| 00:00.04 | shido6 | exten => 100,1,Dial(SIP/boinkee|20|m) |
| 00:00.29 | afe | mgth: did it help? |
| 00:00.53 | mgth | Nope....they had no clue what they were talking about, they told me that it had to with dma not being enabled with my hd |
| 00:01.41 | mgth | afe: as the owner of that bug, you can reopen it. If you do, then will post more information on it |
| 00:02.05 | mgth | unless drumkilla wants to comment on it |
| 00:02.30 | afe | I got recommended to place it as a bug here, so I'll try to reopen it |
| 00:03.06 | mgth | afe: do you ever get a random call.....the phone rings and no one is there? |
| 00:04.24 | afe | yes |
| 00:04.34 | mgth | We have the exact same issue |
| 00:04.47 | afe | after closing a valid call that happens |
| 00:04.49 | *** join/#asterisk RUok (~mike@host-69-19-166-10.rev.o1.com) |
| 00:04.50 | tuxinator_linux2 | if * craps out or freezes, is the rest of the server still stable? |
| 00:05.02 | afe | anyway, the bug is reopened |
| 00:05.04 | mgth | Reopen the bug and place that in there. I will comment with the same and maybe drumkiller or gram can do something or say something |
| 00:06.14 | *** join/#asterisk pulu (~chatzilla@65.77.78.3) |
| 00:06.22 | tuxinator_linux2 | or a better question is, is it safe to play with on a production machine? |
| 00:06.42 | afe | mght: where are you located? |
| 00:06.42 | flewid | _62X[0-9], |
| 00:06.43 | mikeirc | Nothing like blasting classical music through the headphones and a cup of black joe to get the brain going... just thinking out load again... |
| 00:06.55 | flewid | that will match 62[0-9][0-9] right? |
| 00:07.03 | *** part/#asterisk Grooby (~Grooby@12.22.232.212) |
| 00:07.21 | pulu | i just attempted to upgrade to the newest cvs and am now getting segfaults, trying to backtrace and strace but I don't really understand what gdb is saying, does anyone know if there are any problems with the current cvs? |
| 00:07.38 | CoaxD | pulu: The problem is, you're probably not used to using gdb in a threaded environment |
| 00:07.42 | CoaxD | pulu: (neither am i.) |
| 00:08.02 | flewid | pulu: i'm using cvs as of today around 2pm |
| 00:08.04 | flewid | and it's working for me |
| 00:08.06 | mgth | afe: note posted |
| 00:08.37 | pulu | three of the four functions are from chan_sip, so is that a pretty good indication that is where the problem is? |
| 00:09.41 | afe | mgth: and answer from markster as wll ... |
| 00:09.56 | mgth | afe: huh? |
| 00:10.02 | *** join/#asterisk dalabera (~Dalabera@mail.pmrtechnologies.com) |
| 00:10.03 | pulu | ahh, moving sip.conf to something else lets it start... I must have something wrong with the config file |
| 00:10.04 | jsolares | what's the correct syntax for record and silence detection? record(myfile:wav,1/10000) would that wait for 1 second previous and 10 secs after? |
| 00:10.08 | afe | check the bug again |
| 00:10.12 | dalabera | hello gurus... |
| 00:11.03 | tuxinator_linux2 | Hello dalabera |
| 00:11.38 | machinehd | when someone records their personal voicemail greeting where does it save the file? |
| 00:11.51 | tuxinator_linux2 | machinehd, /dev/null |
| 00:12.10 | tuxinator_linux2 | sorry, couldn't resist |
| 00:12.13 | machinehd | :) |
| 00:12.29 | tuxinator_linux2 | need some food, on a quest to find some.... |
| 00:12.36 | ariel_ | afe, have you called back to digium's support department about the non-bug but support issue? |
| 00:12.59 | ariel_ | hello how is everyone doing tonight? |
| 00:13.26 | jsolares | good, just having problems to understand silence detection with record |
| 00:14.17 | afe | ariel_, my tries to the support phone # didn't get an answer |
| 00:14.36 | ariel_ | afe, what times are you calling your in Sweden correct? |
| 00:14.41 | afe | ye |
| 00:14.56 | jsolares | hrm, well the wiki is wrong, no wonder that didnt work |
| 00:14.57 | afe | I tried daytimes as well |
| 00:14.59 | machinehd | so, anyone? where are the voicemail greetings saved? I'm building a new * box and I need to replicate the old one |
| 00:15.16 | afe | (US daytime that is) |
| 00:15.32 | Inv_arp | machinehd: /var/lib/asterisk/sounds i think |
| 00:15.35 | jsolares | machined /vas/spool/asterisk/voicemail |
| 00:15.39 | RUok | machinehd, /var/spool/asterisk/voicemail/<context>/<mailbox>/(busy|unavail).(WAV|gsm|wav) |
| 00:15.43 | ariel_ | jsolares, if the wiki is wrong fix it on there. |
| 00:15.54 | machinehd | awesome, thanks guys |
| 00:15.58 | Mavvie | machinehd: locate voicemail |
| 00:16.18 | afe | but e-mail should work, shouldn'¨t it? :) |
| 00:16.21 | jsolares | ariel_: i'm going to check the code to se if it expects milisecs or secs before screwing up the wiki :) |
| 00:16.29 | ariel_ | afe, yes |
| 00:16.55 | ariel_ | jsolares, I don't use silence detection. |
| 00:17.11 | jsolares | well it's secs, it multiplies by 1000 in the code |
| 00:17.34 | jsolares | i "want"/"need" it to be able to monitor a channel and record only when there's sounds |
| 00:18.14 | Mavvie | to do that, you first need to know how silence sounds. |
| 00:19.03 | jsolares | luckily there's the dsp that has a silence detector with a variable threshold |
| 00:19.07 | *** join/#asterisk kodomo (~memyself@emu.net.informatik.tu-muenchen.de) |
| 00:20.24 | afe | I'll call the support number tomorrow to check if they have anything to say |
| 00:20.47 | ariel_ | afe, are you using the exten 500 to call them via iax2? |
| 00:21.21 | afe | no, I setup a context for iaxtel |
| 00:22.30 | afe | I don't want my children (or wife) calling digium too easily :) |
| 00:23.23 | *** join/#asterisk syslod (~yurplsl@65.114.15.26) |
| 00:23.29 | syslod | Hello |
| 00:23.32 | mikeirc | just curious... what linux/bsd distros are you guys running? Thanks. |
| 00:23.53 | Goshen | mikeirc: Mandrake 10.0 with 2.6.10 kernel |
| 00:24.04 | ariel_ | mikegrb, I use centOS 3.4 |
| 00:24.09 | *** join/#asterisk TekLexus (~Mnemonic@167.206.75.24) |
| 00:24.16 | Inv_arp | mikeirc: debian 350mhz 128ram 2.4.27 |
| 00:24.17 | syslod | Anyone have a good example of IAX2 between two * servers. I can't seem to get them to authenticate. |
| 00:24.57 | ariel_ | syslod, post your settings on pastebin.ca it's not too hard to have them talk to each other. |
| 00:25.43 | Inv_arp | mikeirc: heh 350mhz handles * , zap fxo 2 sips, apache/php and firewall no hiccups try that on another OS |
| 00:25.51 | ctooley | select count(*) from extensions_table return 14672 rows, I now remember why we're using Realtime, I ain't managin those in a flat file. |
| 00:25.54 | TekLexus | hey guys...got a question for you.. I installed my asterisk server 3 weeks ago... havent made alot of calls from it..suddenly my speed from my optimum online cable modem is dead slow.. i call up cablevision, and they tell me they throttled my upload speed because I was using so much bandwidth... i maybe made 10 calls from the server and dont remember uploading anything large anywhere |
| 00:26.02 | TekLexus | any idea if * could be causing it? |
| 00:26.27 | tuxinator_linux2 | porn |
| 00:26.40 | Inv_arp | TekLexus: nope optimum sucks |
| 00:26.44 | ariel_ | use ethereal to see what is going on. |
| 00:26.46 | TekLexus | grrrr |
| 00:26.50 | ctooley | TexLexus: chkrootkit is your friend |
| 00:26.57 | Inv_arp | never heard an isp do that |
| 00:26.58 | TekLexus | chkrootkit? |
| 00:27.31 | RUok | TekLexus, Perhaps someone cracked your box? |
| 00:27.32 | ctooley | chkrootkit will tell you how your box has been rooted... it's just a guess |
| 00:27.33 | Inv_arp | TekLexus: implying u got rooted |
| 00:27.36 | TekLexus | they told me i violated their TOS...when they never sent me any terms of service,and their TOS is impossible to find even on their site |
| 00:28.19 | syslod | http://pastebin.ca/6816 |
| 00:28.20 | ariel_ | TekLexus, have you check for a trojen bot? |
| 00:28.26 | TekLexus | running a server is against their TOS...period |
| 00:28.38 | TekLexus | my systems are virus free |
| 00:28.49 | TekLexus | chkrootkit - command not found |
| 00:28.52 | tuxinator_linux2 | check you /etc/passwd for new users |
| 00:29.04 | jsolares | anyone here know what the threshold for silence in the dsp is? db? |
| 00:29.27 | RUok | http://freshmeat.net/projects/chkrootkit/ |
| 00:29.47 | TekLexus | no new users in /etc/passwd |
| 00:30.36 | TekLexus | checking chkrootkit |
| 00:31.21 | tuxinator_linux2 | I had a old unprotected server and found a few new users add. I found it about an hour after it happened. He got in trough FTP server. |
| 00:32.06 | syslod | ariel_: http://pastebin.ca/6816 |
| 00:32.08 | tuxinator_linux2 | is you syslog running? |
| 00:32.28 | ariel_ | syslod, yes I see it just a sec. |
| 00:32.35 | syslod | Thnks |
| 00:32.43 | TekLexus | i think syslog is running |
| 00:32.51 | TekLexus | i have the logwatch for asterisk in the root email |
| 00:33.04 | TekLexus | i get tries to get rooted all the time, none successful |
| 00:33.04 | mikeirc | TekLexus: That's screwed up... if you're just running SIP like me change your outgoing user-agent string to reflect say that of a vonage voip unit... and tell them it's just a little vonage box hooked up to your phone..... I hate shiot like that. |
| 00:33.32 | TekLexus | thats what i told them |
| 00:33.53 | mikeirc | and that violates their TOS?? |
| 00:33.59 | TekLexus | i said how come you can offer a voice over ip system directly thru your cable modem and then deny people who use other services for makeing phone calls |
| 00:34.11 | TekLexus | any type of server violates their tos |
| 00:34.52 | ariel_ | syslod, are both servers on same lan? |
| 00:35.06 | syslod | They are routed 0 to 1 |
| 00:35.07 | mikeirc | Um... wtf? In the context that they're speaking even a standard pc hooked up to the net via the account would be a server then... idiots |
| 00:35.54 | pulu | does anyone know what might cause an asterisk system to segfault anytime it has anything to do with chan_sip? (register, or make an outgoing call) I just upgraded to the cvs today and it really doesn't seem to be happy |
| 00:36.18 | mikeirc | ok... tell them that you're making SIP calls via x-ten lite...from a windows xp home pc... now what? hehe |
| 00:36.53 | syslod | ariel_: Both private connected via routed 100meg connection. |
| 00:37.03 | TekLexus | chkrootkit came out clean yall |
| 00:37.45 | ariel_ | yes I am removing all the junk you have on there and also putting the right setup for them.. give a few I will repost it on to the pastebin when I am done. |
| 00:38.16 | syslod | Thanks, |
| 00:38.21 | ariel_ | pulu, cvs head or stable? |
| 00:39.23 | TekLexus | im blockin evry port cept for sshd now |
| 00:41.08 | ariel_ | TekLexus, don't tell them your running a server. |
| 00:41.12 | pulu | ariel_: cvs head, i just downloaded it about 30 minutes ago |
| 00:41.13 | afe | ariel_: just tried calling digium support on iaxtel, but that didn't work very well - I suppose I will have to try again tomorrow on a landline ... |
| 00:41.13 | TekLexus | i didnt |
| 00:41.19 | TekLexus | i told em i had vonage |
| 00:41.20 | *** part/#asterisk kodomo (~memyself@emu.net.informatik.tu-muenchen.de) |
| 00:41.34 | ariel_ | pulu go back to stable |
| 00:42.10 | afe | and I know it's closed by now ;) |
| 00:42.38 | TekLexus | btw... anyone wanna trade me FXO module from my TDM400P for a FXS Module? |
| 00:43.05 | hardwire | heh |
| 00:43.12 | *** join/#asterisk algorithmn (~na@ool-18bce89c.dyn.optonline.net) |
| 00:44.04 | *** join/#asterisk ast_freak (~ast_freak@hades-out.universalsystems.net) |
| 00:46.44 | machinehd | any cisco users having problems with the 7.3 firmware? (7960) I've tried multiple phones and they keep randomly locking up on me. |
| 00:48.25 | afe | exit |
| 00:48.28 | afe | oops |
| 00:48.44 | *** join/#asterisk hermie (~nick@24.236.167.53.bay.mi.chartermi.net) |
| 00:48.48 | Juggie | machinehd, how are they breaking? |
| 00:48.55 | ariel_ | syslod, you have user in both setups? |
| 00:49.02 | flewid | i have a question about the new calling format for iax in HEAD - is it Dial(IAX2/blah:blah@host/ext@context) |
| 00:49.03 | flewid | or |
| 00:49.18 | flewid | Dial(IAX2/blah:blah@host-ext@context) |
| 00:49.35 | Juggie | flewid, read the bugnot on mantis |
| 00:49.56 | tuxinator_linux2 | ~mantis |
| 00:49.57 | jbot | mantis is probably the opie bugtracker located at http://opie.info or something that tille says has a life of its own. at http://www.opie.info/mantis/ |
| 00:50.01 | flewid | Juggie: i'm just checking the cvs list and i have here 'the naming convention has changed from from / to -" |
| 00:50.04 | ariel_ | syslod, here is the beginning of it. http://pastebin.ca/6817 |
| 00:50.06 | flewid | and i'm confused as to what that means |
| 00:50.15 | ariel_ | now show me how you do the dialing between them? |
| 00:51.55 | ariel_ | syslod, I have to run down to the store to restock my frig I should be back in an hour. |
| 00:52.28 | ctooley | How do i play the dtmf tones in Asterisk::AGI? |
| 00:52.58 | ctooley | 0,1,2,3,4,5,6,7,8,9,*,# tones... do i need to playback digits/1.gsm, etc? |
| 00:54.11 | flewid | Juggie: any idea on the bugid? |
| 00:55.05 | *** join/#asterisk mesi (~player@dsl-082-083-145-120.arcor-ip.net) |
| 00:56.09 | *** join/#asterisk Goldenear (~goldenear@d193.dhcp212-198-200.noos.fr) |
| 01:03.15 | Goldenear | Hi. I've a very simple question about asterisk but I haven't find the exact anwser anywhere. The question is: |
| 01:03.16 | Goldenear | When two IAX clients (let's say A and B) are registred to the same asterisk server (let's call it S), if A call B, will all the traffic always go thru S after the call is initiated or can the traffic directly go between A and B (like with SIP/SER) ? I hope somebody here answer my question and can answer it :) thanks |
| 01:03.54 | wildcard0 | it can |
| 01:04.04 | Darwin35 | the call is iniated threw the server and handed off to a direct call I believe |
| 01:04.09 | mesi | A talks directly to B. |
| 01:04.15 | Beirdo | if the codecs match, etc |
| 01:04.49 | Goldenear | ok, so no bandwidth is needed on S during the call ? |
| 01:04.59 | mikegrb | Darwin35: iniated? threw? |
| 01:05.00 | wildcard0 | i believe you just have to set notransfer=no |
| 01:05.11 | Darwin35 | well it still uses bandwidth between the 2 units |
| 01:05.39 | Darwin35 | bad spelling |
| 01:05.45 | Goldenear | great :) |
| 01:05.51 | Darwin35 | they still send packets |
| 01:06.00 | mikeirc | Gotta love broadband... downloading an iso at a steady 579KB/sec! |
| 01:06.09 | Goldenear | and if A and B are behind a NAT ? |
| 01:06.10 | *** join/#asterisk jeofrey (~jeofrey@espeed19-153.brunet.bn) |
| 01:06.21 | Darwin35 | no problem |
| 01:06.24 | mikegrb | s/a// |
| 01:07.20 | wildcard0 | forward udp/4569 at least one way |
| 01:07.22 | *** part/#asterisk ctooley ([U2FsdGVkX@199.89.146.18) |
| 01:07.41 | bjohnson | can I use ssh to forward from my localhost, through the machine I am ssh'd to, and to a third machine (inside the lan) for vnc? |
| 01:07.56 | mikegrb | ... |
| 01:08.28 | wildcard0 | yes |
| 01:08.57 | bjohnson | I can't seem to get it to work |
| 01:09.00 | wildcard0 | ssh -Llocalport:<far end ip>:<far end port> <intermediary box> |
| 01:09.15 | kpfleming | add -g |
| 01:09.18 | pulu | bjohnson: -L localport:remotehost:vncport |
| 01:09.58 | wildcard0 | don't add -g :) |
| 01:10.01 | Goldenear | So two NATed IAX clients cant directly talk together when a call is starded ? In this case, asterisk could manage as many connexions as SER, right ? |
| 01:10.21 | wildcard0 | Goldenear, forward udp/4569 at least one way |
| 01:10.45 | wildcard0 | asterisk can be configured to do that |
| 01:10.50 | Essobi | Anyone use the ASterisk TAPI interface? |
| 01:11.05 | bjohnson | hmm .. must be some other problem. It looks like it starts to connect and the vncviewer says end of stream |
| 01:11.54 | bjohnson | and the ssh connection says channel 3: open failed: connect failed: No route to host |
| 01:12.21 | flewid | + |
| 01:12.26 | wildcard0 | are you sure the port is open? |
| 01:12.33 | wildcard0 | no firewalling issues? |
| 01:12.37 | Goldenear | wildcard0: I don't get it ? do you mean A -> S -> B -> A ? |
| 01:12.39 | bjohnson | on the loclhost? |
| 01:13.21 | wildcard0 | bjohnson, on the boxes you're trying to connect to the ports of |
| 01:13.46 | flewid | hmm, if i have two asterisk servers, connected via iax, an _62XX in the 'incoming' context on server A (the master), to dial the other server, i can hit 6200 and it'll ring to phones on the other server |
| 01:14.06 | Essobi | Damnit |
| 01:14.09 | wildcard0 | Goldenear, i mean you need to forward udp/4569 through at least one firewall to the box behind it |
| 01:14.11 | Essobi | TAPI is busted. |
| 01:14.13 | flewid | now if i want to add that to a call group, where i have dial(sip/7000&SIP/7001) |
| 01:14.28 | flewid | can i make it SIP/7000&SIP/7001&6200 |
| 01:14.35 | flewid | or do i have to put the technology in there? |
| 01:15.03 | *** join/#asterisk twisted (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) |
| 01:15.03 | *** mode/#asterisk [+o twisted] by ChanServ |
| 01:15.16 | wildcard0 | SIP is the technology |
| 01:15.18 | pulu | is anyone here using the new jitter buffer? Does it help? |
| 01:15.24 | flewid | wildcard0: i know |
| 01:15.45 | wildcard0 | then i dun understand the question :) |
| 01:15.48 | Darwin35 | now sip2 i believe |
| 01:16.13 | flewid | i'm using _62XX,1,Dial(IAX2/user:pass@host/${EXTEN}@${CONTEXT}) |
| 01:16.24 | flewid | but i have call groups, and want to add those users on server B to the call groups |
| 01:16.38 | flewid | so i was wondering if i could just toss 6200 in there and it'd match, and dial the second server |
| 01:16.44 | syslod | Anyone have suggestions on debuging auth problems between IAX2 trunked machines? |
| 01:17.18 | wildcard0 | flewid, no reason why you can't put that in if im understanding correctly |
| 01:17.28 | wildcard0 | syslod, ethereal? :) |
| 01:17.32 | Essobi | Ahh fuckit |
| 01:17.33 | flewid | wildcard0: so, if i have my pattern matching defined within my group context |
| 01:17.37 | flewid | then just |
| 01:17.38 | Essobi | I'll just fix the damn tapi driver myself |
| 01:17.50 | Darwin35 | is kram around |
| 01:17.54 | flewid | dial(SIP/XXXX&SIP/YYYY&6200) |
| 01:17.59 | flewid | it'll match that 6200 and dial it out via iax2? |
| 01:18.04 | flewid | to second server |
| 01:18.29 | Goldenear | wildcard0: ok thanks for your explanation :) If I understand correctly, it won't work if there are several IAX client behind each NAT/firewall. I assume in this case all the calls will have to go thru S... right ? |
| 01:18.40 | syslod | wildcard0: I get a causecode 50. |
| 01:18.45 | *** join/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
| 01:19.00 | wildcard0 | Goldenear, they have to have some path. so something has to be fowarded |
| 01:19.11 | wildcard0 | syslod, i dunno it off the top of my head. google for that? |
| 01:20.12 | Essobi | Has anyone seen and manager patches to lock in only seeing certain events on certain ports |
| 01:20.17 | Essobi | err |
| 01:20.22 | Essobi | on certain logins.. |
| 01:20.31 | syslod | I have. There is an example. Everything registers and it attempts to connect but something is wrong on the auth. |
| 01:20.39 | Essobi | Like I dont' want to send ALL my manager traffic to everyone. |
| 01:21.10 | wildcard0 | syslod, version problem? |
| 01:21.14 | Essobi | Just the phone events they are authorized to see.. |
| 01:21.30 | syslod | same verison. |
| 01:21.36 | Essobi | sys was that reply meant for me? |
| 01:21.52 | Goldenear | wildcard0: ok I get it now, thank to you (and to the others) |
| 01:22.27 | wildcard0 | syslod, i assume you've double checked the username/passwords |
| 01:22.29 | syslod | Essobi: No sorry |
| 01:22.35 | syslod | wildcard0: Yep |
| 01:22.56 | wildcard0 | does it fail on registration? or on dial? |
| 01:23.04 | Essobi | roger that |
| 01:23.05 | *** join/#asterisk Dr-Linux (~sshah@202.125.141.11) |
| 01:24.26 | flewid | syslod |
| 01:24.32 | flewid | i was having a bitch of a time too with iax => iax |
| 01:24.37 | flewid | this page got me working |
| 01:24.56 | flewid | http://lists.digium.com/pipermail/asterisk-users/2004-august/060397.html |
| 01:26.35 | syslod | On dial. Registration works |
| 01:26.42 | syslod | No authority found... |
| 01:27.44 | TSCHAK | what does it mean when you get a disconnect_string of = no resource ? |
| 01:28.13 | TSCHAK | i'm noticing that on my gnugk debug output |
| 01:28.27 | syslod | flewid: I get page not found. |
| 01:28.59 | flewid | hmm |
| 01:29.03 | flewid | sec |
| 01:29.45 | flewid | oops |
| 01:29.46 | flewid | change that to |
| 01:29.49 | flewid | 2004-August |
| 01:31.42 | syslod | Cool. I think I have most of that its likely just something simple I'm missing. I have it registering and it seems to dial since I can see the refusal of the call. |
| 01:34.57 | syslod | flewid: You even have it registering but it will not accept a call? |
| 01:35.52 | *** join/#asterisk bwzb (~wlwzhang@ip67-153-142-73.z142-153-67.customer.algx.net) |
| 01:36.34 | *** join/#asterisk libpcp (libpcp@210.16.20.5) |
| 01:36.38 | flewid | i'm not onto 'part c and d' |
| 01:36.39 | libpcp | hi all |
| 01:36.47 | flewid | right now i'm just making sure my calls make it from one server to server b |
| 01:36.47 | bwzb | Hi, anyone had experience on building asterisk in CentOS? I failed to build zaptel. |
| 01:37.08 | libpcp | i would like to ask, if anyone has an issue with asterisk while using sata drives ? |
| 01:37.32 | footnote | yer all so needy :) |
| 01:38.03 | footnote | can we have a group hug now? |
| 01:38.22 | bwzb | I'm suprise to see that only me have issue with CentOS |
| 01:38.26 | footnote | hrm, nevermind. |
| 01:38.32 | footnote | no chix here. |
| 01:38.56 | *** join/#asterisk jsolares (~jsolares@200.12.33.64) |
| 01:39.20 | footnote | we need more hawt wimmen on asterisk! |
| 01:39.27 | |Vulture| | wow... |
| 01:39.42 | jsolares | anyone here have an acct with voipjet? |
| 01:39.56 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 01:41.49 | bjohnson | yes |
| 01:42.04 | jsolares | hmm, apparently i'm not legally allowed to say i have an account with them, weird TOS |
| 01:42.05 | pulu | jsolares: yeah |
| 01:42.18 | jsolares | i always gets busy when i try calling with voipjet |
| 01:42.41 | pulu | when you do an iax debug what does it show? |
| 01:44.04 | pulu | jsolares: and what do you have for your dial cmd in extensions.conf? |
| 01:44.53 | jsolares | http://pastebin.ca/6820 |
| 01:44.53 | jsolares | that |
| 01:45.37 | jsolares | let me see the debug |
| 01:47.33 | pulu | it's pretty much the same as mine |
| 01:48.11 | jsolares | http://pastebin.ca/6821 |
| 01:48.14 | jsolares | that's my debug |
| 01:49.00 | jsolares | i'm gonna try changing callerid |
| 01:51.33 | pulu | have you tried a number besides that 800 #? |
| 01:51.38 | jsolares | yeah |
| 01:51.53 | jsolares | many, international as well |
| 01:52.00 | jsolares | my 800 did with nufone |
| 01:52.06 | jsolares | all busy according to voipjet |
| 01:53.03 | *** join/#asterisk Gronker (~Gronker2@adsl-220-79-68.ags.bellsouth.net) |
| 01:53.57 | pulu | the only thing I can suggest is trying to change the callerid name because it shows up with unclosed quotes but i doubt that would be the problem... sorry |
| 01:55.04 | jsolares | do you have a name in callerid? |
| 01:57.00 | *** join/#asterisk paulc (paulc@S010600062586a0b4.vc.shawcable.net) |
| 01:58.32 | pulu | i just looked and i don't, at least it doesn't come up i the debug |
| 02:00.11 | jsolares | that was it tho |
| 02:00.21 | pulu | but I know I do as it's in the cdr records on their website |
| 02:00.23 | jsolares | i used SetCIDName and SetCIDNum |
| 02:00.37 | jsolares | instead of SetCallerID, and it worked |
| 02:01.17 | jsolares | and no quotes in the debug |
| 02:01.41 | jsolares | -- Call accepted by 216.118.117.46 (format ulaw) |
| 02:01.44 | jsolares | score! |
| 02:02.46 | *** join/#asterisk Umaro (~umaro@209.140.74.64) |
| 02:02.52 | Umaro | Hey guys |
| 02:02.57 | jsolares | hey |
| 02:03.07 | Umaro | what's your average turnaround time for getting G729 licenses from digium? |
| 02:03.07 | tuxinator_linux2 | hey U |
| 02:03.20 | tuxinator_linux2 | 200 days |
| 02:03.22 | *** join/#asterisk Inv_arp (junya@adsl-3-247-135.mia.bellsouth.net) |
| 02:03.26 | tuxinator_linux2 | just kidding |
| 02:03.47 | tuxinator_linux2 | Hardware only took a few days, don't know about license |
| 02:04.01 | Umaro | well, I suppose in some sick way, it's my fault, but i'm still waiting on my 10 g729 licenses I ordered on the 14th of january. |
| 02:04.10 | Hmm-work | that's a long time |
| 02:04.14 | Umaro | got my email that they dinged my card, but nothing else yet. |
| 02:04.20 | tuxinator_linux2 | hmm |
| 02:04.23 | tuxinator_linux2 | call them? |
| 02:04.33 | Umaro | yeah, i'll have to do that tommorow.. past closing time now |
| 02:04.45 | tuxinator_linux2 | They must of overlooked it |
| 02:04.55 | tuxinator_linux2 | did you provide an email? |
| 02:05.00 | jsolares | yeah, i'd think the g729 licences to be rather instantaneous |
| 02:05.34 | tuxinator_linux2 | as do I |
| 02:07.48 | Umaro | yeah |
| 02:07.54 | Umaro | jsolares: oh really? |
| 02:08.04 | tuxinator_linux2 | off to get some worok done |
| 02:08.05 | Umaro | jsolares: you mean they should be, or they are usually? |
| 02:08.09 | tuxinator_linux2 | Umaro, give them a call |
| 02:08.11 | tuxinator_linux2 | or email htem |
| 02:08.11 | jsolares | should be |
| 02:08.13 | tuxinator_linux2 | email |
| 02:08.18 | Umaro | yeah, i'll do that tommorow morning |
| 02:08.31 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 02:08.45 | tuxinator_linux2 | Good luck |
| 02:09.16 | *** join/#asterisk TheEmperor (~mattn@203.121.47.100) |
| 02:10.24 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 02:10.51 | *** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net) |
| 02:12.13 | *** join/#asterisk RoyKa (~roy@83.80-203-29.nextgentel.com) |
| 02:12.38 | Chuji | 300 people in here and it's this quiet? |
| 02:12.43 | Mavvie | yes |
| 02:12.47 | Mavvie | all bored |
| 02:13.21 | |Vulture| | Anyone install Flash Operator Panel? |
| 02:13.45 | *** join/#asterisk dsmouse (~mouse@rrcs-24-199-146-243.midsouth.biz.rr.com) |
| 02:14.04 | *** join/#asterisk RoyK (~roy@83.80-203-29.nextgentel.com) |
| 02:14.20 | Chuji | |Vulture| sure |
| 02:14.31 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 02:15.07 | Juggie | vulture, have you touched queue/agents lately? |
| 02:16.41 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 02:16.49 | footnote | Chuji: all the needy people went to get drunk together |
| 02:17.04 | footnote | commiserate, etc |
| 02:18.23 | *** join/#asterisk zignig (~simon@203.217.15.10) |
| 02:19.12 | *** join/#asterisk |Vulture| (~Vulture@109.238.204.68.cfl.res.rr.com) |
| 02:19.20 | |Vulture| | sorry my inet went wack |
| 02:19.26 | |Vulture| | did you say anything Chuji |
| 02:19.33 | TheEmperor | hello all |
| 02:19.42 | Juggie | vulture, i asked have you touched the agent stuff lately |
| 02:20.03 | |Vulture| | Juggie: no I haven't been trying to get this stuff working |
| 02:20.20 | TheEmperor | can someone recommend a good security system for my * server? I am using Snort but someone managed to hack into my server :( |
| 02:20.38 | machinehd | everytime my fedora box boots up, kudzu removes the tdm card and re adds it. Any idea why it's doing that? |
| 02:20.39 | |Vulture| | I use APF |
| 02:20.50 | Chuji | TheEmperor : How about firewalling off all of your ports? |
| 02:20.59 | |Vulture| | lol |
| 02:21.06 | TheEmperor | Chuji: then how can calls come in and out? |
| 02:21.12 | *** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net) |
| 02:21.18 | Chuji | TheEmperor : NAT? |
| 02:21.35 | TheEmperor | Chuji:haven't tried that, maybe I should.. |
| 02:21.54 | Chuji | TheEmperor : Either way, at least firewall the non asterisk ports |
| 02:22.06 | Chuji | I doubt anyone hacked your sip or iax ports |
| 02:22.18 | RoyK | TheEmperor: what OS? |
| 02:22.20 | TheEmperor | Chuji: but sometimes I need to SSH into my own machine |
| 02:22.25 | RoyK | TheEmperor: what distro, if linux? |
| 02:22.29 | TheEmperor | RoyK: Redhat 9 |
| 02:22.37 | TheEmperor | RoyK: Kernel 2.4 |
| 02:22.41 | RoyK | mohaha |
| 02:22.43 | Juggie | |Vulture|, what are you working on? |
| 02:22.48 | Chuji | TheEmperor : dont run ssh on port 22 |
| 02:22.49 | RoyK | choose a better distro |
| 02:22.54 | TheEmperor | RoyK: which one is good? |
| 02:22.59 | RoyK | TheEmperor: debian |
| 02:23.01 | TheEmperor | Chuji: which port should I run on? |
| 02:23.01 | RoyK | TheEmperor: gentoo |
| 02:23.07 | RoyK | TheEmperor: SuSE, perhaps |
| 02:23.20 | RoyK | redhat sucks balls |
| 02:23.21 | TheEmperor | RoyK: I installed a new server with a SCSI drive and Debian can't find the harddrive.. |
| 02:23.29 | Chuji | TheEmperor : Something high, over 1200 |
| 02:23.29 | |Vulture| | Juggie: getting this Flash Operator Panel working.. it can't be this hard... I so suck |
| 02:23.42 | RoyK | TheEmperor: don't use woody. sarge'll find it |
| 02:23.45 | Juggie | the flash operator isnt that exciting :P |
| 02:23.46 | TheEmperor | Chuji: interesting. I never thought of that |
| 02:23.47 | RoyK | TheEmperor: and gentoo |
| 02:23.47 | Chuji | |Vulture| : It's pretty easy, what are you struggling with |
| 02:23.50 | TheEmperor | RoyK: sarge? |
| 02:23.59 | JunK-Y | 3.2 |
| 02:24.03 | RoyK | TheEmperor: google for it |
| 02:24.14 | Chuji | TheEmperor : Well, it will keep it from getting port scanned by script kiddies |
| 02:24.27 | TheEmperor | Chuji: yeah |
| 02:24.30 | |Vulture| | Chuji: I have no clue, its just showing all my SIP clients as being registered, etc. but I have tried opening channels, disconnecting them.. nothing changes |
| 02:24.41 | TheEmperor | the interesting bit is they went in and the first thing they looked at were my * files |
| 02:25.10 | TheEmperor | then the cu$ts edited the system so now it can't even boot up |
| 02:25.15 | Chuji | TheEmperor : how can you tell? .history file? |
| 02:25.40 | Chuji | TheEmperor : That's a pretty dumb ass hax0r that would leave a trail |
| 02:25.46 | dsmouse | TheEmperor: what's the boot symptom? init=/bin/sh work? |
| 02:25.46 | TheEmperor | Chuji: I managed to SSH into it and type histroy and saw they downloaded some stuff fro hackr.com.ro |
| 02:25.58 | Beirdo | got my x100p clone FINALLY hooked up |
| 02:25.58 | TheEmperor | dsmouse: no |
| 02:26.14 | TheEmperor | i might try and run knoppix to get my conf files and then reinstall with debian |
| 02:26.14 | Beirdo | finally remembered to go to Rat Shack and get the extension. |
| 02:26.25 | dsmouse | what was it? |
| 02:26.25 | TheEmperor | pain in the a$s |
| 02:26.29 | Chuji | |Vulture| : Is op_server complaining at all? |
| 02:26.41 | TheEmperor | dmouse: even on safe mode it doesn't boot |
| 02:26.48 | Chuji | |Vulture| : You can start it with some debug to see if it's getting the manager events right |
| 02:26.53 | TheEmperor | dmouse: they killed the boot section i think |
| 02:26.53 | dsmouse | TheEmperor: what os was it? |
| 02:27.02 | TheEmperor | dmouse: redhat 9.0 2.4 kernel |
| 02:27.14 | dsmouse | that's kinda old, yea |
| 02:27.35 | TheEmperor | dsmouse: yeah..i think i'll change it to debian prob knoppix with the 2.6 kernel |
| 02:28.05 | dsmouse | if you're used to rh, why not a free copy of the rhel or fedora? |
| 02:28.05 | |Vulture| | Chuji: yea i put it in full debug and it was seeing all event |
| 02:28.44 | Chuji | |Vulture| : You are calling the page with the fqdn that you entered in the config right? |
| 02:28.58 | TheEmperor | dsmouse: i kinda like debian based distros now... |
| 02:29.06 | TheEmperor | dsmouse: hate rpm dependencies |
| 02:29.26 | *** join/#asterisk odie_flocon (~chatzilla@S01060011953994ee.cg.shawcable.net) |
| 02:29.35 | TheEmperor | Chuji:apart from setting a high port for SSH and snort, anything else I should do? |
| 02:29.37 | odie_flocon | who's talking voice? |
| 02:30.14 | Chuji | TheEmperor : you might run portsentry |
| 02:30.21 | odie_flocon | who's talking voice on guest@66.250.68.194/996 |
| 02:30.23 | dsmouse | TheEmperor: how is rpm deps diffrent from app deps? |
| 02:30.59 | odie_flocon | rpm deps are app deps. |
| 02:31.04 | TheEmperor | dsmouse: dunno.. |
| 02:31.15 | |Vulture| | Chuji: in op_server.cfg? |
| 02:31.18 | odie_flocon | all an rpm is apps. |
| 02:31.32 | TheEmperor | Chuji: will portsentry conflict with snort? |
| 02:31.43 | odie_flocon | I love my ChatZilla |
| 02:31.57 | *** join/#asterisk ClayReiche123 (~creiche@60-189.35-65.tampabay.res.rr.com) |
| 02:32.15 | odie_flocon | who's talking voice on guest@66.250.68.194/996 ? |
| 02:32.23 | tzanger | odie_flocon: not sure |
| 02:32.28 | Chuji | TheEmperor : Nah, it shouldn't. Port sentry can listen on any port you choose (or multiple) and it will drop a route if you hit it |
| 02:32.28 | tzanger | I didn't realize lief was in the conf |
| 02:32.30 | odie_flocon | hmm ok. |
| 02:32.46 | odie_flocon | I can hear them typing 2. |
| 02:32.49 | odie_flocon | :D |
| 02:33.00 | dsmouse | mmm |
| 02:33.01 | dsmouse | food |
| 02:33.03 | TheEmperor | Chuji: ok..what distro do you recommend I install? |
| 02:33.04 | dsmouse | later yo |
| 02:33.14 | ClayReiche123 | Can I get some help configuring an Asterisk voicemail server? |
| 02:33.20 | Chuji | TheEmperor : I'm with the others. Use what you are comfortable with |
| 02:33.34 | Chuji | TheEmperor : If you like redhat, it will be fine |
| 02:33.34 | odie_flocon | What do you want Clay? |
| 02:33.37 | TheEmperor | Chuji: ok |
| 02:33.54 | Chuji | ~rtfw |
| 02:33.55 | jbot | rumour has it, rtfw is Read That F*cking Wiki, the one at http://www.voip-info.org/tiki-index.php |
| 02:33.55 | ClayReiche123 | I have several * machines and I want them all to send voicemail to a seperate * machine dedicated to vmail. |
| 02:33.59 | |Vulture| | Chuji: I am not sure what you meant by "calling the page with the fqdn" |
| 02:34.20 | odie_flocon | ok |
| 02:34.21 | ClayReiche123 | Do I use the Transfer app? |
| 02:34.27 | odie_flocon | no |
| 02:34.34 | *** part/#asterisk bwzb (~wlwzhang@ip67-153-142-73.z142-153-67.customer.algx.net) |
| 02:34.36 | odie_flocon | well I don't think so. |
| 02:34.47 | Chuji | ~fqdn |
| 02:34.48 | jbot | [fqdn] Fully Qualified Domain Name, or the entire DNS identifier for a host, 'foo.example.org' |
| 02:34.56 | odie_flocon | I would setup trunking |
| 02:35.03 | |Vulture| | Chuji: yea I did |
| 02:35.07 | ClayReiche123 | I've had no luck with the transfer app...so it's good to here. |
| 02:35.10 | odie_flocon | and then setup the trunk to go straight to vm. |
| 02:35.12 | |Vulture| | I will check it again though |
| 02:35.38 | odie_flocon | what monkey? |
| 02:35.58 | ClayReiche123 | ok...this will help a lot.... at least it's a place to start, I was stuck. Thank you! |
| 02:36.19 | odie_flocon | ok Clay |
| 02:36.33 | odie_flocon | In your dialplan |
| 02:36.36 | JunK-Y | my roomate is crazy, he's watching an old hockey game dated from 93! |
| 02:36.42 | *** join/#asterisk eipi (~eipi@100-172-114-200.fibertel.com.ar) |
| 02:36.47 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 02:36.51 | odie_flocon | hey any hockey is better then no hocky. |
| 02:37.04 | JunK-Y | he could watch junior hockey |
| 02:37.10 | *** join/#asterisk qwerp (~abc@219.95.105.74) |
| 02:37.11 | JunK-Y | with Sidney Crosby and stuff. |
| 02:37.16 | ariel_ | hello everyone |
| 02:37.16 | qwerp | harlo |
| 02:37.21 | JunK-Y | lo ariel_ |
| 02:37.22 | Chuji | Sup ariel_ |
| 02:37.22 | ClayReiche123 | I've never messed with trunking.... I'll read the wiki and hopefully not come back here. |
| 02:37.29 | ariel_ | syslod, did you get your system working? |
| 02:37.33 | ariel_ | hello JunK-Y |
| 02:37.34 | odie_flocon | what you want to do is instead of sending it to VM in your dial plan. |
| 02:37.38 | qwerp | some question here.. |
| 02:38.09 | odie_flocon | you want to setup an IAX trunk between all the boxes. |
| 02:38.51 | qwerp | assuming exten A and B is ringing, how can Exten C pickup call on exten A and Exten D pickup call on exten B? |
| 02:38.53 | odie_flocon | how does jess = yes? |
| 02:38.56 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 02:38.58 | ariel_ | trunking is the most miss understood part of asterisk. |
| 02:39.10 | ariel_ | but it is something that is really simple. |
| 02:39.12 | ClayReiche123 | great.... this should be fun then. |
| 02:39.32 | ClayReiche123 | The word has always scared me a little. |
| 02:39.49 | odie_flocon | well trunking once you understand it will be your greatest tool. :D |
| 02:39.51 | Juggie | ariel_, trunking is so easy tho. |
| 02:39.55 | ariel_ | ClayReiche123, what do you want to do? |
| 02:39.58 | Juggie | and so very powerful |
| 02:40.08 | *** join/#asterisk ionix (~ionix@MTL-HSE-ppp186145.qc.sympatico.ca) |
| 02:40.24 | ClayReiche123 | Central voicemail server. |
| 02:40.25 | ionix | Hey, I am in the process of submitting the idea to my boss to HOST a IAX peer/Sip peer asterisk box. |
| 02:40.30 | ionix | like Nufone |
| 02:40.39 | ionix | but we have killer prices ;) |
| 02:40.52 | Chuji | Trunking won't really help you with a central vm server |
| 02:41.02 | qwerp | assuming exten A and B is ringing, how can Exten C pickup call on exten A and Exten D pickup call on exten B? |
| 02:41.03 | Juggie | why woudnt it? |
| 02:41.14 | ionix | What is important ? Low international rates ? Low north america rates ? DID availability ? Toll Free ? |
| 02:41.19 | Juggie | you could have many boxes and host all the vm centrally |
| 02:41.22 | ariel_ | Qwell, calling groups |
| 02:41.27 | ClayReiche123 | That's the plan |
| 02:41.44 | Chuji | Juggie : Trunking just allows multiple iax channels to run off of one control channel. |
| 02:41.50 | qwerp | ariel: calling groups onli pickups a call in a group.. |
| 02:41.59 | ariel_ | you can send a any call to voicemail any where without trunking. |
| 02:42.01 | qwerp | how about if i want to specify an extension? |
| 02:42.02 | *** join/#asterisk SexyKen (~sexyken@c-67-161-5-149.client.comcast.net) |
| 02:42.14 | SexyKen | Hey -- I need a custom development team for Asterisk. |
| 02:42.21 | SexyKen | Any freelance developers in here? |
| 02:42.22 | Juggie | ariel_, how? |
| 02:42.34 | Chuji | Juggie : How does that overcome the fact that app_voicemail expects it to be on the same box? |
| 02:42.36 | ariel_ | Juggie, how what? |
| 02:42.59 | Juggie | Chuji, MWI isnt going to work no. |
| 02:43.07 | ariel_ | Chuji, no it does not you can have vm in another box you just send it to that context in the dialing macro. |
| 02:43.19 | Juggie | but voice mail can be elsewhere |
| 02:43.33 | |Vulture| | Chuji: that has to be it... its got to be that IP/Domain thing |
| 02:43.34 | ClayReiche123 | I'm ok with no MWI |
| 02:43.50 | SexyKen | Clay -- are you sure? MWI is a very useful feature. |
| 02:43.50 | Chuji | |Vulture| : Yeah, it's sensitive about that |
| 02:44.01 | ariel_ | Juggie, your almost correct. |
| 02:44.07 | ClayReiche123 | Carrier type environment |
| 02:44.32 | ClayReiche123 | It would be nice, but not critical. |
| 02:44.36 | Juggie | ariel_? :P almost |
| 02:44.50 | |Vulture| | Chuji: because all the debug shows its working, just not feeding it to the swf |
| 02:45.08 | Chuji | |Vulture| Is it working now? |
| 02:45.09 | ariel_ | there are ways like you can store the message in other server but still have it on your working servers dialing rules. |
| 02:45.33 | |Vulture| | Chuji: nope |
| 02:46.02 | ClayReiche123 | ariel: you mean like NFS? |
| 02:46.06 | Juggie | i guess if your voicemail box had all the sip phones registered even if it didnt directally handle them |
| 02:46.10 | Juggie | it could send mwi messages |
| 02:46.22 | ariel_ | ClayReiche123, well yes but in linux you can mount another systems drive. |
| 02:47.12 | Juggie | what about if your phones all register with SER |
| 02:47.15 | Juggie | how does MWI work then? |
| 02:47.17 | ClayReiche123 | I'm trying to get the voicemail processing off of the asterisk servers. |
| 02:47.56 | Chuji | |Vulture| Is it creating a variables.txt in your web directory? |
| 02:48.06 | *** join/#asterisk Luhiwu (~marsosa@200.63.89.243) |
| 02:48.53 | odie_flocon | who's trying to be Canadian. |
| 02:49.06 | tzanger | bkw |
| 02:49.11 | Juggie | ariel? |
| 02:49.25 | *** join/#asterisk trig_hm (~jb@home.monkeypr0n.org) |
| 02:49.35 | qwerp | <PROTECTED> |
| 02:49.50 | ClayReiche123 | So, I guess I bone up on trunking? |
| 02:49.50 | trig_hm | I just started watching friday night lights |
| 02:49.55 | trig_hm | sorry wrong window |
| 02:50.05 | |Vulture| | Chuji: yes |
| 02:50.11 | odie_flocon | that's only in ontario |
| 02:50.14 | qwerp | how can i do a Directed pickup? |
| 02:51.22 | *** join/#asterisk SexyKen (~sexyken@c-67-161-5-149.client.comcast.net) |
| 02:51.26 | SexyKen | Hey -- I need a custom development team for Asterisk. |
| 02:51.28 | SexyKen | Any freelance developers in here? |
| 02:51.35 | |Vulture| | Chuji: http://24.129.110.157/panel/ |
| 02:51.38 | Chuji | |Vulture| : And you are calling it with the same URL that is in web_hostname in op_server.cfg |
| 02:51.41 | |Vulture| | x201 is up |
| 02:52.34 | |Vulture| | there we go |
| 02:52.35 | |Vulture| | :) |
| 02:52.43 | |Vulture| | it was a firewall issue |
| 02:52.48 | |Vulture| | doh |
| 02:53.05 | Chuji | had to open the perl server port? |
| 02:53.19 | |Vulture| | Chuji: you got it... port for the flash connection |
| 02:53.24 | SexyKen | what's that Flash Operator Panel thing? |
| 02:53.39 | |Vulture| | its great stuff |
| 02:53.47 | ClayReiche123 | SexyKen: what do you need done. Just curious. |
| 02:53.52 | Chuji | ~fop |
| 02:53.53 | jbot | An XSL formatter written in Java that outputs PDF. URL: http://www.jtauber.com/fop/ |
| 02:54.06 | Chuji | ~asternic |
| 02:54.09 | Chuji | bleh |
| 02:54.15 | Chuji | stupid jbot |
| 02:54.22 | SexyKen | Clay -- Custom coding ;-) Need a login system setup for phones via Asterisk and Web |
| 02:54.29 | Chuji | ~google Asternic fop |
| 02:54.48 | Chuji | that first link |
| 02:55.38 | *** join/#asterisk amir (~amir@shield.guindehi.ch) |
| 02:55.41 | ClayReiche123 | sexyken: basically a front end for Asterisk? |
| 02:55.50 | SexyKen | Clay -- in a sense. |
| 02:56.08 | qwerp | how can i do a Directed pickup? |
| 02:56.13 | ClayReiche123 | rating engine? |
| 02:56.22 | *** join/#asterisk firestrm (~pcott@s206-116-63-85.bc.hsia.telus.net) |
| 02:56.28 | SexyKen | Clay eh? |
| 02:56.56 | ClayReiche123 | Do you need to rate calls or just configure? |
| 02:57.02 | firestrm | anyone here ever gotten asterisk or more specificly zaptel working on susi 9.2? Im having some very strange problems even compiling |
| 02:57.15 | SexyKen | Clay what do you mean rate clals? |
| 02:57.17 | SexyKen | *calls |
| 02:57.35 | SexyKen | I need full reporting for agents, queues, total calls, average calls based on time of day etc. |
| 02:58.11 | ClayReiche123 | sexyken:(that would be a no...) I thought maybe you would need to put $ to calls |
| 02:58.45 | SexyKen | What? |
| 02:58.56 | ClayReiche123 | nevermind |
| 02:59.56 | firestrm | no suse ppl here? bummer... |
| 03:00.26 | firestrm | guess everyone is away to von :( |
| 03:02.54 | Mavvie | <PROTECTED> |
| 03:02.54 | Mavvie | <PROTECTED> |
| 03:03.03 | Mavvie | not sure if I can wait that long :-) |
| 03:03.51 | modulus_ | who wants to see some dialplan perl logic? |
| 03:03.56 | JohnnyC | Anyone, if I call myself and awnser I dont hear music on hold |
| 03:04.16 | Juggie | if you put yourself on hold? |
| 03:06.52 | JohnnyC | yes |
| 03:07.05 | JohnnyC | Juggie: It worked a while ago |
| 03:07.06 | Juggie | in 1.0.6 its broken |
| 03:07.10 | JohnnyC | oh shit |
| 03:07.12 | JohnnyC | that it |
| 03:07.16 | JohnnyC | I installed 1.0.6 |
| 03:07.16 | Juggie | yah |
| 03:07.20 | JohnnyC | on top of 1.0.5 |
| 03:07.21 | Juggie | a fix for something else broke it |
| 03:07.26 | zignig | Juggie: I got that ldap lookup working ( sort of ) , more work this weekend python agi did the trick. |
| 03:07.31 | Juggie | if you know how to use svs |
| 03:07.38 | Juggie | er, cvs |
| 03:07.40 | JohnnyC | hmm Juggie whats the stable version ? |
| 03:07.42 | JohnnyC | 1.0.5 ? |
| 03:07.49 | Dr-Linux | hello |
| 03:07.52 | Juggie | 1.0.6 is stable but has that bug |
| 03:07.53 | JohnnyC | use CVS ? |
| 03:07.56 | JohnnyC | hehe |
| 03:07.58 | JohnnyC | and CVS ? |
| 03:08.01 | JohnnyC | better ? |
| 03:08.05 | Juggie | use cvs and get cvs-stable |
| 03:08.07 | Dr-Linux | anybody can help me regarding how to make conference call ? |
| 03:08.13 | Dr-Linux | using sip |
| 03:08.16 | ClayReiche123 | JohnnyC: you have mpg123? |
| 03:08.20 | JohnnyC | Dr-Linux: Im not there yet |
| 03:08.21 | Juggie | look on www.asterisk.org in download for instructions how |
| 03:08.23 | JohnnyC | yes ClayReiche123 |
| 03:08.25 | JunK-Y | run head. |
| 03:08.27 | JohnnyC | ok |
| 03:08.30 | Juggie | just compile that, and you'll be good |
| 03:08.42 | Juggie | be sure to specify v1-0 so you get stable |
| 03:08.57 | bjohnson | with -r |
| 03:09.05 | *** join/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net) |
| 03:09.22 | bjohnson | then to upgrade in future you go to /usr/src/asterisk and do make update; make upgrade |
| 03:09.29 | Juggie | zignig, thats cool, i went and found a ldap browser the day after out talk |
| 03:09.32 | bjohnson | and restart * to be up-to-date |
| 03:09.41 | Juggie | bjohnson, dont forget reloading modules |
| 03:09.45 | Juggie | for zaptel etc. |
| 03:09.50 | syslod | Anyone have IAX2 <> IAX2 working with CVS HEAD? |
| 03:10.15 | Juggie | i have IAX (head) <=> IAX (stable) working |
| 03:10.26 | syslod | Mind comparing files? |
| 03:10.27 | *** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
| 03:10.36 | JunK-Y | syslod: yes, but head of like 3 weeks ago. |
| 03:10.40 | Grooby | what did I miss? make update? |
| 03:10.42 | Grooby | make upgrade? |
| 03:10.49 | ManxPower | Well, I finally bought a USB Flash Drive |
| 03:11.08 | Umaro | ManxPower: sweet.. I want one but they are a little too expensive for a toy atm |
| 03:11.21 | Juggie | zignig, when i browsed our ldap i discovered the mitel unified messaging system was storing all the voicemail passwords clear text in the active directory |
| 03:11.23 | ManxPower | Umaro: $50 for 512MB |
| 03:11.24 | Umaro | ManxPower: what size and how much? |
| 03:11.24 | bjohnson | I've seen some good deals but can't figure out what I'd use one for |
| 03:11.26 | Juggie | how lame is that. |
| 03:11.35 | Umaro | ManxPower: nice... now to convince the wife I need it for work |
| 03:11.49 | Juggie | the sad thing is with that mitel vm pin, u can hear voice mail and it will also read your emails to you |
| 03:12.07 | Juggie | $50 is nothing.... if you paid in cash she shoudnt notice :P |
| 03:12.15 | Umaro | Juggie: lol |
| 03:12.37 | Umaro | Juggie: $50 doesn't go far in our family of 5 kids ;) |
| 03:13.10 | zignig | Juggie: eek! |
| 03:13.21 | Juggie | zignig, thats what i said ! :) |
| 03:13.30 | Juggie | i sent them off a email this afternoon, asking wtf they were thinking |
| 03:13.36 | Juggie | why is there no md5 even etc. |
| 03:13.41 | syslod | http://pastebin.ca/6827 |
| 03:13.48 | zignig | I store passwords in ldpa , but they are crypted , and they have a _really_ anal access list and tls access only. |
| 03:13.56 | JohnnyC | Juggie: downloading CVS |
| 03:14.25 | syslod | THats what I have and it won't accept the call. |
| 03:14.50 | syslod | I must have something wrong but I can seem to find it. |
| 03:14.52 | *** join/#asterisk IOscanner (~IOscanner@c-24-0-186-72.client.comcast.net) |
| 03:15.08 | Juggie | zignig, not these guys :) |
| 03:15.18 | Juggie | i'm tempted to drive to their offices and slap someone :P |
| 03:15.24 | Juggie | its only a 10min drive from my house to mitel hq |
| 03:15.28 | *** join/#asterisk santiago (~santiago@201.245.167.96) |
| 03:15.33 | bjohnson | just do a press release |
| 03:15.37 | bjohnson | wow |
| 03:15.39 | bjohnson | nasty |
| 03:15.42 | JohnnyC | Juggie: wich version of CVS should I pull ? |
| 03:15.48 | *** part/#asterisk ClayReiche123 (~creiche@60-189.35-65.tampabay.res.rr.com) |
| 03:16.09 | JohnnyC | the release version is 1.0.6 |
| 03:16.19 | JohnnyC | and has this bug OnHold |
| 03:16.22 | Juggie | justt v1-0 |
| 03:16.29 | Juggie | that will give u the latest stable |
| 03:16.36 | Juggie | which u can call pre 1.0.7 |
| 03:16.55 | JohnnyC | so the last stable is not equal to 1.0.6 |
| 03:16.57 | JohnnyC | ? |
| 03:17.04 | Juggie | no |
| 03:17.08 | Juggie | its 1.0.6+fixes |
| 03:17.14 | Juggie | so it will have the moh fix |
| 03:17.15 | Juggie | etc |
| 03:17.19 | Juggie | just download it |
| 03:17.21 | ManxPower | JohnnyC: No. It was at the time 1.0.6 was released, but there have been several fixes since then |
| 03:17.22 | JohnnyC | ok |
