00:00.29 | Slainte | I will use the same script you have included. Let me clear all the old files |
00:00.43 | Slainte | how long should it take for a 400 meg ISO? |
00:02.11 | Darwin35 | depends on your connection |
00:02.26 | Slainte | no it is creating an ISO on the system |
00:02.30 | Slainte | its all local |
00:02.41 | Slainte | its not a download but a backup creation |
00:02.51 | Essobi | CHATZIRRA! CHATZIRRAA! RUUUN! |
00:02.59 | Darwin35 | its downloading a copy of the iso image |
00:03.14 | Darwin35 | onto your system to burn to cd |
00:03.40 | Slainte | its getting one from the web? I thought it was creating one |
00:03.51 | Yatta | hey how can i get a free USA state number for * .. liek a local NYC number?? |
00:04.30 | Qwell | free? good luck with that |
00:04.38 | Yatta | hehehe.... |
00:05.02 | Yatta | somone was tryuing to call my 1700 number and it said it wasn't available |
00:05.13 | Essobi | Well.. Yea. |
00:06.07 | Slainte | running /etc/cron.weekly# ./rapid-scripts |
00:06.23 | Slainte | lemme do a tail -f on the output log |
00:09.42 | *** join/#asterisk Tough_Nuts (~Tough_Nut@204.110.228.254) |
00:10.24 | *** join/#asterisk florz (nobody@odnb-d9baa468.pool.mediaWays.net) |
00:10.35 | *** join/#asterisk Asta2 (~asta2@66.180.175.16) |
00:10.37 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
00:10.37 | *** mode/#asterisk [+o bkw_] by ChanServ |
00:10.43 | Yatta | how can i stup my box from using adpcm..... it sounds crappy :-(( |
00:11.41 | tzafrir_laptop | Runs fine so far. I'll check in an hour or so when it finishes. I figure I need to run it nicer, though |
00:12.06 | Asta2 | Hi. I'm a total asterisk newbie. Anyone knows how do I forward to another address (through one of the sip proxies)? |
00:12.07 | Slainte | mine is running too |
00:12.20 | Slainte | Ill pastebin the logs in half hour |
00:12.38 | Darien | has anyone used AMP, or would recommend a different management interface? |
00:13.29 | tzafrir_laptop | what about an interface for managing a complete system (not only *) ? |
00:13.39 | tzafrir_laptop | anything better than webmin? |
00:14.09 | tzafrir_laptop | e.g: to see the load avarage, to monitor logs |
00:14.37 | Darien | cpanel/plesk/etc |
00:15.01 | tzafrir_laptop | propriatary, and very intusive |
00:15.05 | Slainte | tzaf, any intentions on trying AMP in rapid? |
00:15.13 | tzafrir_laptop | much worse than webmin |
00:15.31 | Slainte | I think a nice SNMP based Nagios config could do the trick |
00:15.37 | tzafrir_laptop | Slainte, I have a half-baked amp package |
00:15.53 | tzafrir_laptop | what can snmp do? |
00:16.00 | Darien | a LOT |
00:16.16 | Slainte | everything you need about your system is avail via nagios/SNMP |
00:16.22 | Slainte | its all about the config. |
00:16.32 | Slainte | I can help you create the core if needed |
00:16.34 | *** join/#asterisk inv_arp (blah@bart.securityone.com) |
00:17.15 | Slainte | a localhost snmp server and preconfigured nagios to keep an eye on your * and server is a nice option for rapid |
00:17.23 | inv_arp | quick need a cheap iax provider with local avail DID (livevoip/VP connect) arent good for me |
00:18.10 | Slainte | teliax.com |
00:18.29 | inv_arp | ahh forgot bout them lemme chk rhx |
00:18.32 | Slainte | but there email server is having trouble all their mail is bouncing because their processing is garbled |
00:18.34 | inv_arp | err thx |
00:18.37 | Slainte | np |
00:19.12 | tzafrir_laptop | well, snmp is nice for pushing and pulling information over the net. But why would I need it for localhost? |
00:19.57 | Darien | unified interface? |
00:20.01 | tzafrir_laptop | The information is already there |
00:20.01 | Slainte | you run it locally and it can then respond to e local query, for all CPU stats etc |
00:20.14 | inv_arp | Slainte: too expensive need not willing to pay more than $10 amonth _inbound only) |
00:20.15 | Slainte | it makes it easier for something like nagios to pull that info |
00:20.18 | Darien | you can run it locally and do one query to get all the info |
00:20.28 | Darien | instead of having to run a bunch of other programs |
00:20.32 | Darien | then you can use MRTG to graph everything |
00:20.36 | tzafrir_laptop | Slainte, sar is also useful for localhost information |
00:20.59 | Slainte | yes but to pull ti and display it. using nagios and rrdtool, or cricket/cacti/mrtg |
00:21.47 | Slainte | I have built a dozen or so NMS's for ISP's cell phone companies etc. SNMP as a core, then custom scripts for everything else |
00:22.05 | *** join/#asterisk Qwell (~north@70-32-102-18.ontrca.adelphia.net) |
00:22.07 | Slainte | using cacti/mrtg/rrdtool/nagios and create a nice pretty front end for the exec idiots to see their pretty cgraphs |
00:22.31 | Slainte | pluss you can have all the evnet handlers you need, so it can autorepair itself, notify etc. |
00:22.50 | tzafrir_laptop | Slainte, anyway, before the graphs I need a slightly more interactive interface |
00:22.56 | Slainte | for security you install and run it locally and if the user decides to accept outside traffic they can change it |
00:23.09 | Darien | can anyone suggest a software VoIP app I can use to test Asterisk ? |
00:23.15 | Slainte | sjphone |
00:23.20 | Slainte | xphone |
00:23.36 | shido6 | xlite |
00:23.41 | Slainte | check the voip-info.org site |
00:23.53 | izo | n/cl |
00:24.02 | Slainte | tzar is it for configuration? |
00:24.09 | Darien | yeah, I've been through voip-info but it's a question of which are good - I'll check out those three now |
00:24.18 | tzafrir_laptop | configuration and monitoring |
00:25.20 | tzafrir_laptop | following the discussion about ssh security in the users mailing list I'm even more convinced that ssh is well understood |
00:25.20 | Slainte | your thoughts on AMP? let me set up a monitoring example for you in a couple of days. |
00:25.42 | Darien | I'm curious about anyone's thoughts on AMP |
00:25.47 | Slainte | did you see my note about removing the root login ability for default |
00:25.49 | tzafrir_laptop | yes, but what about the system itself? |
00:26.32 | tzafrir_laptop | Asterisk is not the only thing on the system. And it is not that perfect that it will run on its own without admin intervention |
00:26.58 | tzafrir_laptop | My guiding principle in the current menu was to make it easy for the admin |
00:27.08 | Slainte | correct, but root users should loging as themselves and then SU to root. not ssh direct as root |
00:27.28 | Darien | sudo to root, then they don't need the root pass |
00:27.29 | Slainte | I dont even use the rapid menu. |
00:27.37 | tzafrir_laptop | rapid-scripts 0.9.8 added that as an option... |
00:27.46 | tzafrir_laptop | maintinance->ssh-sec |
00:28.04 | Slainte | Darien, correct. Lat update I did tzaf the default install permitted root remote ssh by default |
00:28.11 | *** join/#asterisk illek (~mike@ip68-13-238-168.ok.ok.cox.net) |
00:28.28 | tzafrir_laptop | su to root, you mean. You should need root's password |
00:28.38 | Slainte | use sudo, |
00:28.54 | Slainte | regardless root should never be permitted a remote login |
00:28.56 | Slainte | for anything |
00:29.11 | Slainte | my mnod is stuck I think |
00:29.19 | Slainte | mondo |
00:29.21 | Darien | if you use sudo, you don't need the root pass |
00:29.26 | porkchop | sudo has had its sorted history. I still don't trust it. |
00:29.27 | tzafrir_laptop | This default is Debian's default. I think it is sane. The "default installer" (user) will never bother creating a user |
00:29.30 | Slainte | my double-U key is broken also |
00:29.38 | tzafrir_laptop | This will cause too many problems |
00:30.13 | Slainte | your install can force them to create a user |
00:31.09 | Darien | hmm |
00:31.19 | Darien | the Xircom front page says '1.0.2' - it's out of date |
00:31.28 | Darien | says 1.0.5 everywhere else though |
00:32.00 | Darien | just pointing it out |
00:32.05 | Darien | so it can be fixed |
00:32.20 | Darien | xorcom looks interesting |
00:32.40 | Slainte | darien, I have rolled it out to one customer |
00:32.53 | Slainte | I ahve a couple installs out there and I tried rapid for the last one |
00:32.59 | Darien | rapid? |
00:33.07 | Darien | ok, I should say this right now |
00:33.11 | Darien | 'I am a complete and total newbie' |
00:33.13 | Slainte | xorxom is rapid |
00:33.17 | Darien | alright |
00:33.20 | Slainte | xorcom is rapid |
00:33.22 | Darien | right |
00:33.24 | Darien | so you said |
00:33.41 | Darien | I have a Debian machine that I just did apt-get install asterisk on |
00:33.42 | Slainte | fat finger day :) |
00:33.53 | tzafrir_laptop | Darien, http://xorcom.com/rapid.html |
00:34.08 | Darien | but there's no cdr_mysql.so |
00:34.10 | Darien | only pgsql |
00:34.16 | Darien | are the Debian packages good to use? |
00:34.19 | JerJer | no |
00:34.21 | tzafrir_laptop | I should set a redirection from http://xorcom.com/rapid/ to rapid.html |
00:34.25 | JerJer | cvs co asterisk |
00:34.29 | JerJer | cvs co asterisk-addons |
00:34.41 | Darien | ok, well I have the cvs source already |
00:34.45 | Darien | I guess I'll be using that then |
00:35.21 | Slainte | tzaf, let me test the AMP iso at some point. I can also do some NMS stuff for you. |
00:36.05 | Darien | how competitive is the VoIP market at the moment? it seems pretty dormant right now |
00:36.43 | tzafrir_laptop | Slainte, in Debian the nagios packages seem to include their own separate snmp agent |
00:36.47 | Slainte | very very competitive. The * market is just starting to pick up. |
00:37.03 | Slainte | tzafr, they dont base it on UCDSNMP? |
00:37.22 | Slainte | hmmmm |
00:38.38 | nvadekar | anyone got fwd working over iax2 with 1800 numbers? i can get x612 working, but *1800. numbers claim to be invalid extentions |
00:39.04 | tzafrir_laptop | Slainte, hmm, I probably mis-read package descriptions |
00:39.54 | Slainte | my mono looks like it is hanging. |
00:41.00 | Slainte | I have a 1.iso file but I have 2 mondo processes still running |
00:41.12 | Slainte | time stamp on it is 35 minutes old |
00:42.18 | Slainte | [Main] libmondo-archive.c->offer_to_write_floppies#2638: Warning - can't find a 1.44MB floppy device *sigh* |
00:42.42 | Slainte | and stuck on this |
00:42.43 | Slainte | ---evalcall---1--- Running mkisofs to make ISO #1 |
00:42.43 | Slainte | ---evalcall---2--- TASK: [**********..........] 50% |
00:45.32 | tzafrir_laptop | My own cron-weekly is stuck in something quite similar |
00:45.48 | *** join/#asterisk Mike (~mike@201.135.46.182) |
00:46.50 | tzafrir_laptop | anyway, one of the temp files used by mondo is /tmp/mojo-jojo.blah.PucOsX |
00:47.07 | tzafrir_laptop | I do hope those last six chars are random |
00:47.45 | Slainte | lemme check |
00:48.13 | Slainte | I have /tmp/mojo-jojo.blah.FelpUA |
00:48.20 | Slainte | so yes I think it is |
00:48.29 | tzafrir_laptop | the child mondoarchive keepins polling a pipe. How do I check who's on the other side of a pipe? |
00:48.44 | Slainte | pstree |
00:49.03 | Mike | canreinvite and invite =yes will make peers not pass trou asterisk right? |
00:49.23 | Slainte | not even seeing it in the pstree |
00:49.57 | Slainte | ah find it |
00:50.01 | ManxPower | there is no such option as invite= |
00:50.01 | Slainte | -rapid-scripts---mondoarchive--- |
00:50.13 | Mike | ManxPower, reinvite=yes |
00:50.15 | Mike | ManxPower, sorry |
00:50.29 | ManxPower | There is no such option as reinvite= either. |
00:50.30 | tzafrir_laptop | I normally use ps fax to see the tree |
00:50.42 | Mike | ManxPower, i have a zap channel in my local network but my sipphone connect to a remote server and i would like not to pass trou the remote server when using zap |
00:50.46 | Slainte | do you have 2 instances of mondo running? |
00:50.47 | ManxPower | It's canreinvite=yes|no. Read the source of chan_sip.c for proof |
00:51.50 | Slainte | yeah more detail in the fax, i should use it more, just pstree bad habit |
00:52.02 | Mike | im doing iax2 link to my local server |
00:52.10 | Mike | but my sipphone has to be connected on the remote server |
00:53.39 | Slainte | 2 FIFO pipes created on each mondo process |
00:53.42 | Slainte | and it is stalled |
00:54.04 | tzafrir_laptop | the child waits on a pipe from the parent. The parent waits on read(0,... |
00:54.06 | tzafrir_laptop | :-( |
00:55.46 | Slainte | I have to go. Rugby party. I can try different mondo options 2moro |
00:55.54 | tzafrir_laptop | The option -F is missing :-( |
00:56.02 | Slainte | email me if you have ideas or suggestions. |
01:02.54 | *** join/#asterisk Bifrost_ (~bifrost@server01.minions.com) |
01:02.56 | Bifrost_ | bleh! |
01:03.37 | Bifrost_ | uhm, anyone else got broadvoice? |
01:04.12 | porkchop | heh |
01:04.13 | porkchop | I did |
01:04.17 | porkchop | for 20 minutes. |
01:04.27 | Bifrost_ | heh |
01:04.43 | *** join/#asterisk lohelle (slamm@213.161.252.253) |
01:04.43 | Bifrost_ | so far it works, but I can't get it to pass an incoming call to asstrix |
01:04.52 | Bifrost_ | and I put their stupid patch in, but still no dice |
01:05.48 | *** join/#asterisk RoyK (~roy@83.80-203-29.nextgentel.com) |
01:05.56 | lohelle | does anyone run zaphfc on bristuff 0.2.rc7a? not working here.. :\ |
01:06.08 | lohelle | oohh! |
01:06.19 | lohelle | readme file... |
01:06.42 | *** join/#asterisk Shuri (~shuri@dsl.speedline209.226.electronicbox.net) |
01:06.55 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com) |
01:06.55 | *** mode/#asterisk [+o anthm] by ChanServ |
01:07.05 | porkchop | Bifrost_: the reason may or may not be you. broadvoice just sucks. I had it configured correctly and working. Passed two calls in. Then it stopped passing in calls. No config changes. |
01:07.34 | Bifrost_ | haha nice |
01:07.37 | Bifrost_ | what a buncha fookers |
01:07.45 | porkchop | Bifrost_: I could always do calls out tho. |
01:07.47 | porkchop | yeah |
01:07.48 | Bifrost_ | well, ok, I guess I'm in the market for a 702 number |
01:09.54 | porkchop | I'm using stanaphone.com, but they only have NY numbers. Being located in NY, thats of little issue to me. |
01:11.42 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivfmh0.dialup.mindspring.com) |
01:13.24 | Bifrost_ | hrm |
01:13.40 | Bifrost_ | well, anyone got a place to reccomend for a 702 number that works with asstrix? |
01:13.47 | iMediax | they work fine for me |
01:14.12 | Bifrost_ | imediax: broadvoice? |
01:14.16 | iMediax | yep |
01:14.26 | Bifrost_ | hrm, did you have to do anything special? |
01:15.02 | iMediax | followed the wiki for BV |
01:15.24 | Bifrost_ | damn, I did too |
01:15.47 | iMediax | umm, not the main one! the *update* in the user section |
01:18.21 | Bifrost_ | crap, someone stole the SCSI card out of this damn AXi 1U |
01:19.32 | Darien | ouch |
01:19.41 | Darien | I wish I were that good of a thief |
01:20.23 | Bifrost_ | heh |
01:20.42 | Bifrost_ | well, I probably have one in my pile of crap, but I am really not sure if I have a sun compatible one |
01:25.14 | Bifrost_ | I wonder if I should see if I can turn off their voicemail... |
01:26.47 | Bifrost_ | hrm, 500Mhz USparc-IIe |
01:27.24 | *** join/#asterisk goatmilk (~travis@user-69-73-1-138.knology.net) |
01:31.13 | goatmilk | lo |
01:34.50 | Luke-Jr | any ideas why I wouldn't be able to login to an extention from outside my network? |
01:39.35 | tzafrir_laptop | is asterisk-perl alive? |
01:41.21 | tzafrir_laptop | has not been updated since 8-2003 |
01:41.48 | tzafrir_laptop | Luke-Jr, can you connect to that port? |
01:42.03 | tzafrir_laptop | what protocol? |
01:42.16 | tzafrir_laptop | is NAT involved? |
01:44.05 | nvadekar | anyone played with zgsmplay |
01:51.32 | letherglov | perl is dead, long live COBOL |
01:51.49 | *** join/#asterisk jets (~jetsn@xyharley.dsl.xmission.com) |
01:52.15 | tzanger | whoohoo iax2 remote stats added to cvs |
01:52.21 | tzanger | just gotta get that jitter buffer in there now :-) |
01:53.28 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
