| 00:03.24 | orlock | ANybody here have any advice on debugging quality issues on RTP? |
| 00:06.45 | *** join/#asterisk mceGEEK (n=mceGEEK@70-89-196-165-inkleaf-mn.hfc.comcastbusiness.net) |
| 00:07.32 | mceGEEK | i'm getting an error [Dec 3 18:02:59] WARNING[16595]: app_dial.c:1081 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 00:07.33 | mceGEEK | <PROTECTED> |
| 00:08.12 | JT | make sure it can actually contact the sip provider with tcp/ip |
| 00:08.20 | JT | that means it's unable to reach the host |
| 00:08.51 | mceGEEK | when i add the line outboundproxy i'm able to reach the host |
| 00:09.01 | mceGEEK | however incoming calls stops |
| 00:09.13 | JT | you must have a strange network setup |
| 00:09.26 | mceGEEK | anything to do with sunrocket has been strange :) |
| 00:18.09 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 00:24.27 | orlock | Does asterisk have an rtp jitter buffer? |
| 00:26.11 | icyfire0573 | Why would background() not work for an outside caller? |
| 00:33.30 | *** join/#asterisk tim0123 (n=cash247@75.39.58.91) |
| 00:33.36 | tim0123 | Hello guys |
| 00:35.06 | tim0123 | I need some help getting zapbarge to work |
| 00:35.56 | *** join/#asterisk tr2x (n=alvar@80-218-150-90.dclient.hispeed.ch) |
| 00:36.49 | dlynes_laptop | icyfire0573: it does |
| 00:36.58 | dlynes_laptop | icyfire0573: how about telling us what makes you think it doesn't? |
| 00:37.40 | icyfire0573 | I have a dialplan setup and when I call from outside it dosen't work, but if I copy the context internally to internally and give it an extension it does work. |
| 00:38.05 | *** join/#asterisk doolph (i=doo@200.75.198.188) |
| 00:39.27 | [TK]D-Fender | icyfire0573 : Does playback work in tis place? |
| 00:39.56 | icyfire0573 | If I call the extension, it always plays the recorded message, it just refuses to respond to dtmf |
| 00:40.28 | [TK]D-Fender | icyfire0573 : You are talking apples and oranges. |
| 00:40.54 | [TK]D-Fender | icyfire0573 : Put the context the way it should be, and show us the channel config for the origin of your "outside" calls. |
| 00:40.59 | [TK]D-Fender | ~pb |
| 00:41.00 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 00:41.08 | *** join/#asterisk mindCrime_ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 00:41.12 | doolph | anyone have setup sipura 3000 for trunking? |
| 00:42.32 | icyfire0573 | http://pastebin.ca/266312 |
| 00:43.00 | *** join/#asterisk SomeOne1 (n=SomeOne1@pool-71-126-150-231.washdc.fios.verizon.net) |
| 00:43.13 | SomeOne1 | hello |
| 00:43.21 | SomeOne1 | RoyK: sup |
| 00:45.41 | *** join/#asterisk xnon (i=xnon@200.8.85.221) |
| 00:45.52 | *** join/#asterisk bkruse (n=root@69.73.127.92) |
| 00:46.11 | [TK]D-Fender | icyfire0573 : Taking forever to load |
| 00:46.21 | icyfire0573 | yea |
| 00:46.30 | icyfire0573 | its only 15 lines though. |
| 00:47.13 | dlynes_laptop | doolph: you mean as a gateway? |
| 00:47.42 | JT | umm |
| 00:47.44 | JT | so anyway |
| 00:47.57 | JT | icyfire0573: it sounds like you have inbound DTMF recognition problems |
| 00:48.07 | icyfire0573 | yep.\ |
| 00:48.08 | RoyK | SomeOne1: hey! |
| 00:48.10 | JT | what sort of channel is the call coming in over? |
| 00:48.17 | bkruse | icyfire0573: what are the calls coming in on? |
| 00:48.18 | icyfire0573 | SIP |
| 00:48.21 | [TK]D-Fender | icyfire0573 : ok, its up. So what is this about DTMF not working? |
| 00:48.47 | bkruse | icyfire0573: ive seen this before, can you noop the extension dialed one your in a call, to see if it reconized the dtmf? |
| 00:48.52 | icyfire0573 | basically calls that come in to the sippeer extension don't listen to DTMF even though the ones that go to the testing context do |
| 00:48.54 | doolph | dlynes_laptop yes |
| 00:48.59 | [TK]D-Fender | icyfire0573 : And to my awareness you can't do Background followed by a Dial and have it play while dialing is in progress |
| 00:49.00 | icyfire0573 | noop? |
| 00:49.06 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.226) |
| 00:49.18 | JT | icyfire0573: is dtmf inband? |
| 00:49.24 | icyfire0573 | [TK]D-Fender, that I do know, if you don't dial before the timeout it goes to the operator. |
| 00:49.31 | bkruse | JT: i bet it is, and i bet thats the problem |
| 00:49.38 | [TK]D-Fender | icyfire0573 : You don't have a proper IVR setup in there so its not listening for any input. |
| 00:49.38 | icyfire0573 | JT, i was looking at the sip.conf and that channel looks like it does sip-INFO so probably not. |
| 00:49.41 | JT | icyfire0573: or rfc2833? |
| 00:49.45 | dlynes_laptop | doolph: yeah...there's plenty of examples on the linksys/sipura users' group on voxilla on how to set them up with asterisk |
| 00:49.57 | JT | icyfire0573: dtmfmode=??? |
| 00:50.04 | icyfire0573 | dtfmode=info |
| 00:50.07 | [TK]D-Fender | icyfire0573 : Your entire methodolgy for that "menu" is really really wrong. |
| 00:50.09 | JT | icyfire0573: what codec are you using? |
| 00:50.10 | bkruse | exten => s,1,Answer() |
| 00:50.10 | bkruse | exten => s,n,Noop(${1.4 or 1.2 extension syntax) |
| 00:50.11 | icyfire0573 | i just changed it to inband |
| 00:50.17 | bkruse | its not the codec |
| 00:50.21 | bkruse | try inband dtmf. |
| 00:50.25 | JT | bkruse: it can be. |
| 00:50.32 | doolph | dlynes_laptop those gateways are stupids, it answer the call first, then make call, this means that I get wrong cdrs |
| 00:50.37 | JT | icyfire0573: what codec do the sip calls come in over? |
| 00:50.40 | [TK]D-Fender | JT, bkruse : its not SIP settings (yet), the entire context is just... WRONG. |
| 00:50.47 | icyfire0573 | JT, not sure, never checked |
| 00:50.50 | bkruse | oh really? i havnet looked at it |
| 00:50.56 | icyfire0573 | [TK]D-Fender, how so? |
| 00:51.04 | bkruse | show me the extensions.conf part and sip.conf of the phone entry |
| 00:51.10 | bkruse | JT: i would put money on it :], if its what im thinking....... |
| 00:51.11 | dlynes_laptop | doolph: Yes, that's how it works, but i've never gotten incorrect cdrs from them |
| 00:51.14 | JT | <PROTECTED> |
| 00:51.14 | [TK]D-Fender | icyfire0573 : You need to do IVR's starting with the "s" exten. Go read the basics on this again |
| 00:51.18 | JT | thats his pb |
| 00:51.24 | dlynes_laptop | doolph: or do you mean answer the call and pass it off onto the fxs port? |
| 00:51.26 | bkruse | k, let me hceck it out. |
| 00:51.28 | icyfire0573 | http://pastebin.ca/266313 |
| 00:51.31 | [TK]D-Fender | JT : Yes, I'm looking at it. You can't do IVR's off #;'d extens |
| 00:51.36 | tim0123 | Yoo someone help a brotha out with zapbarge |
| 00:51.39 | [TK]D-Fender | This is dialplan 101 |
| 00:51.43 | bkruse | tim0123: what |
| 00:51.43 | dlynes_laptop | doolph: fxo->fxs is the default mode of operation, unless you override it |
| 00:51.46 | icyfire0573 | [TK]D-Fender, I would use S but it didn't work right |
| 00:51.51 | bkruse | just use chanspy |
| 00:52.03 | [TK]D-Fender | icyfire0573 : It works just fine, you just aren't setting it up right. |
| 00:52.04 | tim0123 | chanspy |
| 00:52.12 | tim0123 | How does that work |
| 00:52.16 | icyfire0573 | maybe its the way the sip.conf is setup. But I debugged it for a while and found that the sip calls were coming in with that extension so that's how I made the phone pickup |
| 00:52.18 | doolph | dlynes_laptop, its voip --> asterisk --> spa 3000 |
| 00:52.37 | JT | icyfire0573: he's absolutely right |
| 00:52.42 | JT | you must start at context s |
| 00:52.47 | dlynes_laptop | doolph: ok |
| 00:52.57 | JT | unless you have isdn, then the extension numbers are inbound MSNs |
| 00:53.02 | [TK]D-Fender | icyfire0573 : No its no. If you aren't running your "ivr" off "s" you're DOA. Period. |
| 00:53.03 | bkruse | this is disgusting |
| 00:53.03 | tim0123 | well im using hudlite and it uses Barge as a command |
| 00:53.05 | bkruse | haha |
| 00:53.10 | bkruse | tim0123: |
| 00:53.14 | bkruse | chanspy=awesome |
| 00:53.16 | bkruse | let me link you |
| 00:53.22 | icyfire0573 | but I can call extension 3333 internally and it works ... |
| 00:53.36 | bkruse | tim0123: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ChanSpy |
| 00:53.43 | tim0123 | Go ahead |
| 00:53.44 | bkruse | icyfire0573: can i get ssh? i have a couple minutes before i have to go |
| 00:53.45 | [TK]D-Fender | icyfire0573 : This nic... but its not an IVR, you're just generating another inbound call. |
| 00:53.58 | tim0123 | Appriciate that bkruse |
| 00:54.37 | bkruse | tim0123: no problem, you will come to LOVE chanspy, especially when you are the administer over some funny people + phones |
| 00:54.40 | icyfire0573 | bkruse, rather not do ssh at this point. |
| 00:54.46 | [TK]D-Fender | icyfire0573 : I'd let you work your way through this, but I'm betting this will take a long time so I'm just going to GIVE you the proper way you should have built it. Hold on. |
| 00:55.34 | bkruse | icyfire0573: thats fine, i wouldnt trust some random dude in an irc room anyways |
| 00:55.40 | bkruse | icyfire0573: what version asterisk you using, if trunk, what rev? |
| 00:55.43 | *** join/#asterisk FastFeet (n=fastfeet@CPE0013109fd25b-CM000f9fa60d7a.cpe.net.cable.rogers.com) |
| 00:55.53 | icyfire0573 | Asterisk 1.2.13, |
| 00:56.25 | bkruse | k |
| 00:56.25 | *** join/#asterisk AJayMn (n=aj@24-159-236-181.dhcp.mdsn.wi.charter.com) |
| 00:56.29 | bkruse | is this a production server? |
| 00:56.43 | bkruse | and what dtmf mode are you using again? |
| 00:56.46 | icyfire0573 | nope, just me |
| 00:56.56 | bkruse | k, awesome |
| 00:56.58 | icyfire0573 | I just changed it from dtmfmode = info to dtmfmode = inband |
| 00:56.59 | bkruse | try different dtmf modes |
| 00:57.09 | bkruse | and ALSO MAYBE try the asterisk 1.4 beta3 tarball from digium's website |
| 00:57.22 | bkruse | WARNING: command line is different, and itll scare you at first |
| 00:57.34 | [TK]D-Fender | icyfire0573 : http://pastebin.ca/266329 |
| 00:58.24 | icyfire0573 | Thanks so much [TK]D-Fender |
| 00:58.38 | bkruse | now what do you want to do icyfire0573 |
| 00:58.41 | bkruse | dtmf just not working |
| 00:58.43 | [TK]D-Fender | icyfire0573 : Try it and let me know. |
| 00:58.46 | bkruse | fender what idd you change exactly? |
| 00:59.04 | [TK]D-Fender | bkruse : Again, its not a DTMF problem, its the fact the IVR was completely broken |
| 00:59.13 | [TK]D-Fender | bkruse : Go look. EVERYTHING |
| 00:59.54 | bkruse | [TK]D-Fender: awesome |
| 00:59.54 | bkruse | haha |
| 01:00.00 | bkruse | sorry im just trying to offer my quick asterisk knowledge |
| 01:00.07 | bkruse | but i see what you mean |
| 01:00.18 | bkruse | it just set off an alarm, ive been working with some sip dtmf problems in the 1.2 branch |
| 01:01.35 | dlynes_laptop | doolph: try this: http://forum.voxilla.com/linksys-sipura-spa-users-group/configuration-asterisk-spa-3000-interaction-9452.html |
| 01:01.42 | [TK]D-Fender | bkruse : Step 1 : Actually LOOK at what they show you. Step 2 : Realize the blatantly obvious core errors (never trust their description of what thy THINK is wrong.). :) |
| 01:01.45 | *** join/#asterisk techie (n=techie@c-67-182-22-174.hsd1.ca.comcast.net) |
| 01:02.32 | AJayMn | Anyone know of a low startup cost E911 provider? |
| 01:03.26 | FastFeet | Maybe a silly question, but I need to ask anyway: Is the Linksys SPA3000 and SPA3102 the same, except that the SPA3102 includes a Router? |
| 01:03.30 | bkruse | [TK]D-Fender: i know asterisk pretty well........... |
| 01:03.43 | bkruse | [TK]D-Fender: i just jumped in irc and thought i saw a bug i was working out. |
| 01:03.47 | FastFeet | Also will either of those work from behind a NAT router already? |
| 01:03.57 | [TK]D-Fender | FastFeet : Both. |
| 01:04.05 | bkruse | qualify=yay! |
| 01:04.11 | FastFeet | Both work from behind my current NAT router? |
| 01:04.41 | FastFeet | Which would you choose? Both are similarly Priced. |
| 01:06.38 | *** join/#asterisk mhnoyes_ (n=mhnoyes@dialup-4.246.21.14.Dial1.SanJose1.Level3.net) |
| 01:06.47 | [TK]D-Fender | FastFeet : Take the SPA-3102. It can act as a standalone router if you need it to later, comes with more ram & a bigger processor |
| 01:07.11 | FastFeet | Sweet thanks for you time... I shall order it now.... |
| 01:07.53 | bkruse | [TK]D-Fender: good chose |
| 01:07.59 | bkruse | s/chose/choice |
| 01:08.23 | [TK]D-Fender | bkruse : I've owned both, as well as SPA-2000, 1001, and 941. |
| 01:08.32 | AJayMn | Anyone know of a low startup cost E911 provider? |
| 01:08.47 | dlynes_laptop | AJayMn: You might want to specify in which country |
| 01:08.50 | bkruse | [TK]D-Fender: and that was your favorite |
| 01:08.59 | AJayMn | US E911 Provider |
| 01:09.09 | bkruse | AJayMn: i honestly dont know of any :[ never had to do 911 :P |
| 01:09.16 | [TK]D-Fender | bkruse : Dunno.. they all seemed jsut about the same in the end to me personally (for the similar models) |
| 01:10.44 | bkruse | [TK]D-Fender: isnt the 3000 a little cheaper? |
| 01:11.30 | [TK]D-Fender | bkruse : Depends where, bu the 3000 is being phased out for the 3102. It is more functional as well. I wouldn't touch an old one at this point if yuo can buy the newer one |
| 01:11.45 | bkruse | [TK]D-Fender: right on, thanks :D |
| 01:14.06 | bkruse | this is random, but my friend just asked me what a noob is, what should i tell her? |
| 01:14.18 | bkruse | i dont wana say the usual, old, "you" |
| 01:15.07 | bkruse | nvm......... |
| 01:15.28 | rob0 | http://en.wikipedia.org/wiki/Noob |
| 01:16.23 | bkruse | hehe |
| 01:16.24 | bkruse | thanks! |
| 01:17.54 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:18.01 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:18.18 | bok | are there problems with having multiple voip numbers/accounts registered with the one provider? |
| 01:18.41 | AJayMn | bok depends on the provider.. |
| 01:18.55 | bok | getting a problem where when it tries to bridge sip channels it always uses the credentials from the second account in the proxy-authenticate message |
| 01:19.02 | *** part/#asterisk dasenjo (n=dasenjo@208.195.215.226) |
| 01:19.45 | bok | so the second account works great, obviously, but the first account just wont send the right details |
| 01:20.07 | bok | been trying to figure this out for three days now |
| 01:20.23 | AJayMn | bok who is ur provider? |
| 01:20.34 | bok | engin, in australia |
| 01:21.03 | JT | hrm |
| 01:21.08 | JT | are they all voiper accounts? |
| 01:21.13 | bok | yeah |
| 01:21.18 | JT | ok |
| 01:21.59 | bok | its odd, the From: header in the INVITE message has the right number, but the Proxy-Authenticate one doesnt |
| 01:22.25 | JT | i have heard of problems with multiple accounts on one provider |
| 01:22.35 | JT | there are seperate sip sections for each account? |
| 01:22.39 | bok | yep |
| 01:22.48 | orlock | i have problems with multiple DID's on the one account, multiple accounts is fine though |
| 01:23.19 | bok | im thinking that whenever asterisk looks for auth details, it does so based on the ip/host of the request and picks the first one listed |
| 01:24.13 | orlock | bok: may, it actually does it based on the To: address in the INVIET |
| 01:25.31 | rob0 | I have 3 accounts from ipkall and 2 from stanaphone ... is this the sort of thing you mean? |
| 01:25.32 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 01:25.32 | rob0 | I have each going to its own context, using different extensions therein per account. |
| 01:25.32 | bok | yeah i have that also |
| 01:25.40 | bok | all works great |
| 01:25.53 | bok | until it tries to bridge channels and sends the wrong username bacmk |
| 01:26.35 | mceGEEK | hmm incoming doesn't work again |
| 01:27.19 | bok | ok found a solution |
| 01:27.27 | bok | just need to register with different proxies |
| 01:27.30 | bok | so it is ip based |
| 01:29.04 | bok | hmm |
| 01:29.10 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 01:30.46 | JT | sounds more like a hack than a colution |
| 01:30.48 | JT | solution |
| 01:31.30 | bok | well yeah |
| 01:31.49 | bok | especially since the outbound proxy only wants to allow the call half the time |
| 01:31.58 | bkruse | jt: agreed |
| 01:32.07 | JT | engin has a few bugs itself, bok |
| 01:32.59 | JT | one of the silliest is that it uses the location of the sip proxy you are registered to, not the location of you as set in your account, to determine the local calling area code |
| 01:33.30 | JT | so if you happen to get given a melbourne area code and you are in sydney, you dial locally in melbourne |
| 01:35.19 | JT | i meant get given a melbourne sip proxy in dns rotation |
| 01:35.21 | JT | but yeah |
| 01:36.48 | bok | heh |
| 01:37.28 | JT | specifying the sydney proxy doesn't help, as it still can get melbourne proxies in SRV requests |
| 01:37.36 | JT | bok: have you noticed this? |
| 01:37.49 | bok | not yet no |
| 01:38.11 | bok | havent been using engin long enough |
| 01:38.13 | JT | i'm guessing you haven't come across trying to block CID sending with SIP? |
| 01:38.45 | *** join/#asterisk kyo (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 01:38.46 | JT | i do both of them with automated prefixing in the dialplan atm, although it's not ideal |
| 01:40.23 | bok | ok better solution :) |
| 01:40.41 | bok | left the host as byo, set different outbound proxies for each |
| 01:42.02 | JT | hmm |
| 01:43.08 | orlock | JT: heh, when i am testing sip stuff i will commonly register with a NSW or QLD sip server |
| 01:43.19 | orlock | then dial the home voip phone from it |
| 01:43.29 | orlock | dial from computer..,. hardware voip phone rings on desk |
| 01:43.35 | orlock | GF looks.. "Sydney? WTF!" |
| 01:43.39 | JT | right |
| 01:43.50 | *** join/#asterisk Soul (n=Soul@87-196-111-228.net.novis.pt) |
| 01:43.56 | JT | why would it say sydney unless you get a sydney DID |
| 01:44.07 | JT | well outgoing line moreso |
| 01:44.28 | orlock | i was testing a sydney did |
| 01:44.30 | orlock | :) |
| 01:44.37 | orlock | but yeah, i can see how that engin bug would suck |
| 01:44.59 | JT | i wonder if it affects engin voice box customers |
| 01:45.12 | bok | dont know |
| 01:45.16 | JT | maybe the SPAs have dialplans that override the problem |
| 01:45.23 | bok | but i have one on my desk im supposed to test |
| 01:45.31 | JT | otherwise it would've veen fixed by now |
| 01:45.41 | JT | not sure if they work on the byo network |
| 01:46.02 | bok | should be no reason why not |
| 01:46.08 | JT | well i'm led to believe that both networks are exactly the same |
| 01:46.09 | bok | but i do have a normal account to go with it |
| 01:46.16 | JT | except one has a lower level of tech support |
| 01:46.26 | JT | well it might do something funny |
| 01:46.43 | JT | engin hide the sip password on normal non voiper accounts |
| 01:47.05 | JT | so their softphone must translate the sip password from the user password |
| 01:47.13 | JT | using som algorithm |
| 01:47.15 | JT | is my theory |
| 01:50.08 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 02:05.25 | *** join/#asterisk jayvabb (n=jaycox@ip68-98-170-147.dc.dc.cox.net) |
| 02:07.04 | *** join/#asterisk BSDTech (n=RNeese@ppp-71-128-112-135.dsl.irvnca.pacbell.net) |
| 02:07.08 | BSDTech | hey all |
| 02:07.13 | *** join/#asterisk Newbie___ (n=me@211.24.146.11) |
| 02:07.18 | BSDTech | ok addons fails to compile |
| 02:07.40 | BSDTech | rc/chan_h323.c: In function `ooh323_new': |
| 02:07.40 | BSDTech | src/chan_h323.c:250: error: too few arguments to function `ast_channel_alloc' |
| 02:07.52 | BSDTech | it errors out there |
| 02:07.55 | bkruse | uh oh |
| 02:08.03 | bkruse | BSDTech: made a boo-boo |
| 02:08.17 | bkruse | i honestly have no idea, havent messed with h323 at all |
| 02:08.18 | *** join/#asterisk _SowdaH_ (n=ubuntu@ivr94-1-81-57-178-223.fbx.proxad.net) |
| 02:08.20 | russellb | BSDTech: feel free to open a report on the bug tracker. |
| 02:08.20 | BSDTech | no I ran ./configure and make |
| 02:08.38 | jayvabb | msg NickServ help register |
| 02:08.38 | russellb | there was an API change ... probably a month ago |
| 02:08.42 | russellb | i guess addons never got updated |
| 02:08.47 | bkruse | jayvabb: /msg |
| 02:08.56 | _SowdaH_ | heyo |
| 02:09.07 | ariel_ | so when is the 1.4 non beta going to be released? |
| 02:09.14 | ariel_ | hello everyone |
| 02:09.15 | russellb | yesterday |
| 02:09.16 | _SowdaH_ | got a newbie problem if someone can help |
| 02:09.18 | BSDTech | I just pulled breanch/1.4 svn tonight |
| 02:09.47 | BSDTech | grrr |
| 02:10.09 | ariel_ | what no posting for it? |
| 02:10.11 | ariel_ | wow |
| 02:10.16 | ariel_ | yesterday........ |
| 02:10.17 | russellb | just kidding :-p |
| 02:10.38 | *** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net) |
| 02:10.47 | bkruse | bugs.digium.com down :[ |
| 02:10.54 | russellb | but seriously, soon. just one more thing to fix. |
| 02:11.13 | ariel_ | yes that is always the case just one more thing to fix |
| 02:11.19 | russellb | bkruse: only for nubs. it works for me |
| 02:11.23 | russellb | ariel_: :) |
| 02:11.29 | Newbie___ | hi all, when using a channel bank to connect to a analog phone. does the line polarity matter? |
| 02:11.39 | bkruse | russellb: it wasnt down, i was just testing you russell (refreshes firefox) |
| 02:11.45 | russellb | ariel_: we still have to fix the shared line appearance code. but that's the last thing on our list |
| 02:11.45 | ariel_ | depends on the cb |
| 02:11.46 | bkruse | wireless > bkruse |
| 02:12.01 | Newbie___ | ariel_: i see |
| 02:12.09 | ariel_ | russellb, great to hear it. I was glad to see the wisper mode get in the code. |
| 02:12.10 | _SowdaH_ | got this kind of problem with my asterisk server |
| 02:12.27 | _SowdaH_ | WARNING[6692]: chan_sip.c:1082 __sip_xmit: sip_xmit of 0x816c818 (len 535) to 192.168.0.1:2772 returned -1: Bad file descriptor |
| 02:12.48 | ariel_ | disk space? |
| 02:13.00 | Newbie___ | ariel_: would a wrong polarity cause strange analog phone behavior. ie random hangup |
| 02:13.18 | JT | wisper mode? |
| 02:13.19 | ariel_ | depends on the CB and other things |
| 02:13.22 | BSDTech | O well I can wait |
| 02:13.26 | _SowdaH_ | ar u talkin to me? |
| 02:13.31 | Newbie___ | ariel_:ok |
| 02:13.31 | JT | Newbie___: call progress decetion in the config would |
| 02:13.32 | BSDTech | zaptel libpri and asterisk built |
| 02:13.52 | _SowdaH_ | ariel_: u talkin to me? |
| 02:13.58 | bkruse | disk space or is that ulimit? |
| 02:14.01 | BSDTech | I dont really want h323 I might just comment it out |
| 02:14.34 | bkruse | BSDTech: sounds like a plan |
| 02:14.51 | ariel_ | _SowdaH_, yes it was too you check space available |
| 02:14.54 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 02:14.59 | Newbie___ | JT: my callprogress=no and there are some phone acted strangly ie randon hangup when bridged |
| 02:15.17 | JT | Newbie___: what are your zttest scores? |
| 02:15.30 | ariel_ | Newbie___, what are you using e&m wink kwel start? |
| 02:15.41 | Newbie___ | zttest ? |
| 02:15.48 | Newbie___ | loopstart |
| 02:15.52 | JT | Newbie___: dude, run zttest |
| 02:15.54 | _SowdaH_ | ariel_: still have plenty |
| 02:16.10 | JT | Newbie___: sort of essential if you are setting up any sort of zap hardware |
| 02:16.33 | ariel_ | for the user that asterisk is running on? |
| 02:16.41 | _SowdaH_ | worked before i dont undertstand |
| 02:16.44 | Newbie___ | JT: hmmm what does zttest do |
| 02:16.52 | ariel_ | worked before what? |
| 02:16.55 | JT | measures zap accuracy |
| 02:16.59 | JT | hurry up and do it :) |
| 02:17.16 | Newbie___ | i am getting mostly 100% a few 99.987793% |
| 02:17.26 | JT | that's fine |
| 02:17.33 | JT | below 99.97% is bad |
| 02:17.35 | _SowdaH_ | JT: do u have any suggestion with this plz? |
| 02:17.36 | _SowdaH_ | WARNING[6692]: chan_sip.c:1082 __sip_xmit: sip_xmit of 0x816c818 (len 535) to 192.168.0.1:2772 returned -1: Bad file descriptor |
| 02:17.43 | Newbie___ | damn, i am learning new stuff everyday |
| 02:17.59 | JT | no, if i did i would've said something, _SowdaH_ |
| 02:18.05 | JT | no need to repaste it |
| 02:18.20 | _SowdaH_ | thought u didnt see it |
| 02:18.24 | bkruse | _SowdaH_: df / |
| 02:18.27 | JT | Newbie___: umm, try running zttest while you are having issues |
| 02:18.35 | JT | during call problems |
| 02:18.38 | bkruse | _SowdaH_: what are the results of df / and unlimit |
| 02:18.41 | JT | see if the score spikes downwards |
| 02:18.44 | Newbie___ | so i take it maybe a bad polarity or a bad analog phone |
| 02:19.12 | JT | is there any reason you are using loopstart instead of kewlstart? |
| 02:19.18 | JT | you haven't tried alternative phones? |
| 02:19.32 | _SowdaH_ | careful its french :p |
| 02:19.38 | _SowdaH_ | Sys. de fich. 1K-blocs Occupé Disponible Capacité Monté sur |
| 02:19.38 | _SowdaH_ | /dev/hda1 36969672 2769516 32322160 8% / |
| 02:19.50 | _SowdaH_ | so i have margin |
| 02:20.12 | Newbie___ | JT: because cb is configure a loopstart |
| 02:20.38 | JT | Newbie___: i don't think that should matter |
| 02:20.47 | JT | kewlstart is like an enhanced loopstart |
| 02:20.47 | BSDTech | ok I disable oh323 and it builds fine |
| 02:20.51 | *** join/#asterisk sahafeez (n=sahafeez@ip68-6-223-92.sd.sd.cox.net) |
| 02:20.56 | JT | better progress detection iirc |
| 02:20.56 | ariel_ | _SowdaH_, it can also mean your not able to get to the correct ip address or domain name resolve |
| 02:20.56 | BSDTech | and installs |
| 02:21.29 | bkruse | thats fine |
| 02:21.32 | bkruse | and ulimite? |
| 02:21.35 | ariel_ | are you runing X86_64? |
| 02:21.36 | bkruse | ulimit* |
| 02:22.13 | *** join/#asterisk mogorman (n=mogorman@c-71-207-215-93.hsd1.al.comcast.net) |
| 02:22.13 | *** mode/#asterisk [+o mogorman] by ChanServ |
| 02:22.22 | bkruse | mogorman!!!!!!!! |
| 02:22.23 | _SowdaH_ | ariel_: i agree more on this |
| 02:22.38 | Newbie___ | JT: i have been experiencing random hangup when bridged last few weeks. just last night i figure out the random hangup only cause by a certain extensions. when i removed those extensions from call group. it's been working fine since this morning |
| 02:22.38 | mogorman | woohoo |
| 02:22.48 | file | mogorman: !!!!!!!!!!!!!!!!!!! |
| 02:22.56 | Newbie___ | i therefore suspect a polarity issue or a bad phone |
| 02:23.03 | mogorman | yup yup |
| 02:23.07 | JT | Newbie___: so have you actually checked their polarity? |
| 02:23.13 | _SowdaH_ | but u dont have to specify any ip on basic modifications of sip.conf or extensions.conf |
| 02:23.28 | bkruse | how was the honeymoon mog? |
| 02:23.35 | bkruse | friends dont ask and a man doesnt tell? |
| 02:23.38 | mogorman | fantastic |
| 02:23.41 | mogorman | cruise |
| 02:23.49 | bkruse | right on! |
| 02:23.49 | mogorman | very good way to spend a honeymoon |
| 02:23.54 | bkruse | hehe |
| 02:23.59 | Newbie___ | JT: no not yet, but before i do that. i wanna try to change the phone. from no random hangup to a problem one |
| 02:24.05 | ariel_ | Cruise yes I know them.... |
| 02:24.12 | bkruse | mogorman: no sea sickness for either? |
| 02:24.22 | mogorman | actually a bit of one |
| 02:24.24 | ariel_ | but did you make it out of the cabin? |
| 02:24.35 | mogorman | but i got over it |
| 02:24.37 | _SowdaH_ | ill try to figure it out on a french asterisk forum |
| 02:24.39 | mogorman | oh yeah |
| 02:24.48 | _SowdaH_ | thx all for the help tho |
| 02:24.57 | blitzrage | mogorman: welcome home! |
| 02:24.57 | _SowdaH_ | :) |
| 02:25.26 | ariel_ | _SowdaH_, are your dns and names for your server setup correctly? |
| 02:25.30 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 02:26.08 | mogorman | been crazy weekend |
| 02:26.08 | blitzrage | someone de-nubify me. What does "MUL" under the "KEY" column in the odbcstorage.txt file mean? |
| 02:26.21 | mogorman | all of my computer equipment over heated some how while i was on cruise |
| 02:26.30 | mogorman | hds burned up, computers crashed |
| 02:26.39 | mogorman | all very messy suprise to come home to |
