00:00.28 | _zoomy_ | What i know of there is only /etc/odbc.ini |
00:00.30 | kdas | anyone ? |
00:00.45 | [TK]D-Fender | _zoomy_: res_odbc.conf <- pastebin it. |
00:00.56 | [TK]D-Fender | kdas: ManxPower has already answered you. |
00:01.17 | jaytee | kdas, how does the analog phone connect? a digium card with FXS or an ATA adapter? |
00:01.25 | kdas | [TK]D-Fender, then how do you set rxgain? |
00:01.46 | _zoomy_ | [TK]D-Fender: sorry im sort of new to the slang. pastebin? |
00:01.53 | Qwell | ~bs |
00:01.54 | jbot | methinks bs is Banal Superficiality. So be smart: /ignore bs. |
00:01.56 | Qwell | erm, wow |
00:01.58 | Qwell | ~pb |
00:01.59 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
00:02.03 | kdas | jaytee, its a normal comcast phone which i call into a DID number that acess my sip which in turn is registered with asterisk :D |
00:02.12 | Qwell | why was I thinking bs? /peer |
00:03.05 | jaytee | kdas, Comcast VOIP to another SIP provider for DID to your * box? |
00:03.07 | *** join/#asterisk loca|host (n=tux@196.203.53.221) |
00:03.26 | *** join/#asterisk sircco (n=sircco@dh207-102-87.xnet.hr) |
00:03.34 | LemensTS | TK: the xlite didnt have that option, my polycom had a spot under "Sip" tab, under "local sip port" i changed it to something different and it still does the same thing after 'asterisk restart' |
00:04.15 | kdas | jaytee, Comcast normal pstn that calls a DID that is to a SIP account and my * box regesters to that SIP |
00:04.55 | sircco | hello people, i have misdn trunk on openvox card with 4 bri.When I dial out i want outgoing cid to be some number and when call goes out it's number assigned to bri port. Trunk is mISDN/g:out/$NUMBER$ |
00:05.19 | sircco | how can i get right outgoing number? |
00:06.08 | kdas | jaytee, does that make sence ? |
00:06.20 | jaytee | sense enough |
00:06.50 | _zoomy_ | [TK]D-Fender: http://paste.debian.net/16489/. So I only use lines 12-19 really |
00:07.27 | kdas | jaytee, so is it my * box config ? or thc maybe the SIP don't allow user input or is it the rxrate ? |
00:07.29 | jaytee | kdas, what do you have set for dtmfmode in the general section of your sip.conf and in the config for the softphone |
00:08.02 | *** join/#asterisk genji1981 (n=gej@203.152.122.50) |
00:08.06 | jaytee | kdas, dtmfmode=rfc2833 should work for 99.9% of sip connections. |
00:08.49 | [TK]D-Fender | _zoomy_: there is no section named 'asterisk-connector' in there. You have entries that refer to a DSN by that name, but that is not the name of an entry |
00:08.54 | kdas | jaytee, but does that apply even if i am calling in through a pstn line ? |
00:09.19 | [TK]D-Fender | _zoomy_: [asterisk] , [mysql2] , [sqlserver] |
00:09.25 | jaytee | kdas, yes because the last leg of the journey into your * box is SIP not analog |
00:09.35 | LemensTS | d |
00:10.13 | *** join/#asterisk kj4acm (n=kj4acm@24-183-225-98.dhcp.kgpt.tn.charter.com) |
00:10.25 | _zoomy_ | ok tnx ill try and change it. i thought it got the connection details from /etc/odbc.ini, no? |
00:10.51 | kdas | jaytee, well just tried that but no luck |
00:11.03 | jaytee | really? did you do a reload? |
00:11.05 | [TK]D-Fender | _zoomy_>Hello. Im trying to connect to a postgresql database and do a write via Set() in the dialplan. But I get this error message: [Sep 6 01:42:28] ERROR[8809]: func_odbc.c:128 acf_odbc_write: No database handle available with the name of 'asterisk-connector' (check res_odbc.conf). <-- READ YOUR OWN ERROR. iT TELLS YOU TO YOUR FACE WHERE YOU SHOULD BE LOOKING |
00:11.07 | kdas | calling in with my soft phone and analog still same |
00:11.09 | [TK]D-Fender | Darn caps |
00:11.19 | kdas | dtmfmode=rfc2833 in the general setting of my sip.conf |
00:12.30 | jaytee | do you have relaxdtmf=yes or is that commented out? usually it's a waste of time playing with it but you could try setting it yes or comment it out if it's already set to that. |
00:12.36 | *** join/#asterisk moy (n=moy@189.169.91.147) |
00:12.54 | kdas | jaytee, in my sip.conf ? |
00:12.59 | jaytee | yup |
00:13.07 | kdas | let me try to put it in |
00:13.27 | jaytee | I'm not sure I like that way that was phrased |
00:14.00 | kdas | jaytee, why whats wrong ? |
00:14.22 | *** join/#asterisk mog (n=mog@216-83-246-98.static.networktel.net) |
00:14.22 | *** mode/#asterisk [+o mog] by ChanServ |
00:14.25 | jaytee | it was sarcastic humor for your last statement |
00:15.22 | [TK]D-Fender | kdas: Don't worry, jaytee will help you out. He'd practically bend over backwards to help people here ;) |
00:15.35 | kdas | jaytee, http://pastebin.com/d3bbe38de |
00:15.40 | kdas | that is my sip.conf |
00:16.08 | kdas | [TK]D-Fender, that is re-assuring |
00:16.24 | jaytee | kdas, line 3, comment it out, reload and test again |
00:16.59 | kdas | the dtmf mode line ?? |
00:17.02 | [TK]D-Fender | kdas: Voipbuster uses "dtmfmode=inband" and requires ulaw <- |
00:17.13 | jaytee | the relaxdtmf=yes line |
00:17.32 | jaytee | oh, he's using Voipbuster? |
00:17.52 | kdas | voipbuster for incoming |
00:17.54 | [TK]D-Fender | jaytee: Read his PB will ya.... |
00:17.57 | kdas | terrasip for outgoing |
00:18.24 | jaytee | ah, ok |
00:18.31 | kdas | [TK]D-Fender, i am using ulaw its under the [kdas] context |
00:19.28 | [TK]D-Fender | kdas: wrong MODE for them. |
00:19.33 | kdas | also i am using x-lite for my softphone if that is any help |
00:19.40 | _zoomy_ | [TK]D-Fender: sorry man and thanks... of course it works now. what i dont really get though is why i from func_odbc.conf have to redirect to a dns in res_odbc.conf which sort of has an "in-dns" and an "out-dns" which redirects to odbc.ini where the connection info is. is this just how its set up or have i really misunderstood something? |
00:19.42 | [TK]D-Fender | kdas: 15. - Does TerraSip support the DTMF Mode (Touch Tone)? NO. DTMF (Dual Tone Multiple Frequency) or Touch Tone is unusual within Sip-Telephony, because caller and destination have to support the same SIP DTMF Mode in order to communicate with each other. |
00:19.47 | kdas | [TK]D-Fender, i switched to inband but no help |
00:20.12 | [TK]D-Fender | kdas: http://www.terrasip.com/xoshop/faqdesk_index.php?nav=2&faqPath=2&language=ru&t_country=ru |
00:20.34 | kdas | [TK]D-Fender, i am not using terrasip to access my voicemail though!! |
00:20.35 | [TK]D-Fender | kdas: Terasip doesn't seem to want to support DTMF. I have seen a few other fuck-off carriers like this |
00:21.01 | [TK]D-Fender | kdas: PB your failed call and your new sip.conf |
00:21.47 | jaytee | [TK]D-Fender, he also said his softphone won't work with DTMF either. |
00:22.06 | kdas | [TK]D-Fender, there is no failed call. i dial into voicemail just fine but when it asks me to press 1 or enter my password it don't regonize my input |
00:22.26 | [TK]D-Fender | kdas: Show us the call. |
00:22.44 | [TK]D-Fender | kdas: And what softphone? |
00:22.52 | kdas | [TK]D-Fender, how ?, x-lite |
00:23.08 | [TK]D-Fender | kdas: PB the CLI output at verbose 10, SIP DEBUG enabel;d |
00:24.20 | kdas | [TK]D-Fender, how do i do that ? in the logger.conf file ? |
00:24.32 | [TK]D-Fender | kdas: Asterisk CLI <------------------- |
00:24.51 | kdas | [TK]D-Fender, yea how do i set the verbose to 10 and sip debug enable ? |
00:25.01 | [TK]D-Fender | jaytee: He's all yours |
00:25.08 | kdas | lol |
00:25.09 | [TK]D-Fender | steps out for a while |
00:25.14 | kdas | [TK]D-Fender, i got it |
00:26.18 | kdas | [TK]D-Fender, the sip debug output is a crap ammount you want one bit of it or should i copy it all ? |
00:26.41 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
00:27.58 | kdas | ...? |
00:28.06 | jaytee | kdas, he's stepped out. |
00:28.12 | kdas | oh |
00:28.17 | kdas | jaytee, so no ideas ? |
00:30.24 | jaytee | well, you can do what he asked and set verbose 10 and sip debug and pastebin a call and I'll look at it but I'm not even close to being in the same league as [TK]D-Fender for debugging. |
00:30.59 | kdas | jaytee, but i mean the sip debug output is crazy it every single packet |
00:31.46 | kdas | well i will play arround and try some more and then come back till then thanks |
00:32.01 | jaytee | kdas, just copy and paste from when the call starts to when you try a couple keystrokes and hangup |
00:34.14 | *** part/#asterisk dawalama (n=dawalama@wsip-68-105-250-226.sd.sd.cox.net) |
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00:43.16 | [TK]D-Fender | jaytee: phew... saved myself on that one.... |
00:43.47 | jaytee | [TK]D-Fender, haha, yep got out just in time. |
00:44.33 | jaytee | [TK]D-Fender, did you checkout that link I posted earlier with the guy's ad? |
00:45.06 | [TK]D-Fender | jaytee: If I had a nickel for every twit who wouldn't "show me the money" in this channel, I'd put them in a bag an use it to beat the next one FUCKING SENSELESS as a warning to others... |
00:45.20 | [TK]D-Fender | jaytee: Yes, I watch DIGG like a hawk too.... |
00:45.39 | [TK]D-Fender | breaths out..... |
00:45.40 | jaytee | that guy was really pissed, it was so funny |
00:45.42 | [TK]D-Fender | ouuuuuuuummmm |
00:46.03 | jaytee | om mani padme hum, breathe! om |
00:47.42 | [TK]D-Fender | jaytee: padme... yup, Nathalie Portman... I'd hit that :) |
00:47.47 | *** join/#asterisk althiom (n=pirch@189.237.8.67.cfl.res.rr.com) |
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00:47.55 | jaytee | I loved V for Vendetta |
00:48.03 | [TK]D-Fender | jaytee: Ditto.... |
00:48.08 | tristanbob | anyone work with Vertical sip phones? will they work with asterisk? |
00:48.12 | [TK]D-Fender | and on that note I'm off to play pool for the night |
00:48.13 | tristanbob | http://www.vertical.com/index.html |
00:48.24 | jaytee | and I'd put up with Jar Jar just to see Natalie |
00:48.43 | jaytee | [TK]D-Fender, have fun |
00:54.09 | althiom | does *now support fax over ip?? |
00:54.36 | ManxPower | althiom: I guess it depends on how you define "support". |
00:54.43 | jaytee | althiom, not reliably but then no other SIP based VOIP system does either |
00:54.52 | ManxPower | There's T.38, with all the limitations Asterisk has with T.38, but that's about it. |
00:55.36 | ManxPower | oh, look at that. newbie wrote "*now" instead of AsteriskNow. Well, bless his heart. |
00:56.07 | althiom | I have a voip connectin and would like to use FOIP on it |
00:56.14 | UD | you did too |
00:56.16 | ManxPower | althiom: give up now. |
00:56.34 | ManxPower | UD: got your problem solved yet? |
00:56.41 | UD | ya |
00:56.45 | althiom | and was wondering if it was supported |
00:56.47 | UD | i had to walk away from it for a while |
00:56.51 | ManxPower | good. |
00:56.56 | jaytee | althiom, get a POTS line and a real fax machine or teach people how to attach documents to email. |
00:57.08 | ManxPower | althiom: perhaps you could ask on the #AsteriskNow channel? |
00:57.24 | jaytee | I'm running AsteriskLater, it's much better :-) |
00:57.25 | althiom | alright and thanks |
00:57.34 | UD | sorry to have been a pita |
00:59.42 | UD | i didnt even know there was a debug command |
00:59.43 | UD | heh |
01:00.06 | ManxPower | If you had read the damn book you would have. |
01:00.11 | althiom | alright, then I am having installing asterisk. I can get to the change the password screen and it wil not allow me to do anything |
01:00.19 | UD | dug this hunk of junk computer out of the attic and had so many other issues with it that it was frustrating |
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01:01.12 | UD | i read through the old version a few times |
01:01.49 | UD | it isnt as educational when you dont have a working config to test stuff out on |
01:03.14 | UD | the book isnt the most clear IMO |
01:04.36 | *** join/#asterisk terracon (n=garry@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com) |
01:04.50 | UD | 1st edition i mean |
01:05.02 | UD | im going to to burn it |
01:06.03 | *** part/#asterisk althiom (n=pirch@189.237.8.67.cfl.res.rr.com) |
01:07.15 | UD | ManxPower do you run asterisk on your myvzw connection? |
01:12.53 | *** join/#asterisk Entr4nced (n=IMG001@216.144.37.23) |
01:13.45 | jaytee | I found a link to a pdf of The Asterisk Handbook by Mark Spencer and boy that was really old. |
01:14.09 | Qwell | jaytee: you aren't kidding |
01:16.26 | jaytee | I was googling for hotline and asterisk as keywords and that was one of the hits. I want to setup an analog phone with an FXS ATA adapter that when you go off hook it autodials our security department. |
01:16.28 | *** join/#asterisk Yourname (i=Yourname@unaffiliated/yourname/x-837320) |
01:16.56 | Qwell | jaytee: zapata.conf, immediate=yes |
01:17.55 | jaytee | Qwell, yeah but I need this to be remote on an ATA like a GS Handytone or a Linksys SPA2102, preferably the latter. |
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01:19.39 | beighto | Anybody have experience trunking Cisco Call Manager and Asterisk? |
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01:22.06 | jaytee | I've been digging through the Linksys documentation which is pretty crappy looking for a way to do the same thing as immediate=yes but haven't found anything like it so far. |
01:22.14 | jaytee | may end up having to call Linksys. |
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02:00.34 | Kyler | I need to get a unique (or even close to unique) ID for a call...without a dot in it. Is there a simple way to do that? I see the patch for a negative substring length but I don't want to patch. |
02:01.34 | Kyler | Oh! Is there a float to integer conversion? Looking... |
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02:30.35 | rpm | anyone know where i can find the mitel sip firmware 7.0.0.8? i can only find 6.0.0.19... |
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02:46.47 | jameswf-home | neat http://www.nathanpralle.com/software/ast_masterlist.html |
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02:50.44 | jaytee | jameswf-home, that's a great link, thanks! |
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03:15.03 | jameswf-home | ~itsplist-us |
03:15.03 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
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03:17.06 | jeev | my friends office's people are complaining about hang ups.. a lot lately |
03:17.10 | jeev | i dunno if it's my ITSP or not :/ |
03:23.34 | techman97 | jeev: I'm in the same ballpark, just in the last 4 weeks |
03:25.35 | jeev | :/ |
03:25.41 | jeev | what is that ITSp that mitnick faggot used |
03:26.20 | ManxPower | jeev: what protocol are you using? |
03:26.40 | jeev | ManxPower, from the local server to the main server, iax |
03:26.48 | jeev | then iax2 goes to ITSP via SIP |
03:27.35 | jeev | so -internal network with 30 phones- - -internal asterisk- iax - nat - internet - iax - asterisk server - sip - internet - sip - itsp |
03:28.15 | ManxPower | jeev: I have personally experienced situations where IAX2 randomly disconnected calls. Also, if you keep everything as SIP you can do reinvites internally |
03:28.55 | ManxPower | Now,this "personal experience" was several years ago and I would be shocked if they had not fixed the problem by the recent releases. |
03:28.58 | jeev | ManxPower, i dunno what to do.. i couldn't for all my life, figure out how to use the internal asterisk box to connect to my external one via SIP |
03:29.08 | jaytee | plus you've got two internet hops and routing through 2 asterisk servers. lotta opportunity for latency |
03:29.12 | jeev | if the problem is my dual wan, i will remove one of the ports. |
03:29.23 | jeev | nat just isn't working. |
03:29.34 | jeev | dual wan isn't helping, no matter how my nat is set up. |
03:29.37 | ManxPower | NAT is hard to do in complex situations. |
03:29.51 | jeev | yea, it's so weird, it's like i'm trying a hosted pbx but it's not working via sip.. |
03:29.54 | jeev | and i know it's fully possible |
03:29.54 | vipcarrier | does any one has a php script to provision cisco 79XX |
03:29.58 | ManxPower | jeev: people frequently end up NATing internal traffic by mistake. |
03:30.01 | jeev | jaytee, what's that ITSP that needs a $35 deposit ? |
03:30.05 | vipcarrier | does any one has a php script to provision cisco 79XX |
03:30.32 | jaytee | jeev, you mean FWD? the one that used to be free. |
03:30.33 | jeev | ManxPower, what do you mean by that? internal traffic is just fine |
03:30.36 | jeev | hmm |
03:30.40 | ManxPower | jeev: can you set up a GRE tunnel between the sites or are you using consumer routers? |
03:31.11 | jeev | isn't that a type of tunnel? like vpn / |
03:31.22 | ManxPower | jeev: VPN without the encryption |
03:31.28 | jeev | jaytee, flowroute. |
03:31.33 | jeev | any luck with flowroute? |
03:31.36 | jeev | ManxPower, definitely. |
03:31.40 | jeev | the dual wan router at the office is a freebsd box. |
03:31.43 | jaytee | jeev, never used it. |
03:31.45 | jeev | so is my asterisk box at the datacenter. |
03:31.51 | ManxPower | jeev: Ah. |
03:32.08 | jeev | you suggest? |
03:32.16 | jeev | vpn would be great too.. |
03:32.22 | jeev | but i just feel like it'd add massive latency |
03:32.26 | jeev | but the crap part is the instability. |
03:32.28 | jaytee | lotta people running * on freebsd without trouble, doubt if it's your * boxes. |
03:32.38 | ManxPower | jeev: About 5 suggestions, none of them easy to explain in my somewhat intoxicated state, but I'll try. Give a min. |
03:33.08 | ManxPower | jeev: not VPN. A gre tunnel would not have encryption latency but would allow you to talk between your networks WITHOUT NAT |
03:33.09 | jeev | jaytee, i know.. |
03:33.17 | jeev | ManxPower, i will do that. |
03:33.20 | jeev | i've got to read on it |
03:33.25 | jeev | jaytee, i'm only saying it could be an IAX issue |
03:33.27 | ManxPower | It would have a little added latency from the tunnel. |
03:33.47 | jeev | totally ok, the connection to my servers is ~10ms. |
03:33.58 | ManxPower | also if you did that and switched to sip (since there's no nat) you could have phones doing reinvites anywhere except to the ITSP. |