01:54.20 | bjohnson | iax2 remote stats ? |
01:54.20 | Luhiwu | does anyone knows if res_config_mysql works with 1.0.5 ? |
01:56.06 | CoaxD | i can get it to not share irqs between system devices and my x100p and tdm400p if i dont enable a lot of the onboard bells and whistles. But i need the damn soundcard and when i turn it on, it reconfigures the whole damn system to share every irq in the book |
01:56.12 | CoaxD | (with system devices) |
01:56.19 | *** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net) |
01:56.19 | CoaxD | new mobo time i suspect |
01:59.02 | Bifrost_ | HAH! fucking A |
01:59.53 | *** part/#asterisk kietlak (~kietlak@11-mo3-6.acn.waw.pl) |
02:04.27 | *** join/#asterisk ScythelX (Fleb@pc-24-181-176-166.sbi.ct.charter.com) |
02:04.33 | ScythelX | hello all |
02:04.58 | ScythelX | is anyone using ser along with asterisk - if so do you need a register => in sip.conf so it connects to the ser server? |
02:06.53 | tzanger | CoaxD: shared IRQs aren't so bad |
02:06.55 | tzanger | look at my server |
02:07.09 | tzanger | <PROTECTED> |
02:07.19 | tzanger | this server is the media box (nfs, samba shares) for my houes |
02:07.47 | Luke-Jr | tzafrir_laptop: what port? SIP. NAT can be involved or not, it has two routes possible. |
02:07.58 | CoaxD | tzanger: mmm |
02:08.17 | CoaxD | tzanger: I was told that x100p or tdm100p irq sharing be BAD |
02:08.30 | *** join/#asterisk john8675309tm (~pabt@207.177.124.87) |
02:08.35 | tzanger | it is |
02:08.39 | CoaxD | tzanger: And that x100p specifically - sharing would not operate at all |
02:09.08 | CoaxD | tzanger: That said, i have not tested it |
02:09.34 | john8675309tm | My have a phone that is having an issue authenticating is there a way I can view what password they are using in sip debug? |
02:10.55 | CoaxD | tzanger: shall i give it a try? or? |
02:11.11 | CoaxD | tzanger; Am i wasting my time? (with x100p) |
02:11.19 | tzanger | not sure |
02:11.21 | tzanger | I don't use it |
02:11.30 | CoaxD | tzanger: sigh. k. i'l try |
02:12.00 | goatmilk | twisted says he's a phat philly |
02:12.14 | kram | omg omg make twisted put his clothes back on! |
02:12.16 | kram | :) |
02:13.24 | Silik0n | hah |
02:13.30 | *** join/#asterisk jets (~jetsn@xyharley.dsl.xmission.com) |
02:14.08 | brc_ | bkw says: "hahahhahahhahah, that's not right" |
02:14.50 | *** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || chan_zap in 1.0.4 had a bug, so 1.0.5 has been released || <kram>omg omg make twisted put his clothes back on! |
02:14.51 | Luke-Jr | hmm |
02:15.22 | Luke-Jr | Is it possible to have * authenticate using by IP? |
02:15.27 | brc_ | eh? |
02:15.30 | tzanger | kram: haha |
02:15.31 | brc_ | be more specific |
02:15.35 | brc_ | authenticate to what |
02:15.38 | Luke-Jr | eg, don't ask for passwords while I'm on my LAN |
02:15.42 | Luke-Jr | auth phones |
02:15.46 | goatmilk | RING THE BELL |
02:15.48 | goatmilk | NOW NOW NOW |
02:16.10 | Luke-Jr | only authenticate when I connect from the net |
02:16.12 | john8675309tm | is it possible to view a password using sip debug? |
02:16.12 | goatmilk | the eagles are coming! the eagles are coming! |
02:16.23 | tzanger | kram: got a few moments to explain iax2 bridge optimization? That is specifically what was causing those weird iax2 timestamps |
02:16.23 | *** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || chan_zap in 1.0.4 had a bug, so 1.0.5 has been released |
02:16.25 | Luke-Jr | john8675309tm: I'm glad its not |
02:16.28 | brc_ | who is goatmilk? |
02:16.47 | john8675309tm | Not even the hash of the password? |
02:17.10 | letherglov | what is authentication by IP? |
02:17.18 | letherglov | it's entirely meaningless |
02:17.23 | Luke-Jr | letherglov: no it's not |
02:17.25 | tzanger | CoaxD: I just tested my irq sharing to make sure too |
02:17.33 | goatmilk | bob |
02:17.37 | letherglov | Luke-Jr, ever changed your IP in windows? |
02:17.38 | tzanger | I am *hammering* the network while on a milliwatt() call |
02:17.39 | letherglov | boom. ok |
02:17.42 | file | goatmilk: I'd like you to meet, Bob, Dill, Potato, and OMG MUFFINS |
02:17.42 | letherglov | now you're authenticated? |
02:17.44 | letherglov | what the hell is that? |
02:17.51 | Luke-Jr | letherglov: you overlook physical security |
02:17.59 | goatmilk | file: pimpkin muffins are good. |
02:18.01 | *** join/#asterisk Mother__ (~m@53.Red-217-126-93.pooles.rima-tde.net) |
02:18.14 | letherglov | ok |
02:18.17 | Luke-Jr | letherglov: If I can ensure my LAN is trustworthy, it doesn't hurt |
02:18.18 | letherglov | maybe if it's in your house |
02:18.22 | file | mmm muffins |
02:18.23 | Luke-Jr | exactly |
02:18.29 | letherglov | and the usps guy doesn't break it |
02:18.37 | letherglov | and your nephew doesn't like calling 1-900 #'s late at night |
02:18.39 | letherglov | you're ok |
02:18.45 | Luke-Jr | I don't support 1-900 |
02:18.46 | goatmilk | file: are you on the phone? |
02:18.50 | file | goatmilk: maybe |
02:18.54 | goatmilk | brc_: i was loud on the phone for a reason! |
02:18.58 | letherglov | file, maybe on the phone? |
02:19.03 | file | mayyyyybe |
02:19.08 | goatmilk | file: eh, canadian? |
02:19.16 | file | goatmilk: yesssss |
02:19.19 | letherglov | are you maybe on IRC too? |
02:19.33 | letherglov | oh, do you have to have your headlights on while making calls too? |
02:19.47 | file | POOSE MENIS |
02:20.11 | *** join/#asterisk {-award-} (admin@pD9E0E92C.dip.t-dialin.net) |
02:20.44 | goatmilk | i am not a nascar fan |
02:20.48 | goatmilk | ya'll are retarded |
02:21.05 | file | goatmilk: have you rode the moose penis? I thought so. |
02:21.43 | Mother__ | hi all |
02:21.47 | goatmilk | if by moose penis you mean your mother.. then yes, i have rode the moose penis mucho tiempos |
02:21.52 | kram | lets move it to #asstricks please |
02:23.02 | *** join/#asterisk jtodd (~jtodd@garthim.fox-den.com) |
02:24.30 | letherglov | ok Luke-Jr how much overhead are we talking with the context and username/pass combo? |
02:25.54 | ScythelX | does * need to be registered to SER in order for calls to be passed from a SIP phone client to the * pbx? |
02:26.24 | file | asterisk doesn't need SER in order to operate at all... |
02:26.40 | ScythelX | i know this, but I want to use SER |
02:27.15 | file | uh huh |
02:27.20 | techie | yeah |
02:27.30 | letherglov | eh? |
02:27.32 | letherglov | sipphone.com? |
02:27.45 | letherglov | once you're connected to their SER with a client |
02:27.54 | letherglov | you need to register the * client at the other end |
02:27.59 | letherglov | with the register => in the sip.conf |
02:28.03 | letherglov | works like a charm otherwise |
02:28.11 | ScythelX | ok thats what i thought |
02:28.11 | letherglov | although, they upgraded some shit with some rfc silence something |
02:28.14 | ScythelX | thank you |
02:28.15 | letherglov | and it's all pissy at me |
02:28.33 | letherglov | my university cross connected their pbx to sipphone |
02:28.42 | Luke-Jr | letherglov: overhead? |
02:28.47 | letherglov | so I can call my * box at home for free through the university pbx |
02:28.48 | letherglov | it's nice |
02:34.00 | PoWeRKiLL | someone already got this error Message count requested for mailbox 101 but voicemail not loaded. ??? |
02:35.11 | *** part/#asterisk Frawg (~kamy@frawg.user) |
02:38.05 | Firestrm | PoWeRKiLL, sounds like a voicemail.conf misconfigureation.. most likely a module did not load.. check debug |
02:38.55 | nvadekar | has anyone here played with suse asterisk rpm, they offered it on their 2.6 kernel builds, is it considered stable on 2.6 with suse? |
02:39.11 | *** join/#asterisk chetan (freetibet@24-193-188-21.nyc.rr.com) |
02:39.35 | bjohnson | letherglov: how do authenticate to get access to other extensions, dial out, etc? |
02:39.52 | Firestrm | nvadekar, im running stable on redhat.. no problems to report.. well only a few, but i dont think they have to do with the kernel |
02:40.20 | Firestrm | nvadekar, er.. i mean FC3.. |
02:40.45 | nvadekar | cool, would you consider using it in production, or still staying with 2.4 in the real world? |
02:41.37 | Firestrm | nvadekar, i am about to put 3 FC3 boxes running asterisk 1.0 stable, into productions.. im not sweating.... much ;) |
02:42.14 | JerJer | use head |
02:42.20 | denon | live a little .. |
02:42.25 | denon | and be prepared to die a little <G> |
02:42.44 | nvadekar | please let us know how it goes after the go live. do you feel their is any value in carrier grade linux for asterix? |
02:43.18 | nvadekar | asterisk, sorry about the typo |
02:43.35 | tzanger | JerJer: hold off on that switch-3 update unless you're itching for it.. the remote stats aren't complete yet (i.e. not ocmplete wtih the new jitter buffer) |
02:43.39 | Firestrm | nvadekar, itsallright.. im one big typo.. |
02:44.04 | *** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net) |
02:44.39 | *** join/#asterisk davinder (~gonefishi@xtreme-28-156.dyn.aci.on.ca) |
02:44.50 | davinder | hello all |
02:45.04 | *** join/#asterisk netsurfer (~netsurfer@dreambox.myvnc.com) |
02:45.20 | Firestrm | nvadekar, funny story.. i though i was really cool when i was on my registrar site and i was able to register asterix.ca.. i only noticed my error AFTER i hit the submit key with all my CC info typed in.. |
02:45.41 | davinder | what hardware is needed to run astrisk? |
02:45.42 | PoWeRKiLL | I use voicemail conf with realtime database |
02:46.17 | nvadekar | firestrm, haha.... what city you in |
02:46.25 | *** part/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
02:46.45 | Firestrm | so i guess i will go ahead with my plans for a asterisk newby site on asterix.ca... Im in Victoria |
02:47.21 | nvadekar | firestrm, cool, I am in bermuda, but if I every have to move back to canada, I just might be your neighbour |
02:47.21 | Darien | I'm trying to set up scandsp, and everything says to copy app_txfax and app_rxfax, but I can't figure out where those are |
02:47.22 | bjohnson | fantasterisk.ca ? |
02:47.51 | bjohnson | asterix is likely trademarked |
02:48.04 | bjohnson | French comic book |
02:48.19 | nvadekar | prob right, i read it as a kid. |
02:48.24 | Firestrm | nvadekar, if i could find work in bermuda, i would be your neighbour tomorrow :) I hate cold.. even slightly cold.. i much preffer to be warm |
02:49.00 | nvadekar | firestrm, if you like linux, you would hate it here, all ^&*@@##@ m$ here.... Uggggg. |
02:49.46 | *** join/#asterisk SeaForth (~SeaForth@c-24-1-126-202.client.comcast.net) |
02:50.13 | Firestrm | nvadekar, ive long ago learned how to convert offices over to linux.. just run it.. and eventually ppl will notice that.. hey he never crashes.. i wonder why ;) |
02:50.52 | Firestrm | bjohnson, im using fwd. but i find the audio quality leaves much to be desired |
02:51.21 | nvadekar | firestrm, does not work here, I tried many times, only small clients are ok, big insure co's seem to have a badge of dishonour keeping bill in dough |
02:51.23 | *** join/#asterisk clint_ (~clint@snap.helixsystems.com) |
02:51.42 | letherglov | we're a solaris house |
02:51.44 | letherglov | works nicely |
02:51.49 | bjohnson | Firestrm: in and out through FWD shouldn't cause issues unless big lag |
02:51.54 | *** join/#asterisk Dalion (anon@HSE-QuebecCity-ppp3497081.sympatico.ca) |
02:52.02 | Dalion | hey all |
02:52.10 | Firestrm | bjohnson, i finally got both ports on my spa-3000 working.. and wouldnt you know it.. same no hangup detect problem i had on my x100p.. this HAS to be a telco problem.. no i just have to prove it to them |
02:52.14 | letherglov | if asterisk's t-1 cards worked under solaris, i think there would be a lot more corporate buy-in |
02:52.19 | letherglov | linux-and-asterisk is a little scary |
02:52.23 | Dalion | quick question with sveasoft on a wrt54g.. how can i add Qos .. since only has sip in there... |
02:52.40 | letherglov | Dalion, you buy the latest firmware? |
02:52.47 | nvadekar | bjohnson, you have any luck getting *1800 numbers working with fwd, I got 612,613 working and callback, but says no valid extention with *1800. |
02:53.08 | Dalion | i tried adding in restrictions page.. the * 4569 ports.. but still doesnt shot in wqos.. now if i oonly knew where get_qossvc() function is in those asp |
02:53.08 | Firestrm | Dalion, or check emule.. |
02:53.08 | Dalion | lol |
02:53.30 | letherglov | I'd send you the latest shit |
02:53.37 | letherglov | but they embed security identifiers in there |
02:53.40 | Dalion | Firestrm: ? |
02:53.41 | letherglov | ;-\ |
02:53.44 | letherglov | :-\ |
02:53.44 | bjohnson | nvadekar: I've come to the conclusion that Canadian 800 numbers often don't work from US based voip providers .. so I don't use FWD for that |
02:53.54 | Dalion | lether i got |
02:54.00 | Darien | :( |
02:54.06 | Dalion | fd |
02:54.20 | letherglov | they're always ripping on us for sending mexicans across the border |
02:54.26 | clint_ | Evening, folks. Is anyone running on linux amd64 and has music on hold working? |
02:54.27 | letherglov | ... ;-) |
02:54.33 | nvadekar | bjohnson, I am trying to call 18008584000, it is novell, and a us based 1800 number |
02:54.41 | letherglov | clint_, mpg123 fucked? |
02:54.53 | Dalion | Firmware Version: Alchemy-6.0-RC6a v3.01.3.8sv |
02:54.53 | Dalion | <PROTECTED> |
02:54.56 | clint_ | letherglov: Yessir, that's the fact |
02:55.01 | jets | Who is calling novell :P |
02:55.04 | jets | and why |
02:55.05 | jets | hahaha |
02:55.16 | nvadekar | nvadekar is, they own SuSE, that is why |
02:55.20 | Dalion | letter what u mean embed ? |
02:55.38 | jets | I know they own SuSE :P |
02:55.39 | letherglov | they have some stuff unique to every user for every download |
02:55.47 | clint_ | letherglov: mpg321 works, sounds crappy, mpg123 won't compile on amd64, fixing is beyond my abilities. |
02:55.49 | jets | rumor has it they are working with Asterisk as well *wink* |
02:56.05 | Dalion | ah .i see a reg firle in / calre /reg |
02:56.15 | bjohnson | clint_: check the wiki for sox and madplay .. maybe one of those will work for you |
02:56.17 | Dalion | i dotn think so |
02:56.28 | nvadekar | yeah, I am running their rpm on my suse9.2 laptop for lab testing, but compile csv for customers. |
02:56.37 | Dalion | since its a download from a forum.. idont think they mod the bin file each time.. lol |
02:56.40 | clint_ | bjohnson: thanx, looking now... |
02:56.50 | Dalion | emule whas for what ? |
02:56.50 | jets | oh do u work at novell |
02:57.24 | nvadekar | ,no, just really like suse, I got my RHCE, but I much prefer Suse distro |
02:57.31 | jets | mm i see |
02:57.32 | file[laptop] | OH, MY, GOD, BECKY |
02:57.43 | Dalion | theres only edonkey in there |
02:58.13 | Firestrm | letherglov, your just pissed that canada grows better pot than USA :).. |
02:58.22 | letherglov | it's true :-( |
02:58.37 | letherglov | and has cheaper prescription drugs |
02:58.46 | Firestrm | letherglov, and you can walk down the street smoken one and not get busted |
02:58.57 | nvadekar | I alway though street price was more important, not were you got it from. |
02:59.29 | Firestrm | nvadekar, im not an expert.. i jest smell it everywhere i go now.. |
02:59.35 | letherglov | yeah, but just think, you can buy it in crappy canadian dollars...which are now a lot stronger against the dol...nevermind. |
02:59.44 | Firestrm | lol |
02:59.59 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
02:59.59 | *** mode/#asterisk [+o bkw_] by ChanServ |
03:00.09 | Firestrm | no more northern paseo.. |
03:00.28 | Firestrm | now its northern Yen.. |
03:00.30 | nvadekar | I with it was still a peso, so much for earning us$$$ |
03:01.29 | Firestrm | bjohnson, we just need to design over the border cruze missiles with bails of pot for a warhead.. that will mellow em out |
03:01.44 | clint_ | bjohnson: why wasn't I able to find that in google? Oh yeah, blind as a bat. Works great, thanks for the info! |
