| 00:01.34 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-242-4.buckeyecom.net) |
| 00:02.58 | *** join/#asterisk coppice (n=chatzill@158.198.17.210.dyn.pacific.net.hk) |
| 00:09.29 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-62-148.cybersurf.com) |
| 00:10.12 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 00:10.40 | *** join/#asterisk MajestiK (n=MajestiK@S0106000024c058cc.ed.shawcable.net) |
| 00:13.26 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:13.30 | Ariel_ | hello everyone |
| 00:13.31 | *** join/#asterisk gbdrbob (n=drbob@alltalk.demon.co.uk) |
| 00:13.45 | remibreval | Hello Ariel_ |
| 00:14.56 | gbdrbob | Hmm the zaptel makefile doesn't play nice with debian... every time I recomplie I find I've forgotten the changes I had to make last time. |
| 00:16.02 | gbdrbob | silly assumptions like /usr/src/linux will link to the current kernel source. |
| 00:16.10 | gbdrbob | There, I feel better now |
| 00:16.12 | *** join/#asterisk Ciber1 (n=Ciber@216-211-204-48.firstgate.net) |
| 00:16.41 | drumkilla | maybe everyone should just have a 'wheretheheckarethekernelsources' executable available |
| 00:17.21 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 00:18.26 | remibreval | Ok, I don't have any more idea tonight, it's quite late in france. I should stop. 12 hour of asterisk per day is quite much for such a problem |
| 00:18.31 | remibreval | Thank you everyone |
| 00:18.37 | remibreval | I hope I'll find something |
| 00:18.40 | Ariel_ | I have always wonder what there are so many different styles of where your files belong. |
| 00:18.53 | Ciber1 | can changing the txgain fry your car? |
| 00:19.02 | Ariel_ | your car? |
| 00:19.06 | Ciber1 | err card |
| 00:19.13 | remibreval | If someonee has an idea... http://pastebin.ca/28301 |
| 00:19.20 | Ariel_ | well can sound frie your speakers? |
| 00:19.54 | Ciber1 | cause someone told me to try changing it since i was dialing numbers and kept getting recorded messages from the phone company even though asterisk was dialing the right numbers |
| 00:20.03 | *** part/#asterisk Morex (n=blah@host81-157-127-252.range81-157.btcentralplus.com) |
| 00:20.29 | Ciber1 | so i raised the txgain to 4.5 and i get a dial tone and all but when i dial nothing happens not even the annoying phone company msg |
| 00:20.38 | Ariel_ | remibreval, there using a codec you don't have |
| 00:20.42 | delmar | is the format for tos tos=0x18 or tos=18 or either/or ? |
| 00:20.45 | QbY | I just followed the instructions here http://www.asterisk.org/download to upgrade to the latest release from cvs. i performed all the instructions, yet, my version is still reads Asterisk CVS-v1-0-07/22/05-17:29:07 -- ? |
| 00:20.46 | Ciber1 | lowered it back to 0.0 where it was originally and still nothing |
| 00:21.17 | Ariel_ | Ciber1, that would not help much |
| 00:21.29 | Ariel_ | Ciber1, have you added any ww to the dial string? |
| 00:21.36 | Ciber1 | ww? |
| 00:21.42 | Ariel_ | yes there waits |
| 00:21.47 | Ciber1 | nope |
| 00:22.05 | Ariel_ | dial(zap/1/ww${EXTEN}) |
| 00:22.06 | remibreval | Ariel, in the configuration of iax (web site of terminaison provider) they have same codec like me |
| 00:22.14 | remibreval | At least it seems they got it |
| 00:22.17 | remibreval | I'm using x-lite |
| 00:22.43 | Ciber1 | well ariel right now i'm more worried about the fact that dialing does nothing since i changed toe txgain |
| 00:22.50 | Ciber1 | before it would at least sort of work |
| 00:23.11 | Ciber1 | err the |
| 00:23.14 | Ariel_ | Ciber1, check the card out or stop the service then restart the service. |
| 00:23.16 | remibreval | Ariel_ is there a way to now they use a codec I don't have in the SIP or IAX debug messages ? |
| 00:23.31 | Ciber1 | check the card out? |
| 00:23.33 | Ariel_ | use iax debug |
| 00:23.37 | Ciber1 | i rebooted the server twice |
| 00:23.48 | Ariel_ | why do you need to reboot a linux box for |
| 00:23.57 | Ciber1 | i just did it :P |
| 00:24.11 | remibreval | Ariel_ I know, but I don't know how to find a message which says I don't have the codec. The IAX and SIP debug are there http://pastebin.ca/28301 |
| 00:24.18 | Ciber1 | either i changed the settings back to what they were and now i can't dial at all heh |
| 00:24.29 | gbdrbob | thing is it does it the right way (according to the FHS afaiks) further down the makefile looking in /lib/modules/'uname -r'/build |
| 00:24.34 | Ariel_ | Ciber1, what is the error your getting |
| 00:24.47 | delmar | I see on the wiki it says tos=0x?? for example.. and inthe sample conf files i see usage like tos=18 (without the 0x) .. what is the correct format please anyone? |
| 00:25.26 | Ariel_ | remibreval, Answering with preferred capability 0x4 (ulaw) |
| 00:25.26 | Ariel_ | Answering with non-codec capability 0x1 (telephone-event) |
| 00:25.29 | remibreval | Ariel_ I change to simplify my config with FuriousGeorge. It took 2 hours... but now we are stick |
| 00:25.49 | remibreval | My provider use ulaw |
| 00:25.57 | *** join/#asterisk ken___ (n=ken@host-66-59-246-138.lcinet.net) |
| 00:25.58 | remibreval | I'm calling from a regular phone via my provider |
| 00:26.08 | Ciber1 | here's the last thing i dialed http://pastebin.ca/28314 |
| 00:26.27 | ken___ | hey, i have a problem with automon, i'm at a loss as to how to get it to work correctly |
| 00:26.35 | ken___ | also, i cannot find any documentation on the matter |
| 00:26.36 | Ariel_ | remibreval, just put allow=ulaw disallow=all in all your setup and start from there |
| 00:26.59 | ken___ | does anyone know how to get automon to work correctly ? |
| 00:27.10 | remibreval | Ariel_, My config is done like that right now |
| 00:27.19 | Ariel_ | no it's not |
| 00:27.28 | remibreval | Hum let me check |
| 00:27.47 | Ariel_ | in the iax2 setup for the provider you have almost all codecs there |
| 00:27.52 | remibreval | OK I change it |
| 00:29.26 | Ariel_ | Ciber1, your using amp or aah |
| 00:29.36 | Ciber1 | aah |
| 00:29.41 | remibreval | Ariel_ you know that I did try this before... but IT WORKS !!!!!!!!!!!!!! |
| 00:29.45 | remibreval | Really !!! |
| 00:29.54 | Ciber1 | now it's not answering the phone either grrr |
| 00:29.57 | remibreval | Crazy !! 2 weeks for this !!! |
| 00:30.42 | remibreval | Completely crazy !! I swear we tested it, but the config was A@H before. Since we simplified and do what you say, it's OK !!!!! |
| 00:30.58 | Ariel_ | remibreval, great |
| 00:31.01 | remibreval | Yihahahahahaha !!!! |
| 00:31.07 | Ciber1 | lol ariel ignore that |
| 00:31.11 | Ariel_ | remibreval, no you can sleep |
| 00:31.24 | Ciber1 | seems the damn phone line got unplugged |
| 00:31.33 | Ariel_ | Ciber1, k |
| 00:31.36 | remibreval | But for now, I don't really understand why it doesn't work with other codec, as I have it and in voipprovider ulaw is 1st codec... |
| 00:31.44 | Ciber1 | i'm still having the issue of the numbers not being dialed correctly |
| 00:31.47 | Ciber1 | for example |
| 00:32.09 | Ariel_ | in your setup for the trunk did you put the ww in? |
| 00:32.11 | Ciber1 | i dial 916465094835 from the phone not plugged into the box and it works fine |
| 00:32.27 | Ciber1 | no i havent |
| 00:32.30 | remibreval | Ariel_ I owe you one ! |
| 00:32.34 | Ciber1 | how do i do that? |
| 00:32.46 | Ariel_ | in the zap identifier put g0/ww |
| 00:33.01 | Ariel_ | remibreval, k I take cash via pay pal |
| 00:33.28 | remibreval | lol ;) |
| 00:33.52 | Ariel_ | Ciber1, you dial 9 in your dial string? are you dialing out via a pbx? |
| 00:33.53 | remibreval | I just send cash with sip protocol :) |
| 00:33.59 | Druken | evening everyone... |
| 00:34.06 | Ariel_ | Druken, hello |
| 00:34.07 | Ciber1 | verizon makes me dial 9 on the line |
| 00:34.13 | Ariel_ | yak |
| 00:34.17 | Druken | do we have any providers from regina in the house? |
| 00:34.45 | Ciber1 | ariel the ww thing worked |
| 00:34.48 | Ciber1 | i love you man :P |
| 00:35.14 | ken___ | does anyone know anything about getting the automon (one touch recording) to work ? |
| 00:36.36 | Ariel_ | Ciber1, np |
| 00:38.10 | Druken | no takers? |
| 00:38.22 | Druken | no service in regina sask. canada? :) |
| 00:38.28 | Ariel_ | Druken, never heard of them |
| 00:38.41 | *** part/#asterisk gbdrbob (n=drbob@alltalk.demon.co.uk) |
| 00:39.35 | Ciber1 | ariel |
| 00:39.45 | Druken | Ariel_: that would be regina, sasketchewan, canada... <city, province, country> |
| 00:40.11 | Ciber1 | in the outbound routing thing i have 9|. right now |
| 00:40.25 | Ciber1 | how can i keep that but without that 9 |
| 00:40.31 | Ciber1 | dialing 9 is annoying :P |
| 00:40.58 | Druken | Ciber1: remove the 9, and add the 9 into the dial ?? |
| 00:41.09 | *** join/#asterisk ptiggerdine (n=ptiggerd@c210-49-98-194.rochd1.qld.optusnet.com.au) |
| 00:41.21 | Ciber1 | this is the thing for it to pick that route |
| 00:41.34 | Ciber1 | it's not the dialing 9 one |
| 00:41.38 | Ciber1 | err that makes no sense |
| 00:41.47 | Ciber1 | i need to get out of this office lol |
| 00:42.47 | *** join/#asterisk alexns (n=ibtek04@pa-bethlprk-cad2-grp2a-1-167.pittpa.adelphia.net) |
| 00:43.39 | alexns | anyone doing load balance / redundancy cluster ?? |
| 00:43.50 | remibreval | Ariel_, actually this was the pb : g729 :-) (just for the knowledge) |
| 00:45.29 | alexns | can anyone in here help with a cluster setup? |
| 00:45.34 | Ariel_ | remibreval, I knew that |
| 00:46.26 | remibreval | Ariel_, but I thought codec was chosen in function of the order ? And alaw was the first in voipprovider and in asterisk!!! |
| 00:46.40 | Ariel_ | no not in order |
| 00:46.52 | Ariel_ | first one it see |
| 00:46.57 | remibreval | Ariel_, arff so I hate A@H handbook !!! :-( |
| 00:46.59 | Ariel_ | from the provider not your system |
| 00:47.21 | remibreval | in the provider list, you can choose order but maybe it's bugged |
| 00:48.48 | Ariel_ | asterisk really does not allow ordered except in the newer 1.2 |
| 00:48.54 | Ariel_ | 1.0.9 does not |
| 00:50.25 | remibreval | Ariel_, ok, that make sense now |
| 00:51.53 | Druken | anyone from ontario here? |
| 00:52.48 | Druken | or... let me rephrase, anyone that can give me origination from lets say... guelph... here? :) |
| 00:54.00 | Beirdo | sorry, I'm in Toronto |
| 00:56.04 | Druken | bah! :) |
| 00:56.42 | *** join/#asterisk bsdfreak (n=alex@daemon.bsdgeek.com) |
| 00:56.44 | bsdfreak | heh |
| 00:56.50 | Beirdo | work damn you! |
| 00:58.21 | Druken | Beirdo: but i don't want to work.... i'm a lazy kid... :) |
| 00:58.30 | Beirdo | no, inn |
| 00:58.34 | Beirdo | it's buggered |
| 00:59.05 | Druken | wut's inn ? |
| 00:59.56 | Beirdo | the main news server used... |
| 01:00.25 | Beirdo | got it |
| 01:00.32 | bsdfreak | heh |
| 01:00.40 | Druken | pfft, news.... buy the sun... :) |
| 01:00.46 | Beirdo | blew away the overview file and rebuilt |
| 01:00.55 | Beirdo | usenet news, silly |
| 01:01.10 | Druken | :P |
| 01:05.36 | Beirdo | OK, cut over |
| 01:15.32 | *** join/#asterisk froguz (n=froguz@54-43-112.adsl.terra.cl) |
| 01:15.39 | froguz | hi |
| 01:16.43 | froguz | is there a way to do call pickup in v1.0.9 besides installing the BristuffPickUp function? |
| 01:17.17 | froguz | it seems there's no a "native" way to do so on this version |
| 01:17.46 | remibreval | More generally, my trouble was about exten => s,2,Macro(record-enable,${ARG2},IN) |
| 01:17.50 | remibreval | Now I know ;-) |
| 01:21.03 | froguz | plase, any help with call pickup?? |
| 01:22.38 | *** join/#asterisk genmud (n=genmud@ip68-98-82-206.ph.ph.cox.net) |
| 01:27.23 | *** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
| 01:29.22 | remibreval | frogus, I may, I had 2 weeks problems... resolved :) |
| 01:32.42 | froguz | remibreval, so it's not a piece of cake :( |
| 01:33.03 | froguz | and i need it for tomorrow ='( |
| 01:38.38 | tzanger | ยง |
| 01:41.21 | remibreval | someone can ring me to know if mu number is allowed to be call from other country than France ? |
| 01:41.43 | remibreval | My number is +33870441450 |
| 01:41.57 | remibreval | landline number |
| 01:43.00 | justinu | i'll call you |
| 01:43.02 | justinu | hang on |
| 01:43.56 | remibreval | Ok, It rings :-) |
| 01:44.01 | remibreval | Great !!! Thanks!!!!!!! |
| 01:44.04 | justinu | cool |
| 01:44.07 | justinu | any callerid? |
| 01:44.30 | remibreval | shitfuck, I didn't had time to read it... I don't know ! |
| 01:44.33 | remibreval | can you try again ? |
| 01:44.37 | justinu | yeah |
| 01:44.49 | remibreval | Anonymous |
| 01:45.14 | justinu | interesting, i'm sending it |
| 01:45.19 | *** join/#asterisk Inv_arp (i=junya@adsl-144-17-25.mia.bellsouth.net) |
| 01:45.23 | justinu | btw, that's from USA, via level3 |
| 01:45.32 | remibreval | when I call from france with my mobile I have the IC |
| 01:45.35 | remibreval | ID* |
| 01:45.45 | remibreval | Maybe it is routing problem between france and US |
| 01:46.08 | remibreval | I don't know about level3... |
| 01:46.17 | justinu | could be... i might ask them about it |
| 01:46.47 | sahafeez | not all phone companies exchange the right ss7 id info between countries |
| 01:46.50 | remibreval | For me, when I call via xlite on regular phone, I have ID on regular, but not on mobile |
| 01:47.00 | sahafeez | i have run it to in japan, etc.. |
| 01:47.15 | remibreval | Stupid ss7 ;-) |
| 01:47.36 | sahafeez | exten => _6060,1,NoOp,${CALLERIDNAME} ; will be empty. - FACILITY IE has not been received yet. |
| 01:47.39 | sahafeez | exten => _6060,2,Wait(1) ; wait for the CO to find the name in the database |
| 01:47.42 | sahafeez | exten => _6060,3,NoOp,${CALLERIDNAME} ; now it shows the name |
| 01:47.53 | Druken | isn't there a law or something restricting the use of ANI as CID ? |
| 01:47.53 | sahafeez | before you answer to give it sometime. sometimes it is the delay |
| 01:47.54 | justinu | sahafeez: you work in asia? |
| 01:47.58 | sahafeez | did |
| 01:48.07 | sahafeez | in sunny san diego now |
| 01:48.13 | justinu | you know of any voip providers in taiwan selling DIDs? |
| 01:48.25 | sahafeez | no, this was in the mid 90's |
| 01:48.27 | sahafeez | sorry. |
| 01:48.32 | justinu | ah |
| 01:48.42 | sahafeez | not really to much voip then. i can ask around to some friends |
| 01:49.01 | sahafeez | btw, i have yet to see the name in caller id - just the number |
| 01:49.15 | justinu | sahafeez: if you run across anything, drop me a /msg or something |
| 01:49.21 | sahafeez | will do. |
| 01:49.23 | remibreval | can't find Wiki about insecure=very. You gad it somewhere ? |
| 01:49.35 | justinu | i've searched, but i have a feeling the fact that I can't read chinese is limiting me |
| 01:50.39 | justinu | insecure=very means that asterisk won't challenge them for authentication on invite |
| 01:51.58 | *** join/#asterisk digime (n=drooth@ip68-111-235-172.sd.sd.cox.net) |
| 01:52.45 | remibreval | justinu, ok thanks. That what I was suspected |
| 01:53.21 | sjobeck99 | hi all. |
| 01:53.21 | sjobeck99 | anyone have any experience, good or bad, about redirecting t38/fax to their ITSP? |
| 01:56.50 | *** join/#asterisk Inv_arp (i=junya@adsl-144-17-25.mia.bellsouth.net) |
| 01:57.05 | remibreval | Froguz ? still there ? |
| 01:57.37 | remibreval | What about your pb for picking up a call ? |
| 01:59.50 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 02:00.22 | froguz | remibreval, yep |
| 02:00.32 | froguz | my pb? |
| 02:00.38 | *** join/#asterisk kn0x (n=root@adsl-68-77-37-8.dsl.emhril.ameritech.net) |
| 02:00.45 | kn0x | hello all |
| 02:00.51 | froguz | did you mean my pbx? |
| 02:00.56 | justinu | hey kn0x |
| 02:01.01 | kn0x | long day |
| 02:01.05 | justinu | no shit |
| 02:01.08 | *** join/#asterisk mape2k (i=mape2k@www.pennewiss.de) |
| 02:01.24 | mape2k | doog morning |
| 02:01.26 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-62-148.cybersurf.com) |
| 02:01.30 | mape2k | good morning |
| 02:01.35 | kn0x | heehee |
| 02:01.49 | froguz | i wanted to make "call pickup" to work, i know i have to do some changes in features.conf |
| 02:01.49 | mape2k | german or english here? |
| 02:02.04 | justinu | english |
| 02:02.06 | froguz | but i can find what exaclty do i have to change there |
| 02:02.09 | kn0x | english if you dont mind |
| 02:02.12 | *** join/#asterisk stbain (n=stuart@68-65-115-61.chvlva.adelphia.net) |
| 02:02.18 | mape2k | ^^ |
| 02:02.25 | remibreval | I mean your problem.. |
| 02:02.26 | mape2k | some problems with voicemail... |
| 02:02.32 | kn0x | shoot |
| 02:02.54 | remibreval | Is it SIP to SIP call ? Does it rings ? IAX ? SIP ?... |
| 02:03.08 | froguz | just SIP to SIP |
| 02:03.35 | mape2k | i want to start a script after recording a message to put informations about it to database and copy it to another server etc... |
| 02:03.50 | mape2k | how can i get the filename used for the recorded wavefile |
| 02:03.50 | mape2k | ? |
| 02:04.41 | remibreval | ok, both SIP devices on same lan ? They well register ? |
| 02:05.05 | froguz | aha |
| 02:05.34 | froguz | i don't know what should i put in my features.conf to make it work |
| 02:07.40 | remibreval | humm... why you want to put something in features.conf |
| 02:07.49 | remibreval | you don't really need it to basically answer a call |
| 02:08.13 | justinu | mape2k: does voicemail use AstDB? |
| 02:11.12 | froguz | remibreval, i'm not talking about just pickup an incoming call... there's a feature called "call pickup" wich allow you to pick up any incoming call in your pickup group just pressing *8 |
| 02:11.14 | *** join/#asterisk Corndawg_ (i=whoisit@c-66-176-66-83.hsd1.fl.comcast.net) |
| 02:12.12 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 02:12.41 | remibreval | froguz, sorry, I thought you got problems with basic pickup |
| 02:13.46 | *** join/#asterisk myke420247 (n=myke@69.29.2.130) |
| 02:14.12 | mape2k | justinu - imho not |
| 02:14.28 | mape2k | i've put the mailboxes in voicemail.conf |
| 02:14.29 | justinu | yeah, just a guess |
| 02:14.37 | *** part/#asterisk QbY (n=QbY@adsl-068-209-210-253.sip.cha.bellsouth.net) |
| 02:15.27 | froguz | it's ok |
| 02:16.23 | froguz | it seems i'm a little lost. it supose "call pickup" it's a very required function... please somebody tell me what do i have to do to make it work |
| 02:17.42 | orlok | Hmm. |
| 02:18.03 | orlok | Would a PCI FXS card require external power? |
| 02:18.12 | orlok | I'm just looking at http://www.digium.com/index.php?menu=product_detail&category=hardware&product=DEVKIT&tab=details |
| 02:20.52 | bweschke | orlok: yes. it requires a molex connection |
| 02:21.26 | sjobeck99 | orlok: especially a FXS, it has to provide power over the line to that device & dialtone |
| 02:27.12 | *** join/#asterisk implicit (n=implicit@ip70-181-114-97.oc.oc.cox.net) |
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| 02:43.42 | remibreval | Ok, 4 o'clock, sleeping time. Good configuration to everyone. See you later ! |
| 02:43.52 | justinu | night |
| 02:44.14 | tengulre | ? |
| 02:44.39 | tengulre | it's 10:43 now |
| 02:44.57 | mog_home | g45 |
| 02:45.03 | mog_home | err 845 |
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| 03:10.32 | orlok | Hmm. |
| 03:10.38 | orlok | Anybody here used Sangoma gear? |
| 03:11.05 | litage | orlok: many people probably have. you might want to ask your real question |
| 03:13.20 | *** join/#asterisk YoMama (n=rewt@pcp02689850pcs.roylok01.mi.comcast.net) |
| 03:14.05 | orlok | I've got a PCI ADSL card sitting in front of me that was apparently designed locally |
| 03:14.11 | YoMama | i'm feeling lazy...is there anyplace that's got precompiled kernels with source? |
| 03:14.15 | orlok | but it looks exactly th same as the one on the sangoma page |
| 03:14.41 | *** part/#asterisk litage (n=nick@203.220.55.70) |
| 03:14.57 | orlok | Anyway, i'm having problems finding the driver on their site |
| 03:15.25 | tengulre | Hmm |
| 03:15.28 | bsdfreak | negatory |
| 03:15.30 | tengulre | Anybody active? |
| 03:15.33 | bsdfreak | roger that |
| 03:15.52 | YoMama | who is Roger? |
| 03:16.03 | Qwell | YoMama |
| 03:17.04 | [TK]D-Fender | I just got a Sangoma A104d..... |
| 03:17.42 | YoMama | hello Qwell |
| 03:17.52 | orlok | this is an S518 |
| 03:19.23 | *** join/#asterisk MajestiK (n=MajestiK@S0106000024c058cc.ed.shawcable.net) |
| 03:19.53 | *** join/#asterisk BhaalWTF (n=bhaal@CPE-141-168-108-119.qld.bigpond.net.au) |
| 03:22.37 | Lostfrog | orlok: looks like you just use the wanpipe drivers from their site. |
| 03:23.01 | orlok | Lostfrog: yeah, i was just thinking that |
| 03:23.33 | orlok | Lostfrog: yeah, seems that way from reading through others experiences |
| 03:23.35 | orlok | Hmm. |
| 03:23.52 | orlok | i swear, the picture on their site is NOT a sangoma card! |
| 03:24.37 | mog_home | > |
| 03:26.06 | [TK]D-Fender | There is a slimline 2nd gen 1 port version, and a much taller 2-port version (filtered pass-through ). |
| 03:26.28 | orlok | [TK]D-Fender: yes |
| 03:26.41 | orlok | [TK]D-Fender: I have a card that looks EXACTLY like the slimline one here |
| 03:26.46 | [TK]D-Fender | Both labels as taht model. |
| 03:26.46 | orlok | same chips and layout |
| 03:26.51 | orlok | even the same PCB break point |
| 03:26.53 | [TK]D-Fender | Well there you have it. |
| 03:27.05 | orlok | except this one has another companies brand silkscreened on it |
| 03:27.13 | Lostfrog | Rebranded? |
| 03:27.40 | orlok | http://www.traverse.com.au/productview.do?product_id=21 |
| 03:27.49 | Qwell | they probably all buy them from China anyhow |
| 03:28.00 | orlok | Lostfrog: well, the company has their own sourceforge project for drivers for this card |
| 03:28.29 | orlok | Qwell: i doubt it, i was a beta tester on the prototypes |
| 03:28.45 | orlok | hah |
| 03:28.48 | orlok | it is the same chipset |
| 03:28.52 | orlok | verrrrry interesting |
| 03:30.34 | [TK]D-Fender | Actually the card IS visually different.... |
| 03:31.00 | [TK]D-Fender | For one the rebranded one is not compatible with both 5 & 3.3 volt PCI |
| 03:31.35 | [TK]D-Fender | Missing the back break on that link |
| 03:32.12 | Lostfrog | The LEDs seem to stick out on the sangoma card. |
| 03:32.13 | orlok | [TK]D-Fender: yup, and the capacitor on mine is blue, not red |
| 03:32.32 | orlok | [TK]D-Fender: the LED type actually changed about 8 months ago |
| 03:32.45 | [TK]D-Fender | Well, not just color, but size, and spec. VERY close though... |
| 03:32.55 | [TK]D-Fender | Probably a rip-off... |
| 03:33.21 | orlok | [TK]D-Fender: he thing is, i've run into driver issues with these cards |
| 03:33.29 | YoMama | how do the version numbers work with linux....there's 2.4.20-30.9 |
| 03:33.38 | YoMama | i mean..what the hell..who needs so much branching |
| 03:33.39 | orlok | [TK]D-Fender: and i've actually been invited to go on-site to prove it to one of their engineers |
| 03:33.45 | orlok | proved it, and he had a fix for me in a week |
| 03:33.46 | *** join/#asterisk BhaalWTF (n=bhaal@CPE-141-168-108-119.qld.bigpond.net.au) |
| 03:33.53 | [TK]D-Fender | The "clone" or Sangoma's? (I'd far sooner believe the former given Sangoma's support) |
| 03:34.19 | orlok | [TK]D-Fender: the clone. Traverse is a 5-10 minute drive from here |
| 03:34.36 | orlok | [TK]D-Fender: they have also been doing comms stuff and had drivers in the kernel for heir gear for many years |
| 03:35.37 | {zombie} | YoMama: the -30.9 will be the patch level added by your distribution |
| 03:35.47 | orlok | [TK]D-Fender: its very odd, it will be interesting to see if the drivers work on either |
| 03:35.49 | {zombie} | 2.4 is the major version, .20 is the minor |
| 03:35.58 | *** join/#asterisk santiago (n=santiago@208.195.215.124) |
| 03:36.29 | emdub | anyone here using inoc-dba with asterisk behind nat? |
| 03:36.38 | [TK]D-Fender | Actually... I never eally considered getting one of these before since I bought an ADSL modem, but now its good reason for secon thoughts..... |
| 03:37.14 | orlok | same PCI ID's |
| 03:37.23 | [TK]D-Fender | Space saving for sure, less splitting 100 ways, and no extra brick.... |
| 03:37.38 | orlok | [TK]D-Fender: sync status and speed in dmesg too :) |
| 03:37.43 | orlok | plus CRC errors, etc |
| 03:38.43 | orlok | [TK]D-Fender: http://oss.sgi.com/projects/netdev/archive/2005-03/txtDZzOuxrgug.txt - the Pulsar Globespan pci id's are exactly the same as these cards too |
| 03:39.18 | orlok | goddamn, i've been having intermittant driver issues with these cards for ages |
| 03:39.32 | orlok | they get crap throughput on some Via southbridge chipset revisions |
| 03:39.41 | orlok | another driver branch to try will be good |
| 03:40.20 | *** join/#asterisk junbug (i=junya@adsl-144-17-25.mia.bellsouth.net) |
| 03:41.09 | *** join/#asterisk buu (i=buu@ded430-deb-175-30.netsonic.net) |
| 03:41.09 | YoMama | wha'ts the number after the dash in linux kernel numbering? |
| 03:42.13 | {zombie} | provided by your distribution (ie redhat/debian/mandriva etc) as I said |
| 03:42.28 | {zombie} | which means they took 2.4.20 and have released their own modified versions |
| 03:42.40 | YoMama | ah |
| 03:43.02 | YoMama | so i'm running RH9...2.4.20-8 |
| 03:43.08 | YoMama | means redhat added that 8 to the end of it |
| 03:43.11 | {zombie} | right |
| 03:43.17 | YoMama | gotcha |
| 03:43.34 | {zombie} | first time they released it, it would have been -1 |
| 03:43.40 | {zombie} | then they patched some things and called it -2 |
| 03:43.44 | {zombie} | etc |
| 03:44.14 | justinu | welcome to linux and it's trillion flavors |
| 03:44.43 | *** join/#asterisk epablo (n=epablo@WLL-24-pppoe201.t-net.net.ve) |
| 03:45.06 | YoMama | 2nd question: are there precompiled rpms that'll update me so i don't haveta compile the damn kernel myself? |
| 03:45.15 | *** part/#asterisk zamsler (n=zamsler@c-67-184-232-149.hsd1.il.comcast.net) |
| 03:45.15 | *** join/#asterisk zamsler (n=zamsler@c-67-184-232-149.hsd1.il.comcast.net) |
| 03:45.20 | YoMama | i used to think it was fun hacking and compiling the kernel..now i don't |
| 03:45.34 | *** part/#asterisk zamsler (n=zamsler@c-67-184-232-149.hsd1.il.comcast.net) |
| 03:45.38 | *** join/#asterisk zamsler (n=zamsler@c-67-184-232-149.hsd1.il.comcast.net) |
| 03:45.53 | epablo | Hi to all the people in the house |
| 03:45.56 | {zombie} | of course |
| 03:46.05 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 03:46.17 | {zombie} | just look on your nearest friendly RedHat mirror |
| 03:46.40 | {zombie} | you should update to 2.4.31 (or 2.6.x) while you are at it.. 2.4.20 is ancient |
| 03:46.58 | YoMama | zombie...so i've been told |
| 03:47.27 | epablo | My E1 on a TE110P sudenlly started to give all my cdrs a billsec of zero. |
| 03:47.43 | buu | Hi, I've got a small syntax problem, I'm trying to use the 'realtime' asterisk configuration stored in a database, but when I dial an extension I get the following error: http://erxz.com/pb/223 |
| 03:47.47 | epablo | Anyone have an idea on why that could be happening |
| 03:47.49 | epablo | ? |
| 03:49.00 | epablo | Sorry haven't used that module :( |
| 03:49.29 | buu | Apparently a syntax error involving my equal signs? |
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| 03:51.52 | *** part/#asterisk epablo (n=epablo@WLL-24-pppoe201.t-net.net.ve) |
| 03:53.56 | stbain | so... anyone here have a Mitel SX 2000 that I can pick their brain about integrating it w/ asterisk? |
| 03:54.19 | YoMama | that sounds like fun |
| 03:55.23 | stbain | well... trying to figure out how to hang an asterisk box off of it to act as voicemail, but I want to have a plan in place just in case I need to configure it to be the auto attendant |
| 03:55.56 | stbain | would prefer the ability to transfer to extensions back on the Mitel without pumping DTMF back down the line |
| 03:56.10 | YoMama | hmm |
| 03:56.11 | stbain | but I'm not so sure there's a method for doing that out of band |
| 03:56.31 | YoMama | i've never dealt with mitel before...does it have any VoIP capability? |
| 03:56.39 | stbain | not the model I'm working with :( |
| 03:56.46 | YoMama | ew |
| 03:56.57 | stbain | I'll basically have a 4 port FXO card from Digium |
| 03:57.25 | stbain | but if I have to bridge calls, that means I can only have two callers connected via the auto-attendant at any given time |
| 03:57.31 | stbain | (not good) |
| 03:58.50 | YoMama | why not connect them using a T1 card |
| 03:59.11 | YoMama | give youreself 24 channels |
| 03:59.17 | stbain | $$$$$$$$$$$ |
| 03:59.21 | YoMama | ah |
| 03:59.30 | YoMama | how many extensions |
| 03:59.40 | YoMama | having 4 channels for auto attendant and voicemail might be pushing it |
| 03:59.48 | stbain | 10 handsets |
| 03:59.54 | YoMama | oh...u should be fine |
| 04:00.08 | stbain | yes... should be |
| 04:00.28 | stbain | I'm pouring through the manual for their existing voicemail solution (iVoice) |
| 04:01.24 | YoMama | if you're doing it analog like that..then sending tone down the line is the only way |
| 04:03.38 | marv | twisted: oh master of ldap, fix our svn authenication that I might not be confused with you. |
| 04:04.05 | stbain | hrmm... check this out... looks like Cisco Unity can integrate with it seamlessly: http://www.google.com/url?sa=t&ct=res&cd=5&url=http%3A//www.cisco.com/en/US/products/sw/voicesw/ps2237/products_configuration_guide_chapter09186a00800804f9.html&ei=yxd0Q6uuF6ueaMKcqNsO&sig2=pVv0Jd6O6_bHgnF8rgUQUg |
| 04:04.22 | manxpower | ~docs |
| 04:04.23 | jbot | rumour has it, docs is 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 |
| 04:04.24 | manxpower | ~mailinglist |
| 04:04.26 | jbot | from memory, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 04:06.24 | buu | So, asterisk |
| 04:06.31 | buu | I've got this syntax error |
| 04:08.27 | *** join/#asterisk schuylerdigium (n=Bosco@pcp03052091pcs.huntsv01.al.comcast.net) |
| 04:08.36 | YoMama | huh |
| 04:09.19 | YoMama | stbain..looks like u can follow this unity crap and get it pretty close |
| 04:09.57 | *** join/#asterisk Jzalae (n=sk@dsl-66-63-110-48.gwi.net) |
| 04:10.45 | stbain | yup... that's what I'm seeing |
| 04:11.30 | YoMama | unity blows |
| 04:12.35 | samueltc | hi |
| 04:12.45 | drumkilla | hI! |
| 04:12.58 | YoMama | argh |
| 04:13.06 | YoMama | i think my only option is to install my own freakin' kernel |
| 04:13.20 | samueltc | any idea of how I can set a 'failover/high availability' asterisk (using vserver)? |
| 04:13.25 | stbain | hrmmm.... or not: Mitel does not use tones to transfer - they use softkeys. |
| 04:13.39 | samueltc | (everything is software i.e. no TDM) |
| 04:14.03 | Math` | YoMama: hey.. what was the solution to remibreval's problem? |
| 04:14.06 | YoMama | stbain: the directions tho show how to use DTMF tones to transfer to voicemail |
| 04:14.16 | YoMama | Math: what problem was that? |
| 04:14.25 | Math` | iax hangup on incoming |
| 04:14.56 | YoMama | Math: i saw that in mantis |
| 04:15.07 | *** join/#asterisk jdiskywlkr (n=kvirc@ip68-0-83-251.tu.ok.cox.net) |
| 04:15.08 | Nivex | OT: Can I use the base unit of one of these ( http://tinyurl.com/5jctr ) to power a Cisco 7960? |
| 04:15.34 | Math` | ah so you made him upgrade and it fixed it? lol |
| 04:15.42 | YoMama | Math: i need to upgrade myself...remember how i couldn't get wcfxsusb to compile on my kernel? |
| 04:16.03 | YoMama | the mantis log says "upgrade to the latest 2.4.x kernel and if it still doesn't work, let me know" |
| 04:17.01 | YoMama | Nivex: u gotta make sure the power is gonna be enough |
| 04:17.04 | Math` | nah wasnt that... I scrolled up, was a codec thingy |
| 04:17.08 | YoMama | just compare the specs and you'll know |
| 04:17.18 | YoMama | oh...i haven't gotten that fancy schmancy with asterisk yet |
| 04:17.39 | YoMama | hell..it took me 20 minutes just to figure out how to turn up the volume |
| 04:17.41 | Nivex | YoMama: The power specs look right, but I just found this: http://www.voip-info.org/wiki-Cisco+POE ; indicating that the polarity might be reversed |
| 04:18.35 | YoMama | Nivex: that's messed up |
| 04:19.00 | YoMama | wait..so that means that Cisco POE switches can't power other devices except for their own phones |
| 04:19.28 | YoMama | wait...that might be right...'cause i remember the engineer telling me that those switches were specifically designed to power those phones |
| 04:19.36 | YoMama | Cisco...dirty tricks |
| 04:19.48 | stbain | YoMama: must have missed it, because I don't see what DTMF tones to send back down the line to transfer the call |
| 04:20.03 | YoMama | stbain: to teh voicemail? |
| 04:20.13 | sjobeck99 | you guys see how hard it is now to open a case on Cisco TAC site. it used to be about 4 clicks, now it is next to impossible to open a case. |
| 04:20.17 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
| 04:20.49 | sjobeck99 | any one out there familiar with native MOH, not mpg123, but the option in musiconhold.conf called [native-random] ? |
| 04:21.00 | Nivex | well crap. I got a free phone and no way to power it :( |
| 04:21.14 | sjobeck99 | nivex: esternal adaptor ? |
| 04:21.26 | Nivex | sjobeck99: will have to order |
| 04:22.04 | sjobeck99 | nivex: which aint great, i know, but ....... what model (sorry i missed first past) |
| 04:23.02 | YoMama | Nivex: just buy the damn power brick for it |
| 04:23.12 | stbain | YoMama: in other words, my AA dialplan says "if you know your party's extension, you may dial it at any time... ". When they dial that extension, I'll need to hookflash/dtmf/dance-a-jig/something to tell the PBX to try that extension. |
| 04:23.37 | YoMama | stbain: wait...your AA is asterisk or the Mitel? |
| 04:25.33 | stbain | I have to have a plan for both cases |
| 04:26.00 | stbain | won't know until the (extremely unresponsive) former phone-guy/vendor gives me a brain dump on how the system is set up |
