| 00:00.09 | Downchuck | but no voice back or forth. opened up ports 8k-20k tcp and udp on the asterisk server. i'm behind a nat at home. |
| 00:00.30 | Downchuck | speex and h323 are installed to the best of my knowledge |
| 00:00.42 | JT | asterisk does not need a sound card |
| 00:00.46 | JT | are you using h.323? |
| 00:01.04 | Downchuck | i'd be using speex for testing |
| 00:01.12 | JT | have you even checked any documentation to see what ports need to be opened? |
| 00:01.15 | JT | and h.323? |
| 00:01.38 | *** join/#asterisk entelechy (i=user@mail.beanproducts.com) |
| 00:01.49 | Downchuck | i compiled it for no good reason; yes I checked the docs, but it'd make some sense if i tested outside of my nat |
| 00:02.08 | Downchuck | i've got 5060 open for SIP. and 10k-20k are for rtp |
| 00:02.30 | JT | yet you opened 8-20k tcp |
| 00:02.42 | JT | the correct ports for sip are udp 5060, 10000-20000 |
| 00:02.50 | Downchuck | some minor obscure faq said open up 8k for misconfigured sip phones |
| 00:03.21 | JT | it's wrong, no use opening up 8k for asterisk unless you modify rpt.conf |
| 00:03.27 | JT | yes but why did you open tcp? |
| 00:03.45 | Downchuck | good point.. |
| 00:03.46 | kiscokid | Every now and then I see STUN mentioned as a way to get around NAT problems. Any info? |
| 00:04.18 | JT | stun is a client side hack to use with dodgy sip servers |
| 00:04.23 | JT | asterisk does not need stun |
| 00:04.46 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 00:05.28 | JT | s/rpt.conf/rtp.conf/ |
| 00:05.41 | Downchuck | JT: caught that :) |
| 00:05.58 | JT | Downchuck: sip with asterisk only uses UDP |
| 00:06.18 | Downchuck | JT: I figured as much, but I didn't know how the voice routing works |
| 00:06.24 | Downchuck | clearly I don't .. I've got some good ideas :) |
| 00:06.44 | *** join/#asterisk entelechy (i=user@mail.beanproducts.com) |
| 00:06.47 | JT | media uses the RTP protocol |
| 00:06.51 | Downchuck | The port 8000 was for X-lite.. for some reason i've loaded about 10 SIP clients onto my computer for testing. |
| 00:06.53 | JT | one port in each direction |
| 00:07.12 | karlhaines | Downchuck: have you found a favorite? |
| 00:07.19 | karlhaines | Downchuck: i need to install an IAX client on my box |
| 00:07.25 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:07.30 | Downchuck | karlhaines: No, but I've done well finding some least favorites |
| 00:07.44 | JT | i haven't seen anything better than idefisk for IAX2 |
| 00:07.45 | karlhaines | Downchuck: which ones suck? i'd like to know, lol |
| 00:08.01 | JT | yeah, you can say that, you only find least favourites with softphones |
| 00:08.03 | JT | they all suck |
| 00:08.06 | karlhaines | JT: thats what i use on my windows box, is there a linux version? |
| 00:08.12 | JT | karlhaines: yes |
| 00:08.13 | karlhaines | idefisk seemed KO to me |
| 00:08.17 | karlhaines | err OK.. |
| 00:08.23 | karlhaines | JT: thanks! |
| 00:08.24 | JT | it's very basic |
| 00:08.37 | JT | and has no noise cancellation etc |
| 00:08.43 | JT | like a real sip phone |
| 00:08.50 | Downchuck | karlhaines: there's basic and there's stupid (IM).. most of the basic's are unstable.. most of the IMs are bloated |
| 00:09.28 | JT | if it's important, use a real phone :) |
| 00:09.35 | karlhaines | Downchuck: lol, don't you just love half-done software? |
| 00:09.43 | Downchuck | karlhaines: it's the only kind I'll use :-) |
| 00:10.07 | karlhaines | rofl, sorry for you! |
| 00:10.18 | irule | how can I troubleshoot zaptel? I upgraded from 1.0 to 1.4 and now it does not work using the same configuration :s " [May 15 19:16:06] WARNING[2646]: app_dial.c:1099 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) |
| 00:10.33 | Downchuck | JT: do i need any kind of nat techniques (beyond nat=yes) for the RTP? |
| 00:10.52 | JT | Downchuck: what's behind the nat, and what is connecting to what? |
| 00:11.03 | Downchuck | I'm behind the nat, connecting to a server which is open on the internet |
| 00:11.36 | JT | you don't need to do much then |
| 00:12.28 | Downchuck | i need to figure out what's triggering asterisk to "modprobe: Can't locate module sound-slot-1".. and find a config error somewhere.. that is about it. |
| 00:12.42 | Downchuck | it's good software, low learning curve considering where i was last week |
| 00:12.59 | JT | i don't think asterisk is doing that |
| 00:13.11 | Downchuck | it was trying to do chan_oss |
| 00:13.53 | Downchuck | eh.. that bouncing wengo head is just horrible. project gizmo is crashy. |
| 00:14.11 | JT | tried ekiga? |
| 00:14.29 | Downchuck | no, will now :) |
| 00:14.57 | JT | x-lite on windows is ok, but seriously handicapped due to being the basic version |
| 00:15.16 | karlhaines | any Gentoo users in here interested in getting involved with my asterisk-gentoo project ? |
| 00:15.22 | JT | i use x-lite for testing, but the single server thing shits me to no end |
| 00:15.25 | karlhaines | if so, please join #asterisk-gentoo |
| 00:15.33 | JT | spam :P |
| 00:15.47 | Downchuck | i've got x-lite on.. it seemed ok, but it practically freezes my computer when i dial the first few digits |
| 00:16.00 | Downchuck | ekiga servers are down, but i'm going to try it when it's back up |
| 00:16.30 | JT | Downchuck: once you get a quality sip phone, you'll never turn back |
| 00:16.43 | Downchuck | I started with a crappy sip phone.. that put me back a year. |
| 00:16.50 | LeddyHM | gentoo sucks |
| 00:17.01 | JT | grandstream? |
| 00:17.06 | Downchuck | I got a ZyXEL i-hate-u wifi phone |
| 00:17.12 | JT | hah |
| 00:17.16 | JT | all wifi phones suck |
| 00:17.27 | Downchuck | yeah, but it could suck less |
| 00:17.29 | JT | the protocol is inappropriate for voip |
| 00:17.47 | Downchuck | it's as bad as windows in maintaining a connection |
| 00:18.13 | Downchuck | i bought it as a little attempt at a cell phone, when traveling through parts of europe |
| 00:18.20 | Downchuck | figured i could stand next to an open wifi and check voicemail. |
| 00:18.35 | JT | as in 30cm from the AP ;) |
| 00:18.48 | Downchuck | ended up using skype :) |
| 00:19.05 | JT | evil |
| 00:19.25 | Downchuck | well.. i ended up actually forwarding through skype to expensive cellphones |
| 00:19.39 | Downchuck | and finding a computer every once in awhile to download, install and update my call forwarding |
| 00:19.49 | Downchuck | It was all bad. and expensive. |
| 00:20.27 | *** join/#asterisk evilcyrus (n=evilcyru@bas3-hamilton14-1096561890.dsl.bell.ca) |
| 00:20.32 | evilcyrus | anyone no a program called SJ phone |
| 00:20.37 | evilcyrus | it looks like a phone |
| 00:20.57 | *** join/#asterisk ObsidianX (n=ryan@adsl-71-142-232-190.dsl.scrm01.pacbell.net) |
| 00:21.39 | ObsidianX | im trying to find a good voip termination/origination service in the california, sacramento area |
| 00:21.43 | ObsidianX | anybody have any suggestions? |
| 00:27.13 | Downchuck | maybe i should try a second termination, on the very off chance that it's the terminator and not me with asterisk |
| 00:28.34 | karlhaines | ObsidianX: i'm trying to find one in the US also, are you wanting to port current DIDs or buy new ones? |
| 00:30.22 | irule | http://pastebin.ca/490417 this looks quite standard, I have a similar setup but is working propperly, it used to be working, and I upgraded from an old 1 to 1.4, I already reboot but this error message looks the same as the other server without the cable connected what else could be wrong? |
| 00:30.34 | ObsidianX | im pretty sure we already have some DID's |
| 00:32.01 | Downchuck | well. my sip to sip calls work. |
| 00:35.17 | karlhaines | ObsidianX: so you want to keep your existing numbers, but port them to a VOIP provider? |
| 00:37.01 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 00:39.46 | *** join/#asterisk huslage (n=nnhuslag@hact.net) |
| 00:39.51 | huslage | hello there |
| 00:39.56 | xpot | anyone know if fields can be added to the cdr yet? |
| 00:40.19 | huslage | has anyone gotten the linksys phones (SPA-94x) to do call parking with a button push? |
| 00:40.30 | huslage | i don't have one handy and can't remember if there is magic needed |
| 00:44.22 | huslage | guess not. |
| 00:44.24 | huslage | oh well. |
| 00:45.00 | snuffy22 | xpot.. field have not been added to cdr.. anyfields you might want you will have to make use of the 'userfield' |
| 00:45.33 | snuffy22 | on my cdrs i seperate fields within userfield via a ':' |
| 00:45.38 | JT | xpot: you can add whatever fields you want if you use cdr-custom i thought |
| 00:45.56 | JT | the standard CDRs stay the same |
| 00:46.21 | xpot | JT: thank you, I will check that out |
| 00:46.44 | xpot | snuffy22: -=0) thank you for trying. |
| 00:47.16 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 00:47.38 | snuffy22 | np |
| 00:48.39 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 00:55.19 | xpot | I have my cdr being written to SQL, is there a way to issue a statement after cdr is written? I tried: exten => h,1,Set(ODBC_CDR(${BLAH})=${BLAH2}) but the cdr does not seem to be written untill the session ends, instead of the call |
| 00:57.54 | *** join/#asterisk ctaloi (n=ctaloi@pool-71-176-69-82.syrcny.fios.verizon.net) |
| 00:57.55 | xpot | BLAH2 = uniqueid |
| 00:58.05 | *** join/#asterisk bjohnson (n=bjohnson@i209-195-85-45.cia.com) |
| 01:08.46 | Downchuck | how many of them DECT phones can operate near each other? |
| 01:09.14 | Downchuck | i'm gonna get one for my home.. but it'd be interesting to put a few in the office |
| 01:09.59 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1128667111.dsl.bell.ca) |
| 01:10.08 | *** join/#asterisk coppice (n=chatzill@94.143.17.210.dyn.pacific.net.hk) |
| 01:10.35 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 01:10.41 | ObsidianX | anybody know of a good california termination service, preferrably in sacramento |
| 01:10.47 | *** part/#asterisk huslage (n=nnhuslag@hact.net) |
| 01:11.16 | LeddyHM | ewuu |
| 01:11.25 | LeddyHM | sack-a-tomato |
| 01:12.34 | karlhaines | ObsidianX: well, as i was asking before, if you are keeping numbers, porting takes a LONG time, but if you are purchasing new numbers, les.net is pretty good |
| 01:13.51 | JT | ObsidianX seems to be very good at ignoring questions when asked for clarification, and continually repeating his original question |
| 01:14.09 | JT | ObsidianX: no-one will help you if you don't help them help you |
| 01:16.11 | ObsidianX | JT: sorry, stepped away, forgot he'd asked me :\ |
| 01:16.18 | ObsidianX | karlhaines: where are you located? |
| 01:16.47 | karlhaines | ObsidianX: TN. but les.net does all of us and canada |
| 01:17.17 | *** join/#asterisk unspin (n=unspin@24.82.161.85) |
| 01:17.49 | JT | ObsidianX: the question that has been asked is if you are porting or not |
| 01:18.24 | ObsidianX | i just checked, and no im not |
| 01:18.24 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 01:20.07 | JT | that makes it a lot easier i guess |
| 01:23.27 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 01:23.30 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 01:24.12 | anonymouz666 | hey anthm, STRPTIME() solve my problem yesterday |
| 01:26.30 | *** join/#asterisk phobus (n=phobus@crlspr-69.65.75.232.myacc.net) |
| 01:27.02 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
| 01:27.18 | *** join/#asterisk litage (n=nick@70.55.220.203.static.comindico.com.au) |
| 01:27.38 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 01:29.56 | *** join/#asterisk Zipper_32 (n=None@d154-5-87-181.bchsia.telus.net) |
| 01:30.59 | *** part/#asterisk kiscokid (n=ron@208.106.33.66) |
| 01:33.21 | *** join/#asterisk Black-Kakugane-1 (n=lonewolf@c-71-206-157-165.hsd1.va.comcast.net) |
| 01:39.18 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 01:39.38 | *** join/#asterisk Strom_M (n=strom@12.175.45.218) |
| 01:43.31 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:46.45 | *** join/#asterisk ido (n=ido@unaffiliated/ido) |
| 01:47.18 | *** join/#asterisk luisjose (n=ljd@nelug/coreteam/luisjose) |
| 01:49.05 | Zipper_32 | Does anybody have any ideas why in the case of a Point to Point IAX2 bridge, one side has a status of "OK (4 ms)" and the other displays "UNREACHABLE", while two days ago (no config changes since), they were both 'OK (4 ms)'? |
| 01:49.54 | JT | qualifications have failed on one side |
| 01:49.55 | Zipper_32 | I've tried rebooting each machine, downing the switches between them, but that doesn't work. Pinging between them works fine. There's no firewall in the way. |
| 01:50.19 | Zipper_32 | qualify=5000 for both. They're both well within their range. |
| 01:50.25 | JT | they're on the same LAN? |
| 01:50.33 | Zipper_32 | <PROTECTED> |
| 01:50.39 | JT | eww |
| 01:50.40 | JT | hmm |
| 01:50.48 | Zipper_32 | Wireless Lan spanning 1.2km |
| 01:50.57 | Zipper_32 | Works very well actually, =) |
| 01:51.40 | JT | as long as it's fixed point i guess |
| 01:52.05 | Zipper_32 | Heaven forbid I have wireless clients... *ack* |
| 01:52.21 | Zipper_32 | Strictly a Point-To-Point in this case. |
| 01:54.38 | *** join/#asterisk tawaki (n=steve@adsl-71-142-232-190.dsl.scrm01.pacbell.net) |
| 01:56.41 | *** join/#asterisk JunK-Y (n=junky@modemcable105.205-56-74.mc.videotron.ca) |
| 01:57.42 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 01:57.50 | tawaki | what is the best way to find low cost yet good quality voip termination from the west coast? |
| 01:59.06 | LeddyHM | www.google.com |
| 01:59.17 | LeddyHM | www.voip-info.org |
| 01:59.41 | Zipper_32 | I use vitelity for Vancouver BC. Seems to do the trick here. (vitelity.net) |
| 01:59.47 | Zipper_32 | Not an employee, just a customer. |
| 01:59.52 | tawaki | tried that...many hours later still trying to not reinvent the researcher's wheel! |
| 02:00.08 | Strom_M | tawaki: where on the west coast are you? |
| 02:00.27 | tawaki | I am using vitelity now and it is not good...maybe MOS 3 at best |
| 02:00.45 | tawaki | I am in the Sacramento, CA area |
| 02:01.01 | Strom_M | teliax has been good to me; they have a POP in los angeles |
| 02:01.51 | *** join/#asterisk limbeaux (n=IceChat7@69.17.126.244) |
| 02:02.20 | tawaki | i have checked out teliax...looks good for quality...but they round calls at 60/6 and I need 6/6 to keep it within budget |
| 02:02.55 | Zipper_32 | Sorry to interrupt, but what do you mean by 60/6? |
| 02:03.03 | Strom_M | picky picky picky |
| 02:03.21 | SwK | Zipper_32, billing increment |
| 02:03.37 | SwK | 60/6 means first increment is 60 seconds, each aditional increment is 6 secs |
| 02:03.38 | tawaki | 60 seconds minimum even if call lasted 8 seconds |
| 02:03.52 | Zipper_32 | Ahh, thank you. |
| 02:04.00 | SwK | you see stuff like 60/60, 60/30, 60/6, 30/6, 6/6 etc |
| 02:04.13 | Zipper_32 | Understood. =) |
| 02:05.13 | SwK | most (anyone worth a damn that is) US domestic carriers are 6/6, then international from the US is typically like 30/6... if you can find some really good routes you might get intl at 6/6 |
| 02:05.41 | limbeaux | i have a dlink phone that will not connect to my asterisk server, my xlite clients all connect just fine. any ideas? |
| 02:05.59 | SwK | (really good US carriers are 6/1 or 1/1 if you can find it but you usually have to have a good ALOC (avg length of call) and 0 auto-dialer traffic |
| 02:06.12 | limbeaux | i guess i should say that i have no idea what most of the settings actually mean. |
| 02:06.23 | limbeaux | i just try to mimic xlite settings |
| 02:08.58 | *** join/#asterisk bkw_ (i=brian@adsl-70-143-39-207.dsl.tul2ok.sbcglobal.net) |
| 02:09.58 | Zipper_32 | limbeaux: pastebin your sip.conf section for the phones I'll take a gander |
| 02:10.42 | *** join/#asterisk nick125_lappy (n=nick@atarack/staff/nick125) |
| 02:11.01 | nick125_lappy | evening everyone! |
| 02:11.41 | limbeaux | thanks zipper, i am trying to see where to export the settings |
| 02:12.53 | *** join/#asterisk asteriskguy (n=learnast@cpe-75-80-111-113.socal.res.rr.com) |
| 02:13.27 | limbeaux | well, that sux, there is no way to back up my settings |
| 02:13.38 | limbeaux | so i cant post them in the windows |
| 02:13.41 | limbeaux | window |
| 02:14.01 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 02:14.09 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 02:14.12 | blitzrage | zup |
| 02:14.12 | Zipper_32 | I don't quite understand you... |
| 02:14.17 | blitzrage | me either |
| 02:14.37 | Zipper_32 | You have your settings on your server... You are accessing this server using... putty? |
| 02:14.39 | *** join/#asterisk sudhir492 (n=sudhir@c-71-62-102-201.hsd1.va.comcast.net) |
| 02:14.49 | Zipper_32 | or just a terminal window from *nix? |
| 02:14.51 | sudhir492 | hi all |
| 02:14.59 | *** join/#asterisk Brijn (n=bas@S010600e0b601c51e.vn.shawcable.net) |
| 02:15.05 | blitzrage | ahoi |
| 02:15.06 | limbeaux | oh, i was looking for the settings for my phone. |
| 02:15.07 | Brijn | Good evening all |
| 02:15.37 | Zipper_32 | limbeaux: Just pastebin the sip.conf settings that you're using on your asterisk server. |
| 02:15.39 | limbeaux | so where do i get my sip.conf settings? |
| 02:15.58 | Zipper_32 | limbeaux: Did you install Asterisk yourself? |
| 02:16.06 | nick125_lappy | karlhaines: ping |
| 02:16.22 | Brijn | Anyone a pointer to a nice and easy HOWTO for a Linksys SPA-2102? |
| 02:16.27 | limbeaux | trixbox |
| 02:16.50 | Zipper_32 | "Join #trixbox for trixbox support." |
| 02:16.56 | Zipper_32 | ~trixbox |
| 02:17.06 | jbot | Trixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums & WIKI at http://www.trixbox.org |
| 02:17.07 | limbeaux | ok |
| 02:17.13 | sudhir492 | what does asterisk use to send voicemail in email? |
| 02:17.27 | sudhir492 | For some reason, my box is not sending any emails |
| 02:17.59 | blitzrage | sendmail |
| 02:18.02 | blitzrage | make sure its installed |
| 02:18.27 | sudhir492 | I do see /usr/sbin/sendmail |
| 02:18.38 | blitzrage | I just install it on centos, and it works |
| 02:18.38 | *** join/#asterisk nxu7 (n=nxu7@S0106006097940f68.vw.shawcable.net) |
| 02:18.40 | blitzrage | make sure its running |
| 02:18.49 | sudhir492 | does it look anywhere else for sendmail? |
| 02:18.59 | blitzrage | you have to make sure the server is running |
| 02:19.07 | blitzrage | ps ax | grep sendmail |
| 02:19.17 | sudhir492 | I have qmail running instead of sendmail, and qmail indeed is running |
| 02:19.23 | blitzrage | ok -- thats not sendmail |
| 02:19.27 | blitzrage | so you have to configure qmail |
| 02:19.38 | nxu7 | has anyone had issue with 1.2.18 and call parking ( cmd ParkedCall/ParkAndAnnounce ), sometimes when you go to pick up the call, it complains it can't find that call in the parking lot |
| 02:20.09 | nxu7 | this is using sangoma hardware with asterisk |
| 02:20.19 | *** part/#asterisk limbeaux (n=IceChat7@69.17.126.244) |
| 02:20.49 | Zipper_32 | JT: Just FYI, it appears that the second Asterisk box has a deeper problem. I can't connect via IAX using any machines to it. It can see any other IAX devices, but they can't see it. |
| 02:21.20 | JT | Zipper_32: weird |
| 02:22.10 | Brijn | How many fields are there on the SPA-2102, OMG! |
| 02:22.28 | blitzrage | 42 |
| 02:22.58 | Strom_M | 800km |
| 02:23.21 | blitzrage | yes |
| 02:23.24 | *** join/#asterisk jsmith (n=jsmith@000-262-659.area3.spcsdns.net) |
| 02:23.59 | Brijn | I should have if there was a nice wiki article on how to set it up.. Because I think that 800km comes closest |
| 02:24.32 | Brijn | And I had tried to put the answer to everything in all the fields, did work either.. strange |
| 02:25.01 | tawaki | anyone present using asterisk for production level call room work? either integrated with sugar and/or gnudialer? |
| 02:25.13 | nxu7 | anyone knows whats the reason for having asterisk 1.2.x and 1.4.x branches ? |
| 02:25.53 | blitzrage | nxu7: more features |
| 02:26.11 | blitzrage | nxu7: plus lots of changes to the core |
| 02:26.13 | nxu7 | blitzrage, are they going to close one or another at some point ? |
| 02:26.18 | blitzrage | what do you mean? |
| 02:26.32 | blitzrage | 1.2 will go away at some point, yes |
| 02:26.50 | nxu7 | blitzrage, so everything thats in 1.2 will be available in 1.4 |
| 02:26.54 | blitzrage | once 1.6 comes out, then 1.2 will go into maintenance mode |
| 02:27.03 | blitzrage | nxu7: of course -- except that which has been deprecated |
| 02:27.31 | nxu7 | having two of them with active lines of development creates some confusion |
| 02:27.36 | blitzrage | no it doesn't |
| 02:27.42 | nxu7 | hm |
| 02:27.45 | blitzrage | there is only bug fixes in 1.2 and 1.3 |
| 02:27.47 | blitzrage | 1.4* |
| 02:27.50 | nxu7 | ah |
| 02:27.54 | blitzrage | development is done in trunk |
| 02:27.56 | nxu7 | so no new stuff ? |
| 02:27.59 | blitzrage | never |
| 02:27.59 | nxu7 | in 1.2 ? |
| 02:28.02 | nxu7 | ah ok |
| 02:28.02 | blitzrage | never |
| 02:28.18 | blitzrage | (well thats not exactly true -- but its the general rule) |
| 02:28.22 | nxu7 | do you think this Parked call issue I am having would go away with 1.4 ? |
| 02:28.38 | blitzrage | I don't know -- either you have a configuration error, or you've found a bug |
| 02:29.59 | blitzrage | if its a bug, the developers won't fix it until someone reports it with a clear bug report and the ability to reproduce it |
| 02:30.20 | nxu7 | ok np |
| 02:30.20 | *** join/#asterisk bkw_ (i=brian@adsl-70-143-39-207.dsl.tul2ok.sbcglobal.net) |
| 02:31.05 | anonymouz666 | I am about to sleep on keyboard |
| 02:31.31 | anonymouz666 | tomorrow I need to test chan_cellphone and iLBC chan_gtalk patch for ILBC codec |
| 02:32.09 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 02:32.57 | blitzrage | ok |
| 02:33.19 | anonymouz666 | blitzrage: I think it is.... bedtime! |
| 02:33.20 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 02:33.20 | *** mode/#asterisk [+o anthm] by ChanServ |
| 02:33.23 | anonymouz666 | heh |
| 02:34.02 | blitzrage | go |
| 02:34.21 | *** join/#asterisk hieunm_vips (n=hieunm_v@210.245.57.162) |
| 02:35.33 | *** join/#asterisk entelechy (n=chatzill@mail.beanproducts.com) |
| 02:35.57 | hieunm_vips | I'm reading Dial app source code and I don't understand privacy/screen option (N & n) |
| 02:36.20 | hieunm_vips | Could you show me an use case of privacy option in practice |
| 02:37.43 | entelechy | can anyone point me to a sample/default extensions.conf with the asterisk_guitools context (for asterisk-gui) ? apologies but nobody seems awake in #asterisk-gui to help with this |
| 02:39.26 | jsmith | hieunm_vips: Sure... Alice calls into Asterisk, and Asterisk calls Bob on his cell phone. Asterisk says "Hey Bob, you've got a call from Alice. Do you want to take it, or send her to voicemail?" |
| 02:42.56 | *** part/#asterisk nxu7 (n=nxu7@S0106006097940f68.vw.shawcable.net) |
| 02:45.31 | jsmith | entelechy: http://www.asterisknow.org/developers/gui-guide |
| 02:46.57 | entelechy | jsmith: thank you, i have been staring at that one and the few other docs i could find. makes no mention of how to add the "asterisk_guitools" context entries to extensions.conf if they arent already there. which they arent. does asterisk-gui require asterisk > 1.4.0 ? |
| 02:47.12 | jsmith | Yes |
| 02:47.18 | jsmith | I'm pretty sure it does |
| 02:47.34 | jsmith | I don't think the manager-over-http facility was in Asterisk before 1.4 |
| 02:47.41 | entelechy | im using 1.4.0 |
| 02:47.43 | jsmith | (or the built-in web server, for that matter) |
| 02:47.57 | entelechy | it works, but the CLI gives this message: |
| 02:47.59 | jsmith | I'd use 1.4.4 -- there have been a lot of fixes in that regard |
| 02:48.34 | entelechy | [May 15 21:48:08] NOTICE[12329]: chan_local.c:562 local_alloc: No such extension/context executecommand@asterisk_guitools creating local channel |
| 02:48.35 | entelechy | [May 15 21:48:08] NOTICE[12329]: channel.c:2762 __ast_request_and_dial: Unable to request channel Local/executecommand@asterisk_guitools |
| 02:48.37 | entelechy | mail*CLI> |
| 02:49.09 | jsmith | Right... the GUI is trying to execute a system command, but that extension doesn't exist. |
| 02:49.43 | entelechy | i understand, i was hoping i could find the "extensions.conf" containing the proper context to have a look at it before i start comparing changelogs from 1.4.0 to 1.4.4 :-) |
| 02:50.28 | jsmith | I think that context is added when you install the gui |
| 02:50.38 | jsmith | At least, that's what that page says |
| 02:50.42 | entelechy | oh you mean if i ran "make samples" ??? |
| 02:50.54 | jsmith | That page I pointed to also shows what that context looks like |
| 02:51.48 | entelechy | http://lists.rpath.org/pipermail/asterisk-commits/2006-December/000626.html contains some hints that i dont see in the page you posted? |
| 02:51.53 | entelechy | i.e. extenions.conf.gui.patch |
| 02:52.07 | entelechy | not mentioned anywhere in the gui setup guide... nor are any of the entries the patch would seem to add |
| 02:52.52 | jsmith | Well, that's obvioiusly a *patch*, and not part of the standard gui install |
| 02:53.18 | entelechy | yes - the 1.4.0 extensions.conf doesnt have the proper context. trying to find the default/blank one that comes with 1.4.X that actually INCLUDES that context |
| 02:54.48 | anonymouz666 | using manager do originate calls, does the cdr get right? |
| 02:54.55 | anonymouz666 | callfiles dont |
| 02:55.12 | nick125_lappy | I need an easy way to make an automated call at a specific time, what would be the easiest way to do that? |
| 02:55.29 | entelechy | forgive me if i am missing something obvious, i'd love to upgrade to 1.4.4 but then i need to very carefully compare changelogs to make sure the upgrade doesnt break anything |
| 02:56.08 | Strom_M | it's a bugfix release |
| 02:56.19 | Strom_M | if anything, it will /fix/ problems |
| 03:00.47 | entelechy | no doubt |
| 03:01.20 | entelechy | i was just trying to avoid having to manually redo the entire /etc/asterisk/*.conf affected by this upgrade |
| 03:01.29 | entelechy | since obviously at least one file, extensions.conf, *is* affected |
| 03:04.12 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:04.40 | entelechy | dont see it in 1.4.4 - extensions.conf is identical |
| 03:05.19 | entelechy | i realize im very new to asterisk. but it seems like it should be a fairly simple matter to find where the context asterisk_guitools is supposed to be defined |
| 03:05.32 | entelechy | i already have working providers, dialplan, extensions w/1.4.0, setup by hand |
| 03:06.30 | entelechy | following instructions at http://www.asterisknow.org/developers/gui-guide and in the readme in asterisk-gui/README didnt do anything to set up this context "asterisk_guitools" |
| 03:06.52 | entelechy | nor even mention it |
| 03:08.21 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 03:09.40 | entelechy | and grepping the entire code tree of 1.4.4 results in not one mention of asterisk_guitools, not in the sample configs, the docs, nowhere |
| 03:10.17 | JT | entelechy: the gui stuff is not part of asterisk |
| 03:10.42 | *** join/#asterisk bkw_ (i=brian@adsl-70-143-39-207.dsl.tul2ok.sbcglobal.net) |
| 03:11.12 | entelechy | JT: I understand. I am following the instructions given to set up asterisk-gui. i mentioned above, nobody in that channel awake to help with this issue. i already have asterisk 1.4.0 set up just fine, working, multiple extensions, multiple providers. |
| 03:11.24 | entelechy | 10 cisco 7940's |
| 03:11.33 | entelechy | voicemail |
| 03:11.35 | entelechy | im not stupid |
| 03:11.55 | JT | so what does this have to do with the gui then? :) |
| 03:12.18 | entelechy | the gui is attempting to call a context called asterisk_guitools |
| 03:12.41 | entelechy | but the gui installation scripts dont do anything to set up such a context anywhere in /etc/asterisk/*.conf |
| 03:12.56 | JT | i see |
| 03:19.34 | *** join/#asterisk Math` (n=seb@modemcable037.229-56-74.mc.videotron.ca) |
| 03:19.49 | entelechy | well, i definitely appreciate the help, i realize asterisk-gui represents the bloody tip of the bleeding edge, and i am definitely still very new to anything but a very basic asterisk setup. im sure its either my own ignorance or its an issue that will probably get addressed in the next few days/diffs/patches |
| 03:21.40 | *** join/#asterisk jsmith (n=jsmith@000-262-659.area3.spcsdns.net) |
| 03:22.04 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 03:23.42 | JT | entelechy: most of us don't use it, so don't know |
| 03:24.55 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 03:24.56 | *** mode/#asterisk [+o mog] by ChanServ |
| 03:25.08 | entelechy | well, i definitely appreciate the help either way :) |
| 03:26.30 | *** join/#asterisk kiwoneka (n=kiwoneka@KTNRON06-1168103823.sdsl.bell.ca) |
| 03:26.52 | *** join/#asterisk phobus (n=phobus@crlspr-69.65.75.232.myacc.net) |
| 03:27.09 | kiwoneka | good evening to all |
| 03:27.57 | entelechy | the only reference i can find to an extensions.conf with entries for the asterisk_guitools context, is the patch i mentioned above: http://lists.rpath.org/pipermail/asterisk-commits/2006-December/000626.html ... trying to find a version of extensions.conf with the patch applied, since the one in the link is already almost 6 months old, and i was hoping i could just copy and paste the... |
| 03:27.58 | entelechy | ...relevant section from an actual 200705xx extensions.conf.sample not from a potentially partial patch from 2006 (or worse) :-) |
| 03:28.00 | entelechy | plus |
