| 00:00.24 | interfaithquest | some how i think the idea of peer to peer free voip is not good for wall street ? as it stands today |
| 00:00.44 | interfaithquest | however freedom of communication maybe the best medicine for all |
| 00:00.55 | interfaithquest | let the titans fall i say |
| 00:01.12 | drmessano | Climb every mountain? |
| 00:01.34 | interfaithquest | some reports claim port scanning is a way to punch thru between any two peers.. and is used by skype now days |
| 00:02.10 | interfaithquest | if a punch thru server is launched for all to use.. that may spark the fire |
| 00:02.56 | drmessano | Skype doesnt "port scan and punch holes" |
| 00:03.01 | drmessano | Theres a LOT more to it than that |
| 00:03.08 | drmessano | Maybe you need a good book on NAT |
| 00:03.55 | interfaithquest | well yes, ,they do many tricks, however port scanning seems to solve the symmetric nat barrier ,several sites claim as much |
| 00:04.23 | interfaithquest | i am inching thru a report online that details the skype protocol |
| 00:05.08 | interfaithquest | anyway once a punch thru engine is launched.. the clients simply slave away, they dont need to be smart |
| 00:05.51 | drmessano | Does the punch thru engine run on water? |
| 00:05.53 | interfaithquest | the java punch thru engine can be upgraded to take on the remaining 1% nat brick houses |
| 00:06.15 | coppice | the size of the internet means you'll find sites claiming anything you can think of |
| 00:06.16 | interfaithquest | have you seen www.blacklightpower.com |
| 00:06.58 | interfaithquest | they claim electronic power is the power of the future.. not fission |
| 00:07.06 | [TK]D-Fender | Skype goes so far as to masquerade as HTTP, HTTPS for ports, and even carry over as TCP (nice risk there). Basically Skype will break every rule or sane way it feels like to do its job |
| 00:07.28 | interfaithquest | for now though i will make the most of iax.. as most will be wiling to port forward 4569 |
| 00:07.42 | [TK]D-Fender | It's "protocol" is a complete breach of "protocol". |
| 00:07.47 | interfaithquest | yeah i was thinking of spoofing.. to jump over NAT |
| 00:08.05 | interfaithquest | fake the 'from' address/port |
| 00:08.11 | interfaithquest | dirty punch |
| 00:08.24 | [TK]D-Fender | kicks interfaithquest in the nads |
| 00:08.32 | interfaithquest | ha ha |
| 00:08.40 | coppice | [TK]D-Fender: in a world of wacky firewalls breaking every rule in the book, you gotta fight back |
| 00:09.02 | [TK]D-Fender | coppice: Its sad when other aspire to the crap others resort to ;) |
| 00:09.08 | interfaithquest | as it is though iax seems to have 90% of the nat problem solved, so enhancing that maybe the starting point |
| 00:09.08 | drmessano | You wanna talk about limiting communication.. basing your FOSSLICUP on IAX2 is umm limiting at best |
| 00:10.04 | coppice | iax has exactly the same issues as SIP, and doesn't usually solve them as well as most practical SIP implementations |
| 00:10.08 | drmessano | SIP has the NAT problem solved too |
| 00:10.15 | interfaithquest | XMPP /jingle is claimed to be 'state of the art' by google |
| 00:10.20 | drmessano | Open the ports, config 3 lines in asterisk |
| 00:11.02 | drmessano | Only difference between IAX2 and SIP.. the 3 lines in Asterisk |
| 00:11.03 | interfaithquest | STUN server will not conquer symmetric NAT |
| 00:11.45 | interfaithquest | gtalk seems way out infront of any SIP solution for global peer to peer open system |
| 00:12.09 | drmessano | Oh? |
| 00:12.13 | interfaithquest | though sip has great momentum.. with the big boys on it |
| 00:12.17 | drmessano | I thought a proxy was out of the question |
| 00:12.20 | drmessano | So how does that work? |
| 00:12.41 | interfaithquest | even skype has a location server |
| 00:12.51 | drmessano | Ok |
| 00:13.04 | drmessano | So Jingle and SIP have the same problem for clients |
| 00:13.10 | drmessano | A central server |
| 00:13.27 | WimpMan | The one called DNS? |
| 00:13.33 | drmessano | heh no |
| 00:13.35 | drmessano | That too |
| 00:13.48 | drmessano | Gtalk client needs the Jabber server to communicate |
| 00:14.01 | drmessano | SIP client, a pure client, needs a server |
| 00:14.05 | interfaithquest | so which one is better at scaling to millions of users ? |
| 00:14.24 | WimpMan | SIP, IAX and even H323 don't NEED anything more than DNS. |
| 00:14.32 | drmessano | Either one, theres no scaling limit to DNS |
| 00:14.41 | WimpMan | What's a SIP client supposed to be? |
| 00:14.45 | drmessano | I can run a SIP server or an XMPP server |
| 00:14.52 | drmessano | WimpMan: A softphone |
| 00:14.56 | drmessano | WimpMan: An ATA |
| 00:14.59 | drmessano | WimpMan: A phone |
| 00:15.04 | interfaithquest | so today is there a sip based instant messenger for millions on line now ? |
| 00:15.05 | drmessano | No ports open, ETC |
| 00:15.09 | drmessano | Just like a Gtalk client |
| 00:15.17 | WimpMan | A softphone is not just a client, but a server as well. |
| 00:15.43 | drmessano | Jesus Christ |
| 00:15.44 | WimpMan | Maybe not the well known one. |
| 00:16.06 | drmessano | Hes comparing to GTALK.. and the point was a SIP CLIENT is as useless behind a NAT with identify as GTALK, and vice versa |
| 00:16.21 | drmessano | identity |
| 00:16.53 | WimpMan | Anything behind NAT is useless unless you do something about it. That's not even specific to voip. |
| 00:16.54 | drmessano | You dont just connect an ATA to the world and become sip://joe@domain.com |
| 00:17.15 | WimpMan | But you can. |
| 00:17.19 | drmessano | Yes, well aware of that |
| 00:17.45 | drmessano | Not without ports open and the ATA having some sense of outside IP |
| 00:18.15 | WimpMan | Sure |
| 00:18.34 | drmessano | Anyway |
| 00:18.49 | interfaithquest | hey its good see a little excitement :) |
| 00:18.51 | drmessano | Gtalk still needs a central server |
| 00:18.56 | WimpMan | But what's the point? That's a firewall/nat/whatevernetworking problem. |
| 00:19.15 | drmessano | WimpMan: Scroll the fuck up |
| 00:19.28 | drmessano | WimpMan: Youre arguing over a point that isnt even needing to be made or relevant |
| 00:19.53 | WimpMan | Tha't indeed the impression I got. |
| 00:20.07 | drmessano | Well, you havent been reading then |
| 00:20.34 | interfaithquest | anyway there is a huge potential for a global peer to peer network, the telcos fear that |
| 00:20.39 | WimpMan | It just doesn't make much sense. |
| 00:20.59 | interfaithquest | the days of the pstn are numbered |
| 00:21.20 | interfaithquest | the devil is on the run |
| 00:21.32 | WimpMan | The PSTN is already shrinking at a considerable rate. |
| 00:21.45 | drmessano | HA |
| 00:21.50 | drmessano | Not really, no |
| 00:21.58 | WimpMan | It is. |
| 00:22.07 | [TK]D-Fender | interfaithquest: they hardly fear it. Lack of guaranteed Net Neutrality, ISP meddling and the world's inability to pick any kind of standard and even have those that do do so reliably make it a moot point |
| 00:22.22 | interfaithquest | when the global universal full duplex internet audio arrives.. all big companies will go for it on the web |
| 00:22.27 | drmessano | WimpMan: Ever heard of cellphones? |
| 00:22.32 | drmessano | WimpMan: Really man |
| 00:22.34 | WimpMan | Many telcos won't do it any more. If yo order a phone line they give you DSL and an ATA. |
| 00:22.42 | [TK]D-Fender | interfaithquest: Who is going to create and protect the standard? And ENFORCE it? |
| 00:22.48 | [TK]D-Fender | interfaithquest: This is a pipe dream |
| 00:23.02 | interfaithquest | the internet was a dream back in 1985 |
| 00:23.05 | drmessano | oh god |
| 00:23.09 | [TK]D-Fender | interfaithquest: Why do you think analog has stuck around so long... you just can't get rid of this shit |
| 00:23.24 | [TK]D-Fender | interfaithquest: No, the internet was Arpanet :0 |
| 00:23.39 | WimpMan | Yes, Cellphones are still going. But with mobile internet they will go the same way in a couple of years. |
| 00:23.46 | drmessano | An ATA from your telco is still PSTN.. its not an analog line, but its no more "the open neutral voip network" than a cordless phone is freedom from PSTN wires |
| 00:23.51 | [TK]D-Fender | interfaithquest: And did the internet replace snail-mail? Will it ever? |
| 00:24.22 | interfaithquest | qwest gives 20meg download and .5 meg upload.. THEY FEAR FULL DUPLEX NETWORKING |
| 00:24.33 | drmessano | lol |
| 00:24.48 | interfaithquest | wimax may move in over the established lords |
| 00:25.02 | drmessano | No, they fear that clients uploading puts more of a strain on the network than downloads do.. |
| 00:25.04 | [TK]D-Fender | interfaithquest: Each consumer's upstream is more than enough to suport VoIP, etc for the normal stuff.... frankly, WHO CARES |
| 00:25.23 | drmessano | and most people dont need the large upload capacity |
| 00:25.24 | WimpMan | drmessano: What's PSTn about getting an ATA? People are becoming aware of the fact thay can just go and chose another SIP server than their Telcos one. |
| 00:25.40 | interfaithquest | the mind will adapt. .and use all that is available.. |
| 00:25.54 | [TK]D-Fender | WimpMan: And where does another SIP server terminate to? the PSTN <--- |
| 00:26.04 | [TK]D-Fender | WimpMan: Welcome back to right where you started. |
| 00:26.15 | WimpMan | [TK]D-Fender: That's just an intermediate. |
| 00:26.16 | [TK]D-Fender | WimpMan: the lowest common denomitor is still in the way |
| 00:26.22 | JymmmEMC | Y'all are full of shit! CONNECT 2400 FTW!!! |
| 00:26.28 | drmessano | WimpMan: Last time I checked, the calls I make over a VoIP provider to any other person on the planet go over the PSTN |
| 00:26.33 | [TK]D-Fender | WimpMan: You are never elimiating it, you are just finding more stuff to shove in the middle |
| 00:26.42 | drmessano | [TK]D-Fender: EXACTLY |
| 00:27.03 | [TK]D-Fender | drmessano: people is DUM. D-U-M dumb! |
| 00:27.03 | WimpMan | You can use URL now. And you can use your e-mail address as phone number as well. That's where the thing gets interesting. Universal addresses. |
| 00:27.15 | drmessano | If they replaced your two wire analog connection with Fiber from your back door to a SLC96 down the street, do you have FIBERPHONE? No, not really |
| 00:27.15 | JymmmEMC | FidoNet will rise up again!!! |
| 00:27.28 | [TK]D-Fender | JymmmEMC: Been there, done that... |
| 00:27.34 | drmessano | I can SIP URI dial a vonage user? |
| 00:27.46 | drmessano | They can SIP URI dial me? |
| 00:27.51 | [TK]D-Fender | drmessano: Sure.. if you like the "rejected" message they'll give you ;) |
| 00:27.55 | drmessano | lol |
| 00:28.08 | JymmmEMC | [TK]D-Fender: Hey, it's still going actually =) I've been thinking about tossing up a bbs with TradeWars =) |
| 00:28.21 | [TK]D-Fender | drmessano: See, NOBODY likes competition... thats what will stop worldwide adoption of anything else. |
| 00:28.38 | [TK]D-Fender | JymmmEMC: I have a few Telnet TW2002 links around at hoem... |
| 00:28.38 | Nugget | telnet is eeeeeeevil! |
| 00:28.42 | [TK]D-Fender | home* |
| 00:28.54 | [TK]D-Fender | kicks Nugget-bot in the nads |
| 00:29.01 | interfaithquest | i stirred up a hornets nest |
| 00:29.11 | [TK]D-Fender | JymmmEMC: I was a TW2002 God in my day... |
| 00:29.16 | drmessano | interfaithquest: Dont flatter yourself |
| 00:29.21 | interfaithquest | lol |
| 00:29.26 | drmessano | interfaithquest: You didnt stir anything.. you havent cause a revolution |
| 00:29.33 | JymmmEMC | [TK]D-Fender: I just kept blowing up ships =) |
| 00:29.47 | [TK]D-Fender | pushes interfaithquest out of the clouds and watches him plummet to his demise... |
| 00:29.54 | drmessano | exactly |
| 00:30.14 | drmessano | Like peer to peer calling has NEVER been debated in here before |
| 00:30.19 | interfaithquest | world peace is on the way.. with or without techno babble |
| 00:31.19 | interfaithquest | everyone needs a stake in it |
| 00:31.21 | RB2 | Has anyone attempted to get BLF working on the poly w/ the new 3.1 fw? |
| 00:31.34 | JymmmEMC | interfaithquest: I see you've rejected our reality and substituted your own. |
| 00:31.43 | [TK]D-Fender | RB2: Should be jsut the same |
| 00:32.10 | [TK]D-Fender | JymmmEMC: I'm going to get myself one of Adam's "I do my own stunts" t-shirts... |
| 00:33.10 | RB2 | [TK]D-Fender, at least on the 650, you couldn't do blf without an expansion module. Apparently with 3.1, you can. |
| 00:33.11 | JymmmEMC | [TK]D-Fender: LOL there ya go! One guys keeps pushing me to try and get a job there. Maybe I should *shrug* |
| 00:33.51 | drmessano | Going back to the earlier convo that so rudely became something else completely.. IAX2 is not better suited for being the "God protocol" than SIP is.. The problem lies in no person having any sort of identity with their connection. No end user cares about IP addresses. In the end you do need some association of user@host which means the host on your end becoming meaningful |
| 00:34.05 | [TK]D-Fender | RB2: BS, you could always do buddies ont he phone itself |
| 00:34.33 | [TK]D-Fender | RB2: just leae some line-keys free |
| 00:35.35 | drmessano | Now if you take into consideration services like gmail, hotmail, etc.. and throw in your directory access.. Youre a button click away from telling Gmail "detect my IP now", setting up a router that assigns "hottie17@gmail.com" to port on, and having your calls routed via some big cloud |
| 00:35.36 | interfaithquest | how about an X prize for voip |
| 00:35.47 | [TK]D-Fender | drmessano: the global telco system works because everybody charges each other. P2P will fail to replace because nobody wants to deal with the central authentication, etc it would require. |
| 00:36.00 | drmessano | Google ! |
| 00:36.04 | [TK]D-Fender | interfaithquest: Put. Down. The. Crack. Pipe. (c) JerJer |
| 00:36.10 | interfaithquest | lol |
| 00:36.16 | RB2 | [TK]D-Fender, yes you could add buddies. But, I was referring to the attendant xml element. Maybe I missed something, but from what I've read, it should add a blf. |
| 00:36.22 | drmessano | Really.. Have good handle the auth |
| 00:36.24 | drmessano | google |
| 00:36.37 | drmessano | Associate my IP via some Google client |
| 00:36.58 | [TK]D-Fender | RB2: .... jsut don't allocate all your line-keys to reg's & appearances and you can SEE youre BLF on the free keys on the phone itself |
| 00:36.59 | drmessano | Basically we need a Gtalk appliance |
| 00:37.01 | JymmmEMC | He, it's kinda funny.... you guys are all about moving forward in the digital world and such. While here I am with my ham radio and (newest toy) 36port serial concentrator that show up as /dev/ttyx under nix =) |
| 00:37.09 | interfaithquest | drmessano: exactly |
| 00:37.12 | JymmmEMC | ANALOG RULES! |
| 00:37.27 | drmessano | JymmmEMC: Ham radio is even old to hams |
| 00:37.36 | WimpMan | ... and we're at the same point as with dyndns? |
| 00:37.41 | drmessano | JymmmEMC: Who the hell WANTS to get on 75 meters |
| 00:37.50 | drmessano | WimpMan: No, IPV6 or static IPs |
| 00:37.58 | JymmmEMC | drmessano: Nah, I have APRS + GPS connected up =) |
| 00:38.03 | drmessano | WimpMan: Dynamic IPs are only used for extortion |
| 00:38.10 | interfaithquest | with google video released, can a low cost google video phone be far away? |
| 00:38.13 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 00:38.13 | *** mode/#asterisk [+o russellb] by ChanServ |
| 00:38.17 | drmessano | JymmmEMC: I was on APRS about 12 years ago.. not much since |
| 00:38.27 | WimpMan | Everyone tells me so, bu I bet at least some ISPs will find ways to make it dynamic. |
| 00:38.30 | drmessano | JymmmEMC: Used to teach an APRS class |
| 00:38.47 | drmessano | WimpMan: For extortion |
| 00:39.02 | RB2 | [TK]D-Fender, by enabling bw in the contacts directory or by adding the attendant directive? I can get the blf to show up by adding a buddy watch, but it never does anything. |
| 00:39.19 | drmessano | WimpMan: did you know Comcast dropped their rates for business class service by 40%? and they TRIPLED the price of a static IP? |
| 00:39.24 | JymmmEMC | drmessano: I was trying to get the county OES to use APRS, but it's hard to teach an old dog new tricks |
| 00:39.24 | WimpMan | drmessano: Sure. |
| 00:39.47 | drmessano | $59 for the former $99 service.. and $15 for the former $5 static |
| 00:39.53 | drmessano | Its about money |
| 00:40.05 | drmessano | Not about cost of them supplying a static |
| 00:40.27 | WimpMan | That's why I don't believe in that static-only story. |
| 00:40.59 | interfaithquest | obama should let fiber MAN into each city |
| 00:41.13 | interfaithquest | not superman fiberman METROPOLITAN AREA NETWORK |
| 00:41.27 | JymmmEMC | Mmmmmm fiber to the desktop! |
| 00:41.30 | interfaithquest | CRUSH the arrogan telcos |
| 00:41.35 | interfaithquest | lol |
| 00:41.43 | JymmmEMC | 4G FTW! |
| 00:41.54 | interfaithquest | 1 gig full duplex |
| 00:42.05 | [TK]D-Fender | RB2: yOU'VE DONE SOMETHING WRONG ALONG THE WAY THEN. |
| 00:42.09 | drmessano | WimpMan: Static IPs are going to have to happen.. they wont be able to continue having to spend money to manage static addresses, which now cost them to seperate out and sell |
