| 00:00.32 | florz | NMR_1122: well, I don't have much of a clue as to how toll-free numbers work in your part of the planet, but if the ITSP does the termination from the PSTN for you, there isn't any "forwarding" involved beyond the primary termination point |
| 00:01.48 | NMR_1122 | so the ITSP will just pass them on as calls, concurrently? |
| 00:02.02 | lanning | yes |
| 00:04.25 | florz | NMR_1122: whether you can rely on that depends on your contract with the ITSP, obviously |
| 00:04.41 | florz | NMR_1122: if it says max. 1 call, then not |
| 00:05.56 | florz | NMR_1122: however, given that you most likely pay them per minute of the call, it wouldn't really make much sense to limit the number of concurrent calls (beyond protecting their infrastructure from overload, maybe ;-) |
| 00:06.38 | florz | NMR_1122: in order for it to actually work, you obviously also need sufficient internet bandwidth ... |
| 00:06.47 | NMR_1122 | since when do phone companies do things that make sense? |
| 00:07.11 | florz | NMR_1122: well, that's true ... so better check the contract ;-) |
| 00:09.07 | [TK]D-Fender | They do when they want to keep a client |
| 00:09.12 | NMR_1122 | Why are residential's always unlimited where business is per minute? |
| 00:09.16 | [TK]D-Fender | "My way or GTFO" |
| 00:09.32 | lanning | I know voicepulse was limiting to 4 channels. |
| 00:09.38 | [TK]D-Fender | NMR_1122: Because normal housholds don't talk on the phone 24/7 |
| 00:09.56 | NMR_1122 | unless they have a teenage daughter? |
| 00:10.00 | [TK]D-Fender | NMR_1122: "unlimited" is a marketing term anyways. |
| 00:10.27 | Kumba_ | unlimited is a product, not a definition ala websters :) |
| 00:10.30 | [TK]D-Fender | NMR_1122: Virtually all have either soft-caps, or average out the calculated cost of such a plan against realistic average usage |
| 00:10.34 | Kumba_ | and it's usually aroudn 20K-minutes/mo |
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| 01:12.57 | raden | wow packet jiiter 100-140 my dsl suxs |
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| 01:20.31 | astrobear | how could a free software equivalent of google voice be created? i figured if anyone know, it would be someone in this channel. |
| 01:21.05 | mchou | astrobear: what do you mean? gv uses asterisk afaik |
| 01:21.50 | astrobear | i want actual desktop software instead of having to use my browser |
| 01:22.00 | astrobear | something to integrate into my office suite |
| 01:22.17 | mchou | you've got to be kidding me, right? |
| 01:22.23 | mchou | office suite? |
| 01:22.27 | astrobear | -office |
| 01:22.29 | astrobear | i know i know |
| 01:22.42 | astrobear | more like communication software like kontact |
| 01:23.12 | astrobear | wants it badly in kontact |
| 01:23.37 | mchou | i dont get it |
| 01:23.50 | mchou | use kphone then |
| 01:23.59 | mchou | what's the problem? |
| 01:24.18 | drmessano | Google Voice uses Asterisk? |
| 01:24.30 | mchou | or any other softphone for that matter |
| 01:24.51 | drmessano | You dont use a softphone with Google Voice |
| 01:24.56 | mchou | drmessano: that's what the "user agent" says :) |
| 01:25.08 | drmessano | User agent from what? |
| 01:29.02 | mchou | user agent via gizmo |
| 01:29.31 | drmessano | The call would come from Gizmos proxy, so Gizmo uses Asterisk |
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| 01:29.51 | mchou | ok, maybe that's wwhat it was |
| 01:30.20 | drmessano | Google Voice doesnt touch SIP at all, if it did, why would be have to use Gizmo at all? |
| 01:31.03 | drmessano | I certainly would cut out the middle man.. Gizmo is kinda sucky |
| 01:31.05 | mchou | how does gv call the gizomo #? |
| 01:31.25 | mchou | I mean the gizmo # is not real PSTN |
| 01:31.26 | drmessano | Peering arrangement |
| 01:32.24 | p1mrx | I've hacked up my asterisk box to make "outbound" google voice calls |
| 01:32.43 | mchou | p1mrx: hmm?? |
| 01:32.46 | drmessano | How did you arrange that? |
| 01:32.51 | p1mrx | it actually submits a click2call request over HTTP, and splices in the inbound call |
| 01:33.10 | drmessano | Hmmmm |
| 01:33.12 | p1mrx | that's why "outbound" is in quotes; it's really an inbound call that gets patched over at the server |
| 01:33.13 | mchou | ahh, I suppose that can work |
| 01:33.21 | drmessano | Right |
| 01:33.37 | p1mrx | Gizmo/GV seems to have a problem with DTMF though |
| 01:33.40 | drmessano | So how does a call work.. start to finish |
| 01:33.42 | mchou | p1mrx: care to share the code? :) |
| 01:33.53 | p1mrx | http://www.pmarks.net/posted_links/google-voice-dialout.agi |
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| 01:34.05 | mchou | p1mrx: indded, gizmo dtmf sucks |
| 01:34.28 | mchou | p1mrx: so does gv "press 1" for that matter :) |
| 01:34.29 | p1mrx | is there a way to actually make it work? I don't understand whether the "dtmfmode" parameter refers to inbound or outbound dtmf |
| 01:34.54 | mchou | p1mrx: I doubt it. I tried various ways to sunday |
| 01:35.40 | mchou | p1mrx: heh, not bad. readable code in python :) |
| 01:35.41 | drmessano | Thats fascinating |
| 01:36.01 | mchou | p1mrx: pretty slick. Thank god you didnt use perl :) |
| 01:36.19 | p1mrx | perl? I'm not that masochistic |
| 01:36.56 | drmessano | Nonsense.. With perl, you could do it in one line of code.. Would just take 10 years to figure it out |
| 01:37.02 | mchou | haha |
| 01:37.05 | mchou | indeedy |
| 01:37.19 | drmessano | ------> oeidjj8rj9348jr984j3r9384jr9834j98rjr43 |
| 01:37.23 | drmessano | That MAY be it |
| 01:37.35 | mchou | haha |
| 01:37.37 | drmessano | or that may eject toast from app_toast |
| 01:37.40 | drmessano | Ive no clue |
| 01:39.44 | p1mrx | the HTTP code is somewhat more wordy that it should be, because I wanted to use persistent HTTP/1.1 |
| 01:40.24 | p1mrx | python's httplib supports HTTP/1.1, but urllib2, which wraps around it, does not |
| 01:40.33 | mchou | p1mrx: so GIZMO_NUMBER can be a PSTN # too, correct? |
| 01:40.56 | p1mrx | mchou: yes, that should work |
| 01:41.05 | p1mrx | it has to be one of the numbers registered with your GV account |
| 01:41.13 | mchou | yeah, understood |
| 01:43.01 | p1mrx | the "exten" rule in the comment is a bit simplistic; I've only been using Asterisk since this week. |
| 01:43.26 | mchou | yeah, np |
| 01:44.21 | p1mrx | I kinda wonder if there's a way to make this work without something as heavyweight as Asterisk, though. does anyone know if the SIP protocol supports connecting two INVITEs together? |
| 01:45.51 | mchou | dunno the internals of sip that well, but isnt that effectively a transfer? |
| 01:46.17 | p1mrx | no, a transfer takes an INVITE, and says "go to this address instead" |
| 01:46.33 | p1mrx | I take an INVITE, and another INVITE, and say "here, you talk to each other" |
| 01:47.14 | mchou | I dunno, sounds like you need a conference bridge :) |
| 01:47.29 | p1mrx | yeah, I'm using the Bridge() application in Asterisk 1.6 |
| 01:47.56 | p1mrx | that connects 2 active calls together |
| 01:49.03 | p1mrx | but, I'm assuming that it could probably be done with SIP alone, using some hackery. Like, respond to each INVITE with an ACK that contains the info from the opposite INVITE. and hope the voice protocol is directionless enough to not notice. |
| 01:50.54 | mchou | bah, at this point I'd settle if gv hooked up with gtalk (via asterisk) |
| 01:51.28 | mchou | yet another way to solve the problem w/o resorting to sip hackery |
| 01:51.47 | p1mrx | does gtalk work with asterisk? |
| 01:52.12 | mchou | yeah, there's a gtalk asterisk module (although I've never used it) |
| 01:52.49 | p1mrx | so, if I go into my gizmo section in sip.conf, and set "dtmfmode=inband", then I can send DTMF to the remote caller, but GV's "press 1 to accept this call" menu doesn't work |
| 01:53.07 | p1mrx | if I use the default dtmfmode, then it's the other way around. |
| 01:53.31 | mchou | yeah, and then there's "auto" |
| 01:53.46 | mchou | all faibus w/ gizmo/gv |
| 01:53.55 | mchou | umm, failbus* |
| 01:59.00 | mchou | I dont think sip by itself supports media mixing (hence the bridge) |
| 01:59.38 | mchou | the bridging function resides outside sip |
| 02:00.08 | mchou | so I'm dubious where you can just 'connect up' 2 invites |
| 02:00.24 | mchou | s/where/whether |
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| 02:30.48 | Brack10 | anyone trust their office to use 100% voip to make and receive calls....even for 911? |
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| 02:32.28 | raden | me |
| 02:32.49 | raden | whats the cheaper voip provider thats reliable ? anyone ? |
| 02:33.06 | jaytee | ~itsplist-us |
| 02:33.07 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 02:36.09 | raden | how do i port a 800 # ? |
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| 02:37.41 | shmaltz | helo everyone |
