| 00:00.18 | Seba1111 | hi |
| 00:00.36 | Seba1111 | I have a question about smp_affinity and digium |
| 00:00.44 | Seba1111 | can anyone help me? |
| 00:02.51 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 00:07.11 | *** join/#asterisk oadaeh (n=jason@wsip-24-234-160-51.lv.lv.cox.net) |
| 00:08.51 | Delta239 | Blafasel: yes... but for example my dial plan is exten => _1XXXXXXXXXX,1,Dial(Sip/${EXTEN}@dialout,60,tTo); |
| 00:09.04 | *** join/#asterisk topping (n=topping@207.47.6.136.static.nextweb.net) |
| 00:09.09 | Delta239 | where should i put the setcallerid thing? |
| 00:09.19 | Un1x | delta239 |
| 00:09.20 | Un1x | exten => _NXXNXXXXXX,1,Dial(1${splitinfinity}/${EXTEN}) |
| 00:09.30 | Un1x | change splitfinity to whatever you have set in sip.conf |
| 00:09.38 | Delta239 | ok |
| 00:10.06 | Faithful | znoG: Does busy detect help it? |
| 00:10.10 | Delta239 | mmmm kind of different of what i have |
| 00:10.23 | *** join/#asterisk benjk_ (n=benjamin@f8a01-0357.din.or.jp) |
| 00:10.37 | Blafasel | Delta239: Why don't you set the callerid at prio 1 and dial at prio 2? |
| 00:10.48 | Blafasel | Read: Set it before you call out? |
| 00:10.53 | Delta239 | ahhhhh |
| 00:10.56 | Delta239 | let me see :D |
| 00:10.57 | dlynes_laptop | Un1x, wtf? |
| 00:11.05 | Delta239 | so... i just go |
| 00:11.14 | znoG | Faithful: might do |
| 00:12.25 | Delta239 | exten => _1XXXXXXXXXX,1,SetCallerID("Delta" <2222222222>[|a]); ??? |
| 00:12.38 | Delta239 | do i need that [|a] there? |
| 00:13.24 | benjk_ | do you have PRI? |
| 00:13.26 | ManxPower | Delta239, um, [means_optional} |
| 00:13.31 | Blafasel | Huh? Look at the voip-info.org wiki and at show function CALLERID |
| 00:13.33 | ManxPower | you don't put the [] in |
| 00:13.45 | Blafasel | I think SetCallerID is deprecated |
| 00:13.52 | ManxPower | Delta239, I see you suffer from a common Asterisk disease. |
| 00:14.00 | ManxPower | It's called Excessive Quote Syndrome. |
| 00:14.21 | ManxPower | Fortunatly, some devices refuse to talk to devices and people suffering from EQS. |
| 00:14.33 | ManxPower | Here's a hint: DON'T USE QUOTES IN CALLERID |
| 00:15.03 | benjk_ | if you don't have PRI, you won't need the |a, its for setting ANI |
| 00:15.43 | ManxPower | Blafasel, Did you ask Delta239 what version of Asterisk he/she is using? |
| 00:15.59 | ManxPower | SetCallerID works in all current released versions of Asterisk |
| 00:16.13 | Delta239 | well |
| 00:16.21 | Delta239 | if i go to the cli |
| 00:16.41 | Delta239 | gives me asterisk version 1.2.9.1 |
| 00:16.41 | Blafasel | ManxPower: No, I fear I didn't due to lack of asterisk knowledge and drinking some white wine at 2:15 am.. |
| 00:16.57 | Blafasel | And for that version it's deprecated, afaik ;) |
| 00:17.08 | Blafasel | I didn't say "doesn't work" anywhere. |
| 00:17.09 | ManxPower | Delta239, then you can use SetCallerID, but it will be removed in 1.4 |
| 00:17.33 | Blafasel | And since he didn't use it before I just wanted to suggest to look at CALLERID |
| 00:17.35 | ManxPower | Perhaps a read of README.upgrade (I think that's the file name) would be a good place to start. |
| 00:17.35 | Delta239 | ok thanks |
| 00:17.39 | Delta239 | so i go |
| 00:17.49 | Delta239 | exten => _1XXXXXXXXXX,1,SetCallerID("Delta" <2222222222>); |
| 00:17.58 | *** join/#asterisk snL20 (n=irssi@host-81-191-63-18.bluecom.no) |
| 00:18.10 | Delta239 | exten => _1XXXXXXXXXX,2,Dial(Sip/${EXTEN}@dialout,60,tTo); |
| 00:18.16 | Delta239 | right? |
| 00:18.23 | benjk_ | is this for NANPA numbers? |
| 00:18.46 | dlynes_laptop | Delta239, looks fine to me |
| 00:18.54 | Delta239 | ok |
| 00:19.01 | *** join/#asterisk lo_tek (n=tone_loc@user-1121l4v.dsl.mindspring.com) |
| 00:19.01 | Delta239 | thanks ill see if it works :D |
| 00:19.18 | benjk_ | the format for NANPA numbers is 1NXXNXXXXXX |
| 00:19.31 | dlynes_laptop | benjk_, not for caller id it isn't |
| 00:19.40 | dlynes_laptop | benjk_, NANPA caller id doesn't use the '1' in front |
| 00:19.56 | benjk_ | the matching pattern is 1NXXNXXXXX anyway |
| 00:20.11 | dlynes_laptop | benjk_, yeah |
| 00:20.31 | benjk_ | you can then strip off the leading 1 in the argument string (using EXTEN:1) |
| 00:22.34 | *** join/#asterisk niteowldave (n=dave@202.171.182.120) |
| 00:22.35 | benjk_ | also it depends where you are calling |
| 00:23.12 | benjk_ | if you call overseas and you set your caller ID to a 10 digit number, I would consider that rude |
| 00:24.15 | robl^ | what if you call internationally and set you caller ID to "666"? ;-) |
| 00:24.22 | Blafasel | Hmm.. Interesting. What happens with a callerid of 1234 for overseas calls? |
| 00:24.31 | benjk_ | because it means that the rest of us will need to do extra work to second guess that your incoming call might be from the US and add the leading 1 back to the incoming caller ID |
| 00:24.39 | Blafasel | All my mobile devices prepend the callerid with a 0 or +49 anyway |
| 00:25.01 | benjk_ | if you come in here with caller ID 666 you will get PRI_CAUSE=1 |
| 00:25.55 | Blafasel | If I try caller ID 666 it gets delivered as 0666 or +49666, depending on the device at the end of the call.. Not that I tried to do it.. :-) |
| 00:26.10 | benjk_ | and since phone spam is picking up, more people will look at caller ID |
| 00:26.20 | benjk_ | and trash the call if it looks odd |
| 00:26.28 | robl^ | Blafasel: of course not! NONE of us would try that |
| 00:27.04 | Blafasel | benjk_: I haven't heard about phone spam yet.. Any hints where I could read about it? |
| 00:27.20 | benjk_ | well its called telemarketers |
| 00:27.45 | benjk_ | and some telemarketers have started using recorded messages |
| 00:28.26 | Blafasel | Oh - I know the first party, but I've yet to meet the second.. ;) |
| 00:29.05 | benjk_ | some of my customers now ask for all incoming US caller IDs to be checked and rejected if the check turns out negative |
| 00:29.39 | benjk_ | and they also ask for PRI_CAUSE=1 to be given as a reject reason |
| 00:29.49 | Blafasel | Something like a sender_verify for smtp addresses? |
| 00:29.58 | benjk_ | which means to the calling party this will be like the number is not in service |
| 00:30.15 | benjk_ | because they will get the "not in service" announcement from their local phone company |
| 00:30.20 | dlynes_laptop | benjk_, why would that be rude? |
| 00:30.24 | Blafasel | Ah.. Thanks for the translation PRI -> english |
| 00:30.39 | dlynes_laptop | benjk_, oh...nvm |
| 00:31.12 | benjk_ | it is the same kind of rude that the French don't appreciate if you walk in there and expect everybody to speak English without making any effort to at least say "Bonjour" |
| 00:31.51 | benjk_ | Blafasel, PRI == Primary Rate ISDN |
| 00:32.11 | benjk_ | PRI_CAUSE is an asterisk variable you can set before hanging up a call |
| 00:32.13 | Blafasel | Yep, I was talking about the PRI_CAUSE=1 thing |
| 00:32.42 | benjk_ | which if you are on a PRI or BRI circuit will send 1 as a cause code back to your telco |
| 00:33.45 | benjk_ | the causes are defined in Q.931, they are standardised throughout ISDN and SS7 (which includes mobile telephony), 1 means number not in service |
| 00:33.48 | Blafasel | Sounds neat - but I'm a bloody newbie and not using PRI |
| 00:34.00 | benjk_ | 21 means "called party rejected call" |
| 00:34.07 | Blafasel | Ah - now ss7 is more interesting ;) |
| 00:34.10 | benjk_ | you may not use it |
| 00:34.10 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:34.40 | benjk_ | but if the party you are calling uses it and they don't like your caller ID and reject the call with PRI_CAUSE = 1, then you get "number not in service" |
| 00:35.00 | benjk_ | if you are lucky they will reject with 21 |
| 00:35.10 | Blafasel | Really nice for a better ex-gf matching ;) |
| 00:35.20 | benjk_ | in which case you will be able to find out that the call was rejected |
| 00:35.46 | *** part/#asterisk snL20 (n=irssi@host-81-191-63-18.bluecom.no) |
| 00:36.35 | benjk_ | if you know what you are doing you could actually send 22 (number changed) and have the number of your local McDonalds announced as the new number ;) |
| 00:37.21 | *** join/#asterisk eBody (n=ehernand@207.71.51.162) |
| 00:37.23 | eBody | in sip.con what should callerid= be ?? it was "=Unknown" but after taking that out all my calls are "Private" |
| 00:37.25 | robl^ | benjk_: I just sue zapteller and a few choice sound files for that. :-) |
| 00:38.05 | benjk_ | rob^, that won't do nothing to Japanese phone/fax spam bots |
| 00:38.17 | benjk_ | but PRI_CAUSE = 1 will |
| 00:38.34 | *** part/#asterisk Seba1111 (n=Sebastia@r200-40-68-22.ae-static.anteldata.net.uy) |
| 00:39.13 | robl^ | benjk_: hrmm.. don't think that works with IAX or SIP trunks though |
| 00:39.17 | benjk_ | there is this company in Osaka, they are a commercial spam faxing enterprise |
| 00:39.55 | dlynes_laptop | benjk_, they're all over the place here |
| 00:40.15 | benjk_ | callerids usually start with +81 6 6222 and 7222 |
| 00:40.30 | benjk_ | they try to send faxes all day long |
| 00:40.57 | benjk_ | the only way to shut them up for a week is to actually connect a fax machine and receive the fax |
| 00:41.19 | benjk_ | the only way to permanently shut them up is to send PRI_CAUSE = 1 |
| 00:42.19 | benjk_ | because that looks to their spam bot like the number is out of service and so they remove it from their database |
| 00:43.54 | benjk_ | rob^, it doesn't matter if you get your DIDs by SIP or IAX, at some point in the chain, those numbers are PRI or SS7 and that's where you need to reject them with a proper Q.931 cause code |
| 00:43.56 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 00:44.34 | n9urk | What is the new name of Asterisk Management Panel? |
| 00:44.45 | robl^ | benjk_: right.. but if my trunk is IAX.. and my providers is IAX to PRI, I can't pass PRI_CAUSE via my IAX trunk to the provider |
| 00:44.56 | benjk_ | probably not |
| 00:45.04 | benjk_ | but you could ask your provider to do it for you |
| 00:45.25 | Blafasel | Hmm.. I'd like to see any docs on how to do that with chan_ss7.. |
| 00:45.27 | benjk_ | it will be in their best interest because their capacity is being used up by spam |
| 00:46.31 | benjk_ | SS7 uses the same Q.931 codes, quite possibly it uses libpri |
| 00:46.48 | benjk_ | quite likely it checks PRI_CAUSE |
| 00:46.49 | Qwell | Dr-Linux: He says you should be fine |
| 00:47.07 | Blafasel | I didn't find any references regarding chan_ss7, but I'll give it a try tomorrow |
| 00:47.08 | Qwell | "good luck enforcing US trademark in .pk" |
| 00:48.11 | benjk_ | Qwell, if you have deep pockets, it should be possible, but try China, not even Bill Gates can do it ;) |
| 00:49.14 | eBody | hye guys, what would call bad static or interupts to a tdm2400 with calls over the PSTN??? |
| 00:49.35 | *** join/#asterisk colinm_ (n=colol@VDSL-130-13-10-75.PHNX.QWEST.NET) |
| 00:50.09 | benjk_ | eBody, I'd call that "I told you so" |
| 00:50.28 | eBody | lol why's that benjk_ |
| 00:50.37 | benjk_ | because analog just sucks |
| 00:50.47 | eBody | no choice though :-/ |
| 00:51.08 | benjk_ | why's that? |
| 00:51.50 | eBody | the business i'm with is still under contract with their telco, so we had to use a tdm_card if we really wanted to use asterisk |
| 00:52.07 | eBody | but calls continue to be choppy when both people talk. |
| 00:53.28 | Cresl1n | eBody: are you using the aggressive suppressor on the echo canceler? |
| 00:53.49 | eBody | i don't think so. |
| 00:54.16 | Cresl1n | that'll do it |
| 00:54.26 | eBody | how do i do that? |
| 00:54.29 | Cresl1n | but you have to explicitly enable it as well |
| 00:54.45 | Cresl1n | so it's probably not enabled |
| 00:54.56 | *** join/#asterisk Ahnung (n=symphony@139-25.tr.cgocable.ca) |
| 00:55.34 | eBody | Cresl1n, is it a flag in a .conf?? |
| 00:55.43 | Cresl1n | no, no |
| 00:55.51 | Cresl1n | it's probably not your problem then |
| 00:56.01 | *** part/#asterisk colinm_ (n=colol@VDSL-130-13-10-75.PHNX.QWEST.NET) |
| 00:56.23 | Ahnung | hi, my manager.conf is existing and seems to be configured correctly. However, when I reload the configuration, it says that it is unable to open the manager.conf file and that the call management is thus disabled |
| 00:56.36 | Ahnung | any ideas? |
| 00:56.40 | Snake-Eyes | is softhangup() ment to hang up a call right away ? |
| 01:00.25 | benjk_ | eBody, I'd be surprised if the telco didn't like a customer converting an analog contract into a PRI equivalent |
| 01:00.43 | benjk_ | even if only a fractional PRI |
| 01:00.47 | eBody | how would they know? |
| 01:01.04 | benjk_ | call them up and ask |
| 01:02.57 | anonymouz666 | Cresl1n: how are the libss7? |
| 01:03.09 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
| 01:03.12 | Cresl1n | anonymouz666: ???? |
| 01:03.43 | Cresl1n | it's free for people to start evaluating now :-) |
| 01:05.04 | *** join/#asterisk ben_d (n=ben@cpe-66-67-129-4.rochester.res.rr.com) |
| 01:07.54 | *** part/#asterisk Ahnung (n=symphony@139-25.tr.cgocable.ca) |
| 01:07.57 | anonymouz666 | yeah, sure |
| 01:08.22 | Qwell | Cresl1n: I just knew somebody was gonna ask, if I didn't say something, heh |
| 01:08.43 | Cresl1n | Qwell: that's cool, thanks for catching that :-) |
| 01:08.59 | Delta239 | mmm still shows my company number.. |
| 01:09.13 | Qwell | Cresl1n: very cool though.. I bet you're excited |
| 01:09.26 | Cresl1n | Qwell: like unbelievably excited :-) |
| 01:09.31 | benjk_ | Delta, how are you dialing out? |
| 01:09.40 | Cresl1n | I've been working on this project kind of "undercover" for about a year now |
| 01:09.46 | Qwell | Cresl1n: yeah.. |
| 01:09.47 | benjk_ | is it SIP, IAX, analog? |
| 01:09.57 | Cresl1n | it'll be nice to get some people starting to use it |
| 01:11.24 | benjk_ | Delta, if you are dialing out via analog, you can't set caller ID anyway, if you are using an upstream SIP/IAX provider, they may ignore the caller ID you send |
| 01:11.44 | anonymouz666 | Cresl1n: is it possible to make something like fxotune to X100P card? |
| 01:12.44 | Delta239 | i think is because i had the <2222222222> instead of SetCallerid(2222222222); no??? |
| 01:12.54 | Qwell | Don't use SetCallerID anymore |
| 01:13.02 | Qwell | Set(CALLERID(num)=1234567) |
| 01:13.19 | Delta239 | ok will try... |
| 01:13.31 | Delta239 | do i have to restart asterisk to? with reload on the cli? |
| 01:13.41 | Qwell | just extensions reload |
| 01:13.54 | *** join/#asterisk tuck3r (n=tuck3r@unaffiliated/tuck3r) |
| 01:13.55 | Delta239 | ok |
| 01:13.58 | Delta239 | thanks |
| 01:14.10 | Cresl1n | anonymouz666: I don't think so. It depends on the line interface part that is used. |
| 01:14.21 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 01:15.39 | *** join/#asterisk mog (n=mogorman@gateway.digium.com) |
| 01:15.39 | *** mode/#asterisk [+o mog] by ChanServ |
| 01:16.13 | Delta239 | still not working |
| 01:16.21 | Qwell | then your provider doesn't allow it |
| 01:16.23 | Delta239 | i will deal with this tomorrow :D |
| 01:16.26 | Delta239 | im using telasip |
| 01:16.38 | Delta239 | i dont know i was able to before |
| 01:17.05 | Delta239 | i dont know if they restricted that |
| 01:17.34 | Delta239 | see ya guys tomorrow |
| 01:26.20 | eBody | anyone know about fxotune? |
| 01:26.27 | watchy2 | what about it |
| 01:27.08 | eBody | why i wouldn't have an fxotune.conf |
| 01:27.19 | watchy2 | it aint around by default |
| 01:27.20 | _Vile | touch fxotune.conf |
| 01:27.28 | watchy2 | it only shows up after you run fxotune -i |
| 01:27.37 | watchy2 | why do you need it? |
| 01:28.20 | eBody | just looking at it, i have a tdm2400 |
| 01:28.22 | eBody | w/ fxo mods |
| 01:28.44 | watchy2 | ah |
| 01:28.50 | watchy2 | run ./fxotune -i 5 |
| 01:29.00 | watchy2 | ill take awhile depending on how many fxo ports u got |
| 01:29.08 | watchy2 | but does a tdm2400 need fxotuned? |
| 01:30.07 | eBody | don't know? |
| 01:30.17 | watchy2 | me either |
| 01:30.22 | watchy2 | i know you use it on tdm400s |
| 01:36.46 | [hC] | is it reccomended to run on all fxo ports, regardless of country you're in? |
| 01:41.38 | *** join/#asterisk Katty (n=Administ@dialup-4.244.183.23.Dial1.StLouis1.Level3.net) |
| 01:44.22 | Katty | wow, so quiet. |
| 01:46.27 | *** join/#asterisk bkw_ (n=brian@asterisk/friend-and-developer/bkw) |
| 01:50.58 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:51.12 | paolob | hi guys! Is t.38 supported in asterisk? |
| 01:51.28 | Katty | bkw_: hi (= |
| 01:51.30 | paolob | (t.38 is for fax communication) |
| 01:51.34 | Katty | and bye apparently, heh |
| 01:56.57 | ariel_ | hello everyone |
| 01:57.13 | ariel_ | how is everyone doing tonight? |
| 01:57.34 | *** join/#asterisk tempest1 (n=asf@adsl-153-43-12.chs.bellsouth.net) |
| 01:58.48 | brookshire | good? |
| 01:59.42 | ariel_ | hello file |
| 01:59.46 | brookshire | the bingo of course! |
| 01:59.49 | file | hola |
| 02:00.02 | file | brookshire: are you... at your desk?!? |
| 02:00.13 | brookshire | yes |
| 02:00.16 | file | lame |
| 02:00.21 | brookshire | ooo.. that's a word |
| 02:00.33 | *** join/#asterisk tengulre11 (n=tengulre@222.90.66.156) |
| 02:00.39 | file | brookshire: I'm gonna do mine tomorrow |
| 02:00.48 | brookshire | i had to update it |
| 02:00.53 | brookshire | with more words |
| 02:01.04 | tengulre11 | Hi,all |
| 02:01.07 | brookshire | hi |
| 02:01.20 | file | caller id spoofing! |
| 02:01.24 | tengulre11 | I can not hear any voice from asterisk. why? |
| 02:01.35 | tengulre11 | I m using X-Lite! |
| 02:01.41 | brookshire | firewall? |
| 02:01.46 | ariel_ | nat |
| 02:01.47 | file | tengulre11: is NAT involved anywhere? |
| 02:01.51 | tengulre11 | yes! |
| 02:01.57 | tengulre11 | file: yes |
| 02:02.01 | file | we are psychic |
| 02:02.01 | Katty | file: are you at cluecon? |
| 02:02.04 | tengulre11 | it have NAT |
| 02:02.08 | file | Katty: no m'am |
| 02:02.09 | brookshire | ooo.. NAT is a word :) |
| 02:02.11 | Katty | file: k |
| 02:02.18 | Katty | brookshire: are you at cluecon? |
| 02:02.20 | ariel_ | ~nat |
| 02:02.31 | jbot | nat is, like, Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 02:02.32 | file | I am sitting on my bed, pondering going for a walk and pondering calling brookshire at his desk so as to bother him |
| 02:02.32 | Netgeeks | hrm, anyone give me a quick clue on setting codec preference in a sip peer entry? |
| 02:02.32 | tengulre11 | anybody know why? |
| 02:02.37 | trelane | file, conference in trelane? |
| 02:02.43 | Katty | file: why not call my desk and leave me a voicemail? )= |
| 02:03.01 | file | if it talked back I would be scared |
| 02:03.08 | ariel_ | tengulre11, because you don't have your network router configured and asterisk correctly. |
| 02:03.27 | tengulre11 | ariel_: how to fix? |
| 02:03.28 | *** join/#asterisk bkw_ (n=brian@asterisk/friend-and-developer/bkw) |
| 02:03.51 | brookshire | netgeeks: like what do you want to do |
| 02:04.38 | ariel_ | ~ docs |
| 02:04.41 | jbot | extra, extra, read all about it, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 02:04.51 | *** join/#asterisk CajunTech (n=Anyone@24-119-86-82.cpe.cableone.net) |
| 02:05.21 | tengulre11 | ariel_: can you give me a simple answer, I have no time to read docs now , pls! |
| 02:05.31 | ariel_ | tengulre11, you need to setup your externip and localnet settings look it up on the wiki. Also your going to have to do some port forwading on your fw to your asterisk box. |
| 02:05.33 | CajunTech | Hello everyone. I have a quick question: I am extending an application (in Java) to call out and play a .wav file to the remote end, then hang up. Do I need a full blow asterisk installation to do this or is that overkill? |
| 02:05.45 | ariel_ | tengulre11, nat has not simple reply |
| 02:05.57 | ariel_ | it depends on your network and many factors |
| 02:06.12 | tengulre11 | ariel_, Oh!! |
| 02:06.14 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 02:06.22 | brookshire | Cajun: you can do that with asterisk, why write a whole application in java to do it |
| 02:06.27 | ariel_ | tengulre11, I wish there was a simple reply. |
| 02:06.44 | tengulre11 | ariel_, thank you very much! |
| 02:06.52 | Netgeeks | brookshire: never mind, my brain is mush today, the problem is not what I thought it was |
| 02:06.53 | Katty | hi ariel! |
| 02:06.58 | Katty | hi Netgeeks! |
| 02:07.13 | tengulre11 | ariel_, let me try it now! |
| 02:07.17 | CajunTech | Brookshire: Oh, I didn't know that. So I could have, say, PHP script pass a phone number to the server and it would call out? |
| 02:07.21 | Netgeeks | Good evening Katty! Long time no chat.... |
| 02:07.26 | Katty | Netgeeks: i know! i've been so busy... |
| 02:07.27 | ariel_ | Katty, hope your doing fine. |
| 02:07.32 | *** join/#asterisk dlynes_ (n=dlynes@S01060015e91356bb.vc.shawcable.net) |
| 02:07.37 | brookshire | CajunTech: it's called a call file |
| 02:07.39 | Katty | ariel_: yeah, pretty good. works been keeping me really busy... |
| 02:07.44 | Katty | ariel_: and i have a new boyfriend. |
| 02:07.57 | Katty | ariel_: that always eats up free time. |
| 02:08.05 | Netgeeks | A new boyfriend can consume alot of time! |
| 02:08.07 | CajunTech | Ahh, thanks brookshire. I'm REALLY new to Asterisk. I'll go research it out. |
| 02:08.59 | ariel_ | Katty, wow |
| 02:09.26 | ariel_ | I have been swapped with 2 jobs... ;-( taken up my time right now away from my baby girl. |
| 02:09.28 | rowter | does ast_codec_get_samples is also found on asterisk 1.0.9? am trying to locate it |
| 02:09.59 | Katty | ariel_: )= |
| 02:10.42 | Netgeeks | It seems to me that VoIP work has exploded in the last few months. For a while I was scraping for work, and now I have three full time jobs :( I need a clone |
| 02:10.55 | Katty | Netgeeks: i hear you... |
| 02:11.04 | Katty | Netgeeks: that's basically why i'm not on irc much anymore. |
| 02:11.14 | Katty | Netgeeks: hell, i'm not even at my desk anymore. |
| 02:11.22 | ariel_ | Netgeeks, no clones allowed. More money charge more |
| 02:11.44 | Netgeeks | hehe, I'm moving back to the SF Bay area soon too for one of the jobs, and i have to find two smart VoIP people to pick up for me in the other two jobs |
| 02:12.29 | *** join/#asterisk Iam8up|lpy (n=iam8up@user-0cdv282.cable.mindspring.com) |
| 02:12.40 | ariel_ | Netgeeks, wow, funny thing is I have about 3 good customers out in the bay area.... And I am in the other side of the states. |
| 02:13.07 | Netgeeks | haha, I have 5 major clients in New York, and i'm currently in Oregon... nuts, eh? |
| 02:13.27 | Snake-Eyes | Every time I use softhangup() it doesnt seem to hang up other call/s on the trunk, have tried exten => s,1,SoftHangup(SIP/trunk-sx|a) and exten => s,1,SoftHangup(SIP/trunk-sx-1) |
| 02:13.34 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 02:13.41 | ariel_ | Netgeeks, yes nuts... |
| 02:14.00 | Iam8up|lpy | if i'm wanting to start up a brand new asterisk box...what distro would you guys active atm suggest for me to start with? i've used centos 4.x for a while but only for things like apache, mail services, mysql, etc - i've got a little experience with debian's ubuntu, but not too much command line wise |
| 02:14.15 | ariel_ | Cent OS |
| 02:14.19 | ariel_ | 4.3 is great |
| 02:14.21 | Netgeeks | whichever one you are most familiar with, I'd say |
| 02:14.36 | ariel_ | Netgeeks, he did CentOS |
| 02:14.39 | Iam8up|lpy | so rhel, debian, suse, anything..doesn't much matter? |
| 02:14.50 | Iam8up|lpy | all kernels work just fine? what about the timing modules? |
| 02:14.52 | shmaltz | lam8up|lpy, whatever you like but not CentOS |
| 02:14.58 | bkw_ | I got to meet one of the main CentOS guys here at Cluecon |
| 02:15.05 | bkw_ | another is arriving tomorrow |
| 02:15.11 | Iam8up|lpy | ...why should i/not use centos... |
| 02:15.12 | shmaltz | hi bkw_ |
| 02:15.21 | Iam8up|lpy | bkw_ - lucky bastard =) |
| 02:15.25 | shmaltz | brian, havn't seen you in a while |
| 02:15.33 | Netgeeks | IMHO, the choice of dist is more of a management issue than anything else. So if you are more comfortable managing a CentOS, Gentoo, etc. dist, I'd say use it |
| 02:15.41 | shmaltz | lam8up|lpy, becasue I don't like it |
| 02:15.44 | Iam8up|lpy | ariel_ - what do you LIKE about centos? shmaltz what do you dispise centos? |
| 02:15.55 | Iam8up|lpy | first off, my name begins with an i (the letter 'eye') |
| 02:16.04 | Katty | bkw_: hug junky for me! |
| 02:16.04 | Iam8up|lpy | and by 'you don't like it' you mean just the feel of the OS? |
| 02:16.10 | shmaltz | lam8up|lpy, do you know why I dont like shrimp? |
| 02:16.15 | Katty | also, bye now (= |
| 02:16.19 | ariel_ | I use CentOS due to it works every time. |
| 02:16.23 | *** part/#asterisk Katty (n=Administ@dialup-4.244.183.23.Dial1.StLouis1.Level3.net) |
| 02:16.28 | Iam8up|lpy | ariel_ - cause it's rhel =) |
| 02:16.41 | ariel_ | well most things will support it so yes. |
| 02:16.55 | Iam8up|lpy | i've got relativly old hardware, so i guess it's centos... |
| 02:16.58 | ariel_ | but for my desktop I use Kubuntu. But for my servers I use CentOS |
| 02:17.04 | shmaltz | ariel_, whats the longs uptime you got on any of your centos boxes running asterisk? |
| 02:17.12 | Iam8up|lpy | but why do you say centos 4.3 desktop as opposed to 4.2 server? |
| 02:17.19 | ariel_ | shmaltz, I have some over a year up and running |
| 02:17.19 | shmaltz | ubuntu is nice, but for desktop |
| 02:17.26 | ariel_ | 3.3 boxes |
| 02:17.30 | Iam8up|lpy | shmaltz - the one at work has been there for months... |
| 02:17.37 | shmaltz | ariel_, you mean uptime in the shell shows a year plus? |
| 02:17.45 | ariel_ | shmaltz, I said I only use kubuntu for desktop |
| 02:17.54 | ariel_ | shmaltz, yes |
| 02:17.57 | shmaltz | ariel_, I know, I just agreed with you |
| 02:18.03 | shmaltz | ariel_ nice |
| 02:18.21 | Iam8up|lpy | i hate how kubuntu doesn't support my docking station for my laptop |
| 02:18.32 | ariel_ | I have one box still on .7 asterisk it's been up for over 3 years and just keeps running. It's on CentOS 3.1 |
| 02:18.47 | shmaltz | anyhow, I use slackware, once you like one distro, and it is stable enough, I don't see why one would like another |
| 02:19.00 | Netgeeks | <PROTECTED> |
| 02:19.07 | ariel_ | Iam8up|lpy, well it's part of life. Not everything is supported. With it. But it's getting better. |
| 02:19.19 | shmaltz | in other words, the reason I said no to centos is because I like slackware |
| 02:19.31 | shmaltz | Netgeeks, what distro? |
| 02:19.32 | ariel_ | shmaltz, that OK. |
| 02:19.37 | ariel_ | you use what you know. |
| 02:19.40 | Netgeeks | thats a FC3 box |
| 02:19.57 | Iam8up|lpy | ariel_ - thing is..ubuntu is one of the most compatible distros out there... |
| 02:20.13 | shmaltz | this is mine: |
| 02:20.14 | shmaltz | 22:11:49 up 256 days, 22:56, 1 user, load average: 0.00, 0.00, 0.00 |
| 02:20.16 | shmaltz | its only that little because I took it down back then to change a PCI card |
| 02:20.23 | shmaltz | Netgeeks, running asterisk? |
| 02:20.48 | Netgeeks | buit it's a dev server, where I do alot of asterisk development, so it's not a production server. But I do run lots of load tests on it using sipp on another system against dialplans on it |
| 02:21.18 | shmaltz | Netgeeks, I had tons of trouble running FC3 with asterisk, it kept crashing |
| 02:21.28 | [hC] | white:~# uptime |
| 02:21.28 | [hC] | <PROTECTED> |
| 02:21.29 | [hC] | :) |
| 02:21.40 | shmaltz | just yesterday I upgraded my last FC3 box to slackware 10.2 |
| 02:21.56 | ariel_ | got to go see you all later.... |
| 02:22.04 | Netgeeks | hehe, actually look at the start time on this asterisk process running on it |
| 02:22.16 | Netgeeks | root 8980 0.0 0.7 49528 7612 ? S 2005 0:01 asterisk -vvvg -c |
| 02:22.30 | Netgeeks | lol, that asterisk been running for at least 8 months |
| 02:22.40 | shmaltz | lol |
| 02:23.01 | Netgeeks | well, I'm off gonna go get some dinner |
| 02:23.03 | Iam8up|lpy | Netgeeks - start time..where... |
| 02:23.18 | Netgeeks | that 2005 is the start time coloumn from a ps aux list |
| 02:23.25 | Netgeeks | colomn |
| 02:23.29 | Netgeeks | column |
| 02:23.29 | Iam8up|lpy | oh neat |
| 02:23.31 | Iam8up|lpy | =) |
| 02:23.39 | Netgeeks | damn, I need to go back to english 101 |
| 02:24.04 | Iam8up|lpy | so...any beginners guides for asterisk? |
| 02:24.13 | tengulre11 | ariel_: I can hear music from asterisk now, thank you very much!! :) |
| 02:35.38 | Snake-Eyes | tengulre11, got asterisk running now? |
| 02:35.41 | sp0n9e | Iam8up|lpy: it's hard to get more nooby than me. |
| 02:35.49 | sp0n9e | at asterisk anyway |
| 02:36.27 | Iam8up|lpy | uhm |
| 02:36.28 | Iam8up|lpy | how bout me |
| 02:36.32 | Iam8up|lpy | i've no experience with it |
| 02:36.42 | sp0n9e | read the oreilly book |
| 02:36.45 | sp0n9e | ~docs |
| 02:36.49 | jbot | rumour has it, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 02:36.52 | Iam8up|lpy | i like the pdf of it tbh |
| 02:36.58 | Iam8up|lpy | it is updated much more |
| 02:37.08 | sp0n9e | yeah...it's still the oreilly book :) |
| 02:37.10 | Iam8up|lpy | unless the bookstore got itself an updated copy |
| 02:37.17 | sp0n9e | that's the one i read |
| 02:37.20 | Iam8up|lpy | the pdf is actually more accurate then the book |
| 02:37.36 | Iam8up|lpy | it's the same stuff, but the pdf has been updated to 1.2.x or wahtever version... |
| 02:38.52 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
| 02:38.53 | *** mode/#asterisk [+o mog] by ChanServ |
| 02:40.56 | *** join/#asterisk DrkShdw (n=DrkShdw@unaffiliated/drkshdw) |
| 02:42.45 | brookshire | so yeah |
| 02:42.49 | brookshire | t1 interface |
| 02:42.50 | brookshire | :) |
| 02:42.56 | file | that's cheating |
| 02:43.01 | brookshire | ADSI |
| 02:43.13 | brimstone | i don't have t1 interface |
| 02:43.13 | file | make that 3! |
| 02:43.26 | file | I wonder if I do |
| 02:43.27 | brimstone | nor ADSI :/ |
| 02:43.31 | file | darn, I don't |
| 02:43.36 | brookshire | ooo.. thanks.. i've got that one |
| 02:43.42 | brimstone | hey now... |
| 02:46.02 | fuser | http://www.nhc.noaa.gov/tracks/2005atl.gif |
| 02:46.27 | fuser | looking at the previous years photos, all i can say is: it was good doing asterisk installs in houston |
