| 00:00.10 | drmessano | Seems like a huge waste of time to me |
| 00:00.23 | disposable | drmessano: the server has a public ip and although i dont' think it's in any danger (i use tcp wrappers), learning is my main objective. |
| 00:00.31 | hardwire | fas3r: what phone? |
| 00:00.44 | fas3r | 7942G and 7921 (wifi) |
| 00:00.57 | drmessano | disposable: Don't learn something that's only going to cause you problems when implemented wrong |
| 00:01.23 | drmessano | ebay |
| 00:01.56 | hardwire | fas3r: and you're using call manager? |
| 00:02.22 | fas3r | hardwire: at work yes but now it's for home |
| 00:02.23 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 00:02.32 | drmessano | So they're stolen? |
| 00:02.35 | deeperror | disposable: http://pastebin.ca/1032673 |
| 00:02.38 | disposable | drmessano: i'd say learning how to set up firewall on asterisk box is a pretty useful thing. the reason why i'm asking for somebody to share their working rules is that i DON'T implement it incorrectly |
| 00:02.41 | hardwire | fas3r: typically this channel will have you just use SIP |
| 00:03.07 | hardwire | fas3r: it may be easier to just use a routed tunnel back to work. |
| 00:03.31 | drmessano | fas3r: Is this even for Asterisk? |
| 00:03.31 | hardwire | can you get an ipsec/gre tunnel set up through some edge equipment? |
| 00:03.39 | hardwire | drmessano: haha, I totally wanted to ask that. |
| 00:03.45 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 00:03.50 | disposable | deeperror: thank you. |
| 00:04.04 | drmessano | fas3r: Is this even for Asterisk? |
| 00:04.40 | drmessano | fas3r: Crickets... crickets |
| 00:04.58 | deeperror | just google each of those commands to learn the switches and what is going on |
| 00:05.05 | hardwire | heh :) |
| 00:05.12 | fas3r | drmessano: sorry i don't understand the question .. :s |
| 00:05.20 | hardwire | fas3r: suddenly no english? |
| 00:05.22 | hardwire | haha |
| 00:05.26 | fas3r | :s |
| 00:05.34 | drmessano | fas3r: Will these phones be uses on an Asterisk(R) Telephony System? |
| 00:05.39 | drmessano | used* |
| 00:05.56 | fas3r | drmessano: yes i want tey work on asterisk |
| 00:06.07 | drmessano | fas3r: Or will you be using these to connect back to your CCM at work? |
| 00:06.32 | fas3r | drmessano: i have see on the 7921 it's possible i create different profil |
| 00:06.39 | hardwire | fas3r: If you wanna use asterisk at home to connct to the CCM as well as an SCCP phone, and handle routing.. then yay |
| 00:06.43 | hardwire | fas3r: just nod your head. |
| 00:06.47 | fas3r | but the 7942G stay at home |
| 00:07.08 | fas3r | ha non it's not possible i route it to my company |
| 00:07.17 | drmessano | fas3r: You CAN use the SIP firmware, but the SIP firmware supports less features |
| 00:07.30 | drmessano | fas3r: There's work to getting sccp set up, but worth it |
| 00:07.38 | hardwire | drmessano: does asterisk support any of the SCCP firmwares features anyways? |
| 00:07.56 | fas3r | drmessano: i know exist a ios sip for the 7942G i will download it tomorrow i need CCID user ident |
| 00:08.20 | fas3r | but not for the 7921 and i want to use at term |
| 00:08.31 | drmessano | hardwire: It's not so much the fluff the phone has to offer, it's that SIP firmware leaves off basic shit that you would expect any self respecting IP phone to support |
| 00:08.58 | drmessano | The SIP firmware doesn't even do all ASTERISK can do.. |
| 00:08.58 | hardwire | you believe it is crippled to support SIP but promote SCCP? |
| 00:09.15 | drmessano | I dont know about Crippled.. but it's lacking |
| 00:09.27 | disposable | deeperror: what are ports 5004:5082 for? |
| 00:09.47 | fas3r | hardwire: drmessano : you use sccp ? |
| 00:10.06 | drmessano | fas3r: No, I quit the communist party years ago |
| 00:10.10 | hardwire | drmessano: can he just DNAT in a range of ports and set his phone to only use that range? |
| 00:10.35 | fas3r | because i haven't understand .. i need to patch asterisk or not ... because i don't find how to :) |
| 00:10.40 | coppice | every time I choose Microsoft products, I choose Cisco to go with them |
| 00:10.51 | drmessano | hardwire: Never got that far into implementing SCCP.. I just know that it will and won't do.. Couldn't even tell you the port requirements |
| 00:11.09 | drmessano | fas3r: Its real simple |
| 00:11.51 | fas3r | drmessano: maybe it's just i don't know where i configure it ? in the sip.conf too ? |
| 00:12.02 | drmessano | fas3r: SIP Firmware = Easier setup, less features than say a Polycom would do out of the box.. SCCP Firmware = Harder setup, much closer to supporting all a polycom (or any other phone) will do with asterisk |
| 00:12.15 | deeperror | sip calls |
| 00:12.24 | deeperror | that was open |
| 00:12.25 | drmessano | It's not SIP |
| 00:12.28 | drmessano | It's SCCP |
| 00:12.37 | deeperror | you should updpate it to what you need |
| 00:12.53 | drmessano | I would suggest making SCCP work |
| 00:12.57 | fas3r | deeperror: with http://chan-sccp.berlios.de/ this ? |
| 00:13.01 | *** join/#asterisk anthm (n=anthm@99-201-251-253.area4.spcsdns.net) |
| 00:13.09 | drmessano | Bananaskin can help you if hes around.. hes the man with SCCP |
| 00:13.12 | deeperror | huh? i'm not talking about sccp haha |
| 00:13.15 | fas3r | because it exist a lot of patch :) |
| 00:13.55 | hardwire | A lot of patch. |
| 00:14.01 | hardwire | that sounds like some sort of innuendo. |
| 00:14.14 | drmessano | Sounds like a woman I met at a biker bar |
| 00:14.19 | *** join/#asterisk drummond_ (n=root@firewall.landofhaze.net) |
| 00:14.34 | hardwire | Maybe a Robin Williams movie festival? |
| 00:15.03 | *** join/#asterisk russellb (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 00:15.03 | *** mode/#asterisk [+o russellb] by ChanServ |
| 00:15.21 | drummond_ | why is it everytime i do a yum update on my centos box, i have to rebuild zaptel? |
| 00:15.34 | drmessano | http://sourceforge.net/projects/chan-sccp-b |
| 00:15.44 | drmessano | drummond_: Kernel changes |
| 00:16.10 | drummond_ | no way to make it static? |
| 00:16.18 | drmessano | Dont update? |
| 00:16.22 | hardwire | drummond_: uninstall the kernel |
| 00:16.29 | hardwire | is dead serious |
| 00:17.00 | hardwire | remove it from the installed package list :) |
| 00:18.16 | fas3r | drmessano: i need to use it ? |
| 00:18.41 | drmessano | fas3r: I was giving you a source of SCCP |
| 00:18.46 | drmessano | You have 3 options here |
| 00:18.49 | drmessano | Ready? |
| 00:18.53 | fas3r | :) |
| 00:19.16 | drmessano | 1. Use SIP and have it do FAR less than the phone is capable of, and less than Asterisk supports |
| 00:19.46 | drmessano | 2. Use SCCP and have it do less than the phone is capable of, and much closer to what asterisk supports |
| 00:19.59 | drmessano | 3. Sell them on eBay, get a copy Polycoms, have a beer |
| 00:20.04 | drmessano | couple* |
| 00:20.14 | drmessano | Me, I like Beer |
| 00:20.20 | drmessano | Cisco even less |
| 00:20.20 | fas3r | :) |
| 00:21.30 | hardwire | I vote for ignoring asterisk all together and creating a GRE tunnel back to work. |
| 00:21.31 | [TK]D-Fender | fas3r, You've got chan_skinny, USE IT |
| 00:21.33 | hardwire | I like wormholes |
| 00:21.51 | [TK]D-Fender | fas3r, in the time you've morosed about this you could probably have had it working. |
| 00:22.18 | hardwire | I like beer wormholes. |
| 00:22.21 | fas3r | [TK]D-Fender: where i find it ? |
| 00:22.26 | hardwire | fas3r: you haz it. |
| 00:22.30 | [TK]D-Fender | fas3r, its part of *. |
| 00:22.31 | drummond_ | has anyone ever setup a dial by name system? |
| 00:22.38 | fas3r | oki |
| 00:22.43 | hardwire | drummond_: app_directory? |
| 00:22.50 | fas3r | i will test |
| 00:22.55 | [TK]D-Fender | fas3r, wake up, go look at the sample configs, get your hands off your nuts and SEIZE THE DAY. |
| 00:23.09 | hardwire | off nuts, on day, check. |
| 00:23.28 | fas3r | [TK]D-Fender: you use sccp ? |
| 00:23.33 | fas3r | ok ok i go |
| 00:23.35 | drmessano | I've been told the other SCCP works better |
| 00:24.29 | deeperror | in meetme does the E or d options just select a random unused conference listed in meetme.conf? |
| 00:24.38 | russellb | [TK]D-Fender: the number of variations on "RTFM" that you come up with continues to amaze me |
| 00:24.52 | [TK]D-Fender | drmessano, I believe it was the reverse. When Qwell rouses he can confirm as he used to maintain one of these. As Skinny is the one thats included with 1.4 and I know one isn't maintained any more and was less stable, I'd put my money on the more obvious looking choice... |
| 00:24.52 | hardwire | deeperror: what does voip-info say? |
| 00:25.00 | hardwire | or "show application meetme" in the console |
| 00:25.06 | deeperror | i'm seeing d = dynamically allocates a new conference |
| 00:25.19 | russellb | Qwell is pretty much the maintainer of chan_skinny in asterisk |
| 00:25.28 | deeperror | wondering how dynamic that really is haha |
| 00:25.37 | [TK]D-Fender | russellb, not even an RTFM, more of a "you have sample configs, just start puching buttons randomly if you have to but FFS do SOMETHING" :) |
| 00:25.39 | russellb | dynamic == doesn't have to be in meetme.conf |
| 00:25.40 | hardwire | deeperror: I think it's dynamic as in it doesn't exist to anybody else. |
| 00:25.47 | [TK]D-Fender | russellb, alternative enticement :) |
| 00:25.50 | deeperror | very cool testing it now |
| 00:25.52 | drmessano | [TK]D-Fender: I don't know.. There is one OLD ONE, and then theres the SCCP-B last dated 12/07.. Which isn't THAT old.. but even still, yeah, Qwell would know |
| 00:25.53 | russellb | hehe |
| 00:26.32 | [TK]D-Fender | russellb, I like to think outside the box.... however don't forget it loses most of its value once you break the shrink-wrap ;) |
| 00:26.43 | *** join/#asterisk timothy_jones (n=chatzill@207.90.4.189) |
| 00:26.55 | znoG | [TK]D-Fender: how come you were asking before what led me to choose LDAP? |
| 00:27.19 | drmessano | I like LDAP because I can store my AsteriskWin32 configs in active directory |
| 00:27.22 | drmessano | ducks |
| 00:27.25 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-246-236.balt.east.verizon.net) |
| 00:27.36 | hardwire | I should work |
| 00:27.37 | hardwire | bbl |
| 00:27.37 | [TK]D-Fender | znoG, well its not an RDBMS so there had to be something right? SQL based is just so much friendlier. |
| 00:27.40 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 00:28.19 | frieze | [TK]D-Fender: I've got a weird linux permissions thing vexing me. when I try to start asterisk as a service (or using safe_) it gives a permissions error trying to create the pid file and fails |
| 00:28.37 | frieze | the /var/run/asterisk and /var/log/asterisk dirs belong to asterisk.asterisk and are writable by same |
| 00:28.44 | [TK]D-Fender | frieze, Check the perms to the folder and file |
| 00:29.16 | [TK]D-Fender | frieze, and double check who * is running as. |
| 00:30.20 | frieze | [TK]D-Fender: how? while services are always started by root my understanding is that the whole point of them is setting different users to run the actual services |
| 00:30.29 | frieze | in this case it should be asterisk.asterisk |
| 00:31.04 | timothy_jones | anybody run into this: When calling UK numbers, we randomly get a recording of a guy who says, Hello, hold on let me go get him... and then you wait on the phone and hear some dog barking in background. Then the guy comes back and says, You still there? Hold on... I complained to our provider by they say it must be something in our asterisk box... right... Anybody run into this? |
| 00:31.41 | frieze | looking at the startup script it seems that asterisk is the user and group, as expected |
| 00:31.48 | [TK]D-Fender | frieze, look at the script that it calls. |
| 00:31.57 | frieze | though it does seem to be looking in /var/run not /var/run/asterisk |
| 00:31.58 | [TK]D-Fender | frieze, asterisk:asterisk is NOT the norm. |
| 00:32.08 | [TK]D-Fender | frieze, there's a hint for you... |
| 00:32.16 | frieze | NAME=asterisk |
| 00:32.16 | frieze | USER=$NAME |
| 00:32.17 | frieze | GROUP=$USER |
| 00:32.38 | frieze | I was assuming that the install script chrooted it |
| 00:32.43 | deeperror | ok if i use d for a dynamic conference how do i get the name or number of the conf that was used? Is there a variable that gets created on this? |
| 00:33.25 | hardwire | deeperror: I think dynamic conferences are mostly used by other apps like app_page |
| 00:33.34 | frieze | okay.... |
| 00:33.35 | hardwire | or special super awesome dialplans for doing similar things |
| 00:33.42 | frieze | so the answer to the riddle is? |
| 00:33.51 | hardwire | sex in the city comes out friday night |
| 00:34.07 | [TK]D-Fender | frieze, look at that folder issue you already found |
| 00:34.59 | deeperror | so will i need to use a database to keep track of conferences and just setup a list of conf in meetme.conf? |
| 00:35.05 | frieze | yeah, editing the PIDFILE var to be /var/run/asterisk/asterisk.pid does not help |
| 00:35.14 | frieze | the error message still says it is looking in /var/run |
| 00:35.30 | timothy_jones | I found this on vonage web site, so looks like I am not the only one: http://www.vonage-forum.com/ftopic22408.html |
| 00:35.37 | frieze | root cannot be the script user |
| 00:36.03 | frieze | and I can't see making /var/run belong to asterisk. so I'm a bit stumped. |
| 00:36.09 | hardwire | timothy_jones: we'd throw money at you if you were the only one to ever be affected by anything. |
| 00:36.58 | timothy_jones | hardwire: did you run into this as well? |
| 00:37.04 | hardwire | never |
| 00:37.08 | hardwire | I'm lolling about it tho |
| 00:37.48 | deeperror | i'm dropping the caller into a conference then calling the agent and having them go through an ivr to accept or reject the call...when accepted they goto the conference with the caller but there could be n calls. What is the best way to handle this? setup n+1 rooms in meetme.conf and expand as needed and use a table to lock and assign rooms per call? |
| 00:37.48 | timothy_jones | man it's really weird. Our guys finally caught on after 4-5 calls |
| 00:37.50 | frieze | [TK]D-Fender: okay so is there any useful advice you can give me? |
| 00:38.01 | hardwire | hahahahaah |
| 00:38.07 | hardwire | this is quite amusing. |
| 00:38.24 | timothy_jones | but when you first get, you've no idea and you don't wanna be rude and just hang up... |
| 00:38.27 | hardwire | deeperror: gimme a sec |
| 00:38.36 | deeperror | hardwire: no prob |
| 00:38.41 | drmessano | Vonage.. |
| 00:38.44 | hardwire | timothy_jones: and you pay for it, right? |
| 00:38.45 | drmessano | Oh Jesus |
| 00:39.08 | timothy_jones | you I bet I pay |
| 00:39.20 | timothy_jones | it's gets worse... |
| 00:39.21 | drmessano | How is VONAGE expected to be SUPPORTED HERE? |
| 00:39.27 | *** join/#asterisk x_or (n=cdawson@68.178.72.172) |
| 00:39.40 | x_or | Is Festival the preferred text-to-speech engine for Asterisk? |
| 00:39.44 | deeperror | i thought vonage was the bee's knees until i found asterisk haha |
| 00:39.56 | hardwire | deeperror: there's better ways to deal with queues like that |
| 00:39.57 | deeperror | x_or: try cepstral |
| 00:40.05 | timothy_jones | I don't use vonage, I just found the link to prove to my provider that I am not crazy |
| 00:40.17 | deeperror | i assume there is |
| 00:40.27 | deeperror | but i may need to have 3 or more in a conference |
| 00:40.32 | deeperror | and record it |
| 00:40.36 | drmessano | Vonage wont support my hardware |
| 00:40.45 | drmessano | "Yes, I have a hacked PAP2...." |
| 00:40.50 | drmessano | *CLICK* |
| 00:40.52 | deeperror | haha |
| 00:40.58 | drmessano | motherfathers |
| 00:40.58 | x_or | deeperror: Hmm, cepstral sounds pretty nice. Thanks. |
| 00:41.00 | [TK]D-Fender | ~asterisk-non-root |
| 00:41.01 | jbot | [~asterisk-non-root] Running * as non-root is covered in chapter 13 of Asterisk:The Future of Telephony 2nd Edition (~book) and an article written by Leif Madsen at http://www.taug.ca/node/115 |
| 00:41.03 | [TK]D-Fender | ^^^^ |
| 00:41.04 | frieze | anyone else know why the asterisk startup script would give me a permission denied error? |
| 00:41.05 | [TK]D-Fender | frie |
| 00:41.35 | [TK]D-Fender | frieze, perms are perms. check them all. your * configs in etc and so on as well. they ENTIRE pile. |
| 00:41.40 | hardwire | deeperror: I bet you'd get a pretty comprehensive amount of feedback if you posted your question on the mailing list |
| 00:41.58 | hardwire | esp about how chan_agents is going buhbye and how to compensate if you haven't already. |
| 00:42.22 | *** join/#asterisk rabelais (n=blank@unaffiliated/rabelais) |
| 00:42.27 | hardwire | but yeh, having a queue call agents and ask them to press a key to accept a call works great |
| 00:42.32 | timothy_jones | they say the recording is in my asterisk.... I almost believed them. And then my wife called Italy yesterday and got similar recording in Italian, except it was TV in background instead of dog |
| 00:42.46 | timothy_jones | :) |
| 00:43.02 | hardwire | deeperror: your dynamic conference rooms + recording = app_queue |
| 00:43.18 | timothy_jones | seems like some sort scam... |
| 00:43.21 | deeperror | hardwire: hrmm let me check this out now |
| 00:43.29 | hardwire | deeperror: I'm out for the rest of the day |
| 00:43.35 | deeperror | ok thanks for the help |
| 00:44.48 | timothy_jones | the weird part is, you hang up and call the number again and it works fine! |
| 00:45.12 | *** join/#asterisk rabelais (n=blank@unaffiliated/rabelais) |
| 00:45.42 | timothy_jones | voodoo |
| 00:47.26 | frieze | [TK]D-Fender: so even /usr/sbin/asterisk should belong to asterisk? that seems unsafe |
| 00:47.29 | deeperror | does setglobalvar only last for the duration of the call? or would another call have the same variable available until it is reset? |
| 00:47.45 | [TK]D-Fender | frieze, * should be able to call itself... |
| 00:48.19 | [TK]D-Fender | deeperror, global = global to * until it restarts |
| 00:48.30 | *** part/#asterisk nny_2 (n=Scott@64.20.130.209.dyn-e-pool3.pool.hargray.net) |
| 00:48.56 | frieze | [TK]D-Fender: okay, did that. no change |
| 00:49.18 | [TK]D-Fender | frieze, start * manually first, and work your way up from there. |
| 00:49.23 | deeperror | so using a global variable to pass data between contexts and channels probably isn't the best idea? what would be the better way of handling that using database? |
| 00:49.30 | frieze | also the fact that the error message says /var/run/asterisk.pid while the script says /var/run/asterisk/asterisk.pid seems troubling |
| 00:50.39 | frieze | [TK]D-Fender: can't seem to get very far. trying to run cli fails because it can't find /var/run/asterisk.ctl |
| 00:51.05 | [TK]D-Fender | frieze, go make it. |
| 00:51.11 | frieze | which, as I have stated before, seems weird since all that stuff should be in /var/run/asterisk not /var/run |
| 00:51.18 | frieze | make what? |
| 00:51.27 | [TK]D-Fender | frieze, the FILE |
| 00:52.00 | frieze | well since I have no idea what is supposed to go in asterisk.ctl that seems tricky |
| 00:53.16 | [TK]D-Fender | frieze, make it so it exists and * can modify it |
| 00:53.44 | seanbright | asterisk.ctl is a socket |
| 00:54.07 | seanbright | domain socket, that is. |
| 00:54.36 | frieze | same problem |
| 00:54.57 | [TK]D-Fender | frieze, pastebin is your friend..... |
| 00:54.58 | [TK]D-Fender | ~pb |
| 00:54.58 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 00:54.59 | frieze | unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) |
| 00:55.15 | [TK]D-Fender | frieze, .... WHO SAID REMOTE? |
| 00:55.28 | [TK]D-Fender | frieze, I said START Asterisk manually. |
| 00:55.32 | frieze | right |
| 00:55.35 | frieze | did that |
| 00:55.43 | [TK]D-Fender | frieze, and? |
| 00:55.44 | frieze | it's (apparently) running |
| 00:55.56 | *** part/#asterisk deeperror (n=deeperro@d149-67-253-63.try.wideopenwest.com) |
| 00:55.59 | [TK]D-Fender | frieze, verify the ctl file |
| 00:56.37 | [TK]D-Fender | frieze, and I didn't say as a DAEMON. I said MANUAL. Now stop it, and restart it MANUALLY |
| 00:57.04 | frieze | I did |
| 00:57.16 | frieze | and there was no /var/run/asterisk.ctl file created |
| 00:57.25 | [TK]D-Fender | ~pb |
| 00:57.26 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 00:57.27 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^ |
| 00:57.28 | frieze | so I tried to connect with cli and it said there was no such file |
| 00:57.33 | frieze | then you said to make one |
| 00:57.35 | frieze | so I did |
| 00:57.51 | frieze | I have no idea why you said to do that |
| 00:57.56 | frieze | but there you go |
| 00:58.32 | frieze | it's an empty file and as such not very helpful to asterisk |
| 00:58.37 | frieze | or anyone |
| 00:58.50 | seanbright | it shouldn't be a normal file |
| 00:58.56 | seanbright | mksock |
| 00:59.05 | seanbright | but you shouldn't have to create it manually |
| 00:59.29 | seanbright | its possible (and likely) that it isn't created when you aren't daemonized |
| 00:59.38 | seanbright | looks |
| 00:59.40 | frieze | okay, killed that file, killed asterisk, and started asterisk again. now I can connect with CLI |
| 00:59.58 | seanbright | started asterisk how? |
| 01:00.01 | frieze | and there is an asterisk.pid |
| 01:00.02 | seanbright | service asterisk start? |
| 01:00.08 | frieze | killall -9 asterisk |
| 01:00.15 | *** join/#asterisk unspin (n=unspin@24.82.161.85) |
| 01:00.33 | frieze | though confusingly now I can probably use services to kill it as it has a pid file (apparently) |
| 01:00.52 | x_or | deeperror: do you know where updated installation instructions for cepstral are? I'm not sure what do with the Makefile because it does not fit the instructions I see online for 1.2 installation. |
| 01:01.13 | seanbright | frieze: "and started asterisk again" <- how did you start it? |
| 01:01.22 | frieze | sudo asterisk |
| 01:01.30 | seanbright | ah, so in the foreground |
| 01:01.41 | frieze | since starting it as a user didn't work and starting as a service doesn't work |
| 01:01.49 | frieze | yes |
| 01:02.01 | seanbright | hrm. |
| 01:02.06 | frieze | as root it can create a pid file in /var/run |
| 01:02.16 | seanbright | is this linux? |
| 01:02.20 | frieze | yes |
| 01:02.23 | seanbright | selinux enabled? |
| 01:02.39 | frieze | the problem is that even though the script says to create the PIDFILE in /var/run/asterisk it still seems to be creating it in /var/run |
| 01:02.47 | frieze | is there another conf setting somewhere for that? |
| 01:02.52 | seanbright | what distro? |
| 01:03.25 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-d6cfe6aff57943fa) |
| 01:03.25 | frieze | ubuntu 8.04 64 |
| 01:03.49 | frieze | nevermind |
| 01:03.51 | frieze | found it |