| 02:38.05 | shmaltz | ~hi |
| 02:38.06 | jbot | hello, shmaltz |
| 02:38.23 | raden | wow teliax expensive |
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| 02:42.55 | raden | is it possible to port a 800 # |
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| 02:43.55 | shmaltz | raden what do you mean if it's possible? |
| 02:44.24 | raden | every voip provider i call its a freakin issue they cant do it |
| 02:44.31 | raden | getting very frusterating |
| 02:44.36 | shmaltz | you mean port in? |
| 02:44.46 | shmaltz | thats because they are not real providers |
| 02:44.48 | raden | and the ones that can do it are like $40 a month for it |
| 02:44.52 | shmaltz | raden, where are you located? |
| 02:44.57 | raden | wisconsin |
| 02:45.04 | raden | weyauwega |
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| 02:45.09 | raden | our 800 # through ATT |
| 02:45.21 | shmaltz | what are you currently paying? |
| 02:45.22 | raden | were setting up new office im setting up asterisk |
| 02:45.34 | raden | 4 cents min |
| 02:45.42 | raden | and about 400 mo fort 3 lines |
| 02:45.58 | raden | i want to be down to about 60 a month for 4 lines |
| 02:46.14 | raden | our 6 inbound channels on 800 # 2 unlimited outbounds |
| 02:46.17 | shmaltz | raden, just use any CLEC, and you should be able to get it for $0.039 |
| 02:46.19 | raden | or 2 metered outbounds |
| 02:46.27 | raden | CLEO ? |
| 02:46.44 | shmaltz | CLEC |
| 02:47.01 | raden | whats CLEO ? |
| 02:47.07 | shmaltz | anyhow, how you going to get 4 lines 4 $60.00 a month? |
| 02:47.18 | shmaltz | I said CLEC not CLEO |
| 02:47.25 | shmaltz | ~CLEC |
| 02:47.26 | jbot | it has been said that clec is Created by the Telecommunications Act of 1996, a CLEC is a service provider that is in direct competition with an incumbent service provider. CLEC is often used as a general term for any competitor, but the term actually has legal implications. To become a CLEC, a service provider must be granted "CLEC status" by a state's Public Utilities Commission. In exchange for the time and money spent t |
| 02:48.04 | shmaltz | I meant $0.029 |
| 02:48.07 | shmaltz | not 3.9 |
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| 02:48.25 | raden | it showing up as C L E O on my side space it out |
| 02:48.57 | shmaltz | raden, then change the fonts |
| 02:49.02 | raden | did |
| 02:49.18 | shmaltz | anyhow, so how you going to get 4 lines for just $60.00 a month? |
| 02:49.56 | shmaltz | last time I checked just surcharges and tax was over $15.00 per line |
| 02:50.39 | shmaltz | raden, you there? |
| 02:50.49 | raden | yes |
| 02:51.03 | shmaltz | so how you going to get 4 lines for $60.00 a month? |
| 02:51.36 | raden | callcentric is one i looked at |
| 02:51.44 | shmaltz | oh, thru VoIP? |
| 02:51.45 | raden | primus can do 4 lines unlimited for 80 |
| 02:51.52 | raden | YES |
| 02:51.54 | drmessano | Callcentric isnt bad |
| 02:51.55 | *** join/#asterisk killown (n=nandateb@unaffiliated/killown) |
| 02:51.59 | drmessano | Just expensive outbound |
| 02:52.05 | drmessano | Flowroute is much cheaper |
| 02:52.11 | shmaltz | raden, how do you know how bad they are? |
| 02:52.11 | raden | i thought it was cheap |
| 02:52.16 | raden | drmessano, thanks |
| 02:52.27 | drmessano | Callcentrics termination is expensive |
| 02:52.37 | raden | shmaltz, ??? |
| 02:52.50 | shmaltz | raden, yes? |
| 02:53.04 | shmaltz | how do you know the quality of those companies? |
| 02:53.14 | raden | shmaltz, i didnt say they were or werent bad drmessano did |
| 02:53.31 | shmaltz | when you replcae a business lines with a VoIP provider you better make sure they are damn good quality, or you may lose your job |
| 02:53.35 | raden | we have a pots adapter from primus there service sucks skype works better |
| 02:53.36 | shmaltz | oh sorry |
| 02:54.25 | drmessano | shmaltz: Who the hell said the POTS was bad or not bad |
| 02:54.26 | shmaltz | in my experience if one is too cheap (like callcentrick is 19.95 unlimited) then they are not worth more |
| 02:54.26 | raden | just to test them out but then again packet jitter on other business location was ridiculous highest packet jitter i seen at new place is 19 over 20 hour test period |
| 02:54.40 | drmessano | Ive seen POTS be much less reliable than even a bad SIP provider |
| 02:54.42 | raden | any recomendations ???? |
| 02:54.58 | raden | ATT here sucks |
| 02:54.58 | shmaltz | drmessano, what stuff you on? |
| 02:55.02 | drmessano | Callcentric is high quality |
| 02:55.14 | raden | there are days are lines are down 3 - 5 hours and we take 1500 calls a day |
| 02:55.23 | shmaltz | POTS could be bad, but a bad VoIP providers quality could be good |
| 02:55.44 | drmessano | shmaltz: So all POTS lines are 100% reliable and noise free? Youve never had bad POTS lines at a location... a bad terminal.. or something ma bell couldnt fix? |
| 02:55.47 | drmessano | Give me a break dude |
| 02:55.49 | raden | well 54 people gave callcentric 5 star review but thats only 54 |
| 02:56.08 | shmaltz | drmessano, who said what you just said, stop implying things and make an argument out of it |
| 02:56.09 | raden | ok just looking for reliable providers here :) |
| 02:56.21 | raden | doh :( |
| 02:56.37 | drmessano | shmaltz: Youre implying the transport has anything to do with reliability |
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| 02:56.53 | drmessano | [22:54] <shmaltz> when you replcae a business lines with a VoIP provider you better make sure they are damn good quality, or you may lose your job |
| 02:56.57 | shmaltz | drmessano, NO, I am implying that the transport provider has to do with that |
| 02:57.19 | drmessano | If you replace VOIP lines with POTS line you better make sure they are damn good too |
| 02:57.21 | shmaltz | drmessano, PROVIDER not medium technology |
| 02:57.44 | shmaltz | drmessano, I can't argue on that, but if they are not, you wont lose your job |
| 02:58.05 | shmaltz | OTOH, if you replace POTS with VoIP and they are bad, you will lose your job |
| 02:58.11 | drmessano | LOL |
| 02:58.27 | raden | this isnt helping me :( |
| 02:58.34 | shmaltz | raden, ok, ok |
| 02:59.02 | shmaltz | first my opinion, if one is trying to replcae POTS with VoIP they should go (at least for inbound) with a VoIP provider that is a CLEC |
| 02:59.09 | drmessano | ..... |
| 02:59.14 | shmaltz | raden, what type of Internet connection do you have? |
| 02:59.22 | drmessano | What are you smoking? |
| 02:59.25 | shmaltz | drmessano, I said opionion |
| 02:59.32 | shmaltz | opinion |
| 02:59.55 | raden | Centurytel DSL, 3/4 mile from central terminal steady rate of 560 kbps burst 768 upload solid 6 mb down |
| 03:00.07 | raden | jitter 2 ms average |
| 03:00.08 | shmaltz | drmessano, this is based on lots of experience and includes all factors in it (LNP, Billing, Customer service etc.) |
| 03:00.13 | drmessano | Are you one of those that thinks VoIP service from AT&T would trump all other ITSPs? |
| 03:00.17 | raden | gets up in 4's once and a while |
| 03:00.37 | shmaltz | drmessano, NO |
| 03:00.55 | shmaltz | raden, could you put in a DSL just for the VoIP part? |
| 03:01.12 | raden | i have a netgear router with dual wans could if we had to |
| 03:01.28 | raden | i can seperate traffic that way |
| 03:01.39 | raden | but they nail us almost 80 per dsl line |
| 03:01.39 | shmaltz | raden, use a seperate router for that second DSL |
| 03:01.45 | drmessano | Do you know who the #3 provider of telephone service in the country is right now? |
| 03:01.47 | shmaltz | wow |
| 03:01.49 | drmessano | Comcast |
| 03:01.53 | shmaltz | thats expensive |
| 03:01.55 | drmessano | and their service is horrible |
| 03:01.59 | raden | yeah thats the issue |
| 03:02.00 | shmaltz | raden comcast/cv available? |
| 03:02.16 | raden | nothing else we in the desert for broadband T1 490 a month |
| 03:02.17 | shmaltz | drmessano, and they do an awful bad job at that |
| 03:02.27 | shmaltz | drmessano, where you taking that number from? |
| 03:02.40 | drmessano | It was published the other day in many publications |
| 03:02.45 | raden | is it a good idea to run out computers through switch on the astra 9133 phones ? |
| 03:02.59 | shmaltz | drmessano, many publications, a link to just one? |
| 03:03.03 | raden | or run computers on seperate network ? |
| 03:03.18 | drmessano | You can google it for yourself |
| 03:03.19 | shmaltz | raden, shouldn 't be a problem in most cases |
| 03:03.32 | shmaltz | drmessano, but I didn't mention it |
| 03:03.33 | drmessano | I dont remember which of 100 publications I read that it was in |
| 03:03.52 | drmessano | and I am not here to prove anything to you.. if you want to read it, google it |
| 03:04.40 | shmaltz | drmessano, http://www.reuters.com/article/technologyNews/idUSTRE52A6A920090311 |
