| 00:00.20 | b11d|bbl | what the hell is Dail Up? |
| 00:00.35 | b11d|bbl | seriously.. this is a telco channel.. you need to know it's "dial" :) |
| 00:00.58 | JT | the telco bit is debatable, but yes, he should know that |
| 00:01.07 | b11d|bbl | yeah.. i thought that immdiately after typing it :) |
| 00:01.38 | `Sean | lol i keep fucking up with that word heh |
| 00:01.39 | `Sean | :/ |
| 00:01.49 | `Sean | even in extensions.conf when i type up my dailplan sometimes... |
| 00:01.50 | b11d|bbl | it happens :P |
| 00:02.06 | `Sean | wanted to us asterisk to get internt connection via dial up |
| 00:02.13 | `Sean | like AOL or something of similar type of thing :) |
| 00:02.39 | b11d|bbl | use like, pppd or something with a tdm card.. not that i've done that, but i suspect it could be done. |
| 00:03.02 | PC_Clone | has anyone seen "unknown signalling method 'bri_cpe_ptmp' errors with bristuff 0.4? |
| 00:03.06 | b11d|bbl | doh.. tdm should be fxo |
| 00:06.50 | b11d|bbl | so.. anyone in Amsterdam? I'm heading over there on the 30th.. |
| 00:10.20 | De_Mon | b11d|bbl gonna be smokin the weed out at a cafe? |
| 00:10.57 | *** join/#asterisk Poehali (n=actionma@74.93.5.186) |
| 00:11.46 | Poehali | hey I got a asterisk question. if I have vonage boxes(but no longer signed up for vonage) how do I use them for my asterisk box? |
| 00:12.18 | JT | PC_Clone: sounds like a pretty new version |
| 00:13.42 | aao_pwner | :/ |
| 00:14.41 | PC_Clone | JT: July 25 I think |
| 00:15.01 | JT | 0.4 must be fairly beta |
| 00:15.10 | PC_Clone | I tried to change it to pri_cpe and that didn't even work |
| 00:15.22 | PC_Clone | so I think I have a problem somwhere else |
| 00:18.52 | Poehali | please can someone tell me what to buy or do to make this thing work? I have a box with asterisk linux installed on it but I dunno what to do next |
| 00:19.29 | *** join/#asterisk thansen|laptop (n=thansen@208.14.131.162) |
| 00:21.15 | *** join/#asterisk digimania (n=digimani@24-119-242-39.cpe.cableone.net) |
| 00:24.00 | *** join/#asterisk Freman (n=freman@210.79.26.194) |
| 00:24.13 | JT | Poehali: ex vonage atas need reflashing, not always possible |
| 00:24.18 | Freman | hey folks. I have a little problem with an asterisk server |
| 00:24.37 | Freman | it's located some way away, and my snom is located behind a nat at another network |
| 00:24.40 | Freman | the snom registers |
| 00:24.54 | Freman | and I can place the call, it appears to start making the call but then it just hangs up |
| 00:25.05 | *** part/#asterisk digimania (n=digimani@24-119-242-39.cpe.cableone.net) |
| 00:25.30 | Freman | -- Format for call is ulaw, then immidatly after -- Hungup 'IAX2/faktortel-4' |
| 00:25.38 | Freman | err immediately |
| 00:26.07 | JT | haha |
| 00:26.07 | Freman | if I make a call file and tell it to call my mobile, then transfer the call to the snom extension it goes through fine and both ends can hear each other |
| 00:26.14 | JT | it's probably just faktortel sucking |
| 00:26.18 | *** join/#asterisk remmo (n=junk@203.25.123.250) |
| 00:26.42 | Freman | i've never had a problem with faktortel, at least not in 2 years |
| 00:27.00 | Freman | and the call file test kind of rules out the faktortel sucking |
| 00:27.29 | JT | lots of others have had problems with faktortel though, they're a fairly dodgy ITSP |
| 00:27.54 | Freman | www.faktortel.com.au <- that one? or another one? |
| 00:28.05 | JT | yes, the one and only |
| 00:28.30 | Freman | I've tried everything to rule out the nat |
| 00:30.01 | Poehali | JT: so assuming reflashing can't be done, what should I buy to hookup ethernet to rj11? |
| 00:30.27 | JT | Poehali: a linksys SPA-3102 or similar |
| 00:31.04 | JT | Freman: faktortel get angry if you "make too many calls", their DIDs in sydney were also down to weeks on end |
| 00:31.29 | Poehali | JT: so for 20 phones I'd need to buy 20 of those? |
| 00:31.41 | JT | Poehali: you wouldn't use those for 20 phones |
| 00:31.51 | JT | it would also make more sense to use ip phones instead |
| 00:32.25 | Freman | never had any of those issues |
| 00:32.33 | JT | Freman: are you in sydney? |
| 00:32.38 | Freman | but then... I don't have a sydney did |
| 00:32.43 | JT | right |
| 00:32.44 | Poehali | JT: we can't afford to replace all these existing phones with new ones |
| 00:32.51 | JT | and ther are just dodgily run |
| 00:33.06 | JT | they get pissy if your call usage isn't in line with their estimations |
| 00:33.06 | Freman | I have had problem calling cyprus, but it effected 3 out of the 4 vsps I use - and when I complained to faktortel it got fixed for all 3 vsps (c: |
| 00:33.29 | JT | sounds like faktortel don't own their own infrastructure |
| 00:33.39 | JT | they must all be using a reseller |
| 00:34.36 | Freman | they own most of it, it was a suppliers fault |
| 00:34.52 | JT | an upstream supplier? ;) |
| 00:35.03 | Freman | pennytel and nodephone suffered same fault |
| 00:35.10 | Freman | and I don't imagin nodephone being to dodgy |
| 00:35.30 | JT | clearly they're all using the same wholesaler, at least for some overseas calls |
| 00:35.48 | Freman | or calls to that area |
| 00:36.01 | JT | hey, they're all ITSPs, dodgyness is simply relative, they're all dodgy :P |
| 00:36.02 | Freman | anyway... this isn't solving my current issue )c: |
| 00:36.22 | JT | if the real PSTN links to cyprus were down, the telcos would realise immediately |
| 00:36.52 | Freman | but that's firewalled tighter then... |
| 00:42.33 | Freman | ok... so it works comming out of my home |
| 00:43.58 | *** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net) |
| 00:44.57 | rvhi | hi, on my polycom phone, inbound calls show "xxxxx <sip:xxxxxxx@domain.com>", the xxxxx is the caller number |
| 00:45.15 | *** join/#asterisk webman (n=chatzill@202.154.82.114) |
| 00:45.16 | rvhi | anyone knows how to get rid of <sip... #domain> part |
| 00:45.37 | bkw__ | why? |
| 00:45.42 | webman | anyone know what time it is at digium ? |
| 00:45.44 | bkw__ | you really can't |
| 00:45.48 | bkw__ | its 7:45PM |
| 00:45.52 | _ShrikE | 7:45 |
| 00:45.52 | bkw__ | they are in Central time zone |
| 00:46.08 | webman | damn, my calculations where off by 1 hour..... |
| 00:47.11 | webman | anyone have experience with the B410p card? are the lights on the back meant to be flashing red, solid red, or off when the port is working correctly? |
| 00:47.35 | bkw__ | I think thats bad |
| 00:48.20 | webman | hmmm, actually, one port is now solid green .... |
| 00:49.30 | webman | maybe line in use == green, line not connected == flash red, and line plugged in == solid red ?? I couldn't find this info in the manual, and I'm have heaps of problems, though sometimes it works really well :( |
| 00:51.49 | webman | anyone manage to get one of these cards working properly? especially in australia? |
| 00:52.11 | JT | i avoid the b410p like the plague :) |
| 00:52.37 | webman | JT: why? |
| 00:52.42 | Freman | sweet |
| 00:52.44 | Freman | different error |
| 00:52.48 | JT | only works with icky misdn |
| 00:52.59 | webman | JT: as opposed to ? |
| 00:55.36 | Freman | JT, turns out to be my fault - handing to many digets to faktortel |
| 00:55.44 | JT | webman: bristuff |
| 00:55.47 | Freman | hand the right number, get different error |
| 00:56.41 | Freman | now it's giving me == Everyone is busy/congested at this time (1:0/0/1) |
| 00:57.23 | JT | Freman: try and do a sip debug |
| 00:57.28 | JT | these high level errors aren't as useful |
| 00:57.56 | Freman | no they aren't |
| 00:59.58 | Freman | hmm that wasn't any more useful |
| 01:00.15 | JT | pastebin it in pastebin.ca |
| 01:00.19 | Freman | the sip debugging showed before and after the block that tells me everyone is conjested, but not during |
| 01:00.30 | Freman | maybe iax debugging |
| 01:00.35 | JT | err |
| 01:00.40 | JT | oh |
| 01:00.51 | JT | if you're connecting to the telco with iax2, use iax2 debug |
| 01:01.32 | Freman | http://pastebin.ca/668699 |
| 01:01.37 | Freman | yeh |
| 01:02.11 | JT | switch off sip debug |
| 01:02.16 | JT | and turn on iax2 debug |
| 01:02.50 | Freman | http://pastebin.ca/668701 |
| 01:03.07 | Freman | oooh do I have to switch sip off? I like digging through 4x the output for no reason *(c: |
| 01:04.29 | JT | Freman: why are you dialling with an excessive leading zero? |
| 01:04.53 | Freman | it's a mobile number, they start with 0's |
| 01:05.13 | JT | they don't start with two zeros. |
| 01:05.14 | JT | CALLED NUMBER : 00400794378 |
| 01:05.23 | Freman | where the hell did it find that? |
| 01:05.34 | JT | in the iax2 call setup |
| 01:05.41 | Freman | not you, IT (c: |
| 01:05.56 | Freman | Executing [s@macro-mDial:9] Dial("SIP/9103-081e1be0", "IAX2/faktortel/00400794378") in new stack |
| 01:05.57 | Freman | thanks |
| 01:06.00 | JT | it's plainly obvious you're calling with an excessive zero |
| 01:06.01 | JT | -- Executing [s@macro-mDial:9] Dial("SIP/9103-081e1be0", "IAX2/faktortel/00400794378") in new |
| 01:06.04 | JT | heh |
| 01:06.18 | Freman | it's the same dialplan I use at home (I thought I'd be smart and copy) |
| 01:06.29 | JT | hmm |
| 01:07.38 | Freman | found it |
| 01:07.47 | Freman | it was from part of the plan that wasn't copied from my own |
| 01:08.17 | *** join/#asterisk techie (n=techie@ppp-69-237-91-25.dsl.frs2ca.pacbell.net) |
| 01:08.23 | Freman | &mDial(${GLOBAL(COUNTRY)}${EXTEN}); shoulda been &mDial(${GLOBAL(COUNTRY)}${EXTEN:1}); |
| 01:08.23 | JT | i see |
| 01:08.25 | Freman | thanks JT |
| 01:08.39 | Freman | if not for you, I probably would have missed it for the next 20 minutes |
| 01:09.14 | JT | hah ;) |
| 01:09.25 | JT | it was clear that the ITSP was rejecting the number |
| 01:09.30 | JT | and it was reaching the ITSP |
| 01:09.39 | *** part/#asterisk techie (n=techie@ppp-69-237-91-25.dsl.frs2ca.pacbell.net) |
| 01:09.45 | JT | # |
| 01:09.45 | JT | Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: HANGUP |
| 01:09.45 | JT | # |
| 01:09.45 | JT | <PROTECTED> |
| 01:09.45 | JT | # |
| 01:09.47 | JT | <PROTECTED> |
| 01:10.15 | JT | only when copy and pasting |
| 01:10.16 | Freman | it's part of the line numbering method |
| 01:10.20 | Freman | you have to view raw |
| 01:10.35 | Freman | code on my own site suffers the same fate (c: |
| 01:10.43 | Freman | I think I hurt myself writing this - !^(000)$!sip:\1@\d!d !^(10[0-9])$!sip:\1@\d!d !^(09[0-9]{6})$!sip:\1@\d!d !^(9[0-9]{3})$!sip:\1@\d!d !^(13[1-9][0-9]{3})$!sip:\1@\d!d !^(1[389]0{3}[0-9]{6})$!sip:\1@\d!d !^([2-9][0-9]{7})$!sip:\1@\d!d !^(0[234678][0-9]{8})$!sip:\1@\d!d !^(0011[0-9]+)$!sip:\1@\d! |
| 01:10.51 | JT | i guess you could copy it out of the dialogue box |
| 01:10.53 | JT | wtf |
| 01:10.56 | Freman | snom300 dial plan (c: |
| 01:10.56 | JT | that's ridiculous |
| 01:10.59 | JT | eww |
| 01:11.10 | Freman | now that I think of it |
| 01:11.13 | JT | snoms, bin 'em i say ;) |
| 01:11.14 | Freman | I could probably shrink it |
| 01:11.26 | Freman | but it's better like that, it's in seperate components |
| 01:11.37 | Freman | hey, snom shits on grandstream (I know, cos I binned that) |
| 01:11.55 | JT | yeah everything shits on grandstream |
| 01:12.06 | JT | i don't know why boring snoms are so popular in australia |
| 01:12.10 | JT | better marketing it would seem |
| 01:12.19 | De_Mon | I refuse to shit on something so shitty |
| 01:14.03 | jingles | it came, pre-shitted on |
| 01:14.39 | Freman | !^(000|13[1-9][0-9]{4}|10[0-9]|09[0-9]{6}|9[0-9]{3}|1[389]0{3}[0-9]{6}|[2-9][0-9]{7}|0[234678][0-9]{8}|0011[0-9]+)$!sip:\1@\d! |
| 01:14.43 | Freman | there... I make it shorter |
| 01:14.52 | Freman | nah |
| 01:14.56 | Freman | there's next to no marketing |
| 01:15.05 | Freman | we just don't have much of a handset market yet |
| 01:15.23 | Freman | and... they're "simple" |
| 01:15.36 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 01:15.53 | Freman | alot of folk round these parts barely have the necessary braincells to find the power button on a pc |
| 01:16.37 | [hC] | hey polycom people: know of a way to specify vlan id to associate to in the config files? |
| 01:17.23 | [hC] | im warming up way more to the aastras lately |
| 01:17.40 | *** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com) |
| 01:17.40 | [hC] | and their new line up.. the 51/53/55/57 line |
| 01:17.58 | JT | Freman: polycom phones are much better than the snoms |
| 01:18.15 | Freman | and a lot harder to get their hands on (c: |
| 01:18.29 | JT | the only reason i think snom has any popularity here is that people have seen others using them |
| 01:18.34 | JT | they're not that hard to get |
| 01:18.36 | *** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com) |
| 01:18.51 | JT | it's still hard to get the IP320/330 locally, but otherwise, not too hard |
| 01:19.25 | [hC] | i can get them easily.. the ip430 my distributor seems to have decided to stop carrying for some reason |
| 01:19.51 | JT | i guess there musn't be much point with the price difference to the IP320 |
| 01:19.55 | bminish | I got a SNOM 320 here and I like it just fine |
| 01:19.55 | [hC] | im trying to set up an auto provisioning system for polycom like ive done for aastra, where the phone will auto-find its way onto a special vlan tag |
| 01:20.06 | [hC] | but there doesnt seem to be a way to do it without CDP |
| 01:20.20 | JT | people always like their phones just fine when they haven't used better |
| 01:20.47 | bminish | ok but whats wrong with it? |
| 01:20.48 | fujin | [hC]: so use CDP? :) |
| 01:20.57 | Freman | even with the snoms, to get a decent price on em we get em shipped from WA (other side of the country) |
| 01:21.08 | JT | they're just not very good value for money |
| 01:21.17 | Freman | most aussies are overly happy (even business and schools) to have PAP2's and standard handsets |
| 01:21.25 | [hC] | fujin: know of a switch that does PoE and CDP and isnt cisco? |
| 01:21.28 | JT | they have poor aesthetics, audio and usability is nowhere near polycom |
| 01:21.39 | bminish | at that price range what is better value? |
| 01:21.42 | JT | not real sized businesses... |
| 01:21.43 | JT | polycom |
| 01:21.47 | [hC] | aastra as well |
| 01:21.49 | JT | i know where WA is |
| 01:21.57 | Freman | 10-30 employees |
| 01:22.19 | bminish | I find the audio fine and the usability ok once I finished getting it all setup with asterisk |
| 01:22.20 | JT | if they had a proper digital PABX before, they would come to expect more than crappy analogue handsets |
| 01:22.29 | heelios | so. i got a SPA3102. when I try to place a call to asterisk from the PSTN via the damn thing, I get three "10.12.1.1(Asterisk Box) -> 10.12.1.15(SPA) SIP Status: 407 Proxy Authentication Required" as the only network traffic. any clues? |
| 01:22.36 | JT | bminish: sure, it's just not the best, which is my entire point |
| 01:23.41 | JT | yes |
| 01:23.51 | JT | australian suppliers seem to like stocking junk phones |
| 01:23.51 | Freman | digital pbx's arn't in a majority around these parts )c: |
| 01:23.53 | JT | fools |
| 01:24.09 | JT | why they stock grandshit is beyond me |
| 01:24.18 | JT | if you want to buy polycoms locally, call Westan |
| 01:24.30 | JT | last i checked they didn't have the 320s yet |
| 01:24.33 | [hC] | i remember talking to the snom founder at astricon, as well as two grandstream guys |
| 01:24.48 | [hC] | the snom guy is really tuned in and hes a smart guy |
| 01:24.48 | fujin | [hC]: maybe Linksys? |
| 01:25.01 | [hC] | the grandstream guys were like 'really? you dont like the phone? what would you change?' |
| 01:25.02 | fujin | CDP is pretty standard in a 'good' office environment |
| 01:25.04 | [hC] | i was dumbfounded |
| 01:25.06 | fujin | get a few 3560 switches |
| 01:25.25 | [hC] | $$$ |
| 01:25.34 | [hC] | compared to the netgear poe switches i normally deploy that work GREAT |
| 01:25.50 | bminish | the utstarcom F3000 is a truly crap phone ;-) |
| 01:26.02 | JT | all wifi phones such |
| 01:26.06 | JT | ~wifisip |
| 01:26.06 | jbot | Wi-Fi SIP phones suck. All of them. HARD. Some only slightly less than others... |
| 01:26.11 | JT | suck |
| 01:26.27 | bminish | my nokia e70 only sucks a bit |
| 01:26.38 | *** join/#asterisk axscode (n=axscode@203.213.217.123) |
| 01:27.56 | _ShrikE | Anyone used the aastra sip dect phones? |
| 01:29.44 | _ShrikE | I wouldn't expect they are much better than the other aastra phones. |
| 01:30.08 | [hC] | I have a dect phone on my desk right now |
| 01:30.15 | _ShrikE | aastra? |
| 01:30.16 | [hC] | I'm the first guy to be aastra sip dect certified here |
| 01:30.17 | [hC] | yep |
| 01:30.21 | [hC] | Ive been playing with it all day |
| 01:30.24 | _ShrikE | how is it? |
| 01:30.26 | [hC] | i like it |
| 01:30.27 | [hC] | so far |
| 01:30.50 | _ShrikE | Im not the biggest aastra fan. But I am very curios about the dect phone |
| 01:30.58 | [hC] | have you tried the new phones? |
| 01:31.05 | [hC] | I was a big aastra hater |
| 01:31.07 | _ShrikE | I ahve a 57iCT |
| 01:31.15 | _ShrikE | s/ahve/have |
| 01:31.18 | [hC] | then i spent a day with david sayson and he gave me some phones to try and i dont mind it |
| 01:31.30 | [hC] | I have a 57iCT and 57i on my desk.. I actually have a problem with the 57i I cant figure out yet |
| 01:31.49 | _ShrikE | Mine is actually behaving rather nicely with the new firmware |
| 01:31.54 | _ShrikE | 2.1 I think |
| 01:31.57 | [hC] | when a second call comes in, the thing rings a split second, then dumps to busy |
| 01:32.53 | _ShrikE | I cant say ive had that problem. |
| 01:33.07 | [hC] | huh. it happens on the 57iCT and 480iCT from what I could test today |
| 01:33.13 | [hC] | try it, get on the phone and call your extension again.. |
| 01:33.18 | [hC] | but other than that, whats your beef with the aastra? |
| 01:33.25 | [hC] | i really like how extensible it is |
| 01:33.39 | [hC] | my biggest complaint is the shape/texture of the handset |
| 01:34.01 | _ShrikE | I just tried it and it seems to behave properly |
| 01:34.18 | [hC] | huh. I wonder why mine does it. |
| 01:34.26 | *** join/#asterisk dlynes (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 01:34.27 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 01:34.45 | _ShrikE | My main issues with aastra seem to have been resolved with the latest firmware. Only time will tell though. |
| 01:34.54 | [hC] | what were they before? |
| 01:34.56 | _ShrikE | I just cant take a phone that requires rebooting now and again |
| 01:35.15 | [hC] | yeah i had some lockup/reboot problems, especially with blf involved with 2.0 |
| 01:35.21 | [TK]D-Fender | [hC], the handset has NO wieght and seriously blows and has acoustic echo issues |
| 01:35.24 | _ShrikE | I could lock the phone up by holding calls on the handset and picking up on the console |
| 01:35.38 | _ShrikE | that also breaks muting |
| 01:35.47 | _ShrikE | those rubber buttons suck |
| 01:36.17 | _ShrikE | Im more of a polycom fan |
| 01:36.20 | [TK]D-Fender | [hC], My 57i CT locked up maybe once every 2 wekks or so. Cordless would become unpaired, and the cordless' are all treated as the SAME. <---- BS! |
| 01:36.49 | _ShrikE | aastra is fancy, but polycom is plain reliable. |
| 01:36.56 | *** join/#asterisk enjay5150 (n=chatzill@74.202.4.2) |
| 01:36.57 | [hC] | what do you mean they're all treated as the same? |
| 01:37.11 | [TK]D-Fender | And yes, the Aastra 5i rubber buttons BLOW. 2 things Aastra has going for them : MASSIVELY versatile state-based soft keys, and SUPER expansion modules. |
| 01:37.26 | [hC] | yeah |
| 01:37.31 | [hC] | every brand has their problems |
| 01:37.34 | [hC] | polycom has shitty bugs too |
| 01:37.52 | [hC] | 601 with side modules and a bunch of blf causes the phone to lock up and reboot |
| 01:37.52 | _ShrikE | yeah, but I have been less impaired by this latest firmware.. |
| 01:38.05 | enjay5150 | I have 4 servers "HP DL380" with Sangoma A104-4 and 16 PRI's b8zs/esf/ni2.. out of every server, every T1 when I send out a call I get this obnoxious buzzing on both ends and voice doesnt pass.. has anyone seen this before? |
| 01:38.19 | _ShrikE | [hC]: I have seen that as well. |
| 01:38.20 | [TK]D-Fender | [hC], seems all handsets will have the same features and can't be functionally independant of the base. |
| 01:38.37 | [TK]D-Fender | [hC], As in I wanted the cordless to go to another guy and HIS calls ALWAYS rang on the base. |
| 01:38.42 | [hC] | [TK]D-Fender: oh.. no they cant. thats advertised though |
| 01:39.12 | [hC] | maybe theyll change it soon. |
| 01:39.20 | [TK]D-Fender | [hC], Only good if you are the cordless user as well. Even then... meh I'd sooner go analog + ATA + Polycom in a heartbeat |
| 01:40.08 | enjay5150 | I've tried turning off all echo cancellation, turning it on, playing with gain, tried 3 different versions of zaptel 2 versions of wanpipe 2 version of asterisk.. same issue all the way around.. telco tests clean all the way up to the NIU as soon as its plugged in they get junk.. |
| 01:40.25 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 01:40.38 | enjay5150 | but they T's come up fine d/b chans |
| 01:40.51 | [TK]D-Fender | enjay5150, NO audio passes? |
| 01:40.55 | asterisknerds | <PROTECTED> |
| 01:41.04 | enjay5150 | no audio at all, just a really obnoxious buzz.. |
| 01:41.08 | enjay5150 | loud too.. |
| 01:41.10 | lowlevel | cat pee + carpets sucks. |
| 01:41.11 | [TK]D-Fender | enjay5150, Got EC on this card? |
| 01:41.15 | enjay5150 | non-EC |
| 01:41.40 | enjay5150 | whats wierd is that its happening on 4 different cards in 4 different servers. |
| 01:41.44 | [TK]D-Fender | enjay5150, Ok, the only time I've heard of anything like that is when the EC module bugger gets caught up... I'd call Sangoma support if I were you |
| 01:41.45 | [hC] | [TK]D-Fender: polycom+ata? |
| 01:41.54 | [hC] | [TK]D-Fender: didnt know polycom made a cordless. |
| 01:42.00 | [TK]D-Fender | [hC], Seperate wired/wireless |
| 01:42.15 | [TK]D-Fender | [hC], Even if I DID want both for my "extension" |
| 01:42.30 | enjay5150 | yea Sangoma support is hokie just wanted to see what you guys had to say.. appreciate it! |
| 01:43.00 | [hC] | [TK]D-Fender: model? |
| 01:44.16 | *** part/#asterisk bminish (n=bminish@brenbox.westnet.ie) |
| 01:44.56 | [TK]D-Fender | [hC], ANY! |
| 01:47.42 | _ShrikE | Is the HD audio with polycom really worth it? I havent tried any personally. |
| 01:48.10 | [TK]D-Fender | _ShrikE, since it'd only matter inter-office, I wouldn't bother. |
| 01:48.35 | [TK]D-Fender | _ShrikE, its a bloody phone FFS :) |
| 01:48.52 | rvhi | hi, on my polycom phone, inbound calls show "xxxxx <sip:xxxxxxx@domain.com>", the xxxxx is the caller number |
| 01:48.52 | _ShrikE | right. |
| 01:48.54 | rvhi | anyone knows how to get rid of <sip... #domain> part |
| 01:52.31 | _ShrikE | rvhi: why would you want to do that? |
| 01:53.39 | rvhi | i'd rather show first line name, second line number, not number with a sip address |
| 01:54.21 | *** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com) |
| 01:54.59 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 01:55.07 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 01:55.37 | asterisknerds | <PROTECTED> |
| 01:58.53 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:01.08 | *** join/#asterisk angelv (n=chatzill@190.83.59.204) |
| 02:01.35 | angelv | hi guys... anny room for asterisk in spanish? |
| 02:11.04 | *** join/#asterisk rtejada (n=chatzill@190.83.59.204) |
| 02:12.17 | Freman | well must go now |
| 02:12.39 | Freman | thanks muchly JT for spotting my stupid mistake |
| 02:12.42 | *** part/#asterisk Freman (n=freman@210.79.26.194) |
| 02:25.47 | Dovid | i installd asterisk as well as asterisk-addons but MOH will not work |
| 02:25.50 | Dovid | i am using ztdummy |
| 02:25.54 | Dovid | any clues ? |
| 02:27.40 | *** join/#asterisk saftsack (n=saftsack@p57A77B8E.dip.t-dialin.net) |
| 02:28.36 | *** join/#asterisk saftsack (n=saftsack@p57A77B8E.dip.t-dialin.net) |
| 02:30.01 | *** join/#asterisk CrazyTux (n=CrazyTux@64.149.30.97) |
| 02:39.36 | *** join/#asterisk AJaymn (n=Me@71-82-218-158.dhcp.mdsn.wi.charter.com) |
| 02:43.30 | AJaymn | Anyone used a Linksys WIP-300 ? |
| 02:50.55 | CCFL_Man2 | breaking news!!! |
| 02:51.16 | CCFL_Man2 | my mc3810 concentrator communicates with my channel bank!!! |
| 02:51.38 | weasel00 | any suggestions for a ppc voip client? sjphone audio is lousy at best |
| 02:51.54 | AJaymn | ppc clients suck in general. :P |
| 02:52.06 | AJaymn | ive yet to find one that even works decent with wifi on a ppc |
| 02:52.15 | CCFL_Man2 | thanks :P |
| 02:52.41 | CCFL_Man2 | 4 Line Code Violations <--that a normal error for a clock source? |
| 02:52.43 | AJaymn | i wish there was a good one.. |
| 02:52.43 | JT | AJaymn: wifi + VoIP usually == bad |
| 02:52.54 | AJaymn | JT: Dah! lol |
| 02:53.00 | CCFL_Man2 | my cisco reports that within an interval of time |
| 02:53.09 | *** join/#asterisk marc7 (n=marc@216.19.180.137.novuscom.net) |
| 02:53.29 | weasel00 | i got the sjphone to work through gpmrs... but audio out is the speaker and audio is keeps getting clipped... stupid client problem |
| 02:53.49 | marc7 | in asterisk 1.4, is inbound caller ID still referred to with the global variable ${CALLERIDNUM}, or has that changed? |