| 00:42.24 | interfaithquest | most cities have fiber along all highways for DOT service.. just roll it out |
| 00:42.51 | interfaithquest | what is stopping ipv6 from emerging ? |
| 00:42.55 | drmessano | or dealing with the excess traffic of renewals, etc |
| 00:43.02 | WimpMan | I'm sure they will still want to charge extra even if it costs them less. |
| 00:43.03 | drmessano | IPv6 ha |
| 00:44.03 | drmessano | WimpMan: Comcast already has to deal with too many devices requesting IP renewals.. it wont last |
| 00:44.11 | RB2 | [TK]D-Fender, ok, thanks. I'll check everything over. Just one more question, should I use the attendant setting w/ asterisk or not? |
| 00:44.12 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 00:44.27 | WimpMan | Take a look at POTS vs ISDN. ISDN costs less than PSTN but is sold for more. |
| 00:44.27 | drmessano | The number of IP addressible devices is going to explode in Feb |
| 00:44.53 | drmessano | WimpMan: Thats not true |
| 00:45.04 | interfaithquest | perhaps networking will be the engine of the new economy.. as this one has tanked |
| 00:45.04 | [TK]D-Fender | RB2: what "attendant setting"? There is nothing spcial to the console.. it just spills over directory entries... |
| 00:45.18 | interfaithquest | obama.. fiber to every home now ! |
| 00:45.24 | WimpMan | drmessano: It is |
| 00:45.31 | interfaithquest | fiberman to the resuce |
| 00:45.32 | drmessano | Non-switched pair back to the CO, cost of the cards at the CO.. |
| 00:45.41 | drmessano | ISDN costs more |
| 00:45.49 | [TK]D-Fender | interfaithquest: Who cares about the MEDIUM when the MESSAGE is bullshit? |
| 00:45.58 | [TK]D-Fender | interfaithquest: You still keep missing the point :) |
| 00:46.10 | interfaithquest | i maybe new at this |
| 00:46.12 | WimpMan | Before the SIP over DSL hack Telcos have given ISDN lines with TAs to customers ordering POTS because it's cheaper. |
| 00:46.38 | drmessano | Apparently youre not in the US |
| 00:46.45 | interfaithquest | fiber to every home will generate jobs |
| 00:46.51 | WimpMan | Correct |
| 00:46.52 | drmessano | Because in AT&T land, they use copper |
| 00:47.02 | [TK]D-Fender | interfaithquest: And take away others |
| 00:47.04 | drmessano | and in AT&T land, ISDN costs MORE |
| 00:47.05 | interfaithquest | fiber to every home will bring networking life to the world |
| 00:47.17 | [TK]D-Fender | interfaithquest: There's never a trade-off in DreamLand is there? |
| 00:47.20 | interfaithquest | fiber to every home is the answer |
| 00:47.36 | [TK]D-Fender | interfaithquest: and the only response to that is.... 42 |
| 00:47.44 | drmessano | In the US, no telco is every gonna give someone DSL and an ATA in place of an order for an analog line |
| 00:47.46 | drmessano | No sir |
| 00:47.47 | interfaithquest | as in 42nd street ? |
| 00:47.53 | drmessano | ever* |
| 00:48.10 | interfaithquest | porn has a way of rotting the soul of every human endeavor.. phone sex whatever |
| 00:48.18 | JymmmEMC | market st |
| 00:48.19 | drmessano | DSL uses a copper pair.. thats not cheaper |
| 00:48.27 | WimpMan | Well, tha's just proof of the theory that time and place are interconnected :-) |
| 00:48.48 | WimpMan | It's cheaper to maintain. |
| 00:48.59 | drmessano | No |
| 00:49.03 | drmessano | Its the same copper |
| 00:49.10 | interfaithquest | fiber to the home will save a lot of gas |
| 00:49.28 | interfaithquest | full duplex gigabit ethernet |
| 00:49.34 | WimpMan | You have a device at the customers end that is capable of doing line tests in an ojvetive way. |
| 00:49.39 | drmessano | interfaithquest: and delivery of fast food vs home cooked meals will generate more gas |
| 00:49.42 | interfaithquest | is there a shortage of ip's for that ? |
| 00:49.55 | interfaithquest | lol |
| 00:50.21 | drmessano | WimpMan: Yeah, youve now added an extra device to the line.. How is that LESS? and they can do objective tests of the line |
| 00:50.24 | drmessano | They do it all the time |
| 00:50.47 | RB2 | [TK]D-Fender, there is an attendant setting as per the poly sip admin guide (page A-103 in the 3.0 manual). In 3.0, it only worked w/ expansion modules. In 3.1, it can be enabled without having one. I'll just research it some more. Thanks again. |
| 00:50.51 | drmessano | a DSLAM isn't free |
| 00:51.12 | WimpMan | The point is with DSL or ISDN you can check the line op to the terminating equipment at a mouseclik. And if that results in an ok, it's up to the customer. That saves big bucks. |
| 00:51.25 | drmessano | They can DO THAT NOW |
| 00:51.29 | drmessano | with a MOUSECLICK |
| 00:52.00 | drmessano | They can run a test from the CO and tell you anything you want to know about that copper pair |
| 00:52.10 | WimpMan | It's no where the same accuracy as with digital equipment. |
| 00:52.19 | drmessano | An ATA isnt gonna tell them shit |
| 00:52.33 | drmessano | Regged or not regged |
| 00:52.37 | drmessano | Working or not working |
| 00:52.46 | WimpMan | It doesn't have to. |
| 00:52.57 | [TK]D-Fender | \o/ - Budget file ^%#@$ing completed! And it only ate up my entire weekend! |
| 00:52.57 | drmessano | and they are using digital equipment to test the line |
| 00:52.58 | interfaithquest | now that the car companies have tanked.. what else is there but fiber to the home left to do here ? |
| 00:53.13 | [TK]D-Fender | \o/ - Budget file ^%#@$ing completed! And it only ate up my entire weekend!I'm done... SO very done... |
| 00:53.18 | [TK]D-Fender | I'm outta here... back in a bit... |
| 00:53.21 | drmessano | Have you ever seen the inside of a CO? Do you even know what capabilities modern linesman have? |
| 00:53.27 | WimpMan | All you want to know is if the Modem or NT has a good signal. Everythin eles is usually not the telcos business any more. |
| 00:53.32 | [TK]D-Fender | later all |
| 00:53.37 | interfaithquest | me too |
| 00:54.29 | WimpMan | Do you want to tell me you can make exact measurement with only one end of the wire? |
| 00:54.58 | WimpMan | Sure. Some faults can be found on one end only, but a full quality check sorely requires more. |
| 00:55.02 | drmessano | Yes actually |
| 00:55.12 | drmessano | They can run loop tests on the line for noise, etc |
| 00:55.20 | drmessano | They dont have one end of the line |
| 00:55.22 | drmessano | They have TWO |
| 00:55.30 | drmessano | One big circuit |
| 00:56.00 | WimpMan | And what is at the consumer end ot a POTS line? |
| 00:56.01 | drmessano | They can check CO > User > CO |
| 00:56.29 | drmessano | A closed relay |
| 00:56.34 | drmessano | A switch |
| 00:57.02 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:57.12 | drmessano | If its closed, we test for line condtions.. if its open, they have something off hook or theres a physical line problem that requires a service call |
| 00:57.37 | drmessano | They can run very accurate tests on that loop |
| 00:57.48 | WimpMan | But on just a short cuircuit loop you can not do everything you can with active equipment. |
| 00:57.52 | drmessano | What good is a modem or NT if the device is OFFLINE |
| 00:58.00 | drmessano | or getting inaccurate measurements |
| 00:58.01 | drmessano | or bad |
| 00:58.04 | drmessano | no use AT ALL |
| 00:58.18 | drmessano | WimpMan: There is active equipment on the line |
| 00:58.22 | drmessano | The Central office |
| 00:58.29 | drmessano | As I said to you before |
| 00:58.32 | drmessano | Its a BIG LOOP |
| 00:58.35 | drmessano | Not just ONE END |
| 00:58.46 | drmessano | They can check that WHOLE circuit |
| 00:58.51 | WimpMan | What? You put a CO in everyones home? |
| 00:58.56 | drmessano | Duh |
| 00:58.59 | drmessano | No |
| 00:59.03 | WimpMan | I said customer end. |
| 00:59.12 | drmessano | Do you understand basics of electrical circuits? |
| 00:59.27 | WimpMan | yes |
| 00:59.36 | WimpMan | And what about you? |
| 00:59.40 | drmessano | That pair with a shorted end is BIG LOOP |
| 00:59.47 | drmessano | Stop being an asshole |
| 00:59.52 | drmessano | Youre not even close on this |
| 01:00.29 | WimpMan | We're talking about a line carrying alternating currents. even more so different frequencies. |
| 01:00.32 | drmessano | If take both wires of that pair at the CO, I have a nice big closed loop to the customer |
| 01:00.35 | drmessano | I can run all I want on it |
| 01:00.43 | drmessano | No |
| 01:01.19 | drmessano | We're talking about a loop that of wire that any number of tests can be run on.. We can check for noise, look for breaks with a simple TDR |
| 01:01.21 | drmessano | Etc |
| 01:01.24 | drmessano | and they do |
| 01:02.10 | WimpMan | Yes, tha's what I referred to above, when I said some tests can be dome from only one end. |
| 01:02.19 | drmessano | Let me ask you a question |
| 01:02.28 | drmessano | If Take that pair from the customer |
| 01:02.31 | drmessano | If I Take that pair from the customer |
| 01:02.45 | drmessano | with it closed off at the other end.. all phones on hook |
| 01:02.47 | drmessano | and I drive to the CO |
| 01:03.11 | drmessano | and I take one end of that wire to the customer and plug it in a testing device |
| 01:03.26 | drmessano | and I take the other end and plug it into another tester |
| 01:03.51 | drmessano | and I make a 1 foot piece of 22 gauge and run it to the second terminal on both testers |
| 01:04.01 | drmessano | So not I have tester ======== tester with a pair between |
| 01:04.04 | drmessano | now* |
| 01:04.13 | drmessano | except one leg is 4 miles long |
| 01:04.16 | drmessano | the other is about 1 foot |
| 01:04.20 | drmessano | Can I not test that line? |
| 01:05.03 | WimpMan | You can do tests. Sure. But Not all or not at the same accuracy. |
| 01:05.08 | drmessano | No |
| 01:05.11 | drmessano | False |
| 01:05.27 | drmessano | I am testing the same 8 miles of copper (4 miles both ways) |
| 01:05.33 | drmessano | I shorted the far end |
| 01:05.39 | drmessano | and I put the SAME tester in line |
| 01:05.39 | WimpMan | Do you think you can get sensible frequency response measurements that way? |
| 01:05.44 | drmessano | yes |
| 01:05.52 | drmessano | No different than 4 miles both ways |
| 01:06.01 | WimpMan | That is definitely not the same. |
| 01:06.04 | drmessano | It is |
| 01:06.21 | drmessano | I can check for all the same line conditions |
| 01:06.29 | WimpMan | Well, it you think so I'd suggest you do some research on that topic. |
| 01:06.39 | drmessano | Naah, I have about 12 years experience |
| 01:06.42 | drmessano | Maybe you should |
| 01:07.28 | drmessano | Anyway.. My work is done here.. I suggest a good reading of basics of electrical circuits to start |
| 01:07.51 | WimpMan | So do I. |
| 01:08.36 | Guest85358 | . |
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| 01:09.09 | [TK]D-Fender | whee |
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| 01:29.27 | WimpMan | Ok, I need to correct that one. With basic electrical circuits, drmessano is off course right, But this is about signal transmission and that's where it no longer fits. |
| 01:29.36 | WimpMan | And now off to bed. |
| 01:29.49 | drmessano | and thats where youre wrong |
| 01:30.08 | drmessano | They can test signal transmission just fine, they're not jusy ohm'ing it out |
| 01:30.10 | drmessano | just* |
| 01:30.26 | drmessano | Just the same as having testers on "both ends" |
| 01:30.28 | JymmmEMC | ringing iyt out |
| 01:30.32 | drmessano | which they effectively do |
| 01:30.32 | JymmmEMC | ringing it out |
| 01:31.26 | drmessano | You would surprised what you can learn about a pair of wire by only having "one end" |
| 01:31.35 | drmessano | Its a pair, so you never really have "one end" |
| 01:32.25 | JymmmEMC | Yeah? Hook that pair to 220vac and see what happens =) |
| 01:32.46 | drmessano | Shorted or unshorted? |
| 01:32.59 | JymmmEMC | one way to find out =) |
| 01:33.01 | drmessano | unshorted, not a problem.. |
| 01:33.25 | drmessano | I've seen 220 run over Belden 291 before lol |
| 01:34.13 | jblack | wow the yen is strong. |
| 01:34.29 | drmessano | When it comes to running signals over wire.. theres nothing ma-bell can do to trump those running RF over copper |
| 01:34.35 | drmessano | Analog audio? please |
| 01:34.57 | kerx | what is a good open source session border controllers? I've never really seen any of these. |
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| 03:03.31 | adilwali | hello, i have a asterisk server running on UK, and my customers are on the US, i also have customers on the UK and i need them to talk to each other but the quality is really bad, because there is lot of latency between the countries... is there a way i could make this better? i use gsm/ulaw/alaw codecs... thanks |
| 03:04.05 | SkramX | are your service providers top notch? |
| 03:04.20 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 03:04.40 | SkramX | im not an expert, but perhaps have a server in the US too and have US-US and UK-UK calls route through their respective servers to make /those/ conversations better.. |
| 03:05.28 | adilwali | SkramX: i use teliax but only for pstn termination |
| 03:05.31 | *** join/#asterisk troubled (n=troubled@unaffiliated/troubled) |
| 03:05.51 | adilwali | SkramX: and how US-US and UK-UK would make these people talk to each other? |
| 03:06.26 | SkramX | im saying.. people in the US talking to people in the US would be better routed through the US than a server in the UK and then back to US |
| 03:06.32 | SkramX | idk |
| 03:08.36 | adilwali | the problem is that people in the US wants to talk with people in the UK |
| 03:09.17 | jer_ | find a better peer in either the UK or the US (or both) ... where latency is lower |
| 03:09.25 | SkramX | yeah |
| 03:09.40 | SkramX | that's really the only solution for the cross-continent communication |
| 03:09.45 | SkramX | not sure how good teliax is.. |
| 03:10.15 | jer_ | well he's only using teliax from his statement for pstn termination, that doesn't mean squat for his internet connection, which is what i was referring to |
| 03:10.39 | SkramX | ah yes. |
| 03:10.42 | jer_ | have the two server option yes, definitely... but also make sure those two servers are behind peers with lower latency to the UK from the US, and likewise |
| 03:11.35 | jer_ | inter-country communication isn't really terrible between countries like, Canada and the US, or any scandinavian country and another scandinavian country... just as an example |
| 03:11.57 | jer_ | but once you start crossing oceans, be prepared to have to find the right nsp to peer with |
| 03:12.04 | jer_ | and be prepared to pay through the roof for it |
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| 03:12.44 | SkramX | heh |
| 03:13.15 | JymmmEMC | adilwali: you in/near argentina? |
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| 03:14.37 | adilwali | JymmmEMC: yes |
| 03:15.08 | JymmmEMC | adilwali: That's the problem it seems. huge latency at that point directly. |
| 03:15.29 | JymmmEMC | almsot 3x more than the previous hop |
| 03:16.05 | adilwali | wont codecs like g729 solve that? |
| 03:17.01 | [TK]D-Fender | adilwali: Codecs don't make your latency any better... |
| 03:17.05 | JymmmEMC | you're talking 225ms latency at your end point. |
| 03:17.08 | [TK]D-Fender | adilwali: delay is delay |
| 03:17.41 | JymmmEMC | adilwali: consider a different local connection if nothing more than a test |
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| 03:24.48 | adilwali | JymmmEMC: yeah, the problem is that with a local connection there will still be delay between the US <-> UK |
| 03:24.57 | adilwali | maybe a TIER-1 connection between the two continents will be good? |
| 03:25.29 | JymmmEMC | even from .nl to you is bad is the problem =) |
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| 03:33.38 | adilwali | can i use teliax to connect my 200 users to it, or i will need like 200 accounts for that? maybe they have better connection than what i have |
| 03:33.45 | adilwali | i mean connect directly |
| 03:39.57 | x86 | voip across the internet is suck |
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| 03:40.43 | adilwali | voip across what then is better? |
| 03:41.00 | SkramX | two cans and string! |
| 03:46.21 | x86 | adilwali: point to point circuit or LAN is the only way to do VoIP |
| 03:48.01 | ManxPower | Well the only way to do VoIP well is point-to-point or LAN, you can do unreliable VoIP over the internet. |
| 03:51.08 | ManxPower | You really can't tell the difference between standard PSTN and VoIP over a link with real QoS. |
| 03:52.41 | *** join/#asterisk AsterNoob (n=AsterNew@65.29.110.49) |
| 03:55.20 | AsterNoob | Quick question, because I am apparently missing something... I have a grandstream 2000 phone which has 4 extensions setup on my desk, and two more on my employees desks. I'd like it to ring the first available extension on each phone only, for example, if I am on line 1 and a call comes in, then ring line 2, if I place line one on hold and am talking to line 2 and another call comes in ring line 3, etc.. and do that for all phones.. The closest I can |
