| 00:01.39 | *** join/#asterisk propellerhead (n=yogurt2u@host66.190-31-153.telecom.net.ar) |
| 00:02.19 | raden_work | drmessano, u think all these issues could be in the router ? |
| 00:04.18 | raden_work | well im gone |
| 00:04.22 | raden_work | nite everyone |
| 00:13.54 | *** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 00:21.06 | *** join/#asterisk coppice (n=chatzill@193.194.17.210.dyn.pacific.net.hk) |
| 00:21.12 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 00:29.02 | *** part/#asterisk vicscandl (n=vicscand@adsl-99-130-13-113.dsl.irvnca.sbcglobal.net) |
| 00:41.14 | *** join/#asterisk Twotone (n=test@24-158-150-143.dhcp.jcsn.tn.charter.com) |
| 00:42.20 | Twotone | Anyone know how to make 1 extension forward to 2 different extensions at the same time? |
| 00:42.57 | drmessano | SIP/1&SIP/2 |
| 00:43.20 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 00:44.16 | Twotone | I want 300 to forward to 301 and 302 |
| 00:45.29 | sirus | wants a working asterisk setup |
| 00:48.15 | [TK]D-Fender | wants one meeeeelion dollars! |
| 00:48.27 | sirus | me to |
| 00:48.41 | [TK]D-Fender | pulls a Dr. Evil |
| 00:48.52 | sirus | :( |
| 00:49.34 | sirus | Dame a copy of someone elses working extensions.conf and sip.conf doesn't seem to let me connect at all :( |
| 00:50.17 | sirus | Do I need to get a provider first to do IP to IP calling? |
| 00:50.32 | [TK]D-Fender | Sirdepends who's IP you're calling |
| 00:50.41 | [TK]D-Fender | sirus: depends who's IP you're calling |
| 00:51.28 | sirus | mine |
| 00:51.39 | JonCup | hey guys, having a issue with a softphone, it registers with asterisk, and I make calls out, but I cant recieve calls, and sip show peers shows the client UNREACHABLE, any insight? |
| 00:51.50 | sirus | I was calling mine I setup two extenstions I used xlite to connect it just fails |
| 00:51.53 | sirus | ;/ |
| 00:52.13 | [TK]D-Fender | sirus: can you describe how it is you are "calling yourself"? (they have padded rooms for people who do that IRL) |
| 00:52.14 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 00:52.34 | [TK]D-Fender | sirus: If you elaborate on "just fails" we might be able to help. |
| 00:52.48 | JonCup | sipphone in question here is sipdroid, for the android platform. supposedly works for some people |
| 00:53.00 | [TK]D-Fender | JonFirst guess sounds like the X-Lite is behind a remote NAT and it times out |
| 00:53.58 | [TK]D-Fender | s/X-Lite/your softphone/ |
| 00:54.13 | sirus | [TK]D-Fender: Well perhaps I havn't configured the softphone correctly but when I say fail it times out on the registration and if I disable the registration it lets me dial numbers but doesn't actually call anyone it just says the person you called can't be reached or is unavailable |
| 00:54.36 | [TK]D-Fender | sirus: what "says" that? |
| 00:54.42 | sirus | Xlite |
| 00:54.45 | sirus | the Softphone |
| 00:55.02 | [TK]D-Fender | sirus: enable SIP DEBUG at * CLI and look for the incoming call attempt |
| 00:56.37 | JonCup | Fender: no nat involved, my asterisk server has two interfaces, 192.168.11.1 and 192.168.10.20, my aastra phones are on the .11.x subnet, and my sipdroid softphone is on the .10.x subnet, asterisk server is set to route packets betweek networks, and routers on both sides are set with the correct default routes, and communications work both ways from both subnets as I have tested |
| 00:56.40 | sirus | how do I look for the incomming call |
| 00:56.58 | sirus | [TK]D-Fender: can I give you the login information and you try? |
| 00:57.08 | [TK]D-Fender | JonCup: pastebin your call attempt and your sip.conf masking only passwords |
| 00:57.31 | [TK]D-Fender | sirus: Just work on this yourself for now. "sip set debug" <- enter at * CLI |
| 00:58.08 | Twotone | How would one add extension 300 to forward to 301 and 302 when dialed. What conf files would need edited and what context added to them? |
| 00:58.08 | sirus | I did |
| 00:58.30 | [TK]D-Fender | sirus: And what does it say? |
| 00:58.46 | sirus | it enabled the debugging |
| 00:58.56 | sirus | but I dont see anything happening when I try to connect with xlite |
| 00:59.15 | [TK]D-Fender | Twotone: there is no such thing as "forwarding". That is a logical concept for which when you dial a specific number it makes a decision as to what to ring. |
| 00:59.37 | [TK]D-Fender | sirus: that indicates a networking issue. Where is * relative to your softphone? |
| 00:59.40 | JonCup | Fender: http://pastebin.com/d215d5610 |
| 01:00.08 | [TK]D-Fender | Twotone: and the act of deciding how to handle a call = extensions.conf |
| 01:00.53 | [TK]D-Fender | JonCup: "sip show peer 204" |
| 01:01.10 | sirus | hrmm |
| 01:01.15 | sirus | the network is up |
| 01:01.20 | sirus | on both ends |
| 01:02.22 | [TK]D-Fender | Sirthat is a very generic description. any number of firewalls may exist along this path. If you've enabled SIP DEBUG properly and you see nothing then packets aren't making it to * |
| 01:02.56 | sirus | ok now we are getting some where |
| 01:03.00 | sirus | disabled the firewall |
| 01:03.06 | JonCup | Fender: http://pastebin.com/d43366e66 |
| 01:03.12 | sirus | got lots of stuff happening but still the call is unavailable |
| 01:03.58 | [TK]D-Fender | sirus: "lots of stuff" = important stuff to actually look at |
| 01:04.34 | sirus | hrmm ok i'll pastebin |
| 01:04.48 | [TK]D-Fender | JonCup: enable SIP debug and call again |
| 01:05.14 | sirus | http://pastebin.com/m71b261fc |
| 01:05.18 | [TK]D-Fender | JonCup: and include your sip.conf and every file it links to. |
| 01:05.46 | [TK]D-Fender | sirus: There is a LOT more that that, |
| 01:06.22 | sirus | my sip.conf http://pastebin.com/m33c42a5e |
| 01:06.29 | sirus | Hrmm I'll do it again |
| 01:08.37 | sirus | It wont let catch all of it but i'll paste what I have |
| 01:08.56 | sirus | http://pastebin.com/m973826 |
| 01:09.03 | [TK]D-Fender | sirus: Get a better client if you can't get more scroll-back |
| 01:09.33 | sirus | hrmm |
| 01:09.42 | [TK]D-Fender | sirus: <--- SIP read from 24.33.65.121:56961 ---> <-- where the call came from |
| 01:09.42 | sirus | I thought putty would be the best |
| 01:09.56 | sirus | yep |
| 01:10.06 | [TK]D-Fender | sirus: Retransmitting #2 (no NAT) to 192.168.1.103:33562: <-- where * RESPONDED to |
| 01:10.08 | [TK]D-Fender | sirus: BAD |
| 01:10.18 | sirus | :( |
| 01:10.23 | [TK]D-Fender | sirus: This device is behind a remote NAT and you did not specify nat=yes |
| 01:10.40 | sirus | hrmm in sip.conf? |
| 01:11.02 | [TK]D-Fender | sirus: yes |
| 01:11.15 | Twotone | [TK]D-Fender, What would a sample of what I'm wanting look like in extensions.conf? |
| 01:11.18 | sirus | damn how did i miss this |
| 01:11.23 | sirus | I'll retry again |
| 01:11.33 | [TK]D-Fender | SirDoes your * have a public IP? |
| 01:12.17 | [TK]D-Fender | Twotone: Depends where you store the information about where you want the call to go, along with the fact that it should indeed take that action. |
| 01:12.58 | *** join/#asterisk cryptanthus (n=newview@ip98-161-25-9.om.om.cox.net) |
| 01:14.06 | Twotone | If I could log into my asterisk box I'd show you how 1 extension was layed out but my IP changed here so I can't. I'll see what I can do about adding it |
| 01:14.41 | *** join/#asterisk Kumbang (n=enit@rusnas.paume.itb.ac.id) |
| 01:15.18 | sirus | [TK]D-Fender: yes my asterisk box has a static public ip |
| 01:15.21 | sirus | http://pastebin.com/m2e35815b |
| 01:15.29 | sirus | Now it just goes to a busy signal |
| 01:15.38 | sirus | soon as I call an extension |
| 01:15.40 | DigitalProcessor | So, I have 100,000 bucks I can play with, I am thinking of getting polycom phones and setting up an asterisk server? What do you guys think? |
| 01:15.41 | [TK]D-Fender | sirus: OK, then only your remote client needs "nat=yes", "qualify=yes" |
| 01:15.54 | sirus | Ahh ok |
| 01:15.56 | *** join/#asterisk Alfio (n=Amunoz@adsl-55-172.tricom.net) |
| 01:16.06 | [TK]D-Fender | DigitalProcessor: How many do you need? |
| 01:16.18 | *** join/#asterisk Deeewayne (n=dwayne@76.29.245.9) |
| 01:16.18 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 01:16.18 | DigitalProcessor | i will need about 320 phones |
| 01:16.27 | [TK]D-Fender | sirus: Oh, and "canreinvite=no" |
| 01:16.53 | [TK]D-Fender | DigitalProcessor: More than enough |
| 01:17.31 | DigitalProcessor | Fender, I mean should I just do a aterisk setup (I love linux) or should I just go with hosted site? |
| 01:17.46 | [TK]D-Fender | DigitalWhere will all the phones be? |
| 01:17.53 | DigitalProcessor | All in NY |
| 01:17.58 | DigitalProcessor | all our stores are in NY |
| 01:18.08 | DigitalProcessor | within the 5 boroughs of NY |
| 01:18.10 | sirus | Let me make the changes |
| 01:18.12 | [TK]D-Fender | DigitalProcessor: I recommend a server per store |
| 01:18.35 | DigitalProcessor | fender that is EXACTLY what I was thinking |
| 01:18.37 | DigitalProcessor | but why? |
| 01:18.57 | [TK]D-Fender | didBecause if centra goes down they can still call inside of the building, and have their own failover |
| 01:19.16 | [TK]D-Fender | DigitalProcessor: Think 911 requirements, etc |
| 01:19.34 | DigitalProcessor | Fender: hosted service provides that |
| 01:19.39 | DigitalProcessor | op;' |
| 01:19.46 | DigitalProcessor | my question is |
| 01:19.47 | [TK]D-Fender | DigitalProcessor: if your INTERNET fails, you're DEAD. |
| 01:20.01 | DigitalProcessor | no, hosted site will failover to POTS line |
| 01:20.02 | [TK]D-Fender | DigitalProcessor: Think bigger |
| 01:20.18 | DigitalProcessor | but if internet fial |
| 01:20.19 | DigitalProcessor | fail |
| 01:20.22 | [TK]D-Fender | DigitalIf you can't readch the bloody hosting server because the STORE's internet fails <-------- |
| 01:20.22 | DigitalProcessor | even with asterisk |
| 01:21.06 | DigitalProcessor | Fender, If store internet fails, HOSTED service see's this. Then it will call POTS line. |
| 01:21.18 | DigitalProcessor | forward calls to POTS line |
| 01:21.25 | [TK]D-Fender | DigitalProcessor: And what will receive the call? |
| 01:21.36 | DigitalProcessor | The phone that is hooked up on the other end. |
| 01:21.41 | DigitalProcessor | =/ |
| 01:21.49 | [TK]D-Fender | DigitalProcessor: If all you have a re polycom phones at the store, what will make the call reach them? |
| 01:22.20 | DigitalProcessor | No, no, we have polycoms for VoIP right. But we will have 1 REGULAR phone hooked up to POTS JUST IN CASE |
| 01:22.30 | DigitalProcessor | if internet goes down |
| 01:22.45 | [TK]D-Fender | DigitalProcessor: Yes, and people will have to know the connection is down and run to it in an emergency. |
| 01:22.50 | [TK]D-Fender | DigitalProcessor: not wonderful... |
| 01:23.17 | DigitalProcessor | okay so with asterisk it will forward calls to polycom phone? |
| 01:23.18 | [TK]D-Fender | DigitalProcessor: thats why you should have an * server per site to handle failover within each store |
| 01:23.24 | DigitalProcessor | okay okay |
| 01:23.28 | DigitalProcessor | but |
| 01:23.33 | *** join/#asterisk phunyguy (n=phunyguy@h69-130-65-35.kgldga.dsl.dynamic.tds.net) |
| 01:23.39 | DigitalProcessor | will polycom phone ring even when internet is down |
| 01:23.54 | sirus | hrmm now I can only call myself |
| 01:23.56 | DigitalProcessor | and outgoing calls will go through POTS in the event of internet fail over |
| 01:23.59 | [TK]D-Fender | DigitalProcessor: when 1 store calls an extension that should be at another location, the local * will call out to a remote *. |
| 01:24.04 | DigitalProcessor | internet failure* |
| 01:24.04 | sirus | no other extensions |
| 01:24.31 | [TK]D-Fender | sirus: "sip show peer [theotherdeviceherewithoutbraces]" |
| 01:25.13 | [TK]D-Fender | DigitalProcessor: more than that witha local server calls to a local device doesn't have to go out the internet only to come right back in wasting bandwidth |
| 01:25.24 | *** join/#asterisk phunyguy (n=phunyguy@h69-130-65-35.kgldga.dsl.dynamic.tds.net) |
| 01:25.43 | sirus | hrmm ok |
| 01:26.01 | sirus | [TK]D-Fender: the person using the other extension would have to be signed into the softphone? |
| 01:26.36 | sirus | oh I see what that does |
| 01:26.38 | [TK]D-Fender | sirus: I don't know.. can you make your friend's cell-phone ring while its off & the batter is removed? :) |
| 01:26.45 | sirus | hrmm my damn extension doesn't even exist |
| 01:26.46 | DigitalProcessor | Fender. The only way stores will be connected to other stores is via internet. So.. Polycom--asterisk-server-at-local-store--------------------internet----------------asterisk-server-at-another-store-----------polycomm |
| 01:26.58 | sirus | [TK]D-Fender: good point |
| 01:27.28 | [TK]D-Fender | DigitalProcessor: Sure. |
| 01:27.33 | DigitalProcessor | if one internet goes down at one point. no, way to reach other end. |
| 01:27.48 | KavanS | if internet goes bye bye, you will want to get "high high" |
| 01:27.54 | [TK]D-Fender | DigitalProcessor: Use POTS <- |
| 01:28.12 | DigitalProcessor | Okay, so that's possible if I have asterisk at each location? |
| 01:28.27 | sirus | [TK]D-Fender: can I give you the login info for my other extension? |
| 01:28.47 | [TK]D-Fender | sirus: No friends to test with, huh? |
| 01:28.54 | DigitalProcessor | Fender, keep in mind, i love linux, and I love asterisk. |
| 01:28.55 | sirus | nope |
| 01:29.02 | [TK]D-Fender | sirus: PM me the details |
| 01:29.03 | sirus | Unless I hop in the car |
| 01:29.04 | DigitalProcessor | just never played with asterisk |
| 01:29.05 | sirus | and go for a drive |
| 01:29.36 | [TK]D-Fender | DigitalProcessor: Good, then I've given you some strategy info, confirmed your budget. Now go DL * and start learning :) |
| 01:29.46 | [TK]D-Fender | ~book |
| 01:29.47 | infobot | [~book] 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://astbook.asteriskdocs.org or see ~buybook |
| 01:29.48 | [TK]D-Fender | ^^^^^^^^^ |
| 01:30.14 | DigitalProcessor | ^^^ I bought that book |
| 01:30.17 | DigitalProcessor | but it's 2005 |
| 01:30.25 | DigitalProcessor | alittle outdated. But will give me the basics |
| 01:30.52 | sirus | downloads the book to |
| 01:31.54 | WindowsUser | dont mine this |
| 01:31.55 | WindowsUser | ~pb |
| 01:31.56 | infobot | [~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/, or apt-get install pastebinit |
| 01:32.06 | DigitalProcessor | Fender, Just a quick qestion. Should I purchase the echo cacellation card and the FXO FXS cards? |
| 01:32.31 | DigitalProcessor | ^^^ my final question |
| 01:32.31 | [TK]D-Fender | DigitalProcessor: Depends how many lines and what kind of usage. |
| 01:32.42 | DigitalProcessor | POTS will be only for failover |
| 01:32.58 | DigitalProcessor | Will need fax machine |
| 01:33.14 | DigitalProcessor | and transmit CC information |
| 01:33.19 | DigitalProcessor | CC = credit cards |
| 01:33.26 | [TK]D-Fender | DigitalProcessor: @ 1 line per site I would recommend a Linksys SPA-3102 for this |
| 01:33.43 | DigitalProcessor | okay, why? if you don't mind me asking |
| 01:34.22 | DigitalProcessor | oh for fax/voice? |
| 01:35.03 | [TK]D-Fender | DigitalProcessor: for your failover. |
| 01:35.17 | DigitalProcessor | okay, I thought asterisk would be doing this. |
| 01:35.31 | [TK]D-Fender | DigitalProcessor: its a cheap 1 port FXS/FXO that autofailovers in the case of a complete power failure bridging the two |
