.202.72 (S) 255.255.255.255 4569 Unmonitored |
| 00:49.52 | Carp1 | 1 iax2 peers [0 online, 0 offline, 1 unmonitored] |
| 00:49.55 | Carp1 | menas I'm connected? |
| 00:51.23 | pdthome | iax2 show registry |
| 00:55.49 | TripleFFFF | yes |
| 00:56.26 | TripleFFFF | you can add a qualify =2000 to to see the ping time responce.. but means if you go over it disconenct form host |
| 00:56.30 | TripleFFFF | i dont use htat crap |
| 00:56.59 | *** join/#asterisk MoutaPT (n=MoutaPT@a83-132-239-109.cpe.netcabo.pt) |
| 00:57.55 | MoutaPT | I've Wav 44khz 16 bit stereo file, to upload it to asterisk should convert to 8khz 8bit mono? |
| 00:58.09 | MoutaPT | to work as native sound file |
| 00:58.10 | MoutaPT | ? |
| 00:58.14 | drray | gsm |
| 00:58.44 | MoutaPT | could you advice me sox command line for this downsample? |
| 00:59.57 | Carp1 | you tell me its connected but nowthing shows in the screen in CLI when I dial |
| 01:00.08 | Carp1 | and I get a quick busy signal then disconnect |
| 01:03.06 | *** join/#asterisk beyond (n=beyond@201-0-103-146.dsl.telesp.net.br) |
| 01:03.41 | Carp1 | Oh |
| 01:03.47 | Carp1 | Maybe I have my account settings wrong |
| 01:03.56 | Carp1 | Does anyone have a NuFone acct here? |
| 01:07.13 | Bullseye_Network | Is Nufone working again? |
| 01:07.41 | *** join/#asterisk Brijn (n=bas@S0106004063c0fa1f.vn.shawcable.net) |
| 01:07.52 | Brijn | Hello all |
| 01:10.22 | *** join/#asterisk knarfly (n=Knarf980@c-69-180-98-189.hsd1.fl.comcast.net) |
| 01:13.31 | Carp1 | Yes |
| 01:13.52 | Brijn | Our current phones (Panasonic) show a led that blinks for the lines your have programmed under the quick-dial buttons when the line is in use.. Does * provide similar functionality? And what SIP phones do?? If I have to trust the image, it seems that theAastra phone have the buttons/led? |
| 01:15.52 | *** join/#asterisk Jaxxan (n=jaxxan@202.70.125.60) |
| 01:17.43 | *** join/#asterisk userdefined (n=jross@cpe-24-169-142-23.rochester.res.rr.com) |
| 01:20.11 | SplasPood | Brijn: search for asterisk presence on www.voip-info.org |
| 01:20.45 | foo | Is there anyway I could set up asterisk on my home phone line to basically answer the phone, prompt the caller with 4 options, and each option would transfer them to a different cell number? But, instead of going out on analog, it would go out through IP to a number or something? Hm |
| 01:22.39 | Brijn | SplasPood: Tx, I didn't know the proper keyword, will have a look |
| 01:25.06 | SplasPood | Brijn: I use polycom phones /w 'hint' extensions (you can search for that too) |
| 01:26.24 | SplasPood | I have a client who wants to provide people with the ability to record a voicemail message, then be given the option to cancel, re-record, and mark as urgent.... Does anything like this exist? |
| 01:27.17 | SplasPood | foo: yes, it's called asterisk :) |
| 01:27.40 | drray | note to self, don't help anyone in email |
| 01:27.59 | foo | SplasPood: haha. I'm looking into maybe setting things up I home. I have 1 analog phone line to work with. hmm. Thanks :) |
| 01:28.08 | foo | Will the calling out that is done over IP cost? |
| 01:30.17 | SplasPood | if you're calling cell phones then yes, most likely |
| 01:30.24 | foo | How much are we talking a month? |
| 01:30.33 | SplasPood | how much are you calling? :) |
| 01:30.38 | foo | ah, I see. |
| 01:30.44 | foo | Hmm. |
| 01:30.53 | SplasPood | lots of providers don't charge a monthly fee for just outbound |
| 01:30.58 | SplasPood | you can pay per minute |
| 01:31.02 | foo | ah, I see |
| 01:31.06 | SplasPood | or |
| 01:31.16 | SplasPood | a lot of people will offer flat rate domestic calling |
| 01:31.21 | SplasPood | like $20-$40/mo |
| 01:31.22 | SplasPood | USD |
| 01:31.25 | SplasPood | for domestic US |
| 01:31.31 | SplasPood | I cannot speak of deals in other countries |
| 01:32.05 | foo | I'm in Los Angeles, CA. |
| 01:32.11 | Brijn | SplasPood: Do I have to enable hinst somewhere? Version 1.2.9.1, show application hint |
| 01:32.13 | foo | Hm. I see |
| 01:32.21 | Brijn | Your application(s) is (are) not registered |
| 01:32.27 | Bullseye_Network | for the mixmonitor would this be correct MIXMONITOR(/var/recordings|b|sh /usr/local/process.sh) |
| 01:32.28 | SplasPood | Brijn: hint is a type of extension.. search voip-info for 'hint |
| 01:33.35 | foo | What are some of the main advantages to moving of moving to VoIP? |
| 01:33.40 | SplasPood | foo: I use Voicepulse Connect for inbound.. they used to be flaky, but I think they've cleaned up their act somewhat... For outbound I've used at various times voxee, broadvoice, umm... nufone.. but I wouldn't recommend that to anyone anymore... |
| 01:34.00 | foo | thanks. I had a local guy just recommend broadvoice. |
| 01:34.06 | mitcheloc | foo, i'm in the oc ;) |
| 01:34.34 | foo | mitcheloc: Nice! |
| 01:34.41 | mitcheloc | foo, there is a user's group in oc that you are welcome to come to |
| 01:34.42 | foo | mitcheloc: What kind of voice set up do you have? |
| 01:34.46 | foo | Hmmm. |
| 01:34.49 | mitcheloc | lots heh |
| 01:34.49 | foo | Where at? You guys have a site? |
| 01:35.47 | SplasPood | I wonder.. is there a users group in NYC? |
| 01:36.18 | *** join/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx) |
| 01:36.23 | Brijn | SplasPood: I set it exactly as in the example but get: |
| 01:36.25 | mitcheloc | Kerry_G runs it, not sure what the URL is... |
| 01:36.37 | Brijn | Hmm, is it ok to paste lines or shoul I pastebin it? |
| 01:36.45 | SplasPood | Brijn: show me exactly what you have.. if its more than a couple lines, use pastebin |
| 01:36.49 | SplasPood | ~pb |
| 01:36.54 | jbot | it has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 01:37.22 | mitcheloc | foo, found it -- http://www.socalasteriskug.org/ |
| 01:37.39 | rene- | does anybody know the irc handle of steve totaro |
| 01:37.55 | *** join/#asterisk flujan (i=flujan@201-42-102-87.dsl.telesp.net.br) |
| 01:40.39 | Brijn | SplasPood: http://pastebin.ca/75327 |
| 01:40.52 | Brijn | Restarted asterisk already, I thought a reload might not be enough |
| 01:41.20 | foo | mitcheloc: haha, thanks! |
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| 01:45.27 | Brijn | SplasPood: I'm blind, forget it :) |
| 01:46.01 | SplasPood | haha |
| 01:46.04 | SplasPood | was just about to paste |
| 01:46.09 | SplasPood | I got distracted |
| 01:46.56 | SplasPood | Brijn: fyi, with the polycom 601s you need 1.6.6 to handle the sidecar and more than 8 monitored "lines" |
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| 02:01.59 | trelane | anyone happen to know a location for default logins/password recovery for adtran kit? I've got a TA616 I can't get into |
| 02:02.09 | *** part/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
| 02:04.56 | *** join/#asterisk edavis10 (n=eric@68-116-84-232.dhcp.trlk.ca.charter.com) |
| 02:05.13 | postel | trelane: http://www.governmentsecurity.org/articles/DefaultLoginsandPasswordsforNetworkedDevices.php |
| 02:05.53 | obiwanmikenolte | http://artofhacking.com/etc/passwd-adtran.htm |
| 02:07.45 | postel | adtran or ADTRAN is pretty much standard on almost all adtran models |
| 02:08.49 | drray | does that include adit? |
| 02:10.46 | drray | no clearly not |
| 02:10.47 | drray | pardon me |
| 02:11.53 | obiwanmikenolte | You wish. You're the only one typing, so now EVERYONE knows that you're wrong |
| 02:12.11 | obiwanmikenolte | Are all of these users bots? |
| 02:12.16 | Qwell | yes |
| 02:12.35 | obiwanmikenolte | Figures. Someone just set them up to make them feel like they're in the cool channel |
| 02:12.56 | obiwanmikenolte | I would NEVER consider that. |
| 02:13.26 | *** join/#asterisk phalacee (n=Sunforge@202.3.110.65) |
| 02:13.44 | drray | I'm a self correcting bot |
| 02:13.54 | obiwanmikenolte | I'm a CAPSLOCK bot |
| 02:14.08 | drray | numnuts er, numlock |
| 02:14.13 | drray | :) |
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| 02:17.37 | mitcheloc | net split? |
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| 02:18.26 | netoguy | does anyone know what the PRIEXCLUSIVE setting is used for in zapata.conf? |
| 02:19.34 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 02:19.44 | obiwanmikenolte | So, good bots of the #asterisk IRC: I'm trying to figure out some way to indicate DND status on a snom to all of the other happy snoms. I have two snoms, and I've set up their LED indicators to subscribe to each other's SIP channels. When I call one from the other, the LED flashes while ringing, then it goes solid once the call is connected. What I'd like to do is have a phone notify Asterisk when an extension hits the DND button then make a |
| 02:21.37 | obiwanmikenolte | netoguy: http://wiki.sangoma.com/ast-original-zapata |
| 02:22.37 | *** join/#asterisk niZon (n=ilt@S010600080db4ab60.wp.shawcable.net) |
| 02:24.08 | netoguy | obiwanmikenolte: thanks for the link, but I've already read that little comment that is in the zapata file |
| 02:24.25 | netoguy | i don't know much about PRI so I don't know what the comment means |
| 02:24.47 | netoguy | do you know any more about it? |
| 02:24.50 | obiwanmikenolte | If anyone can even point me to some solid documentation for sip_notify.conf, I'd be very grateful. I tried messing with it, and I can send notifications using sip notify from the CLI, which is cool, but I'm really just guessing, based on a sip debug of a call, on what I'm supposed to set things to. I'm really guessing about the whole thing, and the phones know it. They keep freaking out, and I have to reboot them, and every time I do, a litt |
| 02:25.42 | *** join/#asterisk userdefined (n=jross@cpe-24-169-142-23.rochester.res.rr.com) |
| 02:26.08 | obiwanmikenolte | netoguy: I'm thinking that you'd know if you needed to override the existing channels selection routine. What's the problem that you're having? Are calls colliding? |
| 02:26.15 | userdefined | rob0: you around? i figured out the network issue i was having the other day |
| 02:26.25 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-63-170.cybersurf.com) |
| 02:26.51 | userdefined | turns out that upgrading the firmware on my linksys wrt54g (v5...*shudder*) took care of it |
| 02:27.13 | obiwanmikenolte | With a bat? |
| 02:27.42 | userdefined | heh. i wish. i'm about thisclose to trying the vxworks killer upgrade to see if i can get it to work =) |
| 02:28.18 | Qwell | vxkindaworksaslongasyoudontlookatitwrong |
| 02:30.53 | obiwanmikenolte | netoguy: Are you having a problem? MAybe someone can help. I've used PRIs. Maybe I can build up some karma so that someone will take a stab at my snom dilemma |
| 02:31.07 | obiwanmikenolte | ...not that I'm motivated only by selfishness |
| 02:31.19 | obiwanmikenolte | I like food, too. And candy. |
| 02:33.38 | netoguy | obiwanmikenolte: no i don't have any real problems right now. I'm new to this and we just got our PRI installed and I'm going to be rolling out the system tomorrow.. and I'm just going through everything with a fine tooth comb right now |
| 02:33.53 | obiwanmikenolte | Gotcha. |
| 02:34.01 | obiwanmikenolte | The biggest problem I've had with PRIs is echo |
| 02:34.15 | netoguy | i do have a quick question that you may be able to shed a little light on for me though...about Caller ID Name |
| 02:34.31 | netoguy | am I able to pass the Name outbound to the PSTN using PRI? |
| 02:34.37 | obiwanmikenolte | Probably |
| 02:34.39 | Qwell | netoguy: no |
| 02:34.42 | netoguy | i've read so many conflicting stories |
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| 02:34.49 | Qwell | name it looked up at the far end |
| 02:34.52 | Qwell | is* |
| 02:35.02 | obiwanmikenolte | We have 3 PRIs, and they allow us to set our own CallerID |
| 02:35.07 | Qwell | cidnum |
| 02:35.10 | Qwell | not cidname |
| 02:35.12 | bugz | ~seen mercestes |
| 02:35.25 | jbot | mercestes <n=merceste@69.15.174.114> was last seen on IRC in channel #asterisk, 15d 11h 51m 4s ago, saying: 'on second thought, Hener...don't ask..just take six a day and pray.'. |
| 02:35.25 | obiwanmikenolte | Oh, yeah. Good call |
| 02:35.29 | netoguy | Qwell, thanks, thats what I thought |
| 02:37.19 | bugz | anyone used sangoma cards in kernel 2.6.16+ ? |
| 02:37.28 | netoguy | just a quick FYI though...our PRI is from McLeod, and it is their Dynamic Integrated VoIP service. They use VoIP to our Cisco IAD which then converts it to a PRI for us. Pretty neat setup.. (Just hope it works ok) .. anyway...they did say that its ok for me to send the CID Name out on our PRI. It will get dropped if it goes off their VoIP network, but if it goes to another customer on their VoIP network then the CID name should |
| 02:37.33 | bugz | on a 64 bit machine? |
| 02:37.55 | Qwell | netoguy: Then why did you ask? :P |
| 02:38.17 | bugz | asterisk to pri to iad to voip |
| 02:38.23 | netoguy | ha, well because I beleived them about the VoIP side of things...but I wasn't clear about the PSTN side |
| 02:38.26 | bugz | kind of defeats the purpose of asterisk doesnt it? |
| 02:39.15 | netoguy | bugzs...its better than that...we actually are currently going from a old Comdial system to a Vina Channel bank then into asterisk and to the PRI / IAD / VoIP ... |
| 02:39.24 | netoguy | wish us luck! |
| 02:39.30 | *** join/#asterisk yxa (n=diablo@58.185.90.101) |
| 02:39.37 | bugz | haha good luck then |
| 02:39.51 | bugz | SIP is not something LEC's like to support anyway |
| 02:40.02 | rob0 | userdefined: vxworks :( ... well, at least it's fixed now. |
| 02:40.15 | bugz | they'll sell it to you but if something goes wrong they have to wait for the security admin to get back from vacation to reinstate your account |
| 02:40.28 | netoguy | obiwanmikenolte: you mentioned you had some problems with echo on your PRI... are there any pointers you could give me if I were to run into that problem? |
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| 02:42.31 | obiwanmikenolte | I've found myself using echocancel = yes (though I've been told that I shouldn't, it works), and I used ztmonitor (in the zaptel tar) to see how channels are behaving, then I adjust the txgain and rxgain |
| 02:42.53 | obiwanmikenolte | It sucks, but it's pretty much taken care of the problem |
| 02:43.37 | netoguy | ok, thanks |
| 02:43.38 | obiwanmikenolte | I'm trying to find the Digium page that addresses echo, but I can't seem to |
| 02:44.02 | netoguy | i'm fingers cross we won't have any problem |
| 02:44.26 | netoguy | i have sent a couple test calls and they seemed fine...but it was only 1 or 2 concurrent calls |
| 02:44.29 | bugz | jeezus.. some hacked web server tried to brute force ssh on one of my pbx's that has a really really really bad connection |
| 02:44.34 | obiwanmikenolte | http://kb.digium.com/19/ |
| 02:44.35 | bugz | must be desperate |
| 02:44.44 | netoguy | would i have noticed the problem by then, or is it something that is caused by load? |
| 02:45.51 | obiwanmikenolte | It all depends. There are all sorts of things that could cause it, and all of the problems I've had have been the fun, intermittent ones |
| 02:45.59 | *** join/#asterisk phalacee (n=Sunforge@202.3.110.65) |
| 02:46.21 | obiwanmikenolte | netoguy: Are you guys using QoS? |
| 02:47.16 | netoguy | we are using some real basic stuff thats built into a linux firewall (physical port based priority) |
| 02:47.19 | bugz | how could qos have any effect on PRI echo |
| 02:47.20 | bugz | ? |
| 02:47.41 | obiwanmikenolte | NEtwork latency. |
| 02:48.31 | bugz | voip providers run their own qos internally |
| 02:49.02 | obiwanmikenolte | But shouldn't they also run their own qos internally? |
| 02:49.03 | bugz | ive found that unless you are dealing with large amounts of phones and computers, multiple T1's etc, qos isnt really necessary and can often lead to more problems than it solves |
| 02:49.38 | *** join/#asterisk h0 (n=h0@ool-44c69453.dyn.optonline.net) |
| 02:49.49 | obiwanmikenolte | Yeah, I'm not a big fan, and the whole echo thing really seems like black magic mumbo jumbo to me, but people have said that it helped |
| 02:50.30 | obiwanmikenolte | I tried to eliminate the placebo effect by secretly switching settings without telling anyone, but it seemed like I got more complaints during that time, so I guess it's really doing something |
| 02:51.08 | drray | I've cured echo complaints by turning volume down |
| 02:51.23 | bugz | obiwanmikenolte: that is a problem i face at work. techs will shut down my firewall to troubleshoot something wierd/sporadic and they wind up thinking it fixes things |
| 02:51.47 | obiwanmikenolte | Also, when I set echocancelwhenbridged to yes, people complained less, even though I'm told by everyone that there should already be echo cancellation on either side and that setting it to yes will probably cause more echo |
| 02:51.50 | bugz | then my pbx is sitting there dealing with nessus while snort and everything else goes nuts stopping it |
| 02:53.11 | bugz | i have this pbx registering via iax about once a minute |
| 02:53.17 | bugz | it gets annoying when working on my own cli |
| 02:53.35 | bugz | and there appears to be no configuration value set to do that so often on either machine |
| 02:53.52 | bugz | any ideas? |
| 02:54.34 | obiwanmikenolte | Turn verbosity to 0? Heh |
| 02:55.02 | drray | change it to more than 1 minute? |
| 02:55.09 | obiwanmikenolte | Also astute. |
| 02:55.44 | drray | I got yelled at by a cow-orker because I did not fix things correctly, I made the problem go away |
| 02:56.08 | mitcheloc | drray: ouch what did you do to her/him? |
| 02:56.12 | drray | nothing |
| 02:56.14 | drray | I was done for the day |
| 02:56.21 | obiwanmikenolte | And then they went away? |
| 02:56.23 | drray | he was still reinstalling windows 2000 |
| 02:56.24 | obiwanmikenolte | Brilliant! |
| 02:56.26 | mitcheloc | well, how did you make him/her "go away"? |
| 02:56.31 | mitcheloc | you didn't kill anyone did you? |
| 02:56.35 | drray | no |
| 02:56.39 | drray | well, not over that |
| 02:56.47 | mitcheloc | bah, no sense of humour =P |
| 02:56.54 | drray | :) |
| 02:57.00 | *** join/#asterisk onweald_tim (n=onweald_@c-67-173-213-205.hsd1.tx.comcast.net) |
| 02:57.06 | bugz | 218.82.202.105 |
| 02:57.21 | obiwanmikenolte | Are we hacking 218.82.202.105? |
| 02:57.31 | obiwanmikenolte | Gentlemen, start your nmaps |
| 02:57.46 | bugz | i prefer the term 'agressively defending myself from' |
| 02:58.15 | drray | yay windows share |
| 02:58.23 | bugz | haha... |
| 02:58.31 | drray | I did not lock |
| 02:58.33 | mitcheloc | lol |
| 02:58.34 | drray | er, look |
| 02:58.58 | bugz | you probably wouldnt be the first |
| 02:59.11 | bugz | its trying to log in as 'steve' on my box at the moment |
| 02:59.19 | drray | I remember the first time I mounted a windows share over the internet (using NT) |
| 02:59.22 | drray | that was something |
| 02:59.37 | drray | took 20 minutes |
| 02:59.44 | obiwanmikenolte | And then the problem went away? |
| 02:59.55 | drray | and then he took off his pants |
| 03:00.51 | bugz | i should let nessus loop on them for a few days |
| 03:00.54 | *** part/#asterisk onweald_tim (n=onweald_@c-67-173-213-205.hsd1.tx.comcast.net) |
| 03:01.02 | bugz | until security@chinatelecom.com emails me |
| 03:01.28 | Brijn | :) |
| 03:04.21 | *** join/#asterisk blebleble (n=ble@d149-67-99-160.col.wideopenwest.com) |
| 03:04.35 | bugz | i like how gentoo doesnt include traceroute in the base install |
| 03:05.49 | bugz | this device looks interesting: 218.1.3.2 |
| 03:06.01 | *** part/#asterisk netoguy (n=skelley@ppp-70-129-186-62.dsl.spfdmo.swbell.net) |
| 03:06.03 | obiwanmikenolte | Oh, that's just drray |
| 03:06.10 | drray | I did not look |
| 03:06.13 | obiwanmikenolte | Haha |
| 03:06.13 | drray | I told you |
| 03:06.51 | SplasPood | Is there any way to dynamically change a meetme room's pin? (other than using Realtime) |
| 03:07.07 | bugz | well it doesnt appear to be running asterisk, ho hum. |
| 03:07.56 | bugz | hehe, its a 2k box running ever service windows has to offer |
| 03:08.08 | bugz | no wonder im getting scans and shit from that network 48234 times a day |
| 03:08.20 | bugz | its probably china telecoms "core router" |
| 03:08.36 | bugz | running a pirated, trojaned version of 2k |
| 03:08.48 | obiwanmikenolte | And squid |
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| 03:08.53 | bugz | heh |
| 03:09.27 | *** part/#asterisk Samoied (n=Samoied@201.47.218.152) |
| 03:11.56 | bugz | person: Wu Xiao Li phone: +86-21-63630562 |
| 03:12.02 | bugz | address: Room 805,61 North Si Chuan Road,Shanghai,200085,PRC |
| 03:15.13 | obiwanmikenolte | Stupid Toasted Spam? http://www.toastedspam.com/stupid/disptext/webbiz2o.com_0002 |
| 03:15.31 | obiwanmikenolte | The Internet is a craaaaaazy place |
| 03:15.46 | obiwanmikenolte | But half the cost of Viagra? Hmm.... |
| 03:19.58 | postel | They're mad in china. They have produced counterfeit meat! A couple of months ago six babies died because of counterfeit milk powder that had no nutritional value at all. Electronics and watches was _almost_ ok, but counterfeit food and medicine is madness :/ |
| 03:20.01 | obiwanmikenolte | Well, I know you'll miss me everyone, but I simply must go. You bots keep that snom DND thing in mind (if any of you actually read that novella of a problem up there) |
| 03:24.41 | *** join/#asterisk d-tech (n=dtc@72.245.233.107) |
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| 03:43.26 | Brijn | What's the best/easiest way to lookup an incoming callerid and change callerid(name) based on the result |
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| 03:46.11 | Qwell | lilo: you broke it! |
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| 03:46.31 | smackus | is anyone out there using redhat enterprise 4? |
| 03:47.10 | Brijn | What's the best/easiest way to lookup an incoming callerid and change callerid(name) based on the result |
| 03:47.33 | *** join/#asterisk kimosabe (n=kimosabe@dsl-200-78-71-105.prod-infinitum.com.mx) |
| 03:48.07 | kimosabe | is it tru that u can songure a sipura 3000 and any other sipura fxo fxs without any asterisk server for transparent calls |
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| 03:56.15 | iceyp | hey guys, anyone using mynetphone.com.au and got DTMF working from the PSTN? |
| 03:58.33 | iceyp | anyone here know of a good DID provider in oz? |
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| 04:05.45 | P-NuT | Hey guys.. |