| 04:26.16 | YoMama | stbain: and you've only got four ports to work with? |
| 04:26.33 | stbain | judging from the way their iVoice (old/soon-to-be-gone) voicemail system is set up, I think asterisk will be the AA |
| 04:26.37 | stbain | four ports to work with |
| 04:26.57 | YoMama | ummm.... |
| 04:27.41 | YoMama | i'm just thinkin' this out...but for AA...you're talking about answering the phone for the company right? |
| 04:27.49 | YoMama | they dial in and u want asterisk to answer the phone |
| 04:27.52 | Nivex | YoMama: I was going to, but the guy I was going to buy it from didn't show up at the LUG meeting :( |
| 04:28.13 | YoMama | Nivex: it'll cost ua bout the same as a POE adapter |
| 04:28.19 | YoMama | or u make that crazy crossover cable in that diagram |
| 04:28.30 | *** join/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
| 04:29.21 | Nivex | YoMama: I just remembered I have ends. Stripping wire now :) |
| 04:29.25 | stbain | here's the psuedo-code for the dial plan: Incoming Call->Ring "the girls" and if one them picks up, great... if not->Goto AA->Ask for extension->Tell PBX to transfer the person to that extension->If busy/noanswer, PBX will transfer back |
| 04:29.38 | stbain | (to voicemail) |
| 04:30.02 | *** join/#asterisk marc324 (n=marc3234@206-248-133-19.dsl.teksavvy.com) |
| 04:30.17 | YoMama | Nivex: or get this thing: http://cgi.ebay.com/Cisco-IP-Phone-POE-Power-Supply-Cube-for-7960-29-95_W0QQitemZ5826462809QQcategoryZ11909QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
| 04:30.51 | YoMama | there's a few for bid on ebay too..power supplies |
| 04:31.23 | YoMama | stbain: i think you're gonna have issues with ports |
| 04:31.31 | Nivex | YoMama: but that won't have me running tonight ;) Thanks for the link tho |
| 04:32.43 | stbain | YoMama: I agree |
| 04:32.57 | YoMama | stbain: i might be talking abotu of my ass...but if u need it to go from PBX -> VM and AA -> PBX....that means both devices have to act as both FXO and FXS |
| 04:33.06 | YoMama | and umm..u only got four ports |
| 04:34.27 | rajiv | how can i play a sound on a call after dial connects but before ringing starts? |
| 04:34.57 | YoMama | rajiv: hell..i just started playing with asterisk and eveni know that |
| 04:35.07 | YoMama | rajiv: like on an outbound call? |
| 04:35.21 | Nivex | w00t! it worked! |
| 04:35.28 | Nivex | wiki++ |
| 04:35.34 | stbain | YoMama: FXO should handle it, but I'm still going to be eating up two ports unless I can hand the transfer back to the PBX.... still having found anything |
| 04:35.55 | YoMama | rajiv: exten => _9NXXXXXX,1,Playback(blahblah) |
| 04:36.15 | rajiv | YoMama: ya on outbound. i want something like 'this call will cost 10 cents per minute' |
| 04:36.21 | YoMama | exten => _9NXXXXXX,2,Dial(SIP/dontcallme,15,r) or whatever |
| 04:36.31 | rajiv | i guess i could check if the chan if available then play only based on that |
| 04:36.51 | rajiv | your examples will always play. i want it conditional if the call will actually go over that trunk or not |
| 04:37.14 | YoMama | rajiv..u didn't say that the first time did ya.. :) |
| 04:38.02 | rajiv | well i said after dial connects but before ringing. ... i'm still getting the terminology down |
| 04:38.31 | YoMama | well...on outbound..if it's connected..then it's up to the PSTN what happens..not you |
| 04:38.48 | YoMama | you want it to tell them it's gonna cost money if they dial long distance...right? |
| 04:39.09 | rajiv | btu if the call is going out over iax or sip surly there must be a way to konw... i see it in my debug logs ("call is making progress") |
| 04:39.14 | rajiv | right |
| 04:39.25 | *** join/#asterisk grimse (n=grimse@p5481EF5D.dip.t-dialin.net) |
| 04:39.27 | rajiv | i want to play something if the "call is making progress" |
| 04:39.52 | YoMama | k...i'm confused...u want it so when someone makes a LD call, it tells them it's gonna cost money and then rings the phone |
| 04:40.17 | rajiv | but only if the call is going to connect over that truk |
| 04:40.21 | rajiv | trunk |
| 04:40.32 | YoMama | so they could make an LD call that doesn't use the trunk? |
| 04:40.40 | rajiv | eg: exten => _011.,2,Dial(${TRUNKVOIPJET}/${EXTEN}) exten => _011.,3,Dial(${TRUNKRCN}/${EXTEN}) |
| 04:41.25 | rajiv | i want it to say the voipjet rate only if it is actually going to call over that |
| 04:41.40 | YoMama | ah |
| 04:41.55 | YoMama | now i get it |
| 04:42.03 | YoMama | so u have two ways to connect to the sam ephone # |
| 04:42.08 | rajiv | yes |
| 04:42.45 | YoMama | ok..and the voip trunk is first |
| 04:43.06 | rajiv | yes |
| 04:43.16 | YoMama | k...NOW i get it |
| 04:44.09 | YoMama | you need something that'll test to see if the trunk will accept the call and if it does,play the message and then actually dial the call |
| 04:44.31 | rajiv | ya. i wonder if that is chanavail() |
| 04:46.09 | YoMama | ChanIsAvail() you mean |
| 04:46.53 | rajiv | ya |
| 04:47.47 | YoMama | u know how many pipes u got right? |
| 04:47.59 | YoMama | u could create a channel group maybe? and use CheckGroup()? |
| 04:48.55 | rajiv | hmm |
| 04:56.30 | Alric | Anyone mess around with the Shared lines the Polycom supports? |
| 04:57.09 | *** part/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 05:12.30 | *** join/#asterisk newmember (n=newmembe@S010600a0c93dce87.cg.shawcable.net) |
| 05:13.38 | *** join/#asterisk zobia (n=laura_sh@218.6.242.212) |
| 05:14.01 | zobia | Hello everyone. anyone knows how to reload the asterisk from web? |
| 05:14.23 | delmar | Well, its official.. Tigernetcom IPPH203 aka TigernetCrap are exactly that.. utter crap. I have spent the last 2 days testing scenarios with their phones, vs grandstream to softphones which are as good as echo-less, but whenever there is an IPPH203 PA168X based phone involved, there is echo...I had a test phone of their new model IPPH202 about to be sent and told them to jam it. |
| 05:14.45 | delmar | Oh and their firmware updates.. wreck more phones than they update. I have phones here that are utterly useless and not firmware recoverable at all. |
| 05:15.57 | delmar | At the end of the day the brand isnt the problem.. its PA168X.. Atcom uses the same chipset from Centrality, and we all know what those phones are like.. utter shit. |
| 05:16.40 | delmar | Seems to me the rule is, if you want a phone that works nice, spend some money, because the cheap ones.. are exactly that, cheap and nasty. |
| 05:20.24 | *** join/#asterisk konfuzed (n=KonfuzeD@H129.C72.B0.tor.eicat.ca) |
| 05:20.50 | *** join/#asterisk Ragnarok_ (n=ramos@mx1.finetooth.com) |
| 05:21.29 | Ragnarok_ | anyone awake? :) |
| 05:22.06 | Math` | ya |
| 05:23.31 | mog_home | no |
| 05:23.42 | Ragnarok_ | I'm stuck at work tinkering with * |
| 05:23.54 | *** join/#asterisk MikeJ__ (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
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| 05:25.30 | Ragnarok_ | seem to be getting garbled audio on my sccp phones when connecting to zap channels... |
| 05:26.22 | YoMama | anyone know of a cheapass switch that does QoS |
| 05:26.42 | Ragnarok_ | IMO none of the cheapass QoS switches do it very well |
| 05:26.45 | YoMama | Ragnarok: what time is it where u are? |
| 05:26.50 | Ragnarok_ | 11:30 |
| 05:26.51 | konfuzed | ovislink.com |
| 05:26.52 | Ragnarok_ | pm |
| 05:27.02 | YoMama | ragnarok: i don't care...i'm using it for home |
| 05:27.09 | YoMama | ragnarok: ah...central time |
| 05:27.12 | Ragnarok_ | yeah.. |
| 05:27.23 | konfuzed | ovislink is commercial gear at residential price |
| 05:27.34 | Ragnarok_ | IMO a cheapass linux box with wondershaper works the best :) |
| 05:27.58 | konfuzed | more or less anyway |
| 05:28.39 | konfuzed | hmmm I looked around to see that I have 8 boxes laying around |
| 05:28.56 | konfuzed | that sould be able to make 4 to 6 of something |
| 05:29.02 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 05:30.12 | konfuzed | hhhmmm i should make one of them astlinux |
| 05:30.42 | *** join/#asterisk YoMama (n=rewt@pcp02689850pcs.roylok01.mi.comcast.net) |
| 05:30.45 | YoMama | blah |
| 05:32.06 | YoMama | i don't see any switches at ovislink that do QoS |
| 05:32.31 | *** join/#asterisk SplasPood (n=sp@paravolve.net) |
| 05:32.41 | Ragnarok_ | linksys makes one for ~$100.. they sell it as a "gamer" router or something |
| 05:32.49 | *** join/#asterisk SplasPood (n=sp@paravolve.net) |
| 05:32.53 | konfuzed | well im inclined to make one of them openbsd with asterisk and a 100p to be the gateway |
| 05:33.29 | konfuzed | but i think compatibility is likely to be a tad annoying |
| 05:34.42 | Ragnarok_ | well is there any reason my phone audio would get all garbled without changing the config? :P it's only from external zap channels and only the incoming audio... |
| 05:35.07 | konfuzed | hm openBSD firewall a-->lan b-->*-->100p PSTN |
| 05:35.09 | *** join/#asterisk clive- (n=pirch@ndn-165-140-22.telkomadsl.co.za) |
| 05:35.36 | YoMama | that's weird |
| 05:36.01 | konfuzed | i just priced one Im sure of it |
| 05:36.04 | Ragnarok_ | sounds like I'm talking all slow |
| 05:37.44 | konfuzed | http://www.ovislink.ca/LifeFSH16T+.html well thats not a gateway but a hell of a switch |
| 05:38.33 | YoMama | konfuzed: it does VLAN but that doesn't mean it prioritizes the packets |
| 05:39.58 | konfuzed | isnt ( Interfaces IEEE 802.3x Flow Control ) QoS |
| 05:41.07 | YoMama | i don't think so |
| 05:41.28 | konfuzed | so far google suggests it to be true |
| 05:42.02 | Ragnarok_ | http://www.linksys.com/servlet/Satellite?childpagename=US%2FLayout&packedargs=c%3DL_Product_C2%26cid%3D1115416832835&pagename=Linksys%2FCommon%2FVisitorWrapper |
| 05:42.09 | YoMama | there's no way to define which VLAN gets priority |
| 05:42.28 | YoMama | Ragnarok: yeah..i saw that...it's kinda scary though |
| 05:42.38 | YoMama | might as well try to find some used thing on ebay |
| 05:42.39 | Ragnarok_ | yeah.. supposedly it works well with VoIP stuff though |
| 05:42.57 | YoMama | Ragnarok: not enough ports for me |
| 05:43.06 | YoMama | oh well |
| 05:43.17 | Ragnarok_ | yeah home gear usually = 4 ports |
| 05:44.02 | YoMama | i'm gonna try zingotel just so i can make a shitload of LD calls for $15/month |
| 05:44.07 | YoMama | now that i got asterisk set up |
| 05:46.47 | konfuzed | well im not getting verification on 302.3x but isnt QoS pretty basic |
| 05:47.14 | YoMama | konfuzed: no |
| 05:47.18 | Ragnarok_ | No QoS is not basic |
| 05:47.27 | YoMama | konfuzed: it can be hellish |
| 05:47.46 | YoMama | konfuzed: there's about 100,000,000 ways to accomplish it and 99,999,999 ways to screw it up good |
| 05:47.51 | Ragnarok_ | yep |
| 05:48.13 | Ragnarok_ | and if you don't set the provisioning amounts just right it will just do nothing :) |
| 05:48.20 | YoMama | konfuzed: i should show u the latest config i put together for a Cisco switch...it'd make u wanna spit your guts out |
| 05:48.52 | YoMama | weights and packet priorities and bandwidth throttling and and and and..it goes on and on |
| 05:49.03 | Ragnarok_ | man cisco is so slimy... they ship their IP phones with broken firmware so you have to buy the support contract to download the update |
| 05:49.17 | konfuzed | i dont mezmorize ieee s tooo often |
| 05:49.22 | YoMama | Ragnarok: ha... |
| 05:49.38 | YoMama | i think i'm gonna buy one of these grandstream phones |
| 05:49.40 | Ragnarok_ | I'm serious, all the soft buttons read <P> </P> |
| 05:49.46 | YoMama | they're ugly as hell...but i hear they sound pretty good |
| 05:49.51 | Ragnarok_ | other than that the phones are awesome and chan_sccp works incredibly well |
| 05:50.09 | YoMama | Ragnarok: oh..the soft buttons pull from a URL |
| 05:50.14 | orlok | Ragnarok_: same with their routers |
| 05:50.17 | Ragnarok_ | only in the services menu |
| 05:50.20 | konfuzed | I gettin another grandstream ata486 tomorrow |
| 05:50.44 | orlok | Ragnarok_: we got shipped 50+ routers, hardware crypto engines, all incapable of doing stable ipsec with the firmware they shipped with |
| 05:50.53 | Ragnarok_ | yeah |
| 05:51.05 | Ragnarok_ | seems typical.. I try to stay away from cisco where I can |
| 05:51.24 | YoMama | Ragnarok: wait...i dunno how sccp works with asterisk...but those phones need a tftp server |
| 05:51.25 | konfuzed | the cisco model is ripped |
| 05:51.28 | Ragnarok_ | my company's still small so we can ge away with it |
| 05:51.33 | YoMama | it's the load on it that defines the soft buttons i believe |
| 05:51.36 | konfuzed | the cisco model is cripped |
| 05:51.54 | konfuzed | the cisco model is crypted |
| 05:52.03 | konfuzed | the cisco model is crypled |
| 05:52.03 | Ragnarok_ | Yo: no.. they're not really soft buttons in "Phone" mode |
| 05:52.06 | konfuzed | ;^) |
| 05:52.23 | YoMama | Ragnarok: you're talking about the buttons right below the screen..right? |
| 05:52.31 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 05:52.43 | konfuzed | bah |
| 05:52.46 | Ragnarok_ | yeah.. chan_sccp doesn't set them nor do the XML config files... I've dug through the source for both :) |
| 05:52.54 | konfuzed | I only need QoS from OpenBSD any way |
| 05:52.59 | konfuzed | thats the clean way |
| 05:53.07 | Ragnarok_ | it's defined by the firmware.. you're thinking of the services menu I think.. which basically is a custom XML browser |
| 05:53.08 | YoMama | Ragnarok: so...why don't u put the SIP image on the phones? |
| 05:53.20 | Ragnarok_ | because the sound quality on the SIP images is terrible |
| 05:53.53 | Ragnarok_ | and that requires a cisco support contract too :) |
| 05:54.08 | YoMama | Ragnarok: no..when Cisco phones start up with the skinny client..they first connect to the VLAN, then grab an IP..then connect to a TFTP server, defined within the DHCP...if that tftp server has a newer "build" than what's on the phone..it grabs it and installs it |
| 05:54.20 | Ragnarok_ | yeah |
| 05:54.30 | Ragnarok_ | but in order to get the firmware updates, you need a cisco support contract |
| 05:54.33 | Ragnarok_ | well, to legally get them |
| 05:54.37 | YoMama | hehe |
| 05:54.44 | YoMama | i'm sure u know someone who has a Cisco TAC account |
| 05:54.48 | konfuzed | YoMama: Im gonna whoop your cisco config with the latest OpenBSD3.8 with two boxes on carp and 6 dsl modems each |
| 05:54.59 | konfuzed | ;^) |
| 05:55.01 | YoMama | haha...go fer it |
| 05:55.08 | Ragnarok_ | well, a 1 year smartnet contract is only $60.. so not that big a deal.. but still slimy business practices |
| 05:55.14 | konfuzed | in about 6 months |
| 05:55.17 | YoMama | i'm lame man..i've been outta the hacking game for years..i'm a suit now |
| 05:55.17 | konfuzed | ;^) |
| 05:55.29 | YoMama | i am only a poser hacker now |
| 05:55.34 | konfuzed | haha |
| 05:55.42 | konfuzed | i did not suggest hack it |
| 05:55.54 | konfuzed | just to be unkonfuzed |
| 05:55.59 | YoMama | Ragnarok: welcome to 2005 |
| 05:56.00 | *** join/#asterisk syle (n=blah@unaffiliated/syle) |
| 05:56.04 | Ragnarok_ | heh |
| 05:56.23 | YoMama | Ragnarok: so buy it for one phone...get the image..and load it onto 200 phones :) |
| 05:56.29 | Ragnarok_ | that's the plan :) |
| 05:56.31 | *** join/#asterisk marc324 (n=marc3234@206-248-133-19.dsl.teksavvy.com) |
| 05:56.44 | konfuzed | then I can put a proper * box on the inside |
| 05:57.01 | marc324 | i get "check debug for more info" ... where is debug info? |
| 05:57.09 | konfuzed | inside of a 100 unit buiding that is |
| 05:57.17 | konfuzed | slow moving project I tell ya |
| 05:59.39 | konfuzed | they should be able to complete the round robin aproval game by the end of the month |
| 05:59.46 | konfuzed | end of december at the latest |
| 05:59.47 | konfuzed | ;^) |
| 06:01.07 | konfuzed | so maybe in six months they'll agree where to put the box |
| 06:01.08 | konfuzed | ;^) |
| 06:02.07 | *** join/#asterisk MSpin (n=jjhuff@sense-sea-MegaSub-1-579.oz.net) |
| 06:02.20 | *** part/#asterisk MSpin (n=jjhuff@sense-sea-MegaSub-1-579.oz.net) |
| 06:02.30 | YoMama | anyone ever try one of these grandstream phones? |
| 06:02.38 | konfuzed | ya |
| 06:02.44 | Flauto | hi there |
| 06:02.44 | konfuzed | pretty good stuff |
| 06:02.53 | Flauto | is anyone here using vbuzzer? |
| 06:03.29 | konfuzed | until the downtown transformer (across street from building) blew and spike all the power in the building |
| 06:03.53 | konfuzed | crappy wiring hurt the gear |
| 06:04.01 | konfuzed | could be heat sensitive too |
| 06:04.03 | YoMama | konfused: wut....isn't the building have conditioned power? |
| 06:04.05 | *** join/#asterisk BhaalWTF (n=bhaal@CPE-141-168-108-119.qld.bigpond.net.au) |
| 06:04.16 | konfuzed | uhm not that building |
| 06:04.25 | konfuzed | like one of the oldest buildings downtown |
| 06:04.25 | YoMama | are the linksys phones any good? |
| 06:04.35 | konfuzed | beside the bus terminal and nothing else |
| 06:04.43 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 06:04.45 | YoMama | why is a 2 line linksys phone $85 and the grandstream 11-line phone $85...seems like a no brainer |
| 06:04.47 | *** part/#asterisk buu (i=buu@ded430-deb-175-30.netsonic.net) |
| 06:05.01 | konfuzed | like the whole downtown core was down for like 2 days |
| 06:05.29 | YoMama | konfuzed: where do u live? |
| 06:05.41 | konfuzed | toront |
| 06:05.46 | konfuzed | tor ont |
| 06:05.50 | YoMama | i like toronto |
| 06:06.01 | YoMama | how the hell can the power be off for 2 days in toronto |
| 06:06.08 | Flauto | i have been there, toronto is pretty good |
| 06:06.19 | YoMama | except for toronto and vancouver..the rest of canada seems sorta...lame |
| 06:06.22 | Flauto | i went there back in may |
| 06:06.26 | konfuzed | bay dundas power transformer |
| 06:06.30 | YoMama | montreal can be pretty...but filled with annoying people |
| 06:06.32 | konfuzed | a small piece of down town |
| 06:06.48 | *** join/#asterisk Juggie (i=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 06:06.55 | Flauto | hey people from canada, do you guys use vbuzzer? |
| 06:07.11 | konfuzed | the area covers eaton center - city hall - previous parliament ype building now a big court house |
| 06:07.11 | YoMama | wut the hell is vbuzzer? |
| 06:07.20 | konfuzed | and the bus terminal |
| 06:07.25 | YoMama | eaton center was outta power for 2 days? |
| 06:07.37 | konfuzed | never herd uv vbuzzzzzer |
| 06:07.52 | Math` | Flauto: vbuzzer? |
| 06:08.04 | konfuzed | crucial transformer |
| 06:08.33 | konfuzed | the bell CO of course has all kinds of power backups but |
| 06:08.54 | YoMama | flauto: oh...it's like skype |
| 06:08.56 | konfuzed | that pipe was like the main pipe of that local power grid |
| 06:09.06 | YoMama | flauto: everyone in europe uses skype |
| 06:09.18 | YoMama | flauto: and if you're looking for an asterisk-friendly softphone..just get xlite |
| 06:09.34 | Flauto | no |
| 06:09.50 | Flauto | vbuzzer is not a softphone |
| 06:09.51 | YoMama | k |
| 06:09.57 | YoMama | flauto: k..then what is it? |
| 06:09.59 | Math` | its a service? |
| 06:10.08 | YoMama | is it bigger than a breadbox? |
| 06:10.09 | Flauto | www.vbuzzer.com |
| 06:10.18 | Flauto | yes, math |
| 06:10.36 | Flauto | i have seem some people are able to set it up to work with asterisk |
| 06:10.36 | YoMama | right..it's like skype |
| 06:10.38 | Flauto | but i tried |
| 06:10.43 | Flauto | it is not working |
| 06:10.48 | YoMama | except skype doesn't work with asterisk |
| 06:10.58 | Math` | its a provider |
| 06:11.45 | Flauto | yes, math |
| 06:11.55 | mog_home | and skype is lame |
| 06:12.35 | Math` | iaxness rule :) |
| 06:12.38 | Flauto | mog, other than fwd, which sucks ass, i dont have anything that i can use iax |
| 06:12.59 | Math` | Flauto: voipjet |
| 06:13.02 | YoMama | if vbuzzer works with asterisk...for $8 unlimitd calling..that kicks ass |
| 06:13.06 | Math` | fwd doesnt suck-ass with sip :P |
| 06:13.06 | mog_home | umm real providers... |
| 06:13.14 | Flauto | i have heard of it and my friends are using it |
| 06:13.16 | Math` | YoMama: yeah thats why Im IDA'ing it now :P |
| 06:13.17 | mog_home | or form your own rag tag network |
| 06:13.18 | Flauto | but i use broadvoice |
| 06:13.21 | mog_home | we have dundi for a reason |
| 06:13.29 | Math` | yeah dundi is nice |
| 06:13.34 | YoMama | Math: IDA? |
| 06:14.00 | Math` | Intelligent DissAssembler |
| 06:14.00 | Flauto | i never know how dundi works |
| 06:14.14 | mog_home | dundi is like dynamic dns |
| 06:14.29 | YoMama | Math: gotcha...you're seeing if vbuzzer uses SIP |
| 06:14.38 | mog_home | err dynamic decentralized dns |
| 06:14.38 | Math` | uhm string "-proto jabber" |
| 06:14.42 | Math` | YoMama: its sip? |
| 06:14.48 | mog_home | -proto jabber? |
| 06:15.06 | konfuzed | SCCP |
| 06:15.08 | Math` | well thats in the strings |
| 06:15.09 | YoMama | Math: uhh...wait...why do you need to dissessemble the code? it's available |
| 06:15.21 | Math` | YoMama: not for the softphone |
| 06:15.42 | Math` | "Vbuzzer is made up of two separate programs: Vbuzzer Softphone (internet telephony) and Vbuzzer Messenger (instant messenger)." |
| 06:15.43 | YoMama | oh..i see... |
| 06:15.48 | YoMama | bastards |
| 06:15.51 | Math` | "The Vbuzzer Softphone is the proprietary work of Softroute and its covered by our Terms of Service. |
| 06:15.51 | Math` | The Vbuzzer Messenger is a program covered by GNU General Public License (GPL). You can download the source code here. |
| 06:15.54 | konfuzed | ah |
| 06:16.03 | Math` | doesnt seem to support nat properly tho lol |
| 06:16.03 | konfuzed | ah |
| 06:16.05 | konfuzed | ah |
| 06:16.06 | konfuzed | aj |
| 06:16.11 | Math` | cuz its frozen at connecting here |
| 06:16.19 | YoMama | Math: i have no idea if it's SIP...but if it is..and u can figure out the proper username, password, and SIP Proxy...then rock on |
| 06:16.51 | konfuzed | one of my projects is to figure out how to have jabber and asterisk talk to each other in useful ways |
| 06:17.23 | mog_home | just wait konfuzed just wait |
| 06:17.25 | mog_home | or donate! |
| 06:17.32 | Math` | er.... SIPStack.dll SIPPhone.dll |
| 06:17.38 | Math` | didnt need to disassemble to see more lol |
| 06:17.56 | Math` | guess I'll try it |
| 06:17.56 | YoMama | hmm |
| 06:18.19 | konfuzed | if i cant get these button heads to use the stuff then these golden widgets are trash |
| 06:18.37 | konfuzed | ;^( |
| 06:19.05 | Math` | think so part of it even has debug symbols :P lol |
| 06:19.16 | mog_home | ? |
| 06:19.34 | Math` | (of vbuzzer's sip dll) |
| 06:19.37 | YoMama | Math: u gotta know the IP address tho and the right way to format the usernamd and password |
| 06:19.40 | YoMama | username |
| 06:19.46 | Math` | I could easily hook it |
| 06:19.55 | Math` | YoMama: well... just some hooks in the SIP dlls and et voilเ |
| 06:20.03 | Math` | I use their software and grab the info from their servers |
| 06:20.41 | Math` | it has debug symbols... I see class names, etc |
| 06:21.23 | Flauto | math, i have all the information |
| 06:21.26 | Flauto | from what i read |
| 06:21.46 | konfuzed | sounds like a race is on |
| 06:21.50 | YoMama | Math: or you dial out and if u got a good firewall..u can see what it's connecting to |
| 06:22.07 | Math` | yeah but if it uses md5 authentications Im screwed :P |
| 06:22.20 | YoMama | ugh |
| 06:22.20 | Flauto | well |
| 06:22.28 | Flauto | i am able to have it registered |
| 06:22.41 | Flauto | but can not call out successfully |
| 06:23.34 | Math` | "Offer ends on December 31 2005" |
| 06:23.47 | Math` | Flauto: your behind a nat probably... |
| 06:23.55 | Math` | support for this is beyond the scope of this channel |
| 06:24.38 | Math` | and it only has the 416 area code for canada |
| 06:24.41 | Flauto | i set nat=yes and i put the externip and local |
| 06:24.55 | YoMama | Math: i only wanna do outbound calling |
| 06:24.59 | YoMama | i could care less about inbound |
| 06:25.27 | Math` | YoMama: and resell it 1.1c/min? lol |
| 06:25.45 | Math` | what does it do when u call out |
| 06:25.45 | YoMama | zingotel supports sip for sure...but it's $15/month and then they hit u with $39 when u disconnect |
| 06:26.05 | Flauto | it rings |
| 06:26.05 | Math` | spectravoice hits me with 40$ if I don't give the mediatrix 2102 back |
| 06:26.10 | Flauto | but when other party answers |
| 06:26.20 | Flauto | i can not hear anything |
| 06:26.22 | Flauto | dead |
| 06:26.28 | YoMama | Math: i forgot to return my vonage box... :( vonage sucked |
| 06:26.31 | Math` | check nat settings for rtp |
| 06:26.39 | Math` | YoMama: what kind of box is it? :) |
| 06:26.44 | YoMama | Flauto: wait..what's the problem you're having? |
| 06:26.48 | konfuzed | oh shit |
| 06:26.51 | YoMama | Math: it's a Motorola..black.. |
| 06:26.59 | konfuzed | better return that vonage box already |
| 06:27.02 | YoMama | Flauto: NAT + RTP = big problems |
| 06:27.15 | Math` | I managed to find the firmware to the mediatrix on a russian site... I flashed it, contacted mediatrix support and they gave me access to the software for managing the units |
| 06:27.25 | YoMama | konfuzed: i'd rather unlock it and use it as my own ATA |
| 06:27.31 | Math` | so I'm using it with * :) |
| 06:27.45 | YoMama | Math: no idea on the motorola boxes? |
| 06:27.54 | Math` | YoMama: the key to that is... find a firmware, sniff for how their firmware auto-update works |
| 06:27.57 | Math` | and google for documentation |
| 06:27.59 | konfuzed | so you cover the charge on my buddies card and I'll send it to ya |
| 06:28.07 | Math` | YoMama: are you close to montreal? |
| 06:28.28 | Math` | YoMama: what motorola model is it |
| 06:29.10 | Flauto | yomama, any solutions? |
| 06:29.22 | Math` | I'd just contact motorola and tell them "I'm very interested at deploying your products. For testing it, I bought the [model here] on eBay, but it didnt come with ahything, could you provide me with the necessary stuff?" |
| 06:29.32 | YoMama | Math: i can't find it for the moment |
| 06:29.44 | Math` | its usually written on it :P |
| 06:29.52 | YoMama | Math: haha...yeah..or i could pay the blood money to vonage..they'll unlock it for a "fee" |
| 06:30.00 | YoMama | Math: I know..i can't find the box..it's buried under some crap |
| 06:30.03 | Math` | riiight |
| 06:30.10 | Math` | open it |
| 06:30.15 | Math` | it must be written on the PCB |
| 06:30.16 | Math` | hehe |
| 06:30.37 | Math` | found another mediatrix on ebay... 30$CAD buy it now +10 shipping |
| 06:30.39 | Math` | jumped on it |
| 06:31.04 | YoMama | Flauto: they can hear you but u can't hear them? |
| 06:31.11 | YoMama | Flauto: the conversatino is only one-way right? |
| 06:31.20 | Flauto | i am not even sure if the other end can hear me |
| 06:31.22 | YoMama | Flauto: u can talk but u can't hear them |
| 06:31.31 | YoMama | Flauto: my guess is they can... |
| 06:31.37 | Flauto | maybe |
| 06:31.45 | Flauto | are you in toronto? |
| 06:31.47 | Flauto | i can call you |
| 06:31.51 | Flauto | and you can tel me |
| 06:31.52 | Math` | YoMama: http://cgi.ebay.com/Mediatrix-2102-Like-Sipura-and-Cisco-ATA186-router_W0QQitemZ5828871640QQcategoryZ3706QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
| 06:31.56 | Math` | they won't go very expensive |
| 06:32.08 | YoMama | Flauto: i had this problem with Cisco IP communicator |
| 06:32.15 | Math` | and its a 2 port |
| 06:32.17 | Math` | (fxs) |
| 06:32.53 | konfuzed | Math`: ever use those mediatrix with a mitel linux server |
| 06:33.04 | konfuzed | with voip blade |
| 06:33.08 | Math` | konfuzed: no but I can pretty much configure anything with SIP |
| 06:33.15 | Math` | or H323 (lemme check the firmwares I have) |
| 06:33.22 | YoMama | Math: hmm |
| 06:33.35 | konfuzed | im thinkin of dumpin those two esmith boxes |
| 06:33.51 | Math` | I got sip, mgcp and h232 |
| 06:34.06 | Math` | YoMama: if you want to pay shipping I can unlock the unit and mail it back to you |
| 06:34.08 | YoMama | u mean h323 right? |
| 06:34.11 | konfuzed | both morons but both boxes on old dell 233 I think off white medium tower |
| 06:34.16 | Math` | YoMama: yeah I mean that |
| 06:34.31 | YoMama | Math: k...lemme find the stupid box first |
| 06:34.35 | YoMama | it wasn't where i thought it was |
| 06:34.41 | Math` | (I mean for the mediatrix) |
| 06:34.47 | YoMama | ah |
| 06:34.54 | Math` | not sure if I can do it for the motorola... but gimme the model number and I'll try to find documentations |
| 06:34.58 | konfuzed | even got or had a spare |
| 06:35.08 | Math` | whats the small fee they ask for? |
| 06:35.17 | YoMama | Math: i think it's like $40 |
| 06:35.20 | YoMama | it's a rip |
| 06:35.26 | Math` | yeah... |
| 06:35.40 | Math` | you got a linux box handy? (which you can unplug the net from it) |
| 06:35.51 | konfuzed | yeah |
| 06:35.51 | Math` | plug the unit directly and sniff it |
| 06:36.08 | konfuzed | well |
| 06:36.12 | Math` | was talking to YoMama |
| 06:36.16 | Math` | konfuzed: what do u need? |
| 06:36.19 | konfuzed | ;) |
| 06:36.40 | konfuzed | a vacation |
| 06:36.48 | YoMama | Math: i think the entire unit has to be reflashed to unlock it |
| 06:37.06 | YoMama | Math: or are u talkingabout this vbuzzer thing? |
| 06:37.09 | konfuzed | aahh tryin to figure out a database setup |
| 06:37.17 | Math` | YoMama: unit flash |
| 06:37.24 | konfuzed | and what to do with these 8 boxes in the corner |
| 06:37.34 | Math` | YoMama: those ATA usually have an automatic firmware update |
| 06:37.48 | YoMama | Math: hmmm...oooh..i gotcha now |
| 06:37.50 | Math` | you google and try to find the firmware (or obtain it by social engineering) and you build a little tftp server |
| 06:37.56 | YoMama | Math: i can get a box to pretend it's the firmware updater |
| 06:38.00 | Math` | :) |
| 06:38.03 | YoMama | riiight |
| 06:38.11 | YoMama | and then it flashes itself with the unlocked software |
| 06:38.13 | konfuzed | edubuntu finishes downloading in 5 mins |
| 06:38.24 | Math` | for mediatrixes its kinda easy, you press reset for 20 seconds and it reboots and attempt to download the firmware from tftp://192.168.0.10/Setup.inf |
| 06:38.26 | konfuzed | gonna set it up with asterisk |
| 06:38.56 | Math` | guess I'm gonna write a HOWTO: Unlock mediatrixes |
| 06:39.09 | Math` | (its the same firmware for all of their ATAs) |
| 06:39.24 | YoMama | i'm sure someone has posted the image for my motorola somewhere |
| 06:39.32 | Math` | try to dig out to find the model |
| 06:39.38 | YoMama | usually in some eastern european country :-P |
| 06:39.46 | Math` | lol |
| 06:40.05 | Math` | my site was so slow that I wget -m'd it overnight |
| 06:40.52 | Flauto | math |
| 06:41.05 | Math` | YoMama: is it dual-port? |
| 06:41.07 | Math` | Flauto: yeah? |
| 06:41.10 | Flauto | have you figured out what is going on with that vbuzzer thing? |
| 06:41.27 | Math` | Flauto: its an RTP nat problem |
| 06:41.37 | Math` | make sure you forward ports 10000 to 20000 |
| 06:41.40 | Math` | udp, taht is |
| 06:41.41 | Math` | that* |
| 06:41.54 | Flauto | i did |
| 06:41.56 | YoMama | Mathl; yeah |
| 06:42.13 | Math` | does it looks like http://images.amazon.com/images/P/B0000WZWTM.01.LZZZZZZZ.jpg |
| 06:42.16 | YoMama | Flauto: i know what the problem is...voice is going out..but not coming back in because RTP is like IPSec |
| 06:42.29 | YoMama | Math: YES..that's it |
| 06:42.32 | Math` | its a Motorola VT1005v |
| 06:42.35 | YoMama | yup |
| 06:42.42 | YoMama | that's it..i remember the model number |
| 06:42.43 | Math` | did a google for "motorola vonage" hehe |
| 06:43.14 | Flauto | and also incoming calls are not working |
| 06:43.20 | Math` | uh |
| 06:43.30 | Flauto | it does not come in at all |
| 06:44.05 | YoMama | Flauto: right |
| 06:44.15 | Flauto | it is also rtp? |
| 06:44.28 | Math` | no thats SIP |
| 06:44.32 | Math` | SIP is for signalling |
| 06:44.32 | Flauto | i have nat and forwarded the rtp ports |
| 06:44.36 | Math` | and RTP is for audio traffic |
| 06:44.52 | Math` | Flauto: you know what the best is.... get a second nic and use your linux box as router |
| 06:47.34 | Math` | Flauto: uhmm I got a vt100v on motorola's site, I wonder if they are compatible |
| 06:47.37 | *** join/#asterisk L|NUX (n=linux@202.5.145.56) |
| 06:47.41 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:47.46 | YoMama | oh...they'll unlock it for $15 |
| 06:47.49 | YoMama | still |
| 06:47.56 | YoMama | Math: i think u meant that for me |
| 06:48.14 | *** part/#asterisk newmember (n=newmembe@S010600a0c93dce87.cg.shawcable.net) |
| 06:48.20 | Math` | YoMama: yeah |
| 06:49.32 | Math` | they're compatible, its in the user manual |
| 06:49.44 | Math` | vt1000v is the "suite" 1003 is 1 line 1005 is 2 line |
| 06:51.09 | YoMama | Math: and u grabbed the firmware? |
| 06:51.27 | Math` | no its not on their site |
| 06:51.38 | YoMama | no...i wouldn't think so |
| 06:52.22 | *** join/#asterisk tclark_ (n=TC@S0106000f66c5d294.gv.shawcable.net) |
| 06:53.11 | Math` | ok there's a single firmware for all motorola voip appliances |
| 06:56.04 | Math` | telnet & snmp interface for management |
| 06:57.07 | YoMama | eh...if i can't find it soon..i'll just pay them the $15...have a nice unlocked ATA |
| 06:57.16 | YoMama | i'm digging thru usenet right now |
| 06:59.07 | Math` | you can also subscribe as a service provider at their site :P |
| 06:59.16 | Math` | and they're gonna provider you manuals and all you need |
| 06:59.30 | YoMama | haha...that's another way to do it |
| 06:59.53 | Math` | thats the way I prefer.. but even if you pay the 15$.... they're gonna give you the SNMP MIBs to manage it? |
| 07:00.10 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 07:00.13 | Ragnarok_ | psh SNMP is so easy to reverse engineer |
| 07:00.20 | Math` | yeah right |
