| 00:00.29 | Titanous | ok, I've added the sample file to zapata.conf, but still no cigar... |
| 00:04.30 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:04.35 | sandorp | [TK]D-Fender: thanks for the help earlier; you were right on the money |
| 00:05.40 | *** join/#asterisk rogerz (n=highvolt@cpe-74-70-240-44.nycap.res.rr.com) |
| 00:07.23 | Titanous | any ideas?? |
| 00:08.05 | Yourname` | Hi. Someone please come to my rescue. I have 10 dialers with NoOps.. I want to parse data from all those dialers, take those NoOps and give me an answer with the number of those noops, etc. What's the best way to do it? A) Using the manager API to connect to all the dialers and get those NoOps? B) Logging /var/log/asterisk/messages to a central logging and parsing server? C) Using some sort of AGI script (this idea remains sketchy) |
| 00:08.08 | Yourname` | How can I do it? |
| 00:08.18 | ManxPower | you must have a [channels] line before any config options |
| 00:09.30 | *** part/#asterisk ManxPower (n=manxpowe@209.16.72.135) |
| 00:12.47 | *** join/#asterisk brian (n=brian@unaffiliated/brian) |
| 00:13.42 | kiscokid | anyone know a good IP VOIP provider with good rates from the US to Europe especially London? |
| 00:14.02 | riddlebox | can someone tell me what is happening when I call someone from my zap channel it rings one extra time after the called party picks up? |
| 00:17.57 | CCFL_Man2 | son of a bitch |
| 00:18.16 | CCFL_Man2 | i might not get the job because i said i can't work 3rd shif |
| 00:18.19 | CCFL_Man2 | t |
| 00:18.20 | *** join/#asterisk gremzoid (n=gremzoid@d58-111-173-16.rdl5.qld.optusnet.com.au) |
| 00:21.51 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 00:21.51 | *** mode/#asterisk [+o anthm] by ChanServ |
| 00:23.06 | *** part/#asterisk kiscokid (n=ron@208.106.35.66) |
| 00:26.15 | Yourname` | So no body |
| 00:26.16 | Yourname` | ? |
| 00:27.07 | riddlebox | CCFL_Man2, if you dont mind, what kind of job is it |
| 00:27.35 | *** part/#asterisk jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net) |
| 00:28.19 | *** join/#asterisk sts3c (n=bryan@66-43-34-10.misn.com) |
| 00:29.11 | riddlebox | I figured out my extra ring problem, I had a smart switch infront of my phone system, and apparently that was the problem, I no go straight from my provider to asterisk and no extra ring |
| 00:31.39 | tru_`z24 | If someone calls and then dials a menu item (extension) is there a variable that holds their phone number? |
| 00:31.51 | tru_`z24 | because {EXTEN} is the last extension they dialed |
| 00:32.05 | tru_`z24 | I want to repeat their number back to them if they press 1 |
| 00:32.37 | *** join/#asterisk jmacz (n=jmacz@190.24.103.32) |
| 00:32.57 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 00:34.27 | tzafrir_home | Titanous, can you pastebin your current zapata.conf ? |
| 00:35.01 | *** join/#asterisk knarfly (n=knarfly@c-98-203-55-196.hsd1.fl.comcast.net) |
| 00:35.22 | knarfly | help...the GotoIf is not working for me |
| 00:36.16 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
| 00:36.16 | riddlebox | is it better to use the TDM FXS cards for a fax machine and use fax detection? |
| 00:37.05 | *** join/#asterisk ManxPower (n=manxpowe@209.16.72.135) |
| 00:37.05 | ectospasm | Faxing is troublesome at best |
| 00:37.13 | knarfly | exten => 206,1,GotoIf($["${CALLERIDNUM}" = "303"]?3) |
| 00:37.13 | knarfly | exten => s,1,GotoIf($["${CALLERIDNUM}" = "3132340099"]?blocking,s,1) doesn't send the caller to blocking,s,1 |
| 00:38.34 | riddlebox | ectospasm, well I have a fax detection smart switch but form some reason, it adds an extra ring when I call out |
| 00:40.37 | CCFL_Man2 | riddlebox: it's an electronics tech job |
| 00:41.04 | CCFL_Man2 | $12.50 an hour entry level, must work any shift |
| 00:41.29 | CCFL_Man2 | but i don't want 3rd shift |
| 00:41.47 | CCFL_Man2 | that fucks up your brain |
| 00:41.55 | tru_`z24 | Is there an equivalent to ${CALLERIDNUM} that gives the 10 digit number instead of the 11 digit one? |
| 00:43.30 | knarfly | tru_`z24, ${CALLERIDNUM} is giving me fits right now too! |
| 00:43.58 | tru_`z24 | Why so? |
| 00:44.09 | riddlebox | CCFL_Man2, thats not bad for entry level |
| 00:44.25 | knarfly | the extension syntax I'm using above isn't working |
| 00:44.53 | tru_`z24 | well |
| 00:44.55 | knarfly | exten => s,1,GotoIf($["${CALLERIDNUM}" = "3132345678"]?blocking,s,1) |
| 00:45.01 | tru_`z24 | put a 1 in front |
| 00:45.22 | tru_`z24 | calleridnum should return "13132345678" |
| 00:45.25 | tru_`z24 | not 3132345678 |
| 00:45.33 | knarfly | it's a local call in a place where there is no 1 required |
| 00:45.37 | tru_`z24 | i know |
| 00:45.39 | tru_`z24 | do it |
| 00:45.40 | tru_`z24 | try it |
| 00:45.50 | tru_`z24 | same situation here |
| 00:45.50 | knarfly | all calls in my area must be 10 digit even local ones |
| 00:45.52 | *** join/#asterisk [hC] (n=hardcore@65.116.224.30) |
| 00:45.56 | tru_`z24 | i know |
| 00:46.02 | tru_`z24 | just try that |
| 00:46.03 | tru_`z24 | please |
| 00:46.07 | knarfly | let me give it a whirl |
| 00:46.27 | tru_`z24 | doesn't matter how you dial the number, its still going to come up in teh caller id as 13132345678 |
| 00:47.01 | ManxPower | What! |
| 00:47.10 | Yourname` | Hi. Someone please come to my rescue. I have 10 dialers with NoOps.. I want to parse data from all those dialers, take those NoOps and give me an answer with the number of those noops, etc. What's the best way to do it? A) Using the manager API to connect to all the dialers and get those NoOps? B) Logging /var/log/asterisk/messages to a central logging and parsing server? C) Using some sort of AGI script (this idea remains sketchy) |
| 00:47.12 | ManxPower | CALLERID number should ALWAYS be 10 digits in the USA. |
| 00:47.19 | ManxPower | the leading 1 is not part of the phone number |
| 00:47.26 | Strom_C | the 1 is the country code :) |
| 00:47.32 | ManxPower | Yourname`: I'm pretty sure nobody has any idea what you are talking about. |
| 00:47.37 | Yourname` | lol |
| 00:47.54 | Yourname` | ManxPower: I want to parse log information, but I want to do realtime and centrally. How? |
| 00:48.34 | ManxPower | Yourname`: you would not do it in asterisk |
| 00:48.37 | knarfly | tru_`z24, it doesn't do the trick...ManxPower is right on spot...USA needs 10 digits even if you're calling into the same room |
| 00:48.50 | ManxPower | needs? |
| 00:48.54 | Yourname` | ManxPower: Then how could I do it? |
| 00:49.07 | ManxPower | Yourname`: write an application to read /var/log/asterisk/messages |
| 00:49.11 | ManxPower | or the CDR |
| 00:49.35 | ManxPower | callerid in the USA is 10 digits. It has nothing to do with how you dial the number. |
| 00:49.48 | Yourname` | ManxPower: Yeah, that's what I'm doing.. but how do I do it centrally and in real time? I understand I can do tail -F /var/log/asterisk/messages. But to do that in 10 different servers? |
| 00:50.04 | knarfly | Yourname`, reading the CDR is pretty simple...I wrote a sh script that reads/parses it into a much easier to read format |
| 00:50.27 | ManxPower | Yourname`: you would not tail -f you would WRITE AN APPLICATION |
| 00:50.45 | ManxPower | Yourname`: you seem to think that what you want to do is simple. It is not simple. |
| 00:50.49 | Yourname` | knarfly: Don't need CDR, just need to read/parse a few NoOps. Across a few asterisk servers and all in real time. |
| 00:51.08 | knarfly | Yourname`, gotcha...over my head I'm afraid |
| 00:51.19 | Yourname` | ManxPower: Well, I'm here asking about it, aren't I? I've been wondering how else I can do it. So you can help, or keep thinking I have it figured out, lol |
| 00:51.53 | Yourname` | Read my question again, I *dont* have it down. I was wondering what would I need to do in order to achieve what I need to achieve. |
| 00:52.00 | knarfly | My attempts to filter an incoming call based on callerid number are all failing |
| 00:52.09 | knarfly | does anyone have the correct syntax? |
| 00:52.34 | knarfly | I've tried eveything I can find in the docs and it just isn't working for me |
| 00:53.04 | _ShrikE | central syslog server? |
| 00:53.07 | gremzoid | Yourname`, there are known knowns, unknowns knowns, knowns unkowns.... etc |
| 00:53.08 | gremzoid | :P |
| 00:53.39 | knarfly | 8-) but what about unknown unknowns |
| 00:53.59 | Yourname` | _ShrikE: That's kind of what I was thinking. But I stopped somewhere when it came to implementation. :S |
| 00:54.29 | gremzoid | Yourname`, that would be a start... read the syslog manpages... syslog -r and -h i think |
| 00:54.35 | *** join/#asterisk Qwell_ (n=north@pdpc/sponsor/digium/Qwell) |
| 00:54.35 | *** mode/#asterisk [+o Qwell_] by ChanServ |
| 00:54.51 | _ShrikE | We do something similar and dump to the fantastic syslog facility in solarwinds orion. |
| 00:55.04 | gremzoid | you'll need to modify your init scripts, where syslogd is started from... |
| 00:55.12 | ManxPower | Poor guy doesn't know about the ex-girlfriend option |
| 00:55.12 | _ShrikE | you can query and alert on just about anything |
| 00:55.30 | ManxPower | Yourname`: most of what you want to know is linux/unix/system admin stuff, not asterisk |
| 00:55.38 | Yourname` | gremzoid: Thing is, I wondering how to do it over the network with the specific asterisk /messages files. |
| 00:55.42 | Yourname` | ManxPower: Yeah, I guess so. |
| 00:55.58 | gremzoid | google and manpages are your freind |
| 00:56.04 | Yourname` | I was wondering if there's a better, easier way already employed in Asterisk, like manager, AGI, etc? |
| 00:57.03 | gremzoid | mmmm agi rocks |
| 00:57.23 | ManxPower | Yourname`: no matter what you do, you'll end up writing code. |
| 00:58.10 | *** join/#asterisk saftsack (n=saftsack@pD9E0742D.dip.t-dialin.net) |
| 00:58.33 | Yourname` | I'm not trying to avoid writing code. Just trying to go in the right direction of writing code. :) |
| 00:59.20 | tru_`z24 | ManxPower: so is the fact that my ${CALLERIDNUM} is returning 1 + the 10 digit phone a configuration item? |
| 00:59.29 | tru_`z24 | and if so, how do i change it to 10 digit only ? |
| 00:59.49 | gremzoid | ${CALLERIDNUM:1} ? |
| 01:00.00 | CCFL_Man2 | it's not bad for entry level, but no extra money with 3rd shift |
| 01:00.18 | CCFL_Man2 | i don't want to be a zombie |
| 01:00.22 | riddlebox | yeah that sucks |
| 01:00.28 | ManxPower | tru_`z24: I dunno. Where is the call coming from? |
| 01:00.33 | tru_`z24 | my cell |
| 01:00.45 | riddlebox | CCFL_Man2, is there a chance to move out of 3rd shift? |
| 01:00.53 | ManxPower | via sip, zap, carrier pidgen, h323, or mind control rays? |
| 01:00.58 | tru_`z24 | Don't get me wrong, I know it is simple to take the right 10 digits... |
| 01:01.06 | CCFL_Man2 | riddlebox: possibly |
| 01:01.06 | tru_`z24 | but I was just curious if its me or the Telco doing it |
| 01:01.18 | gremzoid | shouldn't the callerid contain the country code tho? |
| 01:01.20 | CCFL_Man2 | but i don't want to do it at all |
| 01:01.26 | tru_`z24 | yeah, 1 is the country code |
| 01:01.29 | ManxPower | Cell <-> ??? <-> Asterisk |
| 01:01.30 | tru_`z24 | but i don't need it :_) |
| 01:01.32 | tru_`z24 | :-) |
| 01:01.34 | ManxPower | I need to know what ??? is |
| 01:01.37 | tru_`z24 | my bad |
| 01:01.42 | tru_`z24 | cell -> vonage -> asterisk |
| 01:01.46 | gremzoid | what if i called you (i'm in au) |
| 01:01.51 | ManxPower | vonage is prolly adding the 1 |
| 01:01.52 | gremzoid | that 1 would become +61 |
| 01:02.02 | tru_`z24 | gremzoid: i'm only interested in same country phones for the DNC :-) |
| 01:02.10 | CCFL_Man2 | tru_`z24: don't farking tell me you are using that shitty vonage adapter |
| 01:02.14 | riddlebox | hey I have a question, I use Charter telephone Service, and I need to know if they have disconnect supervision, when someone calls me it rings until voicemail answers everytime even when you hang up after 1 ring |
| 01:02.21 | tru_`z24 | what shitty vonage adapter? |
| 01:02.24 | ManxPower | you only need the country code if the call comes from a different country |
| 01:02.25 | tru_`z24 | this is just for testing |
| 01:02.29 | tru_`z24 | i'll be hooking up to a T1 |
| 01:02.34 | tru_`z24 | i have a x100p for testing |
| 01:02.53 | CCFL_Man2 | riddlebox: call them and ask |
| 01:03.11 | CCFL_Man2 | tru_`z24: why do you even have shitty vonage? |
| 01:03.22 | ManxPower | tru_`z24: Set(CALLERID(num)=${CALLERID(num):1}) before anything else. then just remove that line when you get a real telco line |
| 01:03.23 | tru_`z24 | What else is there to get? |
| 01:03.25 | riddlebox | CCFL_Man2, the problem is I call and ask their stupid representatives and they saw whats disconnect supervision, I have never heard of that |
| 01:03.50 | CCFL_Man2 | tru_`z24: packet 8, sunrocket, voipbuster, quantumvoice, etc |
| 01:03.53 | tru_`z24 | ok, but is :1 just truncating one from the left? |
| 01:03.59 | tru_`z24 | what if the country code WAS 61? |
| 01:04.02 | ManxPower | I doubt vonage supports disconnect supervision. You should call them and tell them that your ANSWERING machines does not detect when the caller hangs up |
| 01:04.03 | tru_`z24 | then it would just drop the 6? |
| 01:04.09 | CCFL_Man2 | riddlebox: ask for highest level tech |
| 01:04.19 | gremzoid | you should use the country code |
| 01:04.26 | gremzoid | all other telephony equipment does |
| 01:04.31 | gremzoid | so why break the standard? |
| 01:04.49 | ManxPower | gremzoid: Huh? in the USA the standard callerid for calls from NANPA countries do NOT include the leading 1 |
| 01:04.56 | riddlebox | yeah I did but was on hold forever |
| 01:05.06 | gremzoid | typical yanks doing things the opposite way |
| 01:05.12 | gremzoid | in AU and europe its the opposite |
| 01:05.14 | gremzoid | :P |
| 01:05.18 | riddlebox | ManxPower, I use charter cable telephone service |
| 01:05.19 | CCFL_Man2 | riddlebox: well, thats the only way to find out |
| 01:05.25 | ManxPower | gremzoid: all calls come in with the country code for local calls? |
| 01:05.25 | tru_`z24 | gremzoid: i agree with you |
| 01:05.30 | tru_`z24 | but i'm working with a legacy system here |
| 01:05.35 | tru_`z24 | and the dnc we have only holds 10 digits |
| 01:05.53 | CCFL_Man2 | tru_`z24: either hack the vonage box and get the sip account, or go to quantumvoice |
| 01:05.54 | gremzoid | yep... it's up to the device to display it or not... we get country code plus area code |
| 01:06.08 | gremzoid | IE: +61732510000 is a brisbane number |
| 01:06.32 | gremzoid | +61 < AU ... 7 < metro queensland |
| 01:06.39 | tru_`z24 | Vonage is international right? |
| 01:06.42 | ManxPower | The thing about NANPA is that the LD code is the same as the country code |
| 01:06.45 | tru_`z24 | Which is probably why they're showing the country cod |
| 01:06.47 | tru_`z24 | code* |
| 01:07.10 | CCFL_Man2 | tru_`z24: vonage has shitty international rates |
| 01:07.10 | ManxPower | + means "dial whatever it is you dial for an international call" |
| 01:07.25 | gremzoid | yea, just a habit... |
| 01:07.35 | Yourname` | _ShrikE: That's what I'm trying to do. :) |
| 01:07.39 | CCFL_Man2 | gremzoid: don't tell me you have vonage? |
| 01:07.44 | gremzoid | the company i work for installs traditional pabx systems (siemens ones) |
| 01:08.07 | gremzoid | we've been having heaps of fun with * and it's AGI |
| 01:08.36 | CCFL_Man2 | gremzoid: link the systems up over the net or a dedicated circuit, it's cheaper |
| 01:08.48 | ManxPower | "over the net"? HAHAHAHA! |
| 01:09.10 | CCFL_Man2 | ManxPower: vpn of course |
| 01:09.19 | ManxPower | Lets see. Do we want it to be reliable? If so, use dedicated circuits. If not, route the calls over the internet. |
| 01:09.21 | CCFL_Man2 | or my personal favorite |
| 01:09.36 | CCFL_Man2 | voice over frame relay |
| 01:09.52 | ManxPower | CCFL_Man2: we send calls over frame relay every day. |
| 01:09.54 | CCFL_Man2 | if it's just used for voice it should work great |
| 01:09.55 | gremzoid | CCFL_Man2, in our country is not actually cheaper |
| 01:10.09 | gremzoid | damn monopoly over our comms/broadband |
| 01:10.29 | CCFL_Man2 | ManxPower: what do you use to interface the frame relay to asterisk? |
| 01:10.34 | gremzoid | well unless your a fairly decent sized business |
| 01:10.36 | CCFL_Man2 | gremzoid: ahh |
| 01:10.53 | ManxPower | All our networking gear is Cisco |
| 01:11.01 | gremzoid | 10GB 1024/1024kbps ADSL will set you back close to $200AUD a month here |
| 01:11.07 | riddlebox | what is another name for disconnect supervision? |
| 01:11.08 | gremzoid | :P |
| 01:11.22 | ManxPower | riddlebox: "my answering machine does not detect when the caller hangs up" |
| 01:11.32 | CCFL_Man2 | gremzoid: i'm amazed you even have running water :P |
| 01:11.34 | adeel | wow, you can get a dedicated 10 mbit box unmetered for $140 per month |
| 01:11.35 | ManxPower | that is another word for "far end disconnect supervision" |
| 01:12.00 | riddlebox | ok I will tell them that, ManxPower its a shame that I have to dumb my self down to get this to work |
| 01:12.05 | gremzoid | CCFL_Man2, for now... we've almost run out in every major city... all ours water supplies are below 15% |
| 01:12.07 | gremzoid | :P |
| 01:12.13 | CCFL_Man2 | ManxPower: you do voice over frame relay or an ip network over frame relay and voice over that? |
| 01:12.14 | gremzoid | so your almost correct! |
| 01:12.16 | ManxPower | riddlebox: apparently you don't deal with the telco much |
| 01:12.28 | CCFL_Man2 | gremzoid: sorry to hear that |
| 01:12.29 | ManxPower | CCFL_Man2: IP network over frame |
| 01:12.38 | riddlebox | I just call tickets in to them, its all AA stuff now |
| 01:12.54 | ManxPower | CCFL_Man2: It was NOT my idea and I fought it every step of the way. |
| 01:12.59 | gremzoid | does asterisk support callerID/addressbooks via LDAP? |
| 01:13.16 | CCFL_Man2 | ManxPower: it'll work well if bandwidth is used properly |
| 01:13.39 | ManxPower | gremzoid: that is like asking if Linux supports callerid/addressbook via ldap. The answer is "yes, if you write it" |
| 01:14.02 | CCFL_Man2 | ManxPower: why didn't part of the frame be used for voice and the other part for ip? |
| 01:14.06 | gremzoid | ahh bugger, i was hoping for somehting like CDR but oh well |
| 01:14.08 | ManxPower | CCFL_Man2: Our frame relay network is 384k with high utilization |
| 01:14.33 | CCFL_Man2 | ManxPower: you poor, poor soul |
| 01:14.42 | ManxPower | CCFL_Man2: Hey, if you have a suggestion on how to divide 384k part for voice, part for data, I'm happy to listen |
| 01:15.12 | ManxPower | We are working on upgrading it to an amazing, blindingly fast 512K |
| 01:15.17 | CCFL_Man2 | ManxPower: how much bandwidth of data you want? |
| 01:15.24 | CCFL_Man2 | whoa!! |
| 01:15.39 | ManxPower | that will require a 2nd T-1 for Frame traffic at HQ |
| 01:16.06 | *** join/#asterisk delmar (n=delmar@ip-58-28-158-154.ubs-dsl.xnet.co.nz) |
| 01:16.43 | CCFL_Man2 | with cisco you can use a variety of bandwidth saving codecsm but i'd stick with g711u |
| 01:16.46 | ManxPower | CCFL_Man2: this company has 3 IT people and a consultant to handle 20+ offices, 400+ users, across 2 states. They don't spend a lot of money on IT |
| 01:16.50 | delmar | I'm having some issues with an Asterisk box behind Nat. When I set the externip in sip.conf the peers go unreachable. |
| 01:17.07 | CCFL_Man2 | ManxPower: they need atleast |
| 01:17.13 | CCFL_Man2 | ManxPower: 1M |
| 01:17.24 | ManxPower | 1M |
| 01:17.45 | CCFL_Man2 | that'll cost them $10000 more a month though |
| 01:17.50 | ManxPower | You do understand that if you want QoS on a frame network you need the port speed to be the same as a CIR, right? |
| 01:18.10 | CCFL_Man2 | oh, thats right |
| 01:18.48 | ManxPower | so we can fit 3 remote offices at 512K going into the cloud if we have a full T-1 into the cloud at HQ |
| 01:19.13 | CCFL_Man2 | if they are going to have two ds1 circuits comming in them why not use one for voice and one for data |
| 01:19.19 | CCFL_Man2 | or is it cost? |
| 01:19.25 | ManxPower | The reason for CIR = Port speed is that by the time the router gets a congestion message from the frame cloud, audio has already gone to hell |
| 01:19.38 | ManxPower | CCFL_Man2: It is always cost. |
| 01:19.40 | CCFL_Man2 | i know |
| 01:19.46 | CCFL_Man2 | hmm.. |
| 01:20.07 | ManxPower | As it is the policy is "if it goes over the frame don't bother to call us if the call quality sucks" |
| 01:20.08 | CCFL_Man2 | you can do voice over frame relay nicely if it's just used for voice |
| 01:20.17 | CCFL_Man2 | lol |
| 01:20.35 | ManxPower | we have several point-to-point t-1s as well to various other offices. |
| 01:20.50 | CCFL_Man2 | wtf |
| 01:21.14 | CCFL_Man2 | they spend money on that |
| 01:21.33 | ManxPower | CCFL_Man2: Uh, these T-1s are like $400/month. Frame is $350/month |
| 01:21.47 | CCFL_Man2 | those T1s used as backups? |
| 01:21.48 | ManxPower | It all depends on WHERE the two end points are. |
| 01:21.56 | CCFL_Man2 | ahh, thats right |
| 01:22.05 | ManxPower | No, if there is a T-1 from the remote office to HQ, then that office is not on the frame cloud. |
| 01:22.20 | CCFL_Man2 | why can't they just put everything on the frame? no redundancy? |
| 01:22.34 | CCFL_Man2 | or isn't there any redundancy in the first place? |
| 01:22.48 | ManxPower | CCFL_Man2: because putting everything on the frame would be too expensive to get the same bandwidth |
| 01:23.11 | CCFL_Man2 | they added all this up and made sure? |
| 01:23.24 | ManxPower | HOW an office connects to HQ is determined by how much revenue that office generates, the location of the office, and the cost of Frame .vs. point to point T-1. |
| 01:23.36 | CCFL_Man2 | oh dear |
| 01:23.43 | CCFL_Man2 | i quit! |
| 01:23.52 | CCFL_Man2 | :P |
| 01:24.44 | ManxPower | So we could have Mandeville Office connect to Covington using Frame Relay at 384K for $350/month and use 384K of the Covington Frame connection, OR we can put a full point to point T-1 between the Mandeville office and the Covington office for $400/month, since the offices are only like 10 miles apart. |
| 01:24.44 | *** join/#asterisk Buhntz (i=Boones@port-212-202-170-97.dynamic.qsc.de) |
| 01:25.23 | ManxPower | which would you pick? |
| 01:25.29 | CCFL_Man2 | so some offices will have only the frame relay link because it's the one that matches it's revenue? |
| 01:26.01 | ManxPower | CCFL_Man2: most offices have a frame link because a point to point T-1 would be $1,200 or more for a connection to HQ |
| 01:26.12 | CCFL_Man2 | ahh, yeah |
| 01:26.15 | ManxPower | Not most, but many |
| 01:26.32 | ManxPower | So, point to point T-1 for $1,200 or 384K frame for $350 |
| 01:26.36 | CCFL_Man2 | well, i would too, mainly because i'd make little money |
| 01:27.08 | ManxPower | Then there are the REALLY small offices with a Linksys VPN router connecting to HQ |
| 01:27.12 | ManxPower | over the internet. |
| 01:27.22 | CCFL_Man2 | linksys vpn, sweet! |
| 01:27.26 | ManxPower | CCFL_Man2: I don't sell hardware or telecom services. |
| 01:27.35 | ManxPower | I consider it a conflict of interest. |
| 01:28.10 | CCFL_Man2 | well, i understand the reasoning behind it, and with such a situation requires such a solution |
| 01:28.26 | *** join/#asterisk ta^3 (n=tacvbo@65.116.224.30) |
| 01:28.32 | CCFL_Man2 | how much bandwidth of data is needed to link the sites? |
| 01:29.14 | ManxPower | CCFL_Man2: that depends on who you ask. IT says 384K is enough for all company critical applications. Users think 10Mbps is not fast enough. |
| 01:29.28 | ManxPower | But users thing watching Youtube and porn is "mission critical" |
| 01:29.57 | ManxPower | The offices send/receive e-mail, upload a few files, and use windows terminal server. |
| 01:30.12 | CCFL_Man2 | why not have internet provided locally and company/voice provided over the frame? |
| 01:30.33 | ManxPower | CCFL_Man2: because then we would have to manage 20 firewalls, rather than 1 firewall |
| 01:30.42 | CCFL_Man2 | ahh, right |
| 01:31.20 | CCFL_Man2 | so e-mail, small file transfer, winders terminal server, and voip? |
| 01:31.26 | ManxPower | The IT department handles all routers, switches, e-mail, servers, voice, data, and PC support for 400 people...with 3 full time staff (including the manager) and 1 consultant |
| 01:31.38 | CCFL_Man2 | ahh |
| 01:31.52 | ManxPower | so if it makes more work for us, it is not going to happen |
| 01:32.38 | CCFL_Man2 | how many calls max do they want going over the frame relay? |
| 01:32.44 | ManxPower | voip is not a mission critical service |