| 03:56.45 | [TK]D-Fender | AsterNoob: this is a phone config issue. |
| 03:56.56 | adilwali | ManxPower: so in other words, is impossible to do VoIP in a reliable way over the public internet? |
| 03:56.58 | [TK]D-Fender | AsterNoob: You should stop thinking of those as "lines" so much as "appearances" |
| 03:57.50 | [TK]D-Fender | AsterNoob: most phones that can support multiple simultaneous calls can be set up to cascade naturally like that |
| 03:58.32 | AsterNoob | I was thinking of them as apparences, so... it's not handled by asterisk then? wierd.. I would have thought for sure some sort of hunting was needed.. |
| 03:58.41 | AsterNoob | Ok, I'll look into the telephone configuration more |
| 03:59.07 | [TK]D-Fender | AsterNoob: They should all be associated with a single registation |
| 04:03.10 | SkramX | [TK]D-Fender - does that apply to all phones? including Cisco SCCP ones? |
| 04:03.25 | SkramX | because I have several which I want to have two lines and will be setting them up for agent queueing ... |
| 04:03.27 | [TK]D-Fender | SkramX: SCCP is another ball-game |
| 04:03.36 | SkramX | uh oh. |
| 04:03.37 | [TK]D-Fender | SkramX: much like MGCP |
| 04:03.41 | SkramX | :( |
| 04:03.55 | SkramX | if i keep the phones one-line.. do I just do regular agent queueing in *? |
| 04:04.00 | AsterNoob | Hmm, currently I have each "line" (because thats what Grandstream calls them) setup with their own extension # (ie: 101 for line 1, 201 for line 2, 301 for line 3, and 401 for line 4) and was trying to find someway to make it hunt to the next line button when the previous one is busy |
| 04:04.24 | [TK]D-Fender | AsterNoob: you should not be setting them up as separate reg's, but rather as the SAME reg |
| 04:04.36 | AsterNoob | ah |
| 04:04.45 | AsterNoob | actually, that kind of makes sense. ;) |
| 04:04.51 | AsterNoob | lemme try that. |
| 04:08.30 | AsterNoob | hmm.. it apparently doesn't like that. |
| 04:09.42 | [TK]D-Fender | AsterNoob: Go read up on how to properly configure it so it spans lines. |
| 04:09.55 | [TK]D-Fender | AsterNoob: it might just use them all if you only specify the first |
| 04:10.48 | AsterNoob | I'll see what I can find, but it appears that Asterisk is not letting me register twice |
| 04:10.54 | AsterNoob | It's only seeing the last one |
| 04:10.57 | [TK]D-Fender | AsterNoob: it shouldn't be |
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| 04:11.16 | [TK]D-Fender | AsterNoob: so knock off the other reg's and see if it spans naturally.. you may have an option to take in the phone for this |
| 04:12.04 | AsterNoob | I've looked for the option, and the reason I tried registering the others in the first place was because it wasn't spanning |
| 04:12.41 | AsterNoob | I didn't get a manual with the phone, but I just found one online, checking to see if it says anything... Like I said, I assumed it was an asterisk thing originally. |
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| 04:15.22 | [TK]D-Fender | AsterNoob: it isn't |
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| 04:16.21 | jaytee | god I hate flying |
| 04:16.44 | jaytee | my flight from O'Hare got cancelled and I had to wait till 8pm for another flight. |
| 04:17.15 | jaytee | then when I got here they'd opened the new air terminal and the cab ride back to my house was 23 bucks. getting to the airport last week it was 9.50 |
| 04:21.36 | SkramX | lame indeed |
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| 04:24.04 | jaytee | the new air terminal is really "spiffy". it even has a Pacific Outfitters and Brooks Brothers stores for those 1.5% of the population that won't be on government relief and actually able to fly and buy things in 2009. |
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| 04:29.16 | [TK]D-Fender | jaytee: Considering how many people go through there on a daily basis, just think what that 1.5% represents... |
| 04:29.33 | jaytee | a few hundred thou a week |
| 04:30.59 | drmessano | hmm |
| 04:31.26 | [TK]D-Fender | just a little food for thought... |
| 04:32.25 | jaytee | [TK]D-Fender, yeah, it's true and it is a very nicely designed terminal. I've just got the post flight delayed flight PMS crabby poopy pants grouchy kick a puppy blues |
| 04:32.50 | jaytee | but it's nice to be home |
| 04:32.51 | [TK]D-Fender | jaytee: don't let Katty hear you saying that! |
| 04:33.17 | jaytee | [TK]D-Fender, I think she knows I'd never actually kick a puppy |
| 04:34.08 | jaytee | the only thing I'll hurt or kill is a spider or a roach in my apartment. if it's outside I'll ignore it. I even walked around some kind of roach in Huntsville the other day. |
| 04:34.28 | jaytee | and ants inside too |
| 04:34.34 | drmessano | HA |
| 04:34.40 | jaytee | if they're outside I'll leave em be |
| 04:34.45 | SkramX | jaytee - which airport? |
| 04:34.52 | jaytee | Indianapolis |
| 04:34.56 | SkramX | k |
| 04:35.00 | SkramX | never been |
| 04:35.03 | drmessano | [Digg] 20 Things to do after installing Ubuntu Linux <--- 5. Throw out your condoms, buy some new comic books (trust me) |
| 04:35.13 | SkramX | lol |
| 04:35.28 | [TK]D-Fender | drmessano: Finally, a credible review! |
| 04:35.44 | jaytee | and wireless worked awesome in Huntsville at the airport on my lappy but in O'Hare it wouldn't work even though it said it was connected and excellent signal strength |
| 04:36.22 | drmessano | lol |
| 04:36.41 | drmessano | Difference between north and south |
| 04:36.54 | drmessano | In the south, we're too ignerent to have camputers |
| 04:36.58 | drmessano | So free Wifi |
| 04:37.43 | jaytee | it's supposed to be free in O'Hare and they keep saying that over the PA but you just can't get to the web. |
| 04:37.45 | drmessano | At O'Hare, they had Cisco AerPort WAPs and could detect you couldnt afford the $75 a min for internet, so they blocked your MAC |
| 04:37.47 | orkid | haha |
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| 04:38.17 | drmessano | Courtesty of your RFID enabled MasterCard |
| 04:38.29 | drmessano | RFID is gonna RUIN us |
| 04:38.52 | drmessano | Imagine women being able to check YOU credit card balance on a first date |
| 04:38.56 | drmessano | YOUR* |
| 04:38.57 | jaytee | makes note to tell the DHS that they need to "chip" drmessano soon. |
| 04:39.28 | SkramX | haha |
| 04:39.33 | drmessano | "My Blackberry just told me you're a loser" |
| 04:39.37 | drmessano | :( SAD FACE |
| 04:40.17 | drmessano | Blackberry.. So evil, yet so... hang on, getting an email.. brb |
| 04:40.49 | orkid | the kind of women you dont wanna date |
| 04:41.03 | jaytee | i read where they're going to block Obama from using a Crackberry after he gets sworn in. They'll let him have a laptop in the Oval Office and he'll be the first president to do that (since he the first person we've elected that can figure out how to use one) |
| 04:41.07 | drmessano | As if I seriously dont want the same thing |
| 04:41.22 | drmessano | Who wouldnt want a deadbeat detector? |
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| 04:41.54 | SkramX | jaytee - i read that recently as well |
| 04:41.55 | drmessano | Relationships are about love.... and business.. and if you just think with your heart, you'll be out on your ass |
| 04:42.08 | drmessano | So a deadbeat detector, YAY |
| 04:42.13 | SkramX | they dont want him to have a personal phone - everything is considered business and it's easy to say something wrong/risque i guess |
| 04:42.30 | drmessano | Every communication he has is public record.. |
| 04:42.35 | drmessano | Thats the crux of it |
| 04:43.05 | drmessano | Apparently GWB had to give up his AOL account too.. reportedly sent one last message to all his friends just before being sworn in |
| 04:43.13 | jaytee | every potential mate should have a RFID chip embedded that will allow you to poll it for things like STD's, annual maintenance fees, etc. |
| 04:44.25 | drmessano | From: allaboutoil37@aol.com, To: FreindsGrupList, Subject: Hey YAll, Body: Hey Ylal tomorow i gots to go, so holdma beer and wach thsis! |
| 04:44.58 | jaytee | hehe |
| 04:45.19 | drmessano | PS: who wants sum oil! |
| 04:45.43 | drmessano | The same man who nearly lost a battle.... with a pretzel |
| 04:51.06 | jaytee | hey....some pretzels can put up a pretty good fight |
| 04:52.05 | drmessano | So can the Iraqi people |
| 04:56.18 | drmessano | Sorry "Those damn insurgents" |
| 04:56.45 | SkramX | heh |
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| 04:56.51 | SkramX | sigh. I live in Tx. |
| 04:57.28 | jaytee | yeah? big insurgent problem in Tx.? |
| 04:57.40 | SkramX | big texan problem here ;) |
| 04:57.59 | SkramX | oh crap. there's a cowboy with a shotgun behind me |
| 04:58.02 | SkramX | bbiab |
| 04:58.04 | drmessano | Reminds me of Southpark |
| 04:58.24 | drmessano | That its ok to kill something if it attacks you first |
| 04:58.38 | drmessano | I wonder if Bush yelled "LOOK OUT DICK, THEY'RE COMIN RIGHT AT US" |
| 04:58.41 | SkramX | yeapp |
| 05:01.51 | AsterNoob | ok, finally got it to work |
| 05:02.08 | AsterNoob | Apparently I needed to enable call waiting (*51) on the phone |
| 05:02.20 | AsterNoob | easy enough, thanks for pointing me in the right direction |
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| 05:15.56 | jaytee | g'nite all |
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| 06:26.42 | *** join/#asterisk monstertruck (n=monstert@174.149.142.102) |
| 06:28.07 | monstertruck | hey guys and girls, does anyone know of a way to get a remote phone jack that can connect wirelessly to the phone line? |
| 06:28.27 | monstertruck | i know this question probably doesnt belong here, but cant think of a better place and google is not helping ... |
| 06:29.34 | monstertruck | and i mean wireless as in radiofrequency, not through powerlines |
| 06:31.40 | fakhir | an ATA connected to a WiFi bridge? |
| 06:34.18 | monstertruck | no, this is for a regular phone |
| 06:34.41 | monstertruck | well, yeah, that makes sense |
| 06:35.07 | fakhir | :) |
| 06:38.27 | drmessano | Ive done that before |
| 06:38.38 | drmessano | Actually |
| 06:38.41 | drmessano | I have done worse |
| 06:42.44 | monstertruck | does it work well? |
| 06:43.04 | monstertruck | would have to be 2 ata's |
| 06:43.10 | monstertruck | one fxo on the line end |
| 06:43.13 | drmessano | heh |
| 06:43.23 | monstertruck | and one fxs on the end i need the jack |
| 06:43.25 | drmessano | Are you even using Asterisk here? |
| 06:43.36 | monstertruck | yes, on the end where i need the jack |
| 06:43.44 | drmessano | I see |
| 06:43.45 | *** join/#asterisk grEvenX (n=even@apb9hb.ip.ssc.net) |
| 06:44.12 | drmessano | So you need the phone line from --------------------> far away |
| 06:44.17 | drmessano | to asterisk |
| 06:44.22 | drmessano | So what you need is ONE ATA |
| 06:44.23 | monstertruck | yes |
| 06:44.31 | drmessano | with a network connection back to * |
| 06:44.32 | *** join/#asterisk kerx (n=prepro@adsl-68-125-33-231.dsl.irvnca.pacbell.net) |
| 06:44.35 | drmessano | an FXO ata |
| 06:44.40 | drmessano | connected to the PSTN |
| 06:44.50 | monstertruck | and a bridge between those two |
| 06:44.57 | drmessano | and an IP connection back to asterisk over wifi |
| 06:46.47 | monstertruck | i hadnt thought of that solution, and thats probably what i'll end up doing |
| 06:47.12 | drmessano | I have a friend of mine.. check this out.. |
| 06:47.14 | monstertruck | im surprised though, that something so simple as a cordless r11 jack doesnt exist as a product |
| 06:47.19 | drmessano | USB Wifi adapter on his PC |
| 06:47.30 | drmessano | 70 feet to the router |
| 06:47.35 | drmessano | On his PC, two NICs |
| 06:47.43 | drmessano | ATA plugged into one, bridged to the WIFI |
| 06:47.57 | drmessano | Works GREAT, even on a crap 700MHZ PC running XP |
| 06:47.58 | monstertruck | hey |
| 06:48.15 | monstertruck | hey now, thats even better and probably cheaper |
| 06:48.27 | drmessano | except if the PC is turned off |
| 06:48.28 | drmessano | and no |
| 06:48.37 | drmessano | The rj11 remote jack stuff sucks |
| 06:48.52 | drmessano | I would go with the bridge |
| 06:48.56 | drmessano | and the FXO ATA |
| 06:49.13 | monstertruck | yeah, one fxo ata is all i really need |
| 06:49.27 | monstertruck | and a wifi card on the * side |
| 06:49.44 | *** join/#asterisk shinao1 (n=shinao1@62.173.48.42) |
| 06:49.52 | monstertruck | plus no need to send stuff overseas |
| 06:49.55 | monstertruck | i love it |
| 06:50.20 | drmessano | Ah there you go |
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| 07:08.34 | orkid | aha, i have figured out my ISP is to blame for packet loss. other ISP's login works 0.0% packet loss! |
| 07:08.49 | orkid | so something with them or their routing |
| 07:11.02 | *** join/#asterisk joobie (n=joobie@joobie.org) |
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| 07:14.52 | yidiyuehan | hi, anybody knows how to set DYNAMIC_FEATURES globally for all the calls? like under [globals] DYNAMIC_FEATURES = testfeature#test#test2 ? |
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| 07:29.47 | ghostknife | I am having trouble dialing out on my Zap trunks. Whenever I dial nothing happens, the line just stays quiet and the log show: Zap/1-1 answered SIP/16-082535e8 |
| 07:29.55 | ghostknife | Any ideas what I can do to debug this? |
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| 08:36.42 | dynaguy | whois |
| 08:36.45 | getsmart | any experience using nikotel.com accounts with *? |
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| 08:41.40 | *** join/#asterisk TrentCreek (n=kvirc@adsl-75-14-1-196.dsl.hrlntx.sbcglobal.net) |
| 08:43.10 | tzafrir_laptop | A bit cheap, but nice: http://notnews.today.com/2008/09/22/free-software-foundation-announces-gnuphone/ |
| 08:44.23 | drmessano | HAHAHAHHA |
| 08:45.00 | tzafrir_laptop | drmessano, you mean you don't have one already? |
| 08:46.55 | drmessano | dial voice +1-555-1212 ântwk verizon âprot cdma2000 âssh-version 2 -a -l -q -9 -b -k -K 14 -x |
| 08:46.59 | drmessano | that.. pwns |
| 08:47.38 | drmessano | Naw, still waiting for the GPL3Phone... I ordered one, but apparently I am not allowed to actually own it or use it |
| 08:48.55 | drmessano | Oh god |
| 08:48.55 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
| 08:49.18 | drmessano | They should have used "Diggnerd Trollbait" as the name of the dev |
| 08:49.23 | drmessano | âReally, weâre not out to destroy Apple; that will just be a completely unintentional side effect.â |
| 08:50.10 | frozty_sa | that's a modified quote. grrrrrrr |
| 08:50.16 | frozty_sa | troll the twat who said it |
| 08:50.34 | drmessano | what? |
| 08:51.28 | frozty_sa | the original quote was by torvalds, about microsoft |
| 08:52.10 | mvanbaak | indeed |
| 08:52.16 | drmessano | Yeah, and it was used in a notnews article, posted about 15 lines up |
| 08:52.55 | drmessano | Now tell the nice guy at the FSF you didnt mean to call their hotline this late, and put the phone down |
| 08:53.02 | drmessano | ty :) |
| 08:53.35 | mvanbaak | I like the comment "you can do: dial +1-555-*" |
| 08:53.42 | *** join/#asterisk stoffell (n=kristof@48.181-201-80.adsl-dyn.isp.belgacom.be) |
| 08:54.29 | drmessano | Obviously it's not running asterisk.. there would be 8 or 9 more cli options involved |
| 08:54.42 | drmessano | and they would all be deprecated, or at least warn of deprecation |
| 08:55.34 | drmessano | "Warning: The protocol 'SIP' will likely be deprecated in the future" |
| 08:56.14 | drmessano | "... please use the protocol 'IAX2' instead" |
| 08:56.15 | drmessano | heh |
| 08:57.26 | *** join/#asterisk sosperec (n=david@office.axpnet.com) |
| 08:57.29 | sosperec | hello |
| 08:58.40 | joat | at least it comes with it's own bugzilla |
| 08:59.46 | TrentCreek | I have never been able to get IAX to function |
| 09:00.21 | drmessano | Use 1.4.20+ or 1.6 |
| 09:00.24 | drmessano | Works great |
| 09:00.34 | mvanbaak | IAX on 1.0 works great as well |
| 09:00.35 | TrentCreek | I have 1.4.21 |
| 09:00.59 | TrentCreek | Even used the provider examples and refused to work |
| 09:01.08 | drmessano | From Les.net? |
| 09:01.25 | TrentCreek | yes |
| 09:01.26 | drmessano | HAW.. yeah, they have IAX2 setup all wrong.. |
| 09:01.38 | TrentCreek | I htink rapidvox too |
| 09:01.43 | drmessano | I just made my own, and tweaked til it worked |
| 09:02.01 | drmessano | They're also using 1.2, and have some stability issues with it |
| 09:02.25 | TrentCreek | oh...oh |
| 09:02.56 | TrentCreek | I am not quite expereniced enough to do thaat |
| 09:04.02 | drmessano | There's no huge benefit to using IAX2 for a smaller setup.. Really its another tool in the toolbox |
| 09:05.09 | TrentCreek | I hope to not be small |
| 09:05.25 | TrentCreek | do you get better response and sound qualityy? |
| 09:05.29 | drmessano | No |
| 09:05.45 | drmessano | You need to learn a lot more about asterisk before starting an ITSP |
| 09:05.49 | TrentCreek | well then I guess no reason to change |