| 03:28.34 | entelechy | that patch doesnt mention the particular string "executecommand@asterisk@guitools" mentioned in the asterisk CLI notices |
| 03:29.00 | entelechy | thus im guessing its just a partial patch... |
| 03:29.27 | kiwoneka | i need an opinion or two, i need to determine why i should upgrade my polycoms' boot rom |
| 03:30.01 | *** join/#asterisk kiscokid (n=Ron@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 03:40.59 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1128667111.dsl.bell.ca) |
| 03:51.00 | *** join/#asterisk JT_ (n=jon@unaffiliated/jt) |
| 03:52.02 | *** part/#asterisk kiwoneka (n=kiwoneka@KTNRON06-1168103823.sdsl.bell.ca) |
| 03:52.48 | *** join/#asterisk bmg505 (n=leon@196.209.179.240) |
| 03:56.41 | *** join/#asterisk [[blah]asfd (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 03:56.51 | *** join/#asterisk genmud (n=genmud@about/windows/regular/genmud) |
| 03:56.58 | genmud | Helo |
| 03:57.10 | genmud | Anyone in here use cisco phones with asterisk? |
| 03:57.12 | [[blah]asfd | does asterisk use more than 32 bit if i have a 64 bit system? |
| 03:57.44 | Strom_M | genmud: I do |
| 03:57.52 | genmud | Strom_M, what kind of phones you using? |
| 03:57.58 | Strom_M | 7960 |
| 03:58.25 | genmud | i have a 7940 and 7941 and having a terrible time to make them play nice |
| 03:58.26 | genmud | :( |
| 03:58.40 | Strom_M | ok |
| 03:58.40 | genmud | i keep getting a 401 authorization failed on my 7941 |
| 03:58.48 | genmud | is what the packet dump is saying |
| 03:59.13 | Strom_M | well, youve got a typo in your configs somewhere then :) |
| 03:59.34 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:59.44 | genmud | that would be a username/password config issue, right? |
| 03:59.52 | genmud | because I have checked/tripple checked that |
| 03:59.54 | Strom_M | quite likely, yes |
| 04:00.03 | genmud | What other causes could it be? |
| 04:01.24 | Strom_M | pastebin your sip.conf and your phone config file |
| 04:05.17 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 04:06.35 | Strom_M | hello? |
| 04:07.28 | genmud | Strom_M, msg |
| 04:07.43 | Strom_M | here's the problem |
| 04:07.44 | Strom_M | line1_authname: "200" |
| 04:07.56 | Strom_M | your sip.conf says 209, not 200 |
| 04:08.59 | Math` | is it possible to use patterns in CID matching? example *10/33[01] => NoOp(something); |
| 04:10.11 | Math` | guess its gonna be an RTFS case |
| 04:10.23 | *** part/#asterisk [[blah]asfd (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 04:10.42 | Strom_M | Math`: i dont know :) |
| 04:11.08 | Strom_M | why not try it/ |
| 04:11.08 | Strom_M | ? |
| 04:12.06 | Math` | its faster to read the source :P |
| 04:12.43 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:13.19 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 04:13.48 | JT | Math`: if it works, it'll probably be *10/_33[01] though, not *10/33[01] |
| 04:14.12 | Strom_M | genmud: did that resolve the problem? |
| 04:14.15 | Math` | treats it as a pattern default! |
| 04:14.22 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 04:14.24 | Math` | so *10/33XX would work :) |
| 04:15.08 | Math` | comparison is done by ast_extension_match() |
| 04:15.39 | genmud | No, still looking at |
| 04:15.39 | JT | Math`: so underscore not needed? |
| 04:15.52 | genmud | the problem that I was having the 401 was the 7941 |
| 04:15.59 | genmud | which is newer, and very unsupported :/ |
| 04:16.00 | genmud | lol |
| 04:16.03 | Strom_M | genmud: but you changed authname to 209 instead of 200, right? |
| 04:16.07 | genmud | yea |
| 04:16.08 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 04:16.13 | Qwell | unsupported? |
| 04:16.20 | Strom_M | so if you're having the problem with the 7941, why didnt you pastebin that? |
| 04:16.24 | genmud | Well, not very many docs |
| 04:16.33 | Qwell | it's...the same as the 7940 |
| 04:16.39 | genmud | umm, no |
| 04:16.44 | genmud | not even close |
| 04:16.46 | Math` | JT: match is done by _extension_match_core() which checks for a _ |
| 04:16.51 | Qwell | okay, it had LEDs on the side |
| 04:16.51 | Math` | so it would require one |
| 04:16.52 | genmud | configs are all xml |
| 04:16.54 | Qwell | otherwise, it's the same |
| 04:16.57 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 04:17.06 | genmud | very different voip images |
| 04:17.09 | genmud | lots of differences |
| 04:17.39 | Strom_M | genmud: so how about you stop pontificating and start pastebinning, eh? :) |
| 04:17.55 | genmud | Strom_M, gonna get the configs |
| 04:17.59 | genmud | and a full tcpdump for you |
| 04:18.07 | Strom_M | i dont want a tcpdump |
| 04:18.13 | Strom_M | just give me config plz |
| 04:18.19 | Strom_M | also kthx |
| 04:18.21 | Math` | if pattern[0] != '_' it just skips pattern eval to speed it up |
| 04:18.31 | Math` | (optimized) |
| 04:19.08 | JT | Math`: right |
| 04:19.40 | Math` | JT: so basically you were right with *10/_330[01] |
| 04:19.51 | Strom_M | hmm |
| 04:19.52 | *** join/#asterisk bkw_ (i=brian@adsl-70-143-39-207.dsl.tul2ok.sbcglobal.net) |
| 04:20.06 | Strom_M | it's 11:19. Waffle House sounds tempting. |
| 04:20.15 | Math` | Waffle House? |
| 04:20.19 | Strom_M | Waffle House |
| 04:20.47 | gerphimum | waffle, side of scrambled eggs + chz, side of bacon, large choc milk FTW |
| 04:21.02 | JT | hrm |
| 04:21.18 | Strom_M | actually, hash browns are FtW |
| 04:21.33 | gerphimum | if you like hash browns theyre teh bomb |
| 04:21.37 | gerphimum | i personally do not |
| 04:22.03 | JT | word from the wise: never ever eat mcdonalds hash browns + thickshake at the one sitting |
| 04:22.07 | Math` | hmm asterisk need array channel variables, string token splitting and foreaches |
| 04:22.18 | JT | typical symptoms: angry stomach for at least 3 hours |
| 04:22.33 | genmud | Strom_M, ms |
| 04:22.34 | genmud | msg* |
| 04:22.34 | Corydon76-home | fore aches? |
| 04:22.40 | Math` | lol |
| 04:22.48 | Math` | foreach somevar in somearray |
| 04:22.51 | Strom_M | JT: I think you can delete everything beyond the word "mcdonalds" |
| 04:23.10 | Corydon76-home | You can already do all those things with ARRAY(), While(), and CUT() |
| 04:23.12 | JT | Strom_M: heh, sometimes it's convenient, and the only thing open |
| 04:23.24 | Math` | :D |
| 04:23.36 | Math` | then a string split fct is needed I guess lol |
| 04:23.47 | Corydon76-home | That's what CUT() does |
| 04:23.51 | Math` | wow |
| 04:24.05 | Math` | thats nice |
| 04:24.22 | Corydon76-home | and use FIELDQTY to find out how many items are in the variable |
| 04:24.57 | Corydon76-home | You don't think you're the first one who has needed these types of things, do you? |
| 04:25.04 | Math` | no |
| 04:25.05 | Strom_M | genmud: I really wish you'd pastebin your files in the channel |
| 04:25.12 | Math` | I was just wondering if people were just using AGI |
| 04:25.22 | genmud | no worries |
| 04:25.30 | Math` | or if there was some actual stuff in there |
| 04:25.30 | JT | Strom_M: be careful what you wish for ;) |
| 04:25.32 | Corydon76-home | I am the AGI killer |
| 04:25.37 | Math` | hehe |
| 04:25.50 | Math` | I wonder what kind of overhead there is in AGI |
| 04:26.07 | JT | spawning an interpreter with every call i think... |
| 04:26.20 | Math` | guessed that |
| 04:26.33 | Math` | but after that... msgs are passed back using stdin/out I think |
| 04:26.53 | Corydon76-home | Yes, but you lose a good bit of control with that, too |
| 04:26.53 | Strom_M | genmud: I'm not sure, but I would recommend you try removing all the unnecessary settings from sip.conf |
| 04:26.53 | rudholm | and if you use it, spawing PERL, which is no lightweight |
| 04:27.08 | bkw_ | res_perl |
| 04:27.12 | bkw_ | works great |
| 04:27.19 | bkw_ | but has a few issues with concurrency |
| 04:27.23 | bkw_ | but is really fast |
| 04:27.26 | bkw_ | or convert things to FAGI |
| 04:27.30 | Math` | I was wondering for php.... if you link with libphp (just like apache does), it should improve speed a bit |
| 04:27.40 | JT | lol php |
| 04:27.46 | Math` | since a lot of people are using php as agi |
| 04:28.03 | JT | a lot of people are also insane... |
| 04:28.15 | Corydon76-home | You're welcome to use whatever tool you like. I'm just glad it's not running on my system |
| 04:28.23 | Math` | lol |
| 04:28.35 | Math` | I just work everything in ael2 |
| 04:29.06 | bkw_ | ael is just extensions.conf with a new face.. its still the same old thingon the undersie |
| 04:29.15 | rudholm | can I write my AGI programs in RPG-III ? |
| 04:29.16 | bkw_ | er underside |
| 04:29.52 | JT | rudholm: does it do stdin and stdout? |
| 04:29.54 | Math` | yeah it "compiles" in extensions.conf format... well the pbx.c hasn't be rewritten for ael2 |
| 04:30.29 | Corydon76-home | The core doesn't need to be rewritten for ael2. That is a major strength |
| 04:30.57 | genmud | Strom_M, the packet dump is located at: http://pastebin.ca/490666 |
| 04:31.03 | genmud | it is really really strange |
| 04:31.16 | genmud | because it just throws a 401 error after it tries to register :( |
| 04:31.22 | Math` | yeah and you can use it with previous versions of asterisk with aelparse -w |
| 04:31.24 | genmud | username/password settings are corrct |
| 04:31.35 | rudholm | JT: it doesn't run on OSes that have unix file handles |
| 04:31.46 | genmud | Does anyone else in here use the cisco 79x1 phones? |
| 04:31.56 | Corydon76-home | Math`: assuming that the apps it compiles to are available in the earlier versions |
| 04:32.10 | Math` | yeah |
| 04:32.33 | bkw_ | blah |
| 04:32.56 | JT | rudholm: non optimal then |
| 04:33.20 | Corydon76-home | That won't be the case, exactly, when trunk is branched. We're using Gosub for branching in trunk, not Macro |
| 04:33.33 | rudholm | I got my asterisk to send CallerID to my slimdevices Squeezeboxes. very happy about that. |
| 04:33.44 | Math` | yeah I read that thread |
| 04:33.57 | Math` | it will stack-overflowing too easily |
| 04:34.11 | Math` | s/will/will stop/g |
| 04:34.51 | *** join/#asterisk donkilla (n=rogersoc@41.206.35.226) |
| 04:35.00 | donkilla | Hi all |
| 04:35.28 | donkilla | How can i configure DID for my extensions |
| 04:35.44 | Strom_M | by asking less ambiguous questions |
| 04:36.14 | rudholm | by making a correct extensions.conf file |
| 04:36.51 | donkilla | I have an ISDN connection and need external callers to dial an extension directly without going through an operator |
| 04:37.05 | Math` | then make a context answer the call and provide IVR functionality |
| 04:37.10 | Strom_M | donkilla: ISDN PRI or ISDN BRI? |
| 04:37.26 | donkilla | ISDN PRI |
| 04:37.31 | JT | donkilla: then have an extension match for each DID in your isdn incoming context |
| 04:37.35 | donkilla | I'm using E1 |
| 04:37.43 | Strom_M | donkilla: do you have DNIS from your telephone company? |
| 04:38.30 | donkilla | I'm linking with an Alcatel OmniPCX Office |
| 04:38.32 | JT | i think that's something that's there by default on an E1, Strom_M |
| 04:38.39 | JT | oh |
| 04:39.00 | JT | donkilla: OmniPBX, you mean? |
| 04:39.17 | Strom_M | donkilla: in theory, you should just be able to match in the inbound context you have your PRI assigned to |
| 04:39.26 | donkilla | Yes i'm able to see the Caller ID of the Alcatel extenson on my Asterisk operator console |
| 04:39.45 | JT | donkilla: simply use extensions matches for each one |
| 04:39.45 | donkilla | I mean OmniPCX |
| 04:40.40 | donkilla | Let me try it out |
| 04:40.48 | Math` | is relying on chan_local a lot going to considerably degrade performance or is it somewhat lite enough? |
| 04:41.00 | Mercestes | anyone awake? |
| 04:41.12 | Strom_M | ALL DEAD HERE |
| 04:41.20 | Math` | yup, all sleep(y/ing) |
| 04:41.44 | donkilla | I'm amused at how people think in terms of their locality |
| 04:41.45 | rudholm | please deposit ten cents for the next one minute |
| 04:41.51 | Mercestes | What would cause a zap interface to have no audio? |
| 04:41.53 | donkilla | It's a global village! |
| 04:42.07 | Math` | Mercestes: uhm the simplest form includes the wire being unplugged from the card |
| 04:42.10 | Mercestes | two cards, sangoma and digium |
| 04:42.27 | Mercestes | Math` Yea, wire's plugged in, no alarms, all green lights, channels are up |
| 04:42.32 | Math` | pri? |
| 04:42.38 | Mercestes | Sip calls give audio, but no zap audio |
| 04:42.46 | Mercestes | 2 PRI's to be exact |
| 04:43.05 | Math` | where is the call sent to |
| 04:43.06 | rudholm | is Asterisk running? :) |
| 04:43.11 | Math` | lol |
| 04:43.18 | Mercestes | we lost the controller card in the server, (embedded Perc4dc or something). So the dell guy replaced the controller card in this server. |
| 04:43.24 | Mercestes | same configs, same ports, all the same crap. no audio |
| 04:43.25 | *** join/#asterisk Chris-NB (n=chris@home.fuerstaller.com) |
| 04:43.38 | Mercestes | ok, can we be a little serious? |
| 04:43.43 | Mercestes | I'm at work here |
| 04:43.45 | Math` | Mercestes: where is the call sent to |
| 04:43.54 | JT | sure he didn't damage the PRI card PCBs while he was at it? |
| 04:44.02 | Mercestes | MoH, my cell phone, another sip phone, Background of wave files. you name it |
| 04:44.06 | JT | serious, why? you never are, Mercestes ;) |
| 04:44.16 | Mercestes | JT: a sangona and a digium card both? |
| 04:44.22 | Math` | JT: if he damaged TDM hardware he would have alarms |
| 04:44.23 | Mercestes | JT: and I still manage to help out. ;) |
| 04:44.26 | JT | maybe, maybe not |
| 04:44.33 | Mercestes | sometimes |
| 04:44.45 | JT | i wouldn't let a dell tech touch my servers |
| 04:44.53 | Math` | Mercestes: out to PRI on in to PRI |
| 04:45.00 | Mercestes | It's one giant embedded Perc/SCSI/riser card that's all in one on th emobo |
| 04:45.09 | JT | it's a case of "oh, so that's our replacement part? here, hand me the part... ok you can have the old part now" |
| 04:45.17 | Mercestes | the only question he had was maybe the firmware wasn't the same version |
| 04:45.46 | JT | dell techs are not allowed to touch the inside of my servers :P |
| 04:46.15 | Mercestes | ok, point. But I have lots of green lights in my server and no audio on my PRIs |
| 04:46.31 | JT | yeah maybe it's irq sharing or something stupid? |
| 04:47.05 | Math` | Mercestes: did u check the other end of the pri? do you Answer() the call? |
| 04:47.31 | Mercestes | <PROTECTED> |
| 04:47.33 | Mercestes | damnit |
| 04:47.36 | Mercestes | extra space |
| 04:47.52 | Mercestes | JT: Umm, uhci_hcd:usb2 can wanpipe share with usb2? |
| 04:48.03 | Downchuck | yeah why would you let a dell tech in your server |
| 04:48.08 | JT | dunno |
| 04:48.15 | Mercestes | me meither |
| 04:48.29 | Mercestes | Downchuck, because my company pays for gold support |
| 04:48.32 | JT | Mercestes: try getting one card at a time to work? |
| 04:48.37 | Downchuck | Mercestes: that's for the phone |
| 04:48.47 | Mercestes | JT: I am. I unload wct4xxp and wanrouter start and vice verse . |
| 04:48.47 | Downchuck | the phone support, i mean |
| 04:49.02 | JT | Mercestes: by removing one card |
| 04:49.09 | Mercestes | Downchuck, Then yo udon' tknow what gold support is. We have 4 hour on site response |
| 04:49.22 | Mercestes | JT: Alright, let me give that a go. thanks |
| 04:49.26 | Downchuck | that is pretty nice |
| 04:49.27 | Mercestes | BRB |
| 04:49.39 | Math` | Mercestes: why would u ignore people trying to help I dont get that... anyways... back 2 work |
| 04:49.39 | Mercestes | Downchuck, yea, if they don't hose your server. ;) I supervised *closely* \ |
| 04:49.48 | Mercestes | and the guy was like....90.....he had to have some decent experience. |
| 04:50.12 | Mercestes | Math` Sorry, I thought you were being smart with me. Answer() would only solve DTMF and one way audio, not no audio, sorry. |
| 04:50.20 | Mercestes | BRB |
| 04:50.38 | JT | Mercestes: is it Answer()ing or not? |
| 04:50.46 | Math` | depends if the other way accepts early media or not.... |
| 04:54.41 | *** join/#asterisk CoffeeIV (i=rgr@rrcs-71-42-183-82.sw.biz.rr.com) |
| 04:57.04 | CoffeeIV | I have an asterisk server on a non-public IP address that has been used as an office PBX. I would like to add a second asterisk server with a public IP, have it allow client SIP or IAX2 softphones to connect from the internet, and have it somehow forward the connections to the inside asterisk so it is as if those phones were inside the office network. Is there a better way do this besides just forwarding IAX2 ports ? |
| 04:57.54 | Math` | you could just forward iax2 to the inside asterisk without needing a second one |
| 05:01.35 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1128667111.dsl.bell.ca) |
| 05:01.40 | CoffeeIV | I think that's what I will do |
| 05:01.57 | donkilla | Thaks everyone |
| 05:02.13 | Math` | CoffeeIV:make sure you set externip= correctly in your sip.conf file |
| 05:02.23 | Math` | and that you forward the proper RTP ports for sip clients |
| 05:02.28 | donkilla | I just included my subscribers context in incoming |
| 05:02.28 | CoffeeIV | however, if there is a way to use asterisk as a kind of gateway, I could also handle SIP phones without having to figure out SIP / NAT settings |
| 05:02.30 | Math` | IAX2 uses only 1 port but sip uses a wide range |
| 05:02.56 | Math` | sure, just setup that asterisk to forward everything to the 2nd one |
| 05:03.15 | Downchuck | So this dell support.. Do they show up within 2 - 4 hrs with spare parts, plug them in and then leave? |
| 05:03.17 | CoffeeIV | Math: I have already wasted too much of my life on iptables/SIP/NAT/RTP ports etc etc, I will never attempt that problem again -- if I forward ports, I will deal only with IAX2, period |
| 05:03.28 | Math` | ok |
| 05:03.29 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 05:03.36 | Math` | but you have to config each "outside" extension in the local one too (pointing to the outside one from the inside) |
| 05:03.47 | bintut | hello all.. |
| 05:03.50 | CoffeeIV | yes |
| 05:03.50 | Math` | or make a dialplan login (ex: all extensions starting with 21 are external) |
| 05:04.23 | Math` | so forward 4569 for iax2 and then link everything :) |
| 05:04.25 | bintut | i'm building a vanilla kernel 2.6.21.1.. is there a module that i need to enable on my kernel rebuilding? |
| 05:04.36 | Math` | or just make the inside register to the outside one, you dont even have to forward iax2 then :) |
| 05:04.43 | Math` | it will do its nat mapping automatically on registration |
| 05:05.06 | bintut | to support my Digium TDM4xx PCI card ==> 01:00.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
| 05:05.23 | Math` | bintut: the module will be compiled when you install zaptel |
| 05:05.34 | Math` | bintut: make sure you dont delete your kernel sources as you are going to need them |
| 05:06.07 | *** join/#asterisk mightnare (n=mike@s230165.ppp.asahi-net.or.jp) |
| 05:06.09 | donkilla | How can i direct a particular zap channel to a particular extension? |
| 05:06.16 | bintut | Math`: yeah, i do have and i won't delete my kernel source.. but, is there other thing i need to enable on my kernel building? |
| 05:07.13 | bintut | Math`: do i need to enable the ISDN SubSystem of my kernel config? |
| 05:07.30 | bintut | Math`: or anything from my kernel config? |
| 05:08.40 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 05:10.53 | Math` | bintut: I always compiled with my distro's kernel... so cant tell u there |
| 05:11.03 | ohadz | hi, i am trying to setup my dialplan and have incoming and outgoing to work. for incoming i am not getting the error - -the number you have reached is not in service. i know that it is in service |
| 05:11.05 | bintut | Math`: it's ok |
| 05:11.53 | bintut | by the way, anyone here knows where can i find a binary rpms of asterisk-1.4.4 for centos-5 for i386? |
| 05:12.34 | Math` | I just compile everything from scratch... u have more control |
| 05:13.42 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:14.38 | ohadz | anyone? |
| 05:16.28 | JT | bintut: do not enable that isdn subsystem stuff |
| 05:16.33 | JT | it's isdn4linux rubbish |
| 05:16.36 | JT | that's deprecated |
| 05:17.08 | JT | you need crc_ccitt compiled in or modularised |
| 05:17.17 | JT | HZ=1000 can help sometimes too |
| 05:17.25 | JT | some say pre-empt is good for asterisk |
| 05:18.32 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 05:22.22 | Math` | uhm any way to inherit chan vars to chan_local? |
| 05:23.02 | Math` | nvm found it |
| 05:31.15 | *** join/#asterisk DarKnesS_WolF (n=wolf@unaffiliated/sherif) |
| 05:36.21 | *** join/#asterisk mightnare (n=mike@s230165.ppp.asahi-net.or.jp) |
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| 05:53.07 | nick125_lappy | Anyone here know of an easy way to place an automated call (I need to send a wakeup call at 4:30am to two phone numbers)? |
| 05:54.08 | nick125_lappy | I remember there used to be a way involving /var/spool/asterisk |
| 05:55.08 | Math` | yeah |
| 05:55.14 | Strom_M | drop a call file into /var/spool/asterisk/outgoing |
| 05:55.22 | Math` | timstamp the file with the time you want * to call |
| 05:55.46 | Math` | see http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out |
| 05:56.43 | nick125_lappy | It'd be cool if I could get it to play hold music once it picks up, but, its not mandatory |
| 05:57.10 | nick125_lappy | I also need it to hang up within 15 seconds of starting to playing |
| 05:57.12 | JT | there's a sample call file in the sources too |
| 06:00.16 | bintut | gtg now.. thanks all.. :) |
| 06:01.58 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 06:05.11 | *** join/#asterisk Ifaistos (n=stelios@ipa226.211.tellas.gr) |
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| 06:13.55 | *** join/#asterisk Ifaistos (n=stelios@ipa226.211.tellas.gr) |
| 06:14.50 | CrazyTux | Where does asterisk normally store like message information for voicemail, i.e. duration etc? |
| 06:15.31 | *** join/#asterisk Ifaistos (n=stelios@ipa226.211.tellas.gr) |
| 06:19.39 | Math` | in /var/spool/asterisk/voicemail |
| 06:21.58 | *** join/#asterisk Keltus (n=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 06:24.06 | DarKnesS_WolF | how to force the value of caller number in the dialplan ? |
| 06:24.32 | DarKnesS_WolF | i want the callerid=${caller(num)} == ports |
| 06:26.19 | nick125_lappy | DarKnesS_WolF: Set(CALLERID(num)=blah) |
| 06:28.19 | DarKnesS_WolF | nick125_lappy: 1.4 |
| 06:28.23 | DarKnesS_WolF | i don't think this will work on 1.4 |
| 06:29.08 | nick125_lappy | It should |
| 06:29.46 | Mercestes | I'm down to just a sangoma card now, on a new PCIx port, and I still get no audio on PRI. I can call in fine, dial out, i can even dial out one PRI into the other one. no audio |
| 06:30.04 | DarKnesS_WolF | nick125_lappy: hmmm it did ;-) thx |
| 06:30.12 | nick125_lappy | DarKnesS_WolF: no problem |
| 06:37.02 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 06:37.36 | *** join/#asterisk nitrus^ (n=Sarah@cpe-76-166-246-175.socal.res.rr.com) |
| 06:38.02 | nitrus^ | has anyone run asterisk on a conroe based xeon? |
| 06:38.37 | nitrus^ | im thinking about buying a intel 3000 chipset with a xeon but im not sure if people have had good success with it |
| 06:39.38 | JT | Mercestes: try recompiling? |
| 06:39.43 | JT | nitrus^: is that an older series? |
| 06:39.59 | Mercestes | JT: yea, tried wanrouter stock configs, and my own editted configs. nada |
| 06:40.07 | nitrus^ | i believe so |
| 06:40.13 | nitrus^ | 5000 is the newer one |
| 06:40.20 | Mercestes | full rebuild of wanrouter stuff |
| 06:40.23 | nitrus^ | and i believe the 7000 is coming out soon or possibly already out |
| 06:42.26 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 06:42.44 | JT | nitrus^: yeah i'd heard of 5000 |
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| 06:43.46 | nitrus^ | how's the compatibility of that? |
| 06:43.58 | JT | fine as far as i've heard |
| 06:45.18 | nitrus^ | i wonder if the performance on g.729 is good the processor is supposed to be far superior to older generation xeons |
| 06:45.46 | JT | 5000 is just a chipset, a lot of different xeons run on it |
| 06:45.51 | JT | and they should be superior |
| 06:48.10 | Mercestes | JT: Any ideas? |
| 06:48.23 | Mercestes | all on configs that worked before my riser card ate it |
| 06:48.53 | JT | no idea |
| 06:48.59 | JT | quite weird really |
| 06:49.06 | JT | zttest is ok? |
| 06:50.50 | Mercestes | yea |
| 06:50.55 | Mercestes | everything is ok, except....no audio |
| 06:51.00 | Mercestes | maybe I have NAT on my zap channels |
| 06:51.07 | Mercestes | or a firewall |
| 06:51.20 | Mercestes | I can even dial out PRI#3 and come back in on Pri#1 |
| 06:52.03 | Mercestes | holy Mary mother of God it works. |
| 06:52.08 | Mercestes | ....and I don't even know why. :( |
| 06:52.14 | Mercestes | I just test dialed to see if DTMF worked. |
| 06:53.31 | JT | umm so you mean you've only been testing with PRI to voip? |
| 06:54.42 | Mercestes | Dialed from my cell phone |
| 06:54.47 | Mercestes | voip to pri works |
| 06:54.49 | Mercestes | well no. |
| 06:54.52 | Mercestes | voip to server works. |
| 06:54.59 | Mercestes | voip to pri doesn;'t work and pri to pri doesn't work |
| 06:55.02 | Mercestes | s/doesn't/didn't/ |
| 06:55.07 | Mercestes | now I have to reboot to fix moh. |
| 06:56.22 | JT | how will that fix moh? |
| 06:56.44 | Mercestes | because I broke it |
| 06:58.20 | Mercestes | gah, now I get ful audio on span 1 and no audio on span 2 |
| 06:59.00 | Mercestes | ok, audio on span 1 and 3 and no audio on span 2. |
| 07:02.44 | c4t3l | patlooptest |
| 07:06.12 | *** join/#asterisk signius (n=IceChat7@dsl-217-155-69-102.zen.co.uk) |
| 07:08.27 | *** part/#asterisk signius (n=IceChat7@dsl-217-155-69-102.zen.co.uk) |
| 07:11.31 | nitrus^ | anyone here use voicepulse? |
| 07:12.15 | *** join/#asterisk kombi (n=kombi@195.158.185.196) |
| 07:12.26 | kombi | which pri card should i get? |
| 07:12.37 | *** join/#asterisk qdk (n=qdk@193.226.189.115) |
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| 07:18.12 | JT | kombi: one with echo cancellation :) |
| 07:20.04 | Mercestes | <PROTECTED> |
| 07:24.22 | JT | bad zap timing |
| 07:24.25 | JT | usually |
| 07:24.44 | Ifaistos | kombi : Eicon divae server cards have hardware echo cancellation |
| 07:24.53 | Ifaistos | s/divae/diva |
| 07:25.20 | JT | ... |
| 07:25.33 | JT | who the hell uses eicon diva cards for PRIs? |
| 07:25.39 | JT | i thought they were only BRI anyway |
| 07:25.52 | Ifaistos | sorry i read BRI.... |
| 07:26.04 | Ifaistos | too eraly in the morning here |
| 07:26.06 | Mercestes | can someone check my configs please???? |
| 07:26.13 | Mercestes | http://pastebin.ca/ |
| 07:26.18 | Mercestes | damnit |
| 07:26.20 | JT | eicon divas don't have much driver support anyway iirc |
| 07:26.28 | Mercestes | http://pastebin.ca/ |
| 07:26.33 | Mercestes | damnit again |
| 07:26.44 | Mercestes | http://pastebin.ca/490775 |
| 07:26.44 | Mercestes | there |
| 07:27.53 | creativx | creampie |
| 07:28.25 | Mercestes | *squeals* Fixed |
| 07:32.46 | Mercestes | had to reboot the fax server. it was out of whack for some reason |
| 07:33.35 | nitrus^ | how do i lower the number or running asterisk processes? |
| 07:33.45 | JT | so what fixed it? |
| 07:33.56 | Mercestes | rebooting the fax server |
| 07:34.03 | JT | i see.... |
| 07:34.05 | JT | weird as |
| 07:34.06 | creativx | those damn fax servers. |
| 07:34.07 | Mercestes | and fixing sangoma's scripted bad configs |
| 07:34.11 | Mercestes | indeed |
| 07:34.20 | JT | is it taking timing from the fax server? |
| 07:34.50 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
| 07:36.58 | CrazyTux | What exactly do the /var/spool/asterisk/voicemail/MAILBOX/INBOX/*.txt files use for, is it strictly informational, or do they need to be there? I.e. are they a file that I can delete, and nothing will change? |
| 07:37.18 | creativx | they contain the timestamps |
| 07:37.27 | creativx | dont they |
| 07:37.33 | CrazyTux | creativx, does asterisk use them? |
| 07:37.45 | CrazyTux | in other words I'm wondering if * itself uses those files for some reason |
| 07:38.10 | creativx | <PROTECTED> |
| 07:38.27 | creativx | yes asterisk uses them |
| 07:38.32 | creativx | via the voicemailmain app |
| 07:38.36 | CrazyTux | creativx, do you know what specifically it uses it for? |
| 07:40.05 | creativx | yes; reading the timestamp when the message was stored |
| 07:40.12 | creativx | and spewing this back out to the user |
| 07:40.39 | CrazyTux | creativx, does asterisk have the ability to store those .txt files in a db? |
| 07:41.50 | creativx | not that I know of |
| 07:43.12 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 07:44.24 | CrazyTux | creativx, I would think there would be a method to store that in a database... |
| 07:44.45 | mightnare | there's doc/voicemail_odbc_postgresql.txt |
| 07:45.00 | creativx | CrazyTux; http://www.voip-info.org/wiki-Asterisk+voicemail+database |
| 07:45.28 | creativx | most things are possible with this beast pbx |
| 07:45.55 | Mercestes | CrazyTux, they are requried. |
| 07:46.23 | CrazyTux | Mercestes, well, how do I go about storing it in a database? |
| 07:46.32 | Mercestes | CrazyTux, all the info in those text files coudl be saved in a DB. I believe the file format is fairly standard |
| 07:46.37 | CrazyTux | Mercestes, I have mailboxes in the database, just need those .txt files now |
| 07:46.44 | Mercestes | CrazyTux, patch the source. :) |
| 07:47.17 | CrazyTux | Mercestes, it is in ini format... |