| 01:03.59 | frieze | the PIDFILE setting in the service scrip is ignored |
| 01:04.25 | seanbright | you're using the vanilla asterisk init.d script? |
| 01:04.26 | frieze | somewhere it is being overwritten with astrundir from /etc/asterisk/asterisk.conf. which d |
| 01:04.31 | frieze | was /var/run |
| 01:04.35 | frieze | I was |
| 01:04.45 | seanbright | that just calls safe_asterisk |
| 01:04.56 | seanbright | which does this: ASTPIDFILE=/var/run/asterisk.pid |
| 01:05.03 | seanbright | (assuming you are on 1.4) |
| 01:05.08 | frieze | what does |
| 01:05.13 | frieze | I am |
| 01:05.15 | seanbright | safe_asterisk |
| 01:05.28 | seanbright | which is what the vanilla asterisk init.d script calls |
| 01:05.39 | *** join/#asterisk stuffcorpse (n=pickle@60-234-225-122.bitstream.orcon.net.nz) |
| 01:05.48 | frieze | I think I inherited a bad init script somewhere |
| 01:05.55 | frieze | oh well, that fixed it either way |
| 01:07.05 | seanbright | so you're good to go? |
| 01:08.53 | frieze | apparently |
| 01:08.58 | frieze | I'm as confused as anyone |
| 01:09.09 | seanbright | heh |
| 01:09.13 | frieze | there is no reason why that should work, but apparently it does |
| 01:09.16 | seanbright | backup everything right now |
| 01:09.16 | seanbright | :) |
| 01:10.01 | drmessano | Ha |
| 01:10.04 | drmessano | You sound like me |
| 01:10.16 | seanbright | bite your tongue |
| 01:10.16 | drmessano | "Oh shit, it's working!.. Quick, COPY IT SOMEWHERE NOW" |
| 01:10.17 | seanbright | ;) |
| 01:10.22 | *** join/#asterisk shido6 (n=shido6@74-130-56-134.dhcp.insightbb.com) |
| 01:10.32 | drmessano | That was me learning apache |
| 01:10.39 | drmessano | Well, still learning |
| 01:11.28 | frieze | hehe thanks everyone for the help |
| 01:14.17 | frieze | now I'm going to go eat some peking duck and read about sip channels and dialplans. fun! |
| 01:16.16 | [TK]D-Fender | . |
| 01:17.01 | drmessano | Im trying to determine how every video on youtube is six degress seperated from the pina colada song |
| 01:17.04 | hsv-al | . |
| 01:17.05 | hsv-al | d-fender |
| 01:17.08 | hsv-al | your still up eh? |
| 01:17.08 | hsv-al | :) |
| 01:18.49 | drmessano | holy crap |
| 01:19.03 | hsv-al | drmessano, I'm pounding away on a fiveguys bacon-cheese burger, and a sugar free red bull while you starve to death. |
| 01:19.10 | drmessano | Someone made a 12"' remix of "Together in Electric Dreams" |
| 01:19.22 | drmessano | Sugar Free red bull? |
| 01:19.29 | drmessano | Isn't that defeating the purpose? |
| 01:19.36 | drmessano | That implies you like the flavor |
| 01:19.39 | hsv-al | it has phenylalinine in it or something as a replacement. |
| 01:19.42 | hsv-al | which is more potent heh |
| 01:19.44 | drmessano | Ah |
| 01:19.50 | mwalling | aspertame i though |
| 01:19.56 | drmessano | If it doesnt cause cancer, I dont want it |
| 01:20.04 | hsv-al | http://www.astro.sunysb.edu/tracy/whatis.html |
| 01:20.08 | fas3r | someone have CCIE user account on cisco ? because it exist IOS SIP for 7942 but i haven't the good right and this good account is at my work ... :s |
| 01:20.10 | seanbright | lights a cigarette |
| 01:20.11 | drmessano | Bring on the lab rats, bitches |
| 01:20.13 | hsv-al | Phenylketonurics - contains phenylalanine |
| 01:20.17 | hsv-al | ever see that on soda drinks? |
| 01:20.19 | *** join/#asterisk Wi_Fi (n=OUT@cpe-76-169-21-21.socal.res.rr.com) |
| 01:20.25 | drmessano | fas3r: WIZARD! |
| 01:20.34 | drmessano | waits for the wizard to pop out |
| 01:20.43 | *** part/#asterisk Wi_Fi (n=OUT@cpe-76-169-21-21.socal.res.rr.com) |
| 01:20.45 | mwalling | aspertame turns my gut inside out |
| 01:20.50 | hsv-al | mwalling: http://www.astro.sunysb.edu/tracy/whatis.html |
| 01:21.16 | drmessano | fas3r: it is illegal to distribute Cisco firmwares like that.. I should have you arrested |
| 01:21.40 | fas3r | it's just i don't want wait tomorrow ... :) |
| 01:21.42 | drmessano | fas3r: Tell me what country you're in, i'm ringing interpol on Sipbroker as we speak |
| 01:21.44 | hsv-al | drmessano, i use ios images in dynamips/dynagen |
| 01:21.49 | hsv-al | via cco account, so whats he doing wrong? |
| 01:22.04 | drmessano | He's asking for someone else to get the images for him |
| 01:22.11 | hsv-al | ahhh |
| 01:22.18 | fas3r | i have account but at work ... |
| 01:22.24 | mwalling | then go to work |
| 01:22.26 | drmessano | Tell that to the judge |
| 01:22.35 | fas3r | no train at this time :) |
| 01:22.40 | fas3r | yes no problem |
| 01:22.45 | drmessano | Maybe interpol can drive you to work |
| 01:22.54 | fas3r | i wait .. it's just i have nothing to do |
| 01:23.04 | *** part/#asterisk x_or (n=cdawson@68.178.72.172) |
| 01:23.16 | drmessano | They're tracing your IP as we speak |
| 01:23.28 | drmessano | 192.168.1.3.. They got your ass |
| 01:23.33 | [TK]D-Fender | fas3r, get off your ass and try chan_skinny |
| 01:23.56 | fas3r | [TK]D-Fender: nop |
| 01:24.01 | drmessano | (come on, someone laugh at that horrid private IP joke) |
| 01:24.03 | hsv-al | d-fender why arent you watching CSI Las vegas |
| 01:24.04 | hsv-al | on spike now? |
| 01:24.07 | hsv-al | u should be tired by now heh |
| 01:24.07 | drmessano | fas3r: Why the hell not? |
| 01:24.29 | [TK]D-Fender | hsv-al, broadcast television? Haven't had that for almost 5 years now. |
| 01:24.35 | fas3r | because SIP is easy no ? |
| 01:24.39 | drmessano | Skinny is much better than SIP and the phones will DO LESS WITH THE SIP FIRMWARE |
| 01:24.53 | drmessano | Oh, you want easy |
| 01:24.54 | [TK]D-Fender | fas3r, Yeah so easy it'll install itself and so much more. You just keep sitting there... |
| 01:24.59 | hsv-al | drmessano, you know what has always confused me |
| 01:25.05 | hsv-al | why some people use asterisk just for sip functionality |
| 01:25.08 | hsv-al | thats like 1% of it HEH |
| 01:25.17 | drmessano | hsv-al: Laziness |
| 01:25.23 | drmessano | hsv-al: Skype replacment |
| 01:25.27 | hsv-al | lol |
| 01:26.42 | drmessano | Sad thing is |
| 01:26.48 | drmessano | hes gonna read the book for that phone |
| 01:26.57 | drmessano | and wonder why so many features dont work |
| 01:27.00 | drmessano | with SIP |
| 01:27.04 | drmessano | "Oh?" |
| 01:27.11 | hsv-al | im not gutting on fas3r, im just mentioning something |
| 01:27.17 | hsv-al | ive obvserved over weeks of conversation in here |
| 01:27.24 | hsv-al | why dio people use * just for SIP, wtf |
| 01:27.47 | fas3r | well write good paper on sccp and i will use it ... |
| 01:28.00 | fas3r | and over people too ... |
| 01:28.32 | [TK]D-Fender | hsv-al, When * offers enough more via something else and better products are out there affordably to relize this potential....THEN we'll talk. |
| 01:29.16 | hsv-al | Im trying to learn as much as I can, because frankly im sick of observing the costs of CCM |
| 01:29.22 | hsv-al | I bought a tdm411e for home use |
| 01:29.22 | *** join/#asterisk BeeBuu (n=beebuu@61.145.77.201) |
| 01:29.46 | hsv-al | what does * average per node, 200-$300? |
| 01:30.10 | [TK]D-Fender | hsv-al, need to pick your specifics. |
| 01:30.13 | drmessano | fas3r: Theres good documentation on getting SCCP going... you already said you wanted "easy".. I'm not ripping on you, but at least be honest about it. |
| 01:30.34 | [TK]D-Fender | hsv-al, $95 / phone + server |
| 01:30.46 | hsv-al | d-fender |
| 01:30.53 | hsv-al | switchvox deployment? |
| 01:31.13 | fas3r | drmessano: i have understand but i have search and i see many different thing about sccp instead of tell what i need to choice or do help me ... :) |
| 01:31.15 | *** join/#asterisk hmmhesays (n=hmmhesay@71-32-30-233.farg.qwest.net) |
| 01:31.37 | [TK]D-Fender | hsv-al, nope. D-Link 24-port PoE switch + Polycom IP320 phone. |
| 01:32.04 | fas3r | well i think it's time go to sleep and wait tomorrow for sip |
| 01:32.09 | hmmhesays | what up folks |
| 01:32.12 | [TK]D-Fender | hsv-al, averaged over 24 ports. |
| 01:32.47 | hsv-al | ip320's are nice |
| 01:33.06 | drmessano | fas3r: If you say so |
| 01:36.15 | *** join/#asterisk Frogzoo (n=Frogzoo@121.217.196.136) |
| 01:36.51 | rift0r | what would cause me not to hear ringing on my softphone |
| 01:36.55 | rift0r | but I can hear them when they pick up |
| 01:37.05 | rift0r | just no initial ringing or busy or anything like that |
| 01:37.10 | rift0r | it's silence and then bam pick up |
| 01:37.33 | shido6 | is the ringer off on the phone? |
| 01:37.34 | shido6 | :) |
| 01:38.25 | rift0r | no |
| 01:38.26 | rift0r | :P |
| 01:38.47 | [TK]D-Fender | lack of call progress |
| 01:40.48 | timothy_jones | fas3r: search voip-info.org you can probably find links to firmware there. I found for 7940 |
| 01:40.53 | timothy_jones | fas3r: search voip-info.org you can probably find links to firmware there. I found for 7940 |
| 01:41.04 | *** join/#asterisk Pegasus_RPG (n=chatzill@cpe-071-076-024-036.sc.res.rr.com) |
| 01:41.33 | Pegasus_RPG | Hello. |
| 01:41.37 | timothy_jones | making sure his anonymous proxy setting is still on |
| 01:42.29 | fas3r | timothy_jones: i search since 2 days ... |
| 01:42.35 | Pegasus_RPG | With the glut of tutorials in various states of completeness across the Web, would someone please point me to one that allows me to get up & running with two SIP hardphones and a Broadvoice trunk? |
| 01:43.24 | Pegasus_RPG | just installed v1.4.19.1 |
| 01:44.47 | fas3r | timothy_jones: why do you want i find firmware ? |
| 01:45.07 | [TK]D-Fender | Pegasus_RPG, Close enough : |
| 01:45.10 | [TK]D-Fender | ~jerjerguide |
| 01:45.11 | jbot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 01:45.15 | [TK]D-Fender | Pegasus_RPG, ~book |
| 01:45.37 | timothy_jones | fas3r: cuz you said you wanted it (I don't have it) |
| 01:45.43 | [TK]D-Fender | Pegasus_RPG, For everything else, GOOGLE. Broadvoice has their own how-tos Use them. |
| 01:46.24 | fas3r | timothy_jones: ha no i wait tomorrow instead of sccp |
| 01:47.20 | Pegasus_RPG | Thank you for that link. It's just what I'm looking for. (I saw Broadvoice's info as well and will be using it when I get to that point.) |
| 01:47.35 | husimon | does anyone have any idea how to get cisco 7940's to use GMT - X timezones? |
| 01:47.50 | husimon | what do you put in the SIPDefault.cnf under timezone : "x" |
| 01:48.02 | *** part/#asterisk Pegasus_RPG (n=chatzill@cpe-071-076-024-036.sc.res.rr.com) |
| 01:48.03 | husimon | sorry "time_zone: 'x'" |
| 01:48.20 | husimon | apparently cisco is RETARDED and thinks that HST means GMT+8 |
| 01:48.27 | husimon | instead of -10 like the rest of the frigging world |
| 01:49.27 | husimon | does it understand stuff like "GMT-10" ? |
| 01:49.31 | lanning | or maybe it doesn't know "HST" and defaults to PST, since that is Cisco's HQ. |
| 01:50.32 | lanning | wait, GMT+8, not -8? |
| 01:50.51 | *** part/#asterisk timothy_jones (n=chatzill@207.90.4.189) |
| 01:52.12 | husimon | odd since i've seen posts with people using HST |
| 01:52.28 | husimon | i think it's +8 |
| 01:52.33 | husimon | even the cisco docs say +8 |
| 01:52.35 | husimon | for HST |
| 01:52.40 | lanning | hmm |
| 01:52.42 | husimon | i just didn't realize it at the time I used HST |
| 01:52.53 | husimon | so the question is, how do I set an arbitrary timezone |
| 01:53.51 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 01:55.56 | husimon | any ideas? |
| 01:56.22 | husimon | and forget cisco actually making a LIST Of their accepted timezones |
| 01:57.23 | drmessano | fas3r: If you want my help, type /server drmessano |
| 01:58.23 | fas3r | it's not /j # ... :) |
| 01:58.39 | drmessano | no |
| 01:59.11 | *** join/#asterisk w9sh (n=w9sh@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 01:59.13 | fas3r | ok garde la peche alors ;) |
| 01:59.30 | husimon | drmessano, I thought your channel was #1,drmessano |
| 01:59.57 | *** join/#asterisk mikong (n=gones@203.193.37.251) |
| 02:00.21 | husimon | ah nm that bug only works in mirc |
| 02:00.21 | husimon | heh |
| 02:00.21 | fas3r | drmessano: you think i don't use ircc or what ? |
| 02:00.30 | drmessano | It's #0 |
| 02:00.39 | fas3r | pss |
| 02:00.48 | drmessano | Damn |
| 02:00.51 | drmessano | That works |
| 02:01.22 | fas3r | why you left ? |
| 02:01.34 | husimon | drmessano, do you have 7940 phones? |
| 02:01.48 | husimon | if so, what do you put in the SIPDefault.cnf under time_zone: ? |
| 02:02.15 | drmessano | I don't use Cisco phones.. I would rather be caught dead |
| 02:02.29 | husimon | me too |
| 02:02.32 | husimon | but we already had them |
| 02:02.33 | fas3r | drmessano: why have you left ? |
| 02:02.40 | husimon | please anyone? |
| 02:03.26 | fas3r | thank drmessano |
| 02:03.32 | drmessano | fas3r: You argue with me for hours in here about using SIP because it's easy, and that SCCP is so hard, and then PRIVATE MESSAGE ME asking me for SCCP help? WTF dude.. Go take your meds |
| 02:04.24 | husimon | can anyone point me at a online resource which will tell me how to set timezones for cisco 79XX phones in the SIPDeafult.cnf |
| 02:04.32 | husimon | I can't figure out what timezone to use for "hawaii" |
| 02:04.42 | fas3r | husimon: you use sccp ? |
| 02:04.47 | drmessano | GMT-10? |
| 02:04.56 | drmessano | ERR |
| 02:04.59 | drmessano | WTF |
| 02:05.01 | husimon | drmessano, yea it is gmt-10 but that won't work in the config |
| 02:05.14 | husimon | fas3r, no it blows i threw away my old sccp conferences phones because it didn't work. |
| 02:05.43 | drmessano | Im trying to think.. Is it today or tomorrow in Hawaii |
| 02:05.54 | husimon | where are you? |
| 02:06.02 | drmessano | That's irrelevant |
| 02:06.06 | husimon | in gmt? |
| 02:06.14 | husimon | hehe |
| 02:06.19 | drmessano | There's an international date line |
| 02:06.27 | drmessano | That's what is relevant |
| 02:06.30 | husimon | depends what time it is |
| 02:06.43 | husimon | if it's near the boundry it would be yesterday |
| 02:06.50 | husimon | otherwise it's today |
| 02:06.53 | drmessano | The other side of |
| 02:06.54 | husimon | would be my thought |
| 02:06.57 | drmessano | Thats the point |
| 02:07.36 | drmessano | GMY-10 doesnt work? |
| 02:07.39 | drmessano | GMT |
| 02:07.41 | husimon | nope |
| 02:07.51 | husimon | "GMT-10" doesn't work |
| 02:08.00 | husimon | and cisco thinks HST = gmt+8 |
| 02:08.09 | husimon | according to SOME of their docs |
| 02:08.15 | mikong | I looking for a tool which can convert mp3 to gsm . |
| 02:09.16 | drmessano | sox |
| 02:09.42 | husimon | sigh |
| 02:10.01 | drmessano | Maybe its GMT+14 |
| 02:10.28 | [TK]D-Fender | drmessano, join the PM club :) |
| 02:10.59 | husimon | [TK]D-Fender, you have any idea about the time_zone stuff for 79xx ? |
| 02:11.11 | [TK]D-Fender | husimon, no |
| 02:11.12 | husimon | k |
| 02:12.28 | drmessano | Im sure they use the time in hell as a reference |
| 02:13.10 | mikong | drmessano: But I don't know how to use it . |
| 02:13.47 | drmessano | mikong: So go figure it out |
| 02:14.42 | drmessano | man sox |
| 02:15.02 | mikong | OK, let me try . |
| 02:15.11 | drmessano | http://sox.sourceforge.net/ |
| 02:15.26 | mikong | I see . |
| 02:15.53 | drmessano | I got that from Google.. maybe you have heard of it |
| 02:15.58 | drmessano | It's the newer version of Lycos |
| 02:16.14 | drmessano | Lycos 2.0 I like to call it |
| 02:17.42 | mikong | ok , thank you . |
| 02:18.03 | drmessano | I miss Night Tracks |
| 02:18.05 | *** join/#asterisk rvhi (n=chatzill@udp197017uds.hawaiiantel.net) |
| 02:18.06 | drmessano | I want my 80s back |
| 02:18.37 | rvhi | hi, i have a polycom phone, 'show hints' shows the phone is not avaiable |
| 02:18.44 | rvhi | but it is online. |
| 02:18.51 | rvhi | what's the problem? |
| 02:19.44 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 02:23.03 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 02:23.16 | plik | husimon: try "AHST" |
| 02:25.02 | husimon | plik, I did |
| 02:25.21 | plik | oh |
| 02:25.24 | plik | shrugs |
| 02:25.44 | husimon | trying one more time with the dst_offset and dst_auto_adjust fields commented out |
| 02:25.48 | husimon | it was frigging fine |
| 02:25.55 | husimon | and then it was off by an hour |
| 02:25.57 | husimon | and I had it on HST |
| 02:26.06 | husimon | so it apparently went into DST or some shit |
| 02:26.21 | husimon | talk about the worst undocumented crap |
| 02:27.12 | plik | you can make it not change to DST, so if you can get it back how it was you should be ok |
| 02:28.00 | plik | fatiling that you could try CAT and make your phones think they're in Alaska |
| 02:30.51 | husimon | plik wtf now I take out the dst lines and ust AHST and it works |
| 02:30.56 | husimon | AGGGGGGGGHRRHRHRHRH |
| 02:31.07 | plik | yay :) |
| 02:31.12 | husimon | that makes no sense at all |
| 02:31.25 | husimon | unless it is somehow not parsing the .cnf after those lines |
| 02:31.27 | plik | cisco rarely does it seems |
| 02:31.39 | husimon | I want to frigging donkey punch them |
| 02:32.30 | plik | shame I'm not on cisco support rates though, otherwise you'd be paypal-ing me $350 for that ;) |
| 02:33.01 | husimon | the crap thing is I Tried AHST just a few minutes ago but I had dst_offset : "0" and dst_auto_adjust : "0" above it |
| 02:33.05 | husimon | and with those removed it works |
| 02:33.36 | husimon | hmm what was that command to reboot phones from asterisk |
| 02:34.01 | plik | noo idea, I missed that one |
| 02:34.09 | husimon | was some extra thing I added |
| 02:35.48 | husimon | sip notify |
| 02:37.59 | husimon | how do I just execute a command outside asterisk? |
| 02:38.05 | husimon | I thought it was asterisk -x command |
| 02:38.31 | husimon | ah it's -r -x |
| 02:40.39 | *** join/#asterisk PepOSX (n=angeldav@200.93.28.168) |
| 02:40.39 | husimon | pfft |
| 02:40.44 | husimon | you can't use sip notify as a command |
| 02:41.40 | husimon | hmm maybe I need quotes |
| 02:46.06 | fas3r | drmessano: i test this : http://southbrain.com/south/2008/02/cisco-7960-and-asterisk-1417-u.html i think it's possible it work |
| 02:46.44 | husimon | fas3r, any reason why you want to use sccp on a phone that supports sip? |
| 02:46.50 | husimon | ?????? |
| 02:47.38 | drmessano | Oh, here we go |
| 02:47.47 | drmessano | Because SIP sucks on Cisco |
| 02:47.58 | husimon | fas3r, you care to comment? |
| 02:48.31 | husimon | 99% of everyone uses sip with that phone on asterisk..... |
| 02:49.21 | drmessano | We've spent all night telling him to use SCCP and not SIP |
| 02:49.22 | husimon | oh well |
| 02:50.49 | fas3r | husimon: (it'a a 7942G not 7960) well i change because drmessano say it's better than SIP |
| 02:50.58 | [TK]D-Fender | husimon, because all he's doing is whining in here incessently hoping to find someone who'll give him the firmware since he didn't pay for a smartnet contract and can't log in to get it himself. |
| 02:51.32 | husimon | [TK]D-Fender, I figured as much |
| 02:51.39 | husimon | isn't it all of $9? |
| 02:52.04 | *** join/#asterisk sun_moon (n=RaviRaja@122.164.90.92) |
| 02:52.06 | fas3r | [TK]D-Fender: hahah nono tomorrow ( in 5h) i call my company if sccp don't work ;) |
| 02:52.16 | *** part/#asterisk sun_moon (n=RaviRaja@122.164.90.92) |
| 02:52.59 | husimon | I thought cisco gave away version 7 |
| 02:53.01 | husimon | of sip |
| 02:53.21 | fas3r | husimon: i have just find v8 |
| 02:53.40 | husimon | what? |
| 02:56.12 | *** join/#asterisk SwK (n=SwK@user-24-214-55-149.knology.net) |
| 02:57.44 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-d6cfe6aff57943fa) [NETSPLIT VICTIM] |
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| 02:58.54 | fas3r | what's the "Build PARK functions" ? |
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| 03:28.10 | LemensTS | Is phpagi like a use flag for asterisk in gentoo? |
| 03:29.22 | korihor | moy: ping |
| 03:29.56 | hsv-al | interesting |
| 03:30.01 | hsv-al | quiet for the past 1 hour practically |
| 03:35.42 | moy | korihor: pong |
| 03:36.50 | korihor | moy: sorry for delay in testing libopenr2 for Venezuela variant |
| 03:38.03 | korihor | moy: today buy two sangoma cards for test the next week with libopenr2 in testing enviroment |
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| 03:38.47 | lanning | icmp type 0 code 0 |
| 03:41.21 | moy | korihor: what are you going to use for testing? the telco or some other PBX? |
| 03:42.22 | korihor | moy: telco CANTV, MOVILNET, MOVISTAR |
| 03:44.29 | korihor | moy: CANTV is the biggest telco |
| 03:46.05 | moy | korihor: alright, let me know if you need help |
| 03:47.30 | korihor | moy: ok, thanks |
| 03:47.36 | rift0r | So why wouldn't I be able to hear ringing? I have zoiper setup as an iax softphone connecting to asterisk fine, calls can be made fine and received, but I can't hear the ringing of the line on outbound calls |
| 03:47.40 | rift0r | i even dmz'd the box |
| 03:47.42 | rift0r | still no go |
| 03:48.09 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:57.49 | rift0r | r: Generate a ringing tone for the calling party, passing no audio from the called channel(s) until one answers. Without this option, Asterisk will generate ring tones automatically where it is appropriate to do so; however, "r" will force Asterisk to generate ring tones, even if it is not appropriate. For example, if you used this option to force ringing but the line was busy the user would hear "RING RIBEEP BEEP BEEP" (thank you tzanger), which is potentiall |
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| 03:57.55 | rift0r | hmm, that may work |
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| 04:16.41 | rift0r | well that atleast does that fake ring |
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| 04:32.30 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-beta9 (2008/05/14) Asterisk 1.4.20.1, 1.2.28.1 (2008/05/21), *-Addons 1.4.6 (2008/02/21) 1.6.0-beta3 (2008/04/02), Zaptel 1.4.9.2 (2008/02/28), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox |
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| 05:08.22 | fas3r | drmessano: doesn't work |
| 05:11.47 | fas3r | well sip i comming :) |
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| 05:30.12 | philip76 | <PROTECTED> |
| 05:31.15 | Strom_M | what's the error? |
| 05:32.58 | philip76 | wait ill check my cli |
| 05:35.30 | philip76 | Storm_M: here's the error |
| 05:35.31 | philip76 | http://pastebin.com/m10027752 |
| 05:36.10 | Strom_M | <PROTECTED> |
| 05:36.38 | philip76 | ok wait i'll check that 3sec |
| 05:39.13 | philip76 | yeah your right, i thought was im having error, im just a newbie in this stuff but i love it, Thanks Storm_M. |
| 05:41.05 | maqr | can anyone recommend a good hardware sip phone? |
| 05:41.17 | NovceGuru | "good" |
| 05:41.37 | NovceGuru | define |
| 05:41.37 | Strom_M | maqr: polycom, polycom, polycom |
| 05:41.45 | Strom_M | usually, anyway :) |