| 03:17.23 | Juggie | stop asking questions :) |
| 03:17.31 | Juggie | it will work we swear :) |
| 03:17.48 | JohnnyC | another question I have to pull the addons again and compile then right ? |
| 03:18.01 | Juggie | it coudnt hurt |
| 03:18.07 | Juggie | but likely they havnt changed much |
| 03:18.08 | Juggie | and it wont mater |
| 03:18.09 | JohnnyC | for each version of Asterisk I install I have to compile new addons package ? |
| 03:18.16 | JohnnyC | ok |
| 03:18.23 | JohnnyC | thanks |
| 03:18.24 | Juggie | you should yes |
| 03:19.02 | JohnnyC | www.offshore.pt/neve |
| 03:19.09 | JohnnyC | the last snow movie I made |
| 03:19.10 | JohnnyC | :) |
| 03:19.20 | JohnnyC | just giving something back ! :) |
| 03:19.45 | Juggie | damn u.... :) |
| 03:19.52 | Juggie | i broke my colar bone skiing a month ago |
| 03:19.56 | Juggie | i didnt want to see :) |
| 03:20.04 | Juggie | i am so pissed i cant ski anymore this year |
| 03:21.47 | zignig | ouch! |
| 03:21.58 | Dr-Linux | hello |
| 03:22.08 | Dr-Linux | hello |
| 03:22.11 | JohnnyC | Hello |
| 03:22.20 | Grooby | moshi moshi |
| 03:22.25 | Dr-Linux | plzz anybody can help me about Conference call in sip ? |
| 03:22.25 | JohnnyC | Skiing ! nice , we haven't much snow in Portugal |
| 03:22.32 | JohnnyC | just a small mountain |
| 03:22.34 | Dr-Linux | kya karoon ? |
| 03:22.46 | JohnnyC | 2000m high with a very small resort |
| 03:22.52 | JohnnyC | where u from Juggie ? |
| 03:23.19 | Dr-Linux | ManxPower: there ? |
| 03:23.34 | Juggie | originally newfoundland, canada but now ontario |
| 03:24.03 | zignig | Juggie: O canada ! |
| 03:24.04 | zignig | :) |
| 03:24.06 | Dr-Linux | no 1 knows about how to make confernece call ? |
| 03:24.08 | Juggie | 2000metres? |
| 03:24.13 | Juggie | vertical of 2000meters? |
| 03:24.14 | Dr-Linux | google also doesn't know :( |
| 03:25.28 | Dr-Linux | hello ??????????? |
| 03:25.36 | easydone | Dr-Linux: look for 'meetme' |
| 03:26.33 | Dr-Linux | easydone: where is meetme ? |
| 03:28.49 | Mavvie | RCFU |
| 03:29.00 | Mavvie | Dr-Linux: http://voip-info.org. Use the search box. |
| 03:31.21 | Dr-Linux | Mavvie: whats should i type in search box ? |
| 03:31.35 | Mavvie | Dr-Linux: listen to what easydone said. |
| 03:31.54 | Nugget | jesus christ, dr-linux, what are you going to ask next? do you need help locating your keyboard? |
| 03:31.57 | Dr-Linux | acha |
| 03:32.15 | Nugget | I'm beginning to think you have an honorary doctorate. |
| 03:32.27 | Dr-Linux | Nugget: |
| 03:32.55 | *** join/#asterisk loud (~ariel@null0.flapping.net) |
| 03:33.05 | Nugget | Welcome to #asterisk, the world's least efficient google proxy. |
| 03:33.27 | Grooby | roflmao |
| 03:34.10 | mikegrb | Nugget: 10 gold stars |
| 03:34.27 | *** join/#asterisk Legend` (~legend@24.244.142.133) |
| 03:34.52 | *** join/#asterisk marc32344 (~marc32344@69-28-224-214.dsl.teksavvy.com) |
| 03:35.06 | jontow | you guys.. we make a pretty decent google proxy |
| 03:35.06 | jontow | :) |
| 03:36.09 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 03:36.19 | ariel_ | hello everyone again. |
| 03:36.27 | syslod | ariel_: Hello |
| 03:36.43 | ariel_ | syslod, did you get your settings working? |
| 03:37.09 | syslod | No. No matter what I try it doesn't want to work. I found three other examples and they yield same result. |
| 03:38.26 | Juggie | whats the error |
| 03:38.47 | syslod | no authority found |
| 03:38.57 | ariel_ | syslod, what is your dialing rule for going between servers. |
| 03:39.05 | *** join/#asterisk zagaya972 (~d2s-compa@APointe-a-Pitre-102-1-13-25.w81-248.abo.wanadoo.fr) |
| 03:39.25 | Grooby | anyone here live in a concrete-built house? |
| 03:39.35 | ariel_ | I do |
| 03:40.12 | syslod | exten => _1XX,1,Dial(IAX2/pbx:test@192.168.0.25/${EXTEN}@C2008-3) |
| 03:40.24 | Grooby | would a 5.8 ghz phone work throught the walls? |
| 03:40.33 | qwerp | syslod: |
| 03:40.47 | qwerp | is there any way i can do a directed pickup? |
| 03:40.53 | ariel_ | wow what a dial string.... |
| 03:41.52 | yxa | Grooby what are freq in the THz used for? |
| 03:42.04 | Grooby | huh? |
| 03:42.17 | Grooby | my parents live in Taiwan and the buildings there are all concrete |
| 03:42.25 | Grooby | but they want me to buy them a phone from USA |
| 03:42.28 | Grooby | (don't ask me why) |
| 03:42.51 | Grooby | and I am afraid the higher the frequency, the harder it is to penetrate through those walls |
| 03:43.16 | ariel_ | exten => _1XX.,1,Dial,(IAX2/pbx:password@silvia/${EXTEN}) |
| 03:43.23 | Juggie | Grooby, i believe 5.8 gets beter penegration with less distance |
| 03:43.30 | Grooby | ahhh |
| 03:43.31 | Juggie | *better penetration |
| 03:43.32 | Grooby | excellent |
| 03:43.34 | ariel_ | syslod, I don't remember your [silvia] |
| 03:43.37 | Grooby | thanks guys |
| 03:43.56 | Grooby | (and now let's all get out head out of the gutter) |
| 03:44.05 | *** join/#asterisk drumkilla (~russell@12.21.241.80) |
| 03:44.05 | *** mode/#asterisk [+o drumkilla] by ChanServ |
| 03:44.13 | ariel_ | syslog hello are you still around? |
| 03:44.33 | syslod | Yup. I'll pastebin my current configs. |
| 03:45.38 | bjohnson | I think kigher frequency gets worse penetration |
| 03:46.05 | ariel_ | Grooby, yes I am in a concrete house and I can get 5.8 all over the house and even outside. |
| 03:46.24 | ariel_ | Grooby, I am in an hurricane zone. |
| 03:46.29 | Grooby | nigh frequency requires longer fore play? |
| 03:46.42 | Grooby | bjohnson, that's what I was thinking too |
| 03:46.43 | ariel_ | but the 2.4 works better. |
| 03:46.45 | Juggie | http://home.howstuffworks.com/cordless-telephone4.htm |
| 03:47.11 | Grooby | good one Juggie, and who said this is the worst google proxy ;) |
| 03:48.06 | syslod | ariel_: http://pastebin.ca/6829 |
| 03:48.18 | syslod | The error is no authority found |
| 03:48.35 | Grooby | i can tell you right now my parent's current phone is at the 43~50mhz range and it sucks |
| 03:50.18 | ariel_ | syslog why do you have two @ in your dial string? |
| 03:50.52 | bjohnson | that page ha ridiculous ranges listed |
| 03:51.04 | bjohnson | and doesn't talk about interference from walls at all |
| 03:51.14 | ariel_ | also if your using host=dynamic you need to have a registration line. But since your on the same network you should have just plain host=ipaddress |
| 03:53.37 | ariel_ | also if your calling the other machine you need to have a usertype=peer or friend. |
| 03:53.42 | syslod | ariel_: One is for server and the other for context |
| 03:54.00 | ariel_ | and on the machine your calling is where you have your usertype=user. |
| 03:55.01 | syslod | Just tried friend and user with no change. Also tried host= actual IP with no change. |
| 03:55.03 | ariel_ | syslog on both you have them as type=user |
| 03:55.12 | syslod | I tried with both friend also |
| 03:55.42 | *** join/#asterisk Othello (Othello@nusnet-154-210.dynip.nus.edu.sg) |
| 03:55.49 | Grooby | you tried it with both friends? are they female? |
| 03:55.56 | Grooby | (i really need to get my head out of the gutter) |
| 03:56.12 | syslod | I'll trade you a matching set of females for a fix :) |
| 03:56.28 | Grooby | lol |
| 03:57.03 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 03:57.06 | ariel_ | your dial string is: exten => _1XX,1,Dial(IAX2/pbx:test@192.168.0.25/${EXTEN}@C2008-3) |
| 03:57.07 | ariel_ | <PROTECTED> |
| 03:57.44 | syslod | Won't that go to wrong context? I want to goto extensions in context C2008-3 |
| 03:58.58 | ariel_ | your context is setup by the [pbx] setting in the iax.conf |
| 03:59.27 | syslod | Ok I need to add that line. |
| 03:59.33 | ariel_ | in fact if your only doing one way dialing you only need the settings on the server that gets the calls. |
| 04:00.10 | ariel_ | your dialing rules sends the username:password@ipaddress |
| 04:03.09 | *** join/#asterisk MikeJ[Jayden] (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
| 04:03.23 | syslod | ariel_: The extension@context does seem to be broken. I did what you said and removed from the dial while placing in iax.conf it works like a charm. Thanks |
| 04:03.42 | *** join/#asterisk Nukemizer (~Nuke@66.237.85.58.ptr.us.xo.net) |
| 04:03.55 | ariel_ | syslod, ok great |
| 04:04.32 | Juggie | @iax works here |
| 04:04.38 | ariel_ | in the great words of bkw_ next! |
| 04:04.43 | Juggie | i am running cvs head from a day or so ago |
| 04:04.52 | JunK-Y | isnt next!, its NEXT! :P |
| 04:04.53 | Juggie | and cvs stable from a day or so ago on thge other box |
| 04:04.58 | MikeJ[Jayden] | ~seen stevekstevek |
| 04:05.01 | jbot | stevekstevek is currently on #asterisk (10h 33m 40s) |
| 04:05.09 | Grooby | i am bored... |
| 04:05.09 | ariel_ | all you mean shout it out. |
| 04:05.34 | ariel_ | Grooby, I was under the impression your name was Grooby not bored? |
| 04:06.06 | Grooby | bored in chinese is grooby..umm..yeah..that's it.... |
| 04:06.07 | Grooby | :-D |
| 04:06.51 | ariel_ | I got to find my power brick for the laptop before it's batteries go dry. |
| 04:07.42 | Grooby | yup |
| 04:07.45 | Grooby | else it wouda suck |
| 04:08.06 | ariel_ | my icon says 1.17 hours left 45 % remaining.... |
| 04:08.17 | Grooby | what kinda laptop is it? |
| 04:08.41 | ariel_ | HP cheap pavilon ze4904us |
| 04:08.58 | Grooby | is that one of those desktop replacements? |
| 04:08.58 | ariel_ | ./pavilon/pavilion |
| 04:09.23 | ariel_ | no it's a 599 special office depot had. I needed one and I am very poor. |
| 04:09.35 | Grooby | lol |
| 04:10.57 | Grooby | (you just made me google for the laptop specs!!) |
| 04:11.38 | Grooby | it's a celery |
| 04:11.57 | ariel_ | it works has all I need for now. I do asterisk demo's on it. I have a usb external drive with centos and asterisk on it plug it in at the customer and works. |
| 04:12.14 | Grooby | ahhh awesome |
| 04:12.59 | Grooby | and i am running mine of my gf's old laptop |
| 04:13.00 | Grooby | hehe |
| 04:14.41 | Grooby | Initial-D! |
| 04:15.41 | tzanger | omg |
| 04:15.42 | tzanger | http://www.compfused.com/filedl/03cirrek/babygotbible.wmv |
| 04:15.45 | tzanger | toooo fucking funny |
| 04:17.10 | tzanger | my girl is hot and her bible's rockin'... she's got good doctrine... hahahahaha |
| 04:18.51 | MikeJ[Jayden] | wow... tzanger, you worry me |
| 04:19.40 | tzanger | MikeJ[Jayden]: :-) |
| 04:19.51 | tzanger | did you watch it? |
| 04:21.13 | MikeJ[Jayden] | part of it |
| 04:21.36 | *** join/#asterisk VirtualIceman (~VirtualIc@dsl-tmpl-66-18-203-37-cgy.nucleus.com) |
| 04:21.38 | tzanger | anyway with that and the rest of my beer gone, it's bedtime |
| 04:21.58 | MikeJ[Jayden] | hehe |
| 04:22.16 | MikeJ[Jayden] | quick, resolve somthing in mantis before you go to sleep :) |
| 04:23.08 | tzanger | sure, where'd that rm -rf command go |
| 04:23.15 | tzanger | or delete all from bugtable; |
| 04:24.05 | tzanger | eww |
| 04:24.08 | tzanger | make it an export at least |
| 04:24.33 | footnote | That's the good stuff... |
| 04:24.50 | tzanger | haha |
| 04:24.53 | slePP | anyone know of other services like simplewire.com or clickatell? |
| 04:24.54 | tzanger | no I mean molson export |
| 04:25.00 | MikeJ[Jayden] | tzanger likes the beast |
| 04:25.05 | footnote | I know a guy that got a Schlitz tattoo |
| 04:25.12 | footnote | (true story) |
| 04:25.19 | tzanger | the beast? |
| 04:25.29 | footnote | naw, just their logo |
| 04:26.06 | footnote | who's on first |
| 04:26.07 | *** join/#asterisk trig (~jb@xob.neospire.net) |
| 04:26.35 | MikeJ[Jayden] | the beast=milwakees best |
| 04:26.43 | tzanger | oh my |
| 04:26.53 | footnote | High class compared to schlitz |
| 04:26.56 | tzanger | I don't like much "standard" american beer |
| 04:27.00 | Grooby | lol |
| 04:27.06 | Grooby | just like a buddy of mine |
| 04:27.10 | Grooby | only drinks beast |
| 04:27.10 | tzanger | I am a fan of yeungling and some of hte microbrews I've tried |
| 04:27.20 | MikeJ[Jayden] | "standard" american beer = urine water |
| 04:27.26 | *** join/#asterisk atmel (~vlad@ruxi.dynamic.ucsd.edu) |
| 04:27.28 | footnote | I never met a micro I didn't like |
| 04:27.31 | tzanger | how's that go |
| 04:27.36 | slePP | when i was in cali last summer, i couldn't find any good beer at all |
| 04:27.42 | tzanger | americans drink cold, pissy tasting beer |
| 04:27.48 | tzanger | the English drink warm, beery tasting piss |
| 04:27.49 | Nugget | no we don't. |
| 04:27.52 | footnote | haha |
| 04:27.58 | Nugget | america hardly has a monopoly on shitty beer. |
| 04:28.04 | tzanger | Nugget: agreed |
| 04:28.04 | slePP | and Canadians drink stuff that is made on mountains |
| 04:28.09 | tzanger | there are shitty beers all over the world |
| 04:28.11 | slePP | or in fish hatcheries |
| 04:28.17 | footnote | But England has a monopoly on bad teeth! |
| 04:28.20 | tzanger | water? I never touch it. The fish fuck in it. |
| 04:28.45 | footnote | tzanger: children? ahh I love children! boiiiiled |
| 04:28.47 | ariel_ | tzanger, so funny. |
| 04:28.56 | tzanger | footnote: :-) |
| 04:29.01 | footnote | W C Fields was the bomb |
| 04:29.07 | tzanger | footnote: amen |
| 04:29.33 | slePP | has anyone used the SMS function of asterisk yet? |
| 04:29.40 | footnote | ariel_: I'm a grampa, she's safe |
| 04:29.42 | *** join/#asterisk ClayReiche123 (~creiche@60-189.35-65.tampabay.res.rr.com) |
| 04:29.51 | tzanger | heh |
| 04:29.51 | footnote | ariel_: It was a quote! |
| 04:29.59 | tzanger | I hopefully have a couple decades before I'm a grandpappy |
| 04:30.10 | Nugget | I hope I'm never a grandpappy. |
| 04:30.14 | footnote | tzanger: grandchildren are a lot more fun tho |
| 04:30.15 | ariel_ | footnote, I am a grampa too. |
| 04:30.20 | mikegrb | Nugget: but you are my grandpappy! |
| 04:30.25 | footnote | ariel_: ain't it a blast |
| 04:30.25 | Nugget | heh |
| 04:30.30 | mikegrb | IN SPIRIT! |
| 04:30.33 | *** part/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net) |
| 04:30.44 | ClayReiche123 | Can someone help me set up an IAX trunk? |
| 04:30.59 | tzanger | wow |
| 04:31.03 | MikeJ[Jayden] | sure clay |
| 04:31.04 | footnote | ClayReiche123: I think we're all drunk. |
| 04:31.05 | tzanger | I've got an 8,5 and 4 year old |
| 04:31.05 | ClayReiche123 | I keep getting Call rejected by xxx.xxx.xxx.xxx: No authority found |
| 04:31.15 | ariel_ | And there all girls in my house except me. |
| 04:31.17 | footnote | tzanger: great ages |
| 04:31.23 | Mavvie | I have a minus two weeks old. |
| 04:31.29 | tzanger | yup |
| 04:31.29 | Nugget | I'm not drunk, but I'm all hopped up on cold medicine, so that counts for something. |
| 04:31.39 | tzanger | yeah I gotta put some vicks under my nose tonight |
| 04:31.43 | tzanger | I *HATE* stuffy noses |
| 04:31.49 | slePP | ariel_: and i'm 23 :> |
| 04:31.56 | footnote | I just got over flu+cold+pneumonia |
| 04:32.04 | footnote | thought i was a goner this time |
| 04:32.23 | MikeJ[Jayden] | clay: in iax.conf on the receiving machine, set up a user |
| 04:32.28 | footnote | I blame the corporate environment. |
| 04:32.35 | MikeJ[Jayden] | or use guest |
| 04:32.44 | tzanger | night all |
| 04:32.54 | footnote | nite t |
| 04:32.57 | ClayReiche123 | MikeJ: ok I have that |
| 04:33.28 | MikeJ[Jayden] | what is in your dialplan on the other side |
| 04:33.58 | ariel_ | slePP, your young enough to be my kid. ARgh |
| 04:34.11 | MikeJ[Jayden] | dial(IAX2/user:password@host\${EXTEN}) |
| 04:34.45 | ClayReiche123 | MikeJ: exten => 8133435411,2,Dial(IAX2/poseidon:olympus@xxx.xxx.xxx.xxx/8133435411) |
| 04:36.28 | ClayReiche123 | should it be a backslash? |
| 04:36.35 | ariel_ | why make dialing rules hard... exten => 8133435411,2,Dial/iax2/poseidon:olympus@xxx.xxx.xxx.xx/${EXTEN} works if you have a setting for poseidon as user. in the iax.conf |
| 04:36.35 | MikeJ[Jayden] | can you pastebin the dialplan and iax.conf (or at least realavant sections) please |
| 04:36.38 | MikeJ[Jayden] | ~pastebin |
| 04:36.39 | jbot | [pastebin] a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 04:37.25 | MikeJ[Jayden] | from the remote box |
| 04:38.01 | ClayReiche123 | I just have 1 line in my client box |
| 04:38.20 | MikeJ[Jayden] | in ext or iax? |
| 04:38.42 | ClayReiche123 | in extensions.conf |
| 04:39.12 | *** join/#asterisk jaxxan (~jaxxan@202.70.125.109) |
| 04:39.19 | jaxxan | hello |
| 04:39.29 | MikeJ[Jayden] | ok, and the posiden id in iax on that box, points to the context that that line is in? |
| 04:39.30 | jaxxan | how do you tell what codec a certain call is using? |
| 04:39.30 | ClayReiche123 | this is my iax.conf entry on server http://pastebin.ca/6830 |
| 04:39.50 | ClayReiche123 | maybe not... let me see |
| 04:40.08 | MikeJ[Jayden] | add context=somthing to iax.conf on remote box |
| 04:40.29 | ClayReiche123 | and something should be context of the client side ext entry? |
| 04:40.40 | MikeJ[Jayden] | y |
| 04:40.56 | MikeJ[Jayden] | it needs to know where default ext go |
| 04:42.44 | Mavvie | anybody here with an idea how to get an animated progress bar in a webpage? |
| 04:43.09 | JunK-Y | Mavvie: flash? |
| 04:43.24 | jaxxan | is there a way to specify which codecs asterisk chooses first by default ? |
| 04:43.44 | MikeJ[Jayden] | as in codec pref? |
| 04:43.48 | jaxxan | yes |
| 04:43.58 | Mavvie | JunK-Y: yes, that's always a last option. |
| 04:44.11 | JunK-Y | a last? |
| 04:44.14 | Mavvie | I think I have something, I just open a new window with a clock in it while old window reloads. |
| 04:44.15 | *** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
| 04:44.38 | MikeJ[Jayden] | codec pref is in head,not stable, uses the order of codecs on the allow line, but will default to not do transcoding overriding those priorities |
| 04:45.04 | MikeJ[Jayden] | that will likely change again in head in the next day or so to offer a way to force codecs |
| 04:45.15 | MikeJ[Jayden] | that is not in stable |
| 04:45.16 | ClayReiche123 | MikeJ: still getting error. http://pastebin.ca/6831 |
| 04:45.45 | zignig | ariel_: what are you talking about it's one in the afternoon ;P |
| 04:46.58 | ariel_ | zignig, it's 11:46pm here. And besides if I don't get some sleep now my baby will wake up any way at 6am so I get it when I can. |
| 04:47.05 | VirtualIceman | what about Constantine? |
| 04:47.18 | VirtualIceman | clearchannel.com? |
| 04:47.28 | MikeJ[Jayden] | ummmm... what end is that iax.conf from? |
| 04:47.57 | MikeJ[Jayden] | and that is debug from sending end right |
| 04:48.14 | ariel_ | MikeJ[Jayden], and ClayReiche123 are you trying to call from one asterisk box to another? |
| 04:48.15 | ClayReiche123 | both are there |
| 04:48.25 | ClayReiche123 | sending end had most info |
| 04:48.29 | MikeJ[Jayden] | ym he is missing context= |
| 04:48.35 | VirtualIceman | clay are you still working on your trunking |
| 04:48.40 | ClayReiche123 | yes |
| 04:48.51 | ClayReiche123 | context= is there... |
| 04:48.59 | ClayReiche123 | just not in the pastebin |
| 04:49.21 | ClayReiche123 | iax.conf is from server side |
| 04:49.27 | VirtualIceman | not much |
| 04:49.29 | VirtualIceman | I can't talk |
| 04:49.38 | MikeJ[Jayden] | that is remote end, what is debug giving you on remote end, cuz that looks like that ext does not exist on remote end in the context it is going to |
| 04:49.46 | VirtualIceman | no voice in the meetme confrence room |
| 04:49.57 | VirtualIceman | no kidding |
| 04:50.01 | VirtualIceman | :D |
| 04:52.29 | *** join/#asterisk niZon (~ilt@S0106deadbeef6977.wp.shawcable.net) |
| 04:53.52 | ClayReiche123 | MikeJ:http://pastebin.ca/6832 |
| 04:54.10 | ClayReiche123 | MikeJ: http://pastebin.ca/6832 |
| 04:56.52 | ClayReiche123 | MikeJ: http://pastebin.ca/6833 (sorry) |
| 04:58.13 | MikeJ[Jayden] | duh... type user work? |
| 04:58.43 | *** join/#asterisk Faithful (~Faithful@202-6-145-116.ip.adam.com.au) |
| 04:59.19 | ClayReiche123 | Not following? |
| 04:59.22 | ClayReiche123 | Not following |
| 05:00.06 | MikeJ[Jayden] | peer and user backwards, if I am reading which config is which right |
| 05:00.07 | *** join/#asterisk frankyboy (~frank@ip68-100-24-82.dc.dc.cox.net) |
| 05:00.42 | frankyboy | i am compiling in fedora core 3 and i am missing the kerberos libraries, how do i get them? |
| 05:00.44 | ClayReiche123 | Client is being redirected to server |
| 05:00.45 | *** join/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net) |
| 05:00.50 | Faithful | Hey guys, I have copied G729 into modules and registered but when I make calls it still uses gsm even though I have changed that in iax.conf |
| 05:00.53 | ClayReiche123 | I'll try swapping |
| 05:01.37 | MikeJ[Jayden] | peer is what is used for outbound, user, is used for stuff connecting to you |
| 05:01.54 | ClayReiche123 | ahh, then you're right |
| 05:02.03 | MikeJ[Jayden] | so machine you are trying to connect to should have entry for user |
| 05:06.13 | ClayReiche123 | MikeJ: still no good |
| 05:06.41 | ClayReiche123 | MikeJ: maybe I'm missing something else in the iax.conf? |
| 05:06.41 | Trepalium | frankyboy: 'yum install krb5-devel' I believe. |
| 05:06.57 | Trepalium | Provided yum is installed. |
| 05:09.00 | MikeJ[Jayden] | try adding @context to your dial line for s+g's |
| 05:10.52 | *** join/#asterisk gus (~user@46.33.233.220.exetel.com.au) |
| 05:10.52 | ClayReiche123 | MikeJ: rejected |
| 05:10.56 | frankyboy | trepalium THANKS |
| 05:12.00 | gus | can i do E&M (4wire) on a digium TDM/FXS card, or does it actually require different hardware? |
| 05:12.01 | MikeJ[Jayden] | are you seeing the call on the remote side at all? |
| 05:12.58 | ClayReiche123 | MikeJ: yes it shows me this at the CLI>Mar 4 00:06:39 NOTICE[2752]: chan_iax2.c:5441 socket_read: Rejected connect attempt from xxx.xxx.xxx.xxx |
| 05:13.41 | MikeJ[Jayden] | that's it?... |
| 05:13.49 | *** join/#asterisk W1thdraw (~Withdraw@ip68-5-125-44.oc.oc.cox.net) |
| 05:13.58 | MikeJ[Jayden] | try starting asterisk -rvvvvv |
| 05:14.27 | *** join/#asterisk W1thdraw (~Withdraw@ip68-5-125-44.oc.oc.cox.net) |
| 05:15.06 | *** join/#asterisk W1thdraw (~Withdraw@ip68-5-125-44.oc.oc.cox.net) |
| 05:15.09 | MikeJ[Jayden] | silly question, as you are making these changes, have you done reload? |
| 05:15.18 | ClayReiche123 | MikeJ: That's it. |
| 05:15.27 | MikeJ[Jayden] | reload/ |
| 05:15.29 | MikeJ[Jayden] | ? |
| 05:15.33 | ClayReiche123 | MikeJ: yes I've been reloading |
| 05:15.45 | ClayReiche123 | restart gracefully |
| 05:15.59 | ClayReiche123 | both servers |
| 05:16.01 | MikeJ[Jayden] | so even with verbosity 5 you see nothing other than reject? |
| 05:16.15 | MikeJ[Jayden] | try 10 v's |
| 05:16.21 | MikeJ[Jayden] | on remote |
| 05:16.28 | ClayReiche123 | MikeJ: correct I'm sending debug to the console also. |
| 05:16.41 | ClayReiche123 | I use 5 v's |
| 05:16.58 | MikeJ[Jayden] | you are debugging IAX2 on remote box and seeing nothing/ |
| 05:17.03 | MikeJ[Jayden] | just the reject? |
| 05:17.22 | ClayReiche123 | MikeJ: correct |
| 05:17.30 | Trepalium | What the... I'm getting "Remote UNIX connection", "Remote UNIX connection disconnected" repeatedly. What could be causing this? |
| 05:17.37 | MikeJ[Jayden] | what do you have in logger.conf on that machine/ |
| 05:18.03 | ClayReiche123 | MikeJ: I know it's sending debug to the console cause I see other calls debug info occasionally |
| 05:18.06 | ManxPower | "iax2 debug" If that doesn't work, then rename /etc/asterisk/logger.conf to something else and do a "logger reload" at the Asterisk CLI. |
| 05:18.22 | MikeJ[Jayden] | hey Manx... |
| 05:18.44 | MikeJ[Jayden] | am I missing somthing here... this should not be a big deal. |
| 05:19.13 | ManxPower | gus, Digium does not support 4-wire Analog E&M as far as I know. |
| 05:20.09 | gus | ManxPower: is there any (asterisk-supported) hardware around that does? |
| 05:20.31 | MikeJ[Jayden] | gus: channel bank |
| 05:20.33 | mikegrb | you could hire a little man to sit inside the computer |
| 05:20.56 | gus | yep. i only want a single port though, so a channel bank is a little bit of overkill |
| 05:21.13 | gus | the little man idea sounds promising though.. |
| 05:21.13 | MikeJ[Jayden] | well.. you didn't say that ;) |
| 05:21.16 | ManxPower | gus, I don't know. You can check out voicetronix |
| 05:21.27 | zignig | ManxPower: is 4-wire analog the same as PSTN ( pots ) |
| 05:21.28 | zignig | ? |
| 05:21.53 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 05:21.54 | *** mode/#asterisk [+o denon] by ChanServ |
| 05:22.10 | MikeJ[Jayden] | there are external devices for phone systems for bypass that covert 4 wire E&M to standard pots... |
| 05:22.22 | MikeJ[Jayden] | I do not know any by name, but they are out there... |
| 05:22.59 | MikeJ[Jayden] | they would be multiple ports... |
| 05:23.04 | gus | yeah, i'm having lots of trouble searching for things since they're such generic telephony terms (and i don't really know much about it) |
| 05:24.18 | MikeJ[Jayden] | sorry, I have just spoken generally about them as I don't use any E&M, but they have the same thing for ground start and stuff to convert lines for failover in power outage for emergency lines |
| 05:25.10 | MikeJ[Jayden] | clay: any luck getting debug? |
| 05:25.54 | zignig | gus : http://www.dcbnet.com/datasheet/dv1dds.html |
| 05:26.16 | ClayReiche123 | MikeJ: no, It makes me think thatthere is nothing to debug. |
| 05:26.35 | ClayReiche123 | MikeJ: The rejection is simply a NOTICE. |
| 05:27.29 | MikeJ[Jayden] | but it is getting a request that it is rejecting, so you should see some debug... did you do manxpower's recomendation on logger.conf |
| 05:27.41 | ClayReiche123 | MikeJ: I see DEBUG events... just autodestroying calls... |
| 05:27.58 | MikeJ[Jayden] | iax debug |
| 05:28.00 | MikeJ[Jayden] | ? |
| 05:28.03 | ClayReiche123 | MikeJ: yes but I get nothing at the console |
| 05:28.20 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
| 05:28.41 | ClayReiche123 | MikeJ: would that go under [general] or [logfiles] |
| 05:28.50 | ClayReiche123 | in logger |
| 05:30.10 | *** join/#asterisk clive- (~pirch@myw-stp-66-18-82-134.sentechsa.net) |
| 05:30.32 | MikeJ[Jayden] | logfiles |
| 05:31.00 | MikeJ[Jayden] | console => notice,warning,error,debug,verbose |
| 05:32.20 | ClayReiche123 | MikeJ: I've had that set from the beginning. |
| 05:32.26 | MikeJ[Jayden] | hmmmm |
| 05:34.34 | ClayReiche123 | Well, tomorrow is a new day... I think I'll sleep on it. Thank you for your help, I'm sure I'm gonna kick myself when I figure it out. |
| 05:35.28 | MikeJ[Jayden] | clay... this is the kind of thing I do all day w/ no troubles |
| 05:35.29 | MikeJ[Jayden] | http://pastebin.ca/6835 |
| 05:36.02 | ClayReiche123 | MikeJ: The instructions on the wiki looked pretty straight forward too.... |
| 05:36.03 | MikeJ[Jayden] | where 528x is an extension on the remote switch in whatever context I include |
| 05:36.26 | MikeJ[Jayden] | and it really is pretty staight forward to set up.. should be 2 min |
| 05:36.51 | ClayReiche123 | I'll try it fresh tomorrow |
| 05:36.55 | MikeJ[Jayden] | I do those all day long btwn boxes |
| 05:37.03 | *** join/#asterisk Qorky (~goaway@dsl-202-72-146-104.wa.westnet.com.au) |
| 05:37.06 | ClayReiche123 | I'll look for you in here when I get it. |
| 05:37.31 | MikeJ[Jayden] | o, and if you do the full user\pass in dialplan, you dont need iax.conf entry on local, just a user onthe remote |
| 05:37.45 | ClayReiche123 | ...and I'm sure I'll have to fess up to something stupid. :) |
| 05:38.11 | MikeJ[Jayden] | config looks fine fromwhat I saw, but I was getting piecies |
| 05:39.14 | ClayReiche123 | Could it be something else in iax.conf? a global setting of sorts? |
| 05:39.29 | MikeJ[Jayden] | ummmmm |
| 05:39.56 | ClayReiche123 | something like work=no :) |
| 05:40.06 | MikeJ[Jayden] | it's somthing, the fact that you are getting reject on remote but no debugs tells me you have somthing screwing up the debugs at least |
| 05:40.29 | MikeJ[Jayden] | cuz if you are getting a reject, there is somthing that it is rejecting and you should see that in debug |
| 05:40.36 | ClayReiche123 | let me try it on another server. I have one here running HEAD |
| 05:41.33 | MikeJ[Jayden] | do you have a guest account in iax.conf on the remote box? |
| 05:42.11 | MikeJ[Jayden] | try just putting dial(IAX2/guest@remotedbox/exten) |
| 05:42.26 | MikeJ[Jayden] | see if that at least works. |
| 05:42.40 | MikeJ[Jayden] | just make sure guest has context=place where the ext are |
| 05:44.41 | *** join/#asterisk drkludge (~drkludge@ip68-231-34-38.ph.ph.cox.net) |
| 05:45.56 | Qorky | anyone know why i get an odd beep sound on my phone calls. I dont get it IPphone to IPphone, only when I'm going out over my x100p's. maybe every minuite or so (its random) i get a fairly high pitch beep sound that last for like 1 second. |
| 05:46.26 | ClayReiche123 | MikeJ: works perfect on my HEAD server! |
| 05:46.38 | MikeJ[Jayden] | what version on the other box? |
| 05:47.01 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:47.19 | Essobi | Mmm.. I've got to say... |
| 05:47.23 | Juggie | heh my fav simpsons is on |
| 05:47.26 | Juggie | stonecutters |
| 05:47.27 | Essobi | * TAPI is a damn nice try.. |
| 05:47.32 | ClayReiche123 | Stable from Jan 18th |
| 05:47.32 | Essobi | but it sucks. |
| 05:47.45 | Juggie | i have stable & head from march 1st |
| 05:47.49 | MikeJ[Jayden] | kram has returned!!! |
| 05:47.49 | Juggie | talking via iax |
| 05:47.55 | kram | more or less heh |
| 05:47.56 | Juggie | with context in the dial |
| 05:47.56 | Juggie | np |
| 05:47.59 | MikeJ[Jayden] | hehe |
| 05:48.05 | Essobi | But the real problem in * tapi is the manager interface is just bad. |
| 05:48.13 | MikeJ[Jayden] | figure out how to get an ata to accept a codec it wont yet ;) |
| 05:48.15 | Juggie | kram, what are you gonna talk about @ von canada |
| 05:48.32 | Essobi | mike I think it's just a cat walking on Krams KB. |
| 05:48.46 | Essobi | He appears and disappears quick. :) |
| 05:48.49 | Juggie | hah ;) |
| 05:48.57 | Juggie | hes speaking for 1/2 hr at von canada in toronto |
| 05:48.57 | ClayReiche123 | Thank you MikeJ! |
| 05:49.01 | MikeJ[Jayden] | a cat can type mor or less?... impressive cat |
| 05:49.01 | Juggie | i just got told today i was going |
| 05:49.22 | MikeJ[Jayden] | thank me you found issue on your box, or just thanks for the time? |
| 05:49.51 | Essobi | kram Hey.. does the zaptel interface to 4ess support sending the caller name.. I got a PBX I want to hand a PRI to and feed it a full but custom CLID/CNAM database.. |
| 05:49.52 | ClayReiche123 | Thanks for your time. I wish I would have tried that sooner... |
| 05:49.56 | Trepalium | Victorian Order of Nurses Canada? |
| 05:50.10 | Essobi | mehe |
| 05:50.12 | Juggie | www.pulver.com/canada2005 |
| 05:50.16 | zignig | Trepalium: mmmmm , nurses |
| 05:50.22 | Essobi | I'm a honorary member. |
| 05:50.29 | MikeJ[Jayden] | clay, upgrade to 1.0.6 anyway, many bug fixes |
| 05:50.36 | Essobi | Pun intended. |
| 05:50.46 | Juggie | dont upgrade to 1.0.6 |
| 05:50.54 | MikeJ[Jayden] | don't |
| 05:50.56 | Juggie | stick with 1.0.5 or go to cvs-stable |
| 05:50.56 | MikeJ[Jayden] | ? |
| 05:51.01 | Essobi | Ehh? No likey, likey? |
| 05:51.03 | Juggie | moh is broken in 1.0.6 |
| 05:51.07 | Essobi | Oh. |
| 05:51.11 | Essobi | No shit? |
| 05:51.11 | MikeJ[Jayden] | really... missed that |
| 05:51.14 | Essobi | Damnit. |
| 05:51.19 | Essobi | Mehe. |
| 05:51.27 | Juggie | there is a small patch u can apply to your 1.0.6 |
| 05:51.32 | Juggie | if u dont want to download the cvs |
| 05:51.46 | MikeJ[Jayden] | clay, upgrade to current cvs-stable, i hear there is a MOH issue in 1.0.6 :) |
| 05:52.44 | Essobi | I always use -head lately. |
| 05:52.55 | Essobi | I like bleeeding edge, but it's going to bite me. |
| 05:53.23 | Essobi | Damnit. I need a way to push URLs to desktop on-ring that is scalable. |
| 05:53.23 | Juggie | hah |
| 05:53.34 | Goshen | I am trying to get Nikotel incoming to work, but I am getting an error, I posted my config files and the sip debug to http://pastebin.ca/6836 if someone could have a look |
| 05:53.36 | Essobi | The manager interface doesn't scale. |
| 05:53.51 | Essobi | What's the error? |
| 05:53.52 | Essobi | :) |
| 05:53.54 | MikeJ[Jayden] | wtf is nikotel? |
| 05:54.16 | Juggie | i run two, semi-production is cvs-stable and secondary is cvs-head |
| 05:54.17 | Trepalium | Explains why there's so many manager interface proxies. |
| 05:54.25 | Juggie | i'm guessing a sip->iax provider |
| 05:54.27 | mikegrb | Essobi: write a proxy to the manager |
| 05:54.30 | Juggie | er, sip or iax. |
| 05:54.32 | mikegrb | Essobi: like asternic uses |
| 05:54.44 | mikegrb | oh. |
| 05:54.45 | Goshen | SIP/2.0 407 Proxy Authentication Required |
| 05:54.48 | mikegrb | Trepalium said that |
| 05:54.56 | Essobi | Yup. |
| 05:55.01 | Goshen | nikotel is a voip provider |
| 05:55.07 | Essobi | mike I was thinking that. |
| 05:55.10 | MikeJ[Jayden] | O |
| 05:55.11 | Essobi | but I still need a client. |
| 05:55.12 | Essobi | :| |
| 05:55.25 | Juggie | Goshen, try logging in ;) it needs a username/pass |
| 05:55.34 | Juggie | see the wiki for instructions |
| 05:55.40 | *** join/#asterisk atmel (~vlad@ruxi.dynamic.ucsd.edu) |
| 05:55.43 | Goshen | I left out the general section... |
| 05:55.43 | Essobi | First of all. |
| 05:55.48 | Essobi | That's an inbound call right? |
| 05:56.13 | Goshen | register => goshen:Nikotelpass@calamar0.nikotel.com |
| 05:56.15 | Goshen | inbound call yes |
| 05:56.18 | Goshen | outbound works great |
| 05:56.24 | Essobi | Okay.. |
| 05:56.30 | Essobi | Here's what's happeneing. |
| 05:56.32 | MikeJ[Jayden] | peer=outbound, user=inbound |
| 05:56.34 | Essobi | it's sending a call to you |
| 05:56.39 | Essobi | and landing on the peer |
| 05:56.42 | Essobi | which you have a password on |
| 05:56.48 | Essobi | but it isn't transmiting one to you. |
| 05:57.00 | Essobi | setup a user in addition to the peer |
| 05:57.07 | Essobi | the inbound will land on user |
| 05:57.16 | Essobi | outbound should land on peer |
| 05:57.28 | Essobi | I just went through all this with a cisco router |
| 05:57.32 | Goshen | another context with a different name? |
| 05:57.46 | Juggie | no, a sip.conf client |
| 05:58.08 | Goshen | different name then |
| 05:58.14 | Juggie | you add nikotel twice, once as a peer and once as a user |
| 05:58.19 | Goshen | got it |
| 05:58.26 | Essobi | one peer |
| 05:58.29 | Essobi | one user |
| 05:58.34 | Essobi | don't use friend |
| 05:58.38 | Essobi | they are evil |
| 05:58.48 | Essobi | you only need a password on the peer |
| 05:58.54 | Essobi | user you can use an IP |
| 05:58.56 | Essobi | no password |
| 05:59.03 | Essobi | cause they ain't using a password to call you |
| 05:59.11 | Goshen | do I take the context off the peer? |
| 05:59.17 | *** join/#asterisk Tarox (someone@pD9E7AE33.dip.t-dialin.net) |
| 05:59.20 | Essobi | yup |
| 05:59.25 | Essobi | you got it sparky |
| 05:59.32 | Essobi | it's not an inbound anymore |
| 06:00.03 | Essobi | now.. if you were not performing auth in either direction.. just IP based.. you can get away with just a peer |
| 06:00.07 | Essobi | but in your case.. you can't. |
| 06:00.53 | Goshen | it is still hitting the peer |
| 06:01.03 | Goshen | perhaps I should move the user above the peer in the config? |
| 06:01.17 | MikeJ[Jayden] | reload ? |
| 06:01.22 | Goshen | yes, did reload |
| 06:01.54 | *** part/#asterisk ClayReiche123 (~creiche@60-189.35-65.tampabay.res.rr.com) |
| 06:02.53 | Goshen | Found peer 'NikotelAccount' |
| 06:04.12 | *** part/#asterisk gus (~user@46.33.233.220.exetel.com.au) |
| 06:07.37 | Goshen | I commented out the outbound user for testing, and now get this... |
| 06:07.37 | Goshen | Looking for goshen in default |
| 06:07.37 | Goshen | Mar 3 23:05:07 NOTICE[8638]: pbx.c:1329 pbx_extension_helper: Cannot find extension context 'default' |
| 06:07.53 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 06:07.53 | *** mode/#asterisk [+o denon] by ChanServ |
| 06:07.54 | Goshen | I don't see why it isn't going into the context I made for it |
| 06:08.04 | MikeJ[Jayden] | do you have an ext defalt? |
| 06:08.19 | MikeJ[Jayden] | a context defualt that is |
| 06:08.50 | Goshen | no |
| 06:08.55 | MikeJ[Jayden] | and outbound is peer, ibound is user? |
| 06:09.11 | MikeJ[Jayden] | so you commented out the peer? and do you have a context=line in the user |
| 06:09.16 | Goshen | I commented out the outbound peer because it was taking out the call yes |
| 06:09.35 | Goshen | context=inbound-nikotel |
| 06:09.46 | MikeJ[Jayden] | in the user |
| 06:09.51 | Goshen | yes |
| 06:09.54 | Goshen | [inbound-nikotel] |
| 06:09.54 | Goshen | exten => goshen,1,Dial(SIP/21,60,Ttm) |
| 06:09.54 | Goshen | exten => goshen,2,Voicemail2(u12) |
| 06:09.54 | Goshen | exten => goshen,3,HangUp |
| 06:10.11 | MikeJ[Jayden] | is - allowed? |
| 06:10.14 | MikeJ[Jayden] | I dunno |
| 06:10.14 | Goshen | in extensions.conf |
| 06:10.29 | Goshen | - works on my incoming iax stuff |
| 06:10.35 | MikeJ[Jayden] | k |
| 06:10.43 | Goshen | I can try it without |
| 06:10.59 | mikegrb | no difference |
| 06:11.14 | MikeJ[Jayden] | if it is saying cannot find context default then it thinks that you do not have a context= in the user it is matching |
| 06:12.57 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 06:16.25 | MikeJ[Jayden] | sleepy time |
| 06:16.40 | *** join/#asterisk sloPP (slepp@proso.netmonks.ca) |
| 06:16.55 | sloPP | someone want to help me with a disappearing channel bank problem? |
| 06:17.34 | Goshen | I take off the secret on the peer and it goes to default again |
| 06:17.40 | Goshen | guess I am going to have to make a default context |
| 06:18.27 | Goshen | well thats works! shesh |
| 06:18.43 | Goshen | I made a default extension and it works and my sip phone rings |
| 06:20.50 | zignig | Goshen: yay! |
| 06:20.52 | Goshen | however I put the user and peer accounts back, and it selects the peer account first, which requires a password |
| 06:21.21 | Goshen | not sure how to make it prefer the user account over the peer account |
| 06:24.20 | Goshen | Here is my current config now http://pastebin.ca/6839 |
| 06:24.53 | Goshen | the problem being...incoming sip doesn't provide a password and asterisk selects the peer over the user |
| 06:25.05 | Goshen | peer is used for outbound |
| 06:26.43 | Beirdo | Hmmm |
| 06:27.07 | Juggie | goshen, can you name the outbound peer someting else |
| 06:27.10 | Juggie | so it doesnt pickup on it |
| 06:27.20 | Goshen | what makes it pick up on it? the host? |
| 06:27.22 | Beirdo | anyone here have any idea how to tell an x100p card NOT to answer both lines if both are wired in? |
| 06:27.44 | Beirdo | i.e. a second line on the other two pair |
| 06:27.58 | Juggie | beirdo, the card likely doesnt support that configuration |
| 06:28.04 | Beirdo | if I call it, it answers both pairs simultaneously |
| 06:28.14 | Goshen | Beirdo: get a line splitter...$5 at wal-mart |
| 06:28.30 | Beirdo | I could... |
| 06:28.43 | Beirdo | or I could just rewire the jack |
| 06:28.52 | Juggie | it only supports one line per jack i guess |
| 06:28.54 | Goshen | How does Asterisk match incoming calls to a sip user? the host? |
| 06:29.01 | Beirdo | but why does the damn card answer both and bridge them? |
| 06:29.14 | Beirdo | :) |
| 06:29.27 | Juggie | its not designed for two lines in the same cat3 i guess |
| 06:29.45 | Beirdo | well, that's a SHITTY design if so |
| 06:30.13 | Beirdo | extremely shitty |
| 06:31.29 | Juggie | its a one line card |
| 06:31.31 | Juggie | what do u expect |
| 06:31.40 | Beirdo | I could just dig up a crappy piece of phone cord with just one pair used |
| 06:31.48 | Goshen | I have an idea...I will add a fromdomain= |
| 06:31.59 | Beirdo | I expect it to not even electrically connect to the second pair, that's what I expect |
| 06:32.23 | *** join/#asterisk search_learn2005 (~Miranda@adsl-68-127-104-255.dsl.pltn13.pacbell.net) |
| 06:33.30 | search_learn2005 | any suggestions for a good voip phone with good echo cancellation and IAX support |
| 06:35.20 | Goshen | fromdomain= didn't help |
| 06:35.32 | Juggie | goshen, try renameing your peer |
| 06:35.37 | Juggie | whats in the [....] |
| 06:36.46 | Juggie | you can try fromuser |
| 06:36.51 | Goshen | nope, it still went there |
| 06:36.52 | Goshen | Found peer 'Nikoteloutbound' |
| 06:37.08 | search_learn2005 | ANY suggestions for a good voip phone with good ECHO cancellation and IAX support ? |
| 06:37.27 | Goshen | search: IAX voip phones are so new, I don't think many people have them yet |
| 06:37.43 | Qwell | tip #304: restating your question with randomly capitalized words, will increase your chances of an answer. |
| 06:37.51 | Juggie | Goshen, does the incomming call have a sip username? |
| 06:38.03 | Juggie | if you turn on sip debug whats the From: |
| 06:38.19 | search_learn2005 | Goshen: so, would you suggest SIP phones? |
| 06:38.47 | Goshen | From: <sip:18017123381@66.52.2.22>;tag=FDF57E98-1DFE |
| 06:39.01 | Juggie | try |
| 06:39.05 | Goshen | INVITE sip:goshen@67.172.238.57 SIP/2.0 |
| 06:39.15 | Juggie | fromuser=18017123381 |
| 06:39.20 | Juggie | provided that number remains the same |
| 06:39.21 | Goshen | ok |
| 06:39.24 | Goshen | it does |
| 06:39.28 | Juggie | k |
| 06:39.47 | Juggie | i am looking at sip.conf example from head tho |
| 06:39.53 | Juggie | so i dont know if fromuser exists in stable |
| 06:40.25 | Goshen | nope, still going into peer |
| 06:40.31 | Juggie | ah |
| 06:40.31 | Goshen | I am on 1.0.6 |
| 06:40.32 | Juggie | from user |
| 06:40.36 | Juggie | and fromdomain |
| 06:40.46 | Juggie | only take affect when a call is placed |
| 06:40.48 | Juggie | not received |
| 06:40.51 | Goshen | ahh |
| 06:41.06 | Juggie | let me look here for u |
| 06:41.08 | Juggie | hold up |
| 06:41.17 | Goshen | I guess Asterisk is just set to prefer peer over user |
| 06:41.33 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
| 06:41.50 | Goshen | I am still trying to figure out what asterisk uses to match what sip user to use |
| 06:42.39 | Juggie | you know, i dont think you really need the peer |
| 06:42.49 | Juggie | with peer disabled inbound calls work correct? |
| 06:43.05 | Goshen | yes |
| 06:43.12 | Goshen | I will try without it |
| 06:43.19 | Juggie | double check |
| 06:43.26 | Juggie | and then we'll go from there |
| 06:44.19 | Juggie | yeah you only need the one section |
| 06:45.29 | *** part/#asterisk search_learn2005 (~Miranda@adsl-68-127-104-255.dsl.pltn13.pacbell.net) |
| 06:45.31 | Juggie | goshen? |
| 06:45.55 | Goshen | yes? |
| 06:46.00 | Juggie | did u remove peer? |
| 06:46.05 | Goshen | I can't seem to place outbound calls without peer |
| 06:46.10 | Juggie | ok |
| 06:46.17 | Juggie | whats the error |
| 06:46.19 | Juggie | when u try to |
| 06:46.26 | Goshen | I get 404 |
| 06:46.43 | Juggie | whats your dialstring |
| 06:47.12 | Beirdo | bjohnson: you in? |
| 06:47.20 | Goshen | exten => _1NXXNXXXXXX,3,Dial(SIP/${EXTEN}@Nikotelout,60,Ttr) |
| 06:47.34 | Goshen | in sip.conf |
| 06:47.36 | Goshen | [Nikotelout] |
| 06:47.45 | Juggie | try this |
| 06:47.48 | Goshen | secret=Nikotelpass |
| 06:47.55 | Goshen | username=goshen |
| 06:47.55 | Goshen | fromuser=goshen |
| 06:48.01 | Juggie | no pasteing :) |
| 06:48.01 | Goshen | host=calamar0.nikotel.com |
| 06:48.01 | Goshen | canreinvite=yes |
| 06:48.04 | Juggie | you'll get in trouble |
| 06:48.04 | Goshen | ok |
| 06:48.24 | Juggie | try this |
| 06:48.29 | Juggie | in yoru nikotelout |
| 06:48.33 | Juggie | set it to friend |
| 06:48.34 | Beirdo | Juggie: you may be interested too... |
| 06:48.36 | Juggie | instead of user |
| 06:48.41 | Juggie | and then add |
| 06:48.45 | Beirdo | http://lists.syonex.com/mailman/listinfo/asterisk-canada |
| 06:48.50 | Juggie | insecure=very |
| 06:49.01 | Juggie | i think that allows no pass for incomming |
| 06:49.29 | Goshen | you mean in my Nikotel incoming sip user |
| 06:49.33 | Goshen | not the Nikotelout |
| 06:49.37 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 06:49.49 | Juggie | hmm... actually i dont know if that will do it... |
| 06:49.55 | Goshen | or just Nikotelout with the pass |
| 06:50.02 | Juggie | give it a shot |
| 06:50.04 | Juggie | no leave the pass |
| 06:50.08 | Juggie | set it to friend |
| 06:50.09 | Juggie | instead of user |
| 06:50.14 | Juggie | and add |
| 06:50.17 | Juggie | insecure=very |
| 06:50.29 | Goshen | trying |
| 06:50.35 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 06:51.16 | Goshen | inbound works now :) |
| 06:51.39 | Goshen | and outbound gives 404 |
| 06:52.02 | Juggie | go to www.pastebin.ca |
| 06:52.10 | Juggie | and paste in the asterisk console output |
| 06:52.21 | Goshen | ok, let me comment out this user and try again first |
| 06:52.54 | Juggie | remove the fromuser fromdomain stuff |
| 06:53.03 | Zeeek | Good ${arbitrary_named_solar_cycle} |
| 06:53.22 | Zeeek | you guys are up (early|late) |
| 06:53.54 | zignig | Zeeek: we are; good eclipse to you too. |
| 06:54.12 | Zeeek | ziggie long time no see! in fact, never :) |
| 06:54.21 | Juggie | goshen |
| 06:54.24 | Juggie | i will msg u |
| 06:54.50 | zignig | Zeeek: always polite to a fellow Z'ster |
| 06:54.59 | Zeeek | I think so too |
| 06:55.11 | Zeeek | so I'm going to look into SMS again |
| 06:55.27 | Zeeek | since the staff is out of office, the tech staff can play |
| 06:55.44 | zignig | Zeeek: cool , do you asterisk at work ? |
| 06:55.58 | Zeeek | we use asterisk as our pbx - small office |
| 06:56.21 | Zeeek | it was a hard sell to my partner, who is also my life partner |
| 06:56.45 | Zeeek | my only other pbx experience was with a "home pbx" we rented from thephone co |
| 06:57.02 | Zeeek | two ISDN lines and two phones |
| 06:57.35 | Zeeek | we've evolved now to two analog lines, 10 voIP providers and three FXS-connected phones plus SIP and IAX remote phones |
| 06:58.06 | zignig | Zeeek: shweet, what distro are you running on |
| 06:58.07 | zignig | ? |
| 06:58.13 | Zeeek | Slackware 9.1 |
| 06:58.21 | zignig | ah , slackmeister ....... |
| 06:58.26 | Zeeek | and I don't change around because this is after all "production" |
| 06:58.28 | zignig | I'm a debian man myself. |
| 06:58.47 | Zeeek | I know little about linux distros - come from FreeBSD |
| 06:59.01 | zignig | 10 voip providers ? , doesn't that cost a lot to maintain ? |
| 06:59.09 | Zeeek | and speaking of OS I need to reboot this box |
| 06:59.20 | Zeeek | no they are all pay as you go except one |
| 06:59.37 | Zeeek | and that one is 8eu/month for unlimited local and national |
| 06:59.53 | zignig | Zeeek: nice , most of the voip providers here (.au) are contract. |
| 07:00.30 | Zeeek | for international I use nufone, voicepulse, ich, voiptalk, voipjet and a bunch of free ones like fwd, sipgate... |
| 07:00.40 | Beirdo | why so many? |
| 07:00.50 | Beirdo | and how do ya do your LCR? |
| 07:00.51 | Zeeek | testing them, constantly testing them |
| 07:00.58 | Zeeek | no need for lcr |
| 07:00.59 | zignig | Zeeek: does it save much money compared to PSTN service ? |
| 07:01.14 | *** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net) |
| 07:01.15 | Zeeek | LD is mostly personal to friends and family so I do it manually. |
| 07:01.28 | Beirdo | gotcha |
| 07:01.32 | Zeeek | the international stuff is 1.5-2.5c/min |
| 07:01.37 | Zeeek | no monthly |
| 07:01.43 | Zeeek | three of the above do IAX |
| 07:02.20 | Zeeek | incidentally I listened to the conference last night for 90 minutes straight - I guess that was nufoine hosted? (not sure) |
| 07:02.34 | Zeeek | anyone know who hosted the conference? |
| 07:03.53 | Juggie | it was hosted via iax.... i dont know if they officially provided dial in numbers |
| 07:04.12 | Zeeek | sure look abovein the mod |
| 07:04.16 | clive- | does anyone know which modules in modules.conf would make my asterisk look for a zap card...I get tons of errors saying "no ISA tormenta card found" |
| 07:04.19 | Zeeek | the number is right there |
| 07:04.37 | Zeeek | anyway, the quality of that (one way) call was excellent for 90 minutes straight |
| 07:04.48 | *** join/#asterisk Red_6 (~alex@m174.net81-66-29.noos.fr) |
| 07:04.54 | Juggie | it went on for like 8 hours ;) |
| 07:05.03 | Zeeek | you didn't listen? |
| 07:05.15 | *** join/#asterisk jaxxan (~jaxxan@202.70.125.109) |
| 07:05.22 | jaxxan | hey ya'll |
| 07:05.55 | Zeeek | that ip belongs to cogentco - does that ring a bell? |
| 07:06.08 | Zeeek | oh well, gotta reboot bbl |
| 07:06.21 | *** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 07:06.53 | *** part/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 07:09.04 | PTG123 | is thee any way to see what codec a particular channel is using between a sip device |
| 07:09.17 | *** join/#asterisk peted20 (~chatzilla@d2-55.rb.gh.centurytel.net) |
| 07:10.23 | clive- | PTG show channels maybe |
| 07:10.36 | PTG123 | doesn't say the codec |
| 07:10.57 | Juggie | http://www.unlimitel.ca/voiptermination.html |
| 07:11.00 | Juggie | wow, damn cheap :) |
| 07:11.18 | PTG123 | yah thats not bad |