| 01:32.44 | delmar | ok this is wierd. If I set the externip= in sip.conf my sip peers will go unreachable, but if it's not set, they are reachable but calls won't work. |
| 01:33.24 | ManxPower | CCFL_Man2: Um, most of the offices get free long distance to all over offices using the standard telco lines. |
| 01:33.30 | ManxPower | voip is not a major issue. |
| 01:33.53 | ManxPower | VoIP lets users dial 4-digit extensions, it does not save money. |
| 01:34.30 | ManxPower | on the point to point t-1 connections we use GSM as the codec and reserve 256k for VoIP |
| 01:34.41 | ManxPower | maybe it is 384K on those links |
| 01:34.48 | CCFL_Man2 | so dedicating X amount of bandwidth on the frame for VoFR won't be something they'll want to do? |
| 01:34.58 | ManxPower | CCFL_Man2: nope. |
| 01:35.47 | CCFL_Man2 | ManxPower: because when voice bandwidth is not in use data bandwidth will be limited and they don't want that? |
| 01:36.15 | riddlebox | ManxPower, I got it to tier 2 and now they say that the "problem I have" is a setting that cannot apply without tier 2 doing it |
| 01:36.20 | riddlebox | I would have never thought to just act stupid and I will get my way, I guess ignorance is bliss |
| 01:37.33 | ManxPower | CCFL_Man2: We are happy to reserve bandwidth for voice on links that are T-1s. Not acceptable on 512K or 384K links |
| 01:38.03 | ManxPower | and only 6 of the offices even have Asterisk |
| 01:38.07 | CCFL_Man2 | ManxPower: then i don't think there is a solution |
| 01:38.17 | ManxPower | well if you count that bastard Asterisk/Nortel beast we have at HQ |
| 01:38.24 | ManxPower | CCFL_Man2: I wasn't looking for one. |
| 01:38.51 | CCFL_Man2 | ManxPower: i know, but i wanted to be "the man" and find you one :P |
| 01:38.58 | ManxPower | you mentioned voice over frame relay and I said we do voice over ip over frame relay |
| 01:39.13 | ManxPower | CCFL_Man2: I've been doing this stuff for over 12 years. |
| 01:39.15 | CCFL_Man2 | which is just wrong |
| 01:39.36 | ManxPower | not Asterisk, of course, but WAN and LAN stuff. |
| 01:39.37 | CCFL_Man2 | ahh |
| 01:39.46 | ManxPower | Asterisk is a very small part of what I do. |
| 01:39.57 | ManxPower | anyway I need to get home. |
| 01:40.06 | CCFL_Man2 | i don't blame you |
| 01:40.27 | CCFL_Man2 | shit, i need to call HR tomorrow to get told i won't get the job |
| 01:41.47 | jsaunders | You could always.... not call? |
| 01:42.00 | adeel | when configuring polycom phones for failover, what should the 'reg.x.auth.optimizedInFailover' setting be? i've read the administrator's guide, and its still not clear to me |
| 01:42.12 | CCFL_Man2 | jsaunders: they want me to call back |
| 01:42.23 | CCFL_Man2 | i should be a consultant |
| 01:42.55 | jsaunders | Of course they do. Certain people derive satisfaction from telling other people they didn't get the job. It's sick I tell you, sick. |
| 01:43.33 | jsaunders | Course, it's all about your confidence level. Were you nervous? That's a no-no. |
| 01:43.47 | jsaunders | Gotta walk in there like you own the place. :D |
| 01:46.14 | delmar | Here is my setup. Asterisk-SIP ------> ST608WL(NAT) ------------>Multiple Providers. rtp.conf has ports 10000to20000 set and the router is forwarding 5060 and 10000-20000 to the ASterisk box. If I set externip= in my sip.conf the SIP peers all go unreachable. if I don't set externip= then calls don't go through. Any ideas? |
| 01:46.29 | delmar | I really hate SIP & NAT :( |
| 01:47.51 | *** join/#asterisk bkruse_home (n=bkruse@user-24-214-44-217.knology.net) |
| 01:48.03 | *** join/#asterisk shido6 (n=shido6@74-130-59-184.dhcp.insightbb.com) |
| 01:48.10 | CCFL_Man2 | jsaunders: actually i was very confident, but i told the guy i don't want to work 3rd shift |
| 01:48.30 | CCFL_Man2 | and they want availablity for all shifts |
| 01:51.56 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 01:52.17 | CCFL_Man2 | sucks how dedicated links cost so much |
| 01:57.34 | MaliutaWrk | delmar: you could run siproxd on the nat box |
| 01:57.53 | delmar | Maliuta, sure. whats that gonna do? |
| 01:58.27 | delmar | MaliutaWrk, ^ |
| 01:58.35 | MaliutaWrk | delmar: I was doing that until just recently ... I have added a some specific NAT rules for my asterisk box and changed the sip.conf files |
| 01:58.38 | delmar | MaliutaWrk, err. not on the router thats doing NAT |
| 01:59.10 | delmar | MaliutaWrk, I thought there were ways to make my above solution play nice with SIP ? |
| 01:59.37 | delmar | MaliutaWrk, like.. by just port forwarding those ports and setting the externip setting... but this is not so |
| 01:59.38 | MaliutaWrk | delmar: you will need to have the externip set in the general section of sip.conf - you should also have set localnet |
| 01:59.54 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 01:59.57 | delmar | MaliutaWrk, done both yes |
| 02:00.07 | MaliutaWrk | delmar: and you should have nat=yes for the external providers and nat=no for the local phones |
| 02:00.16 | delmar | MaliutaWrk, when I set externip to my public static... the SIP peers go unreachable |
| 02:00.35 | delmar | MaliutaWrk, ok lets go over it all. just a sec and I will check it all out |
| 02:00.38 | MaliutaWrk | delmar: pastebin your sip.conf |
| 02:01.27 | MaliutaWrk | of course obscure anything incriminating like the passwords and IP's |
| 02:02.21 | delmar | MaliutaWrk, will see if we get that far. ok so.. what about the canreinvite= setting. can canreinvite be used in that situation? |
| 02:02.35 | MaliutaWrk | delmar: and is the router DNATing incoming SIP and RTP? |
| 02:02.36 | delmar | will set to no for no wanyway |
| 02:02.46 | MaliutaWrk | delmar: simple answer is no |
| 02:03.15 | delmar | MaliutaWrk, 5060 and and 10000:20000 are forwarding to the Asterisk box. |
| 02:03.34 | MaliutaWrk | delmar: I have it set to no for all my hosts (but I am keeping asterisk in the loop for other reasons) |
| 02:04.00 | delmar | MaliutaWrk, yep. pain in the butt when u are trying to record things and the call vanishes :P |
| 02:04.37 | MaliutaWrk | delmar: pastebin the config so I can have something solid to work on :) |
| 02:05.17 | delmar | MaliutaWrk, ok. give me a minute to edit out passwds and such and ill PM u the link |
| 02:05.35 | *** join/#asterisk kavelot (i=x@201-68-27-12.dsl.telesp.net.br) |
| 02:05.43 | *** join/#asterisk nighty^ (n=nighty@72.0.72.123) |
| 02:07.00 | kavelot | what do I need to make and receive external calls (telephone system) using ASTERISK? Is it something like Skype (and examples?)? |
| 02:07.03 | kavelot | and = any |
| 02:07.44 | MaliutaWrk | kavelot: that is very general question |
| 02:07.55 | kavelot | yes, I didn't understand the concept exactly |
| 02:08.12 | *** join/#asterisk techie (n=techie@adsl-68-127-127-133.dsl.frsn02.pacbell.net) |
| 02:08.19 | MaliutaWrk | kavelot: you can recieve calls on an asterisk box that come over the PSTN |
| 02:09.57 | MaliutaWrk | kavelot: I for example use asterisk like an old iron PBX. At home I have a physical phone line coming into my asterisk box, I also have 2 seperate SIP providers (giving me separate "phone lines" in and out over the net) |
| 02:10.12 | *** join/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 02:10.19 | MrTelephone | has anyone tried vmux-110 by rad? |
| 02:10.39 | kavelot | hm, how do you plug the physical phone lines in asterisk? something like a modem? |
| 02:11.27 | MrTelephone | analog fxo /fxs cards |
| 02:11.37 | MrTelephone | digium or sangoma fxo cards |
| 02:11.38 | MaliutaWrk | I have an digium TMD400P card with on FXO and one FXS ... so the phone line plugs into that port like it would to a normal phone, and my handset plugs into the other port |
| 02:11.59 | MrTelephone | phone line to fxo and handset to fxs |
| 02:12.25 | kavelot | got it |
| 02:13.03 | kavelot | i'm reading more, thanks for the start up |
| 02:13.15 | kavelot | i was lost :) |
| 02:13.25 | MaliutaWrk | kavelot: I have cordless handsets that could be plugged into a normal phone line. the connect to the asterisk box and I use it to direct the calls how I want (out over the physical phone line or to one of the VoIP providers depending on what I want to do) |
| 02:14.23 | MaliutaWrk | kavelot: I also have a Cisco IP phone that is attached to my network that I use in the same way, and I use "softphones" off of desktop and laptop computers |
| 02:15.26 | MrTelephone | what about faxing |
| 02:15.36 | MrTelephone | what kind of device can you use that supports t38 |
| 02:17.14 | gremzoid | i'd be keen to know as well.. from what i've read asterisk (or more so VoIP in general) dosn't seem to handle faxing to well... |
| 02:17.21 | MrTelephone | those ricoh fax machines seem towork good with asterisk |
| 02:17.36 | MrTelephone | im trying to provide service and faxing is a hit and miss |
| 02:17.56 | MrTelephone | i was looking at rad vmux-110 to is t1 -> ethernet -> t1 |
| 02:18.24 | MrTelephone | they say it compresses t1 16:1 but they are talking about the difference between g711 and g729 |
| 02:22.00 | *** join/#asterisk tru_`z24 (n=truz_`24@74-129-166-232.dhcp.insightbb.com) |
| 02:25.40 | *** join/#asterisk rozelli (n=eu@201.53.93.182) |
| 02:27.23 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 02:27.24 | rozelli | can some1 help me?Ive copied the g729 sound files to /var/lib/asterisk/sounds/a2billing but a2billing keeps saying file prepaid-enter-dest does not exist in any format |
| 02:30.58 | bkruse_home | rozelli: /var/lib/asterisk/sounds |
| 02:31.00 | bkruse_home | but it there! |
| 02:31.04 | bkruse_home | put* |
| 02:31.05 | bkruse_home | and try |
| 02:31.36 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 02:31.36 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 02:32.35 | rozelli | bkruse_home its already there |
| 02:32.43 | rozelli | all .g729 files |
| 02:35.21 | gremzoid | reloaded? |
| 02:35.48 | rozelli | yes |
| 02:36.06 | gremzoid | sounds like a stupid question... but i got a $450 urgent site visit last week which involved plugging a power pack back in :P |
| 02:36.21 | blitzrage | handy |
| 02:36.26 | d3wayne | bkruse_home: mooooooooo |
| 02:36.44 | MrTelephone | gremoid, why did you charge 450 for that? |
| 02:36.49 | bkruse_home | d3wayne: hows everything that side of the US? |
| 02:37.07 | d3wayne | which side are you on ? |
| 02:37.31 | d3wayne | are you 'there' ? |
| 02:38.09 | gremzoid | MrTelephone, costs money to drop everything on other jobs... we don't sit around doing nothing all day :P |
| 02:38.16 | MrTelephone | canadian pri's don't carry ANI II codes, how do you block collect calls? |
| 02:39.00 | MrTelephone | emergency call out ofr me is 100 bucks |
| 02:40.28 | bkruse_home | d3wayne: no, are you 'there'? |
| 02:40.29 | bkruse_home | ll |
| 02:40.53 | d3wayne | bkruse_home: no I'm here |
| 02:41.06 | bkruse_home | d3wayne: well then nvm lol |
| 02:41.35 | d3wayne | oh, you meant 'that' side of the country. I actually thought you meant this side, but that you were on that side |
| 02:44.33 | *** part/#asterisk rozelli (n=eu@201.53.93.182) |
| 02:48.53 | *** join/#asterisk brian (n=brian@unaffiliated/brian) |
| 02:53.12 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 02:55.45 | *** part/#asterisk b1ch0 (n=ralabiso@static-200-105-209-46.acelerate.net) |
| 02:55.59 | *** part/#asterisk techie (n=techie@adsl-68-127-127-133.dsl.frsn02.pacbell.net) |
| 02:57.34 | *** join/#asterisk Goldfisch (n=gregturn@user-0c6t46t.cable.mindspring.com) |
| 02:58.01 | *** join/#asterisk microcape (n=microcap@c-68-42-66-199.hsd1.mi.comcast.net) |
| 03:07.10 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 03:10.49 | PSU_Boss | Hello, |
| 03:12.57 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 03:21.36 | *** join/#asterisk brian (n=brian@unaffiliated/brian) |
| 03:22.14 | *** join/#asterisk brian_ (n=brian@unaffiliated/brian) |
| 03:24.54 | *** join/#asterisk marcan (i=1337@host214-134.cvd.fit.edu) |
| 03:27.13 | *** join/#asterisk israr (n=israr@210-56-12-34.DSL.isb.comsats.net.pk) |
| 03:37.40 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:38.35 | *** join/#asterisk THX2000 (n=the@netblock-208-127-150-56.dslextreme.com) |
| 03:39.08 | THX2000 | Anyone know what the charge (if any) is to forward on busy from an AT&T pots line to an ITSP? |
| 03:39.44 | *** join/#asterisk Daejeo1 (n=chatzill@211.177.189.60) |
| 03:42.47 | CCFL_Man2 | skype is total shit |
| 03:47.18 | Daejeo1 | what is default password polycom 501 |
| 03:47.47 | Daejeo1 | default username polycom password? |
| 03:47.52 | *** join/#asterisk brian (n=brian@unaffiliated/brian) |
| 03:48.04 | CunningPike | PlcmSpIp |
| 03:51.40 | wunderkin | that's the ftp password.. admin is 456, its all in the admin manual :P |
| 03:52.29 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-e766dd4e6fdd1555) |
| 03:52.52 | *** join/#asterisk bmg505 (n=leon@196.209.183.188) |
| 03:55.17 | CCFL_Man2 | Strom_M: my F1 handset sounds muffled, the speaker bad? |
| 03:59.02 | *** join/#asterisk PepOSX (n=pepOSX@190.72.151.23) |
| 04:02.19 | Daejeo1 | wunderkin: how can i reset the phone? |
| 04:03.06 | *** join/#asterisk Un1x_Laptop (i=Sean@CPE002211997865-CM000f9fac821a.cpe.net.cable.rogers.com) |
| 04:05.15 | *** join/#asterisk brian (n=brian@unaffiliated/brian) |
| 04:11.21 | MaliutaWrk | Does anyone have an idea on why - when I have specifically set nat=yes on peers in the sip.conf file, and sip is behaving properly - a 'sip show peers' lists the NAT'd peers as N in the NAT field? |
| 04:12.34 | CunningPike | Maliuta: Because asterisk has determined that NAT is not required |
| 04:13.37 | *** join/#asterisk nate3472 (n=nate@host-72-174-96-98.mtr-co.client.bresnan.net) |
| 04:14.11 | MaliutaWrk | CunningPike: but it is still actually putting the externip into the SIP headers (otherwise all my SIP from my providers would be trying to go to a 10.x.x.x IP |
| 04:14.43 | CunningPike | Maliuta: But your SIP peers are in the same network as your asterisk? |
| 04:14.48 | nate3472 | anybox know how i can make # used as a 'send' key, so users can dial an extension like 303 and press # to speed it up? :D |
| 04:15.03 | CunningPike | nate3472: What phone? |
| 04:15.18 | nate3472 | CunningPike: outside callers, calling into a menu |
| 04:15.55 | nate3472 | CunningPike: all my phone's have send keys, and my linksys adapters already work with extension+'#' |
| 04:16.40 | MaliutaWrk | CunningPike: no, they are my DID providers on the real internets |
| 04:16.57 | CunningPike | Maliuta: So, you're registering to them? |
| 04:17.04 | *** join/#asterisk israr (n=israr@210-56-12-34.DSL.isb.comsats.net.pk) |
| 04:17.12 | CunningPike | With register statements in your sip.conf? |
| 04:17.16 | MaliutaWrk | CunningPike: the registers are going out to them via the NAT box and coming back fine |
| 04:17.33 | MaliutaWrk | CunningPike: and incoming and outgoing calls are working fine |
| 04:17.49 | CunningPike | Maliuta: If you're using register= and externip=, then that's not the same as nat= |
| 04:18.15 | CunningPike | Maliuta: I take it that your asterisk is registering with them, rather than the other way around |
| 04:18.22 | MaliutaWrk | yeah |
| 04:18.38 | MaliutaWrk | the packets still have to traverse the DNAT on the way out |
| 04:18.50 | MaliutaWrk | and the SNAT on the way in |
| 04:19.31 | CunningPike | OK - so the NAT column in sip show doesn't apply - it is a setting for sip peers/users that tells asterisk whether to trust the ip address sent by the UA (the ip address that the UA thinks it's at) or to trust the src ip as reported by the tcp headers |
| 04:19.52 | *** join/#asterisk Teln1100A (i=hello123@bas2-toronto12-1167860932.dsl.bell.ca) |
| 04:21.18 | MaliutaWrk | kewl |
| 04:26.14 | nate3472 | CunningPike: I have people calling into my pbx, who try and dial my extension and then they press # ....but when they include # my asterisk box tries to connect them to 205# which is not a valid extension... |
| 04:26.23 | nate3472 | can i just drop the #, or ignore it? |
| 04:33.29 | *** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
| 04:33.46 | nate3472 | anyone? |
| 04:37.42 | *** join/#asterisk grimsy (n=chatzill@203-206-220-214.perm.iinet.net.au) |
| 04:43.32 | *** part/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
| 04:45.08 | *** join/#asterisk brian (n=brian@unaffiliated/brian) |
| 04:47.17 | *** join/#asterisk brian (n=brian@unaffiliated/brian) |
| 05:03.49 | luke-jr | Can AGI use functions? |
| 05:03.55 | luke-jr | eg, CALLERID(num) |
| 05:06.36 | *** join/#asterisk metfan2007 (n=metfan20@189.180.216.55) |
| 05:06.38 | kaldemar | nate3472: you can remove it with function CUT. exten => _X.#,1,Set(num=${CUT(EXTEN,#,1)}), exten => _X.#,2,Goto(${num},1) or something like that. |
| 05:07.45 | *** join/#asterisk andrew` (n=andrew@69-12-140-101.dsl.dynamic.sonic.net) |
| 05:08.08 | metfan2007 | Hi all, I have a lot of problems installing zaptel in a new server with CentOS 4.4, it does not create /dev/zap and I'm using a TE120P card... any clue??? |
| 05:08.23 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 05:15.30 | nate3472 | kaldemar: awesome, thanks |
| 05:15.32 | kaldemar | nate3472: actually just exten => _X.#,1,Goto(${EXTEN:0:$[${LEN(${EXTEN})} - 1]}) should also do. that removes the last digit. |
| 05:15.53 | nate3472 | thats sweet |
| 05:16.17 | [TK]D-Fender | no, there should be no # at all <- |
| 05:16.30 | *** join/#asterisk yxa (n=lonari@58.185.90.101) |
| 05:16.32 | [TK]D-Fender | and that pattern doesn't work. |
| 05:16.33 | kaldemar | that one needs ",1" (the priority) in there too. |
| 05:16.45 | nate3472 | hmm |
| 05:16.48 | [TK]D-Fender | it does NOT "# terminate" like you think |
| 05:16.49 | nate3472 | i confused now |
| 05:17.03 | [TK]D-Fender | nate3472, pastebin your ACTUAL dialplan |
| 05:17.05 | [TK]D-Fender | ~pb |
| 05:17.05 | jbot | A Pastebin is a place to paste your stuff 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 |
| 05:17.07 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^ |
| 05:18.29 | nate3472 | [TK]: its a real simple dialplan, its just that when i tell my sister to dial my extension when she calls into the pbx, she dial 205# instead of 205 |
| 05:18.52 | nate3472 | I'm just wanting it to either ignore it or work as a send key |
| 05:18.56 | ^JimmyRidge^ | dang why cant i call my pbx :( |
| 05:19.07 | ^JimmyRidge^ | this is driving me crazy |
| 05:20.23 | [TK]D-Fender | nate3472, stop TALKING about your dialplan and SHOW ME. If she can even DIAL # and it gets accepted when you don't want it to, you've clearly done something wrong. |
| 05:20.26 | kaldemar | oh yes, that pattern will match all numbers, whether there is a # or not. |
| 05:21.12 | kaldemar | so the second one will remove the last digit from numbers that don't have # => not good. |
| 05:21.54 | *** join/#asterisk Delmar (n=delmar@ip-58-28-158-154.ubs-dsl.xnet.co.nz) |
| 05:22.13 | [TK]D-Fender | kaldemar, You gain wisdom child :) |
| 05:23.00 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 05:24.00 | kaldemar | [TK]D-Fender: i'm not all stupid, i blame a sleepless night for that. :) |
| 05:25.52 | metfan2007 | When you install a digium card like TE120P, linux detect the card and autocatically create /dev/zap right?? even if zaptel is not installed, right?? |
| 05:26.17 | Qwell_ | metfan2007: no |
| 05:27.18 | yxa | anyone have any ideas why would a digium PRI card with Octasic H/W EC I am still getting echo intermittently? any suggestions greatly appreciated |
| 05:27.18 | metfan2007 | Qwell_:I can't get linux create /dev/zap with my new TE120P, any clue? |
| 05:27.26 | Corydon76-dig | Mmmm, crack. |
| 05:27.26 | Qwell_ | metfan2007: install zaptel |
| 05:27.54 | yxa | metfan2007 did zaptel load properly? sometimes it required a cold reboot |
| 05:28.00 | metfan2007 | Qwell_:Zaptel is installed, no errors during make, make install |
| 05:28.08 | Qwell_ | is it loaded? |
| 05:28.16 | [TK]D-Fender | MODPROBE <-------------- |
| 05:28.26 | kaldemar | actually, linux will automatically install asterisk and create your custom dialplan upon the card detection. |
| 05:28.30 | Qwell_ | [TK]D-Fender: You came up in conversations tonight at the astricon party :p |
| 05:28.36 | Qwell_ | because you rock |
| 05:28.37 | [TK]D-Fender | :O |
| 05:28.41 | Qwell_ | for the reason above |
| 05:28.58 | Qwell_ | juggie was giving you props, heh |
| 05:29.04 | metfan2007 | lsmod shows zaptel and wcte12xp |
| 05:29.24 | [TK]D-Fender | Qwell_, Like my unabashed love of capitalization for emphasis, and my rapier wit? ;) |
| 05:29.41 | Qwell_ | yxa: I would highly recommend calling Digium support in the morning. They'll be quite happy to give you installation support. :) |
| 05:29.48 | Qwell_ | [TK]D-Fender: something like that ;) |
| 05:30.17 | metfan2007 | Qwell_: when I start zaptel (service zaptel start) it does not show anything.... |
| 05:30.38 | Qwell_ | metfan2007: You could also call Digium support. |
| 05:30.48 | Qwell_ | When you buy our hardware, we give you free install support |
| 05:31.04 | metfan2007 | Qwell_: and when I try a ztcfg -vvv says that /dev/zap missing |
| 05:31.06 | metfan2007 | mmmm |
| 05:31.30 | metfan2007 | Qwell_: 24/7? |
| 05:31.39 | Qwell_ | no, like 5/9 or something |
| 05:31.45 | Qwell_ | erm, 9/5? whatever |
| 05:31.55 | metfan2007 | mmm |
| 05:31.59 | metfan2007 | ok |
| 05:32.05 | yxa | Qwell i did. just hoping for some suggestions |
| 05:32.07 | Qwell_ | but, call during the day tomorrow, they'll be glad to help you |
| 05:32.13 | Qwell_ | yxa: they weren't able to help? |
| 05:32.52 | yxa | Qwell they have not replied yet :) |
| 05:32.59 | Qwell_ | ahh |
| 05:33.28 | Corydon76-dig | Qwell_: you back yet? |
| 05:33.38 | Corydon76-dig | Or tomorrow? |
| 05:33.57 | Qwell_ | saturday |
| 05:34.03 | Qwell_ | like 9pm |
| 05:34.04 | Corydon76-dig | Ah |
| 05:34.22 | Corydon76-dig | Wow, that's pretty late for Digiumites and conferences... |
| 05:34.49 | Corydon76-dig | When I've gone, I'm like the last one left |
| 05:35.05 | Qwell_ | file usually ends up staying late... |
| 05:35.25 | nate3472 | exten => s,n,WaitExten <---- should this prompt not be taking # literally? |
| 05:35.44 | Corydon76-dig | So you've volunteered to spoon and otherwise keep him company? ;-) |
| 05:36.09 | [TK]D-Fender | nate3472, I asked you pastebin you entire dialplan 20 minutes ago |
| 05:36.23 | nate3472 | hmmm, what does that mean.... |
| 05:36.34 | nate3472 | i dont want to paste it, hehe |
| 05:36.49 | yxa | sorry for repeat: anyone have any ideas why would a digium PRI card with Octasic H/W EC I am still getting echo intermittently? any suggestions greatly appreciated |
| 05:37.16 | yxa | i'm using linksys SPA-942s |
| 05:38.11 | [TK]D-Fender | nate3472, If you keep hiding the problem we can't help you |
| 05:38.59 | [TK]D-Fender | yxa, have you tried changing your firmware? |
| 05:39.28 | yxa | [TK]D-Fender you mean the card's firmware can be updated? |
| 05:39.38 | nate3472 | [TK]: i'm gonna cut alotta junk outta this dialplan, if it dont work i'll paste it |
| 05:39.41 | [TK]D-Fender | yxa, Yes |
| 05:39.45 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 05:40.26 | [TK]D-Fender | nate3472, it should take very little time to find out where the error is. So just go ahead an pastebin it or you're wasting our time |
| 05:41.26 | yxa | [TK]D-Fender where can i find such firmware? |
| 05:41.36 | nate3472 | wasting your time? |
| 05:41.39 | [TK]D-Fender | yxa, www.sangoma.com <---------- |
| 05:41.53 | [TK]D-Fender | nate3472, asking for our help and not showing us the problem. |
| 05:41.56 | *** join/#asterisk Delmar (n=delmar@ip-58-28-158-154.ubs-dsl.xnet.co.nz) |
| 05:42.07 | nate3472 | that doesnt fucking mean i'm wasting your time |
| 05:42.17 | [TK]D-Fender | nate3472, thats like driving to the mechanic, asking him to fix your car, and then not letting him LOOK AT IT. |
| 05:42.29 | nate3472 | i asked a question |
| 05:42.35 | nate3472 | didn't ASK you to look at it |
| 05:42.36 | yxa | [TK]D-Fender i have a digium card, not sangoma |
| 05:42.47 | [TK]D-Fender | nate3472, Yeah "whats wrong with my dialplan" and then not showing us. |
| 05:42.50 | nate3472 | see what crack does? |
| 05:43.02 | nate3472 | your quoting me wrong too |
| 05:43.22 | metfan2007 | do you know how to debug or trace an init.d script?? |
| 05:43.27 | [TK]D-Fender | <nate3472> [TK]: its a real simple dialplan, its just that when i tell my sister to dial my extension when she calls into the pbx, she dial 205# instead of 205 |
| 05:43.33 | [TK]D-Fender | <nate3472> I'm just wanting it to either ignore it or work as a send key |