| 02:54.28 | *** join/#asterisk trbldwine (i=troubled@71.194.161.170) |
| 02:57.39 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
| 03:05.07 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
| 03:16.57 | *** join/#asterisk SwK (n=Silik0nJ@204.250.115.130) |
| 03:17.34 | brookshire | PBX Roadmap |
| 03:24.20 | *** join/#asterisk angom_h (n=angom@red-corp-200.76.226.132.telnor.net) |
| 03:27.03 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
| 03:28.17 | *** join/#asterisk lorinc (n=ang@caracas-0797.adsl.interware.hu) |
| 03:31.02 | *** join/#asterisk brc_ (n=brc_@pdpc/supporter/basic/brc) |
| 03:39.03 | *** join/#asterisk bkw_ (n=brian@asterisk/friend-and-developer/bkw) |
| 03:44.49 | brimstone | alaw |
| 03:45.45 | *** join/#asterisk xxoxx (n=xxoxx@tor/regular/xxoxx) |
| 03:46.37 | brookshire | ENUM |
| 03:51.29 | *** join/#asterisk bmg505 (n=leon@c1-206-4.rndf.isadsl.co.za) |
| 04:04.30 | *** join/#asterisk javar (n=javar@200.118.174.253) |
| 04:10.09 | brimstone | Remote Office Support |
| 04:24.51 | niteowldave | file: I found that the t.38 re-invite fails as it does not provide any audio codecs, have you seen the passthrough working in trunk? |
| 04:26.32 | file | niteowldave: that's what I told you two times before, and what you reported in the bug note and on the mailing list - and yes it does work, it was tested before it was put in |
| 04:26.53 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 04:27.02 | file | the Sipuras negotiated initially with audio though, and then reinvited to T.38 |
| 04:28.08 | niteowldave | file: I am seeing the initial call set up and running in alaw ok, when the re-invite comes in asterisk drops it though |
| 04:30.13 | *** join/#asterisk SuperLag (n=aaron@gentoo/developer/SuperLag) |
| 04:34.52 | *** join/#asterisk lorinc (n=ang@caracas-3610.adsl.interware.hu) |
| 04:42.49 | *** join/#asterisk rikstah (n=rick@c-24-17-81-231.hsd1.wa.comcast.net) |
| 04:43.43 | Faithful | Grrr!!! What am I supposed to d with Tel$tra? I just discovered the issues we are having with our * box is that they haven't activated "Disconnect Supervision". What else do we need that they haven't activated? You know like CID, which they did not activate... you know like these are business lines and they don't activate what I think are elementary services for a business. Like a business doesn't need to know who's calling??? |
| 04:44.18 | JT | a lot don't |
| 04:44.37 | JT | it has always been an additional service on top of line rental with telstra |
| 04:44.56 | JT | and callerid has only been around for a few years |
| 04:45.06 | JT | that said, telstra can be pretty hopeless |
| 04:45.13 | Faithful | But it's madness... we need to compile a checklist of thinks to request from telstra when deploying new services. |
| 04:45.17 | JT | what sort of lines are they? |
| 04:45.27 | Faithful | PSTN business lines |
| 04:45.37 | JT | analogue, gotchya |
| 04:46.09 | Faithful | They didn't even terminate them... just left us with a wire hanging out a conduit... |
| 04:46.14 | JT | is disconnect supervision a fee service? |
| 04:46.15 | hads | Disconnect supervision and callerid are about the only ones. |
| 04:46.36 | JT | hads: ? |
| 04:47.02 | hads | There aren't that many other useful 'services' on POTS lines. |
| 04:47.17 | JT | what is telstra analogue disconnect supervision? i've never had to deal with it |
| 04:47.18 | Faithful | The tech said where's your MDF??? we say we don't have one... he says I don't terminate if you haven't got an MDF... Man these are new lines in a new building... what game are they playing. |
| 04:47.20 | JT | true |
| 04:47.35 | *** join/#asterisk LuckySeven (n=lucky7@218.208.204.128) |
| 04:47.35 | JT | well you need an MDF |
| 04:47.46 | JT | i think that may be the demarcation point |
| 04:47.49 | JT | can't remember |
| 04:48.07 | Faithful | It is indeed... but they should supply it if it isn't there. |
| 04:48.30 | LuckySeven | hi, i have asterisk setup with two FXO channels. |
| 04:48.47 | JT | depends on the building |
| 04:48.51 | LuckySeven | when someone calls in to the first channel, phones on the system will ring. however, after a while, if no one answers, |
| 04:48.52 | *** join/#asterisk adorah (n=Administ@87.68.169.196.cable.012.net.il) |
| 04:48.53 | JT | they might have to charge for it |
| 04:48.54 | Faithful | Point is the tech only gets .5hr to terminate... Supply & terminate MDF takes longer... so he walks. |
| 04:49.04 | LuckySeven | the call will be diverted to a cell phone. |
| 04:49.09 | LuckySeven | (via the other zap channel) |
| 04:49.12 | Faithful | They are all subcontractors |
| 04:49.23 | JT | yeah mostly |
| 04:49.39 | LuckySeven | I am having a problem whereby if I end the call on the cellphone, the zap channel will get my cellphone's voicemail. |
| 04:49.40 | JT | so what is disconnect supervision? |
| 04:49.59 | LuckySeven | How do I detect if it has entered voicemail so I can do a Hangup() appropriately? |
| 04:50.09 | Faithful | So they told us to break the law basically. Buy an MDF and termiate it yourself... no one ever checks. |
| 04:50.16 | *** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie) |
| 04:50.20 | LuckySeven | Or is it not a good idea even to do such a forwarding arrangement? |
| 04:51.01 | Faithful | LuckySeven: get "Disconnect Supervision" and you shouldn't have any problems... |
| 04:51.30 | LuckySeven | Ok thank you. I will google on that. :) |
| 04:51.34 | Faithful | if they hit # to end message the cell network hangs up |
| 04:51.37 | JT | Faithful: so what is telstra disconnect supervision and does it cost money? |
| 04:51.48 | Faithful | Doesn't cost |
| 04:51.56 | hads | It'll either be a line reversal or open loop. |
| 04:52.26 | LuckySeven | Faithful: Well. What if they don't hit the # key? |
| 04:52.29 | Faithful | It basically closes the line after a short period if the indial caller hangs up. |
| 04:52.46 | LuckySeven | F: Thats because I have to wait for my cellphone provider's voicemail system to timeout right? |
| 04:52.49 | Faithful | LuckySeven: that's what disconnect supervision is for... |
| 04:53.38 | JT | Faithful: is that not what normally happens when the calling party hangs up? |
| 04:53.53 | Faithful | it's like the old days... a prank caller calls you up... but won't hang up the phone and keeps your line open until they hang up. |
| 04:55.15 | JT | so disconnect supervision allows the callED party to hang up instantly? |
| 04:55.32 | LuckySeven | So disconnect supervision is a service that I have to subscribe from my telco? |
| 04:55.33 | Faithful | JT: not without disconnect supervision... what happens is it sends a disconnect beep beep down the line... now if the other equipment understand the beep it might hang up... but lots of equipment doesn't and in asterisk we have to compensate for the lack of the service but it is not the right way. |
| 04:56.28 | Faithful | LuckySeven: yes or play with * http://www.voip-info.org/wiki-Asterisk+Disconnect+Supervision |
| 04:56.38 | JT | because from my experience, the callING party can always immediately terminate the call |
| 04:57.18 | Faithful | Yes but how does your end know that it is terminated? |
| 04:57.31 | Faithful | Unless there is something on the line telling it |
| 04:57.49 | Faithful | besides a beep. |
| 04:57.51 | JT | ah right, i wasn't sure what you mean until now, wasn't very clear |
| 04:58.08 | Faithful | but it works both ways |
| 04:58.11 | LuckySeven | lol, I am getting a one minute long busy signal voicemail message courtesy of Asterisk when I check my cell phone's voicemail. |
| 04:58.15 | JT | and i guess open circuit or line reversal could always be used |
| 04:58.42 | JT | the joys of lines with no signalling channels :/ |
| 04:58.56 | Faithful | JT: Yes asterisk know what to do with that... but not a beep beep |
| 04:59.09 | JT | ah, is that all telstra will provide? |
| 04:59.23 | Faithful | esp in AU the STD pips and the disconnect beeps are not too disimilar |
| 04:59.39 | JT | out of curiosity how many lines does the business have? |
| 04:59.58 | Faithful | JT: no, if you request... when one party hangs up they close the line. |
| 05:00.14 | Faithful | JT: only 3 for PSTN |
| 05:00.27 | JT | ah ok |
| 05:00.31 | JT | considered ISDN2? |
| 05:00.39 | Faithful | Backups (supposed to be) ... but that raises another issue |
| 05:01.09 | Faithful | JT need ADSL for VoIP |
| 05:01.25 | LuckySeven | Faithful: callprogress=yes and busydetect=yes doesn't seem to do the trick for me. |
| 05:01.33 | Faithful | Besides Wildcard for PSTN is cheaper than a Fritz Card |
| 05:01.44 | JT | ah right hmm |
| 05:02.00 | JT | fritz cards are too small for what i have planned |
| 05:02.03 | Faithful | and then they are not tied to ISDN service either |
| 05:02.04 | JT | need an 8 port card |
| 05:02.10 | Faithful | Yup. |
| 05:02.19 | Faithful | An E1? |
| 05:02.21 | JT | looking at junghanns |
| 05:02.27 | JT | nah, 8 ISDN ports |
| 05:02.32 | JT | 4 set as TE |
| 05:02.35 | JT | 4 as NT |
| 05:02.50 | JT | pabx voip gateway/interceptor/least cost router |
| 05:03.36 | LuckySeven | http://www.asteriskguru.com/tutorials/resolving_hangup_detection_problems_fxo_tdm_voicemail.html |
| 05:07.11 | *** join/#asterisk angom_h (n=angom@red-corp-200.76.226.132.telnor.net) |
| 05:07.27 | Faithful | Meanwhile the company I installed for point the finger at "that gadget box you installed" |
| 05:09.10 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
| 05:09.36 | LuckySeven | Busy detection and all doesn't work for me. |
| 05:09.44 | LuckySeven | Its because I am not trying to detect a busy tone. |
| 05:09.51 | LuckySeven | Rather, I am trying to detect entering voicemail |
| 05:12.16 | JT | sounds like one for the too hard basket, to be honest |
| 05:14.05 | Qwell | What, you mean like app_amd? |
| 05:23.07 | *** join/#asterisk juice (i=1000@mo-71-50-31-176.dhcp.embarqhsd.net) |
| 05:34.49 | adorah | Hi is any one can comment on this CDR log? http://pastebin.ca/111122 |
| 05:34.59 | adorah | =does.. |
| 05:39.14 | adorah | <LuckySeven>I send unanswered calls to a custom-recordme inactive extension 5678 so it "answers" and hangs up after 5 sec.. |
| 05:39.41 | adorah | <LuckySeven>Than the fxo hangs up |
| 05:42.12 | *** join/#asterisk parag_ast (n=root@dxb-b16259.alshamil.net.ae) |
| 05:42.16 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 05:42.22 | parag_ast | macro-exten-vm,s-BUSY,2 |
| 05:42.33 | kmilitzer | Morning everyone ... |
| 05:43.14 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 05:46.55 | *** join/#asterisk WildPikachu (n=WildPika@unaffiliated/wildpikachu) |
| 05:47.23 | parag_ast | my asterisk is not hanging up the channels due to busy voice mails.. |
| 05:47.25 | parag_ast | <PROTECTED> |
| 05:47.25 | parag_ast | <PROTECTED> |
| 05:47.25 | parag_ast | <PROTECTED> |
| 05:47.26 | parag_ast | <PROTECTED> |
| 05:47.26 | parag_ast | <PROTECTED> |
| 05:47.26 | parag_ast | <PROTECTED> |
| 05:47.28 | parag_ast | <PROTECTED> |
| 05:47.30 | parag_ast | i m getting like this |
| 05:48.49 | parag_ast | dlynes_laptop, any idea |
| 05:48.59 | dlynes_laptop | how'd i get elected? |
| 05:49.14 | parag_ast | ahh pls help |
| 05:49.47 | dlynes_laptop | It was a hot day...figured you could cool down... |
| 05:50.28 | parag_ast | Could u pls look into the above matter ?? |
| 05:50.40 | dlynes_laptop | Your asterisk is not hanging up the calls because it's stuck on 'Busy()' |
| 05:50.49 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 05:51.00 | dlynes_laptop | And whoever's connected to Zap/1-1 hasn't hung up the call yet |
| 05:51.51 | parag_ast | so tell me the way so that asterisk should never reach this condition, i hv disabled the voicemail also |
| 05:52.12 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:52.17 | dlynes_laptop | Has nothing to do with the voicemail |
| 05:52.24 | dlynes_laptop | It has to do with your dialplan |
| 05:52.32 | dlynes_laptop | pastebin your extensions.conf file |
| 05:52.50 | parag_ast | okk let me do it |
| 05:52.54 | dlynes_laptop | good afternoon, X-Rob |
| 05:53.05 | X-Rob | heya dlynes_laptop |
| 06:00.37 | WildPikachu | is it possible to make incoming calls ring all extensions until a person picks one up? |
| 06:00.51 | benjk_ | yes it is |
| 06:01.02 | WildPikachu | could you point me to the right place to read up on it? |
| 06:01.03 | LuckySeven | adorah: sorry, I was afk. |
| 06:01.13 | benjk_ | voip-info.org |
| 06:01.30 | benjk_ | search for dialplan |
| 06:01.50 | benjk_ | in particular the dial command |
| 06:01.53 | LuckySeven | adorah: what if I wish for the call to be rererouted to my mobile. e.g. (When I am out of the office) |
| 06:03.16 | parag_ast | dlynes_laptop, http://pastebin.ca/111162 |
| 06:03.46 | adorah | <LuckySeven>I was not here..sorry..so reroute it than in y'r dial plan what is the problem? |
| 06:04.26 | dlynes_laptop | parag_ast, i'm not debugging that |
| 06:04.39 | dlynes_laptop | parag_ast, I have no idea what's in all that amp crapola |
| 06:04.51 | parag_ast | what, |
| 06:05.04 | dlynes_laptop | parag_ast, you're best off asking someone in #freepbx why it's doing what it's doing |
| 06:05.31 | dlynes_laptop | parag_ast, I'm not going to be much help to you, unfortunately, because I don't know what's in all those files AMP includes |
| 06:06.06 | dlynes_laptop | not only that, but that file is one big, huge mess |
| 06:06.10 | dlynes_laptop | It's not even readable |
| 06:06.22 | parag_ast | its a default asterisk@home file |
| 06:06.28 | adorah | <dlynes_laptop>Hi what do u think about that CDR log? my machine went mad..http://pastebin.ca/111122 |
| 06:06.32 | dlynes_laptop | yeah...it's an AMP file |
| 06:07.21 | parag_ast | so kindly let me know what is the best way to overcome this problem |
| 06:07.27 | dlynes_laptop | adorah, looks to me like you've got an extension that calls itself in your dialplan |
| 06:07.39 | dlynes_laptop | parag_ast, I would talk to someone in #freepbx |
| 06:07.47 | dlynes_laptop | parag_ast, they can help you a lot more than I could |
| 06:08.00 | parag_ast | Thanks a lot |
| 06:08.06 | dlynes_laptop | parag_ast, Have you read the topic in this channel, lately? |
| 06:08.47 | adorah | <dlynes_laptop>Only that is not the case..it just happens from time to time..that is the first ext in the ring group.. |
| 06:09.10 | dlynes_laptop | there's a reason it's in the topic...it's not to be difficult...we just can't help much because of all the stuff that gets included from other files in freepbx, and the fact that it's write-only config files |
| 06:09.31 | dlynes_laptop | adorah, do you have an extensions.conf file? |
| 06:09.41 | adorah | <dlynes_laptop>sure |
| 06:10.10 | adorah | I checked it over and over.. |
| 06:11.07 | dlynes_laptop | pb it, please? |
| 06:11.09 | kmilitzer | I just found some strange error about not able to lock SIP channels and ignoring message. See here http://pastebin.ca/111171 |
| 06:12.05 | dlynes_laptop | kmilitzer, cool...that's the first time i've seen anything like that |
| 06:13.18 | kmilitzer | dlynes_laptop: It's a first one for me too. This is my gateway to the pstn. I started to send live traffic over it the day before yesterday and then got this errors tonight. This is not good ... |
| 06:14.33 | *** join/#asterisk daysmen3 (n=primus@host217-44-108-65.range217-44.btcentralplus.com) |
| 06:16.15 | adorah | dlynes_laptop>http://pastebin.ca/111175 |
| 06:16.27 | Un1x | is there a way i can increase volume on asterisk? |
| 06:16.38 | dlynes_laptop | ummmm |
| 06:16.39 | dlynes_laptop | wtf? |
| 06:16.45 | dlynes_laptop | Does nobody read the topic? |
| 06:16.50 | dlynes_laptop | adorah, type /topic |
| 06:17.22 | Un1x | dlynes.. how can i turn up the volume on asterisk |
| 06:17.28 | Un1x | so the output is louder... |
| 06:17.30 | dlynes_laptop | Un1x, define increase the volume? |
| 06:17.33 | adorah | :) I thought It has nothing to do with freepbx.. |
| 06:17.34 | dlynes_laptop | the output on what? |
| 06:17.41 | Un1x | well the volume on my phone line is very low |
| 06:17.42 | dlynes_laptop | adorah, AMP == FreePBX |
| 06:17.47 | Un1x | i guess asterisk is having it in the middle |
| 06:17.57 | Un1x | is there a way to increase volume |
| 06:18.19 | dlynes_laptop | adorah, if it's anything to do with the dialplan in AMP, there's little chance of anyone being able to help you in a timely fashion here |
| 06:18.37 | dlynes_laptop | Un1x, change the volume on what? |
| 06:18.41 | Un1x | Dlynes how do i increase the volume |
| 06:18.43 | adorah | ok thx.. |
| 06:18.46 | Un1x | well see when im talking on the phone |
| 06:18.47 | dlynes_laptop | Un1x, on a SIP phone line, on a ZAP line? |
| 06:18.50 | Un1x | i can barely hear the guy |
| 06:18.52 | Un1x | even if he yells |
| 06:18.58 | dlynes_laptop | Un1x, you haven't stated what type of phone line it is |
| 06:19.34 | adorah | <dlynes_laptop>In my other asterisk non-TB it never happend indeed.. |
| 06:19.44 | Un1x | dlynes it's sip |
| 06:19.51 | Un1x | but im using it via a Zaptel tdm400P card |
| 06:19.52 | Un1x | .. |
| 06:19.53 | dlynes_laptop | adorah, non-TB? |
| 06:20.04 | dlynes_laptop | Un1x, increase your gains on your tdm400p card |
| 06:20.10 | Un1x | how can i do that? |
| 06:20.12 | adorah | <dlynes_laptop>I mean regular *... |
| 06:20.18 | dlynes_laptop | Un1x, edit your zapata.conf file? |
| 06:20.40 | Un1x | dlynes can you give me the syntax on how to add it please? |
| 06:20.45 | dlynes_laptop | Un1x, no |
| 06:20.49 | dlynes_laptop | Un1x, do me a favor |
| 06:21.03 | dlynes_laptop | Un1x, read your sample zapata.conf file and read voip-info.org |
| 06:21.20 | dlynes_laptop | Un1x, then you'll learn about all the other settings in zapata.conf as well |
| 06:22.17 | rowter | anyone know a good way to detect answer on FXO? |
| 06:22.54 | Un1x | rxgain=4.2 |
| 06:22.57 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 06:23.22 | dlynes_laptop | Un1x, yeah...don't set it too high though, or you'll get distortion |
| 06:23.43 | dlynes_laptop | rowter, it's automatic |
| 06:23.43 | Un1x | dylnes but voip doesn't show wich number to edit to make it go higher and wich not to edit... |
| 06:23.50 | Un1x | like if 6.0 is loader then 4.0 |
| 06:23.51 | Un1x | etc |
| 06:23.53 | dlynes_laptop | Un1x, just play with it |
| 06:24.02 | dlynes_laptop | Un1x, higher gains mean louder volume |
| 06:24.03 | Un1x | http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf |
| 06:24.09 | Un1x | so 6.0 is higher thanks |
| 06:24.10 | dlynes_laptop | Un1x, lower gains mean lower volume |
| 06:24.17 | rowter | dlynes_laptop, nop, am trying to detect outgoing answers |
| 06:24.25 | dlynes_laptop | Un1x, so too high will make it distorted; too low will make it so you can't hear it |
| 06:24.34 | Un1x | dylnes does 6.0 sybolize a higher gain then 4.0 |
| 06:24.54 | kmilitzer | dlynes_laptop: Just searched some more in my loges. Found that before the "BAD! BAD! BAD!" message: Aug 1 21:13:31 DEBUG[5014] chan_sip.c: Failed to grab lock, trying again... |
| 06:25.02 | dlynes_laptop | rowter, oh..if the other end answers, asterisk will automatically try bridging the call |
| 06:25.25 | kmilitzer | Do you or anyone else know under what circumstances asterisk is not able to lock a channel? |
| 06:25.32 | dlynes_laptop | rowter, if you want to find out the status of the call (after the call), you can check the ${DIALSTATUS} variable |
| 06:26.04 | dlynes_laptop | kmilitzer, if something else already has control of the channel |
| 06:26.08 | rowter | dlynes_laptop, on my E1 it detects answers and rings, but on FXO it does not.. all calls are answer |
| 06:26.24 | kmilitzer | dlynes_laptop: But how should that happen? It makes no sense to me ... |
| 06:26.42 | dlynes_laptop | rowter, do you have answer supervision enabled on the analog line? |
| 06:27.04 | *** join/#asterisk asterboy (n=kevin@S010600485480f4be.ed.shawcable.net) |
| 06:27.16 | rowter | dlynes_laptop, progess? it does not work well or it does? |
| 06:27.29 | dlynes_laptop | rowter, it does |
| 06:27.35 | dlynes_laptop | rowter, oh...you mean for outgoing calls |
| 06:27.58 | rowter | dlynes_laptop, yeah |
| 06:28.24 | dlynes_laptop | rowter, are you in north america? |
| 06:28.47 | rowter | dlynes_laptop, nop.. but it must be very similar here on mexico |
| 06:28.52 | dlynes_laptop | kmilitzer, have you tried searching for that error string in the source code? |
| 06:29.30 | dlynes_laptop | rowter, callprogress is only for fxs though, not fxo |
| 06:29.44 | rowter | dlynes_laptop, check this "Quote¨: the only POTS hardware that i've used that does support/work with answer supervision is the Sangoma AFT A200 ... as long as your telco provides it. then you can use "answeronpolarityswitch=yes" in zapata.conf. |
| 06:30.09 | rowter | dlynes_laptop, I have digium :( |
| 06:30.38 | dlynes_laptop | rowter, ah...yeah...only digium card i've used is the x100p |
| 06:30.38 | Un1x | dlynes this isn't working... |
| 06:30.39 | Un1x | http://pastebin.ca/111189 |
| 06:30.51 | dlynes_laptop | rowter, everything else has been sangoma hardware |
| 06:31.08 | rowter | dlynes_laptop, ooh |
| 06:31.16 | *** join/#asterisk madlynx (n=sleever@195.206.63.10) |
| 06:31.46 | rowter | dlynes_laptop, you can detect outgoing answer with sagnoma then.. its true? |
| 06:31.58 | hads|home | rowter: You seem to be talking about callprogress on analog lines. |
| 06:32.06 | dlynes_laptop | rowter, he is |
| 06:32.12 | dlynes_laptop | hads|home, he is |
| 06:32.17 | rowter | yeah |
| 06:32.19 | kmilitzer | dlynes_laptop: Yeah ... I found it. It's line 11324 (wow!) in chan_sip. If says it /* Process request, with netlock held */ ... |
| 06:32.31 | dlynes_laptop | rowter, yeah, never had any issues with it |
| 06:32.39 | hads|home | Which is not really reliable on any POTS hardware. |
| 06:32.57 | Un1x | dlynes can youy have a look at this please... |
| 06:32.57 | Un1x | 27.36 |
| 06:32.58 | Un1x | err |
| 06:33.01 | Un1x | http://pastebin.ca/111189 |
| 06:33.45 | kmilitzer | dlynes_laptop: I have the suspicion, that my connection may have lags from time to time, so SIP packages don't come in on time. I will monitor the connection with smokeping and see, if the error will surface again ... |
| 06:34.02 | rowter | hads so neither sagnoma nor digium will detect correctly the outgoing answer? |
| 06:34.47 | *** join/#asterisk sai2 (n=sai@203-97-220-141.cable.telstraclear.net) |
| 06:35.32 | sai2 | hi folks, sorry to ask a dumb question but I'm a bit stumped and can't find an answer online. I'm trying to figure out how to disable native bridging between two sip phones. Any suggestions? |
| 06:35.54 | X-Rob | that's not what you watn to do |
| 06:36.04 | X-Rob | 'native bridging' is an inetneral asterisk thing about the voice path |
| 06:36.10 | X-Rob | you want to disable reinvite, I guess |
| 06:36.24 | hads|home | rowter: As far as I know POTS lines cannot detect call progress at all. You may want to look into the callprogress option in zapata.conf but from memory it's experimental. |
| 06:36.35 | X-Rob | in fact, if you're seeing 'native bridging' that pretty much means that asterisk is in the voice path anyway. |
| 06:36.40 | sai2 | The situation is one in which one of the sip phones cannot talk directly to another, so the call will only work if it goes via asterisk. If it hands off the call, the payload cannot get through. |
| 06:36.52 | dlynes_laptop | Aug 1 15:25:10 VERBOSE[9256] logger.c: -- Called g1/6046619400 |
| 06:36.52 | dlynes_laptop | Aug 1 15:25:13 VERBOSE[9256] logger.c: -- Zap/1-1 answered SIP/104-8fb1 |
| 06:36.52 | dlynes_laptop | Aug 1 15:32:36 VERBOSE[9256] logger.c: -- Hungup 'Zap/1-1' |
| 06:36.53 | dlynes_laptop | A |
| 06:37.00 | dlynes_laptop | Looks like Sangoma detects it just fine |
| 06:37.19 | *** part/#asterisk madlynx (n=sleever@195.206.63.10) |
| 06:37.35 | sai2 | my understanding is that reinvites is not the same as a sip native bridging hand off... (?) |
| 06:37.48 | hads|home | dlynes_laptop: Yes, but on a POTS line the call is considered answered straight away, even if the far end never answers. |
| 06:37.54 | hads|home | Unlike ISDN |
| 06:37.55 | X-Rob | native bridging means that asterisk isn't transcoding the streams. Eg, they're both alaw or g729. |
| 06:38.02 | dlynes_laptop | hads|home, well, yeah |
| 06:38.13 | rowter | dlynes_laptop, check your cdr, all are answers.. |
| 06:38.18 | dlynes_laptop | hads|home, there's no possible way for it to signal that on an analog line |
| 06:38.30 | hads|home | Yeah, exactly. THat is what I was trying to say :) |
| 06:38.42 | dlynes_laptop | It's not a problem with the hardware |
| 06:38.47 | hads|home | I know. |
| 06:38.49 | dlynes_laptop | it's a problem with the technology |
| 06:38.54 | rowter | dlynes_laptop, the thing is I need to detect when someone talks.. after rings ofcourse. |
| 06:38.58 | dlynes_laptop | analog lines don't have q.931 |
| 06:39.00 | sai2 | it doesn't have anything to do with handing off the call so the sip phones talk to each other directly once the call is established? Asterisk coordinates the call, and then washes its hands of it because it thinks the sip phones will play nicely together. |
| 06:39.12 | sai2 | Perhaps I've confused something |
| 06:39.15 | dlynes_laptop | rowter, then you need to use Monitor() |
| 06:39.19 | JT | is it fairly easy in asterisk to configure whether or not for it to drop straight to an outbound line when an extension dials digits, or to have it wait for all the digits? |
| 06:39.26 | dlynes_laptop | rowter, and listen to the monitored conversation |
| 06:39.37 | dlynes_laptop | rowter, and have some kinda voice detection |
| 06:39.46 | hads|home | Or move to ISDN :) |
| 06:39.52 | dlynes_laptop | or a PRI |
| 06:40.03 | dlynes_laptop | or GSM |
| 06:40.03 | JT | which is still a form of isdn |
| 06:40.07 | hads|home | :) |
| 06:40.17 | rowter | hads I have idsn pri also, but we have better rates hehe |
| 06:40.31 | rowter | dlynes_laptop, monitor and then detect mmh |
| 06:40.46 | Un1x | dlynes can youy have a look at this please... |
| 06:40.47 | Un1x | http://pastebin.ca/111189 |
| 06:41.07 | sai2 | I thought I read that asterisk hands off the calls. All I want to do is disable that for the line in question. Am I barking up the wrong tree? |
| 06:41.28 | rowter | dlynes_laptop, I need to detect when it answers and then send them to a queue.. |
| 06:41.47 | hads|home | sai2: canreinvite=no |
| 06:41.50 | rowter | dlynes_laptop, am trying with app_waitforsilnce and some newmans.. |
| 06:42.07 | sai2 | I have that. But I thought canreinvite was for a different purpose anyway |
| 06:42.27 | *** join/#asterisk r0d3nt (i=r0d3nt@tinfoilhat.net) |
| 06:42.29 | hads|home | That will make the RTP go through Asterisk |
| 06:42.38 | sai2 | hmmm... ok |
| 06:42.45 | hads|home | sai2: BTW, hi from another NZer :) |
| 06:42.58 | sai2 | hi :) |
| 06:43.04 | *** join/#asterisk qdk_ (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
| 06:43.09 | dlynes_laptop | rowter, yeah...what you're wanting to do, i'm pretty sure isn't supported in asterisk |
| 06:43.16 | Un1x | can someone have a look at this please... |
| 06:43.17 | Un1x | http://pastebin.ca/111189 |
| 06:43.23 | Un1x | see if i set the rxgain correctly |
| 06:43.28 | dlynes_laptop | rowter, you'd have to write your own app, or something to achieve that |
| 06:43.42 | dlynes_laptop | Un1x, rxgain=n.nn |
| 06:43.54 | dlynes_laptop | Un1x, you can't possibly screw up setting it correctly |
| 06:44.05 | rowter | dlynes_laptop, trying to :) but there are some already that might help.. let me look on mantis |
| 06:44.09 | sai2 | yes, canreinvite sounds like the right thing... my mistake. Oddly, I already have that. |
| 06:44.13 | Un1x | ok i'll set it again see if it corrects |
| 06:44.21 | hads|home | rowter: There is the answer confirmation option to the dial command, but it's probably not something you want to use for external people. |
| 06:45.26 | rowter | hads, press 1 something like that? |
| 06:45.37 | hads|home | Yeah, # from memory |
| 06:49.27 | rowter | hads, dialogic can detect busy and answer on POT, but maybe is also a software thing than hardware.. |
| 06:50.46 | kmilitzer | Again my question to everyone: How can it happen that the lock of a SIP channel fails? Or should I better ask this question in -dev? |
| 06:56.18 | Un1x | ok rxgain=6 works but i cant hear the difference, and rxgain=6.0 doesn't work rxgain => 60% dont work |
| 06:56.52 | X-Rob | Un1x, restart asterisk |
| 06:57.46 | Un1x | i reloaded |
| 06:57.50 | Un1x | everytime i made a change |
| 06:57.53 | Un1x | didn't make a differnce |
| 06:58.33 | *** join/#asterisk madlynx (n=sleever@195.206.63.10) |
| 06:58.36 | *** part/#asterisk madlynx (n=sleever@195.206.63.10) |
| 07:04.22 | *** join/#asterisk af_ (n=af@ip-164-6.sn2.eutelia.it) |
| 07:07.28 | parag_ast | X-Rob, I m using asterisk@home |
| 07:07.32 | parag_ast | <PROTECTED> |
| 07:07.32 | parag_ast | <PROTECTED> |
| 07:07.32 | parag_ast | <PROTECTED> |
| 07:07.32 | parag_ast | <PROTECTED> |
| 07:07.32 | parag_ast | <PROTECTED> |
| 07:07.32 | parag_ast | <PROTECTED> |
| 07:07.34 | parag_ast | <PROTECTED> |
| 07:07.45 | parag_ast | my channels are always getting hanged |
| 07:08.07 | parag_ast | they are not hanging up properly |
| 07:08.44 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 07:08.55 | hads|home | ~AAH |
| 07:08.59 | jbot | well, aah is AMP and amp is, like, NOT supported here! people using it should join #freepbx (the new name of amp) |
| 07:09.35 | X-Rob | hads|home, except for like, I'm the AMP/freePBX/a@h guy. |
| 07:09.56 | X-Rob | and what does that have to do with zaptel hangup detection, anyway? |
| 07:10.12 | parag_ast | Its an asterisk question |
| 07:10.13 | X-Rob | (running A@H) |
| 07:10.20 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 07:10.24 | X-Rob | exactly my point. |
| 07:10.33 | Juggie | Un1x, you have to restart the zaptel for gain to take affect |
| 07:10.36 | Juggie | not just * |
| 07:11.03 | Juggie | er, nm |
| 07:11.04 | parag_ast | my zap channels are always remaing busy |
| 07:11.06 | Juggie | gain is * |
| 07:11.45 | Juggie | parag_ast, perhaps your telco doesnt support out of band notifications |