| 02:26.46 | blitzrage | ewwww |
| 02:27.03 | bkruse | woah! |
| 02:27.25 | russellb | mogorman: i pwned your desk friday |
| 02:27.29 | blitzrage | | Field | Type | Null | Key | Default | Extra | |
| 02:27.34 | mogorman | you were here? |
| 02:27.37 | mogorman | or are here? |
| 02:27.39 | russellb | mogorman: was |
| 02:27.49 | mogorman | aww |
| 02:27.50 | bkruse | agreed with russellb, he owned it |
| 02:27.54 | mogorman | i almost came in friday |
| 02:28.16 | mogorman | i had to finish lots of last minute junk last friday and was way to tired |
| 02:28.52 | mogorman | you didnt take your water bottle russellb ? |
| 02:28.53 | _SowdaH_ | ariel_: in my client i set it up for my asterisk server so 192.168.0.2 |
| 02:29.01 | russellb | mogorman: i didn't even see it |
| 02:29.09 | blitzrage | no one huh? :) |
| 02:29.11 | mogorman | its still was on my desk |
| 02:29.13 | file | mogorman: you also missed a sexy party! |
| 02:29.16 | mogorman | before i left |
| 02:29.22 | mogorman | no i dont think i did file ^_^ |
| 02:29.22 | russellb | mogorman: i didn't see it ... |
| 02:29.43 | russellb | someone stole it while you were gone!!!!! |
| 02:29.48 | mogorman | eep |
| 02:29.52 | mogorman | that is depressing |
| 02:30.45 | *** join/#asterisk Tond (n=tond@CPE0014bf30c190-CM0010954a5ffa.cpe.net.cable.rogers.com) |
| 02:32.22 | Tond | Hi, i need to do routing for my national and international calls to different carrier. I will be perhaps passing over 2 million minutes per month of traffic. However i am not sure if I should let my asterisk box handle this routing and proxying task or look into another software? Any tips are appreciated... tnx |
| 02:33.40 | blitzrage | if you don't have to handle audio, then use OpenSER |
| 02:33.49 | blitzrage | if you want to do fancy things, then add asterisk into the mix |
| 02:33.59 | bkruse | Tond: asterisk can handle that no prob |
| 02:34.20 | mogorman | so what has happened in asterisk whilst i was gone |
| 02:34.30 | Tond | I have OpenSer installed and looking at it, but it doens't look very easy to configure... |
| 02:34.30 | blitzrage | mogorman: file fixed all the bugs |
| 02:34.42 | JT | umm |
| 02:34.51 | mogorman | yay! |
| 02:34.53 | blitzrage | Tond: yah, I find asterisk easier, but that's because I've used asterisk for many years, and SER for a few mins :) |
| 02:34.55 | Tond | I dont want to do anythign fansty simple call route and if it fails, it will jump to the next carrier in route |
| 02:35.02 | JT | should something that handles 2 million minutes of calls a month be "easy" to setup?! |
| 02:35.13 | blitzrage | mogorman: oej and I found the CANCEL/BYE bug in 1.4 |
| 02:35.13 | Tond | ha ha |
| 02:35.16 | bkruse | hey Tond |
| 02:35.18 | bkruse | http://asterisknow.org/ |
| 02:35.29 | blitzrage | bkruse: oh yah, I keep forgetting about that thing :0 |
| 02:35.42 | Tond | thanks... |
| 02:35.45 | mogorman | nice blitzrage |
| 02:36.11 | file | mogorman: it is nifty to have you back online! |
| 02:36.15 | blitzrage | and I got func_odbc updated so I can use the HASH() function in it, and upgraded all my test servers to it :) |
| 02:36.19 | blitzrage | quite |
| 02:36.23 | bkruse | blitzrage: its easier for people coming off of trixbox :] |
| 02:36.23 | blitzrage | mogorman: have a good cruise? |
| 02:36.28 | bkruse | its a good enticement |
| 02:36.29 | mogorman | well spent all saturday rebuilding everything |
| 02:36.35 | mogorman | yeah real nice |
| 02:36.38 | mogorman | and good wedding too |
| 02:36.38 | blitzrage | bkruse: I don't use gui's, so I have no idea :) |
| 02:36.47 | blitzrage | mogorman: excellent. good to hear. |
| 02:37.12 | bkruse | blitzrage: same here |
| 02:37.18 | blitzrage | incase you missed it, it's not very obvious what MUL under the Key column in the odbcstorage.txt file... :) |
| 02:37.19 | Tond | I mean i am very comfortable using Asterisk and doing stuff with it. However iw as a bit spectical about passing 2-3 mil minutes of traffic on it. and thought maybe i can install a sip proxy in the way to do the routing, but SER looks more complicated specially if i have few hundred routes in place.. |
| 02:37.23 | bkruse | blitzrage: give me a console and thats it, links http if i have to :] |
| 02:37.27 | blitzrage | means* |
| 02:37.54 | bkruse | Tond: i think with the right hardware and network, it could deffinitly handle it |
| 02:38.09 | blitzrage | well... SER might be easier to learn than to cluster asterisk :) |
| 02:38.30 | blitzrage | I've been working on a carrier system to do that for the last 2-3 months... almost done :) |
| 02:38.30 | Tond | I see.. Well as for hardware I am thinking of a dual Xeon and lots of ram on a SuperMicro hardware and Cent OS |
| 02:38.37 | Tond | with version 1.2.13 |
| 02:38.45 | bkruse | Cent OS? |
| 02:38.53 | Tond | CentOS 4 |
| 02:38.53 | bkruse | any reason for Cent OS? |
| 02:39.01 | blitzrage | its easy, well supported, stable |
| 02:39.10 | JT | supermicro is good |
| 02:39.11 | *** join/#asterisk mceGEEK (n=mceGEEK@70-89-196-165-inkleaf-mn.hfc.comcastbusiness.net) |
| 02:39.13 | Tond | no, just found it easy and seems very stable |
| 02:39.15 | JT | but buy a brand name server |
| 02:40.10 | Tond | blitzrage, how do you setup routes in SER becsuse form what i undrestand a route that can be as short as exten => in asterisk can turn into a few lines of code in SER, and where do i keep my routing? |
| 02:40.30 | blitzrage | Tond: no idea.. as I said before, years on asterisk, minutes on ser |
| 02:40.36 | bkruse | blitzrage: bleh |
| 02:40.40 | bkruse | blitzrage: :] |
| 02:40.41 | Tond | Oh.. ok thnaks.. |
| 02:40.49 | JT | and yeah, you wouldn't use a single server for that sort of call volume |
| 02:41.00 | bkruse | voip-info has some good info on SER from what i remember |
| 02:41.32 | Tond | well I much rather keep my Asterisk box if I can.. I mena after all if my minutes gro I can dedicate a seperate box for each part of the world instead of loadbalacing which I am not sure is stable enough yet |
| 02:41.53 | blitzrage | according to my tests on a dual xeon 2.4 w/ 2GB of ram, you can get around 120 simultaneous calls. At 120 calls * 60 mins * 24 * 31, you can get around 5356800 minutes in a 31 day month on a single box. |
| 02:41.55 | Tond | I looked at those, wanted to see what others here think too |
| 02:42.00 | JT | well as long as you're sure your boxes will stay up |
| 02:42.03 | blitzrage | so yes -- I'd say 2 million minutes and you're good. |
| 02:42.15 | bkruse | yep |
| 02:42.16 | blitzrage | oh, and that's on 1.2.12.1 |
| 02:42.23 | blitzrage | with fancy DB lookup stuff |
| 02:42.26 | *** join/#asterisk NeonLevel (i=Otto@189.169.43.197) |
| 02:42.46 | JT | an average month is 30 days :) |
| 02:42.58 | Tond | Ya I am planning to have the extension to be loaded from MySQL... |
| 02:42.59 | bkruse | :P |
| 02:43.01 | blitzrage | so I picked the longest billing month :) |
| 02:43.11 | JT | pump up those figures |
| 02:43.52 | blitzrage | 5184000 |
| 02:43.57 | Tond | I have used DynamicSoft SIP proxy before and it worked really well. Simple and easy. But not felxible at all |
| 02:43.58 | NeonLevel | is there a step guide on deploying and use realtime? |
| 02:44.01 | bkruse | Tond: if it gets to intense, make a mysql cluster :D |
| 02:44.11 | blitzrage | NeonLevel: nah, I haven't written one yet |
| 02:44.13 | bkruse | NeonLevel: voip-info.org is your friend |
| 02:44.20 | blitzrage | heh |
| 02:44.23 | JT | mysql cluster, why, so you want to kill yourself? ;) |
| 02:44.28 | NeonLevel | thanks guys |
| 02:44.30 | blitzrage | postgresql is way better |
| 02:44.41 | blitzrage | I've got the DBs clustered too |
| 02:44.43 | bkruse | JT: haha |
| 02:44.47 | Tond | will it make a difference if i don't handle the audio wit Asterisk? you think i'll be bale to pass more calls through? |
| 02:44.53 | bkruse | mysql does crush under intense data flow. |
| 02:45.04 | JT | it makes a huge difference |
| 02:45.06 | bkruse | i would NOT deploy a mysql cluster in an enterprise type situation |
| 02:45.07 | Tond | hrm.. |
| 02:45.21 | bkruse | Tond: i bet so |
| 02:45.32 | bkruse | especially if you kept the same codec and technology across the board |
| 02:45.56 | Tond | I mean I don't really need to handle Audio on a signaling serevr which all that Astersik box is gonna be. to tell other routers and boxes where to send the call (route engine) |
| 02:46.42 | Tond | well if i do pass-throughs and not get * to handle the rtps then I think i should be good.. |
| 02:46.58 | Tond | i ahve never used postgres, is it as easy as MySQL to install? |
| 02:47.10 | bkruse | i think so |
| 02:47.23 | bkruse | you mean have asterisk handle the actual sip sessions and route rtp otherwise |
| 02:47.25 | bkruse | ? |
| 02:48.39 | Tond | ya.. so lets say a call is made, the asterisk will tell the originator where to send the call and let the to end point establish their rtps and when the call is ended log it for billing |
| 02:49.30 | Tond | so it will just handle the signaling and routing, it the voice packets will always be going from point 1 to pint 2. only the signalings will be going 1 <-> asterisk <2-> |
| 02:50.05 | NeonLevel | anyone has used frw-out ? |
| 02:50.11 | Tond | <PROTECTED> |
| 02:51.04 | NeonLevel | anyone has used fwdout, http://www.fwdout.net/ ?? |
| 02:51.14 | Tond | as i said, make asterisk a simple stateless proxy. |
| 02:53.19 | mceGEEK | anyone using sunrocket as their voip provider ? |
| 02:54.03 | blitzrage | never heard of them |
| 02:54.58 | mceGEEK | whats the difference between => and = in sip.conf |
| 02:56.58 | blitzrage | = is an assignment, => is for objects |
| 02:57.32 | blitzrage | typically you don't have to know or care why, but just be aware of the syntax difference on some options |
| 02:57.51 | russellb | actually ........ i think the parser treats them the same |
| 02:58.00 | blitzrage | type=friend, and register => user:pass@host/exten |
| 02:58.04 | blitzrage | really? |
| 02:58.12 | blitzrage | haha, I suppose that's true |
| 02:58.25 | blitzrage | I think register=user:pass@host/exten works the same |
| 02:58.28 | russellb | yeah, it's more just common usage |
| 02:58.36 | Juggie | russellb! |
| 03:01.05 | *** part/#asterisk Ryanw (n=cableguy@ge0-0-15-lns0.207alg.qx21.net) |
| 03:03.41 | Tond | anyone has used the vovida's SIP load balancer? |
| 03:05.13 | benjk | blitzrage, if you look at the code though you will find that => is treated exactly the same as = |
| 03:06.00 | benjk | the distinction between = and => is only in the early documentation, it was probably intended originally but not actually implemented that way |
| 03:10.34 | mceGEEK | Native bridging SIP/4483-081c5b08 and SIP/fwd-081caa60 |
| 03:10.38 | mceGEEK | and it stops |
| 03:10.42 | mceGEEK | i can't hear anything |
| 03:11.04 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 03:11.15 | bkruse | really? |
| 03:11.16 | bkruse | what codecs |
| 03:11.19 | bkruse | ulaw and ulaw/ |
| 03:11.24 | bkruse | ?* |
| 03:11.42 | mceGEEK | yup |
| 03:12.48 | bkruse | wierd, not sure |
| 03:12.54 | bkruse | turn debug up and look for messages ;] |
| 03:12.55 | mceGEEK | oh kruz sorry to trouble you :) |
| 03:13.22 | bkruse | mceGEEK: it is no problem :] i got a couple minutes before i have to do physics homework *bleh* |
| 03:13.29 | bkruse | any kind of debug i can see? :[ |
| 03:14.51 | bkruse | haha didnt realize it was you, ya IM me ;] |
| 03:17.36 | *** join/#asterisk leoncamel (n=leoncame@219.238.107.107) |
| 03:21.06 | *** join/#asterisk aa8ia-away (n=mtindor@noc.qwikhost.com) |
| 03:24.48 | *** join/#asterisk momelod (n=momelod@bas5-toronto12-1088924960.dsl.bell.ca) |
| 03:24.52 | *** part/#asterisk BSDTech (n=RNeese@ppp-71-128-112-135.dsl.irvnca.pacbell.net) |
| 03:25.37 | momelod | hello channel. can anyone point me to some info regarding the setup/config of the digium wildcard te207p |
| 03:26.45 | momelod | the quick setup guide on digium.com says i should be loading the wct2xxp module. but i dont see this module included in any of the zaptel source i've downloaded |
| 03:27.36 | bkruse | momelod: voip-info.org |
| 03:27.37 | bkruse | ;] |
| 03:27.43 | momelod | i didnt see anything there |
| 03:27.54 | momelod | is there a keyword u used? |
| 03:28.49 | russellb | momelod: it's not its own source file |
| 03:28.57 | russellb | momelod: but it will be there after you install zaptel ... |
| 03:29.27 | russellb | it's included in wct4xxp.c |
| 03:30.30 | momelod | okay, are there any tools i can use to interact with the card so that i know its working? |
| 03:30.34 | momelod | like zttool |
| 03:32.03 | *** join/#asterisk goatmilk (n=goatmilk@130-127-44-113.chouse.resnet.clemson.edu) |
| 03:33.07 | momelod | when i search for TE207P all i see it was a news topic a few months ago, but i cant ready there article b/c theres no link to it |
| 03:36.44 | *** join/#asterisk X-Rob_ (n=rob-x@dsl-58-6-69-193.vic.westnet.com.au) |
| 03:38.05 | *** join/#asterisk apardo (n=apardo@87.217.147.146) |
| 03:43.50 | *** join/#asterisk Juggie (n=Juggie@wiley-459-29776.roadrunner.nf.net) |
| 03:46.58 | *** join/#asterisk mindCrime__ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 03:48.52 | mtindor | momelod: did you get my message? |
| 03:52.11 | bkruse | mtindor: need help? |
| 03:52.29 | *** join/#asterisk bmg505 (n=leon@c1-223-14.rndf.isadsl.co.za) |
| 03:52.40 | mtindor | nosir. i was trying to assist momelod by providing him to a link i googled on some other guys configuration of asterisk with a te207 |
| 03:52.54 | mtindor | i had pm'd him but didn't get a response so i asked in pub |
| 03:53.19 | bkruse | ahh, ya PM owned me earlier |
| 03:53.27 | mtindor | thanks tho - you can ask me that question again in a few weeks when I attempt to get a server going |
| 03:53.39 | bkruse | haha, awesome :] |
| 03:53.40 | *** join/#asterisk raina (n=raina@pdpc/supporter/active/ro3159) |
| 03:53.41 | bkruse | ill be here :D |
| 03:53.46 | mtindor | good deal :) |
| 03:53.51 | *** part/#asterisk raina (n=raina@pdpc/supporter/active/ro3159) |
| 03:55.10 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 04:04.33 | mtindor | right there |
| 04:06.44 | *** join/#asterisk sdfkj3478nek (n=philipp@ACAE18DE.ipt.aol.com) |
| 04:08.17 | *** join/#asterisk sdfkj347ek (n=philipp@ACAE18DE.ipt.aol.com) |
| 04:09.13 | *** join/#asterisk asdf123 (n=philipp@ACAE18DE.ipt.aol.com) |
| 04:09.44 | *** join/#asterisk asdf1234 (n=philipp@ACAE18DE.ipt.aol.com) |
| 04:10.24 | *** join/#asterisk qaywsx (n=philipp@ACAE18DE.ipt.aol.com) |
| 04:10.58 | *** join/#asterisk _X-Rob_ (n=rob-x@dsl-58-6-69-193.vic.westnet.com.au) |
| 04:11.50 | *** part/#asterisk qaywsx (n=philipp@ACAE18DE.ipt.aol.com) |
| 04:13.18 | *** join/#asterisk bjohnson (n=bjohnson@i209-195-64-213.cia.com) |
| 04:13.33 | *** join/#asterisk stuq (n=stuq@user-12lcqia.cable.mindspring.com) |
| 04:27.40 | jayvabb | Good evening channel - I have an Adtran TA9xx registered with * and traditional analog phones hanging off the fxs ports. When a call comes in to the TA, from *, and the calling party hangs up before the phone is answered, the phone attached to the TA fxs port continues to ring. Any ideas? |
| 04:29.57 | *** join/#asterisk icyfire0573 (n=icyfire@u1016342.ul.warwick.net) |
| 04:30.07 | bkruse | uh, do you use a page in teh dialplan? |
| 04:30.43 | jayvabb | Not familar with page |
| 04:30.58 | bkruse | k |
| 04:31.01 | bkruse | i know there is a page bug |
| 04:31.31 | bkruse | but im not sure jayvabb |
| 04:31.41 | bkruse | if youve researched it, and think that its asterisk, bugs.digium.com |
| 04:32.14 | jayvabb | OK, TY |
| 04:41.25 | *** join/#asterisk sahafeez (n=sahafeez@ip68-6-223-92.sd.sd.cox.net) |
| 04:42.30 | *** join/#asterisk lengau (n=lengau@c-68-53-53-39.hsd1.tn.comcast.net) |
| 04:42.35 | [TK]D-Fender | jayvabb : Comes in from where? |
| 04:42.56 | bkruse | an adtran TA9xx? |
| 04:43.29 | [TK]D-Fender | bkruse : No, he said a call comes in FROM *, TO the TA. So I want to know the exact origin of the call. |
| 04:44.06 | *** join/#asterisk Mattwj2005 (n=Matt@c-76-17-131-68.hsd1.mn.comcast.net) |
| 04:44.12 | Mattwj2005 | hey guys :) |
| 04:44.48 | Mattwj2005 | how is everyone doing tonight? |
| 04:46.20 | blitzrage | good good... voicemail odbc storage now working! I can get rid of my only NFS partition now :) |
| 04:46.44 | Mattwj2005 | that is cool |
| 04:46.49 | Mattwj2005 | how much storage do you have? |
| 04:47.00 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 04:48.24 | jayvabb | Back. The call comes from * to a sip trunk in the Adtan TA |
| 04:48.28 | [TK]D-Fender | NFS.... No File Security.... *shudder* ;) |
| 04:48.32 | [TK]D-Fender | blitzrage : I just want...... |
| 04:48.41 | file | wasn't me |
| 04:48.47 | blitzrage | ! ! ! |
| 04:49.01 | Mattwj2005 | lol nice |
| 04:49.01 | blitzrage | Mattwj2005: lots of gigs right now |
| 04:49.03 | file | blitzrage: I don't want to know your name |
| 04:49.09 | blitzrage | I just want |
| 04:49.13 | [TK]D-Fender | ! ! ! |
| 04:49.20 | Mattwj2005 | I want to build a 1 TB NAS one of these days :) |
| 04:49.25 | *** join/#asterisk AJayMn (i=AJayMn@24-159-236-181.dhcp.mdsn.wi.charter.com) |
| 04:49.26 | blitzrage | high-5's all around! |
| 04:49.26 | Mattwj2005 | raid 5 |
| 04:49.39 | blitzrage | Mattwj2005: one of these days shortly I'm going to have to get one myself |
| 04:50.02 | blitzrage | right now I need to get the cluster running first :) |
| 04:50.04 | Mattwj2005 | everyone could use at 1 TB |
| 04:50.06 | [TK]D-Fender | blitzrage : Raid 5 cards are getting pretty reasonable, and I've seen it on-board dispicably cheap.... |
| 04:50.07 | blitzrage | close... very close |
| 04:50.14 | Mattwj2005 | cluster? |
| 04:50.18 | blitzrage | aye |
| 04:50.42 | Mattwj2005 | for what? |
| 04:50.42 | file | a cluster of blitzrages |
| 04:50.42 | blitzrage | umm.... for a carrier environment |
| 04:50.42 | file | very scary thought |
| 04:50.42 | blitzrage | obviously |
| 04:50.44 | Mattwj2005 | interesting |
| 04:50.45 | blitzrage | I self divide every 7 years |
| 04:50.54 | blitzrage | Mattwj2005: with DB clustering too |
| 04:51.10 | Mattwj2005 | geez how many nodes? |
| 04:51.16 | blitzrage | i.e. replication, mutl-master |
| 04:51.37 | blitzrage | right now 4, but I'm in building phase. Will test scaling phase shortly. |
| 04:51.47 | blitzrage | but it should scale up to maybe 64 nodes+ |
| 04:51.58 | Mattwj2005 | I am jealous.....I have a 600 Mhz system running Gentoo |
| 04:52.00 | Mattwj2005 | :) |
| 04:52.01 | blitzrage | lol |
| 04:52.17 | blitzrage | I have 2 phyisically separated colocation facilities :) |
| 04:52.20 | russellb | i did development on a 400 mhz pention 2 for a long while :/ |
| 04:52.22 | file | [TK]D-Fender: all your katanas are belong to me |
| 04:52.27 | blitzrage | russellb: for way too long |
| 04:52.30 | russellb | pentium* |
| 04:52.36 | Mattwj2005 | nice |
| 04:52.44 | blitzrage | russellb: it was a chinese made Pentium knockoff |
| 04:52.48 | russellb | lol |
| 04:52.51 | AJayMn | Anyone know of a low startup cost E911 US provider? I have 24 DIDs all over the US and need to offer E911 with them |
| 04:53.22 | blitzrage | AJayMn: I can do that. Msg me if interested. |
| 04:54.48 | Mattwj2005 | hey anyone try the asterisk gui yet? |
| 04:54.56 | blitzrage | I saw some pictures of it once :) |
| 04:55.08 | Mattwj2005 | it looks pretty good :) |
| 04:57.27 | bkruse | its awesome! |
| 04:57.32 | bkruse | warning, only use firefox. |
| 04:57.43 | Mattwj2005 | yeah? |
| 04:57.51 | [TK]D-Fender | file : If you REALLY want them.... I can give them to you ;) |
| 04:58.01 | [TK]D-Fender | file : How many lumps? ;) |
| 04:58.02 | file | ...no |
| 04:59.12 | [TK]D-Fender | file : I've gotten my first cert in my art, and am working on bo a lot right now. Gotta head to Reno-Depot and get myself 2 cut up custom. |
| 04:59.15 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 05:00.15 | file | [TK]D-Fender: ooh |
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| 05:00.42 | blitzrage | hrmmm... ok, so reading doesn't work with my ODBC storage. Just to verify, can I use storage when keeping the configuration set in a flatfile, or does it all have to be in the DB? (i.e. RT voicemail) |
| 05:02.04 | Qwell | blitzrage: you can, yeah |
| 05:02.33 | blitzrage | Qwell: ok, thats what I was thinking. I see it get stored in the DB, but I'm getting: |
| 05:02.34 | blitzrage | [Dec 3 23:57:26] WARNING[23246]: format_wav_gsm.c:109 check_header: Read failed (type) |
| 05:02.35 | blitzrage | [Dec 3 23:57:26] WARNING[23246]: file.c:311 fn_wrapper: Unable to open format wav49 |
| 05:03.00 | blitzrage | I see it go into the DB... maybe its because its pgsql and I have the table wrong.... ? |
| 05:04.53 | yassine | anyone of you guys have an idea why my compiling breaks here : /usr/src/zaptel-1.4.0-beta2/zconfig.h:9:26: error: linux/config.h: No such file or directory ? |
| 05:05.49 | blitzrage | yah, around that line, remove the include for that file |
| 05:06.32 | file | yassine: Ubuntu? FC6? |
| 05:06.42 | yassine | file, debian etch |
| 05:06.43 | blitzrage | #include linux/config.h is what you want to del |
| 05:06.54 | file | hrm, interesting |
| 05:07.03 | yassine | blitzrage, where ? |
| 05:07.12 | blitzrage | in zconfig.h I believe |
| 05:07.20 | yassine | let me have a look there |
| 05:07.25 | blitzrage | it'll be close to the top |
| 05:07.43 | Mattwj2005 | I love gentoo but emerge takes a long time |
| 05:07.55 | blitzrage | I hate gentoo and emerge takes a long time |
| 05:08.02 | Mattwj2005 | lol |
| 05:08.04 | file | I hate gentoo |
| 05:08.07 | Mattwj2005 | why do you hate gentoo? |
| 05:08.18 | Sproket45 | i love gentoo but can't get zaptel to load my clone x100p so i'll ask a question next. hahah. |
| 05:08.18 | yassine | blitzrage, its here : vi zconfig.h +10 |
| 05:08.20 | blitzrage | annoying to use |
| 05:08.25 | file | peer pressure! |
| 05:09.00 | blitzrage | and I've had more servers that were running gentoo crash than those running CentOS |
| 05:09.25 | Mattwj2005 | each to their own |
| 05:09.37 | blitzrage | yep, just run whatever you're comfortable with |
| 05:09.48 | Mattwj2005 | debian, ubuntu, and OpenSuSe are also good |
| 05:09.51 | blitzrage | just not that comfortable with Gentoo, and no reason to learn its weirdisms |
| 05:09.56 | *** join/#asterisk j0 (n=dan@S01060016b6b541d2.va.shawcable.net) |
| 05:10.28 | blitzrage | all OSs have some weirdisms.... I know CentOS's and FC's :) |
| 05:10.33 | momelod | :q |
| 05:10.42 | Mattwj2005 | actually Windows is the best |
| 05:10.56 | *** join/#asterisk niZon (i=bleh@S0106beefd4cecc3d.wp.shawcable.net) |
| 05:10.56 | blitzrage | I run windows inside a window :) |
| 05:11.03 | blitzrage | (a vmware window) |
| 05:11.09 | Mattwj2005 | go Microsoft :P |
| 05:11.10 | blitzrage | its good for running quickbooks... that's about it |
| 05:11.15 | j0 | how accurate is the jitter time in 'iax2 show channels'? it's showing some really high jitter (120ms) but running mtr i only see jitter as high as 20 |
| 05:11.18 | yassine | blitzrage, in zaptel.h +35 too |
| 05:11.32 | j0 | also, when a call is routed to my sip phone, no stats are shown with that command |
| 05:11.33 | blitzrage | is there not a beta3 for zaptel out? |
| 05:11.45 | Mattwj2005 | I was just trying to see what type of reaction I could get out of you guys....it didn't go anywhere though |
| 05:11.46 | blitzrage | gues not |
| 05:11.48 | blitzrage | just beta2 |
| 05:12.02 | blitzrage | although I'd just use the SVN co of the zaptel-1.4 branch to get latest bug fixes |
| 05:12.23 | Mattwj2005 | I have 4 computers only one of them has Windows on it |
| 05:12.23 | blitzrage | Mattwj2005: yah, we don't care about petty shit like other people :) |
| 05:12.29 | yassine | vi torisa.c +22 too |
| 05:12.35 | Mattwj2005 | lol your right :) |
| 05:12.39 | orlock | j0: nfi, but i a having jitter/quality issues with rtp/sip |
| 05:12.40 | Sproket45 | since we're on the distro topic, does anyone have any suggestions for my fresh build of gentoo with 2.6 kernel? I can see my clone x100p installed with lspci -v but DON't see it in /proc/interrupts. When I modprobe all the necessary zaptel modules, I don't see a "1" under /dev/zap/ i've read SO many usenet postings and guides online but can't figure it out. should I use a 2.4 kernel? |
| 05:13.24 | blitzrage | sounds like an issue with udev not having its config updated |
| 05:13.25 | *** part/#asterisk jayvabb (n=jaycox@ip68-98-170-147.dc.dc.cox.net) |
| 05:13.45 | blitzrage | there are sample files for the udev.conf and permissions.d files (I think those are the file names) |
| 05:13.53 | Sproket45 | i only have limited knowledge of udev, but i followed some docs i found online and udev looked like it has the necessary configs |
| 05:14.09 | Sproket45 | what should i see in /proc/interrupts exactly? I know the IRQ or name of card is NOT listed there |
| 05:15.01 | *** join/#asterisk lengau (n=lengau@c-68-53-53-39.hsd1.tn.comcast.net) |
| 05:15.49 | yassine | does this look like a bug to be submitted or am doing somthing wrong : http://rafb.net/paste/results/Rk5D1C26.html |
| 05:17.26 | blitzrage | yassine: might be a bug... |
| 05:17.36 | blitzrage | tzafrir would be the man about xpp :) |
| 05:17.40 | *** join/#asterisk infernix (i=nix@spirit.infernix.net) |
| 05:17.52 | blitzrage | using -netsec? |
| 05:18.08 | blitzrage | pretty sure that's what xpp is for |
| 05:18.13 | Qwell | blitzrage: nope |
| 05:18.19 | blitzrage | really? |
| 05:18.23 | hads | Astribank |
| 05:18.32 | blitzrage | errrr... right... astribank |
| 05:18.35 | Qwell | xpp is the astribank stuff - netsec is the ranch networks security stuff |
| 05:18.39 | Qwell | midcom |
| 05:18.48 | blitzrage | for some reason was picturing the astribank, but thinking ranch networks.... |
| 05:19.11 | blitzrage | must be getting tired :) |
| 05:19.17 | bkruse | ranch networks........ |
| 05:19.32 | blitzrage | bkruse: back to your homework |
| 05:20.27 | JT | is there a way to have a certain extension within a context spawn asterisk to try and rematch that extension within the same context after stripping off some leading digits? |
| 05:21.08 | bkruse | blitzrage: physics :[ |
| 05:21.16 | hads | JT: Local channel? |
| 05:21.19 | file | physics is cooool, so is calculus |
| 05:21.31 | JT | i guess it's a local channel |
| 05:21.35 | blitzrage | calculus sucks ass |
| 05:21.43 | blitzrage | I suck at math soooo badly |
| 05:21.43 | bkruse | blitzrage: i got some pre-cal homework |
| 05:21.57 | bkruse | blitzrage: you can have my open source programs for the TI series calculators :D |
| 05:21.59 | blitzrage | and my mom is like a math genius... guess it skips a generation |
| 05:22.01 | file | blitzrage: remind me never to ask you anything math related |
| 05:22.13 | file | even what time of day it is! |
| 05:22.19 | blitzrage | 3pm |
| 05:22.25 | file | lies! |
| 05:22.30 | blitzrage | 42 o'clock |
| 05:22.32 | file | might be true if I were in Newfoundland |
| 05:22.32 | JT | bzt, it's 1623 |
| 05:22.39 | blitzrage | 4:20am |
| 05:22.57 | orlock | 1165210476 here |
| 05:23.08 | blitzrage | ${EPOCH} :) |
| 05:23.13 | file | #asterisk After Hours - Where VoIP goes out the window and insanity strolls in! |
| 05:23.16 | yassine | blitzrage, its a bug indeed but that seem to be fixed in the trunk |
| 05:23.27 | JT | hads: any ideas? |
| 05:23.32 | blitzrage | yassine: yah, I don't even both with releases.... bleeding edge all the way |
| 05:23.53 | yassine | blitzrage, http://readlist.com/lists/lists.digium.com/asterisk-users/8/40533.html |
| 05:24.19 | hads | JT: I don't quite get what you are trying to do. |
| 05:24.41 | JT | hads: well let's say one extension pattern matches a prefix |
| 05:24.56 | JT | but after the prefix can be any normal number that could be dialed without the prefix |
| 05:25.04 | blitzrage | exten => _X.,n,Dial(Local/${EXTEN:1}@context) |