03:34.11 | jeev | hm |
03:34.22 | ManxPower | You won't notice less than like 150ms |
03:34.30 | jaytee | jeev, yeah possibly. any weird errors for IAX in the logs? |
03:34.46 | jeev | i can go back to the log, i have debug enabled in both places |
03:34.50 | jeev | but i've got to step out REALLY soon |
03:34.55 | jeev | i can't believe i'm being helped but have to leave |
03:34.56 | jeev | this sucks |
03:35.29 | ManxPower | jeev: I'll be around this weekend |
03:35.34 | ManxPower | usually late mornings for a while |
03:35.36 | jaytee | ManxPower's suggestion sounds like a more reliable solution with your setup. |
03:35.38 | jeev | hmm |
03:35.41 | jeev | yea yea |
03:35.52 | jeev | i'm aboutto deposit $35 into flowroute to give them a full shot for about a week |
03:35.57 | ManxPower | My suggestion should be much much simpler, assuming you can easily to a tunnel |
03:36.01 | jeev | it's already annoying, they text me every day saying this got hung up, that got hung up |
03:36.04 | jeev | they can't hear me.. this and that |
03:36.14 | ManxPower | jeev: welcome to MY world. |
03:36.16 | jeev | i'm sure tunneling via the two bsd boxes would be great. |
03:36.22 | jeev | i'm almost to the point of getting ATT linse here |
03:36.22 | jeev | lines |
03:36.30 | ManxPower | you can easily encrypt the tunnel later if you want. |
03:36.32 | jeev | if you get an unmetered $50 |
03:36.35 | jeev | per month line |
03:36.39 | jeev | how many concurrent's could there be? 2? |
03:37.22 | ManxPower | As much as I think FreeBSD is way over rated, it's far, far better then a linksys or netgear. |
03:37.41 | jblack | for $50/unlimited, 2 concurrent is common |
03:37.50 | jeev | hmm |
03:37.55 | jeev | i dunno what to do man, i'm getting TIRED of this crap. |
03:37.59 | jblack | ManxPower: yeah, I can believe that. |
03:39.11 | jaytee | I had a real weird one today. Guy at DID number 630-2031 keeps getting calls that aren't for him and the callers have said they get 630-203-1111 on their callerid which is Hinsdale, Illinois. I'm in Indianapolis 317 area code. All I can think of is they see the number and try to call it back without a 1 and matches 7 digits before they can finish dialing the whole thing. but if I dial from a cell either with a 1 or without I get a message from MC |
03:39.11 | jaytee | I that the number isn't in service. If I go through my telco I get my telco's not in service message. |
03:40.07 | ManxPower | jeev: try sip, it can't hurt and will be easy with a tunnel |
03:40.42 | jaytee | I think it's MCI's routing screwing up or some call center is calling people in my area code and they're trying to dial the number back without a 1 |
03:41.16 | ManxPower | he could ask him what number they dialed. |
03:41.24 | jeev | weird |
03:41.26 | jeev | ok i've got to go |
03:41.32 | jeev | ManxPower,i will try SIp asap and let you know. |
03:41.32 | jaytee | ManxPower, he did |
03:41.33 | jeev | bbiab |
03:42.56 | jaytee | I told him to get me more specifics. Did they dial a 1? did they dial the full 11 digits? what area code are they calling from, etc. says if he gets another call like that he'll get more info for me. Started happening 3 days ago and has happened several times now. |
03:42.59 | vipcarrier | guys |
03:43.20 | vipcarrier | any one has a php scrtip that can generate configs for cisco 79XX |
03:43.29 | vipcarrier | I'm willing to pay for it |
03:43.34 | vipcarrier | it is an urgent |
03:43.40 | ManxPower | vipcarrier: I doubt anyone different from the 10 other times you asked. |
03:43.54 | jaytee | lol, I was just thinking that |
03:44.19 | vipcarrier | i can't belive that u guys provisiong every singel phone manualy |
03:44.26 | ManxPower | vipcarrier: not that many people use Cisco phones here. |
03:44.45 | jaytee | vipcarrier, I don't need php scripts for Polycoms |
03:45.01 | ManxPower | Polycoms are just as good quality and they don't have weird firmware restrictions. |
03:45.07 | vipcarrier | jaytee how do u provision them every phone manyaly |
03:45.29 | ManxPower | vipcarrier: you change 3 lines in a template config file and plug in the phone |
03:45.33 | vipcarrier | i don't like polycom's becouse they have a web gui and it is not secure |
03:45.36 | jaytee | vipcarrier, I provision them using FTP and the built in firmware logic in the phone. |
03:45.45 | jaytee | you don't have to use the web gui |
03:46.00 | vipcarrier | but it is not secure in corporate envirement |
03:46.12 | jaytee | and the web gui doesn't give you the ability to modify all the phone's capabilities. |
03:46.14 | ManxPower | in fact, the web gui on the polycoms is only good for doing your first phone config and generate the template file. |
03:46.27 | ManxPower | vipcarrier: you can disable the web gui. |
03:46.37 | vipcarrier | how can u disabel it? |
03:47.04 | vipcarrier | i remember all Asterisk@Home had a provisioning script for Cisco's but I can't find any where that source |
03:47.10 | ManxPower | read the docs |
03:47.11 | jaytee | yeah, like if you want to enable ring-answer so the phone will autoanswer a call for paging. |
03:47.20 | ManxPower | the polycoms have extensive docs for their phones |
03:47.26 | jaytee | can't do that in the damn web gui |
03:47.41 | ManxPower | jaytee: I've not touched the Polycom gui in YEARS. |
03:47.53 | vipcarrier | I have few polycoms that I'v got for testing but not really happy with them |
03:47.54 | ManxPower | I suppose we really should disable it. |
03:47.58 | vipcarrier | I hav 330 and 650 |
03:48.19 | jaytee | ManxPower, I've only done it for a couple phones for initial testing but so much of the feature set capabilities are lacking in it. |
03:48.19 | ManxPower | vipcarrier: the more you use the polycoms the more you will like them. The more you use Ciscos the less you'll like them. |
03:48.43 | vipcarrier | look I had an aastra for 2 weeks and i cracked it |
03:48.47 | vipcarrier | polycom 650 dead |
03:48.51 | vipcarrier | cisco 7960 works |
03:48.58 | ManxPower | best of luck |
03:49.18 | vipcarrier | u can traw the phone in to the brick stone wall and it will keep working |
03:49.27 | vipcarrier | aastra 57i CT dead |
03:49.34 | vipcarrier | after i droped a had set on it |
03:49.42 | jaytee | if you add a block in the phone's config to handle the ring-answer you can setup paging using SIPAddHeader and the Page application and it'll ring all the phones once, they'll autoanswer and voila. real sweet |
03:51.40 | vipcarrier | $200 for web gui script to provision Cisco 79XX |
03:52.23 | jaytee | if I knew Cisco phones and PHP I'd take you up on that offer but I doubt if you'll find anyone in here with both those skillsets at the level you need. |
03:52.44 | Qwell | where's bkruse.. |
03:53.27 | jaytee | good question, he's not here or in asterisk-gui but then you already know that |
03:53.56 | jaytee | we should "chip" that dude so we can track him with a GPS unit. |
03:54.30 | Qwell | vipcarrier: trixbox has such a script. Feel free to send the money via paypal to my email address |
03:55.53 | *** join/#asterisk sah-work (n=Bawbatos@adsl-76-227-17-18.dsl.pltn13.sbcglobal.net) |
03:56.45 | Qwell | crickets... |
03:57.35 | jaytee | well, it's night. rather have crickets than cicadas |
04:01.11 | *** join/#asterisk moy (n=moy@189.169.91.147) |
04:03.59 | Qwell | guess not |
04:06.39 | *** join/#asterisk Thorn (n=thorn@unaffiliated/thorn) |
04:08.58 | *** join/#asterisk c4t3l (n=root@74.95.210.124) |
04:09.14 | *** join/#asterisk amessina (n=amessina@2001:470:1f11:68:20e:cff:fe01:d5ec) |
04:13.16 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.34.7) |
04:14.45 | jameswf-home | Qwell: any personal itp recomendations |
04:14.51 | jameswf-home | *itsp |
04:14.52 | Qwell | itp? |
04:15.03 | Qwell | dunno, I've always liked nufone |
04:15.35 | Qwell | (Jeremy isn't paying me to say that) |
04:16.48 | jameswf-home | should do this " * Asterisk configuration assistance is $99 per hour, one hour minimum." |
04:16.53 | mchou | jameswf-home: all recs are based on your call volume |
04:17.45 | jaytee | jameswf-home, where did you see that? |
04:18.12 | mchou | $99 is probably a deal :) |
04:18.13 | jameswf-home | I like les net but wife wants a local DID |
04:18.27 | mchou | jameswf-home: where you live? |
04:18.31 | jameswf-home | az |
04:18.42 | mchou | jameswf-home: call volume? |
04:18.44 | jaytee | McCain country |
04:19.18 | mchou | jameswf-home: minutes/month or week? |
04:19.23 | jameswf-home | well she talks to me which would be over an iax trunk to work... heh probably ledss than 500 min/mo |
04:19.29 | jameswf-home | *less |
04:19.46 | mchou | jameswf-home: diamondcard.us |
04:20.04 | mchou | jameswf-home: strictly paygo |
04:20.26 | mchou | jameswf-home: they have DIDs nationally for all of UD |
04:20.30 | mchou | US* |
04:20.32 | jameswf-home | I should just trunk in to work and use company lines... |
04:20.39 | jblack | I'm using callwithus, diamondcard and voipstreet. |
04:20.56 | mchou | more than what's listed on their (crappy) web site |
04:21.17 | jblack | CWU is .0125, diamon is .015 and voipstreet is .017. CWU has less DIDs, diamond and voipstreet had 570 area code numbers, which are hard to get. |
04:21.29 | mchou | jblack: of the three, which is most feature complete? |
04:21.37 | techman97 | voipstreet...*shudder* |
04:21.47 | jameswf-home | I found callwithus to be latency ridden |
04:21.49 | jblack | They're all quirky. |
04:22.04 | techman97 | I've been good with callcentric for the last year, but their fax service is broken permanently |
04:22.15 | mchou | diamondcard is a bit sucky becase they dont set the CNAME field in callerID |
04:22.17 | jblack | They all seem reliable, but none of them have that professional sheen to 'em. |
04:22.36 | jblack | That's true. Diamoand card doesn't do CNAME. I forgot to mention that. |
04:22.48 | mchou | CNAM* |
04:22.57 | jblack | yah ayah yah. That's still the case. |
04:23.02 | jameswf-home | what is fax? |
04:23.05 | mchou | lol |
04:23.08 | jblack | as in, "this morning". |
04:23.23 | mchou | I dont understand why ppl still use faxes |
04:23.24 | techman97 | the only kickers about CallCentric are that they don't do CNAM and their DIDs are done via SIP RFC, not traditional DNIS/ANI |
04:23.26 | jblack | jameswf-home: A legally binding way to electronically transfer images of paper. |
04:23.35 | mchou | I bought a house w/o using a fax :) |
04:23.40 | techman97 | jblack: exactly |
04:23.41 | jblack | mchou: because the law doesn't know what teh internet is yet. |
04:24.00 | mchou | jblack: wha??? |
04:24.05 | jblack | Remember the lawmakers themselves, a "series of tubes"? |
04:24.05 | techman97 | either it's legal, or your customer base are 3,000 60+ year old geezers who are scared of the internet |
04:24.07 | techman97 | rofl |
04:24.19 | mchou | jblack: you mean all that stuff I ordered on line I can have for free? |
04:24.21 | jameswf-home | I hear email hasn't caught on and no one really has an email address |
04:24.31 | techman97 | yeah, the interwhat? |
04:24.35 | techman97 | that's just a fad. |
04:24.36 | jblack | Laws move slow. |
04:24.58 | techman97 | I drove my sherman tank across Europe at 30MPH without no "interweb" |
04:25.02 | mchou | jblack: so internet mechanidizing is legalized robbery? :) |
04:25.16 | mchou | merchandizing* |
04:25.19 | jaytee | it is technically impossible to attach a digital document to an email, just ask any higher up exec that is not in a technical field, they'll tell ya! |
04:25.30 | jblack | Why are you guys picking on me? I'm not in congress! |
04:25.36 | jblack | Hell, I know how to use all 110 keys! |
04:25.47 | mchou | jblack: cause you bought up the law |
04:25.57 | techman97 | jblack: where's the "any key"? |
04:26.00 | jblack | because someone asked why fax hasn't died. What's why. |
04:26.50 | jblack | It hasn't died because legal documents remain legally intact over a fax. |
04:27.17 | mchou | jblack: I bought a house w/o using a fax :) |
04:27.36 | mchou | jblack: that better be legal! |
04:27.51 | jaytee | my first PC was a piece of crap Packard Bell 286 12mhz machine. when most systems had 104 key keyboards this one came with a 105 key keyboard, the extra key labeled Macro. Nothing in the owners manual. Called their tech support. "What's it do? How do I use it?" "Um, it doesn't work and we don't ship those anymore" |
04:27.55 | mchou | jblack: or you're gonna hear from my lawyers :) |
04:28.22 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
04:29.05 | jameswf-home | ok so voipstreet what are the cons |
04:29.09 | mchou | techman97: callcentric doesnt do CNAM either??? |
04:29.23 | techman97 | you're hard pressed to find a provider that does |
04:29.34 | mchou | techman97: ipkall does |
04:29.37 | techman97 | I had one that did, but they dumped their commercial support - voiceeclipse |
04:29.44 | mchou | techman97: and they are "free" |
04:29.58 | mchou | ok, I take that back |
04:30.07 | mchou | CNAM is geographic info |
04:30.18 | mchou | not "full name" |
04:30.42 | mchou | but even geographic info is better than just the phone # |
04:31.00 | jameswf-home | eh just use agi who needs cnam |
04:31.10 | mchou | what??? |
04:31.27 | jblack | callwithus, voipstreet and teliax do CNAM. |
04:31.34 | mchou | jameswf-home: tell me how that works |
04:31.50 | mchou | jameswf-home: how does agi provide CNAM? |
04:31.51 | jameswf-home | voip street doesnt says so in their faq |
04:32.00 | jblack | You want me to call you right now and verify? |
04:32.15 | mchou | jblack: stop smoking dope |
04:32.32 | mchou | jblack: we are talking inbound CNAM |
04:32.35 | jblack | Pardon? Why do you think I'm smoking dope? |
04:32.42 | mchou | jblack: we are talking inbound CNAM |
04:32.53 | mchou | jblack: that's why you smoking dope |
04:33.22 | jameswf-home | mchou: Google, whitepages.com |
04:33.22 | jblack | Ok. I know that voipstreet does incoming CNAM, as does Teliax. IPKall cid is completely broken. I don't know about the other two. |
04:33.36 | mchou | jameswf-home: screw that |
04:33.52 | mchou | jameswf-home: is outdated and unreliable |
04:33.59 | jameswf-home | whocalled.us is valuble |
04:34.14 | mchou | jameswf-home: on whocalled.us we can agree |
04:34.17 | jblack | We need to make a voipinfo table. |
04:34.24 | jameswf-home | if your in whocalled.us you go to telemarketer torture |
04:34.48 | mchou | jblack: wtf are you talking about?? Ipkall has geographic CNAM |
04:35.34 | jblack | It's badly broken. One of their internal hops eats incoming CID and replaces it with SEATTLE WA, and the number with a seattle number. |
04:35.47 | mchou | jblack: in fact it works better than other *paid* ITSP as far as CNAM is concerned |
04:36.04 | jameswf-home | geographic wouldnt be to hard |
04:36.19 | mchou | jblack: maybe, but I've NEVER had that happen |
04:36.26 | jblack | You can find more information on voxilla. |
04:37.13 | jblack | They tend to work when calls are originated with pstn, and tend to break when they come in on voip. |
04:37.35 | mchou | jameswf-home: yeah, except apparently no ITSPs seem to do it |
04:38.14 | mchou | jblack: maybe, but I have those calls and it has NEVER happened to me |
04:38.34 | jblack | Call IPKall with callwithus, voipstreet or teliax if you'd like to verify. |
04:38.52 | mchou | jblack: it could be DID exchange related but I'm dubious |
04:39.06 | mchou | jblack: lol |
04:39.12 | jblack | I had trouble believing it too. It took me several months to become certain. |
04:39.45 | jblack | I found out the hard way, when I was using IPKall to test which providers did CID, and they were all breaking. |
04:39.49 | mchou | jblack: how the hell do you KNOW that's not an issue with the othe ITSPs you mentioned? |
04:40.05 | mchou | jblack: put down the vrack pipe man |
04:40.23 | jblack | I'm getting tired of your insults. Be nice. |
04:40.41 | mchou | jblack: especially since we pretty much concluded that those ITSP dont do CNAM |
04:40.51 | mchou | who is insulting you? |
04:41.03 | jblack | KNOW is a big word. Those three ITSPs work fine when calling other ptsn's, and when calling each other. When IPKall is the destination, all the CID breaks down to SEATTLE WA, with a seattle number. |
04:41.07 | mchou | I'm challenging you on points of fact |
04:41.24 | jblack | Forget it. Plonk |
04:41.38 | mchou | jblack: good riddance |
04:42.07 | techman97 | ok, so here's a weird problem guys....so I set my SIP phone to register to my asterisk box via IP, works JUST fine. I set it to register by the internal DNS name, "asterisk", and it doesn't register. Syslog shows me that it like 1/2-way registers, but when I dial, an address of 0.0.0.0 shows with an error. I know the DNS is setup right, because I can ping it from other computers by name...but the phones just don't deal with |
04:42.53 | techman97 | I have 3 other PBXs up and running in the same manner (on different networks)...no problems there |
04:43.17 | techman97 | I have sip debug on, and I see messages coming from the phone in question with the DNS name in the SIP registration info... |
04:43.17 | mchou | techman97: I'm dubious your DNS is working right based on your description |
04:43.26 | jblack | techman97: Perhaps you forgot to tell the phones where the dns server is, or what domain to default to. |
04:43.48 | jblack | they may be looking up "asterisk" versus "asterisk.yourdomain.com", to be more clear. |
04:43.53 | techman97 | phones get address via DHCP, and I have all of the options set. |
04:43.59 | techman97 | hmmm, let me try the FQDN |
04:44.03 | techman97 | maybe you're right jblack |
04:44.05 | jblack | dhcp doesn't set the domain, iirc. |
04:44.05 | mchou | lol |
04:44.13 | jblack | or is that ppp..... |
04:44.17 | rob0 | ping(1) does not necessarily use DNS. In fact it doesn't use DNS directly, it uses system gethostbyname() function. |
04:44.27 | techman97 | jblack, yeah it does, DHCP option 15 |
04:44.43 | jblack | You may not be setting it, or it may be overridden in the phone? |
