| 00:00.13 | KingDavidNYC | yes, there is |
| 00:00.22 | manxpower | ~answers |
| 00:00.23 | infobot | [answers] Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 00:00.26 | manxpower | read that NAT link |
| 00:00.41 | *** join/#asterisk Dovid (n=annon@213.8.121.90) |
| 00:00.47 | KingDavidNYC | actually, it was working for about 2 years, then someone messed up with the router and external firewall setup |
| 00:00.51 | Dovid | hi. does anyone know what would cause this ? http://pastebin.com/m6187fc16 |
| 00:01.04 | KingDavidNYC | great, I am gonna test that, thank you guys |
| 00:01.24 | djabbour | manxpower, isn't zaptel necessary for the clock for conference call support? |
| 00:01.56 | KingDavidNYC | djabbour: ztdummy is what you need |
| 00:02.23 | manxpower | djabbour: if you want MeetMe then you need zaptel/dahdi. Just hope it works in your VM. |
| 00:02.28 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 00:03.25 | *** join/#asterisk geneticx (n=geneticx@adsl-2-215-240.mia.bellsouth.net) |
| 00:04.10 | djabbour | manxpower, ah... I see. I am reading that zaptel won't compile under xen... so i guess I'm out of luck there |
| 00:05.12 | drmessano | <PROTECTED> |
| 00:05.56 | *** join/#asterisk jack___ (n=jack@ip91358001.speed.planet.nl) |
| 00:06.57 | jack___ | hi |
| 00:06.58 | jack___ | http://nopaste.com/p/agSXpqd6p |
| 00:07.06 | drmessano | There appears to be several sets of RPMs for Dahdi in Xen |
| 00:07.21 | jack___ | can anyone point me to the error in this setup? the AGI script isn't running |
| 00:07.30 | jack___ | atleast, i see no debug what so ever |
| 00:09.00 | p3nguin_ | erieslab233: You could configure a local extension to ring the international number. exten => 7765,1,Dial(SIP/01165612342153@outbound) for example. You could use Authenticate() before it to secure it with a PIN if you want. |
| 00:10.01 | p3nguin_ | erieslab233: Or you could get the sister in Thailand to use a softphone and register to your PBX, then just dial her extension. No phone minutes are used and there would be no charge for it. |
| 00:10.12 | djabbour | drmessano, ah you're right. Looks like Dahdi and 1.6 will make this easier under Xen |
| 00:10.28 | djabbour | drmessano, i was going to stick with an older release for stability reasons, but i guess i can live on the bleeding edge ;-) |
| 00:10.31 | drmessano | Yes, the future is now |
| 00:10.38 | drmessano | 1.6 isnt bleeding edge |
| 00:10.59 | drmessano | Its bleeding old.. We're nearly 3 branches into 1.6.x already |
| 00:10.59 | djabbour | drmessano, well i had my heart set on 1.4.21 |
| 00:11.17 | drmessano | 1.4.21 is old as dirt on an ancient release |
| 00:11.21 | phix | hey hey |
| 00:11.25 | phix | Morning |
| 00:11.29 | Dovid | morning |
| 00:11.42 | drmessano | I would say 1.6.1.x branch is the best to go with right now |
| 00:11.50 | Chainsaw | agrees |
| 00:11.57 | drmessano | 1.6.2.x is "bleeding edge" as "bleeding edge" as an RC can be |
| 00:12.20 | Chainsaw | Yeah, if you don't want excitement... too early. |
| 00:12.36 | drmessano | IMO, 1.4 was never really stable.. So i dont know where somehow 1.4 got to be the new 1.2 |
| 00:13.05 | drmessano | 1.2 was a good fallback with 1.4s issues, but 1.4 is best skipped over now for the much improved 1.6.x |
| 00:15.14 | drmessano | 1.4 went from "unstable, buggy piece of crap" to "safety net" before the body of 1.2 was even cold.. it shows you how much piece get hung up on some idea of safe harbor for releases |
| 00:16.21 | drmessano | I think Digium should release 1.8 TOMORROW and see who dumps 1.4 for 1.6 overnight because "its the new old stable" |
| 00:16.39 | drmessano | </rant> |
| 00:20.46 | *** join/#asterisk Katty (n=Angela@70.253.172.140) |
| 00:20.52 | Katty | hi |
| 00:21.03 | Chainsaw | wb Katty |
| 00:21.51 | Katty | thanks (= |
| 00:22.03 | jblack | wtf? Michael Jackson did a rap? http://www.youtube.com/watch?v=Yu_moia-oVI |
| 00:22.36 | *** join/#asterisk Deeewayne (n=dwayne@c-71-228-179-90.hsd1.al.comcast.net) |
| 00:22.36 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 00:23.26 | Katty | hi Deeewayne |
| 00:23.31 | Deeewayne | hugs Katty |
| 00:24.17 | *** join/#asterisk voipmonk (n=voipmonk@66.49.238.52) |
| 00:25.43 | *** join/#asterisk trnzmeta (n=bleh@secure27.lnk.telstra.net) |
| 00:38.05 | drmessano | If you BING for Asterisk, you should be Pounded. |
| 00:38.25 | erieslab233 | Can one DISA have multiple PIN's? For example I want all my staff to use one DID to use as a DISA with each employee having their own PIN. |
| 00:38.28 | jblack | is that still running? |
| 00:39.04 | drmessano | Sadly |
| 00:39.21 | drmessano | But you know the cool thing |
| 00:39.31 | drmessano | Not that M$ has an unfair advantage or anything |
| 00:39.33 | drmessano | But.. |
| 00:39.33 | jblack | That's interesting. Search for "linux better than windows" at both google and bing |
| 00:39.44 | erieslab233 | the pretty hotspots on bing? lol |
| 00:39.45 | jblack | ok, not really interesting. |
| 00:39.59 | jblack | Katty: have you ever heard of fried cucumbers? |
| 00:40.03 | drmessano | If the keys for your search providers get corrupt in the registry, IE will warn you: |
| 00:40.10 | Katty | jblack: yes. |
| 00:40.20 | jblack | Good thing, or bad thing? |
| 00:40.20 | Katty | jblack: and cucumber chips too |
| 00:40.29 | Katty | jblack: mmmm...i like it sauted, but not deepfried. |
| 00:40.32 | drmessano | "There is a problem with your list of search providers. Your default search provider has been reset to Bing" |
| 00:40.38 | erieslab233 | pickle or cucumber? |
| 00:40.40 | drmessano | Really now? Why not google? |
| 00:41.10 | jblack | Oh, I meant sauteed anyways. |
| 00:41.47 | jblack | I'm gonna try that. |
| 00:42.21 | Katty | i'd cook it with onion and squash too |
| 00:42.24 | Katty | and ham. |
| 00:43.08 | drmessano | Ubuntu user driving down a narrow road, has a small fenderbender with a Windows user.. so the Ubuntu guy gets out, walks to the side of the car with the damage, and finds an irate windows user. Ubuntu guy looks at the damage and and says to the Windows guy "Hey man, chill out. It's just a little BING" |
| 00:43.43 | drmessano | Rimshot? |
| 00:43.51 | drmessano | :( |
| 00:45.52 | Katty | wonders what to do for the next couple hours |
| 00:47.17 | drmessano | http://www.bing.com/search?q=what+to+do+for+the+next+few+hours&go=&form=QBLH&qs=n |
| 00:47.52 | Dovid | anyone see this ? |
| 00:47.52 | Dovid | http://blog.voipsupply.com/aastra-6739i-2 |
| 00:48.07 | drmessano | Apparently you should check up on your contraceptives before going to Gaza |
| 00:48.16 | Dovid | lol |
| 00:48.37 | Dovid | i like how they advertise that they dont give usless crap and here they do. great job M$ |
| 00:50.29 | drmessano | Yeah, the search results remind me of the very commercials they use to sell their damn search |
| 00:50.53 | *** join/#asterisk chendy (n=chatzill@113.91.37.203) |
| 00:51.20 | Dovid | lol |
| 00:51.21 | drmessano | If I didn't actually like Windows 7, I would start a "Why settle for 7, upgrade to 9.10 now!" campaign |
| 00:51.26 | geneticx | I know this is offtopic but does anyone know of a good trustworthy company that provides VPN services? |
| 00:51.37 | drmessano | Cisco |
| 00:51.40 | drmessano | guffaws |
| 00:51.40 | wastrel | some people like ubuntu |
| 00:51.41 | Dovid | i am off to bed. goto be up in 4 hours. |
| 00:51.59 | drmessano | later |
| 00:54.04 | drmessano | I've really liked Karmic.. enough to switch from XP on my desktop and laptop |
| 00:54.43 | *** part/#asterisk ruben23 (n=RPL@122.55.48.243) |
| 00:55.32 | *** join/#asterisk mumtazah1 (n=mumtazah@203.82.79.104) |
| 00:57.38 | erieslab233 | Can I use the same DID for multiple DISA's if I use a different PIN for each DISA? |
| 00:58.46 | erieslab233 | Can a DISA have a caller ID? |
| 00:59.05 | wastrel | i want to upgrade to koala at work but i don't have time |
| 01:00.10 | erieslab233 | ok I will try but if I break it I warn you all that I will only blame myself and bathe in my own self pity for minutes on end! |
| 01:00.22 | erieslab233 | maybe even seconds |
| 01:01.19 | *** join/#asterisk Kumbang (n=kumbang@167.205.24.69) |
| 01:03.58 | voipmonk | callerid auth , too, erieslab233 |
| 01:04.39 | *** join/#asterisk tribex (n=tribal@190.167.60.74) |
| 01:10.05 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 01:13.37 | *** join/#asterisk MichaelGG (n=Michael@190.148.164.198) |
| 01:13.54 | MichaelGG | What kinds of things would cause Asterisk to ignore a re-invite (from a remote session timer) |
| 01:15.02 | ska | What rtp ports does callcentric require? |
| 01:16.37 | p3nguin_ | ska: http://www.callcentric.com/faq/9#226 |
| 01:16.56 | ChannelZ | probably whatever ones your phone asks for |
| 01:17.28 | p3nguin_ | You need two ports per call, so do the math. |
| 01:20.28 | *** join/#asterisk DavidR2008 (n=chatzill@71.0.16.133) |
| 01:22.10 | *** join/#asterisk dmz (n=dmz@h96-61-156-43.mtjltn.dsl.dynamic.tds.net) |
| 01:26.00 | *** join/#asterisk voipmonk (n=voipmonk@66.49.238.52) |
| 01:27.33 | *** join/#asterisk natlonehat (n=natloneh@202.170.42.67) |
| 01:36.39 | *** join/#asterisk sahafeez (n=sahafeez@68.167.234.130) |
| 01:37.21 | *** join/#asterisk ming_zym (n=ming_zym@121.0.29.237) |
| 01:55.29 | *** part/#asterisk MichaelGG (n=Michael@190.148.164.198) |
| 01:56.24 | erieslab233 | omg I can make the caller id say anything i want???!! |
| 01:56.47 | erieslab233 | I am calling papa johns from the fire department!!! |
| 01:56.50 | loather-work | pretty much, yes |
| 01:56.51 | loather-work | heh |
| 01:57.00 | erieslab233 | I am so gonna love prison! |
| 01:57.08 | loather-work | depends on what your telco is setup to accept |
| 01:57.14 | loather-work | if you're on an analog line, you can't set the CID |
| 01:57.18 | loather-work | (generally) |
| 01:57.28 | erieslab233 | teliax and vitelity |
| 01:57.38 | erieslab233 | only one let me change the caller id from asterisk |
| 01:57.44 | erieslab233 | the other is from their cp |
| 01:57.46 | loather-work | with teliax you can set it to pretty much whatever you want. |
| 01:57.53 | loather-work | as long as it's not a tollfree number |
| 01:58.11 | erieslab233 | no no it did with my toll free from ringcentral |
| 01:58.24 | erieslab233 | apparently new rules |
| 01:58.31 | p3nguin_ | erieslab233: Which one forces you to set it in the cp? |
| 01:58.44 | erieslab233 | It must have been vitelity? |
| 01:58.48 | mumtazah1 | hello, can i ask something? Does asterisk voip work with dhcp ip (not static ip) |
| 01:58.53 | erieslab233 | I think my primary trunk out is teliax |
| 01:59.07 | erieslab233 | yes mum |
| 01:59.13 | loather-work | really... interesting. i'll have to fiddle with teliax some more then and see if it'll let me set my 800 number outbound. |
| 01:59.23 | erieslab233 | but only well if all your phones are on the same network |
| 01:59.25 | loather-work | that was a problem last year.. |
| 01:59.35 | p3nguin_ | erieslab233: That's odd. I use VoIP.ms (a Vitelity reseller), and I set my CID in Asterisk. |
| 01:59.50 | erieslab233 | dude I kid you not my 877 number is showing on all cell phones I have dialed |
| 02:00.08 | loather-work | queue show |
| 02:00.09 | loather-work | er. |
| 02:00.10 | p3nguin_ | It's supposed to if you set CALLERID(). |
| 02:00.15 | erieslab233 | perhaps it is both then working in asterisk? |
| 02:00.35 | erieslab233 | only number though no alpha |
| 02:00.51 | p3nguin_ | erieslab233: CNAM lookup is done on the receiving end anyway. |
| 02:00.54 | erieslab233 | on a landline I think it displays alpha as 'toll free' |
| 02:01.06 | loather-work | CNAM is evil. |
| 02:01.13 | p3nguin_ | erieslab233: If that's what the receiving phone company wants to put on it, then it is. |
| 02:01.26 | erieslab233 | none of the providers can write to that database yet? |
| 02:01.38 | erieslab233 | I figured vonage would have made a big enough push |
| 02:01.52 | loather-work | only the issuing authority for the number in question can write a record for that number into CNAM. |
| 02:01.59 | p3nguin_ | They can submit it to "a" database, but there's nothing ensuring that the telco you are calling looks in that database. |
| 02:02.05 | erieslab233 | right |
| 02:02.08 | erieslab233 | sucks |
| 02:02.22 | loather-work | you'll have to call the provider that issued you the number and see if first, they have CNAM write access, and second, if they'll change the database for you |
| 02:02.48 | erieslab233 | fat chance I'm sure |
| 02:02.48 | p3nguin_ | No, you must call the receiving phone company. |
| 02:02.56 | erieslab233 | for real???! |
| 02:03.07 | loather-work | CNAM lookups are done via SS7 now. |
| 02:03.12 | erieslab233 | like all the little a$$ providers in the middle of montana??? |
| 02:03.30 | loather-work | if they're hooked into the central SS7 system, they'll get centrallized CNAM access. |
| 02:03.53 | erieslab233 | how many telco's are there? that couldn't just be done nationwide because they choose which database to reference? |
| 02:06.18 | *** join/#asterisk dmz (n=dmz@h96-61-156-43.mtjltn.dsl.dynamic.tds.net) |
| 02:07.03 | *** join/#asterisk maxagaz (n=maxagaz@124.205.74.34) |
| 02:09.32 | dlynes | mumtazah1: yes...it all depends on your communications architecture |
| 02:10.08 | loather-work | Can you do complex logic expressions, e.g. $[ "${VAR1}" = "something" | "${VAR1}" = "somethingelse" ] ? |
| 02:15.34 | erieslab233 | Does anyone know if it is possible to use a single DID for multiple DISA logins/PIN's? |
| 02:25.09 | *** join/#asterisk mchou (n=quassel@unaffiliated/mchou) |
| 02:26.11 | Katty | named her little facebook restaurant Thai Me Up |
| 02:27.07 | *** join/#asterisk QaDeS_ (n=mklaus@p4FC72C9D.dip0.t-ipconnect.de) |
| 02:28.39 | *** join/#asterisk manxpower (n=ewieling@24.42.221.26) |
| 02:29.45 | *** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com) |
| 02:35.53 | *** join/#asterisk OrNix (n=ornix@91.151.249.47) |
| 02:36.51 | ska | chan_sip.c:14079 in handle_request_invite: Call from '' to extension '1777XXXXXXX' rejected because extension not found. |
| 02:37.18 | ska | I cant find the problem.. please advise. |
| 02:38.02 | *** join/#asterisk RB2 (n=dayres@pool-71-172-130-214.nwrknj.east.verizon.net) |
| 02:38.11 | RB2 | ~book |
| 02:38.12 | 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 |
| 02:38.18 | Katty | i advise you to eat a banana. |
| 02:38.52 | ska | DO I need to match the incoming extension as well as local one? |
| 02:40.58 | *** join/#asterisk EnrageC (n=natloneh@202.170.42.67) |
| 02:41.23 | manxpower | ska: you do not have a 1777XXXXXXX extension defined IN THE CONTEXT the call is coming into |
| 02:41.39 | manxpower | ska: there is no such thing as an incoming extension |
| 02:41.59 | manxpower | ska: are you trying to match the whole number or do you have a pattern match? |
| 02:46.29 | *** join/#asterisk baijum (n=baiju@122.167.75.218) |
| 02:48.33 | erieslab233 | I have a very basic question...When I have calls forwarded to my blackberry I have no way to transfer correct? If correct is there another way to 'forward' so that I can stay within the system to allow me to transfer calls if needed? |
| 02:48.47 | erieslab233 | like a reverse DISA? |
| 02:49.04 | *** join/#asterisk mumtazah1 (n=mumtazah@203.82.79.103) |
| 02:49.12 | *** join/#asterisk geneticx (n=geneticx@adsl-2-215-240.mia.bellsouth.net) |
| 02:49.33 | manxpower | erieslab233: technically they are not "forwarded". It's just another destination to Asterisk. If you are on a phone that is too stupid to be able to transfer (cell phones for example) you an use the T or t option to Dial to allow DTMF based transfers. |
| 02:50.19 | mumtazah1 | hello, can i ask something? can we manage phone call through web? Any recommendation? |
| 02:50.24 | manxpower | don't use both. one allows the call-ER to transfer, one allows the call-EE to transfer. "core show application dial" for more information. caller/callee is all from the point of view of Asterisk |
| 02:51.24 | erieslab233 | since the phones in buildings are 7960G's that would mean that this option is likely disabled by default on the asterisk server correct? |
| 02:51.48 | erieslab233 | by enabling this would it 'damage' my transfer buttons on the cisco phones? |
| 02:53.48 | [TK]D-Fender | erieslab233: it has nothing to do with SIP transfer functionality |
| 02:54.25 | erieslab233 | ok |
| 02:54.34 | erieslab233 | will go looking now thanks |
| 02:55.55 | p3nguin_ | erieslab233: There is no such thing as "default." Either you configured it or you didn't. |
| 02:56.42 | erieslab233 | I assumed with the various distros some possibly had settings preconfigured for people like me |
| 02:57.08 | erieslab233 | that being said p3nguin you just made me feel like a voip guru, cause I had no idea what I was doing! |
| 02:57.39 | erieslab233 | still don't and yet it works lol |
| 02:57.39 | ska | manxpower: Im not exactly sure because Im using askozia to the contexts are a bit hidden. |
| 02:57.52 | p3nguin_ | erieslab233: The preconfigured extensions are usually useless since they don't know what phone numbers you'll be using. |
| 02:57.56 | [TK]D-Fender | skawell GUI's are NOT supported her |
| 02:57.58 | [TK]D-Fender | here |
| 02:58.06 | Kobaz | LOCK and TRYLOCK seem to be backwards in functionality |
| 02:58.08 | [TK]D-Fender | ska: well GUI's are NOT supported here |
| 02:58.09 | [TK]D-Fender | gah |
| 02:58.20 | Kobaz | the help says LOCK will try 3 times, TRYLOCK, will say it will just try and lock it |
| 02:58.20 | ska | manxpower: because it rings right after i boot, but not on 2nd try. |
| 02:59.25 | p3nguin_ | erieslab233: And you don't need to bother with t or T Dial options if you just have Cisco phones with the transfer softkey on them. |
| 02:59.40 | [TK]D-Fender | p3nguin_: which is not the case |
| 03:00.01 | Kobaz | so the real question is... should i use LOCK, or TRYLOCK |
| 03:00.08 | [TK]D-Fender | Kobaz: YES |
| 03:00.15 | p3nguin_ | erieslab233: But for a phone without it, you'll want to study the information that manxpower directed you to so that you can determine where to implement DTMF transferring. |
| 03:00.24 | erieslab233 | ok :) |
| 03:00.25 | Kobaz | [TK]D-Fender: it's an a/b question :P |
| 03:00.49 | dlynes | Does 1.6.1.1 (as a client) have issues registering with 1.4.26.2 (as a server)? |
| 03:01.00 | p3nguin_ | doubtful |
| 03:01.10 | dlynes | It comes up immediately as unreachable, but if I have the wrong password, it tells me about it right away |
| 03:01.35 | p3nguin_ | Since SIP and IAX2 work in both versions, where would the incompatibility lie? |
| 03:02.22 | dlynes | Well, fwiw, 1.4.22 will not talk iax2 to a 1.6.1.1 box (at all) |
| 03:02.35 | p3nguin_ | Oh I bet it will. |
| 03:02.38 | dlynes | or 1.4.22rc5 for that matter...it doesn't seem to work until 1.4.26 |
| 03:03.06 | p3nguin_ | I would have to see the bugs and changelogs to know for sure if IAX2 was broken back then. |
| 03:03.08 | dlynes | p3nguin_: sorry...let me clarify....using t.38 |
| 03:03.17 | drmessano | .... |
| 03:03.17 | p3nguin_ | hmm |
| 03:03.29 | loather-work | isn't there a byte swap issue between those versions? |