| 03:04.42 | shmaltz | cmon thats home based service |
| 03:04.43 | shmaltz | not business |
| 03:05.15 | raden | shmaltz, here the setup westel DSL router telco provided -> FVX538 -> core duo 3.0 Ghz asterisk server w/ 8 gb Dual channel ram @ 1066 mhz -> 10 x astra 9133 phones 3 wifi phones 4 buffalo airstations running dd-wrt |
| 03:05.32 | raden | plan on running g.729 unless you dont think thatd be wise |
| 03:05.37 | shmaltz | raden, that sounds ok to me |
| 03:05.47 | shmaltz | hates g.729 |
| 03:06.00 | raden | what u recomend |
| 03:06.03 | shmaltz | thinks g729 is worse quality then a bullhorn |
| 03:06.08 | shmaltz | raden, ulaw |
| 03:06.15 | raden | 711 |
| 03:06.17 | shmaltz | you are on a damn local network so use |
| 03:06.22 | shmaltz | so use 711 |
| 03:06.45 | raden | 711 on network what about over the wan ? |
| 03:07.16 | shmaltz | raden, 711 over WAN will use around 90k per channel (more like 85) |
| 03:07.40 | raden | ouch |
| 03:07.43 | p1mrx | that's kbits/sec |
| 03:07.48 | shmaltz | which means that with 5 channels you are using 450k, thats why I mentioned using a seperate DSL |
| 03:08.01 | shmaltz | p1mrx, yes |
| 03:08.33 | shmaltz | raden, no cable service? |
| 03:08.45 | raden | cost 3 grand to get the line to us |
| 03:08.54 | raden | 2 MB wifi link |
| 03:08.56 | shmaltz | raden, which provider? |
| 03:09.03 | raden | charter |
| 03:09.13 | raden | i can hook a wifi link to a direct T3 |
| 03:09.15 | shmaltz | raden, with wifi you must measure the latency |
| 03:09.23 | raden | 3 ms |
| 03:09.31 | shmaltz | no I meant the cable, which provider |
| 03:09.38 | raden | charter communications |
| 03:09.44 | raden | 3 ms on the wifi link |
| 03:09.46 | shmaltz | raden, then go for the wifi if you can get it at 3ms |
| 03:09.49 | raden | 3 miles away |
| 03:10.01 | shmaltz | how much for the wifi link? |
| 03:10.04 | raden | shit i was 1 mile away when i did that |
| 03:10.10 | raden | $44 month |
| 03:10.18 | shmaltz | raden, you insane, just grab it |
| 03:10.33 | shmaltz | raden, you'll need to test for both latency and packet lost |
| 03:10.52 | raden | packet loss 0 over 3 weeks |
| 03:11.00 | shmaltz | if you can get less than 70ms latency then you are on very good quality |
| 03:11.05 | raden | i have a co locate repeater setup in boon docks |
| 03:11.17 | raden | off that signal |
| 03:11.34 | shmaltz | raden, thats your best choice then |
| 03:11.50 | shmaltz | anyone watched duplicity? |
| 03:12.22 | raden | crap they getting up there now its 1.5 mbps bi-directional for $46/mo |
| 03:13.21 | shmaltz | raden, still a good price |
| 03:14.02 | raden | they have restrictions im reading only 2 computers may be connected yada yada |
| 03:14.25 | raden | ill have to call them and negotiate that'd be like having a t1 |
| 03:14.26 | shmaltz | I pay $125 per month for 30mb down by 7mb up including one line and 5 statick IPs |
| 03:14.42 | shmaltz | raden, or you could use it just for the VoIP |
| 03:14.54 | raden | lag over wifi link 5 ms using my colocate to ping there radio |
| 03:14.58 | *** part/#asterisk NMR_1122 (n=rahl@adsl-068-209-105-089.sip.mia.bellsouth.net) |
| 03:15.02 | shmaltz | cool |
| 03:15.11 | raden | thats over 5 miles |
| 03:15.29 | raden | deliberant radio & 23 db gain yagi |
| 03:15.31 | Talkradio | shmaltz nice price |
| 03:15.55 | shmaltz | Talkradio, if you are in cablevision area you can get the same |
| 03:15.57 | raden | my big issue no one wants to sell over a 512 kb uplink |
| 03:16.13 | Talkradio | i'm in timewarner |
| 03:16.24 | shmaltz | talkradio, should work as well |
| 03:16.30 | shmaltz | talkradio, which city? |
| 03:16.39 | Talkradio | studio city |
| 03:16.45 | raden | 512k up/down service is $39.95 |
| 03:16.53 | raden | 15mb down 512 up is 59.95 |
| 03:17.03 | raden | + you must have a phoneline so another 31 |
| 03:17.13 | Talkradio | i have a hacked cable modem with ext usb serial cable i bought online but never actually used it |
| 03:18.09 | Talkradio | thought it would be somehting fun to play with and lost interest right after i bought it |
| 03:18.33 | raden | oh and the kicker there is a DS-3 line 40 ft in front of our building on our property |
| 03:19.21 | raden | DS-4 sorry |
| 03:19.43 | shmaltz | talkradio, it's not worth playing with their networks anymore, they are very strict and will shut you down |
| 03:19.49 | shmaltz | gtg |
| 03:19.51 | shmaltz | cya guys |
| 03:19.53 | shmaltz | l8r |
| 03:19.55 | shmaltz | bye |
| 03:20.32 | drmessano | idiit |
| 03:20.37 | drmessano | :P |
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| 03:27.39 | p1mrx | hm... is it possible to set a call's dtmfmode in the dialplan? |
| 03:28.22 | p1mrx | that would help solve the google voice problem; use inband for outgoing calls only |
| 03:29.34 | p1mrx | since it's very rare that anyone would need to enter DTMF tones for an incoming call (except for the "press 1 to accept" menu) |
| 03:32.52 | *** join/#asterisk neurosys (n=vinix@c-71-196-19-254.hsd1.fl.comcast.net) |
| 03:34.32 | p1mrx | answer: yes! |
| 03:35.26 | p1mrx | exten => s/6502650000,1,SipDtmfMode(inband) |
| 03:35.50 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 03:36.02 | p1mrx | so, calls coming directly from my GV number use audio signaling, while calls from anyone else use rfc signaling, so that the GV menu can understand it |
| 03:37.40 | drmessano | Awesome |
| 03:37.56 | p1mrx | it's hardly a fix, but it's a practical workaround |
| 03:38.18 | drmessano | You hacked the Google |
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| 03:39.24 | drmessano | Thats bigger than the Gibson |
| 03:39.42 | naxxfish | am I right in thinking that in order to get DAHDI support, i need to configure --with-dahdi ... |
| 03:40.06 | naxxfish | and that the default ./configure doesn't have it enabled by default? |
| 03:40.39 | [TK]D-Fender | naxxfish: in order to have support it has to be installed first |
| 03:41.03 | naxxfish | it was installed - i just didn't specify it in the ./configure line |
| 03:41.28 | naxxfish | i figured it'd automatically look for it by default |
| 03:42.33 | [TK]D-Fender | naxxfish: it would. Standard "./configure" would pick it up |
| 03:42.41 | naxxfish | hm ... that's odd then |
| 03:42.48 | [TK]D-Fender | naxxfish: perhaps you didn't clear the last time you ran it off. |
| 03:43.07 | naxxfish | seems unlikely, this is a completely fresh install |
| 03:43.11 | [TK]D-Fender | naxxfish: Which is why I always reextract the tarball from scratch |
| 03:44.07 | naxxfish | i've checked out the subversion just now, doing a make on that currently |
| 03:44.24 | naxxfish | with any luck i'll get app_sms working :) |
| 03:45.21 | *** join/#asterisk Spirits-Sight (n=christop@c-71-192-91-123.hsd1.ma.comcast.net) |
| 03:45.39 | Spirits-Sight | why would I get this " -- Got SIP response 489 "Bad event" back from 87.229.111.190"? |
| 03:45.59 | Spirits-Sight | I don't even know what this ip address is? |
| 03:46.30 | [TK]D-Fender | Spirits-Sight: perhaps you should actually look at the entire SIP conversation. |
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| 03:47.58 | Spirits-Sight | [TK]D-Fender: I did, I was just trying figure out this, it been doing it for a few days now, also right now I can make calls but when calls come in it goes right to my menu which it was not doing this the other day, I have not changed any thing |
| 03:48.27 | [TK]D-Fender | Spirits-Sight: And I don't see your failed calls like usual. |
| 03:49.26 | Spirits-Sight | the way the sys is setup is it ring my ext then 20 sec later goes to your being transfered then it calls outside number I will pastebin it |
| 03:54.14 | Spirits-Sight | [TK]D-Fender: I hope got all privit stuff hidden :-) http://pastebin.com/d4daa0315 |
| 03:55.51 | [TK]D-Fender | Spirits-Sight: Well clearly your phone is rejecting the call * is trying to send it and the next priority is continuing on to do the rest of what you have it do |
| 04:02.19 | naxxfish | unf |
| 04:03.03 | naxxfish | just compiled out of svn, and this happens when I run it :/ http://pastebin.com/m145e987d |
| 04:05.00 | Spirits-Sight | [TK]D-Fender: why would it do this? |
| 04:05.38 | [TK]D-Fender | Spirits-Sight: You configured your phone wrong. |
| 04:06.13 | Spirits-Sight | I have not touch the setup of phone at all |
| 04:06.22 | [TK]D-Fender | naxxfish: I would go from the release versions, not SVN |
| 04:06.38 | [TK]D-Fender | Spirits-Sight: You are hiding stuff and that same excuse means NOTHING. |
| 04:06.50 | [TK]D-Fender | Spirits-Sight: Change or accept continued failure. |
| 04:07.31 | naxxfish | [TK]D-Fender: mmkay, just the 1.6.0.6 release was giving me problems |
| 04:07.37 | Spirits-Sight | what you talking about, I gave what it said, minue person name and phone numbers |