| 07:47.24 | CrazyTux | Mercestes, I could do a few different things, just wondering if it's already been done. |
| 07:47.38 | Mercestes | I don't think so, would be a nice addition tho |
| 07:47.59 | mightnare | why don't you checkout doc/voicemail_odbc_postgresql.txt first from 1.4.4? |
| 07:52.32 | Mercestes | goodnight lady and gentlemen |
| 07:52.38 | Mercestes | my thanks. :) |
| 07:52.39 | CrazyTux | Mercestes, goodnight |
| 07:54.51 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
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| 08:05.06 | reber | hi all. What does mean : Auto fallthrough, channel 'SIP/10001-0817a5b8' status is 'UNKNOWN' |
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| 08:37.00 | DarKnesS_WolF | guys can i add another time value in teh CDR record of mysql ? |
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| 08:57.24 | snuffy22 | DarKnesS_WolF, the easiest way to add another value is to put it in the userfield |
| 08:57.27 | [hC] | what the heck is going on... when i execute Busy() in my dial plan when taking a call over IAX, as soon as busy() is called the call hangs up instead of waiting for the caller to hang up |
| 08:57.28 | [hC] | and since the channel is closed the busy tones never play |
| 09:01.03 | *** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net) |
| 09:02.39 | DarKnesS_WolF | snuffy22: yes did it |
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| 09:03.19 | kombi | digium now has a TE120P, voipsupply.com still carries TE110P for $100 less, does it make sense to go for 120? |
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| 09:15.26 | JT | kombi: not really |
| 09:15.29 | JT | but voipsupply is shit |
| 09:19.40 | *** join/#asterisk DrukenLPY (n=jdumais@CPE001346f4961f-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 09:22.46 | kombi | JT: like they don't deliver kind of shit? |
| 09:24.56 | kombi | JT: I wonder which card to buy anyway, $500 is a somewhat hefty price tag |
| 09:26.25 | kombi | who should I buy from (being in the New York metro area? |
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| 09:32.21 | Daejeo1 | does trixbox offer SIP/IAX SERVICE PROVIDER wizard? |
| 09:35.10 | *** join/#asterisk cayorde (n=flexable@host184-111-dynamic.17-87-r.retail.telecomitalia.it) |
| 09:46.45 | JT | kombi: yeah and they screw up orders |
| 09:46.50 | JT | Daejeo1: wrong channel |
| 09:49.10 | Daejeo1 | JT: can you give some basic idea that how I can connect my asterisk box with a sip termination provider? |
| 09:49.34 | Daejeo1 | I just want some basic idea |
| 09:50.04 | Daejeo1 | I want to do it manually |
| 10:01.33 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:01.36 | *** join/#asterisk ecthelion_italy (n=ecthelio@host111-132-dynamic.9-87-r.retail.telecomitalia.it) |
| 10:03.13 | *** join/#asterisk ta^3 (n=tacvbo@189.146.195.139) |
| 10:07.48 | ecthelion_italy | hi |
| 10:11.38 | creativx | ugh |
| 10:12.16 | creativx | what would be best codec config: iax2 to sip phones, bandwidth no problem... ulaw? |
| 10:12.16 | kippi617 | hey |
| 10:12.37 | kippi617 | has anyone here used phone genie? |
| 10:13.14 | *** join/#asterisk tr2x (n=alvar@80-218-185-55.dclient.hispeed.ch) |
| 10:19.27 | *** join/#asterisk VijayG (i=VijayG@202.131.145.247) |
| 10:19.32 | VijayG | Hello everyone |
| 10:19.45 | VijayG | whats the command, if i need to hangup all going calls in asterisk |
| 10:19.56 | VijayG | i do not want to restart asterisk; |
| 10:20.03 | VijayG | just the calls should get disconnected |
| 10:21.11 | VijayG | is there any command like that |
| 10:22.21 | kippi617 | what does this error Unable to re-open DSP |
| 10:27.53 | *** join/#asterisk funkmaster (n=funky1@vhe-364501.sshn.net) |
| 10:31.45 | *** join/#asterisk Mavvie (n=edwin@ppp5-159.lns4.syd7.internode.on.net) |
| 10:32.19 | *** join/#asterisk tenzind (n=tenzind@202.144.144.77) |
| 10:34.08 | *** join/#asterisk Rienzilla (i=rien@sinas.rename-it.nl) |
| 10:34.22 | Rienzilla | wow :) |
| 10:36.29 | Rienzilla | hey guys, I am trying to use my isdn card in nt mode under linux, but I'm having a horrible time getting the drivers to work. I tried both a zaptel driver, but that won't compile against my kernel apparently; and misdn 1.1.3, which compiles, but crashes the system seconds after I load the module. Any ideas what I should do? (hardware is a simple hfc pci card in a soekris embedded pc) |
| 10:42.29 | *** join/#asterisk ccesario (n=ccesario@201-0-124-86.dsl.telesp.net.br) |
| 10:48.37 | *** join/#asterisk zapp-branigan (n=zapp-bra@81.202.137.86.dyn.user.ono.com) |
| 10:49.24 | *** join/#asterisk shinao1 (n=shinao1@196.3.63.252) |
| 10:49.38 | zapp-branigan | hi i have a notice : refected connect from xxxxxx request 1@default does not exist |
| 10:49.52 | zapp-branigan | the number who is try to call is 1 |
| 10:49.53 | zapp-branigan | ? |
| 10:50.56 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 10:55.13 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:01.06 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 11:01.22 | *** join/#asterisk zdrulio (n=krlozano@82.119.72.130) |
| 11:01.25 | zdrulio | hello all |
| 11:02.00 | zdrulio | IAX2 work with TCP or UDP ? |
| 11:02.07 | backblue | can i do iax trunk from 1.4 to 1.2? |
| 11:02.14 | backblue | zdrulio: udp. |
| 11:02.42 | zdrulio | backblue: are u sure ? |
| 11:04.01 | backblue | zdrulio: y |
| 11:04.12 | zdrulio | thx |
| 11:05.15 | *** part/#asterisk zdrulio (n=krlozano@82.119.72.130) |
| 11:06.24 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:25.46 | *** join/#asterisk shinao1 (n=shinao1@196.3.63.252) |
| 11:37.18 | *** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 11:37.27 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 11:43.04 | *** join/#asterisk friedrich| (n=friedric@e177242207.adsl.alicedsl.de) |
| 11:47.24 | JT | Daejeo1: sorry, was away, take a look at the book |
| 11:47.24 | JT | ~thebook |
| 11:47.34 | jbot | i guess thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 11:48.51 | *** join/#asterisk penguinFunk (n=penguin@87.224.86.46) |
| 11:49.57 | Daejeo1 | thanx mate |
| 11:51.39 | *** join/#asterisk Polis_ttt (n=your@194-237-172-225-no48.business.telia.com) |
| 11:54.24 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 12:06.18 | tzanger | wow that was a cool storm |
| 12:06.26 | tzanger | anyone else in ontario/quebec here? |
| 12:06.34 | funkmaster | is there a command, which displays me the received calls during the last hour or day, which i can use from the CLI asterisk console? |
| 12:07.26 | tzanger | funkmaster: nope, put your CDR into a DB and pull it out htat way |
| 12:09.21 | Mavvie | is there a way to find out which room number a meetme conference is in? |
| 12:09.33 | *** join/#asterisk Hmmhesays (n=Neg@24-119-176-74.cpe.cableone.net) |
| 12:10.12 | *** join/#asterisk SoftIce (n=psmith@dsl-242-110-160.telkomadsl.co.za) |
| 12:10.18 | SoftIce | hi, how would I do this with sip |
| 12:10.19 | SoftIce | exten => 1680,1,dial(iax2/9043211:9043211@10.0.0.2) |
| 12:10.37 | SoftIce | just replace iax2 with sip or is the structing different? |
| 12:10.49 | Mavvie | SoftIce: nah, it all is more or less the same |
| 12:11.02 | Mavvie | technology/number@host |
| 12:11.38 | tzanger | SELECT calldate,src,dst,disposition FROM cdr WHERE calldate >= 'now'::timestamp - interval '1 hour' ORDER BY calldate DESC; |
| 12:11.42 | SoftIce | strange im getting no route to host if i do it that way now |
| 12:11.44 | tzanger | funkmaster: ^^ |
| 12:12.26 | SoftIce | sorry I did have this at the end |
| 12:12.40 | SoftIce | exten => 1680,1,dial(iax2/9043211:9043211@10.0.0.2/s@sdcs) |
| 12:13.02 | SoftIce | and now i'm getting when I dial out No such host: 10.0.0.2/s@sdcs |
| 12:13.15 | SoftIce | in asterisk CLI, cant you use /s@context with SIP ? |
| 12:14.45 | *** join/#asterisk jkiff (n=jkiffmey@unaffiliated/vorondil) |
| 12:20.14 | *** join/#asterisk Rienzilla (i=rien@sinas.rename-it.nl) |
| 12:20.36 | Rienzilla | anyone here able to help me with chan_misdn? |
| 12:21.25 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:21.59 | funkmaster | tzanger: thx dude :) |
| 12:22.24 | tzanger | funkmaster: adjust to taste; I generally format my queries a little more than that, but that's the basics |
| 12:22.58 | *** join/#asterisk jkiff (n=jkiffmey@unaffiliated/vorondil) |
| 12:24.00 | *** join/#asterisk ToyMan (n=Stuart@72.168.167.241) |
| 12:25.57 | SoftIce | hmm, do I need a register string first before I can dial out sip ? |
| 12:31.03 | iCEBrkr | Always register! |
| 12:31.09 | iCEBrkr | $$$ pay up sucka! |
| 12:31.14 | funkmaster | tzanger: allright thx a bunch that was what i needed :) |
| 12:31.46 | *** join/#asterisk DrukenLPY (n=jdumais@CPE001346f4961f-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 12:32.00 | *** join/#asterisk axisys (n=axisys@72-255-17-136.client.stsn.net) |
| 12:32.06 | Rienzilla | piece of crap :x |
| 12:32.36 | iCEBrkr | Hey! I'm on channel, you know, you don't have to talk about me like that :) |
| 12:35.12 | [TK]D-Fender | SoftIce: No. Technically Registrations has nothing to do with placing calls or the ability to receive calls |
| 12:35.25 | Rienzilla | hehe |
| 12:36.13 | iCEBrkr | [TK]D-Fender: Are you sure? :P |
| 12:36.40 | iCEBrkr | I suppose 'technically' you could make a call without being authenticated. |
| 12:36.51 | [TK]D-Fender | iCEBrkr: Yes. One of my clients is with a provider that doesn't SUPPORT Register even. Needs a fixed IP or Domain :) |
| 12:36.59 | iCEBrkr | Ahhh |
| 12:37.03 | [TK]D-Fender | iCEBrkr: Registering is NOT auth! |
| 12:37.07 | Hmmhesays | you can do proxy auth |
| 12:37.08 | iCEBrkr | Well that's just a different form of 'authentication' I guess |
| 12:37.22 | Hmmhesays | registration is more for location |
| 12:37.26 | Hmmhesays | than auth |
| 12:37.29 | [TK]D-Fender | iCEBrkr: Register is jsut a way (using auth itself) to update the provider as to your *IP*. Thats all |
| 12:37.38 | iCEBrkr | Hmmhesays: Hmm, yeah... |
| 12:37.55 | iCEBrkr | "Hey! Send calls over here!" |
| 12:38.03 | Hmmhesays | no bitches |
| 12:38.10 | jeremy_g | :) |
| 12:38.13 | iCEBrkr | yes bitches! |
| 12:38.19 | iCEBrkr | lots of bitches! |
| 12:38.20 | [TK]D-Fender | and hos! |
| 12:38.24 | [TK]D-Fender | :D |
| 12:38.26 | iCEBrkr | oh yeah |
| 12:38.32 | jeremy_g | what is this nonce thing required in the register. is it a must? |
| 12:38.53 | jeremy_g | i am writting a small server that would register + auth a client |
| 12:39.04 | Rienzilla | god |
| 12:40.09 | [TK]D-Fender | Rienzilla: .... Yes my child? ;) |
| 12:40.16 | iCEBrkr | [TK]D-Fender: I wasn't going to say it |
| 12:40.33 | Rienzilla | hehe |
| 12:40.50 | tzanger | my father my father, I've come to confess, I just left a girl in a terrible mess... |
| 12:41.19 | Hmmhesays | whoa |
| 12:42.07 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 12:42.14 | *** join/#asterisk neuwald (n=neuwald@200.199.198.61) |
| 12:43.23 | *** join/#asterisk Ebola (n=Ebola@host86-137-4-175.range86-137.btcentralplus.com) |
| 12:43.33 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 12:43.51 | drako | hey |
| 12:43.53 | drako | HFC-MULTI: Card 'HFC-4S Digium Card' found, but not given by module's options, ignoring... |
| 12:44.36 | drako | :~# misdnportinfo |
| 12:44.37 | drako | Found no card. Please be sure to load card drivers. |
| 12:46.06 | iCEBrkr | drako: kick it |
| 12:46.07 | *** join/#asterisk DrukenHME (n=jdumais@CPE001346f4961f-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 12:47.35 | *** join/#asterisk MORA_ (i=MORA@0x50a58735.hsnxx1.adsl-dhcp.tele.dk) |
| 12:48.23 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-72-65-27-9.bflony.east.verizon.net) |
| 12:48.45 | MORA_ | How do you change +xx to 00xx in extensions.conf, and how do you catch the +xx in the exten part ? |
| 12:48.56 | *** join/#asterisk cnbrk (n=canberk@teknopet.com) |
| 12:48.59 | cnbrk | hello |
| 12:49.16 | cnbrk | i have a question, i have 4 pstn lines coming from local telecom operator |
| 12:49.22 | cnbrk | and i want to switch to voip |
| 12:49.40 | cnbrk | how can i make my asterisk server ring whenever those pstn lines ring |
| 12:49.54 | MindTheGap | i using reltime ldap lookup for sip peers but asterisk complains of not finding records of a static sip entry on sip.conf. shouldn´t it lookup realtime only entries not found on sip.conf? |
| 12:50.12 | creativx | uh... how come I get "process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible." even if on my IP10s swissvoice phones i have set "Silence suppression: off" ? |
| 12:50.44 | [TK]D-Fender | cnbrk: your PSTN lines have nothing to do with VoIP here. If you want to take those lines into your * server you'd use an analog interface card or gateway |
| 12:50.50 | cnbrk | YES |
| 12:50.56 | cnbrk | this is the answer i am looking for |
| 12:51.01 | cnbrk | where can i find these gateways |
| 12:51.12 | cnbrk | how should i search for it, what is its name |
| 12:51.13 | [TK]D-Fender | cnbrk: ex. Digium TDM400P, Sangoma A200(d), and their higher density versions |
| 12:51.23 | cnbrk | i see, are there any external ones? |
| 12:51.35 | [TK]D-Fender | cnbrk: The above are PCI cards you'd install into your sever to plug those lines in. |
| 12:52.10 | cnbrk | so are there any external ones that i can put to a remote location which will bring all the traffic to our local asterisk server |
| 12:52.16 | [TK]D-Fender | cnbrk: Yes, there are many kinds of external gateways. ones that take the line in and talk SIP direct to *. Or channel banks attached to T1 interface cards, etc. |
| 12:52.51 | matsk | http://voip-info.org/wiki/index.php?page=VOIP+Gateways |
| 12:52.54 | [TK]D-Fender | cnbrk: For a pure remote external gateway look at the AudioCodes MP-XXX series gateways, or Mediatrix |
| 12:53.01 | cnbrk | what i want to do is there are four pstn lines far from our asterisk server, i want a device that will bring all those external lines to my asterisk server over internet |
| 12:53.12 | cnbrk | an audiocodes mp-xxx devices to that? |
| 12:53.20 | cnbrk | can* |
| 12:54.33 | iCEBrkr | I use my spiffy little Intel Voicemodem for analog! HIGH TECH! |
| 12:54.44 | cnbrk | which one am i looking for |
| 12:54.50 | drako | iCEBrkr, howso |
| 12:54.50 | cnbrk | fxo ones or fxs ones or both |
| 12:55.02 | iCEBrkr | drako: it's a hack, it's not all that great honestly. |
| 12:55.53 | iCEBrkr | You'll be there a long time. |
| 12:56.11 | [TK]D-Fender | cnbrk: You the AudioCodes is the kind of product you're looking for, and you'd be looking for the FXO variety |
| 12:56.12 | DrukenHME | most likely :) |
| 12:56.21 | cnbrk | OKAY |
| 12:56.23 | [TK]D-Fender | ~fxsfxo |
| 12:56.37 | jbot | extra, extra, read all about it, fxsfxo is an FXO port expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 12:57.00 | cnbrk | 4x FXO one will bring four pstn lines to my asterisk server |
| 12:57.05 | cnbrk | will it do the vice versa? |
| 12:57.30 | *** join/#asterisk coppice (n=chatzill@94.143.17.210.dyn.pacific.net.hk) |
| 12:57.39 | *** join/#asterisk jkiff (n=jkiffmey@unaffiliated/vorondil) |
| 12:58.31 | DrukenHME | you mean will it allow asterisk to dial out of those 4 lines? |
| 12:58.35 | cnbrk | yes |
| 12:58.42 | DrukenHME | there's your answer :) |
| 12:58.55 | iCEBrkr | It's a 2-way street ya know! |
| 12:59.03 | [TK]D-Fender | cnbrk: Actually,, another choice would be the Linksys SPA-400. Pretty inexpensive. |
| 12:59.03 | cnbrk | thats nice |
| 12:59.11 | DrukenHME | asterisk swings BOTH ways baby... |
| 12:59.13 | cnbrk | yes i need the cheapest one |
| 12:59.41 | [TK]D-Fender | cnbrk: FXO ports are for using PSTN lines in/out. FXS ports are for PHONES (in/out) |
| 13:00.04 | cnbrk | ok i see |
| 13:00.17 | cnbrk | for linksys, it says it's made only for linksys voice system |
| 13:00.23 | cnbrk | will it work with asterisk too? |
| 13:00.29 | [TK]D-Fender | cnbrk: http://www.voipsupply.com/index.php?cPath=94_286_120 |
| 13:00.51 | [TK]D-Fender | cnbrk: I've seen a few articles mentioning its use with *, so I'd presume "yes". |
| 13:01.11 | [TK]D-Fender | cnbrk: and following that link, IGNORE GrandStream. |
| 13:01.13 | [TK]D-Fender | ~gs |
| 13:01.25 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 13:01.59 | cnbrk | what does a fxs gateway do? |
| 13:02.19 | tzanger | it's a gateway... TO THE UNKNOWN |
| 13:02.29 | cnbrk | what? :) |
| 13:02.56 | tzanger | it allows you to ocnnect FXS devices (telephones, faxes, etc.) |
| 13:02.58 | cnbrk | is fxs gateway something similar to an analog telephone adapter |
| 13:03.13 | cnbrk | yes it is |
| 13:03.16 | cnbrk | great now i got it |
| 13:03.41 | cnbrk | and how long should an asterisk compilation take, mine is compiling for 3 hours... |
| 13:04.04 | tzanger | wtf, are you on a 386 or something? |
| 13:04.05 | [TK]D-Fender | cnbrk: Depends what you're compiling on. |
| 13:04.32 | Hmmhesays | well that was pretty much awesome |
| 13:04.34 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 13:04.40 | [TK]D-Fender | cnbrk: for me, about 3-5 minutes on a 2 ghz machine |
| 13:04.58 | cnbrk | freebsd |
| 13:04.59 | Hmmhesays | walker was chasing this people in his truck and the car just happened to jump through a plate glass window on a trailor |
| 13:05.05 | cnbrk | freshly installed freebsd |
| 13:05.23 | [TK]D-Fender | cnbrk: what class of COMPUTER? |
| 13:05.28 | cnbrk | it is compiling from ports collection and installing lots of additional stuff |
| 13:05.36 | cnbrk | 3.2ghz 2gb |
| 13:06.04 | Corydon76-home | You'd need to check with the ports maintainer, then |
| 13:06.05 | drako | \ |
| 13:06.16 | drako | iCEBrkr, i was talking about the card |
| 13:06.24 | cnbrk | maybe |
| 13:06.29 | drako | anyone else? |
| 13:06.32 | drako | HFC-MULTI: Card 'HFC-4S Digium Card' found, but not given by module's options, ignoring... |
| 13:06.35 | cnbrk | is there a cheaper solution than Linksys SPA-400 |
| 13:06.37 | drako | Found no card. Please be sure to load card drivers. |
| 13:08.12 | iCEBrkr | drako: Yea, it's an old-school hack for using an Intel voice modem for an analog interface |
| 13:08.33 | [TK]D-Fender | cnbrk: Nothing worth considering. The SPA is likely not the best in quality as it is, but likely passable |
| 13:08.43 | drako | iCEBrkr, oh well, this is a B410P Digium, for RDSI. |
| 13:08.43 | JT | cnbrk: spa-400.... that's a pbx |
| 13:09.08 | cnbrk | what? |
| 13:09.28 | JT | iirc the spa-400 has an integrated pbx |
| 13:10.36 | [TK]D-Fender | JT, no, not a PBX. A VM app server apparently. Not relevent |
| 13:10.40 | MindTheGap | i using reltime ldap lookup for sip peers but asterisk complains of not finding records of a static sip entry on sip.conf. shouldn´t it lookup realtime only entries not found on sip.conf? |
| 13:10.45 | Qwell | http://www.zipzoomfly.com/jsp/ProductDetail.jsp?ProductCode=85023-4&prodlist=dc |
| 13:10.53 | Qwell | 2gb ram, $76.90 |
| 13:11.18 | JT | [TK]D-Fender: you know it runs asterisk? |
| 13:13.29 | coppice | that's overpriced |
| 13:13.43 | *** join/#asterisk ToyMan (n=Stuart@72.168.167.241) |
| 13:13.45 | coppice | unless its DDR, when it will be twice that |
| 13:13.50 | Qwell | coppice: ddr2 |
| 13:13.58 | Qwell | pc-5300 |
| 13:14.11 | coppice | the price difference for DDR is getting really silly |
| 13:14.12 | Katty | morning lovables! |
| 13:14.37 | [TK]D-Fender | JT : so I've heard |
| 13:16.54 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:20.18 | *** join/#asterisk Fieldy (i=iyBnbowf@gentoo/contributor/Fieldy) |
| 13:22.00 | *** join/#asterisk Mordor (n=Nosfe@157-233-246-201.adsl.terra.cl) |
| 13:26.28 | cnbrk | i've found Linksys SPA-400 for 234 usd |
| 13:26.39 | *** join/#asterisk ToyMan (n=Stuart@72.168.167.241) |
| 13:26.43 | cnbrk | sounds great |
| 13:26.58 | [TK]D-Fender | Katty: Mew. |
| 13:27.03 | Hmmhesays | holy Katty |
| 13:27.13 | Katty | [TK]D-Fender: one of the members of the board liked the idea of you being a consultant, btw. |
| 13:27.14 | [TK]D-Fender | Katty: Going to try and find a few minutes today to kick your telco in the nads :) |
| 13:28.18 | *** join/#asterisk Avero (n=Avero@216.186.253.120) |
| 13:28.42 | Avero | Are there any known issues with doing a "moh reload" on the CLI that would cause Asterisk to crash? |
| 13:29.00 | Katty | [TK]D-Fender: don't think there's time today. |
| 13:29.05 | Katty | [TK]D-Fender: but maybe |
| 13:29.16 | Katty | Hmmhesays: hey you (= |
| 13:30.21 | [TK]D-Fender | Katty: Well its minutes of MY day, not yours.... so we'll see |
| 13:30.26 | Hmmhesays | How are you? its been forever since I talked to you |
| 13:31.00 | iCEBrkr | [TK]D-Fender as a consultant?! RUN AWAY!! RUN AWAY!! |
| 13:31.02 | *** join/#asterisk bkw_ (i=brian@adsl-70-143-39-207.dsl.tul2ok.sbcglobal.net) |
| 13:31.19 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 13:31.25 | [TK]D-Fender | :O |
| 13:31.32 | iCEBrkr | :P |
| 13:31.34 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:31.34 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:31.40 | [TK]D-Fender | :D |
| 13:32.02 | MORA_ | Anyone knows how do you change +xx to 00xx in extensions.conf, and how to catch the +xx in the exten part ? |
| 13:32.12 | iCEBrkr | oh I really need to get Damin's billing report thing finished. |
| 13:32.23 | Katty | [TK]D-Fender: yeah, but you're not going to call the telco about us unless the boss is on the phone with us (= |
| 13:32.33 | [TK]D-Fender | MORA_: exten => _00xx |
| 13:32.43 | MORA_ | yes, but for +xx |
| 13:32.48 | MORA_ | to be able to change it |
| 13:32.52 | *** join/#asterisk truz_`24 (n=truz_`24@74-129-166-232.dhcp.insightbb.com) |
| 13:32.54 | [TK]D-Fender | Katty: I already told you thats how I was going to play it yesterday :) |
| 13:32.58 | Katty | k |
| 13:33.05 | *** join/#asterisk Strom_M (i=strom@nat/digium/x-7431884083937583) |
| 13:33.13 | Hmmhesays | exten => _+XX,1,Set(foo=00${EXTEN:1}) |
| 13:33.17 | [TK]D-Fender | MORA_: Whats missing from what I just provided? |
| 13:33.31 | DrukenHME | [TK]D-Fender: what is katty's telco NOT doing? :) |
| 13:33.50 | [TK]D-Fender | DrukenHME: They are trying to rob her blind for PRI vs Voice T1. |
| 13:34.04 | MORA_ | I tried that, exten => _+XXXXX,1,NoOp |
| 13:34.08 | Qwell | Katty: [TK]D-Fender: I want my finders fee :P |
| 13:34.10 | DrukenHME | uhmm... aren't they the same damn thing?? :) |
| 13:34.14 | iCEBrkr | Umm. |
| 13:34.28 | iCEBrkr | MORA_: last time I looked, I couldn't dial a + sign |
| 13:34.32 | Qwell | I didn't really find anything, but that's okay |
| 13:34.46 | MORA_ | right, thats why I need to change it before sending it out |
| 13:34.52 | MORA_ | since the users demand to be able to use it |
| 13:34.53 | Qwell | I'm sure you could convince them ;) |
| 13:35.05 | Avero | MORA_: In that case, unless they send + and five digits down then that's not going to match. |
| 13:35.06 | [TK]D-Fender | MORA_: Perhaps you should rethink your wording of CHANGING what was dialed to changing what you DO with it. Please pastebin exactly what you're doing now and describe how you want it changed. |
| 13:35.45 | [TK]D-Fender | DrukenHME: No, Voice T1, is sometimes not jsut CAS @ 24 channels. |
| 13:35.59 | [TK]D-Fender | DrukenHME: But should be counted as effectively the same. |
| 13:37.17 | CrazyTux | Hey guys, I'm forwarding a sip request to Asterisk, and I basically want to playback a sound file and then exit the call, I have exten => some_exten,1,Playback(some_audio) but dosent seem to work right |
| 13:37.17 | JT | DrukenHME: PRI != CAS T1 |
| 13:38.11 | [TK]D-Fender | CrazyTux: Pastebin the output of the failed attempt at verbose 10 |
| 13:38.35 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 13:38.35 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:41.40 | CrazyTux | [TK]D-Fender, one second :) |
| 13:42.28 | *** join/#asterisk MrWup (i=root@80-46-38-124.static.dsl.as9105.com) |
| 13:42.39 | MrWup | does anyone know if BT POTS lines have answer supervision enabled on them? |
| 13:42.58 | Katty | eep |
| 13:43.22 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.140.114) |
| 13:43.49 | MrWup | im having problems when making outgoing calls to the PSTN via my FXO card. asterisk does not detect when the remote party picks up and the dial command times out with NOANSWER while im talking to the remote user |
| 13:44.24 | [TK]D-Fender | Katty: Go caffeinate! |
| 13:44.24 | MrWup | hangup detection etc are all fine |
| 13:44.25 | MrWup | ive tried answeronpolarityswitch=yes but this doesnt seem to make a difference |
| 13:44.25 | CrazyTux | [TK]D-Fender, is there an easy way to send stdout to a file? |
| 13:44.31 | MrWup | any ideas greatly appreciated |
| 13:44.41 | [TK]D-Fender | CrazyTux: jsut cut & paste |
| 13:44.45 | MrWup | have been stuck on this problem for weeks now |
| 13:45.26 | MORA_ | Avero right, . was missing, works now, thanks :) |
| 13:48.14 | CrazyTux | [TK]D-Fender, http://pastebin.ca/491154 |
| 13:48.26 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:48.40 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 13:49.31 | *** join/#asterisk Strom_M (i=strom@nat/digium/x-ad15bd3e407e4592) |
| 13:49.33 | [TK]D-Fender | CrazyTux: Looks like a NAT problem. |
| 13:50.44 | drako | Damn! |
| 13:50.48 | drako | anyone please... |
| 13:51.00 | drako | HFC-MULTI: Card 'HFC-4S Digium Card' found, but not given by module's options, ignoring... |
| 13:51.36 | CrazyTux | [TK]D-Fender, I get a 500 server error back, im looking at sipcap |
| 13:54.58 | *** join/#asterisk johann8384 (n=johann83@206.80.75.240) |
| 14:02.39 | *** join/#asterisk postel_ (n=jp@wikimedia/Postel) |
| 14:03.07 | *** join/#asterisk Zefk (n=Zefk@wsc-fo.b.astral.ro) |
| 14:03.58 | *** join/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu) |
| 14:04.49 | Hmmhesays | nat is fun |
| 14:04.50 | Hmmhesays | crazy fun |
| 14:04.54 | Zefk | Hi. I'm using asterisk 1.4.2 with Avaya 4610SW phones. The phones' lamp remains off all the time. Any hints ? Thx. |
| 14:05.41 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:06.02 | *** join/#asterisk coolbeans (n=null@adsl-074-247-038-249.sip.bhm.bellsouth.net) |
| 14:06.40 | coolbeans | Hi all. If I have 80 extensions and I want to have a wildcard extensions.conf entry to dial them (4-digit), how would I do that? exten => 8... ?? |
| 14:07.09 | MORA_ | 8XXX |
| 14:07.28 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 14:08.13 | coolbeans | Oh! Duh. Thanks! :) |
| 14:08.59 | *** join/#asterisk ghenry (n=ghenry@212.159.59.85) |
| 14:09.23 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:11.06 | [TK]D-Fender | coolbeans: I advise against that if a number in the match is not valid. Bad things will happen... |
| 14:14.03 | truz_`24 | Where can i get the xp100 module in ubuntu fiesty? |
| 14:14.21 | truz_`24 | did apt-get install zaptel and asterisk but modprobe zaptel fails |
| 14:15.44 | cnbrk | what does jitter buffer do? |
| 14:16.18 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
| 14:17.47 | penguinFunk | cnbrk: tries to control jitter by buffering data and sending it at a controlled amount |
| 14:18.22 | coolbeans | TK: It's actually 7XXX and I have nothing in the 7's in the dial plan. How do you typically handle large numbers of internal extensions? (And thanks for any advice/help) |
| 14:18.48 | Zeeek | how can I prevent asterisk from starting mp3 processes? |
| 14:18.57 | coolbeans | BTW: My exten-> _7XXX,s,macro.. is a macro |
| 14:19.06 | Qwell | Zeeek: don't use mpg123 |
| 14:19.09 | penguinFunk | so like instead of sending [data] + delay + [data] (no jitter buffer) you get [data] + [data] since delays arent buffered |
| 14:19.23 | Zeeek | I don't. I just started asterisk and for some reason it starts them |
| 14:19.24 | penguinFunk | send the buffer contents when it is all ready |
| 14:19.40 | Zeeek | musiconhold module isn't loaded |
| 14:19.48 | Qwell | eh? then asterisk isn't loading it |
| 14:19.51 | coolbeans | err.. s = 1 exten=> 7XXX,1,macro |
| 14:19.56 | Zeeek | but I made that change later. Is that where it would start them? |
| 14:20.09 | *** join/#asterisk ccesario (n=ccesario@201-0-124-86.dsl.telesp.net.br) |
| 14:20.12 | Zeeek | ok, this was a remnant of an eariler stupidity |
| 14:21.32 | *** join/#asterisk unixlike (n=spid3r@31.67.modemcable.oricom.ca) |
| 14:23.02 | errr | anyone happen to know of a way to get the output of sip show peer <peer> in php with out having to use an exec or system command?? |
| 14:24.11 | killfill | wired.. i set language=es, and the girl is talking the digis in english.. files are there tho |
| 14:24.17 | killfill | how can i debug this?.. |
| 14:29.02 | Rienzilla | hmm |
| 14:29.48 | coolbeans | On asterisk $ARGx variables, do they start with "0" or "1" when passing? |
| 14:30.08 | SuPrSluG | is there a command line speed test app |
| 14:31.03 | Rienzilla | hmm |
| 14:31.12 | *** join/#asterisk brookshire (i=mbrooks@hijacked.us) |
| 14:31.16 | Rienzilla | what is wrong in misdn when it tells me my port has a l1 link up, but the l2 link is down? |
| 14:31.53 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 14:31.54 | Rienzilla | did I miswire something? or is it a software issue |
| 14:33.55 | Rienzilla | or does it just mean im not using it :) |
| 14:37.54 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:38.59 | [TK]D-Fender | coolbeans: Your laste few lines look like gibberish. pastebin your dialplan. |
| 14:40.06 | *** join/#asterisk Chris-NB (n=chris@home.fuerstaller.com) |
| 14:40.29 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-b7502393bc208dc3) |