| 05:43.48 | nate3472 | your wasting your own time man |
| 05:44.37 | kaldemar | nate3472: err... he is voluntarily trying to help you. |
| 05:44.39 | kiscokid | nate: you could have pastebined the thing already instead of wasting our time arguing about it |
| 05:44.40 | [TK]D-Fender | nate3472, Well I guess if you actually want help you'll come around to letting us. You seem to think we're psychic and simply "know" whats wrong. Sorry to disappoint. |
| 05:46.07 | nate3472 | kaldemar: i know... but i'm not here wasting anyone's time. i'm just here chating |
| 05:46.07 | [TK]D-Fender | anyways, guess I've more productive things to be doing.... |
| 05:56.25 | nate3472 | kiscokid: do you really feel like i wasted your time too? |
| 05:57.18 | flenders | nate3472: shut up mate... you aked for help. |
| 05:57.49 | flenders | s/aked/asked/ |
| 05:58.38 | kiscokid | nate: yes |
| 06:00.04 | nate3472 | 14? |
| 06:00.09 | *** part/#asterisk nate3472 (n=nate@host-72-174-96-98.mtr-co.client.bresnan.net) |
| 06:11.10 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 06:26.15 | *** join/#asterisk grndslm (n=grndslm@24-116-87-97.cpe.cableone.net) |
| 06:26.32 | ^JimmyRidge^ | dang FX* cards are a bit pricey |
| 06:27.14 | ^JimmyRidge^ | know of any cards with just one FXO and one FXS? |
| 06:29.09 | metfan2007 | just to let you know... the problem with zaptel is ok now, there is a bug in zaptel 1.2.19, make config create the zaptel file to load modules in /etc/default/zaptel, but the init script try to get it in /etc/sysconfig/zaptel |
| 06:29.12 | bmg505 | must it be a card? |
| 06:29.36 | bmg505 | sipura 3000 could do the job |
| 06:30.42 | ^JimmyRidge^ | nah i want it to go through asterisk |
| 06:30.56 | ^JimmyRidge^ | IVR from the outside line |
| 06:35.19 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 06:35.39 | Strom_M | ^JimmyRidge^: digium TDM11B |
| 06:37.05 | ^JimmyRidge^ | dang man thats like 200$ |
| 06:37.20 | Strom_M | well, yeah |
| 06:37.26 | Strom_M | this stuff isn't cheap |
| 06:37.32 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 06:37.59 | styelz | you can get cheap x100p cards for around $10 |
| 06:38.42 | ^JimmyRidge^ | whats with the red/green modules strom? |
| 06:39.07 | ^JimmyRidge^ | dif voltages? |
| 06:39.45 | Strom_M | one is FXO, one is FXS |
| 06:40.01 | Strom_M | styelz: i'd rather stick my head in a bucket of hyaena offal |
| 06:40.15 | styelz | green if FXS red is FXO |
| 06:40.58 | ^JimmyRidge^ | oh so that card even though it has 4 ports... only 2 modules... but can upgrade to more lines!? |
| 06:42.46 | styelz | shit x100p works ok with hpec |
| 06:45.42 | Strom_M | ^JimmyRidge^: yes |
| 06:48.08 | *** join/#asterisk gardo (n=gardo@203.82.42.106) |
| 06:59.10 | denon | anyone awake from NZ? |
| 07:01.35 | *** join/#asterisk Dandan (i=dandan@wsip-70-167-100-158.ri.ri.cox.net) |
| 07:02.10 | Dandan | any1 alive? :) |
| 07:03.03 | creativx | yeah, the guy before you who asked if anyone was alive |
| 07:03.18 | Dandan | hehehe |
| 07:03.19 | Dandan | :) |
| 07:03.49 | DarKnesS_WolF | Dandan: luck ya ! |
| 07:04.12 | Dandan | hey DarKnesS_WolF :) |
| 07:04.22 | DarKnesS_WolF | i have a question now i'm using _01XX.,1,Dial(user:password@domain/${EXTEN}) |
| 07:04.39 | DarKnesS_WolF | can i have something like 011${EXTEN} ?? |
| 07:04.46 | DarKnesS_WolF | Dandan: how is everything there? |
| 07:04.50 | DarKnesS_WolF | Dandan: and ? |
| 07:04.53 | Dandan | there will be a free version of it |
| 07:05.09 | DarKnesS_WolF | Dandan: i meet mark when he was in egypt and i meet schuler and some other guys in gitex from digim |
| 07:05.13 | Dandan | but they will continue selling their product for the forseeable future |
| 07:05.14 | DarKnesS_WolF | Dandan: sweeeeeeeeeeeeeeeeeeeeeeeeeeeeet ! |
| 07:05.47 | DarKnesS_WolF | Dandan: ur orginaly from USA? |
| 07:05.50 | Dandan | they are still rethinking how to integrate it with asterisknow/asterisk for busines |
| 07:05.54 | Dandan | DarKnesS_WolF: indeed |
| 07:06.06 | Dandan | u? |
| 07:06.09 | DarKnesS_WolF | Dandan: egypt ! |
| 07:06.40 | DarKnesS_WolF | actually makr half egyptian and he is from same city alexandra and too close to my home :D |
| 07:06.53 | Dandan | oh sweet :) |
| 07:07.04 | Dandan | meet him today at the party |
| 07:07.15 | Dandan | tomorrow he is going to give his keynote speech |
| 07:07.33 | Dandan | btw. I have a few mp3's of the presentations I was at |
| 07:08.01 | DarKnesS_WolF | Dandan: pleasee upload :-) |
| 07:08.01 | Dandan | if anyne interested |
| 07:08.13 | DarKnesS_WolF | Dandan: if u see him again tell him sherif from alexandria egypt sending his regards :-) |
| 07:08.15 | Dandan | when I get home, I will upload it somewhere... |
| 07:08.24 | DarKnesS_WolF | Dandan: mmm okay ur always here ? |
| 07:08.30 | Dandan | i will be |
| 07:08.40 | DarKnesS_WolF | sweet i can have 011${EXTEN} |
| 07:08.45 | Dandan | I met many ppl from sangoma/digium etc... |
| 07:09.04 | DarKnesS_WolF | i meet david the CEO of sangoma when i was in cebit |
| 07:09.09 | DarKnesS_WolF | last march hannover one |
| 07:09.11 | Dandan | I will probably see him tomorrow I do not think I will have a chance to talk to him... |
| 07:09.24 | awk | hrm, tell me if anyone has experienced this, got a queue, setup, everyone falls to the queue, right.. this morning clients phones, they can see calls coming in, they cant pick them up |
| 07:09.27 | awk | also they cant dial out |
| 07:09.36 | awk | the only thing I can see from the cli when checking the outputs was this |
| 07:09.43 | awk | SIPPeer/SIP/8430-09c s@default-iquad:1 Down (None) |
| 07:09.43 | awk | Parking/Local/8430@d 8430@default-local-d Down (None) |
| 07:09.50 | awk | any idea why the parking would be down? |
| 07:10.12 | Dandan | awk: eeew... what did you do? |
| 07:10.12 | Dandan | :) |
| 07:10.26 | awk | I wish it was something I did :P |
| 07:10.35 | awk | had to restart asterisk for it to kick the queue back in |
| 07:10.36 | Dandan | no idea, after so many beers and colombian vodka, I really can't tell you |
| 07:10.48 | awk | i just wish I was high |
| 07:10.49 | awk | :( |
| 07:10.51 | Dandan | but it is strange indeed |
| 07:10.53 | awk | and today was not happening |
| 07:11.00 | Dandan | i think bugs.digium is your friend |
| 07:11.07 | Dandan | just give them lotsa debug info |
| 07:11.10 | Dandan | if you haved |
| 07:11.13 | Dandan | *have |
| 07:11.15 | awk | I wish I had |
| 07:11.18 | awk | no debug info |
| 07:11.27 | awk | so its a pointless post |
| 07:11.31 | awk | but this client is so fucking touchy |
| 07:11.41 | awk | and the problems have to hapen with him :P |
| 07:11.56 | awk | they a trading company they cant be down for a sec :) |
| 07:12.09 | awk | cant bame him for wanting to kill me |
| 07:12.28 | Dandan | eeew |
| 07:12.32 | *** part/#asterisk DarKnesS_WolF (n=sherif@unaffiliated/sherif) |
| 07:12.33 | Dandan | well, gluck man |
| 07:12.35 | *** join/#asterisk DarKnesS_WolF (n=sherif@unaffiliated/sherif) |
| 07:12.49 | Dandan | c u all later 8) |
| 07:12.58 | awk | later |
| 07:16.53 | awk | OMG! |
| 07:16.57 | awk | asterisk just died on thjis client |
| 07:16.58 | awk | :( |
| 07:18.14 | creativx | nice awk |
| 07:18.19 | creativx | looks like you've got something to do this weekend then |
| 07:18.20 | creativx | =) |
| 07:22.24 | ru_wing|wrk | lol |
| 07:24.13 | *** join/#asterisk Delmar (n=delmar@ip-58-28-158-154.ubs-dsl.xnet.co.nz) |
| 07:27.03 | awk | it looked like it caused asterisk to crash, the parking |
| 07:27.14 | awk | let me send a bug report |
| 07:32.50 | *** join/#asterisk brc_ (n=brc__@pdpc/supporter/basic/brc) |
| 07:33.53 | brc_ | I'm having a very strange issue with 7905's. If the phone is off the hook at dialtone it sends I believe a SIP 486 back immediately, which would be okay except the phones are queue members which means queue calls go to voicemail main. |
| 07:34.03 | brc_ | anybody seen this behaivor? |
| 07:34.13 | *** join/#asterisk Zefk (n=Zefk@wsc-fo.b.astral.ro) |
| 07:44.55 | *** join/#asterisk ussrback (n=MAX@81.95.160.147) |
| 07:44.59 | ussrback | Hi all |
| 07:45.19 | ussrback | whats a difference between Asterisk are 1.2.24 and 1.4.11. |
| 07:45.42 | ussrback | releases? |
| 07:45.42 | ^JimmyRidge^ | u could try the changelogs? |
| 07:46.33 | ^JimmyRidge^ | ussually in the source packages |
| 07:47.09 | ussrback | in ore detail? they both have the same functions ? |
| 07:47.17 | creativx | ussrback: read the changelog |
| 07:47.20 | creativx | lots of deprecated functions |
| 07:47.31 | creativx | read the readme too |
| 07:47.46 | ussrback | XQz me im new in asterisk, and im askin stupid questions |
| 07:48.06 | ussrback | but such things is not noted in READMEs |
| 07:48.33 | ussrback | so first of all i need info whats a difference between this two releases |
| 07:48.34 | creativx | since you claim that i assume you have read the corresponding readmes |
| 07:48.45 | awk | bla bla bla jackidie smackidie |
| 07:48.46 | creativx | what you need is already been told |
| 07:48.51 | creativx | -- > changelog |
| 07:48.57 | awk | ok anyone having issues in here anyone here right now? having issues with 1.4.11 parking? |
| 07:49.03 | awk | i think its very broken |
| 07:49.09 | creativx | i dont use parking awk |
| 07:49.14 | creativx | i made my own ghetto parking |
| 07:49.27 | creativx | strangely it works.. every time =) |
| 07:51.56 | awk | is this possible |
| 07:51.59 | awk | I have a pickup group |
| 07:52.04 | awk | but it only works on extern numbers |
| 07:52.12 | awk | local transfers wont allow pickup? |
| 07:52.19 | awk | i dont want people picking up transfered calls |
| 07:52.22 | awk | ponly exten numbers |
| 07:53.21 | *** join/#asterisk nfi|ermes (n=ErMeS@217.220.121.62) |
| 07:53.44 | brc_ | anybody know what the latest firmware version is for 7905g's? |
| 07:56.33 | hmmhesays | google might know |
| 07:56.45 | hmmhesays | i have a 7940 on my desk and it pisses me off daily |
| 07:57.42 | brc_ | google ain't helping much |
| 07:57.52 | brc_ | my damn cisco login expired since I didn't log in in 6 months apparently |
| 07:58.04 | `Sean | lol |
| 07:58.11 | brc_ | hate cisco...HATE |
| 07:58.19 | `Sean | get revence |
| 07:58.23 | `Sean | *revenge |
| 07:58.27 | ussrback | anybody have good tutor for meetme with mysql? |
| 07:58.30 | brc_ | can you make sense of the naming scheme here? I think it's datebased but I can't figure it out http://www.xs4all.nl/~graver1/cisco/7905/ |
| 07:59.42 | *** join/#asterisk Aurs (n=Aurs@1ult2p8.ip.hipercom.no) |
| 08:00.40 | *** join/#asterisk qdk_ (n=qdk@213.150.62.32) |
| 08:02.06 | hmmhesays | looks like phone model, protocol, version |
| 08:02.53 | *** join/#asterisk darkskiez (n=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 08:05.48 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 08:06.26 | *** join/#asterisk dongs (i=500@l212168.ppp.asahi-net.or.jp) |
| 08:06.37 | dongs | lol. it seems someone actually made a voip to ISDN adapter. |
| 08:06.43 | dongs | http://www.voxtream.com/p104.asp |
| 08:06.49 | dongs | time to get some |
| 08:08.01 | *** join/#asterisk delmar (n=delmar@ip-58-28-158-154.ubs-dsl.xnet.co.nz) |
| 08:09.32 | brc_ | GAAHAAAAAA |
| 08:09.36 | brc_ | I could kill cisco |
| 08:09.57 | brc_ | I can't believe nobody else has run into this problem. it makes callback queues totally useless |
| 08:10.31 | awk | what do yo u mean |
| 08:10.35 | awk | I have issues with queues and parking |
| 08:10.38 | awk | tell me you have the same |
| 08:10.46 | awk | hrm, who had that BLF issue with asterisk 1.2? |
| 08:11.04 | awk | tell me does asterisk paid support work on 1.2 asterisk? |
| 08:11.12 | *** join/#asterisk mihinomenest (i=cE10@66.255.220.17) |
| 08:13.39 | dongs | http://www.alexon.co.jp/products/hds5000/hds5000_spec2.html woohoo, and 2 analog -> ISDN converter |
| 08:13.43 | dongs | good shit. |
| 08:16.50 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 08:17.26 | awk | somebody answer me now |
| 08:17.35 | dongs | you can pay me to answer you |
| 08:17.57 | awk | why dont you just stfu |
| 08:19.25 | awk | hrnm |
| 08:19.36 | awk | does digium do asterisk paid support? |
| 08:19.42 | dongs | yeah. |
| 08:19.48 | awk | for bugs? |
| 08:19.49 | dongs | but lemme tell you something |
| 08:19.54 | dongs | by the time you require "paid support" |
| 08:19.58 | dongs | you're better off looking elsewehre. |
| 08:20.02 | dongs | (than asterisk taht is) |
| 08:20.04 | dongs | you mean fixing bugs? |
| 08:20.25 | dongs | they might look into it. |
| 08:20.34 | dongs | I think I have a open bug in asterisk for > 3 years now |
| 08:20.37 | dongs | still hasnt been fixed |
| 08:20.44 | awk | yes but I want this fixed today |
| 08:20.46 | awk | I wil pay |
| 08:20.57 | awk | I want a parking bug fixed |
| 08:21.02 | awk | its a bug in 1.2 and 1.4 |
| 08:21.05 | awk | 1.4 it crashed asterisk |
| 08:21.16 | dongs | you're in the wrong timezone for "today", unless they've outsourced to india lately |
| 08:21.21 | awk | in 1.2 there is no BLF, so no parking lights, wel the lights are inverted |
| 08:21.29 | awk | what do u mean |
| 08:21.31 | awk | its 10am |
| 08:21.37 | awk | and in america they still sleeping |
| 08:21.44 | dongs | thats what I meant. |
| 08:21.50 | awk | good, so can this be done |
| 08:21.52 | awk | and at what cost |
| 08:22.05 | delmar | welll.....I think todays drama has proven to me that Asterisk behind a router (nat) using SIP for connecting to peers... is a bloody joke |
| 08:22.21 | brc_ | gasp |
| 08:22.22 | dongs | assuming you find a developer familiar with the feature and can produce a test environment it can probably be done today. |
| 08:22.23 | awk | make sure you have externhost and externip set |
| 08:22.35 | awk | also dont use sip behing nat |
| 08:22.35 | dongs | delmar: just dont do it. |
| 08:22.37 | delmar | awk, yup yup did ALL of that |
| 08:22.40 | delmar | dongs, exactly |
| 08:22.44 | awk | iax works like a dream |
| 08:22.48 | awk | delmar set your refresh time too |
| 08:22.58 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 08:22.59 | awk | also set your internal address |
| 08:23.02 | delmar | I dropped an old DSL300 modem on the second interface... and SIP works mint right away now... since the linux box now has the public IP |
| 08:23.29 | awk | heh |
| 08:23.31 | awk | well there we go |
| 08:23.35 | delmar | I dont want to use SIP at all... IAX _was_ working mint |
| 08:23.42 | delmar | but now IAX is screwed |
| 08:23.42 | awk | so why change |
| 08:23.42 | dongs | http://img133.auctions.yahoo.co.jp/users/8/4/3/8/dpgreen071jp-img600x450-1190640882newyahoo-pic100_002.jpg |
| 08:23.46 | dongs | check this awesome shit. |
| 08:23.49 | dongs | 2 analog to 1U isdn interface. |
| 08:23.53 | delmar | awk, cuz IAX broke |
| 08:23.59 | awk | delmar how so? |
| 08:24.03 | awk | what version of asterisk |
| 08:24.25 | dongs | how does iax break |
| 08:24.32 | dongs | i've had iax working since before asterisk was 1.0 |
| 08:24.34 | delmar | awk, outgoing calls via IAX.. no problems.. but every time someone rings in, its all chopped up and really really bad |
| 08:24.38 | delmar | and this wasnt always the case |
| 08:24.49 | delmar | here is the general call path... |
| 08:24.53 | awk | and you think thats due to iax? |
| 08:24.59 | awk | id very much doubtg it |
| 08:25.00 | dongs | heh |
| 08:25.11 | delmar | awk, yes.. les me explain further... |
| 08:25.25 | awk | ok while you explain how can I get to digium paid support |
| 08:25.28 | awk | I need to phone them now |
| 08:25.53 | creativx | how about digium.com for starters |
| 08:26.22 | dongs | http://www.digium.com/en/company/contact.php < |
| 08:26.36 | delmar | Polycom501 sip/g729 ==== Local Asterisk box ==== IAX/g729 ====> via ST608WL(nat router) =====> My Asterisk Colocation server ====SIP/ulaw====> DID provider |
| 08:27.00 | delmar | the problem is the IAX connection between by local asterisk and my Colocation server.. |
| 08:27.08 | delmar | if I change that to SIP/g729 there are no problems |
| 08:27.21 | dongs | where the hell is it 10am |
| 08:27.22 | delmar | if I change it to IAX... incoming calls are great going out, but shit coming in |
| 08:27.22 | dongs | israel? |
| 08:27.35 | delmar | dongs, UK its morning |
| 08:27.40 | awk | south africa |
| 08:27.47 | *** join/#asterisk marexz (n=marexz@marexz.mil.lv) |
| 08:27.48 | awk | and uk is an hour behingus |
| 08:27.53 | awk | or 2 hours is it winter there? |
| 08:27.57 | awk | ye, 2 hours now |
| 08:28.05 | delmar | So .. I have changed codecs and all sorts... my first thing to blame was the g729 so I got rid of it... |
| 08:28.17 | delmar | no matter what I do, incoming calls via IAX are just rubbish. |
| 08:28.34 | delmar | I gotta hangup and call the person back |
| 08:29.12 | dongs | well i dont think its iax |
| 08:29.22 | delmar | dongs, ok. what are your thoughts? |
| 08:29.26 | dongs | routing / you got capped / you got hacked / whatever, but not iax. |
| 08:29.51 | delmar | dongs, yet when I switch to SIP it's ok .. |
| 08:30.13 | delmar | ok well here is where I am at now.... |
| 08:30.26 | delmar | the ST608WL router... got replaced for an old faithful speedtouchpro |
| 08:30.32 | delmar | no change. |
| 08:30.52 | delmar | and no real way to diagnose it with either solution... |
| 08:30.56 | delmar | so lastly... |
| 08:31.10 | delmar | I ripped that out too.. and slapped a DSL300 modem on the second interface on the Linux box... |
| 08:31.21 | delmar | so now the Asterisk box has a public IP directly on the interface... |
| 08:31.27 | delmar | so if nothing else.. SIP now works mint. |
| 08:31.31 | delmar | but... |
| 08:31.35 | delmar | IAX is still.. no better |
| 08:31.45 | delmar | so I can rule out my DSL hardware |
| 08:31.48 | delmar | at least |
| 08:32.38 | dongs | weird. |
| 08:33.24 | *** join/#asterisk und3r (i=iwoop@76.76.103.18) |
| 08:33.37 | delmar | im not much of a guru at sniffing out packets and things but I figure having the Linux box where it is now .. will help |
| 08:34.15 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 08:36.19 | brc_ | I've got a list like 123 firstname lastname macaddress0000, My 7905 provision script takes in 4 cmd line args, I'm trying to write a bash script to iterate over my list of devices and feed it into the provision script but I'm not having any luck |
| 08:36.58 | brc_ | using cat 7905maclist | while read COMMAND; do |
| 08:36.58 | brc_ | <PROTECTED> |
| 08:37.01 | *** join/#asterisk vgster (n=vgster@h14866.navonline.net) |
| 08:52.53 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:40.57 | *** join/#asterisk saftsack (n=saftsack@pD9E06C13.dip.t-dialin.net) |
| 09:43.23 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 09:55.21 | luke-jr | How can you get out of a macro? ☺ |
| 09:56.18 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:08.16 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 10:10.29 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 10:11.26 | *** join/#asterisk ussrback (n=MAX@81.95.160.147) |
| 10:38.55 | *** join/#asterisk appelza (n=d@dsl-240-129-197.telkomadsl.co.za) |
| 10:39.27 | appelza | Lets say extention 100 is busy, but someone calls it..how can I have it fall back to extention 110? |
| 10:43.41 | Strom_M | if busy, goto 110 |
| 10:43.46 | Strom_M | or something along those lines |
| 10:45.34 | *** join/#asterisk nighty^ (n=nighty@72.0.72.123) |
| 10:54.47 | *** join/#asterisk Sinist3r (n=Trotsky@cpe-67-49-8-68.socal.res.rr.com) |
| 10:56.18 | Sinist3r | Anyone know where I can get some info on starting up a CLEC? |
| 10:57.51 | Strom_M | step 1: obtain at least five million dollars |
| 10:58.11 | Strom_M | step 2: perform a paperwork dance with the california public utilities commission |
| 10:58.30 | Strom_M | step 3: perform a paperwork dance with the north american numbering plan administraton |
| 10:58.43 | Strom_M | step 4: buy a DMS-500 |
| 10:58.50 | Strom_M | step 5: ??? |
| 10:58.55 | Strom_M | step 6: bankruptcy |
| 11:01.37 | Sinist3r | umm |
| 11:01.44 | Sinist3r | 5 million? |
| 11:02.05 | Sinist3r | I've heard of people running CLECs with a few computers from their own home. |
| 11:02.11 | Sinist3r | for conferencing services |
| 11:04.38 | *** join/#asterisk ussrback (n=MAX@81.95.160.147) |
| 11:04.47 | Sinist3r | Whats the 5 million for? |
| 11:04.48 | *** join/#asterisk tomodachi (n=matmoj@fw.packetfront.com) |
| 11:05.18 | tomodachi | im wondering if someone more experienced could give me their opinion on building asterisk from source vs using a disg (debian) |
| 11:05.41 | tomodachi | im planning to set up a asterisk pbx (still a noob) but would like to stick do the vesions in my distributions |
| 11:05.49 | tomodachi | to get the updates /security fixes automatically |
| 11:05.50 | ussrback | is it possible to call away someone in private room, with meetme? |
| 11:06.11 | ussrback | for example if in conference room there are 6 persons |
| 11:06.33 | ussrback | and person 1 wants to call way person 2 in provate room |
| 11:06.39 | ussrback | to talk tet-a -tet |
| 11:06.48 | ussrback | is it possible? |
| 11:07.12 | rob0 | tomodachi: not sure I qualify as "more experienced", but I have an opinion. I use Slackware, which does not package Asterisk. But I build it with all defaults (no special configuration arguments), and all is well. So ... |
| 11:07.38 | rob0 | ... I don't see how a distro-built package would be so bad, if likewise, it uses the defaults. |
| 11:08.14 | rob0 | Upgrades via automated processes are very often a cause of disasters. |
| 11:08.56 | tomodachi | when using a stable distribution though |
| 11:08.59 | rob0 | I hang out in #postfix, and we see on average several Debian people every who have borked things with an apt-get upgrade. |
| 11:09.01 | tomodachi | its usually quite tested |
| 11:09.31 | rob0 | Okay, the next victim, I'll ask if they're using stable. |
| 11:10.21 | rob0 | What YOU need to do is to look inside your .deb and find out how it was built. |
| 11:10.22 | Sinist3r | Can I get some practical CLEC info from someone? |
| 11:11.05 | rob0 | Sinist3r: I think Strom_M's practical advice to you was "forget it". |
| 11:11.26 | Sinist3r | Doesn't sound practical at all. |
| 11:13.03 | Sinist3r | I hate when people discourage you. |
| 11:13.24 | rob0 | Not everything should/can be encouraged. |
| 11:13.30 | Sinist3r | It's almost as if they fail so they want everyone else to fail too. |
| 11:13.50 | ussrback | is it possible to call away someone in private room, with meetme? |
| 11:14.11 | Sinist3r | you can use sub confs |
| 11:14.28 | Sinist3r | but you need 5 million dollars. |
| 11:29.04 | juuva | anyone using h323? |
| 11:29.26 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:32.13 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 11:39.31 | *** join/#asterisk gremzoid (n=gremzoid@d58-111-173-16.rdl5.qld.optusnet.com.au) |
| 11:42.57 | *** join/#asterisk khronos (n=khronos@c-66-229-159-201.hsd1.fl.comcast.net) |
| 11:45.33 | *** join/#asterisk gardo (n=gardo@121.97.249.68) |
| 11:51.46 | awk | ~pb |
| 11:51.47 | jbot | A Pastebin is a place to paste your stuff 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 |
| 11:51.47 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 11:52.52 | awk | http://paste.debian.net/38257 |
| 11:53.03 | awk | anyone have any idea why im getting that warning, about needing a function |
| 11:55.59 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 11:57.49 | creativx | awk: the warning is that the function is in need of an argument |
| 11:58.03 | creativx | set "group(myvar)=" |
| 11:58.10 | creativx | you cant set it to nothing it seems |
| 12:00.46 | *** join/#asterisk coppice (n=chatzill@16.196.17.210.dyn.pacific.net.hk) |
| 12:03.00 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:04.22 | awk | creativx tell me hjas anyone experienced this before |
| 12:06.18 | awk | http://paste.debian.net/38260 |
| 12:06.19 | creativx | I can only answer on behalf of myself, and I dont use that function. |
| 12:06.23 | awk | this happens all the time |
| 12:06.30 | awk | not that, this :) |
| 12:06.35 | creativx | its called the internet |
| 12:06.43 | awk | thats a lan |
| 12:06.49 | creativx | then your equipment is bad |
| 12:06.49 | awk | it happens every few minutes |
| 12:06.50 | creativx | :) |
| 12:07.01 | awk | 60k cisco switches |
| 12:07.03 | awk | I doubt that |
| 12:07.09 | awk | everything is top of the range |
| 12:07.10 | *** join/#asterisk michael-i (n=michael-@141.41.40.55) |
| 12:07.16 | awk | I have a feeling its a duplex problem |