| 07:11.51 | Juggie | for busy and the like |
| 07:11.55 | Juggie | you might have to use inband. |
| 07:12.27 | parag_ast | inband for all sip |
| 07:12.32 | parag_ast | sip.conf |
| 07:13.01 | *** join/#asterisk vgster (n=vgster@217.78.147.238) |
| 07:13.39 | Juggie | not sip |
| 07:13.42 | Juggie | for zap |
| 07:13.49 | Juggie | are you using analog or digital lines |
| 07:13.57 | parag_ast | Analog |
| 07:13.58 | parag_ast | lines |
| 07:14.02 | Juggie | ok never mind then |
| 07:14.28 | parag_ast | I hv Digium 400P FXO |
| 07:16.43 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 07:18.13 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 07:25.40 | parag_ast | Juggie, any comments for Digium fxo card |
| 07:28.35 | *** join/#asterisk tengulre11 (n=tengulre@222.90.66.156) |
| 07:28.41 | tengulre11 | HI,all |
| 07:29.28 | *** join/#asterisk operat0r (n=h0msar@adsl-19-78-64.asm.bellsouth.net) |
| 07:29.34 | tengulre11 | what time should using agent.conf? |
| 07:29.39 | operat0r | droops http://0pencircuit.net/t0c/index.php?topic=163 |
| 07:31.30 | *** part/#asterisk operat0r (n=h0msar@adsl-19-78-64.asm.bellsouth.net) |
| 07:35.21 | *** join/#asterisk bintut (n=bintut@203.177.82.20) |
| 07:36.35 | *** join/#asterisk tengulre (n=tengulre@61.185.224.66) |
| 07:37.29 | tengulre | why I offine often!! |
| 07:43.27 | *** join/#asterisk Un1x (i=Sean@72.61.82.242) |
| 07:45.04 | *** join/#asterisk sarath (n=Sarath@202.83.55.32) |
| 07:48.24 | *** part/#asterisk sarath (n=Sarath@202.83.55.32) |
| 07:57.05 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 07:57.50 | *** join/#asterisk saftsack (n=saftsack@p54A7C874.dip.t-dialin.net) |
| 07:59.32 | *** join/#asterisk [Airwolf] (n=airwolf@dsl5402DE03.pool.t-online.hu) |
| 07:59.38 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 08:05.39 | *** join/#asterisk Astinus- (n=aa@85.19.143.16) |
| 08:10.46 | JT | anyone got a good idea on how to get asterisk to determine if a SIP provider is down, and switch to backup landlines if so? |
| 08:11.04 | JT | SIP provider being down could mean Internet down, error at SIP provider, etc |
| 08:11.54 | *** part/#asterisk mitcheloc (n=mitchelo@69-167-145-62.lmdaca.adelphia.net) |
| 08:12.54 | *** join/#asterisk Arno[Slack] (i=100@master.infinityperl.org) |
| 08:13.26 | l-fy | put an extension with Dial with priority 2 |
| 08:24.23 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 08:25.40 | *** part/#asterisk parag_ast (n=root@dxb-b16259.alshamil.net.ae) |
| 08:32.33 | *** join/#asterisk WildPikachu (n=WildPika@unaffiliated/wildpikachu) |
| 08:32.49 | WildPikachu | ok so if i make 2 DIAL SIP's the same priority, both phones will be dialled? |
| 08:39.18 | E-bola | When i call ppl PSTN lines on my linksys spa922, which is connected to my local asterisk server, which again is connected to a sip->pstn provider. They often have trouble hearing me |
| 08:39.25 | E-bola | While i can hear everything they say perfectly |
| 08:39.31 | E-bola | Whats the most likely problem? |
| 08:39.41 | l-fy | E-bola > you have to setup the frame size at 20 ms not at 30 ms |
| 08:39.49 | l-fy | in linksys you can do that in advanced mode |
| 08:40.09 | l-fy | WildPikachu > not both, one first and than the other |
| 08:40.12 | E-bola | so if its at 30ms frames/packets get fragmented or what? |
| 08:40.22 | l-fy | i don't remeber |
| 08:40.27 | WildPikachu | l-fy, would that make it ring both? |
| 08:40.41 | l-fy | i don't know |
| 08:40.48 | l-fy | i'm not an asterisk user actually |
| 08:41.08 | l-fy | i just remeber things since i was young and i was using asterisk |
| 08:43.53 | *** join/#asterisk Newbie___ (n=me@211.24.146.11) |
| 08:43.59 | *** join/#asterisk shaZwaz (n=chatzill@203.81.222.122) |
| 08:44.14 | Newbie___ | hi, any one using zhone channel banks with * ? |
| 08:52.43 | shaZwaz | hi, my T1+Adit CB is behaving awkwardly on incoming calls, no calls appear |
| 08:53.03 | *** join/#asterisk eset (n=eset@sjs-130-65-215-38.sjsu.edu) |
| 08:53.06 | shaZwaz | anyone has any idea ? |
| 08:53.34 | eset | ola, can i enable L16 PCM Wideband on asterisk in sip.conf? |
| 08:56.55 | *** join/#asterisk ousk (n=c02ol@login-15.efd.lth.se) |
| 08:57.21 | ousk | hi I have a question about NOTIFY sip messages ... |
| 08:57.53 | ousk | why is Asterisk sending out NOTIFY before there has been any subscribe messages sent? |
| 08:58.00 | ousk | just a register |
| 08:59.21 | shaZwaz | please help T1+Adit CB behaving randomly on incoming calls |
| 08:59.28 | *** join/#asterisk Mw3 (i=mw3@national.t-error.hu) |
| 09:00.11 | *** join/#asterisk Arno[Slack] (i=100@master.infinityperl.org) |
| 09:01.32 | *** join/#asterisk Dovid (n=dovi5988@pool-71-250-15-227.nwrknj.east.verizon.net) |
| 09:03.09 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 09:03.30 | ousk | anyone have a clue? |
| 09:04.49 | daysmen3 | HELP - how can i use hunt groups but call forwarding, dnd and call waiting - since youd have to iterate thru the numbers and change them accordingly - can this be without using AGI |
| 09:06.44 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 09:07.39 | *** join/#asterisk Arno[Slack] (i=100@master.infinityperl.org) |
| 09:10.39 | WildPikachu | is there a reason i don't hear Ringing() on a sip phone when testing my dial plan? |
| 09:16.58 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 09:18.35 | *** join/#asterisk baaaaa (i=barnie@chemical-reaction.de) |
| 09:25.31 | *** join/#asterisk Assid (n=assid@203.212.216.40) |
| 09:32.59 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 09:39.20 | *** part/#asterisk Newbie___ (n=me@211.24.146.11) |
| 09:40.12 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 09:41.29 | *** join/#asterisk saftsack (n=saftsack@p54A7C874.dip.t-dialin.net) |
| 09:41.35 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 09:42.52 | *** join/#asterisk r_marvin (n=marvin@83.166.220.142) |
| 09:46.02 | r_marvin | hello, i set up an asterisk 1.2.10 with chan_h323 and i try to test in in the following config: sipphone -> asterisk -> h323_gateway . i can place calls but there's no voice, apparently chan_h323 reports 0.0.0.0 as remote ip. Is there something i am missing? |
| 09:47.06 | key2 | r_marvin: yate |
| 09:47.13 | key2 | would do it for u |
| 09:48.11 | r_marvin | unfortunately, i need to use asterisk, but i'll give yate a try when possible |
| 09:48.25 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 09:48.28 | l-fy | r_marvin > usually people use yate as a sip-h323 translator |
| 09:48.44 | l-fy | so you can add yate on the same box |
| 09:57.05 | *** join/#asterisk Gunnar (n=gunnar@62.97.242.6) |
| 09:59.35 | *** join/#asterisk lorinc (n=ang@caracas-1359.adsl.interware.hu) |
| 10:08.40 | WildPikachu | darn |
| 10:08.44 | WildPikachu | my call queue is broken |
| 10:08.54 | WildPikachu | Aug 2 10:08:45 WARNING[5565]: file.c:512 ast_openstream_full: File "queue-youarenext" does not exist in any format |
| 10:08.59 | WildPikachu | but its there :( |
| 10:11.30 | *** join/#asterisk mvdk (i=mvdk@60-240-70-160-nsw-pppoe.tpgi.com.au) |
| 10:12.12 | WildPikachu | aha, it wants .gsm |
| 10:12.37 | *** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 10:13.09 | mvdk | How would I use the zaptel drivers to set up an ISDN data call? |
| 10:15.47 | mvdk | The key is, it will be a backup service, so using a statically assigned number of channels is The Wrong Thing (tm), particularly in Australia, where charges for these things can be extortionate |
| 10:18.42 | mvdk | Ah, sorry, just found the answer - it's in how you phrase the question - ZapRAS is the answer :) |
| 10:24.03 | *** join/#asterisk expat_iain (n=expat_ia@194.204.99.131) |
| 10:24.47 | e-ddie | Unable to open (format alaw): No such file or directory |
| 10:25.00 | e-ddie | anyone had such error messages? |
| 10:25.27 | e-ddie | makes our sound choppy, and the server core dumping in between |
| 10:34.28 | WildPikachu | is it possible for an agent to rejoin a queue after he was removed for not answering a call? |
| 10:42.50 | Modcuts | Is there any iax providers offering global toll free numbers? |
| 10:52.30 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 11:03.13 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 11:05.48 | shaZwaz | T1 interface card + Adit 600 donot detect incoming ring ...any guess ? |
| 11:07.10 | *** join/#asterisk potsboy (n=chrisg@196.211.16.202) |
| 11:08.53 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 11:09.56 | *** join/#asterisk jeffjohnson (n=Jeffjohn@unaffiliated/jeffjohnson) |
| 11:10.05 | jeffjohnson | howdy |
| 11:11.11 | jeffjohnson | how much rtp ports I need to forward, for VoIP gateway with 2 channels? 4? 2 rtp ports per call? |
| 11:11.20 | l-fy | 2 |
| 11:11.27 | jeffjohnson | why 2? |
| 11:11.34 | jeffjohnson | for 2 calls at the same time? |
| 11:11.37 | l-fy | jeffjohnson > because rtp is bidirectional in sip |
| 11:11.47 | *** join/#asterisk champster (n=asterisk@AH.tescogroup.com) |
| 11:12.02 | jeffjohnson | l-fy: ah ok thx |
| 11:13.14 | WildPikachu | l-fy, can you offer me advice if i have 1 member in a ringall, its ringed once, doesn't answer ... then never again |
| 11:13.52 | l-fy | WildPikachu > i don't use the * queues |
| 11:14.01 | l-fy | i use a database based acd system |
| 11:14.01 | WildPikachu | :) |
| 11:14.20 | l-fy | which is not using asterisk |
| 11:14.35 | l-fy | and which is handeling your problem in a different way |
| 11:15.12 | l-fy | bassically in my acd system each call is resend to another operator if one dosen't answer and it will do it every time until the operator disconnect |
| 11:15.22 | l-fy | disconnect = unregister himself |
| 11:15.37 | l-fy | which may happend in sip in about 10 minutes if him closes him sip client |
| 11:18.31 | *** join/#asterisk oej (n=oej@h-66-167-184-122.nycmny83.covad.net) |
| 11:19.32 | l-fy | hi oej |
| 11:19.44 | *** join/#asterisk eset (n=eset@sjs-130-65-215-38.sjsu.edu) |
| 11:22.05 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 11:22.27 | ousk | anyone who is 1337 here? |
| 11:23.48 | ousk | jeffjohnson: don't you need to forward rtcp also? |
| 11:24.57 | ousk | hmm or maybe it is set automatically .. |
| 11:25.49 | backblue | ousk: you came to #asterisk asking for 1337?? |
| 11:25.54 | jeffjohnson | ousk: with rtp only it works, i only the to call outside |
| 11:26.53 | ousk | well I have some problem with the behaivor with asterisk .. |
| 11:27.08 | ousk | jeffjohnson: what do you mean? |
| 11:27.23 | ousk | backblue: why not? |
| 11:27.41 | ousk | elite in asterisk and sip ... |
| 11:27.49 | ousk | that would help me a bit .. |
| 11:29.08 | ousk | I have never been here before so I do not know what to expect ,,, |
| 11:29.49 | ousk | My hope was that some developor was here to answer my questions ... |
| 11:30.04 | ousk | just a shot in the dark ... |
| 11:31.24 | ousk | backblue: or did you just diz me? |
| 11:34.10 | *** join/#asterisk AsteriskAlbania (n=info@217.24.244.130) |
| 11:35.01 | AsteriskAlbania | any one has used a ASTERISK + RADIUS implementation |
| 11:37.36 | *** part/#asterisk SuperLag (n=aaron@gentoo/developer/SuperLag) |
| 11:40.43 | AsteriskAlbania | any one has used a ASTERISK + RADIUS implementation , I need some help |
| 11:40.50 | AsteriskAlbania | even paid hours |
| 11:41.15 | *** join/#asterisk nettie (i=[U2FsdGV@85-18-54-38.ip.fastwebnet.it) |
| 11:41.56 | AsteriskAlbania | anybody here |
| 11:42.55 | *** join/#asterisk willy_1234 (n=icechat5@62.231.36.101) |
| 11:43.12 | nettie | hi guys, I would like to use g729 on my phone and alaw with the sip carrier. When I had 1.2.x I bought a license from digium, now I'm using trunk and seems it cant transcode call. Anyone know what I should check please? |
| 11:44.00 | nettie | <PROTECTED> |
| 11:45.01 | nettie | figured out I think |
| 11:46.07 | nettie | no I dont :) |
| 11:46.42 | willy_1234 | so can i modify the record enable macro to record all calls made and recieved as when i transfer calls the recording stops |
| 11:46.48 | willy_1234 | http://pastebin.ca/111549 |
| 11:47.33 | nettie | damn seems it's not supported on trunk |
| 11:48.33 | willy_1234 | argh i am can u recoard calls from queues?? |
| 11:49.02 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 11:51.06 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 11:51.53 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 11:56.48 | willy_1234 | So is it possible to modify the file? |
| 11:57.50 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
| 11:59.05 | backblue | ousk: what do you need? |
| 11:59.24 | backblue | AsteriskAlbania: what do you need? |
| 12:02.56 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
| 12:02.57 | willy_1234 | i got this answer http://pastebin.ca/111569 but i cant find the /b switch |
| 12:06.04 | daysmen3 | is it possible to implement call forwarding with hunt groups using asterisk extensions without resorting to AGI |
| 12:06.54 | *** join/#asterisk bkw_ (n=brian@asterisk/friend-and-developer/bkw) |
| 12:08.08 | [TK]D-Fender | daysmen3 : Yup |
| 12:08.41 | willy_1234 | i guess everyone is annoying me |
| 12:09.03 | jeffjohnson | why it won't be an init script created on asterisk compilatin? |
| 12:09.39 | [TK]D-Fender | jeffjohnson : there is anotehr make option to do that IIRC. |
| 12:10.28 | jeffjohnson | [TK]D-Fender: "make samples" didn't create an init script |
| 12:10.49 | [TK]D-Fender | jeffjohnson : That one wasn't supposed to... |
| 12:14.25 | jeffjohnson | [TK]D-Fender: uhm, in the README I don't find an option for creating an init script. I've looked in the Makefile, there exists "make config" but it can't find the init script to copy |
| 12:14.52 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 12:16.44 | [TK]D-Fender | jeffjohnson : there are init scripts in our source package.... go look.. |
| 12:16.45 | AsteriskAlbania | backblue: I need to make asterisk compatible with my billing ALEPO RBS |
| 12:18.11 | ousk | backblue: |
| 12:18.21 | [TK]D-Fender | jeffjohnson : I'll even give you a clue : /contrib/init.d |
| 12:18.21 | *** join/#asterisk Alpha232 (i=alpha232@198-144-137-40.dyn.megbb.net) |
| 12:18.47 | ousk | I was woundering why asterisk does not follow the sip standard (rfc) |
| 12:18.50 | jeffjohnson | [TK]D-Fender: ah thx |
| 12:18.55 | ousk | regarding the notify .. |
| 12:19.07 | daysmen3 | <[TK]D-Fender> please please please tell us how :-) |
| 12:19.38 | [TK]D-Fender | daysmen3 : Who is "us" and which amongst you am I addressing? |
| 12:20.19 | E-bola | lol |
| 12:20.58 | ousk | According to rfc 3265 ... should not Asterisk send out notify with out a subcription first |
| 12:21.43 | daysmen3 | <[TK]D-Fender> just me here - my girl friend gets kinda rank when i say us when referring to myself - i guess its an East London thing - your addressing ME |
| 12:22.09 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 12:22.53 | daysmen3 | E-bola: london slang for "angry" :-) |
| 12:23.14 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 12:23.22 | [TK]D-Fender | daysmen3 : You can't me ME.... *I'M* ME! You're YOU! |
| 12:24.13 | [TK]D-Fender | daysmen3 : Ok, so enough of my silly bit this morn' Explain what you'd like to do and IIRC your definition of "call hunting is a bit skewed. |
| 12:24.29 | *** join/#asterisk cytrak (n=kvirc@adelphi.geofocus.com) |
| 12:24.48 | Alpha232 | hrrm... I know i saw a posting from back in 2003 about using a USB voice modem with Asterisk... though google cache has since purged it... anyone have suggestions on where to find out if the Aztech UM9100 can work? |
| 12:25.07 | Alpha232 | though right now i'm just having a heck of a time getting it installed just as a plain modem lol |
| 12:25.08 | cytrak | does anyone know if the "duration" field on a cdr table of call records is in seconds or minutes ? |
| 12:25.34 | Vulture- | sec |
| 12:25.40 | cytrak | thanks |
| 12:25.42 | [TK]D-Fender | Alpha232 : No. Get a proper FXO device of some kind unless you feel like writing drivers for it yourself. Believe the odds that anyone else would have are astronomically against you. |
| 12:26.06 | Vulture- | wow... drivers for a new * device.. no thanx |
| 12:26.45 | *** part/#asterisk antiPosix (n=jkezar@159.105.109.200) |
| 12:26.53 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:27.14 | Alpha232 | [TK]D-Fender: grumble... looking to avoid throwing good money after bad and making the best with what I have... the sad part is, it works perfectly with NCH's software under winblows... |
| 12:28.04 | Alpha232 | [TK]D-Fender: plus i havn't seen many USB FXO devices out there |
| 12:28.07 | daysmen3 | <[TK]D-Fender>: i guess so - implementing a follow me so when a users extension is dialed it will go head and ring numbers one at a time or simultaneously |
| 12:28.54 | [TK]D-Fender | Alpha232 : Nope.... only the AStribank (but by its name you can tell it was built for *) |
| 12:29.01 | *** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 12:29.13 | [TK]D-Fender | daysmen3 : Sure all doable... |
| 12:29.58 | *** join/#asterisk yxa (i=lonari@cm121.gamma228.maxonline.com.sg) |
| 12:30.23 | daysmen3 | <[TK]D-Fender> - could you help me then - give me an idea how to accomplish this? |
| 12:31.12 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 12:32.26 | [TK]D-Fender | daysmen3 : "show application dial", the "DIALSTATUS" variable, and "show application gotoif" is all you need for this. |
| 12:33.40 | [TK]D-Fender | daysmen3 : If youwant variable forwarding in there as well that'll be some use of AstDB for setting some permanent values and some little scripts to set it up. |
| 12:33.44 | Alpha232 | [TK]D-Fender: it looks like the smaller Astribank is an FXS device though, providing a port to plug a set into, argh i need fxo so i can connect this to the telco side and answer inbound calls |
| 12:34.08 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 12:34.51 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 12:34.57 | Alpha232 | it seems like some manufactures also reverse the meaning of fxo and fxs... some by what the port does, others by what the port connects to, thats confusing |
| 12:37.29 | [TK]D-Fender | Alpha232 : Wel you have a decent choice of relatively inexpensive FXO options. TDM01B, X100P, SPA-3000, SPA-3102, A200 take your pick. |
| 12:38.24 | Sonderblade | i've installed asterisk from the debian package and it seems like the meetme module isn't included. anyone know why? |
| 12:38.42 | l-fy | Sonderblade > i think that meetme is dependable on zaptel |
| 12:38.55 | Sonderblade | l-fy: why? |
| 12:39.11 | *** join/#asterisk oej (n=oej@h-66-167-184-122.nycmny83.covad.net) |
| 12:39.32 | l-fy | Sonderblade > something weird with time samples |
| 12:40.04 | Sonderblade | l-fy: that IS weird |
| 12:40.07 | Alpha232 | [TK]D-Fender: well for the project I am playing with, we were looking for usb devices, because we can get a deal on a very small linux box but it lacks PCI or ISA, everything is onboard, all other expansion is by USB. |
| 12:40.18 | l-fy | Sonderblade > don't ask me, i use a conference server |
| 12:40.31 | Sonderblade | l-fy: whats that? |
| 12:40.45 | *** join/#asterisk javar (n=javar@200.118.174.253) |
| 12:41.51 | *** join/#asterisk TheCompWiz (n=TheCompW@wsip-68-109-200-102.mc.at.cox.net) |
| 12:42.16 | TheCompWiz | well... I'm baaaaaaaa'aaaaaaaaack! |
| 12:42.34 | TheCompWiz | anyone familiar with the trixbox distro? |
| 12:44.35 | *** join/#asterisk coppice (n=chatzill@136.202.17.210.dyn.pacific.net.hk) |
| 12:44.38 | daysmen3 | <[TK]D-Fender> : thanks ill chew on this - :-) |
| 12:44.55 | nortex | TheCompWiz, Still having problems? |
| 12:45.27 | *** part/#asterisk Alpha232 (i=alpha232@198-144-137-40.dyn.megbb.net) |
| 12:45.48 | [TK]D-Fender | TheCompWiz : That'd be CentOS. |
| 12:46.04 | TheCompWiz | [TK]D-Fender .... why would that make a difference? |
| 12:46.45 | [TK]D-Fender | Sonderblade : MeetMe requires Zaptel for timing. You can try the 3rd pary app_conference if you want an alternative. |
| 12:47.02 | [TK]D-Fender | TheCompWiz : What difference? |
| 12:47.30 | TheCompWiz | [TK]D-Fender CentOS vs other distro? |
| 12:48.36 | *** join/#asterisk creativx (n=creadure@196.82-134-19.bkkb.no) |
| 12:48.36 | [TK]D-Fender | TheCompWiz : If you've got a point, I'm not getting it..... |
| 12:49.10 | TheCompWiz | [08:46] [TK]D-Fender: TheCompWiz : That'd be CentOS. <-- why would CentOS be a problem? |
| 12:49.25 | [TK]D-Fender | TheCompWiz: You just asked if anyone was familiar with the trixbox distro. Who said anything about it being a PROBLEM?! |
| 12:49.44 | [TK]D-Fender | TheCompWiz : Lay off the crack man! |
| 12:50.15 | TheCompWiz | dang... too many channels... not enough brain cells in my head. I'm still having problems with all dialogs from the PBX not playing. |
| 12:50.36 | yxa | are the drivers for B410T out? |
| 12:50.38 | *** join/#asterisk potsboy (n=chrisg@196.211.16.202) |
| 12:50.42 | Sonderblade | [TK]D-Fender: which one is best? |
| 12:50.57 | rob0 | ~braincells |
| 12:51.39 | [TK]D-Fender | TheCompWiz : Oh well in that case.... read the channel topic. |
| 12:52.05 | [TK]D-Fender | Sonderblade : Whichever one you are most comfortable with that will compile * and any associated modules you want. |
| 12:54.00 | Sonderblade | [TK]D-Fender: so have you used any of them? |
| 12:54.20 | [TK]D-Fender | Sonderblade : Any of "what"? |
| 12:54.40 | [TK]D-Fender | ALL OF YOU LAY OFF THE CRACK! |
| 12:55.47 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 12:55.50 | Sonderblade | [TK]D-Fender: which of the conferencing modules |
| 12:56.10 | *** join/#asterisk Shark_y (n=paoloc@217-133-16-124.b2b.tiscali.it) |
| 12:56.14 | *** join/#asterisk bkw_ (n=brian@asterisk/friend-and-developer/bkw) |
| 12:56.32 | [TK]D-Fender | Sonderblade : I've only use app_meetme, but am preparing to test app_conference |
| 12:56.55 | willy_1234 | http://freepbx.org/trac/ticket/1052 |
| 12:57.01 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 12:57.58 | Sonderblade | [TK]D-Fender: guess ill try that then |
| 12:58.10 | [TK]D-Fender | willy_1234 : Not the place to be advertising that really... |
| 12:58.23 | [TK]D-Fender | Sonderblade : Whats wrong with trying Meetme? |
| 12:58.33 | *** part/#asterisk antgel (n=antony@pdpc/supporter/monthlybyte/antgel) |
| 12:58.48 | Sonderblade | [TK]D-Fender: i meant app_meetme |
| 12:58.57 | willy_1234 | i know but i was wondering if u might know the answer |
| 13:00.26 | [TK]D-Fender | willy_1234 : Nope... * is fine... FreePBX is broken, and most wouldn't care to touch it. |
| 13:00.49 | trelane_ | EVIL! |
| 13:00.58 | trelane_ | BANE OF SATAN BE GONE WITH THEE! |
| 13:01.04 | trelane_ | err... did someone say freePBX? |
| 13:01.24 | [TK]D-Fender | trelane : Actually... God would be the "bane of Satan". You would vie against Him? |
| 13:01.33 | willy_1234 | is there a way of seeting it up manually |
| 13:01.42 | willy_1234 | without freepbx |
| 13:01.52 | [TK]D-Fender | willy_1234 : obviously. |
| 13:01.58 | trelane_ | bane of satan is the demon at satan's left hand in demonology, of as in posession, not of as in Satan's bane (God) |
| 13:02.01 | willy_1234 | is it difficult |
| 13:02.02 | Sonderblade | i sure would like a free pbx... |
| 13:02.11 | [TK]D-Fender | willy_1234 : but the problem is you have to modify their crap, and GL with that... |
| 13:02.58 | willy_1234 | guess ill have to buy a manual on how to program asterisk then |
| 13:03.34 | trelane_ | willy_1234, www.voip-info.org? |
| 13:03.37 | trelane_ | there's a TON there |
| 13:03.39 | trelane_ | ~book |
| 13:03.41 | jbot | somebody said book was a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 13:03.45 | trelane_ | use that too. |
| 13:03.49 | [TK]D-Fender | willy_1234 :see the bot... |
| 13:04.20 | *** part/#asterisk javar (n=javar@200.118.174.253) |
| 13:04.28 | [TK]D-Fender | willy_1234 : Join the world of running-wheel-liberated hampsters! |
| 13:04.41 | trelane_ | willy_1234, I generally recommend a hard copy of that book, I always keep one on hand for the command reference in the rear |
| 13:04.53 | jbalcomb | willy_1234: Also, Digium offers a class on programming Asterisk. It's very good. |
| 13:04.55 | *** join/#asterisk viperdude (n=jon@195.74.96.117) |
| 13:04.57 | willy_1234 | think ill forget about queue recording |
| 13:05.15 | *** join/#asterisk mikebwilliams (n=mike@64.184.24.94) |
| 13:05.21 | willy_1234 | is that in the UK or ireland? |
| 13:05.21 | mikebwilliams | hey all |
| 13:05.36 | mikebwilliams | quick question: |
| 13:05.46 | viperdude | hi, i have a * box with disallow=all, allow=g726 in the peer for incoming calls, however when the call comes in it is ulaw? any ideas? |
| 13:05.50 | jbalcomb | trelane: by hard copy you mean download the pdf onto your laptop, PDA, or phone right? cause i know you aren't killing trees with /electronic/ documents. |
| 13:05.54 | [TK]D-Fender | jbalcomb : Should have seent he lightning storm here last night. It was f'n psychotic. 450K people out power. It was like Armadeggon. |
| 13:06.02 | mikebwilliams | is anyone else having problems with Asterisk formatting bad CANCEL messages? version 1.2.10 |
| 13:06.28 | *** join/#asterisk gursikh (n=guriskh1@dsl254-123-245.nyc1.dsl.speakeasy.net) |
| 13:06.33 | [TK]D-Fender | viperdude : peer is for OUTGOING calls. USER/friend is for incoming |
| 13:06.56 | [TK]D-Fender | jbalcomb : I did.... duplexed! |
| 13:07.03 | jbalcomb | [TK]D-Fender: nice! we had a similar storm here last thursday. several 'disaster areas' just east of me. |
| 13:07.17 | viperdude | <[TK]D-Fender> aha.... so if i have a seperate sip entry it should work right? |
| 13:07.26 | mikebwilliams | the message looks like this: CANCEL :18126633390@64.184.24.13:5070 SIP/2.0 |
| 13:07.34 | jbalcomb | [TK]D-Fender: did you run around with your sword raised high screaming "Bakaraiyo!!"? |
| 13:07.34 | mikebwilliams | shouldn't it be sip:18126633390? |
| 13:07.35 | [TK]D-Fender | jbalcomb : IT's original promise is that of a paperless society. BULLSHIT. I've singlehandedly killed entire forests with my print jobs.... |
| 13:08.14 | jbalcomb | [TK]D-Fender: Odd that you would be so EVIL yourself but not know of the 'bane of satan'. ;) |
| 13:08.31 | *** join/#asterisk asteriskmonkey (n=phil@h216-235-8-130.host.egate.net) |
| 13:09.23 | [TK]D-Fender | jbalcomb : No I was driving home and was hit by torrential downpour like God flushing the toilet on me. From 100 to 0% visibility in 2 seconds flat. Then I watched 3 power grids go dark as I was passing by.... my home included and its all KEYLESS ENTRY. I had to sleep on a friends couch last night. |
| 13:09.49 | gursikh | NO key at all? |
| 13:10.10 | [TK]D-Fender | gursikh : RFID cards & garage door opener (backed by another reader). |
| 13:10.18 | jbalcomb | [TK]D-Fender: haha.. you just learned about how the future will work for everyone. |
| 13:10.24 | gursikh | Lol |
| 13:10.49 | [TK]D-Fender | "Shame we have to die my dear, no-ones getting out of here alive. This time" |
| 13:11.52 | jbalcomb | [TK]D-Fender: I recommend a manual override 'just in case'. Commonly supported by the /old guy/ in many scifi movies as the failure of the future. |
| 13:12.15 | jbalcomb | [TK]D-Fender: oh wait... ummm... "that'll never happen" |
| 13:12.28 | jbalcomb | ;) |
| 13:12.33 | trelane_ | [TK]D-Fender, err you don't have your keyless entry on UPS? |
| 13:12.55 | willy_1234 | that seems to be a really good book |
| 13:12.55 | trelane_ | [TK]D-Fender, www.apc.com sheesh |
| 13:13.08 | trelane_ | willy_1234, don't tell me, tell jsmith when he's here |
| 13:13.11 | jbalcomb | trelane: seems not, i beleive [TK]D-Fender prefers the cometing technology 'OOPS' for his fail-over plan |
| 13:13.13 | trelane_ | he helped write the damn thing |
| 13:13.22 | willy_1234 | did he write it? |
| 13:13.28 | trelane_ | willy_1234, one of 3 authors |
| 13:13.34 | jbalcomb | s/cometing/competing |
| 13:13.35 | trelane_ | jbalcomb, seems so |
| 13:13.57 | trelane_ | jbalcomb, isn't OOPS slogan "Damn I fucked up, could I please sleep on the couch tonight?" |
| 13:14.42 | [TK]D-Fender | jbalcomb : Oh forgot to mention... no emergency lighting either. I'm sure that violates plenty of building codes..... |
| 13:15.14 | jbalcomb | [TK]D-Fender: actaully, emergency lighting makes me sad because we don't have any in our server room. :/ |
| 13:15.31 | trelane_ | jbalcomb, that |
| 13:15.35 | jbalcomb | trelane: haha, yes, i do think that was their tag line |
| 13:15.38 | trelane_ | s a network priority |
| 13:16.01 | *** join/#asterisk roche (n=roche@200.122.154.250) |
| 13:16.24 | jbalcomb | trelane: yeah, especially because without it i have three numbskulls with flashlights running around, pointing them in my face and asking if everythings going to be alright |
| 13:16.25 | [TK]D-Fender | jbalcomb : Yeah... neither do I... but I *DO* have a handy-dandy fire sprinkler above my server rack incase it goes up a degree or two by chance! |
| 13:16.54 | jbalcomb | [TK]D-Fender: oh...... now that is the most fabulous thing i can imagine |
| 13:17.21 | gursikh | degree or two? Isn't that a bit low tolerence? I've been in server rooms where temp fluctuates 5,6 degrees depending on time of day (amount of usage) |
| 13:17.45 | jbalcomb | [TK]D-Fender: that server might make a nice shelf to keep your incense burner on eh? |
| 13:17.53 | [TK]D-Fender | gursikh : Hypothetical/comical extrapolation. |
| 13:17.57 | trelane_ | jbalcomb, I've always been a firm believer that any main server room should have emergency lights, flashlights, the little floor tracklights similar to theaters that activate when the lights go out outlining where the racks are, food and water provisions (c rations work well here) and a small arms locker |
| 13:18.06 | [TK]D-Fender | jbalcomb : AAAAAOOOOUUUUUMMMM |
| 13:18.23 | trelane_ | jbalcomb, and scale that based on your company size |
| 13:18.38 | [TK]D-Fender | trelane : I should just bring my katana here.... |
| 13:18.46 | roche | People, If I want to do a Conferences when I have a Call from outside, I have to transfer the call to the extension where the conference is, and then I have to call the conference? There is another way to do it |
| 13:18.48 | roche | ? |