03:02.22 | PoWeRKiLL | what is the best way to do cdr billing ? |
03:02.35 | *** join/#asterisk mtqh (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net) |
03:02.38 | Firestrm | as china's dollar goes up, so will canada.. |
03:02.40 | bjohnson | send "Guido" |
03:02.53 | Darien | it's not that our dollar is going up, it's that the US dollar is going down |
03:03.13 | Firestrm | Darien, i think its a bit of both isnt it? |
03:03.28 | Darien | a little of both, but our dollar really isn't that strong |
03:03.34 | Darien | it's just getting stronger in relation to the dollar |
03:03.41 | nvadekar | every 100 billion on iraq seems to be another 10c off the us dollar |
03:03.43 | Darien | so we go up as they go down |
03:04.19 | Nugget | yay iPods. |
03:04.38 | *** join/#asterisk DaLion (anon@Toronto-HSE-ppp3771217.sympatico.ca) |
03:04.46 | bjohnson | haven't checked .. but I've been told that CND was equal or greater value than USD pretty much ALL the time before the 1970s |
03:04.49 | Nugget | my iPod does everything I want it to do. why would I hack it? |
03:05.02 | Darien | bjohnson: not all the time, but for a period of time, yes |
03:05.05 | Firestrm | P2P over bluetooth.. swaps songs when in range |
03:05.22 | *** join/#asterisk dhewg (~dhewg@dsl-082-082-137-235.arcor-ip.net) |
03:05.26 | Darien | bjohnson: it rose above the US dollar for a few months (years?) then went back down |
03:05.28 | nvadekar | chan_iax2.c:5352 socket_read: Call rejected by 65.39.205.121: No such context/extension is what I get trying to call *1800 on fwd, does anyone have this working? |
03:05.32 | Nugget | bluetooth is too slow for that, and it doesn't sound very interesting. |
03:05.57 | nvadekar | but 612,613 and callback all work.... |
03:06.37 | Darien | so where can I get app_rxfax.c? it doesn't seem to come with spandsp |
03:07.11 | Darien | or at least, find says so |
03:07.12 | Firestrm | anyone here know how hangup signals from telco are supposed to work.. i have to prove to telus that their line is broken.. |
03:07.14 | tzanger | Darien: it's in the same dir as the spandsp stuff |
03:07.31 | Darien | hmm |
03:08.03 | Firestrm | arnt they supposed to reverse polarity or something? |
03:08.36 | nvadekar | firestrm, is there alot of opensource acceptance in vic these days, how about van? |
03:08.54 | *** join/#asterisk anthm (~anthm@CPE-69-76-83-52.wi.rr.com) |
03:08.54 | *** mode/#asterisk [+o anthm] by ChanServ |
03:09.04 | Firestrm | bc in general seems to be Opensource friendly.. |
03:09.36 | nvadekar | just servers, or are there people accepting it on the desk as well? |
03:09.49 | bjohnson | Firestrm: the SPAs info page shows the line voltage |
03:09.57 | bjohnson | watch it during a call |
03:10.37 | bjohnson | tzanger can give you some voltages .. I think it was -48VDC on hook and -6VDC offhook |
03:11.08 | Firestrm | nvadekar, most ppl use windows for desk, but im seeing more and more switching.. every time M$ makes some stupid announcement about not supporting older versions or not allowing patches.. |
03:11.11 | bjohnson | tzanger: I had a problem (not the same as yours) that by watching the line voltages I determined was a reverse wired jack |
03:11.20 | mikegrb | Firestrm: reverse polarity is called disconnect supervision, sometimes extra |
03:11.20 | Firestrm | bjohnson, i'll try that.. |
03:11.29 | tzanger | bjohnson: ok |
03:11.52 | mikegrb | costs extra that is |
03:12.18 | Mother__ | reverse polarity comes standard in Spain, but it's very short and * misses it if it's inside an IVR or voicemail |
03:13.11 | Essobi | you could make an inline piece of hardware that watches for the reverse and holds it longs. |
03:13.14 | Essobi | loger. |
03:13.16 | Essobi | longer |
03:13.17 | Essobi | baaah |
03:13.25 | Mother__ | hehe yes indeed |
03:13.37 | Essobi | would be a farily simple circuit |
03:14.10 | Mother__ | I still would like to know if * is even watching for a reverse polarity when inside voicemail |
03:14.25 | Firestrm | bjohnson, it was 50v until it picked up, then went to 28ma and 6V. but after hang up it stayed there for an additional 10 sec before returning to onhook |
03:14.56 | Mother__ | it doesn't look like it at least, if the hangup comes before VM kicks in, it's detected fine, otherwise it's a the timeout of silence |
03:16.49 | brc_ | vim RULES |
03:16.58 | brc_ | it's the shiznit maan! |
03:17.05 | bkw_ | no emacs is the shiznit man... you know it.. |
03:17.21 | brc_ | emacs is a great os dude |
03:17.29 | bkw_ | and you wanna run it |
03:17.34 | brc_ | too bad it lacks a decent text editor |
03:17.34 | Silik0n | GENTOO IS FOR RICERS |
03:17.39 | brc_ | ya think? |
03:17.47 | brc_ | FUNROLLLLL LOOOOPS BAABY! |
03:17.59 | brc_ | OPTIMIZED! |
03:18.09 | bkw_ | stty erase ^H |
03:18.11 | brc_ | watching stuff scroll by for hours makes me a linux expert! |
03:18.26 | brc_ | mandrake expatriate syndrome. |
03:20.31 | DaLion | lshey |
03:20.54 | DaLion | anyone can figuere out why i ALWAYS get a unable to create channel of type SIP cause 3 . everyone busy |
03:21.02 | bkw_ | well lets see |
03:21.05 | bkw_ | BECAUSE ITS BUSY |
03:21.11 | DaLion | on a dial(SIP/bob|30|tr) |
03:21.15 | DaLion | nah |
03:21.20 | DaLion | its not even rigning my xten |
03:21.20 | anthm | perl beats php into a bloody pulp |
03:21.21 | bkw_ | check include/asterisk/cause.h |
03:21.36 | bkw_ | perl R0CKS!!! |
03:21.43 | Silik0n | perl < * |
03:21.44 | brc_ | ruby baby! |
03:21.45 | DaLion | heeh sure perl rocks |
03:21.51 | Silik0n | linsun |
03:21.53 | DaLion | nah it r0cks |
03:21.54 | Silik0n | BSD BABY |
03:21.55 | nvadekar | exten => _393.,2,Dial(IAX2/${FWDNUMBER}:${FWDPASS}@iax2.fwdnet.net/${EXTEN:3},60,r) is there anything wrong with this use of variables? it does not work |
03:21.58 | Silik0n | linsux |
03:21.58 | bkw_ | bad baby bad |
03:22.37 | netsurfer | nvadekar - u regged for an IAX2 account with fwd ? |
03:23.04 | nvadekar | yup, the 612 works, call back works, but these variables are not being read from my globals properly |
03:23.39 | nvadekar | if I hard type in my user/pass in that line, it seems to work |
03:24.11 | mikegrb | nvadekar: you shouldn't put them there anyway, create a user/peer in iax2.conf |
03:24.13 | netsurfer | thats odd |
03:24.18 | DaLion | sip show peers donest show me |
03:25.12 | *** join/#asterisk eipi (~eipi@OL128-44.fibertel.com.ar) |
03:26.19 | nvadekar | if I setup a peer in iax.conf, what do I put for my dial statement? |
03:27.57 | DaLion | nah it r0cksfd |
03:28.15 | Silik0n | <PROTECTED> |
03:29.58 | *** join/#asterisk Ranma_ (~ranma@pool-64-223-113-75.burl.east.verizon.net) |
03:31.42 | Ranma_ | has anyone gotten spandsp to work? |
03:31.45 | Ranma_ | to receive faxes? |
03:32.00 | bkw_ | Ranma_, get out |
03:32.03 | bkw_ | you can't say Hi |
03:32.04 | bkw_ | get out |
03:32.05 | bkw_ | rude |
03:32.06 | bkw_ | rude |
03:32.12 | brc_ | bkw_, that is idiotic |
03:32.13 | brc_ | idiotic |
03:32.18 | *** kick/#asterisk [brc_!~brian@bkw.developer.and.friend.of.asterisk] by bkw_ (bkw_) |
03:32.21 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
03:32.24 | Ranma_ | pardon me... |
03:32.24 | *** kick/#asterisk [brc_!~brian@bkw.developer.and.friend.of.asterisk] by bkw_ (bkw_) |
03:32.28 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
03:32.29 | Ranma_ | good evening everyone :) |
03:32.29 | *** kick/#asterisk [brc_!~brian@bkw.developer.and.friend.of.asterisk] by bkw_ (bkw_) |
03:32.33 | *** join/#asterisk DaLion (anon@69.156.64.191) |
03:32.39 | bkw_ | haha |
03:32.41 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
03:32.46 | Silik0n | brc_ is bouncy tonight |
03:32.53 | brc_ | it's SILLY when people say hello just for the hell of it |
03:33.01 | bkw_ | chan_ss7 should be out soon too guys |
03:33.03 | Silik0n | hello |
03:33.04 | bkw_ | :P |
03:33.05 | brc_ | people rarely reply to it anyway |
03:33.10 | brc_ | OMFG HELLO EVERYBODY! |
03:33.13 | brc_ | how are you tonight! |
03:33.18 | brc_ | so nice to see you! |
03:33.28 | porkchop | I seem unable to register (SIP) register with iaxtel...I get "wrong password" Pretty sure I have it right...is there a problem over at iaxtel or am I wrong? |
03:33.42 | Ranma_ | apprently, I've spawn some heated conversation now :) |
03:34.59 | CoaxD | figured out the callerid probs |
03:35.12 | CoaxD | also, figured out a bug along the way with busydetect |
03:35.43 | CoaxD | ...which explains the hours and hours of busy recordings sufficed by chad having to restart asterisk cuz he doesnt know what the hll is the deal |
03:36.27 | CoaxD | there are times when asterisk doesn't detect a busy. so, there she sits, beeping busy for hours at a time.. and asterisk doesnt see it |
03:37.04 | CoaxD | re: callerid probs... had rxgain set WAY too high. had to set it at like -7.5 in order to make it go. (I'm about 200 feet from the telco switch. no resistance whatsoever. *lol*) |
03:38.05 | brc_ | <PROTECTED> |
03:38.05 | brc_ | <PROTECTED> |
03:38.05 | brc_ | <PROTECTED> |
03:38.05 | brc_ | <PROTECTED> |
03:38.05 | brc_ | <PROTECTED> |
03:38.06 | brc_ | <PROTECTED> |
03:38.08 | brc_ | <PROTECTED> |
03:38.27 | CoaxD | it should be noted that i think the morons out here in the sticks really need to get with the times and use a switch that at least supports kewlstart |
03:38.48 | letherglov | hmm |
03:38.50 | CoaxD | the switch is a nortel dms100. |
03:38.54 | letherglov | chan skype using their modules under linux? |
03:39.03 | CoaxD | you'd think it had the capability of using kewlstart |
03:39.53 | brc_ | I've said too much already |
03:40.48 | eipi | yes too much |
03:40.51 | eipi | wow |
03:40.55 | DaLion | hehe |
03:40.59 | DaLion | skype works good ? |
03:41.10 | brc_ | yup |
03:41.21 | letherglov | is that a reverse of their protocol |
03:41.27 | eipi | and how it works? getting last cvs and configuring it? |
03:41.28 | letherglov | or a skype-based implementation? |
03:41.31 | brc_ | :) |
03:41.33 | eipi | ;) |
03:42.00 | DaLion | holyl valance is hot |
03:42.07 | brc_ | well what we did is just [REDACTED] |
03:43.01 | nvadekar | does this mean we can call skype users with a channel, or we can soon use the skype protocol for interpbx? or both |
03:43.32 | bkw_ | Skype/username |
03:43.33 | bkw_ | sure |
03:43.40 | brc_ | who is this "we" |
03:43.49 | eipi | haha |
03:44.15 | Nugget | heh |
03:44.16 | nvadekar | we = u and your mama |
03:44.20 | brc_ | chan skype will be available for purchase at some point in the future |
03:44.35 | eipi | near? |
03:44.39 | nvadekar | cool |
03:44.39 | brc_ | I can't comment on the exact price, but we expect it to be around .5 |
03:44.51 | letherglov | purchase?! |
03:44.57 | eipi | 50 cents? |
03:45.02 | brc_ | M |
03:45.13 | letherglov | 1/2 a million?! |
03:45.19 | nvadekar | 500,000 pesos! cool |
03:45.23 | tzafrir_laptop | rubbals? |
03:45.24 | brc_ | USD |
03:45.31 | *** join/#asterisk figfig (~figfig@adsl-66-218-45-62.dslextreme.com) |
03:45.37 | eipi | booohh |
03:45.40 | eipi | ;) |
03:45.43 | DaLion | darn |
03:45.46 | brc_ | delivery would be about 6 months after purchase |
03:45.52 | letherglov | who's selling it? |
03:46.02 | eipi | it comes with fries? |
03:46.10 | eipi | big coke? |
03:46.14 | brc_ | yes, a super size order of frys is included |
03:46.19 | file[laptop] | BKW TOUCHED ME IN THE NAUGHTY PLACE |
03:46.20 | eipi | good |
03:46.22 | Nugget | sure, but they'll be those nasy new Burger King crunchy fries. |
03:46.23 | brc_ | you'll have to BYOC |
03:46.25 | letherglov | from digium? |
03:46.27 | Nugget | nasty, even. |
03:46.27 | letherglov | or from skype? |
03:46.40 | bkw_ | file NO I DID NOT |
03:46.46 | bkw_ | atleast "NOT YET" |
03:46.52 | file[laptop] | HA |
03:47.01 | file[laptop] | you're going to engulf my muffin! |
03:47.12 | brc_ | file, want a alpha copy of chan_skype? |
03:47.16 | file[laptop] | brc_: not really |
03:47.17 | *** join/#asterisk hellop (~hellop@cpe-69-75-47-251.hawaii.rr.com) |
03:47.19 | brc_ | it's 1337 kewl dood |
03:47.24 | hellop | hello |
03:47.26 | DaLion | lol |
03:47.28 | letherglov | hahah |
03:47.30 | brc_ | bkw_, LOOK! |
03:47.30 | letherglov | who wants skype? |
03:47.33 | letherglov | it's nasty ilbc |
03:47.36 | letherglov | yickeee |
03:47.36 | anthm | doesn't digium have a deal with skype to make a hardware dsp fluxx capcitor? |
03:47.38 | brc_ | bkw_, somebody said hello, and NOBODY SAID HELLO BACK! |
03:47.48 | DaLion | lether sure |
03:47.49 | netsurfer | Qwell - u here? |
03:47.53 | figfig | I am trying to set up asterisk to forward calls to my main number simultaniously to my softphone, land line, and cell phone. Despite the support for this in the Dial command, in reality I don't think it can be done because 1). The cell phone picks up right away and goes to voicemail if the cell phone is off, and 2). caller id is not set properly (Broadvoice doesn't let me change the caller id). Does anyone know a way to solve these pr |
03:47.53 | figfig | oblems? |
03:47.54 | letherglov | capacitors don't have flux |
03:48.02 | eipi | isnt the protocol what we would like, its possibility to communicate with skype users |
03:48.03 | brc_ | anthm, yes I believe so, but I think it's going to use an inverse tackyon pulse |
03:48.11 | brc_ | s/tackyon/tachyon |
03:48.23 | hellop | I have kernetl 2.6.3-7mdk but the kernel sources from urpmi is 2.6.3-25mdk do I have to recomplie before installing asterisk? |
03:48.41 | letherglov | well, why is the skype group a bunch of non-interoperable cocksuckers? |
03:48.55 | brc_ | bkw_, so what exactly is the point of smacking people who don't say hello, if when people do say hello they are ignored? |
03:48.59 | hellop | please bare with my bad typing.. |
03:49.06 | letherglov | hello |
03:49.10 | hellop | hi |
03:49.13 | anthm | those stupid shield harmonics were the tough part to crack |
03:49.18 | nvadekar | brc_, do you still limit yourself to 2.4 kernel, or are you comfortable with asterisk on 2.6 now? |
03:49.23 | letherglov | how are you? |
03:49.26 | brc_ | 2.6 works well for the most part |
03:49.48 | letherglov | 2.6 works nicely afaik |
03:49.55 | letherglov | except that whole devfs udev thing with zaptel |
03:49.56 | anthm | asterisk 5.1 runs like a charm on 128bit greenhat linux |
03:49.58 | letherglov | that's just irritating |
03:49.59 | hellop | So, whatcha think? Recompile first? |
03:50.16 | brc_ | hellop, are you going to use zaptel (digium)( hardware? |
03:50.20 | brc_ | if not, then no |
03:50.22 | letherglov | only problem with it |
03:50.23 | letherglov | that I have |
03:50.29 | hellop | brc, yes X100P |
03:50.29 | letherglov | is that I'm running asterisk on ppc |
03:50.36 | brc_ | hellop, then yes |
03:50.38 | letherglov | and my 3ware card won't work under ppc |
03:50.39 | letherglov | :-( |
03:50.40 | CoaxD | why the hell is it that every single time my hard drive clatters, i hear a jitter in my zaptel fxs? |
03:50.43 | letherglov | I love 3ware |
03:50.52 | letherglov | so I called them |
03:50.55 | letherglov | and I'm bringing them a mac |
03:50.56 | brc_ | CoaxD, tdm? |
03:50.57 | letherglov | :-) |
03:50.58 | figfig | So, by the silence can I assume that asterisk can not forward the way I am setting it up to? |
03:51.00 | figfig | do? |
03:51.04 | hellop | ok, tkankyou |
03:51.08 | nvadekar | do zaptel's work find on ppc and 2.6? I am thinking of openpower for asterisk, perhaps a 510? |
03:51.09 | netsurfer | warning guys.. DO NOT buy VOIP hardware from seller "nkans" on ebay - very uncooperative seller |
03:51.11 | CoaxD | brc: Ya |
03:51.23 | brc_ | nvadekar, I no clue |
03:51.25 | CoaxD | brc: I think its RF interference somehow, but i dunno.. |
03:51.41 | CoaxD | brc: Could also be ground interference |
03:51.42 | brc_ | CoaxD, contact digium...yeah there's a known problem with unshielded led traces |
03:51.50 | letherglov | oh |
03:51.56 | letherglov | better pour a bottle of lead on it |