| 12:07.18 | creativx | everything except your latency |
| 12:07.18 | creativx | =) |
| 12:07.19 | awk | set to auto duplex |
| 12:07.29 | awk | think why creativx |
| 12:07.37 | awk | calls are getting cut up too |
| 12:07.40 | awk | once it reaches 2000ms |
| 12:07.44 | awk | it drops back to 15ms and its fine |
| 12:08.16 | awk | I have a feeling the phones are set to auto detect for what dduplex to use |
| 12:08.19 | awk | and I feel the switches are too |
| 12:08.28 | awk | so 1 wayaudio is a case of using half duplex |
| 12:08.54 | awk | so im wondering if its not dropping to half duplex for that second when it requires a pong with the ping sent, then switches back too full duplex |
| 12:09.08 | awk | no vlan, we running above the vlan |
| 12:09.09 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:09.17 | awk | so it can only be a switch issue in my opiniun |
| 12:09.27 | awk | but somebody must have experienced this problem before |
| 12:09.36 | awk | its seen it at another site ive worked at too |
| 12:09.44 | awk | they using hp switches there.. |
| 12:09.53 | creativx | welll |
| 12:09.58 | creativx | only one way to find out |
| 12:09.58 | creativx | test it |
| 12:10.04 | creativx | or force duplex mode on the port |
| 12:10.12 | awk | wsell I have set 10 extensions to full duplex |
| 12:10.22 | awk | and out of those 10 7 have not had that time out problem |
| 12:10.27 | awk | normally it happens every few minutes |
| 12:10.30 | awk | ive run the test for 2 days now |
| 12:10.35 | awk | but what about those other 3 phones? |
| 12:10.59 | awk | thats why im boiling it down to the duplex, but cant be 100% sure its that as why didn't those phones sort out? |
| 12:11.03 | creativx | well |
| 12:11.10 | creativx | i have my sturdy 3com switch which treats me well |
| 12:11.14 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 12:11.16 | lirakis | morning |
| 12:11.24 | awk | creativx unmanaged? |
| 12:11.47 | awk | creativx hy this is an issue, they saying its our problem |
| 12:11.51 | awk | we have swopped the server out 3 times |
| 12:12.01 | awk | its not the server.. i take that box back to our test env and it works |
| 12:12.10 | awk | but to try and prove to them its their network config is difficult |
| 12:12.13 | awk | so I told them to use full duplex |
| 12:12.20 | awk | thing is they said they did for those 10 extensions |
| 12:12.27 | awk | but I have a feeling they trying to trick us on the 3 that isn't working |
| 12:12.32 | awk | and we cant then say its their end |
| 12:12.36 | awk | they ow allot f cash |
| 12:12.41 | awk | so maybe they doing this on purpose |
| 12:13.51 | creativx | awk: its a superstack |
| 12:14.06 | creativx | and you have no other network problems? |
| 12:14.34 | awk | no, their network works fine |
| 12:14.52 | awk | even if they had issues on their tcp stack, it retransmits |
| 12:14.53 | awk | not like udp |
| 12:15.02 | awk | so that second its off we have break up calls |
| 12:15.22 | *** join/#asterisk dijungal (n=kdaniel@63.175.159.171) |
| 12:15.31 | awk | I still dont understand why they set it just for those few extensions, why not just turn the whole switch to full duplex |
| 12:15.37 | awk | its not like we have any 10base devices in there |
| 12:15.41 | dijungal | does anyone know of any IAX voip provider i can terminate my calls to? |
| 12:16.31 | awk | dijungal what sort of volume |
| 12:17.43 | *** join/#asterisk brian (n=brian@unaffiliated/brian) |
| 12:17.47 | dijungal | little |
| 12:17.52 | dijungal | just for testing for now... |
| 12:18.05 | dijungal | lol.... awk ... u should try gawk :) |
| 12:18.30 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 12:18.51 | *** join/#asterisk ifrank (n=ifrank@c-24-3-172-199.hsd1.mn.comcast.net) |
| 12:22.12 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 12:22.37 | dijungal | i guess they aren't any |
| 12:22.58 | *** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
| 12:23.02 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:23.06 | creativx | wb [TK]D-Fender |
| 12:24.17 | awk | dijungal on massive volume I can help you |
| 12:24.29 | Sinist3r | Anyone know where I can get some info on starting up a CLEC? |
| 12:25.47 | [TK]D-Fender | Sinist3r: http://www.fcc.gov |
| 12:26.19 | dijungal | awk: ok tell me about what you would recommend for massive volume |
| 12:26.25 | Sinist3r | FCC has guides on how to setup CLECs? |
| 12:26.27 | coppice | ULECs seems more popular |
| 12:26.36 | *** join/#asterisk minkus (n=minkus@pool-72-84-53-31.clrkwv.east.verizon.net) |
| 12:26.37 | Sinist3r | whats a ULEC? |
| 12:26.46 | coppice | uncompetitive |
| 12:27.01 | Sinist3r | how would that work? |
| 12:27.11 | [TK]D-Fender | Unbundled Local Exchange Carrier (FCC) |
| 12:28.10 | [TK]D-Fender | Sinist3r: Go read up about what you're dealing with and hearing you just spout out questions like that gives me a solid impression you have no clue what you're doing.... |
| 12:28.50 | coppice | isn't having no clue what you're doing is a prerequisite for starting a CLEC? |
| 12:28.54 | Sinist3r | TK, I have somewhat of an understanding but I wish to know more before I actually do it, that's why I was asking for a resource. |
| 12:29.08 | Sinist3r | thank you coppice |
| 12:29.32 | creativx | heheh |
| 12:29.42 | creativx | Sinist3r, it never occured to you to google for it? |
| 12:29.52 | Sinist3r | creativx, I have been for hours. |
| 12:30.01 | coppice | well, if you swim with the great whites, you might expect to be eaten |
| 12:30.06 | Sinist3r | I found one thing, and it wasn't very detailed. |
| 12:30.53 | *** join/#asterisk saftsack (n=saftsack@pD9E06C13.dip.t-dialin.net) |
| 12:30.54 | [TK]D-Fender | Sinist3r: http://www.google.ca/search?hl=en&q=ULEC&btnG=Google+Search&meta= <----- #6 I had to look VERY far apparently |
| 12:32.13 | Sinist3r | TK, again, I'm not looking for what the acronym means, I was looking for a "guide" on how to start one up. |
| 12:32.34 | Sinist3r | let's see how far you have to look to find that. |
| 12:32.45 | [TK]D-Fender | coppice: Ever seen one breach while taking out a seal? Scary shit.... |
| 12:32.56 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 12:33.17 | [TK]D-Fender | Sinist3r>whats a ULEC? <--- funny sure LOOKS like you were asking. Perhaps we're not both actually speaking english... |
| 12:33.46 | Sinist3r | TK, <Sinist3r> Anyone know where I can get some info on starting up a CLEC? |
| 12:33.50 | Sinist3r | that was my main question. |
| 12:33.53 | coppice | I think anything to do with a great white looks like scary shit |
| 12:33.56 | Sinist3r | the other was a side note. |
| 12:34.10 | Sinist3r | are you that stupid or are you just trying to act all big and bad? |
| 12:34.15 | awk | Sep 28 14:32:30 ERROR[19808]: chan_sip.c:11691 sipsock_read: SIP MESSAGE JUST IGNORED: BYE |
| 12:34.15 | awk | Sep 28 14:32:30 ERROR[19808]: chan_sip.c:11692 sipsock_read: BAD! BAD! BAD! |
| 12:34.21 | awk | what could be the reason for this? |
| 12:35.47 | [TK]D-Fender | Sinist3r: Well I linked you to the answer for that one too, FIRST in fact. |
| 12:36.32 | Sinist3r | and I was just asking politely did the fcc have a setup guide or something? |
| 12:37.11 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 12:37.36 | [TK]D-Fender | Sinist3r: Dunno... how long did you look? |
| 12:37.55 | Sinist3r | you mean on the fcc site or on google? |
| 12:38.27 | Sinist3r | on google for hours |
| 12:38.36 | Sinist3r | fcc I just got on it after you told me |
| 12:43.25 | *** join/#asterisk blinky42 (n=me@67.106.175.130.ptr.us.xo.net) |
| 12:46.33 | muh-die-kuh | can anyone of you recomend me an wireless voip phone, which can sync its phonebook with a server? if you know many of this kind, just tell me the cheapest ;) |
| 12:48.17 | [TK]D-Fender | ~wifisip |
| 12:48.18 | jbot | Wi-Fi SIP phones suck. All of them. HARD. Some only slightly less than others... |
| 12:48.27 | _x86_ | [TK]D-Fender: do you ever sleep? ;) |
| 12:48.36 | [TK]D-Fender | _x86_: Possibly ;) |
| 12:49.02 | fenlander | muh-die-kuh: nokia N95 :) |
| 12:49.22 | muh-die-kuh | [TK]D-Fender: dect would be okay, too :P |
| 12:49.42 | muh-die-kuh | wait. |
| 12:49.45 | muh-die-kuh | i said nothing |
| 12:49.47 | [TK]D-Fender | muh-die-kuh: take a look at Seimens then. They seem rather well regarded in that category |
| 12:50.07 | _x86_ | what is dect? |
| 12:50.24 | muh-die-kuh | [TK]D-Fender: but i guess i wont be able to sync my phone book with dect |
| 12:50.31 | [TK]D-Fender | _x86_: Digital Enhanced Cordless Telecommunications - Wikipedia, the free ... |
| 12:50.54 | _x86_ | ass ;) |
| 12:50.55 | [TK]D-Fender | Seriously people! Get a clue and do the freebie Google searches YOURSELVES! |
| 12:51.23 | _x86_ | its much more fun having others do it for you though ;) |
| 12:51.37 | [TK]D-Fender | muh-die-kuh: its possible, but you may be asking in the "would you l;ike fries with that, sir?" category... |
| 12:53.16 | *** join/#asterisk Corydon76-dig (i=pink@pdpc/supporter/sustaining/Corydon76-home) |
| 12:53.16 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 12:56.19 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 12:56.42 | *** join/#asterisk ming_zym (n=ming_zym@124.254.57.194) |
| 12:58.08 | *** join/#asterisk champster (n=asterisk@AH.tescogroup.com) |
| 12:58.17 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:02.07 | *** join/#asterisk Goldfisch (n=gregturn@user-0c6t46t.cable.mindspring.com) |
| 13:05.20 | *** join/#asterisk famicon (i=redz@c51447ddc.cable.wanadoo.nl) |
| 13:07.28 | lirakis | [TK]D-Fender: lol |
| 13:08.45 | *** join/#asterisk adeel (n=adeel@c-24-7-132-155.hsd1.ca.comcast.net) |
| 13:09.10 | [TK]D-Fender | muh-die-kuh: And if you call in the next 30 minutes we'll even through in a lovely free gift from RonCo! |
| 13:09.19 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.186.199) |
| 13:09.35 | adeel | is there a way to 'revive' an apparent bricked polycom 330? |
| 13:11.13 | coppice | And if you call in the next 30 minutes you'll find the entire call centre is watching cricket |
| 13:11.33 | adeel | probably watching the pakistan v india match |
| 13:12.04 | lirakis | adeel: ... probably not unless you are an electrical engineer |
| 13:12.36 | lirakis | adeel: but that depends what 'apparent bricked' means... |
| 13:12.44 | ThoMe | hallo? |
| 13:12.45 | ThoMe | wer da? |
| 13:12.49 | dijungal | awk?? |
| 13:12.59 | dijungal | awk: never got that answer |
| 13:13.29 | adeel | lirakis, well it wasn't really turning on...but now it's turning on, but not moving pass the 'updating initial configuration' screen...nor does it ever find my dhcp server...i'm thinking there's a loose connection in the internal switch |
| 13:13.51 | creativx | give it the well known fist of fury |
| 13:14.01 | adeel | lirakis, is it possible to default the phone settings upon boot? |
| 13:14.18 | Katty | herro |
| 13:16.12 | lirakis | adeel: sure... search the internet... ive found it before |
| 13:16.27 | dijungal | any suggested iax providers? |
| 13:17.00 | adeel | lirakis, i came across a few different methods, but they all require actually being able to get to the menu's and all...it might just be easier to RMA this phone |
| 13:17.29 | lirakis | adeel: .. ive seen several methods that allow you to do it before the phone boots up |
| 13:17.51 | adeel | lirakis, interesting, i'll search some more |
| 13:19.49 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 13:27.14 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 13:31.31 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 13:32.39 | adeel | so the phone's switch is broken...yay |
| 13:32.57 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 13:33.05 | *** join/#asterisk codejunky (n=jan@codejunky.org) |
| 13:37.22 | [TK]D-Fender | adeel: Tried setting a fixed IP and browse to it? |
| 13:37.25 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 13:37.33 | flujan | hi all... |
| 13:37.54 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:37.54 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:38.04 | flujan | Hi [TK]D-Fender ... I implement the socket to listen for events on the AMI and its is working well. Thanks for that tip. :D |
| 13:38.13 | [TK]D-Fender | flujan: np |
| 13:39.06 | *** join/#asterisk Aeudian (n=Aeudian@74.92.134.190) |
| 13:39.39 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 13:41.32 | flujan | Guys, prior to version 1.2 of asterisk I found two configuration options... incoming call limit and outgoing call-limit... 1.4 just have the call-limit option. is it possible to control the incoming and outgoing call-limits in 1.4? |
| 13:42.48 | flujan | The problems is that I have two levels of technical support... the first level answers a the end users. Sometimes the first level needs to contact the second level to acquire some information... |
| 13:43.17 | creativx | flujan: it got another name |
| 13:43.39 | creativx | there was this forever going bug tracker about the BLF problem and device states with call-limit |
| 13:43.43 | creativx | that spurred some new config options |
| 13:43.49 | creativx | that i cannot for the life of me remember right now. |
| 13:44.12 | flujan | Since the first level answers a call queue, i set up the call-limit to 1. This way, asterisk do not tries to deliver incoming calls from the queue to agents that already are on A call. |
| 13:45.33 | flujan | but with this call-limit I lost the option to make then receive one call and make other call to acquire information... To solve this issue, or a call_queue should have a limit per agent. ( if the agent is on a call, does not include it on the deliver algorithm) or I need to set up incoming and outgoing call-limits... |
| 13:45.47 | flujan | what do you suggest to solve my issue? |
| 13:46.10 | creativx | find the 1.4 setting that limits inbound and outbound separately :-) |
| 13:46.37 | creativx | busy-limit |
| 13:47.14 | creativx | flujan: http://bugs.digium.com/view.php?id=7433 more info there |
| 13:52.12 | flujan | creativx: appear that this bug is closed... they have solved the issue... I am currently running 1.4.11 |
| 13:55.36 | *** join/#asterisk elriah (n=null@adsl-070-148-053-190.sip.bhm.bellsouth.net) |
| 13:55.40 | *** join/#asterisk cnet2 (n=nada@196.40.38.130) |
| 13:56.25 | cnet2 | hi, does anyone have had asterisk stoping service for no aparent reason something like once everyday..? |
| 13:56.29 | elriah | [TK]D-Fender: TK, did you have to change anything with your .cfg file ntp settings when going to the latest Polycom firmware? |
| 13:58.51 | [TK]D-Fender | elriah: nope |
| 13:59.13 | elriah | [TK]D-Fender: Thanks. Weird issue with a handful of phones not pulling the time from pool.ntp.org. |
| 13:59.53 | Katty | :> |
| 13:59.54 | [TK]D-Fender | ;) |
| 14:00.06 | Katty | sorry, i think outside the box. |
| 14:00.07 | [TK]D-Fender | Katty: Mew. |
| 14:00.08 | Katty | that just won't work. |
| 14:00.29 | *** join/#asterisk voipnet-tech (n=voipnet-@216.195.128.62) |
| 14:01.57 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 14:03.09 | *** join/#asterisk minkus (n=minkus@static-141-153-94-2.clrk.east.verizon.net) |
| 14:03.26 | voipnet-tech | hi all, I'm having a problem with No Audio from my freepbx setup, it's a new install (my first time setting up asterisk/, zaptel and freepbx (didn't use a trixbox ISO) extension to extension calling/audio works fine. When I call anything that plays a system recording like VM , IVR, *43 echotest etc, The system appears to play the recording, but I get no audio |
| 14:03.37 | voipnet-tech | I also have no audio doing conferencing/paging |
| 14:03.58 | voipnet-tech | i've asked in freepbx, but I don't think this is a freepbx problem, sounds like a problem with zaptel maybe? |
| 14:04.03 | Katty | i also have no audio from my laptop |
| 14:04.07 | Katty | but it's a driver issue :< |
| 14:05.19 | creativx | god damn |
| 14:05.23 | Katty | creativx: god's too busy to do your damning. |
| 14:05.28 | Katty | creativx: do your own damning :P |
| 14:05.29 | [TK]D-Fender | voipnet-tech: Sorry, but * works, you configuration and networking does not. |
| 14:06.03 | creativx | Katty: just lost 30 minutes of a changelog summary |
| 14:06.10 | creativx | nice way to start ze weekend |
| 14:07.07 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 14:07.39 | Katty | creativx: :<<< |
| 14:07.41 | *** part/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 14:08.09 | *** join/#asterisk Aeudian (n=Aeudian@74.92.134.190) |
| 14:09.02 | *** part/#asterisk Aeudian (n=Aeudian@74.92.134.190) |
| 14:09.08 | *** join/#asterisk Aeudian (n=Aeudian@74.92.134.190) |
| 14:09.36 | *** join/#asterisk tsurko (n=tsurko@213.91.216.130) |
| 14:09.50 | creativx | thanks |
| 14:10.03 | creativx | i will manage to fix it by stealing back my notes from the ceo |
| 14:10.08 | creativx | and be out of here by 16:30 |
| 14:10.11 | creativx | and start drinking beer! |
| 14:10.13 | Aeudian | I have several clients complaining about the volume output on the default music on hold by asterisk over VoIP. Is there a way to quiet down the default moh audio from within asterisk without having to use Audacity or some other audio editing program |
| 14:11.45 | Katty | alchamahols!!! :> |
| 14:11.49 | Katty | oh man |
| 14:11.53 | Katty | i want margaritas over lunch |
| 14:12.12 | creativx | Katty: oh yes |
| 14:12.16 | creativx | and the best part comes afterwards |
| 14:12.21 | Katty | oh? |
| 14:12.21 | creativx | we're going out for dinner and more drinks |
| 14:12.23 | Katty | billiards? |
| 14:12.25 | Katty | oh :> |
| 14:12.28 | creativx | paid by... the employer |
| 14:12.30 | Katty | YAY more drinks!! |
| 14:12.33 | Katty | oooooooooh |
| 14:12.35 | Katty | free drinks |
| 14:12.36 | creativx | freebie |
| 14:12.53 | Katty | we have an open bar at our company christmas party. |
| 14:13.04 | Katty | always much fun :> |
| 14:13.22 | creativx | hehe indeed |
| 14:13.31 | creativx | the company mastercard tends to be red glowing |
| 14:13.39 | Katty | hehe |
| 14:13.54 | Katty | my company is the owner's hobby |
| 14:14.02 | Katty | he's got plenty of monies (= |
| 14:14.58 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:15.13 | [TK]D-Fender | Aeudian: You referring to the FPM samples? |
| 14:17.47 | flujan | someone here already used the URL option from the dial command? I am trying this: |
| 14:18.30 | flujan | exten => _XXXXX,1,Dial(SIP/${EXTEN},20,tT,www.voip-info.org) |
| 14:18.33 | flujan | without success.. :( |
| 14:20.31 | [TK]D-Fender | flujan: On what phone? |
| 14:20.44 | flujan | X-lite/Eyebeam... |
| 14:21.01 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:21.10 | flujan | [TK]D-Fender: I know you hate softphones... But I have a boss that loves it... :D |
| 14:21.24 | [TK]D-Fender | flujan: Only works on Ciscos...... |
| 14:21.41 | flujan | [TK]D-Fender: :( |
| 14:22.14 | Katty | creativx: you ever scooped most of a pineapple out, and filled it with vodka, and then freezed it? |
| 14:22.35 | Katty | creativx: well, you don't really have to scoop most of it out |
| 14:23.35 | *** join/#asterisk mindCrime__ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 14:23.36 | coppice | you do if you want to fill it with fried rice |
| 14:23.45 | Katty | oooooh |
| 14:23.48 | Katty | i never thought about doing that |
| 14:23.53 | Katty | what an awesome centerpiece!! |
| 14:24.03 | coppice | its OK. half the world did |
| 14:24.06 | Katty | coppice: do you have a fried rice recipe? |
| 14:24.28 | Katty | coppice: i don't actually buy whole pineapples often :P |
| 14:26.23 | *** join/#asterisk mindCrime__ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 14:26.42 | creativx | Katty: no |
| 14:26.45 | creativx | Katty: im more of a beer person |
| 14:27.00 | creativx | but i do like super chilled vodka |
| 14:27.02 | creativx | vikingfjord is nice |
| 14:30.03 | Katty | i still don't like beer. |
| 14:30.12 | creativx | im out :) have a nice weekend |
| 14:30.16 | Katty | buhbye |
| 14:30.17 | Katty | enjoy! |
| 14:31.01 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 14:31.26 | Katty | wow, bull mastiffs are /huge/ |
| 14:31.55 | syzygyBSD | how can you not like beer? that is like saying I don't like air |
| 14:32.25 | *** join/#asterisk brut- (n=brut@66.173.4.254) |
| 14:32.31 | syzygyBSD | it is a staple of life |
| 14:33.08 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 14:33.13 | Katty | syzygyBSD: i dunno, i just like girly drinks and malt stuff |
| 14:34.01 | syzygyBSD | I have to admit I did have a couple gin n tonics last night too |
| 14:34.32 | Katty | gin :< |
| 14:35.01 | syzygyBSD | you don't like gin either? |
| 14:35.05 | *** join/#asterisk |dennis| (n=dennis@200.32.236.18) |
| 14:35.50 | syzygyBSD | grr, firefox is taking up 500MB of ram.. gotta restart |
| 14:39.05 | elriah | mmmmm... gin |
| 14:43.45 | Katty | oh boy! |
| 14:43.57 | Katty | i've not had home-made anything in awhile :/ |
| 14:44.12 | Daejeo1 | anyone tried the service from clickdigits.com? |
| 14:44.29 | rob0 | I think itsh as good as Bailey'sh, whaddaya think? |
| 14:44.36 | `Sauron | moin moin |
| 14:44.39 | Daejeo1 | 9.99$ is it good provider? |
| 14:44.58 | `Sauron | home made generally > bailey's |
| 14:46.10 | Katty | herro `Sauron (= |
| 14:46.19 | Katty | bailey's is good with milk |
| 14:46.23 | Katty | can't drink it straight. |
| 14:46.33 | Katty | tastes too much like coffee to me. |
| 14:50.38 | *** join/#asterisk Voicemeup (n=VoiceMeU@modemcable132.108-83-70.mc.videotron.ca) |
| 14:51.31 | Voicemeup | is the reason the MOH plays on box 1 , in phone -> AS 1 -> AS 2 -> out scenario, because of canreinvites ? |
| 14:51.38 | Voicemeup | i mean it playes on box 2 |
| 14:53.47 | *** join/#asterisk coppice (n=chatzill@16.196.17.210.dyn.pacific.net.hk) |
| 14:54.51 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 14:55.45 | karleeto | has anyone used any of the cordless handset IP phones?? i've been looking around at some of them, this aastra 480i looks pretty nice, but i was hoping to get someone's opinion or reccomendations |
| 14:55.57 | *** join/#asterisk Goldfisch (n=gregturn@user-0c6t46t.cable.mindspring.com) |
| 14:55.58 | [TK]D-Fender | Voicemeup: So Phone puts "Out' on hold and AS2's MoH plays? |
| 14:56.11 | karleeto | i'm not going to use a wifi ip phone, so i'm looking for a cordless phone with an ip base station, i guess |
| 14:56.22 | elriah | The 480i works great, stay away from wifi sip phones if you're using nat, most wireless access points drop the nat state and after a few minutes call's won't go through to your wifi phones. |
| 14:56.40 | elriah | Or re-register your wifi phones every 30 seconds. |
| 14:56.45 | [TK]D-Fender | karleeto: Aastra's are ok if you're planning on using only a single registration with them |
| 14:56.54 | karleeto | elriah: yeah, wasnt planning on using wifi phones |
| 14:57.24 | voipnet-tech | I've got a ton of Aastra Phones all doing multiple registrations |
| 14:57.48 | voipnet-tech | even a 480iCT registered on 5 different SIP accounts for 4 handsets |
| 14:57.48 | karleeto | [TK]D-Fender: so you mean, having multiple handsets is not a good idea? it says it can support up to 3 additional handsets, does that mean they all would run off of one extension? |
| 14:57.56 | voipnet-tech | the Aastra SIP DECT system is nice too, but pricy |
| 14:58.00 | Voicemeup | yes |
| 14:58.01 | Voicemeup | TK |
| 14:58.23 | karleeto | voipnet-tech: hmm, so they work well for you? |
| 14:58.26 | Voicemeup | sorry was denying over 5000 in payments some hacker trying to put lol |
| 14:58.35 | voipnet-tech | Wifi phones work pretty good too if you have them on a good Wifi Access Point, and don't run any DATA on the wifi, just voice its OK |
| 14:58.50 | voipnet-tech | karleeto, Aastra phones work great for us |
| 14:58.58 | karleeto | voipnet-tech: wonderful |
| 14:58.59 | voipnet-tech | but we use them on broadsoft, not asterisk |
| 14:59.06 | [TK]D-Fender | karleeto: Sorta... the BASE will ring for all of them gauranteed |
| 14:59.19 | karleeto | [TK]D-Fender: hmm |
| 14:59.33 | karleeto | voipnet-tech: you've never tried them with asterisk? |
| 15:00.11 | voipnet-tech | karleeto, yes, but not in the same quantity, I've got about 8 aastra phones on asterisk, and 2500 on broadsoft |
| 15:00.43 | karleeto | voipnet-tech: ok, i'm only doing a 15 phone system, and only 3 or 4 would need to be cordless |
| 15:01.01 | voipnet-tech | erm I have tried them* My aastra 57iCT on my desk now has 4 registrations to Broadsoft, 4 registrations to 1 asterisk server, and 1 registration to a 2nd asterisk server |