| 04:05.58 | P-NuT | if I want to dial multiple extensions, do I do this? |
| 04:05.59 | P-NuT | exten => s,3,Dial(${PHONE1}&${PHONE2}&${PHONE3},30,Tt) |
| 04:08.12 | SplasPood | yep |
| 04:08.22 | SplasPood | presuming those vars held the proper strings.. |
| 04:08.47 | SplasPood | Brijn: lookup.. where? |
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| 04:43.50 | smackus | I am still getting "Warning[5924]: channel.c:787 channel_find_locked:Avoided initial deadlock for '0x7d3d80' , 10 retries!" |
| 04:43.54 | *** join/#asterisk Jaxxan (n=jaxxan@leone-canopy05.bluelink.as) |
| 04:44.05 | smackus | what the hell does that mean? I have been chasing that now for a week with no success |
| 04:44.32 | NotJohnDavid | go grep "Avoided initial deadlock" in the source |
| 04:44.48 | smackus | in which source? in the asterisk directory? |
| 04:45.02 | smackus | "/usr/src/asterisk-1.2.9"? |
| 04:45.20 | NotJohnDavid | yeah |
| 04:45.29 | NotJohnDavid | it should be in channel.c |
| 04:46.30 | smackus | it seems to be looking really hard for something... does that sound like it is doing what I want? |
| 04:47.51 | NotJohnDavid | well i was going to tell you to figure the source out and see what's happening |
| 04:47.59 | NotJohnDavid | that may not happen though |
| 04:48.33 | smackus | I am also getting another error... just noticed this one. |
| 04:49.22 | smackus | blah, blah, blah... already blocked by thread <number string> in procedure ast_waitfor_nandfds |
| 04:49.29 | smackus | does that mean anything to you/ |
| 04:49.30 | smackus | ? |
| 04:51.03 | smackus | posting some output from cli to pastebin....... |
| 04:51.37 | smackus | http://pastebin.ca/75423 |
| 04:52.05 | smackus | please help... my system has been unstable now for 2 weeks. I have finally reinstalled everything... still getting a ton over errors |
| 04:53.00 | NotJohnDavid | what card are you using |
| 04:53.32 | smackus | TE411P |
| 04:55.11 | NotJohnDavid | i don't really know what the deal with that pseduo channel is but you may want to check it out |
| 04:55.43 | smackus | how do i check it out? |
| 04:55.55 | smackus | that is why i am here.... i have no idea where to go from here. |
| 04:57.01 | NotJohnDavid | you know i'm not really the one to ask. i'm new to asterisk but i'd start with the zaptel.conf file. it's /etc/zaptel.conf I think |
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| 05:03.06 | FuriousGeorge | noticed a wierd thing. when * cant find the cid it pulls the last entry from my zapata.conf |
| 05:05.29 | Brijn | Any SQL heroes here? (and yes, it's * related ;-) |
| 05:05.56 | smackus | ok, what about a message that says "junk at the beginning of frame 00000000" |
| 05:08.06 | userdefined | Brijn: i'm definitely not a sql hero, but fairly handy depending on the issue (and the db platform) |
| 05:09.52 | Brijn | Let me pastebin something, one sec |
| 05:12.48 | iceyp | anyone here using mynetphone.com.au ? |
| 05:14.10 | iceyp | can anyone suggest a cheap sydney australia DDI provider ? |
| 05:14.42 | Brijn | userdefined: Can you have a look at http://pastebin.ca/75432 |
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| 05:19.52 | userdefined | Brijn: which version of mysql ? |
| 05:21.03 | Brijn | userdefined: 4.1.14 |
| 05:21.51 | userdefined | just the cdr.src needs to not be in callerid.cid_number? |
| 05:22.41 | Brijn | would be nice if it can check for both cdr.src and cdr.dst, but just cdr.src is fine as well.. IE just show incoming calls |
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| 05:26.16 | littleRalph | aanyone with nslu2 experience? |
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| 05:35.35 | *** join/#asterisk VeNoMouS_ (n=jj@spam.turnstone.net.nz) |
| 05:36.28 | VeNoMouS_ | omfg it is eye! |
| 05:36.39 | VeNoMouS_ | ltns ppls |
| 05:37.04 | iceyp | VeNoMouS_ |
| 05:37.04 | iceyp | ;) |
| 05:38.52 | Brijn | userdefined: Any luck :) |
| 05:39.14 | userdefined | Brijn: http://pastebin.ca/75444 |
| 05:39.36 | userdefined | tested on mysql5, but iirc 4.1 supports subselects so should work (crosses fingers) |
| 05:40.50 | VeNoMouS_ | hay u guys been having issues with video and eyebeam l8ly? |
| 05:40.52 | userdefined | fwiw, the third one is borked at the 'and' ... each of the statements alone works, joining them breaks it |
| 05:41.00 | VeNoMouS_ | i can see the rtp udp streaming to asterisk |
| 05:41.04 | *** join/#asterisk P-NuT (n=P-Nut@fw.office.unitedip.net.au) |
| 05:41.04 | VeNoMouS_ | but asterisk is ignoring it |
| 05:41.37 | mosty | i'm having trouble with receiving some sip calls, when the person at the destination end picks up they hear nothing, while the person at the originating end continues to hear rings. what could be wrong? both client and server are asterisk machines but the same happens when the client is a sip phone |
| 05:42.11 | VeNoMouS_ | mosty : nat'ing, rtp ports |
| 05:42.15 | VeNoMouS_ | could be a number of things |
| 05:42.24 | *** join/#asterisk defy (n=defy@60-234-234-98.bitstream.orcon.net.nz) |
| 05:42.39 | mosty | VeNoMouS_: can you suggest a search string to plug into google or voip-info.org ? |
| 05:42.57 | VeNoMouS_ | not really, u have a firewall? |
| 05:42.58 | Brijn | userdefined: Super!! Time for me to go to bed now.. But i'll see that I can turn it into something useful next week.. Tx! |
| 05:43.05 | VeNoMouS_ | make sure ure allowing more then just 5060 through |
| 05:43.12 | VeNoMouS_ | the audio rtp is dynamic |
| 05:43.19 | VeNoMouS_ | you can set a port range for it |
| 05:43.40 | mosty | VeNoMouS_: ok. would i be correct in assuming this must be a problem at the receiving server's end? i can make calls to other sip servers without problem |
| 05:44.24 | VeNoMouS_ | well it could be if they dont have stateful established or related type rules |
| 05:44.45 | VeNoMouS_ | the rtp goes via udp |
| 05:45.01 | VeNoMouS_ | or it could be your end |
| 05:45.39 | mosty | VeNoMouS_: well with the same local server setup i can call sip phones on a different sip provider. when i switch the part of my dialplan to dial via this other server, we get the problem |
| 05:46.05 | VeNoMouS_ | yea id say its the remote end |
| 05:47.06 | Netgeeks | anyone off the top of thier head know the file size per minute (specifically speaking voicemail files) for the different codecs? wav, gsm? |
| 05:47.12 | mosty | cool, thanks. i will investigate this part more thoroughly |
| 05:48.10 | VeNoMouS_ | Netgeeks wav is like 100k a min from memory |
| 05:48.13 | VeNoMouS_ | or was it a meg |
| 05:48.14 | VeNoMouS_ | hang on |
| 05:48.29 | VeNoMouS_ | it also depends what codec u are using |
| 05:49.03 | *** join/#asterisk pengyong (n=lala@222.188.141.139) |
| 05:49.57 | VeNoMouS_ | asterisk01:/var/lib/asterisk/monitor# ls -lah 23-04-2006-16:30:20-093797759-099705560.wav |
| 05:49.57 | VeNoMouS_ | -rw-r--r-- 1 root root 795K Apr 23 16:31 23-04-2006-16:30:20-093797759-099705560.wav |
| 05:49.57 | VeNoMouS_ | asterisk01:/var/lib/asterisk/monitor# ./read.pl |
| 05:49.57 | VeNoMouS_ | input is 50.82 seconds long |
| 05:50.03 | VeNoMouS_ | bout a meg a min |
| 05:50.10 | Netgeeks | cool, thanks |
| 05:50.21 | VeNoMouS_ | thats using 729 i think or ulaw |
| 05:50.48 | Netgeeks | it should be the same, as it's converted into .wav whatever that is... pcm I think |
| 05:51.32 | VeNoMouS_ | well the pcm on ulaw/alaw is 8000 |
| 05:52.22 | P-NuT | hey guys, |
| 05:52.40 | P-NuT | what port does asterisk accept IAX connections on? |
| 05:53.07 | Netgeeks | 4569 |
| 05:53.07 | VeNoMouS_ | 4569 |
| 05:53.26 | *** join/#asterisk nagl (n=nagl@86.59.54.237) |
| 05:54.25 | P-NuT | yeah... |
| 05:54.36 | VeNoMouS_ | .... what |
| 05:55.02 | P-NuT | well, when I netstat -a it says " udp 0 0 *:iax *:* " |
| 05:55.25 | P-NuT | how do i tell what the IAX port its looking at is? |
| 05:55.43 | Qwell | P-NuT: -n |
| 05:55.49 | P-NuT | k |
| 05:56.04 | xbmodder_newlapp | How do I change my beacon interval on my Orinoco AP-500? |
| 05:56.14 | P-NuT | ok.. |
| 05:56.15 | P-NuT | hmm.. |
| 05:56.22 | P-NuT | its listening thenn.. |
| 05:56.23 | P-NuT | ok |
| 05:56.27 | P-NuT | well thanks guys |
| 05:57.23 | VeNoMouS_ | xbmodder_newlapp this is not #support_for_everything_on_the_planet_including_my_toaster!! |
| 05:57.28 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 06:00.06 | mosty | VeNoMouS_: ok, i have checked the firewall at the remote end, the asterisk server there is set as the dmz host, any other ideas for things to check? |
| 06:02.00 | mosty | VeNoMouS_: the remote asterisk server never shows "SIP/123 is ringing" when the call is being put through, it just says "called 123" |
| 06:03.07 | VeNoMouS_ | thats due to the remote end not sending back sip data |
| 06:03.19 | VeNoMouS_ | nmap -sU remoteip 5060 |
| 06:03.26 | VeNoMouS_ | does it say open or filtered? |
| 06:03.41 | VeNoMouS_ | oh wait u said the other person can hear it ringing right |
| 06:03.46 | VeNoMouS_ | then the sip data is going through |
| 06:03.49 | VeNoMouS_ | tcpdump |
| 06:03.52 | VeNoMouS_ | and look @ the rtp stream |
| 06:04.41 | VeNoMouS_ | fucking eyebeam and its piece of fucking shit h263 |
| 06:04.56 | *** join/#asterisk Ouch-\ (n=rahail1@209-19-88-238.detroit.mi.D-Conn.net) |
| 06:04.58 | defy | lol |
| 06:05.07 | mosty | VeNoMouS_: nmap says the syntax is wrong, it says 5060 is an invalid host |
| 06:05.26 | VeNoMouS_ | dude remotehost is the ip! |
| 06:05.50 | Ouch-\ | is there any ast guru here who would like provide some support our current s erver |
| 06:06.37 | VeNoMouS_ | Ouch-\ for a price |
| 06:06.45 | mosty | VeNoMouS_: i know, i copy+pasted and subbed in the ip |
| 06:06.57 | Ouch-\ | i dont know how much it will be great |
| 06:07.01 | Ouch-\ | if you tell what is your rate |
| 06:08.38 | VeNoMouS_ | $150 an hr |
| 06:08.56 | Ouch-\ | do you have any other option |
| 06:09.00 | stephane_ | joru |
| 06:09.06 | Ouch-\ | like getting montly recuring payment |
| 06:09.47 | Ouch-\ | to mataince they server if something goes wrong FYI server is working like churm only thing I am scared I am new at ast so i dont want any down time incase anything goes down |
| 06:10.35 | mosty | VeNoMouS_: nmap -sU host -p 5060, says 5060/udp open|filtered sip |
| 06:10.41 | VeNoMouS_ | you new to asterisk as well |
| 06:10.46 | VeNoMouS_ | err |
| 06:10.52 | VeNoMouS_ | s/asterisk/english/ |
| 06:14.26 | smackus | anyone using meetme and not getting deadlock errors? |
| 06:19.30 | VeNoMouS_ | does anyone in here have eyebeam working with asterisk? if so what version of eyebeam are you using? |
| 06:19.34 | VeNoMouS_ | err |
| 06:19.38 | VeNoMouS_ | sorry eyebeam with VIDEO |
| 06:20.09 | kruz_ | VeNoMouS_: thats my next project, coming up here soon(tomorrow) |
| 06:20.31 | smackus | apparently no one uses meetme and is not getting Avoided initial deadlock errors. are they really that common? |
| 06:20.38 | VeNoMouS_ | kruz_ .....? |
| 06:20.53 | VeNoMouS_ | smackus : yea |
| 06:21.58 | kruz_ | VeNoMouS_: im going to start working with eyebeam tomorrow |
| 06:22.16 | smackus | VeNoMouS_: its that common, or you dont have errors? |
| 06:22.46 | VeNoMouS_ | its common |
| 06:22.54 | smackus | gothca... |
| 06:23.18 | smackus | how are people getting around it? its crashing my server. |
| 06:23.26 | smackus | there has to be someone who has this figured out |
| 06:25.12 | Strom_C | smackus: try app_conference |
| 06:26.00 | VeNoMouS_ | man this video problem is fucking me right off |
| 06:27.21 | smackus | ok... is it a straight replacement? |
| 06:28.43 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 06:28.49 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 06:28.52 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 06:36.58 | kruz_ | does asterisk have to run on kernel 2.4 as suggest? i get alot of people saying that |
| 06:37.09 | kruz_ | then i just read that it works fine and comfortably in 2.6 also |
| 06:37.43 | mosty | i think it depends which hardware you use with it |
| 06:38.01 | kruz_ | is the 2.4 for the zaptel cards(and the irq delay problems they were having?) |
| 06:38.47 | mosty | i'm not sure what you're asking. but just try it with your preferred kernel, and if that fails try the other branch |
| 06:38.54 | *** join/#asterisk nexstar (n=nexstar@ip68-111-77-138.oc.oc.cox.net) |
| 06:38.55 | kruz_ | sorry, im asking |
| 06:39.00 | kruz_ | is 2.4 necessary |
| 06:39.18 | Strom_C | kruz_: I run on 2.6 without problems |
| 06:39.22 | Strom_C | I set up clients on 2.6 |
| 06:39.29 | Strom_C | I have no compunction about 2.6 :) |
| 06:39.31 | kruz_ | good, thats what i needed to know, thank you strom |
| 06:39.35 | kruz_ | do you use zaptel cards? |
| 06:39.40 | Strom_C | yes |
| 06:39.46 | kruz_ | with pots lines? |
| 06:39.58 | kruz_ | and its all good? no obvious delay?echo? |
| 06:40.05 | nexstar | trying to install asterisk-addons with using command... svn co http://svn.digium.com/svn/asterisk-addons/branches/1.2 asterisk-addons |
| 06:40.08 | Strom_C | pots lines are tricky to get the audio right on |
| 06:40.22 | Strom_C | I usually do larger offices, so I recommend PRI |
| 06:40.26 | nexstar | and its giving me the return of.... svn: 'asterisk-addons' is already a working copy for a different URL |
| 06:40.31 | nexstar | what does that mean? |
| 06:40.38 | kruz_ | thats not my problem, one of my problems is getting CVS to work through my proxy, to checkout asterisk, instead of using gay 6 month old apt-get asterisk |
| 06:41.09 | Strom_C | kruz_: svn has supplanted CVS, you know |
| 06:41.15 | Strom_C | CVS has been deprecated forever now |
| 06:41.24 | kruz_ | cvs.digium.com works does it not? |
| 06:41.34 | kruz_ | does svn work through a proxy? that would sell me now. |
| 06:41.39 | Strom_C | its merely a daily mirror of svn |
| 06:41.51 | Strom_C | at the least it wouldnt work any worse than cvs |
| 06:42.01 | kruz_ | does svn work through proxies? |
| 06:42.11 | Strom_C | worth a shot |
| 06:42.42 | mosty | kruz_: what features that you need was the 6 month old asterisk lacking? |
| 06:43.04 | kruz_ | not sure, but i dont want to get comfortable then install a new one and be stuck |
| 06:43.09 | kruz_ | besides its good to build from source up. |
| 06:43.23 | kruz_ | good habbit, that doesnt necesarrily mean quick unfortunatly. |
| 06:43.32 | Strom_C | yes, and asterisk is changing fast enough that its a good idea to always compile stable from source |
| 06:43.52 | mosty | well debian/ubuntu both provide support for a long time on their packages, i don't see the point in continuously upgrading unless you need some new feature |
| 06:43.58 | mosty | or a bugfix or something |
| 06:44.13 | kruz_ | security issues in the latest * i believe |
| 06:44.15 | kruz_ | err |
| 06:44.21 | kruz_ | fixed security issues rather. |
| 06:44.46 | Strom_C | mosty: i cant tell you how many people run into stupid bugs or try to use new features and then come in here and complain about it...and then we discover they're running some ancient version of asterisk they installed from a debian package |
| 06:44.47 | *** join/#asterisk _4d4m_ (n=adam@62.69.102.99) |
| 06:44.56 | mosty | yeah but you should be getting debian/ubuntu security fixes |
| 06:45.43 | mosty | strom_c: debian aims for stability, which doesn't mesh well with rapidly changing software i guess |
| 06:45.56 | Strom_C | mosty: thats why theres a STABLE branch of asterisk |
| 06:46.05 | Strom_C | separate from the development branch |
| 06:46.30 | nexstar | how do i upgrade addons? |
| 06:46.34 | Strom_C | I use debian. I quite like debian. I still install asterisk stable from digium's SVN server |
| 06:46.39 | nexstar | with out loosing settings? |
| 06:46.46 | mosty | strom_c: what goes into stable? just bug/security fixes? |
| 06:46.51 | kruz_ | i prefer bleeding edge, thats me. |
| 06:46.54 | VeNoMouS_ | its svn btw |
| 06:47.06 | Strom_C | mosty: yes, theres a feature freeze, and then incremental bugfixes |
| 06:47.10 | kruz_ | VeNoMouS_: do you know if svn does http/socks proxy support? |
| 06:47.29 | nexstar | anyone ... how do i upgrade asterisk-addons with out loosing settings? |
| 06:47.30 | mosty | strom_c: so just more frequent freezes than debian's release cycle? |
| 06:47.34 | smackus | how do i disable music on hold system wide? |
| 06:47.38 | smackus | I am using native. |
| 06:48.00 | smackus | I tried to just comment out anywhere calling for moh, but that sent it to default. |
| 06:48.09 | VeNoMouS_ | kruz_ : would depend on ure client |
| 06:48.15 | Strom_C | mosty: there's a feature freeze for the 1.x version, and then any updates after that are bugfixes. The developers aim for a new version every six months or so |
| 06:48.47 | Strom_C | we're on 1.2.9.1; 1.4 is supposed to be released soon |
| 06:48.56 | litage | what's port 0 (zero) used for? i just found this in my logs: list 155 permitted tcp 219.129.237.67(0) -> 202.168.41.171(0), 1 packet |
| 06:49.06 | kruz_ | VeNoMouS_: how so? i need a way to reach the internet through my proxy(ubuntu, it works with wget and apt-get using the proxy) |
| 06:50.47 | mosty | strom_c: that's about on target with ubuntu's release schedule, i wonder if their server release has asterisk |
| 06:51.10 | Strom_C | supposedly it does |
| 06:51.11 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 06:51.12 | mosty | if so it could make a nice pair for a production machine |
| 06:51.20 | Strom_C | but why are you so addicted to the packages? |
| 06:51.21 | *** join/#asterisk P-NuT (n=P-Nut@fw.office.unitedip.net.au) |
| 06:51.29 | Strom_C | scared of subversion or something? :) |
| 06:51.53 | smackus | so i have commented out all of the contexts in musiconhold.conf, still gives me music on hold. how do i just turn it off? |
| 06:52.04 | mosty | strom_c: no not all all, but i maintain lots of machines, and i prefer to let distributions do as much of the work as possible on production machines |
| 06:52.18 | kruz_ | server release 6.06 daper drake does NOT have asterisk preinstalled |
| 06:52.28 | nexstar | can someone please tell me how to upgrade asterisk-addons? |
| 06:52.33 | kruz_ | smackus: did u restart asterisk? |
| 06:52.41 | kruz_ | smackus: to make the new conf take place? |
| 06:53.20 | mosty | kruz: it's not important if it's preinstalled or not, just that it's available and has security fixes available |
| 06:54.08 | kruz_ | mosty:i prefer to build from source, i can analyze, and i know im getting the latest and greatest. |
| 06:54.31 | *** join/#asterisk billy-jo (n=billy-jo@AClermont-Ferrand-251-1-63-223.w81-251.abo.wanadoo.fr) |
| 06:54.36 | kruz_ | anyone: im new to svn, how do u use it to dl asterisk? |
| 06:55.09 | Strom_C | kruz_: there are instructions on asterisk.org |
| 06:55.23 | smackus | i did a reload, do you have to actually restart to affect moh? |
| 06:55.24 | kruz_ | k,ty |
| 06:55.32 | Strom_C | smackus: sure, try it |
| 06:55.38 | kruz_ | smackus: i believe so, to reload the .conf's |
| 06:55.52 | smackus | hmmm. ok. 108 active channels. I will let you know. :-D |
| 06:56.10 | Strom_C | smackus: do your maintenance after the business day ends |
| 06:56.24 | smackus | it is 1am here :-D |
| 06:56.30 | smackus | business never ends |
| 06:56.45 | mosty | kruz: fair enough. i'm more interested in stability personally |
| 06:56.46 | smackus | i am going on 22 hours of work for today |
| 06:56.57 | kruz_ | mosty: well then apt-get is def the way to go |
| 06:57.05 | Strom_C | feh |
| 06:57.06 | kruz_ | i just dont want to get EVERYTHING setup and findout something has changed |
| 06:57.14 | Strom_C | apt-get will get you 1.0.9 or something |
| 06:58.12 | kruz_ | yes, and personally thats wha tim using now |
| 06:58.21 | Strom_C | holy catsex |
| 06:58.26 | Strom_C | thats really really ancient |
| 06:58.35 | Strom_C | considering we're on the verge of 1.4 now |
| 06:58.59 | *** join/#asterisk af_ (n=af@ip-170-209.sn1.eutelia.it) |
| 06:59.10 | kruz_ | yes, if thats hat apt-get has, im just using it for phone configuration testing currently |
| 06:59.34 | Strom_C | kruz_: just do me a favor and dont put 1.0.9 into production |
| 07:00.09 | kruz_ | Strom_C: how stupid do you think i am |
| 07:00.11 | kruz_ | hehe |
| 07:00.20 | kruz_ | im using it while im at home doing configurations |
| 07:00.47 | kruz_ | but i must admit, when im installing asterisk, i LOVE sudo apt-get build-dep asterisk |
| 07:00.51 | Strom_C | kruz_: spend enough time on this channel and youll see some fairly mind-boggling stupidity |
| 07:00.51 | kruz_ | then install asterisk from source |
| 07:00.53 | kruz_ | makes it much easier |
| 07:01.00 | Strom_C | kruz_: asterisk from source is easy as shit |
| 07:01.04 | kruz_ | Strom_C: i will be spending time here |
| 07:01.08 | Strom_C | make clean; make install |
| 07:01.12 | Strom_C | voila |
| 07:01.16 | kruz_ | Strom_C: its not always that easy |
| 07:01.27 | kruz_ | Strom_C: im working on, when things go wrong |
| 07:01.34 | Strom_C | kruz_: yes it is that easy |
| 07:01.43 | kruz_ | not on a DSL install |
| 07:01.43 | Strom_C | i install asterisk boxes for a living |
| 07:01.49 | Strom_C | im actually at a client's site now ;) |
| 07:01.52 | kruz_ | really? do u work for digium? |
| 07:02.03 | Strom_C | i contract for digium, yes |
| 07:02.08 | kruz_ | sweet |
| 07:02.11 | Strom_C | but right now I'm in tarzana, california |
| 07:03.29 | *** join/#asterisk chipach (n=chip@chocolate.chip.net) |
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| 07:20.30 | Strom_C | alright, time to head home |
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| 07:49.22 | creadurx | hm.. is there any requirement to get the callerid(DNID) variable set? |
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| 08:07.12 | RoyK[de] | morgen |
| 08:08.45 | creadurx | guten morgen |
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| 08:28.38 | BertZ | hello there |
| 08:31.00 | BertZ | asterisk is working fine, but I can see that in my logs : SIP/anonymous.invalid-081ced98 |
| 08:31.05 | BertZ | why 'invalid' ??? |
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| 08:49.07 | *** join/#asterisk ramtha (n=ramtha@195.14.234.162) |
| 08:49.10 | ramtha | hey ho |
| 08:49.26 | ramtha | someone got snom conference btton to work? |
| 08:49.44 | ramtha | i can press it thousend times and nothing will haben.... |
| 08:49.48 | ramtha | any ideas? |
| 08:51.04 | RoyK[de] | ramtha: what does sip debug say when you press it? |
| 08:51.17 | ramtha | mom i take a look |