| 13:19.00 | trelane_ | [TK]D-Fender, katanas cut cabling, I generally recommend beanbag shotguns |
| 13:19.03 | [TK]D-Fender | jbalcomb : speaking of which, sensei asked me to bring my blade next week :) Looks like its time... |
| 13:19.07 | roche | Is the another way I mean? |
| 13:19.10 | jbalcomb | trelane: i just bring my practice sword, my harmonica, and some green tea. everything works out fine. |
| 13:19.17 | [TK]D-Fender | trelane : Oh... I was thinking PEOPLE... |
| 13:19.35 | trelane_ | [TK]D-Fender, if you're exceptionally adept fine |
| 13:19.41 | *** join/#asterisk apardo (n=apardo@87.217.147.51) |
| 13:19.45 | jbalcomb | trelane: that is why the practice sword is so handy |
| 13:19.47 | [TK]D-Fender | jbalcomb : What style do you practice? |
| 13:20.07 | jbalcomb | [TK]D-Fender: 70's martial arts movies retard style. you? |
| 13:20.15 | trelane_ | HAHAHAHAHA |
| 13:20.16 | *** join/#asterisk jero (n=jero@savoirfairelinux.net) |
| 13:20.22 | trelane_ | jbalcomb, well spotted! :) |
| 13:20.30 | jbalcomb | trelane: ;) |
| 13:20.34 | jero | hi |
| 13:21.07 | bkw_ | low |
| 13:21.07 | trelane_ | inbetweenish |
| 13:21.07 | jero | hehe |
| 13:21.07 | jbalcomb | bkw_: no shit, you gonna do some cool crap and make it public? |
| 13:21.07 | [TK]D-Fender | trelane : You haven't seen my golf swing then... NOONE IS SAFE. |
| 13:21.09 | [TK]D-Fender | jbalcomb : Katori Shinto. |
| 13:21.16 | bkw_ | jbalcomb, Kevin said I could make voices that sound almost 100% real for festival if you spend the HOURS it takes to tweak a voice |
| 13:21.26 | bkw_ | he showed me all the tools I need to make it possible |
| 13:21.29 | bkw_ | and how to get started |
| 13:21.37 | jbalcomb | [TK]D-Fender: all the same, your katana is not well suit to indoor use and a server is perhaps a less suited environment. |
| 13:21.44 | trelane_ | bkw_, but festival does sound like a real person! |
| 13:21.55 | bkw_ | no the default voices aren't tweaked very much |
| 13:22.00 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:22.04 | bkw_ | you can spend HOURS tweaking them |
| 13:22.15 | trelane_ | but they _do_ sound like a real person |
| 13:22.24 | bkw_ | yes but they are a bit raspy and robotic |
| 13:22.26 | Pj_ | trelane_: no, you can spend HOURS we tell you |
| 13:22.30 | *** join/#asterisk hmmhesays (n=ohyeah@66.173.103.110) |
| 13:22.38 | jbalcomb | bkw_: that is very cool. i've been meaning to mess with it more and more but it won't stick to the priorities list. |
| 13:22.41 | trelane_ | but by default they sound like a fairly well known public speaker |
| 13:22.49 | trelane_ | I've actually been to one of his talks |
| 13:22.53 | bkw_ | thats kevnin's voice |
| 13:22.57 | [TK]D-Fender | trelane : dubbyah?! |
| 13:22.58 | bkw_ | Kevin Lenzo |
| 13:23.11 | jbalcomb | bkw_: Can you make one sound like George W. Bush? |
| 13:23.18 | trelane_ | [TK]D-Fender, well I was thinking Steven W Hawkings, but if you want :) |
| 13:23.24 | trelane_ | jbalcomb, now THAT would take effort |
| 13:23.31 | bkw_ | If I have enough samples that sound about the same in tone and pitch |
| 13:23.35 | hmmhesays | so i'm looking for a small smtp program I can send emails with attachments on the command line |
| 13:23.35 | [TK]D-Fender | jbalcomb : that requires "codec_dopey.so" |
| 13:23.49 | jbalcomb | haha.. |
| 13:23.49 | bkw_ | but i'm going to do my voice first |
| 13:23.49 | hmmhesays | i tried uuencode with mailx and it is hit and miss |
| 13:24.07 | bkw_ | everyone wants a gay boy's voice in their PBX/SWITCH |
| 13:24.19 | jero | :) |
| 13:24.21 | hmmhesays | bkw_: lol |
| 13:24.30 | bkw_ | plus I can use it to mess with people |
| 13:24.31 | bkw_ | haha |
| 13:24.32 | hmmhesays | how about a flaming heterosexual? |
| 13:24.36 | jbalcomb | that cross legged chic from fatal attraction would be a good IVR voice |
| 13:24.59 | bkw_ | hmmhesays, I can have you record some stuff if you want me to make one of your voice :P |
| 13:25.04 | bkw_ | I need test subjects :P |
| 13:25.07 | jbalcomb | personally i would use dennis leary if i could |
| 13:25.17 | rob0 | hmmhesays: nail(1), often installed in place of mailx(1) |
| 13:25.20 | bkw_ | he doesn't have a voice that would make a voice well |
| 13:25.30 | hmmhesays | sure, what are we doing again?... |
| 13:25.37 | bkw_ | he has a very unique voice.. lots of tone and pitch changes |
| 13:25.56 | jbalcomb | [TK]D-Fender: i made that damn test queue for the admins to see if it worked the way they wanted.. now they are talking about having me do the recordings because i have a 'claming voice' |
| 13:26.06 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 13:26.12 | jbalcomb | s/claming/calming |
| 13:26.13 | MrChimpy | clamming voice? |
| 13:26.18 | *** part/#asterisk nettie (i=[U2FsdGV@85-18-54-38.ip.fastwebnet.it) |
| 13:26.24 | MrChimpy | clammy voice? |
| 13:26.29 | rob0 | nail(1) has the advantage of being a complete MUA. It can send via SMTP (even with TLS) and also read mail via IMAP (even with TLS!) |
| 13:26.33 | jbalcomb | yes, very /wet/ sounding.. ;) |
| 13:26.41 | [TK]D-Fender | jbalcomb : You should get some penecillin for that ;) |
| 13:27.01 | jbalcomb | [TK]D-Fender: i dont use drugs, hippie |
| 13:27.15 | hmmhesays | i just need to send mail, and no, nail is not installed it is mailx |
| 13:27.43 | hmmhesays | uuencode myfile.txt | mailx -s "booyeah" matty@matty.com doesn't seem to be working too well |
| 13:27.43 | rob0 | so, check out nail and perhaps get rid of mailx? |
| 13:28.03 | rob0 | what OS is this? |
| 13:28.26 | hmmhesays | fc5 |
| 13:28.47 | hmmhesays | yes i'm the devil |
| 13:29.00 | jbalcomb | hmmhesays: isn't there a switch to mailx for attachments and format? |
| 13:29.14 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
| 13:29.16 | hmmhesays | jbalcomb: in mutt there is |
| 13:29.22 | hmmhesays | mailx I don't think so |
| 13:29.32 | hmmhesays | although if you prove me wrong i'd be happy |
| 13:30.55 | rob0 | I'd think FC would have nail, at least as an optional package. In Slackware, nail replaced mailx back in 9.1 or so. |
| 13:31.00 | [TK]D-Fender | hmmhesays : No... only semi-evil. Quasi-evil. The Diet-Coke of evil.... just 1 calorie, not quite evil enough! |
| 13:31.41 | *** join/#asterisk iriga (n=adu@LSt-Amand-152-31-4-219.w82-127.abo.wanadoo.fr) |
| 13:32.01 | rob0 | yeah, I think mutt(1) can also be configured to work like mailx from the CLI, but that's a lot of work. |
| 13:33.27 | hmmhesays | i think i'm just missing something in my syntax here, uuencode does its thing, but mailx doesn't grab the attachment |
| 13:33.30 | *** join/#asterisk dioedu (n=dioedu@200.207.150.85) |
| 13:34.02 | dioedu | hello, what is the difference TDM2460B and TDM2460E ? |
| 13:34.28 | dioedu | There is no have any information in digium's site |
| 13:34.35 | rob0 | Right, mailx does not know how to construct MIME headers. You need a MUA which understands MIME. |
| 13:35.03 | rob0 | (Or, implement all that yourself and pipe to sendmail(1) directly.) |
| 13:35.08 | hmmhesays | i'm not using mime for the attachment |
| 13:35.41 | rob0 | oh, so it will be manually uudecoded on the receiving end? |
| 13:37.00 | hmmhesays | by the mail client |
| 13:38.01 | [TK]D-Fender | rob0 : Nobody understands mimes.....they have nothing to say.... |
| 13:38.22 | hmmhesays | ok... here is what I can get to work. I can uuencode the file, ex: uuencode myfile.txt myencodedfile.txt; mailx -s "my subject" matty@matty.com < myencodedfile.txt |
| 13:38.58 | hmmhesays | i don't want to have to write the myencodedfile.txt to the disk to use it |
| 13:40.12 | *** join/#asterisk coppice (n=chatzill@180.192.17.210.dyn.pacific.net.hk) |
| 13:40.44 | hmmhesays | i should be able to do this with pipes damnit |
| 13:41.44 | asteriskmonkey | whens 1.4 comming out? |
| 13:44.09 | [TK]D-Fender | asteriskmonkey : When its ready. |
| 13:44.26 | jbalcomb | asteriskmonkey: tomorrow. |
| 13:44.27 | [TK]D-Fender | asteriskmonkey : Oh, and Don keeps sending your customers to me for referrals :) |
| 13:44.37 | [TK]D-Fender | jbalcomb : lol.... sick bastard :) |
| 13:44.46 | jbalcomb | :) |
| 13:44.53 | mut | mew |
| 13:46.07 | *** join/#asterisk TeePOG (n=1234@dsl-145-156-117.telkomadsl.co.za) |
| 13:46.16 | TeePOG | good afternoon |
| 13:46.17 | *** join/#asterisk davidcsi (n=david@213.201.53.222) |
| 13:46.17 | jbalcomb | Asterisk 1.4 will be ready Real Soon Now. |
| 13:47.27 | jbalcomb | TeePOG: heya. How's SA today? |
| 13:47.27 | trelane_ | jbalcomb, lies! |
| 13:47.47 | davidcsi | hello people, quick question: i want to apply the polarity patch to chan_zap but some lines come back rejected, the patch i think was for 1.2.0, is i am applying it to 1.2.9, could that be the reason of the error?? |
| 13:48.16 | jbalcomb | trelane: (Real Soon Now) DEF: http://www.outpost9.com/reference/jargon/jargon_33.html |
| 13:48.17 | davidcsi | and is there a patch for 1.2.9? |
| 13:48.24 | TeePOG | i'm at my wits' end... i need a windows driver for a bloody Jabra a320s bluetooth dongle, but i've been googling all day with no result. All I could think of by now was to ask you people, who might be using them on your sites. |
| 13:48.49 | asteriskmonkey | [TK]D-Fender: good :) i want 10% |
| 13:49.41 | asteriskmonkey | im working on some bleeing edge aastra firmware :) gearing it for asterisk and call centrs |
| 13:50.14 | davidcsi | anyone? |
| 13:50.49 | jbalcomb | TeePOG: i dunno know about bloody bluetooth drivers. sorry. nothing on jabra.com? |
| 13:50.59 | [TK]D-Fender | asteriskmonkey : I mean as a reference you HIM to sell that FireWoks (or whatever its called now). |
| 13:51.05 | [TK]D-Fender | for* |
| 13:51.41 | *** join/#asterisk Slugs_ (i=1000@eacb02-00-cmmgga-70-34-178-219.atlaga.adelphia.net) |
| 13:51.45 | Slugs_ | morning. |
| 13:51.56 | hmmhesays | ahh got it |
| 13:51.58 | hmmhesays | sweetness |
| 13:52.53 | asteriskmonkey | [TK]D-Fender: yes its still call fireworxs |
| 13:53.21 | asteriskmonkey | i like calling it fworxs for short |
| 13:54.12 | asteriskmonkey | im writing a wicked script at the moment to traul networks and bring back all the sip handset macs and types :D |
| 13:54.28 | jbalcomb | TeePOG: you don't have the CD that came with the Jabra A320 device? |
| 13:54.37 | TeePOG | no jbalcomb... no, and no |
| 13:54.39 | *** join/#asterisk benjk_ (n=benjamin@f8a01-0357.din.or.jp) |
| 13:54.46 | TeePOG | it was a present |
| 13:55.20 | TeePOG | and i have a lovely jabra 620s headset standing by, but no go until i have the dongle going |
| 13:55.22 | [TK]D-Fender | asteriskmonkey, jbalcomb : sounds like you two should collaborate |
| 13:56.11 | asteriskmonkey | [TK]D-Fender: whats jbalcomb doing? |
| 13:57.35 | *** join/#asterisk IvyUK (n=mark@194.201.148.200) |
| 13:58.00 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 13:58.09 | jbalcomb | asteriskmonkey: I have written a script that collects my MACs and IPs, puts them in a MySQL DB |
| 13:58.40 | *** join/#asterisk FlyboySR22 (n=rsears@gateway.americanis.net) |
| 13:59.07 | asteriskmonkey | ah cool... are you using nmap? |
| 13:59.10 | jbalcomb | asteriskmonkey: I have another script that use the AMI to collect all peer information from the server and updates the DB with the additional information for username and extention |
| 13:59.24 | jbalcomb | asteriskmonkey: I'm using arpwatch |
| 13:59.58 | asteriskmonkey | i like using namp as it shows dev type also |
| 13:59.59 | asteriskmonkey | :D |
| 14:00.15 | jbalcomb | asteriskmonkey: it runs as a daemon and simply monitors for arp traffic. when it sees a change or a new device it updates its log. my script monitors its log. |
| 14:00.44 | jbalcomb | asteriskmonkey: you mean make and model? |
| 14:01.01 | asteriskmonkey | manufacturer |
| 14:01.12 | asteriskmonkey | the first 3 octets of the mac are the manu |
| 14:01.32 | jbalcomb | asteriskmonkey: you can match the first half of the MAC to get vendor and you can simple grab the info from the AMI in much more detail |
| 14:01.38 | asteriskmonkey | plus i can can collect any other crappy thing on the network |
| 14:01.41 | *** join/#asterisk s0lid (n=jlq@202.73.171.28) |
| 14:02.05 | *** join/#asterisk mvdk (i=mvdk@60-240-70-160-nsw-pppoe.tpgi.com.au) |
| 14:02.09 | asteriskmonkey | yes i like the id of using tha ami for extended detail |
| 14:02.21 | jbalcomb | asteriskmonkey: well, there is certainly more than one way to skin a cat |
| 14:02.25 | mvdk | How much do people here now about ZapRAS? |
| 14:02.44 | jbalcomb | asteriskmonkey: what are you doing with the information your collecting? |
| 14:02.55 | asteriskmonkey | yes indeed : hats of though , nice to see someone else making cool things |
| 14:03.13 | *** join/#asterisk HolyGod (i=nobody@got.securebinary.com) |
| 14:03.43 | asteriskmonkey | i think main purpose of mine was just to get all polycom/aastra phones on a network as all polycoms and aastras have a base xml config (one for each brand) but covers there entire lines |
| 14:03.45 | jbalcomb | TeePOG: Given the way the Docs read I'd say you might try search for Jabra or A320 on a P2P. |
| 14:03.47 | mvdk | The reason is that I'd like to initiate a connection from the Zaptel end..... |
| 14:03.58 | mvdk | (to another asterisk server) |
| 14:04.33 | mvdk | And yes, only as a backup connection |
| 14:04.37 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 14:04.54 | jbalcomb | asteriskmonkey: ah, I'm working on Z.I.P.P. - Ze IP Phone Provisioner. Sounds like similar projects. |
| 14:04.59 | *** join/#asterisk ManxPower (i=ewieling@170.sub-70-210-32.myvzw.com) |
| 14:05.08 | mvdk | (There's already a fast internet connection, and telco data calls are extortionate over here....) |
| 14:05.30 | jbalcomb | asteriskmonkey: so far I have my GXP-2000s and IP 501s auto configurating OoB. |
| 14:05.39 | TeePOG | yah jbalcomb, i was referred to the Widcomm drivers... that did'nt work at all, and now i have a bluetooth icon winking red at me in the system tray because the software can find no bluetooth hardware |
| 14:06.00 | jbalcomb | asteriskmonkey: I'm trying to make a web interface for it though so the helpdesk guys can add, edit, delete. |
| 14:06.27 | asteriskmonkey | jbalcomb: cool , i already made a web interface for mine.. the mac big was the last part :) |
| 14:06.46 | jbalcomb | TeePOG: the other option would be to guy buy one, install SW, and return it. :/ |
| 14:06.58 | IvyUK | hi all, does anyone have Inbound caller ID working from a BT PRI. the line is presenting it as i can see it when using a protocol analyzer but all i get is "Accepting overlap call from '' to '123456' on channel 0/1, span 1" |
| 14:07.30 | jbalcomb | asteriskmonkey: shall we exchange code and notes? /j #ZIPP if you'd like to talk more. i gotta get back on my syslog server. |
| 14:07.42 | asteriskmonkey | k |
| 14:08.01 | *** join/#asterisk Waverly360 (n=mirc@65.169.134.2) |
| 14:08.23 | Pj_ | anyone using BT200 ? (or 100 I'd bet) my users complain that when calling outside the callee says he hears my users "far away"... any way to up the volume sent ? |
| 14:08.50 | mvdk | Pj_: are you certain that's what they mean? |
| 14:09.22 | Pj_ | that it's a volume issue ? yeah |
| 14:09.39 | Pj_ | I got -some- smart users and they confirmed it |
| 14:09.43 | TeePOG | thanks jbalcomb, last resort, yeah, but thanks |
| 14:09.44 | mvdk | Oh, OK, that's sometimes how I've heard echo issues described.... :) |
| 14:10.01 | Pj_ | but no problem between the phones themselves |
| 14:10.15 | mvdk | So which paths does the call cross? |
| 14:10.20 | Pj_ | mvdk: I'll keep that in the back of my head though ;) |
| 14:10.26 | hmmhesays | now my rxfax is freaking out man |
| 14:10.27 | mvdk | Is it entirely IP? |
| 14:10.30 | Pj_ | sip phone - * - E1 line |
| 14:10.36 | mvdk | Ah, OK |
| 14:10.45 | mvdk | Check out the gain on the E1 |
| 14:10.46 | Pj_ | but with a softphone it works ok |
| 14:10.51 | mvdk | Oh, sorry |
| 14:11.01 | Pj_ | I tried but it doesn't change the volume as much as the sound compression |
| 14:11.12 | mvdk | OK, you need to check your phone, the manual would have it somewhere, surely.... |
| 14:11.24 | Pj_ | hmmm |
| 14:11.25 | Pj_ | :) |
| 14:11.28 | hmmhesays | <PROTECTED> |
| 14:11.56 | mvdk | Oh, BTW, I managed to get Hylafax and Asterisk working together, I'm quite stoked :) |
| 14:12.04 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:12.24 | *** join/#asterisk dandan (i=dandan@pacanka.com) |
| 14:12.26 | dandan | hello all :) |
| 14:12.58 | mvdk | I don't suppose that anyone has had a chance to see my question about Asterisk data call facilities? |
| 14:13.31 | dandan | my users complained that when retrieving Vm, the sound is REAAAALY REAAAALY LOW... i can see that that has been discussed before on the mailing list but I do not know if that has been resolved. Anyone can comment? |
| 14:14.12 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 14:16.20 | *** join/#asterisk jalsot (n=tamas@abacus.eworldcom.hu) |
| 14:17.18 | *** join/#asterisk gandhijee (n=gandhije@65.169.134.2) |
| 14:17.44 | hmmhesays | TIFF directory is missing required "StripOffsets" field. anyone run into that with spandsp? |
| 14:17.53 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 14:18.09 | *** join/#asterisk pnlarsson (n=niklas@c83-248-2-120.bredband.comhem.se) |
| 14:19.19 | Waverly360 | there once...was a man....from nantucket... |
| 14:20.27 | bkw_ | who compiled asterisk in a bucket |
| 14:20.47 | Waverly360 | but the bits fell through the cracks... |
| 14:21.21 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 14:22.07 | sevard | the cpu could just not last |
| 14:22.16 | Waverly360 | heh |
| 14:22.17 | sevard | so he bought windows and said ohfuckit. |
| 14:22.24 | Waverly360 | w0000. |
| 14:22.29 | dandan | :> |
| 14:22.33 | sevard | ;) |
| 14:22.45 | eKo1 | lol |
| 14:22.51 | Waverly360 | hrm...too much alcohol last night |
| 14:23.11 | sevard | my tummy is happy at me right now - i avoided it last night. |
| 14:23.39 | *** join/#asterisk anthm (n=anthm@65.169.134.2) |
| 14:23.39 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:23.46 | Waverly360 | hah...yeah...hard to do that visiting Chicago..I swear it was the hot women's fault at that bar. |
| 14:23.48 | *** join/#asterisk Jason99 (n=jason@jason.unitz.ca) |
| 14:24.01 | eKo1 | Are you sure they were women? |
| 14:24.03 | sevard | every error you ever make is some hot chick's fault |
| 14:24.06 | sevard | understand that. |
| 14:24.42 | Waverly360 | eKo1: if they weren't...I'm going to be re-evaluating myself. |
| 14:24.57 | Waverly360 | yeah well...prolly right sevard |
| 14:25.05 | hmmhesays | bah |
| 14:25.12 | hmmhesays | anyone else get this spandsp TIFF directory is missing required "StripOffsets" field with spandsp? |
| 14:25.34 | Waverly360 | Nah...never have hmmhesays |
| 14:25.37 | Waverly360 | sorry |
| 14:25.46 | hmmhesays | something funky going on here |
| 14:28.03 | *** join/#asterisk lorinc (n=ang@caracas-1632.adsl.interware.hu) |
| 14:28.12 | *** join/#asterisk littleball (n=littleba@cm134.epsilon174.maxonline.com.sg) |
| 14:28.17 | dandan | my users complained that when retrieving Vm, the sound is REAAAALY REAAAALY LOW... i can see that that has been discussed before on the mailing list but I do not know if that has been resolved. Anyone can comment? |
| 14:28.28 | sevard | hmmhesays: I know how to call forward on busy on say an ATA or other sip device.. it still fails me how to do it in the asterisk dialplan though, do you k now? |
| 14:28.32 | littleball | hello, does codec translation introduce the lose of DTMF tone? |
| 14:28.44 | littleball | do g729/g711 translation |
| 14:28.52 | eKo1 | littleball: it shouldn't |
| 14:31.44 | *** join/#asterisk trbldwine (i=troubled@adam.ur.northwestern.edu) |
| 14:32.42 | hmmhesays | ahh that is what happens when a fax doesn't come through properly |
| 14:32.49 | littleball | eKol, but in my situation, it does affect for some route |
| 14:33.05 | littleball | hmmhesays? what? |
| 14:33.53 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 14:33.58 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.124) |
| 14:34.02 | *** join/#asterisk viler (i=1000@200.114.70.228) |
| 14:35.01 | eKo1 | lilalinux: if you use ulaw/alaw, does the problem go away? |
| 14:35.34 | *** join/#asterisk coppice (n=chatzill@185.202.17.210.dyn.pacific.net.hk) |
| 14:36.02 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 14:36.43 | hmmhesays | littleball: different subject |
| 14:37.05 | *** join/#asterisk ghinckley6 (n=Spam@209.26.206.110) |
| 14:37.23 | hmmhesays | bitch |
| 14:37.29 | sevard | ^^ question above, whore. |
| 14:37.39 | littleball | eKok. yes |
| 14:37.43 | sevard | what happened to your big H? you feeling submissive today? |
| 14:37.43 | Waverly360 | yeah...whore... |
| 14:37.45 | hmmhesays | sevard: call forward? |
| 14:37.49 | hmmhesays | cf on busy? |
| 14:37.54 | sevard | right, in the dialplan |
| 14:38.01 | littleball | eKok, if using g711, the problem go away |
| 14:38.06 | n9urk | how can I get the caller id info or other info before it is inserted into the cdr log? |
| 14:38.08 | ghinckley6 | got a script plays dial and extension then waits. Question how do i get to dial and exten after say 20 sec |
| 14:38.14 | sevard | there no b extension is there? and i don't see anything in Dial() |
| 14:38.17 | littleball | but if using g729, some phone works and others doesn;t |
| 14:38.41 | n9urk | is there any other way besides inserting it into my own db during the call? |
| 14:38.41 | [TK]D-Fender | sevard : AstDB to store forwarding info and a good stdexten macro |
| 14:39.02 | eKo1 | littleball: are you using inband dtmf? |
| 14:39.22 | hmmhesays | exten => 123,1,Dial(SIP/sevard); exten => 123,1,Goto(s-${DIALSTATUS}); exten => s-BUSY,1,Dial(SIP/SomewhereElse) |
| 14:40.03 | *** part/#asterisk davidcsi (n=david@213.201.53.222) |
| 14:40.32 | sevard | oh yeah. |
| 14:40.52 | sevard | forgot about that bit. |
| 14:41.02 | littleball | no |
| 14:41.05 | hmmhesays | you owe me one |
| 14:41.06 | littleball | eKol. |
| 14:41.31 | sevard | you want a bj? |
| 14:41.32 | eKo1 | littleball: what are you using then. And on what phones does it work and not work? |
| 14:41.38 | Waverly360 | .... |
| 14:41.52 | Waverly360 | ooookay |
| 14:42.01 | littleball | eKol,rfc2833 |
| 14:42.15 | hmmhesays | sevard: no |
| 14:42.24 | hmmhesays | get me some biznaz, i'm out of a job soon |
| 14:42.38 | eKo1 | littleball: is that set globally? |
| 14:43.11 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.21.68.Dial1.SanJose1.Level3.net) |
| 14:44.23 | sevard | hmmhesays: qualifications / job wanted / previous job / reason for leave |
| 14:45.56 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 14:45.56 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 14:49.33 | dandan | my users complained that when retrieving Vm, the sound is REAAAALY REAAAALY LOW... i can see that that has been discussed before on the mailing list but I do not know if that has been resolved. Anyone can comment? |
| 14:50.15 | Waverly360 | only voicemail? |
| 14:50.25 | *** join/#asterisk wiseguy_ (n=chivilis@infospalvos.lt) |
| 14:50.26 | [TK]D-Fender | dandan : there is a VM gain option in voicemail.conf, otherwise up the volume on your incoming call interface. |
| 14:50.27 | dandan | yes, while checking from the outside |
| 14:50.28 | wiseguy_ | hello |
| 14:50.47 | *** join/#asterisk Arnar (n=abi2@landi.oddi.is) |
| 14:50.50 | Waverly360 | hrm |
| 14:50.55 | wiseguy_ | i want to replace the standart ringing tone (to my music)? which command i should use? |
| 14:50.58 | Waverly360 | using a pri or analog lines? |
| 14:50.58 | dandan | [TK]D-Fender: can't find it... |
| 14:51.05 | dandan | btw. hey :) |
| 14:51.28 | dandan | [TK]D-Fender: all other outside calls are fine... |
| 14:51.35 | *** join/#asterisk Arnar_ (n=arnarb@landi.oddi.is) |
| 14:51.48 | dandan | oh one more thing, where should I look, where callers complain about voice fading in and out? |
| 14:52.10 | tzanger | dandan: first things first, record the audio and capture the data traffic (if applicable) |
| 14:52.28 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 14:52.43 | wiseguy_ | anybody, help :) |
| 14:52.47 | tzanger | I never ever ever trust the description a user gives of voice trouble. Yes there may be trouble but I find that "crackle" doesn't mean crackle, "echo" doesn't mean echo, and "fading" very likely never ever ever means fading |
| 14:52.49 | dandan | tzanger: i will, it is just that I heard this complain today, I have never experienced it though :/ |
| 14:53.09 | tzanger | tell them to tell the other person on the line to stop fucking around with the handset :-) |
| 14:53.17 | dandan | :) |
| 14:53.20 | dandan | too easy huh :) |
| 14:53.42 | [TK]D-Fender | wiseguy_ : "show application dial" |
| 14:53.59 | Jason99 | I have three Asterisk servers (A,B,C). Server A has a Sangoma card and is connected to a PRI. Server C is a hosted PBX system. Server B simply routes calls. In this specific case, a Linksys phone attached to Server C hangs up and sends a BYE to C. Server C sends a BYE to server B. Server B doesn't send BYE to server A, it sends another INVITE for some reason and it doesn't hangup the PSTN call. Anyone know what would cause this and how to fix it? |
| 14:55.18 | *** join/#asterisk bpiper (n=bpiper@70.159.49.40) |
| 14:55.21 | *** join/#asterisk ys4321 (i=ys4321@61.247.229.200) |
| 14:55.28 | hmmhesays | sevard:hahaa |
| 14:56.50 | ys4321 | Hi! Has anyone used the app_speech_utils.c? |
| 14:56.59 | wiseguy_ | [TK]D-Fender: really, i wan't to stop background music when the call is answered |
| 14:58.03 | tzanger | Jason99: interesting but no, I haven't got a clue on that one :-) |
| 14:58.11 | littleball | eKol. yes |
| 14:58.15 | littleball | it is set globally? |
| 14:58.17 | littleball | why? |
| 14:58.51 | ys4321 | Has anyone used voice recognisiton with asterisk? |
| 14:59.02 | *** join/#asterisk ESCulapio__ (n=ESCulapi@200.88.44.66) |
| 14:59.09 | wiseguy_ | ys4321: ivr? |
| 14:59.16 | Jason99 | tzanger : Thanks for reading it :) hehe |
| 14:59.17 | *** part/#asterisk Arnar (n=abi2@landi.oddi.is) |
| 14:59.30 | ys4321 | yes! |
| 14:59.42 | sevard | hmmhesays: why isn't this very simple dialplan working? http://pastebin.ca/111921 |
| 15:00.06 | *** join/#asterisk jbalcomb (n=jbalcomb@216.28.180.158) |
| 15:00.28 | [TK]D-Fender | sevard : FUGLY |
| 15:00.29 | wiseguy_ | ys4321: waitexten |
| 15:00.40 | [TK]D-Fender | sevard : Maybe some SLI output of it would be helpful |
| 15:00.43 | tzanger | [TK]D-Fender: hey that's enough of the personal attacks :-) |
| 15:01.12 | ys4321 | wiseguy_: The asterisk documentation talks about a speech recognition module which can be used in the ivr. |
| 15:01.17 | sevard | [TK]D-Fender: how is that fugly? |
| 15:01.24 | sevard | [TK]D-Fender: it's hmmhesays's stuff |
| 15:01.43 | [TK]D-Fender | tzanger : Amazing how the words "don't take this personally but..." are almost invariably followed by a rather direct and peronsl insult, no? :) |
| 15:01.52 | [TK]D-Fender | hmmhesays : FUGLY! |
| 15:01.53 | tzanger | [TK]D-Fender: yep :-) |
| 15:02.07 | *** join/#asterisk pnlarsson (n=niklas@c83-248-2-120.bredband.comhem.se) |
| 15:02.54 | ys4321 | Anyone has used speech_create(engine) application? |
| 15:03.33 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 15:04.08 | hmmhesays | so, spandsp is not much good at accepting multipage faxes |
| 15:04.21 | sevard | hmmhesays: that doesn't work... |
| 15:04.21 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 15:05.55 | [TK]D-Fender | hmmhesays : Works decent for me... |
| 15:06.08 | [TK]D-Fender | sevard : PB up the CLI output. |
| 15:06.29 | nestar | [TK]D-Fender: same goes for "i'm not trying to be a dick, but...." |
| 15:06.36 | [TK]D-Fender | sevard : Actually... the Goto is clearly bad |
| 15:06.48 | sevard | yes, it's the GoTo that's failing |
| 15:07.04 | [TK]D-Fender | sevard : http://pastebin.ca/111926 |
| 15:07.09 | [TK]D-Fender | sevard : there |
| 15:08.11 | sevard | stupid syntax. |
| 15:08.27 | ys4321 | requesting help for speech recognition application on asteisk. Can anyone throw some light on how to use the speech_create application? |
| 15:09.02 | hmmhesays | so multi page faxes, rx_fax |
| 15:10.08 | *** part/#asterisk gursikh (n=guriskh1@dsl254-123-245.nyc1.dsl.speakeasy.net) |
| 15:10.40 | [TK]D-Fender | hmmhesays : Yup. |
| 15:11.02 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 15:11.35 | hmmhesays | [TK]D-Fender: not working for me, does it write more than one tiff file for you? |
| 15:11.51 | [TK]D-Fender | hmmhesays : Nope. MTIFF that I convert to PDF |
| 15:12.20 | hmmhesays | what are you using |
| 15:12.34 | macTijn | xtc perhaps ? :) |
| 15:13.07 | hmmhesays | cause rx_fax seems to write 1 tif file |
| 15:13.31 | macTijn | a tiff can have multiple pages in it |
| 15:13.35 | macTijn | it's a container format |
| 15:16.07 | *** join/#asterisk momelod (n=momelod@HSE-Montreal-ppp134142.qc.sympatico.ca) |
| 15:16.15 | momelod | greetings people. |
| 15:16.30 | momelod | im having some issues with polycom ip501 sets |
| 15:16.43 | momelod | the volume is too low, for transmit and recieve |
| 15:16.59 | momelod | how could i pump it up a smidge? |
| 15:18.10 | ManxPower | momelod, there are volume keys on the phone. |
| 15:18.14 | coppice | hmmmhesays: maybe you should look at what is in the TIFF file |
| 15:20.03 | momelod | ManxPower yes, but those dont effect my voice leaving the phone |
| 15:20.17 | momelod | they only increase/decrease what i hear |
| 15:20.26 | momelod | the people i call complain that my voice is too low |
| 15:20.55 | ManxPower | momelod, then you would have to look it up in the Polycom Admin Manual, but if you have Zap interfaces to the PSTN, it would be easier to just increase the txgain |
| 15:21.16 | momelod | ahh |
| 15:21.26 | [TK]D-Fender | hmmhesays : MTIFF ;multi-page TIFF |
| 15:21.46 | momelod | i couldnt find anything in the polycom admin manual, ill try the txgain tweak |
| 15:21.48 | momelod | thank you |
| 15:22.06 | momelod | is that something i would configure in zapata.conf? |
| 15:22.15 | ManxPower | Yes. |
| 15:23.33 | momelod | your the best :) |