04:44.50 | jblack | Anyways, using fqdn will verify. |
04:45.02 | techman97 | I pinged the hostname from another linux box and a few other misc. windows boxes - works fine as far as I can see |
04:45.03 | techman97 | :S |
04:45.06 | techman97 | yeah, FQDN test we go. |
04:45.30 | mchou | techman97: dont use ping. use dig :) |
04:45.44 | techman97 | ok, phone rebooting... |
04:45.56 | techman97 | syslog and CLI debug rockin' |
04:46.09 | techman97 | c'mon you pig |
04:46.45 | techman97 | crapsticks...that was the deal. FWDN |
04:46.48 | techman97 | FQDN* |
04:46.52 | jblack | suspected as much. |
04:46.55 | techman97 | man I feel stupid...ROFL |
04:47.00 | jblack | Nah, don't. |
04:47.12 | mchou | stupid is as stupid does |
04:47.25 | techman97 | now to figure out what that whole deal is about |
04:47.26 | techman97 | rofl |
04:47.29 | jaytee | what does baffled mean? |
04:47.35 | jblack | confused. |
04:47.40 | jaytee | hehe |
04:47.48 | jblack | Or drapped with layers of cloth in a rippling fashion. |
04:48.11 | techman97 | jblack: Nice Trading Spaces reference |
04:48.25 | jaytee | that was one of my favorite lines from Highlander |
04:48.31 | *** join/#asterisk obnauticus_ (n=obnautic@about/windows/regular/obnauticus) |
04:48.33 | jblack | is confused. |
04:48.46 | jblack | Trading spaces? Is that a stock market movie or something? |
04:48.53 | mchou | lol |
04:49.01 | mchou | that's Trading Places |
04:49.18 | techman97 | hehehe |
04:49.44 | jblack | turns pink |
04:49.45 | techman97 | Trading Spaces = home redesign show on TLC. My wife is nuts over those shows. :S |
04:49.52 | jblack | Yeah. Just googled it. |
04:50.11 | jblack | My idea of fashion is "Hey kid. See those spiderwebs on the ceiling? I'll let you eat dinner if you get rid of them" |
04:50.25 | techman97 | amen to that. |
04:50.42 | techman97 | hell, my daughter redecorated our house recently with crayons on the walls....=/ |
04:50.49 | techman97 | I'm perfectly fine with that |
04:50.52 | jblack | Did you whip her with a chain? |
04:51.11 | jblack | Oh geeze. You're such a wimp. :) |
04:51.11 | techman97 | not initially no, started with the cat...then moved to the chain |
04:51.27 | techman97 | cat makes great noises while using it as a beating stick |
04:51.31 | jblack | You whipped your kid with a cat? That's cruel! Those things have claws |
04:51.56 | techman97 | not mine...it would fight back, but we got rid of his balls too... |
04:52.08 | jblack | You want to cut into their phsyche, not their skin. |
04:52.13 | jblack | psyche, that is |
04:52.30 | techman97 | =) |
04:52.45 | jblack | I trained mine to jump up and put her shoes on every time my keys rattle. |
04:53.12 | techman97 | irl, I just smiled and started looking at buying some crayon-resistant paint...yeah, wuss am I. My daughter has me wrapped around her little 3yr old pinky |
04:53.27 | techman97 | pavlovian kid - love it |
04:53.30 | jaytee | awwww |
04:54.00 | jblack | Yeah, it's great. |
04:54.21 | jblack | She's learned that I have no problem leaving her (she's 14, after all). She also knows that if she can beat me to the car, I'll probably get a milkshake on the way. |
04:54.37 | techman97 | hahaha |
04:55.05 | jblack | So yeah. A tiny jingle, and she's a lightning bolt. |
04:55.09 | techman97 | oh man...boy phase. I thankfully have a decade before that really starts. I swear I'm going to chain my kid to the basement floor until she's married |
04:55.41 | jblack | All you have to do is tell her that anyone that gets her pregnant will die. And then convince her it's true. |
04:56.11 | techman97 | obedience through intimidation...uncle buck style. "Hey bug, I have a hatchet in the trunk, wanna see it?" |
04:58.15 | mchou | techman97: lol. dude, that wont work. You can be Governor of some State and it still wont work :) |
04:58.27 | techman97 | =) |
04:58.51 | jblack | Absolutely. I'm calm and mellow when she gets straight As, is respectiful, polite and does what she needs to do. |
04:58.58 | mchou | techman97: apparainte you can fire state troopers though :) |
04:59.05 | techman97 | ha! |
04:59.07 | mchou | apparantly* |
04:59.27 | jblack | I work hard to be a great dad when she's a good kid. Deviation, though, results in a 240 force of nature with a bad attitude. |
04:59.43 | techman97 | as it should. |
04:59.53 | jblack | She once took a dollar out of my wallet for ice cream without telling me. That resulted in her bedroom being turned upside down. |
05:00.05 | jblack | I have no idea how she righted the bed. |
05:01.04 | jblack | She learned quick that she can have a dollar from dad if she asks. Not asking is not a good idea. |
05:02.20 | jblack | That only works because she's a good kid. If she weren't naturally good... <shudder> |
05:03.36 | techman97 | =) |
05:04.46 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
05:09.32 | mchou | I wonder how telemarketer equipment works. Apparently they have some tech that knows a fake SIT from a real one. |
05:10.21 | mchou | taht effectively renders Zapateller function relatively useless |
05:10.27 | mchou | that* |
05:14.46 | jblack | Why won't you move, Mr Bubbles? |
05:14.46 | techman97 | damn, I was kinda hoping for a Wootoff tonight...o well. |
05:14.55 | jblack | nope. no wootoff. Not even a bag of crap |
05:15.04 | techman97 | telemarketing stuff - they dump amazing amounts of money into ensuring that you hear their blather. |
05:15.41 | mchou | techman97: yeah, but _how_ does that equipment work? |
05:15.51 | techman97 | that's the magic, and how they can get companies to buy it |
05:15.52 | techman97 | lol |
05:16.02 | techman97 | some of it is the PRI result code that is sent back... |
05:16.15 | techman97 | Asterisk as far as I know just plays audio |
05:16.24 | techman97 | it doesn't actually return a PRI code |
05:16.47 | mchou | techman97: yup |
05:17.29 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@ip68-111-67-4.oc.oc.cox.net) |
05:17.31 | techman97 | I've installed my fair share of those kind of dialers |
05:17.49 | CrazyTux[m] | techman97, what kind is that :) |
05:17.49 | mchou | techman97: pls stop :) |
05:17.53 | scooby2 | with ChanSpy() do you just hangup when done? |
05:18.25 | techman97 | the majority of what i did was for a company (been bought and sold 5+ times since 2001) was CellIT in Miami - their own dialer. |
05:18.32 | techman97 | then some Buffalo dialers and whatnot |
05:18.46 | mchou | techman97: tapeworms have more value than telemarketers |
05:18.46 | techman97 | they became Concerto I think...then was bought by Davox maybe? |
05:18.59 | techman97 | mchou: correct! |
05:19.27 | techman97 | you know what has even less value than telemarketers? |
05:19.44 | mchou | not really |
05:19.53 | techman97 | hard account debt collectors...man, being a fly on the wall in some of those places makes you really never want to be late on a bill |
05:20.43 | techman97 | they have a value to the people they're collecting for, yes....but I swear those people ENJOY being dicks on the phone LOL |
05:20.45 | mchou | well, I'm sure those folks have their uses |
05:21.03 | mchou | they at least purport to go after deadbeats |
05:21.32 | mchou | who do telemarketers go after? Innocent bystanders |
05:21.40 | techman97 | the industry does have value...especially in the current US economy...but the PEOPLE that perform the task are just...interesting people |
05:22.02 | mchou | techman97: that's bullshit |
05:22.14 | techman97 | what, the economy comment? lol |
05:22.15 | mchou | a fake economy is NO economy |
05:22.54 | mchou | that's like saying prositutes do good since they let men let off steam. |
05:23.00 | techman97 | this could be an all-nighter discussion...hehehehehehehe.... |
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05:24.29 | mchou | also especially considering even in this economy all the telemarketer calls I get is "reduce your mortgage" or some such BS |
05:24.47 | techman97 | going after the perceived pain points.... |
05:25.33 | mchou | that and the dish network :) |
05:25.37 | mchou | haha |
05:25.41 | techman97 | LOL! |
05:26.17 | mchou | PITA is more like it |
05:26.24 | techman97 | I just can't wait for the Cialis / Viagra telemarketer calls @ dinner time |
05:26.44 | mchou | I'm BUSY right NOW! :) |
05:26.52 | techman97 | "Oh NOW you call..." |
05:27.44 | mchou | speaking of which I just found out how those drugs work |
05:27.50 | mchou | nitrous |
05:28.13 | mchou | I'd rather go to the dentist, thanks :) |
05:28.42 | techman97 | I have this idiot "eternal frat boy" friend of mine that brags about a Viagra and Red Bull drink he does |
05:28.48 | techman97 | why oh why |
05:29.09 | mchou | sigh |
05:29.31 | techman97 | 32 years old, drank himself though college |
05:29.42 | techman97 | "Frank the Tank" from Old School? yeah. |
05:29.42 | rue_mohr | sorry, can you repeat that in context of a voip phone system? |
05:30.06 | jblack | Bioshock is a great fps. |
05:30.16 | techman97 | Goto(toilet_fast,s,1) |
05:31.01 | mchou | exten => s,1,DrivePorcelainBus |
05:31.12 | [TK]D-Fender | techman97>I have this idiot "eternal frat boy" friend of mine that brags about a Viagra and Red Bull drink he does <- so he's proud that he needs help getting it up or even staying awake? Must be the stud of the retirement community |
05:31.28 | techman97 | aye aye |
05:31.49 | techman97 | he could absolutely fit on this website: http://www.hotchickswithdouchebags.com/ |
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05:36.46 | oilinki | <PROTECTED> |
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05:55.52 | [TK]D-Fender | ok, bedtime. |
05:55.55 | [TK]D-Fender | Later all |
05:56.08 | techman97 | I think I should head out too - l8r all |
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06:47.16 | scooby2 | hrm so I'm supposed to use exten => *96,n,Playback(agent-loggedoff) |
06:47.16 | scooby2 | exten => *96,n,Playback(goodbye) |
06:47.16 | scooby2 | exten => *96,n,Hangup() |
06:47.21 | scooby2 | ack |
06:47.33 | scooby2 | sorry about the paste |
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06:55.28 | vipcarrier | any one wanna job at clifton NJ/ |
06:55.39 | vipcarrier | looking for expert Asterisk/SER |
06:55.48 | vipcarrier | full time |
06:55.54 | vipcarrier | or part time |
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06:57.30 | jblack | vipcarrier: Hmm. |
06:57.46 | jblack | checks the distance from wilkes.barre |
06:58.27 | jblack | Can you take a part timer that would primarily telecommute? |
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07:04.59 | scooby2 | man why does AgentCallbackLogin() say agent logged in when I am trying to logoff? |
07:08.17 | jblack | wow. ser looks pretty low level. |
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07:36.41 | FabiOne | hi all |
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09:45.00 | sircco | anyone here using trixbox with multiple bri ?i need idea how to work out something |
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11:40.33 | BarthezZ | hmmfg weird, I've updated my asterisk from 1.2 to 1.4 (bristuff version 0.3 to 0.4) and now i get some additional warnings i can't really place |
11:40.51 | BarthezZ | [Sep 6 13:39:46] WARNING[3647]: app_queue.c:3930 queue_exec: Unable to join queue 'mainqueue' |
11:40.53 | BarthezZ | for example |
11:41.22 | gr0mit | a lot has changed from 1.2 to 1.4 |
11:41.24 | mvanbaak | well, that message is very clear isn't it ? |
11:41.48 | BarthezZ | mvanbaak not really, the queue is there when using show queues etc.... |
11:42.07 | mvanbaak | BarthezZ: are there any agents logged in ? |
11:42.23 | BarthezZ | yes 6 |
11:42.28 | mvanbaak | BarthezZ: if not, check the queueconfig. the thing you want to check is joinwhenempty |
11:42.52 | mvanbaak | any free agent ? |
11:43.53 | BarthezZ | all are free |
11:44.19 | BarthezZ | hmmfg, i commeted out joinempty and it now seems to work |
11:44.24 | BarthezZ | thanks for the pointer :) |
11:45.16 | mvanbaak | :) |
11:45.31 | mvanbaak | if only all problems would be that easy to fix |
11:45.41 | BarthezZ | WARNING[3548]: chan_sip.c:13003 handle_response: Remote host can't match request NOTIFY to call <-- isn't rining any bell inside of me ;p |
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11:51.12 | BarthezZ | hmmfg, seems to be coming from one of my dect handsets |
11:52.02 | BarthezZ | correction :x all my dect handsets |
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12:40.04 | knarfly | 8-) |
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14:26.49 | fordfrog | hi, anybody knows why I get this error when starting asterisk? /usr/lib/asterisk/modules/res_config_odbc.so: undefined symbol: SQLFetch |
14:27.06 | fordfrog | I tried with asterisk 1.2.17 and 1.2.27 and it behaves the same |
14:28.01 | synchris | configure options? |
14:29.28 | fordfrog | what exactly? I am not asterisk expert so not sure what you exactly mean :-) anyway I can connect to the database using isql from unixodbc |
14:30.13 | synchris | u used any package? distro specific? |
14:30.19 | fordfrog | gentoo |
14:30.25 | fordfrog | unixODBC 2.2.12 |
14:31.24 | synchris | use flagas? |
14:31.27 | fordfrog | I recall I had the problem last time I installed asterisk too but do not recall what was the problem |
14:31.28 | synchris | flags* |
14:31.31 | fordfrog | mmt |
14:31.55 | fordfrog | net-misc/asterisk-1.2.27 USE="hardened mmx odbc postgres ssl -alsa -bri -curl -debug -doc -gtk -h323 -lowmem -nosamples -osp -pri -speex -sqlite -zaptel" |
14:32.18 | synchris | this is probably |
14:32.25 | synchris | something missing from odbc |
14:33.29 | synchris | fordfrog, try revdep-rebuild and some more extensive option |
14:34.27 | fordfrog | synchris: the backend database is postgresql, could that be related to the pg driver? |
14:34.31 | synchris | fordfrog, plus check with ldd |
14:34.48 | synchris | i think is odbc problem |
14:35.19 | synchris | as the odbc ask for this undefined symbol |
14:35.47 | synchris | fordfrog, be sure check with ldd too |
14:35.56 | synchris | ldd the library which problem exist |
14:36.54 | fordfrog | synchris: just `ldd lib`? |
14:37.38 | fordfrog | synchris: revdep-rebuild rebuilt nothing |
14:39.22 | fordfrog | synchris: this is what I got from ldd: http://pastebin.osuosl.org/22013 |
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14:49.16 | jaytee | mornin [TK]D-Fender |
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15:07.06 | BarthezZ | Hey, I'm trying to recieve a fax over an zap channel (junghanns quad bri card) using rxfax... After some time my log looks like http://paste.barthezz.name/?show=318 and obviously my fax isn't received :x |
15:07.52 | BarthezZ | when i debugged an open zap channel trying to receive a fax it only displayed Null Frame(5) Subclass: n/a alot |
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15:25.15 | jeev | :< |
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15:48.13 | justinnnnnn | hey ppls |
15:48.35 | justinnnnnn | if i have 2 asterisk servers.. whats the best way to share the filesystem for redundancy ? |
15:50.43 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
15:51.37 | Alton2 | Not sure what you're getting at there..... |
15:51.50 | Assid | anyone suggest a good termination provider? around 1.2-1.5c/min? |
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15:57.55 | Assid | i mean i do see broadvoice and others which have unlimited.. but i prefer going a-la-carte |
16:02.55 | justinnnnnn | alton2 me? |
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16:18.54 | Alton2 | justinnnnn yes |
16:19.40 | fordfrog | synchris: seems noload res_config_odbc.so solved the problem |
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16:47.37 | jeev | warez russellb |
16:48.36 | outtolunc | trying to make russell_jr .. who knows |
16:49.40 | jeev | outtolunc, how can i take a variable that the headers pass to me and then set another variable with the results? |
16:49.48 | *** join/#asterisk tobias (n=tobias@cpe-076-182-118-165.nc.res.rr.com) |
16:50.38 | lmadsen | Set(HEADER_VAR=${SIP_HEADER(<name_of_header>)}) |
16:50.44 | kdas | I am linking ipkall to a voipbuster account and then ii register on that account with my * box and i have my *box act as a voicemail server, but when i call the server ethier with a analog phone or softphone my *box dosent reconize my input. If some one can help it would be much apreshated. |
16:51.09 | jeev | jhmm |
16:51.58 | jeev | leif, can i msg you the example? it's one line |
16:52.24 | lmadsen | if its one line, I'd rather you just do it here |
16:52.30 | lmadsen | I'm doing a system upgrade, so I might not be able to help |
16:52.36 | lmadsen | I'm barely look at this window :) |
16:52.43 | lmadsen | s/look/looking |
16:52.57 | jeev | 2,Set(HEADER_VAR=${SIP_HEADER(${CALLERID(num)}) |
16:53.00 | jeev | i dont know which to replace |
16:53.16 | lmadsen | uhhhh... CALLERID(num) is not a header |
16:53.22 | lmadsen | that is another function |
16:53.23 | outtolunc | sip header is a function |
16:53.48 | lmadsen | don't use CALLERID(num) -- that is already giving you a value |
16:54.04 | jeev | i have SipAddHeader(P-Asserted-Identity:<sip:${CALLERID(num)}@hostname>) |
16:54.05 | jeev | which works |
16:54.23 | jeev | i'm then wanting to insert the caller id unmber and send it to the guys cell phone |
16:54.27 | lmadsen | Set(HEADER_VALUE=${SIP_HEADER(P-Asserted-Identity)}) |
16:55.18 | jeev | ok |
16:56.37 | jeev | tghanks, now i have to fix the incoming call acceptance, jesus |
16:56.47 | lmadsen | allah |
16:56.54 | lmadsen | buddah |
16:57.01 | lmadsen | leif |
16:57.06 | lmadsen | all the same shit |
16:57.07 | lmadsen | :) |
16:57.15 | jeev | hahah |
16:57.16 | jeev | bastard |
16:57.21 | jeev | leif ERICCSON for life |
16:57.21 | jeev | ! |
16:57.22 | jeev | ericson |
16:57.36 | jeev | i can't seem to remember why it does this thing |
16:57.50 | Qwell | Leif must be what the L. stood for in LRH |