| 03:03.40 | loather-work | i remember glossing over something in one of the UPGRADE or README files. |
| 03:03.42 | dlynes | p3nguin_: using t.38, 1.6.1.1 refuses a call from a 1.4.22 or 1.4.22rc5 box |
| 03:04.00 | dlynes | p3nguin_: however, the issue I'm against right now has nothing to do with t.38 |
| 03:04.26 | dlynes | I'm having issues getting a 1.6.1.1 box to register to a 1.4.26.2 box |
| 03:04.31 | p3nguin_ | Are you using a register statement on the client box? |
| 03:04.37 | dlynes | the 1.4.26.2 is not behind a firewall |
| 03:04.40 | dlynes | p3nguin_: yes |
| 03:04.46 | dlynes | the 1.6.1.1 box is behind a router |
| 03:05.16 | drmessano | Why are you using such old versions of every combination imaginable? |
| 03:05.19 | p3nguin_ | I would start by examining the register statements and contexts on both sides. |
| 03:05.28 | drmessano | "I have 1.6.7.. and 1.2.5" |
| 03:05.35 | drmessano | errr |
| 03:05.35 | dlynes | drmessano: ? 1.4.26.2 is not old...it's the latest release |
| 03:05.38 | drmessano | Left out a 1 there |
| 03:05.45 | drmessano | 1.6.1.1 box to register to a 1.4.26.2 box |
| 03:06.00 | dlynes | drmessano: and 1.6.1.1 isn't the latest release, but does it really matter? |
| 03:06.02 | drmessano | or the 1.4.22 boxes |
| 03:06.14 | drmessano | Actually, it does when it comes to tracking down bugs |
| 03:06.24 | *** join/#asterisk evil_gordita (n=evilgord@ip70-188-50-186.rn.hr.cox.net) |
| 03:06.31 | dlynes | drmessano: 1.4.22 is not part of the current problem...1.4.22 is a known problem, and it's one that 1.4.26 fixed |
| 03:06.46 | Katty | naptime for me folks. later |
| 03:06.58 | erieslab233 | is the 'core show application dial' found with the exit dial application? |
| 03:07.07 | dlynes | woah...katty was still up? thought she was a morning person |
| 03:07.10 | p3nguin_ | uh, what? |
| 03:07.22 | p3nguin_ | erieslab233: On the console, core show application dial |
| 03:07.23 | erieslab233 | doing my best with google mate ;) |
| 03:07.28 | erieslab233 | ohh lol |
| 03:07.30 | erieslab233 | sorry |
| 03:07.40 | dlynes | drmessano: lemme upgrade to 1.6.1.8 then...see if that fixes the problem |
| 03:07.54 | dlynes | drmessano: but the 1.4 box is already at the latest 1.4 version |
| 03:08.23 | drmessano | I'm well aware of that.. but all your scenarios seem to involve some version on one side thats is OLD |
| 03:08.24 | dlynes | drmessano: and i can't upgrade it to 1.6, because that'll break compatibility with the box that some idiot decided to install elastix onto |
| 03:08.31 | drmessano | Not sure how thats a puzzle |
| 03:08.55 | erieslab233 | would that T or t enable all of the functionality as if I were at my SIP phone even though I am on a dumb phone? |
| 03:09.08 | erieslab233 | the features I mean |
| 03:09.11 | erieslab233 | feature codes |
| 03:09.22 | drmessano | You stand a better chance of having less issues if you're at least using the most current of a specific branch, especially if a change was made to IAX2 in general |
| 03:09.33 | p3nguin_ | erieslab233: t allows for the called person to transfer, T allows for the caller to transfer. |
| 03:09.44 | dlynes | drmessano: you mean like 1.4 branch or 1.6 branch? |
| 03:09.48 | erieslab233 | ok |
| 03:09.53 | erieslab233 | thanks again |
| 03:09.55 | dlynes | drmessano: not the latest stable release of either? |
| 03:10.06 | drmessano | Thats not what I said |
| 03:10.14 | drmessano | This isnt rocket science |
| 03:10.16 | erieslab233 | I really am trying guys not meaning to spam with noob questions |
| 03:10.24 | dlynes | drmessano: one side is already using the latest 1.4 series |
| 03:10.33 | drmessano | The other side IS FUCKING NOT |
| 03:10.40 | dlynes | drmessano: I just need to upgrade the other side from 1.6.1.1 to 1.6.1.8 |
| 03:10.50 | drmessano | [23:10] <drmessano> You stand a better chance of having less issues if you're at least using the most current of a specific branch, especially if a change was made to IAX2 in general |
| 03:10.52 | dlynes | drmessano: look up the screen...you'll see I already started downloading it |
| 03:11.07 | drmessano | Look up the screen and re-read what I said |
| 03:11.45 | dlynes | drmessano: yes, but you used the term 'branch'...so I was just double checking that you didn't mean branch with respect to subversion |
| 03:30.23 | drmessano | That doesnt make any sense.. What does subversion have to do with the use of the term "branch" |
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| 03:35.49 | drmessano | ~bitchslap [TK]D-Fender |
| 03:35.50 | infobot | ACTION beats the sh*t out of [TK]D-Fender |
| 03:35.50 | drmessano | w00000t |
| 03:35.50 | drmessano | ~slap [TK]D-Fender |
| 03:35.51 | infobot | ACTION slaps [TK]D-Fender, keep your grubby fingers to yourself! |
| 03:35.51 | drmessano | ~gah |
| 03:35.52 | infobot | extra, extra, read all about it, gah is one of the top favourite words |
| 03:35.52 | drmessano | ~pie |
| 03:35.53 | infobot | methinks pie is NOT a mathematical term. It's a food. or about 8.539734222677. |
| 03:35.53 | drmessano | Ah |
| 03:35.53 | drmessano | waits for the queue to flush |
| 03:35.53 | mchou | this logitech sepeaker is some lame ass shit |
| 03:35.53 | drmessano | I am all out of soy milk and I want some Abuelita |
| 03:35.53 | mchou | cant trust newegg revies |
| 03:35.53 | mchou | reviews* |
| 03:35.54 | drmessano | Logitech is lame ass shit, period |
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| 03:35.55 | drmessano | Last few logitech purchases have been crap |
| 03:35.55 | [TK]D-Fender | mchou: M-Audio Studiophile AV-30's monitors here..... very nice with the home studio... |
| 03:36.07 | mchou | lol |
| 03:36.07 | drmessano | would prefer some 24 inch JBLs |
| 03:36.07 | mchou | [TK]D-Fender: too rich for my blood |
| 03:36.16 | [TK]D-Fender | mchou: Not that bad really... $100 |
| 03:36.43 | mchou | $100 each or pair? |
| 03:36.51 | [TK]D-Fender | mchou: the pair |
| 03:37.02 | mchou | that's a good deal |
| 03:37.05 | [TK]D-Fender | mchou: And they get very good reviews |
| 03:37.13 | [TK]D-Fender | mchou: And pretty much deserve it... |
| 03:37.30 | [TK]D-Fender | mchou: I run them off my Behring mixer which my comp feeds into. |
| 03:37.36 | [TK]D-Fender | Behringer* |
| 03:39.30 | Kobaz | ah okay |
| 03:39.38 | drmessano | Behringer makes some of the nicer 2 to 12 channel mixers |
| 03:40.15 | Kobaz | so the documentation is right in asterisk... trylock just runs pthread_mutex_trylock(), which will do a non-blocking attempt at a lock... and then pthread_mutex_lock() will do a blocking attempt |
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| 03:42.44 | Kobaz | i wind up with deadlocked channels though |
| 03:42.48 | [TK]D-Fender | drmessano: I'm running a MX-1604A right now |
| 03:42.54 | Kobaz | even though i'm using LOCK() which is supposed to give up after 3 seconds |
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| 03:58.31 | mnemonix | hi, does anyone know of a way to get a physical listing of network site codes for a particular switch |
| 03:58.40 | mnemonix | a physical location |
| 03:58.55 | mnemonix | is such information published |
| 03:59.34 | drmessano | Which switches are you referring? |
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| 04:01.12 | mnemonix | drmessano, various switches |
| 04:01.16 | mnemonix | owned by a telco |
| 04:01.33 | mnemonix | i need to go pour thermite on them |
| 04:02.10 | mnemonix | assholes dont get the message: stop selling your services to telefraudsters |
| 04:03.09 | mnemonix | haha |
| 04:03.19 | mnemonix | which would be stupid. the fraudster call center could be anywhere. |
| 04:03.34 | mnemonix | would need to trace the route |
| 04:04.02 | mnemonix | would be a nice thing to know how to do on ss7 |
| 04:04.17 | mnemonix | probably something the police can only access |
| 04:04.41 | mnemonix | the switch routing tables |
| 04:11.33 | drmessano | So you're a terrorist then? |
| 04:12.13 | manxpower | dslreports used to have a way to look up central offices, the LERG has that info. The LERG is expensive. |
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| 04:29.22 | OrNix | tell me please, is it able to reload only say.conf without full reload ? |
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| 04:53.30 | ChannelZ | OrNix: module reload app_playback |
| 04:53.57 | OrNix | ChannelZ, thank you |
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| 05:03.10 | TJNII | waves goodbye |
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| 07:45.05 | joobie | guys i have an asterisk box on a static ip where all my sip phones auth and make calls from.. i've another remote site, which has a dynamic ip, where i can setup asterisk also.. I want to somehow get the asterisk where the static ip is, and have it route calls through the asterisk box at the dynamic ip site.. reason being is from the site with the dynamic ip, i can make free local calls - only catch is it has to come from the dynamic ip.. |
| 07:45.05 | joobie | <PROTECTED> |
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| 07:46.26 | Pikoro | can anyone point to any kind of docs dealing with inbound japanese CID? |
| 07:46.43 | Pikoro | been googling for a week and found nothing |
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| 08:02.34 | DND | hi guys, i have a AEX800 card. i want to know what if the ISP is down? will it still be functional at least as a local extension calls? |
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| 08:21.25 | mchou | joobie: both of these asterisk boxes are under your control? |
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| 08:23.27 | joobie | mchou, yup |
| 08:23.48 | mchou | joobie: there are at least two options |
| 08:23.48 | joobie | mchou, because the asteris box at the dynamic ip site changes though, i think it would need to establish a connection to the static ip site? |
| 08:24.06 | joobie | either that or i'd need to customize something so it updates the hosts file on the static-ip asterisk box |
| 08:24.12 | joobie | mchou, cool :P |
| 08:24.52 | mchou | joobie: in the ideal world, would you want to run both * boxes or just one box? |
| 08:25.40 | joobie | i need to run both |
| 08:25.54 | joobie | the problem is the * box at the dynamic ip, which has the free local calls, is not stable |
| 08:26.16 | joobie | so i want the static ip * box to take the call request from the phones and try to send it to the dynamic ip asterisk if it's available, if not then route out another way |
| 08:26.24 | mchou | if you need to run both, you can trun two asterisk boxes together via IAX |
| 08:26.30 | mchou | trunk* |
| 08:26.38 | mchou | http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/I_sect14_tt670.html |
| 08:26.45 | joobie | ahh i read about that jazz.. IAX is asterisk only protocol ya? |
| 08:27.05 | mchou | both asterisks will inherit each others dialplan |
| 08:27.27 | mchou | the dynamic ip address isnt much of a problem |
| 08:27.47 | mchou | go get a free account at dyndns.org |
| 08:28.24 | mchou | that way you can always resolve the dynamic ip address by name (regardless of it's actual ipaddress) |
| 08:28.55 | mchou | now lets pretend you just want to run one * box |
| 08:29.04 | mchou | (say at the remote site) |
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| 08:29.19 | joobie | is there some heartbeat with IAX though? like say the dynamic ip * box drops its uplink.. will the static-ip * box know this instantly and not try to route a call through it? |
| 08:29.30 | mchou | you can also have your sip clients register with that * box |
| 08:29.53 | joobie | mchou, sorry to clarify |
| 08:30.06 | joobie | i dont *need* to run an * box at the dynamic-ip site |
| 08:30.18 | mchou | no, I'm using that as an example |
| 08:30.29 | mchou | only one asterisk box |
| 08:30.31 | joobie | i just figured I need to because in order to make free local calls from the dynamic-ip site, the call must originate from the dynamic-ip site and go to a specific sip provider |
| 08:30.37 | mchou | less admin overhead, etc |
| 08:30.45 | *** join/#asterisk larin (n=larin@mail.cs-service.by) |
| 08:30.48 | joobie | yea - i would actually prefer to run one * box if possible |
| 08:31.10 | mchou | so that's what I asked you in the first place |
| 08:31.16 | joobie | yea my bad :) |
| 08:31.29 | mchou | I dont understand why you cant make free calls from the static ip |
| 08:31.30 | joobie | sry.. presumed too much ;P |
| 08:31.37 | joobie | ahh |
| 08:31.39 | joobie | well |
| 08:32.05 | joobie | i have to use a specific sip server to make the free calls, and that sip server will only accept calls originating from the dynamic-ip |
| 08:32.14 | joobie | and it must also go to a specific sip server |
| 08:32.25 | *** part/#asterisk larin (n=larin@mail.cs-service.by) |
| 08:32.28 | joobie | im starting to think NAT - just not sure if it'll play nice with this whole SIP setup |
| 08:32.38 | mchou | is this sip server "public?" |
| 08:32.55 | mchou | nat is not an issue with asterisk |
| 08:33.18 | joobie | hmm.. public in the sense that it can only be accessed if you are on the ISP that offers it |
| 08:33.24 | mchou | because half of asterisk hangs outside the nat anyways |
| 08:33.31 | joobie | if you are on another ISP you can't connect to their SIP server |
| 08:34.02 | mchou | gotcha on the ISP |
| 08:34.31 | mchou | well, you could always use VPN :) |
| 08:35.04 | mchou | that would make it seem it's coming from within that ISP |
| 08:35.11 | joobie | yea |
| 08:35.14 | joobie | would prefer to avoid VPN |
| 08:35.19 | joobie | can it be done with NAT? |
| 08:35.36 | mchou | no |
| 08:35.49 | joobie | how come? |
| 08:35.57 | mchou | nat only applies to LANs, not to wans |
| 08:36.13 | joobie | I can do NAT on my WAN |
| 08:36.38 | mchou | nat is not gonna rewrite IPs on the WAN |
| 08:36.41 | joobie | ie. take the connection coming into my dynamic IP and translate the SRC to appear like the dynamic ip and then change the DST to the public sip |
| 08:36.51 | joobie | sec mchou fone |
| 08:39.27 | *** join/#asterisk TSM (n=the_soft@87-194-32-212.bethere.co.uk) |
| 08:42.26 | joobie | sry |
| 08:42.27 | joobie | back |
| 08:42.34 | joobie | srill alive mchou ? |
| 08:42.39 | joobie | -srill +still |
| 08:42.48 | mchou | yeah |
| 08:42.51 | joobie | with the NAT - I can do the rewriting fine with iptables |
| 08:42.57 | mchou | anyway forget about that nat |
| 08:43.02 | joobie | my concern is that there will be headers in the packets that asterisk forms |
| 08:43.09 | joobie | which might cause a problem |
| 08:43.26 | joobie | how come?:P I really really like it ;P |
| 08:43.32 | mchou | nat aint goona work nowhere nohow |
| 08:43.32 | joobie | dont have to bother with a local asterisk box |
| 08:43.48 | joobie | whyyyy ?:P seirously, i can rewrite the src and dst of that packet |
| 08:43.53 | mchou | that's my point |
| 08:43.55 | joobie | and have the linux box bounce it |
| 08:44.05 | mchou | you dont even need another box |
| 08:44.10 | joobie | ok |
| 08:44.15 | joobie | what's the next option |
| 08:44.19 | joobie | apart from NAT with one box? |
| 08:44.30 | mchou | just register all you sip devices to that (remote) * box an be done with it |
| 08:44.50 | mchou | you're making this needlessly complicated |
| 08:44.51 | joobie | the remote asterisk box needs to be on the static ip |
| 08:45.04 | mchou | wtf are you talking about?? |
| 08:45.22 | mchou | I just told you to get an account at dyndns.org |
| 08:45.31 | joobie | yea |
| 08:45.45 | joobie | but mchou the problem is the dynamic ip box is a DSL connection - it's not stable |
| 08:46.02 | joobie | the static IP box is sitting in the datacenter and hasn't dropped for years |
| 08:46.05 | mchou | well, too bad |
| 08:46.18 | joobie | i also have ISDN tails in the datacenter, so if the voip links fail, they can fallback to the ISDN |
| 08:46.27 | *** join/#asterisk oej (n=olle@static-213-115-251-100.sme.bredbandsbolaget.se) |
| 08:46.27 | joobie | so i need the phones to terminate in the static ip * |
| 08:46.28 | mchou | no ISP is gonna be 100% reliable |
| 08:46.49 | joobie | mchou, i just want to use the dynamic ip site as a "good to have" |
| 08:46.51 | joobie | it's not reliable |
| 08:47.36 | mchou | I told you how to achieve this already |
| 08:47.41 | mchou | IAX trunking |
| 08:47.46 | mchou | get over it |
| 08:47.53 | joobie | mchou, that reuqires two asterisk boxes |
| 08:48.01 | joobie | you got me tempted on the 1 asterisk box solution |
| 08:48.09 | joobie | and I think NAT is the solution.. i duno why yu are against it |
| 08:48.19 | mchou | because it wont work |
| 08:48.28 | mchou | stop smoking crack |
| 08:48.46 | joobie | mchou, i've done this exact type of NAT before |
| 08:49.06 | mchou | joobie: thenit's clear you dont need any help from me |
| 08:49.13 | mchou | then it's* |
| 08:49.32 | mchou | fucking no good deed goes unpunished |
| 08:49.42 | joobie | dood |
| 08:49.50 | joobie | you came up with some good ideas |
| 08:49.56 | joobie | relax.. this NAT idea is yrosu i think |
| 08:50.01 | joobie | it's the winner |
| 08:50.14 | joobie | iptables can do some complex NAT |
| 08:50.17 | mchou | yea whatever |
| 08:50.20 | joobie | it's not a netgear |
| 08:50.39 | mchou | you're a retard |
| 08:51.00 | joobie | because you think you can't NAT out the same WAN int? |
| 08:51.25 | mchou | how the hell is you ISP gonna route the packet you falsified? |
| 08:51.42 | mchou | your* |
| 08:51.44 | mchou | freaking lamer |
| 08:52.02 | mchou | it's gonna stop DEAD |
| 08:52.18 | joobie | it's a wonderful concept called SOURCE-NAT |
| 08:52.24 | mchou | lol |
| 08:52.26 | joobie | this needs a combination of DNAT and SNAT |
| 08:52.32 | mchou | source nat my ass |
| 08:52.34 | joobie | so the ISP will see it coming from my src ip |
| 08:52.40 | joobie | and route it back to my dynamic ip |
| 08:52.48 | joobie | and iptables will then route that back to the static ip |
| 08:52.52 | joobie | it's quite simple |
| 08:53.04 | mchou | yeah whatever |
| 08:53.05 | joobie | you need to play with NAT other than what comes iwth your netgear router |
| 08:53.20 | joobie | it's a whole can of worms you obviously havent opened |
| 08:53.20 | mchou | dude, I use openwrt |
| 08:53.24 | joobie | woopee |
| 08:53.26 | *** join/#asterisk BentLee (n=BentLee@196.210.224.166) |
| 08:53.29 | joobie | i bet you use the GUI |
| 08:53.44 | mchou | I'm done |
| 08:53.51 | joobie | good |
| 08:53.54 | joobie | co si want some dinner |
| 08:53.58 | mchou | openwrt doesnt have a gui to speak of |
| 08:54.00 | BentLee | hi, any south africans here? local issue, i think... |
| 08:54.06 | joobie | thanks for the help none the less.. ill tell you when i get it going, with NAT. |
| 08:54.25 | joobie | it has a web based GUI |
| 08:54.51 | mchou | yeah, whatever |
| 08:55.11 | joobie | mchou, relax man |
| 08:55.12 | joobie | sheesh |
| 08:56.57 | florz | joobie: do you even have to send the packets from that machine that the dynamic address is being routed to? |
| 08:57.51 | joobie | yes |
| 08:58.04 | joobie | it needs to originate from their network |