| 08:51.21 | skeffling | ramtha: you need to press it 1001 times, no really I just tried it here phone 1 called phone 2. phone 1 put phone 2 on hold. Phone 1 called phone 3. the pressing CONFERENCE on phone 3 joined them all up |
| 08:51.26 | ramtha | is here english or german prefered? |
| 08:51.38 | skeffling | it doesn't use asterisk's meetme though |
| 08:51.48 | RoyK[de] | ramtha: english |
| 08:51.55 | ramtha | ok perhaps there are some park/hold features in asterisk i do not aktivated? |
| 08:52.06 | ramtha | transfer and hold does work |
| 08:52.21 | skeffling | ramtha: so you press hold, and nothing happens? |
| 08:52.31 | *** join/#asterisk trixter (n=trixter@65-165-167-217.du.volcano.net) |
| 08:52.39 | ramtha | jo |
| 08:52.43 | ramtha | oeh yes ;) |
| 08:52.47 | ramtha | hold works |
| 08:52.55 | skeffling | I'm using a 360's and 300's here, on version 6.x.x firmwares |
| 08:53.01 | ramtha | the caller gets the musiconhold |
| 08:53.22 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 08:53.33 | skeffling | ramtha: hold music is good, you then dial the 2nd person, put them on hold, dial the 3rd and then press conference |
| 08:53.41 | skeffling | (in theory) |
| 08:53.51 | ramtha | ahhhhh+ |
| 08:53.52 | ramtha | ok |
| 08:54.13 | ramtha | i thought that pressing conference sets the first on use |
| 08:54.15 | ramtha | i try |
| 08:54.29 | skeffling | it should work...! |
| 08:54.41 | trixter | so who is going to cluecon? |
| 08:55.10 | ramtha | thx @ all |
| 08:55.14 | ramtha | it works |
| 08:55.24 | skeffling | ramtha: cool! |
| 08:55.28 | ramtha | the problem is infront of this computer ;) |
| 08:55.29 | trixter | I still think that cluecon should give out nerf bats that say 'cluecon' on em so at least people there can get cluebats :) |
| 08:55.43 | ramtha | it´s me.. |
| 08:55.47 | BertZ | hmmm |
| 08:56.37 | BertZ | if I want a different music for each queue, should I define several classes, or just set music=my..mp3 in each queue ? |
| 08:57.02 | ramtha | several classes should work.... |
| 08:57.12 | BertZ | let me try |
| 08:57.59 | BertZ | but what if I have several mp3 files in the directory ? |
| 08:58.05 | BertZ | how to specify the one I want ? |
| 08:58.25 | BertZ | I can define a moh classe with a directory, but I can set a mp2 file ? |
| 08:58.33 | BertZ | cant |
| 08:58.57 | trixter | you specify a class for each directory that you want |
| 08:59.02 | trixter | that is the easiest way |
| 08:59.22 | trixter | create a new directory for each group, symlinks if you are short on diskspace and wnat to share music between them if that is something you want |
| 09:01.31 | BertZ | yep but I would like to have all my mp3 in the same dir, and be able to choose the mp3 depending on the selected queue |
| 09:01.48 | BertZ | and I'm not sure it is possible this way |
| 09:02.19 | ramtha | hmm |
| 09:02.35 | trixter | you might be able to specify each file instead of a directory foir the class |
| 09:02.39 | ramtha | if there is a funktion to call the file in the dilplan you can use that way |
| 09:02.45 | trixter | but that is ugly and harder to maintain, but it should work if you do that |
| 09:02.48 | ramtha | but i dont know something like this |
| 09:03.34 | BertZ | I will use subdir for each queue, with the correct mp3 |
| 09:03.43 | BertZ | if it works, then fine :) |
| 09:04.30 | trixter | that makes it easier to dynamically change without reloading, and makes it easier to configure since you wont have a million character command line, which is likely to cause other problems |
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| 09:04.48 | trixter | you should still be able to keep all files in one dir and symlink to them from the different class dirs |
| 09:05.41 | BertZ | yep you right :) |
| 09:06.06 | trixter | I get lucky every now and again |
| 09:14.28 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 09:17.12 | trixter | has anyone made an appreciably longer than 8 hour call with skype? most of mine die less than 8 hours 1 minute, many within seconds after 8 hours |
| 09:17.22 | trixter | so I am thinking its a call limit, but not entirely sure |
| 09:17.35 | trixter | better than internetcalls with 1 hour caps :P |
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| 09:20.03 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 09:22.38 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 09:26.21 | *** join/#asterisk jhiver (n=jhiver@LReunion-151-20-4.w193-253.abo.wanadoo.fr) |
| 09:26.23 | jhiver | hi all |
| 09:26.54 | jhiver | I was wondering if there was a way in sip.conf to limit the number of simultaneous channels for a given group of users rather than per user basis |
| 09:27.05 | jhiver | I have a group of 5 phones, and I need to limit it to 5 channels |
| 09:27.10 | jhiver | sorry |
| 09:27.19 | jhiver | a group of 10 phones, not 5 :) |
| 09:30.47 | trixter | you need to look at call groups |
| 09:30.53 | ramtha | hmm |
| 09:30.53 | ramtha | there is an option calllimit for every sip peers |
| 09:30.53 | ramtha | set each to 1 |
| 09:30.54 | ramtha | or set calllimit var in extensions.conf |
| 09:31.06 | trixter | you can name a group anything you want and put all those phones into that group, and setgroup/checkgroup would be the dialplan functions to look at |
| 09:31.27 | trixter | afaik no asterisk doesnt support that |
| 09:31.35 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 09:31.51 | trixter | but there are a TON of patches that were never implemented into asteriks itself, there might be one out there |
| 09:31.52 | smackus | I have tried everything. I cannot get zaptel installed. |
| 09:32.06 | smackus | I have messed with the spinlock file. |
| 09:32.20 | ramtha | smakus: error message? |
| 09:32.24 | smackus | as far as I can tell, I have the correct kernel files installed |
| 09:32.29 | smackus | .... ok, hang on |
| 09:32.38 | trixter | http://www.voip-info.org/wiki-Asterisk+sip+incominglimit |
| 09:33.03 | smackus | warning: 'fcstab' defined but not used |
| 09:33.57 | trixter | ramtha: aparently it does have that feature :) |
| 09:34.16 | trixter | which is better I think than setgroup/checkgroup however read the notes carefully as it appears to have bugs in 1.2 |
| 09:34.23 | trixter | so test early test often |
| 09:34.42 | smackus | http://pastebin.ca/75570 |
| 09:36.28 | ramtha | smackus: wich distri? |
| 09:36.42 | smackus | red hat ent 4 |
| 09:36.53 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 09:37.26 | joelsolanki | Hi all. i want to use asterisk for providing whole voip. which free billing system should i use ? |
| 09:37.42 | joelsolanki | billing incremental should be configurable. |
| 09:37.43 | joelsolanki | ? |
| 09:37.46 | joelsolanki | any plz |
| 09:37.53 | ramtha | smackus: take a look at: http://lists.digium.com/pipermail/asterisk-users/2006-March/144504.html |
| 09:38.28 | smackus | yep, i tried that one, and it did not work. either i did it wrong, or it is not the issue |
| 09:39.40 | *** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 09:39.44 | ramtha | you tried mv spinlock.h spinlock.h.old ...etc? |
| 09:39.50 | smackus | yep |
| 09:39.54 | ramtha | ok |
| 09:39.55 | ramtha | hmm |
| 09:40.22 | smackus | how do i know for sure that I have the kernel source and kernel headers installed in linux? |
| 09:40.28 | smackus | new to this distro |
| 09:40.42 | smackus | in fedora i just use yum |
| 09:40.58 | ramtha | yum is your friend at this point |
| 09:41.08 | ramtha | install debian, that works out of the box ;) |
| 09:41.15 | smackus | pain in the ass to install yum on redhat because of all of the dependancies |
| 09:42.13 | trixter | use yum to install it, it will resolve the dependancies for you :P |
| 09:42.33 | ramtha | *g* |
| 09:42.35 | smackus | the trouble is all of the dependencies require to install yum |
| 09:42.57 | ramtha | smackus: i can remember that i had the same problems half a year ago |
| 09:42.57 | trixter | yes but if you install it with yum it will take care of all that for you |
| 09:43.04 | ramtha | but i can not remember the solution |
| 09:43.21 | smackus | dammit its too late for jokes :-D i dont understand them |
| 09:43.44 | trixter | heh |
| 09:46.31 | ramtha | thunder of the guns... |
| 09:46.49 | _4d4m_ | smackus: what distro? |
| 09:47.14 | ramtha | redhat.... |
| 09:47.21 | ramtha | so above.. |
| 09:47.49 | _4d4m_ | rpm -qa | grep kernel-source |
| 09:48.00 | jhiver | trixter, thanks |
| 09:48.06 | smackus | red hat ent 4 |
| 09:48.37 | trixter | jhiver: oddly that was just said in here by someone else so it was fresh in my mind from looking it up for them |
| 09:48.43 | trixter | I still suggest you at least try option 3 :) |
| 09:48.58 | trixter | it may be least effective but it is fun! |
| 09:49.05 | smackus | comes back with nothing.... |
| 09:49.33 | jhiver | trixter, I don't think doing it in the dialplan is an option I'm afraid |
| 09:49.39 | smackus | where can i get this from? |
| 09:50.15 | jhiver | I was hoping maybe it's possible to create a "fake" user and then direct the SIP phones calls to this "fake" user, and limit the "fake" user? |
| 09:50.20 | trixter | jhiver: the limit (option 1) is done on the peer level |
| 09:50.23 | trixter | so its not a dialplan thing |
| 09:50.31 | smackus | <PROTECTED> |
| 09:50.31 | trixter | option 3 isnt a dialplan thing either :) |
| 09:50.36 | jhiver | ? |
| 09:50.41 | jhiver | option 3 isn't? |
| 09:50.45 | trixter | did you read the url I provided? |
| 09:50.53 | trixter | option 3 is smacking them with a rotten fish |
| 09:50.57 | jhiver | exten => _0.,1,Set(GROUP()=SOME_PROVIDER) ;Set Group |
| 09:50.59 | trixter | no dialplan anything |
| 09:51.03 | jhiver | aaah I see :) |
| 09:51.03 | trixter | that is option 2 |
| 09:51.15 | trixter | read up on option 1, that is the url I provided |
| 09:51.24 | trixter | you can set a limit on the peer level, ie to the ITSP |
| 09:51.33 | trixter | in 1.2 you have seperate incoming and outgoing limits |
| 09:51.57 | jhiver | yeah but I AM the ITSP |
| 09:52.10 | jhiver | I have a customer who has 10 phones but wants only 5 lines |
| 09:52.20 | trixter | then you can do a limit on a per peer basis, ie your customers |
| 09:52.28 | trixter | so long as you can aggregate all users into one account it will work |
| 09:52.34 | jhiver | yeah but each phone is a peer... |
| 09:52.45 | trixter | if you are trying to take 5 seperate accounts and aggregate them into one acct you must use a group in asterisk |
| 09:52.55 | _4d4m_ | smackus: try rpmfind.net |
| 09:52.56 | jhiver | it's not like there's a sip gateway on his side or anything |
| 09:53.02 | jhiver | I wish :) |
| 09:53.53 | trixter | afaik asterisk doesnt have a way to group users together for the purpose of a call limit |
| 09:54.13 | trixter | so you have to use setgroup/checkgroup in the dialplan if you want to do that across multiple accounts |
| 09:54.19 | jhiver | okay... so back to square 1 :) |
| 09:54.23 | trixter | or get a sip proxy that will sit in between |
| 09:54.33 | trixter | there is always a rotten fish! |
| 09:54.35 | ramtha | sit, not sleep... |
| 09:54.37 | jhiver | :) |
| 09:54.46 | ramtha | hmm its friday i get confused ;() |
| 09:55.10 | jhiver | OK So I *could* in the dialplan, have Set(GROUP()=MY_CUSTOMER_PHONES) |
| 09:55.20 | *** join/#asterisk benjamin7062 (n=benjamin@mailserver.photodex.com) |
| 09:55.29 | trixter | yes, perhaps based off acctcode if you use that as a unique identifier |
| 09:55.35 | jhiver | and then use the strange $${GROUP_COUNT()} function |
| 09:55.39 | jhiver | I do |
| 09:55.43 | smackus | all i can find for my kernel version is the smp version of the kernel. isnt that just for 64bit machines? |
| 09:55.58 | trixter | that way all 10 accounts will have the same groupname, which is how asterisk seperates that out |
| 09:56.13 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:56.30 | mosty | smackus: no smp is multiple cpu/multiple core machines |
| 09:56.56 | trixter | specifically where all cpus are the same speed |
| 09:56.56 | jhiver | well I think I'm gonna tell the customer to accept a "flexivariable" lines deal because I have a way of knowing how many lines were simultaneously but no clean / easy way of doing what he requires |
| 09:57.01 | trixter | thus symetric multi processing |
| 09:57.10 | trixter | as opposed to the much more rare AMP |
| 09:57.13 | smackus | would not work on a single proc machine then... .i cant find anything else. ERRRR! |
| 09:57.43 | mosty | smackus: smp kernels work fine on single cpu machines too |
| 09:58.00 | smackus | really? |
| 09:58.15 | trixter | jhiver: you could look into a sip proxy, which in all honesty sip proxies are single task functions that are very good at what they do, and they do it FAR better than asterisk because they are specialized for their task. many support 'virtual trunks' where you can limit groups of users |
| 09:58.25 | mosty | yes. they are just slightly larger. not really a problem unless you're booting from a floppy or something |
| 09:58.37 | jhiver | trixter, SER won't do that for sure |
| 09:58.47 | jhiver | not on it's own anyway |
| 09:58.53 | jhiver | any other recommendation? |
| 09:58.53 | smackus | ok... so tell me this. I went to the other machine that I have already installed zaptel on. It also has no output for rpm -qa | grep kernel-source. |
| 09:58.55 | _4d4m_ | smackus: why not just download the kernel sources and compile your own? |
| 09:59.16 | smackus | new to this... not sure if I dare. |
| 09:59.20 | smackus | :-D |
| 09:59.32 | smackus | where do i get them at? |
| 09:59.53 | mosty | smackus: i dont know about redhat/fedora but debian has seperate packages from zaptel (utils) and zaptel-source (kernel modules) |
| 10:00.13 | *** part/#asterisk defy (n=defy@60-234-234-98.bitstream.orcon.net.nz) |
| 10:00.23 | trixter | jhiver: nah I am just about out of ideas. you could hack something together based on the call-limit stuff to be able to create a group of users and not have it just on the peer individually, however that is gonna be tricky becuase 1. you are going to have to deal with locks which will degrade performance when you have a bunch of call setup/tear down, 2. you have to be able to accurately sync this up, and user based stuff is harder to d |
| 10:00.24 | trixter | o |
| 10:00.50 | jhiver | don't worry about it |
| 10:00.53 | trixter | asterisk really doesnt have good user support anyway, a zap user is totally seperate from a sip user who is seperate from a iax user and so on |
| 10:01.05 | trixter | users shoiuld be their own entities and you just list the technologies they are allowed to access |
| 10:01.14 | smackus | is there a way to tell if the kernel sources are installed if they were not installed via rpm? |
| 10:01.26 | smackus | ie, my other machine that has zaptel installed and working. |
| 10:01.34 | trixter | smackus: are htey on your filesystem? |
| 10:01.38 | jhiver | I'll just tell the customer "you need an ip pbx to do that" and no problem, if it can't be done it can't be done, not a big deal |
| 10:01.43 | trixter | normally stuff looks in /usr/src for it |
| 10:01.43 | smackus | where do i check |
| 10:01.52 | trixter | but that isnt guaranteed |
| 10:02.36 | jhiver | speaking of SMP machines, forgive my ignorance but how does * benefit from them? Since it's a single process I thought it would be running on a single CPU but obviously it's very naive |
| 10:02.38 | smackus | nothing other than the basic kernel |
| 10:02.43 | trixter | jhiver: by cutting down your registrations you can have more cpu free for other stuff :) although tweaking with the timeouts may be a good thing as well, so that you arent constantly hammered |
| 10:02.53 | jhiver | trixter, yeah |
| 10:03.10 | jhiver | trixter, eventually I'll stick SER back in the equation as well |
| 10:03.10 | trixter | smackus: generally if you dont have a /usr/src/linux/include/whatever then you dont have em |
| 10:03.26 | jhiver | trixter, but NAT support is a bit trickier with it |
| 10:03.48 | smackus | lol, ok so how did zaptel get installed without it? I am stumped. I cannot get zaptel to install on this new machine. same os |
| 10:04.05 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 10:04.36 | smackus | here is what I get when i try: http://pastebin.ca/75570 |
| 10:04.44 | trixter | jhiver: mod your rtp.c to change the IP based on what is received, while you use the sip headers initially, if they send you something on the rtp port for that call and its valid rtp you know that is them |
| 10:05.04 | trixter | so just ignore what they said in the headers and use the IP that sends to you, the code is there but only if nat=yes which has other issues |
| 10:05.32 | trixter | that solves MOST nat issues and less than 0.01% of the time causes a problem (ie where the media gateway on the remote end sends from IP X but must receive on IP Y - very rare) |
| 10:05.48 | *** join/#asterisk nothinman (i=shakey@aczm30.neoplus.adsl.tpnet.pl) |
| 10:05.49 | jhiver | trixter, yeah, I think nextone is like that |
| 10:05.53 | trixter | that way even if SIP sends a rfc1918 addr in the via line or whatever the call still works |
| 10:05.56 | jhiver | it uses two IP addresses |
| 10:05.59 | nothinman | hello |
| 10:06.14 | trixter | nextone is sending me 1 IP for RTP - their class 5 switch that is |
| 10:06.48 | jhiver | I have a customer who required I allow from two distinct IPs, I assumed that was it |
| 10:07.07 | trixter | I have seen some stuff where it will send on one port rx on another, which would break this -- although you dont have to update the port its a good idea if you are going to do that to do all or none |
| 10:07.20 | jhiver | trixter, you managed to interconnect with nextone? this other customer of mine is having the most horrible time trying to get it to work |
| 10:07.42 | trixter | jhiver: yeah but do you send to A and he sends from B for the same call? or is one RTP session using the same IP for that whole sesison? |
| 10:08.02 | jhiver | I'm not sure, let me check the archives |
| 10:08.15 | trixter | jhiver: the switch I am connecting to isnt generally available yet, its doing final trials now.. its their class 5 telco switch |
| 10:08.25 | jhiver | oh yeah |
| 10:08.37 | jhiver | basically they are having to ring tone, no audio, nothing :) |
| 10:08.42 | trixter | but if they can get that to work well then it shouldnt be a problem for their other products |
| 10:09.17 | jhiver | it's really strange this interconnect business. Sometimes in 5 minutes you're done and sometimes you never manage to do it |
| 10:09.41 | trixter | heh |
| 10:10.04 | trixter | speaking of nextone I need to add more DIDs.. adding like 5 more states this week, just gotta test and make sure they really work as well as they should |
| 10:10.33 | jhiver | wow |
| 10:10.44 | jhiver | well, good work then |
| 10:10.48 | trixter | :) |
| 10:11.14 | trixter | adding new york, new jersey, mass, rhode island, and um new hampshire I think |
| 10:11.30 | jhiver | yey |
| 10:11.38 | jhiver | plenty of places I will never need a DID in :) |
| 10:11.39 | trixter | since they are free it shouldnt matter too much where it is but people are picky :P |
| 10:11.49 | jhiver | :) |
| 10:12.06 | trixter | even though I pay people to receive calls people are still picky! sheesh is nothing good enough? :P |
| 10:13.36 | *** join/#asterisk benjamin7062 (n=benjamin@mailserver.photodex.com) |
| 10:15.06 | trixter | so who all is going to cluecon.com ? no one really answered earlier :P |
| 10:15.53 | trixter | chicago, aug 1-3, one of the authors of asterisk the future of telephony will speak, as well as some other asterisk and non asterisk stuff.. tons of open source as well as some that isnt (like cepstral and voxeo) |
| 10:28.18 | *** join/#asterisk kay2 (n=ashdown@sd-420.dedibox.fr) |
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| 10:59.22 | *** join/#asterisk nXOR (n=drade@pdpc/supporter/sustaining/nXOR) |
| 10:59.46 | nXOR | hello good people, can u please tell me where i can get help with visdn ? |
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| 11:13.14 | ramtha | have a nice day all..... |
| 11:13.16 | ramtha | bye |
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| 11:24.38 | *** join/#asterisk PK (n=PK@pdpc/supporter/sustaining/PK) |
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| 11:27.11 | k_ober | hi ther |
| 11:28.13 | k_ober | is there a way in a blind transfer, to get the id of the extensio who made the transfer? |
| 11:32.48 | PK | hey, I have a short question: if I have two hw ip phones, is it possible to see on phone A if there is a call to phone B and pick it up if the owner of phone B left his place? is that what they call 'call monitoring' on the feature list? |
| 11:35.10 | *** join/#asterisk Baffelmae (n=root@211-11-156-165.withe.ne.jp) |
| 11:36.06 | Baffelmae | hello has anyone gotten a TDM400P to work in Japan? |
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| 11:57.45 | *** join/#asterisk lorinc (n=ang@caracas-0815.adsl.interware.hu) |
| 11:58.05 | B4 | hello people |
| 11:58.30 | B4 | anyone active at this time? |
| 11:58.45 | Baffelmae | konichiwa |
| 11:58.49 | Baffelmae | not really |
| 11:59.01 | B4 | konichiwa |
| 11:59.25 | trixter | konbonwa |
| 11:59.39 | trixter | although there really should be some spaces in there :P |
| 11:59.41 | B4 | heh first lessons in japanese |
| 11:59.59 | Baffelmae | hai so desuyo |
| 12:00.02 | B4 | but need a first lesson in configuring a euroisdn pri |
| 12:00.17 | Baffelmae | what is the problem? |
| 12:00.32 | B4 | no problem ... just do not know how to go about it lol |
| 12:00.46 | B4 | never configured a PRI with * before |
| 12:00.50 | Baffelmae | where have you got to? |
| 12:00.54 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 12:01.31 | B4 | I looked at the asterisk guru page for zaptel.conf ... but I am a bit confused as he is only using one D channel on a pri |
| 12:01.31 | Baffelmae | basically it (depending on your provider) is not so hard |
| 12:01.40 | B4 | should'nt there be 2? |
| 12:02.12 | Baffelmae | no just 1 |
| 12:02.17 | B4 | most of the stuff on the net is for US T1 ... not E1 |
| 12:02.20 | B4 | why 1? |
| 12:02.23 | Baffelmae | sure |
| 12:02.30 | Baffelmae | but it is not that different |
| 12:03.35 | Baffelmae | where are you at in the EU? |
| 12:05.14 | B4 | talking in the other window :) |
| 12:05.21 | Baffelmae | lost that one |
| 12:05.23 | Baffelmae | start again |