03:51.57 | brc_ | also, try grounding the box to earth ground |
03:51.58 | letherglov | that'll help |
03:52.01 | letherglov | non-conductive lead |
03:52.12 | CoaxD | brc: When was the issue uncovered? Has it been fixed or someshit? |
03:52.22 | brc_ | apparently the fix is...in the works...... |
03:52.25 | letherglov | I had am radio on my adtran...I'm tired of interference |
03:52.25 | CoaxD | brc: ahhhh. hehe |
03:52.26 | anthm | shh there is a little known secret you can pop the resistors off your motherboard and convert the onbord nic to a t1 interface |
03:52.35 | CoaxD | brc: it might also be x100p.. |
03:52.45 | brc_ | I believe they figured out the static discharge issue a few weeks ago |
03:52.45 | CoaxD | brc: (There's one of those in the mix, too, and the remote caller can hear it) |
03:53.07 | CoaxD | brc: All these little "problems" are gonna yield a hell of an expense for digium to replcae this shit |
03:53.13 | brc_ | yep |
03:53.15 | Mother__ | CoaxD: how did you get around the hangup problem then? |
03:53.34 | CoaxD | Mother: Set an AbsoluteTimeout() after it is clear the user is headed into the voicemail system to record |
03:53.41 | brc_ | CoaxD, I have a tdm40b that would have....*WEIRD* problems about twice a week...only way to fix it was to unload and reload the modules |
03:53.44 | CoaxD | Mother: Regardless of busy or NOT, they're gonna get hung up on :) |
03:53.50 | Mother__ | CoaxD: thanks :D |
03:53.51 | CoaxD | brc: hmmm weird! |
03:53.52 | brc_ | and I'd get an "Ouch....blahblah" message in the kern log |
03:53.52 | Mother__ | lol |
03:53.54 | CoaxD | Mother Sure! |
03:54.09 | DaLion | whanyone here use a pyro external lappydrive / |
03:54.10 | DaLion | ? |
03:54.11 | brc_ | anyhow, good luck |
03:54.15 | CoaxD | brc: Dude... i think i remember tht.. did they fix that? or? |
03:54.25 | brc_ | CoaxD, did you have that problem too? |
03:54.28 | CoaxD | brc: No |
03:54.31 | CoaxD | brc: Never |
03:54.37 | CoaxD | brc: The card's been stable as all getout for me |
03:54.40 | brc_ | I've chatted with a few people about it so I didn't remember |
03:56.19 | hellop | I never know what to do on a kernel recompile. How can I "optomize" it? |
03:56.27 | *** part/#asterisk NatRH (~Nat@dargo.trilug.org) |
03:56.35 | hellop | I've tried removing stuff that looked useless, which just broke stuff. |
03:56.46 | hellop | Should I change anything? |
03:56.58 | netsurfer | hellop - lol me too |
03:57.20 | brc_ | hellop, might try in ##linux |
03:57.31 | DaLion | friggin shit |
03:57.34 | brc_ | <PROTECTED> |
03:57.37 | DaLion | or is it chicken shit |
03:57.51 | *** join/#asterisk Juggie (agony@CPE00c049d9f271-CM014270110981.cpe.net.cable.rogers.com) |
03:58.22 | hellop | k |
04:00.12 | *** join/#asterisk bjohnson_ (~bjohnson@ip226-181.tor.istop.com) |
04:01.04 | *** join/#asterisk jdims (~sleepbsd@24-119-121-6.cpe.cableone.net) |
04:05.08 | anthm | hey bkw , didn't you say you can start asterisk -g729unlock |
04:05.26 | anthm | oh fuck that was meant to be private =0 |
04:05.57 | *** join/#asterisk leo360 (Leo@Ottawa-HSE-ppp4058236.sympatico.ca) |
04:06.20 | bkw_ | <PROTECTED> |
04:06.24 | bkw_ | nobody should have know about that |
04:06.48 | jets | oh shit |
04:07.34 | tzanger | isn't it in the source already |
04:08.46 | Nugget | heh |
04:10.25 | nvadekar | weee |
04:10.56 | loud | only 253 people saw that. |
04:11.15 | letherglov | and 252 care, since I use the IPP version :-P |
04:12.36 | tzanger | I'll give g729 a shot somtiem |
04:12.43 | tzanger | maybe after I put codec limits in iax |
04:12.58 | letherglov | eh? |
04:12.59 | letherglov | there arn't? |
04:13.04 | tzanger | i.e. if I only have 8 g729 licenses and I get a 9th call, GRACEFULLY fall back to gsm or something |
04:13.12 | letherglov | oh, gotcha |
04:13.13 | letherglov | tha'd be smart |
04:13.21 | letherglov | degrade to the next tier license |
04:13.25 | tzanger | yup |
04:13.46 | tzanger | I'm working with the steves on the new jitter buffer though |
04:13.53 | jets | uhm why run g729 over gsm |
04:13.57 | tzanger | except nobody seems willing to help me thorugh the bridge optimizations in iax2 |
04:14.04 | letherglov | hmm |
04:14.06 | tzanger | jets: better MOS, better PLC |
04:14.09 | tzanger | lower bandwidth |
04:14.10 | letherglov | I do a lot of bridging with my iaxy |
04:14.15 | letherglov | since it can't compress anything |
04:14.21 | tzanger | letherglov: no native bridging |
04:14.31 | letherglov | oh, gotcha |
04:14.39 | letherglov | to get it to release the call once you get the route? |
04:14.54 | letherglov | a->b->c = a->c |
04:15.21 | tzanger | no that's transfer |
04:15.42 | tzanger | a->b b->c ==> a->b->c |
04:15.45 | tzanger | that's a bridge :-) |
04:15.53 | letherglov | hehe |
04:15.53 | letherglov | ok |
04:15.58 | letherglov | but has nothing to do with tdm transfer |
04:16.06 | letherglov | since it doesn't make it to the iax level, afaik? |
04:16.07 | tzanger | exactly |
04:17.40 | jets | i can't get the unlock to work any way *shrug* |
04:17.41 | jets | heh |
04:17.45 | tzanger | but like I said I was talking about bridging |
04:17.53 | tzanger | not transfers |
04:17.56 | letherglov | right |
04:18.04 | tzanger | and why iax2 bridge optimization totally shitcans timestamps |
04:18.06 | letherglov | so you've got jitter when you transfer? |
04:18.11 | tzanger | NO |
04:18.18 | letherglov | ok, that's good :-P |
04:18.20 | letherglov | oh |
04:18.21 | tzanger | the word 'transfer' does not exist in my vocabulary |
04:18.27 | letherglov | it's eating the timestamp |
04:18.28 | letherglov | sorry |
04:18.29 | letherglov | bridge |
04:18.29 | letherglov | :-P |
04:18.35 | letherglov | s/transfer/brige/gm |
04:18.35 | tzanger | it's weird |
04:18.44 | letherglov | that's a multiline regex...so go back and fix 'em all :-P |
04:18.47 | tzanger | A timestamps are 20 40 60 80 100 120 140 160... |
04:19.02 | tzanger | B recodes the timestamps as 20 40 41 42 43 44 140 16... |
04:19.04 | tzanger | er 160 |
04:19.17 | letherglov | looks like someone's adding i to the after 40 |
04:19.18 | letherglov | :-P |
04:19.18 | eipi | it works!!! |
04:19.19 | eipi | *CLI> show g729 |
04:19.23 | letherglov | in a for loop? |
04:19.25 | eipi | <PROTECTED> |
04:19.48 | letherglov | you lie eipi |
04:19.54 | eipi | just kidding.... |
04:19.55 | eipi | :D |
04:20.15 | *** join/#asterisk hermie (~nick@24.236.167.53.bay.mi.chartermi.net) |
04:20.21 | tzanger | letherglov: no |
04:20.25 | tzanger | it's even listed as a FIXME? |
04:20.31 | letherglov | oh |
04:20.31 | letherglov | hmm |
04:21.16 | tzanger | basically the code in calc_fakestamp |
04:21.53 | nvadekar | is there any benefit to carrier grade linux while running asterisk |
04:21.55 | tzanger | takes one side of the bridge's timestamps and tries to make them match up to the other side's |
04:22.01 | tzanger | 'carrier grade linux' ?? |
04:22.37 | nvadekar | osdl has a cgl certification 1, 1.1, 2 and now just version 3, all the big telcos are sponsoring it |
04:22.41 | tzanger | and if the timestamp it ocmes up with is earlier than the last one it will add 1 to the last one |
04:22.48 | nvadekar | it is for realtime systems |
04:22.48 | tzanger | nvadekar: meh |
04:25.36 | leo360 | hello, i just joined today, need help config iaxy |
04:26.08 | tzanger | <PROTECTED> |
04:26.09 | nvadekar | tzanger, sorry for ignorance, but what is meh short for? |
04:26.12 | tzanger | <PROTECTED> |
04:26.14 | tzanger | <PROTECTED> |
04:26.17 | tzanger | <PROTECTED> |
04:26.32 | letherglov | looks like you just found your plus one |
04:26.32 | tzanger | that's what is cauisng the problem... or rahter it's the fact that the calculated fake timestamp is coming out odd |
04:26.32 | letherglov | :-P |
04:27.07 | tzanger | nvadekar: it means I'm nonplussed |
04:27.35 | tzanger | realtime patches and O(n) schedulers and so on don't make a whit of difference if all the system's doing is running asterisk |
04:27.44 | tzanger | now if you're trying ot push a system then yes |
04:28.01 | tzanger | but you can very easily get * working perfectly well with most favourite distros |
04:28.20 | nvadekar | ok, thanks |
04:31.31 | *** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net) |
04:32.29 | bjohnson_ | can someone tell me how an incoming call from fwd comes in? I have my own account # setup in my extensions.conf, but my * is refusing an incoming call from another FWD account |
04:32.54 | tzanger | bjohnson_: you need an appropriate type=user entry in iax or sip.conf (i forget which fwd uses) |
04:33.07 | tzanger | bjohnson_: and you need to dump the incoming call into a specified context= |
04:33.24 | bjohnson_ | I use iax2 for fwd and i can call fwd extensions |
04:33.37 | tzanger | bjohnson_: you calling fwd is a type=peer entry |
04:33.40 | tzanger | you need a type=user |
04:33.47 | tzanger | to accept calls |
04:33.49 | bjohnson_ | I have [iaxfwd-in] that uses the rsa in * source for fwd |
04:34.57 | bjohnson_ | tzanger: I have 2 fwd entries in iax.conf .. one for peer and one for user |
04:35.24 | bjohnson_ | I wonder where I got the host=iax2.fwdnet.net line .. maybe I should check that |
04:35.54 | tzanger | bjohnson_: is that where the incoming call is coming from? |
04:36.02 | bjohnson_ | yes |
04:36.06 | bjohnson_ | another fwd account |
04:36.19 | tzanger | no |
04:36.25 | tzanger | is that the HOST that the call is coming from |
04:36.28 | tzanger | use tcpdump and verify |
04:36.36 | tzanger | or even just look at the console |
04:36.41 | tzanger | is the call coming from iax2.fwdnet.net |
04:37.00 | bjohnson_ | Rejected connect attempt from 65.39.205.121 .. dig iax2.fwdnet.net shows 65.39.205.121 |
04:37.12 | tzanger | ok |
04:37.22 | *** join/#asterisk Nethab (~Anon5864@adsl-67-113-141-170.dsl.sntc01.pacbell.net) |
04:37.24 | tzanger | my next guess is that you have a secret in there |
04:37.27 | tzanger | get rid of it and reload |
04:38.23 | Nethab | "i've got a secret", "I'll never tell..." |
04:38.27 | bjohnson_ | nope. type=user, auth=rsa, inkeys=freeworlddialup, disallow=all, allow=ulaw, host=iax2.fwdnet.net, context=incoming-public-voip, accountcode=voip_fwd_in |
04:38.49 | tzanger | get rid of all the auth shit |
04:38.51 | tzanger | get it working |
04:39.10 | tzanger | auth=, inkeys=, get rid of 'em and reload |
04:39.35 | tzanger | you might also want ot get rid of the disallow/allow and only say allow=all |
04:39.41 | bjohnson_ | still rejected |
04:39.45 | tzanger | and what was the reason for the reject |
04:39.56 | bjohnson_ | doesn't say at verbose =3 |
04:40.01 | tzanger | you are reloading right? |
04:40.10 | bjohnson_ | of course |
04:40.24 | tzanger | show dialplan incoming-public-voip |
04:40.42 | tzanger | make sure there's both an s exten and a _. exten |
04:40.50 | tzanger | I don't know what fwd's trying to hit on your side |
04:41.03 | tzanger | but if you have both of those, it should get through |
04:41.06 | bjohnson_ | I have the s .. but no _. |
04:41.15 | bjohnson_ | I guess I can stick it in for now |
04:41.29 | tzanger | I'd say |
04:41.30 | bjohnson_ | pretty messy habit though |
04:41.48 | tzanger | exten => _.,1,NoHup(Exten is ${EXTEN}), _.,2,Hangup |
04:41.52 | tzanger | bjohnson_: this is just ofr testing |
04:42.40 | bjohnson_ | I guess a wait will be enough to test |
04:42.58 | tzanger | I say to echo the ${EXTEN} so you can see what they're hitting you with |
04:43.14 | Nethab | you mean NoOp? |
04:43.21 | tzanger | ye |
04:43.24 | tzanger | haha |
04:43.25 | tzanger | NoHUP |
04:43.26 | tzanger | hahaha |
04:43.32 | Nethab | kill -HUP 1 |
04:44.13 | tzanger | Nethab: nah |
04:44.16 | tzanger | kill -9 1 :-) |
04:44.27 | bjohnson_ | still rejected |
04:44.38 | tzanger | hmm |
04:44.41 | Nethab | it's saying unauthenticated? |
04:44.43 | bjohnson_ | I even have a guest account that should be kicking in |
04:44.52 | bjohnson_ | Feb 12 23:44:02 NOTICE[28773]: chan_iax2.c:5405 socket_read: Rejected connect attempt from 65.39.205.121 |
04:44.56 | tzanger | bjohnson_: what's the [] line you're trying ot hit from |
04:45.07 | tzanger | i.e in iax.conf what is the name in [] for the type=user |
04:45.39 | Nethab | well line 5405 will have the answer |
04:46.03 | bjohnson_ | tzanger: i don't understand the question |
04:46.09 | JerJer | [bob] |
04:46.10 | JerJer | type=user |
04:46.13 | tzanger | yes |
04:46.14 | JerJer | secret=bobsecret |
04:46.17 | tzanger | I'm looking for the [] ine |
04:46.19 | tzanger | what is it |
04:46.20 | *** join/#asterisk paulhuynh (~paul@pcp05051905pcs.univde01.de.comcast.net) |
04:46.21 | JerJer | context=blah |
04:46.41 | bjohnson_ | tzanger: I'm getting a special char from you |
04:46.55 | bjohnson_ | in ___ for the type=user |
04:47.01 | tzanger | ?? |
04:47.04 | tzanger | in iax.conf |
04:47.06 | tzanger | [blah] |
04:47.09 | tzanger | what is the "blah" |
04:47.15 | paulhuynh | i need help my cisco 7940 keep saiding configuring ip |
04:47.19 | paulhuynh | y? |
04:47.33 | bjohnson_ | I'm getting a rectangle from you in this xchat |
04:47.40 | tzanger | then get a real irc client :-) |
04:47.45 | paulhuynh | it show that it have an i from dhcp of 192.168.1.105 |
04:47.50 | tzanger | it's trying ot be fancy |
04:48.01 | tzanger | I am typing left square bracket, right square bracket |
04:48.05 | paulhuynh | but i can't ping it |
04:48.09 | bjohnson_ | ohhh |
04:48.10 | tzanger | what is the entry name |
04:48.29 | bjohnson_ | [iaxfwd-in] |
04:48.38 | paulhuynh | how do i get tftp to the damn phone to upload new firmware |
04:48.41 | tzanger | is that what fwd is trying to authenticate as? |
04:48.49 | bjohnson_ | the type=peer is [iaxfwd] |
04:48.50 | tzanger | if it isn't, make the name what it's supposed to be |
04:48.52 | brc_ | paulhuynh, the phone has to request the firmware |
04:49.05 | paulhuynh | oh ok |
04:49.07 | brc_ | paulhuynh, you'll have to go read the wiki and search google |
04:49.21 | bjohnson_ | tzanger: that matters? I thought id would use the rsa stuff? |
04:49.30 | tzanger | it does for nonrsa |
04:49.30 | brc_ | star, six, settings will unlock the phone so you can set it up |
04:49.32 | paulhuynh | can i use the console port to fix my problem |
04:49.36 | nvadekar | does anyone here know about a polycom xml file configration program being made available? |
04:49.38 | tzanger | I don't know anything about rsa |
04:49.38 | brc_ | er |
04:49.39 | brc_ | no |
04:49.41 | tzanger | so let's go back to basics |
04:49.41 | brc_ | that's wrong |
04:49.47 | *** join/#asterisk lilneon (~tj_r3@cuscon10482.tstt.net.tt) |
04:49.49 | brc_ | it's star star pound |
04:49.52 | lilneon | hey good night everyone |
04:49.59 | paulhuynh | i need help my cisco 7940 keep saiding configuring ip |
04:50.10 | tzanger | if you're doing rsa do you not need to do some kind of loadkeys command when you start up *? |
04:50.26 | Mother__ | paulhuynh: do you have a dhcp server there? |
04:50.26 | bjohnson_ | tzanger: that was it |
04:50.34 | nvadekar | inkeys=freeworlddialup |
04:50.37 | paulhuynh | i can now ping the phone but very intermite |
04:50.43 | paulhuynh | yes |
04:50.54 | paulhuynh | i use linksys wrt54g |
04:50.56 | bjohnson_ | I changed the context name to iaxfwd and uncommented the rsa and inkeys line again .. now get the incoming call |
04:51.04 | tzanger | ok |
04:51.22 | tzanger | now remove the garbage I made you put in extensions.conf and get rsa working :-) |
04:52.37 | paulhuynh | is my cisco phone broke? |
04:55.05 | bjohnson_ | mmmmmm |
04:55.39 | nvadekar | anyone have or know of a easy way to configure polycom xml files? |
04:55.50 | paulhuynh | hello? |
04:56.04 | paulhuynh | anyone here can help me with cisco 7940 |
04:56.13 | bjohnson_ | <paulhuynh> i can now ping the phone but very intermite |
04:56.27 | bjohnson_ | you have some kind of network problem I think |
04:56.35 | bjohnson_ | maybe the phone |
04:56.52 | bjohnson_ | but then maybe it's just busy trying to connect to a tftp server |