| 02:53.51 | CCFL_Man2 | Total Data (last 19 15 minute intervals): |
| 02:54.05 | *** join/#asterisk DeepY0X (n=DeepY0X@201.240.45.49) |
| 02:55.11 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 02:55.41 | asterisknerds | <PROTECTED> |
| 02:56.05 | Dovid | how do i set the socket for mysql to connect to ? |
| 02:56.11 | Dovid | the defualt does not seem to be working |
| 02:56.26 | JT | ${CALLERID(num)} |
| 02:57.21 | *** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 02:57.54 | *** join/#asterisk waKKu (n=worth@unaffiliated/wakku) |
| 02:58.09 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1088929214.dsl.bell.ca) |
| 03:00.48 | JT | marc7: |
| 03:00.55 | Dovid | JT: y would MOH strt and stop right away ? i bult the asterisk add ons |
| 03:01.03 | Dovid | i do not wana use mpg123 |
| 03:01.16 | Dovid | ztdummy is loaded. any ideas ? |
| 03:02.52 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 03:03.11 | marc7 | JT: thanks, I obviously wasn't googling hard enough... our carrier is also sending us ANI information and so it's been a little tricky to get them to send it to us in the right context |
| 03:03.40 | JT | (ani) |
| 03:03.43 | JT | README.variables |
| 03:04.52 | marc7 | nice. they found a way to send us ANI information in caller ID format... so I don't know if we'll be able to use that variable |
| 03:05.00 | marc7 | glad to see it's clear cut these days |
| 03:07.38 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:09.34 | Dovid | JT: did u see my question ? |
| 03:10.18 | JT | Dovid: not sure why you asked me |
| 03:12.13 | Dovid | cause u know a lot. |
| 03:12.19 | Dovid | i just reformated the box |
| 03:12.30 | Dovid | b4 it was working. I must be missing something small (like always) |
| 03:13.29 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 03:13.48 | bintut | tzafrir: still there? :) |
| 03:14.03 | JT | Dovid: no idea |
| 03:14.15 | JT | look for error messages |
| 03:14.26 | CCFL_Man2 | anyone know how to clear an alarm on the adit 600? |
| 03:16.28 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 03:17.17 | bintut | tzafrir_laptop: still there? :) |
| 03:20.41 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
| 03:21.30 | fujin | hi |
| 03:21.45 | fujin | can anyone tell me if Set(DB(blah/lol=blah/newb); will OVERWRITE the previous value? |
| 03:22.23 | [TK]D-Fender | fujin, No, that won't set ANYTHING. |
| 03:26.03 | *** join/#asterisk tengulre11 (n=tengulre@222.90.66.10) |
| 03:26.14 | Dovid | fujin: yes it will |
| 03:30.41 | fujin | thanks |
| 03:30.46 | fujin | I just needed to know if DB puts would overwrite. |
| 03:31.57 | Dovid | fujin: u have exp. wit the db ? |
| 03:32.00 | Dovid | for some reason this wont work |
| 03:32.01 | Dovid | DB(cellular/123456)=${EXTEN}) |
| 03:33.04 | fujin | um |
| 03:33.06 | fujin | should work fine |
| 03:33.14 | fujin | I'm only using it to store the location of a virtual number |
| 03:33.17 | Dovid | not for me |
| 03:33.19 | Dovid | i keep getting |
| 03:33.30 | fujin | with DB(Location/${agent}=SIP/${CALLERID(num)}) |
| 03:33.34 | Dovid | Aug 23 20:33:09 NOTICE[999]: pbx.c:1508 pbx_substitute_variables_helper_full: Error in extension logic (missing '}') |
| 03:33.43 | [TK]D-Fender | fujin, getting COLDER.... |
| 03:33.51 | Dovid | TK: can u help ? |
| 03:33.57 | [TK]D-Fender | Dovid, Pastebin it.... |
| 03:35.10 | Dovid | nm |
| 03:35.12 | Dovid | got it working |
| 03:35.17 | Dovid | also TK once I have u |
| 03:35.23 | Dovid | I cant get MOH to work. it starts and stops |
| 03:35.34 | Dovid | I am using Ztdummy and I dont want to use mpg123 |
| 03:35.43 | Dovid | amd yes I built asterisk addons |
| 03:35.52 | [TK]D-Fender | Dovid, timing isn't required for PMoh |
| 03:36.23 | JT | pmoh? |
| 03:36.36 | [TK]D-Fender | typo, DEAL with it! |
| 03:37.21 | JT | i thought you were refering to a new form of moh |
| 03:38.02 | russellb | res_musiconhold will use zaptel for non-files based MOH (such as mpg123 or another app feeding in the MOH) |
| 03:38.24 | russellb | it uses zaptel ticks to determine how often to wake up and read from the stream that is coming from the external app |
| 03:41.08 | [TK]D-Fender | russellb, I had Moh running fine with native & mpg123 each without zaptel at all..... |
| 03:47.15 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
| 03:51.38 | fujin | are the voicemail prompts recorded in any other languages? |
| 03:51.54 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 03:51.56 | fujin | we have a korean and chiense helpdesk queue and it'd be cool if Voicemail(..); could play back to the end-customer in their specific language |
| 03:52.05 | fujin | I mean, out of the box |
| 03:52.10 | fujin | without my dudes recording them |
| 03:55.05 | [TK]D-Fender | fujin, Nothing done by digium. Go check the WIKI, but thats a language I've heard NO news about |
| 03:55.33 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 03:55.49 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
| 03:56.03 | asterisknerds | <PROTECTED> |
| 03:58.14 | fujin | [TK]D-Fender: appreciate it, thanks |
| 03:58.18 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 03:58.26 | fujin | I'll just get my guys to re-record it if they don't like the asterisk womans voice |
| 04:02.58 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 04:06.08 | Sweeper | Allison is getting on my nerves lately |
| 04:07.57 | *** join/#asterisk ptiggerdine (n=ptiggerd@123-243-144-208.tpgi.com.au) |
| 04:09.52 | [TK]D-Fender | I did prefer the pre-trashed recordings... |
| 04:11.33 | Dovid | TK: What would be the 'correct' way to do this |
| 04:11.35 | Dovid | ? |
| 04:11.35 | Dovid | asterisk -rx 'Set(DB(current/my_trunk)=RNK1)' |
| 04:13.32 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 04:15.55 | [TK]D-Fender | Dovid, that is a DIALPLAN app, not a way to call from CLI. perhaps you should try typing in "help" and SEEING what yuo can do :) |
| 04:17.09 | Dovid | lol |
| 04:17.11 | Dovid | just did that |
| 04:17.20 | Dovid | been up for way too long |
| 04:17.23 | Dovid | deadlines !!!!! |
| 04:17.28 | [TK]D-Fender | Dovid, ps... they hideit in the BIG PRINT ;) |
| 04:18.34 | Dovid | hehe |
| 04:29.48 | russellb | [TK]D-Fender: yeah, it should work fine without it, too. It's just a little more efficient if you do have it |
| 04:29.50 | *** join/#asterisk Tunku (n=Tunku@37.64.48.60.trm02-home.tm.net.my) |
| 04:30.17 | Tunku | hi |
| 04:30.18 | russellb | [TK]D-Fender: also, digium does distribute spanish and french prompts in addition to english |
| 04:30.23 | [TK]D-Fender | russellb, ok, why not! :) |
| 04:30.43 | [TK]D-Fender | russellb, "Got Korean" |
| 04:31.05 | russellb | ah. |
| 04:31.27 | russellb | wait, you know what, i *do* have the korean promps. |
| 04:31.32 | Tunku | I am looking for an Operator Station to work with Asterisk. |
| 04:31.35 | russellb | i just forgot to upload them to the ftp site. |
| 04:31.48 | russellb | Tunku: have you tried flash operator panel? |
| 04:32.04 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 04:32.05 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 04:32.34 | Tunku | russellb: read about it...is there any alternative? |
| 04:32.50 | russellb | probably plenty of them |
| 04:33.08 | russellb | i don't know names though |
| 04:33.14 | Tunku | any suggestion before I really test on each of them? |
| 04:33.18 | russellb | it's really hard to keep up with all of the 3rd party apps ... |
| 04:33.30 | Tunku | haha..I understand. |
| 04:34.14 | [TK]D-Fender | Tunku, Go check out the WIKI's GUI list |
| 04:34.53 | Tunku | k. will do. Thanks guys |
| 04:36.16 | heelios | there's, hm, a little bit too much "comfort noise" in my SPA3102-connected handset. Could it be because I used cheap-ass telephone cables to the handset and PSTN? |
| 04:37.13 | jql | I just got one of those in the mail today. I hope not... |
| 04:42.08 | fujin | would Answer();ing an already answered call be bad? |
| 04:42.48 | jql | no |
| 04:44.17 | fujin | how does asterisk mysteriously have the ability to send audio to calls which haven't been answered? |
| 04:44.25 | fujin | does it detect that it should have answered them or something? |
| 04:44.32 | jql | yes |
| 04:44.36 | fujin | seriously? |
| 04:44.48 | jql | while there is a PSTN capability to play audio without answering, Asterisk doesn't let you use it |
| 04:45.10 | jql | asterisk will Answer in order to play your audio |
| 04:45.14 | jql | for the most part |
| 04:45.16 | fujin | I see |
| 04:45.25 | fujin | yes, I seem to be able to do most things without answer(); |
| 04:45.46 | jql | in order to wait for dtmf or things like that, you'd do best to Answer |
| 04:45.56 | jql | but most people play some audio |
| 04:45.59 | jql | which Just Works |
| 04:46.18 | *** join/#asterisk webman (n=chatzill@124.246.8.196.static.nexnet.net.au) |
| 04:46.53 | Tunku | jql: Are you saying Asterisk can play 'voice prompt' without answering a PSTN call? |
| 04:47.21 | jql | no way. I'm saying it will in some cases Answer it implicitly |
| 04:47.40 | Qwell | Tunku: That's how SIT tones are done. |
| 04:47.40 | Qwell | ~sit |
| 04:47.40 | jbot | ACTION sits |
| 04:47.44 | denon | haha |
| 04:47.47 | Qwell | ... |
| 04:47.52 | Qwell | Special Information Tones |
| 04:47.58 | Qwell | kinda redundant there |
| 04:48.13 | denon | http://en.wikipedia.org/wiki/Special_information_tone |
| 04:48.26 | Tunku | kk.will check on it. |
| 04:49.16 | Qwell | also the messages like "Your call could not be completed as dialed" |
| 04:49.25 | Qwell | that's all early media, in most cases |
| 04:49.52 | jql | essentially, Answer starts billing, and SIT and such noises don't require payment |
| 04:49.57 | Tunku | oh. ok. This msg usually play by the telco |
| 04:50.10 | jql | some tricky people at AT&T do entire IVR systems without answering |
| 04:50.13 | Tunku | can we do it in Asterisk? how? |
| 04:50.23 | Qwell | AT&T hell.. a lot of toll free numbers do it |
| 04:50.27 | Qwell | banks, etc |
| 04:50.28 | jql | yeah |
| 04:50.32 | denon | Qwell: hey, this is a handy site: http://www.thisisarecording.com/sounds19.shtml |
| 04:50.43 | Qwell | nice |
| 04:50.50 | jql | their opinion is that if you don't actually connect the call with a human, they don't *necessarily* have to pay for it |
| 04:50.51 | jql | very funny |
| 04:51.18 | denon | yeah, until you get a knock on the door from the fcc |
| 04:51.24 | jql | exactly |
| 04:51.26 | Tunku | this is good (i means from biz point).. |
| 04:51.32 | Qwell | http://www.thisisarecording.com/sounds/wav/9164400030 |
| 04:51.35 | Qwell | disturbing |
| 04:51.53 | denon | haha |
| 04:52.06 | jql | however, the FCC isn't the real telco whippers. it's the local governments that want a piece of you. hence why 800-numbers get away with that stuff. :) |
| 04:52.11 | jql | no local government to piss off |
| 04:52.14 | denon | Qwell: I guess they didnt like hookers setting up inbound call centers |
| 04:52.16 | Tunku | how can I test it with asterisk? |
| 04:52.19 | Qwell | heh |
| 04:52.43 | JT | hrm |
| 04:52.58 | JT | i thought asterisk could playback without answering implicitly |
| 04:53.35 | [TK]D-Fender | ~areyouagdog ? |
| 04:53.45 | [TK]D-Fender | ~areyouadog ? |
| 04:53.46 | jbot | Bark! Bark! |
| 04:53.52 | [TK]D-Fender | ~botsnack |
| 04:53.52 | jbot | [TK]D-Fender: thanks |
| 04:54.11 | jql | if asterisk had early media capability, there would be a ringback.conf file. :) |
| 04:54.16 | jql | well, configurable |
| 04:54.24 | jql | not that it doesn't have *any* |
| 04:54.33 | JT | jql: or you could just not Answer |
| 04:54.43 | JT | i'm not sure if an answer is implicit or not |
| 04:54.43 | jql | doesn't work for me, PRI or SIP |
| 04:54.55 | jql | sip gets the 200, and PRI gives the stupid answer thing |
| 04:54.57 | JT | hmm okay |
| 04:55.00 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 04:55.30 | asterisknerds | <PROTECTED> |
| 04:56.29 | webman | jt: on a pri you can send early media, but that depends on if your telco will allow you |
| 04:56.40 | Qwell | ahh, a classic.. http://www.thisisarecording.com/sounds/mp3/8003998378 |
| 04:57.20 | JT | webman: i realise, the discussion is with relation to asterisk though |
| 04:57.41 | webman | yes, using asterisk, I can send audio on a pri without answer |
| 04:58.01 | jql | really? and you don't get billed for it? |
| 04:58.27 | webman | jql: correct |
| 04:58.39 | jql | what version of asterisk? |
| 04:58.55 | webman | however, if I call from some (not all) mobile phone providers then the remote end doesn't hear the audio |
| 04:59.26 | webman | jql: all versions of asterisk since 1.0.x |
| 04:59.48 | webman | not that I have specifically tested this recently on 1.4, but I doubt there is anything that has removed this feature |
| 05:05.00 | Tunku | webman: Am I right saying that u just play some audio with answering the call to make it happen and this is eventually depends whether the telco allow it or not? |
| 05:05.09 | Tunku | without* |
| 05:05.51 | webman | tunku: yep, something like exten => 12345678,1,playback(something) and then in step 2, do the answer/etc |
| 05:06.12 | Tunku | k. thanks. will try it |
| 05:09.38 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-243-162-33.bflony.east.verizon.net) |
| 05:10.12 | *** join/#asterisk Strom_M (n=strom@static-68-236-161-53.ny325.east.verizon.net) |
| 05:10.44 | SuPrSluG | hi |
| 05:11.22 | SuPrSluG | i'm get no ringing sound when i call another internal extension |
| 05:11.29 | SuPrSluG | any ideas |
| 05:12.14 | SuPrSluG | i've tried cmd Ringing and r after dial cmd. nothing |
| 05:16.35 | JT | SuPrSluG: you haven't even told us what is doing the ringing |
| 05:17.30 | SuPrSluG | when the caller places a call they hear nothing. not even crickets |
| 05:17.40 | JT | yawn |
| 05:17.45 | JT | stop being obtuse |
| 05:17.48 | JT | DETAILS please |
| 05:18.01 | SuPrSluG | the called phone rings fine |
| 05:18.07 | JT | great |
| 05:19.09 | webman | suprslug: paste a copy of the console output to pastebin.ca |
| 05:19.30 | JT | before that |
| 05:19.33 | JT | DETAILS :) |
| 05:19.39 | JT | calling what to what |
| 05:19.39 | JT | etc |
| 05:19.55 | webman | then we might have some details to actually decide what is going wrong ... or as JT said, don't be obtuse :) |
| 05:22.35 | SuPrSluG | http://pastebin.ca/668851 |
| 05:23.13 | JT | SuPrSluG: you still haven't explained your setup |
| 05:23.26 | JT | until you do so, you are wasting both your own time and ours |
| 05:24.10 | *** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com) |
| 05:25.50 | SuPrSluG | i'm just calling between 2 sip phones. everything works, but no ringtones are heard by the caller. callee's phone rings normally |
| 05:26.07 | JT | what phones? |
| 05:27.28 | Tunku | This is the warning "Aug 24 01:19:58 WARNING[1376]: channel.c:2097 ast_indicate: Unable to handle indication 3 for 'SIP/2002-10043088'" |
| 05:28.54 | *** join/#asterisk jarod14 (n=jarod14@mail.viatelecom.com) |
| 05:28.58 | *** join/#asterisk PC_Clone (n=pc_clone@c-76-16-79-154.hsd1.il.comcast.net) |
| 05:29.54 | jarod14 | hi guys |
| 05:30.19 | CCFL_Man2 | x86: so you have regular pots phones connected to a channel bank for your channel bank? |
| 05:31.05 | Tunku | webman: Follow your method but Asterisk will answer the call eventhought the Answer() was not used. |
| 05:31.17 | *** join/#asterisk zutoo (n=zutoo@74.94.57.209) |
| 05:31.36 | zutoo | Anyone know how to get the PC port working on an Aastra phone? |
| 05:31.55 | webman | tunku: what interface are you using on both ends?? ie, calling from pri to pri or mobile to pri or what ?? |
| 05:32.20 | Tunku | oh..tested using SIP to Asterisk |
| 05:32.31 | Tunku | it must be on PRI? |
| 05:33.05 | webman | tunku: I've never tested with sip, only PRI |
| 05:33.23 | Tunku | in that case, I can only test it on Monday...back in office |
| 05:34.51 | CCFL_Man2 | so with isdn, what is "fxo" and "fxs", what are they called? |
| 05:35.09 | CCFL_Man2 | one receives isdn battery, one generares it |
| 05:35.15 | webman | fxo and fxs are not isdn they are pstn |
| 05:35.33 | webman | oh, maybe you mean TE and NT ?? |
| 05:35.37 | CCFL_Man2 | webman: i know, but what is the isdn equivilent? |
| 05:35.48 | Tunku | network or CPE |
| 05:36.08 | CCFL_Man2 | network |
| 05:36.38 | Tunku | ?? in ISDN, you need to configure your port as Network or CPE |
| 05:36.47 | Tunku | Network= provide clocking |
| 05:37.13 | CCFL_Man2 | oh |
| 05:37.33 | JT | webman: not PSTN |
| 05:37.40 | JT | s/pstn/POTS/ |
| 05:39.23 | CCFL_Man2 | well, if i were to put isdn card in my channel bank, they would provide clocking, whats the name for than? |
| 05:40.30 | Tunku | sry. no experience with Channel Bank. Others can help? |
| 05:41.07 | JT | CCFL_Man2: what's the real question? |
| 05:41.47 | CCFL_Man2 | JT: whats it called, an isdn trunk |
| 05:41.58 | zutoo | Anyone have experience with the PC port on Aastra phones? |
| 05:42.42 | CCFL_Man2 | a pots trunk would be fxo |
| 05:44.19 | *** join/#asterisk saftsack (n=oliver@p54A7C28A.dip.t-dialin.net) |
| 05:44.25 | JT | no |
| 05:44.30 | JT | only one end is fxo |
| 05:44.33 | JT | it is POTS |
| 05:44.36 | JT | analogue |
| 05:45.29 | CCFL_Man2 | right, but what would be the isdn equivilent |
| 05:47.50 | JT | for what |
| 05:47.53 | JT | it's isdn |
| 05:47.58 | JT | one side acts are CPE |
| 05:48.01 | JT | the other as network |
| 05:48.31 | Dovid | thanks guys for all the help |
| 05:49.11 | JT | s/are/as/ |
| 05:49.38 | CCFL_Man2 | JT: well, usually with a T1, the end hats providing the clocking technically is the one that's supplying voltage |
| 05:50.02 | JT | what voltage? |
| 05:50.12 | CCFL_Man2 | in a short haul ds x signal |
| 05:50.14 | JT | T1/E1 is just signalling |
| 05:50.20 | JT | well data |
| 05:50.41 | JT | yeah sorry, you're not making much sense |
| 05:50.42 | CCFL_Man2 | i'm talking about the electrical interface |
| 05:50.45 | JT | both ends send data |
| 05:50.51 | JT | that is a small amount of voltage |
| 05:51.25 | CCFL_Man2 | actually thats not the right analogy |
| 05:51.44 | JT | what analogy? |
| 05:52.04 | CCFL_Man2 | network and cpe electrical interface |
| 05:52.28 | CCFL_Man2 | because ds1 uses two channels |
| 05:52.28 | JT | sense not maketh |
| 05:52.57 | CCFL_Man2 | i'm talking about the electrical interface, not the signalling |
| 05:53.06 | JT | yes then talk about it |
| 05:53.14 | JT | then vaguley babling stuff about voltage |
| 05:53.38 | *** join/#asterisk The_LightSide (n=JBouncer@dsl-241-79-152.telkomadsl.co.za) |
| 05:53.40 | CCFL_Man2 | with a DS1 interface, it uses two channels |
| 05:53.57 | JT | electrical has no concept of channels |
| 05:54.27 | CCFL_Man2 | two electrical channels, tx and rx |
| 05:54.36 | JT | ... |
| 05:54.45 | JT | it uses 2 PAIRS, 4 WIRES |
| 05:54.47 | JT | not 2 channels |
| 05:54.48 | JT | fark |
| 05:54.58 | CCFL_Man2 | right, pairs |
| 05:55.10 | CCFL_Man2 | channe;s is the wromg term |
| 05:55.21 | CCFL_Man2 | isdn uses 1 pair |
| 05:55.22 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 05:55.31 | JT | wrong |
| 05:55.59 | CCFL_Man2 | to the cpe it does |
| 05:56.27 | fujin | isdn is two pair dude |
| 05:56.30 | fujin | 4 wires |
| 05:56.39 | JT | he should be more specific |
| 05:56.44 | JT | perhaps he means BRI |
| 05:56.46 | fujin | mm |
| 05:56.49 | fujin | a PRI = 4wires, 2pair |
| 05:57.00 | CCFL_Man2 | oh |
| 05:57.00 | JT | in which case BRI uses 1 pair to the NT1 |
| 05:57.10 | JT | but the S/T-bus is 2 pair |
| 05:57.11 | CCFL_Man2 | right |
| 05:57.21 | JT | the U bus is 1 pair |
| 05:57.31 | JT | PRI is often also delivered over 1 pair |
| 05:57.39 | JT | but the actual interface is 2 pair |
| 05:58.05 | CCFL_Man2 | JT: delivered over hdsl? |
| 05:58.15 | JT | yeah SHDSL/HDSL |
| 05:58.23 | CCFL_Man2 | ahh |
| 05:58.24 | JT | BRI does a similar thing |
| 05:58.46 | JT | the NT1 has the modem in it |
| 05:59.45 | JT | for future reference, both BRI and PRI are ISDN |
| 05:59.53 | CCFL_Man2 | JT: so it goesfrom the telco, to the nt1 over the coper plant to another nt1, then the cpe? |
| 06:00.03 | CCFL_Man2 | right |
| 06:00.21 | JT | it would not be an NT1 at the exchange |
| 06:00.32 | CCFL_Man2 | right |
| 06:03.05 | CCFL_Man2 | isdn i never could fully understand |
| 06:03.17 | JT | why, it's simple |
| 06:03.40 | hermuli | i have a small problem with mysql&asterisk... asterisk keeps freezing when the mysql connection fails to connect... |
| 06:04.21 | CCFL_Man2 | JT: i want to understand the telco and cpe interfaces |
| 06:04.39 | *** join/#asterisk saftsack (n=oliver@p54A7FAD6.dip.t-dialin.net) |
| 06:04.48 | hermuli | i think the problem might be with mysql or debian settings since there's no such problem with a mysql db hosted by a web sote hosting company... |
| 06:05.12 | hermuli | so if anyone has tackled this kind of thing or has experience i'd really love to get some help |
| 06:06.16 | JT | CCFL_Man2: there's heaps of material online |
| 06:07.34 | CCFL_Man2 | JT: reading wikipedia now |
| 06:08.36 | CCFL_Man2 | so from the telco it's the ISDN Uk0 interface, that goes into the network terminator 1 then into the cpe |
| 06:08.46 | CCFL_Man2 | the two wire interface |
| 06:09.33 | JT | U interface, but sure |
| 06:09.38 | JT | you're talking about BRI |
| 06:10.20 | CCFL_Man2 | exactly |
| 06:10.54 | CCFL_Man2 | with PRI, it's isdn signalling and carried over a DS1 |
| 06:11.19 | JT | make sure you refer specifically |
| 06:11.25 | CCFL_Man2 | right |
| 06:11.31 | JT | isdn doesn't specify anything in particular |
| 06:11.39 | JT | PRI can be over G.703 |
| 06:11.48 | JT | in fact it is in most parts of the world |
| 06:12.15 | CCFL_Man2 | because on a DS1, you can have ISDN PRI, CAS, etc |
| 06:12.21 | CCFL_Man2 | right |
| 06:12.39 | CCFL_Man2 | see, when i think of isdn, i think of the two wire interface |
| 06:13.24 | JT | just call it a T1, much easier |
| 06:14.39 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 06:16.10 | CCFL_Man2 | i just need to familiarize myself with the interfaces |
| 06:17.13 | CCFL_Man2 | because i think of a T1 as a DS1, either a DS1 for long haul, DSX1 for the short haul |
| 06:17.48 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 06:18.22 | JT | too much american terminology crap for T1 that no-one else bothers with :) |
| 06:18.32 | JT | if you want to learn telecommunications http://web.archive.org/web/20040419051144/www.ericsson.com/support/telecom/part-b/index.shtml |
| 06:18.47 | JT | sorry |
| 06:18.50 | JT | http://web.archive.org/web/20040413074912/www.ericsson.com/support/telecom/index.shtml <---- |
| 06:22.43 | CCFL_Man2 | thanks |
| 06:24.10 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 06:24.33 | CCFL_Man2 | i want to find an isdn to pots adapter |
| 06:24.56 | JT | you mean an NT1+? |
| 06:26.07 | CCFL_Man2 | thats what they are called? |
| 06:26.36 | *** join/#asterisk saftsack (n=oliver@p54A7FA6D.dip.t-dialin.net) |
| 06:27.10 | CCFL_Man2 | i thought there would be one that connects to the NT1 |
| 06:29.17 | CCFL_Man2 | http://cgi.ebay.com/Polycom-NT3620-Triple-NTI-for-ISDN_W0QQitemZ300141981037QQihZ020QQcategoryZ162QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
| 06:29.27 | CCFL_Man2 | something like that |
| 06:30.14 | *** join/#asterisk jkrueger (n=jkrueger@ip66-104-41-250.z41-104-66.customer.algx.net) |
| 06:32.53 | JT | NT1+ have POTS breakout as well as S-bus |
| 06:34.29 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:34.30 | CCFL_Man2 | ahh, ok |
| 06:35.25 | Wonka | hrm. i've just read about autofill in queues.conf and it sounds somehow stupid to me... shouldn't the queue stuff just ring either all or as many agents as there are callers waiting and the first agent to pick up gets connected the first caller, the second the second, and so on? |
| 06:37.29 | CCFL_Man2 | can't find any on ebay |
| 06:38.13 | mial | good morning |
| 06:39.49 | *** join/#asterisk RsaMan (n=aa@196.210.154.3) |
| 06:40.02 | RsaMan | hi |
| 06:40.11 | jql | howdy |
| 06:40.24 | RsaMan | is cdr odbc required when i connect my cdr to a postgres database? |
| 06:40.31 | CCFL_Man2 | so ISDN BRI S/T plugs into say a cisco isdn bri wic? |
| 06:41.31 | CCFL_Man2 | or any isdn bri cpe? |
| 06:41.34 | Tunku | RsaMan: No |
| 06:41.34 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 06:41.56 | Tunku | RsaMan: r u using cdr_postgres? |
| 06:42.03 | *** join/#asterisk coldsteal (n=coldstea@unaffiliated/coldsteal) |
| 06:42.05 | coldsteal | hello |
| 06:42.27 | CCFL_Man2 | Tunku: were you the one with the adit 600 stuff? |
| 06:42.40 | coldsteal | i notice in my councle i keep getting this error [Aug 24 01:27:03] WARNING[7846]: chan_sip.c:12147 handle_response_register: Got 200 OK on REGISTER that isn't a register |
| 06:42.56 | *** join/#asterisk Aurs (n=Aurs@1ult2p8.ip.hipercom.no) |
| 06:42.59 | Tunku | CCFL: no. wrong person. :) |
| 06:44.01 | RsaMan | Tunku : yes |
| 06:44.08 | CCFL_Man2 | ahh, you don't play with channel banks? |
| 06:44.11 | RsaMan | Tunku : it does not seem to connect to my postgres databse |
| 06:44.27 | CCFL_Man2 | they are cool little toys that are carrier class :P |
| 06:44.43 | RsaMan | Tunku : if i set my verbosity high will it give an error if it cannot connect |