| 01:35.32 | DigitalProcessor | Fender, are you part of the development team? |
| 01:35.38 | DigitalProcessor | by any chance? |
| 01:35.47 | [TK]D-Fender | DigitalProcessor: No, but I am an OP, and an * consultant |
| 01:35.53 | DigitalProcessor | hmmm |
| 01:36.11 | DigitalProcessor | we should talk |
| 01:36.29 | DigitalProcessor | I mean I can get you paid |
| 01:36.35 | DigitalProcessor | one way or another |
| 01:36.37 | DigitalProcessor | =) |
| 01:37.31 | jaytee | he can get you paid.....but can he get you l...? |
| 01:37.45 | DigitalProcessor | nah, I don't roll like htat |
| 01:37.46 | DigitalProcessor | that |
| 01:38.01 | DigitalProcessor | it's always about the paper |
| 01:38.09 | DigitalProcessor | In the ned |
| 01:38.11 | DigitalProcessor | end* |
| 01:38.25 | DigitalProcessor | people need $$$ |
| 01:38.35 | jaytee | yeah, I recommend the 2-ply myself |
| 01:39.19 | [TK]D-Fender | jaytee: Mr. FancyPants CitySlicker.... |
| 01:39.26 | jaytee | lol |
| 01:39.36 | [TK]D-Fender | breaks off a brach w/ some leaves and cleans up the old fashioned way... |
| 01:39.48 | DigitalProcessor | fender, so for now, your suggesting. asterisk at each location and that's it? |
| 01:40.30 | DigitalProcessor | says make sure it's not posion oak. =/ |
| 01:40.34 | [TK]D-Fender | DigitalProcessor: and a central for inter-store dispatch |
| 01:40.45 | DigitalProcessor | correct. |
| 01:40.48 | [TK]D-Fender | DigitalProcessor: Poison Sumac.... much better |
| 01:41.00 | [TK]D-Fender | DigitalProcessor: I did say "tree", and not "bush". |
| 01:41.01 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 01:41.12 | DigitalProcessor | woah |
| 01:41.14 | DigitalProcessor | that woudl suck |
| 01:41.38 | DigitalProcessor | Fender, I seriouly appreciate your time bro. |
| 01:41.43 | DigitalProcessor | thank you for your help |
| 01:42.16 | [TK]D-Fender | I am the very model of a modern Major-General, I've information vegetable, animal, and mineral...... |
| 01:44.11 | [TK]D-Fender | jaytee: Rockin' it old-school y0 |
| 01:44.26 | jaytee | hahahahaa |
| 01:45.04 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
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| 01:51.09 | *** part/#asterisk H3XiL3D (n=H3X@unaffiliated/hexiled) |
| 01:51.55 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-3582aedea147c573) |
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| 01:58.24 | sirus | taps [TK]D-Fender |
| 01:59.19 | [TK]D-Fender | gimme a minutes |
| 02:02.21 | sirus | k |
| 02:05.10 | *** join/#asterisk PhunTelTek (n=PhunTelT@cpe-76-188-233-188.neo.res.rr.com) |
| 02:05.44 | *** join/#asterisk Kumbang (n=ennit@rusnas.paume.itb.ac.id) |
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| 02:12.56 | *** join/#asterisk duckz (n=duckz@86.107.84.186) |
| 02:16.36 | *** join/#asterisk JorgeStraightEdg (n=aaristar@189.181.202.161) |
| 02:17.17 | JorgeStraightEdg | Hi hey can someone help me with a macro im trying to set a rule that when there is no entry of digits continue with the macro but always ends on a failover or something like that |
| 02:17.21 | JorgeStraightEdg | ???? |
| 02:22.40 | Corydon76-dig | JorgeStraightEdg: Are you trying to use WaitExten or Read? |
| 02:23.57 | *** join/#asterisk Defraz (n=T0tal@24-117-156-215.cpe.cableone.net) |
| 02:30.40 | *** join/#asterisk OrNix (n=ornix@78.40.81.34) |
| 02:32.00 | JorgeStraightEdg | Corydon i allready use WaitExten and everything works fine i can enter digits but when i end the line does not pass to the next see: |
| 02:33.18 | JorgeStraightEdg | [lumenvox-did] |
| 02:33.18 | JorgeStraightEdg | exten => s,1,Answer |
| 02:33.18 | JorgeStraightEdg | exten => s,n,Wait(1) |
| 02:33.18 | JorgeStraightEdg | exten => s,n,Background(newvist) |
| 02:33.18 | JorgeStraightEdg | exten => 1,1,dbDel(${BLKVM_OVERRIDE}) |
| 02:33.19 | JorgeStraightEdg | exten => 1,n,Set(__NODEST=) |
| 02:33.21 | JorgeStraightEdg | exten => 1,n,Goto(ext-group,101,1) |
| 02:33.23 | JorgeStraightEdg | exten => 2,1,dbDel(${BLKVM_OVERRIDE}) |
| 02:33.25 | JorgeStraightEdg | exten => 2,n,Set(__NODEST=) |
| 02:33.27 | JorgeStraightEdg | exten => 2,n,Goto(ext-group,61,1) |
| 02:33.29 | JorgeStraightEdg | exten => 3,1,dbDel(${BLKVM_OVERRIDE}) |
| 02:33.31 | JorgeStraightEdg | exten => 3,n,Set(__NODEST=) |
| 02:33.33 | JorgeStraightEdg | exten => 3,n,Goto(ivr-8,s,1) |
| 02:33.35 | JorgeStraightEdg | exten => 4,1,dbDel(${BLKVM_OVERRIDE}) |
| 02:33.36 | _ShrikE | ! |
| 02:33.37 | JorgeStraightEdg | exten => 4,n,Set(__NODEST=) |
| 02:33.39 | JorgeStraightEdg | exten => 4,n,Goto(ext-group,161,1) |
| 02:33.39 | *** mode/#asterisk [+b %JorgeStraightEdg!*@*] by Corydon76-dig |
| 02:33.43 | Corydon76-dig | ~pb |
| 02:33.44 | infobot | [~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/, or apt-get install pastebinit |
| 02:34.24 | *** mode/#asterisk [-b %JorgeStraightEdg!*@*] by Corydon76-dig |
| 02:34.39 | JorgeStraightEdg | http://pastebin.com/m5031b6d7 |
| 02:34.45 | JorgeStraightEdg | sorry here is http://pastebin.com/m5031b6d7 |
| 02:35.28 | Corydon76-dig | You have 2 "s,1" in the same context, for starters |
| 02:36.24 | Corydon76-dig | Fix that first |
| 02:36.50 | JorgeStraightEdg | :D i cant change the second s,1 to s,2 right?? |
| 02:37.24 | Corydon76-dig | s,4, perhaps |
| 02:37.40 | JorgeStraightEdg | ok :D |
| 02:37.44 | JorgeStraightEdg | let me change this line |
| 02:40.45 | JorgeStraightEdg | Corydon you are a genius man you have a coder view something like matrix :D |
| 02:40.57 | JorgeStraightEdg | thanks man really thanks :D:D:D:D:D works like charm |
| 02:41.09 | *** join/#asterisk AussieGuy (n=AussieGu@r220-101-171-79.cpe.unwired.net.au) |
| 02:42.40 | AussieGuy | hi, if I want to use asterisk with my SPA-3102 (which ive configured to forward calls from the PSTN network over VOIP), could I simply make the spa3102 route calls to asterisk? |
| 02:42.56 | AussieGuy | via a sip number for the asterisk server |
| 02:44.54 | AussieGuy | then make asterisk decide what to do with the call, place it on hold until theres an operator available, etc |
| 02:45.28 | JorgeStraightEdg | @Corydon one last question why you tell me that i use s,4 instead of s,2 |
| 02:45.41 | JorgeStraightEdg | is a reason for that?? |
| 02:49.35 | [TK]D-Fender | AussieGuy: Certainly |
| 02:59.36 | *** join/#asterisk s14ck (n=s14ck@190-76-126-155.dyn.movilnet.com.ve) |
| 03:10.14 | *** join/#asterisk Kizmet (n=Kizmet@ppp59-167-37-126.lns2.cbr1.internode.on.net) |
| 08:38.03 | *** join/#asterisk infobot (i=ibot@rikers.org) |
| 08:38.03 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.1.1 (2009/06/05), Asterisk 1.6.0.10 (2009/06/05), 1.4.26 (2009/07/21), *-Addons 1.6.1.1 (2009/07/24), 1.6.0.3 (2009/07/24), 1.4.9 (2009/07/24), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.1 (2009/07/02) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev #asterisk-commits |
| 08:38.27 | *** join/#asterisk Maximo (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 08:43.18 | VladTheImpaled | I think I just fixed it..... |
| 08:43.29 | VladTheImpaled | This is weird, but its to do with Grandstream's settings |
| 08:43.38 | TommyBotten | Ah |
| 08:44.09 | VladTheImpaled | Looks like it requires the port no to be suffixed on the Asterisk server IP address even if its the default 5060 SIP port |
| 08:44.18 | VladTheImpaled | ie 192.168.0.2:5060 |
| 08:44.42 | VladTheImpaled | as soon as I did that and rebooted the phone, it now connects and updated the message indicator correctly |
| 08:44.49 | VladTheImpaled | man that was driving me nuts |
| 08:45.25 | TommyBotten | Strange |
| 08:45.30 | TommyBotten | I haven't had that experience |
| 08:45.59 | TommyBotten | Probably different firmware though. |
| 08:46.07 | VladTheImpaled | I found that little gem lurking in a comment at the bottom of a tutorial on asteriskguru.com from a user |
| 08:46.32 | VladTheImpaled | I upgraded this to the latest firmware yesterday, so you could be right about the firmware being a factor |
| 08:47.07 | VladTheImpaled | ok, its 2 in the morning here so I'm calling it a day |
| 08:47.10 | VladTheImpaled | Thanks for your help |
| 08:52.02 | *** join/#asterisk WeazelON (n=deazel@mail2.tikalnetworks.com) |
| 08:52.09 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 08:57.29 | *** join/#asterisk angryuser (n=angryuse@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 09:09.56 | *** join/#asterisk Psychobilly (n=moi@adsl117-205.kln.forthnet.gr) |
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| 09:28.33 | *** join/#asterisk devyll (n=paul@89.36.24.2) |
| 09:29.49 | devyll | can anybody explain to me why do I get this messages in log file : http://pastebin.com/d122a924 |
| 09:29.51 | devyll | please |
| 09:31.02 | TommyBotten | When do you get this error? |
| 09:31.32 | *** join/#asterisk Dovid (n=annon@tony09-118-62.inter.net.il) |
| 09:33.12 | devyll | TommyBotten: it's showing recursively .. every 1 minute or so for every available sip peer |
| 09:37.36 | kaldemar | devyll: do you have ringinuse=no in queues.conf? that would explain device state checking for queue members. |
| 09:38.39 | devyll | yes kaldemar. are you saying those messages are completely normal ? |
| 09:39.15 | kaldemar | yes, they are just debug messages to let you know what asterisk is doing. not an error, unless the check result is unexpected. |
| 09:39.38 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 09:40.37 | kaldemar | you might be better off not logging debug to a file unless you're debugging an actual problem. |
| 09:43.02 | devyll | kaldemar: but is it normal to check the state every 2 minutes ? http://pastebin.com/d2fda7a28 (I have 201, 202, and 205 sip users active) and no activity on the asterisk. |
| 09:43.29 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 09:46.50 | kaldemar | devyll: i don't know what the check intervall is, but sounds normal to me. |
| 09:49.53 | devyll | to me it sounds normal to check only once. only before an action is to be made on that channel or device. so .. If I want to use device x , before this action to verify it's state and if it's busy then ... else ... |
| 09:50.29 | devyll | but, If it's a normal behaviour .. than no problem. I just want to be sure it's nothing to worry about |
| 09:50.48 | kaldemar | i don't know if you have activity on the server. |
| 09:51.33 | kaldemar | but anyway, those are just debug messages, whatever triggers the state check. |
| 09:52.57 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 09:54.42 | *** join/#asterisk fbnts (n=root@213.106.147.11) |
| 10:01.47 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 10:03.29 | *** join/#asterisk jojo^ (i=jol@ph4.se) |
| 10:04.28 | jojo^ | How can I connect two ongoing calls/channels? |
| 10:04.38 | *** part/#asterisk Zhad (n=tom@server30261.uk2net.com) |
| 10:06.18 | *** join/#asterisk andyd (n=andy@chilli.nosignal.org) |
| 10:06.45 | kaii | jojo^: we do not understand? |
| 10:06.46 | andyd | is there a way to register more than one handset at a time to a sip.conf SIP account in Asterisk, so that when someone calls the extension all phones ring |
| 10:07.00 | andyd | I have done this with openser but not asterisk, i assumed it was possible but it doesn't work in my test scenario |
| 10:07.18 | kaii | andyd: multiple registration is not possible in asterisk |
| 10:07.24 | jojo^ | kaii, Suppose I have two incoming calls. How can I connect/bridge these calls to the callers can speak to eachother? |
| 10:07.26 | kaii | in asterisk one peer is one device. |
| 10:07.33 | andyd | Ok, thanks for confirming. |
| 10:07.52 | kaii | andyd: you can do tricks with the database |
| 10:08.39 | kaii | andyd: _XXX,1,Dial(SIP/${EXTEN}&${DBGet(MULTICALL/${EXTEN})}) |
| 10:09.03 | kaii | andyd: where MULTICALL/100 (for example) is set to "SIP/101&SIP/102" |
| 10:09.10 | andyd | Oh yuck |
| 10:09.36 | kaii | quite more sexy than having multiple devices registering for the same user |
| 10:09.59 | kaii | jojo^: you could put them into a meetme() |
| 10:10.03 | kaii | jojo^: "show application meetme" |
| 10:10.30 | kaii | jojo^: otherwise you could do a "pickup" from channel A to channel B |
| 10:11.33 | jojo^ | kaii, Thank you very much. |
| 10:20.07 | *** part/#asterisk andyd (n=andy@chilli.nosignal.org) |
| 10:27.41 | *** join/#asterisk csiadmin (n=csiadmin@81.144.152.52) |
| 10:36.49 | *** join/#asterisk grEvenX (n=even@188.113.114.118) |
| 11:03.42 | pa | i have an ISDN line i use with asterisk through zapata.. how can i send faxes with it? |
| 11:12.34 | *** join/#asterisk grey-monkey (n=ericshel@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 11:12.52 | *** join/#asterisk phunyguy (n=phunyguy@h69-130-68-88.kgldga.dsl.dynamic.tds.net) |
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| 11:17.01 | *** join/#asterisk TSM (n=the_soft@fw-lon1.wenn.com) |
| 11:18.29 | *** join/#asterisk phunyguy (n=phunyguy@h69-130-68-88.kgldga.dsl.dynamic.tds.net) |
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| 11:23.56 | *** join/#asterisk Han (n=han@unaffiliated/han) |
| 11:26.51 | kaii | pa: sending faxes with asterisk is a complicated story. read up on http://www.voip-info.org/wiki/view/Asterisk+fax |
| 11:29.51 | *** join/#asterisk grEvenX (n=even@188.113.114.118) |
| 11:32.42 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:32.42 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 11:35.44 | *** join/#asterisk Psychobilly (n=moi@adsl117-205.kln.forthnet.gr) |
| 11:40.03 | *** join/#asterisk micols (n=mio@rlogin.dk) |
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| 11:42.06 | *** join/#asterisk Josvds (n=josvds@ip5657c69f.direct-adsl.nl) |
| 11:42.24 | Josvds | hello |
| 11:42.51 | Josvds | i am new here but can some one tell me how to install asterisk as a SMS gateway |
| 11:43.13 | Josvds | or if someone knows a good documentation about it? |
| 11:44.01 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 11:45.39 | *** join/#asterisk datacompboy (n=datacomp@213.187.251.250) |
| 11:46.54 | *** join/#asterisk DelphiWorld (n=Miranda@41.201.115.151) |
| 11:47.00 | DelphiWorld | hello all |
| 11:47.10 | DelphiWorld | please how we can use SIP over TCP in asterisk 1.6? |
| 11:47.15 | datacompboy | pffrrr... one stupid question:) i have two remote sip accounts on one provider. if incoming call on one leg comes, it pass fine, if on other leg -- it fails, since host= not dynamic. But if host=dynamic, i can't do outgoing call, since SIP/<number>@<number> fails |
| 11:57.05 | leifmadsen | wow... he waited a whole 6 mins for an answer |
| 11:57.49 | kaldemar | maybe read the samples with the same intensity |
| 11:58.04 | *** join/#asterisk lathos (n=simon@alibi.simon-cozens.org) |
| 11:58.45 | lathos | Anyone know why the features processor might not be firing even though my DTMF digits are being recognised and emulated? |
| 12:00.48 | *** join/#asterisk xrmx__ (n=rm@87.13.48.2) |
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| 12:10.03 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 12:11.53 | datacompboy | insecure=very fixes mine prob:) |
| 12:15.13 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:18.55 | *** join/#asterisk israr (n=israr@119.153.29.143) |
| 12:19.02 | *** part/#asterisk israr (n=israr@119.153.29.143) |
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| 12:29.19 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