| 04:08.51 | [TK]D-Fender | Spirits-Sight: no sip debug, and hearing "it used to work" is just annoying at this popint. |
| 04:08.56 | [TK]D-Fender | point* |
| 04:09.26 | Spirits-Sight | if you don't mind how do you do a sip debug? never did that before |
| 04:13.15 | [TK]D-Fender | Spirits-Sight: "help sip" |
| 04:15.04 | Spirits-Sight | I already got it on :-) thanks, how much of it do you need it change alot? also is there stuff in it that I should mask |
| 04:20.17 | Spirits-Sight | http://pastebin.com/d2f8eac14 do you need more? |
| 04:22.02 | [TK]D-Fender | Spirits-Sight: ! tiny piece in the middle of a conversation. Complete waste of time. |
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| 04:22.24 | Spirits-Sight | [TK]D-Fender: sorry now the phone works again, this is strange, I changed nothing on the phone or * |
| 04:22.39 | naxxfish | http://pastebin.com/m702d2b10 here's the issue i was getting with 1.6.0.2 |
| 04:22.50 | naxxfish | (and DAHDI) |
| 04:23.50 | naxxfish | apparently there is a patch, but i've yet to actually find it |
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| 05:04.21 | mchou | p1mrx: so does SipDtmfMode work for you with google voice/gizmo? |
| 05:04.50 | mchou | p1mrx: I just tried and it's still no joy for me :( |
| 05:17.57 | p1mrx | mchou: it seemed to work, yes. |
| 05:18.11 | p1mrx | but, the key is, you need to use inband to send DTMF to a remote caller |
| 05:18.19 | p1mrx | and rfc2833 to send DTMF to the GV menu system |
| 05:18.36 | p1mrx | so, you need to set inband on outbound calls, and rfc2833 on inbound calls |
| 05:18.58 | mchou | p1mrx: ahh, ok. I misread you earlier (explains why I had no luck) |
| 05:19.15 | mchou | p1mrx: I'll give it another shot.... |
| 05:19.23 | p1mrx | as I said, it doesn't really fix anything, but it works for the most common cases |
| 05:19.45 | mchou | I just want it to work for "press 1" for GV/gizmo |
| 05:19.58 | mchou | i.e. inbound |
| 05:21.12 | p1mrx | for me, I don't seem to have to set anything for DTMF to work on the GV menu |
| 05:21.20 | p1mrx | so, I assume rfc2833 is the default |
| 05:21.27 | mchou | what?? |
| 05:21.43 | mchou | "press 1" works for you for GV/Gizmo? |
| 05:21.57 | p1mrx | it did last I checked |
| 05:22.06 | p1mrx | but, my SIP client only supports rfc2833 |
| 05:22.13 | mchou | it's _never_ worked for me |
| 05:22.25 | p1mrx | try setting "dtmfmode=rfc2833" in your sip.conf, for the proxy01.sipphone.com secion |
| 05:22.33 | mchou | I did :) |
| 05:22.51 | mchou | dtmfmode=inband and "auto" too |
| 05:23.18 | p1mrx | I just tried, and it worked. |
| 05:23.59 | mchou | yeah, I'll give it a shot tomorrow |
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| 05:24.27 | p1mrx | oh, actually, I was looking at the wrong config file when I said I didn't set anything. |
| 05:24.36 | p1mrx | I actually have "dtmfmode=rfc2833" |
| 05:25.13 | p1mrx | and then I use SipDtmfMode(inband) for outgoing calls (i.e. calls directly from my GV number) |
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| 05:27.39 | mchou | see, that never worked for me |
| 05:28.19 | mchou | my dtmfmode in sip.conf was always rfc2833 |
| 05:28.46 | mchou | and no matter what "press 1" inbound on GV/gizmo was fail |
| 05:29.08 | p1mrx | maybe it has to do with the configuration on the phone itself... is that set to rfc2833 also? |
| 05:29.38 | p1mrx | I'm using Ekiga, if it matters |
| 05:31.06 | NovceGuru | what are those little black tube things in hot and sour soup |
| 05:31.49 | mchou | p1mrx: I'm using pap2 |
| 05:32.10 | mchou | p1mrx: hmm, maybe pap2 firmware is fubar |
| 05:33.20 | p1mrx | some searches indicate that the PAP2 has a DTMF Tx option |
| 05:33.28 | mchou | it does |
| 05:33.45 | mchou | but the choices are inband, avt,info |
| 05:34.10 | mchou | I suppose info==rfc2833 |
| 05:34.48 | mchou | I could have sworn it had option for rfc2833 b4 |
| 05:35.26 | inckie | even with your life? |
| 05:35.52 | mchou | inckie: come on man |
| 05:36.17 | inckie | :P |
| 05:36.24 | p1mrx | mchou: I'd say try them all, and see what happens |
| 05:37.13 | p1mrx | AVT might be the same as rfc2833, I'm not completely sure though |
| 05:37.14 | mchou | p1mrx: screw it, been there, done that |
| 05:37.34 | mchou | at this point I've lost interest in gizmo |
| 05:37.55 | mchou | no wonder it's such a non starter for google voice |
| 05:38.20 | mchou | makes skype look easy |
| 05:38.41 | mchou | and I dont mean that in a good way |
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| 05:59.18 | drmessano | Hmmmm |
| 05:59.45 | p1mrx | is it possible to make Asterisk send both inband and rfc2833 tones simultaneously? |
| 06:00.09 | p1mrx | I feel embarrased for suggesting that :-) |
| 06:05.10 | p1mrx | looks like you probably can't do it without changing the program |
| 06:10.05 | jql | I would concur with that |
| 06:10.53 | p1mrx | I see where to change chan_sip.c to make it send both, but my current inbound/outbound works well enough for now. |
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| 08:21.45 | hi365 | a bit off topic: im looking for a source guide on specifications for home wiring. specifily, if I were to run adsl over cat 5e, how far would I need to keep it form the power lines? |
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| 08:22.20 | _omer | any asterisk developer there? (Programmer / Coder) |
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| 13:03.45 | Pegasus_RPG | hello |
| 13:04.48 | Pegasus_RPG | I'm trying to use QuteCom 2.2 with an Asterisk PBX using Speex @ 8kHz and Asterisk is refusing to communicate with this codec |
| 13:05.11 | Pegasus_RPG | If I force it to only allow speex, it gives SIP/199-00af8c70 is circuit-busy when I answer on the softphone |
| 13:05.22 | Pegasus_RPG | (It works fine with GSM, uLaw, and other codes) |
| 13:05.26 | Pegasus_RPG | codecs |
| 13:06.02 | Pegasus_RPG | Any idea what the problem could be? (I need Speex because my wife is using a really low bandwidth satellite connection.) |
| 13:13.09 | *** join/#asterisk s0lid (n=s0lid@122.53.98.50) |
| 13:13.48 | NovceGuru | Pegasus_RPG: is speex allowed in the sip.conf in the context of the extension you are trying to register |
| 13:13.56 | Pegasus_RPG | it is |
| 13:14.47 | Pegasus_RPG | if I allow another codec too after speex, it'll use that instead |
| 13:14.48 | NovceGuru | what else does the console spitting out |
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| 13:15.02 | NovceGuru | does the sip client support speeex |
| 13:15.17 | Pegasus_RPG | Yes. |
| 13:15.18 | NovceGuru | is speex support compiled into asteirsk |
| 13:15.19 | NovceGuru | etcetc |
| 13:15.41 | Pegasus_RPG | using Asterisk from Debian repo which includes speex |
| 13:16.15 | Pegasus_RPG | console output: http://pastebin.com/d7e5bfba5 |
| 13:16.32 | *** join/#asterisk Daejeo (n=chatzill@114.201.159.235) |
| 13:17.19 | NovceGuru | paste show codecs |
| 13:17.35 | NovceGuru | maybe speex doesnt like the ~3000msec latency of the sat link? |
| 13:18.00 | Pegasus_RPG | http://pastebin.com/d39bf08d5 |
| 13:18.07 | Pegasus_RPG | I'm just testing on the LAN right now |
| 13:19.15 | Pegasus_RPG | (And the client can connect with GSM and uLaw over the sat link, but with 32kbps upstream, I can't hear much. :) ) |
| 13:20.19 | NovceGuru | gsm can be ~13kbps |
| 13:20.34 | Pegasus_RPG | I thought so but how can I configure that? |
| 13:21.04 | Pegasus_RPG | (COurse I'd rather get speex working since it's supposed to have better quality for a given bandwidth) |
| 13:21.19 | NovceGuru | yes |
| 13:21.21 | NovceGuru | test on the lan |
| 13:21.24 | NovceGuru | maybe its the cleint |
| 13:21.38 | Pegasus_RPG | qutecom.com |
| 13:22.03 | NovceGuru | she on a mac? |
| 13:22.06 | Pegasus_RPG | no, Linux |
| 13:22.14 | NovceGuru | ah |
| 13:22.23 | Pegasus_RPG | But I'm testing on the LAN in Windows |
| 13:22.32 | Pegasus_RPG | calling from a Snom 300 using ulaw |
| 13:22.38 | NovceGuru | i've always had decent luck with x-lite |
| 13:22.46 | NovceGuru | working? |
| 13:23.02 | Pegasus_RPG | If i use GSM on the softphones, yes |
| 13:23.18 | Pegasus_RPG | * seems either unable to talk using speex or doesn't know how to transcode |
| 13:23.39 | NovceGuru | speex to speex should work |
| 13:23.43 | NovceGuru | can the snom300 do speex? |
| 13:23.46 | Pegasus_RPG | no |
| 13:23.49 | Pegasus_RPG | unfortunatle |
| 13:23.51 | Pegasus_RPG | y |
| 13:24.22 | NovceGuru | could be the linux client vs windows client |
| 13:24.33 | NovceGuru | can you call from the windows softphone to her :) |
| 13:24.38 | Pegasus_RPG | Haven't tried that |
| 13:24.58 | NovceGuru | you do know it'll be like a walkie talkie :P |
| 13:25.15 | Pegasus_RPG | Sortof, Skype works decently |
| 13:25.24 | Pegasus_RPG | after it realizes the link is so slow |
| 13:25.28 | NovceGuru | yeah |