| 05:41.47 | maqr | NovceGuru: one that's not made poorly? |
| 05:41.51 | maqr | Strom_M: never heard of them |
| 05:42.17 | Strom_M | maqr: www.polycom.com |
| 05:42.20 | NovceGuru | wow |
| 05:42.22 | philip76 | Storm_M: is there a way that when i call an extension instead on ringing, i can hear music on hold like music? |
| 05:43.37 | Strom_M | why the fuck does everyone think my name is Storm |
| 05:43.41 | Strom_M | philip76: yes |
| 05:43.47 | Strom_M | the "m" option to Dial() |
| 05:43.51 | NovceGuru | Strom_M: tab complete?! |
| 05:43.58 | Strom_M | NovceGuru: you'd think people would use it |
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| 05:46.06 | philip76 | Storm_M: so how can i do that? i want to use the sample music in my /var/lib/asterisk/mohmp3, where shoul i put it |
| 05:46.15 | philip76 | where can i edit that |
| 05:46.31 | Strom_M | philip76: do you have regular music on hold working? |
| 05:46.43 | maqr | Strom_M: polycom IP 330? |
| 05:46.49 | Strom_M | philip76: and for the love of god, learn to use tab complete. My name is not storm. |
| 05:46.56 | Strom_M | maqr: that's a good one |
| 05:47.50 | maqr | Strom_M: do i need a special router for power over ethernet, btw? |
| 05:48.16 | Strom_M | maqr: no, just a PoE switch or an injector |
| 05:48.27 | maqr | Strom_M: that sounds painful |
| 05:48.50 | Strom_M | ? |
| 05:48.59 | maqr | nevermind :p |
| 05:49.47 | philip76 | i dont have regular music on hold working at this time but i have sample in my pbx, i just want to know how |
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| 05:50.36 | maqr | Strom_M: do you think the linksys ones are any good? |
| 05:51.07 | Strom_M | philip76: get the sample music on hold working, and then the "m" flag to Dial() will produce music instead of audible ring tone |
| 05:51.12 | Strom_M | maqr: eh |
| 05:51.14 | Strom_M | they're ok |
| 05:51.24 | Strom_M | ive used a wide variety, and i like polycom the best |
| 05:51.44 | maqr | k, i guess i'll go with that, $130 isn't too bad |
| 05:51.47 | maqr | thanks ) |
| 05:51.48 | maqr | :) |
| 05:54.19 | philip76 | Strom_M: to make that is it in musiconhold.conf? |
| 05:54.32 | Strom_M | philip76: yes. read the documentation and the wiki |
| 05:54.43 | philip76 | Strom_M: ok thanks |
| 05:55.21 | fiddur | I actually prefer Snom 300 towards Polycom 301... not the linksys 922 I tried though... |
| 05:55.47 | Strom_M | fiddur: 301 != 330 |
| 05:56.20 | fiddur | I know... but I've just tried the halv-cheap ones :P |
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| 06:50.57 | docelm0 | Say anyone know how to enable sip re-invite in a cisco media gateway? |
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| 06:58.16 | docelm0 | Is this automatic or is there a setting somewhere? |
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| 07:16.11 | sysadmin-lb22 | hi all I need to get ilbc to work with Asterisk..I am using the latest 1.4 ...where can I get the sources ..are they still available ? |
| 07:21.11 | dudes | I think all you do is add a .c file in the codec directory to get ilbc to compile |
| 07:21.53 | dudes | probably wrong, I haven't had to think about ilbc not work so I'm not sure if it doesn't just work |
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| 08:12.23 | sohail_y | Hello everyone |
| 08:12.39 | sohail_y | can any one tell me that Unix is the best for asterisk or CentOS |
| 08:14.25 | Strom_C | sohail_y: use whatever linux you like best |
| 08:14.35 | Strom_C | it doesnt matter from an asterisk perspective |
| 08:14.50 | dominic1 | anybody used h.323? |
| 08:14.52 | sohail_y | hmmm |
| 08:14.54 | Strom_C | but it will make a difference in your ability to do all the system administration |
| 08:15.07 | sohail_y | doesnt it will be going to messup in frivers |
| 08:15.15 | sohail_y | driver* |
| 08:15.20 | tzafrir | thinks UnixWare is the best Unix for CentOS |
| 08:15.26 | Strom_C | sohail_y: huh? |
| 08:15.30 | tzafrir | ;-) |
| 08:15.35 | sohail_y | :) |
| 08:17.30 | tzafrir | sohail_y, do you have any experince with any Linuxx distribution? (or BSD, or whatever) |
| 08:17.54 | sohail_y | hmm i have been using asterisk in CentOS and RHEl4 from last 3 yrs |
| 08:17.59 | tzafrir | s/xx/x/ |
| 08:18.10 | sohail_y | but thinking to implement Unix now so looking for reviews |
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| 08:18.32 | tzafrir | that's generally fine. Though for a new installs today I'd go with 5.x |
| 08:18.45 | sohail_y | hmm me too |
| 08:19.05 | tzafrir | any special reason you use the word "Unix"? Any specific OS you have in mind? |
| 08:19.22 | sohail_y | nop |
| 08:19.27 | sohail_y | jus tfor expderiment |
| 08:19.42 | tzafrir | CentOS is as good as any |
| 08:19.47 | dominic1 | is it possible to use G.722 for internal communications and if I dial a number external it should use G.711? |
| 08:20.30 | tzafrir | dominic1, sure |
| 08:21.02 | tzafrir | Asterisk transcodes if it has to |
| 08:21.18 | sysadmin-lb22 | hi all I need to get ilbc to work with Asterisk..I am using the latest 1.4 ...where can I get the sources ..are they still available ? |
| 08:21.36 | dominic1 | what will I have to do? Cause I am using asterisk 1.4 and transcoding is not possible. How can I tell the phone or asterisk to use a specific codec for a connection? |
| 08:21.38 | tzafrir | There's a script in the sources |
| 08:21.44 | Strom_C | sysadmin-lb22: your linux distro should have ilbc packages |
| 08:21.48 | tzafrir | yells "use speex!" |
| 08:21.50 | Strom_C | just install those libraries |
| 08:22.00 | dominic1 | @sysadmin: /usr/src/asterisk-1.4.20.1/contrib/scripts/get_ilbc_source.sh |
| 08:22.02 | tzafrir | Strom_C, which distro has ilbc? |
| 08:22.12 | Strom_C | tzafrir: IIRC, debian has it |
| 08:22.20 | Strom_C | but i havent touched the codec in a while |
| 08:22.24 | Strom_C | so I may be misremembering |
| 08:22.24 | sysadmin-lb22 | Strom_C, could not found those for CentOS |
| 08:22.40 | sysadmin-lb22 | Strom_C, I read in the debian mail list that they removed it ..even in non-free |
| 08:22.44 | tzafrir | Debian and Fedora ripped ilbc from the sources of Asterisk before Digium did |
| 08:22.47 | dominic1 | under debian you must use the script under contrib too. |
| 08:22.54 | dominic1 | did that a few minutes before |
| 08:23.02 | tzafrir | for patent issues |
| 08:23.27 | sohail_y | tzafrir do you have experience of MozIAX ? |
| 08:23.31 | tzafrir | no |
| 08:23.46 | sohail_y | hmm any other Click to Dial sofphone |
| 08:24.02 | sohail_y | like skype has |
| 08:24.48 | sysadmin-lb22 | ok gout the ILBC source.. |
| 08:24.53 | sysadmin-lb22 | what about SPEEX ? same issue ? |
| 08:24.59 | tzafrir | that takes browser integration with some client. What browser do you use? |
| 08:25.13 | sohail_y | right now mozilla is my choice |
| 08:25.27 | tzafrir | (very minimal browser integration - registering a program to handle some specific types of links) |
| 08:25.38 | sohail_y | i found some click to dial plugin from info@hand but wont worked on my xten |
| 08:26.08 | tzafrir | write a script to translate the URL into something xten understands |
| 08:26.18 | tzafrir | Or use a better phone... |
| 08:26.36 | sohail_y | do you know any other fone with free services :) |
| 08:27.41 | dominic1 | Can anybody help me with my problem? Asterisk is not able to transcode G.722 without patch. I only want to use G.722 for internal phonecalls. How can I tell asterisk, that calls to external destinations sould be initiated with G.711 from phone to asterisk? |
| 08:29.17 | dominic1 | So asterisk should dynamically set the right codec. If the user dials a number starting with 0 the call should be initiated with G.711 if the user dials a number like 111 the call should be initiated with G.722 |
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| 08:44.58 | LuisTorres | Howdy |
| 08:45.17 | Strom_C | howdy howdy |
| 08:45.24 | LuisTorres | :-) |
| 08:49.20 | LuisTorres | smaybe you guys can help me.., sometimes when I execute a dial(sip/xxx&sip/xxx&ip/xxx) I got a massive decline from all phones. Any ideas that could cause this? |
| 08:49.41 | LuisTorres | Im using 1.4.17 with polycoms |
| 08:52.04 | sysadmin-lb22 | hi as mentioned above I am installaling ILBC..got the source..then make clean ./configure and now make..after that is it enought to just copy the .so file or do I have to do make install ? |
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| 08:55.38 | Vec | Hi, I have a strange problem, when a agent is in a queue and they push * it hangsup the call, if the SIP phone is added to the queue it does not hangup the phone when pushing *, no h or H options are set in the Queue application when a call comes in ? |
| 08:55.49 | Vec | Any ideas ? |
| 09:01.16 | Vec | Got to disconnect bbl |
| 09:01.44 | fas3r | how to verify if the ip phone go in tftp to load conf from the xml file ? |
| 09:07.25 | tzafrir | fas3r, check the logs of the tftp server, or sniff network traffic |
| 09:07.43 | fas3r | yes i have look i syslog |
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| 09:08.13 | fas3r | it didn't find trying to get file: .tlv file |
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| 09:26.41 | sysadmin-lb22 | ntek albi min el jibne |
| 09:26.43 | sysadmin-lb22 | woul labne |
| 09:29.53 | styelz | ÊÉnÉ ÉÉ¥Ê ÊÉÉ¥Ê |
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| 09:47.53 | _gm | is it possible to transfer/redirect a call from cli |
| 09:49.16 | Strom_C | _gm: as far as I'm aware, no. |
| 09:49.31 | _gm | hm ok |
| 09:52.44 | sysadmin-lb22 | hi All I did get the speex libraries and I did get the ilbc code using hte sh file..did make disclean. ./configure make and make install..but show translation still does not show numbers for either speex nor ilbc..and there are no .so file for them I even used ./configure --with-speex |
| 09:53.44 | Strom_C | sysadmin-lb22: is speex enabled when you make menuselect? |
| 09:54.42 | sysadmin-lb22 | Strom_C, thx for the help.. I did make all |
| 09:54.56 | sysadmin-lb22 | but I did not change the MakeFile if that is what you meant |
| 09:54.59 | Strom_C | that's not what I asked |
| 09:55.24 | sysadmin-lb22 | Strom_C, ooopps |
| 09:55.34 | sysadmin-lb22 | Strom_C, checking that utility out..did not know about it |
| 09:55.39 | sysadmin-lb22 | give me a second ..thanks..again |
| 09:56.03 | sysadmin-lb22 | Strom_C, there are 3 XXX on it in menuselect |
| 09:56.06 | *** join/#asterisk casix (n=casix@pbxedifici.adamvozip.es) |
| 09:56.08 | casix | hello |
| 09:56.41 | Strom_C | hello |
| 09:58.38 | casix | I have a trouble with a zaptel driver (zaptel-1.4.10.1) in logs files: chan_zap.c: Write returned -1 (Resource temporarily unavailable) on channel 96, any idea? is a hardware error? |
| 09:59.00 | Strom_C | casix: iz zaptel actually crashing? |
| 09:59.11 | Strom_C | s/iz/is/ |
| 09:59.30 | casix | well some calls yes and some not |
| 09:59.46 | tzafrir | casix, hmm... this is not an error condition of zaptel, IIRC |
| 09:59.46 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 10:00.58 | tzafrir | casix, what did you run there? Asterisk? |
| 10:01.27 | casix | yes asterisk 1.4.20.1 |
| 10:02.04 | atis_work | huh, anybody has idea why zaptel install fails on "make install" saying *** No rule to make target `/kernel/Makefile'. Stop. |
| 10:03.38 | sysadmin-lb22 | Strom_C, allright ilbc is compiling now but I am getting this error codec_ilbc.c:66: error: syntax error before "iLBC_Enc_Inst_t" |
| 10:03.39 | sysadmin-lb22 | codec_ilbc.c:66: warning: no semicolon at end of struct or union |
| 10:04.06 | Strom_M | sysadmin-lb22: you still havent answered my question |
| 10:04.47 | sysadmin-lb22 | Strom_M, yes I have used menu select to choose iLBC as for Speex there is XXX infront of it indicating some conflict |
| 10:05.22 | Strom_M | sysadmin-lb22: well, i dont know; perhaps you installed a crappy set of librarie |
| 10:05.31 | Strom_M | why is ilbc so important, anyway? |
| 10:05.59 | sysadmin-lb22 | Strom_M, my boss wants it ... best codec around for him..at least that is his opinion. |
| 10:06.25 | Strom_M | for what purpose? |
| 10:06.39 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 10:10.09 | Strom_M | sysadmin-lb22: hello? |
| 10:10.29 | sysadmin-lb22 | low bandwidth voip calls... |
| 10:13.32 | sysadmin-lb22 | Strom_M, I kown why the ilbc is throwing some compile error..I executed the sh for ilbc and it fetched the awk file..I even got it directly for the site to make usre it is not corrupt..when it tries to akw the file I get awk: extract-cfile.awk.1:1: BEGIN { srcname = "nothing"; } |
| 10:13.52 | sysadmin-lb22 | ' in expressionile.awk.1:1: ^ invalid char ' |
| 10:14.05 | Strom_M | congratulations |
| 10:14.10 | Strom_M | it's all gibberish to me |
| 10:14.28 | Strom_M | file a bug report |
| 10:14.31 | Strom_M | etc |
| 10:14.51 | sysadmin-lb22 | Strom_M, never did that where can I file it in ? |
| 10:15.19 | Strom_M | god forbid you should do a web search on "asterisk bug tracker" |
| 10:15.33 | Strom_M | bugs.digium.com |
| 10:17.45 | drmessano | There's a shell script in the source that facilites downloading ilbc now |
| 10:18.35 | drmessano | oh nm |
| 10:19.39 | tzafrir | atis_work, what version? |
| 10:19.44 | tzafrir | (of zaptel) |
| 10:20.01 | atis_work | tzafrir: 1.4.19.2, but it worked |
| 10:20.12 | tzafrir | that's asterisk, not zaptel |
| 10:20.33 | atis_work | tzafrir: mv /kernel /kernel-dist ; ln -s /usr/src/kernels/2.6.24.7-92.fc8-x86_64 /kernel |
| 10:20.34 | tzafrir | zaptel is 1.4.10.1 or 1.4.11 |
| 10:20.37 | atis_work | sorry 1.4.9.2 |
| 10:20.45 | atis_work | is there .10 and .11 already? |
| 10:20.59 | tzafrir | 1.4.11 released yesternight |
| 10:21.08 | atis_work | habit of typing 1.4.19 :) |
| 10:21.19 | atis_work | oh, going to try that out :) |
| 10:21.27 | atis_work | some bug in Fedora dir layout |
| 10:22.21 | tzafrir | atis_work, any bug opened for that in bugzilla.redhat.com? |
| 10:22.24 | atis_work | thanks for trying to help anyway :) |
| 10:22.29 | tzafrir | (if it is their bug) |
| 10:22.42 | atis_work | tzafrir: not really sure bug of what is it |
| 10:23.15 | patrick-- | Hey all! Im still coping with this faxerror. im using spandsp and rxfax to receive faxes. i can receive 4 faxes in a row and the fifth from the exact same number is not viewable. what could cause this? |
| 10:23.59 | Strom_M | patrick--: you only have enough fax juice for four faxes |
| 10:24.25 | patrick-- | wow, that was amazingly funny |
| 10:24.27 | patrick-- | ... not |
| 10:26.23 | *** join/#asterisk s0lid (n=s0lid@58.69.92.178) |
| 10:27.09 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
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| 10:27.10 | *** mode/#asterisk [+o file] by irc.freenode.net |
| 10:33.41 | tzafrir | Was this in protest of a bad joke? |
| 10:34.11 | Strom_M | tzafrir: I'm inclined to believe my joke was terrible enough to cause a netsplit |
| 10:42.41 | buckhill`away | LOL |
| 10:46.54 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 10:48.21 | *** join/#asterisk defswork (n=andy@mx1.3gcomms.co.uk) |
| 10:53.55 | *** join/#asterisk r0land (n=roland@193.227.191.91) |
| 10:53.58 | r0land | hello all |
| 10:54.19 | r0land | coule some1 help out with voicemail plz! as soon as i dial the extension, it immediatly opens up instead of letting it ring at the destination |
| 10:55.39 | Strom_M | r0land: pastebin the relevant section of your dialplan |
| 10:55.48 | r0land | Strom_M k just a sec |
| 10:57.36 | r0land | Strom_M http://pastebin.ca/1032992 |
| 10:58.40 | Strom_M | your pattern matching is a complete mess |
| 10:58.48 | r0land | :S |
| 10:58.52 | Strom_M | i can't even follow it |
| 10:59.29 | r0land | well i got incoming/outgoing working perfectly.. i could hear the welcoming msg and get transfered to the extension.. but the voicemail issue when i aded it! it messed everything up |
| 11:00.02 | Strom_M | i'm not surprised. your extensions.conf is like a house of cards |
| 11:00.04 | Strom_M | fix it |
| 11:00.13 | Strom_M | plan it out |
| 11:00.17 | r0land | thats why im here for .. |
| 11:00.34 | r0land | i just need to figure out how to set the voicemail.. and ill b done |
| 11:00.35 | Strom_M | clean up your extensions.conf so it's actually logical and easy to follow |
| 11:02.37 | r0land | Strom_M http://pastebin.ca/1032996 thts the best i could do |
| 11:03.44 | Strom_M | i can't even tell where you're going to voicemail |
| 11:04.30 | Strom_M | or what you're trying to do at all, really |
| 11:05.11 | Strom_M | now, of course, the pattern matching in the suipura-line context is completely wrong |
| 11:05.15 | Strom_M | you're missing the preceding _ |
| 11:05.36 | r0land | hmm k lemme figure it out for a sec |
| 11:06.36 | Strom_M | even so, your attempt at an IVR menu needs improvement |
| 11:06.49 | Strom_M | conflicting extension numbers... |
| 11:06.52 | Strom_M | etc etc |
| 11:08.51 | tzafrir | atis_work, tried a newer zaptel? |
| 11:09.18 | atis_work | tzafrir: yes, but it still searches for /kernel somehow |
| 11:10.08 | tzafrir | could you please pastebin the build log? |
| 11:10.19 | atis_work | huh, timing is great on new core2quad.. worst is 99.995 |
| 11:10.37 | atis_work | tzafrir: when i have nothing in /kernel ? |
| 11:10.49 | atis_work | i mean - when it fails? |
| 11:11.07 | tzafrir | yes (if it's still that way with recent zaptel) |
| 11:12.15 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 11:12.21 | atis_work | tzafrir: you mean console output or some file? can't find any |
| 11:12.29 | *** join/#asterisk nain (n=naain@58-27-208-74.wateen.net) |
| 11:12.45 | nain | Hi |
| 11:12.53 | atis_work | tzafrir: there's config.log |
| 11:13.38 | nain | I need to configure a SIP user with a different Authorization Username from Username; ? |
| 11:15.42 | atis_work | tzafrir: http://www.pastebin.ca/1033004 |
| 11:16.28 | nain | Same question has been asked in digium forum but no reply! http://forums.digium.com/viewtopic.php?p=70657&sid=978e49e5c234ad0a1e156ebbb2e58cf8 |
| 11:16.53 | nain | Can any one put a light on it, how to set different username then authorization username? |
| 11:17.40 | r0land | Strom_M in voicemail.conf the servermail= should it b a normail mail or for example @bla.com or wht |
| 11:18.08 | atis_work | nain: there's name and username properties for SIP peer |
| 11:18.26 | atis_work | nain: name is in [name] in sip.conf |
| 11:18.34 | atis_work | nain: try playing with that |
| 11:18.50 | atis_work | nain: not really sure if it helps, but just giving you a hint ;) |
| 11:18.53 | tzafrir | atis_work, my first guess is that $PWD has not survived 'sudo' well |
| 11:19.05 | atis_work | tzafrir: i tried also as root |
| 11:19.38 | atis_work | tzafrir: oh, true |
| 11:20.05 | tzafrir | either that, or something is wrong with "uninstall-modules"? |
| 11:20.31 | tzafrir | can you try the target "uninstall-modules" instead of "install"? |
| 11:20.37 | atis_work | tzafrir: it worked when i did "sudo su -" and then "make install" |
| 11:20.47 | atis_work | tzafrir: i'm just used to do everything as non-root |
| 11:20.56 | atis_work | tzafrir: and use sudo whenever i need something |
| 11:21.06 | atis_work | tzafrir: usually worked with installing asterisk, etc |
| 11:22.04 | tzafrir | atis_work, the makefile there has: SUBDIRS=$(PWD)/kernel |
| 11:22.10 | atis_work | tzafrir: no output, but worked |
| 11:22.25 | atis_work | tzafrir: oh, that could be cause |
| 11:22.25 | tzafrir | This is an issue with $(PWD) becoming empty for some reason |
| 11:22.51 | atis_work | tzafrir: let me check with "sudo make install" for asterisk on same machine |
| 11:23.26 | atis_work | tzafrir: well, "sudo pwd" works |
| 11:23.58 | atis_work | tzafrir: "sudo echo $PWD" too |
| 11:26.17 | tzafrir | atis_work, those don't count, as $PWD is set by the shell even before executing sudo |
| 11:26.53 | atis_work | tzafrir: asterisk install trough sudo worked too |
| 11:27.12 | atis_work | tzafrir: even sounds are installed - i found some references to $(PWD) in sounds makefile |
| 11:27.57 | tzafrir | $(PWD) is supposed to be set by make normally |
| 11:28.39 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 11:34.30 | fiddur | Hi. If the band width is too small for a call, does asterisk say that out loud in e.g. verbosity 3, or where should I look? (I just had a call with bad quality, om a sip-trunk u-law, and I'm guessing our band width was too small.. we haven't gotten our QoS-switches yet...) |
| 11:36.02 | atis_work | fiddur: no, but you can enable "rtcp stats" to see something like that in logs |
| 11:36.16 | atis_work | fiddur: or "rtcp debug" |
| 11:37.02 | fiddur | I have an ordinary debug-log switched on... is it included there as well, or do I have to enable rtcp debug explicitly? |
| 11:37.12 | atis_work | fiddur: yes |
| 11:37.33 | atis_work | fiddur: you will also have to analyze that, as there are lot of different parameters that could go wrong |
| 11:37.44 | atis_work | fiddur: but as i recall there was something called - packets lost |
| 11:37.48 | *** part/#asterisk r0land (n=roland@193.227.191.91) |
| 11:38.43 | fiddur | atis_work: ok thanks, I'll look there.... |
| 11:38.52 | atis_work | fiddur: i mean - you have to enable it explicitly |
| 11:40.26 | fiddur | atis_work: ah... can I do that in logger.conf, or just in cli? |
| 11:40.37 | atis_work | fiddur: only in cli |
| 11:40.57 | fiddur | but with rtcp debugging enabled, it will add to loggers debug-log? |
| 11:42.31 | fiddur | catching it in the cli without ability to grep or search isn't that easy... |
| 11:43.51 | fiddur | or I will have to look when I'm talking... I was hoping I could check the logs after someone said they had a bad quality call... |
| 11:45.54 | DarKnesS_WolF | any idea why the notifications LED not working on * 1.4 ? |
| 11:46.04 | DarKnesS_WolF | i did everything correctly but not working still |
| 11:46.08 | atis_work | fiddur: it should be in your full.log |
| 11:46.11 | DarKnesS_WolF | i can hint zaptel channels |
| 11:46.15 | DarKnesS_WolF | but not hint SIP channels ! |
| 11:46.43 | atis_work | fiddur: but there was some problems, ip is not registered in "stats", only in debug, so you have to match session IDs.. |
| 11:46.57 | fiddur | ah it came in verbose log, not debug log... |
| 11:47.23 | fiddur | atis_work: now I've got something to work with at least; thank you. |
| 11:47.38 | atis_work | tzafrir: seems like "sudo make install" for zaptel doesn't work anywhere. just tried on different Fedora and also on Gentoo |
| 11:47.58 | atis_work | tzafrir: do you think it's reasonable that i post it in digium bugtracker? |
| 11:48.14 | atis_work | fiddur: you're welcome :) |
| 11:48.22 | tzafrir | any idea if it worked before? |
| 11:48.51 | tzafrir | atis_work, yes, but please also report the workaround |