| 08:58.17 | florz | that's pretty unlikely, I guess |
| 08:59.13 | florz | if anything, your static provider may be filtering source addresses they haven't assigned to you, but I'd be surprised if your dynamic provider had any such inbound filters |
| 09:00.15 | florz | but anyhow, a little tunneling and SNAT totally should be able to solve that |
| 09:02.07 | *** join/#asterisk defswork (n=andy@195.58.91.66) |
| 09:08.07 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 09:09.47 | *** part/#asterisk BentLee (n=BentLee@196.210.224.166) |
| 09:11.50 | Pikoro | I have 4 fxs lines, is there a way to display what line an inbound call is coming in on instead of the inbound caller id? |
| 09:11.57 | *** part/#asterisk rabbit7 (n=rabbit7@ds1789722.dedicated.solnet.ch) |
| 09:19.04 | *** join/#asterisk dandate2 (n=mangy@121.1.37.145) |
| 09:19.17 | dandate2 | where cna I get free toll-free minutes or minutes for less than 3 cents? |
| 09:20.18 | dandate2 | i hear that iowa has some kind of government subsidy on toll-free... |
| 09:21.51 | *** join/#asterisk pietro (n=pietro@nat.xsec.it) |
| 09:23.10 | joobie | florz, all i need to do is connect my static * box to the dynamic ip .. then just do DNAT & SNAT at the dynamic ip end to make it go out to the sip provider |
| 09:24.04 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:e00d:f6e7:8ef3:ae84) |
| 09:24.09 | *** join/#asterisk af_ (n=getsmart@88.149.240.72) |
| 09:37.29 | *** join/#asterisk QaDeS_ (n=mklaus@213.157.13.70) |
| 09:42.20 | TSM2 | would there be any issues including multiple ring groups within another ring group |
| 09:47.27 | *** join/#asterisk soman (n=somnath@118.102.130.6) |
| 09:48.29 | *** join/#asterisk Da-Geek (n=Da-Geek@nat/redhat/x-avigasxeenoxbgkl) |
| 09:49.48 | *** join/#asterisk decimalz (n=pabx1800@203.171.196.123) |
| 09:57.51 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 10:02.00 | *** join/#asterisk coppice (n=chatzill@host86-139-183-205.range86-139.btcentralplus.com) |
| 10:02.24 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/professional/sulex) |
| 10:11.20 | *** join/#asterisk nephilinturbo (n=victorsa@189.28.0.19) |
| 10:13.25 | *** join/#asterisk Tim_Toady (n=moi@adsl38-48.kln.forthnet.gr) |
| 10:22.32 | angryuser | TSM2, have you tested ? |
| 10:23.56 | TSM2 | yup, does not seem to work |
| 10:25.29 | angryuser | TSM2, well this is it xD |
| 10:26.19 | TSM2 | still annoying, would have been nice to include numbers from other lists dynamicly, hopfully in another version of FreePBX |
| 10:28.08 | angryuser | TSM2, freepbx ? so it has nothing to do with asterisk maybe but with scripting if freepbx |
| 10:28.17 | angryuser | of* |
| 10:32.13 | TSM2 | i think it may be core to * though, if you called groups within groups i could see a problem with loops |
| 10:32.38 | TSM2 | freepbx accpets the number, just nothing happens when you call it |
| 10:34.38 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 10:42.43 | *** join/#asterisk soman (n=somnath@118.102.130.6) |
| 10:43.09 | Chainsaw | Do SIP useragents show for anyone on 1.6.1.8? |
| 10:51.52 | Chainsaw | Never mind, a handset reboot fixes it up. |
| 10:52.06 | Chainsaw | (A SIP qualify doesn't. $#@* Cisco handsets) |
| 10:56.53 | *** join/#asterisk penguinFunk (n=admin@unaffiliated/penguinfunk) |
| 11:00.14 | *** join/#asterisk louben (n=lou@212-70-216-131.ath.static.tee.gr) |
| 11:04.41 | nephilinturbo | Good morning guys... What do you think about work with ael ? |
| 11:05.24 | nephilinturbo | My parent was talking about use ael, and not asterisk |
| 11:12.01 | dlynes | Do iax2 registrations work from a 1.6.1.1/1.6.1.8 box to a 1.4.2.6.2 box for anyone else? |
| 11:13.23 | dlynes | If I give it an address for a box that doesn't do iax, it's sure to tell me, but then when I give it an address of one that does do IAX2 (the 1.4.26.2 box), it never registers, and tells me it's unreachable |
| 11:14.05 | dlynes | Otoh, in the other direction it works just fine (1.6.1.1 registering to a 1.4.22-rc5 box) |
| 11:15.22 | dlynes | Pikoro: have different contexts for every line, and then as part of your dialplan for each context, set the caller id to the line number before ringing an extension |
| 11:32.19 | dlynes | mchou: maybe he just wants a shitload of ports open on his firewall to allow sip through, and doesn't mind that his firewall isn't as much of a firewall anymore |
| 11:35.51 | TSM2 | if you want to create an extn that dials an outbound number do you just put DIAL/1234 etc... |
| 11:35.51 | dlynes | TSM2: Is the outbound number on iax2, sip, or some other technology? |
| 11:36.10 | TSM2 | dlynes: its a standard outbound number, could be via a SIP trunk or ZAP trunk |
| 11:37.00 | dlynes | TSM2: So say something like either: Dial(Zap/channelname/number), or Dial(SIP(or IAX2)/username:password@hostname/number) or Dial(SIP(or IAX2)/peername/number) |
| 11:38.18 | TSM2 | ahh, so no way to just make it follow the outbound route rules? |
| 11:38.36 | dlynes | TSM2: so some practical examples: Dial(Zap/g1/6045551212) and Dial(SIP/trunk/6045551212) |
| 11:38.48 | dlynes | TSM2: These would be in your outbound route rules |
| 11:39.31 | dlynes | TSM2: i.e. [extensions] exten => 4122,1,Dial(SIP/peername) |
| 11:39.40 | dlynes | TSM2: Goto(extensions,4122,1) |
| 11:40.38 | dlynes | TSM2: I generally set it up as macros myself, though...just for simplicity of handling dialstatus |
| 11:40.45 | nephilinturbo | Guys ...What do you think about to use ael ? |
| 11:41.05 | dlynes | TSM2: however, if you're new to asterisk, you might want to try it without macros just so you can understand the basics first |
| 11:41.36 | TSM2 | dlynes: im not new realy, just mostly using FreePBX so that when im no longer here at least other peeps can easly manage the phone system |
| 11:42.05 | dlynes | TSM2: ah...in that case, it's a whole other ball of wax |
| 11:42.18 | dlynes | TSM2: freepbx has its own unique way of doing things |
| 11:43.16 | dlynes | TSM2: I haven't delved into it enough to find out everything you need to set up to use the infrastructure of freepbx to call an extension that's in sip_custom.conf or iax_custom.conf |
| 11:43.33 | dlynes | TSM2: i.e. from extensions_custom.conf |
| 11:44.05 | dlynes | TSM2: but for normal cases, just use 'from-internal' for your context on your custom sip and iax2 extensions |
| 11:44.10 | TSM2 | dlynes: i was thinking along that idea but was trying a way to do it in the GUI using custom extention dial command |
| 11:44.16 | TSM2 | ile play more |
| 11:44.41 | dlynes | TSM2: but for those kind of questions, you're better off asking in #freepbx...most people in this channel aren't able to help on freepbx issues |
| 11:44.59 | TSM2 | yup im on both boards |
| 11:45.19 | dlynes | I only started using freepbx in the last year, and even then, try to avoid it like the plague...I'm just forced to use it, against my will |
| 11:45.44 | TSM2 | horses for courses |
| 11:45.46 | TSM2 | :) |
| 11:45.55 | dlynes | however...that being said |
| 11:46.31 | dlynes | I've got one person finally agreeing with me that elastix is a crap shoot, and certainly nothing you should be running your income stream on |
| 11:46.38 | *** join/#asterisk greysd (n=oae2@ns.plasma-trade.com) |
| 11:46.49 | TSM2 | now i dont agree |
| 11:47.00 | dlynes | so i'm currently in the process of replacing the elastix install with vanilla asterisk |
| 11:47.26 | dlynes | TSM2: have you ever tried do a yum update on an elastix box? |
| 11:47.37 | dlynes | TSM2: or upgrade the version of asterisk? |
| 11:47.40 | TSM2 | yup, works fine so far for me |
| 11:47.51 | TSM2 | if it breaks i can just compile asterisk myself and install it |
| 11:48.11 | dlynes | TSM2: are you using all the fax modules for elastix? |
| 11:48.29 | TSM2 | nop, dont want to, we have hard faxes in the office |
| 11:48.41 | tzafrir__laptop | dlynes, "vanila" as in replacing Asterisk? FreePBX? |
| 11:49.08 | dlynes | tzafrir__laptop: no...vanilla as in swapping out elastix and replacing it with plain old asterisk on top of centos |
| 11:49.16 | dlynes | tzafrir__laptop: no freepbx |
| 11:49.31 | TSM2 | i used elastix as a quick off the ground thing, the only things i use in elastix, is the endpoint manager, the conference manager, calendar |
| 11:49.48 | dlynes | tzafrir__laptop: a number of things he's been wanting to do for a while he hasn't be able to do because freepbx doesn't offer the flexibility |
| 11:50.21 | dlynes | tzafrir__laptop: I've gotten used to freepbx now, so I don't mind it for a simple endpoint install |
| 11:50.41 | dlynes | tzafrir__laptop: I just find it's a pain in the ass (or impossible) to do anything out of the ordinary with |
| 11:51.11 | leifmadsen | morning all! |
| 11:51.17 | greysd | hi! anybody know how to route list of extensions on alternate way i.e default route goes to first provider and list of extensions routes go to second, |
| 11:51.20 | dlynes | TSM2: i see |
| 11:51.31 | leifmadsen | greysd: you mean in your router, or in asterisk? |
| 11:51.50 | greysd | in asterisk |
| 11:52.15 | leifmadsen | greysd: in asterisk, that's basically just dialplan; you can create the route lists using many technologies, such as cURL or ODBC (database) |
| 11:52.29 | dlynes | greysd: It's just a matter of putting different extensions in different contexts, and saying everything in one context goes to one provider and vice-versa |
| 11:52.42 | leifmadsen | dlynes: haha, that is the simple way! :D |
| 11:52.51 | dlynes | leifmadsen: why overcomplicate things? |
| 11:53.06 | leifmadsen | dlynes: no reason too, I've just been working in clusters too long so I tend to centralize |
| 11:53.13 | leifmadsen | s/too/to |
| 11:53.25 | leifmadsen | dlynes: your way is perfectly correct :) |
| 11:53.29 | greysd | different context is now way in my case |
| 11:53.30 | dlynes | leifmadsen: ah |
| 11:53.50 | leifmadsen | dlynes: when you get to a 2nd server though, you need to centralize the information so you can update once, populate many |
| 11:54.00 | leifmadsen | I love asterisk for that; many ways to solve the same problem |
| 11:54.08 | dlynes | leifmadsen: true, but when he doesn't even know the basics, i doubt he's clustering :) |
| 11:54.47 | leifmadsen | entirely possible :) |
| 11:54.53 | greysd | it is not very basic. may be i didnt explain good becouse of my poor english |
| 11:54.57 | leifmadsen | dlynes: I've seen stranger things :D |
| 11:55.49 | leifmadsen | greysd: I think what you do is have a context that includes a couple of other contexts. The first included context would contain your "specific routes" with pattern matches and unique numbers, then you have a following include that is [default_routes] which is essentially a catch-all, which routes to your default provider |
| 11:57.09 | dlynes | greysd: you might want to define your different routes using constants as well, just to make writing the rules a little less cumbersome |
| 11:58.46 | *** join/#asterisk ming_zym (n=ming_zym@218.108.49.4) |
| 11:59.53 | dlynes | greysd: what is your native language? maybe someone else here also speaks/types it? |
| 12:00.10 | *** join/#asterisk baijum (n=baiju@203.123.174.162) |
| 12:00.24 | greysd | i have wcte121. calls come from E1 to asterisk. i need route them to 1 Prov. and about 50 extensions to 2 Prov. extensions cannot be templated |
| 12:00.29 | greysd | Russian |
| 12:00.29 | leifmadsen | greysd: http://www.pastebin.ca/1645016 |
| 12:00.52 | leifmadsen | greysd: what do you mean "extensions"? |
| 12:01.26 | greysd | PBX --->E1--->asterisk. extensions on PBX |
| 12:01.42 | greysd | callerID |
| 12:01.51 | leifmadsen | right, and what do you mean "route to different providers" ? |
| 12:02.09 | leifmadsen | they are coming from the E1... which is the provider.... which can only be 1 per E1 |
| 12:02.27 | *** join/#asterisk ska (n=ska@cpe-70-124-88-58.austin.res.rr.com) |
| 12:02.46 | greysd | no ia hve PBX with connection to aasterisk. |
| 12:02.57 | greysd | an then calls go to VOIP |
| 12:03.27 | dlynes | greysd: yeah...can't help you with russian here...french or chinese maybe though |
| 12:03.38 | greysd | :) |
| 12:04.26 | dlynes | well...not even chinese...don't have any chinese typing software installed on this box :) |
| 12:05.02 | greysd | may i try to explain in others words? ^) |
| 12:05.08 | leifmadsen | greysd: I still don't get what you're trying to do -- I don't get where the 2 Providers come into the picture; you haven't provided them in your description in a meaningful way |
| 12:05.31 | greysd | ok. wait a mom |
| 12:05.32 | leifmadsen | greysd: ya, use the pastebin to do some ASCII art to visually show what you're trying to do |
| 12:05.40 | leifmadsen | don't talk about my mom! |
| 12:05.45 | leifmadsen | j/k |
| 12:05.49 | leifmadsen | (seriously, don't though) |
| 12:06.04 | greysd | mom is moment i mean |
| 12:06.40 | leifmadsen | I figured :) |
| 12:06.43 | leifmadsen | I'm just being silly |
| 12:07.57 | *** part/#asterisk decimalz (n=pabx1800@203.171.196.123) |
| 12:11.11 | greysd | may be so? http://www.pastebin.ca/1645027 |
| 12:11.22 | dlynes | greysd: let me understand this. You have an E1 connecting your phone system to your asterisk box? |
| 12:11.36 | greysd | yes |
| 12:12.02 | dlynes | greysd: i guess I should have just read your pastebin before asking :) |
| 12:12.57 | leifmadsen | greysd: it needs to either be done in a list on asterisk (via callerID I presume), or you have to do something more dynamic (which is essentially still a list) by routing based on information in a database, or some other data abstracted location |
| 12:12.59 | dlynes | greysd: anyways...as we mentioned before...you slice and dice your dial plan according to which voip provider you want to send calls out to |
| 12:13.12 | dlynes | greysd: based on extension dialed |
| 12:13.15 | leifmadsen | you have to match on SOMETHING in order to provide logic to determine where to route a call |
| 12:14.46 | dlynes | greysd: you're wanting to do LCR, I take it? |
| 12:14.59 | greysd | what is LCR |
| 12:15.03 | dlynes | least cost routing |
| 12:15.19 | greysd | no in this case |
| 12:15.47 | dlynes | ah....leifmadsen==blitzrage...you changed your name! |
| 12:15.54 | leifmadsen | :D |
| 12:16.03 | leifmadsen | yes sir |
| 12:16.08 | leifmadsen | <-- the real leifmadsen |
| 12:16.09 | greysd | VOIP Prov is high availibility and list of PBX exts is VIP |
| 12:16.38 | dlynes | greysd: again, but without the vip/prov abbreviations? |
| 12:16.46 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:16.59 | dlynes | greysd: not sure what you mean by 'prov' and 'vip'? |
| 12:17.40 | dlynes | I'm assuming voip provisioning, but doesn't seem to make much sense, and very important person, which makes no sense at all |
| 12:18.02 | greysd | VOIPProvider 1 is fail-safe and i need to route extension of very important persons to it/ and all others to VOIPProvider 2 |
| 12:18.18 | dlynes | ah |
| 12:18.54 | dlynes | greysd: gotoif, based on extension to the appropriate dialplan context |
| 12:19.11 | greysd | but VIPs is 50! |
| 12:19.21 | greysd | 50 gotoif? |
| 12:19.24 | dlynes | greysd: So write an AGI script to do it, then |
| 12:19.53 | dlynes | greysd: or write a dialplan module |
| 12:20.16 | dlynes | greysd: so that it will get all the extension numbers from a database |
| 12:21.18 | greysd | is it possible just to read file? |
| 12:21.23 | dlynes | greysd: or do the vips have extensions in a given range? |
| 12:21.35 | dlynes | greysd: you can do whatever you want inside of an agi, or inside of a dialplan module |
| 12:22.16 | dlynes | greysd: i would actually suggest the dialplan module, personally...it should be more stable than the agi script...agi has had stability issues in the past |
| 12:22.39 | dlynes | greysd: mind you all the stability issues with agi in the past have been at high call volumes |
| 12:22.47 | greysd | no it is not range |
| 12:23.12 | dlynes | greysd: so write a module, then |
| 12:23.27 | greysd | have you link how to? |
| 12:24.58 | dlynes | greysd: /usr/local/src/1.6.1.8/asterisk-1.6.1.8/apps/app_skel.c |
| 12:25.07 | dlynes | greysd: or wherever your asterisk source drop is |
| 12:25.30 | dlynes | greysd: you do know how to write code in C, right? |
| 12:26.07 | greysd | yes |
| 12:26.30 | greysd | btw i have 1.4.26 aster |
| 12:26.47 | dlynes | greysd: the nice thing about writing a module is that after you've got it finished, you can submit it to digium for inclusion in asterisk |
| 12:27.00 | dlynes | greysd: because someone else might have a use for the module as well |
| 12:27.25 | dlynes | greysd: that app_skel.c has been in asterisk source distro since before v1.0 I think |
| 12:28.02 | [TK]D-Fender | dlynes: AGI is overkill... single Gotoif <- |
| 12:28.14 | dlynes | [TK]D-Fender: single gotoif for 50 extensions? |
| 12:28.22 | [TK]D-Fender | dlynes: Yes |
| 12:28.28 | dlynes | [TK]D-Fender: how so? |
| 12:28.55 | [TK]D-Fender | dlynes: Set a variable in the PEER to ID them and you'll have a flag you can track for who has it and who doesn't |
| 12:29.14 | [TK]D-Fender | dlynes: We're talking 50 values VS adding AGI processing load to everyone's calls |
| 12:29.30 | [TK]D-Fender | dlynes: Thats absurb... You're STILL going to haev to create that list anyway |
| 12:30.07 | dlynes | [TK]D-Fender: you do realize these extensions are coming in off of a pri from a phone system, right? |
| 12:30.09 | [TK]D-Fender | greysd: Set a var in your PEER to ID them and do a single GotoIF checking for it |
| 12:30.22 | *** join/#asterisk telnettech (n=bsimpson@office.callcopy.com) |
| 12:30.23 | dlynes | [TK]D-Fender: not from a sip peer? |
| 12:30.35 | greysd | not sip |
| 12:30.48 | [TK]D-Fender | greySo PRI > ITSP direct? |
| 12:31.15 | dlynes | [TK]D-Fender: phone extensions -> phone system -> pri/e1 -> asterisk -> two different sip trunks |
| 12:31.44 | dlynes | [TK]D-Fender: and depending on which phone extension, will determine which sip trunk to go out on |
| 12:31.56 | dlynes | [TK]D-Fender: i.e. caller id will determine the trunk |
| 12:32.02 | [TK]D-Fender | dlynes: Then sure, 51 lines of dialplan it is. |
| 12:32.12 | greysd | :)) |
| 12:32.19 | dlynes | [TK]D-Fender: yeah...that's what I said too, but he doesn't want to do it that way |
| 12:32.22 | *** join/#asterisk voipmonk (n=voipmonk@66.49.238.52) |
| 12:32.42 | dlynes | [TK]D-Fender: and that's when i suggested agi or a dialplan module |
| 12:32.54 | [TK]D-Fender | dlynes: Really? I think he just thinks there is a better way.... Really... there isn't... |
| 12:33.11 | [TK]D-Fender | dlynes: You're talking about adding external stuff to modify and processing load to every call. |
| 12:33.34 | [TK]D-Fender | dlynes: And it will take a LOT more lines of work to accomplish and give you nothing for the added effort |
| 12:33.53 | dlynes | [TK]D-Fender: well, if he's expecting the list to increase significantly, or have constant changes to it all the time, writing a dialplan module might be more time effective |
| 12:34.11 | [TK]D-Fender | dlynes: I supopse I could use the Ferrari to do my groceries... it'll jsut take 50 trips. that makes sense.... |