| 15:23.55 | hmmhesays | my bad, kview doesn't show the multipage |
| 15:24.23 | coppice | there is no such thing as MTIFF. TIFF is always multi-page |
| 15:25.56 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 15:26.05 | ManxPower | Many TIFF viewers will only display the first page. |
| 15:26.12 | *** join/#asterisk vooduhal (n=vooduhal@tc-proxy2.catt.com) |
| 15:27.05 | vooduhal | Hey guys. Question. If an agent is a member of multiple queues, and I set the weight on one queue, will that queue take priority over the other? |
| 15:27.09 | *** join/#asterisk iq (n=IQ@unaffiliated/iq) |
| 15:27.18 | hmmhesays | well my fax to email is working |
| 15:27.43 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 15:28.59 | hmmhesays | ManxPower: yeah that was my problem |
| 15:29.09 | hmmhesays | now I have a nice little script to resize the image and email me |
| 15:29.11 | hmmhesays | sweetness |
| 15:29.22 | eKo1 | I'm amazed faxing is working for you period. |
| 15:29.40 | hmmhesays | pretty good success rate |
| 15:29.41 | hmmhesays | it seems |
| 15:29.45 | nestar | i never could get spandsp to compile |
| 15:29.56 | hmmhesays | sixtel and vonage both for incoming seem to be working ok |
| 15:30.12 | eKo1 | Cool. |
| 15:30.15 | nestar | i always envisioned having it spit the fax directly to my printer |
| 15:30.46 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 15:30.49 | hmmhesays | nestar: i just compiled it fine with 1.2.10 |
| 15:30.56 | [TK]D-Fender | nestar : I'm working to have my GUI provider integrate that.... |
| 15:31.20 | hmmhesays | that shouldn't be a problem |
| 15:31.31 | watchy | hey tk it seems some folks are saying when they dialout it doesnt seem to be catching all of the digits |
| 15:31.34 | [TK]D-Fender | hmmhesays : Nope, jsut a small task |
| 15:31.40 | nestar | hmmhesays: dunno. |
| 15:31.42 | watchy | any idea why what would cause that |
| 15:31.49 | [TK]D-Fender | watchy : Cahnge your local dialplan |
| 15:32.12 | watchy | how would that affect missing #s? |
| 15:32.20 | watchy | im just curious im a newb remember |
| 15:32.36 | momelod | ManxPower: when i make an adjustment to the txgain, do i need to reload the zaptel module? and do i need to reboot asterisk? |
| 15:32.37 | [TK]D-Fender | watchy : Polycom's automatically start reformatting invalid #'s to match valid string in wierd ways. |
| 15:32.47 | [TK]D-Fender | momelod : zaptel |
| 15:32.57 | momelod | ty |
| 15:33.13 | watchy | so format the local dialplan in sip.cfg? |
| 15:33.36 | [TK]D-Fender | watchy : yup |
| 15:33.50 | watchy | ok thats a great answer you know i think ill have to do that |
| 15:33.58 | watchy | its using the default one |
| 15:34.03 | *** join/#asterisk Dr-Linux|work (n=Linux@202.125.139.198) |
| 15:34.10 | watchy | you should get paid to irc here |
| 15:34.34 | [TK]D-Fender | watchy : My rates are very accessable :) |
| 15:34.44 | watchy | haha |
| 15:34.55 | hmmhesays | As are mine |
| 15:34.58 | hmmhesays | <snipe> |
| 15:35.05 | [TK]D-Fender | watchy : I do full-service remote installs. |
| 15:35.32 | watchy | tk: nice what os do you use? |
| 15:35.37 | watchy | well distro |
| 15:36.42 | *** join/#asterisk champster (n=champste@AH.tescogroup.com) |
| 15:37.59 | *** part/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 15:40.45 | *** join/#asterisk champster (n=asterisk@AH.tescogroup.com) |
| 15:40.53 | vooduhal | In queues.conf, if I set timeout=0 will that disable the timeout? |
| 15:40.55 | asterboy | mornin |
| 15:41.33 | hmmhesays | yes it is |
| 15:42.57 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 15:44.09 | momelod | whats the best way to reload the zaptel module? when i try to unload it, i cant because other modules depend on it |
| 15:44.53 | momelod | i just noticed an /etc/init.d/zaptel script.. could i use that to reload it? |
| 15:45.39 | cytrak | hey guys I have added a menu on my dial plan as I image a lot of you may have also done so... but I have some serious issues with the DTMF tones not been completly recognized when some people call |
| 15:46.19 | cytrak | I had vonage at home (which I no longer have) and that sucked I would enter an extension for a phone like 1555 and asterisk would only see 555 |
| 15:47.11 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 15:47.17 | cytrak | the same goes for my menu ... I have options 1 through 5 and also give the user the option to enter the extension number if they know it and I get the same issue |
| 15:47.58 | cytrak | I have set the digit and response timeout to over 15 to see if it makes any difference |
| 15:48.32 | asterboy | anyone here connecting to irc.linuxfromscratch.org? |
| 15:48.40 | asterboy | can't seem to get on this morning |
| 15:48.43 | *** join/#asterisk brif8 (n=Administ@ns1.ttienterprises.org) |
| 15:49.15 | *** join/#asterisk marly (i=marly@CPE-24-209-183-224.wi.res.rr.com) |
| 15:49.47 | brif8 | What is the current going rate for PBX co-location? something that can handle 50 incoming numbers +/- 30 concurrent calls ? |
| 15:49.58 | websae | hey brif8 |
| 15:50.50 | n9urk | what do I need to do to get musiconhold to work in 1.2.10? |
| 15:50.55 | n9urk | do I have to install mpg123? |
| 15:50.58 | watchy | yes |
| 15:51.08 | watchy | goto ast source dir type make mpg123 |
| 15:51.22 | watchy | then i installed it by going into mpg123 dir and make install |
| 15:51.30 | watchy | but there maybe another way i didnt look i was in a hurry |
| 15:52.21 | n9urk | watchy: is there a dep lib that I need? I get this error |
| 15:52.22 | n9urk | : undefined reference to `xfermem_done' |
| 15:52.22 | n9urk | collect2: ld returned 1 exit status |
| 15:52.22 | n9urk | make[2]: *** [mpg123] Error 1 |
| 15:52.37 | momelod | sweet that txgain thing worked |
| 15:52.43 | watchy | what os? |
| 15:52.49 | n9urk | deb sarge |
| 15:52.53 | momelod | next question :) |
| 15:52.58 | watchy | hmm |
| 15:53.08 | momelod | where can i download ringtones for my polycom ip501? |
| 15:53.21 | n9urk | watchy: do I need mpg123 to pay gsm files for moh? |
| 15:53.24 | hmmhesays | i am so sick of people freaking changing things while i'm troubleshooting |
| 15:53.30 | watchy | no idea i play mp3s |
| 15:53.39 | watchy | i think everyone plays mp3s for moh if im not mistaken |
| 15:53.43 | watchy | im a newb |
| 15:53.49 | n9urk | watchy: same here, but I think I saw that it had changed |
| 15:53.54 | watchy | i guess it converts them on the flt to gsm though |
| 15:54.00 | n9urk | so you could play them |
| 15:54.03 | watchy | it mighta i havent checked :/ |
| 15:54.12 | n9urk | what ver you have? |
| 15:54.18 | watchy | i know my setup is mp3s that come with asterisk |
| 15:54.33 | watchy | its ghetto music but it works im probably gonna have it stream a shoutcast eventually |
| 15:54.37 | n9urk | anybody know why mpg123 won't compile? |
| 15:54.48 | hmmhesays | why do you need it? |
| 15:54.50 | watchy | google that error n9urk i bet you find the answer |
| 15:54.58 | watchy | hmm: MOH |
| 15:55.01 | n9urk | hmmhesays: Music oh Hold |
| 15:55.25 | watchy | if you setup MOH to work and dont got mpg123 prepare for transfer nightmares |
| 15:55.28 | watchy | it breaks them |
| 15:55.29 | hmmhesays | asterisk supports mp3's natively now |
| 15:55.51 | watchy | n9urk: look into that now that its native |
| 15:55.58 | n9urk | hmmhesays: all I get is static |
| 15:56.01 | *** join/#asterisk LoneShadow (n=duh@59.92.163.63) |
| 15:56.03 | watchy | i had mine setup to mpg123 and moved it to a new box and had it all setup |
| 15:56.28 | n9urk | hmmhesays: what do you have to do to get it to play the mp3s then? I do have 1.2.10 |
| 15:56.33 | LoneShadow | Hello |
| 15:56.39 | n9urk | watchy: Google is not my friend today |
| 15:56.48 | n9urk | it didn't return anything usefull |
| 15:56.49 | LoneShadow | does "Got SIP response 400 "Bad Request -- cc1-n05" back from xxx.xxx.xxx.xxx" mean anything ? |
| 15:56.57 | watchy | n9urk: i never even installed debian or i would try to help |
| 15:57.07 | n9urk | thanks anyway ;)watchy |
| 15:57.11 | watchy | i use gentoo because once its installed it doesnt even come with ftp |
| 15:57.18 | watchy | its plain as you can get i hate bloat |
| 15:57.25 | n9urk | deb is pretty plain |
| 15:57.30 | n9urk | it does not have ftp with it |
| 15:57.35 | watchy | wow |
| 15:57.39 | watchy | hows the installer? |
| 15:57.50 | watchy | i install gentoo manually |
| 15:57.57 | watchy | its kinda annoying but its ok |
| 15:58.01 | Qwell | pfft, installers are for chumps |
| 15:58.06 | LoneShadow | deb is addictive :D |
| 15:58.06 | n9urk | pretty easy, ubuntu is better for installing, but a little more bloated. Debian was straightforward to isntall |
| 15:58.24 | watchy | well these new live cds come with a X installer on gentoo |
| 15:58.30 | n9urk | hmmhesays: do you have any more advice on how to get MOH to play and not just have static? |
| 15:58.40 | watchy | and i havent messed with it enough to make it not freaking install X when it installs gentoo |
| 15:58.46 | watchy | i dont want X installed |
| 15:58.49 | watchy | x sucks |
| 15:58.53 | LoneShadow | you can install ubuntu from debootstrap and add packages you need, I have a pretty good asterisk + icewn + ubunutu dapper on 1 gb CF card |
| 15:59.01 | Qwell | watchy: use -X |
| 15:59.05 | Qwell | USE -X rather |
| 15:59.06 | vooduhal | In queues.conf is there a timeout default or can I just set timeount=0 for no timeout? |
| 15:59.28 | [TK]D-Fender | watchy : Preferences for Slackware & CentOS |
| 15:59.33 | LoneShadow | oops, ubunty breezy actually, but dont see any diff installing dapper as well on 1GB card |
| 15:59.51 | *** join/#asterisk truz_`24 (n=truz_`24@74.129.166.232) |
| 15:59.55 | Qwell | vooduhal: default timeout appears to be 15s |
| 16:00.03 | n9urk | anyone able to help me with MOH all I get is static |
| 16:00.10 | n9urk | I have 1.2.10 |
| 16:00.19 | vooduhal | Qwell, can it be disabled? |
| 16:00.25 | Qwell | vooduhal: just set timeout=0 |
| 16:00.45 | vooduhal | Qwell, thanks. That's what I thought. |
| 16:01.22 | Qwell | bbl |
| 16:02.29 | *** join/#asterisk RoyK (n=roy@a217-118-45-74.bluecom.no) |
| 16:02.40 | n9urk | nobody knows? |
| 16:02.41 | marly | I'm setting up IAX2, and I'm trying to set the minregexpire and maxregexpire from 60 seconds to about 2 seconds, and whatever I change it to doesn't seem to take effect. Any clues on on what I'm doing wrong, or am I going about it the wrong way? |
| 16:03.12 | n9urk | thanks anyway have a good day watchy and all |
| 16:03.29 | hmmhesays | i just improved my faxing script about a million times |
| 16:04.53 | *** join/#asterisk tomlobato (n=tomlobat@201-43-224-3.dsl.telesp.net.br) |
| 16:07.24 | *** join/#asterisk burnproof (n=jsharryp@210.213.240.52) |
| 16:10.39 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 16:10.51 | CunningPike | n9urk: What's your setup? |
| 16:10.58 | CunningPike | n9urk: Are you a ham? |
| 16:11.06 | n9urk | CunningPike: Yeah |
| 16:11.14 | n9urk | CunningPike: deb sarge 1.2.10 |
| 16:11.23 | CunningPike | n9urk: VA7IRL |
| 16:12.06 | n9urk | CunningPike: you not update your QRZ profile? |
| 16:12.17 | CunningPike | n9urk: What does it say? |
| 16:12.33 | n9urk | it just has your basic demographic info |
| 16:12.44 | n9urk | and your birth year |
| 16:12.56 | CunningPike | n9urk: Oh, ya - what more do people need to know :) |
| 16:13.11 | n9urk | you don't have a bio? ;) |
| 16:13.19 | n9urk | its more fun to read a bio on someone |
| 16:13.59 | n9urk | CunningPike: what license class do you hold? |
| 16:14.10 | CunningPike | n9urk: Umm. Anyway, back to your MOH problem - do you have any timing source in your box? |
| 16:14.19 | CunningPike | n9urk: Basic + 5wpm |
| 16:14.47 | n9urk | ztdummy is compiled into the kernel |
| 16:15.20 | n9urk | CunningPike: you goign to upgrade? |
| 16:15.26 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
| 16:15.26 | *** mode/#asterisk [+o mog] by ChanServ |
| 16:15.33 | n9urk | CunningPike: I am a general. Will upgrade eventually |
| 16:15.43 | CunningPike | n9urk: Eventually - like you :D |
| 16:15.53 | n9urk | CunningPike: ;0 |
| 16:15.58 | n9urk | ;) |
| 16:16.16 | n9urk | CunningPike: will it not work with ztdummy? |
| 16:16.46 | CunningPike | n9urk: It should - 2.6 kernel, right? |
| 16:16.55 | n9urk | yeah |
| 16:17.00 | CunningPike | n9urk: Hmmm |
| 16:17.01 | *** join/#asterisk lorinc (n=ang@caracas-0312.adsl.interware.hu) |
| 16:17.31 | n9urk | CunningPike: What I get is static when the file plays |
| 16:18.28 | CunningPike | n9urk: We had that, but with Digium hardware. We removed the onboard EC and the problem went away. That won't help you though |
| 16:18.33 | benjk_ | you dont need a timing source for MOH |
| 16:19.04 | n9urk | benjk_: CunningPike: that is what I thought, that you didn't need a timing source |
| 16:19.10 | benjk_ | you only need it for two things: IAX call bundling (aka trunking) and app_meetme |
| 16:19.23 | tomlobato | hello all! please, can I put two sections [iaxfwd] as says the book asterisk of the oreilly? or I didnt understand? |
| 16:19.40 | n9urk | CunningPike: benjk_ when I tried to compile mpg123 that comes with * it won't compile, I just get an error |
| 16:19.50 | CunningPike | n9urk: Try native MOH |
| 16:19.51 | tomlobato | sorry, put it in the file iax.conf |
| 16:20.01 | n9urk | CunningPike: how do I use native MOH? |
| 16:20.09 | CunningPike | ~nativeMOH |
| 16:20.13 | benjk_ | don't use mpg123 |
| 16:20.25 | n9urk | CunningPike: I don't get it, how do I do that? |
| 16:20.32 | CunningPike | ~moh |
| 16:20.34 | jbot | [moh] Music On Hold. Good information about how to set it up in the various possible ways can be found at http://www.voip-info.org/wiki/index.php?page=Asterisk+config+musiconhold.conf |
| 16:20.34 | benjk_ | you can use sox or madplay instead |
| 16:20.42 | CunningPike | Aha - there we go |
| 16:20.43 | *** join/#asterisk _alex_mx_ (n=_alex_mx@200.78.229.18) |
| 16:21.08 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 16:21.08 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 16:21.19 | CunningPike | n9urk: That link has a good example of a native MOH config - it's simple to set up and works way better than omg123 |
| 16:21.19 | n9urk | CunningPike: how do I use that in the dial() cmd? |
| 16:21.33 | *** join/#asterisk smackus (n=ckwall@63.149.122.93) |
| 16:21.40 | CunningPike | n9urk: Exactly the same - it's a change to the musiconhold.conf |
| 16:21.55 | smackus | is anyone out there using hp dl320 or 380s? |
| 16:22.06 | CunningPike | smackus: 360s - any good? |
| 16:22.12 | n9urk | CunningPike: ok, let me read that link a little closer |
| 16:22.26 | hmmhesays | wicky wicky wild wild |
| 16:22.29 | CunningPike | n9urk: OK |
| 16:22.43 | smackus | CunningPike: yes... regarding irq's do you know how to change things so that you have more than only 2 irqs to choose from? |
| 16:22.58 | smackus | i have had to turn off usb and one of my nics to be able to get my t1 card on its own t1 |
| 16:23.10 | smackus | i throw in a pci nic and it takes the same irq as the t1 card. |
| 16:23.16 | CunningPike | smackus: Yes, we had to disable USB |
| 16:23.23 | benjk_ | smackus, don't buy a POS excuse for a computer next time |
| 16:23.35 | n9urk | CunningPike: This is all greek to me. lol. I don't quite get what the difference is between the old mpg123 way and the native moh way. |
| 16:23.37 | smackus | my thoughts exactly. |
| 16:23.43 | CunningPike | benjk_: ABE is certified on DL-380 |
| 16:23.51 | eKo1 | n9urk: have you ever used mpg123? |
| 16:24.00 | n9urk | yes |
| 16:24.05 | benjk_ | computers which have a thing called BIOS are all a bunch of dog poo |
| 16:24.11 | n9urk | eKo1: yes, I used that until upgrading to 1.2.10 |
| 16:24.16 | mog | whats that benjk_ ? |
| 16:24.21 | mog | whats wrong with a bios |
| 16:24.26 | eKo1 | n9urk: well then you'll realize that is sucks major donkey balls |
| 16:24.42 | CunningPike | n9urk: It uses a new ability of asterisk to play moh files natively ie using its own internal player, instead of having to call an external app |
| 16:24.43 | watchy | mog i love you |
| 16:24.45 | smackus | my e-machine can do 12 irqs but my 2500 dollar hp server can only do 2... lame |
| 16:24.46 | benjk_ | BIOS is 50% of what makes PC crappy |
| 16:24.52 | n9urk | eKo1: so what do I need to do to use native moh. I don't quite get how to make the config changes |
| 16:24.53 | benjk_ | the other 50% being Windoze |
| 16:25.01 | mog | its like three seconds of boot |
| 16:25.09 | mog | everything working for ya watchy ? |
| 16:25.11 | benjk_ | no its not |
| 16:25.20 | *** join/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
| 16:25.21 | TripleFFFF | <PROTECTED> |
| 16:25.22 | jbalcomb | benjk_: what computer system doesn't have a BIOS? |
| 16:25.22 | mog | it goes bnannana beep and then your booted |
| 16:25.24 | mog | macs |
| 16:25.30 | TripleFFFF | exept hard rtp timeout |
| 16:25.32 | smackus | there has to be a jumper or something that enables more flex with the irqs |
| 16:25.32 | benjk_ | it's messing with how your OS talks to the hardware and how the hardware is controlled |
| 16:25.32 | mog | benjk_, only uses macs |
| 16:25.40 | mog | psha |
| 16:25.43 | benjk_ | not true |
| 16:25.50 | watchy | yes except he removed some context from my zapata.conf for 4 lines and they got redirected somewhere else |
| 16:25.51 | mog | what are you gonna do now by the way? |
| 16:25.57 | mog | cling to your ppc boxen? |
| 16:26.03 | mog | or is efi clean enough for you? |
| 16:26.06 | watchy | but i fixed that in 1 minute. he probably thought it was a conf mistake |
| 16:26.07 | benjk_ | I use WRAPs |
| 16:26.13 | watchy | they got a wierd ass setup at this company |
| 16:26.16 | benjk_ | they don't have BIOS |
| 16:26.25 | benjk_ | not the kind that PCs have anyway |
| 16:26.27 | watchy | they have like a total of 30 phones 8 lines, 4 lines goto 1 secretary 4 goto another |
| 16:26.36 | n9urk | eKo1: CunningPike: would one of you guys mind sending me an example moh config file? |
| 16:26.39 | watchy | its a wierd deal |
| 16:26.40 | mog | yes they do |
| 16:26.43 | jbalcomb | benjk_: Where do you buy WRAPs? |
| 16:26.48 | mog | they are just like the soekris |
| 16:26.51 | benjk_ | in Switzerland |
| 16:26.52 | *** part/#asterisk sevard (n=sev@adsl-71-129-115-244.dsl.irvnca.pacbell.net) |
| 16:26.53 | mog | which also has a bios |
| 16:27.02 | *** join/#asterisk wangster (n=wangster@static-64-201-170-129.ptr.terago.ca) |
| 16:27.07 | mog | just glad its working for you watchy |
| 16:27.09 | CunningPike | n9urk: Sure hang on a sec...... |
| 16:27.09 | jbalcomb | mog benjk_: what about SPARCs? |
| 16:27.10 | smackus | i cannot find an irc channel for anything relating to hp... is there one that anyone knows of? |
| 16:27.11 | benjk_ | Soekris have their own firmware, its not BIOS |
| 16:27.19 | watchy | yea its great i just gotta tweak out the polycoms dialstring |
| 16:27.23 | benjk_ | WRAPs have their own firmware, its not BIOS |
| 16:27.25 | mog | yes it is a bios |
| 16:27.28 | mog | its bios man |
| 16:27.32 | mog | it comes up and says bios |
| 16:27.37 | mog | has all normal bios operations |
| 16:27.40 | mog | is a bios |
| 16:27.53 | n9urk | CunningPike: eKo1: will the native moh support mp3s or not? |
| 16:27.54 | mog | i mean its not really worth arguing over |
| 16:27.56 | mog | but meh |
| 16:27.58 | TripleFFFF | rtptimeout |
| 16:28.07 | benjk_ | its custom made for the WRAP, none of the ballast that vanilla Intel PCs carry around |
| 16:28.14 | CunningPike | n9urk: Here we go |
| 16:28.16 | CunningPike | [default] |
| 16:28.16 | CunningPike | mode=files |
| 16:28.16 | CunningPike | directory=/var/lib/asterisk/mohmp3/dnv/default |
| 16:28.16 | CunningPike | random=yes |
| 16:28.19 | Qwell[] | mog: If it walks like a duck, and looks like a duck...it must not be a duck |
| 16:28.21 | mog | it doesnt matter its still a bios |
| 16:28.25 | mog | Qwell[], ! |
| 16:28.29 | CunningPike | n9urk: That's all you need |
| 16:28.30 | benjk_ | aka Phoenix BIOS and derivatives |
| 16:28.31 | n9urk | CunningPike: eKo1 it appears as if the link the bot sent me earlier about moh is not exactly consistant. Thanks |
| 16:28.33 | *** part/#asterisk brif8 (n=Administ@ns1.ttienterprises.org) |
| 16:28.40 | benjk_ | that's a piece of junk |
| 16:28.49 | benjk_ | it was appropriate in the late 70s |
| 16:28.51 | n9urk | CunningPike: thanks |
| 16:28.55 | benjk_ | for home computers of the time |
| 16:29.00 | CunningPike | n9urk: For mp3 files, you need format_mp3.so from asterisk-addons |
| 16:29.03 | benjk_ | it's not anymore today |
| 16:29.09 | CunningPike | n9urk: But we convert ours to ulaw |
| 16:29.25 | *** join/#asterisk Delta239 (n=adfadsf@200.124.18.171) |
| 16:29.44 | jbalcomb | ben_dj mog: perhaps the channel is a bit busy for a such an off-topic conversation? |
| 16:29.48 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 16:29.53 | mog | yeah its a bit much |
| 16:29.53 | benjk_ | its not off-topic |
| 16:30.14 | CunningPike | Thanks, jbalcomb |
| 16:30.21 | benjk_ | because the question was posed how to prevent the BIOS from assigning only those 2 available IRQs |
| 16:30.26 | CunningPike | Gotta keep these young kids in check ;) |
| 16:30.30 | jbalcomb | :) |
| 16:30.44 | benjk_ | and the answer to that is that you most likely don't unless you get a computer with proper firmware |
| 16:30.47 | *** join/#asterisk SwK (n=Silik0nJ@204.250.115.248) |
| 16:30.55 | mog | okay and now we are done |
| 16:31.02 | n9urk | I have asterisk-addons. do I need to load another module for playing gsm CunningPike? |
| 16:31.14 | TripleFFFF | guys ? |
| 16:31.18 | *** part/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 16:31.20 | jbalcomb | benjk_: that seems like a reasonable and fairly complete answer. |
| 16:31.36 | CunningPike | n9urk: I don't think so, because the ability to play gsm files already exists within asterisk |
| 16:31.48 | n9urk | ok, tnx. I am working on it now |
| 16:31.54 | jbalcomb | TripleFFFF: sorry, i'm not sure i follow you there. |
| 16:32.06 | n9urk | CunningPike: I did what you said and I still get static |
| 16:32.45 | TripleFFFF | i had a call that went over 13 hours |
| 16:32.55 | TripleFFFF | im sure its a false online.. or hangup not catched |
| 16:33.01 | TripleFFFF | so anyway to hangup these calls ? |
| 16:33.02 | jbalcomb | TripleFFFF: there is an rtpholdtimeout for calls on hold |
| 16:33.06 | TripleFFFF | scripted or using rtptimeout ? |
| 16:33.11 | TripleFFFF | not on hold |
| 16:33.13 | TripleFFFF | i think |
| 16:33.44 | jbalcomb | TripleFFFF: my conference phone does that sometimes. was the device on the call an ata? |
| 16:34.01 | [TK]D-Fender | benjk_ : I remember the days of having to care about IRQ allocation.... no more! |
| 16:34.14 | droops | http://www.remote-exploit.org/index.php/BackTrack_Downloads |
| 16:34.14 | Dr-Linux|work | Qwell[], around |
| 16:34.17 | n9urk | CunningPike: Goter working |
| 16:34.22 | droops | err wrong channel |
| 16:34.23 | TripleFFFF | no idea |
| 16:34.23 | n9urk | CunningPike: I had to do a hard restart |
| 16:34.25 | Qwell[] | Dr-Linux|work: nope |
| 16:34.26 | TripleFFFF | its a client |
| 16:34.29 | Dr-Linux|work | ok good |
| 16:34.30 | n9urk | of * |
| 16:34.31 | Qwell[] | Dr-Linux|work: he said you should be fine |
| 16:34.36 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.100) |
| 16:34.39 | n9urk | CunningPike: Thanks for your help, OM. |
| 16:34.39 | benjk_ | well, now it seems that EFI is going to go mainstream, so that will become a thing of the past |
| 16:34.39 | TripleFFFF | func_hello |
| 16:34.42 | TripleFFFF | i cant find htis lol |
| 16:34.45 | CunningPike | n9urk: Static gone? |
| 16:34.52 | n9urk | CunningPike: Got the MOH now |
| 16:34.56 | Dr-Linux|work | Qwell[], wow, so what he suggested? |
| 16:35.26 | CunningPike | n9urk: Perfect - we really have to remove mpg123 from the documentation - it's just not necessary any more |
| 16:35.28 | Dr-Linux|work | Qwell[], : should i just remove the whole site and put something like "This is not Red Hat Linux" |
| 16:35.42 | Dr-Linux|work | Qwell[], or what? |
| 16:35.43 | [TK]D-Fender | CunningPike : it has its place... |
| 16:36.08 | n9urk | CunningPike: I have to run. Maybe I can try to add somethign to the voip-info wiki after bit. * is an awesome program but the documentation really limits its usefullness |
| 16:36.13 | CunningPike | [TK]D-Fender: At the bottom of a deep, dark hole |
| 16:36.28 | CunningPike | n9urk: It's like any OSS project :) |
| 16:36.32 | [TK]D-Fender | CunningPike : not THAT far.... just not for most users |
| 16:36.38 | benjk_ | D-Fender, has its place or *had* its place? |
| 16:36.41 | n9urk | CunningPike: OSS strikes again ;). |
| 16:36.46 | [TK]D-Fender | benjk_ : both |
| 16:36.46 | Dr-Linux|work | Qwell[], can i /msg you? |
| 16:36.48 | n9urk | CunningPike: It will evolve, I am sure |
| 16:36.54 | CunningPike | n9urk: It's part of the fun :D |
| 16:36.56 | n9urk | CunningPike: it just takes time |
| 16:36.58 | benjk_ | why would you still use mpg123? |
| 16:37.11 | [TK]D-Fender | benjk_ : Volume control, streaming, etc. |
| 16:37.13 | jbalcomb | [TK]D-Fender: How can you say that? alsa makes mpg123 useless, madplay actually still works, and asterisk apparently needs niether anymore. |
| 16:37.15 | benjk_ | its not being maintained anymore and there are known security issues |
| 16:37.23 | benjk_ | you can just as easily use madplay |
| 16:37.32 | [TK]D-Fender | jbalcomb : Guess Madplay can take on a chunk then |
| 16:37.41 | n9urk | CunningPike: Tnx and 73. I will see you in here later, I guess. Enjoy the Beautiful BC country ;) |
| 16:37.46 | TripleFFFF | <PROTECTED> |
| 16:37.48 | [TK]D-Fender | just don't know it that well |
| 16:37.51 | CunningPike | n9urk: Will do. 73 |
| 16:38.03 | Dr-Linux|work | Qwell[], should be fine with doing what? |
| 16:38.16 | jbalcomb | [TK]D-Fender: its easy. the important part to keep in mind is that mpg123 is out for linux not just asterisk |
| 16:38.53 | benjk_ | but most Linux distros don't even ship mpg123 anymore |
| 16:38.57 | [TK]D-Fender | jbalcomb : thats fine I guess.. just didn't know much about the alternatives. |
| 16:39.00 | jbalcomb | CunningPike: i think you're right about removing mpg123. we've had severally people start lengthy support conversation before realizing that it just isn't an option. |
| 16:39.07 | benjk_ | madplay has pretty much taken its place |
| 16:40.07 | jbalcomb | [TK]D-Fender: check out our moh.conf. its in there. apt-get install madplay OR emerge madplay OR whateverpkg madplay and you're done. |
| 16:41.09 | tomlobato | is it allowed to use two sections with same name in the iax.conf file? |
| 16:41.19 | benjk_ | yes |
| 16:41.27 | benjk_ | one friend and one peer |
| 16:41.43 | [TK]D-Fender | jbalcomb : Well I haven't needed it personally so I'll just stick to learning my RHEL stuff for now :) |
| 16:41.44 | benjk_ | correction: one user and one peer |
| 16:42.13 | jbalcomb | [TK]D-Fender RHEL? didn't you give me shit when i said my asterisk server was gonna be RHEL 4? |
| 16:42.50 | tomlobato | benjk_, oh cool, cause I read on the asterisk oreilly book two with [iaxfwd] and fwd server didnt allowed me, so I think this possible problem |
| 16:43.00 | [TK]D-Fender | jbalcomb : Only becuase you PAID for it :) |
| 16:43.09 | [TK]D-Fender | jbalcomb : CentOS = RHEL so WTS ;) |
| 16:43.11 | jbalcomb | TripleFFFF: have you read the voip-info pages for rtptimeout and rtpholdtimeout? |
| 16:43.24 | jbalcomb | [TK]D-Fender thrrp. |
| 16:43.35 | benjk_ | [foobar] |
| 16:43.35 | [TK]D-Fender | jbalcomb mrmble |
| 16:43.39 | benjk_ | type=user |
| 16:43.44 | benjk_ | [foobar] |
| 16:43.47 | benjk_ | type=peer |
| 16:43.52 | benjk_ | works for me |
| 16:43.59 | tomlobato | benjk_, perfect, thank you |
| 16:44.10 | tomlobato | benjk_, did you get trouble for register on fwd server? |
| 16:44.25 | benjk_ | not that I recall |
| 16:44.34 | tomlobato | Aug 2 13:50:59 NOTICE[3829] chan_iax2.c: Registration of '789843' rejected: 'Registration Refused' from: '192.246.69.186' |
| 16:45.07 | benjk_ | iax2 debug is your friend |
| 16:45.46 | tomlobato | oh yes, thank you... interesting.... debug session in the book is near fwd =) |
| 16:45.59 | tomlobato | ok, thank you, I ll try again |
| 16:47.40 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 16:47.40 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 16:47.58 | momelod | how do i use a wav file as a ringtone for the polycom ip501? I placed the wave in the ftp, the ftp log reports the file was downloaded by the phone. but i dont see the wav in the list of available ringers |
| 16:48.49 | jbalcomb | momelod: http://www.google.com/search?hl=en&sa=X&oi=spell&resnum=0&ct=result&cd=1&q=polycom+ip+501+ringtone&spell=1 |
| 16:49.00 | [TK]D-Fender | momelod : You need to set it in the SAMPLES section in sip.cfg |
| 16:49.28 | momelod | thank u |
| 16:49.43 | jbalcomb | damn, i didn't realize the google URL says i need a spellchecker. |
| 16:52.21 | Qwell[] | Dr-Linux|work: You're fine, as in, tell them no if you want |
| 16:52.33 | Qwell[] | Dr-Linux|work: Then can try to come after you, but...good luck enforcing US trademark law there |
| 16:52.35 | SplasPood | Hrm... ProjectMf... Its been ages since I'd heard anything from phiber optik |
| 16:52.42 | Qwell[] | s/then/they/ |
| 16:58.01 | *** join/#asterisk JunK-Y (n=junky@modemcable205.175-81-70.mc.videotron.ca) |
| 16:58.35 | [TK]D-Fender | JunK-Y : Not at Cluecon? |
| 16:58.55 | JunK-Y | yes i am. |
| 16:59.00 | [TK]D-Fender | Scrren :) |
| 16:59.04 | [TK]D-Fender | heh |
| 16:59.11 | Qwell[] | scrren? |
| 16:59.12 | [TK]D-Fender | JunK-Y :Cool. |
| 16:59.20 | JunK-Y | yes screen |
| 16:59.40 | [TK]D-Fender | Qwell : whut ar u toking abowt my tiping is fyn! |
| 17:00.38 | Dr-Linux|work | Qwell[], okey should i remove the site and put "This is not red hat linux site" |
| 17:00.46 | Dr-Linux|work | ? |
| 17:01.28 | Qwell[] | Dr-Linux|work: it's up to you |
| 17:01.28 | [TK]D-Fender | Dr-Linux|work : Get your own cool name and stop ripping off others. |