| 11:49.12 | atis_work | tzafrir: yes, seems that it's working on zaptel 1.4.1 |
| 11:49.31 | atis_work | tzafrir: oops, isn't |
| 11:51.12 | atis_work | oh, no - i got differenent error on 1.4.1 |
| 11:52.13 | nain | atis_work: Hi Back, I have tried different variation of username in sip.conf but no success, I am able to login to remote Server using eyebeam with username different them auth user, but not with asterisk |
| 11:53.17 | nain | atis_work: it works in eyebeam like, username=xyz, autusername=xyz@domain.net, how to stimulate it in sip.conf ? |
| 11:53.52 | atis_work | nain: is'nt authusername just something to use instead of username? |
| 11:54.05 | *** join/#asterisk stuffcorpse (n=pickle@60-234-225-122.bitstream.orcon.net.nz) |
| 11:54.08 | atis_work | nain: i'm probably not spec in this, just suggested what i thought could work |
| 11:54.59 | nain | atis_work: i have studied it to be almost same, as username but provider doesn't accept the way we send in sip.conf |
| 11:55.30 | *** join/#asterisk Tili (n=tili@58.27.154.27) |
| 11:55.45 | nain | I am facing same problem like this post: http://forums.digium.com/viewtopic.php?p=70657&sid=978e49e5c234ad0a1e156ebbb2e58cf8 |
| 11:56.06 | atis_work | nain: you want to change username/authusername in outgoing call from asterisk to provider? |
| 11:56.08 | nain | Remote Device is also IMS that work with eyebeam but not with asterisk |
| 11:56.23 | nain | atis_work: yes |
| 11:56.51 | *** join/#asterisk ZaVoid (n=zavoid@c-69-141-238-214.hsd1.nj.comcast.net) |
| 11:57.01 | atis_work | nain: do you have register=> line in sip.conf? |
| 11:57.03 | atis_work | http://www.voip-info.org/wiki-Asterisk+config+sip.conf |
| 11:57.16 | atis_work | nain: it can have authuser parameter |
| 11:57.31 | nain | atis_work; Actually I am trying to pass username string like this: username=username@domain.net |
| 11:59.11 | nain | atis_work; alternate i have found that it can be passed to provide by setting three parameter like this: username=username, fromuser=username, fromdomain=domain.net and asterisk will send it to provide like username@domain.net |
| 11:59.46 | DarKnesS_WolF | guys any idea why a sip don't send a HINT ? |
| 12:00.33 | nain | atis_work: i haven't find authuser parameter under peer or user defination ? |
| 12:01.26 | atis_work | nain: i suppose that authuser is just what asterisk matches to user, it's valid only for client, not for server |
| 12:04.13 | nain | Ok |
| 12:04.59 | nain | atis_work: What does this mean Re-invite to non-existing call leg on other UA. SIP dialog, when i send call to provider |
| 12:05.32 | atis_work | nain: i'm really not spec in all this, especially reinvites to provider :p |
| 12:05.55 | atis_work | nain: i'm avoiding reinvites all the time, all media goes trough my installations :) |
| 12:06.11 | nain | atis_work: thanks for your suggestions |
| 12:07.06 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-246-236.balt.east.verizon.net) |
| 12:07.32 | *** part/#asterisk dudes (n=james@12.192.85.19) |
| 12:09.28 | *** join/#asterisk drehlecom (i=ircbnc@unaffiliated/drehlecom) |
| 12:15.17 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:19.49 | *** join/#asterisk s0lid (n=s0lid@58.69.92.178) |
| 12:24.01 | *** join/#asterisk codehaxor (n=hellowor@222.127.118.88) |
| 12:24.03 | codehaxor | hi guys |
| 12:24.10 | codehaxor | how do i make asterisk register to openser |
| 12:24.58 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:28.25 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 12:29.11 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:29.40 | *** join/#asterisk waKKu (n=ugabuga@unaffiliated/wakku) |
| 12:31.01 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 12:33.01 | codehaxor | ? |
| 12:33.05 | styelz | http://www.spreadfirefox.com/en-US/worldrecord/ |
| 12:35.13 | *** join/#asterisk coppice (n=chatzill@106.198.17.210.dyn.pacific.net.hk) |
| 12:36.27 | *** join/#asterisk albertoandrade (n=alberto@200.195.161.164) |
| 12:41.31 | *** join/#asterisk adr3nalin3 (n=afink@asa.redglaze.com) |
| 12:44.01 | [TK]D-Fender | codehaxor: REGISTER => user:pass@sersiporfqdn/extentodial |
| 12:46.08 | codehaxor | ok |
| 12:46.27 | [TK]D-Fender | codehaxor: jsut like every other SIP server |
| 12:46.38 | codehaxor | ok |
| 12:47.42 | *** join/#asterisk jetlagmk2 (i=jetlag@pool-70-17-48-245.pskn.east.verizon.net) |
| 12:48.09 | eric2 | is using dns srv a viable way to setup for failover if you have 2 asterisk servers (primary & secondary) ? |
| 12:55.29 | *** join/#asterisk macros73 (n=cs@c-24-131-77-140.hsd1.pa.comcast.net) |
| 12:58.59 | *** join/#asterisk delparnel (n=delparne@KTNRON06-1168103470.sdsl.bell.ca) |
| 12:59.11 | Slashman | has someone already set an Asterisk to work with ldap backend where the user's password are NOT in md5 (crypt or ssha), I only saw possibilities about md5 password |
| 12:59.34 | *** join/#asterisk kannan (n=kann@123.201.60.110) |
| 13:00.30 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 13:09.02 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 13:14.29 | Katty | wonders how to find wanpipe version |
| 13:20.22 | [TK]D-Fender | Katty: "wanrouter version" |
| 13:20.36 | *** join/#asterisk hsv-al (n=ccvp@66.0.46.210) |
| 13:20.39 | *** join/#asterisk JenniferAkemi (n=akemi@76-10-181-11.dsl.teksavvy.com) |
| 13:20.44 | Katty | neat. |
| 13:20.51 | coppice | its between 1 and 4 |
| 13:21.07 | hsv-al | Hello, are we all looking forward to another long & glorious day on irc? :) |
| 13:21.18 | hsv-al | turn the crank, spin the cycle, all over again, we are in the matrix. |
| 13:21.20 | Katty | are you kidding me |
| 13:21.22 | Katty | i never leave |
| 13:22.08 | Katty | [TK]D-Fender: did i ever tell you want happened to that long drawn out choppy audio problem we had with the telco? |
| 13:22.08 | *** join/#asterisk ghento (n=ghento@CPE001cf067ec65-CM001868e2c5f4.cpe.net.cable.rogers.com) |
| 13:22.44 | [TK]D-Fender | Katty: Nope, nor did you continue on that "guess what happened" bit yesterday afternoon. |
| 13:23.02 | Katty | [TK]D-Fender: well you were supposed to guess. |
| 13:23.04 | *** join/#asterisk Vec (n=Vec@host-87-74-7-50.dslgb.com) |
| 13:23.05 | Katty | [TK]D-Fender: and you never did. |
| 13:23.25 | [TK]D-Fender | Katty: Left to go home, and when I come back the first thing you say is "get out" |
| 13:23.28 | Katty | [TK]D-Fender: but... the telco tried to blame sangoma hardware for the choppy audio problem. we had sangoma and the telco in a conference call, and the tech just let them have it. *hee* |
| 13:23.29 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 13:23.41 | Katty | [TK]D-Fender: the telco gave us two t1s |
| 13:23.44 | [TK]D-Fender | Katty: Who'd you get from Sangoma? |
| 13:23.47 | Katty | [TK]D-Fender: instead of the sing one... |
| 13:24.15 | Katty | [TK]D-Fender: Marc... |
| 13:24.29 | Katty | [TK]D-Fender: Marc Celsie (sp?) |
| 13:24.33 | hsv-al | katty, any grinding in Ahn'Qiraj |
| 13:24.34 | hsv-al | ? :) |
| 13:24.43 | Katty | doesn't ring a bell. |
| 13:24.49 | hsv-al | AQ1,2,3 etc |
| 13:24.52 | Katty | [TK]D-Fender: and i got a new 4gb flash drive. |
| 13:25.00 | Katty | hsv-al: mew? ^_- |
| 13:25.10 | hsv-al | http://www.wowwiki.com/Ahn'Qiraj |
| 13:25.16 | [TK]D-Fender | hsv-al ? |
| 13:25.25 | hsv-al | hello fender, whats going on |
| 13:25.31 | *** join/#asterisk Slashman (n=Slash@ariane.fimasys.fr) |
| 13:25.56 | [TK]D-Fender | hsv-al :same old... |
| 13:25.58 | hsv-al | katty, your addiction will be unlocked even more soon |
| 13:26.04 | hsv-al | level 80 is a new cap i hear in wow soon heh |
| 13:26.15 | Vec | Does AddQueueMember care if a agent is a member of a queue in agents.conf ? |
| 13:26.53 | Katty | hsv-al: isn't that an old pre-BC instance? |
| 13:27.18 | hsv-al | dunno, I don't play it |
| 13:27.31 | hsv-al | I never really got into mmorpgs except D2X 8 years ago |
| 13:27.49 | Katty | yeah i don' tthink anyone does that instance anymore |
| 13:28.08 | hsv-al | plus, wow's base is going to drop by 50% they predict |
| 13:28.15 | hsv-al | if Diablo3 comes out this december as some are saying. |
| 13:28.18 | hsv-al | no subscription. |
| 13:28.26 | Katty | yeah... and some other ones |
| 13:28.40 | Katty | which Ry's been talking about...but i can't remember the name |
| 13:28.41 | Katty | Warhammer |
| 13:28.45 | Katty | Age of Conan |
| 13:28.47 | Katty | something else... |
| 13:29.09 | Katty | i'll still play wow |
| 13:29.23 | hsv-al | | drmessano ³ idle / 57m 56s |
| 13:29.33 | *** join/#asterisk Slashman (n=Slash@ariane.fimasys.fr) |
| 13:29.44 | hsv-al | drmessano/d-fender i expect being active |
| 13:29.49 | hsv-al | by 8:30am cst, if not ,something is aloof |
| 13:29.51 | Katty | i haz rice crispies! |
| 13:30.21 | hsv-al | d-fender, don't tell me your just staring at the monitor waiting for a question :) |
| 13:30.29 | hsv-al | Do you have anything to talk about until a legitimate question arises? |
| 13:30.44 | Katty | hsv-al: be nice. |
| 13:30.49 | hsv-al | i am, he knows it :) |
| 13:30.50 | Katty | hsv-al: fender is a very well rounded individual. |
| 13:30.58 | *** join/#asterisk moy (n=moyhu@nat/ibm/x-a066774de22b70b8) |
| 13:31.32 | Katty | oui! |
| 13:31.53 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:32.35 | [TK]D-Fender | vec: no |
| 13:33.08 | Katty | hugs anthm |
| 13:33.17 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:33.37 | anthm | hi |
| 13:33.48 | hsv-al | anthm, you didnt play qw/dm |
| 13:33.51 | hsv-al | back in the day? :) |
| 13:34.42 | *** join/#asterisk quaqo (n=quaqo@89-97-102-198.ip17.fastwebnet.it) |
| 13:36.43 | anthm | i guess not cos i don't know what it means lol |
| 13:37.20 | hsv-al | odd |
| 13:37.28 | kannan | I am using g729 , licensed from digium. Does this mean i am not supposed to use MySQL? I just read the licensing update for why it is kept as addons |
| 13:39.14 | Katty | [May 29 08:30:14] ERROR[11876]: cdr_csv.c:313 csv_log: Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : Read-only file system <- anyone know what that means? |
| 13:39.46 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:40.13 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 13:40.50 | anthm | Katty, it means the disk where your file is has read only mode set on it |
| 13:41.01 | anthm | so it cannot write to it |
| 13:41.28 | Katty | that's the same partition that asterisk is on |
| 13:41.43 | anthm | did your server crash at all recently, some stuff puts the filesystem in read-only when it decides the disk is really messed up |
| 13:41.50 | Katty | hmm |
| 13:41.52 | Katty | well |
| 13:42.01 | Katty | my postgres database recently got screwed up royally |
| 13:42.07 | Katty | i was going to drop the database and start over... |
| 13:42.13 | [TK]D-Fender | Katty: Happened to me once. Thats what forced be to do a rebuild to CentOS5.1 |
| 13:42.15 | Katty | i don't know that it crashed tho |
| 13:42.22 | Katty | great. |
| 13:42.35 | anthm | try "mount" as root with no args |
| 13:42.50 | anthm | look for the filesystem where your stuff is |
| 13:42.58 | anthm | and note the status and attributes |
| 13:43.00 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 13:43.05 | Katty | http://pastebin.ca/1033092 |
| 13:43.20 | Katty | it's /dev/sda2 on / type ext3 (rw,errors=remount-ro) |
| 13:43.33 | anthm | yah it's messed up |
| 13:43.39 | Katty | oh. |
| 13:43.43 | Katty | that's not good. |
| 13:43.46 | anthm | how much stuff you have on it? |
| 13:43.57 | Katty | not a whole lot... |
| 13:44.01 | Katty | i can backup the csv files.. |
| 13:44.06 | anthm | back it up |
| 13:44.10 | Katty | and backup a few conf files.. and revert to a g4u clone i did a month ago |
| 13:44.11 | anthm | then you can run fsck on it |
| 13:44.14 | Katty | and just dump the new stuff on there |
| 13:45.08 | Katty | do you think fsck will fix it? |
| 13:45.12 | anthm | probably just rebooting it will trick linux into doing it for you |
| 13:45.18 | Katty | hrmm |
| 13:45.23 | Katty | but if it kernel panics on the way up |
| 13:45.24 | anthm | either fix it or erase most of it =p |
| 13:45.26 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:45.26 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 13:45.30 | Katty | yeah |
| 13:45.32 | Katty | heh |
| 13:45.33 | [TK]D-Fender | Katty: back up what you can now... |
| 13:45.36 | Katty | k |
| 13:45.41 | [TK]D-Fender | Katty: Mine never woke up on reboot... |
| 13:46.05 | anthm | if its a prod box |
| 13:46.17 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 13:46.20 | anthm | i'd install OS again on a new hd and never use that one again |
| 13:46.45 | Katty | it's a brand new hard drive.. |
| 13:46.53 | Katty | the machine was put into place a little over a month ago >.< |
| 13:47.01 | anthm | brand new is the most likely time for a hd to die |
| 13:47.33 | anthm | just saying what i would do, you don't have to but I am paranoid that way with prod stuff |
| 13:47.41 | anthm | if a box looks at me wrong i never use it again |
| 13:47.57 | Katty | backing this stuff up is going to be a pain. i can't copy anything into /var/www |
| 13:48.15 | anthm | but if you install os on a new disk you can mount the messed up one |
| 13:48.21 | anthm | still ro |
| 13:48.25 | anthm | and cp everything |
| 13:48.29 | Katty | i've got a flash drive, but i don't know how to mount it. maybe gnome will just see it. heh |
| 13:50.18 | glaz | gnome does not "see" drives. |
| 13:50.38 | Katty | well x just crashed. heh |
| 13:50.44 | anthm | anyway you may want to consider the data on that drive "in jeopardy" =D |
| 13:50.54 | Katty | yeah |
| 13:51.02 | Katty | any thoughts on mounting an NTFS flash drive? |
| 13:51.04 | anthm | the os can't run very well on a ro disk |
| 13:51.17 | Katty | all i need to get is a couple files |
| 13:51.36 | Katty | let me guess, i can't mount it |
| 13:51.42 | anthm | can you move them over the network? |
| 13:51.57 | Katty | i don't have the csv directory shared out |
| 13:52.04 | Katty | or the /etc/asterisk directory |
| 13:52.10 | Katty | and i don't think i can update smb.conf |
| 13:52.13 | Katty | or samba.conf |
| 13:52.19 | [TK]D-Fender | Katty: SCP from another box. |
| 13:52.23 | Katty | [TK]D-Fender: SCP? |
| 13:52.27 | jaytee | what about sftp using Filezilla? |
| 13:52.33 | *** join/#asterisk AfricAnne (n=AfricAnn@41.204.130.161) |
| 13:52.38 | [TK]D-Fender | Katty: like FTP over SSH |
| 13:52.50 | Katty | i don't know how to do that. |
| 13:53.04 | jaytee | Katty, download FIlezilla and use port 22 |
| 13:53.09 | Katty | yeah i have it... |
| 13:53.22 | [TK]D-Fender | Katty: http://winscp.net/eng/index.php |
| 13:55.42 | *** part/#asterisk AfricAnne (n=AfricAnn@41.204.130.161) |
| 13:56.11 | Katty | [TK]D-Fender: i'm getting my stuff. thanks |
| 13:56.32 | [TK]D-Fender | Katty: warning : be sure to use BINARY mode. |
| 13:56.46 | Katty | oh? |
| 13:56.47 | [TK]D-Fender | Katty: Otherwise your text files will come over with Windows CRLF instead of CR |
| 13:57.07 | [TK]D-Fender | Katty: You'll end up doing a LOT of dos2unix to recover them |
| 13:57.22 | Katty | ok |
| 13:57.28 | Katty | anything else i need to make sure of? |
| 13:59.36 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:03.03 | *** join/#asterisk spokra (n=spokra@host093-179-178.sea0.speakeasy.net) |
| 14:03.45 | [TK]D-Fender | Katty: Thats about it. |
| 14:03.56 | [TK]D-Fender | Katty: I recovered jsut about everything important off of mine. |
| 14:04.14 | [TK]D-Fender | I have to make a backup scipt for my current setup.... |
| 14:04.42 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 14:04.55 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 14:12.58 | *** join/#asterisk ming_zym (n=ming_zym@123.103.29.231) |
| 14:17.38 | Katty | [TK]D-Fender: yeah i got everything but a handful of unimportant voicemails. |
| 14:17.53 | [TK]D-Fender | Katty: Didn't feel like grabbing those? |
| 14:21.30 | *** join/#asterisk _Krieger_ (n=krieger@193.39.118.158) |
| 14:23.07 | *** join/#asterisk killmel8tr (n=IceChat7@ip-198-22-65-131.quickconnectusa.com) |
| 14:24.08 | *** join/#asterisk ghento (n=ghento@CPE001cf067ec65-CM001868e2c5f4.cpe.net.cable.rogers.com) |
| 14:26.33 | ix33 | is there a way to automatically cause one particular agent to be logged into the queue on system startup? |
| 14:26.45 | ix33 | *an incoming call queue, that is |
| 14:26.56 | seanbright | ix33: besides making them static? |
| 14:27.23 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 14:29.00 | seanbright | ix33: if you are using 1.6, you can use cli.conf |
| 14:29.34 | seanbright | ix33: otherwise you can define the agent statically, or put /usr/sbin/asterisk -rx 'queue add member blah blah' in rc.local |
| 14:29.53 | Katty | [TK]D-Fender: it bombed out |
| 14:30.05 | Katty | [TK]D-Fender: do you know why my ethernet card is getting a new eth0 number everytime i reboot? |
| 14:30.10 | Katty | [TK]D-Fender: it's up to eth5 now |
| 14:30.11 | [TK]D-Fender | Katty: well since you say you got the rest I guess it'll do |
| 14:30.19 | [TK]D-Fender | Katty: no ida. |
| 14:30.21 | Katty | [TK]D-Fender: yeah, everyone has voicemails on email anyway. |
| 14:30.40 | tzafrir | well, until I can think of something better, it's now http://docs.tzafrir.org.il/ |
| 14:30.41 | spokra | any suggestions for a voip provider allows asterisk and has unlimited calling |
| 14:30.56 | ix33 | seanbright: where do you define an agent statically? is this agents.conf? |
| 14:31.06 | seanbright | ix33: queues.conf |
| 14:31.24 | seanbright | ix33: member => Agent/foo |
| 14:31.32 | seanbright | ix33: member => SIP/whatever |
| 14:31.33 | seanbright | etc |
| 14:31.49 | ix33 | hmmm. don't know why i haven't done that yet... |
| 14:31.51 | ix33 | let's see |
| 14:32.04 | seanbright | there are examples in the queues.conf.sample file |
| 14:32.46 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:32.48 | jaytee | tzafir, that's awesome! bookmarked it. |
| 14:33.16 | ix33 | well "persistentmembers=yes" doesn't seem to be doing what i thought it would either |
| 14:34.23 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 14:35.32 | seanbright | ix33: what did you think it would do? |
| 14:35.54 | ix33 | persist logged-in agents through a * restart |
| 14:36.11 | *** join/#asterisk mort___ (n=mort@user-54445d40.lns3-c10.dsl.pol.co.uk) |
| 14:36.17 | ix33 | does that have to be set per-queue, or in [general]? |
| 14:36.29 | ix33 | (can it be in [general]) |
| 14:36.40 | seanbright | ix33: it can be in general, yes. |
| 14:37.02 | seanbright | ix33: in fact |
| 14:37.05 | ix33 | well ok it's not persisting through a restart... |
| 14:37.08 | seanbright | ix33: it *has* to be in general |
| 14:38.38 | seanbright | won't work at the [queue] level |
| 14:38.48 | seanbright | or the [foo] level |
| 14:38.54 | seanbright | if you catch my drift. |
| 14:39.58 | ix33 | check |
| 14:40.08 | ix33 | ok statically saying SIP/100 seems to work fine |
| 14:40.12 | ix33 | in queues.conf |
| 14:42.36 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584377.dsl.bell.ca) |
| 14:42.43 | seanbright | cool |
| 14:42.46 | *** join/#asterisk oserr (n=oserr@c-b774e255.016-134-73746f32.cust.bredbandsbolaget.se) |
| 14:43.25 | *** join/#asterisk mihinomenest (n=argh@cerebus.clandestineresearch.com) |
| 14:46.27 | jbeez | hallo |
| 14:49.37 | ix33 | the fact that dynamic agents are automatically logged off at night is now a 'feature.' :V |
| 14:52.10 | seanbright | ix33: what version of asterisk? |
| 14:52.45 | [TK]D-Fender | ix33: "member =>" are not dynamic... |
| 14:52.53 | *** join/#asterisk adr3nalin3 (n=afink@asa.redglaze.com) |
| 14:53.09 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 14:53.09 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:53.27 | hsv-al | uh oh, end of the world |
| 14:53.59 | seanbright | ix33: is it possible to pastebin your queues.conf? |
| 14:54.00 | seanbright | ~pb |
| 14:54.01 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 14:59.59 | *** join/#asterisk stmaher (n=stmaher@mail.voxpilot.com) |
| 15:00.03 | stmaher | Hello everyone.. |
| 15:00.27 | stmaher | is there a way in asterisk to identity if a call is Video or Audio when it arrives? |
| 15:00.57 | stmaher | eg.. if I want to use the same inbound DNIS but route it to different IVRs/sipphones depending if the call is plain audio or video? |
| 15:01.08 | stmaher | Thank joo |
| 15:02.38 | *** join/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
| 15:06.04 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:06.04 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:08.43 | TJNII | ~book |
| 15:08.44 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 15:09.27 | *** join/#asterisk francogwapito (n=chatzill@125.252.90.5) |
| 15:10.21 | francogwapito | has anyone here tried using the followme cmd? |
| 15:11.21 | francogwapito | do i need to include => followme on extensions.conf? |
| 15:13.14 | Katty | [TK]D-Fender: hm. that was fun. |
| 15:13.41 | *** join/#asterisk seanmh (i=seanmh@216.31.101.25) |
| 15:16.37 | [TK]D-Fender | francogwapito: Where do you see any mention of a context that ming need to be included? |
| 15:16.42 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 15:17.11 | [TK]D-Fender | might* |
| 15:17.12 | francogwapito | in voip-info |
| 15:17.20 | francogwapito | tried that though, that didnt work |
| 15:17.34 | *** part/#asterisk RobH (n=RobH@cpe-72-184-129-136.tampabay.res.rr.com) |
| 15:17.49 | [TK]D-Fender | francogwapito: link the page please. |
| 15:18.36 | francogwapito | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+followme |
| 15:19.10 | francogwapito | i'm getting this error No such extension/context XXXX@default creating local channel |
| 15:19.31 | francogwapito | Unable to allocate a channel for Local/XXXX@default cause: Unknown |
| 15:19.35 | francogwapito | any idea? |
| 15:19.55 | jblack | francogwapito: Sounds to me like you don't have a default context in your extensions.conf, or at least don't have extension XXXX in it |
| 15:20.06 | [TK]D-Fender | francogwapito: Yes, you clearly don't have an exten to match what is being dialing in the [default] caontext. |
| 15:21.12 | francogwapito | in my extensions.conf, the context i used is from-internal then in followme.conf i set [XXXXX] context => from-internal |
| 15:22.06 | [TK]D-Fender | francogwapito: Looks like you missed one somewhere. pastebin your configs. |
| 15:22.07 | [TK]D-Fender | ~pb |
| 15:22.08 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:22.09 | francogwapito | i dont understand why asterisk is calling default context when i defined from-internal context |
| 15:22.09 | [TK]D-Fender | ^^^^^^^^^^^ |
| 15:22.14 | francogwapito | ok |
| 15:22.28 | patrick-- | hey [TK]D-Fender :) i know know why my faxscript didnt work as it should |
| 15:22.44 | [TK]D-Fender | patrick--: Why is that? |