| 13:25.37 | NovceGuru | still can't do anything about speeding up the speed of light though |
| 13:25.43 | Pegasus_RPG | heh |
| 13:25.44 | NovceGuru | unless its not an orbiting satellite you mean |
| 13:26.52 | *** join/#asterisk egypcio (n=egypcio@unaffiliated/egypcio) |
| 13:27.01 | Pegasus_RPG | so I'm first trying to get my Snom 300 w/ uLaw to talk to QuteCom w/ Speex via * on the LAN before I make her compile the latest version of QuteCom |
| 13:27.28 | Pegasus_RPG | (Since the older version she has only supports wideband speex) |
| 13:27.42 | Pegasus_RPG | (So that wasn't working) |
| 13:31.16 | Pegasus_RPG | I gotta run now. Will mark myself away so hopefully if anyone else has any suggestions, I'll get them. :) THanks NovceGuru ! |
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| 15:40.49 | bl4 | Can someone see if they can access sip/morse@kd7ike.no-ip.biz ? |
| 15:43.55 | *** part/#asterisk Ericounet (n=Ericoune@ACaen-151-1-34-194.w86-215.abo.wanadoo.fr) |
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| 16:00.13 | *** join/#asterisk patrick-- (n=patrick@eos.openroot.de) |
| 16:00.51 | patrick-- | Hey all, I remember there beeing a feature where the caller has to say its name, its recorded, then the target extension is called, played the sound and given a variety of options what to do with the call. |
| 16:01.01 | patrick-- | what is that feature called? i seem to have "lost" it :) |
| 16:01.59 | [TK]D-Fender | patrick--: "core show application record" , "core show application dial" <- M() |
| 16:02.38 | jaytee | there's no feature that does exactly that but someone posted a macro that does call screening like that. It does use the record and playback apps |
| 16:02.56 | patrick-- | i remember it beeing some part of "freepbx" |
| 16:04.30 | [TK]D-Fender | patrick--: And this isn't #freepbx nor is it supported here |
| 16:04.51 | patrick-- | ok, no offence. |
| 16:04.57 | [TK]D-Fender | patrick--: Yes * can do this rather easily... can FreePBX allow you to do it easily? Don't know, don't care. |
| 16:05.26 | jaytee | patrick--, here: http://www.asteriskextras.com/index.php?option=com_content&task=view&id=15&Itemid=2 |
| 16:05.49 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 16:05.56 | patrick-- | thats exactly what i ment :) thank you |
| 16:07.03 | jaytee | you'll still need to adapt the code to fit your dialplan but it's a start |
| 16:09.58 | [TK]D-Fender | slaps jaytee's hand |
| 16:10.03 | [TK]D-Fender | jaytee: DON'T FEED THE ANIMALS! |
| 16:10.10 | jaytee | and pay attention to the encoding, the example uses gsm. for my purposes I'd have to change that to ulaw |
| 16:11.06 | jaytee | [TK]D-Fender, well....if he's running FreePBX he'll be spending the next 6 months trying to get that to work :-) |
| 16:13.25 | x86 | there's some |
| 16:13.43 | x86 | there's some Privacy thing that will record() if there is no CID passed in |
| 16:13.54 | x86 | some built-in app... lemme find |
| 16:13.56 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 16:15.25 | x86 | http://www.voip-info.org/wiki/view/Asterisk+cmd+PrivacyManager |
| 16:16.59 | jaytee | yeah, wow! like that's really up to date :-) |
| 16:17.50 | x86 | hehe |
| 16:18.19 | jaytee | although the book does detail it for 1.4 at least |
| 16:20.16 | beek | Hey jaytee and [TK]D-Fender -- don't you have anything better to do than work on a Sunday? |
| 16:20.25 | beek | realizes that I'm working too.... |
| 16:20.37 | jaytee | who's workin? I'm whackin bad guys in Mafia Wars |
| 16:20.39 | patrick-- | [TK]D-Fender jaytee: ive been running asterisk for a couple of years now. i was just wondering what the native function is called.. nothing else.. didnt mean to cause any stress :) |
| 16:21.07 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-34-194.w86-215.abo.wanadoo.fr) |
| 16:21.20 | jaytee | patrick--, I eat stress for breakfast, usually with a 4 egg omelet and a half a pound of bacon. gives my doc nightmares but screw em! |
| 16:21.36 | patrick-- | right then :D |
| 16:21.41 | patrick-- | thanks for your effort anyways |
| 16:21.46 | [TK]D-Fender | beek: Work? What work? I'm watching BSG, and eating fondue :) |
| 16:22.03 | jaytee | Daybreak Part 2? |
| 16:22.17 | beek | [TK]D-Fender: I just say my first episode of the new season. I have the rest on TiVo, awaiting me. |
| 16:22.22 | beek | s/say/saw/ |
| 16:22.25 | [TK]D-Fender | jaytee: 4x16 Deadlock |
| 16:22.56 | [TK]D-Fender | I have through Daybreak 1/3 on me... |
| 16:23.29 | jaytee | I was chatting with Mark Sheppard last nite (he plays Rolo) about it. He's a funny guy. |
| 16:28.46 | *** join/#asterisk specialist1 (n=research@119.160.105.172) |
| 16:29.05 | specialist1 | hi.. any one needs an asterisk consultant |
| 16:29.09 | *** join/#asterisk Mog (n=mog@c-68-62-218-186.hsd1.al.comcast.net) |
| 16:29.09 | *** mode/#asterisk [+o Mog] by ChanServ |
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| 16:31.43 | [TK]D-Fender | specialist1: Very bad form to whore yourself here.... |
| 16:37.20 | *** part/#asterisk Ericounet (n=Ericoune@ACaen-151-1-34-194.w86-215.abo.wanadoo.fr) |
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| 16:52.06 | Talkradio | i'd suggest learning to form a proper sentence before soliciting work heh |
| 16:57.53 | Pegasus_RPG | I'm trying to get my Snom 300 w/ uLaw to talk to QuteCom w/ Speex via * on the LAN and * drops the call (saying congestion) when QuteCom answers it. Using other codecs in QC work fine (uLaw, GSM) What could be the problem? |
| 17:00.28 | Pegasus_RPG | narrowband, Speex, that is |
| 17:00.36 | [TK]D-Fender | Pegasus_RPG: pastebin the complete failed call attempt from beginning to end w/ SIP DEBUG enabled |
| 17:00.41 | Pegasus_RPG | ok |
| 17:01.26 | [TK]D-Fender | pb |
| 17:01.28 | [TK]D-Fender | ~pb |
| 17:01.29 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 17:02.19 | Pegasus_RPG | http://pastebin.com/d1423f316 |
| 17:03.15 | Pegasus_RPG | For this test, I have speex being the only allowed codec for the QC client |
| 17:03.25 | Pegasus_RPG | (Otherwise it would fall through to another) |
| 17:03.31 | [TK]D-Fender | SIP/2.0 488 Not Acceptable Here |
| 17:03.43 | [TK]D-Fender | Speex = no good |
| 17:04.31 | Pegasus_RPG | argh |
| 17:04.46 | *** part/#asterisk patrick-- (n=patrick@eos.openroot.de) |
| 17:05.11 | Pegasus_RPG | So even though the software says it supports it, it really doesn't? |
| 17:05.24 | [TK]D-Fender | Pegasus_RPG: or you've configured it wrong |
| 17:05.53 | Pegasus_RPG | or that. :) unfortunately, there's no codec config in the client other than the order of codecs |
| 17:05.59 | Pegasus_RPG | Ok, thank you for your time! |
| 17:08.28 | Pegasus_RPG | Oh, one other thing: how can I decrease the bandwidth used by the GSM codec? |
| 17:09.02 | [TK]D-Fender | Pegasus_RPG: You can't |
| 17:09.11 | [TK]D-Fender | Pegasus_RPG: All * codecs are pretty much fixed |
| 17:09.15 | Pegasus_RPG | oh ok |
| 17:09.25 | [TK]D-Fender | Pegasus_RPG: * uses GSM 6.10 @ 13kbps |
| 17:09.31 | mort_gib | Pegasus_RPG: Use another codecs :-) |
| 17:09.41 | [TK]D-Fender | Pegasus_RPG: G.729 is lighter at about 8kbps IIRC |
| 17:09.43 | Pegasus_RPG | REally... I wonder why it's not working acceptably with a 32kbps uplink |
| 17:09.51 | Pegasus_RPG | (on the client) |
| 17:10.04 | [TK]D-Fender | Pegasus_RPG: Because thats the CODEC weight. You have left out UDP overhead |
| 17:10.35 | [TK]D-Fender | GSM + UDP maxes out your upsteam and a tiny bit more |
| 17:11.13 | Pegasus_RPG | ah ok |
| 17:11.22 | Pegasus_RPG | arer there any other codecs that might work? |
| 17:11.34 | [TK]D-Fender | G729 |
| 17:12.40 | mort_gib | Pegasus_RPG: Make sure latency and Jitter is acceptable too... |
| 17:13.04 | Pegasus_RPG | well latency is the pits, but I can deal with that |
| 17:13.09 | [TK]D-Fender | mort_gib: They never are :) |
| 17:13.12 | Pegasus_RPG | (it's a sat connection) |
| 17:13.15 | [TK]D-Fender | And don't forget to sacrifice a goat... |
| 17:13.31 | Pegasus_RPG | haha |
| 17:13.34 | MaliutaLap | [TK]D-Fender: black or wite? |
| 17:13.42 | mort_gib | True TK, and splash the blood on Thor, Freja and Odin |
| 17:13.54 | [TK]D-Fender | MaliutaLap: Mottled for best effect |
| 17:13.55 | Pegasus_RPG | crap, QuteCom doesn't support G.729. Any other Linux softphone recommendations? |
| 17:14.06 | MaliutaLap | [TK]D-Fender: aren't black goats best for SCSI termination? |
| 17:15.05 | [TK]D-Fender | MaliutaLap: I always go with mottled.... your general "all-purpose" sacrifice. |
| 17:15.07 | *** join/#asterisk mazpe (n=mazpe@c-71-196-32-22.hsd1.fl.comcast.net) |
| 17:15.39 | mort_gib | MaliutaLap: Color dosen't matter, the goods you choose do, Choose Odin for SCSI, he has the bad temper that seems to most help on SCSI issues |
| 17:15.48 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 17:17.18 | mort_gib | Freja and Frej are good for SIP/IAX problems |