| 09:07.15 | TrentCreek | yes, I need to know about the metrics so I can pinpoint audio problem spots |
| 09:08.10 | drmessano | and how to config basic peers |
| 09:08.17 | drmessano | etc etc |
| 09:09.20 | TrentCreek | yes, but at this point sound quality would be most important |
| 09:11.59 | *** join/#asterisk feeds (n=chatzill@85-135-225-22.adsl.slovanet.sk) |
| 09:12.05 | protocols | hmm I have problems with receiving faxes from pstn. when watching ztmonitor it seems I am getting signal.. but the sending fax machine gets stuck at "sending..." |
| 09:12.17 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 09:15.58 | TrentCreek | now I am working on a rate card |
| 09:16.21 | TrentCreek | anyone know how to paste single value into multiple rows? |
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| 09:19.15 | angryuser | has anyone tryed downetworks predictive or progressive dialer ? |
| 09:20.31 | TrentCreek | sounds like telemarketing |
| 09:20.48 | *** join/#asterisk ggiusti (n=giovanni@lpierucci.micc.unifi.it) |
| 09:20.49 | angryuser | i am searching a premise solution for outbound call center , which is user friendly ;) not a case of vicidal |
| 09:21.20 | TrentCreek | I doubt you will find anyone to help you with that on here. |
| 09:22.17 | angryuser | TrentCreek: yes, it can be used for that, but in our case we need to call our client's to tell them their account status, ect |
| 09:22.33 | *** part/#asterisk feeds (n=chatzill@85-135-225-22.adsl.slovanet.sk) |
| 09:22.58 | TrentCreek | maybe install FreePBX on the install |
| 09:23.25 | angryuser | *predictive* |
| 09:26.53 | TrentCreek | guess you will have to come back during US normal hours when a lot of people are here |
| 09:26.58 | *** join/#asterisk astrOdz (n=astrOdz@ppp-58-8-59-246.revip2.asianet.co.th) |
| 09:27.00 | astrOdz | hey |
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| 09:30.08 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
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| 09:50.10 | *** join/#asterisk pcrack (n=pcrack@116.50.213.210) |
| 09:51.28 | pcrack | hi, i wanted to ask how to create an IVR that queries to a database. if i enter my pin it will query that pin and the IVR will tell what info does it has...? |
| 09:55.03 | joat | that sounds like an agi script |
| 09:58.10 | mark_csi | joat: I agree, I've something similar |
| 09:59.10 | mark_csi | pcrack: there's a sample php file on asteriskguru that will play back the numbers you put in. I'd get that going and then fire your queries into it. |
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| 10:00.39 | pcrack | thanks |
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| 10:34.22 | shazaum | hi all |
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| 11:18.46 | Karlitoo | question, if I want to add a peer for h323 (h323,conf) do I do it the same way as in sip.conf |
| 11:18.48 | Karlitoo | ?? |
| 11:23.33 | Karlitoo | hi I'm new to asterisk and I would like to know after I made a trunk between asterisk and avaya g350 media gateway trough h323 trunk, how do I add a h323 user and test if the trunk works |
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| 11:59.55 | hi365 | anyone have any ideas for video over sip of sorts? |
| 12:00.06 | hi365 | i have a client that want to be able to see who's at the door |
| 12:02.41 | yang | h.264 video |
| 12:02.59 | yang | I tested it with ekiga works wel loer sip |
| 12:03.06 | yang | well over |
| 12:05.27 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 12:10.50 | Karlitoo | do I need a rtp proxy or something with asterisk |
| 12:11.02 | Karlitoo | so that the sip softpohne can connect |
| 12:12.01 | Karlitoo | and how can I start the CLI so that it outputs errors and when some 1 is tyrinig to login into and extension |
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| 12:24.39 | hi365 | yang: i am actualy looking for a hardphone |
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| 12:33.47 | ratmandu | Anyone have any ideas what is going wrong here? http://pastebin.ca/1259119 |
| 12:34.12 | ratmandu | I'm running Gentoo, and asterisk 1.27.7 |
| 12:34.50 | hi365 | ratmandu: i have no idea what res_smdi.so is for, but you can always try adding a noload => res_smdi.so to modules.conf |
| 12:35.11 | hi365 | as far as the music on hold, i think that one is quite obvious |
| 12:35.26 | ratmandu | done that, and another one fails with the same undefined symbol |
| 12:35.35 | hi365 | shrugs |
| 12:35.46 | hi365 | sorry, I dont know |
| 12:35.58 | hi365 | uses centos and has never had such issues |
| 12:36.24 | ratmandu | no prob, I just got a digium card from a friend and decided to try asterisk out |
| 12:36.52 | hi365 | google should be able to point you to some live cd's that support asterisk |
| 12:37.03 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 12:37.37 | ratmandu | well, the system that this is running on also runs several VMs for a few other people |
| 12:37.56 | hi365 | then you cant reboot it... but you could set up your own vm |
| 12:38.21 | ratmandu | no, i can reboot it, but I just cant use some livedisk to run it |
| 12:38.55 | ratmandu | plus, I dont think openvz supports using external hardware in a vm |
| 12:39.07 | hi365 | :( |
| 12:39.23 | ratmandu | or, if it does, it would likely be a pain to get it working that way |
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| 12:59.03 | hi365 | can anyone recomend an asterisk compatable gsm pci card? |
| 13:00.28 | WimpMan | vlines or junghanns spring to mind |
| 13:02.26 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
| 13:05.26 | hi365 | WimpMan: thanks. nothing much shows on their site and nothing about pricing or purchasing |
| 13:06.00 | WimpMan | Yes, I know. You have to check your favourite reseller. |
| 13:06.09 | *** join/#asterisk SuPrSluG (n=SuPrSluG@72.237.213.162) |
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| 13:18.14 | stoffell | hi365, where are you from? |
| 13:18.43 | hi365 | steliosk: ME |
| 13:18.59 | *** join/#asterisk MrNaz (n=mrnaz@ppp118-208-142-142.lns10.mel4.internode.on.net) |
| 13:19.11 | hi365 | Y? |
| 13:19.40 | stoffell | if u were located in .eu i could give you more info on the gsm pci cards |
| 13:20.01 | hi365 | stoffell: let me hear them anyway (if you dont mind :)) |
| 13:21.33 | WimpMan | I think Maple Leaf do business in quite a number of countries. |
| 13:22.21 | hi365 | links guys, links! |
| 13:22.53 | WimpMan | I'd also have to google. Add some vendor names to get the right page. |
| 13:23.18 | stoffell | hehe :d |
| 13:23.53 | hi365 | WimpMan: no bother than |
| 13:26.43 | WimpMan | http://www.mapleleaf-technologies.de/sitemap.php |
| 13:27.09 | WimpMan | Hmm. I always thought they were british or something. But looks they're in Germany. |
| 13:27.13 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:28.36 | WimpMan | +like |
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| 13:57.16 | PrimeHaxor | <PROTECTED> |
| 13:58.01 | [TK]D-Fender | PrimeHaxor: Means * never got an answer back to its requests |
| 13:58.52 | *** join/#asterisk flush (n=SYN_SENT@ip216-239-73-148.vif.net) |
| 13:59.16 | *** join/#asterisk Daejeo (n=chatzill@118.219.208.186) |
| 14:02.18 | *** join/#asterisk codefreeze-lap (n=murf@72.21.67.40) |
| 14:04.37 | *** join/#asterisk mort_gib (n=mjensen@adsl-2-234.gibnet.gi) |
| 14:04.49 | *** join/#asterisk rnst (n=Ernzt@teisa.netvision.com.py) |
| 14:06.06 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:08.50 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:08.52 | [TK]D-Fender | PrimeHaxor: Networking typically due to improper NAT settings or other firewall issues |
| 14:10.36 | PrimeHaxor | [TK]D-Fender, it's something wrong im looking for network traffic is so slow, i've done a nat for the asterisk server |
| 14:10.59 | *** join/#asterisk Mshadow_ (i=mshadow@rl0.net) |
| 14:11.02 | Mshadow_ | .wu |
| 14:11.55 | [TK]D-Fender | PrimeHaxor: if your * is behind NAT, go read : |
| 14:11.56 | [TK]D-Fender | ~sipnat |
| 14:11.57 | 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 |
| 14:12.19 | Katty | good morning lovables. |
| 14:12.25 | PrimeHaxor | so i'll try isolate the asterisk server and the adsl link in one vlan |
| 14:12.46 | PrimeHaxor | i'll read the link lemme see |
| 14:13.39 | *** join/#asterisk pluesch0r (n=pluesch0@iwein.devoteam.at) |
| 14:13.59 | *** join/#asterisk Thorn (n=thorn@unaffiliated/thorn) |
| 14:19.43 | *** join/#asterisk BBHoss (n=bbhoss@c-68-62-168-242.hsd1.al.comcast.net) |
| 14:22.03 | [TK]D-Fender | Katty: Mew. |
| 14:22.52 | *** join/#asterisk delgaudio (n=delgaudi@cpe-66-108-242-45.nyc.res.rr.com) |
| 14:23.12 | PrimeHaxor | [TK]D-Fender, this problem of exceeded transmission, can drop the call when it established? |
| 14:23.40 | [TK]D-Fender | PrimeHaxor: depends when it happens. |
| 14:24.11 | PrimeHaxor | when i make local call, i don't got any erros, but when i'm try to make a call to other states i've got the drop |
| 14:24.43 | Katty | [TK]D-Fender: mew. |
| 14:24.51 | [TK]D-Fender | PrimeHaxor: you should still ensure that your system is configured properly. |
| 14:25.25 | PrimeHaxor | i'll show u |
| 14:26.56 | *** join/#asterisk Mshadow_ (i=mshadow@rl0.net) |
| 14:29.06 | PrimeHaxor | [TK]D-Fender, http://www.binpaste.com/v.php?id=1aoo0 |
| 14:30.13 | [TK]D-Fender | PrimeHaxor: You said your * box is behind NAT. Is it, or isn't it? |
| 14:30.49 | PrimeHaxor | software voip > asterisk > ATA > ADSLMODEM |
| 14:32.22 | [TK]D-Fender | PrimeHaxor: What is * doing behind an ATA? |
| 14:32.32 | [TK]D-Fender | PrimeHaxor: and is that NAT that it is behind? |
| 14:32.55 | PrimeHaxor | i dunno lol! don't need it ? |
| 14:33.10 | [TK]D-Fender | PrimeHaxor: If you don't know what you have, then what you have is a real problem. |
| 14:33.15 | PrimeHaxor | im very noob when the subject is voip |
| 14:33.25 | [TK]D-Fender | primtNAT is not a voip question! |
| 14:34.08 | PrimeHaxor | if i get out the ATA and connect directly on asterisk will work? |
| 14:34.25 | PrimeHaxor | the ADSL MODEM > ASTERISK |
| 14:34.39 | Katty | :< |
| 14:35.54 | PrimeHaxor | :< = yes ? :( |
| 14:36.08 | [TK]D-Fender | PrimeHaxor: I cannot answer because you don't even know what your networking equipment is doing. |
| 14:36.30 | PrimeHaxor | i'll try one thing |
| 14:36.30 | PrimeHaxor | brb |
| 14:38.02 | [TK]D-Fender | load chan_clue.so |
| 14:39.23 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:42.08 | *** join/#asterisk tobias (n=tobias@cpe-076-182-095-118.nc.res.rr.com) |
| 14:50.20 | *** join/#asterisk LND (n=chatzill@92.41.205.191.sub.mbb.three.co.uk) |
| 14:50.32 | kaldemar | there should be a chan_clue that would print "You should not be doing this" upon load and then segfault * for security's and peace of mind's sake. |
| 14:51.18 | kaldemar | or divide by zero and make the whole box disappear. |
| 14:54.21 | [TK]D-Fender | kaldemar: Anyone who can't answer how their computers get to the internet should not be allowing to use it |
| 14:57.04 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 14:57.24 | kaldemar | that would free the rest of us from a huge amount of spam and lack of bandwidth. and generate a nice amount of customer service jobs. |
| 15:00.50 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 15:01.52 | coppice | Anyone who can't answer how their computers get to the internet is a good source of revenue |
| 15:03.19 | *** join/#asterisk jer_ (n=jtregunn@unaffiliated/jer) |
| 15:03.27 | angryuser | huh ? what is internet |
| 15:03.47 | coppice | for $250 an hour I'll tell you |
| 15:04.27 | angryuser | coppice: tell me your credit call number so i could transfer funds |
| 15:05.29 | coppice | let me guess. you have a mutually beneficial offer for me? |
| 15:05.54 | [8none1] | coppice: Should I use the Dell internet or AOL internet? |
| 15:06.12 | angryuser | coppice: no i am searching the ultilmate answer what is internet ;) |
| 15:06.51 | coppice | use mine. its similar to their's, but has 30% added cost for an improved internet experience |
| 15:06.58 | *** join/#asterisk n3hxs (n=HAMming@63.68.135.4) |
| 15:10.18 | *** join/#asterisk LND (n=lee@92-233-208-244.cable.ubr08.gate.blueyonder.co.uk) |
| 15:10.58 | *** join/#asterisk mindCrime (n=chatzill@216.27.62.2) |
| 15:11.47 | awk_r | [8none1], I just grow my own internet and use that...its way cheaper...i can send you internet seeds if you want? |
| 15:12.35 | coppice | I guess if you grow your own it has a high fibre content |
| 15:13.37 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 15:15.25 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:15.25 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 15:15.59 | x86 | offtopic i know, but does anyone have experience setting up ntpd on linux as a server for the whole subnet? |
| 15:16.25 | lmadsen | google has readily told me the answer to that question... it wasn't very hard |
| 15:16.25 | tzafrir_laptop | x86, apt-get install ntpd |
| 15:16.32 | *** join/#asterisk [8none1] (n=[8none1]@cerberus.franklinamerican.com) |
| 15:16.41 | x86 | tzafrir_laptop: err, yeah, i'm a little past that point ;) |
| 15:16.54 | x86 | tzafrir_laptop: I'm trying to get the config to a point where it allows queries |
| 15:16.55 | tzafrir_laptop | And the specific problem is? |
| 15:17.20 | tzafrir_laptop | It allows queries by default. Unless you block it through firewalls |
| 15:17.29 | x86 | tzafrir_laptop: I've got restrict dsefault nomodify and restrict 127.0.0.1 as the only two restrict lines |
| 15:17.37 | x86 | there is no firewall |
| 15:18.12 | x86 | it does not allow queries by default |
| 15:18.19 | x86 | otherwise there would have never been a problem ;) |
| 15:19.07 | x86 | I've got Polylcom IP330 --> linksys switch --> Asterisk server |
| 15:19.11 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:19.35 | x86 | iptables -vnxL on the Asterisk box shows no rules at all, and default policy values of "ACCEPT" on all tables |
| 15:20.33 | *** join/#asterisk drmessano^ (n=nonya@pdpc/supporter/active/drmessano) |
| 15:21.18 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-ab56e4cdf405d814) |
| 15:21.18 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:22.10 | x86 | wish ntpd would log why the hell it's dropping queries |
| 15:23.23 | *** join/#asterisk ziram19 (n=chatzill@196.203.52.254) |
| 15:23.24 | x86 | tzafrir_laptop: do you have a working config I could borrow? |
| 15:23.38 | x86 | mine seems to load fine, but not do what is intended |
| 15:24.04 | ziram19 | sip show peers don't work on * 1.6? |
| 15:24.13 | x86 | if I use "default" in the restrict line or actually define the local subnet with no restrictions, both ways fail |
| 15:24.39 | [TK]D-Fender | x86: You remember to provision your phones to point to your server for NTP in the first place? |
| 15:25.41 | x86 | [TK]D-Fender: YEP |
| 15:25.46 | x86 | [TK]D-Fender: damn caps |
| 15:26.19 | x86 | [TK]D-Fender: the logs say "failed to set time from server" or something, and also I have a linux laptop that I'm trying to use as an NTP client too (for testing), no dice there either |
| 15:26.38 | [TK]D-Fender | x86: netstat shows it listening? |
| 15:26.49 | [TK]D-Fender | x86: pastebin your configs |
| 15:26.55 | ziram19 | no response for sip show peers on * 1.6? |
| 15:27.00 | x86 | hmm seems like it just started working |
| 15:27.02 | [TK]D-Fender | x86: dhcpd.conf & ntpd.conf |
| 15:27.15 | [TK]D-Fender | x86: it fears me :) |
| 15:27.18 | x86 | I did a query on the laptop with -u, and now regular queries from the laptop seem to work |
| 15:27.21 | *** join/#asterisk Un1x (n=Shaanay@CPE001d7e1e2ddb-CM001692fa22fa.cpe.net.cable.rogers.com) |
| 15:27.24 | [TK]D-Fender | x86: just like allthe computers at my office :) |
| 15:27.30 | x86 | there is no dhcpd on this network ;) |
| 15:29.29 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 15:30.41 | x86 | ugh, now the polycom is trying to sync with an outside NTP server, even though it's configured for the local server |
| 15:30.57 | x86 | 164.67.62.194 |
| 15:31.20 | x86 | SNTP: initial soMain sync failed with server 164.67.62.194. |
| 15:31.30 | x86 | (from the ip330 logs) |
| 15:32.20 | *** join/#asterisk kippi (n=chriso@83-244-164-130.cust-83.exponential-e.net) |
| 15:32.21 | kippi | hey |
| 15:32.39 | kippi | has anyone used a poloycom IP 6000 with asterisk and got it to work? |
| 15:34.20 | dominic1 | after installing zapte I get this information: http://pastebin.com/d6f371f23 |
| 15:34.27 | coppice | kippi: we've tested those with Freeswitch and had no problems |
| 15:34.31 | dominic1 | Is it possible to use my beronet misdn card with zaptel? |
| 15:34.43 | Un1x | YOUR supposed to be using dahdi not zaptel |
| 15:35.03 | IsUp | BAAAAAAAAMMMMM |
| 15:35.08 | dominic1 | not yet @Un1x |
| 15:35.14 | kippi | with freeswitch was you using openser |
| 15:35.16 | IsUp | forget Zaptel, USE DAHDI! save the world asdasf asfasfasfas |
| 15:35.18 | IsUp | bored. |
| 15:35.22 | dominic1 | I first need to use the latest zaptel |