| 12:06.16 | trixter | most E1 have 1 D and 1 'dead' channel, many people say that 2nd dead channel (used for syncing but nothing of value goes over it) is a 2nd D, or so I have been informed |
| 12:06.40 | B4 | ah ok |
| 12:06.50 | B4 | yes it is used for synching ... |
| 12:06.51 | trixter | thus 32 channels goes down to 30 voiuce (B) and 1 signalling (D) with one slack for syncing |
| 12:06.59 | trixter | er voice |
| 12:07.17 | trixter | 16 is the common D and 32 is the common dead channel from what I have read |
| 12:08.09 | trixter | but I really dont know much about E1s, never researched it since I currently am in north america |
| 12:08.22 | trixter | so anything I read was in passing more by accident than anything else |
| 12:08.23 | B4 | k :) |
| 12:08.43 | B4 | there is very little on the net about E1s and * |
| 12:09.50 | B4 | thanks :) |
| 12:12.56 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 12:13.45 | trixter | really? I kept finding stuff by accident when I looked for other things, why I know what I do about them |
| 12:14.52 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
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| 12:15.04 | jhiver | trixter, weren't u supposed to work :) |
| 12:15.13 | jhiver | IRC junkies :) |
| 12:15.59 | jhiver | thanks for you msg on the ml anyway, I like the rotten fish approach |
| 12:16.40 | jhiver | allo? Hi, how are you doing? This is john from *SLAP* *hangup* |
| 12:17.11 | trixter | jhiver: I am working, my boss is perfectly happy with what I am doing, course I work for myself so that makes it easier but meh :P |
| 12:18.14 | trixter | I do need to finish rewriting the control software, especially the carrier provisioning stuff, so that ITSPs can integrate my products (which are free, I even pay you to use em!) to their existing stuff |
| 12:18.49 | *** join/#asterisk Sg-fr (n=Tristan@fe0-0-c2621-fw.nce.telemaque.fr) |
| 12:18.55 | Sg-fr | Hi everybody |
| 12:19.07 | PK | does anyone know if it's possible to see if there is a call on another phone and pick that up? line monitoring or what they call that? |
| 12:20.27 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:22.05 | trixter | PK: what channel type? and you can generally look at 'hints' to see from a client if there is a line in use, as far as picking it up that depends on exactly what you mean by that |
| 12:29.06 | znoG | so does anyone use/own a SPA-841/941? |
| 12:31.28 | *** join/#asterisk beyond (n=beyond@200.192.160.100) |
| 12:32.01 | PK | trixter: well, without former TVA (PBX), I could see when the phone of my collegue rang and if he wasn't here, I could press a key on my phone to answer his calls |
| 12:36.18 | trixter | you can cause it to check to see if he is available, via DND, you can have it goto multiple extensions at the same time, etc |
| 12:38.12 | *** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br) |
| 12:40.40 | PK | trixter: doesn't sound like what I need :( I want it to ring normally, he might just have left to get a new cup of coffee and didn't turn on the DND. And I want to answer the call to his phone with my phone (of course only if I have to rights to) |
| 12:41.16 | Mw3 | create a pickupgroup |
| 12:41.27 | Mw3 | and press *8 or the configured keys |
| 12:41.31 | trixter | that becomes a little harder with asterisk, you could make it dial both his phone and yours, but if you dont have a multi line phone that makes it a little more bothersome becuase you dont know which extension is ringing |
| 12:41.39 | trixter | you can make it ring yours if its not answered after a couple seconds |
| 12:41.50 | trixter | ie after 1-2 ring cycle yours starts ringing |
| 12:42.17 | trixter | yeah but then its all sorts of extra buttons, it was my understanding that he wanted something a little more simple than that |
| 12:42.34 | trixter | but if that works for what he wants then great. |
| 12:43.11 | trixter | course you may want to make that a speed dial or soemthing so its easier to remember and dials faster, assuming your phone does have that without some convoluted method ... |
| 12:43.33 | trixter | hmm I gotta check on a domain |
| 12:43.51 | trixter | Domain Name: chicken.coop |
| 12:44.14 | trixter | heh its taken, just saw a commercial for a .coop domain (cooperative) and thought instantly of chicken.coop :P |
| 12:44.26 | *** join/#asterisk AltnTab (n=ecs@nrjsoft13.networx-bg.com) |
| 12:44.59 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:46.46 | *** join/#asterisk Sg-fr (n=Tristan@LNeuilly-152-23-33-115.w193-252.abo.wanadoo.fr) |
| 12:46.57 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 12:47.33 | *** part/#asterisk B4 (n=B4@202.69.48.245) |
| 12:49.30 | *** part/#asterisk Baffelmae (n=root@211-11-156-165.withe.ne.jp) |
| 12:54.38 | PK | trixter: basically I need to know if a line is ringing (seeing it's status) and being able to pick its rings (for example *8#[phonenumber]). Do you how much of that is implemented already and if it would be a lot of effort to do that? |
| 12:55.36 | trixter | status is done via hints and a BLF enabled phone (if sip, dunno what other channel types support that) |
| 12:55.48 | trixter | you can see if a line is ringing, in use, or idle |
| 12:56.06 | PK | ok, good |
| 12:56.32 | PK | then I only need to add a way to tell the asterisk server to reroute a ring? |
| 12:56.43 | trixter | so that takes care of that part of it, then you need to set it up so you can answer that line if you want, which can be done a few different ways, that part of it might be what mw3 said earlier |
| 12:59.49 | *** join/#asterisk arguile (i=user224@66.38.201.234) |
| 13:01.07 | *** join/#asterisk Katty (n=aisaacs@64.82.232.54) |
| 13:01.26 | *** join/#asterisk speedwagon (n=Ariel@70.46.87.158) |
| 13:02.29 | Katty | morning |
| 13:04.03 | trixter | hi |
| 13:05.44 | *** join/#asterisk ClayReiche12 (i=fwuser@rrcs-71-41-73-130.se.biz.rr.com) |
| 13:07.06 | *** join/#asterisk littleball (n=littleba@cm52.epsilon174.maxonline.com.sg) |
| 13:07.06 | *** join/#asterisk Oins (n=Hartmut@p549208F9.dip0.t-ipconnect.de) |
| 13:07.06 | littleball | hello |
| 13:07.07 | X-Gen | hey ho |
| 13:07.20 | Oins | hi |
| 13:08.38 | ClayReiche12 | Can anyone tell me why I'm getting the privacy screen when running the "Dial" command with an "&"? When I launch the command from extensions.conf it works as expected, (dials both parties) but when I launch the same command from an AGI script (using perl) I get the privacy screener and hear Allisons prompts... |
| 13:10.52 | Oins | Can anyone explain me the differences between Zaptel and Dialogic Hardware? |
| 13:11.23 | ClayReiche12 | The exact command is: Dial(IAX2/username:password@76.33.88.2/8134166290|30|gM(screen^${SCREEN_FILE})&IAX2/username:password@76.33.88.2/8137491444|30|gM(screen^${SCREEN_FILE})) |
| 13:12.14 | X-Gen | Oins, dialogic = plenty of $$ where processing is dont on the card's DSP. Zaptel uses the PC's cpu |
| 13:12.33 | *** join/#asterisk djPepse (n=pepse@ip68-109-169-37.ph.ph.cox.net) |
| 13:13.55 | *** join/#asterisk AltnTab1 (n=ecs@nrjsoft13.networx-bg.com) |
| 13:14.14 | Oins | X-Gen, ok thank you... and what means DSP :) ? |
| 13:14.51 | X-Gen | Digital Signal Processor |
| 13:16.47 | Oins | ok, have i understand that right (sorry, my english is not the best) that the Dialogic Cards have there own CPU, the Zaptel usw the PC cpu? |
| 13:17.44 | X-Gen | correct |
| 13:18.00 | *** join/#asterisk B4 (n=B4@202.69.48.245) |
| 13:18.48 | B4 | trixter: it worked :) |
| 13:18.50 | Oins | ok and this should mean that the dialogic cards are much dearer then zaptel? |
| 13:18.59 | X-Gen | correct |
| 13:19.04 | Oins | ok, thank you |
| 13:19.10 | B4 | was very simple to configure indeed ... 2 mintues total |
| 13:19.17 | *** join/#asterisk popvoxdave (n=popvoxda@c-71-206-59-174.hsd1.md.comcast.net) |
| 13:22.06 | *** join/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com) |
| 13:22.12 | Oins | and a last question :) I want to use asterisk as SIP Phone with a normal analog phone. That means that i use a X100P Compatible modem, connect it with my analog phone and can use the analog phone in combination with asterisk as SIP Phone.. is that right? |
| 13:22.39 | X-Gen | correct |
| 13:23.00 | Oins | ok, thank you ! |
| 13:25.59 | Hmmhesays | Smoked way too many cigarettes last night |
| 13:27.08 | *** part/#asterisk B4 (n=B4@202.69.48.245) |
| 13:27.29 | *** join/#asterisk MrChimpy (n=MrChimpy@212.158.8.162) |
| 13:27.40 | MrChimpy | hi guys |
| 13:28.46 | Katty | Hmmhesays: you gotta stop that before you can't sing anymore. |
| 13:29.36 | Hmmhesays | I know |
| 13:29.38 | Hmmhesays | trying to quit |
| 13:30.33 | MrChimpy | i'm dialling externally by going from one asterisk box over IAX2 to another. in the client box I have this bit of dialplan : |
| 13:30.36 | MrChimpy | exten => _123XXXXXX,1,Dial(IAX2/ivroutdial:odpass@10.1.232.31/${EXTEN:3}) |
| 13:31.22 | MrChimpy | looks like i'm reaching some sort of length limit as the CDR on the other box only shows me trying to dial the first 6 digits of what I dial after 123 |
| 13:31.44 | MrChimpy | any suggestions? |
| 13:31.52 | Hmmhesays | Turn your computer off? |
| 13:32.11 | MrChimpy | thanks. |
| 13:33.01 | MrChimpy | anyone more useful around? |
| 13:33.23 | Hmmhesays | LOL |
| 13:34.36 | fenlander | uh, MrChimpy, aren't you striping the 123 with EXTEN:3? |
| 13:34.39 | Hmmhesays | Now i'm not going to help you |
| 13:35.00 | MrChimpy | yes. that's what the exten:3 is for. |
| 13:35.07 | MrChimpy | I dial 123<phonenumber> |
| 13:35.12 | fenlander | then you are only matching 6 digits |
| 13:35.25 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.235.254.Dial1.SanJose1.Level3.net) |
| 13:35.30 | MrChimpy | but <phonenumber> when it gets to the other end is truncated to only 6 digits |
| 13:35.43 | fenlander | which is what you match with _123XXXXXX |
| 13:35.44 | MrChimpy | ah! |
| 13:35.45 | MrChimpy | I see! |
| 13:35.52 | MrChimpy | silly me |
| 13:35.56 | fenlander | :-) |
| 13:35.57 | Hmmhesays | I think i've reached the end of the internet |
| 13:36.17 | MrChimpy | I need . instead |
| 13:36.32 | fenlander | _123X. will match any number of digits - yes |
| 13:36.47 | MrChimpy | X. or .? |
| 13:37.37 | *** join/#asterisk hypnox (n=dan@cornelyn.force9.co.uk) |
| 13:37.58 | hypnox | hi guys, how can asterisk be told to try and register with an iax peer stored in realtime ? |
| 13:38.01 | fenlander | depends if you want to dial 123 by itself I guess |
| 13:38.03 | ClayReiche12 | Anyone have any idea about my problem? |
| 13:38.30 | fenlander | either should work for you |
| 13:38.46 | MrChimpy | aye, thanks |
| 13:39.04 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 13:40.34 | MrChimpy | secondary question, when I specify stuff like Zap/g1 in dial commands, I assume those groups are those defined in zapata.conf, and I can put my 4 E1s in just one group? |
| 13:42.40 | hypnox | anyone? |
| 13:44.09 | *** join/#asterisk ionix (i=ionix@p5008-ipbfp04miyazaki.miyazaki.ocn.ne.jp) |
| 13:44.22 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 13:45.16 | *** part/#asterisk nortex (n=nortex@64.136.65.142) |
| 13:46.02 | ClayReiche12 | hypnox: maybe you could reload iax module with a -rx command? |
| 13:46.40 | ClayReiche12 | probably not what your looking for though.... |
| 13:48.27 | *** join/#asterisk engie (n=se204@servalan.ecs.soton.ac.uk) |
| 13:50.00 | engie | Hi. I'm looking to hook two cheap dect systems together to allow an intercom type system between many handsets. If I use a X101P card for each dect receiver, would a 400Mhz machine be beefy enough to pipe calls between them? |
| 13:52.31 | Hmmhesays | how many |
| 13:53.00 | *** join/#asterisk fishboy1669 (i=proxyuse@62.69.81.129) |
| 13:53.43 | engie | Hmmhesays: I guess there will only be 1 call at a time between two sets of dect handsets |
| 13:54.12 | ClayReiche12 | Hmmhesays: You seem to be the MAN in here... any ideas about my Privacy screener problem? |
| 13:54.32 | *** join/#asterisk [TK]D-Fender (n=joe@CPE000d3a2c3061-CM00080d8dba84.cpe.net.cable.rogers.com) |
| 13:55.08 | Hmmhesays | I am? |
| 13:55.24 | ClayReiche12 | The only one responding.... |
| 13:55.40 | Hmmhesays | paste your problem |
| 13:56.01 | ClayReiche12 | <PROTECTED> |
| 13:56.22 | Hmmhesays | the sun and vaginas are only mythical things i've read about on the internet |
| 13:56.36 | ClayReiche12 | Dial(IAX2/username:password@76.33.88.2/8134166290|30|gM(screen^${SCREEN_FILE})&IAX2/username:password@76.33.88.2/8137491444|30|gM(screen^${SCREEN_FILE})) |
| 13:57.04 | ClayReiche12 | That's the command |
| 13:57.09 | Hmmhesays | Ugh, why are you calling dial from an agai |
| 13:57.10 | Hmmhesays | *agi |
| 13:57.40 | ClayReiche12 | Because the "ring-to" numbers are dynamic and pulled from a database |
| 13:57.43 | MrChimpy | nothing wrong with agi! |
| 13:57.53 | Hmmhesays | I didn't say there was anything wrong with an agi |
| 13:57.57 | MrChimpy | well, apart from it being marginally insane :) |
| 13:58.11 | MrChimpy | i use dial lots from agi |
| 13:58.17 | Hmmhesays | I'm sorry |
| 13:58.30 | Hmmhesays | I've always found it to be a fantastic pain in the ass |
| 13:58.42 | ClayReiche12 | When I dial one right after the other it works great.... when I try to "Blast" with the "&" it doesn't. |
| 13:58.46 | MrChimpy | can't drop in and out of dialplan in some apps |
| 13:59.12 | file | ClayReiche12: probably because you're using it wrong |
| 13:59.20 | *** join/#asterisk funxion (n=nunya@63.214.236.169) |
| 13:59.23 | ClayReiche12 | I'm trying to avoid Forking a new process for each "Ring-To" number.... |
| 13:59.45 | file | Dial(IAX2/username:password@76.33.88.2/8134166290&IAX2/username:password@76.33.88.2/8137491444|30|gM(screen^${SCREEN_FILE}) |
| 13:59.47 | Hmmhesays | oh come on file, its friday I wanted to play a little |
| 14:00.13 | ClayReiche12 | file: That may be... however when I run the same command from extensions.conf it works great. Only seems to happen with the agi |
| 14:00.33 | file | show me the Dial line that you see on the CLI when done from extensions.conf |
| 14:00.47 | MrChimpy | clay: tried using dial via EXEC? |
| 14:00.51 | MrChimpy | just an idea |
| 14:02.28 | MrChimpy | i had to do it that way to get any sense out of it |
| 14:02.42 | MrChimpy | straight AGI dial is broken |
| 14:04.09 | Hmmhesays | Seriously Strider rocks |
| 14:04.30 | Hmmhesays | No IF's AND's or But's about it |
| 14:04.44 | ClayReiche12 | file: from cli from extensions.conf:"IAX2/username:password@76.33.88.2/8134166290|30|gM(screen^/tmp/8137491400-1151675565)&IAX2/username:password@76.33.88.2/8137491444|30|gM(screen^/tmp/8137491400-1151675565)" |
| 14:05.18 | file | I mean what do you see on the CLI |
| 14:05.22 | file | when you actually use that |
| 14:05.24 | MrChimpy | clay: seriously, use exec. broken stuff suddenly starts working |
| 14:05.40 | ClayReiche12 | MrChimpy: I am using exec. |
| 14:05.50 | MrChimpy | ah, i'm no use then |
| 14:05.55 | ClayReiche12 | MrChimpy: didn't work at all with dial |
| 14:06.07 | file | see, the format for a dial line is Dial(Tech/blah&Tech/blah&Tech/blah|timeout|options) |
| 14:06.34 | ClayReiche12 | file: that is from the cli output |
| 14:06.41 | ClayReiche12 | you want more? |
| 14:06.50 | Hmmhesays | formatting it right might help you |
| 14:06.56 | MrChimpy | clay: ah, ok. that's all i've got :) |
| 14:06.57 | file | did you try doing it the right way like I said? |
| 14:07.07 | Hmmhesays | File, will you buy me a new guitar? |
| 14:07.09 | ClayReiche12 | Thanks MrChimpy. |
| 14:07.13 | file | Hmmhesays: nah |
| 14:07.22 | Hmmhesays | Fine |
| 14:07.26 | file | Hmmhesays: those don't come complimentary at Telcomjoshvoxmart |
| 14:08.03 | Hmmhesays | Hmm, I don't have $2500 to drop on the one I want though... and I'm too ugly to be a hooker |
| 14:09.43 | ClayReiche12 | file: I'm sorry. I don't see the difference between what you typed and what I typed. |
| 14:09.54 | file | you can't have two sets of timeouts and options |
| 14:10.05 | file | which you do |
| 14:10.06 | Hmmhesays | well... you can but it won't work right |
| 14:10.30 | *** join/#asterisk coppice (n=chatzill@61.197.17.210.dyn.pacific.net.hk) |
| 14:10.46 | ClayReiche12 | file: ahh... I see now... sorry. |
| 14:10.52 | ClayReiche12 | let me try that |
| 14:11.05 | *** join/#asterisk }btorch{ (n=btorch@208.63.19.179) |
| 14:11.33 | }btorch{ | hello |
| 14:11.48 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 14:11.55 | *** join/#asterisk Synthe (i=Synthe@odo.synthe.net) |
| 14:12.27 | }btorch{ | is a T1 enough for say 60-80 SIP phones ? |
| 14:12.53 | Hmmhesays | what a fantastically generic question |
| 14:12.58 | jbalcomb | }btorch{ yes |
| 14:13.05 | }btorch{ | hehe :-) |
| 14:13.11 | fishboy1669 | is there anyone here who uses tdm400p cards in uk? |
| 14:13.41 | Hmmhesays | how many simultaneous calls, what codec to plan on using |
| 14:13.49 | }btorch{ | what about 30 of those phones make cocurrent calls |
| 14:13.54 | Hmmhesays | s/to/do you |
| 14:14.15 | file | and are you talking for internet access, or a channelized T1 for phone calls... |
| 14:14.27 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 14:14.27 | *** mode/#asterisk [+o russellb] by ChanServ |
| 14:14.47 | coppice | if the 60-80 SIP phones are in a trapist monastery, then probably a single channel FXO is enough |
| 14:14.49 | MrChimpy | it's like "is a car big enough for my family?" |
| 14:15.00 | file | MrChimpy: if the car contains a black hole, yes |
| 14:15.12 | Hmmhesays | well honestly most of the time if that question is being asked, not enough research has been done |
| 14:15.24 | coppice | dumping my kinds into a black hole - now that's tempting :-) |
| 14:16.16 | iCEBrkr | T-Mobiles MDA sucks rocks. |
| 14:16.23 | iCEBrkr | Random FYI |
| 14:16.30 | coppice | why? |
| 14:16.35 | file | god everything sucks for people, no matter what you can't please every single person out there |
| 14:16.37 | iCEBrkr | The interface sucks ass |
| 14:16.42 | fishboy1669 | icerbreke |
| 14:16.46 | iCEBrkr | It relies on the damn stylus too much |
| 14:16.52 | coppice | its WinCE. of course it sucks |
| 14:16.54 | iCEBrkr | I'm spoiled I guess with my SK-2 |
| 14:16.54 | fishboy1669 | u need to upgrade it |
| 14:17.04 | iCEBrkr | fishboy1669: Upgrade it? I just got it |
| 14:17.14 | fishboy1669 | the new fw is really good |
| 14:17.17 | iCEBrkr | I'm thinking about returning this PoS and getting the SK3 |
| 14:17.17 | coppice | WinCE only exists to make other Microsoft products look good |
| 14:17.32 | fishboy1669 | wince is ok |
| 14:17.37 | iCEBrkr | fishboy1669: It's Windows Mobile 5.0 |
| 14:17.52 | coppice | that's worse than 2003 |
| 14:18.00 | fishboy1669 | my mate got one was peed off with it got it upgraded and works really well now |
| 14:18.06 | fishboy1669 | doesnt even have to overclock it |
| 14:18.26 | file | iCEBrkr: you're spoiled with the Sidekick since you're used to not using a stylus |
| 14:18.59 | iCEBrkr | file: well how the hell do you use this thing with one hand? |
| 14:19.03 | coppice | The WinCE philosophy: you have to buy a calculator for your phone :-) |
| 14:19.17 | iCEBrkr | There's no 'back' button on this bitch |
| 14:19.22 | file | carefully! |
| 14:19.34 | iCEBrkr | I'm so not impressed. |
| 14:19.51 | iCEBrkr | I may have to just suck it up and deal with having to hack my SK3 to install apps and free ringers. |
| 14:20.30 | coppice | HTC is growing very fast making basically the only WinCE machines to have ever sold well. However, i've never met a buyer who does say, nice hardware but the OS is a POS |
| 14:20.43 | *** join/#asterisk knobo (n=Knut@85.196.83.87) |
| 14:20.45 | iCEBrkr | coppice: True |
| 14:20.48 | iCEBrkr | The UI sucks ass |
| 14:21.01 | iCEBrkr | You have to scroll menus to do shit |
| 14:21.17 | knobo | which driver is used for Eicon Networks Corporation Diva 2.02 PCI S/T ? |
| 14:21.29 | coppice | Any Symbian Series 60 is pretty hard to be worse than :-) |
| 14:21.53 | *** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
| 14:21.53 | coppice | But WinCE sails past |
| 14:22.23 | Hmmhesays | I think winCE used to run on the sega dreamcast |
| 14:23.00 | *** join/#asterisk mog (n=mogorman@gateway.digium.com) |
| 14:23.13 | iCEBrkr | fishboy1669: apparently 'upgrades' are only available for UK versions |
| 14:24.34 | }btorch{ | file, no it's a T1 for internet access .. the reason I ask is because right now I got two offices with seperate T1s since they are a couple of other offices apart and I'm planning to connected them using a wireless bridge and get rid of one T1 |
| 14:25.51 | *** join/#asterisk Torginator (n=Chris@c-67-190-204-43.hsd1.mn.comcast.net) |
| 14:26.13 | Torginator | Morning all! |
| 14:26.31 | *** join/#asterisk RoyK[de] (n=roy@static.148.bras.breisnet.com) |
| 14:26.48 | Torginator | me again, searching for answers to life's questions.... about Asterisk, anyway. |
| 14:26.54 | ClayReiche12 | file: Thanks! I'm on the right track now... don't have it working yet but it's no longer sending me to Allisons provacy screening... |
| 14:26.54 | Torginator | hehehe just like my cat. |
| 14:27.00 | ClayReiche12 | MrChimpy: Thnks! |
| 14:27.20 | Torginator | (the licking part, not the life's questions part) |
| 14:27.30 | Torginator | oh.... kick. nevermind. |
| 14:27.35 | Torginator | she does that too. |
| 14:28.03 | Hmmhesays | Cat is good on the grill |
| 14:28.06 | Torginator | Anyway. I have a phone that tries to register with Asterisk, which gives Registration from '<sip:601@192.168.1.25;user=phone>' failed for '192.168.1.26' - Username/auth name mismatch |
| 14:28.21 | Hmmhesays | I bet there is NAT in there |
| 14:28.38 | Torginator | uhm... the asterisk box is natted to the internet, yes, but the phone is on the same hub. |
| 14:29.16 | Torginator | Asterisk takes incoming cals just fine. "Congratulations! You have .... blah blah" but the phone won't register. |
| 14:29.49 | Hmmhesays | don't use a secret is the easiest way around that |
| 14:30.19 | *** join/#asterisk pnlarsson (n=niklas@c83-248-2-120.bredband.comhem.se) |
| 14:30.46 | Katty | oh god. |
| 14:30.54 | Katty | someone kidnap me, quick. |
| 14:30.56 | Torginator | ok.... so I took it out of sip.conf... and it's blank on the phone. Still same response |
| 14:31.36 | Torginator | But on the phone (Grandstream Budgetone 100 series), I'm not sure which of SIP User ID, Authenticate ID or Name are matched to which fields in sip.conf. |