| 17:17.27 | *** join/#asterisk FuriousGeorge (n=Brian@ool-4354d18c.dyn.optonline.net) |
| 17:17.33 | FuriousGeorge | hey all |
| 17:17.49 | FuriousGeorge | im guessing this should be somewhat obvious, but i cant find the solution: |
| 17:18.18 | FuriousGeorge | i just switched my provider from IAX to SIP and my remote sip clients arent receiving audio now |
| 17:18.45 | FuriousGeorge | checked my ports (5060,8000,10000-20000), have externhost set |
| 17:18.56 | FuriousGeorge | nat=yes in that clients sip.conf... still, nothing |
| 17:19.35 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 17:20.15 | FuriousGeorge | i just dont see how changing the provider could affect the leg between asterisk and my client, which has always been nat |
| 17:21.13 | *** join/#asterisk hi365 (n=hi365@bzq-79-176-69-239.red.bezeqint.net) |
| 17:21.56 | mort_gib | [TK]D-Fender: How much does G729 actually take up, including UPD overhead?? (more or less) |
| 17:22.23 | ricko73 | FuriousGeorge: canreinvite=? |
| 17:22.52 | FuriousGeorge | ricko73: whatever the default is... no? |
| 17:23.00 | [TK]D-Fender | mort_gib: http://www.voip-info.org/wiki-Bandwidth+consumption |
| 17:23.19 | mort_gib | [TK]D-Fender: THanks |
| 17:23.33 | FuriousGeorge | actually its specifically set to know |
| 17:24.11 | FuriousGeorge | *set to "no" |
| 17:24.23 | FuriousGeorge | hmmm, neither party hears each other |
| 17:24.37 | ricko73 | you said you changed providers? |
| 17:24.45 | [TK]D-Fender | FuriousGeorge: And there is no reason I'd trust that you actually set any of this right without seeing it, and a failed call in a pastebin w/ SIP debug for myself... |
| 17:24.59 | ricko73 | some providers use a wider range for rtp than others |
| 17:25.33 | ricko73 | but, [TK]D-Fender speaks the truth. Without seeing more information, anyone is just guessing |
| 17:25.56 | mort_gib | [TK]D-Fender: Wow, never read up on that! I knew overhead added some, but not that much |
| 17:28.58 | FuriousGeorge | sorry ill get a pastebin together |
| 17:29.56 | FuriousGeorge | im wondering if my phone is just broken... now that i think of it why would changing the providers matter? all the phones at the location work. this phone at my house has just stopped working, but nothing has changed that should affect that |
| 17:30.56 | [TK]D-Fender | FuriousGeorge: Yeah only your protocols and all of the configs around it. |
| 17:31.07 | [TK]D-Fender | FuriousGeorge: Now stop with the distractions and show us the problem. |
| 17:31.24 | *** join/#asterisk [intra]lanman (n=intralan@freeswitch/developer/intralanman) |
| 17:31.25 | FuriousGeorge | sorry, brb with that pb |
| 17:32.26 | jaytee | [TK]D-Fender, you'd save yourself alot of time if you just flogged yourself with a knotted rope or chain for 15 minutes instead of spending hours in here :-) |
| 17:34.10 | Pegasus_RPG | lol |
| 17:35.21 | mazpe | on asterisk 1.6.0, how can do i add options to a Dial()? for example moh instead of the ring. option m |
| 17:35.35 | mazpe | i was trying DIAL(SIP/1${EXTEN}@provider|m) |
| 17:35.38 | russellb | mazpe: the same way you do in 1.4 :-) |
| 17:35.45 | russellb | mazpe: except use "," instead of "|" |
| 17:35.54 | russellb | mazpe: you should check out UPGRADE.txt |
| 17:36.11 | russellb | and you need two delimiters instead of 1 |
| 17:37.05 | mazpe | first is for the timeout. |
| 17:37.07 | mazpe | got it |
| 17:37.53 | [TK]D-Fender | russellb: load res_omfgwehavedocs.so :p |
| 17:38.22 | drmessano | res_rtfm.so.what |
| 17:38.26 | mazpe | thanks russellb |
| 17:38.30 | Pegasus_RPG | [TK]D-Fender: But it says "file not found" when I do that. ;) |
| 17:38.31 | russellb | np |
| 17:38.38 | *** join/#asterisk roe (n=roe___@207-172-35-242.c3-0.eas-ubr15.atw-eas.pa.cable.rcn.com) |
| 17:38.49 | [TK]D-Fender | Pegasus_RPG: I'm sure it does for res_clue.so ;) |
| 17:38.50 | FuriousGeorge | [TK]D-Fender: ricko73: http://pastebin.ca/1368199 i verified that it actually has nothing to do with the provider because it doesnt work when i call voicemail either |
| 17:39.13 | Pegasus_RPG | lol |
| 17:39.18 | jaytee | Colonel Mustard, in the Library, with....YO MOMMA!!!! |
| 17:39.38 | FuriousGeorge | so something gets lost between asterisk <nat> wan <nat> sip_phone |
| 17:39.40 | [TK]D-Fender | FuriousGeorge: several errors in there |
| 17:39.57 | [TK]D-Fender | FuriousGeorge: You don't have "nat=yes" for [general] |
| 17:39.59 | FuriousGeorge | [TK]D-Fender: really? |
| 17:40.22 | [TK]D-Fender | FuriousGeorge: register => blah has stuff AFTER IT. These get IGNORED so all your extern settings = meaningless |
| 17:40.34 | [TK]D-Fender | FuriousGeorge: Move them up |
| 17:41.08 | [TK]D-Fender | FuriousGeorge: And you should have "careinvite=no" under [general] as well for safety |
| 17:41.14 | drmessano | jaytee: Followup to your FB comment.. If you opened all the cans, the cat would likely eat all of them and throw up, showing their approval for your giving in, and at the same time, their dislike for all of the cans |
| 17:41.20 | FuriousGeorge | [TK]D-Fender: where do you see nat=yes in general? i actually have it set to no but i truncated that with the comments when i made that pb by accident |
| 17:41.44 | [TK]D-Fender | FuriousGeorge: I DON'T see it. Thats the problem <- |
| 17:41.54 | [TK]D-Fender | ~sipnat |
| 17:41.55 | 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 |
| 17:42.06 | jaytee | drmessano, sounds like you've been "staff" to a cat or two yourself :-) |
| 17:42.43 | FuriousGeorge | [TK]D-Fender: gotcha. also, as far as stuff coming after register, that's how the conf is set up by default |
| 17:43.00 | [TK]D-Fender | FuriousGeorge: "default"? Pardon? |
| 17:43.11 | [TK]D-Fender | FuriousGeorge: vi has no concept of "default" |
| 17:43.14 | FuriousGeorge | [TK]D-Fender: when you make sampledocs, |
| 17:43.18 | [TK]D-Fender | .... |
| 17:43.37 | [TK]D-Fender | FuriousGeorge: Just go fix it all |
| 17:44.54 | FuriousGeorge | i already moved it, and im reloading sip, im just saying that the bit in the sampledocs about ";----------------------------------------- NAT SUPPORT ------------------------ |
| 17:44.54 | rbd | hey guys...I have an asterisk PBX with some cisco 79xx phones, linksys SPA922 and SPA962 phones connected. On the 79xx and 922, outbound calls work fine everytime...with the 962s, I see that RTP is coming from the phone to the asterisk box (it is a nat setup) to the trunk, but not the other way (one way audio). |
| 17:44.54 | FuriousGeorge | " comes after the ;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------ |
| 17:45.12 | rbd | I've tried firmware updates, and all sorts of config changes on the 962s, no luck...any ideas? |
| 17:46.34 | roe | anyone experience different latencies as reported by asterisk based on which SIP hard phone is in use? We see a 50ms difference and at the moment the only difference we can see is an Aastra vs Linksys phone |
| 17:47.41 | FuriousGeorge | [TK]D-Fender: anyway, you were right, moving the section above the register worked |
| 17:58.11 | *** join/#asterisk raden (n=jon@adsl-99-139-235-165.dsl.applwi.sbcglobal.net) |
| 17:59.00 | mazpe | anyone is awared of a2billing having a call duration limitation? for some reason all my calls are been droped at 870secs |
| 17:59.20 | mazpe | around 870secs |
| 17:59.50 | *** part/#asterisk Pegasus_RPG (n=chatzill@cpe-071-076-024-036.sc.res.rr.com) |
| 18:01.49 | drmessano | mazpe: Common sense would tell me something is set wrong |
| 18:02.38 | mazpe | yeah |
| 18:03.00 | mazpe | I just cannot find it anywhere... anything that sets a call duration in a2billing |
| 18:03.58 | mazpe | among many things.. non that seem to seem to be relevant to the issue i get this: |
| 18:04.01 | mazpe | a2billing.php,2: file:Class.RateEngine.php - line:1139 - -> dialstatus : ANSWER, answered time is 870 |
| 18:04.15 | mazpe | and the previous one: |
| 18:04.16 | mazpe | a2billing.php,1: file:Class.RateEngine.php - line:1139 - -> dialstatus : ANSWER, answered time is 869 |
| 18:10.51 | *** join/#asterisk voxter (n=voxter@24.84.56.44) |
| 18:11.07 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) |
| 18:23.30 | mazpe | very strange |
| 18:30.51 | *** join/#asterisk raden (n=jon@adsl-99-139-235-165.dsl.applwi.sbcglobal.net) |
| 18:44.35 | *** join/#asterisk CapriCoRN^80 (n=int@209.8.41.156) |
| 18:52.01 | *** join/#asterisk path_ (n=path_@pc-15-190-86-200.cm.vtr.net) |
| 18:55.34 | mazpe | does asterisk have any other call duration aside from L(x[:y][:z]) ? |
| 18:55.53 | *** join/#asterisk shyam_k (n=user@unaffiliated/shyam-k/x-8459115) |
| 19:05.52 | *** join/#asterisk mrspinx (n=shawn@clamwin/tester/mrspinx) |
| 19:06.36 | mrspinx | Hi |
| 19:07.46 | mrspinx | what works better port fwding, stun, or xtunnels |
| 19:08.34 | [TK]D-Fender | mrspinx: read : |