| 05:25.38 | blitzrage | at least thats the format if it works... I don't quite get what you're doing either |
| 05:25.48 | hads | Yeah, that's something like what I was thinking. Why not just strip the prefix in the extension though? |
| 05:26.04 | blitzrage | I was thinking it sounded more like a Goto() to me |
| 05:26.06 | file | you could use Goto |
| 05:26.09 | blitzrage | :D |
| 05:26.17 | file | blitzrage: get off my brainwaves! |
| 05:26.30 | blitzrage | file: sorry man... getting better at this psychic stuff |
| 05:26.58 | yassine | blitzrage, forget about above |
| 05:27.09 | yassine | the bug still there even in the trunk |
| 05:27.18 | yassine | just checked it out and it remains there |
| 05:29.03 | JT | well |
| 05:29.06 | JT | there are prefixes that do things like force blocking or sending caller id |
| 05:29.36 | JT | i do not want to have to recreate the entire pstn dialplan with all of these prefixes for every phone number pattern |
| 05:29.47 | JT | it seems inefficient and not very manageable/modular |
| 05:29.55 | hads | Well either of those ideas would work then. Local channel or Goto. |
| 05:30.19 | JT | ah ok, never heard of the local channel |
| 05:30.28 | hads | Goto probably is nicer. |
| 05:30.49 | hads | (in this situation) |
| 05:30.58 | JT | alright |
| 05:31.32 | JT | basically i'd need to set a variable and strip digits, then re-execute pattern matching |
| 05:31.51 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 05:32.12 | joelsolanki | <PROTECTED> |
| 05:32.14 | hads | Goto(context|${EXTEN:1}) |
| 05:32.22 | joelsolanki | this is my /etc/asterisk/zapata.conf |
| 05:32.25 | blitzrage | that won't work :) |
| 05:32.32 | blitzrage | always need a priority number |
| 05:32.32 | joelsolanki | when i call from group0 to mobile1 and simulteneously call from other extension to mobile 2 from group3 then both calls are working good. |
| 05:32.42 | joelsolanki | but when i call from group3 to mobile1 and simulteneously call from other extension to mobile2 from group3 then both calls dont work. |
| 05:32.53 | joelsolanki | show g729 shows 2 to be in use. |
| 05:33.03 | joelsolanki | group3 has channel 3 / 4 assinged |
| 05:33.11 | joelsolanki | any hints ? |
| 05:33.22 | hads | blitzrage: True, bad memory. |
| 05:33.30 | blitzrage | going to bed.... |
| 05:33.54 | JT | _1234X.,1,Set(Silent=1) |
| 05:34.17 | JT | _1234X.,2,GoTo(context|${EXTEN:4}) |
| 05:34.19 | JT | wont work? |
| 05:34.29 | hads | GoTo(context|${EXTEN:4}|1) |
| 05:34.46 | JT | ah easy |
| 05:35.04 | yassine | i still get the same error even with 1.2 |
| 05:35.44 | yassine | http://rafb.net/paste/results/Rk5D1C26.html can someone help guys ? |
| 05:35.53 | JT | so i didn't have digit timeouts |
| 05:36.50 | JT | so it would look for pattern matches with all the prefixes without having to specifically write a pattern for every possible pstn pattern |
| 05:38.03 | bkruse | yassine: update your build-tools? |
| 05:38.23 | yassine | bkruse, what do you mean ? |
| 05:38.40 | JT | or disa with no dial tone, that would be cool :) |
| 05:39.24 | bkruse | yassine: make, gcc, etc |
| 05:39.26 | bkruse | just a thought |
| 05:39.35 | bkruse | what system you running yassine |
| 05:39.46 | yassine | bkruse, debian etch |
| 05:39.52 | bkruse | debian ehh? |
| 05:39.56 | bkruse | do you need zaptel 1.4 |
| 05:40.02 | bkruse | do |
| 05:40.09 | bkruse | asterisk build-dep zaptel asterisk libpri |
| 05:40.16 | bkruse | well |
| 05:40.17 | bkruse | just |
| 05:40.21 | bkruse | asterisk build-dep zaptel |
| 05:40.32 | hads | apt-get maybe? :) |
| 05:40.40 | bkruse | haha! |
| 05:40.44 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 05:40.45 | bkruse | s/asterisk/apt-get/g |
| 05:40.47 | bkruse | tired. |
| 05:40.51 | hads | Asterisk on the brain :) |
| 05:40.54 | bkruse | indeed |
| 05:40.55 | bkruse | then try to make && make install zaptel again from src |
| 05:41.00 | yassine | bkruse, i think i solved it |
| 05:41.07 | bkruse | what seemed to be the problem |
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| 05:41.29 | yassine | bkruse, the type bool has been indeed defined twice |
| 05:41.36 | bkruse | wow |
| 05:41.40 | bkruse | if its a for sure bug, report it |
| 05:41.47 | bkruse | of course, make SURE it exists in trunk first. |
| 05:41.57 | yassine | one in /linux/config.h and once in xdefs.h |
| 05:42.05 | bkruse | if you can |
| 05:42.08 | yassine | so i comments one |
| 05:42.10 | bkruse | see if thats the same in trunk |
| 05:42.11 | yassine | and it works |
| 05:42.22 | bkruse | if it is, and its an error, bug report it |
| 05:42.24 | yassine | bkruse, its in the trunk too |
| 05:42.32 | bkruse | bugs.digium.com |
| 05:42.43 | yassine | let me do that at first |
| 05:42.52 | bkruse | k |
| 05:43.33 | bkruse | i gota go, see you guys later, pre-cal calls, ah! |
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| 07:11.44 | Newbie___ | hi all, just wondering, is it possible to change line polarity in zapata.conf? |
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| 07:13.45 | tzafrir | yassine, here? |
| 07:14.02 | yassine | tzafrir, yes |
| 07:14.22 | *** join/#asterisk probonono (n=User@ppp203-125.lns1.adl4.internode.on.net) |
| 07:14.29 | tzafrir | have you managed to build zaptel despite the issue with the xpp driver? |
| 07:14.50 | yassine | yes |
| 07:15.09 | yassine | now trying to install the gui |
| 07:15.30 | tzafrir | ok. It seems that the fix is not trivial (use the kernel typdef instead of ours) as I get some strange errors. |
| 07:15.37 | *** join/#asterisk diclophis-work (n=jbardin@c-69-181-70-186.hsd1.ca.comcast.net) |
| 07:15.49 | diclophis-work | howdy all |
| 07:15.51 | diclophis-work | what is "Packet2Packet" ? |
| 07:16.05 | *** join/#asterisk olinux (i=olinux@ip68-107-4-202.sd.sd.cox.net) |
| 07:16.32 | yassine | tzafrir, thats what i exactly did |
| 07:16.57 | yassine | i commented the definition of the second bool in the zaptel sources |
| 07:17.26 | yassine | but for now im lost in the gui configurations |
| 07:17.37 | tzafrir | the 1.4 gui? |
| 07:17.54 | tzafrir | what seems to be the problem? |
| 07:18.04 | tzafrir | Do you have it listening on port 8000? |
| 07:18.12 | tzafrir | netstat -lntp | grep 8000 |
| 07:18.17 | yassine | let me see |
| 07:18.32 | *** join/#asterisk af_ (n=af@ip-156-221.sn2.eutelia.it) |
| 07:18.37 | yassine | no its not up |
| 07:18.48 | nextime | i was trying the gui yesterday, it randomply cause crash to my test environment * install |
| 07:18.52 | yassine | do i need to start some extra services ? |
| 07:19.58 | nextime | yassine : read the README file, you need the * http enabled and 2 or 3 lines of config |
| 07:20.33 | yassine | nextime, i enabled it in manager.conf and i created the user there |
| 07:20.56 | nextime | yassine : have you restarted * or reloaded the http/manager modules? |
| 07:21.31 | tzafrir | the manager is not a module |
| 07:21.52 | tzafrir | basically, reload. |
| 07:21.53 | nextime | it's the same |
| 07:22.04 | yassine | okay |
| 07:22.41 | tzafrir | please pastebin the [general] section of manager.conf and the [general] section of http.conf |
| 07:22.50 | yassine | okay one sec |
| 07:23.07 | yassine | i have asterisk listning on 8088 but i can not connect to it via browser |
| 07:23.09 | yassine | :s |
| 07:23.24 | *** join/#asterisk VibroMax (n=phisto@83.229.70.154) |
| 07:24.03 | nextime | yassine : point your browser at http://<your * ip>/asterisk/static/config/setup.html |
| 07:24.04 | yassine | now its loading |
| 07:25.07 | tzafrir | yassine, "cannot connect via browser": when you connect, do you get an error page from the "Asterisk server"? |
| 07:25.33 | yassine | no seems like if im hitting the wrong port number |
| 07:25.37 | tzafrir | (with messages that are a bit funnier than the default Apache ones) |
| 07:27.38 | yassine | strange the make checkconfig claims everything is as it should be and that i can connect to the port i specified 8088 |
| 07:27.47 | yassine | but i hit the wall there |
| 07:35.05 | *** join/#asterisk Flusher (n=anon@filer.euroserv.com) |
| 07:38.10 | diclophis-work | anyone know where app_flite.c is ? |
| 07:41.11 | probonono | Is there any support for a serial voice-modem as a FXO for something like a automated response system, or maybe a voicemail system? |
| 07:41.45 | *** join/#asterisk KeRneL (n=ePona@88.240.47.39) |
| 07:48.55 | dlynes_laptop | probonono: only the intel and motorola voicemodems that were rebranded as the X100P and the X101P |
| 07:53.29 | *** join/#asterisk _X-Rob_ (n=rob-x@dsl-124-150-115-171.vic.westnet.com.au) |
| 07:54.54 | benjk | probomone, depends |
| 07:55.04 | benjk | er probomono, I meant |
| 07:58.38 | probonono | benjk, care to elaborate? :) |
| 07:59.37 | benjk | there is a chan_modem module (now deprecated but sources still available somewhere) |
| 08:00.02 | benjk | this is only good for half=duplex, not for real phone conversations |
| 08:00.03 | *** join/#asterisk pourriture (n=pourritu@c-69-242-205-193.hsd1.sc.comcast.net) |
| 08:00.17 | benjk | but you seem to want only answering machine/voicemail type scenarios |
| 08:00.29 | benjk | those could be handled with half-duplex only |
| 08:01.28 | *** join/#asterisk mr50__ (n=scott@dyn-dsl-mb-66-79-234-152.nexicom.net) |
| 08:04.06 | probonono | ah ok, thanks. I wonder if anyone has tried using a pstn modem with audio in/out jacks in combination with a sound card as a codec? |
| 08:04.30 | nextime | anyone is using chan_gtalk or chan_jingle with a recent svn trunk? |
| 08:04.43 | *** join/#asterisk Nobbie (n=corne@wbs-41-208-220-81.wbs.co.za) |
| 08:04.48 | Nobbie | heya =) |
| 08:05.52 | Nobbie | what could cause this problem: in part of my dialplan, i use Wait(20), but execution doesn't continue the the next priority after 20 seconds, the 't' timeout is encountered at least 30 seconds later instead |
| 08:05.53 | *** join/#asterisk Growly (n=himself@125-238-1-176.broadband-telecom.global-gateway.net.nz) |
| 08:06.26 | dlynes_laptop | Nobbie: pastebin your dialplan |
| 08:06.28 | dlynes_laptop | ~pb |
| 08:06.37 | jbot | methinks pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 08:12.42 | Nobbie | pastebin url ? |
| 08:13.12 | EmleyMoor | Nobbie: jbot gave you some |
| 08:13.25 | *** join/#asterisk marexz (n=marexz@195.20.127.222) |
| 08:19.17 | *** join/#asterisk Growly|PissedOff (n=himself@125-236-141-65.broadband-telecom.global-gateway.net.nz) |
| 08:19.23 | *** join/#asterisk petit-toon (n=petit-to@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 08:19.32 | *** join/#asterisk lorinc (n=ang@caracas-3376.adsl.interware.hu) |
| 08:19.45 | Nobbie | pastebin seems broken ? |
| 08:19.55 | EmleyMoor | Which pastebin? |
| 08:20.12 | sbingner | pastebin.ca works for me |
| 08:22.54 | *** join/#asterisk Emrah (n=Me@adslgva0491.worldcom.ch) |
| 08:23.07 | Emrah | 'morning everybody |
| 08:23.19 | *** join/#asterisk tr2x (n=alvar@80-218-150-90.dclient.hispeed.ch) |
| 08:23.38 | Nobbie | ahh, .com doesn't work too well. .ca works ok. |
| 08:23.39 | Nobbie | http://pastebin.ca/266637\ |
| 08:23.45 | Nobbie | without the trailing \ |
| 08:25.24 | Emrah | I just received 40 Snom 360 and 80 7970g... I'm experiencing a disturbing echo on my snom phones... Anyone has an idea if I can enable an echo cancellation somewhere or do something to avoid this echo problem? |
| 08:25.51 | dlynes_laptop | Emrah: is it happening on all the phones, or only certain ones? |
| 08:25.56 | Emrah | all phones |
| 08:26.00 | Emrah | I think |
| 08:26.10 | dlynes_laptop | Emrah: you've got analog paths somewhere in your setup? |
| 08:26.27 | Emrah | Hold on I'll see |
| 08:27.26 | Emrah | No |
| 08:27.44 | Emrah | In the setup section of the Web Interface I have no analog path |
| 08:27.56 | dlynes_laptop | Emrah: so no analog lines or pri lines or t1/e1/j1 lines? |
| 08:28.11 | Emrah | Oh lol |
| 08:28.14 | dlynes_laptop | Emrah: everything's going over voip? |
| 08:28.20 | Emrah | I didn't undestand it like that |
| 08:28.23 | Emrah | Oh no |
| 08:28.43 | dlynes_laptop | Emrah: I would start looking at your analog/pri/... lines first |
| 08:28.51 | dlynes_laptop | Emrah: that's where your echo is probably being generated |
| 08:28.55 | JT | pri isn't analog fwiw |
| 08:29.03 | Emrah | No no |
| 08:29.10 | Emrah | Excuse-me |
| 08:29.11 | dlynes_laptop | JT: Yeah, but echo still gets generated on pri, right? |
| 08:29.17 | Emrah | I didn't express myself correctly |
| 08:29.19 | JT | no |
| 08:29.19 | Emrah | I mean |
| 08:29.21 | JT | maybe at far end |
| 08:29.24 | dlynes_laptop | JT: oh, ok |
| 08:29.25 | JT | not on the pri line |
| 08:29.26 | Emrah | It is a completely internal communication |
| 08:29.28 | florz | dlynes_laptop: No. PRI just connects you to analog lines more often than not. |
| 08:29.40 | Emrah | I'm not talking about any pstn connection |
| 08:29.46 | JT | and every system has analogue in it |
| 08:29.48 | JT | the handpiece |
| 08:29.54 | dlynes_laptop | florz: yeah, but because you've got analog on the far end, you still need hwec or swec on the pri card, right? |
| 08:30.11 | dlynes_laptop | JT: yeah...that's why i asked him if his echo was occurring on every phone, or just some of them |
| 08:30.14 | pourriture | JT ... the users vocal cords are analog too |
| 08:30.19 | pourriture | :| |
| 08:30.25 | florz | dlynes_laptop: That depends on whether someone else is doing EC before the signal reaches you :-) |
| 08:30.29 | Emrah | Hey |
| 08:30.30 | Emrah | Ok |
| 08:30.39 | Emrah | Now I have one which works without any echo |
| 08:30.44 | Emrah | Except when I put the speakerphone |
| 08:31.03 | Emrah | sorry my English is really bad |
| 08:31.06 | EmleyMoor | That sounds environmental |
| 08:31.10 | dlynes_laptop | Emrah: So you can only talk from one snom phone to another? |
| 08:31.18 | Emrah | I have here a 360 working with no echo if I use the handset |
| 08:31.22 | dlynes_laptop | Emrah: You can't talk to anyone outside of the office? |
| 08:31.28 | Emrah | Yes I can |
| 08:31.31 | Emrah | sure I can lol |
| 08:31.44 | Emrah | But I have already 500 7960 / 7940 working with no problem |
| 08:31.48 | dlynes_laptop | Emrah: And how do you achieve that? You said "it is a completely internal communication" |
| 08:31.51 | florz | And the key point, after all, is "analog on a shared medium", not just "analog" ... |
| 08:32.02 | Emrah | No I'm testing only a couple of 360 internally |
| 08:32.30 | *** join/#asterisk KermitTheFragger (n=ktf@118-197.bbned.dsl.internl.net) |
| 08:32.33 | dlynes_laptop | Emrah: are they close to each other? |
| 08:32.36 | Emrah | Currently >I'm only doing internal calls to make tests |
| 08:32.40 | Emrah | no |
| 08:32.46 | dlynes_laptop | Emrah: how close are they? |
| 08:32.59 | Emrah | They arenot in the same room |
| 08:33.03 | dlynes_laptop | ok |
| 08:33.31 | Emrah | Even when I place a call IP 2 IP the echo thing happends... |
| 08:33.47 | Emrah | Anyway thanks a lot for the help :) |
| 08:33.55 | Emrah | I'll see what's the problem with that |
| 08:33.57 | shellshark | use IAX2's jitter buffer |
| 08:35.02 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 08:35.04 | JT | <PROTECTED> |
| 08:35.05 | JT | --- Results after 2732 passes --- |
| 08:35.05 | JT | Best: 100.000000 -- Worst: -2730.517578 -- Average: 98.938861 |
| 08:35.07 | *** join/#asterisk Growly (n=himself@125-236-141-65.broadband-telecom.global-gateway.net.nz) |
| 08:35.08 | JT | go zttest |
| 08:35.17 | JT | zttest really sucks at long term operation |
| 08:36.52 | *** join/#asterisk frogzoo (n=frogzoo@202.155.165.25) |
| 08:37.03 | *** join/#asterisk sam33 (n=sam@LAubervilliers-151-12-81-84.w193-252.abo.wanadoo.fr) |
| 08:37.35 | dlynes_laptop | Try replacing the handsets then |
| 08:39.41 | shellshark | JT: how do you get a 100.000000? |
| 08:39.57 | shellshark | 99.9633 was my best |
| 08:40.18 | shellshark | 99.9389 was my worst |
| 08:40.31 | florz | shellshark: It's basically mere luck. |
| 08:40.41 | shellshark | ah |
| 08:40.49 | shellshark | what does zttest "test" anyway |
| 08:40.58 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 08:45.55 | dlynes_laptop | shellshark: how close to being compatible with an IBM Netfinity your machine is |
| 08:46.21 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 08:46.23 | dlynes_laptop | shellshark: basically if it dips below 99.875, your machine isn't name brand enough |
| 08:46.37 | florz | *g* |
| 08:46.41 | shellshark | i dont get why that would be variable then ;) |
| 08:47.04 | florz | shellshark: Basically, it tests how close zaptel timing and the system clock are |
| 08:47.11 | dlynes_laptop | shellshark: it's digium's way of determining whether you're running a clone, or not |
| 08:47.18 | shellshark | florz: ah |
| 08:47.28 | shellshark | florz: why doesnt it just use the system's RTC anyway? |
| 08:47.36 | dlynes_laptop | florz: in that case, it should always be way off |
| 08:47.51 | ShadowHntr | is Asterisk available on IA-64 (Itanium) arch? |
| 08:47.52 | dlynes_laptop | florz: the system clock is 1024Hz; the zaptel clock is 1000Hz |
| 08:47.55 | *** join/#asterisk daysmen3 (n=primus@host86-141-241-223.range86-141.btcentralplus.com) |
| 08:48.47 | dlynes_laptop | Emrah: if you're getting echo from ip to ip, you've got a handset issue |
| 08:48.51 | ShadowHntr | oh wait |
| 08:48.54 | ShadowHntr | it's in source |
| 08:48.55 | ShadowHntr | DUH |
| 08:49.00 | florz | shellshark: or, more exactly, how close their, erm, "frequency precision"?, is ... |
| 08:49.02 | dlynes_laptop | Emrah: or your echo is generated by the room you're in |
| 08:49.37 | Emrah | dlynes_laptop: Thanks for your answers |
| 08:49.50 | florz | dlynes_laptop: What "system clock" do you mean? |
| 08:50.12 | dlynes_laptop | florz: the system clock chip on your motherboard |
| 08:50.24 | dlynes_laptop | florz: it's a binary not a decimal clock |
| 08:50.38 | florz | dlynes_laptop: Well, that depends pretty much on the hardware architecture, no? |
| 08:51.17 | dlynes_laptop | florz: from what I understood, the AT architecture dictates that it must be 1024Hz |
| 08:51.18 | florz | dlynes_laptop: And at least the PC RTC actually does have second resolution only |
| 08:51.52 | dlynes_laptop | florz: I believe the Pentium IV's and Athlon XP's still use the standard AT clock |
| 08:51.52 | sam33 | Hello all, I'm trying to get a fax in Asterisk, does somebody know a working version of spandsp lib (my Asterisk: 1.2)? |
| 08:52.00 | dlynes_laptop | florz: that might have changed with the new 64-bit chips, however |
| 08:52.01 | shellshark | florz: no it has ms resolution (mille), but iirc it's lacking microsecond |
| 08:52.22 | dlynes_laptop | florz: but even the millisecond doesn't count all milliseconds |
| 08:54.36 | *** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it) |
| 08:54.45 | shellshark | dlynes_laptop: iirc, xeon uses AT clock also, while itanium2 is different |
| 08:54.52 | shellshark | i believe i read that somewhere |
| 08:55.03 | shellshark | opteron is still AT clock too |
| 08:55.08 | florz | shellshark: Uh? http://www.st.com/stonline/products/literature/ds/4557.pdf <- page 12, I don't see any register with a higher resolution than a second!? |
| 08:55.25 | dlynes_laptop | florz: there's two registers on AT-based chips |
| 08:55.51 | dlynes_laptop | florz: erm AT-based chipsets, I mean |
| 08:56.45 | florz | dlynes_laptop: Erm this is an "AT-compatible" component, no? Otherwise, you've got some datasheet of some PC RTC with higher-precision registers? |
| 08:57.06 | dlynes_laptop | florz: one second |
| 08:58.42 | florz | But I was speaking of Linux's system clock anyway, not any hardware RTC ... |
| 08:58.50 | dlynes_laptop | florz: irq0, the system timer gets updated 18.2 times per second by channel 0 of the 8254 system timer; it's used to keep the time-of-day clock updated |
| 08:59.53 | florz | dlynes_laptop: Erm, no, that's a completely different matter. That is what was used by DOS for its system clock - the hardware RTC runs completely independently from the processor. |
| 09:01.04 | dlynes_laptop | florz: And also, the second fast interrupt from the AT real-time clock chip is generated once every 977 microseconds (approx. 1000Hz) |
| 09:01.24 | dlynes_laptop | florz: it's 1/1024th of a second |
| 09:01.45 | dlynes_laptop | florz: adding in the latency, gives you about once every 977 microseconds |
| 09:01.56 | dlynes_laptop | florz: on most machines |
| 09:02.11 | florz | dlynes_laptop: You probably mean jitter? Latency doesn't change the spacing between events. |
| 09:02.33 | dlynes_laptop | florz: i'm thinking more latency within the chipset between timing of interrupts/messages |
| 09:02.37 | florz | dlynes_laptop: But anyway: What does that have to do with the system clock? |
| 09:02.53 | dlynes_laptop | florz: that is the system clock (the AT real-time clock chip) |
| 09:03.46 | dlynes_laptop | It's well documented within Ralf Brown's interrupt list |
| 09:04.34 | *** join/#asterisk VibroMax (n=phisto@83.229.70.154) |
| 09:04.50 | florz | dlynes_laptop: Erm, it is a signal that can be generated by the same device that contains (or rather contained) a PC's RTC, yes. But how does that make it a "system clock"?! |
| 09:05.25 | dlynes_laptop | What do you call the clock in the RTC then? |
| 09:05.32 | dlynes_laptop | Everyone I know calls it the system clock |
| 09:05.48 | dlynes_laptop | It's part of the PC's internal system |
| 09:06.32 | florz | dlynes_laptop: Well, first of all: do you mean "clock" as in "wall time clock" or "clock" as in "synchronisation signal"? |
| 09:06.45 | *** join/#asterisk jm|home (n=jamiem@dilbert.jamiem.com) |
| 09:06.49 | florz | erm, s/wall time clock/wall clock time/ |
| 09:06.54 | dlynes_laptop | florz: sync signal |
| 09:07.11 | dlynes_laptop | Nobbie: The wait is never executed...it gets stuck on Busy() first |
| 09:07.44 | dlynes_laptop | Nobbie: Or Ringing(), depending on which part of the dial plan youi're talking about |
| 09:07.46 | florz | dlynes_laptop: Well, then that certainly isn't any "system clock", as the "system clock" would most likely be the oscillator driving the processor, no? |
| 09:08.59 | dlynes_laptop | Now I have absolutely no idea what you're talking about...an oscillator can be just about anything all the way down to a resistor and a capacitor hooked up to an NE555 chip |
| 09:09.44 | florz | dlynes_laptop: Sure, but what would make this particular oscillator so important/central/whatever, that you'd call it the "system clock"? |
| 09:10.09 | dlynes_laptop | because the real-time clock syncs on it |
| 09:10.27 | dlynes_laptop | Which is what drives the timing of all the kernel timing-sensitive code |
| 09:10.45 | *** join/#asterisk TonyM (n=TonyM@softins.claranet.co.uk) |
| 09:12.43 | florz | dlynes_laptop: Erm, yes and no. The (hardware) RTC of course is driven by the same oscillator. But no, this does not drive all the kernel's timing-sensitive code, and occasionally it doesn't even drive any of it. |
| 09:13.46 | dlynes_laptop | Well, yeah...some of the kernel's timing sensitive code is driven by interrupts from certain hardware, too |
| 09:14.47 | dlynes_laptop | But, anyways...what does what constitute being a system timer or not have to do with what the resolution of the timer that asterisk ultimately needs to sync to? |
| 09:15.55 | florz | dlynes_laptop: That anyway, but not even scheduling is necessarily driven by the RTC's clock anymore - nor is the system clock (as in "linux's idea of wall clock time") necessarily dependent on the scheduling timer alone ... |
| 09:18.24 | florz | dlynes_laptop: Erm, well, actually the fact that it doesn't have to do anything with one another, is the very reason why I said that zttest showing "100%" is mere luck :-) |
| 09:18.30 | *** join/#asterisk HaMYaI (n=HaMYaI@ppp-58.8.5.41.revip2.asianet.co.th) |
| 09:18.52 | Nobbie | dlynes_laptop:, i can see Ringing and Wait(20) being executed, but after 30 seconds a 't' timeout is encountered |
| 09:19.41 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 09:19.47 | HaMYaI | anyone knows what's the appropriate value for "offset samples" in agi -> record_file ? |
| 09:20.08 | florz | dlynes_laptop: That is, given I parsed that question of yours correctly ... =:-) |
| 09:21.44 | *** join/#asterisk VistacDyna (n=vistac@59-125-115-136.HINET-IP.hinet.net) |
| 09:22.01 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 09:25.57 | dlynes_laptop | florz: yeah...my mind and my fingers were typing two different things |
| 09:31.51 | Nobbie | please check: http://pastebin.ca/266696 |
| 09:32.09 | Nobbie | i've added a log trace to show that Wait(20) doesn't do what it's supposed to |
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| 09:36.59 | Nobbie | when running Wait(), will the caller still hear ring tone without me having to run Ringing() before it ? |
| 09:39.06 | *** join/#asterisk solutions (n=ben@124.197.29.4) |
| 09:39.14 | solutions | Hello everybody |
| 09:39.26 | solutions | I was hoping to get some advice |
| 09:39.34 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 09:39.56 | dlynes_laptop | Nobbie: I don't understand why you want to put that wait in after ringing() and busy(), anyways |
| 09:39.59 | dlynes_laptop | Nobbie: what is the purpose? |
| 09:40.00 | sbingner | Nobbie: You have no priority 13, so yes... it would go to timeout after the wait |
| 09:40.22 | Nobbie | sbingner: won't it go to 30 ? |
| 09:40.30 | Nobbie | the next highest priority ? |
| 09:40.32 | sbingner | not without a goto |
| 09:40.42 | Nobbie | for sure ? |
| 09:40.47 | joelsolanki | dlynes_laptop: daniel ? |
| 09:40.53 | dlynes_laptop | joelsolanki: hanjee? |
| 09:41.06 | joelsolanki | hehe |
| 09:41.08 | joelsolanki | how r u |
| 09:41.11 | dlynes_laptop | good |
| 09:41.12 | dlynes_laptop | and you? |
| 09:41.14 | DaPrivateer | I am trying to setup a queue such that the caller hears ringing instead of MOH. is this possible? |
| 09:41.19 | joelsolanki | i m also good :) |
| 09:41.21 | Nobbie | dlynes_laptop: when an extension is busy, i want the caller to hear ringing for 20 seconds, then transfer to reception queue or retry the busy extension |
| 09:42.36 | dlynes_laptop | Nobbie: i c |
| 09:43.07 | sbingner | DaPrivateer: see 'show application queue' |
| 09:43.14 | sbingner | DaPrivateer: in short 'yes' |
| 09:43.17 | DaPrivateer | ty |
| 09:44.23 | JT | shellshark: that's a terrible zttest result |
| 09:44.30 | JT | do you have any hardware in the machine at all? |
| 09:44.32 | solutions | Hello? |
| 09:44.50 | JT | i usually get 99.97 or 99.98 |
| 09:44.58 | JT | not 100, that's only every so often |
| 09:45.22 | *** join/#asterisk slayer192 (n=slayer19@pirus.securax.be) |
| 09:46.21 | shellshark | i get 99.97 sometimes |
| 09:46.32 | shellshark | no zaptel hardware at all |
| 09:46.32 | joelsolanki | daniel |
| 09:46.41 | dlynes_laptop | joelsolanki: ? |
| 09:46.42 | joelsolanki | i m facing some stupid problem |
| 09:46.45 | joelsolanki | yes |
| 09:46.46 | shellshark | maybe i should get an X110P or something just for timing? :) |
| 09:46.58 | joelsolanki | let me describe the problem and give more details |
| 09:47.12 | dlynes_laptop | shellshark: if you want your interrupts slammed and a good %'age of your cpu usage up |
| 09:48.00 | solutions | Hi Guys, |
| 09:48.17 | joelsolanki | http://pastebin.ca/266719 |
| 09:48.18 | solutions | I need a little advice on setting up a new asterisk box |
| 09:48.31 | solutions | Is there any chance someone can give me some guidence |
| 09:48.31 | joelsolanki | this is config for 2 customers |
| 09:48.38 | Nobbie | On an HP DL380 with 3.20GHZ HT Xeon, 3GB of RAM: Best: 99.829102 -- Worst: 97.827148 -- Average: 99.148837 |
| 09:48.45 | Nobbie | that seems bad then .. ? |
| 09:49.16 | *** join/#asterisk mattfletcher (n=matt@heysham-mail.nshc.co.uk) |
| 09:49.30 | shellshark | Nobbie: my results are from a white box, dual xeon 2.8ghz with HT enabled (kernel sees 4 procs), 2GB RAM |