| 07:11.21 | PTG123 | who i sthat comapny? |
| 07:11.24 | PTG123 | is that candian prices? |
| 07:11.35 | Juggie | yes.... |
| 07:11.44 | Juggie | it was some users in a * group i just signed up for |
| 07:11.47 | Juggie | in my city |
| 07:11.53 | Juggie | one of the people in the group owns it |
| 07:12.15 | pashah | morning |
| 07:12.26 | jaxxan | If i'm using: TRUNKGROUP=Zap/G1 ... and wanted calls to go out my eth0 interface as SIP, what would TRUNKGROUP equal? |
| 07:12.53 | Inv_arp | ok want to setup normal call conferencing , where does on start? |
| 07:13.29 | Inv_arp | jaxxan: what provider u using? |
| 07:13.35 | Inv_arp | for sip outbound? |
| 07:14.06 | jaxxan | myself, pointing to a AS5350 |
| 07:14.48 | Juggie | jaxxan, it would be SIP/ |
| 07:14.52 | brc_ | PTG123, sip show channels |
| 07:14.58 | brc_ | format |
| 07:15.06 | Juggie | but its just as easy to put Dial(SIP/host ... ) |
| 07:15.08 | Juggie | then it is |
| 07:15.23 | Juggie | Dial(${TRUNKGROUP}host ... ) |
| 07:16.05 | jaxxan | TRUNKGROUP=SIP/ipaddress |
| 07:16.06 | jaxxan | or |
| 07:16.11 | *** join/#asterisk Koshatul (~evangelio@202.9.38.223) |
| 07:16.16 | jaxxan | TRUNKGROUP=SIP/sip.conf entry |
| 07:17.12 | Inv_arp | jaxxan: eg.. Dial(SIP/${EXTEN:1}@proxy01.sipphone.com) for me |
| 07:17.39 | RestLessGemini | ping |
| 07:19.43 | *** part/#asterisk santiago (~santiago@201.245.167.96) |
| 07:21.04 | jaxxan | i dont understand this |
| 07:21.11 | jaxxan | i normally send all calls to zap/g1 |
| 07:21.34 | jaxxan | but now i want to redirect through my ethernet port |
| 07:22.43 | jaxxan | i create a sip.conf entry |
| 07:23.07 | Inv_arp | jaxxan: then set an outbound context that points out to sip |
| 07:23.13 | jaxxan | [AS5350] |
| 07:23.13 | jaxxan | type=peer |
| 07:23.13 | jaxxan | host=xxx.xxx.xxx.xxxdtmfmode=rfc2833 |
| 07:23.31 | jaxxan | then in extensions.conf |
| 07:23.41 | Goshen | does md5secret=4a8e71480c5b1ef0a5d502a8eb98576a go in [general] of sip.conf? |
| 07:23.46 | jaxxan | i point my TRUNKGROUP Variable from zap/g1 |
| 07:23.54 | jaxxan | to SIP/AS5350 |
| 07:23.55 | jaxxan | right ? |
| 07:24.05 | Goshen | the wiki is kind of general |
| 07:25.50 | ta[i]nted | my voice prompts keep getting cut off.. is there a way to stop that? |
| 07:25.53 | RaYmAn-Bx | Goshen: I would think it has to be in the section for the unit in question. At least that works :) |
| 07:26.05 | Goshen | ok |
| 07:26.27 | RaYmAn-Bx | it's equivalent to secret in that sense I believe |
| 07:26.47 | Goshen | how does auth=md5 work then? |
| 07:27.50 | Inv_arp | jaxxan: paste your extesnion.conf on pastebin.ca |
| 07:27.59 | jaxxan | k |
| 07:28.20 | RaYmAn-Bx | sorry, I don't use that at all...It seems to work just fine with just having an md5secret instead of a secret :) |
| 07:28.32 | Goshen | ok |
| 07:29.10 | jaxxan | http://voip.bluesky.as/extensions.conf |
| 07:30.35 | RaYmAn-Bx | Goshen: it obviously only offer limited security though..if people get the md5sum they can still use your account (but can't change pw and stuff)..And I think you still need the actual passwords in register => statements(although I'm not entirely sure of that) |
| 07:31.01 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 07:31.42 | Juggie | worst fialplan ever :P |
| 07:31.53 | Juggie | dialplan |
| 07:31.59 | jaxxan | ok |
| 07:32.06 | jaxxan | that doesn't help |
| 07:32.08 | jaxxan | (= |
| 07:32.18 | jaxxan | lemme explain what i'm trying to do |
| 07:32.22 | Juggie | jaxxan, if you dont understand dialplan |
| 07:32.25 | Juggie | dont copy someone elses |
| 07:32.27 | Juggie | start a small one |
| 07:32.41 | modulus_ | what's the web ui port on gs101? |
| 07:32.43 | Juggie | Goshen, did u get it working? |
| 07:33.32 | jaxxan | my landline destined calls go to out zap/g1 to my dms100, from there it's directed to a cisco AS5350 media gateway which is then routed VOIP to our VOIP Provider for termination |
| 07:33.44 | jaxxan | we wanna bypass the dms100 and go to a new AS5350 via ethernet |
| 07:33.59 | Zeeek | The dialplan is the heart of an Asterisk system, as it defines how Asterisk should handle calls. |
| 07:33.59 | Zeeek | http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN650 |
| 07:34.05 | jaxxan | i'm not sure how to send it out the ethernet interface destined for the AS5350 |
| 07:34.26 | jaxxan | FYI, my dialplan is custom fit to our needs |
| 07:34.35 | Goshen | juggie: not yet incoming is fine...but the minute I put a secret on outgoing it locks out the incoming |
| 07:34.46 | Goshen | pisses me off... |
| 07:34.47 | Inv_arp | jaxxan: TRUNKGROUP=SIP/123456@host |
| 07:35.01 | Juggie | even with insecure=very ? |
| 07:35.04 | *** join/#asterisk rainM (~nic3try@p3.pub.ro) |
| 07:35.10 | rainM | re all |
| 07:35.12 | rainM | help |
| 07:35.17 | Goshen | yup |
| 07:35.25 | Juggie | did u try both examples from the wiki? |
| 07:35.27 | Goshen | it always goes to the outgoing peer |
| 07:35.28 | Goshen | yup |
| 07:35.32 | rainM | WARNING[8924]: file.c:550 ast_readaudio_callback: Failed to write frame |
| 07:35.41 | Goshen | not the first one, just the more complicated one |
| 07:35.57 | Goshen | trying first one now |
| 07:36.00 | modulus_ | damn grandstream... i can't connect web interface |
| 07:36.02 | modulus_ | grrrr |
| 07:36.19 | rainM | can anyonet tell me what that error is ? |
| 07:37.55 | Juggie | Goshen, that 404 u get, i want to see the error |
| 07:38.01 | ta[i]nted | what is the reserved extension name for when user dials zero |
| 07:38.16 | zignig | jaxxan: I would be more carefull about putting files like that on the internet. |
| 07:38.18 | ta[i]nted | is it exten => 0,1,Application() |
| 07:38.40 | zignig | jaxxan: the cdr/voice mail passwords are expoesed. |
| 07:38.59 | Juggie | and i have no idea why u have every person in the system |
| 07:39.05 | Juggie | defined in the dialplan |
| 07:39.08 | Juggie | it makes no sense |
| 07:39.54 | jaxxan | cause, that's where it goes |
| 07:39.59 | Goshen | IT works! |
| 07:40.02 | Goshen | inbound and outbound |
| 07:40.04 | zapa | hi all can i configure DND or Call Forward for my Budgetone 100 phone? or is possible to implent in dial plan ? |
| 07:40.06 | *** join/#asterisk firestrm (F1r3570rm@S010600047577bccd.gv.shawcable.net) |
| 07:40.10 | Goshen | with that simple little sip.conf amazing |
| 07:40.22 | Juggie | hah.... i said that an hour ago :) |
| 07:40.29 | Goshen | going to add my experience to the wiki...untested my foot! |
| 07:40.34 | *** join/#asterisk vinmoh (~vinmohfx@pcp0010311885pcs.avenel01.nj.comcast.net) |
| 07:40.50 | Juggie | :) good stuff |
| 07:40.58 | firestrm | anyone here familiar with fxotune? Im trying to run it and it reports that it ran ok, but i just ran too fast for me to believe it ran correctly |
| 07:41.05 | Juggie | allways satisfying to win and get something working :P |
| 07:41.30 | rainM | anyone had erros like that file.c:550 ast_readaudio_callback: Failed to write frame |
| 07:41.46 | rainM | my asterisk keeps krasing :(( |
| 07:41.54 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 07:41.58 | Goshen | crashing? |
| 07:41.59 | Inv_arp | rainM: did u chge anything? |
| 07:42.14 | Goshen | rainM: never had that error |
| 07:42.28 | rainM | no.. i have an IVR |
| 07:43.20 | vinmoh | hey I'm new to Irc |
| 07:43.26 | rainM | an i have someting playing in background |
| 07:43.51 | rainM | i have it recodet with a phone.. coul this be a problem ? |
| 07:44.01 | Goshen | juggie: I added that working config to the nikotel page too |
| 07:44.04 | vinmoh | I joined in cause I got a new task at my company to set up a small VOIP Solution with :asterisk" and I'm new to asterisk uhh |
| 07:44.26 | Goshen | vinmoh: welcome :) |
| 07:44.43 | vinmoh | Thanks "goshen" |
| 07:44.49 | rainM | i just have 10 chanels on my asterisk :) playing that ivr |
| 07:44.55 | vinmoh | Hope all are having a great time in here |
| 07:45.15 | Goshen | when not pounding my head against the desk in frustration, yes :) |
| 07:45.35 | RaYmAn-Bx | vinmoh: a good place to start is the asterisk section at www.voip-info.org :) It contains a lot of information..and of course download asterisk and get used to the configuration files |
| 07:45.39 | vinmoh | I have a question on Asterisk + adit 600 implimentation |
| 07:46.12 | vinmoh | I did download asterisk and played with config files and comands ..today |
| 07:46.26 | vinmoh | I conected to chanel bank and mod probed it |
| 07:46.48 | slePP | does anyone know why a channel bank connected to asterisk just disappears from time to time? give it about 6 hours, and it stops responding at all. the channel bank does its thing, but asterisk never notices if a line gets picked up |
| 07:47.17 | vinmoh | But the chanel bank have some problem so we are trying to impliment with adit 600 |
| 07:48.36 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
| 07:48.48 | vinmoh | our company is a small start up..we have 4 at&t Lan lines coming in ..so I need 1 FXO bank in adit + 1 FXS bank + a T1 cable conected to a PCI card on my Linux asterisk right? |
| 07:49.12 | |Vulture| | vinmoh: you mean land right? |
| 07:49.21 | |Vulture| | hahah |
| 07:49.31 | |Vulture| | I was gunna say... 4 PRIs... wow |
| 07:50.03 | ta[i]nted | how do i transfer someone in the middle of a call in asterisk? |
| 07:50.04 | vinmoh | Look we are a start up company trying to do a small research on VOIP |
| 07:50.08 | vinmoh | :) |
| 07:50.13 | ta[i]nted | in a SIP channel.. |
| 07:51.47 | vinmoh | Vulture: sorry LAN means Regular analog connection (we use LAN @ office as a slang) :) |
| 07:52.00 | Zeeek | anyone here using SMS ? |
| 07:52.07 | vinmoh | Dont confuse with Local Area Net :) |
| 07:52.11 | |Vulture| | vinmoh: oh wow... that could get confusing |
| 07:52.20 | vinmoh | yeh sorry vulture |
| 07:52.27 | |Vulture| | vinmoh: so you want to run VoIP phones or regular phones? |
| 07:52.51 | vinmoh | regular phone connected to FXS as extentions |
| 07:53.05 | firestrm | anyone ever have problems getting fxotune to work properly? i just terminates immediatly |
| 07:53.19 | firestrm | i cant seem to make it work :( |
| 07:53.23 | vinmoh | Later IP phones implimentation (sip) |
| 07:53.41 | |Vulture| | vinmoh: how many? |
| 07:54.00 | vinmoh | We need 6 extention now at office |
| 07:54.31 | vinmoh | Then later ip phone connected through Internet @ our guest house |
| 07:54.42 | |Vulture| | yea then you prolly need a bank for the FXS and a TDM4** for the 4 FXOs |
| 07:54.44 | vinmoh | 5 ip phones |
| 07:54.57 | |Vulture| | IP phones are easy, they work VERY well with * |
| 07:55.05 | vinmoh | cool |
| 07:55.25 | vinmoh | I need to do the sip configuration for Ip phones right? |
| 07:55.26 | |Vulture| | go with the Polycom IP500s they are cheap for what they do |
| 07:55.31 | |Vulture| | yup |
| 07:56.10 | vinmoh | but we already have adit 600 contributed by our friend |
| 07:56.35 | vinmoh | So I like to set up with adit 600 and I dont have any idea about adit uhh |
| 07:56.47 | |Vulture| | ah oky |
| 07:56.58 | |Vulture| | me neither but if you already have a bank then go with that |
| 07:57.52 | Inv_arp | |Vulture|: cheap for what they do? like |
| 07:58.01 | vinmoh | I think I should connect 4 incomings Analoge to FXO module on adit 600 ..then connect the extentions at FXS and run a T1 cable to the T1 PCI card at Asterisk right? |
| 07:58.26 | Goshen | Polycom IP500 - H.323 VoIP Phone ??? |
| 07:58.28 | |Vulture| | Inv_arp: they are great phones, cisco quality, great config |
| 07:58.30 | |Vulture| | SIP |
| 07:58.37 | |Vulture| | they come in both h323 and SIP |
| 07:58.42 | Goshen | $299.00 ??? |
| 07:58.45 | |Vulture| | the 323s can be converted it just takes work |
| 07:58.52 | |Vulture| | $180 |
| 07:58.58 | |Vulture| | IP600 is 300 |
| 07:59.05 | Inv_arp | aahh nice |
| 07:59.18 | Goshen | eeks...if you are a startup...get a nice black grandstream for each desk |
| 07:59.26 | vinmoh | So which channel bank would you sugest other than adit? |
| 07:59.43 | |Vulture| | vinmoh: I don't use channel banks so I can't be much help |
| 07:59.54 | |Vulture| | I promote IP phones... I am not the one to talk to lol |
| 07:59.55 | Goshen | http://voipstore.pulver.com/product_info.php?products_id=33&osCsid=679d0f6caab6b87ccd85d9f2de03f5a4 |
| 07:59.58 | Goshen | $65 |
| 08:00.17 | vinmoh | oh ok vulture |
| 08:00.31 | Goshen | vinmoh: look at the phone at that url |
| 08:00.34 | Goshen | I have one, works great |
| 08:00.43 | vinmoh | sure Goshen |
| 08:01.01 | Goshen | the display flashes when I have voicemail waiting which is cool |
| 08:01.19 | Goshen | bedtime, night all |
| 08:03.16 | vinmoh | wow goshen..they have a preconfigured PBX too |
| 08:03.18 | slePP | nono.. 4 _lan_ lines i'd expect to be 40mbit/s of goodness |
| 08:03.18 | slePP | :> |
| 08:03.18 | slePP | vinmoh: yes, that sounds about right. how many local phones d you want? and you won't be using SIP/IAX phones? |
| 08:03.21 | slePP | ta[i]nted: using what device for the call? a SIP phone, PSTN/PRI call, analog phone on a channel bank..? |
| 08:03.29 | slePP | |Vulture|: fix my channel bank for me, okay? :> |
| 08:04.01 | ta[i]nted | slePP analog phone |
| 08:04.02 | vinmoh | we will be using sip phones too once we set up the extentions through fxs |
| 08:04.39 | Zeeek | what's the best way to read the first line of a file from shell? |
| 08:04.39 | vinmoh | night goshen |
| 08:05.03 | slePP | hmm... |
| 08:05.07 | slePP | lost internet threr |
| 08:05.11 | Zeeek | In fact the second line |
| 08:05.25 | slePP | ta[i]nted: try usin T or t in the Dial() command |
| 08:05.32 | slePP | and then you can use # to transfer |
| 08:05.36 | vinmoh | Hey where would I get a best step by step instructions on setting up asterisk with chanel banks for the first time on his/her life :) |
| 08:05.40 | slePP | check: show application Dial |
| 08:05.41 | slePP | for more info |
| 08:05.51 | slePP | vinmoh: http://www.voip-info.org/ |
| 08:06.12 | vinmoh | thanks slePP |
| 08:06.21 | Zeeek | I have a file and need to read only the second line. What's the best way fromthe shell pls? |
| 08:06.23 | slePP | just search for channel bank |
| 08:06.33 | slePP | it pretty much walks you through step by step of the T1 stuff you need to do |
| 08:06.40 | slePP | the rest of the asterisk config info is there as well |
| 08:07.54 | vinmoh | nice |
| 08:08.33 | vinmoh | But is there any books or cd which I can buy which will help me to find step by step instruction ? |
| 08:08.50 | PTG123 | anyone here use realtime? |
| 08:09.55 | slePP | vinmoh: dunno |
| 08:10.02 | slePP | i think there is an asterisk book |
| 08:10.38 | firestrm | anyone here ever used fxotune? |
| 08:10.59 | *** join/#asterisk cp5 (~samy@chcgil2-ar7-4-3-040-086.chcgil2.dsl-verizon.net) |
| 08:11.01 | cp5 | hi |
| 08:11.51 | cp5 | if a call comes in, it executes an agi, does a SET VARIABLE, then dials another machine over IAX, any way i can pass that set variable to that other machine? |
| 08:12.26 | vinmoh | I got the info on book through google... |
| 08:13.00 | vinmoh | slePP" I dont know if its worth its "VoIP Telephony with Asterisk" |
| 08:13.14 | vinmoh | Any one ever tried that book ? |
| 08:13.19 | *** join/#asterisk Astinus_ (~abba@213.167.111.138) |
| 08:14.28 | slePP | not i, said the cow |
| 08:15.35 | *** join/#asterisk jas_williams (~Jason@host81-155-66-178.range81-155.btcentralplus.com) |
| 08:15.59 | *** part/#asterisk Astinus_ (~abba@213.167.111.138) |
| 08:18.16 | slePP | Judge Judy is on! |
| 08:18.17 | slePP | woohoo! |
| 08:22.03 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
| 08:23.18 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 08:23.41 | *** join/#asterisk Mother_ (~mother@93.Red-80-32-127.pooles.rima-tde.net) |
| 08:23.47 | Mother_ | ha! |
| 08:24.28 | Mother_ | I know what the problem with this double NAT is, for some reason, * is giving the remote Cisco it's LAN IP instead of the WAN IP to send RTP to |
| 08:25.42 | RaYmAn-Bx | externip configuration option in sip.conf might be what you are looking for? :P As well as localnet |
| 08:25.51 | Mother_ | I have that set |
| 08:26.02 | Mother_ | I mean I have externip, but localnet? |
| 08:26.19 | Mother_ | something like localnet=10.0.0.1 ? |
| 08:26.38 | RaYmAn-Bx | well, hmm. localnet just defines who to tell the real ip address to afaik |
| 08:26.59 | RaYmAn-Bx | localnet=10.0.0.1/255.255.255.0 for example |
| 08:27.13 | Mother_ | OK thanks, will try that now |
| 08:27.30 | vinmoh | Good night all |
| 08:29.19 | Mother_ | SUCCESS!!!! |
| 08:29.26 | Mother_ | argh gotta run now, thanks a lot! |
| 08:30.44 | *** part/#asterisk vinmoh (~vinmohfx@pcp0010311885pcs.avenel01.nj.comcast.net) |
| 08:32.09 | *** join/#asterisk clive- (~pirch@myw-stp-66-18-86-216.sentechsa.net) |
| 08:32.49 | *** join/#asterisk JamesDotCom (~james@202.92.213.226) |
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| 08:35.02 | *** join/#asterisk RoyK (~roy@83.80-203-29.nextgentel.com) |
| 08:37.10 | firestrm | anyone know why fxotune dies with this? Skipping non-TDM / non-FXO |
| 08:37.10 | firestrm | fxotune: successfully set echo coeffecients on FXO modules |
| 08:39.06 | Inv_arp | whats the syntax to inlide a file again? |
| 08:39.10 | Inv_arp | err include |
| 08:41.30 | *** join/#asterisk TrevMeister (~thammonds@ip68-4-223-70.oc.oc.cox.net) |
| 08:43.14 | shido6 | #include ? |
| 09:01.34 | Zeeek | . |
| 09:01.41 | JamesDotCom | .,. |
| 09:04.18 | ta[i]nted | what does it mean if i cannot hear my moh? |
| 09:04.37 | ta[i]nted | I see -- Started music on hold, class 'default' ... but no music |
| 09:04.48 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 09:04.52 | ta[i]nted | and i did 'make mpg123' in /usr/src/asterisk |
| 09:04.59 | ta[i]nted | restarted asterisk |
| 09:06.22 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 09:07.12 | ta[i]nted | are there mpg123 alternatives for moh? mpg123 likes to hang |
| 09:11.00 | *** join/#asterisk modulus_ (modulus@rm-f.net) |
| 09:11.01 | modulus_ | wotcher |
| 09:17.15 | |Vulture| | ta[i]nted: are you running 1.0.6? |
| 09:17.35 | *** join/#asterisk afe ([QBiitF5XB@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
| 09:18.55 | Zeeek | exit |
| 09:19.35 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 09:25.25 | *** join/#asterisk terrapen (~cjs@cpe-66-25-86-139.satx.res.rr.com) |
| 09:25.46 | terrapen | evening |
| 09:28.52 | ta[i]nted | |Vulture| how do i tell.. |
| 09:32.54 | |Vulture| | start * |
| 09:33.06 | |Vulture| | what does it say as the version |
| 09:33.42 | |Vulture| | ta[i]nted: it will say "Connected to ASterisk (version)..." |
| 09:34.49 | *** join/#asterisk fabrizioxxx (FABRIZIO@81.208.26.86) |
| 09:35.03 | Inv_arp | anyone have the iax2 dial string for fwd? |
| 09:35.17 | Zeeek | yes, fwd does on their site |
| 09:35.41 | Zeeek | <PROTECTED> |
| 09:36.02 | Zeeek | there's a whole page on it |
| 09:36.46 | fabrizioxxx | hello everyone .. i'm setting a variable in my dialplan with the SetVar comand. Is this var global or is it referres only to the current extension .. eg ... if from extension 444 i do a SetVar,foo=555 then from extension 999 i do a SetVar,bar=888 .. will this last one overwrite the first one? |
| 09:37.14 | Inv_arp | hmm i get Call rejected by 65.39.205.121: No authority found from fwd |
| 09:37.43 | Zeeek | fab note that there is a setvarglobal or something |
| 09:38.04 | Zeeek | SetGlobalVar |
| 09:38.11 | Zeeek | that kinda answers the question |
| 09:38.25 | fabrizioxxx | ok thks ... i'll check the manual straight away.. |
| 09:38.35 | ta[i]nted | |Vulture| CVS-HEAD-02/12/05 |
| 09:38.39 | Zeeek | in fact type show application stevar |
| 09:38.44 | Zeeek | you'd see this: |
| 09:38.45 | Zeeek | Setvar(#n=value): Sets channel specific variable n to value |
| 09:38.46 | *** part/#asterisk terrapen (~cjs@cpe-66-25-86-139.satx.res.rr.com) |
| 09:38.51 | Zeeek | which is the exact asnwer |
| 09:39.07 | Zeeek | heh, well SetVar, not stevar |
| 09:39.36 | Zeeek | show application stevar would probably be about coppice |
| 09:39.48 | fabrizioxxx | yes ... exaclty waht i was looking for ... thank you very much .. |
| 09:40.13 | *** join/#asterisk JohnnyC (~JoaoCorre@81.193.116.63) |
| 09:41.30 | RoyK | wtf does this mean? |
| 09:41.30 | RoyK | <PROTECTED> |
| 09:41.44 | Zeeek | it means they don't want you |
| 09:41.51 | Zeeek | ~lart RoyK |
| 09:41.58 | RoyK | ~lart Zeeek |
| 09:42.05 | Zeeek | I love fish heads |
| 09:42.15 | Zeeek | you wouldn't want that |
| 09:42.21 | *** join/#asterisk A-Tuin|work (~A-Tuin@nat.office.legend.net.uk) |
| 09:42.36 | *** join/#asterisk glLoadIdentity (~tuyan@dsl81-214-9283.adsl.ttnet.net.tr) |
| 09:42.51 | Zeeek | hey Roy my phone still works after 18 hours |
| 09:43.06 | fabrizioxxx | one more question ... when my users dial out they add a 0 in front of the number to dial eg: num -> 02566897, they dial 002566897.. is there some way so that when they dial the first 0 before the number that they get some sort of tone in the phone.. just to let them know that they caught the external line? Is this controlled in * or within the phone i am using? |
| 09:43.41 | Zeeek | common question but I can't rememeber the answer :) |
| 09:43.48 | Zeeek | I'd guess no |
| 09:44.09 | Zeeek | unless you want the to hit 0 and then get a real telco dialtone |
| 09:44.15 | Zeeek | (assuming POTS) |
| 09:44.37 | fabrizioxxx | well .. at the moment i'm using a ISDN modem .... |
| 09:45.03 | fabrizioxxx | supposing i use POTS could this be possible...? |
| 09:45.24 | Zeeek | you can dial 0 and give them a line that asterisk no longer controls easily enuf |
| 09:45.39 | Zeeek | as if they were dialing direct - they will be |
| 09:45.45 | glLoadIdentity | exten => 0,1,PlayBack(dialtone) or there is an application to provide dialtone.. |
| 09:46.12 | Zeeek | nice but the tone won't stop when they dial |
| 09:46.18 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
| 09:46.27 | ta[i]nted | glLoadIdentity what are u trying to do? |
| 09:46.27 | Zeeek | unless you want just like 1 second of tone |
| 09:46.31 | fabrizioxxx | yes .. already tried that .. it doen't stop .. |
| 09:46.43 | Zeeek | it stops if the soundfile ends |
| 09:46.45 | fabrizioxxx | Zeeek, even 1 sec is ok |
| 09:47.05 | Zeeek | you want just a tone for 1 second? No you want a tone that stops when they hit the first key |
| 09:47.23 | Zeeek | one sec, just record the tone in a 1 sec file |
| 09:47.39 | fabrizioxxx | yes .. actually you're right .. maybe it is better that it stops with the next keypress.. |
| 09:47.51 | Zeeek | picking up the phone, there should already be a dialtone |
| 09:48.08 | Inv_arp | fabrizioxxx: ignorepat => 0 |
| 09:48.24 | fabrizioxxx | Inv_arp, ok .. i'l try it now ... |
| 09:48.31 | Zeeek | I don't think that's the question :) |
| 09:48.43 | Zeeek | or is it ??? |
| 09:48.43 | mamcinty | Zeek, what kind of phones are you using? |
| 09:48.57 | fabrizioxxx | .. ignorepat removess the leading digit right? |
| 09:49.05 | Zeeek | me? GS, Farfon and soon many more |
| 09:49.16 | Inv_arp | fabrizioxxx: http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html/x862.html |
| 09:49.26 | Inv_arp | read towards bottom |
| 09:49.44 | Inv_arp | actually center |
| 09:50.21 | Zeeek | you are so right! |
| 09:50.33 | Zeeek | I'll have to remember this for the next time it comes up.... tomorrow |
| 09:50.42 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 09:50.48 | mamcinty | I asked because I know my Sipura SPA-2000 has the ability to set a tone to play when a certain pattern is matched it its internal dialplan |
| 09:51.20 | Zeeek | thx mam but I'm not the one asking about this |
| 09:51.22 | ta[i]nted | is there a way to bypass moh during call transfer? |
| 09:51.45 | mamcinty | So I was wondering if the phones you were using might have a similar feature |
| 09:54.34 | *** part/#asterisk powD (~atob100@83.146.53.34) |
| 09:54.36 | *** join/#asterisk powD (~atob100@83.146.53.34) |
| 09:55.32 | Zeeek | mam i don' think the gs do that |
| 09:56.08 | mamcinty | I am trying to think of an easy way to do it within the asterisk dialplan as you guys were saying |
| 09:56.50 | *** join/#asterisk dyer_ru (dyer@cube.sut.ru) |
| 09:57.52 | *** join/#asterisk neopher (~crazy@mail.techhelpresources.com) |
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| 09:58.37 | Zeeek | to what end ? |
| 09:58.55 | Delvar | Zeeek: the world! |
| 09:59.02 | mamcinty | haha |
| 09:59.33 | Delvar | hmmm cafeen injection |
| 10:00.10 | glLoadIdentity | ta[i]nted: me ? i am answering the questions.. |
| 10:00.13 | mamcinty | which GS do you use? |
| 10:00.25 | *** join/#asterisk terrapen (~cjs@cpe-66-25-86-139.satx.res.rr.com) |
| 10:00.39 | terrapen | anybody up for some phisher hunting? |
| 10:00.44 | Zeeek | I have a BT101 and a 102 |
| 10:01.23 | Zeeek | nespresso machine is the greatest invention since asterisk. In fact it predates asterisk |
| 10:02.04 | Zeeek | SO I got SMS working and now I can deluge my partner with SMS while she's on the train to make it look like I'm working |
| 10:02.26 | *** join/#asterisk cereal_ (~nico@gifu.newel.net) |
| 10:02.30 | cereal_ | morning all |
| 10:02.51 | Zeeek | cereal in the morning |
| 10:02.58 | Zeeek | where's eggs_ ? |
| 10:03.29 | *** join/#asterisk neopher (~crazy@mail.techhelpresources.com) |
| 10:07.35 | *** join/#asterisk neopher (~crazy@mail.techhelpresources.com) |
| 10:10.37 | *** join/#asterisk Jas_Williams (~Jason@host81-155-66-178.range81-155.btcentralplus.com) |
| 10:11.53 | *** join/#asterisk Savage-S (~savage@c514701e0.cable.wanadoo.nl) |
| 10:12.00 | clive- | whats does the module GTK consol so ? |
| 10:12.02 | clive- | do |
| 10:18.18 | Zeeek | do you have GTK? |
| 10:19.37 | *** join/#asterisk dg1nsw (~schulte@gate.sympat.de) |
| 10:22.18 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 10:23.20 | cereal_ | I m looking for real good compatible SIP phones with asterisk for affordable price any advices ? |
| 10:23.54 | Zeeek | Grandstream BT102 |
| 10:24.01 | Zeeek | or Sipura |
| 10:24.36 | clive- | Zeek I dont even know what GTK is..:) |
| 10:24.59 | Zeeek | you use X ? |
| 10:25.15 | clive- | zeek no |
| 10:25.18 | Zeeek | cereal_ Sipura SPA-841 |
| 10:25.45 | Zeeek | then you don't need the GTK thingie... are you getting make errors with that in it? |
| 10:27.01 | Zeeek | anyone familiar (as in ordered from) voipsupply.com ? |
| 10:27.19 | *** join/#asterisk shaZwaz (~adnans@203.81.196.167) |
| 10:27.31 | clive- | I am using chan_capi, and for some strange reason it keeps trying to load zap "torisa" stuff |
| 10:29.18 | RoyK | er |
| 10:29.18 | RoyK | SIP/2.0 487 Request Terminated |
| 10:29.21 | RoyK | wtf is this? |
| 10:29.45 | shaZwaz | 1.0.6 ? |
| 10:30.02 | RoyK | shaZwaz: 1.0.3 |
| 10:30.26 | *** join/#asterisk DEVILoper (~x@202.5.145.50) |
| 10:30.30 | DEVILoper | Hi All |
| 10:30.44 | shaZwaz | unload chan_sip.so and load again |
| 10:30.45 | Zeeek | Roy? |
| 10:30.51 | DEVILoper | any pointer from where i can read about GSM protocol ? |
| 10:30.54 | Zeeek | If the UAS has not issued a final response for the original request, its behavior depends on the method of the original request. If the original request was an INVITE, the UAS SHOULD immediately respond to the INVITE with a 487 (Request Terminated). |
| 10:31.05 | RoyK | Zeeek? |
| 10:31.09 | *** join/#asterisk owh (~onno@OptusSatelliteServices.22bjc76f09.optus.net.au) |
| 10:31.13 | Zeeek | Roy? ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 10:31.19 | RoyK | shaZwaz: doesn't help. tried restarting asterisk |
| 10:33.06 | *** join/#asterisk Dibbler_ (~Dibbler@zidane.pi-net.net) |
| 10:33.08 | *** join/#asterisk oej (~oej@216.65.229.136) |
| 10:33.22 | cereal_ | zeek Sipura very good for registring with asterisk ? (cause i had issues already for grandstream to registrer) |
| 10:33.26 | owh | Hey all, this may sound like an ignorant question, but I've been googling 'til my fingers are blue and I cannot find an answer. I'm using an IAX2 channel across a satellite link. The call takes around 30 seconds to settle. I want to play a message across the link while this is happening. Using either a macro or an announce in the dial I can play the message to the called party, but on this end I have silence, how do I play the me |
| 10:34.10 | Zeeek | cereal_ I don't have a Sipura yet but I plan to get one to evaluate. I have 2 GS that have no problem with htis; it's firmware dependent on GS |
| 10:34.42 | *** join/#asterisk ta[i]nted (~ta_i_nted@65-60-70-243-cust.telepacific.net) |
| 10:34.54 | shaZwaz | owh did u check dial options ? |
| 10:35.23 | owh | shaZwaz: Yes, that is how the recipient is hearing the message, but on this side I hear silence. |
| 10:35.29 | *** join/#asterisk brc__ (~brian@brc.base.supporter.pdpc) |
| 10:35.53 | cereal_ | zeek : wich firmware u use in your GS ? |
| 10:35.55 | shaZwaz | there is one option for playing it on both ends I guess |
| 10:36.11 | owh | (This side, being on the far end of the satellite link, with a local Asterisk box) |
| 10:36.29 | owh | shaZwaz: I've not found that option. |
| 10:37.14 | shaZwaz | di u check m ? |
| 10:37.37 | owh | Yes, I can play to the other end with either A() or M(). |
| 10:38.10 | shaZwaz | m is for ur end |
| 10:38.21 | Zeeek | cereal_ I'm stuck at 5.16 which works fine but is missing a few features |
| 10:38.21 | shaZwaz | the Caller that is |
| 10:38.28 | owh | AFAIK, M() is a macro. |
| 10:39.03 | shaZwaz | the dial option m |
| 10:39.06 | shaZwaz | not M |
| 10:39.08 | RoyK | hmmmmm |
| 10:39.18 | owh | shaZwaz: The one to play a macro? |
| 10:39.24 | RoyK | seems I couldn't bind to the box's secondary IP |
| 10:39.27 | owh | s/play/execute/ |
| 10:39.29 | *** join/#asterisk mady (~root@61.11.24.250) |
| 10:39.29 | shaZwaz | not the macro |
| 10:39.38 | RoyK | that is, it worked somehow, but iax-sip-calls didn't work |
| 10:40.14 | shaZwaz | it play music to the caller until connected |
| 10:40.21 | shaZwaz | to the callee |
| 10:40.32 | cereal_ | Zeeek what kind of features missing ? |
| 10:40.36 | mady | hello all i am mady and new user of asterisk can anyone help me out |
| 10:40.41 | owh | shaZwaz: The call is connected immediately, it doesn't take 30 seconds to connect, it takes 30 seconds to settle after it's connected. |
| 10:40.52 | owh | shaZwaz: Thus the music on hold doesn't work. |
| 10:41.31 | shaZwaz | then see it WaitforSilence can help |
| 10:41.37 | cereal_ | mady shoot :) |
| 10:42.02 | mady | thanks cereal now |
| 10:42.35 | Zeeek | cereal_ I'm not where that phone is so now I can't rember which firm I have |
| 10:42.49 | Zeeek | but the missing stuff is like attended transfer I think |
| 10:42.53 | shaZwaz | or u can transfer the caller to some extension for 30 secs and reconnect him back |
| 10:43.03 | Zeeek | I can't remember now if I'm at .12 or .16 |
| 10:43.05 | owh | shaZwaz: Hmm, if I take the satellite link out of the equation for testing, calling between two ATA186, then the recipient only gets the message, not the caller. |
| 10:43.08 | mady | when i am using command modprobe wcfxs it is there is error as |
| 10:43.10 | cereal_ | Zeeek : that doesnt bother me ; will try that 5.16 firmware then |
| 10:43.21 | Zeeek | I think 5.18 is the recommended level |
| 10:43.27 | Zeeek | there is a beta 22 already |
| 10:43.32 | shaZwaz | like park him for 30 secs ans then after the timeout he will be connected back |
| 10:43.44 | cereal_ | last officail stabvle on grandstream website is 5.16 |
| 10:43.44 | Zeeek | the issues with the earliers ones were registering |
| 10:43.45 | owh | shaZwaz: Eeek. |
| 10:43.53 | Zeeek | ok |
| 10:43.58 | cereal_ | mady : wich error u have at modprobe ? |
| 10:44.03 | Zeeek | I tried everything from .11 to .22 |
| 10:44.26 | owh | shaZwaz: That sort of defeats the purpose, seeing that the traffic between the handsets needs to settle... |
| 10:44.47 | mady | yaa unable to open master device /dev/zap/ct1 |
| 10:44.48 | owh | shaZwaz: The message is just to indicate that something is happening... |
| 10:45.31 | cereal_ | mady u got that device listed with ls in /dev/ ? |
| 10:45.37 | owh | Should the playback() not play to everyone on the call? |
| 10:45.50 | cereal_ | Zeeek where do u find this 5.16+ firmwares ? |
| 10:46.19 | shaZwaz | once u dial I guess u can't jump into something else |
| 10:46.21 | mady | cereal:actually am not sure can u pls giude me how do i check |
| 10:47.02 | cereal_ | mady : cd /dev/zap |
| 10:47.08 | cereal_ | ls -al ct* |
| 10:47.43 | Zeeek | <PROTECTED> |
| 10:47.59 | Zeeek | or their tftp : TFTP server: 168.75.215.189 |
| 10:48.08 | shaZwaz | I think parking the call is one good option |
| 10:48.08 | mady | cereal_ : it is showing no such file or directory |
| 10:48.18 | owh | So, has no-one come up with a mechanism that says something like: "One moment while your call is being connected?" |
| 10:48.46 | cereal_ | ok mady so it didnt create the device |
| 10:49.02 | cereal_ | mady wich linux are u using ? |
| 10:49.13 | shaZwaz | owh, u can do that b4 u dial |
| 10:49.48 | mady | cereal_:i am using Digium TDM400P card and Kernel version 2.6.X.X |
| 10:49.51 | owh | shaZwaz: But then the caller would need to wait 60 seconds and they'd still have silence while the message was being played to the recipient would they not? |
| 10:50.04 | mady | cereal : it fedora core |
| 10:50.09 | shaZwaz | I guess |
| 10:50.33 | owh | Or would a background() before dial play to me and then a playback() after dial play to them? |
| 10:51.08 | shaZwaz | owh u can't jump into playback or background once u execute dial |
| 10:51.13 | cereal_ | mady there should be a packet rpm with zaptel tools |
| 10:51.18 | shaZwaz | uless u trasfer that call |
| 10:51.31 | cereal_ | i dont know the fedora packet system it s rpm right ? |
| 10:52.00 | owh | shaZwaz: If that was true, the recipient would never have heard the message at all... |
| 10:52.29 | shaZwaz | its in the Dial Cmd |
| 10:52.33 | mady | cereal_:actually i am not sure but if can guide me i can give u the exact information |
| 10:53.33 | ta[i]nted | how do i include another context in my dialplan again? |
| 10:53.41 | ta[i]nted | something like include => [context] ?? |
| 10:53.55 | shaZwaz | without the [ ] |
| 10:53.59 | cereal_ | mady : on debian there is a "zaptel" packet |
| 10:54.01 | *** join/#asterisk LarsAC (~chatzilla@134.130.124.227) |
| 10:54.08 | cereal_ | not the module source i mean |
| 10:54.16 | cereal_ | try to find on rpm list zaptel |
| 10:55.28 | mady | cereal:it is showing pckage zaptel is not installed |
| 10:56.01 | *** join/#asterisk justnulling1 (justnullin@ool-18bc013e.dyn.optonline.net) |
| 10:56.08 | justnulling1 | hey |
| 10:56.09 | owh | shaZwaz: I've just changed the dial plan to background(), then dial(), this plays the message to me, gives me a ringing, then rings the other hand set, plays the message, then connects the call. Halfway there... |
| 10:57.05 | owh | shaZwaz: Now all I need to have happen is that the call starts dialling as soon as the message starts playing on my end. |
| 10:57.40 | *** join/#asterisk amer (~aaa@203.99.60.27) |
| 10:58.18 | shaZwaz | use the M(X) and do it is the macro ...may be another background |
| 10:58.19 | cereal_ | and libmady : install it :) |
| 10:58.46 | mady | cereal :what should i do? |
| 10:59.06 | RoyK | http://www.citypages.com/databank/26/1264/article12985.asp <-- america the best? |
| 11:00.23 | Zeeek | the sun does orbit the earth |
| 11:00.52 | owh | shaZwaz: I tried it in the macro, but only the recipient gets it, I get ringing. |
| 11:01.16 | shaZwaz | u did what |
| 11:01.17 | justnulling1 | i need a pbx gateway in ny that supports did, any sugestions |
| 11:01.31 | shaZwaz | can u pastebin that ? |
| 11:01.41 | cereal_ | mady something like rpm -i zaptel |
| 11:01.41 | owh | shaZwaz: Background and playback both in the macro... Yup, one mo... |
| 11:03.13 | RoyK | Zeeek: I know |
| 11:03.21 | ta[i]nted | where can i find 1.0.6 ChangeLog? |
| 11:03.22 | mady | cereal: could u please tel me from where i should download this? |
| 11:03.49 | Zeeek | is there really a 1.0.6 ? |
| 11:03.56 | Zeeek | how is it? |
| 11:03.58 | owh | shaZwaz: http://www.pastebin.com/249160 |
| 11:04.13 | shaZwaz | ta[i]nted, asteriskdocs |
| 11:04.34 | ta[i]nted | http://dev.asteriskdocs.org/ is down |
| 11:06.16 | shaZwaz | try sineapps |
| 11:06.20 | Zeeek | not down, b0rken |
| 11:06.41 | *** join/#asterisk nazgool (~nazgoool@port-83-236-180-106.static.qsc.de) |
| 11:06.43 | nazgool | hi |
| 11:07.02 | Zeeek | iznogood |
| 11:07.53 | owh | shaZwaz: Did that pastebin make sense to you? |
| 11:09.27 | cereal_ | mady : give me second |
| 11:09.59 | mady | cereal_ Ok |
| 11:10.13 | cereal_ | mady : rpmfind.net , just type zaptel and take package for fedora |
| 11:10.15 | shaZwaz | owh everything u do in the macro is audible to the callee only ? |
| 11:10.28 | shaZwaz | thats strange |
| 11:10.30 | owh | Yes |
| 11:10.45 | nazgool | if i'd like to get a log of what asterisk would see as console messages, is there any better solution than to try something like asterisk -vvvvvc 2>&1 >mylog.txt ? |
| 11:11.27 | shaZwaz | owh check the WaitforSilence cmd |
| 11:12.44 | owh | shaZwaz: The only reference I can find to that is this: http://www.voip-info.org/wiki-Asterisk+auto-dial+out |
| 11:13.35 | shaZwaz | wud have worked with the macro thou |
| 11:14.02 | owh | Am I perhaps missing something else? |
| 11:15.30 | shaZwaz | u need to fill in ur announcement for the 30 secs period .....and pass no audio |
| 11:15.58 | owh | Preferably, the audio runs across the link during the 30 seconds... |
| 11:16.49 | shaZwaz | try parking the call in the macro |
| 11:20.29 | owh | shaZwaz: I just executed the macro before the call, changed the playback to background and still have the same result. It seems that the playing of the sound file is blocking the continuation of the dial plan, so it won't do anything until the sound is completed, which makes no sense to me. |
| 11:21.13 | shaZwaz | ofcourse |
| 11:21.22 | owh | How so? |
| 11:21.27 | shaZwaz | did u try parking the call in the macro |
| 11:21.45 | owh | Sorry, missed that, for which call? |
| 11:22.13 | owh | With the macro running for both, or just for the recipient? |
| 11:22.14 | shaZwaz | the one u dialing |
| 11:22.52 | shaZwaz | just park it in the macro |
| 11:23.38 | amer | Hey guys how can I make asterisk relay all error messages |
| 11:23.43 | amer | SIP |
| 11:24.12 | owh | shaZwaz: Can you hit me with a cluestick and tell me how I park the call in the macro? |
| 11:24.12 | amer | My setup is sip proxy=========Asterisk==========SIP proxy=======PSTN |
| 11:24.34 | shaZwaz | see parkandannounce cmd |
| 11:25.01 | *** join/#asterisk eipi (~eipi@200.114.172.100) |
| 11:25.22 | amer | When I get an error msg from PSTN side Asterisk blicks that message and sends a 403 msg back to the SIP proxy |
| 11:25.31 | amer | blicks=blocks |
| 11:27.25 | owh | shaZwaz: It eludes me... |
| 11:27.29 | *** join/#asterisk raarts (~chatzilla@fw.office.netland.nl) |
| 11:27.58 | shaZwaz | what ? |
| 11:28.22 | raarts | I am having problems with the hint extension |
| 11:28.28 | owh | shaZwaz: I don't understand what it is you want me to do or how to achieve it. |
| 11:28.56 | shaZwaz | phew |
| 11:29.02 | raarts | I have a SNOM that is subscribed, but asterisk does not send NOTIFY messages to that SNOM |
| 11:29.06 | raarts | How do I debug this? |
| 11:31.59 | shaZwaz | owh I am mixed up myself |
| 11:32.11 | shaZwaz | dunno if that will work |
| 11:33.14 | Zeeek | are we still on the satelite link? |
| 11:33.22 | TheEmperor | hello |
| 11:33.35 | shaZwaz | yup k |
| 11:33.36 | owh | I don't understand why a background() sound is blocking the call progress until it is completed, unless it's not that at all, but that the ringing noise is being queued and nothing happens until that can actually play. |
| 11:33.38 | TheEmperor | can anyone help me with musiconhold? i get this error message |
| 11:33.42 | owh | Zeeek: You betcha... |
| 11:33.45 | TheEmperor | -- Executing Answer("SIP/3036-dbc5", "") in new stack |
| 11:33.45 | TheEmperor | <PROTECTED> |
| 11:33.45 | TheEmperor | Mar 4 19:31:10 WARNING[1225991360]: res_musiconhold.c:305 moh0_exec: Unable to start music on hold (class '') on channel SIP/3036-dbc5 |
| 11:33.45 | TheEmperor | <PROTECTED> |
| 11:34.22 | ta[i]nted | TheEmperor i think you don't have a class defined in musiconhold.conf |
| 11:34.39 | Zeeek | good call |
| 11:34.49 | Zeeek | unless the class is name '' |
| 11:34.54 | Jas_Williams | TheEmperor: You need to post your musiconhold.conf to pastebin.ca but it looks like you have not defines a moh class |
| 11:34.55 | ta[i]nted | true |
| 11:34.56 | TheEmperor | ta[i]nted: ihave using loud=> |
| 11:35.11 | TheEmperor | [classes] |
| 11:35.11 | TheEmperor | ;default => quietmp3:/var/lib/asterisk/mohmp3 |
| 11:35.11 | TheEmperor | loud => mp3:/var/lib/asterisk/mohmp3 |
| 11:35.11 | TheEmperor | ;random => quietmp3:/var/lib/asterisk/mohmp3,-z |
| 11:35.24 | shaZwaz | Zeeek what owh needs is a sound to fill in (30 secs) after the call has been connected, audible to both sides |
| 11:35.37 | ta[i]nted | TheEmperor but u didn't specify the class in your musiconhold() |
| 11:36.00 | TheEmperor | ta[i]nted: erm you mean in the extensions file? |
| 11:36.07 | ta[i]nted | yea |
| 11:36.08 | Zeeek | owh send an SMS saying there will be 30 sec of silence while the call settles |
| 11:36.24 | owh | How do I do that? |
| 11:36.28 | TheEmperor | ta[i]nted: so in extensions it should be MusicOnHold(default) ? |
| 11:36.29 | Jas_Williams | TheEmperor: Hane you placed musicclass=loud in sip.conf |
| 11:36.35 | ta[i]nted | TheEmperor if u omit it, it will go to default |
| 11:36.40 | Zeeek | <PROTECTED> |
| 11:36.48 | shaZwaz | well in that case the announcement thing works fine in the beginning of the call |
| 11:36.52 | owh | Ah, I understand, no the audio needs to play across the link to *make* it settle :-) |
| 11:37.07 | TheEmperor | Jas_Williams: will check |
| 11:37.14 | ta[i]nted | MusicOnHold("loud") is what you want i think |
| 11:37.36 | ta[i]nted | are there alternatives to mpg123 for MOH? |
| 11:37.44 | owh | shaZwaz: The message followed by 30sec of ringing is pretty disconcerting... |
| 11:37.50 | ta[i]nted | i can't have hung processes on the production box |
| 11:38.08 | TheEmperor | ta[i]nted: MusicOnHold("loud") doesn't work.. |
| 11:38.30 | ta[i]nted | has your MOH ever worked? |
| 11:38.35 | ta[i]nted | is mpg123 installed? |
| 11:38.48 | TheEmperor | i don't think mpg123 is installed... |
| 11:39.04 | shaZwaz | if the audio doesn't pass how can they hear the music ? |
| 11:39.18 | shaZwaz | for the intial 30 secs ? |
| 11:39.32 | TheEmperor | ta[i]nted: show i install that first? |
| 11:39.32 | owh | Huh? |
| 11:39.55 | ta[i]nted | TheEmperor yea |
| 11:40.01 | ta[i]nted | you need it to play mp3s |
| 11:40.04 | owh | shaZwaz: Are you talking to me? |
| 11:40.07 | shaZwaz | owh ... what does settle exactly mean here |
| 11:40.29 | TheEmperor | ta[i]nted |
| 11:40.35 | TheEmperor | ta[i]nted:i go download that then :) |
| 11:40.46 | ta[i]nted | TheEmperor hurry back or i will miss you |
| 11:40.51 | shaZwaz | if there is a 30 sec gab in the conversation, how come they can hear the announcements ? |
| 11:40.54 | owh | shaZwaz: Ah, well, lots of jitter and breakup. For some reason a played sound file does not exhibit that problem. So the idea is, play the sound file, get the call stable, then get the parties to talk. |
| 11:41.36 | TheEmperor | ta[i]nted:hahaha :) |
| 11:43.05 | shaZwaz | okay now I guess that can work |
| 11:43.28 | Zeeek | Play some music like MIdnight Express |
| 11:43.35 | Zeeek | it will sync to it |
| 11:43.48 | owh | All I want is that both ends hear the same thing... |
| 11:44.15 | owh | Zeeek: Something with a better beat would make it sync even better :-) |
| 11:44.48 | Zeeek | dum dum dum dum dum dum DUM dum .... |
| 11:45.08 | Zeeek | isn't that Midnight Express? |
| 11:45.13 | shaZwaz | make an extension for the caller to ring extract the number, park the caller, dial the number using the A(30 sec file) reconnect the both calls |
| 11:45.17 | mady | cereal: hi unfortunetly i didnot found any rpm package name zaptel in rpmfind.net |
| 11:45.32 | visik7 | mady build it by your own |
| 11:45.38 | Jas_Williams | TheEmperor: do a make mpg123 in the src/asterisk directory |
| 11:45.40 | owh | At the moment the message is: "Hi, one moment while your call in being connected. This is an experimental call across a satellite link using VoIP... etc." |
| 11:45.50 | Jas_Williams | then a make install |
| 11:45.58 | Zeeek | ok, both callers go to meetme as well as a channel that is playing moh ? |
| 11:46.19 | Zeeek | a call from space |
| 11:46.23 | owh | shaZwaz: Say that again? |
| 11:46.29 | jeofrey | ohmmmmmmmm Zeeek not working the advice of you grandson regarding the port forwarding of wrt router to asterisk..... :( |
| 11:46.30 | owh | Zeeek: Yup. |
| 11:46.48 | Zeeek | jeofry why not? |
| 11:46.53 | jeofrey | hahahaha |
| 11:47.03 | jeofrey | i try everything there also cannot |
| 11:47.04 | shaZwaz | I think that will work |
| 11:47.37 | ta[i]nted | no one here has mpg123 process problems? |
| 11:47.43 | Zeeek | owh ok you direct a call file: extension one that plays music into meetme |
| 11:48.02 | Jas_Williams | ta[i]nted: not when using mpg123r |
| 11:48.09 | mady | visik7:could u pls help me from i will get src image |
| 11:48.16 | ta[i]nted | Jas_Williams what is mpg123r |
| 11:48.25 | Jas_Williams | version r |
| 11:48.36 | ta[i]nted | i am using .59r |
| 11:48.51 | Zeeek | jeof what *is* wroking ? Anything ? |
| 11:48.52 | ta[i]nted | and there are rogue mpg123 processes after * exits |
| 11:48.57 | Jas_Williams | that one :) High Performance MPEG 1.0/2.0/2.5 Audio Player for Layer 1, 2 and 3. |
| 11:48.57 | Jas_Williams | Version 0.59r |
| 11:49.03 | shaZwaz | why the call file zeeek that can be done in the dial plan |
| 11:49.05 | Zeeek | jeofrey have you tried FWD ? |
| 11:49.12 | jeofrey | Zeeek maybe can ask again ur grandson that i am using asterisk-h323 connecting to my voip providers and grandstream connected to asterisk using sip |
| 11:49.26 | Zeeek | h323? No way |
| 11:49.30 | jeofrey | hahahahha |
| 11:49.30 | owh | shaZwaz: I'm not yet sure I understand what you are trying to tell me. |
| 11:49.36 | Zeeek | he's way too modern to be using that old stuff |
| 11:49.44 | Jas_Williams | h323 Yuk, you need an alg on your nat gateway |
| 11:49.54 | jeofrey | nyahhhhhhhhhhh |
| 11:49.56 | shaZwaz | needs a firm understanding of the dialplan language |
| 11:49.59 | mady | visik7: hello visik7 could you tel me which version of redhat linux support this by default |
| 11:50.04 | Zeeek | but is the SIP part working or not? Can yoiu dial echo() ? |
| 11:50.34 | jeofrey | can dial |
| 11:50.39 | visik7 | mady none |
| 11:50.44 | jeofrey | but they cannot heard me |
| 11:51.01 | Zeeek | without h323 in thepoicture, right? |
| 11:51.11 | visik7 | mady maybe some fedora |
| 11:51.13 | mady | visik7 : oobs it is news for me |
| 11:51.14 | Zeeek | [defau |
| 11:51.18 | Zeeek | jeez |
| 11:51.29 | shaZwaz | its kinding of puting the caller on hold for 30 secs soon as he dials, then dial the number of the callee make him listent he announcement A() and then reconnect him to the caller |
| 11:51.45 | Zeeek | owh wacky problem - good luck |
| 11:51.48 | owh | shaZwaz: Ah... |
| 11:52.00 | mady | visik7 now i want to build my own but i need your help |
| 11:52.03 | owh | shaZwaz: I understand, now I get to make it happen :-) |
| 11:52.10 | Zeeek | wait send your callee to meetme alone with moh |
| 11:52.14 | owh | Zeeek: Seeing that I'm wacky, that's OK :-) |
| 11:52.27 | visik7 | mady read the docs |
| 11:52.34 | shaZwaz | but u will need to understand programming the dialplan |
| 11:52.34 | Zeeek | sedn yourself to another meetme with moh |
| 11:52.48 | shaZwaz | or programming in the dialplan |
| 11:52.49 | Zeeek | then after 30 s, go to the first meetme |
| 11:53.44 | owh | shaZwaz: After being a developer for 24 years, this is pretty straight forward :-) |
| 11:53.51 | mady | visik7 ok |
| 11:54.02 | shaZwaz | good then it will be smooth |
| 11:54.02 | *** join/#asterisk d-tech (~dtc@node-423a1ebb.cle.onnet.us.uu.net) |
| 11:54.17 | owh | shaZwaz: Now all I need to do is figure out why the manual and reality don't match :-) |
| 11:54.40 | shaZwaz | u can have a look at AGI too |
| 11:55.24 | Zeeek | and writing modules in C |
| 11:55.57 | Zeeek | in fact that would be the best solution for a programmer |
| 11:56.25 | Zeeek | show application WackyPlaybackMOF_for_two() |
| 11:56.43 | Zeeek | nope, hasn't been thought of yet |
| 11:58.24 | *** join/#asterisk Mother_ (~m@53.Red-217-126-93.pooles.rima-tde.net) |
| 11:58.29 | Zeeek | Mutha! |
| 12:00.28 | Mother_ | yo! |
| 12:00.54 | Mother_ | I solved the 7960 problem |