| 06:44.44 | RsaMan | ? |
| 06:45.01 | Tunku | can u connect to Postgresql manually? |
| 06:45.14 | RsaMan | Tunku : yes , i have a test php file |
| 06:45.17 | Tunku | on the OS command prompt |
| 06:45.20 | RsaMan | Tunku : it works fine |
| 06:45.29 | RsaMan | using su postgres? |
| 06:45.32 | RsaMan | psql |
| 06:46.23 | Tunku | ya. make sure user hv the rights to access Postgresql |
| 06:46.53 | Tunku | dun use Trust authentication |
| 06:47.10 | RsaMan | i am logged on as root? |
| 06:47.21 | RsaMan | wouldnt that be enough? |
| 06:47.54 | *** join/#asterisk saftsack (n=oliver@p54A7F755.dip.t-dialin.net) |
| 06:48.21 | Tunku | RsaMan: wait.. accessing my Linux |
| 06:48.22 | coldsteal | i keep getting this in my apterisk -r http://rafb.net/p/whx9f860.html |
| 06:48.48 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 06:49.18 | Tunku | hv u configure properly the pg_hba.conf ? |
| 06:50.26 | Tunku | and also hv you create CDR table in postgresql? |
| 06:50.46 | RsaMan | yes, but i will double check |
| 06:51.09 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 06:51.35 | *** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com) |
| 06:52.06 | Tunku | check also the postgresql.conf to ensure asterisk server allow to access. Default only localhost can access |
| 06:53.20 | CCFL_Man2 | what kind of connectors are they called that look like low density centronix connectors which are used in telephony? |
| 06:53.46 | RsaMan | arg |
| 06:53.51 | RsaMan | pastebin.ca isdown |
| 06:55.09 | RsaMan | http://pastebin.com/m3e5bd4a5 |
| 06:55.16 | RsaMan | it seems to be correct |
| 06:55.22 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 06:55.23 | RsaMan | and my postgres database is on my server |
| 06:55.33 | RsaMan | so localhost should not be an issue |
| 06:55.41 | RsaMan | localhost access |
| 06:55.52 | asterisknerds | <PROTECTED> |
| 06:56.04 | *** join/#asterisk McDouglas (n=mcd@mmcomp.adsl.datanet.hu) |
| 06:57.51 | Tunku | try change trust to password authentication for localhost |
| 06:58.05 | McDouglas | hi |
| 06:58.10 | McDouglas | i'm having some problems with my dialplan |
| 06:58.11 | McDouglas | http://pastebin.com/d6f9896bb |
| 06:58.38 | McDouglas | if i enter the extension number after the greeting message nothing happens |
| 06:58.41 | RsaMan | is that setup in pg_hba.conf |
| 06:59.53 | Tunku | ya. hv u done with cdr_pgsql.conf in /etc/asterisk? |
| 07:00.34 | RsaMan | yes |
| 07:00.47 | Tunku | can show us? |
| 07:01.16 | Tunku | I mean without the actual user & pwd |
| 07:01.30 | RsaMan | yes |
| 07:02.07 | RsaMan | http://pastebin.com/m681cc7d8 |
| 07:02.16 | RsaMan | no password, |
| 07:02.25 | RsaMan | just using the standard postgres user for now |
| 07:03.06 | Tunku | where is your table? |
| 07:03.17 | RsaMan | in the asterisk databse |
| 07:03.25 | RsaMan | database called asterisk * |
| 07:03.57 | Tunku | in cdr_pgsql.conf, there should be an entry "table=cdr" |
| 07:04.04 | RsaMan | o |
| 07:04.37 | RsaMan | i will add that |
| 07:05.03 | Tunku | if still can not, then pls use a password |
| 07:05.59 | RsaMan | need to go, thanks for the help,, i hope that resolves the issue |
| 07:06.01 | RsaMan | have a nice day |
| 07:06.02 | RsaMan | bye bye |
| 07:06.13 | Tunku | bb |
| 07:08.32 | flenders | McDouglas: are you using sip? |
| 07:09.33 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 07:09.58 | coldsteal | i keep getting this in my apterisk -r http://rafb.net/p/whx9f860.html |
| 07:10.19 | coldsteal | how do i fix it? |
| 07:10.58 | *** join/#asterisk pkunkra (n=chris@cpe-74-73-28-89.nyc.res.rr.com) |
| 07:11.16 | *** join/#asterisk saftsack (n=oliver@p54A7E5DA.dip.t-dialin.net) |
| 07:15.07 | McDouglas | flenders: well, one extension is a zap channel the other is sip |
| 07:15.14 | McDouglas | but it doesnt matter which one i call |
| 07:15.53 | flenders | the one dialing in is sip? |
| 07:16.07 | flenders | try changing the dtmfmode on sip.conf |
| 07:16.26 | McDouglas | no, i'm calling from a cell phone |
| 07:16.31 | flenders | dtmfmode=rfc2833 |
| 07:16.57 | McDouglas | there is an analog line connected to the zap channel |
| 07:17.08 | flenders | ok |
| 07:17.16 | McDouglas | actually, it hangs up even before i would dial an extension |
| 07:17.23 | RGi_26 | hmm anyone using zap-hfc og isdn here ? |
| 07:17.30 | flenders | what about the sip number |
| 07:17.44 | McDouglas | as i said, it hangs up before i could dial the extension |
| 07:18.36 | flenders | do you wait for the enter-ext-of-person sound to finish? |
| 07:18.41 | JT | CCFL_Man2: what is the connector used for? |
| 07:18.57 | McDouglas | yeah it does finish |
| 07:19.12 | *** join/#asterisk af_ (n=getsmart@81-174-47-128.dynamic.ngi.it) |
| 07:19.15 | flenders | try adding: exten => s,n,WaitExten(10) |
| 07:19.31 | McDouglas | actualy, its funny because it rings 2 times, and by the time it picks it up first half of the speech is missing |
| 07:19.45 | flenders | that will wait for input for 10 seconds |
| 07:19.51 | CCFL_Man2 | JT: connecting the tip and ring pairs to a punch block |
| 07:20.07 | McDouglas | after s,2 line? |
| 07:20.11 | flenders | also add: exten => s,n,Wait(2) before playing the file |
| 07:20.14 | JT | dunno, maybe a telco connector |
| 07:20.20 | JT | or 80 pin centronics |
| 07:20.22 | JT | who knows |
| 07:20.34 | flenders | waitexten has to be after playing the file |
| 07:20.42 | flenders | the Wait(2) has to be before |
| 07:21.16 | CCFL_Man2 | JT: i was gonna get them on ebay but i think i can get a few from work, i just want the cable |
| 07:22.07 | McDouglas | flenders: thank you, got it fixed ;) |
| 07:22.39 | McDouglas | wonder why it was not in that o'reilly asterisk bok... |
| 07:22.40 | JT | ah ok |
| 07:23.19 | CCFL_Man2 | JT: want one of those rj11 panels or boxes to connect it to |
| 07:23.23 | flenders | McDouglas: not everything is in there |
| 07:23.39 | JT | CCFL_Man2: a breakout box |
| 07:23.44 | JT | <PROTECTED> |
| 07:23.46 | JT | panel |
| 07:23.50 | JT | <PROTECTED> |
| 07:23.52 | CCFL_Man2 | err, yeah |
| 07:23.58 | flenders | McDouglas: book is great to learn concepts, it's not meant to be used as a guide |
| 07:24.06 | CCFL_Man2 | well, breakout panel |
| 07:24.27 | McDouglas | what is meant to be a guide then? |
| 07:24.54 | McDouglas | where is the manual then, so to speak? :P |
| 07:25.05 | flenders | McDouglas: there isn't such a thing |
| 07:25.19 | McDouglas | cool :\ |
| 07:25.25 | JT | show application blah |
| 07:25.39 | flenders | well, I'm guessing you're a linux admin? |
| 07:25.52 | McDouglas | yes |
| 07:25.58 | flenders | I've never seen a manual for iptables, for example |
| 07:26.14 | flenders | man pages are just for reference |
| 07:26.32 | flenders | that's what makes us special. |
| 07:26.47 | flenders | and that's why it's so damn difficult for windows admins to become linux admins |
| 07:27.17 | flenders | no manuals, no intructions. read man pages and LEARN |
| 07:27.36 | McDouglas | well, its kinda hard to migrate to a new platform if the only guide i have are some man pages :\ |
| 07:27.42 | McDouglas | i mean pbx platform |
| 07:29.07 | flenders | well, you can find instructions on how to do everything, even building a decent dialplan |
| 07:29.47 | Tunku | go wiki for asterisk. u can find the sky |
| 07:30.19 | flenders | but you won't find a printed manual |
| 07:30.29 | flenders | if you buy asterisk business you get one, I think |
| 07:30.35 | flenders | but not on the free version |
| 07:31.10 | JT | the book is good enough for most manualish stuff |
| 07:31.35 | Tunku | well...if a person need to learn, then he will hv to spend time reading from the screen |
| 07:31.42 | *** join/#asterisk tsurko (n=tsurko@213.91.216.130) |
| 07:31.54 | Tunku | I learned all my stuf from wiki |
| 07:32.13 | *** join/#asterisk saftsack (n=oliver@p54A7D894.dip.t-dialin.net) |
| 07:32.25 | flenders | well, I read the book about 9 months ago |
| 07:32.28 | flenders | it was a great start |
| 07:32.32 | SwK | is CCFL_Man2 trolling again? |
| 07:32.59 | CCFL_Man2 | SwK: always |
| 07:33.48 | *** part/#asterisk hank (n=hank@netwichtig.de) |
| 07:33.50 | CCFL_Man2 | SwK: i got the voip router connected to the channel bank without any alarms |
| 07:34.20 | SwK | you passed any calls thru it yet? |
| 07:35.28 | CCFL_Man2 | CAS signalling, b8zs line coding, esf framing, the voip router provides the clock source to the channel bank |
| 07:35.46 | CCFL_Man2 | not yet, i need to upgrade the ios first |
| 07:36.25 | CCFL_Man2 | i can't set the dial peers yet |
| 07:36.39 | CCFL_Man2 | ios doesn't have that feature |
| 07:36.57 | SwK | so head over to cco and get it |
| 07:37.15 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 07:37.21 | CCFL_Man2 | i did, but i need to upgrade the ram to 64mb |
| 07:37.43 | CCFL_Man2 | and to do that i need to upgrade the prom |
| 07:38.19 | CCFL_Man2 | so i ebayed the prom and ram and waiting for bith of those |
| 07:38.42 | CCFL_Man2 | then i can set up a dial peer and test out calls |
| 07:40.24 | CCFL_Man2 | with this ios when you set themode to cas it doesn't get you into a cas sub configuration |
| 07:41.52 | CCFL_Man2 | pain in the ass silly shit |
| 07:42.25 | CCFL_Man2 | oh, and i'm not sure if i need the compression module or not |
| 07:44.52 | SwK | didnt I tell you to just buy a as5300 and be done with it? heh |
| 07:46.12 | Tunku | bye everyone |
| 07:46.36 | McDouglas | why is that, if I use Hangup( ), my cell phone does not realise that the call has ended? |
| 07:47.17 | JT | McDouglas: you shouldn't have whitespace in there, but it should still work, maybe your telco just sucks |
| 07:53.50 | *** join/#asterisk saftsack (n=oliver@p54A7C938.dip.t-dialin.net) |
| 07:55.33 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 07:56.03 | asterisknerds | <PROTECTED> |
| 07:56.07 | *** join/#asterisk Bladerunner05 (n=feelme@81-174-56-54.static.ngi.it) |
| 07:57.19 | Bladerunner05 | Hello I'm using * 1.4.9 and TDP400p (4FXO) when I call * from outside and without respond hang up the call my extension ring for other 2 or 3 time before stop |
| 07:57.38 | Bladerunner05 | What I have to check for this problem ?= |
| 08:00.04 | *** join/#asterisk shtoom (n=shtoom@59.181.111.61) |
| 08:02.54 | *** join/#asterisk ajohnstone (n=ajohnsto@85.189.117.98) |
| 08:03.56 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
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| 08:12.13 | *** join/#asterisk saftsack (n=oliver@p54A7F196.dip.t-dialin.net) |
| 08:14.41 | *** join/#asterisk dharrigan (n=david@host12.williamhill.co.uk) |
| 08:17.38 | *** join/#asterisk Airwolf- (n=ibro@221.132.242.183) |
| 08:18.39 | Airwolf- | i write a macro to get peer ip address who picks up the call in a ring group. it's working, ijust want to know if this is the proper way to do it. http://pastebin.com/d36c08912 |
| 08:20.26 | Airwolf- | also, an extension in a macro is local to the macro itself or not ? |
| 08:34.16 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 08:36.30 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 08:36.40 | s0ck | hmm |
| 08:36.45 | s0ck | should it be possible to pick up a queue |
| 08:36.55 | s0ck | as opposed to picking up an extension |
| 08:37.08 | s0ck | if the queue is ringing, could be ringing any/all extensions |
| 08:38.03 | Airwolf- | pick up a queue ? |
| 08:39.05 | s0ck | say i have 4 people in a queue |
| 08:39.13 | s0ck | persons 5 and 6 sit outside the queue |
| 08:39.18 | s0ck | queue rings |
| 08:39.25 | s0ck | 5 or 6 want to pick up the call |
| 08:39.36 | s0ck | how would one achieve this |
| 08:40.08 | Airwolf- | queueaddmember ? |
| 08:40.19 | s0ck | i dont want them to be members of the queue tho |
| 08:40.26 | s0ck | just be able to pick up if need be |
| 08:40.34 | Airwolf- | transfer the call ? |
| 08:40.39 | s0ck | **queuenumber doesn't seem to work |
| 08:43.16 | *** join/#asterisk yassaccan (n=yassacca@admin130.hgo.se) |
| 08:44.02 | Bladerunner05 | Why I see Hungup 'Zap/1-1' but it really don't hang up ? |
| 08:44.10 | Airwolf- | add and then remove them ? |
| 08:45.51 | *** join/#asterisk thansen|laptop (n=thansen@69.233.sfcn.org) |
| 08:48.25 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:49.43 | *** join/#asterisk dds (n=dds@41.98.156.220.st.bbexcite.jp) |
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| 08:55.33 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 08:56.03 | asterisknerds | <PROTECTED> |
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| 09:44.15 | hermuli | anyone has any idea how i should configure mysql for asterisk? |
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| 09:55.22 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 09:55.52 | asterisknerds | <PROTECTED> |
| 09:56.01 | sty|bong | <PROTECTED> |
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| 10:07.38 | dds | any developers around? |
| 10:07.56 | dds | if so, if you have a moment, please review http://bugs.digium.com/view.php?id=10546 |
| 10:38.35 | RGi_26 | tzafrir : here ? |
| 10:43.02 | Bladerunner05 | Hello I'm using * 1.4.9 and TDP400p (4FXO) when I call * from outside and without respond hang up the call my extension ring for other 2 or 3 time before stop |
| 10:50.48 | tzafrir_laptop | yes |
| 10:52.31 | tzafrir_laptop | RGi_26, here |
| 10:55.13 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 10:55.43 | asterisknerds | <PROTECTED> |
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| 11:46.43 | Wonka | my queues.conf "joinempty" doesn't work as expected - i want callers to join a queue with possibly no agents - if there are no agents for 300s, they may drop back out, but they should stay on these 300s. all i get is a QUEUESTATUS of TIMEOUT or JOINEMPTY. |
| 11:54.09 | *** join/#asterisk zepmantra (n=sdadsa@125.212.110.114) |
| 11:54.24 | *** join/#asterisk lirakis (n=etamme@65.200.191.253) |
| 11:54.33 | lirakis | morning |
| 11:55.17 | zepmantra | good day, we have a setup adtran(atlas 550) connected to a asterisk server with TE205P, when we try to call the asterisk server all we get is busy signal no output from the CLI |
| 11:56.45 | *** join/#asterisk RsaMan (n=aa@196.210.154.3) |
| 11:56.51 | RsaMan | hi |
| 11:57.11 | RsaMan | has anyone successfully connected asterisk cdr to a postgres database? |
| 11:57.14 | *** join/#asterisk masus (n=tet@88.248.14.186) |
| 11:57.17 | masus | hi all |
| 11:57.36 | masus | how to parse the year from ${DATETIME} . THANKS A LOT :) |
| 11:58.37 | masus | for use as variable |
| 12:00.07 | RsaMan | is anyone familiar with transfering calls |
| 12:00.16 | RsaMan | my blind transfer seeems to drop both parties |
| 12:00.22 | RsaMan | i am looking for an alternative |
| 12:00.39 | RsaMan | because i am running out of ideas to fix my blind transfer issue |
| 12:00.45 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:00.49 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:01.02 | McDouglas | i installed a b410p card |
| 12:01.07 | McDouglas | asterisk can detect it |
| 12:01.18 | McDouglas | but if i connect the isdn lines |
| 12:01.22 | McDouglas | the redas are still red |
| 12:01.24 | McDouglas | what gives? |
| 12:01.29 | McDouglas | *leds |
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| 12:03.20 | RsaMan | lol |
| 12:03.24 | RsaMan | all the gurus are mia |
| 12:04.47 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:04.47 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 12:04.48 | mial | good morning |
| 12:04.52 | masus | morning |
| 12:05.42 | masus | any IDEA how to parse the year from ${DATETIME} for use as variable ? |
| 12:06.08 | mial | hm |
| 12:06.13 | mial | what looks DATETIME like ? |
| 12:06.54 | masus | one mom i'll paste |
| 12:07.00 | kaldemar | masus: DATETIME is deprecated, use STRFTIME in stead: http://www.voip-info.org/wiki/view/Asterisk+variables |
| 12:07.06 | masus | ${DATETIME} * Current date time in the format: DDMMYYYY-HH:MM:SS |
| 12:07.11 | Wonka | ${DATETIME} * Current date time in the format: DDMMYYYY-HH:MM:SS (Deprecated; use ${STRFTIME(${EPOCH},,%d%mNaVH:NaVS)}) |
| 12:07.29 | masus | yes i have see it but can't parse the year value for example 2007 |
| 12:07.44 | Wonka | i'd not parse DATETIME, but use STRFTIME with %Y or whatever |
| 12:07.57 | masus | have test it but always the same |
| 12:08.50 | kaldemar | masus: but if you have to use DATETIME, ${DATETIME:4:4}. |
| 12:08.52 | *** join/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 12:08.55 | masus | hmm |
| 12:08.58 | Wonka | http://www.voip-info.org/wiki/index.php?page=Asterisk+func+strftime |
| 12:08.58 | masus | that is it |
| 12:09.02 | masus | i think |
| 12:09.11 | masus | thanks all |
| 12:09.17 | Wonka | %Y The year as a decimal number including the century. |
| 12:09.50 | masus | ${DATETIME:4:4} that was the answer ... thanks all |
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| 12:25.27 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
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| 12:36.18 | *** join/#asterisk Rsaman2 (n=aa@196.210.154.3) |
| 12:36.20 | Rsaman2 | bleh |
| 12:36.23 | Rsaman2 | was disonnected |
| 12:36.34 | Rsaman2 | have real issues with my blind transfer |
| 12:36.37 | Rsaman2 | :( |
| 12:36.43 | Rsaman2 | which function does it use ? |
| 12:37.20 | Rsaman | my blind transfer cuts both partys off |
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| 12:42.35 | Rsaman | :( |
| 12:42.38 | Rsaman | i am going to cry |
| 12:43.30 | waKKu | :D |
| 12:43.47 | Rsaman | stuck with this transfer issue all week |
| 12:43.47 | Rsaman | no solution |
| 12:43.49 | Rsaman | so far |
| 12:44.16 | Rsaman | and blind transfer seems to be built in |
| 12:44.19 | Rsaman | and mine does not work |
| 12:44.32 | hmmhesays | what is the zaptel command for building the init script? |
| 12:46.32 | waKKu | Rsaman did u config it on features.conf ? |
| 12:46.34 | [TK]D-Fender | hmmhesays: make config |
| 12:46.40 | waKKu | set DYNAMIC_FEATURES=blindxfer ? |
| 12:46.47 | [TK]D-Fender | Rsaman: what phones are you using? |
| 12:47.01 | hmmhesays | [TK]D-Fender that is right |
| 12:47.07 | hmmhesays | but the init script seems to be broken |
| 12:47.10 | hmmhesays | it doesn't load ztdummy |
| 12:47.25 | Rsaman | waKKu : havent changed anything so far |
| 12:47.34 | Rsaman | [TK]D-Fender : IAX clients and zap channels |
| 12:47.50 | [TK]D-Fender | Rsaman: Zaptel FXS = ass |
| 12:48.28 | Rsaman | [TK]D-Fender : i am trying to transfer a calls on fxs channels mainly |
| 12:49.03 | [TK]D-Fender | Rsaman: I understand... just saying that I would never use Zaptel FXS for analog desk phones. |
| 12:49.50 | Rsaman | <PROTECTED> |
| 12:49.53 | Rsaman | oops |
| 12:49.57 | Rsaman | [TK]D-Fender : |
| 12:50.12 | [TK]D-Fender | Rsaman: Nope, you're stuck with zaptel for processing.... thats all there is. |
| 12:50.30 | robl^ | [TK]D-Fender: don't hate me.. I am setting up a new system using Aastra (15x 9133i and 4x 480i) instead of Polycom |
| 12:50.58 | Rsaman | [TK]D-Fender : so basically i am screwed? |
| 12:51.11 | Rsaman | [TK]D-Fender : my iax clients come with transfer buttons |
| 12:51.28 | Rsaman | [TK]D-Fender : zaptel i have to use # or watever the blind transfer key is |
| 12:51.36 | *** join/#asterisk guillote_GNU (n=bancaria@host185.190-136-203.telecom.net.ar) |
| 12:52.01 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 12:53.46 | Rsaman | http://forums.digium.com/viewtopic.php?t=17557&highlight= |
| 12:54.07 | hmmhesays | anyone else have a problem with the zaptel init script not starting ztdummy? |
| 12:54.09 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 12:55.23 | tzafrir_laptop | hmmhesays, not me, but what problems? |
| 12:55.40 | hmmhesays | it just exits with no output |
| 12:55.51 | tzafrir_laptop | Rsaman, what's the problem? |
| 12:56.20 | tzafrir_laptop | you hav no hardware? |
| 12:56.30 | tzafrir_laptop | and ztdummy still doesn't get loaded? |
| 12:56.35 | hmmhesays | tzafrir_laptop: bingo |
| 12:56.50 | tzafrir_laptop | what version of zaptel? |
| 12:56.56 | hmmhesays | 1.4.4 |
| 12:57.26 | Rsaman | tzafrir_laptop : hey man , still stuck with my blind transfer problem |
| 12:57.42 | Rsaman | tzafrir_laptop : dropping both parties |
| 12:57.59 | hmmhesays | what is this zaptel-perl utilities and do I need it? |
| 12:58.28 | tzafrir_laptop | hmmhesays, the are required for the astribank, and at the moemnt nice to have for others |
| 12:58.40 | hmmhesays | but the script should work without |
| 12:58.41 | tzafrir_laptop | hmmhesays, check zaptel_hardware and lszaptel |
| 12:59.05 | tzafrir_laptop | zaptel_hardware could be much more potent once there's a decent sysfs interfce in place |
| 12:59.48 | tzafrir_laptop | Rsaman, what exactly do you know and what don't you know? |
| 13:00.11 | tzafrir_laptop | I recall that your problem was that you transfered to the wrong context, right? |
| 13:00.28 | tzafrir_laptop | [TK]D-Fender, this has nothing to do with the channel, actually |
| 13:00.42 | Rsaman | tzafrir_laptop : thats wat we thought, but it was the right context |
| 13:01.30 | tzafrir_laptop | in the trace you showed me, the number was accepted correctly, but then asterisk compained "no number 101 in context incoming" or something |
| 13:01.50 | tzafrir_laptop | to debug the detected digits, use the "dtmf" log priority |
| 13:01.59 | tzafrir_laptop | you'll have to also log "debug" |
| 13:02.09 | tzafrir_laptop | in logger.conf |
| 13:02.13 | hmmhesays | yeah I get no output what so ever from that init script |
| 13:02.26 | tzafrir_laptop | hmmhesays, so trace it: |
| 13:02.40 | tzafrir_laptop | add a 'set -x' before the suspicious parts |
| 13:02.58 | tzafrir_laptop | 'set +x' stops tracing |
| 13:03.13 | Rsaman | ok |
| 13:03.13 | *** join/#asterisk zeeesh (i=zeeesh@202.166.161.45) |
| 13:03.14 | Rsaman | thanks |
| 13:03.36 | tzafrir_laptop | hmmhesays, or run: sh -x /etc/init.d/zaptel start |
| 13:04.16 | hmmhesays | + MODULES='ztdummy tor2 wct4xxp wct1xxp wcte11xp wcfxo wctdm wctdm24xxp' |
| 13:04.16 | hmmhesays | + '[' '' = yes ']' |
| 13:04.16 | hmmhesays | + exit 0 |
| 13:04.58 | Rsaman | Transferring IAX2/102-5 to '444' (context incoming) priority 1 |
| 13:04.59 | Rsaman | <PROTECTED> |
| 13:04.59 | Rsaman | <PROTECTED> |
| 13:05.06 | Rsaman | transferes |
| 13:05.10 | Rsaman | then hangs up |
| 13:07.13 | tzafrir_laptop | hmmhesays, so you have ztdummy explicitly in the modules list |
| 13:07.24 | tzafrir_laptop | I'd say that the modprobe of ztdummy fails |
| 13:07.32 | hmmhesays | it doesn't though |
| 13:07.36 | tzafrir_laptop | lsmod | grep ztdummy |
| 13:08.01 | tzafrir_laptop | if it doesn't, then: no problems |
| 13:08.02 | hmmhesays | i can load it manually tzafrir_laptop |
| 13:08.05 | hmmhesays | but the script doesn't load it |
| 13:08.44 | tzafrir_laptop | rmmod ztdummy; /etc/init.d/zaptel start; lsmod | gerp ztdummy |
| 13:09.19 | hmmhesays | tzafrir_laptop; as I said the script does not load it |
| 13:09.33 | tzafrir_laptop | ahh... |
| 13:09.44 | tzafrir_laptop | hmmhesays, you have zaptel 1.4.4 ? |
| 13:09.52 | hmmhesays | tzafrir_laptop: yep just built |
| 13:09.58 | tzafrir_laptop | ls -l /etc/default/zaptel /etc/sysconfig/zaptel |
| 13:10.13 | rody | exit |
| 13:10.20 | zeeesh | i downloaded app_conference in /usr/src/asterisk-1.4.8 .. i just unzip it and now trying to install it by "make clean" or make .. but getting error "rm -f *.so *.o app_conference.o conference.o member.o frame.o cli.o libspeex/preprocess.o libspeex/misc.o libspeex/smallft.o |
| 13:10.21 | tzafrir_laptop | rody, /quit |
| 13:10.21 | zeeesh | "? |
| 13:11.05 | tzafrir_laptop | hmmhesays, in short: a screwup of zaptel 1.4.4 (sorry) - |
| 13:11.17 | tzafrir_laptop | mv /etc/default/zaptel /etc/sysconfig/zaptel |
| 13:11.31 | [TK]D-Fender | tzafrir_laptop: I never said it was the fault of the channel |
| 13:11.48 | hmmhesays | there we go |
| 13:11.51 | tzafrir_laptop | Though frankly I think that this test for $TELEPHONY=yes is pointless |
| 13:12.08 | tzafrir_laptop | hmmhesays, but then again, use 1.4.5.1 if you install now |
| 13:12.11 | tzafrir_laptop | it's nicer |
| 13:12.13 | [TK]D-Fender | tzafrir_laptop: I'm saying that if was using anything else he wouldn't be RESTRICTED to that means of acheiving his ends :) |
| 13:12.23 | hmmhesays | nicer? I will never have any zaptel hardware on this box |
| 13:13.21 | tzafrir_laptop | That can be fixed :-) |
| 13:13.58 | *** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 13:14.39 | tzafrir_laptop | [TK]D-Fender, restricted? both zaptel and SIP offer a channel-specific transfer. You could also use the generic transfer |
| 13:15.35 | tzafrir_laptop | infact, various SIP hadsets take quite some tuning to implement such features as well |
| 13:15.43 | tzafrir_laptop | if it is possible at all |
| 13:16.04 | *** join/#asterisk jfitzgibbon (n=IceChat7@64.72.237.130) |
| 13:18.43 | x86 | morning |
| 13:20.13 | lirakis | he |
| 13:20.15 | lirakis | *y |
| 13:20.45 | jfitzgibbon | morning folks; any PRI gurus up this early? |
| 13:20.54 | *** join/#asterisk sakic (n=sakic@adsl-146-182-113.clt.bellsouth.net) |
| 13:21.56 | deeperror | not a guru but up |