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| 12:31.27 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:33.31 | TSM | is it possable to test calls on a T1 when using a loopback connector |
| 12:36.01 | mbrevda | if you have two port, yes |
| 12:36.35 | *** join/#asterisk micols (n=mio@rlogin.dk) |
| 12:37.52 | jaytee | 2 port card with a crossover, one port configured for PRI_NET and one for PRI_CPE. have to play with your outbound and inbound routes in dialplan also to test |
| 12:37.52 | jaytee | but single port card, no. that would not work for testing calls |
| 12:38.06 | *** join/#asterisk Psychobilly (n=moi@adsl8-176.kln.forthnet.gr) |
| 12:39.59 | TSM | jaytee: ahh ok, poo, ile have to wait and test on a live E1 line then |
| 12:40.16 | TSM | jaytee: have you used the sangoma cards? |
| 12:40.30 | jaytee | TSM, no I use Digium's cards |
| 12:40.53 | TSM | ahh ok |
| 12:41.06 | TSM | no worries then, ile chug along |
| 12:41.18 | jaytee | but pretty much the same scenario, use a crossover cable and code in your dialplan to route a test extension out one span and in the other to another phone. |
| 12:47.16 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
| 12:47.43 | *** join/#asterisk stefano (n=stefano@ip-87-238-2-99.static.adsl.cheapnet.it) |
| 12:47.59 | *** part/#asterisk stefano (n=stefano@ip-87-238-2-99.static.adsl.cheapnet.it) |
| 12:49.38 | Han | Can the hypermedia hg-5000 UMTS gateway also cooperate with Asterix? It's not really mentioned in their documentation. |
| 12:52.50 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 12:56.10 | [TK]D-Fender | Han: Looks like "yes". Supports H323 native (which * sucks at but might work OK). And SIP is "optional" according to their brochure. That would be the better idea |
| 12:57.10 | Han | Thank you. It's all rather not my expertise. |
| 13:01.59 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:05.44 | *** join/#asterisk Stese (n=Someone@adsl.ntsols.com) |
| 13:06.01 | Stese | Ok, anyone care to do an idiot check for me? |
| 13:06.07 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:06.49 | Kizmet | Stese, what kind of idiot check do you require? |
| 13:08.36 | Stese | just installed Cent0S 5 and *1.6.1, using the assistance of http://www.voip-info.org/wiki/index.php?page_id=5608 for asterisk, created a basic sip.conf and extenions.conf |
| 13:09.33 | Stese | yet after a reload, dialplan show doesn't show my dialplan |
| 13:09.59 | Stese | and I've disabled ael |
| 13:11.03 | Stese | hang on....I think i've spotted it |
| 13:11.47 | Stese | helps if extenisons.conf is spelt correctly! |
| 13:11.51 | Stese | *blush* |
| 13:13.02 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 13:13.24 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:15.21 | [TK]D-Fender | just found an idiot! |
| 13:15.23 | [TK]D-Fender | \o/ |
| 13:15.27 | [TK]D-Fender | what do I win? |
| 13:15.28 | Stese | lol |
| 13:15.42 | Stese | idiot check, not hunt.... |
| 13:15.56 | [TK]D-Fender | pulls out his compound bow, just in case... |
| 13:16.02 | Stese | ie... tell me what idiotic mistake i've made |
| 13:18.54 | Stese | urg... spelling corrected, and it's still not working... |
| 13:20.49 | Kizmet | hmm |
| 13:21.17 | Stese | I know it's gonna be something really daft |
| 13:21.30 | Kizmet | I might not be the best person to assist, I hate the original extensions.conf and love AEL heh. |
| 13:21.39 | leifmadsen | is opposite of that |
| 13:22.21 | leifmadsen | do you have pbx_config.so loaded? |
| 13:22.28 | leifmadsen | (or whatever that module is... I think that is the name) |
| 13:24.22 | Stese | apparently not |
| 13:25.12 | *** join/#asterisk bobJR (n=bob@adsl-153-124-150.tys.bellsouth.net) |
| 13:25.25 | Stese | oh no... it is |
| 13:26.22 | Kizmet | Stese, what do you get with a: module show like config & dialplan show |
| 13:26.25 | *** join/#asterisk Katty (n=asterisk@mail.copi-rite.com) |
| 13:26.26 | Katty | :> |
| 13:26.33 | Kizmet | pastebin it mebbe? |
| 13:26.40 | Kizmet | hi Katty ;) |
| 13:26.50 | [TK]D-Fender | Stese: Go check your permissions on your configs <- |
| 13:26.52 | Katty | umm hi randompersonidon'trecognize |
| 13:26.58 | [TK]D-Fender | Katty: Mew. |
| 13:27.02 | Katty | [TK]D-Fender: mew. |
| 13:27.06 | *** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903) |
| 13:27.33 | Kizmet | Im new, well this year I am (havent floated around Freenode for several years) |
| 13:27.56 | Katty | oh. well hi there! |
| 13:28.02 | Katty | i've been ehre for... |
| 13:28.06 | Katty | asks nickserv |
| 13:28.13 | Stese | [TK]D-Fender > does the fact that at the moment i'm running * as root make any difference? |
| 13:28.15 | kaii | leifmadsen: just for my personal interest.. what are the reasons you prefer classic extensions over ael ? |
| 13:28.21 | Katty | since 05 |
| 13:28.25 | Katty | 4 years. |
| 13:28.27 | Katty | it's 09, right? |
| 13:28.33 | Kizmet | yep |
| 13:28.46 | file | Katty: it's you, and me, moving at the speed of light into eternity! |
| 13:29.02 | Kizmet | Hi file! :) |
| 13:29.13 | file | waves |
| 13:29.14 | Katty | file: is that the song of the day?! |
| 13:29.26 | file | Katty: it can be for you! |
| 13:29.28 | Kizmet | file, still talk with grumpydog? |
| 13:29.35 | file | Kizmet: nyet |
| 13:29.38 | [TK]D-Fender | Stese: How are you starting it? |
| 13:29.42 | Kizmet | nods |
| 13:29.47 | Katty | dances |
| 13:30.22 | *** join/#asterisk grandpapadot (n=no@99-175-248-81.lightspeed.brhmal.sbcglobal.net) |
| 13:30.49 | Kizmet | thinks file might just barely remember him :o |
| 13:30.50 | Stese | it's done at boot... |
| 13:30.56 | [TK]D-Fender | kaii: AEL gets parsed back to extensions.conf anyway and can only do less that you are capable of doing yourself directly. Its also one more layer that can get hit with syntax changes, and have bugs of its own before even reaching standard dialplan parsing. |
| 13:31.22 | [TK]D-Fender | kaii: It only looks a little more familiar and neater to other programmers |
| 13:31.26 | file | Kizmet: at this time of day I'm lucky if I remember my own name |
| 13:31.34 | kaii | i see. |
| 13:31.40 | [TK]D-Fender | kaii: and fewer docs to explain. |
| 13:31.47 | Katty | file: i remembered your muffins. |
| 13:31.48 | Kizmet | file, ditto here. 11pm :-/ |
| 13:31.54 | [TK]D-Fender | file: .... Hi Steve!!!!!! |
| 13:31.59 | kaii | is it possible to write functions in AEL and "compile" them to "extensions.conf" to manually check what AEL really did? |
| 13:32.01 | file | [TK]D-Fender: Hi Bob! |
| 13:32.04 | Stese | [TK]D-Fender > used "chkconfig asterisk on" |
| 13:32.22 | Kizmet | [TK]D-Fender, Im a programmer by nature so it looks nicer to me and I have been using it since AEL was released. *shrugs* |
| 13:32.31 | [TK]D-Fender | kaii: You can see it in "dialplan show" |
| 13:32.43 | [TK]D-Fender | Stese: that doesn't prove who the script starts * as |
| 13:32.56 | Kizmet | Stese, ps aux | grep asteriskl |
| 13:33.02 | Kizmet | - the l |
| 13:33.24 | Katty | lllllllllllllllll |
| 13:33.32 | [TK]D-Fender | Kizmet: As am I, but I want the lowest level control and the best documented with fewest bugs. |
| 13:33.32 | Kizmet | oooooooooo |
| 13:33.36 | Katty | lllllllll |
| 13:33.42 | Katty | *hee* |
| 13:34.03 | Stese | all in here |
| 13:34.03 | Stese | http://pastebin.com/d29fcaeb |
| 13:34.49 | [TK]D-Fender | Stese: ls -la /etc/asterisk |
| 13:35.14 | *** join/#asterisk DelphiWorld (n=Miranda@41.201.117.204) |
| 13:35.29 | DelphiWorld | any ubuntu repository to install the 1.6.X branche of asterisk? |
| 13:35.47 | [TK]D-Fender | DelphiWorld: Highly doubt it. |
| 13:36.47 | DelphiWorld | [TK]D-Fender: |
| 13:36.52 | kaii | [TK]D-Fender: well of course i can see it there. question was about "compiling" AEL to extensions.conf |
| 13:36.53 | DelphiWorld | i don't understand |
| 13:36.57 | *** join/#asterisk neurosys (n=vinix@c-66-229-80-246.hsd1.fl.comcast.net) |
| 13:37.15 | Stese | [TK]D-Fender > http://pastebin.com/d43caea41 |
| 13:37.19 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 13:37.21 | Kizmet | DelphiWorld, I cant find any * 1.6 .debs on the mirror server @ the isp I work for nor on the google machine. |
| 13:38.03 | [TK]D-Fender | Stese: -rw-r--r-- 1 root root 1214 Jul 29 14:00 extenisons.conf <-- you still can't spell... |
| 13:38.05 | Kizmet | Stese, Your extensions.conf is still spelt incorrectly |
| 13:38.08 | DelphiWorld | Kizmet: then i must build it from source; |
| 13:38.11 | Stese | LOL |
| 13:38.21 | Stese | *hangs head in shame* |
| 13:38.23 | ariel_ | Morning |
| 13:38.31 | Kizmet | Morning ariel_ |
| 13:38.51 | [TK]D-Fender | grabs his bow and shoots Stese in the ass. |
| 13:38.52 | Kizmet | Stese, also you're still loading pbx_ael.so ;) |
| 13:39.01 | DelphiWorld | ariel: morning |
| 13:39.19 | [TK]D-Fender | Kizmet: He's entitles... at least its config file is spelled right ;) |
| 13:39.33 | [TK]D-Fender | -s+d |
| 13:39.51 | [TK]D-Fender | (except for being .sample) |
| 13:39.52 | Katty | discovered blueberry propel water :> |
| 13:39.56 | Stese | Kizmet > Disabled via Modules.conf? |
| 13:40.17 | [TK]D-Fender | Stese: Try getting something to WORK, before wasting time STOPPING other things :) |
| 13:40.26 | Katty | does not approve of the pomegranate flavoring in her blueberry water. |
| 13:40.36 | Katty | :< |
| 13:40.39 | Kizmet | Katty, ohes noes :< |
| 13:40.44 | Katty | i knoes :< |
| 13:40.56 | Katty | it are ruint |
| 13:41.02 | Kizmet | hugs Katty |
| 13:41.06 | leifmadsen | and doesn't asterisk complain with a WARNING that it is missing an extensions.conf (or whatever .conf) when it loads the modules? |
| 13:41.10 | Kizmet | poor Katty ruint water :( |
| 13:41.17 | leifmadsen | this is why running 'asterisk -cvvvg' is a good thing when you're testing |
| 13:41.26 | Stese | [TK]D-Fender > I think learning to spell would be more helpful |
| 13:41.38 | *** part/#asterisk DelphiWorld (n=Miranda@41.201.117.204) |
| 13:41.43 | Stese | @leifmadsen > I did that, missed it totally |
| 13:41.49 | Katty | Miranda!!! THE REAVERS |
| 13:41.55 | Katty | AHH |
| 13:41.56 | Katty | runs |
| 13:42.09 | grandpapadot | lol, Reavers |
| 13:42.10 | ariel_ | did you paint red on your ship Katty ???? |
| 13:42.24 | Katty | sadly, i don't have a ship. |
| 13:42.24 | grandpapadot | Firefly rocks. |
| 13:42.27 | Katty | it hasn't come in yet. |
| 13:42.30 | Kizmet | Katty, As long as your kitchen appliances arent named 'smeg' like mine :( |
| 13:42.32 | ariel_ | misses that show |
| 13:42.35 | Stese | I'm out to confound these bungers... |
| 13:42.40 | Katty | smeg? |
| 13:42.45 | Katty | no they say fisher price (= |
| 13:42.53 | grandpapadot | FIrefly got cancelled but somehow Lost is still on? |
| 13:43.04 | leifmadsen | w00t Lost |
| 13:43.06 | Katty | grandpapadot: yeah i haven't figured that out yet |
| 13:43.07 | *** join/#asterisk AndyML (n=AndyML@pool-173-49-144-213.phlapa.fios.verizon.net) |
| 13:43.16 | Stese | grandpapadot > Sucks, doesn't it |
| 13:43.22 | Katty | grandpapadot: other than Firefly probably had competition with another mainstream series ona nother channel. |
| 13:43.22 | Kizmet | Katty, its a reddwarf reference ;) |
| 13:43.39 | Katty | Kizmet: ah. didn't watch that whole series. |
| 13:43.44 | Stese | ~Smega |
| 13:43.46 | Stese | ~Smegma |
| 13:43.47 | infobot | from memory, smegma is blue |
| 13:43.47 | Katty | bablyon 5 is my favorite of all time. |
| 13:43.57 | Stese | fine |
| 13:44.04 | Kizmet | ~smeg |
| 13:44.05 | infobot | you're a smeghead |
| 13:44.08 | Katty | tho star trek was damn fine. |
| 13:44.10 | grandpapadot | I didn't even know FireFly existed until after it was cancelled, I watched the entire series and couldn't believe it got cancelled, absolutely great show with perfect cast interaction. |
| 13:44.11 | Kizmet | there you go :) |
| 13:44.12 | Katty | for several years. |
| 13:44.30 | grandpapadot | Babylon 5 was a good show. SG-1 rocks. But FireFly beats them all IMHO. |
| 13:44.36 | Stese | grandpapadot > Same here... I did get an advanced showinf of serenity tho... |
| 13:44.44 | coppice | I think the writer of red dwarf must have had a oven that sucked |
| 13:44.45 | Kizmet | grandpapadot, I can agree. *sigh* damn network execs have no idea! |
| 13:44.52 | *** join/#asterisk Dovid (n=annon@tony09-118-62.inter.net.il) |
| 13:45.11 | Kizmet | coppice, probobly. |
| 13:45.12 | [TK]D-Fender | Katty: "We also have an open air-lock policy!" |
| 13:45.27 | Stese | Kizmet > Do i need to disable AEL, if i'm not using ti? |
| 13:45.29 | Stese | *it |
| 13:45.34 | Kizmet | Stese, no |
| 13:45.35 | coppice | or a vacuum cleaner that didn't |
| 13:45.36 | [TK]D-Fender | Stese: No |
| 13:45.38 | ariel_ | has no use fro AEL |
| 13:45.51 | Stese | ok, not going to go change things where I don't need to :) |
| 13:46.03 | Katty | [TK]D-Fender: :> |
| 13:46.04 | ariel_ | leifmadsen: got my dynamic menus done thank you for your input yesterday. |
| 13:46.22 | leifmadsen | ariel_: np! glad it worked out |
| 13:46.44 | Kizmet | I hate my job.... Just finished a 15 level deep menu :-/ |
| 13:47.03 | leifmadsen | Stese: well, you don't *have* to, but make sure you either remove the extensions.ael file, or disable the pbx_ael.so module, or you might get some extra dialplan stuff that you don't expect |
| 13:47.14 | leifmadsen | (since asterisk can use both AEL and dialplan at the same time, it merges them together) |
| 13:48.22 | Stese | Renamed the extensions.ael file... |
| 13:49.15 | [TK]D-Fender | Katty: Ivanova > All |
| 13:49.26 | Katty | i AM the expanding borders of Russia! |
| 13:49.40 | [TK]D-Fender | :D |
| 13:50.05 | Kizmet | [TK]D-Fender, so whats your view on LUA |
| 13:50.07 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 13:50.07 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:50.12 | Kizmet | ducks |
| 13:51.03 | [TK]D-Fender | Kizmet: give you a hint.... its through a Bushnell scope ;) |
| 13:51.15 | Kizmet | hehehe *snickers* |
| 13:51.26 | leifmadsen | if I knew lua, I might like it |
| 13:51.47 | Kizmet | leifmadsen, it looks 'OK |
| 13:51.48 | leifmadsen | all the other abstraction languages, or modules, seem to be limited in functionality though |
| 13:51.53 | Kizmet | but I like AEL heh. |
| 13:52.21 | leifmadsen | it looks ok in principle, but I know dialplan so well, I see no reason to change |
| 13:52.23 | leifmadsen | but that's just me |
| 13:52.31 | leifmadsen | nothing wrong with AEL if you now it |
| 13:52.33 | leifmadsen | know it* |
| 13:52.47 | [TK]D-Fender | leifmadsen: Biggest problem = extensions.conf. Every other layer simply makes matter worse... |
| 13:53.08 | [TK]D-Fender | leifmadsen: So... when do we sit down to reinvent the whole mess? ;) |
| 13:53.18 | Stese | isn't lua the scripting language that WoW uses for it's Addons? |
| 13:53.28 | Kizmet | Stese, yes. dont ask me how I know. |
| 13:53.42 | [TK]D-Fender | Kizmet: How do you know? |
| 13:53.54 | [TK]D-Fender | notes the directional request... |
| 13:54.07 | Kizmet | [TK]D-Fender, I wrote a few bits and bobs for my *sigh* Mother.... *sigh* |
| 13:54.51 | Kizmet | hides in the corner in shame! |
| 13:54.58 | Stese | why shame? |
| 13:55.07 | leifmadsen | [TK]D-Fender: go nuts, I have zero problems with dialplan |
| 13:55.21 | leifmadsen | if there was a "better" method, I'd probably still use dialplan |