| 12:01.32 | Mother_ | basically I changed the bind from 0.0.0.0 to the actual LAN IP of the asterisk box, and added the localnet parameter |
| 12:02.00 | Mother_ | I did some sniffing and the 7960 was sending the RTP packets to the LAN IP of the * box, which of course would never work |
| 12:02.17 | *** join/#asterisk meppl (~mephisto@p3E9E2C22.dip.t-dialin.net) |
| 12:02.29 | meppl | guten morgen |
| 12:02.40 | Mother_ | morgen guten |
| 12:02.52 | meppl | guten morgen mother |
| 12:03.38 | *** join/#asterisk jcims (~jcims@cpe-69-135-121-57.columbus.rr.com) |
| 12:11.19 | *** join/#asterisk jerlique (jerlique@lnk254.adl0.adsl.esc.net.au) |
| 12:11.28 | Zeeek | sounds like a nat=yes where it shouldn't have been? |
| 12:12.12 | Mother_ | nope |
| 12:12.35 | Mother_ | well, I believe nat=yes should be there, no? |
| 12:12.39 | Zeeek | what's the best way to send a short email a shell script? |
| 12:12.52 | Zeeek | not in the client sip.conf |
| 12:13.23 | Zeeek | I have a script that send mail except when called from * |
| 12:13.34 | Mother_ | so the client's entry in sip.conf should not have nat=yes? I don't think it made a difference, as I had two lines in the 7960, one as local and one as nat and none worked |
| 12:13.40 | Zeeek | others called from asterisk use mime-construct |
| 12:14.12 | Zeeek | nat=yes would need to be where, um, nat is yes :) |
| 12:14.25 | *** join/#asterisk TheEmperor (TheEmperor@218.111.50.103) |
| 12:14.44 | Mother_ | hehe well then nat=yes was correct for this setup |
| 12:15.08 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 12:15.11 | Mother_ | still it was when * became aware of it's local IP and subnet that it stopped sending it's own IP as the RTP destination to the 7960 |
| 12:16.00 | Mother_ | before I had bind 0.0.0.0 and no localnet parameter |
| 12:16.46 | Mother_ | btw how about sendmail email@domain.com < ~/maildir/mailmessage |
| 12:17.53 | Zeeek | ya for some reason that isn't working in th script called by system |
| 12:17.55 | ariel_ | morning |
| 12:18.03 | Mother_ | hmmmm |
| 12:18.06 | Mother_ | morning |
| 12:18.53 | Mother_ | Zeeek: maybe echo "Message Content" | mail -s "Subject goes here" you@domain-name.com ? |
| 12:19.01 | Mother_ | without the ? |
| 12:19.03 | Mother_ | hehe |
| 12:19.27 | Zeeek | I was considering that option |
| 12:19.49 | Zeeek | or even cat $file | mail .... |
| 12:20.00 | Zeeek | oh, cat is more "expensive" |
| 12:20.08 | Mother_ | lol |
| 12:20.15 | jcims | anyone know why my cisco 7960 isn't picking up 'next-server' from dhcp as its tftp server? |
| 12:20.21 | Zeeek | but then I'll bet mail ain't cheap either |
| 12:20.31 | Mother_ | because it's Cisco and can do whatever it wants |
| 12:20.33 | jcims | i can see it sending it via a sniffer |
| 12:20.34 | jcims | haha |
| 12:20.35 | Zeeek | jcims I hate when that happens |
| 12:20.36 | jcims | i'm learning that |
| 12:20.46 | Mother_ | I can't get my 7960 to take alternative RTP ports, so I'm in a similar spot :) |
| 12:21.32 | Mother_ | but I got it doing SIP through a double NAT, so I'm quite happy today |
| 12:21.34 | jcims | kinda sucks..they're nice phones...just a bit stubborn! :) |
| 12:22.11 | *** join/#asterisk shaZwaz (~adnans@203.81.196.167) |
| 12:22.58 | Zeeek | Interesting: cat $file | mail -s "subj here" works, mail doesn't |
| 12:23.09 | dyer_ru | strange problem. firefly-3dparty, iax, ilbc or gsm codec. incoming/outgouing call from/to firefly crash my asterisk box. |
| 12:23.15 | Mother_ | really? |
| 12:23.20 | Zeeek | I vonder vy ? |
| 12:23.28 | Mother_ | who knows ;) |
| 12:23.39 | Zeeek | vrom asterisk dialplan ONLY |
| 12:23.48 | Zeeek | vrom zystem call |
| 12:23.52 | Mother_ | I know a friend had problems sending mail under certain circumstances and never found out why |
| 12:24.20 | Zeeek | and the system admin said "don't worry... a lot men have this problem?" |
| 12:24.21 | Mother_ | in the end by trying different ways like you found one that worked, and left it at that |
| 12:24.25 | Mother_ | LOL |
| 12:24.35 | Mother_ | bbiab |
| 12:24.43 | Zeeek | FOOD needs me |
| 12:24.50 | Zeeek | to prcess it |
| 12:26.05 | ta[i]nted | i am having problems getting MusicOnHold to play |
| 12:26.05 | ta[i]nted | it executes the application |
| 12:26.05 | ta[i]nted | but i don't hear audio |
| 12:26.51 | ariel_ | ta[i]nted, if you go to a linux prompt and put mpg123 what version do you get |
| 12:27.04 | ta[i]nted | 0.59r |
| 12:27.33 | ta[i]nted | i haven't replaced the default mp3s in /var/lib/asterisk/mohmp3 |
| 12:27.54 | ariel_ | ta[i]nted, what version of asterisk are you running? |
| 12:28.19 | ta[i]nted | Asterisk CVS-HEAD-02/12/05-15:01:06 |
| 12:30.18 | *** join/#asterisk Jas_Williams (~Jason@host81-155-66-178.range81-155.btcentralplus.com) |
| 12:30.40 | ariel_ | ta[i]nted, there was a bug in asterisk about that update they applied the fix to the current cvs |
| 12:31.19 | ta[i]nted | is current cvs stable? i've been running cvs-head on my production box lol |
| 12:31.37 | ta[i]nted | probably not the best idea... |
| 12:33.02 | ariel_ | ta[i]nted, I don't run head at all unless I am testing something. |
| 12:34.33 | ta[i]nted | ariel_ have u ever tried asteriskathome? |
| 12:34.44 | ariel_ | yes |
| 12:34.51 | ta[i]nted | was it any good? |
| 12:35.12 | ariel_ | ta[i]nted, it's actually very good. |
| 12:35.17 | ta[i]nted | i'm interested in it for AMP |
| 12:35.23 | ta[i]nted | hmm |
| 12:35.28 | Zeeek | how good is 1.0.6 ? |
| 12:35.42 | shaZwaz | Zeeek I have been looking for this fairly simple functionality, but I feel very restricted by Asterisk dialplan |
| 12:36.02 | ta[i]nted | shaZwaz what functionality |
| 12:36.23 | shaZwaz | trying to load some variables to be used as prefix for a outgoing line |
| 12:36.37 | shaZwaz | I need to these variables to be loaded from astDB |
| 12:36.39 | ta[i]nted | use AGI |
| 12:37.03 | ariel_ | Zeeek, it's good. |
| 12:37.27 | Zeeek | ariel real good ? |
| 12:37.47 | shaZwaz | ta[i]nted, can u gimee some specific pointers ? |
| 12:38.15 | ta[i]nted | shaZwaz do u know perl? |
| 12:38.25 | Zeeek | anyone in the conf call last nite ? |
| 12:38.28 | shaZwaz | go om |
| 12:38.30 | shaZwaz | on |
| 12:38.41 | ta[i]nted | that's it |
| 12:38.49 | ta[i]nted | if u know perl, agi is very easy |
| 12:39.07 | shaZwaz | so I need something external to do that |
| 12:39.13 | jerlique | tainted: can agi answer a call on a particular extension? |
| 12:39.20 | ta[i]nted | jerlique sure |
| 12:39.26 | jerlique | thank! |
| 12:39.31 | jerlique | s ;) |
| 12:40.11 | *** join/#asterisk EC-ASP (~alfredo@Intelideas-Avanzia.Mesena.MAD.ES.INTELIDEAS.NET) |
| 12:40.13 | ta[i]nted | jerlique $objAGI->answer(); i think |
| 12:40.17 | EC-ASP | good day |
| 12:40.30 | EC-ASP | wondering if anybody can help me with a rebel module |
| 12:40.35 | EC-ASP | for a Digium 110p |
| 12:40.59 | EC-ASP | I get ZT_SPANCONFIG failed on span 1: No such device or address (6) |
| 12:40.59 | EC-ASP | FATAL: Error running install command for wcte11xp |
| 12:41.09 | EC-ASP | when executing ztcfg |
| 12:41.10 | jerlique | ok - I ask, casue I want to develop an application which staff can use to transfer calls and answer calls by clicking on other staff names.... |
| 12:41.50 | ta[i]nted | jerlique i think there are a few applications for that already |
| 12:42.00 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-200-128.dsl.scarlet.be) |
| 12:42.34 | ta[i]nted | jerlique http://www.asternic.org/ |
| 12:42.38 | ta[i]nted | flash operator panel |
| 12:42.42 | jerlique | ok, I've had a look at gastman and another couple, but they dont seem to be able to do it. |
| 12:42.45 | ta[i]nted | i don't know how good it is though |
| 12:43.01 | jerlique | ok seem this, havent tested it yet. |
| 12:43.16 | jerlique | The demo online is iirc not active.... |
| 12:43.25 | EC-ASP | any experts in Digium cards around? I'm puzzled, this has worked well before... |
| 12:43.29 | ta[i]nted | jerlique let me know if u find/develop a good one.. i need something for the receptionist |
| 12:43.30 | jerlique | or fully function I should say |
| 12:45.06 | ta[i]nted | did http://dev.asteriskdocs.org/ get hosed? |
| 12:46.07 | EC-ASP | ta[i]nted, kind of scary, isn't it? |
| 12:46.09 | EC-ASP | :) |
| 12:46.54 | ta[i]nted | yea! |
| 12:46.58 | ta[i]nted | hope they backed up |
| 12:47.00 | ta[i]nted | haha |
| 12:47.12 | Zeeek | coffee |
| 12:47.41 | EC-ASP | It's almost lunch time here |
| 12:47.49 | EC-ASP | I think I'll leave my problems for l8r... |
| 12:48.18 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@bzq-207-183.red.bezeqint.net) |
| 12:51.34 | dyer_ru | hmm, iaxclient 1.0rc working successfully |
| 12:54.17 | Mother_ | back |
| 12:56.18 | Zeeek | mozzer ... |
| 12:56.40 | Zeeek | where were we before the sechuan beef arrived? |
| 12:56.50 | Mother_ | so, what kind of things would be missing in 1.0.6 from HEAD? |
| 12:57.04 | Mother_ | since I hear it's very good :) |
| 12:57.17 | Zeeek | and good head should never be passed up |
| 12:57.26 | Mother_ | hmmmm |
| 12:57.28 | Zeeek | so I'll think I'll get down with it |
| 12:57.38 | Zeeek | ooops, misread |
| 12:57.56 | ariel_ | Mother_, many new features. like Dundi, Realtime |
| 12:58.19 | Zeeek | what if we don't care about those two? and dialplan changes? |
| 12:58.30 | *** join/#asterisk bpoint (~bpoint@cn220.opt2.point.ne.jp) |
| 12:58.32 | Mother_ | ariel_: thanks, it's just that I've been running an early january head on a production box to test a few things |
| 12:58.54 | Mother_ | Zeeek: agreed, I was more interested in for example attended transfers |
| 12:59.19 | Zeeek | yea, dial changes? |
| 12:59.22 | ariel_ | Zeeek, check the changelog but yes they do have allot of new things. |
| 12:59.23 | Zeeek | or |
| 12:59.31 | Mother_ | ok |
| 12:59.33 | Zeeek | ok thanks I'll maybe check it out |
| 12:59.38 | Mother_ | same here |
| 12:59.38 | Zeeek | RealSoonNow (tm) |
| 12:59.42 | bpoint | sorry to barge in, but I'm having problems compiling asterisk 1.0.6... can anyone help out? |
| 13:00.02 | ariel_ | but it's still in beta so there are many big bugs |
| 13:00.16 | ariel_ | bpoint, what seems to be the problem? |
| 13:00.24 | *** join/#asterisk oej (~oej@216.65.237.25) |
| 13:00.25 | Zeeek | bpoint please tell us... |
| 13:00.35 | bpoint | ok here goes :) |
| 13:00.53 | bpoint | from what I can tell it looks to be libedit related |
| 13:01.07 | bpoint | from a fresh untar, I do a make |
| 13:01.22 | bpoint | and the first source file (editline.c) gives a whole lot of errors |
| 13:01.31 | bpoint | the first one being |
| 13:01.31 | Zeeek | been while - remind me someone, gunzip option to untar at the same time? |
| 13:01.32 | bpoint | emacs.c: In function `em_inc_search_next': |
| 13:01.32 | bpoint | emacs.c:474: error: `ED_SEARCH_NEXT_HISTORY' undeclared (first use in this function) |
| 13:01.38 | bpoint | tar -xzvf :) |
| 13:01.45 | Zeeek | thanks |
| 13:02.12 | bpoint | I found that there is an automatically generated "fcns.h" file |
| 13:02.29 | bpoint | unfortunately, I seem to be lacking _any_ EM* functions in mine |
| 13:02.34 | bpoint | er, ED, sorry |
| 13:02.42 | bpoint | I have EM and VI... |
| 13:03.20 | bpoint | any clues? :) I could copy/paste more of the make output if you like |
| 13:03.22 | ariel_ | bpoint, what tar file are you using and what distro |
| 13:03.37 | bpoint | the tar I just downloaded from asterisk.org, 1.0.6 |
| 13:03.55 | bpoint | my distro is well, custom... it's not any normal linux dist |
| 13:04.35 | ariel_ | bpoint, then don't use the tar. use the cvs instructions on your custom linux |
| 13:05.06 | bpoint | should I presume the cvs version contains more bugfixes then? |
| 13:05.10 | Mother_ | argh |
| 13:05.19 | ariel_ | bpoint, yes |
| 13:05.23 | bpoint | I was hoping to use this in a production environment :) |
| 13:05.30 | Mother_ | now I'll have to remember to reinsert the fax patch in 1.0.6 |
| 13:05.33 | Mother_ | grrrr |
| 13:06.00 | bpoint | okie, I'll give the cvs a try first |
| 13:06.01 | ariel_ | most that use a custom linux make there own tar from the cvs. |
| 13:06.14 | bpoint | *nod* thanks |
| 13:06.56 | *** part/#asterisk EC-ASP (~alfredo@Intelideas-Avanzia.Mesena.MAD.ES.INTELIDEAS.NET) |
| 13:06.59 | Zeeek | now I'm waiting for whatever I forgot to change to make the make die |
| 13:07.56 | gambolputty | I am having a dtmf problem in the cvs version of *. When I release a keypad button, only the tail end of the dtmf tone is output. |
| 13:08.09 | gambolputty | The tone is never heard while the keypad button is pressed |
| 13:08.40 | Zeeek | we used to try for that in the old radio days |
| 13:09.04 | gambolputty | what do you mean? |
| 13:09.28 | Zeeek | that used to be a desired effect on walkie-talkies |
| 13:09.49 | gambolputty | the dtmf tone is not long enough at all to be recognized at all by any service that needs it |
| 13:10.09 | Zeeek | I knew there'd be at least one this I forgot: warning: implicit declaration of function `gtk_init_check' |
| 13:10.09 | gambolputty | like my pstn voicemail service |
| 13:10.34 | Zeeek | did check the usual like the mailing list? |
| 13:10.45 | Zeeek | I haven't heard of the problem here |
| 13:10.53 | gambolputty | will try |
| 13:11.25 | ariel_ | bpoint, forgot to let you know your better off with the stable vesion instead of head |
| 13:12.37 | ariel_ | gambolputty, are you using sip phones or zap ports |
| 13:12.50 | bpoint | mm, I'll check to see if the cvs head compiles first or not |
| 13:12.51 | gambolputty | sip phones and a sipura 3000 |
| 13:12.57 | Mother_ | that is something that really boggles me, if there are many fixes in head, plus new features, how stable can stable really be considered? |
| 13:13.06 | gambolputty | I have tried setting different dtmfmode settings in sip.conf |
| 13:13.11 | gambolputty | nothing seems to work |
| 13:13.26 | Mother_ | i.e. how much testing is made on stable to call it stable so-to-speak? |
| 13:13.36 | ariel_ | gambolputty, if your using ulaw as a codec inband is the only one that works. |
| 13:13.58 | gambolputty | I do know that, I use ulaw only on my internal lan |
| 13:14.05 | gambolputty | and going out to the Internet too |
| 13:14.13 | ariel_ | Mother_, your the tester. It's not even going into a feature freeze yet. |
| 13:14.29 | ariel_ | ./going/gone |
| 13:14.31 | Mother_ | ariel_: that's very comforting LOL |
| 13:14.44 | bpoint | ariel: the cvs head just compiled editline.c without a problem |
| 13:14.44 | gambolputty | The one thing I don't understand is why a dtmf tone is not heard at all while I press down a keypad key |
| 13:14.57 | gambolputty | It seems like * is suppressing it |
| 13:15.04 | bpoint | ariel: either I'm gonna have to make a patch or stick with head :) |
| 13:15.27 | ariel_ | bpoint, it's something left out of the tar that you need. |
| 13:15.39 | Mother_ | gambolputty: depends on how the DTMF tone is sent, it can be mixed in the audio stream, generated by the phone, or it can be signalling that tells the other end a key is being pressed |
| 13:15.46 | ariel_ | gambolputty, is the service your going to via zap outbound or iax or sip? |
| 13:16.04 | bpoint | ariel: hmm, okay.. at least it compiles. I'll play with it some more. thanks! |
| 13:16.06 | gambolputty | it is a sipura 3000 no fxo card |
| 13:16.30 | ariel_ | gambolputty, do you have internal voicemail? |
| 13:16.34 | gambolputty | yes |
| 13:16.40 | ariel_ | does it work? |
| 13:17.00 | gambolputty | right now no |
| 13:17.13 | gambolputty | I may have to switch dtmf settings just to get that to work |
| 13:18.04 | gambolputty | dtmfmode=rfc2833 allows voicemail to work again |
| 13:18.07 | gambolputty | in sip.conf |
| 13:18.51 | gambolputty | but calling some number like Dell doesn't allow dtmf to work |
| 13:19.10 | gambolputty | their menus don't recognize my dtmf |
| 13:19.59 | Mother_ | gambolputty: what codec are you using? |
| 13:20.07 | gambolputty | ulaw |
| 13:20.17 | Mother_ | ok |
| 13:20.22 | gambolputty | all points in the chain are ulaw for sure |
| 13:20.27 | Mother_ | compression codecs would affect inband |
| 13:20.32 | gambolputty | between my snom phone and * |
| 13:20.43 | gambolputty | and from the sipura 3000 out |
| 13:21.52 | *** join/#asterisk strace (~strace@ADSL-F49-S197-critical-coi.nortenet.pt) |
| 13:22.08 | *** join/#asterisk nomad_ (nomad@delta-labs.packetstorm.at) |
| 13:22.35 | strace | http://www.pastebin.com/249196 |
| 13:22.38 | strace | help? |
| 13:22.45 | Mother_ | hello? |
| 13:22.55 | Zeeek | so far so good: Connected to Asterisk 1.0.6 currently running on r |
| 13:23.09 | Zeeek | now to see if it answers and calls :) |
| 13:23.30 | strace | hello |
| 13:23.31 | strace | help? |
| 13:23.32 | strace | :) |
| 13:23.41 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 13:23.46 | *** join/#asterisk ctooley ([U2FsdGVkX@199.89.146.18) |
| 13:24.40 | ariel_ | strace, what is that from? |
| 13:25.02 | strace | ariel_: misdn |
| 13:25.25 | ariel_ | hummm I have not seen that before. |
| 13:25.28 | PoWeRKiLL | hi |
| 13:25.28 | strace | :( |
| 13:25.32 | bpoint | hmm... I don't suppose there is an "at-least" required version for bison in order to get asterisk to compile properly? |
| 13:25.46 | strace | it's working now |
| 13:25.48 | strace | thank you |
| 13:26.00 | PoWeRKiLL | last stable cvs is buggy cause I don't know why the server is crashing with too many open files twice a day ? |
| 13:26.34 | ariel_ | bpoint, I don't know the required one. |
| 13:26.42 | Zeeek | the SMS app changed the directory where it stores msgs in 1.0.6 |
| 13:27.04 | bpoint | ariel: could you tell me your version? :) I have 1.28... *hides* |
| 13:29.11 | ariel_ | Version: 1.875c |
| 13:29.12 | ariel_ | Release: 2.EL |
| 13:29.24 | bpoint | ok, thanks |
| 13:29.34 | ariel_ | I use CentOS which is RHEL 3 |
| 13:30.28 | bpoint | np, there's a bunch of versions available on gnu.org... just wanted to know what would be a good version to try it with |
| 13:33.43 | *** join/#asterisk neopher (~crazy@mail.techhelpresources.com) |
| 13:34.22 | neopher | hmmm, my dial command is missing from the asterisk CLI, any ideas |
| 13:35.20 | bjohnson | anyone here want my iconnecthere free minutes from my voxilla purchase? |
| 13:37.31 | Grooby | w00t |
| 13:37.40 | Grooby | speex 1.1.7 is released |
| 13:37.57 | Mother_ | I use CentOS too, it has been the most trouble-free distro I've tried for * |
| 13:38.03 | Jas_Williams | bpoint: bison 1.85 |
| 13:38.16 | Jas_Williams | I think |
| 13:38.23 | bpoint | jas: thx... I'm gonna try 2.0 :) |
| 13:38.25 | Mother_ | apart from a minor mishap with hyperthreading being detected as a second processor but that was OK |
| 13:40.10 | bpoint | my 1.28 was released in 1999... *sigh* |
| 13:41.31 | *** join/#asterisk jtodd (~jtodd@adsl-146-36-69.mia.bellsouth.net) |
| 13:41.53 | *** join/#asterisk YoYo (~YoYo@pool-151-199-125-240.roa.east.verizon.net) |
| 13:43.13 | Zeeek | well,, whatever 1.0.6 fixed, it didn't change my callerid on the siemens phone |
| 13:43.50 | Zeeek | is *8 still the pickup a ringing line? |
| 13:43.56 | *** part/#asterisk drkludge (~drkludge@ip68-231-34-38.ph.ph.cox.net) |
| 13:44.24 | bpoint | btw: if I only intend to use sip with *, do I really need zaptel and libpri? |
| 13:44.37 | Zeeek | no unless you want meetme |
| 13:44.42 | Grooby | zaptel for timing device |
| 13:44.46 | Grooby | what's lbpri for? |
| 13:44.47 | Zeeek | then you'll need the - yeah |
| 13:44.54 | Zeeek | PRI |
| 13:44.55 | bpoint | timing? |
| 13:45.01 | bpoint | what kind of timing? |
| 13:45.22 | Zeeek | music on hold and meetme |
| 13:45.24 | bpoint | and what's meetme? :) |
| 13:45.30 | shaZwaz | which IAX client has G729 ? |
| 13:45.31 | Zeeek | Starter tutorial: |
| 13:45.31 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 13:45.31 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 13:45.31 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 13:45.31 | Zeeek | THE reference of the moment: |
| 13:45.32 | Zeeek | http://www.asteriskdocs.org |
| 13:45.37 | bpoint | ok |
| 13:45.53 | Zeeek | or show application meetme |
| 13:46.08 | bpoint | everything I've been reading so far has been on voip-info.org :) |
| 13:46.36 | Zeeek | a good technique to learn fast is to find the list of applications and scan them a few times |
| 13:46.43 | Zeeek | then you will at least have heard of them |
| 13:46.49 | bpoint | *nod* |
| 13:46.56 | Zeeek | that's what I did and I never regretted the few mintues |
| 13:47.04 | shaZwaz | Zeeek gimme some goos starter for AGI |
| 13:47.12 | Zeeek | it will generate a lot more questions |
| 13:47.24 | Zeeek | I don need no stinking AGI |
| 13:48.03 | Grooby | gonna google up on PRI |
| 13:48.11 | Zeeek | anyway bpoint - if you want conference rooms, you'll need zaptel |
| 13:48.32 | julianjm | Zeeek: or ztdummy |
| 13:48.37 | Zeeek | ya |
| 13:48.43 | bpoint | my only concern is that I read somewhere that the kernel sources are necessary... |
| 13:48.45 | Zeeek | but isn't that in zaptel? |
| 13:48.47 | *** join/#asterisk hemant (hemant@220.226.37.183) |
| 13:49.00 | Zeeek | I'd guess at least the includes ya |
| 13:49.02 | bpoint | that's something I don't have access to |
| 13:49.04 | julianjm | Zeeek:ahh, ok... i thouth you meant zaptel hardware ;) |
| 13:49.23 | Zeeek | I don't know what I meant |
| 13:49.30 | bpoint | hehe |
| 13:49.39 | bjohnson | woops .. can someone go to iax.cc and tell me if they see a sign in form |
| 13:50.03 | Zeeek | by the way, here's a picture of Saturn taken a few weeks ago : http://www.wineloverspage.com/graphics1/saturn.jpg |
| 13:50.15 | Zeeek | I'll try to get Uranus next |
| 13:50.23 | bpoint | bjohnson: yeah, there's a login form there |
| 13:50.29 | bjohnson | hey buddy .. stay away from my anus |
| 13:50.38 | Zeeek | I won't go there |
| 13:50.46 | Zeeek | not even with a probe |
| 13:50.47 | bpoint | saturn... pretty |
| 13:50.58 | Zeeek | nice shot, eh? Not me a friend took it |
| 13:50.59 | bpoint | there were some nice shots on space.com a few weeks ago |
| 13:51.00 | bjohnson | bpoint: oops .. I asked them to add a link to the signon .. not rplace their page with a sign on |
| 13:51.05 | *** join/#asterisk tzafrir_home (~tzafrir@bzq-179-40-134.cust.bezeqint.net) [NETSPLIT VICTIM] |
| 13:51.06 | *** join/#asterisk KryoStoffer (~kri@helium.kri.dk) [NETSPLIT VICTIM] |
| 13:51.21 | bpoint | zeeek: ooh! |
| 13:51.27 | bpoint | that is nice, then :) |
| 13:51.38 | Zeeek | and now a stupid question: |
| 13:51.50 | Zeeek | do I have to find and remake spandsp yet again? |
| 13:51.59 | Zeeek | I fear so |
| 13:54.19 | *** join/#asterisk YoYo (~YoYo@pool-151-199-125-240.roa.east.verizon.net) |
| 13:54.27 | *** join/#asterisk goatmilk (~goatmilk@130-127-45-50.chouse.resnet.clemson.edu) |
| 13:55.03 | *** join/#asterisk Lashkar (~Lashkar@12.44.215.11) |
| 13:56.20 | ta[i]nted | is festival okay to put on a production box? |
| 13:56.55 | *** join/#asterisk pjm_uk (~pjm_uk@cpc1-pool3-3-0-cust116.sot3.cable.ntl.com) |
| 14:00.03 | jontow | doing what? |
| 14:01.22 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
| 14:01.54 | Zeeek | spandsp worked! |
| 14:01.56 | Grooby | 1900 number |
| 14:02.02 | *** join/#asterisk RoyK (~roy@213.160.242.69) |
| 14:02.07 | Zeeek | shit this is the most productive day I've spent doing nothing in weeks |
| 14:02.34 | bpoint | congrats :) |
| 14:02.49 | Grooby | zeek how does it work? |
| 14:02.52 | Zeeek | amazing things happen when the female staff is out on appointments |
| 14:02.56 | Grooby | do you need a zaptel card? |
| 14:03.05 | Zeeek | for what? fax? |
| 14:03.06 | Grooby | or can it work over ulaw |
| 14:03.10 | Grooby | yeah |
| 14:03.17 | Zeeek | some say you can |
| 14:03.40 | Zeeek | but spandsp sometimes has trouble depending on whose machine is sending |
| 14:03.51 | Zeeek | ~seen coppice |
| 14:04.01 | jbot | coppice <~chatzilla@245.195.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 5d 53m 38s ago, saying: 'Zeeek: an engineer in India probably gets about 1/5 as much as a US one, so 500xcost would be a little expensive :-)'. |
| 14:04.01 | Grooby | interesting |
| 14:04.01 | Mother_ | Zeek: you'll have to remake spandsp |
| 14:04.02 | Grooby | i'll try to fax something to myself |
| 14:04.02 | Zeeek | I didn't and it worked |
| 14:04.04 | Zeeek | nya nya nya |
| 14:04.08 | Mother_ | really? |
| 14:04.10 | Mother_ | nice |
| 14:04.22 | Zeeek | God know why though |
| 14:04.26 | Mother_ | I had to reapply the patch manually and rebuild |
| 14:04.34 | Zeeek | the patch isn't a problem |
| 14:04.36 | Mother_ | first time around I had to patch manually too |
| 14:04.39 | Mother_ | yeah |
| 14:04.43 | Zeeek | but see I'm not sure which version to use |
| 14:04.56 | Mother_ | I tried the latest, as was told it was OK to use |
| 14:04.58 | Zeeek | because he always has the advanced beta stuff too |
| 14:05.05 | *** join/#asterisk Cherebrum (~jgarland@216.32.77.10) |
| 14:05.06 | Mother_ | exactly |
| 14:05.09 | Zeeek | as in 2pre9 ? |
| 14:05.11 | Mother_ | yep |
| 14:05.16 | Zeeek | hmmm |
| 14:05.35 | Mother_ | but incoming faxes were cut after a few seconds, and no matter what I tweaked it wouldn't work so I dropped it |
| 14:05.40 | Zeeek | so if I make that - oh yeat I'll need to remake asterisk |
| 14:05.51 | Mother_ | yes, patch asterisk then remake |
| 14:05.58 | Zeeek | well it works perfectly every time with j2.com (jfax) |
| 14:06.03 | Cherebrum | What time zone is this conference? |
| 14:06.04 | Zeeek | but many machines don't |
| 14:06.11 | Zeeek | it was yesterday |
| 14:06.13 | Mother_ | yeah |
| 14:06.20 | Zeeek | today is March 4th |
| 14:06.31 | Zeeek | (everywhere on the planet?) |
| 14:06.34 | Mother_ | Cherebrum: for me it's March 4th too |
| 14:06.41 | Mother_ | we can tell you in Beats |
| 14:06.56 | Mother_ | do they still make them swatches? |
| 14:07.04 | \Grooby\ | missed the dev conference yesterday |
| 14:07.06 | Zeeek | PHP still has the format in dat() |
| 14:07.24 | Zeeek | I listened for 90 minutes, mainly to see if the call quality would vary :) |
| 14:07.46 | Zeeek | and I got kicked after a few min, but I think everyone did - then 88 min of call |
| 14:07.58 | bpoint | it's almost march 5th here :) |
| 14:08.03 | Zeeek | worked perfectly and I'm far from the server |
| 14:08.07 | bpoint | after 51min :P |
| 14:08.21 | \Grooby\ | zeek, you talking about the dev conf right? |
| 14:08.39 | Zeeek | yowsuh |
| 14:08.50 | Zeeek | it was very enlightening |
| 14:09.04 | Zeeek | I don't dare to say much |
| 14:09.05 | \Grooby\ | so are they still talking about forking off or something? |
| 14:09.24 | Zeeek | I'm told it was three hours long |
| 14:09.40 | Zeeek | I wouldn't tell anyone to fork off |
| 14:09.44 | Zeeek | not polite |
| 14:09.52 | Mother_ | lol |
| 14:09.53 | \Grooby\ | hahahahaha |
| 14:09.54 | \Grooby\ | :P |
| 14:10.08 | Zeeek | now I can't remember who wrote the SMS stuff |
| 14:10.09 | \Grooby\ | kinda like fsck huh? |
| 14:10.42 | Zeeek | anyway, developers are like... oh well... I can't say |
| 14:11.00 | Zeeek | I used to work in integration of a european product in an american company |
| 14:11.16 | Zeeek | too weird |
| 14:11.24 | *** join/#asterisk Hmmhesays (negative3k@66.173.103.108) |
| 14:13.05 | \Grooby\ | bbiab |
| 14:13.34 | boch | could anyone giveme an example of AGI written in C? |
| 14:14.12 | jontow | boch; you can just use magical printf()'s :) |
| 14:14.16 | Zeeek | boch there is one in many languages incl c - I think on the wiki |
| 14:14.19 | *** join/#asterisk santiago (~santiago@201.245.167.78) |
| 14:14.27 | Hmmhesays | perl, php, python, c |
| 14:14.31 | Zeeek | yeah |
| 14:14.41 | Zeeek | there's a hello world in all those |
| 14:14.43 | Hmmhesays | php is my drug of choice for agi's |
| 14:14.54 | Zeeek | me too - but c is great too |
| 14:15.05 | boch | Zeeek yes, but ive tryed it and didnt work |
| 14:15.13 | Zeeek | tried wot? |
| 14:15.16 | jontow | i have a 7 line example in shell script :) |
| 14:15.17 | Hmmhesays | just one try? |
| 14:15.31 | Zeeek | did you sacrifice the chicken? |
| 14:15.35 | boch | Hmmhesays more than one |
| 14:15.53 | Zeeek | did you face Huntsville and turn three times chanting Geeee Peeeee Elllll ? |
| 14:16.07 | Zeeek | because I tried those examples and they work |
| 14:16.15 | boch | if you look at the code of that AGI, there is a while, well, its always in that while |
| 14:16.31 | bpoint | finally... bison-2.0 compiled and tested |
| 14:16.32 | Zeeek | fix it |
| 14:17.10 | boch | Zeeek ok |
| 14:17.23 | boch | Zeeek thanks a lot, i apreciate your help a lot |
| 14:17.25 | *** join/#asterisk rumba (~ropawa@cpe-68-201-148-205.sw.res.rr.com) |
| 14:17.37 | Zeeek | you're not asking any questions |
| 14:17.51 | boch | yes i did |
| 14:17.52 | Zeeek | I think those AGI examples work fine |
| 14:17.53 | Mother_ | LOL |
| 14:18.03 | Zeeek | been a while, but I'm sure I tested the C and the PHP |
| 14:18.47 | Zeeek | boch - your question was can anyone give you an example, right? It was answered |
| 14:18.53 | boch | im trying with xlite and it never connects |
| 14:19.18 | Mother_ | trying what? |
| 14:19.22 | Zeeek | show us the page in pastebin or something |
| 14:19.53 | boch | its doesnt say anythings, only when hungup prints Mar 4 11:09:36 WARNING[6127]: res_agi.c:1683 agi_exec: If you want to run AGI on hungup channels you should use DeadAGI! |
| 14:20.11 | nazgool | <PROTECTED> |
| 14:20.17 | Zeeek | boch what example? |
| 14:20.24 | boch | C |
| 14:20.35 | Zeeek | on the wiki? |
| 14:21.13 | boch | http://home.cogeco.ca/~camstuff/agi.html |
| 14:21.14 | boch | there |
| 14:21.59 | Mother_ | bbl |
| 14:23.02 | Zeeek | hmmmm that isn't exazctly the same |
| 14:23.32 | Zeeek | ah wait I modified it when I was trying to do something else, then gave up to write an app module instead |
| 14:24.00 | Zeeek | but I know that one works too |
| 14:24.44 | Zeeek | yes, it is always in the while() it's an infinite loop |
| 14:25.00 | Zeeek | until astrisk finishes talking to it, then it exits |
| 14:25.17 | boch | it seems, * never stops talking to it |
| 14:26.15 | Zeeek | did you slightly modify it? |
| 14:26.59 | boch | not yet |
| 14:29.47 | Zeeek | ok, I just compiled it it works perfectly |
| 14:30.05 | Zeeek | how else can I be of service to you? |
| 14:30.51 | jontow | http://mno.bsd.st/~jontow/systat.txt |
| 14:31.45 | Zeeek | cute - I wasq just thinking about doing this in a limited wau=y - to watch memory |
| 14:32.27 | *** join/#asterisk adjacent (~scott@office.bftwave.com) |
| 14:32.29 | jontow | real quick example using shell scripting |
| 14:32.44 | Zeeek | boch I suggest you keep plauying because that C example works perefectly |
| 14:32.45 | bjohnson | wow. that was fast. ordered my wrt54g yesterday afternoon and it showed up today |
| 14:32.55 | Zeeek | you won't regret it |
| 14:33.00 | jontow | gota do your own voice recording though :) |
| 14:33.11 | Zeeek | except if it has a serial number of 144F5431423142KL |
| 14:33.29 | Zeeek | that's the one stolen from my van |
| 14:33.41 | Zeeek | and I planted a mean backdoor in it |
| 14:34.11 | Hmmhesays | speaking of agi's i'm having a helluva time with mine looking in the wrong path for a text file |
| 14:34.14 | bjohnson | not unless staples repackaged and sold it as new |
| 14:34.29 | Zeeek | which they prolly often do with returns like every other store |
| 14:34.42 | Zeeek | but I digress |
| 14:35.16 | Zeeek | jontow how can I get the RAM usage stats from shell? |
| 14:35.17 | nestAr | sweet.. polycom's support has improved i think.. now instead of giving me the wrong answer.. they just don't reply |
| 14:35.23 | nestAr | i think that's an improvement |
| 14:35.26 | Zeeek | heh |
| 14:36.00 | jontow | OS dependant |
| 14:36.05 | jontow | some use 'vmstat' or similar |
| 14:36.11 | Zeeek | less irritating than the "I hope this has helped" answer having absolutely nothing to do with the question |
| 14:36.17 | jontow | some would argue its easier to parse it out of 'top's output |
| 14:36.29 | Zeeek | that was my first idea |
| 14:36.36 | *** join/#asterisk YoYo (~YoYo@pool-151-199-125-240.roa.east.verizon.net) |
| 14:36.40 | jontow | and some OS' are .. woo, who knows :D |
| 14:36.40 | *** join/#asterisk DHuang (~DHuang@203.41.13.154) |
| 14:36.45 | DHuang | hi... :-p |
| 14:36.49 | SexyKen | Hey guys -- Anyone have a problem like this: Sometimes when Agents login to the queue, they wont get the queue calls but Asterisk shows the agents as logged in and all that -- but it just says they're busy when it trys to give the agent a call....but if I restart asterisk, it works. |
| 14:36.55 | jontow | ok, updated that txt file with the spoken prompts in text :) |
| 14:37.11 | bjohnson | so any pointers on how to flash it and get * on it? |
| 14:37.15 | YoYo | Ken, what phones? |
| 14:37.26 | DHuang | anyone know why I'm getting - Answering with non-codec capability 0x1 (telephone-event) - even both SIP phone and asterisk are set to g729,iLIBC, gsm? |
| 14:37.47 | Zeeek | wow RAM usage went up hugely with 1.0.6! I better reboot to see what that means |
| 14:37.54 | *** join/#asterisk yaout (eric@CPE-65-30-220-56.wi.rr.com) |
| 14:38.22 | *** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net) |
| 14:40.09 | DHuang | anyidea? |
| 14:42.38 | *** join/#asterisk AgiNamu (~AgiNamu@216.230.151.230) |
| 14:42.53 | AgiNamu | Hey all |
| 14:43.10 | AgiNamu | How can I change things to save 'international' caller ID to my CDRs? |
| 14:43.27 | AgiNamu | My problem is that incoming caller id for the states is 10 digits, no country code |
| 14:43.41 | AgiNamu | so I'd like asterisk to add the 1 |
| 14:43.53 | BrianR___ | You can use the NXX rule... |
| 14:44.15 | AgiNamu | That'll work? there's no country codes starting with NXX? |
| 14:44.41 | *** part/#asterisk DHuang (~DHuang@203.41.13.154) |
| 14:45.05 | SexyKen | YoYo - Sorry for the late response. The phones are SoundPoint IP 600's |
| 14:45.31 | meppl | sexyken, are you sexy? |
| 14:46.03 | BrianR___ | AgiNamu: Do your international calls come in with dialable dn (ie, 011 prefix) or regular DN? |
| 14:46.40 | AgiNamu | well, the international calls come in a little screwy too. sometimes they come in with a prefix, sometimes clipped to 10 digits |
| 14:46.59 | AgiNamu | But given a 10 digit number, there's no way to know if it's a USA number missing the prefix, or a full international number eh? |
| 14:47.37 | AgiNamu | 5022157822 could be my cell in guatmeala, or someone in the middle of kentucky |
| 14:48.51 | BrianR___ | AgiNamu: That depends on the minimum DN length in guatmeala or whatever 50 is... |
| 14:49.05 | AgiNamu | 502 is guatemala country code |
| 14:49.17 | AgiNamu | and Guatemala used to be 7 digits after the country code |
| 14:50.27 | BrianR___ | Hmm.. That does kinda suck.. |
| 14:50.33 | BrianR___ | What kind of line are the calls coming in on? |
| 14:51.20 | AgiNamu | a DS3 |
| 14:51.27 | AgiNamu | I think it's a Telica |
| 14:51.32 | AgiNamu | and then IAX'd to me |
| 14:51.50 | AgiNamu | Part of the problem is the local Guatemalan telcos. they're flaming idiots. |
| 14:52.07 | AgiNamu | So there's no magic algorithm I don't know about to figure this out? :\ |
| 14:52.26 | *** join/#asterisk Red_6 (~alex@m174.net81-66-29.noos.fr) |
| 14:52.30 | *** join/#asterisk Casper_UA (~casper@as-2-22.ar43-2x.kharkov.ukrtel.net) |
| 14:53.51 | *** join/#asterisk nix000 (~nixman@66.11.190.225) |
| 14:54.09 | nix000 | anyone deakt with sangoma e1/t1 cards in here ? |
| 14:55.08 | Casper_UA | nix000: what's the prob? |
| 14:55.36 | nix000 | Casper_UA: i am actively looking for a t1/e1 card that supports ss7 thats all. digium does not seem to support it. |
| 14:56.22 | nix000 | Casper_UA: did you deployed them with asterisk ? |
| 14:57.27 | Casper_UA | never used sangoma in ss7 env |
| 14:57.41 | *** join/#asterisk jsolares (~jsolares@200.30.141.85) |
| 14:57.42 | Casper_UA | sangoma runs with * in "transparent" mode |
| 14:59.48 | *** join/#asterisk cbachman (~chatzilla@victory.ece.northwestern.edu) |
| 15:01.42 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-40-134.cust.bezeqint.net) |
| 15:02.11 | nix000 | Casper_UA: still there ? |
| 15:03.55 | Zaw | does anyone have an example extensions.conf and/or iax.conf snip for setting up an incoming 1-700 IAXTel number to ring a specific SIP phone connected to your asterisk server? |
| 15:07.29 | bjohnson | yes |
| 15:07.45 | bjohnson | iaxtel incoming example in the sample config files |
| 15:08.00 | bjohnson | to dial a specific sip phone use the dial command |
| 15:08.36 | Zaw | ok |
| 15:09.47 | bjohnson | iaxtel example just goes to default context |
| 15:10.24 | bjohnson | but default context in example don't dial a phone on an incoming call |
| 15:10.34 | bjohnson | add exten => s,1,wait(2) |
| 15:10.54 | bjohnson | exten => s,2,dial(SIP/2203) |
| 15:11.05 | bjohnson | (replace 2203 with the sip phone you want to ring |
| 15:11.34 | bjohnson | I think the superdial macro on the wiki has an example of this that incorporates voicemail usage too |
| 15:11.40 | *** join/#asterisk ctooley ([U2FsdGVkX@199.89.146.18) |
| 15:13.43 | Zaw | got it |
| 15:13.57 | nix000 | did anyone at digium noticed the need for interfacing with legacy tdm protocols like c7/ss7 ? |
| 15:14.45 | *** join/#asterisk fugitivo (~ajf@201.255.105.251) |
| 15:15.04 | eKo1 | Hmm...for some reason, even with AbsoluteTimeout set, the channel won't hang up. |
| 15:15.12 | eKo1 | Hmm.... |
| 15:15.33 | ctooley | How did the Dev Conference call go yesterday? |
| 15:16.05 | tzanger | loooooooooooong |
| 15:17.15 | *** join/#asterisk W1thdraw (~Withdraw@ip68-5-125-44.oc.oc.cox.net) |
| 15:17.19 | *** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net) |
| 15:17.53 | *** join/#asterisk W1thdraw (~Withdraw@ip68-5-125-44.oc.oc.cox.net) |
| 15:19.19 | Moc____ | HAIL |
| 15:21.52 | Hmmhesays | wow, trying to use php5 for agi is a pain |
| 15:22.19 | ManxPower | LOL! The USA Govt is saying that unless Amtrak (USA Passenger Rail) restructures, they won't get any money. |
| 15:22.20 | ctooley | tzanger, anything useful come of it? |
| 15:22.39 | Hmmhesays | i'd like to see amtrack get some money |
| 15:22.48 | Hmmhesays | the airlines get enough, why shouldn't the train |
| 15:22.55 | [TK]D-Fender | Hmmhesays : Whats the change that hurst in your transition? |
| 15:23.02 | ManxPower | Hmmhesays, I would like Amtrak to get 1% of the USA transportation tax. |
| 15:23.09 | ctooley | ManxPower, Well, it could be worse, we could divert that money to the airlines... oh wait.... |
| 15:23.19 | Hmmhesays | Fender: in reference to php? |
| 15:24.01 | ctooley | You know Amtrak wouldn't be so bad if they got enough money to buy the right of way on some of the tracks, it wouldn't take 3 days to take a train from Austin to Dallas |
| 15:24.17 | ManxPower | ctooley, Yeah! |
| 15:24.19 | [TK]D-Fender | Hmmhesays : yeah |
| 15:24.55 | ctooley | And with 802.11 network connectivity it actually would be a relatively productive way to travel. It takes an entire day to fly anywhere at this point anyway. |
| 15:24.56 | Hmmhesays | Fender: php5 doesn't change the working directory to where the script resides. so all of your non explicit paths no longer work |
| 15:25.03 | [TK]D-Fender | Hmmhesays : I've done a few small things with PHP for AGI but never seemed to work well in 4.3 |
| 15:25.07 | ManxPower | It used to take me 8 hours to get from Pensacola FL to New Orleans LA. It's a 3 hour drive. |
| 15:25.08 | tzanger | ctooley: yes |
| 15:25.46 | [TK]D-Fender | Hmmhesays : are you calling your scripte like "php thescript.php"? |
| 15:26.00 | Hmmhesays | asterisk is calling the script |
| 15:26.18 | [TK]D-Fender | can you show me the exact line? |
| 15:26.39 | Hmmhesays | it's not an error, it's how php 5 operates |
| 15:27.00 | [TK]D-Fender | hmm. |
| 15:27.05 | [TK]D-Fender | (no pun intended) |
| 15:27.14 | Hmmhesays | fopen('readme.txt', 'r'); doesn't return /var/lib/asterisk/agi-bin any longer |
| 15:27.23 | Hmmhesays | fopen('readme.txt', 'r'); doesn't return /var/lib/asterisk/agi-bin/readme.txt any longer |
| 15:28.15 | [TK]D-Fender | Can you use an env var to see where the script found itself and force a chdir? |
| 15:28.35 | Hmmhesays | probably, but what a pain in the ass |
| 15:28.44 | *** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || 1.0.6 Released || Dev Conf 1PM CST MARCH 10th -> IAX2/guest@66.250.68.194/996 || ClueCon Dev Conf Aug 3rd - 5th |
| 15:28.50 | Hmmhesays | there should be a command line switch |
| 15:29.33 | eKo1 | Say I make a call using SIP and it returns a 4XX message. Will it jump to priority n + 1? |
| 15:29.49 | ManxPower | I got an e-mail back from the immigration lawyer I contacted. Maybe I'll just sneak into Canada. |
| 15:30.07 | Hmmhesays | ManxPower: make sure to leave all your hookers and weed in the states though |
| 15:30.29 | eKo1 | Yes, Canada has enough of those to go around. |
| 15:30.31 | ManxPower | Hmmhesays, No need to take them. Easy enough to get them in Canada. |
| 15:30.39 | Hmmhesays | good point |
| 15:31.27 | Hmmhesays | i think i'll stick with php 4.3 for now |
| 15:31.36 | Hmmhesays | even though the classes are kind of farked up in it |
| 15:32.32 | Goshen | ok, I just got my IAXy in the mail...it didn't come with any documentation |
| 15:32.37 | Goshen | and there doesn't seem to be any one the wiki |
| 15:32.55 | ManxPower | Goshen, Correct. It was hard to build, it should be hard to use. |
| 15:33.09 | ManxPower | Goshen, Check out "iaxprov" from Digium CVS. |
| 15:33.13 | zipp | http://www.digium.com/downloads/product_sheets/IAXy.pdf |
| 15:33.21 | ManxPower | ~mailinglist |
| 15:33.22 | jbot | from memory, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. |
| 15:33.48 | zipp | no, that is not right, that is the info about it |
| 15:33.57 | ManxPower | Ugh. After using GNOME, KDE sucks. |
| 15:34.13 | Goshen | ~IAXy |
| 15:34.14 | jbot | from memory, iaxy is pronounced "Eeks-ee", a small ATA produced by Digium for using the IAX protocol in place of SIP. |
| 15:34.46 | zipp | ManxPower, I will agree w/ you |
| 15:35.09 | ManxPower | zipp, I'm trying out different LiveCDs |
| 15:35.19 | Goshen | ok, that pdf file is just a spec sheet.. |
| 15:35.25 | Goshen | I still don't have the ip address of the thing |
| 15:35.35 | Goshen | or a sample iax.conf |
| 15:35.37 | ManxPower | Goshen, That is correct. |
| 15:35.44 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 15:35.46 | ManxPower | iaxprov has much of the info you need. |
| 15:35.53 | Goshen | I don't use CVS, I installed from tarball |
| 15:36.05 | ManxPower | Goshen, Then you cannot use the IAXy. |
| 15:36.06 | Goshen | is the information not included with the stable distribution? |
| 15:36.25 | ManxPower | Goshen, The software to configure the IAXy is seperate. |
| 15:36.35 | Zeeek | I finally go the cojones to install 1.0.6 - for now it seems to be working |
| 15:36.49 | Zeeek | Goshen you have a new IAXy? |
| 15:37.00 | Goshen | 1.0.6 works fine for me, only thing that doesn't is music on hold when pressing the hold button on the phone |
| 15:37.08 | Goshen | yes, just came in the mail this morning |
| 15:37.10 | Zeeek | ah didn't try that yet |
| 15:37.16 | zipp | farfone: Low budget phone announced for Q1 2004 <-- now that is funny |
| 15:37.20 | Zeeek | and what is your rpoblem? cause I have one too |
| 15:37.21 | zipp | on the wiki |
| 15:37.21 | Goshen | it is plugged in, lights are on... |
| 15:37.32 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 15:37.34 | Zeeek | which lites? |
| 15:37.35 | Zeeek | all? |
| 15:37.46 | Goshen | my problem? I need to know how to log in to it...but I guess I have to download iaxprov form the CVS |
| 15:37.55 | Goshen | eithernet port lights |
| 15:38.05 | Goshen | when I pick up the phone, a light comes on inside |
| 15:38.13 | tzanger | Goshen: like the fridge |
| 15:38.18 | Goshen | :) |
| 15:38.28 | ManxPower | Goshen, Um, the IAXy is almost as dumb as a rock. It doesn't have a web interface, it does not have a telnet interface, it does not have an FTP interface. |
| 15:38.34 | Zeeek | true - hey ur rightn moh is broken! |
| 15:38.53 | Zeeek | Goshen you need to provision it as I'm sure Manx has been saying |
| 15:39.09 | Goshen | allrighty, have a command line for that iaxprov checkout? I think I have cvs installed |
| 15:39.14 | zipp | gnophone still is IAX1? |
| 15:39.15 | Goshen | although I usually use svn |
| 15:39.16 | ManxPower | Zeeek, The MoH was fixed in post 1.0.6 |
| 15:39.20 | Zeeek | shit |
| 15:39.30 | Zeeek | not that anyone cares who calls us, frankly |
| 15:39.43 | Zeeek | but I like to play with it |
| 15:39.47 | *** join/#asterisk search_learn2005 (~Miranda@209.68.139.150) |
| 15:39.50 | Zeeek | since I'm playing all my own moh |
| 15:39.58 | Goshen | moh works when you call it directly, or when transfering calls |
| 15:40.25 | search_learn2005 | Hi Everyone ! What is the difference between PRI and PSTN (POTS) ? |
| 15:40.25 | Zeeek | Goshen IIRC there was just a web or ftp download on digium.com for all the iaxyprov stuff |
| 15:40.33 | *** join/#asterisk B0ngFrOg (~wsmith@c-24-9-253-203.client.comcast.net) |
| 15:40.51 | Zeeek | I just tried it on a transfer and heard nada |
| 15:40.52 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com) |
| 15:40.52 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:41.57 | B0ngFrOg | anyone have an idea why I get 10 - 15 occurences of asterisk running when it is started from the init.d script??? |
| 15:42.09 | Zeeek | anyway, there is a lmot of stuff about provisioning the IAXy on the mailing list - recent stuff including a Windows program to so it if you need that |
| 15:42.15 | Goshen | aha! |
| 15:42.27 | tzafrir_laptop | B0ngFrOg, look at their PIDs. Any chance some of them are older copies? |
| 15:42.28 | Zeeek | I grabbed it but never tried it |
| 15:42.34 | jcims | is there a manager API (or CLI ) command i could use to determine if a sip client is on a call? i want to build a 'dashboard' service for our cisco phones that shows who's on a call.. |
| 15:42.35 | ManxPower | Zeeek, That was why I referred Goshen to the mailing list archives. |
| 15:42.46 | Goshen | http://www.digium.com/downloads/Iaxy_Installation_Guide.pdf |
| 15:42.52 | ManxPower | jcims, Anything wrong with Flash Operator Panel? |
| 15:42.54 | Zeeek | http://asterisk.gnuinter.net/files/digium/iaxyprov/ |
| 15:43.15 | jcims | no, we're actually using it now. it's just that having it on the phone would be more convenient for some of our managers |
| 15:43.18 | Zeeek | this is an example of incoherent |
| 15:43.19 | Goshen | I have been subscribed to the mailing list for two months, so I can search the last two months in my gmail :) |
| 15:43.37 | jcims | i was thinking of talking to op_server.pl to get the state rather than trying to go right to asterisk, but FOP may not be around forever |
| 15:43.50 | Zeeek | yeah me too and I have an iaxy filter so it's easier still |
| 15:43.53 | ManxPower | Does anyone have recommendations for LiveCDs w/GNOME? |
| 15:43.53 | *** join/#asterisk X-Gen (~x-gen@rrba-146-120-223.telkomadsl.co.za) |
| 15:44.24 | X-Gen | hey freaks© |
| 15:44.29 | Zeeek | Manx this post 1.0.6 moh fix - is there a patch? |
| 15:44.49 | ManxPower | Zeeek, No idea. Just pull down -r v1-0 |
| 15:45.00 | Zeeek | that isn't in the tarball? |
| 15:45.02 | B0ngFrOg | tzafrir_laptop - on boot up I get 10 - 15 processes running with conscutive pids |
| 15:45.03 | *** join/#asterisk coppice (~chatzilla@245.195.17.210.dyn.pacific.net.hk) |
| 15:45.23 | tzafrir_laptop | B0ngFrOg, then those are probably separate threads of the same process |
| 15:45.30 | tzafrir_laptop | What kernel version is it? |
| 15:46.04 | B0ngFrOg | tzafrir_laptop - 2.6.10 (fedora 3) |
| 15:46.18 | tzafrir_laptop | Do they have exactly the same memory sizes? If so: they're probably separate threads of the same process |
| 15:46.49 | tzafrir_laptop | 2.6.10's tools should be aware of threads... |
| 15:47.23 | tzafrir_laptop | Do you see all of them with top? |
| 15:47.32 | B0ngFrOg | tzafrir_laptop just a sec |
| 15:47.39 | tzafrir_laptop | try top -p pid1,pid2,... |
| 15:48.16 | eKo1 | B0ngFrOg: do a pstree |
| 15:48.20 | B0ngFrOg | tzafrir_laptop i see multiple lines for asterisk in top |
| 15:49.37 | *** join/#asterisk wizhippo (~wizhippo@Quebec-HSE-ppp233869.qc.sympatico.ca) |
| 15:50.31 | B0ngFrOg | tzafrir_laptop - in pstree i get asterisk - asterisk 14*[asterisk] |
| 15:51.21 | NirS | hey all |
| 15:51.28 | B0ngFrOg | tzafrir_laptop looks like safe_asterisk is starting 14 copies ??? or is this a diffrent behavior ( from 2.4.x kernels) |
| 15:51.29 | NirS | anyone has an idea how to use patlooptest ? |
| 15:52.30 | NirS | bong, it's not a question of the kernel |
| 15:52.36 | NirS | it's a question of procps |
| 15:52.51 | NirS | I have machines running kernel 2.4.x that show a single process |
| 15:52.57 | NirS | and others that show 14 processes |
| 15:53.05 | NirS | depending on the version of procps on the machine |