| 17:03.03 | Dr-Linux|work | [TK]D-Fender, i was unaware of this trademark system, so i just need compensation that i spent time and stuff on my site. |
| 17:03.05 | watchy | hey tk in the dialstring whats a T do? |
| 17:03.19 | asterboy | anyone going to LinuxWOrld San Fran? |
| 17:03.22 | [TK]D-Fender | watchy : Timeout |
| 17:03.26 | watchy | ah |
| 17:03.50 | [TK]D-Fender | watchy : What I use : X.T|*.T|#.T |
| 17:03.59 | Dr-Linux|work | if they still use strong arm with me and pay me nothing then today i gonna remove my whole site |
| 17:04.05 | [TK]D-Fender | watchy : as in "accept anything with a wait and STFU" |
| 17:04.11 | ManxPower | This is starting to piss me off. My Cisco 1750 can't get an IP from the Cablemodem. PCs can. |
| 17:04.52 | ManxPower | Many similar problems reported, no solutions reported. |
| 17:06.52 | ManxPower | ~ecfo |
| 17:06.56 | jbot | Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as "screeching", "feedback", "static", or other useless terms. If users report "static" on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. |
| 17:08.36 | hmmhesays | ok a sales person just gave away support AGAIN |
| 17:08.39 | hmmhesays | what the fsck |
| 17:09.36 | Dr-Linux|work | Qwell[], what's trademark law there, if i remove the site and put "something its not redhat linux site"? |
| 17:10.21 | *** join/#asterisk arkonadev (n=chatzill@65.203.186.131) |
| 17:10.48 | arkonadev | so if i call and AGI script on an IP address but that address cant be resolved how do i handle that elegantly |
| 17:10.52 | hmmhesays | anyone remember what version of zaptel goes with asterisk 1.2.7.1? |
| 17:11.11 | Qwell[] | Dr-Linux|work: It'd still be illegal in US law, BUT, you're Paki :) |
| 17:11.30 | ManxPower | #charter |
| 17:11.31 | Qwell[] | hmmhesays: any should work, afaik |
| 17:11.32 | ManxPower | oops |
| 17:11.39 | Qwell[] | ManxPower: give em hell for me |
| 17:11.51 | ManxPower | Qwell, who? |
| 17:11.55 | Qwell[] | charter |
| 17:12.01 | ManxPower | Qwell, Ha! |
| 17:12.20 | ManxPower | I need to convince them to swap out the cablemodem for a different brand |
| 17:12.20 | asterboy | no one going to linux world? |
| 17:12.37 | Qwell[] | ManxPower: yeah...you'll end up having to buy your own... |
| 17:12.41 | Qwell[] | (probably) |
| 17:12.54 | ManxPower | Qwell, *nod* Sucks. |
| 17:13.03 | Qwell[] | give it a shot though..who knows |
| 17:13.20 | Dr-Linux|work | Qwell[], www.whitehouse.com is not an official domain .. it had same problem |
| 17:13.35 | Qwell[] | Dr-Linux|work: "whitehouse" isn't a trademark |
| 17:13.40 | watchy | whys it suck a cablemodem is $30 |
| 17:13.46 | watchy | just buy your own |
| 17:14.38 | ManxPower | watchy, I guess I could be late on yet another credit card payment. |
| 17:14.50 | watchy | time for a 2nd job |
| 17:14.51 | ManxPower | I assume Best Buy has cable modems? |
| 17:15.00 | watchy | yea i got a linksys it works great |
| 17:15.07 | ManxPower | watchy, I'm working on reducing expenses. |
| 17:15.10 | Qwell[] | get a list of approved ones... |
| 17:15.11 | CunningPike | Has anyone had any luck getting Polycoms to allow a person to answer a call that rang on multiple sets, place the call on hold and then allow someone to pick it up on another set? Or even a way to simulate it? |
| 17:15.16 | ManxPower | Hmmm...I think WalMArt has them too. |
| 17:15.24 | tzanger | does Rogers allow you to use your own modem? |
| 17:19.12 | [TK]D-Fender | CunningPike : Call Parking. |
| 17:19.42 | CunningPike | [TK]D-Fender: We're using call parking now, but the users hate it |
| 17:20.09 | [TK]D-Fender | CunningPike : TFB until * gets SIP-B support (was hopeful for 1.4 but didn't make it. Who know when....) |
| 17:20.58 | CunningPike | [TK]D-Fender: I was hoping someone had a dialplan trick I could use....... |
| 17:21.26 | [TK]D-Fender | CunningPike : Actually you could use other line appearances for buddy watch on them. Then if you know they parked a call have it so the exte it dials looks for a parked call by that exten before dialing them. |
| 17:21.35 | [TK]D-Fender | CunningPike : Try that. |
| 17:23.02 | CunningPike | [TK]D-Fender: Interesting - but they would still have to park the call first - they hate that too - "too many keypresses" |
| 17:23.28 | [TK]D-Fender | CunningPike : *2* |
| 17:24.00 | CunningPike | [TK]D-Fender: (sigh) - I know. Two presses too many for our folks...... |
| 17:24.11 | SplasPood | Question all.. if I do a Dial() that returns BUSY and I don't exec Busy(), will the SIP client see the BUSY response, or...? |
| 17:24.20 | bcnl | CunningPike: get the Aaster 480i CT phones, I have a park button defined |
| 17:24.47 | bcnl | you git the button, it announces the parking spot, the user hangs up |
| 17:24.52 | [TK]D-Fender | CunningPike : Actually Polycom has a parking feature... haven't tested it though... |
| 17:25.22 | bcnl | things *must* be very very simple |
| 17:25.38 | CunningPike | bcnl: Already committed to Polycoms - and have call parking working. Users don't like it. Our existing Nortel allows them to place a call on hold and pick it up somewhere else. They want that |
| 17:25.42 | arkonadev | so if a fastagi script fails in the dial plan what is returned |
| 17:25.53 | [TK]D-Fender | CunningPike : TFB |
| 17:26.08 | CunningPike | [TK]D-Fender: :D |
| 17:26.16 | bcnl | CunningPike: perhaps call pickup could be your friend? |
| 17:26.31 | CunningPike | bcnl: Perhaps...... |
| 17:26.33 | *** join/#asterisk pigpen2 (n=mark@207.71.33.114) |
| 17:26.34 | bcnl | send the call into a long ring to a extension with no one on it |
| 17:26.35 | [TK]D-Fender | CunningPike : SIP-B is a while away for *, so don't blame the phone. They should be rather happy with the savings * offers and the great functionality. Time to grow and learn. |
| 17:26.43 | bcnl | and then let people just *8 it to get the call |
| 17:26.58 | bcnl | course that seems like just as much work as parking, but it might get past the users mental block |
| 17:27.01 | [TK]D-Fender | bcnl : Intersting idea.... worth keeping a back-burner for... |
| 17:27.51 | [TK]D-Fender | bcnl : And radically annoying if you want ot hold it for someone else while remainging at your desk. Of course with Polycom you could set the Alert-Info for a silent ring, but there's still a risk of pulling up the call again.... |
| 17:28.32 | bcnl | I agree, call park is the proper solution |
| 17:29.53 | smackus | question... I am moving to a sangoma card instead of the digium card. I am using the a104d - 4 port t1 card. Do I still use zaptel? |
| 17:32.03 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 17:32.04 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 17:33.47 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:34.32 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 17:34.57 | [TK]D-Fender | smackus : Yup. Compile Zaptel, then wanpipe, then recompile zaptel (wanpipe should do this when you run the install script anyways) |
| 17:35.23 | noname32 | what would be proper synax for extenstion.con for when some one presses 0 in vm for operator? |
| 17:35.28 | hmmhesays | wiki wiki wild wild |
| 17:35.41 | noname32 | do you use dial to pass it to an extention? |
| 17:36.06 | [TK]D-Fender | noname32 : lookup "config voicemail.conf" on the WIKI. |
| 17:36.14 | hmmhesays | so my script now is pretty good |
| 17:36.18 | hmmhesays | not bad for a days work |
| 17:39.00 | noname32 | [TK]D-Fender, didnt give much help mate |
| 17:39.58 | *** join/#asterisk deb_user (n=debian_u@albuquerque.agroinnovations.com) |
| 17:40.03 | noname32 | my question is simple i think do you use dial or is there a better cmd or method to transfer a call to an extention |
| 17:40.14 | tzanger | bah |
| 17:40.20 | tzanger | 32MB/sec is slow on a gigabit network :-( |
| 17:40.47 | noname32 | lol blam the hardware ;p |
| 17:40.54 | noname32 | blame* |
| 17:40.55 | *** join/#asterisk lorinc (n=ang@caracas-4182.adsl.interware.hu) |
| 17:40.59 | deb_user | could somebody help me learn how to use the zttool please? |
| 17:43.13 | hmmhesays | ruh roh, i'm getting invalid module format trying to modprobe zaptel |
| 17:44.22 | watchy | stab it |
| 17:44.27 | *** join/#asterisk jwei (n=jwei@64.77.225.68) |
| 17:44.37 | jwei | hi |
| 17:44.40 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 17:44.44 | jwei | i have a question |
| 17:44.56 | nortex | ask it |
| 17:45.00 | jwei | i am trying to optimize Asterisk and RedHat EL4 so I can increase the number of concurrent calls |
| 17:45.15 | *** join/#asterisk marly (i=marly@CPE-24-209-183-224.wi.res.rr.com) |
| 17:45.25 | jwei | I am running Asterisk 1.2.10 on a box with dual Xeon 3.4 GHz + 2GB RAM + SATA drives |
| 17:45.52 | jwei | I have enabled realtime for voicemail accounts and ODBC for voicemail storage and have a MySQL call prior to dialing |
| 17:45.54 | watchy | how many calls are you doing |
| 17:45.56 | benjk_ | tzanger: 32MB/sec == 256Mbit/sec => 1/4 the capacity of a Gigabit network == actually very fast |
| 17:46.17 | tzanger | yeah I know it's 256mbit but on a switched network I'd liked to have seen much higher |
| 17:46.28 | jwei | the total I can make is 200 but with choppy audio...with good audio about 120 |
| 17:46.32 | jwei | using G.711u |
| 17:46.36 | tzanger | benjk_: it looks like this one server is taking a shitkicking though... 12k interrupts/sec |
| 17:46.37 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 17:46.42 | tzanger | I think I need some NAPI lovin |
| 17:46.45 | benjk_ | ouch |
| 17:46.49 | marly | I'm setting up IAX2 and when I do an iax2 show registry the Refresh says 60, I want to set the Refresh to say 2 seconds, and I tried setting minregexpire and maxregexpire in the iax.conf however, it hasn't changed anything. Am I going about this the wrong way? |
| 17:46.51 | tzanger | the other system is at 6k interrupts/sec |
| 17:46.58 | jwei | at 200 calls, the load avg is 15 approx |
| 17:47.03 | jwei | spiking at 20 |
| 17:47.06 | charles___ | jwei: reduce your disk access |
| 17:47.17 | jwei | with 120, it's less than 5 |
| 17:47.32 | jwei | I'm using SIP + RTP only |
| 17:47.37 | charles___ | jwei: try to reduce the disk access to 0 |
| 17:48.07 | watchy | you running sql on the same box? |
| 17:48.19 | jwei | no...SQL on a seprate box |
| 17:48.51 | watchy | what do you find thats eating cpu? |
| 17:48.57 | charles___ | jwei: logs ? |
| 17:49.02 | jwei | asterisk process |
| 17:49.08 | jwei | sucks up allt he CPU |
| 17:49.12 | charles___ | jwei: logs won't be a problem unless you're debuging too much |
| 17:49.12 | jwei | other than that, I don't know |
| 17:49.25 | benjk_ | transcoding perhaps? |
| 17:49.28 | charles___ | jwei: it can suck your cpu waiting for slow deviecs as disks |
| 17:49.31 | jwei | also, I have about 200k for max file descriptor |
| 17:49.48 | charles___ | jwei: yes, like benjk_ said, transcoding will do too |
| 17:49.49 | jwei | i am logging in debug mode |
| 17:49.57 | jwei | also storing CDRs on MySQL |
| 17:50.07 | jwei | but I can't stop Asterisk from saving CDRs on text files |
| 17:50.21 | benjk_ | sure you can |
| 17:50.30 | jwei | there are 2 text files it saves to..one Master.csv and another for custom.csv |
| 17:50.50 | CunningPike | noname32: exten => o,1,WhatEver() |
| 17:50.52 | charles___ | benjk_: but it would'nt use all that hardware cdr's once per call |
| 17:51.17 | charles___ | jwei: look at transcoding |
| 17:51.34 | charles___ | jwei: and any other disk access on your system |
| 17:51.37 | jwei | end points are g.711u |
| 17:51.48 | charles___ | jwei: you may try having a ramdrive for anything extra |
| 17:52.00 | jwei | ok... |
| 17:52.10 | jwei | let me try with logging off |
| 17:52.13 | charles___ | jwei: strip the code |
| 17:52.16 | jwei | or minimal |
| 17:52.19 | charles___ | jwei: and gcc -O9 |
| 17:52.41 | charles___ | jwei: HD led's blink ? |
| 17:53.05 | charles___ | jwei: using Hyper Threading and threads on kernel ? |
| 17:53.06 | jwei | not sure...the server is in the back room |
| 17:53.08 | jwei | i can take a look |
| 17:53.12 | jwei | yes |
| 17:53.13 | jwei | using HT |
| 17:53.31 | charles___ | jwei: there is threads on the kernel and GCC that new linux versions are comming with |
| 17:53.32 | arkonadev | what would be the syntax to set the result of an AGI call to a variable SetVar(result=${AGI(agi://host/file.agi)})? |
| 17:53.37 | charles___ | jwei: one sec |
| 17:55.00 | TripleFFFF | so there a t38 patch |
| 17:56.58 | charles___ | jwei: instead of using threads you may use NPTL that will get more efficient kind of 10% increase |
| 17:57.07 | *** join/#asterisk ZPertee (n=zach@cpe-65-25-51-117.neo.res.rr.com) |
| 17:57.46 | *** join/#asterisk dlynes_ (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 17:57.53 | charles___ | jwei: do a /lib/libc.so.6 |
| 17:57.57 | luke-jr_ | charles___: erm, you do know that NPTL *is* threads, right? |
| 17:58.05 | charles___ | luke-jr_: I do! |
| 17:58.34 | charles___ | jwei: instead of Linux Threads you should have Native POSIX Threads Library |
| 17:58.40 | *** join/#asterisk oadaeh (n=jason@las-static-208.57.199.83.mpowercom.net) |
| 17:58.41 | luke-jr_ | ;) |
| 17:58.48 | charles___ | luke-jr_: linux and POSIX :P |
| 17:58.49 | TripleFFFF | ok |
| 17:58.52 | charles___ | luke-jr_: threads :P |
| 17:59.08 | TripleFFFF | now i need asterisk t38 patch... |
| 17:59.11 | luke-jr_ | "fax working really good" = ulaw =p |
| 17:59.12 | charles___ | TripleFFFF: T38 ? whree ? |
| 17:59.16 | TripleFFFF | but was that on latest branch or svn or where ? |
| 17:59.26 | Qwell[] | T38 support is in trunk |
| 17:59.35 | TripleFFFF | so itssvn |
| 17:59.40 | jwei | so compile asterisking using POSIX thread? |
| 17:59.42 | charles___ | wow really ? or is it a joke ? wow |
| 17:59.47 | charles___ | wow wow wow :P |
| 17:59.52 | Qwell[] | RTFML |
| 17:59.54 | Qwell[] | :p |
| 18:00.00 | luke-jr_ | charles___: trunk has XMPP and Jingle support too =p |
| 18:00.01 | charles___ | jwei: first you gotta have your system usin Posix Threads |
| 18:00.06 | Qwell[] | (ml=mailing lists) |
| 18:00.11 | luke-jr_ | charles___: which means Google Talk |
| 18:00.24 | mog | ^_^ |
| 18:00.24 | mog | yes |
| 18:00.25 | charles___ | luke-jr_: XMPP ? is it the name of the Jingle singer ? |
| 18:00.30 | Qwell[] | jingle sucks |
| 18:00.37 | luke-jr_ | charles___: XMPP is the IETF standard for instant messaging |
| 18:00.40 | Qwell[] | mog: sowwy :( |
| 18:00.50 | mog | call it jabber |
| 18:00.52 | mog | it sounds better |
| 18:00.54 | mog | its res_jabber |
| 18:00.57 | mog | not res_xmpp |
| 18:00.58 | mog | ^_^ |
| 18:00.59 | luke-jr_ | but Jabber is just one brand |
| 18:01.03 | charles___ | cool jabber is cool |
| 18:01.04 | luke-jr_ | like Google Talk or Gizmo |
| 18:01.07 | *** join/#asterisk tracinet (n=tracinet@216.242.235.2) |
| 18:01.16 | charles___ | what about jingle ? |
| 18:01.18 | mog | jabber is xmpp |
| 18:01.20 | mog | period |
| 18:01.24 | luke-jr_ | Google Talk is XMPP too |
| 18:01.25 | mog | we do jingle |
| 18:01.31 | mog | google talk is jabber |
| 18:01.36 | luke-jr_ | charles___: Jingle is basically SIP for XMPP |
| 18:01.38 | mog | its all called jabber |
| 18:01.38 | Qwell[] | xmpp > jabber > jingle :D |
| 18:01.46 | jwei | charles__:so recompile kernel to use POSIX? |
| 18:01.53 | mog | jingle sits inside of jabber |
| 18:01.55 | Qwell[] | mog: not better than |
| 18:02.01 | Qwell[] | oh |
| 18:02.10 | charles___ | cool |
| 18:02.18 | charles___ | but people aren't that much in mixing Voice and IM |
| 18:02.23 | luke-jr_ | ROFL |
| 18:02.26 | luke-jr_ | most are |
| 18:02.26 | charles___ | I don't really know what that can do :P |
| 18:02.28 | mog | ? |
| 18:02.29 | mog | most are |
| 18:02.32 | mog | its spiffy |
| 18:02.34 | mog | and easy |
| 18:02.38 | mog | no wonder its number 1 |
| 18:02.50 | charles___ | mog: but voice and IM ? |
| 18:02.54 | mog | both |
| 18:03.07 | tracinet | hey guys! |
| 18:03.09 | luke-jr_ | ... |
| 18:03.12 | TripleFFFF | ok so t38 from trunk is where .. do i need special makefile patching ? etc |
| 18:03.18 | luke-jr_ | charles___: you've obviously never heard of Skype? |
| 18:03.28 | charles___ | jwei: did you get to see if you have NPTL on GLIBC already ? |
| 18:03.53 | tracinet | quick question - on dual proc machine - anyone found benefit using smp kernel with asterisk? |
| 18:03.54 | charles___ | luke-jr_: yes I do, but there is nothing special with skype, my ICQ on 1997 could place a PC to PC call |
| 18:04.09 | luke-jr_ | tracinet: not like you have a choice... |
| 18:04.11 | charles___ | luke-jr_: and still back on the past there was FreeTel :P |
| 18:04.22 | luke-jr_ | charles___: Skype made voice+IM popular |
| 18:04.23 | tracinet | did some basic benchmarks and on CentOS when running non-smp kernel - results were higher |
| 18:04.23 | jwei | no..give me a min |
| 18:04.34 | tracinet | made me think.... |
| 18:04.41 | mog | bah voice+im was popular for a long time |
| 18:04.47 | asterboy | can someone please check if irc.linuxfromscratch.org is available? |
| 18:04.48 | luke-jr_ | tracinet: non-SMP ignores the 2nd CPU |
| 18:04.53 | charles___ | jwei: /lib/libc.so.6 |
| 18:04.57 | asterboy | can't seem to get connected. |
| 18:05.09 | luke-jr_ | mog: it wasn't *really* popular till Skype tho |
| 18:05.13 | tracinet | luke: agreed - just weird that general server performance was lower .... |
| 18:05.15 | luke-jr_ | which is a shame since Skype sucks |
| 18:05.18 | TripleFFFF | Update Feb 2006: bug/patch 5090 now contains a T.38 pass-thru patch for Asterisk v1.2.4 and SIP |
| 18:05.29 | asterboy | All I did was tell the op to fuck off...why can't I connect? |
| 18:05.29 | wangster | How can you remotely setup a call from the asterisk console? I want asterisk to place a call to an outside line sending Milliwatt so I can tune tx gain |
| 18:05.36 | mog | eh /me doesnt care about em that much to be honest |
| 18:05.40 | asterboy | just kidding |
| 18:05.42 | mog | non open and thus they will die |
| 18:05.45 | mog | at some point |
| 18:05.57 | jbalcomb | mog: agreed. voice+im with MSNM was easy sheezy. |
| 18:06.08 | TripleFFFF | how can i tell if t38 pass trough is on server ? |
| 18:06.10 | luke-jr_ | anyone run a Google Talk->POTS service yet? |
| 18:06.12 | charles___ | mog: are you gaucho ? |
| 18:06.15 | mog | ? |
| 18:06.18 | mog | mog is mog |
| 18:06.18 | TripleFFFF | oh that in zaptel ? |
| 18:06.20 | TripleFFFF | shit |
| 18:06.24 | luke-jr_ | or is everyone waiting for ebay to pull free Skype? |
| 18:06.24 | mog | you can luke-jr_ |
| 18:06.28 | mog | i dont think anyone is |
| 18:06.32 | luke-jr_ | mog: I know I can ;) |
| 18:06.43 | luke-jr_ | I plan to, once 1.4 is out =p |
| 18:06.49 | mog | heh ^_^ |
| 18:06.59 | charles___ | luke-jr_: festival + Jabber may be interesting but with one way audio :P |
| 18:07.06 | luke-jr_ | maybe sooner if I get the free time |
| 18:07.07 | mog | heh |
| 18:07.18 | Dr-Linux|work | Qwell[], around? here is a news |
| 18:07.37 | luke-jr_ | charles___: at my new job, they're working on full text<->voice bridging for XMPP/Asterisk |
| 18:07.43 | charles___ | I'm waiting for paskambing mcc v 1.5 :P |
| 18:07.44 | luke-jr_ | charles___: hopefully I'll get to work on it |
| 18:08.04 | luke-jr_ | assuming it's not already done |
| 18:08.11 | mog | heh |
| 18:08.17 | mog | im not working on that |
| 18:08.18 | mog | yet |
| 18:08.27 | luke-jr_ | they were at least planning it in March |
| 18:08.47 | charles___ | luke-jr_: so you can make an IM call with a woman voice and persuade people :P |
| 18:09.06 | luke-jr_ | charles___: >_< |
| 18:09.09 | jwei | charles: i jsut msged you the output |
| 18:09.17 | charles___ | luke-jr_: we are going to have phone sex also moved to india |
| 18:09.29 | watchy | why is there no universal install cd for x86 for gentoo? |
| 18:09.45 | charles___ | jwei: you are using linux threads get NPTL and you will get some extra performance |
| 18:09.59 | ZPertee | what does auth=rsa mean? |
| 18:10.03 | charles___ | watchy: fits most x86 :P |
| 18:10.14 | *** part/#asterisk tracinet (n=tracinet@216.242.235.2) |
| 18:10.17 | luke-jr_ | watchy: ask in #gentoo ? |
| 18:10.24 | ZPertee | ok thanks |
| 18:10.42 | jwei | charles: ok..thanks |
| 18:10.47 | watchy | i got this stupid ass gentoo cd that boots into X and has gui/cmd line installers but i dunno how to make it not install X |
| 18:10.52 | charles___ | luke-jr_: girls sexy voice is going to be generated by key strokes from a 40 year old indian. |
| 18:11.14 | charles___ | watchy: no make.conf -x |
| 18:11.27 | watchy | ah |
| 18:11.30 | charles___ | watchy: dude, RTFM, gentoo is very good documented |
| 18:11.39 | charles___ | very well |
| 18:11.40 | charles___ | :P |
| 18:11.51 | watchy | i cant read |
| 18:11.55 | [TK]D-Fender | charles___ : Grammar Rangers... ATTACK!!!!! |
| 18:11.58 | TripleFFFF | watchy |
| 18:12.11 | TripleFFFF | when you first boot... prolly asks press enter to enter gui install or blah for not |
| 18:12.14 | mog | luke-jr_, i wonder how good it would sound with cepstral |
| 18:12.18 | charles___ | watchy: get festival to speak to you the gentoo manual. |
| 18:12.18 | TripleFFFF | start wih taht lol |
| 18:12.29 | TripleFFFF | OR.. when adding packages remove all X11 related crap |
| 18:12.36 | charles___ | watchy: and LTTFM (LISTEN TO THE F. MANUAL) |
| 18:12.38 | jwei | charles: is there anyway to stop CDRs from saving to textfiles and only have it saved to MySQL? |
| 18:12.50 | TripleFFFF | remove module |
| 18:12.53 | TripleFFFF | or yes |
| 18:13.05 | TripleFFFF | jwei ..rm cdr-custom.blah; then |
| 18:13.15 | TripleFFFF | ln -s /dev/null > cdr-customfilename |
| 18:13.18 | TripleFFFF | i mean |
| 18:13.24 | TripleFFFF | <PROTECTED> |
| 18:13.32 | TripleFFFF | will log to null |
| 18:13.33 | charles___ | [TK]D-Fender: yeah, I saw that, but sometimes I get those glitches. I need a new native girlfriend. |
| 18:13.51 | charles___ | jwei: unload module for text cdr's |
| 18:13.59 | luke-jr_ | mog: ? |
| 18:14.12 | jwei | ok..thanks |
| 18:14.13 | charles___ | jwei: but it won't change much, cdr's are only one line, and after the call ends |
| 18:14.15 | mog | the text to speach engine |
| 18:14.18 | mog | has good voices |
| 18:14.24 | charles___ | jwei: get NPTL first |
| 18:14.28 | luke-jr_ | ah |
| 18:14.33 | jwei | ok |
| 18:14.36 | mog | i think the delay in typing to talking |
| 18:14.39 | mog | would be too much though |
| 18:14.45 | jwei | not logging in debug helps a lot |
| 18:15.02 | jwei | 200 calls has avg load of 12 instead of 15-20 |
| 18:15.12 | luke-jr_ | mog: read fast? =p |
| 18:15.18 | jwei | let me get NPTL |
| 18:15.21 | charles___ | jwei: noload cdr-custom |
| 18:15.33 | mog | reading shouldnt be problem, typing would be |
| 18:15.35 | charles___ | jwei: on asterisk.conf |
| 18:15.35 | mog | i would think |
| 18:15.45 | jwei | i spoke too soon..it's up to 15-18 now... |
| 18:16.19 | *** join/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br) |
| 18:16.33 | charles___ | mog: are you gaucho ? |
| 18:16.54 | mog | am i a cowboy? |
| 18:16.59 | TripleFFFF | error: syntax error before "torisa" |
| 18:17.02 | TripleFFFF | any idea ? |
| 18:17.05 | TripleFFFF | zaptel compile |
| 18:17.05 | charles___ | mog: no, a gaucho is different to a cowboy |
| 18:17.06 | TripleFFFF | centos |
| 18:17.25 | mog | i thought a gaucho was a cowboy from south america |
| 18:17.29 | TripleFFFF | ./usr/src/zaptel/torisa.c:132: error: syntax error before "torisa" |
| 18:17.32 | mog | cant remember which country |
| 18:17.34 | charles___ | mog: just asked because your expression : "bah" |
| 18:17.39 | denon | mog: argentina |
| 18:17.44 | mog | thanks denon |
| 18:17.49 | charles___ | mog it's not a cowboy |
| 18:17.57 | mog | not meaning to offend |
| 18:18.07 | mog | just always had it explained to me thats what a gaucho was |
| 18:18.25 | TripleFFFF | darn |
| 18:18.51 | charles___ | mog get the cowboy bill multiply by 10 put him alone and let him do everything by himself than he will be a gaucho |
| 18:19.08 | watchy | hey tk: these folks are trying to blind transfer and when they do it puts like 20 at the top and leaves a digit at the bottom and wont do it |
| 18:19.11 | watchy | whats the issue? |
| 18:19.12 | TripleFFFF | hever ind |
| 18:19.14 | watchy | it used to work apparently |
| 18:19.16 | TripleFFFF | ~centosbug |
| 18:19.18 | jbot | from memory, centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 18:19.18 | charles___ | mog: and don't let him listen to shania twain or any cowboy shift |
| 18:19.18 | mog | ahh okies charles___ |
| 18:20.02 | TripleFFFF | fixed |
| 18:20.15 | charles___ | mog http://www.geocities.com/ulysses_leal/gaucho.jpg |
| 18:20.34 | mog | looks like a cowboy to me |
| 18:20.35 | hmmhesays | bah fedora thrwarts me |
| 18:20.36 | jwei | charles: I'm having a hard time finding source files for nptl |
| 18:20.44 | mog | but i understand what you were saying |
| 18:20.48 | TripleFFFF | wahts nptl |
| 18:20.49 | *** join/#asterisk DaveHope (n=Dave@internal.davehope.co.uk) |
| 18:20.52 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-6.rcmdva.fios.verizon.net) |
| 18:21.05 | SpaceBass | hey folks |
| 18:21.05 | watchy | how the hell do you turn off sip debugging |
| 18:21.12 | TripleFFFF | sip no debug |
| 18:21.17 | TripleFFFF | set debug 0 |
| 18:21.17 | charles___ | mog The inhabitant of Pampas, vast open fields of northeast Argentina, south Brazil, Uruguay and part of Paraguay known as Gaucho, was a cultural and a blood mestizo. |
| 18:21.19 | TripleFFFF | set verbose 0 |
| 18:21.23 | TripleFFFF | ;) |
| 18:21.25 | charles___ | mog On the battlefield the Gaucho was a fierce opponent, mainly when he took part as member of cavalry. During the War of Cisplatine, the cavalry of both sides were formed of these valiant warriors. |
| 18:21.43 | watchy | thanks |
| 18:21.53 | charles___ | jwei: do cowboys fight on wars too ? |
| 18:21.57 | SpaceBass | trying to track down a bandwidth issue....If I am downloading a bit torrent file at about 1.5mbs/256k (hard limit) on my 15mbs/5mbs connection, called parties complin that Im breaking up ...but I can hear them perfectly |
| 18:21.59 | mog | yes |
| 18:21.59 | charles___ | ops |
| 18:22.03 | mog | or they have |
| 18:22.06 | charles___ | mog cowboys ? |
| 18:22.09 | mog | yes |
| 18:22.13 | mog | see texas |
| 18:22.14 | SpaceBass | how can I determine if the problem is my connection, my ISP, broadvoice, etc |
| 18:22.15 | mog | etc |
| 18:22.18 | mog | its not important though |
| 18:22.20 | mog | seriously |
| 18:22.26 | mog | i know what you are saying |
| 18:22.47 | wangster | what are the proper levels to shoot for in rxgain and txgain when watching ztmonitor? |
| 18:23.18 | TripleFFFF | hey |
| 18:23.18 | TripleFFFF | udptl.c: if ((s = ast_variable_retrieve(cfg, "general", "T38FaxUdpEC"))) { |
| 18:23.23 | tzanger | 14667 I think (from my [admittedly poor] memory) |
| 18:23.23 | TripleFFFF | hmm |
| 18:24.13 | TripleFFFF | trunk rules |
| 18:24.16 | TripleFFFF | i like new install |
| 18:24.31 | wangster | Actually, how would you go about measuring txgain? ztmonitor shows the same value for the tx meter no matter what you txgain to. |
| 18:25.34 | TripleFFFF | Qwell[] so t38 is installe by default on trunk ? |
| 18:27.33 | tzanger | wangster: I use -v so it's numeric |
| 18:27.47 | charles___ | 1800's giving me trouble |
| 18:28.06 | TripleFFFF | 1800'/s |
| 18:28.06 | TripleFFFF | ? |
| 18:28.25 | *** join/#asterisk ste82uk (n=ste82uk@cpc1-kemp2-0-0-cust228.lutn.cable.ntl.com) |
| 18:28.29 | charles___ | can call some can't call others, over vonage |
| 18:28.42 | charles___ | having a ANI set but still doesn't go thru |
| 18:29.05 | charles___ | when I place the call from a Budgetone-100 direct to vonage without asterisk the call get's completed |
| 18:31.02 | *** join/#asterisk blaylock (n=seth@snap.helixsystems.com) |
| 18:31.10 | ste82uk | Hi all. I'm currently banging my head against Shtoom/Doug for something I can probably do directly in Asterisk, but I'm not too sure where to begin.... |
| 18:31.51 | charles___ | TripleFFFF: sounds weird doesn't it ? |
| 18:31.51 | ste82uk | I'd like to place a call through my PSTN provider to a specific number, send DTMF signals at specific intervals and then hang up, having recorded the exchange. |
| 18:32.24 | SpaceBass | ste82uk, check out the wiki and search for call file |
| 18:32.29 | charles___ | ste82uk: asterisk will do it |
| 18:32.39 | charles___ | ste82uk: get asterisk and use AGI to make your inputs |
| 18:32.43 | *** join/#asterisk ikey (i=ikey@220.226.55.159) |
| 18:33.03 | ste82uk | Thanks guys, I'll check that out. Been googling around for "IVR client" for ages! |
| 18:33.03 | charles___ | ste82uk: cron + call file + agi |
| 18:33.11 | ste82uk | Excellent, thank you. |
| 18:33.40 | *** join/#asterisk Mercestes (n=Merceste@216.54.143.2) |
| 18:34.04 | charles___ | ste82uk: thank you ? what ? it's 2 cents dude plus taxes |
| 18:34.37 | ste82uk | LOL, I'll wire it from my Russian bank account if you send me your details ;) |
| 18:34.38 | jwei | charles__: i've found nptl-devel-2.3.4-2.19.i686.rpm and installed it |
| 18:34.47 | dandan | payable in e-gold? :D |
| 18:34.57 | jwei | but it doens't look like it did anything |
| 18:35.12 | *** join/#asterisk aaugustj (n=aaugustj@pool-71-251-28-244.nycmny.fios.verizon.net) |
| 18:35.28 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-241-188-23.buff.east.verizon.net) |
| 18:35.35 | *** join/#asterisk WildPikachu (n=WildPika@unaffiliated/wildpikachu) |
| 18:35.38 | SuPrSluG | hello |
| 18:35.45 | WildPikachu | how can i make asterisk not try bridge a sip outbound call? |
| 18:35.49 | aaugustj | hi everyone |
| 18:35.59 | SuPrSluG | anyone having problems with voxee? |
| 18:36.11 | file | WildPikachu: what do you mean? you mean reinvite? |
| 18:36.16 | SuPrSluG | i can't register or show as a peer |
| 18:36.27 | aaugustj | i moved my working asterisk dev install to a new server, but my X100P card no longer takes incoming calls |
| 18:36.48 | WildPikachu | file, i'm behind NAT connecting to my asterisk box, if i call out the other side can hear me, i can't hear them |
| 18:36.49 | aaugustj | anyone have any idea what moving my install over to a new system could have done to the config? |