04:57.03 | paulhuynh | what is the console port for? |
04:57.07 | bjohnson_ | try watching the network packets |
04:57.36 | paulhuynh | how do i know what tftp address it try to contact? |
04:57.43 | bjohnson_ | don't know. don't have one. but lots of people here use them .. maybe just not the people here right now |
04:57.44 | paulhuynh | it a use phone |
04:57.54 | bjohnson_ | try watching the network packets |
04:57.57 | paulhuynh | and have preset network infor on it |
04:59.17 | bjohnson_ | do they have a web interface? try nmap on it |
05:01.49 | darkskiez | nvadekar: sure you dont mean sipura configuration thing instead of polycom ? |
05:02.38 | *** join/#asterisk lilneon (~tj_r3@cuscon10482.tstt.net.tt) |
05:02.43 | lilneon | hi again. . |
05:02.45 | bjohnson_ | nvadekar: buch of polycom users here normally .. maybe just not now |
05:03.07 | bjohnson_ | anyone ever tried the fwd "welcome line" ? 55555 |
05:03.31 | nvadekar | bjohnson, yeah, been trying today several times, always says no extention |
05:03.38 | lilneon | hey guys i downloaded httptunnels on my linux box to try to get a sip client to wrk behind a nat.. but when i type hts it tells me no command exists.. anyone here used httptunnel? |
05:03.53 | nvadekar | 612, 613, callback and NOW!!! 1800 are all working for me though |
05:04.04 | bjohnson_ | nvadekar: for me it just rings |
05:04.17 | bjohnson_ | nvadekar: you can try calling me on my new fwd account if you want |
05:04.28 | nvadekar | bj, what is you number? |
05:04.56 | nvadekar | I am using kphone attaches to asterisk to fwd all on my laptop |
05:05.01 | bjohnson_ | 613834 |
05:05.08 | nvadekar | inasec.... |
05:06.45 | nvadekar | says you are unavailable |
05:07.12 | *** join/#asterisk Darwin35 (~Darin@c-24-3-241-22.client.comcast.net) |
05:07.20 | nvadekar | bj, funny, did not hear ringing, but asterisk said it was rining, then it says you are busy |
05:07.20 | bjohnson_ | hmm |
05:07.30 | nvadekar | you can try me, 608004 |
05:08.34 | Darwin35 | ok |
05:10.34 | *** join/#asterisk Rick_Hunter (~rhunter@01-109.008.popsite.net) |
05:16.24 | Mavvie | is it just me or is there no way to reload the voicemail configuration? |
05:18.18 | Mother__ | I'm having a weird problem...two of the PSTN channels on a TDM400 go into the same zap channel |
05:18.50 | Mother__ | i.e. call PSTN that's connected to Zap/2, and Zap/3 triggers, call the PSTN connected to Zap/3, and Zap/3 also triggers |
05:19.54 | letherglov | Mother__, I had a weird issue where I called cisco and got connected to another customer when I was on their support hotline |
05:20.06 | letherglov | we discussed how much cisco sucks |
05:20.08 | Mother__ | hehehe |
05:20.19 | letherglov | hmm |
05:20.28 | letherglov | maybe you've got pins 3-4 connected to zap/3 also? |
05:20.32 | letherglov | and 1-2 on zap-2? |
05:21.17 | Mother__ | no, I have four individual terminators from the telco's switch, which I then wired individually to the RJ11 ports on the TDM |
05:21.29 | letherglov | interesting... |
05:21.40 | Mother__ | checking with a phone it rings as it should |
05:27.18 | *** join/#asterisk quigleymd (~quigleymd@69-174-22-36.chvlva.adelphia.net) |
05:27.27 | Mother__ | grrrr just switched Zap/3 with Zap/4 and it looks cured....still very strange |
05:33.19 | bjohnson_ | Mavvie: i think that is correct |
05:33.55 | nvadekar | asterisk is way cool, fwd seems to work fine, good night all! |
05:38.24 | tzafrir_laptop | anybody used kphone? |
05:39.11 | tzafrir_laptop | any idea why it won't accept calls and practically believes it has failed to register to the server? |
05:39.39 | *** join/#asterisk iMediax (lklk@00045a809589.click-network.com) |
05:39.57 | bjohnson_ | sorry .. linphone and iaxcomm here |
05:40.56 | tzafrir_laptop | linphone works OK for you? what version? from where? |
05:41.46 | bjohnson_ | err .. I'm not at that computer right now |
05:42.21 | bjohnson_ | but I think it was available from apt-get on my fc2 and fc3 systems .. so likely atrpms |
05:44.02 | *** part/#asterisk illek (~mike@ip68-13-238-168.ok.ok.cox.net) |
05:44.33 | tzafrir_laptop | Well, time to freeze my laptop and go to work... |
05:47.05 | *** join/#asterisk quigleymd (~quigleymd@69-174-22-36.chvlva.adelphia.net) |
05:48.11 | Mother__ | nite all |
05:49.42 | Luke-Jr | What is the technical difference between having two identical user and peer entries and having a single friend entry? |
05:53.08 | bjohnson_ | different things .. |
05:53.17 | bjohnson_ | you can set callerid differently |
05:53.20 | bjohnson_ | accountcodes |
05:53.30 | bjohnson_ | authentication |
05:53.35 | bjohnson_ | don't know what else |
05:54.09 | Luke-Jr | so, the only difference is that you can have different config settings for them? |
05:54.10 | bkw_ | sdf |
05:54.26 | Luke-Jr | so if they'd have the same settings, I might as well use friend? |
05:54.37 | Luke-Jr | bjohnson: what doesn't work about it? |
05:54.43 | bjohnson_ | no sound |
05:54.45 | bjohnson_ | hehe |
05:55.23 | bjohnson_ | fedora core 3 .. tried running sound-config and alsaconf .. they sense my sound card .. but no sound |
05:57.37 | *** join/#asterisk jtodd (~jtodd@garthim.fox-den.com) |
05:58.37 | quigleymd | ive got a system-config-soundcard on fc3 and that worked for me |
06:01.28 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
06:04.19 | *** join/#asterisk criptos (~criptos@201.135.127.28) |
06:04.27 | *** part/#asterisk lilneon (~tj_r3@cuscon10482.tstt.net.tt) |
06:04.53 | postel | bjohnson: modprobe the kernel module for your card, or if its not there install it, sound-config might work might not, take the canonical way |
06:05.17 | criptos | why digium is no longer making the x100p / x110p fxo card? |
06:05.22 | criptos | sniff |
06:06.15 | Sedorox | you can find them on ebay for like $20 w/ shipping |
06:06.55 | postel | Sedorox: those are clones (The Tiger-something version) some ppl have terrible echo trouble |
06:07.28 | bjohnson_ | postel: I think the correct one is there but I am not able to remove it .. says it's in use |
06:07.47 | criptos | I have deployed over 20 x100p, what should I do now ? those tiger.. humm... well.. |
06:07.52 | Sedorox | I dunno which one I have.. or my friend has.. mines a intel something I think... but apparently he just started getting echo problems.. but it might be the phone.. so dunno.. otherwise haven't had problems |
06:08.13 | Sedorox | pull 4 at a time and get a new modular card :-p |
06:10.09 | postel | bjohnson: pastebin your lsmod |
06:10.25 | criptos | Mostly of the configurations made are single card configuration.. really using the TDMxxp card, increases the cost a lot |
06:10.37 | Sedorox | yea |
06:10.52 | bjohnson_ | http://pastebin.ca/5744 |
06:10.54 | Mavvie | bjohnson_: that's what I thought. |
06:10.59 | Sedorox | dunno.. I doubt you'll have problems with them |
06:11.16 | *** join/#asterisk harryvv (~noyb@S010600055d210201.vs.shawcable.net) |
06:11.25 | Sedorox | damn.... and ipv6.. nice :-p |
06:13.18 | Luke-Jr | bjohnson: killall artsd? |
06:13.26 | Luke-Jr | esd, too |
06:15.34 | postel | bjohnson: cool, remove everything, modinstall soundcore and ac97_codec and throw a mixer on top of it |
06:16.04 | postel | you threw every single thing against it.. surprise surprise it wont play |
06:17.22 | harryvv | artsd and asterisk work against each other. |
06:17.24 | bjohnson_ | I can't remove anything .. says it's in use |
06:17.42 | harryvv | bj, ive seen this same problem |
06:18.01 | harryvv | trying to record or play a gsm ? |
06:18.27 | bjohnson_ | naw .. just a movie |
06:18.38 | harryvv | okay |
06:18.48 | postel | bjohnson: yeah dude, its in use by the 23 mixers you're running, comment them out from modules.conf or bring down the rc that loads them on boot |
06:18.52 | harryvv | Personally if it was me the box would only run asterisk and thats abou tit. |
06:20.37 | bjohnson_ | I'll give it a try. Don't remember adding mixers .. |
06:20.45 | harryvv | I dont run anything but asterisk,sendmail and qsl on the debian box. All that are required for asterisk. |
06:20.52 | *** join/#asterisk Nukemizer (~Nuke@66.237.85.58.ptr.us.xo.net) |
06:20.59 | harryvv | bj, on the command prompt type alsamixer |
06:21.35 | bjohnson_ | ok |
06:22.23 | bjohnson_ | this is a home box .. so trying to run desktop too on occasion .. like right now |
06:22.28 | harryvv | btw, anyone know how to make musiconhold play when switching from line one to line two on xlite? i have the settings right in both config files. |
06:23.36 | bjohnson_ | well .. I'll restart and give it another try tomorrow |
06:23.41 | bjohnson_ | night |
06:23.41 | harryvv | bj, I have a greeting message stating if there is any emergency or work releated issue to press in extention 100 otherwise leave a message. Well someone did and I answered my cell phone via voip. |
06:23.55 | bjohnson_ | neat |
06:24.00 | bjohnson_ | forwarded out fxo? |
06:24.10 | harryvv | no, to iax.cc :) |
06:24.23 | harryvv | in from zap to asterisk then out to iax.cc |
06:24.33 | bjohnson_ | ahhh |
06:24.38 | bjohnson_ | how was quality? |
06:24.43 | harryvv | very good. |
06:24.52 | bjohnson_ | cool |
06:24.57 | bjohnson_ | good to know |
06:25.15 | harryvv | I was thinking of actually ring both mine and wifes phone at same time. |
06:25.25 | harryvv | in case one is down or out of range. |
06:25.37 | postel | bjohnson: ebay a pIII, throw some sdrams on it and a NIC, put yr FXS/FXO over it and forget all bout it, ssh on it, cvs asterisk, job done, desktop + services is never a good idea |
06:26.19 | harryvv | postal, I do the same. The only thing I have on my asterisk box is a power cable, phone/nick and that it. |
06:26.27 | postel | good man |
06:26.30 | harryvv | nic I mean |
06:26.42 | letherglov | no no no |
06:26.44 | letherglov | a ppc! |
06:27.03 | harryvv | btw, netsurfer made a fail over asterisk system. When one fails the second one takes ove.r |
06:27.27 | postel | safe_asterisk was not good enough? |
06:27.45 | postel | failling process-wise or load-wise? |
06:27.48 | harryvv | I dont know. whats the benifits of that? what if memory/hd/ps fail in the first one? |
06:27.48 | Sedorox | if you loose power? |
06:28.03 | harryvv | Im talking all cases |
06:28.26 | harryvv | processes, errors, hardware and power :) |
06:29.20 | harryvv | btw, how can i make my asterisk go moh when going from line one to line 2 on xlite? |
06:29.37 | harryvv | I know how it works in start priorities |
06:29.47 | harryvv | but thats not what I want. |
06:29.48 | harryvv | ;) |
06:30.53 | harryvv | http://news.zdnet.com/2100-1035_22-5571679.html |
06:31.47 | Sedorox | damn it.. I hate the network here.... |
06:32.11 | *** part/#asterisk leo360 (Leo@Ottawa-HSE-ppp4058236.sympatico.ca) |
06:34.16 | harryvv | why is that |
06:35.32 | Sedorox | my phone doesn't want to register with my * server here... and I think its part of the network. because I had the problem when I first got the phone.. and registered with a remote box.. wouldn't work with DHCP.. worked fine if I hard coded a IP addy in... |
06:36.01 | Sedorox | and they limit stuff.. so untill the CTO gets back to me about opening 5060 and IAX2 ports.. its gonna be like shit calling outside the network |
06:36.12 | Sedorox | 'cause it gets dumped in with the P2P stuff in the PIX |
06:38.01 | harryvv | how many asterisk phones |
06:38.12 | harryvv | that is ip phones interact with asterisk |
06:38.38 | *** part/#asterisk bubulubugoth (~criptos@201.135.127.28) |
06:39.22 | Sedorox | I'm on a college network.. this is the resnet.. its only my phone with the * box sitting right next to me... |
06:39.36 | Sedorox | I also think there is something wrong with that box too.. old POS HP |
06:40.53 | *** join/#asterisk jets (~jetsn@xyharley.dsl.xmission.com) |
06:41.31 | harryvv | i see |
06:41.46 | netsurfer | hi harryvv |
06:42.05 | harryvv | hi |
06:42.07 | harryvv | :) |
06:42.42 | Luke-Jr | Is it possible to wait one second, but if the person starts dialing, let them finish? |
06:42.48 | harryvv | netsurfer, any idea how to play moh on line one when switching to line two on xlite? |
06:43.10 | netsurfer | park the call ? |
06:43.17 | netsurfer | ah.. u cant.. on xlite |
06:43.21 | harryvv | mm is that whats it refered to do? |
06:43.26 | harryvv | ohh |
06:43.32 | harryvv | on clite pro you can? |
06:43.34 | harryvv | xlite |
06:43.46 | netsurfer | sure... when u switch from 1 to 2 it should play music auto |
06:43.49 | netsurfer | mine does |
06:44.06 | netsurfer | u got moh setup properly ? |
06:45.01 | harryvv | yea |
06:45.28 | netsurfer | if i accept a call on line #1 and then click line #2 the music starts straight off |
06:45.35 | harryvv | thats the idea. |
06:47.24 | netsurfer | default => mp3:/var/lib/asterisk/mohmp3 |
06:47.29 | netsurfer | u got that line uncommented ? |
06:47.41 | netsurfer | in musiconhold.conf |
06:47.44 | harryvv | it should be. |
06:48.16 | harryvv | <PROTECTED> |
06:48.16 | harryvv | ; |
06:48.16 | harryvv | ; Music on hold class definitions |
06:48.16 | harryvv | ; |
06:48.16 | harryvv | [classes] |
06:48.16 | harryvv | default => quietmp3:/var/lib/asterisk/mohmp3 |
06:48.26 | netsurfer | #flood ;) |
06:48.27 | *** join/#asterisk goatmilk (~travis@user-69-73-1-138.knology.net) |
06:48.31 | harryvv | :) |
06:48.55 | netsurfer | remove the 'quiet' from the start of that line |
06:49.15 | netsurfer | plays it so quiet u would think there was no moh |
06:49.18 | harryvv | mp3 in its self a binary? |
06:49.32 | netsurfer | default => mp3:/var/lib/asterisk/mohmp3 |
06:51.02 | netsurfer | lmao i had more luck with failover than with windows file syncronisation |
06:51.14 | Nukemizer | has anyone gotten a Digium T1 card to work with a PBX ? I get getting D channel error message and I can not tell if the PBX is the problem or the asterisk |
06:51.50 | wildcard0 | Nukemizer, i have many of them |
06:51.56 | harryvv | netsurfer what do you have for silence settings? |
06:51.59 | harryvv | in xlite |
06:52.03 | wildcard0 | be more specific about the error? |
06:52.07 | netsurfer | send silence = on |
06:52.13 | harryvv | okay |
06:52.34 | netsurfer | actually, that can do it. |
06:52.36 | harryvv | its acually no or yes |
06:52.41 | netsurfer | yes |
06:52.47 | netsurfer | send silence = yes |
06:53.12 | netsurfer | or rather, transmit silence |
06:53.52 | Nukemizer | I just can not seem to tell if the PBX just does not like the Asterisk or what |
06:54.05 | Nukemizer | it is a new TE110P |
06:54.11 | Juggie | Nukemizer, what kind of pbx? |
06:54.23 | harryvv | going from one answered line on the called xlite to another line plays no sound on the other what sound binary does it run? |
06:54.27 | Nukemizer | Toshiba |
06:54.51 | Nukemizer | <PROTECTED> |
06:54.54 | harryvv | mpg123? |
06:54.55 | Juggie | you have to set all the rught frameing settings and such |
06:55.17 | Nukemizer | that is my Asterisk Error -- Toshiba has its own Dmessage error |
06:55.50 | Nukemizer | <U1>16;19'533 Rx:[SAPI]00 C [TEI]000 [FRAME]RR P [N(R)]000 |
06:55.50 | Nukemizer | <U1>16;19'543 Tx :[SAPI]00 R [TEI]000 [FRAME]RR F [N(R)]000 |
06:56.14 | netsurfer | harryvv - mpg123 |
06:56.14 | Nukemizer | that is my D cahnnel form the Toshiba - and it looks good for being idel thats the norm |
06:56.20 | Juggie | Nukemizer, check your settings in /etc/zaptel.conf |
06:56.33 | netsurfer | harryvv - go to /usr/src/asterisk and type make mpg123 |
06:56.39 | harryvv | that was probebly the problem it was not installed its installed now. |
06:56.45 | netsurfer | doh |
06:56.47 | netsurfer | :P |
06:56.53 | Juggie | be sure your codeing & frameing id configured |
06:56.59 | Juggie | properly |
06:57.03 | Juggie | *os |
06:57.04 | Nukemizer | span=1,1,0,esf,b8zs |
06:57.04 | Nukemizer | loadzone=us |
06:57.04 | Nukemizer | defaultzone=us |
06:57.04 | Nukemizer | bchan=1-8 |
06:57.04 | Nukemizer | dchan=24 |
06:57.07 | Juggie | grrr |