| 13:55.24 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 13:55.46 | Kizmet | Stese, dunno. heh. Im sure there are some in this channel rofling :P |
| 13:56.42 | Stese | WoW isn't so bad, if you take it with a pinch of salt... anyway... I'll stop being off topic :P |
| 13:56.44 | [TK]D-Fender | leifmadsen: Yeah it "works", but we know the issues with not having typed data, sorting issues, lack of actual structured language, delimiter issues, maximum line-length, etc. Many paging scenarios are serious PITA..... |
| 13:57.06 | leifmadsen | I guess |
| 13:57.10 | Kizmet | I reason I ask about lua is because they just hired another 'PBX Admin' who has decided that he is going to write everything in lua from now on... When I told him to use AEL he chucked a wobbly :( |
| 13:57.28 | leifmadsen | Kizmet: make mnicholson his new best friend then |
| 13:57.37 | leifmadsen | Kizmet: that's who wrote pbx_lua |
| 13:57.44 | Kizmet | leifmadsen, ahaha. |
| 13:57.57 | leifmadsen | oh. I'm not laughing sir. |
| 13:57.59 | Kizmet | leifmadsen, im his box so he'll either do as I say or gtfo. |
| 13:58.06 | Kizmet | s/box/boss/ |
| 13:58.07 | leifmadsen | box! |
| 13:58.11 | Kizmet | *sigh* tired. |
| 13:58.13 | [TK]D-Fender | Kizmet: And I'd watch out about mentioning "wobbly" around here... |
| 13:58.18 | leifmadsen | wobbly pops? |
| 13:58.24 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 13:58.54 | lirakis | I dont know about using an abstraction layer for a "dialplan" alone |
| 13:58.59 | lirakis | for applications - sure |
| 13:59.25 | Kizmet | [TK]D-Fender, I would have said fit but my partner has Epilepsy and finds it offensive ;) |
| 13:59.44 | lirakis | if you get used to a rails like structure and a db abstraction layer like datamapper / active record, adhearsion really makes doing apps a heck of a lot easier. |
| 13:59.57 | [TK]D-Fender | Kizmet: When he makes a deal with someone, does he shake on it? |
| 14:00.19 | Kizmet | [TK]D-Fender, nope. |
| 14:00.24 | [TK]D-Fender | ;) |
| 14:01.52 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-241-77-24.msy.bellsouth.net) |
| 14:02.16 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 14:02.42 | Kizmet | wants to cry about the next change I have to make. |
| 14:02.55 | Stese | ? |
| 14:03.10 | *** join/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda) |
| 14:03.22 | Kizmet | Stese, Add another 3 levels to the 15 level menu for 'statistics collection' purposes. :( |
| 14:03.32 | Stese | ouch |
| 14:03.56 | Stese | i'm glad I've not looked at IVR's yet... |
| 14:04.32 | Kizmet | Stese, they are pretty simple until you get into running huge menu arrays for an executive team that has no idea. |
| 14:04.51 | Stese | My boss is dropping * and going with OCS... with * as a SIP TCP to UDP gateway |
| 14:05.14 | Kizmet | finds his rifle. |
| 14:05.20 | leifmadsen | that seems dumb |
| 14:05.23 | grandpapadot | OCS? |
| 14:05.32 | leifmadsen | especially since Asterisk supports SIP TCP in 1.6 |
| 14:05.33 | Stese | Office Comunications Server |
| 14:05.35 | leifmadsen | MS OCS |
| 14:05.39 | leifmadsen | (Microsoft) |
| 14:05.40 | grandpapadot | Ick. |
| 14:05.52 | *** join/#asterisk codefreeze-lap (n=murf@72.21.67.40) |
| 14:06.05 | Kizmet | feels for Stese :( |
| 14:06.47 | iewebguy | Will the cisco 7960 do TCP SIP ? |
| 14:07.35 | [TK]D-Fender | iewebguy: We're jsut wondering when cisco will support SIP UDP properly ;) |
| 14:07.47 | [TK]D-Fender | :F |
| 14:08.10 | Kizmet | iewebguy, yes it does but dont depend on it actually working with *, Seems to work OK with CCM |
| 14:08.49 | iewebguy | thanks! What is CCM? |
| 14:08.58 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 14:09.05 | *** part/#asterisk lathos (n=simon@alibi.simon-cozens.org) |
| 14:09.17 | Kizmet | iewebguy, Cisco Call Manager |
| 14:09.52 | iewebguy | Oh, gotcha. |
| 14:09.56 | Kizmet | iewebguy, If you have a spair Cisco 1760 or similar with some DSP's use it as a Skinny to SIP gateway heh. (I've never really had much luck with chan_skinny) |
| 14:10.24 | *** join/#asterisk blee01 (n=blee01@237.177.205.68.cfl.res.rr.com) |
| 14:10.46 | Stese | Asterisk supports TCP and UDP Sip, but OCS only supports SIP over TCO |
| 14:10.49 | Stese | *TCP |
| 14:11.07 | Stese | and our current sip provider only supports UDP |
| 14:11.27 | blee01 | Anyone know how to manually reload changes made in the features.conf file? I'm trying to remove automon or at the very least change its mapping but it wont budge |
| 14:11.35 | *** join/#asterisk jeff (i=jeff@unaffiliated/jeff) |
| 14:11.40 | Kizmet | Stese, Is your boss changing for any particular reason? |
| 14:12.30 | Stese | Well, OCS provides integration into Exchange, I think is the main reason# |
| 14:12.45 | Stese | for instance, I can call, and listen to my emails |
| 14:13.59 | Kizmet | to me thats silly. really! I have an email capable phone for that! |
| 14:14.08 | Kizmet | thinks someone is lazy |
| 14:14.18 | Stese | indeed, I do as well, as does he, |
| 14:14.20 | leifmadsen | Stese: I think you can still integrate with exchange via IMAP voicemail storage |
| 14:14.24 | iewebguy | How can I unset caller ID name and number in asterisk 1.4 my telco is seeing my extension name as caller id right now. |
| 14:14.41 | leifmadsen | you mean re-set? |
| 14:14.42 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 14:14.42 | Stese | unset? or force it to use something? |
| 14:14.49 | Kizmet | iewebguy, Set(CALLERID(all)=""); |
| 14:14.54 | blee01 | Set right? |
| 14:14.56 | blee01 | yeah |
| 14:15.02 | leifmadsen | use the CALLERID() function |
| 14:15.03 | iewebguy | I want to make sure its not sending any caller id to the telco |
| 14:15.12 | leifmadsen | well, it probably needs to send "something" |
| 14:15.21 | leifmadsen | most telcos will reject if you send "nothing" |
| 14:15.26 | Stese | send (Withheld) |
| 14:15.27 | leifmadsen | set it to something like 0000000000 |
| 14:15.45 | iewebguy | its getting a NAME right now. |
| 14:16.01 | leifmadsen | set reset the name |
| 14:16.02 | Kizmet | leifmadsen, something like that in AU is too simmilar to emergency svcs which is 000 people get confused, its bad.. :( |
| 14:16.03 | iewebguy | I;ll try sending an empty string. |
| 14:16.04 | leifmadsen | CALLERID(name) |
| 14:16.16 | leifmadsen | Kizmet: ah -- I'm in N.A. |
| 14:16.40 | Kizmet | leifmadsen, ;) just a hint is all hehe. |
| 14:16.41 | blee01 | Do you guys any idea why even after i reload asterisk changes i made in the features.conf arent taking? |
| 14:16.59 | blee01 | either changing the automon mapping or commening it out |
| 14:17.08 | leifmadsen | ;) |
| 14:17.40 | Kizmet | blee01 are you using the w/W param in the dial() app? |
| 14:17.47 | grandpapadot | blee01: I think features.conf requires a restart. |
| 14:18.07 | blee01 | i restarted as well |
| 14:18.51 | *** join/#asterisk shido6 (n=shido6@dsl-67-212-24-92.acanac.net) |
| 14:18.58 | blee01 | not sure about that param |
| 14:19.01 | blee01 | let me see |
| 14:19.36 | Kizmet | blee01, as I beleive that the default being *1 will be enabled upon using w/W. someone feel free to correct me! |
| 14:20.42 | [TK]D-Fender | grabs an 18 gallon cannister of Liquid Paper and tosses Kizmet in head first |
| 14:21.09 | blee01 | I dont see that parameter |
| 14:21.11 | KavanS | drops a deuce on it! |
| 14:21.13 | Kizmet | [TK]D-Fender, huhn, use metric! :P |
| 14:21.37 | blee01 | Even then, i am still having the same problem even if i remove that line from features.conf |
| 14:22.03 | [TK]D-Fender | upgrades to a kiloliter basin, submerges Kizmet and closes the lid |
| 14:22.33 | Kizmet | finds a slimy old tuna and slaps [TK]D-Fender arounds... |
| 14:23.03 | Stese | Kizmet > Is that a RSPCA Tuna? |
| 14:23.14 | [TK]D-Fender | ~cluebat Kizmet |
| 14:23.15 | infobot | ACTION pulls out a ClueBat (tm) and thwaps Kizmet. |
| 14:23.16 | Kizmet | Stese, yes.. |
| 14:23.17 | [TK]D-Fender | VICTORY IS MINE! </stewie> |
| 14:24.05 | Kizmet | ok so have we all had our fun now? Im pretty sure blee01 still has an issue with features.conf ;) |
| 14:24.06 | *** join/#asterisk jmacz (n=mcorb@190.144.75.22) |
| 14:24.13 | blee01 | rofl indeed :-/ |
| 14:24.36 | blee01 | The thing is, even with the wW parameter on the dial, even when i change the mapping or comment it out |
| 14:24.52 | blee01 | it still is holding the old mapping |
| 14:25.19 | blee01 | currently *7 ive changed it to *5 and commented it out. Ive reloaded and restarted asterisk with the same result |
| 14:25.29 | [TK]D-Fender | blee01: Pastebin is your friend... |
| 14:25.31 | [TK]D-Fender | ~pb |
| 14:25.32 | infobot | [~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/, or apt-get install pastebinit |
| 14:25.53 | blee01 | haha sorry bad habit |
| 14:26.43 | *** join/#asterisk krivenok_dmitry (n=krivenok@85.235.219.42) |
| 14:28.11 | krivenok_dmitry | Hello! |
| 14:28.11 | krivenok_dmitry | I've just installed asterisk-1.4.26. When I start "asterisk -r" and type "sip" I get an error: |
| 14:28.11 | krivenok_dmitry | No such command 'sip' (type 'help sip' for other possible commands) |
| 14:28.17 | krivenok_dmitry | What's wrong? |
| 14:28.32 | *** join/#asterisk cryptanthus (n=newview@wsip-72-214-233-12.om.om.cox.net) |
| 14:28.43 | Stese | krivenok_dmitry > What are you expecting that to do? |
| 14:29.20 | grandpapadot | try "sip show peers" or type "sip<space>" then hit "tab" |
| 14:29.28 | Stese | krivenok_dmitry > it's also telling you what is wrong " No such command 'sip' " ¬_¬ |
| 14:30.08 | krivenok_dmitry | No such command 'sip show peers' (type 'help sip show' for other possible commands) |
| 14:30.47 | Stese | krivenok_dmitry > try 'help' |
| 14:30.54 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 14:31.47 | krivenok_dmitry | "help" doesn't show me anything about SIP. |
| 14:31.47 | krivenok_dmitry | I haven't seen any SIP-related configuration options in the output of ./configure --help. |
| 14:31.57 | krivenok_dmitry | Maybe SIP support isn't enabled |
| 14:32.10 | grandpapadot | krivenok_dmitry: It is by default, did you install libpri and zaptel? |
| 14:32.16 | grandpapadot | (or dahdi as it were) |
| 14:32.22 | [TK]D-Fender | krivenok_dmitry: What do you have in modukles.conf? |
| 14:32.24 | kaldemar | krivenok_dmitry: what does "module load chan_sip.so" say? |
| 14:32.28 | [TK]D-Fender | krivenok_dmitry: What do you have in modules.conf? |
| 14:32.33 | *** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 14:33.11 | Kizmet | krivenok_dmitry, run a > show modules |
| 14:33.12 | [TK]D-Fender | ~pb |
| 14:33.13 | infobot | [~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/, or apt-get install pastebinit |
| 14:33.15 | [TK]D-Fender | krivenok_dmitry: ^^^ |
| 14:33.17 | krivenok_dmitry | Oh, I see the problem |
| 14:33.19 | *** join/#asterisk blackest_mamba (n=blackest@c-71-239-160-143.hsd1.il.comcast.net) |
| 14:33.22 | krivenok_dmitry | [Jul 29 18:32:50] WARNING[8211]: loader.c:409 load_dynamic_module: Error loading module 'chan_sip.so': /usr/local/asterisk-1.4.26/lib/asterisk/modules/chan_sip.so: undefined symbol: ast_parking_ext |
| 14:33.22 | krivenok_dmitry | [Jul 29 18:32:50] WARNING[8211]: loader.c:721 load_resource: Module 'chan_sip.so' could not be loaded. |
| 14:35.25 | *** join/#asterisk Keizer (n=keizer@64.238.20.94) |
| 14:42.20 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 14:42.43 | Kizmet | so what is everyone upto where they are be it day, night, morning, midnight? |
| 14:43.06 | Stese | I'm @ work... on IT Support... |
| 14:43.34 | Kizmet | I'm @ home, Doing work at midnight :-/ |
| 14:43.48 | *** part/#asterisk blee01 (n=blee01@237.177.205.68.cfl.res.rr.com) |
| 14:43.52 | Stese | Why.... |
| 14:44.14 | Kizmet | Training new staffers takes up too much of my time *sigh* |
| 14:44.50 | Kizmet | stares at his nearly empty bottle of Jim Beam :-/ |
| 14:45.03 | Stese | Do you admin a large system, or provide support for others using *] |
| 14:45.12 | Kizmet | Both :( |
| 14:45.18 | *** part/#asterisk krivenok_dmitry (n=krivenok@85.235.219.42) |
| 14:45.28 | Stese | oh right, ok |
| 14:46.01 | Kizmet | Its not as bad as my last job... supporting CCM :-/ |
| 14:46.02 | Stese | along side my normal job, i'm support our system of 14ish phones and a few softphones |
| 14:46.52 | Kizmet | Stese, doesnt sound to bad :) (I admin a fleet of 200ish Polycom 670 phones around the country) |
| 14:46.58 | Stese | and I deal with the few customers that we've sold * to |
| 14:47.37 | Stese | and I know very little about * and Linux... I'm getting there |
| 14:47.50 | ariel_ | I am admin to over 10K users and over 600 asterisk systems, There are 3 in our team, but it's all fairly easy if there setup correctly....hehehe |
| 14:48.07 | Stese | 0-o |
| 14:48.19 | Kizmet | (0_o) |
| 14:48.59 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 14:48.59 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:50.07 | *** join/#asterisk Gnarfy (n=mirc@213.144.244.115) |
| 14:50.22 | Kizmet | Poly phones are good for management *grins* Anycast FTP servers ftw. |
| 14:50.49 | Stese | we use Linksys SPA922's |
| 14:51.13 | Gnarfy | Ah finally, managed to connect :) Got a question if anyone is around to help! |
| 14:51.29 | Kizmet | plenty of us around :) |
| 14:52.21 | Gnarfy | Goody, here goes... i've got a sip trunk at mcxess, they have 2 telephone numbers connected to it, but i can't figure out which call is made to which phone number |
| 14:52.31 | Gnarfy | The sip headers say: To: <sip:s@myserversipadres> |
| 14:52.49 | Gnarfy | I was under the impression the s, should be the phonenumber dialed... |
| 14:53.19 | Kizmet | Gnarfy, certainly should, Is there a seperate trunk per number or just one trunk? |
| 14:53.51 | Gnarfy | 1 trunk, several numbers... otherwise i would just fix it a register with seperate /gohere behind it... |
| 14:54.14 | Kizmet | care to pastebin the relevant sip.conf entries? |
| 14:54.16 | Kizmet | ~pb |
| 14:54.17 | infobot | [~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/, or apt-get install pastebinit |
| 14:55.17 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:55.58 | Gnarfy | sip.conf |
| 14:55.58 | Gnarfy | [general] |
| 14:55.58 | Gnarfy | register=>account:passwrd@voip.mcxess.nl |
| 14:55.58 | Gnarfy | [mcxess] |
| 14:55.58 | Gnarfy | type=friend |
| 14:55.58 | Gnarfy | host=voip.mcxess.nl |
| 14:56.00 | Gnarfy | username=account |
| 14:56.02 | Gnarfy | fromuser=account |
| 14:56.04 | Gnarfy | secret=passwrd |
| 14:56.06 | Gnarfy | insecure=port,invite |
| 14:56.06 | Stese | Gnarfy.... |
| 14:56.08 | Gnarfy | ;trustrpid = yes |
| 14:56.10 | Gnarfy | ;sendrpid = yes |
| 14:56.12 | Gnarfy | context=0906xxxx |
| 14:56.13 | Stese | use Pastebin |
| 14:56.26 | Gnarfy | oeps sorry :x lets have a look :x |
| 14:58.45 | Gnarfy | like this then? http://pastebin.com/m5e6edc2a |
| 14:58.56 | Gnarfy | kinda new to the whole mirc thing :p |
| 14:58.56 | Kizmet | much better |
| 14:58.58 | Kizmet | thanks :) |
| 14:59.11 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 15:00.05 | Kizmet | extensions.conf under [0906xxxx] try defining a separate entry for each number instead of using s :) |
| 15:00.30 | Kizmet | remember to comment the s entried out for the moment. |
| 15:01.16 | Kizmet | s/entried/entries/ |