| 15:53.52 | NirS | helllllooooooo |
| 15:53.52 | NirS | anybody hommmmeeeeee ? |
| 15:53.53 | B0ngFrOg | NirS - OK ... I see so the util ps or /proc/ps displays threads or not ??? |
| 15:53.55 | *** join/#asterisk MikeJ[Jayden] (~ircatjerr@65.170.43.34) |
| 15:54.11 | NirS | ps isn't supposed to show threads |
| 15:54.17 | NirS | just processes, as far as I can recall |
| 15:54.30 | *** join/#asterisk styx2005 (~styx2005@a-line138.supra.net) |
| 15:54.40 | zipp | anyone know the debian pkgs for gtk-config and glib-config? |
| 15:54.43 | zipp | gastman wants them |
| 15:55.33 | bpoint | *sigh* smsq.c doesn't compile... :/ |
| 15:55.33 | *** join/#asterisk hemant (hemant@220.226.42.82) |
| 15:55.33 | search_learn2005 | Can people call me even if I don't sign up with Vonage or FreeWorld Dial , but use asterisk server to SBC PSTN |
| 15:55.33 | *** part/#asterisk styx2005 (~styx2005@a-line138.supra.net) |
| 15:55.41 | NirS | <<-- Mandrake Man, can't stand Debian and Slackware |
| 15:55.54 | bjohnson | B0ngFrOg: mine does that too. I thought it was normal |
| 15:56.00 | zipp | search_learn2005, yes |
| 15:56.03 | B0ngFrOg | NirS - on all of the other boxes I have only seen one instance of asterisk . This is the fire 2.6 and I was scratching my head... |
| 15:56.13 | NirS | sl2005, in general terms yes via direct IP2IP dial |
| 15:56.16 | bjohnson | ManxPower: I thought there was a gnommix |
| 15:56.27 | Zeeek | Manx since I didn't use cvs for 1.0.6 can I still use some kind of make option to get the newere version? |
| 15:56.55 | Goshen | I got the IAXy provisioned....but I must have something wrong in my iax.conf because I get this error |
| 15:56.55 | Goshen | *CLI> Mar 4 08:55:46 NOTICE[9065]: chan_iax2.c:3910 register_verify: No registration for peer 'goshen' (from 67.172.238.57) |
| 15:57.05 | bjohnson | search_learn2005: sure |
| 15:57.30 | Zeeek | Goshen you have [iaxyusername] in you iax.conf ? |
| 15:57.45 | Zeeek | same one as in the iaxyprov? |
| 15:57.56 | *** join/#asterisk LoRez (lorez@lorez.staff.freenode) |
| 15:58.09 | CleanerX | strange |
| 15:58.20 | CleanerX | i cannot create a sip channel within the same server |
| 15:58.24 | search_learn2005 | I want to use IAX because my asterisk server is behind a NAT, but I do not see any IAX phones, any ideas ? Is routing SIP through NAT very cumbersome? |
| 15:58.36 | *** part/#asterisk [Sim] (florian@clio.obsimref.com) |
| 15:58.38 | CleanerX | 16:55:50 NOTICE[22652]: app_dial.c:927 dial_exec_full: Unable to create channel of type 'SIP' (cause 3) |
| 15:58.48 | Goshen | http://pastebin.ca/6854 |
| 15:58.51 | Goshen | is my config |
| 15:58.59 | CleanerX | but calling from a sipphone works |
| 15:59.11 | Goshen | ok, username goes in the [] I will try that |
| 15:59.16 | CleanerX | == Everyone is busy/congested at this time (1:0/1/0) |
| 15:59.24 | CleanerX | but actually it's not |
| 15:59.33 | CleanerX | anyone got a clue for this? |
| 15:59.54 | Zeeek | Goshen and what's in iaxyprov.conf |
| 15:59.59 | CleanerX | a few days earlier everything worked fine |
| 16:00.49 | bjohnson | search_learn2005: sip through nat sometimes works, sometimes doesn't |
| 16:01.09 | Goshen | zeeek: http://pastebin.ca/6855 |
| 16:01.10 | bjohnson | search_learn2005: there are ways to make sip work if you put in a proxy |
| 16:01.23 | Goshen | Zeeek: that worked... |
| 16:01.46 | Goshen | I am in :) ring ring...thanks guys |
| 16:01.47 | bjohnson | search_learn2005: also, there are iax phones starting to be available to consumers now eezeephone.com has some |
| 16:01.48 | zipp | libgtk1.2-dev |
| 16:01.55 | Goshen | that was far less painless then I thought it would be |
| 16:02.03 | Zeeek | good! |
| 16:02.07 | Goshen | time to head to the clinic and test it out |
| 16:02.09 | Goshen | bye |
| 16:02.38 | search_learn2005 | bjjohnson: Which one would you prefer an IAX phone or an analog adapter IAXy from digium? |
| 16:04.21 | Zeeek | You didn't ask me, but I'm learning that each IAX2 device has very different usability |
| 16:04.38 | Zeeek | they aren't like SIP phones |
| 16:05.07 | ManxPower | I prefer an ATA. Then I can hook up my fave analog phone to it. Like my VTech cordless dual handset cordless phone. |
| 16:05.25 | Zeeek | except that there is only one IAX ATA, right? |
| 16:05.45 | Zeeek | no actually those chinese ones exist |
| 16:06.01 | Hmmhesays | ahh the chinese, always with the knock off hardware |
| 16:06.06 | Zeeek | iaxtalk.com says "all out voIP products talk IAX" |
| 16:06.25 | Zeeek | only what is it knowcked off? Since no others exist yet |
| 16:06.35 | ManxPower | Zeeek, eezeephone has basically the same phones in the USA |
| 16:06.45 | Zeeek | oh yea? And ATA? |
| 16:07.04 | *** part/#asterisk jcims (~jcims@cpe-69-135-121-57.columbus.rr.com) |
| 16:07.10 | Zeeek | I'm still trying to test all these phones |
| 16:07.47 | ManxPower | I didn't look for an ATA |
| 16:07.56 | X-Gen | i'm still getting these dam errors "PRI got event: HDLC Abort" ..and my card is not sharing an IRQ with anyone :( |
| 16:08.26 | ManxPower | X-Gen, You have an interrupt latency problem. Have you tried the stuff listed in the mailing lsit archives. |
| 16:08.52 | X-Gen | ManxPower: nope, what should the latency be 32 ? |
| 16:09.02 | search_learn2005 | Anybody who has a name of a good phone that uses IAX, or would you say get an analog IAX adapter? |
| 16:09.17 | ManxPower | X-Gen, You misunderstand. OTHER devices are locking interrrupts. IDE, RAID, Video, etc. |
| 16:09.51 | X-Gen | ManxPower: but my cards are on their OWN interupts, not shared with anyone |
| 16:09.56 | ManxPower | search_learn2005, Um, there's only 2 phones that use IAX (same company) and two ATAs that use IAX. It's not like there's a lot of stuff to evaluate. |
| 16:10.15 | ManxPower | X-Gen, So what? Any kernel driver can disable all other interrupts. |
| 16:10.47 | X-Gen | ManxPower: currently the latency is set to 32 ? should i fiddle with that ? |
| 16:11.08 | ManxPower | X-Gen, only after you've tried everything else other than replacing the motherboards. |
| 16:11.09 | *** join/#asterisk Cresl1n (~matt@216.207.245.23) |
| 16:11.15 | shaZwaz | ManxPower any idea how I can take control of the call after it has been dialed and do something else with it ? |
| 16:11.28 | search_learn2005 | MaxPower: What is the name of that company which has the IAXphones? Would you prefer those phones or an adapter if you were asked? |
| 16:11.49 | Zeeek | ManxPowerisn't this what was supposed to be done? : cvs update -r v1-0 asterisk |
| 16:11.59 | ManxPower | X-Gen, i.e. unmask ide interrupts, remove IDE RAID, enable IDE DMA, stop using graphics, etc. |
| 16:12.10 | ManxPower | Zeeek, Huh? |
| 16:12.20 | eKo1 | Hmm...128 postgresql procs. all belonging to asterisk... |
| 16:12.21 | Zeeek | to get the moh working in 1.0.6 |
| 16:12.36 | ManxPower | search_learn2005, I cannot recommend any IAX hardphone/ATA since I've not used them. |
| 16:12.37 | X-Gen | ManxPower: i disabled DMA and it seamed to get better :( |
| 16:12.41 | CleanerX | anybody knows what this is : http://pastebin.ca/6856 |
| 16:12.42 | ManxPower | Zeeek, Yes. |
| 16:12.47 | Zeeek | didn't do anything |
| 16:13.00 | Jas_Williams | Zeeek: make update in the src/asterisk dir should get the latest verion of Stable |
| 16:13.01 | ManxPower | Zeeek, Maybe it was zaptel |
| 16:13.07 | X-Gen | whats the switch for hdparam to unmask ide interupts please ? |
| 16:13.13 | search_learn2005 | MansPower: What is the name of the company who offers them then? |
| 16:13.16 | Zeeek | ah zaptel broke the moh? |
| 16:13.36 | ManxPower | X-Gen, Fuck you. There are a billion mailing list messages easily searchable via Google, there is also a man page for hdparm. |
| 16:13.40 | Jas_Williams | Zeeek: It was a change in sip.c |
| 16:13.47 | Zeeek | ok |
| 16:13.59 | ManxPower | Zeeek, I don't know. I ALWAYS update asterisk, zaptel, libpri, and asterisk-sounds at the same time. |
| 16:14.00 | X-Gen | ManxPower: hehehe, k, that was a cheap shot :P |
| 16:14.13 | Zeeek | I did all that today but with tarball |
| 16:14.25 | ManxPower | ~google site:lists.digium.com unmask interrupts |
| 16:14.30 | [TK]D-Fender | Feel the love..... :/ |
| 16:14.36 | Jas_Williams | Zeeek: The tarball amy well not be fixed yet |
| 16:14.37 | ManxPower | ~google site:lists.digium.com hdparm |
| 16:15.01 | Cresl1n | ~~ |
| 16:15.12 | Zeeek | you can't update a tarbell build with make update I guess (duh) |
| 16:15.23 | ManxPower | Zeeek, Oh! Of course you can't. |
| 16:15.41 | Zeeek | that's why I xwas surprised but blindly followed your advice :) |
| 16:16.02 | *** join/#asterisk JerJer (~jj@feth100-fw.fament.net) |
| 16:16.11 | ManxPower | Zeeek, Give me all your money. |
| 16:16.19 | Zeeek | I think they were talking about htis last night - that most projects generate a nightly tarball |
| 16:16.28 | ManxPower | Sorry, Zeeek, you should give me all your money. |
| 16:17.12 | ManxPower | "The money's on the dresser, Chocolate." |
| 16:17.35 | Zeeek | Usually I fall asleep in front of the TV with beautiful girls... lst night it was the dev conf |
| 16:17.54 | Zeeek | Tarrentino? |
| 16:17.59 | zipp | enlighten me |
| 16:18.08 | ManxPower | Zeeek, Gene Hackman in The Birdcage. |
| 16:18.09 | Zeeek | Quentin Crisp? |
| 16:18.17 | search_learn2005 | ManxPower: Can you give me the name of the company that offers IAX phones please? |
| 16:18.21 | Zeeek | Just saw it agin a few weeks ago - brilliant |
| 16:18.37 | zipp | search_learn2005, farfon has shipped beta units |
| 16:18.38 | Zeeek | One of my all time favorite movies |
| 16:18.43 | zipp | search_learn2005, digium sells iaxy ata's |
| 16:18.44 | Zeeek | zipp are you in a hurry? |
| 16:18.51 | Zeeek | I have one |
| 16:19.01 | zipp | Zeeek, in a hurry for what? |
| 16:19.06 | Zeeek | a farfon |
| 16:19.16 | zipp | I wouldn't mind having one :) |
| 16:19.20 | jsolares | a fartfon |
| 16:19.21 | Zeeek | that's why I have nothing left for ManxPower |
| 16:19.30 | Zeeek | ~lart fartfon |
| 16:19.44 | Zeeek | mine was hit by an anvil |
| 16:19.57 | Zeeek | allthough that is stupid - the anvil is hit by the hammer |
| 16:20.28 | Zeeek | so, to end this misery... does the tarball get updated like one a week or what? |
| 16:20.35 | Beirdo | yeah, you are supposed to drop anvils on things... Doesn't anyone know their Animaniacs? |
| 16:20.47 | Zeeek | drop, maybe, but ni hit |
| 16:21.13 | ManxPower | ~doc |
| 16:21.14 | jbot | i guess doc is the command is ~docs, moron. |
| 16:21.18 | ManxPower | ~docs |
| 16:21.19 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 16:21.21 | Beirdo | hehe |
| 16:21.31 | Beirdo | a bot with a 'tude |
| 16:21.39 | Zeeek | LOL |
| 16:21.42 | Jas_Williams | Zeeek: Here's the bug report may be manualy patch yours http://bugs.digium.com/bug_view_page.php?bug_id=0003701 |
| 16:21.49 | Zeeek | thanks Jas_ ! |
| 16:22.08 | Zeeek | I looked in bugs earlier but didn't see it |
| 16:23.16 | *** join/#asterisk mhess_ (~chatzilla@66.185.98.148) |
| 16:24.02 | zipp | http://pastebin.ca/6857 |
| 16:24.04 | MikeJ[Jayden] | thats not attitude... |
| 16:24.07 | MikeJ[Jayden] | ~rtfw |
| 16:24.08 | jbot | from memory, rtfw is Read That F*cking Wiki, the one at http://www.voip-info.org/tiki-index.php |
| 16:24.15 | MikeJ[Jayden] | that's attitude |
| 16:24.34 | zipp | anyone know what I need to install on debian to fix this: http://pastebin.ca/6857 |
| 16:24.42 | zipp | I have gdk-pixbuf-dev |
| 16:24.45 | zipp | still errors |
| 16:25.24 | Beirdo | hehe |
| 16:29.13 | ctooley | zipp, "updatedb && locate gdk-pixbuf.h" |
| 16:29.41 | *** join/#asterisk oej (~oej@216.65.237.25) |
| 16:30.15 | Zeeek | moh fixed, thanks folks - any other 1.0.6 surprises? (yes, I've scanned the ML) |
| 16:30.15 | ManxPower | ~google gdk-pixbuf.h rpm |
| 16:30.25 | MikeJ[Jayden] | ~troutslap Beirdo |
| 16:30.27 | jbot | ACTION slaps Beirdo around with a large trout |
| 16:30.34 | Beirdo | hehe |
| 16:30.51 | Beirdo | hey, that's what I do to people in #mythtv-users :) |
| 16:30.53 | X-Gen | ~logs |
| 16:30.54 | jbot | apt/ibot/jbot/purl all log to http://ibot.rikers.org/<channelname>/ where channelname is html encoded ie: %23debian | lines that start with a space are not shown | some channels have stats at http://ibot.rikers.org/stats/<channelname>.html.gz |
| 16:30.57 | Zeeek | ctooley were you on the conf last nite? I heard part of it |
| 16:31.38 | Zeeek | I missed one part I'm interested in now: WHEN is the tarball updated? |
| 16:32.08 | ManxPower | Zeeek, The tarball is NEVER updated. |
| 16:32.15 | ManxPower | There will be a 1.0.7 release at some point. |
| 16:32.22 | Zeeek | what good is it then? :) |
| 16:32.36 | jontow | reference point? |
| 16:32.40 | Zeeek | I don't undertsand why it isn't even updated. |
| 16:32.47 | Zeeek | even/ever/ |
| 16:33.07 | jsolares | it's already released, they would have to re release a 1.0.6.1 |
| 16:33.08 | Zeeek | why not apply fixes nightly |
| 16:33.09 | jontow | -r v1-0 is a moving target, even though its the stable branch.. its tagged as 1.0.X and a tarball is made, as a reference point |
| 16:33.26 | Zeeek | that isn't the way mozilla works, is it? |
| 16:33.32 | jontow | sure it is |
| 16:33.43 | jontow | binaries are only built as reference points |
| 16:33.49 | Zeeek | forthat matter, who cares... it wouldbe good if it were 1.0.3.18 |
| 16:33.50 | jontow | used to be milestones, now they're real releases |
| 16:33.50 | tzafrir_laptop | dpkg -S gdk-fixbuf.h || http://packages.debian.org/ |
| 16:34.20 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
| 16:34.31 | Zeeek | ok, well there's no good argument for staying with tarballs then. This is the first time I have used one for asterisk |
| 16:34.38 | jontow | cvs will get you exactly what you want, so do this: cvs -d :pserver:anoncvs@cvs.digium.com:/usr/cvsroot login ; cvs -d .... co -r v1-0 asterisk ; tar zcvf asterisk-r-v1-0.tar.gz asterisk/ |
| 16:34.52 | ManxPower | Zeeek, In theory the released tarballs would not have any major bugs. |
| 16:34.58 | jontow | that'll get you your nightly tarball.. cron it if you find it necessary ;) |
| 16:35.01 | Zeeek | except no moh? |
| 16:35.10 | tzafrir_laptop | Zeeek, the argument for using a tarball is when you need to redistribute... |
| 16:35.13 | ManxPower | HOWEVER, drumkilla is new at this sort of stuff and so testing is not done as well as it could be before release. |
| 16:35.29 | Zeeek | yeah, ok I'm getting it, slowly but surely |
| 16:35.41 | ManxPower | Maybe he should have had a 1.0.6rc1, 1.0.6rc2, etc. |
| 16:35.43 | tzafrir_laptop | Not to mention that a CVS client is not always available |
| 16:36.03 | ctooley | Zeeek, No, I wasn't on the conference, in the middle of this stupidly huge development project. |
| 16:36.03 | tzafrir_laptop | ManxPower, I figure that there are not enough testers. |
| 16:36.17 | Zeeek | funny thing about this - except for a few people in devel (well maybe not a few) the folks in prod all have different scenarios |
| 16:36.19 | X-Gen | what the maximum channels ever used in a * box ? ISDN channels ? 3Xquad pri cards ? |
| 16:36.25 | Zeeek | heh |
| 16:36.38 | Juggie | tzafrir, no there was no formal stable release policy |
| 16:36.46 | ctooley | X-Gen, 1 billion dollars... err... channels |
| 16:36.47 | Zeeek | ctooley I finally printed the latest version of the "book" |
| 16:36.47 | ManxPower | X-Gen, 8 T-1s |
| 16:36.49 | Juggie | but there is now, at least in principal |
| 16:37.13 | Zeeek | anyone here on that call last night? |
| 16:37.20 | Zeeek | it was odd to listen from a user pov |
| 16:37.23 | X-Gen | ManxPower: they use 2 quad cards ? |
| 16:37.28 | Zeeek | of course it was for dev |
| 16:37.38 | Zeeek | did you.... speak? |
| 16:37.43 | zipp | nope |
| 16:37.52 | Zeeek | was that you breathing hard at the end? |
| 16:37.59 | zipp | ha, nope |
| 16:38.08 | Zeeek | I listened for 90 whole minutes almost |
| 16:38.22 | zipp | I listened to it all |
| 16:38.22 | Zeeek | call quality was great |
| 16:38.27 | zipp | yes |
| 16:38.43 | Zeeek | considering that I'm 3000miles for the server |
| 16:38.45 | jsolares | what did they talk about? |
| 16:38.51 | Zeeek | politics |
| 16:38.59 | zipp | it was nice, using the iaxcli code from simpleclient in iaxclient :) |
| 16:39.01 | jsolares | and this was a dev conf? |
| 16:39.01 | ManxPower | X-Gen, Hmm? |
| 16:39.03 | jsolares | hehehe |
| 16:39.10 | Zeeek | of course |
| 16:39.18 | zipp | had a lot to do with growing pains, initially |
| 16:39.33 | Zeeek | mutiny on the Bounty! |
| 16:39.37 | Zeeek | ho ho ho |
| 16:39.38 | ManxPower | Of course, if you were on the right mailing list you would have seen a message last night or this morning from Paul Mhaler(sp!) saying he has had 500 calls. |
| 16:39.48 | X-Gen | ooh fek, i just broke my * box "ast_load_resource: /usr/lib/asterisk/modules/pbx_dundi.so: undefined symbol: pbx_substitute_variables_varshead" |
| 16:40.02 | hacim | Zeeek: book? |
| 16:40.14 | Zeeek | asteriskdocs.org downloadable PDF |
| 16:40.20 | ManxPower | X-Gen, Welcome to CVS-HEAD. Get on the asterisk-cvs mailing list if you use CVS. |
| 16:40.29 | Zeeek | The Reference - Whole Earth Guide to Asterisk |
| 16:40.41 | Zeeek | Includes coupns for 50% at cooperating spas |
| 16:40.52 | Hmmhesays | how long has res_perl been around? |
| 16:41.14 | Hmmhesays | and I never noticed |
| 16:41.27 | X-Gen | ManxPower: looks like i'm gonna haveto, i been lurking arround for long enuff |
| 16:42.05 | *** join/#asterisk Ahewes (~rsb@209.81.2.58) |
| 16:42.44 | MikeJ[Jayden] | yesterdays call went for about 10 hrs |
| 16:43.37 | tzanger | wow it's quiet today |
| 16:43.44 | tzanger | all the people talking yesterday are too tired :-0 |
| 16:43.51 | tzanger | it was a pretty good conf though |
| 16:44.01 | Hmmhesays | heh |
| 16:44.02 | jsolares | i'd say those 10hrs would've been better spent coding? hehe *ducks* |
| 16:44.03 | ManxPower | tzanger, Lots of lazy people have burned us out this morning |
| 16:44.06 | Hmmhesays | i was sick, i missed it |
| 16:44.07 | Zeeek | heh |
| 16:44.14 | tzanger | ManxPower: that sucks |
| 16:44.44 | MikeJ[Jayden] | did you see how much stuff that has been sitting forever either moved or hit cvs yesterday? |
| 16:44.46 | Hmmhesays | are they going forward with the plans from last weeks conf? |
| 16:45.00 | MikeJ[Jayden] | a good deal of that time was spent coding |
| 16:45.01 | coppice | Beirdo: nah. get something that does T.38 :-) |
| 16:45.16 | MikeJ[Jayden] | at least after about 5 or 6 CST |
| 16:45.16 | *** join/#asterisk algorithmn (~na@ool-18bce89c.dyn.optonline.net) |
| 16:45.22 | jsolares | hehe |
| 16:45.25 | MikeJ[Jayden] | KRAM has returned! |
| 16:45.27 | Beirdo | I'm not too concerned about faxing right now |
| 16:45.38 | tzanger | rooooooooooooooooooooooooot has ... ahem I mean kram has landed! |
| 16:45.38 | *** join/#asterisk G0shen (~Goshen@70-57-80-147.slkc.qwest.net) |
| 16:46.15 | *** join/#asterisk Yoda-BZH (~yoda-bzh@213.223.51.13) |
| 16:47.57 | Yoda-BZH | re |
| 16:48.08 | zipp | http://dev.asteriskdocs.org./ |
| 16:48.14 | zipp | ha |
| 16:48.19 | zipp | phpinfo(); |
| 16:48.28 | tzanger | php |
| 16:48.29 | tzanger | ewwwww |
| 16:48.32 | Zeeek | well at least they updated |
| 16:48.32 | tzanger | nastyass |
| 16:48.38 | *** join/#asterisk km- (~pgrace@brdgw1.rttx.com) |
| 16:50.09 | Zeeek | anyone having memory leak problems in asterisk 1.0.6? |
| 16:50.24 | Zeeek | it keeps on growing even with no calls |
| 16:50.51 | tzanger | get it on Atkins |
| 16:51.01 | tzanger | I find i don't grow without beer but man am I miserable |
| 16:51.14 | Zeeek | I wish I could have a beer rightnow |
| 16:52.15 | tzanger | heh |
| 16:54.06 | Zeeek | but the memory footprint of * with nothing much else running grows and grows |
| 16:56.10 | zipp | Zeeek, I am getting 1.0.6 now, will try it here |
| 16:56.20 | Zeeek | cvs? |
| 16:56.27 | MikeJ[Jayden] | aparently there is a MOH bug in 1.0.6?? |
| 16:56.31 | Zeeek | yes |
| 16:56.41 | Zeeek | small bug. No music |
| 16:56.47 | eKo1 | I downloaded it yesterday. I have experienced no memory leaks of any kind. |
| 16:57.00 | eKo1 | At least not yet. |
| 16:57.01 | Zeeek | well what is the normal behavior? |
| 16:57.11 | eKo1 | I had to restart today though. |
| 16:57.16 | Zeeek | thopugh use will vary, there must be some typical scenario |
| 16:57.21 | eKo1 | We'll see what happens be tomorrow. |
| 16:57.23 | zipp | Zeeek, cvs head |
| 16:57.25 | *** join/#asterisk dsfr (~dsfr@216.207.244.183) |
| 16:57.35 | *** join/#asterisk fantomax1 (~fanto@81.208.114.250) |
| 16:57.37 | fantomax1 | hi all |
| 16:57.52 | Zeeek | when I first started it is was around 67M used in RAM - now two hours later it's 134M |
| 16:58.03 | fantomax1 | can anyone tell me where I can find the content of an INVITE sent from * ? |
| 16:58.04 | Zeeek | almost no calls or any other activity |
| 16:58.20 | fantomax1 | is there any place in * I can see it ? |
| 16:58.34 | Zeeek | do you see it with sip debug on? |
| 16:58.53 | fantomax1 | how ? |
| 16:59.00 | fantomax1 | asterisk -vvvvr ? |
| 16:59.06 | *** join/#asterisk Grooby (~Grooby@12.22.232.212) |
| 16:59.06 | Zeeek | heh I said no calls, maybe it isn't showing or answering calls :) |
| 16:59.18 | Grooby | zeeek; you got spandsp to work right? |
| 16:59.19 | Zeeek | on CLI - sip debug |
| 16:59.23 | fantomax1 | ok |
| 16:59.27 | Zeeek | it si for now |
| 16:59.28 | fantomax1 | thanks |
| 16:59.38 | Zeeek | Grooby it did a few minutes ago |
| 16:59.43 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 16:59.55 | Grooby | let see if i can get it to work here |
| 17:00.02 | Zeeek | yeah check it out |
| 17:00.03 | Grooby | can you use it to receive fax over ulaw? |
| 17:00.13 | Zeeek | depends on a few things |
| 17:00.22 | Zeeek | some people have I think |
| 17:00.29 | Grooby | depends on how many virgin goats I have to sacrafice? |
| 17:00.46 | Zeeek | more the accuracy of the turning towards Huntsville |
| 17:00.58 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 17:01.07 | mishehu | boom. |
| 17:01.10 | zipp | bounty for t.38 is huge |
| 17:01.25 | jsolares | it makes me want to learn and code it |
| 17:01.37 | Grooby | mishehu, speex 1.1.7 |
| 17:02.03 | coppice | zipp: if you think the bounty for T,38 is huge, you must work for peanuts |
| 17:02.51 | spackle | I had spandsp working on 1.0.4 when I started manually, but when I started from init it would no longer work. weird. |
| 17:02.57 | jsolares | coppice: i do actually, the joys of living on the uncivilized world |
| 17:03.03 | Grooby | spackle, wierd |
| 17:03.07 | *** join/#asterisk Mearth (~Mearth@65.38.211.195) |
| 17:03.27 | CleanerX | anybody got a clue for this : http://pastebin.ca/6856 |
| 17:03.31 | *** join/#asterisk GiabboO (~GiabboOo@host101-246.pool8173.interbusiness.it) |
| 17:03.39 | GiabboO | hi all |
| 17:03.50 | *** join/#asterisk JerJer[mobile] (~jj@feth100-fw.fament.net) |
| 17:03.51 | CleanerX | I cannot create sip-channels |
| 17:04.12 | spackle | CleanerX, I can't either, I use Asterisk for that. |
| 17:04.19 | jsolares | hehehe |
| 17:04.22 | coppice | I have elements of T,38 working now, but there are so many buggy ATAs |
| 17:04.40 | Hmmhesays | anyone having trouble compiling the latest asterisk-addons ? |
| 17:04.48 | *** join/#asterisk steakhouse (~steakhous@mail.austingraphics.com) |
| 17:04.50 | Hmmhesays | app_addon_sql_mysql.c:164:64: macro "AST_LIST_REMOVE" requires 4 arguments, but only 3 given |
| 17:04.56 | zipp | coppice, 10K isn't big to you? |
| 17:05.05 | spackle | cleanerX, seriously, what's going on. |
| 17:05.25 | coppice | its only 4K |
| 17:05.26 | *** join/#asterisk Gerrath (~Gerrath@shanev.lifecor.com) |
| 17:05.35 | zipp | coppice, it is 10 |
| 17:05.36 | Zeeek | coppice what is the best version of spandsp for asterisk 1.0.6 please? |
| 17:05.41 | GiabboO | i setted up 2 iax users, [200] and [201], i create also an extension like this: exten => 2XX,1,Dial(IAX2,${EXTEN}) , why it cant reach 200@default when I dial 200 ? |
| 17:05.44 | coppice | or, rather 4k |
| 17:06.20 | |Vulture| | Zeeek: use latest not stable |
| 17:06.23 | coppice | Zeeek the last, which is spandsp-0.0.2pre10, but the patch for the makefile hasn't been updated yet |
| 17:06.23 | |Vulture| | urg sorry |
| 17:06.27 | |Vulture| | I was thinking speex |
| 17:06.37 | *** join/#asterisk Essobi (kstone@75.137.26.216.host.teledvance.com) |
| 17:06.39 | Zeeek | coppice thanks, I usually patch by hand |
| 17:06.46 | Essobi | Howdy coppice. |
| 17:07.09 | coppice | hi |
| 17:07.22 | Essobi | I prefer to hand patch. I feel better when I RTFS and know what's going on. |
| 17:07.30 | Grooby | w00t..compiled ok |
| 17:07.36 | steakhouse | question re: "IAXy prov" anyone know where I can find a copy of this app? |
| 17:07.38 | Grooby | how the hell do i know if the module's loaded |
| 17:07.42 | coppice | zipp: where did you get 10k from? wiki says 4k |
| 17:07.43 | Grooby | is there a CLI command to check? |
| 17:07.51 | *** part/#asterisk santiago (~santiago@201.245.167.78) |
| 17:07.56 | JerJer[mobile] | steakhouse: digium cvs |
| 17:07.57 | Essobi | Grooby show applications |
| 17:08.09 | spackle | steakhouse, it is in the CVS, follow the IAXY instructions at digium. |
| 17:08.29 | Grooby | i am in love |
| 17:08.31 | Grooby | see them |
| 17:08.50 | Essobi | I swear I had remote sip peers in my queues last week.. I don't know if I was just imaging it, or a cvs update broke the ability to have a remote SIP peer in as a static member in a queue. |
| 17:08.54 | spackle | steakhouse, you can also search the list archives, there is a windows-based provisioning tool somebody wrote that may be of interest to you. |
| 17:09.02 | steakhouse | sorry, I don't follow... cvs? |
| 17:09.16 | Essobi | spackle SIP provisioner? |
| 17:09.18 | Zeeek | http://asterisk.gnuinter.net/files/digium/iaxyprov/ |
| 17:09.21 | GiabboO | can anyone help me to set up a workin exten => for iax users ? |
| 17:09.34 | spackle | no, IAXY provisioning tool. |
| 17:09.42 | Essobi | What were you talking about then? |
| 17:09.51 | GiabboO | there is any problem on asterisk 0.9.0 using ${EXTEN} ? |
| 17:09.59 | steakhouse | thanks zeeek... sorry all, i'm noob ^-^ |
| 17:10.15 | JerJer[mobile] | steakhouse: then start reading |
| 17:10.24 | JerJer[mobile] | only after can you ask questions |
| 17:10.45 | Essobi | GiabboO exten,5551212,1,Dial(IAX/mynameisbob) <--- That's how I do it with SIP.. replacing IAX.. I havn't used IAX. |
| 17:10.49 | spackle | here is where the windows IAXY provisioning tool lives: http://dacosta.dynip.com/asterisk |
| 17:11.06 | GiabboO | exten, or exten => ? |
| 17:11.12 | GiabboO | im editing extensions.conf |
| 17:11.25 | km- | howdy! |
| 17:11.26 | tzanger | http://www.contentvulture.com/pictures/pc/20053411575.tetris-up.jpg |
| 17:11.27 | tzanger | hahahaha |
| 17:11.33 | *** join/#asterisk BuckRogers (~none@ool-18bce89c.dyn.optonline.net) |
| 17:11.37 | GiabboO | im tryin ${EXTEN} variable |
| 17:11.37 | steakhouse | heh... jer, i'm on it |
| 17:11.44 | algorithmn | realtime sip any1? |
| 17:11.49 | *** join/#asterisk ast_freak (~ast_freak@hades-out.universalsystems.net) |
| 17:11.52 | JerJer[mobile] | algorithmn: no |
| 17:11.58 | km- | tzanger: what's cookin? |
| 17:11.58 | BuckRogers | yeah i played with realtime |
| 17:12.01 | GiabboO | if i set up exten => 200,1,Dial(IAX2/200) it work |
| 17:12.05 | tzanger | km-: nada u |
| 17:12.10 | BuckRogers | but i havent gotten it working properly |
| 17:12.11 | algorithmn | JerJer[mobile]: its nice to know your listening.. |
| 17:12.16 | GiabboO | when i use ${EXTEN} nothing work |
| 17:12.26 | BuckRogers | mysql is not my fortay |
| 17:12.27 | km- | tzanger: same shit different day -- I never got back to you on sunday about figuring out this winking problem -- I apologize, I got hornswaggled into family life |
| 17:12.41 | JerJer[mobile] | realtime is absolutely not the answer |
| 17:12.44 | tzanger | no worries km- |
| 17:12.44 | BuckRogers | any one else here play with realtime |
| 17:12.48 | BuckRogers | they say its the shit |
| 17:12.55 | tzanger | I'm just going for lunch, ttyia |
| 17:12.56 | tzanger | b |
| 17:12.56 | JerJer[mobile] | more like just 'shit' |
| 17:13.00 | algorithmn | JerJer: no, but its a good start, i say |
| 17:13.02 | km- | tzanger: later dude! |
| 17:13.07 | km- | I need some munchies myself |
| 17:13.09 | JerJer[mobile] | algorithmn: no its not |
| 17:13.25 | BuckRogers | y do u say this Jer Jer |
| 17:13.27 | algorithmn | JerJer: please explain your opinion if you may |
| 17:13.41 | BuckRogers | do u not understand it as i dont |
| 17:13.54 | JerJer[mobile] | why force asterisk to depend on a database? |
| 17:13.59 | algorithmn | BuckRogers: quite noble honesty |
| 17:14.27 | algorithmn | JerJer: asterisk<->backend<->database<->web interface |
| 17:14.32 | JerJer[mobile] | then do that |
| 17:14.41 | tzanger | JerJer[mobile]: I don't agree with making * depend on a db for anything, but having CDRs go out to a DB is very handy |
| 17:14.43 | JerJer[mobile] | but don't FORCE asterisk to depend on the database to be there |
| 17:14.46 | Zeeek | JerJer scalability I guess - personally I don't want no stinking databases |
| 17:14.51 | JerJer[mobile] | we use a database all day long |
| 17:15.02 | JerJer[mobile] | but our asterisk systems have no clue about it |
| 17:15.07 | algorithmn | JerJer: too late, just looking for others to talk about its potential applications |
| 17:15.07 | BuckRogers | really JerJer how so |
| 17:15.09 | JerJer[mobile] | and if the db dies, asterisk doesn't care |
| 17:15.15 | tzanger | JerJer[mobile]: how do you put CDR into DB reliably |
| 17:15.19 | tzanger | just rotate and inject? |
| 17:15.26 | BuckRogers | why would a MYSql data base "die" |
| 17:15.30 | JerJer[mobile] | tzanger: trival - my own cdr backend |
| 17:15.32 | BuckRogers | <PROTECTED> |
| 17:15.34 | tzanger | BuckRogers: BWAAAAAAAAAAHAHAHAHAHAHHAHAHAHA |
| 17:15.34 | Zeeek | they do sometimes |
| 17:15.35 | *** join/#asterisk pif (ldm@zenon.apartia.fr) |
| 17:15.38 | JerJer[mobile] | hard drive failure |
| 17:15.43 | JerJer[mobile] | power supply failure |
| 17:15.44 | algorithmn | BuckRogers: you smart arse |
| 17:15.45 | JerJer[mobile] | AC failure |
| 17:15.54 | km- | when configuring sip.conf |
| 17:15.55 | tzanger | MySQL data consistency problems |
| 17:15.55 | JerJer[mobile] | innodb corruption |
| 17:15.56 | km- | I have to do |
| 17:15.58 | Zeeek | heart attack |
| 17:15.58 | BuckRogers | wouldnt asterisk die too then? |
| 17:16.03 | km- | disallow=all\nallow=gsm\nallow=ilbc |
| 17:16.05 | km- | instead of |
| 17:16.06 | outtolunc | operator error <G> |
| 17:16.11 | km- | disallow=all\nallow=gsm,ilbc |
| 17:16.13 | km- | right? |
| 17:16.13 | Zeeek | asterisk has very low cholerterol |
| 17:16.14 | tzanger | JerJer[mobile]: ahh okay but it doesn't block if it can't hit the db |
| 17:16.21 | JerJer[mobile] | BuckRogers: if asterisk depended in that database, most certainly |
| 17:16.25 | tzanger | Zeeek: true, but it's triglycerides are way up |
| 17:16.39 | Zeeek | too many sweet drinks |
| 17:16.46 | BuckRogers | but if they are both running on the same machine? |
| 17:16.54 | spackle | JerJer wants telecom style availability. |
| 17:16.58 | tzanger | BuckRogers: now why would you do something silly like that |
| 17:17.04 | tzanger | anyway lunchtime |
| 17:17.05 | tzanger | later all |
| 17:17.07 | Zeeek | yeah besides, don't do you billing on the asterisk box |
| 17:17.08 | BuckRogers | y not |
| 17:17.14 | Zeeek | (my motto) |
| 17:17.39 | Zeeek | ask accounting to buy their own box |
| 17:18.21 | algorithmn | BuckRogers: hows the gnu project comin along |
| 17:18.51 | BuckRogers | algorithmn: Ehh its comming haveing some issues with mysql |
| 17:19.10 | BuckRogers | besides that fine and dandy |
| 17:19.28 | algorithmn | i'd still like to help.. but i still would want the case of jim beam |
| 17:19.28 | BuckRogers | its parcailly up and moving |
| 17:19.41 | BuckRogers | hey man if thats ur price u got it |
| 17:19.44 | algorithmn | for my services |
| 17:19.52 | BuckRogers | im sure accounting will part with the funds |
| 17:19.57 | algorithmn | beyond the price... so i know you appreciate my help |
| 17:20.16 | BuckRogers | hell pecker yeah |
| 17:20.17 | BuckRogers | ! |
| 17:20.42 | algorithmn | call my cell later today, i've got a hang over to nurse till then |
| 17:20.49 | BuckRogers | he's a buggy cat but algorithmn knows his stuff |
| 17:20.59 | BuckRogers | u got it |
| 17:21.16 | algorithmn | as long as my code aint buggy.. |
| 17:21.37 | BuckRogers | buggy "i"eeeeeee |
| 17:21.44 | BuckRogers | with the fade away |
| 17:22.04 | algorithmn | if you get me some red label.. there may be some inherent flaws in your system.. |
| 17:22.27 | BuckRogers | yeah u want black or blue |
| 17:22.36 | algorithmn | yes sir |
| 17:22.37 | BuckRogers | if blue i got you for the weekend |
| 17:22.43 | BuckRogers | <PROTECTED> |
| 17:22.45 | CleanerX | spackle, sorry got a call |
| 17:22.53 | CleanerX | are you still there |
| 17:23.16 | spackle | Still here. |
| 17:23.25 | km- | what is the best full-featured IAX softclient available? |
| 17:23.34 | CleanerX | have you read the pastebin stuff? |
| 17:23.36 | km- | iaxcomm?> |
| 17:23.48 | jontow | i like diax, but i don't know if i'd call it full-featured by any means :) |
| 17:23.50 | spackle | CleanerX, no have you got stuff in there? |
| 17:23.50 | zipp | km-, look in the simpleclient dir of iaxclient |
| 17:24.01 | CleanerX | http://pastebin.ca/6856 |
| 17:24.12 | km- | zipp: the simpleclient works best? gotcha |
| 17:24.32 | zipp | km-, tkphone, iaxcli, ... |
| 17:24.36 | coppice | zipp: I had a look around. all I can see is 4k on offer for T.38 |
| 17:24.39 | CleanerX | * won't open up a channel for sip |
| 17:24.49 | zipp | coppice, me too, just 4K |
| 17:24.59 | Essobi | Anyone have static queue members in queue.conf that are statically dialed numbers on a peer? |
| 17:25.00 | CleanerX | i can place calls from sip phone -> zap, iax , whatever |
| 17:25.04 | zipp | yesterday someone was sure it was 10, I guess I had misinformation... |
| 17:25.10 | spackle | CleanerX, I think maybe a look at your sip.conf and dialplan would be more helpful. |
| 17:25.15 | km- | zipp: yah, I'm looking for the best one to roll out to the coworkers |
| 17:25.17 | CleanerX | can't figure out what the problem is |
| 17:25.19 | Essobi | I swear I was dialing them before.. |
| 17:25.26 | zipp | km-, ah |
| 17:25.28 | coppice | zipp: some put it up to 8k for a little while, and then withdrew |
| 17:25.43 | CleanerX | exten => _94,1,Dial(Sip/aunten,20) |
| 17:25.46 | zipp | km-, iaxcomm |
| 17:25.56 | CleanerX | for problem reduction that's all |
| 17:25.57 | zipp | coppice, are you working on something? |
| 17:26.07 | coppice | zipp: yep |
| 17:26.10 | zipp | I have a spa2100 and can send you data if you need |
| 17:26.11 | CleanerX | i'm using ast_data for user management |
| 17:26.15 | bkw_ | WASABI ASTERISK FOLK!!! |
| 17:26.23 | km- | zipp: yeah, everyone here loves asterisk, now I just gotta get them voipified |
| 17:26.25 | km- | bkw: MOOSE. |
| 17:26.27 | coppice | zipp spa2100 doesn't do t.38 |
| 17:26.29 | Essobi | WAAASABI? |
| 17:26.30 | Essobi | :) |
| 17:26.31 | bkw_ | mk-: PENIS. |
| 17:26.34 | km- | hehehe |
| 17:26.51 | algorithmn | what up mr west |
| 17:26.56 | km- | hehe, we're having a confrence call stress test this afternoon |
| 17:26.59 | G0shen | coppice: I read that spa2100 did do T.38 |
| 17:27.01 | km- | I e-mailed the entire company |
| 17:27.08 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
| 17:27.17 | G0shen | the spa2000 doesn't do T.38 |
| 17:27.23 | km- | I'm hoping we saturate the voice T1 lines and it still sounds good |
| 17:27.29 | zipp | http://www.voip-info.org/wiki-SPA-2100 |
| 17:27.32 | zipp | the wiki says it does |
| 17:27.37 | km- | Hey, is the farfon ready for sale? |
| 17:27.42 | bkw_ | ok |
| 17:27.43 | zipp | km-, nope |
| 17:27.56 | PBXtech | can asterisk to a SQL query from the extension.conf file? does it need an agi script to do this? |
| 17:28.02 | bkw_ | everyone take note that we pushed back ClueCon to Aug 3rd to the 5th |
| 17:28.10 | Essobi | Hey coppice.. I got a question.. you might know off the top of your head.. or anyone else for that matter.. I got a PBx I was to hand a PRI to.. I believe it's 4ESS.. does the zaptal/* drivers support sending caller-ID name in addition to number across the PRIs? If not, do any protocols other then zaptel protocls support it? |
| 17:28.15 | G0shen | www.iaxtalk.com has IAX hardphones, there is another company too I believe |
| 17:28.23 | Essobi | PBXtech see cmd mysql on voip-info |
| 17:28.27 | Essobi | I'm doing it now |
| 17:28.30 | bkw_ | ewww mysql |
| 17:28.33 | bkw_ | get a real database |
| 17:28.35 | bkw_ | :P |
| 17:28.37 | Essobi | Mehe. |
| 17:28.45 | Essobi | It's "REAL" enough for me. |
| 17:28.45 | bkw_ | mysql is ok.... |
| 17:28.46 | zipp | G0shen, shipping per phone is 30 dollars, they are just PA168 chips |
| 17:28.49 | *** join/#asterisk coppice (~chatzilla@245.195.17.210.dyn.pacific.net.hk) |
| 17:28.54 | bkw_ | I use it for small projects that i'm workin on |
| 17:28.59 | Essobi | D'oh.. coppice left. |
| 17:29.05 | Essobi | bkw_ I'm warehousing in mysql |
| 17:29.18 | Essobi | Hey coppice.. I got a question.. you might know off the top of your head.. or anyone else for that matter.. I got a PBx I was to hand a PRI to.. I believe it's 4ESS.. does the zaptal/* drivers support sending caller-ID name in addition to number across the PRIs? If not, do any protocols other then zaptel protocls support it? |
| 17:29.19 | km- | have the budgettones gotten any better lately? |
| 17:29.22 | bkw_ | no |
| 17:29.25 | bkw_ | CRAPSTREAM |
| 17:29.25 | Essobi | maha |
| 17:29.34 | Essobi | They are called budget for a reason. |
| 17:29.39 | km- | haahaha |
| 17:29.42 | MikeJ[Jayden] | anyone have a te110p card out there? |
| 17:29.44 | zipp | km-, they are fine if you have them on the same lan /24 as asterisk, then have asterisk IAX2 out |
| 17:29.46 | Zeeek | very good phone. Very good |
| 17:29.58 | km- | zipp: the phone supports IAX2 now? |
| 17:30.04 | Zeeek | nothing better for occasional use |
| 17:30.07 | MikeJ[Jayden] | hey bkw... wassup |
| 17:30.12 | zipp | km-, SIP to asterisk, then asterisk does IAX2 |
| 17:30.16 | jsolares | i have my budgetone connected remotely to asterisk behind nat, it works :) |
| 17:30.25 | km- | zipp: oh. We have T1's, so I dont need to use voip out |
| 17:30.26 | zipp | jsolares, stun? |
| 17:30.37 | Zeeek | jsolares me too. Wirks very well, very well indeed |
| 17:30.38 | zipp | km-, if you are on the same /24, just use sip |
| 17:30.39 | jsolares | well no |
| 17:30.41 | jsolares | no stun |
| 17:30.45 | zipp | what then? |
| 17:30.47 | spackle | I have budgetone over VPN and it works OK too. Prefer the new sipura phones though. |
| 17:30.51 | Zeeek | no STUN here either |
| 17:30.51 | jsolares | nfi, it just works XD |
| 17:30.54 | zipp | I have setup ipip tunnels and mine works fine |
| 17:31.06 | coppice | Essobi: I'm not sure. some of the pri protocols support name, but i'm not sure which off hand |
| 17:31.14 | km- | I'd love to put cisco 7940's on everyone's desks, but they're $$$ |
| 17:31.27 | zipp | coppice, the spa2100 does support t.38, according to them |
| 17:31.29 | G0shen | I love my new IAXy, just plug it in anywhere that has dhcp, and it works |
| 17:31.29 | Essobi | coppice Cool.. |
| 17:31.34 | MikeJ[Jayden] | 5ess and NI2 off the top of my head support name |
| 17:31.42 | G0shen | I have it behind a double(might be tripple) nat right now |
| 17:31.43 | zipp | G0shen, yes, I wish I could get a wifi phone like that :0 |
| 17:31.46 | Essobi | I got a guy with a pbx that wants me to hand him a pri. |
| 17:31.54 | G0shen | IAXy with wireless bridge :) |
| 17:31.57 | coppice | zipp: Sipura in big characters say T.38, and in rather small characters say pending |
| 17:32.03 | Essobi | Mike UNF.. I suppose * ain't supporting either yet. |
| 17:32.04 | G0shen | It would be like carrying around one of those old brick cellphones :) |
| 17:32.04 | Essobi | :| |
| 17:32.04 | fantomax1 | hi again .. another question... a 488 response to an INVITE to *, w |
| 17:32.04 | jsolares | lol |
| 17:32.05 | MikeJ[Jayden] | do iaxy's support poe? |
| 17:32.08 | zipp | coppice, ah |
| 17:32.13 | JerJer[mobile] | MikeJ[Jayden]: no |
| 17:32.24 | Essobi | TE cards can only handle 4ess thus far. |
| 17:32.25 | MikeJ[Jayden] | :( |
| 17:32.25 | fantomax1 | 488 Response to INVITE, What is the meaning ? |
| 17:32.40 | Zeeek | I just received an absolutely perfect 3 page international fax with spandsp and the tones sounded like shit. Bravissimo coppice! |
| 17:32.42 | zipp | MikeJ[Jayden], they are 12vdc, wouldn't be hard to do |
| 17:32.45 | km- | do the grandstream phones have speakerphone? |
| 17:32.50 | Zeeek | km ya |
| 17:32.52 | G0shen | Yes |
| 17:32.53 | MikeJ[Jayden] | I need an IAXy, w/ wireless bridge, built in power supply, and phone.. :) |
| 17:32.59 | G0shen | the speakerphone works really well |
| 17:33.04 | km- | cool. |
| 17:33.07 | G0shen | can turn it up really loud |
| 17:33.09 | Zeeek | very, very good speakerphone |
| 17:33.12 | MikeJ[Jayden] | or a wireless 802.11x IAXY phone would be cool |
| 17:33.13 | km- | I need to get a phone that has speakerphone |
| 17:33.14 | zipp | MikeJ[Jayden], me too |
| 17:33.23 | G0shen | and the message waiting indicator works too...flashes the screen |
| 17:33.24 | Essobi | I love my 7960s. |
| 17:33.27 | zipp | km-, sounds like budgetone would work for you |
| 17:33.27 | coppice | 488 usually means "I don't know how to handle that" |
| 17:33.32 | Essobi | I should take a picture for you guys |
| 17:33.35 | G0shen | km: they have black grandstream, which look better |
| 17:33.46 | Zeeek | yeah at night I sometimes wake up and think I'm living near a bar with a flashing neon sign |
| 17:33.46 | G0shen | IMHO |
| 17:33.48 | Essobi | I got 5 7960's hanging on the wall behind me. |
| 17:33.49 | Essobi | :) |
| 17:33.50 | MikeJ[Jayden] | I am going to go make one... my cordless phone will be as big as my head, and way 8 pounds including batteries |
| 17:34.04 | Essobi | maha |
| 17:34.09 | MikeJ[Jayden] | but the duct tape exterior will be so shieke |
| 17:34.22 | km- | essobi: why have 5 hanging there when you can have 1 with five lines into it? :) |
| 17:34.24 | roamer323 | Mike - sell the plan to Nokia afterwards :-) |
| 17:34.28 | spackle | fiberglass it to a motorcycle helmet! |
| 17:34.33 | Essobi | Zeeek What version you running of span DSP? |
| 17:34.36 | Zeeek | coppice that is the most beautifly fax I've ever seen. |
| 17:34.47 | Essobi | km- 4 different softswitch platforms and my deskphone. :) |
| 17:34.50 | Zeeek | now to try the bad machine on it.... |
| 17:34.55 | km- | essobi: nice! |
| 17:35.06 | Zeeek | spanDSP .2pre10 |
| 17:35.12 | Essobi | on -head? |
| 17:35.15 | *** join/#asterisk JohnnyC (~JoaoCorre@81.193.116.63) |
| 17:35.16 | Zeeek | 1.0.6 |
| 17:35.20 | Essobi | Damn. |
| 17:35.22 | Zeeek | with the moh patch |
| 17:35.26 | Essobi | I'm running -head still. |
| 17:35.35 | Zeeek | may work, ask him |
| 17:35.38 | JohnnyC | Anyone knows why using a PC micro the audio quality is poor. I hear echo from the room |
| 17:35.53 | JohnnyC | thing that doesnt happen when I'm using a normal phone |
| 17:36.00 | JohnnyC | but using the PC micro its bad |
| 17:36.22 | Essobi | That's just silly.. why can I put a member in queues that are SIP/EXT@PEERNAME but not a member that is SIP/PEER/NUMBER, like when I use Dial? |
| 17:36.22 | Zeeek | does he PC mic suck by any chance? |
| 17:36.23 | *** join/#asterisk dahunter (~joe@lsanca1-ar8-4-60-068-194.lsanca1.dsl-verizon.net) |
| 17:36.33 | Essobi | Has peer/number been/being depreciated? |
| 17:36.36 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
| 17:36.55 | roamer323 | johnnyC - depends on your soundcard and headset (if no headset - u're asking for trouble) |
| 17:37.21 | JohnnyC | roamer323: headset ... hmm I see |
| 17:37.46 | JohnnyC | its a portable PC . The audio board is nothing special |
| 17:38.01 | JohnnyC | This wouldn't happen on a HARDWARE ip phone right ? |
| 17:38.02 | Zeeek | headset or just mic? |
| 17:38.11 | JohnnyC | just mic |
| 17:38.15 | Zeeek | sucks |
| 17:38.19 | Zeeek | get a headset |
| 17:38.24 | km- | if your headset volume is high, and your mic gain is high |
| 17:38.25 | Zeeek | $20 Radio Shack |
| 17:38.38 | Zeeek | not even $20 |
| 17:38.40 | km- | you'll pickup the sound from your headphone in the mic |
| 17:38.56 | JohnnyC | thats bad |
| 17:38.57 | km- | USB headsets work wonders if you're using windows clients like X-Lite |
| 17:39.20 | JohnnyC | USB headsets are compatible with MAC ? |
| 17:39.26 | JohnnyC | can you give me an example ? |
| 17:39.30 | km- | I don't know |
| 17:39.33 | km- | I mentioned windows, not mac |
| 17:39.38 | km- | my guess is it'd be fine though. |
| 17:40.09 | Zeeek | the original question, no a hardphone won't do that. |
| 17:40.32 | km- | Zeeek's right, when you've actually got a handset those sorts of things dont happen |
| 17:40.33 | *** join/#asterisk nextime (~nextime@danex.i-m-c.it) |
| 17:40.36 | Zeeek | If you do want to use a laptop and have a software phone for it that you like, just use a headset - nodrivers needed |
| 17:40.50 | Zeeek | I'm not a big fan of USB |
| 17:40.57 | JohnnyC | hmm ok |
| 17:40.59 | JohnnyC | thanks guys |
| 17:41.12 | km- | I used to be like that, but the nice thing about USB is that you can just have your phone conversation through the headset |
| 17:41.23 | km- | and tell windows to use the normal audio board for system sounds and the like |
| 17:41.44 | km- | if you just use a headset, you hear the "bongs, beeps and chirps" through the headset with your phone conversation |
| 17:42.01 | km- | USB headset eliminates that problem |
| 17:43.47 | roamer323 | hey Zeeek - is sandsp mainly a "receive fax on an * extension" solution? |
| 17:44.21 | *** join/#asterisk Nebukadneza (~daddel9@i3ED6E5C0.versanet.de) |
| 17:44.28 | Nebukadneza | hi ... |
| 17:44.30 | ManxPower | roamer323, spandsp is a software fax data DSP library. The rx_fax and tx_fax apps use that library to send and receive faxes. |
| 17:44.33 | *** join/#asterisk TheEmperor (TheEmperor@218.111.49.254) |
| 17:44.43 | Zeeek | I'm trying on the machine that never worked before |
| 17:44.50 | Zeeek | in a few moments we'll know |
| 17:44.52 | Zeeek | muhahahaha |
| 17:45.07 | km- | I'd like to play with spandsp |
| 17:45.08 | Nebukadneza | i got a small problem with my sip.conf ... when someone trys calling phone1@my-ip (phone1 is a type=friend in my sip.conf) i get the following error on the asterisk cli: http://rafb.net/paste/results/RbvbUC68.html |
| 17:45.13 | km- | when I first was fiddling with it it crashed asterisk a lot though |
| 17:45.19 | TheEmperor | hi, can anyone recommend a good ssh program besides putty? |
| 17:45.33 | km- | why not putty? |
| 17:45.34 | roamer323 | ManxPower - what sort of load on a PC, as compared to say G.729 ? |
| 17:45.38 | coppice | km- : sort out yout libtiff |
| 17:45.46 | km- | coppice: yeah, it was a while ago too |
| 17:45.51 | TheEmperor | km-: just wondering..i can't seem to log into my server remotely anymore |
| 17:45.52 | Zeeek | coppice you once said I could fax you - is that offer still on? |
| 17:46.03 | km- | theemperor: putty crashes when you try to login? |
| 17:46.08 | TheEmperor | km- am wondering if it is putty or my server |
| 17:46.15 | coppice | Zeeek: not right now |
| 17:46.19 | Zeeek | cause this softfax still won't send to spandsp |
| 17:46.31 | Zeeek | and I'm not at the office so can't do much tracing |
| 17:46.50 | TheEmperor | km- no, it doesn't even bring up the login screen.. |
| 17:46.51 | Zeeek | coppice if you have a number and a day and time, I'd like to do it |
| 17:46.51 | spackle | TheEmperor, did you check what version of SSH you are set to? |
| 17:47.03 | TheEmperor | spackle: you mean on my * machine? |
| 17:47.07 | nestAr | man this "why should i answer a newbie question" thread on the users list is annoying.. |
| 17:47.16 | nestAr | welcome to my filter file! |
| 17:47.19 | spackle | On asterisk and on putty. |
| 17:47.23 | Nebukadneza | hehe |
| 17:47.23 | km- | theemperor: try telnet <ip> 22 |
| 17:47.24 | Zeeek | yes, but it happens every few months |
| 17:47.36 | TheEmperor | km- ok will try that.. |
| 17:47.38 | km- | theemperor: if you see a banner advertising SSH, then the problem is with putty configuration |
| 17:47.47 | km- | theemperor: if you dont get the banner, the problem is on your server |
| 17:48.35 | ManxPower | Actually, putty does not display a banner. |
| 17:48.44 | TheEmperor | km-network error:connection timed out |
| 17:48.51 | ManxPower | Come to think of it I believe putty displays the banner AFTER you login. |
| 17:49.10 | TheEmperor | never had this problem before.. |