06:57.07 | eipi | who can recommend a good CD for music on hold? |
06:57.10 | Juggie | is |
06:57.20 | netsurfer | Nukemizer stop flooding |
06:57.29 | Nukemizer | that is all there is i am hoping to have NI2 and 8 Channels PRI |
06:57.31 | netsurfer | use pastebin.ca or something |
06:57.42 | Nukemizer | sorry |
06:57.49 | Juggie | Nukemizer, if you are only using 8 chans shoudnt your d chan be like 9? |
06:58.02 | Juggie | if its a partial |
06:58.12 | Nukemizer | not neccessarily - is that what Asterisk likes ? |
06:58.22 | Nukemizer | <PROTECTED> |
06:58.54 | *** join/#asterisk ta[i]nted (~tainted@65-60-70-242-cust.telepacific.net) |
06:59.06 | Juggie | asterisk will work with what ever the tishiba is providing |
06:59.25 | Juggie | can you dial a did and receive calls? |
06:59.42 | Juggie | a DID for the pri that is |
06:59.48 | harryvv | netsurfer it works |
06:59.50 | ta[i]nted | quick question |
06:59.51 | harryvv | thanks very much |
06:59.52 | harryvv | :) |
06:59.58 | ta[i]nted | how do i checkout a specific version of asterisk from CVS? |
06:59.59 | netsurfer | harryvv - thats good :) |
07:00.00 | Nukemizer | is the re a way t have D channel messaging with Asterisk ? so i can tell if astersik is doing ? |
07:00.11 | Juggie | (excuse the typos i broke my collarbone, i am working one handed) |
07:00.34 | netsurfer | Juggie - wow.. nasty |
07:00.34 | harryvv | :) |
07:01.05 | Juggie | yes skiing kid cut me off :) |
07:01.22 | harryvv | nice to have virtual outgoing lines ;) netserver my next project it to ring several iax to pstn lines and have a chat meeting with familly or who knows what else. |
07:02.17 | `Sauron | Juggie: What'd you do that for? |
07:02.45 | Sedorox | night |
07:03.31 | Juggie | `Sauron i love skiing i was passing a kid 2nd run @ 9am crowded and all of a sudden he cut left i tried to avoid... and i put my right ski between his legs which slingshotted me |
07:03.43 | Juggie | and planted me shoulder first |
07:04.38 | *** join/#asterisk sjaak538 (~sjaaknabu@d5c53145.dsl.concepts.nl) |
07:04.40 | *** join/#asterisk BoRiS (~boris@S01060040ca1e5b54.wp.shawcable.net) |
07:04.59 | harryvv | well, im going to bed |
07:05.00 | harryvv | :) |
07:05.10 | Juggie | it didnt hurt tho |
07:05.12 | netsurfer | u had the right idea gettin the ski between his legs.. unfortunate about the outcome ;) |
07:05.19 | netsurfer | nite harry |
07:05.28 | harryvv | long day. wife is asleap purpindicular on bed she is beat :) |
07:05.31 | Juggie | yah sucks |
07:05.32 | harryvv | thanks net |
07:05.32 | harryvv | ;) |
07:05.47 | *** join/#asterisk BoRiS (~boris@S01060040ca1e5b54.wp.shawcable.net) |
07:05.50 | netsurfer | well, tis 7am guess i should get to bed too |
07:05.58 | netsurfer | g'nite dudes |
07:06.18 | jets | boris :P |
07:06.53 | BoRiS | hi jets! |
07:06.57 | BoRiS | wasssup? |
07:07.12 | jets | not a whole lot |
07:08.59 | BoRiS | Any luck finding the ChanSpy author? |
07:10.16 | jets | ya i e-mailed bkw he was busy tonight |
07:10.21 | jets | *sigh* |
07:10.35 | jets | if i knew enough c i would rewrite it or write it myself but i suck at c |
07:10.56 | BoRiS | :( |
07:12.57 | jets | O well :) |
07:13.37 | *** join/#asterisk Tili (~Tili@202-133-65-132-dialup.sat.net.pk) |
07:14.34 | brc_ | BoRiS, anthm |
07:16.15 | BoRiS | hey brc_ :) |
07:16.20 | brc_ | . |
07:21.20 | *** join/#asterisk Fanguin (~Fanguin@p50818601.dip0.t-ipconnect.de) |
07:25.11 | Nukemizer | I have a questions about framing and coding.. does the Astersik want to make a PRI have B8ZS ESF, like a non PRI T1 ? |
07:25.27 | drumkilla | it is configurable |
07:26.46 | Nukemizer | I ask ecause when I turn up PRI's on the PBX side i never set the Framing and Coding like that . Only on E&M wink T1 circuits. |
07:27.24 | Nukemizer | just trying to figure out why i can not get my B channels up |
07:28.15 | Nukemizer | I know i must be getting close, becuase Asterisk keep reseting th B channels every 20 seconds |
07:28.27 | letherglov | it's hard to have phone sex while it's doing that |
07:29.19 | letherglov | well |
07:29.24 | letherglov | in my understanding |
07:29.30 | letherglov | the t-1 itself is b8zs esf framed |
07:29.40 | letherglov | but you're running 23 clear-channel (non-robbed bit) channels |
07:29.47 | letherglov | and a 24th clear-channel d-channel |
07:29.57 | letherglov | you really only need that robbed-bit for signalling info |
07:30.09 | letherglov | and if you're running it over the d-channel, then get rid of it ;-) |
07:31.42 | Nukemizer | hum.. |
07:31.45 | Nukemizer | checking |
07:33.10 | *** join/#asterisk X-Gen (~x-gen@rrba-146-67-74.telkomadsl.co.za) |
07:38.34 | *** join/#asterisk beto75 (~ha@201.128.177.84) |
07:38.39 | beto75 | hello guys |
07:39.10 | beto75 | excuse me can someone say me why , at the few seconds in a meetme asterisk crashes? |
07:39.36 | quigleymd | Nukemizer: just an off shot.. theres a bit in the span parameter of your zaptel.conf that tells the card to either attempt to set its own timing or try to pull timing from the CO, try toggling that for kicks... |
07:40.13 | beto75 | hrrmm seems to crash when mpg123 enters to the picture |
07:40.22 | letherglov | which version of mpg123 is it? |
07:40.27 | letherglov | are you on a 64-bit platform? |
07:40.47 | beto75 | mpg123-0.59r.tar.gz |
07:40.58 | letherglov | hmm |
07:41.00 | letherglov | that's correct |
07:41.06 | letherglov | from make mpg123 in the asterisk dir? |
07:41.08 | beto75 | <letherglov> no is 32 plat |
07:41.14 | beto75 | but |
07:41.15 | Nukemizer | quigeymd: I will ook for that ia try. Thank you so much for your help |
07:41.22 | beto75 | is a suse linux quite old |
07:41.26 | beto75 | 8.1 |
07:41.39 | letherglov | should be fine |
07:41.45 | letherglov | does mpg123 work independently of asterisk? |
07:41.48 | letherglov | can you play an mp3 |
07:41.54 | beto75 | what you mean here : from make mpg123 in the asterisk dir? |
07:41.54 | letherglov | and output it to a wav file or stdout or something? |
07:42.03 | letherglov | did you get asterisk from a tarball? |
07:42.04 | letherglov | and install it? |
07:42.10 | tzafrir | 8.1 is not officially supported (security updates) anymore |
07:42.21 | beto75 | well , I enter the meetme , I hear teh welcome msg, then 1 second of the MOH |
07:42.29 | tzafrir | letherglov, yes, it can. |
07:42.48 | letherglov | huh? |
07:42.50 | letherglov | no no |
07:42.52 | letherglov | I know it can |
07:42.56 | letherglov | but on his system can it |
07:43.00 | letherglov | e.g. does mpg123 run at all |
07:43.07 | tzafrir | sorry |
07:43.10 | beto75 | yes it runs |
07:43.10 | letherglov | obviously asterisk is another layer on top of mpg123 |
07:43.12 | quigleymd | Nukemizer: np, hope it does the trick :) |
07:43.18 | modulus_ | jbot weather klax? |
07:43.22 | letherglov | ok...hmm |
07:43.35 | letherglov | jbot weather kpao? |
07:43.45 | letherglov | hmm |
07:43.49 | letherglov | mostly cloudy eh |
07:43.54 | letherglov | i guess it's not mostly sunny at night ;-) |
07:44.10 | beto75 | the strange is that why I hear 1 second music |
07:44.14 | letherglov | right |
07:44.17 | letherglov | then it takes a crap |
07:44.24 | letherglov | you've got a zaptel device on the machine for timing? |
07:46.15 | beto75 | letherglow : MOH is working perfectly ,, just meetme has the touble |
07:46.25 | letherglov | right |
07:46.31 | letherglov | do you have a zaptel device or ztdummy installed? |
07:46.33 | beto75 | do you thinkk i s caused by ztdummy |
07:46.35 | beto75 | :) |
07:46.36 | BoRiS | jbot weather YWG? |
07:46.39 | beto75 | ztdummy |
07:46.45 | letherglov | hmmmmmmmm |
07:46.53 | letherglov | and you've got the usb modules installed too? |
07:46.59 | letherglov | since ztdummy uses those, right? |
07:47.11 | beto75 | GNFI |
07:47.24 | beto75 | I just compile zap. with ztdummy |
07:47.36 | beto75 | I didnt install anything more |
07:47.47 | beto75 | how do I do that? (sorry newbie) |
07:47.49 | letherglov | jbot weather kl21? |
07:48.11 | letherglov | jbot weather knkx? |
07:48.14 | beto75 | jbot weather mty |
07:48.27 | beto75 | no jbot doesnt have my weather :( |
07:48.29 | tzafrir | beto75, what distro, what kernel version? |
07:48.44 | beto75 | Suse 8.1 2.4.19 |
07:49.21 | modulus_ | jbot weather ZKPY? |
07:49.22 | tzafrir | beto75, is this a production server? exposed to the internet? (e.g: voip from the internet?) |
07:49.38 | tzafrir | if so: consider a newer distro |
07:49.50 | beto75 | <tzafrir> yes ,, is for testing purposes but its public in internet |
07:50.02 | beto75 | <tzafrir> you mean security? |
07:50.50 | beto75 | you are right , I will tell my customer to do that ,, he is a Novell-suse Old water wolf |
07:51.04 | beto75 | but he can change distro :) |
07:51.26 | modulus_ | jbot weather hkjk? |
07:51.31 | tzafrir | current SUSE (not SuSE anymore) is 9.2 |
07:51.36 | beto75 | better for me to stop my hair going down :) |
07:51.39 | modulus_ | i tooted |
07:53.03 | tzafrir | anyway, have you tried building zaptel with it? |
07:53.12 | beto75 | <letherglov> FYI.. when asterisk crash I get this |
07:53.14 | beto75 | Ouch ... error while writing audio data: : Broken pipe |
07:53.15 | beto75 | Segmentation fault |
07:54.03 | beto75 | <tzafrir> compiling zaptel was a real pain but it is compiled |
07:54.39 | quigleymd | Nukemizer: i think i remembered incorrectly, what i was thinking of was the setting for 'signalling' in zapata.conf - pri_net vs pri_cpe... sorry |
07:54.47 | tzafrir | so have you managed to load ztdummy? |
07:55.13 | beto75 | how can I see if its correctly loaded? |
07:55.34 | beto75 | I just take out the # in the Makefile |
07:56.28 | letherglov | hm |
07:56.30 | letherglov | bad ram? |
07:57.49 | beto75 | do you think is RAM? |
07:57.51 | beto75 | :S |
07:59.03 | letherglov | try getting a bunch of mp3's in a directory |
07:59.06 | letherglov | and run mpg123 for a while |
07:59.07 | letherglov | on the machine |
07:59.16 | letherglov | to see if it crashes indepedently of asterisk |
07:59.40 | *** join/#asterisk af_ (~af@ip-148-227.sn1.eutelia.it) |
08:06.18 | *** join/#asterisk Firestrm (F1r3570rm@S010600047577bccd.gv.shawcable.net) |
08:09.56 | eipi | distraction: http://www.rathergood.com/first_drink/ |
08:17.53 | beto75 | letherglov, can you be so kind to tell e how do I play a mp3 outside asterisk |
08:22.13 | letherglov | sure |
08:22.18 | letherglov | mpg123 <the mp3's name> |
08:22.27 | beto75 | :) let me test |
08:23.13 | beto75 | Can't open /dev/dsp! |
08:23.17 | letherglov | ok |
08:23.21 | beto75 | :s |
08:23.23 | letherglov | you don't have an audio device |
08:23.24 | letherglov | that's file |
08:23.25 | letherglov | fine |
08:23.26 | beto75 | thats the trouble? |
08:23.30 | letherglov | you should be able to dump it to a file |
08:23.31 | letherglov | no no |
08:23.37 | letherglov | run uhhh |
08:23.52 | letherglov | mpg123 -w test < file name> |
08:24.01 | letherglov | it'll write the output to a file called test |
08:24.05 | beto75 | ahh ok |
08:24.07 | letherglov | and < file name> is the name of the mp3 |
08:24.13 | letherglov | just so it decodes it, you know? |
08:24.24 | beto75 | let me test |
08:25.05 | beto75 | MPEG 1.0 layer III, 128 kbit/s, 44100 Hz joint-stereo |
08:25.05 | beto75 | [2:41] Decoding of fpm-sunshine.mp3 finished. |
08:25.55 | letherglov | hmm |
08:25.57 | letherglov | it worked |
08:25.58 | letherglov | damnit. |
08:26.01 | letherglov | now I'm really confused |
08:26.09 | letherglov | what version of mpg123 did it report running? |
08:26.10 | letherglov | if you run |
08:26.11 | letherglov | mpg123 |
08:26.21 | letherglov | Version 0.59s-r9 (2000/Oct/27). Written and copyrights by Michael Hipp. |
08:26.23 | letherglov | something like that |
08:26.40 | beto75 | Version 0.59r |
08:26.51 | letherglov | :- |
08:26.52 | letherglov | :-\ |
08:26.54 | letherglov | ok, I'm lost |
08:26.57 | beto75 | Version 0.59r (1999/Jun/15). |
08:27.17 | beto75 | do you think its the ztdummy? |
08:27.43 | beto75 | well the mos probable issue iis the Damn Old Suse |
08:30.02 | beto75 | hey ,, letherglov : why whatever mp3 file I test (as per your instructions) all are 2:41 secs |
08:31.24 | *** join/#asterisk gdsm (~gdsm@mk-ns500-1.uk.tiscali.com) |
08:35.33 | beto75 | thank you letherglov..... |
08:35.47 | beto75 | I need to sleep ,, Bye for now |
08:39.47 | Fanguin | hello, is somebody here who knows something about the meetme source? |
08:43.27 | *** part/#asterisk beto75 (~ha@201.128.177.84) |
08:45.02 | brc_ | Fanguin, not at the moment I don't think |
08:45.09 | brc_ | might try tomorrow afternoon if you can |
08:45.17 | brc_ | or, the -dev list |
08:48.19 | Fanguin | brc_, ok, i will choose one of the options. thank you. |
08:48.27 | brc_ | no problem |
08:48.34 | brc_ | the -dev list is your best bet |
08:48.55 | Fanguin | brc_, ok, good. |
08:57.54 | Firestrm | here is a brain teasing question.. is it possible to have distinctive ring patterns outgoing from asterisk to phones connected via fxs ports for example? |
08:58.16 | brc_ | yes |
08:58.39 | Firestrm | brc, where would i find info on this. (ive checked the wiki to no avail) |
08:58.49 | brc_ | depends on the phone |
08:59.01 | brc_ | and there IS info on the wiki |
08:59.09 | brc_ | use google, site:voip-info.org |
08:59.16 | Firestrm | in my case im using a standard consumer cordless phone connected to a sipura spa-3000 |
08:59.45 | brc_ | that ain't an fxs port to asterisk |
08:59.47 | brc_ | now is it |
08:59.48 | brc_ | nope |
08:59.51 | brc_ | it's a sip device |
08:59.55 | brc_ | http://www.google.com/search?num=100&hl=en&safe=off&client=firefox&rls=org.mozilla:en-US:unofficial&q=distinctive+ring+site:voip-info.org&spell=1 |
09:00.12 | brc_ | second result |
09:00.20 | brc_ | er |
09:00.21 | brc_ | 3'rd |
09:00.51 | Firestrm | thanks.. reading it now.. cant believe i didnt find this before.. |
09:00.55 | brc_ | yup |
09:00.59 | brc_ | very simple |
09:01.06 | brc_ | btw |
09:01.09 | brc_ | I don't think it says |
09:01.21 | brc_ | well it does |
09:01.24 | brc_ | but it's on another page |
09:01.45 | brc_ | you can generally use bellcore_1 to 6 |
09:02.24 | brc_ | Firestrm, http://www.voip-info.org/wiki-Sipura+SPA-2000 |
09:03.18 | Firestrm | nice.. thats perfect!! thanks |
09:06.42 | *** join/#asterisk zoa (zoa@82.103.76.147) |
09:06.47 | zoa | yooooooooooooooooooooooooooo |
09:06.51 | zoa | hey look whos here |
09:06.53 | zoa | its bkw!!!! |
09:10.43 | *** join/#asterisk Mw3 (mw3@daisy.chains.ch) |
09:25.39 | *** join/#asterisk w0w0 (~apardo@80.26.162.178) |
09:33.20 | *** join/#asterisk HjemmeRoyK (~roy@83.80-203-29.nextgentel.com) |
09:35.11 | Firestrm | brc_, well thats fricking awesome!!, works perfectly on sipura 3k!! |
09:36.39 | Firestrm | now if only i could get hanup to work properly :( |
09:36.51 | Firestrm | but i think that is Telus's problem |
09:49.48 | *** join/#asterisk jetscreamer (~jetscream@adsl-64-219-216-41.dsl.hstntx.swbell.net) |
09:53.24 | *** join/#asterisk darby_t (~tom@dno182.neoplus.adsl.tpnet.pl) |
10:05.29 | *** join/#asterisk implicit (~implicit@ip68-5-148-1.oc.oc.cox.net) |
10:10.27 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l01v-27-233.d4.club-internet.fr) |
10:44.41 | *** join/#asterisk WildPikachu[BED] (~wildpikac@wildpikachu.user) |
10:50.26 | *** join/#asterisk lohelle (slamm@213.161.252.253) |
10:51.22 | lohelle | hello! does anyone have X-lite web embedded (freeware)? It is not on Xten web page any more.. |
10:52.48 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
11:12.33 | Zeeek | wake up |
11:12.38 | djin | Source of Webiax seems to have gone from the web. Does anyone have a copy? |
11:12.39 | *** join/#asterisk benoitv (benoit@3.ds.rdns.acropolistelecom.net) |
11:12.39 | Qwell | no |
11:12.40 | Zeeek | the sykype is falling |
11:12.53 | Zeeek | hi djin! |
11:12.59 | djin | Hi Zeeek, |
11:13.07 | djin | how are you today. |
11:13.17 | Zeeek | I'm ok in spite of all the bad news |
11:13.25 | djin | bad news? |