| 15:35.35 | dominic1 | cause my production system is actually on zaptel |
| 15:35.43 | dominic1 | I am installing a testenv |
| 15:36.10 | dominic1 | is it possible to use my misdn card with dahdi then? |
| 15:36.21 | Un1x | lol wow, your either not reading or your ignorant, dahdi is just renamed zaptel.... |
| 15:36.36 | dominic1 | that's not 100% correct |
| 15:36.54 | coppice | dahdi is zaptel qith added problems |
| 15:36.59 | x86 | unreal..... wtf... why wont this damn phone attempt to contact the NTP server it's configured to talk to? |
| 15:37.15 | Un1x | OKay, if you want to beleive that dahdi works fine for me :) |
| 15:37.38 | [TK]D-Fender | x86: So its either in the bootrom settings or your provisioning... go look. |
| 15:38.21 | dominic1 | I need a system with exactly the same versions as my productionsystem and just wanted to know if it's now possible to use misdn cards with zaptel, cause I got this information Loading zaptel while installing zaptel http://pastebin.com/d6f371f23 |
| 15:41.07 | stoffell | dominic1, you don't need zaptel if you use misdn ?? |
| 15:41.27 | stoffell | replace ?? with ... :p |
| 15:41.52 | dominic1 | okay, I need zaptel for things like meetme |
| 15:42.00 | dominic1 | I always used ztdummy |
| 15:42.52 | dominic1 | I was a little bit confused as I saw the output above while isntalling zaptel the driver found my misdn card. |
| 15:43.55 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
| 15:43.57 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-37-217.w86-215.abo.wanadoo.fr) |
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| 15:44.20 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:44.34 | *** join/#asterisk joako (n=joako@adsl-074-170-252-213.sip.gnv.bellsouth.net) |
| 15:45.17 | joako | how can i reboot polycom phones remotely? sip notify polycom-check-config only reboots if the config file changed |
| 15:45.55 | angryuser | joako: change useless option in config |
| 15:49.43 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 15:50.51 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
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| 15:51.56 | *** join/#asterisk root52 (n=f745082a@ip70-191-120-39.cl.ri.cox.net) |
| 15:52.54 | joako | angryuser: won't work for me |
| 15:54.19 | *** part/#asterisk root52 (n=f745082a@ip70-191-120-39.cl.ri.cox.net) |
| 15:55.39 | *** join/#asterisk ManxPower (n=manxpowe@6.sub-70-220-58.myvzw.com) |
| 15:57.35 | *** join/#asterisk stimpie (n=stimpie@213-73-211-48.cable.quicknet.nl) |
| 16:09.06 | *** join/#asterisk ElDios (n=ElDios@213-140-6-112.ip.fastwebnet.it) |
| 16:09.13 | ElDios | hey guys |
| 16:09.22 | ElDios | I've a music on hold problem, on the caller side... |
| 16:09.31 | ElDios | in the logs I've this rows |
| 16:09.39 | ElDios | http://pastebin.com/m2aee08d5 |
| 16:09.48 | ElDios | but nothing comes out from the caller phone |
| 16:10.00 | ElDios | no ring tone, no music, nothing... |
| 16:10.01 | *** join/#asterisk write_erase (n=Olivier@telindu015615-6.clients.easynet.fr) |
| 16:10.02 | ElDios | any idea? |
| 16:10.04 | joako | Post the dialplan |
| 16:10.27 | ElDios | mmm... where do I get it fromin my conf files? |
| 16:10.44 | joako | extensions.conf |
| 16:10.50 | ElDios | aahh.. oke |
| 16:11.15 | *** join/#asterisk propellerhead (n=yogurt2u@host38.190-136-115.telecom.net.ar) |
| 16:11.21 | write_erase | Hi, modprobe dahdi_dummy returns : dahdi_dummy: Unable to register DAHDI rtc driver ... (I'm running Debian Etch + dahdi svn) I need my timing source, please help ! |
| 16:12.12 | joako | try modprobe dahdi ;modprobe dahdi_dummy |
| 16:12.26 | ManxPower | modprobing dahdi or zaptel does NO good. |
| 16:12.37 | ManxPower | all the hardware drivers load zaptel automatically |
| 16:13.06 | ManxPower | write_erase: is "rtc" listed in the output of "lsmod"? |
| 16:13.22 | ElDios | joako that file is quite long... the extensions specific configuration is in extension_additional.conf |
| 16:13.26 | ElDios | and is much shorter |
| 16:13.29 | ElDios | is that part enough? |
| 16:14.33 | write_erase | ManxPower, Yes... rtc is loaded |
| 16:14.49 | ManxPower | ElDios: Sounds like you are using a GUI with Asterisk. |
| 16:15.04 | ElDios | ManxPower yes -_-' |
| 16:15.18 | ManxPower | write_erase: *_dummy uses the RTC. Does the output of dmesg say anything helpful. |
| 16:15.29 | ManxPower | ElDios: You might have better luck asking on the correct channel. |
| 16:15.32 | ManxPower | ~trixbox |
| 16:15.33 | jbot | methinks trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org/. We do not recommend using it. |
| 16:15.42 | ElDios | :D |
| 16:15.49 | ElDios | lulz |
| 16:16.12 | ElDios | this is the best => We do not recommend using it.<= |
| 16:16.14 | ElDios | oke.. |
| 16:16.45 | write_erase | dahdi: Registered tone zone 0 (United States / North America) |
| 16:16.46 | write_erase | dahdi_dummy: Unable to register DAHDI rtc driver |
| 16:16.52 | ElDios | thnx anyway guys |
| 16:17.16 | ManxPower | ElDios: you will find pro-GUI people on the GUI channels. |
| 16:17.27 | write_erase | ManxPower, dahdi registers, but not dahdi_dummy ... which provides the timing IIRC |
| 16:17.31 | ManxPower | write_erase: try rmmod rtc and see what happens. |
| 16:18.02 | ElDios | ManxPower no ideas in there... that was I came here to see if there's anyone that could help me.. |
| 16:18.06 | ElDios | anyway, no problem |
| 16:18.38 | ManxPower | write_erase: Looks like time for a mailinglist search. 9-) |
| 16:19.13 | ManxPower | ElDios: the problem with most GUI users is that they are using a GUI because they don't want to learn anything. These sorts of people don't provide very good support. |
| 16:19.48 | write_erase | ManxPower, Ok... chrony used /dev/rtc, now I could rmmod rtc & load dahdi_dummy |
| 16:19.55 | ElDios | ahah.. ManxPower I'm a *nix users since years.. I've lots of RTFM-ppl in action... |
| 16:20.35 | ElDios | I'm not an Asterisk specialist, that's why someone passed me a Trixbox-ready system already configured and I'm continuing to use it @ work |
| 16:21.04 | ElDios | if I had more experience with Asterisk I probably had already reimplemented the whole thing... |
| 16:21.14 | ManxPower | *nod* You really need to know Linux (or at least *nix), Networking (including UDP, NAT and ports), Telecom, SIP, and Asterisk. |
| 16:21.31 | ElDios | no problem with the first 3 |
| 16:21.34 | Un1x | Hey guys quick question where can i find some information on setting up asterisk for three way calling? |
| 16:21.43 | ElDios | the last 2 quite new to me |
| 16:21.49 | ManxPower | ElDios: the major issue with the GUIs is that they totally take over Asterisk making it virtually impossible to do any customizing. |
| 16:22.03 | ElDios | as any GUI in the world, AFAIK |
| 16:22.04 | ManxPower | Un1x: Press the CONF button on your phone. |
| 16:22.05 | ElDios | =) |
| 16:22.22 | ManxPower | Un1x: Now are you ready to ask a decent question? |
| 16:22.28 | ElDios | XD |
| 16:23.06 | ManxPower | ElDios: Un1x has been an Asterisk user for years. He doesn't get to ask useless questions. 8-| |
| 16:23.24 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:23.42 | ElDios | ^_^ |
| 16:23.44 | ManxPower | ElDios: If the only things you need to learn is SIP and Asterisk then you are further along than most Asterisk noobs. |
| 16:24.04 | ElDios | in your opinion |
| 16:24.46 | ElDios | how many study/trial&error hours do I have to spend to have a fully configured asterisk system from scratch on debian? |
| 16:24.47 | ManxPower | ElDios: One issue with noobs that they don't understand the source port does not have to be the same as the destination port. |
| 16:25.06 | ManxPower | ElDios: I recommend setting aside at least 2 weeks. |
| 16:25.14 | ElDios | ManxPower oke... at least now I now that I'm not at the zero level =) |
| 16:25.35 | ManxPower | Asterisk is not really a PBX. Asterisk is a PBX TOOLKIT. |
| 16:26.02 | ElDios | so you mean that is Asterix+FreePBX? |
| 16:26.04 | Un1x | ManxPower, what if my phone doesn't have the conference button... |
| 16:26.33 | ManxPower | Un1x: You know you need to provide more information before any decent answer can be given. What version of Asterisk, what protoocl, what phone? |
| 16:26.57 | Un1x | Asterisk 1.4.22 SIP and its a regualr analoge phone.. |
| 16:27.01 | ManxPower | FreePBX is a PBX built on the Asterisk toolkit. |
| 16:27.08 | ManxPower | Un1x: plugged into what kind of card? |
| 16:27.17 | Un1x | TDM400p |
| 16:27.24 | *** join/#asterisk BBHoss_ (n=bbhoss@c-68-62-168-242.hsd1.al.comcast.net) |
| 16:27.33 | ManxPower | Un1x: do you know how to do 3-way calling on a normal PSTN line from the telco? |
| 16:27.42 | Un1x | yes. |
| 16:27.53 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
| 16:28.23 | ManxPower | that is how you do it in Asterisk with an analog port. See the /etc/asterisk/zapata.conf or /etc/asterisk/chan_dahdi.conf and enable transfer and three-way calling. |
| 16:28.30 | ManxPower | see the .sample files for example. |
| 16:28.33 | Un1x | okay |
| 16:30.02 | rwaite | yawn |
| 16:30.28 | [TK]D-Fender | ManxPower>FreePBX is a PBX built on the Asterisk toolkit. <-- not quite as I'd put it... |
| 16:30.30 | ManxPower | Un1x: the flash button is what you would use. |
| 16:30.37 | ManxPower | [TK]D-Fender: I would. 8-) |
| 16:31.01 | ManxPower | Well ok, FreePBX is a pathetic attempt to make a PBX built on the Asterisk toolkit. Better? |
| 16:31.11 | rwaite | FreePBX: parasite who's host is the asterisk toolkit? |
| 16:31.39 | [TK]D-Fender | FreePBX is a set of scripts & apps that builds a fairly complete set of * configs based on its limited structure which accounts for a lot of basics people expect to configure on a closed legacy PBX. |
| 16:31.52 | [TK]D-Fender | rwaite: there you go! |
| 16:32.08 | ElDios | =) |
| 16:32.35 | ElDios | thnx anyway guys... cya soon |
| 16:32.40 | ElDios | thnx ManxPower |
| 16:32.42 | ElDios | bye |
| 16:33.07 | *** part/#asterisk ElDios (n=ElDios@213-140-6-112.ip.fastwebnet.it) |
| 16:36.10 | *** join/#asterisk Thorn (n=thorn@unaffiliated/thorn) |
| 16:36.53 | Un1x | dude i just replaced my phone with a fax just for the fuck of it and guess what it works fine :D |
| 16:37.56 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 16:41.39 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 16:42.20 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 16:46.02 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 16:49.07 | ManxPower | Your fax does 3-way calling? |
| 16:50.58 | *** join/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 16:51.04 | fcois93 | hello all |
| 16:51.14 | fcois93 | How can I insert logs in the dialplan ? |
| 16:51.38 | fcois93 | like a noop but in a logfile |
| 16:54.48 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 16:55.46 | *** join/#asterisk bijit (n=benji@200.122.158.243) |
| 16:56.36 | jsmith | fcois93: Use the Log() application? |
| 16:56.41 | *** join/#asterisk andresmujica (n=andresmu@190.25.104.186) |
| 16:57.00 | *** join/#asterisk erth64net (n=erth64ne@mail.opensourcery.com) |
| 16:57.16 | Un1x | ManxPower no, but im saying i just plugged it in for randomb testing and sure enough it works :) |
| 16:58.09 | fcois93 | jsmith: yes, found it :) sorry |
| 17:07.42 | *** join/#asterisk joako (n=joako@adsl-074-170-252-213.sip.gnv.bellsouth.net) |
| 17:07.47 | *** join/#asterisk BBHoss (n=bbhoss@c-68-62-168-242.hsd1.al.comcast.net) |
| 17:18.07 | *** join/#asterisk erth64net (n=erth64ne@mail.opensourcery.com) |
| 17:21.05 | *** join/#asterisk mark_csi (n=mark@host217-41-18-3.in-addr.btopenworld.com) |
| 17:21.07 | *** join/#asterisk sah-work (n=Bawbatos@wlan-4089.sc08.org) |
| 17:26.03 | mark_csi | hi all - anyone configured a skype channel in asterisk? |
| 17:27.35 | Qwell | mark_csi: save yourself the headaches - wait for the Digium one to be released |
| 17:28.14 | Maliuta | or just avoid skype |
| 17:28.21 | Maliuta | it's a massive security hole |
| 17:28.42 | Maliuta | anything that p2p's and looks to punch out of a firewalled environment is evil |
| 17:28.56 | angryuser | mark_csi: how do you want to implement your skype channels ? |
| 17:29.34 | angryuser | a had a great succes with skip2pbx but it is not gpl nor free |
| 17:29.48 | angryuser | others wre just a crap |
| 17:29.50 | angryuser | ;) |
| 17:30.39 | mark_csi | hmmm - not really thought about it? I put asterisk into a hotel and customed it, now hotel is looking to offer skype to clients |
| 17:31.19 | Maliuta | mark_csi: how do they want to offer skype? to the phones in the room? |
| 17:32.08 | mark_csi | ideally I'd like to offer it through the phones in the room |
| 17:32.27 | *** join/#asterisk kc2tnk (n=fskrotzk@host.textwise.com) |
| 17:33.09 | Maliuta | chan_skype isn't really functional for that |
| 17:33.36 | *** part/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 17:33.42 | mark_csi | that's what I thought - how do users login and then see their contact list without using their own laptop |
| 17:33.59 | Maliuta | exactly how do they want that to work? skype into the room phones with X skype accounts? or just to enable guests to skype out? |
| 17:34.29 | Maliuta | sounds poorly thought out |
| 17:35.01 | mark_csi | I've set up a couple of wireless sip phones that also hold skype accounts |
| 17:35.42 | mark_csi | I suspect they just want to dial skype accounts out |
| 17:35.56 | [TK]D-Fender | mark_csi: they'd have to add * as a friend and so much other BS, the administration overhead on this so they could use * on the interior is jsut ridiculous. |
| 17:35.57 | Maliuta | I guess it's just a matter of letting skype out if the hardware supports it, no real way to bill/control it though |
| 17:36.50 | mark_csi | What we could do is just hire out skype enabled phones from reception - have them sip configured as well |
| 17:36.51 | n3hxs | I don't see how you would be able to allow the caller to designate their Skype account & PW. |
| 17:37.31 | n3hxs | through Asterisk, that is... |
| 17:37.37 | Maliuta | the whole "offer skype" thing sound dubious, if they offer general 'net connections there is automatically the opportunity for guests to use skype there |
| 17:38.27 | mark_csi | Maliuta: perhaps what we could do is offer to set up a couple of wifi phones with their user accounts and then give them to the guests |
| 17:39.08 | mark_csi | the hotel also has a skype account setup - probably looking to receive calls inbound to reception on that user account |
| 17:39.17 | Maliuta | shrugs |
| 17:39.39 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:40.10 | Maliuta | reminds me next time I do up cards I need to get "SIP/" put on them as a contact number |
| 17:40.27 | Maliuta | well sip and/or iax |
| 17:40.52 | *** join/#asterisk maxxim (n=maxxxim@93-96-96-106.zone4.bethere.co.uk) |
| 17:41.35 | maxxim | Hi, how to enable global internal timing? i was using [options] and "internal_timing = yes" from asterisk.conf. But no success. |
| 17:45.43 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 17:46.40 | *** join/#asterisk sah-work (n=Bawbatos@wlan-4089.sc08.org) |
| 17:47.38 | *** join/#asterisk shriven (n=shriven@cpe-076-182-080-161.nc.res.rr.com) |
| 17:48.07 | [TK]D-Fender | maxxim: To acheive what exactly? |
| 17:48.37 | shriven | hey guys I'm looking for some help finding documentation on how to make it so my server can receive a call like "user@my.asterisk.domain.com". I just am not sure of the correct lingo I need to google to find the answer. Anyone able to point me in the right direction? |
| 17:48.49 | *** join/#asterisk hfb (n=hfb@96.247.64.35) |
| 17:49.10 | shriven | or EXTENNUMBER@my.asterisk.domain.com |
| 17:49.29 | [TK]D-Fender | shriven: [general] allowguest=yes context=mymiscsipcalls [mymiscsipcalls] exten => fred,1,Dial(SIP/100) |
| 17:49.59 | shriven | hmmmm |
| 17:50.51 | Maliuta | hmmmmm??? it really is that simple |
| 17:51.07 | WimpMan | And don't forget to drill some holes intop your firewall at appropriate places. |
| 17:52.56 | [TK]D-Fender | WimpMan: No more than you needed to do in the first place <- |
| 17:53.33 | [TK]D-Fender | WimpMan: thats like saying "Oh and don't forget to install Asterisk". Its kind of a given. |
| 17:54.10 | WimpMan | Only if you have external clients. |
| 17:54.21 | *** join/#asterisk PMantis (n=sswitzer@cpe-67-240-239-27.rochester.res.rr.com) |
| 17:54.39 | [TK]D-Fender | WimpMan: or anything external... but then again, this is stuff every user should already knows has to be set up properly in the first place.. |
| 17:55.39 | PMantis | Hello, According to a bug report (http://bugs.digium.com/view.php?id=10226), I shouldcompile with GCC 4.1.x, not 4.2.x. I had this sound issue, too. So, can I set an env variable to cause make to use gcc 4.1? |
| 17:56.14 | WimpMan | [TK]D-Fender: He, having you optimistic day today? :-) |
| 17:56.57 | Yourname` | Hi, 1) after changing a SetMusicOnHold variable in extensions.conf, will a reload help or do I need to restart? 2) Also, I have exten=6,n,SetMusicOnHold(new).. so I'm guessing when I'm transfered to 6, the hold class should be new.. yet somehow the "default" class is played.. where do I make this change? |