| 18:37.15 | file | WildPikachu: is your Asterisk box behind NAT? |
| 18:37.39 | WildPikachu | file, yep |
| 18:37.48 | file | WildPikachu: have you setup sip.conf? |
| 18:37.51 | file | WildPikachu: and forwarded ports? |
| 18:38.11 | WildPikachu | I'm using exten => 2,1,Dial(SIP/613@fwd.pulver.com) to test echo |
| 18:38.13 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 18:41.50 | *** join/#asterisk wunderkin (n=wunderki@216-19-202-13.getnet.net) |
| 18:43.39 | aaugustj | here's the beginning of my zapata.conf, if its any help: http://pastebin.ca/112130 |
| 18:43.40 | TripleFFFF | question... wy does WMI not working behind nat ? |
| 18:44.05 | aaugustj | it looks as though my X100P lost its config/connection to AAH |
| 18:44.30 | aaugustj | i've used genzaptelconf, ztcfg, lspci, etc to troubleshoot this but so far, nothing helpful |
| 18:45.13 | rob0 | X100P and a@h ... not the best combination to talk about in here. :) |
| 18:45.17 | cytrak | anyone knows what this is ? Got reject for frame 8, retransmitting frame 8 now, updating n_r! |
| 18:45.19 | rob0 | Both are junk |
| 18:45.28 | cytrak | it goes on and on |
| 18:45.41 | nortex | And the channel topic is pretty clear |
| 18:45.45 | rob0 | <== uses an x101p, don't tell anyone :) |
| 18:46.17 | aaugustj | rob0: well it IS a dev box - i'd prefer not to spend big bucks on it |
| 18:46.18 | cytrak | I keep on getting it all the time now even after shutting down and restarting asterisk |
| 18:46.36 | cytrak | this on too : Got reject for frame 33, but we have nothing -- resetting! |
| 18:46.38 | cytrak | help |
| 18:46.58 | nortex | aaugustj, then toss AAH and load asterisk from the source :) |
| 18:47.17 | aaugustj | sounds like a fun weekend project... |
| 18:47.30 | aaugustj | or rather...a fun winter project |
| 18:48.07 | nortex | aaugustj, As far as you pastebin you have not told asterisk to configure a ZAP channel. |
| 18:49.05 | *** part/#asterisk lo_tek (n=tone_loc@user-1121l4v.dsl.mindspring.com) |
| 18:49.11 | momelod | can anyone suggest a frontend for cdr entries in mysql? |
| 18:49.17 | momelod | web frontend |
| 18:49.20 | aaugustj | nortex: well, it also loads a zapata-auto.conf |
| 18:49.26 | cytrak | momelod: I've been checking voiceone |
| 18:49.35 | momelod | is it good? |
| 18:49.38 | cytrak | but haven't had a chance to actually install it |
| 18:49.41 | cytrak | look good |
| 18:49.43 | momelod | lol |
| 18:49.49 | momelod | k, ill look it up |
| 18:49.56 | cytrak | it also uses the asterisk real time scheme |
| 18:50.09 | aaugustj | nortex: here's the relevant excerpt from zapata-auto: http://pastebin.ca/112141 |
| 18:50.11 | cytrak | which I'm setting up on a non production server for testing |
| 18:50.40 | momelod | is that freeware? |
| 18:50.43 | cytrak | why do those got frame reject happer n ? |
| 18:50.59 | aaugustj | though, if theres no easy solution for this, i'll likely toss the box and re-install (most likely not from source though) |
| 18:51.17 | cytrak | http://www.voiceone.it |
| 18:51.41 | nortex | aaugustj, WCFXO/0 "Wildcard X100P Board 1" RED try a different IRQ. |
| 18:51.58 | nortex | or PCI slot |
| 18:52.35 | charles___ | did you guys see the patch do have 3d card FPU to do G723 and get extra voice fps ? |
| 18:53.31 | aaugustj | nortex: ive been thinking thats the problem - the box im running it on had it sharing IRQ's - though the current placement avoids that problem |
| 18:53.45 | aaugustj | but what does the RED indicate? |
| 18:53.50 | *** join/#asterisk gandhijee (n=gandhije@65.169.134.2) |
| 18:54.40 | aaugustj | nortex: my lspci -vv command indicates that the card is the only device on IRQ 10 |
| 18:54.44 | *** join/#asterisk hmmhesays (n=ohyeah@66.173.103.110) |
| 18:54.51 | hmmhesays | WARNING: Error inserting zaptel (/lib/modules/2.6.5/misc/zaptel.ko): Invalid module format anyone seen this before? |
| 18:55.02 | mog | dmesg |
| 18:55.13 | momelod | is there any web front end that just does cdr? |
| 18:55.14 | mog | very likely module was not built against your kernel |
| 18:55.16 | mog | or wrong gcc |
| 18:55.38 | nortex | momelod, There is on, I'm looking for the name now . .. |
| 18:55.39 | hmmhesays | mog, .. it was |
| 18:55.45 | momelod | thanx nortex |
| 18:55.54 | mog | ^_^ |
| 18:55.58 | hmmhesays | i just built it, gcc version, same as the kernel build gcc version |
| 18:56.01 | *** join/#asterisk _4d4m_ (n=adam@62.69.102.99) |
| 18:56.35 | hmmhesays | i don't get it |
| 18:57.08 | TripleFFFF | m=image 4866 udptl t38 |
| 18:57.08 | TripleFFFF | a=T38FaxVersion:0 |
| 18:57.08 | TripleFFFF | a=T38MaxBitRate:9600 |
| 18:57.08 | TripleFFFF | a=T38FaxFillBitRemoval:0 |
| 18:57.08 | momelod | did u build it against your existing kernel headers? |
| 18:57.09 | TripleFFFF | ;) |
| 18:57.13 | hmmhesays | yes |
| 18:57.21 | mog | do a dmesg and pastebin it |
| 18:57.26 | momelod | depmod -a |
| 18:57.28 | l-fy | hello |
| 18:57.35 | l-fy | the openvox cards are digium clones? |
| 18:57.48 | nortex | momelod, http://www.voip-info.org/wiki/view/Asterisk+CDR+Areski+GUI |
| 18:57.53 | momelod | nortex: thank you! |
| 18:58.13 | *** join/#asterisk Skaag (n=hintza@212.199.180.157.static.012.net.il) |
| 18:58.34 | nortex | momelod, It takes some kicking to get it up and a new version is supposed to be in the works, but I have had good luck with it since I finished kicking :) |
| 18:58.52 | *** join/#asterisk bmg505 (n=leon@c1-100-16.rndf.isadsl.co.za) |
| 18:58.53 | Skaag | i've installed asterisk on debian, it's now listening on ports 2000 and 1720, which one is the SIP port? |
| 18:59.30 | *** join/#asterisk DaveHope (n=Dave@internal.davehope.co.uk) |
| 18:59.49 | Skaag | ok forget my question |
| 18:59.51 | nortex | aaugustj, Check voip-info for information about zttool |
| 19:00.41 | TripleFFFF | where is t38 module in asterisk cli |
| 19:00.45 | TripleFFFF | how can i see if its in there |
| 19:01.06 | hmmhesays | wooo pastebin is taking its sweet time |
| 19:01.20 | *** join/#asterisk hohum (n=dcorbe@mail.interceltelecoms.com) |
| 19:01.21 | hmmhesays | http://pastebin.ca/112153 |
| 19:03.17 | mog | its part of chan_sip |
| 19:03.20 | mog | and rtp |
| 19:03.23 | mog | not its own module |
| 19:04.54 | *** join/#asterisk Ark_Molt (n=chatzill@65.203.186.131) |
| 19:05.57 | Ark_Molt | can somebody tell me how to set a variable to the return of an AGI funciton call? |
| 19:07.21 | Ark_Molt | i've got something like ... exten=> s,4,SetVar(agiResult=${AGI(agi://${IP}/callerid.agi)}) |
| 19:08.01 | Ark_Molt | then a GotoIf() based on the result, but the agi wont execute to give me a result variable like -1 |
| 19:08.13 | Ark_Molt | any thought anybody? |
| 19:08.47 | *** join/#asterisk budmang (n=budman@12.206.134.162) |
| 19:09.00 | budmang | how do I get the ZAP command to work on asterisk CLI? |
| 19:09.17 | budmang | zttest and zttool work fine. Ive recomplied asterisk twice. |
| 19:10.19 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 19:10.52 | *** join/#asterisk Corydon-w (n=tilghman@pdpc/supporter/sustaining/Corydon76-home) |
| 19:10.52 | *** mode/#asterisk [+o Corydon-w] by ChanServ |
| 19:11.37 | hmmhesays | bah he compiled his own kernel and got the libs off of the yum archive |
| 19:11.40 | *** join/#asterisk SwK (n=Silik0nJ@65.169.134.2) |
| 19:13.55 | *** join/#asterisk muh-die-kuh (n=hco@muh-die-kuh.de) |
| 19:14.08 | budmang | any helpers in? |
| 19:14.10 | muh-die-kuh | hi |
| 19:14.37 | muh-die-kuh | are there any voip phones which can synchronize their phonebook via ldap? |
| 19:14.39 | nortex | budmang, Can you explain a little better? You mena like zap shp channels? |
| 19:14.56 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 19:15.08 | nortex | budmang, I mean zap show channels |
| 19:15.13 | charles___ | muh-die-kuh: get the phone to access a xml |
| 19:15.14 | budmang | correct nortex |
| 19:15.17 | budmang | zap show channels |
| 19:15.17 | muh-die-kuh | and that are capable of use with asterisk, by the way |
| 19:15.18 | nortex | Sticky fingers :) |
| 19:15.23 | budmang | says zap not found |
| 19:15.26 | charles___ | muh-die-kuh: phonebook can get very big and not fit on the phone |
| 19:15.31 | budmang | asterisk*CLI> zap show status |
| 19:15.32 | budmang | No such command 'zap' (type 'help' for help) |
| 19:15.33 | sp0n9e | how's the echo cancellation on the Sangoma Remora A200? |
| 19:15.50 | nortex | Did you make and make install zaptel? |
| 19:15.56 | charles___ | muh-die-kuh: I've get my phones here to sync with egroupware |
| 19:16.07 | charles___ | muh-die-kuh: I mean, search on egroupware |
| 19:16.10 | budmang | zaptell is fully running. |
| 19:16.20 | budmang | zttest and zttool show up and work fine. |
| 19:16.56 | budmang | -bash-3.00# ./zttest |
| 19:16.56 | budmang | Opened pseudo zap interface, measuring accuracy... |
| 19:16.56 | budmang | 99.975586% 99.963379% 99.938965% |
| 19:17.06 | CunningPike | budmang: zapata.conf says.....? |
| 19:17.08 | muh-die-kuh | charles___: that means, the phonebook isned stored on the phone, its just stored on the groupware server? |
| 19:17.22 | nortex | What if you do load chan_zap.so |
| 19:17.26 | charles___ | muh-die-kuh: exactly |
| 19:17.31 | nortex | from the asterisk cli |
| 19:17.40 | budmang | asterisk*CLI> load chan_zap.so |
| 19:17.40 | budmang | Unable to load module chan_zap.so |
| 19:17.43 | charles___ | muh-die-kuh: but tapping 3 buttons I can make a search |
| 19:17.53 | budmang | CunningPike where is that? |
| 19:17.53 | charles___ | muh-die-kuh: or browse it |
| 19:17.56 | *** join/#asterisk samoaz (n=user@shpongle.netissat.bg) |
| 19:18.02 | hmmhesays | where is the default directory gcc looks for libs? |
| 19:18.02 | CunningPike | budmang: /etc/asterisk |
| 19:18.05 | charles___ | muh-die-kuh: egroupware has around 2k entries |
| 19:18.31 | charles___ | hmmhesays: cat /etc/ld.so.conf |
| 19:18.34 | budmang | CunningPike: zapata.conf.template |
| 19:18.46 | CunningPike | budmang: No zapata.conf? |
| 19:18.48 | budmang | should I cp it with out the .template? |
| 19:18.53 | nortex | budmang, did you build zaptel after asterisk ? |
| 19:18.59 | muh-die-kuh | charles___: sounds even better, to be honest. so, does this feature have a special name? or can you tell me some phones which can do this? |
| 19:19.08 | budmang | nortex yes. |
| 19:19.10 | CunningPike | budmang: That, and you should edit it to suit your configuratiob |
| 19:19.16 | budmang | Ive maked it twice |
| 19:19.21 | CunningPike | s/iob/ion/ |
| 19:19.23 | charles___ | muh-die-kuh: I'm using cisco's 7940 and I do recommend them |
| 19:19.26 | charles___ | muh-die-kuh: they work fine |
| 19:19.42 | charles___ | muh-die-kuh: with paging and everything |
| 19:20.13 | nortex | zaptel before asterisk always :) |
| 19:20.25 | sp0n9e | if my pstn lines are about 3 feet long, i shouldn't get any echo on the line, right? |
| 19:20.26 | muh-die-kuh | charles___: yeah i see. and the price is okay, too. you now whether some wireless phones are capable of this, too? |
| 19:21.01 | samoaz | hi, |
| 19:21.08 | charles___ | muh-die-kuh: probably the cisco's wireless but I'm not using them, I use ATA + regular cordless phones |
| 19:21.17 | samoaz | i wonder whether a particular setup is possible? |
| 19:21.18 | charles___ | muh-die-kuh: on the desk people can use regular corded |
| 19:21.29 | budmang | now that zaptel is installed |
| 19:21.33 | muh-die-kuh | charles___: key, thanks :) |
| 19:21.36 | budmang | I should just fresh install you think? |
| 19:21.50 | samoaz | one ser and few asterisk boxes behind it sharing the load of 2k subscribers? |
| 19:22.13 | CunningPike | sp0n9e: Echo has little to do with line length and more to do with the hybrid at the other ens |
| 19:22.17 | CunningPike | s/ens/end/ |
| 19:22.24 | charles___ | muh-die-kuh: out of the desk they get cordless, but here for example we don't originate calls from cordless, just transfer to it and get to work on equipment |
| 19:22.56 | nortex | budmang, You should be able to do a make install in asterisk since zaptel is done and have it pickup the module. |
| 19:23.00 | muh-die-kuh | charles___: okay, sounds senseful |
| 19:23.18 | budmang | nortex trying again now with that zapta conf in place too |
| 19:23.32 | sp0n9e | CunningPike: well, the type of access we're getting is 10 lines from some voip that cox is doing to their datacenter...but we have to use POTS from the cisco box to the asterisk server |
| 19:23.40 | sp0n9e | just wondering what kind of echo there would be |
| 19:23.52 | sp0n9e | and if it would justify hardware echo cancellation |
| 19:24.07 | charles___ | muh-die-kuh: if you need generate xml's to access ldap's database with your cisco phone as frontend I can help you out |
| 19:24.43 | CunningPike | sp0n9e: Depends on the quality of the hybrid in the Cisco gear - should be good. I'd try without and see |
| 19:25.09 | sp0n9e | with the sanoma i could always grab the hardware echo module and plug it in later, right? |
| 19:25.30 | muh-die-kuh | charles___: i just said ldap, cus this was what i thought it would be most sensefull to store this data. i would survive if i would have to save it as an xml file ;) |
| 19:25.50 | CunningPike | sp0n9e: I believe so - you can certainly do that with the Digium cards. Or you can add a Telabs, or some other external EC. |
| 19:25.51 | budmang | same error asterisk*CLI> load chan_zap.so |
| 19:25.51 | budmang | Unable to load module chan_zap.so |
| 19:25.52 | sp0n9e | s/sanoma/sangoma |
| 19:26.08 | charles___ | muh-die-kuh: cisco won't load huge xml's , just 30 entries per xml :P |
| 19:26.20 | samoaz | any idea how exactly to do this ? |
| 19:26.25 | CunningPike | budmang: It won't load without a zapata.conf...... |
| 19:26.34 | nortex | budmang, After doing the make you restarted the asterisk service. |
| 19:26.40 | samoaz | please, is it possible at all? |
| 19:26.42 | budmang | i did restart it |
| 19:26.46 | muh-die-kuh | charles___: uhm, okay... ill have a look at the cisco website |
| 19:26.47 | budmang | zapata.conf is there now |
| 19:26.57 | CunningPike | samoaz: What? |
| 19:27.33 | budmang | should i stop asterisk whiling making / make installing? |
| 19:27.38 | budmang | would that matter? |
| 19:27.40 | samoaz | to do load balancing with few asterisk boxes and a ser in front of the serving as the load balancer ... |
| 19:28.01 | trelane_ | ok, chanspy TOTALLY OWNS |
| 19:28.03 | CunningPike | samoaz: I believe so - you looked in the wiki? |
| 19:28.04 | nortex | Should not matter. Did you edit the zapata.conf file or just rename it. |
| 19:28.20 | budmang | renamed it. Nothing inside looked like it needed to be edited. |
| 19:28.27 | *** join/#asterisk Waverly360 (n=mirc@65.169.134.2) |
| 19:28.46 | samoaz | for example every time a subscriber tries to register to the ser it is redirected to one of the asterisks and registers there ... |
| 19:29.34 | CunningPike | budmang: What verbosity level are you using in asterisk? |
| 19:29.48 | budmang | asterisk -rvvvvc |
| 19:29.49 | samoaz | the asterisks share one database thus knowing which subscriber on which asterisk is registered ... |
| 19:32.04 | samoaz | CunningPike: u mean voip-info.org? |
| 19:32.24 | CunningPike | samoaz: Yes |
| 19:32.54 | nortex | budmang, Check in /usr/lib/asterisk/modules for chan_zap.so |
| 19:32.56 | hmmhesays | what a pain in the arse |
| 19:33.02 | hmmhesays | i wish people wouldn't compile their own kernels |
| 19:33.34 | [TK]D-Fender | OMG another big LCD price drop...... |
| 19:33.49 | charles___ | [TK]D-Fender: 200 for 20 inches ? |
| 19:34.08 | nortex | I'd like 20 for 20 better :) |
| 19:34.29 | [TK]D-Fender | charles___ : 260$CAD for 20" Widescreen 1680*1050 |
| 19:34.40 | [TK]D-Fender | charles___ : retail. |
| 19:34.58 | *** join/#asterisk justnulling2 (n=justnull@ool-182e45b6.dyn.optonline.net) |
| 19:35.27 | *** part/#asterisk sp0n9e (n=sp0n9e@69.12.216.48) |
| 19:35.43 | TripleFFFF | what ? |
| 19:35.44 | TripleFFFF | where |
| 19:36.02 | *** part/#asterisk _alex_mx_ (n=_alex_mx@200.78.229.18) |
| 19:36.04 | budmang | -bash-3.00# ls -la | grep chan_zap |
| 19:36.10 | budmang | nothing nortex |
| 19:36.15 | TripleFFFF | fender ? |
| 19:36.44 | justnulling2 | how do i login to voicemail at another ext then my phone's? i have tried *97 and *98 but it doesn't log me in (doesn't look like it is using ext that i have typed in) any pointers? |
| 19:36.50 | [TK]D-Fender | TripleFFFF : nevermind... I'm blind... I hit the .COM US site, not the .ca one :( |
| 19:36.53 | TripleFFFF | 8500 |
| 19:36.55 | TheCompWiz | budmang you need chan_zap.so |
| 19:36.56 | TripleFFFF | try that |
| 19:37.03 | TripleFFFF | lol |
| 19:37.07 | TripleFFFF | try ncix.com |
| 19:37.12 | TripleFFFF | canada largest |
| 19:37.21 | CunningPike | justnulling2: What extension have you set up for voicemail access? |
| 19:37.26 | TripleFFFF | try 8500 |
| 19:37.29 | [TK]D-Fender | TripleFFFF : tigerdirect = bigger, but they can be real fuckups sometimes |
| 19:37.35 | budmang | TheCompWiz why would that not be there? |
| 19:37.39 | budmang | where can I get it? |
| 19:37.41 | Nivex | newegg > tigerdirect |
| 19:37.57 | [TK]D-Fender | Nivex : Newegg = no CAD division and won't even SHIP here |
| 19:38.05 | TheCompWiz | when you compiled asterisk... you didn't use the right command line switches? or forogot to run .configure... who knows. |
| 19:38.08 | Nivex | hmm, that would be a problem |
| 19:38.16 | nortex | budmang, Try running a make and make install in both Zaptel and asterisk again. |
| 19:38.54 | nortex | [TK]D-Fender, ship to a buddy in the US :) then acroos the boarder. |
| 19:39.08 | [TK]D-Fender | nortex : Costly and time consuming |
| 19:39.14 | [TK]D-Fender | nortex : not worth it, |
| 19:39.34 | nortex | [TK]D-Fender, Do it all the time for then in-laws in Toronto area :) |
| 19:39.45 | hmmhesays | how do i find out the full path this is looking for /bin/sh: scripts/genksyms/genksyms: No such file or directory |
| 19:40.27 | CunningPike | hmmhesays: locate genksyms |
| 19:40.43 | *** join/#asterisk angom_w (n=angom@red-corp-200.79.158.218.telnor.net) |
| 19:41.09 | TripleFFFF | SO off net = tdm and ON net = to sip provider right ? |
| 19:41.26 | justnulling2 | CunningPike: what do you mean? lets say i have this setup my phone ext is 123 (no voicemail) and my voicemail ext is 456 and my pass is 999 but when i try it i get login incorrect |
| 19:41.52 | rob0 | hmmhesays: do you have the kernel source installed? |
| 19:42.12 | hmmhesays | yes, this kernel was compiled from the source |
| 19:42.35 | hmmhesays | the source in in /usr/src/linux |
| 19:42.44 | justnulling2 | CunningPike: in both cases *97/98 it is using my phone ext Executing VoiceMailMain("IAX2/50300-5", "123@disabled") in new stack |
| 19:42.47 | Skaag | i'm trying to connect my Ekiga to my Asterisk, but I get a 401 error. I think I set up the user in sip.conf properly, any ideas? |
| 19:43.07 | Skaag | I get: SIP/2.0 401 Unauthorized |
| 19:43.19 | *** join/#asterisk popvoxdave (n=popvoxda@c-71-206-59-174.hsd1.md.comcast.net) |
| 19:43.22 | rob0 | Anyway, "scripts/genksyms/genksyms" is a relative path to the working directory. |
| 19:43.25 | *** join/#asterisk robl^ (n=robl@dsl093-025-218.hou1.dsl.speakeasy.net) |
| 19:43.27 | [TK]D-Fender | Skaag : bad auth info |
| 19:43.44 | CunningPike | justnulling2: You need another access number - one that calls VoiceMailMain() instead of VoiceMailMain(${EXTEN}@foo) |
| 19:44.25 | hmmhesays | that is odd, since we are in zaptel-1.2.7 |
| 19:44.46 | justnulling2 | CunningPike: freepbx doesn't came with one? |
| 19:44.54 | hmmhesays | the file it is looking for is in /usr/src/linux/scripts/genksyms/ |
| 19:45.06 | CunningPike | ~lart justnulling2 |
| 19:45.33 | justnulling2 | what was that? |
| 19:45.35 | nortex | justnulling2, then you in the wrong channel :) |
| 19:45.42 | *** join/#asterisk Assid (n=assid@203.212.216.40) |
| 19:45.44 | [TK]D-Fender | justnulling2 : Please read the channel topic... |
| 19:45.57 | justnulling2 | oh my bad |
| 19:46.14 | Assid | and once again im down with no net.. and having to use some alternate method of connecting to the internet |
| 19:47.12 | Skaag | [TK]D-Fender: I'm setting the username in [user], and the password in secret=... |
| 19:47.17 | Skaag | what else is there |
| 19:47.19 | *** join/#asterisk Trakkasure (n=Sgemtum@adsl-068-153-217-253.sip.bct.bellsouth.net) |
| 19:47.27 | [TK]D-Fender | News : Polcom SIP 2.0 expected mid-August per my rep. |
| 19:47.42 | CunningPike | [TK]D-Fender: Interesting - what will it bring? |
| 19:47.46 | [TK]D-Fender | Skaag : Something doesn't match... |
| 19:48.18 | [TK]D-Fender | CunningPike : Not a ton of stuff from what I can see just yet. Beta has been replaced by 1.6.7 and not sure what the final outlay will be like. |
| 19:48.28 | [TK]D-Fender | CunningPike : A lot of LCS stuff, and so on. |
| 19:48.34 | CunningPike | [TK]D-Fender: LCS? |
| 19:48.43 | [TK]D-Fender | CunningPike : And just when I was about to mass upgrade us to 1.6.7 here :) |
| 19:48.44 | nortex | [TK]D-Fender, Good deal, you got a good rep, I'm still waiting that 1.6.7 download link from mine :) |
| 19:48.45 | Skaag | Is ACL supposed to be NO for my user? |
| 19:48.50 | budmang | Aug 2 12:51:36 WARNING[19806] loader.c: libpri.so.1.0: cannot open shared objec t file: No such file or directory |
| 19:48.50 | budmang | Aug 2 12:51:36 WARNING[19806] loader.c: Loading module chan_zap.so failed! |
| 19:48.53 | [TK]D-Fender | CunningPike : MS Live Communication Server. |
| 19:49.01 | CunningPike | [TK]D-Fender: Ugh |
| 19:49.12 | TripleFFFF | hey |
| 19:49.20 | TripleFFFF | in trunk theres a bunch of SPEach stuff |
| 19:49.21 | [TK]D-Fender | CunningPike : Never said I liked or needed it for anything.... |
| 19:49.25 | *** join/#asterisk Astinus- (n=a@213.167.111.138) |
| 19:49.26 | TripleFFFF | is that doc'ed ? |
| 19:49.28 | nortex | budmang, did you build libpri? |
| 19:49.37 | CunningPike | budmang: You installed libpri, yes? |
| 19:49.55 | Skaag | ok fixed it |
| 19:49.58 | CunningPike | budmang: zaptel -> libpri -> asterisk |
| 19:50.06 | *** join/#asterisk tomlobato (n=tomlobat@201-43-224-3.dsl.telesp.net.br) |
| 19:50.10 | budmang | wheres a good document on libpri? |
| 19:50.38 | CunningPike | budmang: Don't need one - no user serviceable components inside |
| 19:50.53 | budmang | How do I install it? |
| 19:51.22 | TripleFFFF | doanyone ? |
| 19:51.35 | tomlobato | please, I cant register to FWD account as asterisk oreilly book says, using iax2 debug I get: CAUSE : Registration Refused |
| 19:51.35 | tomlobato | <PROTECTED> |
| 19:51.44 | CunningPike | budmang: Download the tarball, unzip, make clean, make, make install |
| 19:51.57 | tomlobato | someone know whats means error code 29? |
| 19:53.15 | budmang | CunningPike got the svn downloading now thanks to nortex |
| 19:53.48 | CunningPike | budmang: OK - what versions of zaptel and asterisk are you running? svn also? |
| 19:54.46 | Assid | when did 1.6.7 come in? |
| 19:55.00 | TripleFFFF | ? |
| 19:55.15 | TripleFFFF | whats 1.6.7 |
| 19:55.33 | nortex | I head about the 1.6.7 Polycom firmaware yesterday here |
| 19:55.42 | [TK]D-Fender | Assid : July 7th. |
| 19:55.57 | [hC] | Oooo |
| 19:56.02 | [hC] | Anyone have a changelog? |
| 19:56.06 | Assid | wasnt that one 1.6.6 ? |
| 19:56.13 | [TK]D-Fender | [hC] : I have it all, including the 3.2.1 BR |
| 19:56.43 | [TK]D-Fender | Full support for the IP430 between them, and much faster SIP response apparently |
| 19:57.07 | Assid | [TK]D-Fender: any betterment in quality/response etc? |
| 19:57.07 | TripleFFFF | anyone have BTPD yup ? |
| 19:57.52 | [TK]D-Fender | Assid : Haveup/down graded yet. My home is split between SIP1.6.6 and 2.0.0 Beta |
| 19:58.53 | Assid | any particular reason you dont like the 1.6.7 ? |
| 19:59.04 | [TK]D-Fender | Assid : I just GOT it :) |
| 19:59.13 | TripleFFFF | `OCF_READ_LINK_QUALITY' undeclared |
| 19:59.15 | Assid | hehe |
| 19:59.17 | TripleFFFF | getting that nice error |
| 19:59.38 | budmang | CunningPike: Aug 2 13:00:46 WARNING[7451] loader.c: libtonezone.so.1.0: cannot open shared object file: No such file or director |
| 19:59.43 | budmang | now after installing the libri |
| 20:00.43 | *** join/#asterisk ArkonaDev-Mike (n=somewher@65.203.186.131) |
| 20:01.03 | Ark_Molt | in the dial plan, can i set a variable equal to what an AGI() returns? if so how?? |
| 20:02.34 | Ark_Molt | really i need to know if an AGI() successfully went through or not before moving on in my dial plan... any thoughts? any body? |
| 20:02.44 | Skaag | ok I found out that when I do: sip show settings that realm auth is turned off |
| 20:03.06 | CunningPike | budmang: Never seen that before |
| 20:03.09 | Skaag | i've managed to register once, but now I can no longer register. My softphone insists on sending a realm... how do I include the realm in auth? |
| 20:03.27 | [TK]D-Fender | Skaag : "asterisk" <- |
| 20:03.34 | Skaag | I use this |
| 20:03.39 | Skaag | I know it's the default |
| 20:03.55 | Skaag | but it seems like asterisk is not including it in the authentication process |
| 20:04.22 | Skaag | I turned on nat for my peer (i'm in an office behind nat), and canreinvite=no |
| 20:05.21 | budmang | Aug 2 13:00:46 WARNING[7451] loader.c: libtonezone.so.1.0: cannot open shared object file: No such file or director y |
| 20:05.21 | budmang | Aug 2 13:00:46 WARNING[7451] loader.c: Loading module chan_zap.so failed! |
| 20:05.28 | budmang | i get that now and asterisk willnot start. |
| 20:10.48 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 20:11.20 | hmmhesays | anyone remember the command to put selinux in permissive mode? |
| 20:11.44 | *** join/#asterisk proxyfrog1 (n=bluewave@adsl-75-4-195-66.dsl.irvnca.sbcglobal.net) |
| 20:11.49 | proxyfrog1 | watsup ppl |
| 20:11.57 | *** join/#asterisk dasenjo (n=dasenjo@200.21.65.141) |
| 20:11.58 | proxyfrog1 | who's on |
| 20:12.02 | *** join/#asterisk beyond (n=beyond@200.192.160.100) |
| 20:12.03 | syzygyBSD | nobody |
| 20:12.07 | proxyfrog1 | hey dasenjo |
| 20:12.17 | proxyfrog1 | hey syzygyBSD |
| 20:12.23 | proxyfrog1 | hey beyond |
| 20:12.23 | syzygyBSD | hmmhesays: the only way I know how to do that is in the confid |
| 20:12.24 | hmmhesays | nm setenforce PERMISSIVE |
| 20:12.46 | proxyfrog1 | hey ppl.......sendmail is too complex |
| 20:12.52 | proxyfrog1 | imma throw in the towel |
| 20:12.58 | dasenjo | !blame proxyfrog1 |
| 20:13.04 | proxyfrog1 | hehe |
| 20:13.07 | syzygyBSD | lol.. sendmail is too complex? |
| 20:13.10 | dasenjo | Hi everybody! |
| 20:13.20 | budmang | CunningPike: you there? |
| 20:13.22 | proxyfrog1 | hey man, sendmail is like using a hatchet to kill a fly |
| 20:13.24 | SplasPood | [TK]D-Fender: eh.. 1.6.7 holds no thrill for me.. |
| 20:13.28 | CunningPike | budmang: Yup |
| 20:13.30 | budmang | Can I pm you ? |
| 20:13.32 | *** join/#asterisk santiago (i=santiago@debian/developer/santiago) |
| 20:13.34 | CunningPike | Sure |
| 20:13.38 | proxyfrog1 | i just wanna have a simple mta, dude |
| 20:13.52 | rob0 | proxyfrog1: Postfix or Exim |
| 20:13.55 | proxyfrog1 | can someone suggest a simple mta to use |
| 20:14.09 | proxyfrog1 | kool rob0 |
| 20:14.21 | proxyfrog1 | i heard of postfix.........exim is new to me |
| 20:14.34 | dasenjo | I'm trying to print (NoOp) the DIALSTATUS variable after invoking the Dial app without success. Can you help me? |
| 20:14.49 | proxyfrog1 | i got my asterisk box and cant send out voicemail as emails......even tho webmin sends out emails just fine to my isp's mail server |
| 20:14.54 | eKo1 | dasenjo: post the noop here please |
| 20:15.16 | beyond | proxyfrog1, hey dude |
| 20:15.19 | rob0 | Oh yeah, Postfix would be pretty easy in a setup like that. |
| 20:15.21 | proxyfrog1 | hey bro watsup |
| 20:15.23 | eKo1 | proxyfrog1: check your sendmail config |
| 20:15.39 | rob0 | but then ... sendmail shouldn't be that much harder |
| 20:15.46 | *** part/#asterisk bpiper (n=bpiper@70.159.49.40) |
| 20:16.00 | syzygyBSD | proxyfrog1: you probably want exim |
| 20:16.02 | proxyfrog1 | bro, sendmail is working if i send mail using webmin |
| 20:16.06 | syzygyBSD | what distro are you on |
| 20:16.07 | dasenjo | eKo1, exten=s,12,Dial(Zap/1/${ARG1},25|${options}) |
| 20:16.07 | dasenjo | exten=s,13,NoOp(${DIALSTATUS}) |
| 20:16.35 | proxyfrog1 | but when asterisk tries to send email out, it gets a user authetication kickback from my isp........ |
| 20:16.44 | Skaag | I don't get this... my asterisk is trying to contact my phone even though i've disconnected it long ago |
| 20:16.52 | dasenjo | eKo1, I'm calling a busy line, but I got HungUp and congestion without printing anything |
| 20:17.12 | Skaag | how do I turn that behavior off? |
| 20:17.14 | rob0 | oh ... well it probably won't be simple to set up client AUTH in Postfix. |
| 20:17.25 | proxyfrog1 | hey........if i go with postfix or exim, how do i turn off sendmail for asterisk? |
| 20:17.26 | eKo1 | dasenjo: strange |
| 20:17.28 | *** join/#asterisk ronaldl79 (n=chatzill@d198-53-139-22.abhsia.telus.net) |
| 20:17.55 | syzygyBSD | proxyfrog1: the sendmail command is overwritten with those, it is automatic if you do it from a package |
| 20:17.57 | eKo1 | proxyfrog1: alias sendmail to postfix |
| 20:17.59 | Skaag | proxyfrog1: it probably uses 'sendmail' binary so just make sure the sendmail binary it uses points to exim/postfix and not sendmail itself |
| 20:18.00 | rob0 | proxyfrog1 never answered what OS/distro? |
| 20:18.04 | dasenjo | eKo1, I'm using *-1.0.10 |
| 20:18.24 | eKo1 | dasenjo: upgrade to 1.2 |
| 20:18.25 | proxyfrog1 | i'm using trixbox.......so its centos |
| 20:18.35 | ronaldl79 | Anyone experiencing SIP timeouts/connectivity issues when using more than 2-3 active SIP trunks?? (* 1.2.* & TRUNK) |
| 20:18.40 | TripleFFFF | anyone have bluettoth workoing ? |
| 20:18.41 | TripleFFFF | thanks |
| 20:18.51 | *** join/#asterisk s0lid (n=jlq@124.6.176.100) |
| 20:18.56 | dasenjo | eKo1, I can't do it now ... just in this server |
| 20:19.05 | proxyfrog1 | trixbox got it so tat u boot with asterisk and sendmail running |