| 09:49.46 | dlynes_laptop | Nobbie: yep |
| 09:49.51 | joelsolanki | if i dial from my extension to mobile1 from group0 and then dial simultenously from my other extension to mobile2 then both calls works gr8 with g729 |
| 09:50.07 | shellshark | wow... 97.827... ouch |
| 09:50.18 | shellshark | my lowest so far has been 99.93 |
| 09:50.20 | joelsolanki | but when i dial from my extension to mobile1 from group3 and then dial simulteneously from my other extension to mobile2 then both calls drops. i cant hear any thing |
| 09:50.31 | dlynes_laptop | I get about 99.97 to 100% |
| 09:50.33 | joelsolanki | is there any thing to do in zapata.conf ? |
| 09:50.41 | dlynes_laptop | Almost all of them are 100% |
| 09:50.52 | dlynes_laptop | But then again |
| 09:50.56 | dlynes_laptop | I use sangoma...not digium |
| 09:50.56 | shellshark | dlynes_laptop: why would a single-port FXO card cause a huge CPU utilization increase? |
| 09:51.07 | dlynes_laptop | shellshark: cause it generates a lot of interrutps |
| 09:51.25 | shellshark | more so than a PRI card? |
| 09:51.27 | dlynes_laptop | shellshark: out of all the zaptel cards out there, it generates the most number of interrupts |
| 09:51.35 | dlynes_laptop | shellshark: yes, quite a bit more |
| 09:51.38 | shellshark | ah |
| 09:51.46 | dlynes_laptop | shellshark: it's just a really poor card |
| 09:51.55 | shellshark | what would be the cheapest route then? |
| 09:51.56 | solutions | I want to use a 2 X TE110P's with 2 X E1 connections on a new asterisk server using only SIP / G729 to connect to this PBX |
| 09:51.59 | mattfletcher | Hello everyone, if I use the Dial command to call two phones (call them phone1 and phone2) concurrently, I find that if phone1 is on the phone, rather than the system just ringing only phone2, the caller hears the busy voicemail from phone1. How can I avoid this? |
| 09:52.01 | dlynes_laptop | shellshark: you're better off to use the ztdummy driver, instead |
| 09:52.01 | shellshark | for a hardware timing solution? |
| 09:52.13 | dlynes_laptop | shellshark: it uses the system timer |
| 09:52.25 | shellshark | dlynes_laptop: using the ztdummy driver now, but i'm getting a lot of clipping |
| 09:52.41 | Nobbie | sbingner: Thansk, setting a Goto(30) at priority 13 fixed the problem. i didn't know the priorities must follow on each other within the same extension/context |
| 09:52.44 | joelsolanki | dlynes_laptop: i would to talk when u r free :) |
| 09:52.45 | shellshark | i've enabled generic jitterbuffer on IAX, and that helped quite a bit, but it's still present |
| 09:53.01 | Nobbie | so how does one improve the zttest stats ? |
| 09:53.15 | shellshark | and on SIP there's not much you can do at all, since asterisk doesnt support a generic "universal" jitterbuffer like openpbx does |
| 09:53.50 | dlynes_laptop | shellshark: there used to be a usb driver for the 2.4 kernel build of zaptel |
| 09:53.50 | sbingner | Nobbie: see fxotune |
| 09:53.54 | sbingner | er |
| 09:53.56 | sbingner | Nobbie: ignore me |
| 09:53.58 | hads | Nobbie: Luck. My home box is a little 600Mhz MiniITX and gets 100.000000% 100.000000% 100.000000% 99.987793% 100.000000% 100.000000% 100.000000% 100.000000% |
| 09:54.01 | dlynes_laptop | shellshark: but I think it disappeared with asterisk 1.2 |
| 09:54.01 | solutions | <PROTECTED> |
| 09:54.06 | Nobbie | maybe my zttest stats are skewed becuase the box is busy at the moment. 300 registrations, 85 activer channel |
| 09:54.59 | dlynes_laptop | joelsolanki: i wouldn't know...I'd need to see a log of it |
| 09:55.05 | dlynes_laptop | joelsolanki: try pastebinning your log |
| 09:55.22 | joelsolanki | daniel u mean CLI output ? |
| 09:55.33 | dlynes_laptop | joelsolanki: or better yet /var/log/asterisk/full |
| 09:55.35 | solutions | <PROTECTED> |
| 09:55.53 | Nobbie | "nice -n -15 zttest" improves it a bit, but worst is still at 99.14. my asterisk runs at nice -10 |
| 09:55.56 | joelsolanki | oh ok |
| 09:55.59 | joelsolanki | let me do that |
| 09:56.38 | mattfletcher | When using Dial to call two phones concurrently, I find that if one of the phones is in a call, rather than the expected outcome (system rings the other phone only) the caller hears the busy voicemail from the phone in the call. Ideas to avoid this? |
| 09:57.18 | Nobbie | mattfletcher: call watiting |
| 09:57.22 | Nobbie | waiting rather ... |
| 09:57.42 | mattfletcher | nobbie: what of it? enable it? disable it? |
| 09:57.55 | Nobbie | mattflether: understand it, then apply it |
| 09:58.16 | Nobbie | eanbling it, will make a phone able to accept multiple incoming calls, but only talk on 1 at a time |
| 09:58.41 | Nobbie | disabling it will cause CONGESTION and your dialplan is probably going to voicemail then |
| 09:59.34 | mattfletcher | i'm not sure how that would help. i don't want multiple calls on one phone. i'm using the command "exten => s,n,Dial(Local/300&Local/301,20,tro)". If 300 is in a call, a new call will go to 300's vm, not ring 301 |
| 10:00.51 | *** join/#asterisk santibiotico (n=santi@37.Red-83-36-42.dynamicIP.rima-tde.net) |
| 10:01.14 | hads | That's odd. |
| 10:01.15 | santibiotico | does anybody know how to change the format of one-touch-record recordings? |
| 10:01.28 | santibiotico | i want the system to save recordings in mp3 format |
| 10:01.36 | santibiotico | i've heard about the TOUCH_MONITOR_FORMAT variable |
| 10:01.48 | santibiotico | but i don't know exactly what i need to do |
| 10:01.51 | Nobbie | mattfletcher: because the call is being sent to voicemail when the busy extension is called. you can either enable Call Waiting to avoid it, disable the voicemail, or build in a ChanIsAvail() check before dialing |
| 10:01.59 | *** join/#asterisk merbanan (n=Anders@136.240.13.217.in-addr.dgcsystems.net) |
| 10:02.32 | *** join/#asterisk aadilismail (n=aadilism@2-237-154-202.wol.net.pk) |
| 10:02.37 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 10:02.37 | Nobbie | mattfletcher: paste your dialplan into pastebin.ca |
| 10:03.25 | aadilismail | wht is difference between "asterisk -r" and "asterisk -vvvvvr" |
| 10:03.27 | aadilismail | ? |
| 10:03.29 | *** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no) |
| 10:03.58 | mattfletcher | nobbie: http://pastebin.ca/266747 |
| 10:04.21 | EmleyMoor | aadilismail: Verbosity - the vs give more output |
| 10:04.58 | Aurs | anyone here used the gigaset SL75? |
| 10:05.02 | Aurs | ..with asterisk |
| 10:06.53 | joelsolanki | daniel can u get online in msn ? |
| 10:07.28 | dlynes_laptop | joelsolanki: maybe |
| 10:07.41 | dlynes_laptop | joelsolanki: I can't remember if I fixed my msn or not :0 |
| 10:07.42 | joelsolanki | hehe u r there |
| 10:07.47 | dlynes_laptop | Yeah, I guess I did |
| 10:08.01 | dlynes_laptop | The binary package didn't work, so I had to compile from source and reinstall |
| 10:08.22 | tzafrir | aadilismail, -vvvvvr sets (core) verbosity to at least 5 |
| 10:08.41 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
| 10:09.01 | aadilismail | thnx |
| 10:09.50 | dlynes_laptop | tzafrir: because windows users have never heard of it? |
| 10:10.21 | shellshark | tzafrir: i use jabber ;) |
| 10:10.21 | tzafrir | dlynes_laptop, they have, actually. Only they think it is called "google talk" |
| 10:10.45 | shellshark | tzafrir: jabber, aim, yahoo, msn, icq.... i'm fairly reachable ;) |
| 10:10.47 | dlynes_laptop | tzafrir: even then...I don't know anybody on windows using jabber or google talk |
| 10:10.58 | dlynes_laptop | tzafrir: even all the peeps that used to use icq are now using msn |
| 10:11.01 | shellshark | dlynes_laptop: my dad uses google talk on windows |
| 10:11.22 | shellshark | dlynes_laptop: half of my 500+ contact list is ICQ users ;) |
| 10:11.25 | tzafrir | I'm trying to get people around me to use jabber, or at least gtalk... |
| 10:11.41 | shellshark | jabber is nice |
| 10:12.08 | tzafrir | A simple method: I attempt to add to my contact list people with gmil accounts |
| 10:12.20 | tzafrir | gamail |
| 10:12.40 | tzafrir | gmail, actually. I bet all three typos have domains |
| 10:12.44 | dlynes_laptop | I've got gmail |
| 10:12.51 | dlynes_laptop | but I don't use google talk |
| 10:13.58 | mattfletcher | nobbie: did u have any further thoughts on that pastebin? |
| 10:14.02 | tzafrir | What MSN client do you use? |
| 10:14.58 | Nobbie | mattfletcher: sorry, only checked now. trying to figure out what causes it to go to voicemail. do you have a call forward busy set maybe ? or is there something on your dialplan causing this ? |
| 10:17.00 | *** join/#asterisk jm|home (n=jamiem@dilbert.jamiem.com) |
| 10:18.11 | *** join/#asterisk xnon (n=xnon@200.8.85.221) |
| 10:18.56 | Nobbie | mattfletcher: you can probably take out that Ansewer() as well |
| 10:21.52 | *** part/#asterisk jerlique (n=jerlique@lnk6.adl5.adsl.esc.net.au) |
| 10:24.45 | mattfletcher | nobbie: where would i set a call forward busy? you have the entire dialplan, i reduced it to this as i diagnosed things. |
| 10:25.01 | santibiotico | does anybody know how to change the format of one-touch-record recordings? |
| 10:25.50 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 10:31.58 | *** join/#asterisk TonyM (n=TonyM@softins.claranet.co.uk) |
| 10:32.14 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 10:33.16 | Nobbie | mattfletcher: you'll need to set asterisk verbose at least 3 and paste the output of a call as well |
| 10:42.26 | *** join/#asterisk The_Ball (n=alex@203.27.181.55) |
| 10:42.53 | The_Ball | what is the problem when lspci shows unknown device for the wildcard card? |
| 10:43.02 | *** join/#asterisk henrique (n=henrique@201-42-26-16.dsl.telesp.net.br) |
| 10:43.08 | *** part/#asterisk henrique (n=henrique@201-42-26-16.dsl.telesp.net.br) |
| 10:43.19 | The_Ball | "Unknown device 6159:0001", not Tiger Jet Network Inc. Tiger3XX Modem/ISDN |
| 10:45.44 | tzafrir | The_Ball, the card is simply not in the standard PCI devices list. |
| 10:46.03 | tzafrir | hmmmm... sorry, probably wrong device |
| 10:47.03 | The_Ball | this tutorial doesn't mention anything if lspci doesn't show the card http://www.asteriskguru.com/tutorials/wildcard_tdm400p.html |
| 10:47.04 | santibiotico | does anybody know how to change the format of one-touch-record recordings? which value should i assign to the TOUCH_MONITOR_FORMAT variable?? |
| 10:47.29 | The_Ball | it's the correct device, if you google 6159 0001 you will find the tdm card |
| 10:47.37 | tzafrir | The_Ball, sorry: this device is actually identified by the wct1xxp driver |
| 10:48.39 | tzafrir | The_Ball, grep 6159 /lib/modules/`uname -r`/modules.pcimap |
| 10:48.48 | tzafrir | Should give you a clue |
| 10:50.08 | The_Ball | ah, thanks |
| 10:51.19 | lilalinux | how do I dial a dynamic iax from extensions.conf, when I don't know the ip? |
| 10:51.21 | tzafrir | In fact, hotplug should auto-load that module for you, if zaptel is installed |
| 10:51.34 | lilalinux | the dynamic asterisk is "registered" |
| 10:52.05 | tzafrir | lilalinux, Dial(IAX2/username) |
| 10:52.12 | lilalinux | thx |
| 10:52.22 | tzafrir | where username is the name of the Asterisk user/friend |
| 10:54.21 | tzafrir | The_Ball, BTW: I'd appreciate it if you would mention a single detail that you consider missing or wrong in http://voip-info.org (unlike the asterisk-guru site: you can actually fix those pages) |
| 10:55.03 | The_Ball | tzafrir, ah, i used google and landed on the guru site |
| 10:55.46 | zoa | you can fix asteriskguru too, just talk to me :) |
| 11:08.09 | tzafrir | zoa, a random browse of the guides there shows much obsolete content. For instance, try searching for "CVS" |
| 11:08.24 | Emrah | Has anyone ever beenable to mmake VoIP calls through a ssh tunnel? |
| 11:09.14 | Emrah | SIP should be difficult... But IAX may work fine, don't you think? |
| 11:09.15 | tzafrir | EmleyMoor, standard VoIP protocols work normally over UDP. And ssh tunnel gives you a relatively long latency (big packets) anyway |
| 11:09.39 | lilalinux | tzafrir: Dec 4 12:07:20 NOTICE[4090]: chan_iax2.c:6911 socket_read: Rejected connect attempt from xxx.xxx.xxx.xxx, who was trying to reach 's@' |
| 11:09.53 | tzafrir | EmleyMoor, sorry, that was for Emrah |
| 11:09.59 | Greek-Boy | anyone here looking for african routes? I am in the process of setting up a route to Tanzania. msg me for details. |
| 11:10.18 | Emrah | tzafrir: Thanks but I'm not able to use anything else than ssh.... |
| 11:10.25 | Emrah | And I want to register onmy Asterisk |
| 11:10.47 | tzafrir | Emrah, there are nice tunnels over UDP. Try using UDP port 53 |
| 11:10.55 | tzafrir | Many firewalls leave it open |
| 11:11.06 | Emrah | Currently my laptop use mylocal proxy at home for my general tcp connections... But Ican't do much with that |
| 11:11.29 | tzafrir | one such tunnel is openvpn |
| 11:11.30 | Emrah | tzafrir: you're incredible :) |
| 11:11.36 | Emrah | I'll try that!!! |
| 11:12.16 | Emrah | But I think that it won't work... All ports are blocked, only the proxy can access the outside |
| 11:12.27 | Emrah | And I use the proxy to connect to my ssh server at home |
| 11:12.32 | Emrah | and then connect to my home proxy |
| 11:14.10 | *** join/#asterisk petit-toon (n=petit-to@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 11:15.52 | *** join/#asterisk slayer192 (n=slayer19@pirus.securax.be) |
| 11:21.02 | *** join/#asterisk cyberarty (n=cyberart@wbs-196-2-122-183.wbs.co.za) |
| 11:21.06 | cyberarty | asdf |
| 11:21.13 | cyberarty | soz |
| 11:22.25 | *** join/#asterisk De_Mon (n=de_mon@fl-69-69-138-64.dyn.embarqhsd.net) |
| 11:24.28 | *** join/#asterisk Emrah_ (n=Me@adslgva0491.worldcom.ch) |
| 11:26.42 | zoa | tzafrir: thanks, i will have it fixed |
| 11:26.58 | merbanan | ummm, how do I submit patches for inclusion ? |
| 11:27.15 | merbanan | to a tracker or to some mailinglist ? |
| 11:27.19 | zoa | i dont open it up to anyone to avoid too much spam (as we already have too much) |
| 11:27.34 | zoa | anybody who wants a login could though. |
| 11:28.37 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 11:28.37 | zoa | there's a guy updating them one by one at the time (layout is fucked a lot too since the new design) |
| 11:30.07 | mattfletcher | can i use gotoif to goto a context, or just an extension within the current context? |
| 11:30.10 | The_Ball | tzafrir, i got it figured out, it was an interrupt problem |
| 11:30.17 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 11:31.32 | cyberarty | The "Asterisk: The Future of Telephony" book link on the www.asterisk.org/support website is broken. Can anyone pls tell me where to get the book? |
| 11:33.19 | dlynes_laptop | cyberarty: O'Reilly |
| 11:33.35 | dlynes_laptop | merbanan: bugs.digium.com |
| 11:33.53 | dlynes_laptop | merbanan: You'll need to fax in a code release agreement, though |
| 11:33.54 | merbanan | thanks |
| 11:34.09 | merbanan | not for a translation patch |
| 11:34.14 | cyberarty | thanks |
| 11:34.15 | dlynes_laptop | merbanan: you can get that from www.digium.com if I remember correctly |
| 11:34.30 | dlynes_laptop | merbanan: Yeah, for any code change, no matter how small |
| 11:34.58 | dlynes_laptop | merbanan: but i'm sure kevin or whoever will let you know that after you've submitted it |
| 11:35.00 | merbanan | there is no code change, just a translation of comments |
| 11:35.09 | dlynes_laptop | ah |
| 11:35.58 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 11:36.45 | brian | merbanan: that's still a code change |
| 11:37.41 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 11:38.18 | merbanan | ok, I don't agree though but I'll see what I can do about that |
| 11:38.36 | *** part/#asterisk cyberarty (n=cyberart@wbs-196-2-122-183.wbs.co.za) |
| 11:40.11 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 11:47.37 | *** join/#asterisk apardo (n=apardo@87.217.116.214) |
| 11:48.17 | *** join/#asterisk Malcom (n=Malcom@85-18-14-13.fastres.net) |
| 11:51.35 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 11:59.25 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 12:01.48 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:02.45 | santibiotico | is there any known problem with zaptel and kernel 2.6.19 |
| 12:02.46 | santibiotico | ?? |
| 12:03.24 | santibiotico | i get an error when trying to compile zaptel that i don't get when using a previous kernel version |
| 12:03.44 | EmleyMoor | What's th'error? |
| 12:04.01 | santibiotico | make[2]: *** [/usr/src/zaptel-1.2.10/zaptel.o] Error 1 |
| 12:04.01 | santibiotico | make[1]: *** [_module_/usr/src/zaptel-1.2.10] Error 2 |
| 12:04.01 | santibiotico | make[1]: Leaving directory `/usr/src/linux-2.6.19' |
| 12:04.01 | santibiotico | make: *** [linux26] Error 2 |
| 12:04.09 | EmleyMoor | I get one when I try to use anything higher than 2.6.8 but that could just be Deban messing me about |
| 12:04.36 | santibiotico | when i use i.e. 2.6.18-2 it all goes ok |
| 12:05.10 | santibiotico | any idea?? |
| 12:05.26 | *** join/#asterisk hi365 (n=hi365@bzq-167-158.dsl.bezeqint.net) |
| 12:09.44 | *** join/#asterisk queuetue (n=scott@MTRLPQ02-1177745854.sdsl.bell.ca) |
| 12:10.25 | queuetue | Hi. My asterisk server appears to be running fine, but is set up to not allow a console. (Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) when I run asterisk -r ) ... Is there a setting for this? |
| 12:10.31 | *** join/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no) |
| 12:11.24 | dlynes_laptop | santibiotico: what's the error? you're not showing the error...the error will be between 1 and 5 lines above your error 1 line |
| 12:11.34 | dlynes_laptop | good morning, royk |
| 12:11.40 | dlynes_laptop | RoyK: erm afternoon |
| 12:11.54 | RoyK | day |
| 12:11.57 | RoyK | g'day |
| 12:12.38 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 12:13.02 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.226) |
| 12:14.42 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 12:15.09 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 12:15.24 | EmleyMoor | queuetue: Are you sure it's running?# |
| 12:16.00 | RoyK | http://karlsbakk.net/piespy/images/asterisk/asterisk-current.png |
| 12:16.13 | queuetue | EmleyMoor: Yes. So sure, I can make calls on it. :) |
| 12:16.19 | EmleyMoor | Hmmm |
| 12:18.19 | dlynes_laptop | RoyK: ah...I see you have another one for me to manipulate :) |
| 12:18.42 | dlynes_laptop | RoyK: but i don't have to manipulate this one...i'm already at the center of attention :) |
| 12:19.00 | RoyK | hehe |
| 12:19.43 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 12:22.20 | *** join/#asterisk CleanerX (n=nix@p54A39258.dip0.t-ipconnect.de) |
| 12:23.54 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 12:31.53 | queuetue | Aso, how does DEBUG get turned on, and how can I turn it off? (My asterisk.full log is packed with DEBUG messages with a verbosity I do not need.) |
| 12:32.30 | shellshark | logger.conf |
| 12:32.40 | shellshark | or logging.conf? cant remember off the top of my head ;) |
| 12:33.59 | queuetue | shellshark: Thanks. |
| 12:34.34 | hi365 | so wada yo think: whats the best voip phone out there? |
| 12:34.34 | queuetue | (It was logger.conf) |
| 12:34.54 | shellshark | hi365: "best" is relative |
| 12:35.09 | shellshark | hi365: depending greatly upon what features you need |
| 12:35.21 | hi365 | true. most relible. nice. blf. |
| 12:35.33 | shellshark | blf? |
| 12:35.40 | hi365 | busy lite |
| 12:35.48 | shellshark | ah righto |
| 12:36.03 | shellshark | check out a polycom IP601 |
| 12:36.40 | hi365 | will do |
| 12:36.50 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 12:37.29 | hi365 | any echo issues? my grandstream 2000's are terabile |
| 12:37.38 | hi365 | (about as bad as my spelling!) |
| 12:39.43 | hi365 | shellshark: any echo issues? my grandstream 2000's are terabile |
| 12:40.24 | shellshark | my grandstream 101 is horrid too |
| 12:40.36 | hi365 | how are the polycoms? |
| 12:40.37 | shellshark | I've got a polycom IP301 sitting here and it's night and day difference |
| 12:40.45 | hi365 | wow. |
| 12:41.03 | shellshark | i've not touched a 601, but I'm sure it'd be about the same voice quality |
| 12:41.12 | shellshark | 601 has a lot bigger screen, more soft keys, etc |
| 12:41.57 | *** join/#asterisk hello007 (i=hello007@81.169.227.211) |
| 12:42.00 | hello007 | anu one know how to configure ISA 2004 for asterisk to work on outbound and inbound calls on sip ? |
| 12:43.51 | *** join/#asterisk henrique (n=henrique@201-42-26-16.dsl.telesp.net.br) |
| 12:48.33 | mosty | what is isa 2004? |
| 12:48.53 | hi365 | shellshark: interesting that they dont list their phones on their site |
| 12:49.02 | shellshark | they do |
| 12:49.10 | shellshark | SoundPoint IP series |
| 12:50.06 | hi365 | i dont c it under products |
| 12:51.56 | *** join/#asterisk coppice (n=chatzill@40.168.17.210.dyn.pacific.net.hk) |
| 12:55.36 | zoa | hey ho coppice |
| 12:55.38 | razu | does any variable tell which caller hanged up the phone ? |
| 12:56.04 | *** join/#asterisk dwmw2_gone (i=ctrlprox@baythorne.infradead.org) |
| 12:57.16 | hello007 | MICROSOFT isa FIREWALL |
| 12:57.50 | hello007 | i changed the firewall because i was unbale to receive sip calls |
| 12:58.11 | hello007 | now i m able to receive sip calls , but i m unable to make outgoing sip calls |
| 12:58.15 | mosty | hello007: are you doing NAT too, or just firewalling? |
| 12:58.26 | hello007 | no i m doing nat too |
| 12:58.59 | mosty | if you can't make outgoing calls, the first thing i would check is that you are registered ok |
| 12:59.23 | hello007 | yes i m registerd ok |
| 12:59.33 | hello007 | sip show registry show it |
| 12:59.54 | hello007 | and if i call the pbx from outseide i enter the IVR |
| 13:00.07 | hello007 | but i can not call from the inside |
| 13:00.10 | *** join/#asterisk olinuXP (i=olinux@ip68-107-4-202.sd.sd.cox.net) |
| 13:00.40 | coppice | zoa: hi |
| 13:00.58 | mosty | hello007: what does the asterisk console show with debugging and verbose set to say 10, when you make an outgoing call? |
| 13:01.11 | hello007 | is it firewall/port issue? |
| 13:02.00 | hello007 | it shows an i can hear all circuit are busy |
| 13:02.17 | hello007 | immeditaely |
| 13:03.06 | hello007 | i have my asterisk with public ,and my polycom phones behind a nat with private ip |
| 13:04.20 | *** join/#asterisk Tili (n=tili@202.133.67.90) |
| 13:04.51 | synthetiq | polycom + nat = trouble |
| 13:05.00 | hi365 | shellshark: thanks. got the info. it look good. what is the 650? what is the dh thing? |
| 13:05.21 | mosty | hello007: is your asterisk server behind the NAT? ie on the same lan as your phones? |
| 13:05.29 | *** join/#asterisk svenna_ (n=svenna@p548D07A2.dip0.t-ipconnect.de) |
| 13:06.12 | *** join/#asterisk nicky_ (n=doskey@82-39-244-173.cable.ubr02.jarr.blueyonder.co.uk) |
| 13:06.20 | hello007 | no the asterisk have a public ip,and only the phone are behind nat with private ips |
| 13:13.54 | *** join/#asterisk frogzoo (n=frogzoo@202.155.165.25) |
| 13:17.10 | *** join/#asterisk aaqq (n=yytrttry@201-43-57-216.dsl.telesp.net.br) |
| 13:18.42 | Jo211 | hello, i'm getting trouble with a 110p card connect to a pri telecom, perhaps someone can help me |
| 13:19.23 | Jo211 | Dec 4 11:18:48 NOTICE[10836] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 |
| 13:19.23 | Jo211 | Dec 4 11:18:49 NOTICE[10836] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 |
| 13:19.23 | Jo211 | Dec 4 11:18:51 NOTICE[10836] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 |
| 13:19.34 | Jo211 | i all calls are dropped |
| 13:19.46 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
| 13:20.29 | Jo211 | zaptel-1.2.11 / asterisk-1.2.13 |
| 13:20.37 | Jo211 | zaptel.conf: |
| 13:20.38 | Jo211 | span=1,1,0,ccs,hdb3 |
| 13:20.38 | Jo211 | bchan=1-15 |
| 13:20.38 | Jo211 | dchan=16 |
| 13:20.38 | Jo211 | bchan=17-31 |
| 13:20.38 | Jo211 | loadzone = us |
| 13:20.40 | Jo211 | defaultzone = us |
| 13:20.59 | Jo211 | any idea? |
| 13:23.17 | *** part/#asterisk pourritur1 (n=pourritu@c-68-58-198-220.hsd1.sc.comcast.net) |
| 13:25.12 | tzafrir | ~pb |
| 13:25.24 | 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 |
| 13:26.19 | Chris-NB | hi |
| 13:26.28 | Chris-NB | can someone tell me, what that err means? |
| 13:26.31 | Chris-NB | zt_pri_error: 2 !! Got a UA, but i'm in state 1 |
| 13:29.44 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 13:30.17 | shmaltz | anybody know of a simple call log system that reads the cdr records from the csv files? |
| 13:31.47 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 13:31.50 | SomeOne1 | RoyK: sup |
| 13:32.28 | *** join/#asterisk The_Ball (n=alex@203.27.182.167) |
| 13:33.31 | *** join/#asterisk gerhard7 (n=gerhard@82-171-117-191.dsl.ip.tiscali.nl) |
| 13:36.13 | *** join/#asterisk knathraak (n=zach@151.196.142.242) |
| 13:36.16 | flujan | Hi guys, could you please check if my dialplan is right? http://pastie.caboo.se/25651 |
| 13:36.27 | flujan | I am not sure about the gotoif and the len command... :) |
| 13:36.51 | knathraak | hi, all. I'm needing some help with my queues.conf |
| 13:37.17 | mosty | knathraak: be more specific |
| 13:37.18 | knathraak | specifically I'm wondering if it is possible to set global variables in queues.conf, or to import them from somewhere else |
| 13:37.52 | mosty | knathraak: what kind of variables do you want to set? |
| 13:39.00 | knathraak | mosty: the members of my queues consist of a series of Zap channels (fxo) dialing cell phone numbers. Since some of the cell phones are members of more than one queue, i'd like to set them as variables in a global section, like you can in extensions.conf |
| 13:39.17 | knathraak | mosty: this doesn't seem to work, though |
| 13:39.44 | mosty | knathraak: wouldn't you just have member => settings in each queue definition? |
| 13:40.01 | Jo211 | any hw support to 110p? geeting chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 |
| 13:40.30 | knathraak | mosty: that's what I currently have. but i'd like to have member => ${cellphone1} |
| 13:40.58 | knathraak | mosty: that way I could use that same variable in more than one queue definition, and if i need to change it, i could just change it in one place |
| 13:41.15 | mosty | knathraak: you could probably do it with #include |
| 13:41.35 | *** join/#asterisk vooduhal (n=vooduhal@tc-proxy2.catt.com) |
| 13:41.37 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 13:42.00 | knathraak | mosty: so (1) to includes work in queues.conf, and (2) where would I put it, and what would it look look like? |
| 13:42.43 | mosty | #include "cellphone-zap-channel.conf" |
| 13:42.54 | knathraak | mosty: would i put the include at the beginning of each queue definition? |
| 13:43.07 | mosty | put that in each queue, then you would just have to edit cellphone-zap-channel.conf if you want to change them all |
| 13:43.25 | vooduhal | Hey quys. Quick qustion. Do the zaptel drivers need to be reloaded in order for the rx and tx gains in zapata.conf to take effect or can you just do a reload? |
| 13:43.47 | knathraak | mosty: nice... do includes work with queues.conf, that you know of? |
| 13:43.56 | mosty | #include works in all asterisk config files |
| 13:44.10 | knathraak | mosty: nice... thanks....i'll give this a try. |
| 13:46.19 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 13:48.47 | knathraak | mosty: I created a file called "cellphones.conf" and at the beginning of my queue definition, I specified: #include "cellphones.conf" |
| 13:49.29 | knathraak | mosty: cellphones.conf looks like [globals] |
| 13:49.29 | knathraak | CELLPHONE=Zap/2/18885551212 |
| 13:49.58 | knathraak | mosty: is this what you were describing? |
| 13:50.09 | *** join/#asterisk arjan (n=arjan@82-204-26-196.dsl.bbeyond.nl) |
| 13:50.10 | *** join/#asterisk AuPix (n=root@mail.aupix.com) |
| 13:50.15 | arjan | Hello |