| 15:02.43 | Gnarfy | Yeah oke, i'll give that a try... but nonetheless, i would expect it to still show up in the sip headers when looking at the debug info |
| 15:03.00 | *** join/#asterisk Joel (n=jjshoe@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 15:03.38 | Kizmet | Its just a hunch, I remember coming across something like that before. |
| 15:04.20 | [TK]D-Fender | Gnarfy: Pastebin SIP DEBUG from a call attempt. |
| 15:04.52 | [TK]D-Fender | Gnarfy: And I only see one registration there. You have multiple DID's that land on the peer associated with it? |
| 15:06.07 | Gnarfy | Its 1 DID with several numbers connected to it, 10 sip trink (if i'm saying it correct) |
| 15:06.20 | Gnarfy | http://pastebin.com/mbc601b thats the sip debug from last test call |
| 15:07.25 | Gnarfy | Kizmet, i'll try the seperate numbers for the incoming calls as soon as all current calls are done :) |
| 15:07.55 | [TK]D-Fender | Gnarfy: I don't see the DID in any of those headers... |
| 15:08.40 | Kizmet | Gnarfy, Neither here. It does actually look like your provider is dialing s@you instead of did@you. |
| 15:08.47 | [TK]D-Fender | Gnarfy: Which if you do indeed have several landing on that single peer, I would consider a "shitty provider" issue. |
| 15:09.01 | [TK]D-Fender | Kizmet: issue is MULTIPLE DID's supposedly hit that peer |
| 15:09.47 | Joel | in sip.conf, in a register line, can you tell it which context to hit instead of an extension? like: register => user:pass@voip.com/incoming-calls |
| 15:09.48 | Gnarfy | I have 2 phonenumbers connected to the sip account, i raised a ticket at the provider... but they had to escalate it to someone who knew more, was just hoping it was something i did wrong |
| 15:09.53 | *** join/#asterisk moy (n=moy@bas1-unionville55-1177733615.dsl.bell.ca) |
| 15:10.13 | Kizmet | [TK]D-Fender, I understand that ;) I have a trunk that does the same thing they dial did@me not just s@me |
| 15:10.15 | Gnarfy | Joel: ye, but then the 2 phonenumbers on the 1 sip account would both go to incoming-calls still |
| 15:10.31 | jplank | correct me if I'm wrong, but I've always noticed, when * registers to a trunk, the calls come in as TO:s@whatever I've noticed that with multiple providers, and between multiple * boxes, I always assume it had something to do with the way * registers lines |
| 15:10.59 | Joel | jplank right, which is exactly why I'm asking ) |
| 15:11.00 | Joel | :) |
| 15:11.03 | [TK]D-Fender | ... |
| 15:11.37 | jplank | I always thought only peer trunks pass DNIS? |
| 15:12.37 | Gnarfy | Well, thanks for the info guys :) i'll try kizmet's suggestion and poke my provider some more! |
| 15:13.27 | jplank | Joel: you can have the sip.conf's entry point to a specific context, assuming I understand your question |
| 15:14.17 | Joel | jplank I need to poke into this more... for some reason I'm not receiving calls when a certain iax provider isn't in the top of iax.conf beofre any other providers |
| 15:14.24 | Joel | but yeah, we'll see how it goes :) |
| 15:15.11 | jplank | Joel, sounds like a syntax issue then, if asterisk is ignoring everything after the first entry in iax.conf, I bet you have a typo in that entry |
| 15:15.12 | *** join/#asterisk xpot-mobile (n=james@static-66-182-88-85.bbsc.net) |
| 15:15.22 | Joel | it's not ignoring it, it's just not matching |
| 15:15.36 | jplank | is there a difference? |
| 15:16.12 | Joel | yes |
| 15:18.32 | Kizmet | isnt too happy about paying over $100 for one call to an Inmarsat phone! ($19.80 p/m ex tax) :-/ |
| 15:21.20 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 15:22.18 | Joel | hrm weird, yeah, so an incoming call with another provider is somehow matching the top peer listing in my iax.conf :\ |
| 15:22.26 | Joel | and getting rejected since of course it has no authority |
| 15:23.09 | TSM | where does asterisk keep all of the settings for phones such as the DND status on the server? |
| 15:23.33 | Joel | TSM have you tried a database show? |
| 15:23.43 | TSM | mysql? |
| 15:23.50 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 15:23.54 | Joel | TSM what about mysql? |
| 15:24.11 | Kizmet | TSM, in the ast console do a 'database show' |
| 15:24.16 | kaldemar | asterisk doesn't keep such information on VoIP phones unless you tell it to. |
| 15:24.20 | TSM | ok |
| 15:24.42 | Kizmet | TSM, Mind you phones such as Polycom dont actually tell Asterisk about there DND status (by default anyway) |
| 15:24.43 | *** join/#asterisk Linuturk (n=linuturk@fluxbuntu/developer/Linuturk) |
| 15:25.37 | TSM | ive got polycom phones, prob is that i can make calls from it, but when i call it, asterisk always thinks its on DND or somthing like that, ive -r -vvvv and looked at the logs |
| 15:25.44 | [TK]D-Fender | TSM>where does asterisk keep all of the settings for phones such as the DND status on the server? <- no such thing |
| 15:26.06 | [TK]D-Fender | TSM: DND from SIP phones is a response, not a status |
| 15:26.37 | jplank | (he could be using something like freepbx that holds DND status inside astdb) |
| 15:26.45 | [TK]D-Fender | jplank: AKA = BS |
| 15:27.05 | TSM | im using elastix, which yes is freepbx |
| 15:27.19 | [TK]D-Fender | TSM: FreePBX != Asterisk |
| 15:27.30 | [TK]D-Fender | TSM: if its dialplan implements stuff, it isn't supported here. |
| 15:27.30 | TSM | i know freepbx is only a GUI |
| 15:27.34 | [TK]D-Fender | ~freepbx |
| 15:27.35 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 15:27.45 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 15:27.54 | jplank | TSM check out #freepbx |
| 15:29.13 | jplank | TSM it also can't hurt to check the extension inside "sip show peers" and "core show hints" to make sure * even sees it connected |
| 15:29.40 | TSM | ok, i still think its more core to asterisk, i think the phones are not staying logged on, ile play more, it was working before |
| 15:30.07 | [TK]D-Fender | TSM: What gives you that impression? What does it actually say? |
| 15:30.58 | jplank | what does sip show peers say about the extension? |
| 15:31.05 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 15:31.37 | TSM | jplank: it shows both phones there, with status OK |
| 15:32.16 | TSM | 2200 does not accept incomming calls, i get "extension 2200 has do not disturb enabled, or followme pre-ring returned busy" |
| 15:33.04 | jplank | dial *79 from the phone |
| 15:33.14 | jplank | and make sure followme isn't enabled |
| 15:33.29 | jplank | if the CLI says that, then its almost def a freepbx misconfig, not a * issue |
| 15:33.32 | TSM | working now, thanks :) |
| 15:33.35 | [TK]D-Fender | TSM : FreePBX GUI decisions, not * or phone status. |
| 15:33.36 | *** join/#asterisk Mango (n=Mango@S01060016b6853255.vc.shawcable.net) |
| 15:33.57 | jplank | TSM *79 = freepbx feature code to turn OFF DND. |
| 15:34.07 | TSM | i was doing toggle before, must have toggled it the other wayand not noticed |
| 15:34.24 | TSM | i put a new soundpack in for UK English and it did not have audio for the DND toggle etc... |
| 15:35.04 | *** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl) |
| 15:35.19 | TSM | if a audio file does not exist but is ref'd in the extentions conf, what happens, does it continue or drop out at that point in the script |
| 15:35.44 | TSM | i looked at the app-dnd-toggle script and the devstate is only toggled after the sound playback |
| 15:36.09 | *** join/#asterisk manxpower (n=eric@69.73.94.162) |
| 15:37.22 | cusco__ | hi |
| 15:37.25 | Mango | Hello. How can I find out what module I need to load in order to use a particular function? I want to test whether a variable is defined, but Asterisk says "Function EXISTS not registered" |
| 15:37.33 | cusco__ | what is the lightest of the lightest codecs available |
| 15:37.49 | Mango | In terms of Bandwidth, or CPU usage? |
| 15:37.54 | cusco__ | bandwidth |
| 15:38.09 | coppice | LPC10 |
| 15:38.17 | Mango | Check out http://www.bandcalc.com/ |
| 15:39.05 | coppice | for low bit rate codecs bandwidth is affected more by the ptime than the codec itself |
| 15:39.22 | *** join/#asterisk Kizmet (n=Kizmet@ppp59-167-37-126.lns2.cbr1.internode.on.net) |
| 15:39.25 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.229.58) |
| 15:39.40 | kaldemar | Mango: EXISTS comes with func_logic |
| 15:39.41 | *** join/#asterisk Dksaarth (n=Dks@dsl-145-216-83.telkomadsl.co.za) |
| 15:40.39 | Kizmet | night all :) |
| 15:41.23 | Stese | laters Kizmet |
| 15:41.50 | Kizmet | drops and sleeps on the keyboard.... ;) night. ttfn all |
| 15:41.58 | Mango | kaldemar: Thanks. Did you just know that or look it up somewhere? |
| 15:42.00 | *** join/#asterisk ArchGT (n=ArchGT@unaffiliated/archgt) |
| 15:44.58 | mnicholson | oh, i missed a discussion about lua earlier. You guys waited till i left to have it :) |
| 15:45.03 | Dksaarth | Hi guys - getting a kernel panic when trying to load the kernel modules for a TE420B |
| 15:45.30 | Dksaarth | tried many different asterisk versions, two different computers |
| 15:45.33 | Dksaarth | no irq conflicts |
| 15:46.26 | Dksaarth | Next plan of action is to try return it unless somebody can suggest something ? |
| 15:46.45 | mnicholson | btw, pbx_lua is not an abstraction over extensions.conf/pbx_config, it is a completely separate pbx switch |
| 15:49.42 | kaldemar | Mango: had a hunch and looked at the source file |
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| 15:50.29 | Mango | kaldemar: You rock; thanks :) |
| 15:51.52 | Dksaarth | Anybody know what "TE4XXP Version Synchronization error" means / what it could be ? |
| 15:52.02 | manxpower | Dksaarth: Your problem is exactly the issues that Digium will provide free support for. |
| 15:52.18 | Dksaarth | (this is on two different computers, installed from scratch using latest trixbox as well as older ones) |
| 15:52.51 | Dksaarth | manxpower: Thanks - time is a issue here, I have raised the issue with digium - was hoping maybe somebody here had a similar experience |
| 15:53.06 | Dksaarth | <PROTECTED> |
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| 16:01.43 | manxpower | Asterisk is not the tool for you if you are rushed. |
| 16:05.50 | Dksaarth | thanks ManxPower, that is helpful. I am rushed now, because it seems the hardware is broken. I was not rushed when I set up this box originally |
| 16:05.57 | yangp | Is any of you using ENUM technology ? |
| 16:06.41 | *** join/#asterisk bhodder (n=blake@142.166.111.234) |
| 16:06.52 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:07.22 | bhodder | hey can anyone tell me how to set caller id so that it passes to my sip trunk |
| 16:08.08 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 16:08.24 | bhodder | I have been looking at google and have tried a few things but nothing seems to work? |
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| 16:11.59 | bhodder | Is it possible to set caller ID in asterisk and have it sent through the sip trunk? |
| 16:12.14 | Mango | Yes. |
| 16:12.34 | Mango | in sip.conf, in the context of the device in question: |
| 16:12.45 | Mango | callerid="It's Mango!" <5551234567> |
| 16:12.50 | Mango | Does that help? |
| 16:13.18 | bhodder | Yes, I think so. I am going to give that a try now. Thanks Mango |
| 16:13.50 | Mango | There are also examples here of how to set it within the dial plan: |
| 16:13.50 | Mango | http://www.voip-info.org/wiki/view/Asterisk+func+callerid |
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| 16:15.03 | leifmadsen | hmmmm... now I think I should change my CID to something fun like, "It's Leif!" |
| 16:18.42 | bhodder | I have tried both methods but neither seem to work |
| 16:19.39 | leifmadsen | 1) it is possible your provider is blocking it, or resetting it to what they have on file |
| 16:19.51 | leifmadsen | 2) you may need to enable 'sendrpid=yes' in sip.conf |
| 16:20.02 | leifmadsen | didn't read the scrollback, so that may have already been mentioned |
| 16:20.41 | bhodder | I had asked him and he had said that if it is configured right in asterisk it will get passed through |
| 16:21.01 | leifmadsen | then I suggest you enable sendrpid |
| 16:21.10 | bhodder | however it simply shows up as private caller |
| 16:21.37 | bhodder | oh ok |
| 16:22.01 | mmattice | is it "normal" for a pri to take a bit to stablize after a reboot? |
| 16:22.56 | ManxPower | mmattice: PRIs should take less than 1 min to stabilize after asterisk starts. |
| 16:23.47 | *** join/#asterisk bijit (i=1000@190.241.15.48) |
| 16:23.56 | *** join/#asterisk _abc_ (n=no@bas4-toronto02-1167989859.dsl.bell.ca) |
| 16:24.03 | mmattice | ManxPower: that's what I'd think |
| 16:24.16 | bhodder | so I need to enable sendrpid for it to go through |
| 16:25.24 | mmattice | http://pastebin.ca/1511181 |
| 16:26.01 | ManxPower | mmattice: 1) your PRI is screwed up or 2) you screwed up your PRI config |
| 16:26.52 | ManxPower | oh! or 3) you have a Sangoma card and the Sangoma drivers do not support your version of Zaptel/DAHDI,and libPRI |
| 16:26.56 | mmattice | it's nominally stable, but it takes more than a minute. |
| 16:26.57 | bhodder | thanks leifmadsen |
| 16:27.09 | mmattice | it's a brand new te200b with dahdi driving it. |
| 16:27.42 | mmattice | takes closer to 5 minutes to stabilize |
| 16:28.10 | mmattice | well, actually, it passed a call, but now it's not. |
| 16:28.42 | bmoraca | too bad it's not a sangoma...their debug utilities are much superior to digium's |
| 16:29.55 | ManxPower | bmoraca: what is the sync source set for? |
| 16:30.09 | bmoraca | ? |
| 16:30.13 | TSM | has anyone been able to get polycoms to send its DND status to * |
| 16:30.20 | bmoraca | i'm not having a problem |
| 16:30.48 | mmattice | odd... maybe dahdi_cfg isn't getting run properly on boot. |
| 16:31.20 | TSM | i know DND works, but its local to the phone so * needs to check directly with the handset first |
| 16:31.52 | ManxPower | sorry, that was for mmattice |
| 16:31.54 | tzafrir_laptop | mmattice, what's the output of: dahdi_hardware; lsdahdi |
| 16:33.07 | mmattice | tzafrir_laptop: http://pastebin.ca/1511185 |
| 16:34.02 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 16:34.12 | ManxPower | tzafrir_laptop: looks like he has the wrong timing source? |
| 16:34.25 | *** join/#asterisk eans (n=eans@118.100.181.93) |
| 16:34.55 | mmattice | clocksource is the first and only up port |
| 16:35.05 | eans | How to start OP Panel via SSH? |
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| 16:35.24 | mmattice | the second is down as it is just a dsx-1 to our "regular" phone system |
| 16:35.26 | ManxPower | mmattice: clock source should be 1, IIRC |
| 16:35.29 | mmattice | and I pulled it. |
| 16:35.41 | ManxPower | (for the span connected to the telco) |
| 16:35.57 | mmattice | ManxPower: yeah, that's the way lsdahdi reads to me |
| 16:36.29 | mmattice | span 1 is connected to the telco. |
| 16:36.45 | *** join/#asterisk shazaum (n=889uJWER@unaffiliated/shazaum) |
| 16:37.00 | mmattice | anyway, I ran dahdi_cfg -v by hand, it put the span into red alarm, then came right back out and was stable immediately. |
| 16:37.04 | ManxPower | mmattice: what is the span= line for the telco span? |
| 16:37.58 | mmattice | ManxPower: span=1,1,1,esf,b8zs |
| 16:38.05 | mmattice | SPAN 1: ESF/B8ZS Build-out: 133-266 feet (DSX-1) |
| 16:39.02 | ManxPower | mmattice: Your Asterisk server is 133Â -Â 266Â feet from the telco smartjack? |