| 14:40.29 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 14:41.11 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 14:41.41 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 14:42.10 | irule | hi there, what is this? Asterisk died with code 1. |
| 14:42.15 | *** join/#asterisk galeras (n=root@200.31.204.42) |
| 14:42.38 | mosty | irule: that is a script noticing that asterisk died |
| 14:43.19 | irule | thanks mosty, that was enlightening! lol |
| 14:43.52 | mosty | irule: what do you expect? it could be anything, you will have to check your logs |
| 14:44.43 | *** join/#asterisk Alan_Hicks (n=alan@208.62.162.112) |
| 14:44.44 | cnbrk | so with jitter buffer, does it mean i'll get less latency? |
| 14:44.45 | *** join/#asterisk Vec2 (n=Vec@dsl-241-220-134.telkomadsl.co.za) |
| 14:45.32 | mosty | cnbrk: no, jitterbuffer will increase latency |
| 14:46.01 | cnbrk | oh ok |
| 14:46.10 | cnbrk | what can i do to get the least latency |
| 14:46.57 | Zeeek | hi |
| 14:47.02 | mosty | use as short a path as possible, with as fast a connection as possible, with as little traffic on it as possible |
| 14:47.24 | cnbrk | i see |
| 14:47.41 | cnbrk | and also, what is the main difference between asterisk 1.2 and 1.4 |
| 14:47.50 | cnbrk | which one is more stable and better |
| 14:47.58 | mosty | cnbrk: 1.4 is buggy |
| 14:48.08 | mosty | 1.2 is less buggy |
| 14:48.25 | cnbrk | oh shit then |
| 14:48.36 | cnbrk | i deinstalled 1.2 and installing 1.4 now |
| 14:49.46 | Zeeek | heh |
| 14:50.25 | *** join/#asterisk BSD_tech (n=BSDTech@adsl-69-230-174-37.dsl.irvnca.pacbell.net) |
| 14:51.32 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 14:51.52 | *** join/#asterisk bbryant (i=brett@nat/digium/x-991ab7920b23b5ec) |
| 14:52.13 | cnbrk | by the way, thank you guys for all the answers to my newbie questions |
| 14:52.34 | *** part/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu) |
| 14:52.38 | Strom_M | now here's an interesting problem: how would you /induce/ echo into an analog circuit to test the echo canceller? |
| 14:52.54 | Qwell[] | haha |
| 14:52.58 | Qwell[] | it's funny you should ask that |
| 14:53.11 | Qwell[] | there is a new "echo can" in zaptel...jpah |
| 14:53.16 | Qwell[] | Jason Parker's Audio Hoser :P |
| 14:53.20 | Strom_M | hahahaha |
| 14:53.22 | Qwell[] | (seriously) |
| 14:53.22 | Strom_M | is that in trunk? |
| 14:53.28 | Qwell[] | and 1.4, and I think 1.2 |
| 14:53.30 | nomadsoul | lol |
| 14:53.35 | Strom_M | heh |
| 14:53.50 | nomadsoul | Strom_M: call the pope, he always do echo when he talk |
| 14:54.35 | nomadsoul | me want to konquer der world! yavol! |
| 14:54.36 | cnbrk | lmao |
| 14:55.08 | MrWup | anyone here using asterisk in the UK? |
| 14:55.19 | cnbrk | which platform is the best for asterisk, freebsd? or others? |
| 14:55.46 | mosty | cnbrk: linux is the most well tested system |
| 14:55.57 | cnbrk | which linux is the best then? |
| 14:56.01 | [TK]D-Fender | cnbrk: Linux clearly |
| 14:56.03 | Qwell[] | cnbrk: whichever you know best |
| 14:56.13 | mosty | cnbrk: whichever you know how to admin best |
| 14:56.22 | cnbrk | i know freebsd :( |
| 14:56.34 | cnbrk | at least i didn't have any problems with * 1.2 and freebsd |
| 14:56.45 | cnbrk | now we'll see if the same applies for 1.4 too |
| 15:01.03 | Qwell[] | Strom_M: actually, jpah probably isn't *quite* what you want.. it actually drops "packets" |
| 15:01.21 | Zeeek | who's the FreeBSD expert here? |
| 15:01.25 | Strom_M | ah yeah |
| 15:01.28 | Strom_M | i want to make it echo |
| 15:01.49 | Strom_M | or, more accurately, my student wants to make it echo so he can play with the echo can |
| 15:02.36 | Qwell[] | heh, I want the phone in page 5 of the latest VON mag |
| 15:03.01 | Strom_M | Qwell[]: join us for lunch today :) |
| 15:03.16 | Qwell[] | where? |
| 15:03.26 | Strom_M | atrium, suite 100 |
| 15:04.19 | [TK]D-Fender | Strom_M: use a soft-phone on one end, and put a guitar reverb unit on the mic. |
| 15:05.34 | [TK]D-Fender | Strom_M: Or delay. |
| 15:06.26 | Alan_Hicks | Am I the only person that sees IAX and immediately think AIX? |
| 15:06.39 | Strom_M | Alan_Hicks: it's a habit you will soon be broken of |
| 15:06.48 | Alan_Hicks | hehehe |
| 15:06.51 | Strom_M | Qwell[]: anyway, lunch is at noon |
| 15:07.09 | Strom_M | bring whoever else you want to bring with |
| 15:07.17 | Strom_M | these guys are full of questions :) |
| 15:08.28 | BSD_tech | in 1.4.x how do you see what codec is in use |
| 15:08.33 | *** join/#asterisk hfb (n=hfb@pool-72-67-156-130.lsanca.dsl-w.verizon.net) |
| 15:10.44 | [TK]D-Fender | BSD_tech: In use by what? |
| 15:10.49 | Zeeek | most vocalists don't care about the instruments |
| 15:11.04 | BSD_tech | I have a call going |
| 15:11.11 | Zeeek | I have a flikr account. Ijn fact I think you access thru yahoo, no? |
| 15:11.19 | BSD_tech | but I dont know if it used my g729 or ulaw |
| 15:11.20 | Zeeek | shit, wrong win again |
| 15:11.29 | [TK]D-Fender | Zeeek: Top reason Americal Idol (and all shows like it) suck. Singers are pretty much a dime-a-dozen. |
| 15:11.35 | Zeeek | heh |
| 15:11.39 | mosty | BSD_tech: sip show channels |
| 15:11.40 | [TK]D-Fender | BSD_tech: Depends on channel type |
| 15:11.51 | Zeeek | BSD_tech I had a question but have to go. I'll ask you later |
| 15:11.53 | [TK]D-Fender | mosty: there you go ASSUMING again :) |
| 15:11.53 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 15:12.34 | BSD_tech | sip |
| 15:12.56 | BSD_tech | but they took the show g729 out of 1.4 it seems |
| 15:13.03 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 15:13.23 | mosty | [TK]D-Fender: that's why i was voted psychic of the year 2009, i will have you know |
| 15:13.39 | rene- | american idol sucks because tv sucks, when tv is open like the internet and linux, our artists should rise on youtube like tony soberon |
| 15:14.01 | [TK]D-Fender | mosty: Best answer I could have expected from you :) |
| 15:14.14 | [TK]D-Fender | mosty:++ |
| 15:14.20 | [TK]D-Fender | mosty++ |
| 15:14.23 | rene- | mosty: will my incoming call quality improve today? |
| 15:14.38 | [TK]D-Fender | BSD_tech: not removed... RENAMED |
| 15:14.47 | BSD_tech | Renamed to what |
| 15:15.10 | [TK]D-Fender | BSD_tech: That is the correct question :) |
| 15:15.40 | mosty | rene-: my magic 8 ball says don't count on it |
| 15:15.44 | rene- | hehehe |
| 15:15.45 | *** join/#asterisk drrt (n=junior@ip242-64.baltnet.ru) |
| 15:16.10 | Mercestes | Speaking of magic 8 balls. what's a good way to troubleshoot failed DTMF detection on a PRI line. Yea, I have relaxdtmf=yes |
| 15:16.32 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 15:16.56 | rene- | mosty: damn, what about asking the new girl out, or getting some with the old one? |
| 15:17.01 | Mercestes | it works on some phones...not on others. Sangoma card. A104d |
| 15:18.03 | rene- | failed dtmf? incoming or outgoing? |
| 15:18.08 | Mercestes | incoming. |
| 15:18.15 | Mercestes | in my IVR |
| 15:18.23 | mosty | Mercestes: http://web.ics.purdue.edu/~ssanty/images/6.gif |
| 15:18.32 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 15:18.35 | Mercestes | mosty: *cries* |
| 15:19.01 | rene- | jajajajaj |
| 15:19.02 | BSD_tech | i cant find the new command |
| 15:19.05 | BSD_tech | grrr |
| 15:19.12 | rene- | Mercestes: there you go |
| 15:19.30 | *** join/#asterisk mikebwilliams (n=mikebwil@12-218-71-62.client.mchsi.com) |
| 15:19.34 | Mercestes | rene-, there I go what? |
| 15:19.42 | rene- | what do you mean works on some phones |
| 15:19.47 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 15:19.47 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:19.49 | rene- | works from some carriers? |
| 15:20.07 | Mercestes | rene-, heh, it works from one of my sprint cell phones and not on my other sprint cell phone. |
| 15:20.27 | rene- | thats odd |
| 15:20.32 | Mercestes | rene-, I agree. |
| 15:20.55 | Mercestes | but yea, from different carriers/areas it works/not works. Seems to be mostly long distance v/s local but, I wouldn't call that a 100% |
| 15:21.27 | rene- | Mercestes: talk to Sangoma? |
| 15:23.02 | Mercestes | I set it to relaxdtmf before I lost my server and that fixed it then, and all was happiness. Then the evil controller card died and now it's the same issue as before, with relaxdtmf=yes set. |
| 15:23.02 | rene- | poke at relaxdtmf code to 'relax' it even more? |
| 15:23.15 | rene- | try a different card/server? |
| 15:23.16 | Mercestes | =/ |
| 15:23.48 | Mercestes | well, I have 2 seperate carriers.... If I go back to the digium card then span 3 HDLC aborts out the ass because of the multiple clock sources I'm guessing |
| 15:24.06 | Mercestes | I had to do a rain dance to get the sangoma to work with two seperate clock sources. |
| 15:24.10 | *** join/#asterisk vykarian (i=satyrico@200.138.30.10) |
| 15:24.25 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 15:24.38 | MrWup | anyone here using asterisk in the UK? |
| 15:26.29 | rene- | ohh |
| 15:26.33 | rene- | damn |
| 15:26.42 | Mercestes | rene-, yea |
| 15:26.46 | penguinFunk | MrWup: yes |
| 15:26.50 | penguinFunk | why ? |
| 15:27.07 | *** join/#asterisk tecnico (n=tecnico@24.96.146.69) |
| 15:27.08 | MrWup | penguinFunk, analogue card linked to PSTN? |
| 15:27.53 | penguinFunk | yeah |
| 15:28.16 | rene- | i currently have a setup like that, span 1,2 come from CARRIER1 in E1 form and span 3 is from CARRIER2 and it is T1 i see no slips no frame errors, i am only taking clock from one of them |
| 15:28.22 | penguinFunk | we're using the ermitted errors were being caused by the evil middle man |
| 15:28.23 | *** join/#asterisk DBavousett (n=davidb@208.180.247.127) |
| 15:28.30 | penguinFunk | we're using the TDM400P*** |
| 15:28.31 | DBavousett | Good morning. |
| 15:28.46 | *** join/#asterisk af_ (n=getsmart@81-174-8-57.f5.ngi.it) |
| 15:28.47 | MrWup | penguinFunk, when you dial out using asterisk... and the other person picks up, does asterisk generate an answer event? |
| 15:28.48 | Mercestes | rene-, Yea, I'm CPE on two of them and Network to a brooktrout card |
| 15:28.58 | penguinFunk | yeah |
| 15:29.02 | penguinFunk | i think so |
| 15:29.16 | penguinFunk | wait. |
| 15:29.19 | penguinFunk | no it dosent |
| 15:29.25 | MrWup | penguinFunk, i am using TDM as well. when asterisk dials out, with Dial, the other party picks up, we talk, but asterisk times out with NOANSWER |
| 15:29.35 | penguinFunk | thats odd |
| 15:29.40 | penguinFunk | haev you checked your dialplan? |
| 15:29.46 | Mercestes | anyone have any luck/bad experiences with the Dell PE-2850 btw? Seems to be one of the more problematic boxes I've been on thus far. |
| 15:29.46 | MrWup | penguinFunk, because it thinks the other party hasnt picked up at all |
| 15:30.01 | *** join/#asterisk inv_arp[work] (i=junya@c-67-191-12-203.hsd1.fl.comcast.net) |
| 15:30.28 | MrWup | penguinFunk, i have a context [internal] for the SIP phones. and inside that i got exten => 123,1,Dial(zap/g1/9ww123,15) |
| 15:30.36 | MrWup | so when the SIP phone dials 123 it dials the talking clock |
| 15:30.58 | MrWup | the dial executes properly and the talking clock picks up and does time sponsore dby accurist beep beep etc |
| 15:31.08 | MrWup | but asterisk doesnt think they answered |
| 15:31.31 | *** join/#asterisk karlhaines (n=karl@unaffiliated/karlhaines) |
| 15:31.42 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-0dc3af23d3e56d2a) |
| 15:31.49 | rene- | Mercestes: oh i see, i think i am cpe to all of them |
| 15:31.51 | karlhaines | dude who works at a VOIP provider, are you here?? |
| 15:32.21 | rene- | Mercestes: i have a 2850 seems reliable |
| 15:32.26 | MrWup | penguinFunk, that should work right? |
| 15:32.29 | penguinFunk | how long does it take to timeout MrWup? is it the same ever time ? |
| 15:32.34 | penguinFunk | every* |
| 15:32.41 | DBavousett | I'm having issues with a TDM844 dialing out on the FXO ports...anyone have a few minutes to give me a hand? |
| 15:32.42 | MrWup | penguinFunk, it times out after 15 seconds |
| 15:32.50 | MrWup | penguinFunk, or whatever time u specify in the Dial command |
| 15:33.01 | MrWup | basically Dial never thinks the remote person has picked up and so times out |
| 15:33.01 | penguinFunk | yeh |
| 15:33.06 | penguinFunk | dont specify a time you nutter |
| 15:33.16 | penguinFunk | my dialplan for outgoing calls works great |
| 15:33.17 | penguinFunk | i have |
| 15:33.19 | penguinFunk | exten => _9.,1,Dial(${OUTBOUNDTRUNK3}/${EXTEN:1}) |
| 15:33.48 | penguinFunk | OUTBOUNDTRUNK3 being defined as a global variable which == OUTBOUNDTRUNK3=Zap/3 |
| 15:34.00 | MrWup | penguinFunk, ok, but say for billing purposes... i dont want to bill users when the other party hasnt picked up |
| 15:34.01 | [TK]D-Fender | MrWup: make sure to set "callprogress=no" |
| 15:34.02 | penguinFunk | so you have to dial 9(number_to_dial) |
| 15:34.05 | penguinFunk | to get an outside line |
| 15:35.43 | MrWup | http://pastebin.ca/491276 |
| 15:35.48 | MrWup | thats my zapata.conf |
| 15:36.14 | MrWup | even if i do what penguinFunk says and dont specify a timeout, i'll still never be able to tell when the remote party picks up for billing purposes |
| 15:36.16 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 15:36.32 | MrWup | it'll just work on the assumption that every number dialled counts as a call |
| 15:36.54 | MrWup | [TK]D-Fender, setting callprogress=no doesnt seem to change matters |
| 15:39.44 | bkw_ | karlhaines, what up? |
| 15:39.51 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:40.40 | irule | how can I unisnstall *? |
| 15:40.58 | Mercestes | rene-, yea, it makes my life interesting with this setup all in one server. Trying to change that though. |
| 15:41.01 | [TK]D-Fender | MrWup: Call progress on analog lines is trouble. For UK info, go check out the WIKI and mailing lists |
| 15:41.18 | MrWup | yeah ive done that |
| 15:41.31 | rene- | yup get another box |
| 15:41.31 | MrWup | UK uses polarity reversal to signal line hangup or pickup |
| 15:41.34 | bkw_ | irule, you just remove all the stuff it installed. /etc/asterisk /var/spool/asterisk /usr/lib/asterisk |
| 15:41.37 | *** join/#asterisk DrCron (n=rszasz@c-67-174-231-152.hsd1.ca.comcast.net) |
| 15:42.07 | MrWup | yet answeronpolarityswitch=yes seems to be ineffective |
| 15:43.14 | DrCron | is there a recomended flavor of linux for asterisk? |
| 15:43.30 | Qwell[] | DrCron: not really |
| 15:43.48 | MrWup | penguinFunk, do your calls die after 2 mins or so? even without a timeout set, mine will just die with chanunavail after about 2 mins |
| 15:44.01 | MrWup | penguinFunk, is you zapata.conf different to mine? |
| 15:44.47 | *** join/#asterisk Ad-Hoc (n=nimbus@ppp99-60.adsl.forthnet.gr) |
| 15:45.49 | DrCron | and is there a recognised dhcp field for autoconfig of phones? |
| 15:45.50 | *** join/#asterisk DrukenLPY (n=jdumais@CPE001346f4961f-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 15:46.05 | *** join/#asterisk dlynes_laptop (n=dlynes@d207-216-161-56.bchsia.telus.net) |
| 15:46.15 | CunningPike | dlynes_laptop!! |
| 15:46.43 | penguinFunk | MrWup: my calls do not die until someone hangs up |
| 15:46.55 | penguinFunk | here is my zapata.conf: http://pastebin.ca/491283 |
| 15:47.47 | MrWup | thanks |
| 15:47.58 | penguinFunk | no probs :] |
| 15:48.14 | rene- | DrCron: meaning ? |
| 15:48.46 | rene- | DrCron: i guess yoy could add extensions and send info like sip registrar to the phone |
| 15:48.54 | rene- | but the phone should be looking for it |
| 15:49.02 | rene- | i am not aware of anything like that tho |
| 15:49.25 | [TK]D-Fender | DrCron: Depends on the phone |
| 15:50.00 | rene- | DrCron: i guess most phones, will ask for the address of the TFTP server that holds configuration files |
| 15:50.13 | *** join/#asterisk Vec (n=Vec@dsl-244-218-226.telkomadsl.co.za) |
| 15:50.35 | penguinFunk | MrWup: why do you have a timeout in your dial command for ? |
| 15:50.37 | rene- | that address is a standard option in dhcp i think |
| 15:50.44 | penguinFunk | that doesnt make sense |
| 15:50.49 | penguinFunk | to me anyway |
| 15:50.52 | Qwell[] | rene-: option-66 |
| 15:50.57 | rene- | true |
| 15:52.06 | MrWup | penguinFunk, when you dial out and the remote party hangs up, it generates a hangup event, but does it also generate a chanunavail event? |
| 15:52.09 | MrWup | cause with mine it does |
| 15:52.17 | *** join/#asterisk stuntdouble (n=ronald_l@082.143-60-66.DIA-subnet.surewest.net) |
| 15:52.21 | MrWup | (since the channel had never been picked up) |
| 15:53.02 | penguinFunk | when you dial outside it should bridge the sip user connection with a ZAP channel |
| 15:53.21 | rene- | it would be cool |
| 15:53.32 | *** part/#asterisk DBavousett (n=davidb@208.180.247.127) |
| 15:53.40 | rene- | that phones truly would configure themselves in a standard format |
| 15:54.03 | rene- | most read xml but i mean logic format, like a standard syntax |
| 15:54.04 | stuntdouble | I have a question on the g729 licensing? Would I only need 3 licenses, if I have 3 phones? I read on voip-info.org that I would need 6 since there's 3 phones plus the MOH. |
| 15:54.10 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 15:54.16 | cnbrk | moh does not need the license |
| 15:54.23 | cnbrk | as i know |
| 15:54.41 | stuntdouble | How about license for the voicemail? |
| 15:54.45 | penguinFunk | MrWup: here is the output from dialling 9123 (the time) if this helps |
| 15:54.48 | penguinFunk | <PROTECTED> |
| 15:54.48 | penguinFunk | <PROTECTED> |
| 15:54.48 | penguinFunk | <PROTECTED> |
| 15:54.48 | penguinFunk | <PROTECTED> |
| 15:54.48 | penguinFunk | <PROTECTED> |
| 15:54.48 | cnbrk | however if you are using it for non-profit use, you can get the free license |
| 15:54.49 | rene- | if you make your music on hold music g729 format you wouldnt need a license for that |
| 15:54.51 | [TK]D-Fender | stuntdouble: If your MOH is encoded in G.729 then it won't need a license |
| 15:54.55 | penguinFunk | oops |
| 15:55.20 | cnbrk | get the free g729 binary if you are going to use it non-profit |
| 15:55.22 | MrWup | ah |
| 15:55.27 | MrWup | you do get an answered event |
| 15:55.30 | penguinFunk | MrWup: http://pastebin.ca/491298 |
| 15:55.31 | rene- | or live in a not 1 tier country |
| 15:55.33 | MrWup | thats just what im looking for |
| 15:55.35 | MrWup | wit mine... |
| 15:55.36 | MrWup | http://pastebin.ca/491297 |
| 15:55.40 | MrWup | it does executing |
| 15:55.41 | MrWup | it does called |
| 15:55.45 | MrWup | answered never happens |
| 15:55.55 | MrWup | then when the remote party hangs up it does hangup, same as you |
| 15:55.58 | stuntdouble | Thanks |
| 15:56.02 | MrWup | but then does chanunavail etc |
| 15:57.07 | MrWup | inbetween called and hngup, i can talk with the remote party |
| 15:57.54 | penguinFunk | its not detecting the other party 'answering' |
| 15:57.58 | penguinFunk | even tho it does |
| 15:57.59 | MrWup | indeed |
| 15:58.03 | *** join/#asterisk MindTheGap (n=iote@mail.lpj.com.br) |
| 15:58.06 | MrWup | thats my problem |
| 15:58.30 | MrWup | just trying your zapata.conf out |
| 15:59.19 | penguinFunk | MrWup: you shouldn't irc as root either |
| 15:59.36 | MrWup | yeah i know |
| 15:59.39 | penguinFunk | :P |
| 16:00.24 | MindTheGap | at the CLI, if I type "sip show re(TAB)", CLI hangs... anyone experiente this? |
| 16:02.20 | MrWup | ok |
| 16:02.24 | MrWup | tried your zapata.conf |
| 16:02.38 | MrWup | and it does the same as u posted |
| 16:02.38 | MrWup | with the answer event |
| 16:02.42 | MrWup | but the answer happens as soon as the number is dialled |
| 16:02.51 | penguinFunk | MindTheGap: nope, what version of * you using? |
| 16:03.17 | penguinFunk | MrWup: does it time out during the call? |
| 16:03.24 | penguinFunk | it all happens pretty quickly |
| 16:03.42 | penguinFunk | i.e is your original problem still existent? |
| 16:04.16 | *** join/#asterisk wunderkin (i=wunderki@ip68-108-204-139.ph.ph.cox.net) |
| 16:04.19 | MindTheGap | 1.4.4 |
| 16:04.42 | MindTheGap | penguinFunk, 1.4.4 |
| 16:05.57 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 16:07.13 | coolbeans | A higher TTL indicates fewer hops, right? |
| 16:08.26 | penguinFunk | yes coolbeans |
| 16:08.37 | penguinFunk | not necassarily tho i guess |
| 16:08.37 | coolbeans | Thanks. What does it start at? 255? |
| 16:08.47 | penguinFunk | depends on number of hops |
| 16:10.05 | penguinFunk | time to live is measured in seconds, although every host that passes the datagram must reduce the TTL by at least one unit. In practice, the TTL field is reduced by one on every hop. |
| 16:10.43 | coolbeans | So a 244ttl, 35ms over 15 hops is a good connection vs a 68TTL over 15 hops averaging still 35ms? |
| 16:11.11 | *** join/#asterisk adker (n=chatzill@74-33-221-202.br1.glv.ny.frontiernet.net) |
| 16:11.22 | penguinFunk | theoretically yes |
| 16:11.37 | coolbeans | Are there any OS tools that easily measure packet loss? |
| 16:11.41 | penguinFunk | the packet got there quicker since its TTL is still high |
| 16:11.49 | coolbeans | penguinFunk: Thanks. |
| 16:11.57 | penguinFunk | ping measures packetloss |
| 16:12.22 | coolbeans | Ahh. I see it now. Thanks! |
| 16:13.35 | *** join/#asterisk [[blah]asfd (n=ckwall@206.71.78.178) |
| 16:13.40 | penguinFunk | :] |
| 16:14.40 | [[blah]asfd | having trouble with a sangoma installation. I get everything installed and then once it goes to configure the devices I get the message: http://pastebin.ca/491321 |
| 16:14.58 | [[blah]asfd | i have tried a few ways to stop hotplug, but have not been successful. |
| 16:15.08 | [[blah]asfd | could anyone teach me how to disable this? |
| 16:20.26 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 16:21.27 | *** join/#asterisk Waverly360 (n=irc@209.12.249.243) |
| 16:22.27 | [TK]D-Fender | [[blah]asfd: http://wiki.sangoma.com/wanpipe-linux-asterisk-appendix |
| 16:22.31 | MrWup | penguinFunk, it seems to work ok in that the call appears to be answered just after the dialk |
| 16:22.44 | MrWup | as far as i know this is the normal way that asterisk handles things |
| 16:22.54 | MrWup | so thanks for sorting that out =] |
| 16:24.00 | MrWup | only problem now is that since the call is deemed answered when it starts ringing the other party... for my billing purposes, even if the other party doesnt pick up, the call has still been answered |
| 16:24.18 | MrWup | is there some way of listening to the line and detecting when the line stops ringing? (i.e. someone has picked up) ? |
| 16:24.54 | *** join/#asterisk tuan_modulis (n=chatzill@3-82-252-216-static.enter-net.com) |
| 16:25.43 | penguinFunk | no idea |
| 16:25.45 | Strom_M | MrWup: sort of, but it's not terribly reliable |
| 16:25.57 | penguinFunk | and np :) |
| 16:25.58 | Waverly360 | Hey guys, what does it mean to use G.729 in pass-thru mode? |
| 16:26.20 | Strom_M | Waverly360: it means you're passing the audio through the PBX without transcoding it |
| 16:26.59 | MrWup | Strom_M, it might be of some help to me as i record all calls, so i dont want a folder full of wav files of ring-tone when nobody ever picked up |
| 16:27.08 | Waverly360 | Strom_M: How does that work with the licensing? |
| 16:27.27 | Strom_M | Waverly360: you only need a license to transcode |
| 16:27.51 | Strom_M | MrWup: well....if proper call progress information is important to you, I'd recommend ISDN |
| 16:28.13 | MrWup | yeah so would i |
| 16:28.21 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 16:28.22 | MrWup | unfortunately im stuck with POTS |
| 16:28.26 | *** join/#asterisk keulin (n=cray@AMontpellier-152-1-57-204.w83-197.abo.wanadoo.fr) |
| 16:29.32 | Strom_M | where are you located? |
| 16:29.42 | Waverly360 | Strom_M: Wait, so that means if I dial from a Polycom phone to another Polycom, and the phone uses g.729, asterisk can pass it through directly without using a license? |
| 16:29.51 | Waverly360 | Strom_M: or am I misunderstanding? |
| 16:30.09 | Strom_M | Waverly360: that's exactly it |
| 16:30.22 | Waverly360 | Hmm. |
| 16:30.23 | *** join/#asterisk shinao1 (n=shinao1@196.3.63.252) |
| 16:30.48 | Black-Kakugane-1 | however, you will need a liscense if you want to use call monitor or voicemail or the like |
| 16:30.53 | MrWup | anyone know the best way to detect answer (end of ringing) on a POTS line? |
| 16:31.09 | Strom_M | MrWup: put "callprogress=yes" in zapata.conf |
| 16:31.12 | Waverly360 | Black-Kakugane-1: Yeah, cause asterisk would have to transcode it to understand it for that app..I see. :) |
| 16:31.43 | *** join/#asterisk mikebwilliams (n=mikebwil@12-218-71-62.client.mchsi.com) |
| 16:32.14 | mikebwilliams | hey all, what does the "continue" do in GotoIf($[ ${TTL} > 0 ]?continue) this do? Actually, I just can't find an explanation of continue online |
| 16:32.19 | MrWup | thanks |
| 16:32.54 | *** join/#asterisk ZaVoid (n=zavoid@160.79.136.150) |
| 16:33.19 | ZaVoid | hey guys quick question |
| 16:33.38 | ZaVoid | is there a way in sip.conf to append a prefix(4 digits) to every outgoing call to a certain [name] item |
| 16:33.56 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 16:36.20 | *** join/#asterisk renier (n=renier@69.79.111.24) |
| 16:36.38 | renier | hello, have a question about fxotune |
| 16:37.22 | renier | what is the number after the -i options? I've read that it is "the keypress that will clear the dialtone". But I don't understand what that means, though. |
| 16:37.36 | tuan_modulis | mikebwilliams: continue is probably a name associated to the priority |
| 16:38.47 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 16:38.47 | tuan_modulis | exten => s,n(continue),blabla |
| 16:38.47 | [[blah]asfd | mikebwilliams: its a priority label. |
| 16:38.58 | [[blah]asfd | so instead of going 1,2,3... it can go 1,n,n,n(continue) |
| 16:39.44 | unixlike | i'm looking for a way to make asterisk bridge an incoming call with an outgoing number. i have 3 pots lines |
| 16:40.06 | *** join/#asterisk notoriousrab1982 (n=notoriou@207.47.34.74.static.nextweb.net) |
| 16:40.20 | *** join/#asterisk DarKnesS_WolF (n=wolf@unaffiliated/sherif) |
| 16:41.08 | *** join/#asterisk PioneerVM2 (n=IceChat7@ool-45779466.dyn.optonline.net) |
| 16:41.31 | ZaVoid | anyone anyone? |
| 16:42.21 | PioneerVM2 | Hi -- Im using Linksys PAP2T to connect to Asterisk, but have some very slight garbling -- its not a lot but noticable ... is there any parameters I can tweak? I have a Vonage PAP2 box at home as well and its fine and clear -- and have tested the network to my asterisk box and the audio from an analog line to it is fine, so leads me to beleive its a setting in my unlocked PAP2T |
| 16:43.15 | *** join/#asterisk hrmphh (i=patrick@notchill.com) |
| 16:43.21 | hrmphh | any idea what might have caused: Power alarm on module 1, resetting! |
| 16:43.38 | hrmphh | im getting serious static right now too |
| 16:43.40 | hrmphh | on the fxs port |
| 16:43.51 | *** join/#asterisk fx0 (n=fx0@cypher.punk.net) |
| 16:44.06 | renier | unixlike: so you want asterisk to pass the incoming call on to another zap channel? |
| 16:44.13 | [TK]D-Fender | unixlike: Answer the first, dial out another. |
| 16:44.33 | unixlike | yes thats it |
| 16:45.21 | [TK]D-Fender | unixlike: You just dial out like any other dial. there is nothing special about this. |
| 16:45.22 | unixlike | i tried to answer the first, and the next priority is dial(zapblaba/5555555), but it hangup the lines |
| 16:45.35 | [TK]D-Fender | unixlike: pastebin what you've done |
| 16:45.47 | unixlike | ok gimme a sec |
| 16:45.58 | ZaVoid | what about passing 111,222,333,444, in the front of the dialed #'s on specifc peers in sip.conf? adding prefixs basically |
| 16:46.11 | hrmphh | why would i get a power alarm on the module? |
| 16:46.12 | ZaVoid | but i don't wanna do it on my dial commands.. i want to do it based on the sip.conf entries |
| 16:47.23 | unixlike | [TK]D-Fender: first i Answer(), then people do 0, then its exten => 0,1,Macro(dial, 5555555), and here is the macro |
| 16:47.31 | unixlike | exten => s,1,ChanIsAvail(Zap/2&Zap/3&Zap/4) |
| 16:47.40 | unixlike | exten => s,2,Dial(${CUT(AVAILCHAN||1)}/${ARG1}) |
| 16:47.48 | unixlike | exten => s,3,HangUp() |
| 16:48.07 | wunderkin | you should use call groups for that |
| 16:48.12 | [TK]D-Fender | unixlike: PASTBIN it including the failed Attemp. |
| 16:48.33 | PioneerVM2 | anyone know what protocol Vonage uses with their PAP2 boxes? |
| 16:48.45 | unixlike | oh ok hold on |
| 16:49.14 | truz_`24 | What is a good card for 4 line analog? |
| 16:49.36 | [TK]D-Fender | PioneerVM2: SIP, the only protoal that any PAP2 uses |
| 16:49.52 | [TK]D-Fender | truz_`24: Sangoma A200d |
| 16:50.11 | truz_`24 | So there is no good digium card? |
| 16:50.34 | Qwell[] | truz_`24: the TDM400p |
| 16:50.56 | brookshire | or the tdm800p :) |
| 16:51.03 | Qwell[] | true |
| 16:51.03 | PioneerVM2 | sorry i meant algorithm i suppose |
| 16:51.07 | Qwell[] | he did ask for 4 ports though :p |
| 16:51.10 | truz_`24 | 800 is 8 lines? |
| 16:51.10 | PioneerVM2 | compression algorithm |
| 16:51.15 | Qwell[] | yes |
| 16:51.17 | brookshire | up to eight lines |
| 16:51.22 | brookshire | however, you can get it with 4 |
| 16:51.28 | Qwell[] | and TDM2400p with up to 24 |
| 16:51.41 | brookshire | and you can get that with 4 lines as well |
| 16:51.58 | Qwell[] | 2400 doesn't take the single span modules, does it? |
| 16:52.04 | brookshire | no |
| 16:52.09 | Qwell[] | 800 does? |
| 16:52.10 | brookshire | only quad modules |
| 16:52.17 | brookshire | yes, i believe it does |
| 16:52.20 | *** join/#asterisk TUplink (n=TUplink@c-24-126-39-22.hsd1.wv.comcast.net) |