11:13.36 | Zeeek | the good news is our new connection with static ip is up now |
11:13.40 | benoitv | hi all , i am stuffed with fax problems over Iax, local networks, g711, any success story ? |
11:14.10 | Zeeek | benoit fax using what? fax machine? spandsp? |
11:14.57 | benoitv | pstn zap-*-iax-*-zap pstn |
11:15.10 | Zeeek | between two fax machines? |
11:15.36 | benoitv | yes |
11:17.55 | Zeeek | http://lists.digium.com/pipermail/asterisk-users/2003-December/030362.html |
11:20.16 | HjemmeRoyK | benoitv: wait for t.38 support |
11:20.30 | HjemmeRoyK | benoitv: http://www.voip-info.org/tiki-index.php?page=Asterisk+T.38+Bounty |
11:20.39 | Zeeek | funny how fax hangs on |
11:20.41 | HjemmeRoyK | hm |
11:20.42 | HjemmeRoyK | http://www.voip-info.org/wiki-T.38 |
11:20.47 | HjemmeRoyK | yeah |
11:21.04 | HjemmeRoyK | all legal stuff |
11:21.09 | HjemmeRoyK | as in "fax is secure" |
11:21.18 | HjemmeRoyK | as in "windows is secure" |
11:21.21 | Zeeek | there is no good reason why a fax machine that emails transparently coulnd't be developed |
11:21.47 | Zeeek | fax is as secure as me scanning any doc, midifying it as I lki,e and refaxing it |
11:22.06 | benoitv | thanks, If I understand : no solution at the moment, waiting for T38. |
11:22.11 | HjemmeRoyK | Fax over email == T.37 :P |
11:22.14 | Zeeek | unless there were a secured fax machine, which could be done, fax is as insecure as email |
11:22.36 | Zeeek | fax ->PGP -> email |
11:22.38 | HjemmeRoyK | Zeeek: I know, but there are legal stuff that says fax == ok, email == bad |
11:22.48 | Zeeek | ture but bullshit just the same |
11:22.51 | Zeeek | true |
11:23.47 | benoitv | The point is that I am on a Lan... |
11:25.00 | HjemmeRoyK | benoitv: that should work |
11:25.10 | HjemmeRoyK | latency < 10ms may work with T.30 fax |
11:25.32 | HjemmeRoyK | and latency > 10ms on a lan isn't what you'd expect |
11:26.30 | djin | Or in this case: pstn zap-*-TIFF-*-zap pstn |
11:27.07 | HjemmeRoyK | djin: that'll require spandsp |
11:27.12 | djin | yes |
11:27.18 | Zeeek | then you add another variable element |
11:27.24 | benoitv | i use iax and G711 codecs and TE410 |
11:27.28 | HjemmeRoyK | meaning more trouble..... |
11:27.28 | Zeeek | spandsp can't receive from some faxes |
11:27.44 | HjemmeRoyK | benoitv: alaw/ulaw? where are you located? |
11:27.50 | djin | This looks like a controlled environment. |
11:28.09 | benoitv | I tried both but now in alaw (europe) |
11:28.18 | djin | spandsp only has to work with own faxmachine. |
11:29.01 | Zeeek | yeah but what if you have a machine it doesn't work with? |
11:29.16 | Zeeek | hey anyone have a siemens DECT phone? |
11:29.18 | benoitv | right Zeeek |
11:29.24 | *** part/#asterisk X-Gen (~x-gen@rrba-146-67-74.telkomadsl.co.za) |
11:29.39 | benoitv | yes Zeeek |
11:29.59 | Zeeek | benoitv I say this because my customers have machines that I can't receive with spandsp |
11:30.32 | benoitv | I understand and I also cant afford that, my customers wont understand |
11:30.39 | Zeeek | excatly! |
11:31.14 | Zeeek | I asked one to fax me for a test she did and it looped until timeout and tried 3 times. Not very impressive compared to a $40 fax machine |
11:31.36 | benoitv | do we have a way to see jitter, packet loss, and so on between 2 * on a Lan ? |
11:32.01 | Zeeek | doesn't show channels give yuou that? |
11:32.09 | Zeeek | (snapshot) |
11:32.51 | Zeeek | maybe only lag... |
11:33.17 | benoitv | no |
11:33.39 | Zeeek | RoyK do you have phones and FXS ? |
11:34.13 | Zeeek | benoitv are you in Eu? |
11:37.34 | *** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net) |
11:37.54 | benoitv | yes in France |
11:38.37 | Zeeek | me too |
11:38.54 | Zeeek | I'm trying to figure out why my new phones don't show CID |
11:39.19 | Zeeek | asked on mailing list but tout le mode s'en fout de ce qui se passe en France |
11:39.49 | benoitv | siemens dont show cid ? wich model, on what equipement ? |
11:40.24 | Zeeek | siemens C200 - brand new - works on PSTN, asterisk ->TDM400 no |
11:40.40 | Zeeek | my other shitty Alcatel CID works, number and name |
11:40.52 | Zeeek | I can't understand what could have changed |
11:41.43 | Zeeek | I mean in technology, nothing changed on my system |
11:42.00 | Zeeek | benoitv were you at any asterisk lunches in Paris with Mark? |
11:42.08 | benoitv | yes :-) |
11:42.17 | Zeeek | ah we've maybe spoken by email |
11:42.27 | benoitv | I guess so |
11:42.35 | Zeeek | you are from, let me see |
11:42.49 | Zeeek | apolo somthing? |
11:42.55 | Zeeek | adonis? |
11:42.56 | benoitv | acropolis |
11:43.00 | Zeeek | yes, that's it! |
11:43.28 | Zeeek | comment vont les affaires? :) |
11:46.15 | benoitv | I get it, fax is working but with echocanceller set to off. |
11:46.31 | benoitv | The point is I need echo canceller for voice communications, any idea ? |
11:46.53 | Zeeek | no I heard it worked with the echocancel set to off |
11:47.06 | benoitv | voice ? |
11:48.14 | Zeeek | no fax can work if echocancel is off |
11:48.25 | Zeeek | not really helpful |
11:48.28 | benoitv | right |
11:49.23 | benoitv | If I could chang dynamicly echo canceller, I could use fax detection and then chang echo cancelling... hum.. possible ? |
11:50.36 | benoitv | someone have an idea about T38 delivery on asterisk, end of march is realistic ? |
11:52.14 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
11:52.20 | Zeeek | muhahahaha |
12:35.41 | *** join/#asterisk ZX81_Laptop (~chatzilla@222-153-20-92.jetstream.xtra.co.nz) |
12:41.24 | ZX81_Laptop | ~ping |
12:42.01 | jbot | pong |
12:42.01 | Zeeek | pr0n |
12:42.01 | ZX81_Laptop | :) |
12:42.18 | Zeeek | major lag |
12:42.34 | ZX81_Laptop | yeah? |
12:42.37 | ZX81_Laptop | me or you? |
12:42.45 | Zeeek | 3 minutes? not good enuf for voIP |
12:42.53 | ZX81_Laptop | LOL |
12:42.56 | Zeeek | even in Australia |
12:42.59 | ZX81_Laptop | hehe |
12:43.03 | Zeeek | although... |
12:43.22 | Zeeek | why can't my exec() cmd work in Apache/PHP? |
12:44.29 | ZX81_Laptop | permissions |
12:44.45 | *** join/#asterisk Pantanero (~Pantanero@a213-22-84-44.netcabo.pt) |
12:45.17 | Zeeek | yeah |
12:46.38 | Zeeek | except no |
12:46.47 | ZX81_Laptop | lol |
12:46.50 | benoitv | someone has success with beronet 4 or 8 bri cards ? |
12:54.30 | hellop | Gentlemen, we have a go. |
12:54.41 | hellop | I just got asterisk running on my VIA800mgz!!!!!! |
12:54.43 | hellop | woohoo! |
12:56.00 | hellop | had to comment out a few lines from the makefile and recompile asterisk. as per Asterisk Compile on the wiki. |
12:57.45 | Zeeek | ok suid bit solved that problem |
12:58.11 | Zeeek | hellop that's great - I hate to see probalmes because of hardwxare you do,'t wanna replace! |
13:00.32 | tzafrir | what exactly is the license of the client part of the flash operator panel? |
13:00.46 | tzafrir | a/ware/wear/ |
13:01.47 | tzafrir | hellop, what CPU is that exactly? |
13:02.12 | tzafrir | Debian's patch was to use: |
13:02.30 | tzafrir | PROC=$(shell uname -m) |
13:02.41 | tzafrir | which would probably have worked for you |
13:06.17 | hellop | tzafrir, VIA C3 800mgz CPU |
13:06.49 | hellop | Matsonic MSCLE266-F Motherboard. $70 new from online store. (paid $80 from ebay) |
13:08.17 | hellop | tzafrir, yeah I use debian. but I commented that line out to get it to work. |
13:09.03 | tzafrir | hellop, I've just submitted a bug to debian about this. Latest Rapid packages now replace i686 with i586 |
13:09.21 | hellop | now, if only I had written down everything else I did to get it to work... ;) |
13:09.38 | tzafrir | and the dilema now is whether to leave it as 586 or lower it down to 386 |
13:09.52 | tzafrir | (for those 386-users using *) |
13:15.01 | Mw3 | anyone here who is good in fiber optics ? |
13:18.51 | *** join/#asterisk mogorman (~mogorman@146.229.185.151) |
13:19.50 | *** join/#asterisk gbs (~GBS@adsl-5063187d.monradsl.monornet.hu) |
13:24.08 | hellop | fiber optics? easy. cut cable, insert jack. done |
13:24.16 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
13:25.34 | Mw3 | hellop: ah :) |
13:27.06 | mogorman | morning |
13:41.49 | nitram | benoitv: yes, they work with misdn/chan_misdn though nt_pp is not yet supported |
13:43.38 | hellop | the X100P can make phone calls right? |
13:44.24 | benoitv | I dont get how to specify some low level things as isdn protocol |
13:44.37 | benoitv | with chan_misdn |
13:44.47 | benoitv | and I get a Nt stack error -6 |
13:46.01 | benoitv | my config is *-beronet-alcatelpabx |
13:46.13 | hellop | lemme rephrase, I want to configure asterisk to make phone calls, play different messages, and read keypresses, using some type of script. Any suggestions on where to start research/what to search for? |
13:48.58 | *** join/#asterisk ddfire (~ircap751@OL206-180.fibertel.com.ar) |
13:49.33 | ddfire | hi |
13:50.18 | ddfire | i am new whit asterisk and i am looking for a software phone for windows (sorry) i get openphone it is good? |
13:52.08 | hellop | basically, same as voice mail, only ATDT #number instead of ATA |
13:52.20 | hellop | call instead of answer |
13:52.34 | nitram | benoitv: then you probaly have the same problem, nt_pp does still not work/is not yet implemented |
13:54.38 | *** join/#asterisk pif (~pif@zenon.apartia.fr) |
13:57.23 | *** join/#asterisk P-Chan (~pchan@68.142.66.200) |
13:57.34 | ddfire | can someone sugest me a software phone to use whit asterisk for windows? |
13:58.08 | P-Chan | WHew...thank god there are people here... ;) I'm been slaving over a new VoIP box for the last 20 hours straight. I need help (as well as more caffeine)! |
13:59.08 | P-Chan | Long story short: Previous admin built an asterisk server with slack 10, he quit, on reboot server completely died (luckily this was fri night) - I reinstalled a new server w/ Gentoo w/ 2.6 w/ udev and need to get the old config up and running. |
13:59.56 | P-Chan | I'm getting the error: Feb 12 23:56:58 NOTICE[8343]: app_dial.c:749 dial_exec: Unable to create channel of type 'Zap' |
14:00.26 | P-Chan | dmesg says: TE410P: Span 1 configured for ESF/B8ZS |
14:00.26 | P-Chan | Registered tone zone 0 (United States / North America) |
14:01.00 | P-Chan | <PROTECTED> |
14:03.29 | P-Chan | I guess sleep would be an option right about now. I hate to do that and shorten my available time to get this fixed (not only that, but I'd sleep better knowing its working). |
14:05.18 | ddfire | [P-Chan] i thing you are alone i ask something and no one answer |
14:10.22 | P-Chan | Hello? Anyone else awake? |
14:11.11 | hellop | It works!!!! |
14:11.19 | hellop | YEEEHA time to celebrate. |
14:11.57 | P-Chan | Yes! Another living, breathing person! <dances around with hellop> Please tell me you may be able to help me with my Unable to create channel of type 'Zap' problem? |
14:12.54 | P-Chan | Although, I'm guessing from your jubilation that you were fighting with it yourself. |
14:15.07 | hellop | oh P-chan yeah same 20 hours, only my reason was for fun/destructive behaviour |
14:15.19 | hellop | yes possibly |
14:15.35 | hellop | your zaptel drivers not loaded me thinks.. |
14:16.18 | P-Chan | hellop: lemme list my lsmod output: |
14:16.26 | hellop | try: modprobe zaptel; modprobe wcfxo; ztcfg -vv; |
14:16.32 | P-Chan | wct4xxp 70048 0 |
14:16.32 | P-Chan | zaptel 183428 53 wct4xxp |
14:16.32 | P-Chan | ppp_generic 27028 0 |
14:16.32 | P-Chan | slhc 8960 1 ppp_generic |
14:16.46 | P-Chan | I use the T4110P card. |
14:16.56 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
14:16.58 | P-Chan | oh, ztcfg -vv, I need to try that |
14:17.20 | P-Chan | Channel 24: D-channel (Default) (Slaves: 24) |
14:17.20 | P-Chan | 24 channels configured. |
14:17.29 | P-Chan | so according to that its configured. |
14:18.06 | hellop | I have a simple and easy fix for you. |
14:18.26 | hellop | You are a business. Wouldn't it be more cost effective to simple hire an expert? |
14:18.50 | P-Chan | Well, actually I was thrown into this position... |
14:19.04 | P-Chan | and the previous admin left the asterisk server in a non-rebootable state. :/ |
14:19.27 | hellop | yeah.. well, U can get approval to spend $50 on some telphone support right? |
14:19.35 | P-Chan | with who? |
14:19.40 | *** join/#asterisk satlink (satlink@66.178.97.50) |
14:19.40 | hellop | pleanty of people... |
14:19.41 | P-Chan | Digium isn't open, that's who I'd call |
14:19.59 | hellop | One guy here on this list offered help last night... |
14:20.10 | hellop | I have only his e-mail. |
14:20.22 | hellop | There are also sources to be found on the web. |
14:20.37 | *** join/#asterisk sezuan (sezuan@port-212-202-57-119.dynamic.qsc.de) |
14:20.49 | P-Chan | my problem is that I need to get this up and running today...I've been at this for about 20 hours straight and made quite alot of progress (found out devfs patch for 2.6 doesn't work well, switched to udev) |
14:21.46 | ddfire | bye |
14:21.49 | benoitv | nitram:I configured it in mtp, can't I use pristuff from junghans or capi or.. |
14:22.27 | nitram | benoitv: u can use bristuff, but you have to change the vendor id in the driver for it to accept your beronet card |
14:22.36 | satlink | Hi, I'm quite new with asterisk, and I am a bit stuck.. I can call from diax to xpro, everything workes fine, but I cant call from xpro to diax. Any idea where to look? hints? |
14:22.37 | hellop | P-Chan, can you not get ahold of/hire the old tech guy? I'm sure he's not pissed at you. |
14:22.42 | hellop | Well, I assume. |
14:24.52 | hellop | I was getting that error.. hmm how did I fix it. |
14:25.25 | hellop | <PROTECTED> |
14:25.30 | P-Chan | yeah |
14:25.45 | hellop | and you are using his old .conf scripts? |
14:25.49 | P-Chan | yes |
14:25.56 | hellop | 24 line system? |
14:26.00 | P-Chan | pri |
14:26.12 | hellop | ? |
14:26.16 | P-Chan | with 24 lines I'd assume. It's set up with 23 bchans and 1 dchan |
14:26.49 | P-Chan | PRI = digital phone line. It's 1 cat 5 cable that has digital capacity for 24 separate 64k lines |
14:27.18 | P-Chan | it comes straight into a TE410P by digium (into 1 of the 4 ports) |
14:27.28 | hellop | I just got my Asterisk to anser the phone and play a .wav.. Not sure I can really help much. |
14:27.39 | hellop | Sux that U had to start from a new server tho... |
14:27.45 | P-Chan | Your attempt is appreciated tho |
14:27.56 | P-Chan | yeah, I wish I could have pieced the old one together |
14:28.00 | hellop | I'm using Knoppix with the 2.4.27 kernel... |
14:28.39 | P-Chan | actually, I just realized something. Maybe now that I've been working on this for so long, maybe I can attempt to get the old system working now? <sigh> |
14:29.06 | P-Chan | I know alot more about the system now than I did fri night...maybe I was missing something back then. |
14:29.26 | hellop | Well, I was stuck last night. U need sleep, man. You'll be a much better admin with sleep. It would be worth it to get sleep. |
14:30.03 | P-Chan | lol...yeah, but how can I sleep knowing that we could loose a big client of ours Monday morning if I can't get it running? |
14:30.04 | *** join/#asterisk sambal (~sambal@gateway.office.flatbox.nl) |
14:30.24 | P-Chan | big client = call center... <yikes> |
14:31.03 | P-Chan | oh...the box is at the colo...I'm too tired to even realize that.. :/ Maybe I should get some sleep |
14:31.19 | benoitv | nitram : euh... It will work ? how can I change the vendor id ? is it legal ? |
14:31.35 | hellop | do you have the line: [channels] in your zapata.conf? |
14:31.52 | hellop | P-chan tomorrrow is Sunday |
14:32.04 | Mw3 | this day is sunday |
14:32.05 | hellop | Go sleep. |
14:32.12 | Mw3 | it depends on your location :) |
14:32.30 | P-Chan | signalling=pri_cpe |
14:32.41 | Mw3 | P-Chan: can you share your zaptel.conf ? |
14:32.49 | P-Chan | channel => 1-2 |
14:32.54 | P-Chan | yea, just a sec |
14:33.08 | hellop | no just [channels] |
14:33.23 | hellop | I think that was my prob.. |
14:33.41 | hellop | missing "[channels]" |