| 19:08.36 | [TK]D-Fender | ~sipnat |
| 19:08.37 | 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 |
| 19:08.48 | raden | anyone use gafachi or have a place i can get cheap rates on 800 origination with like $2 per channel fee and that can port our existing 800 # ? |
| 19:09.19 | [TK]D-Fender | raden: www.ipkall.com |
| 19:10.59 | raden | how does that help with a 800 # ? |
| 19:14.01 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 19:15.25 | *** join/#asterisk blazomir2 (n=blazko@77.28.221.136) |
| 19:15.55 | blazomir2 | How can I fallback from Disa() if a timeout is reached? |
| 19:16.46 | [TK]D-Fender | blazomir2: You don't. It is too limited. |
| 19:16.57 | [TK]D-Fender | raden: You asked for cheap origination. That's it. |
| 19:18.06 | [TK]D-Fender | blazomir2: Just make your own IVR to provide dialtone. |
| 19:18.49 | raden | [TK]D-Fender, no i asked for cheap 800 # Origination |
| 19:19.16 | [TK]D-Fender | raden: Noone is going to offer you $2/channel for it. Its pretty much always per minute. |
| 19:20.09 | raden | call centric is #2 per channel 0.017 per min just cant port our # |
| 19:20.57 | drmessano | Callcentric is .017? |
| 19:21.02 | *** part/#asterisk CapriCoRN^80 (n=int@209.8.41.156) |
| 19:21.03 | raden | yes |
| 19:21.15 | raden | drmessano, sorry gafachi is |
| 19:21.36 | raden | drmessano, doing way to much research starting to make head spin :) |
| 19:21.38 | drmessano | Callcentric is almost 2 cents per minute |
| 19:21.52 | raden | drmessano, it more than that |
| 19:21.56 | raden | 0.029 |
| 19:21.58 | raden | US |
| 19:22.03 | drmessano | Well, US domestic |
| 19:22.09 | drmessano | Is .0198 |
| 19:22.09 | Kumba_ | I'll sign up with anyone that gives me 1-cent/minute and accepts a 25% ASR :) |
| 19:22.17 | raden | ASR ? |
| 19:22.25 | Kumba_ | and a 3-5 minute ACD |
| 19:22.28 | raden | drmessano, im talking toll free |
| 19:22.33 | drmessano | Ok |
| 19:22.38 | Kumba_ | ASR = Attempted Success Rate |
| 19:22.56 | Kumba_ | ACD = Average Call Duration (in this case) |
| 19:24.02 | raden | anyone here use gafachi ? |
| 19:24.18 | drmessano | I ate some bad gafachi once.. too much garlic and overcooked |
| 19:24.18 | Kumba_ | I'm testing them out |
| 19:24.41 | raden | Kumba_, your opinion so far ? |
| 19:24.53 | drmessano | Swore I would never go back, but that veal mixed with Velveeta was oh so good |
| 19:26.03 | Kumba_ | Ehh, they seem OK so far |
| 19:26.07 | Kumba_ | Just another SIP aggregator? |
| 19:27.46 | raden | kumba ever tested one that really stands out in quality ? |
| 19:29.00 | Kumba_ | Not really stand out in quality, but i've tried some that were horrible |
| 19:29.04 | Kumba_ | broadvoice for one :) |
| 19:29.05 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 19:29.44 | Kumba_ | Most of them are all similar, it's just a matter of how well connected your endpoints are to theirs... |
| 19:29.58 | Kumba_ | the good ones will have a real SBC/Proxy for you to connect to |
| 19:30.13 | raden | broadvox any good ? |
| 19:30.24 | raden | so many choices some are soo expensive |
| 19:30.37 | raden | just need 4 inbound 800 channels on one number 3 outbound unlimited |
| 19:30.52 | raden | i can get all that with call centrix & gafatchi for like $40 a month |
| 19:31.10 | raden | just done like the fact everything done online via support tickets |
| 19:31.26 | raden | do any of these places have like network failure number setups ? |
| 19:31.30 | Kumba_ | try vitelity, i've had good luck with them |
| 19:33.14 | raden | im pretty sure they just resell broadvox |
| 19:33.40 | Kumba_ | more like xo/lvl3 |
| 19:34.36 | Kumba_ | Unless you go with a tier-1 they are all reselling someone... |
| 19:34.51 | Kumba_ | There's only a dozen or so Tier-1's, and even some of them are half-and-half |
| 19:34.56 | raden | who are the tier one providers ? |
| 19:35.31 | Kumba_ | Companies like Level 3, XO, GLobal Crossing, AT&T/Verizon, Qwest, Nuvox, etc. |
| 19:36.52 | raden | the expensive ones in sense |
| 19:37.07 | Kumba_ | The good ones, yes |
| 19:37.07 | raden | how on earth do places offer unlimited nationwide calling for $8.95 then ? |
| 19:37.17 | Kumba_ | Cuse it's not unlimited |
| 19:37.52 | raden | how it not unlimited |
| 19:37.58 | raden | unlimited us calling 8.95 mo |
| 19:38.05 | Kumba_ | You are too much of a consumer if you really need to ask that |
| 19:38.24 | Kumba_ | Unlimited is usually around 20K-minutes/mo |
| 19:38.28 | Kumba_ | they cut you off after that |
| 19:39.16 | raden | well there about 40,000 min in a month soo that ok :) |
| 19:39.21 | j_kroon | if the contract says unlimited you can nail them pretty hard if they do cut you off. |
| 19:39.43 | Kumba_ | No, you have to see what their term definition of unlimited is |
| 19:39.57 | Kumba_ | You assume that the term unlimited is defined ala websters dictionary |
| 19:39.57 | j_kroon | raden, i've got clients doing average call concurrencies of 12 to 16 calls at any given point in time. |
| 19:39.58 | Kumba_ | it's not |
| 19:40.12 | Kumba_ | replace the word unlimited with widget |
| 19:40.15 | j_kroon | oxford's actually :) |
| 19:40.26 | Kumba_ | Cause that's actually correct in this sense... |
| 19:40.40 | raden | j_kroon, i, saying 8.95 per channel unlimited |
| 19:40.40 | Kumba_ | Now, you need to read their contract to see what a widget is |
| 19:40.57 | j_kroon | raden, ah ok. that makes it very, very different. |
| 19:41.38 | drmessano | loves Vonage users shopping for business voip service.. Like a freshmans first day in high school |
| 19:42.13 | raden | drmessano, yeah we wont go there |
| 19:42.21 | raden | they need to stick with there 24.99 |
| 19:42.28 | j_kroon | :). can't wait for the day that i've got sufficient bandwidth, and low enough latencies international to be able to start shopping for intl providers. |
| 19:42.31 | drmessano | I was referring to you, dude.. lol |
| 19:42.51 | raden | drmessano, we have providers now |
| 19:43.44 | raden | just sick of this 30+ per line all lines have to be unlimited or metered for every inbound line you have you get a outbound etc.... just want to find something that fits our business but dont want to get crap |
| 19:43.51 | raden | and support is nice |
| 19:44.03 | raden | NUN would be nice |
| 19:44.29 | drmessano | All decent service is going to be metered |
| 19:45.39 | drmessano | Expecting to dial and get a live human is going to cost $$$$$.. In the long run, being able to submit a priority ticket and get it answered in queue is going to be cheaper for them, which will be passed on to you |
| 19:45.51 | drmessano | You're gonna be paying a lot for "live assistance insurance" |
| 19:46.06 | drmessano | Which, if the company is worth their salt, wont be needed much |
| 19:46.21 | Kumba_ | I shove about 600 concurrent calls through vitelity and have been happy |
| 19:46.24 | Kumba_ | that's my recommendation |
| 19:46.36 | Kumba_ | Tell them James from ViciDial Group sent you... I like the kudos |
| 19:47.15 | raden | lol ok.. |
| 19:47.51 | drmessano | I've just about given up on les.net.. Good god my call quality has been horrid lately |
| 19:51.34 | Kumba_ | well, all SIP providers have some funkiness going on, doesn't matter who it is... but some are inherently more fubar then others :) |
| 19:54.23 | *** join/#asterisk ayeso (n=chatzill@216.65.195.52) |
| 19:54.49 | *** join/#asterisk darkskiez (n=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 19:56.22 | drmessano | Well, I have used les.net for like 2 years now, and this has been horrid lately |
| 20:09.29 | mazpe | anyone recommends a good solutiong for calling cards aside from a2billing? |
| 20:24.55 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 20:25.45 | KavanS | I am trying to adjust a outgoing trunk to append a "1" in front of the area code so it will dial long distance correctly....I am using the macro "trunkdial-failover-0.3" |
| 20:26.05 | KavanS | are there any suggestions on reading so I can adjust per trunk dialing options? |
| 20:30.40 | [TK]D-Fender | KavanS: Its your dialplan, do whatever you want and why does the name you chose for your macro matter? |
| 20:31.04 | KavanS | it's included in the asterisk-gui |
| 20:31.14 | [TK]D-Fender | "thats nice". GUI's not supported here |
| 20:31.15 | KavanS | I thought it would be relevant |
| 20:31.25 | KavanS | ok |
| 20:31.33 | [TK]D-Fender | KavanS: But feel free to go into you macro and jsut shove the 1 where it belings |
| 20:31.42 | [TK]D-Fender | belongs* |
| 20:32.27 | drmessano | I find it VERY hard to believe you cant append a 1 in the GUI somewhere |
| 20:32.32 | drmessano | Thats basic functionality |
| 20:33.49 | [TK]D-Fender | drmessano: But entirely believable that he has not looked at the macro and can't even spot a Dial command change it |
| 20:36.28 | KavanS | ok, I think I found it |