| 16:52.23 | Qwell[] | right |
| 16:52.24 | brookshire | the 800 does i believe |
| 16:53.01 | Strom_M | 800 definitely does |
| 16:53.04 | TUplink | how do i get app_flite...... freebsd..... asterisk 1.4 |
| 16:53.10 | Strom_M | but only four of them |
| 16:53.59 | brookshire | strom: aren't you like teaching a class? |
| 16:54.15 | Strom_M | they're busy figuring out how to get their boxes to talk directly to each other |
| 16:54.23 | Strom_M | plus lunch is five minutes away |
| 16:54.30 | Strom_M | http://dvorak.org/blog/rednecks/redneck1.jpg |
| 16:54.36 | *** join/#asterisk mikebwilliams (n=mikebwil@12-218-71-62.client.mchsi.com) |
| 16:54.36 | brookshire | strom: you student it out here talking about handguns |
| 16:54.40 | brookshire | your also |
| 16:54.49 | brookshire | s/it/is |
| 16:54.52 | truz_`24 | so is Sangoma A200d just as good as the tdm400? |
| 16:54.53 | brookshire | blah.. can't type today |
| 16:55.16 | Strom_M | brookshire: ah |
| 16:55.16 | tuan_modulis | I'm looking up sipX to configure some Polycoms... is it possible to use that to configure the "Messages" button to use *97 |
| 16:55.16 | tuan_modulis | im amenable to other suggestions too since im new at this |
| 16:55.18 | Strom_M | he just came back into the room |
| 16:55.23 | brookshire | yes.. him |
| 16:55.48 | TUplink | app_flite.c how do i get it? |
| 16:56.21 | hrmphh | grr |
| 16:56.23 | hrmphh | had to reboot server |
| 16:56.25 | hrmphh | to clear up static issue |
| 16:56.41 | hrmphh | the TDM400P is a pos |
| 16:56.49 | *** join/#asterisk ToyMan (n=Stuart@72.168.167.241) |
| 16:56.53 | [TK]D-Fender | tuan_modulis: yes |
| 16:57.06 | drako | hrmphh, pos? |
| 16:57.08 | tuan_modulis | thx |
| 16:57.11 | hrmphh | piece of shit |
| 16:57.21 | hrmphh | i get static all the time on an unloaded core 2 duo box |
| 16:57.24 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 16:57.29 | drako | hrmphh, me too |
| 16:57.30 | [TK]D-Fender | hrmphh: Don't hold back... tell us how you REALLY feel about it! ;) |
| 16:57.30 | hrmphh | well not all the time, but intermittently AWFUL |
| 16:57.32 | drako | hrmphh, i agree |
| 16:57.45 | hrmphh | digium support = useless |
| 16:57.50 | hrmphh | theyll email you 4-6 weeks after you email them |
| 16:57.54 | hrmphh | with some silly questions |
| 16:57.55 | hrmphh | you respond |
| 16:57.56 | drako | hrmphh, ahaha we felt the same |
| 16:57.57 | hrmphh | and no answer |
| 16:57.59 | bkruse | brookshire: is the class that bad this time? |
| 16:58.11 | drako | any other real good alternative? |
| 16:58.13 | hrmphh | im about to drop asterisk and replace it with cisco callmanager express |
| 16:58.13 | drako | maybe sangoma? |
| 16:58.48 | [TK]D-Fender | hrmphh: Aside from your TDM400P woes, any other reasons to ditch *? |
| 16:58.49 | hrmphh | it cant even support a 3 line phone system w 1/analog phone on fxs and 2 sip phones |
| 16:58.53 | unixlike | [TK]D-Fender : http://www.pastebin.ca/491381 |
| 16:59.01 | hrmphh | TK; well asterisk itself has locked up on me several times |
| 16:59.19 | [TK]D-Fender | unixlike: -- Executing Dial("Zap/2-1", "Zap/3/REMY") in new stack |
| 16:59.26 | hrmphh | i dont like having to reboot a pbx like i do a winME machine |
| 16:59.29 | [TK]D-Fender | unixlike: Since when is REMY a phone NUMBER?! |
| 16:59.51 | hrmphh | dont get me wrong, i love the flexibility of asterisk, but the stability is to be laughed at |
| 16:59.53 | Strom_M | [TK]D-Fender: obviously it's 7369 |
| 17:00.04 | ZaVoid | winME RULES |
| 17:00.15 | unixlike | [TK]D-Fender : REMY is defined by REMY=5555555 |
| 17:00.15 | hrmphh | like wtf is this "Power alarm on module 1, resetting!" |
| 17:00.22 | hrmphh | ${REMY}? |
| 17:00.32 | unixlike | shit dude |
| 17:00.34 | unixlike | sorry |
| 17:00.53 | [TK]D-Fender | unixlike: No, it is NOT. That is raw text being shoved on that dial. You have not correctly referenced a variable. |
| 17:01.21 | unixlike | sorry it was noob eheh |
| 17:02.43 | brookshire | bkruse: i don't know.. they are really quite |
| 17:02.53 | bkruse | brookshire: thats scary. |
| 17:03.23 | Qwell[] | they are really quite what? |
| 17:03.51 | brookshire | quiet |
| 17:03.53 | brookshire | bah |
| 17:04.09 | bkruse | i gotcha |
| 17:04.52 | *** part/#asterisk [[blah]asfd (n=ckwall@206.71.78.178) |
| 17:05.07 | hrmphh | does digium plan to hire more people or just chill and not answer anything? |
| 17:05.24 | brookshire | hrmphh: what do you want digium to answer? |
| 17:06.03 | hrmphh | a case i put in about a month ago? |
| 17:06.11 | hrmphh | audio quality issues on the TDM400P |
| 17:06.16 | hrmphh | they asked me for lspci and such |
| 17:06.18 | hrmphh | i responded |
| 17:06.19 | TUplink | i would bet on the later |
| 17:06.21 | hrmphh | no answer from them |
| 17:06.31 | bkruse | hrmphh: thats no good |
| 17:06.45 | hrmphh | sitting on hold now |
| 17:06.47 | hrmphh | having meeting in 30mins |
| 17:06.51 | hrmphh | id say about 5% chance i get through |
| 17:06.53 | hrmphh | before meeting starts |
| 17:07.22 | brookshire | hrmphh: do you have a ticket number? |
| 17:08.03 | hrmphh | yes |
| 17:08.08 | hrmphh | do you work for digium? |
| 17:08.14 | brookshire | yes, but in IT |
| 17:08.15 | bkruse | brookshire: run! |
| 17:08.17 | *** join/#asterisk TechBlazer (n=tim@70.88.27.130) |
| 17:08.27 | *** join/#asterisk CrazyTux (n=CrazyTux@spyglass.houston.hostgator.com) |
| 17:08.42 | brookshire | i can at least make a note of it |
| 17:08.45 | hrmphh | oh so you program the Wait(rand(3600,9600)) after the Answer() when calling tech support |
| 17:08.59 | hrmphh | :) |
| 17:09.11 | brookshire | nah.. i think it's just lunch |
| 17:10.00 | bkruse | I have alot of respect for tech support |
| 17:10.02 | brookshire | which means.. fewer people answering phones :( |
| 17:10.38 | *** join/#asterisk alrs (n=lars@pozug.com) |
| 17:10.39 | hrmphh | shrug i do when they are responsive |
| 17:10.48 | hrmphh | may not be their fault if digium is severely understaffed |
| 17:10.53 | hrmphh | or churning out defective hardware |
| 17:11.23 | bkruse | I would go with c) theres alot of dumb people that cannot read help pages |
| 17:11.37 | bkruse | but who wants to read a help page when you can call and talk to someone that knows what they are talking about |
| 17:11.49 | hrmphh | shrug |
| 17:11.52 | hrmphh | this is a HARDWARE issue |
| 17:12.04 | hrmphh | if you dont pay for software support, they shouldnt take your call |
| 17:12.22 | hrmphh | if you sell defective/incompatible hardware |
| 17:12.23 | hrmphh | you should support it |
| 17:12.23 | *** join/#asterisk cnet2 (n=nada@190.10.0.120) |
| 17:12.24 | bkruse | It is good to have more information than "sound quality is bad" |
| 17:12.28 | fx0 | rma the tdm400p and be happy again, simple. |
| 17:12.32 | hrmphh | ... |
| 17:12.36 | bkruse | thats kind of hard to diagnose right off the bat :[ |
| 17:12.37 | hrmphh | fx0; until it breaks again? |
| 17:12.56 | hrmphh | bkruse; oh you mean with lspci output, full debug logs, /proc/interrupts, etc? |
| 17:13.02 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 17:13.03 | hrmphh | its not like ok thanks for the info, lets try this |
| 17:13.15 | hrmphh | its ok, apparently we received it, but dont feel like responding to you |
| 17:14.04 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 17:14.37 | bkruse | I agree though, it should be taken care of quicker than this. For that I am sorry :[ |
| 17:15.24 | hrmphh | maybe digium will swap my tdm400 card for sangoma at their cost |
| 17:16.24 | hrmphh | initial email was 4/20 (yeah yeah) |
| 17:16.28 | hrmphh | initial resposne was 5/6 |
| 17:16.30 | hrmphh | sick response time |
| 17:16.40 | [TK]D-Fender | hrmphh: I doubt that. Best they'd do is bet you a working one if they can find fault with yours, after which yuo can sell off and recoup your $ to buy something else YOU want. Its not their job to buy a competitors product for you. Only to support theirs |
| 17:16.47 | *** join/#asterisk sasch (n=sasch@host102-30-static.107-82-b.business.telecomitalia.it) |
| 17:16.50 | sasch | hi all |
| 17:16.54 | *** part/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 17:16.56 | hrmphh | right but if theirs in unsupportable |
| 17:16.57 | hrmphh | ... |
| 17:16.57 | hrmphh | anyway |
| 17:16.58 | hrmphh | meeting |
| 17:16.58 | hrmphh | bbl |
| 17:17.18 | *** join/#asterisk axisys (n=axisys@72-255-17-136.client.stsn.net) |
| 17:18.20 | *** join/#asterisk scurb (n=scurb@c-25aae355.14-16-64736c13.cust.bredbandsbolaget.se) |
| 17:18.40 | renier | hrmphh: I have a tdm400p. don't get static. two sip devices. two pots lines. |
| 17:19.07 | *** part/#asterisk TUplink (n=TUplink@c-24-126-39-22.hsd1.wv.comcast.net) |
| 17:19.16 | truz_`24 | so when u get the tdm400p, u have to get modules with it as well? |
| 17:21.06 | brookshire | truz_`24: you can buy it with variouse configurations of modules |
| 17:21.06 | brookshire | various also |
| 17:21.06 | truz_`24 | lol |
| 17:21.06 | truz_`24 | so the modules control wether its a VOIP card or an analog card? |
| 17:21.06 | brookshire | i don't think anyone will sell you a tdm400p without them.. |
| 17:21.06 | truz_`24 | someone on ebay is |
| 17:21.07 | [TK]D-Fender | truz_`24: No. |
| 17:21.07 | brookshire | oh wow |
| 17:21.07 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 17:21.09 | [TK]D-Fender | truz_`24: The module determin how many PHONES, or LINES it can take in. |
| 17:21.09 | renier | truz_`24: the modules control how you use each port on the card. its always a voip card. |
| 17:21.10 | cnet2 | hi, i have several VoIP providers, but I want to equally load each provider, how can I set so that after one call is on provider1, the 2nd call is directed through provider2 ? |
| 17:21.13 | brookshire | the tdm400p is an analog card |
| 17:21.19 | brookshire | you do not need a card to do VOIP |
| 17:21.20 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 17:21.22 | [TK]D-Fender | renier: it is NOT a voip card! |
| 17:21.25 | grEvenX | hi all |
| 17:21.48 | truz_`24 | So what modules do i need with the tdm to hook up 4 analog lines to it? |
| 17:21.51 | grEvenX | anyone know if there's any good solution out there for custom CDR fields in a database yet? |
| 17:21.55 | renier | [TK]D-Fender: ah, sorry. got confused by how I use it at my end. |
| 17:22.08 | [TK]D-Fender | truz_`24: You need 4 FXO modules (the Red ones) |
| 17:22.16 | truz_`24 | k |
| 17:22.57 | *** join/#asterisk afrosheen (n=cj@207.71.49.164) |
| 17:22.59 | TechBlazer | <PROTECTED> |
| 17:23.04 | truz_`24 | what is the difference bewteen fxo and fxs modules? |
| 17:23.11 | [TK]D-Fender | renier: The thing you need to keep in mind the card lets * use your line. The card has nothing to do with VoIP. the fact you use * to answer the call and call out on a VoIP channel is completely irrelevent to the nature of the card. |
| 17:23.15 | afrosheen | <PROTECTED> |
| 17:23.16 | [TK]D-Fender | ~fxofxs |
| 17:23.25 | jbot | i heard fxofxs is An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
| 17:23.25 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 17:23.35 | renier | [TK]D-Fender: yep |
| 17:24.02 | renier | jbot, yep |
| 17:24.22 | truz_`24 | ah, thanks jbot |
| 17:25.03 | *** join/#asterisk ms345 (n=mike_sim@64.74.198.10) |
| 17:25.36 | renier | [TK]D-Fender: i used fxotune to deal with my echo and it worked pretty well. Do you know what is the best place to put a call to fxotune -s so that it gets sets on a system restart? |
| 17:25.47 | [TK]D-Fender | TechBlazer: There is no hardware to make * work with voip. For POTS, there are several manufacturers of PCI cards and VoIP gateways to take in analog lines. |
| 17:26.21 | [TK]D-Fender | renier: I'd mod your Zaptel startup script (whichever you use) and call it in the right order there |
| 17:26.32 | renier | ok, will look |
| 17:26.53 | renier | assuming its under /etc/init.d |
| 17:26.53 | truz_`24 | Does anyone use a ubuntu box to run their asterisk setup? |
| 17:27.39 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 17:27.50 | [TK]D-Fender | truz_`24: Many people. |
| 17:28.10 | truz_`24 | Well at the moment i don't have the tdm yet, so i have a single xp100 card, but i cannot seem to get the module loaded. |
| 17:28.16 | truz_`24 | Trying to figure out how to get it modprobe'd in |
| 17:28.22 | truz_`24 | so i can play with single line setups |
| 17:28.23 | ms345 | If I insert * between my phone switch and a long-distance T1 I know I run pri_cpe on the asterisk span than connects to the T1 and pri_net on the span that connects to the phone switch. Can someone confirm that I use a straight patch cable between * and the T1 and a T1 crossover cable between asterisk and the phone switch? Does that sound right? |
| 17:28.28 | *** join/#asterisk angom (n=angom@red-corp-201.143.54.251.telnor.net) |
| 17:28.39 | truz_`24 | It uses the zaptel module apparently... |
| 17:29.04 | aptura | truz_`24 so what is the problem |
| 17:29.25 | *** part/#asterisk angom (n=angom@red-corp-201.143.54.251.telnor.net) |
| 17:29.25 | truz_`24 | modprobe zaptel returns no zaptel module found |
| 17:29.31 | aptura | k |
| 17:29.36 | truz_`24 | but zaptel is installed |
| 17:30.21 | [TK]D-Fender | ms345: Sounds about right |
| 17:30.57 | ms345 | thx - I did it about a year ago and know I needed a crossover somewhere. thx again. |
| 17:32.06 | sasch | anyone have a how to for using SLA with asterisk ?? |
| 17:33.47 | *** join/#asterisk lerat (n=dnormand@bas2-montreal19-1178031196.dsl.bell.ca) |
| 17:33.49 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 17:33.53 | lerat | Hi all |
| 17:34.23 | [TK]D-Fender | sasch: Its documented in your source tarball and on asterisk.org |
| 17:35.13 | Ifaistos | truz_`24: probably when installed the zaptel haven't run depmod |
| 17:35.18 | lerat | I need the exact link to download g729 from the CLI. The link on digium web site (download procedure) doesn t work can someone help me please |
| 17:36.33 | lerat | i ve downloaded the register programme with no problem but the codec....nada ! |
| 17:36.40 | TechBlazer | [Tk]D-Fender: Would I just plug the IP phones into my existing network, configure * to recognize each phone, and sign up with a VoIP provider? |
| 17:37.08 | *** part/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 17:37.13 | [TK]D-Fender | TechBlazer: You could do that and forgo and PSTN hardware at all and let your ITSP terminate for you. |
| 17:37.34 | TechBlazer | Wow, that sounds too easy. |
| 17:37.47 | [TK]D-Fender | TechBlazer: * lets you take in all sorts of devices and handle calls in/out of them any way you choose. |
| 17:38.56 | [TK]D-Fender | TechBlazer: * is not hard so much as dealing with its learning curve, and that of any other hardware/software you choose to use with it. |
| 17:40.41 | TechBlazer | [TK]D-Fender: What is needed to receive calls from POTS? |
| 17:43.20 | anonymouz666 | what represents && logic in dialplan? |
| 17:43.32 | anonymouz666 | gotoif($[] && $[]?duh:blah) |
| 17:43.42 | renier | TechBlazer: you'd need an analog card like the tdm400p. |
| 17:46.15 | *** join/#asterisk osiris250 (i=mc9m3u3a@bsd02.evansengineering.net) |
| 17:47.46 | [TK]D-Fender | TechBlazer: If you're referring to an analog POTS line near your server : |
| 17:48.53 | [TK]D-Fender | TechBlazer: Digium TDM400P, Sangoma A200(d), Linksys SPA-3102 / SPA-400, and much more |
| 17:49.10 | TechBlazer | Yes, the PSTN I guess. Sorry, still new the lingo. |
| 17:50.47 | osiris250 | I am having an issue with Cisco 7960s and Asterisk 1.4.4.. With Asterisk 1.2, I was able to be on the phone and still receive incoming calls (to the same line or a different line configured on the phone.) Since upgrading to 1.4.4, I am unable to get "call waiting" or multiple calls working at the same time. Asterisk reports 'Got SIP response 486 "Busy here" back' when trying to dial the phone while I am on it. Any ideas on how to get ca |
| 17:50.47 | osiris250 | ll waiting working? |
| 17:50.57 | TechBlazer | <PROTECTED> |
| 17:51.17 | [TK]D-Fender | TechBlazer: No, the TDM400P is a PCI card you'd install in your * server PC. |
| 17:52.10 | [TK]D-Fender | TechBlazer: the Linksys ones are external SIP gateways that would talk to * via your lan. other gateway makers are AudioCodes, Mediatrix, and PAtton. Not much else that I would suggest |
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| 17:54.58 | *** join/#asterisk ToyMan (n=Stuart@12.23.30.130) |
| 17:55.25 | TechBlazer | [TK]D-Fender: Okay, I think I got it now. Thanks for the info, you have really cleared some things up for me. |
| 17:55.39 | [TK]D-Fender | TechBlazer: np. |
| 17:55.52 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 17:57.21 | [TK]D-Fender | osiris : thats your phone saying its busy. So thats what need to be corrected, not * |
| 17:57.53 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 17:57.58 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 17:58.54 | osiris250 | [TK]D-Fender: yeah, but what's weird is that it was working under 1.2, but after upgrading to 1.4.4, it stopped. From testing it only does this when I am currently dialing a call. If the dialed call is connected, call waiting then works.. |
| 17:59.06 | osiris250 | [TK]D-Fender: Thanks |
| 17:59.40 | *** join/#asterisk ctaloi (n=ctaloi@nat-66-218-1-147.usadatanet.com) |
| 18:00.49 | Waverly360 | I'm looking to start using some of the AudioCodes stuff |
| 18:01.04 | Waverly360 | Their analog and pri gateways seem to be really nice. |
| 18:02.41 | Waverly360 | I'm on digium's ftp site looking at the available g729 codecs I can download, and there's about 12 of them listed under the asterisk-1.2/x86-32/ directory. They all look to be for specific processor types. |
| 18:03.02 | Waverly360 | Will any of them work or do I have to use one specifically built for my processor? |
| 18:03.31 | Waverly360 | i386 is one of the choices, so I assumed that would would work fine with what I have. |
| 18:04.15 | [TK]D-Fender | Waverly360: pick the closest choice. |
| 18:04.54 | Waverly360 | [TK]D-Fender: Well, the problem is that I have several customers on slightly different hardware. Some are using Intel, some AMD |
| 18:05.19 | hrmphh | hrm |
| 18:05.20 | [TK]D-Fender | Waverly360: What arch is your OS installed as? |
| 18:05.21 | Waverly360 | [TK]D-Fender: I'm looking for something pretty generic that I can install across all customers. |
| 18:05.24 | hrmphh | how much are you guys paying for sangoma cards |
| 18:05.29 | hrmphh | 2 fxs / 4 fx0 |
| 18:05.37 | Waverly360 | [TK]D-Fender: CentOS with the i386 rpms |
| 18:05.44 | hrmphh | about to dump this garbage tdm400p card |
| 18:05.48 | [TK]D-Fender | Waverly360: I always use CentOS x386, so taht about covers it. |
| 18:06.16 | [TK]D-Fender | hrmphh: Price is on par with Digium. |
| 18:06.31 | [TK]D-Fender | hrmphh: www.atacomm.com , www.voipsuply.com , www.telephonydepot.com |
| 18:06.44 | hrmphh | hmm seems like $150+ more |
| 18:06.47 | [TK]D-Fender | hrmphh: I highly recommend getting the HWEC models |
| 18:06.50 | hrmphh | HWEC? |
| 18:06.52 | hrmphh | echo cancel? |
| 18:06.54 | hrmphh | on the card? |
| 18:07.20 | [TK]D-Fender | hrmphh: Yes |
| 18:07.27 | Waverly360 | [TK]D-Fender: Ok, that's what I was hoping. |
| 18:07.50 | *** join/#asterisk saftsack (n=oliver@pD9E05BE2.dip.t-dialin.net) |
| 18:07.55 | hrmphh | jesus |
| 18:07.56 | hrmphh | $840 |
| 18:08.03 | hrmphh | for 2fxs / 4fxo |
| 18:08.06 | aptura | yup |
| 18:08.15 | aptura | :) |
| 18:08.20 | aptura | pass the cost to the customer |
| 18:08.34 | hrmphh | well considering im about to dump asterisk and go w/ccm, looks like a cheaper route |
| 18:08.54 | hrmphh | now if i could only get a refund on the tdm400p |
| 18:08.56 | aptura | w/ccm? |
| 18:08.59 | hrmphh | that has failed me so miserably |
| 18:09.01 | hrmphh | cisco callmanager |
| 18:09.05 | aptura | ohh yea |
| 18:09.11 | aptura | btw |
| 18:09.18 | aptura | asterisk is used in goverment |
| 18:09.29 | aptura | north vancouver city hall uses it to power there phones. |
| 18:10.04 | TechBlazer | Can anyone recommend a good IP phone for business use (~20 users)? I'm looking at the Grandstream GS-102. Has anyone used this one before? |
| 18:10.06 | hrmphh | to power their e911 systems? doubtful :) |
| 18:10.19 | hrmphh | to power their complaint department? prob :) |
| 18:10.44 | PioneerVM2 | Hey anyone found that using SIP to a provider works better than IAX2? Im connected to VoicePulse and find much less choppyness when using SIP |
| 18:10.51 | aptura | hrmphh the NVCH phone admin comes here all the time and told us that it is running most if not all there phones. |
| 18:11.40 | hrmphh | he must be using non-digium hardware |
| 18:11.58 | [TK]D-Fender | TechBlazer: ... |
| 18:11.59 | [TK]D-Fender | ~phones |
| 18:12.11 | jbot | [phones] http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 18:12.24 | [TK]D-Fender | ~gs |
| 18:12.26 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 18:12.43 | *** join/#asterisk lokkju_wrk_ (n=lokkju@unaffiliated/lokkju) |
| 18:13.22 | tuan_modulis | that always makes me laugh |
| 18:13.47 | TechBlazer | jbot: Thanks for the heads up on Grandstream. |
| 18:13.47 | jbot | bitte, TechBlazer |
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| 18:18.43 | *** join/#asterisk rikstah (n=rick@rhamnett.plus.com) |
| 18:18.45 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 18:19.05 | rikstah | Hey, can someone kindly explain what g729 "passthrough" means? |
| 18:19.23 | Strom_M | it means you're passing the audio through the PBX without transcoding it |
| 18:19.39 | rikstah | so this is useful when....? |
| 18:19.46 | [TK]D-Fender | rikstah: it means if you are using 2 phone, each of which support ZG.729, * can let them talk to each other without transcoding and requiring a licence. |
| 18:19.56 | rikstah | ahh great, i see...thanks |
| 18:20.29 | rikstah | so if i call a cisco 79xx from a 79xx, it will use g729 ? |
| 18:20.40 | *** join/#asterisk swyrus (n=ss@user-514c1abb.l1.c5.dsl.pol.co.uk) |
| 18:20.41 | [TK]D-Fender | rikstah: it CAN, if you tell * to allow it |
| 18:20.46 | rikstah | yeah |
| 18:20.50 | [TK]D-Fender | rikstah: and configure your phones accordingly |
| 18:21.03 | rikstah | fine. cheers |
| 18:21.21 | [TK]D-Fender | rikstah: But the moment * has to transcode you'd better have a license available for it |
| 18:21.39 | rikstah | won't it just select the next preferred codec |
| 18:22.05 | *** join/#asterisk Paul_UK (n=foo@email.seatwave.com) |
| 18:22.19 | Waverly360 | rikstah: Hah...so funny that I asked the exact same question about half an hour ago :) |
| 18:22.30 | *** join/#asterisk dasuberdavid (i=root@nat/digium/x-3a0ab0d3c79155a7) |
| 18:22.33 | rikstah | Waverly360, hehe, whoops ....i should been here :) |
| 18:22.52 | rikstah | I did try to look for an explanation but it wasn't clean on voip-info.org |
| 18:23.09 | Waverly360 | rikstah: Yeah, I did the same. Maybe someone should add an explanation of that on there :) |
| 18:23.19 | rikstah | I might do it later |
| 18:23.46 | Waverly360 | I guess I could, I just don't feel worthy enough to alter voip-info just yet :) |
| 18:23.47 | [TK]D-Fender | rikstah: if the phones can agree on a lower codec sure, but if not : transcode time. |
| 18:24.52 | *** join/#asterisk foll (n=chatzill@88-149-177-251.f5.ngi.it) |
| 18:25.03 | rikstah | ok :) FYI I added a paragraph to explain this in the wiki |
| 18:25.05 | *** join/#asterisk marc\cba (n=marc@cpc2-whit2-0-0-cust200.cdif.cable.ntl.com) |
| 18:25.10 | foll | Guys, I have a question.... |
| 18:25.40 | foll | I have been playing with trixbox for some time now, and I need to move my asterisk installation to a data center. |
| 18:25.54 | foll | and so I can start from scratch |
| 18:26.02 | *** join/#asterisk dacter (n=dlittrel@207.200.33.213) |
| 18:26.25 | Sweeper | anyone have an audiocodes mp-118 conf file lying around that works? |
| 18:26.30 | foll | Now, I am wondering if I should go and try to install trixbox on the new VM or just go ahead and do a "raw" installa of asterisk? |
| 18:26.40 | Strom_M | foll: do a raw install :) |
| 18:26.45 | marc\cba | snap, i've also got a great question! what do i need to set, in order for asterisk to take the caller id from the extn/peer, and pass it un-touched out onto its trunk? |
| 18:26.46 | _mm_ | the only answer you'll get in here is raw install :) |
| 18:26.58 | foll | Yeah, I feared that :) |
| 18:27.00 | marc\cba | aye foll |
| 18:27.02 | Waverly360 | foll: All of these guys will tell you to do a raw install. I suppose I would too, but you know trixbox. |
| 18:27.06 | marc\cba | it's not so bad foll |
| 18:27.25 | foll | Trixbox is kinda nice: all fits together, when it works.... |
| 18:27.40 | Waverly360 | foll: Is this something live that you need immediately? |
| 18:27.49 | Waverly360 | foll: or will you have time to tinker with it to get it right? |
| 18:27.56 | foll | When it doesn't, you have to get bloody, because you have to really understand what the authors wanted to achieve in the first place! |
| 18:28.19 | [TK]D-Fender | marc\cba: It will aready do that normally. |
| 18:28.20 | foll | Waverly360: no, I can take my time with it. |
| 18:28.21 | anonymouz666 | codefreeze: exten => s,2,Set(CDR(MyFavoriteBand)=Foo Fighters) --> this should work on 1.2? |
| 18:28.28 | [TK]D-Fender | marc\cba: Assuming your outbound channel supports it |
| 18:28.34 | marc\cba | well - that was what i thought |
| 18:28.36 | Waverly360 | foll: Yeah, maybe you should install asterisk from scratch. |
| 18:28.38 | *** join/#asterisk TUplink (n=TUplink@c-24-126-39-22.hsd1.wv.comcast.net) |
| 18:28.39 | marc\cba | i have two * servers |
| 18:28.48 | marc\cba | an x-lite client and a carrier |
| 18:28.49 | TUplink | how do i go about getting app_flite? |
| 18:28.51 | marc\cba | sip) |
| 18:29.24 | marc\cba | so i'm trying to dial out on extn 200 on *a - which is registered on *b - which can pass out one of several caller id's |
| 18:29.51 | marc\cba | trouble is, the call - when it leaves *b, uses the local extn that *a is registered with, not the caller id passed from *a |
| 18:29.56 | marc\cba | so im confused ;p |
| 18:30.29 | marc\cba | i realise fully this is a problem with something i've done - and know exactly what is wrong |
| 18:30.32 | neverblue | ph + asterisk |
| 18:30.42 | [TK]D-Fender | marc\cba: Do you set "callerid" in your accounts? |
| 18:30.50 | marc\cba | on which * box? |
| 18:30.59 | neverblue | php + asterisk, why is there "not" much documentation on those two topics working together? |
| 18:31.08 | [TK]D-Fender | marc\cba: on either for the interconnecting accounts |
| 18:31.21 | neverblue | and alot of releases of users attempting to get them to "work together" |
| 18:31.55 | marc\cba | fender, bear with me for just a second |
| 18:31.58 | [TK]D-Fender | neverblue: maybe because they inherently have nothing to do with each other? |
| 18:33.03 | neverblue | ok, so what would you use to communicate with your asterisk server to perform different admin tasks? |
| 18:33.51 | [TK]D-Fender | neverblue: Well to pick a specific solution, I'd need a specific task I'd need to accomplish |
| 18:33.54 | ghenry | hi all |
| 18:33.58 | [TK]D-Fender | neverblue: So get SPECIFIC :) |
| 18:34.12 | ghenry | I'm trying to dial a sip url on another internal box, which is not * |
| 18:34.20 | ghenry | it connects but no audio |
| 18:34.20 | Waverly360 | neverblue: tell me you're not using commandline php to do things with asterisk. |
| 18:34.25 | neverblue | well, communicating with the manager, to say, make phone calls |
| 18:34.35 | marc\cba | yes fender, on *a - the first pbx in the call route |
| 18:34.37 | ghenry | my dial is: Dial(SIP/slee@192.168.x.x:5605,10,tTr) |
| 18:34.38 | marc\cba | callerid is set |
| 18:34.55 | ghenry | it's not a sip user on the * system |
| 18:34.57 | [TK]D-Fender | neverblue: AMI is jsut a socket interface. The format of communication is clearly documented, so the language is irrelevent |
| 18:34.58 | marc\cba | and the packet sniffer correctly identifies the "From: "00000000000" header |
| 18:35.18 | ghenry | anythign special I need to be doing? |
| 18:35.27 | ghenry | using a softphone direct is fine |
| 18:35.34 | ghenry | maybe canreinvite=no is the prob? |
| 18:35.35 | [TK]D-Fender | neverblue: if I needed something interactive one might use Java for this. (if you're looking for OS independence, etc) |
| 18:35.37 | neverblue | "format of communication is clearly documented", what do you mean by this? |
| 18:35.51 | [TK]D-Fender | neverblue: the format of AMI messages |
| 18:36.00 | marc\cba | you write to stdin, stdout and stderr i think neverblue |
| 18:36.01 | neverblue | URL? |
| 18:36.03 | marc\cba | is what he means |
| 18:36.10 | [TK]D-Fender | marc\cba: NO, that is AGI <- |
| 18:36.21 | [TK]D-Fender | marc\cba: don't confuse things! |
| 18:36.24 | marc\cba | oh sorry we're talking about AMI |
| 18:36.27 | marc\cba | ;-) |
| 18:36.47 | Waverly360 | neverblue: http://www.voip-info.org/wiki/view/Asterisk+manager+API |
| 18:37.06 | marc\cba | so yes, anyway - sorry |
| 18:37.11 | [TK]D-Fender | neverblue: AMI uses a TCP socket connection. What language you use to make it doesn't matter. The format of the message you will send/receive are well documented on the WIKI and all the programming is up to you. |
| 18:37.21 | marc\cba | *a has callerid set for the peer/extn which creates the call |
| 18:37.24 | neverblue | wiki? |
| 18:37.33 | neverblue | the previously provided one? |