| 12:34.17 | dlynes | [TK]D-Fender: but, otoh if there's only going to be minimal changes on an ongoing basis, then it doesn't make any sense |
| 12:34.25 | [TK]D-Fender | dlynes: Dialplan module? hardly worth it. |
| 12:34.42 | [TK]D-Fender | dlynes: AstDB <- |
| 12:34.51 | [TK]D-Fender | dlynes: at WORST |
| 12:34.55 | dlynes | Or astdb for that matter |
| 12:35.08 | dlynes | I never think about astdb...almost never use it |
| 12:35.31 | dlynes | but yeah...astdb would probably be the better solution in that case |
| 12:35.44 | [TK]D-Fender | dlynes: Yet you're suggesting creating code that will also complicate upgrading * with extra stuff to compile.... |
| 12:36.03 | dlynes | [TK]D-Fender: not if it gets included in the asterisk branch |
| 12:36.04 | greysd | and what functions i need to use |
| 12:36.09 | [TK]D-Fender | dlynes: AGI is a more logical choice.... if you're looking to do something excessive :0 |
| 12:36.46 | [TK]D-Fender | dlynes: Or put it in an external database and do a lookup. |
| 12:37.05 | dlynes | [TK]D-Fender: you can do that from a normal dialplan? |
| 12:37.36 | dlynes | [TK]D-Fender: i.e. asterisk has a lookup dialplan application/function, or something? |
| 12:37.42 | [TK]D-Fender | dlynes: .... holy crap... do you even USE Asterisk? :p |
| 12:37.47 | dlynes | hahaha |
| 12:37.58 | dlynes | I usually jsut set and forget |
| 12:38.07 | [TK]D-Fender | takes dlynes teaching license away... |
| 12:38.17 | [TK]D-Fender | dlynes: NO SOUP FOR YOU!!! |
| 12:38.59 | dlynes | [TK]D-Fender: so what is the application to do a lookup, then? |
| 12:39.06 | *** join/#asterisk garymc (n=garymc@host81-134-0-102.in-addr.btopenworld.com) |
| 12:39.30 | [TK]D-Fender | dlynes: take your Pick between DB() MySQL(), func_ODBC, etc... |
| 12:39.57 | [TK]D-Fender | swats dlynes with a hardcover copy of TFoT |
| 12:40.02 | [TK]D-Fender | ~book |
| 12:40.03 | 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 |
| 12:40.15 | dlynes | greysd: so did you catch all that? |
| 12:40.19 | [TK]D-Fender | dlynes: Trees DIED so you could learn! |
| 12:40.39 | dlynes | [TK]D-Fender: you mean pixels? it's an ebook :) |
| 12:41.00 | *** join/#asterisk retentiveboy (n=pdugas@72.54.144.26) |
| 12:41.03 | greysd | yes/ and i have this book |
| 12:41.12 | dlynes | I suppose I should read up on some of the asterisk functions |
| 12:41.16 | dlynes | I never use the functions |
| 12:41.26 | greysd | so i have one more question :) and it is about range of extensions :) like exten => 450-549,1,NoOp() |
| 12:42.15 | [TK]D-Fender | greysd: Will take multiple patterns for a precise match. |
| 12:42.30 | dlynes | greysd: I meant specifically: [08:39]<[TK]D-Fender>dlynes: take your Pick between DB() MySQL(), func_ODBC, etc... |
| 12:43.47 | greysd | i think it cant be templated with substitutional simbols |
| 12:44.01 | greysd | dlynes: i have seen it thanks |
| 12:44.37 | dlynes | greysd: so say like DB_EXISTS(vips/${CALLERID(num)}) |
| 12:45.42 | greysd | aha. ok. DB_EXISTS... i think it is good |
| 12:45.56 | dlynes | [TK]D-Fender: btw...have you run into any issues getting a 1.6.1.8 box to iax2 register to a 1.4.26.2 box? |
| 12:46.48 | [TK]D-Fender | dlynes: Neither myself, nor any of my clients use IAX2 |
| 12:46.53 | dlynes | Or is iax2 a real crap shoot anymore, trying to get it to work between disparate versions? |
| 12:47.17 | dlynes | ic |
| 12:47.53 | *** join/#asterisk ManxPower-work (n=ewieling@24.42.221.26) |
| 12:48.02 | ManxPower-work | ~answers |
| 12:48.03 | infobot | i heard answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 12:49.07 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:50.09 | *** join/#asterisk dandate2 (n=mangy@112.202.168.28) |
| 12:50.33 | dandate2 | where can i get bulk toll -free minuets that will inevitably cost me less than 3 cents per minute? were handing a pretty popular call center probably getting more volume than comcast |
| 12:51.43 | dlynes | dandate2: what country are you in? |
| 12:51.47 | dandate2 | USA |
| 12:52.14 | dandate2 | dealing with pre-foreclosures, we get like 3000 calls a day and a long ass hold time |
| 12:52.20 | dlynes | dandate2: have you considered a pri? |
| 12:52.42 | *** join/#asterisk hugorebelo (n=hugorebe@200.171.132.124) |
| 12:52.51 | dandate2 | a T1? hows that give me toll-free minutes? |
| 12:53.27 | [TK]D-Fender | dlynes: ..... |
| 12:53.35 | dlynes | dandate2: your toll free calls get dropped onto the pri, and if it's a clec (not an ilec), you can generally get sub 3c/min for your calls |
| 12:53.37 | [TK]D-Fender | dlynes: go caffeinate or something :) |
| 12:54.05 | dlynes | [TK]D-Fender: ? |
| 12:54.22 | [TK]D-Fender | dlynes: the medium the call is carried over has nthing to do with the cost of the SERVICE |
| 12:55.01 | [TK]D-Fender | dlynes: Just like every noob coming in here going, "I'm using VoIP, how can Asterisk give me free calling?" |
| 12:55.04 | dandate2 | yeah im thinking the second they dial i am being charged no matter what. anyone know about iowa subsidies and where to get the advantage? |
| 12:55.26 | dlynes | [TK]D-Fender: true...but if he's getting his pri from his ilec, he's probably paying too much for the pri, too |
| 12:55.42 | [TK]D-Fender | dlynes: 3 cheers for dissociative thinking... |
| 12:55.58 | dandate2 | i can't get a pri anyway i never paid hospital bills from 2001 |
| 12:55.59 | dlynes | ;) |
| 12:56.16 | *** join/#asterisk retentiveboy (n=pdugas@72.54.144.26) |
| 12:56.18 | [TK]D-Fender | dlynes: You're stabbing at a vaguely potentially even slightly related link between the two... |
| 12:56.49 | dlynes | Well, fwiw, in Canada, Allstream's got cheaper than 3c/min for inbound toll free |
| 12:56.50 | [TK]D-Fender | dlynes: At that rate... his PBX probably uses ELECtriciTY... he should look into getting solar panels to lower his telecom related costs... |
| 12:56.59 | dlynes | cheers. |
| 12:57.02 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 12:57.04 | dandate2 | haha |
| 12:57.14 | telnettech | dandate2: try XO communications or PAETEC........they can get you close maybe 5 or 6 cents |
| 12:57.16 | dandate2 | no providers will give me unlimmited toll -free for a monthly rate rather than minute? |
| 12:57.19 | dlynes | Maybe allstream's cheaper in the states, too? |
| 12:57.26 | *** join/#asterisk ketema_ (n=ketema@ketema.net) |
| 12:57.38 | dandate2 | i already got an 866 for 3 cents per minute and $5/month, i've seen it as cheap as 2.7c per minuet but im really aiming for like 1 |
| 12:57.44 | ariel_ | Morning |
| 12:58.03 | telnettech | dandate2: good luck!!! |
| 12:58.12 | dlynes | dandate2: good luck with that...but I get it for 2.7c/min, and no monthly charge |
| 12:58.26 | dlynes | dandate2: 1c/min, you're dreaming |
| 12:58.33 | dandate2 | or a fat monthly charge |
| 12:58.35 | dandate2 | would be nice |
| 12:58.49 | ManxPower-work | nobody is going to do unlimited toll free |
| 12:58.52 | dandate2 | i heard in a forum that iowa subsidizes toll free anyone heard of this? |
| 12:59.16 | ManxPower-work | At least nobody that will be around for a while |
| 12:59.31 | telnettech | no flat monthly with no minute charge unless your a Fortune 100 company.....and that is maybe |
| 12:59.52 | dandate2 | bastards |
| 13:00.02 | dandate2 | well a santa barbara 805 area code might trick em |
| 13:00.06 | dandate2 | bahahahahhaha |
| 13:01.01 | [TK]D-Fender | dandate2: And you keep thinking that the # its terminated on means anything.... |
| 13:01.16 | dandate2 | oh no i meant for consumer appeal |
| 13:01.24 | dandate2 | 805 in the ads might think its toll free |
| 13:01.38 | ManxPower-work | Ah, so you are trying to trick your callers |
| 13:01.42 | dandate2 | lol |
| 13:01.59 | ManxPower-work | It all has the stench of telemarketing |
| 13:02.09 | [TK]D-Fender | ManxPower-work: Don't woory... he's also running on top of the line Optiplex servers too :p |
| 13:02.21 | dandate2 | well your on to me, i gotta raise my price i guess |
| 13:02.28 | dlynes | ManxPower-work: well...he's one of those ambulance chasers....preforeclosure call center |
| 13:02.34 | ManxPower-work | [TK]D-Fender: It didn't seem like much to worry about, poor and all that |
| 13:02.46 | dlynes | ManxPower-work: he's been to too many Robert Allen bootcamps |
| 13:02.49 | ManxPower-work | poor planning, that is |
| 13:03.23 | dandate2 | i just dont know how you plan for toll-free minute charges and overseas workers, they just cant close high enough |
| 13:03.57 | dlynes | dandate2: get somebody that can speak english without an accent |
| 13:04.05 | dlynes | dandate2: and pay them a decent wage :) |
| 13:04.11 | ManxPower-work | and don't route those calls over the internet! |
| 13:04.49 | dlynes | dandate2: and if your profit margins aren't high enough to do that, you're in the wrong business |
| 13:05.10 | *** part/#asterisk ming_zym (n=ming_zym@218.108.49.4) |
| 13:05.12 | dlynes | dandate2: fix and flips in santa barbara are way more profitable than preforeclosures |
| 13:05.15 | ManxPower-work | [TK]D-Fender: at least ITSPs like to kick telemarketers off their networks. |
| 13:07.02 | dandate2 | actually were teaching people to dofix and flips, selling my ebook |
| 13:07.15 | dandate2 | my mistake is paying the overseas workers a human wage |
| 13:07.24 | dandate2 | a call center a few blocks over is paying their workers $100/mo for a $300 sale lol |
| 13:08.27 | dlynes | dandate2: go on a recruiting drive at Walmart :) |
| 13:09.04 | dandate2 | its difficult to get americans to work remotely with x-lite, u mention skype and they think they wont get paid. overseas people see this as a hot opportunity |
| 13:09.13 | *** join/#asterisk fofware (n=chatzill@190.7.25.160) |
| 13:09.28 | dandate2 | skype and x-lite...thats much better than the tiki hut |
| 13:09.40 | garymc | anyone know how to get better ringotnes for these polycoms. The default ones suck |
| 13:09.48 | telnettech | and their is the problem with unemployment in America |
| 13:10.10 | [TK]D-Fender | garymc: Go get your own from anywhere |
| 13:10.11 | dandate2 | well i havent seen it, some guy without even a bank account wouldnt work for me cuz i told him to write his own contract |
| 13:10.12 | dlynes | telnettech: because dandate2's hiring indians and filipinos |
| 13:10.45 | telnettech | cause the profit margin is not big enough to live modest.....we want to live like we are all kings |
| 13:11.15 | dandate2 | telnet: that ones true, but with 100,000 usd you can live for 500 years in thailand |
| 13:11.29 | *** join/#asterisk iscsi (n=iscsi@78.108.73.46) |
| 13:11.46 | dlynes | telnettech: not true...I've seen some neighborhoods in the us that look like you just stepped into a third world country |
| 13:12.01 | dandate2 | the nice neighborhoods in this third world country im in look like those |
| 13:12.23 | dlynes | dandate2: third world country meaning iowa? |
| 13:12.32 | dandate2 | oh no im out here in the philippines |
| 13:12.33 | telnettech | that is cause people in that nieghborhood have no hope and just dont care anymore |
| 13:12.35 | ManxPower-work | Gotta love the South USA. Someone in south TN is using an ink pen to censor bad words in the books of the local library. |
| 13:13.01 | Nivex | ManxPower-work: Maybe they should start with the Bible |
| 13:13.02 | dlynes | dandate2: ah |
| 13:13.22 | ManxPower-work | Nivex: That is an EXCELLENT idea. |
| 13:13.26 | dandate2 | philippines is great if you got a little money though, i saw a house what would cost 2 million in san francisco for only $15,000 |
| 13:14.06 | dlynes | dandate2: yeah...I'm seeing houses in Southern Ontario (in super nice neighborhoods) go for $550K, that would go for 2 to 4 million in Vancouver |
| 13:14.31 | telnettech | thats because the house in SF or Vancouver is overpriced |
| 13:14.33 | dlynes | It's crazy how cheap real estate is in Ontario |
| 13:14.45 | dlynes | telnettech: no...not overpriced...it goes for whatever the market will bear |
| 13:15.03 | dandate2 | im really out here in the phillippines to buy loads of gold jewelry before the dollar collapses. there will be no banks or safe deposit boxes when that happens so we gotta wear our wealth |
| 13:15.07 | dlynes | telnettech: there's a lot of cash rich hong kongese in Vancouver that don't need a mortgage |
| 13:15.35 | dlynes | dandate2: and then the price of gold plummets, and you've lost half your value |
| 13:15.46 | dandate2 | oh gold will never plummet i cannot even fathom how |
| 13:15.59 | dandate2 | unless they nuke a mountain and find so much gold the stuffs not even rare anymore |
| 13:16.11 | telnettech | the devaluation of the dollar is a conspriacy by the super rich and governments to control the world's population |
| 13:16.42 | dandate2 | oh for sure, like this gov bailout crap, i still cant get a loan |
| 13:17.57 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:18.34 | dandate2 | they print money to give to banks to loan to people but they just pay their executives more |
| 13:18.53 | *** join/#asterisk silverglade00 (n=mdkelly@anubis.ddns.ualr.edu) |
| 13:19.12 | garymc | [TK]D-Fender : What format have the ring tones got to be in and how do i add them? |
| 13:19.29 | [TK]D-Fender | garymc: Go read the admin guide & WIKI... |
| 13:19.40 | garymc | ok |
| 13:19.41 | [TK]D-Fender | garymc: 8 khz ULAW |
| 13:19.45 | ManxPower-work | telnettech: I thought it was the rest of the world finally getting pissed off enough at the USA to stop using their money for international transactions. |
| 13:20.36 | dandate2 | well the rest of the worlds money will collapse too eventually, its all fiat anyway printed from nothing and backed only by nuclear weapons |
| 13:20.59 | mchou | anyone ever played with #cx here? |
| 13:21.05 | mchou | 3CX* |
| 13:22.27 | dandate2 | i mean who the hell is for the war in afghanistan? i think thats more unpopular than vietnam but these days noone can afford gas to drive to the protest |
| 13:23.00 | telnettech | Manx: I find myself understanding less about economics and more sympathy to the anarchist protesters at these different "summits" |
| 13:23.06 | ManxPower-work | I don't agree it is more unpopular than Viet Nam. |
| 13:23.17 | telnettech | this global economy is just shit!!!! |
| 13:23.34 | dandate2 | i think that the vietnam war was like 50% unpopular ? |
| 13:23.57 | dlynes | I think it was Bush that was unpopular...not so much Afghanistan |
| 13:24.05 | dlynes | Iraq was unpopular though |
| 13:24.16 | dandate2 | but it was a war against athiest communists so all the christians would back that, what we have here is a war against islam which countries were our allies not but 50 years ago |
| 13:24.39 | dandate2 | cairo was like highclass resort |
| 13:24.40 | dlynes | dandate2: afghanistan was an ally? |
| 13:24.49 | telnettech | the problem with Vietnam and Afghanistan is that it is a war about ideology....people have different opinions and now it is to see who can piss the farther and longer without making other people mad |
| 13:25.02 | dandate2 | yeah remember casablanca, there wasnt any terrorists lol |
| 13:25.14 | dlynes | dandate2: I thought Taliban was a secret insurgent army used as a cia tool against the russians? i.e. not really an ally |
| 13:25.37 | dandate2 | lol |
| 13:25.45 | dandate2 | if you give em money they must be a friend |
| 13:26.03 | dandate2 | but i guess in todays banking world giving money is more along the lines of adjustable rate mortgage |
| 13:26.06 | dlynes | dandate2: well, and besides...the taliban provided the cia with the opium they needed to take care of the hood |
| 13:26.38 | dandate2 | yeah true that, nothing wrong with plants tho no need to be shady about them |
| 13:27.17 | dlynes | opium and cocaine are both good tools to keep civil unrest at rest |
| 13:27.23 | dandate2 | i smoke cigarettes it comes from a plant, prolly more addictive than opium i think. |
| 13:27.45 | dandate2 | and when i tried to quit i almost pulled my hair out tooo |
| 13:27.46 | dandate2 | lol |
| 13:28.06 | dlynes | aaaaaaaaaaaaaaaaaanyways |
| 13:28.31 | garymc | Ok anyone know a place I can get good free ringtones or I gotta pay for them? |
| 13:28.54 | dlynes | has anyone had issues getting a 1.6.1.8 box to register to a 1.4.2.26 box over iax2? Or should I just give up on iax2 and use sip? |
| 13:28.54 | dandate2 | but u know tobaccco has the highest seed yield per gram than any other plant in the world, i realized manjust wasn't meant to run out of it |
| 13:29.24 | dlynes | garymc: what format? |
| 13:29.46 | garymc | ULaw 8khz |
| 13:30.03 | *** join/#asterisk Anth8708 (n=Anth8708@client105.jdcc.edu) |
| 13:30.04 | dlynes | garymc: just go to a free wav file site, and then use sox to convert them |
| 13:30.23 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 13:32.06 | dlynes | nvm...it's registering now, for some crazy reason |
| 13:32.30 | ManxPower-work | dlynes: Use SIP |
| 13:32.53 | ManxPower-work | dlynes: Use SIP. |
| 13:32.58 | ManxPower-work | oops |
| 13:33.30 | dlynes | ManxPower-work: iax2 is really that buggy? |
| 13:33.57 | ManxPower-work | dlynes: using SIP you could, in theory, enable reinvites at some future time. |
| 13:34.24 | dlynes | ManxPower-work: which would do what for me? Sorry...not familiar with the internals of sip |
| 13:35.25 | dlynes | ManxPower-work: does that mean in theory that sometime down the road, you won't have sip peers that are unreachable for up to 4 or 5 seconds at a time? |
| 13:35.26 | ManxPower-work | dlynes: reinvites allow the two endpoints to send their audio direct, instead of thru the server. Signaling is still sent thru the server. |
| 13:35.35 | dlynes | ManxPower-work: ah |
| 13:35.41 | ManxPower-work | dlynes: Oh, they way to fix that is to not enable qualify= |
| 13:36.15 | dlynes | ManxPower-work: but then it doesn't keep the firewall/router open? |
| 13:36.38 | ManxPower-work | dlynes: you can lower the registration interval or enable keepalives in the client. |
| 13:36.47 | dlynes | ic |
| 13:36.53 | dlynes | ok, thanks for the heads up |
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| 13:38.26 | *** join/#asterisk Katty (n=asterisk@mail.copi-rite.com) |
| 13:38.29 | Katty | :> |
| 13:41.20 | *** join/#asterisk Boardy (n=chatzill@kirakira.xs4all.nl) |
| 13:43.19 | Boardy | I want to have Asterisk generate an SMS message and send it to one of the connected IP telephones. It should be SMS capable. |
| 13:43.23 | Boardy | I don't know where to start. |
| 13:43.46 | Boardy | I tried smsq <extension> <msg> but it doesn't work (yet?) |
| 13:43.51 | *** join/#asterisk luckyaba (n=lucky@ip72-194-215-55.sb.sd.cox.net) |
| 13:44.02 | Boardy | Any pointers? |
| 13:45.01 | *** join/#asterisk Anth8708 (n=Anth8708@client105.jdcc.edu) |
| 13:45.18 | *** join/#asterisk Whtsup (n=sssi@WimaxUser3715-154.wateen.net) |
| 13:46.06 | Whtsup | hello |
| 13:46.08 | Whtsup | everyone |
| 13:46.59 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 13:47.03 | Katty | sure is quiet this morning |
| 13:47.14 | Whtsup | yeah |
| 13:48.16 | jaytee | I love these quiet mornings where I can relax, sit back and enjoy a nice toasted bagel with cream cheese |