16:57.53 | jeev | it's ilke the call from the 'user' to the extension '1DID' rejected because extension not found |
16:58.01 | jeev | i always fix it |
16:58.04 | jeev | then forget next time |
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16:58.20 | lmadsen | Qwell: LRH? |
16:58.24 | Qwell | nothing |
16:58.26 | lmadsen | Qwell: I am the way, the truth, and the Leif! |
16:58.31 | Cresl1n | Psssh |
16:58.37 | outtolunc | looks like someone has a DID var that isn't wrapped in ${} |
16:58.41 | lmadsen | Cresl1n: oh no you di'ant |
16:58.42 | Cresl1n | lmadsen, you sure think a lot of yourself |
16:58.43 | Cresl1n | :-) |
16:58.51 | lmadsen | Cresl1n: lol |
16:58.58 | Cresl1n | Long time no see |
16:59.01 | Cresl1n | how's CA doing? |
16:59.02 | lmadsen | Cresl1n: I think more of you than me |
16:59.05 | jeev | [Sep 6 09:57:10] NOTICE[22791]: chan_sip.c:14035 handle_request_invite: Call from '9xxx' to extension '1xxxxxxxxxx' rejected because extension not found. |
16:59.06 | Cresl1n | heh |
16:59.07 | jeev | i hate that crap |
16:59.08 | lmadsen | Cresl1n: CA is beautiful today |
16:59.14 | jeev | lmadsen, wehre in cali |
16:59.14 | Cresl1n | Good to hear |
16:59.17 | jeev | i'll come and steal your next damn book |
16:59.18 | Cresl1n | we're overcast today |
16:59.18 | lmadsen | CA == Canada |
16:59.22 | jeev | BOOOOOOOOOOOOOOOOOOOOOOOOOOOOOO |
16:59.27 | lmadsen | Cresl1n: slightly overcast, but still sunny |
16:59.36 | jeev | that just made leif ericson not proud |
16:59.46 | lmadsen | Canada > California |
17:00.09 | outtolunc | use CAN for can, us california guys like CA |
17:00.10 | Qwell | lmadsen: You are mistaken. |
17:00.10 | Cresl1n | if you're talking about land masss |
17:00.10 | lmadsen | Qwell: I've been to Cali :) |
17:00.13 | Cresl1n | GDP might not be so |
17:00.26 | lmadsen | heh |
17:00.32 | Cresl1n | although, that might be incorrect |
17:00.38 | lmadsen | wikipedia! |
17:00.41 | Cresl1n | <-------- likes people from canada |
17:00.49 | lmadsen | <------- likes people from HSV |
17:00.58 | Cresl1n | hugs lmadsen |
17:01.10 | lmadsen | w00t |
17:01.17 | lmadsen | Cresl1n: gonna be at Astricon? |
17:01.22 | Cresl1n | Yessir |
17:01.25 | lmadsen | w00t |
17:01.25 | Cresl1n | got 3 talks to do |
17:01.26 | jeev | i'm gonna go to astricon and beat every canadian |
17:01.29 | lmadsen | damn G :) |
17:01.40 | jeev | lmadsen, you, twice. |
17:01.40 | lmadsen | jeev: uh huh... I doubt that |
17:01.50 | lmadsen | wonders how big jeev is... :) |
17:02.07 | jeev | 6'1 194 now |
17:02.18 | outtolunc | just put a dialplan in front of him, that'll slow him down |
17:02.24 | jeev | lol |
17:02.27 | lmadsen | kick to the nuts will bring you down to 5'3" |
17:02.35 | lmadsen | outtolunc: lol |
17:02.36 | jeev | lol |
17:02.45 | lmadsen | goes back to his system upgrade |
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17:02.50 | jeev | ok |
17:02.52 | jeev | No |
17:02.53 | jeev | leif! |
17:02.54 | jeev | Set(HEADER_VALUE=${SIP_HEADER(P-Asserted-Identity)}) |
17:02.55 | jeev | last thing |
17:03.01 | jeev | how can i set caller id with that value now |
17:03.07 | jeev | screams adrian.... er leif |
17:03.35 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
17:03.42 | jeev | guessing: Set(CALLERID(number)=$SIP_HEADER) ? |
17:04.28 | jeev | almost. |
17:04.54 | outtolunc | {almost} |
17:05.13 | jeev | hahah |
17:05.26 | jeev | ok, i got it working kind of... except it adds crap to it |
17:05.29 | jeev | which i've got to remove |
17:05.50 | lmadsen | jeev: before you ask... think for just a few seconds, and you'll probably figure most of your questions out |
17:06.55 | jeev | yea, i know bro. |
17:07.15 | lmadsen | bo knows |
17:07.16 | jeev | CALLERID(number)=<sip:1xxxxxxxxx@xxxxxxxxxxxxx>") in new stack |
17:07.21 | jeev | i've just got to fix the thingy |
17:07.23 | jeev | and it'll work |
17:07.33 | lmadsen | CUT() is a wonderful thing |
17:07.47 | jeev | yea, i know |
17:07.56 | jeev | i mean, i helped youw rite the book in the time of your need |
17:07.57 | lmadsen | actually... I'd just use the variable string control |
17:07.57 | jeev | you help me with this |
17:07.59 | jeev | it's mutual |
17:08.20 | jeev | Set(HEADER_VALUE=${SIP_HEADER(P-Asserted-Identity)}) |
17:08.22 | jeev | wghen i do that.. |
17:08.36 | jeev | i dunt understand variable string control |
17:08.40 | jeev | oh well, i've got to step out |
17:08.41 | lmadsen | Set(CALLERID(num)=${HEADER_VALUE:5:10}) |
17:08.44 | jeev | i dunno what to do but i'm sure i'll get it |
17:08.45 | jeev | ahh |
17:08.47 | jeev | is that not cut ? |
17:08.51 | lmadsen | nope |
17:09.00 | lmadsen | CUT() is unecessary there probably) |
17:09.05 | jeev | ahh |
17:09.19 | lmadsen | ${VAR:offset:length} |
17:09.22 | jeev | nice |
17:09.23 | jeev | works |
17:09.31 | jeev | thanks so much leif |
17:09.34 | lmadsen | np |
17:09.38 | jeev | i wont beat you up |
17:09.38 | *** join/#asterisk wgarrett (n=wgarrett@cpe-071-076-032-122.sc.res.rr.com) |
17:09.39 | jeev | at astricon |
17:09.47 | lmadsen | ok... I appreciate that :) |
17:09.50 | jeev | everyone else is going to hell! |
17:09.53 | jeev | how big are you ? |
17:10.03 | lmadsen | 5'9" 170 lbs |
17:10.04 | jeev | ahh |
17:10.16 | jeev | ok dood, i've gotta go pick up my friend and bring him home |
17:10.16 | lmadsen | I'm farm boy strong :) |
17:10.18 | jeev | haha |
17:10.21 | jeev | with a farmers tan |
17:10.46 | outtolunc | us farm kids got to stick together |
17:11.23 | outtolunc | has a perm faint farmers tan |
17:11.41 | *** join/#asterisk jplank (n=GBove@ool-18bb018e.dyn.optonline.net) |
17:12.32 | lmadsen | my black belt would help too :) |
17:15.00 | *** join/#asterisk beasty (i=jan@about/apple/macbook/beasty) |
17:15.02 | beasty | hi there |
17:15.20 | beasty | [Sep 6 19:17:13] WARNING[21864]: pbx.c:2968 ast_register_application: Already have an application 'Directory' |
17:15.27 | beasty | can anyone tell me what that mean ? |
17:15.28 | beasty | s |
17:16.36 | tzafrir_laptop | beasty, you're trying to register Directory from both the module app_directory.so and app_directory_odbc.so ? |
17:16.37 | outtolunc | waits for the next book 'Asterisk-Foo: the Black Arts' |
17:16.50 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
17:17.12 | *** join/#asterisk F00JIN (n=F00JIN@lns-bzn-30-82-253-131-242.adsl.proxad.net) |
17:17.16 | tzafrir_laptop | Happen to use the Debian package? If so, see the included modules.conf that by default disables the odbc variant |
17:17.33 | F00JIN | hi! |
17:18.16 | Qwell | tzafrir_laptop: freepbx |
17:18.34 | tzafrir_laptop | what about it? |
17:18.41 | Qwell | it replaces modules.conf |
17:19.04 | *** join/#asterisk shinao1 (n=shinao1@62.173.48.176) |
17:23.53 | *** part/#asterisk F00JIN (n=F00JIN@lns-bzn-30-82-253-131-242.adsl.proxad.net) |
17:25.19 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
17:28.34 | jaytee | lmadsen, did jeev really help you write the book? |
17:30.04 | lmadsen | jaytee: ummm... not that I know of :) |
17:30.22 | jaytee | another one of his grandios claims then. |
17:30.37 | lmadsen | I was drunk most of the time, so who knows, heh |
17:30.43 | lmadsen | is being sarcastic there |
17:31.22 | lmadsen | outtolunc: when I meant black belt... I meant the black leather belt I typically wear |
17:32.08 | *** join/#asterisk c7g (n=c7g@201.53.189.43) |
17:32.09 | jaytee | "Must stay focus, madsen-san!" |
17:32.50 | lmadsen | totally... this system upgrade is... not the most fun :) |
17:33.02 | lmadsen | but now I get to install my clustering stuff into another production server, so that kinda is exciting |
17:33.16 | jaytee | lmadsen, system upgrade? hardware or OS version? |
17:33.39 | lmadsen | dialplan |
17:33.51 | jaytee | uh-huh |
17:33.52 | lmadsen | multi-site asterisk cluster |
17:35.59 | jaytee | in a couple months I'm going be setting up a second * server as a failover for redundancy. I'm thinking of just using rsync from the "hot" system to the "cold" system and having eth1 active on cold with eth0 inactive with the same IP. then I can just flip them after moving cables. Eventually I'll get a T1 failover box and fancy it up with HA and figure out how to run both hot with failover. |
17:36.00 | kdas | I am linking ipkall to a voipbuster account and then ii register on that account with my * box and i have my *box act as a voicemail server, but when i call the server ethier with a analog phone or softphone my *box dosent reconize my input. If some one can help it would be much apreshated. |
17:36.19 | lmadsen | jaytee: yep, that sounds like a good plan |
17:36.34 | kdas | jaytee, :) |
17:36.36 | jaytee | kdas, appreciated not apreshated |
17:37.00 | jaytee | slaps kdas on the hand with his "teacher's ruler" |
17:37.03 | kdas | jaytee, thanks i knew that looked wrong, sorry i just woke up |
17:37.11 | jaytee | more coffee dude |
17:37.16 | kdas | ouch i am sirry |
17:37.26 | kdas | sorry* |
17:37.32 | kdas | lol see i told you! |
17:37.41 | lmadsen | ok, off to grab some lunch! |
17:37.44 | *** join/#asterisk F00JIN (n=F00JIN@lns-bzn-30-82-253-131-242.adsl.proxad.net) |
17:37.51 | *** join/#asterisk ido (n=ido@unaffiliated/ido) |
17:37.53 | jaytee | kdas, when we were going over your problem last night [TK]D-Fender asked for some stuff and you didnt' want to pastebin it. |
17:38.51 | kdas | jaytee, i don't mind pastebining it just i asked him if he is sure we wants like all the packet data etc. and then he didn't reply and you told me he was afk or something |
17:39.38 | jaytee | question I have regarding your problem. when you call from the softphone is it on the same local net as the * server? and if so, what is it's configuration set for in regards to dtmfmode? |
17:41.10 | kdas | jaytee, well yesterday we established that voipbuster is dtmfmode=inband and then yes it is on same local network and i use the local ip 192.168.1.117 in this particuliar case |
17:42.37 | F00JIN | I'd like to know if there's an easy way to upgrade asterisk from 1.4.2 to 1.6 |
17:42.48 | F00JIN | i'm using ubuntu 8.04 |
17:42.59 | jaytee | yes, voipbuster is inband and I wasn't aware of that till [TK]D-Fender pointed that out but what about the softphone itself? calls from it don't go through voipbuster so I want to know what that softphone is set to for dtmfmode. |
17:43.15 | jaytee | F00JIN, not with packages |
17:43.34 | jaytee | you're better off uninstalling and then compiling the source |
17:44.00 | F00JIN | :-\ |
17:44.12 | jaytee | 1.6 is still beta |
17:44.35 | lmadsen | actually release candidate |
17:44.40 | F00JIN | in fact I want to use an mp3 on hold music |
17:44.42 | lmadsen | rc9 I believe |
17:44.46 | jaytee | and 1.4.2 is pretty old which means whoever packaged it for Hardy isn't keeping up to date. |
17:44.54 | lmadsen | 1.4.2 is REALLY old |
17:44.57 | lmadsen | like... almost 2 years |
17:44.58 | jaytee | lmadsen, yup |
17:45.08 | jaytee | on both old and rc9 |
17:45.17 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
17:45.33 | F00JIN | and with asterisk-gui 2.0 a message notify me I need 1.6 to upload and use mp3 |
17:45.39 | lmadsen | F00JIN: btw -- 1.4 --> 1.6 is a major upgrade |
17:45.46 | jaytee | very major |
17:45.58 | lmadsen | its like going from a 1.0 to a 2.0 |
17:46.13 | lmadsen | or going from kernel 2.4 to 2.6 |
17:46.38 | kdas | jaytee, i have x-lite softphone and dosent seem to give me the information or the option for switching the dtmf mode |
17:46.44 | lmadsen | there is no *easy* path -- read UPGRADE.txt and CHANGES in the 1.6.0-rc9 source as well |
17:47.06 | jeev | farmers |
17:47.07 | jaytee | kdas, it would be under the phone's SIP settings |
17:47.07 | jeev | hee haw |
17:47.27 | lmadsen | goes for lunch for real this time |
17:47.30 | jeev | NO |
17:47.33 | jeev | you can't go when i just came |
17:47.34 | jeev | bastage |
17:48.04 | jaytee | kdas, you wouldn't happen to have a hard IP phone to test with would you? it might be nice to know if dtmf detection is working period on your * server. |
17:48.54 | kdas | jaytee, no and the still can't find no dtmf setting in x-lite |
17:49.09 | kdas | jaytee, let me do a google search to see if that soft phone support that |
17:50.01 | kdas | but if i am using a analog phone to dial in wouldn't not matter about the dtmf modes? |
17:50.09 | jaytee | it's been awhile since I've used x-lite. When I do use a softphone I use Ekiga but that only allows RFC2833, it won't let you set it to anything else. probably the same in x-lite. |
17:51.07 | jaytee | if you're configuration is set to inband for your incoming calls from voipbuster then it should work calling from an analog phone. |
17:51.26 | jaytee | what version of * are you running? |
17:52.10 | kdas | ummmi think 1.4 that what you get when you do a apt-get install with ubuntu |
17:52.26 | kdas | and on my other test system i am running the windows version of it |
17:52.38 | jaytee | of ffs! |
17:52.58 | kdas | what ? |
17:53.23 | jaytee | windows? with cygwin? why not just go out and spend the money on bubble gum? |
17:53.23 | kdas | would it solve the problem if i used something other then voipbuster ? |
17:53.48 | kdas | http://www.asteriskwin32.com/ |
17:53.54 | kdas | the other system is ubuntu |
17:54.06 | jaytee | kdas, I don't really know at this point. you can't even get dtmf detection working right with a softphone. I'd say you might have a bad * package. |
17:54.08 | kdas | i have 2 but both fail |
17:54.35 | jaytee | kdas, I'm familiar with asteriskwin. I'd rather use two cans and some string |
17:54.59 | kdas | jaytee, what about this ? http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SIPdtmfmode |
17:55.24 | kdas | jaytee, ok got the point but what about the ubuntu box ? |
17:55.49 | *** join/#asterisk ManxPower (n=manxpowe@195.sub-75-201-155.myvzw.com) |
17:56.14 | ManxPower | waves to everyone |
17:56.26 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
17:57.04 | kdas | waves to ManxPower |
17:57.10 | jaytee | with the 'buntu box I don't know. you've tried relaxdtmf=yes already and that didn't help (usually doesn't) but you haven't proven that dtmf detection works AT ALL so trying to point the finger at voipbuster at this stage isn't going to accomplish anything. |
17:57.30 | jaytee | waves back at ManxPower |
17:57.44 | *** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk) |
17:57.52 | kdas | jaytee, right i understand, but if i switch up servers then maybe it will be easier ? |
17:58.07 | jaytee | switch up servers? |
17:58.15 | ManxPower | kdas: it is not a hardware problem. |
17:58.15 | jaytee | you mean providers? |
17:58.23 | jaytee | it's a software problem |
17:58.34 | ManxPower | jaytee: what is kdas's problem (I ask you because you would be better at explaining it) |
17:58.39 | kdas | jaytee, wait a second i see what you are saying sorry, because the softphone dosent even use voipbuster |
17:59.04 | kdas | jaytee, server, provider its all the same lol i am so sorry maybe i should go make my self some coffe or something |
17:59.16 | jaytee | ManxPower, it's the same as what you pegged it for last night. dtmf problems with voicemail, a mismatch in settings or his * package for Ubuntu isn't working for dtmf detection at all. |
17:59.19 | kdas | ManxPower, well :P i can explain just fine |
17:59.22 | ManxPower | kdas: yes. do that. step away from it for a few mins. |
17:59.29 | jaytee | he can't even get a softphone on his local net to work with it. |
17:59.59 | ManxPower | Ah. Well getting the local phone and the local server seeing the correct DTMF needs to be solved before ANYTHING else. |
18:00.10 | *** join/#asterisk drcode (n=Gadi@85.65.12.156.dynamic.barak-online.net) |
18:00.14 | drcode | hi all |
18:00.21 | jaytee | he uses voipbuster as an incoming ITSP and that uses dtmfmode=inband but his softphone is x-lite. |
18:00.37 | jaytee | ManxPower, exactly. I told him to test locally first |
18:00.58 | drcode | Asterisk with x100p can answer fax and voice (with iaxmodem)? |
18:01.07 | drcode | or send fax also |
18:01.14 | ManxPower | kdas: if the softphone is on the local LAN (and will stay on the local lan), I would use ulaw as the codec and RFC2833 as the dtmf mode. If you just can't get rfc2833 working between your phone and the server, then look at inband for phone/server link |
18:01.20 | jaytee | Ekiga only does RFC2833. don't remember if x-lite offers any additional options. he doesn't have a hard phone to test with. |
18:01.47 | ManxPower | Even with ulaw you can get audio blips occasionally so even with ulaw rfc2833 is best, but inband would work well enough for most people. |
18:02.07 | kdas | ManxPower, i was thinking i should remove all my sip server settings and just test a loca softphone to server connection till we can get this thing right sounds good ? |
18:02.27 | ManxPower | kdas: make sure you have a backup copy of sip.conf. |
18:02.45 | ManxPower | once you do that put the sip.conf on pastebin.ca masking ONLY the passwords. |
18:03.05 | ManxPower | kdas: you have multiple problems. It is good you are willing to solve them one at a time. |
18:03.11 | jaytee | and you'll still need settings for the softphone in sip.conf. I'm assuming you meant the section that refers to your voipbuster account settings in sip.conf. just comment them out with semicolons. |
18:03.21 | kdas | yea |
18:03.32 | kdas | one second please gentlemen |
18:03.35 | ManxPower | jaytee: x-lite allows at least rfc2833 and inband I think. |
18:04.19 | jaytee | and if you really want to make progress, quit wasting time with asteriskwin and focus on * on linux. Most of us here swear by compiling over package installs. |
18:04.40 | ManxPower | Huh? He's trying to run Asterisk on Windows? |
18:04.58 | kdas | ManxPower, no i have 2 servers a windows and ubuntu |
18:05.00 | jaytee | ManxPower, I thought it did but I haven't messed with it in almost a year. I had a copy of the free version but can't find it anymore. just have eyebeam at work. |