| 17:49.12 | PBXtech | is there a command to kill an entire meet-me bridge? |
| 17:49.13 | km- | theemperor: so you've got a problem with your firewall, your nat settings, or your PC just doesnt have ssh turned on? |
| 17:49.20 | ManxPower | TheEmperor, you don't have sshd running, or you have a NAT problem, or you have a firewall problem. |
| 17:49.21 | km- | pbxtech: shutdown now? :) |
| 17:49.26 | TheEmperor | km-maybe.. |
| 17:49.26 | PBXtech | heh |
| 17:49.36 | nestAr | rofl |
| 17:49.46 | km- | manxpower: beer! :) |
| 17:49.51 | TheEmperor | ManxPower:the funny thing is, when I am in the office it works |
| 17:49.53 | km- | manxpower: I told him to look at the banner in a raw telnet session |
| 17:50.00 | TheEmperor | when I am at home, I can't log in remotely |
| 17:50.07 | nestAr | sounds like a firewall problem |
| 17:50.08 | TheEmperor | although it did work before |
| 17:50.09 | km- | sounds like a firewall problem |
| 17:50.11 | km- | DAMN YOU |
| 17:50.15 | nestAr | :eek3: |
| 17:50.19 | km- | hahaha |
| 17:50.25 | TheEmperor | on my * machine, no firewall.. |
| 17:50.28 | TheEmperor | just snort |
| 17:50.32 | Inv_arp | ok want to setup normal call conferencing , where does on start? |
| 17:50.53 | *** join/#asterisk mbranca (~matteo@81.208.92.210) |
| 17:50.57 | km- | bbiab, I gotta find some lunch |
| 17:51.32 | Inv_arp | TheEmperor: no layer2 firewalls after *? check with local admin |
| 17:52.06 | wizhippo | Inv_arp: meetme |
| 17:52.10 | TheEmperor | Inv_arp: funny thing is it did work before.. |
| 17:52.10 | TheEmperor | i guess it's where the server is |
| 17:52.26 | JerJer[mobile] | the gov't just censored your VoIP |
| 17:52.55 | Inv_arp | wizhippo: ahh so meetme is then |
| 17:53.01 | TheEmperor | my sip phones still work though.. |
| 17:53.15 | *** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 17:53.39 | AgiNamu | Has anyone here licensed FireFly? |
| 17:53.44 | AgiNamu | And know +/- how much it is? |
| 17:53.58 | TheEmperor | AgiNamu: pricing is on their website.. |
| 17:54.07 | outtolunc | the oem pricing is on their site |
| 17:54.29 | TheEmperor | funny thing is I emailed them, and after 3 weeks, they wrote me an email saying that they would get back to me in 4 weeks |
| 17:54.39 | TheEmperor | guess they are too busy.. |
| 17:54.40 | AgiNamu | OEM pricing? It says "email us" |
| 17:54.56 | Essobi | Firefly sucked the last time I used it. |
| 17:55.09 | Essobi | that's been months and months ago |
| 17:55.15 | AgiNamu | essobi, yea |
| 17:55.18 | AgiNamu | its working for me now :) |
| 17:55.26 | Beirdo | ahahah |
| 17:55.48 | Beirdo | I posted to the asterisk-canada mailing list and got a reply from a coworker :) |
| 17:55.52 | Beirdo | heh |
| 17:55.58 | outtolunc | strange, they used to have a 1-999, 1000-xxxx scale with pricing |
| 17:56.00 | TheEmperor | AgiNamu: the prices were there. I guess they took it off now |
| 17:56.06 | TheEmperor | yeah, I just checked. |
| 17:56.27 | AgiNamu | What was the 1-999 pricing? |
| 17:56.31 | AgiNamu | I need like 50-100 liceneses now |
| 17:56.36 | TheEmperor | both freshtel.net and virbiage.com |
| 17:56.39 | AgiNamu | and I need to localize it to Spanish and Korean. |
| 17:56.55 | TheEmperor | i think they only sell it blocks of 0-1000 and then up |
| 17:57.50 | Zeeek | The web site still says their IAX hardphione is coming out RealSoonNow after over a year |
| 17:58.14 | AgiNamu | Zeeek, yea, who cares? I've got IAX hardphones |
| 17:58.21 | AgiNamu | and in a week, they'll have iLBC support on them |
| 17:58.33 | Zeeek | the point is, emailing them I got the same kind of non-answer |
| 17:58.36 | AgiNamu | So the whole benefit of theirs is suddenly negated :) |
| 17:58.51 | AgiNamu | I dont know why they dont go buy a PA168 chip and add some hardware if they want to. |
| 17:58.52 | Zeeek | what IAX hardphones? |
| 17:59.02 | AgiNamu | And their USB phones! LOL $100US!!!! I buy them for a LOT LESS |
| 17:59.02 | *** join/#asterisk santiago (~santiago@63.245.86.93) |
| 17:59.06 | coppice | Zeeek: one of the PA168 phones looks rather like the picture of the firefly, and that really supports IAX now :-) |
| 17:59.11 | AgiNamu | I can sell them for $30 |
| 17:59.20 | Zeeek | coppice still trying to find out how well they work |
| 17:59.22 | AgiNamu | PA168 rocks |
| 17:59.32 | AgiNamu | Zeeek, I have 100 here, rolling them out to customers |
| 17:59.34 | AgiNamu | residential |
| 17:59.36 | AgiNamu | works great |
| 17:59.40 | coppice | Zeeek: vapour works? :-) |
| 17:59.40 | Zeeek | how are they provisioned AgiNamu? |
| 17:59.47 | AgiNamu | anon FTP |
| 17:59.53 | AgiNamu | ti sucks , but we're fixing it |
| 18:00.09 | coppice | AgiNamu: yep. the pa168 is a weird device, but its gonna be big |
| 18:00.13 | AgiNamu | I think Woody said it's in the 1.42 firmware, which should be out in a week |
| 18:00.20 | AgiNamu | and it'll have iLBC support on too |
| 18:00.21 | Zeeek | what is? |
| 18:00.26 | Zeeek | web interface? |
| 18:00.28 | AgiNamu | The provisiniong |
| 18:00.33 | AgiNamu | Provisioning sucks with the current firmware |
| 18:00.48 | AgiNamu | At any rate, I'm just gonna go mod the firmware. not that big a deal |
| 18:00.51 | Essobi | how do you provision? |
| 18:00.58 | Zeeek | my question too |
| 18:01.01 | Essobi | you can automate ANYTHING.. |
| 18:01.01 | AgiNamu | via anonymous FTP |
| 18:01.02 | Essobi | :) |
| 18:01.07 | Nebukadneza | hm ... |
| 18:01.12 | AgiNamu | no, their anon FTP provisionin blows |
| 18:01.26 | Nebukadneza | can someone try calling phone1@nebuk.homelinux.org (via SIP) |
| 18:01.27 | Zeeek | how can it be anon? How does it know the user info? |
| 18:01.32 | AgiNamu | a: it has a few bugs. b: you have ot create special files (binary sdram file) C: there's no encryption |
| 18:01.39 | AgiNamu | it looks for NAME_MAC_Ver.dat |
| 18:01.42 | Zeeek | based on MAC? |
| 18:01.44 | Zeeek | ok |
| 18:01.46 | AgiNamu | or PPP ID |
| 18:01.50 | AgiNamu | or Account name |
| 18:01.50 | Zeeek | thats what I finally guessed |
| 18:02.13 | Zeeek | anon ftp - there's anew slant on things... |
| 18:02.27 | Essobi | so uhh.. Write scripts to automate the binary file generation from a MYsql database then a php front end for the tables. |
| 18:02.32 | Essobi | What's the big deal? |
| 18:02.33 | AgiNamu | well, it can do anon FTP, TFTP or HTTP |
| 18:02.45 | AgiNamu | essobi, it's anonymous FTP |
| 18:02.48 | AgiNamu | and it requires a directory listing |
| 18:02.52 | AgiNamu | and there's no encryption :) |
| 18:02.57 | Essobi | HTTP is even better. |
| 18:02.57 | AgiNamu | ... yet. |
| 18:02.58 | Zeeek | can you give it a fixed ip without DHCP? |
| 18:03.03 | AgiNamu | sure |
| 18:03.08 | Zeeek | how? |
| 18:03.09 | Essobi | pull it directly from a PHP cgi |
| 18:03.16 | AgiNamu | you can config it from the phone, telnet, or web |
| 18:03.27 | Zeeek | there is a web interface, ok |
| 18:03.38 | Zeeek | then you provision IAX |
| 18:03.41 | Zeeek | right? |
| 18:03.55 | AgiNamu | the firmware decides the protocol |
| 18:04.05 | AgiNamu | it was getting too big to have all protocols in one firmware |
| 18:04.15 | AgiNamu | so you load iax firmware, sip firmware, mgcp, net2phone, h323 |
| 18:04.18 | AgiNamu | whatever you need |
| 18:04.23 | Zeeek | logical enough |
| 18:04.27 | AgiNamu | yep |
| 18:04.33 | AgiNamu | So we use IAX, since sip is a PITA and doesnt work |
| 18:05.04 | AgiNamu | I've noticed a few minor issues, say with codec negotiation |
| 18:05.13 | Zeeek | I'm trying to look at one of these phones next time I'm in back the USSR^H^H |
| 18:05.16 | AgiNamu | but no showstoppers, except the provisioning, which will be fixed in a few days |
| 18:05.32 | AgiNamu | want to buy a few samples? |
| 18:05.35 | *** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com) |
| 18:05.35 | Zeeek | "minor issues" = hmmmm reboot your phone? |
| 18:05.50 | AgiNamu | I've got someone going ot the USA next week |
| 18:05.55 | Zeeek | I'll be there in May at an undisclosed location |
| 18:06.07 | AgiNamu | no, codec negotiation, sometimes, when asterisk is ringing, and not using the phones preferred codec |
| 18:06.12 | AgiNamu | it decodes the data incorrectly |
| 18:06.21 | AgiNamu | where are you now, for shipping? |
| 18:06.34 | Zeeek | so instead of "I'm sorry..." the customer hears "Fsck yuuuuuuu" |
| 18:06.42 | Zeeek | Paris, France |
| 18:06.50 | AgiNamu | so intead of "ring" the customer hears "*IH@#*IHNADFS(D!@EIAHSD |
| 18:06.57 | Zeeek | not ever worth shipping cause that adds shipping +20%VAT |
| 18:07.17 | AgiNamu | ouch |
| 18:07.20 | Zeeek | the first time I ever had to pay VAT was on a Digium order - my ass still hurts |
| 18:07.26 | AgiNamu | Well, we can put a low price on the invoice. |
| 18:07.39 | Zeeek | I can buy Digium stuff here as cheaply as from the uS |
| 18:07.40 | AgiNamu | That's how we import 'em :) |
| 18:07.48 | *** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net) |
| 18:07.49 | Zeeek | you are where ? Brazil? |
| 18:07.53 | AgiNamu | Guatemala |
| 18:07.59 | AgiNamu | We buy from China |
| 18:08.04 | jsolares | well fuck |
| 18:08.07 | AgiNamu | have him put a low price on the invoice |
| 18:08.13 | Zeeek | stick some hash in the package, then it'll be sure to get here |
| 18:08.27 | shido6 | should insure shipments like that |
| 18:08.28 | jsolares | AgiNamu: what's your company called? |
| 18:08.39 | Zeeek | don't say it! |
| 18:08.44 | Zeeek | he's a Narc |
| 18:08.47 | *** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net) |
| 18:08.48 | jsolares | :P |
| 18:08.56 | AgiNamu | whats the hash for? bribing officials? or does that only work with Canadian people? |
| 18:08.59 | Zeeek | secret policia |
| 18:09.04 | AgiNamu | uh oh |
| 18:09.07 | Zeeek | no Canadians want guns |
| 18:09.09 | AgiNamu | im gonna have the tax admin on my ass now |
| 18:09.11 | AgiNamu | LOL |
| 18:09.18 | AgiNamu | What the fuck would canadians do with guns? use them as bongs? |
| 18:09.21 | Zeeek | they hang out all day io IRC |
| 18:09.27 | jsolares | do you have a brick & mortar store precense AgiNamu? |
| 18:09.32 | Zeeek | join #teens_with_tits |
| 18:09.35 | AgiNamu | jsolares, hell no |
| 18:09.38 | tzanger | AgiNamu: no we shoot moose |
| 18:09.49 | jsolares | too bad, i wanted to buy a phone :p |
| 18:09.50 | AgiNamu | Rule one: Don't get involved in retarded Central American regulations :) |
| 18:09.52 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 18:09.58 | AgiNamu | Our Compay is Telefinity |
| 18:10.03 | *** join/#asterisk __Sparks_ (ringding@bb-195-172-54-59.ukonline.co.uk) |
| 18:10.05 | jsolares | never heard of them |
| 18:10.06 | AgiNamu | Where are you ? |
| 18:10.13 | AgiNamu | nope, we're just launching. |
| 18:10.16 | jsolares | Ciudad de Guatemala |
| 18:10.17 | Zeeek | you should call it HuggaFinity |
| 18:10.20 | jsolares | :p |
| 18:10.29 | Zeeek | warmer |
| 18:10.55 | *** join/#asterisk mmlj4 (~looseduk@ip68-14-39-201.no.no.cox.net) |
| 18:11.00 | Zeeek | What are these pho,nes worth on the black market? |
| 18:11.09 | Zeeek | (meaning here) |
| 18:11.27 | shido6 | what phones? |
| 18:11.37 | BuckRogers | ahh yes the black market |
| 18:11.37 | Zeeek | AgiPhones |
| 18:11.44 | BuckRogers | the "shadow world" |
| 18:11.51 | coppice | about $30 |
| 18:11.53 | Zeeek | the nether world |
| 18:12.01 | CleanerX | well i figured out there is a problem with ast_data |
| 18:12.06 | Zeeek | that's a years wages where you are coppice, ya? |
| 18:12.18 | BuckRogers | its like middle earth on the border of morrdorn |
| 18:12.34 | Zeeek | clapton in a Rolex ad |
| 18:12.44 | BuckRogers | haha awsome |
| 18:12.47 | coppice | where I am, no. across the border factory workers might get $60-$70 a month |
| 18:12.51 | AgiNamu | We can sell them for about $64, FOB china |
| 18:13.03 | coppice | what a swindle |
| 18:13.14 | Zeeek | $64 is about the price for such a phone if they work, I'd say (in single Q) |
| 18:13.20 | BuckRogers | this is true u can get these objects very inexpensivly at the factories |
| 18:13.21 | coppice | those phones are about $30 |
| 18:13.32 | shido6 | what phones? |
| 18:13.36 | BuckRogers | ive heard of people doing it with memory and made big money |
| 18:13.43 | Zeeek | so the factory workers can buy 2 IAX phones a month |
| 18:13.53 | BuckRogers | ehh maybe more |
| 18:14.13 | coppice | Zeeek: well, the usually save up and get a GSM phone :-) |
| 18:14.14 | Zeeek | then they can go to a voIP show and dounble their investment |
| 18:14.20 | BuckRogers | you would probly get a better deal talking direct with management, they will rip off real easy |
| 18:14.26 | Zeeek | coppice - yeah with video capability |
| 18:14.26 | BuckRogers | corupt as hell |
| 18:14.49 | coppice | An engineer in the same area might get $500-$1000 a month |
| 18:14.52 | AgiNamu | $30? If so, I need a new supplier in China |
| 18:15.23 | BuckRogers | U can do it with AMD CPU's in malaysia |
| 18:15.30 | coppice | I know people who have got small volume samples for $40 |
| 18:15.38 | PBXtech | is there any sort of timer function, so if an agent is talking on a call for 10 minutes i can send an alert? |
| 18:15.38 | BuckRogers | ehh ive delt with China companies |
| 18:15.48 | AgiNamu | coppice, of PA168 phones? |
| 18:15.51 | BuckRogers | they really arnt that easy to deal with |
| 18:15.59 | AgiNamu | Buck, well, you gotta know chinese :) |
| 18:16.06 | coppice | yep. quite nice looking versions, too |
| 18:16.20 | BuckRogers | for negotiations? |
| 18:16.30 | AgiNamu | My provider is the same one who said "IF USE IAX NET . MUST HAVE IAX protocol flat roof" |
| 18:16.34 | *** part/#asterisk dahunter (~joe@lsanca1-ar8-4-60-068-194.lsanca1.dsl-verizon.net) |
| 18:16.43 | AgiNamu | negotiations? who negotiates. that's bad form. |
| 18:16.55 | BuckRogers | there price quotes on qty of 1000's was nearly what you could get by dealing with grandstream direct |
| 18:17.10 | BuckRogers | Bad form |
| 18:17.23 | BuckRogers | with sales people always negociate |
| 18:17.47 | Nebukadneza | hm ... |
| 18:18.12 | *** join/#asterisk lyroy (~lyroy@picachou.csaffluents.qc.ca) |
| 18:18.21 | *** join/#asterisk Imo (~johni@www.moe.co.uk) |
| 18:18.35 | coppice | my 7 year can provide excellent chinese translation services, if any of you need them :-) |
| 18:18.37 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 18:18.50 | lyroy | I'v download the CVS asteris-addons but i cannot find asterisk-addons/mysql-vm-routines.h... Does somehave have clue? |
| 18:19.26 | Zeeek | coppice are you a native speaker of either chinese? |
| 18:19.41 | coppice | bu shi |
| 18:19.49 | Zeeek | is that a no? |
| 18:19.51 | coppice | m'hai |
| 18:19.56 | Zeeek | is that a yes? |
| 18:20.16 | Zeeek | you drink wine? |
| 18:20.25 | coppice | m'hai |
| 18:20.26 | Zeeek | because we do use proof readers |
| 18:20.27 | *** join/#asterisk Gronker (~Gronker2@adsl-220-79-68.ags.bellsouth.net) |
| 18:20.29 | Nebukadneza | hai is a yes :) |
| 18:20.34 | Zeeek | in Japanese |
| 18:20.45 | Nebukadneza | wakatta |
| 18:20.50 | Zeeek | watevah |
| 18:20.56 | modulus_ | zai ni zwo yo |
| 18:21.04 | modulus_ | hai yun dwo jwo |
| 18:21.04 | Zeeek | so everyone, which is the best of these chinese phones? |
| 18:21.18 | coppice | Nebukadneza: but what is m'hai? |
| 18:21.22 | Cherebrum | I have stacks of phones from china here |
| 18:21.25 | Nebukadneza | coppice: no idea :) |
| 18:21.26 | Cherebrum | we get them very cheap |
| 18:21.33 | Cherebrum | Most of them are junk tho |
| 18:21.43 | Zeeek | I said the best |
| 18:21.45 | coppice | Nebukadneza: well, the "m'" negates |
| 18:21.53 | Zeeek | "not yes" |
| 18:21.54 | Nebukadneza | coppice: huh really? |
| 18:21.56 | Cherebrum | I recommend the Polycom phones |
| 18:21.59 | Cherebrum | they seem to work best |
| 18:21.59 | Nebukadneza | didnt know that :? |
| 18:22.27 | Nebukadneza | (but there is no 'single' m Oo) |
| 18:22.37 | Nebukadneza | whats the full form of this? |
| 18:22.41 | coppice | Nebukadneza: hai really means is, but is used for yes too. m'hai means isn't or no |
| 18:22.52 | *** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
| 18:22.56 | coppice | this is cantonese |
| 18:23.00 | *** join/#asterisk TSCHAK (~thomas@69-161-192-45.clspco.adelphia.net) |
| 18:23.04 | Zeeek | I had some of that today |
| 18:23.10 | Nebukadneza | hm kay |
| 18:23.12 | Zeeek | good with the pepper beef |
| 18:23.15 | TSCHAK | is there info I can read about configuring Asterisk to use QSIG ? |
| 18:23.31 | tzanger | TSCHAK: there really isn't a lot to it |
| 18:23.37 | tzanger | switchtype=qsig |
| 18:23.45 | *** join/#asterisk julianjm (~julianjm@250.Red-80-59-67.pooles.rima-tde.net) |
| 18:26.21 | shaZwaz | anyone knows of some AGI script with some web interface to feed in outgoing calls for different extension and dial them ? |
| 18:27.05 | PBXtech | is there any sort of timer function, so if an agent is talking on a call for 10 minutes i can send an alert? |
| 18:27.40 | shaZwaz | in the dial I guess |
| 18:27.43 | JerJer[mobile] | show application dail |
| 18:27.44 | JerJer[mobile] | dial |
| 18:27.45 | *** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com) |
| 18:27.48 | search_learn2005 | how is H.323 when it comes to NAT traverse? Is it less troublesome than SIP? I wish I can just buy a good IAX phone and get on with it. |
| 18:27.57 | Nebukadneza | argh ... |
| 18:28.21 | JerJer[mobile] | search_learn2005: it doesn't |
| 18:28.29 | tzanger | show application DailiLama |
| 18:28.35 | AgiNamu | H323 and "less troublesome" in the same sentences? |
| 18:28.39 | Essobi | I got something goofy going on with queue DTMF options.. |
| 18:28.44 | Zeeek | shaWaz you mean enter a number and a local extension and the call is made to ring the local extension and the external call ? |
| 18:28.45 | AgiNamu | search_learn2005, then buy an IAX2 phone |
| 18:28.54 | AgiNamu | I don't know what hte big deal is. |
| 18:29.12 | coppice | "VoIP is less troublesome without H.323" |
| 18:29.13 | Essobi | when I dial a queue.. and I'm in queue and I hit 0 to land on the operator.. if I hang up before they answer.. it just keeps ringing.. |
| 18:29.50 | Beirdo | Mar 4 13:28:58 WARNING[1060]: chan_sip.c:1829 sip_write: Asked to transmit frame type 2, while native formats is 4 (read/write = 4/4) |
| 18:29.57 | AgiNamu | coppice :) |
| 18:30.14 | Beirdo | is this an easy thing to fix? that was trying to transfer to a SIP phone |
| 18:30.41 | PBXtech | In the manager interface, disable sending events EXCEPT the UserEvents i send? |
| 18:30.49 | PBXtech | err how do it |
| 18:31.46 | search_learn2005 | AgiNamu: What is a good IAX2 phone, and which version of asterisk support IAX2? |
| 18:32.00 | AgiNamu | PA168, v1.40 and above. |
| 18:32.19 | AgiNamu | or did you mean which version of asterisk? um, anything? |
| 18:32.28 | search_learn2005 | Aginamu: Can you tell me the company name? |
| 18:32.38 | JerJer[mobile] | ~google PA168 |
| 18:32.44 | AgiNamu | Sure, Telefinity :). email me mgg@Atrevido.net |
| 18:32.44 | JerJer[mobile] | pick one |
| 18:32.56 | AgiNamu | or you can go find a company in china |
| 18:33.00 | AgiNamu | either way works. |
| 18:33.05 | search_learn2005 | thanks |
| 18:33.12 | Essobi | annyone here a queue wiz? |
| 18:33.19 | JerJer[mobile] | vi queues.conf |
| 18:33.22 | TSCHAK | hey, is it possible to use the asterisk-skinny channel so I can route calls from Cisco Callmanager? |
| 18:33.36 | JerJer[mobile] | TSCHAK: no |
| 18:33.42 | Essobi | TSCHAK No |
| 18:33.53 | JerJer[mobile] | is there an echo in here? |
| 18:33.54 | *** join/#asterisk HitTop (~root@host6614613596.biz.tor.fcibroadband.com) |
| 18:33.56 | TSCHAK | haha |
| 18:34.01 | Essobi | is there an echo in here? |
| 18:34.01 | HitTop | hi all |
| 18:34.10 | outtolunc | TSCHAK: no |
| 18:34.24 | JerJer[mobile] | damn a multi-tap echo |
| 18:34.29 | Essobi | mehe. |
| 18:35.15 | Essobi | Hmm. |
| 18:35.21 | Essobi | that makes no sense.. |
| 18:35.44 | Essobi | when dialing a dtmf while in queue to land dial() somewhere else.. |
| 18:35.58 | search_learn2005 | AgiNamu: What do you mean you have some PA 168 phones? |
| 18:36.01 | Essobi | the caller hangs up, but the callee's phone keeps ringing for quite a while afterwards. |
| 18:36.10 | AgiNamu | I mean, I have 5 boxes sitting 2 metres away from me. |
| 18:36.19 | AgiNamu | IAX2 ATAs and Phones |
| 18:37.50 | ManxPower | The signup page for Voicepulse Connect does not work. |
| 18:37.56 | *** join/#asterisk sunil (~sunil@202.54.37.184) |
| 18:37.59 | tzanger | ManxPower: they're all full |
| 18:38.14 | search_learn2005 | Aginamu: I am going to need around 50 of them, any ideas? |
| 18:38.25 | PBXtech | In the manager interface, how do i disable sending events EXCEPT the UserEvents i send? |
| 18:38.54 | Zeeek | who hosted the conference call last night? |
| 18:39.12 | Juggie | who ever owns 66.250.68.194 :) |
| 18:39.13 | Juggie | figure it out :P |
| 18:39.39 | Zeeek | I don't know who they are but I saw the name |
| 18:40.18 | sunil | need help, does Digium TE410P support R2MF signiling |
| 18:40.44 | shaZwaz | never heard of that |
| 18:41.22 | shido6 | sunil, hold on |
| 18:41.50 | sunil | ok shido |
| 18:42.42 | shido6 | drum roll |
| 18:42.43 | *** part/#asterisk Yoda-BZH (~yoda-bzh@213.223.51.13) |
| 18:42.45 | shido6 | http://www.soft-switch.org/ |
| 18:42.46 | shido6 | BAM! |
| 18:42.52 | shido6 | take a look over there |
| 18:44.04 | Essobi | shido6 Never seen coppice's webpage I guess? |
| 18:44.23 | Zeeek | spandsp.org |
| 18:44.37 | Essobi | he bought another domain |
| 18:44.39 | *** join/#asterisk Bonbon (~bonbon@83.146.53.34) |
| 18:44.50 | Bonbon | guys, has anyone out there done any tapi integration with eyebeam? |
| 18:44.50 | tzanger | yeah what was it |
| 18:45.05 | Essobi | Not with Eyebeam. |
| 18:45.13 | Essobi | I was messing with the * tapi last night |
| 18:45.25 | Essobi | and outlook |
| 18:45.25 | shido6 | Essobi, slow doses of info |
| 18:45.28 | shido6 | slow dosses |
| 18:45.32 | Zeeek | who in the USA has these iax phones for sale ? |
| 18:45.35 | Essobi | and a pop-up |
| 18:45.54 | shido6 | spandsp and www.soft-switch.org are the same |
| 18:46.00 | Essobi | yup |
| 18:46.07 | Essobi | that's coppice |
| 18:46.09 | Zeeek | spandsp is easier to remember |
| 18:46.24 | Essobi | (since it's the name of the app_*.c) |
| 18:46.25 | Essobi | :) |
| 18:46.47 | Zeeek | before you had to Google for spandsp to find opencall |
| 18:47.05 | Inv_arp | how do ya tell * that im dialing a 3 digit ext .. eg..101 when i press 1 it jumps to t, responding invalid ext |
| 18:47.09 | Zeeek | with domain names cheaper than the cheapest chinese IAX phone.... |
| 18:47.29 | Zeeek | t would be timeout |
| 18:48.02 | Zeeek | sounds like a _1. is lurking |
| 18:48.30 | Zeeek | or even the more heinous _. |
| 18:48.39 | Essobi | like.. mod stream dialing? |
| 18:48.43 | Essobi | mid stream |
| 18:48.48 | Essobi | or after you dial the number |
| 18:48.59 | Essobi | BOOP BOOOP BOOOOOP - This ext is invalid |
| 18:49.24 | *** join/#asterisk jaxxan (~jaxxan@202.70.125.109) |
| 18:49.24 | jaxxan | g'morning |
| 18:49.28 | Inv_arp | after first BOOP doesnt wait for others |
| 18:49.35 | Essobi | g'afternoon. |
| 18:49.45 | Essobi | Inv_arp that's in your dial patterns setup on the phone |
| 18:49.47 | Essobi | not * |
| 18:49.52 | Essobi | atleast not with sip. |
| 18:49.57 | Essobi | I dunno about other protocols. |
| 18:50.05 | Essobi | what kind of phone and protocols? |
| 18:50.20 | Inv_arp | Essobi: ahh patterns sip |
| 18:50.28 | Essobi | What kind of phone? |
| 18:50.43 | Essobi | I know the 7960's DL the dial patterns from TFTP. |
| 18:50.49 | Essobi | I dunno about anything else. |
| 18:51.33 | Essobi | so you can get the 2nd dial tone after a 9.. or interupt a dialing if they start a dial with 1, etc, etc. |
| 18:52.11 | Nebukadneza | hm ... |
| 18:52.34 | Zeeek | what web site sells these IAX phones that anyone here has used/heard of? |
| 18:52.35 | Nebukadneza | can anyone help me testing my SIP settings? (just call 01@nebuk.homelinux.org) |
| 18:52.59 | Inv_arp | Essobi: internal lines call each other fine but when calls come from outside BV they get menu and from menu i cant access internal sip phones |
| 18:53.20 | AgiNamu | Zeeek, I have them, and sell them. But dont have a site up for it yet. working on it. |
| 18:53.38 | *** join/#asterisk Tili (~Tili@202-133-65-229-dialup.sat.net.pk) |
| 18:53.42 | Zeeek | but you aren't in the US |
| 18:53.47 | Zeeek | so there are borders |
| 18:53.54 | Zeeek | I don like borders |
| 18:54.28 | *** part/#asterisk wizhippo (~wizhippo@Quebec-HSE-ppp233869.qc.sympatico.ca) |
| 18:56.41 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 18:57.52 | Inv_arp | "if u know the party u wanna reach please dial now" i press 101 ... * stops at 1 and doesnt wait for "01" and says there is no exten at 1 |
| 18:58.56 | shido6 | your dialplan is setup incorrectly |
| 18:58.59 | *** join/#asterisk RoyK (~roy@83.80-203-29.nextgentel.com) |
| 18:59.07 | |Vulture| | Inv_arp: do you have ex101 in the same context? or the context included? |
| 18:59.33 | PBXtech | In the manager interface, how do i disable sending events EXCEPT the UserEvents i send? |
| 18:59.48 | |Vulture| | manager.conf ? |
| 19:00.40 | PBXtech | what do all these mean: write = system,call,log,verbose,command,agent,user |
| 19:00.54 | PBXtech | the call is the one that sends all the events |
| 19:01.14 | Zeeek | AgiNamu who are your customers? |
| 19:01.22 | PBXtech | but if i could get rid of that and send something with one of hte others |
| 19:01.23 | AgiNamu | We're a VSP in Guatemala. |
| 19:01.24 | PBXtech | i dunno |
| 19:01.39 | Zeeek | you sell to individuals or to companies mostly |
| 19:01.50 | Inv_arp | |Vulture|: http://pastebin.ca/6864 |
| 19:01.53 | *** part/#asterisk MikeJ[Jayden] (~ircatjerr@65.170.43.34) |
| 19:02.31 | shido6 | where is office? |
| 19:02.37 | [TK]D-Fender | remove the [] on your include line |
| 19:03.12 | [TK]D-Fender | also you us a "-" in yout context and an "_" in your include. The names don't match! |
| 19:03.21 | Inv_arp | oh wait the include line looks wrong OFFICE-I=SIP/x101 |
| 19:03.22 | [TK]D-Fender | use* |
| 19:03.39 | Inv_arp | :) |
| 19:03.50 | [TK]D-Fender | include => [to_sip] ---------------- [to-sip] |
| 19:03.50 | [TK]D-Fender | 025 exten => 101,1,Dial(${OFFICE-I},20,tr) |
| 19:04.04 | [TK]D-Fender | subtle.... |
| 19:05.57 | AgiNamu | Zeek, mainly foreigners living down here |
| 19:06.14 | Zeeek | and they hook in to your services? |
| 19:07.05 | AgiNamu | yea, we sell them a phone, connected to our servers |
| 19:07.10 | RoyK | hi. anyone that feels inclined to help a bugger with some really bad sound on SIP? problem is, we're getting something sounding like jitter even when connecting the ATA to the server via LAN |
| 19:07.33 | Zeeek | a bugger? onthis channel? |
| 19:07.38 | Zeeek | I'd be careful |
| 19:07.46 | RoyK | please.... |
| 19:07.49 | jlewis | anyone here using IAX with single type=friend entries for both inbound and outbound calling? |
| 19:08.02 | eKo1 | jlewis: yes |
| 19:08.54 | jlewis | eKo1: I'm having some weird issues with (mostly with my misunderstanding of) context= entries |
| 19:09.30 | bjohnson | ~seen JerJer |
| 19:09.31 | jbot | jerjer <~JerJer@dsl-106-170.che.centurytel.net> was last seen on IRC in channel #asterisk, 5d 2h 50m 36s ago, saying: 'mrgoby: sure'. |
| 19:09.39 | jlewis | I thought context= lines were only applied to incoming calls...but they appear to be applied to outgoing as well if the dial command doesn't specify a remote context |
| 19:09.58 | Inv_arp | bjohnson: JerJer[mobile] is usaully here lately |
| 19:10.06 | bjohnson | JerJer[mobile]: what firmware are you using on your WRT54G? |
| 19:10.10 | jlewis | though I found I can override that behavior by dialing IAX2/identifier/exten@ |
| 19:10.11 | eKo1 | jlewis: of course. |
| 19:10.19 | |Vulture| | bjohnson: I am using the Sveasoft |
| 19:10.37 | |Vulture| | bjohnson: I think he uses some Open source project |
| 19:11.23 | bjohnson | openwrt |
| 19:11.26 | |Vulture| | yea |
| 19:11.31 | bjohnson | does it really make any difference? |
| 19:11.35 | |Vulture| | I duno |
| 19:11.58 | bjohnson | ok .. I'll start with openwrt just since it's accessible |
| 19:12.06 | |Vulture| | bjohnson: what version wrt54g do you have? it is on the bottom.. like 2.0,2.1,2.2,3 |
| 19:12.15 | bjohnson | 2.2 |
| 19:12.16 | spackle | bjohnson, I got Asterisk going on Openwrt, works OK |
| 19:12.34 | jlewis | so...next question is what exactly does peercontext= do? |
| 19:12.36 | |Vulture| | k 2.2 should have firmware from openwrt |
| 19:12.41 | spackle | bjohnson, there are pointers to the important stuff from the wiki. |
| 19:12.50 | bjohnson | according to les over at #wrt54g the experimental version of openwrt should work |
| 19:13.09 | *** join/#asterisk joebob2000 (~joebob200@serverd.qn.net) |
| 19:13.15 | bjohnson | I'm just starting it now |
| 19:13.19 | *** join/#asterisk dslx (~jay@network-operations-center.dslx.net) |
| 19:13.26 | Zeeek | "The inexpensive VOIP gateway you are waiting for a long time" |
| 19:13.38 | Zeeek | that must be whyuy it's inexpensive? |
| 19:13.53 | bjohnson | this wrt54g I think is going to cost me $0 |
| 19:14.17 | joebob2000 | are there any good general practices for securing voip usage against eavesdropping? |
| 19:14.18 | bjohnson | MIR, discounts, coupons, price matches .. all bringing the price down |
| 19:14.41 | bjohnson | joebob2000: easiest is vpns or ssh tunnels |
| 19:14.59 | Zaw | what codec should i be using for the least latency and the best quality? |
| 19:15.09 | bjohnson | LAN traffic is uaually not protected and even internet stuff is usually ignored |
| 19:15.14 | dslx | Can * be used to terminate a bunch of IAX connections from branch offices and forward the IAX stream to someone that provides IAM Termination? |
| 19:15.29 | bjohnson | yes but why ? |
| 19:15.31 | dslx | IAM=IAX (sorry) |
| 19:15.41 | joebob2000 | is there any movement toward vendors supporting it? i am currently with voicepulse and dont know of any way to secure the link |
| 19:15.45 | Inv_arp | Zaw: http://voip-info.org/wiki-Bandwidth+consumption |
| 19:15.49 | bjohnson | if going to a voip provider I just connect to them directly from each location |
| 19:15.57 | Zaw | Inv_arp: thanks |
| 19:16.06 | *** join/#asterisk anti (russ@anti.developer.gentoo) |
| 19:16.09 | bjohnson | joebob2000: nope |
| 19:16.38 | dslx | The branch offices will be my customers and I want just one connection to the IAX provider |
| 19:16.43 | joebob2000 | so generally you shouldnt trust a voip call any more than an email |
| 19:17.10 | Inv_arp | joebob2000: just like the pstn |
| 19:17.49 | dslx | Anyone know anyone that is doing IAX termination in the Southeast US? |
| 19:18.01 | *** part/#asterisk xai (~pasta@user-0vvdb42.cable.mindspring.com) |
| 19:18.32 | joebob2000 | true, altho the pstn is generally regarded as more lock-and-key than the internet at large |
| 19:18.34 | bjohnson | joebob2000: generally you shouldnt trust a voip call any more than a pstn call |
| 19:18.52 | bjohnson | might be thgouht of that way .. but it isn't that way |
| 19:19.41 | __Sparks_ | Hi! - I ha having a bit of trouble with recording calls with asterisk. - I have tried the following line, but am not sure if it is actually doing it or not! - exten => 1471,2,Monitor(wav,1471) |
| 19:19.55 | bjohnson | besides, other methods of eavesdropping are easier and don't require access to the wiring at all |
| 19:20.10 | joebob2000 | id say i know of far more security-indifferent internet companies than i do security-indifferent phone companies. at least hardline providers. |
| 19:20.41 | bjohnson | I bet I could tap right into the phone line at your house |
| 19:20.53 | joebob2000 | i bet you couldnt |
| 19:20.53 | bjohnson | how is that for indifferent |
| 19:21.01 | joebob2000 | (only because i dont have a phone line at my house) |
| 19:21.33 | joebob2000 | true, it becomes a physical issue if someone is dead set on compromising your communication |
| 19:21.35 | joebob2000 | BUT |
| 19:21.48 | *** join/#asterisk afe ([XJ6cAANOq@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
| 19:22.26 | joebob2000 | the fact that you rarely talk about worthwhile information on the phone means that the ability to aggregate huge amounts of streams of info becomes much more valuable to someone looking to exploit for profit |
| 19:23.22 | joebob2000 | meaning, that the ability for someone to simply pipe off all voip traffic and filter through it at leisure is much more of a risk than someone looking to compromise my phone line directly |
| 19:24.55 | joebob2000 | im just curious about it since my job has had a security slant to it recently |
| 19:24.59 | joebob2000 | anyone care to comment? |
| 19:25.45 | wildcard0 | sip traffic would be difficult to do that with becaues of the rtp nature. but if it's really a concern for you, run it through an encrypted tunnel |
| 19:26.51 | eKo1 | Does IAX have the ability to encrypt the rtp stream? |
| 19:27.07 | eKo1 | Or did I just dream that? |
| 19:27.12 | wildcard0 | iax doesn't use rtp in the same way. and yes, you can encrypt iax |
| 19:27.25 | wildcard0 | i'd trust ipsec over iax crypto tho |
| 19:27.36 | eKo1 | why? |
| 19:27.47 | wildcard0 | unless of course you could gaurentee the delivery of the key safely |
| 19:28.02 | wildcard0 | it's been looked at by more eyes. less likely to have some hole |
| 19:28.35 | joebob2000 | so iax uses a preexisting key arrangement instead of a SSL like handshake? |
| 19:28.54 | Nebukadneza | which ports do i need for sip? (besides 5060 tcp/udp) |
| 19:29.01 | *** join/#asterisk Mneumonic (Mnemonic@ool-18ba58b4.dyn.optonline.net) |
| 19:29.34 | BuckRogers | yeah i just got the diguim te410 and a amd 64 fx 55 with two gigs ram |
| 19:29.46 | BuckRogers | my system is gonna cook |
| 19:29.52 | eKo1 | Nebukadneza: 10000-20000 |
| 19:29.53 | BuckRogers | like emerl |
| 19:29.57 | wildcard0 | i don't remember off the top of my head. but any symmetric crypto is tuns faster than public key crypto |
| 19:30.19 | wildcard0 | i usually set a 1000 port range instead and specify it in rtp.conf |
| 19:30.38 | Nebukadneza | eKo1: Oo WTF? |
| 19:30.44 | Nebukadneza | thats a big/high number |
| 19:30.56 | BuckRogers | running basic fedora 3 no gui |
| 19:31.25 | wildcard0 | Nebukadneza, that's hte default. look at rtp.conf |
| 19:31.39 | Nebukadneza | hm kay sec |
| 19:32.14 | Zeeek | AgiNamu are you still lurking? |
| 19:32.29 | Nebukadneza | wildcard0: what happens if i decrease the number of ports? |
| 19:33.05 | wildcard0 | it shouldn't make much of a difference unless you're supporting a -huge- number of clients |
| 19:33.12 | BuckRogers | any one running asterisk on 64bit ark |
| 19:33.18 | BuckRogers | here |
| 19:33.32 | eKo1 | IA64? |
| 19:33.37 | BuckRogers | amd64 |
| 19:33.54 | eKo1 | eh, isn't it the same? |
| 19:34.01 | Nebukadneza | wildcard0: okay ... |
| 19:34.03 | BuckRogers | i suppose im not an expert |
| 19:34.08 | wildcard0 | eKo1, vastly different |
| 19:34.14 | wildcard0 | VASTLY |
| 19:34.19 | wildcard0 | so different, i can't stress it enough |
| 19:34.25 | eKo1 | Is the IS different? |
| 19:34.32 | Nebukadneza | wildcard0: are those on tcp or upd |
| 19:34.43 | *** join/#asterisk firestrm (F1r3570rm@S010600047577bccd.gv.shawcable.net) |
| 19:34.44 | BuckRogers | i was told the amd64 fx 55 was the top of the line |
| 19:34.47 | wildcard0 | eKo1, it's as different as x86 vs sparc |
| 19:34.56 | eKo1 | Ah, OK. |
| 19:34.58 | wildcard0 | Nebukadneza, udp |
| 19:35.00 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 19:35.12 | *** join/#asterisk solutions_2005 (~ben@203.100.216.218) |
| 19:35.16 | Mneumonic | Hey, i just got my sipura 841, i made some test calls to friends and it cut out on one call, then he said that it was like i was getting louser and then quieter... i have the codec set to G711u.... should i change the codec? |
| 19:35.19 | solutions_2005 | Hello all |
| 19:35.23 | Nebukadneza | wildcard0: kay |
| 19:35.23 | eKo1 | I just thought it was the same since most AMD processors use Intel architectures |
| 19:35.30 | Nebukadneza | got 10000:12000 open now - should work? |
| 19:35.48 | BuckRogers | deffintly not true with the fx55 |
| 19:36.00 | wildcard0 | if it matches with rtp.conf, yes |
| 19:36.06 | solutions_2005 | I am in desperate need of some advice with asterisk is anybody willing to help me |
| 19:36.08 | BuckRogers | 1mb on cpu cache |
| 19:36.18 | wildcard0 | you -may- need to restart asterisk (not just reload) to read the changes in rtp.conf. i forget |
| 19:36.44 | Inv_arp | solutions_2005: u havent asked anything |
| 19:36.49 | wildcard0 | eKo1, that hasn't really been true since the 486. they use the same instruction set only |
| 19:36.55 | firestrm | anyone know why i cant get fxotune to run.. it comes back with Skipping non-TDM / non-FXO |
| 19:37.01 | solutions_2005 | LOL, im getting there just making sure im in the rite place :) |
| 19:37.19 | firestrm | i have fxo card installed. |
| 19:37.50 | solutions_2005 | Has anyone had any luck using asterisk (SIP) behing cone nat firewall? |
| 19:38.02 | BuckRogers | yes solutions |
| 19:38.12 | spackle | what is "cone nat firewall"? |
| 19:38.13 | wildcard0 | solutions_2005, yes, someone has |
| 19:38.21 | solutions_2005 | :) |
| 19:38.45 | solutions_2005 | I was beginning to think it was impossible |
| 19:39.24 | firestrm | most things are impossible with asterisk.. until you find the right breadcrumb of info :) |
| 19:39.37 | wildcard0 | or you dig out gcc |
| 19:39.52 | solutions_2005 | Well I have read up on so many people having my problems but no solutions |
| 19:39.53 | Inv_arp | solutions_2005: nah h323 is impossible :) |
| 19:39.59 | solutions_2005 | lol |
| 19:40.19 | firestrm | fxotune... now thats impossible.. |
| 19:40.39 | solutions_2005 | Im looking at getting FX0 card once I get sip going properly |
| 19:41.07 | firestrm | solutions_2005, thats when all the fun begins!! |
| 19:41.22 | Mneumonic | anyone got a quick answer to my question? |
| 19:41.28 | *** join/#asterisk HeadachesAbound (user79@wsip-68-99-73-32.tu.ok.cox.net) |
| 19:41.37 | firestrm | maybe :) |
| 19:41.42 | solutions_2005 | I can make outgoing calls but not recieve incoming calls via sip |
| 19:41.44 | firestrm | hows that for quick |
| 19:41.59 | Mneumonic | very quick :) |
| 19:42.10 | __Sparks_ | Can anyone here help me with my audio recording in Asterisk!? |
| 19:42.14 | Mneumonic | Hey, i just got my sipura 841, i made some test calls to friends and it cut out on one call, then he said that it was like i was getting louser and then quieter... i have the codec set to G711u.... should i change the codec? |
| 19:42.17 | solutions_2005 | Except when I changed my sip.conf file, restarted asterisk it worked |
| 19:42.31 | solutions_2005 | then stoped once I restarted it again |
| 19:42.43 | solutions_2005 | Never had it going again |
| 19:42.54 | firestrm | Mneumonic, ive been playing around with a s3k.. same problem.. |
| 19:43.11 | HeadachesAbound | Hypothetical Question..What wouold you do with 28 T1 Connections and how many asterisk boxes would it take to use them? |
| 19:43.21 | wildcard0 | Mneumonic, how's the frequency range on the mic/speaker on that phone? |
| 19:43.26 | firestrm | Mneumonic, there seems to be a bug in the echocancelling algorithm |
| 19:43.35 | wildcard0 | HeadachesAbound, i'd just get a ds-3 instead. |
| 19:43.49 | Mneumonic | dude, the speaker just sucks on this phone |
| 19:44.13 | wildcard0 | i figured that. does it break up if you talk loud? |
| 19:44.13 | Mneumonic | firestrm... theres no way around this bug? |
| 19:44.26 | HeadachesAbound | that's what we are looking at. we got quoted 825 a month for 28 t's on a ds3 |
| 19:44.31 | wildcard0 | turn off echo cancellation? |
| 19:44.32 | Mneumonic | yea, a little |
| 19:44.47 | firestrm | Mneumonic, playing with the gains, does bring some relief, but then you start to get more echo.. |
| 19:45.00 | Mneumonic | wildcard - on the phone or on in *? |
| 19:45.10 | Mneumonic | for turning off echo cancellation |
| 19:45.20 | wildcard0 | Mneumonic, start with both. only use it where you need it |
| 19:45.42 | Mneumonic | i shoulda just got the IP-300 :( |
| 19:46.05 | wildcard0 | i have a few ip500's here. they're nice |
| 19:46.06 | spackle | no speakerphone on ip-300 |
| 19:46.17 | wildcard0 | ya |
| 19:46.32 | wildcard0 | snom 220 is supposed to be nice |
| 19:46.36 | HeadachesAbound | what's the best place to get a good deal on ip500s? |
| 19:46.37 | wildcard0 | never tried it tho |
| 19:46.42 | wildcard0 | HeadachesAbound, ebay? |
| 19:46.50 | Mneumonic | wildcard - where in the sipura config do i turn it off? cant seem to find it |
| 19:46.53 | spackle | Did you get the latest firmware for the 841? I have had no trouble with mine. |
| 19:47.03 | wildcard0 | i believe it's in the line config near the bottom? |
| 19:47.12 | wildcard0 | ok. out for a bit. |
| 19:47.13 | wildcard0 | later |
| 19:47.18 | Mneumonic | silence support enable? |
| 19:47.41 | km- | is there a way to make firefly ring through the headset? |
| 19:48.00 | ManxPower | The SPA-841 defaults to VAD disabled. |
| 19:48.28 | |Vulture| | Anyone use any web based voicemail packages? |
| 19:48.28 | Mneumonic | ManXPower - VAD? |
| 19:48.41 | ManxPower | VAD = Voice Activity Detection |
| 19:48.47 | solutions_2005 | Sure busy in here aint it |
| 19:48.48 | ManxPower | aka Silence Supression |
| 19:48.53 | ManxPower | ~docs |
| 19:48.54 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 19:49.25 | Mneumonic | the silence support enable is off on mine |
| 19:49.38 | firestrm | ManxPower, have you ever used fxotune? i cant seem to get it to detect my x100p card |
| 19:49.58 | ManxPower | firestrm: fxotune only works with the FXO modules on the TDM400P |
| 19:50.18 | firestrm | ManxPower, ahhh,, well that would explain it.. |
| 19:50.28 | *** join/#asterisk kenoca (~ken@207.175.34.200) |
| 19:50.43 | ManxPower | firestrm: fxotune uses tuning features of the specific chip on those modules. |
| 19:51.10 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 19:51.19 | firestrm | ManxPower, im going to have to break down and buy a tdm.. |
| 19:51.36 | ariel_ | afternoon all |
| 19:51.38 | km- | hahaha |
| 19:51.40 | mishehu | ManxPower: don't see fxotune on voip-info, what is it? |
| 19:51.50 | km- | you gotta love it when the VP calls and says "Pete, this iaxcomm shit is really cool!" |
| 19:51.52 | ManxPower | mishehu: It's something to help with echo |
| 19:52.00 | firestrm | makes the echo.. echo.. echo... go away |
| 19:52.07 | mishehu | ManxPower: it's separate from the zaptel drivers I gather right? |
| 19:52.12 | kenoca | need a little help. I am having a problem when calling in to check voicmail of entering an extension. asterisk is not picking up the digits entered. What am i missing |
| 19:52.18 | ManxPower | mishehu: no, it's part of the Zaptel CVS-HEAD |
| 19:52.24 | mishehu | ah ok |
| 19:52.30 | Mneumonic | wow... i just left a voicemail and even internally it has the louder quieter effect |
| 19:52.35 | Zeeek | kenoca what phone? |
| 19:52.35 | mishehu | alright, back to work for me... |
| 19:52.43 | *** join/#asterisk HitTop (~root@host6614613596.biz.tor.fcibroadband.com) |
| 19:52.48 | BrianR___ | Is anyone hwere using Asterisk with a non-mysql database? |
| 19:53.23 | Zeeek | kenoca Grandstream phone? |
| 19:53.26 | kenoca | when calling in from anyphone it seems |
| 19:53.36 | kenoca | calling in on cell phone to asterisk or cisco phone to asterisk |
| 19:53.39 | *** part/#asterisk joebob2000 (~joebob200@serverd.qn.net) |
| 19:53.45 | Zeeek | try the echo test to see if any audio is getting thru |
| 19:53.55 | kenoca | how might i do the echo test |
| 19:54.01 | Zeeek | echo () |
| 19:54.05 | Zeeek | in the dial plan |
| 19:54.14 | Zeeek | show application echo on CLI |
| 19:55.32 | Zeeek | ofr http://www.voip-info.org/wiki-Asterisk+cmd+Echo |
| 19:55.53 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 19:56.01 | kenoca | ok i will give that a try, what am i looking for when i do it. |
| 19:56.37 | Zeeek | since you hear the quadio coming back you can test latency and quality you'll hear dropuit etc |
| 19:56.54 | Zeeek | *quadio is available only in Criso |
| 19:57.00 | kenoca | ok setting up test. |
| 19:57.11 | GiabboO | bye al |
| 19:57.13 | GiabboO | +l |
| 19:57.14 | *** join/#asterisk ropeguru_work (~ropeguru@141.152.37.26) |
| 19:57.14 | *** part/#asterisk GiabboO (~GiabboOo@host101-246.pool8173.interbusiness.it) |
| 19:57.23 | Zeeek | exten => 600,1,Echo ; will do it |
| 19:57.37 | *** join/#asterisk randu (~randy@207.144.38.52) |
| 19:57.50 | *** join/#asterisk calvinhp (~calvinhp@rrcs-24-123-25-236.central.biz.rr.com) |
| 19:58.58 | ropeguru_work | Has anyone gotten any emails from the users list in the past hour or so?? I get a pretty steady stream all day usually and things seen unusually quiet.. |
| 19:59.00 | AgiNamu | if I move a file, say, Master.csv, there's no danger of a partial write, right? |
| 19:59.17 | *** join/#asterisk Nukemizer (~Nuke@65.103.231.131) |
| 19:59.27 | Zeeek | no guarantee IMO |
| 19:59.31 | kenoca | ok the echo test works good. i can hear everything i say right away |
| 19:59.38 | Zeeek | and the tones? |
| 19:59.42 | randu | I have a weird problem. We have an asterisk box setup on high speed DSL. It has 1 broadvoice sip connected to it for incoming calls. If I call from my vonage line it connects just fine, but if they call from the verizon land line it is fast busy. any ideas what could be happening? |
| 19:59.48 | Zeeek | try to hit the # key |
| 19:59.48 | kenoca | tones sound like a quick beep |
| 19:59.55 | kenoca | # key worked fine |
| 20:00.02 | Zeeek | it should end the echo test |
| 20:00.06 | kenoca | pressing numbers did not sound like I expected |
| 20:00.09 | kenoca | it did |
| 20:00.16 | ropeguru_work | randu: sounds like Verizon may not have the prefix in their switch |
| 20:00.25 | Zeeek | AguiNamu is this something like the phones? http://ipphone.eezeephone.com/index_files/Page640.htm |
| 20:00.58 | Zeeek | so kenoca it's only in vmail that you have the problem? |
| 20:01.09 | kenoca | even when dialing an exten |
| 20:01.11 | Zeeek | often cellphones don't do great with tones by the way |
| 20:01.27 | kenoca | ok |
| 20:01.30 | Zeeek | that has nothing to do with a cellphone |
| 20:01.36 | randu | ropeguru_work : hmm, weird... so do we need to call verizon and say add this to your switch? I find it weird that it would not be in their switch |
| 20:01.39 | ropeguru_work | zandu: If you open up the cli on the * box with -vvvvr and watch for incoming calls from Broadvoice, do you only see them from the Vonage line or Vonage and Verizon?? |
| 20:01.45 | Zeeek | dialing an extension on what phone? Crisco? |
| 20:02.00 | kenoca | i also did the test from my cisco 7960 and cell. same results |
| 20:02.11 | *** join/#asterisk TSCHAK (~thomas@69-161-192-45.clspco.adelphia.net) |
| 20:02.13 | randu | just from vonage line and cell phones, I do not see any other lines |
| 20:02.16 | AgiNamu | Wait, so writing CDRs to Master.csv -- is a dumb idea? |
| 20:02.17 | Zeeek | something ain't right |
| 20:02.23 | AgiNamu | ours look nicer |
| 20:02.25 | AgiNamu | but yea, same idea |
| 20:02.26 | kenoca | cisco 7960 is coming from another VoIP provider |
| 20:02.47 | Zeeek | you should have mentioned that earlier! |
| 20:02.51 | kenoca | calling into asterisk |
| 20:03.05 | kenoca | sorry, even my house phone does the same thing though |
| 20:03.08 | Zeeek | the provider may require a different DTMF |
| 20:03.19 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-219-145.dsl.scarlet.be) |
| 20:03.37 | ropeguru_work | randu: So when you are at the CLI and dial from your vonage line, you can see the call coming in from Broadvoice. But is you dial in from a Verizon line, you do not see and incoming activity?? |
| 20:03.43 | Zeeek | your cell calls through what to get to asterisk? you have POTS hw hooked up? |
| 20:04.32 | kenoca | my cell phone is through cingular, the asterisk system is VoIP through voiceeclipse.net not POTS hw on it |
| 20:04.33 | randu | ropeguru_work: I don't think so I will check again. |
| 20:04.33 | Zeeek | AgiNamu where is your colleague going in the US next week? Florida? |
| 20:04.44 | Nebukadneza | hm ... |
| 20:05.00 | Zeeek | so tyour cell is calling asterisk though a SIP provider? |
| 20:05.09 | ropeguru_work | Ok, I am just trying to verify that it is a Verizon issue. It may be a relatively new prefix and they have not entered into their translations yet. |
| 20:05.11 | kenoca | yes |
| 20:05.19 | kenoca | so is my home phone |
| 20:05.40 | Zeeek | and the tones sound clipped off, do they not? |
| 20:05.40 | kenoca | as well as my office voip phone going throught and mgcp provider to a sip provider |
| 20:05.47 | kenoca | yes they do |