| 07:00.33 | Math` | 1.2.3.53.32.1.5.6.2.1.3.5.2.5.7.9.2.4.6.2 = 1 |
| 07:00.45 | Math` | what value did I modify? |
| 07:01.06 | Ragnarok_ | fuck if I know :) you can find things in context tho |
| 07:01.33 | YoMama | Math: hehehe |
| 07:01.34 | Math` | yeah but... for mediatrix... they use smth.enterprise.private... context |
| 07:01.46 | Math` | so without the MIBs you can try to guess but... still |
| 07:01.54 | YoMama | shit..how much is a dual port FXS ata device now anyway? |
| 07:02.03 | Math` | YoMama: new? 150$? |
| 07:02.29 | Math` | YoMama: monitor the mediatrixes on ebay... they won't really go expensive |
| 07:03.08 | YoMama | hmm |
| 07:03.18 | Math` | I'm subscribing to motorola as company wanting to provider voip services and that want to investigate their products |
| 07:03.24 | Math` | (which isnt completely false) |
| 07:03.24 | YoMama | u get anywhere with that vbuzzer thing? |
| 07:03.32 | YoMama | i know someone who works at motorola |
| 07:03.36 | Math` | I dont wanna pay for that now |
| 07:03.36 | YoMama | i'm gonna call him tomorrow |
| 07:03.38 | Math` | oh thats nice |
| 07:04.12 | *** join/#asterisk lehel (n=lehel@82.79.20.17) |
| 07:04.16 | Math` | I'm gonna be publicizing & implementing asterisk in businesses in the next month... I'll probably get myself a centralised server to relay SIP traffic etc... |
| 07:04.36 | Math` | ensure IAX2 (maybe with dundi) connectivity between clients |
| 07:04.52 | YoMama | Math: a buddy of mine makes good money putting in asterisk servers for call centers actually |
| 07:05.05 | Math` | I can charge a low rate and get even-lower rates from the wholesale providers |
| 07:05.13 | Ragnarok_ | I could see that making money |
| 07:05.24 | Math` | yeah... well there's always a lot of money to make in telemarketting |
| 07:05.30 | YoMama | it's the cheapest way to set up a call center |
| 07:05.35 | Ragnarok_ | with phone systems you can charge 10x what it's worth and still be half the price of the "hardware" PBX systems |
| 07:05.37 | YoMama | asterisk that is |
| 07:05.40 | Ragnarok_ | yeah |
| 07:05.45 | Math` | yeah |
| 07:06.02 | YoMama | there are a few companies in my local area that use it for call center crap |
| 07:06.02 | Math` | still.... Bell Canada charges 25 cents/min for a toll-free number |
| 07:06.05 | YoMama | IVR, etc. |
| 07:06.09 | Math` | I freaked out when I got told that |
| 07:06.22 | YoMama | cnada hasn't woken up to deregulation yet |
| 07:06.24 | Ragnarok_ | I'm using a ~$10,000 asterisk buildout to replace a $50,000 PBX, only asterisk has more features |
| 07:06.30 | Ragnarok_ | wow |
| 07:06.31 | Math` | lol |
| 07:06.42 | YoMama | Ragnarok: that's nuts |
| 07:06.44 | Math` | and you can upgrade the features for free |
| 07:06.47 | Ragnarok_ | yeah |
| 07:07.06 | Ragnarok_ | I don't have to pay $2500 for every conference bridge :P |
| 07:07.11 | YoMama | well, let's do the math here...a PRI card is how much |
| 07:07.15 | Ragnarok_ | $500 |
| 07:07.15 | YoMama | $1500? |
| 07:07.18 | YoMama | that's it? |
| 07:07.18 | YoMama | ok |
| 07:07.26 | Ragnarok_ | yeah, TE110P from digium |
| 07:07.40 | Math` | heh |
| 07:07.40 | YoMama | $500..let's say u get some analog lines..another $200 for 2 fxs and 2 fxo's |
| 07:07.52 | YoMama | $700...plus a server that doesn't suck...$1200 |
| 07:07.53 | Math` | just make em buy IP phones if its a new system |
| 07:07.58 | YoMama | so let's round up...$2000 hardware |
| 07:08.02 | YoMama | software...free |
| 07:08.07 | Math` | they most of the time already have a network wired everywhere |
| 07:08.09 | Ragnarok_ | phones are by far the biggest expense |
| 07:08.15 | YoMama | those grandstream phones with 11 freakin' lines is only $85 |
| 07:08.15 | Math` | yeah |
| 07:08.19 | Math` | :o |
| 07:08.30 | Ragnarok_ | well we're doing cisco phones |
| 07:08.36 | YoMama | yeah..but IP phone for $85 versus cisco's $250... |
| 07:08.56 | YoMama | Ragnarok: everyone likes the way those look |
| 07:08.56 | Math` | cisco's 250$? you mean the lowest-end one? :P |
| 07:09.06 | Ragnarok_ | nah the 7940s |
| 07:09.14 | Ragnarok_ | the lowest end ones are like $50 on ebay :P |
| 07:09.18 | Ragnarok_ | new |
| 07:09.20 | Math` | hehe |
| 07:09.21 | YoMama | Math: yeah..u can get the 7940's for under $250 actually |
| 07:09.39 | YoMama | the one good thing about the cisco ones is the speakerphones on 'em use the polycom chips |
| 07:09.41 | YoMama | real good |
| 07:09.50 | Ragnarok_ | we're doing a few of the 7902s actually... they're uber cheap, for cube workers |
| 07:09.56 | Math` | http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet0900aecd8030e546.html |
| 07:10.03 | Math` | I wonder how much it retails heh |
| 07:10.06 | YoMama | i have no idea what the speakerphone sounds like on the grandstream |
| 07:10.19 | ptiggerdine | speaker phone is good. |
| 07:10.34 | Ragnarok_ | yeah my boss uses his speaker phone more than he uses the handset |
| 07:10.40 | YoMama | ptigger..on what..the grandstreams? |
| 07:11.31 | YoMama | Math: what if the vbuzzer thing just takes plaintext login? |
| 07:11.36 | Math` | grandstream got 2-port FXS for less then 70$us |
| 07:11.37 | YoMama | Math: how sweet would that be? |
| 07:11.52 | Math` | YoMama: the guy had all the registration info and everything |
| 07:11.52 | YoMama | $8/month to call anywhere in north america |
| 07:11.56 | Math` | he just had nat problems |
| 07:12.08 | YoMama | Math: flauto? |
| 07:12.11 | Math` | ya |
| 07:12.15 | Math` | I got 500 free ld minutes in north america and I don't even use them |
| 07:12.16 | Ragnarok_ | well, I had to go with a TDM card.. our analog lines are fax/modem stuff so the latency in converting to an IP network would kill them |
| 07:12.45 | YoMama | Math: hmm |
| 07:12.48 | YoMama | maybe he'll come back |
| 07:13.15 | YoMama | well, even zingotel is a good deal |
| 07:13.20 | YoMama | u even get an incoming # with that one |
| 07:13.27 | YoMama | i dunno if tghe service is any good |
| 07:13.30 | Math` | heh grandstream even has tftp servers for upgrades, so you always to latest released firmware :) |
| 07:13.39 | YoMama | but it can't suck as bad as vonage |
| 07:13.52 | Math` | lol I didnt even try vonage |
| 07:13.57 | YoMama | it blew |
| 07:14.00 | Math` | I hear they charge per-minute for using a soft-phone |
| 07:14.02 | Math` | sucks badly |
| 07:14.09 | YoMama | that's ridiculous |
| 07:14.12 | *** join/#asterisk mmmToop (n=chatzill@196.31.11.194) |
| 07:14.26 | Ragnarok_ | heh I haven't had a land line in probably 7 years... |
| 07:14.34 | Math` | I wanted to get vonage but I checked voip-info's wiki first and I got another service :) |
| 07:14.39 | Ragnarok_ | voip or pots |
| 07:14.50 | Math` | the only thing that bugs me is that Im out of phone in case of power-failure |
| 07:14.56 | Math` | but a good UPS will change that pretty soon |
| 07:14.58 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 07:15.37 | YoMama | Math: who do u use? |
| 07:15.47 | Math` | grandstreams are nice... Multiline support of up to 11 lines indicators (expandable to a few dozen more through expansion key-module) |
| 07:15.53 | Math` | YoMama: spectravoice, division of colbanet |
| 07:16.02 | YoMama | Math: u a canuck? |
| 07:16.05 | Ragnarok_ | math: got a picture of the grandstream 11 like phone? |
| 07:16.06 | Math` | lol |
| 07:16.10 | Ragnarok_ | haha |
| 07:16.16 | Math` | http://www.grandstream.com/y-gxp2000.htm |
| 07:16.24 | YoMama | http://www.grandstream.com/ |
| 07:16.28 | YoMama | their front page has one |
| 07:16.31 | Ragnarok_ | oh that's not too bad |
| 07:16.32 | YoMama | it's a big ass phone |
| 07:16.44 | Math` | buy it now @ 88$US http://cgi.ebay.com/Grandstream-GXP-2000-VOIP-Phone-Black_W0QQitemZ5828911104QQcategoryZ61840QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
| 07:17.10 | Math` | anyone here is doing mass-phone-ordering has a good store to refer me? |
| 07:17.10 | YoMama | http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-43410781952.htm <-- $85 |
| 07:17.18 | Math` | im checking voipsupply now |
| 07:17.36 | Ragnarok_ | yeah voipsupply.. ordered phones through them, they were really quick on the shipping |
| 07:18.04 | Ragnarok_ | ordered the phone at 4pm and it was on my desk when I got in the next morning |
| 07:18.17 | Math` | even has power over ethernet, if the cabling is good |
| 07:18.28 | Ragnarok_ | yeah I'm doing PoE with the cisco phones |
| 07:18.29 | Math` | Ragnarok_: awesome |
| 07:18.37 | Ragnarok_ | works really well |
| 07:18.44 | Math` | Ragnarok_: can you deal with them for cheaper prices which you buy a lot? |
| 07:18.59 | Math` | probably... its a store |
| 07:18.59 | Math` | lol |
| 07:19.01 | YoMama | Math: that one link i pasted says they do volume |
| 07:19.06 | *** join/#asterisk GXTi (n=omgwtfbb@freenode/developer/GXTi) |
| 07:19.06 | Math` | oh nice |
| 07:19.16 | YoMama | Math: that one grandstream is $15 cheaper on atacomm's website |
| 07:19.38 | YoMama | although voipsupply has a nice selection |
| 07:20.35 | Math` | is voipsupply us-based |
| 07:20.49 | Ragnarok_ | yes |
| 07:21.06 | Math` | ok so they are both cheap on the shipping |
| 07:21.22 | Math` | (I wont pay for air shipping from australia when I can order from here or from the us ) |
| 07:21.32 | Ragnarok_ | yeah' |
| 07:21.38 | YoMama | 716 is their area code |
| 07:21.40 | YoMama | wherever that is |
| 07:22.20 | Ragnarok_ | whois says they're in buffalo |
| 07:22.54 | Math` | ok |
| 07:23.08 | Math` | YoMama: with voip you base yourself on area codes? :P |
| 07:23.30 | Math` | unlimitel isn't bad... 2.50$cad/month for a DID number (1.1c/min) |
| 07:23.45 | Ragnarok_ | hell even cell phones... most people I know don't have an area code from this city |
| 07:24.24 | YoMama | Math: haha..true..who knows |
| 07:24.43 | YoMama | Math: vbuzzer gives u 30 free minutes when u sign up...hmmm :) |
| 07:24.52 | Math` | hehe |
| 07:25.00 | Math` | enough to make it work with * |
| 07:25.05 | YoMama | yup |
| 07:25.16 | YoMama | all i gotta do is load up vbuzzer and figure out the IP address it's tlaking to |
| 07:25.18 | Math` | but still.... for the LDs Im making its not worth it |
| 07:25.29 | Math` | uhm anyone's on dundi-e164? |
| 07:26.15 | YoMama | nope |
| 07:26.22 | YoMama | haven't gotten that far |
| 07:26.31 | Math` | I need to sign the damn paper and fax it |
| 07:26.43 | Math` | tho I'm on dundi-test |
| 07:26.48 | Math` | (exporting +1-514 and +1-450) |
| 07:27.15 | YoMama | what's that mean? |
| 07:27.25 | YoMama | if someone dials 514 they call you? |
| 07:27.29 | Math` | thease are area codes |
| 07:27.30 | *** join/#asterisk tartar (n=tartar@CPE0004e27b716e-CM014370001917.cpe.net.cable.rogers.com) |
| 07:27.32 | YoMama | right |
| 07:27.32 | Math` | no |
| 07:27.51 | YoMama | what's it mean when u "export" those? |
| 07:27.51 | Math` | if someone looks up 1514NXXXXXX thru dundi-test, I provide the pstn termination |
| 07:27.58 | YoMama | right |
| 07:28.08 | YoMama | so if soemne dials that...they can go thru your phone line? |
| 07:28.12 | Math` | yup |
| 07:28.17 | YoMama | hmm |
| 07:28.27 | Math` | its unlimited so I dont really give a damn |
| 07:28.30 | YoMama | that's cool...except why would u want someone using your phone line |
| 07:28.33 | YoMama | oh...gotcha |
| 07:28.34 | Math` | unless they start complaining |
| 07:28.36 | YoMama | still |
| 07:28.39 | Math` | I got 2 lines |
| 07:28.42 | Math` | I limit free stuff to 1 line |
| 07:29.06 | Math` | well.. if everyone that had unlimited phone services were doing that... it'd be great :) |
| 07:29.28 | YoMama | hmm |
| 07:29.42 | YoMama | well, if i sign up for zingotel..or get vbuzzer working..heheh |
| 07:29.50 | *** join/#asterisk tengulre (n=tengulre@222.90.66.4) |
| 07:29.58 | YoMama | i don't care who uses it as long as i can kick 'em off when i wanna use it |
| 07:30.56 | Math` | you got 2 lines? |
| 07:31.46 | *** join/#asterisk TommyTheKid (n=tommy@63.82.105.244) |
| 07:32.08 | TommyTheKid | so, is there a place to report (bugs) in "the book" ? |
| 07:32.31 | YoMama | Math: i will if i sign up with zingotel and/or vbuzzer |
| 07:32.34 | Ragnarok_ | heh |
| 07:32.43 | TommyTheKid | or let me re-phrase that.. does anyone else thing that the top of page 94 has a bug (typo) ? |
| 07:33.10 | Ragnarok_ | is this the oreilly book you're talking about? |
| 07:33.24 | TommyTheKid | yea |
| 07:33.26 | *** join/#asterisk testmachine (n=assink@ip237-239-58-62.adsl.versatel.nl) |
| 07:33.34 | Ragnarok_ | looks fine to me |
| 07:33.36 | testmachine | HELLO |
| 07:33.41 | testmachine | argg |
| 07:33.42 | testmachine | caps |
| 07:33.49 | testmachine | Nov 11 08:32:07 ERROR[19610]: rate_engine.c:790 poster_worker: Failed to insert into database: Table 'rating.cdr' doesn't exist |
| 07:33.51 | TommyTheKid | was reading thru the pdf.. and it says if a user dialed 888-555-1212 it would match _55512XX |
| 07:34.00 | testmachine | i dont know what i have to add in this table |
| 07:34.11 | YoMama | Tommy: yeah...that's not right |
| 07:34.30 | TommyTheKid | i understand what they are getting at, but the "888" needs to go away |
| 07:34.35 | Ragnarok_ | well.. it depends how the call comes in... |
| 07:34.39 | Math` | is there any in-depth NANPA implementation for *? I mean, rate center distances etc... so I can evaluate whether the call is LD or not, even if its the same area code |
| 07:34.42 | Ragnarok_ | and what your setup is like |
| 07:34.52 | YoMama | Ragnarok: no..i'm looking at the book right now..it is a dialing pattern |
| 07:34.54 | Ragnarok_ | my PRI lines only give me the last 4 digits |
| 07:34.57 | Ragnarok_ | yeah |
| 07:35.04 | YoMama | Ragnarok: not incoming |
| 07:35.09 | Ragnarok_ | if I dialoh |
| 07:35.26 | YoMama | Ragnarok: yeah..phone companies like to give u 4, 7, or 10 digitrs |
| 07:35.29 | YoMama | just for fun |
| 07:35.36 | TommyTheKid | Ragnarok_: my PRI likes give me strange stuff too.. upstream PBX-isms :p 99XXXXX :) |
| 07:35.40 | Math` | tho I could list all NXX that are LD |
| 07:35.40 | Ragnarok_ | hehe |
| 07:36.18 | TommyTheKid | hmm.. I am sleepy.. "my PRI likes to give me strange ..." |
| 07:36.21 | Ragnarok_ | well my phone company is about to go under :) they just sold off all their assets except their austin CO |
| 07:36.24 | Math` | how much does a PRI cost per month? |
| 07:36.27 | Math` | lets say a voice T1 |
| 07:36.36 | Math` | (must depend of many stuff but... lets say the link only) |
| 07:36.40 | YoMama | Math: where? it's different everywhere |
| 07:36.42 | Ragnarok_ | 23 channels? probably in the 5-700 range |
| 07:36.56 | Math` | means 23 simultaneous calls? |
| 07:36.57 | YoMama | Math: i got the company i was working for...2 PRIs for $800/month |
| 07:36.59 | Ragnarok_ | yeah |
| 07:37.09 | Ragnarok_ | it all depends on location though.. |
| 07:37.13 | Math` | is it possible to plug a T3 into a digium card? |
| 07:37.23 | YoMama | Math: i don't think so |
| 07:37.42 | TommyTheKid | Math`: no.. DS3 I think Digium is working on a card... |
| 07:37.46 | Math` | oh :) |
| 07:37.49 | YoMama | Math: a T3 is uhh...i can't remember how many T1s |
| 07:37.49 | Ragnarok_ | I don't think a single PC could handle that many lines anyway ;P |
| 07:37.55 | YoMama | Ragnarok: true |
| 07:37.58 | TommyTheKid | but, IMO they will have to embed DSP to make that work |
| 07:37.59 | Math` | YoMama: 27 T1 |
| 07:38.23 | Ragnarok_ | yeah, but embedded DSPs are expensive and wouldn't use zaptel |
| 07:38.31 | Math` | Ragnarok_: sure? what about a Quad Xeon MP? :P |
| 07:38.32 | TommyTheKid | right 27, or 672 from a US channels perspective |
| 07:38.42 | testmachine | anybody knows what is in the rating.cdr table in the rating database???????? |
| 07:38.54 | testmachine | i have to make this table myself |
| 07:38.55 | Math` | its probably ratings |
| 07:38.57 | YoMama | if i did use a T3 with asterisk..i think i'd deplex it into separate T1s |
| 07:39.02 | Ragnarok_ | math: I think the interrupts from 600+ lines would kill the machine |
| 07:39.08 | Ragnarok_ | if the card were anything like their current ones |
| 07:39.09 | TommyTheKid | agreed |
| 07:39.09 | testmachine | but i dont know wich fields i have too add into it |
| 07:39.21 | Math` | Im gonna start using wholesale providers.... I'll get PRI when I'll have an established business |
| 07:39.36 | YoMama | Math: PRIs in Canada are expensive |
| 07:39.45 | Math` | YoMama: a data T1 is 1200$/m |
| 07:39.50 | Math` | ($cad) |
| 07:39.58 | TommyTheKid | ouch |
| 07:40.00 | Math` | with unlimited transfer quotas |
| 07:40.07 | Ragnarok_ | that's about right actually |
| 07:40.13 | YoMama | Math: exactly...u can get DS1 with internet in the States for like $450-500 now |
| 07:40.28 | Math` | there's probably cheaper |
| 07:40.30 | Math` | its Bell..... |
| 07:40.40 | YoMama | Math: I'll host your PRI for ya :) |
| 07:40.43 | Math` | they rent ip space for hundreds of dollars a month |
| 07:40.59 | TommyTheKid | yea lots of my uh "customers" have service from "Cebeyond" I think where they get a T1 that is part Internet and part channels.. dynamicly |
| 07:41.08 | Math` | unlimitel even allows u to colocate your server in the same room as the PRIs |
| 07:41.16 | Math` | so you can use ulaw to the pstn |
| 07:41.28 | YoMama | Math: there are companies in the US that offer PRIs with non-local #'s |
| 07:41.31 | YoMama | it's pretty sweet |
| 07:41.36 | Math` | heh |
| 07:41.50 | Math` | YoMama: ur in the us? |
| 07:41.52 | YoMama | that way u can have #'s all over the country that all come into one switch |
| 07:41.59 | YoMama | Math: yeah...although not far from canuckland |
| 07:42.00 | Ragnarok_ | hehe actually I'm looking at a quote right now for a point to point T1 for $320/mo : |
| 07:42.15 | YoMama | Math: I'm about 15 minutes outside of Windsor |
| 07:42.18 | TommyTheKid | local loop isl ike $300 last I looked |
| 07:42.21 | Math` | YoMama: once you paid your PRI.. how much you pay per DID or DOD? |
| 07:42.34 | YoMama | Math: I got a block of 100 DIDs for like $20/month |
| 07:42.41 | Math` | YoMama: windsor ontario on windsor quebec? they're 2 cities named windsor :P |
| 07:42.48 | YoMama | ON |
| 07:42.55 | Math` | YoMama: per-minute-rate? |
| 07:43.07 | YoMama | Math: 0.023 |
| 07:43.17 | Math` | thats pretty expensive |
| 07:43.25 | YoMama | 2.3 cents? |
| 07:43.42 | Math` | unlimited has DIDs for 2.5$CAD each , 0.011$/min |
| 07:43.49 | Math` | unlimitel** |
| 07:43.49 | *** part/#asterisk Qwell (n=north@24-205-180-81.dhcp.wsco.ca.charter.com) |
| 07:43.54 | *** join/#asterisk Qwell (n=north@24-205-180-81.dhcp.wsco.ca.charter.com) |
| 07:44.08 | YoMama | Math: that's a good LD rate |
| 07:44.15 | YoMama | Math: billed at six second increments? |
| 07:44.17 | Math` | thats for inbound |
| 07:44.31 | YoMama | wait...inbound? u pay per minute on inbound? |
| 07:44.31 | Math` | they don't specify but I guess so |
| 07:44.41 | hypa7ia | it covers outbound in most of ontario + montreal too |
| 07:44.45 | Math` | thats a wholesale IP provider.... |
| 07:44.53 | hypa7ia | unlimitel rocks. |
| 07:44.54 | YoMama | there's no inbound costs on the stuff i'm talking about |
| 07:45.00 | Math` | ah you don't pay per-minute for inboud on PRIs? |
| 07:45.02 | Math` | nice! |
| 07:45.03 | YoMama | no |
| 07:45.31 | Math` | and for outbound, use voxee, 0.011$/min anywhere in us48 + canada |
| 07:45.42 | Math` | or voipjet.. 0.013$/min |
| 07:45.49 | testmachine | nobody can help me? |
| 07:46.02 | testmachine | just like looking whats in their rating database |
| 07:46.07 | testmachine | in the rating.cdr table |
| 07:46.11 | testmachine | like wich fields? |
| 07:46.17 | YoMama | testmachine: ok...so what's this? |
| 07:46.19 | Math` | hypa7ia: you're a customer of unlimitel? |
| 07:46.20 | *** part/#asterisk Ragnarok_ (n=ramos@mx1.finetooth.com) |
| 07:46.44 | hypa7ia | Math`: yeah, i have a 416 did with them now, going to get another and a 1800 when i get my business running |
| 07:46.49 | Math` | hehe |
| 07:46.52 | testmachine | YoMama: like this message again:P Nov 11 08:46:33 ERROR[19888]: rate_engine.c:790 poster_worker: Failed to insert into database: Table 'rating.cdr' doesn't exist |
| 07:46.56 | Math` | they bill per 6 seconds increments? |
| 07:47.00 | hypa7ia | yup |
| 07:47.06 | hypa7ia | and send a daily CDR email |
| 07:47.12 | Math` | heh |
| 07:47.24 | Math` | from what I can see... as soon as you drop off their network its better to use another provider for DOD |
| 07:47.29 | hypa7ia | yeah |
| 07:47.36 | hypa7ia | i use stanaphone for now |
| 07:47.40 | Math` | its IAX2? |
| 07:47.41 | YoMama | testmachine: umm...i've never seen that before..u using mysql? |
| 07:47.48 | testmachine | YoMama: yes |
| 07:47.53 | hypa7ia | i use unlimitel's iax, stanaphone's sip |
| 07:48.06 | Math` | how much is stanaphone |
| 07:48.15 | hypa7ia | 2.5c US |
| 07:48.33 | hypa7ia | free incoming, which is cool |
| 07:48.42 | YoMama | oh wait...unlimitel is IP based? |
| 07:48.50 | hypa7ia | YoMama: yeah :) |
| 07:48.53 | Math` | YoMama: yes |
| 07:48.56 | YoMama | that stuff i was quoting you was traditional telephony |
| 07:49.03 | YoMama | like..a PRI through SBC |
| 07:49.06 | hypa7ia | yeah, i was wondering :) |
| 07:49.07 | Math` | I know, I wanted to compare :) |
| 07:49.37 | hypa7ia | nice |
| 07:49.46 | testmachine | apt-get install asterisk |
| 07:49.47 | testmachine | :) |
| 07:49.52 | Math` | as from what I see... if you don't have lots of DIDs, you're better off with an IP DID provider, when you get enough DID to pay the PRI, you switch |
| 07:49.58 | YoMama | Math: MCI now offers a "internet + voip gateway" service...have no idea how much it is though |
| 07:50.03 | hypa7ia | jdub gave a talk in toronto on monday |
| 07:50.06 | hypa7ia | was awesome :) |
| 07:50.16 | Math` | who's that? :P |
| 07:50.29 | testmachine | but i only cant find a sollution to my problem anywhere |
| 07:50.31 | Math` | hypa7ia: er stanaphone is 0.0260 for DOD..... |
| 07:50.32 | hypa7ia | he's a gnome / canonical / ubunty guy |
| 07:50.40 | testmachine | like i was searching for a few houres yesterday |
| 07:50.41 | YoMama | testmachine: i don't use mysql with asterisk...sorry |
| 07:50.45 | hypa7ia | Math`: did it go up? :s |
| 07:51.05 | Math` | dunno I just checked that on their frontpage |
| 07:51.13 | Math` | maybe they have a cheaper wholesale servic |
| 07:51.26 | Math` | stanaphone.com? |
| 07:51.32 | hypa7ia | check the pull-down |
| 07:51.44 | Math` | 0.0250$ for canada... |
| 07:51.49 | hypa7ia | ya |
| 07:51.53 | hypa7ia | that's whati said |
| 07:52.00 | *** join/#asterisk pooh_ (n=hfwang@cust.15.241.adsl.cistron.nl) |
| 07:52.14 | Math` | hypa7ia: you should switch to voxee or voipjet |
| 07:52.26 | Math` | (didnt try voxee... I know voipjet is good quality) |
| 07:52.28 | testmachine | YoMama: but you have a database i think |
| 07:52.28 | hypa7ia | meh, when i get unlazy |
| 07:52.28 | testmachine | ? |
| 07:52.36 | Math` | u'll save money :P |
| 07:52.42 | hypa7ia | the guy from voxee gave a great talk at astricon |
| 07:52.42 | Math` | that should get your ass up lol |
| 07:52.45 | delmar | I used to be on voipjet |
| 07:52.50 | delmar | found someone better. |
| 07:52.53 | lehel | Math`: could u ar somebody give me a hint why an IAX2 virtual isn't workin'? i mean: he calls the other answer and they can talk few seconds.. after that sound is down [i checked his router config, but.. i dunno] pls.. |
| 07:52.54 | delmar | their supplier lol :P |
| 07:52.56 | Math` | delmar: who? |
| 07:52.56 | hypa7ia | not when i'm talking about saving $2/mo |
| 07:52.57 | Math` | lollll |
| 07:53.00 | hypa7ia | :p |
| 07:53.18 | delmar | having said that.. John and voipjet provides a fantastic service. |
| 07:53.27 | Math` | John being voxee? lol |
| 07:53.32 | delmar | very hard to find someone like him as dedicated to customer service |
| 07:53.39 | delmar | no.. john at voipjet |
| 07:53.42 | Math` | ah ok |
| 07:53.51 | Math` | who's their supplier? |
| 07:53.51 | hypa7ia | eeks |
| 07:53.58 | YoMama | Math: where are u in Canada? |
| 07:54.02 | hypa7ia | i think my breezy is messed up :/ |
| 07:54.15 | Math` | YoMama: Montreal, Quebec |
| 07:54.16 | delmar | Math`, im in the business of making money, not making competition :P |
| 07:54.27 | Math` | delmar: so am I :P |
| 07:54.28 | YoMama | Math: hehe...sorry..i think i ripped on Montreal earlier |
| 07:54.34 | Math` | lool |
| 07:54.47 | hypa7ia | especially the bell helpdesks there |
| 07:54.48 | hypa7ia | :p |
| 07:54.57 | YoMama | people sound relaly funny up there |
| 07:55.01 | *** join/#asterisk many (n=many@daheim.ukeer.de) |
| 07:55.10 | YoMama | i can understand french too but i can't understand them |
| 07:55.25 | Math` | lol |
| 07:55.39 | lehel | so why one's IAX2 call just muting?.. |
| 07:56.03 | YoMama | quebec city is interesting |
| 07:56.11 | hypa7ia | i love QC |
| 07:56.18 | delmar | Math`, they must be pretty high up the VoIP tree tho, begining of this year i was searching around hardcore.. came accross voipjet only a few days before i got responses to emails from other suppliers... so im testing about 3 all at once right.. and this main one says.. they need a 15min window for an outage... etc.. so down they go at the scheduled time... as well as the other two im testing LOL :P |
| 07:56.19 | YoMama | the farther north u go..the less they wanna see out of towners though |
| 07:56.22 | Math` | quebec city is nice too |
| 07:56.29 | Math` | lollllll |
| 07:56.40 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 07:57.13 | Math` | delmar: who are they? :P |
| 07:57.26 | delmar | Math` :P |
| 07:57.47 | delmar | Math`, as i say.. im not in the business if creating competition for myself lol :P |
| 07:57.50 | Math` | you're a reseller? |
| 07:57.53 | Math` | whats your company |
| 07:58.14 | delmar | currently it's private and more under construction :P |
| 07:58.27 | delmar | but my server is up, and I have connections to it. |
| 07:58.34 | Math` | well as I said earlier, I'm looking for DOD providers |
| 07:58.38 | delmar | but wont be getting anything connected outside of NZ |
| 07:58.53 | Math` | so if you can equal 0.010$/min, I'm with you |
| 07:59.02 | Math` | ah you're in NZ |
| 07:59.08 | delmar | yep. |
| 07:59.15 | delmar | no point in off-shore connecting to here |
| 07:59.18 | Math` | voipjet's on peer1, so is my colocated server |
| 07:59.28 | Math` | so.... 100mbps to voipjet :) |
| 07:59.54 | delmar | i can get a rack at my supplier if I want.. who is voipjet's supplier :P |
| 08:00.01 | delmar | thats the next step |
| 08:00.07 | delmar | when i get some funds |
| 08:00.21 | Math` | you mean a 1U or the whole rack? lol |
| 08:00.36 | delmar | either :P |
| 08:00.43 | delmar | but year.. a few U :P |
| 08:01.04 | delmar | so anyway.. i just told TigerNetcrap to jam their phones up their arse. |
| 08:01.20 | delmar | this PA168X stuff is kids shit. |
| 08:01.26 | lehel | if i have a SIP virtual i'm not able to call through an IAX2 trunk? |
| 08:01.45 | Math` | wow: form voxee: All calls to destinations not listed in our rate sheet are |
| 08:01.45 | Math` | charged at US$10 per minute |
| 08:02.10 | YoMama | $10 per minute?!?!? |
| 08:02.13 | delmar | im gonna stick 5v up one and give it to my 10month old daughter to bash. ill smash a few more.. and send the bits back to china. i might even do a curley grogan in it too. imagine that when they open it a few weeks later. |
| 08:02.21 | Math` | YoMama: guess they do that cuz of voip fraud |
| 08:02.41 | YoMama | Math: voip fraud? what is that? |
| 08:02.43 | Math` | voipjet bills 1$/min for unlisted prefixes |
| 08:02.47 | YoMama | there are phone sex lines that are less than that |
| 08:02.53 | Math` | lol |
| 08:03.00 | YoMama | oh....the NANP "bad" area codes...right? |
| 08:03.17 | hypa7ia | yeah |
| 08:03.23 | hypa7ia | and eastern europe |
| 08:03.24 | Math` | YoMama: uhm, you open up a cell-phone carrier in some third-world country and you charge like 5$ a minute while your voip provider lists 0.10$/min |
| 08:03.25 | hypa7ia | etc |
| 08:03.35 | Math` | then you make a lot of calls. |
| 08:03.39 | Math` | guess the rest :P |
| 08:03.47 | YoMama | Math: ah |
| 08:04.12 | delmar | sounds like something Sahil Gupta would do. |
| 08:04.19 | Math` | who the hell is that? |
| 08:04.28 | delmar | google is your friend :P |
| 08:04.42 | Math` | lol |
| 08:04.49 | Math` | ah btw... for NANPA, here's a nice site: http://members.dandy.net/~czg/lca_index.php |
| 08:05.35 | delmar | Math`, seriously.. google the name .. for voip fraud and stuff.. here in NZ.. he abused callerID spoofing and "hacked" (no u cant even call it that) into peoples VoiceMail |
| 08:05.48 | Math` | lol |
| 08:06.05 | Math` | callerid spoofing is fun to make as a joke.... but... voicemail really use CID as an authentication method? |
| 08:06.06 | *** join/#asterisk lenne_dk (n=mirc@83.72.129.7.ip.tele2adsl.dk) |
| 08:06.16 | delmar | I was speaking with him like.. 2 weeks before it happened.. and we were talking about callerID... and how nice it was ppl could set it to a real phone number of their.. cell phone... whatever.... |
| 08:06.23 | delmar | 2 weeks later.. its in the news... |
| 08:06.30 | delmar | 2 days later.. i had him arrested. |
| 08:06.35 | Math` | like... if someone watches the ring in here I wont hesitate to make the phone ring with 1-666-666-6666 and make a Playback() that says "7 days" to joke someone lol |
| 08:06.50 | Math` | lol |
| 08:07.09 | snitt | lol |
| 08:07.11 | delmar | lol sounds like fun. |
| 08:07.27 | delmar | yeah i have had my CID set to all 6's before for a laugh... |
| 08:07.37 | Math` | delmar: I just make it locally |
| 08:07.39 | delmar | and i mean.. as long as its just some mucking around i think thats fine... |
| 08:07.41 | delmar | yeah |
| 08:07.46 | Math` | I've a devil.sh script to initiate that |
| 08:07.49 | YoMama | delmar: i like to set mine to my friend's cell phone and hten call him on it |
| 08:08.05 | delmar | but this idiot was on the war path... caused massive amount of trouble |
| 08:08.05 | Math` | ahhh nice that site allows you to list local calling prefixes giving a rate center :D |
| 08:08.10 | Math` | anywhere in NANPA |
| 08:08.57 | delmar | YoMama, or ring your mates cell phone from HIS callerID when hes out at work... and have hot sexy noises of some chick (his wife) getting it good :P |
| 08:09.18 | Math` | lol |
| 08:09.23 | Math` | u can do that with voipjet |
| 08:09.41 | delmar | sure, u can do that with most providers but its a breach of the T&C. |
| 08:10.00 | Math` | uhm I wonder.... how are major telcos connecting together |
| 08:10.04 | delmar | personally single/end-users should have to have their callerID set and confirmed/verified by the provider.... |
| 08:10.13 | Math` | there must be some kind of ASN/BGP for phones |
| 08:10.32 | Math` | delmar: ah my provider requires me to bring a legal document to change my CID |
| 08:10.40 | delmar | Math`, well, what the little fuckwit Sahil Gupta has done is what I feared... he caused TelecomNZ to get the shits, and they have now blocked international in-bound callerID. |
| 08:10.48 | Math` | but voipjet is intended for providers |
| 08:10.54 | syle | i'm going braindead |
| 08:11.06 | Math` | hey hi syle |
| 08:11.11 | syle | math how do you copy an integer into a string again in c |
| 08:11.23 | syle | strcpy bitching at me lol |
| 08:11.24 | Math` | I use snprintf |
| 08:11.28 | delmar | syle, thats an improvement tho isnt it? |
| 08:11.28 | syle | right thx |
| 08:11.48 | delmar | braindead.... |
| 08:11.53 | delmar | oh speaking of which... |
| 08:11.59 | delmar | its friday night.. i should be off drinking. |
| 08:12.01 | delmar | see ya |
| 08:12.11 | Math` | lol |
| 08:12.28 | Math` | still... I wonder, how are big telcos interconnected |
| 08:12.30 | lenne_dk | I have trouble with the dialplan. I turn logging on, and the first entry I se is -- Executing Goto("SIP/36946139-f3fc", "leifind|s|1") in new stack |
| 08:12.39 | Math` | thats normal |
| 08:12.46 | Math` | can you pastebin everything that you see? |
| 08:12.47 | Math` | ~pastebin |
| 08:12.49 | jbot | it has been said that pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 08:13.05 | Math` | delmar: er its thursday night |
| 08:13.08 | Math` | friday is tomorrow |
| 08:13.11 | Math` | heh |
| 08:13.14 | *** join/#asterisk nvrs (i=RUR@toronto-HSE-ppp4256323.sympatico.ca) |
| 08:13.31 | lenne_dk | I can see the entry in the dialplan, but I can't see why it goes there. |
| 08:13.47 | lenne_dk | I'd like to see the incoming call |
| 08:14.15 | lenne_dk | Like "Call to 1234@my.host content default |
| 08:14.36 | lenne_dk | I have 4 numbers registered |
| 08:14.53 | lenne_dk | 2 goes directly to an extension |
| 08:14.57 | YoMama | delmar: it's thursday night |
| 08:15.03 | lenne_dk | 1 goes to the main menu |
| 08:15.07 | *** join/#asterisk ComputerWarm (n=dan@rddrpx38-port-68.dial.telus.net) |
| 08:15.18 | lenne_dk | the last should also, but it doesn't |
| 08:15.37 | lenne_dk | Is there a debug switch I can set? |
| 08:16.03 | lenne_dk | sip debug gives much to much data |
| 08:16.30 | *** join/#asterisk Pikoro (n=pikoro@db.sunny-net.ne.jp) |
| 08:16.43 | delmar | YoMama, no its 9:20pm Friday, WAY past Bourbon O'Clock.... right.. im outta here. |
| 08:16.54 | Pikoro | any idea how to get multiple outbound sip trunks to work in like round robin fashion? |
| 08:17.04 | YoMama | delmar: where do u live? |
| 08:17.07 | delmar | NZ |
| 08:17.07 | delmar | l8r |
| 08:17.10 | YoMama | ah |
| 08:17.14 | Math` | delmar 1 thing |
| 08:17.16 | YoMama | always wanted to visit there |
| 08:17.19 | delmar | quick.. |
| 08:17.21 | Math` | its not friday night yet :P |