| 15:22.49 | patrick-- | the script checks for files every minute and as soon as it finds a file, it sends it, regardless of the process of the fax transfer. and if the transfer is still in progress the file arrives chopped up |
| 15:22.55 | *** join/#asterisk ghento (n=ghento@CPE001cf067ec65-CM001868e2c5f4.cpe.net.cable.rogers.com) |
| 15:23.11 | patrick-- | can i create files with some dialplan app? sort of pid files? |
| 15:23.23 | [TK]D-Fender | patrick--: System, etc |
| 15:24.36 | *** join/#asterisk funxion (n=x@63.214.236.169) |
| 15:24.40 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 15:25.52 | funxion | can anyone think of an easy way to report zap channel utilization over a month period, possibly using mysql cdr? |
| 15:26.35 | francogwapito | D-Fender: http://pastebin.com/m1b1784a1 here.. |
| 15:27.22 | *** join/#asterisk ManxPower (n=manxpowe@39.sub-75-201-150.myvzw.com) |
| 15:28.40 | kannan | funxion - phpmyadmin? |
| 15:28.54 | funxion | kannan what about it? |
| 15:29.16 | kannan | well , use it to query the mysql cdrs |
| 15:30.00 | funxion | I can query the cdr through the mysqlclient, just not sure of the query that would show me how many zap channels are being used at any given moment in history |
| 15:30.27 | *** join/#asterisk Asterisk- (n=a@ACB0EF17.ipt.aol.com) |
| 15:30.31 | Asterisk- | Hello |
| 15:30.57 | russellb | O.O |
| 15:31.08 | russellb | who are you to get to be named "Asterisk-" ? |
| 15:31.10 | russellb | :-p |
| 15:31.17 | Asterisk- | :)) |
| 15:31.21 | Asterisk- | it's a free country |
| 15:31.28 | Asterisk- | i cand be whoever i want |
| 15:31.29 | funxion | lol |
| 15:31.29 | Asterisk- | i think |
| 15:31.30 | Asterisk- | :P |
| 15:31.37 | Asterisk- | lol |
| 15:31.55 | kannan | its a nice nick |
| 15:32.10 | Asterisk- | Anyone in here uses vonage business sip trunks? |
| 15:32.33 | kannan | ~vonage |
| 15:32.34 | jbot | from memory, vonage is a bunch of monkeys |
| 15:32.44 | Asterisk- | lol |
| 15:32.45 | [TK]D-Fender | francogwapito: Ok, the formatting of the sample configs and the docs out there are all very inconsistent. I can't tell what "proper" would look like. |
| 15:33.17 | Asterisk- | does anyone know why it allows me only 200 concurrent calls? :( |
| 15:33.21 | Asterisk- | they say unlimited |
| 15:33.30 | Qwell | Asterisk-: call them and yell at them |
| 15:33.33 | [TK]D-Fender | Asterisk-: Really? Link it please. |
| 15:34.13 | francogwapito | D-Fender: hmm.. :( |
| 15:34.51 | francogwapito | D-Fender: have you read my configs? |
| 15:35.23 | [TK]D-Fender | francogwapito: Yes, and its the samples alone I can't comprehend. They don't really make sense and a re poorly explained. |
| 15:36.10 | Asterisk- | any other alternative to vonage? with unlimited concurrent calls? |
| 15:36.34 | Asterisk- | the price doesn't matter |
| 15:36.45 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 15:36.56 | Qwell | Asterisk-: If price doesn't matter, use Qwell Communications. |
| 15:37.06 | Qwell | only $10.95/minute, per channel |
| 15:37.45 | Asterisk- | and the max number of channels is? |
| 15:37.48 | *** join/#asterisk draygon (n=draygon@gateway5-pnap.exigo.com) |
| 15:37.52 | Qwell | unlimited |
| 15:38.20 | Asterisk- | website? :P |
| 15:39.09 | [TK]D-Fender | Asterisk-: call Level3 direct. |
| 15:39.10 | denon | Asterisk-: you know, by saying "price doesn't matter", we know you're using a fradulant credit card, and running a scam |
| 15:39.18 | Qwell | denon: quite |
| 15:39.26 | Qwell | Asterisk-: Please leave |
| 15:39.33 | Asterisk- | are you kidding right? |
| 15:39.53 | denon | no, I'm not kidding -- our largest wholesale sip customers still don't say "price doesn |
| 15:39.55 | denon | 't matter" |
| 15:39.59 | denon | in fact, it matters more |
| 15:40.00 | [TK]D-Fender | Asterisk-: No, I'm sure their assessment is entirely logical. |
| 15:40.35 | Asterisk- | just that one has the money to buy everything doesn't matter that he is a criminal |
| 15:40.37 | Asterisk- | damn |
| 15:40.55 | [TK]D-Fender | Asterisk-: like I said, call Level3 directly. |
| 15:41.29 | Asterisk- | anyway i will live just for your rudeness |
| 15:41.31 | Asterisk- | :) |
| 15:41.41 | Asterisk- | i'll search help in another place |
| 15:41.43 | Asterisk- | take care |
| 15:41.52 | *** part/#asterisk Asterisk- (n=a@ACB0EF17.ipt.aol.com) |
| 15:41.53 | [TK]D-Fender | Asterisk-: http://www.level3.com/services/index.html |
| 15:42.01 | [TK]D-Fender | Schmuck. |
| 15:42.08 | gr0mit | what was all that about? |
| 15:42.33 | Qwell | gr0mit: a moron thinking he can ask us for recommendations on companies to defraud |
| 15:42.40 | [TK]D-Fender | gr0mit: Just read the last dozen lines. |
| 15:42.56 | Yourname` | From Pakistan, I'm sure. |
| 15:45.27 | [TK]D-Fender | Worse yet..... AOL user in Germany : Asterisk- (n=a@ACB0EF17.ipt.aol.com) |
| 15:45.37 | gr0mit | aaaah ok |
| 15:45.41 | Qwell | [TK]D-Fender: germany? |
| 15:45.47 | [TK]D-Fender | Qwell: apparently. |
| 15:45.57 | [TK]D-Fender | Qwell: Unless he's tor'd, etc |
| 15:46.20 | gr0mit | in other words, give me lots of mins paid for by some poor sod |
| 15:46.26 | hsv-al | heh |
| 15:46.51 | hsv-al | just noticed this: |
| 15:46.54 | hsv-al | [10:31am] [Asterisk-] i cand be whoever i want |
| 15:47.00 | Qwell | heh |
| 15:47.06 | [TK]D-Fender | Irony |
| 15:47.27 | *** join/#asterisk dwayne (n=dwayne@76.29.245.9) |
| 15:49.39 | hsv-al | love it starts out |
| 15:49.41 | hsv-al | [10:30am] .-- Join: (Asterisk-) (n=a@ACB0EF17.ipt.aol.com) |
| 15:49.41 | hsv-al | [10:30am] [Asterisk-] Hello |
| 15:49.41 | hsv-al | [10:30am] [@russellb] O.O |
| 15:49.41 | hsv-al | [10:31am] [@russellb] who are you to get to be named "Asterisk-" ? |
| 15:49.49 | hsv-al | how |
| 15:50.04 | patrick-- | why wont this work: exten => _58028799,n,System(touch /var/spool/asterisk/fax/fax.pid) |
| 15:50.07 | patrick-- | ? |
| 15:50.08 | patrick-- | do i need quotes? |
| 15:50.28 | Qwell | patrick--: that should work fine.. does that last fax/ dir exist? |
| 15:50.51 | patrick-- | yes |
| 15:50.56 | patrick-- | it does |
| 15:51.01 | patrick-- | it doesnt create any file though |
| 15:51.01 | Qwell | have permissions to it? |
| 15:51.13 | patrick-- | yes asterisk also writes incoming faxes to it |
| 15:51.24 | funxion | does anyone know a query for assterisk mysql cdr that would give me zaptel channel utilization by hour? |
| 15:52.00 | funxion | I need to justify more digium cards |
| 15:52.07 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:52.30 | patrick-- | Qwell: http://phpfi.com/320534 |
| 15:52.35 | patrick-- | thats the entire process |
| 15:52.42 | patrick-- | ah |
| 15:52.44 | patrick-- | i know why |
| 15:52.53 | patrick-- | i should write the pid before i write the fax file |
| 15:53.06 | patrick-- | mhh |
| 15:53.08 | patrick-- | i do though |
| 15:53.17 | patrick-- | i only set a var above writing the pid |
| 15:53.33 | ManxPower | patrick--: why are you doing the touch and the rm? |
| 15:53.39 | ManxPower | It's not locking anything |
| 15:54.10 | patrick-- | ManxPower: cause i have my own fax delivery script thats running, and it should not process faxes until the receiving of the fax is completed. |
| 15:54.16 | outtolunc | i am wondering why he is using the _ on the exten |
| 15:54.28 | patrick-- | outtolunc: you can talk to me directly, you know.. |
| 15:54.29 | ManxPower | patrick--: Why not run the script in the dialplan? |
| 15:54.58 | patrick-- | i want to try get it working like this first. |
| 15:55.06 | patrick-- | after that i can worry about beauty corrections |
| 15:55.14 | ManxPower | patrick--: it's usually easier to start with less complications, not more. |
| 15:55.30 | ManxPower | patrick--: I would expect the file to not be removed based on your design. |
| 15:55.33 | patrick-- | well the learning process is much higher if you start complicated ;) |
| 15:55.44 | patrick-- | mhh, why is that? |
| 15:55.50 | ManxPower | patrick--: exactly. So do it more simple -- in the dialplan |
| 15:56.04 | patrick-- | ManxPower: do you know why its not writing the pid file? |
| 15:56.05 | ManxPower | patrick--: um, because when the far end hangs up, it will stop processing the dialplan. |
| 15:56.26 | patrick-- | mhh ok |
| 15:56.36 | ManxPower | Which is why we send fax calls to a separate context and use exten => h to postprocess |
| 15:56.50 | ManxPower | patrick--: put the cli output on pastebin.ca |
| 15:58.42 | ManxPower | patrick--: comment out the line that rm's the file |
| 15:59.50 | patrick-- | http://phpfi.com/320536 |
| 16:00.00 | patrick-- | its not removing the file as you thought |
| 16:00.10 | ManxPower | patrick--: good thing I don't need to edit any of that text for you |
| 16:00.31 | ManxPower | patrick--: OK, so it is not creating the file or is it not removing the file? |
| 16:00.36 | patrick-- | last |
| 16:00.55 | ManxPower | patrick--: OK, now paste that info again -- this time without debug. |
| 16:01.10 | patrick-- | sec |
| 16:01.15 | ManxPower | patrick--: I would expect what is happening, as when the far end hangs up the dialplan will stop processing. |
| 16:01.32 | ManxPower | patrick--: don't bother. I just told you what the problem is and how I solved it. |
| 16:01.42 | patrick-- | k |
| 16:02.17 | ManxPower | You told us 5 mins ago that the file was not being created. |
| 16:02.30 | ManxPower | now you are saying the problem is that it is not being deleted. |
| 16:02.32 | ManxPower | pick one. |
| 16:03.18 | patrick-- | ManxPower: forget the "not creating" part. i was beeing stupid. its not beeing removed. |
| 16:03.34 | ManxPower | patrick--: well there is 5 mins of my life I'll never get back. |
| 16:03.45 | patrick-- | my bad :P |
| 16:04.01 | patrick-- | mhh never used h extension before |
| 16:04.37 | ManxPower | patrick--: Use a Goto to send the call to a separate context, do all your stuff in that context |
| 16:04.49 | patrick-- | mhh |
| 16:04.52 | ManxPower | then you can safely use the "h" extension without problems |
| 16:05.01 | patrick-- | goto on hangup.. how do i do that? |
| 16:05.03 | *** join/#asterisk kamh (n=q@host-81-190-236-85.wroclaw.mm.pl) |
| 16:05.11 | ManxPower | NO, that is not what I said. |
| 16:05.22 | patrick-- | where do i use the goto then? |
| 16:05.28 | kamh | hi |
| 16:05.44 | kamh | does someone know something about "asterisk's QoS"? |
| 16:05.52 | patrick-- | if its in a low priority in the faxextension it will not be processed after the far end hung up |
| 16:05.57 | russellb | asteirsk is not in charge of QoS ... |
| 16:05.57 | patrick-- | -low +high |
| 16:06.05 | ManxPower | REMOVE: exten => _58028799,1,Answer() and everything after it, ADD: exten => _58028799,1,Goto(fax,${EXTEN},1) |
| 16:06.24 | patrick-- | ah, shift the whole process to another context.. |
| 16:06.32 | ManxPower | That is what I said. |
| 16:06.42 | patrick-- | sry |
| 16:06.45 | ManxPower | patrick--: you may not understand why yet, but you will. |
| 16:07.09 | patrick-- | i do |
| 16:07.23 | patrick-- | cause otherwise all hanged up calls with be processed to h ext |
| 16:07.29 | patrick-- | if ill do it within the old context |
| 16:07.52 | kamh | because when I registered two IPphones to the asterisk (connected to the same network by hub) the sound was not so good, but when I registerd them to some VoIP provide sound was perfect |
| 16:07.58 | kamh | it was strange to me |
| 16:08.20 | ManxPower | patrick--: from a working system: http://pastebin.ca/1033195 |
| 16:08.23 | patrick-- | ManxPower: I assume "fax" is the target extension in your example, right? |
| 16:08.38 | patrick-- | erm |
| 16:08.41 | patrick-- | target context* |
| 16:08.42 | ManxPower | no, fax is the target CONTEXT |
| 16:09.09 | ManxPower | patrick--: maybe you should grab a cup of coffee and go for a walk. You are too frustrated to accomplish anything. |
| 16:09.56 | ManxPower | In my pastebin, I used the context [in-fax] in the example I gave you on the channel uses [fax] as the context. |
| 16:10.06 | patrick-- | got that part |
| 16:10.49 | patrick-- | so in that case the h extension will always be processed? if the far end hangs up too early or just in time? |
| 16:11.14 | ManxPower | correct. |
| 16:11.25 | ManxPower | there is no real way to tell if the far end hung up early |
| 16:11.30 | patrick-- | thanks. ive stuck the removing process into that ext. |
| 16:11.35 | patrick-- | lets try again |
| 16:20.33 | patrick-- | okay got that bit all worked out. now its all in my delivery script |
| 16:21.33 | ManxPower | patrick--: You want a copy of my script |
| 16:21.36 | ManxPower | ? |
| 16:21.44 | patrick-- | no thanks |
| 16:21.48 | patrick-- | you well at bash? |
| 16:21.59 | ManxPower | It's written in perl |
| 16:21.59 | patrick-- | maybe you could look at mine and tell me where im wrong |
| 16:22.12 | ManxPower | patrick--: no. I don't debug other people's code. |
| 16:22.56 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 16:23.29 | ManxPower | For anyone that wants it: http://www.fnords.org/~eric/fax2email.txt |
| 16:23.38 | Assid | hi . i just tried to install asterisk on a VPS running within a OpenVZ environment on ubuntu. however after installing asterisk (With apt) i get the following error: http://assid.pastebin.com/d396d75e6 |
| 16:24.23 | Assid | someone told me i should be able to run asterisk within a vps environment |
| 16:24.37 | ManxPower | Assid: is Asterisk started as root? |
| 16:24.45 | ManxPower | Assid: Which someone told you that? |
| 16:25.01 | ManxPower | Also, we don't support Asterisk packages. Install from source. |
| 16:25.33 | Assid | personally i dont think it will work cause another error is "Failed to read hardware config: pcilib: Cannot open /proc/bus/pci" |
| 16:25.38 | tzafrir | asterisk from debs will not start as root |
| 16:25.54 | ManxPower | tzafrir: So it can't enable realtime priority? |
| 16:26.04 | tzafrir | -U asterisk |
| 16:26.29 | ManxPower | tzafrir: Right, that tells asterisk to SWITCH to that user, right? Not START as that user? |
| 16:26.32 | Assid | hrmm interesting.. asterisk starts tho |
| 16:26.44 | Assid | i wonder if MoH will work |
| 16:26.45 | ManxPower | Assid: Not interesting at all |
| 16:26.46 | tzafrir | ManxPower, right |
| 16:27.00 | ManxPower | tzafrir: So my question stands, is he starting Asterisk as root? |
| 16:27.38 | Assid | ManxPower: its within a vps.. and well openvz doesnt let you do custom kernels.. means shouldnt load zaptel and ztdummy |
| 16:27.40 | tzafrir | Assid, which module needs that? |
| 16:27.46 | tzafrir | vpb? |
| 16:27.47 | Assid | ManxPower: yeah its running as asterisk |
| 16:28.04 | ManxPower | Assid: As far as I can tell that has nothing to do with your error |
| 16:28.11 | ManxPower | Assid: START it as root |
| 16:28.17 | Assid | ok ok |
| 16:28.32 | ManxPower | I don't really give a flying weasel what it's running as, you need to START it as root. |
| 16:28.36 | tzafrir | the permission error is also to root? |
| 16:28.42 | ManxPower | Asterisk will switch to it's configured error. |
| 16:28.57 | ManxPower | tzafrir: only root can give a process realtime priority. |
| 16:29.00 | tzafrir | Assid, disable chan_vpb |
| 16:29.53 | tzafrir | Assid, what version of Ubuntu? |
| 16:30.08 | tzafrir | (some of us do support Asterisk from packages) |
| 16:30.22 | Assid | tzafrir: hardy - 8.04 |
| 16:30.30 | ManxPower | tzafrir: That's just because you built the package 8-0 |
| 16:31.50 | Assid | i havemt really played with the updated version of asterisk .. last i think was in early 1.4 if im not mistaken. |
| 16:32.21 | Assid | is zaptel still required for MoH and meetme ? |
| 16:32.39 | ManxPower | Assid: zaptel was never required for MoH |
| 16:32.57 | Assid | err..hrmm okay .. meetme then |
| 16:33.11 | ManxPower | yes, you need zaptel timer for Meetme |
| 16:33.11 | Qwell | Assid: yes |
| 16:33.51 | Assid | zaptel : /etc/zaptel.conf fails test for exists and readable |
| 16:34.23 | Assid | thought as much.. there is no way to load a module |
| 16:35.00 | Assid | although i wonder if the host node loads a module , will the VE get access to it |
| 16:35.04 | ManxPower | Assid: you are running in a VM, what did you expect? |
| 16:35.08 | tzafrir | touch /etc/zaptel.conf |
| 16:35.21 | Kobaz | man, cleaning is exhausting |
| 16:35.47 | hardwire | account add xmpp spencer_shane@216.67.99.230 *xfnr58a |
| 16:35.51 | hardwire | account list |
| 16:35.56 | hardwire | hi |
| 16:35.59 | hardwire | wrong uh.. |
| 16:36.01 | hardwire | ducks |
| 16:36.18 | Assid | tzafrir: great there is no lsmod |
| 16:36.46 | tzafrir | no lsmod? become root properly |
| 16:36.50 | Assid | i am root |
| 16:36.59 | Assid | tzafrir: openvz virtual environment |
| 16:37.07 | tzafrir | properly, and your PATH will be set properly |
| 16:37.20 | *** part/#asterisk paul0 (n=paulo@201-2-255-30.fnsce701.dsl.brasiltelecom.net.br) |
| 16:37.54 | *** part/#asterisk dwayne (n=dwayne@76.29.245.9) |
| 16:39.19 | ManxPower | Assid: "su -l root" |
| 16:39.25 | ManxPower | the -l sets the paths |
| 16:39.42 | Assid | -su: lsmod: command not found |
| 16:39.51 | Assid | im telling you its something to do with the virtual environment |
| 16:39.59 | Assid | or the way its setup |
| 16:40.17 | ManxPower | Assid: What about /sbin/lsmod |
| 16:40.35 | ManxPower | Assid: then maybe you need to go get your VM set up properly |
| 16:40.41 | Assid | no such file or directory |
| 16:40.52 | ManxPower | Assid: Then leave and go fix your VM |
| 16:50.45 | Qwell | O.o |
| 16:51.18 | Qwell | silly freenode bots |
| 16:55.02 | *** mode/#asterisk [-r] by Qwell |
| 16:55.05 | *** join/#asterisk dikdust (n=dikdust@77.43.42.95) |
| 16:55.30 | Qwell | somebody remind me to reset that mode later, after services come back up? |
| 16:57.58 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:02.08 | *** join/#asterisk [1]hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 17:03.01 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 17:04.02 | ManxPower | Qwell: remember to reset the channel mode after services comes back up. 8) |
| 17:05.08 | Katty | and remember to hug Manx! |
| 17:05.10 | Katty | hugs ManxPower |
| 17:05.13 | Qwell | ~kick ManxPower |
| 17:05.14 | jbot | ACTION kicks ManxPower |
| 17:06.32 | *** join/#asterisk ming_zym (n=ming_zym@123.103.29.231) |
| 17:07.44 | Katty | jbot_: hug qwell |
| 17:07.51 | Katty | ~hug qwell |
| 17:07.51 | jbot | ACTION sneaks up on qwell and suddenly hugs qwell tightly |
| 17:07.52 | Qwell | he knows better |
| 17:07.59 | Katty | mhmm. |
| 17:08.00 | Katty | right. |
| 17:08.02 | Katty | hugs Qwell too |
| 17:08.06 | Qwell | ~kick jbot_ |
| 17:08.07 | jbot | ACTION kicks jbot_ |
| 17:08.10 | Katty | offers lettuces |
| 17:08.12 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 17:10.53 | Katty | file: lettuces. |
| 17:10.55 | Katty | jbot_: lettuces? |
| 17:11.14 | Qwell | can I have a lattice? |
| 17:11.34 | Katty | no lattices for you! |
| 17:11.40 | Katty | your morning glories will climb up the HOUSE |
| 17:11.43 | file | moo |
| 17:11.52 | ManxPower | hands Qwell some pieces of green lattice |
| 17:11.53 | Katty | feeds file lettuces |
| 17:11.57 | TJNII | notes that the conversation is almost always bizzare when Katty is in |
| 17:12.15 | Katty | want to know the rest? |
| 17:12.17 | Katty | buy the rights! |
| 17:12.20 | Katty | how bizzare |
| 17:12.40 | TJNII | Oh damnit, not I'm going to have that song stuck in my head! |
| 17:12.41 | file | tickles Katty |
| 17:12.44 | [TK]D-Fender | Katty: the backing track is a ripoff of "Spanish Harlem" |
| 17:12.57 | Qwell | TJNII: You did this to yourself, you know |
| 17:13.30 | Katty | TJNII: http://youtube.com/watch?v=J6tS6FsY4Ms |
| 17:13.32 | Katty | nice. |
| 17:13.45 | Katty | that troll is at least lvl 38, he's wearing herod shoulders from SM |
| 17:13.53 | Katty | and they're standing outside RFD instance in southern barrens |
| 17:14.13 | Katty | oh wait, he has a mount. at least lvl 40 |
| 17:14.53 | Katty | [TK]D-Fender: is it? i never noticed. |
| 17:15.41 | Katty | [TK]D-Fender: also, i REALLY like that program you gave me that does ftp over ssh |
| 17:15.43 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 17:15.48 | Katty | [TK]D-Fender: you're my hero today. |
| 17:15.54 | *** join/#asterisk RobH (n=RobH@rob.tech.wikimedia.org) |
| 17:16.03 | [TK]D-Fender | Katty: Glad to help. |
| 17:16.38 | sp00kz | scp wins |
| 17:16.49 | *** join/#asterisk fedya (n=fedya@75.112.143.226) |
| 17:16.58 | patrick-- | *sigh* |
| 17:16.59 | patrick-- | done :) |
| 17:17.09 | Katty | horays! time to take it out of oven |
| 17:17.12 | Katty | what did you make me? |
| 17:17.21 | patrick-- | a nice turkey |
| 17:17.28 | patrick-- | *hands over hot plate* |
| 17:17.51 | *** join/#asterisk timothy_jones (n=chatzill@207.90.4.189) |
| 17:18.12 | vector | hey, anyone seen my pet turkey? |
| 17:18.17 | patrick-- | *whistles* |
| 17:18.23 | fedya | i'm trying to build my first asterisk box from scratch, i was looking at visual dialplan program, but i don't want to spend the $170 quite yet, i installed the sample conf files, where would i start? should i create some extensions first then some providers then deal with ivr and such? or are there better sample conf files i could use? |
| 17:18.37 | Qwell | ~book |
| 17:18.37 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 17:18.41 | Qwell | fedya: start there :) |
| 17:19.00 | fedya | i have one |
| 17:19.06 | vector | patrick--, oh no.. not AGAIN!.. 3rd time this week |
| 17:19.07 | fedya | oh man i forgot to bring it to the office :0 |
| 17:19.12 | Qwell | fedya: pdf |
| 17:19.22 | patrick-- | *looks away* |
| 17:19.40 | fedya | is there a quickstart guide in there? |
| 17:19.53 | [TK]D-Fender | fedya: for some inspriation : |
| 17:19.55 | [TK]D-Fender | ~jerjerguide |
| 17:19.56 | jbot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 17:20.14 | fedya | that sounds better |
| 17:20.20 | patrick-- | ... |
| 17:20.24 | [TK]D-Fender | fedya: read that for a sample of a simple setup, and get cracking on the book and go try stuff |
| 17:20.28 | patrick-- | wrong attitude |
| 17:20.47 | Qwell | patrick--: attitude? |
| 17:20.48 | fedya | perfect, thanks |
| 17:20.57 | fedya | just needed somewhere to start |
| 17:21.16 | patrick-- | yes Qwell |
| 17:21.23 | patrick-- | "19:20 < fedya> that sounds better" |
| 17:21.29 | patrick-- | thats how i started.. didnt get me far |
| 17:21.36 | patrick-- | you need to understand how things work |
| 17:22.22 | fedya | yea i've made some tweaks before and walked through dialplans, but have no understanding of the basic concepts as of yet |
| 17:22.38 | patrick-- | then its best to start with the book |
| 17:22.45 | patrick-- | it takes you right through it |
| 17:22.49 | Qwell | there's no "best" way to learn :) |
| 17:22.55 | [TK]D-Fender | fedya: Chapter 5 is your friend/ |
| 17:23.06 | patrick-- | shuts up |