| 13:22.53 | jfitzgibbon | ever seen a problem where after a reboot a TE4xxxP that was working fine refuses to sync to the Telco? All I see is my side sending SABME with no response from the switch |
| 13:23.53 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:33.07 | [TK]D-Fender | tzafrir_laptop: All the SIP phones I've used have done blind transfer jsut fine, as have Linksys/Sipura ATA's |
| 13:33.54 | tzafrir_laptop | well, flash transfer works quite well on Zaptel |
| 13:34.28 | tzafrir_laptop | Just a different way of doing things |
| 13:34.31 | [TK]D-Fender | tzafrir_laptop: For BLIND? That is what he was asking about. |
| 13:34.33 | *** join/#asterisk AsteriskProblems (n=pbarnsle@81.171.174.178) |
| 13:34.47 | Bladerunner05 | Hello I'm using * 1.4.9 and TDP400p (4FXO) when I call * from outside and without respond hang up the call my extension ring for other 2 or 3 time before stop |
| 13:35.10 | AsteriskProblems | hey guys, new to asterisk, ive just ssh'd into the box from a vista computer - it says its running bash but if i type ls or dir it doesnt respond - just a new line |
| 13:36.05 | tzafrir_laptop | [TK]D-Fender, I was trying to get him to provide me some useful debugging output (which Asterisk provides) to eliminate channel-speicifc issues from dialplan setup issues |
| 13:36.13 | tzafrir_laptop | he hasn't so far provided me |
| 13:37.07 | AsteriskProblems | its ok - sorted it |
| 13:37.07 | tzafrir_laptop | AsteriskProblems, and this is related to Asterisk because? |
| 13:37.16 | robl^ | AsteriskProblems: learning Linux / bash would be a good palce to start before running Asterisk. |
| 13:37.17 | AsteriskProblems | i had to type enable cd for some reason |
| 13:37.37 | tzafrir_laptop | start at bash.org |
| 13:38.05 | *** join/#asterisk ManxPower (n=manxpowe@032-441-472.area7.spcsdns.net) |
| 13:38.48 | [TK]D-Fender | tzafrir_laptop: Yeah, all too many people come in with nothing to backup their problems. |
| 13:40.36 | AsteriskProblems | anyone know a link to somewhere that shows you how to set up a basic auto attendant? |
| 13:41.37 | [TK]D-Fender | AsteriskProblems: WIKI + BOOK |
| 13:41.39 | [TK]D-Fender | ~wikis |
| 13:41.39 | jbot | from memory, wikis is http://www.voip-info.org |
| 13:41.41 | [TK]D-Fender | ~book |
| 13:41.41 | jbot | methinks book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 13:42.28 | AsteriskProblems | yeh ive looked at the wiki's but they tend to be written by people that dont explain everything too well :D i will check the second link you give |
| 13:42.57 | AsteriskProblems | when i get my system fully working i might write my own blog or something in easy language |
| 13:45.40 | tzafrir_laptop | AsteriskProblems, bash.org is a place to waste you time. not to learn Linux, bash or whatever :-) |
| 13:46.04 | *** part/#asterisk telmich (i=telmich@gpm/telmich) |
| 13:46.15 | HarryR | AsteriskProblems, most of the blogs out there are by people who had the same intentions as you |
| 13:46.29 | HarryR | AsteriskProblems, the biggest problem is with the initial learning curve for Asterisk which some people find difficult |
| 13:46.34 | ManxPower | I hate blogs |
| 13:46.53 | AsteriskProblems | yeh its a bit of a curve |
| 13:47.14 | HarryR | the best thing you could do is write what you know _now_, converting from asterisk speak to normal person speak |
| 13:47.19 | HarryR | and use it as a reference later |
| 13:47.26 | AsteriskProblems | my linux experience is very rusty, so im coming to asterisk with quite old experience and trying to pick up asterisk |
| 13:47.37 | HarryR | yeah I know how you feel |
| 13:47.52 | AsteriskProblems | ive got it calling out, calling between extensions and incoming calls so im not doing too bad |
| 13:47.56 | *** join/#asterisk rickross (n=rickross@supporter/active/rickross) |
| 13:48.04 | AsteriskProblems | just trying to figure out how to set up the auto attendandt now |
| 13:48.13 | *** join/#asterisk Strom_M (n=strom@160.79.29.94) |
| 13:48.30 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 13:48.30 | *** mode/#asterisk [+o file] by ChanServ |
| 13:48.46 | HarryR | ah cool |
| 13:48.58 | *** join/#asterisk MACscr (n=MACscr@adsl-75-23-94-138.dsl.peoril.sbcglobal.net) |
| 13:49.02 | MACscr | Hello everyone |
| 13:49.40 | *** join/#asterisk Enz0gfx (n=Enz0@24.248.220.72) |
| 13:49.53 | MACscr | Is there a open source solution to enable asterisk to be able to communicate with skype? |
| 13:50.20 | AsteriskProblems | that would be cool |
| 13:52.54 | Enz0gfx | that would be sweet |
| 13:54.51 | [TK]D-Fender | MACscr: No. |
| 13:55.31 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 13:55.56 | MACscr | [TK]D-Fender : there is some commercial option for it isnt there? |
| 13:56.14 | [TK]D-Fender | MACscr: Yes, and they are all dirty hacks. |
| 13:56.49 | [TK]D-Fender | Skype is the bastard child of telephony ; used & promoted by chumps who think their new cool app is "the shiznit". |
| 13:56.53 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 13:56.56 | MACscr | I find it odd that there isnt an asterisk option thats free when other scripts out there that do interface with skype and are not official skype tools |
| 13:57.12 | MACscr | I agree. I dont use it myself, but some of my clients do |
| 13:57.25 | MACscr | Would be nice if they could call us with skype |
| 13:57.40 | MACscr | For free that is |
| 13:59.34 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:59.34 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:04.03 | rickross | does WaitExten() recognize conference room extensions as valid? |
| 14:04.44 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:05.52 | rickross | I have an IVR that backgrounds a message, then calls WaitExten(), but if I enter a valid conference extension it gets bounced as invalid |
| 14:09.46 | jfitzgibbon | rick: defining conference room numbers in meetme.conf doesn't make them available in the dialplan - you need to have a matching extension that calls MeetMe with the room number as an argument |
| 14:09.53 | Dr-Linux | i'm using Monitor application for recordings, but it doesn't record the whole conversation, i can see on the CLI: |
| 14:09.54 | Dr-Linux | ast_write: Failed to write data to channel monitor write stream |
| 14:10.02 | Dr-Linux | any recommendations? |
| 14:10.02 | Rsaman | http://pastebin.com/m76d4739e |
| 14:10.07 | Rsaman | i am running out of ideas |
| 14:10.11 | Rsaman | this is my cli output |
| 14:10.16 | Rsaman | of my blind call transfer |
| 14:10.23 | Rsaman | along with my dialplan |
| 14:10.31 | rickross | jfitz - thank you |
| 14:10.32 | Rsaman | both parties still get hung up on |
| 14:10.41 | rickross | can I keep that from being visible in the directory |
| 14:11.21 | jfitzgibbon | again, having an extension in the dialplan doesn't make it visible to the Directory application - that reads voicemail.conf |
| 14:11.27 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 14:11.41 | rickross | super - thx! |
| 14:13.38 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.172.179) |
| 14:14.04 | Rsaman | anyone manage to look ? |
| 14:14.05 | Rsaman | http://pastebin.com/m76d4739e |
| 14:14.09 | Rsaman | i am soo sad |
| 14:14.14 | Rsaman | cant blind transfer |
| 14:14.44 | [TK]D-Fender | Rsaman: You are transferring to an extension that DOESN'T EXIST |
| 14:14.57 | jfitzgibbon | since your example expects the zap channel to be transferred to Dial(IAX2/...), have you tried doing it with iax debugging turned on |
| 14:15.01 | [TK]D-Fender | Rsaman: its COMMENTED OUT. |
| 14:15.06 | [TK]D-Fender | -- Transferring IAX2/888-2 to '102' (context incoming) priority 1 |
| 14:15.12 | [TK]D-Fender | ;exten => 102,1,Dial(SIP/nico) |
| 14:15.16 | jfitzgibbon | d-fender: I thought so too, but he includes 'office' which pattern matches |
| 14:15.18 | Rsaman | [TK]D-Fender : but if u look in the office context |
| 14:15.19 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 14:15.45 | Rsaman | [TK]D-Fender : it does the same if i transfer to 444 |
| 14:15.54 | Rsaman | [TK]D-Fender : which is in both contexts |
| 14:16.12 | Dr-Linux | [TK]D-Fender: any clue? |
| 14:16.13 | jfitzgibbon | i'm wondering if the hangup happens before or after any attempt to place a new call over IAX is made |
| 14:16.13 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 14:16.38 | [TK]D-Fender | :/ |
| 14:20.05 | Rsaman | could someone please show me the cli output of a successful blind call transfer |
| 14:21.16 | Rsaman | gonna flood 3 lines .. sorry |
| 14:21.24 | Rsaman | <IAX2/102-13> Playing 'pbx-transfer' (language 'en') |
| 14:21.25 | Rsaman | <PROTECTED> |
| 14:21.25 | Rsaman | <PROTECTED> |
| 14:21.25 | Rsaman | <PROTECTED> |
| 14:21.27 | Rsaman | 4 |
| 14:21.34 | Rsaman | thats if the context does not exist |
| 14:21.54 | bkw__ | no 33 in context internal |
| 14:21.54 | Rsaman | so it is finding the context but dropping after the transfer |
| 14:21.57 | Mercestes | I would say that's if the extension "33" does not inext in that context. |
| 14:22.12 | Mercestes | s/inext/exist/ |
| 14:22.54 | Rsaman | yes |
| 14:22.54 | Rsaman | http://pastebin.com/m76d4739e |
| 14:22.54 | Rsaman | but if u look here |
| 14:22.54 | Rsaman | i was just showing that in my prev post , the extension does exist |
| 14:22.54 | Rsaman | but the blind call transfer does not work |
| 14:22.58 | Rsaman | <PROTECTED> |
| 14:23.51 | Bladerunner05 | when hang up and incoming call the caller line remains open |
| 14:24.04 | Mercestes | I see 88 no where in here. |
| 14:24.11 | Mercestes | nor do I see 33 |
| 14:24.17 | Mercestes | what line matches 33? |
| 14:24.22 | Rsaman | none |
| 14:24.25 | Rsaman | i was just showing |
| 14:24.32 | Mercestes | ... |
| 14:24.35 | Mercestes | Showing us WHAT? |
| 14:24.44 | Mercestes | The extension doesn't exist. |
| 14:24.55 | Rsaman | well that 102 does exist,, |
| 14:25.03 | Rsaman | because i was told the extension was not found |
| 14:25.15 | Rsaman | so i showed the output of when an extension does not exist |
| 14:25.16 | Mercestes | then why are you pasting 33? |
| 14:25.33 | Rsaman | http://pastebin.com/m76d4739e |
| 14:25.43 | Rsaman | i have my other output here |
| 14:25.51 | Rsaman | where i try to blind transfer to 102 |
| 14:25.56 | Mercestes | Can you call 102? |
| 14:25.58 | Rsaman | but it hangs both parties up |
| 14:25.59 | Rsaman | yes |
| 14:26.03 | Mercestes | what type of phones? |
| 14:26.35 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 14:26.35 | *** mode/#asterisk [+o file] by ChanServ |
| 14:26.36 | Rsaman | i am calling a desktop phone from an IAX client and transfering to another IAX client |
| 14:26.49 | Rsaman | desktop from on zap channel |
| 14:26.50 | Rsaman | fxs |
| 14:26.55 | Mercestes | Try swapping your canreinvite value to it's opposite, reload and try again. |
| 14:27.04 | Mercestes | Rsaman, FXS? Or modem? |
| 14:27.44 | Rsaman | fxs |
| 14:28.05 | *** join/#asterisk zpertee (n=chatzill@cpe-24-166-81-113.neo.res.rr.com) |
| 14:28.16 | Rsaman | canreinvite in iax.conf? |
| 14:28.17 | Mercestes | try swapping canreinvite then and see if that has any effect. |
| 14:28.36 | Mercestes | hrm...I guess I normally see canreinvite in sip.conf huh? |
| 14:28.44 | Mercestes | you would have to make it difficult |
| 14:29.18 | Rsaman | yeah its usually there |
| 14:29.29 | Mercestes | notransfer instead |
| 14:29.35 | Mercestes | try yes and no |
| 14:29.43 | datachomper | What was the name of the built in http service thing for asterisk? |
| 14:29.52 | Mercestes | don't ask me why the changed the name. |
| 14:30.04 | Mercestes | datachomper, a web app? |
| 14:30.09 | Mercestes | apache? |
| 14:30.20 | Mercestes | Asterisknow? |
| 14:30.23 | Mercestes | Trixbox? |
| 14:30.27 | Mercestes | The manager? |
| 14:30.30 | Mercestes | FoP |
| 14:30.34 | Mercestes | I give up. |
| 14:30.51 | datachomper | The web service thing that's new in 1.4 |
| 14:30.56 | Mercestes | asterisknow |
| 14:31.09 | *** join/#asterisk ajohnstone (n=ajohnsto@85.189.117.98) |
| 14:31.38 | datachomper | AJAM, found it |
| 14:31.55 | *** join/#asterisk bmg505 (n=leon@196.209.183.203) |
| 14:33.26 | *** join/#asterisk riddlebox (n=JamesMid@75-128-170-26.static.stls.mo.charter.com) |
| 14:36.10 | *** join/#asterisk DarkFlib (n=mike@host90-152-23-30.ipv4.regusnet.com) |
| 14:36.56 | *** join/#asterisk Seldon75 (n=chatzill@69.77.161.2) |
| 14:37.12 | Seldon75 | hello, |
| 14:37.26 | Seldon75 | whats the * console command to apply changes to the dialplan file? |
| 14:37.54 | Rsaman | dialplan reload |
| 14:37.57 | Mercestes | You type in this |
| 14:37.58 | Mercestes | ~book |
| 14:37.59 | jbot | i heard book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 14:38.01 | Rsaman | lol |
| 14:38.18 | Rsaman | i read the book, if u have blind call transfer issues , it wont help |
| 14:38.24 | Mercestes | I'm pretty sure it's extensions reload unless they decided to change that too in 1.4 |
| 14:38.34 | Rsaman | *they did |
| 14:38.43 | Mercestes | whoever did that is going to hell. |
| 14:38.54 | DarkFlib | got a little issue with the regex function... seems carets and dollars (and commas I'm told) need to be escaped, but while I can put a dollar in a variable I can't do anything with the caret in the regex http://pastebin.ca/669218 |
| 14:38.54 | Rsaman | lol |
| 14:39.10 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
| 14:39.16 | DarkFlib | any known workaround? |
| 14:39.31 | Mercestes | DarkFlib, escape the caret |
| 14:39.32 | Rsaman | only if u can help me with my blind transfer issue |
| 14:39.40 | Rsaman | :) |
| 14:39.44 | DarkFlib | tried that... see the pastebin entry |
| 14:40.13 | DarkFlib | unless you have an alternative to \^ |
| 14:40.25 | ManxPower | DarkFlib: not many people use regex with asterisk. I assume you read README.variables (or whatever it is called in 1.4) |
| 14:41.16 | DarkFlib | I had a read of the wiki... but not that file yet... |
| 14:41.57 | ManxPower | you must read that file. The wiki is full of wrong information |
| 14:42.51 | DarkFlib | no README.variables in 1.4.10 from what I can see |
| 14:43.08 | ManxPower | I'm sure they renamed it. |
| 14:43.15 | ManxPower | /path/to/src/asterisk/docs |
| 14:43.38 | AsteriskProblems | anyone know how to access the demo auto attendant in asterisknow? |
| 14:44.12 | ManxPower | AsteriskProblems: as on #asterisknow |
| 14:44.18 | ManxPower | that is not Asterisk |
| 14:44.24 | AsteriskProblems | yeh if anyone bothered to reply :D |
| 14:44.29 | AsteriskProblems | they are all dead |
| 14:44.37 | ManxPower | I guess you'll have to wait then |
| 14:44.48 | *** join/#asterisk mog (i=mog@nat/digium/x-2f3663febbfa2d83) |
| 14:44.48 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:45.10 | riddlebox | what is the best way to send emails out when a message is left in a voicemail box? |
| 14:45.36 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 14:45.57 | ManxPower | riddlebox: your question is a FAQ |
| 14:46.13 | ManxPower | Should be covered in ATFOT, The Wiki, and a billion other places. |
| 14:46.30 | ManxPower | Hell, it's covered in the voicemail.conf.sample |
| 14:46.53 | DarkFlib | ManxPower: seems to be now called channelvariables.txt |
| 14:47.05 | riddlebox | ManxPower: for some reason what used to work for me is no longer working, I just used exim4 -t and it worked fine for a long time |
| 14:47.24 | ManxPower | riddlebox: then you have an MTA issue, not an Asterisk issue. |
| 14:47.45 | puzzled | hi |
| 14:47.50 | ManxPower | All Asterisk does when you set the voicemail.conf options is run "sendmail" to send the message. All MTAs have a "sendmail", if only for compatability. |
| 14:48.39 | DarkFlib | ahhh... there is another regex syntax with the expressions... |
| 14:49.08 | ManxPower | DarkFlib: there are 2 places to look for docs BEFORE ALL OTHERS. "show application X" in the CLI and /path/to/src/asterisk/docs |
| 14:49.18 | ManxPower | THEN check the Wiki, mailinglist archives, etc. |
| 14:50.20 | DarkFlib | yeah, I know about show application... I've only just moved to 1.4 (been using 1.2 since it became stable tho (and before that 1.0 and the pre 1.0 cvs versions)) |
| 14:50.41 | Mercestes | Yea, this is probably the last place you should ask. We give out wrong information pretty reliably. |
| 14:50.52 | Mercestes | I do anyways. |
| 14:51.03 | ManxPower | Mercestes: not even close to the mailing lists, however! |
| 14:51.11 | Mercestes | lmao |
| 14:51.14 | Mercestes | Nice. |
| 14:51.18 | DarkFlib | I don't need answers most of the time... just pointers of where I should have looked to RTFM... |
| 14:51.29 | DarkFlib | <PROTECTED> |
| 14:51.54 | ManxPower | At least Digium sometimes monitors -users and corrects the worst of the misinformation. |
| 14:51.57 | Mercestes | Yea, our manual is kinda like a book.....with no binder, set in front of one of those giant industrial fans. yea, it's all documented......somewhere. |
| 14:52.18 | umdstu | hey all |
| 14:52.35 | umdstu | i was wondering if there was a request to send to an asterisk server, and have it reply with its version |
| 14:52.45 | Mercestes | asterisk -help |
| 14:52.54 | umdstu | in the CLI ? |
| 14:53.11 | ManxPower | COMMAND LINE INTERFACE |
| 14:53.14 | Mercestes | in your os CLI |
| 14:53.19 | ManxPower | Aka "the asterisk console" |
| 14:53.26 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 14:53.27 | DarkFlib | umdstu: asterisk -V |
| 14:53.29 | umdstu | ManxPower: thank you... |
| 14:53.49 | Seldon75 | CLI in telecoms also means Calling Line Identifier |
| 14:53.58 | umdstu | DarkFlib: alright thanks ! |
| 14:54.02 | Seldon75 | that's the problem with telecoms |
| 14:54.06 | Seldon75 | too many acronyms |
| 14:54.09 | jfitzgibbon | the asterisk CHUI then? |
| 14:54.27 | DarkFlib | also asterisk -r the first line is Connected to Asterisk 1.4.10 currently running on host (pid = pid ) |
| 14:54.37 | umdstu | yea |
| 14:54.40 | umdstu | i see all that |
| 14:54.40 | Mercestes | Seldon75, and most acronyms are context sensitive and change meanings as you change contexts |
| 14:54.41 | *** join/#asterisk PepOSX (n=pepOSX@190.72.146.161) |
| 14:54.53 | umdstu | i'm looking for something more remote |
| 14:54.57 | DarkFlib | also.... on the CLI itself... show version |
| 14:54.59 | umdstu | when your not on the computer running the server |
| 14:55.10 | Mercestes | you want to remotely discover the asterisk version? |
| 14:55.17 | umdstu | Mercestes: is that possible? |
| 14:55.34 | Mercestes | Sure, Festival(System(asterisk -V)) in yoru dial plan. |
| 14:55.37 | Mercestes | why would you want that??? |
| 14:55.42 | jfitzgibbon | you might be able to use SIP_HEADER once you're in a call with that * |
| 14:55.50 | ManxPower | umdstu: ssh into the server, issue "asterisk -rvvv" do a "show version" |
| 14:55.53 | DarkFlib | umdstu: yes, use ssh with a key pair to run asterisk -V no password required... |
| 14:56.06 | umdstu | hmm |
| 14:56.06 | jfitzgibbon | but anonymously without creds on that system, no |
| 14:56.13 | Mercestes | No, you cannot covertly discover someone else's asterisk vresion. |
| 14:56.18 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:56.21 | umdstu | jfitzgibbon: yea i understand that |
| 14:56.24 | DarkFlib | you *might* be able to fingerprint it... |
| 14:56.27 | umdstu | it doesn't have to be covert |
| 14:56.43 | umdstu | DarkFlib: You see what I want to do |
| 14:56.45 | Mercestes | Oh, well the obvious manner would be to call them up and go "Hey, what version asterisk do you run?" |
| 14:56.57 | Wonka | i do a Queue(qname,,300); qname has no agents registered. it returns immediately with "Exiting on time-out cycle", which should only apply when the n option is given - or do i misunderstand the comments in app_queue.c? |
| 14:56.58 | umdstu | hah |
| 14:57.40 | jfitzgibbon | what version are you using wonka? |
| 14:57.48 | Wonka | jfitzgibbon: 1.4.9 |
| 14:57.54 | jfitzgibbon | yeah, it's broken there |
| 14:58.00 | DarkFlib | umdstu: you should be able to see the difference between 1.0 1.2 and 1.4 easily enough with fingerprinting techniques... but minor versions might be a little more complex |
| 14:58.07 | jfitzgibbon | I reported it and it was fixed in 1.4.10.1 |
| 14:58.21 | Wonka | jfitzgibbon: ok, so i'd need to rebuild... |
| 14:58.23 | jfitzgibbon | but there's a standalone patch for 1.4.10 (maybe applicable to 1.4.9) in the tracker |
| 14:58.38 | jfitzgibbon | http://bugs.digium.com/view.php?id=10320 |
| 14:58.52 | umdstu | DarkFlib: Minor versions aren't too important |
| 14:59.11 | jfitzgibbon | yeah, but they did a lot of block-level cleanup of app_queue in 1.4.8 - 1.4.10 |
| 14:59.26 | jfitzgibbon | makes it near impossible to cleanly apply a patch designed for 1.4.7 or earlier to 1.4.9 or later |
| 14:59.39 | [TK]D-Fender | And its not like we're up to 1.4.11 anyways.... |
| 14:59.42 | [TK]D-Fender | oh wait... we are :) |
| 14:59.50 | [TK]D-Fender | UPGRADE! |
| 14:59.58 | [TK]D-Fender | </sarcasm> |
| 15:00.01 | umdstu | lol |
| 15:00.07 | umdstu | i'm at uh |
| 15:00.38 | Wonka | jfitzgibbon: thx |
| 15:01.14 | Mercestes | umdstu: UH as in University of Houston? |
| 15:01.14 | umdstu | 1.4.4 |
| 15:01.22 | ManxPower | I thought 1.4.11 didn't even COMPILE. |
| 15:01.30 | umdstu | uh as in a pause |
| 15:01.31 | umdstu | hah |
| 15:01.43 | umdstu | umdstu= University of Maryland |
| 15:01.50 | umdstu | student |
| 15:01.56 | Mercestes | what's the d mean? |
| 15:02.05 | DarkFlib | md == maryland |
| 15:02.07 | umdstu | MD |
| 15:02.08 | umdstu | lol |
| 15:02.11 | Mercestes | ah. |
| 15:02.16 | umdstu | funny how people always ask that |
| 15:02.19 | DarkFlib | I'm english and even I could see that |
| 15:02.22 | DarkFlib | :P |
| 15:02.24 | umdstu | lol |
| 15:02.37 | umdstu | i work at NDU tho |
| 15:02.53 | deeperror | any ideas why after leaving a 3way call the 2 channels would stay open even after their call has ended? |
| 15:02.53 | umdstu | and no, not north dakota |
| 15:03.10 | jfitzgibbon | umdstu: I just did a quick test, and exten => whatever,1,SayAlpha(${SIP_HEADER(User-Agent)}) works properly when I call it from my Polycom desk phone |
| 15:03.30 | jfitzgibbon | so you could do that and try to parse out the * version (might be better done in an AGI if you're not doing it for every call) |
| 15:03.44 | jfitzgibbon | given the annoying regex stuff built into the dialplan |
| 15:05.56 | DarkFlib | use sipsak and analyse the returned packet... |
| 15:06.00 | DarkFlib | :P |
| 15:06.05 | umdstu | jfitzgibbon: thanks for that, i'll give it a try! |
| 15:06.15 | DarkFlib | an options request shoudl work |
| 15:06.32 | *** join/#asterisk FlatFoot (n=simon@80.88.192.83) |
| 15:06.34 | umdstu | i havn't used sipsak |
| 15:06.55 | DarkFlib | sip swiss army knife... nice tool for all sorts of stuff... |
| 15:07.03 | umdstu | neato |
| 15:07.29 | *** join/#asterisk b11d (n=no@234-200-29-134.hcc.mnscu.edu) |
| 15:08.06 | b11d | Does anyone know if it is possible to override the "Forward" option on a Polycom 501 if a call is received from a specific CallerID number? |
| 15:08.33 | b11d | Also.. the same question,but replace "Forward" with "Do Not Disturb" |
| 15:09.29 | ManxPower | no |
| 15:09.35 | datachomper | I don't think so |
| 15:09.42 | b11d | drat :) |
| 15:09.54 | *** join/#asterisk jstew (n=jstew@adsl-76-210-118-28.dsl.klmzmi.sbcglobal.net) |
| 15:09.58 | b11d | that would be nice if i could.. oh well :P |
| 15:09.59 | [TK]D-Fender | b11d: No. |
| 15:10.04 | ManxPower | you could tell phone phone to do DND on a PER LINE basis, then the override would send the call to a line without DND on it. |
| 15:10.15 | jstew | Hi can anyone tell me if zaptel works fine on PPC? |
| 15:10.29 | datachomper | I didn't think 501's had line level resolution for DND |
| 15:10.34 | ManxPower | they do |
| 15:10.41 | [TK]D-Fender | datachomper: They do |
| 15:10.42 | datachomper | Is that set in the conf? |
| 15:10.45 | b11d | thats an interesting idea.. |
| 15:11.01 | [TK]D-Fender | datachomper: Yes, thats why its called a CONFIGURATION file ;) |
| 15:11.21 | b11d | basically my problem is this.. im trying to use the asterisk system to alert all phones in the event of an emergency.. but phones with forward options enabled, or DND, do not get that message.. |
| 15:11.22 | jstew | <PROTECTED> |
| 15:11.25 | [TK]D-Fender | b11d: Then again to get throught this person would ahve to call another reg's exten. |
| 15:11.36 | datachomper | Ok, so what I meant was, where is it set in the conf. |
| 15:11.42 | [TK]D-Fender | jstew: Sad waste... |
| 15:11.54 | jstew | haha, not my sad waste :) |
| 15:11.58 | jstew | my employers idea |
| 15:12.02 | b11d | thanks for the advice all.. |
| 15:12.13 | [TK]D-Fender | datachomper: I believe its actually default in the newer firmwares where ti was introduced |
| 15:12.30 | datachomper | Ah, knew I wasn't crazy |
| 15:12.36 | datachomper | (for that reason, anyway) |
| 15:13.41 | Mercestes | b11d: disable forward and DND |
| 15:13.48 | Mercestes | b11d, it breaks random shit. |
| 15:14.02 | *** join/#asterisk exvito (n=exvito@195.245.132.93) |
| 15:14.46 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 15:15.11 | b11d | haha |