| 13:48.17 | ManxPower-work | Boardy: IP phones are not SMS compatible. |
| 13:48.32 | Whtsup | hello |
| 13:48.35 | Whtsup | guys |
| 13:48.42 | Whtsup | any one here who will help me ? |
| 13:48.42 | Katty | Whtsup: if you need help, please ask in channel. don't send me private messages. that's not polite. |
| 13:48.47 | ManxPower-work | Boardy: Asterisk's SMS support only works in the parts of the works where the carrier supports PSTN SMS. |
| 13:48.47 | Katty | infobot: ask |
| 13:48.48 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 13:49.10 | Katty | jaytee: that sounds yummy. what sort of bagel is it? |
| 13:49.31 | Boardy | Manx: Hmmmm... it says in the description of the telephone it can send and receive SMS messages |
| 13:49.38 | Whtsup | i need some help about asterisk dialplan |
| 13:49.39 | Boardy | (Siemens A580IP) |
| 13:49.53 | *** join/#asterisk LemensTS (n=customgt@adsl-70-238-136-66.dsl.stlsmo.sbcglobal.net) |
| 13:50.02 | LemensTS | exten => s,n,Set(i=${MATH(2.6+0,int)}) i got that to work, but it rounds it down to 2 instead of up to 3...is there no way around that? |
| 13:50.10 | ManxPower-work | Boardy: Correct. It can send SMS using FSK over PSTN. |
| 13:50.38 | Whtsup | ??? |
| 13:50.46 | ManxPower-work | LemensTS: Maybe you should start off by trying to add 0.5 instead of 0 |
| 13:51.11 | ManxPower-work | Whtsup: Are you going to ask your question or not? |
| 13:51.38 | [TK]D-Fender | LemensTS: INT != ROUND |
| 13:51.47 | *** join/#asterisk pblake71 (n=Phil@173-12-112-209-miami.hfc.comcastbusiness.net) |
| 13:51.51 | Boardy | Manx: So this means I can only set up my telephone to work with some specific SMS centres? |
| 13:51.53 | Katty | Question--what is the square root of Pie. |
| 13:51.54 | dlynes | Boardy: what he's saying is it's only for pstn, not sip |
| 13:51.54 | [TK]D-Fender | LemensTS: And yes... +.5 |
| 13:52.03 | LemensTS | ManxPower: 3.0+.5 = 3 |
| 13:52.05 | ManxPower-work | Boardy: or specific PSTN SMS phones. |
| 13:52.11 | LemensTS | whoops sorry didnt mean to send that |
| 13:52.19 | [TK]D-Fender | LemensTS: Yes, but 3 is what you are looking to round <- |
| 13:52.26 | LemensTS | Manxpower yea that will work perfectly |
| 13:52.30 | dlynes | Boardy: if you want to get what you call sms to work with that phone, you'll need to use sip headers, or some other work around |
| 13:52.35 | LemensTS | Thanks |
| 13:53.10 | dlynes | Boardy: it might even do sms via some proprietary method that only the siemens phone system knows how to do |
| 13:53.14 | Boardy | So then I would need to dig deep into some heavy scripting to get it working |
| 13:53.23 | Whtsup | guys |
| 13:53.31 | Boardy | Hmmmm... Not worth the effort, I guess. Pitty |
| 13:53.37 | ManxPower-work | LemensTS: have you even read up on how to round numbers? |
| 13:53.46 | dlynes | Whtsup: and gals ... *points* to Katty. |
| 13:53.59 | LemensTS | ManxPower, yea for a couple hours last night. Thats the best i could get. |
| 13:54.06 | dlynes | Whtsup: ... |
| 13:54.08 | dlynes | infobot: ask |
| 13:54.09 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 13:54.15 | LemensTS | There was a ROUND(x) expression but couldnt get it to work |
| 13:54.33 | Whtsup | i want to remove prefix from my dial plan |
| 13:54.45 | dlynes | Whtsup: be more specific |
| 13:54.51 | ManxPower-work | LemensTS: Where EXACTLY did you find a ROUND expression? |
| 13:54.52 | Whtsup | well |
| 13:54.57 | Whtsup | i will paste ma dial plan |
| 13:55.07 | ManxPower-work | Whtsup: DO NOT PASTE TO THE CHANNE: |
| 13:55.15 | ManxPower-work | ~pb |
| 13:55.16 | 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://bin.cakephp.org/ , or apt-get install pastebinit |
| 13:55.16 | LemensTS | http://www.voip-info.org/wiki/view/Asterisk+Expressions |
| 13:55.36 | ManxPower-work | LemensTS: that is not valid place to find the stuff availble in YOUR asterisk. |
| 13:56.04 | ManxPower-work | "core show applications" and "core show FUNCTIONS" (remember functions are all upper case) is the correct place to find what is available in YOUR asterisk. |
| 13:56.09 | LemensTS | ManxPower: yea core show functions didnt show it |
| 13:56.28 | LemensTS | how do i add those into my asterisk if they are not there? |
| 13:56.34 | LemensTS | im on 1.6 |
| 13:56.41 | ManxPower-work | LemensTS: YOU CAN'T. The WIKI IS WRONG> |
| 13:56.48 | ManxPower-work | The Wiki is usually wrong. |
| 13:56.53 | LemensTS | Ahh ok |
| 13:57.02 | Whtsup | exten => _0092X.,1,NoCDR() |
| 13:57.02 | Whtsup | exten => _0092X.,2,Wait(1) |
| 13:57.02 | Whtsup | exten => _0092X.,3,Dial(Dahdi/r0/${EXTEN},45,) |
| 13:57.19 | LemensTS | whtsup: paste to pastebin.com |
| 13:57.21 | Whtsup | this is ma dial plan i want to remove 0092 and add o |
| 13:57.36 | ManxPower-work | Whtsup: To strip off the leading char of a variable use ${VARIABLE:1} |
| 13:57.44 | dlynes | Whtsup: o${EXTEN:4} |
| 13:58.02 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 13:58.07 | ManxPower-work | Whtsup: for more information see channelvariables.txt in your Asterisk source code. |
| 13:58.52 | dlynes | Whtsup: Also read the following page: http://www.voip-info.org/wiki/view/Asterisk+variables |
| 13:59.04 | Whtsup | ok |
| 13:59.21 | ManxPower-work | dlynes: channelvariables.txt will give more informtation and won't be wrong like the wiki frequently is. |
| 13:59.38 | dlynes | ManxPower-work: not everyone has the source installed though, either |
| 13:59.55 | dlynes | ManxPower-work: although 1-1/2 to 2 years ago, I think everyone was using source, because binary distros were useless |
| 14:01.24 | dlynes | otoh, channelvariables.txt probably mentions some built in variables that exist in newer versions that the wiki won't have |
| 14:03.10 | [TK]D-Fender | [09:59]<dlynes>ManxPower-work: not everyone has the source installed though, either <- No excuse. |
| 14:03.30 | jaytee | how can 1.6.1.8 be released when they just released 1.6.1.7-rc2 less than a week ago? |
| 14:03.34 | [TK]D-Fender | dlynes: That is like saying "there is good documentation and I can't be bothered to download it" |
| 14:03.49 | [TK]D-Fender | jaytee: Jump because of a security issue |
| 14:03.52 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 14:03.52 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:04.05 | [TK]D-Fender | jaytee: This has happened a few times in this past year |
| 14:05.10 | beek | Morning [TK]D-Fender / jaytee |
| 14:05.29 | jaytee | [TK]D-Fender, the ACL being ignored on SIP invites? yeah, I read that but now I'm confused as to whether I should jump to 1.6.1.8 or will 1.6.1.7 ever actually be released or is it now dropped in favor of 1.6.1.8 |
| 14:05.59 | beek | I'm glad that they're reverting to their older numbering scheme. |
| 14:06.12 | ManxPower-work | LOL! The Asterisk PBX at work has stopped accepting DTMF. My response was basically "Digium analog cards suck, a reboot should fix it" |
| 14:06.20 | *** join/#asterisk slinksh0t (n=slinksh0@adsl-9-177-182.mia.bellsouth.net) |
| 14:09.15 | ManxPower-work | dlynes: The Asterisk documentation is part of the Asterisk source. If you want docs you need the source. |
| 14:10.11 | ManxPower-work | If you want documentation that is full or errors or does not apply to your version of Asterisk then feel free to use other sources of documentation |
| 14:11.35 | Chainsaw | jaytee: 1.6.1.7 release candidates? That code will eventually be released as 1.6.1.9 |
| 14:11.48 | jaytee | Chainsaw, when? |
| 14:12.00 | Chainsaw | jaytee: The security update was released as 1.6.1.8 to make sure nobody expected the 1.6.1.7 code to be in there. |
| 14:12.10 | ManxPower-work | The frickin' security advisory tells you what version the fix is in. |
| 14:12.16 | Chainsaw | jaytee: You'd have to ask leifmadsen that. I don't work for Digium. |
| 14:12.25 | ManxPower-work | Chainsaw: neither does leifmadsen |
| 14:12.41 | leifmadsen | true |
| 14:12.44 | ManxPower-work | At least I don't think he does. They might have hired him at some point. |
| 14:12.54 | Chainsaw | He does seem like an in-charge kind of guy. So still a good bet :) |
| 14:13.49 | ManxPower-work | ~leifmadsen |
| 14:13.50 | infobot | i guess leifmadsen is blitzrage |
| 14:13.59 | ManxPower-work | ~blitzrage |
| 14:13.59 | infobot | rumour has it, blitzrage is a super cool fellow |
| 14:13.59 | jaytee | I hear he's a terrific dancer |
| 14:14.04 | ManxPower-work | ~manxpower |
| 14:14.05 | infobot | ManxPower has been using Asterisk in production since late 2001. Currently works at InterGlobe Communications, a CLEC based in NYC with service in NY, NJ, FL, and TX. http://www.nyigc.com |
| 14:14.16 | ManxPower-work | silly bot |
| 14:14.21 | jaytee | ~jaytee |
| 14:14.22 | infobot | it has been said that jaytee is likes to play with the animals in the zoo |
| 14:14.50 | jaytee | see, that's it! that's my sole claim to fame. I am gonna be sooooo forgotten 10 seconds after I'm dead |
| 14:17.03 | beek | jaytee: not true! |
| 14:17.07 | eppigy | hello |
| 14:17.09 | eppigy | i am dave |
| 14:17.48 | jaytee | morning beek |
| 14:17.57 | jaytee | dave! |
| 14:19.17 | wastrel | clec based in nyc eh |
| 14:19.22 | wastrel | you can come over and help me with my asterisk box |
| 14:19.53 | wastrel | ~wastrel |
| 14:19.53 | infobot | i guess wastrel is a funny monkey |
| 14:19.58 | wastrel | ! |
| 14:20.41 | wastrel | is infobot actually running infobot |
| 14:23.52 | *** join/#asterisk moy (n=moy@74.12.131.148) |
| 14:25.03 | telnettech | self promoting with the infobot is such a sin!!!! |
| 14:26.23 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 14:34.23 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 14:35.06 | *** join/#asterisk pblake71 (n=Phil@173-12-112-209-miami.hfc.comcastbusiness.net) |
| 14:35.18 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:36.49 | *** join/#asterisk chendy (n=chatzill@116.25.81.155) |
| 14:38.15 | tzafrir__laptop | ~telnettech |
| 14:38.15 | infobot | i heard telnettech is someone who thinks about self-promotion as some sort of mathematical function |
| 14:38.52 | pblake71 | I am currently running Asterisk on a Ubuntu box with Freepbx. I would like reinstall using Asterisknow or trixbox on Centos. Is it possible to bring my setting from the old server to the new install? |
| 14:39.26 | tzafrir__laptop | infobot, tell wastrel about ask |
| 14:39.47 | ariel_ | pblake71: there is a channel that can help you with that #freepbx |
| 14:40.23 | ariel_ | pblake71: but yes you can save your settings via the backup program and then instal AsteriskNOW and recover your settings. But not to Trixbox |
| 14:41.45 | pblake71 | ariel - thank you. |
| 14:48.59 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 14:54.09 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
| 14:56.37 | *** join/#asterisk friartuck (n=pmccary@66.162.90.56) |
| 14:57.11 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
| 14:59.37 | Boardy | dlynes: Update... It works. When I use smsq with the correct parameters and chown the files to asterisk.asterisk before putting them in the spool dir. |
| 14:59.53 | p3nguin_ | When I use PlayTones(dial), DTMF does not cause the tone to stop, so I figured StopPlayTones might be required. But how do I get it to exit when it detects DTMF being entered? |
| 15:00.47 | ManxPower-work | p3nguin_: why not use DISA? |
| 15:01.13 | p3nguin_ | You know, that's probably a better idea. |
| 15:02.48 | *** join/#asterisk swiff (n=swiff@0x5da24e0a.cpe.ge-1-1-0-1309.vbrnqu2.customer.tele.dk) |
| 15:04.03 | *** join/#asterisk pirulo (n=andres@65.102.99.5) |
| 15:04.13 | swiff | Hello. I'm having issues related to hot desking on my SNOM 820 phones. http://wiki.snom.com/FAQ/Why_is_there_no_prompt_after_entering_a_wrong_password clearly states that asterisk should sent back 401/407 when login is not correct - however, it does not. It only gives me "Registration from '<sip:xxx@phones.domain>' failed for '172.16.x.x' - Wrong password" in the logs - Any clues? |
| 15:04.22 | p3nguin_ | I'm trying to read DTMF into a variable, and using DISA will probably force me to add another context, but it will certainly get the job done. |
| 15:06.35 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 15:08.37 | [TK]D-Fender | p3nguin_: Then background a recording of dialtone, and stop trying to complicate things |
| 15:08.51 | [TK]D-Fender | facepalms |
| 15:09.12 | p3nguin_ | I'm going to try the DISA way first, without using an additional context. |
| 15:09.28 | p3nguin_ | I think it might just allow me to NOT even need that variable. |
| 15:10.39 | p3nguin_ | WTF, why is my toll-free DID failed over?! |
| 15:11.32 | *** join/#asterisk bluOxigen (n=asad@static-host119-73-66-30.link.net.pk) |
| 15:17.49 | *** join/#asterisk Dovid (n=annon@tony09-118-62.inter.net.il) |
| 15:18.16 | *** join/#asterisk pawpro (n=Miranda@213.166.12.34) |
| 15:18.23 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 15:20.32 | pawpro | hello everybody. I need to be able to poke my asterisks every now end then for monitoring purposes but if i send them OPTIONS i always get "SIP message could not be handled, bad request". Can anybody help? The message i send it is http://pastebin.com/d53e6f1c1 it's written by hand so please point what is to be fixed. Thank you |
| 15:20.45 | diatonic1 | Can anyone tell me how to add extensions to a spygroup so that Chanspy() can be used to spy on that specific group? |
| 15:20.53 | Naikrovek | pawpro: use snmp? |
| 15:21.00 | Naikrovek | despite it's complex setup |
| 15:21.25 | garymc | why would the time be an hour out on my phones. I know here in UK clocks went back 1 hour Sunday gone. But my ip650 is displaying the correct time, and in its config it is pointing to the same time server |
| 15:21.25 | pawpro | Naikrovek: it's not only asterisk at the end so it must be SIP monitrong. I have other aspects of the system covered |
| 15:21.39 | Naikrovek | ah |
| 15:21.42 | Naikrovek | no idea then |
| 15:23.15 | rene- | mines also went stupid i am in mexico GMT-6 |
| 15:24.33 | [TK]D-Fender | pawpro: its fine <- |
| 15:24.44 | [TK]D-Fender | pawpro: there is nothing to fix. |
| 15:25.01 | pawpro | so why is asterisk saying that it's bad request? |
| 15:25.30 | [TK]D-Fender | pawpro: Because * doesn't care to respond with any detail because its a PBX, not a dumb phone. Functionality isn't the same |
| 15:25.32 | diatonic1 | garymc: Time server has no awareness of daylight savings time. Only GMT. It is up to the client device to account for daylight savings time. Hit the web server on the IP address of your phone and you can set DST dates on the general section |
| 15:25.43 | [TK]D-Fender | pawpro: the answer doesn't matter. That fact that it does answer is what's important |
| 15:26.24 | pawpro | [TK]D-Fender: ohh no. It does not respond. This is what I got in logs. I would be delighted to get any SIP response |
| 15:27.12 | [TK]D-Fender | pawpro: pastebint he complete SIP debug from * CLI |
| 15:28.42 | garymc | just disabled Daylight savings and set the time accordingly |
| 15:29.13 | pawpro | [TK]D-Fender: Thank you for looking at it http://pastebin.com/d17d41786 |
| 15:29.42 | [TK]D-Fender | pawpro: that isn't proper * SIP debug |
| 15:30.41 | *** join/#asterisk ryduh (n=ryduh@204.16.143.186) |
| 15:30.44 | *** join/#asterisk Caplain (i=shayne@2001:470:5:fb:c4c6:d4ba:1f33:6923) |
| 15:31.29 | *** join/#asterisk bsaxon (n=bsaxon@12.68.234.174) |
| 15:31.38 | Katty | stretches |
| 15:31.54 | diatonic1 | garymc: It's an easy fix. See settings here: http://is.gd/4Eqov otherwise it will be a pain in your ass next March |
| 15:32.00 | thehar | tosses a snowball at Katty |
| 15:32.28 | Katty | :< |
| 15:32.44 | Katty | chunks bigger snowball at thehar :> |
| 15:32.53 | garymc | Are the daylight savings at the same days every year? |
| 15:33.00 | Katty | no |
| 15:33.06 | Katty | a couple years ago it got switched |
| 15:33.09 | garymc | surely I gotta get them right otherwise it will be wrong every time |
| 15:33.10 | thehar | hahaha |
| 15:33.12 | thehar | it's snowing outside! |
| 15:33.17 | Katty | WHAT?! |
| 15:33.23 | thehar | yessss |
| 15:33.25 | thehar | in da slc |
| 15:33.26 | thehar | snow |
| 15:33.27 | Katty | eww. keep that stuff over there. |
| 15:33.27 | diatonic1 | garymc: No, it goes by Sundays on certain weeks here in the US |
| 15:33.37 | thehar | but it's so pretty |
| 15:33.46 | diatonic1 | garymc: that's why "Fixed Day" is disabled in my config |
| 15:34.10 | garymc | yeah but you got November? |
| 15:34.21 | diatonic1 | garymc: DST ends for us first Sunday in November |
| 15:34.56 | garymc | right i need to find out uk times then, cos we switch last sunday of the month which was sunday gone |
| 15:36.19 | diatonic1 | garymc: Do you have a phone that is working right that you can look at? |
| 15:36.24 | garymc | its ok ill just google the UK DST times and dates |
| 15:36.24 | garymc | then ill put them in |
| 15:37.28 | *** join/#asterisk gardo (n=gardo@121.96.137.100) |
| 15:38.21 | Chainsaw | garymc: http://en.wikipedia.org/wiki/Daylight_saving_time_around_the_world#In_general |
| 15:38.45 | Chainsaw | garymc: As usual there's United States and rest-of-world. So look at rest-of-world and you'll be fine :) |
| 15:39.12 | IBC_jkenney | I have a fast question about meetme. If I want to set it up that everyone hears music/or cannot speak to each other until the admin logins to the conference is that too difficult I looked at the options but nothing is jumping out at me |
| 15:39.14 | IBC_jkenney | i have the music |
| 15:39.50 | IBC_jkenney | and what not just not the part where asterisk waits for the admin before giving everyone a voice |
| 15:43.15 | *** join/#asterisk krionux (n=scostes@unaffiliated/krion) |
| 15:43.18 | krionux | hi |
| 15:44.02 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 15:44.57 | Chainsaw | IBC_jkenney: I don't believe meetme offers that limbo state where you have recorded your greeting, but are not on the conference. |
| 15:45.46 | Chainsaw | IBC_jkenney: (So you'd have to bounce them off if your admin has not surfaced yet, I don't believe you can queue them up for when the admin gets there) |
| 15:46.04 | IBC_jkenney | yeah thats what i'm looking for |
| 15:46.37 | [TK]D-Fender | IBC_jkenney: "core show application meetme" <- |
| 15:48.10 | Chainsaw | There's an interesting one in there, IBC_jkenney. The w option allegedly waits for a specific user to enter, but the parameter it takes is in seconds. |
| 15:48.15 | Chainsaw | (This is on 1.6.1.8) |
| 15:49.01 | [TK]D-Fender | Chainsaw: Missed a key word there... |
| 15:49.33 | *** join/#asterisk Chaytan (n=chatzill@modemcable091.107-22-96.mc.videotron.ca) |
| 15:49.47 | Chainsaw | [TK]D-Fender: Which one is that? |
| 15:50.12 | *** join/#asterisk niekie (i=quasselc@dreamworld.bergnetworks.com) |
| 15:50.19 | Chaytan | hey guys... please can someone tell me how some companies become long distance carriers and are able to route their client's 011 calls through their softswitch? |
| 15:51.08 | *** join/#asterisk QaDeS (n=mklaus@213.157.13.70) |
| 15:51.12 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 15:51.26 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 15:52.12 | ManxPower-work | Does anyone know how to make a Polycom play a howler tone when the user forgets to hangup? |