| 08:17.25 | delmar | u have until i hit send on this email :p |
| 08:17.26 | YoMama | bourbon o'clock |
| 08:17.27 | YoMama | hahaha |
| 08:17.33 | Math` | ah maybe in NZ |
| 08:17.34 | Pikoro | Math`, is friday night here |
| 08:17.35 | Math` | whats your TZ |
| 08:17.41 | YoMama | drinky drinky |
| 08:17.45 | Math` | nvm, go drink and have fun |
| 08:17.46 | delmar | we are like.. GMT+12 or GMT+13 |
| 08:17.52 | Pikoro | and i'm in japan |
| 08:17.53 | delmar | eh |
| 08:17.55 | delmar | lol. |
| 08:17.55 | delmar | l8r |
| 08:17.59 | Math` | here its GMT-5 |
| 08:17.59 | Math` | heh |
| 08:18.00 | YoMama | he's sittin' on the other side of the date line |
| 08:18.16 | lenne_dk | http://pastebin.ca/28345 |
| 08:18.18 | Math` | yeah I was like... er, 1. we're thursday, 2. its past 3am lol |
| 08:18.18 | Pikoro | so anyone tried multiple outbound trunk rotation? |
| 08:18.42 | YoMama | Pikoro: it's WAY too early to be askin' stuff like that |
| 08:18.48 | Pikoro | only 5pm here :D |
| 08:18.54 | Math` | Think I'll write an application which gets the rate list from multiple DOD providers and picks up the cheapest one |
| 08:19.02 | YoMama | Math: LCR |
| 08:19.19 | Math` | url? |
| 08:19.26 | Math` | cuz google is giving me a list to a communist league site |
| 08:19.26 | Math` | lol |
| 08:19.27 | YoMama | Math: Least Cost Routing |
| 08:19.36 | syle | sunsaturn:/etc/asterisk/sunsaturn-module# cat sunsaturn.c|wc -l |
| 08:19.37 | syle | <PROTECTED> |
| 08:19.37 | syle | sunsaturn:/etc/asterisk/sunsaturn-module# |
| 08:19.43 | Math` | hehe |
| 08:19.45 | Pikoro | i got 3 sip trunks i need to have like, fail over.. so 2 people can make outbound calls at the same time |
| 08:19.45 | YoMama | commies are cool |
| 08:19.46 | syle | god |
| 08:19.59 | Math` | syle: now strip empty lines from that :P |
| 08:20.02 | YoMama | Pikoro: use a group or something |
| 08:20.18 | syle | how do you do that |
| 08:20.24 | Pikoro | a group of outbound trunks and point the users at the group? |
| 08:20.35 | syle | without actually doing it :) |
| 08:20.37 | YoMama | Pikoro: yes |
| 08:21.05 | Pikoro | hmm... ok.. i"ll look that up.. want to point me at a config file so i can start my search? |
| 08:21.17 | lenne_dk | Do I report compile faiures here, head fails on BSD, didn't a few days ago. Or just file it at bugs.digium.com |
| 08:21.21 | syle | well i might give a 100 lines to blank lines and comments |
| 08:21.29 | Math` | syle: er gotta make a regex heh |
| 08:21.53 | syle | i did run the coding standard on it recently |
| 08:21.54 | Math` | nevermind, 1613 is a cool number |
| 08:22.23 | syle | indent -i4 -ts4 -br -brs -cdw -cli0 -ce -nbfda -npcs -npsl foo.c |
| 08:22.28 | Math` | syle: I'll be implementing my ccache+distcc lab tomorrow :) |
| 08:23.00 | Math` | the xeon server builds asterisk in 3 minutes ( 1 cpu... no SMP in the kernel lol, its a dual ) |
| 08:23.06 | YoMama | Pickoro: try callgroup |
| 08:23.08 | Math` | using cache hits takes 40 seconds |
| 08:23.12 | Pikoro | thanks |
| 08:23.15 | Math` | YoMama: callgroup is for Pickup() |
| 08:23.25 | YoMama | Math: c'mon man..it's 3:24 |
| 08:23.29 | YoMama | umm |
| 08:23.35 | YoMama | i just read about this today |
| 08:23.58 | syle | idk if asterisk is even setup to take advantage of SMP during a make |
| 08:24.21 | Math` | yeah it works fine using make -j |
| 08:24.23 | Pikoro | hmm |
| 08:24.28 | Pikoro | i need this for outbound sip calls |
| 08:24.42 | Math` | you have 3 SIP trunks? |
| 08:24.49 | Pikoro | yah, will have 10 |
| 08:25.13 | Math` | anyone tried astbill and can give feedback? |
| 08:25.28 | Math` | Pikoro: damn... are they all equivalent or do you have some kind of routing logic |
| 08:25.29 | YoMama | wait |
| 08:25.34 | YoMama | uhhh |
| 08:25.45 | Pikoro | basically they're all the same |
| 08:25.45 | YoMama | Pikoro: just put one right aftger the other in priority |
| 08:25.59 | YoMama | if one is used...and it's busy it'll go to teh second one |
| 08:26.01 | Math` | 10 providers? |
| 08:26.12 | YoMama | until u hit the fourth call..u do a Congestion() |
| 08:26.18 | lenne_dk | Is that nessecary? I can make multipe calls to my VOIP-ISP on the same "line" |
| 08:26.27 | YoMama | if u want it to rotate..then u gotta do some incremeting math thing using variables and stuff |
| 08:26.38 | YoMama | but if u don't mind tghat it uses the first channel for the first call every single time |
| 08:26.42 | YoMama | then that's all u gotta do |
| 08:26.56 | Pikoro | what i did was like: |
| 08:26.56 | YoMama | pikoro: get it? |
| 08:27.07 | Pikoro | exten => _9.,1,Dial(SIP/${EXTEN:1}@trunk1,30,r) |
| 08:27.07 | Pikoro | exten => _9.,2,Dial(SIP/${EXTEN:1}@trunk2,30,r) |
| 08:27.15 | Pikoro | then make 1 oubound call |
| 08:27.23 | YoMama | k |
| 08:27.28 | Pikoro | when i try to make a seond outbound, it fails... like 486 busy here |
| 08:27.49 | YoMama | @trunk2 |
| 08:28.10 | YoMama | umm |
| 08:28.20 | Math` | I wonder how dundi handles multiple routes to the same destination |
| 08:28.53 | Pikoro | is that wrong? |
| 08:29.05 | lenne_dk | Doesn't the busy jump 100, so the next exten should be 101? |
| 08:29.06 | YoMama | no..i'm trying to think and my brainis already off |
| 08:29.08 | Pikoro | well, i assume it is cause it doesn't work |
| 08:29.09 | Pikoro | heh |
| 08:29.24 | YoMama | lenne: right |
| 08:29.43 | Pikoro | ? |
| 08:29.43 | YoMama | Pikoro: it only goes to 2...u gotta do 102 |
| 08:29.45 | YoMama | hold on |
| 08:30.05 | Pikoro | oh, for the priority |
| 08:30.08 | shanky | anyone knows is the T1 wiring is the same for an E1? |
| 08:30.14 | Pikoro | what about n? |
| 08:30.32 | syle | more wires on a E1 |
| 08:30.44 | Math` | syle: what are you working on already? |
| 08:30.58 | syle | i'm working on the dialplan application now |
| 08:30.59 | lenne_dk | But what if the called is really busy. Then you'll make 3 or 10 calls, all busy |
| 08:31.00 | YoMama | Pickoro: yeah |
| 08:31.09 | Math` | syle: which will do? |
| 08:31.11 | YoMama | Pickoro: u need to make it 102 |
| 08:31.25 | YoMama | Pikoro: if 1 fails 'cause 1 is busy..it'll go to 102 |
| 08:31.28 | Pikoro | ok, so first trunk is 1, second is 102, third is 203,, etc... |
| 08:31.34 | YoMama | right |
| 08:31.43 | syle | returns seconds_left allowed for caller on pre-paid |
| 08:31.59 | Pikoro | ok... and how about timeouts? will they have to wait 30 seconds for the first trunk to timeout before it goes to the second? |
| 08:32.00 | Math` | ok |
| 08:32.06 | syle | Sunsaturn(blah); in your dialplan sets ss_seconds |
| 08:32.23 | Math` | ok |
| 08:32.36 | Math` | Pikoro: don't forget to set the 'j' option |
| 08:32.38 | lenne_dk | If it's busy it'll jump as soon as * knows it |
| 08:32.51 | Pikoro | and only one last quick question.... can I use s instead of _9. so that nobody has to dial 9 to get out? |
| 08:32.52 | Math` | as jumping to +101 will be deprecated soon, the j option is going to force this behavior |
| 08:33.03 | Pikoro | rj? |
| 08:33.04 | Pikoro | or just j |
| 08:33.09 | syle | j never worked for me wheni tried it |
| 08:33.12 | YoMama | Pikoro: s??? |
| 08:33.23 | Math` | <PROTECTED> |
| 08:33.30 | Math` | rj |
| 08:33.30 | Pikoro | well i don't to have to dial 9 to get out |
| 08:33.41 | Math` | no but you can use _X.,1,Dial(...) |
| 08:33.42 | YoMama | pikoro: if u don't want them to dial 9...then don'tput the 9 |
| 08:33.49 | Pikoro | heh.. ok |
| 08:33.52 | YoMama | Math: that's a real bad idea |
| 08:33.57 | YoMama | that matches anything |
| 08:34.10 | Math` | except stuff that starts by * # A B C D |
| 08:34.14 | Math` | thats what he wants :P |
| 08:34.15 | lenne_dk | Is it possible to see why it's busy? if it's the VOIP that's denying it, or if the called? |
| 08:34.28 | lenne_dk | Some returned variable? |
| 08:34.30 | Math` | DIALSTATUS |
| 08:34.33 | YoMama | Math: insane in the membrane |
| 08:34.37 | ComputerWarm | Anyone know where i can find a list of all Country codes/ City codes? |
| 08:34.44 | shanky | syle: do you know if the wiring is as: 1-4,2-5,3-3,4-1,5-2,6-6,7-7,8-8 ? |
| 08:35.31 | lenne_dk | It is not needed to have a 9 to dial out. |
| 08:35.31 | Pikoro | ok.. gonna test.. brb |
| 08:35.32 | YoMama | ComputerWarm: do a google search for it..it'll take u 20 sweconds to fidn it |
| 08:36.07 | ComputerWarm | YoMama oh ok thanks |
| 08:36.13 | lenne_dk | I have extension 5, 55 and 555, and anything with more than 7 digits is an external call |
| 08:36.49 | YoMama | _NXXXXX. |
| 08:36.58 | lenne_dk | Just have to dial in 4 seconds, then * or the phone thinks I'm done. |
| 08:36.58 | YoMama | 7 digits or more |
| 08:37.30 | YoMama | is teh second port on the grandstream for your 'puter? |
| 08:37.43 | Math` | lenne_dk: thats a phone-based setting |
| 08:37.53 | Math` | you can change the timeout value |
| 08:38.14 | syle | shanky I'm not from europe, i am sure in your country you can just buy a straight through or crossover E1 cable , or search google to make it yourself |
| 08:38.25 | Math` | my ATA is configured so that if I press #, it means I've finished dialig (and it strips off the # before sending the call) |
| 08:38.43 | lenne_dk | I might change it when Im getting geriatric :-) 4 secs is good enough |
| 08:38.55 | Math` | I just press # |
| 08:39.10 | Math` | (and I match NPA-7digits too) |
| 08:39.20 | YoMama | math: most phone companies in teh US recognize that as the end of dialing too for international |
| 08:39.27 | lenne_dk | Me too. Just wanted to point out it is not needed to "dial 9 for a line" |
| 08:39.30 | YoMama | 'cause there's no standard #of digits for international |
| 08:39.36 | YoMama | saves ya about 3-4 seconds |
| 08:40.04 | lenne_dk | Or press the "send" on the Grandstream BT |
| 08:40.27 | Math` | :) |
| 08:40.39 | Math` | YoMama: with that locally reachable prefix list, I'll writh smth to generate a dialplan automaticaly |
| 08:40.51 | YoMama | lenne: u got grandstream stuff? |
| 08:40.54 | Math` | syle: you tried astbill? |
| 08:41.12 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) |
| 08:41.20 | lenne_dk | Yep. HT-286 and BT101 |
| 08:41.28 | syle | i looked at it, seems just full of crap people don't need |
| 08:41.33 | YoMama | lenne: does the BT101 have dual ethernet? |
| 08:41.36 | lenne_dk | And a FritzBox fon |
| 08:41.44 | lenne_dk | no, that's BT102 |
| 08:41.51 | YoMama | lenne: what's the second port for? |
| 08:42.06 | YoMama | lenne: for a computer? is one a switch port and the other a computer port? |
| 08:42.12 | Math` | syle: you have any better Least cost routing solution? |
| 08:42.12 | YoMama | like the cisco's? |
| 08:42.29 | lenne_dk | YoMama: Yes. |
| 08:42.30 | syle | yeah http://software.sunsaturn.com when its complete :) |
| 08:42.35 | Math` | lol |
| 08:43.21 | lenne_dk | Or for residential use, put it first, connect it to the DSL, gets some priority to the voip, I guess. |
| 08:43.21 | lehel | why i don't have any output inmy CLI> if i'm trying to make a call from SIP (virtual) |
| 08:43.25 | YoMama | lenne: k |
| 08:43.36 | lehel | however i have verbosity.. debug.. |
| 08:43.55 | lehel | and i cannot make the call either |
| 08:44.43 | YoMama | k |
| 08:44.45 | YoMama | i'm going to bed |
| 08:44.48 | YoMama | goodnight |
| 08:45.01 | konfuzed | ;^( |
| 08:45.07 | lenne_dk | Got 5 extensions on *. Me, wife, kitchen, child |
| 08:45.19 | lenne_dk | And temperature in fridge |
| 08:45.20 | konfuzed | cat |
| 08:45.33 | Math` | wtf, an astbill live cd |
| 08:46.03 | lehel | Math`, do you know a2billing? |
| 08:46.12 | Math` | lehel? |
| 08:46.27 | Math` | Im just trying astbill for least cost routing and to see if it'd bit some clients |
| 08:46.32 | Math` | er customer I mean |
| 08:46.46 | Math` | I do not know a2billing no, but I'll look at it |
| 08:46.53 | lehel | http://www.voip-info.org/wiki/view/A2Billing |
| 08:46.59 | lehel | nice thing |
| 08:48.05 | Math` | lehel: did u try both? |
| 08:49.37 | syle | more slow software in php :) |
| 08:49.57 | lehel | i did try astbill some time ago, but didn't work out for me; a2billing trying now |
| 08:50.18 | Math` | syle: what do u suggest then? make a C application for lcr? |
| 08:50.26 | Math` | did you finish your LCR module for your solution? |
| 08:50.34 | syle | well thats what i'm proposing |
| 08:51.19 | Math` | its AGI anyways... it has to be slow |
| 08:51.25 | syle | i have a routing part of it complete yeah, i really don;t want to go into to much details on how powerful it is yet |
| 08:51.30 | syle | 2 weeks |
| 08:51.54 | ptiggerdine | tease |
| 08:53.30 | syle | i had a bitch of a time with the math algorithms for allowing people to do stuff like : |
| 08:54.05 | syle | rate 2 cents for first min of call 1 cent for next 5 min, .50 cents for remainder of call |
| 08:54.23 | syle | that was very difficult math but i got it working :) |
| 08:54.31 | Math` | nice stuff :) |
| 08:55.39 | syle | but define lcr |
| 08:55.44 | Math` | least cost routing |
| 08:55.45 | syle | what do you mean by it so i know |
| 08:55.47 | Math` | ok |
| 08:56.02 | Math` | I download the rates of different provider in csv format |
| 08:56.27 | *** join/#asterisk testmachine (n=assink@ip237-239-58-62.adsl.versatel.nl) |
| 08:56.29 | testmachine | hm |
| 08:56.41 | testmachine | wich sql daemon can i install as best? |
| 08:56.46 | Math` | I import those lists somewhere, and outgoing routes are checked upon the "big-list"(tm) to get the cheapest rate |
| 08:56.50 | lehel | a2billing has the rates included |
| 08:57.00 | Math` | are they up to date? |
| 08:57.06 | testmachine | i am doing a total reinstall |
| 08:57.07 | lehel | absolutely |
| 08:57.08 | testmachine | so |
| 08:57.20 | syle | testmachine: http://dev.mysql.com/downloads/mysql/5.0.html |
| 08:58.05 | testmachine | mysql |
| 08:58.06 | testmachine | ? |
| 08:58.12 | syle | math are these lists in the same format ? |
| 08:58.19 | testmachine | :) |
| 08:58.29 | Math` | syle: yeah but that doesnt really matter... I can convert from one format to another pretty easily |
| 08:58.37 | Math` | the best would be to store that in an *sql db |
| 08:58.59 | Math` | select .... order by rate asc limit 1 |
| 08:59.09 | Math` | selects the cheapest one |
| 08:59.19 | Math` | (based on the prefix) |
| 08:59.25 | lehel | why i can't make a SIP call from a virtual? no output either |
| 08:59.34 | Math` | from a virtual? |
| 08:59.37 | syle | but why not just update over the other one then? |
| 08:59.43 | lehel | soft-phone Math` |
| 09:00.05 | Math` | think I gotta use some sleep (4am) |
| 09:00.06 | pooh_ | anybody care to explain what the word 'prune' means exactly pls ? |
| 09:00.19 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 09:00.22 | syle | or are you using multiple routes |
| 09:00.34 | lehel | pooh_ it's a fruit |
| 09:00.39 | pooh_ | lehel: :-) |
| 09:00.43 | lehel | :P |
| 09:00.45 | Math` | syle: I just want lets say to put voxee's rates and voipjet's rates and use the cheapest for where I want to call |
| 09:01.06 | Math` | tought this doesn't really matter for the beginning since its a flat rate for CA+US |
| 09:01.21 | pooh_ | e.g. Sip prune relatime |
| 09:01.41 | syle | so you want interface to support importing their formats then automatically? |
| 09:01.53 | lehel | pooh_, and "to prune" means: to clip, to pare, to trim, to shear |
| 09:02.07 | Math` | yeah, but as long as your engine supports choosing the right one, I can manage changing the formats |
| 09:02.12 | pooh_ | lehel: hmm, so what does sip prune realtime mean pls? |
| 09:02.22 | Math` | like... voxee gives an excel file, I doubt your gonna open it |
| 09:03.28 | lehel | pooh_: maybe you can find out: http://bugs.digium.com/view.php?id=3509 |
| 09:03.47 | lehel | pooh_: the docs for 'iax2 prune realtime' say that it takes a username, but it only handles peers (and the completion function will only return peer names) |
| 09:05.25 | pooh_ | lehel: thx |
| 09:05.34 | lehel | np, pooh_ |
| 09:09.53 | *** join/#asterisk cjk (n=cjk@80.92.64.103) |
| 09:13.53 | lenne_dk | I'd like to read telephone numbers in groups of two digits. 12345678 as twelve thirtyfour fiftysix seventyeight. (Danish standard) |
| 09:14.25 | lenne_dk | Should I make it a new function in say.c? |
| 09:14.31 | shanky | mmm, lenne_dk do you know telsome? |
| 09:14.54 | lenne_dk | telsome.dk voip-supplyer? |
| 09:15.08 | shanky | lenne_dk: yes |
| 09:15.25 | lenne_dk | Of name, but I'm on another voisp |
| 09:16.04 | syle | lenne if your gonna do that let people specify amount of chars in the array to grab |
| 09:16.23 | shanky | amm, ok, it's just because we are talking to them to use it as voip-supplier as soon as they install it platform in Spain |
| 09:16.45 | syle | actually you can do that already |
| 09:17.00 | lenne_dk | syle: Perhaps a format? How already? |
| 09:17.04 | *** join/#asterisk p1tst0p (n=will@82-38-106-54.cable.ubr03.donc.blueyonder.co.uk) |
| 09:17.05 | syle | ${variable}:0:2 |
| 09:17.06 | syle | etc |
| 09:17.28 | *** join/#asterisk tobiasWolf (n=konversa@195.162.255.10) |
| 09:17.52 | syle | i mean |
| 09:17.52 | lenne_dk | Ok, but I rather have it as a new function, so the voicemail can use it too. |
| 09:18.02 | syle | ${variable:0:2} |
| 09:18.03 | syle | etc |
| 09:18.18 | syle | in your case |
| 09:18.20 | syle | ${variable:0:2} |
| 09:18.20 | p1tst0p | hi, i am using a Avaya IP phone with my asterisk, its has a small 2 lign LCD display, which displays caller ID etc. I have set a few queue's up, is there a way for asterisk to display a queue name on my phones LCD ? |
| 09:18.26 | syle | ${variable:2:4} |
| 09:19.22 | syle | actually no for lenne it would be ${variable:0:2} ${variable:2:2} ${variable:4:2} etc |
| 09:19.28 | p1tst0p | ill add to that, the model is 4602 |
| 09:20.21 | lenne_dk | How's us numbers pronounced? 555-1234-5678 five five five pause one two three four pause five six seven eight or five five five pause twelve thirtyfour pause fiftysix seventyeight? |
| 09:21.38 | *** join/#asterisk Bonzai009 (n=Bonzai00@wbs-146-190-189.telkomadsl.co.za) |
| 09:21.45 | Bonzai009 | hi all |
| 09:21.59 | *** join/#asterisk newmember (n=newmembe@S010600a0c93dce87.cg.shawcable.net) |
| 09:22.08 | Bonzai009 | i have a amd opteron 4 way 2.6ghz server |
| 09:22.13 | Bonzai009 | with 8gb memory |
| 09:22.26 | Bonzai009 | how many concurrent voip calls can it handle?// |
| 09:22.36 | Bonzai009 | is there a app to test the call load |
| 09:22.37 | many | "depends" |
| 09:23.02 | Bonzai009 | many running x86_64 |
| 09:23.08 | Bonzai009 | fedora core 4 |
| 09:23.19 | lenne_dk | Depends on codecs and if * needs to transcode. |
| 09:23.19 | many | It still depends. |
| 09:23.37 | Bonzai009 | hmm i'll be using gsm here as bandwidth is at a premium |
| 09:23.54 | many | see if you do sip<->sip with same codecs itll take much less load as if it needs to do pstn<->sip for example. |
| 09:24.16 | Bonzai009 | many i'll only do iax to iax and use gsm codec |
| 09:24.26 | Bonzai009 | pstn to iax as well |
| 09:24.45 | pooh_ | anybody familiar with sip_notify.conf ? |
| 09:25.00 | lenne_dk | Set up the phones to use gsm then |
| 09:25.35 | pooh_ | Bonzai009: that way it should take a couple of hudred |
| 09:25.55 | Bonzai009 | hmm pooh_ |
| 09:26.10 | Bonzai009 | only couple hundred ?.. |
| 09:26.23 | pooh_ | using 1 codec for both peers should make them natively bridge |
| 09:26.42 | pooh_ | dunno about the all together usage of bandwidth tho |
| 09:26.42 | many | well, iax<->iax with restriction of peers to gsm takes up almost no cpu at all. |
| 09:26.58 | Igbothom_III | and less bandwidth! :) |
| 09:27.00 | pooh_ | bandwidth will be the biggest issue |
| 09:27.10 | pooh_ | Igbothom_III: morning |
| 09:27.15 | Bonzai009 | dunno i have a 2mb line but i wanna know how much the server it self can handle before the cpu's etc starts taking strain |
| 09:27.16 | many | i assume so, too. |
| 09:27.18 | Igbothom_III | g'day pooh_ |
| 09:27.23 | many | bonzay: 2M? |
| 09:27.30 | many | uhm. |
| 09:27.57 | many | a GSM codec takes how much bandwidth? 10kbit? |
| 09:28.49 | Bonzai009 | many but i have a quad pri digium card so calls comming into the * system will be from the pstn |
| 09:28.55 | Igbothom_III | 8 kbps each way |
| 09:29.00 | Igbothom_III | plus overhead |
| 09:29.26 | Bonzai009 | hmm so we looking at about what 16kbps per call ?.. |
| 09:29.54 | Igbothom_III | prolly aboot 24 Kbps in total, I'd guesstimate |
| 09:30.31 | Bonzai009 | Igbothom_III, if you look @ a 8kbps with out header and footer you can add on 4 kbps for header and footer |
| 09:30.37 | Igbothom_III | if you calculate at 16 kbps each way, then you'd have enough headroom |
| 09:30.54 | many | so a 2M is 2048kbit, with 24kbps i get... 85 calls simult. |
| 09:31.03 | Igbothom_III | yes, that makes 12 kbps each way, or 24 kbps - as I said :) |
| 09:31.07 | many | well, atleast if i understood riese. :) |
| 09:31.12 | Bonzai009 | Igbothom_III, hmm that should be ok then hehehe :> |
| 09:31.22 | Bonzai009 | thanks many Igbothom_III |
| 09:31.30 | Bonzai009 | now to stress test my optron box |
| 09:31.34 | clive- | bonzai allow more than less bandwidth |
| 09:31.34 | Igbothom_III | np - my figures ain't gospel, tho |
| 09:33.15 | Igbothom_III | bandwidth is like blowjobs - you can never have too much |
| 09:33.38 | syle | more like lucky to get any |
| 09:34.09 | RoyK | argh |
| 09:34.11 | RoyK | <PROTECTED> |
| 09:34.11 | RoyK | <PROTECTED> |
| 09:34.12 | Igbothom_III | those two answers are not mutually exclusive :) |
| 09:34.17 | RoyK | these spam my console |
| 09:34.58 | sjobeck99 | royk: Remote UNIX connection = you might have two safe_asterisk's running |
| 09:35.26 | RoyK | sjobeck99: thanks :) that was it |
| 09:35.41 | sjobeck99 | royk: 99.9 is the case |
| 09:36.11 | RoyK | there should be a lock/check in safe_asterisk |
| 09:36.20 | sjobeck99 | perhaps |
| 09:36.25 | RoyK | probably |
| 09:36.32 | sjobeck99 | sure |
| 09:36.38 | RoyK | absolutely |
| 09:37.44 | *** join/#asterisk p1tst0p (n=will@82-38-106-54.cable.ubr03.donc.blueyonder.co.uk) |
| 09:38.08 | many | this is the lock check. |
| 09:38.30 | p1tst0p | how can i tell if my channel to my sip provider ( sipgate.co.uk ) is using the g.629 codec i just registered ? |
| 09:40.06 | testmachine | so i should use just mysql |
| 09:40.08 | pooh_ | hmmm.... |
| 09:40.09 | pooh_ | Beginning asterisk shutdown.... |
| 09:40.09 | pooh_ | Executing last minute cleanups |
| 09:40.09 | pooh_ | <PROTECTED> |
| 09:40.09 | pooh_ | Nov 11 10:13:29 DEBUG[7199]: res_musiconhold.c:1066 ast_moh_destroy: killing 7267! |
| 09:40.10 | pooh_ | Nov 11 10:13:29 DEBUG[7199]: res_musiconhold.c:1074 ast_moh_destroy: mpg123 pid 7267 and child died after 4160 bytes read |
| 09:40.11 | testmachine | daemon? |
| 09:40.11 | pooh_ | Asterisk cleanly ending (15). |
| 09:40.19 | p1tst0p | the documentation said, 'show g729' but that doesn't seem to work, |
| 09:40.20 | pooh_ | * died on me |
| 09:41.09 | pooh_ | p1tst0p: during a call do a show channels |
| 09:41.40 | pooh_ | p1tst0p: do a show translation to see if the codec is loaded/activated at all |
| 09:41.55 | syle | p1tstop : i think you mean "show translation" |
| 09:42.36 | syle | if you don't see g729 in there then you did something wrong |
| 09:44.54 | pooh_ | p1tst0p: when the codec in show traslation is all set to '-' Then something is wrong |
| 09:45.02 | pooh_ | p1tst0p: it will show tho |
| 09:47.02 | lehel | pooh_: what * ver do you use? |
| 09:47.35 | Igbothom_III | rc1 |
| 09:47.36 | p1tst0p | hmm if i do show translation, and follow g729 on the left, across to g729 on the top, that has '-' but every other has a number |
| 09:48.09 | syle | then it works |
| 09:48.11 | syle | what is the problem? |
| 09:49.14 | p1tst0p | syle, there is no problem i wanted to check if it was working on my chanel, i expected somewhere to be able to see what codec a channel. |
| 09:49.23 | p1tst0p | *was using |
| 09:50.12 | syle | umm just start asterisk in verbose mode and make a call |
| 09:50.16 | p1tst0p | show channels just shows, Channel Location State Application(Data) |
| 09:50.20 | testmachine | Can't connect to MySQL server on 'server.sigmasoft.com' (110) |
| 09:50.26 | testmachine | it has to connect with localhost |
| 09:50.32 | testmachine | where can i change it? |
| 09:50.42 | syle | #mysql |
| 09:50.52 | testmachine | syle: no |
| 09:51.02 | lehel | p1tst0p: sip/zap/iax2/ show channels |
| 09:51.06 | testmachine | syle: its an asterisk message |
| 09:51.18 | syle | in other words.../join #mysql |
| 09:51.26 | testmachine | asterisk wants to connect with sigmasoft.com |
| 09:51.30 | testmachine | instead of localhost |
| 09:51.55 | lehel | testmachine: look in cdr_mysql.conf |
| 09:52.13 | testmachine | i dont have this file.... |
| 09:52.24 | lehel | oh ;) that's a problem |
| 09:52.28 | syle | hes trying to setup mysql not even do anything with asterisk yet |
| 09:52.58 | lehel | oh.. then i shut up;) srr |
| 09:53.00 | testmachine | syle: somewhere i have to say let asterisk connect with mysql on localhost |
| 09:53.08 | testmachine | and not on sigmasoft.com |
| 09:53.25 | p1tst0p | lehel, is it the Form column ? |
| 09:53.46 | syle | testmachine i suggest you learn abit about databases |
| 09:54.23 | syle | that error could mean many things, maybe its not running, maybe you haven;t allowed localhost to connect from localhost in grant tables etc |
| 09:54.23 | testmachine | syle: i know something about databases, not much but ok |
| 09:54.35 | lehel | testmachine u installed asterisk-addons? |
| 09:54.55 | testmachine | lehel: i did an ubuntu install |
| 09:55.02 | testmachine | apt-get install asterisk |
| 09:55.02 | lehel | 'coz there it is: /usr/src/asterisk-addons/configs/cdr_mysql.conf.sample |
| 09:56.07 | syle | well then why don;t you try apt-get install mysql-server |
| 09:56.21 | *** join/#asterisk A-Tuin|work (n=A-Tuin@nat.office.legend.net.uk) |
| 09:56.35 | testmachine | syle: mysql is running |
| 09:56.41 | testmachine | root@testserver:/etc/asterisk # /etc/init.d/mysql start |
| 09:56.42 | testmachine | Starting MySQL database server: mysqld...already running. |
| 09:57.04 | syle | so what happens when your type "mysql" |
| 09:57.28 | testmachine | i come in the mysql console |
| 09:57.35 | syle | so whats the problem |
| 09:57.49 | testmachine | asterisk doesnt connect with mysql on localhost |
| 09:57.52 | lehel | syle: his asterisk isn't connecting in the right place.. |
| 09:57.53 | many | grep sigma /etc/asterisk/* |
| 09:57.56 | testmachine | but on sigmasoft.com |
| 09:58.19 | testmachine | in the rate_engine |
| 09:58.20 | testmachine | :) |
| 09:58.25 | testmachine | thanks many |
| 09:59.26 | many | report a bug against bts if it shouldnt be so (i dont know rate_Engine) |
| 10:02.36 | joelsolanki | does any body know ..how to disable music on hold ? when i call any number and keep the phone off then that phone rings back and on asterisk console it tells me music on hold ..what does that mean |
| 10:03.32 | syle | --/etc/asterisk/musiconhold.conf |
| 10:04.52 | joelsolanki | ok let me check |
| 10:05.00 | syle | i'm guessing it means you didn;t call hangup in your dialplan properly |
| 10:05.15 | cjk | does anyone know how i can register with sipsak? |
| 10:05.18 | *** join/#asterisk szw2001 (n=vip@218.1.209.67) |
| 10:06.10 | joelsolanki | syle: so it is related to extensions? |
| 10:07.01 | joelsolanki | exten => _X.,1,Dial(SIP/${EXTEN}@isphone,30) i have only this line. |
| 10:07.30 | syle | exten => _X.,2, Hangup |
| 10:07.45 | joelsolanki | exten => _X,2,Hangup |
| 10:07.57 | joelsolanki | hmm yes i was going to try that :) |
| 10:09.05 | lehel | joelsolanki: _X. < with the dot |
| 10:09.13 | joelsolanki | yes |
| 10:15.48 | *** join/#asterisk duckz (n=duckz@193.192.46.26) |
| 10:16.40 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 10:16.52 | p1tst0p | Hi, im sorry to be a pain, but i cant see where it specifiy's if my connection to sipgate from asterisk is using the g.729 codec i just registered.. is it the Form column in, sip show channels that specify's the codec used ? |
| 10:29.06 | mistral | is this the correct syntax for Dial ? Dial(SIP/sipname/${EXTEN:1),60,tr) ? where sipname is the name in sip.conf |
| 10:30.48 | syle | what |
| 10:30.55 | syle | you met her? |
| 10:31.28 | lehel | mistral: "sipname is the name in sip.conf" what name? |
| 10:31.55 | *** join/#asterisk Aze` (n=aze@85-18-136-114.ip.fastwebnet.it) |
| 10:32.25 | mistral | well theres a few of them so lets call it draytel |
| 10:33.21 | lehel | p1tst0p: sip show channel <channel> |
| 10:34.27 | lehel | mistral: or "sipname" or ${EXTEN}.. i think |
| 10:36.03 | lenne_dk | cvs question: I want to checkout a previous version of a single file |
| 10:36.12 | lenne_dk | cvs checkout -r 1.374 channels/chan_iax2.c |
| 10:36.23 | lenne_dk | What's the right syntax? |
| 10:37.59 | lenne_dk | cvs -d :pserver:anoncvs@cvs.digium.com:/usr/cvsroot checkout -r 1.374 asterisk/channels/chan_iax2.c |
| 10:38.11 | lenne_dk | was right. Sorry for disturbing :-) |
| 10:39.21 | Aze` | Anyone has patch for polarity hangup (bug 2847) TDM400P for asterisk 1.0.9 ? |
| 10:40.52 | p1tst0p | lehel, so, sip show channel then the channel ( thats active ) which part, is telling me the codec ? is it Format, if so mine says 'ulaw' |
| 10:41.34 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 10:42.05 | lehel | p1tst0p, what ver of * do you use? |
| 10:42.24 | Igbothom_III | syle; no, I'd not WANT to met her - can't stand the crap she comes out with |
| 10:42.59 | Igbothom_III | I just managed to have a dig at her while spewing about the Sony Rootkit Copy Protection |
| 10:43.10 | Igbothom_III | Celine is married to Mr Pure Evil |
| 10:43.23 | Igbothom_III | whatever he is in your religious fantasies |
| 10:48.13 | syle | ummm |
| 10:49.43 | syle | you realize she was the youngest of 14 children, and dated older guys so that she could afford to get where she is today, think guy mortaged his house to pay for her music things |
| 10:50.24 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 10:51.11 | syle | she'll always be married to mr.evil(people with power and money) , she knows no other way now i think |
| 10:51.15 | Igbothom_III | so, she shagged her way to wherever it is she's at today? Now, that's something to be proud of - she's a common prostitute! |
| 10:51.53 | syle | well she did it a better way than pamela anderson and marilyn monroe |
| 10:52.01 | Igbothom_III | lol |
| 10:52.04 | Igbothom_III | :) |
| 10:52.32 | Igbothom_III | not that I have an issue with those who choose to work in the sex industry - it is a legitimate job serving a legitimate need |
| 10:53.01 | Igbothom_III | it predates our industry by a few thousand years |
| 10:53.10 | Igbothom_III | and has remained largely unchanged |
| 10:54.07 | syle | well beats jerking off to virtual reality :) |
| 10:54.14 | Igbothom_III | lol |
| 10:54.24 | Igbothom_III | if its not scratch 'n' sniff, I'm not interested |
| 10:58.03 | syle | i stay away from these issues, its its one day your the old fart that is 60 years old and starts banging some young girl, makes her into someone famous, why shoudl people critisize you hehe |
| 10:59.14 | Igbothom_III | they should immortalize you! |
| 11:02.20 | Igbothom_III | want to get REALLY SCARED? Have a listen to http://www.npr.org/templates/story/story.php?storyId=4989260, click on the "Listen" link, and concentrate from about 1 min 45 sec for the next 20 - 45 seconds |
| 11:06.57 | *** join/#asterisk ful|work (n=fulgas@209.8.233.197) |
| 11:07.50 | syle | listend |
| 11:07.57 | syle | yeah more bs |
| 11:08.59 | syle | can't beleive they would even try to do that |
| 11:10.24 | ful|work | hey |
| 11:12.22 | shanky | well, at the end, I have get working my E1 connection |
| 11:12.56 | shanky | now it's time to edit zapata.conf and so on |
| 11:13.49 | mistral | ok i give in how do i get something to dial out a trunk ? currnetly i have it starting a dial but it the sip provider seems to reject me. I have a dialplan like exten => _9.,1,Dial(SIP/${EXTEN:1}@provider,60,tr) the provider is in sip.conf with the user the secrect etc.. but it says wrong password on authentication for INVITE when i attempt to dial |
| 11:14.02 | mistral | oh yeah inbound works fine |
| 11:19.29 | Igbothom_III | syle; that kind of position from the President of Sony BMG's Global Digital Business is simply frightening |
| 11:21.01 | syle | yes a big company like that should know to play by the rules, but i imagine they have been getting pressure to from singers to do something about protecting their cd content |
| 11:21.44 | shanky | mmm, I can't get my inbounds calls to work: |
| 11:21.45 | shanky | Nov 11 12:24:56 NOTICE[26504]: app_dial.c:764 dial_exec: Unable to create channel of type 'SIP' |
| 11:23.46 | testmachine | woei |
| 11:23.50 | testmachine | no errors anymore |
| 11:24.13 | testmachine | :) |
| 11:29.02 | lehel | bv testmachine |
| 11:29.26 | testmachine | :) |
| 11:29.40 | testmachine | i did like tail -f the asterisk log file |
| 11:29.50 | Igbothom_III | syle; the artists are the ones in the pie who get to smell the food and taste very little of it - unless they are U2-sized. The record companies are the ones driving copy-protection, not the artists. The record companies make billions out of artists who struggle to make ends meet. Unfortunately (not) there are more and more artists joing smaller, boutique labels and/or releasing their music independently on CD or th |
| 11:29.50 | Igbothom_III | e Internet and taking money out of the mouths of the fatcat record company execs. I don't feel sorry in the slightest, having worked in the music industry for the past 20+ years |
| 11:29.54 | testmachine | and i saw all the cdr table fields coming by |