18:05.29 | lesouvage | kdas: if you use xlite don't to disable the option "silence surpression". I'm nt sure under what name it is part of the configuration option but as I remember it is something like "don't send silence". |
18:05.33 | ManxPower | AsterWin was an april fools joke that got out of hand. It is and never was an actual port. |
18:05.37 | *** part/#asterisk drcode (n=Gadi@85.65.12.156.dynamic.barak-online.net) |
18:05.43 | lesouvage | don't = don't forget |
18:06.04 | jaytee | lesouvage, good point |
18:06.09 | ManxPower | "transmit silence" I think was the term |
18:06.16 | ManxPower | It would not have an effect on DTMF, however. |
18:06.52 | lesouvage | MaxPower: that is through I guess but it results in a general sound quality problem. |
18:06.54 | jaytee | no, but it sure messes with voice recognition. |
18:07.43 | kdas | ok here is how sip.conf looks now to edit my extensions |
18:07.44 | kdas | http://pastebin.com/d3ca12548 |
18:08.25 | ManxPower | kdas: why are you using non-standard port? |
18:09.35 | kdas | ManxPower, should i change from 5070 to 5060 ? |
18:09.41 | ManxPower | kdas: http://pastebin.com/m2ba3a5d6 |
18:09.44 | ManxPower | leave it out. |
18:10.10 | lesouvage | I would start with allow=all and if that works narrow the codes available. |
18:10.18 | ManxPower | lesouvage: then you would be wrong. |
18:10.31 | ManxPower | allow=all can easily make things not work. |
18:10.47 | lesouvage | Manxpower: why is that? |
18:10.48 | ManxPower | In fact it is the worst thing you can do during troubleshooting, next to formatting your system. |
18:11.09 | kdas | ok here is extensions.conf http://pastebin.com/d7da4647b |
18:11.13 | ManxPower | lesouvage: because Asterisk supports several codecs in passthru only modes and if your call uses one of those it will fail. |
18:11.45 | kdas | ManxPower, LOL @ formatting system |
18:11.50 | lesouvage | ManxPower: thanks for the expenation. I think I have been lucky in the past. |
18:11.51 | ManxPower | kdas: that should work. Now you need to go into your softphone and set it up for the port and the DTMF mode. |
18:12.00 | ManxPower | lesouvage: you have. |
18:12.04 | lesouvage | and the codec |
18:12.22 | kdas | ManxPower, ok well my softphone don't have a dtmf setting unless i havent disconvered the easter egg yet |
18:12.36 | ManxPower | lesouvage: no, just leave all the codecs enabled one the softphone and let Asterisk's allow/disallow settings determine the codec. |
18:12.38 | jaytee | damn! I didn't even spot the non-standard port number last nite. |
18:12.53 | kdas | jaytee, you need coffee too! |
18:12.53 | ManxPower | kdas: all softphones have dtmf modes that I know of. |
18:13.02 | jaytee | sucks when you're old and tired |
18:13.07 | ManxPower | which softphone are you using? |
18:13.14 | kdas | ManxPower, i know i just don't know were to find setting or which dtmf mode it uses |
18:13.23 | kdas | ManxPower, x-lite |
18:13.41 | jaytee | x-lite for linux or windows? |
18:13.49 | ManxPower | lesouvage: if you manually force your phone to a specific codec then allow=all does not matter. I prefer to force the stuff on the asterisk side and leave as much of the phone config alone as I can. |
18:14.43 | kdas | ManxPower, how am i to call my systems vmail if i don't have any outgoing rule in the extensions.conf ? |
18:15.00 | jaytee | ? |
18:15.12 | jaytee | from a softphone the call isn't outgoing |
18:15.13 | kdas | http://pastebin.com/d7da4647b is my extensions.conf |
18:15.32 | ManxPower | kdas: because you are not making an outgoing call |
18:15.45 | ManxPower | kdas: however "s" is not what you want. |
18:15.56 | ManxPower | I would make it exten => 1234 or something like that. |
18:16.06 | kdas | ok i am a little confused i have x-lite up and it is registered to my * box now what ? |
18:16.09 | ManxPower | "s" is almost never used unless you have FXO signalled ports. |
18:16.21 | ManxPower | kdas: looks like x-lite defaults to rfc2833. |
18:16.21 | kdas | ok let me change |
18:16.48 | ManxPower | kdas: http://pastebin.com/m1cd22534 |
18:16.56 | ManxPower | then dial 66 from your x-lit |
18:17.13 | ManxPower | make sure to restart Asterisk as we made some sip.conf changes that should require it. |
18:17.40 | jaytee | that looks like an 88 to me with these bifocals. maybe I need a new prescription? |
18:17.54 | kdas | it is 88 last time i checked |
18:17.56 | ManxPower | jaytee: you do. |
18:18.15 | ManxPower | http://pastebin.com/m1cd22534 <-- has 66 |
18:18.45 | jaytee | I'm goin to Walmart next time. Lenscrafter raped me for 300 bucks for the last pair and their scratch resistant coating is worthless. |
18:18.46 | kdas | ok when i dial 88 on my x-lite phone it gives me the "the person you calling is unavaliable" message |
18:19.01 | ManxPower | kdas: correct. it's 66 not 88 |
18:19.18 | ManxPower | kdas: and is that a text message on the screen or an audio message? |
18:19.45 | ManxPower | do you get anything on the CLI? |
18:19.47 | kdas | ManxPower, sorry it is 66 |
18:19.59 | ManxPower | ext message on the screen or an audio message? |
18:20.04 | ManxPower | text message that is. |
18:20.06 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
18:20.06 | jaytee | swear to god it's 88 in the extensions.conf he pasted |
18:20.08 | kdas | ok one sec |
18:20.17 | ManxPower | jaytee: it is 66 in the extensions.conf *I* pasted. |
18:20.21 | kdas | YAY it works!!!! |
18:20.41 | ManxPower | kdas: now try logging into voicemail. |
18:20.50 | ManxPower | see if it recognizes your DTMF |
18:20.54 | jaytee | ah, ok that one. yes. I was looking a the one he pasted with the s extension |
18:20.56 | kdas | it does |
18:21.07 | jaytee | well, that's progress. |
18:21.28 | kdas | yea i was confused because i didn't notice my self the differance |
18:21.33 | ManxPower | create a mailbox and password that is more than 2 digits. Sometimes DTMF can work 4 out of 5 digits and stuff like that. |
18:21.47 | ManxPower | Just to be sure and it takes only a few moments |
18:22.19 | ManxPower | We want to make 100% sure that we have good DTMF on the local side before we try the service provider side. |
18:22.22 | kdas | well it is working now yesterday and just a lil before it was dead like a corpse |
18:22.37 | kdas | ok so i am to make a password longer then 4 didgits ? |
18:22.56 | kdas | just for a side note you kept the |s option in your pastebin ;) |
18:23.15 | ManxPower | kdas: you can make the password as long as you want (subject to Asterisk's internal limits which I suspect is something abour a 64 digit password |
18:23.34 | kdas | ok but what did you want me to test now ? |
18:23.48 | ManxPower | kdas: |s doesn't really matter either way. you should use , instead of | as pipe(|) is going away in 1.6 |
18:24.18 | *** join/#asterisk rconnect__ (n=rconnect@85.138.128.75) |
18:24.23 | ManxPower | I guess now we should work on your provider. Do you have a cell phone or analog telco phone to test incoming calls with? |
18:24.46 | ManxPower | The provider part is the hard part. |
18:24.53 | ManxPower | what provider do you use? |
18:25.21 | kdas | ok thanks for head ups with the | |
18:25.43 | jaytee | he uses voipbuster |
18:25.51 | kdas | i have normal tele phone |
18:26.02 | kdas | i am using voipbuster temp |
18:26.03 | jaytee | which needs dtmfmode=inband |
18:26.34 | kdas | but i am looking for a good sip provider ... if you know any ? |
18:26.53 | kdas | terrasip and voipbuster i am using now because they are free for a while... |
18:27.10 | *** join/#asterisk trumee (n=trumee@cpc1-seve11-0-0-cust723.popl.cable.ntl.com) |
18:27.23 | trumee | guys, i am trying to setup * using pbxinaflash. I have setup an IAX trunk. I can call out successfully, but do not receive any incoming calls. any ideas? |
18:27.30 | ManxPower | I don't use voip over internet (I work in the business industry and they want rock solid and reliable). |
18:27.47 | jaytee | PRI is the only way to go :-) |
18:27.48 | ManxPower | When I have to use an ITSP I usually recommend Vitelity |
18:27.53 | trumee | i am using freepbx btw. |
18:28.03 | TJNII | ~freepbx |
18:28.03 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
18:28.04 | jaytee | trumee, wrong channel |
18:28.05 | ManxPower | jaytee: I don't do installs unless they are PRI |
18:28.12 | kdas | ManxPower, so what do you use asterisk for hooking up normal lines ? |
18:28.16 | trumee | jaytee:why is that? |
18:28.18 | ManxPower | kdas: PRI |
18:28.32 | kdas | PRI = what ? forgive my ignorance |
18:28.33 | jaytee | ManxPower, if I was doing installs I'd be take the same approach. |
18:28.51 | ManxPower | PRI is a type of voice T-1 with advanced signalling on it. |
18:28.55 | trumee | jaytee:is there a way to debug iax2 incoming calls. |
18:29.07 | jaytee | ITSPs are too unreliable for the most part for real world businesses |
18:29.20 | ManxPower | jaytee: It's less the ITSPs and more the internet. |
18:29.28 | jaytee | ManxPower, true |
18:29.29 | trumee | jaytee:i did do a iax2 debug on. But if call my DID it doest give any trace at all. |
18:29.31 | TJNII | ~freepbx |
18:29.32 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
18:29.36 | TJNII | trumee: ^^^^^^ |
18:29.42 | trumee | i did do a iax2 debug on. But if call my DID it doest give any trace at all. |
18:29.42 | ManxPower | heck even without voip my customers frequently have problems even connecting to some sites because some carrier is having a spat with another carrier. |
18:30.18 | ManxPower | trumee: If #freepbx support is so terrible maybe you should consider a different product? |
18:30.21 | kdas | ManxPower, well i was thinking if i shutoff my normal phone i can upgrade my internet speed to 16mb download and X upload should i try using that and something differnt then voip ? |
18:30.42 | jaytee | even with SIP over TCP instead of RDP the internet isn't setup to prioritize voip traffic so using the PSTN network is the only way to guarantee a higher reliablity. |
18:30.49 | kdas | ManxPower, I love your subtle little jokes like the formating and the another product LOL |
18:30.57 | trumee | ManxPower:freepbx irc channel is so dead. is there any other GUI which has better community support? |
18:31.04 | jaytee | quick and funny, ain't he cool! |
18:31.05 | ManxPower | kdas: voip is OK for home use. You are not gong to be screaming at yourself because you screwed up and you are going to lose a 4 million dollar contract becaus the phones are down. |
18:31.15 | ManxPower | trumee: none of us here use GUIs. |
18:31.21 | kdas | point |
18:31.50 | jaytee | gui's are sticky and you end having to wash your hands and recode your dialplan constantly. yuch! |
18:31.53 | ManxPower | My significant other uses a SIPura ATA, a cordless phone, and VoIP to Vitelity. |
18:32.05 | trumee | ManxPower:ok. what basic files should i replace to get the basic asterisk back. i have another system (gentoo) on which i have asterisk setup |
18:32.15 | ManxPower | works 9 out of 10 times he picks up the phone. |
18:32.32 | ManxPower | trumee: you must replace everything in /etc/asterisk |
18:32.36 | kdas | * uses 5060udp correct, i am trying to setup my DID to contact my *box directly |
18:33.11 | trumee | ManxPower:ok, i guess i will gzip /etc/asterisk and copy over vanilla asterisk files |
18:33.29 | ManxPower | kdas: correct. You need to port forward 5060/UDP and 10000 - 10020 and make sure that range is reflected in /etc/rtp.conf |
18:34.07 | ManxPower | trumee: but almost nobody here will want to help you. The reason most people use a GUI is they don't know enough to not use a GUI. |
18:34.29 | ManxPower | so if you want to take the time to learn some stuff then go read the Asterisk book |
18:34.31 | ManxPower | ~book |
18:34.32 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
18:34.52 | ManxPower | kdas: all voip with Asterisk is UDP, BTW. |
18:35.29 | kdas | ManxPower, ok so 10000 -10020 udp is for what ? |
18:35.35 | lmadsen | RTP |
18:35.38 | lmadsen | (media) |
18:36.21 | lmadsen | jbot: no, book is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook |
18:36.22 | jbot | lmadsen: okay |
18:36.38 | lmadsen | that comma after the tfot link seemed to break for me |
18:36.49 | kdas | ManxPower, to use ipkall i just need to put HOSTNAME:5060 no need 10000-10020 right ? |
18:36.54 | ManxPower | kdas: Audio |
18:37.19 | ManxPower | kdas: huh? Don't get ahead of yourself. we havea lot of work to do before we start making calls. |
18:37.36 | jeev | hip hip, HORRAYYYYYYYYYYYYY |
18:37.39 | jeev | ManxPower, i'm gonna do GRE |
18:37.41 | jeev | in a smidge |
18:37.44 | ManxPower | do you have the ports forwarded in your firewall and have you set the /etc/asterisk/rtp.conf to match? |
18:37.58 | jeev | http://www.pointless.net/~jasper/consume/docs/my-docs/tunneling.html looks easier than jaytee's sister |
18:37.59 | ManxPower | jeev: Awesome! Let me know if it works well for you. |
18:38.36 | kdas | ManxPower, well on loacal net all ports open so no problem i was just trying to deal with ipkall |
18:38.57 | ManxPower | kdas: now we need to set the localnet= and externip= in sip.conf |
18:39.02 | kdas | ManxPower, i was just going to test the analog phone |
18:39.21 | ManxPower | kdas: we are not even registered to a provder yet, no incoming calls will work. |
18:39.30 | ManxPower | kdas: what is your local lan network and netmask? |
18:39.54 | ManxPower | lets get outgoing calls working first and get registration working to your provider. |
18:40.01 | kdas | ManxPower, isn't if you have ipkall go directly to your asterisk box you don't need a provider right ? |
18:40.28 | ManxPower | ipkall IS a provider |
18:40.40 | kdas | ipkall = DID provider |
18:40.43 | ManxPower | but it's NOT GOING TO WORK until we have the NAT issues worked out. |
18:41.00 | ManxPower | kdas: you did so well, don't screw up by getting ahead of your self. |
18:41.00 | kdas | ok well lets do that then you know best :) |
18:41.25 | ManxPower | (1:39:29 PM) ManxPower: kdas: what is your local lan network and netmask? |
18:41.33 | kdas | ok so i am on 192.168.1.X 255.255.255.X network |
18:41.43 | ManxPower | and with is your external IP address |
18:41.58 | ManxPower | so you would put localnet=192.168.1.0/255.255.255.0 |
18:42.09 | ManxPower | as the last line in [global] in sip.conf/ |
18:42.14 | kdas | 67.161.44.222 = externip |
18:42.27 | kdas | external* |
18:42.33 | ManxPower | we also have to set externip=67.161.44.222 in sip.conf right before or right after the localnet line |
18:43.42 | kdas | ok |
18:43.49 | ManxPower | kdas: looks like IPkall does not give you any setup instructions. |
18:44.08 | kdas | does asterisk respect the placement of the lines? |
18:44.34 | ManxPower | kdas: sort of. |
18:44.39 | kdas | lol ok |
18:44.52 | ManxPower | isipkall your only provider? |
18:44.59 | ManxPower | their setup seems like a nightmare |
18:45.14 | drmessano | LOL |
18:45.18 | kdas | ipkall is DID provider and as far as the sip goes i need to find a reliable good one |
18:45.19 | drmessano | IPKALL is easy to set up |
18:45.49 | ManxPower | standby, I'm looking at some stuff. |
18:45.51 | drmessano | You tell it which extension to send the calls to, and which proxy.. |
18:46.05 | kdas | but as of now i am just using voipbuster and terrasip becasue it is free as of now |
18:46.07 | drmessano | All calls come from voiper.ipkall.com |
18:46.21 | drmessano | Just need to accept the calls |
18:46.43 | ManxPower | drmessano: how do they know your IP address? |
18:46.48 | drmessano | From the Web UI |
18:47.00 | drmessano | <drmessano> You tell it which extension to send the calls to, and which proxy.. |
18:47.15 | kdas | ManxPower, you set it on there site on voip-info it tells how to set it up |
18:47.15 | ManxPower | kdas: ok, first you say you are only using ipkall, now you are saying you use voipbuster and terrasip. Which is it? |
18:47.24 | ManxPower | kdas: how 1980s of them |
18:47.30 | kdas | ManxPower, lol |
18:47.43 | ManxPower | kdas: put your current sip.conf on pastebin as well as your current extensions.conf. |
18:47.52 | kdas | ManxPower, will you be here like in 10 mintues ? |
18:47.59 | ManxPower | kdas: maybe. why? |
18:48.08 | drmessano | IPKALL works great for free |
18:48.10 | kdas | ManxPower, i need to do something real fast |
18:48.19 | drmessano | He has to poop |
18:48.21 | ManxPower | kdas: sure. can you put up the pastebins first? |
18:48.25 | kdas | ok |
18:48.44 | kdas | so far only sip.conf changed i will put it up |
18:48.58 | ManxPower | OK. |
18:49.08 | kdas | http://pastebin.com/m4acfb9a |
18:49.18 | kdas | ok brb |
18:55.44 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
18:55.52 | jeev | ManxPower, you suggest doing ipsec over gre for this? |
18:56.09 | ManxPower | jeev: not at this time. Maybe later. |
18:56.42 | ManxPower | jeev: we are not doing the tunnel for security. We are doing it to make your stuff 10x less complicated. 8-) |
18:57.06 | ManxPower | jeev: ipsec would just add latency at this point. |
18:57.24 | jeev | hm |
18:57.34 | jeev | only problem is that |
18:57.40 | jeev | the internal asterisk is dynamic. |
18:57.48 | jeev | ip |
18:58.15 | drmessano | NO-IP |
18:58.24 | ManxPower | jeev: that sucks. |
18:58.30 | *** join/#asterisk shinao1 (n=shinao1@62.173.48.176) |
18:58.50 | ManxPower | getting a static IP would make things easier as well. |
18:59.14 | jeev | hmm, or maybe this will help ease the dual wan situation |
18:59.16 | jeev | let me see |
19:00.10 | ManxPower | jeev: there are other ways to accomplish what you want without tunnels, but they are much more complicated on the Asterisk setup. |
19:00.51 | kdas | ManxPower, ok back so what is your verdict on ipkall? |
19:00.57 | ManxPower | http://pastebin.com/m707d7690 use this sip.conf |
19:01.05 | kdas | me? |
19:01.13 | ManxPower | give me the url for the past extensions.conf so I can modify it. |
19:01.15 | ManxPower | kdas: yes |
19:01.33 | jeev | yea, i know |
19:01.59 | kdas | ok cool |