| 15:01.01 | karleeto | voipnet-tech: how is the range and quality with the aastra phones? |
| 15:01.40 | voipnet-tech | I'd recommend you buy 3 or 4 480iCT's, and the rest 480i, or even 9133i |
| 15:02.15 | voipnet-tech | the range is pretty good, I don't know the specs, but I can go anywhere in our 7000 sq ft building with no problem, and works fine outside around the building |
| 15:02.37 | voipnet-tech | talking back to the base on my desk in the basement |
| 15:02.57 | [TK]D-Fender | Aastra's only real selling point is the DECT handsets. As phones themselves I would never choose them to deploy for anything else. |
| 15:03.12 | [TK]D-Fender | Polycom > All |
| 15:03.27 | [TK]D-Fender | I **LOATHED** my Aastra 57i CT <- |
| 15:03.32 | voipnet-tech | I respectfully disgree though :-p |
| 15:03.38 | [TK]D-Fender | I'd have preferred my home bedside IP 310 to it. |
| 15:04.25 | tzanger | coppice: sir |
| 15:04.28 | [TK]D-Fender | Second rate audio quality, finding the DECT I gave to my warehouse manager WASN'T independent of the base and his calls rang on MINE pissed me off. 5i's Rubbery-ass buttons too |
| 15:04.36 | [TK]D-Fender | IP 301 * |
| 15:04.53 | tzanger | coppice: http://www.mixdown.ca/~andrew/dump/x.wav - 6s clip of a congestion tone - can you help me identify the *type* of distortion you hear? |
| 15:05.04 | karleeto | [TK]D-Fender: yeah, we've ordered 10 polycom 501s, and i guess the other 4 people that want the cordlesses will get the aastra 480iCTs |
| 15:05.06 | [TK]D-Fender | And the 5i's have a new pixel based display but still using char-matrix firmware is BS. |
| 15:05.07 | Voicemeup | TK know anything about that ? |
| 15:05.09 | _x86_ | so would yall recommend an IP330 or an IP501? |
| 15:05.09 | tzanger | I describe it as a periodic "metallic" or "flanger" type of noise |
| 15:05.15 | tzanger | it's most certainly not a frame slip |
| 15:05.24 | karleeto | _x86_: i LOVE my 501s |
| 15:05.28 | _x86_ | I've only used the 301, 501, and 601's, never messed with the 320/330 or 430 phones |
| 15:05.31 | _x86_ | karleeto: me too |
| 15:05.32 | [TK]D-Fender | karleeto: IP 501's are very nice, but very hard to find them to be the right choice... |
| 15:05.58 | karleeto | [TK]D-Fender: we use them almost exclusively |
| 15:06.01 | Voicemeup | MOH plays on box 2 when phone A puts someone in hold from a call to a cell phone that passing trough OUT , in ( phone A -> BOX1 -> BOX 2 -> OUT ) |
| 15:06.01 | tzanger | [TK]D-Fender: a clear case of want vs need? |
| 15:06.03 | [TK]D-Fender | karleeto: as a mid-range without native PoE its hard to justify |
| 15:06.04 | voipnet-tech | [TK]D-Fender, you could have made his calls not ring on your phone, just set his lineX ringtone: none and it would not have rang on the base |
| 15:06.29 | karleeto | [TK]D-Fender: well, we cant afforce PoE switches, so thats not an issue with us, lol |
| 15:06.44 | [TK]D-Fender | voipnet-tech: That might be tone, but I could accidently hijack his call as it rings silently, no? |
| 15:06.45 | voipnet-tech | i do admit we have about 10x more weird problems with the 5i series phones vs the 9133i and 480is' |
| 15:07.08 | karleeto | voipnet-tech: are you on any IM services? |
| 15:07.18 | _x86_ | [TK]D-Fender: I need a Polycom phone with full duplex speakerphone -- the IP301 is out... what would you suggest? |
| 15:07.23 | Katty | someone set linksys on fire for treating me like a moron |
| 15:07.25 | [TK]D-Fender | karleeto: IP 320/330 w/ power brick is much cheaper, and the IP 430 (only place of value) fits right below it supporting both... |
| 15:07.28 | _x86_ | I've been just buying IP501's |
| 15:07.48 | *** join/#asterisk Qwell_ (n=north@pdpc/sponsor/digium/Qwell) |
| 15:07.48 | *** mode/#asterisk [+o Qwell_] by ChanServ |
| 15:07.48 | _x86_ | 330 has full-duplex speakerphone? |
| 15:07.49 | karleeto | _x86_: me too.. |
| 15:07.52 | coppice | tzanger: you have some DC offset, and substantial 100Hz noise on a 425Hz + 550Hz signal |
| 15:07.54 | [TK]D-Fender | karleeto: But the only place for IP 501 is in non-PoE Environments |
| 15:07.59 | [TK]D-Fender | _x86_: Yes. |
| 15:08.06 | [TK]D-Fender | _x86_: only the 301 has 1-way |
| 15:08.07 | Voicemeup | let me know anyhow |
| 15:08.07 | voipnet-tech | [TK]D-Fender, you can always hijack from the base as it's the master, but if you just use Line 5,6,7,8, or 9 for the cordless, you won't see it blink your Line keys either |
| 15:08.22 | voipnet-tech | karleeto, MSN |
| 15:08.23 | tzanger | coppice: what did you use to get that result so fast? |
| 15:08.28 | _x86_ | [TK]D-Fender: should i buy a 330 or 501? |
| 15:08.33 | coppice | by ear, of course |
| 15:08.35 | [TK]D-Fender | voipnet-tech: True but if its ringing and I want to place a call I'll accedentally steal his... |
| 15:08.44 | [TK]D-Fender | _x86_: You need the pass-through? |
| 15:08.52 | _x86_ | [TK]D-Fender: for a PC? yes |
| 15:08.55 | karleeto | voipnet-tech: may i get your associated email? i'd like to contact you when my aastras get in |
| 15:08.59 | [TK]D-Fender | _x86_: Got PoE? |
| 15:09.08 | _x86_ | [TK]D-Fender: not yet, but planning on it someday |
| 15:09.15 | voipnet-tech | rpurinton@voipnettechnologies.com |
| 15:09.24 | [TK]D-Fender | _x86_: Any ETA on "someday"? |
| 15:09.30 | _x86_ | [TK]D-Fender: 3-6 months |
| 15:09.39 | _x86_ | well within the life cycle of the phone ;) |
| 15:09.49 | Katty | [TK]D-Fender: i have an urge to look at rings. |
| 15:09.57 | Katty | [TK]D-Fender: what's coming over me :< |
| 15:10.03 | tzanger | coppice: but the distortion is not constant... it's got a low metallic noise and a high metallic nosie |
| 15:10.13 | tzanger | the 425+550 is the congestion signal |
| 15:10.54 | tzanger | http://www.mixdown.ca/~andrew/dump/tsfill-fromstamp.wav is another example of when I'm sending constant data and just receiving the background noise of whatever room the phone is in, but you can hear the distortion again about 2s and 16s in I think |
| 15:10.57 | coppice | oh, it varies through the file. the DC and 100Hz is there all the time, the other components seem to change |
| 15:11.01 | [TK]D-Fender | _x86_: I would suggest the 330 + 1 PoE injector (more expensive than the brick, but RECYCLABLE). That way when you go PoE you'll have an injector around that can be used with other PoE-needing devices. |
| 15:11.18 | [TK]D-Fender | _x86_: At which point make sure its Cisco compatible as well. |
| 15:11.56 | tzanger | coppice: hmm, I wonder where teh DC's coming from, I recreated the wav from pulling the actual ulaw frame data out of the TDMoE frame and converting it |
| 15:12.01 | [TK]D-Fender | Katty: You're going way to giddy over those 3 word's you've been waiting so long for. He isn't ready yet.... |
| 15:12.53 | *** join/#asterisk r3zon8 (n=hg@64.80.101.162) |
| 15:13.26 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 15:13.28 | coppice | tzanger the second file is DC + 100Hz + a bit of 50Hz and 200Hz |
| 15:13.48 | coppice | in the first file there seem to be some bursts of 1.1kHz |
| 15:14.08 | Katty | [TK]D-Fender: i know. |
| 15:14.11 | Katty | [TK]D-Fender: i'm not ready yet |
| 15:14.16 | Katty | [TK]D-Fender: but the shiny is so pretty :/ |
| 15:14.41 | tzanger | coppice: it's clearly not frame slips, but I'm wondering if it's bits within the timestamps shifting |
| 15:15.03 | Corydon76-dig | Ooo, look, a bicycle! |
| 15:15.08 | coppice | nope. its just a damned ugly signal |
| 15:15.09 | tzanger | there are some errata notes for the DSP that say if the rx clock is "too slow" you can get corruption like that |
| 15:15.20 | tzanger | coppice: haha, well I"ll have a hardware loop on it shortly I think |
| 15:15.24 | tzanger | I can then eliminate the analog bits and see |
| 15:15.47 | Katty | goshdangitanyhow |
| 15:15.54 | Katty | stupid linksys people |
| 15:16.15 | Katty | i just /told/ you i'm doing a cascading wan...why are you asking me about usernames and passwords to my internet connection?!?! |
| 15:16.18 | Katty | RAWR |
| 15:16.27 | Katty | :< |
| 15:16.33 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 15:16.47 | Katty | GET EM! |
| 15:17.19 | tzanger | coppice: where can I download your ears to analyse this on my own? |
| 15:17.48 | coppice | there are lots of tools for looking at wave files |
| 15:18.12 | [TK]D-Fender | Notepad FTW! |
| 15:18.24 | tzanger | and your sliptest application only works when it's receiving audio from an unterminated line (strong untainted echo), correct? auto-correlation doesn't work well otherwise? |
| 15:18.37 | tzanger | [TK]D-Fender: my MSN name is "IRC is multi-player notepad" |
| 15:18.54 | coppice | you nedd some echo. it doesn't need to be that strong |
| 15:19.14 | [TK]D-Fender | tzanger: My MSN client overrides people's silly nicknames and shoves their real names at all times :) |
| 15:19.34 | tzanger | ok... sliptest didn't work at all with the echo I did get back, but again it's the echo from the ear to mic of a phone offhook in a noisy room :-) |
| 15:20.18 | *** join/#asterisk Yourname`` (n=Miranda@unaffiliated/yourname/x-837320) |
| 15:20.38 | coppice | ah, well, that wouldn't help |
| 15:21.18 | _x86_ | tzanger: haha |
| 15:21.29 | _x86_ | multi-player notepad... that's teh awesomeness |
| 15:22.19 | *** join/#asterisk hfb (n=hfb@pool-71-106-223-11.lsanca.dsl-w.verizon.net) |
| 15:22.49 | [TK]D-Fender | _x86_: wanna play a game of solitaire with me? :) |
| 15:23.20 | *** join/#asterisk ivrc (n=chatzill@adsl-074-228-054-164.sip.bct.bellsouth.net) |
| 15:23.25 | tzanger | coppice: indeed. the hardware loop will help figure this out I think |
| 15:23.50 | tzanger | I'm baffled though, but at least I think the elastic buffers and the driver are working correctly, this seems like data corruption |
| 15:24.44 | *** join/#asterisk galeras (n=galeras@200.31.204.42) |
| 15:26.01 | ivrc | Need some help with Zaptel - getting errors on 'make config' -- gives a batch of 'Unknown line at line' with line numbers of 5959 to 5970 - suggestions? |
| 15:28.47 | Katty | hahahahahahahahahha |
| 15:28.52 | Katty | linksys didn't know what a cascading wan was. |
| 15:29.00 | Katty | sigh. |
| 15:29.11 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 15:29.28 | Katty | and these are who people consider the 'professionals' :/ |
| 15:33.03 | [TK]D-Fender | Katty: umm.. I'm not quite clear with what you mean there... |
| 15:33.43 | [TK]D-Fender | ivrc: What version of Zaptel? |
| 15:33.57 | Voicemeup | ivr now rorry |
| 15:34.02 | Voicemeup | its always does that |
| 15:34.11 | ivrc | zaptel 1.4.5.1 |
| 15:34.14 | Voicemeup | jsut cant find those modules from config |
| 15:34.20 | Voicemeup | ivrc no worry i said lol |
| 15:34.25 | Voicemeup | you have a zap device ? |
| 15:34.41 | Voicemeup | uncomment it in the zap config.. then ztcfg -vvv |
| 15:34.42 | ivrc | no worry, but it doesn't seem to see the TDM400 |
| 15:35.07 | ivrc | appreciate the help - please bear with the noob |
| 15:35.10 | Katty | [TK]D-Fender: then you're mental. |
| 15:35.27 | [TK]D-Fender | Katty: Or maybe your term isn't clear or appropriate :) |
| 15:35.29 | Katty | [TK]D-Fender: firewall hands out 192.168.0.x the router beneath it hands out 1.x |
| 15:35.52 | [TK]D-Fender | Katty: I've never seen a Linksys that didn't NAT across its WAN port.... |
| 15:36.06 | Katty | me either. |
| 15:36.09 | [TK]D-Fender | Katty: Except when OpenWRT'd :) |
| 15:36.23 | [TK]D-Fender | Katty: Guess what, it isn't supported natively :) |
| 15:36.43 | [TK]D-Fender | Katty: Linksys doesn't make routers, them make NAT Toasters :p |
| 15:36.48 | [TK]D-Fender | they* |
| 15:37.24 | Katty | yeah, but it's not like i can throw my clients router away |
| 15:37.25 | [TK]D-Fender | Katty: Go set up a Linux box as a gateway then. |
| 15:37.41 | Katty | yeah yeah |
| 15:37.42 | Katty | silly males |
| 15:37.45 | Katty | always with your solutions |
| 15:37.45 | *** join/#asterisk xezz (n=sdd@87.203.215.213) |
| 15:37.49 | [TK]D-Fender | Katty: Correct, its THEIR job to dispose of waste products :) |
| 15:38.34 | [TK]D-Fender | Katty: Oh I'm sorry... "Yes I understand your pain and its OK.we still love you and you'll figure it out when you're ready" |
| 15:38.37 | Katty | [TK]D-Fender: don't you start annoying me too |
| 15:38.48 | Katty | there you go! |
| 15:38.49 | [TK]D-Fender | :) |
| 15:38.49 | Katty | much better. |
| 15:39.17 | Katty | :P |
| 15:39.26 | [TK]D-Fender | Katty: I sometimes forget to flip that "doesn't actually want help, just wants someone to listen" switch.... I learn fast through :p |
| 15:39.46 | Katty | you need a wireless transmitter so i can use a remote |
| 15:39.47 | xezz | hello, is there an other opensource call center like trixbox ? |
| 15:39.56 | Katty | ok |
| 15:40.23 | ivrc | voicemeup: ztcfg shows the 4 channels - is there any easy way of testing without setting up extensions and trunks? (tried making an inbound call with verbosity set high, but nothing shows) |
| 15:41.06 | [TK]D-Fender | xezz: No, employees don't tend to work for free.... |
| 15:41.56 | [TK]D-Fender | ivrc: Yes you have to setup your channels or you'll get nothing. |
| 15:42.25 | *** join/#asterisk blq (i=Bl@dslb-088-064-141-083.pools.arcor-ip.net) |
| 15:42.27 | blq | hi |
| 15:42.55 | ThoMe | hello |
| 15:43.02 | ThoMe | iss dialstatus "DONTCALL" == DND ? |
| 15:43.17 | ThoMe | i have a "dnd" button at my phone |
| 15:43.37 | ThoMe | is the dnd button / status == dontcall if i fetch the $DIALSTATUS ? |
| 15:44.25 | *** join/#asterisk hrmphh (i=patrick@notchill.com) |
| 15:44.27 | booray | what would be the featd compliment of /etc/zaptel.conf? |
| 15:44.48 | hrmphh | what asterisk config steps need to be taken to add a sangoma a101d card? i currently am using a POS digium analog car |
| 15:46.21 | hrmphh | wanpipe: AFT-A101-SH T1/E1 card found (HDLC (DS) rev.31), cpu(s) 1, bus #3, slot #13, irq #10 |
| 15:46.27 | hrmphh | its recognized as a "wanpipe"? |
| 15:46.44 | rob0 | Put that in your wanpipe and smoke it! |
| 15:47.04 | [TK]D-Fender | ThoMe: No. |
| 15:47.12 | coppice | evidently not a wanpipe of peace |
| 15:47.59 | [TK]D-Fender | hrmphh: You setup your wanpipe drivers, start wanrouter and the rest is jusk like a Digium card |
| 15:48.16 | ThoMe | [TK]D-Fender: no? hmm? |
| 15:48.20 | ThoMe | [TK]D-Fender: what is DND ? |
| 15:48.37 | [TK]D-Fender | ThoMe: DND = Do Not Disturb. |
| 15:48.39 | hrmphh | cool |
| 15:48.42 | hrmphh | where do i add the asterisk config |
| 15:48.45 | hrmphh | im perusing sangoma site now |
| 15:48.49 | coppice | dungeons and dragons |
| 15:48.54 | rob0 | Maybe it's a wannabepipe |
| 15:49.11 | [TK]D-Fender | hrmphh: What do you mean "add the asterisk config"? You jsut make your zaptel.conf & zapata.conf like normal |
| 15:49.23 | errr | with asterisk 1.2.x is it possible to speed up the rate that the voicemail info is read to you.. "You have one new message and one old message" that stuff.. and the envolope stuff |
| 15:50.04 | errr | Id like it to sound more like talking and less like poor reading |
| 15:50.46 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
| 15:50.53 | [TK]D-Fender | errr: Go re-record the prompts |
| 15:51.30 | hrmphh | wanpipe1 | AFT HDLC | N/A | Connected | |
| 15:51.32 | hrmphh | woot :) |
| 15:51.33 | errr | [TK]D-Fender: you cant just speed up the rate? |
| 15:51.58 | Zeeek | Oyé, oyé, the Voip Users Conference starts in 38 minutes: http://VoipUsersConference.org IRC: #voip-users-conference |
| 15:52.00 | [TK]D-Fender | errr: Sure, grab an audio editing tool and have fun |
| 15:52.17 | errr | [TK]D-Fender: we dont mind it sounding like a computer, it just that the reading is done slowly |
| 15:52.39 | *** join/#asterisk jprater (n=jprater@cpe-72-185-204-251.tampabay.res.rr.com) |
| 15:52.41 | Zeeek | I may have a $60 dollar phone bill from bridging two ZAP channels, hurray :( |
| 15:52.43 | errr | [TK]D-Fender: like in M$ you can speed up and slow down how fast Microsoft sam reads a line of text.. |
| 15:53.15 | hrmphh | tk; yeah just wasnt sure how to set up the channels in zaptel, its just fxoks=1 and fxsks=2-4 right now for teh analog card |
| 15:53.35 | [TK]D-Fender | hrmphh: You set them up the same as you would for their Digium equivalent |
| 15:54.08 | hrmphh | as if they were fxs chans? |
| 15:54.29 | [TK]D-Fender | hrmphh: I've said it TWICE ALREADY. |
| 15:56.03 | hrmphh | yes yes |
| 15:56.12 | hrmphh | ill rtfm :) |
| 15:56.34 | hrmphh | just trying to figure out how to tell it which hardware/channel to use |
| 15:56.48 | hrmphh | because id like to keep both running |
| 15:56.51 | hrmphh | the analog is simply a backup |
| 15:57.40 | hrmphh | nm i found this: http://wiki.sangoma.com/wanpipe-linux-asterisk-install |
| 15:58.06 | jprater | I'm having some trouble with the basic install of a TE120P card. It looks as though it's failing to enumerate by the lsipci output. Does anyone know how to deal with that? |
| 15:58.07 | [TK]D-Fender | hrmphh>tk; yeah just wasnt sure how to set up the channels in zaptel, its just fxoks=1 and fxsks=2-4 right now for teh analog card <- I can promise you this is wrong however |
| 15:58.10 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 15:58.28 | *** join/#asterisk qdk_ (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 15:58.55 | hrmphh | yeah im saying thats how my analog is |
| 15:59.08 | ivrc | edited zapata.conf -- tried a module reload chan_zap.so - that threw errors on reload - rebooted, and now I get "ZT_CHANCONFIG failed on channel 1: No such device or address (6)" |
| 15:59.10 | hrmphh | http://wiki.sangoma.com/wanpipe-linux-asterisk-appendix#sampleZaptel |
| 15:59.12 | hrmphh | sangoma has legit docs |
| 15:59.13 | hrmphh | im all set |
| 15:59.14 | *** join/#asterisk Lawbringer (n=Lawbring@212.183.136.192) |
| 15:59.19 | hrmphh | they even provide a script to create the files |
| 16:00.06 | [TK]D-Fender | hrmphh: Hold on... A101= with mixed FXO/FXS signalling? You running a channel bank? |
| 16:00.20 | hrmphh | no no |
| 16:00.22 | *** join/#asterisk ManxPower (n=manxpowe@209.16.72.135) |
| 16:00.23 | hrmphh | thats my old setup |
| 16:00.29 | hrmphh | a digium analog card |
| 16:00.33 | hrmphh | only used when T1 is down |
| 16:00.47 | hrmphh | im using PRI B8ZS ESF for the digital |
| 16:00.52 | *** join/#asterisk pasquall (n=pasquall@200-160-115-020.static.spo.ctbc.com.br) |
| 16:00.53 | [TK]D-Fender | hrmphh: Ok, your setup description was terribly spotty. |
| 16:00.57 | hrmphh | sorry |
| 16:01.08 | [TK]D-Fender | hrmphh: Start over and list the cards & modules you are using NOW. |
| 16:01.35 | hrmphh | lol ok |
| 16:02.02 | hrmphh | the 3fxs, 1fxo card |
| 16:02.15 | hrmphh | from digium |
| 16:02.24 | hrmphh | used exclusively right now to run our system |
| 16:02.30 | hrmphh | added a sangoma a101d-x card |
| 16:02.46 | hrmphh | which we're migrating to for our primary voice connection, the analog card will be kept for backup |
| 16:03.34 | *** part/#asterisk pasquall (n=pasquall@200-160-115-020.static.spo.ctbc.com.br) |
| 16:04.36 | [TK]D-Fender | hrmphh: Ok, then your Digium config should stay the smew and channel on the ned for your sangoma |
| 16:06.40 | [TK]D-Fender | ivrc: pastebin your zaptel.conf , zapata.conf, "ztcfg -vvvv", and "can /proc/interrupts" |
| 16:06.42 | [TK]D-Fender | ~pb |
| 16:06.43 | jbot | A Pastebin is a place to paste your stuff 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 |
| 16:06.44 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^ |
| 16:07.13 | hrmphh | span=1,0,0,esf,b8zs |
| 16:07.13 | hrmphh | bchan=1-23 |
| 16:07.13 | hrmphh | dchan=24 |
| 16:07.20 | ivrc | fender: will put it up in just a few |
| 16:07.21 | hrmphh | have got that to add to my zaptel.conf |
| 16:07.31 | [TK]D-Fender | hrmphh: 1,1,0 <0 |
| 16:07.38 | [TK]D-Fender | <- * |
| 16:07.41 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:07.50 | [TK]D-Fender | hrmphh: you want to take timing from the telco |
| 16:08.03 | hrmphh | 1,1,0 instead of 1,0,0? |
| 16:08.17 | hrmphh | oh and this is an integrated T1 too, so we only have 12 channels |
| 16:08.42 | hrmphh | so ill prob need to bchan 1-11 and chan=12, will ask teh L3 tech today |
| 16:09.11 | [TK]D-Fender | hrmphh: You're going to be running Data over it as well? |
| 16:09.18 | hrmphh | yeah |
| 16:09.21 | hrmphh | but thats already split off |
| 16:09.27 | hrmphh | ethernet handoff for data |
| 16:09.35 | Zeeek | ë“‘{¶«¡Çø≠÷÷…∞~ß◊©≈‹‡Ò∂ƒï¬ÃŒÃȬµπœîºÚ†®êÂæ›â„¢√∫ı¿••\Ó‰|ËÃÎfl·∆∑ÔâˆÅ’犯™‚ÊÅÆ |
| 16:09.38 | [TK]D-Fender | hrmphh: D is usually still 24. so: 1-11 + 24. 12-23 data |
| 16:09.41 | hrmphh | when no voice channels are in use, we've got full 1.5mb |
| 16:09.43 | hrmphh | k |
| 16:10.19 | [TK]D-Fender | hrmphh: Ph so you've got a T1 router to sit between * & telco already? |
| 16:10.43 | hrmphh | no they have an IAD |
| 16:10.51 | [TK]D-Fender | hrmphh: Oh let me guess a Cisco 2430 or so dynamic T1, right? |
| 16:10.58 | hrmphh | nah its adtran |
| 16:11.05 | hrmphh | ive used cisco iads w/ctc before, but this is l3 |
| 16:11.09 | [TK]D-Fender | hrmphh: Seen the same service offered by XO |
| 16:11.14 | hrmphh | yup |
| 16:11.16 | hrmphh | they ahve similar |
| 16:11.22 | hrmphh | its not bad, $400/mo |
| 16:11.24 | ivrc | fender: posted on pastebin.com under ivrc |
| 16:12.00 | [TK]D-Fender | hrmphh: Excellent deal if everything works out. I can't get that kind of service here yet. My telco tech is checking things out because its not a service that scales to the market around here. |
| 16:12.02 | hrmphh | hrm which switchtype keyword do i use for NI2? |
| 16:12.06 | [TK]D-Fender | ivrc: link please. |
| 16:12.12 | *** join/#asterisk revcane (n=dng@cpe-76-186-113-159.tx.res.rr.com) |
| 16:12.25 | jprater | Can anyone offer some help with "insmod: error inserting 'wcte12xp.ko': -1 Unknown symbol in module"? |
| 16:12.35 | [TK]D-Fender | ivrc: And the are HUNDREDS of pastebin sites. Do you really think we're going to go LOOKING for which one you used and hope to find your post? |
| 16:12.41 | ivrc | fender: http://pastebin.com/d1c978742 |
| 16:12.55 | hrmphh | lol |
| 16:13.27 | revcane | do any of you guys know of a good howto or tutorial on setting on a r1t1 card for data and voice ? |
| 16:13.28 | ivrc | sorry |
| 16:13.45 | [TK]D-Fender | ivrc: TDM400P? |
| 16:13.54 | ivrc | fender: yes |
| 16:14.23 | tzanger | coppice: this is WEIRD |
| 16:14.31 | [TK]D-Fender | ivrc: "modprobe zaptel" , "modprobe wctdm", "ztcfg -vvvv" , "cat /proc/interrupts" |
| 16:14.37 | tzanger | I modified sliptest to send constant data (ulaw 0xaa) -- I get back ulaw 0xaa without fail |
| 16:14.40 | tzanger | no corruption |
| 16:14.48 | tzanger | I use unmodified sliptest to send awgn... I can't correlate |
| 16:15.01 | tzanger | I use asterisk and listen, and I send good audio but get that same weird corruption |
| 16:16.42 | [TK]D-Fender | ivrc: And you don't have your channels defines in zapata.conf unless the included file you DIDN'T provide has those settings. At which poitn everything following the include is superfluous. |
| 16:16.55 | ivrc | fender: the modprobes showed nothing - posted the others http://pastebin.com/d54db966c |
| 16:17.49 | ivrc | fender: wouldn't the defines be group=0 and channel=1? |
| 16:18.12 | ivrc | fender lines 100-101 of http://pastebin.com/d54db966c |
| 16:19.52 | hrmphh | Using a combination of Analog Cards and T1/E1 Cards Analog Cards register 24 channels, even if less ports are used, so the first T1/E1 channel will start at 25. |
| 16:19.55 | hrmphh | hmm interesting |
| 16:19.56 | hrmphh | have you heard that before? |
| 16:21.24 | ivrc | fender: zapata_additional.conf is empty |
| 16:21.29 | [TK]D-Fender | ivrc: Ah, my mistake on the "channel" it was buried and not in the same format or spacing I might have expected. |
| 16:22.26 | [TK]D-Fender | ivrc: on redoing "ztfcg -vvvv" following the modprobes yous till dont see wctdm in there? |
| 16:22.47 | [TK]D-Fender | hrmphh: Yup |
| 16:23.02 | [TK]D-Fender | hrmphh: Depending on which modules load first |
| 16:24.07 | ivrc | fender: just reran the modprobes - now the ztcfg shows 4 channels configured |
| 16:24.33 | ivrc | fender: Channel 01: FXS Kewlstart (Default) (Slaves: 01) and so on... |
| 16:25.09 | [TK]D-Fender | ivrc: no errors out of ztcfg? Do you see the module in your interrupts dump? |
| 16:26.02 | ivrc | fender: it shows in the interupt -- 169: 315555 310073 IO-APIC-level wctdm |
| 16:26.12 | *** join/#asterisk t3rror (n=t3rror@gateway.sscgp.com) |
| 16:26.41 | [TK]D-Fender | ivrc: Ok, looking good now. Now try to start * |