| 18:37.36 | marc\cba | the correct callerid is passed to *b as confirmed by packet sniffer |
| 18:38.06 | marc\cba | *b sets up the call on its outbound trunk, but passes the callerid of the peer *a is registered to it as |
| 18:38.15 | Waverly360 | neverblue: What exactly are you wanting an answer for? I don't think any of us are really sure. |
| 18:38.50 | Waverly360 | neverblue: It sounds like maybe you're looking for example php scripts that interact with the Manager interface of asterisk...is that it? |
| 18:38.54 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 18:38.59 | hrmphh | SIP/2.0 404 Not Found |
| 18:39.09 | hrmphh | what does that mean exactly when im seeing it in the sipd ebug output |
| 18:39.20 | neverblue | we lose Fender? |
| 18:39.41 | Waverly360 | neverblue: He might be bashing his head into a desk somewhere. He does that from time to time I think. |
| 18:40.13 | neverblue | Waverly360, well, I have seen an example or two, but I am not sure if I am missing the point around working with the * Manager, or working with sockets in PHP |
| 18:40.28 | ghenry | just missing rtp I think |
| 18:40.54 | hrmphh | Looking for s in default (domain mysipdomain) |
| 18:41.06 | hrmphh | <--- Transmitting (no NAT) to mypcip:5060 --> |
| 18:41.11 | hrmphh | SIP/2.0 404 Not Found |
| 18:41.45 | Waverly360 | neverblue: Well, the point is having a method of interacting with asterisk aside from the CLI. |
| 18:42.20 | truz_`24 | how are u supposed to compile the zaptel module when the current zaptel source looks for linux/config. |
| 18:42.22 | truz_`24 | config.h |
| 18:42.54 | Qwell[] | truz_`24: yell at your distro for backporting kernel changes, and keeping the version number the same |
| 18:42.54 | *** join/#asterisk Miadon (n=Miadon@82.153.141.97) |
| 18:43.03 | *** join/#asterisk juanjoc (n=jcomella@38.100.34.2) |
| 18:43.11 | neverblue | Waverly360, can I pm for a few questions? |
| 18:43.12 | Waverly360 | neverblue: I'm not sure if I'm answering your question though. |
| 18:43.16 | Waverly360 | sure |
| 18:43.18 | *** part/#asterisk Miadon (n=Miadon@82.153.141.97) |
| 18:43.19 | neverblue | thank you |
| 18:45.20 | *** join/#asterisk someninjamaster (n=chatzill@016-047-749.area7.spcsdns.net) |
| 18:46.05 | *** join/#asterisk ccesario (n=ccesario@200-158-227-79.dsl.telesp.net.br) |
| 18:47.13 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 18:47.33 | *** join/#asterisk NirS (n=Nir@84.95.71.191.cable.012.net.il) |
| 18:48.02 | NirS | hey all |
| 18:48.02 | NirS | anybody home ? |
| 18:48.22 | marc\cba | so - a different approach |
| 18:48.49 | marc\cba | can anyone suggest a good way of connecting to asterisk servers together so that .. |
| 18:49.08 | marc\cba | the first server can place calls on the second servers sip trunks |
| 18:49.23 | marc\cba | with the second server, respecting the callerid - that the first server passes? |
| 18:49.32 | NirS | anyone compiled spandsp with 1.2.18 ? |
| 18:49.40 | [TK]D-Fender | neverblue: There are a number of exampless out the for PHPAGI library that also has an AMI set of functions. |
| 18:50.46 | [TK]D-Fender | hrmphh: As blatant as it sounds, there is no exten => s,1,..... in [default] like the call is expecting. |
| 18:50.53 | hrmphh | there is no [default] |
| 18:51.11 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:51.13 | [TK]D-Fender | hrmphh: Hence the source of your problem. Look ar your user setup, and [general] |
| 18:51.21 | hrmphh | i have ael-internal |
| 18:51.23 | [TK]D-Fender | hrmphh: in sip.conf |
| 18:51.28 | hrmphh | and ael-incoming |
| 18:51.28 | hrmphh | ok |
| 18:51.45 | hrmphh | the context of every sip device is listed as ael-internal |
| 18:51.49 | [TK]D-Fender | hrmphh: Ifs your SIP config and your extensions (AEL/regular). You should see where you are telling calls to go... |
| 18:52.03 | hrmphh | ok so general context is wrong |
| 18:52.04 | [TK]D-Fender | hrmphh: pastebin it |
| 18:52.11 | hrmphh | by the specific devices all have correct |
| 18:52.26 | marc\cba | i thought i could just setup an extn on server b, use that to register server a onto server b (this works..) |
| 18:52.27 | [TK]D-Fender | hrmphh: then the call coming in ISN"T using those peers/users |
| 18:52.33 | marc\cba | but server be won't respect the callerid |
| 18:52.39 | marc\cba | and i've got no idea what to look for |
| 18:52.47 | *** join/#asterisk RaYmAn-Bx (i=rayman@skumler.dk) |
| 18:52.48 | [TK]D-Fender | marc\cba: pastebin your setup. |
| 18:52.52 | [TK]D-Fender | ~pb |
| 18:53.03 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 18:53.03 | *** part/#asterisk RaYmAn-Bx (i=rayman@skumler.dk) |
| 18:53.16 | marc\cba | lol orly |
| 18:54.14 | marc\cba | fender - i will, but in your mind - should asterisk pass a callerid specified by a client/peer/phone? |
| 18:54.32 | [TK]D-Fender | marc\cba: And while you're at it, the CLI output of your failed attempt NoOp-ing CID info on both sides. |
| 18:54.39 | marc\cba | i would suspect there is a setting in sip.conf to enable this - as it is a bit of a security? risk |
| 18:54.59 | [TK]D-Fender | marc\cba: CID != auth |
| 18:55.08 | marc\cba | how do you mean? |
| 18:56.17 | [TK]D-Fender | marc\cba: taht was the simplest stateme I can make. CallerId is NOT auth info. |
| 18:56.37 | [TK]D-Fender | and my typing ability is fading by the second. |
| 18:57.36 | *** join/#asterisk FABN (n=teste@200-168-100-68.dsl.telesp.net.br) |
| 18:57.45 | FABN | Hi to all! |
| 18:58.08 | marc\cba | yes understood; but server b gets the correct sip packet to setup the call - if i cut B out of the equation, and go direct to the carrier, the CID carries from server A |
| 18:58.21 | marc\cba | simmilarly - if i originate the call, from B - the CID carries |
| 18:58.39 | FABN | It's my first time here! Does anybody could help me in configuration of 2 e1 in a TE407p?? |
| 18:58.48 | *** join/#asterisk matt_ (n=matt@2001:770:168:1:220:edff:feb4:7c9d) |
| 18:59.06 | hrmphh | how can you "give someone two lines" ... they are using a linksys 941 and logged in to the first extension with their own sip |
| 18:59.19 | hrmphh | can they put a call on hold and make calls on the second line if its not set up? |
| 18:59.31 | hrmphh | assuming no, but does it need a different sip login? |
| 18:59.38 | [TK]D-Fender | hrmphh: Ummm.. SUH. Set up the 2nd registration? :) |
| 18:59.54 | hrmphh | yeah |
| 18:59.56 | [TK]D-Fender | s/SUH/DUH/ |
| 18:59.58 | hrmphh | but the 2nd reg has to be diff right? |
| 19:00.16 | [TK]D-Fender | hrmphh: Line = identity. |
| 19:00.20 | hrmphh | s/has/stfujbotkthxbai:)/gi # meh! |
| 19:00.49 | hrmphh | ok so if line1 is "userAsip" line2 should be something like "userAsip2" |
| 19:00.53 | hrmphh | diff identity |
| 19:01.16 | hrmphh | can that 2nd registration be a shared line with a softphone? |
| 19:01.32 | [TK]D-Fender | hrmphh: use of one of those "light-up button thingiys on the right of your phones screen" represent you ability to place/receive a call against the identity (registration) to which it is allocated :) |
| 19:01.35 | hrmphh | i.e. line1 = main line, line2 = userAsip, userA is also logged in on their pc using softphone w/userA sip? |
| 19:02.02 | [TK]D-Fender | hrmphh: Do you actually WANT this phone to have distinct identities, or jsut have 1 identity handling multiple simultaneous calls? |
| 19:02.16 | hrmphh | actually want the former |
| 19:02.21 | hrmphh | distint identities |
| 19:02.27 | hrmphh | so its receptionist answering main line |
| 19:02.27 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:02.29 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 19:02.32 | hrmphh | but she also has her personal line on line#2 |
| 19:02.46 | hrmphh | and shes logged in on her pc softphone to her personal as well |
| 19:02.48 | hrmphh | in case she wants to use that |
| 19:03.03 | hrmphh | but from watching sip debug looks like asterisk is getting confused as that same reg is bouncing between ips |
| 19:03.52 | hrmphh | http://www.voip-info.org/wiki/view/SPA-941 <-- hmm looks like doesnt play well w/shared lines |
| 19:03.55 | [TK]D-Fender | hrmphh: You should be able to set up 2 regs on that phone just fine. |
| 19:04.39 | hrmphh | yes |
| 19:04.40 | hrmphh | i can do that |
| 19:04.41 | hrmphh | but |
| 19:04.43 | [TK]D-Fender | hrmphh: Forget the idea of shared lines. SLA support is not ready for that. |
| 19:04.52 | hrmphh | that 2nd reg cant also be logged in on the pc |
| 19:04.55 | hrmphh | so im abandoning that |
| 19:05.12 | [TK]D-Fender | hrmphh: that is *'s fault. And no, you can't be in 2 places at once. once the 941 is set up for the reg, no more soft-phone |
| 19:05.51 | hrmphh | ok |
| 19:06.11 | hrmphh | so heres what i want to do, make that just a main reception phone ... have 2 diff regs on it, but i only want one to ring if nobody is on either line |
| 19:06.14 | hrmphh | is that possible? |
| 19:06.20 | hrmphh | basically on incoming calls i ring ${ALLPHONES} |
| 19:06.23 | hrmphh | which would include both |
| 19:06.31 | hrmphh | however im assuming that will simultaneously ring both of th soe lines |
| 19:06.54 | [TK]D-Fender | hrmphh: Sure, thats all jsut dialplan work. |
| 19:07.09 | brad_mssw | think 1.6 added ${DEVSTATE(sip/whatever)} support that makes checking to see if a line is in use easier |
| 19:07.19 | brad_mssw | s/added/adds/ |
| 19:08.35 | hrmphh | yow 1.4.4 is out? |
| 19:08.37 | hrmphh | think im running 1.4.2 |
| 19:08.43 | hrmphh | some good bug fixes? |
| 19:08.58 | *** part/#asterisk Rienzilla (i=rien@sinas.rename-it.nl) |
| 19:10.05 | cnile | I have a Wildcard (crap card I know) X101P and the caller-id isnt working...any suggestions (other than try another card). |
| 19:10.36 | bkw_ | cnile, those aren't bad |
| 19:10.40 | bkw_ | they are just INTEL 537 modems |
| 19:10.45 | bkw_ | with two resistors popped off |
| 19:10.51 | bkw_ | and a heat sink put on the ambient chip |
| 19:10.56 | bkw_ | to hide the fact its really a modem |
| 19:11.51 | cnile | bkw_, I messed with asterisk years ago and you helped me a little bit, just picked it back up again |
| 19:12.12 | cnile | seems like you lived in Franklin, TN maybe? |
| 19:12.51 | *** join/#asterisk keulin (n=cray@nat30-2-88-160-17-233.fbx.proxad.net) |
| 19:13.04 | *** join/#asterisk scurb (n=scurb@c-25aae355.14-16-64736c13.cust.bredbandsbolaget.se) |
| 19:13.39 | Qwell[] | cnile: You may be thinking of Corydon that's in TN |
| 19:13.46 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 19:13.50 | cnile | I have usecallerid=yes |
| 19:13.55 | cnile | in zapata.conf |
| 19:14.13 | cnile | does Steven Critchfield hang in here? |
| 19:14.13 | Qwell[] | and it takes 2 rings before calls hit your IVR? |
| 19:14.25 | cnile | I think 1 ring |
| 19:14.33 | cnile | need to increase it? |
| 19:14.35 | *** join/#asterisk lokkju_wrk (n=lokkju@unaffiliated/lokkju) |
| 19:15.02 | cnile | well, actually, the inbound route just points to an extension, and that extension rings about 4 times before it goes to vm |
| 19:15.03 | Qwell[] | cnile: You just need to put in a Wait() before anything |
| 19:15.17 | Qwell[] | usually about 2 seconds does the trick |
| 19:15.22 | [TK]D-Fender | Qwell[]: ummm... nope :) |
| 19:15.40 | Qwell[] | [TK]D-Fender: eh? |
| 19:15.46 | [TK]D-Fender | Qwell[]: * will automatically wait if you tell it to use CID. You don't need to wait manually |
| 19:16.08 | Qwell[] | [TK]D-Fender: x100p :p |
| 19:16.08 | [TK]D-Fender | Qwell[]: "usercallerid=yes" , "callerid=asreceived" |
| 19:16.20 | [TK]D-Fender | Qwell[]: well aware of it :) |
| 19:16.21 | Qwell[] | lord knows what it'll decide to do |
| 19:16.31 | [TK]D-Fender | load chan_eightball.so! |
| 19:16.33 | Qwell[] | I guess the callerid=asreceived would help too :p |
| 19:16.43 | Qwell[] | s/lord/eightball/ |
| 19:16.43 | [TK]D-Fender | Qwell[]: :p |
| 19:16.44 | cnile | will try |
| 19:19.18 | cnile | Verbosity is at least 1 |
| 19:19.18 | cnile | recordingcheck|20070516-141845|1179343121.4: Inbound recording not enabled |
| 19:19.18 | cnile | dialparties.agi: Starting New Dialparties.agi |
| 19:19.18 | cnile | dialparties.agi: priority is 1 |
| 19:19.18 | cnile | dialparties.agi: Caller ID name is 'unknown' number is 'unknown' |
| 19:19.21 | *** join/#asterisk rva (n=rafael@200-206-132-201.dsl.telesp.net.br) |
| 19:19.36 | cnile | thats shows up after about 2 rings |
| 19:20.40 | rva | hi, could anyone helpme with channelRedirect? seems that after redirecting, the other lines from the dialplan are not ran |
| 19:24.26 | [TK]D-Fender | cnile: ... |
| 19:24.26 | [TK]D-Fender | ~freepbx |
| 19:24.41 | jbot | extra, extra, read all about it, freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 19:24.41 | cnile | yah, thats what I am using |
| 19:24.51 | gerphimum | whats the max verbosity you can go to ? |
| 19:24.53 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 19:25.40 | Qwell[] | gerphimum: 2 billion or so, technically |
| 19:25.54 | Qwell[] | gerphimum: realistically, most people use 3-4 |
| 19:25.59 | gerphimum | hmm. thats alot of v's ;) |
| 19:26.07 | Qwell[] | gerphimum: let bash do the work for you |
| 19:26.45 | gerphimum | well i just installed asterisk yesterday as a hobbyist thing and i like seeing what its doing |
| 19:27.11 | Qwell[] | asterisk -<esc>2000000000v |
| 19:27.25 | gerphimum | lol should i be scared of what i see |
| 19:27.55 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 19:27.58 | Katty | afternoon lovables. |
| 19:28.13 | Katty | eep. |
| 19:28.16 | Mercestes | :D |
| 19:29.50 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 19:30.23 | tzanger | hello ManxPower |
| 19:31.28 | *** join/#asterisk nomadsoul (n=nomadsou@unaffiliated/nomadsoul) |
| 19:33.28 | *** join/#asterisk clive- (n=pirch@dsl-242-139-161.telkomadsl.co.za) |
| 19:33.53 | *** join/#asterisk _deg_ (n=deg@200.195.161.164) |
| 19:34.12 | clive- | hi, does anyone here know of a good itfs supplier? |
| 19:34.24 | marc\cba | can anyone suggest how i might alter my dialplan, so that the callerid is not set explicity - but whatever i specifty it to be in user agent / phone ( x-lite for example )? |
| 19:34.30 | _deg_ | Is there anybody facing problems with sip.conf useragent parameter? |
| 19:34.38 | _deg_ | It is not working for me on 1.4.2 |
| 19:34.40 | Katty | Mercestes: i'm all sleepy inside :< |
| 19:34.43 | _deg_ | 1.4.* |
| 19:35.01 | Mercestes | marc\cba, Since it requires a line to set CallerID explicitly I would say...delete those lines and set callerid in sip.conf |
| 19:35.17 | Mercestes | Katty: I sorry. how can I wake you up inside? |
| 19:35.56 | marc\cba | merc- i don't want to set it in sip.conf - i want to specify the callerid on my phone, and for asterisk to respect this and pass it to my sip provider |
| 19:35.56 | Katty | i just wanna complain. |
| 19:36.03 | Mercestes | ah, okies. I shall listen then |
| 19:36.17 | tzanger | marc\cba: what I typically do is to have the internal callerID set in sip.conf |
| 19:36.30 | tzanger | and then any call leaving the location hits a macro which sets CID explicitly |
| 19:36.36 | Mercestes | marc\cba, you shall get no respect from asterisk. |
| 19:36.43 | marc\cba | hmm |
| 19:36.59 | marc\cba | can i get access to the callerid my phone / user-agent passes to asterisk in my dial plan as a variable? |
| 19:37.22 | tzanger | marc\cba: check sipd ebug to make sure asterisk is seeing it |
| 19:37.35 | _deg_ | marc\cba, there was a variable called USERAGENT |
| 19:37.59 | _deg_ | marc\cba, I think it has been removed on 1.4 |
| 19:38.05 | rva | could anyone help me setting a different conference room? |
| 19:38.15 | marc\cba | hmm - i was more after the callerid - over the useragent anyway deg |
| 19:38.26 | rva | i want to someone dial and all others ring....and goes to the conf\ |
| 19:38.26 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 19:38.32 | _deg_ | * \par Deprecated stuff |
| 19:38.32 | _deg_ | <PROTECTED> |
| 19:38.32 | _deg_ | <PROTECTED> |
| 19:38.33 | _deg_ | <PROTECTED> |
| 19:38.48 | _deg_ | from chan_sip.c |
| 19:39.04 | Katty | anyone ever heard of Fonality. |
| 19:39.08 | Katty | and if so...details please. |
| 19:39.22 | marc\cba | why, Katty? |
| 19:39.56 | clive- | hi OEj? |
| 19:40.06 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 19:40.09 | oej | evening |
| 19:40.14 | tuan_modulis | it's a company! |
| 19:40.15 | clive- | how you doing? |
| 19:40.24 | Katty | tuan_modulis: fancy that :P |
| 19:40.34 | [TK]D-Fender | Katty: A non-free * distro using an antiquated version of * and configured via its own GUI. |
| 19:40.58 | [TK]D-Fender | Katty: Basically "to be avoided" for you :) |
| 19:41.00 | Katty | hmm, i see. |
| 19:41.10 | Katty | i don't like antiquated. |
| 19:41.18 | Katty | i most definately like Shiny. |
| 19:41.21 | Katty | and New. |
| 19:41.29 | _deg_ | Guys... And what about the useragent parameter on sip.conf? |
| 19:41.40 | Katty | and lots of libri compiling errors too, apparently *sigh* |
| 19:41.41 | _deg_ | it is not working for me. Any trick to do that? |
| 19:41.44 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 19:42.35 | Katty | _deg_: if i were you, i'd try a lighter and a bat. |
| 19:42.35 | Katty | _deg_: and if that doesn't work, well... maybe some ice cream and a hug. |
| 19:42.46 | neverblue | Fender have you had experience with php+* ? |
| 19:42.58 | Mercestes | Katty, I'm antiquated, but a classic. do you still like me? :( |
| 19:43.03 | [TK]D-Fender | neverblue: I've done a little bit of AMI via PHP, yes. |
| 19:43.06 | Katty | Mercestes: of course, dear. |
| 19:43.10 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 19:43.42 | neverblue | i guess since its a UDP connection is the reason why there must be a two way communication for it to work? |
| 19:44.25 | neverblue | if I use fputs, I am finding my script will not work, until I add in an fgets (when doing the Action: Login |
| 19:45.02 | [TK]D-Fender | neverblue: No, AMI = TCP, not UDP. |
| 19:45.49 | neverblue | hmm |
| 19:46.00 | Katty | [TK]D-Fender: speaking of tcp and upd... |
| 19:46.06 | neverblue | wonder why i need to have an fgets() along with my fputs() then |
| 19:46.11 | Katty | [TK]D-Fender: how easy is video conferencing? is it based on meetme? |
| 19:46.14 | _deg_ | Katty, If I were you I try to put you finger in your hole. |
| 19:46.41 | [TK]D-Fender | Katty: Don't know about "conferencing it", but * can pass calls including video on just fine. |
| 19:47.09 | Katty | [TK]D-Fender: i might have to play with that sometime. |
| 19:47.28 | Katty | _deg_: I'm sorry, but you do not parse. |
| 19:47.29 | [TK]D-Fender | Katty: Just intall X-Lite on 2 PCS w/ webcams and away you go |
| 19:47.43 | Katty | [TK]D-Fender: eh, but that does allow for lots of users. |
| 19:47.51 | Katty | [TK]D-Fender: just thinkin ahead n all |
| 19:48.05 | Katty | [TK]D-Fender: multilocation conference calls and such |
| 19:48.09 | [TK]D-Fender | Katty: aren't you having enough trouble with NOW? ;) |
| 19:48.26 | Katty | [TK]D-Fender: minor detail ;) |
| 19:48.37 | SwK | katty |
| 19:48.39 | Katty | [TK]D-Fender: i'm having problems cause this is my second one. |
| 19:48.46 | Katty | [TK]D-Fender: i'm not all sorts of smrt like you. |
| 19:48.49 | Katty | SwK: hello dear. |
| 19:48.58 | SwK | hows it going |
| 19:49.03 | Katty | it's going (= |
| 19:49.03 | [TK]D-Fender | Katty: I am so smrt. S-M-R-T |
| 19:49.06 | Katty | hey `Sauron (= |
| 19:55.20 | *** join/#asterisk clive-- (n=pirch@dsl-242-139-161.telkomadsl.co.za) |
| 19:56.32 | Hmmhesays | Katty Katty Katty |
| 19:56.42 | Katty | matt matt matt |
| 19:56.55 | Hmmhesays | for about the 11th time, how are you? its been forever |
| 19:57.04 | Katty | oh, well. ya know. |
| 19:57.07 | Katty | same as last forever. |
| 19:57.13 | Hmmhesays | haha |
| 19:57.28 | Katty | i dyed my hair black. |
| 19:57.28 | Hmmhesays | yeah about the same here. gf just dumped me, band broke up |
| 19:57.54 | Katty | ouch. |
| 19:57.54 | Katty | about the band, anyway |
| 19:57.54 | Katty | there will be other girls ;) |
| 19:57.55 | Hmmhesays | and other bands ;) |
| 19:58.00 | Katty | true, true. |
| 19:58.07 | Katty | hard to put a good one together tho. |
| 19:58.10 | tuan_modulis | fender... ur name refers to the guitar brand? |
| 19:58.29 | Hmmhesays | black hair huh? I have blue black dye in my closet I was going to use tonight or tomorrow |
| 19:58.34 | Katty | Hmmhesays: finding a human metronome isn't exactly easy. |
| 19:58.52 | Qwell[] | Katty: drum machine |
| 19:58.54 | Qwell[] | problem solved |
| 19:58.56 | Hmmhesays | Katty: yeah probably the hardest to find, drums need space |
| 19:58.56 | Katty | Hmmhesays: and a bass player that doesn't want to over power the lead is also a pain. |
| 19:59.03 | [TK]D-Fender | Katty: Is that what the kids are calling it these days? ;) |
| 19:59.15 | Katty | [TK]D-Fender: oh you hush. |
| 19:59.31 | Hmmhesays | i just wanna be, wanna be lo-o-oved |
| 19:59.49 | [TK]D-Fender | tuan_modulis: No, its a gaming nick I've used since I started with CTF |
| 19:59.49 | Hmmhesays | the new papa roach cd brings the rock, like the rock should be brung |
| 20:01.10 | gerphimum | hey guys, whats required to register a sip phone thats on the internet ? ive forwarded port 5060 to the asterisk box but there must be more to it than that |
| 20:01.27 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 20:01.31 | Hmmhesays | if you have dynamic port mapping. or upnp you really don't have to forward any ports |
| 20:01.38 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 20:01.47 | Hmmhesays | I'm thinking about buying a schecter lady luck |
| 20:02.29 | *** join/#asterisk chefrs (n=joe@c-24-8-226-145.hsd1.co.comcast.net) |
| 20:02.47 | Hmmhesays | http://media.schecterguitars.com/imgs/hr_c1_ladyluck_rr.jpg |
| 20:02.53 | chefrs | Any idea why about 90% of my faxes (in/out) don't work? And when they do (in) chances are half the page is cutoff? |
| 20:03.06 | [TK]D-Fender | Hmmhesays: iBANEZ HAS A PRETTY NICE THROUGH-NECK 24 FRET GUITAR i WAS CONSIDERING PICKING UP OVER THE cARVIN CUSTOM JOB.... |
| 20:03.13 | gerphimum | Hmmhesays >> sweet lookin guitar, mang |
| 20:03.45 | Hmmhesays | [TK]D-Fender: cool |
| 20:03.48 | Hmmhesays | gerphimum: damn right |
| 20:04.29 | gerphimum | grr... i cant figure out why the hell ssh wont work :( |
| 20:04.29 | [TK]D-Fender | Hmmhesays: how much for this one? |
| 20:04.32 | bkw_ | Hmmhesays, http://upnp.sourceforge.net/ |
| 20:04.47 | bkw_ | brb |
| 20:05.33 | Hmmhesays | [TK]D-Fender: 450 |
| 20:05.36 | Hmmhesays | US |
| 20:05.39 | Hmmhesays | bkw_ cool |
| 20:05.57 | [TK]D-Fender | Hmmhesays: Pretty economical. |
| 20:06.10 | Hmmhesays | yeah not bad, its all schecter hardware, thats why the price is low |
| 20:06.23 | [TK]D-Fender | Hmmhesays: What do the reviews say? |
| 20:06.29 | gerphimum | can either of you try to register to my asterisk box to see if it works across the internet |
| 20:06.49 | Hmmhesays | not many out there, but the hardware is the same as a schecter c-1 |
| 20:07.35 | Hmmhesays | good multipurpose guitars with hot pickups |
| 20:08.57 | [TK]D-Fender | Hmmhesays: Right fret count/style? Like the fixed bridge? (typically good for sustain). Tried one personally? |
| 20:09.37 | Hmmhesays | right fret count? |
| 20:09.53 | [TK]D-Fender | Hmmhesays: My own style sort of demands a dual-locking system w/ floating trem & 24 fret |
| 20:10.08 | Hmmhesays | ahh, 24 jumbo frets |
| 20:10.11 | ghenry | What is the actual RTP server component(s) used by Asterisk? How do I replace the default RTP server in the standard Asterisk distribution with another one, whether I downloaded it or wrote it myself? |
| 20:10.32 | [TK]D-Fender | Hmmhesays : I play in E a LOT, so I use it from top to bottom :) |
| 20:10.49 | Hmmhesays | tonepros bridge, string through body for lots of sustain |
| 20:11.08 | *** join/#asterisk tuan_modulis (n=chatzill@3-82-252-216-static.enter-net.com) |
| 20:11.23 | [TK]D-Fender | Hmmhesays: I recently brought my old Ibanez EX-360 back into service, and being 22 fret I found myself doing full-tone bends to hit notes it didn't have :) |
| 20:11.44 | Hmmhesays | I don't ever play up that high |
| 20:11.59 | [TK]D-Fender | Hmmhesays: You are also much more of a rythm player than I am, so the fixed bridge wouldn't faze you much. |
| 20:12.19 | `Sauron | Hmm he said. |
| 20:12.35 | Hmmhesays | well I don't play a lot of 80's rock, but I play plenty of lead |
| 20:12.44 | karlhaines | what country code is 'cn' ? |
| 20:12.52 | [TK]D-Fender | karlhaines: China |
| 20:12.55 | *** join/#asterisk Paul_UK (n=foo@cpc1-acto7-0-0-cust402.brnt.cable.ntl.com) |
| 20:13.22 | *** join/#asterisk hijacked (i=vPps@cerebus.clandestineresearch.com) |
| 20:13.50 | karlhaines | i got bruteforced on my ssh from someone in china yesterday |
| 20:13.56 | karlhaines | they never got in of course |
| 20:14.02 | karlhaines | but they made 3000+ tries |
| 20:14.13 | Hmmhesays | secure passwords are good that way |
| 20:14.23 | karlhaines | Hmmhesays: certainly |
| 20:14.43 | Paul_UK | hey guys, as i understand it, there is no failover on the wanside ip with regards to using 2 internet connections. say for example, you have 86.53.113.25 with 1st provider and 192.168.0.100 with 2nd provider. for asterix to failover from 1 ip to another, a config file has to be changed and then reloaded.. am i right? there is no automatic failover? |
| 20:14.45 | xkev | I once admined a box w/ 500 users on it. 300 had password=username |
| 20:14.46 | Hmmhesays | you'd have to have one fscked up dictionary to have my passwords in it |
| 20:14.55 | xkev | romanians owned that in short order |
| 20:15.04 | xkev | also, you should hosts.allow |
| 20:15.26 | xkev | and disable ~user for apache, if active etc |
| 20:15.41 | xkev | and lockdown ftp client hosts etc |
| 20:15.59 | karlhaines | xkev: i didn't think of that, ~user would help them fish working login's huh? |
| 20:16.32 | xkev | if they can exploit php code etc, they can get kits in that way too |
| 20:17.05 | xkev | ..for example, this 300 user=pass box that got owned had no shells for the users |
| 20:17.22 | karlhaines | ahhh |
| 20:17.24 | xkev | ..they managed to drop in some junk with ftp, then execute it via http |
| 20:17.49 | xkev | I allowed users a personal webspace, but like 1 in 500 actually knew it was even there |
| 20:17.59 | Paul_UK | are we all running just one wan link with our asterisk boxes? |
| 20:18.42 | karlhaines | Paul_UK: they way to do a failover is have two seperate askterisk boxes |
| 20:19.14 | karlhaines | Paul_UK: one would be a secondary SIP |
| 20:21.07 | Paul_UK | having 2 seperate boxes isnt an issue. so where in the higher tier does the failover occur? |
| 20:23.05 | Hmmhesays | ok something is using my sound card and I can't figure out what |
| 20:23.48 | PioneerVM2 | anyone know if there is a way to have "subfiles" for extension.conf so i dont have to keep making it bigger? |
| 20:23.53 | PioneerVM2 | can i "include" other files? |
| 20:23.57 | Hmmhesays | yes |
| 20:23.58 | Hmmhesays | you can |
| 20:24.12 | PioneerVM2 | what is the format? |
| 20:24.19 | PioneerVM2 | (or command) |
| 20:24.34 | Strom_M | #include filename.conf |
| 20:24.40 | PioneerVM2 | thanks |
| 20:24.49 | Strom_M | whee |
| 20:25.04 | chefrs | Any idea why about 90% of my faxes (in/out) don't work? And when they do (in) chances are half the page is cutoff? |
| 20:25.20 | gerphimum | is there a way to include two contexts for sip channels ? |
| 20:25.31 | Paul_UK | karlhaines? |
| 20:26.04 | gerphimum | context=tech \n context=troyswanson |
| 20:26.18 | Hmmhesays | chefrs: sounds like a network problem |
| 20:26.34 | Strom_M | gerphimum: why would you want to? |
| 20:27.25 | gerphimum | you could have a set of things for techs, but also have a set of things for an individual user (stuff the rest of the techs cant do) |
| 20:27.25 | chefrs | Hmmhesays: Fax machine is on an FXS connection. |
| 20:27.42 | Hmmhesays | yeah definately networking issues |
| 20:27.52 | Hmmhesays | I get that also sometimes |
| 20:27.52 | Strom_M | gerphimum: yeah, but in that case you'd want separate sip accounts |
| 20:27.53 | chefrs | Can you explain a bit further? |
| 20:28.26 | Hmmhesays | packet loss on the network makes the faxes fail |
| 20:28.49 | chefrs | But it's an FXS, it doesn't use the network. |
| 20:29.05 | Hmmhesays | an fxs ata has to use some network somewhere |
| 20:29.14 | chefrs | T1 > * box > Fax machine. |
| 20:29.51 | *** join/#asterisk mihinomenest (i=DtZ5@cerebus.clandestineresearch.com) |
| 20:29.51 | gerphimum | Strom_M >> ok, another example - i'd have a default context in extensions.conf that handles stuff that any sip account can do (calling outbound, making calls to extensions, etc), but then other more user specific contexts |
| 20:30.40 | PioneerVM2 | Hmm can you give an example of #include |
| 20:30.46 | PioneerVM2 | it's not working for some reason |
| 20:31.02 | PioneerVM2 | I tried #include ext_re.conf and #include <ext_re.conf> |
| 20:33.04 | *** join/#asterisk deeperror (n=deeperro@mail.banctel.com) |
| 20:34.23 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 20:34.25 | |ryan| | anybody know about what a PRI T1 costs in california? |
| 20:34.34 | syzygyBSD | free if you steal it |
| 20:34.45 | Qwell[] | I don't think you can just steal a PRI |
| 20:34.46 | chefrs | syzygyBSD: Your telco could probably tell you |
| 20:35.05 | syzygyBSD | Qwell: think outside the box |
| 20:35.10 | PioneerVM2 | sip.conf refers to "voicepulse-in" which use to be in extension.conf but i now put it in ext_incoming.conf and #include that from extensions.conf |