| 17:58.12 | ManxPower | You just have to do a reload to change the MoH class in extensions.conf. However if you are ADDING a new MoH class to musiconhold.conf it would not suprize me if you had to do a restart |
| 17:59.24 | jsmith | ManxPower: a "moh reload" should be enough |
| 17:59.43 | WimpMan | 'd go for dialplan reload. |
| 17:59.49 | [TK]D-Fender | Yourname`: PASTEBIN <- |
| 17:59.51 | Yourname` | jsmith: Or a reload should suffice, no need for restart? Because ManxPower is right, I added a new MOH class |
| 18:00.11 | ManxPower | Yourname`: Did you TRY jsmith's suggestion? |
| 18:00.14 | jsmith | Yourname`: A reload *won't* suffice for musiconhold... it has to be "moh reload" |
| 18:00.23 | Yourname` | AH! |
| 18:00.25 | Yourname` | Let me try |
| 18:00.33 | maxxim | [TK]D-Fender: [Nov 17 20:00:16] DEBUG[18974]: channel.c:2780 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=1048576 chan->timingfd=-1) |
| 18:00.51 | maxxim | [TK]D-Fender> core show settings: Internal timing: Enabled |
| 18:02.03 | shriven | maliuta, [TK]D-Fender: Ok, I am not able to determine what is wrong for my setup... I had not had "allowguest" I have turned that on and reloaded, I am getting this error: http://pastebin.com/d366e8e95 |
| 18:02.06 | bijit | does busydetect has a value by default even though not on zapata.conf? |
| 18:02.59 | shriven | that extension does exist |
| 18:03.23 | ManxPower | bijit: it defaults to OFF |
| 18:03.23 | bijit | ar242 |
| 18:03.27 | shriven | I can make calls to if from other phones served by the asterisk by dialing 6000 |
| 18:04.21 | [TK]D-Fender | shriven: IAX? Who uses IAX? :) |
| 18:04.25 | bijit | ManxPower: so its the same as busydetect=no right/ |
| 18:04.32 | ManxPower | bijit: correct |
| 18:04.34 | [TK]D-Fender | shriven: Sorry, taht was for SIP... dunno how it works for IAX |
| 18:04.36 | shriven | just connecting from an iax client for testing |
| 18:04.38 | shriven | it's easier |
| 18:04.45 | shriven | the sip account exists for extension 6000 |
| 18:04.46 | bijit | ManxPower: thx |
| 18:04.59 | Un1x | is there a new book now, since weve gone through soo much changes? |
| 18:05.35 | ManxPower | Un1x: just the addendum that's included in the Asterisk source. UPGRADE.txt and UPGRADE-1.2.txt I believe |
| 18:05.40 | *** join/#asterisk Segnale007 (n=Pietro@host146-242-dynamic.9-79-r.retail.telecomitalia.it) |
| 18:05.41 | bijit | ManxPower: is there like a site where I can see the indications of tones for my country or do I have to get it from my telco? |
| 18:06.02 | ManxPower | bijit: using busydetect or callprogress is a bad idea and will lead to random hangups |
| 18:06.33 | bijit | ManxPower: I don't have that in my pri setting and still get random hangups |
| 18:06.44 | ManxPower | bijit: NEVER EVER use those options with PRI. |
| 18:07.18 | ManxPower | You have some OTHER problem causing the hangups |
| 18:07.28 | *** join/#asterisk gazzerh (n=garryh@host-212-158-227-149.dslgb.com) |
| 18:08.13 | bijit | ManxPower: That is what I am trying to troubleshoot. |
| 18:08.20 | *** join/#asterisk TrentCreek (n=kvirc@ppp-70-244-17-140.dsl.hrlntx.swbell.net) |
| 18:08.24 | bijit | Any Ideas? Where I can start? |
| 18:08.41 | ManxPower | bijit: what is the value of HANGUPCAUSE for the dropped calls? |
| 18:08.53 | *** join/#asterisk gazzerh (n=garryh@host-212-158-227-149.dslgb.com) |
| 18:09.06 | shriven | [TK]D-Fender: Hmm ok I think it may have just been because I was originating the test call from an iax client..... If I originate it from a sip client it seems to work. Thanks. |
| 18:09.18 | Yourname` | jsmith: That worked, thanks! |
| 18:09.27 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-199-83-rrdg-esr-2.dynamic.isadsl.co.za) |
| 18:09.37 | [TK]D-Fender | shriven: Who are you looking to give this address to anyway? |
| 18:09.47 | shriven | ipkall.com |
| 18:09.55 | shriven | get a free DID. : ) |
| 18:10.09 | [TK]D-Fender | shriven: OH... then FFS use SIP if you know whats good for you! |
| 18:10.20 | shriven | Yeah I am using sip. |
| 18:10.20 | [TK]D-Fender | shriven: and that's exactly what I did |
| 18:10.29 | shriven | : ) |
| 18:10.35 | [TK]D-Fender | shriven: Except I did make a non-authing peer to receive the call. |
| 18:10.50 | shriven | Oh hmm... Yeah I'll have to do that. |
| 18:10.52 | shriven | Thanks. |
| 18:10.56 | [TK]D-Fender | shriven: Directs the context and I set the exten to the DID so it looks "natural" |
| 18:11.31 | bijit | ManxPower: Let me look for it. |
| 18:12.04 | shriven | [TK]D-Fender: I'm going to use it to have my mobile provider redirect calls that I miss to my asterisk... Then I can do fun stuff. : ) |
| 18:12.40 | [TK]D-Fender | shriven: If you want to bypass cell VM sure, but is your IPKall # LOCAL to you? |
| 18:13.04 | [TK]D-Fender | shriven: because redirecting your cell there would incur LD normally if it isn't |
| 18:13.06 | bijit | <PROTECTED> |
| 18:13.06 | shriven | No, but no one that calls my original number is local either.. |
| 18:13.18 | shriven | Right, free nationwide long distance. |
| 18:13.21 | ManxPower | bijit: so the Q.931 cause code is 17. look it up. |
| 18:13.23 | [TK]D-Fender | shriven: I mean is your CELL local to your IPKAall # |
| 18:13.37 | bijit | ManxPower: ok |
| 18:13.40 | [TK]D-Fender | shriven: shriven I'd double check if I were you. |
| 18:13.44 | shriven | [TK]D-Fender: Yeah shouldn't be a problem with free nationwide. |
| 18:13.45 | *** join/#asterisk hi365_m (n=hi365@213.151.57.227) |
| 18:13.54 | shriven | [TK]D-Fender: Indeed, that is a good thought. |
| 18:13.54 | ManxPower | Cause 17 is The number dialed is busy and cannot receive any more calls. |
| 18:13.55 | [TK]D-Fender | shriven: But if you are covered for LD on your cell, then you should be fine. |
| 18:14.15 | ManxPower | you should be running Busy() when you get that cause back |
| 18:15.53 | bijit | that us why it issues this: == Everyone is busy/congested at this time (1:1/0/0) |
| 18:16.06 | ManxPower | bijit: PRIs don't normally play those tones, it is up to the PBX to capture the code and do what is correct for that code. |
| 18:16.18 | ManxPower | bijit: the number you dialed was busy. |
| 18:19.18 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 18:20.08 | bijit | ManxPower: So that is not a call drop. |
| 18:20.46 | ManxPower | bijit: there is no indication that the call dropped so I assume it did not drop |
| 18:23.38 | bijit | [Nov 14 12:04:37] VERBOSE[24474] logger.c: q931.c:2764 q931_disconnect: call 31970 on channel 21 enters state 11 (Disconnect Request) |
| 18:24.29 | ManxPower | bijit: I do not read Q.931 debug |
| 18:25.19 | jaytee | ManxPower, the ending sucks. thank god there wasn't a sequel |
| 18:25.46 | [TK]D-Fender | jaytee: You mean BloodRayne? |
| 18:25.59 | Katty | anyone have thoughts about why DTMF isn't recognize when i call an IVR through a sip trunk and hit 1 a billion times with no response. |
| 18:26.05 | jaytee | [TK]D-Fender, no Q.931 debug :-) |
| 18:26.18 | maxxim | hi, i've loaded 'ztdummy' but anyway i got [Nov 17 20:25:02] DEBUG[24450]: channel.c:2780 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=1048576 chan->timingfd=-1) |
| 18:26.19 | [TK]D-Fender | jaytee: oh... hat too ;) |
| 18:26.19 | *** join/#asterisk erth64net (n=erth64ne@mail.opensourcery.com) |
| 18:26.27 | Katty | outgoing calls. through sip, to another phone system in East Jesus... and no love on their IVR. more speicifically |
| 18:27.25 | jaytee | Katty, what's the DTMF setting for the phones and SIP trunk? RFC2833? INBAND? |
| 18:27.33 | Katty | looks |
| 18:27.48 | [TK]D-Fender | jaytee: except... they MADE a sequel : http://www.imdb.com/title/tt0896036/ |
| 18:27.49 | Katty | jaytee: which conf would that be in? |
| 18:28.37 | Katty | jaytee: in sip.conf for sip trunk it's rfc2833 |
| 18:29.34 | Katty | jaytee: zapata claims relaxdtmf=yes |
| 18:30.28 | PMantis | Katty: That's in sip.conf |
| 18:32.36 | _ShrikE | Is the buggymwi option in sip.conf available on a per peer basis, or just general? |
| 18:40.40 | Un1x | Hey, was wondering is there a way i can setup asterisk to start and stop recording by using like an extension like lets say im in a call and i can press #blah and it starts and then #duh to stop? |
| 18:41.30 | _ShrikE | google asterisk automon |
| 18:42.02 | Un1x | thanks |
| 18:42.26 | _ShrikE | np |
| 18:43.08 | Un1x | ahh so just *1 to start and *1 to stop interesting |
| 18:43.55 | shriven | [TK]D-Fender: you set this up with ipkall right? Did you give it a sip number like extension@yourasterisk.com or did you have another did for it to use? The sip phone number field seems to not have enough room for an exten@ format. |
| 18:44.33 | shriven | [TK]D-Fender: Or.... maybe that is what sip proxy is for. ;) |
| 18:44.37 | shriven | < n00b. |
| 18:44.46 | [TK]D-Fender | shriven: it does for me... |
| 18:44.52 | [TK]D-Fender | shriven: Indeed.. 2 pieces. |
| 18:45.01 | [TK]D-Fender | shriven: exten in one, proxy in the other |
| 18:45.02 | Un1x | _ShrikE, i enabled it in features.conf and reloaded asterisk and for some odd reason agent pressed *1 in a call and 2 minutes later *1 again and asterisk console showed nothing,... |
| 18:45.03 | *** join/#asterisk sah-work (n=Bawbatos@wlan-4089.sc08.org) |
| 18:45.14 | shriven | yeah... got it now |
| 18:45.15 | shriven | thanks |
| 18:45.40 | *** join/#asterisk bmoraca (n=bmoraca@209.60.253.58) |
| 18:46.19 | *** join/#asterisk s34n (n=chatzill@ip-208-76-93-125.mvdsl.com) |
| 18:47.57 | bmoraca | has anyone here had success getting fax tone detection over a sip trunk from another asterisk server? |
| 18:48.08 | _ShrikE | Un1x: Do you have the w or W option in your dial statement? |
| 18:49.35 | Un1x | exten =>_3xxx,2,Dial(SIP/${EXTEN},30,wth) |
| 18:49.37 | Un1x | thaty right? |
| 18:51.31 | maxxim | yhhaa, i have soleved the problem with ringback!!! it was problem in timing :D |
| 18:52.29 | *** join/#asterisk rdgr (n=rich@82-46-0-91.cable.ubr01.aztw.blueyonder.co.uk) |
| 18:52.29 | Un1x | _ShrikE i have this in my dial plan... exten => _X.,2,Dial(${splitinfinity}/${EXTEN},30,wth) |
| 18:52.33 | Un1x | and it still doesn't work |
| 18:52.36 | *** join/#asterisk bbryant (n=brett@68.208.65.50) |
| 18:53.07 | _ShrikE | Did you set DYNAMIC_FEATURES=automon? before you dialed, or globally? |
| 18:53.56 | Un1x | _ShrikE. where do i set that? |
| 18:54.26 | _ShrikE | exten => _X.,1,Set(DYNAMIC_FEATURES=automon) |
| 18:55.20 | Un1x | exten => _X.,2,Dial(${splitinfinity}/${EXTEN},30,Wth) |
| 18:55.24 | Un1x | so that is wrong then |
| 18:55.33 | *** join/#asterisk [netman] (n=netman@238.Red-88-23-119.staticIP.rima-tde.net) |
| 18:56.18 | _ShrikE | That woudl allow the calling party to initiate recording. |
| 18:56.23 | _ShrikE | err.. would |
| 18:56.30 | s34n | is there a fedora package for ztdummy? |
| 18:57.16 | *** join/#asterisk erth64net (n=erth64ne@mail.opensourcery.com) |
| 18:57.17 | Un1x | Ya, i need the calling party to be able to initiate the recording.. |
| 18:57.24 | Un1x | but _ShrikE its not working.. |
| 18:57.57 | Un1x | _ShrikE my dialplan http://pastebin.com/d56728084 |
| 18:58.29 | *** join/#asterisk fransman (n=frans@a80-127-14-241.adsl.xs4all.nl) |
| 18:59.12 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:00.52 | [TK]D-Fender | Un1x: Go prove that DTMF even works with that peer <- |
| 19:00.59 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 19:01.22 | [TK]D-Fender | Un1x: Next you need to confirm the pattern to initiate it and realize that you have VERY little time between digits to trigger it |
| 19:01.37 | Un1x | [TK]D-Fender what peer the provider, ? or the called party? |
| 19:01.54 | [TK]D-Fender | Un1x: the side of the call who is trying to trigger it |
| 19:01.59 | Un1x | so how can i fix that so it triggers when i want it to |
| 19:02.21 | [TK]D-Fender | Un1x: useless open ended question... |
| 19:02.22 | Un1x | Well, the calling side is me |
| 19:02.33 | [TK]D-Fender | Un1x: Go prove DTMF works. then test another way |
| 19:02.56 | [TK]D-Fender | Un1x: then ensure your actual recording test is done somewhere where you can assure that the user enters the code fast enough |
| 19:03.13 | [TK]D-Fender | Un1x: And of course confirm thaqt your features.conf is set up right |
| 19:03.22 | *** part/#asterisk oej (n=olle@ns.webway.se) |
| 19:03.52 | Un1x | [TK]D-Fender, Well, you mean i was one who was hitting the code and i was doing it fast enough now i just tried on the other end of the phone and soon as i pressed * it said user disconnectd the call |
| 19:04.27 | [TK]D-Fender | Un1x: Well I guess you'd better show us more... |
| 19:04.46 | [TK]D-Fender | Un1x: Somewhere "*" is considered a disconnect. Go look at the big picture |
| 19:04.55 | Un1x | http://pastebin.com/d61fe2bde |
| 19:05.33 | *** join/#asterisk mog (n=mog@nat/digium/x-20c8f4daf44a534b) |
| 19:05.33 | *** mode/#asterisk [+o mog] by ChanServ |
| 19:06.13 | [TK]D-Fender | Un1x: Go look in features.conf |
| 19:07.15 | *** part/#asterisk PMantis (n=sswitzer@cpe-67-240-239-27.rochester.res.rr.com) |
| 19:08.52 | _ShrikE | Un1x: Take the h out of your dial command |
| 19:09.31 | Un1x | i took the H out and the T and tried and still no look |
| 19:09.31 | Un1x | :( |
| 19:10.53 | LeddyHM | Received SIP subscribe for peer without mailbox: *PEER* <-- This is a secondary "remote" login. Is there a way to suppress these messages? |
| 19:12.04 | [TK]D-Fender | LeddyHM: point the secondary login to the same box as the primary |
| 19:12.20 | jsmith | LeddyHM: Sure... tell that phone to stop subscribing to a mailbox (that doesn't exist!) |
| 19:13.02 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 19:19.24 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 19:21.15 | *** join/#asterisk jtodd (n=jtodd@blob.fox-den.com) |
| 19:21.31 | LeddyHM | I have mailbox=extension for both |
| 19:21.52 | _ShrikE | mailbox=extension@context |
| 19:21.53 | LeddyHM | doh, wrong user |
| 19:22.02 | LeddyHM | mailbox= <-blank |
| 19:22.04 | LeddyHM | correcting |
| 19:22.48 | Un1x | [TK]D-Fender is there a guide you can point me to for the proper way to setup Automon? |
| 19:24.26 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 19:25.35 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 19:25.57 | [TK]D-Fender | Un1x: You probably are using it right, you just completely missed the point that "*" is configured to HANG UP ON YOU. |
| 19:26.09 | [TK]D-Fender | Un1x: Now go look for WHERE that is configured. |
| 19:26.23 | *** join/#asterisk stochastik (n=ircfs@204.246.139.68) |
| 19:26.59 | stochastik | Can anyone recommend a place to buy Polycom IP650? |
| 19:27.20 | jaytee | telephonydepot.com |
| 19:27.30 | stochastik | thanks |
| 19:28.16 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 19:28.19 | jaytee | for a stochastik, anytime. if you were a heuristik I might have reservations :-) |
| 19:29.11 | jameswf | OMFG bri on 1.6 it's so beautiful..... |
| 19:29.15 | stochastik | lol |
| 19:29.54 | jaytee | ZOMG!!! PRI on 1.4.15 it's so ............... normal |
| 19:33.42 | *** join/#asterisk xieliwei (i=dcff041e@gateway/web/ajax/mibbit.com/x-b1fd365c44f3f5ea) |
| 19:34.11 | xieliwei | Agk... Need help with chan_mobile |
| 19:34.34 | xieliwei | it does not seem to work with any of my mobiles, even those listed as compatible |
| 19:34.54 | xieliwei | it just lists them as headsets and says they're not usable |
| 19:35.33 | xieliwei | have tried a motorola V3, nokia e51 and htc p3600 running windows mobile 6.1 |
| 19:35.36 | *** join/#asterisk obmit (n=fanti___@dslb-088-072-102-108.pools.arcor-ip.net) |
| 19:36.05 | *** join/#asterisk AlexTO (n=alex@173.9.143.137) |
| 19:36.14 | xieliwei | tried it on different systems with both asterisk 1.4 and 1.6 trunk running opensuse 10.2 and 10.3 |
| 19:36.23 | AlexTO | hi everyone... |
| 19:37.44 | xieliwei | the only thing i can't change is the dongle, but its csr based and I've got many of them used for other linux bluetooth apps on other systems |
| 19:38.00 | xieliwei | i did try using a different dongle (but same model), no difference |
| 19:46.05 | AlexTO | I have an issue setting up DUNDi, between 2 * boxes, i can make that box one reach extensions to the second one but not in the other direccion, this is my log, if some has any ideas please let me not, thanks http://pastebin.com/m2d04ac10 |
| 19:46.57 | *** join/#asterisk newtonglez (n=username@189.164.131.247) |
| 19:47.40 | *** join/#asterisk n3hxs (n=HAMming@63.68.135.4) |
| 19:47.56 | *** join/#asterisk dynaguy (n=dynaguy@d154-20-51-140.bchsia.telus.net) |
| 19:49.28 | *** join/#asterisk neurosys (n=neurosys@c-66-229-81-161.hsd1.fl.comcast.net) |
| 19:50.52 | AlexTO | there is someone there that knows about DUNDI ?? |
| 19:51.12 | mog | uh hu |
| 19:51.50 | AlexTO | Hi...@mog.. |
| 19:52.11 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 19:52.11 | AlexTO | have you ever setup DUNDi?? |
| 19:53.07 | Un1x | [TK]D-Fender where would that be configured |
| 19:54.20 | mog | yes AlexTO |