| 16:27.42 | ivrc | fender: still nothing showing on the console when I make an incoming call |
| 16:28.13 | tzanger | coppice: I got it to corrupt on my terms |
| 16:28.21 | tzanger | if I send static data, it comes trhough fine |
| 16:28.31 | tzanger | so I send 3 specific values over and over |
| 16:28.48 | tzanger | and you can watch the auto-correlator find and lose it |
| 16:29.07 | ivrc | fender: want to pick up a few $$$ and remote in? |
| 16:29.18 | *** join/#asterisk zydrunas (n=Miranda@216.49.228-ip-184.ckt.net) |
| 16:29.50 | *** join/#asterisk kuku5 (n=kuku5@c-71-201-219-72.hsd1.il.comcast.net) |
| 16:30.10 | kuku5 | Is voicepulse down for everyone else too ? |
| 16:30.21 | hrmphh | fender; any way to tell which module loads first? |
| 16:31.21 | Zeeek | digium guys? |
| 16:32.11 | [TK]D-Fender | hrmphh: Not sure 100% how to tell myself. I usually invert my config to test. |
| 16:33.22 | *** join/#asterisk hypa7ia (i=hypatia@judecca.aculei.net) |
| 16:40.27 | cpm | kuku5, my vp lines aren't connecting. |
| 16:40.29 | *** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net) |
| 16:45.10 | kuku5 | cpm: their main number doesnt work either |
| 16:45.31 | cpm | hrmm |
| 16:45.39 | cpm | must be having issues today |
| 16:45.44 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 16:46.48 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 16:47.12 | kuku5 | yah |
| 16:47.18 | *** join/#asterisk horsesgofaster (n=dcantera@pool-72-82-220-61.cmdnnj.east.verizon.net) |
| 16:47.57 | cpm | everything's registered, and all that, but no calls going through. |
| 16:48.27 | *** part/#asterisk horsesgofaster (n=dcantera@pool-72-82-220-61.cmdnnj.east.verizon.net) |
| 16:49.40 | kiscokid | /channels |
| 16:54.11 | CCFL_Man2 | this fxs card i got today |
| 16:54.16 | CCFL_Man2 | it's dead |
| 16:54.24 | CCFL_Man2 | it's for my channel bank |
| 16:54.43 | rob0 | 'E's not dead! 'E's stunned! You stunned 'im!! |
| 16:55.10 | CCFL_Man2 | might have been stunned before hand by lightning |
| 16:55.42 | coppice | how does and fxs card work with a channel bank? :-\ |
| 16:55.57 | [TK]D-Fender | coppice: its a card FOR the CB, not to USE the CB |
| 16:56.10 | [TK]D-Fender | coppice: AKA modular CB |
| 16:56.10 | CCFL_Man2 | coppice: because it's designed to plug right into the chassis? :P |
| 16:56.13 | cpm | That is an EX-fxs card |
| 16:56.27 | CCFL_Man2 | it is |
| 16:56.39 | CCFL_Man2 | $40 down the drain |
| 16:56.42 | rob0 | Look, is there anything you can do? |
| 16:56.54 | CCFL_Man2 | nope, listed it as is |
| 16:57.08 | rob0 | I know, a mixed Python reference, but I can't help myself. |
| 16:57.13 | CCFL_Man2 | here i thought these cards don't die |
| 16:57.30 | CCFL_Man2 | but they do |
| 16:57.49 | cpm | I know a dead fxs 'ard when I see one |
| 16:57.58 | CCFL_Man2 | heh |
| 16:58.03 | hmmhesays | before or after its been hit with a hammer |
| 16:58.24 | rob0 | Old FXS cards never die; they only crumble away. |
| 16:58.43 | CCFL_Man2 | rob0: none of the mights light up :P |
| 16:58.48 | outtolunc | usually it is the fxo card that gets hit by lightening |
| 16:58.51 | CCFL_Man2 | lights |
| 16:59.07 | CCFL_Man2 | outtolunc: not in a channel bank |
| 16:59.13 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 16:59.25 | outtolunc | haha |
| 16:59.31 | outtolunc | think whatever you like <G> |
| 17:00.30 | drako | why mixmonitor on queues does not merge the in and out files? |
| 17:01.17 | CCFL_Man2 | outtolunc: if this is put in a CO more than likely the loop has access to lightning |
| 17:03.20 | CCFL_Man2 | bastard |
| 17:07.13 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 17:07.21 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 17:07.22 | coppice | access to lightning will be a good selling point to Dr Frankenstein |
| 17:07.43 | hmmhesays | thats funny |
| 17:07.47 | tzanger | haha |
| 17:07.53 | tzanger | coppice: I found the problem, but solving it's a bitch |
| 17:07.56 | tzanger | it's my elastic buffers |
| 17:08.02 | tzanger | basically a variable-width ring buffer |
| 17:08.34 | tzanger | if my max buffer size is an integer multiple of the data repeat value, everything works |
| 17:08.41 | tzanger | so when I round the corner of the buffer, it misses |
| 17:08.48 | tzanger | should be easy to solve |
| 17:11.37 | *** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 17:14.23 | CCFL_Man2 | sucks |
| 17:14.37 | CCFL_Man2 | someone sell me an fxs card for an adit 600 cheap |
| 17:16.49 | *** join/#asterisk techie (n=techie@adsl-68-127-127-133.dsl.frsn02.pacbell.net) |
| 17:17.28 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 17:17.36 | *** part/#asterisk jprater (n=jprater@cpe-72-185-204-251.tampabay.res.rr.com) |
| 17:18.38 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 17:23.09 | [TK]D-Fender | CCFL_Man2: ebay it |
| 17:25.37 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.186.199) |
| 17:26.16 | Voicemeup | wahts the flag to choose a music cals in a sip.conf def on 1.4.11 |
| 17:26.20 | Voicemeup | class i mean |
| 17:26.51 | drako | why mixmonitor on queues does not merge the in and out files? |
| 17:29.25 | booray | I have a theory, and that is that the Verizon Flexgrow T1 service is incompatible with Asterisk. |
| 17:29.45 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 17:30.06 | hmmhesays | what makes you come to this theory? |
| 17:31.19 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 17:31.33 | tzanger | Sanne: /last coppice |
| 17:31.37 | tzanger | oop |
| 17:32.51 | booray | hmmhesays: two days on the phone with verizon and digium, and trying every single type of signalling remotely possible, getting just about nothing but rings at the cli. verizon expecting 1's when we go off hook and a never changing state, but no alarms, etc etc |
| 17:33.38 | booray | the closest I have gotten is a ring, but when asterisk answers, it doesn't _really_ answer, but thinks it does, and then complains about weird ring/off-hook states on the channel |
| 17:35.55 | Jameno123 | Oh for those interested |
| 17:36.03 | Jameno123 | I leave last night, issues galore |
| 17:36.11 | Jameno123 | come back today, asterisk is operating perfectly fine :( |
| 17:36.24 | Jameno123 | (for those following my issues over the past 2 days) |
| 17:36.39 | Corydon76-dig | booray: That sounds like Verizon set up a data T1, not a voice T1 |
| 17:36.43 | Jameno123 | Now to figure out what changed since i left last night :( |
| 17:36.55 | rob0 | cpm cracked your root password, came in and fixed it for you. |
| 17:37.24 | Corydon76-dig | booray: A lot of the CLECs won't provision voice T1s anymore... they do their own proprietary voip over a T1 link |
| 17:37.28 | rob0 | I tried to tell him it was a naughty thing to do. |
| 17:37.45 | booray | Corydon76-dig: that would make sense, but why would they claim that I'm supposed to be getting six voice channels? |
| 17:38.01 | Corydon76-dig | booray: signalled how? |
| 17:38.21 | booray | Corydon76-dig: the only stuff on their paperwork is esf/bz8s loop start |
| 17:38.29 | *** part/#asterisk javar (n=javar@69.79.134.24) |
| 17:38.48 | Corydon76-dig | booray: are you plugged directly into the quickjack or into their channel bank? |
| 17:39.00 | booray | directly in. they didn't provide a channel bank |
| 17:39.22 | booray | some initial paperwork indicated a shark unit, which I believe is a channel bank, but it never ended up coming |
| 17:39.36 | booray | and the people we talk to now indicate that we shouldn't have gotten one |
| 17:40.10 | Corydon76-dig | I betcha they're supposed to have a channel bank in place that translates their voip back into voice for 6 channels |
| 17:40.26 | *** part/#asterisk Voicemeup (n=VoiceMeU@modemcable132.108-83-70.mc.videotron.ca) |
| 17:40.34 | Corydon76-dig | The ones down here usually put in an Adtran of some sort, then break out the T1 with the 6 voice channels, and also hand you an Ethernet uplink |
| 17:41.20 | Corydon76-dig | booray: I bet they've proviisioned 100% as data and don't realize that they've misprovisioned |
| 17:41.47 | Corydon76-dig | The provisioning engineer saw the Shark unit and provisioned as if it was in place |
| 17:42.10 | booray | Corydon76-dig: I think you're right |
| 17:42.40 | booray | Corydon76-dig: there was enough confusion between individual verizon departments the other day that I would suspect that it was misprovisioned as you're suggesting |
| 17:43.00 | CCFL_Man2 | [TK]D-Fender: i got screwed buying a fxs card on ebay |
| 17:43.31 | Corydon76-dig | I've dealt with this at the telco level. If you suggest that's the problem and ask them to get the switch engineer on the line, they can fix the provisioning in 5 minutes |
| 17:43.57 | Corydon76-dig | The two week "normal provisioning" is bullshit |
| 17:44.01 | booray | Corydon76-dig: ha, well, we'll see if I can anyone on the phone |
| 17:44.14 | booray | I'll report back soon |
| 17:44.32 | Corydon76-dig | It's 5 minutes to do the actual work and 2 weeks to dawdle around |
| 17:45.26 | Corydon76-dig | booray: and after they change the provisioning, make sure they do a "full restart" on the circuit |
| 17:45.26 | Corydon76-dig | Lots of stuff doesn't go into effect otherwise |
| 17:46.42 | Corydon76-dig | I had to put a PRI in debugging once and tell the switch engineer what they turned off was bullshit, I was still seeing the packets come across the circuit |
| 17:47.00 | De_Mon | wtf! |
| 17:47.01 | Corydon76-dig | They did a full restart and magically, everything worked |
| 17:47.17 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 17:47.18 | *** mode/#asterisk [+o russellb] by ChanServ |
| 17:47.44 | Corydon76-dig | And hey, that was the same company, too... MCI or "Verizon Business" |
| 17:48.04 | De_Mon | calls are now failing to yet another customer with this bloody Unsupported SDP t38 crap. |
| 17:49.20 | *** join/#asterisk [hC] (n=hardcore@wsip-70-184-124-51.ph.ph.cox.net) |
| 17:49.23 | *** join/#asterisk Humblgrumpf (n=humblgru@p54B08664.dip0.t-ipconnect.de) |
| 17:53.24 | De_Mon | I see an update to zaptel around when this started happening, leme see what downgrading does |
| 17:55.57 | De_Mon | ok good, still happening. Didn't think zaptel had anything to do with this... |
| 17:56.48 | booray | Corydon76-dig: and of course now I can't get ahold of anyone... :-/ |
| 18:01.23 | Aeudian | i have multiple voip accounts coming inbound on my asterisk server and i want each to have their own sip port like 5060, 5061, 5062, 5063. under sip.conf how to i tell asterisk to bindport=5060 to do multiple ports or do i just repeat code 3x to match my ports |
| 18:01.38 | ThoMe | spricht hiwer wer deutsch? |
| 18:01.59 | ThoMe | what is the best alternative for a call-group? (without agents) |
| 18:02.23 | *** join/#asterisk metfan2007 (n=metfan20@189.135.156.38) |
| 18:02.27 | ThoMe | i would like check if busy or dnd .. per member |
| 18:02.56 | metfan2007 | Anybody from Digium here?? I'm calling tu installation support service but nobody answer... and I nees urgent help.... :( |
| 18:03.00 | [TK]D-Fender | Aeudian: you can't |
| 18:03.15 | ThoMe | [TK]D-Fender: do u have a idea for me? :-) |
| 18:03.23 | De_Mon | ThoMe dynamic queue members |
| 18:03.32 | *** join/#asterisk deb_user (n=debian_l@70-59-107-53.albq.qwest.net) |
| 18:03.40 | Aeudian | TK: so i need to make all sip invites go through 5060 on the same ip? |
| 18:03.42 | ThoMe | De_Mon: how? do u have a docu? |
| 18:03.45 | [TK]D-Fender | ThoMe: Your question is very vague |
| 18:04.28 | deb_user | anybody out there willing to peer with me and get our organization linked into the dundi network? |
| 18:04.28 | De_Mon | ThoMe yeah voip-info.org talks about dynamic queue members |
| 18:04.28 | [TK]D-Fender | Aeudian: To a single port... as to IP, * will bid to each IP on your system if you use 0.0.0.0 |
| 18:04.48 | mvanbaak | metfan2007: did you try more then once ? it's 1 PM there so maybe they were out to lunch |
| 18:04.54 | ThoMe | De_Mon: ok :-) |
| 18:06.32 | r3zon8 | whats the default password for admin account on the vmware image? |
| 18:06.52 | r3zon8 | it said i would be allowed to set it, but i was never prompted? |
| 18:06.52 | mvanbaak | r3zon8: what vmware image ? |
| 18:06.55 | De_Mon | the vmware image? |
| 18:06.58 | [TK]D-Fender | lol |
| 18:07.01 | r3zon8 | sorry..asterisk now beta 6 |
| 18:07.03 | Aeudian | TK: i have 4 VoIP accounts inbound from the same carrier with 4 numbers, 1 per company. under users.conf i could set the ports to 5060 (or whatever i want to) My carrier will not allow me to register the same ip/port to a different account. Thats why i waanted to use 5060-5063. How would i go about this |
| 18:07.12 | [TK]D-Fender | r3zon8: wrong channel... it isn't supported here |
| 18:07.26 | mvanbaak | r3zon8: try #asterisknow |
| 18:07.44 | r3zon8 | thanks :) |
| 18:07.55 | deb_user | does anybody here even use dundi? |
| 18:08.02 | mvanbaak | deb_user: yup |
| 18:08.15 | [TK]D-Fender | Aeudian: Go setup SER in front or something then |
| 18:08.26 | r3zon8 | is there a big diff between using astNow, and asterisk? |
| 18:08.28 | [TK]D-Fender | deb_user: Realistically speaking? Nearly irrelevent |
| 18:08.41 | deb_user | fender: do you think it will take off? |
| 18:08.43 | [TK]D-Fender | r3zon8: thats like comparing a car to an ENGINE |
| 18:08.53 | rob0 | deb_user: why would it? |
| 18:09.04 | [TK]D-Fender | deb_user: with enough high-explosives, sure :p |
| 18:09.06 | mvanbaak | wtf is wrong with dundi ? |
| 18:09.08 | deb_user | well...does anybody here have a record in e164.org? or an ISN number? |
| 18:09.19 | rob0 | deb_user: termination services are very inexpensive. |
| 18:09.27 | deb_user | mvanbaak: i think dundi is sweet |
| 18:09.38 | [TK]D-Fender | r3zon8: AsteriskNOW is a full distro CD for which Asterisk is only a PIECE. |
| 18:09.40 | mvanbaak | I think my stuff is in e164.org |
| 18:09.50 | mvanbaak | but I dont care actually |
| 18:09.58 | [TK]D-Fender | r3zon8: We don't support the other 95% |
| 18:10.01 | deb_user | rob0: true, but why wouldn't we want to move more towards pure voip? |
| 18:10.24 | [TK]D-Fender | deb_user: because the world at large does not give a shit about * and Dundi. |
| 18:10.27 | deb_user | mvanbaak: have you EVER actually observed a call that was routed to you via e164.org? |
| 18:10.45 | [TK]D-Fender | deb_user: e164 is gaining in popularity naturally |
| 18:10.47 | DarKnesS_WolF | i have a problem with TDM400 it has 4 FXS modiles i do load wctdm and the leds are one |
| 18:11.07 | deb_user | fender: so you don't think dundi will ever amount to anything? |
| 18:11.07 | mvanbaak | deb_user: I dont know. my number is in a couple of ENUM trees and I point them all to the same place inside my asterisk |
| 18:11.11 | hmmhesays | I just found out i can get 7mbps DSL here |
| 18:11.13 | hmmhesays | rock |
| 18:11.26 | mvanbaak | so I dont know what calls come via e164.org, my srv records or other means |
| 18:11.32 | hrmphh | switchtype=national |
| 18:11.33 | hrmphh | used for NI2? |
| 18:11.37 | DarKnesS_WolF | it configured in zaptel.cofnf with fxo_ks but when i do plug phones i don't hear any tomne |
| 18:11.44 | DarKnesS_WolF | tone * any idea why ? |
| 18:11.50 | mvanbaak | but to answer your question: I get 8% of my calls directly via voip |
| 18:11.52 | deb_user | mvanbaak: how about dundi? do you route outgoing calls via dundi lookups? |
| 18:12.05 | [TK]D-Fender | hmmhesays: Yeah, pray your central is on your street-corner |
| 18:12.10 | deb_user | or enum.org lookups? |
| 18:12.12 | mvanbaak | so 8% of the callers use enum or some other voip routing to me without going thru the PSTN |
| 18:12.16 | r3zon8 | TK- ahh i see, im assuming now is a quicker way to get started, or at least experiment |
| 18:12.33 | deb_user | mvanbaak: that's not too bad, considering how young the technology is |
| 18:12.35 | mvanbaak | deb_user: I use dundi between my own boxen. I dont connect to some public dundi cloud |
| 18:12.44 | deb_user | mvanbaak: why not? |
| 18:12.55 | [TK]D-Fender | DarKnesS_WolF: because "fxo_ks" is not a valid mode for zaptel.conf |
| 18:13.03 | mvanbaak | deb_user: because I dont meet the requirements for the dundi clouds |
| 18:13.11 | [TK]D-Fender | r3zon8: Started with WHAT is the question... |
| 18:13.18 | hmmhesays | [TK]D-Fender: anything is better than the cable i have now, they limit data transfer to 2gig a day |
| 18:13.37 | deb_user | mvanbaak: o really? what requirements are there? I would have thought the barrier to entry would be low, because they would want as many people as possible participating |
| 18:13.39 | DarKnesS_WolF | [TK]D-Fender: sorry in zaptel fsoks=1-4 |
| 18:13.44 | mvanbaak | and all the numbers we terminate come from speakup. and speakup already participates in the dundi cloud where digium is in as well |
| 18:13.46 | DarKnesS_WolF | fxoks=1-4 |
| 18:13.55 | [TK]D-Fender | DarKnesS_WolF: is * started? |
| 18:14.00 | mvanbaak | deb_user: I think you are mixing dundi and enum |
| 18:14.17 | DarKnesS_WolF | [TK]D-Fender: yes it is |
| 18:14.26 | DarKnesS_WolF | and it can detect the zap channel open |
| 18:14.44 | [TK]D-Fender | DarKnesS_WolF: well PB your configs |
| 18:14.54 | mvanbaak | ~pb |
| 18:14.54 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 18:15.13 | deb_user | mvanbaak: but dundi is a peering type of thing, can't I lookup voip routes when I dial a number using dundi? |
| 18:15.18 | [TK]D-Fender | mvanbaak: I know he knows PB, I've beaten him over the head with it plenty of times :) |
| 18:15.27 | deb_user | (i am new to dundi, I admit) |
| 18:15.32 | deb_user | i don't even have it configured yet |
| 18:15.41 | DarKnesS_WolF | [TK]D-Fender: ther eis something strange i just find |
| 18:15.47 | mvanbaak | lol [TK]D-Fender |
| 18:15.55 | [TK]D-Fender | deb_user: Learn to STAND before considering walking let alone running.... |
| 18:16.02 | DarKnesS_WolF | when i did call one of the zaptel channel i can't hear it rining on the dialing one just noise |
| 18:16.19 | [TK]D-Fender | DarKnesS_WolF: Did you plug in the molex power connector? |
| 18:16.47 | deb_user | fender: what's that supposed to mean? don't try learning about something new by talking to other people without reading all the technical specifications first?? |
| 18:16.59 | mvanbaak | http://saynotovista.electricmonk.nl/ |
| 18:17.24 | DarKnesS_WolF | [TK]D-Fender: sure i did |
| 18:17.34 | DarKnesS_WolF | i can hear the phones talking |
| 18:17.46 | DarKnesS_WolF | it just no dial tone when i pick up the phone |
| 18:18.18 | [TK]D-Fender | DarKnesS_WolF: .... CONFIGS please |
| 18:18.40 | deb_user | mvanbaak: so what would you recommend to get started with dundi for a really small organization? |
| 18:18.45 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 18:18.48 | DarKnesS_WolF | [TK]D-Fender: okay one min |
| 18:19.14 | mvanbaak | deb_user: the question is wether you need dundi in a really small org. |
| 18:19.27 | metfan2007 | have anybody experiment that zaptel startup script does not load zaptel and wcxxxx modules correctly?? |
| 18:19.46 | deb_user | mvanbaak: i'm just an early adopter, I like to mess around with new stuff |
| 18:20.22 | [TK]D-Fender | Anyone got an HTC / UTSTARCOM PPC 6700 (cell phone) they could give me some opinions on? |
| 18:20.38 | *** join/#asterisk neverblue2 (n=neverblu@unaffiliated/neverblue) |
| 18:20.39 | mvanbaak | [TK]D-Fender: the phone rox |
| 18:20.53 | [TK]D-Fender | mvanbaak: You upgraded to WM6 on it? |
| 18:20.58 | mvanbaak | it's fast, 100000000+ times more stable then previous versions |
| 18:21.07 | mvanbaak | [TK]D-Fender: mine came with WM6 |
| 18:21.22 | mvanbaak | WM5 is the worst WM release I've seen |
| 18:21.29 | [TK]D-Fender | mvanbaak: Any caveats? |
| 18:21.39 | mvanbaak | [TK]D-Fender: as always: battery life |
| 18:21.40 | neverblue2 | does anyone manage a call centre that uses VOIP, I want to compare setups/ask questions ? |
| 18:21.58 | mvanbaak | it's a phone, it should run at least a week on a battery with normal usage |
| 18:22.17 | [TK]D-Fender | mvanbaak: lets say only 20-30 min voice per day, no PDA usage (slow week), how long would you estimate battery life? |
| 18:22.19 | mvanbaak | but mine runs 3 days with 10 calls a day and 2 hours wireless internet usage |
| 18:22.37 | neverblue2 | if you do, then please drop me a private message, thanks ! :) |
| 18:22.45 | [TK]D-Fender | mvanbaak: ok, that one covers it pretty well I guess |
| 18:22.53 | hmmhesays | no caps with qwest |
| 18:23.05 | mvanbaak | [TK]D-Fender: sjphone works pretty neat on it ;) |
| 18:23.18 | mvanbaak | I use this phone as portable at home with my asterisk setup |
| 18:23.32 | [TK]D-Fender | mvanbaak: they are now discontinued on my carrier but I can get used and am considering it.... |
| 18:23.56 | [TK]D-Fender | mvanbaak: of the WM6 IE / WMP : anything better about those? |
| 18:24.08 | [TK]D-Fender | mvanbaak: usable alternatives? |
| 18:24.22 | mvanbaak | I tried familiar linux, but that was no success |
| 18:24.34 | DarKnesS_WolF | [TK]D-Fender: sorry for delay i'm just making sure that the power is connected to the card |
| 18:24.43 | mvanbaak | all the browsers I tried were slow or non-free |
| 18:24.52 | mvanbaak | the opera one is nice, but not free |
| 18:24.58 | hmmhesays | opera is not free? |
| 18:25.09 | mvanbaak | minimo is freaking slow, even on the tytn (new HTC model) |
| 18:25.17 | mvanbaak | hmmhesays: not the one for mobile devices |
| 18:25.27 | mvanbaak | the one for mobile devices that uses opera's proxy is free |
| 18:25.38 | [TK]D-Fender | mvanbaak: ok, last test i think I'll have to to up close & personal is a couple of PDF's |
| 18:25.43 | mvanbaak | but the one that runs on the mobile device and goes directly to the webpage is not free |
| 18:25.54 | hmmhesays | gotcha |
| 18:26.01 | mvanbaak | [TK]D-Fender: I never looked at a pdf on the device |
| 18:26.16 | mvanbaak | I use it for making phonecalls and ssh sessions most of the time |
| 18:26.22 | mvanbaak | and sjphone of course |
| 18:27.06 | [TK]D-Fender | mvanbaak: SSH would be nice too... PDF is because I'd insist on carrying some maps with me, and TFOT to boot :) |
| 18:28.14 | mvanbaak | [TK]D-Fender: I bought a thinkpad and a 12' ibook. I use those for that ;) |
| 18:28.56 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 18:29.07 | r3zon8 | what handsets do most use with asterisk? |
| 18:29.22 | mvanbaak | cisco 7960 |
| 18:29.23 | [TK]D-Fender | mvanbaak: Mine is for making sure I don't get lost around town where I don't want to lug a briefcase with me for have a laptop :) |
| 18:29.35 | [TK]D-Fender | Polycom > All |
| 18:29.40 | mvanbaak | [TK]D-Fender: ah, I use tomtom for that on my HTC |
| 18:29.49 | mvanbaak | food, brb |
| 18:29.50 | kiscokid | Polycom 430 |
| 18:29.54 | DarKnesS_WolF | [TK]D-Fender: http://pastebin.com/m1682bce7 |
| 18:30.52 | [TK]D-Fender | Polycom IP430/501 are suggestable in rare scenarios. |
| 18:31.06 | r3zon8 | rare meaning? |
| 18:31.22 | [TK]D-Fender | r3zon8: very few cases. Tell us what your needs are. |
| 18:31.31 | r3zon8 | home office |
| 18:31.33 | [TK]D-Fender | DarKnesS_WolF: you have no loadzone in zaptel.... |
| 18:31.37 | r3zon8 | 2 maybe 3 handsets |
| 18:31.39 | [TK]D-Fender | r3zon8: single phone? |
| 18:31.49 | r3zon8 | 2 line |
| 18:32.13 | r3zon8 | im not sure whether to go with voicepulse/etc or buy my own hardware |
| 18:32.16 | [TK]D-Fender | r3zon8: do you need to plug it in-lie with a PC or can each phone have its own jack to your switch? |
| 18:32.30 | r3zon8 | it just seems those pots lines are more expensive from local carrier than these online places |
| 18:32.37 | r3zon8 | each phone have jack |
| 18:32.38 | frenzy | is there an open source multi-tenat pbx manager available for asterisk? |
| 18:33.12 | hmmhesays | what a fantastically vague question |
| 18:33.16 | [TK]D-Fender | r3zon8: Betting that you don't have PoE I'd suggest Polycom IP 320's + their Power Brick. |
| 18:33.30 | frenzy | LOL |
| 18:33.51 | r3zon8 | you guessed right, no PoE..thanks |
| 18:34.02 | r3zon8 | what do you suggest i do about lines? |
| 18:34.32 | frenzy | what i meant was a front end like freepbx but that which supposers multi-tenant |
| 18:34.32 | *** join/#asterisk dug (n=chatzill@c-76-102-23-25.hsd1.ca.comcast.net) |
| 18:34.34 | *** part/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 18:34.35 | [TK]D-Fender | r3zon8: http://www.telephonydepot.com/Polycom_s/25.htm |
| 18:34.39 | DarKnesS_WolF | [TK]D-Fender: damn it how can i forget it :-s |
| 18:34.39 | DarKnesS_WolF | thx dude |
| 18:34.52 | [TK]D-Fender | r3zon8: 87.50 + 17.95 ea |