| 17:23.16 | fedya | i'll read this article first and continue when i have the book in front of me tonight, i can't read pdf of books |
| 17:23.28 | fedya | [TK]D-Fender: thx |
| 17:23.50 | [TK]D-Fender | Qwell: Sure there is but I think he'd have a cerebral hemorrhage if we attempt a direct telepathic transfer :p |
| 17:24.09 | patrick-- | mindmelt? :O :O |
| 17:24.14 | patrick-- | mr spock ?!?!?! |
| 17:24.29 | Qwell | hands [TK]D-Fender a hammer |
| 17:24.49 | *** join/#asterisk mihinomenest (n=argh@cerebus.clandestineresearch.com) |
| 17:26.25 | [TK]D-Fender | terminates patrick-- with his Vulcan Death-Grip |
| 17:30.40 | Katty | gosh. |
| 17:30.49 | Katty | i'm gone for 10 minutes, and fender whips out star trek lore |
| 17:30.56 | patrick-- | :) |
| 17:31.29 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:33.07 | sp00kz | Qwell, please change the channel mode back |
| 17:33.23 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:33.27 | Qwell | in a few minutes, after people get a chance to re-ident |
| 17:33.30 | Qwell | thanks though |
| 17:33.48 | sp00kz | ;] |
| 17:33.52 | patrick-- | assimilates [TK]D-Fender |
| 17:33.57 | *** part/#asterisk lirakis_work (n=lirakis@65.200.191.241) |
| 17:33.58 | *** join/#asterisk hubguruJR (n=hubguruJ@mail.ntegratedsolutions.com) |
| 17:34.03 | *** join/#asterisk rene- (n=renemend@200.34.66.137) |
| 17:34.09 | russellb | oh wtf |
| 17:34.14 | Katty | ohai russellb |
| 17:34.19 | russellb | nickserv doesn't accept ident as an alias for identify anymore |
| 17:34.21 | russellb | that's silly |
| 17:34.29 | Katty | :< |
| 17:34.31 | Qwell | tell christel! |
| 17:34.36 | russellb | too lazy |
| 17:34.42 | Qwell | tsk, tsk |
| 17:34.50 | sp00kz | not too lazy to bitch at us about it though :P |
| 17:34.51 | hubguruJR | Hi All |
| 17:35.07 | Qwell | sp00kz: it's a fine line |
| 17:35.09 | Katty | bitching takes no effort |
| 17:35.09 | russellb | sp00kz: zing |
| 17:35.10 | Katty | i would know. |
| 17:35.16 | patrick-- | lol |
| 17:35.16 | Katty | i'm a pro at bitching |
| 17:35.20 | patrick-- | youre female |
| 17:35.24 | patrick-- | thats what makes u pro |
| 17:35.27 | Katty | exactly. |
| 17:35.41 | *** mode/#asterisk [+r] by Qwell |
| 17:35.56 | Katty | yay |
| 17:36.08 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 17:36.16 | Katty | so what's the super duper software phone these days? i'm still using zoiper |
| 17:36.21 | rene- | hello, i am trying to make sense on the way unique ids are managed and or generated by asterisk, i am seeing that in outbound calls with an A party and a B party, B's uniqueid would be X.X+1 for a given A uniqueid |
| 17:36.28 | Qwell | Katty: asterisk with chan_oss's video support |
| 17:36.34 | Katty | ooh |
| 17:36.37 | Katty | video support now eh |
| 17:36.47 | Katty | well ain't that just fancy schmancy |
| 17:39.40 | rene- | in calls that came in from a queue that has local channels as members you have 4 uniqueids involved, and then you would have base unique id of X.X and then X+11.X+1, third uniqueid would be X.X+1 and so would forth |
| 17:39.52 | jjshoe | freenode is such a 'noisey' network |
| 17:39.58 | rene- | i meant fourth |
| 17:40.04 | Qwell | 'open' |
| 17:40.28 | Qwell | jjshoe: I don't see wallops when people are about to hose efnet :p |
| 17:40.31 | rene- | is there any logic to it? i am trying to make sense of AMI.status output |
| 17:40.44 | Qwell | would rather know about it, than see a 3-hour long netsplit |
| 17:42.02 | *** part/#asterisk Assid (n=assid@unaffiliated/assid) |
| 17:42.08 | *** join/#asterisk ghento (n=ghento@CPE001cf067ec65-CM001868e2c5f4.cpe.net.cable.rogers.com) |
| 17:43.50 | rene- | umm |
| 17:43.56 | *** join/#asterisk adr3nalin3 (n=afink@asa.redglaze.com) |
| 17:46.37 | atis_work | rene-: each call invovles 2 channels |
| 17:47.16 | *** join/#asterisk Deeewayne (n=Deeewayn@216.207.245.1) |
| 17:47.16 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 17:49.37 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 17:51.28 | jjshoe | Qwell I don't need to know about it. my client puts me on the larger piece of the net *shrug* |
| 17:51.35 | jjshoe | some say tomato... |
| 17:52.12 | *** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 17:59.31 | sp00kz | other says I pity the fool. |
| 18:00.50 | jjshoe | sp00kz lol |
| 18:00.52 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 18:01.18 | sp00kz | :p |
| 18:06.07 | funxion | does anyone know a query for assterisk mysql cdr that would give me zaptel channel utilization by hour? |
| 18:06.58 | *** join/#asterisk tobias (n=tobias@c-67-165-106-220.hsd1.pa.comcast.net) |
| 18:10.31 | *** join/#asterisk steliosk (n=Stelios@79.131.123.31) |
| 18:11.16 | seanbright | well you would need a select |
| 18:11.19 | seanbright | a like |
| 18:11.20 | seanbright | and a group by |
| 18:11.23 | seanbright | err |
| 18:11.27 | seanbright | not a like. a left. |
| 18:11.50 | seanbright | goes to write it |
| 18:14.23 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 18:14.35 | Qwell | right outer cross join |
| 18:17.06 | *** join/#asterisk vgster (n=vgster@93.96.221.240) |
| 18:17.23 | seanbright | ~pb |
| 18:17.24 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 18:18.16 | seanbright | funxion: this does it for sql server. you should be able to modify it a bit to work with mysql --> http://pastebin.ca/1033302 |
| 18:19.50 | *** join/#asterisk xacatecas (n=jkroon@dsl-240-130-76.telkomadsl.co.za) |
| 18:20.44 | xacatecas | hi guys, is there any way to do an GotoIf() based on whether a certain number ${CALLERID(num)} matches a certain pattern such as 1XX ? I'm already pretty convinced that there is ... I just don't know how. |
| 18:24.24 | xacatecas | hmm, i can probably use the exten => _dst/_src,... type pattern ... |
| 18:27.16 | *** join/#asterisk [Akemi] (n=akemi@76-10-181-11.dsl.teksavvy.com) |
| 18:31.58 | Katty | are crontab entries seperated by spaces or tabs? |
| 18:32.23 | styelz | both will work |
| 18:32.24 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 18:33.15 | rob0 | "Entries?" Separated by newline. |
| 18:34.27 | rob0 | If you mean "elements", whitespace, space[s] or tab[s]. |
| 18:34.28 | styelz | i presumed the bits between the numbers.. not the actual enrties... |
| 18:34.35 | *** join/#asterisk [hC] (n=hardcore@S0106001346a4b813.vc.shawcable.net) |
| 18:34.38 | styelz | as return is obvious |
| 18:34.50 | rob0 | yes you probably guessed right :) |
| 18:35.41 | Katty | yes |
| 18:35.55 | Katty | between the 30 8 1 * * |
| 18:36.06 | Katty | 8:30, every weekday, ever month |
| 18:36.09 | [TK]D-Fender | xacatecas: as a pattern no, but you cal compare the length and starting digit |
| 18:37.13 | xacatecas | [TK]D-Fender, i just thought of another approach that should work as well, just run through a context with _./_pat,Goto(...) with a _. extension at the end. probably not the cleanest but it just might work. |
| 18:38.05 | [TK]D-Fender | not sure if you can CID pattern match like that, but if you can, ok. |
| 18:39.41 | xacatecas | well, it may be patched, but i found lines like these in trixbox (which is what i'm trying to move away from): s/_2XX,1,Set(__FROM_DID=${EXTEN}) |
| 18:39.51 | xacatecas | so I'm going to give it a shot. |
| 18:44.48 | fedya | would i be better off starting from scratch than using the sample conf files that were installed? |
| 18:47.10 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 18:50.27 | *** join/#asterisk kannan (n=kann@123.201.60.116) |
| 18:53.23 | Strom_M | fedya: use sample conf files as guidelines, but you're better off writing them from scratch |
| 18:53.33 | fedya | thx |
| 18:55.39 | *** join/#asterisk RobH (n=RobH@cpe-72-184-129-136.tampabay.res.rr.com) |
| 18:56.09 | funxion | does anyone know a query for assterisk mysql cdr that would give me zaptel channel utilization by hour? |
| 18:57.27 | Strom_M | funxion: a mysql query alone isnt going to tell you that |
| 18:57.41 | x86 | it could |
| 18:57.48 | x86 | depends on how badass your mysql skills are |
| 18:58.07 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:58.26 | Katty | i've got the pgsql database, csv log files, the monitor, voicemails, /etc/asterisk stuff |
| 18:58.29 | Katty | what else shoudl i backup |
| 18:58.37 | funxion | Also, I have a weird problem my * box connects to a Nortel Meridian option 81c viz cisco 5400 PRI. When calling out from notel ext 4000 the proper callerid shows up, but if I create an extension 4000 on * and mak the call through the nortelno callerid shows up. Whats makes this weird is that this problem only occures in the local area code, but all calls are going out to the same carrier via the same trunk group |
| 18:58.49 | funxion | x86 exactly |
| 18:58.56 | funxion | thats what I was looking for |
| 18:59.31 | funxion | I was hoping that someone has already spent the time thinking up the query and woulndt mind sharing |
| 19:00.13 | *** join/#asterisk iratik (n=itariki@12-226-116-3.client.mchsi.com) |
| 19:00.25 | iratik | This sphinx stuff looks pretty difficult... even the quick start is 3 pages |
| 19:01.14 | funxion | anyone have a clue about the callerid issue? |
| 19:01.23 | funxion | that one's got me bent |
| 19:05.24 | xacatecas | funxion, once you've got the cdr's in mysql you can get data out of it with sql queries ... so yea. shouldn't be too hard. |
| 19:05.34 | xacatecas | you want incoming utilization or outbound? |
| 19:05.38 | xacatecas | or a combination? |
| 19:06.04 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 19:06.25 | seanbright | funxion: i gave you a SQL Server query to base your mysql one off of. i'm guessing you don't know sql? |
| 19:08.27 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
| 19:10.13 | *** join/#asterisk r0land (n=df@194.146.154.54) |
| 19:10.47 | r0land | hello all |
| 19:11.13 | r0land | could some1 help me in setting up voice mail to send an email with a wav attachment to each user tht gets a voice msg ! |
| 19:12.05 | [TK]D-Fender | r0land: go read the sample config, its all in there. |
| 19:12.05 | Strom_M | r0land: set up your mail transfer agent, then put the email address and "attach=yes" in the voicemail box's configuration entry in voicemail.conf |
| 19:12.28 | xacatecas | funxion, select channel, sum(duration) from cdr where channel LIKE 'ZAP/%' group by channel; (incoming time), and select dstchannel, sum(billsec) from cdr where dstchannel LIKE 'ZAP/%' group by dstchannel; (outbound time). |
| 19:12.38 | Qwell | Strom_M: transport* |
| 19:12.42 | xacatecas | that gives it in seconds, just divide by 3600 to get hours. |
| 19:12.53 | Strom_M | Qwell: details details ;) |
| 19:13.53 | Qwell | wikipedia is wrong. stupid wikipedia |
| 19:14.11 | Katty | can i cp -r /etc/* without anything bombing? |
| 19:14.21 | Katty | or is there anything i need to turn /off/ like in windows |
| 19:14.41 | Qwell | Katty: doing exactly that command would result in the last argument being the last file, so every file would be (individually) copied to a random file in /etc/ |
| 19:14.41 | r0land | Strom_M i just hacing prb with servermail= ?? do i set it as @hotmail for exampe! or i give it a specific mail! or an smtp server! or wht.. |
| 19:14.49 | Qwell | since * expands to everything in that dir |
| 19:14.53 | Katty | Qwell: well i'm going to put it there |
| 19:14.57 | Katty | not |
| 19:14.59 | Katty | going to put it there |
| 19:15.03 | Katty | it's going to /backup/stuff/date/thing |
| 19:15.03 | Qwell | put what where? |
| 19:15.07 | Qwell | I see |
| 19:15.10 | Qwell | then yeah |
| 19:15.13 | Katty | k |
| 19:15.19 | r0land | Strom_M i could leave a msg now, but it just gets saved in /...../tmp/extension... and doesnt do anything with it |
| 19:16.05 | r0land | Strom_M i wanna b able to call my voicemail.. enter my pass! and listen to my msg.. AND get it as a wav attachment to my mail |
| 19:16.19 | Strom_M | r0land: I already told you how to do it |
| 19:16.53 | r0land | Strom_M mail transfer agent as in.. ?! i installed mailx.. |
| 19:17.16 | Qwell | like exim, postfix, sendmail |
| 19:18.22 | *** join/#asterisk StevenBrady (i=d04e65ca@libervis/StevenBrady) |
| 19:18.41 | *** join/#asterisk docelm0 (n=chatzill@206.248.239.194) |
| 19:18.50 | docelm0 | Say anyone in here know anything about rxfax? |
| 19:19.12 | Strom_M | "anyone in here know anything about rxfax" |
| 19:19.15 | Strom_M | there, i said it |
| 19:19.18 | Strom_M | do I win a prize? |
| 19:19.25 | Qwell | ~say anyone in here know anything about rxfax? |
| 19:19.25 | jbot | anyone in here know anything about rxfax |
| 19:19.47 | docelm0 | Strom_M well if you do yes.. Im having issues with it crashing |
| 19:19.57 | Deeewayne | anyone ~in ~here know anything about rxfax |
| 19:20.06 | Deeewayne | hmmmm.... |
| 19:20.16 | docelm0 | channel.c keeps telling me: channel.c:3059 set_format: Unable to find a codec translation path from ulaw to unknown |
| 19:20.21 | Qwell | ~in |
| 19:20.21 | jbot | somebody said in was India |
| 19:20.30 | docelm0 | now I have looked at the SDP and only see ulaw.. So Im baffled |
| 19:20.47 | StevenBrady | I'm interested in ordering some equipment and setting up a new phone system at my office, but my manager wants me to check the compatibility of our lines with Asterisk. We have multiple PRI lines coming in for our phone lines, and apparently (as I'm not versed in this area) the CID is handled via DNIS outpulse. Is Asterisk compatible with DNIS outpulse? Are there configuration options I'm supposed to be aware of? |
| 19:21.15 | docelm0 | StevenBrady: dude.. you will be fine.. |
| 19:21.15 | Strom_M | StevenBrady: asterisk can be configured to work with that, yes |
| 19:21.27 | docelm0 | I use asterisk in my office w/ Cisco 3660's as my media gateways |
| 19:21.42 | Strom_M | StevenBrady: your PRI can also be reconfigured to work in a more normal ISDN-like way |
| 19:21.58 | StevenBrady | Is there extra configuration required? Or is that auto configured when I install the card? |
| 19:22.05 | StevenBrady | A more normal ISDN-like way? |
| 19:22.26 | Strom_M | StevenBrady: DNIS outpulse doesn't sound like PRI; it sounds like channelized T1 |
| 19:22.37 | Strom_M | are you sure your circuit is ISDN PRI? |
| 19:22.46 | hsv-al | sbrady, you have seen the light, dont be brainwashed by those dumbass |
| 19:22.49 | hsv-al | nortel commercials |
| 19:22.56 | hsv-al | more $$$ then ccm :) |
| 19:23.09 | Katty | i want to see how many files total are in a directory |
| 19:23.14 | Katty | and what the size of it is |
| 19:23.20 | Katty | right now i'm doing du -ha |
| 19:23.26 | StevenBrady | I was told it was PRI. But the vendor sent me my site surveys. I can look more deeply into it. |
| 19:23.27 | Katty | but i can't find a tag that shows me how many files |
| 19:23.39 | StevenBrady | I'm not super knowledgable on these types of connections. |
| 19:23.44 | *** join/#asterisk mike-ekim (n=mike@adsl-072-151-207-108.sip.mia.bellsouth.net) |
| 19:23.55 | r0land | Qwell ive installed sendmail, and agent is started... |
| 19:24.15 | Strom_M | StevenBrady: regardless of whether it's regular T1 or ISDN, asterisk can likely be configured to work with it |
| 19:24.15 | *** join/#asterisk styelz (n=yoohoo@2001:388:f000:0:0:0:0:20b) |
| 19:24.43 | seanbright | Katty: just regular files? |
| 19:24.50 | seanbright | Katty: not counting sockets/named pipes/etc? |
| 19:25.00 | Katty | yeah just basically conf, txt, etc |
| 19:25.20 | seanbright | Katty: find -type f | wc |
| 19:25.34 | StevenBrady | I'm calling the provider for verification. |
| 19:26.07 | StevenBrady | And of course I get voicemail. |
| 19:26.36 | ManxPower | Once a carrier has you locked into a contract they stop answering calls |
| 19:26.43 | StevenBrady | haha |
| 19:26.52 | Katty | seanbright: how di i give it a directory? |
| 19:26.58 | seanbright | first param |
| 19:27.03 | seanbright | Katty: find /etc -type f | wc |
| 19:27.10 | Katty | and what's this wc thingy? |
| 19:27.16 | Qwell | word count |
| 19:27.27 | Strom_M | water closet |
| 19:27.27 | seanbright | 'man wc' |
| 19:27.28 | seanbright | :) |
| 19:27.36 | StevenBrady | Thanks for the quick reply all. |
| 19:27.39 | StevenBrady | I appreciate it. |
| 19:29.05 | Katty | oh ah. |
| 19:29.13 | seanbright | Katty: the first (of 3) numbers listed is the number of files |
| 19:29.30 | Katty | and the last is the number of characters? |
| 19:29.34 | seanbright | aye |
| 19:29.38 | Katty | is this recursivez? |
| 19:29.43 | seanbright | find is, yes. |
| 19:29.47 | Katty | k |
| 19:29.55 | seanbright | (by default, anyway) |
| 19:30.04 | Katty | neat. |
| 19:30.09 | Katty | tank you berry much |
| 19:30.15 | seanbright | welcome |
| 19:31.52 | *** join/#asterisk grEvenX (n=even@pc107-102.ktv.no) |
| 19:34.09 | *** join/#asterisk Zougloub (n=cJ@zougloub.eu) |
| 19:40.08 | *** part/#asterisk StevenBrady (i=d04e65ca@libervis/StevenBrady) |
| 19:42.46 | *** join/#asterisk mintee (n=mintone@72-165-177-94.dia.static.qwest.net) |
| 19:43.17 | mintee | Can I query the DB some other way then Set()ing it to a variable? |
| 19:43.30 | seanbright | mintee: you can do it from the cli |
| 19:43.34 | seanbright | mintee: database show |
| 19:43.35 | mintee | I tried $[key/value] |
| 19:43.48 | mintee | seanbright, yeah, i mean from an extension |
| 19:44.05 | seanbright | mintee: ${DBGet(Key/Value)} |
| 19:44.18 | ManxPower | mintee: there is no point in doing that |
| 19:44.24 | *** join/#asterisk grEvenX (n=even@pc107-102.ktv.no) |
| 19:44.33 | seanbright | mintee: that was a lie anyway |
| 19:44.35 | ManxPower | But if you really want to, just wrap it in a noop |
| 19:44.37 | mintee | DBGet is deprecated |
| 19:44.43 | mintee | I tried that ManxPower. |
| 19:44.49 | ManxPower | mintee: then you did something wrong |
| 19:44.54 | seanbright | mintee: NoOp(${DB(Key/Value)}) |
| 19:45.17 | *** join/#asterisk marc7 (n=marc@S0106001c10243803.gv.shawcable.net) |
| 19:45.33 | mintee | I just wanna Authenticate(DB(${THISNUM}/pass)) |
| 19:45.41 | xacatecas | which decent soft-phones exists for linux? |
| 19:45.47 | xacatecas | wengo is just a pain. |
| 19:45.59 | seanbright | mintee: Authenticate(${DB(${THISNUM}/pass)}) |
| 19:46.03 | *** join/#asterisk techie (n=techie@adsl-76-214-30-46.dsl.lsan03.sbcglobal.net) |
| 19:46.16 | ManxPower | mintee: It will work right if you don't have syntax errors. |
| 19:46.18 | *** join/#asterisk rcahilig (i=ca4e4bf5@gateway/web/ajax/mibbit.com/x-5189a97bb0feb031) |
| 19:46.33 | seanbright | mintee: Authenticate(${DB(${THISNUM}/pass)}) <--------------------------------<< |
| 19:46.51 | seanbright | mintee: >>----> Authenticate(${DB(${THISNUM}/pass)}) <----<< |
| 19:46.54 | ManxPower | mintee: what about when there is no pass for THISNUM? |
| 19:47.01 | seanbright | mintee: >>----> Authenticate(${DB(${THISNUM}/pass)}) <----<< |
| 19:47.09 | Katty | passes out |
| 19:47.27 | [TK]D-Fender | mintee: Go read up on how to reference functions & variables again. You clearly missed the boat. |
| 19:47.32 | marc7 | simple order-of-operations question with http://pastebin.ca/1033389 ... If a SIP user is part of "Context-internal" and dials 0, who gets called? user1 or user2? |
| 19:48.00 | ManxPower | marc7: duplicate extensions should generate an error |
| 19:48.08 | mintee | ah, seanbright, hum... yeah, it works now.. must have missed a { or ( something |
| 19:48.10 | mintee | thanks |
| 19:48.13 | marc7 | ManxPower: it doesn't... it calls user1 when I'm expecting it to dial user2 |
| 19:48.14 | seanbright | mintee: no problem |
| 19:48.21 | mike-ekim | i installed asterisk, and when i try to issue sip reload, or sip show peers, it says that command doesnt exist... ?? |
| 19:48.28 | RobH | I am honestly not sure where to begin on this issue. My asterisk server is emailing out voicemail notifications with the file attached. Occasionally, rather than sending out the email correctly, it sends out a MIME email that is of course all jumbled and unreadable. |
| 19:48.31 | ManxPower | marc7: no, it should call user1 |
| 19:48.35 | Strom_M | marc7: yes,user1 should be called |
| 19:48.36 | ManxPower | first match gets it |
| 19:48.48 | [TK]D-Fender | mike-ekim: Running a softphone on your server as well? |
| 19:48.51 | mintee | ManxPower, there will always be a pass for THISNUM |
| 19:48.56 | mintee | I'll be setting it manually |
| 19:49.08 | Strom_M | first it matches within the current context, then it checks the included contexts in the order they were included |
| 19:49.10 | marc7 | crap. how do I override that context so that dialing "0" from the outside (eg: Context) reaches the receptionist, but if the *receptionist* dials 0 (eg: Context-internal), it reaches our technical support? |
| 19:49.25 | Strom_M | marc7: that's a dumb idea |
| 19:49.31 | seanbright | mike-ekim: make sure /etc/asterisk/modules.conf contains either a) autoload = yes or b) load => chan_sip.so |
| 19:49.33 | marc7 | Strom_M: not my idea ;) |
| 19:49.41 | seanbright | mike-ekim: also make sure that /etc/asterisk/sip.conf exists |
| 19:49.43 | Strom_M | marc7: it's still a dumb idea |
| 19:49.46 | ManxPower | marc7: put users in a differnet context than the operator |
| 19:49.51 | ManxPower | well receptionist |
| 19:49.54 | mike-ekim | will do! |
| 19:50.31 | marc7 | ManxPower: again, there's that separation between Context and Context-internal..... part of what I'm trying to avoid is tons of duplication (eg: I don't want to repeat extensions 101, 102, 103, 104.... etc in a public/private context |
| 19:50.34 | RobH | Exim is getting a single email from asterisk and sending it out, its not splitting it into a multi-part message, asterisk is... |
| 19:50.40 | RobH | just no clue where to fix it. |
| 19:50.45 | ManxPower | marc7: you don't need to. |
| 19:50.46 | mike-ekim | No softphone |
| 19:50.54 | mike-ekim | autoreload = yes is set |
| 19:50.55 | marc7 | i'd just like the the second include to take precedence |
| 19:51.01 | mike-ekim | but not load => chan_sip.wo |
| 19:51.04 | mike-ekim | so |
| 19:51.06 | seanbright | mike-ekim: autoreload? |
| 19:51.10 | marc7 | ManxPower: how so? |
| 19:51.11 | ManxPower | marc7: You put the devices in a context that only include =>'s extensions. |
| 19:51.11 | Strom_M | marc7: if you want it to take precedence, include it first |
| 19:51.27 | ManxPower | marc7: give me 10 mins and I'll give you a pastebin example |
| 19:51.31 | Strom_M | marc7: or set up your device contexts in a sane and sensible way |
| 19:51.40 | Strom_M | yeah, listen to ManxPower :) |
| 19:51.41 | marc7 | no need Manx, I understand what you're suggesting. |
| 19:51.50 | seanbright | mike-ekim: you mean autoload? |
| 19:52.36 | marc7 | at least, I understand you're suggesting [Company-External], [Company-Internal], and [Company-Devices]... and both external/internal include the devices context |