| 15:53.50 | ariel_ | that would be funny and useful thing to do, maybe send via the hangup context from a call to there 2nd line a buzz tone via info? |
| 15:58.34 | angryuser | Chaytan, what represent 011 calls ? urgent calls ? (medical ect...) ? |
| 15:59.22 | [TK]D-Fender | Chainsaw: "OPTIONAL" <- |
| 16:00.33 | *** part/#asterisk Boardy (n=chatzill@kirakira.xs4all.nl) |
| 16:00.46 | ManxPower-work | ariel_: Asterisk does not have a call active in this case. |
| 16:01.10 | ManxPower-work | angryuser: in the USA 011 indicates "international call" |
| 16:01.15 | ariel_ | was it an inbound call? |
| 16:01.39 | ManxPower-work | ariel_: My problem is the the user fails to hang up the handset properly after a call |
| 16:02.15 | ryduh | what's a cheap voip provider other than voip.ms? |
| 16:03.11 | ariel_ | cheap, good, inbound or outbound? both from where to where, there are so many options that I really don't know how to reply |
| 16:03.34 | ariel_ | cheap free inbound ipkall example |
| 16:03.42 | [TK]D-Fender | ManxPower-work: Whats the problem with that? |
| 16:04.04 | [TK]D-Fender | ManxPower-work: It won't hold a dead call off-line |
| 16:04.14 | ManxPower-work | [TK]D-Fender: a call comes in and she picks up the handset and nobody is there, she has to press a line key and then she says "sometimes I have to press the line key" |
| 16:04.20 | ryduh | outbound for now. I have a problem sending DTMF tones and I want to test it with another provider. Cheap I guess |
| 16:04.20 | [TK]D-Fender | ManxPower-work: If the other side terminates the phone will treat it as on-hook |
| 16:04.35 | ManxPower-work | [TK]D-Fender: EXACTLY! Even if the handset is offhook. |
| 16:04.51 | diatonic1 | ryduh: Have you tried changing your DTMF method to info or inband? |
| 16:04.52 | [TK]D-Fender | ManxPower-work: Yes, thell them they can hit the hookswitch as well. |
| 16:05.02 | [TK]D-Fender | ManxPower-work: Because you ahve to take an action to answer a call |
| 16:05.18 | Chaytan | Angryuser, sorry just saw your message, 011 represents long distance calls here in Canada |
| 16:05.23 | ryduh | diatonic1: yes, i've tried all 3 modes and set all three modes in the settings for voip.ms |
| 16:05.25 | [TK]D-Fender | ManxPower-work: picking up a handset that isn't sitting on a sensor isn't going to cut it |
| 16:05.51 | ManxPower-work | [TK]D-Fender: we are talking about someone that can't manager to hangup the handset. |
| 16:05.53 | diatonic1 | ryduh: I use vitelity and 'info' works flawlessly |
| 16:06.01 | ryduh | diatonic1: ariel_: it currently works SOMETIMES.. definitely not reliable |
| 16:07.15 | diatonic1 | ryduh: When I was having DTMF issues I would call my cell phone and press digits from my sip phone to listen to DTMF. Helped see what worked and what didn't |
| 16:07.20 | [TK]D-Fender | ManxPower-work: tell "fuck off, what do you expect?" |
| 16:07.27 | [TK]D-Fender | ManxPower-work: Seriously.... |
| 16:07.44 | ManxPower-work | [TK]D-Fender: I prefer to work around user stupidity -- less argument then. |
| 16:07.52 | p3nguin_ | I use dtmfmode = rfc2833 with VoIP.ms. And VoIP.ms is a Vitelity reseller, so I would expect anything on Vitelity to work on VoIP.ms. |
| 16:08.01 | [TK]D-Fender | ManxPower-work: Sorry... doesn't work that way... esp with Polycom |
| 16:08.01 | *** join/#asterisk timeshell_atwork (n=chatzill@142.46.193.194) |
| 16:08.02 | ryduh | diatonic1: I've been using that method as well. I've got it to send the # sign correctly so that it tries to check my voicemail but it won't accept my pin |
| 16:08.38 | ManxPower-work | I can put up with the constant (incorrect) reports of problems receiving calls or I can send 10,000 volts to a user that screws up hanging up the handset. Which do you think is more fun AND effective? |
| 16:08.48 | ryduh | p3nguin_: the only way I've been able to send DTMF at all is with inband/ulaw. I'm using voip.ms' us3 server. |
| 16:08.57 | angryuser | Chaytan, what's the problem then ? nobody will teach you here how to become another Voip provider out of the box, it is not that easy and requires at least basic knowledge in voip telephony, its like me asking a bmw forum "hey how can i build a sell a cars" |
| 16:09.04 | ryduh | I'll try again with another server, and rfc2833 |
| 16:09.29 | p3nguin_ | I use us2 and us4. |
| 16:09.35 | diatonic1 | p3nguin_: I've never gotten dtmfmode = rfc2833 to work reliably on my vitelity service |
| 16:09.58 | diatonic1 | but dtmfmode = info with ulaw works great |
| 16:10.06 | diatonic1 | (for me) |
| 16:10.12 | p3nguin_ | I also run it on auto in the provider's web portal. |
| 16:10.15 | ryduh | p3nguin_: diatonic1 it's important to note that transferring dtmf from a soft phone works fine most of thetime if not all of the time |
| 16:10.26 | ryduh | the problem is when * generates the dtmf |
| 16:10.38 | [TK]D-Fender | Chaytan: I can set up an account with any ITSP that supports international calling and give you an account on my server. Congratulations, I am now "able to route their client's 011 calls through their softswitch" |
| 16:10.41 | p3nguin_ | Why would * generate DTMF? |
| 16:10.46 | ryduh | SendDTMF |
| 16:10.47 | diatonic1 | ryduh: Mine are Polycom endpoints and Audiocodes FXS gateways |
| 16:11.08 | ManxPower-work | Asterisk 1.4 totally rewrote DTMF support on RTP/SIP. It should work better now. |
| 16:11.37 | ryduh | maybe I used the wrong word. not Generate but send. ManxPower-work I'm running 1.4.16.2 |
| 16:12.32 | ManxPower-work | ryduh: maybe your carrier is still using 1.2? |
| 16:12.34 | ryduh | ManxPower-work: Would it be worth it to upgrade to 1.6? I've tried to avoid using anything * claimed was deprecated in 1.6 (using gosub's etc) |
| 16:12.39 | p3nguin_ | ryduh: Why not just send your own DTMF from your phone? |
| 16:12.52 | ryduh | p3nguin_: because it's an automated call ;/ |
| 16:12.59 | ManxPower-work | ryduh: I'm not aware of *MAJOR* DTMF changes in 1.6, but I've not looked closely |
| 16:13.26 | *** part/#asterisk xmitter (n=xmitter@c-24-20-192-165.hsd1.or.comcast.net) |
| 16:17.52 | *** join/#asterisk JKac3BEq (n=JKac3BEq@173-20-68-31.client.mchsi.com) |
| 16:21.44 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 16:23.27 | leifmadsen | ya, the DTMF should "just work" |
| 16:23.31 | leifmadsen | 1.4.16 is OLD though |
| 16:23.45 | ryduh | Ok so I just changed my servers to us1. Changed my dtmf mode to rfc2833 and just dialed my cell phone |
| 16:24.18 | ManxPower-work | ryduh: Vitelity? |
| 16:24.36 | ManxPower-work | The Support Tab in your Vitelity portal will tell you which server to use. click on one of the sample configs |
| 16:24.49 | ryduh | ManxPower-work: still with voip.ms |
| 16:25.05 | ryduh | Here's the context that's run when my automated call is answered http://pastebin.com/d36a21c17 |
| 16:25.30 | ryduh | when I just called my cell phone, I heard the first dtmf tone, and nothing for the second two |
| 16:26.01 | p3nguin_ | What's the point of this exercise? |
| 16:26.45 | ryduh | p3nguin_: I need to dial a restaurant's phone number and get past their IVR to a real person and notify them that we have a new order for them |
| 16:27.32 | p3nguin_ | And they don't have any other DID without the IVR? |
| 16:28.19 | ryduh | I play a soundfile once they answer. Nope. We don't even have a DID w/o IVR on the old Panasonic system we have in this office ;/ |
| 16:28.57 | p3nguin_ | I'm going to try your dialplan on my system and see what I can hear. |
| 16:29.11 | ryduh | Thanks p3nguin_ |
| 16:30.20 | ryduh | I've tried different stuff with SendDTMF. I've used the SendDTMF(107). SendDTMF(1,1000) SendDTMF(0,1000) SendDTMF(7,1000) but my current dialplan is the only one that gave ANY results. |
| 16:30.47 | p3nguin_ | ryduh: How you are originating the call? |
| 16:31.07 | ryduh | A call file. I'll pastebin my script now |
| 16:31.23 | p3nguin_ | okay |
| 16:31.51 | ryduh | http://pastebin.com/d5289ce57 |
| 16:32.26 | ryduh | usage: ./call.sh 8058889999 |
| 16:32.41 | *** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
| 16:35.41 | *** part/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
| 16:36.23 | p3nguin_ | ryduh: I'm just using originate on the console, and it's only playing one tone just like you described. |
| 16:36.58 | ryduh | hmmm. which provider & what * version are you using? |
| 16:37.07 | *** join/#asterisk TimToady_ (n=moi@adsl248-94.kln.forthnet.gr) |
| 16:37.47 | *** join/#asterisk Athahira (n=chatzill@81.10.119.23) |
| 16:38.30 | ryduh | Is it just a misunderstanding of my own for how DTMF tones are sent or should that work? |
| 16:38.58 | p3nguin_ | ryduh: http://pastebin.com/d1cb7769d |
| 16:39.13 | p3nguin_ | ryduh: The highlighted line was the only one that I heard. |
| 16:39.35 | ryduh | p3nguin_: That's the one I hear as well |
| 16:39.37 | p3nguin_ | VoIP.ms, 1.4.24.1 |
| 16:39.55 | p3nguin_ | It should be working. |
| 16:40.08 | ryduh | RFC2833? |
| 16:41.28 | *** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
| 16:41.39 | p3nguin_ | yep |
| 16:43.00 | ryduh | Would anyone not using voip.ms mind testing this dialplan? http://pastebin.com/d36a21c17 |
| 16:45.50 | *** join/#asterisk Peaceful (n=Peaceful@70.102.57.178) |
| 16:46.01 | p3nguin_ | ryduh: Dialing an internal phone over SIP, I don't even get any DTMF at all. |
| 16:46.01 | ryduh | Vitelity's lowest deposit is $35. Any other voip termination providers out there with a low starting investment? |
| 16:46.40 | p3nguin_ | VoIP.ms uses Vitelity's network, so why even consider them at all? |
| 16:47.11 | ryduh | p3nguin_: good point. I won't. |
| 16:47.30 | p3nguin_ | Maybe they have some additional hardware of their own, but I don't know. |
| 16:47.31 | Peaceful | Can someone point me to which component of asterisk I should be looking at to block outbound caller id on specific calls going out to the pstn? A dialplan function/app? A dahdi setting? A sip setting? I'm happy to read up on it, I'm just not sure where to start... |
| 16:48.49 | ManxPower-work | Peaceful: that is protocol specific. |
| 16:49.18 | ManxPower-work | Peaceful: Are you trying to send calls out analog, T-1 CAS, PRI, SIP? |
| 16:49.41 | ryduh | Whoa. I just learned how close I live to Level3 |
| 16:50.22 | [TK]D-Fender | Peaceful: "core show function CALLERID" , "core show appliction setcallerpres" |
| 16:51.03 | p3nguin_ | ryduh: I tested dtmfmode = info, and get WARNING[1093]: chan_sip.c:13288 handle_response: Host '192.168.192.26' does not implement 'INFO' |
| 16:51.38 | ryduh | p3nguin_: Is that your other sip phone? |
| 16:51.50 | Peaceful | ManxPower-work: T-1 |
| 16:52.03 | p3nguin_ | ryduh: On the other hand, inband works perfectly and I heard all the tones. |
| 16:52.04 | ManxPower-work | Peaceful: you can't set outgoing callerid on a T-1. You can on a PRI |
| 16:52.06 | Peaceful | [TK]D-Fender: I'll read up on all of those |
| 16:52.27 | p3nguin_ | ryduh: I used originate SIP/201 extension s@ryder-test |
| 16:52.33 | ryduh | p3nguin_: I see the same thing. But for you does it work every time? |
| 16:52.46 | Peaceful | ManxPower-work: I'm a computer guy. Which ones are PRI's? I've got local trunks and LD trunks to the regular telco plugged into a digium card. |
| 16:53.31 | p3nguin_ | ryduh: Just tested it five times in a row and I hard all digits. I'll test through the ITSP now. |
| 16:54.17 | ryduh | p3nguin_: which server will you use? |
| 16:54.32 | p3nguin_ | ryduh: I only hear the first one that way. Let me make sure the peer is also using that mode. us2 |
| 16:58.59 | p3nguin_ | ryduh: Well, IAX2 apparently has only one say to send DTMF, so I can't force it to anything else. |
| 16:59.09 | *** join/#asterisk b14ck (n=comradeb@cpe-24-24-136-239.socal.res.rr.com) |
| 16:59.55 | ManxPower-work | Peaceful: A PRI is a specific form of T-1. The other kind is CAS |
| 17:00.48 | Peaceful | ManxPower-work: Ah, well we have PRIs then. The telco techs have called them that many times over the years and I've never heard of "CAS". |
| 17:00.52 | *** join/#asterisk sahafeez (n=sahafeez@68.167.234.130) |
| 17:01.13 | ManxPower-work | Peaceful: then it is up to your provider of they allow you to set or block your callerid |
| 17:02.45 | Peaceful | ManxPower-work: Well, ya, I just want to know how to block it from asterisk's side. Assuming they let me do that. Which I assume I'll figure out how to do by reading up on CALLERID and setcallerpres that [TK] mentioned. |
| 17:03.03 | ryduh | What is a decent voip provider with a low initial deposit? |
| 17:03.53 | ryduh | p3nguin_: Thank you for testing what you did. |
| 17:07.42 | leifmadsen | Unlimitel |
| 17:07.52 | thehar | snow snow snowwwwww |
| 17:19.11 | hardwire | no |
| 17:19.20 | hardwire | where are you thehar? |
| 17:19.41 | ManxPower-work | hardwire: he's one of those perverts that likes snow. |
| 17:21.11 | hardwire | I'm in Alaska.. so am I a fetish pervert? |
| 17:21.34 | ManxPower-work | hardwire: Alaska? That would he Hardcore Pervert. |
| 17:21.34 | thehar | Salt Lake city |
| 17:21.40 | hardwire | really? |
| 17:21.42 | thehar | sorry i was gazing outside my office window |
| 17:21.43 | hardwire | thehar: neat. |
| 17:21.46 | thehar | hardwire: yes |
| 17:21.58 | angryuser | ryduh, lots of providers propose you to deposit at least 10$ and then pay-as-you-go but dont be surprised being cut when 0 on balance |
| 17:22.22 | thehar | http://photos-h.ak.fbcdn.net/hphotos-ak-snc1/hs117.snc3/16433_298548180229_630620229_9578314_4472842_n.jpg << outside my office |
| 17:24.35 | ryduh | Has anyone had any experience with FlowRoute? |
| 17:25.48 | hardwire | ryduh: they design good websites! :) |
| 17:26.10 | hardwire | thats the limit of my experience with them |
| 17:26.43 | ryduh | hardwire: yes they do! haha. Their prices are kind of ridiculous though. Termination in California: $0.0098 per min |
| 17:26.55 | hardwire | rediculous low? |
| 17:27.31 | ryduh | hardwire: yes. Has anyone seen prices lower that that without some kind of wholesale contract? |
| 17:27.45 | ryduh | p3nguin_: DTMF worked fine on flowroute first time |
| 17:29.15 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 17:29.18 | p3nguin_ | ryduh: Interesting. |
| 17:29.56 | ManxPower-work | 0.98/cents/min is just a little lower than 1.2/cents/min that many people offer |
| 17:32.41 | ryduh | ManxPower-work: I think it's the extra 0 that throws me off. Kind of like like $x.99 idea |
| 17:41.44 | *** join/#asterisk superbeef (n=superbee@74.84.194.4) |
| 17:46.10 | *** join/#asterisk sahafeez (n=sahafeez@68.167.234.130) |
| 17:47.45 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 17:47.59 | hardwire | wonders how many wholesalers there are in this channel. |
| 17:51.33 | *** join/#asterisk btsteve (n=tstevens@12.44.147.40) |
| 17:51.38 | rene- | vitelity <=> teliax |
| 17:54.21 | *** join/#asterisk mumtazah1 (n=mumtazah@203.82.79.102) |
| 17:56.08 | ecrane | is a wholesaler |
| 18:01.36 | ecrane | Can anyone recommend some good SIP books and Asterisk books? (Obviously 'Asterisk: The Future of Telephony' is one of them...) I'm looking for a few books that explain the SIP protocol and Asterisk configs in plain english, peppered (but not saturated) with practical/real world examples. |
| 18:03.51 | jaytee | SIP Demystified by Gonzalo Camarillo |
| 18:07.03 | Qwell | SIP can't be explained in English. |
| 18:07.25 | Qwell | s/ in English// |
| 18:07.52 | jaytee | no, but it can be "demystified" |
| 18:09.25 | ecrane | Anyone read "Carrier Grade Voice Over IP (Daniel Collins)"? |
| 18:10.23 | ecrane | jaytee: Thanks I'll look for that SIP Demystified at the local bookstore and I see it on Amazon... |
| 18:10.33 | telnettech | SIP Demystified by Gonzalo Camarillo |
| 18:10.39 | telnettech | that is the best book on SIP |
| 18:10.55 | telnettech | there is a new Asterisk book out but forget the name |
| 18:11.05 | LemensTS | openSER book had some good sip information |
| 18:11.39 | *** join/#asterisk gardo (n=gardo@121.96.137.100) |
| 18:11.57 | *** join/#asterisk voipmonk (n=voipmonk@66.49.238.52) |
| 18:12.28 | ryduh | Why would monitor not recored the beginning of the call? |
| 18:13.25 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 18:13.26 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 18:17.10 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:17.52 | *** join/#asterisk umay (n=chris@174-16-31-61.hlrn.qwest.net) |
| 18:21.25 | kaldemar | nyt saa riittää, perkele. |
| 18:22.28 | kaldemar | oops, wrong window :D |
| 18:29.04 | Naikrovek | i wish i could fluidly switch between languages. also, i wish i knew two languages. |
| 18:29.21 | *** join/#asterisk buttons840 (n=buttons8@63.230.20.246) |
| 18:32.32 | buttons840 | In the AMI, OriginateResponse events. I'm having trouble with OriginateResponse events which have Response: Failure, and the reason is some number, like 0, 1, 3, 5 (some of the number we are getting). I can't find any information about what these number mean? |
| 18:33.02 | rene- | buttons840: use the source |
| 18:33.10 | rene- | as in read C files |
| 18:33.13 | rene- | it is there |
| 18:33.29 | rene- | look for states extension states |
| 18:33.38 | ManxPower-work | buttons840: in the time you have worked on this you could have read the source code. |
| 18:33.38 | rene- | use doxygen |
| 18:33.39 | buttons840 | I know that calling a Verizon cell phone which rings long with no answer, gives reason 0, and calling a Telco disconnected message gives reason 0 as well. |
| 18:33.51 | rene- | the online asterisk source code viewer |
| 18:34.34 | buttons840 | I don't know C, but I know most open source projects have some documentation. |
| 18:34.58 | buttons840 | which file would i read? |
| 18:35.51 | rene- | ok you should look somewhere in manager.c to search for the originateresponse | failure part, it should have a reference to extension states |
| 18:35.55 | rene- | just follow from there |
| 18:36.00 | rene- | dont read the C files directly |
| 18:36.04 | rene- | use doxygen |
| 18:36.08 | rene- | use google |
| 18:36.14 | Qwell | russellb: ping here! |
| 18:36.31 | russellb | WHAT |
| 18:36.34 | Qwell | <PROTECTED> |
| 18:36.38 | Qwell | can you do that? |
| 18:36.45 | Qwell | I can't. ChanServ is MIA. |
| 18:36.47 | *** join/#asterisk ChanServ (ChanServ@services.) |
| 18:36.47 | *** mode/#asterisk [+o ChanServ] by irc.freenode.net |
| 18:36.49 | russellb | done |
| 18:36.51 | Qwell | thanks |
| 18:36.53 | [TK]D-Fender | \o/ |
| 18:36.54 | Qwell | [TK]D-Fender: ^ |
| 18:39.08 | leifmadsen | russellb: wb |
| 18:39.11 | russellb | thanks |
| 18:39.26 | leifmadsen | have a good trip? |
| 18:39.39 | russellb | yes! |
| 18:39.43 | leifmadsen | awesome |
| 18:40.10 | buttons840 | rene-, http://www.asterisk.org/doxygen is this what your referring to? |
| 18:40.25 | rene- | yes |
| 18:40.38 | diatonic1 | I want to execute something in the dialplan if the extension placing the call is 201-370. Is there a way to do an execif([${CALLERID(num)} = 201-370], Set(SPYGROUP=monitor)) |
| 18:40.55 | diatonic1 | I don't think I can check if it is in a range like that |