| 20:35.19 | PioneerVM2 | and it keeps saying it does not exist |
| 20:35.21 | syzygyBSD | dirty internets causing problems the last week |
| 20:35.22 | PioneerVM2 | or it "can't find it" |
| 20:36.04 | bkruse | PioneerVM2: thats a total freepbx style extensions :[ |
| 20:36.06 | bkruse | i frown :[ |
| 20:36.23 | PioneerVM2 | i figured it out, sorry |
| 20:36.28 | bkruse | haha, its all good |
| 20:36.29 | PioneerVM2 | bad file permission on ext_incoming |
| 20:36.32 | bkruse | what was the problem? |
| 20:36.35 | bkruse | ahh |
| 20:36.42 | bkruse | your not running asterisk as root? |
| 20:36.43 | PioneerVM2 | and no warning from asterisk |
| 20:36.51 | PioneerVM2 | it has its own user, asterisk/asterisk |
| 20:38.26 | |ryan| | yes, stealing a PRI is kind of hard. |
| 20:39.01 | |ryan| | and I don't wanna call the telco, cause I'll have to talk to thier sales weasels. |
| 20:39.36 | |ryan| | who will have to 'look in to it and call me back' (thinly veiled attempt to get contact info) then never leave me alone! |
| 20:39.46 | |ryan| | they're scary :( |
| 20:43.24 | chefrs | Well at some point I imagine you'd have to talk to them to purchase one. |
| 20:44.28 | deeperror | ha yea |
| 20:45.46 | `Sean | FUCK DAMN |
| 20:45.49 | `Sean | asterisk is broke |
| 20:46.10 | `Sean | honestly, when you call numbers and they requyire caller input to lets say press option 2 for blah when you press 2 nothing happens |
| 20:46.16 | `Sean | the server dont send the msg's |
| 20:46.18 | *** join/#asterisk galeras (n=root@200.31.204.42) |
| 20:46.27 | deeperror | dtmf inband |
| 20:46.38 | `Sean | dtmfinband? |
| 20:46.43 | `Sean | doesn't it dtmf-outband? |
| 20:47.03 | high-rez | there's a configuration issue between your handset and asterisk. a sip handset? |
| 20:47.18 | chefrs | Who said they used callwave? |
| 20:48.06 | `Sean | high-rez no, a Anologe phone using TDM400p |
| 20:48.53 | `Sean | dtmfmode=rfc2833 |
| 20:48.55 | `Sean | ~dtmf |
| 20:49.07 | jbot | DTMF: Dual Tone Multi-Frequency. The technical term describing Touch Tone dialing. Basically the combining of two tones, one low frequency and one high frequency. |
| 20:49.33 | high-rez | Ok, that's using SIP, sean. |
| 20:49.59 | `Sean | so where do i add dtmf=inband |
| 20:50.55 | deeperror | sip.conf |
| 20:51.17 | high-rez | deeperror: He says he's not using a sip device though. |
| 20:51.30 | deeperror | oh |
| 20:51.31 | high-rez | Sean: If you're really not using a sip device, try relaxdtmf in iax.conf |
| 20:51.55 | `Sean | im using sip for calls not iax |
| 20:51.59 | `Sean | let me paste my sip.,conf |
| 20:52.07 | high-rez | Stop |
| 20:52.14 | high-rez | You said a moment ago you were using all analog. |
| 20:52.16 | deeperror | HAHA |
| 20:52.21 | high-rez | <`Sean> high-rez no, a Anologe phone using TDM400p |
| 20:52.38 | `Sean | i did |
| 20:52.46 | `Sean | but, you can use sip as a protocal..... |
| 20:52.53 | `Sean | it doesn't neccessarly need to be iax |
| 20:53.05 | high-rez | Dude, you are lost. |
| 20:53.09 | high-rez | Sip is not an analog protocol |
| 20:53.21 | deeperror | using sip to send the call to termination |
| 20:53.26 | `Sean | high-rez you are lost not me, i never said it was youre just having trouble comprehending.. |
| 20:53.31 | `Sean | yes deeperror |
| 20:53.33 | deeperror | analog phone connected to FXS somewhere? |
| 20:53.40 | `Sean | Yes |
| 20:53.56 | high-rez | Sean: I asked you specificall if you were using a sip handset. |
| 20:53.58 | deeperror | can you call your voicemail on phone and login with tones? |
| 20:54.06 | `Sean | deeperror yes |
| 20:54.19 | deeperror | pastebin sip.conf |
| 20:54.44 | galeras | Hi!, sometimes, after reload command, asterisk is crashing at at app_macro.c:311 (* 1.2.18), (please see gdb at http://www.pastebin.ca/491764 ). Any suggestion? |
| 20:55.50 | `Sean | http://pastebin.ca/491768 |
| 20:55.53 | `Sean | deeperror http://pastebin.ca/491768 |
| 20:56.04 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:56.21 | Paul_UK | so has anyone implemented wan failover? |
| 20:56.24 | `Sean | d-fender wusup |
| 20:56.51 | deeperror | Sean: try changing rfc2833 to inband |
| 20:56.57 | deeperror | test it |
| 20:57.07 | *** join/#asterisk nick125_lappy (n=nick@atarack/staff/nick125) |
| 20:58.32 | `Sean | Ahh that worked ;) thanks deeperror |
| 20:58.32 | *** join/#asterisk tecnico (n=tecnico@24.96.146.69) |
| 20:58.46 | deeperror | nice |
| 20:59.07 | `Sean | i guess i should update to 1.4 since its stable now |
| 20:59.32 | deeperror | could but there are some changes to be aware of |
| 20:59.34 | `Sean | nvm i will wait for 1.6 |
| 20:59.36 | deeperror | should look into that |
| 20:59.46 | `Sean | yea im reading now but i wont update |
| 20:59.49 | `Sean | 1.6 is around the corner |
| 21:01.20 | *** join/#asterisk CunningPike_ (n=CunningP@204.239.12.189) |
| 21:02.36 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 21:02.38 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 21:03.20 | *** join/#asterisk phillipk (n=pkey@fw.datafax.net) |
| 21:07.07 | syzygyBSD | that is like saying you will wait till vista becomes stable... |
| 21:07.32 | syzygyBSD | sure it will happen, but is it really worth the wait? |
| 21:08.05 | deeperror | have to wait to save the money for new hardware |
| 21:08.59 | syzygyBSD | get a soldering iron and build your own |
| 21:09.17 | deeperror | 5 years would give you plenty of time for that |
| 21:12.13 | *** join/#asterisk savaticus (i=savaticu@sta-206-168-96-69.rockynet.com) |
| 21:12.59 | nick125_lappy | syzygyBSD: Don't even joke like that |
| 21:13.19 | syzygyBSD | nick125_lappy: not part of the homebrew computer club? |
| 21:13.42 | nick125_lappy | syzygyBSD: No, about the previous comment about vista |
| 21:13.43 | savaticus | Hello Asterisk people ^^, I was wondering what the general feeling is on the readines of 1.4.x for use at an ITSP |
| 21:13.54 | nick125_lappy | bbl guys :p |
| 21:14.48 | savaticus | readines/readyness |
| 21:15.03 | Paul_UK | hry guys, another question. Has anyone implemented failover with asterix? i.e one server fails and another takes over? |
| 21:16.14 | deeperror | Paul have you looked into ser? |
| 21:16.18 | savaticus | doing instant failover with no call interuption would be a challange but I have two 1.2.17 boxes running with realtime that can place or terminate any call that the other can. |
| 21:16.34 | Sweeper | oi, oss softphone with legit g729? |
| 21:16.45 | Sweeper | paying for it is not a problem :3 |
| 21:16.51 | Paul_UK | deeperror what is ser? |
| 21:17.36 | Paul_UK | savaticus, so if one places a call, it turns off for some reason or loses connectivity with the network, the call can continue and the other can terminate it? |
| 21:17.56 | *** join/#asterisk a1fa (n=a1fa@unaffiliated/a1fa) |
| 21:18.12 | a1fa | hm.. is there a site where ip providers can bid for your business? |
| 21:18.27 | deeperror | sip exchange router google that openser |
| 21:21.18 | Hmmhesays | sip express router |
| 21:21.55 | deeperror | thanks hmmhesays |
| 21:22.13 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:22.56 | *** part/#asterisk deeperror (n=deeperro@mail.banctel.com) |
| 21:24.07 | Hmmhesays | i use openser for faxing |
| 21:24.49 | a1fa | does anybody know of a good sip provider in europe whos TOS allows re-sale? |
| 21:25.04 | Paul_UK | well alot of that has just gone over my head lol |
| 21:27.44 | *** part/#asterisk ms345 (n=mike_sim@64.74.198.10) |
| 21:28.06 | *** join/#asterisk sharp (i=sharp@gateway/tor/x-f5fe194aab45c914) |
| 21:30.10 | *** join/#asterisk hfb (n=hfb@pool-72-67-156-130.lsanca.dsl-w.verizon.net) |
| 21:31.03 | savaticus | no the call doesnt continue, but new calls can be placed. |
| 21:31.11 | a1fa | yoooo? |
| 21:32.06 | a1fa | fucking future market with phone minutes!!! |
| 21:34.51 | bkw_ | Hmmhesays, you don't use OpenSER for faxing.. its a sip proxy :P |
| 21:35.01 | bkw_ | you might route sip messages with OpenSER but you do not fax with it :P |
| 21:35.13 | *** join/#asterisk alrs (n=lars@pozug.com) |
| 21:36.46 | Paul_UK | savaticus, could you explain how you do that for me? in a nutshell? |
| 21:37.52 | savaticus | Well you just setup asterisk using the realtime extensions, and make sure your sip is in realtime |
| 21:38.09 | savaticus | that way any of your boxes can do sip |
| 21:38.40 | savaticus | for any client regardless of who it registered with |
| 21:39.08 | savaticus | loadbalancing between the boxes is up top you, I just use a lame DNS roundrobin |
| 21:39.43 | savaticus | the main stuff you need to know can be obtained from voip-info.org/wiki |
| 21:40.26 | savaticus | but this does not make the system fault proof, only fault tolerant |
| 21:40.39 | Paul_UK | well that gives me more than enough information, thanks very much for that |
| 21:40.40 | bkw_ | savaticus, that doesn't work if your clients are behind nat |
| 21:40.51 | savaticus | if a box goes down all calls that where on that box die |
| 21:41.17 | savaticus | My clients arent behind nats, and additionally, why? |
| 21:41.35 | bkw_ | because some NAT will not allow an invite to come from a box that the initial port setup was to |
| 21:41.41 | bkw_ | security issues |
| 21:41.47 | bkw_ | and no nat should allow that |
| 21:41.52 | savaticus | right |
| 21:41.56 | Qwell[] | should...heh |
| 21:41.58 | savaticus | thats why nat is a PITA |
| 21:42.01 | Qwell[] | sadly, many do :p |
| 21:42.03 | bkw_ | no NAT works great |
| 21:42.07 | bkw_ | its a good thing |
| 21:42.09 | bkw_ | nat isn't bad at all |
| 21:42.10 | savaticus | if you have alot of nAT clients use a VIP |
| 21:42.11 | bkw_ | I used to hate it |
| 21:42.17 | bkw_ | VPN? |
| 21:42.18 | afrosheen | bkw_, I want some of what you're smoking |
| 21:42.30 | bkw_ | afrosheen, you have to understand nat, sip and rtp |
| 21:42.38 | bkw_ | 99% of the world doesn't get it |
| 21:42.48 | afrosheen | yeah, mark spencer understood it well enough to build iax |
| 21:42.59 | bkw_ | IAX suffers the same issue |
| 21:43.06 | bkw_ | multiple clients behind the same nat |
| 21:43.19 | bkw_ | the problem just comes from a different angle |
| 21:43.30 | *** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net) |
| 21:43.41 | bkw_ | their are ways around it.. but its a bloody mess |
| 21:43.52 | bkw_ | their are tricks you can do with RTP to bust nat like a hot knife thru butter |
| 21:44.27 | bkw_ | busting double and triple nat is fun |
| 21:44.35 | afrosheen | yeah double nat is a pain in the ass with sip |
| 21:44.36 | bkw_ | the biggest problems are these so called "smart" routers that do SIP fixups for you |
| 21:44.47 | savaticus | Real Time Protocol, the protocol that controls the timing of the RTSP stream? |
| 21:44.54 | Qwell[] | bkw_: ugh |
| 21:45.00 | afrosheen | bkw_, hah..I agree with that, our Sonicwall breaks SIP with that stuff turned on, I've seen Cisco PIX do the same thing |
| 21:45.04 | Qwell[] | yeah, let's just go ahead and change the IP, and break the hash! |
| 21:45.17 | bkw_ | Qwell[], yep |
| 21:45.19 | bkw_ | its crazy |
| 21:45.29 | bkw_ | also I have seen two nat busting techniques break each other |
| 21:45.37 | Hmmhesays | haha |
| 21:45.38 | Hmmhesays | fun |
| 21:45.38 | bkw_ | as a result no media flows |
| 21:46.02 | bkw_ | if one end would just stop trying so damn hard it would just work |
| 21:46.04 | bkw_ | hehe |
| 21:46.21 | bkw_ | nat is a fact of life.. its not going away.. I thing everyone getting a better understanding is the only way to fix it |
| 21:46.27 | savaticus | whats that smell? |
| 21:46.31 | Hmmhesays | i need a good way in perl to parse this config file |
| 21:46.39 | afrosheen | bkw_, just wait for ipv6 to hit mainstream, nat will go away for awhile |
| 21:46.50 | Mercestes | afrosheen, have you lost your mind? |
| 21:46.56 | afrosheen | yes |
| 21:47.35 | bkw_ | l-fy did a good write up on NAT and voip on freshmeat |
| 21:47.37 | Qwell[] | I need to get me an ipv6 assignment |
| 21:47.50 | afrosheen | ipv6 gives a shitload more public IPs, which is half the reason NAT is used, because the world is running out of IP addresses, right? |
| 21:47.59 | Qwell[] | afrosheen: not really |
| 21:48.03 | bkw_ | afrosheen, the problem still exists |
| 21:48.12 | bkw_ | because they give out /32's like they are going out of style |
| 21:48.17 | bkw_ | so we'll end up in the same boat before long |
| 21:48.18 | Qwell[] | yeah, that's a bit silly |
| 21:48.34 | bkw_ | I mean really what do I need 96 bits of address space for? |
| 21:48.37 | afrosheen | hunted to extinction theory huh |
| 21:48.47 | Qwell[] | bkw_: VMs. Lots and lots of VMs |
| 21:48.54 | bkw_ | haha |
| 21:48.59 | bkw_ | spam.. lots of spam |
| 21:49.04 | Qwell[] | ugh, yeah |
| 21:49.21 | Qwell[] | all I used mine for, was reverse DNS |
| 21:49.24 | Mercestes | yea. As soon as "IPV6" comes out we'll be out of address space again because all the companies will jump on it |
| 21:49.32 | afrosheen | I find I learn more when I make purposefully idiotic statements |
| 21:49.43 | Mercestes | there wouldnt' be a *need* for IPV6 if it wasn't so damned easy to get entire blocks of public IPs. |
| 21:49.57 | bkw_ | afrosheen, knowing they are idiotic statements is one step in the right direction |
| 21:50.29 | rudholm | Mercestes: how easy is it? |
| 21:50.33 | afrosheen | bkw_, I don't really believe NAT is going away, but rather than poke around the issue of why won't ipv6 solve it, I just said what I said :) |
| 21:51.09 | rudholm | Mercestes: it might have been easy many years ago, but it hasn't been in a long time, not from ARIN, anyway (dunno about other regional NICs) |
| 21:51.17 | Mercestes | rudholm, Pretty easy from an infrastructure standpoint, if you configure your internal routing for internal IPs and just use one external IP (or a set of IPs) on your BGP/outside interface. |
| 21:51.40 | Qwell[] | ARIN doesn't give out perm assignments anymore either, does it? |
| 21:52.03 | Qwell[] | ...perm |
| 21:52.18 | rudholm | Mercestes: how does that make it easy to obtain large blocks of numbers? |
| 21:52.27 | Mercestes | we had 3 large blocks of numbers. |
| 21:52.38 | Mercestes | they are leased out by the bushel |
| 21:52.51 | Mercestes | phone numbers are a bigger issue than IPV4 |
| 21:52.55 | rudholm | last I checked, you had to make a case to ARIN |
| 21:53.12 | rudholm | I work for a really big internet company, we have about 6000 CIDR blocks |
| 21:53.30 | rudholm | it'd be great if we could get fewer huge blocks, but we can't. |
| 21:54.02 | afrosheen | Mercestes, you think so |
| 21:54.16 | afrosheen | Mercestes, all they do around here (North Texas) is keep creating new area codes |
| 21:54.35 | Mercestes | Yea, I know, I livein North texas but those new area codes aren't giong to last forever. |
| 21:54.44 | Mercestes | we've already gone from 7 digit to 10 digit dialing pretty much everywhere. |
| 21:55.03 | rudholm | 10-digit dialing doesn't evidence number exhaustion, though. |
| 21:55.07 | afrosheen | Mercestes, where abouts |
| 21:55.15 | rudholm | it's just convenient if you live in an area with a lot of area codes |
| 21:55.25 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 21:55.41 | Mercestes | afrosheen, n-htown, not n-texas. Sorry, I keep forgetting there is more Texas outside of Houston |
| 21:55.43 | rudholm | People in Alaska can do 11-digit dialing. |
| 21:55.57 | Mercestes | I can do 11 digit dialing |
| 21:56.02 | rudholm | that's my point |
| 21:56.06 | rudholm | everyone can |
| 21:56.06 | Mercestes | 1 NPA NXX XXXX |
| 21:56.18 | rudholm | it doesn't evidence numberspace exhaustion |
| 21:56.29 | rudholm | unless that wasn't your point |
| 21:56.36 | PioneerVM2 | anyone here do any AGI programming? |
| 21:56.37 | Mercestes | well of course not, we can go to 15 or 20 digit dialing.... |
| 21:56.46 | PioneerVM2 | i think i've hit some kind of glitch or bug |
| 21:56.50 | Mercestes | but given 10 digits only, yea, that's not goign to last too much longer |
| 21:57.05 | savaticus | but more likely its the code. |
| 21:57.17 | rudholm | eh, I'm not so sure |
| 21:57.27 | Mercestes | well, we'll see. :) |
| 21:57.32 | *** join/#asterisk hrmphh (i=patrick@notchill.com) |
| 21:57.51 | rudholm | we were burning through numbers fast for a while because CLECs needed a whole NXX in every rate center they operated in |
| 21:57.59 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 21:58.01 | hrmphh | can someone recommend a known-good motherboard to use with a 3fxo/1fxs asterisk system? will be box just for that purpose so please spec accordingly |
| 21:58.12 | rudholm | now the minimum assignments are only 1000 DNs rather than 10,000 DNs |
| 21:58.15 | rudholm | which helps a lot |
| 21:58.32 | Qwell[] | they gave out NX's? |
| 21:58.48 | rudholm | plus, some technologies could mitigate number consumption. I only need/want one DN, yet I have many (home, work, cell, etc, etc) |
| 21:59.04 | savaticus | I would use something lower end such as a Dell SC420 |
| 21:59.11 | Mercestes | I have abuot 4 #'s myself tho. Average is two numbers. |
| 21:59.15 | rudholm | Qwell[]: I'm referring to the fact that the minimum assignment used to be 10,000 DNs (a whole prefix) now it's one tenth of a prefix. |
| 21:59.23 | hrmphh | what about a dell precision workstation? |
| 21:59.24 | hrmphh | like a 420? |
| 21:59.28 | hrmphh | should that work well? |
| 21:59.34 | savaticus | should |
| 21:59.36 | Qwell[] | So, it was NXX, now it's NXX-X |
| 21:59.41 | rudholm | yes |
| 21:59.43 | savaticus | thats not a highload environment |
| 21:59.44 | Qwell[] | gotcha |
| 21:59.48 | Qwell[] | still pointless |
| 21:59.50 | hrmphh | ive got a tdm400p card now on a gigabyte ds3 mobo |
| 21:59.52 | hrmphh | and its awful |
| 22:00.00 | hrmphh | intermittent clicking, popping, etc. |
| 22:00.08 | savaticus | your biggest issue is transcode speed so a 800mhz+ FSB helps |
| 22:00.11 | rudholm | Qwell[]: that change is what made it possible to postpone the split/overlay of 310 here in LA |
| 22:01.24 | PioneerVM2 | any agi programmers here? |
| 22:01.26 | Qwell[] | that still hasn't happened? |
| 22:01.31 | rudholm | nope |
| 22:01.38 | *** join/#asterisk dr0ck (i=dr0ck@nat/digium/x-4d3be1f2440bfa9a) |
| 22:01.39 | savaticus | what language is your agi code in? |
| 22:01.45 | dan__t | 'afternoon. |
| 22:01.45 | PioneerVM2 | perl |
| 22:01.51 | PioneerVM2 | it may not just be a perl issue |
| 22:01.52 | rudholm | they've gone back to mandatory 11-digit dialing in 310 however (as a preliminary step) |
| 22:02.05 | Qwell[] | silly |
| 22:02.08 | rudholm | yes |
| 22:02.14 | hrmphh | what about an older system |
| 22:02.14 | hrmphh | like an ibm desktop thats p3-1000mhz |
| 22:02.14 | hrmphh | could that handle 4 lines? |
| 22:02.17 | savaticus | I have written a few agi scripts all in perl |
| 22:02.18 | *** part/#asterisk galeras (n=root@200.31.204.42) |
| 22:02.19 | PioneerVM2 | im using "record_file" |
| 22:02.30 | hrmphh | savat; you think? |
| 22:02.34 | rudholm | Qwell[]: the good news is that it's going to be an overlay, as it should be, rather than a geographic split. |
| 22:02.46 | Qwell[] | splain |
| 22:02.50 | savaticus | I think it could "handle it" but you WILL get echo etc |
| 22:02.53 | *** join/#asterisk CunningPike_ (n=CunningP@204.239.8.157) |
| 22:02.59 | Qwell[] | explain what both are, actually.. |
| 22:03.00 | hrmphh | got it |
| 22:03.03 | PioneerVM2 | if my AGI does this: festival "enter msg", record_file, festival "thank you", stream_file |
| 22:03.07 | hrmphh | unless i got a sangoma card w/hw ec? |
| 22:03.15 | PioneerVM2 | and I pass an incoming call directly to that script |
| 22:03.24 | savaticus | when you spec for n * box, think mediua server |
| 22:03.27 | hrmphh | im about to throw this pos digium card out the window |
| 22:03.28 | PioneerVM2 | the record_file part does not work, i get silence, its like the script cant hear it |
| 22:03.40 | rudholm | Qwell[]: geographic split is the standard way an area code is split. they divide the geographic area it covers into two, leaving the old code in one part and moving everyone in the other part to a new area code. |
| 22:03.50 | PioneerVM2 | if i do something in the extensions.conf like "background" or do a festival from there, then it works |
| 22:03.51 | Qwell[] | ahh |
| 22:04.00 | Qwell[] | and an overlay, is just new numbers for new people? |
| 22:04.10 | Qwell[] | or, numbers in the new areacode, rather |
| 22:04.11 | rudholm | Qwell[]: an overlay is where they simply start assigning new numbers in the new area code |
| 22:04.14 | rudholm | yeah |
| 22:04.16 | Qwell[] | gotcha |
| 22:04.27 | Qwell[] | yeah, the 951 split was retarded |
| 22:04.32 | rudholm | Qwell[]: the advantage is that people don't have their phone numbers taken away from them. |
| 22:04.39 | Qwell[] | (as was the 714 split way back when) |
| 22:04.50 | rudholm | as was the 213/323 split |
| 22:04.57 | Qwell[] | didn't live in LA at the time :D |
| 22:04.59 | rudholm | that was crying out to be an overlay |
| 22:05.04 | Qwell[] | actually, that was recent, wasn't it? |
| 22:05.06 | rudholm | yep |
| 22:05.09 | rudholm | relatively |
| 22:05.13 | Qwell[] | sameish time as 951? |
| 22:05.31 | rudholm | it's annoying, the phone number my parents had since the 1960s is now assigned to some downtown law firm. |
| 22:05.37 | Qwell[] | heh |
| 22:05.52 | rudholm | the phone companies like to say "your phone number isn't changing, just your area code." |
| 22:05.55 | Qwell[] | but, you know... |
| 22:05.55 | rudholm | which is entirely nonsensical |
| 22:05.59 | Qwell[] | the 951 split wasn't all bad |
| 22:06.09 | Qwell[] | we all know who true 909ers are anyhow ;) |
| 22:06.13 | rudholm | haha |
| 22:06.18 | Qwell[] | Fontucky can keep it! |
| 22:06.35 | rudholm | yeah, Riverside, Corona, Norco are 951 now |
| 22:06.52 | rudholm | Diamond Bar |
| 22:06.53 | rudholm | et |
| 22:06.53 | rudholm | c |
| 22:06.56 | PioneerVM2 | what is the suggested forum or mailing list for asterisk discussion |
| 22:07.05 | rudholm | this one |
| 22:07.15 | PioneerVM2 | somewhere that i can reach a larger group |
| 22:07.28 | rudholm | 320 people isn't enough? |
| 22:07.29 | PioneerVM2 | IE: to discuss a programming issue -- im limited here to who is on at the time |
| 22:07.33 | savaticus | there are forums on digiums website |
| 22:07.40 | Qwell[] | PioneerVM2: lists.digium.com |
| 22:07.42 | Qwell[] | http:// |
| 22:07.49 | PioneerVM2 | thank you |
| 22:07.50 | savaticus | also voip-info.org |
| 22:08.08 | savaticus | did you get my /msg? |
| 22:08.36 | savaticus | http://gundy.org/asterisk/agi.html is what I use as an agi ref |
| 22:13.00 | Corydon-w | Heh, looks just like my old reference |
| 22:13.43 | Corydon-w | http://asterisk.drunkcoder.com/agi.cgi |
| 22:13.46 | savaticus | I found it lin ked on the wiki |
| 22:14.01 | Corydon-w | except I haven't touched mine since 2003 |
| 22:16.01 | syzygyBSD | how can I kill tcpdump without kill or ctrl-c |
| 22:16.41 | syzygyBSD | ie I want to pipe the output and analyze it on the fly, but don't want to kill the anlyzer until after it prints out the results |
| 22:16.50 | savaticus | technically your phone number is only the last 4 digits, its (country) (area) (exchange) (number) so when the telco tells you that, they arent lying |
| 22:17.37 | syzygyBSD | savaticus: if you live in the US |
| 22:17.57 | syzygyBSD | or you don't own your own exchange... |
| 22:18.38 | carrar | Any suggestions on what is better? |
| 22:18.38 | carrar | ; exten => s,n,SetMusicOnHold(a20010724) |
| 22:18.38 | carrar | ; exten => s,n,Set(MUSICCLASS()=a20010724) |
| 22:18.38 | carrar | ; exten => s,n,Set(CHANNEL(musicclass)=a20010724) |
| 22:18.52 | carrar | last one doesn't seem to work |
| 22:18.52 | *** join/#asterisk nestAr (i=nester@makes.all.the.girlies.go.wewt.wewt.net) |
| 22:18.57 | savaticus | true that wa sa us specific exchange |
| 22:19.12 | savaticus | true that was a us specific example |
| 22:19.17 | nestAr | greetings, and long time no chat |
| 22:19.18 | Qwell[] | carrar: does the latter work? |
| 22:19.25 | carrar | last one does not work |
| 22:19.25 | Qwell[] | nm |
| 22:19.29 | carrar | heh |
| 22:19.30 | Qwell[] | I totally need to read the last line |
| 22:19.47 | Qwell[] | I guess Set(MUSICCLASS() |
| 22:19.56 | Qwell[] | check UPGRADE.txt |
| 22:19.56 | nestAr | anyone using this junky DIV-140 from D-Link? |
| 22:19.56 | carrar | 1st two seem to do the same thing, but not sure what the diff is |
| 22:20.03 | carrar | checking... |
| 22:20.11 | Qwell[] | the first is probably deprecated syntax |
| 22:20.21 | Qwell[] | ie; it's a function now, instead of an app |
| 22:20.42 | Mercestes | Are you guys ever going to decide on a syntax? |
| 22:21.03 | Qwell[] | Mercestes: no |
| 22:21.08 | carrar | yeah Qwell, decide already ;) |
| 22:21.09 | Mercestes | damnit |
| 22:21.24 | Qwell[] | well, Set* applications don't make sense |
| 22:21.35 | *** join/#asterisk Downchuck (n=downchuc@c-24-22-20-80.hsd1.or.comcast.net) |
| 22:21.39 | Mercestes | I was more referring to "core" that's aesthtic only as far as I can tell |
| 22:21.40 | Qwell[] | Set by itself is all that's really needed. the rest can be functions |
| 22:21.55 | Qwell[] | Mercestes: yeah...I'm gonna have to defer that one ;) |
| 22:22.05 | Mercestes | ;) |
| 22:22.35 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.169.195) |
| 22:22.50 | Mercestes | hehe |
| 22:23.08 | Mercestes | Well, just because you commit doesn't mean it was your idea, only that you submitted. |
| 22:23.55 | *** join/#asterisk ToyMan (n=Stuart@cpe-68-175-3-72.hvc.res.rr.com) |
| 22:24.43 | *** join/#asterisk hads (n=hads@reef80.anchor.net.au) |
| 22:25.08 | Mercestes | yea, I don't know where that came from, it doesn't even rhyme |
| 22:25.47 | Mercestes | I do submit that we should commit to Qwell being permitted to commit submissions |
| 22:25.57 | *** join/#asterisk data23 (i=data@92.b6.3845.static.theplanet.com) |
| 22:32.35 | *** join/#asterisk watchy (n=watchy@70.238.56.9) |
| 22:32.39 | watchy | yo i got a problem |
| 22:32.43 | *** join/#asterisk karlhaines (n=karl@unaffiliated/karlhaines) |
| 22:32.48 | Corydon-w | ~ask |
| 22:32.55 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 22:32.55 | watchy | how wold i tell if lightning blew up one of my TDM400s |
| 22:33.22 | Corydon-w | Does the card smell like ozone? Is it black and smoking? |
| 22:33.30 | watchy | 3 transformers exploed out at the .gov place that i got a * system, it was working fine, took phone lines down till 3:pm today |
| 22:33.37 | watchy | fones are up * is up, cant dialout |
| 22:34.06 | rudholm | do you get dialtone? |
| 22:34.59 | Corydon-w | It's highly unlikely that the computer survived but the card was killed |
| 22:35.00 | *** join/#asterisk KeltusEx (n=Keltus@c-76-21-119-166.hsd1.ca.comcast.net) |
| 22:35.00 | watchy | well it was working %100 fine yesterday |
| 22:35.00 | rudholm | I've seen my TDM400 do weird things, like disable a channel if it didn't like some electrical noise on the connected pair |
| 22:35.00 | watchy | now after all this shit blew up it dont dialout |
| 22:35.00 | rudholm | do you get dialtone? |
| 22:35.10 | watchy | the lines give me a dialtone |
| 22:35.13 | watchy | how do i tell if my tdm is getting a dialtone |
| 22:35.28 | rudholm | does the dialtone break if you send a dtmf tone? |
| 22:35.46 | Corydon-w | You might consider doing a reload on the drivers |
| 22:35.51 | rudholm | can you receive calls? |
| 22:35.53 | watchy | i rebooted the box |
| 22:35.59 | savaticus | ztcfg -vvv |
| 22:36.06 | Corydon-w | Okay, also try a cold reboot |
| 22:36.28 | watchy | Channel map: |
| 22:36.28 | watchy | Channel 01: FXS Kewlstart (Default) (Slaves: 01) |
| 22:36.28 | watchy | Channel 02: FXS Kewlstart (Default) (Slaves: 02) |
| 22:36.55 | savaticus | then in * do zap status, any alarms? |
| 22:37.14 | savaticus | also you can look for alarm states in zttool |
| 22:37.36 | watchy | nope no alarms |
| 22:37.38 | Qwell[] | savaticus: if the card isn't working properly anymore, I don't know that I'd trust zttool |
| 22:38.19 | watchy | it shows me dialin in |
| 22:38.21 | watchy | but it just rings forever |
| 22:38.57 | watchy | * aint seeing me dialin anymore |
| 22:38.59 | watchy | nothin |
| 22:39.02 | rudholm | what happens if you try to dial? |
| 22:39.08 | rudholm | does the dial tone stop? |
| 22:39.34 | Corydon-w | Try: Dial(Zap/1/w)... you should hear dialtone... |
| 22:39.37 | watchy | pbx*CLI> dial 98366866@default |
| 22:39.37 | watchy | <PROTECTED> |
| 22:39.37 | watchy | <PROTECTED> |
| 22:39.37 | watchy | <PROTECTED> |
| 22:39.37 | watchy | <PROTECTED> |
| 22:39.38 | watchy | <PROTECTED> |
| 22:39.49 | watchy | thats all i get |
| 22:40.15 | Corydon-w | If you don't hear dialtone with that, then try a cold reboot, to ensure the firmware is cleaned out |
| 22:40.27 | watchy | the box was off for 10 hours |
| 22:40.31 | watchy | 3 transformers exploded |
| 22:40.40 | watchy | very bad weather lastnight |
| 22:40.47 | Corydon-w | Then the daughtercard at the very least is probably dead |
| 22:40.48 | watchy | it took all day to repair it |
| 22:40.50 | savaticus | swap the card |
| 22:41.02 | watchy | im having someone go out there right now to swap the car |
| 22:41.03 | watchy | d |
| 22:41.07 | watchy | luckily i realised i had an extra |
| 22:41.25 | hrmphh | corydon; thats actually HIGHLY likely |
| 22:41.27 | Corydon-w | You might consider putting in a telephony surge protector on your line in |