| 19:52.57 | rcahilig | Hi, I created a macro in my Dialplan but its not working? Please help me, here is my dialplan http://pastebin.ca/1033395 |
| 19:53.23 | mike-ekim | hm, load => chan_sip.so didnt load anything on startup |
| 19:53.46 | *** join/#asterisk shido6 (n=shido6@209.114.208.192) |
| 19:53.47 | seanbright | mike-ekim: you said "autoreload" is set |
| 19:53.50 | *** join/#asterisk rch1 (i=devnull@c-68-81-28-33.hsd1.pa.comcast.net) |
| 19:53.51 | seanbright | mike-ekim: there is no "autoreload" |
| 19:54.11 | ManxPower | marc7: http://pastebin.ca/1033397 |
| 19:54.19 | *** join/#asterisk jsidhu2 (n=atomik@mail2.simplybhangra.com) |
| 19:54.30 | rch1 | I did a stupid thing and deleted astdb in /var/lib/asterisk - with asterisk still running and the DB still in memory is there any way I can restore it? |
| 19:55.00 | seanbright | rch1: restart asterisk :) |
| 19:55.14 | jsidhu2 | what is the best way to have two phones register to one extension? Should I just create two seperate extensions and then make a ring group to ring both extensions? |
| 19:55.25 | marc7 | manx: gotcha. i agree that's slightly cleaner. i'll try that approach. thank you. |
| 19:55.30 | rch1 | kinda can't right now, would bring our callcenter offline - if it do a put will it re-write the db? |
| 19:55.35 | marc7 | (sometimes it's best to suggest not cutting corners) |
| 19:55.38 | rch1 | erm, sorry, if I do a put :) |
| 19:55.58 | Strom_M | jsidhu2: create two separate accounts and then have one extension dial both |
| 19:55.59 | seanbright | jsidhu2: have them register as two different devices (SIP/Phone1 and SIP/Phone2) |
| 19:56.03 | deeperror | jsidhu2: would create 2 extensions and dial them both if you want them to both ring |
| 19:56.03 | ManxPower | marc7: contexts are one of the most important things to learn, they are also one of the hardest things to learn |
| 19:56.09 | seanbright | jsidhu2: then do a Dial(SIP/Phone1&SIP/Phone2) |
| 19:57.01 | jsidhu2 | ok, let me play around.. thanks |
| 19:57.33 | seanbright | rch1: i don't know. i doubt it would re-write the entire db file. |
| 19:57.47 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:58.05 | sp00kz | jsidhu2: what I do is use queues |
| 19:58.11 | seanbright | ughhh |
| 19:58.20 | seanbright | o.v.e.r.k.i.l.l. |
| 19:58.21 | seanbright | heh |
| 19:58.47 | sp00kz | maybe for just two phones, but for 10+ and maintainablity its so much better :P |
| 19:58.50 | jsidhu2 | sp00kz: then what do you do about voicemail? can the voicemail box be the same extension as the queue #? if not then I have to train users to remember two numbers, one for extension and one for voicemail |
| 19:59.22 | [TK]D-Fender | jsidhu2: its whatever you want it to be. Its your server, and your job. |
| 19:59.25 | seanbright | jsidhu2: sure, they can have the same extension for VM purposes |
| 19:59.41 | seanbright | jsidhu2: but if you really only have 2 phones you want to work with, queue is a bit much |
| 19:59.50 | seanbright | (imho) |
| 20:00.10 | seanbright | which is extra h because its parenthesized |
| 20:01.00 | jsidhu2 | i'll play around with ur suggestion, i only have 2 devices to ring |
| 20:01.06 | [TK]D-Fender | Dial <----- |
| 20:01.13 | seanbright | ^^^^ |
| 20:01.59 | jsidhu2 | all im trying to do is get to use a decent headset.. i cant any good wireless headset to work with my aastra480i.. so i got a usb bluetooth headset and using it with x-lite.. so i need to ring both devices.. softphone + aastra 480i.. (anyone recommend a better solution?) |
| 20:02.00 | mike-ekim | seanbright: i meant auto load |
| 20:02.19 | RobH | jsidhu2: remember that dial can be used to dial two extensions at the same time. |
| 20:02.19 | seanbright | mike-ekim: ok, then you don't need the 'load => chan_sip.so' line than |
| 20:02.31 | mike-ekim | oh |
| 20:02.42 | seanbright | mike-ekim: you have a sip.conf in /etc/asterisk? |
| 20:02.49 | RobH | its what i did back before i knew about queues. =P |
| 20:03.11 | [TK]D-Fender | mike-ekim: I asked you earlier if you were running a soft-phone on your server in addition to *. Are you? |
| 20:04.37 | hsv-al | heh |
| 20:04.47 | hsv-al | just got back from Target and theres a drink called |
| 20:04.49 | hsv-al | Bawls |
| 20:04.55 | seanbright | yup |
| 20:04.58 | seanbright | its salty |
| 20:04.59 | seanbright | ba dum dum |
| 20:05.03 | hsv-al | http://www.bawls.com/ |
| 20:05.17 | sp00kz | i *think* thinkgeek started that |
| 20:05.20 | RobH | hsv-al: thinkgeek.com been selling it for awhile, was big back in the lan party days. |
| 20:05.36 | hsv-al | g33k b33r |
| 20:05.38 | hsv-al | the hell |
| 20:05.42 | seanbright | OpenCola |
| 20:05.47 | RobH | sp00kz: either started it, or was one of the first major distributors of it yea |
| 20:06.49 | Strom_M | hsv-al: that drink is old news |
| 20:07.00 | hsv-al | never saw it till today |
| 20:07.02 | hsv-al | was pretty cheap too |
| 20:07.33 | JenniferAkemi | in sip.conf even when i put deny=0.0.0.0/0.0.0.0 then permit=192.168.0.0/255.255.255.0 under my [192.168.0.198] thing i get "Failed to authenticate user "Extension 1325" <sip:1325@192.168.0.198>;tag=8y5gc373rggNc" |
| 20:07.54 | *** join/#asterisk snapple42 (n=snapple4@h216-18-80-132.gtconnect.net) |
| 20:07.55 | JenniferAkemi | i thought that was supposed to mean that people from those IPs don't need to authenticate |
| 20:08.01 | Strom_M | JenniferAkemi: nope |
| 20:08.06 | JenniferAkemi | darn. |
| 20:08.16 | Strom_M | you want people to authenticate. always. |
| 20:08.16 | JenniferAkemi | what's it mean Strom_c? |
| 20:08.32 | [TK]D-Fender | JenniferAkemi: thats like saying I won't even ask the password unless its coming in from the right place and instead reject outright |
| 20:08.36 | Strom_M | JenniferAkemi: it means "allow connections from this IP range, and deny them outright from everyhting else" |
| 20:08.55 | JenniferAkemi | gotcha. |
| 20:09.16 | JenniferAkemi | how do i make it not require authentication if it comes from that? is there a way? |
| 20:10.22 | JenniferAkemi | i want asterisk to act like a sip provider to that IP |
| 20:10.33 | Strom_M | JenniferAkemi: why do you want to turn authentication off in the first place? |
| 20:11.28 | JenniferAkemi | Strom_M: extension 1325 is a phone on the other voip box. i can't add all the extensions into asterisks from the other box |
| 20:11.41 | Strom_M | JenniferAkemi: sounds like you want a sip proxy |
| 20:11.44 | Strom_M | try SER |
| 20:13.51 | deeperror | got the a104d on the way |
| 20:14.09 | hsv-al | deep |
| 20:14.14 | hsv-al | q t1/e1? |
| 20:14.20 | deeperror | yea |
| 20:14.32 | hsv-al | u get one under 2500? |
| 20:14.40 | deeperror | 2175 |
| 20:14.43 | hsv-al | heh |
| 20:14.48 | hsv-al | ebay i bet |
| 20:14.57 | deeperror | brand new |
| 20:15.10 | deeperror | 888voipstore |
| 20:15.19 | [TK]D-Fender | http://www.telephonydepot.com/product_p/105-052-003.htm |
| 20:15.23 | [TK]D-Fender | $2000 even |
| 20:15.40 | deeperror | haha |
| 20:15.43 | *** join/#asterisk moy (n=moyhu@nat/ibm/x-652fd4271344de5d) |
| 20:15.57 | *** part/#asterisk moy (n=moyhu@nat/ibm/x-652fd4271344de5d) |
| 20:16.09 | Qwell | $1,883.00 for the Digium card there :p |
| 20:16.14 | deeperror | didn't come out of my checking acct |
| 20:16.19 | deeperror | ;) |
| 20:16.59 | deeperror | it was a tossup between digium and sangoma |
| 20:17.32 | *** join/#asterisk moy (n=moyhu@nat/ibm/x-652fd4271344de5d) |
| 20:17.36 | deeperror | i've got a quote for te412p also but going to see how this one fairs |
| 20:18.14 | [TK]D-Fender | deeperror: I'm very happy with mine. |
| 20:18.25 | deeperror | which one a102? |
| 20:18.39 | [TK]D-Fender | deeperror: No, A104d |
| 20:18.44 | deeperror | ahh very nice |
| 20:19.01 | deeperror | yea i read on it a bit regarding the drivers and installation seemed fairly straight forward |
| 20:19.21 | [TK]D-Fender | deeperror: a few extra steps,but nothing to write home about. |
| 20:19.21 | deeperror | seems like just wanpipe install ? |
| 20:19.27 | [TK]D-Fender | deeperror: yup |
| 20:19.49 | deeperror | very cool...i've been using this rhino r4t1 and it just is not cutting it |
| 20:20.14 | *** join/#asterisk zippytech (n=ron@mail.zippytech.com) |
| 20:20.21 | deeperror | i'm running 4x 3 of the r1t1's but think it needs to get on one card instead of adding a 4th r1t1 haha |
| 20:20.47 | zippytech | any one here use a cisco 7960 G phone on asterisk? |
| 20:20.55 | xacatecas | how would i go about setting up a timer to terminate a call after a certain time? |
| 20:21.17 | Qwell | xacatecas: look at the L() option to Dial() |
| 20:21.18 | deeperror | [TK]D-Fender: right now my banks used rj48 is that what the a104 uses or is it some different cable? |
| 20:21.31 | xacatecas | Qwell, thanks! |
| 20:21.33 | Strom_M | zippytech: I've done so in the past |
| 20:21.46 | zippytech | do they work good? |
| 20:22.06 | Strom_M | zippytech: they're not bad...have you already purchased the set? |
| 20:22.15 | deeperror | i just looked it up...they look like its the same stuff |
| 20:22.16 | zippytech | no |
| 20:22.23 | Strom_M | zippytech: i'd recommend polycom |
| 20:22.28 | zippytech | hmm |
| 20:22.32 | Strom_M | overall, it's a much better experience |
| 20:22.42 | spokra | any opinion on broadvoice? |
| 20:22.55 | [TK]D-Fender | deeperror: yup, just Rj48 |
| 20:23.01 | deeperror | [TK]D-Fender: what is your card plugged into? lines or banks? |
| 20:23.01 | zippytech | i have grandstream and so far most are ok but there are little odd problem here and there with them |
| 20:23.18 | sp00kz | ~gs |
| 20:23.19 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 20:23.22 | *** join/#asterisk M1s3ry (n=M1s3ry@216.207.245.1) |
| 20:23.22 | [TK]D-Fender | deeperror: 1 PRI (straight) and 1 Channel Bank (Crossover) |
| 20:24.16 | mintee | is there an easy way for READ() to count a minimum amount of digits, i see there's a max, but no min |
| 20:24.34 | deeperror | very nice...i'm excited to get that card in the mail going to attempt an install into a power edge 1750 see how that goes |
| 20:25.05 | [TK]D-Fender | deeperror: the 1750 is exactly where I'm about to port my to as well in about a month. |
| 20:26.09 | deeperror | ha very cool |
| 20:26.58 | deeperror | i got pci so it should work in the pci-x ports and have it ready to plug in and start the install...that was another issue with rhino and the dsp on multi core chips was not working. ec would not work blah mess |
| 20:27.53 | [TK]D-Fender | deeperror: Not sure if Rhino suffers the same PCI issues Digiums did on PE's ,etc. thats what Sangoma got their rep on ; working in just about anything. |
| 20:28.23 | drumkilla | which the digium cards don't have problems with anymore |
| 20:28.29 | drumkilla | and haven't for a long time |
| 20:28.40 | deeperror | i worked with rhino for about 7 months and they were very great with us but at this point i'm just tired and want something that i can put in production and forget about ha |
| 20:28.44 | [TK]D-Fender | drumkilla: Yeah I heard 1.4 Zaptel helped that a lot. |
| 20:29.03 | drumkilla | a combination of drastic driver improvements, as well as new hardware designs |
| 20:29.09 | drumkilla | it's not the same story anymore. |
| 20:29.13 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 20:29.21 | *** join/#asterisk SplasPood (i=jwb@jwb.sh) |
| 20:29.34 | [TK]D-Fender | drumkilla: Well, I'll leave the "new hardware design" part out for now. If you need a new card, then the older one isn't getting better :) |
| 20:29.43 | deeperror | and i was thinking about getting the digium for that reason since they kinda make the pbx i'm using but not sure....sangoma seemed to get a better response, great warranty, and all the people i talked to on the phone were friendly and helpful....digium sales is just that...sales |
| 20:29.46 | [TK]D-Fender | drumkilla: Moderns stuff though... thats better |
| 20:30.26 | [TK]D-Fender | drumkilla: I've got to canvas for opinions these days to keep myself a bit more up to dATE. |
| 20:30.28 | Qwell | deeperror: great warranty? with Digium, if it doesn't meet your expectations, we'll take it back. Period. |
| 20:30.40 | [TK]D-Fender | BUT ALAS, TIME HERE AT THE OFFICE HAS RUN OUT. HEADING HOME, bbiab |
| 20:30.44 | [TK]D-Fender | darn caps :) |
| 20:30.49 | deeperror | yea thats what i heard |
| 20:30.52 | drumkilla | [TK]D-Fender: yep, i would encourage you to check stuff out and refresh your facts. :) |
| 20:31.22 | [TK]D-Fender | deeperror: If Sangoma doesn't meet your expectations, then you're probably trapped in Bizarro-world ;) |
| 20:31.40 | deeperror | Qwell: i've still got the quote for the digium haha may be placing that order too just have to see how this one performs first |
| 20:31.52 | drumkilla | and they don't do anything to help asterisk development |
| 20:31.53 | drumkilla | period |
| 20:31.54 | drumkilla | at all |
| 20:32.01 | drumkilla | if that means anything to you |
| 20:32.12 | deeperror | hrmm |
| 20:32.45 | Yourname` | Hi. How can I make sure that when calls go to this particular extension, that I want a special CDR flag, or something like that? Basically.. I want logging on when this extensions is dialed. http://pastebin.ca/1033423 |
| 20:32.45 | drumkilla | I don't say that you should ever use a worse product |
| 20:32.48 | drumkilla | but I _do_ ask that you give Digium a chance. |
| 20:33.12 | drumkilla | and let us try to prove that we're the best. |
| 20:33.25 | deeperror | I am with the pbx ;) just don't have the cash to throw down on something i can't try first |
| 20:33.30 | deeperror | esp when it's a few grand |
| 20:33.43 | drumkilla | if the card doesn't work, digium will completely refund your money |
| 20:33.47 | drumkilla | regardless of where you buy it |
| 20:33.59 | hardwire | Do US PRI send congestion status over the d-chan when establishing a call? |
| 20:34.07 | hardwire | Does asterisk have to support the tones to know if it's congested? |
| 20:34.38 | hardwire | sorry, to know if it's got a "fast busy" on the other end of the answered call. |
| 20:40.12 | *** part/#asterisk moy (n=moyhu@nat/ibm/x-652fd4271344de5d) |
| 20:40.41 | *** join/#asterisk moy (n=moyhu@nat/ibm/x-b24abdbdcf5e3b5f) |
| 20:42.55 | ManxPower | hardwire: yes, no, doesn't make sense |
| 20:43.11 | ManxPower | you don't normally get tones on a USA PRI |
| 20:43.30 | ManxPower | You check the status of HANGUPCAUSE after the Dial, do whatever is correct for each value of HANGUPCAUSE |
| 20:46.31 | ManxPower | hardwire: you can see an example using DIALSTATUS instead of HANGUPCAUSE in extensions.conf.sample Look for stdexten |
| 20:46.47 | ManxPower | HANGUPCAUSE just gives more information than DIALSTATUS |
| 20:48.08 | ManxPower | deeperror: As far as I can tell from reports, modern Digium cards are on par with Sangoma for compat and IRQ latency. Older cards did have problems on some systems. |
| 20:48.50 | deeperror | ok i'll be testing them out for sure |
| 20:49.01 | deeperror | may be a month or so before i can get one |
| 20:51.11 | Yourname` | What's the replacement for SetAccount in 1.4? |
| 20:56.51 | hardwire | ManxPower: interesting |
| 20:56.59 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 20:57.15 | hardwire | ManxPower: we use a sip provider that answers, then continues bringing up the rest of the call on it's own trunks. |
| 20:57.30 | hardwire | I need to detect the status instead of rely on them to tell me the status. |
| 20:58.00 | hardwire | they aren't doing any sort of tone detection to give back a good response before a real Answer() occurs. |
| 20:58.17 | jjshoe | Qwell sangoma has a lifetime warranty, does digium? |
| 20:58.23 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:58.39 | hardwire | jjshoe: explain lifetime.. :) your life? my life? sangoma's life? |
| 20:58.48 | hardwire | isn't time relative? |
| 21:00.52 | deeperror | digium said 2 years |
| 21:03.04 | Qwell | 5 |
| 21:03.32 | sp00kz | new appliance :o |
| 21:04.17 | Yourname` | If I use SetCDRUserfield(hello), where shall the 'hello' show up? In cdr-custom's Master.csv's userfield? |
| 21:04.42 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 21:06.48 | drmessano | Qwell: |
| 21:06.48 | *** join/#asterisk Shotygun (n=thorn@82.166.244.147) |
| 21:07.15 | Qwell | ? |
| 21:07.41 | drmessano | I want you to be honest with me.. |
| 21:08.26 | drmessano | I heard you maintain the SCCP implementation in *.. including the others out there, is it the better one to use right now? |
| 21:08.36 | Qwell | it's the only one to use |
| 21:08.58 | Qwell | chan_sccp hasn't had a release in over 2 years, and doesn't support 1.4 |
| 21:10.10 | Qwell | is the berlios site even still up? |
| 21:10.22 | Qwell | 20060408 |
| 21:10.23 | Qwell | last release |
| 21:10.48 | drmessano | What about that one --> http://sourceforge.net/project/showfiles.php?group_id=186378 |
| 21:11.03 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 21:11.17 | Qwell | okay, so it's had 2 releases ever :p |
| 21:11.21 | Qwell | it's a fork of chan_sccp |
| 21:11.24 | drmessano | oh ok |
| 21:11.37 | Qwell | also unmaintained |
| 21:11.41 | drmessano | You would know much better than I would.. I just google |
| 21:11.50 | Qwell | there are many many many critical crash (and security) bugs |
| 21:12.11 | Qwell | putting a call on hold = crash, parking = crash, dialing randomly = crash |
| 21:12.20 | drmessano | nice |
| 21:12.32 | Qwell | it's a joke. it's had like 5 maintainers, who have all given up after a few months |
| 21:12.59 | Qwell | Sergio seemed active for a while, but.. |
| 21:13.43 | drmessano | Is the one your maintain stable enough to drop 30 or 40 Cisco phones on a box and expect the reliability of say Polycom SIP phones? |
| 21:13.48 | Qwell | no |
| 21:13.53 | Qwell | buy polycom phones |
| 21:13.56 | drmessano | lol |
| 21:14.06 | drmessano | Well, some people are Cisco whores |
| 21:14.25 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 21:14.29 | drmessano | What are the issues.. scalability, or just the usual need to hammer out the BS crap |
| 21:14.46 | Qwell | the protocol is ever changing, and undocumented |
| 21:14.55 | drmessano | I see |
| 21:14.59 | Qwell | it *IS* better than SIP, but.. |
| 21:15.09 | Qwell | I mean, the protocol - not the implementation(s) |
| 21:15.10 | jbeez | but its alot of work |
| 21:15.14 | jbeez | that needs to be done for free |
| 21:15.36 | Qwell | well, I have 1 single Cisco phone here, and it's very very stupid (firmware issues) |
| 21:16.15 | Qwell | Cisco told me that "less than 1/100th of 1% "(odd math, but whatever)"of our phones are used with Asterisk" |
| 21:16.39 | [hC] | there is a new fork of chan_sccp |
| 21:16.44 | [hC] | at sourceforge, chan_sccp-b |
| 21:16.50 | [hC] | already, its better |
| 21:16.52 | Qwell | [hC]: see above |
| 21:16.56 | jbeez | I have a phone guy tell me, cisco voip phones are not made by a phone company, and lack(ed) several standard pbx features |
| 21:17.03 | drmessano | Yeah... you're late to the convo |
| 21:17.09 | [hC] | Sorry. i'll read back. |
| 21:17.09 | [hC] | heh. |
| 21:17.12 | Qwell | jbeez: Cisco phones are actually quite fully featured |
| 21:17.27 | Qwell | SIP no. Skinny, absolutely |
| 21:17.43 | jbeez | he was bitching about doing a doorphone, he may have been more targetting their headend portion, the pbx portion than the phone itself |
| 21:17.55 | Qwell | dynamic softkeys, the pbx handles everything, rather than the phone |
| 21:18.02 | Qwell | it's technically MUCH better than SIP ever could be. |
| 21:19.09 | drmessano | One thing I like about the phones at work is the one-touch signing in and out of the queue |
| 21:19.17 | Qwell | easy with xml services |
| 21:19.52 | [hC] | I'm running chan_sccp-b 20071213, and its alot better than the old version sergio had |
| 21:20.17 | [hC] | honestly i like alot of the functionality sccp carries, and the SIP firmwares are just so limited and screwed up (on purpose?) |
| 21:20.48 | [hC] | Aastra is the only other phone that comes close in terms of programability to a cisco.. The one killer feature i love about cisco is their BLF implementation |
| 21:20.50 | hsv-al | heh |
| 21:20.53 | [hC] | You can see who the other person is TALKING TO. |
| 21:21.45 | hsv-al | dont think you can dl blf on linksys btw |
| 21:21.47 | hsv-al | if your a poor home user |
| 21:22.06 | hsv-al | s/dl/do |
| 21:22.33 | ManxPower | hardwire: you need a new ITSP!!! |
| 21:22.41 | [hC] | I dont really bother with linksys much unless the customer really wants it for some reason |
| 21:22.42 | ManxPower | A telco should NEVER EVER answer the line for no reason |
| 21:23.00 | [hC] | For the price of aastra and polycom phones these days, linksys isnt worth it |
| 21:23.04 | hardwire | ManxPower: I totally agree |
| 21:23.13 | hsv-al | hc |
| 21:23.16 | hsv-al | u mess with grandstreams? |
| 21:23.25 | [hC] | hsv-al: I'll tell you one thing, and one thing only. |
| 21:23.25 | ManxPower | hardwire: the option you are looking for is busydetect and callprogress, they are commonly known in the Asterisk community as "randomlydisconnectmycalls=yes" |
| 21:23.28 | hardwire | ManxPower: I'm still trying to diagnose if it was one of their trunks that did it for them, or not. |
| 21:23.46 | [hC] | hsv-al: The only grandstream i have is mounted in my bathroom. It plays music, and sometimes intercoms. It doesnt even do a good job of that. |
| 21:23.51 | hsv-al | :) |
| 21:24.09 | hsv-al | well, your looking for cheapness |
| 21:24.10 | hardwire | they are a homebrew joint and I wanna give em my business, hopefully I won't be shooting myself in the foot. |
| 21:24.12 | hsv-al | why did i say it heh |
| 21:24.39 | ManxPower | hardwire: then teach them the right things to do. |
| 21:24.45 | [hC] | hsv-al: aastra and polycom both have phones in the $100 and possibly sub $100 area.. why bother with grandstream. |
| 21:25.04 | ManxPower | Honestly, I don't use providers that I have to teach them what to do |
| 21:25.33 | [hC] | The Aastra 51i is $98 on google, and the polycom IP320 Is $90 |
| 21:25.43 | ManxPower | [hC]: maybe they are all smokers and want to use the cost difference between the GS and a real phone and spent it on a couple of packs of smokes |
| 21:25.57 | ManxPower | ok, maybe a carton of smokes |
| 21:26.03 | [hC] | hahah |
| 21:26.13 | gr0mit | gave up with Grandstreams a looooong time ago |
| 21:26.21 | lesouvage | Can I use the res_features.c from asterisk 1.4.18.1 with asterisk 1.4.20.1 or will this cause serious problems. |
| 21:26.31 | [hC] | wow |
| 21:26.36 | [hC] | The budgetone 101's are $35 |
| 21:26.41 | [hC] | no wonder you cheap bastards are clawing at them |
| 21:26.42 | drmessano | HA |
| 21:26.58 | [hC] | Any time you can buy an IP phone for $35, you have to ask yourself "Why is this so cheap?" |
| 21:27.09 | ManxPower | Heck you can buy TWO GS for the price of a carton of name brand smokes in NY |