19:02.40 | kdas | your sip phone number here = my DID number? |
19:03.12 | ManxPower | kdas: I don't think so, the docs I found are vague. It says "Where 508 is the SIP phone number you specified when setting up IPKall." |
19:03.20 | ManxPower | [508] is on the example. |
19:03.49 | ManxPower | kdas: if you don't know what it is, we can find out when we debug a failed call |
19:04.02 | kdas | oh you must have read wrong docs one sec let me find the ones i read |
19:04.14 | jeev | k, pf is blocking the packets, let me add the rule and :D |
19:04.54 | ManxPower | kdas: doesn't really matter, just use the sip.conf I gave you and get me a copy of your extensions.conf so I can add the required lines for that. |
19:04.54 | hardwire | Anybody ever used Alepo RBS? |
19:05.00 | kdas | http://www.voip-info.org/wiki-IPKall |
19:05.14 | kdas | ok if you read that it will tell you how to set it up |
19:05.30 | ManxPower | kdas: the exact same document I was looking at. |
19:05.36 | ManxPower | and you see the [508] in the example. |
19:05.42 | ManxPower | and I quoted off that page as well. |
19:05.42 | kdas | lol your right |
19:05.52 | kdas | i was using the top half though |
19:05.55 | ManxPower | so we are back exactly where we started, me waiting for your extensions.conf. |
19:05.58 | kdas | let me re-read it real fast |
19:06.17 | kdas | my extensions.conf is same as before rememeber ? |
19:06.32 | ManxPower | kdas: yes, but I don't remember the 100 char URL for me to find it on pastebin again. |
19:06.41 | *** join/#asterisk rhombus (n=sfbosch@dsl-vlan435-66-18-218-36.nucleus.com) |
19:06.50 | ManxPower | so either tell me the URL again or paste it again. |
19:06.58 | ManxPower | I really don't care which. |
19:07.41 | jeev | 1. 001027 rule 102/0(match): block in on gre0: 192.168.20.38 > 192.168.20.37: ICMP echo request, id 58689, seq 210, length 64 |
19:07.42 | jeev | yayayay |
19:08.08 | jeev | yayyyyyyyyy |
19:08.16 | jeev | 64 bytes from 192.168.20.37: icmp_seq=250 ttl=64 time=7.016 ms |
19:08.17 | jeev | :D |
19:08.27 | jeev | ManxPower, i try the rest later. |
19:08.32 | kdas | ManxPower, http://pastebin.com/m7152c559 |
19:08.32 | jeev | the reconfiguration |
19:08.33 | jeev | :( |
19:08.41 | ManxPower | jeev: you are making major progress,. |
19:08.44 | jeev | hmm wtf |
19:08.53 | jeev | i am confused on my next step |
19:08.57 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
19:08.58 | jeev | i'm converting from iax to sip, that's it ? |
19:09.00 | kdas | ManxPower, let me get the extension.conf for you |
19:09.04 | rhombus | I have a callcentric trunk that is working for inbound calls, but for outbound calls, I hear no sound. |
19:09.21 | ManxPower | kdas: I'm not doing or saying anything more until I get extensions.conf |
19:09.29 | rhombus | ... in either direction. |
19:09.35 | jeev | run a sip debug |
19:09.39 | jeev | and look to see for errors |
19:09.46 | kdas | http://pastebin.com/d2bfe5d9 = extensions.conf |
19:11.15 | rhombus | jeev: ok |
19:12.22 | *** join/#asterisk ElSonico (n=tav@xdsl-179-75.nblnetworks.fi) |
19:12.33 | ManxPower | http://pastebin.com/m394c3063 notice all the extra stuff, make sure all the . and , are correct when you copy it to your system |
19:13.00 | ManxPower | kdas: then do a reload and try calling your number. chances are it won't work, but we are doing things step by step. |
19:13.21 | kdas | ManxPower, i love your way of thinking ;) step by step |
19:13.21 | ManxPower | kdas: ignore the stuff about _. not being good when you do the reload. |
19:13.31 | kdas | lol ok |
19:13.54 | ManxPower | _. is almost never a good idea, except when it is a catch all and no other extensions exist in the context. |
19:14.08 | TJNII | ManxPower: [INVALID} <- Typo? |
19:14.22 | lmadsen | yes |
19:14.23 | ManxPower | TJNII: good catch! yes it should be [INVALID] |
19:14.31 | kdas | ok |
19:15.05 | ManxPower | kdas: if you get any CLI output when you try calling the DID, pastebin it. |
19:15.35 | kdas | ok sec |
19:15.45 | *** join/#asterisk kj4acm (n=kj4acm@24-183-225-98.dhcp.kgpt.tn.charter.com) |
19:15.55 | kdas | manx it works!!! |
19:16.06 | kj4acm | my googlefoo isn't very strong here... is there a way to subscribe 1 extension to multiple mailboxes ? |
19:16.09 | kdas | ManxPower* |
19:16.11 | ManxPower | kdas: nifty! |
19:16.33 | ManxPower | pastebin the cli output so I can see if I can find any potential future problems. |
19:16.52 | ManxPower | kdas: I've been using Asterisk heavily since late 2001. I've done this a few times before. 8-) |
19:17.11 | kdas | bows to ManxPower the * guru |
19:17.12 | kdas | lol |
19:17.20 | kdas | u sure know your shit |
19:18.15 | ManxPower | kdas: Asterisk pretty much accepts any incoming connections, that's why we use INVALID in [general] to make sure those calls NEVER get into your real dialplan. |
19:18.48 | kdas | ManxPower, yea i understood that i just never thought of it, its a nice security touch ;) |
19:18.50 | ManxPower | kdas: [TK]D-Fender is also a guru, but he sometimes likes sip debug information a little too much. |
19:19.03 | ManxPower | If it had not worked, we would have used sip debug to diagnose it further. |
19:19.39 | kdas | ManxPower, umm the CLI output is prettly clear it just has the verbose msg that we have about the incoming call and then it gives us a waring on using "_." |
19:19.40 | jplank | which side of the call does pickup() pick up? |
19:19.53 | ManxPower | jplank: the asterisk side. |
19:20.14 | ManxPower | of whatever the first call that is ringing in the ring/call group |
19:20.31 | jplank | I can't use it to pick up a specific extension? |
19:20.49 | *** join/#asterisk Defraz (n=T0tal@24-117-156-21.cpe.cableone.net) |
19:21.04 | ManxPower | jplank: not with Pickup(). If your versions as DirectedPickup() use that. I'll double check for you |
19:21.33 | ManxPower | kdas: the verbose message is something like Noop(Call to XXX) |
19:22.10 | kdas | exten => 66,1,Verbose(1|--Random Incoming Call--${CALLERIDNUM}) |
19:22.35 | jplank | I thought pickup() was a directed pickup? |
19:22.35 | ManxPower | jplank: I was wrong. Pickup is directed pickup. I was thinking of park/pickup. |
19:22.57 | ManxPower | kdas: nifty. |
19:23.02 | jplank | I can grab a call from a pickup group |
19:23.14 | ManxPower | kdas: now MAKE A BACKUP COPY of the files. |
19:23.14 | jplank | I just can't pick up a specific extension |
19:23.17 | kdas | ManxPower, ;) ok so what sip/voip provider do you suggest for me ? |
19:23.30 | ManxPower | jplank: remember extension@context |
19:24.01 | kdas | ManxPower, backup complete |
19:24.24 | jplank | I get a unknown, no matter what context I use |
19:24.25 | *** join/#asterisk UD (i=OHNOUDID@unaffiliated/underdawg) |
19:24.30 | ManxPower | kdas: I really don't have a suggestion other than Vitelity, but there are many providers out there. Vitelity's web account management and DID management is a little wonky at first, but you get used to it. |
19:24.41 | ManxPower | jplank: weird. |
19:24.55 | kdas | ManxPower, does it support CID spoofing ? |
19:25.09 | ManxPower | And they were VERY helpful when I totally screwed up my account settings. |
19:25.09 | jplank | if I don't put a context, doesn't it use the context that I'm calling from? |
19:25.14 | ManxPower | kdas: I have no idea. |
19:25.30 | ManxPower | jplank: always specify the context 8-) |
19:25.38 | kdas | ManxPower, i read alot about CID/ANI spoofing and i have a quick qestion, a lil off topic but you seem to have the knolege.. |
19:25.51 | kdas | knowledge* |
19:26.14 | jplank | the context the ringing extension is in, or the one where the call came from? (I assume the ringing extension) |
19:26.21 | ManxPower | jplank: once you get it working, try it without the context. The "calling context" can be pretty complicated when you have includes, etc. |
19:26.35 | ManxPower | no, pickup does not know what the ringing extension is |
19:26.40 | ManxPower | until you tell it. |
19:26.44 | kdas | ManxPower, when a cell provider offers free incoming and out going to mobiles on the same network is that based on ANI or CID ? |
19:26.51 | ManxPower | kdas: no. |
19:26.57 | jplank | this is whats confusing me |
19:27.04 | jplank | when I do extension@context |
19:27.16 | jplank | do I do the ringing extension @ the calling from context? |
19:27.19 | kdas | ManxPower, huh ? |
19:27.23 | ManxPower | give us the ACTUALL stuff |
19:27.40 | jplank | or the ringing extension @ the ringing extensions context |
19:28.01 | ManxPower | lets say a call comes in and is sent to exten 666 in context [hell] If you want to pick up that extension using pickup you would use Pickup(666@hell) |
19:28.30 | Strom_M | 666 hell? sounds familiar |
19:28.34 | ManxPower | kdas: no, the carriers do not use CLID/ANI to determine if the call is on the same network |
19:28.34 | jplank | I'll try it again, but I tried that and kept getting a unknown |
19:28.49 | ManxPower | jplank: I've not used pickup much. |
19:29.06 | ManxPower | Strom_M: yes, I just copied my customer's info. 8-) |
19:29.12 | jplank | so something like exten => _**.,1,Pickup(${EXTEN:2}@from-test) *should* work |
19:29.13 | Strom_M | heheh |
19:29.24 | kdas | ManxPower, what do they use ? |
19:29.34 | jplank | assuming ${EXTEN} is in from-test |
19:29.51 | ManxPower | kdas: the tower, the ESSN, the PRL, any number of things cell networks have that users never see. |
19:30.04 | ManxPower | jplank: that's what the docs say. |
19:30.34 | jplank | yea thats what I read |
19:30.38 | jplank | also doesn't work :( |
19:30.41 | ManxPower | is the ACTUAL exten => line in [from-test] or is there an include => realcontext in the [from-test] context? |
19:30.47 | kdas | ManxPower, i see so no way in tricking the bill system... smart |
19:31.41 | jplank | the actual exten => line is in from-test |
19:31.50 | ManxPower | kdas: a few carriers use CLID/ANI to bypass your voicemailbox auth requirement (the idea is that if you are calling voicemail from your cell, you must want to log in and a password would be just so much of a bother for the customer). Not many these days do it that way. |
19:31.58 | ManxPower | jplank: I'm stumped. |
19:32.07 | jplank | as am I |
19:32.21 | jplank | the last thing I need to do to get BLF working perfectly :( |
19:32.29 | kdas | ManxPower, ok i understand thanks |
19:33.35 | kdas | ManxPower, i can't seem to find the rates for USA and stuff on the Vitelity website |
19:33.36 | ManxPower | ~manxpower |
19:33.36 | jbot | somebody said manxpower was NOT an employee of Digium. He is looking for a training/teaching job in networking and/or Asterisk. Currently doing Asterisk and WAN consulting. Contact: eric@fnords.org |
19:34.22 | ManxPower | kdas: all of use is one rate. |
19:34.28 | ManxPower | ..er.. all of the USA are one rate. |
19:34.41 | ManxPower | 1.9 cents, I think |
19:35.00 | ManxPower | http://vitelity.com/index.php?p=retailserv |
19:35.19 | ManxPower | 1.44 cents/min actually. |
19:35.37 | kdas | ManxPower, but what about services like skype or voipbuster witch is free as long as you have 5$ on account ? |
19:35.49 | ManxPower | kdas: what about them? |
19:36.17 | ManxPower | Skype is a closed protocol closed source, proprietary protocol. You get what you pay for. |
19:36.19 | jeev | flowroute.com is cheap |
19:36.26 | kdas | ManxPower, lol just asking why pay 1.44 rather then free ? |
19:36.38 | ManxPower | you get what you pay for. |
19:36.51 | kdas | ManxPower, point |
19:37.23 | ManxPower | my significant other's phone bill is like $5-$10/month and that includes a DID, incoming/outgoing, e911, CNAM, and several other features. |
19:37.46 | kdas | ManxPower, so my next step is in getting a Voip provider and using it for my outbound calls and then fixing up my dialplan so people don't just get my vmail each time correct? |
19:38.05 | ManxPower | kdas: yup. Go read The Book. |
19:38.07 | ManxPower | ~book |
19:38.08 | jbot | well, book is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook |
19:38.21 | *** join/#asterisk j0 (n=dan@S01060016b6b541d2.va.shawcable.net) |
19:38.54 | kdas | ManxPower, just the DID is 7.XX$ and everything else might be extra 5$ so i am guessing a 14-20$ bill right ? |
19:38.56 | j0 | on samgona analog cards with an additional remora board, should the red led's turn on for the additional boards right away? i've hooked power up to the main board too |
19:40.06 | kdas | ManxPower, i would to thank you sooooo much for your help you kick ass i been wondering from channel to channel and person to person and they all could not help me and you just straight up kicked ass with out making me reporgram the source to asterisk so 2 thumbs up for you |
19:40.20 | Strom_M | kdas: send him some cash |
19:40.22 | jplank | I can't think of a logical place to set pickupmark to use with pickup(), seems like it would be a pain to use with DID routing, would have to add it to ever DID translation |
19:40.32 | ManxPower | kdas: huh? dids are lile $1.99/month |
19:40.38 | kdas | Strom_M, how much and how ? |
19:40.49 | Strom_M | ask him |
19:40.51 | ManxPower | paypal donations are welcome to eric@fnords.org |
19:41.06 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
19:41.07 | kdas | errrg need to setup paypal account |
19:41.40 | ManxPower | kdas: the $7.xx/month is for unlimited inbound. |
19:41.49 | kdas | ManxPower, yea |
19:42.08 | kdas | ManxPower, so what she gets the 1.2cents a minute incoming ? |
19:42.37 | jplank | residential DID's charge inbound? |
19:42.51 | ManxPower | so to break even you would have to have 64 incoming mins per month to break even on the $7.95/month |
19:43.05 | ManxPower | jplank: yes, in the voip world. |
19:43.12 | jplank | hmmmm |
19:43.17 | jplank | thats kind of messed up |
19:43.54 | jplank | most telco's make money off the inbound call from the sending carrier |
19:43.58 | kdas | ManxPower, that is 1hr 4min for 7$ |
19:43.59 | jplank | and they are charging the end user |
19:44.03 | jplank | screwed up |
19:44.13 | kdas | ManxPower, i talk more then that every month |
19:44.18 | ManxPower | oh wait! you would need 662 mins per month to break even on the $7.95 plan |
19:44.30 | kdas | ManxPower, that is better |
19:44.39 | ManxPower | you can calc it if you want. 7.95 / 0.012 |
19:44.57 | ManxPower | My first number did 12 cents/min, not 1.2 cents/min |
19:45.23 | jplank | 0.12 and 0.012 biiiig difference :) |
19:45.47 | ManxPower | jplank: yeah, I should know better. |
19:45.54 | jplank | everyone does it |
19:45.59 | jplank | I find myself doing it all the time |
19:46.16 | ManxPower | jplank: in the telcom world a simple mistake can cost you thousands of dollars |
19:46.29 | jplank | forget about it |
19:46.44 | ManxPower | 8-) |
19:46.50 | jplank | VZ missed rated 200 of our calls by one decimal place |
19:47.07 | jplank | turned 400000 minutes into 1.1 million |
19:47.11 | ManxPower | jplank: how much extra did they try to bill you for? |
19:47.16 | jplank | a lot |
19:47.36 | jplank | turned calls that lasted minutes into hours |
19:48.11 | jplank | we only noticed it because when we did our bill run our minutes jumped almost 300% in one month |
19:48.16 | ManxPower | jplank: you have to wonder how many people had the same thing happen to them and did not catch it. Did you at least report the issue to the PUC/PSC? |
19:48.54 | jplank | it was a simple mistake I guess, my company is a CLEC and VZ is our interconnect for local calls |
19:49.01 | jplank | so we just had them fix it |
19:49.14 | jplank | I could only imagine how many people get screwed by it |
19:49.22 | jplank | we only noticed because we have a high volume of calls |
19:49.30 | jplank | someone with a small volume would never notice |
19:49.33 | vipcarrier | what do u use for CDR mediation |
19:49.53 | jplank | times that by 1000's of end users and VZ makes out with $$$ |
19:50.04 | jplank | well we use a billing company calls profitec |
19:50.14 | ManxPower | jplank: and VZ should be nailed to the wall by the PSC/PUC |
19:50.27 | Strom_M | "times that by"? what the fuck ever happened to the word "multiply" |
19:50.52 | vipcarrier | what do u use for CDR mediation |
19:51.03 | rhombus | Strom_M: " |
19:51.10 | jplank | I can't say if it happened to everyone, I just know it happened to us, actually twice, first time it was on one call, that happened to be international, which was easy to spot, and the next time it happened on a ton of calls |
19:51.27 | jplank | vipcarrier: if your talking to me, like I said, we use a company called profitec |
19:51.28 | rhombus | Strom_M: "times" is shorter than "multiply" -- easier to type, too. |
19:51.32 | kdas | ManxPower, do you use any Instant messengers ? yahoo google etc? |
19:51.43 | Strom_M | rhombus: and dumber-sounding, to boot! |
19:51.45 | drmessano | Strom_M: "times that" is from 4th grade |
19:51.46 | ManxPower | kdas: a few, occasionally |
19:51.54 | jplank | lol |
19:51.54 | rhombus | Strom_M: but gets the point across :) |
19:52.02 | drmessano | Strom_M: Where most people seem to stop learning nowadays |
19:52.09 | kdas | ManxPower, so it would be better to catch you here ? |
19:52.10 | vipcarrier | how does it work? |
19:52.12 | Strom_M | I didn't know stupidity was in such vogue these days |
19:52.27 | jplank | rhombus: what did I just say about drmessano :) |
19:52.29 | drmessano | Strom_M: You think that's bad, times that by 6 billion people |
19:52.37 | jplank | ROFL |
19:52.37 | Strom_M | oh em gee |
19:52.43 | rhombus | jplank: yes |
19:52.55 | vipcarrier | slaps jplank around a bit with a large trout |
19:53.01 | drmessano | zee oh em gee |
19:53.14 | drmessano | Sorry Canada |
19:53.18 | drmessano | ZED oh em gee |
19:53.22 | kdas | ManxPower, i want to send some money i just got to wait for my friend who is in the bathroom to use his paypal don't expect alot but it is something if i had more to spare i would give more ;) |
19:53.39 | kdas | so i will come back in a few to get your details |
19:55.14 | jplank | lol |