| 22:41.27 | watchy | im ordering an 8 port sangoma fxo to replace these 2 digiums though |
| 22:41.28 | hrmphh | that the computer survived |
| 22:41.31 | hrmphh | and card was killed |
| 22:41.36 | hrmphh | you see lightning strikes all the time |
| 22:41.39 | hrmphh | that hit the copper wiring |
| 22:41.47 | hrmphh | and fry whats connected to it directly, but not the rest of the machine |
| 22:41.50 | watchy | ive seen dead modems in pcs alot with the pc surving |
| 22:41.57 | hrmphh | ive seen lightning strikes at hospitals take out 30+ NICs |
| 22:42.05 | hrmphh | and leave all but 1 or 2 computers without issues |
| 22:42.14 | JT | watchy: can't you get a T1? |
| 22:42.28 | Corydon-w | It's unfortunately difficult to predict, though |
| 22:42.30 | watchy | jt'; they only use 8 lines im not sure if thats practical |
| 22:42.43 | Corydon-w | surge suppressors do a nice job of protecting equipment, though |
| 22:42.46 | JT | you don't have to get a direct lightning hit to get damaged |
| 22:42.49 | Qwell[] | watchy: some places, it ends up being more cost effective to go with T1 over 8 analog lines |
| 22:42.55 | Qwell[] | not always though, of course... |
| 22:42.59 | JT | watchy: usually 8 channels is the fractional minimum |
| 22:43.06 | hrmphh | actual legit surpressers |
| 22:43.15 | hrmphh | that are in line with the blocks |
| 22:43.17 | hrmphh | are uberexpensive |
| 22:43.28 | watchy | jt: ah i wonder if it would be cheaper |
| 22:44.04 | JT | watchy: it's definitely better, sometimes cheaper, depends if you live in the sticks |
| 22:44.05 | JT | digital signalling is far superior |
| 22:44.29 | watchy | yea i imagine |
| 22:44.33 | JT | there are 3 types of currents induced by a lightning strike: 1. direct hit, 2. capacitive coupling, 3. inductive coupling |
| 22:44.53 | JT | usually it's not a direct hit that affects phone lines |
| 22:45.12 | JT | capacitive coupling affects underground mainly |
| 22:45.30 | SwK | altho direct hits can be pretty spectacular |
| 22:45.38 | Qwell[] | if you survive them ;) |
| 22:45.43 | JT | as the potential of the surrounding earth increases, and cause a capacitive effect to increase the charge in a cable |
| 22:45.44 | rudholm | JT: where'd you get that little list? (I'm just curious because of the wording they use) |
| 22:45.54 | JT | rudholm: out of my head |
| 22:45.57 | rudholm | oh |
| 22:46.02 | rudholm | :) |
| 22:46.04 | *** join/#asterisk cnbrk (n=canberk@212.156.213.131) |
| 22:46.06 | cnbrk | hi |
| 22:46.13 | cnbrk | how can i transfer a call to another asterisk user |
| 22:46.18 | rudholm | I was gonna say, that it'd be more correct to say a potential is induced, not a current. |
| 22:46.20 | JT | might not be the perfect terms :P |
| 22:46.24 | JT | heh |
| 22:46.36 | rudholm | but one is a function of the other, so it's really semantics |
| 22:47.23 | JT | watchy: t1 over fibre is the best lightning protection ;) |
| 22:47.32 | mosty | cnbrk: yes, use the transfer button on your phone if it has one, or set up attended or blind transfer in features.conf |
| 22:47.47 | watchy | jt: haha i got 15mbit inet over fiber at my NOC |
| 22:47.53 | rudholm | JT: that reminds me of my ground hum solution --I switched from coax S/PDIF to fiber :) |
| 22:47.55 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 22:47.55 | watchy | but theres no way they will run fiber to this place |
| 22:48.02 | JT | rudholm: nice |
| 22:48.07 | watchy | this company is in the middle of no where because they make explosives |
| 22:48.12 | watchy | big explosives for the .gov |
| 22:48.19 | watchy | so if they blow up it wont kill anyone but the workers |
| 22:48.21 | JT | rudholm: does that actually induce a hum, or errors, seeing as it's digital? |
| 22:49.03 | rudholm | it wasn't a hum in the bitstream, it was affecting the analog stages of the amp |
| 22:49.23 | JT | ah |
| 22:49.36 | rudholm | my house was built in 1961, before every outlet had to have ground, so I get weird ground offsets between my gear |
| 22:49.59 | JT | funny american regulations ;) |
| 22:50.07 | afrosheen | rudholm, that's the tough thing about old houses, you know they're rock solid compared to the junk builders make today, but they're wired like crap |
| 22:50.35 | savaticus | lol like when you find a place with tin wire |
| 22:50.36 | rudholm | yeah, mine is kind of the worst of both worlds because it was built after they stopped using conduit but before they started using ground |
| 22:50.43 | rudholm | tin, or aluminum |
| 22:50.44 | cnbrk | what is an attended transfer and blind transfer |
| 22:50.45 | JT | they didn't wire them that badly in this country |
| 22:50.53 | rudholm | JT: which country is that? |
| 22:50.58 | JT | australia |
| 22:51.02 | rudholm | ah |
| 22:51.03 | JT | they've always had to be grounded |
| 22:51.04 | afrosheen | cnbrk, attended means you as a person babysit the transfer, wait for someone to pick up and hit transfer again |
| 22:51.16 | cnbrk | i see |
| 22:51.16 | afrosheen | cnbrk, blind means you just send the call to the recipient and forget about it |
| 22:51.30 | cnbrk | if nobody picks up what happens? |
| 22:51.30 | afrosheen | blind is usually preferable |
| 22:51.43 | JT | preferable in what sense? |
| 22:51.48 | afrosheen | cnbrk, if nobody picks up then your dialplan kicks in and dumps them to that extension's voicemail or whatever |
| 22:51.54 | JT | certainly not from a customer service point of view |
| 22:52.00 | JT | from a lazyness sense, yes |
| 22:52.04 | cnbrk | :) |
| 22:52.26 | afrosheen | jt: depends on if you have really busy employees and a misdial..or no secretary, etc. |
| 22:52.29 | watchy | jt: id like going digital/t1 whaever, but i use at&t so it wold take 5 weeks to get a line put in |
| 22:52.30 | cnbrk | for ex. can i use the # button of my phone to transfer the call? |
| 22:52.37 | watchy | jt: they need the phone system asap |
| 22:52.44 | afrosheen | cnbrk, maybe, depends on how you set your call features and dialplan |
| 22:52.50 | JT | watchy: do they have Internet? |
| 22:52.59 | cnbrk | hm okay, now it makes sense |
| 22:53.07 | watchy | jt: yea through me, im thier WISP |
| 22:53.08 | watchy | heh |
| 22:53.08 | afrosheen | cnbrk, most of the time people set a key combination or a hotkey up for that, like *59 or whatever |
| 22:53.13 | JT | ah |
| 22:53.19 | watchy | i supply them 6mbit |
| 22:53.34 | watchy | you thinkin i should brigde them over a voip supplier for now? |
| 22:53.43 | JT | not sure |
| 22:53.43 | watchy | for outgoing it would work, incoming it wouldnt though |
| 22:53.45 | cnbrk | i need to google a lil bit for the config. of extensions.conf |
| 22:53.46 | afrosheen | watchy, pstn card for now |
| 22:53.50 | JT | t1 might not even be available there though |
| 22:53.59 | watchy | jt: it is for like a billion amonth |
| 22:54.00 | JT | you could probably get the numbers redirected |
| 22:54.08 | watchy | thats why i supply them wireless internet nd they dont got a T1 |
| 22:54.19 | watchy | because of the insane cost |
| 22:54.23 | JT | right |
| 22:54.34 | afrosheen | watchy, I've seen some nice microwave wireless T1's and above before that would easily support voice |
| 22:54.41 | JT | t1s use less infrastructure than 8 lines |
| 22:54.48 | afrosheen | jt: s /stupid/greedy |
| 22:54.51 | watchy | my wireless should support voice |
| 22:55.00 | *** join/#asterisk kukhuvud (n=io@65.39.69.38) |
| 22:55.05 | JT | afrosheen: well they save valuable copper going to t1 |
| 22:55.08 | afrosheen | watchy, wireless is usually pretty flaky |
| 22:55.20 | cnbrk | this is the best helping group i've ever seen |
| 22:55.34 | afrosheen | cnbrk, you should have been here 4 hours ago...a ghost town |
| 22:55.40 | cnbrk | :) really |
| 22:55.58 | watchy | kukhuvud is a thier he stole me heart |
| 22:56.01 | cnbrk | maybe because of lunchtime |
| 22:56.28 | watchy | afro: our wireless is pretty decent were using 5.7ghz |
| 22:57.55 | afrosheen | watchy, microwave towers and whatnot right? |
| 22:58.05 | watchy | nah it aint microwave i dont think |
| 22:58.21 | kukhuvud | canopy? |
| 22:58.28 | afrosheen | watchy, well as long as it doesn't drop too many packets and the latency stays down it should work for voip |
| 22:58.30 | watchy | our backend is alvarion 5.7 |
| 22:58.31 | cnbrk | here is an interesting question for me, let's say there is a 20ms ping time between two points, what will be the latency of the calls? |
| 22:58.40 | cnbrk | 20ms or 20+20=40? |
| 22:58.44 | *** join/#asterisk grndslm (n=grndslm@host-64-139-246-34.nctv.com) |
| 22:58.44 | watchy | afro: im getting 5ms to my router which is 20miles away |
| 22:58.46 | Qwell[] | cnbrk: neither |
| 22:58.47 | afrosheen | cnbrk, 40 |
| 22:58.55 | afrosheen | watchy, impressive |
| 22:58.56 | Qwell[] | icmp != udp |
| 22:58.56 | cnbrk | hm |
| 22:59.05 | Qwell[] | icmp has a *VERY* low qos |
| 22:59.09 | watchy | brb my asterisk box is coming back up |
| 22:59.10 | afrosheen | but again, packet loss is the concern |
| 22:59.18 | watchy | with a new tdm |
| 22:59.24 | cnbrk | no packet loss i think since this is a private line |
| 22:59.26 | watchy | maybe this will give this company something temporary |
| 22:59.29 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 22:59.34 | Qwell[] | cnbrk: you can always lose packets |
| 22:59.34 | afrosheen | cnbrk, with asterisk, you really don't start worrying about latency until you get around 200ms and higher |
| 23:00.06 | *** join/#asterisk NovceGuru (n=asdf@oh-71-50-248-25.dhcp.embarqhsd.net) |
| 23:00.08 | cnbrk | up to how many milliseconds a human being will not understand the latency? |
| 23:00.16 | afrosheen | haha, that was my next answer |
| 23:00.20 | cnbrk | :) |
| 23:00.25 | afrosheen | 500ms is probably intolerable |
| 23:00.31 | afrosheen | like an overseas satellite call |
| 23:00.35 | cnbrk | yeah definetely |
| 23:00.36 | Qwell[] | cnbrk: "understand" isn't the right word.. you want "perceive" |
| 23:00.53 | cnbrk | english is not my mother tongue sorry :) |
| 23:00.55 | Qwell[] | but, 400ms round-trip is the "acceptable" standard, I believe |
| 23:01.20 | Qwell[] | and really, what you'd be most worried about is jitter |
| 23:01.33 | cnbrk | what does jitter do in this case |
| 23:01.42 | cnbrk | what if i don't turn the jitter on? |
| 23:01.47 | savaticus | I think the latency that is acceptable relies on your codec somewhat as well. |
| 23:01.50 | afrosheen | jitter compensates for packet loss right? |
| 23:01.54 | *** join/#asterisk waltz (n=walterbr@bas1-toronto44-1177949638.dsl.bell.ca) |
| 23:01.58 | cnbrk | yeah as i learned today |
| 23:01.58 | Qwell[] | jitter is a bad thing |
| 23:02.07 | cnbrk | it increases ping time rite> |
| 23:02.24 | Qwell[] | jitter means that you have varying times for packets to arrive |
| 23:02.34 | Qwell[] | like, one is 10ms, the next is 40ms, then 20ms, then 100ms |
| 23:02.36 | mosty | jitterbuffer helps with voice packets that arrive out of order |
| 23:02.41 | Qwell[] | that is very bad for audio |
| 23:02.50 | cnbrk | i see |
| 23:02.59 | savaticus | http://en.wikipedia.org/wiki/Delay_jitter |
| 23:03.01 | cnbrk | what do you think an ordinary pstn line's latency? |
| 23:03.02 | NovceGuru | I wanted to verify that this is possible. Would I be able to take 2 ATA's and basically make an "analogue extension" over the internet? I.E., have a line coming from the telco into the ATA, having it go through the internet to the other ATA, and ring on an analogue phone on the other end |
| 23:03.03 | russellb | not just out of order, but at varying amounts of delay |
| 23:03.25 | mosty | yes |
| 23:03.48 | cnbrk | NovceGuru i think you'll need a fxo voip gateway in your case |
| 23:03.48 | mosty | NovceGuru: yes |
| 23:04.12 | waltz | an ordinary PSTN line can do about 6000 mi in 1s |
| 23:04.16 | mosty | NovceGuru: er well you need an internet connection in there somewhere of course |
| 23:04.37 | NovceGuru | of course :) |
| 23:04.37 | Qwell[] | afrosheen: You're thinking plc - packet loss concealment |
| 23:04.45 | JT | watchy: 5.7GHz is most definitely microwave. |
| 23:04.59 | NovceGuru | I was thinking about the linksys SPA3102 |
| 23:05.23 | JT | waltz: what do you mean? |
| 23:05.48 | waltz | JT: I was answering cnbrk's question (re: PSTN latency) |
| 23:06.31 | NovceGuru | the SPA3102 has "Hot line and Warm Line Calling" which I believe is what I need |
| 23:06.47 | JT | cnbrk: ping time is return trip time.... time to get to destination, time for destination to process, as well as time for it to get back, and processing on your end |
| 23:07.13 | JT | waltz: what pstn lines are 6000mi long? |
| 23:07.41 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 23:08.15 | waltz | JT: ? |
| 23:08.36 | waltz | JT: I'm talking about going through exchanges and across the oceans |
| 23:08.41 | JT | waltz: you said pstn lines 6000mi long have 1s latency |
| 23:08.51 | JT | i have no idea what gave you that idea |
| 23:08.52 | waltz | I know what I said |
| 23:09.04 | JT | waltz: that does not take 1 second |
| 23:09.13 | waltz | oh, perhaps you know better? |
| 23:09.54 | JT | 6000mi is about 31.25 msec each direction |
| 23:10.08 | JT | 0.03125 seconds |
| 23:10.14 | waltz | well then |
| 23:10.42 | waltz | guess I was wrong |
| 23:10.48 | swyrus | why can i dial extensions from outsise my lan but not use my voip providers in asterisk? |
| 23:10.57 | JT | 2 way geostationary satellite is the only thing that gives you over half a second latency point to point |
| 23:11.23 | watchy | oh my fucking god |
| 23:11.25 | watchy | i think its sbc |
| 23:11.27 | JT | as geostationary satellites orbit 35000km (3 times diameter of Earth) above the equator |
| 23:11.34 | watchy | im going to kill these sons of bitches |
| 23:11.47 | *** join/#asterisk sandorp (n=sandor@firewall2.wsi.net) |
| 23:11.50 | NovceGuru | do it |
| 23:11.58 | waltz | okay |
| 23:12.02 | nick125_lappy | waltz: take pictures |
| 23:12.03 | nick125_lappy | and videos |
| 23:12.07 | nick125_lappy | and put it on youtube |
| 23:12.11 | nick125_lappy | :-) |
| 23:12.20 | waltz | stop confusing me with watchy for chrissakes |
| 23:12.28 | waltz | ;< |
| 23:12.32 | nick125_lappy | I hate tab complete |
| 23:12.35 | nick125_lappy | Really, I do. |
| 23:12.40 | savaticus | see yall laters |
| 23:12.47 | *** join/#asterisk Vec2 (n=Vec@dsl-241-215-86.telkomadsl.co.za) |
| 23:12.53 | nick125_lappy | s/waltz/watchy |
| 23:13.06 | Qwell[] | s/<tab>/<tab><tab>/ |
| 23:13.16 | sandorp | I just installed a TDM808 card and compiled the very latest source but I can't get asterisk to start ... it complained a number of add-ons were not compiled for the current version when I did a "make install" ... how do I fix that? |
| 23:13.17 | Downchuck | asterisk refuses to register SIP when I have bindaddr = 127.0.0.1 instead of 0.0.0.0 |
| 23:13.30 | Downchuck | Any ideas what i'm doing wrong? openser is on the public 5060 |
| 23:13.33 | mosty | i like partial tab completion, not first match tab completion |
| 23:13.39 | afrosheen | Qwell[], I see |
| 23:14.05 | watchy | holy shit |
| 23:14.11 | watchy | my tech arrived on site of my ofne system |
| 23:14.17 | watchy | he said my TDMs had no capacitors |
| 23:14.21 | watchy | they poped off |
| 23:14.28 | Qwell[] | nice |
| 23:14.29 | waltz | wtg |
| 23:14.36 | watchy | i guess that tells me they are dead |
| 23:14.42 | Qwell[] | I'd think so |
| 23:14.56 | watchy | how depressing |
| 23:15.27 | watchy | and i live in arkansas so its impossible to gets cards by tommorow |
| 23:15.31 | JT | you should probably use a surge protector on the lines at a site like that |
| 23:15.34 | watchy | this company is fucked |
| 23:15.54 | JT | i bet the lines are long |
| 23:15.56 | afrosheen | watchy, yep, you're right |
| 23:16.09 | watchy | well they will survive |
| 23:16.12 | watchy | they have no choice |
| 23:16.14 | afrosheen | watchy, where did you get those TDMs from |
| 23:16.20 | watchy | voipsupply |
| 23:16.29 | afrosheen | watchy, yeah but are they Digium TDMs |
| 23:16.38 | watchy | yea |
| 23:16.41 | Qwell[] | afrosheen: I don't think lightning damage is covered.. if that's what you're implying |
| 23:16.42 | watchy | will digium replace them |
| 23:16.57 | afrosheen | I think Digium tdms are junk, we had 3 out of 8 fxo modules die on 2 cards |
| 23:17.02 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 23:17.07 | afrosheen | personal experience of course |
| 23:17.10 | watchy | im not to happy with these TDMs personally |
| 23:17.28 | watchy | lots of issues out at this client, but who knows if its the tdms |
| 23:18.10 | mosty | watchy: sangoma card are better, imo |
| 23:18.39 | watchy | ive thought about replacing these tdms with A200Ds |
| 23:18.40 | watchy | but i dunno |
| 23:18.41 | *** join/#asterisk linagee (n=linagee@unaffiliated/linagee) |
| 23:18.50 | afrosheen | watchy, or a handful of iaxys |
| 23:18.52 | *** part/#asterisk hads (n=hads@reef80.anchor.net.au) |
| 23:18.58 | NovceGuru | So would anybody here happen to be an NEC hardware specialists? I'll be needing some config done on an DS1000 :\ |
| 23:19.03 | afrosheen | watchy, depending on if you need an fxo or an fxs... |
| 23:19.14 | watchy | afro: just 8 fxos |
| 23:20.08 | JT | watchy: a 8 FXOs, you start getting into channel bank territory |
| 23:20.24 | watchy | jt: they shouldnt really need anymore then 8 |
| 23:20.32 | watchy | they might goto 12 but that would be max |
| 23:20.39 | afrosheen | 8 fxos, it's time for a single Rhino card with the dual modules |
| 23:20.54 | watchy | afro: whats that |
| 23:21.06 | JT | what drivers do rhino use? |
| 23:21.07 | mosty | sangoma have a card that will do 12 fxo/fxs on a single board |
| 23:21.24 | JT | mosty: which card? |
| 23:21.27 | afrosheen | http://www.rhinoequipment.com/R8FXXcard.html |
| 23:21.40 | *** join/#asterisk qdk (n=qdk@85.9.7.139) |
| 23:21.53 | watchy | are those cards any good afro |
| 23:22.04 | afrosheen | I can't vouch for them personally but they look decent |
| 23:22.15 | mosty | jt: a400 series i think |
| 23:24.22 | JT | yeah you're right |
| 23:24.22 | JT | i was getting confused with a200 |
| 23:24.22 | watchy | they look nice |
| 23:24.22 | afrosheen | the module connectors on the boards look better than the crap digium uses |
| 23:24.38 | mosty | jt: and they're available for pci-express too |
| 23:24.46 | JT | but i don't think many people use rhino.... |
| 23:24.48 | JT | mosty: indeed |
| 23:25.21 | mosty | unfortunately they are not yet authorised for australian use yet :( |
| 23:25.28 | JT | correct |
| 23:25.44 | afrosheen | I think most people use their channel banks |
| 23:26.26 | cnbrk | what do you guys think about asterisknow and trixbox |
| 23:26.43 | JT | ~trixbox |
| 23:26.54 | jbot | Trixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums & WIKI at http://www.trixbox.org |
| 23:27.25 | afrosheen | trixbox is good if you want to get your feet wet |
| 23:27.30 | cnbrk | ~asterisknow |
| 23:27.38 | Qwell[] | wet from LAVA |
| 23:27.38 | afrosheen | i.e. no previous asterisk experience |
| 23:27.44 | cnbrk | i see |
| 23:27.47 | cnbrk | which one is better |
| 23:27.50 | cnbrk | asterisknow or trixbox |
| 23:27.51 | Qwell[] | asterisknow |
| 23:27.59 | cnbrk | it is still beta |
| 23:28.02 | Qwell[] | #include <stdbias.h> |
| 23:28.03 | JT | even better is to not use either |
| 23:28.07 | JT | and read the book |
| 23:28.23 | afrosheen | lol @ Qwell[] |
| 23:28.25 | cnbrk | will asterisknow do everything that an asterisk installation do? |
| 23:28.33 | JT | no |
| 23:28.36 | JT | no gui will |
| 23:28.47 | afrosheen | cnbrk, here's the deal, and pay close attention |
| 23:28.50 | watchy | man my customer cant goto hell |
| 23:28.51 | cnbrk | doesn't it have a cli |
| 23:28.55 | watchy | its almost time to go play poker |
| 23:28.59 | afrosheen | cnbrk, any gui-powered asterisk will get you up and running quick |
| 23:29.02 | watchy | poker is more important |
| 23:29.04 | JT | they make the easy things sort of easy, and the hard things hard |
| 23:29.16 | afrosheen | cnbrk, but in the end it'll limit you and you'll get frustrated as you try to do more complicated things |
| 23:29.18 | watchy | i want to kiss a puppy |
| 23:29.37 | afrosheen | cnbrk, then you'll come in to #asterisk and get flamed |
| 23:29.42 | cnbrk | i see |
| 23:30.00 | cnbrk | for ex, can i upload a mp3 and make asterisknow play it on moh? |
| 23:30.02 | afrosheen | cnbrk, so, I reiterate, gui asterisk stuff is cool to get a feel for it, i.e. is asterisk right for me/my business |
| 23:30.22 | afrosheen | but in the end you will write dialplans by hand |
| 23:31.12 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 23:31.25 | cnbrk | can i switch to cli on asterisknow? |
| 23:31.38 | afrosheen | Qwell[], wanna field this one |
| 23:32.01 | pigpen | Hi all, question about faxdetect.... I would like it to detect and redirect faxes before it rings the extension....currently it is doing it afterwards.. |
| 23:32.01 | Qwell[] | well, the asterisk-gui was designed to allow that |
| 23:32.04 | pigpen | ideas |
| 23:32.11 | Qwell[] | unlike certain *cough*other*cough* GUIs |
| 23:32.20 | JT | pigpen: you need an ivr then |
| 23:32.32 | JT | pigpen: it's impossible to determine if a call is a fax without answering it |
| 23:32.42 | pigpen | JT yeah..I kinda figured... |
| 23:33.23 | pigpen | k, so I need to answer it before it hits the ivr. |
| 23:33.23 | afrosheen | pigpen, unless you know the specific fax numbers you will always receive from, like in a bank or something |
| 23:34.01 | afrosheen | pigpen, yep |
| 23:34.17 | pigpen | k, I knew it was something stupid. |
| 23:37.33 | pigpen | much better....tks. |
| 23:38.03 | afrosheen | pigpen, I wondered why freepbx had that as a setting :) |
| 23:38.40 | pigpen | ha.. |
| 23:39.02 | pigpen | I have been working out of town on vpn's and M$ for the last 3 weeks....I am a bit out of it....hard to get focused. |
| 23:39.21 | pigpen | hence, probably stupid questions from me. |
| 23:40.18 | *** join/#asterisk angom (n=angom@red-corp-201.143.54.251.telnor.net) |
| 23:40.19 | JT | yes working with microsoft products will cause illogical thinking ;) |
| 23:40.19 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 23:40.30 | *** part/#asterisk angom (n=angom@red-corp-201.143.54.251.telnor.net) |
| 23:40.31 | pigpen | yeah...especially when software vendors say, "Oh..64bit W2K3....you need to run the normal one...." |
| 23:40.34 | pigpen | dumbasses. |
| 23:40.39 | NovceGuru | Could somebody point me in the right direction to finding an NEC hardware specialists :) |
| 23:40.47 | blitzrage | call NEC? |
| 23:41.16 | NovceGuru | Well I wanted to give the community a chance first before turning to that |
| 23:42.24 | linagee | NovceGuru: say it in english |
| 23:42.34 | linagee | NovceGuru: "well i wanted free help before calling for support" :P |
| 23:42.35 | kukhuvud | has anyone here used SRTP before on handsets? |
| 23:43.00 | NovceGuru | linagee no I would pay |
| 23:43.04 | JT | i don't asterisk even supports SRTP |
| 23:43.21 | linagee | NovceGuru: hrm. too bad i don't know NEC then |
| 23:43.23 | afrosheen | lol |
| 23:43.29 | afrosheen | call NEC...we have some helpers tody |
| 23:44.21 | linagee | NovceGuru: NEC PBX? |
| 23:44.38 | NovceGuru | yes, DS1000 |
| 23:46.00 | pigpen | I actually have a customer wanting me to make all 200 of their polycom phones ring the "CTU 24" ring.... |
| 23:46.07 | snuffy22 | JT.. i think SRTP is being developed in trunk.. |
| 23:46.28 | JT | pigpen: ask [TK]D-Fender |
| 23:46.29 | kukhuvud | i'm trying to read up more on it |
| 23:46.36 | JT | pigpen: he has the CTU ringtone for polycom |
| 23:46.43 | JT | it's a cisco ringtone |
| 23:46.51 | kukhuvud | it could be a passed through type of encryption like zfone |
| 23:46.52 | pigpen | really? I was going to look into it...but a cisco..funny. |
| 23:47.08 | JT | pigpen: haven't you noticed all the phones at CTU are ciscos? :) |
| 23:47.24 | pigpen | [TK]D-Fender, if I ask pretty please, could I get a copy for the poly? |
| 23:47.34 | pigpen | yeah..I noticed on the 2 shows I watched. |
| 23:48.01 | pigpen | ...and also the $$ my local school district forked out for their Cisco setup ..... |
| 23:48.07 | JT | heh |
| 23:48.09 | pigpen | no wonder my property taxes are so high. |
| 23:48.11 | JT | yeah |
| 23:48.21 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 23:48.26 | pigpen | bastards. |
| 23:48.36 | kukhuvud | http://en.wikipedia.org/wiki/Secure_Real-time_Transport_Protocol (says * doesn't support SRTP) :( |
| 23:48.40 | kukhuvud | ok nevermind |
| 23:48.41 | mosty | NovceGuru: i think the problem is that there probably is no open community for proprietary pbx's |
| 23:49.05 | *** join/#asterisk Fercho (i=fer@200.63.82.199) |
| 23:49.20 | mosty | for most of them, anyway |
| 23:49.20 | JT | kukhuvud: only real option is putting voip over a udp vpn |
| 23:49.41 | kukhuvud | i'm still eagerly waiting for some handset phones to support ZRTP, the pbx doesn't have to support ZRTP only SIP and line will be encrypted |
| 23:49.48 | Fercho | hi. Does Asterisk show CallerID information from PSTN lines? |
| 23:50.08 | mosty | Fercho: it can |
| 23:50.19 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 23:50.20 | JT | kukhuvud: why won't the pbx have to support it? |
| 23:50.31 | kukhuvud | JT: i think you're right for handsets. software phones can be easily encrypted with zfone (http://en.wikipedia.org/wiki/Zfone) |
| 23:50.34 | Fercho | (I think I already asked here but I didn't make myself clear) |
| 23:50.48 | Fercho | thanks mosty |
| 23:51.02 | kukhuvud | JT: because ZRTP encrypts the packets and sends them as regular sip/rtp. the phone on the other end has to be able to decrypt them |
| 23:51.24 | JT | kukhuvud: what about if the media gets terminated at the pbx? |
| 23:51.38 | kukhuvud | JT: so if you have a handset phone with ZRTP support, and you call a regular cell phone the line wont be encrypted. but if you call another handset ot software phone with ZRTP it'll be encrypted. the PBX doesn't even now any better |
| 23:52.08 | JT | what about if a prompt plays from the pbx |
| 23:52.24 | JT | i imagine zrtp would be quite annoying if you wanted to record the calls at the pbx |
| 23:52.30 | kukhuvud | yes |
| 23:52.39 | kukhuvud | you couldn't recrod the calls if it was encrypted |
| 23:52.44 | kukhuvud | thats kinda the point though |
| 23:53.01 | JT | i wonder if you can disable zrtp at the pbx |
| 23:53.26 | kukhuvud | what makes ZRTP the best is that you can enable it on one end and still makes and recieve calls from non-encrypted phones without changing anything, and without changing anything on the PBX |
| 23:53.37 | kukhuvud | since it's transparent to the PBX I dont know |
| 23:53.49 | kukhuvud | maybe some mod that looks specifically for it and kills the sip session |
| 23:53.53 | Fercho | the basic Asterisk is command-line, right? |
| 23:53.58 | JT | it has to negotiate at the endpoint somehow |
| 23:54.04 | *** join/#asterisk mightnare (n=mike@s230165.ppp.asahi-net.or.jp) |
| 23:54.11 | kukhuvud | exactly |
| 23:54.29 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 23:55.02 | kukhuvud | i've tested it with softphones and it worked great through my * box |
| 23:55.11 | kukhuvud | but no hardset phones support it yet :( |
| 23:55.14 | JT | what softphones? |
| 23:55.38 | kukhuvud | any, it installs as a transparent driver that encrypts all rtp traffic from your host |
| 23:55.52 | kukhuvud | free download |
| 23:56.31 | kukhuvud | http://zfoneproject.com/ i tihnk |
| 23:56.36 | *** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au) |
| 23:57.08 | kukhuvud | the dude that wrote PGP write ZRTP , so that makes me feel better |
| 23:57.15 | kukhuvud | *wrote |
| 23:57.26 | JT | do you actually need encryption? |
| 23:57.36 | kukhuvud | man all handsets should come with this its so easy and transparent |
| 23:57.41 | kukhuvud | actually i do |
| 23:57.54 | hrmphh | hrm |
| 23:57.58 | hrmphh | how is the sugarcrm integration? |
| 23:58.01 | hrmphh | work well? |
| 23:58.08 | JT | kukhuvud: why? |
| 23:58.27 | NovceGuru | mosty, I know, I just didnt know if someone here used asterisk but also delt with NEC boxes |
| 23:58.38 | NovceGuru | and I could pay for their services |
| 23:58.40 | kukhuvud | legal reasons, sometimes i have to be able to talk to a person and promise it can't be recorded |
| 23:58.44 | kukhuvud | cells can be tapped |
| 23:58.47 | NovceGuru | makes them sound like prostitutes |
| 23:58.47 | NovceGuru | heh |
| 23:58.48 | kukhuvud | land lines even easier |
| 23:58.53 | sandorp | I just installed a TDM808 card and upgraded to the very latest asterisk; calling internal extensions via x-lite works fine but I still can't dial an outside number ... I get a "All circuits are busy now" message ... any ideas? |
| 23:59.05 | Qwell[] | kukhuvud: It can still be recorded |
| 23:59.24 | kukhuvud | if the 2 points encrypting and decrypting are the handsets, i'd feel a lot better |
| 23:59.24 | Qwell[] | zrtp doesn't mean it's 100% safe, no matter what, always |
| 23:59.31 | kukhuvud | they can record the encrypted packets all they want |
| 23:59.43 | rudholm | you can record an encrypted conversation |
| 23:59.46 | Qwell[] | they can also record the audio right off the line-out jack |
| 23:59.51 | kukhuvud | true |
| 23:59.54 | Qwell[] | trivial |
| 23:59.54 | kukhuvud | i can't help that though |
| 23:59.57 | JT | kukhuvud: actually they can install a trojan on the pc that monitors the speaker and mic lines |