| 15:15.18 | *** join/#asterisk SECGOD (i=SECGOD@c-24-14-83-172.hsd1.il.comcast.net) |
| 15:15.21 | b11d | i cant disable them.. too many people here use those options |
| 15:15.30 | b11d | and now that they've had a taste, they wont want to give them up |
| 15:15.51 | exvito | Hi... I was browsing the wiki but found no info related to my specific need. What kind of system (cpu/ram/etc) would I need to setup a SIP <-> IAX gateway with no transcoding ? |
| 15:16.06 | exvito | ...say for 100 simultaneous calls. |
| 15:16.10 | Mercestes | b11d, then let them die. |
| 15:16.13 | *** join/#asterisk jhiver (i=jhiver@164-242.206-83.static-ip.oleane.fr) |
| 15:16.15 | jhiver | hi |
| 15:16.19 | b11d | holy shit thats a good idea Mercestes :) |
| 15:16.25 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-aa19b15e15777dd4) |
| 15:16.25 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:16.29 | jhiver | is there a way to check what is the bandwith used by a given IAX trunk? |
| 15:16.30 | Mercestes | exvito, a normal server. 2 gigs of ram. |
| 15:16.38 | b11d | I totally heart you Mercestes |
| 15:17.00 | b11d | and now im stalking you full time |
| 15:17.02 | [TK]D-Fender | b11d: If you want selective DND, thats what your DIALPLAN is for :p |
| 15:17.13 | b11d | it's really not so much the DND as it is the forwarding TK.. |
| 15:17.13 | jfitzgibbon | there's a lesson to be learned there about letting the lusers taste blood |
| 15:17.20 | exvito | Mercestes: ...that would be any current server-class dual core cpu, I belive. Correct ? |
| 15:17.38 | b11d | hrm.. it's probably a really bad idea to allow people to foward to off-campus phones though.. |
| 15:17.39 | wunderkin | quad dual dual quad core |
| 15:17.45 | b11d | i just realized the potential for abuse there |
| 15:18.31 | Mercestes | exvito, pretty much |
| 15:18.33 | [TK]D-Fender | b11d: Only if the can call somewhere you don't want them to at some point in time. |
| 15:18.41 | exvito | Mercestes: thanks ! |
| 15:18.57 | [TK]D-Fender | b11d: like rigging it to call LD family members after hours and phoning in from the outside ;) |
| 15:19.04 | b11d | yeah.. all i need is someone to forward their office phone to a house in .au :P |
| 15:19.12 | b11d | exactly TK! |
| 15:19.17 | [TK]D-Fender | b11d: yup ;) |
| 15:19.44 | Mercestes | b11d, use multiple lines.... |
| 15:19.50 | riddlebox | has anyone installed the Asterisk Appliance 50? |
| 15:19.56 | Mercestes | b11d, and then disable DND and forwarding on the secondary "emergency" line |
| 15:19.59 | b11d | that is an option.. but it sounds cumbersome |
| 15:20.01 | [TK]D-Fender | b11d: then again the only option is to have the dialplan control everything. |
| 15:20.11 | Mercestes | exvito, just don't get cheap. |
| 15:20.17 | [TK]D-Fender | b11d: And check office hours, destinations, etc. |
| 15:20.28 | Mercestes | exvito, most mid-grade to business grade servers come with multi-core and 2-4 gigs with internal arrays. That's what you want. |
| 15:20.32 | b11d | yeah.. that'll be a lot of work but likely necessary.. |
| 15:20.52 | [TK]D-Fender | Mercestes: a 2nd line isn't a 1-stop solution. Wh is the call going to REACH the 2nd? |
| 15:20.58 | [TK]D-Fender | Mercestes: Thats another # to dial. |
| 15:21.00 | exvito | Mercestes: ...I was considering something like the proliant ML3xx / DL3xx servers. |
| 15:21.09 | b11d | haha compaq |
| 15:21.25 | Mercestes | [TK]D-Fender, that's pretty easily done in the dialplan, he will likely know when it's an emergency |
| 15:21.36 | [TK]D-Fender | b11d: Last I heard HP servers were very well respected.... |
| 15:21.44 | b11d | are they, these days? |
| 15:21.48 | [TK]D-Fender | Mercestes: Yes, in dialplan, not the phone + reg. |
| 15:21.51 | b11d | i had terrible experiences with them back in the day |
| 15:21.51 | HarryR | exvito, on the tests I've done the SAS Dell 19xx and 29xx series outperform similar spec HP DL series servers |
| 15:21.59 | HarryR | but their still behind the SCSI models :( |
| 15:22.04 | Mercestes | b11d, yea, HP bought them so they aren't really compaq anymore. |
| 15:22.06 | [TK]D-Fender | exvito: "Nobody ever got fired for buying IBM" <--- |
| 15:22.10 | b11d | Hewlett-Paquard now or what? |
| 15:22.14 | umdstu | [TK]D-Fender: Thanks for trying to help yesterday |
| 15:22.28 | umdstu | i gave up on that NAT'ing and put my server inside the LAN |
| 15:22.34 | [TK]D-Fender | umdstu: For what litte I was able to while being dragged away often.... you're welcome :) |
| 15:22.37 | umdstu | maybe one day there will be an easy fix |
| 15:22.51 | exvito | HarryR: ...as a gateway the HD access won't be critical: no recording, no music, no prompts; all I need is an IAX <-> SIP gateway. Anyhow, thanks for the info. |
| 15:22.54 | umdstu | hah, yea well you tried |
| 15:23.26 | [TK]D-Fender | exvito: 100 calls w/ no transcoding isn't much, its mostly media.... |
| 15:23.45 | umdstu | i'll write a new protocol that piggy backs on SIP but bypasses NAT |
| 15:23.47 | s0ck | HarryR: ever use IBM servers? |
| 15:23.53 | HarryR | nope |
| 15:23.55 | umdstu | i should have that ready by Sunday |
| 15:24.14 | umdstu | who wants to help |
| 15:24.34 | HarryR | s0ck, are they good? (pSeries right?) |
| 15:26.14 | umdstu | does anyone know of open solaris channel? |
| 15:27.24 | HarryR | #opensolaris |
| 15:27.26 | deeperror | why would 2 channels opened after a three way call not drop after both parties hang up? |
| 15:29.20 | umdstu | HarryR: thank you good sir |
| 15:29.44 | [TK]D-Fender | umdstu: SIP works jsut fine with most NAT's in my experience |
| 15:30.07 | umdstu | [TK]D-Fender: I dont know what I was doing then |
| 15:30.15 | [TK]D-Fender | ~sipnat |
| 15:30.15 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 15:30.17 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^ |
| 15:30.30 | umdstu | jbot: I'll take a look at that, thank you |
| 15:31.10 | umdstu | we've got several wireless networks doing NAT thru a PIX box |
| 15:35.03 | [TK]D-Fender | umdstu: AH! |
| 15:35.08 | [TK]D-Fender | umdstu: PIX = Death! |
| 15:35.09 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 15:35.10 | umdstu | yea |
| 15:35.15 | [TK]D-Fender | worst NAT... EVER |
| 15:35.17 | b11d | I hate PIX so much |
| 15:35.20 | umdstu | lol |
| 15:35.32 | umdstu | so do you think i'm screwed? |
| 15:35.35 | [TK]D-Fender | there is an entire WIKI page on it..... |
| 15:35.36 | b11d | it was totally screwing over SMTP because of it's "great" protocol fixup.. |
| 15:35.51 | umdstu | oh really? |
| 15:36.16 | aptura | I should be at work right now but the employee I work with is having heart troubles. |
| 15:36.34 | s0ck | HarryR: x series i had in mind |
| 15:36.43 | umdstu | i am at work right now |
| 15:36.51 | umdstu | fortunately, i get paid to do exactly what i'm doing here |
| 15:37.24 | aptura | umd, Well I could be at this job site but lifting a network cage up to the cieling is not exactly a one man job :) |
| 15:37.35 | Dr-Linux | when we dial local channel, it changes codec to "slin" which creates problem to record the files, how can i change this format to ulaw? |
| 15:38.02 | Dr-Linux | it doesn't record the whole conversation, i'm using "monitor" application for recordings |
| 15:38.06 | *** join/#asterisk coppice (n=chatzill@132.192.17.210.dyn.pacific.net.hk) |
| 15:38.08 | umdstu | aptura: weak.... it builds character |
| 15:38.17 | umdstu | :) |
| 15:38.19 | [TK]D-Fender | aptura: Time to learn about pullys and leverage ;) |
| 15:38.24 | umdstu | haha |
| 15:38.37 | datachomper | Or rockets and ductape |
| 15:38.40 | umdstu | [TK]D-Fender: the wiki you speak of, remember wher eto find it? |
| 15:38.45 | [TK]D-Fender | umdstu: Standing underneath and failing = data compression! |
| 15:38.49 | datachomper | ~wiki |
| 15:38.50 | [TK]D-Fender | ~wikis |
| 15:38.51 | jbot | i heard wikis is http://www.voip-info.org |
| 15:39.44 | umdstu | voip-info.org is great |
| 15:39.52 | umdstu | i wonder how many hundred times i've visited it |
| 15:40.34 | aptura | yea I know. I once watched how a fire pipping installer used a drill driver to mount a clevis into the concreate ceiling and then hooked on a 1 tone chain winch to lower a 100 bls halon bottle. He did a great job doing this. Last thing was to have this thing drop and then discharge. Halon displaces oxygen and can kill. |
| 15:41.13 | [TK]D-Fender | aptura: only if the place is really enclosed |
| 15:41.17 | Mercestes | yess....beautiful halon |
| 15:41.31 | Mercestes | silly broken lock...don't know how it jammed like that. |
| 15:41.35 | ManxPower | kind of pointless if it is not enclosed |
| 15:41.48 | Mercestes | I like chlorine better. |
| 15:41.55 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 15:42.01 | coppice | halon sucks as a fire extinguisher |
| 15:42.04 | aptura | TK it was enclosed! :) it was a federal CSI/Food inspection service. Would not look good if this did happen. |
| 15:42.29 | ManxPower | You'd think simple CO2 would do the job too |
| 15:42.59 | aptura | coppice I used to tote 200 bls hallon bottles on the flighline in the AF but never used them. Why do you think thay are not that effective? |
| 15:43.18 | *** join/#asterisk keulin (n=cray@AMontpellier-152-1-4-136.w81-251.abo.wanadoo.fr) |
| 15:43.33 | aptura | the USAF did use Co2 then switched to Hallon. |
| 15:43.45 | [TK]D-Fender | aptura: Without oxygen you wouldn't decompose anywhere NEAR as fast! So much better for bacteria reduction! |
| 15:43.58 | coppice | the problem with halon is nobody ever came up with reliable equipment to use it. long before most places banned it, lots of insurance companies insisted it was removed, and replaced with sprinklers |
| 15:44.33 | aptura | I heard it was replaced because it was a ozone depleating agent. |
| 15:44.47 | _ShrikE | I am getting TC400B: firmware zaptel-fw-tc400m.bin not available from userspace when I modprobe wctc4xxp. |
| 15:44.48 | coppice | that's the reason for the ban |
| 15:44.49 | [TK]D-Fender | coppice: I LOVE the sprinkler sitting right over my server rack... I wonder which would be better... fire or FLOOD ;) |
| 15:45.00 | _ShrikE | followed by wctc4xxp: probe of 0000:00:09.0 failed with error -1 |
| 15:45.09 | _ShrikE | ideas? |
| 15:45.17 | aptura | TK is your sprinkler system a wet or dry pipe system? |
| 15:45.27 | coppice | sprinklers have an excellent record for going off when needed, and doing no more than give the odd drip when not. |
| 15:45.30 | [TK]D-Fender | aptura: NO clue |
| 15:45.44 | Qwell[] | pfft, ozone...who cares? |
| 15:45.50 | [TK]D-Fender | coppice: regardless, my rack would be jsut as dead :) |
| 15:45.54 | Qwell[] | I want heat *now* |
| 15:45.54 | Mercestes | who needs ozone? |
| 15:46.04 | aptura | Well if there is a air compressor attached to it then its a dry pipe. Meaning there is no water in that pipe above your head. |
| 15:46.17 | coppice | you should be more concerned about the death of the building's occupants |
| 15:46.36 | ManxPower | coppice: people are cheaper than equipment. |
| 15:46.42 | [TK]D-Fender | coppice: they are replacable... this is my server rack you're talking about! |
| 15:46.52 | coppice | are your servers in cambodia, or something? |
| 15:47.39 | coppice | computer people have this idiotic fear of sprinklers. if they go off there is a fire, for goodness sake |
| 15:48.02 | aptura | well the fire is contained to that one room. |
| 15:48.25 | coppice | not with crappy halon that doesn't fire when needed |
| 15:48.53 | Mercestes | coppice: computers > people. |
| 15:49.03 | Mercestes | they give us everythign we need, love, companionship, entertainment... |
| 15:49.25 | Mercestes | people? People are evil, contrite, backstabbing bastards who should all die by anal electrocution. |
| 15:49.25 | coppice | and in the case of the P4, warmth in winter |
| 15:49.26 | Mercestes | all of them. |
| 15:49.26 | ManxPower | I thought that was cats |
| 15:49.34 | Mercestes | If they don't have an AC-line up their backend...put one there. |
| 15:49.56 | Mercestes | All things good plug into good ole AC. |
| 15:49.58 | Dr-Linux | ManxPower: i'm using "monitor" application for recording the calls, when i dial directly the extensions, it records the calls, but when i send the calls through local channel, it records little but don't record the whole conversation |
| 15:49.58 | b11d | i've got water sprinklers above my rack and a tiny little halon extinguisher on the wall.. thats what OFF-SITE backups are for i guess |
| 15:50.24 | x86 | so I've got a question... is there a way to make ChanSpy tell the spying party the source channel, and not the destination channel, of the bridged call? |
| 15:50.59 | puzzled | coppice: if you have a sec, any idea what's causing this unicall segfault: http://pastebin.ca/669284 |
| 15:51.03 | x86 | my managers spying on agents are getting confused because when they spy on agents, they randomly hear different channel numbers for the same agent |
| 15:51.15 | Dr-Linux | ManxPower: so should i use some other "mixmonitor" instead or anything your suggest? |
| 15:51.23 | x86 | (because it's reporting the destination channel, not the source channel) |
| 15:51.26 | coppice | puzzled: its probably the bugs |
| 15:51.45 | ManxPower | Dr-Linux: I suggest you read the fucking file I told you to read. |
| 15:52.01 | deeperror | is there anyway to setup a timeout on a native bridge between 2 sip channels? |
| 15:52.23 | aptura | [TK]D-Fender what happens if the fire alarm goes off is the power to the NOC cut off and then it goes into UPS mode? I worked for a large network company that had that arangment once. Unfortunaly the idiotic sprinker techs went to test the alarm and the UPS for some reason did not come on and all 100 servers shut down costing the company 1 million in lost sales for the day. |
| 15:52.29 | umdstu | harsh |
| 15:52.31 | *** join/#asterisk minkus (n=minkus@pool-72-84-53-31.clrkwv.east.verizon.net) |
| 15:52.50 | ManxPower | deeperror: yes. Just like you do with a non-bridge. only the audio is bridged, not the signaling |
| 15:52.51 | aptura | Testing is normally done on sundays. |
| 15:53.02 | puzzled | coppice: it probably is yes :) is there perhaps a simple fix or is asterisk 1.2 with unicall not something I should pursue? |
| 15:53.07 | umdstu | i wish they would let me in the NOC |
| 15:53.11 | umdstu | they don't trust me |
| 15:53.22 | coppice | puzzled: |
| 15:53.24 | coppice | #0 0x00c89bca in r2_mf_rx_init (s=0x100, fwd=1) at bell_r2_mf.c:773 |
| 15:53.25 | coppice | since s is a pointer to an R2 tone receiver context, I guess it shouldn't be 0x100 |
| 15:53.27 | Mercestes | <PROTECTED> |
| 15:53.31 | umdstu | lol |
| 15:53.52 | umdstu | Mercestes: what do you do |
| 15:54.03 | aptura | noc is a noc ;) biggest one other then microsoft is amazon when I was being interviewd for a cisco position. |
| 15:54.08 | Mercestes | I'm an infrastructure administrator |
| 15:54.25 | deeperror | ManxPower: I setup an absolute timeout for the call and this works. The bigger issue what woudl cause the channels to remain open even after the callers have hung up the phone? |
| 15:54.28 | aptura | Mercestes nifty |
| 15:54.45 | ManxPower | deeperror: are you experiencing this? |
| 15:54.50 | Mercestes | Yea, that title means, "that guy that took over everyone else's job when they quit." |
| 15:55.20 | deeperror | yes after transfers occur they stay open indefinately until asterisk is restarted |
| 15:55.35 | deeperror | transfers = 3 way calling transfers with flash |
| 15:56.08 | deeperror | so last night i had 12 channels open for 11 hours haha |
| 15:56.11 | ManxPower | deeperror: weird. a SIP phone with a FLASH button?: |
| 15:56.23 | ManxPower | are the calls connected to the PSTN? |
| 15:56.28 | deeperror | analog phones over analog |
| 15:56.46 | aptura | Mercestes ever change out large batteries before? |
| 15:56.49 | deeperror | these are SIP connections |
| 15:56.55 | Mercestes | aptura, many times. |
| 15:57.01 | Mercestes | I do everything. |
| 15:57.06 | deeperror | to bandwidth.com for sip trunking |
| 15:57.18 | ManxPower | deeperror: In many locations analog lines to not signal asterisk when the far end hangs up. |
| 15:57.25 | Mercestes | FR, MPLS, Asterisk, VoIP, SIP, PSTN, POTS, T1s, PRIs, C#, VB.Net, Linux, Unix, Microsoft. et al |
| 15:57.32 | Mercestes | UPS's. |
| 15:57.37 | puzzled | coppice: uhm ok. my lack of C voodoo doesn't help me here. don't have enough knowledge to fix this. guess I'll remove unicall from the rpms then |
| 15:57.42 | aptura | I had my first experaince a few weeks ago. Each weighing in at 450 bls each. Good experaince. |
| 15:58.10 | Mercestes | a nice one ton floor-jack comes in handy sometimes. |
| 15:58.17 | *** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 15:58.22 | deeperror | ManxPower: is there a bridge timeout or a configuration area for how to handle those calls similar to the rtptimeout? |
| 15:59.15 | aptura | Mercestes yea this was a special jack for brining these huge things to waist height. I would not mind doing more of these. Low stress and can pace your self ;) |
| 15:59.22 | ManxPower | deeperror: I don't even know what the hell you are doing. First you say SIP, then you say analog, then you say "sip trunking", which does not exist in Asterisk. |
| 16:00.46 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 16:00.51 | Mercestes | Manxpower: "Are you using, Sip, IAX, or Analog." "Yes." |
| 16:01.04 | ManxPower | deeperror: Asterisk will disconnect a call if it sees a disconnect message/signal. |
| 16:01.06 | deeperror | analog phone makes a call - out over SIP - 3 way flash line - call party b - all three are talking when agent hangs up then callee A & B remain in the call to talk it out....after they hang up the two SIP channels remain open that have been bridged together |
| 16:01.06 | Mercestes | aptura, I wanna come work with you |
| 16:01.14 | aptura | :) |
| 16:01.26 | ManxPower | I guess if you were doing REINVITES and the person unplugged their phone, asterisk would not hangup the call. |
| 16:01.33 | aptura | Mer this was a three week job. But I personally would not mind doing more. |
| 16:01.39 | Mercestes | deeperror, so you have an analog phone hooked to an ATA running to asterisk over SIP to terminate out over broadvoice?? |
| 16:01.57 | Mercestes | deeperror, and then they hit hte flash button? |
| 16:02.03 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 16:02.07 | deeperror | analog phone - channel bank - t1 card in asterisk box |
| 16:02.15 | ManxPower | deeperror: I would suspect a phone issue. |
| 16:02.18 | deeperror | outbound calls go out over SIP |
| 16:02.26 | ManxPower | since asterisk is not seeing the phones hangup. |
| 16:02.55 | deeperror | however the 12 lines that were active last night were 12 different numbers that were dialed |
| 16:03.01 | deeperror | 12 different phones |
| 16:03.29 | ManxPower | deeperror: I have never, in all my years of using asterisk heard of this issue. |
| 16:04.00 | deeperror | first time for everything right? |
| 16:04.07 | ManxPower | set canreinvite=no in all the sip.conf entries to disable reinvites, but I don't think this is your problem. |
| 16:04.09 | deeperror | how can i prove this |
| 16:04.23 | deeperror | ok i'll set to no |
| 16:04.45 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
| 16:05.07 | *** join/#asterisk variable_office (n=variable@cerberus.iswan.net) |
| 16:05.39 | variable_office | can the RealTime command be used to retreive multiple rows? |
| 16:05.54 | deeperror | i've made that change now |
| 16:06.00 | Zeeek | In 25 minutes, the weekly Asterisk Users Conference will begin. #asterisk-users-conference for more info |
| 16:06.10 | deeperror | i had a test run last night with 986 calls being made out |
| 16:06.36 | deeperror | we had in that time 6 transfers occur and when i came in this am there were 12 channels 'sip show channels' still open |
| 16:06.38 | Zeeek | further, http://AsteriskUsersConference.org |
| 16:08.34 | ManxPower | rickross: time for a new provider, eh? |
| 16:08.34 | rickross | bonus? is it free? are these guys total jokers? |
| 16:09.01 | Mercestes | Of coruse they are. |
| 16:09.07 | Mercestes | They can't even spell "Viltality" |
| 16:09.10 | Mercestes | ... |
| 16:09.12 | Mercestes | Vitality. |
| 16:09.22 | Mercestes | I would like to mention...i do *NOT* run a phone company. |
| 16:09.30 | Qwell[] | I do |
| 16:09.36 | rickross | ok, where should we be then? I need a reliable, cost-effective business voip provider |
| 16:09.38 | Zeeek | Mercestes: I thought you did! |
| 16:09.52 | Mercestes | nah, just a consulting business. |
| 16:10.10 | rickross | it completely baffles me that a provider would blow off such an important part of business telcom |
| 16:10.13 | ManxPower | rickross: All ITSPs suck. Teliax seems to suck less than most. |
| 16:10.23 | aptura | Sounds like awfull jitter in the asterisk users conferance login session. |
| 16:10.32 | *** join/#asterisk Strom_C (n=strom@160.79.29.94) |
| 16:10.40 | JunK-Y | after qwest, here's qwell :) |
| 16:13.10 | *** join/#asterisk el_critter (n=chatzill@190.74.100.35) |
| 16:13.40 | Strom_C | haha |
| 16:13.43 | Strom_C | qwellst |
| 16:13.51 | Qwell[] | qwellcomm |
| 16:13.56 | Strom_C | or, if Sprint buys Qwest: |
| 16:13.58 | Strom_C | Sprest |
| 16:14.08 | Strom_C | the refreshing taste of telecom |
| 16:14.14 | Zeeek | Consultants are like hookers... what they earn, they spend on makeup |
| 16:14.25 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-243-162-33.bflony.east.verizon.net) |
| 16:14.27 | Strom_C | makeup? |
| 16:14.28 | Zeeek | if you see the analogy that is |
| 16:14.40 | Mercestes | .... |
| 16:14.42 | Zeeek | explaining jokes never works |
| 16:14.45 | Mercestes | that was just shy of retarded. |
| 16:14.55 | Zeeek | I resemble that remark |
| 16:15.03 | Strom_C | retorted |
| 16:15.08 | Qwell[] | ... |
| 16:15.59 | bkw__ | Strom_C, it'll be Pest |
| 16:16.03 | bkw__ | if sprint and qwest merge |
| 16:16.47 | coppice | Zeeek: explaining to jokes has its problems too :-) |
| 16:16.47 | Zeeek | the enemy always merges to defend against open software, eh? |
| 16:17.06 | Mercestes | huh? |
| 16:17.16 | aptura | Would not suprise me if these phone companies CEO's are buddies. Seems to happen here in British Columbia. |
| 16:17.21 | Mercestes | to "defend against" ...does that mean you are pro open source or anti open source? |
| 16:17.55 | jhiver | hi |
| 16:18.08 | jhiver | how do i check that my IAX2 trunk is effectively trunking? |
| 16:18.16 | jhiver | i mean |
| 16:18.17 | Zeeek | salut jhiver. Tu devrais couter la conférence |
| 16:18.27 | Zeeek | Ecouter |
| 16:18.28 | b11d | non! |
| 16:18.34 | Zeeek | (pas d'accents ici) |
| 16:18.43 | JunK-Y | n'écoute pas la conférence, il y a des femmes en chaleur :) |
| 16:18.50 | b11d | you take that back! |
| 16:18.54 | jhiver | hey Zeeek =) |
| 16:19.04 | jhiver | i have this IAX2 'trunk' |
| 16:19.13 | jhiver | but i don't think it's stripping UDP headers at all |
| 16:19.29 | jhiver | how do i check wether this is active (other than doing bandwith measurements) |
| 16:20.08 | jhiver | et comment on fait pour l'ecouter cette conf chaude? =) |
| 16:20.27 | *** join/#asterisk tsurko (n=tsurko@77.70.15.52) |
| 16:20.36 | Zeeek | jhiver: http://AsteriskUsersConference.org/join.php |
| 16:21.10 | Zeeek | Call using SIP or PSTN. Dans ton cas, SIP me semble plus interessant :) |
| 16:21.52 | jhiver | flemme de configurer =) |
| 16:22.16 | Zeeek | Bon, va sur la page et ecoute en direct |
| 16:22.20 | Zeeek | via l web |
| 16:22.45 | aptura | Not Found |
| 16:22.45 | aptura | The requested URL /join.php was not found on this server. |
| 16:22.46 | Zeeek | <PROTECTED> |
| 16:23.02 | jhiver | oh lala faut s'enregistrer |
| 16:23.03 | Zeeek | heh, I must not have called it that, then |
| 16:23.07 | Zeeek | MAIS NON |
| 16:23.09 | jhiver | j'écouterais l'archive |
| 16:23.14 | Zeeek | jhiver anyone can listen |
| 16:23.19 | Zeeek | ah |
| 16:23.35 | Mercestes | Mille Bournes |
| 16:23.35 | Zeeek | true the player is either flash or QT |
| 16:23.37 | jhiver | ok |
| 16:23.44 | Zeeek | http://www.talkshoe.com/talkshoe/web/talkCast.jsp?masterId=22622&cmd=tc |
| 16:23.58 | Zeeek | Click on listen when the conf is open which is in aout 8 minutes |
| 16:24.41 | jhiver | a chaque fois me demande de me logguer |
| 16:24.59 | jhiver | c nul =) |
| 16:25.35 | jhiver | any ideas how i can check that asterisk is effectively trunking? |
| 16:25.54 | Mercestes | FOP maybe? |
| 16:26.13 | jhiver | because right now my IAX2/g729 trunk seems to be using 25kbps / call, which is about right if the trunking doesn't work |
| 16:26.29 | jhiver | but it did define trunk=yes and trunkfreq=40 |
| 16:26.31 | ManxPower | jhiver: how many calls? |
| 16:26.38 | ManxPower | jhiver: on BOTH sides of the connection? |
| 16:26.39 | jhiver | 11-12 |
| 16:26.43 | jhiver | i think so |
| 16:26.46 | jhiver | lemme check |
| 16:27.24 | jhiver | on on receiveing end under [general] i have trunkfreq=40 and trunktimestamps=yes |
| 16:27.34 | jhiver | and under [dedibox] i have trunk=yes |
| 16:27.38 | jhiver | now on sending end |
| 16:28.17 | jhiver | well, i have the same |
| 16:28.40 | jhiver | so, i don't understand |