| 18:41.28 | leifmadsen | diatonic1: there is possibly a dialplan function to accomplish that |
| 18:41.35 | leifmadsen | that'd be the most likely place to do that |
| 18:41.45 | leifmadsen | which would then probably return 1 or 0 |
| 18:42.13 | rene- | exwhere is doxygen |
| 18:42.31 | rene- | http://www.asterisk.org/doxygen is not the source code viewer |
| 18:43.52 | diatonic1 | leifmadsen: Do you know what function? |
| 18:43.59 | leifmadsen | diatonic1: if I did, I'd have said it |
| 18:44.03 | diatonic1 | lol |
| 18:44.43 | *** join/#asterisk ehsjoar (n=ehsjoar@98.245.155.132) |
| 18:45.19 | rene- | does anybody has a new url for the asterisk doxygen viewer? |
| 18:46.10 | rene- | buttons840: i dont know man it has been taken offline |
| 18:46.26 | rene- | that url you are referring is not what i was talking about |
| 18:46.52 | buttons840 | so there is no documentation |
| 18:47.13 | buttons840 | unless you consider the code documentation, in which case i will remove "undocumented" from my vocabulary |
| 18:47.16 | rene- | well it was there a couple months ago |
| 18:47.39 | rene- | let me search in my notes |
| 18:49.04 | leifmadsen | the asterisk.org site changed 2 weeks ago -- if you can't find an existing link to a document, or it is broken, then email webmaster@digium.com |
| 18:49.19 | Qwell | hmm |
| 18:49.28 | buttons840 | thanks rene- , although you should know that i've seen a few different mailing lists and forum posts about these originateresponse reasons, and in general i've never found an explination of originateresponse reasons |
| 18:49.30 | leifmadsen | Qwell: yay, another page missing? :) |
| 18:49.32 | Qwell | yeah, doxygen != documentation |
| 18:49.33 | Qwell | silliness. |
| 18:49.37 | leifmadsen | that too :) |
| 18:49.40 | rene- | there you go http://pastie.org/672062 |
| 18:49.44 | leifmadsen | doxygen is developer documentation |
| 18:49.54 | rene- | well the ami is not very well documented |
| 18:50.00 | buttons840 | Qwell, that's how i feel, but i am greatful for any help, even if it's not just laying the answer in my lap :) |
| 18:50.17 | rene- | they might be totally wrong but hey they work for me |
| 18:51.25 | *** join/#asterisk brad_mssw (n=brad@216.155.101.90) |
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| 18:52.03 | buttons840 | rene-, the real reason i'm scratching my head is, as i said, a telco message gives reason 0, and so does a verizon cell phone with does a ring no answer |
| 18:52.33 | rene- | hmm |
| 18:52.34 | buttons840 | seeing this i'm beginning to wonder if there is any way to determine why a call failed, and for what reason |
| 18:52.41 | rene- | are you sure its their ring? or your ring |
| 18:52.55 | LemensTS | i hear whitehouse.gov is going drupal...wow doing something free and open source thats not like us |
| 18:53.16 | buttons840 | rene-, i'm not sure what your asking? if your refering to a dummy ring, then yes, i think that has something to do with it |
| 18:53.36 | rene- | what is your carrier? |
| 18:53.49 | buttons840 | qwest ipld service, dsl lines, sip protocol |
| 18:54.08 | *** join/#asterisk denon (i=denon@sassinak.net) |
| 18:54.08 | *** mode/#asterisk [+o denon] by ChanServ |
| 18:54.28 | rene- | weird, they are for the most part reliable for me |
| 18:54.56 | buttons840 | they are reliable, but not playing nicely with asterisk and originateresponse reasons |
| 18:55.08 | *** join/#asterisk jpcansa (n=jpbenavi@201.237.157.34) |
| 18:55.16 | buttons840 | have you had experience with originateresponse reasons? |
| 18:55.36 | rene- | thats what i said, response reasons have been reliable for me |
| 18:55.52 | jpcansa | hi, any idea why VM is not picking up here? http://pastebin.com/m73968188 |
| 18:56.16 | rene- | buttons840 they seem to work for me |
| 18:56.21 | rene- | why do u need to get specific with failures? |
| 18:56.29 | rene- | if it is SIP |
| 18:56.39 | rene- | you might as well check the SIP codes you are getting |
| 18:56.43 | rene- | maybe with ethereal |
| 18:56.50 | rene- | and then look on the internet for the error codes |
| 18:58.10 | leifmadsen | jpcansa: because that is not the right formatting for the dialplan? |
| 18:58.37 | leifmadsen | jpcansa: what version are you using? that might have been valid in like 0.9 ... |
| 18:58.44 | buttons840 | so if i call the PSTN the sip codes will still indicate what the failure reason is, whether it be busy or a telco message? |
| 18:58.54 | jpcansa | @leifmadsen: version of what? |
| 18:59.01 | leifmadsen | jpcansa: the software you are using |
| 18:59.15 | leifmadsen | I presume you're using asterisk |
| 18:59.19 | jpcansa | @leifmadsen: this is asterisk 1.4 |
| 18:59.30 | leifmadsen | jpcansa: ya, that formatting for the dialplan is not right |
| 18:59.43 | jpcansa | what is wrong? |
| 18:59.51 | leifmadsen | jpcansa: and priority jumping isn't the right way either |
| 18:59.57 | leifmadsen | jpcansa: lots... I'll see if I can fix it up... |
| 19:00.29 | jpcansa | if i remove the HINT line it works, but oviously i have no hint to my blf |
| 19:00.56 | *** join/#asterisk Skeeter- (i=Skeeter-@190-141.cgocable.ca) |
| 19:00.59 | leifmadsen | jpcansa: http://pastebin.com/m1d276c2c |
| 19:01.06 | leifmadsen | jpcansa: that's because the hint shouldn't be where you put it |
| 19:01.20 | Skeeter- | i recorded a message for my VM, i wanna know where is it located on my server |
| 19:01.34 | leifmadsen | it should be in the [subscriptions] context, or wherever you have subscriptions setup in your sip.conf |
| 19:01.34 | ManxPower-work | Skeeter-: /var/spool/asterisk/voicemail |
| 19:02.09 | jpcansa | i see |
| 19:02.19 | leifmadsen | jpcansa: the 'hint' priority is totally independent of actually dialing something |
| 19:02.44 | leifmadsen | also, using '102' is incorrect, because +101 priority jumping is not the method to use for busy detection |
| 19:03.09 | jpcansa | @leifmadsen: what should i use instead of 102 |
| 19:03.11 | jpcansa | ? |
| 19:03.18 | leifmadsen | ${DIALSTATUS} variable |
| 19:03.31 | leifmadsen | and parse on the value returned, such as BUSY |
| 19:03.45 | leifmadsen | hence what I did in the Voicemail() example I showed, but you could have done a GotoIf() instead |
| 19:03.56 | leifmadsen | GotoIf($[${DIALSTATUS} = BUSY]?busy_status) |
| 19:05.09 | jpcansa | ok let me try that |
| 19:05.18 | *** part/#asterisk ehsjoar (n=ehsjoar@98.245.155.132) |
| 19:05.18 | buttons840 | rene-, you say the originateresponse reasons work for you; are you calling over the pstn to various phones with various providers, or are you on a smaller and more controlled network? |
| 19:05.21 | jpcansa | thanks |
| 19:05.23 | leifmadsen | jpcansa: http://astbook.asteriskdocs.org |
| 19:06.52 | jpcansa | cool |
| 19:07.37 | rene- | yes calling to the pstn to varios phones |
| 19:07.41 | rene- | various |
| 19:08.27 | ryduh | p3nguin_: Do you have any problems when DTMF tones are pressed real fast when using INFO? |
| 19:09.06 | p3nguin_ | ryduh: My phones apparently don't support info mode, so yeah, big problems. |
| 19:10.34 | Skeeter- | ManxPower-work: thank you |
| 19:10.52 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 19:10.56 | ryduh | p3nguin_: what about when pressing keys real fast with RFC2833? |
| 19:11.33 | p3nguin_ | ryduh: Not sure. I just pressed the keys normally. |
| 19:11.54 | ryduh | p3nguin_: how fast is normally? |
| 19:12.12 | p3nguin_ | maybe half a second between presses. |
| 19:13.39 | p3nguin_ | ryduh: I tried SendDTMF(107) and I only heard the 1. |
| 19:14.05 | ManxPower-work | If you only hear a 1 when sending 107 then I suspect you are doing INBAND on a COMPRESSED codec. |
| 19:14.07 | p3nguin_ | ryduh: It's weird that I can call other IVRs and my key presses are accepted correctly. |
| 19:14.40 | ManxPower-work | Too bad you're not using a PRI (or even analog or CAS T-1). You could manually set the outgoing doneduration |
| 19:14.48 | buttons840 | rene-, would you be able to call 2 numbers for me and let me know what responses you get with your configuration? |
| 19:16.12 | p3nguin_ | manxpower-work: I was originally using rfc2833 and ulaw, originating from the console to an internal SIP phone. Only 1 tone was heard. I then switched to inband and originated to the same SIP phone and heard all the tones. |
| 19:16.30 | buttons840 | so i can determine if this is unique to our config, or the same accross all configurations? |
| 19:17.06 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:17.09 | ManxPower-work | For the most part if you have DTMF problems with a carrier your only options are 1) use inband DTMF w/ulaw or alaw or 2) get a diffferent carrier. Not anything else you can do. |
| 19:17.13 | p3nguin_ | manxpower-work: But when originating out to an external number via IAX2, I still only hear the first tone. |
| 19:17.49 | p3nguin_ | Since there's apparently no way to change dtmfmode on IAX2, there was no further testing that I could do. |
| 19:18.21 | ManxPower-work | p3nguin_: Chances are your CARRIER has DTMF with THEIR carrier. |
| 19:18.29 | ManxPower-work | so---there is nothing you can do. |
| 19:18.52 | ManxPower-work | since if you are using IAX2 you are guarnteed for the dtmf to arrive at the far end. |
| 19:19.05 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 19:20.39 | *** join/#asterisk ChUbB (n=IceChat7@62-31-213-230.cable.ubr12.aztw.blueyonder.co.uk) |
| 19:20.47 | ryduh | ManxPower-work: So I'm using INBAND w/ ulaw on voip.ms. I call my DID and try to dial ext 630. I see this http://pastebin.com/d797e95ef |
| 19:21.26 | leifmadsen | diatonic1: why not just do something like: $[${CALLERID(num)} > 201 & ${CALLERID(num)} < 301] |
| 19:22.17 | *** join/#asterisk bmoraca (n=chatzill@66.242.174.254) |
| 19:23.01 | *** join/#asterisk TSM (n=the_soft@87-194-32-212.bethere.co.uk) |
| 19:23.28 | ManxPower-work | ryduh: CONFIRM the mode of the call using the sip show channel command |
| 19:24.53 | *** join/#asterisk freckle (n=chatzill@195.74.96.123) |
| 19:26.08 | freckle | have just upgraded to 1.6 and the console is telling me Dial command does not accept S(0), what has replaced it? |
| 19:27.25 | ryduh | ManxPower-work: It's confirmed http://pastebin.com/da2bc4e |
| 19:27.33 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 19:27.57 | p3nguin_ | ryduh: That's the incoming... how about the outgoing? |
| 19:29.06 | ryduh | p3nguin_: Right now, I'm just trying to get incoming dtmf to work well. |
| 19:29.26 | *** join/#asterisk dysinger (n=tim@cpe-75-85-135-191.hawaii.res.rr.com) |
| 19:29.26 | p3nguin_ | ryduh: From your cell phone? |
| 19:29.38 | ryduh | p3nguin_: a landline |
| 19:29.43 | p3nguin_ | okay |
| 19:29.49 | buttons840 | https://issues.asterisk.org/view.php?id=15104 this bug appears very similar to the one i'm having. i receive response 0 (frames.h says this means failure apparently), from a verizon cell phone with does ring |
| 19:30.47 | buttons840 | if i upgrade from asterisk 1.4.21.2 will the latest version break some thing, or are the version backwards compatible? |
| 19:31.27 | ryduh | buttons840: that's a hard question to answer. take a look at UPGRADE-*.txt |
| 19:32.23 | rene- | buttons840, maybe you can apply the patch |
| 19:32.31 | rene- | and see it the responses you get make more sense |
| 19:32.36 | rene- | than what you are currently getting |
| 19:32.47 | buttons840 | do you have the latest asterisk rene? |
| 19:32.56 | ManxPower-work | ~answers |
| 19:32.57 | infobot | it has been said that answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 19:32.58 | rene- | no |
| 19:33.16 | rene- | i have systems with 1.4.21~1.4.24 |
| 19:33.42 | diatonic1 | leifmadsen: Ended up doing this and it works: ExecIf($[${CALLERID(num)} >= 201 & ${CALLERID(num)} <= 370],Set,SPYGROUP=cati) |
| 19:33.57 | diatonic1 | leifmadsen: The I saw your post :) |
| 19:33.59 | buttons840 | i've narrowed it down to reason 0's that appear to be the real problem, if it does mean failure as it says in frames.h (apparantly), then 0 is correct for a telco disconneted message, but why a verizon cell phone would give the same thing? |
| 19:34.00 | leifmadsen | diatonic1: :) |
| 19:35.03 | ryduh | bbl lunch :) |
| 19:35.03 | ManxPower-work | buttons840: why don't you just originate to a Local/ channel and then let Dial and HANGUPCAUSE help you deal with disconnects? |
| 19:35.12 | diatonic1 | I need to use a real IRC client instead of Pidgin. This is kinda painful |
| 19:35.41 | buttons840 | ManxPower-work, i am originating to a local channel so i can use the local routing, but what is this hangup cause you speak of? |
| 19:36.46 | buttons840 | guess i'll look into that |
| 19:36.46 | ManxPower-work | buttons840: Dial sets 2 variables: DIALSTATUS (very generic) and HANGUPCAUSE (Q.931 cause codes) |
| 19:36.46 | ManxPower-work | For SIP, Asterisk maps the SIP responses to Q.931, there's an RFC for that. |
| 19:36.58 | buttons840 | ManxPower-work, and in a case where originate calls a disconnected phone, or a ring no answer, these variables would still be set and usable? |
| 19:37.15 | ManxPower-work | buttons840: it's not Originate anymore, it's Dial |
| 19:37.27 | buttons840 | because it goes through the local? i see |
| 19:37.28 | ManxPower-work | So BOTH legs of your call are in your dialplan, right? |
| 19:37.50 | buttons840 | define leg? |
| 19:38.59 | *** join/#asterisk telnettech (n=bsimpson@office.callcopy.com) |
| 19:39.06 | buttons840 | a script reads a text file, originates through the ami to a local route, then the script continues passively watching the call through the ami and records it progress as significant events occur, the logic of the call is all dialplan |
| 19:39.08 | ManxPower-work | buttons840: All calls have two legs, meeting at the Asterisk server. |
| 19:39.28 | *** part/#asterisk telnettech (n=bsimpson@office.callcopy.com) |
| 19:39.39 | ManxPower-work | OK, maybe not ALL calls. |
| 19:40.11 | ManxPower-work | buttons840: so you have a call going out to a number somewhere. Where's the other end of that leg? The dialplan? |
| 19:40.12 | *** join/#asterisk telnettech (n=bsimpson@office.callcopy.com) |
| 19:40.15 | buttons840 | so in the case that both legs were in the dialplan, what would that mean. |
| 19:40.32 | ManxPower-work | buttons840: that would mean you should get good HANGUPCAUSES |
| 19:40.47 | *** part/#asterisk telnettech (n=bsimpson@office.callcopy.com) |
| 19:41.21 | *** join/#asterisk telnettech (n=bsimpson@office.callcopy.com) |
| 19:41.41 | buttons840 | i originate through ami to local route, which then uses dial, the call proceeds then ends... does this account for one leg? what would the other leg be then? |
| 19:42.02 | telnettech | diatonic1: you sure do.....i tried with pidgin and couldnt keep up with a heavy conversation load |
| 19:44.17 | buttons840 | i guess i'm asking if a "leg" is a conceptual thing, or if it's a literal part of the code (ie, this code is leg one, and this code is leg 2) |
| 19:45.59 | ManxPower-work | buttons840: conceptual |
| 19:49.04 | Skeeter- | is there any other good monitoring software, like ASTassistant out there?? |
| 19:50.27 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 19:50.41 | bmoraca | never heard of astassistant, but i've always liked isymphony |
| 19:50.46 | *** join/#asterisk ttl- (n=patrick@94-224-78-192.access.telenet.be) |
| 19:54.13 | *** join/#asterisk Tim_Toady (n=moi@adsl13-36.kln.forthnet.gr) |
| 19:55.07 | *** join/#asterisk _brent_ (n=_brent_@166.70.142.225) |
| 19:55.34 | p3nguin_ | I want me failover * system to first try to direct the call to the primary * system just in case there was some type of failure at the provider, but then still accept the call if the primary is really unavailable. If I use switch in the incoming context, followed by the rest of the dialplan, will I be able to achieve what I'm after, or should I use Dial(IAX2/primary-server/${EXTEN}) instead? |
| 19:56.32 | beek | 1.6.0 AMI -- Voicemailuserslist Is there any way to send this beastie the CONTEXT you want it to look in for email? |
| 19:57.27 | beek | I can do the list with no problem but on the console I'm getting: ERROR[14072]: app_voicemail.c:1487 messagecount: Couldn't find mailbox 309 in context default |
| 19:57.35 | beek | 309 is in another context. |
| 19:57.59 | beek | I'm getting: -1 new messages |
| 19:58.06 | beek | Or is this a bug? |
| 19:58.14 | ManxPower-work | did you try using 309@thecontectyouwant? |
| 19:58.22 | *** part/#asterisk _brent_ (n=_brent_@166.70.142.225) |
| 19:58.37 | beek | Hi ManxPower-work ... I'm trying to get the entire list with the number of messages for each mailbox. |
| 19:58.58 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 19:59.04 | beek | I'm getting the entire list, but number of new messages = -1 even though I'm getting the correct context back in the response. |
| 19:59.18 | beek | It's looking like a bug since it knows the context, it should know to look there for new messages. |
| 19:59.21 | ManxPower-work | beek: might be a bug then |
| 19:59.55 | beek | It might be. The console voicemail show users gets it right. |
| 20:11.24 | gandhijee | anyone here happen to have the SIP firmware for a cisco 7970? |
| 20:13.51 | superbeef | Katty: make this for lunch http://www.notmartha.org/archives/2009/10/27/meat-hand/ |
| 20:15.42 | *** join/#asterisk tzafrir (n=tzafrir@bzq-218-155-170.cablep.bezeqint.net) |
| 20:17.53 | *** join/#asterisk BlackSLik (n=jamesmar@64.191.115.29) |
| 20:23.50 | beek | WTF? This just gets better. |
| 20:24.23 | ManxPower-work | beek: what is the value of HANGUPCAUSE |
| 20:24.52 | beek | ManxPower-work: I'm not sure what you mean. |
| 20:25.33 | ManxPower-work | in your dialplan as the priority after Dial put Noop(HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is ${DIALSTATUS}) This is basic dialplan troubleshiiting |
| 20:25.39 | ManxPower-work | and shOOting too. |
| 20:25.46 | BlackSLik | any know any good DID provider with iax and sip capability |
| 20:26.10 | beek | Ummm.... there's no dialplan involved here. I'm using AMI Action: VoicemailUsersList to get a list of users. That report is wrong. Everything else is right. |
| 20:26.32 | beek | CLI reports "no mailbox found in context 'default'" |
| 20:26.44 | beek | IMAP has no voicemail context. |
| 20:26.44 | ManxPower-work | beek: sorry, I thought that issue was CLOSED when we relaized it's a bug. |
| 20:27.11 | beek | No, I'm just muddling through the source code looking at it right now to see why this thing thinks it should worry about a context. |
| 20:27.17 | *** join/#asterisk Skarmeth (n=Skarmeth@201.57.179.27) |
| 20:28.00 | beek | I'd like to submit a patch with the bug report. |
| 20:35.26 | *** join/#asterisk jeff (i=jeff@unaffiliated/jeff) |
| 20:38.06 | Skeeter- | with polycoms, can we see the called id of the 2nd line with using the 1st line |
| 20:38.43 | ManxPower-work | Skeeter-: what is normally on the first line and on the 2nd line? |
| 20:40.25 | Skeeter- | well, when its a call on the 1st line, i can see the caller ID. while speaking ono the 1st line, i can see the little light of the 2nd line annoncing me that there is someone on the 2nd line, but i cant see the called ID of that 2nd line on the little LCD on the right |