| 18:34.59 | DarKnesS_WolF | now to the billing issue ;-) |
| 18:35.12 | [TK]D-Fender | DarKnesS_WolF: I accept paypal :) |
| 18:35.30 | coppice | I accept deeds to large properties |
| 18:35.41 | hmmhesays | and for the ones coppice doesn't want... |
| 18:35.42 | [TK]D-Fender | r3zon8: Lines really depends on your needs & budget |
| 18:35.56 | dug | I have two extensions both show up in sip show peers as ok but when I call the extension it doesnt ring and goes straight to voicemail |
| 18:36.02 | r3zon8 | feel that i need 2 lines, tight budget :) |
| 18:36.09 | DarKnesS_WolF | [TK]D-Fender: paypal forbidding in egypt :P |
| 18:36.22 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 18:36.32 | [TK]D-Fender | DarKnesS_WolF: Wire transfer it is :) I did that with 1 internation client of mine.... |
| 18:36.58 | [TK]D-Fender | r3zon8: Ok, rethinking time maybe... have you considered ATA's ? |
| 18:37.38 | r3zon8 | ata's? |
| 18:38.05 | DarKnesS_WolF | dose a2billing supports zaptel :-s? |
| 18:38.08 | [TK]D-Fender | ~ata |
| 18:38.08 | jbot | i guess ata is Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 18:38.14 | r3zon8 | analog terminal adapters? |
| 18:38.29 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 18:38.30 | [TK]D-Fender | r3zon8: little box that'll let you use a normal analog phone as a SIP phone |
| 18:38.40 | [TK]D-Fender | r3zon8: very cost effective. |
| 18:38.48 | [TK]D-Fender | r3zon8: about $35/port |
| 18:39.03 | [TK]D-Fender | r3zon8: provided you've got the phones to plug into them :) |
| 18:39.03 | r3zon8 | i see, so i can use 'regular' analog phones |
| 18:39.08 | [TK]D-Fender | r3zon8: yup |
| 18:39.16 | r3zon8 | yea i got those |
| 18:39.21 | r3zon8 | hmm |
| 18:39.31 | [TK]D-Fender | r3zon8: http://www.telephonydepot.com/Linksys_ATA_s/33.htm |
| 18:39.37 | [TK]D-Fender | r3zon8: SPA-2102 recommended |
| 18:40.08 | [TK]D-Fender | r3zon8: Honestly I rarely need more than they offer |
| 18:41.03 | jm|laptop | not that I need the router side of things |
| 18:41.28 | r3zon8 | ok, this supports 2 handsets |
| 18:41.29 | _Sam-- | [TK]D-Fender : thanks again for your help. just a quick status report: System uptime: 1 week, 23 hours, 1 minute, 35 second |
| 18:41.39 | _Sam-- | alls well. |
| 18:41.44 | [TK]D-Fender | _Sam--: Glad to hear.... |
| 18:41.44 | jm|laptop | [TK]D-Fender: here's a nice vague question for you - with those Linksys SPAs, why might incoming calls be cut off (by the Linksys) when a DECT phone is connected to its FSO? |
| 18:42.00 | jm|laptop | I can't remember if it's exactly the same time - but I worry it might be 60 secs into a call |
| 18:42.16 | jm|laptop | and afaik I have turned off all silence detection |
| 18:42.23 | [TK]D-Fender | jm|laptop: and the dect based uses analog? |
| 18:42.28 | jm|laptop | iirc all I have set is the disconnect /tone/ |
| 18:42.34 | jm|laptop | [TK]D-Fender: yes, analogue |
| 18:42.58 | [TK]D-Fender | jm|laptop: You want the dect to auto-hangup on reorder tone basically? |
| 18:43.13 | jm|laptop | [TK]D-Fender: er ... do I? |
| 18:43.27 | [TK]D-Fender | jm|laptop: basically so whent he call ends the DECT will disconnect. Correct? |
| 18:43.29 | jm|laptop | [TK]D-Fender: I want the Linksys to not hang me up mid-incoming-call ! |
| 18:43.45 | jm|laptop | [TK]D-Fender: no; it's disconnecting when it shouldn't be |
| 18:43.57 | [TK]D-Fender | jm|laptop: first prove that its the Linksys at fault. Plug in a normal phone and test |
| 18:44.02 | jm|laptop | Detect CPC [no] Detect Polarity Reversal [no] |
| 18:44.20 | jm|laptop | [TK]D-Fender: GF won't let me :( heh under the thumb |
| 18:44.34 | [TK]D-Fender | jm|laptop: sure ask for help then tie my hands..... |
| 18:44.40 | jm|laptop | Detect PSTN long silence [no] Detect VOIP long silence [no] |
| 18:44.49 | [TK]D-Fender | jm|laptop: you need relationship help more apparently :p |
| 18:44.56 | jm|laptop | hehehe! |
| 18:45.11 | [TK]D-Fender | jm|laptop: But I've had my "intervention of the week" already, so you'll have to take a number.... |
| 18:45.14 | jm|laptop | I was going to say "If I turned on Long Silence detection I'd *always* get cut off to the GF" ;) |
| 18:45.29 | jm|laptop | Detect Disconnect Tone [yes] |
| 18:45.42 | jm|laptop | Disconnect tone 400@-30,400@-30;2(*/0/1+2) wtf is that?! |
| 18:45.53 | [TK]D-Fender | jm|laptop: My guess : trouble |
| 18:46.40 | jm|laptop | [TK]D-Fender: when it annoys GF enough she'll let me try with a non-DECT |
| 18:46.46 | jm|laptop | just hoped you'd have heard of this before |
| 18:46.51 | jm|laptop | [TK]D-Fender === Google |
| 18:48.10 | *** join/#asterisk elriah (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 18:48.24 | elriah | Hi all. Is res_speech.so in asterisk 1.2.24? I've searched high and low and can't find it. |
| 18:49.08 | *** join/#asterisk blitz[astricon] (n=blitz[as@65.116.224.30) |
| 18:49.22 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 18:50.00 | *** join/#asterisk afrosheen (n=cj@207.71.49.137) |
| 18:50.14 | r3zon8 | what are other carriers besides Voicepulse/iaxtel..? |
| 18:50.25 | elriah | vitelity, les.net |
| 18:50.30 | elriah | heavylogic |
| 18:50.42 | elriah | voicepulse sux |
| 18:50.43 | afrosheen | anyone know why zap show channels isn't showing me all active PRI channels? |
| 18:54.19 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-3.dllstx.fios.verizon.net) |
| 18:55.05 | elriah | Hi all. Is res_speech.so in asterisk 1.2.24? I've searched high and low and can't find it. |
| 18:57.00 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 18:57.06 | afrosheen | it's part of 1.4.11 |
| 18:57.17 | elriah | hrm... |
| 18:57.27 | afrosheen | of course there may be dependencies you didn't satisfy which caused it to skip building that.. |
| 18:57.29 | elriah | Is 1.4.11 ready for prime time? (high volume) |
| 18:57.30 | ManxPower | elriah: why do you think it is part of 1.2.x? |
| 18:57.46 | elriah | afrosheen: I can't even find it in the 1.2.24 source tarball ... |
| 18:57.48 | elriah | or addons |
| 18:58.11 | afrosheen | elriah, it may not be a part of the 1.2.x branch |
| 18:58.16 | dug | I have am trying to test an extension, it shows the extension in sip show peers as status ok but I cannot call the extension? it goes straight to voice mail? |
| 18:58.27 | elriah | ManxPower: I don't, I wan to implement lumenvox on 1.2 and I was reading through the docs, doing my due dillegence, and couldn't find the module.. hit google, asterisk.org, no luck... |
| 18:58.39 | elriah | So I figured I'd ask because sometimes I miss the obvious... |
| 18:59.07 | elriah | Apparently, it's a 1.4 only module then.. |
| 18:59.48 | ManxPower | dug: you do not call extensions. You call devices |
| 19:00.05 | ManxPower | sip show peers shows devices, not extensions. |
| 19:00.15 | ManxPower | dug: and what MESSAGES do you get on the console. |
| 19:00.30 | [TK]D-Fender | dug: And your discription is worthless without seeing the CLI output at high verbose & SIP debug enabled. |
| 19:00.33 | [TK]D-Fender | ~pb |
| 19:00.34 | jbot | A Pastebin is a place to paste your stuff 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 |
| 19:00.35 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^ |
| 19:00.43 | afrosheen | elriah, if you're trying to do lumenvox on 1.2.x, I know it can be done because Digium sells the business edition with that setup |
| 19:00.57 | r3zon8 | whats a good carrier to use for home/residential service? heavylogic seems mid-large business.. |
| 19:01.29 | elriah | afrosheen: Maybe it comes with the lumenvox tarball... |
| 19:01.55 | elriah | r3zon8: If you just want a sip trunk to play with, use les.net, easy and pay as you go. Quality is good. |
| 19:02.23 | elriah | afrosheen: Thanks, at least I know it's possible ... |
| 19:03.33 | *** join/#asterisk Defraz (n=t0tal@65.121.20.50) |
| 19:03.35 | dug | When I call the device 100 from the device 101 I get http://pastebin.com/m690891d5 |
| 19:03.52 | blq | hi, is it possible to connect a analog-phone via an analog-modem to a asterisk server? |
| 19:04.00 | elriah | blq: freepbx |
| 19:04.02 | elriah | ? |
| 19:04.17 | ManxPower | dug: We do not support trixbox here. |
| 19:04.27 | elriah | ahh.. tribox |
| 19:04.28 | blq | elriah: freepbx? sry I don't know that :/ |
| 19:04.31 | ManxPower | we cannot help you with your problem because the issue is with trixbox, not Asterisk |
| 19:04.42 | elriah | blq: try #tribox |
| 19:04.43 | dug | ManxPower: I know... its not trixbox... its amp BTW |
| 19:05.02 | dug | like a extension isnt a device ;) |
| 19:05.04 | elriah | blq: oops, I meant dug |
| 19:05.05 | ManxPower | dug: it's still a gui and we still can't help you and it is still a problem with the gui scripts |
| 19:05.08 | afrosheen | there's a whole channel dedicated to freepbx/amp/trix |
| 19:05.27 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 19:05.35 | ManxPower | dug: Asterisk is not even trying to call device 100 because the scripts you are using are refusing to even try dialing it. |
| 19:05.49 | *** join/#asterisk saint_ (n=saint@c-69-242-118-124.hsd1.nj.comcast.net) |
| 19:05.52 | saint_ | hi all... |
| 19:06.04 | saint_ | anyone connected an asterisk to an alcatel pbx, using sip or ip trunk ? |
| 19:06.19 | dug | ManxPower: got it |
| 19:06.40 | blq | [21:03:40] <elriah> blq: freepbx << was this also for dug? |
| 19:06.46 | ManxPower | saint_: Asteirsk does not support ip trunks |
| 19:06.50 | elriah | Yep, sorry. |
| 19:07.03 | saint_ | ok, i ll give it a shot at SIP trunks then ... |
| 19:07.09 | elriah | blq: Did you get what you needed answered? |
| 19:07.13 | ManxPower | Asterisk also does not support SIP trunks. |
| 19:07.13 | saint_ | i am trying to install it on a brand new centos 5 |
| 19:07.18 | saint_ | hu ? |
| 19:07.23 | saint_ | r u kidding ? |
| 19:07.26 | ManxPower | perhaps you mean "sip connection" or "sip peer" or "sip device" |
| 19:07.35 | ixx | How do you detect when someone answers a call if you have been transferred. Specifically I am wanting to dial a number... wait some period of time for the other end to give me their IVR menu.. Send DTMF for an extension.. |
| 19:07.46 | ManxPower | saint_: since there is no such thing as a "sip trunk" don't be suprized that astrerisk does not support it. |
| 19:08.00 | ixx | Then wait to send more audio until someone answers at the extension |
| 19:08.27 | *** part/#asterisk dug (n=chatzill@c-76-102-23-25.hsd1.ca.comcast.net) |
| 19:08.33 | saint_ | ok.. well.. it says to install the GUI to run SVN. what's SVN ? |
| 19:08.34 | elriah | ixx: A receptionist? |
| 19:08.46 | elriah | saint_: source code repository |
| 19:08.56 | elriah | saint_: svn = subversion |
| 19:09.00 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 19:09.57 | *** join/#asterisk mindCrime__ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 19:09.59 | ManxPower | what says use a gui? |
| 19:10.11 | hrmphh | hey |
| 19:10.14 | saint_ | oh |
| 19:10.14 | hrmphh | emergency |
| 19:10.19 | hrmphh | production system down :( |
| 19:10.28 | saint_ | elriah, so i can install svn with yum install subversion then ? |
| 19:10.31 | elriah | hrmphh: Why you wasting time in here them? |
| 19:10.32 | elriah | lol |
| 19:10.38 | elriah | Thanks for letting us know, though. |
| 19:10.42 | hrmphh | [Sep 28 12:10:35] WARNING[2725]: app_dial.c:1099 dial_exec_full: Unable to create channel of type 'Zap' (cause 66 - Channel not implemented) |
| 19:10.47 | hrmphh | need help =] |
| 19:10.55 | mvanbaak | I'm off for today |
| 19:10.56 | elriah | hrmphh: Did you restart asterisk? |
| 19:10.56 | mvanbaak | latero |
| 19:10.58 | hrmphh | yes |
| 19:11.01 | hrmphh | and the machine itself |
| 19:11.11 | elriah | hrmphh: Want me to take a look? |
| 19:11.16 | blq | elriah: not really - is it possible to connect a analog-phone via an analog-modem to a asterisk server? |
| 19:11.33 | hrmphh | hrm the fact that i cant 'show zap' |
| 19:11.35 | hrmphh | is prob not good |
| 19:12.03 | afrosheen | show modules then see if zap is loaded |
| 19:12.04 | elriah | blq: No. |
| 19:12.09 | ManxPower | hrmphh: you installed asterisk and asterisk did not detect zaptel installed and so did not build zap support |
| 19:12.13 | afrosheen | and lsmod | grep zap while you're at it |
| 19:12.17 | ManxPower | meetme is prolly not there either |
| 19:12.23 | elriah | ManxPower: He said it was running zap before ... |
| 19:12.30 | ManxPower | elriah: he is confused. |
| 19:12.48 | elriah | ManxPower: Ahh, then I bow-out to your expertise . |
| 19:12.54 | ManxPower | hrmphh: can you "load chan_zap.so" |
| 19:13.01 | outtolunc | that or he's misconfig'd and it unloaded chan_zap |
| 19:13.23 | hrmphh | <PROTECTED> |
| 19:13.26 | elriah | hrmphh: can you 'local chan_zap.so' ? |
| 19:13.31 | hrmphh | ok that works |
| 19:13.33 | elriah | local = locate |
| 19:13.34 | hrmphh | now i can call |
| 19:13.35 | hrmphh | after load |
| 19:13.41 | afrosheen | yeah |
| 19:13.44 | hrmphh | weird |
| 19:13.45 | ManxPower | hrmphh: look at /etc/asterisk/modules.conf |
| 19:13.46 | hrmphh | why wouldnt it load |
| 19:13.47 | hrmphh | auto |
| 19:13.48 | afrosheen | so it's not loading the zap module |
| 19:13.57 | afrosheen | that was easy TM |
| 19:14.05 | hrmphh | it says autoload |
| 19:14.20 | booray | alright... so... patlooptest getting lots of errors could be a problem, right? |
| 19:14.24 | afrosheen | cat /var/log/asterisk/full | grep error ? |
| 19:14.30 | *** join/#asterisk kpreid (n=kpreid@cpe-24-59-154-165.twcny.res.rr.com) |
| 19:14.34 | afrosheen | booray, pri hardware? |
| 19:14.47 | hrmphh | there is no full |
| 19:14.51 | booray | afrosheen: loopback plug on a te120p |
| 19:15.01 | elriah | blq: Try a T100P, you can find them all day long on ebay for < $10. I have a bunch of them just sitting here, they work great for learning asterisk and small home apps... |
| 19:15.14 | afrosheen | hrmphh, weird..well you should have a log somewhere on your server for asterisk |
| 19:15.57 | afrosheen | booray, my sangoma a102d is kinda flaky too, I need to patlooptest it once I build a cable. Channel 1 was noisy, now channel 2 is, but only on the far side of the conversation |
| 19:15.57 | blq | elriah: thanks! |
| 19:16.36 | elriah | blq: If you send me a SASE I'll send you one of these cards. |
| 19:17.09 | blq | elriah: I think that won't be that easy since I'm from germany :( |
| 19:17.19 | blq | elriah: but thankyou ! |
| 19:17.21 | elriah | Ahh.. include lots of stamps ;) |
| 19:18.18 | blq | elriah: hm.. I could only find some "TE100P" fpr 189,99$ on ebay -but I'm still searching |
| 19:18.38 | afrosheen | any reason why on my PRI, I have a channel that is permanently "resetting" |
| 19:18.41 | [TK]D-Fender | elriah: I believe you're thinking of the *X*100P... |
| 19:18.51 | elriah | I'm sure that's an OK card, but for your purposes, X100P. |
| 19:19.02 | elriah | What [TK]D-Fender said. |
| 19:19.04 | hrmphh | i do have a messages log afro |
| 19:19.44 | De_Mon | I have compiled asterisk 1.4 with imap support but don't want to use it at this time.. Can I disable the IMAP parts? |
| 19:20.12 | booray | ah, forgot clear=1-24 |
| 19:20.55 | elriah | Well, I've done enough damage to the Asterisk community today, time for the weekend, later all. |
| 19:22.02 | blq | elriah: looks like noone in Germany is using those cards - I found some from US some from Great Britain but none from Germany :/ is there a alternative card? |
| 19:24.13 | blq | [TK]D-Fender: looks like noone in Germany is using those cards - I found some from US some from Great Britain but none from Germany :/ is there a alternative card? |
| 19:25.17 | hrmphh | anyone know if you have a digium 4 port analog card AND a sangoma a101d card in the same box, which will take span1/2? |
| 19:25.41 | *** join/#asterisk brc_ (n=brc__@pdpc/supporter/basic/brc) |
| 19:25.47 | ixx | elriah, yes... dropped directly into a receptionist (menu system) |
| 19:25.48 | rob0 | Probably depends on PCI bus order. |
| 19:26.21 | hrmphh | k |
| 19:26.25 | hrmphh | so if the analog comes first |
| 19:26.27 | hrmphh | does it use all 24 chans? |
| 19:26.36 | ixx | elriah, need to wait a bit then send DTMF.. (Dial with D() option works)... then wait for final answer before playing any audio |
| 19:27.17 | jm|laptop | my X100P scared me |
| 19:27.23 | hrmphh | http://pastebin.com/m67c8be5b |
| 19:27.36 | hrmphh | theres the cat /proc/zaptel/* output |
| 19:27.43 | hrmphh | so looks like span2 is the t1 card? |
| 19:27.56 | *** join/#asterisk bkruse_home (n=bkruse@69.73.127.92) |
| 19:27.57 | hrmphh | which bchan should i start with? |
| 19:29.18 | [TK]D-Fender | bchan=5-27 dchan=28 |
| 19:30.07 | hrmphh | yeah i tried that |
| 19:30.12 | hrmphh | Level-3 said go fuck yourself :) |
| 19:31.00 | hrmphh | wouldnt it be 5-28 and dchan=29? |
| 19:31.28 | hrmphh | they said chan 1-12 is voice, 13-23 is data, and 24 is d |
| 19:31.37 | *** join/#asterisk Op3r (n=Op3r@210.4.60.88) |
| 19:31.37 | hrmphh | data portion is up, and handed off from iad |
| 19:32.24 | hrmphh | so voice should be bchan=5-16 dchan=28? |
| 19:33.00 | *** join/#asterisk drwelby (n=mpfister@mail.enplan.com) |
| 19:33.46 | hrmphh | do i need to put a span statement for the analog or would this suffice: http://pastebin.com/m717c5b8b |
| 19:35.03 | [TK]D-Fender | hrmphh: Should be span 1 I believe |
| 19:36.57 | hrmphh | what should be? |
| 19:37.01 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 19:37.13 | hrmphh | its span=2 because of the order in /proc/zaptel |
| 19:37.16 | bkruse_home | hrmphh: cat /proc/zap/chan_nu |
| 19:37.23 | hrmphh | see http://pastebin.com/m67c8be5b |
| 19:37.33 | bkruse_home | hrmphh: You will be able to see what is an analog/digital and what card |
| 19:37.49 | hrmphh | there is no /proc/zap/chan_nu |
| 19:37.58 | hrmphh | thereis /proc/zaptel/1 and 2 |
| 19:38.03 | hrmphh | see my pastebin |
| 19:38.24 | hrmphh | hmm i have no 'pri' in asterisk |
| 19:38.35 | hrmphh | i added libpri from apt, but maybe i need to load? |
| 19:38.53 | ThoMe | how i can set in my "queue" without music ? |
| 19:38.55 | [TK]D-Fender | hrmphh: Screw packaging recompile everything from scratch |
| 19:38.58 | ThoMe | only "kling"... and wait... |
| 19:38.59 | ThoMe | ? |
| 19:39.30 | [TK]D-Fender | ThoMe: Don't set a MoH class in you queue definition, or set it to a siltent one. |
| 19:40.06 | outtolunc | and don't forget to do the same with the agents.conf if using agentlogin/agentcallbacklogin |
| 19:40.15 | hrmphh | [Sep 28 12:39:54] ERROR[2977]: chan_zap.c:10789 process_zap: Unknown signalling method 'pri_cpe' |
| 19:40.20 | hrmphh | ah hah! |
| 19:40.28 | ThoMe | outtolunc: hmm. have no agents. only members |
| 19:40.36 | ThoMe | member => SIP/8 |
| 19:40.36 | ThoMe | member => SIP/9 |
| 19:40.37 | ThoMe | etc.. ? |
| 19:40.40 | outtolunc | keyword: if |
| 19:41.03 | ThoMe | ah :-) |
| 19:41.07 | ThoMe | outtolunc: hihi |
| 19:41.12 | hrmphh | does that indicate i dont have a correct module installed? |
| 19:41.15 | hrmphh | the unknown signalling? |
| 19:41.34 | [TK]D-Fender | hrmphh: Means zaptel has no clue about PRI. Go recompile everyhitng. |
| 19:41.39 | ThoMe | [TK]D-Fender: u mean musicclass = default ? |
| 19:41.46 | [TK]D-Fender | ThoMe: duh |
| 19:41.51 | hrmphh | yeah |
| 19:41.54 | hrmphh | i need to get libpri |
| 19:42.01 | hrmphh | and compile in |
| 19:42.04 | ThoMe | [TK]D-Fender: hm? |
| 19:42.09 | hrmphh | i want libpri 1.4.1? |
| 19:44.40 | *** join/#asterisk tsurko (n=tsurko@213.91.216.130) |
| 19:44.44 | ixx | crap.. just noticed he left... |
| 19:45.29 | ixx | so anyone else have any idea on how to deal with a receptionist/menu on the callee end... sending DTMF for an extension.. then when is it answered again? |
| 19:46.32 | *** join/#asterisk tripps (n=ss@66.60.235.100) |
| 19:46.34 | *** part/#asterisk tripps (n=ss@66.60.235.100) |
| 19:46.49 | *** join/#asterisk tripps (n=ss@66.60.235.100) |
| 19:47.22 | *** join/#asterisk FinboySlick (n=Miranda@207.134.8.4) |
| 19:47.23 | *** join/#asterisk DeeJayTwo (n=deejay2@office.abi.ca) |
| 19:47.39 | *** join/#asterisk frenzy_ (n=frenzy@unaffiliated/frenzy) |
| 19:47.42 | DeeJayTwo | we're using polycom soundpoint ip phones with asterisk... |
| 19:47.55 | DeeJayTwo | it looks like the polycom doesn't send a ptime in the sdp part of the invite... |
| 19:48.01 | *** join/#asterisk tc3driver-nii (n=chatzill@adsl-75-49-241-185.dsl.irvnca.sbcglobal.net) |
| 19:48.06 | FinboySlick | [TK]D-Fender: Greetings. |
| 19:48.09 | DeeJayTwo | it appears in the web browser...and in the sip configuration file.. |
| 19:48.10 | hrmphh | do i compile libpri directly or as part of asterisk? |
| 19:48.21 | hrmphh | there doesnt seem to be a ./config or anything |
| 19:49.05 | hrmphh | nm 'make clean && make' |
| 19:49.10 | [TK]D-Fender | ixx: .... huh? |
| 19:49.27 | [TK]D-Fender | FinboySlick: huh? |
| 19:49.31 | FinboySlick | I'm having problems with zaptel fax detection (not nvfaxdetect) on a Wildcard TDM800P. Is it even possible? |
| 19:50.01 | FinboySlick | [TK]D-Fender: Heh, I've bugged you often enough, I figured I'd greet you first. |
| 19:50.36 | [TK]D-Fender | FinboySlick: Sorry, intended to write "y0" and was still in shock from other requests :) |
| 19:50.53 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:51.05 | hmmhesays | haha |
| 19:55.08 | tc3driver-nii | Is there a different use of 'SetVar' in asterisk 1.4? |
| 19:55.31 | [TK]D-Fender | tc3driver-nii: SetVar = GONE. You should have been using Set since 1.2 |
| 19:57.01 | tc3driver-nii | well that would explain why it is complaining about setvar... thanks.. |
| 20:01.47 | saint_ | hey |
| 20:01.52 | saint_ | anyone installed the asterisk GUI ? |
| 20:02.16 | FinboySlick | [TK]D-Fender: With considerable thanks for helping me out in my Sangoma days. You know if there are faxdetect issues on TDM800P cards? |
| 20:02.30 | tc3driver-nii | now if I can just figure out why my AA quit working I'll be good to go |
| 20:07.03 | ThoMe | spricht hier wer deutsch? |
| 20:07.28 | FinboySlick | nein ;) |
| 20:07.39 | ThoMe | FinboySlick: schade :-) danke trotzdem :-) |
| 20:08.01 | ThoMe | FinboySlick: mit queues kennst du dich ned aus hm? |
| 20:08.22 | FinboySlick | ThoMe: I was speaking the truth :P I don't speak deutsch... |
| 20:08.44 | [TK]D-Fender | FinboySlick: nothing i'm aware of... |
| 20:09.05 | *** join/#asterisk [hC] (n=hardcore@wsip-70-184-124-51.ph.ph.cox.net) |
| 20:09.20 | ThoMe | how i can play "you are next.. " if i have a new call in my queue? |
| 20:09.46 | FinboySlick | [TK]D-Fender: Given that it appears just as deaf as my Sangoma, I'm guessing that something else may be the issue. Outside of gain and relaxdtmf, I'm totally stumped :( |
| 20:10.00 | *** join/#asterisk tomcontr3 (n=tomcontr@231-161-28.dial.terra.cl) |
| 20:10.05 | tomcontr3 | hi does anyone knows where can I find an IP-Phone for an Operator.... I want something like the Flash Operator but not in Software... I need hardware something for a Secretary |
| 20:10.21 | [TK]D-Fender | ThoMe: Go read the sample configs |
| 20:10.40 | [TK]D-Fender | tomcontr3: How many phones withh this person have to watch? |
| 20:10.46 | FinboySlick | tomcontr3: If you want to go the cheap way, there's an add-on for GrandStream GXP2000 that seems to do that. |
| 20:10.51 | [TK]D-Fender | ~gs |
| 20:10.52 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 20:10.56 | [TK]D-Fender | ~grandstream |
| 20:10.57 | jbot | grandstream is probably the Yugo of VoIP hardware. Run. Run away now. |
| 20:11.16 | [TK]D-Fender | ~cheap |
| 20:11.17 | jbot | somebody said cheap was a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 20:11.25 | [TK]D-Fender | FinboySlick: .... pwned |
| 20:11.30 | tomcontr3 | 30-50 |
| 20:12.07 | [TK]D-Fender | tomcontr3: Guess its the Aastra 57i + LCD console |
| 20:12.14 | tc3driver-nii | I like the Aastra series of phones personally, they do have consol extensions |
| 20:12.15 | tomcontr3 | it can bee less too, first I need to find out if this exist for Ip telephony |
| 20:12.42 | FinboySlick | Pff, we have six here and it seems that it's only the shiny awesome hardware giving us trouble. Those cheap bastards worked without a problem from the get go. |
| 20:12.55 | [TK]D-Fender | tomcontr3: http://www.telephonydepot.com/Aastra_s/44.htm |
| 20:12.57 | tomcontr3 | mm it looks nice... |
| 20:13.20 | [TK]D-Fender | tomcontr3: http://www.telephonydepot.com/product_p/105-057-560m.htm |
| 20:14.22 | *** join/#asterisk aninoSAdilim (n=a@58.69.243.203) |