| 13:50.23 | vooduhal | Hey quys. Quick qustion. Do the zaptel drivers need to be reloaded in order for the rx and tx gains in zapata.conf to take effect or can you just do a reload? |
| 13:50.29 | *** join/#asterisk markit (n=konversa@host119-245-static.72-81-b.business.telecomitalia.it) |
| 13:50.34 | arjan | I'm trying to make asterisk to connect to a sip service that I have with the register => command |
| 13:50.45 | markit | hi, seems that asterisk 1.2svn does not recognize dtmf anymore, and I've enabled debug channel sip, and it logs 2 events when I press *2... how can it be then? |
| 13:50.57 | arjan | Only this service wants me to connect to port 38383 instead of the standard port |
| 13:51.01 | markit | << [ TYPE: DTMF (1) SUBCLASS: * (42) ] [SIP/1001-081bfaa0] |
| 13:51.01 | markit | << [ TYPE: DTMF (1) SUBCLASS: 2 (50) ] [SIP/1001-081bfaa0] |
| 13:51.07 | markit | what could it be? |
| 13:51.09 | arjan | How can I put that in the register => command? |
| 13:52.27 | markit | arjan: I don't know, have you checked the wiki if the syntax address:port is supported by "register"? |
| 13:52.54 | fantasio | someone here knows how to connect a asterisk (digium t110p) with a Siemens Hicom? |
| 13:53.06 | zoa | we did before i think |
| 13:53.15 | fantasio | i need some assistance by configuring the Siemens :D |
| 13:53.48 | arjan | yes, I checked that, I don't see it anywhere |
| 13:54.32 | markit | http://www.voip-info.org/wiki-Asterisk+config+sip.conf |
| 13:54.53 | markit | arjan: read there, there is a "port" option |
| 13:55.00 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 13:56.14 | fantasio | zoa: You did? |
| 13:56.35 | arjan | ok tnx |
| 13:57.44 | vooduhal | Hey quys. Quick qustion. Do the zaptel drivers need to be reloaded in order for the rx and tx gains in zapata.conf to take effect or can you just do a reload? |
| 13:59.01 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 13:59.09 | *** join/#asterisk mistermocha (n=espresso@adsl-75-40-188-46.dsl.irvnca.sbcglobal.net) |
| 13:59.19 | vooduhal | exit |
| 13:59.24 | *** join/#asterisk |oranjia| (n=kvirc@dsl-243-168-195.telkomadsl.co.za) |
| 13:59.38 | |oranjia| | helloo peeps :) |
| 13:59.42 | mistermocha | hi... |
| 13:59.53 | |oranjia| | mistermocha: :) |
| 14:00.00 | mistermocha | this is strange... outbound dialing isn't presenting a phone number to my pri line... any thoughts? |
| 14:00.33 | mistermocha | it's not presenting a dialed number.... it is presenting a caller ID |
| 14:00.38 | |oranjia| | try NoOping CALLERID(num) |
| 14:00.45 | |oranjia| | ah |
| 14:00.49 | |oranjia| | you mean the dnid? |
| 14:00.59 | *** join/#asterisk docelmo (n=vircuser@c-68-32-143-73.hsd1.de.comcast.net) |
| 14:01.24 | *** join/#asterisk Tili (n=tili@202.133.67.3) |
| 14:01.44 | mistermocha | *blink* it's early... which is the dnid? |
| 14:01.48 | |oranjia| | Can someone help me? for some reason using Progress() and then Playback with noanswer doesn't work |
| 14:02.05 | |oranjia| | mistermocha: i am not sure you get the dnid on zap channels |
| 14:02.14 | |oranjia| | but it works with sip...its gives the number dialled |
| 14:02.24 | |oranjia| | but you can also use ${EXTEN} |
| 14:02.33 | |oranjia| | i think |
| 14:04.18 | mistermocha | how else is the pri going to know the number dialed? |
| 14:04.40 | lilalinux | what does "No authority found" mean? |
| 14:04.45 | Jo211 | any hw support to 110p? geeting calls dropped and: chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 |
| 14:04.48 | hi365 | i dont understand: does the polycom 601 have a blf feature? |
| 14:07.33 | mistermocha | wait... |
| 14:07.40 | mistermocha | where do I set the dnis digits? |
| 14:07.49 | mistermocha | hi365: I don't think so |
| 14:08.10 | hi365 | damn. so whats the best phone taht does? |
| 14:08.43 | mistermocha | aastra |
| 14:09.33 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 14:10.03 | hi365 | wow this is confusing! |
| 14:10.23 | hi365 | b4 sum1 mentiond the polycom |
| 14:12.00 | markit | anyone can tell me how to svn check the 1.2 stable of some days ago? just to test when dtmf broken |
| 14:12.22 | *** part/#asterisk BlackRatchet (n=ratchet@curleypu.be) |
| 14:14.04 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:14.14 | SomeOne1 | sum1? |
| 14:14.18 | SomeOne1 | i am sum1!! |
| 14:14.29 | SomeOne1 | hi365: i did not mention polycom! |
| 14:15.25 | hi365 | lol. didnt relise that sum1 is called sum1! |
| 14:15.25 | hi365 | http://webpages.marshall.edu/~hartwel1/humor/misc/everybody_somebody_anybody_nobody.html |
| 14:18.45 | knathraak | mosty: i tried your idea of including a cellphones.conf in the queue definition. |
| 14:18.58 | knathraak | mosty: unless i'm doing it wrong, it doesn't seem to work |
| 14:19.40 | mosty | did you include quotes? #include "foo" |
| 14:19.47 | knathraak | mosty: my cellphones.conf has a single entry: CELLPHONE1=Zap/2/18885551212 |
| 14:20.01 | *** join/#asterisk peterme2005 (n=petere@browse.net-serv.co.uk) |
| 14:20.04 | mosty | don't try and set a variable like that |
| 14:20.18 | knathraak | mosty: okay how? |
| 14:20.24 | mosty | in cellphones.conf put member => Zap/2/12343543etc |
| 14:20.49 | peterme2005 | Hi guys does anyone know how to perform an attended transfer operation that doesnt involve using meetme or parked calls? |
| 14:21.15 | peterme2005 | if anyone can share a bit of time for a private chat that would be great!!! |
| 14:21.23 | knathraak | mosty: okay i see... but that still doesn't solve my problem of needing to use variables |
| 14:21.50 | mosty | knathraak: but it solves the "i want to edit this in once place" problem |
| 14:22.35 | mosty | ie you don't need to use variables |
| 14:22.51 | knathraak | mosty: hmm... i guess I could do cellphones_1.conf and cellphones_2.conf, etc, and just include the files I want to use in each queue. |
| 14:23.02 | knathraak | mosty: is that what you mean? |
| 14:23.08 | mosty | yes |
| 14:23.38 | mistermocha | dumb question... what denotes a comment in a .conf file? |
| 14:23.54 | mistermocha | in particular a trunk file |
| 14:23.57 | knathraak | mistermocha: semicolon, i believe |
| 14:24.09 | mistermocha | I'll try .. thx |
| 14:25.12 | Tili | anybody every got this |
| 14:25.15 | Tili | wanpipe1: Critical: AFT Chip Security Compromised: Disabling Driver!(0EBA0189) |
| 14:25.15 | Tili | wanpipe1: Please call Sangoma Tech Support (www.sangoma.com)! |
| 14:25.16 | Tili | wanpipe1: Error: Card Critically Shutdown! |
| 14:25.16 | Tili | wanpipe1: Clearing E1 Interrupts |
| 14:25.31 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 14:27.08 | peterme2005 | guys if anyone knows how to perform an attended transfer in asterisk please message me...desperate for some help or a point in the right direction |
| 14:27.20 | mosty | see features.conf |
| 14:27.46 | *** part/#asterisk [Wiebel] (i=wiebel@wiebel.nl) |
| 14:29.32 | shido6 | http://www.voip-info.org/wiki/index.php?page=Asterisk+config+features.conf |
| 14:29.48 | peterme2005 | ive had a look at the feats.conf |
| 14:29.53 | shido6 | see atxfer |
| 14:29.57 | *** join/#asterisk henrique (n=henrique@201-42-26-16.dsl.telesp.net.br) |
| 14:30.01 | peterme2005 | not what i want though apparently it puts too much load on the network |
| 14:30.23 | peterme2005 | so one of the guys here wrote a patch that simulates DTMF tones |
| 14:30.29 | peterme2005 | with this -t flag |
| 14:30.31 | peterme2005 | but...... |
| 14:31.07 | peterme2005 | again on the network it sucks the upsteam |
| 14:31.08 | peterme2005 | :-( |
| 14:31.12 | peterme2005 | therefore |
| 14:31.24 | peterme2005 | what we want is the phones to do the work for us |
| 14:31.32 | *** join/#asterisk allankardec (n=root@201.45.22.130) |
| 14:31.37 | peterme2005 | so joe blogs phones reception |
| 14:31.43 | peterme2005 | reception picks up |
| 14:31.51 | peterme2005 | joe blogs asks to speak to the manager |
| 14:31.52 | shido6 | spa-941 can do attended xfers on the phone |
| 14:32.06 | peterme2005 | i know you can do xfers on the phone |
| 14:32.12 | peterme2005 | i want to do it via the manager or CLI |
| 14:32.21 | peterme2005 | and i cant figure out how to do this elegantly |
| 14:32.38 | peterme2005 | i want to do the xfer on the phone for my desktop application |
| 14:32.52 | shido6 | some number combo to execute the script using user input |
| 14:33.18 | peterme2005 | im not sure i follow that last bit shido6 |
| 14:33.55 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:34.14 | shido6 | you could use agi, or the "System" application.... or... |
| 14:34.29 | peterme2005 | well we are using asterisk -java |
| 14:34.44 | peterme2005 | which if the interface for java to asterisk |
| 14:34.53 | peterme2005 | but again we can do a blind transfer |
| 14:35.07 | peterme2005 | but we want something that doesnt use this -t flags |
| 14:35.18 | shido6 | can you show me how you are doing the blind xfer? |
| 14:35.33 | peterme2005 | ok hang on need to give me 1 min |
| 14:36.56 | peterme2005 | i think my colleague uses a REDIRECT |
| 14:38.19 | peterme2005 | http://asterisk-java.org/0.3-m1/apidocs/org/asteriskjava/manager/action/RedirectAction.html |
| 14:38.30 | peterme2005 | from looking at the code thats what he has implemented |
| 14:39.58 | peterme2005 | its all a bit annoying really at this point |
| 14:40.39 | peterme2005 | ive found loads of ppl asking about this sort of issue online but few have answers |
| 14:41.14 | shido6 | so you want to call someone speak with them for a bit.... then enter some number combination to put them on hold and call someone else& speak with them for a bit then join the two |
| 14:41.27 | peterme2005 | pretty much yea |
| 14:41.39 | peterme2005 | but with minima use or workload for asterisk |
| 14:41.54 | peterme2005 | i basically want to perform the same operation that the phones use |
| 14:42.31 | shido6 | oh, and the challenge is not to use "t or T" |
| 14:42.36 | *** join/#asterisk jm|home (n=jamiem@dilbert.jamiem.com) |
| 14:42.39 | peterme2005 | YEP |
| 14:43.25 | shido6 | we have Progomate |
| 14:43.27 | shido6 | err |
| 14:43.43 | peterme2005 | progomate? |
| 14:43.48 | shido6 | we have Originate, Redirect, Transfer, Command to use... |
| 14:43.56 | peterme2005 | yep |
| 14:43.58 | shido6 | u originate the cal on the phone... |
| 14:44.12 | shido6 | ahhh |
| 14:44.22 | shido6 | so you will need to start with executing your application first |
| 14:44.32 | shido6 | since you dont want to straight into using the "Dial" application |
| 14:44.42 | peterme2005 | well the application will act as the middle person.... the receptionist if you will |
| 14:45.11 | peterme2005 | then she will transfer the incoming caller "joe blogs" say to the manager "john smith" |
| 14:45.12 | shido6 | is the receptionist asking for input and asking a question? |
| 14:45.35 | peterme2005 | yes she will say who do you want to speak to and he will say "john smith the manager" |
| 14:45.46 | *** join/#asterisk shinux__ (n=shinux@196.220.26.93) |
| 14:45.46 | peterme2005 | then she will hit the transfer button |
| 14:46.10 | peterme2005 | be but through to john smith and say ive got joe blogs on the phone do you want to take it ....he accepts |
| 14:46.10 | shido6 | do you have a speech to text application or will the user input DTMF in response to some prerecorded message? |
| 14:46.34 | peterme2005 | and when the receptionist puts down the physical phone then behind the scenes joe blogs is connected to john smith |
| 14:46.57 | peterme2005 | no dtmf stuff...well not yet anyway but there might be later not sure |
| 14:47.16 | shido6 | err... I thought the application was replacing the live person........ is there a live person asking this which is doing the transfer? |
| 14:47.26 | shinux__ | hello guys ... we are trying out asterisk in our office... i wonder if we can connect it to our existing pbx in the office? and if theres a guide for that somewhere? |
| 14:47.46 | shido6 | shinux... what kind of interfaces does your existing pbx have? |
| 14:47.52 | shinux__ | so as to benefit from the layout we already have |
| 14:47.59 | shinux__ | rj11 |
| 14:48.10 | shinux__ | the regular analog kind |
| 14:48.12 | shido6 | FXO or FXS? |
| 14:48.26 | shido6 | for handset or for telephone lines from the co |
| 14:48.35 | shinux__ | for telephone lines |
| 14:48.41 | shinux__ | from the company |
| 14:48.44 | shido6 | yes |
| 14:48.46 | shinux__ | sorry |
| 14:48.47 | shido6 | you can |
| 14:48.49 | shido6 | how many |
| 14:48.50 | shido6 | ? |
| 14:48.53 | shinux__ | let me make that striaght |
| 14:48.55 | peterme2005 | no the receptionist uses out application alongside the phone.... but we (the application) override the phone because when a call comes through to that extension the application assumes control and displays the call on the inbound list |
| 14:49.03 | peterme2005 | the receptionist then clicks answer |
| 14:49.06 | peterme2005 | her phone rings |
| 14:49.12 | shido6 | shinux__: how many? |
| 14:49.16 | shinux__ | about 6 extesions |
| 14:49.24 | shinux__ | and 3 lines from the telco |
| 14:49.41 | *** join/#asterisk mo3 (n=odeame@141.155.251.202) |
| 14:50.14 | shinux__ | i think thats right shido6 |
| 14:50.16 | peterme2005 | any ideas shido6 |
| 14:50.47 | *** join/#asterisk syn (i=syn@kenobi.sceen.net) |
| 14:50.48 | syn | hello |
| 14:51.06 | syn | is three-way calling a feature managed by the PBX or the client phones (or both) ? |
| 14:51.32 | shido6 | shinux__: you have a few options.... http://www.thevoipconnection.com/store/catalog/product_16276_Digium_Wildcard_TDM2400P.html or http://www.thevoipconnection.com/store/catalog/product_16185_Digium_TDM400P_FXO_FXS_Interface_Card.html configured with 4 FXO and another one with 2 FXO |
| 14:51.36 | nibbler_de | syn: depends on the origin of the individual calling parties |
| 14:51.50 | peterme2005 | guys ill be right back |
| 14:52.04 | syn | nibbler_de: let's say it's asterisk's job then |
| 14:52.07 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.235.181.Dial1.SanJose1.Level3.net) |
| 14:52.11 | shido6 | you can use SIP FXO gateways... |
| 14:52.22 | syn | nibbler_de: i didn't see any way to configure this (except meetme, but this is not what i want :/) |
| 14:52.24 | shido6 | you can use Digium Cards |
| 14:52.46 | syn | shido6: are you suggesting this to me ? |
| 14:52.54 | nibbler_de | syn: it's actually very simple - if you call two people via sip your phone does the job |
| 14:53.03 | shido6 | no, this is for shinux__ |
| 14:53.09 | syn | shido6: ok |
| 14:53.10 | syn | :) |
| 14:54.21 | syn | nibbler_de: when does asterisk do the job then ? |
| 14:55.11 | knathraak | got a question about call queues... |
| 14:55.32 | knathraak | members consist of zap channels which dial technicians' cell phones |
| 14:55.59 | knathraak | i need to set priorities on the members such that it tries the technicians in the same order each time |
| 14:56.33 | knathraak | is this possible? if so which strategy would work |
| 14:57.26 | *** join/#asterisk delta (i=delta@217.113.15.254) |
| 14:58.14 | shinux__ | ok shido6 |
| 14:58.36 | shinux__ | so shido6... how is it configured? |
| 14:59.45 | shinux__ | is there a guide somewhere i can use? |
| 15:03.08 | knathraak | hi, i'm wondering if anyone knows how to make a queue dial its members in the same order every time, regardless of who last answered a call. round robin seems to skip the most recently used member |
| 15:10.58 | *** join/#asterisk MASK666 (n=pRiMe_eV@ip68-230-133-70.ri.ri.cox.net) |
| 15:11.06 | *** join/#asterisk Katty (n=angela@hera.copi-rite.com) |
| 15:11.31 | *** join/#asterisk Katty (n=angela@hera.copi-rite.com) |
| 15:12.08 | *** join/#asterisk elriah (i=elriah@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 15:12.36 | *** join/#asterisk slayer192 (n=slayer19@pirus.securax.be) |
| 15:12.43 | elriah | Hi all. Is there a way to set a variable in a extensions.conf context without doing exten => x,x,set() ?? I need to set a variable for the entire context for outbound calls ... |
| 15:12.51 | elriah | asterisk 1.2.x |
| 15:13.01 | Katty | morning. |
| 15:14.22 | nibbler_de | syn: when the lines terminate in asterisk - for example you have a pstn call and a phone connected via misdn in nt-mode and one via sip |
| 15:15.34 | knathraak | elriah: this might help: http://www.voip-info.org/wiki-Asterisk+variables |
| 15:15.45 | syn | nibbler_de: ah, ok |
| 15:15.55 | elriah | Thanks. |
| 15:15.56 | syn | nibbler_de: so i guess there is no need to configure anything, right ? |
| 15:16.01 | syn | it just works out of the box |
| 15:16.07 | *** join/#asterisk Katty_ (n=angela@hera.copi-rite.com) |
| 15:16.17 | *** join/#asterisk ming_zym (n=zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 15:16.58 | nibbler_de | syn: yup |
| 15:17.05 | syn | nice |
| 15:17.07 | syn | nibbler_de: thanks :) |
| 15:17.16 | syn | have a nice day :) |
| 15:17.20 | *** part/#asterisk syn (i=syn@kenobi.sceen.net) |
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| 15:22.45 | *** join/#asterisk bkw__ (n=brian@24.249.5.154) |
| 15:23.00 | santibiotico | i have an asterisk server with a private ip address connected to a router, which has one public ip address...then i want a sip phone in another location to register with my asterisk and make calls...i've configured nat for udp port 5060, and i get the phone registered and can make calls; i can dial numbers and they ring. but when the call is stablished, neither of the peers can hear each other. |
| 15:23.26 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 15:23.28 | santibiotico | i've configured sip.conf with nat=yes option |
| 15:23.37 | santibiotico | any idea of what could be happening?? |
| 15:23.52 | *** join/#asterisk mindCrime__ (i=chatzill@nat/redhat/x-6966319f058e6996) |
| 15:29.41 | shido6 | RTP |
| 15:29.45 | shido6 | RTP audio ports |
| 15:29.59 | *** join/#asterisk ManxPower (n=manxpowe@71-8-11-73.dhcp.leds.al.charter.com) |
| 15:30.00 | shido6 | there is more to sip then just port 5060 |
| 15:30.49 | shido6 | you arent letting the RTP audio through which can be between 10,000 and 20,000 linksys devices like 16284-16484 ish |
| 15:30.52 | shido6 | 16384 |
| 15:31.02 | shido6 | i need breakfast........ brb |
| 15:31.43 | santibiotico | i've mapped udp ports from 10000 to 20000 |
| 15:33.54 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 15:34.23 | ManxPower | shido6: Cisco defaults to 16384 - 32768. I don't recall if it is odd ports or even ports. |
| 15:34.24 | *** join/#asterisk shinux_ (n=shinux@196.220.27.23) |
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| 15:51.17 | ManxPower | ASUS Motherboard Turns House Phones into Skype™ Phones http://www.asus.com/news_show.aspx?id=5116 |
| 15:53.43 | *** join/#asterisk Juggie (n=Juggie@wiley-459-29776.roadrunner.nf.net) |
| 15:55.09 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:55.14 | devel | hey all, anybody with an audiocodes FXO media gateway can answer a few questions caller id related? |
| 15:55.36 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.149) |
| 15:56.20 | ManxPower | devel: your extensive search of the mailinglist archives was not helpful? |
| 15:56.56 | knathraak | got a question about voicemail...how to set the default unavailble message to a custom sound? |
| 15:57.04 | *** join/#asterisk marv[work] (n=timr@br0.asteriasgi.com) |
| 15:57.06 | devel | ManxPower, i'm not going to lie to you: i'm helping somebody else, so i haven't gone there yet. allow me to remedy that. |
| 15:57.42 | SomeOne1 | ~sex |
| 15:57.44 | jbot | updatedb; locate; talk; date; cd; strip; look; touch; finger; unzip; uptime; gawk; head; apt-get install condom; mount; fsck; gasp; more; yes; yes; yes; more; umount; apt-get remove --purge condom; make clean; sleep, or super extractor, http://sf.net/projects/sex/ |
| 15:59.38 | HarryR | ahah |
| 15:59.45 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.89) |
| 16:01.05 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 16:01.05 | *** mode/#asterisk [+o anthm] by ChanServ |
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| 16:08.01 | Jo211 | Please, anyone can help-me with 110p support? i'm geeting calls dropped and: chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 |
| 16:11.25 | ManxPower | Jo211: that is one of the hardest problems to fix. It is caused by latency on the PCI bus of the motherboard. |
| 16:11.56 | *** part/#asterisk zoa (n=d@pirus.securax.be) |
| 16:12.03 | *** join/#asterisk zoa (n=d@pirus.securax.be) |
| 16:12.11 | ManxPower | Things that can cause this are On Board Ethernet, Onboard RAID, running in graphics mode, and some IDE controllers |
| 16:12.38 | ManxPower | Jo211: it can SOMETIMES be cause by a problem with the T-1/E-1 |
| 16:13.11 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
| 16:13.59 | ManxPower | BTW, does anyone have recommendations for a cheap rack mount server with many PCI slots? |
| 16:14.22 | peterme2005 | guys do any of you know who to perform an attended transfer using asterisk i..e via the manager interface?! |
| 16:14.57 | Jo211 | maxpower: what you suggest to correct this problem? If I put a external ethernet (using internal now) can help? Or change the slot of the 110p board? |
| 16:15.35 | ManxPower | Jo211: disable onboard LAN and put in a PCI ethernet card. That is the easiest thing to try. |
| 16:15.44 | ManxPower | make sure you are not running in graphics or framebuffer mode. |
| 16:16.01 | Jo211 | ok, what's framebuffer mode? |
| 16:16.22 | ManxPower | Jo211: anything this isn't 80x24 characters on the console. |
| 16:16.35 | ManxPower | many distros enable it by default to make things look "pretty" |
| 16:17.03 | ManxPower | Jo211: put the output of "cat /proc/interrupts" on pastebin.ca |
| 16:17.14 | Jo211 | maybe this can be a tiP: this is a asteriskathome 2.4 |
| 16:17.41 | *** join/#asterisk lenne_dk (n=Miranda@83.72.129.7.ip.tele2adsl.dk) |
| 16:17.47 | ManxPower | Jo211: I cannot help you with Asterisk@Home. /join #freepbx for that. The HDLC error is NOT an Asterisk@Home specific thing. |
| 16:17.51 | Jo211 | anyway, if I change to a better cpu i probaly the problem goes away and it's not related to telco, wright? |
| 16:18.25 | ManxPower | Jo211: no. the problem is not CPU related. The problem is that SOMETHING is locking interrupts for so long that data from the TE110P is being lost. |
| 16:18.51 | ManxPower | As I said this is one of the hardest problems to solve. |
| 16:19.58 | Jo211 | cat /proc/interrupts -> http://pastebin.ca/267012 |
| 16:21.05 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 16:21.08 | lenne_dk | Good [$timezone.time.greeting()] group. Anyone familiar with sphinx2? |
| 16:21.20 | RoyK | <PROTECTED> |
| 16:21.23 | lenne_dk | voice recognition |
| 16:21.24 | RoyK | Jo211: that's BAD |
| 16:21.28 | RoyK | Jo211: unload usb drivers |
| 16:21.50 | RoyK | bad bo Jo211 |
| 16:21.57 | RoyK | s/bo/boy/ |
| 16:22.00 | tzafrir | I saw a link on voip-info to a site called www.1bizcom.com . Looked into it. Looks like it is made by a web2.0 template site, with just about zero content |
| 16:22.01 | RoyK | :) |
| 16:22.14 | KermitTheFragger | lenne_dk: never heard of it, but it looks interesting :) |
| 16:22.51 | tzafrir | They claim to be "open source" and provide support and such. There is even a silly install guide. But there is no actual this to download. |
| 16:23.51 | tzafrir | Why should you unload USB? It generates just aobut zero interrupts if you have no device connected |
| 16:24.26 | tzafrir | And if you have a device connected, you can't really unload the drivers :-p |
| 16:24.38 | *** join/#asterisk GerjanT (i=gerjan@frontgate.watchthe.net) |
| 16:25.49 | lenne_dk | There is an /agi-bin/eagi-sphinx-test installed, and i have installed sphinx2, but there must be some configuration missing, because nothing happens wheren I execute Executing EAGI("SIP/36949608-086d8000", "eagi-sphinx-test "yes*no"") |
| 16:26.21 | *** join/#asterisk hfb (n=hfb@pool-71-118-252-158.lsanca.dsl-w.verizon.net) |
| 16:26.52 | *** join/#asterisk jubei_ (n=Stormtro@147.27.47.88) |
| 16:27.11 | jubei_ | anybody know of a good way to make asterisk talk to a cisco router who only works with h.323 ? |
| 16:27.45 | *** join/#asterisk mike1211 (n=mike1211@bas1-malton22-1167904974.dsl.bell.ca) |
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| 16:33.06 | peterme2005 | hi guys if anyone know how to perform an attended transfer please message me |
| 16:33.46 | lenne_dk | I know. |
| 16:33.59 | *** join/#asterisk sigmounte (n=sigmount@81.56.234.199) |
| 16:35.05 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 16:35.46 | lenne_dk | peterme, still here? |
| 16:35.52 | nextime | jubei_ : you can try to use chan_h323 or chan_ooh323 or chan_oh323 or chan_woomera. Both shuld work with cisco, anyway, if you need to have a gatekeeper, you can use gnugk. Another option is to use yate as h323 to SIP signaling proxy, do from asterisk you can speak sip that work better than h323, and yate work better that * in my experience on h323 |
| 16:35.54 | markit | peterme2005: the xfer of features is not good for you? |
| 16:36.00 | markit | hi nextime :) |
| 16:36.14 | nextime | hi markit |
| 16:36.29 | markit | nextime: still "overbusy"? :) |
| 16:36.56 | nextime | markit : more that "over" |
| 16:37.02 | nextime | s/that/than |
| 16:37.12 | nextime | (i'm tired and nervous today) |
| 16:37.13 | markit | btw, is there are developers lissening, going back to an older svn version fixed the dtmf problem, so is a BUG |
| 16:37.19 | knathraak | anybody know how to set default unavailable message for voicemail to a custom sound? |
| 16:37.27 | *** join/#asterisk BSDTech (n=RNeese@ppp-71-128-112-135.dsl.irvnca.pacbell.net) |
| 16:37.38 | peterme2005 | markit_ the features.conf is ok but.... it added load to the adsl line meaning that calls are tromboned |
| 16:38.00 | EmleyMoor | "tromboned"? |
| 16:38.07 | peterme2005 | basically asterisk is doing much of the work for an attended transferred call |
| 16:38.27 | lenne_dk | Tromboned? |
| 16:38.31 | peterme2005 | so the adsl (phone line) line becomes swamped |
| 16:38.52 | peterme2005 | ok can i perform an attended transfer via the manager api |
| 16:40.48 | RoyK | it't not an api.... |
| 16:41.47 | lenne_dk | so if one calls you from outside, you want to call another outside caller, and then connect the two callers, and leave your asterisk out of the call? |
| 16:42.16 | *** join/#asterisk hohum (n=dcorbe@host-12-195-58-235.iad1.interceltelecoms.net) |
| 16:42.24 | peterme2005 | yes |
| 16:42.37 | peterme2005 | but i want this to be an attended transfer |
| 16:42.51 | peterme2005 | so i say hey ive got joe blogs on the phone do you want to speak to him |
| 16:43.00 | ManxPower | I thought the "Manager Interface" was for maximizing synergy thru cohesive partnerships with forward thinking vendors |
| 16:43.44 | peterme2005 | then i hang up and asterisk connects the two people |
| 16:43.47 | peterme2005 | sounds simple enough |
| 16:43.52 | peterme2005 | but asterisk cant seem to do it |
| 16:44.45 | lenne_dk | Asterisk can't connect the two people without being in the middle of the call, especially if the parties are on POTS. |
| 16:44.59 | ManxPower | peterme2005: we do it all the time. We press the transfer button on the Polycom phone, dial the other party, consult, then complete the trensfer. |
| 16:45.27 | peterme2005 | yea ok thats cool but i want to do this via an application i have been developing thus i need to use the manager interface |
| 16:45.31 | hi365 | its called atended transfer |
| 16:45.31 | lenne_dk | But aren't asterisk in the loop then? |
| 16:45.34 | ManxPower | peterme2005: your extensive search of the mailing list archives was not helpful? |
| 16:45.39 | peterme2005 | the manager interface doesnt seem to give me an attended xfer |
| 16:46.07 | *** join/#asterisk hmmhesays (n=Neg@24-117-135-28.cpe.cableone.net) |