| 16:39.16 | *** join/#asterisk Defraz (n=T0tal@24-117-156-215.cpe.cableone.net) |
| 16:44.00 | mmattice | yeah, it's on the other side of the building |
| 16:44.30 | mmattice | it's probably just around 140-150 ft of cable |
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| 16:56.27 | mmattice | yeah, if I reboot the machine, it takes another execution of dahdi_cfg to straighten things out |
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| 17:00.38 | mmattice | and within moments of it coming out of red I get the D-channel and all the B's get restarted |
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| 17:21.06 | WhIteSidE | Hello all, I'm having a bit of trouble with 1.6.1.1 and a digium TDM 400 fxs port. The port works fine, I can dial to it (i.e. the phone rings, call bridges, conversation works fine), but I cannot dial from it. When I pick up the handset I immediately get a dialtone, however, as soon as I dial even a single digit, the phone "hangs up" |
| 17:22.13 | *** join/#asterisk werdan7 (n=w7@freenode/staff/wikimedia.werdan7) |
| 17:24.58 | leifmadsen | WhIteSidE: pastebin what is going on at the console |
| 17:25.09 | WhIteSidE | Hang on |
| 17:25.09 | leifmadsen | sounds like what you're dialing can't be matched in the dialplan |
| 17:25.32 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 17:26.31 | WhIteSidE | http://pastebin.com/d1294313b |
| 17:26.54 | leifmadsen | show the dialplan that the channel is associated with |
| 17:27.03 | WhIteSidE | The fxs is on "port 3" of a TDM 400 |
| 17:28.26 | WhIteSidE | http://pastebin.com/de32ecfc |
| 17:28.44 | WhIteSidE | The problem is, it doesn't let me dial a second digit |
| 17:29.21 | leifmadsen | try dialing faster? |
| 17:29.29 | leifmadsen | the default digit timeout could be pretty short |
| 17:29.46 | WhIteSidE | It's "instantaneous", where can I change the timeout? |
| 17:29.49 | leifmadsen | I think you can modify that in features.conf -- unless there is an option in chan_dahdi.conf for that |
| 17:30.05 | leifmadsen | never seen that before though, so I'm out of ideas |
| 17:30.21 | [TK]D-Fender | WhIteSidE: We need to see your chan_dahdi.conf |
| 17:30.24 | leifmadsen | could also set it with the TIMEOUT() function |
| 17:31.18 | [TK]D-Fender | leifmadsen: Think thats a little off. the function is dialplan, but he's gets a reorder on DTMF. Thats a dialplan thing all but certainly. |
| 17:31.23 | [TK]D-Fender | leWhich we're about to see. |
| 17:31.25 | WhIteSidE | Also, it may be my chan_dahdi, the problem is, I'm instructed to use "#include dahdi-channels.conf" and then I'm setting context=internal. The problem is, I'd really prefer to set the context from each port manually. What I've been thinking is that I need to copy dahdi-channels into chan_dahdi and then manually set the context in each section? |
| 17:31.32 | [TK]D-Fender | leifmadsen: Which we're about to see. |
| 17:31.38 | *** join/#asterisk theron (n=theron@216.51.246.211) |
| 17:32.00 | [TK]D-Fender | WhIteSidE: Please pastebin it. |
| 17:32.25 | WhIteSidE | http://pastebin.com/d159fa9e7 |
| 17:32.30 | WhIteSidE | Sorry, took a moment to get it |
| 17:32.50 | WindowsUser | I might not be up to speed on terminology, if I answer a call on one phone and want to send it somewhere else, thats a "transfer"? |
| 17:33.01 | leifmadsen | yes |
| 17:33.07 | [TK]D-Fender | WhIteSidE: As shown, that will not work <- |
| 17:33.29 | [TK]D-Fender | WhIteSidE: the context line has to appear before the Channel =>" line for the channels youa re assigning |
| 17:33.35 | WhIteSidE | Ok, can you point to to a document on chan_dahdi? |
| 17:33.54 | leifmadsen | everything above a channel => is what gets assigned to it |
| 17:33.58 | [TK]D-Fender | WhIteSidE: Look for any for zapata.conf. the layout is the same |
| 17:34.27 | [TK]D-Fender | WhIteSidE: bunch or parameters, channels to apply to. More and/or different parameters, more channels. |
| 17:34.33 | WhIteSidE | [TK]D-Fender: So if I copy dahdi-channels into chan_dahdi and then add the context to each block? Thanks, I thought that the info went after Channel => |
| 17:34.48 | [TK]D-Fender | WhIteSidE: parms must appear before to apply |
| 17:34.56 | WhIteSidE | Thanks, I'll try it out |
| 17:35.10 | [TK]D-Fender | WhIteSidE: this will require reloading chan_dahdi.so, or easier, a restart of * |
| 17:35.14 | [TK]D-Fender | (not reload) |
| 17:35.19 | WhIteSidE | Yep |
| 17:38.32 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:41.02 | WhIteSidE | Thanks a LOT guys! Just putting the directives before the assignment made everything start working as expected |
| 17:41.47 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 17:44.31 | [TK]D-Fender | WhIteSidE: You're welcome |
| 17:45.01 | *** join/#asterisk bmg505 (n=leon@196-209-79-155-rndf-esr-5.dynamic.isadsl.co.za) |
| 17:48.15 | TSM | grr, trying to figure out why when i park a call of someone i have called internaly, the system then tries to connect me to the VM of that user |
| 17:48.25 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 17:49.07 | [TK]D-Fender | TSM: Not the place to ask... #freepbx <- |
| 17:49.25 | TSM | ok |
| 17:50.33 | levity | [TK]D-Fender: sorry for the personal question, are you a payed Digium employee? |
| 17:50.50 | [TK]D-Fender | levity: Nope. Independant Consultant |
| 17:51.06 | levity | you should be! |
| 17:51.13 | eppigy | TRABAJO |
| 17:52.53 | WhIteSidE | This is slightly OT, but does anyone have a wifi sip handset that they really like? |
| 17:54.11 | [TK]D-Fender | WhIteSidE: So far the Polycom/Kirk models are reported to be prettey decent as those go, howevere in general : |
| 17:54.17 | [TK]D-Fender | ~wifivoip |
| 17:54.18 | infobot | [~wifivoip] Wi-Fi (802.11a/b/g) VoIP phones on the market may possess any of the following drawbacks : poor battery life, limited range, lack of call features (many lack transfer, etc), poor or no NAT traversal, shoddy construction, etc. Because of this they are usually not recommended. Some alternatives are : VoIP ATA + cordless, DECT w/ VoIP base, etc |
| 17:55.17 | WhIteSidE | I've been considering the cordless + fxs option. Seems kind of "hacky", though |
| 17:55.50 | bmoraca | WhIteSidE: i've used the Snom M3, and while it "worked", it was expensive and didn't do anything that ATA+Cordless phone could do |
| 17:55.54 | [TK]D-Fender | WhIteSidE: Not really... |
| 17:55.57 | WhIteSidE | Ok |
| 17:56.15 | [TK]D-Fender | bmoraca: Hav you had the range & volume issues other have reported? |
| 17:56.18 | bmoraca | WhIteSidE: i've heard good things about the polycom spectralinks, but I've not had the chance to use them. Overall, ATA+Cordless phone works very well |
| 17:56.28 | *** join/#asterisk grandpapadot (n=no@99-175-248-81.lightspeed.brhmal.sbcglobal.net) |
| 17:56.42 | bmoraca | [TK]D-Fender: no...biggest complaint is battery life. though the office where it's used isn't very big |
| 17:57.02 | [TK]D-Fender | bmGah |
| 17:57.17 | WindowsUser | is dect w/ voip expensive compared to ATA+cordless? |
| 17:57.23 | bmoraca | basically, the receptionist who uses it just needs to set it in the charger when she sits at her desk. |
| 17:57.27 | bmoraca | WindowsUser: yes. very. |
| 17:57.51 | WhIteSidE | bmoraca: Ok, the only problem I'm facing is allowing call transfers. I've never gotten it working from an analog extension. |
| 17:57.58 | WhIteSidE | to a sip extension |
| 17:58.14 | bmoraca | WhIteSidE: you need to work out your features.conf then. it works perfectly well. |
| 17:58.32 | bmoraca | both blind and attended transfers are possible from a phone connected to an ATA |
| 17:58.36 | WhIteSidE | Ok, I've got the wiki article up now |
| 17:59.44 | bmoraca | WhIteSidE: you also need to make sure that you have the appropriate dial option on your dial string to allow transfers to that phone. on outbound calls, you need to allow CALLER transfers, on inbound calls, you need CALLEE transfers. they are two different options for your dial string |
| 18:00.03 | [TK]D-Fender | WhIteSidE: You can use flash transfers with your DAHDI channels. |
| 18:00.13 | [TK]D-Fender | WhitNo need for DTMF transfers. |
| 18:00.38 | WhIteSidE | Ok, I'm reading up on this now, I'll probably be back with questions in a few |
| 18:02.09 | [TK]D-Fender | WhIteSidE: cantransfer=yes , threewaycalling=yes , callwaiting=yes |
| 18:02.57 | *** join/#asterisk AlmightyOatmeal (n=jamie@66-190-59-171.dhcp.mdsn.wi.charter.com) |
| 18:03.35 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 18:03.52 | AlmightyOatmeal | how would i go about having an inbound call ring all extensions at once and whomever picks up the call gets it? |
| 18:04.14 | grandpapadot | AlmightyOatmeal: Dial(SIP/801&SIP/802&SIP/803|30) |
| 18:04.31 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 18:04.38 | grandpapadot | AlmightyOatmeal: Just use "&" to join your peers with Dial() |
| 18:04.57 | AlmightyOatmeal | can i do what i want without manually specifying those extensions? :) |
| 18:04.57 | jaytee | anyone work for a non-profit organization out there? |
| 18:05.16 | [TK]D-Fender | AlmightyOatmeal: No |
| 18:05.27 | AlmightyOatmeal | oh, okay |
| 18:05.32 | [TK]D-Fender | jaytee: Atheism : A non-prophet organization |
| 18:05.57 | jaytee | [TK]D-Fender, sweet looking deal here if you're a 501c http://www.techsoup.org/stock/product.asp?catalog%5Fname=TechSoupMain&category%5Fname=Sun&product%5Fid=HW%2D44165&Cat1=Sun&CatCount=1 |
| 18:06.05 | KavanS | bmoraca, I've been using siphon for iphone, it's a pretty nice SIP/WIFI solution |
| 18:06.32 | grandpapadot | siphon does work pretty good |
| 18:06.44 | [TK]D-Fender | jaytee: OMMFG |
| 18:06.54 | jaytee | 190 bucks includes shipping |
| 18:07.14 | [TK]D-Fender | grandpapadot: siphon sucks :p |
| 18:07.15 | jaytee | and it'll run CentOS if you don't like Solaris |
| 18:07.32 | KavanS | [TK]D-Fender, heh bad experience with it? |
| 18:07.33 | grandpapadot | [TK]D-Fender: Compared to? (is there a better iphone sip app I've missed?) |
| 18:07.45 | jplank | AlmightyOatmeal: I guess you can create a macro that rings all the extensions you want, and then just point the did to that macro, but that will be the same is just using the dial command and "&" |
| 18:07.48 | KavanS | grandpapadot, I've tried a couple ...weephone and the others in the app store are a joke |
| 18:08.03 | grandpapadot | [TK]D-Fender: Oh, they all pretty much suck, but in a pool of suckage, syphone seems to float to the top (for me at least).. |
| 18:08.09 | KavanS | grandpapadot, use insomnia + siphon and you can get incoming calls after the iphone "sleeps" |
| 18:08.20 | *** join/#asterisk dajhorn (i=dajhorn@sueno.dreamhost.com) |
| 18:08.32 | AlmightyOatmeal | jplank: i suppose, but i think manually adding them with & is an easier idea than creating additional scripts ;) |
| 18:08.33 | *** join/#asterisk boynas (n=gflores@wsip-98-190-136-194.ph.ph.cox.net) |
| 18:08.35 | grandpapadot | ... float to the top in a pool of shit-apps and broken 1.0's |
| 18:08.36 | KavanS | I guess it's a hack...but I don't know of too many other solutions, maybe WinMO (gasp) has something swell for the SIP side |
| 18:09.01 | jplank | I agree, you just asked for an alternative, and that is one. |
| 18:09.18 | AlmightyOatmeal | much appreciated hehe :) |
| 18:09.20 | jplank | You could also make a queue and just point the call to the queue, then only people logged into the queue would get the call |
| 18:09.40 | AlmightyOatmeal | i thought about that but then it hit me, theres like less than 10 extensions so... hehe |
| 18:09.45 | [TK]D-Fender | jplank: And he'd still have to add them as MEMBER's, not just as part of a dial. |
| 18:10.03 | jplank | well, he could make it that they log in and out of the queue themselves |
| 18:10.29 | jaytee | doesn't the term "pool of suckage" pretty much sum up the whole Apple experience? |
| 18:10.29 | jplank | I'm not saying its a better solution (I'm not a fan of replacing ring groups with queues) I'm just saying |
| 18:10.44 | DigitalDaz21 | Can anyone recommend a good book on sip or a decent online tutorial please? |
| 18:10.47 | AlmightyOatmeal | jaytee: more or less yes |
| 18:10.54 | jplank | ~book |
| 18:10.55 | infobot | [~book] 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://astbook.asteriskdocs.org or see ~buybook |
| 18:10.58 | jaytee | DigitalDaz21, SIP Demystified |
| 18:11.07 | DigitalDaz21 | cool |
| 18:11.09 | boynas | Anybody nows how the faxdetect=incoming option works??? I am trying to use it to disable echocancellation when asterisk detects fax tone before it forwards the fax to an IAX trunk... Any ideas? The thing is that when I enable that and asterisk detects a fax it tells me that there is no fax channel in the machine.. and really it shoudnt.. the fax termination is in a remote asterisk server via IAX. |
| 18:11.25 | KavanS | jaytee, no...I disagree |
| 18:11.39 | KavanS | jaytee, I'd say the iphone is an exception to this... |
| 18:11.54 | DigitalDaz21 | My aim is to be able to debug traffic like register and invites, etc |
| 18:11.58 | KavanS | I try to leave out their other pieces of work when evaluating 1 |
| 18:12.03 | jaytee | KavanS, in which direction? Apples are great but iPhone is suckage or the reverse? |
| 18:12.10 | KavanS | jaytee, iphone is a great piece of hardware |
| 18:12.23 | KavanS | jaytee, I dropped my blackberry for it... |
| 18:12.28 | WindowsUser | would features.conf apply to using an ATA or are all the features done on the ATA? |
| 18:12.52 | KavanS | there are many things you could find that are incomplete...however I think it's definitely an improvement of portable computing |
| 18:13.33 | jaytee | KavanS, seems to be the general consensus. I know alot of people love the iPhone but hate Macs as personal computers and then there are the fanbois that if you stamped the Apple logo on a dried turd and slapped a $999.99 price tag on it they'd buy it without even asking what it does. |
| 18:13.57 | [TK]D-Fender | WindowsUser: ATA's do their own tranfers etc and * doens't have to do anything special |
| 18:14.07 | KavanS | jaytee, yeah, several years ago I approved of their products...but honestly, I'm just partial to the phone and overall user interface |
| 18:14.41 | KavanS | openmoko is a great idea, but just that...a great idea...from my understanding it's not even "working" |
| 18:14.58 | KavanS | so until someone offers me some other portable device I can ssh into, I'm going to stick with this guy ;) |
| 18:15.04 | jaytee | KavanS, nope, not anymore. some hope to revive it but only time will tell |
| 18:15.50 | [TK]D-Fender | KavanS: Too bad the dev imploded |
| 18:16.15 | BlargMaN00 | is safe_asterisk depricated, or is there a better solution for automatic restart?? |
| 18:17.23 | *** join/#asterisk Orbixx (i=Orbixx@office.exoware.net) |
| 18:17.24 | *** join/#asterisk dmz (n=dmz@64.203.233.195.dyn-cm-pool35.hargray.net) |
| 18:17.26 | leifmadsen | why do you think it is deprecated? |
| 18:17.39 | Orbixx | Why would DTMF be recognised over IP but not (or just the first number) when called from landline/mobile/cellphone? |
| 18:17.51 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 18:18.11 | KavanS | Orbixx, what provider? |
| 18:18.30 | Orbixx | KavanS: Multiple providers, British Telecom and Hutchinson 3G in the UK. |