| 20:20.11 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 20:21.36 | afrosheen | FinboySlick, for us, it's polycom or nothing, and they haven't failed us yet |
| 20:22.27 | *** join/#asterisk kiscokid (n=ron@208.106.35.66) |
| 20:27.44 | *** join/#asterisk RipeR-81 (n=ircap8@190.53.33.3) |
| 20:28.12 | RipeR-81 | good afternoon anyone |
| 20:28.29 | RipeR-81 | im having problem setting a conexion between an asterisk server 1.2 and a cisco 2801 |
| 20:28.45 | RipeR-81 | the call is droppign when we pickup our cisco ip phone 7941 |
| 20:29.23 | *** join/#asterisk jozu (i=torrent@84.79.51.163) |
| 20:29.29 | jozu | hi |
| 20:29.32 | tc3driver-nii | I am ready to pull my hair out... I'll start with this... where is the place you can post code snippets? |
| 20:30.08 | booray | ~pb |
| 20:30.09 | jbot | A Pastebin is a place to paste your stuff 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 |
| 20:31.01 | tc3driver-nii | thank you. |
| 20:32.34 | tc3driver-nii | ok, now I have pasted a portion of my extensions.conf file, the problem is that it is not registering key presses for IVR. |
| 20:32.35 | tc3driver-nii | http://pastebin.com/d1ea4bd11 |
| 20:34.09 | Strom_M | tc3driver-nii: let me guess: you're usig SIP |
| 20:34.20 | tc3driver-nii | yes. |
| 20:34.33 | Strom_M | can you talk both ways across that connection? |
| 20:34.41 | *** join/#asterisk putnopvut (n=putnopvu@wsip-70-184-124-51.ph.ph.cox.net) |
| 20:35.00 | tc3driver-nii | yes. |
| 20:35.11 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 20:35.20 | Strom_M | tc3driver: check your dtmfmode settings in sip.conf for that peer |
| 20:35.41 | *** part/#asterisk Aeudian (n=Aeudian@74.92.134.190) |
| 20:36.23 | tc3driver-nii | no setting. |
| 20:37.07 | jozu | someone can helpme, please? |
| 20:37.23 | RipeR-81 | ? |
| 20:37.28 | Strom_M | well, now would be a good time to figure out what your provider is expecting you to set that to |
| 20:37.29 | ixx | tkd-fender, i need to call a number which goes to a menu system where you enter a extension to reach someone.. then I send DTMF for that extension... |
| 20:37.48 | *** join/#asterisk blackhole (n=Mishu@unaffiliated/blackhole) |
| 20:37.51 | ixx | then i need to wait until someone picks up at that extension before playing audio... |
| 20:37.57 | jozu | i have a gsm gateway in a sip extension (300), the asterisk server as registered into voip sip provider |
| 20:38.27 | blackhole | Is there some way i can configure sip to talk to Skype Or Asterisk to talk to skype using sip |
| 20:38.41 | Strom_M | blackhole: no |
| 20:38.43 | jozu | i want to put a IVR into the 300 sip extension (idea gsm call --> ivr --> 1 (office) --> 2 (DISA?) |
| 20:38.50 | jozu | its possible? |
| 20:38.58 | Strom_M | ~skype |
| 20:38.59 | jbot | well, skype is stupid worthless junk. |
| 20:39.04 | *** join/#asterisk |omni| (n=rob@net82.allied-security.com) |
| 20:39.11 | blackhole | Strom_M, Why its not possible? |
| 20:39.39 | _x86_ | blackhole: it is possible, but you have to pay for a channel driver... google for chan_skype |
| 20:39.55 | Strom_M | and those channel drivers are all horrendous kludges |
| 20:40.26 | blackhole | _x86_, Okay Thanks But will i need to configure them with asterisk in some manner or just get them and install? |
| 20:41.08 | _x86_ | blackhole: RTFM ;) |
| 20:41.10 | _x86_ | blackhole: http://chanskype.com/ |
| 20:41.43 | _x86_ | blackhole: i've never had to buy it, so i have no idea the quality and/or installation procedure |
| 20:41.50 | _x86_ | blackhole: they do have a trial |
| 20:41.54 | blackhole | Hmm, Okay Thanks. |
| 20:42.11 | rob0 | Were they found guilty? |
| 20:42.16 | jozu | i put and inbound call route with the DID number of the mobile sim |
| 20:42.27 | jozu | destination IvR but, nothing |
| 20:42.45 | jozu | only hear the second tone dial |
| 20:43.04 | jozu | any idea? |
| 20:45.21 | lirakis | later everyone |
| 20:46.00 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 20:46.11 | *** join/#asterisk schattengolfer (n=fromm@fromm.omnis.com) |
| 20:46.55 | blackhole | _x86_, & Strom_C, What exactly would channel driver do internally any idea? Also any idea about any limit of calls that can be made on spot |
| 20:48.39 | afrosheen | anyone here have trouble with sangoma cards? |
| 20:48.44 | _x86_ | blackhole: dude, seriously, read the documentation kplzthx |
| 20:49.00 | _x86_ | afrosheen: i use sangoma cards exclusively |
| 20:49.24 | afrosheen | _x86_, we're seeing weirdness, like channels on our PRI going "bad" one by one, DID calls not hitting the phone system, etc. |
| 20:49.32 | afrosheen | I guess it's time to patlooptest it |
| 20:51.34 | _x86_ | afrosheen: I've got (7) asterisk boxes in my organization, (5) of them have an A20002D-x + A102D-x, (2) of them have an A104D-x + A102D-x, for a total of (7) A102D-x's, (2) A104D-x's, and (5) A20002D-x's |
| 20:51.59 | _x86_ | afrosheen: I've spent a _lot_ of money on sangoma cards ;) |
| 20:53.20 | denon | you have our condolences |
| 20:54.21 | *** join/#asterisk digime (n=digime@70.230.202.243) |
| 20:56.03 | RipeR-81 | have anyone had problems connecting asterisk to cisco ip phones thru cisco routers? |
| 20:58.56 | tzafrir_home | afrosheen, sorry for the silly question, but what exactly is patlooptest for? |
| 20:59.24 | tzafrir_home | (I know that there's a patlooptest.c in zaptel, but not sure what it is there for) |
| 20:59.48 | coppice | its a rather dumb loopback pattern test |
| 21:00.00 | Corydon76-dig | tzafrir_home: it's essentially testing whether your T1 loop is clean |
| 21:00.04 | coppice | instread of something sensible, like a BERT test |
| 21:00.16 | tzafrir_home | what type of loopback? to where? |
| 21:00.56 | Corydon76-dig | Anywhere, really |
| 21:01.04 | CCFL_Man2 | tzafrir_home: thanks again for the help setting up asterisk on my sun netra |
| 21:01.19 | Corydon76-dig | To whatever initiates the loopback |
| 21:03.58 | *** join/#asterisk Spida (n=timo@spinnennetz.org) |
| 21:04.06 | Spida | hi |
| 21:06.57 | afrosheen | tzafrir_home, it's an internal loopback test for the card in my case |
| 21:07.23 | afrosheen | tzafrir_home, I contacted Sangoma support before because channel 1 went static-y on us, he suggested I run that test and report back |
| 21:07.55 | tzafrir_home | coppice, speaking of smarter tests, I am still unable to get anything useful from sliptest |
| 21:07.57 | afrosheen | _x86_, ever had any fail, or specific channels on them fail? |
| 21:08.04 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 21:08.10 | _x86_ | afrosheen: never |
| 21:08.23 | _x86_ | afrosheen: i have seen channels fail on a crappy channel bank before |
| 21:08.33 | _x86_ | afrosheen: but you're doing PRI to the PSTN right? |
| 21:09.12 | afrosheen | _x86_, yeah, PRI from TWTC to the pstn |
| 21:09.24 | CCFL_Man2 | TWTC? |
| 21:09.39 | _x86_ | it's a carrier, iirc |
| 21:09.43 | CCFL_Man2 | ahh |
| 21:09.52 | CCFL_Man2 | not time warner i hope |
| 21:10.07 | _x86_ | afrosheen: might also contact the carrier to see if they can loop to the smartjack and run patterns |
| 21:10.13 | Corydon76-dig | Yes, that's Time Warner Tele Com |
| 21:10.29 | CCFL_Man2 | Corydon76-dig: they are Bell as well? |
| 21:10.30 | afrosheen | twtc has been great until this happened |
| 21:10.37 | _x86_ | heh... get your Animaniacs and PRI from the same company! |
| 21:10.49 | tc3driver-nii | Is there anything in the extensions.conf file that would be suppressing dtmf? When I call in asterisk from my Cell It doesn't respond to any key presses (or from a standard phone) if I call out to my cell phone, I cannot hear key presses from my cell phone or land line... |
| 21:10.55 | CCFL_Man2 | what more do you need |
| 21:11.00 | afrosheen | well actually TWTC isn't Time Warner, some kind of separate division |
| 21:11.12 | _x86_ | tc3driver-nii: it's not extensions.conf that has anything to do with that, it's sip.conf and/or iax.conf |
| 21:11.19 | *** join/#asterisk CVirus (n=GoD@196.205.191.113) |
| 21:11.24 | Corydon76-dig | CCFL_Man2: well, a CLEC |
| 21:11.25 | _x86_ | tc3driver-nii: your dtmf signalling method is out of whack |
| 21:11.40 | CCFL_Man2 | like the cable companies, do they charge the ass out of you? |
| 21:11.43 | CCFL_Man2 | Corydon76-dig: ahh |
| 21:11.44 | _x86_ | tc3driver-nii: progressinband=yes, dtmfmode=auto will usually fix it |
| 21:11.55 | afrosheen | CCFL_Man2, actually their rates are reasonable |
| 21:12.34 | CCFL_Man2 | afrosheen: they give you a full PRI? are they DIDs? |
| 21:13.07 | afrosheen | CCFL_Man2, yeah we have a full 23 channel pri with x200-x399 for our DIDs |
| 21:13.19 | afrosheen | so everyone here gets a direct line |
| 21:13.28 | *** join/#asterisk r0d3nt (n=astrutt@foster.stonedcoder.org) |
| 21:13.45 | CCFL_Man2 | nice |
| 21:13.56 | afrosheen | _x86_, the card in my server is an a102d but I'm only using port 1, what would it take for me to switch everything to port 2? |
| 21:14.37 | afrosheen | CCFL_Man2, like I said, we've been happy with them so far, service has been great but tech support could be faster |
| 21:14.49 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:15.04 | _x86_ | afrosheen: not a lot |
| 21:15.29 | CCFL_Man2 | afrosheen: did it go down? |
| 21:15.34 | CCFL_Man2 | loss of frame? |
| 21:15.36 | _x86_ | afrosheen: just delete wanpipe1.conf from /etc/wanpipe, and run wancfg again |
| 21:15.43 | afrosheen | _x86_, gotcha, thanks |
| 21:16.00 | _x86_ | afrosheen: if you set it up as the first channel group, you dont even have to touch zaptel.conf, zapata.conf, or extensions.conf for asterisk |
| 21:16.05 | afrosheen | luckily we just bought a new a101x to use for development, I can swap that in if worst comes to worst |
| 21:16.17 | _x86_ | hmm wait, you may still have to fudge with zaptel.conf |
| 21:16.19 | afrosheen | _x86_, yeah that's how I did it |
| 21:17.06 | afrosheen | CCFL_Man2, not loss of frame, just really, really bizarre stuff, like DIDs getting answered then receiving a hangup from TWTC, meanwhile the caller's phone shows "connecting" |
| 21:17.26 | afrosheen | that and noisy channels, but it's only noisy for the caller |
| 21:18.02 | CCFL_Man2 | afrosheen: they using hdsl? |
| 21:18.21 | afrosheen | CCFL_Man2, no clue, what's HDSL |
| 21:18.21 | CCFL_Man2 | the line is basically, just screwed up on their end? |
| 21:18.29 | afrosheen | I don't know at this point |
| 21:18.45 | CCFL_Man2 | modulation scheme to send the DS1 signal to you |
| 21:18.47 | afrosheen | it appears that way but they see nothing wrong |
| 21:19.00 | afrosheen | I've had this setup for months and never had trouble like this |
| 21:19.05 | CCFL_Man2 | have you checked your smart kack? |
| 21:19.09 | CCFL_Man2 | jack |
| 21:19.17 | afrosheen | nope, guess it wouldn't hurt |
| 21:19.29 | tzafrir_home | coppice, all I can see is that the ZT_IOMUX ioctl retuns "no events". Which is probably understandable, as nobody actually writes to the channel. |
| 21:19.32 | afrosheen | although the pri isn't throwing any kind of alarms |
| 21:19.33 | CCFL_Man2 | see if there are any alarms |
| 21:19.41 | CCFL_Man2 | ahh |
| 21:19.42 | afrosheen | at least not on the asterisk box |
| 21:19.44 | tzafrir_home | I'm at lost at how this program should do something |
| 21:21.17 | tzafrir_home | Unless I'm supposed to run sliptest /dev/zap/pseudo |
| 21:22.30 | afrosheen | smartjack is happy |
| 21:22.38 | CCFL_Man2 | i need a 684A subset ringer box |
| 21:22.56 | CCFL_Man2 | afrosheen: they bitch at them until the problem is fixed |
| 21:23.09 | afrosheen | CCFL_Man2, I still gotta run patlooptest on this card |
| 21:23.10 | CCFL_Man2 | you checked your logs on the asterisk box? |
| 21:23.32 | afrosheen | CCFL_Man2, yeah, been pri debugging it also, it looks happy as far as I can see |
| 21:23.44 | coppice | tzafrir_home: you run sliptest with a channel device |
| 21:24.19 | tzafrir_home | If the channel is used by Asterisk, sliptest cannot open it. If it is unused, then I see no output |
| 21:24.20 | tc3driver-nii | my sip porvider says dtmf should be inband |
| 21:24.25 | CCFL_Man2 | afrosheen: you have other equipment to test the line? say a cisco router with a T1 WIC and fxs VWIC? |
| 21:24.26 | *** join/#asterisk sacitec (n=tobi@189.149.103.123) |
| 21:24.29 | *** part/#asterisk sacitec (n=tobi@189.149.103.123) |
| 21:24.44 | afrosheen | CCFL_Man2, nope |
| 21:24.52 | WilliamK | is there a way I can add a queuemember from the console of *? |
| 21:24.53 | afrosheen | I just gotta craft a cable and patlooptest both ports |
| 21:25.06 | afrosheen | then swap in our new card, see what happens |
| 21:25.09 | CCFL_Man2 | oh, T1 crossover? |
| 21:25.13 | afrosheen | if all that fails, TWTC is getting an ass chewing |
| 21:25.20 | coppice | the port needs to be an unused one. it should be written and read |
| 21:25.43 | afrosheen | coppice, you referring to the patlooptest? |
| 21:26.05 | tzafrir_home | sliptest, from http://soft-switch.org/downloads/ |
| 21:26.28 | tzafrir_home | The port *is* unused |
| 21:26.49 | tzafrir_home | Is it supposed to work with analog channels? |
| 21:28.22 | coppice | it works with analogue channels |
| 21:28.22 | afrosheen | well thanks for your advice guys, gonna tackle this tomorrow (ugh) |
| 21:28.28 | CCFL_Man2 | afrosheen: since i'm a home user, i would get an old cisco voice gateway, update it's ios with my $8 smart net cco login, and terminate the PRI with SIP |
| 21:28.34 | tzafrir_home | WilliamK, using '!' and a call to sed should do the trick :-( |
| 21:28.53 | CCFL_Man2 | but also being a home user, i really couldn't afford a PRI |
| 21:29.45 | coppice | tzafrir_home: the port looks very much used to me |
| 21:30.13 | tzafrir_home | If anybody uses it, it is not asterisk |
| 21:30.30 | tzafrir_home | sliptest managed to open it. So how can it be used? |
| 21:31.32 | coppice | sliptest opens the file name you give on the command line, and reads and writes it |
| 21:32.57 | coppice | it writes AWGN, and reads back the echo of that. if there more than a very little, and not too much sound from the far end, sliptest will work out the loop delay |
| 21:35.22 | tzafrir_home | Right. And Astribank drivers will not send over PCM when the line is not off-hook... |
| 21:36.38 | coppice | take it off hook, then. just put a resistor across, to loop the line |
| 21:36.57 | jozu | someone speak spanish? |
| 21:37.00 | jozu | alguien habla español? |
| 21:37.22 | coppice | ä¸æ˜¯ |
| 21:38.35 | CCFL_Man2 | jozu: engrish plz |
| 21:39.03 | jozu | can anyone helme with a gsm-gateway? |
| 21:39.15 | jozu | i call to them and y ear the second tone |
| 21:39.35 | tzafrir_home | This is why it worked with channel 1 and not the others |
| 21:39.47 | jozu | but i want a ivr into gsmgateway sip extension |
| 21:40.25 | tzafrir_home | we have such "devices" at work, but not here |
| 21:40.31 | jozu | i made a inbound call route but nothing (destination ivr), only second tone |
| 21:40.32 | coppice | tzafrir_home: why doesn't audio work on hook? that is gonna kill CLID |
| 21:40.59 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 21:41.09 | tzafrir_home | there is also audio when CLID is sent |
| 21:41.28 | tzafrir_home | see /proc/xpp/XBUS-*/XPD-*/summary |
| 21:41.42 | tzafrir_home | It saves a whole lot of traffic |
| 21:41.49 | tzafrir_home | and CPU time |
| 21:43.03 | *** join/#asterisk sacitec (n=tobi@189.149.103.123) |
| 21:43.07 | tzafrir_home | jozu, what specific gsm-gateway are you talking about? Which vendor? What model? |
| 21:43.28 | sacitec | hello, anyone using aastra 9133i with TFTP + asterisk ? |
| 21:43.29 | tzafrir_home | How do you connect it to Asterisk? |
| 21:44.16 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583653.dsl.bell.ca) |
| 21:45.03 | jozu | i register the gsm-gateway in a sip extension in the server |
| 21:45.23 | jozu | and the server have a sip trunk to the voip provider |
| 21:46.43 | jozu | i want a ivr when gsm call arrive to the gsm gateway |
| 21:46.52 | *** join/#asterisk Ryushin (n=Ryushin@windwalker.openinnovations.com) |
| 21:47.11 | *** join/#asterisk gabbernaster (n=jshanks@69.10.147.2) |
| 21:47.24 | jozu | example --- 1 call office, 2 second tone for external number (DISA?) |
| 21:48.34 | jozu | any idea? |
| 21:54.17 | WilliamK | anyone know why when you reload from the console it knocks everyone out of the queues? |
| 21:55.47 | sacitec | aastra 9133i and TFTP ? |
| 21:56.54 | tc3driver-nii | who do you guys recommend for a good sip/voip provider? |
| 21:57.29 | hmmhesays | depends on what you need |
| 21:57.44 | hmmhesays | I use vitelity for my home/personal use |
| 21:57.53 | kiscokid | tc3: voicepulse |
| 21:59.31 | tc3driver-nii | for use with asterisk, and for business use (166 phones) |
| 22:00.36 | *** join/#asterisk moprilo (n=jjohn@201.192.35.138) |
| 22:01.54 | moprilo | hi,.. in my newly install asterisk i don't have the zap show .. even though I have my zaptel drivers good to go (ztcfg) any ideas |
| 22:03.36 | Deeewayne | ~thebook |
| 22:03.37 | jbot | [thebook] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 22:03.56 | *** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net) |
| 22:06.21 | tzafrir_home | moprilo, what is the output of: cat /proc/zaptel/* |
| 22:06.39 | tzafrir_home | moprilo, see also http://svn.digium.com/svn/zaptel/branches/1.4/README (look for PROCFS) |
| 22:08.07 | tzafrir_home | sacitec, probably someone does. Though I suspect Asterisk tends to communicate with that phone through SIP/RTP ;-) |
| 22:08.13 | tzafrir_home | Just ask your question |
| 22:11.42 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 22:11.42 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 22:12.45 | sacitec | i'm unable to work with FTP, i have xinetd/tftp running on my * server, on the directory /tftpboot i have both files, aastra.cfg and mac.cfg, both of them with -rw-r--r--, and root.root |
| 22:12.59 | sacitec | phones freezes when triying to download config |
| 22:13.14 | sacitec | on the screen apears "retriying config downlad" |
| 22:13.17 | sacitec | and that's all |
| 22:18.10 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 22:18.10 | *** mode/#asterisk [+o russellb] by ChanServ |
| 22:21.48 | *** join/#asterisk Qwell_ (n=north@pdpc/sponsor/digium/Qwell) |
| 22:21.48 | *** mode/#asterisk [+o Qwell_] by ChanServ |
| 22:32.43 | *** join/#asterisk elriah (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 22:33.21 | elriah | [TK]D-Fender: Ever seen a polycom phone display "application is not present" after an attempted firmware update? This is a 500. |
| 22:34.17 | *** part/#asterisk kiscokid (n=ron@208.106.35.66) |
| 22:34.40 | [TK]D-Fender | elriah, what ver? |
| 22:34.56 | elriah | Latest. 4.0/2.2 |
| 22:35.04 | elriah | From unknown firmware version |
| 22:35.15 | [TK]D-Fender | elriah, Hope you noted the change in support for 500/300...... |
| 22:35.31 | [TK]D-Fender | elriah, now's a good time to read your release notes... |
| 22:35.45 | elriah | [TK]D-Fender: Apparently not... |
| 22:36.05 | elriah | Oh crud. |
| 22:36.33 | ManxPower | Actually, a good time to read the release notes would have been BEFORE you tried upgrading the firmware. |
| 22:38.51 | *** join/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com) |
| 22:40.02 | mcab | elriah: it's ok, you haven't pooched the 500 - I think there's a tech bulletin on how to have 500/300s coexist peacefully with 4.0.0 & 2.2.0 |
| 22:44.37 | *** join/#asterisk Defraz (n=t0tal@65.121.20.50) |
| 22:50.18 | tzafrir_home | sacitec, I really don't know those phones, but you can check if you actually get requiests to that port |
| 22:50.26 | tzafrir_home | (in e.g. the logs) |
| 22:50.46 | tzafrir_home | You can also try to connect with an independent tftp client to check it's possible |
| 22:52.01 | elriah | Yea, found it, scrambling... thanks, mcab |
| 22:53.39 | *** join/#asterisk codefreeze (n=steve_mu@wsip-70-184-124-51.ph.ph.cox.net) |
| 22:53.39 | *** mode/#asterisk [+o codefreeze] by ChanServ |
| 22:56.12 | CCFL_Man2 | best way to interface fxs and fxo lines to asterisk is with a channel bank |
| 22:58.46 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 22:59.43 | lesouvage | I need an ata with poe and vpn support. Is there an ata on the market that fits this requirements? |
| 23:00.46 | [TK]D-Fender | lesouvage, PoE ATA w/ VPN? lol |
| 23:01.38 | schattengolfer | would someone be so kind as to explain what the "Act" item means on a T1 connected to a Digium TE110P? |
| 23:01.59 | schattengolfer | when viewing the interface in zttool |
| 23:02.08 | lesouvage | [TK]D-Fender: is that a no? |
| 23:02.45 | [TK]D-Fender | lesouvage, I honestly haven't even seen a PoE ATA period, let alone one with VPN |
| 23:03.54 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 23:04.21 | lesouvage | I tried to connect a fax today on a vpn for voip with a sipura ata but that was a disaster. I couldn't even get the webenebled configuration page on the screen. |
| 23:05.05 | [TK]D-Fender | lesouvage, Oh, and now your adding FAXING to the mix? |
| 23:05.18 | lesouvage | I can't imagine that I'm the first one having this problem. |
| 23:05.24 | *** join/#asterisk |omni| (n=rob@net82.allied-security.com) |
| 23:05.40 | coppice | every good system *must* do fax |
| 23:06.26 | lesouvage | [TK]D-Fender: Otherwise they have to bike to the headoffice 1 km away to send a fax. |
| 23:07.41 | lesouvage | [TK]D-Fender: Any suggestion to get faxing working on a location with only glassfiber and no isdn or pstn? |
| 23:13.45 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 23:15.37 | ectospasm | lesouvage: good luck with that. You'd probably be better off with issuing digital signature pads and pass documents via e-mail, or ftp et al... |
| 23:24.29 | lesouvage | ectospasm: maybe a scanner that sends the scan to something that can send the fax from the headoffice. |
| 23:25.37 | ectospasm | Only reason why I can see faxing being an issue in this day and age is to trade signatures... but that suggestion could work |
| 23:26.19 | coppice | whether is makes sense is irrelevant. its heavily used anyway |
| 23:26.19 | ectospasm | actually, you could probably talk to a Xerox document management salesperson to get ideas from... |
| 23:27.12 | ectospasm | ipp was supposed to make faxing irrelevant, hasn't really done so... |
| 23:28.12 | coppice | it hasn't really gone anywhere |
| 23:30.04 | ectospasm | Oh, well. An analog phone line used strictly for faxing is fairly cheap anyway |
| 23:30.34 | ectospasm | just slap a basic long distance plan on it, and you should fit 90% of business needs... |
| 23:31.44 | *** join/#asterisk putnopvut (n=putnopvu@wsip-70-184-124-51.ph.ph.cox.net) |
| 23:32.37 | lesouvage | lesouvage: but it is kind of strange that there aren't any ata's with vpn support. The poe is just for not having powersupplies everywhere but the vpn is simply there. |
| 23:33.09 | coppice | an ata with vpn doesn't make much sense |
| 23:34.21 | [TK]D-Fender | An ATA with VPN & PoE considerably less so. |
| 23:34.48 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 23:35.03 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 23:35.35 | *** join/#asterisk Cresl1n (n=matt@wsip-70-184-124-51.ph.ph.cox.net) |
| 23:35.35 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 23:38.42 | aninoSAdilim | can asterisk send a faxtone? |
| 23:40.21 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
| 23:41.45 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 23:42.11 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 23:42.58 | lesouvage | coppice: why not, there is a vpn dedicated to all the voip traffic (will be 400 phones in short time) and fax traffic is part of that. btw: http://www.voip-info.org/wiki/view/YGW60+ATA might be the solution. |
| 23:43.53 | *** join/#asterisk [hC] (n=hardcore@wsip-70-184-124-51.ph.ph.cox.net) |
| 23:45.14 | coppice | its probably a good solution if you like trouble |
| 23:46.26 | *** join/#asterisk anthm (n=anthm@mb90736d0.tmodns.net) |
| 23:46.26 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:46.30 | lesouvage | coppice: what do you mean? |
| 23:47.20 | coppice | well, that box uses a myson century CS6220 chip, and all the boxes using that use the same software, and the software sucks |
| 23:48.00 | coppice | a lot of those boxes advertise functions they don't even have |
| 23:48.48 | lesouvage | coppice: thanks |
| 23:52.21 | *** join/#asterisk remmo (n=junk@203.32.47.250) |
| 23:55.32 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 23:55.32 | *** mode/#asterisk [+o russellb] by ChanServ |
| 23:57.07 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.154.118) |