| 20:39.56 | *** join/#asterisk nima0102 (n=nima@91.98.216.108) |
| 20:39.58 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 20:46.15 | beek | [TK]D-Fender: I had a POTS line supplied via a channel bank that the telco hung on the wall. Said telco has been replaced with a Verizon line back to the CO. Asterisk no longer picks up incoming calls on that line, tho' I can see the LED indicate ringing on our Adit600. Am I looking at a GS vs LS issue? |
| 20:50.21 | lanning | ask verizon what they are using, and set the channel bank (and asterisk) accordingly. |
| 20:51.53 | beek | lanning: I have a trouble-ticket in but, of course, they'll get back to me when they get back. I'm not taking calls on that line at the moment but I found it interesting that it wasn't working. |
| 20:51.53 | beek | lanning: My incoming/outgoing calls are via PRI. I just need to use that POTS line for 911 backup. |
| 20:53.10 | lanning | then, you can either wait, or you can experiment. there are not that many choices. |
| 20:53.45 | lanning | my AT&T line at home is LS. |
| 20:54.07 | beek | lanning: I changed things to LS both on the Adit600 and in chan_dadhi.conf -- no joy. I'm expecting Loop Start and will really get involved with it as soon as I get confirmation from Verizon. |
| 21:03.38 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:07.34 | *** join/#asterisk jeff_phillips (n=jeff_phi@209-206-132-52.dyn.centurytel.net) |
| 21:07.38 | jeff_phillips | hi |
| 21:11.32 | jeff_phillips | does anyone know of any ITSPs that offer a DID # that can take both incomming VoIP calls and be able to receive SMS messages from ordinary cell phone networks? |
| 21:12.43 | drmessano | Gizmo5, Google Voice |
| 21:15.15 | jeff_phillips | Google Voice says it isn't even available yet |
| 21:16.56 | drmessano | Will be within a few weeks |
| 21:17.41 | jeff_phillips | i just put my e-mail in to get notified when it is |
| 21:18.10 | drmessano | The SMS stuff works, i've tested it |
| 21:18.24 | drmessano | It actually transcribes your voicemails too |
| 21:18.32 | jeff_phillips | i'm looking for a good way of receiving SMS messages on numerous different DIDs that I can have voice calls routed through asterisk |
| 21:18.34 | drmessano | With VERY good accuracy |
| 21:18.57 | jeff_phillips | Yeah I just saw that on their list of features -- right now I'm paying $6.99 a month to YouMail just for the voice transcription feature on my own cell |
| 21:19.43 | drmessano | I had a spam voicemail that it transcribed 100 or so words of, and it missed maybe 2 |
| 21:19.58 | drmessano | I was shicked |
| 21:20.03 | drmessano | shocked too |
| 21:21.19 | jeff_phillips | hmm |
| 21:21.28 | jeff_phillips | gizmo only lets me buy one number per week. that won't work |
| 21:21.53 | jeff_phillips | looking for a wholesaler I guess |
| 21:21.59 | jeff_phillips | most of the DID providers don't give a rip about SMS |
| 21:22.17 | drmessano | You can get Gizmo business |
| 21:22.21 | drmessano | Buy as many as you like |
| 21:22.43 | jeff_phillips | yeah I'm looking at the gizmo business page and it says I can add as many #'s as I wish but am limited to only buying one new number per week |
| 21:23.23 | drmessano | Hmmm I dont recall that bit.. weird |
| 21:23.51 | jeff_phillips | "You can add as many Call In numbers to your account as you like; however, you are currently limited to add a new number once a week." from http://gizmo5.com/pc/network/callin-numbers/ |
| 21:24.50 | drmessano | That applies to standard accounts |
| 21:26.20 | jeff_phillips | funny i clicked business and it took me to that page |
| 21:27.49 | drmessano | The gizmo business stuff on the site was thrown together and doesnt always link correctly |
| 21:29.22 | jeff_phillips | well thanks, i'll look into what they have to offer |
| 21:45.02 | *** join/#asterisk keith4_ (n=keith@216-164-144-176.c3-0.eas-ubr9.atw-eas.pa.cable.rcn.com) |
| 21:45.08 | roe | anyone experience different latencies as reported by asterisk based on which SIP hard phone is in use? We see a 50ms difference and at the moment the only difference we can see is an Aastra vs Linksys phone |
| 21:46.10 | *** join/#asterisk joobie (n=joobie@mx01.anric.com.au) |
| 21:47.22 | bkw_ | roe: yes some phones will respond slower to the pings/options when in use |
| 21:47.50 | bkw_ | remember its not latency but response time on qualify |
| 21:48.22 | roe | bkw_, nothing documented though? I'd like to put this matter to bed with some assuredness. |
| 21:48.36 | bkw_ | its the phone responding slower when in use |
| 21:48.38 | bkw_ | nothing more |
| 21:49.00 | bkw_ | its not pinging the phone |
| 21:49.47 | [TK]D-Fender | Polycom's have a low OPTIONS priority. |
| 21:50.06 | bkw_ | polycom also supports the PING method |
| 21:50.08 | [TK]D-Fender | Every make is different and isn't a solid measure of much |
| 21:50.20 | roe | ok, thanks for the info |
| 21:51.23 | *** join/#asterisk hepta (i=cso@78.156.12.251) |
| 21:52.30 | hepta | how do you say "to call someone" in a way that unambiguously refers to telephony? "dial someone"? |
| 21:53.16 | hepta | or is the word "dial" archane? |
| 21:53.24 | bkw_ | "to place a telephone call" |
| 21:53.26 | bkw_ | :P |
| 21:53.31 | keith4_ | depends where you are |
| 21:54.10 | hepta | would "dial contact" be a good text for a GUI button? |
| 21:54.25 | drmessano | Place Call |
| 21:54.30 | drmessano | or Call |
| 21:54.33 | keith4_ | i vote for "ring ring" |
| 21:54.35 | Chainsaw | hepta: Our phones just state "Dial" |
| 21:54.43 | hepta | lets say US english. ok, so "place call to contact"? |
| 21:54.54 | Chainsaw | hepta: (For recently placed call lists, missed calls, personal speed dials etc) |
| 21:55.04 | hepta | Chainsaw: in GB, or US? |
| 21:55.12 | drmessano | hepta: Too many words |
| 21:55.14 | Chainsaw | hepta: GB |
| 21:55.25 | drmessano | hepta: Call Contact |
| 21:55.35 | Chainsaw | I wouldn't go overboard with it, at any rate. If I see a button that states 'call' and it's next to a phone number, that's obvious to me. |
| 21:55.41 | drmessano | if the entry is a "contact" |
| 21:56.04 | Chainsaw | You could always have a little phone icon in front of the word dial. |
| 21:56.04 | hepta | yes. thanks. |
| 21:56.20 | Chainsaw | (Or call, for that matter) |
| 21:56.33 | drmessano | If youre looking at a contact list, and the button is off to the side, "Call Contact" would work, and what Chainsaw said, if its next to a number, "Call" is more than enough |
| 21:57.21 | hepta | unless people think thats for calling the person something else, hehe |
| 21:57.41 | hepta | call me Back. |
| 21:57.43 | hepta | James Back |
| 21:58.26 | hepta | just call me back ... |
| 21:58.51 | hepta | ok, "back", now ... |
| 21:59.25 | joobie | guys is there a way i can disconnect / reconnect my pennytel sip connection asterisk is holding? |
| 21:59.33 | joobie | i tried restarting asterisk, duno if that worked.. doesnt look like it |
| 21:59.36 | joobie | preferably from console |
| 22:08.49 | hepta | do people in the US think "dial" sounds funny? |
| 22:10.14 | jeff_phillips | "dial me back" would sound funny |
| 22:10.27 | jeff_phillips | "what number do I dial?" sounds perfectly normal |
| 22:12.24 | hepta | jeff_phillips: oh, so its _not_ good to say "dial contact" then |
| 22:13.03 | jeff_phillips | hepta: No, that's fine |
| 22:13.20 | jeff_phillips | Its hard to explain |
| 22:13.47 | hepta | ok. conclusion is "dial contact" is fine, and does not sound funny to anybody |
| 22:14.05 | jeff_phillips | it's okay for a person to tell a device to dial. But for me to tell you to dial joe, it sounds wierd. You'd say call joe. |
| 22:14.23 | jeff_phillips | yeah a dial contact option is fine |
| 22:14.56 | hepta | ok. a machine can be told to dial, not a human |
| 22:15.12 | jeff_phillips | yeah pretty much. We'd all "get it", but we'd think someone's not from around here |
| 22:15.41 | jeff_phillips | gotta run, ttyl |
| 22:15.51 | hepta | ok. thanks! |
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| 23:21.25 | tompaw | HI. |
| 23:22.43 | tompaw | Guys, I have migrated from 1.6.0.1 to 1.6.0.6 and now a strange thing happens. My GSM phone doesn't recognize the call's been answered. |
| 23:22.55 | tompaw | So when I dial 1234p5678, the '5678' isn't sent. |
| 23:23.07 | tompaw | It takes around 15 seconds (!) for it to realize it's on. |
| 23:25.29 | tompaw | rfc2833compensate=yes << that was the solution! |
| 23:25.38 | tompaw | it should be written in a bold huge font somewhere. |
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| 23:56.55 | lesouvage | perhaps ot but bayhamsystems stops with the option to set the sender and instead there is a uk phone number. Cany anybody suggest an alteranative that can be used from a linux box, asterisk system? |
| 23:57.17 | lesouvage | Sorry, this is about sms messages |