| 20:05.54 | kenoca | on all calls no matter where from |
| 20:05.56 | Nebukadneza | i got a * on my server which is my router at the same time ... when trying to phone via SIP (someone calls me on 01@my-external-ip) my softphone which runs on a machine in the lan rings but there is no sounds on both sides of the connection ... i got the ports 5060 and 10000-20000 on tcp and upd open on my router(=* server) |
| 20:05.56 | Zeeek | it sounds like the provider isn't sending them through |
| 20:06.33 | kenoca | i see, so voiceecplise might be a bad provider for my asterisk system |
| 20:06.43 | Zeeek | I use voicepulse IAX |
| 20:06.52 | Zeeek | works great (outgoing) |
| 20:07.10 | Zeeek | I'm a little puzzled by your problem |
| 20:07.12 | ropeguru_work | Zeek: I have to agree. I have not had a minutes issue with VP Connect service. |
| 20:07.21 | kenoca | outgoing is great and incoming is great too when it works. It's just dialing numbers on the asterisk prompts is were the problems are at. |
| 20:07.28 | kenoca | could this be a bandwidth problem |
| 20:07.40 | Zeeek | ??? |
| 20:07.47 | *** join/#asterisk heison (~heison@dns.somanetworks.com) |
| 20:07.51 | Zeeek | outgoig...gibberish.?...??? |
| 20:08.00 | kenoca | no sounds fine |
| 20:08.12 | afe | anyone knows of a reliable and easy way to automatically email a recorded call sound file? |
| 20:08.26 | Zeeek | but the cisco is talking direct to asterisk? |
| 20:08.42 | kenoca | 1 cisco is 1 cisco is not |
| 20:08.53 | Zeeek | and the one that is dials ok? |
| 20:09.02 | kenoca | yes |
| 20:09.14 | afe | could it be done in the dialplan, or do I need a cron script that checks a directory? |
| 20:09.16 | Zeeek | are you on the Michael Jackson jury by and chance? |
| 20:09.28 | kenoca | no but i am close to it. |
| 20:09.29 | KalD|Work | ok - 100 dollars to the first person that can make my friend's furby say the incoming caller id from asterisk! =) |
| 20:09.49 | shido6 | hehe |
| 20:09.55 | Zeeek | well, the phone connected to asterisk dials fine - calls coming from from providr(s) don't ... |
| 20:09.59 | kenoca | it's 30 minutes up the road from me |
| 20:10.06 | shido6 | get an IR transceiver |
| 20:10.19 | Zeeek | I like S.B. for vacations - the most good restaurants per square mile |
| 20:10.21 | shido6 | with a serial interface |
| 20:10.30 | kenoca | yep |
| 20:10.32 | shido6 | and point the IR to the firbee's eyes |
| 20:10.32 | Zeeek | go spend the night at Neverland |
| 20:10.41 | shido6 | and we can work out a deal |
| 20:10.43 | Zeeek | but you're too old for that |
| 20:10.47 | kenoca | i don't feel like getting felt up tonight |
| 20:10.52 | kenoca | that too |
| 20:10.52 | Zeeek | heh anyway |
| 20:10.57 | [TK]D-Fender | KalD|Work : What interface do we have to this "Kirby"? |
| 20:11.14 | kenoca | phone dials fine provider incoming not so fine. |
| 20:11.15 | [TK]D-Fender | errr.. Furby |
| 20:11.32 | Zeeek | I have a zillion voIP accounts but they are all connected to my asterisk box - that's what it's good at. So my phones are all connected to it |
| 20:11.38 | KalD|Work | [TK]D-Fender, Furby - it is a little toy my friend has - it speaks spanish! We thought it'd be funny to teach it things - but it takes forever to get it to repeat things - it only has an IR port |
| 20:11.50 | Zeeek | Furbys are cool |
| 20:12.12 | firestrm | Zeeek, idea.. how about a furby sip phone? |
| 20:12.14 | Zeeek | plus they do IAX2 |
| 20:12.18 | kenoca | all the phones in the users office are connected to it and they love the system |
| 20:12.22 | *** join/#asterisk LarsAC (~chatzilla@pD9501934.dip0.t-ipconnect.de) |
| 20:12.27 | Zeeek | connected to what? |
| 20:12.32 | LarsAC | hi |
| 20:12.32 | Zeeek | yuor asterisk box? |
| 20:12.36 | firestrm | yes. |
| 20:12.39 | [TK]D-Fender | Well... you could patch in your speaker-out and use chan-alsa |
| 20:12.43 | kenoca | the people calling them are having a problem with dialing extens and the users of the system are having a hard time checking voicemail from outside the office |
| 20:12.50 | kenoca | yes the asterisk box |
| 20:13.01 | Zeeek | what version asterisk? |
| 20:13.07 | kenoca | 1.0.5 |
| 20:13.11 | *** join/#asterisk Aviator (~ask@ip-65-111-77-10.customer.accelacom.net) |
| 20:13.14 | ManxPower | kenoca: You know that's a DTMF problem, right? |
| 20:13.19 | Zeeek | you could try 1.0.6 |
| 20:13.21 | firestrm | connect to the mic and speaker in him.. and move the mouth to the outgoing sound |
| 20:13.42 | Zeeek | Manx yes |
| 20:13.42 | kenoca | maxpower 1.0.5 has a dtmf problem |
| 20:13.42 | kenoca | or just my problem in general |
| 20:13.43 | *** join/#asterisk BrainStormerToo (~bcbrown@c-24-20-119-153.client.comcast.net) |
| 20:13.48 | Zeeek | no yours |
| 20:13.54 | kenoca | ok |
| 20:14.03 | Zeeek | I just mentioned 1.0.6 since I just changed to it |
| 20:14.06 | kenoca | that is what i thought, dtmf is my problem |
| 20:14.08 | LarsAC | what needs to be done to have * listen on dyndns names ? |
| 20:14.11 | Zeeek | and it appears to work (except moh) |
| 20:14.21 | kenoca | no moh, that sucks |
| 20:14.33 | Zeeek | but kenoca-san, it seems to be a provider only problem |
| 20:14.41 | afe | will you guys kill me if I ask a question that's more about Windows/sound settings and X-pro than astersisk? |
| 20:14.44 | Zeeek | there is a patch to make it work |
| 20:14.52 | Aviator | I have an X100P card, does Asterisk generate a dialtone for mr to call into the system with an analog phone?, or is this card only for an incoming PSTN line |
| 20:14.55 | Zeeek | no because I use WInXP and X-Lite |
| 20:15.31 | Zeeek | X100P is FXO - for the phone line only |
| 20:15.34 | afe | I have a creative Audigy2 platinum pro zs (or whatever it's called) |
| 20:15.53 | Zeeek | call it a sound card |
| 20:15.58 | afe | and for some reason it keeps changing the sound input from microphone to analog mix |
| 20:16.09 | Zeeek | is it built in to the mobo? |
| 20:16.18 | kenoca | well it looks as if my asterisk version is cvs-v1-0-02 not 1.0.5 |
| 20:16.25 | Aviator | thanks, I'm also trying to get Xlite client working as a softphone, where do I set the usernames/passwords to authenticate to the Asterisk server? |
| 20:16.37 | Ahewes | I have a question about the directory in the voicemail application if anyone has time. |
| 20:16.45 | Zeeek | X-Lite happens to have an execllent manual |
| 20:16.45 | kenoca | i used the asterisk@home install, does that suck |
| 20:16.58 | Zeeek | yes if you want to learn anything about asterisk, it might |
| 20:17.03 | afe | zeek: nope, its one of those with a zillion ports (external and on the card) |
| 20:17.03 | Ahewes | I have heard that the directory can be listed by first name instead of last name, but I don't see a config option for that. |
| 20:17.25 | Ahewes | Anyone know where the option is to list the directory by first name? I'm missing it. |
| 20:17.37 | Nugget | kenoca: what's the full version string? |
| 20:17.39 | Zeeek | afe I have a bult in sound but it seems to me I constantly have to screw with it - it has built in "intelligence" |
| 20:17.52 | kenoca | <PROTECTED> |
| 20:18.06 | Nugget | that doesn't mean 1.0.2 it means 1-0 built on 15-Feb-2005. |
| 20:18.07 | Zeeek | no they all say 1-0 |
| 20:18.22 | Zeeek | prolly 1.0.5 like you said |
| 20:18.24 | kenoca | oh yeah i see that now |
| 20:18.46 | Zeeek | Ahewes isn't the option to put the names in backwards? |
| 20:18.52 | Zeeek | (no joke) |
| 20:18.53 | afe | it's not a big issue, but I hate everytime I forget to set the input to microphone, and people I call only get the sound from my TV card :P |
| 20:18.58 | BrainStormerToo | Bug or just logic, If Sipura is set to use DTMF Mode AVT+INFO, Asterisk uses both, IE. The key is bounced. |
| 20:19.03 | Zeeek | afe me too |
| 20:19.12 | *** join/#asterisk randy998 (~randy@pool-68-238-41-51.scr.east.verizon.net) |
| 20:19.25 | Zeeek | and I have a TV tuner and when I record I get no sound cause I forget about changing the source |
| 20:19.44 | *** part/#asterisk ropeguru_work (~ropeguru@141.152.37.26) |
| 20:20.08 | randy998 | anybody been getting this lately? using Broadvoice Got SIP response 404 "Not found" back from 147.135.0.128 |
| 20:20.14 | Ahewes | Zeeek actually, there is supposed to be an option that changes the prompts and the order things are listed in, so you get "Dial by first name..." as the prompt. |
| 20:20.21 | *** join/#asterisk footnote (~jhicks@67.141.135.121) |
| 20:20.26 | afe | I just tried changing the TV tuner software to use microphone for recording - that will screw up tv recroding, but maybe the phone works better :) |
| 20:20.33 | Zeeek | maybe in voicemail.conf? |
| 20:21.27 | Zeeek | the trap on mine is I can hear the sound of TV, but when I record it doesn't record unless I change something I can't remember even now what |
| 20:21.28 | Ahewes | Zeeek one would suppose, but I don't see it. |
| 20:21.36 | Zeeek | so by the time I find it, the show is over |
| 20:21.51 | Zeeek | check the mailing list, that sounds familiar to me |
| 20:22.16 | Zeeek | Directory(vm-context | dial-context | options) |
| 20:22.22 | Zeeek | wouldn't be around there? |
| 20:22.38 | Zeeek | http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Directory |
| 20:23.10 | Zeeek | options L for Last name and F for first |
| 20:23.27 | afe | zeeek will restart my computer now and see if it stays on microphone (I never record anything from the computer anyway) :) |
| 20:23.34 | Zeeek | except lower case |
| 20:23.49 | Zeeek | Ahewes? Are you not worthy? |
| 20:24.07 | Zeeek | turn east and bow three times |
| 20:24.14 | *** join/#asterisk randu (~randy@pool-68-238-41-51.scr.east.verizon.net) |
| 20:24.20 | Ahewes | AH! Thank You! |
| 20:24.43 | Zeeek | si ManxPower, even if you don't laugh at my stupid jokes, I serve as your filter from time to time :) |
| 20:24.55 | Ahewes | the crowd chants manically! Zeeek! Zeeek! Zeeek! |
| 20:24.58 | Zeeek | and you helped me with my important issue today |
| 20:25.09 | Zeeek | three bows - I am east of you I think? |
| 20:25.27 | Zeeek | God I feel emptied now |
| 20:25.31 | Zeeek | whew |
| 20:27.08 | *** join/#asterisk pht (~PaulT@xob.neospire.net) |
| 20:28.10 | *** join/#asterisk RaYmAn-Bx (user@213.237.12.147.adsl.vby.tiscali.dk) |
| 20:28.56 | Aviator | how do I setup SIP usernames/passwords for softphones to connect? |
| 20:29.12 | Zeeek | Starter tutorial: |
| 20:29.12 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 20:29.12 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 20:29.12 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 20:29.12 | Zeeek | THE reference of the moment: |
| 20:29.12 | Zeeek | http://www.asteriskdocs.org |
| 20:29.33 | G0shen | my new IAXy that I provisioned this morning(thank you Zeeek and others) is working FANTASTICLY! |
| 20:29.48 | Zeeek | Cool! What you using as a power supply? |
| 20:29.51 | *** join/#asterisk TSCHAK (~thomas@69-161-192-45.clspco.adelphia.net) |
| 20:29.55 | BrianR___ | dev.asteriskdocs.org seems to be busted again |
| 20:29.55 | G0shen | the one that came with it |
| 20:29.59 | jsolares | hehe |
| 20:30.12 | Zeeek | mine didn't so I use the radio shack |
| 20:30.24 | Zeeek | which is great cause it does 220v |
| 20:30.31 | G0shen | I found the best price at pullver store, AND it came with a power supply :) |
| 20:30.34 | jsolares | the power supply on it is almost 10 times as heavy as the iaxy :X |
| 20:30.44 | G0shen | yup! |
| 20:30.47 | __Sparks_ | Is there somewehre that lists USA Fixed line prefixes and mobile prefixes? - or is is as simple as the UK's 00447 for mobiles? |
| 20:30.52 | Zeeek | Well 1A is a lot of current |
| 20:31.27 | *** join/#asterisk eborn (~eborn@linda.fambus.nl) |
| 20:31.44 | BrianR___ | __Sparks_: US prefixes can contain a mix of mobile and non-mobile numbers. |
| 20:31.55 | BrianR___ | __Sparks_: Also, it's possible for a number to be mobile one day and fixed the next :( |
| 20:32.04 | jsolares | fun, isnt it |
| 20:32.04 | PTG123 | anyone here configured an ata186 to work with asterisk? |
| 20:32.22 | __Sparks_ | :( so I cant get Asterisk to route US mobile and fixed number differnetly then? |
| 20:32.31 | BrainStormerToo | exten => 9,2,Dial(ZAP/2/) |
| 20:32.37 | BrainStormerToo | oops. |
| 20:33.18 | BrainStormerToo | Bug or just logic, If Sipura is set to use DTMF Mode AVT+INFO, Asterisk uses both, IE. The key is bounced. |
| 20:33.23 | Zeeek | god the iaxy is cheap at pulver |
| 20:33.25 | BrianR___ | Biggest pain in the ass is that it's now impossible to easily do least-cost-routing using unlimited mobile-to-mobile service. |
| 20:33.32 | BrainStormerToo | of couse setting the sipura to info works... |
| 20:33.37 | BrianR___ | Since it's impossible to tell what numbers belong to what carriers. |
| 20:33.42 | G0shen | yea, cheap, with cheap shipping :) |
| 20:33.47 | G0shen | and comes with power supply |
| 20:33.53 | BrainStormerToo | but it'd be nice to be idiot proof too.. |
| 20:34.13 | Zeeek | why do they list "transfer" and then right under it "blind transfer" |
| 20:34.20 | Zeeek | that's bogus! |
| 20:34.30 | BrianR___ | One lets you introduce the call. The other doesn't... |
| 20:34.53 | __Sparks_ | BrianR___, there is a company here in the UK that charges nothing for calls to fixed US numbers, and GBP0.03 per minute for mobiles - how do they know what's what! |
| 20:34.54 | Zeeek | it does attended? (can't remember) |
| 20:35.10 | BrianR___ | __Sparks_: Maybe they guess? |
| 20:35.19 | __Sparks_ | are there any loose rules? |
| 20:35.21 | *** join/#asterisk afe ([Aur3q9tLE@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
| 20:35.24 | *** join/#asterisk roamer323 (~sing@67.71.60.238) |
| 20:35.25 | Zeeek | at that level they know somehow |
| 20:35.56 | BrianR___ | __Sparks_: One used to be able to assume that certain area code / npa combinations were mobile. But that hasn't been true for over a year now. |
| 20:35.57 | Zeeek | the numbers aren't even area dependent anymore |
| 20:36.22 | Zeeek | people can get a Manhattan number no matter where they are - except a lot of people want one |
| 20:36.53 | Zeeek | "These devices are a little strange to configure for the first time, but once you do one it easy" IAXy review |
| 20:37.03 | afe | zeeek: it helped to change the tv tuner to use microphone input - now X-Pro uses the right channel for input :) |
| 20:37.11 | Zeeek | oh, cool |
| 20:37.43 | __Sparks_ | BrianR___, is there a list of the old area codes somwhere? - and what are npa's? |
| 20:37.55 | Zeeek | be sure to get the detailed X-Pro manual PDF - it has like 50 pages about SIP, NAT etc |
| 20:38.40 | Zeeek | G0shen how much was the shipping? |
| 20:38.47 | afe | zeeek: why would I want that - it works perfectly fine as it is now (but it's still ugly as hell) |
| 20:39.00 | *** join/#asterisk file (~file@mctnnbsah25-142166093180.nb.aliant.net) |
| 20:39.26 | Zeeek | for the wisdom contained therein |
| 20:39.31 | afe | hehe, ok |
| 20:39.37 | __Sparks_ | BrianR___, Is it the same with Canadian numbers too? |
| 20:39.47 | Zeeek | there is no SIP phone ever with that many configurable parameters |
| 20:39.57 | G0shen | Zeeek : 8.99 I think? |
| 20:40.00 | PBXtech | how do you send a message to the manger API from AGI? |
| 20:40.06 | G0shen | they ship from NY |
| 20:40.16 | G0shen | FedEx |
| 20:40.31 | jsolares | PBXtech: you make your agi talk to the telnet port, there might be other ways |
| 20:40.31 | Zeeek | no tax, right? |
| 20:40.35 | G0shen | right |
| 20:40.46 | PBXtech | jsolares how? |
| 20:41.18 | jsolares | there are plenty of examples on the wiki for using the manager api, how? well it depends on what language you are using for the AGI |
| 20:41.28 | PBXtech | ok |
| 20:41.48 | Zeeek | the only trouble with any of these phones is getting them to work without going there to install it yourself |
| 20:41.49 | Pinhole | PBXtech, use the manager class that comes with phpagi |
| 20:41.57 | jsolares | i think perl has a module for talking directly to the manager api as well |
| 20:42.32 | Pinhole | my $ok = getopts('a:b:?', \%Options); |
| 20:42.50 | *** part/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 20:43.45 | Pinhole | if ($suffix =~ m/([^=]+)=([^,]+)/) { |
| 20:43.50 | jsolares | hehehe |
| 20:44.21 | Pinhole | it's kinda fun to grab random perl files on your system and echo out a random line. ... lol |
| 20:45.31 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 20:51.10 | *** join/#asterisk walnuck (~jack@modemcable106.82-200-24.mc.videotron.ca) |
| 20:51.27 | walnuck | what's a good modem to go along for a telephony gateway to ip? |
| 20:52.02 | jontow | walnuck; perhaps you really want an FXO card (?) |
| 20:53.26 | walnuck | jontow: i'm pretty new to this, shouldn't a regular phone modem work ? |
| 20:53.43 | km- | woo hoo, Meetme r0x |
| 20:53.55 | km- | we had 12 people connected via the T1's and the audio was pristine |
| 20:53.59 | km- | this box is a P2 266 |
| 20:54.10 | PBXtech | is there any CTI or Manager wrapper to make the events into a single line? |
| 20:54.15 | *** join/#asterisk Dr-Linux (~sshah@202.125.141.6) |
| 20:54.32 | Dr-Linux | tzanger |
| 20:54.43 | walnuck | jontow: i yet don't have a modem in the machine and I have successfully done ip-2-ip with open phone, can an fxo card help me dial-in to my station so I can make an ip call? |
| 20:55.02 | Nukemizer | Does anyone here use Mandrake as the OS for running Asterisk ? |
| 20:55.05 | bjohnson | so has anyone that has installed * on a wrt54g documented any of the steps? |
| 20:55.14 | bjohnson | Nukemizer: not too many |
| 20:55.22 | bjohnson | Nukemizer: likely one or two |
| 20:55.26 | Nukemizer | yikes |
| 20:55.41 | bjohnson | it's never mentioned |
| 20:56.42 | Nukemizer | bummer for me then I guess I will install reshad then and abandon my Mandrake on this server |
| 20:56.53 | Nukemizer | redhat :) |
| 20:57.31 | X-Gen | ja freaks© |
| 20:58.25 | [TK]D-Fender | Nukemizer : What is that PC do for you besides *? |
| 20:58.53 | Nukemizer | Fender: just * |
| 20:59.41 | jontow | walnuck; of course.. an FXO card is just 'like' a modem in the respect that it answers the line when you call it (well, asterisk does..) and then you're in the pbx.. its all dialplan configuration at that point. |
| 21:00.07 | jontow | its just like having a T1 card in your PC answering PRI-type calls.. except that its only 1 line |
| 21:00.16 | jontow | i just bought 3 of them not long ago :) |
| 21:00.23 | [TK]D-Fender | Might want to try something a little "simpler" and direct than RHunless its a support issue |
| 21:01.07 | *** part/#asterisk pht (~PaulT@xob.neospire.net) |
| 21:01.10 | *** join/#asterisk bile_one (~Bile@pcp03281999pcs.gillst01.ar.comcast.net) |
| 21:01.22 | walnuck | jontow: so if I go out shopping in my neighborhood, does it matter if I get a voice modem or a regular modem for asterisk? |
| 21:01.29 | jontow | yes |
| 21:01.32 | jontow | they probably will not work |
| 21:01.42 | jontow | you probably won't find what you need in your neighborhood :) |
| 21:01.48 | Nukemizer | I haev been instructed by Digium support to load RH to see why PRI will not stay up.. I can get my TE11oP card to work as a clear channel T1 just not PRI |
| 21:01.48 | JonR800 | im too stupid to use a real package manager :) |
| 21:02.30 | [TK]D-Fender | Nukemizer : Oh, well do as they say then :) |
| 21:03.51 | footnote | jontow |
| 21:03.57 | afe | zeeek: darn... it didn't work. If the tuner uses the microphone X-Pro won't work at all :/ |
| 21:04.08 | walnuck | jontow: i don't have a t1 line, i don't see how I can make use of an fxo card |
| 21:04.40 | X-Gen | Nukemizer: 'not stay up' what do u mean ?u get some evt in the log saying it is now down ? |
| 21:04.43 | footnote | #freebsd jontow? |
| 21:04.44 | [TK]D-Fender | FXO is for a regular line, not a T1 |
| 21:05.14 | walnuck | ok |
| 21:05.31 | walnuck | so - would it be that an fxs can receive as well as the fxo? |
| 21:05.36 | [TK]D-Fender | For instance I use a Digium TDM22B Card. 2 FX, 2 FXS. That means I can plug in 2 regular lines from the telco into it, and 2 regular phones as extensions |
| 21:05.44 | Nukemizer | X-Gen : I get Red alamrs and must run the PBX with no timming if I have any hopes of grabbing a B chennel |
| 21:06.01 | [TK]D-Fender | FXS cards let you plug in phones. FXO cards let you plug in LINES. |
| 21:06.07 | *** join/#asterisk csim (~root@mtnuser.icgws.com) |
| 21:06.10 | *** join/#asterisk buddah (~hnic@208.179.86.5) |
| 21:06.15 | Nukemizer | X-Gen: D channel goes up and down constantly |
| 21:06.35 | X-Gen | Nukemizer: u sure it is not a cable issue ? |
| 21:06.56 | walnuck | [TK]D-Fender: when you say 2 FX, what are you referring to? |
| 21:07.07 | [TK]D-Fender | 2 FXO. type. |
| 21:07.09 | [TK]D-Fender | typo( |
| 21:07.11 | [TK]D-Fender | * |
| 21:07.12 | [TK]D-Fender | ugh! |
| 21:07.17 | Nukemizer | yeah, get wierdness even with loop back |
| 21:07.23 | walnuck | [TK]D-Fender: what does the S stand for in fxs? |
| 21:07.38 | eKo1 | service |
| 21:07.48 | [TK]D-Fender | FXS = Foreign Exchange Station. FXO = Foreign Exchange Office |
| 21:07.59 | [TK]D-Fender | eKo1 : close |
| 21:08.00 | csim | does anyone know where I can find openh323 v1.15.1 and the appropriate pwlib source as well? |
| 21:08.06 | eKo1 | oh well |
| 21:08.11 | Nukemizer | X-Gen: if I put the card in E&M Wink mode, the T1 works |
| 21:08.14 | *** join/#asterisk andrewct (~root@jumbo.ntplx.net) |
| 21:08.23 | walnuck | [TK]D-Fender: are they protocols or defactos? |
| 21:08.24 | eKo1 | Too many acronyms to keep track of. |
| 21:08.27 | Nukemizer | which should rule out the xover calble |
| 21:08.45 | andrewct | Is there an issue with MOH for parking transfers? |
| 21:09.19 | [TK]D-Fender | FXS ports PROVIDE voltage to the port. Thats what PHONES need, something to drive them. FXO ports require voltage (as provided by your telco). |
| 21:09.36 | Nukemizer | But -- what the heck do I know tis a learning proccess this journey is :) I just gratefull for help where i can get it |
| 21:09.46 | footnote | ok, lemme see if i can make postgres cdr's work |
| 21:09.46 | [TK]D-Fender | PLug them in wrong and BZORTCH (afaik) |
| 21:10.05 | walnuck | [TK]D-Fender: so I basically can't get a card that does fxs and fxo? |
| 21:10.09 | csim | [TK]D-Fender any chance you might know the answer to my qustion? |
| 21:10.10 | jontow | footnote; yes, that is me. |
| 21:10.42 | andrewct | A port is FXS or FXO, You can get cards that have BOTH FXS and FXO ports |
| 21:10.51 | jontow | yeah.. don't use cheap dish-detergent as handwashing soap.. it makes your hands smell funny.. just for reference :/ |
| 21:10.52 | andrewct | You can get an FXO<-->FXS converter box |
| 21:10.53 | [TK]D-Fender | walnuck : Digium's TDM400 series cards hold 4 modules. They can be any mix of FXO/FXS |
| 21:10.55 | jontow | oh.. woah :D hi |
| 21:10.59 | footnote | hey man |
| 21:11.16 | walnuck | andrewct: so what is an fxo card, are they usually towards fxo-> or fxs-> oriented ? |
| 21:11.16 | [TK]D-Fender | csim : Maybe, ask away |
| 21:11.22 | csim | does anyone know where I can find openh323 v1.15.1 and the appropriate pwlib source as well? |
| 21:11.27 | *** join/#asterisk neopher (~crazy@mail.techhelpresources.com) |
| 21:11.32 | andrewct | A |
| 21:11.37 | footnote | csim: sorta old ain't it? |
| 21:11.38 | bjohnson | |Vulture|: did you build * on your wrt or is there a package? Do you do nfs mounts for logs/cdr records, etc? |
| 21:11.40 | csim | just trying to get h323 support working |
| 21:11.45 | [TK]D-Fender | csim : No idea, never touched H.323 since learning about SIP/IAX |
| 21:11.54 | Nugget | ask again in another three minutes. |
| 21:11.58 | andrewct | The Digium TDM400 card has for slots, each slot can have an FXS or FXO port module |
| 21:11.59 | [TK]D-Fender | walnuck : What is your goal? |
| 21:12.05 | jontow | FXO = plug a phone line into it.. it does all of the signalling voodoo that one has to do on a phone line.. from there you simply let asterisk know "i have an fxo card" a la zaptel drivers (wctfxo) and set it up via the wiki's notes |
| 21:12.08 | footnote | csim: hrm. 1.0.6 build isn't working? |
| 21:12.09 | csim | according to the readme in the /etc/asterisk/chanels/h323 folder |
| 21:12.17 | footnote | -stable? |
| 21:12.29 | jontow | i'd really like a TDM400 :/ but instead i have cheap X100P clones |
| 21:12.29 | walnuck | [TK]D-Fender: my goal is to extend my ip2ip connection I can make with open phone with my regular phone- possibly a third party caller |
| 21:12.39 | csim | according to the readme only that specific version is supported unless i am mistaken |
| 21:12.47 | jontow | ... but they really were cheap |
| 21:12.57 | jontow | paid $40 for 3 of them, vs. a pile for a TDM400 and the modules |
| 21:12.58 | neopher | hmmm, tring to get G.729 to work, asterisk sees it registered but when i make calls it is not being used, and yes, i did buy a licence from digium |
| 21:13.01 | bile_one | jontow, I got my TDM40P installed and I get a dial tone, I can call it from a sip, but I can't call a sip? |
| 21:13.08 | [TK]D-Fender | What is "ip2ip"? |
| 21:13.08 | csim | is anyone is here using h323 on asterisk? |
| 21:13.24 | jontow | bile_one; what modules? |
| 21:13.29 | bile_one | FXS |
| 21:13.38 | footnote | csim: hang with me, i'm gonna dig in |
| 21:13.44 | PBXtech | i see lots of examples but i dont see anything on sending a command to an already connected Manager API connection from AGI.. grr |
| 21:13.46 | jontow | ok, must be dial-plan issues? i haven't ever configured FXS cards |
| 21:13.46 | bile_one | jontow, FXS is a 40B |
| 21:13.48 | csim | ryte on |
| 21:13.54 | X-Gen | Nukemizer: i hope u get your problem solve, sorry i cant help :( i had issues with getting my E1 working, but it was just the learning curve |
| 21:13.54 | footnote | brb |
| 21:14.09 | bile_one | jontow, sorry I typed P instead. my mistake |
| 21:14.12 | jontow | :) np |
| 21:14.26 | walnuck | [TK]D-Fender: o, I meant that since I have done an ip 2 ip conversation with openphone (openh323.org) client, I'd like to extend the conversation to someone who could be calling in from a regular phone |
| 21:14.28 | *** join/#asterisk afe ([T+Fm4ttmr@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
| 21:14.52 | Nukemizer | X-Gen: am thankful for guidance , appricate the help |
| 21:15.20 | footnote | csim: how did you get the sources you're building from? |
| 21:15.31 | csim | the asterisk srouces? |
| 21:15.36 | footnote | yeah |
| 21:15.36 | [TK]D-Fender | walnuck : the point of being here is pretty much to get rid of open H323 and go SIP between IP softphones and then use an FXO card to have access to a land line |
| 21:15.43 | csim | cvs |
| 21:15.47 | footnote | csim: good, k |
| 21:16.04 | footnote | csim: i just want to make sure we're looking at the same stuff :) |
| 21:16.13 | csim | ryte on |
| 21:16.29 | *** join/#asterisk dalabera (~Dalabera@mail.pmrtechnologies.com) |
| 21:16.31 | *** join/#asterisk redder86 (~lee@gateway.howardsilvan.com) |
| 21:16.51 | redder86 | JerJer: around? |
| 21:16.55 | redder86 | shido6: around? |
| 21:16.57 | walnuck | [TK]D-Fender: well, I don't have the intention of coding a project- I'm here to understand how I can create this sort of gateway |
| 21:17.08 | afe | <PROTECTED> |
| 21:17.18 | bjohnson | walnuck: some people have been able to get the h323 code to work with * but it is a big pita |
| 21:17.19 | csim | walnuck I'm trying to do somewhat the same thing |
| 21:17.26 | walnuck | [TK]D-Fender: I'm here to try asterisk :) |
| 21:17.27 | afe | 22:16 < redder86> shido6: around? |
| 21:17.39 | csim | but i am trying to send calls via h323 carrier |
| 21:17.44 | bjohnson | walnuck: most people find it easier to change their software to SIP or IAX softphones than to get * to work with H323 |
| 21:17.45 | [TK]D-Fender | walnuck : * does this all pretty easy. |
| 21:18.03 | footnote | csim: damn phone, brb |
| 21:18.07 | [TK]D-Fender | h323 = PITA |
| 21:18.11 | csim | ryte on |
| 21:18.15 | csim | PITA? |
| 21:18.16 | walnuck | [TK]D-Fender: h323 vs what? |
| 21:18.24 | *** part/#asterisk redder86 (~lee@gateway.howardsilvan.com) |
| 21:18.47 | neopher | Is there a way to change the codec prority |
| 21:18.53 | walnuck | [TK]D-Fender: I've read that in the docs, that it is difficult to setup |
| 21:18.55 | csim | [TK]D-Fender what are the real advantages of using sip over h323 on a private network? |
| 21:19.21 | walnuck | [TK]D-Fender: do I look into SIP then to link the idea I'm trying to tell you? |
| 21:19.29 | Grooby | ping |
| 21:19.34 | Grooby | wierd |
| 21:19.40 | [TK]D-Fender | Pain In The Ass |
| 21:19.41 | csim | its /ping |
| 21:19.45 | csim | ahh |
| 21:19.47 | csim | thnx |
| 21:19.50 | Grooby | anyone know of spandsp will work with sip channels? |
| 21:19.53 | csim | I agree |
| 21:19.58 | walnuck | [TK]D-Fender: pita, yes, I thought you were referring to mexican flatcakes.. |
| 21:20.12 | csim | but none the less I need to make thwem work together if I am ever going to make the switch |
| 21:20.18 | csim | I have an existing h323 network |
| 21:20.19 | [TK]D-Fender | walnuck : Yes, ditch the H.323 clients, go with SIP clients (easy), get an FXO card/ATA and you're set. |
| 21:20.23 | neopher | Grooby: u need t.38 support |
| 21:20.25 | JonR800 | Grooby: probably not.. maybe on a lan.. sometimes.. :) |
| 21:20.30 | walnuck | [TK]D-Fender: can the bot spit? |
| 21:20.37 | Grooby | that sucks |
| 21:20.39 | Grooby | :( |
| 21:20.53 | *** join/#asterisk jaxxan (~jaxxan@202.70.125.109) |
| 21:20.54 | [TK]D-Fender | walnuck : ?! |
| 21:20.58 | jaxxan | hey ya'll |
| 21:21.20 | walnuck | [TK]D-Fender: its okay, i suppose there's not a bot- does sip offer security? |
| 21:21.52 | neopher | Grooby: once someone fills the bounty for t.38 you will be able to |
| 21:21.54 | csim | it can be sniffed just as easy as h323 |
| 21:22.00 | [TK]D-Fender | inherently? No. There are ways of encrypting the channel, but I don't know the details. |
| 21:22.09 | [TK]D-Fender | What csim said :D |
| 21:22.29 | csim | :P |
| 21:22.50 | *** join/#asterisk jsmith (~jsmith@pip.drs1.omniture.com) |
| 21:22.52 | bjohnson | I don't know h323 |
| 21:22.52 | *** join/#asterisk Koshatul (~evangelio@202.9.38.223) |
| 21:22.54 | *** join/#asterisk miguellinux (~miguellin@200.47.223.190) |
| 21:22.56 | neopher | ahhhhh, stupid call won't use G.729 |
| 21:22.59 | Grooby | thanks guys |
| 21:23.03 | neopher | np |
| 21:23.06 | walnuck | [TK]D-Fender: but say sip doesn't work out, I can still make use of iax or h323 right? |
| 21:23.10 | bjohnson | SIP is a little tricky on encryption becuse it's not just one port |
| 21:23.13 | Grooby | i was beating my head against the wall trying to figure out why spandsp is not detecting my incoming fax |
| 21:23.13 | Grooby | hehehe |
| 21:23.21 | Grooby | I guess that's for zapa device only |
| 21:23.29 | miguellinux | hi, anyone uses broadvoice on asterisk? |
| 21:23.34 | *** part/#asterisk randu (~randy@pool-68-238-41-51.scr.east.verizon.net) |
| 21:23.35 | *** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
| 21:23.37 | Grooby | i use BV w/ * |
| 21:23.39 | bjohnson | walnuck: unless you REALLY need h323 .. use SIP and IAX |
| 21:23.46 | neopher | Grooby: yes, spandsp is zap only |
| 21:23.49 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 21:23.57 | [TK]D-Fender | walnuck : SIP will work as easily as H.323 if not better. |
| 21:24.09 | neopher | Grooby: me too, BV rocks |
| 21:24.17 | miguellinux | Hi Grooby any troubles with that? |
| 21:24.17 | [TK]D-Fender | H.323 was an idea whose time came and went when NAT came along. |
| 21:24.19 | csim | but can someone here tell me where to find the sources code for openh323 and pwlib ? |
| 21:24.28 | Grooby | not really..been w/ them for couple months now |
| 21:24.36 | andrewct | Yes, BV works with asterisk. |
| 21:24.38 | walnuck | [TK]D-Fender: shouldn't i go with iax, since it seems built for asterisk? |
| 21:24.42 | andrewct | I have both local lines and BV |
| 21:24.42 | [TK]D-Fender | Forget H.323. That idea took a short while for me to understand but the whole world is aligning that way. |
| 21:24.43 | G0shen | on my IAXy I hear a beep every few minutes, it sounds like the beep that cordless phones make when they are about to run out of batteries |
| 21:24.49 | neopher | Grooby: Don't know if you know this, they support G.729 as i was told by one of their reps |
| 21:25.01 | Grooby | no kidding |
| 21:25.05 | [TK]D-Fender | walnuck :IAX is good, SIP is a gaurantee |
| 21:25.07 | walnuck | [TK]D-Fender: is this only here, or is that a trending norm that h323 should "go" ? |
| 21:25.18 | Grooby | neopher, did you get that to work? also don't you need license for that? |
| 21:25.24 | neopher | tring to get it to work now |
| 21:25.26 | [TK]D-Fender | if you can influence both side, IAX through FireFly is the easiest way |
| 21:25.28 | G0shen | SIP is to VCR, as IAX is to DVD :) |
| 21:25.35 | csim | [tk]d-fender I agree but I need to be able to get asterisk to work with h323 because I already have an existing h323 network I need to make this work, its my first step to killing off h323 |
| 21:25.47 | neopher | i have a couple licences for G.729 |
| 21:25.56 | miguellinux | Grooby, did you patch the source code? which is the failure if not patch? |
| 21:25.56 | [TK]D-Fender | walnuck : Worldwide trend. 99% of VoIP providers = SIP. Ask yourself why...... |
| 21:26.01 | walnuck | G0shen: o really? so IAX is more of a >>pita to setup? |
| 21:26.07 | Grooby | no need to patch |
| 21:26.08 | walnuck | hmm |
| 21:26.17 | neopher | is cheap, digium sells for $10 a channel |
| 21:26.22 | Grooby | no kidding |
| 21:26.29 | [TK]D-Fender | csim : Good luck with that :/ a challenge I'm sure |
| 21:26.29 | csim | I like IAX better than sip |
| 21:26.36 | Grooby | i'll have to look into it |
| 21:26.37 | csim | but no one else supports it |
| 21:26.49 | hardwire | its good for inter .. asterisk .. |
| 21:26.52 | eKo1 | $10 is a lot. I can buy myself two happy meals with that. |
| 21:26.52 | csim | yes |
| 21:26.52 | G0shen | I got my first IAXy today, and I had it up and configured in 15 minutes, even thought I had never done it |
| 21:26.57 | csim | saves bandwidth |
| 21:26.57 | neopher | I'll let you know when i get it wirking |
| 21:27.07 | CleanerX | i found a bug in ast_data - shell I put this into the bugtracker? |
| 21:27.08 | miguellinux | Grooby, because I follow the instructions I can call but hear nothing |
| 21:27.11 | andrewct | anyone use parking here? |
| 21:27.25 | eKo1 | andrewct: I use valetparking. |
| 21:27.27 | [TK]D-Fender | csim : thats why SIP is still king. |
| 21:27.39 | neopher | I don't hear the beeps either, but i use a grandstream with * and BV |
| 21:27.42 | andrewct | How about just what's included by default? |
| 21:27.46 | csim | until asterisk takes over |
| 21:27.47 | csim | lol |
| 21:27.56 | miguellinux | Grooby, I can call, it rangs, but when the call is answer... I hear nothing... |
| 21:28.02 | Grooby | miguel, google for braodvoice and asterisk. |
| 21:28.02 | eKo1 | andrewct: nah, valetparking is better. |
| 21:28.10 | G0shen | Monitor doesn't play recording beeps in the call does it? |
| 21:28.19 | Grooby | also are your ports open? NAT properly and such? |
| 21:29.07 | walnuck | i don't like the fact that the defn of iax mentions udp, udp means anyone can listen to your conversation.. |
| 21:29.17 | csim | hey walnuck |
| 21:29.17 | miguellinux | Grooby, that is what I did... but... nobody talks about the supported codecs by BV |
| 21:29.36 | Grooby | currently ulaw |
| 21:29.41 | andrewct | I only use Ulaw with BV because they only seem to support inband DTMF |
| 21:29.45 | csim | the easieswt thing to do is use a cisco ip2ip to transcode from h323 to sip back to ur * box |
| 21:29.48 | *** join/#asterisk nextime (~nextime@ns0.nexlab.net) |
| 21:29.57 | [TK]D-Fender | walnuck : UDP is required to ensure voice quality. TCP breaks too easily worldwide to guarantee your connections don't time out and lag like a bitch |
| 21:30.01 | walnuck | G0shen: outside calls have been tested already? |
| 21:30.29 | walnuck | [TK]D-Fender: I know..but I'd like to try both routes :).. |
| 21:30.36 | [TK]D-Fender | walnuck : Who would be using the "soft-phones" in your scenario? |
| 21:30.52 | walnuck | [TK]D-Fender: does IAX offer encryption services? if not I guess I can find someway to wrap it if necessary |
| 21:31.13 | footnote | hrm |
| 21:31.16 | neopher | Digiums PBX is messed up, keep calling and get hung up on |
| 21:31.17 | [TK]D-Fender | walnuck : if you're paranoind, use an IPSec VPN |
| 21:31.35 | [TK]D-Fender | or something even more esoteric like Open VPN |
| 21:31.42 | *** join/#asterisk zoa (~zoa@ip-212-239-162-26.dsl.scarlet.be) |
| 21:31.44 | zoa | help |
| 21:31.47 | footnote | What is the nature of the 'h323 is a PITA' proble,? |
| 21:31.50 | kram | neophor, what are you calling? |
| 21:31.52 | zoa | any digium people here |
| 21:31.54 | zoa | aaah kram |
| 21:31.55 | zoa | help kram |
| 21:31.59 | footnote | er, problem. |
| 21:32.03 | miguellinux | Grooby, and you have tried BV on Sipura? |
| 21:32.10 | Grooby | yup |
| 21:32.16 | neopher | 877-5468963 |
| 21:32.21 | footnote | Is it "code is b0rk3n?" |
| 21:32.32 | eKo1 | footnote: Lack of support. |
| 21:32.43 | file | VON! |
| 21:32.44 | footnote | eKol: sounds like an opportunity! |
| 21:32.45 | file | VON VON VON |
| 21:32.48 | footnote | woop |
| 21:32.49 | bkw_ | VON |
| 21:32.54 | km- | VON! |
| 21:32.55 | km- | VON? |
| 21:32.57 | eKo1 | footnote: for what? |
| 21:33.04 | footnote | eKol: to support it! |
| 21:33.07 | footnote | woohoo |
| 21:33.10 | Grooby | i use voxilla.com's tool to configure my SPA |
| 21:33.13 | km- | file: hey, havent chatted with you in ages! What's cookin? |
| 21:33.15 | bkw_ | file I have a fist full of dollar bills.. NOW DANCE BOI DANCE!!! |
| 21:33.16 | andrewct | BKW is MOH with a parking transfer broken....it does not seem to work any more for me... |
| 21:33.18 | Grooby | was using Spa2k w/ BV before i got my * running |
| 21:33.22 | footnote | hehe |
| 21:33.24 | file | bkw_: NOOOOOOOO! |
| 21:33.30 | bkw_ | andrewct, how are you doing it? |
| 21:33.32 | file | km-: getting ready for VON |
| 21:33.38 | km- | file: rock on. |
| 21:33.42 | bkw_ | andrewct, and let me guess your music class is not called "default" |
| 21:33.56 | miguellinux | Grooby, Im getting crazy with BV, they use to take control of the sipura externally, but when I ask for the SIP parameters... everithing goes wrong... |
| 21:34.16 | walnuck | what about in terms of creating a voice menu- can I adjust asterisk to work with other software or can asterisk already supply this? |
| 21:34.23 | file | bkw_: my Hitachi got delivered to the neighbor's house |
| 21:34.30 | walnuck | anybody? |
| 21:34.32 | miguellinux | Grooby, the most extrange problem was the ability to call toll fre numbers but not normal numbers... !!???!? |
| 21:34.33 | bkw_ | walnuck, you need to go read |
| 21:34.35 | [TK]D-Fender | walnuck : * does everything AND the kitchen sink. |
| 21:34.35 | bkw_ | yes it can do this already |
| 21:34.38 | andrewct | I have a call, I transfer to 700, it tells me 701 and plays music. I hear music, they hear music, I finish the transfer to parking and the other end drops the music. |
| 21:34.50 | Grooby | miguel, look at voxllia.com |
| 21:34.50 | [TK]D-Fender | walnuck : it even make me COFFEE. I'll explain that when I get home in 15 mins |
| 21:34.54 | bkw_ | andrewct, IS YOUR MUSIC class called "default" |
| 21:34.54 | km- | haha, you guys know whats funny... watching the console in asterisk while my coworkers who are working from home, they start dialing themselves from X-lite, then they call their extension and answer it via X-Lite, I can only imagine the "Whoa, that's cool!" that's coming out of their mouths |
| 21:34.56 | bkw_ | or not |
| 21:34.58 | Grooby | they might be able to help you more w/ it |
| 21:35.08 | andrewct | YES, it is "default" |
| 21:35.15 | bkw_ | what version of asterisk? |
| 21:35.21 | miguellinux | Grooby, ok thanks... |
| 21:35.22 | PTG123 | anyone here use cisco ata186s? |
| 21:35.27 | km- | wait |
| 21:35.30 | zoa | brian!!!! |
| 21:35.31 | km- | why would andrewct hear music |
| 21:35.34 | andrewct | <PROTECTED> |
| 21:35.37 | CleanerX | bkw_, do you know who supports ast_data? |
| 21:35.39 | bkw_ | zoa go to #clueniversity |
| 21:35.46 | andrewct | default => custom:/var/lib/asterisk/moh-raw,/usr/bin/rawplayer |
| 21:35.48 | bkw_ | CleanerX, rm -rf usually supports it on my system |
| 21:35.55 | bkw_ | andrewct, ding ding ding cvs-head |
| 21:35.57 | bkw_ | ok |
| 21:36.03 | CleanerX | :-( |
| 21:36.06 | km- | bkw: shouldn't andrew not get Moh? |
| 21:36.07 | walnuck | bkw_: http://asterisk.org/index.php?menu=features - I don't see a faq on answering services, that's why I'm asking |
| 21:36.17 | km- | his caller should, yes |
| 21:36.22 | km- | but he should not be hearing it himself, right? |
| 21:36.31 | file | walnuck: it's called an IVR |
| 21:36.44 | andrewct | My caller should still hear music while they wait in parking |
| 21:36.50 | Grooby | i am off guys |
| 21:36.50 | bkw_ | walnuck, does voicemail not cover it? |
| 21:36.53 | *** part/#asterisk Grooby (~Grooby@12.22.232.212) |
| 21:37.07 | bkw_ | file, you tell em |
| 21:37.27 | km- | bkw: you still work at that ISP? |
| 21:37.29 | walnuck | bkw_: no i guess file was smart enough- IVR |
| 21:37.39 | walnuck | acronyms..they're fun! |
| 21:38.11 | kram | where did neophor go? |
| 21:38.29 | km- | hey hey markster |
| 21:38.35 | file | I must go get my phone |
| 21:39.18 | km- | kram: can I carve out a few minutes of your schedule in the next week or two? I'm having a T1 oddity that tzanger was looking into -- for some reason asterisk isn't winking |
| 21:39.25 | km- | kram: even though I have em_w in the settings |
| 21:39.26 | |Vulture| | wow ZapBarge is great :P |
| 21:39.29 | andrewct | bkm, so I should just open a bug on it |
| 21:39.34 | kram | km- did you try calling tech support |
| 21:39.49 | km- | kram: no, I didn't think it was a card issue but an asterisk issue |
| 21:39.57 | kram | go through tech support |
| 21:40.00 | kram | please |
| 21:40.04 | Juggie | krm, are u in von toronto for the entire 3 days? |
| 21:40.04 | km- | kram: sure, I'll do that. |
| 21:40.09 | Juggie | or just the day u talk |
| 21:40.13 | kram | if it involves the digium card and talking to something, that's the way to go |
| 21:40.30 | Juggie | er, kram i ment, not krm |
| 21:40.42 | walnuck | ok, so I'm on Fedora core 3, should I do a custom install- (compile/make) or go for a package? The issue now with SELinux is that it might interfere, anyone have a feedback on this? |
| 21:40.46 | km- | kram: just didnt think the people at tech support would know stuff as arcane as why I've got state 15/2 15/1 15/1 as opposed to 0/1 1/1 0/1, you know |
| 21:40.59 | km- | kram: but if you've got crack tech team there I'll try it after tzanger and I bang at it a bit more |
| 21:41.08 | Juggie | walnuck, compile from source. |
| 21:41.15 | Juggie | follow instructions on the wiki for FC3 |
| 21:41.24 | firestrm | walnuck, SeLinux is nothing but trouble on FC3 IMHO |
| 21:41.32 | walnuck | Juggie: ok, but I'm assuming you're not using selinux though.. |
| 21:41.34 | KalD|Work | kram, give your rma folks a raise! they kick ass |
| 21:41.36 | Juggie | yah, u can disable SELinux when u install. |
| 21:41.57 | Juggie | i didnt but i got head working with little problem |
| 21:42.12 | walnuck | firestrm: i know, I had to bitch around why UserDir on apache was failing- SELinux had to modified to allow userdirectory access.. SELinux requires *reboot >:| |
| 21:42.20 | Juggie | except for like the network, usb, hdd, and zaptel all shareing irq 9 :) |
| 21:42.21 | firestrm | walnuck, stay away from LVM as well.. makes trashed filesystems very well.. |
| 21:42.30 | Juggie | which works but has lots of misses |
| 21:42.35 | Juggie | i have to go back to the lab to fix it |
| 21:42.50 | PBXtech | In the manager interface, how do i disable sending events EXCEPT the UserEvents i send? |
| 21:42.59 | walnuck | firestrm: I did a fc3 install on a machine, it went fine on with lvm.. |
| 21:43.05 | |Vulture| | hey anyone know the name of that patch that bkw_ did that allows you to ZapBarge style a SIP/IAX2 channel? |
| 21:43.15 | Juggie | PBXtech, i dont think you can selectively chooze the events you see |
| 21:43.26 | firestrm | walnuck, it allways works at first.. then data starts getting corrupted |
| 21:43.27 | Juggie | although there was some talk of that on the bugs list i think |
| 21:43.39 | PBXtech | grr |
| 21:43.40 | PBXtech | <PROTECTED> |
| 21:43.51 | firestrm | walnuck, then your are screwed because recovery is painful and sketchy at best |
| 21:44.22 | PBXtech | would be nice to turn off the events and send custom events from agi somehow but i cant figure that either |
| 21:44.33 | Juggie | for * theres no reason you coudnt do Raid1 |
| 21:44.36 | Juggie | its not hdd intensive |
| 21:44.39 | Juggie | and it doesnt need alot of space |
| 21:44.45 | Juggie | so two hdds in mirror would be fine |
| 21:44.47 | Pinhole | phpagi manager interface allows you to select the events you want. |
| 21:45.03 | PBXtech | you got a url Pinhole I cant find anything about it |
| 21:45.19 | Juggie | its on sourceforge i think but the web page is down |
| 21:45.22 | Pinhole | phpagi.sourceforge.net get version 2 out of cvs |
| 21:45.36 | walnuck | Juggie: where's that wiki? |
| 21:46.11 | Juggie | www.voip-info.org |
| 21:46.29 | PBXtech | so does phpagi do just that? you turn off the asterisk events and let phpagi send custom events? |
| 21:46.31 | heison | does anyone here use app_postgresql? |
| 21:46.39 | Juggie | walnuck, i think u have to fix the udev in the make file, and edit some selinux permissions for fc3 |
| 21:46.43 | |Vulture| | Anyone know where ChanSpy went? its off mantis |
| 21:46.52 | Juggie | though in cvs-head the udev editing may be resolved. |
| 21:46.56 | Pinhole | PBXtech, it allows you to filter events and do things with the ones you care about. |
| 21:47.01 | cbachman | wow, stripping .o's and then linking then stripping results in much smaller than if you just link and then strip. |
| 21:47.12 | Pinhole | It also allows you to connect to the manager interface from an agi. |
| 21:47.24 | PBXtech | thanks Pinhole |
| 21:47.43 | Juggie | why would u want to connect to the manager if from an agi? |
| 21:48.19 | Pinhole | I want to get information about a peer. agi wont give me that information. the manager will. |
| 21:49.17 | *** join/#asterisk sd-tux (~Shadow@emasq.stusta.mhn.de) |
| 21:49.17 | Juggie | why not just pass it into the agi |
| 21:49.18 | heison | i've uncommented APPS+=app_sql_postgres.so in apps/Makefile, recompiled, made install; just before I restart asterisk, I did a show applications and see PGSQL, is that supposed to replace what I'm trying to do (obselete)? Or is something totally different? |
| 21:49.58 | Pinhole | how do I pass the ip address of the peer to the agi? |
| 21:50.06 | walnuck | Juggie: I'd like an ivr setup with my car, so I can tell my car to startup her engines and blink the lights, can this be done? |
| 21:50.23 | Juggie | hahahah |
| 21:50.31 | *** join/#asterisk [TK]D-Fender (~chatzilla@66.11.164.239) |
| 21:50.36 | Juggie | wheres my crack pipe |
| 21:50.55 | *** join/#asterisk darby_t (~tom@dnc209.neoplus.adsl.tpnet.pl) |
| 21:50.58 | walnuck | Juggie: [TK]D-Fender: said he would mention something like this when he returns, coincidently he must be drinking his coffee by now.. |
| 21:51.06 | [TK]D-Fender | hehe |
| 21:51.11 | walnuck | [TK]D-Fender: how's the coffee, so did it work? |
| 21:51.12 | *** join/#asterisk n4y (~tmalkut@fire2.orasoft.net.pl) |
| 21:51.32 | [TK]D-Fender | I use a lot of X-10 home automation stuff and can control my lights, door buzzer, and coffe with it :D |
| 21:51.42 | Juggie | but not a car :P |
| 21:51.43 | Juggie | hah |
| 21:51.53 | *** join/#asterisk Goldenear (~goldenear@d193.dhcp212-198-200.noos.fr) |
| 21:52.03 | walnuck | [TK]D-Fender: you should get a satellite phone to this channel, tell us about the X10 glory, it would be overkill :) |
| 21:52.35 | [TK]D-Fender | You forgot to add "route the call through 15 international exchanges, and a tin-cup-string setup) ;: |
| 21:52.40 | [TK]D-Fender | ;) |
| 21:52.51 | PBXtech | [Pinhole]: so this phpagi connects to the manager api and filters and my app server would connect to the phpagi? |
| 21:53.16 | walnuck | just don't go crazy with defense mechanisms, apparently I heard someone triggering Tasers and injuring innocent children in the homes.. |
| 21:53.18 | Juggie | i think using php to connect to the manager interface from within agi |
| 21:53.21 | bjohnson | |Vulture|: did you build * on your wrt or is there a package? Do you do nfs mounts for logs/cdr records, etc? |