| 19:55.07 | AlexTO | Can you give a hand of this.. I have already setup this but i can reach extessions just in one way.. |
| 19:55.33 | AlexTO | Box B --> Reach Ext Ok from Box A |
| 19:56.27 | xieliwei | I have to go, if anyone has any idea, please leave me a message at ytalrselho@mailinator.com thanks! |
| 19:56.34 | *** join/#asterisk lou_gr (n=lou@212-70-216-131.ath.static.tee.gr) |
| 19:56.57 | AlexTO | http://pastebin.com/m2d04ac10 |
| 20:00.10 | *** join/#asterisk ManxPower (n=manxpowe@59.sub-75-249-173.myvzw.com) |
| 20:02.49 | *** join/#asterisk nosbig (n=nosbig@cpe-75-185-59-24.columbus.res.rr.com) |
| 20:06.52 | Katty | has been all anti-socially today |
| 20:09.34 | jameswf | Read today the preseident of ASU makes more $$$ than the president of the USA |
| 20:11.11 | Un1x | LOL but again the power the office of president holds is alot better :P) |
| 20:11.16 | Un1x | some say most powerfull office in the world |
| 20:14.48 | *** join/#asterisk harry_v (n=pcsuppor@S0106001d7e52cc78.vs.shawcable.net) |
| 20:20.17 | AlexTO | Hi ManxPower |
| 20:22.58 | *** join/#asterisk magumbade (n=magumbad@ppp-82-135-93-192.dynamic.mnet-online.de) |
| 20:23.00 | *** join/#asterisk rcy` (n=rcy@S01060002553240a8.vc.shawcable.net) |
| 20:23.47 | *** part/#asterisk fransman (n=frans@a80-127-14-241.adsl.xs4all.nl) |
| 20:23.52 | *** join/#asterisk hi365_m (n=hi365@213.151.44.194) |
| 20:28.11 | harry_v | what are the reasons chan_zap.so not to load? |
| 20:28.30 | Un1x | try lloading, it with via console |
| 20:28.34 | Un1x | you'll see errorr messages |
| 20:28.37 | Un1x | most likely your config |
| 20:28.59 | [TK]D-Fender | harry_v: Bad config, modules not loaded, etc |
| 20:29.39 | harry_v | loader.c:666 load_resource: Module 'chan_zap.so' could not be loaded. |
| 20:29.39 | harry_v | config is fine |
| 20:31.20 | harry_v | dispite that ztcfg-v does say one channel to be configured. I have configured it correctly and reverified even against other known working configs. reloaded asterisk. BTW I am running 1.4 |
| 20:32.48 | *** join/#asterisk erth64net (n=erth64ne@mail.opensourcery.com) |
| 20:33.28 | *** join/#asterisk Aurumzx (n=aurum@router.banktrade.com) |
| 20:33.46 | harry_v | this may be a old probem that did not go away duing the second recompile. I mistakenly compiled ast before zap then relized it should have been the other way around. So the next time, i recompiled zap then asterisk. |
| 20:34.02 | harry_v | did all the steps needed by README |
| 20:34.45 | [TK]D-Fender | harry_v: And you've shown us exactly nothing. |
| 20:35.05 | AlexTO | Hi.. someone can help me http://pastebin.com/m2d04ac10 |
| 20:41.31 | *** part/#asterisk unpaidbill (i=bill@420nugs.info) |
| 20:43.09 | [TK]D-Fender | Yup... some people just really don't want help... |
| 20:43.47 | *** join/#asterisk Greek-Boy (n=greek@41.222.92.254) |
| 20:46.34 | AlexTO | yes... I seem that :-( |
| 20:48.27 | jaytee | anyone ever used astograph.py to get a visual view of their dialplan? |
| 20:49.04 | AlexTO | [TK]D-Fender, do you know how can i track error on Dundi? |
| 20:50.00 | [TK]D-Fender | AlexTO: No. |
| 20:51.09 | *** join/#asterisk TOrrIeri (n=torrieri@nelug/crew/torrieri) |
| 20:56.45 | AlexTO | OK thanks... my problem now is that only works one way.... |
| 20:57.56 | *** join/#asterisk Ritzerisk (n=Ritztech@nv-65-40-156-46.sta.embarqhsd.net) |
| 20:58.30 | Ritzerisk | anyone know of how to get into the ADMIN portion of a Linksys sipura device |
| 21:02.25 | *** join/#asterisk mitcheloc (n=mitchel@adsl-249-77-176.hsv.bellsouth.net) |
| 21:04.05 | hardwire | heya. |
| 21:04.07 | hardwire | punk. |
| 21:06.26 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 21:06.53 | mitcheloc | you callin' me a punk? =P |
| 21:07.09 | Micc | my asterisk is finally back to its rock solid self now that I removed that sql log file that was 2gb. |
| 21:07.58 | neurosys | I am calling into my asterisk box thru one ITSP. Once im connected to my box, i try to make an outgoing call thru another ITSP, but it tells me faliure to invite. |
| 21:08.03 | Micc | It just so happened that the one day it was most important to work right, it had that 2gb limit problem. Out of all the other times in the last few years it had to be that day. |
| 21:08.10 | AlexTO | lsmod |
| 21:08.11 | neurosys | If i connect direct to the 2nd ITSP, it goes thru fine. |
| 21:08.58 | Micc | If anyone wonders how stable asterisk is, its more stable than the OS it runs on. |
| 21:09.17 | *** join/#asterisk mercutioviz (n=chatzill@freeswitch/developer/msc) |
| 21:16.11 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 21:17.43 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
| 21:17.53 | lesouvage | anybody in Europe interested in ca. 70 SNOM320 phones lightly used for a year (normal office setting with a phone call once in a while). |
| 21:18.12 | jasonwoot | anyone have any leads on a company that might actually BUY my old Nortel, instead of just pretending to be interested so they can try and sell me hardware? |
| 21:19.53 | s34n | how can I install ztdummy without stepping on my distro's zaptel pakage? |
| 21:20.24 | s34n | jasonwoot: good luck |
| 21:20.50 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 21:21.03 | *** join/#asterisk LND (n=LND@89.193.210.156) |
| 21:21.33 | *** join/#asterisk CrazyTux (n=brandon@user-vcaur3m.dsl.mindspring.com) |
| 21:22.00 | jasonwoot | why? why must they tease me so? |
| 21:24.00 | lesouvage | jasonwoot: there seem to be a market for it, different models are offered on ebay. Maybe you could give that a try. |
| 21:26.58 | jasonwoot | you know I had it listed twice, first time no bids, second time, bidder asked me to end early and work a deal... they wanted to sell me stuff :^( |
| 21:27.14 | jasonwoot | $75,000 paperweight |
| 21:27.27 | jasonwoot | I should list it that way |
| 21:32.20 | *** join/#asterisk joobie (n=joobie@mx01.anric.com.au) |
| 21:32.39 | n3hxs | I had a CO grade Harris switch and just after I gave it away, the guy calls wanting to buy it. |
| 21:33.43 | s34n | jasonwoot: try Uruguay |
| 21:35.19 | lesouvage | s34n:any advice for me where to sel 70 snom 320 phoes for a fair price? |
| 21:36.27 | s34n | lesouvage: find somebody with a metaswitch. and hurry |
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| 21:42.02 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com) |
| 21:43.44 | tzafrir_laptop | s34n, and your distro is? |
| 21:44.09 | s34n | tzafrir_laptop: fedora |
| 21:45.58 | tzafrir_laptop | I suspect that if you just install the modules (make modules-install) it will happen to work |
| 21:48.05 | s34n | tzafrir_laptop: is modules-install a valid target for make? |
| 21:49.07 | *** join/#asterisk sam555 (n=chatzill@udp136618uds.hawaiiantel.net) |
| 21:49.20 | sam555 | hello all! |
| 21:49.30 | sam555 | I'm possibly going to be new to asterisk. |
| 21:49.51 | sam555 | I was wondering if you were noob, is asterisk easy to understand to run for a company? |
| 21:50.09 | s34n | sam555: it depends on what other experience you have |
| 21:50.37 | sam555 | hmm, just a bit with linux and very little with panasonic pbx kd-1232 |
| 21:51.18 | lesouvage | sam55: start with reading the book. |
| 21:51.23 | lesouvage | ~book |
| 21:51.24 | jbot | somebody said book was probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 21:52.30 | lesouvage | sam55L: there are examples available of ready to go dialplans that will fit your basic requirements after some adjustments. |
| 21:53.20 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 21:53.35 | sam555 | i see |
| 21:54.06 | lesouvage | sam55L: Asterisk isn't something that you learn on a late friday afternoon, it is not a ready to go solution. |
| 21:54.15 | sam555 | i guess what I'm trying to figure out is if I should go with the nec dsx rather than an asterisk because it's easier to run.... |
| 21:54.21 | s34n | sam555: your question is pretty vague. |
| 21:54.33 | *** join/#asterisk dynaguy (n=gao@S0106001346c0e2a7.vc.shawcable.net) |
| 21:54.45 | s34n | How big is the company? How critical is the phone system to the company? |
| 21:55.00 | s34n | How big is your staff? |
| 21:55.19 | s34n | Have you ever admin'ed a unix server before? |
| 21:55.24 | sam555 | the company is about 90 volunteers and 30 staff people |
| 21:55.28 | lesouvage | Do they feel that life without a gui for the phonesystem is not posiible? |
| 21:55.47 | sam555 | the phones are a very high priority for running a retreat center |
| 21:56.13 | *** join/#asterisk LND (n=LND@89.193.214.123) |
| 21:56.24 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 21:56.32 | sam555 | the most admin'ing of a unix server is a Suse P2P file server that was already put together before I got here so all I have to do is run occasional backups |
| 21:57.11 | sam555 | well I was looking at tribox and it was claiming that it had a gui, although I'm not afraid of command line operations, lesouvage |
| 21:57.24 | s34n | sam555: occasional backups? no updating? no log analysis? no basic admin grunt work? |
| 21:58.09 | sam555 | s34n: yeah, no updating or log analysis, although that may change as I transfer the file server to ubuntu |
| 21:58.30 | jaytee | you'd have better results probably running AsteriskNOW 1.5 beta with the FreePBX gui than you would the shipping version of Trixbox |
| 21:58.30 | sam555 | so, yeah, very very basic linux background |
| 21:58.47 | lesouvage | sam555: You should not start your Asterisk carier with setting up a critical system with 120 phones (and a more or less complex dialplan). My advice is to hire an expert and pick up the expertise during the proces of setting up the system. Be sre to have expertise available when the system is running. |
| 21:58.48 | sam555 | hmmm |
| 21:59.09 | s34n | sam555: Properly administrating any server is usually much more work than the casual observer understands. |
| 21:59.09 | sam555 | lesouvage: that's what I was thinking |
| 21:59.20 | sam555 | s34n: agreed |
| 21:59.29 | s34n | sam555: you can admin a linux server with very little effort |
| 21:59.40 | s34n | sam555: but you can't do it properly |
| 21:59.44 | jaytee | trixbox is just * as a core with a forked version of FreePBX gui glommed on and when they forked the gui they mucked it up, then they laid off several of their better engineers |
| 21:59.55 | sam555 | There is a co worker that says he's willing to admin the asterisk, but he knows a lot less than I |
| 22:00.10 | s34n | sam555: (same goes for Windows, or anything else) |
| 22:00.24 | sam555 | There is someone willing to fly out here and set up the asterisks, but after he leaves, I want to be sure that me or my co worker can maintain the asterisks with little for knowledge then what the installer teaches us |
| 22:00.28 | *** join/#asterisk unpaidbill (i=bill@420nugs.info) |
| 22:01.15 | s34n | sam555: you can't |
| 22:01.20 | s34n | don't do it |
| 22:01.30 | unpaidbill | fxs ports should be using the fxoks= option in dahdi/system.conf, correct? and in chan_dahdi.conf it should be using signaling=fxoks? |
| 22:01.36 | lesouvage | sam555: If you don't start using Asterisk for its flexibility in the end it is a matter the total cost of ownership picture. You mentioned an alternative. Do you ahve any idea about the investment that has to be made to get that system up and running for 120 phones? |
| 22:01.41 | s34n | if the phones are important, treat them like they are important |
| 22:01.41 | sam555 | s34n: thanks for info |
| 22:02.30 | sam555 | s34n: indeed! |
| 22:02.55 | lesouvage | sam555: and if you don't have the expertise yoursele you have to count with costs for hiring an asterisk consultant and pay regular fee to have the expertise available during operation if needed. |
| 22:03.52 | sam555 | lesouvage: this is what I figured, although I did find someone yesterday who said they'd be willing to do such a thing, but up until yesterday was wary... |
| 22:03.59 | s34n | sam555: I wouldn't suggest that you try to admin your own mail server either |
| 22:04.08 | sam555 | this is the other system we were possibly going to go for http://www.necdsx.com/index.html |
| 22:04.42 | s34n | or your own physical plant |
| 22:04.45 | sam555 | s34n: the person who is willing to set up the asterisk actually manages are web and mail server |
| 22:05.25 | n3hxs | you can always go take the classes. |
| 22:05.39 | sam555 | n3hxs: true |
| 22:05.58 | s34n | sam555: or your own payroll |
| 22:06.09 | s34n | sam555: the point is, start small and learn |
| 22:06.27 | sam555 | s34n: gotcha |
| 22:06.30 | s34n | sam555: as you gain competence, go ahead and grow if you like |
| 22:06.56 | n3hxs | I have been messing with it for a couple of years or so, and I have broken it badly which isn't good when you have 120 mad people standing behind you. |
| 22:07.10 | s34n | sam555: but don't helicopter into the middle of the Atlantic to learn how to surf. |
| 22:07.12 | sam555 | n3hxs: indeed! |
| 22:07.23 | sam555 | s34n: gotcha |
| 22:07.28 | n3hxs | Lucky I can unplug and plug the home phone directly to keep the wife happy :) |
| 22:07.37 | lesouvage | sam555: if you have a budget and want to make use of a proper graphical user interface you should check out scopserv. It isn't perfect but to my experience it is the best gui available. |
| 22:07.38 | sam555 | s34n: i just needed outside advise because of course I'm being told it's "easy" |
| 22:07.58 | n3hxs | easy for home when you have time to learn. |
| 22:08.14 | *** join/#asterisk JayTee52 (n=jforde@unaffiliated/jaytee) |
| 22:08.25 | n3hxs | but in the long run, you will pay for support with either system for a commercial endeavor. |
| 22:08.35 | n3hxs | or "should" |
| 22:08.56 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 22:09.02 | s34n | sam555: it's very cool, and sometimes it's fun. But you don't hobyy around when hundreds of people are depending on it |
| 22:09.10 | JayTee52 | quittin time, be back later |
| 22:09.25 | sam555 | n3hxs: that's what I was figuring, with the dsx we will have support, with asterisk, I would need to find support other than the 2 of us |
| 22:09.38 | sam555 | s34n: that's exactly what I'm thinking! |
| 22:09.39 | s34n | sam555: I'm not saying that asterisk can't do the job for you. I'm sure it can. |
| 22:10.01 | n3hxs | they are out there. but there may be more needed than just the server and IP phones. |
| 22:10.34 | sam555 | s34n: what I'm hearing you say is it requires admin work. You can't just install it and leave it alone. |
| 22:10.47 | *** part/#asterisk mercutioviz (n=chatzill@freeswitch/developer/msc) |
| 22:10.57 | s34n | sam555: most organizations with 100+ people pay tens of thousands of dollars a year or more on their communications systems |
| 22:10.59 | n3hxs | Network infrastructure can really kill a good VOIP system if it does not meet the requirements, that could cost even more. |
| 22:11.22 | s34n | sam555: you could install it and leave it alone |
| 22:11.38 | s34n | sam555: if you don't care about the security of your system |
| 22:11.51 | s34n | sam555: do you leave any of your servers alone? |
| 22:12.04 | lesouvage | sam555: One of the thinks that often needs improvement when setting up a voip telephone system is the network. A proper network assesment before migrating to voip for internal phonetraffic is not a luxery and it can help a lot to avoid very serious problems later on in the proces. |
| 22:12.12 | sam555 | s34n: ? |
| 22:12.16 | *** join/#asterisk AJayMN (i=0cc08974@gateway/web/ajax/mibbit.com/x-f3fe3a9fc930c590) |
| 22:12.29 | AJayMN | Anyone using AsteriskNow 1.5? wondering how stable it is. |
| 22:12.58 | s34n | sam555: this will be a linux server with access from the public, right? |
| 22:13.15 | sam555 | s34n: lesouvage n3hxs you guys are VERY helpful. I needed this before I go to the meeting to discuss our options |
| 22:13.19 | sam555 | s34n: no, access from the public |
| 22:13.21 | s34n | sam555: any public facing network is a security risk |
| 22:13.37 | lesouvage | sam555; you are going to the meeting now? |
| 22:13.40 | s34n | sam555: you won't take any phone calls from outside your building? |
| 22:14.12 | sam555 | lesouvage: at 3pm hawaii time |
| 22:14.24 | sam555 | s34n: i see what you're saying |
| 22:14.36 | sam555 | s34n: yes, we will be taking outside calls |
| 22:14.42 | n3hxs | How do you get the calls into your building now? |
| 22:14.47 | s34n | sam555: are you in Hawaii, or is your consultant? |
| 22:15.22 | sam555 | n3hxs: we have 2 main lines people call into and then that is directed to extensions. All this is on a panasonic kd-1232 pbx |
| 22:15.25 | lesouvage | That sounds like an interesting place to do some Asterisk consultant work ;-) |
| 22:15.27 | s34n | isn't it already after 3pm in Hawaii? |
| 22:15.29 | *** part/#asterisk AJayMN (i=0cc08974@gateway/web/ajax/mibbit.com/x-f3fe3a9fc930c590) |
| 22:15.31 | sam555 | s34n: i'm in hawaii, the consultant is on the mainland |
| 22:15.44 | *** join/#asterisk CrazyTux (n=brandon@user-vcaur3m.dsl.mindspring.com) |
| 22:15.51 | sam555 | no i'ts 12:15 pm |