| 16:29.51 | jhiver | right now it shows 9 channels and network throughput (incoming + outgoing) to the trunk is about 434kbps |
| 16:30.06 | jhiver | so about 200+ kbps / direction |
| 16:30.20 | jhiver | which for 9 channels sounds about right if IAX2 trunking isn't working |
| 16:30.31 | jhiver | since it's all g729 |
| 16:31.00 | jhiver | i don't get it! |
| 16:31.45 | jhiver | not so long ago i had 22 channels and throughput was about doubled, so... |
| 16:31.49 | jhiver | any ideas? |
| 16:31.54 | *** join/#asterisk Lawbringer (n=Lawbring@212.183.134.66) |
| 16:32.22 | jhiver | or pehaps trunk=yes needs to be a global parameter? |
| 16:35.29 | russellb | Zeeek: is there an IRC channel for the conf? |
| 16:35.59 | Zeeek | #asterisk-users-conference |
| 16:36.52 | russellb | ah, forgot the dashes |
| 16:37.28 | Zeeek | Dash *this* |
| 16:37.41 | jhiver | ok, apparently trunking is NOT active |
| 16:37.58 | jhiver | if you do IAX2 show peers it should show (T) |
| 16:38.00 | Zeeek | OOPS |
| 16:38.08 | jhiver | and when i do it it shows (S) |
| 16:38.23 | jhiver | so the question is |
| 16:38.27 | jhiver | WHY WHY WHY OH WHY |
| 16:38.29 | jhiver | =) |
| 16:39.51 | hmmhesays | well I just ordered a macbook from amazon |
| 16:40.18 | Qwell[] | hmmhesays: when will it arrive at my doorstep? |
| 16:40.20 | aptura | Who knows what is the most bug free and stable version of asterisk running in a production enviroment right now? I met up with a client the other day and know who installed asterisk in there office and its had issues. Unfortunaly thay are a channel regular here. |
| 16:40.30 | hmmhesays | Qwell[]: more likely at my doorstep |
| 16:40.37 | jhiver | has somebody got a working IAX2 trunk sample configuration file i could use? |
| 16:40.41 | hmmhesays | everywhere is sold out of these things except amazon |
| 16:40.43 | Qwell[] | stupid amazon, always shipping to the wrong place |
| 16:41.02 | hmmhesays | 5 mac stores around msp area and none had the 1799 model |
| 16:42.07 | Netgeeks | wow, 5 stores sold out of them at once? wtf is up with that? |
| 16:42.35 | Netgeeks | oh, back to school.... duh |
| 16:42.48 | [TK]D-Fender | jhiver: You running zaptel? |
| 16:42.56 | jhiver | yes |
| 16:42.59 | jhiver | ztdummy |
| 16:43.08 | hmmhesays | should I pay the extra money for 128 more video memory? |
| 16:43.10 | hmmhesays | 300 bucks |
| 16:43.15 | Qwell[] | no, heh |
| 16:43.18 | [TK]D-Fender | jhiver: should have trunk on both ends. Got timing on BOTH? |
| 16:43.21 | jhiver | I'll cut and paste my iax2 config files in pastebin |
| 16:43.22 | jhiver | yes |
| 16:43.26 | hmmhesays | and the free ipod |
| 16:43.29 | hmmhesays | ipod nano |
| 16:43.30 | jhiver | i've installed debian etch on both |
| 16:43.33 | Qwell[] | eh... |
| 16:43.35 | jhiver | same kernel version |
| 16:43.40 | jhiver | same everything =) |
| 16:44.06 | Mercestes | mooooooooggggg! |
| 16:44.09 | jhiver | if you know a way to check wether timing is enabled... |
| 16:44.13 | jhiver | i'll take it |
| 16:44.21 | Mercestes | mog: I sent the thing you told me to send to the place you told me to send it too. |
| 16:44.37 | mog | ? |
| 16:44.50 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 16:45.18 | jhiver | i can run zttest on both boxes mind you |
| 16:45.42 | jhiver | ugh |
| 16:45.52 | jhiver | unable to open zap interface |
| 16:45.55 | jhiver | what happened? |
| 16:46.10 | jhiver | ok |
| 16:46.15 | [TK]D-Fender | jhiver: go modprobe both, ztcfg and restart * :p |
| 16:46.26 | jhiver | doing a modprobe ztdummy fixed it |
| 16:46.41 | jhiver | ztcfg doesn't seem to do jack |
| 16:47.50 | jhiver | ok zttest works for both |
| 16:48.03 | jhiver | so an asterisk -rx 'restart now' + a test seems in order |
| 16:48.34 | jhiver | only 4 calls |
| 16:48.38 | jhiver | let's drop em =) |
| 16:49.34 | jhiver | ok now i see (T) |
| 16:49.36 | jhiver | hurray |
| 16:49.39 | jhiver | i need to test |
| 16:50.04 | *** join/#asterisk Peaceful (n=peaceful@70.98.162.62) |
| 16:50.31 | Peaceful | Does anyone have experience buying from www.ipphone-warehouse.com ??? Are they ok? |
| 16:51.02 | jhiver | ok |
| 16:51.07 | jhiver | test successful! |
| 16:51.16 | jhiver | at the beginning the call was a bit jitter-ish |
| 16:51.20 | jhiver | but then it worked |
| 16:51.22 | jhiver | cool |
| 16:51.50 | [TK]D-Fender | Peaceful: Excellent pricing |
| 16:52.36 | robl^ | Peaceful: I use Attacomm or TelephonyDepot most of the time.. never tried ipphone-warehouse |
| 16:52.45 | jhiver | [TK]D-Fender : you the man =) |
| 16:53.02 | *** join/#asterisk Victor_Yure (n=aaaa@dagsrv08.guarany.com.br) |
| 16:53.04 | [TK]D-Fender | robl^: Telephonydepot.com is almost the same pricing and I've bought from them as have my customers. |
| 16:53.12 | [TK]D-Fender | jhiver: You're welcome |
| 16:53.34 | robl^ | Yeah. TD is my favorite vendor. Just got 15 Aastra 9133is from them yesterday |
| 16:54.00 | aptura | Attacom is good. |
| 16:54.31 | rickross | any Teliax customers here? |
| 16:54.44 | [TK]D-Fender | SPA8000 = sick price... |
| 16:54.47 | rickross | someone above mentione dthey are pretty goo - just had a good talk with a rep |
| 16:54.55 | [TK]D-Fender | robl^: AASTRA's?! ew |
| 16:56.04 | *** join/#asterisk NOT_guru (n=chatzill@24-241-103-142.static.stls.mo.charter.com) |
| 16:57.08 | robl^ | [TK]D-Fender: yeah. the client has Norstar currently.. The older Norstars phones are similar to the Aastras 9133is.. so the client really wanted those instead of Polycom |
| 16:58.25 | [TK]D-Fender | robl^: SCHMUCKS |
| 16:58.29 | *** part/#asterisk Victor_Yure (n=aaaa@dagsrv08.guarany.com.br) |
| 16:58.31 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 16:58.33 | riddlebox | [TK]D-Fender: do you like any phones lol |
| 16:58.47 | robl^ | [TK]D-Fender: better than they demanded Grandstream BT101 |
| 16:58.47 | [TK]D-Fender | riddlebox: Yes... Polycom :) |
| 16:59.01 | [TK]D-Fender | robl^: Shit looks very good... when compared to CRAP :p |
| 16:59.55 | robl^ | My favorites (in order) are Polycom, Aastra 9133i/480, then Snom, then the NEW Aastras.. everything else isn't worth using |
| 17:00.01 | *** join/#asterisk alerios (n=alerios@190.24.96.105) |
| 17:00.22 | aptura | robl, in other words the other phones can ruin your companys reputation ;) |
| 17:01.14 | riddlebox | I have some Grandstream phones at home the GXP-2000 and the work perfectly, but if we sell an asterisk system it will be using polycom phones because they actually say asterisk on the phones |
| 17:01.32 | robl^ | aptura: basically. They don't work too well and make asterisk look like not such a good platform to build on.. |
| 17:01.43 | *** join/#asterisk Bladerunner05 (n=feelme@81-174-56-54.static.ngi.it) |
| 17:01.46 | [TK]D-Fender | robl^: For normal use I put Linksys above the New Aastra's |
| 17:01.48 | alerios | Hi... silly question: is it possible to configure a PRI line in something like read-only mode? like to spy/monitor a PRI line connected to another PBX |
| 17:02.10 | [TK]D-Fender | robl^: Aastra never learned how to to a phone UI and the 5i screen usage and keys SUCK |
| 17:02.20 | robl^ | Grandstreams WORK, but the quality is pretty poor. Sound quality is marginal. Firmware tends to be very buggy from one version to the next. I have some Grandstream for test phones.. never put one in production though |
| 17:02.38 | [TK]D-Fender | alerios: No, but you can put * BETWEEN the two transparently. |
| 17:02.39 | jhiver | I saw one GXP-2000, it was sending a local ringtone even before receiving a 180 or 183 |
| 17:02.47 | jhiver | thought that was a bit of a joke to be honest |
| 17:02.51 | riddlebox | robl^: thats kinda what I use them for, my home phones |
| 17:03.16 | [TK]D-Fender | I use an IP 501 & 301 for my home phones :) |
| 17:03.19 | alerios | [TK]D-Fender: I wonder how some hardware makes it |
| 17:03.45 | alerios | [TK]D-Fender: they just derive the VNC cables and record the PRI calls |
| 17:03.45 | riddlebox | [TK]D-Fender: I just dont have alot of money to invest in my IP phones yet |
| 17:04.14 | robl^ | I haven't tried any of the Linksys phones.. Also, I haven't tried cisco in a LONG time.. feature-wise / price wise / and licensing issues.. they just were not worth the effort |
| 17:04.48 | riddlebox | it does seem like we are loosing a ton of bids lately to Cisco vendors |
| 17:05.04 | coppice | the price you pay for a cisco can vary a lot |
| 17:05.06 | flujan | someone using asterisk over windows? it is not to production enviroment. |
| 17:05.06 | robl^ | My "home" phones are Polycom 601s and an Aastra 480i. My "test phones" are much more varied. |
| 17:05.20 | aptura | robl, yea and gives it a bad name. BTW I talked to one of our clients who also happens to be a client of one of the channel users here and he was not to impressed with the new phone install. Has jitter echo ect. Last thing is I want to hear this. The phones thay are using are the aastra 51i |
| 17:05.24 | flujan | just my boss want to try it... and he actually knows nothing about linux. :D |
| 17:05.31 | riddlebox | but then we get hired to help the Cisco vendors during the cut, because they dont know how to setup PRI's or T1's |
| 17:06.10 | robl^ | aptura: yup. The new Aastras have firmaware glitches... and they are just plain ugly |
| 17:06.23 | aptura | really |
| 17:06.34 | aptura | well thay should have done there home work and avoid them. |
| 17:06.41 | riddlebox | I got into an argument with a Cisco vendor on a cut he hired us to be there, he said I had to run a cat 3 cable for a fax machine, I told him no we didnt cause there was a cat 5 cable there, and he laughed at me and that a fax cannot go over cat 5 |
| 17:07.07 | coppice | ugly is normal for phones |
| 17:07.44 | ManxPower | riddlebox: did you check the wire and discover that is is in fact cat5? |
| 17:07.44 | ManxPower | ..er... cat3? |
| 17:07.44 | aptura | coppice thay are OKAY not in looks though. Its really what the customer wants as long as its EASY to configure and RELIABLE. |
| 17:08.00 | robl^ | coppice: I think the newer Polycoms 650 and 320s are sexy.. |
| 17:08.25 | tzanger | heh |
| 17:08.35 | riddlebox | ManxPower: I ran the cable, the customer is a technology company and they only wanted cat5 in their building, the customer even looked at the Cisco vendor and asked him if he knew what he was doing |
| 17:08.38 | aptura | Looks like day is going to be shot again :| this bites. I am hunting for a new company to work for this has happened way to many times. |
| 17:08.40 | coppice | the polycom 320 is certainly reasonably priced. probably the first IP phone that looks almost worth the money |
| 17:09.15 | *** join/#asterisk guillote_GNU (n=guillote@190.7.27.17) |
| 17:12.59 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@216-110-94-230.static.twtelecom.net) |
| 17:16.20 | jhiver | now this is cool |
| 17:16.42 | jhiver | for 10 calls currently, the inbound + outbound throughput is 257kbps |
| 17:17.12 | jhiver | that's ~= 13kbps per call |
| 17:17.15 | jhiver | not bad =) |
| 17:17.34 | datachomper | How are you measuring throughput? |
| 17:17.34 | jhiver | trunking does seem to work after all =) |
| 17:17.36 | [TK]D-Fender | riddlebox: PolycomIP 320/330 cost LESS than Aastra 9133's unless you've got some sort of sick deal not seen in retail.... |
| 17:17.40 | jhiver | using ntop |
| 17:18.22 | *** join/#asterisk ToTo (n=ToTo@host72-142-dynamic.8-87-r.retail.telecomitalia.it) |
| 17:18.38 | jhiver | load average on box: 0.00 |
| 17:18.40 | jhiver | cool =) |
| 17:19.08 | jhiver | well there's no transcoding or anything like that |
| 17:19.12 | riddlebox | [TK]D-Fender: I havent looked at the Aastra phones in a long time |
| 17:19.47 | [TK]D-Fender | riddlebox: Just mixed you androbl up for a sec |
| 17:19.59 | jhiver | so if you're having 10 g729 calls IAX2 trunking divides your req. bandwith usage by a factor of 2 |
| 17:20.09 | jhiver | and i suppose the more calls, the more interesting it becomes |
| 17:20.14 | jhiver | nice |
| 17:20.30 | jhiver | now I'll be able to squeeze out a lot more capacity out of this 2 Mbps link =) |
| 17:20.31 | *** join/#asterisk ptiggerdine (n=ptiggerd@123-243-144-208.tpgi.com.au) |
| 17:21.04 | jhiver | i hope it'll cope just as fine with 100 calls |
| 17:22.18 | [TK]D-Fender | jhiver: 100calls over IAX2 *trunk* ....lol sounds NASTY, and there's no way I can picture this surviving... |
| 17:22.36 | jhiver | really? why not? |
| 17:23.01 | jhiver | seems to work fine for 10 calls =) |
| 17:23.14 | aptura | thats 10 calls |
| 17:23.20 | jhiver | i'll go progressively |
| 17:23.30 | jhiver | i'm not going to switch over all the gateways overnight |
| 17:23.32 | aptura | best to have more then one server running |
| 17:23.54 | jhiver | yeah you're right, pehaps i'll do 2 trunks with loadbalancing on both + failover |
| 17:24.01 | jhiver | just in case one box crashes |
| 17:24.12 | aptura | Are there reasonable blade servers on the market to split up voip trunks? |
| 17:24.45 | jhiver | these look cheap |
| 17:24.45 | jhiver | http://www.colomachine.com/ |
| 17:25.16 | jhiver | 10 2.6 celeron blades for about 6k USD |
| 17:25.35 | jhiver | with chassis |
| 17:26.32 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-189-7-234.lsanca.fios.verizon.net) |
| 17:26.37 | jhiver | anyway, you had issues with IAX2 trunking? because if it's known not to work, i could be better off spending some cash on RAD solutions or something |
| 17:26.44 | jhiver | those are pretty impressive |
| 17:27.12 | jhiver | they manage to squeeze about 1 E1 with =~ 160kbps |
| 17:27.22 | jhiver | when using silence suppression + g.723 |
| 17:27.29 | dahunter3 | I just signed up for voip service and it seems to work fine but the iax provisioning that they gave me seems sort of half-duplex. Is that my fault with the configuration or something I should mention to them? |
| 17:27.34 | aptura | btw downtown vancouver had a telco cable intrusion with the canada line being built. The construction crew acidently drilled into one of the major copper or fiber lines bring down 200k customers. The news reporter said it was difficult to repair because of the comment he read "it is like trying to sew up jello together" I wonder what this means? I think these lines are packed with some kind of weather resitant jell or something. Anyone |
| 17:28.28 | jhiver | any ideas when asterisk will support VAD BTW? It would be nice to get rid of these pesky "VAD frame at the end" warnings =) |
| 17:28.39 | aptura | The cost of the repairs were in the hundreds of thosuands of dollars. I am not sure if this includes the lost business the customers may have experainced. |
| 17:29.26 | *** join/#asterisk GlubZor (n=yann@vig38-3-82-228-184-159.fbx.proxad.net) |
| 17:31.23 | flujan | guys, does asteriskwin32 supoorts digium cards? |
| 17:32.21 | jhiver | asteriskwin32 still exists? =) |
| 17:32.36 | jhiver | my guess would be /NO/ |
| 17:32.41 | flujan | yeap... dunno if it works... |
| 17:32.42 | flujan | :D |
| 17:32.42 | aptura | never consider asterisk to work on a OS that hangs or crashes. |
| 17:32.59 | flujan | it is not to a production mode... |
| 17:33.02 | jhiver | but then why would you want to use a) windows for a server or b) digium hardware ? =) |
| 17:33.06 | aptura | kiss your consulting income good bye if it does ;) |
| 17:33.12 | many | it doesnt matter, the interesting thing would be if there are windows drivers for the HW. |
| 17:33.14 | flujan | both of them |
| 17:33.28 | many | and afaik its a clear no. |
| 17:33.44 | flujan | i need a tdm card to run on a windows machine with a simple pbx server that |
| 17:33.48 | jhiver | IMHO it's best to use dedicated SIP gateways anyway |
| 17:33.59 | jhiver | i've never had very good results with digium cards |
| 17:34.05 | blitzrage | hardware does not work on Windows |
| 17:34.09 | flujan | i) tries to dial through a sip line... if it fails, tries to dial from a pstn line attached to i |
| 17:34.26 | flujan | blitzrage: nor software. LOL |
| 17:34.28 | blitzrage | and why would you want to run on Windows in the first place? I guess some people are more masochistic than me |
| 17:34.37 | [TK]D-Fender | flujan: Put. Down. The. Crack. Pipe. |
| 17:34.47 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 17:34.54 | jhiver | well, pehaps a windows only company policy =) |
| 17:34.58 | flujan | blitzrage: my dammed boss want to get project for a bank |
| 17:35.15 | many | now |
| 17:35.16 | blitzrage | flujan: well, tell him it's not possible :) |
| 17:35.19 | many | that scares me |
| 17:35.21 | jhiver | and they want WINDOWS =) |
| 17:35.48 | jhiver | yes it's possible, use your win32 asterisk port with server 2003 (which is fairly stable) and get a SIP <-> PSTN gateway |
| 17:35.51 | jhiver | problem sorted |
| 17:36.05 | *** join/#asterisk ToTo (n=ToTo@host72-142-dynamic.8-87-r.retail.telecomitalia.it) |
| 17:36.58 | flujan | jhiver: bah, thinks are not so easy. |
| 17:37.09 | jhiver | why not? |
| 17:37.47 | jhiver | what's wrong with this solution? you're not gonna tell me that a BANK can't afford a SIP <-> PSTN gateway, especially if they can afford windows server! |
| 17:38.23 | jhiver | or you could setup a linux box and run windows in WMWare to make them believe it's windows 2003 server =) |
| 17:38.39 | jhiver | and then your inbuilt hardware would work |
| 17:38.44 | blitzrage | asteriskwin32 was running 1.2..... is it even being kept up to date to run 1.4 ? |
| 17:38.52 | aptura | anyone listing to the live asterisk discussion on talkshoe? |
| 17:39.11 | [TK]D-Fender | jhiver: TDM card on a SEPERATE server. Stop trying to screw them over... |
| 17:39.31 | jhiver | i'm just kiddin [TK]D-Fender |
| 17:39.42 | jhiver | I've made it clear i wouldn't even consider a TDM card |
| 17:40.16 | jhiver | you get a hell of a lot better echo cancellation and codec conversion with an audiocodes or something anyways |
| 17:40.30 | flujan | the dammed bank loves the analog line |
| 17:41.04 | jhiver | then get an audiocodes Mediapack |
| 17:41.23 | jhiver | these little units work great |
| 17:41.35 | jhiver | and if they can afford T0 ISDN, go get a patton smartnode |
| 17:41.44 | jhiver | great hardware, rock solid, great sound |
| 17:42.21 | jhiver | if you use digium X100P clones on analog line, well... good luck |
| 17:42.40 | jhiver | that's just my experience |
| 17:42.42 | jhiver | anyways |
| 17:42.49 | *** join/#asterisk djs_2_6 (n=djstillm@cpe-075-182-081-167.nc.res.rr.com) |
| 17:43.59 | *** join/#asterisk CVirus (n=GoD@212.12.250.74) |
| 17:45.04 | datachomper | So what's the generally reccomended PSTN -> SIP gateway brand? |
| 17:45.14 | marc7 | jhiver: digium gear (even straight FX0 cards) have a noticeable value-added over other random clones? |
| 17:45.15 | aptura | sangoma |
| 17:45.21 | marc7 | have/has |
| 17:45.37 | jhiver | for analog audiocodes mediapacks are good |
| 17:45.49 | jhiver | for T0, patton or vegastream are ok |
| 17:46.02 | jhiver | for T2, audiocodes or vegastream, preferably audiocodes mediant 2000 |
| 17:46.13 | jhiver | all have worked fantastic for me |
| 17:46.18 | jhiver | never crashed |
| 17:46.52 | robl^ | or use a channel bank --- |
| 17:46.56 | robl^ | like Adtran |
| 17:46.57 | datachomper | What if I'm looking for a call center sized router that could perform T1 -> sip gateway, plus interface with SS7 ? |
| 17:48.20 | aptura | jhiver never crashed in how many months or years under what load? |
| 17:51.48 | riddlebox | can you use a RDSP/432 4-line voice board, the Rhetorex with asterisk? |
| 17:52.36 | *** part/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 17:57.03 | [TK]D-Fender | riddlebox: DEFINITELY rhetorical ;) |
| 17:57.31 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 17:57.41 | *** part/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 17:57.56 | riddlebox | [TK]D-Fender: just wondering, we have a bunch of these boards that use with Amanda voicemails |
| 18:00.57 | robl^ | what's the sox option to convert to .ulaw? |
| 18:02.40 | Mercestes | http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk |
| 18:02.55 | *** join/#asterisk KW-CHS (n=none@208.189.33.150) |
| 18:03.30 | KW-CHS | Anyone done SLA? I'm trying to get it setup and I'm lost. |
| 18:03.32 | robl^ | Mercestes: that doesn show ulaw native.. |
| 18:03.43 | Mercestes | so it doesn't. =/ |
| 18:03.52 | Mercestes | someone should add that. |
| 18:03.54 | robl^ | Mercestes: checked earlier. Otherwise I wouldn't have asked ;-) |
| 18:04.22 | *** join/#asterisk dsfr (i=dsfr@pdpc/sponsor/digium/dsfr) |
| 18:04.25 | Mercestes | lol. Sorry, forgot it was you. ;) |
| 18:04.53 | robl^ | Mercestes: you should be punished! Maybe I'll just convert to sln for now |
| 18:05.08 | dahunter3 | my iax2 connection seems half-duplex. where should I look for the problem? |
| 18:05.13 | Mercestes | http://forums.whirlpool.net.au/forum-replies-archive.cfm/535002.html |
| 18:05.20 | robl^ | I hate transcoding on the fly. prefer to have everything in the native format of the codec |
| 18:05.21 | riddlebox | [TK]D-Fender: I guess I would need drivers for the cards then huh |
| 18:05.48 | [TK]D-Fender | riddlebox: get coding! |
| 18:06.07 | riddlebox | lol what a project huh |
| 18:06.09 | robl^ | Mercestes: thanks, mate!!! |
| 18:06.15 | Mercestes | np. |
| 18:06.26 | [TK]D-Fender | KW-CHS: * SLA = 2-bit hack only marginally usable in a handful of cases |
| 18:06.28 | Mercestes | googled asterisk wav to ulaw |
| 18:06.34 | robl^ | now if I can remember my wiki password, I'll add it |
| 18:07.07 | robl^ | Mercestes: hrmm.. I tried googling but ended up with 50,000 irrelevant hits |
| 18:07.55 | KW-CHS | We're needing a key system functionality using asterisk |
| 18:08.06 | Mercestes | KW-CHS, killl your users instead. |
| 18:08.16 | ManxPower | KW-CHS: Asterisk is not a key system -- anything that tries to make it one is an ugly hack |
| 18:08.33 | Mercestes | KW-CHS, welcome them to the 21st century. |
| 18:09.32 | robl^ | KW-CHS: you will spend 20x more time trying to get Asterisk to work like a Key System than the 30 minutes it will take to train your users on how a PBX works |
| 18:10.04 | robl^ | (20x is a VERY VERY VERY VERY low estimate) |
| 18:11.43 | KW-CHS | I'm needing this for a staffing situation -- it needs to be able to park call and grab from lamps -- that's why I have the Snom 360's, for that. |
| 18:12.05 | ManxPower | KW-CHS: then I suggest you use something other than Asterisk |
| 18:12.29 | KW-CHS | ?? |
| 18:13.03 | b11d | it wasnt that hard to understand :) |
| 18:13.11 | robl^ | KW-CHS: thats not key. you can do that now with Asterisk. You can PARK by transfering to 700.. Then setup hints for park:701 through park:7XX .. set up the Snoms to BLF and yer done |
| 18:13.26 | ManxPower | b11d: the issue is the WILLINGNESS to accept it. |
| 18:13.52 | ManxPower | robl^: I don't know if you can hint parking slots without BRIStuff patches |
| 18:14.19 | ManxPower | or trunk |
| 18:14.19 | KW-CHS | Can I grab parked calls from any phone in office by hitting button that way? |
| 18:14.35 | robl^ | ManxPower: in 1.4 you can. I usue it extensively.. |
| 18:14.42 | Peaceful | Would a Polycom 550 be considered a step up from Cisco 796x's? |
| 18:14.52 | ManxPower | robl^: Ah, I thought was a trunk-only thing |
| 18:14.53 | KW-CHS | I'm on 1.4.5, so that should work |
| 18:15.13 | robl^ | 1.45 is buggy. Go with 1.4.11 |
| 18:15.14 | ManxPower | Peaceful: from a technology standpoint or an everything else standpoint? |
| 18:15.22 | Peaceful | ManxPower, both |
| 18:15.41 | KW-CHS | I'm using AsteriskNow right now . . |
| 18:15.44 | Peaceful | ManxPower, although I don't really use many features except for multiple lines and a headset |
| 18:15.54 | ManxPower | from a tech standpoint I would say "no", (they are pretty similar from a tech standpoint) from an everything else standpoint I would say "yes" |
| 18:16.07 | robl^ | ManxPower: in 1.4.x you can hint meetme as well |
| 18:16.14 | KW-CHS | It's four stations. I just want to park a call and show them on all phones and then use that to pick up on another. |
| 18:16.25 | Peaceful | so, even on a technology standpoint? So their about the same to use with asterisk? |
| 18:16.49 | Peaceful | "even on" -> "they're about the same on" |
| 18:17.05 | ManxPower | Peaceful: it is the licensing and lack of included power supply that makes Cisco unacceptable to many people |
| 18:17.57 | robl^ | Cisco also tends to be lacking in features for SIP, compared to Polycom.. Cisco's baby is SCCP/Skinny.. so SIP is just an afterthought |
| 18:18.04 | Peaceful | Cisco's 7961 doesn't seem to have a documented config and a buggy firmware, according to voip-info. That's why I'm wondering if I should go with Polycom instead. |
| 18:18.16 | b11d | yes you should |
| 18:18.18 | b11d | i love my polycoms |
| 18:18.21 | b11d | i hate the cisco phones |
| 18:18.31 | Peaceful | We have 40 7960's that we've been using for years (not 7960g's), and we finally ran out and I have to buy 20 more phones |
| 18:18.58 | Peaceful | b11d, are they easier to use with asterisk? tftp provisioning, firmware, etc? |
| 18:19.02 | robl^ | I had a bunch of Cisco 796xes.. Swapped them out for Snom... Then went to Polycom for my primay install |