| 18:18.34 | BlargMaN00 | just curious cause it's not in the default init.d script that asterisk creates by default on a debian install... (from source) |
| 18:18.37 | Orbixx | And probably others. |
| 18:18.44 | Orbixx | KavanS: I don't think it's a carrier problem. |
| 18:18.59 | KavanS | Orbixx, I've experienced some of the same....I've had at&t not send the correct DTMF's to my SIP provider |
| 18:19.03 | KavanS | I'm not sure why... |
| 18:19.23 | Orbixx | It could be carrier I suppose, but it does seem odd to be affecting a significant number of them. |
| 18:19.36 | WindowsUser | [TK]D-Fender: good to know |
| 18:19.36 | BlargMaN00 | i will just rewrite the init.d script to use it instead... |
| 18:19.48 | *** join/#asterisk batphone (n=wclayton@66.219.32.14) |
| 18:19.52 | *** join/#asterisk voxter (n=voxter@190.241.15.56) |
| 18:20.01 | batphone | anyone have the XML for a cisco 7916 expansion module? |
| 18:20.06 | batphone | i cant seem to find it anywhere |
| 18:20.27 | [TK]D-Fender | Orbixx: Unless its inband, and those other sources degradation pushes things over the edge |
| 18:21.22 | Orbixx | [TK]D-Fender: Yeah, so there's no common issue with DTMF working IP -> IP but not with any other combination? |
| 18:21.32 | Orbixx | (in your experience) |
| 18:21.34 | BlargMaN00 | batphone: are you running SIP or Skinny on that phone? |
| 18:21.55 | grandpapadot | inband DTMF via SIP will always be an issue as long as there's such a thing as packet loss. |
| 18:22.01 | grandpapadot | rather, via RTP |
| 18:22.12 | grandpapadot | inband is basically just sending the tones in the audio |
| 18:22.18 | BlargMaN00 | batphone: the expansion modules will *ONLY* work with Skinny... *NOT* SIP |
| 18:22.45 | [TK]D-Fender | Orbixx: Line what? |
| 18:22.58 | batphone | SIP |
| 18:23.01 | batphone | lol |
| 18:23.05 | batphone | what the crap |
| 18:23.08 | Orbixx | [TK]D-Fender: Sorry? |
| 18:23.09 | batphone | thanks BlargMaN00 |
| 18:23.30 | BlargMaN00 | batphone: no worries... |
| 18:23.32 | bmoraca | batphone: polycom's expansion modules look better anyway |
| 18:23.32 | [TK]D-Fender | [14:21]<Orbixx>[TK]D-Fender: Yeah, so there's no common issue with DTMF working IP -> IP but not with any other combination? <-- not sure what you;'re getting at |
| 18:23.38 | batphone | bmoraca: agreed |
| 18:23.48 | [TK]D-Fender | bmoraca: Aastra's = Godly |
| 18:24.05 | [TK]D-Fender | bmoraca: I just hate the phones they're attached to :D |
| 18:24.21 | bmoraca | was gonna say..only every used one aastra phone and i never want to use another |
| 18:24.29 | Orbixx | [TK]D-Fender: I'm asking you if you know of any common issue that causes DTMF to malfunction *outside* of IP to IP communications, basically. (other than what you've already mentioned) |
| 18:24.47 | grandpapadot | Orbix: If you're IP->IP DTMF is inband via SIP/RTP and there is any packet loss that changes the actual "tone" sent then there will be problems. |
| 18:24.48 | batphone | gonna get my receptionist a new polycom with a couple of expansion modules |
| 18:24.50 | [TK]D-Fender | Orbixx: Well what are you using outside of this? |
| 18:24.58 | AlmightyOatmeal | how hard would it be to integrate something in the inbound dialplan to check for the time and be like if before 0900 or after 1700 go to context blah else ring all phones? |
| 18:25.12 | grandpapadot | Orbixx: Best to get it from the PSTN then to rfc2833 as soon as possible. |
| 18:25.15 | bmoraca | batphone: why not use iSymphony or something of the like? |
| 18:25.31 | [TK]D-Fender | AlmightyOatmeal: "core show application gotoiftime" |
| 18:25.36 | batphone | anyone know if skinny on asterisk supports that module? |
| 18:26.14 | jaytee | <8 GotoIfTime |
| 18:26.21 | AlmightyOatmeal | [TK]D-Fender: sweet! :D |
| 18:26.53 | Orbixx | [TK]D-Fender: I'm connecting to a trunk via IAX which gives me an incoming/outgoing number. |
| 18:26.57 | batphone | bmoraca: to be precise, i am running a switchvox system right now |
| 18:26.59 | Orbixx | IAX2* |
| 18:27.08 | batphone | bmoraca: soon to be replaced with 1.6 >=| |
| 18:27.26 | [TK]D-Fender | Orbixx: So how is that not IP-IP? |
| 18:27.31 | batphone | i take it this isymphony stuff is the predecessor of the flash operator panel? |
| 18:27.39 | grandpapadot | Orbixx: Think of inband dtmf as the actual "tone" being recorded as part of the audio stream and rfc2833 dtmf sends the actual "digits" in the SIP packets, hence, rcf2833 is inherently more reliable. |
| 18:27.48 | [TK]D-Fender | batphone: Similar role, and much newer and bigger |
| 18:27.55 | Orbixx | [TK]D-Fender: Because at some point, it starts with my analogue phone. |
| 18:28.04 | bmoraca | batphone: predecessor? hahaha. isymphony is to FOP as an aston marton is to a pinto |
| 18:28.05 | [TK]D-Fender | Orbixx: ..... |
| 18:28.19 | [TK]D-Fender | Orbixx: thats every schmuck you call on the PSTN. |
| 18:28.30 | [TK]D-Fender | Orbixx: You're link is still IP until your ITSP |
| 18:28.46 | [TK]D-Fender | Orbixx: And on IAx, DTMF is always done at the far point. |
| 18:29.15 | batphone | sweet |
| 18:29.15 | batphone | is this free?! |
| 18:29.26 | batphone | apparently.. |
| 18:29.57 | bmoraca | batphone: they offer a free, limited, 5-user version of iSymphony, yes |
| 18:31.04 | batphone | ahh |
| 18:31.09 | batphone | 5 users, as in 5 phones? |
| 18:31.26 | batphone | thats gay-ish |
| 18:31.26 | bmoraca | no, as in 5 people logged in at once |
| 18:31.33 | batphone | thats less gay-ish |
| 18:31.46 | batphone | ugh java |
| 18:31.50 | batphone | im liking this less and less |
| 18:32.03 | WindowsUser | can externhost and register => in sip.conf use dnsmgr? I'm assuming i can avoid asterisk 1.6.1.1 hanging while a dns query fails |
| 18:32.05 | bmoraca | batphone: it's by far the best PC-based operator panel |
| 18:32.33 | *** part/#asterisk levity (i=canuck@unaffiliated/canuck) |
| 18:33.15 | *** join/#asterisk cusco (n=tralala@2001:0:53aa:64c:3427:8713:2ac0:762d) |
| 18:33.18 | cusco | hello |
| 18:33.23 | cusco | sorry |
| 18:33.45 | bmoraca | batphone: if you need more than 5 users or need some of the other advanced features (queue monitoring, etc), then you can pay for it. it's a great product and everyone deserves to make a little money. i've sold it to a number of people, and they all enjoy it. |
| 18:34.54 | cusco | as somebody said before, LPC10 is the lightest codec available (regarding bandwidth) right? |
| 18:35.16 | cusco | but voice sounds robotic... is there one that voice is not as robotic, and can be cosidered the lightest? |
| 18:35.24 | WhIteSidE | Ok, I think I'm going with a Linksys PAP2T and a 2 line cordless. Any thoughts? |
| 18:35.30 | *** join/#asterisk sjobeck (n=Adium@c-71-193-223-71.hsd1.or.comcast.net) |
| 18:35.43 | bmoraca | WhIteSidE: PAP2Ts aren't bad |
| 18:35.45 | *** part/#asterisk sjobeck (n=Adium@c-71-193-223-71.hsd1.or.comcast.net) |
| 18:35.47 | grandpapadot | cuso g729a |
| 18:36.13 | cusco | is it free? |
| 18:36.20 | cusco | to use in comercial use, grandpapadot ? |
| 18:36.28 | grandpapadot | cusco: No, $10 per channel, cheap. |
| 18:36.29 | WindowsUser | is the PAP2T the one with two g729 licenses? |
| 18:36.44 | [TK]D-Fender | LPC10 = Domo-arrigato Mr. Roboto! |
| 18:36.49 | [TK]D-Fender | WindowsUser: No |
| 18:37.04 | cusco | grandpapadot: yes, I heard about that one, isn't there a free implementation? |
| 18:37.05 | grandpapadot | WindowsUser: No, the PAP2T only does a single g729 channel at a time |
| 18:37.08 | [TK]D-Fender | PAP2 = 1 G.729. subsequent calls = G.711 |
| 18:37.13 | grandpapadot | cusco: Yea, but it's crap |
| 18:37.30 | grandpapadot | cusco: For $10/channel, the licensed codec is cheap and it's very good quality considering. |
| 18:37.45 | cusco | can i use g729a for testing purposes before I get a licence? |
| 18:38.00 | grandpapadot | cusco: Yea, you can spend $30 and test with 3 channels all you want. |
| 18:38.01 | cusco | and what is the free implementation called? I would like to compare... |
| 18:38.26 | grandpapadot | cusco: The free implementation can't be used commercially, it's encumbered. |
| 18:38.37 | cusco | how come? |
| 18:38.50 | grandpapadot | Dunno, I just spend the $$ and use the one everybody else does from digium. |
| 18:39.24 | cusco | what is the free implementation called? |
| 18:39.29 | grandpapadot | cusco: Are you going asterisk->asterisk? |
| 18:39.37 | grandpapadot | cusco: over an internet or wan link? |
| 18:39.37 | cusco | no |
| 18:39.39 | WindowsUser | cusco: digium charges for g729a due to patent licensing, we're not going to help you with that, search yourself |
| 18:39.44 | cusco | yes |
| 18:40.18 | cusco | i don't mind paying for it. |
| 18:40.25 | grandpapadot | cusco: have you tried ilbc? |
| 18:40.28 | cusco | but if there is a free legal implementation... why not use it? |
| 18:40.37 | [TK]D-Fender | cusco: It isn't legal <- |
| 18:40.39 | grandpapadot | cusco: It sucks, use the $10 one. |
| 18:40.42 | cusco | grandpapadot: no... |
| 18:40.55 | bmoraca | wow |
| 18:41.06 | grandpapadot | cusco: if you're going from asterisk->asterisk, ilbc may be a good alternative if you're too cheap to spend $10/channel. |
| 18:41.18 | cusco | grandpapadot: I believe, tho I would like to try it |
| 18:41.24 | boynas | Anybody nows how the faxdetect=incoming option works??? I am trying to use it to disable echocancellation when asterisk detects fax tone before it forwards the fax to an IAX trunk... Any ideas? The thing is that when I enable that and asterisk detects a fax it tells me that there is no fax channel in the machine.. and really it shoudnt.. the fax termination is in a remote asterisk server via IAX. |
| 18:41.30 | cusco | grandpapadot: im going asterisk -> internet |
| 18:41.42 | cusco | I don't mind paying for the codec |
| 18:41.44 | [TK]D-Fender | iLBC =13/15kbps. Slightly heavier than GSM |
| 18:41.44 | grandpapadot | cusco: The Internet doesn't support sip, I just called and asked them. |
| 18:42.18 | cusco | grandpapadot: sorry, I mean, voip over a ISP to a landline phone |
| 18:42.57 | bmoraca | cusco: if you don't mind paying for the codec, what's the problem? the best solution is digium's g729...use it! |
| 18:42.59 | *** join/#asterisk _abc_ (n=no@69-165-134-188.dsl.teksavvy.com) |
| 18:43.15 | grandpapadot | cusco: What is your Asterisk server connecting to? An ITSP? |
| 18:43.39 | cusco | dunno. its optic fibre from our isp |
| 18:43.39 | grandpapadot | cusco: TK says GSM may be a better choice, and it's free. |
| 18:43.43 | _abc_ | does anyone here know about the meetme application on asterisk@NetBSD ? it seems to be ignored and there is no such application in the pbx |
| 18:43.57 | grandpapadot | cusco: Your asterisk server has fiber connected to it somehow? |
| 18:44.42 | grandpapadot | cusco: Or are you trying to say your internet connection is fiber, if so, what type of VOIP server or service are you connecting your asterisk server to? |
| 18:44.43 | [TK]D-Fender | G.729 is better quality for basic voice. GSm does better with music, but is heavier (a bit), but free and low CPU usage |
| 18:44.55 | [TK]D-Fender | CPU may be a consideration depending. |
| 18:46.46 | cusco | grandpapadot: thats what I meant. I actually don't know what plan does the ISP offer us, boss set the asterisk server up... |
| 18:47.01 | _abc_ | does anyone here know about the meetme application on asterisk@NetBSD ? it seems to be ignored and there is no such application in the pbx |
| 18:47.06 | _abc_ | so what does one do to compile qemu+kqemu ? is there a script ? i was told here than kqemu is not supported here, and to use kvm ?! |
| 18:47.08 | [TK]D-Fender | cusco: Guess you'd better go and ask. |
| 18:47.09 | cusco | cpu is allright for now... bandwidth is not |
| 18:47.14 | _abc_ | i run asterisk on qemu |
| 18:47.18 | cusco | [TK]D-Fender: he is not in, but I will later |
| 18:47.33 | cusco | meanwhile I will see how to get 2 of those licenses |
| 18:48.01 | grandpapadot | cusco: http://store.digium.com/productview.php?product_code=G729CODEC |
| 18:48.06 | cusco | gotta go home now.. thank you for your patience |
| 18:51.41 | *** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001) |
| 18:52.19 | *** join/#asterisk propellerhead (n=yogurt2u@host62.200-117-118.telecom.net.ar) |
| 18:57.37 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 19:00.55 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 19:02.48 | *** join/#asterisk Victor_Yure_ (n=victor@unaffiliated/victoryure/x-837844) |
| 19:02.52 | jaytee | man, this job is just sooooo weird. walking back from our server room I pass one of our elephants, Kubwa, out for a walk with one of the keepers on the access road and she starts pissing while I pass her. you really gotta give them distance :-) |
| 19:03.16 | KavanS | hehe interesting |
| 19:03.20 | bmoraca | anyone used a TC400B? any input on how well it works? |
| 19:14.01 | *** join/#asterisk wasserja (n=wasserja@corporate.volstate.net) |
| 19:14.09 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 19:15.30 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:18.43 | *** join/#asterisk Dovid (n=annon@213.8.121.90) |
| 19:19.08 | Dovid | hi. Asterisk seems to crash because of moh |
| 19:19.18 | Dovid | http://pastebin.com/m32cf294e |
| 19:19.21 | Dovid | is a back trace |
| 19:22.10 | *** part/#asterisk seb- (n=seb@li30-51.members.linode.com) |
| 19:26.27 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 19:27.37 | Orbixx | [TK]D-Fender: Just done a tcpdump on incoming and outgoing IAX data - packets stop transmitting both ways when I send DTMF. |
| 19:30.56 | *** join/#asterisk hohum (n=dcorbe@206.71.169.115) |
| 19:32.18 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 19:33.52 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 19:37.41 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 19:39.51 | *** join/#asterisk korcan (n=korcan@76.234.161.17) |
| 19:40.09 | lesouvage | Is it possible to use Meetme() on an Asterisk system on a XEN vertiual machine. Dahdi loads and everything seems to be ok but after the "you are the only person in this conference" sound stops altough I used the m parameter. I run a Xen vm from Linode. |
| 19:40.47 | DigitalDaz21 | Can anyone tell me please how to get asterisk to send Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO with the REGISTER request? |
| 19:41.54 | [TK]D-Fender | DigitalDaz21: vi chan_sip.c |
| 19:42.12 | Dovid | lol |
| 19:43.51 | DigitalDaz21 | Do you think it not sending this with the REGISTER could cause a problem? |
| 19:49.00 | [TK]D-Fender | DigitalDaz21: Hasn't for for anyone else I've ever heard |
| 19:52.05 | *** part/#asterisk awkfu (n=awkfu@66.162.90.56) |
| 19:52.32 | WindowsUser | ~dnsmgr |
| 19:52.37 | WindowsUser | :( |
| 19:52.56 | *** join/#asterisk korcan_ (n=korcan@adsl-76-234-161-17.dsl.sfldmi.sbcglobal.net) |
| 19:54.29 | DigitalDaz21 | Aterisk shows as registering but still doesn't seem to receive any sip traffic when a call is placed, both eyebeam and office communicator send that line and work fine, thats the only difference I can see, also when I fire up either the eyebeam or communicator client, the asterisk box now receives sip calls fine, even though the softclients and asterisk are in two different locations |
| 19:56.15 | WindowsUser | is dnsmgr not really used by anything? |