| 14:31.52 | RaYmAn-Bx | leave auth name blank on the phone (but keep username)..and make sure the name of the section in sip.conf is the name as the name in username= |
| 14:32.51 | *** join/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx) |
| 14:32.53 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 14:32.59 | Torginator | RaYmAn-Bx: uhm... no "auth name" or uhm... let me work this out. |
| 14:33.14 | rene- | hello, i am looking for a whisper mode integrator |
| 14:33.19 | RaYmAn-Bx | Torginator: authenticate id |
| 14:33.29 | rene- | for asterisk |
| 14:33.58 | Torginator | yeah. No "Auth name" or "user name"... so Authenticate should be blank? And the "Name" field on the phone should match the [blah] section name? |
| 14:34.30 | RaYmAn-Bx | and the [blah] section name should match the username in sip.conf in that section |
| 14:34.46 | Torginator | cool. trying... |
| 14:35.57 | *** join/#asterisk Kokey (n=jramirez@201.123.192.227) |
| 14:36.01 | *** join/#asterisk l-fy (n=pchitesc@yate/developer/l-fy) |
| 14:36.03 | l-fy | hello |
| 14:36.23 | l-fy | i need some help to setup the iax trunking in asterisk |
| 14:36.33 | l-fy | actually i want to tell to a certain extension on iax to do trunking |
| 14:36.38 | l-fy | how can i do that |
| 14:37.26 | Torginator | RaYmAn-Bx: Thanks. I have to wait for someone on site to reset the phone. |
| 14:38.06 | MikeJ[Laptop] | l-fy, setup a peer |
| 14:38.17 | l-fy | MikeJ[Laptop] > ok but how? |
| 14:38.24 | MikeJ[Laptop] | in iax.conf |
| 14:38.37 | MikeJ[Laptop] | like the samples |
| 14:38.38 | l-fy | ok |
| 14:38.41 | l-fy | i have something like this |
| 14:38.56 | l-fy | exten => 899,1,Dial(IAX2/demo:abc@192.168.168.12/100) |
| 14:39.01 | l-fy | how can i setup a peer? |
| 14:39.06 | *** join/#asterisk erik2 (n=eanders@65-102-92-135.sxfl.qwest.net) |
| 14:39.11 | *** join/#asterisk nortex (n=nortex@64.136.65.142) |
| 14:39.23 | MikeJ[Laptop] | you want it just for that one ext |
| 14:39.28 | MikeJ[Laptop] | or for the whole host? |
| 14:39.49 | l-fy | for just extension |
| 14:39.58 | l-fy | when i send a call there at that number to be trunk |
| 14:40.04 | l-fy | or at that prefix |
| 14:40.59 | Hmmhesays | Torginator: sip reload dude |
| 14:41.27 | *** join/#asterisk ptinsley (n=ptinsley@209.12.249.243) |
| 14:41.29 | *** join/#asterisk Luke-Jr (n=luke-jr@2002:1891:f657:0:20e:a6ff:fec4:4e5d) |
| 14:43.08 | *** join/#asterisk fourcheeze (n=rich@82.153.215.21) |
| 14:43.21 | fourcheeze | any tips on getting MOH to sound reasonable over g729? |
| 14:44.56 | coppice | music will always sounds awful over G.729 |
| 14:45.00 | *** join/#asterisk gandhijee (n=gandhije@mail.win-ent.com) |
| 14:45.07 | coppice | you can always subtract 18 |
| 14:45.12 | file | <PROTECTED> |
| 14:45.28 | coppice | singing won't sound so bad |
| 14:45.42 | coppice | as long as its just your voice, and no backing |
| 14:45.54 | file | coppice: you ruined my joke |
| 14:45.59 | fourcheeze | hehe |
| 14:46.12 | *** join/#asterisk OuterSpace (n=me1@168.226.4.248) |
| 14:46.16 | fourcheeze | got a customer wants Beethoven |
| 14:46.19 | fourcheeze | sounds terrible |
| 14:46.30 | fourcheeze | I find trance stuff sounds best |
| 14:46.31 | cypromis | I doubt it makes a big difference to hear file singing over G711 or g729 or even g723.1 |
| 14:46.31 | file | well it's a codec designed to compress voice |
| 14:46.41 | cypromis | :P |
| 14:46.47 | MikeJ[Laptop] | ilbc? |
| 14:46.50 | file | cypromis: LPC10! |
| 14:46.59 | fourcheeze | sure, I realise g729 is going to be bad but there must be some tips for getting the most out of it |
| 14:47.05 | fourcheeze | I tried compressing which helped a bit |
| 14:47.22 | fourcheeze | then I boost the low and high frequencies |
| 14:47.27 | fourcheeze | which also helps a bit |
| 14:47.27 | coppice | subtract 18 is the only tip I can give |
| 14:47.33 | OuterSpace | hi, i have no problems with iax clients, but now installed a sip ip phone, when i call him, he can listen to me, i cant listen him, he cant call me either, tips ? |
| 14:47.41 | fourcheeze | coppice: that one doesn't work any better the next time |
| 14:48.05 | rene- | coppice: what does subtract means? |
| 14:48.06 | fourcheeze | is it possible to change codec for putting on hold? |
| 14:48.17 | rene- | in that context? |
| 14:48.27 | fourcheeze | rene-: take away in a mathematical sense |
| 14:48.28 | coppice | rene-: its kinda the converse of addition |
| 14:48.32 | fourcheeze | 729-18=711 |
| 14:48.38 | rene- | ahh |
| 14:48.41 | rene- | ulaw alaw |
| 14:48.50 | fourcheeze | yeah |
| 14:48.51 | rene- | is that what you meant? |
| 14:48.53 | *** join/#asterisk nexstar (n=nexstar@adsl-67-112-181-25.dsl.lsan03.pacbell.net) |
| 14:48.54 | rene- | ok |
| 14:49.06 | fourcheeze | ok, so just out of interest how would I change codec during a call? |
| 14:49.43 | *** part/#asterisk nexstar (n=nexstar@adsl-67-112-181-25.dsl.lsan03.pacbell.net) |
| 14:49.46 | *** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 14:50.04 | *** join/#asterisk nexstar (n=nexstar@adsl-67-112-181-25.dsl.lsan03.pacbell.net) |
| 14:50.21 | OuterSpace | please dont ignore my question :-( |
| 14:50.34 | *** join/#asterisk anonymouz666 (i=anonymou@200.218.196.5) |
| 14:50.38 | coppice | it doesn't matter much what you do with filtering and other fun and games G.729 is gonna sound bad for anything which is not a single clean voice. The best you can do is get closer to that. Really simple things like a solo acoustic guitar don't screw up nearly as much as more complex music |
| 14:50.58 | nexstar | anyone here use trixbox |
| 14:51.01 | nexstar | ? |
| 14:51.04 | rene- | you would need to put the first call on hold an establish a new one i think |
| 14:51.08 | file | OuterSpace: are you behind NAT? |
| 14:51.21 | OuterSpace | yes, both of us are |
| 14:51.47 | file | is your Asterisk behind NAT? |
| 14:51.59 | OuterSpace | yes, but i forwared all open ports |
| 14:52.06 | file | did you configure sip.conf? |
| 14:52.09 | nexstar | there is an option within trixbox to create files for cisco phones, but im using polycom phones, is there a module i can load for that? just to make it easier in the future? |
| 14:52.21 | OuterSpace | yes, he can connect, and he hear me talking |
| 14:52.31 | OuterSpace | when i call him |
| 14:52.54 | file | did you set externip/externhost and localnet? |
| 14:53.12 | rene- | is gsm good enough for music on hold? |
| 14:53.15 | Torginator | Hmmhesays: sip reload huh? |
| 14:53.26 | OuterSpace | its dinamic ip |
| 14:53.27 | *** part/#asterisk h0g (n=jharley@216.235.10.210) |
| 14:53.32 | OuterSpace | it changes |
| 14:53.37 | file | you still have to tell Asterisk what your IP is somehow |
| 14:54.00 | file | it sends that to the phone, and the phone sends audio to there... so right now the phone is probably sending the audio to your private IP, which is inaccessible |
| 14:54.23 | OuterSpace | no, its sending to public ip, fixed that on ip phone |
| 14:54.47 | OuterSpace | on debug i got: chan_sip.c: Auto destroying call 'vpUspCnspeo7yo5d@201.240. |
| 14:54.47 | nexstar | there is an option within trixbox to create files for cisco phones, but im using polycom phones, is there a module i can load for that? just to make it easier in the future? |
| 14:54.47 | file | pastebin a sip debug of a failing call |
| 14:54.59 | file | and I'll show you. |
| 14:55.00 | OuterSpace | call is sucessfull |
| 14:55.14 | OuterSpace | its just i can hear him, but he can hear me |
| 14:55.16 | file | apparently not if you're getting one way audio |
| 14:55.18 | coppice | gsm is fairly nasty for music, but compared to G.729 you could consider it Hi-Fi :-) |
| 14:55.37 | file | one would consider that a failed call, unless you don't want to hear him |
| 14:56.21 | *** join/#asterisk Spy000007 (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
| 14:56.24 | OuterSpace | heh, sorry, ill check nat conf, thanks a lot |
| 14:56.36 | coppice | half the time people complain the audio echoes back, and then when it doesn't echo back they still complain |
| 14:56.50 | nexstar | what causes the echo back? |
| 14:56.50 | Torginator | heh |
| 14:57.52 | fourcheeze | coppice: yeah I like gsm - if our outbound SIP people supported it I would use it |
| 15:00.24 | fishboy1669 | any guys from uk here? |
| 15:00.55 | darkskiez | bit sexist |
| 15:01.01 | fishboy1669 | ? |
| 15:01.15 | fishboy1669 | any humans from uk here? |
| 15:01.22 | fishboy1669 | is that better ;-) |
| 15:01.24 | darkskiez | yes, 'sup |
| 15:01.29 | darkskiez | :] |
| 15:01.40 | fishboy1669 | asl? lol |
| 15:01.41 | fishboy1669 | he he |
| 15:01.56 | fishboy1669 | back to reality |
| 15:02.07 | fishboy1669 | have u experience of using a tdm400p card |
| 15:02.21 | darkskiez | yes |
| 15:02.22 | fishboy1669 | im trying to get mine working with a bt line but having issues |
| 15:02.25 | *** join/#asterisk eKo1 (n=bernd@190.4.7.90) |
| 15:02.32 | darkskiez | ah, only used it with handsets |
| 15:02.40 | fishboy1669 | you wouldnt happen to have a zapata.conf file |
| 15:02.42 | *** join/#asterisk vechers-away (n=svecher@64.61.117.139) |
| 15:02.42 | fishboy1669 | oh dow |
| 15:02.59 | darkskiez | whats the prob anyway |
| 15:03.05 | fishboy1669 | hangups |
| 15:03.20 | fishboy1669 | but dont know if its the zap or the sip |
| 15:03.27 | fishboy1669 | think i have multiple issues |
| 15:03.30 | fishboy1669 | so hard to track down |
| 15:03.36 | darkskiez | the debug console will tell you that |
| 15:03.54 | fishboy1669 | ye been using that a bit but faults are speradic |
| 15:04.13 | fishboy1669 | i was using a tecom ip2006 phone |
| 15:04.24 | fishboy1669 | but it wansnt sending out hangup signals |
| 15:04.33 | fishboy1669 | now im on a polycom 300sip |
| 15:04.43 | fishboy1669 | but similar issues and some extra ones |
| 15:04.53 | fishboy1669 | also changed location so different bt line |
| 15:05.20 | fishboy1669 | just wanted a config so i knew that that wasnt the issue cuts down on the fault tinding then |
| 15:05.20 | darkskiez | bt use polarity reversal afaik, so you should have hanguponpolarityswitch=yes, but that shouldnt mis-trigger. |
| 15:05.52 | fishboy1669 | ah i looked in the bt sin/spin and they say they dont |
| 15:05.57 | *** part/#asterisk sevard (n=sev@adsl-71-129-115-244.dsl.irvnca.pacbell.net) |
| 15:05.59 | fishboy1669 | no wonder im confused lol |
| 15:06.17 | fishboy1669 | ill put the porarity back on then |
| 15:06.35 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 15:07.14 | fishboy1669 | would be nice if i could find a standard zapata.conf that works and go from there |
| 15:08.01 | *** join/#asterisk ToyMan (n=stuq@74-32-9-135.dsl1.mdl.ny.frontiernet.net) |
| 15:08.42 | eKo1 | fishboy1669: that is impossible given the amount of different setups... |
| 15:10.02 | *** join/#asterisk CoffeeIV_ (n=CoffeeIV@www.airlinksystems.com) |
| 15:10.30 | fishboy1669 | but there could be a basic set up for bt |
| 15:11.19 | eKo1 | bt? |
| 15:11.38 | fishboy1669 | british telecom |
| 15:11.51 | fishboy1669 | the standard telephone line in uk |
| 15:11.58 | eKo1 | fishboy1669: I see. Well, if and when you do get it working, feel free to put the setup somewhere on the web. |
| 15:12.10 | fishboy1669 | i will |
| 15:12.13 | fishboy1669 | wiki |
| 15:12.17 | eKo1 | Great. |
| 15:12.27 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:12.39 | fishboy1669 | the big question is if i ever get the dam thing to work |
| 15:12.41 | fishboy1669 | :( |
| 15:13.03 | fishboy1669 | feel im getting nowhere at mo out of my depth of knowlege |
| 15:13.24 | *** join/#asterisk jamincollins (n=jcollins@ptech7-44.acdmis.com) |
| 15:13.28 | eKo1 | fishboy1669: I feel like that everyday... |
| 15:13.34 | fishboy1669 | lol |
| 15:13.53 | fishboy1669 | sometimes microsoft is appealling |
| 15:13.58 | fishboy1669 | just gui point and click |
| 15:14.07 | fishboy1669 | and so many manuals to give u info |
| 15:14.23 | fishboy1669 | linux and open source sometimes seems such a black art |
| 15:14.36 | Hmmhesays | fc5 is not |
| 15:14.45 | coppice | i've always found windows a far worse black art |
| 15:14.56 | fishboy1669 | im using fc5 |
| 15:15.07 | fishboy1669 | suse is my prefered distro |
| 15:15.21 | *** part/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 15:15.38 | coppice | i prefer the one that's set up and working |
| 15:15.48 | jamincollins | I'm trying to connect to Asterisk boxes via a T1 tie-line between them. Using a Sangoma A101 in one and a TE110P in the other... I'm able to pass calls between the two, but the circuit periodically alarms and resyncs |
| 15:16.30 | coppice | is one configured as clock master, and one as slave? |
| 15:16.52 | eKo1 | jamincollins: sounds like a clock source issue |
| 15:16.57 | jamincollins | I've looked for documentation on configuring one side or the other as a master clock source, but so far haven't found anything |
| 15:17.02 | fishboy1669 | cioouce if only life was that easy lol |
| 15:17.25 | jamincollins | I have them both configured as em_w for signalling |
| 15:17.29 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 15:17.40 | coppice | look in the sample zaptel.conf. it tells you how to set the clocking |
| 15:18.43 | jamincollins | Isn't that just for that box... not for the circuit? |
| 15:19.08 | coppice | you need to make one box the master and one box the slave |
| 15:19.33 | jamincollins | for instance, if I configure span 1 with the second parameter of "1" then it's the primary clock source for that system... |
| 15:20.04 | jamincollins | right... but how is that accomplished... I don't see anything in the zaptel.conf that appears to do that |
| 15:20.21 | *** join/#asterisk variable_office (n=variable@Adv-Proprietary-Systems.s7-0-0.2-15-0.ar4.CHI1.gblx.net) |
| 15:20.32 | variable_office | when people say DID is that the same as phone number? |
| 15:20.43 | jamincollins | not entirely |
| 15:20.59 | Torginator | ok guys. same thing: Jun 30 10:20:05 NOTICE[30785] chan_sip.c: Registration from '<sip:601@192.168.1.25;user=phone>' failed for '192.168.1.26' - Username/auth name mismatch |
| 15:21.00 | jamincollins | DID (afaik) stands for Direct Inward Dial |
| 15:21.06 | variable_office | i guess i am confused on what a DID is then? |
| 15:21.59 | eKo1 | variable_office: To put it simply, it is a phone number. |
| 15:22.04 | variable_office | in what ways is it not the same as a phone number? |
| 15:22.29 | jamincollins | coppice: what am I missing on the clock source |
| 15:22.35 | eKo1 | a DID is a phone number with special properties |
| 15:22.40 | coppice | jamincollins: you want one box set something like: |
| 15:22.42 | coppice | <PROTECTED> |
| 15:22.43 | coppice | and one set something like |
| 15:22.45 | coppice | <PROTECTED> |
| 15:23.23 | coppice | the parameters will vary, depending whether you are using E1 or T1, CAS or ISDN, etc |
| 15:23.47 | Torginator | http://pastebin.ca/75767 |
| 15:23.47 | jamincollins | I had them set as: span=1,1,0,esf,b8zs and span=1,0,0,esf,b8zs |
| 15:24.05 | eKo1 | jamincollins: that's fine |
| 15:24.08 | *** join/#asterisk Nobbie (n=no@fwb003.fw.is.co.za) |
| 15:24.22 | jamincollins | and yet every few minutes the digium side would red alarm and resync |
| 15:24.28 | Torginator | where is the "user=phone" coming from? |
| 15:24.43 | jamincollins | could it simply be a case of an under powered proc? |
| 15:24.59 | *** join/#asterisk zoa (n=kkk@pirus.securax.be) |
| 15:25.02 | eKo1 | jamincollins: or it could be the cable... |
| 15:25.05 | jamincollins | note: the circuit would alarm even when the system was completely idle |
| 15:25.14 | zoa | hey ho |
| 15:25.35 | Torginator | who's a ho? </groan> |
| 15:25.38 | Torginator | sorry. |
| 15:26.34 | Torginator | ...and the room went silent. |
| 15:26.52 | jamincollins | trying to think of a way to rule out the cable... |
| 15:26.58 | jamincollins | with what I have available... |
| 15:27.34 | jamincollins | voice crossover is what TA68? |
| 15:28.31 | eKo1 | If you want to rule out the cable, try another cable that you know works. |
| 15:28.40 | eKo1 | If you still get the red alarm, then it is not the cable. |
| 15:29.16 | jamincollins | don't have another on hand that I /know/ works... that's the problem... |
| 15:29.46 | Torginator | RaYmAn-Bx still around? |
| 15:29.48 | eKo1 | Make or buy one then. |
| 15:30.15 | RaYmAn-Bx | Torginator: if it didn't work, I'm all out of ideas, but yeah, I'm around |
| 15:30.41 | Torginator | I dunno if it worked, exactly, or what's "on the wire" here's the sip debug: http://pastebin.ca/75767 |
| 15:30.53 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 15:30.57 | Torginator | OK, I do know it didn't work, but not why. |
| 15:31.17 | *** join/#asterisk [Airwolf] (n=airwolf@cp656687-a.landg1.lb.home.nl) |
| 15:31.18 | RaYmAn-Bx | but you get a different error now? |
| 15:31.48 | Torginator | no. Same thing. |
| 15:32.41 | Torginator | Anyone know the difference between "SIP Server" and "Outbound Proxy"? I have them both set to the Asterisk box, on my BT100. |
| 15:33.03 | *** join/#asterisk littleball (n=littleba@cm52.epsilon174.maxonline.com.sg) |
| 15:33.53 | *** part/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 15:34.08 | eKo1 | Think of outbound proxy as an HTTP proxy. |
| 15:34.31 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 15:35.03 | eKo1 | Unless you have one, leave it blank. |
| 15:35.25 | Torginator | ok. And the "SIP User ID"? That should have what? |
| 15:36.09 | Torginator | probably what's in the "username=" field of the section for that phone.... |
| 15:36.51 | eKo1 | Torginator: I think it is time for you to read up on SIP. |
| 15:37.40 | Torginator | maybe. But it would be nice if the names that the phone uses were the same as the ones in sip.conf. |
| 15:38.16 | eKo1 | Torginator: world peace would be nice to but then again... |
| 15:39.35 | Torginator | OK. Got a link to a "everything YOU need to know about SIP" web page? :) |
| 15:40.00 | eKo1 | ~sip |
| 15:40.02 | jbot | i guess sip is http://www.cs.columbia.edu/sip/ X11 PPP dialer interface written in gtk+. URL: http://www.geocities.com/SiliconValley/Campus/3104/sip/ Session Initiation Protocol (see RFC 3261) |
| 15:40.07 | *** join/#asterisk dandan (i=dandan@pacanka.com) |
| 15:40.17 | dandan | re all :) |
| 15:40.19 | Torginator | cool |
| 15:40.28 | dandan | anyone having a polycom software 1.6.6/BR 3.1.3? |
| 15:40.40 | Torginator | the geocities link is dead. |
| 15:40.59 | dandan | ~polycom |
| 15:41.01 | jbot | somebody said polycom was the manufacturer of one of the best IP phones in the market. http://polycom.com - Note: Here is where you can get some downloads: http://www.polycom.com/resource_center/0,,pw-6812-12612,00.html |
| 15:41.04 | *** join/#asterisk slayer192 (n=slayer19@wookie.sundownertrailer.com) |
| 15:42.44 | Hmmhesays | anyone know if the snom 320 supports http proxy's? |
| 15:43.55 | littleball | hello, anyone using spandsp to fax? |
| 15:43.56 | *** join/#asterisk nortex (n=nortex@64.136.65.142) |
| 15:44.16 | littleball | what is the function of digium card used with spandsp to fax out? |
| 15:44.16 | dandan | not too many users today... |
| 15:44.46 | skeffling | Hmmhesays: looks like it, the 320 I have here offers HTTP proxy text box under the Advanced settings |
| 15:46.06 | coppice | littleball: well, how could it work without a telephony card? |
| 15:46.31 | littleball | coppice, of course i see this. just want to introduction about its function |
| 15:46.42 | littleball | just as PSTN interface? |
| 15:46.50 | coppice | yes |
| 15:47.21 | littleball | i am going to try on E1/ulaw |
| 15:47.36 | littleball | coppice, how about the quality? |
| 15:47.56 | coppice | the quality of what? |
| 15:48.03 | Hmmhesays | skeffling can you screenshot that for me? or let me take a look? |
| 15:48.04 | littleball | fax |
| 15:48.25 | littleball | (1)stability. fax out quality.... |
| 15:49.17 | skeffling | Hmmhesays: all it has is a title of HTTP: and under that, User, Password, Auth Scheme, HTTP Proxy, HTTP Port, Register HTTP contact, webserver connection type, Auto Logout.... |
| 15:49.29 | coppice | on a well setup system it works fine. modems don't like anything quirky, though |
| 15:49.46 | skeffling | Hmmhesays: pm me your email, and I'll send a screeshot |
| 15:49.58 | *** join/#asterisk n3glv (n=n3glv@monrovll-cuda1-24-53-251-235.pittpa.adelphia.net) |
| 15:53.45 | Hmmhesays | gmail |
| 15:53.54 | Hmmhesays | you can guess the addy I bet |
| 15:54.29 | gandhijee | where would someone start in examing where zaptel is gettin messed up when its being ported to a diff arch? |
| 15:55.13 | gandhijee | *examining |
| 15:55.32 | Qwell | gandhijee: the code? |
| 15:57.12 | Torginator | eKo1: I had to set "authname=phone" in sip.conf. Dialtone. Thanks for the help, you and all. |
| 15:57.53 | *** join/#asterisk Heimidal (n=Heimidal@phpbb/styles/heimidal) |
| 15:57.59 | *** join/#asterisk IMG-SD (n=IMG-SD@tserv6.imperialgroup.ca) |
| 15:59.04 | gandhijee | yeah most of it compiled cleanly... |
| 15:59.22 | gandhijee | i get a failed to initalize DAA error, in googling that |
| 15:59.33 | gandhijee | it just returned mostly stuff about that error on *BSD |
| 15:59.57 | IMG-SD | Question: Has anyone gotten around the issue where when dialing multiple extensions at once, and one of the dialed extensions has a call forward, that only the forwarded extension will ring? Any way to have Asterisk ignore call forwards (302 moved temporarily) when dialing multiple extensions so that the other extensions will still ring? |
| 16:01.41 | l-fy | ok |
| 16:01.50 | l-fy | how can i get a clock source for zaptel? |
| 16:02.00 | l-fy | zaptel-dummy is still the right way? |
| 16:03.03 | littleball | <PROTECTED> |
| 16:05.17 | twisted[asteria] | l-fy, buy hardware, install hardware, configure hardware, if you want a REAL clock source |
| 16:05.24 | kruz_ | or |
| 16:05.25 | kruz_ | l-fy:ztdummy |
| 16:05.30 | kruz_ | but i agree with twisted[asteria] |
| 16:05.35 | twisted[asteria] | ztdummy is teh evil :P |
| 16:05.46 | kruz_ | ztdummy: is teh gay |
| 16:05.52 | russellb | is the suck |
| 16:06.00 | russellb | darn you text replacement ... |
| 16:06.00 | twisted[asteria] | zaprtc is the better way |
| 16:06.08 | kruz_ | ztdummy read your clock bios if i remember correctly |
| 16:06.15 | kruz_ | bios clock rather* |
| 16:06.21 | kruz_ | and can always come off |
| 16:06.34 | *** join/#asterisk lorinc (n=ang@caracas-0815.adsl.interware.hu) |