| 16:46.09 | peterme2005 | ManxPower_ no otherwise i wouldnt be in here |
| 16:46.39 | lenne_dk | Excuse me, but aren't there two issues here? management interface, and transfer without asterisk in the loop? |
| 16:46.53 | *** part/#asterisk ming_zym (n=zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 16:47.00 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 16:47.30 | peterme2005 | ok first issue i want to fire off a command to the manager api and say performAttendedXfer between callerA callerB (ME) and callerC |
| 16:48.02 | ManxPower | lenne_dk: did he say asterisk had to be out of the loop? |
| 16:48.06 | peterme2005 | we have a patch that simulates DTMF codes to perform an attended transfer |
| 16:48.08 | peterme2005 | it works |
| 16:48.11 | peterme2005 | but..... |
| 16:48.19 | ManxPower | peterme2005: Originate can't do it. |
| 16:48.21 | Emrah_ | Is it normal to hear echo when using the speakerphone with a Snom 360 |
| 16:48.26 | peterme2005 | it cripples the adsl upstream die to the setting of a -t flag |
| 16:48.32 | lenne_dk | "it added load to the adsl line meaning that calls are tromboned" |
| 16:48.52 | peterme2005 | yep |
| 16:49.03 | ManxPower | Ah. He needs to get attended transfer via AMI working before he tries to optimize it. |
| 16:49.37 | lenne_dk | So you wants the calls connected somewhere outside your system, at the public exchange. |
| 16:49.57 | ManxPower | lenne_dk: no, via an internet telephone company |
| 16:50.12 | ManxPower | no local PSTN interface or he would not be caring about his adsl bandwidth |
| 16:50.26 | Emrah_ | Anyone uses Snom phones here? |
| 16:50.44 | SheriF_SpacE | anyone know any news about meetme -v " video in 1.4 " ? |
| 16:50.58 | ManxPower | Emrah_: not I. They seemed too expensive and buggy for our use. |
| 16:51.09 | Emrah_ | You're true :) |
| 16:51.28 | Emrah_ | Anyone else? |
| 16:51.58 | blitzrage | nada, Polycom here |
| 16:52.22 | ManxPower | Polycom here too. |
| 16:52.51 | peterme2005 | we have an inbound call on a zap channel which will be answered by 'reception_phone' (SIP Phone) and we want to transfer the call to another SIP phone, say 'sales_1' but would like to ask 'sales_1' if they want to accept the call first - An Attended/consultative transfer. However we want to direct this transfer from a piece of software using the manager API |
| 16:53.03 | peterme2005 | alas we only seem to be able to perform blind xfers |
| 16:53.21 | mattfletcher | I have a question regarding the Dial() application. I'm trying to dial two extensions at the same time (using the "&" syntax). What I want to do is for the command to be treated as busy (jumping 101 commands) if EITHER extension is busy, not just BOTH. Is this possible? |
| 16:53.22 | *** join/#asterisk spunz (n=spunz@h081217096236.dyn.cm.kabsi.at) |
| 16:53.23 | lenne_dk | first you must be sure the IP-phone company supports that xfer. Who should pay for the call anyway, if caller C is long distance? If both callers are IP, there might be a possibility, but if either are PSTN (and converted to IP at the IP-phone company) I don't think it is possible. |
| 16:53.23 | ManxPower | peterme2005: if all else fails ask on the malinglists. |
| 16:53.32 | ManxPower | mattfletcher: no. |
| 16:53.50 | mattfletcher | arse biscuits |
| 16:53.54 | peterme2005 | i think if i can manipulate the channels at the lowest level then i might be able to bridge the two together but this is not as easy as it sounds |
| 16:53.56 | ManxPower | lenne_dk: you are making this much more complicated than it is. |
| 16:54.03 | *** join/#asterisk Mw3 (i=mw3@89.147.75.209) |
| 16:54.04 | peterme2005 | am i? |
| 16:54.10 | mattfletcher | manxpower: Is there another way? |
| 16:54.12 | ManxPower | mattfletcher: you can use ChanIsAvail to accomplish something similar. |
| 16:54.18 | ManxPower | peterme2005: no lenne_dk is |
| 16:54.47 | Emrah_ | Thanks ManxPower and blitzrage. |
| 16:55.01 | mattfletcher | cool thanks, i will look it up |
| 16:55.21 | Emrah_ | Maybe you can answer just a last question. I have a polycomIP300 which sudanly... The 7, * and the mute key stoped working |
| 16:55.34 | lenne_dk | Sorry, but I cannot se how two POTS calls can be connected, without going down his adsl. |
| 16:55.39 | Emrah_ | Do you think that it is a hardware problem or a software matter? |
| 16:56.39 | ManxPower | Emrah_: what firmware release? |
| 16:56.46 | peterme2005 | well ok take this.... if i use the xfer button on the aastra phone it seems to work ok and the system (asterisk) handles it ok |
| 16:56.55 | ManxPower | lenne_dk: HE IS NOT USING POTS |
| 16:57.07 | Emrah_ | ManxPower: No Idea I disconnected the phone for the moment... Well I'll replug it :) |
| 16:57.21 | *** join/#asterisk ShipHead (i=ShipHead@gateway/tor/x-804f55176ae89746) |
| 16:57.37 | *** join/#asterisk inv_Arp (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 16:57.55 | peterme2005 | however if i perform the same operation via the manager api it cripples the adsl upstream which has the repercussion of effecting the number of concurrent calls that can take place |
| 16:58.03 | mattfletcher | manxpower: i see the idea behind this chanisavail, but i cannot get my head around what i should be checking and how i should react to those checks. does anyone have an example of chanisavail being used in this way? |
| 16:58.34 | ManxPower | mattfletcher: you checked extensions.conf.sample? |
| 16:58.55 | ManxPower | peterme2005: are the calls coming in / going out via IAX2 or SIP? |
| 16:59.13 | peterme2005 | Zap in Sip out |
| 16:59.18 | *** join/#asterisk topping_ (n=topping@207.47.6.182.static.nextweb.net) |
| 16:59.39 | peterme2005 | i.e. Joe Blogs calls (ZAP) reception of the company |
| 16:59.43 | peterme2005 | reception answers |
| 16:59.58 | peterme2005 | reception transfers to the manager by performing an Attended Transfer |
| 17:00.20 | peterme2005 | and asks the manager if he wants to take the call |
| 17:00.23 | peterme2005 | he accepts |
| 17:00.28 | peterme2005 | then as the receptionist hangs up |
| 17:00.29 | *** join/#asterisk alamantia (i=alamanti@nat/digium/x-e26874cf504832af) |
| 17:00.40 | *** join/#asterisk emann (n=emann@74.136.146.15) |
| 17:00.42 | peterme2005 | the phone connects to the manager |
| 17:00.43 | *** join/#asterisk syn (i=syn@kenobi.sceen.net) |
| 17:00.45 | syn | hello again |
| 17:01.07 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 17:01.09 | peterme2005 | this (yes) can be achieved via the Dial() and setting the -t (transfer ) flag |
| 17:01.36 | emann | i have setup an server, and i want the people that work with me to be able to use it from there home. I read somewhere that doing a VPN solution adds alot of overhead, |
| 17:01.39 | puzzled | hi |
| 17:01.42 | peterme2005 | however and again this causes asterisk to manage the calls thus putting strain on the line as asterisk is then managing the transfer |
| 17:01.44 | syn | when using Queue(), and one of the agents is already on the phone, and a new call arrives, the already onthephone agent is ringing too. Is there a way to avoid this ? |
| 17:01.58 | mattfletcher | where culd i find the extensions.conf.sample file online? i've stupidly overwritten mine i fear |
| 17:01.59 | peterme2005 | as asterisk does this is swallows up most of the ADSL bandwidth |
| 17:02.00 | syn | emann: depends |
| 17:02.02 | peterme2005 | which is BAD |
| 17:02.08 | syn | emann: we use a VPN here and it works perfectly |
| 17:02.17 | *** part/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 17:02.26 | peterme2005 | most solutions have suggested using meet me or parked calls..... |
| 17:02.30 | peterme2005 | which yes will work but..... |
| 17:02.34 | emann | what is a secure way to allow people to access the asterisk server? I am using iax2 |
| 17:02.37 | ManxPower | peterme2005: I can't see how it would matter at all since the call is PSTN/ZAP -> Internet/SIP no matter if you do an attended or non-attended transfer. |
| 17:02.49 | peterme2005 | im back to the same issue that these swallow up adsl upstream bandwidth |
| 17:03.08 | emann | syn: ipsec vpn or pptp or ssl? |
| 17:03.10 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 17:03.21 | *** part/#asterisk BSDTech (n=RNeese@ppp-71-128-112-135.dsl.irvnca.pacbell.net) |
| 17:03.32 | ManxPower | peterme2005: it still won't make any difference since one leg of the call is ZAP |
| 17:03.44 | peterme2005 | ok here is a question...... |
| 17:03.46 | syn | emann: pptp |
| 17:03.48 | peterme2005 | via the manager api |
| 17:03.52 | emann | ok |
| 17:03.52 | peterme2005 | well the CLI |
| 17:04.04 | *** join/#asterisk ToTo (n=ToTo@host108-163-dynamic.2-87-r.retail.telecomitalia.it) |
| 17:04.17 | peterme2005 | can i do "hold channel:SIP/212blah" |
| 17:04.22 | peterme2005 | or something similar? |
| 17:04.23 | nextime | i use a vpn too, both with ipsec and openvpn, and both work great |
| 17:04.42 | ManxPower | peterme2005: as I said before ask on the mailing list if you can't find an answer here |
| 17:04.43 | peterme2005 | i.e. how to the dect phones do it...these aastra 480i phones i have sat on my desk |
| 17:04.57 | peterme2005 | ive asked and posted my issue on the mailing list |
| 17:04.58 | emann | nextime: thanks. i can do that. |
| 17:05.12 | peterme2005 | there are lots of post on various sip forums asking the same thing as im asking |
| 17:05.16 | ManxPower | peterme2005: I don't know, but they cannot magically turn a PSTN call into a VoIP call |
| 17:05.17 | peterme2005 | and there are no answers |
| 17:05.58 | ManxPower | Now if your calls were coming into the system via a SIP ITSP, then the entire scenario changes |
| 17:06.05 | peterme2005 | ok but i can take the channel for a call (SIP/ZAP) and connect it to another channel i.e. a local extension |
| 17:06.14 | ManxPower | peterme2005: I did not say "forums". I said mailing lists |
| 17:06.30 | ManxPower | peterme2005: yes, but the call is still coming in on your POTS line. |
| 17:06.50 | peterme2005 | well ive posted my question on the asterisk mailing list |
| 17:06.50 | peterme2005 | correct |
| 17:06.51 | peterme2005 | so ZAP in this case |
| 17:08.18 | peterme2005 | one of the guys here wrote a patch for asterisk which does work and simulated the DTMF tones for a a transfer #2 this gives us via the manager api an AttendedXfer |
| 17:08.19 | peterme2005 | cool |
| 17:08.21 | peterme2005 | but..... |
| 17:08.41 | peterme2005 | again this is now not an option as using the -t flag causes some serious bandwidth issues |
| 17:08.53 | peterme2005 | which affect the service |
| 17:09.06 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 17:09.38 | peterme2005 | so if say 10 concurrent calls can be handled then if we use the -t flag with the patch then the adsl line becomes swamped as asterisk is handling the management of the transfer between the phones |
| 17:09.57 | peterme2005 | which is a bummer as this causes a big bottleneck |
| 17:10.12 | ManxPower | peterme2005: That is incorrect. |
| 17:10.29 | peterme2005 | ? |
| 17:10.36 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 17:10.42 | ManxPower | There is no difference between 10 POTS -> Internet calls with or without "t" |
| 17:11.06 | ManxPower | Asterisk still has to convert the POTS to SIP no matter what. |
| 17:11.07 | peterme2005 | ok then why are we having an issue with bandwidth when we use the -t flag? |
| 17:11.14 | *** join/#asterisk knathraak (n=zach@151.196.142.242) |
| 17:11.20 | peterme2005 | all the calls are talking place over SIP |
| 17:11.22 | peterme2005 | forget POTS |
| 17:11.35 | peterme2005 | well forget it for now anyway |
| 17:11.48 | ManxPower | peterme2005: Well there is 30 mins of my life I won't ever get back. I asked you how the calls got to asterisk and you said ZAP |
| 17:12.05 | peterme2005 | no the call comes on on ZAP |
| 17:12.06 | knathraak | hi, i got disconnected...don't know if anyone saw my question earlier--does anyone know how to set a custom default unavailable message for voicemail? |
| 17:12.08 | peterme2005 | read my scenario above |
| 17:12.10 | ManxPower | Let me ask you again. How are the calls getting to Asterisk? |
| 17:12.58 | ManxPower | peterme2005: There is no -t flag, btw. It is "t" |
| 17:13.16 | peterme2005 | ok from the top.... a call can come in via ZAP so external or via SIP ...so any way really is doesnt matter |
| 17:14.12 | ManxPower | peterme2005: it sure does. If it comes into zap then audio will go via the ADSL no matter what you do. If it comes in via SIP/IAX2 and the call goes out the same protocol then Asterisk (and your ADSL) can get out of the media stream. |
| 17:14.13 | peterme2005 | i know i simply used -t as a reflection of the fact is it a flag variable |
| 17:14.33 | ManxPower | peterme2005: -t will confuse people. |
| 17:14.42 | peterme2005 | sorry t/T flags then |
| 17:14.44 | peterme2005 | :-) |
| 17:14.55 | peterme2005 | im a java programmer what can i say |
| 17:15.02 | ManxPower | peterme2005: now the people that read the various logs of the channel should not be confused. |
| 17:15.04 | peterme2005 | java -classpath -something -somethingelse |
| 17:15.17 | peterme2005 | fair play |
| 17:15.26 | peterme2005 | i shall keep that in mind then |
| 17:15.44 | peterme2005 | but quite simply if we are say talking about 10 extensions..... |
| 17:16.09 | peterme2005 | if any of those 10 extensions press the xfer button then then can transfer and there is no major load on the system |
| 17:16.18 | peterme2005 | im assuming that the phones are sending out SIP HEADERS |
| 17:16.21 | peterme2005 | however..... |
| 17:16.28 | ManxPower | peterme2005: for this discussion it does matter at all what technology the extensions use, BTW. |
| 17:16.45 | peterme2005 | i want to perform an xfer (all be it an attended transfer) via my application using the manager api |
| 17:16.59 | ManxPower | peterme2005: my advice is to wait for a response on the mailing lists. I assume you read manager.txt? |
| 17:17.21 | peterme2005 | no to do this my colleague wrote a patch that uses two main things really Dial() and the t flag to transfer |
| 17:17.44 | ManxPower | peterme2005: I'll bet there is an easy way to do what you want, but I'll bet it will be via some other method than Transfer |
| 17:17.54 | ManxPower | Like Originate or Redirect or something else. |
| 17:18.01 | peterme2005 | but this causes a bottle neck on the adsl line so the networking engineer is telling me |
| 17:18.14 | peterme2005 | then therefore how do the SIP phones we have do the xfer |
| 17:18.22 | peterme2005 | and can i simply simulate this |
| 17:18.37 | ManxPower | peterme2005: I'll bet they do it with an INVITE or REINVITE |
| 17:18.37 | peterme2005 | but in the form of an attended transfer |
| 17:18.49 | peterme2005 | they do and i think a REFER command? |
| 17:18.55 | ManxPower | Maybe |
| 17:18.57 | peterme2005 | so....question is ..... |
| 17:19.08 | peterme2005 | can i generate SIP PACKETS and inject them into asterisk??? |
| 17:19.18 | peterme2005 | if so it would be a major security flaw |
| 17:19.22 | ManxPower | But since the manager is a protocol agnostic thing I strongly doubt you can do protocol specific stuff in the manager API |
| 17:19.32 | peterme2005 | i agree |
| 17:19.56 | ManxPower | peterme2005: well you can generate SIP PACKETS and inject them into astersk all you want, as long as your packets have the correct auth info. |
| 17:20.17 | peterme2005 | (sorry if i appear agitated but its an issue thats been bugging me the last week...so if im snappy its not you :-)) |
| 17:20.42 | peterme2005 | i suggested this to one of the network guys here....... |
| 17:21.11 | peterme2005 | how about i get the phone to perform the xfer operation and releave asterisk or us of using the t flag which is causing this bottlenexk |
| 17:22.00 | peterme2005 | so.... a behind the scenes magic finger that when the user presses my transfer button to perform the attended transfer operation is sends a signal to the phone to 'press' the xfer button |
| 17:22.03 | ManxPower | peterme2005: I'm not aware of any SIP phones that can be controlled that way. |
| 17:22.03 | peterme2005 | that was one idea |
| 17:22.11 | peterme2005 | excatly |
| 17:23.45 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 17:23.50 | peterme2005 | i mean i could use a sipsoftphone.jar library (assuming one exists) for my application so that it performs an attended transfer the same as the hard phone on my desk..... only we dont want to engineer a softphone this is simply a call manager |
| 17:24.39 | peterme2005 | the other option was to place a call into a holding bay and then as we transfer we hook up the caller who has been put in the holding bay to the person we transferred to |
| 17:25.14 | peterme2005 | however the major flaw in this solution is that on the last part of the scenario where the 3rd person is waiting to be connected to the caller.....instead the call comes through to him on another channel |
| 17:25.27 | ManxPower | peterme2005: have you looked at Flash Operator Panel? |
| 17:25.30 | peterme2005 | so he effectively has to hang up too with me for the call to go through |
| 17:25.32 | peterme2005 | YES |
| 17:25.43 | ManxPower | Maybe that application can do it. IF you you might be able to figure out how. |
| 17:25.54 | peterme2005 | it doesnt do it this way and besides FOP only performs a blind transfer |
| 17:26.01 | peterme2005 | which is what we can currently do |
| 17:26.35 | peterme2005 | you would have thought this this concept would have been considered from the start!??? |
| 17:26.56 | peterme2005 | two of the other guys here are also stumped at the moment on this one |
| 17:27.18 | peterme2005 | i thin for now it might be a case of generating the packets ourselves |
| 17:28.12 | peterme2005 | but this has a number of issues 1. is a security hole 2. leads to the possibility of generating ill packets which can do all sorts of stuff i.e. connecting callers to random ppl??! |
| 17:28.52 | *** join/#asterisk avalone (i=avalone@83.239.191.57) |
| 17:30.49 | peterme2005 | so in summary i have posted to the mailing-list(s) both the asterisk one and various others.... i have posted to the forums and not got anything yet |
| 17:30.53 | peterme2005 | but its early days anyway |
| 17:31.14 | peterme2005 | however I have read posts from people that date back to 2005 asking the same thing |
| 17:31.20 | peterme2005 | and they havent got a response either |
| 17:31.31 | peterme2005 | i assume people have just given up with attended transfers |
| 17:31.40 | IOscanner | I seem to have a problem with app_addon after I upgraded to mysql 5. I have upgraded all mysql client, common, server, lib and dev. I rebuilt addons and I get an error. |
| 17:31.41 | *** part/#asterisk syn (i=syn@kenobi.sceen.net) |
| 17:31.57 | IOscanner | I get this error: WARNING[21596]: app_addon_sql_mysql.c:235 aMYSQL_connect: mysql_real_connect(......) failed |
| 17:32.04 | file | peterme2005: assuming things is dangerous |
| 17:32.09 | IOscanner | any ideas what I might be missing? |
| 17:32.38 | peterme2005 | well im assuming that no one knows the answer due to not getting a valid answer from anyone |
| 17:33.01 | peterme2005 | im also assuming that asterisk is inherently crap |
| 17:33.09 | zoa | you could do such a transfer |
| 17:33.14 | zoa | with a phone i think |
| 17:33.17 | peterme2005 | i mean have you ever attempted to look at the billing CDR stuff asterisk churns out |
| 17:33.30 | peterme2005 | yes you can do an attended transfer with a phone |
| 17:33.38 | shido6 | and a blind xfer |
| 17:33.39 | peterme2005 | i want to do it via my software application |
| 17:33.45 | peterme2005 | via the manager API |
| 17:33.46 | peterme2005 | however |
| 17:34.01 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:34.06 | peterme2005 | it doesn't support this feat |
| 17:34.11 | *** part/#asterisk dasenjo (n=dasenjo@208.195.215.226) |
| 17:34.16 | IOscanner | just prompt for a # from person you are trasfering from so your application will know to transfer |
| 17:34.30 | IOscanner | You will have to code an agi program to do this |
| 17:34.35 | peterme2005 | we have |
| 17:34.46 | peterme2005 | and it gives us an AttendedXfer operation |
| 17:34.50 | peterme2005 | brilliant |
| 17:34.51 | zoa | <peterme2005> ok first issue i want to fire off a command to the manager api and say performAttendedXfer between callerA callerB (ME) and callerC |
| 17:34.59 | zoa | i think you could actually do that through the manager |
| 17:35.00 | peterme2005 | however this adds load to the system |
| 17:35.10 | peterme2005 | meaning it sucks up the adsl upstream |
| 17:35.23 | peterme2005 | which effects the placement of any other concurrent calls |
| 17:35.24 | IOscanner | on a DSL line up |
| 17:35.29 | IOscanner | change to g729 |
| 17:35.32 | IOscanner | or get a better link |
| 17:35.50 | ManxPower | his calls are coming in and going out his ADSL. He wants to try to let the audio go direct and not over his DSL line. |
| 17:35.51 | peterme2005 | we have a bonded adsl line we supply clients with |
| 17:35.54 | IOscanner | implment QoS on your border |
| 17:36.07 | peterme2005 | i dont want to say oh if you want to do transfers you need 10 extra adsl lines |
| 17:36.16 | peterme2005 | QoS in enabled |
| 17:36.19 | peterme2005 | its what we offer |
| 17:36.26 | ManxPower | IOscanner: So you are recommending he re-engineer his entire WAN just because Asterisk can't do an attended transfer via the manager interface. |
| 17:36.26 | monsted | QoS is overrated |
| 17:36.30 | peterme2005 | over CISCO hardware |
| 17:36.34 | peterme2005 | i agree |
| 17:36.42 | monsted | just get enough bandwidth instead :) |
| 17:36.50 | peterme2005 | "So you are recommending he re-engineer his entire WAN just because Asterisk can't do an attended transfer via the manager interface." |
| 17:36.52 | peterme2005 | no way |
| 17:37.09 | IOscanner | put the server in a colo and do the transfers and RTP off network |
| 17:37.58 | IOscanner | then he can use SER at his location to route the calls to the correct asterisk box that has the correct bandwidth needed to route the call. Then he is just routing the calls and dealing with SIP headers. |
| 17:37.58 | peterme2005 | well i think that the best solution for now to be generate sip headers |
| 17:38.08 | *** join/#asterisk Ebola (n=Ebola@host86-134-167-28.range86-134.btcentralplus.com) |
| 17:38.33 | IOscanner | Then you put the bandwidth cost on there end and not yours. |
| 17:38.55 | ManxPower | peterme2005: get used to this. People never understood half the things I wanted to do either. Eventually I stopped trying to do cool stuff and stick with the easy stuff. |
| 17:40.04 | peterme2005 | this is true but with pressure from the sales guys i cant |
| 17:40.07 | peterme2005 | :-( |
| 17:40.29 | mattfletcher | manxpower: i've had a play with chanisavail. It won't work though. Whether the first phone is busy or not, I get a return code of 0 (AST_DEVICE_UNKNOWN - "Unknown"; channel is valid, but unknown state) every time. |
| 17:41.04 | IOscanner | It is hard to keep the sales guys from making sales of feature we don't have. |
| 17:41.09 | peterme2005 | all the ways i can perform an attended transfer... using Meet Me...Parking the Call.....simulating dtmf tones all suck up the upstream bandwidth |
| 17:41.28 | ManxPower | mattfletcher: paste the CLI output of the failed chanisavail |
| 17:41.42 | peterme2005 | anyway guys im off been a long (12 hour) day and i need rest |
| 17:41.44 | *** join/#asterisk dacleric (n=dacleric@p54820CF8.dip0.t-ipconnect.de) |
| 17:41.47 | ManxPower | ONLY the 2 or 3 lines |
| 17:42.00 | peterme2005 | im pooped.....thanks anyway and you have all been help regardless |
| 17:42.02 | peterme2005 | :-) |
| 17:42.11 | IOscanner | good look |
| 17:42.14 | IOscanner | luck |
| 17:42.23 | peterme2005 | i may well be back on tomorrow |
| 17:42.29 | peterme2005 | thanks IOscanner |
| 17:42.38 | peterme2005 | bye everyone :-) |
| 17:42.39 | mattfletcher | http://pastebin.ca/267161 |
| 17:42.39 | IOscanner | np |
| 17:43.44 | *** join/#asterisk Tili (n=tili@202.133.67.234) |
| 17:43.46 | ManxPower | mattfletcher: show me the Verbose line from the dialplan |
| 17:44.12 | mattfletcher | exten => s,3,Verbose(LYNN AVAILABLE STATUS: ${AVAILSTATUS}) |
| 17:44.59 | *** join/#asterisk puzzled_ (n=patrick@puzzled.xs4all.nl) |
| 17:45.10 | ManxPower | mattfletcher: why not ChanIsAvail(SIP/201&SIP/202,s) and check the status of ${AVAILCHAN} |
| 17:46.08 | ManxPower | or ChanIsAvail(SIP/201&Zap/1,s) as the case may be. Also stop using tro dial options |
| 17:46.16 | ManxPower | they just screw things up for testing |
| 17:46.38 | mattfletcher | manxpower: that's not what i'm trying to achieve. i want to know whether sip/201 is active. if it is, then i want to ring neither of those two phones and continue on with other things |
| 17:46.54 | IOscanner | anyone using addons and realtime with mysql 5? I upgraded to mysql 5 and rebuild addons and now addons doesn't want to work. Any ideas is there any reported bugs with 1.2? |
| 17:47.56 | ManxPower | Ah. Try ChanIsAvail(SIP/201,s) and check the status of ${AVAILCHAN} |
| 17:50.25 | CunningPike | Has anyone had trouble getting a Polycom IP4000 to do CDP? |
| 17:50.58 | ManxPower | CunningPike: I've never managed to get any polycom to do CDP |
| 17:51.13 | monsted | as in cisco discovery protocol? |
| 17:51.26 | CunningPike | ManxPower: All our 501/601s do no problem - our 4000 is being a PITA |
| 17:51.26 | ManxPower | You know that CDP stands for, right? Crappy Damn Protocol |
| 17:51.42 | CunningPike | monsted: As in that |
| 17:51.52 | ManxPower | CunningPike: how do you set the CDP up in your switch? |
| 17:52.13 | CunningPike | ManxPower: I have absolutely no idea - as I say, it works for 100+ 501s |
| 17:52.19 | monsted | ManxPower: "no cdp run" in global mode and "no cdp enable" on the switch ports :) |
| 17:52.26 | CunningPike | ManxPower: I'm not the network dude |
| 17:53.09 | ManxPower | monsted: that's all? Even with multiple VLANs on a port? |
| 17:53.18 | ManxPower | It magically figures out which one to use? |
| 17:55.12 | monsted | cdp doesn't use vlans, i believe |
| 17:55.14 | *** join/#asterisk Modcuts (n=Moducts@88-109-72-123.dynamic.dsl.as9105.com) |
| 17:55.26 | mattfletcher | manxpower: I got for 3 tests: SIP/201-e351 , SIP/201-805a , SIP/201-21d0. The first one SIP/201 was free for, the other two it wasn't. SIP/201 btw is a AAstra 480i with multiple lines support i ought to mention. I've used call-limit=1 to restrict it to using one line though. Is this relevant? |
| 17:55.48 | monsted | cdp is also a completely useless protocol :) |
| 17:55.54 | *** join/#asterisk delta1 (i=delta@217.113.15.254) |
| 17:56.26 | CunningPike | monsted, ManxPower: We use CDP to have the phones discover the voice VLAN on our network - works great, except for this IP4000 |
| 17:56.35 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
| 17:57.07 | monsted | we just put them in a vlan and ignore that possibility |
| 17:57.11 | ManxPower | mattfletcher: Ah. the -e351 is the call instance, not a channel |
| 17:58.13 | mattfletcher | manxpower: if i understand correctly though this suggests that for all three tests asterisk saw SIP/201 to be available to call though, no? |
| 17:59.01 | ManxPower | try it without call-limit |
| 18:03.21 | mattfletcher | manxpower: ur a star |
| 18:03.31 | mattfletcher | works a treat |
| 18:03.41 | *** join/#asterisk RoyK (n=roy@ti211310a080-14619.bb.online.no) |
| 18:04.12 | *** join/#asterisk asberger (n=nb@cpe-76-170-93-79.socal.res.rr.com) |
| 18:04.22 | mattfletcher | for reference i'm using the j argument to jump 101 lines if the check is false. thank you so much. i can now go home before my tea gets any colder |
| 18:04.32 | *** join/#asterisk underzsod (n=g3443@ppp176-244.adsl.forthnet.gr) |