| 22:16.03 | s34n | lesouvage: I think the company hosting voip-info is in Hawaii. |
| 22:16.05 | n3hxs | sam555 I would fly to Hawii to install... If I knew more. |
| 22:16.30 | s34n | compartners has a strong presence in Hawaii |
| 22:16.43 | sam555 | n3hxs: we already have someone willing to come here and install is, i just want to know how much upkeep it requires after it's installed |
| 22:17.25 | n3hxs | changes can be done remotely if needed, with good people on site, which would be you. |
| 22:17.25 | s34n | sam555: * requires adding and dropping extensions, etc |
| 22:17.30 | s34n | usual stuff |
| 22:17.50 | s34n | sam555: linux requires regular updating, etc. usual stuff |
| 22:17.56 | n3hxs | sorry I keep forgetting to add the nick :( |
| 22:17.58 | *** join/#asterisk am88b (i=siim@uba.linux.ee) |
| 22:18.09 | s34n | sam555: publicly accessible network requires serious attention |
| 22:18.14 | n3hxs | sam555 the bad part is when an upgrade goes bad. |
| 22:18.22 | am88b | Hello. Can someone please explain me why http://bugs.digium.com/view.php?id=13907 was closed in such way? I read bug guidelines and didn't find anything wrong with my bug report. |
| 22:19.10 | *** join/#asterisk vicom (n=Sam@ves1.vicomnet.com) |
| 22:19.17 | sam555 | s34n: can the updating and adding extensions be done remotely? Remote admin? |
| 22:19.37 | s34n | sam555: have you ever done remote admin? |
| 22:19.47 | s34n | sam555: it's all good until it isn't. |
| 22:20.06 | s34n | Then you have a serious problem until you can get on site |
| 22:21.02 | s34n | remote admin of anything usually requires a local body who can take dictation over the telephone when things go bad |
| 22:21.09 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 22:21.26 | n3hxs | s34n I have that problem often with 32 sites across the US and Canada, and no one at the site knows where the damn machine is! |
| 22:21.33 | s34n | and a willingness to pay last minute airfair, overtime, and per diem |
| 22:22.32 | s34n | n3hxs: I hate it when they bounce the AC instead of the network switch. |
| 22:22.50 | sam555 | excuse me if I get booted |
| 22:22.54 | sam555 | will be back though |
| 22:22.55 | n3hxs | <PROTECTED> |
| 22:22.59 | s34n | Or when they have no idea how to open the panel on the generator |
| 22:23.03 | n3hxs | later, goin home. |
| 22:25.42 | kerx | what does outgoingspoolfailed mean? |
| 22:26.42 | sam555 | well I feel I know the basics, but I just shot and email to the guy who lives here and is willing to admin if necessary |
| 22:26.42 | sam555 | so if I have him as a back up, I will feel must more inclined to have the asterisk system in place |
| 22:28.13 | *** join/#asterisk jcordell (n=jcordell@79-75-199-3.dynamic.dsl.as9105.com) |
| 22:28.25 | *** join/#asterisk trnzmeta (n=bleh@iinet.guard.com.au) |
| 22:28.32 | *** join/#asterisk awk_r (n=rawk@nat/digium/x-ce9cab9dd9b36115) |
| 22:29.37 | jcordell | I have written a little patch for asterisk made a diff and want to submit it. my question is how? |
| 22:30.44 | s34n | FATAL: Error inserting ztdummy (/lib/modules/2.6.26.6-49.fc8/misc/ztdummy.ko): Invalid module format |
| 22:30.53 | seanbright | jcordell: http://bugs.digium.com/ |
| 22:31.36 | seanbright | jcordell: you'll have to sign the contributers license before uploading your diff, though. |
| 22:32.49 | *** join/#asterisk rickross (n=rickross@supporter/active/rickross) |
| 22:32.55 | s34n | tzafrir_laptop: make install-modules seems to have stepped on my zaptel |
| 22:33.12 | rickross | anyone know a command to kill iax2 channels that seem to be stuck open? |
| 22:33.19 | s34n | tzafrir_laptop: and didn't build a working ztdummy |
| 22:33.30 | mog | rickross, soft hangup iax2/chan |
| 22:33.46 | rickross | thank you - is that per channel? |
| 22:34.26 | mog | yes |
| 22:34.37 | rickross | if I use "iax2 show channels" it lists 7 of them, no calls are presently active |
| 22:34.42 | sam555 | thanks again, everyone for your help! |
| 22:35.03 | rickross | and none of the channels it lists actually have a name/number in the first column of the results |
| 22:35.32 | *** part/#asterisk beek (n=klinebl@65.211.106.242) |
| 22:37.16 | shriven | Might anyone be able to shed some light on this error message? The peer has a host=76.182.80.161 specified... I can't figure why it thinks it should be dynamic. |
| 22:37.17 | shriven | register_verify: Peer 'bubbletastic' is not dynamic (from 76.182.80.161) |
| 22:39.21 | jtodd | Anyone know what Polycom phones support Siren7 and/or Siren14? I want to do some end-to-end tests. |
| 22:40.14 | *** join/#asterisk carpenike (n=ryanholt@82.ecb7d1.client.atlantech.net) |
| 22:44.09 | [8none1] | jtodd: I believe it's just the 550/560/650/670 it's their G.722.1 impl |
| 22:44.35 | *** join/#asterisk echelon (i=Unknown@gateway/tor/x-f31c0e8f808e6778) |
| 22:44.53 | echelon | hi, does the PAP2T-NA allow you to change the user-agent string or headers? |
| 22:45.07 | *** join/#asterisk TOrrIeri (n=torrieri@nelug/crew/torrieri) |
| 22:46.47 | *** join/#asterisk `Sean (n=Shaan@CPE0016d3fe20a0-CM0014045acc3c.cpe.net.cable.rogers.com) |
| 22:51.03 | jtodd | 8none1: Thanks. |
| 22:51.54 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 22:55.51 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:58.04 | *** join/#asterisk philippel (n=p_lindhe@pool-98-111-70-106.sttlwa.fios.verizon.net) |
| 22:59.46 | philippel | question - does the Pickup() application 'hangup' if a call to Pickup() fails on asterisk 1.2? On 1.4 you can have a series of attempts (different contexts) that will get executed one after the other, on a couple 1.2 systems I tested, it goes to hangup after the first call to Pickup() if not successful? |
| 23:01.07 | *** join/#asterisk fujin (i=aj@junglist.gen.nz) |
| 23:01.34 | fujin | Anyone recommend a manufacturer of 'hotphones' / 'batphones'? need to program some and stick them to the wall throughout the datacentre here |
| 23:02.19 | [TK]D-Fender | fujin: Any analog phone + ATA |
| 23:02.25 | fujin | would rather something that doesn't have buttons. |
| 23:02.30 | [TK]D-Fender | philippel: Depends who Pickup is getting called |
| 23:02.43 | [TK]D-Fender | fujin: You can get analog handsets without any buttons |
| 23:02.48 | [TK]D-Fender | fujin: Easy enough |
| 23:02.54 | fujin | hrm |
| 23:03.17 | fujin | was hoping for PoE SIP, no-button wallmountable phones |
| 23:03.21 | philippel | [TK]D-Fender directed call pickup and in the instruction Pickup() - are there more then one (other than the BRI version which I'm not refering to)? |
| 23:05.05 | [TK]D-Fender | fujin: You know what the likely hood of finding a SIP phone (deluxe to the rest of the world) with NO buttons or features? yOU'VE INSANED ;) |
| 23:05.20 | [TK]D-Fender | philippel: aND HOW IS THE APP BEING CALLED? |
| 23:05.21 | fujin | I've seen PoE SIP phones with red flashy lights |
| 23:05.25 | fujin | just don' trecall the manufacturer. |
| 23:05.26 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
| 23:05.46 | *** join/#asterisk CunningPike_ (n=arodgers@204.239.8.157) |
| 23:05.53 | fujin | like a president call, yo |
| 23:05.55 | [TK]D-Fender | fujin: I'm sure there must be one out there, but your odds are low, and your cost high..... |
| 23:06.24 | philippel | [TK]D-Fender for example exten => **200,1,Pickup(200@ext-local) |
| 23:06.40 | philippel | and then an instruction after that if that one fails, which is never reached |
| 23:06.45 | [TK]D-Fender | philippel: well if you want it to fail over add more priorities. |
| 23:07.03 | [TK]D-Fender | philippel: pastebin is your friend <- |
| 23:07.09 | [TK]D-Fender | ~pb |
| 23:07.09 | 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, or , http://bin.cakephp.org/ |
| 23:08.00 | philippel | [TK]D-Fender here's the crux of the question, the exact same code on 1.4 will execute all the instructions, on 1.2 it fails after the first and hangsup |
| 23:08.18 | [TK]D-Fender | philippel: PASTEBIN |
| 23:09.22 | philippel | for example: http://pastebin.ca/1259810 |
| 23:09.34 | philippel | runs through fine on 1.4 but not on 1.2 |
| 23:11.54 | fujin | ~batphone |
| 23:12.16 | [TK]D-Fender | philippel: and the call? and the dialplan dump? |
| 23:14.01 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 23:14.01 | philippel | [TK]D-Fender unfortunatley I don't have time right now, I thought someone might know of something specific to 1.2 vs. 1.4, I'll have to come back to that and get more details when I can spend more time, since it sounds like there is not a "yes things changed in 1.4" answer... |
| 23:14.40 | [TK]D-Fender | philippel: The proof is often hidden in little things that actual physical evident bring to light. |
| 23:14.58 | [TK]D-Fender | philippel: Maybe when you come back with some we can see if things are the way you think & claim them to be |
| 23:15.18 | [TK]D-Fender | evidence* |
| 23:16.48 | *** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 23:17.23 | philippel | [TK]D-Fender I know -which is why I need to wait on this, since it may be more time consuming then I have right now (and don't want to ask others to take their time to help me unless I can concentrate on it as well :) |
| 23:19.35 | fujin | [TK]D-Fender: just got off the phone with my polycomm rep, he's going to have someone call me back |
| 23:19.38 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:19.38 | fujin | maybe they have something for me |
| 23:19.42 | fujin | [TK]D-Fender: thanks, as usual |
| 23:20.07 | [TK]D-Fender | fujin: Well I'd be surprised if you find anything other than what you can see right on their site... |
| 23:20.55 | fujin | yeah, at this stage it'd be cool to see how they've met other customers requirements (if at all) |
| 23:20.58 | fujin | worth a shot I suppose |
| 23:22.15 | fujin | anyway, thanks |
| 23:22.16 | *** part/#asterisk fujin (i=aj@junglist.gen.nz) |
| 23:34.40 | *** join/#asterisk jonasb (n=jonas@87.112.83.4.plusnet.ptn-ag2.dyn.plus.net) |
| 23:35.06 | *** join/#asterisk BBHoss (n=bbhoss@user-24-214-210-231.knology.net) |
| 23:39.39 | jonasb | hi all, <newbie alert/> if I were get a number of "virtual" telephone numbers, and redirect calls to some proper telephone numbers, would Asterisk be the way to go? |
| 23:40.32 | awk_r | jonasb, yes |
| 23:40.38 | awk_r | s/yes/sure/ |
| 23:40.55 | awk_r | hugs jbot. |
| 23:42.35 | BBHoss | jonasb: might be overkill for just redirecting media, but its probably easier to configure to do so |
| 23:43.05 | jonasb | when I say "virtual" telephone numbers what do I actually mean? :-) |
| 23:43.28 | [TK]D-Fender | jonasb: Odds are whatever provider you get your DID's from might be able to do the switch directly without you even having to configure anything |
| 23:43.33 | [TK]D-Fender | ~did |
| 23:43.33 | jbot | hmm... did is Direct Inward Dialing, or just a phone number |
| 23:43.36 | [TK]D-Fender | ^^^^ |
| 23:44.36 | BBHoss | jonasb: all you want is simple call routing, asterisk is a full blown PBX. [TK]D-Fender is correct about your provider possibly supporting those routing options |
| 23:45.03 | [TK]D-Fender | BBHoss: And no, * is not a PBX. It is a toolkit you can use to build a PBX |
| 23:45.56 | jonasb | thanks, will google for DID. there are two things I want 1) to easily configure the redirections, and 2) to monitor the call charges for each number and disconnect when a certain limit is reached |
| 23:46.20 | BBHoss | [TK]D-Fender: but you must agree that it is geared toward pbx duties rather than high volume call routing |
| 23:47.23 | drmessano^ | BBHoss: Most of the horrid performance cited in blogs/wikis with Asterisk in a high volume arena were gathered during the early 1.2 era |
| 23:47.24 | awk_r | [TK]D-Fender, "Asterisk is the world's leading open source PBX" -- http://www.asterisk.org |
| 23:47.32 | drmessano^ | Even the last year has different |
| 23:47.38 | drmessano^ | Even the last year has been different |
| 23:47.48 | awk_r | only because this is the 2nd time I've heard you say that...and yes...i agree with you and understand what you are trying to say |
| 23:47.56 | BBHoss | drmessano^: its not like i could actually judge anyways as i don't have a high volume environment |
| 23:48.04 | [TK]D-Fender | BBHoss: Yes, but who said anything about high volume? |
| 23:48.18 | [TK]D-Fender | awk_r: MARKETING <- |
| 23:48.56 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 23:49.12 | BBHoss | [TK]D-Fender: i was giving to extreme examples, pbx only on one side, and high volume call router on the other, asterisk lies somewhere in the middle but my point was it leans more toward pbx duties. |
| 23:49.33 | drmessano^ | People run ITSP's off asterisk |
| 23:49.34 | awk_r | BBHoss, that is severely limiting Asterisk |
| 23:50.17 | [TK]D-Fender | PPBX is what something does. "Volume" is how much it can do of something bfore it chokes. Like comparing the weight of one car against the top speed of another. Which is better? |
| 23:50.20 | awk_r | BBHoss, "most applications of Asterisk lean toward pbx duties", not necessarily the majority of Asterisk's features or development |
| 23:50.41 | [TK]D-Fender | BBHoss: PBX is what something does. "Volume" is how much it can do of something bfore it chokes. Like comparing the weight of one car against the top speed of another. Which is better? |
| 23:50.48 | BBHoss | awk_r: i wasn't trying to limit asterisk, just make a point |
| 23:51.16 | drmessano^ | You werent making a point, you were making a statment |
| 23:51.16 | [TK]D-Fender | You can use * as a JUKEBOX, or a CRON replacement, or to have your computer schedule COFFEE. Or just about anything else. |
| 23:51.29 | *** join/#asterisk harry_v (n=pcsuppor@S0106001d7e52cc78.vs.shawcable.net) |
| 23:51.42 | drmessano^ | Saying "Asterisk leans towards being a PBX" is not a *point*, it's a *statement* |
| 23:51.45 | jaytee | I usually use mine as a hybrid PBX with VOIP and PRI :-) |
| 23:51.51 | awk_r | [TK]D-Fender, mine reads my gmail for me...can i call it my email inbox? |
| 23:52.03 | drmessano^ | A point would be <bunch of facts used to bring together a single thought> |
| 23:52.19 | [TK]D-Fender | awk_r: No... your inbox is still with gmail. That would make * your e-mail READER :) |
| 23:53.26 | awk_r | BBHoss, sorry...you hit a touchy subject :-P |
| 23:54.27 | drmessano^ | Its not a matter of it being "touchy" |
| 23:54.31 | BBHoss | ok, maybe I am wrong, but personally, when I think Asterisk, I think highly versatile PBX/Voice app builder, but when I think about something like YXA or OpenSER, I think high volume call router |
| 23:54.35 | drmessano^ | why does everyone thing everything is about emotions.. |
| 23:54.38 | drmessano^ | think* |
| 23:55.20 | awk_r | sighs at drmessano^. |
| 23:55.38 | awk_r | s/touchy/commonly argued/ |
| 23:55.45 | coppice | personally when I think Asterisk, I think something that needs combining with "rm -rf" to piss off someone I dislike |
| 23:55.47 | drmessano^ | Everyone is entitled to their opinion, but fact is, there no truth in the statement that 'Asterisk leans towards being a PBX" |
| 23:56.17 | drmessano^ | Its could be an opinion, maybe a somewhat common opinion |
| 23:56.30 | drmessano^ | But theres no basis for the statement to be fact |
| 23:56.31 | rickross | can zaptel be uninstalled simply by deleting the zaptel.ko files in /lib? |
| 23:56.42 | harry_v | TK, as stated earlier "Not enough info provided" I suspected at least you would mention if Dahdi was installed. I was totally unaware of the change from zap to Dahdi BUT you could have mentioned it. |
| 23:57.52 | [TK]D-Fender | drmessano^: Well take a list of the features that * has built in that are of PBX origin (VM app, call processing, IVR, etc. That would be PBX handling features. Now look at the parts that AREN'T PBX related necessarily (many of the generic dialplan apps). I would say that the amount of PBX type features is quite prominent. Now what you DO witht he pieces you have is another matter |
| 23:58.45 | rickross | this page suggests there is a "make uninstall" for zaptel, but I don't see it - http://www.mail-archive.com/asterisk-users@lists.digium.com/msg211854.html |
| 23:58.45 | drmessano^ | I wasnt aware open source software was capable of expressing it's use using action verbs |
| 23:58.52 | [TK]D-Fender | harry_v: You had nothing to show us at all. No configs, not "ztcfg -vvvv", no interupts lists, DEV dumps proving your cards driver was loaded or ANYTHING of value. You want to tell me that you got off your ass in the slightest in looking for help? |
| 23:58.54 | awk_r | I still say the 'proper'y statement would be: "most Asterisk applications are PBX-related" |
| 23:59.39 | [TK]D-Fender | harry_v: "it doesn't work". I looked at 'stuff'". This doesn't offer us much, does it? |
| 23:59.49 | jaytee | harry_v, if you're running 1.6 you'd know about the zaptel/DAHDI change if you'd read the UPGRADE.TXT file or did a little research up front. |
| 23:59.57 | BBHoss | one thing i want to know is why can't we get a decent dialplan language? Most sane people now just drop into AGI to write things, but should we really have to do that? |