| 11:29.59 | testmachine | so i could add them all |
| 11:30.26 | Igbothom_III | oh, no - its a testie machine! |
| 11:32.41 | syle | i think many feel that way considering they lost alot of court cases trying to stop peer sharing |
| 11:33.14 | testmachine | yes its a test machine |
| 11:33.21 | testmachine | where i am installing asterisk on |
| 11:33.23 | testmachine | :P |
| 11:34.08 | syle | think i;ll go play dead or alive on xbox , need a break from coding |
| 11:34.41 | testmachine | hehe |
| 11:35.03 | lehel | syle: Dead or Alive Ultimate ? |
| 11:35.11 | syle | yeah, my favorite game |
| 11:35.22 | syle | can;t wait for dead or alive 4 with xbox 360 nov 22nd |
| 11:35.33 | lehel | cool indeed |
| 11:35.39 | testmachine | hehe |
| 11:35.48 | testmachine | nah almost lunch here |
| 11:36.05 | testmachine | and its friday |
| 11:36.08 | testmachine | so it will be fastfood |
| 11:36.09 | lehel | yap |
| 11:36.30 | lehel | testmachine u live in NL.. |
| 11:36.52 | testmachine | lehel: yes i do |
| 11:37.11 | *** join/#asterisk coppice (n=chatzill@178.194.17.210.dyn.pacific.net.hk) |
| 11:37.33 | lehel | i work 4 an nl co. |
| 11:37.46 | testmachine | lehel: ahha |
| 11:38.44 | testmachine | nah, co-work |
| 11:38.47 | testmachine | like for school |
| 11:39.18 | *** join/#asterisk mut (n=animenod@65.111.201.79) |
| 11:41.55 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 11:42.22 | *** join/#asterisk stevie20 (n=stevie@mini.fdknet.de) |
| 11:42.35 | stevie20 | hello |
| 11:43.04 | testmachine | pooh_: lol |
| 11:43.26 | stevie20 | i think i've got a problem with asterisk... but i don't really know... |
| 11:43.35 | pooh_ | stevme neither ;-) |
| 11:43.55 | stevie20 | do you think, you can help me a little bit? ;_) |
| 11:44.18 | pooh_ | stevie20: maybe, shoot |
| 11:44.36 | testmachine | stevie20: whats the prob |
| 11:44.42 | stevie20 | we are using asterisk 1.0.8 |
| 11:45.00 | stevie20 | and we have got an TE205P E1 interface card |
| 11:45.25 | stevie20 | following call routing setup... |
| 11:45.58 | stevie20 | incoming calls from the PSTN are forwarded to the second zap device.. on the second zap device is an Cisco 7200 |
| 11:46.31 | stevie20 | outgoing calls are accepted from the second zap device and forwarded to an SIP gateway standing in another city |
| 11:46.38 | testmachine | lunch |
| 11:46.39 | testmachine | bye |
| 11:47.24 | stevie20 | now i've expereinced, that some calls are forwarded to the SIP gateway und when the called station begins to ring, sometimes the connections get dropped |
| 11:47.36 | stevie20 | with following error message: |
| 11:47.48 | stevie20 | Nov 11 12:38:21 WARNING[10292]: Unable to forward frame |
| 11:49.01 | pooh_ | stevie20: turn on debug in logger.conf e.g. add the word 'debug' to the end of the 2 lines, restart and see |
| 11:49.08 | pooh_ | I need to go sorry |
| 11:50.31 | stevie20 | np.. |
| 11:51.10 | lehel | stevie20: and set debug 3, then look at asterisk/full log |
| 11:52.16 | shanky | yeah |
| 11:52.44 | shanky | I have finally incoming and outgoing calls with my telco |
| 11:52.47 | shanky | :D |
| 12:06.36 | *** join/#asterisk blop (i=blop@VoIP-with-Asterisk.mgcp.h323.sccp.sip.iax.be) |
| 12:14.18 | *** join/#asterisk JJ[work] (n=ooo@82-37-50-93.cable.ubr03.brom.blueyonder.co.uk) |
| 12:14.33 | JJ[work] | is anyone able to help me with setting up AAH? |
| 12:15.55 | stevie20 | lehel is it possible, to get the debug running, without restarting asterisk? |
| 12:16.30 | JJ[work] | anyone? |
| 12:19.38 | lehel | stevie20: if you "set debug 3" [core debug] it is not necessary to restart asterisk, you just need to check your /var/log/asterisk/full |
| 12:19.38 | *** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8) |
| 12:23.24 | stevie20 | on the cli interface? |
| 12:24.03 | lehel | yes stevie20 |
| 12:26.34 | JJ[work] | so noone here can help... |
| 12:26.38 | JJ[work] | what a useful channel |
| 12:27.31 | shanky | JJ[work]: just tell your problem |
| 12:28.22 | lehel | JJ[work] this chan is not about aah, but speak |
| 12:28.25 | JJ[work] | i have installed asterix@home onto an old pc. i have it hooked up. im registered with sipgate but cannot for the life of me make it work. i did earlier get outgoing calls working but it meant massacrering the extensions.conf |
| 12:28.31 | *** join/#asterisk Tjief (n=Tjief@r0x0r.dk) |
| 12:29.15 | JJ[work] | from xlite all i get is a 404 error |
| 12:29.32 | JJ[work] | and incoming calls just goto sipgate voicemail now. (although at one stage it did connect but nothing happened on the line) |
| 12:36.03 | testmachine | JJ[work]: did you edit extentions.conf? |
| 12:36.41 | JJ[work] | yes |
| 12:36.45 | JJ[work] | and the sip.conf |
| 12:36.47 | testmachine | so yes, what did you edit? |
| 12:37.05 | JJ[work] | i tried 'adding' to it but that didnt seem to work so in the end i pasted in and modified someone elses conf |
| 12:37.12 | JJ[work] | but tbh |
| 12:37.17 | JJ[work] | id prefer to just do it all through the standard interface |
| 12:37.23 | JJ[work] | as its all then 'done for me' instead of lots of coding |
| 12:38.09 | testmachine | you should edit something like exten => sipuser,1,Dial(SIP/sipuser,10,t) |
| 12:38.45 | JJ[work] | i think i might need to reinstall the whole damn thing and start again unless i can find the default sip and extentions conf files for aah |
| 12:41.27 | testmachine | just do a make samples again |
| 12:42.06 | *** join/#asterisk liran_ (n=liran@80.178.120.123.adsl.012.net.il) |
| 12:42.14 | liran_ | hey everyone |
| 12:42.36 | JJ[work] | how do i do a 'make samples'? |
| 12:43.28 | *** join/#asterisk Inkubot (n=inkubot@200.75.4.7) |
| 12:43.41 | liran_ | i was asked to program something that would produce a report of calls/rates based on the asterisk's master.csv log file. is there anywhere i can get some help/refernce regarding it? i haven't had any experience with the asterisk system before |
| 12:43.48 | testmachine | JJ[work]: cd /usr/src/asterisk |
| 12:43.53 | testmachine | make clean |
| 12:44.27 | testmachine | make samples |
| 12:45.01 | JJ[work] | that wont work tho. my prompt is [root@asterix1 root] |
| 12:45.14 | JJ[work] | so typing cs /usr etc doesnt go anywhere :/ |
| 12:45.23 | testmachine | cd |
| 12:45.26 | testmachine | hm |
| 12:45.35 | testmachine | wich linux distro do you use? |
| 12:47.05 | JJ[work] | centro2.5 |
| 12:47.07 | JJ[work] | er |
| 12:47.07 | JJ[work] | 3.5 |
| 12:47.16 | JJ[work] | its all bundled up with the aah thing |
| 12:48.05 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 12:48.17 | [TK]D-Fender | 1 hr to Astricon :) |
| 12:48.35 | testmachine | hmz |
| 12:49.44 | stevie20 | anybody got experience with incoming fax ? |
| 12:54.26 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
| 12:56.09 | [TK]D-Fender | I've my share of PROBLEMS with it, does that count? |
| 12:58.04 | [TK]D-Fender | Personal suggestion : Go analog for your faxes and don't let * near them |
| 12:58.28 | *** join/#asterisk santiago (n=santiago@208.195.215.124) |
| 12:58.59 | *** join/#asterisk grimse (n=grimse@p5481E772.dip.t-dialin.net) |
| 12:59.08 | stevie20 | ok d-fender, that counts ;-) |
| 12:59.36 | coppice | stevie20: what's your problem? |
| 13:00.22 | stevie20 | coppice we normaly get faxes from pstn... |
| 13:00.31 | stevie20 | but they dont work now... |
| 13:01.08 | *** join/#asterisk Druken (n=druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com) |
| 13:01.09 | stevie20 | previously, we don't use asterisk.. we just tesing asterisk... |
| 13:01.18 | stevie20 | and i should enable fax... |
| 13:01.45 | [TK]D-Fender | I've got a PRI on 1 T! port, and a Rhino Channel bank on the other. My faxes are on the Rhino and I get a sad failure rate. |
| 13:02.38 | Druken | [TK]D-Fender: could be worse, i'm sure your fax machine doesn't dissapear from your system... |
| 13:02.45 | mut | yayz |
| 13:02.49 | mut | got another 405p in |
| 13:03.31 | coppice | stevie20: well that was rather vague. what is your setup? |
| 13:03.42 | stevie20 | coppice maybe this is an problem with vad ? is vad enbaled by default for calls coming from a zap dvice and going out to a zap device ? |
| 13:04.01 | stevie20 | coppice we got asterisk 1.0.8 with a TE205P Pri interface card |
| 13:04.14 | *** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
| 13:04.17 | stevie20 | incoming calls are bridged from zap1 to zap2 |
| 13:05.10 | coppice | FAX across an VoIP channel is normally a very bad idea |
| 13:05.47 | stevie20 | but this should be an native TDM channel bridged to the other native TDM channel... |
| 13:05.49 | Druken | coppice: i agree... but it can be done... |
| 13:06.28 | coppice | not with any eral reliability |
| 13:07.50 | [TK]D-Fender | Ok, off to Astricon! Later all..... |
| 13:07.58 | stevie20 | hmm.. could this be a problem of echo chancellation? |
| 13:08.00 | Druken | coppice: fax even on an analog line isn't realiable.... |
| 13:08.36 | stevie20 | cancellation |
| 13:08.50 | coppice | you must have awful analogue lines then. For most people FAX is 99.something percent reliable. usually 99.9something |
| 13:09.15 | liran_ | where can i find a sample master.csv file to work on? cause i dont have access right now to our asterisks servers |
| 13:09.22 | Druken | coppice: no... i've just always had bad luck with fax |
| 13:09.51 | coppice | stevie20: by TDM channel do you mean lines on a TDM400 card? |
| 13:10.04 | stevie20 | lines on an TE205P |
| 13:10.41 | coppice | where is the fax machine? |
| 13:11.47 | stevie20 | connected to an Cisco 7200 |
| 13:12.27 | stevie20 | and the cisco is on the other inteface from the two ones from the TE205P |
| 13:13.24 | Druken | uhmm... isn't a cisco 7200 a router? |
| 13:13.43 | coppice | that's what I was thinking |
| 13:13.45 | Inkubot | damn cisco phone |
| 13:13.52 | Inkubot | Protocol Application Invalid :( |
| 13:14.04 | stevie20 | yes, but our 7200 has got an E1 interface card fpr receiving calls... |
| 13:14.05 | Nivex | whee I just had that last night :) |
| 13:14.24 | Nivex | Now I can't get mine to register. |
| 13:14.39 | Inkubot | Nivex: how you fix it ' |
| 13:14.41 | Inkubot | ? |
| 13:14.45 | stevie20 | and the cisco 7200 routes the calls to analogue/isdn lines or to a cisco call manager... |
| 13:14.50 | Druken | stevie20: and normal calls work ? |
| 13:15.06 | Nivex | Inkubot: look for XMLDefault.cnf.xml on the wiki |
| 13:15.08 | stevie20 | normal calls works, but i expereince some quality "impact"... |
| 13:15.20 | stevie20 | it looks like there is vad activatet |
| 13:15.34 | Inkubot | voip ? |
| 13:15.45 | coppice | stevie20: even "minor quality impact" means "0 fax reliability" |
| 13:16.18 | stevie20 | when the called party doesnt say any words, there totally silence |
| 13:16.47 | stevie20 | coppice yes, thats right... |
| 13:16.55 | stevie20 | how to disable VAD ? |
| 13:17.04 | Druken | ahh, yes... that would be silence suppression |
| 13:17.07 | Druken | that's a bad thing |
| 13:17.09 | *** join/#asterisk MrSip (n=lior@kone12.seclan.vip.fi) |
| 13:17.12 | stevie20 | or silence suppression... |
| 13:17.13 | MrSip | hello |
| 13:17.18 | stevie20 | how to disable it? |
| 13:17.29 | Druken | how the hell would i know... ask cisco |
| 13:17.39 | stevie20 | the cisco is not the problem |
| 13:17.49 | MrSip | i have a question regarding multi ring and multi sip lines who can help |
| 13:17.53 | stevie20 | without asterisk it works |
| 13:18.55 | Druken | well stevie20, i'd bone up on the wiki about silence suppression than |
| 13:20.06 | stevie20 | voip-info ? |
| 13:21.05 | Druken | yes |
| 13:21.47 | MrSip | i have 3 sip trunks on the same provider. i configured on the provider SSW multi ring of one of the account to ring on all the others. on * all this trunks maped to Queue why its not workin???????+ |
| 13:22.13 | *** join/#asterisk zoa (n=kkk@pirus.securax.be) |
| 13:22.16 | zoa | hey ho! |
| 13:23.09 | MrSip | help please |
| 13:26.10 | manxpower | MrSip, ask |
| 13:26.32 | MrSip | hi |
| 13:26.34 | MrSip | thnx |
| 13:27.01 | manxpower | MrSip, when a call goes into a queue it might be considered "answered" |
| 13:27.09 | MrSip | i have 3 tsip trunks account configure on * (3 difrrent peers) |
| 13:27.11 | manxpower | Define: SSW multiring |
| 13:27.25 | manxpower | MrSip, incoming or outgoing? |
| 13:27.37 | manxpower | MrSip, We don't call them trunks, we call them "peers" |
| 13:27.45 | manxpower | or "accounts" |
| 13:28.04 | coppice | we don't call them docks, we call then piers |
| 13:28.13 | MrSip | i configure the first account on the SSW when its get a call to ring simulteinus on all the 3 account until the first answer the call |
| 13:28.30 | manxpower | coppice, trunks and accounts are different. So are docks and piers 8-) |
| 13:28.37 | manxpower | MrSip, WHAT IS SSW |
| 13:28.59 | MrSip | soft swithch |
| 13:29.59 | coppice | our ferries dock at the piers |
| 13:30.00 | MrSip | i have a sip soft switch that the * registre to with 3 accounts |
| 13:30.06 | manxpower | So you set up your account with the first provider. Where do you set "ring simulteinus on all the 3 account"? |
| 13:30.50 | MrSip | on the soft switch so * get 3 invite message when i dial in |
| 13:31.24 | manxpower | MrSip, Sonmewhere one of the calls is being answered. You have to figure out where that is and stop that from happening. |
| 13:31.42 | MrSip | you don't get me |
| 13:31.55 | MrSip | i give you example |
| 13:32.49 | MrSip | on SSW (not *) i opened accponts 358753250900 - 358753250903 (sorry its 4 accounts) |
| 13:33.18 | zoa | hey ho, manx 'n coppice! |
| 13:33.37 | MrSip | in sip.conf * register to those acoounts on the SSW and use them to get cals in and out |
| 13:33.46 | coppice | zoa: hi |
| 13:34.31 | *** join/#asterisk emakris2 (n=emakris@c-24-128-56-2.hsd1.ma.comcast.net) |
| 13:34.57 | MrSip | on the SSW i configur hunt group (multi ring) so if i call to 358750900 its ring on all the others too until some one is answer |
| 13:35.43 | MrSip | is it clear until now |
| 13:39.37 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 13:40.19 | manxpower | zoa, How have you been, Mr Jitterbuffer? |
| 13:40.24 | *** join/#asterisk rculp (n=rculp@66.173.240.20) |
| 13:40.35 | zoa | fine fine |
| 13:40.39 | zoa | working like crazy |
| 13:40.46 | mut | mmm |
| 13:40.53 | mut | thai beef fried rice for breakfast |
| 13:42.48 | ManxPower | stevie20, Asterisk does not support sending silence supression, Asterisk does not support receiving silence supression. Turn off silence supression in any device that connects to Asterisk |
| 13:45.18 | stevie20 | thx ManxPower... this implicied that asterisk isnt responsible for the silence suppression, right? |
| 13:45.29 | ManxPower | stevie20, Correct |
| 13:45.30 | testmachine | bleh |
| 13:45.38 | testmachine | cant fix this dammn voicemail |
| 13:45.43 | ManxPower | Since Asterisk CANNOT be responsible for silence supression |
| 13:47.20 | stevie20 | damn... no the question is, WHO is suppressing my silence.... |
| 13:47.26 | stevie20 | ...now... |
| 13:49.02 | *** join/#asterisk Inkubot (n=inkubot@200.75.4.7) |
| 13:49.27 | ManxPower | stevek, Cisco routers do it by default |
| 13:49.31 | *** join/#asterisk enniof (n=Ennio_F@host130-204.pool82188.interbusiness.it) |
| 13:49.34 | ManxPower | I don't think Cisco phones do, but I don't know for sure. |
| 13:50.00 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 13:52.30 | stevie20 | hmm ManxPower, without asterisk and sip, there is no problem.. |
| 13:52.43 | stevie20 | is think this is a problem on the SIP gateway... |
| 13:52.50 | ManxPower | stevek, correct, since pretty much all devices support silence supression |
| 13:53.00 | enniof | Hi all. I've a problem with grandstream BT100 phone... the phone become MUTE as soon as the SIP link to asterisk is established. Could you help me? |
| 13:53.03 | iCEBrkr | Oh shit, we get GoSub and Return now?? |
| 13:53.10 | stevie20 | i have to search for it... |
| 13:53.14 | iCEBrkr | GW-BASIC HERE WE COME!! W00T! |
| 13:53.28 | Inkubot | the phone is alive again :) |
| 13:57.53 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-242-4.buckeyecom.net) |
| 14:01.02 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:01.09 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 14:02.09 | stevie20 | hhmmm... ManxPower, i changed the dialplan, now any call is directly placed on the PMX... |
| 14:02.20 | *** join/#asterisk serhiy (n=serhiy@65.202.182.2) |
| 14:02.28 | stevie20 | so, SIP is disabled.. now there isnt any silence suppression |
| 14:02.39 | serhiy | Hi everyone! |
| 14:02.47 | stevie20 | but, outgoing faxes still not working.. any idea? |
| 14:03.42 | serhiy | I'm absolutely new to Asterix and PBX and need some little help, so I will know where to start. |
| 14:03.58 | iCEBrkr | serhiy: start here :P |
| 14:04.00 | iCEBrkr | ~docs |
| 14:04.01 | jbot | well, docs is 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 |
| 14:05.06 | serhiy | We have Nortel PBX. Is it possible to add Asterix and connect them together? |
| 14:05.32 | iCEBrkr | serhiy: Possibly, if you have an analog card in there. |
| 14:05.41 | iCEBrkr | I'm not a PBX guy, so I'm not sure of the other options |
| 14:05.46 | serhiy | Tahnks a lot |
| 14:07.34 | serhiy | Can you estimate a cost of implementing Asterix for Company with 5 remote offices (10 phones each, 4 incoming lines per office) and 70 phones main office? |
| 14:07.45 | serhiy | just draft cost |
| 14:08.11 | iCEBrkr | serhiy: Depends on the phones. |
| 14:08.16 | *** join/#asterisk Level2 (i=LEVEL2@chp253.neoplus.adsl.tpnet.pl) |
| 14:08.21 | iCEBrkr | Depends on your PRI providers, etc. |
| 14:08.32 | ManxPower | Depends on how much profit you want to make. |
| 14:08.35 | serhiy | Waht about softphones (using computers) |
| 14:08.55 | iCEBrkr | serhiy: Softphones in an non-geeky office environment? Naaa, skip that idea |
| 14:09.05 | serhiy | no profit. I'm working in that company |
| 14:09.33 | iCEBrkr | serhiy: Polycom's are about $200 on the low end. For a decent phone. |
| 14:10.41 | serhiy | Do i need server in main office only or in remote offices too (we have a vpn connections) |
| 14:11.23 | iCEBrkr | serhiy: I'd put an asterisk box in each office |
| 14:11.25 | ManxPower | SERGEUS, That depends on how you design it. |
| 14:11.28 | iCEBrkr | then trunk'm |
| 14:11.44 | ManxPower | We normally put an Asterisk server in each office, then use local PSTN lines, and only use VoIP for inter-office calls. |
| 14:11.56 | ManxPower | Since you can never depend on the internet for ANYTHING. |
| 14:11.58 | iCEBrkr | ManxPower: That's exactly what I was thinking |
| 14:12.44 | ManxPower | I just did an Asterisk install where we did not put in a local server. The existing office had Asterisk, the new branch office was opening. We got a T-1 between the two offices and run voice and data over it. |
| 14:12.55 | *** part/#asterisk rculp (n=rculp@66.173.240.20) |
| 14:13.00 | iCEBrkr | ManxPower: How'd that work out? |
| 14:13.08 | ManxPower | But that's ONLY because we could put QoS on the link in both directions, because it was a point-to-point T-1 |
| 14:13.15 | ManxPower | iCEBrkr, seems to be working fairly well. |
| 14:13.25 | iCEBrkr | Cool |
| 14:14.06 | serhiy | i think a point-to-point T-1 is pury expensive? |
| 14:14.08 | *** join/#asterisk maik (n=maik@bfs.cs.uni-sb.de) |
| 14:14.38 | *** join/#asterisk without (n=dean_dav@CPE-60-226-176-110.qld.bigpond.net.au) |
| 14:15.33 | serhiy | Is there any local to Jersey city consultants? |
| 14:15.34 | *** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
| 14:16.38 | *** join/#asterisk Maksim (n=max@213.142.207.20) |
| 14:18.34 | ManxPower | serhiy, So are local PSTN lines. |
| 14:18.43 | ManxPower | T-1s are billed based on mileage. |
| 14:19.06 | *** join/#asterisk ast_freak (n=jesse@hades-out.universalsystems.net) |
| 14:19.10 | iCEBrkr | ManxPower: What part of the country you in? |
| 14:19.30 | ManxPower | iCEBrkr, what part of the country am I in, or what part of the country my customers are in? |
| 14:19.41 | iCEBrkr | You. :) |
| 14:19.45 | ManxPower | I am near Texarkana |
| 14:19.54 | iCEBrkr | haha |
| 14:19.58 | ManxPower | My customers are mostly about 30 miles north of New Orleans |
| 14:20.13 | ManxPower | I hope to move out of this god forsaken area soon. |
| 14:20.46 | iCEBrkr | Ever hear of Paetec? They do leased lines/PRI's etc. |
| 14:20.56 | iCEBrkr | I think they're mainly in Florida |
| 14:22.09 | ManxPower | iCEBrkr, nope. |
| 14:22.18 | ManxPower | We use I-55 telecom as our preferred CLEC. |
| 14:22.38 | ManxPower | We use AT&T as our preferred IXC |
| 14:22.49 | ManxPower | We use the local ILEC when it makes sense to do so. |
| 14:23.57 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com) |
| 14:24.20 | nassy | are there any WiFi (not sure if that is the right term) phones that support IAX |
| 14:25.21 | SkramX | nassy: I do not think so. |
| 14:25.26 | iCEBrkr | Zyxel is making WiFi SIP phones?!?! |
| 14:25.27 | nassy | :-( |
| 14:25.39 | SkramX | it cant be too hard to make an iaxy wireless.. but bleh |
| 14:25.55 | nassy | seems like all the demand is for SIP though |
| 14:28.50 | testmachine | thos zyxel phones are pretty good |
| 14:28.51 | testmachine | :) |
| 14:29.25 | *** join/#asterisk yxa (i=empty@cm121.gamma228.maxonline.com.sg) |
| 14:29.25 | testmachine | those even |
| 14:29.29 | iCEBrkr | testmachine: You get in on there beta test? |
| 14:29.34 | iCEBrkr | their even. |
| 14:29.53 | testmachine | we tested it here a while ago |
| 14:30.17 | *** join/#asterisk frenzy (n=frenzy@193.220.82.108) |
| 14:30.21 | iCEBrkr | Cool. |
| 14:30.39 | frenzy | hi |
| 14:30.48 | frenzy | what Numbering format are supported with asterisk ? |
| 14:31.14 | yxa | guys this might be a stupid question but should i get 1.2.0 RC1 or CVS HEAD to learn * ? |
| 14:31.39 | frenzy | prefix (TBA) - E.164 ? |
| 14:32.19 | iCEBrkr | frenzy: numbering formats? |
| 14:32.30 | Lostfrog | Anyone set up an audiocodes MP1XX box for authentication with *? |
| 14:32.31 | iCEBrkr | frenzy: anything you want.. it has pattern matching and expressions |
| 14:32.33 | ManxPower | yxa, it's doesn't really matter. CVS-HEAD is 1.2.0 RC1 + bugfixes |
| 14:32.53 | frenzy | hmm |
| 14:32.59 | frenzy | what does VAD/CNG mean ? |
| 14:33.08 | Nugget | continuing to follow CVS HEAD going forward might not always be so similar to the production release, though. |
| 14:33.15 | yxa | ManxPower thanks. cvs head it is... |
| 14:33.26 | ManxPower | Nugget, correct. |
| 14:33.36 | *** join/#asterisk david09452 (n=davidh@196.34.229.130) |
| 14:33.51 | ManxPower | yxa, it's doesn't really matter. Currently CVS-HEAD is 1.2.0 RC1 + bugfixes, but after 1.2.0 is released, they will diverge. |
| 14:33.57 | david09452 | at last i am registered and here we go |
| 14:34.00 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 14:34.04 | david09452 | hi there people |
| 14:34.20 | Inkubot | how can enable call pickup ? |
| 14:34.23 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 14:34.29 | david09452 | how long has asterisk been available |
| 14:34.31 | david09452 | ?? |
| 14:34.37 | yxa | ManxPower ok. |
| 14:34.38 | ManxPower | david09452, at least 2 years |
| 14:34.48 | ManxPower | maybe 3 years |
| 14:34.56 | *** join/#asterisk kippi (n=chris@untrust-gct.equinoxit.net) |
| 14:34.58 | ManxPower | Inkubot, English is not your native language? |
| 14:35.01 | david09452 | zzzzzzzzzzzeeeeeeesh and i only learned about it today |
| 14:35.17 | Inkubot | haha no :P |
| 14:35.26 | Lostfrog | Don't feel bad.. some companies still don't know about it. |
| 14:35.26 | Inkubot | how can i enable call pickup ? |
| 14:35.30 | Inkubot | is that correct ? |
| 14:35.32 | david09452 | does anyone know if there is any training available |
| 14:35.36 | ManxPower | Inkubot, Asterisk has several different types of call pickup. |
| 14:35.45 | coppice | 5 years. we had bits of * running early in 2000 with the zapata 1 cards |
| 14:35.49 | Inkubot | group and directed |
| 14:35.52 | Inkubot | right ? |
| 14:36.09 | ManxPower | Inkubot, the most common is "pick up any ringing phone by using #8" That is configured by callgroup= and pickupgroup= in zapata.conf or sip.conf |
| 14:36.34 | zoa | i use if for 3 years now i think |
| 14:36.38 | ManxPower | Inkubot, I don't know the status of Directed Pickup. It's a new feature in 1.2.x/CVS-HEAD |
| 14:36.53 | Inkubot | and i can define de #8 in features.conf |
| 14:37.16 | zoa | Sun Nov 28, 1999 3:39 pm |
| 14:37.16 | zoa | markster at marko.net (Ma |
| 14:37.21 | zoa | is the first mail on the mailinglist |
| 14:37.32 | david09452 | where is the best way to learn this program? |
| 14:37.40 | ManxPower | zoa, so 6 years then. |
| 14:37.49 | coppice | i spent the christmas holiday in 1999 getting zapata 1 cards to work with E1 |
| 14:37.50 | iCEBrkr | david09452: Here. |
| 14:37.50 | zoa | http://www.asteriskguru.com/archives/viewforum.php?f=2&topicdays=0&start=42550 |
| 14:37.53 | iCEBrkr | ~docs |
| 14:37.54 | jbot | extra, extra, read all about it, docs is 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 |
| 14:38.22 | david09452 | or should i say where do i start? |
| 14:38.45 | iCEBrkr | ^^^^^^^^^^^^^^^^^^^^^^^^ |
| 14:39.28 | Inkubot | aa great |
| 14:39.32 | david09452 | k is there anyway one can download a basic working model and do your changes on it? |
| 14:39.34 | Inkubot | thanks ManxPower |
| 14:39.40 | Inkubot | now i'm going to try |
| 14:39.44 | iCEBrkr | david09452: www.asterisk.org |
| 14:40.00 | ManxPower | david09452, Telecom is HARD and there is no standard for how things are configured. |
| 14:40.21 | ManxPower | There are silly girly man stuff like Asterisk@Home, but no real geek would use those. |
| 14:40.33 | david09452 | i know i am a philips pabx programmer |
| 14:40.33 | iCEBrkr | ha! I grew up hacking phones/phreaking, and I still don't understand all this telco shit |
| 14:41.11 | zoa | its capt crunch! |
| 14:41.12 | zoa | hide |
| 14:41.14 | zoa | HIDE! |
| 14:41.25 | iCEBrkr | Pfft, I'm hardly Capt'n Crunch |
| 14:41.37 | david09452 | if you set up a model could you create an html interfase to add, change date for instant extention numbers ect? |
| 14:41.44 | zoa | i hope you are not |
| 14:41.50 | zoa | you dont like little boys ? |
| 14:41.51 | iCEBrkr | Tho, I *DID* have a whistle when I was kid. Just too bad I couldn't hang on to it when I figured out all the phreaking stuff :) |
| 14:42.07 | iCEBrkr | zoa: hehe |
| 14:42.15 | iCEBrkr | zoa: I heard he was kinda freaky at Astricon. :) |
| 14:43.13 | SplasPood | ice: stop kidding yourself.. yer not old enough |
| 14:44.40 | iCEBrkr | SplasPood: I'm not old enough? Pfft!! Seriously, I remember having one of those whistles as a child. |
| 14:44.59 | zoa | kinda freaky |
| 14:45.02 | zoa | omg yeah |
| 14:45.06 | iCEBrkr | SplasPood: ...and like I said, it wasn't until later that I found out what they can be used for... |
| 14:45.11 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 14:45.42 | iCEBrkr | SplasPood: By then, it was long gone. I did dig in my attic in all boxes my parents had of my toys. I couldn't find it and I was pissed. |
| 14:46.16 | SplasPood | cept for the fact that I believe 2600hz was no longer used by the 70s or so |
| 14:46.22 | SplasPood | but I could be wrong |
| 14:48.43 | coppice | 2600Hz was used until the 90s |
| 14:48.54 | SplasPood | in the US? |
| 14:49.03 | coppice | you can probably find some around even now. |
| 14:49.35 | SplasPood | I've got a VAXserver 3100 in my apartment... that doesn't mean its still in use :P |
| 14:50.08 | coppice | The US was one of the the last countries to get its backbone fully digital |
| 14:50.23 | coppice | apart from the 3rd world, of course |
| 14:51.29 | *** join/#asterisk rking (n=rking@ip68-1-234-152.dl.dl.cox.net) |
| 14:52.01 | SplasPood | Capt'n Crunch /w the whistle was in the 70s tho |
| 14:52.05 | coppice | the first place to elimiate all analogue kit from its PSTN was Singapore, and that was in something like 1995. HK came soon after |
| 14:52.30 | zoa | in belgium it was all digital in 95 too i think |
| 14:53.02 | *** join/#asterisk maik (n=maik@bfs.cs.uni-sb.de) |
| 14:53.11 | SkramX | :) |
| 14:53.23 | coppice | could be. within a year of so of us, quite a few countries reached the all digital status |
| 14:53.23 | iCEBrkr | SplasPood: I understand that, but do you understand the cool-factor of having one? |
| 14:53.35 | coppice | RoyK: most phones still are analogue |
| 14:53.40 | SplasPood | iCE: Maybe... I'm just implying you never had one :P |
| 14:54.13 | david09452 | is it possible to create a voicemail server with asterisk using bro ports? |
| 14:54.20 | iCEBrkr | SplasPood: Fucker, I did too. |
| 14:54.25 | david09452 | sorry bri port |
| 14:54.36 | zoa | royk, i used that too |
| 14:54.36 | RoyK | david09452: yes, of course |
| 14:54.38 | SplasPood | iCE: its ok man.. yer lame... lots of people are.. Just learn to accept it :P |
| 14:54.38 | SkramX | bro? |
| 14:54.40 | RoyK | zoa: :) |
| 14:54.43 | zoa | to make fun of teachers |
| 14:54.56 | Lostfrog | I wonder if you could get asterisk to respond to 2600. |
| 14:54.57 | zoa | i called them and "forgot" to hang up |
| 14:55.07 | david09452 | anyone from south africa here? |
| 14:55.08 | zoa | to make sure they lost their phone for the evening |
| 14:55.09 | zoa | :) |
| 14:55.19 | RoyK | zoa: hehe |
| 14:55.25 | coppice | LostFrog: I have tone signalling for *, though it has never been used |
| 14:56.07 | david09452 | do you need a sip proxy or can asterisk handle the routing? |
| 14:56.31 | znoG | if I boot up a system and the Zaptel FXO card is not detected, there isn't much else you can try is there? |
| 14:56.36 | coppice | its amazing how much old stuff with weird protocols is still in use. old stuff lasts so much better than the newer stuff :-( |
| 14:57.34 | *** part/#asterisk MrSip (n=lior@kone12.seclan.vip.fi) |
| 14:57.48 | iCEBrkr | SplasPood: Don't be jealous, fag. |
| 14:58.03 | SplasPood | iCEBrkr: :P |
| 14:58.13 | iCEBrkr | SplasPood: Just cuz you didn't pwn a 800 VMB service when you were 15... :D |
| 14:58.33 | iCEBrkr | Actually, I pwned any ASPEN VMB system back in the day |
| 14:59.27 | SkramX | LOL |
| 14:59.36 | SplasPood | c0dez |
| 15:00.19 | SplasPood | iCE: when I was 15 I was crackin vaxen |
| 15:00.45 | coppice | when I was 15 there were no vaxen |
| 15:01.08 | SplasPood | coppice: how old are ya? |
| 15:01.12 | zoa | when i was 15 i was drinken milk |
| 15:01.25 | coppice | and none of them microprocessor thingies, either |
| 15:01.39 | SplasPood | coppice: ahh yer an old fart :P |
| 15:01.47 | coppice | we had to make do with counting wooden blocks |
| 15:01.51 | SplasPood | you were hackin stone ta... |
| 15:02.09 | coppice | we counldn't afford stone |
| 15:02.34 | Lostfrog | Clay tablets? |
| 15:03.03 | coppice | we had a garden that seemed like a solid chunk of clay :-) |
| 15:03.59 | coppice | we dipped our pens in our ink-wells and wrote on paper |
| 15:04.07 | Lostfrog | Happy US die for your country day. |
| 15:04.22 | stbain | Obligatory Monty Python Reference... |
| 15:04.25 | stbain | Well, of course, we had it tough. We used to 'ave to get up out of shoebox at twelve o'clock at night and lick road clean wit' tongue. We had two bits of cold gravel, worked twenty-four hours a day at mill for sixpence every four years, and when we got home our Dad would slice us in two wit' bread knife. |
| 15:04.42 | *** join/#asterisk _Sam-- (n=sam@phone2.kneedraggers.com) |
| 15:04.50 | *** part/#asterisk david09452 (n=davidh@196.34.229.130) |
| 15:04.55 | coppice | that python sketch wasn't around when I was 15, either |
| 15:05.00 | stbain | heheh |
| 15:05.07 | stbain | http://www.phespirit.info/montypython/four_yorkshiremen.htm |
| 15:05.27 | _Sam-- | if a call comes in on asterisk, and an SIP client answers it and the call is put on hold...is there any way for another SIP extension to answer the call without having to transfer it? |
| 15:05.59 | stbain | Sam: would call parking satisfy your needs? |
| 15:06.15 | _Sam-- | probably..i am just not familiar with it, let me check it out...thanks. |
| 15:07.25 | iCEBrkr | SplasPood: Aren't you like 20-something.. :) |
| 15:07.52 | Lostfrog | Aren't we all 20-something? |
| 15:07.55 | Lostfrog | I'm 20-11. |
| 15:07.58 | stbain | easy concept... Example: call comes in for Sam, Stbain says "please hold" and transfers the call to extension 600. The * server says back to stbain "six zero seven". Stbain pages Sam and says, "you have a call on 607". If Sam doesn't pick up a phone and dial 607 within a specified amount of time, it rings back to Stbain's desk. |
| 15:07.58 | iCEBrkr | haha |
| 15:08.31 | stbain | Lostfrog: I'm also 20-11 |
| 15:08.36 | _Sam-- | thats perfect....im reading up on it, sorry to seem stupid, i just didnt know what it was. |
| 15:09.39 | stbain | call parking especially works great in retail or manufacturing environments with paging systems. That's why when you're in department stores, you hear "Mr. Sam, you have a call on 605" |
| 15:10.14 | *** join/#asterisk maik (n=maik@bfs.cs.uni-sb.de) |
| 15:10.32 | mistral | when i try to dial out to my sip provider the call starts and then * rejects it with 403 Forbidden why is this ? |
| 15:10.39 | sivana | once you park a call, how do you get it back? |
| 15:11.10 | Lostfrog | sivana: i.e. dial 607 |
| 15:11.12 | stbain | sivana: you pick up the phone and dial the number to where it was parked. In the retail example, Sam would pick up a handset and dial 607 from any phone in the system to get his call |
| 15:11.32 | stbain | err.... 605... you get the point |
| 15:11.36 | sivana | ya |