| 21:27.11 | hardwire | ManxPower: eventually I will be |
| 21:27.15 | gr0mit | would rather pay $75 and have something which worked |
| 21:27.17 | hsv-al | heh |
| 21:27.35 | drmessano | Does a BT-101 make a good paging speaker for short distances? |
| 21:27.40 | hsv-al | grandstreams give us all "saftsacks" |
| 21:27.41 | hsv-al | lol |
| 21:27.47 | ManxPower | I'll bet these people spend $150 on a bag of weed, but not $100 on a decent phone. |
| 21:28.00 | sp00kz | lol |
| 21:28.18 | gr0mit | grandstremas are in my opinion great as a first start when you think this voip lark can't possible work |
| 21:28.33 | ManxPower | gr0mit: GS gives VoIP a bad name. |
| 21:28.40 | sp00kz | they're great when you want to buy two sets of phones |
| 21:28.44 | gr0mit | but when you realise it can and does, you soon give up with grandstrams and go for proper kit |
| 21:28.45 | sp00kz | and then throw one out |
| 21:29.18 | gr0mit | has had good success with snoms and Thomson phones |
| 21:29.31 | ManxPower | hugs his Polycoms, all 300 of them. |
| 21:29.34 | gr0mit | but am feeling Aastra might be be a better bet |
| 21:29.42 | sp00kz | I love my polycoms too, 650's are sexy |
| 21:29.57 | [hC] | I wonder how many phones ive deployed so far |
| 21:29.59 | [hC] | Its got to be over 1000 |
| 21:30.04 | gr0mit | has not tried them |
| 21:30.06 | sp00kz | same here :/ |
| 21:30.25 | gr0mit | do they have decent BLF like snoms? |
| 21:30.40 | [hC] | I have run into my fair share of weird fucking problems with polycom let me tell you |
| 21:30.44 | sp00kz | they imo are 10x better than snoms |
| 21:30.49 | *** join/#asterisk [TK]D-Fender (i=Joe@64.235.218.194) |
| 21:30.58 | sp00kz | haha, yeah, there are a lot of issues |
| 21:31.02 | sp00kz | but when you get them running right |
| 21:31.14 | [hC] | even today, I have an issue where turning on "ringing" for call waiting (ie ringing the base) instead of beeping in your ear, after a couple hours the ringer reduces its own volume until you cant hear it any more (yet the volume is turned up) |
| 21:31.16 | sp00kz | they're the bomb diggity |
| 21:31.27 | [hC] | its this sort of crap that i hate dealing with, it makes us look bad to our clients |
| 21:31.31 | sp00kz | ya, i came across that before |
| 21:31.38 | [hC] | you did? |
| 21:31.39 | sp00kz | had to turn it off |
| 21:31.40 | sp00kz | yeap |
| 21:31.42 | [hC] | did you fix it? |
| 21:31.52 | sp00kz | I don't know |
| 21:31.52 | sp00kz | lol |
| 21:31.55 | sp00kz | lemme find out |
| 21:31.57 | [hC] | Ive found one other person who has the problem and cant figure out how to fix it |
| 21:32.06 | sp00kz | yeah, the ringers go way down |
| 21:32.08 | [hC] | I would love to have the feature turned on, but it seems to break shit. |
| 21:32.11 | sp00kz | when the base is set to ring instead of beep |
| 21:32.28 | sp00kz | breaks a lot it seemed when i was doing it |
| 21:32.54 | sp00kz | Oh, we turned it off because it made volumes not save |
| 21:32.55 | sp00kz | lol. |
| 21:33.13 | sp00kz | like ringer, speaker, handset |
| 21:33.26 | [hC] | Ive also had a lot of lockups/freezes/reboots when using a 601 + sidecars + blf. I'm not sure if its fixed now, but I think it is (people are horrible at reporting fixes).. Seems to have been related to how you construct your directory xml file. If you put in too many <bw> and <sd> tags, the phones lock up |
| 21:33.32 | [hC] | and by too many i mean more than ... 50? |
| 21:33.40 | sp00kz | :/ |
| 21:33.59 | sp00kz | we hooked up an asterisk box to handheld radios |
| 21:34.02 | sp00kz | that was fun project |
| 21:34.17 | sp00kz | 2 way radios that called phones |
| 21:34.17 | sp00kz | :P |
| 21:34.21 | sp00kz | and visa versa |
| 21:35.13 | [hC] | how'd you do that? |
| 21:35.38 | sp00kz | by far the worst problem ive had though, is that randomly phones screw up adtran poe routers, the older revisions and continually reboot :/ |
| 21:35.45 | sp00kz | with a soundcard tied to the headset jacks |
| 21:35.56 | sp00kz | was for lockheed |
| 21:36.13 | [hC] | wait so you'd have one radio plugged in to the server |
| 21:36.21 | [hC] | and it would send audio into like, line in? |
| 21:36.26 | sp00kz | yeap |
| 21:36.38 | sp00kz | sort of |
| 21:36.41 | [hC] | was it just in a meetme or something, or were you actually able to call people |
| 21:37.00 | sp00kz | you have to use a good soundcard, external with like real mic jacks |
| 21:37.12 | *** join/#asterisk RoyK (n=Roy@91.149.38.81) |
| 21:37.16 | sp00kz | i could call the handhelds from my desk phone |
| 21:37.39 | sp00kz | and could hold a convo, with push to talk 2way radios |
| 21:37.45 | [hC] | individual ones, or just the whole group? Im curious how you pulled that off.. |
| 21:37.47 | [hC] | that would be a fun project |
| 21:37.53 | sp00kz | well |
| 21:38.01 | sp00kz | the radios had 4 channels |
| 21:38.05 | sp00kz | so one radio per channel |
| 21:38.21 | sp00kz | need 8 radios to do it |
| 21:38.25 | sp00kz | 4 connected to pbx |
| 21:38.29 | sp00kz | one on each channel |
| 21:39.47 | [hC] | oh i see. |
| 21:39.53 | [hC] | the 4 connected to a pbx would each be on a separate channel |
| 21:40.11 | Yourname` | My only question after 2 got unanswered. If I Set(CDR(userfield)=hello) -> how do I make the userfiel appear in Master.csv in /var/log/asterisk/cdr-custom/Master.csv? A module reload cdr, or what exactly? |
| 21:40.29 | [hC] | and the handhelds that people carried, if there was a 'call' coming in on another channel, would it tell them? or would they have to already be on that channel? |
| 21:41.04 | sp00kz | they would have to be on the channel |
| 21:41.11 | sp00kz | it wasnt designed for real use |
| 21:41.16 | sp00kz | more as a PoC |
| 21:41.20 | sp00kz | and demo |
| 21:42.08 | [hC] | cool.. so there probably wasnt any like.. call setup... (ringing/answered/hangup) it was just, open the channel connected to the right input on the sound card and pass audio... either with a meetme force join, or something similar |
| 21:47.06 | *** join/#asterisk x_or (n=cdawson@or-76-2-6-173.dhcp.embarqhsd.net) |
| 21:47.49 | *** part/#asterisk hubguruJR (n=hubguruJ@mail.ntegratedsolutions.com) |
| 21:47.53 | x_or | Can someone tell me where to look to register with a trunk? I signed up for a provider and am not sure which configuration file I need to modify to specify my connection settings. |
| 21:49.24 | *** join/#asterisk waKKu (n=ugabuga@unaffiliated/wakku) |
| 21:49.27 | waKKu | ~book |
| 21:49.28 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 21:50.32 | *** join/#asterisk golumn (n=golumn@201.220.132.138) |
| 21:52.41 | *** part/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
| 21:53.12 | x_or | waKKu: Was that for me? |
| 21:53.41 | waKKu | x_or see my join just before.... what do u think ? :) |
| 21:54.00 | golumn | Hi guys, Yesterday I ask about asterisk using some meridians phones, but someone told me that they aren't compatible but that there is a gateway. I decide not using that gateway so which card would you recomend for let say 12 extensions and what phones do I need to buy? |
| 21:54.19 | [TK]D-Fender | x_or, ... |
| 21:54.21 | [TK]D-Fender | ~book |
| 21:54.22 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 21:54.29 | [TK]D-Fender | x_or, THAT is for you :) |
| 21:54.48 | x_or | You guys are SO generous! :) |
| 21:55.16 | [TK]D-Fender | golumn, we typically suggest decent SIP phones (Polycom), and those just go on your network switch |
| 21:55.33 | [TK]D-Fender | x_or, we are... thats a great book you get to read for FREE. |
| 21:58.24 | golumn | [TK]D-Fender, one question tell me if I am wrong. By using a sip phone it won't be necesary to buy a digium special card I connect the telephone to the computer nic? |
| 21:58.55 | [TK]D-Fender | golumn, Correct. SIP phones are IP phones. As in a direct TCP/IP device |
| 21:59.47 | golumn | [TK]D-Fender, that is a better option that the one I was thinking |
| 22:00.14 | golumn | I rather have everything VoIP that a analog line |
| 22:01.51 | [TK]D-Fender | golumn, You might still want analog lines. Phones are great to be IP on the INSIDE of your private LAN. having your PSTN connectivity going over the public internet is another matter. |
| 22:02.12 | [TK]D-Fender | golumn, its a question of reliability & privacy |
| 22:02.14 | *** join/#asterisk masus (n=ethemc@78.162.30.135) |
| 22:02.47 | golumn | [TK]D-Fender, I am looking on the web and look some fancy Polycom Sip Phones. What I am looking is for a basic phone. Dial and received |
| 22:03.14 | masus | hi , i have a small problem . if i run a code who connects to an ODBC database with func_odbc my asterisk shutdown automaticaly with no messages, can anybody help? Thanks ... |
| 22:03.16 | golumn | nothing more |
| 22:03.58 | ManxPower | golumn: low end polycom |
| 22:04.04 | ManxPower | the 3xx and 4xx series |
| 22:04.11 | masus | i use asterisk 1.4 and debian etch |
| 22:04.20 | ManxPower | masus: did you compile from source? |
| 22:04.25 | masus | ManxPower: Yes |
| 22:04.33 | lesouvage | When I have to downgrade from Asterik 1.4.20.1 to Asterisk 1.4.18.1 do I have to recompile zaptel too? |
| 22:04.44 | ManxPower | masus: start asterisk as "asterisk -cvvv" then try it, you should get more messages |
| 22:04.46 | golumn | ManxPower, do you know an specific model |
| 22:04.55 | masus | ManxPower: ill try thanks |
| 22:05.09 | ManxPower | golumn: I could go to the polycom web site, and find all the 3xx models for you, but I think you can do that yourself. |
| 22:06.00 | masus | ManxPower: i get this error :) asterisk: symbol lookup error: /usr/lib/asterisk/modules/func_odbc.so: undefined symbol: ast_odbc_request_obj |
| 22:06.59 | masus | but get no answers at google :/ |
| 22:07.07 | masus | can anybody help how to fix this ? |
| 22:07.17 | spokra | masus: what do you get when you do ldd /usr/lib/asterisk/modules/func_odbc.so |
| 22:07.35 | ManxPower | masus: you used a different odbc than the one that came with your asterisk |
| 22:08.21 | *** join/#asterisk hubguruJR (n=hubguruJ@mail.ntegratedsolutions.com) |
| 22:08.24 | masus | a different ODBC ? |
| 22:09.13 | masus | spokra: thats the result -> http://pastebin.com/d24fa728 |
| 22:09.26 | masus | a different func_odbc you mean ? |
| 22:09.29 | Qwell | ~sipnat |
| 22:09.30 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 22:11.10 | spokra | masus.. the error is saying the shared library you linked to during compile is different now and doesn;t have the same entry point.. |
| 22:11.35 | masus | how can i fix this ? |
| 22:12.04 | ManxPower | masus: recompile asterisk |
| 22:12.17 | masus | hmm |
| 22:12.18 | masus | ok |
| 22:12.19 | x_or | masus: What is happening is that the shared library library (so) has different symbols inside it than asterisk expects. This is usually when you compile asterisk yourself. |
| 22:12.20 | masus | one mom |
| 22:12.22 | spokra | make sure you have all the packages required by asterisk loaded.. go into the asterisk source dir.. type make distclean;./configure;make |
| 22:12.22 | x_or | Did you do that? |
| 22:12.31 | ManxPower | on that system, no don't compile it somewhere else and move it to that system -- that's what screwed it up in the first place |
| 22:12.51 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 22:13.09 | masus | if i make a distclean will my sample files removed ? |
| 22:13.20 | x_or | How can I confirm that an IAX2 trunk is registered using the CLI? |
| 22:13.36 | smash- | can someone help me out really quick im trying to setup a dialplan in an ATA device, the device got reset to default. it use to require a 9 to dial out so 918005551234 |
| 22:13.38 | x_or | masus: sample files? You can make a copy of the source tree. |
| 22:13.38 | spokra | no! there is a make command to copy samples to /etc/asterisk |
| 22:13.48 | masus | ok |
| 22:13.54 | smash- | i belive it was something like (9xxxxxxxxxxxx) |
| 22:13.57 | spokra | then do a make install. |
| 22:14.52 | masus | do i have to do the same for asterisk-addons ? |
| 22:15.05 | spokra | you probably should.. |
| 22:15.12 | masus | ok one mom please |
| 22:15.15 | ManxPower | x_or: You can't. You can only tell if an IAX2 user is registered |
| 22:15.44 | Yourname` | Hello hello, If I Set(CDR(userfield)=hello) -> how do I make the userfiel appear in Master.csv in /var/log/asterisk/cdr-custom/Master.csv? A module reload cdr, or what exactly? |
| 22:15.56 | Yourname` | module reload cdr doesn't work btw. |
| 22:16.03 | Yourname` | As in doesn't show the userfield |
| 22:16.51 | x_or | ManxPower: What does this command do: iax2 show registry. I get back "2xx.1xx.2xx.2x:4569 N foobar 2xx.2x.6x.2xx:4569 60 Registered" |
| 22:17.26 | ManxPower | x_or: that shows what peers Asterisk is registered to |
| 22:18.26 | x_or | ManxPower: OK. I'm not clear on Asterisk terminology yet, but I am registering with an IAX2 trunk provider, so that is not the same thing? |
| 22:18.27 | hardwire | "x_or" looks like a beat up emoticon with a funny haircut |
| 22:19.09 | bkw_ | Yourname`: if I recall userfield shows up if you set it. |
| 22:19.17 | x_or | hardwire: I have had this nick for years and that is the funniest thing I have ever heard. I love it. Wow, I do look like an abused housewife at the grocery store with dark sunglasses. |
| 22:19.24 | bkw_ | Yourname`: what asterisk version? |
| 22:19.28 | ManxPower | x_or: no. An IAX2 trunk is a specially configured IAX2 connection to reduce bandwidth usage when 3 or more calls are happening at the same time. If you are not doing that (via trunk=yes on BOTH servers) then you don't have a trunk. you have a connection |
| 22:19.32 | Yourname` | bkw_: 1.4* |
| 22:19.47 | bkw_ | might need to set loguserfield=1 in your config |
| 22:20.05 | Yourname` | bkw_: Which config? |
| 22:20.10 | bkw_ | cdr.conf I think it is |
| 22:20.25 | Yourname` | ohk |
| 22:20.53 | bkw_ | that should get the userfield logged.. no clue what that isn't on by default. |
| 22:21.43 | ManxPower | x_or: "trunk" here means something very very specific |
| 22:21.46 | Yourname` | gotcha.. lemme look, thanks bkw_ |
| 22:21.47 | ManxPower | ~trunk |
| 22:21.47 | jbot | trunk is, like, a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 22:22.19 | bkw_ | sip trunk.. people use that term |
| 22:22.19 | codefreeze-lap | Yourname`: actually, it'd be cdr_custom.conf |
| 22:22.31 | bkw_ | codefreeze-lap: you might wanna check cdr.conf |
| 22:22.34 | bkw_ | for cdr_csv.c |
| 22:23.16 | codefreeze-lap | bkw_: yes, but up above, he's looking in cdr-custom/Mater.csv... so... |
| 22:23.55 | ManxPower | bkw_: People also used to say the earth is flat, they are no more accurate in that than someone saying "sip trunk" |
| 22:24.14 | bkw_ | codefreeze-lap: sorry I seen that.. I should know all about CDR custom |
| 22:24.21 | Yourname` | bkw_: Thanks!! |
| 22:24.26 | Yourname` | codefreeze-lap: thanks!! |
| 22:24.40 | bkw_ | codefreeze-lap: I know all about it.. because we wrote cdr_custom |
| 22:25.23 | codefreeze-lap | bkw_: no prob; as they say around here in deference, "You probably have forgotten more.... than I ever learned!" |
| 22:25.41 | bkw_ | maybe |
| 22:25.46 | bkw_ | I try not to forget but sometimes I can't help it |
| 22:26.09 | codefreeze-lap | oh, and add to that...." and you haven't forgotten much!" |
| 22:26.20 | bkw_ | Yourname`: the default cdr_custom also adds the userfield to the log |
| 22:26.38 | golumn | [TK]D-Fender, One question. If I want to use an analog line for incoming phones, and then use the polycom sip phones for the extensions I will need to buy a asterisk analog card right. Is it possible to do it that way? |
| 22:26.50 | golumn | *incoming calls |
| 22:26.53 | Yourname` | bkw_: Just noticed it.. thanks |
| 22:27.02 | ManxPower | golumn: you sure did read the book fast! |
| 22:27.11 | bkw_ | Yourname`: np |
| 22:27.17 | bkw_ | Yourname`: just pay it forward |
| 22:27.43 | masus | great work, ManxPower and spokra . its work now thank u very much .. |
| 22:28.36 | Yourname` | Great, thanks bkw_ ... you rock! |
| 22:28.43 | bkw_ | :P |
| 22:29.38 | *** join/#asterisk waKKu (n=ugabuga@unaffiliated/wakku) |
| 22:30.11 | hardwire | waKKu waKKu |
| 22:30.45 | waKKu | wacca wacca ;D |
| 22:32.35 | hardwire | raise your hand if you think life should suck less. |
| 22:33.13 | bkw_ | hardwire: if life doesn't suck we never grow as humans |
| 22:36.45 | [hC] | I for one, think life should suck (me) more. |
| 22:36.47 | [hC] | Just aying. |
| 22:36.53 | [hC] | saying* |
| 22:41.08 | *** join/#asterisk stuffcorpse (n=pickle@60-234-225-122.bitstream.orcon.net.nz) |
| 22:47.44 | LuisTorres | vaca? |
| 22:48.52 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 22:49.08 | *** join/#asterisk madduck (n=madduck@debian/developer/madduck) |
| 22:49.13 | madduck | i have a *-server on a public IP out there, and am right now behind a Belkin ADSL router |
| 22:49.30 | madduck | my softphone (twinkle) can register and make calls just fine |
| 22:49.38 | madduck | but my nokia e51 cannot, it seems. |
| 22:49.48 | madduck | it registers, even gets the 200 OK back |
| 22:49.57 | madduck | but then it just says "cannot connect to the connection network" |
| 22:50.06 | madduck | i realise this is probably a nokia braindeadness |
| 22:50.15 | madduck | but maybe someone here has heard of this and knows a way around? |
| 22:51.44 | masus | by all |
| 22:53.23 | ManxPower | madduck: do you have nat=yes for both peers? |
| 22:56.28 | *** join/#asterisk macros73 (n=cs@c-24-131-77-140.hsd1.pa.comcast.net) |
| 22:56.56 | jaytee | "nat"ty dread, mon |
| 22:57.17 | madduck | ManxPower: yes |
| 22:59.09 | madduck | i was thinking it's probably the belkin router |
| 22:59.17 | madduck | since i could use the e51 at home behind my linux router |
| 22:59.35 | seanbright | any polycom digitmaps gurus about? |
| 22:59.44 | madduck | uh, which is actually the asterisk server, so maybe not... |
| 23:00.01 | madduck | but if the softphone can connect but the e51 cannot, then it's hardly the router, huh? |
| 23:04.51 | Strom_C | seanbright: whatcha need? |
| 23:05.04 | *** join/#asterisk delparnel (n=delparne@CPE001839f2889c-CM00137185dfb6.cpe.net.cable.rogers.com) |
| 23:06.01 | seanbright | Strom_C: need my IP 501 to accept 2, 4, and 6 digit star codes |
| 23:06.19 | seanbright | Strom_C: e.g. *97, *1111, and *881111 |
| 23:06.23 | Strom_C | seanbright: why in god's name are you screwing with vertical service code assignments? |
| 23:06.43 | Strom_C | VSCs are only two digits with the exception of two reserved blocks for three digit expansion |
| 23:07.01 | Strom_C | ~vsc |
| 23:07.02 | jbot | it has been said that vsc is Vertical Service Codes such as *67, *69, *72, and *82. These codes are generally reserved for specific uses, and it's a bad idea to conflict with the official assignments. A list of assigned VSCs for North America is at http://nanpa.com/number_resource_info/vsc_assignments.html and http://www.nanpa.com/number_resource_info/vsc_definitions.html |
| 23:07.21 | seanbright | so you aren't going to help? ok |
| 23:07.24 | seanbright | thanks |
| 23:07.27 | Strom_C | i'm asking why |
| 23:07.57 | Strom_C | i'm trying to help you make sure that breaking the rules is exactly what you want to do before i help you break them |
| 23:08.10 | seanbright | Strom_C: gotcha. thanks. |
| 23:08.11 | Strom_C | but having this kind of variable-length numbering system is problematic anyway |
| 23:09.54 | seanbright | Strom_C: when i started here, i inherited an AMP install |
| 23:10.09 | seanbright | Strom_C: which used *97 and *98xxxx for accessing VM |
| 23:10.24 | seanbright | Strom_C: i've since migrated to a vanilla asterisk install |
| 23:10.38 | seanbright | Strom_C: maintained the *foo for backwards compatibility reasons |
| 23:12.11 | Strom_C | hm |
| 23:12.16 | Strom_C | ok |
| 23:12.32 | seanbright | Strom_C: and that link you sent suggests 97 and 98 are for local assignment |
| 23:12.43 | seanbright | i think adding *xx.T will work |
| 23:12.45 | seanbright | testing it now |
| 23:13.10 | Strom_C | seanbright: yeah, but you said *1111 and *881111, which aren't locally assignable ;) |
| 23:13.21 | Strom_C | seanbright: is it only *97 and *98xxxx? |
| 23:14.16 | seanbright | Strom_C: yes |
| 23:14.24 | Strom_C | well that's dead easy then |
| 23:14.33 | seanbright | oh |
| 23:14.34 | seanbright | right |
| 23:14.41 | Strom_C | just put in *97|*98xxxx| etc |
| 23:14.42 | seanbright | explicitly use *97|*98xxxx |
| 23:14.45 | seanbright | right |
| 23:14.48 | seanbright | ok, thanks. |
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| 23:29.23 | *** join/#asterisk FarrisG (n=jrush@gateway.wiquest.com) |
| 23:29.28 | FarrisG | Some of my calls to a specific number are being disconnected after exactly one minute, and then the disposition in the CDR shows "No answer". Are the calls just not receiving a valid pickup, even though the call is successful and I can speak to the other end? |
| 23:30.09 | Strom_C | FarrisG: well, does the other end actually appear to be sending answer supervision? |
| 23:30.21 | Strom_C | because it sounds like they aren't |
| 23:30.46 | directory | is it a toll-free number? |
| 23:30.57 | Strom_C | and usually the IXC will cut your call off after 1-3 minutes without answer supervision |
| 23:31.20 | directory | Strom_C: quick, take out my garbage! |
| 23:31.40 | Strom_C | DONE |
| 23:31.46 | directory | liar |
| 23:31.50 | FarrisG | yes it's a toll-free number. How do I tell if the other end is sending answer supervision? |
| 23:32.06 | Strom_C | FarrisG: i'll make it easy -- PM me the number |
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| 23:32.54 | Strom_C | FarrisG: but if they do send answer supervision, your equipment will show the call as answered and the call timer will begin counting |
| 23:33.54 | FarrisG | Done |
| 23:33.59 | FarrisG | Oh, yeah, you're right. |
| 23:34.05 | FarrisG | My phone doesn't start a call timer |
| 23:34.19 | Strom_C | yup, they're not sending supervision |
| 23:34.24 | Strom_C | what a bunch of fucking cheapskates |
| 23:35.17 | FarrisG | wow, that's fucked up |
| 23:35.18 | Strom_C | stupid idea too, because you're not even supposed to be able to talk forward until you receive supervision |
| 23:36.57 | directory | I should really do something with this geode prototype board... or put it back in storage |
| 23:37.18 | directory | mumbles to himself |
| 23:39.44 | alrs | Strom_C: I've found that audio usually goes two ways when a carrier permits early media, but DTMF definitely breaks |
| 23:40.18 | Strom_C | alrs: yes, but if you read the specs, that's not what's supposed to happen |
| 23:40.42 | Strom_C | all forward audio is supposed to be restricted until the called party sends answer supervision |
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