| 20:02.24 | ManxPower | WindowsUser: it's not integrated into enough of Asterisk to be more good than a local DNS server. I don't know about 1.6, but that was the case in 1.4 |
| 20:03.26 | *** join/#asterisk zeeesh (n=zeeesh@203.215.179.43) |
| 20:04.40 | WindowsUser | does asterisk use /etc/hosts? if not i guess i could go the local dns server route |
| 20:05.53 | leifmadsen | /etc/hosts is part of the network infrastructure, not asterisk |
| 20:05.56 | leifmadsen | asterisk uses the network |
| 20:06.41 | WindowsUser | thats almost a no |
| 20:07.33 | DigitalDaz21 | asterisk does definitely seem to notice the hosts file, I think by default on most linux systems the hosts file is checked before a dns query is done |
| 20:08.25 | *** join/#asterisk korcan_ (n=korcan@adsl-76-234-161-17.dsl.sfldmi.sbcglobal.net) |
| 20:10.16 | Dovid | leifmadsen: can you have a look at a dump ? |
| 20:10.18 | Dovid | http://pastebin.com/m32cf294e |
| 20:10.24 | Dovid | oops. backtrace |
| 20:13.50 | WindowsUser | now to see if /etc/hosts is read in more than once |
| 20:14.43 | *** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu1.dynamic.dsl.tele.dk) |
| 20:17.35 | *** join/#asterisk many (n=many@dslb-092-075-125-033.pools.arcor-ip.net) |
| 20:18.03 | [TK]D-Fender | DigitalDaz21: Lack of packets has nothing to do with the content of the packet |
| 20:21.02 | ManxPower | DigitalDaz21: Asterisk tries to figure out the hostname for each IP in the system. I like to keep that info in /etc/hosts just in case DNS blows up |
| 20:22.29 | DigitalDaz21 | I'm just stabbing in the dark here, just cannot understand why the box doesn't get sent sip traffic, I also get the same with an INVITE, I get a 480 returned but again if I fire up a softclient with the same credentials the asterisk box can then make outgoing too |
| 20:30.09 | [TK]D-Fender | checkout time, bbiab |
| 20:33.35 | *** join/#asterisk sjobeck (n=Adium@c-71-193-223-71.hsd1.or.comcast.net) |
| 20:33.51 | *** part/#asterisk sjobeck (n=Adium@c-71-193-223-71.hsd1.or.comcast.net) |
| 20:34.52 | leifmadsen | Dovid: I didn't know I offered to help :) |
| 20:46.26 | ManxPower | I need a spam filter called "reject if from name has too many consonants" |
| 20:48.39 | *** join/#asterisk [TK]D-Fender (n=zsirc@161.216.183.5) |
| 20:55.33 | *** part/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 20:58.00 | *** join/#asterisk propellerhead (n=yogurt2u@host62.200-117-118.telecom.net.ar) |
| 20:59.16 | defsdoor | ManxPower: ASSP |
| 21:04.21 | *** join/#asterisk Xetrov` (n=xetrov@unaffiliated/xetrov/x-827361) |
| 21:12.09 | *** join/#asterisk ingenius (n=alektro@host215.200-45-165.telecom.net.ar) |
| 21:13.40 | generalhan | Anyone in here use the Xorcom Astribank devices to pass digital lines to analog ports for faxing? im trying to decide if this will be a better route than trying out Sangoma's cards with the Fax Sync Cable |
| 21:16.06 | generalhan | lol, of course he is the one that leaves after i post that question ! |
| 21:16.06 | generalhan | lol |
| 21:20.15 | *** join/#asterisk TSM2 (n=the_soft@87-194-32-212.bethere.co.uk) |
| 21:23.00 | TSM2 | ive been reading lots of things about dialup over asterisk but nothing definitive, whats my best options |
| 21:23.32 | TSM2 | i dont plan to put it over a WAN link, it would be nearly 100% over a LAN link to the local asterisk server then over the T1 link |
| 21:23.36 | *** join/#asterisk garymc (n=gar@host86-164-35-108.range86-164.btcentralplus.com) |
| 21:23.44 | garymc | Hi |
| 21:23.50 | garymc | Im looking for some advice |
| 21:24.03 | garymc | Ive got one isdn line 2e coming into my office |
| 21:24.54 | garymc | If I buy a digium single echo card for my server will i only beable to put voip phone on the network or will the digital lines still beable to go through my card and server? |
| 21:25.58 | garymc | Wow is none here? |
| 21:26.02 | garymc | no one |
| 21:26.33 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 21:26.55 | garymc | yo lesouvage |
| 21:27.04 | garymc | you know anything about asterirsk |
| 21:27.21 | garymc | I need advice and no one in here chatting |
| 21:28.31 | TSM2 | evreyone asleep here :) |
| 21:28.50 | garymc | TSM2 seems to be |
| 21:29.01 | garymc | you know anything about astreisk? |
| 21:29.24 | garymc | im just trying to sus stuff out before i buy in |
| 21:29.29 | TSM2 | haaaa, not good yet, been playing, so far ive heard that sangoma are better than digium |
| 21:29.37 | lesouvage | I just installed asterisk 1.6.1.1 with dahdi-inux and dahdi tools. Dahdi is running fine but lsmod shows that dahdi isn't use by anything. How do I get Asterisk that far that it will use Dahdi? with the 1.4 version it simply worked fter building. |
| 21:29.49 | lesouvage | garymc: give it a try |
| 21:30.27 | garymc | I cant fork out £300 for a card I dont know if I need or not |
| 21:30.36 | TSM2 | what you trying to do? |
| 21:30.52 | TSM2 | isdn2e as pstn lines into asterisk? |
| 21:32.44 | TSM2 | from what i understand, when i was playing with the sangoma PRI card, it was able to be setup to provision some of the circuits and use the others for data, i guess they are able to be used with other stuff, i would not try though |
| 21:32.54 | TSM2 | is this ISDN2E in the UK? |
| 21:33.20 | garymc | TSM2 yes ISDN2e in the uk |
| 21:33.21 | *** join/#asterisk propellerhead (n=yogurt2u@host183.190-136-104.telecom.net.ar) |
| 21:33.23 | lesouvage | Perhaps my question is wrong. How can I check that my Asterisk is Dahdi aware? |
| 21:34.51 | TSM2 | have you tried speaking to sangoma? they offer a guarantee that their cards work on any computer and asterisk, they may tell you how to do what you want |
| 21:35.06 | garymc | yes but i dont get no replys |
| 21:35.14 | garymc | I thought id try here cos people are using it |
| 21:36.06 | *** join/#asterisk theron (n=theron@216.51.246.211) |
| 21:36.38 | TSM2 | is the software you have that you need to access the BRI on the asterisk server or is it another peice of hardware you have? |
| 21:37.05 | garymc | sorry what is BRI? |
| 21:37.28 | lesouvage | ~book |
| 21:37.29 | infobot | [~book] 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://astbook.asteriskdocs.org or see ~buybook |
| 21:37.53 | TSM2 | BRI = Basic Rate Interface = ISDN2E |
| 21:39.00 | garymc | we are currently using BT VERSATILTY hardware but it doesnt record all calls etc which is one of the main things we need to do |
| 21:40.17 | TSM2 | thats handled in asterisk, you can turn it on so all channels are recorded to WAV |
| 21:40.44 | garymc | Whats handled in asterisk? Bt versatility? |
| 21:40.53 | TSM2 | recording |
| 21:40.57 | garymc | i know yes |
| 21:41.33 | garymc | but Bt Verstility give you upto 30 extension lines and not through voip as voip is 100% guarteed to work that well |
| 21:41.45 | garymc | voip isnt 100% |
| 21:42.24 | garymc | so would putting my ISDN2e Line into a ASTERISK machine kill my digital lines and just give me voip |
| 21:42.33 | garymc | what it is I dont really understand it all |
| 21:43.16 | TSM2 | ok, i get it now, you want to keep the versatility for your phones, but you want to use asterisk as a recording system and thats all |
| 21:44.56 | garymc | if i have too, id rather just have the one system that did all |
| 21:45.03 | TSM2 | you need two ports of ISDN2E, 1 goes to the NTE (BT wall), the other goes to the versatility system, when the sangoma cards are installed, one port has to be set as NT and the other as TE, that means that one port acts as a phone the other acts like the wall port |
| 21:45.03 | garymc | being asterisk |
| 21:45.44 | garymc | i only get on port |
| 21:45.49 | garymc | i only got one port |
| 21:45.59 | TSM2 | i know, you will have 2 ports on the asterisk system |
| 21:46.17 | TSM2 | one connects to the ISDN2D BT port, the other then connects to the PBX |
| 21:46.27 | TSM2 | think of it as a man in the middle aproach |
| 21:46.37 | TSM2 | ofcourse this will not record internal calls on your PBX |
| 21:46.49 | TSM2 | if you want that then you will need to go fully to asterisk |
| 21:47.59 | garymc | internal calls as extension to extension |
| 21:48.09 | TSM2 | yup correct |
| 21:48.18 | garymc | would all incoming/outgoing calls be recorded? |
| 21:49.02 | TSM2 | if you want, yes or you could do it on a per extention if you wanted |
| 21:53.17 | garymc | hmmm |
| 21:53.26 | garymc | would this be easy to setup do you think? |
| 21:53.48 | garymc | Cos I already have a Office server with linux installed and I have client computers connect to this |
| 21:54.10 | TSM2 | look at full GUI distos of asterisk, trixbox and elastix |
| 21:54.20 | TSM2 | use a sep machine |
| 21:54.32 | TSM2 | you dont want any problems with your calls |
| 21:56.08 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 21:56.10 | TSM2 | ive currently got a small hp dc7900 comp next to my desk with a sangoma PRI card, currently testing, but looks all good |
| 21:56.55 | TSM2 | oh and the reason i like sangoma cards at the mo is that at least the PRI card comes with lifetime warranty |
| 21:57.11 | garymc | whats the echo like on them? |
| 21:57.36 | TSM2 | get the hardware echo cancelation with them |
| 21:58.06 | garymc | how much are they? |
| 21:58.08 | TSM2 | look around, the general word is that sangoma is one of the best out there and you have no compatability issues unlike digium |
| 21:58.10 | *** join/#asterisk stacker (n=petr@ip-85-160-104-89.eurotel.cz) |
| 21:58.29 | drmessano | compatability issues? |
| 21:58.39 | drmessano | What is digium incompatible with? |
| 21:59.09 | [TK]D-Fender | drmAll the old MB issues |
| 21:59.13 | garymc | TsM2: so what sangoma card would i need? |
| 21:59.17 | [TK]D-Fender | drmessano: IRQ fighting, etc |
| 21:59.19 | TSM2 | they have issues with IRQs and ive heard issues with dell servers too, many issues, digium hardware is picky |
| 21:59.21 | garymc | b600? anolougue? |
| 21:59.31 | TSM2 | http://www.voipon.co.uk/sangoma-asterisk-cards-c-23_101.html |
| 21:59.51 | TSM2 | A500 = BRI |
| 22:00.05 | TSM2 | you then have to add modules for the ammount of BRI ports you want |
| 22:00.08 | garymc | Jesus £650 |
| 22:00.51 | TSM2 | yup, but thats prolly fully loaded with 3 ports |
| 22:01.10 | garymc | nope its £227 |
| 22:01.12 | TSM2 | about £515 with a single module and echo cancelation on board |
| 22:01.19 | TSM2 | nop thats just the base card |
| 22:01.20 | garymc | but what the diff between express version? |
| 22:01.37 | TSM2 | PCI Express |
| 22:01.40 | TSM2 | vs PCI |
| 22:01.47 | garymc | yes |
| 22:02.10 | TSM2 | nothin, depends on you mobo, if your computer has PCIe then get the PCIe card |
| 22:02.12 | garymc | what country you in TSM2 |
| 22:02.24 | TSM2 | London |
| 22:02.29 | garymc | ahh cool |
| 22:03.24 | garymc | ok well thanks for your help TSM2 |
| 22:03.43 | TSM2 | i was looking a Swyx for VoIP then the company wanted to save money, i always wanted to play with asterisk but thought they wanted proper branded stuff, but 45k for swyx vs 25k for asterisk when rolled out changed their mind |
| 22:03.50 | TSM2 | ahh hees gone, oh well |
| 22:11.06 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 22:11.36 | *** join/#asterisk sjobeck (n=Adium@208-151-246-17.dq1sn.easystreet.com) |
| 22:12.37 | *** part/#asterisk sjobeck (n=Adium@208-151-246-17.dq1sn.easystreet.com) |
| 22:13.07 | TSM2 | how well does dial up modems work on asterisk |
| 22:13.22 | Qwell | TSM2: for what purpose? |
| 22:15.08 | TSM2 | pass thru, we have a couple computers in our office that currently use analog lines from our legacy PBX system for accessing the bank, but when we move to asterisk i need to make sure they still work |
| 22:15.45 | [TK]D-Fender | DON'T |
| 22:15.49 | Qwell | You can't |
| 22:16.33 | TSM2 | its only over our LAN with local server, so even G711 with no silence supp will it still not work |
| 22:16.52 | TSM2 | what if i use sangoma cards in the server instead of ATAs |
| 22:16.57 | [TK]D-Fender | TSMYou are asking for tragic failure. |
| 22:17.41 | TSM2 | grrr :( |
| 22:17.49 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 22:19.42 | TSM2 | im already having a nightmare with Faxes, but ive been assured that by using sangoma cards for both PRI and FXS that it will work fine because they use a backplane link to keep the cards in sync |
| 22:19.51 | TSM2 | im still in trials |
| 22:20.39 | TSM2 | so i guess ile put an ata on the modem and see if it works, i know the banks communicate at a slow speed, ours are at 9600baud or 14k |
| 22:20.52 | [TK]D-Fender | TSM2: Just don't get it, do you? |
| 22:21.01 | [TK]D-Fender | TSM2: Fax = mode = fail |
| 22:21.09 | [TK]D-Fender | TSM2: ... |
| 22:21.11 | [TK]D-Fender | ~wglwat |
| 22:21.12 | infobot | from memory, wglwat is well, good luck with all that |
| 22:21.12 | TSM2 | even with T38 |
| 22:22.35 | TSM2 | http://wiki.sangoma.com/t1e1analogfaxing |
| 22:33.59 | *** join/#asterisk kb3ien (n=kb3ien@ool-45766a2d.dyn.optonline.net) |
| 22:34.07 | kb3ien | any way to ring 2 extensions at the same time? but allow the secord one to keep ringing even if the first is answered. then let the second one steal the call from the first? |
| 22:34.58 | kb3ien | seems that allowing callpickups does it, but not elegantly. still it doesnt make the phone ring. |
| 22:35.03 | generalhan | [TK]D-Fender: how do you handle your faxing ? |
| 22:35.23 | [TK]D-Fender | generalhan: With a fax machine on a separate line |
| 22:35.27 | generalhan | :( |
| 22:36.29 | kb3ien | iaxmodem allows one to convert a fax signal to the bits that would flow across the modems rs488 line, if you're trying to deal with jitter and delay outside what fax/voip will allow. |
| 22:36.33 | generalhan | i have no choice but to run my faxes through Asterisk ... im in the same boat as TSM2. i'm hoping that the Xorcom device im getting will be, at least, better than the digium based solution im trying now |
| 22:37.21 | kb3ien | i'm hoping to setup a 200 USD box to do raw-realtime-fax over ssh and mitigate that disaster. |
| 22:38.09 | kb3ien | i'd like to have the operator phone keep ringing even after the aa has picked up. |
| 22:38.36 | *** join/#asterisk iewebguy (n=mark@c-75-68-250-121.hsd1.vt.comcast.net) |
| 22:38.42 | kb3ien | the aa is envoked by dial(LOCAL/aa-delayed&SIP/phone); |
| 22:39.03 | kb3ien | aa-delayed starts with Wait(12) to give the operator a chance. |
| 22:39.31 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 22:39.47 | *** join/#asterisk Linuturk (n=linuturk@fluxbuntu/developer/Linuturk) |
| 22:39.53 | kb3ien | i'd like to allow the operator a means to steal the call, even after the AA has kicked in. |
| 22:41.07 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 22:43.34 | TSM2 | generalhan & [TK]D-Fender: look at sangoma, they guarantee fax working when you mix their PRI/BRI/Analog cards together with a cable between to sync the cards clock together |
| 22:43.51 | TSM2 | i know you dont want to beleave it but they do gurantee it will work |
| 22:44.21 | TSM2 | i posted the link above |
| 22:44.30 | TSM2 | <PROTECTED> |
| 22:47.58 | *** join/#asterisk phunyguy (n=phunyguy@h69-130-67-17.kgldga.dsl.dynamic.tds.net) |
| 22:49.07 | *** join/#asterisk Linuturk (n=linuturk@fluxbuntu/developer/Linuturk) |
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