| 18:20.32 | Peaceful | So polycom SIP is better? Hmm... and it looks like a Polycom 550 can be had for ~$230, where my Cisco rep is pushing 7961's for $292 |
| 18:20.32 | b11d | if i didnt hate the name snom so much, i might have tried them out |
| 18:20.34 | KW-CHS | BLF with hinting would I be able to park call to a user chosen button (SNOM 360) and allow other phone stations to pick up that same call using their button on their phone? |
| 18:20.56 | robl^ | KW-CHS: yes / no.. |
| 18:21.10 | KW-CHS | ok ?? |
| 18:21.12 | Peaceful | Is there a good reseller that would give me a Polycom unit to evaluate? |
| 18:21.23 | robl^ | KW-CHS: usually.. you park.. it goes to the next available button. 701, 702, etc... |
| 18:21.39 | KW-CHS | Ok, that's fine. |
| 18:21.55 | KW-CHS | Does it show up on all the phones then and they can all grab that line? |
| 18:21.58 | robl^ | KW-CHS: any phone that has a BLF key for those will be able to see a light if a call is parked on that number.. and can pick up |
| 18:22.21 | robl^ | KW-CHS: but only one phone can have that call at any time. |
| 18:22.35 | *** join/#asterisk CVirus (n=GoD@212.12.250.74) |
| 18:22.38 | KW-CHS | Problem solved then -- now how do I do that . .. ? |
| 18:22.53 | [TK]D-Fender | Peaceful: IP 550 = waste. |
| 18:23.24 | [TK]D-Fender | Peaceful : That and the IP 30X. |
| 18:23.32 | robl^ | KW-CHS: not sure with AsteriskNOW. I don't use it. I edit the configuration files with a text editor |
| 18:23.41 | [TK]D-Fender | IP430 = ALMOST a complete waste. |
| 18:24.05 | Peaceful | [TK]D-Fender, why do you say that? |
| 18:24.22 | Peaceful | what would be better than the 550? |
| 18:24.32 | KW-CHS | I can get in there and edit . . . already have to fix some rollover on outbound line issues that *now doesn't do well. |
| 18:24.35 | Peaceful | I'm looking for 7960 equivalents, remember... |
| 18:24.41 | [TK]D-Fender | Peaceful: What does it offer over the IP 320/330 and warrant costing the same as an IP 601 which supports more reg, line-keys and is EXPANDABLE? |
| 18:24.51 | robl^ | KW-CHS: ask in #asterisk now. Tell them you want to set up hint priorities for parking. Then just setup keys on your SNOM for BLF with 701, 702, etc. default Asterisk is 701-720 |
| 18:25.18 | [TK]D-Fender | Peaceful: What do you NEED in a phone? |
| 18:25.45 | Peaceful | headset |
| 18:25.55 | Peaceful | 2-4 lines |
| 18:26.14 | Peaceful | daisy-chainable |
| 18:26.16 | Peaceful | POE |
| 18:26.22 | Peaceful | works well with asterisk |
| 18:26.26 | Qwell[] | sounds like a polycom to me |
| 18:26.35 | *** join/#asterisk msetim (n=marcos@200.195.161.164) |
| 18:26.46 | Peaceful | well, the 320/330 doesn't seem to have a headset jack |
| 18:27.34 | bkw__ | http://www.internetisseriousbusiness.com/ |
| 18:27.37 | Peaceful | The 430 has a max of 2 lines, which isn't too big of a deal, I don't have many users using more than two lines |
| 18:27.55 | msetim | hi :) how can I send a ignore command ( to a call ) to asterisk? |
| 18:27.57 | riddlebox | have you guys seen this? http://www.rowetel.com/ucasterisk/ip04 |
| 18:28.10 | [TK]D-Fender | Peaceful: IP330 = $109 |
| 18:28.15 | [TK]D-Fender | Peaceful: There you go. |
| 18:28.29 | Peaceful | It doesn't have a headset jack, it's not even in the running. |
| 18:28.58 | [TK]D-Fender | Peaceful: ALL polycoms hav RJ9 headset jacks, and the 320/330 have a 2.5mm jack in ADDITION |
| 18:29.38 | Peaceful | Really. Show me. http://www.polycom.com/usa/en/images/products/voice/desktop/soundpoint_ip/alt_soundpoint_ip330_320.jpg |
| 18:30.09 | Strom_C | [TK]D-Fender: WRONG |
| 18:30.19 | Strom_C | no RJ9 headset jack on the 320 |
| 18:30.24 | [TK]D-Fender | Peaceful: 2.55 is on the SIDE. |
| 18:30.25 | Strom_C | sorry :) |
| 18:30.41 | Strom_C | the only RJ9 jack is for the handset |
| 18:31.13 | Peaceful | I need RJ9 headset jacks. |
| 18:31.22 | Strom_C | Peaceful: check out the IP430 |
| 18:31.31 | [TK]D-Fender | Peaceful: IP 430 then = $145 |
| 18:32.26 | [TK]D-Fender | Didn't know the 320/330 dropped the RJ9.... |
| 18:32.29 | [TK]D-Fender | darnit :) |
| 18:33.05 | [TK]D-Fender | Guess they realized the % of cheap consumers using then :) |
| 18:33.27 | [TK]D-Fender | Ok, IP430 not has a new reason to live! |
| 18:33.30 | [TK]D-Fender | now* |
| 18:33.57 | Peaceful | The 430 is definitely an option. The problem is that it seems like a step down from the 7960's with its smaller screen. |
| 18:34.29 | Braxus | 320/330 also doesn't come with the power/ethernet cable if I recall. |
| 18:34.49 | Qwell[] | ethernet cables are 10 cents |
| 18:35.10 | Strom_C | Braxus: the 320 includes an ethernet cable |
| 18:35.13 | Braxus | the polycom phones use a semi-PoE injected setup no? |
| 18:35.27 | mcab | Braxus: only the 50x & 30x |
| 18:35.31 | Braxus | ah |
| 18:35.32 | Strom_C | Braxus: the new polycom phones use straight 802.3af |
| 18:37.19 | msetim | I can't send a hangup to a call, i would like to send a ignore, it's possible? |
| 18:37.27 | [TK]D-Fender | Braxus: And they always came with an ethernet cable ;) |
| 18:37.51 | msetim | i won't * :) |
| 18:38.01 | Braxus | http://www.voiplink.com/Polycom_24v_Power_Supply_p/polycom-24v-power-supply.htm < this is what I was talking about which is missing from the 320/330s. The phone's ethernet cable then has a place to inject that power near the wall. |
| 18:38.19 | [TK]D-Fender | Peaceful: How much time do you spend staring at your phone? Also, how many phones are you planning? |
| 18:38.33 | [TK]D-Fender | Braxus: No, it doesn't |
| 18:38.47 | [TK]D-Fender | Braxus: they don't use the 501 style injector cable |
| 18:39.02 | Braxus | plug strait into the phone? |
| 18:39.03 | [TK]D-Fender | Braxus: ONLY the IP 50X used that in-line cable for wall-warts |
| 18:39.08 | [TK]D-Fender | Braxus: correct |
| 18:39.28 | [TK]D-Fender | Braxus: it was kinda hideous for the IP 501 really... |
| 18:40.04 | [TK]D-Fender | Braxus: but if you broke the safety pin off you could have the brick near your patch panel and the phone would APPEAR to be PoE to the suer ;) |
| 18:40.09 | [TK]D-Fender | user* |
| 18:40.44 | Braxus | yeah but why not just go true PoE in that case. :3 |
| 18:40.46 | Braxus | hee hee |
| 18:41.04 | Peaceful | [TK]D-Fender, I don't spend too much of my personal time on the phone, but I've got over 50 employees, and some of them spend a TON of time on the phone |
| 18:41.13 | Peaceful | and 20 more phones |
| 18:41.27 | Braxus | was considering the 320/330s at one point to replace the 501s on cubicle installations but hear the speakerphone quality on the 320/330s suck? |
| 18:41.29 | Peaceful | I've got either 40 or 50 7960's currently, can't remember which |
| 18:41.44 | [TK]D-Fender | Peaceful: ok, well you don't normally sit around staring at it, just TALKING on it. |
| 18:41.48 | *** join/#asterisk truescot (n=jaja@62.212.129.230) |
| 18:41.54 | truescot | does anyone have any experience using asterisk desktop manager? i am having problem getting the browser popup to work, the rest all works fine like slide in popup and calling etc just brower doesnt |
| 18:43.00 | Peaceful | [TK]D-Fender, true, but I've got not only the beauty of the phone to consider, but also office politics :) Getting a phone with a smaller screen for some people may cost more than the price difference in the phones :-P |
| 18:43.48 | [TK]D-Fender | Peaceful: This is a harder thing to valuate I guess. |
| 18:44.09 | jfitzgibbon | question for anyone familiar with Zaptel PRI. After a reboot, my TE412P refuses to sync with the telco on one of the four ports. If I do a pri debug, I see my side sending SABME (with C/R set to 0), but I see nothing coming from the Telco side. Any ideas? |
| 18:44.09 | Peaceful | very true |
| 18:44.12 | [TK]D-Fender | Peaceful: Because the IP 430 can support at least 5 calls per line-key with 2 regs. |
| 18:44.43 | [TK]D-Fender | Peaceful: But it you want aestetics, sure the IP 550 wins, but the 601 is about the same price and is expandable and has more line-key buttons. |
| 18:45.14 | Qwell[] | what's the 550? |
| 18:45.18 | [TK]D-Fender | jfitzgibbon: I think you should pastebin your configs as well as "pri show span [spannumber]" |
| 18:45.19 | [TK]D-Fender | ~pb |
| 18:45.20 | 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 |
| 18:45.22 | Qwell[] | oh, right |
| 18:45.31 | Qwell[] | umm, yeah, 550 > 601 |
| 18:45.51 | [TK]D-Fender | Qwell : IP 550 = IP 650 with 4 line keys (2 jsut MISSING), and no expansion |
| 18:45.53 | Peaceful | enter politics again. My Boss likes Cisco, but he is also price concious. Cisco is telling me either $240 for a 7960g, or $290 for a 7961g. |
| 18:46.08 | [TK]D-Fender | Qwell[]: not IMO. 601 = more regs, and expandable |
| 18:46.27 | rudholm | the 550 supports G.722 |
| 18:46.30 | Peaceful | So if a Polycom 550 is comparable to a 7961g, and I can get it for ~ $240 -> Win/Win |
| 18:46.32 | rudholm | which the 601 does not |
| 18:46.35 | [TK]D-Fender | Peaceful: 7960G = Cisco PoE, and seriously, who needs that many lines? |
| 18:46.38 | puzzled | Peaceful: just tell him that Cisco licensed the stuff from Polycom and that Polycom works better with Asterisk than Cisco who only wnat to sell CallManager |
| 18:46.41 | Qwell[] | so just get a 650 :p |
| 18:46.47 | Qwell[] | and you'll *STILL* be paying less than a cisco |
| 18:47.02 | [TK]D-Fender | Qwell[]: Yes, I WOULD sooner pay a few more bucks for the 650 rather than consider the 550 :) |
| 18:47.11 | Qwell[] | how much more? |
| 18:47.14 | rudholm | I selected a 550 over a 650 because I didn't need all the extra buttons and I think the 650 is physically bigger (and more expensive) |
| 18:47.24 | Qwell[] | [TK]D-Fender: seriously though - you need to start getting commissions from polycom |
| 18:47.37 | [TK]D-Fender | Peaceful: Biig thing is that Cisco's SIP sucks <- Polycoms call-handling is unmatched in my experience. |
| 18:47.44 | [TK]D-Fender | Qwell[I should ;) |
| 18:48.09 | Peaceful | Haven't looked at a 650, cheapest appears to be ~ $272, I'll go pull the datasheet... |
| 18:48.25 | *** join/#asterisk apardo (n=apardo@bonemachine.ipv6.theprimusproject.com) |
| 18:48.34 | Mercestes | comissions?? |
| 18:48.35 | rudholm | I paid 230$ for my 550 (telephonydepot.com) |
| 18:48.36 | jfitzgibbon | TK - cli output plus zaptel/zapata configs are at http://pastebin.com/d3731adf5 |
| 18:48.36 | Peaceful | That'll be a hard sell -- boss man expected closer to $200 |
| 18:48.45 | Mercestes | I thought [TK]D-Fender was the president of Polycom. =/ |
| 18:48.48 | Peaceful | ...which is why I'm able to break from Cisco |
| 18:48.48 | [TK]D-Fender | Qwell[]: 550 = $232, 650 = $276. I'd probably do it. Save s the resale value. |
| 18:49.18 | yakkop | anyone know of any voip providers that terminate to toll-free numbers for free? |
| 18:49.19 | Qwell[] | is there a 350 yet? |
| 18:49.22 | [TK]D-Fender | Peaceful: IP 430 still seems a good choice. |
| 18:49.47 | [TK]D-Fender | Qwell[]: I can't picture a place for it... personally the 550 is overpriced. they should have done a completely new based for the 550.... |
| 18:50.00 | [TK]D-Fender | Qwell[]: it was a poorly placed solution. |
| 18:50.35 | [TK]D-Fender | Qwell[]: as it is the 320/330 should have been *1* phone to kill the IP 301 / 430. |
| 18:51.29 | [TK]D-Fender | Qwell[]: 1 low, 1 medium, 1 receptionist/high all with dual powered, dual headset, backlight. |
| 18:51.55 | *** join/#asterisk gardo (n=gardo@203.82.42.106) |
| 18:52.16 | Peaceful | I don't see the benefit of a 650 over a 550. I don't need more than 4 lines, and so all I get is two extra buttons? |
| 18:52.31 | rudholm | Peaceful: that was my thinking, and is why I selected the 550 |
| 18:52.58 | Peaceful | rudholm, how do you like it? |
| 18:53.22 | rudholm | Peaceful: I love it |
| 18:53.48 | CCFL_Man2 | 03:44 < SwK> didnt I tell you to just buy a as5300 and be done with it? heh |
| 18:53.53 | CCFL_Man2 | yes you did |
| 18:54.01 | SwK | :P |
| 18:54.04 | CCFL_Man2 | as5300 cosats a ton though |
| 18:54.05 | rudholm | Peaceful: works flawlessly, supports PoE, love the ringtones, sounds great on regular G.711, and sounds amazing with G.722 |
| 18:54.07 | [TK]D-Fender | Peaceful: are you sure you need 4 LINES, and not the ability to handle 4 CALLS? |
| 18:54.13 | SwK | as5300 not really considering what you get |
| 18:54.30 | SwK | T38, HWEC< SIP/H323, plus other things |
| 18:54.39 | CCFL_Man2 | SwK: oh i know, but for a fry cook budget like i have :P |
| 18:54.56 | SwK | CCFL_Man2, maybe you should hit the track to make some extra cash then |
| 18:55.05 | Peaceful | G.722, eh? That sounds interesting. Do you only notice the difference on in-office calls (all G.722?) |
| 18:55.19 | rudholm | Peaceful: the 550 is in my house (in my office room) so the quality of the ringtone counts a lot (can't have loud/annoying ringing) |
| 18:55.28 | [TK]D-Fender | Peaceful: * has TRANSCODE issues.... you may want to reconsider that ;) |
| 18:55.32 | rudholm | Peaceful: yeah, you have to be talking to another G.722 device |
| 18:55.58 | [TK]D-Fender | rudholm: Since you can use any old recording I do jsut fine :) I run the Cisco "24" warble as it is |
| 18:55.59 | rudholm | Peaceful: but my employer is planning to move to Asterisk, so calls to people at work will likely support G.722. |
| 18:56.17 | CCFL_Man2 | SwK: i'm not allowed to gamble on anything my company does, according to my company policy, and gambling on slots is pretty much throwing your money away |
| 18:56.30 | CCFL_Man2 | i'm waiting for tables |
| 18:56.45 | SwK | CCFL_Man2, I said the track not the casino |
| 18:56.50 | SwK | the track as in the ho track |
| 18:56.54 | SwK | not the race track |
| 18:57.09 | rudholm | the Trunk version of asterisk supports G.722 transcoding. I back-ported it to SVN 1.4 and it works without problems. |
| 18:57.15 | Peaceful | [TK]D-Fender, transcoding issues? As in, you have to have the CPU to handle it? |
| 18:57.16 | Qwell[] | rudholm: Do you guys use asterisk at work? |
| 18:57.30 | CCFL_Man2 | SwK: i know, i'm not allowed to :P |
| 18:57.44 | CCFL_Man2 | i can sell myself on the street :P |
| 18:57.45 | rudholm | Qwell[]: we're looking into it. nothing official yet. |
| 18:57.47 | *** join/#asterisk Sci_05 (i=Sci_05@ts.bestserversllc.net) |
| 18:57.49 | Qwell[] | ahh |
| 18:57.58 | Qwell[] | You still work for Big Company, right? |
| 18:58.02 | Qwell[] | or |
| 18:58.03 | Qwell[] | Big Company! |
| 18:58.09 | rudholm | Qwell[]: yes, Big Company! |
| 18:58.12 | Qwell[] | heh |
| 18:58.18 | SwK | CCFL_Man2, at $5 a pop you should be able to pay for a AS5300 in the short term |
| 18:58.24 | Qwell[] | or would it be Big! Company |
| 18:58.31 | CCFL_Man2 | i spent all my money on my collectable WECO phones :P |
| 18:58.32 | rudholm | Big! Inc. |
| 18:58.33 | rudholm | :) |
| 18:58.36 | Qwell[] | there you go |
| 18:58.38 | *** join/#asterisk kwame (n=kwame@209.213.194.7) |
| 18:58.58 | kwame | Hi, I just finished installing and configuring an asterisk 1.4 |
| 18:59.17 | Qwell[] | rudholm: well, you need to hurry up, so you can issue a nice press release saying how much money you saved, etc |
| 18:59.21 | kwame | When I configure my ekiga clients that are on the same network (192.168.98.x) the connection and communication works great |
| 18:59.24 | Qwell[] | media eats that up |
| 18:59.40 | kwame | but when I try to connect an ekiga client that it's on a vpn that connects to my 192.168.98.x network registration fails |
| 19:00.11 | kwame | there is a route from the vpn network to my 192.168.98.x network, and I don't see any file that I should modify in my asterisk /etc/asterisk directory |
| 19:00.14 | kwame | any help on this? |
| 19:00.29 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584025.dsl.bell.ca) |
| 19:00.31 | CCFL_Man2 | SwK: cheapest as5300 is $650 on ebay, but doesn't have all the needed voip stuff i don't think |
| 19:00.37 | rudholm | Qwell[]: yeah, I know. I've already agreed to notify certain interested entities when it happens. presumably there will be joint press releases. |
| 19:00.39 | sty|bong | kwame: add nat=yes tp that users extension |
| 19:00.44 | Qwell[] | neat |
| 19:00.52 | Qwell[] | is it actually bound to happen? |
| 19:00.55 | CCFL_Man2 | though atleast it has 100mbit ethernet :P |
| 19:01.05 | [TK]D-Fender | kwame: You need to do "make samples" in your source folder to install the samples from which you can begin to develop your setup |
| 19:01.08 | rudholm | Qwell[]: unless something goes really badly. |
| 19:01.13 | SwK | CCFL_Man2, you need one for VoIP for 650 you're gettign the dailup box |
| 19:01.15 | Qwell[] | sweet |
| 19:01.16 | rudholm | Qwell[]: the idea is to start with a single office |
| 19:01.17 | SwK | and that will not work |
| 19:01.23 | [TK]D-Fender | sty|bong: he has no CONFIG FILES <-------- |
| 19:01.32 | msetim | how can I send a busy signal to a channel that is ringing? |
| 19:01.57 | kwame | [TK]D-Fender: was that for me? |
| 19:02.04 | CCFL_Man2 | SwK: oh, cheaper ones are for dialup modem pools? |
| 19:03.28 | [TK]D-Fender | kwame: Yes |
| 19:03.48 | *** join/#asterisk Delvar (n=Delvar@77.240.56.18) |
| 19:04.09 | SwK | CCFL_Man2, yes |
| 19:04.19 | kwame | :? what does that have to do with the ekiga client? |
| 19:05.19 | msetim | somebody knows? |
| 19:06.05 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 19:09.32 | *** join/#asterisk poehali (n=actionma@74.93.5.186) |
| 19:09.53 | poehali | anyone know where I can buy a sipura 3000? not the linksys type |
| 19:10.01 | poehali | if not which device should I buy? |
| 19:10.18 | poehali | I need a box with 1 incoming jack and 1 outgoing jack |
| 19:10.32 | [TK]D-Fender | kwame: Nothing. It has to do with * being set up. |
| 19:10.52 | [TK]D-Fender | kwame: You tell me you have no configs then what the hell is Ekiga supposed to connect TO? |
| 19:11.24 | kwame | [TK]D-Fender: ok, here is my scenario again |
| 19:11.25 | [TK]D-Fender | poehali: You should go for the Linksys SPA-3102. Sipura was bought out YEARS ago. |
| 19:11.43 | kwame | [TK]D-Fender: the asterisk is setup and working with my ekiga clients connecting from the same network |
| 19:12.08 | poehali | [TK]D-Fender: is that the best box for asterisk? |
| 19:12.21 | kwame | [TK]D-Fender: when I configure one ekiga client on a computer that is connected through a vpn the registration fails |
| 19:12.25 | [TK]D-Fender | poehali: For exactly that need, yes |
| 19:13.08 | poehali | [TK]D-Fender: is there a box that has enough jacks for 20 or more incoming and equal outgoing? |
| 19:13.10 | [TK]D-Fender | kwame: Well it could be a VPN issue, it could be a SIP/IP issue. pastebin the [general] section of your sip.conf |
| 19:13.45 | [TK]D-Fender | poehali: that is a LOT.. you should sreiously consider getting a PRI |
| 19:13.53 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 19:14.05 | [TK]D-Fender | poehali: and why would you want so many PHONES running analog? |
| 19:14.07 | sty|bong | or just try nat=yes |
| 19:14.12 | kwame | sty|bong: I modified the nat=yes in the stip.conf, restarted asterisk and I got the same thing |
| 19:14.27 | sty|bong | cool |
| 19:14.28 | [TK]D-Fender | sty|bong: No, thats LAN, not INTERNET. |
| 19:14.30 | poehali | [TK]D-Fender: primary rate interface? |
| 19:14.34 | [TK]D-Fender | poehali: Yes |
| 19:14.52 | sty|bong | right, so its a stun issue then ? |
| 19:15.22 | [TK]D-Fender | sty|bong: NO |
| 19:15.42 | poehali | [TK]D-Fender: I'm trying to setup a PBX that handles voicemails and hunting list with the existing 20 at&t accounts |
| 19:16.08 | kwame | [TK]D-Fender: here is the general part of the sip.conf |
| 19:16.09 | kwame | http://pastebin.com/m376bf5f5 |
| 19:16.11 | sty|bong | mm. most likely firewall eh |
| 19:16.16 | poehali | [TK]D-Fender: and I can't use voip because in the past sales department kept losing customers due to bad reception |
| 19:16.23 | [TK]D-Fender | sty|bong: you are guessing blind. |
| 19:16.33 | sty|bong | just chatting |
| 19:16.55 | [TK]D-Fender | poehali: what are the 20 FXS for though? |
| 19:17.07 | *** join/#asterisk bisybackson (n=de@196.23.50.30) |
| 19:17.14 | kwame | [TK]D-Fender: what do you see in that sip.conf ? something missing? |
| 19:17.59 | [TK]D-Fender | kwame: check your routes on that bok, and set "localnet=192.168.xx.0/24" for EACH of your corporate subnets. |
| 19:18.05 | [TK]D-Fender | box |
| 19:18.38 | poehali | [TK]D-Fender: for sales |
| 19:19.15 | [TK]D-Fender | poehali: Why analog PHONES within that building though? |
| 19:19.32 | poehali | why not? |
| 19:19.37 | marc7 | I'm trying to setup a voicemail access extension (let's say 700) where people can dial from their IP phones to check their voicemail without authenticating.... but let's say person X has a two-line SIP phone... how would I build logic to say if they're from *this* sip peer, then redirect them to this voicemailbox? |
| 19:20.23 | [TK]D-Fender | marc7: hint, "CALLER ID" <- |
| 19:20.27 | Mercestes | marc7: You can't do that without authenticating unless you don't use a password. |
| 19:20.43 | [TK]D-Fender | Mercestes: ummm.. nope :p |
| 19:20.47 | bisybackson | hello - I am unable to disable call waiting by *71, or "database put CW XXXX DISABLE". Running 1.2.22, Snom300 and 320 phones. Any other options? |
| 19:20.49 | Mercestes | How do you skip the password then? |
| 19:20.52 | Nugget | of course you can do that. |
| 19:20.59 | marc7 | Mercestes: VoiceMailMain(s****) |
| 19:21.10 | Mercestes | That asks for a password doesn' tit? |
| 19:21.21 | [TK]D-Fender | bisybackson: this isn't *'s job, its the PHONES. |
| 19:21.35 | marc7 | Mercestes: where the asteriskes are the voicemailbox... the "s" in front of it doesn't ask for a password |
| 19:21.47 | Mercestes | oh...the "S" |
| 19:21.50 | Mercestes | nice.... |
| 19:21.54 | [TK]D-Fender | s - Skip checking the passcode for the mailbox. |
| 19:22.09 | marc7 | [TK]D-Fender: i thought about that... but basically there are two people with let's say a "company line" ... so the caller ID from Line 1 will be the same for multiple people.... but the caller ID from line 2 on each phone will be different... that's why i'm trying to assign a variable denoting the SIP peer |
| 19:22.25 | Nugget | marc: you can set the voicemailbox as a variable in sip.conf instead of using, say, callerid(number) for passing to voicemailmain() |
| 19:22.25 | kwame | [TK]D-Fender: set that localhost=xxx in where? |
| 19:22.29 | *** join/#asterisk drwelby (n=mpfister@mail.enplan.com) |
| 19:22.31 | bisybackson | [TK]D-Fender: thanks, but I've given up on the Snoms... |
| 19:22.43 | [TK]D-Fender | marc7: Stop thinking about your OUTSIDE caller id and think about the PHONE's <- |
| 19:22.51 | *** part/#asterisk msetim (n=marcos@200.195.161.164) |
| 19:23.00 | Nugget | in sip.conf use setvar=VMBOXNUM=200 |
| 19:23.02 | marc7 | Nugget: the trick is that I'm using 7960's which have a "Messages URI"... or a number that's dialled to check voicemail |
| 19:23.05 | [TK]D-Fender | bisybackson: you could try "call-limit=1" in sip.conf |
| 19:23.12 | Nugget | or something like that, then call voicemailmain(s${VMBOXNUM}) |
| 19:23.24 | marc7 | oh... oh nice! that's exactly what i'm looking for |
| 19:23.44 | [TK]D-Fender | marc7: Though I like using CALLERID, nugget's way is great too. |
| 19:23.48 | marc7 | Fender: the phones can only have one Messages URI... so I want 700 to work for both... that's pim |
| 19:23.58 | marc7 | er, that's pimp Nugget, thanks |
| 19:24.14 | bisybackson | [TK]D-Fender: thank you, will try. |
| 19:24.17 | [TK]D-Fender | marc7: what phones? |
| 19:24.22 | marc7 | 7960's |
| 19:24.29 | [TK]D-Fender | marc7: Poor you :| |
| 19:25.20 | marc7 | [TK]D-Fender: not the worst thing in the world... any suggestions on new hardware-based SIP phones? we're getting a couple of Linksys SPA-962's... any other suggestions? |
| 19:25.24 | *** join/#asterisk zcionn_ (n=a@58.69.243.203) |
| 19:25.38 | [TK]D-Fender | marc7: Stop the SPA's before they arrive and jsut go Polycom :) |
| 19:25.53 | [TK]D-Fender | Polycom = phones of great happiness |
| 19:27.18 | marc7 | hmm... any specific model, or are they just one of those vendors that "gets it" and all their gear is awesome |
| 19:27.30 | *** join/#asterisk southtel (n=southtel@68-114-23-151.dhcp.gwnt.ga.charter.com) |
| 19:28.17 | marc7 | and does anybody else want to weigh in on the subject? |
| 19:28.23 | southtel | Does anyone know if analog phone lines normally send the dialed DID number? |
| 19:28.32 | Strom_C | southtel: no, they generally don't |
| 19:28.35 | sty|bong | gee those polycom's are explensive |
| 19:28.36 | [TK]D-Fender | marc7: What do you need? |
| 19:28.49 | southtel | Strom_C: Thanks, that's what I thought. |
| 19:29.00 | [TK]D-Fender | marc7: Polycom quality is rock solid, they feature set rather strong. |
| 19:29.11 | Braxus | I have a couple Linksys SPA-941s. The polycom 501s I have here blow them away in virtually all aspects. |
| 19:30.12 | [TK]D-Fender | Braxus: I own a 501, 430, 301, I use 301, 501, 600, 601 here at the office and have owned an SPA-941. 94X = only if you're outside NA & cheap |