| 16:06.45 | kruz_ | i would use it MAYBE if i had a REALLY GOOD NTS, but even then if that part of the network went down, uh oh |
| 16:07.39 | hypnox | eh its not that kind of clock..? |
| 16:07.54 | kruz_ | well, i know |
| 16:08.02 | kruz_ | but u can do clock timing over networks |
| 16:08.13 | kruz_ | not like 8:30 clocks, but real clocks |
| 16:08.14 | Katty | my brain has suddenly insaned. what's the format of an iax2/call@thingy |
| 16:08.18 | Katty | is it that? |
| 16:08.24 | Katty | iax2/username@ip/ext |
| 16:08.50 | kruz_ | does anyone know the video compression used for the new and upcoming video technology and forwarding? i bet that SUCKS bandwith, ima have to fix that. |
| 16:09.03 | Katty | twisted[asteria]: i know you know. |
| 16:09.19 | Katty | twisted[asteria]: and if you'd stop trying to get me to drink quad-espressio shots, you could tell me! |
| 16:10.10 | twisted[asteria] | Katty, lol |
| 16:10.48 | twisted[asteria] | iax2/sucks:balls@server/1234 |
| 16:10.56 | Katty | fantastic, thanks |
| 16:11.07 | twisted[asteria] | hehe |
| 16:11.07 | ptinsley | anybody done a gateway to verizon wireless? |
| 16:11.39 | eKo1 | kruz_: h.264 ? |
| 16:12.01 | Katty | twisted[asteria]: give me a call? |
| 16:12.01 | kruz_ | eKo1: is that one of them, that and iax2 supports it doesnt it |
| 16:12.08 | Katty | twisted[asteria]: i'll get the iax number to ya |
| 16:12.09 | kruz_ | eKo1: is that the actual compression? |
| 16:12.19 | Katty | twisted[asteria]: or is your firewall still hurling? |
| 16:12.23 | twisted[asteria] | mmm...booty call^H^H^H^H^H^H^H^H^H^H |
| 16:12.30 | Katty | twisted[asteria]: i swear, if it's not your email server it's your firewall |
| 16:12.40 | n3glv | I thought it was h.323? |
| 16:12.47 | kruz_ | i think it is |
| 16:12.48 | Katty | twisted[asteria]: you've downed one too many shots of starbucks liquor |
| 16:12.52 | twisted[asteria] | actually, we have iax blocked at the firewall |
| 16:12.55 | Katty | )= |
| 16:12.57 | kruz_ | but is that the protocol or compression?? |
| 16:13.00 | Katty | that makes me all sad inside. |
| 16:13.07 | Katty | file, dear, i need you to call me |
| 16:13.16 | Katty | file: I MISS YOUR VOICE |
| 16:13.17 | n3glv | connection protocol, llike sip |
| 16:13.20 | twisted[asteria] | WHOA |
| 16:13.32 | kruz_ | right, thats what i thought |
| 16:13.34 | file | Katty: ooh? |
| 16:13.37 | variable_office | what is the most commonly accepted codec amongst ip phones? |
| 16:13.39 | kruz_ | who regulates the video compression? |
| 16:13.41 | Hmmhesays | and they'll make you call fellation a trouser friendly kiss |
| 16:13.42 | Katty | file: yes please. do you still have my iax number? |
| 16:13.46 | kruz_ | is it mpeg2? |
| 16:13.49 | Hmmhesays | lpc10 |
| 16:13.50 | n3glv | ulaw |
| 16:13.51 | file | I might |
| 16:13.53 | file | lemme looksee |
| 16:13.56 | Katty | k |
| 16:14.10 | Hmmhesays | I do if you're at the same IP |
| 16:14.23 | kruz_ | what is it resolution/kbps wise, pretty decent? |
| 16:14.23 | *** join/#asterisk PerlStalker (n=PerlStal@firewall.falconsroost.alamosa.co.us) |
| 16:14.29 | Katty | Hmmhesays: i think we might have actually changed ips. |
| 16:14.32 | jamincollins | anyone know if a fluke 620 can test a t1 cross over cable? and if so, what the configuration would be? |
| 16:14.45 | twisted[asteria] | www.fluke.com |
| 16:15.12 | jamincollins | twisted[asteria]: thanks, but I've already looked there |
| 16:15.18 | file | Katty makes weird sounds |
| 16:15.24 | variable_office | jamincollins you could make an adapter to make it work |
| 16:15.26 | eKo1 | kruz_: http://en.wikipedia.org/wiki/H.264 |
| 16:15.38 | kruz_ | eKo1: thanx man |
| 16:15.58 | n3glv | ahh |
| 16:16.08 | n3glv | arigato |
| 16:16.09 | kruz_ | eKo1: could you implement mpeg streaming also? or is it not routable like h264, and is the video compressioned defined in the protocol standards? |
| 16:16.13 | n3glv | domo |
| 16:16.30 | kruz_ | eKo1:nvm, i just read the first line ;] |
| 16:16.46 | PerlStalker | Quick question. Is is possible to disable the ability for forward voice mail to other users? |
| 16:16.56 | eKo1 | jamincollins: what is a fluke 620? |
| 16:17.06 | n3glv | what abt gabcast? mp3? |
| 16:17.16 | jamincollins | eKo1: it's a cable tester |
| 16:17.31 | *** join/#asterisk marv0997 (i=marv0997@190.4.2.83) |
| 16:18.08 | n3glv | so why is iax firewalled? it's a great protocol |
| 16:18.24 | twisted[asteria] | it's also holy |
| 16:18.27 | twisted[asteria] | er hole-y |
| 16:18.29 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 16:18.52 | n3glv | insecure? |
| 16:19.22 | variable_office | how can you tell what codecs are installed? |
| 16:19.22 | eKo1 | n3glv: read the topic |
| 16:19.40 | n3glv | ahhhhh |
| 16:19.44 | n3glv | eeek |
| 16:20.00 | ptinsley | that reminds me I think I forgot about a pbx, whoops |
| 16:20.15 | twisted[asteria] | n3glv, heh... it's only the 2nd major security hole found in iax |
| 16:20.19 | twisted[asteria] | otherwise, it's a great protocol |
| 16:20.27 | l-fy | http://yate.null.ro/pmwiki/index.php?n=Main.Routing |
| 16:20.36 | n3glv | yeah, almost as secure as SENDMAIL .-0 |
| 16:20.41 | twisted[asteria] | haha |
| 16:20.41 | l-fy | damn |
| 16:20.41 | ptinsley | lol |
| 16:20.43 | l-fy | wrong window |
| 16:20.45 | l-fy | sorry |
| 16:20.57 | ptinsley | or MS dcom |
| 16:22.15 | ptinsley | i wonder which one would win if you added up all the vulns dcom or sendmail |
| 16:23.13 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 16:23.31 | *** join/#asterisk salviadud (n=ralfalfa@201.135.2.210) |
| 16:23.45 | kruz_ | or myspace. |
| 16:24.02 | kruz_ | fusion+aspx=??what where you thinking? |
| 16:24.20 | kruz_ | no field entry filtering=come on. |
| 16:24.29 | kruz_ | its a little better now. |
| 16:25.25 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 16:28.40 | *** join/#asterisk mtaht4 (n=m@reserve-64-79-113-254.wiline.com) |
| 16:31.29 | *** join/#asterisk Trazz (i=Trazz@c-67-163-92-37.hsd1.il.comcast.net) |
| 16:32.08 | Trazz | does anyone recommend any companies that can put together the configuration files you need for a small system? |
| 16:32.20 | Qwell[] | Trazz: any of the hundreds of consultants on the wiki |
| 16:32.20 | n3glv | it's not that hard |
| 16:32.23 | n3glv | what u need? |
| 16:32.24 | Qwell[] | ~asterisk consultants |
| 16:32.27 | n3glv | and where u at? |
| 16:32.31 | Trazz | chicago |
| 16:32.34 | Qwell[] | ~consultants |
| 16:32.35 | jbot | i heard consultants is http://www.debian.org/consultants/ |
| 16:32.39 | n3glv | why not do yourself? |
| 16:32.40 | Qwell[] | nope |
| 16:32.50 | n3glv | how fancy u need? |
| 16:32.54 | Qwell[] | Trazz: there are a bunch of people here too |
| 16:32.54 | Trazz | i started it but not much time in my schedulue now.. :( |
| 16:33.10 | *** join/#asterisk Soul (n=Soul@82.102.1.42) |
| 16:33.46 | n3glv | if u could get the hw together there's people who would set the rest up via the Internet |
| 16:34.02 | Qwell[] | ~asteriskconsultants |
| 16:34.03 | jbot | asteriskconsultants is probably http://www.voip-info.org/wiki/view/Asterisk+consultants |
| 16:34.07 | Qwell[] | there it is :p |
| 16:34.28 | Trazz | i have hardware together and have it talking to broadvoice.. etc.. just need it finished up |
| 16:34.43 | Qwell[] | Trazz: what is needed exactly? |
| 16:34.51 | n3glv | that's not a big deal then |
| 16:35.12 | n3glv | I heard BV has some hidden limits, overage etc. |
| 16:35.21 | CrashHD | is there a way to add a prefix to a voicemails callback? |
| 16:35.22 | n3glv | but for biz I guess it's ok |
| 16:35.40 | Trazz | to start with i need basic IVR going with voicemail |
| 16:35.48 | Trazz | then in the future the ability to have queues |
| 16:36.05 | Qwell[] | queues are evil |
| 16:36.08 | Trazz | i am using cisco ip phones and softphones to access pbx |
| 16:36.20 | Corydon76-home | Cisco is evil |
| 16:36.31 | Trazz | what phones do you recommend? |
| 16:36.38 | Corydon76-home | Polycom |
| 16:36.41 | n3glv | friend on headset here (fisk) was joking our new IVR is going to be, For English, press ONE, for English, press 2 Etc... |
| 16:36.42 | Trazz | i already have some phones is why i opted to use this |
| 16:37.02 | Corydon76-home | which makes Cisco phones the spawn of evil |
| 16:37.25 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 16:37.43 | Trazz | ya i dont like having to have a tftp server after phone reboot |
| 16:37.46 | Trazz | but its cisco |
| 16:38.49 | Trazz | so for now need to have IVR inbound and then using two diferent voip providers for dialing out with dialplans and ability to check voice mail remotely |
| 16:39.25 | SplasPood | when using the features.conf #1 transfer, how does it determine which context to drop the outgoing call into? |
| 16:41.48 | littleball | hello, in the Makefile.patch, how to interpret the lines? |
| 16:42.24 | Trazz | n3glv, still there |
| 16:42.25 | littleball | some lines with + at the begining |
| 16:42.37 | n3glv | yes |
| 16:42.38 | littleball | some lines without "+" at the begining |
| 16:42.41 | n3glv | was away for a sec |
| 16:42.53 | n3glv | two dialplans depending on area etc? toll rules? |
| 16:42.59 | Trazz | yes |
| 16:43.03 | littleball | hello, in the Makefile.patch, how to interpret the lines? |
| 16:43.49 | *** join/#asterisk blitz[laptop] (n=blitzrag@CPE0040f44a40c4-CM00122570228c.cpe.net.cable.rogers.com) |
| 16:44.10 | Trazz | i am on 1.2.8 now |
| 16:44.11 | MikeJ[Laptop] | blitz rag? |
| 16:44.13 | MikeJ[Laptop] | heh |
| 16:44.22 | blitz[laptop] | ? |
| 16:44.40 | blitz[laptop] | GO GERMANY! |
| 16:44.41 | MikeJ[Laptop] | you dropped an e? |
| 16:44.54 | blitz[laptop] | ? |
| 16:45.01 | MikeJ[Laptop] | blitzrag |
| 16:45.02 | blitz[laptop] | I have my nick showing as blitz[laptop] |
| 16:45.09 | MikeJ[Laptop] | yah... |
| 16:45.16 | MikeJ[Laptop] | n=blitzrag@CPE0.... |
| 16:45.21 | blitz[laptop] | ? |
| 16:45.23 | blitz[laptop] | no idea |
| 16:45.36 | blitz[laptop] | probably cut off or something |
| 16:45.38 | MikeJ[Laptop] | oh.. are those limited to 8 or somthing? |
| 16:45.43 | blitz[laptop] | probably |
| 16:45.46 | MikeJ[Laptop] | heh |
| 16:45.50 | MikeJ[Laptop] | wassup |
| 16:46.32 | blitz[laptop] | nada much -- just watching Germany vs. Argentina |
| 16:46.54 | blitz[laptop] | gotta setup some monitoring software today... |
| 16:47.33 | n3glv | Trazz, there are some cool ways to avoid running min on tht pstn acct |
| 16:48.04 | Trazz | great :) |
| 16:48.45 | n3glv | enum is part of it, several services to resolv phone numbers to ip addys for direct ip to ip dial |
| 16:50.10 | n3glv | especially if you know the numbers u will be calling most and know they have voip, u are in like flynn |
| 16:50.43 | n3glv | guy from Sprint told a friend of mine that his calling will be upwards of 97% ip to ip |
| 16:51.10 | Trazz | luckily we are going to be inbound mainly.. |
| 16:51.11 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-6.rcmdva.fios.verizon.net) |
| 16:51.30 | n3glv | is it a commercial BV account? |
| 16:51.36 | n3glv | how many legs they allow you? |
| 16:51.44 | Qwell[] | 1 :P |
| 16:51.54 | n3glv | yeah, on consumer, that's what I heard |
| 16:52.03 | Trazz | 2 legs per # |
| 16:52.31 | n3glv | I get that from consumer unlimited viatalk |
| 16:52.40 | Trazz | they require seperate account per number right now.. i need to find another provider that is more friendly |
| 16:52.42 | SpaceBass | with bv I've had more than 2 incoming calls |
| 16:52.45 | SpaceBass | never more than one outgoing |
| 16:52.59 | n3glv | axvoice seemed to be unlimited, but puked on friend at 2200 min (comsumer unlim acct) |
| 16:53.06 | SpaceBass | Telasip allows a few concurrent outgoing calls |
| 16:53.37 | n3glv | I was not sure on incoming for BV, know for sure they limit private accts to 1 out |
| 16:53.41 | *** join/#asterisk Crshman (n=crshman@netblock-68-183-62-163.dslextreme.com) |
| 16:53.53 | SpaceBass | like I said, I've had at least 3 concurrent incoming |
| 16:53.58 | Qwell[] | ~unlimited |
| 16:53.59 | jbot | well, unlimited is <Nugget> unlimited voip == punch the monkey to win a free ipod |
| 16:54.03 | n3glv | shellshark has 4 out limit, and 2 DID's, true unlimited too |
| 16:54.17 | *** part/#asterisk littleball (n=littleba@cm52.epsilon174.maxonline.com.sg) |
| 16:54.18 | *** part/#asterisk blitz[laptop] (n=blitzrag@CPE0040f44a40c4-CM00122570228c.cpe.net.cable.rogers.com) |
| 16:54.37 | SpaceBass | Im trying to move my home office to telasip if we can work out the billing |
| 16:54.41 | n3glv | we had 11 legs going through axvoice |
| 16:54.44 | n3glv | ;-) |
| 16:54.57 | Trazz | what is the site for shellshark> |
| 16:55.18 | n3glv | umm have to dig him up, he has a signup fee I didn't like |
| 16:56.06 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 16:56.07 | Trazz | how is axvoice |
| 16:56.09 | Trazz | ? |
| 16:56.22 | Qwell[] | Trazz: all voip sucks :p |
| 16:56.32 | n3glv | axvoice was ok |
| 16:56.39 | n3glv | viatalk is nice |
| 16:57.04 | n3glv | that's only 2 I have personally tried |
| 16:57.15 | *** join/#asterisk Blackthorn (i=blacktho@72.236.88.10) |
| 16:57.26 | n3glv | VT has fallback to some number, it's great if the pbx drops it's load on you... kick the wire out the wall etc. |
| 16:57.45 | n3glv | so my cell rings if the pbx is not answering |
| 16:57.47 | n3glv | or full |
| 16:58.14 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 16:58.19 | n3glv | if anyone go's viatalk, let me get u as a referral! ;-) |
| 16:58.29 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 16:58.54 | Trazz | how much for unlimited business ? |
| 16:59.04 | Trazz | do they let you pass your own callerid? |
| 16:59.13 | n3glv | they have some kind of thing on their consumer unlim that wants u to use no more than a 3 to 1 ratio of out to in |
| 16:59.32 | n3glv | I don't know abt the cid, on biz they might |
| 16:59.37 | n3glv | their biz rates are good |
| 16:59.40 | n3glv | have 2200 markets |
| 17:01.09 | *** join/#asterisk Spy000007 (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
| 17:01.28 | jamincollins | yay for bad cable! |
| 17:04.49 | Blackthorn | Hi All. When using my Sipura units calls get cut off when I call a local number that is maped through my local pri, calling through the long distace provider works great. Thus I know my problem is with my pri. How do i log or get more detailed info on whats wrong with the pri's ? |
| 17:06.12 | Blackthorn | Checking zttools it shows no irq misses |
| 17:06.18 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.235.254.Dial1.SanJose1.Level3.net) |
| 17:10.25 | SplasPood | is there any way to increase the gain on voicemail recordings? |
| 17:10.40 | Qwell[] | SplasPood: check voicemail.conf |
| 17:10.44 | *** join/#asterisk Druken (n=Druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com) |
| 17:10.51 | SplasPood | Qwell[]: I did, maybe I missed something? |
| 17:10.56 | SplasPood | Qwell[]: Whats the option called? |
| 17:11.30 | SplasPood | Do you know that such an option exists, or... ? |
| 17:11.48 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 17:11.48 | *** mode/#asterisk [+o anthm] by ChanServ |
| 17:12.06 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 17:12.07 | Qwell[] | SplasPood: show application voicemail |
| 17:12.09 | Qwell[] | g() |
| 17:12.16 | stephane_ | re |
| 17:12.52 | SplasPood | Qwell[]: ahh! I knew I had seen it somewhere.. danke |
| 17:13.00 | SplasPood | i was stuck on it being a conf option |
| 17:13.23 | SplasPood | Kinda annoying to go change all my Voicemail() calls, but it works |
| 17:14.08 | Blackthorn | I found an artical on digium that talks about the most common cause of my issue is that i have busydetect=yes. And I do, so i just truned it off, and i'll see what happens now. |
| 17:15.17 | Druken | who wants to help my mush brain today? god i feel like a born again newb |
| 17:16.01 | Druken | having trouble getting me PA168V IAX2 ATA to connect to asterisk... |
| 17:16.08 | Bert- | guys |
| 17:16.18 | Bert- | I've to thank you all very very much |
| 17:16.35 | Bert- | today I made a presentation of Asterisk as IVR |
| 17:16.41 | Bert- | to my bosses |
| 17:16.48 | Bert- | they don't like open sources |
| 17:16.55 | Bert- | they wanted to buy cisoc call manager |
| 17:17.01 | nexstar | sip responce :" internal server error 500" invalid sip responce |
| 17:17.07 | Bert- | they showed my demo |
| 17:17.09 | nexstar | anyone know what may be happining |
| 17:17.20 | Bert- | then they want Asterisk ASAP :) |
| 17:17.32 | Bert- | so real thank you for support ! |
| 17:18.00 | nexstar | when transfering goes to hold music for a sec, then nothing when it should be ringing |
| 17:19.00 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 17:19.11 | Bert- | nexstar, what are you tryingto do ? |
| 17:19.23 | nexstar | transfer a call |
| 17:19.35 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 17:20.10 | Bert- | incoming call answered by someone, which transfert call to another agent ? |
| 17:20.35 | nexstar | 1 sec |
| 17:20.57 | Bert- | did you set 't' option in your Dial() cmd ? |
| 17:21.32 | nexstar | i figured it out but |
| 17:21.33 | nexstar | 1 sec |
| 17:23.02 | Druken | iax2 is 4569 right? |
| 17:23.14 | russellb | yup |
| 17:23.38 | Qwell[] | russellb: took a couple more tweaks for make install to go through |
| 17:23.51 | russellb | s/install/ginstall/ ? |
| 17:24.02 | russellb | or, INSTALL=ginstall |
| 17:24.03 | russellb | whatever |
| 17:24.03 | Qwell[] | it was trying to use install-sh actually |
| 17:24.04 | nexstar | polycom 501 |
| 17:24.09 | Qwell[] | but yeah, that was one of them |
| 17:24.12 | nexstar | latest updates |
| 17:24.17 | nexstar | bad echos... on and off |
| 17:24.18 | Qwell[] | res/install-sh didn't exist |
| 17:24.21 | nexstar | not all the time |
| 17:24.26 | nexstar | internal and external calls |
| 17:24.43 | Qwell[] | and then something in build_tools or something, with sed |
| 17:24.55 | Qwell[] | and sounds/Makefile - tar xvf :D |
| 17:24.55 | smackus | I am trying to set up different T1s than I am used to. Could someone help me out getting them set up? I am switching from what I am used to with pri and have gone to a regular t1 and i cannot get to a point where my configs work |
| 17:25.15 | Qwell[] | the tar fix is easy, and probably committable...will show you later |
| 17:25.27 | russellb | k |
| 17:25.31 | nexstar | anyone? |
| 17:25.35 | russellb | smackus: support@digium.com |
| 17:25.36 | Qwell[] | gzcat file.tar.gz | tar xf - |
| 17:25.44 | smackus | awesome... thats right |
| 17:26.05 | Qwell[] | oh, and if ! test -f blah vs if test ! -f blah |
| 17:27.28 | nexstar | anyone echo noise? |
| 17:27.43 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 17:27.51 | Qwell[] | nexstar: find x-rob...he's the self proclaimed king of echo |
| 17:27.59 | doughecka_ | king of echo? |
| 17:28.05 | doughecka_ | lol |
| 17:28.20 | coppice | is there a queen of echo? |
| 17:28.23 | nexstar | well its not all the time sometimes it will do it and others it wont |
| 17:28.28 | nexstar | internal and external calls |
| 17:28.32 | Qwell[] | coppice: No, but feel free to proclaim yourself |
| 17:28.38 | doughecka_ | I was about to ask that |
| 17:28.44 | nexstar | any idea what would cause something like that? |
| 17:29.32 | Qwell[] | anybody have any idea if any OSs don't have gzcat? |
| 17:29.47 | Qwell[] | ie; gunzip but not gzcat |
| 17:30.01 | doughecka_ | what does gzcat do? |
| 17:30.03 | russellb | gzcat is not on my linux box, heh |
| 17:30.09 | Qwell[] | russellb: d'oh |
| 17:30.20 | Qwell[] | okay, cat | gunzip it is then :p |
| 17:30.27 | Qwell[] | doughecka_: decompress |
| 17:30.31 | doughecka_ | ah |
| 17:30.33 | Qwell[] | gzcat == cat | gunzip |
| 17:30.40 | doughecka_ | ah |
| 17:31.03 | Qwell[] | the latter is a uuoc though |
| 17:31.43 | coppice | everyone has gzip. what's wrong with using that? |
| 17:31.50 | Qwell[] | coppice: nothing |
| 17:32.01 | Qwell[] | now, should I use gunzip or gzip -d? |
| 17:32.15 | Qwell[] | friggen solaris :p |
| 17:32.35 | coppice | few people have anything but gzip. use it |
| 17:32.43 | Qwell[] | ok |
| 17:32.46 | russellb | at least it's not windows |
| 17:32.54 | Qwell[] | heh |
| 17:33.15 | russellb | i wonder if we'll ever have a windows port of asterisk ... |
| 17:33.31 | Qwell[] | ugh |
| 17:33.44 | Qwell[] | let's just pretend you didn't say that :P |
| 17:33.52 | russellb | no, i said it |
| 17:33.53 | doughecka_ | we do! |
| 17:33.58 | russellb | doughecka: lies |
| 17:34.07 | Qwell[] | :p |
| 17:34.16 | doughecka_ | I thought some dood from digium made a version |
| 17:34.27 | russellb | not a real windows port |
| 17:34.31 | doughecka_ | :P |
| 17:34.37 | doughecka_ | its called vmware |
| 17:34.38 | russellb | there was astwind ... asterisk under colinux |
| 17:34.53 | russellb | and then asterisk in cygwin |
| 17:34.54 | doughecka_ | someday vmware will have direct PCI support... |
| 17:35.09 | russellb | we tried doing conferencing under vmware, it blew up |
| 17:35.14 | doughecka_ | hah |
| 17:35.19 | doughecka_ | asterisk likes its cpucycles |
| 17:35.35 | doughecka_ | now how can cisco do conferecing under vmware |
| 17:36.03 | russellb | i mean ... it kinda worked |
| 17:36.04 | doughecka_ | I heard about a guy who had a conference going, and vmotioned a server from one host to another without a hickup |
| 17:36.26 | anonymouz666 | russellb: the most up to date doc about developing is in doxygen documentation? |
| 17:36.31 | rob0 | Don't they still recommend not using * on a machine with a GUI? Windows without GUI is not possible. |
| 17:36.47 | *** join/#asterisk anto9us (n=anthony@cpc1-ptal1-0-0-cust555.swan.cable.ntl.com) |
| 17:36.49 | russellb | anonymouz666: yeah, for the most part ... some stuff at http://www.asterisk.org/developers |
| 17:41.25 | *** join/#asterisk Nodren (n=nodren@adsl-75-8-201-246.dsl.frs2ca.sbcglobal.net) |
| 17:41.55 | Nodren | Hey everyone, for anyone interested, I have a $100-500 dollar asterisk job posted on rent-a-coder https://www.rentacoder.com/RentACoder/misc/BidRequests/ShowBidRequest.asp?lngBidRequestId=487490p |
| 17:45.49 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 17:48.11 | Qwell[] | russellb: http://bugs.digium.com/view.php?id=7463 - don't hate me :D |
| 17:49.17 | russellb | i don't hate you |
| 17:49.19 | russellb | ... just solaris |
| 17:49.21 | Qwell[] | heh |
| 17:49.27 | russellb | can i commit it with the message "more reasons why solaris sucks" |