| 18:04.33 | underzsod | THE BEST WAREZ SITE IN THE PLANET! UPLOADING BATTLE BEGAN 2DAY,3 WINNERS TAKE ONE MONTH RAPIDSHARE PREMIUM! ONLY AT--> WWW.UNDERZSOFT.COM |
| 18:04.35 | *** part/#asterisk underzsod (n=g3443@ppp176-244.adsl.forthnet.gr) |
| 18:05.30 | Corydon-w | Wish the feds would quit it with the entrapment like that. |
| 18:07.38 | ManxPower | mattfletcher: feel free to send a paypal to eric@fnords.org |
| 18:08.19 | ManxPower | Corydon-w: you'd think they would have more important things to do like payoffs and assassinations. |
| 18:08.37 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 18:09.22 | Corydon-w | ManxPower: the conspiracy theorists are going mad again, thanks to a CIA operative near Bobby Kennedy on the day he was shot |
| 18:09.42 | *** join/#asterisk RoyK (n=roy@ti211310a080-14619.bb.online.no) |
| 18:10.35 | Corydon-w | You'd think cia employees couldn't support political candidates |
| 18:10.49 | RoyK | <PROTECTED> |
| 18:12.13 | *** join/#asterisk X-Gen (n=X-Gen@dsl-242-17-185.telkomadsl.co.za) |
| 18:12.41 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 18:13.10 | *** join/#asterisk DTE (n=pier@213-156-52-125.fastres.net) |
| 18:13.17 | DTE | hi all |
| 18:13.21 | DTE | i have a question |
| 18:13.38 | DTE | i'd like to use an asterisk server |
| 18:13.45 | DTE | to connect the internal phones |
| 18:14.08 | DTE | and to go ou with a voispeed server |
| 18:14.13 | DTE | is that possible? |
| 18:15.51 | ManxPower | What protocol does a voispeed server use? |
| 18:16.39 | DTE | eh...a second..i gicve a look |
| 18:17.21 | ManxPower | If it supports SIP then it SHOULD work with Asterisk. |
| 18:17.55 | SomeOne1 | ~love |
| 18:18.00 | jbot | If you love <insert item> so much, why don't you marry it |
| 18:18.00 | DTE | it supports sip and it's protocol |
| 18:18.00 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 18:18.09 | SomeOne1 | ~time |
| 18:18.11 | jbot | 2006.12.04 18:18:11 GMT |
| 18:18.11 | *** join/#asterisk alerios_ (n=alerios@190.24.98.197) |
| 18:18.21 | SomeOne1 | GMT?? |
| 18:18.24 | SomeOne1 | who the heck follows GMT? |
| 18:18.29 | *** mode/#asterisk [+o mog] by ChanServ |
| 18:18.32 | SomeOne1 | RoyK: sup!! |
| 18:20.37 | DTE | and another question |
| 18:20.59 | DTE | the new asterisk does it have a web interface? |
| 18:21.32 | ManxPower | DTE: Join #asterisk-gui to learn about the new Asterisk GUI framework |
| 18:21.56 | DTE | ahh |
| 18:22.09 | lenne_dk | ~time |
| 18:22.11 | jbot | You are educated stupid and therefore too dumb to understand nature's perfect time cube! (2006.12.04 18:22:11 GMT) |
| 18:22.27 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 18:23.47 | lenne_dk | Does asterisk have any problems with IAX-phones? I have two in the mail from an ebay-auction. |
| 18:24.17 | benjk | LOL |
| 18:24.20 | diclophis-work | well.. IAX is the inter-asterisk-exchange protocol |
| 18:24.27 | diclophis-work | so.. my guess is no? |
| 18:24.49 | diclophis-work | unless IAX is a brandname, with a shotty SIP support |
| 18:24.53 | puzzled | lenne_dk: no but my experience is that the phones do not work very well and look like crap |
| 18:25.28 | devel | hey all, anybody with an audiocodes FXO media gateway can answer a few questions caller id related? i've looked over list archives and don't see anything seemingly relevant. |
| 18:26.14 | *** join/#asterisk xAD (n=xAD@host144-199.pool8290.interbusiness.it) |
| 18:28.49 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-e93a683f9b4f1538) |
| 18:30.51 | *** join/#asterisk rr-- (n=rr@cpe-66-69-217-206.austin.res.rr.com) |
| 18:32.08 | rr-- | can asterisk act as a sip server? |
| 18:32.45 | linlin | define sip server |
| 18:33.21 | bkruse | linlin: good point |
| 18:33.28 | bkruse | rr--: almost deffintly anything you need it for. |
| 18:33.34 | rr-- | the machine that one points one's ATAs to |
| 18:34.03 | bkruse | absolutly |
| 18:34.03 | bkruse | (sp). |
| 18:34.06 | *** join/#asterisk Sed[PCT]_ (n=Brandon@2001:4830:2403:0:0:0:0:1) |
| 18:34.08 | *** part/#asterisk alamantia (i=alamanti@nat/digium/x-e26874cf504832af) |
| 18:34.18 | *** join/#asterisk alamantia (i=alamanti@nat/digium/x-e26874cf504832af) |
| 18:34.23 | *** join/#asterisk DrkShdw (n=scorpio@unaffiliated/drkshdw) |
| 18:35.12 | linlin | yes |
| 18:35.19 | linlin | a sip proxy on the other hand, it is not |
| 18:36.04 | rr-- | oh, i thought sip server and proxy were same thing |
| 18:37.08 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 18:37.40 | linlin | a sip proxy allows a sip device inside a NAT or firewall to communicate with a sip server on the other side of that nat or firewall |
| 18:38.54 | *** join/#asterisk hoobastoob2 (n=ckwall@63.149.122.93) |
| 18:38.55 | ChkDigit | Has anyone been having problems with IAX2 going from a 1.2.1 server to a 1.4.0-beta3? |
| 18:39.27 | ChkDigit | 1.4.0-beta3 crashes (probably a segfault) after spewing a bunch of chan_iax2.c: No private structure for packet? |
| 18:39.57 | hoobastoob2 | i am looking to see how to disable some of the buttons on a Linsys SPA 942. Things like removed the ability to use DND and call forward. does anyone know how to do this? |
| 18:40.23 | hoobastoob2 | I am just using the html page. I dont know where to find any of the cfg files for it. |
| 18:40.58 | file | ChkDigit: use the 1.4 branch |
| 18:42.22 | hmmhesays | does asterisk have native postgresql support? |
| 18:42.32 | hmmhesays | for sip, dp and vm ? |
| 18:42.44 | nibbler_de | hmmhesays: just use chan_postgres |
| 18:42.56 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 18:43.22 | *** join/#asterisk Zork_ (n=Zork_@j214090.upc-j.chello.nl) |
| 18:44.37 | rr-- | at the moment, i have many ATAs behind a NAT pointing to Gizmo SIP server, acting as Gizmo clients only. i am thinking of getting a Sangoma 6-FXO/0-FXS card. Will I be able to have asterisk direct calls from PSTN to the ATAs (and vice versa) while still preserving the ability of ATAs to make/receive Gizmo calls? |
| 18:45.07 | *** join/#asterisk mfroes (n=froes@200-162-218-81.corp.ajato.com.br) |
| 18:45.15 | linlin | yes thats possible |
| 18:45.28 | mfroes | do anyone have a newer version of chan_oh323.so ??? |
| 18:45.29 | Zork_ | Hello, I have the following situations: PSTN line --> Linksys 3102 --> Asterisk --> Voip Phone. Now when I get called, I occasionally hear echo on my phone. The other party doesn't seem to notice it, but for me it sometimes is really bad. Any ideas as to where to look at? I tried playing with the gain values, but that didn't seems to help. |
| 18:45.34 | mfroes | to fedora core 5 ? |
| 18:45.40 | linlin | you would make the asterisk box act as a client to gizmo instead, and the ATAs talk to asterisk |
| 18:46.28 | hoobastoob2 | Zork_: have you verified that your zaptel device is not sharing irqs with anything else? |
| 18:47.05 | *** join/#asterisk Socket (n=CiraK@88.224.166.181) |
| 18:47.09 | *** part/#asterisk knathraak (n=zach@151.196.142.242) |
| 18:47.11 | Zork_ | hoobastoob2 > I don't think I'm using zaptel... It's PSTN --> Linksys 3102 (which connects through SIP to Asterisk) |
| 18:47.23 | *** join/#asterisk knathraak (n=zach@151.196.142.242) |
| 18:47.32 | Zork_ | Although I do have an ISDN card in the asterisk computer. |
| 18:47.47 | hoobastoob2 | Zork_: walk me through this... |
| 18:47.49 | *** join/#asterisk tim27 (i=tim27@97-70.dr.cgocable.ca) |
| 18:48.02 | hoobastoob2 | you have a pstn provider, right? |
| 18:48.15 | Zork_ | hoobastoob2: I have an PSTN (analog) line. yes. |
| 18:48.27 | Zork_ | That's connected to my LinkSys 3102 VoIP router. |
| 18:48.43 | Zork_ | Which is capable of converting the PSTN signal to SIP/RTP. |
| 18:48.56 | Zork_ | Using that, it connects to my asterisk server. |
| 18:49.12 | Zork_ | And I also have a SIP phone, which is also connected to that asterisk server. |
| 18:49.30 | knathraak | hi...I posted this question before, but didn't get any replies--anybody know how proceed down a list of queue members in *the same order* every time? So the member (a zap channel in this case) at the top gets the first crack each time? |
| 18:49.32 | hoobastoob2 | with the linsys i have no idea. |
| 18:49.36 | *** join/#asterisk jgoo (n=64a25640@foodtecsolutions.com) |
| 18:49.44 | Zork_ | k, thanks for trying. |
| 18:49.53 | Zork_ | Anyone else got a clue? |
| 18:50.01 | hoobastoob2 | knathraak: read the samples in the queues.conf |
| 18:50.13 | knathraak | i looked at the samples. |
| 18:50.15 | jgoo | hrm, guys - I am using asterisk-java and I am getting a hard limit 60 second timeout... I googled and checked the docs, is this in AGI or is this an extensions default? (call must pass out of extensions control in 60 seconds) ?? |
| 18:50.24 | Zork_ | The thing I find strange is that a call can sound okey, then get some echo, and then okay again. |
| 18:50.39 | knathraak | hoobastoob2: i tried using penalties, in combination with roundrobin |
| 18:51.04 | hoobastoob2 | knathraak: all you want is for the same person to get the call first and the same people to follow in succession after that? |
| 18:51.06 | knathraak | hoobastoob2: and I tried roundrobin with no penalties |
| 18:51.16 | knathraak | hoobastoob2: yes |
| 18:51.22 | linlin | Zork_ you have good phone lines? |
| 18:51.31 | hoobastoob2 | roundrobin with no penalties is exactly what you want |
| 18:51.38 | knathraak | hoobastoob2: roundrobin seems to prefer to hit the least recently used member |
| 18:51.40 | hoobastoob2 | make sure you add the members in the order you want them rung |
| 18:51.45 | hmmhesays | so whats the best way for me to get realtime postgresql? |
| 18:51.46 | Zork_ | Well, if I connect a normal analog phone to the PSTN line, there's to problem. |
| 18:52.03 | Zork_ | to=no |
| 18:52.03 | hoobastoob2 | that is roundrobin with memory... whatever the name for that one is |
| 18:52.13 | *** join/#asterisk s1gny|wrk (n=s1gny@p54916A3E.dip.t-dialin.net) |
| 18:52.21 | knathraak | hoobastoob2:rrmemory, I think |
| 18:52.40 | Zork_ | So I guess the phone line is okay. |
| 18:52.44 | *** part/#asterisk s1gny|wrk (n=s1gny@p54916A3E.dip.t-dialin.net) |
| 18:52.54 | hoobastoob2 | yeah, thats the one |
| 18:53.30 | knathraak | hoobastoob2: okay, i'll give it another try. maybe i'm doing something wrong. |
| 18:53.54 | hoobastoob2 | good luck |
| 18:54.50 | *** join/#asterisk ambriento (n=melcon@200.192.160.100) |
| 18:55.26 | tim27 | i have a sip trunk with babytel.ca... and also 3 DID on this account... i want to seperate did to ring proper extension. Babytel told me that they send the DID number in the TO: of the sip header... but my freepbx (asterisk) seem to not detect the did correctly... (i verified the sip header, on the cli with sip debug... and TO: show my correct DID) but it seem that asterisk detect DID based on the account and not the DID |
| 18:55.34 | knathraak | hoobastoob2: just tried it. when I took the call with the first member, then hung up, then called back, it immediately went to the second member. |
| 18:55.39 | *** join/#asterisk tim0123 (n=cash247@adsl-71-158-168-130.dsl.rcsntx.sbcglobal.net) |
| 18:55.53 | tim0123 | hello everybody |
| 18:56.07 | jgoo | anyone else know about a 60 second limit for scripts? |
| 18:56.09 | knathraak | hoobastoob2: under the specific queue definition, i have 'strategy = roundrobin' |
| 18:56.56 | tim0123 | where do you define barge for monitoring calls |
| 18:56.59 | *** join/#asterisk zoqute (n=pilopos@85.137.126.95) |
| 18:57.59 | tim0123 | Does it go in the same context as your phones do? |
| 18:58.33 | bkruse | tim0123: same context as the phone your barging FROM |
| 18:59.16 | *** join/#asterisk elriah (i=elriah@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 18:59.33 | elriah | Hi all. How do I tell if a bridged call is using g729 in asterisk 1.2? |
| 19:00.46 | *** join/#asterisk ToTo (n=ToTo@host108-163-dynamic.2-87-r.retail.telecomitalia.it) |
| 19:01.11 | *** join/#asterisk santiago (n=santiago@208.195.215.22) |
| 19:02.45 | elriah | Hi all. How do I tell if a bridged call is using g729 in asterisk 1.2? |
| 19:05.11 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 19:06.22 | benjk | anthm, here's an idea for your anti-asterisk creativity .... get a copy of a "Lost" episode and Final Cut Pro on your Mac, then edit the scene where they have to enter the weirdo numbers into the computer, replace that with "asterisk sucks" |
| 19:06.24 | elriah | Maybe the question is How do I monitor codecs in general in asterisk? |
| 19:06.47 | tim0123 | anybody Spawn extension (barge, 1201, 0) exited non-zero on 'SIP/201-bbe1' |
| 19:06.47 | tim0123 | <PROTECTED> |
| 19:08.22 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.226) |
| 19:10.00 | *** join/#asterisk RoyK (n=roy@ti211310a080-14619.bb.online.no) |
| 19:10.01 | CunningPike | knathraak: You need each agent on it's own penalty - the first agent has penalty 0, the next penalty 1 and so on. Also, when testing, don't forget to leave the 'clean-up' time between subsequent calls |
| 19:10.54 | *** join/#asterisk SwK[Work] (n=SwK@br0.asteriasgi.com) |
| 19:11.50 | *** part/#asterisk markit (n=konversa@host119-245-static.72-81-b.business.telecomitalia.it) |
| 19:11.59 | Qwell[] | SwK[Work]: y0 |
| 19:12.56 | knathraak | CunningPike: unfortunately, by using penalties, it hangs on the member with the lowest penalty, and calls that one over and over. I've seen some mention of this problem on mailing lists as well as the voip-info asterisk wiki, but i've not yet seen a work-around |
| 19:14.03 | CunningPike | knathraak: You need to make sure that your members can only accept one call at a time - either in the phone (so it returns busy) or in sip.conf using call-lmit |
| 19:14.53 | rob0 | Be more cunning? |
| 19:15.05 | mfroes | how can i compile asterisk with oh323 support ??? |
| 19:15.06 | CunningPike | I'm as cunning as a pike can be |
| 19:15.11 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:15.24 | rob0 | Then be less pikeful? |
| 19:15.49 | *** join/#asterisk Assid (n=assid@59.183.23.248) |
| 19:16.26 | nextime | mfroes : asterisk-oh323 seem to be outdated, try the asterisk-ooh323c included in asterisk-addons, in my opinion is the better chan h323 for * |
| 19:17.13 | mfroes | nextime: thanks. .. can i compile only ooh323??? |
| 19:17.19 | *** join/#asterisk Modcuts- (n=Moducts@88-109-72-123.dynamic.dsl.as9105.com) |
| 19:17.31 | CunningPike | I'm pike to the cartilage...... |
| 19:17.34 | CunningPike | Can't change |
| 19:18.48 | nextime | mfroes : you can use ooh323 or h323, or even woomera or oh323, anyway, i don't know if woomera is working good, i know that oh323 is outdated, h323 don't have good performance and have some issues with many devices, ooh323 is the better one |
| 19:18.51 | *** join/#asterisk knathraa1 (n=zach@151.196.142.242) |
| 19:19.12 | nextime | you can compile only ooh323 from asterisk-addons, yes |
| 19:19.28 | nextime | and it don't require openh323lib and pwlib |
| 19:19.30 | knathraa1 | CunningPike: got disconnected. did you get my last comment.. about zap channels & cell phones? |
| 19:19.53 | *** part/#asterisk knathraa1 (n=zach@151.196.142.242) |
| 19:20.13 | *** join/#asterisk knathraak (n=zach@151.196.142.242) |
| 19:20.18 | CunningPike | There he is |
| 19:20.19 | knathraak | test |
| 19:20.35 | awannabe | is there any reason after a call is on hold for so long that it disconnects? |
| 19:20.35 | knathraak | keep getting disconnected |
| 19:20.55 | CunningPike | knathraak: No - the last thing I saw from you was about penalties not working - did you get my reply? |
| 19:21.01 | knathraak | sorry.. |
| 19:21.20 | knathraak | CunningPike: you said to cconfigure the phones to accept only one call at a time |
| 19:21.28 | CunningPike | knathraak: Correct |
| 19:21.50 | knathraak | CunningPike: I don't think that's the problem. I'm calling into the queue, and I'm the only caller for testing. |
| 19:22.15 | knathraak | CunningPike, so there's not more than one call trying to go to a single queue member |
| 19:22.44 | CunningPike | knathraak: And you're leaving your tidy up time between calls? |
| 19:22.54 | *** join/#asterisk ManxPowe1 (n=manxpowe@93.sub-75-202-248.myvzw.com) |
| 19:23.02 | knathraak | CunningPike: how should that work? |
| 19:23.27 | knathraak | CunningPike: i mean there's about 30 sec or so in between calls |
| 19:23.43 | knathraak | CunningPike: here's what happens... |
| 19:24.31 | knathraak | CunningPike: I call in as a test customer, and get put in the queue, then the queue rings cellphone #1 over a zap channel. |
| 19:24.40 | CunningPike | knathraak: A queue member will appear to be unavailable for wrapuptime seconds after the end of the call |
| 19:24.45 | CunningPike | knathraak: OK |
| 19:25.10 | knathraak | CunningPike: I answer cellphone#1 , then hang up both calls |
| 19:25.37 | CunningPike | knathraak: And does that Zap channel appear busy the next time you call in? |
| 19:25.53 | CunningPike | knathraak: In other words, does the queue try that zap channel next time? |
| 19:25.59 | knathraak | CunningPike: when I call back, if i'm using penalties (eg. 1 for one member 2 for another) it goes to the same channel each time |
| 19:26.13 | knathraak | CunningPike: the one with the lowest penalty |
| 19:26.18 | CunningPike | OK - and that's not what you want? |
| 19:26.26 | knathraak | CunningPike: yes, but... |
| 19:26.59 | knathraak | CunningPike: If cell phone #1 times out, it should then hang up, and attempt to call cellphone #2 which as a higher penalty. |
| 19:27.05 | *** part/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 19:27.07 | knathraak | CunningPike: but that's not what happens |
| 19:27.44 | knathraak | CunningPike: instead it tries again on cell phone #1, times out, hangs up, and tries again on cell phone #1, because it has the lowest penalty, and wont try anything with a higher penalty. |
| 19:27.58 | CunningPike | knathraak: Ah - OK. If an agent logs in, it is assumed that they are available to take calls. You need to manage your agents so that they answer the calls presented to them |
| 19:28.36 | *** part/#asterisk reber (n=reber@cl-157.dub-01.ie.sixxs.net) |
| 19:28.39 | CunningPike | knathraak: We went through this with our folks on an identical queue set up - if you're logged in, you must be available to take calls |
| 19:28.58 | CunningPike | knathraak: If you can't ensure that, then you need a different queue strategy |
| 19:29.21 | *** join/#asterisk DrAk0SX (n=luisjose@unaffiliated/luisjose) |
| 19:29.26 | DrAk0SX | hey |
| 19:29.27 | DrAk0SX | can I paste 3 lines? |
| 19:29.56 | CunningPike | DrAk0SX: In the CLI? ;) |
| 19:29.59 | rob0 | You forfeit one limb per line pasted. ;) |
| 19:30.03 | knathraak | CunningPike: okay for us the point of the whole system, is so that if one technician's cell phone is off network, or off etc. it will try the next phone in the list. But it should *always* try the first cell phone in the list every time. |
| 19:30.23 | rob0 | I should know ... just call me "Lefty". |
| 19:30.28 | CunningPike | knathraak: Hmmm |
| 19:31.08 | knathraak | CunningPike: so in theory the techs with cell are available 24x7 |
| 19:31.19 | *** join/#asterisk adorah (n=admin@87.68.146.112) |
| 19:31.51 | adorah | Hi anyone with experience in astlinux/soekris net4801? |
| 19:32.03 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool4-183.mtco.com) |
| 19:32.09 | CunningPike | knathraak: I think that, the way app_queue works, your Zap channel will need to return busy for your present queue strategy to work |
| 19:32.32 | knathraak | CunningPike: what is app_queue? |
| 19:32.39 | docelmo | sigh |
| 19:32.45 | docelmo | you are kidding right? |
| 19:32.52 | CunningPike | knathraak: The code that provides the Queue() application |
| 19:32.55 | knathraak | CunningPike: oh the part of asterisk that does cueing? |
| 19:33.02 | knathraak | CunningPike: ah oky |
| 19:33.53 | docelmo | hmmm |
| 19:35.33 | knathraak | CunningPike: okay so seem like there should be a way to make Queue() forget who it called last, so that it starts at the top of the list. is there a variable that can be fiddled or anything? other than that, how do i force the zap channel to return busy if not answered (and not actually busy). |
| 19:36.50 | CunningPike | knathraak: You've tried roundrobin? I forget the difference between that and rrmemory (and it's being deprecated anyway) |
| 19:36.52 | *** join/#asterisk ToTo (n=ToTo@host108-163-dynamic.2-87-r.retail.telecomitalia.it) |
| 19:36.59 | *** join/#asterisk RoyK (n=roy@ti211310a080-14619.bb.online.no) |
| 19:38.17 | DrAk0SX | Dec 4 15:28:11 WARNING[61644]: translate.c:88 powerof: Powerof 0: No power?? |
| 19:38.18 | DrAk0SX | Dec 4 15:28:11 WARNING[61644]: translate.c:133 ast_translator_build_path: No translator path from gsm to unknown |
| 19:38.19 | knathraak | CunningPike: roundrobin remembers who answered last. |
| 19:38.40 | CunningPike | knathraak: Ah right |
| 19:39.27 | knathraak | CunningPike: a bit unintuitive. |
| 19:39.27 | adorah | Hi anyone with experience in astlinux/soekris net4801? |
| 19:39.49 | CunningPike | adorah: You might be better off with a mailing list posting for something that specific |
| 19:40.08 | knathraak | CunningPike: there's actually an informative section at the bottom of http://www.voip-info.org/wiki-Asterisk+config+queues.conf |
| 19:40.19 | rr-- | is it electrically OK to split an incoming PSTN line onto two FXO ports (say the FXO port of a PBX and the FXO port of a Sangoma/Digium card) |
| 19:40.35 | adorah | manual and support for Astlinux is sooo poor. |
| 19:40.36 | CunningPike | knathraak: I'm not sure what to suggest for your purposes |
| 19:40.44 | knathraak | CunningPike: unfortunately it mentions penalties as a solution. |
| 19:41.05 | knathraak | CunningPike: is there away to make the zap channel return busy if not answered? |
| 19:41.14 | docelmo | rr-- in theory yes.. should you do it? no |
| 19:41.19 | *** join/#asterisk evisu (i=hIRC@bzq-88-152-176-54.red.bezeqint.net) |
| 19:41.24 | CunningPike | knathraak: Not to my knowledge..... |
| 19:41.56 | docelmo | adorah what is your problem? I will call the guy who wrote it now |
| 19:42.39 | rr-- | docelmo: the idea would be to configure asterisk and the analog PBX as to who would answer ... would that not work? |
| 19:43.14 | adorah | <docelmo>I just don't get how to log into the net4801..installed a flash card but can't access it either with console or theu ethernet |
| 19:43.17 | docelmo | yes it would but eh.. its your nightmare |
| 19:43.55 | docelmo | if its turned on you should be able to get to the console if its not bad. |
| 19:45.07 | adorah | well I just got the machine today and it seems to work just that with the hyperterminal I get nothing on screen and the network connection doesn't work either with a regular or a crossed cable connection.. |
| 19:46.07 | adorah | <docelmo>I supposed to log in using the default ip with ssh or https but connection is refused |
| 19:46.21 | docelmo | http://karlsbakk.net/piespy/images/asterisk/asterisk-current.png |
| 19:47.04 | docelmo | I would say check the documentation.. Kristian has put alot of time and energy into this project. |
| 19:47.18 | *** join/#asterisk Deeewayne (n=dwayne@adsl-070-145-146-225.sip.mgm.bellsouth.net) |
| 19:47.33 | adorah | well there is very poor ducumentation and in the voip-info it simly not a correct one |
| 19:47.36 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 19:47.59 | linlin | can the cisco 7910 phone be patched to support sip and not only cisco proprietary format? |
| 19:48.14 | Qwell[] | no, I don't think so |
| 19:48.26 | *** join/#asterisk shidan (n=chatzill@CPE0013107d30c4-CM001371871af0.cpe.net.cable.rogers.com) |
| 19:48.28 | IOscanner | I don't think Cisco has release sip for 7910 or 7912 |
| 19:48.38 | docelmo | linlin no it cant.. Cisco doesnt have a SIP firmware for it yet |
| 19:48.49 | docelmo | QWELL! |
| 19:48.54 | IOscanner | You can use it you will have to use sccp |
| 19:48.55 | linlin | bummer, i just bought one |
| 19:48.56 | Qwell[] | I'm not here. |
| 19:49.06 | Qwell[] | linlin: it should work fine with chan_skinny in 1.4 |
| 19:49.14 | linlin | on, well any way for me to get it to talk to asterisk |
| 19:49.16 | rob0 | adorah: Sounds like your problem is missing prerequisites ... you have to have a pretty strong understanding of Unix/Linux here. |
| 19:49.21 | docelmo | Qwell[] hay can you throw something at Kevin and ask him to drop me a message on IRC |
| 19:49.30 | Qwell[] | docelmo: he's not here |
| 19:49.39 | docelmo | ohh really? Where's he at? |
| 19:49.40 | linlin | 1.4 is the new release of asterisk in a few weeks right? |
| 19:50.32 | awannabe | is there a setting that would make a caller on hold to be hung up after x number of minutes? |
| 19:50.37 | adorah | <rob0>well it is just getting into a console..the system supposed to enable access via ethernet.. |
| 19:50.54 | *** join/#asterisk heh_v_water (n=heh_v_wa@70-57-200-16.hlna.qwest.net) |
| 19:51.04 | IOscanner | What would cause addons to error: WARNING[21596]: app_addon_sql_mysql.c:235 aMYSQL_connect: mysql_real_connect(.....)failed? |
| 19:51.13 | heh_v_water | does moh not work with IAX extensions? |
| 19:51.21 | IOscanner | I thought I had everything upgraded for Mysql 5 |
| 19:51.26 | *** join/#asterisk xnon_ (n=xnon@200.8.85.221) |
| 19:51.57 | shidan | anyone has seen or played with products from sutus? |
| 19:52.30 | IOscanner | realtime show it working, but I am using MYSQL calls from extensions.conf for dynamic calls. It worked with mysql 4 but not mysql 5 |
| 19:53.43 | *** join/#asterisk Defraz (n=t0tal@fw.centrisys.com) |
| 19:54.06 | *** part/#asterisk axisys (n=axisys@c-69-143-190-152.hsd1.va.comcast.net) |
| 19:55.10 | adorah | <rob0>well it is just getting into a console..the system supposed to enable access via ethernet.. |
| 19:55.25 | adorah | Hi anyone with experience in astlinux/soekris net4801? |
| 19:55.47 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 19:56.09 | *** join/#asterisk yassine (n=yassine@xdsl-87-78-33-167.netcologne.de) |
| 19:56.16 | *** join/#asterisk RichiH (i=richih@freenode/staff/richih) |
| 19:56.24 | *** part/#asterisk RichiH (i=richih@freenode/staff/richih) |
| 20:01.22 | *** join/#asterisk olinux_ (i=olinux@ip68-107-4-202.sd.sd.cox.net) |
| 20:01.26 | *** join/#asterisk Zeeek (n=Zeeek@80.125.80.38) |
| 20:02.16 | *** part/#asterisk hoobastoob2 (n=ckwall@63.149.122.93) |
| 20:04.29 | Zeeek | clean |
| 20:04.55 | IOscanner | yep it is clean...lol |
| 20:05.37 | shidan | so no one has heard anything about sutus? is it vapourware? |
| 20:05.40 | Zeeek | what is? I just woke up and it's bed time |
| 20:07.59 | *** join/#asterisk reza_ (i=reza@abort.boom.net) |
| 20:09.05 | *** join/#asterisk voidans (i=adam@dieor.freelive.org) |
| 20:10.32 | Zeeek | quiet night, quiet starz |
| 20:10.47 | Zeeek | quiet chords on my guitarz |
| 20:11.20 | Zeeek | or not |
| 20:11.43 | *** join/#asterisk BlepsoaF (n=pbaker@nnat-gw.adeptra.com) |
| 20:12.04 | BlepsoaF | hell all, can someone take a peak at http://pastebin.ca/267340 - trying to get a try again prompt to happen, but the goto statement is failing |
| 20:13.08 | BlepsoaF | ^hello |
| 20:13.51 | Zeeek | and the meetme is working? |
| 20:14.10 | BlepsoaF | yes, but if the meetme conf doesnt exist I want them to attempt to re-enter instead of hanging up |
| 20:14.41 | Zeeek | polly need to test and see if it exists first |
| 20:16.54 | Zeeek | s/polly/prolly/ |
| 20:17.09 | Zeeek | yes we all knew that idiot bot |
| 20:17.47 | BlepsoaF | Zeeek: is there an asterisk function for that? |
| 20:18.29 | Zeeek | dunno. Have you read the meetme doc? |
| 20:18.54 | BlepsoaF | yes, but I guess my question is why auto fall through isnt working with meetme |
| 20:18.54 | Zeeek | I seem to remember that there is a way to see if a room exists |
| 20:19.36 | Zeeek | I haven't looked at meetme for ages |