| 21:53.23 | Juggie | is asking for trouble |
| 21:54.13 | Cherebrum | you could use NFS |
| 21:54.16 | Cherebrum | I prefer MySQL |
| 21:54.26 | [TK]D-Fender | walnuck: basically * is a full-bown PBX. You can use it as a simply softphome exchange system that loks as "sumb" as netmeeting beteen users, or to just about anytihing you can envision. |
| 21:54.27 | Pinhole | you can use the manager standalone or from within an agi. The manager uses call-backs for events, so you set up a call-back for the events you care about. |
| 21:54.32 | walnuck | [TK]D-Fender: does SELinux cause a problem with asterisk? |
| 21:55.00 | Cherebrum | Cisco 7912 is JUNK! |
| 21:55.11 | [TK]D-Fender | Depends on your SEL policies. its as strict as you are. By default with FC3 I presume it might be. But never having use SEL personall, couldn't verify |
| 21:55.34 | PBXtech | hmm not sure if i like the extra step in the process like that :/ |
| 21:55.35 | walnuck | [TK]D-Fender: naked babe-lines... but I have some better vision than that..at the moment I'd keep it simple ..I don't want to kill a man :) |
| 21:55.38 | [TK]D-Fender | SEL should be understood well before being implemented |
| 21:55.56 | Juggie | it worked for me without any unsurmountable problems |
| 21:56.05 | [TK]D-Fender | Tasers are NON lethat! (at least that what the box said....) |
| 21:56.06 | tuxinator_linux | Cherebrum: What is wrong with the 7912 |
| 21:56.37 | tuxinator_linux | Tasers are made in my state |
| 21:56.46 | [TK]D-Fender | lethal. Beside which, there is very little one can't do with really, Really, REALLY....... BIG LASERS |
| 21:57.12 | [TK]D-Fender | X-10 stuff is fun though. Dial-A-Light..... |
| 21:57.17 | walnuck | [TK]D-Fender: well I love exagerrating, but I think NOW that they may become so popular that the TASER makers are afraid, legislation is on demand by police forces saying they can't control crowds that bear them and are labelling them as a Lethal weapon.. |
| 21:57.39 | *** join/#asterisk LarsAC (~chatzilla@pD95004EB.dip0.t-ipconnect.de) |
| 21:57.45 | footnote | Cops always say stuff like that |
| 21:57.50 | Goldenear | I have a question about notransfer=no in iax.conf : how does asterisk know when it has to directly connect two IAX client or not ? |
| 21:57.59 | PBXtech | im worried if/when you get 100 calls sending events that the port will bog down. |
| 21:58.00 | walnuck | [TK]D-Fender: according to some yahoo news router story.. doesn't make a good bedtime story.. but the police hate citizens carrying it around.. |
| 21:58.09 | LarsAC | do i have to do something in extensions.conf to route a call to a mailbox ? |
| 21:58.14 | [TK]D-Fender | US Legislation = suck. Doomed to create the wussiest of humanity who need to be "protected" from EVERYTHING. |
| 21:58.27 | footnote | [TK]D-Fender: amen |
| 21:58.32 | tuxinator_linux | I agree [TK]D-Fender |
| 21:58.32 | footnote | buncha sissies |
| 21:58.50 | [TK]D-Fender | its 1984 & Brave New World, all in one..... |
| 21:59.02 | walnuck | [TK]D-Fender: well I live up here in Canada and we're a peaceful country, down south everyone has a gun.. but its okay I have a few american friends and they don't have guns..so I am happy :) |
| 21:59.03 | tuxinator_linux | The liberal demacrats are a "buncha sissies" |
| 21:59.12 | [TK]D-Fender | Then look at international standings in education & arts.... |
| 21:59.22 | footnote | tuxinator_linux: yeah and the liberal republicans suck too! |
| 21:59.25 | footnote | hehe |
| 21:59.31 | [TK]D-Fender | I'm in Montreal, the epitome of NOT USA. |
| 21:59.38 | tuxinator_linux | that's true, footnote |
| 22:00.07 | [TK]D-Fender | footnote: where you from? |
| 22:00.16 | footnote | I've lived all over |
| 22:00.20 | footnote | California |
| 22:00.21 | walnuck | [TK]D-Fender: coincidently mlug comes to mind |
| 22:00.24 | [TK]D-Fender | Currently? |
| 22:00.28 | tuxinator_linux | Where poeple are not allowed to carry guns, there is the highest crime rate, and where people can carry, it's lower |
| 22:00.28 | footnote | Georgia |
| 22:00.51 | walnuck | [TK]D-Fender: so when said TDM can do the job I hear that there are "series" with modules? what type of complexity is that? |
| 22:00.57 | footnote | I'm seceding only a 100ft circle around where ever I am :) |
| 22:00.58 | [TK]D-Fender | Succession is one of the few Federal Death Penalties as I recall.... |
| 22:01.02 | tuxinator_linux | bad guys carry no matter what the law is |
| 22:01.10 | footnote | [TK]D-Fender: bah :) |
| 22:01.13 | [TK]D-Fender | I've been to a MLUG meeting on VOIP about 10 months ago |
| 22:01.45 | footnote | [TK]D-Fender: hey, is the Fender in you nick a reference to your ax? |
| 22:01.52 | footnote | er, your nick |
| 22:02.07 | Goldenear | Is there a place (documents) where I could find the answer to my question ? |
| 22:02.11 | [TK]D-Fender | walnuck: you need a TDM01B at the minimum. That will give you 1 phonle line from your wall into your * box. |
| 22:02.16 | Beirdo | ~docs |
| 22:02.17 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 22:02.34 | [TK]D-Fender | footnote: nope, its from my playing style in FPS games. I play on a Dean personally :D |
| 22:02.42 | walnuck | [TK]D-Fender: I guess if it worked for you, it should for me too... you better not be a mountie ;-) |
| 22:02.48 | footnote | Dean is good |
| 22:03.01 | tuxinator_linux | Time to shower, see ya |
| 22:03.07 | footnote | later tux |
| 22:03.13 | G0shen | When I listen to my .wav file produced with a call that has IAX on one end and ZAP on the other, the ZAP is super quiet...if I adjust the volume in zapata.conf will that help you think? |
| 22:03.15 | [TK]D-Fender | I liked it because its a 24-fret w/ floating trem and at 1/2 price :D |
| 22:03.29 | footnote | [TK]D-Fender: what kind of amp? |
| 22:03.41 | G0shen | my rxgain and txgain on zapata.conf is already 1.5 |
| 22:03.52 | [TK]D-Fender | none, I use a Boss VF-1 rack direct to computer |
| 22:03.56 | footnote | ah |
| 22:04.08 | footnote | you know, I've got a GT3 I like a lot |
| 22:04.24 | footnote | Mackie mixer, computer, etc. |
| 22:04.37 | footnote | But it's hard to beat a LOUD tube amp :) |
| 22:04.52 | [TK]D-Fender | I'm looking to buy a GT-6 this year. Very swweet |
| 22:04.57 | footnote | yes |
| 22:05.03 | footnote | You'll enjoy it |
| 22:05.16 | footnote | the GT3 i've had for six years |
| 22:05.16 | [TK]D-Fender | I've used on at the local music store...... very nice |
| 22:05.25 | footnote | It's indestructible |
| 22:05.33 | footnote | GT6 even better |
| 22:05.44 | [TK]D-Fender | thats what I like. Unlike Zoom, its not a piece of cheap plastic. |
| 22:05.48 | *** join/#asterisk hotoke (~hnic@208.179.86.5) |
| 22:05.48 | footnote | yeah |
| 22:05.55 | footnote | road worthy |
| 22:06.06 | [TK]D-Fender | indeed, and I've done shows on occasion |
| 22:06.26 | footnote | I get a really good tone considering my apartment setup |
| 22:06.39 | footnote | you wouldn't believe the power amp i'm using |
| 22:06.51 | footnote | a friggin Peavey Rage 157 |
| 22:06.54 | footnote | er, 158 |
| 22:07.02 | footnote | just going through the CD jack |
| 22:07.11 | [TK]D-Fender | I'm actually just goind through a Cambrige SoundWorks 4.1 (turned 2.1) setup direct through PC (Audigy) and it sounds GREAT |
| 22:07.15 | jaxxan | on the cisco 7960, the displayed logo via html is a .bmp ..... can you use a .gif or .jpg? |
| 22:07.26 | footnote | jaxxan: don't think so |
| 22:07.31 | [TK]D-Fender | I have a Bandit 112 in storage I haven't used in years |
| 22:07.53 | andrewct | the 7690 needs what it wants.....bmp |
| 22:08.11 | G0shen | Who has the highest rxgain, txgain settings in zapata.conf? I am reading on the mailing list someone had 15! |
| 22:08.13 | footnote | I've got two yamaha speakers on poles I wired into the little practice amp and it will actually get the cops called on me. |
| 22:08.22 | footnote | 15 watts |
| 22:08.44 | BrianR___ | The Cisco 7940 needs a windows BMP. It needs to be 8 bit grayscale and uncompressed. Only colors black and white will be shown on the LCD. |
| 22:08.47 | [TK]D-Fender | My Bandit's POT is messed and everything past 2 is measured in RICHTER |
| 22:08.47 | G0shen | 15 db |
| 22:09.15 | BrianR___ | (In other words, the bmp is needlessly large) |
| 22:09.39 | *** join/#asterisk walnuck (~jack@modemcable106.82-200-24.mc.videotron.ca) |
| 22:09.47 | [TK]D-Fender | walnuck: SFound the TDM01B for sale yet? |
| 22:10.10 | [TK]D-Fender | Oh, see that you're local too.... |
| 22:10.22 | *** join/#asterisk kuj (~kuj@c-67-165-241-16.client.comcast.net) |
| 22:10.49 | kuj | howdy |
| 22:11.14 | walnuck | [TK]D-Fender: well on cyberspace, I'm actually a porn star.. they call me the shemale :) |
| 22:11.21 | [TK]D-Fender | TMI |
| 22:11.28 | walnuck | [TK]D-Fender: I have both genders >:) |
| 22:12.42 | *** part/#asterisk km- (~pgrace@brdgw1.rttx.com) |
| 22:13.06 | Goldenear | I'm sorry but I can't find any answer to my question in the docs :( |
| 22:13.24 | [TK]D-Fender | Goldenear: what was it? |
| 22:13.29 | Goldenear | how does asterisk know when it has to directly connect two IAX client or not ? |
| 22:13.52 | Goldenear | (with notransfer=no in IAX.conf) |
| 22:13.55 | [TK]D-Fender | It never HAS to, but can choose whether or not to. |
| 22:14.02 | [TK]D-Fender | thats the "canreinvite" option |
| 22:14.21 | Goldenear | I know that |
| 22:14.55 | Goldenear | it's "canreinvite" in sip.conf and "notransfer=no" in iax.conf |
| 22:15.04 | JerJer[mobile] | Goldenear: nostransfer=no is the default |
| 22:15.22 | bjohnson | JerJer[mobile]: hey |
| 22:15.35 | bjohnson | JerJer[mobile]: do you use vm on your wrt *? |
| 22:15.42 | JerJer[mobile] | no |
| 22:16.24 | Goldenear | so with notransfer=no (the default) asterisk will always make two clients directly speak together ? |
| 22:16.29 | bjohnson | did you use the Brian Capouch ipk? |
| 22:16.33 | JerJer[mobile] | simple dumb, sip to iax converter |
| 22:16.46 | JerJer[mobile] | no i have my own firmware build |
| 22:16.53 | Goldenear | (two IAX clients I mean) |
| 22:16.59 | JerJer[mobile] | Goldenear: you don't have to even put anything |
| 22:16.59 | bjohnson | crap |
| 22:17.11 | JerJer[mobile] | that ipkg that is out there is rather lame |
| 22:17.21 | JerJer[mobile] | they build everything known to man and put it on the flash |
| 22:17.28 | *** join/#asterisk kodomo (~memyself@emu.net.informatik.tu-muenchen.de) |
| 22:17.34 | kodomo | hi folks :) |
| 22:17.34 | JerJer[mobile] | when 99.999999999% of those .so's will never get touched |
| 22:17.46 | bjohnson | did you build it on a pc ? |
| 22:17.49 | JerJer[mobile] | yes |
| 22:18.13 | bjohnson | I guess I should look for a howto on creating a build environment for openwrt |
| 22:18.19 | Goldenear | JerJer[mobile]: so, by default a call between two IAX clients doesn't go thrue the asterisk after the call is initiated, right ? |
| 22:18.21 | kodomo | Does anybody know if it's possible to make kphone or iaxcomm use arts or esound? |
| 22:18.28 | sivana | Goldenear: correct |
| 22:18.33 | sivana | I learnt the hard way.. hehe |
| 22:18.46 | JerJer[mobile] | Goldenear: if possible |
| 22:18.52 | JerJer[mobile] | and operating the same codec |
| 22:19.11 | Goldenear | ok |
| 22:19.56 | cbachman | I'm currently messing with building asterisk for openwrt, trying to trim it down |
| 22:20.05 | Goldenear | so if two IAX client can use the same codec the call will go directly between them without going thru the server (and so saving some bandwidth on the server) |
| 22:20.06 | cbachman | Definitely don't need ALL the .so's to just do sip->iax |
| 22:20.14 | JerJer[mobile] | is there an echo in here? |
| 22:20.18 | sivana | heh |
| 22:20.34 | sivana | JerJer[mobile]: do you have echo cans on your network? |
| 22:20.41 | Goldenear | JerJer[mobile]: and what about NAT ? |
| 22:20.49 | JerJer[mobile] | sivana: does a cat have an ass? |
| 22:20.57 | sivana | JerJer[mobile]: my customers say that they sometimes get echo on calls routed to you |
| 22:21.05 | sivana | and I'm not sure what to make of it |
| 22:21.12 | JerJer[mobile] | Goldenear: if the two endpoints cannot talk diirectly to each other it will work |
| 22:21.23 | JerJer[mobile] | sivana: that is not our problem |
| 22:21.40 | sivana | JerJer[mobile]: anyway for me to can it? |
| 22:22.16 | Cherebrum | We have been having problems calling certain 800#s thru nufone |
| 22:22.35 | Cherebrum | the 800#s work fine on other phones but thru nufone I get congestion |
| 22:22.46 | Goldenear | JerJer[mobile]: Does asterisk knows that the clients are NATed ? |
| 22:23.09 | scrubb | Ok, I have a privacy application that I'd like to pursue that involves "voice modulation". What is the best place to look in the * code to start creating a waveform filter module into the bridged call? |
| 22:23.15 | gambolputty | How about a step by step tutorial on how to properly cross compile * for the WRT54G? |
| 22:23.17 | G0shen | I route my toll free calls through FWD IAX, Haven't had a problem yet |
| 22:23.20 | gambolputty | that would really help |
| 22:23.31 | JerJer[mobile] | Cherebrum: you have to sent a valid US48 Caller*id to call toll-free numbers |
| 22:23.37 | JerJer[mobile] | gambolputty: um yeah ok |
| 22:24.08 | Cherebrum | JerJer[mobile]: I am sending my nufone DID as my caller id |
| 22:24.13 | walnuck | Cherebrum: you can use nufone products without use of the service? |
| 22:24.16 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 22:24.39 | Cherebrum | walnuck: ? |
| 22:24.52 | JerJer[mobile] | yeah ??? |
| 22:25.06 | scrubb | anyone? |
| 22:25.15 | Cherebrum | so should I not use my nufone DID for my callerid? |
| 22:25.15 | walnuck | Cherebrum: nvrmind, I'm still quite new to this, I'm looking for a product for fx |
| 22:25.21 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 22:25.47 | Cherebrum | fx? |
| 22:25.52 | Cherebrum | fxo or fxs? |
| 22:26.16 | JerJer[mobile] | Cherebrum: if it is a Michigan DID, yes |
| 22:26.17 | Cherebrum | JerJer[mobile]: I am setting my nufone toll free DID as my CALLERIDNUM |
| 22:26.19 | walnuck | G0shen: have you tested softphones from nonlinux machines? |
| 22:26.21 | JerJer[mobile] | if it is a toll-free did, no |
| 22:26.24 | Cherebrum | aha |
| 22:26.27 | Cherebrum | then that is the problem |
| 22:26.39 | JerJer[mobile] | you cannot set a toll-free number as a caller*id and call other toll-free numbers |
| 22:26.42 | Cherebrum | do I have to set it as a DID that nufone owns? |
| 22:26.43 | walnuck | Cherebrum: I guess I'd like to get both, what's a good product ? |
| 22:26.49 | JerJer[mobile] | our switch and upstream providers wll not allow it |
| 22:26.57 | Cherebrum | walnuck: sipura 3000 |
| 22:27.11 | Cherebrum | JerJer[mobile]: ok. thanks |
| 22:27.17 | Cherebrum | I'll make some changes then |
| 22:27.19 | Goldenear | JerJer[mobile]: I'm confused... did you said that two NATed clients (behind two diferent NAT) can be directly be interconnected after the call has been initiated ? |
| 22:27.20 | *** join/#asterisk habakuk (~chatzilla@24-117-8-113.cpe.cableone.net) |
| 22:27.36 | JerJer[mobile] | 248-724-0700 is a good number to use if you don't have any other number to use |
| 22:27.43 | G0shen | walnuck : I played around with four different soft phones on win98 |
| 22:28.07 | JerJer[mobile] | Goldenear: i didn't say |
| 22:28.16 | walnuck | G0shen: mic+headset counts as one of your soft phones? |
| 22:28.35 | JerJer[mobile] | or 248-724-9999 if you are really in a good mood :) |
| 22:28.36 | habakuk | hi is anyone using ODBC_STORAGE? |
| 22:28.41 | JerJer[mobile] | no |
| 22:29.14 | Goldenear | JerJer[mobile]: you daid "if the two endpoints cannot talk diirectly to each other it will work" :/ |
| 22:29.29 | JerJer[mobile] | exactly |
| 22:29.41 | JerJer[mobile] | asterisk will determine if the two endpoints can rap directly |
| 22:29.47 | walnuck | Cherebrum: how much did you pay for it? |
| 22:29.49 | Goldenear | ok |
| 22:29.54 | G0shen | I just used the laptop built in mic and headphones, just to dink around with before I dove in and bought an IAXy and an Sipura2100 |
| 22:30.00 | jsolares | JerJer[mobile]: for iax2 or for any kind of channel? |
| 22:30.10 | JerJer[mobile] | if so, a native bridge will take place and asterisk will optmize the iax call path |
| 22:30.32 | Goldenear | so if they can, asterisk we directly connect them and if they can't, the call will go thru the server, right ? |
| 22:30.40 | JerJer[mobile] | unless otherwise configured |
| 22:30.46 | JerJer[mobile] | correct |
| 22:30.53 | Goldenear | ok |
| 22:30.55 | JerJer[mobile] | test |
| 22:30.57 | JerJer[mobile] | test |
| 22:31.04 | JerJer[mobile] | then test again |
| 22:31.16 | sivana | once it works.. test again! |
| 22:31.17 | Goldenear | so asterisk has a way to know if the clients are nated or not. |
| 22:31.25 | jsolares | Goldenear: are your client phones iax or sip? |
| 22:31.35 | Goldenear | both |
| 22:31.42 | walnuck | Goshen: what's a good softphone for a M$win station - I have 1 xp station |
| 22:31.43 | Goldenear | and I'm currently testing |
| 22:31.49 | Goldenear | :) |
| 22:32.00 | jsolares | if they are sip, you have to use nat=yes and maybe qualify=yes and maybe canreinvite=no to have two sips behind nat connect |
| 22:32.18 | JerJer[mobile] | qualify ?! |
| 22:32.20 | G0shen | I can't remember which one worked, I think it was the IAXtelefon, but I tried Firefly too |
| 22:32.21 | G0shen | ~firefly |
| 22:32.22 | jbot | [firefly] http://virbiage.com/firefly/download/firefly-thirdparty.exe |
| 22:32.29 | Goldenear | I want to switch to IAX only :) |
| 22:32.34 | jsolares | you should |
| 22:32.35 | G0shen | good idea :) |
| 22:32.45 | JerJer[mobile] | qualify has nothing to do with NAT |
| 22:33.01 | walnuck | ~scream |
| 22:33.02 | jbot | aaaaaaaaaaaaaaahhhhhhhhhhhh!!! |
| 22:33.08 | G0shen | I really want one of those 4 port IAX devices when they "get back from chinese spring festival" whenever that is |
| 22:33.12 | walnuck | Well there is a bot.. hmm |
| 22:33.14 | walnuck | ~iax |
| 22:33.15 | jbot | rumour has it, iax is 4569 and 5036, or pronounces "Eeks" |
| 22:33.17 | G0shen | but my IAXy works good |
| 22:33.30 | walnuck | ~jbot |
| 22:33.31 | jbot | well, jbot is the shipboard computer, but you may call me eddie if it helps you relax |
| 22:33.34 | G0shen | its my geek badge....I have been showing it around |
| 22:33.50 | Pinhole | ~eddie |
| 22:33.51 | jbot | Robust, clustering, load balancing, high availability, web server tool.. URL: http://www.eddieware.org/ |
| 22:33.51 | JerJer[mobile] | G0shen: more like 2FXO / 6FXS or 4FXO / 4FXS |
| 22:34.02 | walnuck | !moderate |
| 22:34.34 | jsolares | JerJer[mobile]: it helps keep the udp session from my grandstream open, so it can always receive a call, and not just make calls |
| 22:35.13 | jsolares | so even if it has nothing to do with nat, it is extremely helpfull in my case atleast |
| 22:35.36 | Cherebrum | JerJer[mobile]: ok it works now. Thanks alot! |
| 22:35.48 | G0shen | rxgain txgain of 4 seems to be a nice number :) nice and loud |
| 22:36.40 | machinehd | anyone know the default unlock password for cisco 7960? I've tried "cisco" and "**#" |
| 22:36.49 | Cherebrum | **#* |
| 22:36.54 | Cherebrum | or **##** |
| 22:36.56 | Cherebrum | something like that |
| 22:37.04 | Cherebrum | just keep hitting **##**##**## |
| 22:37.08 | Cherebrum | eventually it will unlock |
| 22:37.09 | loud | **# only |
| 22:37.15 | Cherebrum | I can never remember it |
| 22:37.15 | Cherebrum | :p |
| 22:37.26 | machinehd | invalid :/ |
| 22:37.29 | loud | tftp is easier |
| 22:37.34 | Cherebrum | and then 1234 I thinks |
| 22:37.48 | walnuck | machinehd: check the doc, there's also a way to reset it- (cisco site's docs) |
| 22:38.24 | *** join/#asterisk MikeJ[Jayden] (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
| 22:38.40 | machinehd | well, it was **# with the old firmware. Now that I just upgraded to 7.1 it doesn't work. |
| 22:38.46 | *** join/#asterisk fafnir (~hello@tdds-gw.Moscow.gldn.net) |
| 22:38.46 | habakuk | use a tftp server |
| 22:38.51 | walnuck | machinehd: rtfm |
| 22:38.54 | MikeJ[Jayden] | I love pri hotcuts, especially by the dozen, all in an hour. |
| 22:39.29 | walnuck | machinehd: password recovery should be done under the table, how stupid can you be? |
| 22:39.31 | G0shen | does anyone know how to properly format this? this is from the example |
| 22:39.40 | G0shen | callerid=.My IAXy. <(801) 255-3925> |
| 22:39.51 | G0shen | are the . part of the displayed name? |
| 22:40.04 | machinehd | walnuck, blow me |
| 22:40.07 | G0shen | or are they required, and you put your name between? |
| 22:40.31 | Goldenear | do you think IAX will replace SIP or will the both have different applications ? |
| 22:40.46 | G0shen | I think it will |
| 22:40.57 | footnote | de facto |
| 22:41.04 | Goldenear | more and more endpoints support IAX |
| 22:41.41 | Goldenear | and some harphones now have a IAX firmware update |
| 22:41.53 | Goldenear | that's a good sign :) |
| 22:42.22 | walnuck | G0shen: IAXtelefon, is that somewhere in the public domain? |
| 22:42.37 | footnote | I wanna set up a mesh network in polynesia and live on a yacht |
| 22:42.39 | footnote | ahhhh |
| 22:42.44 | G0shen | http://www.geocities.com/babarnazmi/ |
| 22:43.13 | walnuck | JerJer[mobile]: 2FXO / 6FXS or 4FXO / 4FXS - are those standards? |
| 22:44.02 | walnuck | Goshen: thanks |
| 22:44.28 | G0shen | welcome |
| 22:45.54 | sivana | quit |
| 22:45.58 | sivana | oops |
| 22:46.15 | AgiNamu | OK, guys, PA168 is the phone that does IAX |
| 22:46.18 | AgiNamu | no big deal here :) |
| 22:46.22 | AgiNamu | Just get a 1.40+ firmware |
| 22:46.28 | AgiNamu | works great |
| 22:46.35 | AgiNamu | and tons of people make phones based on the PA168 |
| 22:46.36 | Goldenear | can an asterisk server manage as many IAX (ONLY) clients than a SER server can manage SIP clients ? |
| 22:48.59 | Goldenear | (I'll need a VoIP server for a very large free public mesh network) |
| 22:49.23 | AgiNamu | "Very large"? |
| 22:49.40 | Nukemizer | Can I get the * to provide fake Dialtone to a connected PBX ? when using a T1 card ? |
| 22:50.05 | Goldenear | (That's why I initialy choose SER because I read that it needs less bandwidth than * |
| 22:50.07 | Goldenear | ) |
| 22:50.20 | footnote | less flexible too |
| 22:50.32 | modulus_ | anyone know the default Admin password for linksys pap2 web interface? |
| 22:50.42 | AgiNamu | SER and Asterisk are fundamentally different things. |
| 22:50.45 | Goldenear | AgiNamu: like 10 000 possible registered enpoints |
| 22:51.18 | *** join/#asterisk ckruetze (ckruetze@cpc1-cmbg7-5-0-cust31.cmbg.cable.ntl.com) |
| 22:51.19 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 22:52.16 | AgiNamu | Asterisk gets its hands dirty with the media |
| 22:52.23 | AgiNamu | SER just handles registrations |
| 22:52.47 | Goldenear | AgiNamu: I know that. What I want is that all the users of the meshnetwork to be able to phone each other without using bandwidth on the server |
| 22:52.55 | modulus_ | anyone? |
| 22:53.05 | AgiNamu | goldenear, well, that requires public IPs |
| 22:53.06 | modulus_ | friggin' i can't reset the linksys pap2 w/o the admin password |
| 22:53.11 | modulus_ | default no configs yet |
| 22:53.34 | AgiNamu | i.e., nat will screw you up |
| 22:54.07 | walnuck | Goshen: mind me asking, what card are you using? |
| 22:54.10 | Goldenear | AgiNamu: the server will be in the private area... |
| 22:54.30 | AgiNamu | "the private area"? damn, sounds like my manual "dont use phone in the exploder place" |
| 22:54.39 | AgiNamu | what is that supposed to mean? |
| 22:54.41 | G0shen | generic FXO card from digitnetworks.com |
| 22:54.43 | Goldenear | lol |
| 22:54.46 | footnote | Goldenear: like a community system |
| 22:54.52 | Goldenear | the private network I mean :) |
| 22:54.57 | footnote | with a gateway out, etc |
| 22:55.01 | Goldenear | footnote: exactly |
| 22:55.03 | AgiNamu | *a* private network? |
| 22:55.07 | jsolares | and will all of your clients be in the same network? |
| 22:55.08 | footnote | that can work |
| 22:55.11 | PyroSteve | hey guys |
| 22:55.13 | AgiNamu | oh, well , then you should be OK with Asterisk or SER |
| 22:55.22 | AgiNamu | Asterisk supports IAX and others, and SER is just SIP AFAIK |
| 22:55.29 | PyroSteve | this is a non - asterisk related questions |
| 22:55.36 | Goldenear | ok |
| 22:55.44 | AgiNamu | pyrosteve, no, that was a statement |
| 22:55.53 | footnote | PyroSteve: how are we supposed to know if you'll ever get laid? |
| 22:56.08 | footnote | hehe |
| 22:56.15 | AgiNamu | Pyro, don't be nervous. Just try to enjoy yourself. |
| 22:56.18 | PyroSteve | is a sipura 3000 capable to connecting to key systems over an internet connection ? |
| 22:56.29 | PyroSteve | or simply over IP |
| 22:56.46 | AgiNamu | I think the sipura pretty much relies on IP |
| 22:56.53 | Goldenear | so if all the clients are in the same network Asterisk can only handle registrations as SER does ? |
| 22:56.58 | AgiNamu | I dont think it has PPP or anything |
| 22:57.28 | jsolares | Goldenear: if all the clients are on the same network, you cand do anything you want |
| 22:57.58 | AgiNamu | Goldenear, it'll try to proxy the media |
| 22:58.06 | AgiNamu | BUT you can have them re-invite (or IAX transfer) |
| 22:58.20 | AgiNamu | and Asterisk will just run control (for SIP) or completely step out of it (for IAX) |
| 22:58.41 | AgiNamu | which is the one annoying thing of IAX's design |
| 22:58.42 | Goldenear | AgiNamu: proxy = a lot of bandwith needed for the server ... and I don't want this |
| 22:58.57 | *** part/#asterisk mogorman (~mogorman@dhcp-162.digium.com) |
| 22:59.03 | AgiNamu | yes, i understand. |
| 22:59.11 | AgiNamu | Pyro, doesnt the Sipura only do SIP |
| 22:59.35 | AgiNamu | Goldenear, so thats why you need re-invites or transfers |
| 22:59.40 | walnuck | G0shen: I'm planning to buy a card, I think an fxs card would be better wouldn't you think? |
| 22:59.42 | AgiNamu | so it connects the devices directly to each other |
| 22:59.53 | Goldenear | AgiNamu: exactly :) |
| 22:59.57 | AgiNamu | wlnuck, what do you want to do |
| 23:00.02 | G0shen | walnuck: it depends what you need it for |
| 23:00.04 | AgiNamu | so just turn on reinvite/transfer |
| 23:00.08 | G0shen | that a silly question ;) |
| 23:00.09 | PyroSteve | AgiNamu: Well I want to interconnect two phone systems over IP |
| 23:00.18 | G0shen | no, a FXO card would be better, if you are going all soft phones |
| 23:00.19 | AgiNamu | an FXS is gonna suck if you want to connect a POTS line :) |
| 23:00.38 | G0shen | Need FXO card to connect to the telco lines |
| 23:00.41 | walnuck | G0shen: well if I go with an fx0 card I'm restricted to just a softphone rather than all the ptsn phones in the residence, is this correct? |
| 23:00.53 | AgiNamu | Hmm.... I *THINK* you can have one Sipura dial another Sipura |
| 23:01.02 | G0shen | no, because you could get a sipura, or IAXy to connect the house phones |
| 23:01.06 | AgiNamu | if your key system supports sIP, then it should work too |
| 23:01.22 | G0shen | you don't need a card to connect the house phones, but you could |
| 23:01.29 | AgiNamu | walnuck, you can also connect a VOIP phone :) |
| 23:01.31 | G0shen | you could get a develouper card for $200 that gives you one of each |
| 23:01.40 | AgiNamu | get the TDM card |
| 23:01.51 | *** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au) |
| 23:01.52 | AgiNamu | and you can mix up to 4 FXS/FXO daughtercards |
| 23:02.10 | *** join/#asterisk Shido6 (~shido6@d57-87-253.home.cgocable.net) |
| 23:02.20 | AgiNamu | but its probably cheaper to buy voip phones instead of FXS ports |
| 23:02.27 | Goldenear | AgiNamu: and if a call need to go out the "private" network be routed to the internet, I want it to be NATed and directly connected to the external endpoint/asterisk pbx |
| 23:02.55 | AgiNamu | if you're using sip, you want to stay the f*k away from NAT |
| 23:03.03 | Shido6 | IAXys |
| 23:03.06 | Shido6 | get an IAXy |
| 23:03.12 | AgiNamu | PA168s :) |
| 23:03.15 | AgiNamu | get a PA168 |
| 23:03.17 | FaithX | hey what's the CVS passwd? |
| 23:03.22 | AgiNamu | anoncvs |
| 23:03.27 | Shido6 | brb |
| 23:03.29 | *** part/#asterisk Shido6 (~shido6@d57-87-253.home.cgocable.net) |
| 23:03.33 | PyroSteve | i have sipura 2100 and 2000 behind nat work an asterisk server on a public ip working fine |
| 23:03.45 | AgiNamu | oh, you can get it to work |
| 23:03.51 | Goldenear | AgiNamu: that's why I don't want to use sip but only IAX |
| 23:03.58 | AgiNamu | it's just that it can be a bitch, or not work at all sometimes |
| 23:04.01 | AgiNamu | depending on type of NAT |
| 23:04.02 | Mavvie | yay! got the patton 2977 for US$ 710.- on ebay. |
| 23:04.03 | walnuck | so basically internal fxs and fxo cards for the pc aren't hot..and I should go with a mother hub.. |
| 23:04.07 | Mavvie | dual PRI :-) |
| 23:04.08 | Goldenear | any other endpoint than iax won't be supported |
| 23:04.23 | AgiNamu | wlanuck, completely depends on your needs |
| 23:04.36 | AgiNamu | FXO cards are great if you have a pots line to connect to |
| 23:05.37 | AgiNamu | FXS... well, with a channelbank, it makes more sense |
| 23:05.38 | *** join/#asterisk Shido (~greg@d57-87-253.home.cgocable.net) |
| 23:06.09 | AgiNamu | Goldenear, well, if you dont want sIP, then Ser is outta the question :) |
| 23:06.20 | AgiNamu | So get your asterisk box installed , and then worry about forwarding calls out |
| 23:06.49 | walnuck | AgiNamu: what are you referring to as an asterisk box? |
| 23:06.50 | AgiNamu | depending on your accounting needs, you might need to disable transfer when making a call "out" |
| 23:07.00 | G0shen | anyone have an example of how to ring two extensions at the same time on incoming call? just need the dial string |
| 23:07.05 | AgiNamu | walnuck: A computer running asterisk |
| 23:07.21 | AgiNamu | isnt it DIAL(iax2/123|iax2/456)?> |
| 23:07.26 | G0shen | isn't it (IAX2/goshen&SIP/21) |
| 23:07.30 | G0shen | ahh right the | |
| 23:07.30 | AgiNamu | oh, & |
| 23:07.31 | AgiNamu | :) |
| 23:07.33 | AgiNamu | lol |
| 23:07.41 | G0shen | is it & or | ? |
| 23:07.44 | Shido | & |
| 23:07.46 | AgiNamu | dunno. use the wiki |
| 23:07.49 | Goldenear | AgiNamu: I said I was initialy thinking about using SER ... but now I know asterisk can manage as many users, I'll go for asterisk :) |
| 23:07.55 | Shido | | is a replacement you can use for , |
| 23:07.59 | walnuck | AgiNamu: that part is probably the easiest.. right now I'm a little confused as to what my hardware should be.. |
| 23:08.01 | Shido | at the end of the statement |
| 23:08.11 | AgiNamu | walnuck, tell us your scenario |
| 23:08.15 | AgiNamu | thanks shido |
| 23:08.24 | Shido | for seperating the amount of milliseconds u want to ring a phone or try an extension |
| 23:08.24 | *** join/#asterisk nextime (~nextime@ns0.nexlab.net) |
| 23:08.35 | Goldenear | AgiNamu: Why would I have to worry about forwarding calls out ? |
| 23:08.46 | walnuck | AgiNamu: basic-> ip gateway to all in-house pot phones |
| 23:09.01 | AgiNamu | Goldenear, you asked about making public calls. |
| 23:09.22 | AgiNamu | walnuck, you have incoming pots lines? |
| 23:09.35 | walnuck | AgiNamu: I'd like the pot phones to ring as well which is why I keep mentioning the term fxs |
| 23:09.37 | Shido | exten => 100,1,Dial(IAX2/user@peer|20|r) for example rings 100 for 20k milliseconds |
| 23:09.51 | Goldenear | AgiNamu: public = to the internet not to the POTS :) |
| 23:09.57 | Shido | how many pots? |
| 23:10.01 | AgiNamu | golden, same thing |
| 23:10.15 | AgiNamu | going out to the internet, it might not be able to do the transfer |
| 23:10.26 | AgiNamu | or more importantly, you might not WANT it to do the transfer |
| 23:10.28 | AgiNamu | so you keep the CDRs |
| 23:10.29 | Goldenear | AgiNamu: why shoult it be a problem with asterisk ? |
| 23:10.29 | walnuck | Shido: I'm assuming a pot is just the resident phone service, i have no pbx service |
| 23:10.51 | AgiNamu | because if your phone does the native transfer, Asterisk wont record the CDR |
| 23:11.08 | AgiNamu | walnuck, ok, so you have analog lines coming in |
| 23:11.10 | AgiNamu | how many lines? |
| 23:11.36 | walnuck | AgiNamu: 1 analog line at the time being.. |
| 23:11.51 | Shido | walnuck , yep Plain old telephone |
| 23:11.52 | AgiNamu | OK, and what existing wiring do you have? |
| 23:12.00 | walnuck | AgiNamu: none |
| 23:12.11 | AgiNamu | Because, if you go with FXS cards, then you need to run telefone wire out to your phones |
| 23:12.12 | walnuck | AgiNamu: well the basic copper pair of course.. |
| 23:12.27 | AgiNamu | if you buy a VoIP phone/ATA, then you can use ethernet |
| 23:12.34 | Shido | on your LAN |
| 23:12.49 | Shido | or at some other place with a net connection to your Asterisk box |
| 23:12.50 | AgiNamu | Oh yea, good point. you'd have to have a netowrk :) |
| 23:12.55 | Shido | hehe |
| 23:13.22 | AgiNamu | im getting a Cisco Catalyst 2500 for my house soon :D |
| 23:13.34 | loud | you mean, 2900. |
| 23:13.34 | Shido | catalyst... |
| 23:13.39 | Shido | drop that on someones head |
| 23:13.41 | Shido | Fatality |
| 23:13.44 | AgiNamu | lol |
| 23:13.51 | walnuck | AgiNamu: well, I do have a Lan I should of mentioned it, that's where I have my private 192.168.x.x network, 3 computers are on it, I have currently fc3 running with a static ip which I plan to put asterisk on it |
| 23:13.54 | Shido | I like das blinken lights but damn |
| 23:13.55 | AgiNamu | loud yea, i dont know the model. maybe they told me wrong, or i forgot |
| 23:14.01 | AgiNamu | price was right :) |
| 23:14.25 | AgiNamu | walnuck, well, you have to decide. First: do you want to connect existing analog handsets? |
| 23:14.38 | Goldenear | I would like an IAX wifi phone :) |
| 23:14.54 | AgiNamu | Goldenear: get a standard wireless + a PA168 ATA :) |
| 23:15.05 | walnuck | AgiNamu: well I assume pot handsets or plain old jackass term "Old Analogue Crap" suits it.. |
| 23:15.17 | AgiNamu | Walnuck, if so, you need FXS. you use FXS cards, means you have to run cable from the card to the phone. If you get an FXS ATA , then you can net to the ATA, and then analog telefone from there. |
| 23:15.25 | walnuck | AgiNamu: unfortunately for the time being, yes I have to make use of the analogues.. |
| 23:15.36 | Goldenear | AgiNamu: I want to be able to connect from any hotspot :) |
| 23:15.54 | AgiNamu | Goldenear, ok, so bring the ATA with you and plug it into your laptop |
| 23:15.57 | modulus_ | anyone have any experience with linksys pap2? |
| 23:15.59 | AgiNamu | OK, so your decision is mainly either to buy the FXS modules for the TDM card |
| 23:16.06 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
| 23:16.08 | AgiNamu | walnuck, or , buy ATAs |
| 23:16.13 | modulus_ | i have this really annoying linksys pap2 that i can't configure the sip on |
| 23:16.44 | AgiNamu | FXS module for the TDM card is $75 |
| 23:16.52 | AgiNamu | so it's near the price you'd pay for an ATA |
| 23:16.59 | AgiNamu | so ... mainly convenience. |
| 23:17.07 | AgiNamu | whatever works easiest for you |
| 23:17.08 | walnuck | AgiNamu: how about I place the fxs connection where the main entry copper comes in, shouldn't that avoid getting an fxs connection hookup for each phone? |
| 23:17.20 | AgiNamu | yes |
| 23:17.28 | walnuck | AgiNamu: yes what? |
| 23:17.30 | Goldenear | AgiNamu: http://www.zyxel.com/product/P2000W.php |
| 23:17.37 | AgiNamu | Yes, that'll work. but that means all the phones will be "one |
| 23:17.41 | AgiNamu | one extension |
| 23:17.42 | Goldenear | I want that with IAX2 ! :) |
| 23:17.51 | AgiNamu | I've heard the zyxel sucks |
| 23:17.53 | walnuck | AgiNamu: extension to what? |
| 23:17.57 | AgiNamu | to asterisk |
| 23:18.05 | walnuck | AgiNamu: ok |
| 23:18.10 | AgiNamu | like, Phone1 can't call Phone2 in your house |
| 23:18.14 | walnuck | AgiNamu: thanks for telling me :) |
| 23:18.19 | AgiNamu | nor can they make separate calls :). |
| 23:18.48 | AgiNamu | weak encryption |
| 23:18.51 | AgiNamu | no WPA AFAIK |
| 23:18.58 | AgiNamu | bad battery too, I think. |
| 23:18.59 | BoRiS | Anyone have a newer firmware for the Senao SI-7800 wifi phone then 0.03.0008? |
| 23:19.16 | AgiNamu | i dont get why more manufacturers dont open up their firmware |
| 23:19.17 | walnuck | AgiNamu: but if I have 1 copper pair residence from the bell corp, what's the deal if most calls have to rely on this? |
| 23:19.28 | AgiNamu | We were gonna buy 1000 devices from someone, and we were even gonna add IAX2 ourselves |
| 23:19.40 | AgiNamu | walnuck, cause you have voip, right? |
| 23:19.57 | AgiNamu | so you can place calls with the single line from bell, or place them with nufone or someone |
| 23:19.58 | walnuck | AgiNamu: I don't have any services |
| 23:20.09 | AgiNamu | well, you can order them in 2 minutes :) |
| 23:20.21 | AgiNamu | www.nufone.net, for example |
| 23:20.40 | PBXtech | what am i doing wrong here: exten => h,2,MeetMeCount(1111|var5) |
| 23:20.40 | PBXtech | exten => h,3,GotoIf($[${var5} > 0]?4:5) |
| 23:21.02 | JerJer[mobile] | using the h exten |
| 23:21.21 | *** join/#asterisk owh (~onno@OptusSatelliteServices.22bjc76f09.optus.net.au) |
| 23:21.34 | PBXtech | huh it works fine, the gotoif isnt working |
| 23:21.50 | PBXtech | or the variable isnt passing |
| 23:21.53 | AgiNamu | well im out of here |
| 23:21.54 | *** join/#asterisk malcolmd (~malcolmd@malcolmd.digium.sponsor.pdpc) |
| 23:21.55 | AgiNamu | cya round |
| 23:22.03 | Goldenear | thanks you for all your answers :) |
| 23:22.08 | AgiNamu | good luck, |
| 23:22.12 | AgiNamu | . |
| 23:22.22 | modulus_ | stupid pap2 |
| 23:22.32 | walnuck | AgiNamu: thanks |
| 23:22.49 | owh | How do I play the same message at the start of a call to both the caller and the callee? |
| 23:23.36 | walnuck | AgiNamu: i guess I'll go with an fxs card, but does that mean that my * box can make calls with a mic? |
| 23:24.43 | Shido | fxs card means u have a regular phone |
| 23:24.56 | Shido | connected to the fxs card in your asteriskbox |
| 23:25.22 | walnuck | Shido: I understand that part, but what about outgoing calls |
| 23:25.29 | walnuck | Shido: from that box |
| 23:25.33 | Shido | outgoing calls, if u have 1 pots then u get an fxo module |
| 23:25.41 | Shido | and plug into that card with the fxs module |
| 23:25.48 | Shido | u can buy a modular card with 4 slots on it |
| 23:25.55 | Shido | u can mix and match fxo and fxs module |
| 23:25.55 | Shido | s |
| 23:26.03 | Shido | fxo are for CO lines or your analog lines from the telco |
| 23:26.10 | Shido | FXS are for your station devices or analog phones |
| 23:26.16 | walnuck | Shido: so the fxs card plugs into the computer or into something like tdm? |
| 23:26.27 | Shido | plugs into your puter |
| 23:26.36 | Shido | running Asterisk |
| 23:27.33 | *** join/#asterisk clark_ (~bclark@64.171.107.5) |
| 23:27.59 | owh | Perhaps some background to my question will assist. I can play a sound with the dial command, only the callee hears it. I can play a sound before I dial, but it won't dial until the sound is finished and the caller hears ringing while the callee hears their sound. |
| 23:28.44 | modulus_ | stupid pap2 |
| 23:28.46 | modulus_ | i hate it |
| 23:28.46 | *** join/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
| 23:29.13 | walnuck | Shido: my question is: Is that fxs card capable of allowing me to make outgoing calls on the analog network that my bell corp provides- so that I can call other local people on their traditional phones..? |
| 23:29.53 | owh | walnuck: NAFAIK, you use an FXO card for that. |
| 23:30.38 | walnuck | owh: i'm wondering wtf I should care to buy both if some other product can do both, is this what tdm does? |
| 23:31.02 | owh | walnuck: I am not familiar with tdm, so I do not know. |
| 23:31.25 | walnuck | owh: what can i use that can do this? |
| 23:32.32 | owh | walnuck: FYI: http://lists.digium.com/pipermail/asterisk-users/2004-October/067823.html |
| 23:32.39 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 23:34.53 | owh | walnuck: Did you read that? |
| 23:37.28 | walnuck | owh: well I did again, but that has to do with the fact that THAT product which is referred can never to both at the same time... I'd like to be able to do both, I guess I have to get some kind of hub, is there a term to this?(outside box for fxs and fxo) |
| 23:37.59 | owh | walnuck: Uhm why would you need both at the same time? What are you trying to do? |
| 23:38.38 | jsmith | If you want to connect both a telephone *line* and an analog *telephone*, you need a TDM11B card. It has one FXO port and one FXS port. |
| 23:39.06 | walnuck | jsmith: thanks |
| 23:39.38 | jsmith | walnuck: No problem. |
| 23:39.45 | jsmith | walnuck: Glad I could help out. |
| 23:39.46 | walnuck | jsmith: you're probably one of the smartest here.. ;-) .. good one heheheh.. |
| 23:39.57 | owh | I've been googling until I'm blue in the face. According to the docs I find, I should be able to play a sound in the background while doing something else, only the sound playing stops the dialing from happening, what am I missing? |
| 23:40.01 | jsmith | walnuck: More useful info at www.asteriskdocs.org/ |
| 23:40.37 | wildcard0 | owh, when you dial on top of Backrground(), it transfers the call to what you dial |
| 23:41.13 | owh | wildcard0: So, should I not hear the sound as the caller while that is happening? |
| 23:41.15 | jsmith | wildcard0: Yes, that's the point. |
| 23:41.44 | wildcard0 | jsmith, yes it is. so? |
| 23:41.50 | jsmith | I meant to send that to owh |
| 23:41.53 | jsmith | Typed the wrong nick |
| 23:41.54 | wildcard0 | oh |
| 23:41.55 | wildcard0 | heh |
| 23:42.14 | jsmith | owh: You have two choices -- play a sound and ignore input (the Playback() app) |
| 23:42.28 | wildcard0 | owh, as soon as you hit a key, it transfer to another extension. so yes...it stops playing |
| 23:42.37 | jsmith | owh: Or play a sound *UNTIL* the user hits a key (the Background() app) |
| 23:42.42 | owh | No, this is not what is happening... |
| 23:42.55 | jsmith | owh: Which do you want? |
| 23:43.03 | owh | Neither |
| 23:43.22 | owh | jsmith: I need to play the same sound to both parties at the start of the call. |
| 23:43.31 | wildcard0 | both? |
| 23:43.32 | *** join/#asterisk _mwoodj_ (~MWoodJ@hyper-eye.digium.sponsor.pdpc) |
| 23:43.40 | owh | wildcard0: The caller and the callee |
| 23:43.41 | jsmith | owh: Seems like there's an argument to the Dial() app to do just that... |
| 23:43.44 | wildcard0 | ok |
| 23:44.00 | owh | jsmith: The A() only seems to play to the callee, not the caller. |
| 23:44.00 | jsmith | owh: Seems like it's announcement argument. |
| 23:44.35 | owh | jsmith: The caller gets silence during the A() |
| 23:44.36 | wildcard0 | ya you could use 'm' |
| 23:44.39 | jsmith | 'A(x)' -- play an announcement to the called party, using x as file |
| 23:44.53 | wildcard0 | ya and 'A' |
| 23:45.08 | owh | wildcard0: That only plays while the call is being answered, but the call gets answered immediately, so the sound never plays. |
| 23:45.11 | jsmith | Yup, that's only to the called party. |
| 23:45.22 | wildcard0 | ah |
| 23:45.24 | jsmith | owh: Does it really need to be to both parties? |
| 23:45.43 | wildcard0 | ok so...hack... put them both in a conference room and bridge in a playback |
| 23:45.44 | owh | jsmith: Yup, it's a warning that says: "This is a satellite call, be patient." |
| 23:46.16 | owh | wildcard0: Hmm... |
| 23:46.32 | wildcard0 | not rally elegant, but it might work for what you're doing in all cases |
| 23:46.56 | owh | jsmith: What happens when I have multiple calls happening simultaneously, multiple people ringing out? |
| 23:47.17 | owh | Uhm, that should have been for wildcard0. |
| 23:47.33 | jsmith | Yeah, the music on hold wouldn't work. |
| 23:47.46 | owh | I mean the conference call... |
| 23:47.47 | wildcard0 | you could just bringe them into multiple dynamic meetme rooms |
| 23:47.48 | jsmith | Why not tell the caller before doing the dial, and use the A() for the called party |
| 23:47.57 | jsmith | Something like this: |
| 23:48.07 | wildcard0 | ya. i like jsmith's solution really |
| 23:48.09 | owh | jsmith: Because while the A() is playing for the called party, the caller gets silence. |
| 23:48.10 | jsmith | exten=>123,1,Playback(sattelite) |
| 23:48.18 | *** join/#asterisk iceyp (~icepick@max.unix.co.nz) |
| 23:48.28 | jsmith | exten=>123,2,Dial(${CALLEDPARTY},,A(satellite)) |
| 23:48.33 | *** join/#asterisk TSCHAK (~thomas@69-161-192-45.clspco.adelphia.net) |
| 23:48.37 | jsmith | owh: That works, right? |
| 23:48.42 | owh | jsmith: That is exactly what I have at the moment. |
| 23:48.48 | jsmith | They're not hearing it at exactly the same time, but who cares? |
| 23:48.48 | iceyp | hey guys, having issue with an inbound service and getting NU tone... could someone please test IAX2/2202@voip.fast.co.nz so I know the issue isn't my end |
| 23:49.24 | Shido | NU tone? |
| 23:49.26 | owh | jsmith: Well, the message lasts 30 seconds, because it is actually also being used to settle the call across the satellite link, which is dynamically changing bandwidth allocations. |
| 23:49.38 | iceyp | Shido you talking to me? |
| 23:49.47 | wildcard0 | im out. later |
| 23:49.53 | owh | wildcard0: Tah |
| 23:50.11 | Shido | thought it was a reference to NuFone, misunderstood |
| 23:50.25 | *** part/#asterisk clark_ (~bclark@64.171.107.5) |
| 23:50.27 | jsmith | owh: Then rewrite Dial to play the annoucement to both callers. |
| 23:50.36 | TSCHAK | who here has used the IAXy's ? |
| 23:50.40 | TSCHAK | I just have some simple questions. |
| 23:50.42 | *** join/#asterisk buddah (~hnic@208.179.86.5) |
| 23:50.57 | Shido | yep |
| 23:51.00 | Shido | shoot , TSCHAK |
| 23:51.00 | TSCHAK | I want to see if I can replace a fleet of Cisco ATA-186s with these buggers. |
| 23:51.05 | Shido | u can |
| 23:51.06 | Shido | :) |
| 23:51.09 | TSCHAK | how simple are they to set up? |
| 23:51.17 | Shido | 40 seconds each box |
| 23:51.20 | Shido | max |
| 23:51.24 | TSCHAK | ... |
| 23:51.26 | TSCHAK | wow. |
| 23:51.27 | owh | jsmith: Hmm... I have also been playing with macros and instead of playing, I macro before I dial, that gives me a ringing during the playback for the called party. |
| 23:51.33 | Shido | user, peer, provision |
| 23:51.42 | TSCHAK | Shido, is there any configuration involved on the iaxy itself? |
| 23:51.48 | Shido | yes |
| 23:52.01 | *** join/#asterisk cbachman (~chatzilla@victory.ece.northwestern.edu) |
| 23:52.02 | Shido | u edit a file and shoot it to the iaxy |
| 23:52.03 | Shido | with iaxyprov |
| 23:52.04 | TSCHAK | basically telling it where the asterisk server(s) are? ok. |
| 23:52.10 | TSCHAK | ok. |
| 23:52.18 | Shido | user, ip, altip, whether u want dhcp or static and codec |
| 23:52.20 | Shido | basically |
| 23:52.25 | TSCHAK | wow. |
| 23:52.26 | Shido | then all the power of the iaxy is done in the dialplan |
| 23:52.45 | TSCHAK | beats the shit out of TFTPing those god awful ata186 configs. |
| 23:52.46 | Shido | consider the iaxy a dumb device that registers as a user and peer to asterisk |
| 23:52.53 | TSCHAK | that's what I Was hoping |
| 23:52.54 | Shido | and plows through nat'd environmens |
| 23:52.55 | TSCHAK | :-) |
| 23:52.55 | Shido | +t |
| 23:53.05 | Shido | so u dont have the same 1 way audio problems with sip |
| 23:53.07 | *** join/#asterisk mesi (~player@dsl-082-083-054-206.arcor-ip.net) |
| 23:53.29 | TSCHAK | Shido, I'm going to put together a plan, and just blow the ISP away with the cost savings. |
| 23:53.50 | TSCHAK | Shido, because the more I look at these iaxys.. i'm just like... geez... who needs a 186 anymore? :-P |
| 23:54.08 | TSCHAK | if they made a dual line version, it'd be over :-) hehe |
| 23:54.15 | iceyp | can somone please test my guest accounts work.... SIP/2202@voip.fast.co.nz & IAX2/2202@voip.fast.co.nz , thanks |
| 23:54.37 | Shido | yep |
| 23:54.46 | Shido | I sent 2 faxes through my iaxy today |
| 23:54.50 | TSCHAK | Shido, cool |
| 23:54.58 | TSCHAK | Shido, we're basically in a remote area of colorado... |
| 23:55.16 | Shido | yeah, digium rocks |
| 23:55.21 | TSCHAK | Shido, and we're looking to do a mid size deployment of about 500 customers, giving them dialtone. |
| 23:57.05 | JunK-Y | anyone knows who's working to port SS7 to * ? |
| 23:57.22 | jsmith | Steve Underwood said something about it a while ago... |
| 23:57.31 | jsmith | I'm not sure if he's done much on it though |
| 23:57.37 | mesi | How can I send a '*' out on my isdn-card to use those special codes like *31*telnr |
| 23:58.01 | mesi | iceyp: I will test them. HOld on. |
| 23:58.02 | JunK-Y | jsmith: thx |
| 23:58.14 | mesi | iceyp: Oh, I can only test the iax one. ;-) |
| 23:59.42 | Shido | incoming |
| 23:59.56 | Shido | -- Called 2202@voip.fast.co.nz |
| 23:59.56 | Shido | <PROTECTED> |
| 23:59.56 | Shido | <PROTECTED> |