| 17:49.36 | Qwell[] | I hate <insert non-linux unix> |
| 17:49.49 | russellb | or is that too bitter ... |
| 17:49.58 | Qwell[] | not at all |
| 17:49.59 | file | not bitter enough |
| 17:50.00 | *** join/#asterisk barros (n=barros@89.106.66.150) |
| 17:50.23 | rob0 | Bit, bitter, bittest. |
| 17:50.25 | russellb | sun may be a digium partner or something ... i might get in trouble |
| 17:50.26 | barros | does anyone succed using h323 in asterisk-64bits? |
| 17:50.26 | russellb | :) |
| 17:50.32 | Qwell[] | heh |
| 17:50.45 | file | russellb: I'll protect you |
| 17:52.50 | russellb | done |
| 17:53.05 | Qwell[] | qwell_karma++ :D |
| 17:53.17 | Assid | wassup |
| 17:53.23 | russellb | forgot to give you credit in the commit msg, sorry |
| 17:53.39 | Qwell[] | no worries |
| 17:53.56 | file | russellb: you owe him an onion ring now |
| 17:54.04 | russellb | at cheeburger? |
| 17:54.04 | Qwell[] | omg |
| 17:54.07 | file | yes! |
| 17:54.15 | russellb | cheeburger onion rings r0x0r |
| 17:54.26 | Qwell[] | must try them then |
| 17:54.40 | file | it's going to be fun :D |
| 17:54.43 | russellb | yayz |
| 17:55.13 | OuterSpace | hi, how can i configure sip with dynamic ip ? can i use a dyndns domain on externip ? |
| 17:55.29 | Qwell[] | OuterSpace: externhost |
| 17:55.39 | Qwell[] | and externrefresh, or whatever it is |
| 17:55.54 | OuterSpace | thanks |
| 17:55.56 | Qwell[] | heh |
| 17:56.00 | Qwell[] | funny... |
| 17:56.21 | Qwell[] | somebody fixed that user who abused the karma bug...and now the hall of fame only has 14 :D |
| 17:56.45 | Qwell[] | brookshire: ! |
| 17:58.21 | *** join/#asterisk pbx1 (n=pbx1@58.69.92.24) |
| 17:59.03 | *** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br) |
| 18:00.57 | *** join/#asterisk tdonahue (n=tdonahue@207.138.151.58) |
| 18:02.00 | *** join/#asterisk Beighto (n=chatzill@64.160.113.130) |
| 18:03.18 | anonymouz666 | russellb could code an app_whisper :) |
| 18:04.00 | Beighto | Can a CLI command be run in the dialplan? For example, I want the "database put conferences {variable}" command to be automated |
| 18:06.04 | [TK]D-Fender | Beighto : "show application exec" |
| 18:08.07 | *** join/#asterisk Strom_C (n=strom@gateway.digium.com) |
| 18:08.57 | Beighto | [TK]D-Fender: Looks promising, thanks you |
| 18:19.17 | *** join/#asterisk [TK]D-Fender (n=joe@CPE000d3a2c3061-CM00080d8dba84.cpe.net.cable.rogers.com) |
| 18:19.58 | gmfm | can anyone think of a reason the ${CALLERID} and ${CALLERIDNAME} variables would not contain the caller id name? I know it is received because it shows up when i do 'show channel...' |
| 18:21.28 | Nugget | your asterisk has decided to take friday off. |
| 18:22.15 | gmfm | it took thursday off moreso with the crashing and the dying and the pain |
| 18:22.16 | [TK]D-Fender | gmfm : Those 2 vars are long since deprecated |
| 18:22.55 | gmfm | so... i take it the variable reference on the wiki is a bit outdated |
| 18:24.30 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 18:24.54 | gmfm | TK, do you know what the new way to access the CID name is? |
| 18:25.46 | Qwell[] | gmfm: ${CALLERID(name)} |
| 18:26.32 | *** join/#asterisk SexyKen (n=Ken@c-24-5-129-114.hsd1.ca.comcast.net) |
| 18:26.51 | SexyKen | Hey guys - if you had a box that kept NOT RESPONDING like - everything just freezes up, what would you look into |
| 18:26.59 | gmfm | thanks Qwell |
| 18:27.22 | gmfm | for some odd reason that doesn't return it to me either... i think it hates me |
| 18:29.39 | *** part/#asterisk n3glv (n=n3glv@monrovll-cuda1-24-53-251-235.pittpa.adelphia.net) |
| 18:30.22 | brookshire | qwell: ????????????????????????????????????? |
| 18:30.29 | Qwell[] | brookshire: mantis :( |
| 18:30.34 | brookshire | boo! |
| 18:30.50 | Qwell[] | brookshire: any idea how the karma hall of fame works? It's only listing 14 people, when it says 15, heh |
| 18:30.55 | brookshire | qwell: /etc/init.d/apache2 stop |
| 18:30.58 | brookshire | :) |
| 18:31.02 | Qwell[] | (ie; I want to be on it!) |
| 18:31.03 | Qwell[] | :p |
| 18:31.19 | Qwell[] | so close, I am |
| 18:31.20 | gmfm | bah ${CALLERID(all)} returns "" <949861****> |
| 18:31.21 | Qwell[] | I can taste it |
| 18:31.31 | Qwell[] | gmfm: Then you aren't getting name.. |
| 18:31.39 | brookshire | i'll take a look |
| 18:31.49 | Qwell[] | brookshire: You rock |
| 18:32.11 | gmfm | when i show the channel i get Caller ID Name: ENTERPRISE MORT |
| 18:32.43 | brookshire | i increased it one place ;) |
| 18:32.45 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 18:32.51 | PakiPenguin | hello everyone |
| 18:32.52 | Qwell[] | am I on it?! |
| 18:33.07 | Qwell[] | damn, I'm not |
| 18:33.29 | PakiPenguin | can i have seperate group and context for all fxo ports? |
| 18:34.12 | Qwell[] | wait...not it shows 16 :P |
| 18:34.16 | Qwell[] | s/not/now/ |
| 18:35.51 | brookshire | qwell: you made me find a bug with karma |
| 18:35.53 | brookshire | now i have to fix it |
| 18:35.54 | brookshire | boo! |
| 18:36.00 | PakiPenguin | how do i know if i have a 66 punch block or 110 |
| 18:36.05 | Qwell[] | :D |
| 18:36.14 | Qwell[] | w00t, 16th place |
| 18:36.18 | Qwell[] | I knew I was close :D |
| 18:36.19 | file | brookshire: :( |
| 18:36.48 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 18:37.04 | *** join/#asterisk qstax (n=bob@net-252-14.northwestel.net) |
| 18:37.13 | brookshire | there fixed |
| 18:37.22 | Qwell[] | excellent |
| 18:37.27 | brookshire | now it actually shows the top 25 |
| 18:37.35 | brookshire | instead of 27 |
| 18:37.38 | brookshire | like it was |
| 18:37.40 | Qwell[] | heh |
| 18:37.44 | brookshire | eventhough it was set to 25 |
| 18:37.49 | Qwell[] | weird |
| 18:38.10 | brookshire | they did a distinct karma_scores |
| 18:38.33 | brookshire | so if two people had the same score, it only counted once |
| 18:38.36 | Qwell[] | ha |
| 18:38.45 | Qwell[] | that's classic |
| 18:38.56 | brookshire | there is still another bug.. |
| 18:39.03 | wunderkin | PakiPenguin, images.google.com |
| 18:39.12 | Qwell[] | brookshire: btw, I sent you a message |
| 18:39.14 | PakiPenguin | i just did that |
| 18:39.31 | *** join/#asterisk Manipura (n=chatzill@S01060011954c9c46.cg.shawcable.net) |
| 18:39.52 | wunderkin | 110 is bigger |
| 18:40.03 | Manipura | Whats better for running asterisk, dual rank or single rank DIMMS? |
| 18:40.05 | brookshire | heh |
| 18:40.07 | brookshire | 14:36 -!- Private messages from unregistered users are currently blocked due to spam problems, |
| 18:40.10 | brookshire | <PROTECTED> |
| 18:40.18 | Qwell[] | woops |
| 18:41.58 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 18:43.19 | PakiPenguin | what do you call that small box , where you can connect wires at one end and it gives you rj-11 at the other end |
| 18:43.21 | PakiPenguin | :p |
| 18:43.22 | PakiPenguin | hehe |
| 18:43.49 | brookshire | phone jack? |
| 18:43.54 | PakiPenguin | ummm nope |
| 18:44.03 | PakiPenguin | its a small box typa thing :p here |
| 18:44.24 | Qwell[] | Is there a jack in the box? |
| 18:44.30 | Qwell[] | (pun definitely intended) |
| 18:44.32 | PakiPenguin | :p |
| 18:44.39 | PakiPenguin | hmmm |
| 18:44.48 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 18:44.52 | brookshire | phone company phone jack hookup? |
| 18:45.00 | wunderkin | jumbo jack hehe |
| 18:45.57 | PakiPenguin | :p haha |
| 18:45.58 | PakiPenguin | maybe |
| 18:46.08 | nortex | Smart Jack |
| 18:46.24 | PakiPenguin | :p |
| 18:46.29 | brookshire | Dumb Jack |
| 18:46.29 | brookshire | ;) |
| 18:46.52 | brookshire | ha.. port is also a good work to describe it |
| 18:47.01 | gmfm | does anyone know when callerid name is transmitted on a PRI? i thought it was sent in the call setup along with the ani so it would be available immediately |
| 18:47.21 | PakiPenguin | :p |
| 18:47.33 | devicenode | brookshire: you can plug into my port... |
| 18:47.45 | brookshire | orly? thanks! |
| 18:47.54 | Qwell[] | ... |
| 18:47.56 | devicenode | I only have 10Mbps uplink though :( half duplex |
| 18:48.01 | brookshire | is it hot? |
| 18:48.10 | devicenode | yesssss |
| 18:48.43 | *** join/#asterisk sevard (n=sev@adsl-71-129-115-244.dsl.irvnca.pacbell.net) |
| 18:48.45 | sevard | You are cordially invited to the party in my pants. |
| 18:48.46 | *** part/#asterisk vechers (n=svecher@64.61.117.139) |
| 18:49.07 | PakiPenguin | this http://www.wptele.com/rj11.jpg |
| 18:49.08 | PakiPenguin | lol |
| 18:49.25 | Qwell[] | PakiPenguin: That's called a phone jack :P |
| 18:49.34 | PakiPenguin | lol no |
| 18:49.36 | PakiPenguin | haha |
| 18:49.55 | PakiPenguin | Qwell, we call it a dabbi here :p |
| 18:49.56 | PakiPenguin | haha |
| 18:50.18 | PakiPenguin | http://www.mavromatic.com/images/cat5e.jpg <-- more like it but for rj-11 :p |
| 18:50.38 | Qwell[] | still a phone jack |
| 18:50.50 | PakiPenguin | yup yup i know |
| 18:51.39 | PakiPenguin | i have to tell the guy specifically what to get , he needs to get cable from a 66block to a 2400p |
| 18:51.42 | *** join/#asterisk Un1x (i=seann@CPE00016c29e15a-CM00080d40ee4c.cpe.net.cable.rogers.com) |
| 18:51.43 | Un1x | hey |
| 18:51.52 | PakiPenguin | and also for the astribank |
| 18:51.59 | Un1x | anyone around was wondering if this gsm gateway will work with asterisk |
| 18:52.00 | Un1x | http://store.voxilla.com/customer/product.php?productid=16234&cat=276&page=1 |
| 18:52.08 | Qwell[] | the tdm2400p uses an amphenol connector |
| 18:52.30 | PakiPenguin | Un1x, use junghanns cards , they work great with asterisk |
| 18:52.47 | Un1x | dude i dont need a card i need a GSM Gateway! |
| 18:52.59 | PakiPenguin | i see |
| 18:53.12 | brookshire | un1x: if it talks sip, they probably |
| 18:53.18 | brookshire | s/they/then |
| 18:53.29 | devicenode | I talk SIP, after a few drinks |
| 18:53.30 | *** join/#asterisk postel_ (n=jp@unaffiliated/postel) |
| 18:53.37 | brookshire | qwell: actually... amphenol is a brand :/ |
| 18:53.49 | Qwell[] | oh |
| 18:58.38 | rob0 | Would someone help me test my new Iaxtel account, please? |
| 19:00.02 | *** join/#asterisk brijn (n=brijnier@204.244.176.116.net-conex.com) |
| 19:00.26 | Druken | anyone gotten ahold of one of those wip300 cisco wifi phones? |
| 19:00.41 | Qwell[] | Druken: friend of mine did |
| 19:00.51 | Druken | good or shit? |
| 19:00.52 | Qwell[] | and I had one for about 45 seconds |
| 19:00.55 | Qwell[] | it looked good |
| 19:01.17 | Nivex | oh man, I'd love to get my 'rents on one of those GSM gateways |
| 19:01.21 | Druken | looks like a nice phone... |
| 19:01.31 | Qwell[] | Druken: just don't get a 330, heh |
| 19:01.36 | Nivex | my Dad has a hatred of Verizon |
| 19:01.38 | Qwell[] | friggen windows ce |
| 19:01.58 | Qwell[] | Druken: and, it's linksys, not cisco |
| 19:02.57 | Druken | same shit.. |
| 19:03.29 | Nivex | linksys bought sipura, cisco bought linksys. Ciscsysura? |
| 19:03.55 | Qwell[] | linksys and cisco are run seperately |
| 19:04.40 | Druken | i got with linksys and cisco switches, identical |
| 19:05.26 | CoffeeIV_ | It used to be that if I did MixMonitor instead of Monitor I needed to do StopMixMonitor instead of StopMonitor. However I just upgraded my * to 1.2.8 and now there is no StopMixMonitor -- should StopMonitor be used in both cases ? |
| 19:06.52 | *** join/#asterisk normsteel (n=nathank@69.17.44.81) |
| 19:07.05 | Druken | bah! i think the bastards at gentek went home early... |
| 19:08.30 | normsteel | having some problem getting zaptel dummy to compile. i patched the ztdummy and spinlock.h but i still get an error (http://pastebin.ca/75918) |
| 19:08.39 | *** join/#asterisk backblue (n=moo@87-196-69-21.net.novis.pt) |
| 19:09.34 | Nivex | whoa, when did wifi phones get so inexpensive? atacomm has a zyxel for $170 US |
| 19:11.58 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 19:15.06 | Druken | that wip300 looks sexy.... |
| 19:15.14 | Druken | i think i just found my new portable phone |
| 19:16.09 | Manipura | So what is TCP/IP Offload Engine and Do I need it? |
| 19:16.20 | Manipura | I can't find any info with it for running voip |
| 19:18.33 | *** join/#asterisk Defraz (n=t0tal@fw.centrisys.com) |
| 19:19.48 | brijn | Manipura, as far as I know TOE is the processor on the NIC doing some work that otherwise would be done by the CPU == a good thing |
| 19:20.03 | Manipura | Awesome, thank you |
| 19:21.19 | *** join/#asterisk RF_MIA (n=mw1@adsl-070-147-214-250.sip.mia.bellsouth.net) |
| 19:21.47 | gandhijee | ToE's are GOOD |
| 19:22.09 | gandhijee | Druken: they feel cheaply made.... |
| 19:22.47 | Druken | oh yeah? |
| 19:22.58 | *** join/#asterisk asterisknewbiezz (n=asterisk@rrcs-67-52-187-18.west.biz.rr.com) |
| 19:23.05 | gandhijee | yeah |
| 19:23.13 | gandhijee | very very light |
| 19:23.19 | Druken | so you wouldn't reccomend one? |
| 19:23.25 | gandhijee | and the back cover slides off way to easily |
| 19:23.27 | asterisknewbiezz | can someone help me put an IVR? |
| 19:23.33 | gandhijee | one of the guys here has a 330... |
| 19:23.42 | gandhijee | it might be a better choice |
| 19:23.46 | asterisknewbiezz | on top of my existing vicidialer. |
| 19:23.47 | Druken | that's the one with winse |
| 19:23.55 | gandhijee | yea |
| 19:24.21 | Druken | i just have a hard time swallowing windows.. and voip in the same region... |
| 19:24.21 | *** join/#asterisk [Airwolf] (n=airwolf@cp656687-a.landg1.lb.home.nl) |
| 19:24.46 | asterisknewbiezz | can someone help me put an IVR? on top of my existing vicidialer. |
| 19:24.50 | gandhijee | at least its not MS Communications server or what ever the hell its called. |
| 19:25.17 | Druken | touche |
| 19:25.25 | *** join/#asterisk arguile (i=user224@66.38.201.234) |
| 19:25.42 | Druken | argh... i am sooo bored.... |
| 19:25.59 | gandhijee | wanna help me port zaptel to Xscale/ARM then? |
| 19:26.14 | klasstek | Has anyone used the Sangoma A108 EC yet? |
| 19:26.36 | Druken | gandhijee: i would be of no use or help to you in that matter |
| 19:26.45 | gandhijee | =/ |
| 19:27.48 | Druken | maybe i'll go take the blade off my lawn mower, i need a new blade... |
| 19:27.57 | PerlStalker | Perhaps you could tell me how to prevent users from forwarding their voicemail messages to random people. |
| 19:28.00 | asterisknewbiezz | can someone help me put an IVR? on top of my existing vicidialer. |
| 19:28.28 | *** part/#asterisk sevard (n=sev@adsl-71-129-115-244.dsl.irvnca.pacbell.net) |
| 19:28.33 | gandhijee | yes, at the cost of $1,000,000 US |
| 19:28.48 | asterisknewbiezz | thats it lol |
| 19:31.47 | marv0997 | hi all, can someone help me troubleshoot a call i'm trying to make from my * to fwd that keeps getting dropped |
| 19:32.54 | asterisknewbiezz | Anyone can help me with putting an IVR on an existing dialplan.? |
| 19:35.52 | ptinsley | so ya, it's bad when asterisk dumps core right |
| 19:36.00 | ptinsley | :) |
| 19:37.09 | *** part/#asterisk [Airwolf] (n=airwolf@cp656687-a.landg1.lb.home.nl) |
| 19:40.17 | *** join/#asterisk angelad (n=angela@gateway.digium.com) |
| 19:41.23 | *** part/#asterisk Torginator (n=Chris@c-67-190-204-43.hsd1.mn.comcast.net) |
| 19:42.13 | rob0 | Does Digium still have an IAXtel number? I tried 1-700-428-6000 and got rejected. "No such context/extension". |
| 19:42.49 | devicenode | rob0: I was... mucking with the server |
| 19:42.51 | devicenode | it'll work now |
| 19:42.55 | rob0 | thanks |
| 19:43.19 | brookshire | angelad: hi! |
| 19:43.26 | brookshire | rob0: :( |
| 19:43.40 | brookshire | rob0: do that again! |
| 19:44.59 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 19:45.56 | rob0 | I got through. |
| 19:46.14 | brookshire | oh.. nm |
| 19:46.19 | brookshire | with iaxtel? |
| 19:46.23 | rob0 | yes |
| 19:46.27 | brookshire | hotness |
| 19:46.36 | devicenode | I was trimming down the loaded modules |
| 19:46.36 | brookshire | i don't have to fix anything then ;) |
| 19:46.47 | devicenode | caught it at a time when pbx_config.so wasn't loaded |
| 19:47.01 | rob0 | I should have asked him to call me back to test incoming :) |
| 19:47.23 | rob0 | (I got Randall in the sales queue) |
| 19:47.30 | brookshire | call 1-700-428-6069 ;) |
| 19:47.41 | rob0 | thanks, is that you? |
| 19:47.44 | brookshire | yah |
| 19:47.51 | devicenode | noes don't call that number |
| 19:47.58 | rob0 | no? |
| 19:48.07 | devicenode | it's a trick! |
| 19:48.09 | brookshire | haha.. call 6068 instead |
| 19:48.10 | rob0 | aha |
| 19:48.14 | rob0 | 69 |
| 19:48.16 | rob0 | :) |
| 19:48.20 | brookshire | 1-700-428-6068 |
| 19:48.47 | rob0 | tt-monkeys :) |
| 19:48.55 | brookshire | HAHA |
| 19:48.57 | brookshire | 6069 is me |
| 19:50.11 | Qwell[] | rob0: I would call 6066 if I were you |
| 19:51.00 | Qwell[] | :D |
| 19:51.24 | angelad | brooks your needed with an IT matter on 2nd floor |
| 19:51.27 | angelad | pronto |
| 19:52.57 | MikeJ[Laptop] | STAT |
| 19:53.16 | rob0 | :) |
| 19:53.34 | MikeJ[Laptop] | wassup devicefile |
| 19:53.50 | devicenode | the amount of water falling from the sky |
| 19:53.51 | devicenode | you? |
| 19:54.35 | MikeJ[Laptop] | ummm |
| 19:54.37 | MikeJ[Laptop] | soda time |
| 19:58.23 | ptinsley | ouch 0x002c22eb in _IO_new_file_overflow () from /lib/tls/libc.so.6 |
| 20:00.03 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 20:01.35 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 20:08.49 | terrapen | has anybody heard of a device that will mute a stereo and broadcast an overhead page over a sound system? |
| 20:10.59 | Qwell[] | terrapen: a robot |
| 20:13.15 | *** join/#asterisk anjaana_ladk (n=abhinav@59.176.29.112) |
| 20:13.39 | terrapen | heh |
| 20:13.46 | terrapen | starting to believe it. |
| 20:14.15 | terrapen | maybe this Polycom auto answer config stuff is what i want |
| 20:14.22 | terrapen | it just seems a little kludgy |
| 20:15.23 | *** part/#asterisk anjaana_ladk (n=abhinav@59.176.29.112) |
| 20:16.59 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 20:17.47 | *** join/#asterisk anjaana_ladka (n=anjaana_@59.176.29.112) |
| 20:17.55 | anjaana_ladka | hello room |
| 20:18.13 | anjaana_ladka | can any body pls help me in configuring asterisk.... |
| 20:18.34 | anjaana_ladka | i am new ... here.. and have installed asterisk on my machine... |
| 20:18.37 | *** join/#asterisk IronHelixz (n=irc@ool-45785cfe.dyn.optonline.net) |
| 20:18.43 | anjaana_ladka | and have also installed the client xlite.. |
| 20:19.08 | anjaana_ladka | but i am not able to set up the extensions.conf and other sip and iax.conf files.. |
| 20:20.42 | *** part/#asterisk jamincollins (n=jcollins@ptech7-44.acdmis.com) |
| 20:20.45 | Strom_C | anjaana_ladka: read the book |
| 20:20.46 | Strom_C | ~book |
| 20:20.48 | jbot | [book] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 20:21.12 | Strom_C | ~docs |
| 20:21.13 | jbot | i heard docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 20:25.09 | *** part/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 20:28.06 | *** join/#asterisk tamp4x (n=tampon@64.201.13.51) |
| 20:28.32 | tamp4x | i have low level sip/zap debugging turned on, how do i turn it off? |
| 20:28.42 | Qwell[] | tamp4x: sip no debug |
| 20:28.44 | Qwell[] | ? |
| 20:29.00 | tamp4x | no |
| 20:29.23 | tamp4x | its telling me dtmf digits that are dialed, |
| 20:29.31 | *** join/#asterisk Jaxxan (n=jaxxan@202.70.125.60) |
| 20:29.36 | Strom_C | SIX! |
| 20:29.37 | tamp4x | when retranmissiosn are stopped |
| 20:29.37 | Strom_C | TWO! |
| 20:29.47 | tamp4x | if frames are missed and wha tnot |
| 20:32.31 | Qwell[] | edit logger.conf |
| 20:34.25 | tamp4x | that not do it |
| 20:34.28 | tamp4x | =[ |
| 20:34.55 | Corydon76-home | Remove the ,dtmf in logger.conf and 'logger reload' |
| 20:36.23 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 20:37.51 | tamp4x | not in there |
| 20:37.52 | tamp4x | =] |
| 20:38.16 | *** join/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx) |
| 20:38.24 | tamp4x | looks like logger reload did it tho |
| 20:38.30 | tamp4x | thought general reload would do it |
| 20:39.25 | rene- | hello, i have done an asterisk installation without doing make samples first. startup fails at the [chan_phone.so] => (Linux Telephony API Support) |
| 20:39.25 | rene- | <PROTECTED> |
| 20:39.44 | Strom_C | it's not too late to do make samples, you know |
| 20:40.12 | rene- | Strom_C: i guess its not, can i avoid it? |
| 20:40.51 | Strom_C | rene-: sure, if you know how to write the config files from scratch |
| 20:41.51 | rene- | ok |
| 20:43.03 | twisted[asteria] | silly tampon, reload is for apps |
| 20:46.02 | Corydon76-home | twisted[asteria]: they also make pads |
| 20:47.34 | rene- | i just needed a modules.conf to ditch chan_[phone|alsa|oss] |
| 20:47.40 | rene- | .so |
| 20:48.01 | rene- | in mac os x a modules.conf is not needes |
| 20:48.04 | rene- | neeeded |
| 20:48.10 | rene- | shit needed |
| 20:49.25 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 20:52.42 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 20:53.44 | *** part/#asterisk smackus (n=smackus@c-67-169-248-217.hsd1.ut.comcast.net) |
| 20:55.17 | *** join/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
| 20:55.47 | FuriousGeorge | hey all. i noticed a wierd phenomenon. when * cant get the callerid info it just takes the last entry in my zapata.conf |
| 20:56.10 | FuriousGeorge | also, if i have no voicemail, and i hit the mail retrieve button on my snom. it calls one of my sip extensions |
| 20:56.14 | FuriousGeorge | wierd stuff |
| 21:01.18 | *** part/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com) |
| 21:02.34 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-246-145.buckeyecom.net) |
| 21:09.09 | rene- | weirs |
| 21:09.11 | rene- | weird |
| 21:09.49 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 21:15.24 | rene- | this is really stupid stuff software patents are teh suck |
| 21:15.28 | rene- | http://www.infoq.com/news/RedHat-Sued-Due-to-Hibernate-3-O;jsessionid=054E20D345CCFF3D3C82849DB2E6A588 |
| 21:19.58 | *** join/#asterisk tgrman (n=jcmoore@picard.ojc.nuvio.com) |
| 21:20.32 | FuriousGeorge | anyone know anything about using the snom 3xx series with asterisk? |
| 21:21.02 | Hmmhesays | I got a snom200 running fine with asterisk, the config pages look similar |
| 21:21.28 | FuriousGeorge | Hmmhesays: what happens if you have no voicemail and you hit the retrieve button |
| 21:22.18 | Qwell[] | then you go to voicemail still? |
| 21:22.22 | *** join/#asterisk pythos (i=pythos@unaffiliated/pythos) |