| 15:11.45 | Lostfrog | I am seriously thinking about using call parking instead of assisted transfers. |
| 15:11.49 | sivana | what is he was parked on 505? |
| 15:11.51 | *** join/#asterisk c0w (n=c0w@staff-ns50-3.as25178.net) |
| 15:12.37 | Lostfrog | Call parking uses a range of extensions.. |
| 15:12.40 | sivana | or what about 803? |
| 15:12.41 | stbain | call parking is usually set up in a block of extensions... and you transfer the call to the starting extension (say... 600). Then the asterisk server will repeat back to you the actual extension it's parked on (say... 607) so you know where to redirect the person to. |
| 15:12.42 | Lostfrog | By default 700-720 |
| 15:13.02 | sivana | what happens if the party hangs up or the core network goes down? |
| 15:13.11 | stbain | call gets dropped |
| 15:13.24 | Lostfrog | lol.. if the network goes down, what good is * going to be? |
| 15:13.28 | sivana | why |
| 15:13.50 | _Sam-- | the wiki says "While in a conversation, press # to initiate a transfer, then dial 700"....so while you in a conversation, you dont even need to put the person on hold? |
| 15:13.55 | _Sam-- | <it didnt work for me , yet> |
| 15:14.03 | _Sam-- | http://www.voip-info.org/wiki-Asterisk+call+parking |
| 15:14.17 | stbain | Sam: make sure you have call parking configured |
| 15:14.23 | Lostfrog | features.conf |
| 15:14.24 | tzanger | _Sam--: make sure you have 'T' in your Dial() if you are the caller and 't' if you're the callee |
| 15:15.23 | sivana | tzanger: why |
| 15:15.27 | Lostfrog | Ok.. I have to tear up our telecom closet and rebuild it.. luckily we only have 6 physical phone lines. |
| 15:15.39 | Lostfrog | I hope I don't screw up. |
| 15:15.49 | tzanger | sivana: uh |
| 15:16.03 | tzanger | because 'T' means the caller of Dial() can initiate a #-transfer, and 't' means the callee can? |
| 15:16.06 | SkramX | Any VOIP people in Texas? PM me. |
| 15:16.13 | sivana | unable to comply... |
| 15:16.18 | Lostfrog | SkramX = kram? |
| 15:16.24 | tzanger | Lostfrog: no |
| 15:16.28 | SkramX | Lostfrog: No. |
| 15:16.41 | Lostfrog | ok. thanks. |
| 15:16.44 | coppice | SkramX: TI are :-) |
| 15:16.55 | Lostfrog | I didn't see kram up there. |
| 15:17.01 | SkramX | I had this name before I even knew of Asterisk && Mark Spencer |
| 15:17.07 | sivana | ya right |
| 15:17.09 | SkramX | coppice: whom? |
| 15:17.18 | Lostfrog | Texas Instruments? |
| 15:17.29 | SkramX | Oh,yeah. |
| 15:17.40 | SkramX | THey have a pretty big building somewhat close to my house. |
| 15:17.44 | coppice | Texas Instruments make the silicon for about 90% of VoIP ports |
| 15:17.55 | SkramX | I am possibly looking for an internship.. |
| 15:18.30 | SkramX | 'Have experience doing online-customer support and tech stuff for WebHosting companuies and also have a pretty vast knowledge in Asterisk and VoIP in general. |
| 15:19.49 | Lostfrog | Anyone want to hire a Lazy SOB for */Linux work in Pittsburgh? :) |
| 15:21.22 | ManxPower | sivana, you don't want anyone that you call to be able to press # and transfer themselves off your PBX, do you? |
| 15:21.29 | mistral | why does asterisk send back a forbidden message to a sip provider its trying to call though ? |
| 15:21.48 | ManxPower | This is why you should NOT USE t or T UNLESS YOU UNDERSTAND THEM. |
| 15:22.16 | mistral | manx: is that directed at me ? |
| 15:22.24 | _Sam-- | if i pick up an extension and dial "700" i get the call parking, and it says "701"...but i cant pick up the call if i dial 701...and i cant hit # while in conversation to inititation the parking |
| 15:22.37 | _Sam-- | er to initiate too |
| 15:22.42 | sivana | ManxPower: no, I do not |
| 15:23.02 | sivana | I want them to press * and then transfer themselves off |
| 15:23.18 | *** join/#asterisk demetrio (n=demetrio@62.173.180.182) |
| 15:23.20 | ManxPower | sivana, why? |
| 15:23.30 | sivana | I don't know.. just to be difficult :) |
| 15:23.46 | ManxPower | Whoo! Whoo! Free long distance service from sivana! |
| 15:23.52 | *** join/#asterisk _mountie (n=mountie@trb229.travel-net.com) |
| 15:24.12 | sivana | I don't use t or T actually.. but that's interesting though |
| 15:24.19 | demetrio | I have asterisk behind NAT. When I try to pass an incoming call to a sipphone on the same private subnet, the remote server will say "we don't forward to private IP addresses". How can I solve this? |
| 15:24.48 | ManxPower | demetrio, you need canreinvite=no in sip.conf |
| 15:25.15 | demetrio | under the remote server definition or the sipphone's |
| 15:25.16 | demetrio | ? |
| 15:25.27 | demetrio | I guess the server |
| 15:25.28 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 15:25.40 | demetrio | thanks |
| 15:26.04 | ManxPower | of course maybe you just need externip= and localnet= defined. |
| 15:26.20 | *** join/#asterisk skeffling (n=Andrew_H@andrew.1ec.aaisp.net.uk) |
| 15:26.27 | *** join/#asterisk marc324 (n=marc3234@206-248-133-19.dsl.teksavvy.com) |
| 15:26.28 | *** join/#asterisk }cytrak{ (n=kvirc@208.63.19.172) |
| 15:27.23 | *** join/#asterisk lobbin (n=lobbin@212.247.86.106) |
| 15:27.46 | skeffling | Hi, has anyone got the MWI on a SNOM 360 to work? I've got it to blink when there are messages, but pressing the Retrieve button trys calling "sip:1571@" (1571 is out vm extension), which gets 'not found' |
| 15:27.57 | *** join/#asterisk shog (n=rene@62.8.240.129) |
| 15:28.20 | Lostfrog | There is some stuff on the wiki for that.. I haven't read it because I am afraid. :) |
| 15:28.49 | bon | what does canreinvite= do? |
| 15:28.53 | skeffling | Lostfrog, I've read through that, and I'm on HEAD, it's half working! |
| 15:29.15 | Lostfrog | bon: It stops phones/providers from talking directly to each other. |
| 15:29.19 | yxa | is there anything wrong with this? exten => 516,1,Dial(SIP/18005558355@proxy01.sipphone.com,30,rT) I cant seem to connect |
| 15:29.22 | Lostfrog | They have to talk through * |
| 15:29.36 | shog | hello! i use asterisk to transfer a call, but when the recipient is busy asterisk returns error code 503. is it possible to alter the returned error code? |
| 15:32.42 | lobbin | I haven't used my asterisk for a few month, and when I try it now, my provider gives my "SIP/2.0 403 Use from=id". I haven't changed anything but apparently my provider has. What does it mean? |
| 15:34.04 | *** join/#asterisk bored2 (i=roadster@d58-105-55-54.dsl.vic.optusnet.com.au) |
| 15:34.05 | bored2 | hey |
| 15:34.09 | bored2 | anyone using a http proxy ? |
| 15:34.15 | bored2 | at the moment.. ? |
| 15:34.28 | lobbin | You probably? :) |
| 15:34.45 | bored2 | ya |
| 15:34.46 | *** join/#asterisk shanky (n=shanky@82.159.214.9) |
| 15:34.59 | bored2 | but im not being proxied enough for apache to think its not me tho |
| 15:35.32 | shog | no ideas how to change error codes? |
| 15:35.51 | bored2 | just need someone to goto a website from me from there proxy |
| 15:35.54 | bored2 | must be someone |
| 15:36.12 | SkramX | bored2: fpr what? |
| 15:36.36 | bored2 | im using htaccess to block access in apache |
| 15:36.45 | bored2 | and ppl with proxies it doesnt work for |
| 15:36.51 | bored2 | cause it comes up with the proxies ip obviously |
| 15:37.00 | SkramX | and... |
| 15:37.04 | bored2 | so i made a little script to load the x_forwarded_for as the remote_addr |
| 15:37.08 | bored2 | trying to test it tho.. |
| 15:37.11 | bored2 | but i dont have access to a proxy hehe |
| 15:37.39 | Lostfrog | set up squid. |
| 15:37.48 | bored2 | nah cant be stuffed |
| 15:37.50 | bored2 | its like 2am |
| 15:37.53 | bored2 | hehe |
| 15:37.58 | Lostfrog | I need to install squid. |
| 15:38.10 | Lostfrog | My boss told me he wants to filter traffic by URL. |
| 15:38.18 | Inkubot | anyone knows how can i make a threewaycalling with a cisco phone ? |
| 15:38.29 | Inkubot | i only can do it with hook-flash |
| 15:38.33 | bored2 | squids good |
| 15:38.40 | bored2 | useful for finding things to blackmail ya boss with :) |
| 15:39.07 | Lostfrog | He's too clean for that. |
| 15:39.10 | ManxPower | Inkubot, you press the conference button on the cisco |
| 15:39.19 | Inkubot | emm... |
| 15:39.30 | Inkubot | it's kind of wire.. |
| 15:39.54 | Inkubot | the button isn't working maybe |
| 15:39.56 | ManxPower | Flash based transfers only apply to ANALOG FXS PORTS |
| 15:40.14 | Inkubot | i know |
| 15:40.38 | Inkubot | it's there a way to change the flash button, maybe to something line *xxx |
| 15:40.46 | Inkubot | line/like |
| 15:42.22 | *** join/#asterisk YoMama (n=rewt@pcp02689850pcs.roylok01.mi.comcast.net) |
| 15:42.32 | YoMama | waah |
| 15:42.37 | testmachine | YoMama: hey, i have this database fixxed |
| 15:42.38 | testmachine | :) |
| 15:42.51 | YoMama | testmachine: holy crap..you're still on? geez |
| 15:42.55 | testmachine | haha |
| 15:42.59 | testmachine | nah, i am at work |
| 15:43.02 | YoMama | anyone running rc1 on 2.4.x kernel? |
| 15:43.05 | testmachine | bussy with asterisk |
| 15:43.26 | testmachine | but i cant get voicemail on yet |
| 15:43.38 | YoMama | voicemail? what's wrong wiht that? |
| 15:44.13 | testmachine | dont know whats wrong with it |
| 15:44.13 | testmachine | i wish i knew it |
| 15:44.29 | YoMama | i meant...what's it doing? |
| 15:44.35 | testmachine | nothing |
| 15:44.35 | testmachine | hehe |
| 15:44.46 | YoMama | anyone here know what crap i haveta recompile if i'm installing a new kernel? |
| 15:44.54 | YoMama | i'm assuming zapata and libpri |
| 15:44.58 | YoMama | but not asterisk |
| 15:45.11 | testmachine | you are using fedora? |
| 15:45.16 | testmachine | i dont like fedora |
| 15:45.20 | YoMama | no...i'm on RH 9 |
| 15:45.33 | testmachine | i dont like rh either |
| 15:45.34 | testmachine | hehe |
| 15:45.45 | YoMama | yeah...they make it harder by trying to make it easier |
| 15:45.51 | testmachine | yep |
| 15:45.53 | YoMama | reminds me of SCO |
| 15:45.59 | testmachine | debian is the future |
| 15:46.00 | testmachine | :P |
| 15:46.11 | lobbin | RH9 is old as hell |
| 15:46.32 | testmachine | but i am running ubuntu on mu asterisk machine |
| 15:46.38 | testmachine | becouse its easy for asterisk |
| 15:46.41 | testmachine | apt-get install asterisk |
| 15:46.42 | testmachine | :) |
| 15:46.49 | YoMama | lobbin: thank you....i know this... :) |
| 15:47.06 | YoMama | man...maybe i should just grab the 2.4.31 kernel and install it ghetto style |
| 15:47.12 | *** join/#asterisk dalfry (n=dalfry@gateway.ishisystems.com) |
| 15:47.22 | YoMama | i hate compiling kernels though |
| 15:47.25 | testmachine | hehe |
| 15:47.32 | testmachine | cp your old .config file :) |
| 15:47.34 | iCEBrkr | zoa: You do know why capt'n crunch is all jacked up, right? |
| 15:47.35 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 15:47.43 | testmachine | too your new kernel directory |
| 15:47.52 | testmachine | make oldconfig |
| 15:47.56 | Ariel_ | Morning everyone |
| 15:47.59 | testmachine | make menuconfig |
| 15:48.03 | testmachine | or whatever you use |
| 15:48.37 | YoMama | test...if i do a make oldconfig...why do i need make menuconfig |
| 15:48.45 | YoMama | doesn't make oldconfig just use my oldconfig to set everything up? |
| 15:49.20 | testmachine | nah with make menuconfig |
| 15:49.28 | testmachine | you can add or remove some things after all |
| 15:49.40 | testmachine | maybe there are some important changes |
| 15:49.50 | YoMama | testmachine...i guess since i'm in there...i might as well take the time to make the kernel as small as i can |
| 15:50.37 | YoMama | well...2.4.31 here i come |
| 15:50.55 | YoMama | oh boy...it's been about....5 years since i compiled a linux kernel |
| 15:51.11 | *** join/#asterisk robtro (n=lol@unaffiliated/robtro) |
| 15:51.11 | YoMama | i better test boot the new one :-P |
| 15:51.15 | robtro | hai |
| 15:51.38 | *** join/#asterisk kisu (n=kisu@2001:5c0:8fff:fffe:0:0:0:3bdb) |
| 15:52.53 | testmachine | lol |
| 15:54.09 | YoMama | or i just get a real FXS port on this box |
| 15:54.59 | *** join/#asterisk MuppetMaster (n=MuppetMa@62.37.170.132) |
| 15:57.55 | *** join/#asterisk alvariux (n=unky@201.138.135.221) |
| 15:58.00 | alvariux | hello |
| 15:58.11 | alvariux | anybody have used realtime extensions? |
| 15:59.47 | testmachine | bye |
| 16:00.09 | *** join/#asterisk DJ-Pyro (n=abrezins@12-208-113-85.client.insightBB.com) |
| 16:00.50 | DJ-Pyro | question about a voicemail problem...a user of mine moved a message from one folder to another and now it's gone (this is version 1.2 beta1) |
| 16:00.52 | MuppetMaster | alvarisux: Yes |
| 16:00.59 | DJ-Pyro | when I try to switch into his directories I get these fun log messages |
| 16:01.06 | MuppetMaster | Anyone noticeed any anamolies with IAX in RC2? |
| 16:01.12 | MuppetMaster | RC1 that is... |
| 16:01.13 | *** join/#asterisk tim|mac (n=tim|mac@kennelvanstoopy.xs4all.nl) |
| 16:01.34 | DJ-Pyro | Nov 11 10:01:22 DEBUG[3001]: app.c:1150 ast_lock_path: Locked path '/var/spool/asterisk/voicemail/internal/1028/Family' |
| 16:01.34 | DJ-Pyro | Nov 11 10:01:22 DEBUG[3001]: app.c:1162 ast_unlock_path: Unlocked path '/var/spool/asterisk/voicemail/internal/1028/Family' |
| 16:01.34 | DJ-Pyro | Nov 11 10:01:22 WARNING[3001]: file.c:493 ast_openstream_full: File does not exist in any format |
| 16:01.34 | DJ-Pyro | Nov 11 10:01:22 WARNING[3001]: file.c:805 ast_streamfile: Unable to open (format ulaw): No such file or directory |
| 16:01.35 | DJ-Pyro | <PROTECTED> |
| 16:01.51 | MuppetMaster | DJ-Pyro, you might want to use pastebin.ca for posting such things here. |
| 16:01.53 | DJ-Pyro | That's when I press '3' for change folders |
| 16:02.46 | *** join/#asterisk santiago (n=santiago@208.195.215.124) |
| 16:03.01 | tim|mac | hi people, this might not be the correct channel to ask this, but since I'm trying to get this to work with asterisk, I'm going to give it a go :) I'm using a HFC-PCI ISDN modem and when I load the module into the kernel, the ttyI* devices aren't created... |
| 16:03.07 | tim|mac | I'm using debian sarge |
| 16:03.19 | robtro | if i've got Playback(file)... instead of doing multiple lines for each "Playback()" can i put them in like Playback(1,2,3) |
| 16:03.20 | robtro | ? |
| 16:03.22 | tim|mac | any idea where I should look for an answer? |
| 16:03.25 | *** part/#asterisk MuppetMaster (n=MuppetMa@62.37.170.132) |
| 16:04.37 | *** join/#asterisk rculp (n=rculp@66.173.240.20) |
| 16:05.09 | ManxPower | Has anyone had problems with ZAP FXS FLASH Transfers? |
| 16:05.14 | ManxPower | With 1.2rc1 |
| 16:05.41 | rculp | do I need to worry about having fxoks set in my zaptel.conf if I'm using a pri? |
| 16:05.54 | alvariux | muppetmaster how can i mixed static and realtime extensions, i have a default context with some static extension, and i added switch => Realtime/default@extensions |
| 16:05.56 | ManxPower | rculp, it will be ignored |
| 16:06.21 | alvariux | but i cant dial to the realtime extension |
| 16:06.29 | *** join/#asterisk mariogamboa (n=sudaikdd@201.138.150.15) |
| 16:06.35 | mariogamboa | hiall |
| 16:06.52 | demetrio | sometimes when I answer a call and then hang up the phone will immediatly ring again, and if I pick up I'll hear the line tone. What could be causing this? |
| 16:06.52 | mariogamboa | i have some doubt about asterisk with isdn modem |
| 16:06.53 | rculp | here's my zaptel.conf http://pastebin.ca/28375 |
| 16:07.20 | *** join/#asterisk docelmo (n=docelmo@66.237.242.41.ptr.us.xo.net) |
| 16:07.22 | rculp | I'm just trying to make sure I don't have any problems with that config |
| 16:07.34 | docelmo | Greetings! |
| 16:07.40 | rculp | I can receive incoming, but just looking to set it up optimally |
| 16:07.52 | *** join/#asterisk mountie (n=mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com) |
| 16:09.03 | mariogamboa | i have a modem isdn that give me to me 2 fxs this 2fxs connect to the asterisk pbx i need the call incoming rings in 2 extension is done but when i answer the call in one extension the another extension continue ring exist a command to give asterisk to say when the call answer in one extension cut the another line |
| 16:09.04 | mariogamboa | ? |
| 16:09.37 | *** part/#asterisk tim|mac (n=tim|mac@kennelvanstoopy.xs4all.nl) |
| 16:09.38 | snitt | omgwtf? |
| 16:10.08 | ManxPower | snitt, mariogamboa is not a native english speaker. |
| 16:10.53 | snitt | im not an native speaker |
| 16:11.04 | mariogamboa | ok i know |
| 16:11.59 | zoa | hmm |
| 16:12.11 | zoa | its either that or whatever he is having, the same for me :) |
| 16:12.42 | rculp | is there a benefit to using fxsks over just having bchan=1-23 dchan=24 |
| 16:12.44 | rculp | ? |
| 16:12.45 | mariogamboa | the problem i have is my isdn modem when incoming call comes ring in 2 fxs this 2 fxs is connected to my asterisk the problem here is when i pickup one extension to answer the call the another extension continue rings i need to pickup the another extension to hangup the second call of isdn modem |
| 16:13.19 | snitt | mariogamboa: ok i undestand, but i don't have a solution personally, im sorry |
| 16:13.51 | mariogamboa | no problem |
| 16:14.07 | snitt | mariogamboa: ask mw3 |
| 16:14.22 | snitt | but he is away at this time |
| 16:15.10 | ManxPower | rculp, If you have a PRI then only bchan= and dchan= will work for the PRI channels |
| 16:15.32 | rculp | manx: ok |
| 16:15.40 | rculp | I'll restore the config |
| 16:21.25 | *** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca) |
| 16:22.23 | *** join/#asterisk JJ[work] (n=ooo@82-37-50-93.cable.ubr03.brom.blueyonder.co.uk) |
| 16:22.30 | JJ[work] | hi |
| 16:22.40 | *** join/#asterisk Laerte (n=io@82.112.222.200) |
| 16:22.43 | Laerte | hy hall |
| 16:23.06 | JJ[work] | why would an incoming call be going straight to the voicemail service and saying 'the extension is busy' when its not |
| 16:24.20 | ManxPower | JJ[work], So many reasons. |
| 16:24.27 | *** join/#asterisk bweschke (n=bweschke@pcp09754274pcs.narlington.nj.comcast.net) |
| 16:24.27 | JJ[work] | im using AMP to configure it |
| 16:24.31 | JJ[work] | and xlite is the client |
| 16:24.38 | JJ[work] | it logs in on xlite fine nad i can dial out |
| 16:24.48 | JJ[work] | but nothing seems to be able to dial into the xlite client |
| 16:25.02 | ManxPower | device is not registered, device is not reachable, there are no codecs available in common with the device and the call, device rejects the call. |
| 16:25.35 | ManxPower | But if you want help with AMP, then ask on #ampportal or whatever the name of the channel is. |
| 16:26.01 | JJ[work] | not the right chan :L/ |
| 16:26.01 | ManxPower | JJ[work], Really, pasting the console output of a failed call on pastebin.ca might be a good place to start. |
| 16:26.11 | JJ[work] | ok ill go take a look for it |
| 16:27.51 | JJ[work] | i have a debug log |
| 16:28.24 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 16:28.49 | JJ[work] | where can i paste ? |
| 16:29.07 | rculp | pastebin.ca |
| 16:29.23 | rculp | paste into the top box and give a description in the bottom |
| 16:29.24 | rculp | submit |
| 16:29.29 | rculp | and it will generate a url for you |
| 16:29.37 | rculp | paste url here |
| 16:30.04 | ManxPower | ~pastebin |
| 16:30.05 | jbot | hmm... pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 16:30.05 | JJ[work] | http://pastebin.ca/28377 |
| 16:30.06 | JJ[work] | ta |
| 16:30.48 | ManxPower | JJ[work], you have a config error in AMP. Go ask on amp. |
| 16:30.57 | ManxPower | This isn't good, the lights just dimmed. |
| 16:31.07 | JJ[work] | ok whats the channel for amp on here |
| 16:31.10 | JJ[work] | sorry for being a pain int he ass |
| 16:31.18 | JJ[work] | it is about <---> close to working |
| 16:31.19 | JJ[work] | heh |
| 16:31.20 | shog | hello! i use asterisk to transfer a call, but when the recipient is busy asterisk returns error code 503. is it possible to alter the returned error code? |
| 16:31.21 | ManxPower | Try #amportal |
| 16:31.25 | rculp | we're all pitas at some point |
| 16:31.26 | rculp | :) |
| 16:31.31 | rculp | I'm still in the pita phase |
| 16:31.52 | Qwell | tmi |
| 16:31.53 | *** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com) |
| 16:31.59 | ManxPower | Nov 11 11:28:03 VERBOSE[1329]: -- Executing NoOp("SIP/7908091-96bf", "Returned from dialparties with no extensions to call") in new stack That is an AMP error |
| 16:32.04 | Laerte | anyone know about http://bugs.digium.com/bug_view_page.php?bug_id=0003067 (my * HEAD as the same problem and it have the channel.c patched) |
| 16:33.02 | Qwell | ManxPower: do you happen to know what it takes to become a Cisco reseller? Their ICA says I can only buy for myself and clients. ;/ |
| 16:34.08 | *** join/#asterisk bgtroll (n=bgtroll@pirus.securax.be) |
| 16:34.31 | ManxPower | Qwell, I think it takes an Act of God and a CCIE on staff. |
| 16:34.44 | Qwell | ahh |
| 16:34.58 | Qwell | so, two acts of god then |
| 16:36.07 | *** join/#asterisk ful|work (n=fulgas@209.8.233.197) |
| 16:36.09 | ful|work | hey |
| 16:36.24 | ful|work | anyone knows the extension to the pickupgroup on amp? |
| 16:37.58 | kippi | anyone know where i can get a red phone |
| 16:38.02 | stevie20 | qwell, no, its possible to get the CCIE... one of my colleagues has this certificate |
| 16:38.22 | *** join/#asterisk xtr (i=01928375@S01060012174cc0e1.vf.shawcable.net) |
| 16:38.22 | darkskiez | like batman ? |
| 16:39.01 | *** join/#asterisk XTR-II (n=xtr@staff-nat.netnation.com) |
| 16:39.42 | Laerte | anyone know about http://bugs.digium.com/bug_view_page.php?bug_id=0003067 (my * HEAD as the same problem and it have the channel.c patched) |
| 16:39.47 | docelmo | stevie20 CCIE is one of the hardest tests out. |
| 16:39.56 | docelmo | Many take and only 20% pass |
| 16:40.09 | *** join/#asterisk [Lamer] (i=Lamer@221.128.98.183) |
| 16:40.48 | [Lamer] | anyone had the RC1 compiled successfully on FC3? |
| 16:40.52 | demetrio | I'm experiencing strange stuff whith call hold |
| 16:42.12 | demetrio | twho phones on the same subnet with asterisk: during conversation, if one end puts the call on hold and then returns on line again, it won't be able to hear the other end speaking whilst the other can hear the first one |
| 16:42.21 | pooh_ | Who sees the difference: |
| 16:42.21 | pooh_ | <PROTECTED> |
| 16:42.22 | pooh_ | res_config_mysql.c: MySQL RealTime: Update SQL: UPDATE SIP_accounts SET ipaddr = '192.168.5.118', port = '5060', regseconds = '1131563860', username = '9011' WHERE name = '9011' |
| 16:43.19 | pooh_ | and if you see it, please tell me why ;-) |
| 16:43.29 | stevie20 | 20% is a good pass rate ;-) |
| 16:44.19 | pooh_ | ok I will tell: fullcontact = 'sip:9011@192.168.5.118:5060' |
| 16:44.47 | pooh_ | now please tell me why :-) |
| 16:45.13 | [Lamer] | stevie20 : have you got the RC1 running? |
| 16:45.27 | stevie20 | but not everybody ot an explanation ;-) |
| 16:45.37 | stevie20 | RC1 ? no.. |
| 16:45.54 | pooh_ | stevie20: 1 is enough ;-) |
| 16:46.24 | *** join/#asterisk S4w (n=asas@adsl-3-71-130.mia.bellsouth.net) |
| 16:46.27 | stevie20 | [Lamer] why? |
| 16:46.35 | S4w | hey hello, how do I use the pastebin thing? |
| 16:46.59 | stevie20 | hello S4w |
| 16:47.16 | S4w | hello :-) |
| 16:47.21 | [Lamer] | stevie20: just tried to compile it on FC3 |
| 16:47.22 | S4w | i need to paste an error |
| 16:47.26 | S4w | how to proceed? |
| 16:47.32 | pooh_ | S4w: go to pastebin.ca |
| 16:47.35 | S4w | ok |
| 16:47.45 | pooh_ | submit it and paste the pastebin number you get in return |
| 16:48.25 | stevie20 | and? does it work [Lamer] ? |
| 16:50.32 | pooh_ | Any RealTime junkie around pls? |
| 16:51.04 | S4w | http://pastebin.ca/28380 |
| 16:51.06 | S4w | this is it |
| 16:51.19 | pooh_ | S4w: and what is the question ? |
| 16:51.44 | [Lamer] | stevie20: just tried the asterisk-addons, but couldn't get through |
| 16:52.04 | pooh_ | S4w: as it says: wrong password on authentication |
| 16:52.12 | [Lamer] | i'm only running the 1.0.9 |
| 16:52.15 | S4w | well, the calls are not going out. I want to know why, because I am totally lost why this is hppening? :-( |
| 16:52.24 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 16:53.00 | S4w | pooh_: but my clients are connected to me successfully and I have connected to my provider ok |
| 16:53.17 | S4w | my clients dont have to authentificate with my provider |
| 16:53.48 | pooh_ | the response from your provider is 'forbidden' |
| 16:53.54 | *** join/#asterisk bgtroll (n=bgtroll@pirus.securax.be) |
| 16:54.16 | *** join/#asterisk _T3_ (n=rposada@200.63.231.210) |
| 16:54.21 | pooh_ | S4w: whi is: 8887@65.3.71.130 |
| 16:54.24 | pooh_ | who |
| 16:54.36 | _T3_ | Hi |
| 16:54.39 | S4w | 8887 is the username that I use to login to my provider |
| 16:54.57 | S4w | and that IP is my ip address |
| 16:55.20 | pooh_ | Is 8887 routed to your provider on your * box ? |
| 16:55.24 | _T3_ | anybody knows a linux iso that install asterisk but just asterisk?? or how to build one?? |
| 16:55.31 | *** join/#asterisk RickTick (n=rpulido_@71.196.2.172) |
| 16:55.41 | pooh_ | _T3_: try asterisk@home |
| 16:55.49 | S4w | but "R.Bilbao" is my client |
| 16:55.52 | S4w | I dont get it |
| 16:55.56 | _T3_ | i do, but install a lot of thing too |
| 16:56.10 | pooh_ | username is not know to your provider, change it to match yours |
| 16:56.25 | pooh_ | _T3_: me not understand ;-) |
| 16:56.52 | S4w | pooh_: the client's username to match the provide's? |
| 16:56.56 | S4w | :-O |
| 16:57.27 | pooh_ | If YOU mae a call, what does the CLI tell you when you mae a good call? |
| 16:57.30 | *** join/#asterisk dalfry (n=dalfry@gateway.ishisystems.com) |
| 16:58.16 | stevie20 | so... i leave ou work now... |
| 16:58.36 | RickTick | hello all: Anyone has an idea what this error msg is saying --- WARNING[2826]: chan_sip.c:695 retrans_pkt: Maximum retries exceeded on call 25b5e8846c1da82b2b8f5d7360c6b3c4@206.225.87.172 for seqno 102 (Non-critical Request) |
| 16:58.38 | stevie20 | we'll meet again on monday, i believe |
| 16:58.57 | stevie20 | good weekend |
| 16:58.59 | stevie20 | cya |
| 16:59.03 | pooh_ | cya |
| 16:59.30 | ManxPower | It turns out that sometime shortly before rc1 FLASH transfers changed from being in the context of the called channel to being in the context of the calling channel |
| 16:59.36 | S4w | pooh_: sorry I dont get what you are saying |
| 16:59.57 | pooh_ | When not your client but yourself is call through your provider, does that work ? |
| 17:00.25 | stbain | ManxPower: thought that was defined in the Dial() option T/t |
| 17:00.45 | ManxPower | stbain, no, T/t are for DTMF transfers, not FLASH transfers |
| 17:00.50 | S4w | I dont accept any incomming calls from this provider I only use it to make calls to PTSN network |
| 17:00.56 | stbain | ManxPower: I stand corrected |
| 17:01.04 | stbain | well... actually, I'm sitting |
| 17:01.52 | pooh_ | S4w: yes, I understand, but can YOU yourself make a good call through this provider ? |
| 17:02.19 | S4w | yes |
| 17:02.27 | pooh_ | PASTEBIN, THE CALL FLOW PLEASE |
| 17:02.30 | pooh_ | sorry |
| 17:02.34 | pooh_ | no password pls |
| 17:04.03 | S4w | pooh_: callerid="R.Bilbao" <1011> this is how I defined my client's section I am starting to suspect about it maybe I should set it to callerid="8887" <1011> being 1011 the username that the client uses to login to my server |
| 17:05.46 | S4w | pooh_: what do you think? |
| 17:05.46 | pooh_ | S4w: that's why I want to see a good call ,og |
| 17:05.49 | pooh_ | log |
| 17:06.00 | S4w | hmm |
| 17:06.06 | Inkubot | if i call to a extension, and is busy, how can i make when he hungs up automaticly asterisk make a call ? |
| 17:06.59 | Rawplayer | re |
| 17:06.59 | S4w | pooh_: I will make a call using my * where do I find the logfiles in the default debian instalation, any idea? :-O |
| 17:07.10 | pooh_ | on the CLI |
| 17:07.15 | pooh_ | set verbose 100 |
| 17:07.18 | Lostfrog | var/log/asterisk? |
| 17:07.43 | *** join/#asterisk jpablo (n=jpablo@200.94.130.194) |
| 17:07.46 | S4w | let me check |
| 17:07.57 | darkskiez | Inkubot: when the call is busy, set a database entry to point to the person who wants the call, in the hangup handler for an extension, have it check to see if a database entry for that extension has been added to invoke a ringback, and use that to originate a call, maybe via an agi script or something. |
| 17:08.47 | ManxPower | S4w, don't use quotes in callerid stuff |
| 17:09.46 | S4w | :o |
| 17:09.49 | S4w | ok |
| 17:10.25 | pooh_ | S4w: there are 305 other listeners |
| 17:10.39 | S4w | in my logs only dissasters exist :-P there is not such thing as a successfull call :-O |
| 17:11.23 | S4w | pooh_: sorry |
| 17:12.00 | *** join/#asterisk ful|work (n=fulgas@209.8.233.197) |
| 17:12.15 | ful|work | anyone got any issues with a bridge on zap and h323 ? |
| 17:12.35 | lesouvage | Does somebody wants to try to register a sip phone on my asterisk box? It should work but I can't test it from within the lan. |
| 17:12.49 | demetrio | So, if after transferring a call or putting it on hold & picking it up again communication is one-way (one end won't be able to hear the other but not vice-versa) what am I doing wrong? |
| 17:12.59 | *** join/#asterisk astcryz (n=cryzeck@www2.gathering.org) |
| 17:13.06 | astcryz | Anyone here familiar with phpagi? |
| 17:13.29 | astcryz | i try to run agi like this *70,1,agi(dtmf.php) |
| 17:13.39 | astcryz | but it returns 0, and nothing happens |
| 17:14.15 | pooh_ | astcryz: do you have the PHP agi stuff installed ? |
| 17:14.34 | astcryz | phpagi.php etc? |
| 17:14.55 | astcryz | i have copyed the files into agi-bin? |
| 17:15.16 | pooh_ | astcryz: check voip-info.org on hwo to install |
| 17:15.23 | jpablo | anyone knows how stable is an asterisk with two e1s using a digium card, with channel unical in a modified r2 for mexico? |
| 17:16.03 | *** join/#asterisk xtr (n=01928375@S01060012174cc0e1.vf.shawcable.net) |
| 17:16.37 | *** part/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 17:16.52 | astcryz | pooh_: where on voip-info? |
| 17:17.14 | ManxPower | ~docs |
| 17:17.19 | jbot | i heard docs is 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 |
| 17:19.17 | *** join/#asterisk rjuan (n=rjuan@233.Red-217-127-61.staticIP.rima-tde.net) |
| 17:27.18 | *** join/#asterisk Kokey (n=Kokey@201.153.63.79) |
| 17:30.44 | *** join/#asterisk Mother (n=mother@93.Red-80-32-127.staticIP.rima-tde.net) |
| 17:33.49 | *** join/#asterisk justinu (n=justinu@cpe-72-129-86-208.socal.res.rr.com) |
| 17:34.50 | *** join/#asterisk dippo (n=cwage@quietlife.net) |
| 17:35.04 | *** join/#asterisk SpaceBass (n=sp@c-24-125-184-203.hsd1.va.comcast.net) |
| 17:35.06 | SpaceBass | hey folks |
| 17:36.30 | mocker | What are some recommended softphones? |
| 17:36.40 | mocker | For a call center type environment. |
| 17:36.46 | ManxPower | All softphones suck. |
| 17:36.58 | mocker | ManxPower: That's what I've been noticing. :( |
| 17:37.17 | SpaceBass | tried xten and portrate...both sucked |
| 17:37.41 | implicit | SpaceBass, what's new :) |
| 17:37.47 | SpaceBass | ;) |
| 17:38.19 | SpaceBass | i know what IS new... FiOS service starting monday! 30mbs down 10mbs up! |
| 17:39.22 | justinu | SpaceBass: stfu ;) |
| 17:39.35 | mocker | I wish voip-info had a star rating next to the phones. |
| 17:39.40 | mocker | Hard to tell which ones are good. |
| 17:39.47 | ManxPower | All softphones suck. |
| 17:40.01 | mocker | ManxPower: Which hardphones do you like? |
| 17:40.29 | SpaceBass | justinu thats what they claim... we'll see about reality... |
| 17:40.32 | ManxPower | mocker, for ATA I use SIPura, for phones I use Polycom IP30x/50x/60x |
| 17:40.57 | SpaceBass | I need to get (and unlock) an pap2 to try fax |
| 17:41.23 | SpaceBass | i like the cisco hard phones, but I dont like dealing with cisco |
| 17:41.30 | SpaceBass | i like my hitachi wifi phone a lot too |
| 17:41.45 | demetrio | Nov 11 18:38:51 NOTICE[3545]: app_dial.c:777 dial_exec: Unable to create channel of type 'SIP' |
| 17:42.29 | astcryz | i cannot find out whats wrong with phpagi |
| 17:42.38 | astcryz | everything seems like it is installed correctly |
| 17:43.05 | }cytrak{ | hey guys what ports do I need to have open on a linux firewall for asterisk to work ? I know about 5060 but is there another one ? |
| 17:43.31 | }cytrak{ | I can connect using x-lite from within the asterisk box but from another linux box I can't |
| 17:43.32 | SpaceBass | }cytrak{ depends on what you want to pass... sip needs RTP ports and IAX has its own too |
| 17:43.57 | SpaceBass | by default rtp is 10000-20000 but can be limited in rtp.conf |
| 17:44.06 | SpaceBass | iax2 is 5469 |
| 17:44.06 | }cytrak{ | I need RTP and IAX |
| 17:44.13 | }cytrak{ | cool |
| 17:44.33 | astcryz | anyone here who is using phpagi? |
| 17:45.37 | }cytrak{ | are those udp ? |
| 17:45.44 | yxa | isnt iax 4569? |
| 17:45.47 | rjuan | no TCP |
| 17:45.53 | SpaceBass | sip is udp i think |
| 17:46.16 | }cytrak{ | SpaceBass: Where do you have your rtp.conf ? it's not on my /etc/rtp.conf |
| 17:46.24 | rjuan | i've read that all ports are TCP |
| 17:46.32 | rjuan | in asterisk |
| 17:46.41 | }cytrak{ | no SIP is udp |
| 17:46.47 | rjuan | ok |
| 17:46.53 | }cytrak{ | i think i read that |
| 17:47.01 | mut | all ports are udp |
| 17:47.11 | mut | sip can be tcp but asterisk only does it in udp |
| 17:47.25 | wunderkin | yes its all udp and iax is 4569 |
| 17:47.30 | }cytrak{ | ok |
| 17:47.47 | }cytrak{ | is it 4569 or 5469 ? |
| 17:47.59 | rjuan | in iax.conf is 4569 |
| 17:48.44 | demetrio | Why am I getting "unable to create channel of type SIP" errors after adjusting system time&date ? |
| 17:49.07 | *** join/#asterisk Math` (n=math@modemcable157.3-81-70.mc.videotron.ca) |