| 19:31.06 | Braxus | most cases we were happy with the 941s. The main gripe was the speakerphone quality and not having the ability to auto-provision directories easily. |
| 19:31.08 | sty|bong | what polycom would be comparable to a spa 492 ? |
| 19:31.30 | kwame | [TK]D-Fender: is that localnet=192.168.x.x/24 a configuration that should go in an asterisk config file? |
| 19:32.03 | Braxus | hmmm. anyone able to recommend a couple of RJ-9 headsets? |
| 19:32.08 | bisybackson | [TK]D-Fender: that appears to work, but no engaged tone on second call. Do I need something like "exten => _XXXX,n,Congestion" for each extension? |
| 19:32.58 | CCFL_Man2 | when configuring a voice group in cas mode, what do i use for "timeslot-list" |
| 19:34.52 | *** part/#asterisk riddlebox (n=JamesMid@75-128-170-26.static.stls.mo.charter.com) |
| 19:34.56 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@216-110-94-230.static.twtelecom.net) |
| 19:35.19 | CCFL_Man2 | and then the type, ground start and loop start fxs ans fxo, but what if that channel will be used for say ISDN U? |
| 19:35.22 | sty|bong | kwame: slap it in sip.conf |
| 19:35.40 | viKing78 | [TK]D-Fender: I just bought 10 of the Polycom 330s do you have anything to say good or bad about one Polycom model or another? |
| 19:37.00 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:37.08 | *** join/#asterisk Voltage3 (n=someone@www.bishopfixtures.com) |
| 19:37.08 | kwame | sty|bong: so should I place in there both networks? my 192.168.98.x and 10.1.6.x ? |
| 19:37.39 | *** join/#asterisk guillote_GNU (n=bancaria@host185.190-136-203.telecom.net.ar) |
| 19:38.36 | sty|bong | kwame: only the local network |
| 19:39.13 | mvanbaak | anyone here looked at linuxmce ? |
| 19:39.38 | mvanbaak | following the thread on -users mailinglist, I think linuxmce already has some nice home automation with asterisk |
| 19:39.40 | *** join/#asterisk heelios (n=heelios@onyx.6pixies.com) |
| 19:40.44 | [TK]D-Fender | viKing78: Nothing bad to say for them so much as advising specific models for specific scenarios to maximise your purchase value |
| 19:40.56 | sty|bong | kwame: you may also want to set externip=<your-ext-ip> . |
| 19:41.21 | Voltage3 | Hello everyone, I was wondering if someone could help me with a TDM PCI master abort (on a TDM400P) , and if its related to an IRQ problem. zttest does not produce output, i'm assuming because of that. |
| 19:42.31 | [TK]D-Fender | Voltage3: pastebin "cat /proc/interrupts" and what version are you running? |
| 19:42.33 | [TK]D-Fender | ~pb |
| 19:42.33 | 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 |
| 19:42.35 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 19:42.54 | [TK]D-Fender | kwame: You should have 1 line per subnet with the appropriate mask. |
| 19:43.48 | Voltage3 | zaptel 1.2.20 |
| 19:43.50 | Voltage3 | zaptel-1.2.20.1 |
| 19:44.08 | Voltage3 | <PROTECTED> |
| 19:44.08 | Voltage3 | <PROTECTED> |
| 19:44.08 | Voltage3 | <PROTECTED> |
| 19:44.08 | Voltage3 | <PROTECTED> |
| 19:44.08 | Voltage3 | <PROTECTED> |
| 19:44.09 | Voltage3 | <PROTECTED> |
| 19:44.11 | Voltage3 | <PROTECTED> |
| 19:44.13 | Voltage3 | <PROTECTED> |
| 19:44.15 | Voltage3 | <PROTECTED> |
| 19:44.17 | Voltage3 | <PROTECTED> |
| 19:44.19 | Voltage3 | <PROTECTED> |
| 19:44.21 | Voltage3 | <PROTECTED> |
| 19:44.23 | Voltage3 | <PROTECTED> |
| 19:44.25 | Voltage3 | NMI: 0 0 |
| 19:44.27 | Voltage3 | LOC: 415799 415798 |
| 19:44.29 | Voltage3 | ERR: 0 |
| 19:44.31 | Voltage3 | MIS: 0 |
| 19:44.48 | marc7 | so much for pastebin |
| 19:45.05 | Mercestes | lol |
| 19:45.14 | Deeewayne | O.O |
| 19:45.19 | sty|bong | its the IRC version |
| 19:45.48 | Voltage3 | sorry i haven't used irc in a while |
| 19:46.39 | *** join/#asterisk Fulk (n=test@87-194-176-39.bethere.co.uk) |
| 19:47.17 | yakkop | anyone know of any voip providers that terminate to toll-free numbers for free? |
| 19:47.57 | mog | fwd |
| 19:47.59 | sty|bong | i use freeworlddialup for that |
| 19:48.11 | sty|bong | yea FWD |
| 19:48.51 | yakkop | how about for a business? |
| 19:49.02 | mog | ? |
| 19:49.31 | sty|bong | just set it up as a seperate trunk |
| 19:49.33 | yakkop | I thought FWD was primarily for indidual consumers.. |
| 19:49.42 | sty|bong | oh |
| 19:49.44 | tzafrir_laptop | Voltage3, so the card does not send any interrupts, right |
| 19:51.30 | bisybackson | exten => _1000,1,Dial(SIP/poohbear) |
| 19:51.31 | bisybackson | exten => _1000,2,Congestion |
| 19:51.41 | Voltage3 | right, and unfortunatly, my bios doesn't let me set irqs |
| 19:52.12 | bisybackson | is that the right way of doing it? I wish to have an engaged tone when the second call comes in to extension 1000 |
| 19:52.36 | CoffeeIV_ | I have an asterisk-iaxmodem-hylafax setup. I am migrating my old RxFax/TxFax to iaxmodem/hylafax. In my old setup, a script uploaded the fax to a CRM, and that script had access to asterisk's $UNIQUEID variable; I need to either get $UNIQUEID into Hylfax's FaxDispatch script, or get the filename of the received fax up into the asterisk dialplan after the fax is received |
| 19:53.33 | yakkop | sty|bong: are you using FWD to terminatte calls for a biz? |
| 19:53.57 | sty|bong | no |
| 19:55.53 | *** join/#asterisk rickross (n=rickross@supporter/active/rickross) |
| 19:57.14 | CoffeeIV_ | In an asterisk-iaxmodem-hylafax setup, is there a way, after the Dial(IAX2/iaxmodem/) part, when hylafax hangs up, to get control back int he dialplan, and have the filename of the received fax available to you ? |
| 19:58.04 | Mercestes | CoffeeIV_, exten => h,1, and some System() magic. |
| 19:58.11 | marc7 | hey is there any way to have Record() create a directory to record in if one doesn't exist? |
| 19:58.11 | *** join/#asterisk Delvar (n=Delvar@77.240.56.18) |
| 19:58.42 | marc7 | like if the full path name provided doesn't exist, it could then happen via an mkdir |
| 19:59.02 | Mercestes | marc7, a nice shell script with a test -r |
| 19:59.03 | mvanbaak | marc7: you can use an agi script for that |
| 19:59.11 | CoffeeIV_ | MErcestes: it's the System() magic I need . .. if multiple faxes are received at once, It won't simply be the newest tif file in the spool directory |
| 19:59.22 | marc7 | okay, thanks |
| 19:59.39 | Mercestes | CoffeeIV_, The naming conventing is entirely seperated from asterisk. Could use a "key" file since the faxes are named incrementally. |
| 19:59.49 | mvanbaak | CoffeeIV_: cant you put the fax in the CRM using hylafax ? |
| 19:59.50 | Mercestes | CoffeeIV_, after you check the file, just increment yoru seq file. |
| 20:00.16 | Mercestes | I hate doing it that way though because it is an indirect method, I prefer 1:1 ratios. |
| 20:00.29 | mvanbaak | so instead of running the CRM script in asterisk use the CRM script in hylafax |
| 20:00.38 | bisybackson | yay, BUSY works. Thanks all. |
| 20:00.41 | CoffeeIV_ | I can put hte fax in the CRM using Hylafax, but the CRM is setup to have everything keyd to an asterisk $UNIGUEID, so it can find the callerid and etc from a CDR record |
| 20:00.59 | CoffeeIV_ | so I do it from Hylafax, I need to get the uniqueid into Hylafax |
| 20:01.19 | mvanbaak | hylafax does not know about callerid ? |
| 20:01.35 | CoffeeIV_ | either way, I need the uniqueid and the filename in hte same place at the same time, whether in asterisk or hylafax |
| 20:02.19 | mvanbaak | if hylafax can handle cidnames you can do: Set(CALLERID(name)=${UNIQUEID}) right before you send it to hylafax |
| 20:02.20 | CoffeeIV_ | hylafax knows about callerid, but you can get multiple faxes from one callerid . . . the CRM keeps everything together by tracking the uniqueid of that call |
| 20:02.37 | CoffeeIV_ | mvanbaak: I'm going to try that |
| 20:02.39 | CoffeeIV_ | thanks |
| 20:02.44 | mvanbaak | ur welcome |
| 20:04.08 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 20:07.08 | *** join/#asterisk Ebola (n=Ebola@host86-138-6-48.range86-138.btcentralplus.com) |
| 20:08.31 | lirakis | any one here in NYC ? .. im trying to find an asterisk users group locally and havent hit much] |
| 20:09.16 | *** join/#asterisk Olobola (n=sfsdsdfs@74.95.13.57) |
| 20:12.02 | *** join/#asterisk Tako-san (n=Tako-san@24.108.162.254) |
| 20:13.17 | Olobola | so if I'm able to dial voicemail from xten, DTMF must be working... correct? When I punch in my userID/pass nothing is sent: "Incorrect password '' for user..." |
| 20:13.31 | *** part/#asterisk Voltage3 (n=someone@www.bishopfixtures.com) |
| 20:16.51 | *** join/#asterisk Tako-san (n=Tako-san@24.108.162.254) |
| 20:21.35 | *** join/#asterisk brannfenix (i=brannfen@ip68-230-133-70.ri.ri.cox.net) |
| 20:23.54 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@216-110-94-230.static.twtelecom.net) |
| 20:26.14 | NOT_guru | lirakis: best I can do is these guys http://tech.groups.yahoo.com/group/ny-aug/ |
| 20:26.32 | lirakis | NOT_guru: yeah .. already joined that |
| 20:30.26 | [TK]D-Fender | heading home, later all |
| 20:33.33 | *** join/#asterisk Galeras (n=Galeras@201.245.103.169) |
| 20:34.54 | NOT_guru | right on |
| 20:35.00 | NOT_guru | just wanted to throw it out there |
| 20:35.44 | yakkop | anyone know what's wrong with this: GET VARIABLE ENUMLOOKUP(18883336651,sip,,1,freenum.org) |
| 20:36.15 | yakkop | from an AGI... returns 200 result=1 () |
| 20:37.46 | lirakis | later |
| 20:37.49 | *** part/#asterisk lirakis (n=etamme@65.200.191.253) |
| 20:40.11 | Galeras | Could be some issue sending 30 simultaneous faxes using txfax? |
| 20:40.31 | Galeras | over a E1-PRI |
| 20:42.08 | Olobola | voicemail question: asterisk is recieving correct DTMF tones, but will not authenticate. voicemail.conf = 1234 => 4242.. I tried reloading and stopping. any suggestions? |
| 20:45.28 | ^majik^ | Olobola, make sure your handset/etc is on the same (outbound) context as specified in your voicemail config |
| 20:46.04 | ^majik^ | Olobola, for zap channels, check zapata.conf |
| 20:48.22 | Galeras | Or better i may ask: Is possible to send 30 simultaneous faxes over a E1? |
| 20:49.44 | *** join/#asterisk markit (n=konversa@88-149-177-66.static.ngi.it) |
| 20:51.25 | Olobola | ^majik^: ok, thanks |
| 20:51.50 | yakkop | anyone know who operates voipmich.com? |
| 20:52.10 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:52.41 | carrar | yakkop, Amanda Robinson? |
| 20:54.59 | yakkop | One person? or is there company? Just trying to understand what it is and if I can rely on it... |
| 20:56.40 | carrar | try calling them and asking |
| 20:57.34 | carrar | 248.724.0600 |
| 20:59.33 | carrar | yakkop, seeing how their web page doesn't work, no company Email address. I might suscpect you can't |
| 21:01.05 | Olobola | ^majik^: that was it, wrong context. Thank you very much. |
| 21:01.46 | carrar | yakkop, Amanda Robinson 248-393-3297 |
| 21:01.58 | carrar | looks like another phone |
| 21:05.42 | Galeras | How may simultaneous faxes can i send with spandsp? |
| 21:08.15 | *** join/#asterisk engrxyz (n=fgsfgfs@82-34-18-23.cable.ubr03.basi.blueyonder.co.uk) |
| 21:08.33 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 21:15.13 | kwame | sty|bong: and [TK]D-Fender, thanks bros! it worked |
| 21:20.46 | *** join/#asterisk dahunter3 (n=dahunter@static-71-189-78-103.lsanca.dsl-w.verizon.net) |
| 21:22.25 | *** join/#asterisk KW-CHS (n=none@208.189.33.150) |
| 21:23.40 | KW-CHS | What's the park code for Asterisk? |
| 21:23.48 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 21:23.55 | *** join/#asterisk anderiv (n=anderiv@207-67-87-34.static.twtelecom.net) |
| 21:24.49 | Nugget | "the" park code? how cute. Check features.conf |
| 21:25.00 | JerJer | anyone see anything wrong with this Set(device_trigger=${SIP_HEADER(X-Dev-Trigt)}) I see the header in the cli sip debug but its always null when it runs thru the dialplan - 1.4.11 |
| 21:26.03 | KW-CHS | Ok -- now how can I assign that to a button on my SNOM 360? |
| 21:26.26 | *** join/#asterisk kkn088 (n=kikoun@88-136-56-85.adslgp.cegetel.net) |
| 21:27.04 | KW-CHS | So I want to have the call, put it to Hold button and that'll send it to 700 (park) and then each phone will have a 701-704 pickup button. |
| 21:28.05 | *** join/#asterisk rbord (n=ricardo@AC9F7509.ipt.aol.com) |
| 21:28.21 | [TK]D-Fender | kwame, you're welcome |
| 21:28.28 | *** join/#asterisk neverblue2 (n=neverblu@unaffiliated/neverblue) |
| 21:32.14 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 21:35.39 | *** join/#asterisk Vorondil (n=vorondil@unaffiliated/vorondil) |
| 21:37.18 | KW-CHS | Can someone explain what I need to do to park a call? |
| 21:38.38 | marc7 | KW-CHS: http://www.voip-info.org/wiki-Asterisk+call+parking |
| 21:41.11 | *** join/#asterisk djs_2_6 (n=djstillm@cpe-075-182-081-167.nc.res.rr.com) |
| 21:46.57 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.188.154) |
| 21:59.44 | *** part/#asterisk thx2000 (n=evan@netblock-208-127-94-59.dslextreme.com) |
| 22:10.22 | KW-CHS | Ok -- I'm not seeing this "By default extension 700 is used to park a call. While in a conversation, press # to initiate a transfer, then dial 700. " |
| 22:10.51 | KW-CHS | I have a call . . . I press # and nothing happens??? |
| 22:14.40 | [TK]D-Fender | KW-CHS, that is ASSUMING that you're using * based DTMF transfers. |
| 22:14.49 | [TK]D-Fender | KW-CHS, What kind of phone are you using? |
| 22:15.27 | fujin | lol, funtimes |
| 22:15.33 | fujin | just cut my entire callcentre over to the voip system. |
| 22:17.37 | heelios | fujin: what did you use before? |
| 22:18.05 | fujin | um, a 'Gateway PABX' system from Telstraclear |
| 22:19.37 | heelios | sounds australian. ;p |
| 22:19.51 | fujin | nz actually, but rubbish all the same. |
| 22:20.05 | brookshire | an unlocked iPhone is up to $10,200 - http://tinyurl.com/26br8w :( |
| 22:20.20 | Qwell[] | brookshire: I'll mod yours |
| 22:20.31 | heelios | i wish I had been smart enough to do it. <_< |
| 22:20.49 | Sweeper | brookshire: 10k? must be a joke |
| 22:20.57 | brookshire | actually.. it's up to 15K now |
| 22:21.05 | Qwell[] | Sweeper: Don't forget, these ARE mac fanatics we're talking about |
| 22:21.18 | brookshire | probably from europe |
| 22:21.22 | heelios | ridiculous. |
| 22:21.36 | KW-CHS | D-Fender -- I'm using SNOM 360's |
| 22:21.37 | brookshire | because that's like $2 in euros :) |
| 22:21.55 | Sweeper | Qwell[]: eh, $10 says he doesn't get paid |
| 22:22.08 | Qwell[] | I'll take that bet. |
| 22:22.20 | Sweeper | I take paypal, and major credit cards |
| 22:22.25 | Sweeper | or cash~ |
| 22:22.27 | *** join/#asterisk gmalsack (n=gmalsack@h69-128-239-198.69-128.unk.tds.net) |
| 22:22.29 | Qwell[] | I take cash only |
| 22:22.34 | gmalsack | hello all |
| 22:22.41 | [TK]D-Fender | KW-CHS, Just use your phone's transfer feature then. Not a BLIND transfer, but an ATTENDED one. |
| 22:22.41 | Qwell[] | actually, I take redbull too |
| 22:22.51 | gmalsack | I have the same problem as bug issue 1223 |
| 22:23.03 | Sweeper | I'll bet some apple douchebags are running up the auction on fake accounts |
| 22:23.04 | gmalsack | However I don't see a resolution listed. can anyone give me a clue? |
| 22:23.17 | brookshire | for $15,100, I would deliver it personally |
| 22:23.21 | KW-CHS | How do I make it an attended one? |
| 22:23.37 | [TK]D-Fender | KW-CHS, Read your phone's manual |
| 22:23.54 | Sweeper | hmm |
| 22:24.00 | Sweeper | quite a few new users in the auction lists |
| 22:24.11 | Sweeper | but the top one is a blue star.... |
| 22:24.33 | brookshire | is blue like the lowest? |
| 22:24.47 | Sweeper | nah |
| 22:24.57 | Sweeper | blue = 50 positive feedbacks |
| 22:25.02 | Sweeper | and a memeber for 5+ years |
| 22:25.08 | Sweeper | well, blue is just 50 |
| 22:25.13 | brookshire | yeah.. but it's weird.. he's only ever bidded on 2 things |
| 22:25.21 | Qwell[] | seller |
| 22:25.21 | Sweeper | in the last month, anyways |
| 22:25.27 | brookshire | oh. ok |
| 22:25.30 | brookshire | <-- not an ebayer |
| 22:25.50 | marc7 | gmalsack: i can't see your privmsgs for some reason... what did you want? |
| 22:26.22 | Sweeper | anyways, there's a website with software unlock that's gonna be out in a couple of weeks |
| 22:27.15 | gmalsack | does a bug related to a queue hanging up after a minute sound familiar? |
| 22:28.23 | gmalsack | . |
| 22:28.45 | gmalsack | Does anyone here know why a queue might disconnect after exactly 60 seconds? |
| 22:28.54 | Qwell[] | gmalsack: We heard you the first 3 times. |
| 22:28.58 | Sweeper | maybe you told it to |
| 22:28.59 | Sweeper | ? |
| 22:29.26 | gmalsack | thanks for the non help.. have a good weekend. :) |
| 22:29.31 | *** part/#asterisk gmalsack (n=gmalsack@h69-128-239-198.69-128.unk.tds.net) |
| 22:29.45 | Sweeper | I will, sans extraneous periods |
| 22:30.31 | [TK]D-Fender | Qwell[], Yup, its Friday... when all the fuck-tards come out to play :) |
| 22:34.18 | KW-CHS | Ok -- I'm trying all this and looking at phone docs . . . nothing is happening when I put call to 700?? What can I trace or track down to see where the call is vapourizing? |
| 22:38.01 | KW-CHS | I've got this in extensions.conf: [general] static=yes |
| 22:38.01 | KW-CHS | <PROTECTED> |
| 22:38.01 | KW-CHS | <PROTECTED> |
| 22:38.01 | KW-CHS | <PROTECTED> |
| 22:38.22 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
| 22:38.45 | [TK]D-Fender | KW-CHS, "include => parkedcalls" needs to be in the context your PHONE uses, or in another linked to it |
| 22:38.58 | [TK]D-Fender | KW-CHS, This is not something you can just cram in [general] |
| 22:39.26 | KW-CHS | Ok . . . let me change that. |
| 22:40.42 | KW-CHS | so in my sip.conf file? |
| 22:41.28 | KW-CHS | ??? Where do I put that? |
| 22:41.32 | [TK]D-Fender | KW-CHS, Whatever context your phones use should have that INCLUDE in it. |
| 22:41.46 | KW-CHS | Freaking asterisknow setup . . . |
| 22:41.55 | [TK]D-Fender | KW-CHS, indeed. Ditch that shit :) |
| 22:44.54 | *** join/#asterisk JoseBravo (n=jbravo@190.156.225.15) |
| 22:45.17 | JoseBravo | Im receiving this error, Dial failed due to CONGESTION - failing through to other trunks |
| 22:45.29 | JoseBravo | But I don't know why is that? |
| 22:46.16 | *** join/#asterisk lirakis (n=lirakis@cpe-68-175-38-65.nyc.res.rr.com) |
| 22:46.44 | KW-CHS | Unfortunately I'm sitting at location, phones supposed to go live monday. :-) |
| 22:47.25 | *** join/#asterisk jedaustin (n=chatzill@wsip-66-210-241-251.ph.ph.cox.net) |
| 22:50.09 | marc7 | this is where paid support would come in handy |
| 22:51.10 | scooby2 | what would stop me from having > 199 extensions? I've searched everywhere yet am still at a loss. I dont really have that many but I want to add extension 999. |
| 22:52.34 | jedaustin | scooby2: the I/O limits of the box would limit how many actual connections you have. As far as I know there are no limits on how many extensions you can create - you may need to modify your dialplan to allow 4 digit extensions |
| 22:53.37 | scooby2 | 999 would be considered four digit? |
| 22:53.46 | jedaustin | Anyone successfully installed an astribank without Xorcom's help? |
| 22:53.51 | jedaustin | scooby2: no |
| 22:54.04 | [TK]D-Fender | KW-CHS, yup.... |
| 22:54.39 | JoseBravo | Im trying to do a call using a SIP trunk, but I get this error: -- Executing [s-CONGESTION@macro-dialout-trunk:1] NoOp("SIP/120-081f6620", "Dial failed due to CONGESTION - failing through to other trunks") in new stack |
| 22:54.41 | JoseBravo | Any idea? |
| 22:54.54 | [TK]D-Fender | scooby2, exten => 28376487236487263,1,NoOp(I'm a HUGE digit extension!!!) |
| 22:55.35 | [TK]D-Fender | JoseBravo, show us the line that GENERATED the error. |
| 22:58.18 | scooby2 | [TK]D-Fender: thanks |
| 22:58.51 | JoseBravo | [TK]D-Fender http://www.pastebin.ca/669680 |
| 22:59.59 | KW-CHS | In asterisk, what should I be looking for (conf files) to get the parking working . . . I've looked at the voipinfo page and it was clear as mud . . . |
| 23:02.26 | [TK]D-Fender | JoseBravo, -- SIP/MLV-081fb000 is circuit-busy <--- maybe this is a clue |
| 23:02.58 | *** join/#asterisk n00dle (n=ccraft@204.10.248.123) |
| 23:03.01 | [TK]D-Fender | KW-CHS, I told you pretty blatantly. You have have that include statement in the context your phone uses. |
| 23:03.16 | n00dle | Any grandstream fans? |
| 23:03.37 | KW-CHS | Ok . . sorry, this is all so "hand held" with the *Now crap . . . |
| 23:06.26 | *** join/#asterisk bryanfe2 (n=chatzill@wsip-70-169-190-173.sb.sd.cox.net) |
| 23:06.37 | KW-CHS | The phone context would be in the sip.conf file?? |
| 23:07.24 | bryanfe2 | hey folks, I have a SIP client behind a NAT firewall. When the SIP client places a call, Asterisk is trying to send the audio to it's internal 192.168.xxx.xxx address, which is wrong (it's not the external IP). I thought that "nat=yes" in sip.conf is supposed to correct for this, but it doesn't seem to be doing it. Is there something else I'm missing? |
| 23:09.37 | jedaustin | bryanfe2: This is from memory... make sure nat=YES in your configuration, set externip to whatever the real ip |
| 23:09.56 | bryanfe2 | "tethereal" shows that all SIP traffic is correctly going the client's external IP address, but all RTP traffic is going to it's internal IP address. |
| 23:13.09 | [TK]D-Fender | KW-CHS, no, CONTEXTS are part of your dialplan (extensions.conf) |
| 23:13.35 | [TK]D-Fender | bryanfe2, you need a lot more than just "nat=yes". Read this : |
| 23:13.37 | [TK]D-Fender | ~sipnat |
| 23:13.37 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 23:13.38 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^ |
| 23:14.59 | KW-CHS | Thanks D-Fender . . . looking into that. |
| 23:18.10 | bryanfe2 | thanks TKJD-Fender -- isn't there just an option somewhere in Asterisk to say "for the RTP channel, use their external Ip address?", and if there isn't, then it's not clear to me what good "nat=yes" does in the first place. |
| 23:18.47 | jedaustin | bryanfe2: nat=yes adds more info to the header information |
| 23:20.32 | JoseBravo | [TK]D-Fender but what can I do with that? |
| 23:21.17 | [TK]D-Fender | JoseBravo, the other side is BUSY, there's nothing YOU can do about it. |
| 23:21.38 | [TK]D-Fender | bryanfe2, its all documented in that first link, now go READ |
| 23:22.01 | bryanfe2 | reading now! |
| 23:22.45 | [TK]D-Fender | ok, I'm out later all |
| 23:23.07 | bryanfe2 | thank you fender |
| 23:23.09 | JoseBravo | [TK]D-Fender im connecting using xlite, and I can do calls. |
| 23:26.22 | Sweeper | Qwell[]: well, unlocked iphone is up to 200k :) |
| 23:26.26 | Sweeper | <PROTECTED> |
| 23:26.28 | Sweeper | I lied |
| 23:26.34 | Sweeper | 19million |
| 23:26.47 | Sweeper | make that 20 |
| 23:27.00 | Sweeper | Qwell[]: paypal to aleks.clark@gmail.com :D |
| 23:27.12 | Qwell[] | highest real bidder will get it ;) |
| 23:27.44 | jedaustin | Sweeper: and that has what to do with Asterisk? |
| 23:28.00 | Sweeper | jedaustin: I bet Qwell[] the dude wouldn't get 19k for his phone |
| 23:28.25 | Sweeper | also, if you can't see what an unlocked iphone has to do with asterisk, I suggest you go read the 1.4 changelog |
| 23:33.43 | bryanfe2 | hey folks.. Fender's article didn't help although it was useful to read.. implemented that configuration, and asterisk is still sending RTP data to the sip client's internal IP address, even though it's sending SIP data correctly to it's external IP address. |
| 23:33.55 | bryanfe2 | seems weird to me |
| 23:38.42 | KW-CHS | Ok -- I can now park a call . . . wahoo! Now, can I get that call that's parked light a light on the phone? |
| 23:39.14 | KW-CHS | I see on the display -- 701 . . I transfer to 700 and that comes up as the place it got parked. |
| 23:40.44 | jedaustin | Sweeper: I see NOTHING in the 1.4 changelog related to the iphone. |
| 23:41.11 | jedaustin | Sweeper: so how is your selling an unlocked iphone related to Asterisk again? |
| 23:41.14 | bryanfe2 | can anyone throw out any ideas about how to get asterisk to send RTP data back to a SIP client's external IP address rather than it's internal? Stun is not an option and knowing their external IP is not an option. |
| 23:41.37 | KW-CHS | I want the light to come on . . . I have a SNOM 360 phone . . so I have to do anything in the web gui on the phone or is all asterisk? |
| 23:41.40 | jedaustin | bryanfe2: why would you want it to go to the external IP? |
| 23:42.06 | bryanfe2 | jedaustin because right now it's sending RTP data to the SIP client's internal Ip address, and the SIP client is behind a nat firewall (typical home router) |
| 23:42.29 | jedaustin | bryanfe2: nat=yes and qualify=yes fixes my nat problems with sip devices |
| 23:43.07 | bryanfe2 | thanks jed, I have those settings in place but still no go, Asterisk keeps sending audio data (RTP) to 192.168.x.x which is the SIP client's internal IP address at it's own LAN far away... |
| 23:43.23 | bryanfe2 | pulling my hair out here |
| 23:45.18 | *** join/#asterisk boch (n=fran@190.48.243.161) |
| 23:45.41 | boch | hallo |
| 23:48.27 | boch | is it possible to make two simultaneous connections to the AMI with the same user ? |
| 23:48.42 | jedaustin | Anyone installed a Xorcom astribank? |
| 23:52.03 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@216-110-94-230.static.twtelecom.net) |
| 23:54.06 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