| 20:19.23 | eKo1 | dasenjo: you're going to have to do something then because I'm running 1.0.10 and it works fine |
| 20:19.29 | rob0 | just use the distro packaging tools (yum?) to remove sendmail and install postfix. |
| 20:19.50 | proxyfrog1 | i see........kool......tat would work i guess |
| 20:20.21 | proxyfrog1 | hey when u get a voicemail r u suppose to hear a stutter tone when u pick up the phone? |
| 20:21.00 | rob0 | Did a stutterer call and leave voice mail? |
| 20:21.07 | [TK]D-Fender | rob0 : lol |
| 20:21.12 | *** join/#asterisk nettie (i=[U2FsdGV@85-18-54-38.ip.fastwebnet.it) |
| 20:21.13 | proxyfrog1 | wat i mean is.....instead of hearing a steady tone like usual, u'd hear the dialtone stutter 3 times |
| 20:21.25 | proxyfrog1 | lol........ |
| 20:21.44 | *** join/#asterisk mitcheloc (n=mitchelo@69-167-145-62.lmdaca.adelphia.net) |
| 20:21.55 | dasenjo | eKo1, thanks anyway |
| 20:21.57 | proxyfrog1 | yea might as well route tat stutterer to my sendmail |
| 20:22.20 | hmmhesays | is there any conferences that don't require a timer, besides app_conference? |
| 20:22.22 | proxyfrog1 | dude.......imma look into postfix and exim......... |
| 20:22.46 | syzygyBSD | meetme doesnt' require a timer does it |
| 20:23.01 | proxyfrog1 | see which is easier to configure with using isp's mail server to send outbound mail.......... |
| 20:23.10 | syzygyBSD | exim is |
| 20:23.12 | CunningPike | syzygyBSD: Yes, it does |
| 20:23.23 | proxyfrog1 | ..and i guess inbound too eventually...... |
| 20:23.33 | proxyfrog1 | kk...........sounds good.......thanks bro |
| 20:23.34 | nettie | Hi guys, I have a couple of Polycom phones connected to my asterisk server, when I issue a "sip show peers" the Status column show me that the phones have a latency of 130ms. I can reach my asterisk server with a roundtrip of 35ms, it's like 100ms more, is that normal? |
| 20:23.43 | proxyfrog1 | imma try it right now.....dam |
| 20:23.51 | *** part/#asterisk proxyfrog1 (n=bluewave@adsl-75-4-195-66.dsl.irvnca.sbcglobal.net) |
| 20:24.11 | CunningPike | nettie: Are you comparing ICMP with RTP? |
| 20:24.24 | Skaag | how do I get rid of sip objects? |
| 20:24.48 | Skaag | it looks like my sip phone connected once, and it "seems" connected since |
| 20:24.49 | nettie | CunningPike ahh those values are based on RTP? that's the reason it's so high then.. |
| 20:24.52 | Skaag | unregistration failed |
| 20:25.01 | nettie | CunningPike does it includes the JB? |
| 20:25.37 | CunningPike | nettie: Yup - it's an RTP 'ping' - no idea if the jitterbuffer is included - I rather doubt it |
| 20:25.48 | nettie | CunningPike well it's kinda strange |
| 20:26.01 | nettie | CunningPike if it's not included |
| 20:26.16 | nettie | CunningPike considering I get 50ms from the pap2 |
| 20:26.28 | nettie | polycom rtp pings at the double |
| 20:27.03 | *** join/#asterisk fearnor (n=alex@66.250.55.42) |
| 20:27.52 | ronaldl79 | Has anyone noticed any strange SIP behavior (registration timeouts/no connectivity to * via SIP) when connecting to 5 or more SIP servers simultaneously? This just started happening yesterday, for no apparent reason. |
| 20:28.34 | ronaldl79 | For * to be responsive via SIP internally/externally, I now have to register to 2-3 SIP servers for * to behave as usual .. strange. |
| 20:29.24 | *** join/#asterisk Mercestes (n=Merceste@216.54.143.2) |
| 20:29.27 | ronaldl79 | I've reproduced this in TRUNK and 1.2.9-10 |
| 20:31.34 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 20:31.43 | Dr-Linux | hi all |
| 20:31.58 | n9urk | does the Husic on hold now restart from 0 on each call? |
| 20:32.57 | n9urk | It used to not restart because it would stop once the callers are released from hold and then would start at that point once the next caller was placed on hold |
| 20:33.04 | n9urk | any idears? |
| 20:33.28 | file | n9urk: which one? mpg123 based or files? |
| 20:33.40 | n9urk | native moh |
| 20:33.52 | vader-- | any of you guys create an ivr type menu yet? |
| 20:33.55 | n9urk | file: is native moh what you mean by files? |
| 20:34.03 | file | mpg123 worked like you said, native did not afaik |
| 20:34.10 | n9urk | vader--: we had one going. |
| 20:34.11 | Skaag | vader--: i'm working on it, if I can get my softphone to actually register with my server |
| 20:34.35 | vader-- | im trying to create one to administer an auto attendant and it seems like they take up more space in the extensions file than they should |
| 20:34.52 | vader-- | it's kinda huge for this somewhat small menu im trying to build |
| 20:35.11 | n9urk | vader that is normal |
| 20:40.24 | Skaag | I have this message once in a while, how do I unregister this user by force: Registered SIP 'skaag' at 88.154.77.103 port 5060 expires 120 |
| 20:41.06 | nortex | Has anyone ever seen a way to take a call and connect it to a soundcard's lineout? |
| 20:41.26 | Qwell[] | nortex: chan_alsa/chan_oss |
| 20:42.27 | nortex | Thanks |
| 20:43.02 | Skaag | i'm doomed |
| 20:44.07 | Seba_soy | guys, have yate some console to see what is happening? |
| 20:44.29 | Seba_soy | bad window |
| 20:44.45 | Skaag | Realm. auth: No <-- how do I turn this on? |
| 20:45.14 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 20:48.12 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 20:51.02 | Dr-Linux | Qwell[]: :) |
| 20:51.52 | Dr-Linux | Qwell[]: i just talk to a lawyer friend |
| 20:51.55 | Dr-Linux | taked |
| 20:52.00 | Dr-Linux | talked* |
| 20:52.35 | Skaag | he's going to sue me for asking too many questions? |
| 20:52.35 | Assid | talked? |
| 20:52.44 | Skaag | spoke |
| 20:52.45 | Assid | spoke? |
| 20:52.48 | Assid | hehe |
| 20:52.53 | Skaag | spock |
| 20:52.57 | Skaag | this is illogical |
| 20:53.01 | Assid | dammit.. you just had to hit enter 1 second before me didnt you |
| 20:53.10 | TripleFFFF | Skaag try YES |
| 20:53.29 | Dr-Linux | discussed |
| 20:53.37 | Skaag | TripleFFFF: i'm desperate, it registered once and now I can no longer register, and nothing i do fixes it |
| 20:54.12 | nortex | OT, But is there a way from linux to stream to a windows media server? |
| 20:54.26 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 20:55.10 | Skaag | asterisk is giving me a headache... i'm going home. |
| 20:55.45 | Dr-Linux | ? |
| 20:55.47 | *** join/#asterisk sevard (n=sev@adsl-71-129-115-244.dsl.irvnca.pacbell.net) |
| 20:55.59 | sevard | I have a question... another one :) |
| 20:56.18 | sevard | My telco wants ground start on my fxo lines so I changed it from fxsls to fxsgs and ztcfg -vv freaks out |
| 20:56.22 | sevard | Changing signalling on channel 65 from FXS Loopstart to FXS Groundstart |
| 20:56.22 | *** join/#asterisk deb_user (n=debian_u@albuquerque.agroinnovations.com) |
| 20:56.22 | sevard | ZT_CHANCONFIG failed on channel 65: Invalid argument (22) |
| 20:56.31 | TripleFFFF | http://pastebin.ca/112322 |
| 20:56.39 | TripleFFFF | trunk btp doesn compile it seems |
| 20:56.40 | sevard | erm, fxsls to fxsgs |
| 20:56.52 | sevard | no i said that right, nm heh |
| 20:56.54 | deb_user | can anybody point me to a key that tells you what each file actually says in asterisk-sounds? |
| 20:56.56 | sevard | anyone know? |
| 20:56.57 | TripleFFFF | ~btp |
| 20:56.59 | jbot | i guess btp is .BTP is a creative approach to solving the problem of having one phone that rings you everywhere. Normally, if you want to have a single number that can reach you everywhere, you have to have it call all locations simultaneously, unless you want to manually forward your phone. BTP takes advantage of relatively common Bluetooth devices to allow ... |
| 20:57.18 | TripleFFFF | ~bot btp = broken |
| 20:58.01 | *** part/#asterisk Skaag (n=hintza@212.199.180.157.static.012.net.il) |
| 20:58.54 | deb_user | anybody know where I can find a key to asterisk-sounds? |
| 20:59.01 | deb_user | instead of having to play all the sounds... |
| 20:59.16 | nortex | deb_user, There is a little bit of a list on voip-info.org |
| 20:59.25 | deb_user | ok... |
| 20:59.27 | deb_user | I'll check it out |
| 21:00.04 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 21:01.59 | sevard | Is there something else someone must enable to get Ground Start working with a TDM2400P? |
| 21:02.26 | sevard | I change my FXO channels over to fxsgs and ztcfg freaks out. |
| 21:02.33 | mog | you cant do gs on tdm400p or 2400p |
| 21:02.34 | budmang | collect2: ld returned 1 exit status |
| 21:02.34 | budmang | make: *** [asterisk] Error 1 |
| 21:02.34 | budmang | [root@bmf asterisk-1.2.10]# |
| 21:02.35 | sevard | fails on those channels. |
| 21:02.36 | mog | or on sangoma hw |
| 21:02.44 | mog | it only works for t1/e1 cards |
| 21:02.54 | sevard | mog: So it's a hardware issue then? On the TDM2400P you can only do kewlstart and loopstart? |
| 21:02.59 | mog | yes |
| 21:03.30 | sevard | Thank you for your help. |
| 21:04.01 | *** join/#asterisk trixter (n=trixter@65-165-167-217.du.volcano.net) |
| 21:04.53 | mog | no problems |
| 21:05.04 | mog | sorry it wont work for ya |
| 21:05.07 | sevard | buttrub! |
| 21:05.15 | trixter | Jim Van Meggelen (one of thge authors of asterisk: the future of telephony) is giving his speach at cluecon.com and you can hear him live at +1 712 432 7800 (skype, gizmo, and others can give you free calls to this number if you want to listen but dont have unlimited US calling) |
| 21:05.21 | trixter | he just started |
| 21:05.28 | mog | thanks for the spam trixter ^_^ |
| 21:05.37 | sevard | trixter: is that right now? |
| 21:05.39 | *** join/#asterisk tempest1 (n=asf@adsl-0-234-179.chs.bellsouth.net) |
| 21:05.45 | trixter | no problem, since it is asterisk related |
| 21:05.53 | trixter | yes he just started like 1 minute ago |
| 21:05.55 | Qwell[] | trixter: What is the topic? |
| 21:06.11 | trixter | something on asterisk I am not yet sure he is doing his introduction on who he is and stuff that way |
| 21:06.13 | sevard | that's awesome |
| 21:06.32 | deb_user | nortex: that's a pretty comprehensive list... |
| 21:06.33 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 21:06.42 | deb_user | is there anywhere to download additional .gsm files? |
| 21:06.52 | deb_user | for setting up a phone menu... |
| 21:07.10 | sevard | damn my phone for only having one bar :( |
| 21:07.20 | sevard | haha "woohoo" |
| 21:07.29 | trixter | he said they are working on a 2nd book as well |
| 21:07.42 | nortex | deb_user, Did you already download asterisk-sounds? |
| 21:07.46 | trixter | an asterisk cookbook |
| 21:07.52 | budmang | anyone know how to force asterisk to think zaptel is installed? |
| 21:07.59 | TripleFFFF | oh welljust bought another bluetooth earset |
| 21:08.02 | budmang | VPS thing im working on :-) |
| 21:08.11 | deb_user | nortex: yeah, installed it as a debian package |
| 21:08.13 | sevard | budmang: there's ztdummy |
| 21:08.39 | deb_user | but there's no: please dial the extension of the person you which to reach |
| 21:08.44 | budmang | sevard i know. |
| 21:08.52 | sevard | mog: I don't consider that spam, I call that related information. |
| 21:08.54 | budmang | I just need asterisk to make the zap modules. |
| 21:08.58 | budmang | with out them being there. |
| 21:09.23 | mog | i only consider it spam as same group of people come in and advertise for said conference etc repeatidly |
| 21:09.25 | deb_user | i suppose I could just record it myself |
| 21:09.32 | mog | without provocation |
| 21:09.39 | mog | but to each their own |
| 21:09.58 | nortex | deb_user, try enter-ext-of-person.gsm or go to digium and pay allison to record it :) |
| 21:10.37 | *** join/#asterisk gandhijee (n=gandhije@65.169.134.2) |
| 21:10.53 | nortex | deb_user, it is on this list. http://www.nathanpralle.com/software/ast_masterlist.html |
| 21:13.33 | deb_user | nortex: cools, thanks |
| 21:13.48 | deb_user | the debian package only has a few of the sounds...not nearly as comprehensive as the list just sent me |
| 21:14.37 | bcnl | is 1.4 ready(ish) for production yet? |
| 21:15.39 | *** part/#asterisk trixter (n=trixter@65-165-167-217.du.volcano.net) |
| 21:15.39 | *** part/#asterisk gandhijee (n=gandhije@65.169.134.2) |
| 21:15.39 | watchy | dude 1.2.10 is fucked |
| 21:15.39 | watchy | you dudes need to tell folks not to use it |
| 21:15.39 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 21:15.39 | hads | bcnl: 1.4 doesn't exist |
| 21:15.39 | bcnl | watchy: howso |
| 21:15.39 | bcnl | hads: well the code that shall soon be known as 1.4 |
| 21:15.39 | nortex | deb_user, I know almost nothing about debian, but could you not download the sounds package extract it to the folder? |
| 21:15.40 | watchy | it totally fucked up a install i had at a .fov contractor |
| 21:15.40 | watchy | .gov' |
| 21:15.43 | watchy | mega fucked transfer bug |
| 21:15.45 | nortex | Funny my 1.2.10 servers work fine :) |
| 21:16.13 | watchy | nortex look it up at digium its reported |
| 21:16.20 | budmang | does anyone know the flag to make asterisk force bulid the zaptel modules? |
| 21:16.22 | watchy | it totally racked my fucking brain all day |
| 21:16.33 | watchy | i downgrade and bam it works |
| 21:16.39 | Qwell[] | watchy: feel free to submit a patch |
| 21:16.48 | watchy | i cant im no coder |
| 21:16.48 | *** part/#asterisk SkramX (n=MarkS@70.86.176.56) |
| 21:16.56 | nortex | Got a bug number? |
| 21:16.57 | Qwell[] | feel free to hire a developer to submit a patch |
| 21:17.04 | watchy | asterisk rocks i just think 1.2.10 is bad |
| 21:17.14 | JunK-Y | this is docelm0 watchy, why the fuck would you use the newest version.. Have you not followed asterisk version wowes? You NEVER run the newest.. Im still running 1.2.7.1 patched.. |
| 21:17.15 | watchy | yea hold nortedx |
| 21:17.42 | bcnl | budmang: that's a different source tarball |
| 21:17.43 | watchy | junk: i replaced a server to fix crackling and put 1.2.10 without even thinking about it |
| 21:17.47 | bcnl | budmang: look on the ftp server |
| 21:17.52 | deb_user | nortex: yeah, maybe that would be the best strategy |
| 21:18.06 | bcnl | watchy: got a bug ID I can lookup? |
| 21:18.09 | hmmhesays | is there any conference besides app_conference that doesn't need a timer? |
| 21:18.21 | JunK-Y | not junk docelmo. :) If you know asterisk then you should know not to use the newest.. |
| 21:18.35 | CunningPike | hmmhesays: Why not provide a timer and use MeetMe()? |
| 21:18.40 | watchy | im loooing for the bug # now |
| 21:18.40 | JunK-Y | I mean hell there are all kinds of commits then reversals.. its SO not worth it.. |
| 21:18.55 | watchy | it aint like 1.2.10 is SVN |
| 21:19.29 | JunK-Y | close enough |
| 21:19.29 | hmmhesays | CunningPike: unfortunately that is not possible |
| 21:19.46 | CunningPike | hmmhesays: I'm interested as to why |
| 21:20.00 | TripleFFFF | coz box to far away to put a sound card |
| 21:20.02 | TripleFFFF | i guess |
| 21:20.08 | TripleFFFF | if a sound card can indeed provide it |
| 21:20.09 | watchy | http://bugs.digium.com/view.php?id=7438 |
| 21:20.15 | watchy | theres the bug |
| 21:20.17 | budmang | bcnl: which one? |
| 21:20.27 | watchy | 0007438: A transfer SIP call present one way audio problem until hold and resume the call |
| 21:21.14 | bcnl | budmang: ftp.digium.com |
| 21:21.22 | budmang | bcnl im in there. |
| 21:21.23 | hmmhesays | CunningPike: can't compile zaptel, kernel incompatibilities |
| 21:21.31 | CunningPike | hmmhesays: Ahh - I see |
| 21:22.04 | hmmhesays | can't do anything with the kernel cause the server is 1500 miles away, and a screw up is not an option |
| 21:22.49 | watchy | that bug caused me alot of hell today so if you got transfer issues with 1.2.10 you know why |
| 21:23.28 | budmang | There has to be a way |
| 21:23.36 | charles___ | hmmhesays: kernel incompatibilities ? |
| 21:23.37 | *** join/#asterisk Wazb^ (n=wazb@199.243.74.220) |
| 21:23.41 | Wazb^ | hi to all |
| 21:23.45 | charles___ | hmmhesays: what's the problem ? |
| 21:23.47 | watchy | what kernel? |
| 21:24.03 | *** join/#asterisk Bobcat_1966 (n=chatzill@cpe-069-132-139-254.carolina.res.rr.com) |
| 21:24.14 | hmmhesays | nothing that will be solved by discussing it in here |
| 21:24.33 | *** join/#asterisk websae (n=websae@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
| 21:24.34 | watchy | well lets hug then |
| 21:24.37 | Wazb^ | does anyone know about CONVERT command in asterisk ? |
| 21:24.55 | folder | what does ael stand for again? |
| 21:24.56 | Wazb^ | for codec conversion |
| 21:25.03 | watchy | hey man i spent all day fight that hold audio issue dont make me kill you |
| 21:25.07 | watchy | im a ninja today |
| 21:25.08 | *** join/#asterisk ffang (n=ffang@porcuswine.cbnco.com) |
| 21:25.10 | blaylock | anyone happen to know off hand what txgain value starts to bork up DTMF? |
| 21:25.13 | CunningPike | folder: Asterisk Extension Language |
| 21:25.14 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 21:25.20 | folder | CunningPike: ah, thanks |
| 21:25.23 | CunningPike | ~ael |
| 21:25.24 | jbot | methinks ael is Asterisk Extension Language - a dialplan language with 'c like' syntax? |
| 21:25.25 | blaylock | im currently using txgain=-3.0 |
| 21:25.36 | nortex | watchy, That is odd that we have not seen that, but I try to teach our users to use the pbxtransfer. |
| 21:25.50 | folder | CunningPike: thanks. I'm just going through my new AstLinux install and I notice an extensions.ael file there along with the expected extensions.conf |
| 21:25.54 | nortex | watchy, I will keep an eye out for it though. |
| 21:25.58 | budmang | does anyone know what I would change in the Makefile to force it to think zaptel is installed? |
| 21:26.06 | watchy | nortex: whats pbxtransfer transfering though *? |
| 21:26.16 | Wazb^ | does anyone know about CONVERT command in asterisk ? |
| 21:26.20 | hmmhesays | app_conference segfaults using ilbc |
| 21:26.21 | hmmhesays | lol |
| 21:26.36 | mog | yeah meetme does not... |
| 21:26.46 | hmmhesays | yeah |
| 21:26.50 | hmmhesays | meetme isn't an option |
| 21:26.51 | bcnl | budmang: download the zaptel sources, and install them |
| 21:26.53 | CunningPike | budmang: You might try the wiki to see if there is anything there about virtualization |
| 21:27.04 | ffang | Hi, should Asterisk restart itself after adding a SIP extension? |
| 21:27.08 | mog | hmmhesays, why no meetme |
| 21:27.17 | nortex | watchy, You mean what is pbxtransfer? Like dialing # then asterisk takes over the transfer. |
| 21:27.24 | CunningPike | ffang: Uh, no.......... - how did you add it? |
| 21:27.32 | watchy | nortex: oh is it better then using the phone? |
| 21:27.43 | *** join/#asterisk folder (n=carl0s@compsup.demon.co.uk) |
| 21:27.53 | *** join/#asterisk Ethon (i=arne@Oldman.steinkamm.com) |
| 21:28.00 | ffang | by add extensions in extensions.conf file |
| 21:28.19 | CunningPike | ffang: That shouldn't make asterisk restart by itself..... |
| 21:28.48 | nortex | watchy, I think it is, but the attended transfer is still mainly done here by pressing transfer on the polycom then transfer again after announcing the call. |
| 21:29.09 | ffang | well, to let the new extension take effect, I have to restart the Asterisk? |
| 21:30.04 | CunningPike | ffang: No - 'sip reload' in the CLI will work |
| 21:30.13 | watchy | nortex: yea thats how they use it |
| 21:30.19 | watchy | they like announcing |
| 21:30.33 | ffang | ok. thanks a lot |
| 21:30.48 | *** join/#asterisk ivanfm_ (n=ivanfm@201.52.129.236) |
| 21:30.54 | nortex | watchy there is away to do attended through asterisk, but I have not done it. |
| 21:31.14 | CunningPike | ffang: You may also want to amend your extensions.conf and then it's 'extensions reload' |
| 21:31.41 | watchy | what i got now works, so i wont bother replacing it |
| 21:32.39 | ffang | cool |
| 21:33.32 | *** join/#asterisk supjigatr (n=syslod@152.53.17.26) |
| 21:33.47 | CunningPike | ffang: The only time I restart asterisk is when I upgrade |
| 21:34.16 | smackus | has anyone run into this error when setting up their servers in redhat? PCI: Failed to allocate mem resource... |
| 21:36.40 | supjigatr | ANyone have trouble shooting tips for PRI that works on inbound but on outbound call from SIP gets hangup? |
| 21:37.24 | CunningPike | supjigatr: pri debug span n for starters |
| 21:39.50 | supjigatr | Is that for the D channel? |
| 21:40.05 | *** join/#asterisk bkw__ (n=brian@asterisk/friend-and-developer/bkw) |
| 21:40.05 | folder | that's strange. AstLinux seems to be running the digium g729 codec... but I haven't licensed/paid for it on this install. |
| 21:41.28 | supjigatr | Hmm. Invalid number format |
| 21:41.46 | CunningPike | supjigatr: Yes - it will provide you with the D-channel signalling |
| 21:42.13 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 21:42.34 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 21:43.11 | budmang | libtonezone.so.1.0 |
| 21:45.21 | *** join/#asterisk RoyK (n=roy@ti211310a080-10269.bb.online.no) |
| 21:45.55 | *** join/#asterisk arkonadev (n=chatzill@65.203.186.131) |
| 21:46.16 | arkonadev | hey does an AGI script on the server have to be perl or could i use a bash script |
| 21:46.50 | crochat | Hello ! |
| 21:46.55 | charles___ | arkonadev: any language, bash, csh, php, c |
| 21:47.33 | crochat | Anybody knows a website from where I can send a free fax in order to test my Asterisk server ? |
| 21:47.58 | eKo1 | crochat: yeah right |
| 21:48.23 | fearnor | www.tpc.int |
| 21:48.24 | charles___ | crochat: ask some marketing company to put your name on the list |
| 21:48.36 | fearnor | yes, as a matter of fact and tpc.int has been around for 10 years. |
| 21:48.50 | charles___ | crochat: or call someone and ask them for fax quote |
| 21:48.52 | Assid | oh crap.. that site stil exists? |
| 21:49.12 | fearnor | or call one of the fax-back services |
| 21:49.28 | crochat | charles___: mmm, no thanks, I'm not testing SpamAssassin over Asterisk ;-) |
| 21:49.39 | supjigatr | Anyone take a look at the PRI trace and tell me why it won't connect? http://pastebin.ca/112387 |
| 21:50.59 | folder | hmm. that tcp.int looks like a remarkable service. I wonder how they feel about people using them as a full on outbound fax trunk.. |
| 21:52.01 | *** join/#asterisk MGSsancho (n=user@adsl-67-122-138-5.dsl.irvnca.pacbell.net) |
| 21:52.31 | *** join/#asterisk ayano (n=ayano_@72.24.212.226) |
| 21:53.46 | *** part/#asterisk Astinus- (n=a@213.167.111.138) |
| 21:53.47 | crochat | fearnor: tpc.int does not cover my inbound phone number :-( |
| 21:53.53 | fearnor | oh well. |
| 21:54.00 | fearnor | demand a refund. |
| 21:54.24 | crochat | I have a sipphone.com phone number |
| 21:55.35 | budmang | http://phpfi.com/136790 |
| 21:55.38 | budmang | im almost there |
| 21:57.53 | *** join/#asterisk DrkShdw (n=DrkShdw@unaffiliated/drkshdw) |
| 21:58.20 | folder | hmm. Asterisk for Dummies. |
| 21:58.45 | folder | Is that being written by anybody well-known within the community? |
| 22:00.10 | folder | http://images.amazon.com/images/P/0470098546.01._SS500_SCLZZZZZZZ_V62406428_.jpg |
| 22:00.33 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 22:03.55 | [hC] | Welcome to #asterisk, here is your book. |
| 22:03.57 | [hC] | :) |
| 22:04.35 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 22:04.53 | budmang | http://phpfi.com/136790 - anyone have any ideas? |
| 22:05.29 | *** join/#asterisk gursikh (n=guriskh1@dsl254-123-245.nyc1.dsl.speakeasy.net) |
| 22:09.36 | *** join/#asterisk viler (i=1000@200.114.70.228) |
| 22:10.03 | *** part/#asterisk viler (i=1000@200.114.70.228) |
| 22:12.39 | *** join/#asterisk mivck (i=1000@200.114.70.228) |
| 22:14.32 | *** join/#asterisk gmfm (i=gmfm@rtr.enterprisemtg.net) |
| 22:16.27 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-91-202.cybersurf.com) |
| 22:17.05 | *** join/#asterisk mtaht4 (n=m@dc-ns.rockliffe.com) |
| 22:20.22 | *** part/#asterisk ayano (n=ayano_@72.24.212.226) |
| 22:22.58 | *** join/#asterisk tempest1 (n=asf@adsl-0-234-179.chs.bellsouth.net) |
| 22:24.04 | *** part/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
| 22:25.09 | *** join/#asterisk bkw_ (n=brian@asterisk/friend-and-developer/bkw) |
| 22:27.47 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 22:31.35 | folder | Coolio. Just ordered some new reading material for me. Asterisk TFOT, and Unix Power Tools. I already have chapter 4 of TFOT printed out, but I keep falling asleep reading it and waking up with peices of paper all over the place. |
| 22:32.08 | folder | (that's not to say it's boring. I just get sleepy when reading in bed) |
| 22:32.34 | folder | budmang: Is that a virtual server thingy then? |
| 22:32.46 | budmang | yeap |
| 22:32.48 | budmang | 2 day mission |
| 22:32.49 | folder | cool |
| 22:32.49 | budmang | jesus |
| 22:32.53 | folder | lol :D |
| 22:33.02 | budmang | now i can use my own conference room :-) |
| 22:35.32 | SpaceBass | arrruuggg this phone won't update its firmware |
| 22:36.36 | *** part/#asterisk skac (n=skac@kakistocracy.co.uk) |
| 22:37.02 | CunningPike | budmang: Now for your next task - write it up for the wiki |
| 22:38.45 | budmang | practically done |
| 22:38.46 | budmang | already |
| 22:38.54 | budmang | but i want everyone to have that much fun :-) |
| 22:39.01 | CunningPike | budmang: lol |
| 22:39.09 | budmang | thanks for your help tho CunningPike |
| 22:39.18 | CunningPike | budmang: de nada |
| 22:43.11 | anonymouz666 | I would say "por nada" |
| 22:47.48 | smackus | asked earlier and did not get an answer (or did not see an answer) what do I do differently to configure a sangoma t1 card instead of a digium. I have replaced my digium card with a sangoma card. |
| 22:51.28 | l-fy | smackus > you did an wise thing |
| 22:51.40 | l-fy | now you have to download the wanpipe drivers from ftp.sangoma.com |
| 22:52.31 | smackus | ok |
| 22:52.34 | *** join/#asterisk burnproof (n=jsharryp@210.213.198.37) |
| 22:52.41 | smackus | so do i move away from zaptel then? |
| 22:52.47 | l-fy | no |
| 22:52.53 | l-fy | wanpipe will patch your zaptel drivers |
| 22:53.11 | smackus | gotcha, so i just download and install the driver... then the rest is as usual? |
| 22:53.29 | l-fy | the only way to use native wanpipe driver without zaptel is to use yate insted of * |
| 22:53.32 | burnproof | smackus: just download the wanpipe-3.4-beta drivers then prepare your zaptel source also |
| 22:53.33 | l-fy | yes |
| 22:53.39 | l-fy | and use the TDMV interface |
| 22:54.04 | *** join/#asterisk ManxPower (n=ewieling@71-8-61-45.dhcp.leds.al.charter.com) |
| 22:54.11 | smackus | ok... i will download and take a look.. may come back to you |
| 22:54.49 | ManxPower | [15 censored bad words] Charter ISP [12 censored bad words] |
| 22:57.26 | *** join/#asterisk zeppelin_ (n=zeppelin@201.66.133.218) |
| 22:59.23 | smackus | not seeing wanpipe 3.4... only 2.3, 4, and 7 |
| 22:59.35 | Qwell[] | ManxPower: 39 bad words in a row? |
| 22:59.41 | Qwell[] | 29 too |
| 23:00.18 | *** join/#asterisk LoneShadow (n=duh@59.92.130.129) |
| 23:00.22 | burnproof | smackus: 2.3.4 rather sorry |
| 23:00.35 | smackus | ok, thanks |
| 23:00.37 | burnproof | smackus: if forget really the version but get the recent beta drivers |
| 23:01.01 | smackus | that would be beta7-2.3.4 |
| 23:01.15 | burnproof | smackus: ok |
| 23:02.00 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 23:02.37 | burnproof | smackus: do you have ec on your sangoma card? |
| 23:04.04 | smackus | do you have ec on your sangoma card? |
| 23:04.13 | smackus | sorry... wrong. |
| 23:04.20 | smackus | dunno why i did that |
| 23:04.23 | smackus | yes, i do have ec |
| 23:04.33 | smackus | i got the a104d |
| 23:05.52 | Dr-Linux | Qwell[]: i decided to not give them now. |
| 23:06.55 | burnproof | guys, anyone here has experience on MFC r2 chan_unicall driver by steve? is it safe to use in production? |
| 23:07.36 | burnproof | i'm eyeing on a project integrating MERIDIAN 1 to asterisk any idea? |
| 23:08.59 | *** join/#asterisk slima (i=slima@syriusz.slnet.uu3.net) |
| 23:09.02 | carrar | Send me details and I'll write a PO for you |
| 23:11.03 | burnproof | come again? |
| 23:11.18 | burnproof | carrar: are you refering with me? |
| 23:12.00 | carrar | How much you paying? |
| 23:15.10 | *** join/#asterisk tempest1 (n=asf@adsl-0-234-179.chs.bellsouth.net) |
| 23:15.20 | burnproof | carrar: it depends on how much work we will pass you |
| 23:15.43 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 23:22.03 | sevard | What is AVT? |
| 23:23.40 | Qwell[] | ~avt |
| 23:25.09 | sevard | ~anus |
| 23:25.12 | jbot | I still have yet to figure out where the vaseline on my anus comes from... |
| 23:25.21 | sevard | Qwell[]: do you have two CDs? |
| 23:25.22 | *** join/#asterisk ManxPower (n=ewieling@71-8-11-111.dhcp.leds.al.charter.com) |
| 23:25.23 | sevard | erm |
| 23:25.26 | sevard | do you want two CDs? |
| 23:26.23 | sevard | Qwell[]: do you want two CDs? |
| 23:33.02 | Qwell[] | Why would I want two CDs? |
| 23:33.04 | *** join/#asterisk brif8 (n=Administ@ns1.ttienterprises.org) |
| 23:34.33 | *** part/#asterisk brif8 (n=Administ@ns1.ttienterprises.org) |
| 23:34.41 | *** join/#asterisk brif8 (n=Administ@ns1.ttienterprises.org) |
| 23:37.15 | *** join/#asterisk JohnJacob (n=JohnJaco@pool-71-246-132-81.aubnin.fios.verizon.net) |
| 23:37.21 | Qwell[] | I guess I should ask which CDs first, eh? |
| 23:38.05 | sevard | DEEZ NUTS BITCH~! |
| 23:38.07 | fuser | 197.x.x.x <-> *10.2.1.2* <-> 10.2.1.1 <-> 12.x.x.x <-> 12.y.y.y <-> 10.1.1.1 <-> *10.1.1.2* <-> 197.y.y.y |
| 23:38.13 | fuser | this is a net i built today on at&t |
| 23:38.28 | fuser | the last endpoint is the first endpoints default gateway |
| 23:38.31 | fuser | its a vpn tunnel |
| 23:38.32 | fuser | ;D |
| 23:38.34 | sevard | J0 BILT ATNT |
| 23:39.07 | *** part/#asterisk angom_w (n=angom@red-corp-200.79.158.218.telnor.net) |
| 23:45.55 | *** join/#asterisk trbldwine (i=troubled@71.194.161.170) |
| 23:46.33 | fuser | whats the name of that webite where you can draw networks? |
| 23:47.07 | *** join/#asterisk GREEKMAN2 (i=greekman@c-24-34-146-108.hsd1.ma.comcast.net) |
| 23:49.56 | *** part/#asterisk GREEKMAN2 (i=greekman@c-24-34-146-108.hsd1.ma.comcast.net) |
| 23:49.59 | *** join/#asterisk GREEKMAN2 (i=greekman@c-24-34-146-108.hsd1.ma.comcast.net) |