| 17:49.23 | }cytrak{ | the rtp.conf port rage settings that depends really on how many users right ? like if I have 20 users I might give a range of 50 |
| 17:49.49 | Math` | yeah, calls will start to fail if you are out of rtp ports |
| 17:49.59 | }cytrak{ | is rtp udp or tcp ? |
| 17:50.03 | Math` | udp |
| 17:50.12 | rjuan | udp i was wrong |
| 17:50.50 | Math` | RTP is like TCP, without the retransmissions and with less overheads |
| 17:50.54 | }cytrak{ | nice thanks everyone |
| 17:51.09 | }cytrak{ | got it logged in |
| 17:52.20 | *** part/#asterisk alvariux (n=unky@201.138.135.221) |
| 17:52.45 | synthetiq | why would a phone lose registration after oen call? |
| 17:52.49 | rjuan | i'm trying to play a mp3 file in musiconhold(), what i have to install? |
| 17:53.01 | }cytrak{ | mpg123 |
| 17:53.04 | }cytrak{ | i think |
| 17:53.05 | Math` | mpg123, version 0.59r |
| 17:53.09 | Math` | NOT mpg321 |
| 17:53.20 | rjuan | i have both |
| 17:53.32 | }cytrak{ | sorry |
| 17:53.35 | synthetiq | how can i force a reregistraion of a phone in asterisk |
| 17:53.37 | Math` | check mpg123's version... and make sure its not a symlink to mpg321 |
| 17:53.46 | rjuan | i'm using ubuntu |
| 17:54.00 | }cytrak{ | asterix comes with correct version as addon |
| 17:54.20 | rjuan | Paquete: mpg123 |
| 17:54.20 | rjuan | Nuevo: sรญ |
| 17:54.21 | rjuan | Estado: instalado |
| 17:54.21 | rjuan | Instalado automรกticamente: no |
| 17:54.21 | rjuan | Versiรณn: 0.59r-19 |
| 17:54.21 | rjuan | Prioridad: opcional |
| 17:54.34 | rjuan | sorry it's in spanish |
| 17:54.35 | Math` | sounds like the good on |
| 17:54.37 | Math` | one* |
| 17:54.43 | *** join/#asterisk DaPrivateer (i=Privatee@CRIMSON.OFF-HOURS.COM) |
| 17:55.26 | synthetiq | ?? |
| 17:55.44 | yxa | is there a small voip gateway like iaxy that supports both iax and sip? |
| 18:02.14 | docelmo | YAY! |
| 18:02.42 | SkramX | docelmo: ? |
| 18:02.44 | Math` | may you share your joy with us? :P |
| 18:03.22 | docelmo | Its friday.. I aint doing shit @ work except playing ya.. I love my job! |
| 18:03.35 | Lostfrog | bitch. :) |
| 18:03.42 | rjuan | jaj |
| 18:04.46 | docelmo | dont hate tha playa hate tha game.. :) |
| 18:05.00 | Math` | lol |
| 18:05.00 | justinu | we have today offf |
| 18:05.08 | docelmo | ok you just fucking suck |
| 18:05.11 | justinu | lol |
| 18:05.24 | Math` | Im working on my own so I'm pretty much rarely off |
| 18:05.32 | Math` | the fun of starting a business |
| 18:05.48 | justinu | yeah, that's always a bitch |
| 18:07.38 | Math` | is DUNDi properly handling multiple routes? (like... if more than a route is published on the dundi mesh, if one fails, is it going to use the other?) |
| 18:07.49 | docelmo | ya.. Im there now.. I converted NetDoctor Goup, Inc. from hosting to 10% hosting and 90% consulting.. I have so many clients I am about to pull my hair out.. I dont have enough work to justify another person.. but too much work for just me.. |
| 18:08.06 | Math` | docelmo: subcontract some of the work |
| 18:08.26 | hypa7ia | hosting is a brutal, brutal business |
| 18:08.37 | docelmo | Hense why I am getting out of it. |
| 18:08.42 | *** join/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
| 18:08.56 | hypa7ia | i got out like a year ago |
| 18:08.56 | hypa7ia | :) |
| 18:08.56 | docelmo | I am doing purely asterisk work now and its working out pretty well |
| 18:08.57 | hypa7ia | nice |
| 18:09.08 | hypa7ia | i'm done at my current employer at the end of the month |
| 18:09.10 | Math` | yeah setting up PBXs sounds like a lucrative industry |
| 18:09.18 | hypa7ia | gonna start doing asterisk work and security work :) |
| 18:09.22 | Alric | Sales can be difficult :) |
| 18:09.24 | Math` | nice |
| 18:09.42 | docelmo | Well I have a friend of mine sell them then I do out and install and give him half the labor profits |
| 18:09.42 | Math` | Alric: thats why u need someone in marketting when you do that |
| 18:09.51 | hypa7ia | yxa: this one can be flashed to sip or iax: http://www.voipdepot.ca/index.php?main_page=product_info&cPath=1&products_id=28&zenid=8d71a89ba6eb8abecbd74245e27ee5a5 |
| 18:09.54 | lunk | webpages and all support is below AOL phone tech support |
| 18:09.58 | Math` | well someone that fetches the clients, so you can concentrate on setting up solutions |
| 18:09.58 | hypa7ia | or h323 or mgcp :p |
| 18:10.05 | hypa7ia | lunk: yup |
| 18:10.30 | Math` | GNet's making ata? didnt know |
| 18:10.32 | Alric | Math`: Small operations don't always have that luxury :) |
| 18:11.10 | hypa7ia | Math`: yup, the manufacturer was at the last toronto asterisk UG meeting |
| 18:11.17 | Math` | hehe |
| 18:11.32 | Math` | Mediatrixes can be flashed to sip/h323/mgcp too... but no iax :( |
| 18:11.51 | hypa7ia | so in theory it should only take an hour to move 200gb over usb2 |
| 18:11.54 | hypa7ia | WE SHALL SEE |
| 18:11.54 | hypa7ia | :) |
| 18:12.05 | Math` | in theory, communism works |
| 18:12.26 | hypa7ia | in theory, your mom |
| 18:12.29 | hypa7ia | ;p |
| 18:12.49 | lunk | lol |
| 18:12.51 | hypa7ia | too much data recovery and too little play makes hypa7ia something something |
| 18:12.55 | Math` | parse_english.c:134: Missing verb |
| 18:13.01 | hypa7ia | haha |
| 18:13.28 | Math` | gnet ata's are damn cheap |
| 18:13.37 | Math` | less expensive than SIPuras :o |
| 18:13.41 | rjuan | anybody knows a good asterisk-GUI |
| 18:13.43 | rjuan | ?? |
| 18:13.47 | lunk | rjuan: AMP |
| 18:14.01 | rjuan | where I download it? |
| 18:14.19 | lunk | someone probably has a hot link |
| 18:14.28 | Math` | ~AMP |
| 18:14.29 | jbot | well, amp is a web based interface for configuring Asterisk. See http://amp.coalescentsystems.ca/ |
| 18:14.29 | Math` | lets check if the bot has one |
| 18:14.30 | lunk | yea, where's the bot? |
| 18:14.30 | lunk | haha |
| 18:14.37 | Math` | it did :) |
| 18:14.41 | hypa7ia | jbot is mad slow |
| 18:15.54 | rjuan | thanks |
| 18:16.09 | lunk | it's not even winter yet and the molasses has coagulated |
| 18:16.17 | lunk | that makes me a sad panda |
| 18:19.35 | synthetiq | how can i force a reregistraion of a phone in asterisk ? |
| 18:19.52 | synthetiq | or send a sip ping or whatever to it |
| 18:20.01 | Math` | call it? |
| 18:20.22 | Math` | or check if the qualify= option does what you want |
| 18:21.07 | synthetiq | i ahve qualify set to yes |
| 18:24.23 | *** join/#asterisk SplasPood (n=sp@paravolve.net) |
| 18:25.42 | *** join/#asterisk Math` (n=math@modemcable157.3-81-70.mc.videotron.ca) |
| 18:26.31 | javo | i have not been able to find any information on this anywhere, so i am not sure if it is possible. But, can I make an extension that I could dial when my zap phoneline is ringing to answer it? |
| 18:26.45 | justinu | i can't believe sipura doesn't have gsm support |
| 18:27.07 | javo | i know about zapbarge to listen in, but is it possible to listen and speak? |
| 18:28.03 | Lostfrog | grr.. I hate g729. |
| 18:28.22 | SpaceBass | javo why not just have * pick up the line and ring a phone? |
| 18:28.37 | Lostfrog | javo: wouldn't that be a conference? |
| 18:28.41 | javo | because i want to have a chance to answer it on my cordless phone |
| 18:28.43 | Lostfrog | Or a 3-way call? |
| 18:28.44 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 18:29.10 | SpaceBass | javo still not following... not being difficult, just think there might be another answer |
| 18:29.16 | justinu | why do you hate g729? because of the lame ass licenensing stuff? |
| 18:29.29 | SpaceBass | javo whats the set-up like currently? |
| 18:29.30 | Lostfrog | No.. it just sounds like sh*t to me. |
| 18:29.46 | Lostfrog | I'll pay $50 for the license.. |
| 18:29.55 | Lostfrog | I just hate it. |
| 18:30.15 | justinu | it does kinda sound like shit |
| 18:30.33 | javo | my situation is: i have a roommate who has a regular cordless phone connected to the phoneline. I have my * box connected to thesame line. I want the answering machine on the regular phone to pick up if no one answers, but i also want the ability to answer the call with my ip phone |
| 18:30.40 | rjuan | Try running ./install_amp --username=user --password=pass (using your own user and pass) |
| 18:30.53 | ManxPower | javo, "show application WaitForRing" |
| 18:30.56 | javo | eventually i will buy an ata, but until then, i want a good way to handle this |
| 18:30.57 | SpaceBass | javo got it! ... leme think |
| 18:31.17 | rjuan | any idea? |
| 18:31.32 | rjuan | i'm installing AMP |
| 18:31.44 | ManxPower | rjuan, *nod* This is an Asterisk channel, not an AMP channel. |
| 18:31.50 | ManxPower | Try #amportal or something like that |
| 18:32.00 | SpaceBass | javo one thought... in the dial plan you might be able to have an extension that dials the zap channel but doesnt send any digits... basically takes it off hook |
| 18:32.08 | rjuan | sorry |
| 18:32.32 | ManxPower | Yes. Dial(Zap/1/) (notice the trailing /( |
| 18:32.36 | ManxPower | ..er..trailing / |
| 18:32.39 | javo | SpaceBass, i think then it will tell me that the zap is busy |
| 18:32.45 | SpaceBass | javo exten -> 1234,1,Dial(ZAP,1,) |
| 18:32.49 | *** join/#asterisk BladeRunner05 (n=feelme@adsl-ull-58-69.44-151.net24.it) |
| 18:32.51 | SpaceBass | what ManxPower said |
| 18:32.59 | Lostfrog | I'm going to cry.. I can't find my crimping tool. |
| 18:33.09 | BladeRunner05 | IronHeli: Hi my friend |
| 18:33.11 | SpaceBass | Lostfrog I'll trade you, I cannot find my rj-45 ends |
| 18:33.25 | javo | ManxPower, will that work if the phone is ringing already? |
| 18:33.36 | Lostfrog | I can give you some of them. |
| 18:33.36 | ManxPower | javo, Try it and see |
| 18:34.00 | *** join/#asterisk Kero (n=kero@a197111.upc-a.chello.nl) |
| 18:34.05 | BladeRunner05 | Hi all |
| 18:34.19 | BladeRunner05 | I need information using AVM B1 card |
| 18:34.38 | SpaceBass | javo another, crazy idea... see if your phone company will do a call forward and ring... mine will... then get a broadvoice byod lite plan (unlimited incoming for $5/month) and have it forward and ring to that |
| 18:34.52 | SpaceBass | then both the regular POTS and the sip 'lines' will ring |
| 18:35.32 | Lostfrog | Or get a second FXS port and plug is phone into it. |
| 18:35.37 | Lostfrog | his phone, even. |
| 18:36.12 | SpaceBass | might be the best bet... get a pap2 for like $40 from compusa or something, unlock it and use it |
| 18:36.25 | javo | hrmm that would probably work, but i was hoping for a asterisk config fix for it |
| 18:36.29 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 18:36.34 | Dr-Linux | justinu: :) |
| 18:36.37 | justinu | broadvoice byod DID's seemed a lot more expensive than 5 bucks |
| 18:36.45 | justinu | hey Dr-Linux |
| 18:36.47 | Lostfrog | The lite plan. |
| 18:36.58 | justinu | i thought that was just for outbound |
| 18:37.05 | Dr-Linux | system specs |
| 18:37.08 | SpaceBass | justinu after tax, I pay like $9.00 for one (with a 1800) and $7.95 for the other |
| 18:37.11 | justinu | it started at 5 bucks, but as soon as I added a DID, it went to like 12 |
| 18:37.26 | justinu | hmmm |
| 18:37.37 | Lostfrog | <PROTECTED> |
| 18:37.37 | Lostfrog | You'll get your pick of numbers, all of our basic and advanced features, including 100 minutes of outbound calls in the United States or Canada, AND the option to upgrade to our unlimited plans anytime for no additional charge. Additional outbound minutes in the US and Canada are just 3.9 cents per minute. |
| 18:37.46 | SpaceBass | all of mine have a DID for the $5... it got expensive ($9.95) when I added the 1800 DID |
| 18:37.54 | justinu | cool, thanks! |
| 18:38.11 | SpaceBass | what I DESPERATLY want is a local DID so I can get rid of my ZAP channels all together |
| 18:38.47 | Math` | you don't like zap channels? |
| 18:38.52 | SpaceBass | HATE THEM |
| 18:39.03 | wunderkin | he means the tdm card |
| 18:39.04 | Math` | why? |
| 18:39.08 | SpaceBass | get really bad sound from time to time..not just echo, but distortion |
| 18:39.10 | Lostfrog | I'm not getting rid of my physical lines. |
| 18:39.12 | javo | well i did the Dial(Zap/1/1) it works when the phone is on the hook, but not when it is ringing |
| 18:39.13 | SpaceBass | x100p clone cards |
| 18:39.27 | javo | er Dial(Zap/1/) |
| 18:39.33 | SpaceBass | javo I've found that * has a hard time sharing POTS lines |
| 18:39.41 | justinu | that's what sucks about broadvoice, no DIDs in my area code |
| 18:39.42 | *** join/#asterisk |cleric| (n=dacleric@p5482A44A.dip0.t-ipconnect.de) |
| 18:40.05 | SpaceBass | justinu I have the same problem, but since my phone company has unlimited long distance, forwarding to a LD number wasn't an issue |
| 18:40.07 | javo | is there nothing like zapbarge that would allow me to talk as well as listen? |
| 18:40.24 | marcus2 | how do i put a "any number" in my dialplan? |
| 18:40.26 | SpaceBass | jav I searched for just that when I had call waiting issues and never found anything |
| 18:40.34 | Math` | marcus2: _X. |
| 18:40.35 | javo | hrmm |
| 18:40.37 | justinu | SpaceBass: interesting, but I'm totally off the RBOC network |
| 18:40.46 | SpaceBass | RBOC? |
| 18:40.55 | BladeRunner05 | I need information using AVM B1 card |
| 18:41.06 | justinu | regional bell operating company, aka theives |
| 18:41.12 | SpaceBass | oh |
| 18:41.25 | *** join/#asterisk mog_work (n=mogorman@gateway.digium.com) |
| 18:41.27 | SpaceBass | i agree... I'd LOVE to get rid of my POTS lines and just use voip... but cannot find a local DID |
| 18:41.57 | SpaceBass | well... i can get a T-1 with a PRI for a few hundred a month :) |
| 18:42.05 | Lostfrog | Get a T1 card and a decent channel bank.. |
| 18:42.40 | SpaceBass | yeah, wife would LOVE that |
| 18:42.52 | justinu | it's l337, yo |
| 18:42.53 | SpaceBass | she already doesnt understand "why we have to have office phones" |
| 18:42.56 | shido6 | how many lines? |
| 18:43.11 | marcus2 | math; i mean, literally, any number. not just any single digit. |
| 18:43.20 | Math` | ah |
| 18:43.24 | Lostfrog | marcus2: LISTEN! _X. |
| 18:43.27 | SpaceBass | POTS numbers cannot be ported to VoIP providers can they? |
| 18:43.30 | justinu | SpaceBass: what's wrong with office phones? |
| 18:43.33 | Math` | isnt that what I said? |
| 18:43.33 | marcus2 | er my ad, sorry |
| 18:43.34 | justinu | SpaceBass: yes, they can. |
| 18:43.35 | marcus2 | i missed the . |
| 18:43.37 | Math` | hehe |
| 18:43.51 | SpaceBass | justinu does the VoIP provider have to have DIDs in my area? |
| 18:43.54 | marcus2 | thanks :) |
| 18:43.56 | justinu | don't think so |
| 18:44.11 | SpaceBass | justinu so in theroy I could call BroadVoice and port my number to them? |
| 18:44.15 | marcus2 | we're turning up a PRI in 15 minutes, i wonder how this will go |
| 18:44.20 | justinu | SpaceBass: in theory |
| 18:44.23 | Lostfrog | And actually, that will only match numbers 2 digits and longer. |
| 18:44.52 | Lostfrog | But _. is a bad idea. |
| 18:45.16 | *** join/#asterisk _iCeb0x_ (n=sysadmin@200.246.26.130) |
| 18:45.26 | marcus2 | this is just for testing |
| 18:45.26 | _iCeb0x_ | Hello, all |
| 18:45.32 | marcus2 | i just want to answer any incoming call on the pri |
| 18:45.49 | _iCeb0x_ | I need some help with E1 cards, connectors and figuring some stuff... |
| 18:46.21 | _iCeb0x_ | Is this the right place? |
| 18:46.30 | _iCeb0x_ | I mean, to ask about this kind of stuff? |
| 18:47.08 | shido6 | you're here |
| 18:47.13 | shido6 | this is it :) |
| 18:47.35 | Lostfrog | No.. you need to go to #E1cardsconnectorsandfiguringsomestuff |
| 18:47.36 | Lostfrog | lol |
| 18:47.43 | _iCeb0x_ | Good one |
| 18:48.51 | _iCeb0x_ | Well... Let's go... I have this E1 connection, 32 voice channels via a 34GHz radio link. There's a lot of equipment here and a pair of coax cables come out of the whole shebang and connect to a hardware PBX we're using now |
| 18:49.22 | _iCeb0x_ | All E1 cards I have seen use RJ-45 connectors... |
| 18:50.03 | _iCeb0x_ | I want to use Asterisk and I an sure I need an E1 card, but I can't figure out how to connect the stuff together |
| 18:50.55 | Lostfrog | Yes T1/E1 cards have what looks like RJ45 sockets, and into which RJ45 connectors fit. |
| 18:51.08 | Lostfrog | What are you connecting to your E1 card? |
| 18:51.45 | _iCeb0x_ | I am still planning on using an Asterisk server. I am not using it, nor do I have hardwware for it *yet* |
| 18:52.04 | javo | hrmm this is interesting. i added a zapbarge just for fun, and it actually does let me talk |
| 18:52.14 | Lostfrog | well.. a T1/E1 card would be used for one of two things. |
| 18:52.35 | javo | in fact it does exactly what i was looking for |
| 18:52.45 | Lostfrog | Connect to your telco provider, or connect to a channel bank/PBX. |
| 18:53.09 | javo | thanks for the help though everyone |
| 18:53.22 | Lostfrog | SpaceBass, come crimp some RJ11 for me. |
| 18:53.25 | SpaceBass | javo nice! |
| 18:53.31 | _iCeb0x_ | I Lostfrog: Yes... I want to connect to the telco |
| 18:53.36 | SpaceBass | Lostfrog you bring the ends, I'll bring the crimper |
| 18:53.45 | Lostfrog | Got them right here. |
| 18:54.05 | SpaceBass | gotta find my wireing diagram too.... |
| 18:54.36 | Lostfrog | Then, _iCeb0x_, your telco provider will provide you with a plug into with you plug a straight through E1/T1 cable to your E1 card. |
| 18:54.45 | Lostfrog | There are pinouts for that cable on the wiki. |
| 18:55.15 | _iCeb0x_ | Lostfrog: As you can see, I am a complete newbie on telephony and Asterisk. I have been reading a lot of stuff and I am pretty confident I can set uo an asterisk server. I just need to know how to connect the server to the telco supplied equipment |
| 18:56.23 | _iCeb0x_ | Great, Lostfrog |
| 18:56.35 | _iCeb0x_ | Gonna check the Wiki now! |
| 18:56.38 | _iCeb0x_ | Thanks a lot! |
| 18:56.46 | Lostfrog | Not a problem. |
| 18:57.14 | justinu | _iCeb0x_: what you want is a 75-120ohm E1 adapter cable |
| 18:57.36 | _iCeb0x_ | justinu: Seems so. :) |
| 18:57.40 | justinu | _iCeb0x_: something like this: http://www.cisco.com/univercd/cc/td/doc/product/core/7206/port_adp/multich/6728cabl.htm |
| 18:58.25 | Lostfrog | ok.. There used to be a windows wav editor recommended on the wiki. |
| 18:58.30 | Lostfrog | Anyone remember what it was? |
| 18:58.40 | justinu | i like using the old ass cooledit 96 |
| 18:58.46 | _iCeb0x_ | Exactly. Just saw the picture. A pair of coax cables with bnc-style connectors on one end and a female RJ48c on the other |
| 18:59.24 | Inkubot | bye |
| 19:00.42 | Lostfrog | cooledit was bought out by Adobe. :( |
| 19:01.22 | marcus2 | to dial out on a zap line, i just do Dial(8885551212@Zap/1) ? |
| 19:01.50 | Lostfrog | I believe Zap/1/888... |
| 19:04.14 | Lostfrog | I haven't dialed out on a Zap line for so long. :) |
| 19:04.38 | *** part/#asterisk SpaceBass (n=sp@c-24-125-184-203.hsd1.va.comcast.net) |
| 19:05.45 | justinu | i've never dialed out on a zap line |
| 19:07.44 | oogle | Lostfrog: did you take my frog? him name is hopkin green frog |
| 19:08.48 | lunk | haha |
| 19:08.51 | ManxPower | <PROTECTED> |
| 19:09.17 | justinu | if i had a frog, i'd name it clydefrog |
| 19:09.19 | synthetiq | what would be the reason that a phone keeps losing registration after it has been called? |
| 19:09.22 | *** join/#asterisk davidg (i=Austin@adsl-66-141-55-204.dsl.austtx.swbell.net) |
| 19:09.35 | justinu | synthetiq: nat bindings closing, possible |
| 19:09.39 | davidg | hi :) |
| 19:09.54 | oogle | it's definately the latter for zaptel, ex: Zap/g3/number |
| 19:10.01 | synthetiq | i dont beleive there is nat |
| 19:10.08 | iCEBrkr | doooooo-bee-do-be-doooooooo |
| 19:10.35 | synthetiq | it makes no sense cause a phone with the same set up on the enxt desk stays registered |
| 19:11.14 | justinu | synthetiq: turn on sip debug peer <peername> |
| 19:11.30 | SkramX | ATX Represent! |
| 19:11.31 | justinu | pastebin the output from the time the phone stops working |
| 19:11.46 | synthetiq | the peer has to be reigstered for that lol |
| 19:12.08 | justinu | well, it's working, then it stops right? |
| 19:12.31 | synthetiq | when a call coems in and finishes |
| 19:12.50 | Lostfrog | audacity. |
| 19:14.14 | davidg | new question ok? |
| 19:14.46 | _iCeb0x_ | Lostfrog, justinu: thanks for the help. That link saved my day. I'll be back in a few days when my Digium E1 card arrives |
| 19:14.56 | _iCeb0x_ | I have to go now. See ya |
| 19:14.59 | justinu | _iCeb0x_: enjoy |
| 19:15.50 | davidg | Q: using portaone radius asterisk add-in behind a NAT, cannot initiate call accounting, any suggestions? |
| 19:16.17 | shido6 | dont use radius. |
| 19:16.18 | marcus2 | holy shit it works! |
| 19:16.24 | justinu | synthetiq: look at the sip debug during the time the phone stops working |
| 19:16.32 | bweschke | mog_work: s/n 59990 on a t/e410P - is that 1st gen or 2nd gen t/e410P? |
| 19:16.34 | marcus2 | i've never had an actual pstn line plugged into an asterisk server before, hehe |
| 19:16.45 | shido6 | make a dialout context, marcus2 |
| 19:16.52 | shido6 | sign up for a voip provider |
| 19:16.55 | shido6 | and make a dialout context |
| 19:16.56 | oogle | marc324: welcome to the wonderful world of zapata telephony |
| 19:16.59 | shido6 | call in from a cell phone |
| 19:17.04 | shido6 | and dial out through your voip provider |
| 19:17.14 | shido6 | or setup voicemail , you'll get a kick out of that |
| 19:17.20 | oogle | Junction Networks offers pretty good IAX termination |
| 19:18.29 | sahafeez | anyone recommend a good voip provider for the SoCal area? it is just for my home use - want to have asterisk running with a few ploycom phones |
| 19:18.43 | davidg | I'm using Asterisk as SIP server |
| 19:18.52 | iCEBrkr | sahafeez: nufone? voicepulse connect? |
| 19:19.04 | sahafeez | cool. i will have a look. |
| 19:19.22 | SkramX | voipjet.com |
| 19:21.02 | sahafeez | is there away to do 911/callerid based on a extension. let say if i have 3 asterisk boxes at remote sites with one in the center with the pri, how do i tell 911 that the call came from site 3 and not the term point of the pri? |
| 19:21.10 | marcus2 | oh i've been using asterisk for over two years |
| 19:21.28 | marcus2 | just always with iax/sip providers, never with a pri plugged right into the box |
| 19:21.33 | Igbothom_III | broadvoice is supposed to be decent, too |
| 19:21.42 | ManxPower | sahafeez, Generally you CAN'T |
| 19:21.54 | marcus2 | i have nufone, connect.voicepulse, and terravon accounts |
| 19:22.04 | sahafeez | and un-generally? is there a way |
| 19:22.59 | *** join/#asterisk Habakuk (n=andy@64.1.15.130.ptr.us.xo.net) |
| 19:23.14 | ManxPower | sahafeez, it depends on the specifics of how your local E911 center aquires it's physical address information. |
| 19:23.26 | ManxPower | Most centers do NOT use CallerID for that. |
| 19:23.27 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 19:23.29 | puzzled | hi |
| 19:23.59 | iCEBrkr | Yea, E911 is ANI based |
| 19:24.04 | sahafeez | ManxPower: can you point me to something that explains how they do it? |
| 19:24.15 | ManxPower | iCEBrkr, It could be ANI based, it could be BTN based. |
| 19:24.20 | iCEBrkr | sahafeez: Well, that's the problem with VoIP |
| 19:24.23 | ManxPower | sahafeez, Contact your local E911 center and ASK THEM. |
| 19:24.25 | iCEBrkr | ManxPower: Ahhhh |
| 19:24.44 | ManxPower | BTN would be the most accurate before VoIP existed. |
| 19:25.11 | marcus2 | now i get to see if i have any luck hooking our merlin magix system up to our asterisk server with a pri |
| 19:25.21 | marcus2 | anyone here have experience with magix pbxs? :) |
| 19:25.53 | iCEBrkr | ManxPower: I thought I read somewhere it's mainly ANI based. Isn't that basically the same as BTN? |
| 19:26.07 | ManxPower | iCEBrkr, no. ANI can be changed, BTN cannot. |
| 19:26.10 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 19:26.13 | iCEBrkr | Ahh ok |
| 19:26.33 | ManxPower | But it ALL DEPENDS on the E911 center. |
| 19:26.36 | tzanger | well BTN's where # the bill goes to |
| 19:26.45 | tzanger | BTN has nothing to do with location |
| 19:26.54 | tzanger | it's strictly for accounting control if I'm not mistaken |
| 19:26.59 | ManxPower | tzanger, perhaps it's a different TLA I'm thinking of. |
| 19:27.15 | tzanger | i.e. a company with 16 offices can have the same BTN which gives the right billing address |
| 19:27.25 | ManxPower | Regardless, if it was easy, don't you think all the VOIP companies would be providing 911 service? None of them do, even with an FCC mandate. |
| 19:27.47 | tzanger | ANI is supposed to be physical location based |
| 19:28.15 | iCEBrkr | ManxPower: Well, that's only cuz the VoIP <-> PSTN interconnects-- the BTN would be whatever the address the interconnect is located. |
| 19:28.25 | ManxPower | If you can change your ANI, then it's not reliable |
| 19:28.29 | iCEBrkr | It's not like they can figure out where the hell you are based on your IP? |
| 19:28.37 | ManxPower | iCEBrkr, correct. |
| 19:28.56 | iCEBrkr | Unless the VoIP would take your billing record ( Which could be innacurate anyhow ) And pass that information along as BTN/ANI... |
| 19:29.10 | iCEBrkr | Which is a jackedup situation regardless. |
| 19:29.31 | sahafeez | well i just did some reading. you can change your ANI. and ANI is listed at the method for E911. |
| 19:29.34 | ManxPower | They could pass a POTS line number as the ANI for the call. The emergency people will still not arrive at the right location. |
| 19:29.36 | iCEBrkr | Spoof 911 calls would be a big problem. |
| 19:29.48 | ManxPower | That's why I always keep at least 1 POTS line at EVERY location |
| 19:30.01 | ManxPower | Even it's only connected to a big red phone on the wall. |
| 19:30.05 | iCEBrkr | ManxPower: The good ol' red phone :P |
| 19:30.11 | docelmo | lathos thats like me hugging my Cisco AS5850 |
| 19:30.26 | *** join/#asterisk Rowter (n=SilverDr@201.135.26.195) |
| 19:30.27 | sahafeez | so I would need to make sure the lines from the remote sites when dialing 911 always pickup the same line, then i could have the ANI recorded changed |
| 19:30.40 | ManxPower | sahafeez, good luck with that. |
| 19:30.56 | iCEBrkr | sahafeez: your best bet is to do what ManxPower is doing. |
| 19:31.03 | Rowter | there is any way to know what pri line is in use, or how many are free? show zap channel ? |
| 19:31.19 | sahafeez | if i have a 4 floor office with 1000 people that does not work. |
| 19:31.22 | iCEBrkr | sahafeez: I'd suggest making _911 go through a PSTN line, but if the server fails, you're screwed. |
| 19:31.48 | sahafeez | unless i put RED PHONES in 5 places on each floor. then there is the issue of the panic'd person remembering to run to the RED PHONE |
| 19:31.49 | ManxPower | Rowter, asterisk -rx "show channels" | grep Zap | wc -l |
| 19:32.09 | Rowter | thanks ManxPower |
| 19:32.28 | iCEBrkr | sahafeez: Well, that's why the FCC has their panties in a bunch over this E911 + VoIP providers. |
| 19:32.32 | stbain | ok... found the hookflash codes I need to integrate asterisk with my existing SX-2000 Mitel (via the ONS/Analog ports). Question is, can I use the dialogic card out of my old voicemail system? |
| 19:32.40 | stbain | ~dialogic |
| 19:32.42 | jbot | i heard dialogic is no more. it has ceased to be. it shuffled off its moral coil and went to meet Intel |
| 19:32.48 | stbain | thnx, jbot |
| 19:33.57 | sahafeez | hum, the solutions is static IPs and a registration database. |
| 19:34.26 | iCEBrkr | sahafeez: Nope. |
| 19:34.43 | sahafeez | iCEBrkr: i am all ears for your solution |
| 19:34.57 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
| 19:34.57 | iCEBrkr | sahafeez: Cuz unless you take ownership of those IPs, the address on those IPs is your providers. |
| 19:35.14 | iCEBrkr | sahafeez: And if you're a multi-office corp. Typically the ownership of those IPs will have home offices address. |
| 19:35.37 | iCEBrkr | GeoIP isn't reliable. |
| 19:36.25 | sahafeez | iCEBrkr: you seem to miss the point, static ip with a ddns type setup - were you have to update a registry when you connect. you can ignore who owns the ips |
| 19:36.32 | iCEBrkr | sahafeez: You'll need PSTN in each office. Setup red-phones and you could setup exten => _911,1,Dial(Zap/1).... |
| 19:36.37 | *** join/#asterisk colinm_ (n=colol@VDSL-130-13-9-157.PHNX.QWEST.NET) |
| 19:36.44 | iCEBrkr | But like I said, if the PBX fails and someone goes to dial 911.. You're screwed. |
| 19:36.44 | sahafeez | and they can be non-static at that point |
| 19:37.07 | iCEBrkr | sahafeez: No, I'm talking liability here. |
| 19:39.31 | marcus2 | so, no merlin/magix pbx experts here? :) |
| 19:40.04 | ManxPower | marcus2, In spite of what Merlin says, there's nothing special about their PBXs. |
| 19:40.12 | ManxPower | Their PRI is the same as everyone elses. |
| 19:40.15 | marcus2 | manx; clearly |
| 19:40.23 | marcus2 | well, i dont expect it to be |
| 19:40.32 | justinu | i thought merlin was a key system |
| 19:40.34 | marcus2 | i just have no clue how to make the damn thing send certain calls to the pri |
| 19:41.55 | *** join/#asterisk |cleric| (n=dacleric@p5482A44A.dip0.t-ipconnect.de) |
| 19:42.00 | Lostfrog | ick.. 'sox test.wav test.gsm' ends up sounding like crap. |
| 19:42.10 | Lostfrog | And yes.. the wav is mono 8k. |
| 19:42.18 | iCEBrkr | Lostfrog: So read the Wiki and put some command line parameters in there |
| 19:42.28 | Lostfrog | will do. |
| 19:43.29 | iCEBrkr | Lostfrog: I'd personally use a higher quality wave file so in the end sox can sort it all out and make it 8k mono |
| 19:43.34 | marcus2 | if i'm configuring zaptel.conf for span 2 of my te410p, do i refer to channels 1-24 or 25-48? |
| 19:43.57 | ManxPower | marc324, you can never have more than 1 channel with the same channel number |
| 19:44.04 | Lostfrog | I recorded it using Monitor. |
| 19:44.09 | ManxPower | Sp 25-48 would be right |
| 19:44.11 | marcus2 | i kind of figured that for zapata.conf |
| 19:44.12 | iCEBrkr | Lostfrog: Use Record |
| 19:44.18 | marcus2 | just wondering if zaptel.conf was the same |
| 19:44.20 | iCEBrkr | Lostfrog: and make it save as GSM by default |
| 19:44.45 | marcus2 | so my config should look like this: |
| 19:44.45 | marcus2 | span=2,0,0,esf,b8zs |
| 19:44.45 | marcus2 | bchan=25-47 |
| 19:44.45 | marcus2 | dchan=48 |
| 19:44.58 | marcus2 | (i'm getting sync from span 1, and i want to provide sync to the equipment at the other end of span 2) |
| 19:45.15 | iCEBrkr | exten => *55,2,Record(/var/lib/asterisk/sounds/test:gsm) |
| 19:45.15 | iCEBrkr | exten => *55,3,Playback(test) |
| 19:45.24 | iCEBrkr | or something |
| 19:50.20 | Lostfrog | I recorded the sound off of my existing PBX.. It's not very easy to use Record that way. |
| 19:51.12 | Lostfrog | I'll just use it as a wav for now. |
| 19:53.22 | marcus2 | mmm |
| 19:53.24 | marcus2 | winspm at 2400 baud |
| 19:53.26 | marcus2 | love it |
| 19:56.07 | mocker | I don't even know who makes it. |
| 19:56.11 | hypa7ia | crappytone? |
| 19:56.16 | mocker | A-100 |
| 19:56.21 | mocker | If that means anything to anyone. |
| 19:56.42 | hypa7ia | yeah |
| 19:56.48 | hypa7ia | that sounds like a budgetone |
| 19:57.55 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 19:58.09 | Igbothom_III | Budgie Tone |
| 20:01.06 | Lostfrog | Clipcom? |
| 20:01.06 | *** join/#asterisk n4y (n=tmalkut@fw.orasoft.net.pl) |
| 20:01.33 | Lostfrog | http://www.clipcomm.co.kr/eng/e_product/e_product_bs_a100.html |
| 20:03.35 | marcus2 | can i do d4/ami with a zaptel line? |
| 20:04.26 | *** join/#asterisk hugo-v6 (n=hugo@ns1.bundesunixminister.de) |
| 20:04.32 | hugo-v6 | hiho |
| 20:05.14 | hugo-v6 | q: which *-variable contains the extension called? wass it ${DNID} |
| 20:07.42 | justinu | finally got the polycoms uploading logs/directories onto the HTTP server |
| 20:07.42 | *** join/#asterisk paxr0 (n=jjhhjkh@200-126-68-249.bk5-dsl.surnet.cl) |
| 20:07.48 | ManxPower | hugo-v6, it's ${EXTEN} |
| 20:08.07 | hugo-v6 | ManxPower: d'oh. thank you verry much. |
| 20:10.13 | *** part/#asterisk Habakuk (n=andy@64.1.15.130.ptr.us.xo.net) |
| 20:12.36 | mocker | Lostfrog: atcom.cn it looks like. |
| 20:13.28 | mocker | Lostfrog: AT-320 series. |
| 20:13.33 | mocker | Not A100, ;) |
| 20:13.58 | fugitivo | atcom sucks! |
| 20:14.10 | fugitivo | don't buy that phone |
| 20:14.24 | mocker | fugitivo: It's already on my desk. :( |
| 20:14.55 | fugitivo | sorry :( |
| 20:14.56 | Lostfrog | Not having RJ-25 plugs suck. |
| 20:15.18 | Lostfrog | Not that I'm going to run out and buy some. |
| 20:16.20 | ManxPower | RJ-25? |
| 20:16.34 | Lostfrog | 6pin modular RJ. |
| 20:16.51 | ManxPower | What country are you in that doesn't use RJ-21X, RJ-48, RJ-11 or RJ-12? |
| 20:17.16 | Lostfrog | Hmm.. wait a second.. These RJ-11s are RJ-25. :) |
| 20:17.34 | Lostfrog | RJ-11/RJ-14/RJ-25 |
| 20:18.04 | ManxPower | I think I have a phone line problem. |
| 20:18.44 | Lostfrog | How so? |
| 20:19.39 | mocker | fugitivo: The buttons on this are silly. |
| 20:19.51 | mocker | fugitivo: Why do I need a button dedicated to telling me my IP? |
| 20:19.51 | fugitivo | mocker: and it doesn't hangup sometimes |
| 20:19.52 | mocker | ;) |
| 20:19.55 | fugitivo | it's too light |
| 20:28.37 | marcus2 | so i took a soldering gun to my linksys pap2 |
| 20:28.53 | marcus2 | now line2 is actually the second line on the jack for line1 |
| 20:29.05 | marcus2 | i cant believe they didnt put those traces on the pcb |
| 20:29.11 | }cytrak{ | how come I can't get the any of the proxy 1- 9 setting of x-lite to work ? |
| 20:29.27 | }cytrak{ | I disable the default but it still won't work |
| 20:29.49 | }cytrak{ | nevermind |
| 20:35.42 | *** join/#asterisk zeedo (n=zeedo@80.68.92.188) |