| 20:40.44 | Skeeter- | what do u mean by normaly |
| 20:40.56 | *** join/#asterisk syuroff (n=syuroff@216.47.253.194) |
| 20:41.48 | ManxPower-work | Skeeter-: nevermind. |
| 20:42.30 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 20:42.31 | Skeeter- | ok.. |
| 20:43.53 | *** join/#asterisk diatonic1 (n=diatonic@mail.clearwater-research.com) |
| 20:47.13 | *** join/#asterisk sfr33man (n=sfreeman@64.183.147.98) |
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| 20:58.02 | p3nguin_ | blackslik: VoIP.ms has SIP and IAX2 support, and they are very inexpensive. |
| 21:04.28 | *** join/#asterisk sahafeez (n=sahafeez@68.167.234.130) |
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| 21:21.36 | *** part/#asterisk grey-monkey (n=ericshel@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 21:27.44 | jblack | linode is down |
| 21:28.27 | ryduh | I'm calling my DID at voip.ms and then try to dial an extension 630 I get this http://pastebin.com/d3d12d7bc WTF? |
| 21:29.15 | TSM | dtmf mode looks wrong |
| 21:29.19 | trogs | jblack: yeah, there was some update they did and they're now rebooting a bunch of hosts. |
| 21:30.09 | ManxPower-work | ryduh: I don't see a sip show channel for that call |
| 21:30.18 | *** join/#asterisk Chodorenko (n=chodoren@ext.one.by) |
| 21:30.46 | ManxPower-work | ryduh: maybe they are still running 1.2 and you need the rfc compensate option |
| 21:32.29 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:35.37 | ryduh | ManxPower-work: http://pastebin.com/d402ab086 I added the rfc2833compensate option and it did not help :( |
| 21:35.41 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 21:36.25 | TSM | ryduh: look at your dtmf mode, its still set as inband |
| 21:37.49 | Katty | ohmaniamsotireddd |
| 21:38.07 | ryduh | Whoops. Setting that option wouldn't do much then would it lol. I do have it set on inband. |
| 21:39.10 | *** join/#asterisk sahafeez_ (n=sahafeez@65-119-47-100.dia.static.qwest.net) |
| 21:39.25 | ryduh | I've tried RFC2833 and INFO and I'm currently on inband. |
| 21:39.29 | [TK]D-Fender | Katty: we can tell... you didn't even have the energy to target the biggest key on your keyboard in between words ;) |
| 21:40.13 | ryduh | [TK]D-Fender: you could make an argument that it took *more* effort to not hit the biggest key |
| 21:40.25 | Katty | yeahnotreally |
| 21:40.37 | Katty | thespacebarisjustirratingactually |
| 21:41.17 | ryduh | hmmmimgoingtotryitoutforthisline.itsnottoobadactuallyalthoughtitdoesfeelweird |
| 21:41.24 | Katty | :P |
| 21:41.27 | [TK]D-Fender | Katty: IcAnThInKoFaFeWtHiNgStHaReWoRsE |
| 21:41.27 | jblack | whyarewedoingthis |
| 21:41.34 | Katty | argh |
| 21:41.35 | Katty | stop that |
| 21:41.43 | *** join/#asterisk sahafeez__ (n=sahafeez@68.167.234.130) |
| 21:41.43 | jblack | youstartedit |
| 21:41.52 | Katty | WELLI"MGOINGTOFINISHITTOO |
| 21:42.00 | jblack | you shit too? |
| 21:42.04 | Katty | yes. |
| 21:42.13 | dlynes | Does anyone know if there's a trick to getting vitelity to work with Asterisk 1.6.1.8? |
| 21:42.20 | Katty | my, that induced giggling. |
| 21:42.24 | jblack | wow. Does it stink? |
| 21:42.33 | Katty | perhaps i have newfound energies! |
| 21:42.47 | jblack | vision blackinging |
| 21:42.52 | Katty | :< |
| 21:43.03 | ryduh | ManxPower-work: I'm having my problems with voip.ms. I tried using flowroute and it seemed to work immediately. |
| 21:43.30 | jblack | I walked 4.5 miles today. |
| 21:43.33 | Katty | :> |
| 21:43.36 | ryduh | Is there anyway to forward a DID to another VoIP provider and not get caught up in the first voip provider's DTMF? |
| 21:43.39 | Katty | did your car break down? |
| 21:43.46 | jblack | Nope. |
| 21:43.49 | Katty | excellent. |
| 21:43.53 | Katty | how do you feel? |
| 21:44.25 | jblack | fun-key |
| 21:44.28 | Katty | hmmmmmm. |
| 21:44.32 | Katty | perhaps you need dinner. |
| 21:44.40 | ryduh | I'm not looking to port it to another voip provider yet because we're on a time-sensative deadline |
| 21:44.54 | jblack | I did 0.8 and had coffee, then 1.6 and had chinese, then 2.1 and had sugar free koolaid |
| 21:45.12 | Katty | chinese? |
| 21:45.18 | Katty | tell me it wasn't takeout |
| 21:45.31 | jblack | Don't be silly! |
| 21:45.34 | jblack | All you can eat buffet. |
| 21:45.38 | Katty | facepalm |
| 21:45.41 | ryduh | lol |
| 21:46.08 | jblack | it's my new rule. I'm allowed chinese if I'm willing to walk over 2 miles to get there. |
| 21:46.23 | Corydon76-dig | Katty: http://www.flickr.com/photos/matthewdillon/4045823110/in/set-72157622657249246/ |
| 21:46.37 | Katty | jblack: http://www.youtube.com/watch?v=13eDfrMgFQM <- i dedicate this to you |
| 21:46.50 | TSM | so its the new pre-eating exercise thats the order of play |
| 21:47.00 | Katty | jblack: see 0:33 |
| 21:47.03 | jblack | my bg 45 minutes after eating was only 160. |
| 21:47.19 | Katty | Corydon76-dig: woahhh |
| 21:47.29 | Katty | Corydon76-dig: i gotta get me some of those decorations |
| 21:48.11 | jblack | rotf |
| 21:48.47 | jblack | Katty: Hey, cmon. I walked 2 miles to get there, 2 miles to get back, and my bg was ok! |
| 21:49.05 | *** join/#asterisk coppice (n=chatzill@host86-139-183-205.range86-139.btcentralplus.com) |
| 21:49.37 | Katty | jblack: :>>>>>>>>>> |
| 21:49.39 | Katty | no. more like.. |
| 21:49.45 | Katty | jblack: :>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> |
| 21:50.03 | jblack | That was nearly an hour and half of walking. And 160 is a perfectly reasonable level. |
| 21:50.10 | jblack | What does :> mean anyways? |
| 21:50.29 | Katty | umm like... |
| 21:50.32 | Katty | a very happy excited smile |
| 21:51.13 | jblack | Oh, ok |
| 21:52.06 | coppice | I think it means . since . is less than : |
| 21:52.07 | ManxPower-work | 8-) <-- smile with eyeglasses. @*@ <-- Princess Leia Tribble |
| 21:52.55 | ryduh | When I port a DID to a new provider and it takes 2-4 weeks to switch, what happens as it switches? To switch I would need to have a replacement system set up on the receiving provider corrext? |
| 21:53.09 | [TK]D-Fender | ($)($) <- Pamela Anderson |
| 21:53.27 | ManxPower-work | ryduh: when it ports, calls will simply go to the new provider. |
| 21:53.39 | jblack | <PROTECTED> |
| 21:53.45 | Katty | http://icanhascheezburger.files.wordpress.com/2008/07/funny-pictures-kitten-is-excited-about-bacon.jpg <- :> |
| 21:54.04 | implicit | ryduh, which carrier |
| 21:54.15 | jblack | dat kitty iz mee! |
| 21:54.27 | ManxPower-work | ryduh: make sure you have ALL YOUR paperwork correct. |
| 21:54.47 | ryduh | implicit: voip.ms to someone else? not sure yet. I'm having DTMF problems with them |
| 21:54.50 | Katty | http://icanhascheezburger.files.wordpress.com/2009/06/funny-pictures-cat-is-excited-about-ribbons.jpg <- :> |
| 21:55.04 | implicit | ryduh: switching to someone like flowroute? :) |
| 21:55.22 | ryduh | implicit: possbily. do you work there? |
| 21:55.28 | implicit | i'm a founder |
| 21:55.39 | implicit | yep |
| 21:56.38 | ryduh | implicit: awesome! your site is well designed. although I did run into an error when signing up. When I pushed submit, the page reloaded with no errors and all blank fields. tried it again and it worked |
| 21:56.50 | implicit | thanks |
| 21:56.58 | implicit | hm, let me have someone look into that |
| 21:57.07 | *** join/#asterisk manxpower (n=ewieling@24.42.221.26) |
| 21:57.16 | ryduh | I can give you my IP to look in the logs if you wish |
| 21:57.30 | *** part/#asterisk ManxPower-work (n=ewieling@24.42.221.26) |
| 21:57.31 | implicit | let me see if that's necessary, we can take it into private chat if you want too |
| 21:57.39 | implicit | was that today? |
| 21:57.44 | Katty | http://www.youtube.com/watch?v=rUwb4RmvVyM <- also :> |
| 21:58.05 | ryduh | yep. actually nvm. I think the IP is available through irc info |
| 21:58.17 | ryduh | nvm on sending you the ip* |
| 21:58.24 | jblack | katty: I can outdo that. |
| 21:58.26 | implicit | thanks |
| 21:58.31 | implicit | I have some of our web guys looking into it now |
| 21:58.55 | jblack | Katty: http://www.youtube.com/watch?v=cJOZp2ZftCw |
| 21:59.23 | jblack | If you don't enjoy that one, you're dead inside, and possibly outside too |
| 21:59.25 | Katty | :>>> |
| 21:59.26 | implicit | ryduh: I guess they know about it and there is a reworked signup process that's been finished. They just haven't deployed it yet |
| 21:59.28 | implicit | sorry for the trouble |
| 21:59.33 | Katty | grooves |
| 21:59.38 | implicit | It was happening when the captcha was wrong i guess |
| 22:00.05 | Katty | alright, i'm going home. |
| 22:07.09 | *** join/#asterisk hexbomber (n=hexbombe@S0106004854811dff.wp.shawcable.net) |
| 22:07.42 | hexbomber | I installed asterisk from the repos, and it ran fine, until I rebooted my computer, now I can't get it to start... |
| 22:07.54 | Qwell | hexbomber: "the repos"? |
| 22:08.13 | hexbomber | apt-get install asterisk from debian based systems. |
| 22:08.22 | TSM | deb :( |
| 22:08.52 | TSM | check the * logs |
| 22:09.06 | hexbomber | I have... nothing out of the ordinary. |
| 22:10.50 | Athahira | hello all |
| 22:11.06 | Athahira | can i ask about installing chan_ss7 here ? |
| 22:13.26 | Athahira | make: *** [l4isup.o] Error 1 still can't install successfully :-( |
| 22:16.38 | ecrane | I'm having trouble getting x-lite (2.0, linux) to REGISTER w. asterisk. I have my x-lite configured as the book suggests, with a 'secret' and have the secret in my sip.conf. Did sip enable debug. It looks like x-lite is NOT sending the secret over. Any ideas on how to get x-lite to send the secret? |
| 22:18.45 | *** join/#asterisk darkadan (n=darkadan@190.240.37.84) |
| 22:18.48 | darkadan | HI |
| 22:19.04 | darkadan | problem DBdel: Error deleting key from database |
| 22:19.07 | darkadan | DBdel: Error deleting key from database |
| 22:19.07 | ecrane | HI DARKADAN! |
| 22:19.54 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 22:20.04 | darkadan | tengo este problema en los log de asterisk |
| 22:20.09 | darkadan | I have this problem in the log of asterisk |
| 22:20.40 | darkadan | is an issue in the Mennecke starters, there some solution |
| 22:22.35 | darkadan | please... |
| 22:22.56 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 22:23.08 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 22:26.10 | *** join/#asterisk metfan2007 (n=metfan20@201.103.36.143) |
| 22:27.58 | IBC_jkenney | I have a problem running asteirsk 1.6.6 on ubuntu 9.04 and its forking and creating tons of child processes |
| 22:28.03 | IBC_jkenney | utnil it runs out of ram |
| 22:28.36 | Qwell | No such version exists. Double-check what you're running.. |
| 22:28.53 | IBC_jkenney | Asterisk 1.6.1.6 |
| 22:29.00 | IBC_jkenney | sorry |
| 22:32.09 | *** part/#asterisk darkadan (n=darkadan@190.240.37.84) |
| 22:32.52 | Athahira | is there a libss7 works with asterisk 1.4 ? |
| 22:33.29 | IBC_jkenney | So what i'm having a problem with is asterisk 1.6.1.6 is forking out a ton of child processes and eating up all the ram in the server 8Gigs |
| 22:35.29 | *** join/#asterisk [netman] (n=netman@249.Red-88-8-164.dynamicIP.rima-tde.net) |
| 22:35.55 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 22:35.57 | *** join/#asterisk albertoandrade (n=albertoa@189.101.111.35) |
| 22:36.29 | manxpower | IBC_jkenney: chances are you are seeing THREADS not processes. Obviously a problem, but at least you know a little more about it |
| 22:37.33 | IBC_jkenney | how do i fix it |
| 22:37.34 | IBC_jkenney | ? |
| 22:38.07 | manxpower | IBC_jkenney: no idea. Maybe you have runaway AGIs? |
| 22:38.47 | IBC_jkenney | no not that i am aware of |
| 22:39.06 | Qwell | What are you doing that causes it? |
| 22:39.12 | IBC_jkenney | nothing |
| 22:39.15 | IBC_jkenney | it runs |
| 22:39.21 | IBC_jkenney | threw out the day |
| 22:39.26 | ChannelZ | ps ax |grep asterisk is showing multiplies? |
| 22:39.52 | IBC_jkenney | 9883 ? Ssl 0:06 asterisk |
| 22:39.52 | IBC_jkenney | 10248 pts/22 S+ 0:00 grep asterisk |
| 22:40.09 | jblack | http://www.youtube.com/watch?v=_vE9Zq7Gs-U&feature=rec-LGOUT-exp_stronger_r2-HM |
| 22:41.23 | ChannelZ | hmm so what are the child processes you are seeing then? |
| 22:42.01 | *** part/#asterisk bsaxon (n=bsaxon@12.68.234.174) |
| 22:42.10 | *** join/#asterisk bsaxon (n=bsaxon@12.68.234.174) |
| 22:42.46 | *** part/#asterisk bsaxon (n=bsaxon@12.68.234.174) |
| 22:43.23 | IBC_jkenney | all asterisk |
| 22:43.52 | IBC_jkenney | each taking up 621 |
| 22:43.53 | Qwell | Where are you seeing child processes? |
| 22:43.53 | IBC_jkenney | megs |
| 22:44.00 | ChannelZ | no.. I mean where are you seeing them because they didn't just sho up in your ps |
| 22:44.01 | IBC_jkenney | ntio |
| 22:44.24 | Qwell | Is it causing problems? |
| 22:44.26 | ChannelZ | ??? |
| 22:44.44 | ChannelZ | what is ntio |
| 22:44.45 | IBC_jkenney | Yes |
| 22:45.05 | IBC_jkenney | the box is running out of ram |
| 22:45.05 | Qwell | What problems is it causing? |
| 22:45.05 | IBC_jkenney | and crashing |
| 22:45.09 | Qwell | The box is crashing? |
| 22:45.10 | ChannelZ | is this WIndows? |
| 22:45.16 | ChannelZ | er no.. nevermind |
| 22:46.41 | IBC_jkenney | umm no |
| 22:46.54 | Qwell | What specific problems are you seeing? |
| 22:46.58 | IBC_jkenney | i have a box running 1.4 with less resources that is not giving me this problem |
| 22:47.53 | *** join/#asterisk Katty (n=Angela@adsl-70-253-172-140.dsl.stlsmo.swbell.net) |
| 22:48.11 | Katty | hi (= |
| 22:51.41 | jblack | I wish |
| 22:52.19 | Qwell | IBC_jkenney: Good luck with the not answering questions approach. |
| 22:52.21 | Qwell | runs off |
| 22:53.58 | *** join/#asterisk slinksh0t (n=slinksh0@c-76-108-31-194.hsd1.fl.comcast.net) |
| 22:57.13 | *** join/#asterisk SaiSoma (n=SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |
| 22:59.16 | IBC_jkenney | http://pastebin.com/d4467084a |
| 22:59.20 | IBC_jkenney | sorry it took so long |
| 22:59.25 | IBC_jkenney | i had to go handle soething |
| 22:59.27 | IBC_jkenney | something |
| 23:00.43 | Qwell | That didn't answer my question.. |
| 23:01.47 | Katty | hugs jblack |
| 23:01.57 | Katty | jblack: i'm having problems finishing my dinner lately :< |
| 23:02.09 | IBC_jkenney | Qwell could you please ask it differently |
| 23:02.18 | IBC_jkenney | maybee i'm not understanding it |
| 23:03.03 | hardwire | Katty: too many hugs. |
| 23:03.45 | ecrane | IBC_jkenney: what is "ntio"? Why did you write that earlier? |
| 23:03.54 | Katty | hugs hardwire |
| 23:04.00 | Katty | hardwire: no, not too many hugs...to many veggies. |
| 23:04.13 | Katty | hardwire: 2 cups of them. |
| 23:04.21 | ryduh | thank god for flowroute sending my dtmf tones correctly |
| 23:04.40 | hardwire | Katty: 2 smooshed cups? |
| 23:04.48 | Katty | well. mostly smooshy. |
| 23:04.51 | hardwire | steamed? |
| 23:04.57 | Katty | sauted mushrooms, broccoli, peas, carrots, and celery |
| 23:05.03 | hardwire | nom |
| 23:05.11 | hardwire | now I want stroganoff |
| 23:05.12 | Katty | yes, it's very nom. |
| 23:05.19 | Katty | but i cannot seem to nom it all. |
| 23:05.27 | hardwire | tried snorting? |
| 23:05.37 | Katty | especially with the tuna and bread crumbs in there |
| 23:05.45 | hardwire | that's too much food! |
| 23:05.54 | Katty | but i have to get my vitamins! |
| 23:05.59 | Katty | :< |
| 23:06.07 | [TK]D-Fender | Katty: ... |
| 23:06.19 | Katty | [TK]D-Fender: you hush |
| 23:06.22 | Katty | [TK]D-Fender: before you get started. |
| 23:06.24 | [TK]D-Fender | Katty if you ate everything they told you you had to you'd end up 800 lbs |
| 23:06.48 | [TK]D-Fender | Katty: And have to eat something ELSE "special" to go down. |
| 23:07.40 | jblack | Katty: Why so? |
| 23:07.56 | kyleh | Hey anyone use festival with asterisk? |
| 23:08.14 | Katty | jblack: because i've not been eating enough vegetables. |
| 23:08.23 | jblack | That's easy to do |
| 23:08.31 | Katty | besides drinking them? |
| 23:08.39 | [TK]D-Fender | Katty: Warning about vegetables.... you are what you eat ;) |
| 23:09.10 | jblack | Try spicing them up just a little? A pinch of salt goes a long way |
| 23:09.10 | Katty | i have noodly appendages. |
| 23:09.14 | Katty | jblack: oh, no, it's not a problem about taste. |
| 23:09.18 | kyleh | im having a problem with festival, when festival is called in the extension.conf it connects then disconnects and asterisk just hangs |
| 23:09.26 | Katty | jblack: it's just hard to eat enough without getting full |
| 23:09.58 | jblack | Veggies do that. :) |
| 23:10.00 | [TK]D-Fender | Katty: Have you had enough before getting "full"? |
| 23:10.23 | jblack | veggies are good for two reasons... nutrients, which you can replace with vitamins, and filling up (mission accomplished there) |
| 23:10.31 | Katty | i think i could eat bananas all day long and still not get enough potassium |
| 23:10.35 | jblack | so pop a one-a-day with your veggies, and don't feel bad. |
| 23:10.50 | jblack | How do you know you're not getting enough potassium? |
| 23:11.01 | Katty | because i've been keeping track of everything i eat. |
| 23:11.06 | Katty | including the nutrtional value of each item. |
| 23:11.10 | [TK]D-Fender | Katty: Apparently not getting enough potassium doesn't stop people from living beyond 100 years old... |
| 23:11.22 | jblack | Are you anorexic? |
| 23:11.29 | Katty | no |
| 23:11.36 | [TK]D-Fender | jblack: No, just neurotic sadly ;) |
| 23:11.49 | [TK]D-Fender | Katty: But its so CUUUUUUUUUTTTTTTTTEEEEEEEEE@ |
| 23:11.58 | Katty | >.< |
| 23:12.21 | jblack | katty: Unless you're drinking a huge amount of fluids, I wouldn't worry about your potassium |
| 23:12.31 | *** join/#asterisk [netman] (n=netman@3.Red-88-23-116.staticIP.rima-tde.net) |
| 23:12.42 | Katty | and what about my b2? |
| 23:12.47 | Katty | i get like 2% of that a day. |
| 23:13.17 | Katty | and i don't wanna eat Liver :< |
| 23:13.51 | jblack | so take a daily vitamin. |
| 23:14.37 | Katty | sadly, i think i'm going to have to. |
| 23:14.38 | jblack | unless you're seing things like muscle weakness and arrhythmia, you're probably fine on potassium |
| 23:14.50 | jblack | bbias. I want to talk with you more about this |
| 23:15.06 | Katty | k |
| 23:15.51 | jblack | -> |
| 23:29.03 | *** join/#asterisk shmaltz (n=chatzill@mail.dmaven.com) |
| 23:29.11 | shmaltz | hi everyone |
| 23:29.49 | ecrane | is firing xten-xlite 2.0 for linux and using Ekiga to test with. |
| 23:40.41 | *** join/#asterisk ishchatul (i=ishchatu@bzq-80-22-199.red.bezeqint.net) |
| 23:41.30 | ryduh | ecrane: what about voiper? |
| 23:44.52 | *** join/#asterisk tzafrir__laptop (n=tzafrir@212.179.75.202) |
| 23:50.02 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 23:51.11 | ecrane | ryduh: I never heard of it. I'll check it out, thanks! |
| 23:54.29 | ecrane | Wow this voiper thing is interesting but I think I'll need to read a sip book and get my * working better before I can take advantage of what voiper can do. |
| 23:55.08 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 23:59.00 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