19:57.20 | *** join/#asterisk kdas (n=ME@c-67-161-44-222.hsd1.ca.comcast.net) |
19:57.39 | kdas | ok ManxPower what info do i need to send you money to your paypal ? |
19:58.15 | jplank | you can send money to my paypal |
19:58.27 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
19:58.31 | drmessano | Send $10 to all of our paypal's |
19:58.37 | drmessano | times that by 241 |
19:58.42 | jplank | lol |
19:58.49 | drmessano | That's almost $10000 or something |
19:59.04 | Strom_M | no, just divide it by .005 and send the result directly to me |
19:59.10 | jplank | you paypal me $10 a month and i'll set you up with a DID with free incoming :) |
19:59.29 | drmessano | Strom_M: What is that times by 100? |
19:59.58 | drmessano | A nice greasy pork sandwich, served in a dirty ashtray.. that's what.. |
20:00.38 | *** join/#asterisk wiscados (n=mint@81.25.184.155) |
20:00.42 | jplank | errr I can't get pickup to work, this is really getting to me, what use is BLF if you can't pick up the call |
20:01.40 | jeev | sets up manxpower@hisdomain.com and on paypal as an alias and msgs to kdas. |
20:02.14 | ManxPower | jeev: I've received less than $300 in paypal donations since 2002 |
20:02.27 | ManxPower | I'm not in this channel for the money 8-) |
20:02.38 | drmessano | I'm here for the Ho's |
20:02.39 | jeev | heh |
20:03.02 | drmessano | Still waiting for the women to arrive :( |
20:03.09 | drmessano | Sausage fest ------------------------------------------------------------------> |
20:03.18 | drmessano | All 241 of them |
20:03.43 | SpaethCo | hopefully your MoH is working so you won't sit in silence while you're waiting for them to show up |
20:04.12 | drmessano | I got that working first.. I knew it was gonna be a while |
20:04.24 | drmessano | Mark Spencer promised me free telephony and Ho's |
20:04.34 | drmessano | I'm beginning to dislike him :( |
20:04.43 | jaytee | he promised me a Winnebago |
20:04.51 | rhombus | drmessano: only him? I thought you disliked everybody. |
20:05.03 | jplank | lol |
20:06.05 | ManxPower | drmessano: I don't think you'd like the hos mark would send your way/ |
20:06.26 | drmessano | rhombus: No.. I dislike a lot of people, bit everyone.. So if you think I dislike you, and are wondering if it's just me disliking everyone or if it's personal, it is indeed personal. Sorry. |
20:06.50 | drmessano | errr |
20:07.08 | drmessano | I also dislike thinking 3 or 4 words ahead too |
20:07.37 | jaytee | I hate everyone equally without regard to race, creed, nationality or sexual orientation |
20:07.59 | ManxPower | jaytee: that's what I do. |
20:08.09 | jaytee | it's all about the equality :-) |
20:08.33 | jplank | jaytee: does that include hot female blonds who are bi? |
20:08.53 | jaytee | jplank, well......there are exceptions to every rule :-) |
20:09.57 | ManxPower | sends Major Iceborg to jaytee |
20:10.07 | jaytee | OMG!!! |
20:10.07 | jplank | lol, except the rule of exceptions I suppose |
20:10.59 | jaytee | "Leeloo Dallas Multipass" |
20:11.04 | ManxPower | 8-) |
20:11.17 | [TK]D-Fender | ManxPower: SIP debug does wonderful stuff like show that calls are even mathing te peers we think they should.... |
20:11.20 | jaytee | I love that movie |
20:11.33 | rhombus | drmessano: well, forgive me. It does indeed seem like you hate everybody. Personally. |
20:11.34 | ManxPower | jaytee: 5th Element, Matrix, and Dogma are what I call "jesus movies" |
20:11.39 | ManxPower | [TK]D-Fender: I know. |
20:11.55 | ManxPower | [TK]D-Fender: but you love sip debug enough to be...creapy. *tease* |
20:11.55 | jaytee | ManxPower, jesus movies? |
20:12.11 | ManxPower | jaytee: main character has to save the world, but has to die first. |
20:12.25 | jaytee | I loved all three but the Matrix sequels sucked. |
20:12.27 | [TK]D-Fender | ManxPower: and it usually eve saves you looking at useless dialplan contexts people think should work but are never being used |
20:13.06 | jaytee | only part of Matrix 2 that I liked was Merovingian when he said, "I love ze french language, it's excellent to curse with. Like wiping your ass with silk." |
20:13.40 | jplank | fender: do you have much experience with pickup()? |
20:14.02 | [TK]D-Fender | jplank: nope, but what's your actual question |
20:14.03 | *** join/#asterisk j0 (n=dan@S01060016b6b541d2.va.shawcable.net) |
20:14.12 | jaytee | ManxPower, when I think of jesus movies in that context The Omega Man with Charlton Heston is the first thing that comes to mind. |
20:14.28 | ManxPower | Leeloo dies right at the beginning, Neo has to die before he can control the matrix, in Dogma Bethany has to die when she saves the world. |
20:14.30 | jplank | no matter how I use it, I get a error inside the console |
20:14.34 | jaytee | speaking of which I guess now we can finally pry his gun from his cold, dead hands. |
20:14.49 | jplank | I'm pretty sure I'm using it correctly, but it doesn't work |
20:14.58 | jplank | pickupexten works though |
20:15.27 | [TK]D-Fender | jplank: "pretty sure" doesn't mean much. pastebin <- |
20:15.32 | jplank | i know |
20:15.55 | jaytee | Dogma was excellent. I actually own a Buddy Christ dashboard figure. |
20:16.09 | jaytee | and now they have Buddy Christ bobbleheads. |
20:16.50 | ManxPower | I'm a devout atheist, but I'd consider a Buddy Jesus on my dashboard |
20:17.03 | jaytee | then go to www.viewaskew.com |
20:17.08 | jplank | http://pastebin.com/m77904f3e |
20:17.09 | [TK]D-Fender | %$##ing sliced my thumb at martial arts practice today. Just messed my whole week up... |
20:17.13 | [TK]D-Fender | dammit |
20:17.21 | jplank | bot extensions are inside the from-test context |
20:17.23 | jaytee | ouch! |
20:17.24 | jplank | both* |
20:17.57 | UD | gcc is god |
20:18.00 | [TK]D-Fender | jay took most of my thumb-print though 3 layers but left a flap so it should reseal.... |
20:18.37 | *** join/#asterisk terracon (n=garry@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com) |
20:18.38 | jaytee | [TK]D-Fender, yeah but it's like a paper cut but worse and they take longer to heal than a scrape or gouge |
20:19.15 | jaytee | is it on your writing hand? |
20:19.19 | [TK]D-Fender | jay it was real clean fortunately.... Mr. Pointy was very good at its job... |
20:19.19 | jplank | fender: did you catch the pastebin? |
20:19.28 | [TK]D-Fender | jaytee: left fortunately |
20:19.40 | UD | hey ManxPower I can't seem to get a 2 person voip conversation going |
20:19.43 | jaytee | that's not as bad but still |
20:19.44 | UD | on this connection |
20:19.57 | UD | does your wireless connection suck as bad as mine? |
20:19.59 | [TK]D-Fender | jplank: it doesn'rt match |
20:20.19 | jplank | what do you mean? |
20:20.48 | [TK]D-Fender | jplank: exten => _**.,1,Pickup(${EXTEN:2}@from-test) and -- Executing [**2003@from-test:1] Pickup("SIP/2002-08e02888", "2003") in new stack <-- no context in execution |
20:20.50 | UD | not that i was actually planning on using this connection for real world asterisk use |
20:20.58 | jaytee | I've finally rewritten my code for the speech recognition IVR so I'm not using the n+101 deprecated crap anymore. Yay!!!! |
20:21.19 | [TK]D-Fender | jaytee: whats the vr component? |
20:21.19 | jplank | err |
20:21.20 | jplank | sorry |
20:21.28 | jaytee | [TK]D-Fender, Lumenvox |
20:21.29 | jplank | that wasn't the right error |
20:21.42 | jplank | that was the error from when I tried doing the pickup without a context |
20:21.45 | jblack | Heh. Looks like New Orleans is gonna have to evacuate again. |
20:21.49 | jplank | hold on, I'll update the pastebin |
20:21.59 | ManxPower | jblack: maybe, still too far in the future to know. |
20:22.09 | [TK]D-Fender | jaytee: how accurate is it performing? |
20:22.11 | jaytee | aw, man! those poor people in LA just can't catch a break |
20:22.19 | j0 | the additional remora board on my sangoma a200 doesn't power up (yes power is connected). any ideas? |
20:22.39 | *** join/#asterisk Levonk (n=lk@adsl-75-62-133-138.dsl.lsan03.sbcglobal.net) |
20:22.49 | *** part/#asterisk rhombus (n=sfbosch@dsl-vlan435-66-18-218-36.nucleus.com) |
20:23.20 | jaytee | [TK]D-Fender, actually very well so far. I need to test with people who have odd accents but for the simple grammars I'm using the recognition is usually 80% or better so it matches properly. |
20:23.46 | jplank | fender, I updated the pastebin http://pastebin.com/m7fc41949 |
20:24.08 | jaytee | I wished I could have imported the grammars from my Locus Liaison system but those are compiled grammar files and Lumenvox only seems to support raw text grammar files. |
20:24.44 | jblack | Like I said, "It looks like", not will. N.O. is dead center for the predicted path for a H3 on Friday. |
20:25.23 | [TK]D-Fender | jplank: please include more cli incuding proof the exten you are trying to pickup is ringin |
20:25.28 | jaytee | so I've got the menu tree tested for both voice rec and dtmf and both options work for each keyword. Just need to record the custom prompts and I'm good to go. |
20:25.33 | jplank | ok |
20:26.38 | jplank | fender: http://pastebin.com/m20b93f0b |
20:27.17 | jaytee | [TK]D-Fender, if you ever decide to mess with Lumenvox their starter kit for one port license is only 50 bucks but comes "without support" and even then their reps will answer questions anyways if you're evaluating it. |
20:28.10 | jaytee | and they've got alot of example configs and online training vids for it but I haven't really had to dig into the documentation to deep to make it work. |
20:28.18 | jplank | fender: the actual ringing is line 81 |
20:30.41 | [TK]D-Fender | jplank: "This application can pickup any ringing channel that is calling the specified extension" |
20:31.02 | [TK]D-Fender | jplank: the extension being diales i "s" |
20:31.10 | [TK]D-Fender | jplank: not gonna work |
20:31.18 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
20:31.18 | jplank | ahhh |
20:31.47 | jplank | so if I did a plain old dial it should work? |
20:32.39 | [TK]D-Fender | jplank: if you used a valid target as per the apps instructopns, I'm sure it does its job |
20:38.30 | *** join/#asterisk russellb (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
20:38.30 | *** mode/#asterisk [+o russellb] by ChanServ |
20:38.46 | ManxPower | jplank: I think you are confused. An extension is the first field after exten => |
20:39.13 | ManxPower | the extension is not the sip account name. If you set the sip userid and the extenson to be the same....well you see the confusion that results. |
20:40.39 | jplank | I get it, thank you |
20:46.28 | jplank | aastra 57i ct is a pretty sick phone |
20:48.12 | [TK]D-Fender | jplank: I hated mine. made me yearn for my bedside Polycom IP 301 |
20:51.35 | jplank | really why? |
20:52.03 | jplank | I've been testing it all day, and everything seems to work perfectly (aside from direct call pickup from the phone) |
20:52.47 | jplank | so theres not way to grab a call from the device thats ringing directly? |
20:52.51 | jplank | no way* |
20:54.24 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@ip68-111-67-4.oc.oc.cox.net) |
20:54.29 | *** join/#asterisk bmg505 (n=leon@196-209-78-72-tbnb-esr-2.dynamic.isadsl.co.za) |
20:56.38 | [TK]D-Fender | jplank: sure there is. "pickup", "pickupexten", etc |
20:57.02 | [TK]D-Fender | jplank: unfortunately FreePBX owns your sorry ass. |
20:57.12 | jplank | doesn't pickupexten pickup the ring group? |
20:57.20 | jplank | errr your telling me |
20:57.47 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
21:00.54 | jplank | I guess I could use pickupmark, but that seems like a pain in the ass |
21:01.45 | jplank | hmm maybe not |
21:04.10 | [TK]D-Fender | jplank: I'm saying that FreePBX'S SPAGHETTI |
21:04.18 | jplank | yea it is |
21:04.22 | jplank | I totally agree |
21:04.48 | [TK]D-Fender | & meatball dialplan and you lack of understanding of it, and those dial plan apps = failure |
21:06.01 | jplank | well, not as much as not understanding it, I use very few of freepbx's dialplans, but DID routing is one of them, and I forgot that, so I forgot that the inbound call was going through 100 macros before ringing the extension |
21:06.23 | *** part/#asterisk Cresl1n (n=matt@nat/digium/x-1751bd4058c47317) |
21:06.36 | jplank | but if I set pickupmark at the last macro, it *should* work I think |
21:08.08 | ManxPower | jplank: the PROBLEM with FreePBX is that all calls go thru like 100 macros. |
21:09.10 | jplank | I really use freepbx for creating extensions, all the context's I've created in extensions_custom, I know its stupid, but I originally started with freepbx, so now I'm so deep into it, its easy working around it then rebuilding |
21:09.29 | jplank | easier* |
21:10.07 | [TK]D-Fender | I love it when people send good money after bad... |
21:12.13 | *** join/#asterisk EI5GTB-macbook (n=EI5GTB@78.16.207.184) |
21:13.26 | ManxPower | jplank: At least you are not a total idiot, unlike the vast majority of GUI users. |
21:15.09 | _trine | well that's a backhanded compliment if ever I heard one |
21:15.15 | _trine | :)) |
21:15.30 | jplank | my problem is I jumped into asterisk before learning exactly how it worked, so I was stuck with a gui for a while, and then I realised all of freepbx's limitations so I had to start compensating for them, luckily with the custom conf's its not THAT hard to do |
21:15.54 | jplank | the plan is to sooner then later rebuild the whole box on a test box without freepbx, get it running and them swap them |
21:16.02 | jplank | unfortunately easier said then done |
21:18.19 | j0 | i started with raw asterisk for a few years.. but got tired of editing configs and need to have some way for un-educated techs to make changes... so now i use ...... trixbox! barf |
21:19.15 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
21:19.40 | jplank | trixbox is good if you don't want to do anything really cool, or pretty much anything but make a couple phone calls |
21:19.50 | [TK]D-Fender | salmonella : the state of constantly swimming upstream only to get fucked & die. |
21:20.44 | j0 | jplank: yeah.. and it *just works* as long as u dont try to add all the cool features.. :) i just started using aastra phones with trixbox and it's made my life so much easier |
21:21.30 | jplank | I just started testing aastra today, fendor doesn't seem to like them, but I'm loving it |
21:22.13 | jplank | don't know if we are going to start swapping out polycom users any time soon, but if the testing keeps going so well, might give us another option |
21:22.31 | jplank | tried grandstream, but I was ready to throw them away after a day |
21:22.58 | j0 | haha. i started with grandstream for my personal system just to test with... they've been terrible |
21:23.51 | ManxPower | we tested GS BT101, Cisco 7940, Polycom 500, and some Uniden SIP phone (aastra was not an option at the time -- their IP phones did notexist) |
21:23.57 | ManxPower | settled on Polycom |
21:24.02 | [TK]D-Fender | aastra 5i's have many disappointments, and nothing I'd pick one over a polycom for except in extremely specific scenarios. |
21:24.16 | [TK]D-Fender | I wouldn't want one for any of my users |
21:24.49 | [TK]D-Fender | ManxPower: uniden uip-200 = z0mg kill me now |
21:25.18 | ManxPower | [TK]D-Fender: It was a UIP somethething, it was about as good as a grandstream |
21:25.53 | [TK]D-Fender | ManxPower: I have 2. glad they run and I don't have to think about them ever |
21:26.11 | jplank | fender: what disappointed you about 5i's? |
21:27.25 | jeev | Fender is the canadian chuck norris, everything disappoints him and then he will head butt it |
21:28.44 | *** join/#asterisk Yourname (i=Yourname@unaffiliated/yourname/x-837320) |
21:28.56 | [TK]D-Fender | jplank: no handset weight, tinny audio, no base weight, slippery rubber base that gets yanked around by the handset, shit viewing angle on the LCD, pixel screen used in dumb char matrix mode designed by retards, cryptic button icons to name a few |
21:29.23 | [TK]D-Fender | jeev: no, I have almost no complaints about polycom.... |
21:29.37 | RypPn | sounds like my first phone, the atcom 530 |
21:29.56 | [TK]D-Fender | jeev: atcom = crap, just like GS |
21:30.00 | jeev | Fender, i've been getting a lot of complaints about hang ups.. so ManxPower figured i should try a gre tunnel and use SIP.. |
21:30.13 | jeev | i heard grandstream's = satan |
21:30.23 | Yourname | Hi guys, sometimes an agent is logged into a queue via SIP, and his internet connection drops. The agent is logged in for several minutes afterwards. And so when the agent comes back online in 2-3 mins, he cant log back on cuz Asterisk thinks he's still logged on |
21:30.24 | [TK]D-Fender | jeev: nobody wants to talk to you... it isdn't the phone ;) |
21:30.25 | jplank | fender: I can't speak for the 5i, but the CT version doesn't seem to have (all of) those problems |
21:30.30 | jplank | the weight is fine |
21:30.30 | jeev | hahah |
21:30.32 | jeev | you suck |
21:30.39 | jplank | they replaced the icons with labeled |
21:30.43 | ManxPower | [TK]D-Fender: I'm helping him as the networking piece is interesting. |
21:30.46 | [TK]D-Fender | jplank: 5i = the whole SERIES. |
21:30.47 | jplank | the feet have grips |
21:30.53 | jeev | hopefully that is it. |
21:30.57 | [TK]D-Fender | jplank: and I specifically had the 57i CT |
21:30.58 | ManxPower | and most of what I *do* is networking. |
21:31.06 | jplank | thats what I'm testing here |
21:31.24 | jplank | the manual shows different buttons and feet then came with it |
21:31.35 | [TK]D-Fender | jplank: yet it has grips.... THEY SUCK |
21:31.39 | jplank | lol |
21:31.42 | jeev | ManxPower, so now i just change all iax crap to sip and viola, eh? |
21:32.09 | jplank | I def can kind of agree with the LCD comments, but the backlight helps a lot I think |
21:32.11 | ManxPower | jeev: sounds simple, huh? It's not, but I can help, as I have a similar Asterisk SIP to Asterisk SIP in production. |
21:32.28 | jeev | yea, i think i'll get that fine, i haven't tried. |