| 20:36.37 | p1tst0p | Hi, wonder if someone can point me in the right direction.. i have a Avaya 4602 IP phone with sip firmware, connected to my asterisk.. all working great.... however, it has a 2 line lcd display, and i was wondering if i can make that display the queue name when calls come into my queues ? |
| 20:43.47 | }cytrak{ | I got a problem with the sip.con setup ... If I create a sip extension like [joe] and I want to dial that user extension shouldn't I be able to dial the extension number ? |
| 20:44.08 | }cytrak{ | I used regexten=7878 |
| 20:44.19 | }cytrak{ | [joe] regexten=7878 |
| 20:44.19 | marcus2 | ok, has anyone here ever hooked an asterisk server up to an existing pbx? |
| 20:44.36 | marcus2 | do i need a crossover cable or something between my digium card and the pri board in the pbx? |
| 20:44.45 | }cytrak{ | marcus2: I will be doing that soon .. waint for the digium card |
| 20:45.02 | }cytrak{ | marcus2: I think you need a T1 cable |
| 20:45.11 | marcus2 | i cant get this stupid magix to do anything useful |
| 20:45.16 | marcus2 | what is a 't1 cable' ? |
| 20:45.27 | marcus2 | i just use normal straight-thru rj45 cables between my telco PRIs and the digium card |
| 20:45.42 | *** join/#asterisk Phantom-X (n=there@c-212-031-186-047.cust.broadway.se) |
| 20:45.44 | }cytrak{ | that's what I meant sorry |
| 20:46.09 | }cytrak{ | a good cat5e or cat6 |
| 20:46.27 | marcus2 | but do i need a crossover cable between the pbx and the digium card? |
| 20:47.54 | wunderkin | yes and i think yes |
| 20:47.55 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 20:48.02 | bsdfreak | heh |
| 20:48.18 | marcus2 | i am assuming the pinouts are not the same as an ethernet crossover |
| 20:48.25 | wunderkin | i think it would follow the same rules as ethernet, for a t1 you can use an ethernet cable but you cant use a regular ethernet crossover, you need a t1 crossover |
| 20:48.25 | stbain | if Irecall correctly, t1 uses 1,2,4,5 |
| 20:48.50 | stbain | so... going by color |
| 20:49.40 | stbain | T1 crossover would be... WO,O,X,WB,B,X,X to WB,B,X,WO,O,X,X where W = White, O = Orange, B = Blue, and the X represents the green & brown pairs |
| 20:49.40 | marcus2 | ah yes, you're right, 1/2/4/5 |
| 20:50.04 | stbain | and slap an extra X on the end of both of those for the last brown wire |
| 20:50.17 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 20:50.56 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 20:55.00 | *** join/#asterisk sigwerk (n=sigwerk@athena.rootednetworks.com) |
| 20:57.39 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 20:57.47 | }cytrak{ | does anyone know what is the difference between username and the authentication user |
| 20:57.51 | }cytrak{ | on the x-lite ? |
| 20:58.04 | justinu | username goes in the sip URI |
| 20:58.15 | justinu | auth user is used for the authentication challenge |
| 21:00.38 | rculp | I have a quick question about getting outside numbers to go into a voicemail tree |
| 21:00.50 | rculp | I know how to make a phone ring a direct sip extension |
| 21:01.12 | rculp | but having issues making it answer without setting anything |
| 21:04.56 | Lostfrog | Is there a way to set a variable when a call enters a context even though it will match an included context extension? |
| 21:05.00 | marcus2 | woohoo, the crossover cable gives me some green |
| 21:05.27 | *** join/#asterisk riksta (n=rick@62.6.163.90) |
| 21:05.34 | *** join/#asterisk pooh_ (n=hfwang@cust.15.241.adsl.cistron.nl) |
| 21:05.37 | shido6 | good to go |
| 21:05.41 | pooh_ | evening |
| 21:05.44 | marcus2 | no to figure out this stupid magix pbx |
| 21:06.02 | Phantom-X | hmmm I got a little proch check it here : http://www.gophi.rotfl.pl/i/misc/blind.gif :) |
| 21:06.17 | marcus2 | asterisk still shows the channels as being in alarm |
| 21:06.20 | pooh_ | looking for a RealTime guru pls |
| 21:07.16 | pooh_ | I see different resut when using Realtime e.g.: |
| 21:07.16 | pooh_ | (17:42:21) pooh_: res_config_mysql.c: MySQL RealTime: Update SQL: UPDATE sip_accounts SET ipaddr = '192.168.5.118', port = '5060', regseconds = '1131531041', username = '9011', fullcontact = 'sip:9011@192.168.5.118:5060' WHERE name = '9011' |
| 21:07.17 | pooh_ | (17:42:21) pooh_: res_config_mysql.c: MySQL RealTime: Update SQL: UPDATE SIP_accounts SET ipaddr = '192.168.5.118', port = '5060', regseconds = '1131563860', username = '9011' WHERE name = '9011' |
| 21:07.35 | }cytrak{ | man I hate this username/auth mismatch stuff .... |
| 21:08.14 | }cytrak{ | If I use the username that I setup on the sip.com for [7878] it doesn't work |
| 21:08.19 | SkramX | cd /web |
| 21:08.20 | SkramX | oops |
| 21:09.05 | stbain | marcus2: check clock source and make sure only one side of that T1 connection has clocking set up |
| 21:09.45 | marcus2 | the pbx says it is getting clock from the digium card |
| 21:09.45 | marcus2 | i think |
| 21:09.48 | marcus2 | "clock: loop" |
| 21:10.00 | stbain | tell it clock: line |
| 21:10.28 | stbain | (if it works like a Cisco) |
| 21:11.31 | *** join/#asterisk Flauto (n=zhao@c-24-14-197-214.hsd1.il.comcast.net) |
| 21:11.49 | Flauto | hi all |
| 21:12.22 | iCEBrkr | blah blah blah |
| 21:12.27 | iCEBrkr | Is it time to go home yet? |
| 21:12.32 | Lostfrog | Stupid question: Can you do a dial like Dial(sip/101@provider.com/8005551212)? |
| 21:12.52 | Flauto | i cannot modprobe when i was installed my x100p |
| 21:12.55 | Flauto | any idea |
| 21:13.17 | pooh_ | 'was installed'?? what are the errors ? |
| 21:13.25 | *** join/#asterisk MajestiK (n=MajestiK@S0106000024c058cc.ed.shawcable.net) |
| 21:13.41 | iCEBrkr | Lostfrog: If there's a 101 @ provider |
| 21:13.47 | Flauto | pooh, you remember that you were telling me that there is something i need to change? |
| 21:14.05 | pooh_ | Flauto: not exactly, refresh my memory pls |
| 21:15.22 | *** join/#asterisk n4y (n=tmalkut@fw.orasoft.net.pl) |
| 21:15.32 | marcus2 | uh mci screwed up |
| 21:15.39 | marcus2 | they gave us 450 DIDs with this PRI |
| 21:15.55 | iCEBrkr | sweet |
| 21:16.08 | marcus2 | we asked for 200. nice. |
| 21:16.20 | *** join/#asterisk rene- (n=rene-@201.154.240.182) |
| 21:16.20 | iCEBrkr | I wish I could get into a colo that has a PRI I could use for a few weeks for testing... |
| 21:16.27 | Lostfrog | That didn't work, iCEBrkr. |
| 21:16.41 | Lostfrog | Complained of congestion. |
| 21:16.48 | iCEBrkr | Lostfrog: then there's no 101 extension |
| 21:17.00 | iCEBrkr | I do Asterisk->Asterisk calls like that |
| 21:17.25 | ManxPower | Our CLEC can't even find an available block of 100 DIDs between 36xx and 79xx |
| 21:17.27 | rene- | hello, where does echocancelwhenbridged applies? zap to zap calls or zap to iax/sip calls? |
| 21:17.45 | *** join/#asterisk Spacebar (n=stingray@stingr.net) |
| 21:17.48 | Lostfrog | Hmm.. Dial(sip/101@provider.com) works. |
| 21:17.48 | ManxPower | rene-, as it says, zap to zap calls |
| 21:17.51 | Lostfrog | I get a dial tone. |
| 21:17.52 | marcus2 | they gave us 2251 thru 2699 |
| 21:17.54 | marcus2 | sweet. |
| 21:18.18 | pooh_ | ManxPower: it says BRIDGED, al channels can be bridged right ? |
| 21:18.31 | iCEBrkr | Lostfrog: um, I think your '101' is a context actually. |
| 21:18.38 | rene- | is it because analog to analog wouldnt need echo cancelling right? |
| 21:18.46 | pooh_ | rene-: wrong |
| 21:18.53 | Spacebar | anybody here having success dealing with Digium TE210P card on x86_64 ? |
| 21:18.54 | marcus2 | how often do telcos screw up like that? |
| 21:19.02 | ManxPower | rene-, generally, analog/analog calls don't have echo |
| 21:19.04 | iCEBrkr | Lostfrog: wait.. I'm confused. nevermind me |
| 21:20.49 | Lostfrog | ok.. then I don't know how to make outbound calls using my audiocodes MP-108 |
| 21:22.27 | Spacebar | :( |
| 21:23.05 | p1tst0p | Hi, wonder if someone can point me in the right direction.. i have a Avaya 4602 IP phone with sip firmware, connected to my asterisk.. all working great.... however, it has a 2 line lcd display, and i was wondering if i can make that display the queue name when calls come into my queues ? |
| 21:23.09 | marcus2 | i wonder if mci will figure this out and take away some of our block |
| 21:23.09 | pooh_ | Spacebar: using AMD ? |
| 21:23.24 | ManxPower | p1tst0p, not normally |
| 21:23.49 | ikarus | Hmmmmmm, an incoming call via ISDN with DID ends up being handled by the exten with the given DID number, right ?, what then happens with DID disabled ? |
| 21:24.29 | p1tst0p | ManxPower ahh shame, hmm i cant think of a way to identify which queue the call came from. hm |
| 21:25.09 | Lostfrog | ManxPower: no way with AGI and sipsak? |
| 21:25.17 | }cytrak{ | justinu: I don't even use the sip url on my xlite setup |
| 21:25.18 | ManxPower | ring them on different call apearences |
| 21:25.45 | p1tst0p | ManxPower, i only have 2 call apperence's on these phones, and about 6 queue's |
| 21:25.46 | ManxPower | Lostfrog, it would ALL, COMPLETLY, and ENTIRELY depend on if the phone supports it and in what way it supports it |
| 21:26.08 | }cytrak{ | justinu: right now I can only get it to register to the * server if I use username=[what is between the brackets] and leave authentication user empty |
| 21:26.26 | }cytrak{ | am I forgeting somethins |
| 21:26.28 | }cytrak{ | am I forgeting something |
| 21:27.26 | *** join/#asterisk Pegger (i=pegasus@66.92.40.210) |
| 21:27.44 | Spacebar | pooh_: Intel Pentium 4 630 cpu |
| 21:28.16 | Spacebar | pooh_: when installing 32bit kernel and userlang, everything is working fine |
| 21:28.21 | }cytrak{ | btw I'm using 1.2 |
| 21:28.50 | stbain | p1tst0p: have you experimented with SendText? (http://www.voip-info.org/wiki/view/Asterisk+cmd+SendText) |
| 21:29.09 | Spacebar | pooh_: with 64bit I cannot see anything going in with pri intense debug span 1 |
| 21:29.35 | Spacebar | pooh_: now I've reinstalled 64bit system and attached it to cisco, will try do debug this issue |
| 21:30.01 | Spacebar | pooh_ somebody said that I just need to run 32bit asterisk and 32bit libpri |
| 21:30.22 | Spacebar | iow, chan_zap isn't 64bit safe ;) |
| 21:30.25 | pooh_ | Spacebar: possibly, dunno if * is FULLY 64 bit 'aware'/comaptible |
| 21:30.58 | Spacebar | ;( |
| 21:31.12 | *** join/#asterisk Pegger (i=pegasus@66.92.40.210) |
| 21:31.34 | Pegger | how do I make * call you at a pree determined time like mornign wake up |
| 21:31.59 | stbain | Pegger: create a call and use cron to put it in the queue |
| 21:32.16 | Pegger | stbain well how am i supos to creat a call? |
| 21:32.20 | *** join/#asterisk Math` (n=math@modemcable148.4-81-70.mc.videotron.ca) |
| 21:32.57 | pooh_ | Pegger: voip-info.org search for wakeup |
| 21:33.22 | Pegger | i found it |
| 21:33.31 | Pegger | Asterisk auto-dial out |
| 21:34.09 | jhb | hi *. I just got my asterisk online. Yuhu! No I ran a test with about 5 people calling in using sipgate to connect to * (register =>), and I got a ton of echo. Any ideas? |
| 21:34.37 | jhb | oh, and the test was using meetme, of course |
| 21:34.57 | jhb | if only two people speak everything is fine |
| 21:35.03 | stbain | jhb: http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation |
| 21:37.44 | *** join/#asterisk Starmaker (n=magnus@85.8.2.169) |
| 21:37.51 | jhb | pooh_: thx and sorry. I googled quickly around, but didn't find anything adaequate |
| 21:37.56 | *** join/#asterisk mcn (n=mcn@81.187.239.184) |
| 21:37.58 | Starmaker | does anyone have any experience with chan_bluetooth? |
| 21:40.29 | pooh_ | jhb: no offence made, just in general |
| 21:40.57 | p1tst0p | stbain thanks man, i shall try SendText shortly ! |
| 21:42.14 | jhb | pooh_: no worries. I was just really exited, hoping to get a clean conference (waiting for this for quite a while). |
| 21:42.22 | stbain | p1tst0p: good luck w/ that. I know the Cisco phones feed off of XML snippets that get sent to it. Not sure about yours though. |
| 21:42.25 | Lostfrog | I can't find one single example of destination number mangling -> group for MP-10X. |
| 21:42.43 | jhb | reads like I really need to dive into things... |
| 21:42.48 | jhb | o well, so be it |
| 21:42.54 | ManxPower | The only thing SendText has ever done for me is disconnect my calls when it's run |
| 21:43.32 | ManxPower | Whoo! Whoo! SBC just called up and offered me DSL. |
| 21:43.43 | ManxPower | Apparently they don't realize I'm 22,000 ft from the CO. |
| 21:43.44 | harryvv | we cannot get dsl out here |
| 21:43.46 | ManxPower | So I ordered it. |
| 21:43.58 | harryvv | yet its a full sized area of the city |
| 21:44.19 | Juggie | ManxPower, that otta provide some entertainment over the comming weeks |
| 21:44.24 | ManxPower | Juggie, Yup! |
| 21:45.22 | harryvv | make sure its not made in china |
| 21:45.53 | ManxPower | My car died today, I got it on monday. The extended warrenty info has not been processed yet. |
| 21:46.03 | harryvv | what died on it? |
| 21:46.15 | ManxPower | stbain, I need paper to fax |
| 21:46.19 | ManxPower | harryvv, sounds like the starter. |
| 21:46.36 | harryvv | If I had a car that the engine died, I would yank it out and convert it to a electric car. |
| 21:46.50 | stbain | ManxPower: you have papers you need to fax? |
| 21:47.04 | *** join/#asterisk MarkT1982 (n=MarkT@tcc.demon.co.uk) |
| 21:47.12 | harryvv | manx, put a volt meter test lead on the starter then attempt to start the car. if you see voltage but it does not start its likly the starter. |
| 21:47.14 | jhb | haha - ' There is not a echo=no switch' from the faq |
| 21:47.19 | jhb | ok, I see |
| 21:47.30 | jhb | sorry to have have not read that first |
| 21:47.31 | jhb | ;-) |
| 21:47.47 | harryvv | our kilowatt rate here is 2.2 cents |
| 21:48.00 | tzanger | nice |
| 21:48.01 | harryvv | cheap at best |
| 21:48.06 | tzanger | I think in Ontario it's 4.7 |
| 21:48.15 | harryvv | makes powering a electric car very cheap |
| 21:48.42 | MarkT1982 | Has anyone managed to get chan_capi operating with Asterisk 1.2.0? I'm battling with it and losing and I thought I might pop in and see if anyone would be willing to give me a hand? |
| 21:48.54 | stbain | buried an extension cord to my neighbor's house... makes running an electric car even more cheap |
| 21:49.32 | tzanger | hahahahaha |
| 21:50.28 | ManxPower | It sure would be nice if this message "Nov 11 15:49:34 NOTICE[5794]: chan_sip.c:3569 process_sdp: No compatible codecs! |
| 21:50.28 | ManxPower | " provided the device or channel number |
| 21:51.41 | MarkT1982 | Has anyone ever seen this kinda output from chan_capi? Manager registered action ParkedCalls |
| 21:51.42 | MarkT1982 | <PROTECTED> |
| 21:51.43 | harryvv | stabain, untill the neighbor pulls up the cord traces it to your car then tourches it ;) |
| 21:52.03 | MarkT1982 | just bombs out, no error, no nothing, even in high verbosity logging! |
| 21:52.18 | shido6 | so using the trim argument in sox we can trim off the first 7 seconds from cepstral and then a little editing of app_cepstral.c no more license |
| 21:52.33 | shido6 | thats evil |
| 21:53.38 | Lostfrog | Come one.. You can't tell me noone has managed to dialout from * to a MP-104 or 108 FXO to PSTN.. |
| 21:57.18 | rculp | what's a good place to get sample asterisk configs of working setups? |
| 21:57.50 | Lostfrog | ~wiki |
| 21:58.01 | Lostfrog | ~docs |
| 21:58.04 | jbot | rumour has it, docs is 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 |
| 21:58.10 | rculp | wow |
| 21:58.22 | Flauto | i hve a problem, here. i compiled the zaptel and installed |
| 21:58.42 | Flauto | and wheni need to modprobe zaptel and wcfxo |
| 21:58.56 | Flauto | it tells me fatal module zhaptel not found |
| 21:59.19 | Flauto | i checked the directory of lib/modules/ |
| 21:59.29 | Flauto | there are two directories |
| 21:59.52 | Flauto | 2.6.11-6mdk and 2.6.11-6mdkcustom |
| 22:00.09 | Flauto | which the 2.6.11-6mdk has misc |
| 22:00.20 | Flauto | and it has all the .ko files |
| 22:00.50 | Flauto | pooh told me to try insmod |
| 22:00.55 | Flauto | still, it is not working |
| 22:02.54 | Flauto | never mind |
| 22:02.56 | Flauto | i got it |
| 22:02.59 | Flauto | thanks to all though |
| 22:03.01 | Igbothom_III | you did "make install" in the zaptel dir after "make lilnux26"? |
| 22:03.03 | Flauto | you guys are the best |
| 22:03.09 | Flauto | yes, sire |
| 22:03.11 | Flauto | sir |
| 22:03.17 | Flauto | igbothom, i did |
| 22:03.20 | Flauto | thanks very much |
| 22:05.05 | rene- | has anyone tried modem pass tru using an ATA on a LAN? |
| 22:05.45 | rene- | going out to an fxo interface tru asterisk i mean |
| 22:15.05 | *** join/#asterisk dos000 (n=dos000@CPE00119572fd49-CM00137186e53a.cpe.net.cable.rogers.com) |
| 22:16.02 | *** join/#asterisk ozant (n=ozan@81.215.121.100) |
| 22:17.05 | *** join/#asterisk niZx (n=ilt@S0106deadbeefbeef.wp.shawcable.net) |
| 22:26.10 | Lostfrog | I don't want to buy another 8 port FXO or 2 TDM400s or TE2xx and a CB.. :( |
| 22:26.56 | MikeJ[Laptop] | you can do just make.. make linux26 shouldn't be necessary anymore |
| 22:27.14 | mog_work | get a tdm24xxp |
| 22:27.34 | Lostfrog | lol.. yeah |
| 22:28.25 | Lostfrog | I would buy another CB and TE110 first. |
| 22:28.45 | Lostfrog | Not to mention the fact that the tdm24 isn't released or priced. |
| 22:29.52 | Lostfrog | ~$1000 |
| 22:30.17 | justinu | why not just get an audiocodecs 24 port SIP gateway? |
| 22:30.20 | justinu | audiocodes |
| 22:30.37 | Lostfrog | I have a MP-108.. I can't figure out how to dial out on a specific port. |
| 22:31.03 | fugitivo | Lostfrog: yes it's priced, i have prices for my country |
| 22:31.15 | fugitivo | it's cheaper than a audiocodes gateway |
| 22:31.21 | Lostfrog | yeah.. voipsupply is pre-selling. |
| 22:31.43 | justinu | i see pci cards as a big liability |
| 22:31.51 | justinu | if you need to change something, server is powered off |
| 22:32.09 | fugitivo | justinu: just like fxo cards |
| 22:32.12 | Lostfrog | I've never been happy with my TDM4xx's in the past. |
| 22:32.21 | justinu | yeah, anything that you stick in the server |
| 22:32.39 | fugitivo | it's nice to have all the hardware inside one box |
| 22:32.48 | fugitivo | what is better? i don't know |
| 22:33.02 | justinu | i'm always concerned with scalability |
| 22:33.10 | justinu | so I go with external devices |
| 22:33.24 | Damin | Hmm.. |
| 22:33.25 | fugitivo | you don't lose scalability with pci cards |
| 22:33.36 | fugitivo | you can always add external hardware |
| 22:33.53 | justinu | yeah, but if that card were to fail, or something |
| 22:33.59 | justinu | it's downtime to replace |
| 22:34.12 | Damin | I'm getting weird SIP Pops/Cracks on a CCM 4.1 -> Asterisk CVS Head call.. |
| 22:34.19 | Starmaker | I'm trying asterisk 1.2.0 RC1 out :) |
| 22:34.28 | fugitivo | if the gateway fails it's the same, it's downtime, not full downtime maybe, that depends on the size of the implementation |
| 22:34.42 | justinu | Damin: can you tcpdump the rtp stream and analyze it in etherreal? |
| 22:34.49 | Lostfrog | Ok.. I'm willing to pay someone to figure this out for me. |
| 22:34.55 | Lostfrog | There has to be a solution. |
| 22:34.58 | Damin | justinu: Probably.. |
| 22:35.24 | Damin | justinu: It sounds like out of order packets.. |
| 22:35.33 | *** join/#asterisk Frawg (n=Frawg@unaffiliated/frawg) |
| 22:35.33 | mocker | Anyone have any thoughts on the Grandstream GXP-2000? |
| 22:35.41 | justinu | asterisk RTP wouldn't just discard them? |
| 22:35.57 | Lostfrog | After my experiences the the BT-100.. I wouldn't touch GS with a 20' pole. |
| 22:36.01 | Lostfrog | +with |
| 22:36.06 | justinu | mocker: yeah, i have a few |
| 22:36.15 | mocker | justinu: Any luck with them?> |
| 22:36.16 | justinu | weren't you in here before asking about them? |
| 22:36.29 | justinu | maybe 2 weeks ago |
| 22:36.32 | mocker | No. |
| 22:36.38 | mocker | I was in this morning asking about a different phone. |
| 22:36.42 | justinu | they're simply OK phones |
| 22:36.55 | justinu | don't expect the speakerphone to work very well |
| 22:37.03 | justinu | there's no "missed calls" directory |
| 22:37.07 | justinu | or any directory, for that matter |
| 22:37.16 | Frawg | hi all |
| 22:37.18 | Frawg | anyone here used SER? |
| 22:37.28 | justinu | yes |
| 22:37.38 | mocker | justinu: Have you messed w/ the BLF feature at all? |
| 22:37.44 | Frawg | justinu: pm? |
| 22:37.50 | justinu | mocker: not on that phone yet |
| 22:37.58 | justinu | that's the latest unreleased firmware |
| 22:38.11 | justinu | frawg: ok |
| 22:40.46 | justinu | mocker: bottom line... the GS software sucks |
| 22:40.50 | mocker | justinu: Wonderful. |
| 22:40.50 | justinu | but the hardware seems decent |
| 22:40.59 | justinu | give me a phone with sipura software, and GS hardware |
| 22:41.14 | justinu | because sipura has great software, but the 841 is kind of a piece of shit |
| 22:41.18 | mocker | What phones do people recommend for $150 |
| 22:41.22 | justinu | polycom |
| 22:41.46 | *** join/#asterisk zotz (n=zotz@24.231.47.168) |
| 22:41.53 | mocker | justinu: What do you think of the SoundPoint IP 301? |
| 22:42.02 | justinu | it's great... just no full duplex speaker |
| 22:42.11 | justinu | gotta go to the 501 for that, which is like $175 |
| 22:42.21 | *** join/#asterisk bweschke_ (n=bweschke@pcp09754274pcs.narlington.nj.comcast.net) |
| 22:42.22 | justinu | polycom sound is "amazing" |
| 22:42.46 | Frawg | i got s snom 190 here |
| 22:42.48 | Frawg | it's fairly nice |
| 22:42.53 | Frawg | dunno how much it costs though |
| 22:42.57 | justinu | i have a snom 360 |
| 22:43.02 | justinu | doesn't sound near as good as polycom |
| 22:43.07 | mocker | justinu: Okay, for the test bed we're going to pick GXP-2000, Polycom SoundPoint 301, and Cisco 7905G |
| 22:43.22 | mocker | justinu: That sound like a good set of phones to choose from? |
| 22:43.30 | justinu | yeah |
| 22:43.47 | justinu | you could also try and get the new linksys/cisco/sipura 941 |
| 22:47.54 | *** join/#asterisk |freat| (n=freat@h-69-3-229-184.chcgilgm.covad.net) |
| 22:49.24 | *** join/#asterisk BladeRunner05 (n=gianni@adsl-ull-244-65.44-151.net24.it) |
| 22:51.40 | harryvv | mocker, dont forget to include a ftp server |
| 22:51.54 | harryvv | or put one on the asterisk box |
| 22:58.33 | mocker | harryvv: ? |
| 22:58.53 | mocker | justinu: Sent those off to my boss, thanks for the insights. |
| 23:02.03 | BladeRunner05 | Please suggest me an expensive ip_phone |
| 23:02.19 | YoMama | an expensive one? |
| 23:02.21 | YoMama | any cisco one |
| 23:02.33 | ender | interesting. CentoS4's comps.xml references 'redhat-logos' package. |
| 23:02.37 | YoMama | the polycom conference phone...costs like $700 |
| 23:02.39 | BladeRunner05 | any expensive |
| 23:02.45 | ender | er wrong chan. |
| 23:03.11 | BladeRunner05 | sorry, the cheapest |
| 23:03.22 | snitt | lohl |
| 23:03.31 | bsdfreak | heh |
| 23:03.36 | bsdfreak | anyone tried asterisk 1.2.0rc1? |
| 23:04.00 | Starmaker | yeah |
| 23:04.01 | *** join/#asterisk zagaya971 (n=almeli@APointe-a-Pitre-102-1-11-9.w81-248.abo.wanadoo.fr) |
| 23:04.11 | bsdfreak | any major improvements you've noticed? |
| 23:04.32 | Starmaker | nope, but I havn't tested it very much yet |
| 23:05.34 | justinu | i found a bug in the dial macros |
| 23:05.40 | justinu | but that's fixed in head, apparently |
| 23:06.26 | Starmaker | what kind of bug? |
| 23:06.36 | justinu | they just don't work |
| 23:06.52 | Starmaker | huh, I havn't had any trouble with those, i think |
| 23:07.06 | justinu | option "M" of the dial app |
| 23:07.18 | Starmaker | lemme check my extensions.conf |
| 23:07.22 | bweschke_ | starmaker: it was a bug with parsing options that had additional arguments with it |
| 23:07.30 | Druken | has anyone used the "privacy manager" ?? |
| 23:07.36 | bweschke_ | as justinu said, it is indeed fixed already in head and will be in rc2 |
| 23:07.40 | justinu | bweschke_: i dunno, my setup doesn't even use options |
| 23:07.51 | justinu | err macro doesn't use arguments |
| 23:09.05 | Starmaker | ah, no wonder i havn't found any problems with that, I don't use them :) |
| 23:09.07 | bweschke_ | you macro didn't have to use options... your arguments in dial is what exposed the error, but in some cases, there's code reuse going on to read and parse those arguments now |
| 23:09.20 | justinu | bweschke: yeah, i misread what you had sid |
| 23:09.21 | justinu | said |
| 23:09.23 | justinu | my mistake |
| 23:09.45 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-60-57.cybersurf.com) |
| 23:10.26 | Starmaker | now it's time to fix a polka-extension |
| 23:11.04 | bweschke_ | nice |
| 23:11.09 | Starmaker | yeah :p |
| 23:11.11 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 23:11.28 | Starmaker | it would be even nicer if I had some polka / death metal fusion |
| 23:11.36 | Starmaker | but i havn't found anything like that |
| 23:11.46 | BladeRunner05 | does Grandstream GXP-2000 supported by * ? |
| 23:12.02 | justinu | BladeRunner05: it works fine |
| 23:12.15 | BladeRunner05 | justinu:tnx |
| 23:13.58 | bsdfreak | heh |
| 23:14.05 | *** join/#asterisk harmonium (n=andreas@zz195070.cipherkey.net) |
| 23:14.10 | wunderkin | crap, every so often i'm getting a red alarm.. sometimes its on span 2, sometimes on span 1, both are connected to same telco.. timing is 1 for 1st, 2 for 2nd.. usually happens later at night.. but just saw it happen now.. not very frequently.. it always comes right back up on the next cycle in 5 sec.. hmm |
| 23:14.28 | *** join/#asterisk brc_ (n=Brian@pdpc/supporter/basic/brc) |
| 23:14.28 | justinu | odd |
| 23:14.38 | wunderkin | it was behaving fine when i had it connected via xover to another * box |
| 23:14.40 | *** join/#asterisk e-Hernick (n=ncc@modemcable228.85-70-69.mc.videotron.ca) |
| 23:14.42 | BladeRunner05 | does Snom 320 supported ? and wich is the best one between grandstream gpx-2000 and Snom 320 ? |
| 23:15.03 | justinu | wunderkin: can you leave a loop up on one span for testing? |
| 23:15.09 | justinu | BladeRunner05: snom is much nicer |
| 23:15.18 | wunderkin | yeah i havent been using them so yeah i can do whatever i want at this point |
| 23:15.20 | harmonium | anyone had a problem with udp packets being queued, but they don't seem to go anywhere? ex. udp 74992 0 0.0.0.0:5060 0.0.0.0:* |
| 23:15.32 | wunderkin | the only problem is im remote |
| 23:15.35 | justinu | wunderkin: i would ask your LEC to do some BERT testing |
| 23:15.43 | bweschke_ | wunderkin: carriers don't often cause a red alarm.... that's usually more often a continuity problem on your prem... do you have fiber coming into your prem or are your ds1's term'd by copper? |
| 23:16.06 | bsdfreak | heh |
| 23:16.13 | wunderkin | its in a colo facility, all i know is it comes in over dsl and theres a smartjack, then it goes into a 66 block and then into the cabinet |
| 23:16.14 | justinu | lol, idiot CLECS like I have cause red alarms all the time |
| 23:17.00 | wunderkin | i know that one of the lines i had a problem with before (thanks again justinu) that was caused by incorrect signalling on their side, before that they said nothing was wrong, then sent the lec, then they fixed the signalling |
| 23:17.09 | wunderkin | so they were doing a lot of testing on the 1st line at that point |
| 23:17.18 | justinu | oh yeah, i remember that |
| 23:17.32 | harmonium | the udp packets seems to be timeout issues with SIP client authentication. Anyone know what the problem might be? |
| 23:17.37 | wunderkin | it usually only happens on the 2nd.. sometimes it does both.. and now it just did the 1st.. i usually only saw it at night so i thought the provider may be doing something |
| 23:17.55 | wunderkin | justinu: yeah that was lots of fun :D |
| 23:18.08 | justinu | heh, ain't telcom wonderful? |
| 23:18.12 | wunderkin | really |
| 23:18.18 | justinu | can you see now why i'm sick of TDM? |
| 23:18.22 | wunderkin | heh |
| 23:18.57 | wunderkin | not our problem, not our problem, oops its our problem! |
| 23:19.23 | wunderkin | he was trying to hide it but i got him to admit it.. that was sweet |
| 23:20.29 | wunderkin | i know i can try new cables but i dont really think thats it |
| 23:20.45 | wunderkin | unless its from the interference from everything else in there |
| 23:20.55 | justinu | how long are the cables? |
| 23:20.56 | bweschke_ | red alarm = loss of frame .... so ya... misconfigured framing could cause it, but I'd suspect it's cabling at that prem before anything else... of all the hundreds of ds1's I managed that were muxed down from ds3's in my prior job, over 90% of the time, replacing the cable/connection fixed the problem |
| 23:21.13 | Druken | justinu: no it's not wonderful, it's a pain in the ass |
| 23:21.17 | Druken | but i still love it |
| 23:21.25 | justinu | tdm? |
| 23:21.27 | wunderkin | they are short ethernet jumper cables, a foot maybe |
| 23:21.30 | Druken | telecom |
| 23:21.36 | justinu | oh, i love telcom too |
| 23:21.56 | justinu | wunderkin: how about the 66 punch downs? |
| 23:21.56 | ManxPower | I just confirmed the problem. |
| 23:22.11 | justinu | wunderkin: make sure they're clean and not double punched or anything idiotic like that |
| 23:22.32 | wunderkin | justinu: the distance from the smartjack to the block is just a few feet |
| 23:23.03 | wunderkin | from the block to the rack is a little ways, its in another room |
| 23:23.16 | bweschke_ | justinu had a good pt.... 66 block connections.... can be a lot more flakey than an rj48 connection |
| 23:23.34 | ManxPower | wunderkin, Well, what lights are lit on the smartjack. |
| 23:23.52 | ManxPower | Ask your telco to "run a 30 min loopback from the telco to the smartjack" |
| 23:23.52 | wunderkin | i dont know, im remote.. it always comes back up in 5 sec |
| 23:24.03 | Lostfrog | ManxPower: what problem did you confirm? |
| 23:24.19 | bweschke_ | are you collecting LCV's in between the red's ? |
| 23:24.28 | justinu | wunderkin, if you could loop up the smartjack towards your PBX, that would tell you if the CPE wiring is OK |
| 23:24.45 | wunderkin | bweschke_: umm.. no? how? |
| 23:24.49 | ManxPower | Lostfrog, "The forwarding of a voicemail only works when you |
| 23:24.50 | ManxPower | press 8 (to forward) and then 2 (no prepend). It does not work when you |
| 23:24.50 | ManxPower | prepend message. |
| 23:24.50 | ManxPower | " I confirmed this is a problem with today's CVS. |
| 23:24.50 | BladeRunner05 | justine: does |
| 23:24.50 | BladeRunner05 | Polycom Soundstation IP 501 |
| 23:24.58 | BladeRunner05 | supported by * ? |
| 23:25.05 | justinu | yes, and that is an excellent phone |
| 23:25.31 | ManxPower | BladeRunner05, No. Only the SoundPoint IP 501 is supported, as far as I know. |
| 23:26.39 | bweschke_ | Manx: did you post a bug? I will start looking at it for u. |
| 23:26.40 | wunderkin | looks like more fun testing is in order :P |
| 23:27.52 | Starmaker | something seems to have gone odd with my queues |
| 23:28.14 | Starmaker | if there are no agents logged in, the call gets hanged up after like 5 seconds |
| 23:29.34 | Lostfrog | Why does Dial(sip/blah,60,D(1234)) dial the leading '(' before 1234? |
| 23:29.51 | bweschke_ | Lostfrog. it's a bug that was fixed in CVS-HEAD last night |
| 23:30.00 | bweschke_ | update your code to the latest CVS |
| 23:30.38 | justinu | manxpower: i assumed he meant sounpoint, because there isn't a soundstation ip501, iirc |
| 23:31.47 | Lostfrog | ok.. can I update my ftped 1.2RC1 directory via CVS? |
| 23:32.37 | bweschke_ | lostfrog - no, probably not |
| 23:32.39 | snitt | i think u can |
| 23:32.57 | snitt | bud i dont know personally how cvs work |
| 23:33.07 | *** join/#asterisk kuku5 (n=macova@thesocket.net) |
| 23:33.27 | bweschke_ | lostfrog: I don't know if the tarball on the ftp site contains the CVS folders itself. I don't think it does. If it doesn't, CVS is going to complain loudly when it sees all the same named files in it's way |
| 23:33.32 | kuku5 | Anyone have a te110p card in chicago to purchase tonight? |
| 23:33.39 | justinu | heh |
| 23:33.43 | kuku5 | the one i got is bad |
| 23:37.10 | ManxPower | kuku5, always keep a spare in stock |
| 23:39.14 | wunderkin | justinu: how would i loop the smartjack towards *? the telco would do that? |
| 23:39.42 | justinu | they should be able to |
| 23:39.43 | Lostfrog | That's easy for you to say, ManxPower. |
| 23:39.56 | wunderkin | ok, i didn't think there was a way for me to do anything :D |
| 23:40.14 | ManxPower | wunderkin, The telco only cares about the telco<->smartjack connection. |
| 23:40.35 | wunderkin | yeah, well i just call my provider and they take it from there |
| 23:41.17 | ManxPower | wunderkin, if the 30 min telco loopback from them to the smartjack doesn't fix it, then it's prolly a wire problem. We had a problem with a smartjack where the port was not quite right and so you could bump the cable and the T-1 would bounce. |
| 23:41.33 | wunderkin | ah |
| 23:42.01 | *** join/#asterisk colinm_ (n=colol@VDSL-130-13-9-157.PHNX.QWEST.NET) |
| 23:44.40 | harryvv | Telus has locked out most of its employees and now outsourcing alot of its work. Thay have lowered the cost of there local phone services from 39 per month to 13 per month. I get a hint this is voip and the extra overhead "the employees" cost to much. |
| 23:50.51 | ManxPower | http://bugs.digium.com/view.php?id=5718 |
| 23:52.29 | *** join/#asterisk brc_ (n=Brian@pdpc/supporter/basic/brc) |
| 23:54.17 | *** join/#asterisk nagl (n=nagl@213.235.241.6) |
| 23:57.07 | *** join/#asterisk Corndawg_ (i=whoisit@c-66-176-66-83.hsd1.fl.comcast.net) |
| 23:57.24 | *** join/#asterisk Brixius (i=Brixius@c-69-180-132-70.hsd1.mn.comcast.net) |
| 23:57.48 | Brixius | Hello |
| 23:59.10 | Brixius | hmm quiet tonight |
| 23:59.31 | pooh_ | Brixius: man, you jst got in |