| 20:20.30 | BlepsoaF | should I use something else? |
| 20:22.44 | Zeeek | no I pean I just looked at the show application info and there's a lot of new ones. |
| 20:23.48 | Zeeek | you want to ask for the number and then fallthru if it's empty? |
| 20:24.40 | BlepsoaF | no fall through if it exists |
| 20:24.42 | BlepsoaF | instead of haning up |
| 20:24.55 | BlepsoaF | hanging up |
| 20:25.03 | BlepsoaF | IE conf room = 123 |
| 20:25.08 | BlepsoaF | but someone enters 1234 |
| 20:25.18 | BlepsoaF | it would say that its incorrect, and re route to entering the conf # again |
| 20:25.46 | BlepsoaF | do I need to turn on priority jumping for that context? |
| 20:25.56 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 20:28.40 | Zeeek | I can't quite figure out what you're trying to do |
| 20:29.17 | Zeeek | gotta run |
| 20:29.27 | Zeeek | sorry I couldn't help |
| 20:29.30 | BlepsoaF | its ok |
| 20:29.36 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
| 20:30.38 | *** join/#asterisk ozoneco (n=stanp@CPE-24-27-138-124.neb.res.rr.com) |
| 20:34.35 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 20:34.36 | Assid | hrmm anyone know a cheap place besides sipphone for incoming did? |
| 20:34.44 | Assid | preferably 201(hackensack) |
| 20:35.02 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 20:35.08 | Assid | /j trxtel |
| 20:35.22 | hmmhesays | vitelity is cheap |
| 20:35.34 | Assid | vitelity ? |
| 20:35.36 | rob0 | 119 active SIP channels ... yikes ... and no calls? |
| 20:35.45 | hmmhesays | sounds familiar rob0 |
| 20:35.50 | rob0 | Any suggestions where to look? |
| 20:35.53 | hmmhesays | Assid vitelity.com I think |
| 20:36.18 | hmmhesays | rob0: that happened in my dp because i had commands after cmd dial that didn't match a goto |
| 20:36.27 | rob0 | aha, thanks. |
| 20:36.37 | hmmhesays | ie my goto went nowhere |
| 20:36.40 | grEvenX | any self-proclaimed GoSub experts here? :P |
| 20:36.52 | hmmhesays | i use gosub sometimes |
| 20:36.54 | grEvenX | is it OK to use GoSub inside a GoSub |
| 20:36.55 | grEvenX | ? |
| 20:37.19 | Assid | 7.95.. sipphone is 35$/yr |
| 20:37.22 | hmmhesays | I would imagine so, unless the return path is overwritten |
| 20:37.33 | Corydon-w | Yes, Gosub is safe to 100,000 levels |
| 20:37.40 | Corydon-w | at least |
| 20:37.46 | grEvenX | Corydon-w: nice, thanks :) |
| 20:37.58 | hmmhesays | where did you get that number from? |
| 20:38.00 | Corydon-w | codefreeze tested it to 100,000 levels |
| 20:39.16 | *** join/#asterisk jmacz (n=jmacz@190.24.98.197) |
| 20:39.55 | *** join/#asterisk docelmo (n=vircuser@c-68-32-143-73.hsd1.de.comcast.net) |
| 20:40.14 | grEvenX | I also received a question about wether one could somehow make the Gosub return after a timer has run out |
| 20:40.27 | Corydon-w | What timer? |
| 20:41.17 | Corydon-w | You could do a WaitExten(15), then a Return, and it will return if a new extension isn't entered within 15 seconds |
| 20:41.26 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 20:41.56 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 20:42.04 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 20:45.58 | *** join/#asterisk Scoundrel (n=toggy@host-81-191-169-198.bluecom.no) |
| 20:46.22 | grEvenX | Corydon-w: thanx, I'll need to discuss it further with the guy that asked me about the info... I din't really get what he meant with it either |
| 20:46.34 | Scoundrel | anyway to fix this without reinstalling the hole os part: Asterisk ended with exit status 1 |
| 20:46.34 | Scoundrel | Asterisk died with code 1. |
| 20:46.55 | *** join/#asterisk Dr-Linux|home (n=nono@202.59.73.131) |
| 20:47.45 | Manfish | Scoundrel sounds like your zaptel is stuffed up |
| 20:48.06 | Scoundrel | so reinstalling zaptel would fix it? |
| 20:48.20 | Manfish | or the config is wrong |
| 20:48.40 | Scoundrel | zaptel config? |
| 20:49.22 | Manfish | zaptel.conf |
| 20:49.28 | *** join/#asterisk luke-jr (n=luke-jr@CPE-24-31-246-32.kc.res.rr.com) |
| 20:49.57 | Scoundrel | ok, i saw in the log file it was bitching about the chan_capi ( Dec 4 21:46:12 WARNING[8024] loader.c: Loading module chan_capi.so failed! |
| 20:49.58 | *** part/#asterisk Zork_ (n=Zork_@j214090.upc-j.chello.nl) |
| 20:50.05 | *** part/#asterisk nettie (n=nettie@85-18-54-38.ip.fastwebnet.it) |
| 20:50.19 | Scoundrel | tried to reinstall chan_capi but just got alot of errors |
| 20:51.18 | Scoundrel | hm, yeah zaptel.conf says us on both i guess that is wrong |
| 20:51.21 | BlepsoaF | hell all, can someone take a peak at http://pastebin.ca/267340 - I cant figure out why the goto statement isnt working upon an invalid conference number being entered |
| 20:51.25 | BlepsoaF | it just disconnects |
| 20:51.28 | Manfish | what does zttool report? |
| 20:51.45 | Manfish | and ztcfg |
| 20:52.41 | Scoundrel | 0 channels 0 configured no errors.. hehe |
| 20:53.02 | Scoundrel | ztcfg gives no output |
| 20:53.12 | Manfish | ztcfg -vv |
| 20:53.14 | *** join/#asterisk luke-jr (n=luke-jr@CPE-24-31-246-32.kc.res.rr.com) |
| 20:53.36 | Scoundrel | 0 channels configured. |
| 20:53.54 | Manfish | what card are you using? |
| 20:54.06 | Scoundrel | eicon bri4 diva server |
| 20:54.49 | Manfish | is it installed correctly? sorry I never tried with an Eicon |
| 20:55.39 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-82-81-215-87.cablep.bezeqint.net) |
| 20:55.44 | Scoundrel | yeah i recompiled the kernel with the drivers and had it showing with capi info inside asterisk -r was also abel to dial inn but then i died with wasnt abel to find the asteris.ctl? |
| 20:55.46 | *** join/#asterisk djflux (n=djflux@mm.shermfin.com) |
| 20:56.33 | Scoundrel | rebooted and was still the same, so i tried to load in asterisk,lib,zaptel again then im stuck her with it wont start at all |
| 20:56.45 | Manfish | Like I say I never used an eicom card |
| 20:57.11 | Scoundrel | ok, got any clue what i could do without reinstalling ? im working remote on the server :) |
| 20:57.35 | *** join/#asterisk CleanerX (n=nix@p54A39258.dip0.t-ipconnect.de) |
| 20:57.43 | Manfish | you could remove and reinstall astrisk and the drivers |
| 20:59.11 | Scoundrel | ok know any howtos for it? |
| 20:59.26 | Scoundrel | (so i get it right) |
| 21:00.13 | Manfish | just download the latest from astrisk.org then extract compile and install |
| 21:00.36 | Scoundrel | ok, ill give it try.. thanks |
| 21:01.32 | Manfish | is this a trixbox? |
| 21:02.59 | *** join/#asterisk luke-jr (n=luke-jr@CPE-24-31-246-32.kc.res.rr.com) |
| 21:03.16 | Scoundrel | yes |
| 21:03.35 | Manfish | if it is see the following link to untrixbox it http://www.freepbx.org/2006/09/28/un-trixbox-your-trixbox/#more-7 |
| 21:04.14 | Manfish | there are a couple of obvious thins missing like there is another repo that needs editing out for trixbox |
| 21:04.50 | *** join/#asterisk dasenjo (n=dasenjo@190.24.177.199) |
| 21:04.59 | Manfish | and after you upgrade to * some of the email scripts for voice mail dont work |
| 21:05.16 | Scoundrel | okay, know what repo? |
| 21:05.38 | Manfish | the latest trix add the beta repo in yum |
| 21:06.20 | SwK[Work] | ok someone riddle me this |
| 21:06.28 | Scoundrel | okay, how do i remove it? |
| 21:06.48 | SwK[Work] | in the 1.2branch from SVN did someone "Fix" notifies for the polycom 2.0 sip software? |
| 21:07.25 | Manfish | Scoundrel the link i sent you is step by step |
| 21:07.51 | Scoundrel | ok just wondered since u said there where something missing there |
| 21:08.03 | Manfish | just read the extra comments at the bottom and when it comes to the disable repo step do the change to the other one as well |
| 21:08.24 | Scoundrel | ah okay, havent read all the way down yeat.. :) |
| 21:08.28 | Scoundrel | thx again |
| 21:08.31 | Manfish | np |
| 21:09.12 | *** join/#asterisk mrichmanM (n=richmanm@70.89.184.1) |
| 21:12.55 | *** join/#asterisk dasenjo_ (n=dasenjo@190.24.179.39) |
| 21:17.23 | *** join/#asterisk jarrod (i=nobody@dont.juniperyour.net) |
| 21:17.52 | jarrod | is there a way to leave the default musiconhold class for everyone, except one context uses another? |
| 21:18.06 | jarrod | i try SetMusicClass in the context dialplan but it sets it for everyone |
| 21:24.17 | *** join/#asterisk findlay (n=justin@67.137.24.115) |
| 21:26.30 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 21:27.21 | syzygyBSD | Hmm, why do we have the version and date for zaptel in the topic but not the stable asterisk? |
| 21:29.47 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 21:34.16 | *** join/#asterisk |Vulture| (n=Vulture@101.222.121.70.cfl.res.rr.com) |
| 21:34.23 | |Vulture| | Anyone here using an IP-301? |
| 21:36.05 | *** join/#asterisk adorah (n=admin@87.68.146.112.cable.012.net.il) |
| 21:36.28 | BlepsoaF | using a 430 and 501's and 601 |
| 21:36.49 | |Vulture| | yea I use the 501 didn't know if the 301s had a messages button |
| 21:36.57 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 21:36.58 | |Vulture| | or if you have to go through menu/features/messages |
| 21:37.07 | BlepsoaF | hmm not sure |
| 21:37.18 | adorah | Hi anyone with experience with astlinux/net4801? |
| 21:41.10 | BlepsoaF | is there a way to make meetme do something else after entering an invalid conference # |
| 21:41.25 | BlepsoaF | IE. right now it just hangs up no matter what I put in the dialplan |
| 21:41.58 | [hC] | anyone running the new 2.x firmware for polycoms? |
| 21:42.31 | *** part/#asterisk knathraak (n=zach@151.196.142.242) |
| 21:42.54 | *** join/#asterisk mut (n=ana@65.111.222.120) |
| 21:43.30 | BlepsoaF | [hC]: I am |
| 21:43.47 | [hC] | BlepsoaF: worth the upgrade? anything cool added? |
| 21:44.06 | BlepsoaF | i think the latest stuff is configuration related which you cant actually do anything with yet |
| 21:44.50 | BlepsoaF | actually a lot of bug fixes |
| 21:45.02 | BlepsoaF | 9 bug fixes |
| 21:45.14 | BlepsoaF | the rest is |
| 21:45.15 | BlepsoaF | The following configuration file changes have been included in this build in preparation for |
| 21:45.15 | BlepsoaF | future inclusion of the IP 650 platform in a software release. Support for the IP 650 is not |
| 21:45.15 | BlepsoaF | currently included in this release. |
| 21:45.39 | CunningPike | |Vulture|: There is no Messages button, according to the diagram in the Admin Guide |
| 21:45.59 | *** join/#asterisk fourcheeze (n=rich@82.153.23.79) |
| 21:46.15 | BlepsoaF | so does anyone know how to make meetme NOT hangup after an invalid conference number is entered? |
| 21:47.00 | Nugget | create every possible conference number in advance so that ther are no invalid numbers. |
| 21:47.19 | BlepsoaF | Nugget: thats a little extremem |
| 21:47.22 | BlepsoaF | extreme |
| 21:47.28 | [hC] | BlepsoaF: ahh gotcha.. just curious if the bugfixes were significant enough to upgrade |
| 21:47.32 | Nugget | I like to think outside of the box. |
| 21:47.42 | *** part/#asterisk alerios (n=alerios@190.24.98.197) |
| 21:47.54 | rob0 | [_] think |
| 21:47.55 | BlepsoaF | i would need a more viable solution |
| 21:48.00 | [hC] | you can pass a variable to meetme |
| 21:48.06 | [hC] | to dynamically create a room if it doesnt exist |
| 21:48.11 | CunningPike | BlepsoaF: Ours doesn't - it creates whatever conference I enter, even if it's not listed..... |
| 21:48.14 | syzygyBSD | Nugget: there is also a way to dynamically create them... |
| 21:48.15 | [hC] | show application meetme |
| 21:48.25 | syzygyBSD | so one is created for the number they enter |
| 21:48.53 | *** join/#asterisk Juggie (n=Juggie@74.105.235.221) |
| 21:49.13 | BlepsoaF | Looks like I will have to make an agi |
| 21:49.31 | *** join/#asterisk Dude34 (n=Aces1UP@ip68-96-234-176.lv.lv.cox.net) |
| 21:49.55 | Dude34 | does anyone here run a calling card business? |
| 21:51.02 | *** join/#asterisk converx (n=locid@206-248-132-2.dsl.teksavvy.com) |
| 21:51.12 | converx | ? |
| 21:52.11 | linlin | whats the best way to use a sip device behind a NAT, excluding DMZ or any port forwarding if possible |
| 21:52.28 | linlin | server is outside, no port restrictions, fully public dedicated server |
| 21:52.36 | linlin | phone is inside nat |
| 21:53.25 | *** join/#asterisk maverickbna (i=sentinel@wikipedia/Shadowhntr) |
| 21:55.57 | *** join/#asterisk debian_gnu_linux (n=wjrojas@165.98.229.210) |
| 21:56.32 | *** join/#asterisk dasenjo_ (n=dasenjo@190.24.177.209) |
| 21:58.05 | *** join/#asterisk sigmounte (n=sigmount@81.56.234.199) |
| 21:58.55 | Nugget | linlin: you should be ok with just putting nat=yes in sip.conf for that device. |
| 22:00.10 | linlin | ok |
| 22:00.14 | debian_gnu_linux | hi |
| 22:00.16 | linlin | anything to do on the nat side? |
| 22:00.36 | debian_gnu_linux | i have a pc installed asterisk |
| 22:01.15 | BlepsoaF | the sip device itself will need to support NAT, so it can rewrite its IP address as the public facing one |
| 22:01.52 | linlin | cisco phones, possibly other brands |
| 22:01.56 | linlin | support it you think? |
| 22:03.33 | BlepsoaF | last time I dealt with sip devices and a gateway I had to use SER |
| 22:04.07 | BlepsoaF | otherwise you'll run into one way audio situations |
| 22:06.55 | *** join/#asterisk icel (n=icel@63.78.162.41) |
| 22:08.01 | *** part/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 22:09.43 | icel | Question: If I hook up asterisk to a Voice T1 how many conversations should be able to simultaneously happen? |
| 22:09.48 | cy3o3 | ugh, anyone around to help me out with some zaptel shizz? |
| 22:10.22 | Assid | anyone seen trixter? |
| 22:11.13 | *** join/#asterisk jgoo (n=8d9j823d@foodtecsolutions.com) |
| 22:11.15 | jgoo | hrm |
| 22:11.24 | jgoo | so in agi there is SET AUTOHANGUP |
| 22:11.25 | converx | anyone knows how I can do 'group voicemail' with asterisk? |
| 22:11.42 | hmmhesays | read a little |
| 22:11.47 | jgoo | however, I would like to set this on the channel at the start, as calling this agi command over fastagi doesn't work |
| 22:12.00 | hmmhesays | the voicemail box that answers is whatever you set |
| 22:12.06 | debian_gnu_linux | i'll like to configure asterisk on my pc |
| 22:12.13 | jgoo | it still dies at 60 seconds, even though I set it to 0, or 10*60 |
| 22:12.13 | hmmhesays | mailboxes aren't bound to sip peers |
| 22:12.16 | debian_gnu_linux | to test |
| 22:12.23 | debian_gnu_linux | what i need ?? |
| 22:12.32 | Qwell[] | debian_gnu_linux: a computer |
| 22:12.35 | Qwell[] | ~docs |
| 22:12.45 | jbot | somebody said docs was Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
| 22:12.46 | *** join/#asterisk zmef420_ (n=zmef420@metarb3-pool4-183.mtco.com) |
| 22:12.46 | Qwell[] | ~book |
| 22:12.48 | jbot | rumour has it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 22:13.39 | jgoo | Qwell[]: how would you set the autohangup in extensions.conf? |
| 22:13.49 | debian_gnu_linux | but any hardware special |
| 22:13.52 | debian_gnu_linux | ?? |
| 22:14.01 | jgoo | debian_gnu_linux: what phone line do you have? |
| 22:14.22 | jgoo | you can buy an isdn card for 15 euros, HFC chipset, and use zaphfc |
| 22:15.56 | debian_gnu_linux | let me to see |
| 22:16.02 | *** join/#asterisk conico (n=chatzill@85.107.25.28) |
| 22:16.19 | *** join/#asterisk ShipHead (i=ShipHead@gateway/tor/x-e1e37bcaa8474c64) |
| 22:17.30 | debian_gnu_linux | i have a INNOMEDIA VoIP |
| 22:22.01 | *** join/#asterisk remmo (n=chatzill@202.172.106.161) |
| 22:22.16 | *** part/#asterisk remmo (n=chatzill@202.172.106.161) |
| 22:22.57 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 22:24.05 | *** join/#asterisk nextime (n=nextime@unaffiliated/nextime) |
| 22:27.08 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 22:33.03 | *** join/#asterisk jgoo (n=8d9j823d@foodtecsolutions.com) |
| 22:33.05 | *** join/#asterisk maverickbna (i=sentinel@wikipedia/Shadowhntr) |
| 22:34.35 | jgoo | just to clear things up SetVar is now Set.... but SetGlobalVar is still SetGlobalVar... right? |
| 22:38.42 | jgoo | trying to set / remove autohangup - anyone dealt with that before |
| 22:38.43 | jgoo | ? |
| 22:40.06 | icel | anyone know much about T1's? |
| 22:43.58 | j0 | my music on hold is EXTERMELY choppy when I'm doing it locally, I have an extension that just has MusicOnHold and it's terrible. It also happens on 1 of my IAX trunks, but all other IAX and SIP trunks play hold music perfectly!!! any ideas where to start? i'm all out |
| 22:46.35 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-43-253.lsanca.dsl-w.verizon.net) |
| 22:46.39 | icel | any recommendations on SIP phones, anyone? |
| 22:47.07 | j0 | ice1: theres tons out there.. what do u want it for? |
| 22:47.15 | [TK]D-Fender | icel : Polycom, Aastra, Linksys. In that order |
| 22:47.18 | j0 | what have you looked at so far? |
| 22:47.21 | icel | business use |
| 22:47.28 | j0 | yeah.. and grandstream last.. hehe |
| 22:47.44 | j0 | icel: what he said ;) |
| 22:47.48 | icel | I haven't looked at any yet |
| 22:47.53 | icel | k |
| 22:49.38 | j0 | AREWG!@#$ i'm gonna pull my hair out.. locally, and 1 of my IAX trunks has impossible hold music |
| 22:49.59 | j0 | on my IAX trunk, if i talk or just blow into the mic, the hold music keeps playing... but as soon as i stop making any noise, it cuts out |
| 22:50.14 | j0 | locally i can't figure out how to make hold music work at all |
| 22:50.36 | *** join/#asterisk LordBacon (n=kvirc@unaffiliated/frb) |
| 22:50.53 | LordBacon | I'm having trouble finding someplace to buy voip handsets for our office |
| 22:50.59 | LordBacon | anyone got a recommendation? |
| 22:51.10 | |Vulture| | LordBacon: whats your price range for headsets? |
| 22:51.13 | linlin | voipsupply |
| 22:51.30 | |Vulture| | I always use plantronics |
| 22:51.32 | LordBacon | the company I was told, we were getting them for like $80/phone |
| 22:51.44 | LordBacon | but they won't ship to a different address than the billing address |
| 22:51.47 | |Vulture| | any clue on model or features? |
| 22:51.56 | LordBacon | grandstream 2000 iirc |
| 22:52.22 | |Vulture| | ah Ive only purchased for Polycom IP-* phones |
| 22:52.58 | LordBacon | surprising to me, amazon doesn't have much in the line of voip phones |
| 22:54.19 | LordBacon | http://www.voipsupply.com/product_info.php?products_id=331 |
| 22:54.22 | ShadowHntr | www.atacomm.com |
| 22:54.22 | ShadowHntr | :) |
| 22:54.56 | ShadowHntr | from the reviews i've read for a good price check out the Tornado M5 and M20 |
| 22:55.26 | linlin | anyone have experence in repairing ip phones? |
| 22:55.35 | linlin | i have an ATCOM phone that doesnt do anything |
| 22:55.42 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.89) |
| 22:55.44 | linlin | probally useless, figured i'd give it a try |
| 22:57.17 | LordBacon | who makes the Tornado? |
| 22:57.23 | ShadowHntr | Lenoxa |
| 22:57.30 | ShadowHntr | voip-info reviewed it nicely |
| 22:57.39 | ShadowHntr | http://www.lenoxa.com/ |
| 22:58.28 | LordBacon | the M5 looks ok, but I'm worried about the angle of the LCD |
| 22:59.57 | ShadowHntr | wouldn't know |
| 23:00.13 | ShadowHntr | but the Tornado M5 or M20 is probably what i'll go when i implement hard phones |
| 23:00.22 | *** join/#asterisk aao_pwner (i=asd@c-24-21-91-140.hsd1.wa.comcast.net) |
| 23:00.56 | *** part/#asterisk debian_gnu_linux (n=wjrojas@165.98.229.210) |
| 23:01.28 | *** join/#asterisk adorah (n=admin@87.68.172.221) |
| 23:02.02 | *** join/#asterisk SparFux (n=raoul@e182019049.adsl.alicedsl.de) |
| 23:06.03 | *** join/#asterisk Prelius (n=pzotov@70.88.213.25) |
| 23:06.53 | *** join/#asterisk oneeyedelf1 (n=knic@cpe-69-205-162-208.stny.res.rr.com) |
| 23:07.26 | LordBacon | ShadowHntr: where is this rating you saw? I'd like to read the reviews |
| 23:07.31 | *** part/#asterisk dasenjo (n=dasenjo@190.24.177.209) |
| 23:07.37 | ShadowHntr | it was a short one |
| 23:07.37 | *** join/#asterisk peyote (n=kvirc@port-83-236-4-51.dynamic.qsc.de) |
| 23:07.56 | ShadowHntr | http://www.voip-info.org/wiki/view/VOIP+Phones+Reviews |
| 23:08.03 | ShadowHntr | second to last one |
| 23:08.19 | oneeyedelf1 | I want to get an fxs network adapter for christmas, but dont know what one I should ask for, all I know is someone said Grandstreams are crap |
| 23:09.37 | LordBacon | bleh |
| 23:09.40 | LordBacon | there is no review there |
| 23:10.02 | icel | are snom ip phones decent? |
| 23:10.14 | JT | ShadowHntr: pci card or ata? |
| 23:10.21 | ShadowHntr | JT: eh? |
| 23:10.28 | JT | your fxs adapter |
| 23:10.30 | JT | ATA i assume |
| 23:10.38 | ShadowHntr | i didn't say anything about an fxs adapter |
| 23:10.43 | ShadowHntr | i was talking about a hard phone |
| 23:10.44 | JT | oh |
| 23:10.50 | JT | i meant oneeyedelf1 |
| 23:10.57 | Prelius | Folks, I was wondering if someone can help me to configure two VoIP cards: |
| 23:10.58 | Prelius | I have a Sangoma A100 with at PRI int, and Digium 400 with 4 FXS interfaces for FAX machine... My zaptel.conf looks like this: |
| 23:10.58 | Prelius | span=1,1,0,esf,b8zs |
| 23:10.58 | Prelius | bchan=1-23 |
| 23:10.58 | Prelius | dchan=24 |
| 23:11.07 | JT | for some reason your nicks looked exactly the same |
| 23:11.14 | JT | i haven't fully woken up yet |
| 23:11.24 | Prelius | When I power the TDM card Asterisk stops working... |
| 23:11.31 | oneeyedelf1 | JT: network, external |
| 23:11.39 | JT | oneeyedelf1: so an ATA |
| 23:11.43 | oneeyedelf1 | okay |
| 23:11.55 | JT | helps to know the name of what you want :P |
| 23:12.01 | shido6 | dood |
| 23:12.03 | shido6 | thats scary |
| 23:12.04 | *** join/#asterisk sloth (n=josh@cpe-69-203-212-182.nyc.res.rr.com) |
| 23:12.07 | LordBacon | do they make IAX2 hardphones? |
| 23:12.11 | shido6 | when u power the tdm card asterisk stops working? |
| 23:12.18 | JT | common wisdom seems to be that for consumer use, sipura stuff is the way to go |
| 23:12.24 | Prelius | Yes... |
| 23:12.25 | shido6 | why is asterisk running before you power the TDM card? |
| 23:12.46 | Prelius | It works fine with just sangoma card... |
| 23:13.01 | JT | you didn't answer the question |
| 23:13.10 | oneeyedelf1 | JT: I was looking at this, http://www.voipsupply.com/product_info.php?products_id=320 but was wondering why it said unlocked, I heard some linksys stuff was hacked but after poweroutage it could revert |
| 23:13.16 | JT | why would you be applying power to the card after asterisk has started |
| 23:13.19 | Prelius | I want to plug in fax machines into digium card... |
| 23:13.20 | *** join/#asterisk Soul (n=Soul@87-196-111-228.net.novis.pt) |
| 23:13.37 | oneeyedelf1 | JT: do you konw if that will revert if it looses pwoer? |
| 23:13.42 | JT | Prelius: we aren't asking what you want to do |
| 23:13.58 | JT | Prelius: we are asking why on earth you are applying power to the card with the computer already on |
| 23:14.12 | Prelius | No, I shut down the astersik box, plug in power into digium card, turn on the box, and asterisk would not start... |
| 23:14.36 | JT | do you know why it would not start? |
| 23:14.56 | *** part/#asterisk peyote (n=kvirc@port-83-236-4-51.dynamic.qsc.de) |
| 23:15.11 | Prelius | Yes, my ztcfg complains about incorrect spans... |
| 23:15.20 | *** join/#asterisk morex (i=morex@host81-157-4-188.range81-157.btcentralplus.com) |
| 23:15.24 | morex | Evening all |
| 23:15.28 | JT | oneeyedelf1: the sipura 3000 or 3102 (i think, newer model) does FXS and FXO with failover power outage relay |
| 23:15.29 | xheliox | Any dCAPs around? |
| 23:15.53 | morex | I think safe_asterisk is blocking my daily anacron run |
| 23:15.53 | JT | Prelius: then your /etc/zaptel.conf is incorrectly configured |
| 23:15.53 | *** join/#asterisk GaVak (n=denniso@adsl-074-228-124-003.sip.sav.bellsouth.net) |
| 23:15.54 | morex | Anybody seen this before? |
| 23:15.58 | JT | Prelius: obviously it needs to be reconfigured when you add a new card |
| 23:16.10 | shido6 | yep I see a lot of "?" 's , morex :) |
| 23:16.10 | GaVak | What is a suggested windows call viewer/monitor software for *? |
| 23:16.11 | LordBacon | does the SNOM 300 have a tilting lcd? |
| 23:16.17 | robin_sz | Prelius, you have to be careful with spans, as they are rleated ot the order in which the kernel modules are loaded |
| 23:16.20 | oneeyedelf1 | alright, i think my parents can spring for that, THanks JT |
| 23:16.21 | morex | Shido: I'll bet you do :-) |
| 23:16.32 | shido6 | heheh, let me scroll up maybe I missed it |
| 23:16.50 | morex | well, I have a script in /etc/cron.daily to restart asterisk at midnight |
| 23:16.58 | Prelius | So digium card loads first, Sangoma second... |
| 23:17.03 | JT | oneeyedelf1: apparently it's not worth getting the 3000 anymore with the price difference to the newer model |
| 23:17.08 | morex | Which it does, but the /etc/init.d/asterisk script hangs around as a defunct process |
| 23:17.19 | JT | oneeyedelf1: make sure you do not buy one locked to a voip provider |
| 23:17.19 | morex | which prevents the rest of my /etc/cron.daily scripts from running |
| 23:17.32 | JT | most of the ones in normal consumer shops are locked to a provider |
| 23:17.34 | morex | and prevents anacron from running the next day (as the defunct process is still there) |
| 23:17.45 | Prelius | my zaptel.conf specifies sangoma chanels as 1-23... |
| 23:18.03 | robin_sz | mmm no |
| 23:18.10 | JT | Prelius: look at dmesg output to see which driver loads first |
| 23:18.33 | Prelius | JT: digium forst, then Sangoma |
| 23:18.38 | JT | Prelius: zaptel numbers spans and channels in the order they are initialised |
| 23:18.44 | *** join/#asterisk ozoneco (n=stanp@CPE-24-27-138-124.neb.res.rr.com) |
| 23:18.46 | JT | well you'll need to modify it then |
| 23:18.54 | robin_sz | it tries to set up 23 channels on the first device ... if the first device changes .... |
| 23:19.39 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
| 23:19.53 | robin_sz | chan_mISDN hasnt crashed for four days now! |
| 23:20.16 | JT | heh |
| 23:20.20 | JT | good to hear i guess |
| 23:21.26 | Prelius | I see, so I need to set up my digium card first: number channels 1-4. and them my sangoma card, channels 5-28... |
| 23:21.41 | JT | yes, and renumber the d channel too |
| 23:21.46 | JT | i assume you are using a pri |
| 23:21.55 | JT | considering there's only 23 chans |
| 23:21.58 | Prelius | ON sangoma, not on digium |
| 23:22.15 | JT | yes, on the digital card obviously |
| 23:22.23 | Prelius | I have 4 fxs modules |
| 23:22.34 | JT | yeah i read that bit |
| 23:23.07 | Prelius | Ok, thank you very much for your help... Gonna try it now... |
| 23:25.26 | *** join/#asterisk jm|home (n=jamiem@zen.jamiem.com) |
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| 23:31.20 | *** join/#asterisk puddingpimp_work (n=puddingp@gateway.quickcircuit.co.nz) |
| 23:31.29 | puddingpimp_work | Where does asterisk drop it's core files? |
| 23:31.55 | [TK]D-Fender | puddingpimp_work : What do you define as "core"? |
| 23:32.09 | puddingpimp_work | whatever -g is supposed to make it drop |
| 23:32.22 | [TK]D-Fender | puddingpimp_work : "-g"? |
| 23:32.24 | puddingpimp_work | Asterisk is crashing without any log messages |
| 23:32.45 | [TK]D-Fender | AH, core dump... |
| 23:32.53 | puddingpimp_work | -g Remove resource limit on core size, thus forcing Asterisk to |
| 23:32.54 | puddingpimp_work | <PROTECTED> |
| 23:33.00 | [TK]D-Fender | in /var/log/asterisk I think.... |
| 23:33.04 | *** join/#asterisk [1]HarryR (n=Harry@host-83-146-53-46.bulldogdsl.com) |
| 23:33.21 | hmmhesays | the core is dumped to where you called asterisk from I believe |
| 23:34.43 | puddingpimp_work | do you know the pwd rc.local is exec'd in on RH9? |
| 23:35.49 | file | [TK]D-Fender: it's you! |
| 23:42.58 | *** join/#asterisk jm|home (n=jamiem@zen.jamiem.com) |
| 23:52.53 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 23:53.35 | icel | does anyone use a digium TE4xxP with a voice T1? |
| 23:53.56 | *** join/#asterisk So3kris (n=jan-will@ids.netland.nl) |
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