21:32.48 | ManxPower | jeev: /join #asterisk-cli so we don't start flooding the channel with talk. |
21:32.49 | [TK]D-Fender | jpbacklight gives you something to look at when the writing disappears as of 10deg view |
21:35.16 | jplank | I need to get some more time under my belt with this phone to experience your dislikes it seems, but features wise, it seems solid so far |
21:35.23 | jplank | thats a lot more then I can say about most phones |
21:36.36 | Yourname | So....nobody |
21:41.50 | jplank | I can't remember how to trim a string from the right instead of left, anyone know off the top of their head? |
21:43.12 | [TK]D-Fender | jplank: channelvariables.txt |
21:44.11 | jplank | thanks |
21:44.51 | jplank | heh, that should of been obvious, thanks |
21:46.29 | *** join/#asterisk knarfly (n=knarfly@c-75-74-155-198.hsd1.fl.comcast.net) |
21:47.44 | *** join/#asterisk tvirus (i=TheVirus@c-68-54-165-28.hsd1.md.comcast.net) |
21:55.14 | jplank | aha! |
21:55.21 | jplank | screw freepbx and their crazy marco's |
21:55.26 | jplank | got pickup to work! |
21:55.39 | jplank | pain in the ass though |
21:56.17 | jplank | now if I could just get the aastra to send **+exten when pressing the BLF button, I'd be in business |
21:57.55 | *** join/#asterisk Levonk (n=lk@adsl-76-230-111-142.dsl.lsan03.sbcglobal.net) |
21:57.55 | [TK]D-Fender | jplank: it sends what you tell it to |
21:58.05 | jplank | does it? |
21:58.14 | [TK]D-Fender | jplank: yup |
21:58.23 | jplank | the value? |
21:58.49 | jplank | put if I put the value as **+exten, wouldn't it request to notify for that? |
21:59.22 | jplank | yup |
21:59.26 | jplank | errr |
21:59.29 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
21:59.29 | *** mode/#asterisk [+o mog] by ChanServ |
21:59.58 | [TK]D-Fender | jpchange the hint |
22:00.49 | jplank | err backs to hints, thats where I lost it with the polycoms |
22:01.35 | [TK]D-Fender | jplank: hint = notify.... |
22:01.51 | jplank | oh thats it? |
22:04.35 | jplank | wait I still don't get it... |
22:04.35 | *** join/#asterisk rconnect__ (n=rconnect@85.138.128.75) |
22:05.00 | jplank | how would I send the hint? |
22:05.31 | jplank | if I want to do **2003 |
22:07.00 | [TK]D-Fender | jplank: then thats the exten that should have the hint |
22:07.55 | jplank | I get that, but how would I send the hint, exten => hint,SIP/2003 ? |
22:07.59 | jplank | that doesn't make sense |
22:08.32 | jplank | ohhh |
22:08.43 | jplank | exten => hint,SIP/**2003 ? |
22:10.09 | rconnect__ | what is one of the best linksys routers to use with Asterisk? |
22:10.11 | *** join/#asterisk AdamWest (n=Leif@CPE001d7e2f9574-CM0012c9db3d2e.cpe.net.cable.rogers.com) |
22:10.28 | jameswf-home | linksys ewww |
22:11.00 | ManxPower | exten => _**XXXX,hint,SIP/2003 |
22:11.00 | [TK]D-Fender | jplank: see the problem... you don't even know how one is DEFINED |
22:11.23 | [TK]D-Fender | ManxPower: through 1.4 you cant use patterns |
22:11.31 | rconnect__ | <jameswf-home, which one you recomend then? :) |
22:11.43 | jameswf-home | make your own |
22:11.45 | tvirus | Is it possible to change the outgoing caller ID on an analog line (Asterisk with a TDM410). The line is from Verizon. |
22:11.47 | [TK]D-Fender | rcwhicever |
22:11.55 | [TK]D-Fender | rconnect__: whichever |
22:11.56 | jplank | fender: yea I knew that was the problem |
22:12.02 | ManxPower | tvirus: no |
22:12.13 | rconnect__ | ok |
22:12.15 | [TK]D-Fender | tvirus: did verizon tell you a way? |
22:12.23 | ManxPower | [TK]D-Fender: Ah! |
22:12.35 | tvirus | We have yet to contact them, figured it would have to go through them. Thanks. |
22:12.48 | [TK]D-Fender | tvirus: 99.99999999999999999999999999999999999999% = impossible |
22:12.56 | ManxPower | so it would be exten => **2003,hint,SIP/2003 would be correct |
22:13.10 | tvirus | So there is a chance. |
22:13.13 | tvirus | :D |
22:15.43 | jplank | ManxPower: let me see if I follow |
22:16.05 | jplank | that, thrown into my context, should translate the aastra's notify **2003 message to 2003? |
22:16.42 | [TK]D-Fender | jpno |
22:16.45 | [TK]D-Fender | jplank: no |
22:16.55 | jplank | it did work btw |
22:17.13 | jplank | now the question is why? |
22:17.34 | jameswf-home | how to fix any asterisk issue:http://jantman.dyndns.org:10011/wiki/index.php/Generic_Problem_Solving_Method |
22:18.57 | jplank | do you mind explaining? |
22:22.14 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
22:22.14 | *** mode/#asterisk [+o mog] by ChanServ |
22:25.06 | jplank | tvirus: you'd have to be some pretty good telephone freak to spoof your POTS line's caller-ID (without redirecting the call of course) |
22:25.22 | jplank | phreak* |
22:26.35 | [TK]D-Fender | jplank: like the kind that can spontaneously create services that don't exist, and then take advantage of them... |
22:27.42 | jplank | it was sarcasm |
22:28.06 | jplank | and its not something that doesn't exist |
22:28.28 | jplank | but most telco's wont do it for you |
22:28.42 | jplank | youd have to break into their switches and set it up |
22:36.20 | rconnect__ | anyone know why accessing the gui, this message is displayed? The GUI does not have necessary privileges. |
22:36.30 | rconnect__ | i have configured both manager and http |
22:36.44 | rconnect__ | Please check the manager permissions for the user ! |
22:41.35 | jplank | maybe its a NAT problem? |
22:45.28 | [TK]D-Fender | nope, and GUI's are not supported in this channel |
22:46.09 | rconnect__ | ok, i have asked in the proper channel as well, but no answer so far, so i thought maybe someone here would know... sorry |
22:48.38 | jplank | grrr no one gets my sarcasm, I actually thought he was asking a sarcastic question himself |
22:49.20 | jplank | rconnect__: what gui are you using? |
22:51.04 | rconnect__ | digium asterisk gui 2.0 |
22:54.42 | jplank | never used it, so can't help you there, but if its anything like trixbox or freepbx, most privileges issues come down to changing passwords in the gui, and not changing the proper password on your system (example mysql) or vice versa. |
22:55.05 | jplank | can I change the value of ${EXTEN}? |
22:55.09 | rconnect__ | right. thanks jplank |
22:55.17 | rconnect__ | have to go. good night |
22:55.31 | *** join/#asterisk Levonk (n=lk@adsl-76-243-67-57.dsl.lsan03.sbcglobal.net) |
22:55.31 | jplank | I can't find anywhere it says I can't, but it doesn't seem to work |
22:56.17 | jaytee | jplank Set(${EXTEN} = 1234567) |
22:57.12 | jplank | tried that, it doesn't seem to work |
22:58.08 | jplank | http://pastebin.com/m237285cf |
22:58.44 | [TK]D-Fender | lol |
22:58.56 | file | you can not change EXTEN, it is a special variable used to return the current executing extension |
22:59.14 | [TK]D-Fender | AND you're refernecing the variabl |
22:59.28 | [TK]D-Fender | AND adding BS whitespace |
22:59.36 | jaytee | ah, yeah |
23:00.11 | jplank | I was hoping there was a way |
23:00.38 | jaytee | so you'd do something like Set(${newexten} = ${EXTEN:2}) ? |
23:01.04 | jplank | I'm going to have to rewrite a few scripts now to get the ** out of there, and I thought it would be easiest to do it before it hit those scripts |
23:01.46 | jplank | why would it matter if I was referencing the variable? |
23:02.01 | jplank | and where am I adding a whitespace? |
23:02.18 | jaytee | he probably meant my bad example |
23:02.27 | jplank | ahhh |
23:02.40 | jplank | ohhhh |
23:02.59 | jplank | he meant me ${EXTEN}=${EXTEN:2} |
23:03.17 | jplank | I only added that because jaytee said it, it was originally EXTEN= |
23:04.18 | jaytee | try using a unique variable name for the new extension number you want to dial |
23:06.54 | jplank | I'm trying to remove the ** so I can match for 2003 |
23:07.19 | jaytee | so use a new variable name that isn |
23:07.25 | jplank | I'd hate to have to rewrite all my ext lists to account for ** |
23:07.27 | jaytee | isn't a reserved system variable |
23:08.19 | jplank | how would I match for that? |
23:09.22 | jaytee | your paste didn't have any kind of code for matching anything |
23:09.43 | jaytee | you're trying to strip the ** out and dial? |
23:10.03 | jplank | yea |
23:10.12 | jplank | no |
23:10.26 | jaytee | well that's frikken clear as glass |
23:10.28 | jplank | i thought you said ** out of the dial |
23:10.30 | jplank | lol |
23:10.36 | jplank | I'll show you the full code |
23:11.05 | jaytee | no please don't |
23:11.21 | jplank | http://pastebin.com/m39e0539b |
23:11.23 | jaytee | the last time someone showed me the full code I couldn't sleep for weeks and needed therapy |
23:11.25 | jplank | no two mroe liens |
23:11.42 | jplank | not the full code |
23:11.48 | jaytee | look if your bank has liens on stuff that's not my problem |
23:11.49 | jplank | I wouldn't want to scare you away |
23:12.34 | jplank | is that clearer? |
23:12.48 | *** join/#asterisk sanjayb (n=sanj@59.183.52.45) |
23:13.11 | jaytee | jplank, those two pastes look identical to me |
23:13.40 | jplank | added two lines |
23:13.44 | jplank | matching for 2003 now |
23:13.55 | jplank | just to show you why I'm trying to set EXTEN |
23:14.35 | jaytee | yes, I understand and I see those two line IN BOTH pastes and they're identical |
23:15.42 | outtolunc | last time i looked EXTEN was one of those you couldn't change |
23:16.44 | jaytee | so just try Set(${YOURMOMMA} = ${EXTEN:2} and on the next line NoOp(${YOURMOMMA}) and then do a Goto(samecontext,${YOURMOMMA},1) |
23:17.00 | jplank | original pastebin http://pastebin.com/m237285cf |
23:17.01 | jaytee | on the next line |
23:17.10 | jplank | new http://pastebin.com/m39e0539b |
23:17.22 | jplank | how can you say they are the same? |
23:17.27 | outtolunc | http://svn.digium.com/view/asterisk/branches/1.4/doc/channelvariables.txt?view=markup |
23:17.36 | jplank | outtolunc: they told me already |
23:17.37 | outtolunc | <PROTECTED> |
23:17.37 | outtolunc | <PROTECTED> |
23:17.37 | outtolunc | <PROTECTED> |
23:17.49 | outtolunc | ok |
23:17.49 | jplank | I was just trying to show jaytee WHY I was doing it |
23:18.05 | jplank | thank you thought |
23:18.47 | jplank | is there any way to change the digits dialed then? |
23:19.17 | outtolunc | goto(yadda,${your mod'd exten},1) |
23:19.25 | Yourname | Hi guys, sometimes an agent is logged into a queue via SIP, and his internet connection drops. The agent is logged in for several minutes afterwards. And so when the agent comes back online in 2-3 mins, he cant log back on cuz Asterisk thinks he's still logged on |
23:19.32 | jaytee | jplank, ok my mistake on the same stuff. I had too many pastebin tabs open in Firefox. |
23:19.38 | jplank | n/p |
23:19.46 | jplank | I thought I was going crazy at first |
23:20.02 | jplank | I got it though |
23:20.04 | jplank | thank you both |
23:20.30 | jplank | I didn't read jaytee's message at first because I thought he was being sarcastic |
23:20.43 | outtolunc | i missed his also |
23:20.50 | outtolunc | haha |
23:22.36 | outtolunc | needs to build a modified roomba for steam cleaning the carpets |
23:24.00 | jplank | heh I left samecontext thinking that was some cool thing I didn't know about |
23:24.07 | jaytee | completely forgot you couldn't actually change the value of ${EXTEN}, probably because I'd already read that and never atttempted it. I use unique variables to do stuff. |
23:24.39 | jaytee | no, the name of the context, which is optional I believe |
23:25.15 | jaytee | unless you WANT to jump to another context and then it's required |
23:25.32 | jaytee | anyone feel free to jump in and correct me if I'm wrong on that point. |
23:25.46 | outtolunc | yes, context in goto is optional |
23:26.13 | outtolunc | obviously if goto'ing a diff context then its required <G> |
23:26.22 | jaytee | whew! |
23:27.05 | jaytee | be back later |
23:28.19 | outtolunc | note: never let the woman talk you into renting a rug doctor thing |
23:36.27 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
23:37.54 | jplank | is this the correct syntax exten => **2003,Transfer(SIP/2003) ? |
23:38.10 | jplank | I'm trying to redirect a SIP REF for **2003 to 2003 |
23:39.11 | *** join/#asterisk kdas (n=ME@c-67-161-44-222.hsd1.ca.comcast.net) |
23:39.22 | kdas | ManxPower, you arround still? |
23:39.36 | kdas | jaytee, alive? |
23:40.23 | kdas | i have my extensions.conf file setup as _XX.,1, blah but i don't think that is too good is there a better way of doing this ? |
23:42.04 | [TK]D-Fender | kdas: pick real patterns based on what your telco requires |
23:42.45 | kdas | [TK]D-Fender, reall patterns meaning 9 or 10 digets etc? |
23:42.59 | [TK]D-Fender | kdas: is that proper? |
23:43.16 | kdas | whats the use of 's' like s,1,blah it dosent catch anything ? |
23:43.44 | kdas | [TK]D-Fender, you tell me i mean i am not going to be expecting 1-235-2345-2342342 numbers just usa numbers |
23:44.12 | [TK]D-Fender | kdas: "s" is not a catchall |
23:44.57 | [TK]D-Fender | kdas: no i mean to tell you that YOU should know what kind of numbers are valid and not use wide reaching patterns like you have been to do the job |
23:45.21 | kdas | [TK]D-Fender, "s" is the start |
23:45.49 | ManxPower | kdas: "s" is almost never used except for in macros and FXO signalled ports. |
23:46.05 | [TK]D-Fender | kdas: you clearly do not understand the "s" exten |
23:46.10 | kdas | ManxPower, oh thanks |
23:46.42 | kdas | [TK]D-Fender, well thanks for pointing out the obvious |
23:46.54 | [TK]D-Fender | ~stdextens |
23:46.54 | jbot | [~stdextens] The "s" Standard Extension : Where a call goes to when * does not know the destination of the call. Ex : Calls coming in on FXO ports (no DID), or from an ITSP that doesn't specify or where it was not set in the REGISTER line, or FXS port goes off-hook and "immediate=yes" in zapata.conf. "s" is also used to make IVRs & macros. |
23:47.10 | kdas | damn i hate geeks who think they all that because they know just that tad more |
23:47.24 | [TK]D-Fender | kdas: When you dial a number on a phone that lands on a NUMBERED extensions. "s" does not factor into this |
23:47.37 | [TK]D-Fender | kdas: You jsut finished trying to |
23:47.38 | kdas | [TK]D-Fender, thank you |
23:47.39 | ManxPower | kdas: [TK]D-Fender knows at least as much as I do. |
23:47.51 | [TK]D-Fender | "tell me how it is". And then you you FAILED |
23:48.07 | [TK]D-Fender | then* |
23:48.14 | [TK]D-Fender | Anyways, forget "s" right now |
23:48.24 | jblack | He doesn't have to help you, and you don't have to ask for his help. |
23:48.28 | [TK]D-Fender | kdas: And go pick some sensible patterns |
23:48.41 | jplank | fender: how can I compensate for the aastra sending **+extension inside the SIP REFER on a BLF xfer |
23:48.55 | kdas | ManxPower, i am not saying he isn't just i hate when "people" try to show off there knowledge like "you clearly don't understand what hacking into nasa means.. HHAHA.. N00b i am so 31337 |
23:49.30 | jblack | jplank: Look at offsets. Such as ${EXTEN:3} |
23:49.35 | kdas | jblack, fair enough i never asked for his help but he is kind enough to help ;) |
23:49.56 | kdas | [TK]D-Fender, once again thank you i will do some more research and forgive my ignorance |
23:51.00 | [TK]D-Fender | jplank: aastra sends exctly what you tell it to subsribe to. its your job to make sure * matches |
23:51.06 | jplank | I know |
23:51.12 | jplank | thats what I'm trying to do |
23:51.13 | ManxPower | kdas: I get as pissed off at people as [TK]D-Fender when they don't just let the person with experience drive. |
23:51.56 | jplank | how can I get the variable from SIP REFER? |
23:52.04 | jplank | to look at the offsets |
23:52.05 | [TK]D-Fender | jpAnd the problem becomes your FreePBX dialplan. As I said, it owns your ass... If you can dodge this bullet, good luck. |
23:52.21 | jplank | I've gotten pretty far so far |
23:52.24 | [TK]D-Fender | jplank: You don't get aceess to traffic like that |
23:52.58 | jplank | so then how can I tell my * to refer calls destined for **2003 to 2003? |
23:53.07 | jplank | oh |
23:53.23 | jplank | I could just tell my * that **2003 and 2003 is the same thing, can't I |
23:53.28 | jblack | exten => _**XXXX,1,Goto(${EXTEN:2},1) |
23:53.30 | ManxPower | jplank: If you were trying to get the info out of a TRANSFER, that might be able to be done, but not during a REFER that I'm aware of. That requires much lower level access to the SIP protocol that Asterisk allows |
23:53.58 | jplank | jblack: I used the same thing with transfer |
23:54.18 | jplank | ManxPower: so theres no way? |
23:54.54 | jplank | grrr |
23:55.29 | ManxPower | jplank: I suspect you have the wrong approach. Have you checked the wiki? |
23:55.38 | *** join/#asterisk mog (n=mog@74.95.48.254) |
23:55.38 | *** mode/#asterisk [+o mog] by ChanServ |
23:55.49 | jplank | voipinfo? I doesn't have much about REFER |
23:55.57 | ManxPower | jplank: until the call hits the dialplan, you can't know anything about it. |
23:56.04 | jplank | hmmmm |
23:56.06 | ManxPower | jplank: No. What you are trying to accomplish. |
23:56.24 | jplank | basically, i'm working on BLF with a aastra phone |
23:56.28 | ManxPower | Which I suspect is direct accesss to call parking lots using a a BLF for each slot. |
23:56.42 | ManxPower | so that is what you should look on the wiki for. |
23:56.48 | jplank | lol |
23:57.05 | ManxPower | It doesn't matter HOW you do it, as long as you can do it. don't get stuck in the mechanics of it. |
23:57.16 | ido | does anyone here have a good all-software solution for fax detection from incoming SIP or IAX channels? |
23:57.27 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
23:57.33 | ido | preferrably in tutorial format? :) |
23:57.49 | ManxPower | ido: NVFaxDetect if you can find the code. |
23:57.57 | ido | ManxPower: where might one find the code? |
23:58.11 | ManxPower | just remember chances are it won't work on anything except ulaw and alaw codecs. |
23:58.19 | ido | ulaw is fine |
23:58.22 | ManxPower | ido: I have no idea. check the wiki, the link is dead |