| 21:22.26 | pythos | mornin gang |
| 21:22.41 | FuriousGeorge | Qwell[]: one would think or expect. on some of my phones nothing happens, and im fine with that |
| 21:23.03 | pythos | anyone know about having your old POTS phone number moved to an VoIP provider? |
| 21:23.06 | FuriousGeorge | Hmmhesays: Qwell[]: but on the others it calls the extension with the last snom on it |
| 21:23.35 | pythos | FuriousGeorge, U from Madison? |
| 21:23.35 | FuriousGeorge | which is interesting because when asterisk doesnt get any cid info, it pulls the cid of the last entry in zapata,conf |
| 21:23.44 | FuriousGeorge | pythos: madison, nj? |
| 21:24.02 | pythos | no, just a band in Madison WI with that name.. |
| 21:24.09 | FuriousGeorge | ive heard |
| 21:24.13 | pythos | heh |
| 21:24.22 | pythos | I know one of them |
| 21:24.29 | pythos | the drummer |
| 21:24.37 | Strom_C | pythos: it's called "porting" the number |
| 21:24.48 | pythos | Strom_C: is it difficult? |
| 21:25.03 | Qwell[] | pythos: it usually costs |
| 21:25.14 | pythos | hmm, well better cost than get cut off |
| 21:25.17 | FuriousGeorge | pythos: there was also a character in an episode of the simpsons, thats where i got the name |
| 21:25.24 | rob0 | Does the drummer wear a big yellow hat? |
| 21:25.28 | pythos | FuriousGeorge: ahh |
| 21:25.38 | Qwell[] | ~FuriousGeorge |
| 21:25.40 | jbot | furiousgeorge is probably a knife-fighting (cable) monkey last seen with The Man with the Yellow Bat |
| 21:25.41 | pythos | never saw them play :-( |
| 21:25.49 | FuriousGeorge | jbot: that never gets old |
| 21:25.54 | Strom_C | ~Strom)C |
| 21:25.55 | Strom_C | er |
| 21:25.57 | *** join/#asterisk [TK]D-Fender (n=joe@CPE000d3a2c3061-CM00080d8dba84.cpe.net.cable.rogers.com) |
| 21:25.59 | Strom_C | ~Strom_C |
| 21:26.10 | Strom_C | gasp, theres nothing about me |
| 21:26.35 | Qwell[] | jbot: Strom_C is just some nub |
| 21:26.36 | jbot | okay, Qwell[] |
| 21:26.43 | pythos | anyway, I guess before QWest catches up to me, I should probably port the number... ;-) |
| 21:26.46 | Strom_C | hah |
| 21:26.53 | Strom_C | ~strom_c |
| 21:26.55 | jbot | it has been said that strom_c is just some nub |
| 21:27.00 | Qwell[] | ;) |
| 21:27.03 | Strom_C | :) |
| 21:27.07 | Strom_C | ~qwell |
| 21:27.08 | jbot | rumour has it, qwell is a patented liquid formula that contains three plant-based bio-active agents that work together in a perfectly balanced combination. These agents act synergistically to boost your good cholesterol and slash the bad. |
| 21:27.48 | FuriousGeorge | the daunting thing is that its not consistant accross all the phones, so im not sure how to go about addressing it |
| 21:28.03 | Qwell[] | FuriousGeorge: replace them with ciscos |
| 21:28.09 | FuriousGeorge | if they actually /have/ voicemail, then the button works fine |
| 21:28.40 | FuriousGeorge | Qwell[]: i heard that ciscos are really stuffed with old rma'd linksys routers now |
| 21:28.51 | Qwell[] | heh |
| 21:29.52 | FuriousGeorge | you know when i was more newbish this channel was really helpful, now i find it kinda sarcastic :) |
| 21:30.00 | Qwell[] | kinda? |
| 21:30.03 | Strom_C | only kinda? |
| 21:30.06 | FuriousGeorge | kinda really |
| 21:30.10 | rob0 | And you enjoy participating in the abuse! |
| 21:30.18 | droops | Strom_C, no catsex in #asterisk |
| 21:30.33 | Strom_C | but I haven't even said that today! |
| 21:30.46 | Qwell[] | said what? |
| 21:30.50 | Strom_C | catsex |
| 21:30.52 | Qwell[] | ha |
| 21:30.53 | droops | it was a premtive strike |
| 21:30.58 | droops | Strom_C, no catsex in #asterisk |
| 21:31.09 | Qwell[] | postemptive? |
| 21:31.15 | droops | yes |
| 21:32.00 | droops | hey strom where is this so calls slast logo? |
| 21:32.05 | Strom_C | on my laptop |
| 21:32.07 | Strom_C | which is in the car right now |
| 21:32.15 | droops | thats a great place for it |
| 21:32.17 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 21:32.17 | droops | =o) |
| 21:32.41 | *** join/#asterisk ssgt2002 (n=ssgt2002@69.88.77.125) |
| 21:36.50 | Strom_C | droops: I have to go out anyway, so hopefully you'll be here when I return with the laptop |
| 21:38.34 | *** join/#asterisk copantl (n=FreePBX1@190.4.22.82) |
| 21:38.55 | copantl | hi |
| 21:39.26 | copantl | i need help with a te205p |
| 21:40.00 | copantl | some one from digium please |
| 21:40.14 | Strom_C | copantl: if you want someone from digium, call support :) |
| 21:40.25 | Strom_C | but otherwise we can fumble our way through it |
| 21:41.07 | copantl | Strom_C: and the guys from digium send me to #astersk |
| 21:41.19 | *** join/#asterisk pingywon (n=mike@c-71-230-221-39.hsd1.pa.comcast.net) |
| 21:41.43 | Strom_C | what do you need help with, exactly? |
| 21:43.52 | copantl | i tried to install a te205 and give me this error ZT_CHANCONFIG failed on channel 49: No such device or address (6) |
| 21:44.25 | Strom_C | are you sure you talked to digium support about that? sounds like a classic support issue |
| 21:44.26 | Heimidal | hmm.. anyone know why a voicemail message wouldn't work? we've recorded both a busy and unanswered greeting and it still goes to the default greeting |
| 21:44.35 | Strom_C | anyway, show me your zaptel.conf |
| 21:44.40 | Strom_C | and your zapata.conf |
| 21:44.43 | Strom_C | ~pb |
| 21:44.50 | jbot | pb is, like, 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/ |
| 21:45.15 | FuriousGeorge | how would i go about having allison say the CID after the message body |
| 21:45.22 | FuriousGeorge | when checking a message |
| 21:45.34 | FuriousGeorge | i cant find it in voicemail.conf, i could of swore it was there |
| 21:46.50 | *** join/#asterisk nagl (n=nagl@86.59.54.237) |
| 21:47.09 | *** join/#asterisk Jaxxan (n=jaxxan@202.70.125.60) |
| 21:47.13 | copantl | <Strom_C> http://pastebin.ca/76012 |
| 21:48.00 | Qwell[] | 16:41 < copantl> Strom_C: and the guys from digium send me to #astersk |
| 21:48.02 | Qwell[] | I call BS |
| 21:48.06 | pythos | I would like some help with my ata and SIP.conf, anyone game? |
| 21:48.08 | Strom_C | I call BS too |
| 21:48.26 | Strom_C | this is quite clearly a digium installation support issue :) |
| 21:48.27 | Qwell[] | FuriousGeorge: could HAVE.. |
| 21:48.30 | FuriousGeorge | i found saycid, but i guess it always comes before the message |
| 21:48.53 | FuriousGeorge | Qwell[]: yeah i know, i just get phonetic from time to time when i type |
| 21:49.06 | Qwell[] | phonetic? |
| 21:49.15 | FuriousGeorge | i write like it sounds |
| 21:49.16 | Qwell[] | it's clearly "couldave" |
| 21:49.38 | FuriousGeorge | not cud-ahve but cud-of |
| 21:49.41 | FuriousGeorge | cud-uf |
| 21:49.54 | FuriousGeorge | cud-uv |
| 21:50.05 | Strom_C | Qwell[]: don't even bother...I've heard about three hundred excuses for "could of" instead of "could have" |
| 21:50.19 | Corydon76-home | Coulda woulda shoulda? |
| 21:50.31 | Strom_C | every time, the person refuses to admit that they just didn't know that the phrase is "could have" |
| 21:50.36 | Corydon76-home | Or the southernism "might could" |
| 21:50.37 | FuriousGeorge | its like when people say no instead of know from time to time |
| 21:50.50 | FuriousGeorge | that doesnt meant they dont know how to say it correctly. |
| 21:51.02 | Strom_C | speech and written language are two entirely different things :) |
| 21:51.18 | FuriousGeorge | s/write/say |
| 21:51.24 | devicenode | Strommy Boyyyyyy |
| 21:51.27 | Qwell[] | one time, I was trying to spell "windy" |
| 21:51.30 | Strom_C | file |
| 21:51.34 | Qwell[] | and I accidently spelled it "windy".. |
| 21:51.40 | Qwell[] | I was like "omg, lol" |
| 21:51.59 | devicenode | lollerskatez |
| 21:52.05 | Strom_C | roflgibletz |
| 21:52.23 | Corydon76-home | One time, I was trying to spell "freakishly homosexual" and I accidently spelled it "Qwell" |
| 21:52.30 | FuriousGeorge | lol |
| 21:52.31 | Qwell[] | wtf |
| 21:52.32 | *** part/#asterisk variable_office (n=variable@Adv-Proprietary-Systems.s7-0-0.2-15-0.ar4.CHI1.gblx.net) |
| 21:52.33 | devicenode | oh no you didn't |
| 21:52.53 | devicenode | Qwell[]: hide behind me, I have magical powers |
| 21:52.56 | devicenode | I will protect you! |
| 21:52.58 | Qwell[] | just because you spoon a guy, doesn't mean you're gay :P |
| 21:53.07 | FuriousGeorge | one time i was going to say "im going to" and i accidentally said "ima" |
| 21:53.18 | Corydon76-home | Qwell[]: that's true... you could be a woman... |
| 21:53.56 | Qwell[] | devicenode: I'm scared of you too :P |
| 21:55.40 | FuriousGeorge | and go camping |
| 21:55.47 | pythos | so, pretty much its the weekend... no voip stuff allowed anymore today then? |
| 21:56.01 | FuriousGeorge | and wake up covered in vasoline |
| 21:56.20 | Corydon76-home | v-a-s-e-l-i-n-e |
| 21:56.30 | Qwell[] | g-a-s-o-l-i-n-e |
| 21:56.34 | rob0 | I thought he meant petrol, yes |
| 21:56.41 | Strom_C | c-a-t-s-e-x |
| 21:56.41 | FuriousGeorge | sure, thats the name brand, im talking about the generic |
| 21:56.46 | Corydon76-home | Besides, nobody uses vaseline anymore. http://www.boybutter.com |
| 21:57.11 | Strom_C | what the hell. the generic is "petroleum jelly" |
| 21:57.23 | Strom_C | stop trying to cover for your bad spelling! :) |
| 21:57.46 | *** join/#asterisk ajmn (i=AJmn@70.59.126.204) |
| 21:57.47 | FuriousGeorge | maybe where you are it thats what they call it |
| 21:57.49 | Corydon76-home | and nobody uses Kentucky jelly either |
| 21:57.49 | rob0 | It could of been an intentional ploy |
| 21:57.53 | Qwell[] | COULD HAVE |
| 21:57.57 | rob0 | HAHA |
| 21:57.57 | FuriousGeorge | here its on the shelf next tot he melk |
| 21:57.59 | Qwell[] | damnit |
| 21:58.01 | ajmn | Anyone know of a billing system for asterisk that will print out bills? |
| 21:58.09 | Qwell[] | FuriousGeorge: heh, my friend says melk |
| 21:58.15 | FuriousGeorge | and pellow? |
| 21:58.29 | Qwell[] | FuriousGeorge: no, just melk |
| 21:58.43 | Corydon76-home | So is it wadder or wooder? |
| 21:58.48 | FuriousGeorge | ask him to say pillow enxt time you see him |
| 21:58.53 | FuriousGeorge | i was about to mention wudder |
| 21:59.06 | Qwell[] | FuriousGeorge: he's said it...it's JUST milk that he can't say, heh |
| 21:59.19 | FuriousGeorge | Qwell[]: try water |
| 21:59.26 | Qwell[] | just melk :p |
| 21:59.29 | FuriousGeorge | :) |
| 22:00.09 | Qwell[] | he doesn't have any accent, and he pronounces most words thoroughly... |
| 22:00.15 | *** join/#asterisk jsaunders (n=root@S01060060971c5817.va.shawcable.net) |
| 22:00.39 | Corydon76-home | Oh, and is the highway a rode or a rood? |
| 22:00.39 | FuriousGeorge | i got a friend who says all of the above, and hes just from jersey like the rest of us. i dunno where he gets it from |
| 22:02.30 | FuriousGeorge | so what i ended up doing is just adding the lines callerid = "unknown 000" or something to the end of my zapata.conf, since thats what asterisk sees fit to use when it doesnt get CID info |
| 22:03.32 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 22:03.48 | FuriousGeorge | and how come set(__${CALLERIDNAME}=MM-${CALLERIDNAME}) doesnt do what i think it should do |
| 22:04.18 | Qwell[] | because you're setting the variable "__John Doe" |
| 22:04.40 | FuriousGeorge | i should put the var in quotes? |
| 22:04.48 | Qwell[] | no, remove the ${} |
| 22:04.58 | Qwell[] | and don't use CALLERIDNAME... CALLERID(name) |
| 22:05.23 | Qwell[] | Corydon76-home: has that been deprecated for removal in 1.4? :p |
| 22:05.38 | Corydon76-home | Yes |
| 22:05.46 | Qwell[] | already gone in trunk? |
| 22:06.00 | Corydon76-home | CALLERIDNAME, yes |
| 22:06.03 | Qwell[] | excellent |
| 22:06.16 | Corydon76-home | CALLERID I think is still there until after 1.4 is released |
| 22:06.23 | Corydon76-home | It was an oversight in 1.2 |
| 22:06.36 | Qwell[] | CALLERID the var, not the func, I assume |
| 22:07.12 | Strom_C | speaking of which, when the hell are we getting our happy happy joy joy 1.4 release? |
| 22:07.24 | Qwell[] | Strom_C: 3 weeks ago. where have you been? |
| 22:07.38 | Strom_C | under a rock |
| 22:07.44 | Corydon76-home | Right |
| 22:07.49 | Strom_C | with my tv and my toaster and my steel belted radials |
| 22:08.05 | *** join/#asterisk L|NUX (n=linux@202.5.145.56) |
| 22:08.06 | devicenode | 1.4? that doesn't exist... no... |
| 22:08.21 | droops | Strom_C, get your laptop |
| 22:08.35 | dlynes_office | holy heat wave batman |
| 22:08.45 | Strom_C | ok ok |
| 22:08.47 | Strom_C | going now |
| 22:08.48 | Strom_C | back later |
| 22:08.50 | droops | =o) |
| 22:08.51 | Corydon76-home | 1.4 will be released when it's ready |
| 22:08.59 | Corydon76-home | and not a day before |
| 22:09.21 | dlynes_office | 1.4's slated for August 8th, 2008. |
| 22:09.32 | L|NUX | does any one have eyebeam with g723 ? |
| 22:09.40 | *** part/#asterisk nortex (n=nortex@64.136.65.142) |
| 22:12.49 | *** part/#asterisk wintix (n=tobias@pegel-neuburg.de) |
| 22:17.21 | *** join/#asterisk brijn (n=brijnier@204.244.176.116.net-conex.com) |
| 22:21.34 | *** join/#asterisk phigwork (n=phigan@71-209-152-225.phnx.qwest.net) |
| 22:21.53 | phigwork | Anyone have Stanaphone configured with asterisk? |
| 22:23.57 | L|NUX | does any one have eyebeam with g723 ? |
| 22:24.38 | eKo1 | I have eyebeam. I don't remember if it comes with g723 though... |
| 22:27.57 | L|NUX | humm |
| 22:30.20 | eKo1 | L|NUX: have what? eyebeam? |
| 22:30.28 | eKo1 | Go buy it. |
| 22:33.22 | NotJohnDavid | what would it mean of PSTN line voltage is 4V at idle ? |
| 22:34.35 | NotJohnDavid | now it's 10V |
| 22:35.38 | eKo1 | on idle? |
| 22:35.46 | dlynes_office | now it's 50V |
| 22:40.56 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.235.56.Dial1.SanJose1.Level3.net) |
| 22:41.06 | trixter | I buy -48vdc computers so I can run them cheaper by using a pots line as the power cord! |
| 22:41.57 | eKo1 | no no no, you buy a -48vdc to 120vac transformer |
| 22:42.44 | NotJohnDavid | why is it 50. it should be -50ish |
| 22:42.49 | trixter | dlynes_home: in north america a pots line is -48vdc (ring, tip is switched) when nothing is there. When you pick it up it goes down to about 7-10vdc, when it rings its like 90-120vac@20Hz |
| 22:42.51 | trixter | or so |
| 22:43.26 | trixter | if you use like a 10k ohm resistor between your meter and the line you shouldnt trigger the off hook status |
| 22:43.52 | trixter | then it should stay up there about -48 and not drop down |
| 22:44.04 | phigwork | i wanna power my pc through phone power |
| 22:44.25 | NotJohnDavid | codec (711/729a etc) doesn't have anything to do with CallerID does it? |
| 22:44.30 | Spy000007 | anyone know any good SuperMicro resellers on the east coast? |
| 22:44.35 | trixter | notjason: no |
| 22:44.40 | trixter | er NotJohnDavid |
| 22:45.26 | NotJohnDavid | callerid was working on my voip line now it doesn't |
| 22:45.32 | NotJohnDavid | and the only thing i changed was the codec |
| 22:45.33 | tclark | any one tell me a url for t1 cause codes ? |
| 22:45.39 | trixter | are you certain your provider is still sending it? |
| 22:45.53 | NotJohnDavid | not certain but i dont have any other way to know |
| 22:46.21 | trixter | caller id is sent at the signalling level, codecs happen at the media level |
| 22:47.01 | NotJohnDavid | oh... actually i think it may be that i changed the ring pattern |
| 22:47.06 | trixter | analog lines are inband so technically all of that is the same 'channel' but meh that is somewhat different because its still abstracted |
| 22:47.09 | NotJohnDavid | let me check that. maybe the phone was freaked out by a different ring pattern |
| 22:47.27 | NotJohnDavid | i forgot that i had changed the pattern |
| 22:47.41 | trixter | if its an analog phone it must have a specific ring pattern so it knows to get caller id, some phones are good at tolerating a lot of differenvces cheap ones arent |
| 22:48.05 | NotJohnDavid | okay that was the problem. distinctive rings |
| 22:48.09 | NotJohnDavid | i'm not sure why... but it was |
| 22:48.36 | NotJohnDavid | well i've got three phones here. none of them got it. which really bites because now i don't know which line the calls are coming in on |
| 23:01.01 | mitcheloc | does anyone here know how to get chan_jingle to work/compile? |
| 23:01.36 | dlynes_office | mitcheloc: gcc? |
| 23:02.07 | *** join/#asterisk roche (n=roche@200.122.154.250) |
| 23:02.15 | mitcheloc | dlynes_home, i checked out asterisk svn-head and compiled it just fine, when starting asterisk i don't see the chan_jingle module loaded |
| 23:02.33 | dlynes_office | mitcheloc: edit your modules.conf file to tell it to load? |
| 23:02.45 | mitcheloc | my modules.conf has autoload=yes |
| 23:02.46 | dlynes_office | mitcheloc: and then do a load chan_jingle.so from the asterisk cli? |
| 23:03.17 | *** join/#asterisk [TK]D-Fender (n=joe@CPE0080c6fce706-CM00159a09e31c.cpe.net.cable.rogers.com) |
| 23:03.18 | dlynes_office | mitcheloc: do you get any errors if you try load chan_jingle.so from the cli? |
| 23:03.34 | *** join/#asterisk greendisease (n=jack@fedora/greendisease) |
| 23:03.45 | devicenode | [TK]D-Fender: stealing wifi some more? shame on you |
| 23:03.49 | [TK]D-Fender | :D |
| 23:04.11 | [TK]D-Fender | I prefer "picking up other people's leftovers" thank you ;) |
| 23:04.21 | mitcheloc | dlynes_home, it says the file doesn't exist |
| 23:04.30 | dlynes_office | mitcheloc: so maybe you forgot to install it |
| 23:04.48 | mitcheloc | "make install"? i'm pretty sure i did that |
| 23:04.50 | dlynes_office | mitcheloc: ls -al /usr/lib/asterisk/modules/*jingle* |
| 23:05.47 | *** join/#asterisk marv0997 (i=marv0997@190.4.2.83) |
| 23:06.12 | mitcheloc | dlynes_home, nothing in there, i'm checking my asterisk "make install", i thought i ran it |
| 23:07.48 | dlynes_office | anyways...the heat in this office is absolutely stifling |
| 23:07.52 | dlynes_office | I'm heading out |
| 23:08.45 | rene- | intel macs have no SIP phones |
| 23:12.02 | *** join/#asterisk Wi_Fi (n=OUT@c-24-127-12-85.hsd1.ca.comcast.net) |
| 23:14.02 | *** join/#asterisk Helmchen (i=andre@geddert.net) |
| 23:14.12 | Helmchen | hi |
| 23:15.56 | *** join/#asterisk pjenvey (n=pjenvey@groovie.org) |
| 23:16.43 | *** join/#asterisk Kis (i=vlad@p5080FDAE.dip.t-dialin.net) |
| 23:23.49 | *** join/#asterisk mDuff (n=chatzill@user-12lmn28.cable.mindspring.com) |
| 23:24.23 | *** join/#asterisk Skarmeth (n=Skarmeth@201009048002.user.veloxzone.com.br) |
| 23:24.30 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 23:24.30 | *** mode/#asterisk [+o russellb] by ChanServ |
| 23:24.51 | Helmchen | someone (maybe from germany) here who is bored and want to help a *-newbie with a simple dialplan? |
| 23:32.09 | DrkShdw | Helmchen: just ask. lots of people here are helpful. |
| 23:34.04 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 23:35.37 | mitcheloc | is anyone else familiar with chan_jingle? i just recompiled asterisk, still no chan_jingle.so module |
| 23:36.10 | trixter | did you edit the makefile to tell make to build it? |
| 23:38.41 | *** join/#asterisk SwK (n=Silik0nJ@12-214-3-254.client.mchsi.com) |
| 23:39.31 | russellb | mitcheloc: you need libiksemel |
| 23:40.10 | knarfly | Can anyone field an moh question? |
| 23:41.22 | mitcheloc | russellb, is that just from an apt-get/yum repo or in asterisk svn? |
| 23:41.32 | russellb | apt/yum/whatever |
| 23:41.36 | russellb | or google |
| 23:41.47 | mitcheloc | okay =) |
| 23:41.55 | russellb | then re-run configure |
| 23:42.05 | russellb | and run "make menuselect" make sure it's enabled to be built |
| 23:42.12 | russellb | if you see XXX by it, the lib still hasn't been found |
| 23:42.48 | mitcheloc | russellb, okay, i just checked out their code from the jabberstudio repo |
| 23:44.57 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-246-145.buckeyecom.net) |
| 23:44.59 | mitcheloc | russellb, looks like i'll have to find a prebuilt package? their svn code doesn't "./configure" properly |
| 23:45.08 | russellb | heh |
| 23:45.11 | russellb | i don't remember how i installed it |
| 23:45.42 | mitcheloc | www.pastebin.ca/76049 |
| 23:46.03 | russellb | i got it from apt, debian unstable |
| 23:46.06 | Qwell[] | mitcheloc: cat INSTALL |
| 23:46.07 | Qwell[] | :P |
| 23:46.59 | russellb | looks like you need gnu tls :) |
| 23:47.36 | mitcheloc | installed, trying again |
| 23:47.37 | Qwell[] | so...this kinda bothers me |
| 23:47.45 | Qwell[] | (this is incredibly offtopic and random btw) |
| 23:47.59 | Qwell[] | if you've got a 20oz bottle of soda near you, you can follow along |
| 23:48.27 | DrkShdw | will a 16.9oz bottle of water work? |
| 23:48.29 | Qwell[] | serving size: 1 cup (240mL) |
| 23:48.36 | Qwell[] | servings per container: 2.5 |
| 23:48.48 | Qwell[] | now, here's where things get a little...interesting |
| 23:48.59 | Qwell[] | 20 fl oz, 1.25pt, 591mL |
| 23:49.13 | Qwell[] | 240 * 2.5 is not 591 |
| 23:49.35 | Qwell[] | let's continue |
| 23:49.40 | russellb | we need to get rid of our crappy system of units |
| 23:49.45 | Qwell[] | sodium per serving: 35mg |
| 23:49.48 | DrkShdw | wow, when you said 'you can follow along' I thought it was gonna be something fun... |
| 23:49.50 | Qwell[] | sodium per container: 100mg |
| 23:50.02 | Qwell[] | 35mb * 2.5 is not 100mg |
| 23:50.16 | russellb | milli-bits? |
| 23:50.30 | Qwell[] | total carb: 27g/66g |
| 23:50.34 | russellb | i'm not even sure what that would be |
| 23:50.38 | Qwell[] | again...that doesn't match |
| 23:50.47 | Qwell[] | carbs from sugar: 27g/64g |
| 23:50.52 | Qwell[] | hwf does THAT work? |
| 23:50.56 | Qwell[] | htf* |
| 23:51.34 | Qwell[] | </rant> |
| 23:51.56 | Qwell[] | You can all blame devicenode btw, for letting me be bored enough to read the label on my dr. pepper |
| 23:52.01 | *** join/#asterisk reza_ (i=reza@abort.boom.net) |
| 23:52.24 | reza_ | so i just got a IP 601 phone -- it wants a password to get to the advanced setup menu, but came with no manual |
| 23:52.35 | Qwell[] | reza_: google? |
| 23:52.54 | reza_ | any idea what it's supposed to be? and any idea where i can find a link to configure it wrt asterisk |
| 23:53.01 | reza_ | qwell- checking, havn't found it yet |
| 23:54.12 | mitcheloc | i skipped using their svn and used a package from rpmfind.net - iksemel-1.2-1.1.fc3.rf.i386.rpm, it installed fine, make menu select doesnt' detect it |
| 23:57.16 | reza_ | anyone know how to get asterisk to listen on two different networks on the same interface? |
| 23:59.01 | Qwell[] | reza_: bindaddr=0.0.0.0 and a firewall to block other interfaces |