14:34.06 | *** join/#asterisk SeaForth (~SeaForth@c-24-1-126-202.client.comcast.net) |
14:34.28 | P-Chan | copying the file to a websever, just a sec. |
14:34.58 | hellop | use pastebin.ca fast easy |
14:36.41 | P-Chan | http://pastebin.ca/5754 |
14:36.49 | P-Chan | Thanks for the tip, never used it before. ;) |
14:38.48 | P-Chan | channel => 1-23 is at the bottom of the file |
14:40.57 | hellop | Mw3, I notice he has no "signalling=" in his zapata.conf |
14:41.37 | P-Chan | line 74 |
14:41.49 | P-Chan | ^_^ |
14:41.49 | Mw3 | 074 signalling=pri_cpe |
14:42.02 | hellop | oh |
14:43.07 | hellop | give me your e-mail and I'll forward it to the guy that offered to help me.. |
14:43.19 | P-Chan | ok, jpfingstmann@azxws.com |
14:43.30 | P-Chan | btw, http://pastebin.ca/5755 has the exact error I get with asterisk |
14:45.05 | P-Chan | btw, thanks hellop - If I can't get this resolved by noon I'll give him a call. |
14:45.36 | Mw3 | P-Chan: and this is exactly the same config whihc worked before ? |
14:45.50 | *** join/#asterisk Total-Net (~me@user-11famtp.dsl.mindspring.com) |
14:46.01 | Total-Net | Hi all |
14:46.04 | P-Chan | as far as I know, all I did was reboot it, but there may have been changes made that hadn't taken effect. |
14:46.10 | P-Chan | hi Total-Net |
14:46.29 | P-Chan | So the running (working) conf, may not have been this one. /sigh |
14:46.37 | hellop | reboot.. maybe a module isn't being loaded? |
14:46.45 | P-Chan | did it...several times. |
14:46.52 | hellop | is there any .old confs? |
14:47.00 | P-Chan | I've been manually starting everything and loading the modules myself. |
14:47.01 | Total-Net | anyone know what happened to asterisk.xvoip.com ? |
14:47.35 | P-Chan | not of zapata.conf |
14:47.57 | Total-Net | that is the only place I can't look for a possible solution to my problem |
14:49.18 | P-Chan | anyway to get more useful error output from asterisk? |
14:49.29 | hellop | Ok, I sent an e-mail to zoa... he does Asterisk support for a living. |
14:49.42 | P-Chan | ok, cool. Thanks. |
14:50.06 | Mw3 | P-Chan: type zap show channels on CLI |
14:50.32 | P-Chan | <PROTECTED> |
14:50.32 | P-Chan | <PROTECTED> |
14:50.32 | P-Chan | <PROTECTED> |
14:50.36 | P-Chan | all the way through 23 |
14:51.23 | Mw3 | P-Chan: try starting asterisk in verbose mode, asterisk -vvvvvc |
14:51.51 | P-Chan | I was using asterisk -vvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvgr - I'll try that tho now |
14:53.01 | P-Chan | same output as I had... (I added an r to -vvvvvvvc to make it connect to the running asterisk service. |
14:53.30 | P-Chan | I just killed the process and ran with just vvvvvvvvvvvvc |
14:53.43 | Mw3 | a -d may help ... but i dunno |
14:54.02 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
14:54.21 | ariel_ | Morning all hope everyone is going fine today. |
14:54.33 | Mw3 | what does ztcfg -vvvv do ? |
14:55.08 | P-Chan | SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) |
14:55.08 | P-Chan | Channel map: |
14:55.08 | P-Chan | Channel 01: Individual Clear channel (Default) (Slaves: 01) |
14:55.18 | P-Chan | ... Channel 23: Individual Clear channel (Default) (Slaves: 23) |
14:55.18 | P-Chan | Channel 24: D-channel (Default) (Slaves: 24) |
14:55.18 | P-Chan | 24 channels configured. |
14:58.04 | P-Chan | anything else I can try at the CLI? Something that may give me some idea of what its doing? (or not doing)? |
15:00.18 | P-Chan | show channels is blank, is that normal? |
15:00.42 | *** join/#asterisk file[laptop] (~file_lapt@mctn1-1056.nb.aliant.net) |
15:01.30 | Mw3 | it shows the active channels i think |
15:01.38 | Mw3 | so it's normal in this case |
15:01.40 | P-Chan | that's what I figured. :/ |
15:03.35 | P-Chan | It's a good thing the server is about 5 miles away at a colo site...I would have thrown it out the window by now. |
15:03.55 | *** join/#asterisk cc (~cc@byte.fedora) |
15:04.19 | Mw3 | is /etc/zaptel.conf okay ? |
15:06.01 | P-Chan | http://pastebin.ca/5754 - it looks ok, from what I can see |
15:06.35 | P-Chan | oooh...I have a thought. |
15:06.42 | P-Chan | permissions maybe? |
15:07.15 | Mw3 | you pasted /etc/asterisk/zapata.conf |
15:07.20 | Mw3 | i asked /etc/zaptel.conf |
15:07.26 | P-Chan | oh |
15:08.14 | *** join/#asterisk TheEmperor (TheEmperor@218.111.50.121) |
15:08.16 | Mw3 | check the permissions, but i think the problem is not that |
15:08.26 | P-Chan | http://pastebin.ca/5756 |
15:08.31 | *** part/#asterisk asher (ben@62.121.2.101) |
15:08.39 | P-Chan | just checked, the process runs as root. :/ not a permissions issue. |
15:09.08 | *** join/#asterisk Dr-Linux (~sshah@202.125.141.6) |
15:10.45 | Mw3 | P-Chan: this /etc/zaptel.conf is also the old one ? |
15:10.52 | P-Chan | Yes |
15:11.00 | P-Chan | I haven't touched it at all. |
15:13.49 | *** join/#asterisk zoa (zoa@82.103.76.147) |
15:14.31 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
15:14.47 | *** join/#asterisk clinthome (~clinthome@64.241.37.140) |
15:15.17 | *** join/#asterisk Total-Net (~me@user-11fa2gi.dsl.mindspring.com) |
15:15.44 | Total-Net | I hate dynamic leases |
15:16.00 | Dr-Linux | to defiene Holdon Music necessary to make entry with users extention in extenstions.conf ? |
15:16.27 | Mw3 | P-Chan: i have no more idea, i would start to try things like strace :) |
15:16.27 | P-Chan | if it kicked you off, I'd say...normally it shouldn't. (The client should tell it its still being used) |
15:16.53 | P-Chan | I was afraid you were gonna say something like that. |
15:17.11 | P-Chan | oops, that was supposed to be a /msg to Mw3 |
15:19.51 | Dr-Linux | lol |
15:19.59 | Dr-Linux | to defiene Holdon Music necessary to make entry with users extention in extenstions.conf ? |
15:26.33 | *** join/#asterisk mhnoyes (~mhnoyes@user-38lc04v.dialup.mindspring.com) |
15:27.55 | *** join/#asterisk meebey (meebey@meebey.net) |
15:28.20 | meebey | can I make a caller id mapping? |
15:28.38 | meebey | when someone calls me via ISDN, the callerid is ISDN_NUMBER@my_asterisk |
15:28.58 | meebey | but when I want to call that person it must have a 0 in the beginning |
15:29.18 | meebey | I would like to avoid having 2 entries in my hardphone's addressbock |
15:29.23 | meebey | s/bock/book/ |
15:30.17 | Total-Net | I need to ask my question again as I disconnected before I saw anything.. anu one know anything about 'Tundo Telport-16' ? |
15:30.26 | meebey | same of course for SIP callers that come in via a sip gateway |
15:30.42 | file[laptop] | meebey: what about just adding 0 when you dial out through ISDN... |
15:30.46 | file[laptop] | like... |
15:31.08 | meebey | file[laptop]: thats the problem I have, the telephon should be able to call ppl back |
15:31.08 | file[laptop] | well, let's see if you can think of how |
15:31.17 | meebey | file[laptop]: the the missed calls feature of ny telefon |
15:31.22 | file[laptop] | yes... and it can |
15:31.27 | file[laptop] | but your asterisk is what actually places the call |
15:31.33 | *** join/#asterisk oej (~oej@40.186.204.213.sol.worldonline.se) |
15:31.35 | meebey | file[laptop]: it will try to call the callerid which is not the exact right number for me to call |
15:31.42 | file[laptop] | just because your phone says call 5551212 doesn't mean you can't actually 05551212 |
15:31.46 | file[laptop] | it's all in how you write your dialplan |
15:31.52 | meebey | ic |
15:31.54 | file[laptop] | like for example |
15:32.04 | file[laptop] | exten => _NXXXXXX,1,Dial(SIP/0${EXTEN}@myprovider) |
15:32.05 | meebey | you mean I should map in my calling not in the receiving |
15:32.12 | file[laptop] | that would call 05551212 instead of 5551212 |
15:32.25 | file[laptop] | just an example. |
15:32.37 | file[laptop] | you are overcomplicating things a lot :) needlessly |
15:32.39 | meebey | hm |
15:33.15 | meebey | my ISDN actually has to call 5551212 but I use the 0 to indicate its a ISDN call |
15:33.25 | meebey | via a dialplan |
15:33.35 | *** join/#asterisk smurfix (~smurf@smurfix.developer.debian) |
15:33.39 | file[laptop] | see you never said that :) |
15:33.40 | nitram | benoitv: bristuff is gpl, so u can change whatever you want |
15:33.44 | nitram | hi meebey :) |
15:33.47 | meebey | file[laptop]: yes but know I do :) |
15:33.50 | meebey | s/know/now/ |
15:33.52 | meebey | nitram: hiya ;) |
15:33.54 | file[laptop] | you'll have to do some fun stuff... |
15:34.02 | meebey | file[laptop]: I like fun stuff |
15:34.06 | meebey | file[laptop]: tell me more :) |
15:34.14 | file[laptop] | no, I'm not - you're going to figure it out |
15:34.40 | meebey | file[laptop]: what I am trying to do is possible at least? |
15:34.46 | nitram | meebey: i would prepend the zero |
15:34.56 | nitram | in the callerid already |
15:35.06 | file[laptop] | see? now there's a creative idea |
15:35.08 | meebey | nitram: I want this also for the SIP calls |
15:35.15 | nitram | so? |
15:35.23 | meebey | thats what I was trying to ask and do :) |
15:35.23 | file[laptop] | all you have to do is think. |
15:35.34 | meebey | prepending the 0 in the callerid when the call comes in |
15:35.43 | yashax | Guys: Which on would you recommend - Polycom IP500/600 or Cisco 7960? |
15:35.46 | meebey | file[laptop]: all you have to do is listen |
15:35.51 | file[laptop] | meebey: have you looked at the available applications, like... setcallerid? |
15:36.02 | nitram | meebey: btw... i do not have to prepend the zero on incoming capi calls |
15:36.11 | file[laptop] | nitram: it's the way his dialplan is setup |
15:36.34 | nitram | file[laptop]: no, i mean it is already there (the zero) |
15:36.37 | meebey | file[laptop]: hm no, I will take a look |
15:37.01 | file[laptop] | nitram: ^^ he has to put 0 in front so his dialplan knows to send it out ISDN |
15:37.12 | nitram | ih... duh |
15:37.39 | meebey | nitram: my brother just called me via ISDN, 123123@my_asterisk it showed in the display (the caller id), but telephon doesnt know that number |
15:37.57 | meebey | nitram: it knows 0123123@my_asterisk, the 0 is required for going out ISDN |
15:38.11 | meebey | nitram: so I want to control the callerids for incoming calls |
15:38.14 | nitram | meebey: so just prepend the zero on incoming isdn calls |
15:38.34 | meebey | nitram: thats what I will try to do now :) setcallerid is the application I need? |
15:39.01 | nitram | setcidnum/setcidname/setcallerid |
15:39.04 | file[laptop] | there's setcallerid... setcidnum... setcidname... so many to choose from, which do you think? |
15:39.10 | nitram | *g* |
15:39.23 | meebey | allright I will give it a try, thanks so far |
15:39.26 | nitram | choices, choices, choices |
15:39.28 | nitram | ;) |
15:39.30 | meebey | nitram: yeah |
15:40.15 | *** join/#asterisk heka (~fasada@82.114.68.126) |
15:40.35 | heka | Hello, is there any problem with CVS server? |
15:40.45 | Corydon76-home | Lunch at 9:40 a.m.? |
15:41.08 | file[laptop] | 11:40AM |
15:41.19 | yashax | Guys: Which on would you recommend - Polycom IP500/600 or Cisco 7960? |
15:41.33 | Corydon76-home | Polycom, hands down |
15:41.57 | file[laptop] | Corydon76-home: hands down below? |
15:42.15 | Corydon76-home | file[laptop]: heh, maybe after I recover from this cold |
15:42.25 | file[laptop] | I thought so |
15:43.17 | yashax | does anyone else have an oppinion on this? - Polycom IP500/600 or Cisco 7960? |
15:43.34 | Corydon76-home | The Polycom is a much better phone |
15:44.17 | Damin | I'd get the Cisco.. |
15:44.26 | Damin | I have both and I like the Cisco 7960 better.. |
15:44.32 | Damin | Everything works on it.. |
15:44.40 | Damin | The Polycom has some "issues" |
15:44.40 | wwalker | One problem with Polycom 500 is that the maximum volume is very low on the speakerphone. |
15:44.42 | yashax | why do you like it better? |
15:44.44 | Corydon76-home | There's a rumor floating around that Polycom wrote the Cisco phone software |
15:44.56 | Damin | yashax: Cause it works. ;) |
15:45.15 | yashax | what about the sound quality and speakerphone? |
15:45.21 | Damin | Corydon-w: Cisco licenses some portions of the software and speakerphone technology from Polycom. |
15:45.41 | Damin | yashax: Cisco and Polycom are both great speakerphones. |
15:45.50 | wwalker | Don't know about the Cisco, but the lack of a backlight on the LCD is a sore spot with me on the Polycom 500. |
15:46.13 | yashax | does cisco 7960 has backlit? |
15:46.17 | Damin | No.. |
15:46.20 | Damin | Neither of them does. |
15:46.25 | wwalker | bummer |
15:46.25 | Damin | Which is kind of lame.. |
15:46.31 | yashax | what about 600? |
15:46.37 | Damin | Makes it hard to use the phone in the dark. ;) |
15:46.49 | Corydon76-home | Oddly, enough, the Grandstream BT102 has a backlight |
15:46.54 | yashax | yeah strange..... |
15:47.20 | yashax | how is the sound quality compares to Grandstream phones? |
15:47.31 | heka | is there any problem with cvs.digium.com because Im unable to connect! |
15:47.44 | yashax | I am trying to choose whcih phones to buy for our implementation...... |
15:48.05 | Corydon76-home | The sound quality is more dependent on the available bandwidth than on the phone model |
15:48.08 | Damin | Corydon-w: I still have not gotten an answer to this question: http://lists.digium.com/pipermail/asterisk-users/2005-January/084548.html |
15:48.47 | Damin | yashax: Personally, I'd reccomend the Cisco.. I've deployed both.. |
15:49.23 | zoa | heya damin |
15:49.25 | zoa | long time no see |
15:49.28 | Damin | yashax: Don't get me wrong, I like the Polycom phones.. However, on the Cisco, everything works! |
15:49.28 | zoa | are you going to von ? |
15:49.29 | yashax | damin: great, thank you.... |
15:49.29 | Corydon76-home | Damin: Ah. Can't comment on it. I personally prefer a hardphone |
15:49.35 | zoa | who is going to von ??? |
15:49.41 | Damin | zoa: Howdy! Yeah.. I'll be at Von. |
15:49.47 | zoa | Coool |
15:49.53 | yashax | are there cenrtain features that do not work on polycomes> |
15:49.55 | Damin | zoa: Where are you staying? |
15:49.59 | zoa | i will be a day early because flight was a lot cheaper then |
15:50.05 | zoa | dunno kram would arrange me something |
15:50.07 | Damin | yashax: http://lists.digium.com/pipermail/asterisk-users/2005-January/084548.html |
15:50.24 | zoa | damin, how about a preparty ? :) |
15:50.38 | Damin | zoa: I'm flying in on Sunday.. I'd love to.. |
15:51.00 | Damin | zoa: I need to find out where everyone is going to be staying.. Wanna sort of stick with the crew from Astricon.. |
15:51.08 | *** join/#asterisk pcm (~pcm@user-69-73-0-22.knology.net) |
15:51.42 | zoa | i will arrive the fifth |
15:51.45 | zoa | dunno what day that is |
15:51.58 | zoa | in the evening |
15:52.04 | zoa | well evening for you guys :( |
15:52.29 | Damin | zoa: Hehe.. I have a feeling that I'm going to be whipped after that week. :) |
15:52.44 | zoa | i hope its going to be a tough week :) |
15:53.17 | file[laptop] | Damin: we're at the Saint Claire Hyatt |
15:53.38 | file[laptop] | across the street from the conference place |
15:54.59 | file[laptop] | Damin & zoa: I will be at VON too btw |
15:55.02 | Moc | morning |
15:55.10 | Corydon76-home | Have fun, file |
15:55.26 | file[laptop] | getting there will be fun |
15:56.14 | zoa | cheers file |
15:56.18 | zoa | will see you there then i guess |
15:56.20 | nitram | who is going to cebit btw? |
15:56.23 | file[laptop] | yup |
15:56.28 | zoa | nitram my collegue is |
15:56.33 | yashax | damin: did you have the same proble with cisco of having a very low volume on the speaker phone? |
15:56.35 | zoa | im not im stuck again |
15:56.50 | zoa | yashax: i never had that problem |
15:56.59 | heka | any idea why this error: /usr/bin/ld: cannot find -lmysqlclient? |
15:57.09 | file[laptop] | heka: you don't have the mysql client library installed |
15:57.10 | Damin | file: Rockin.. |
15:57.30 | file[laptop] | I'll probably be the last one to show up |
15:57.32 | Damin | file: Same here.. |
15:57.39 | file[laptop] | ooh what time? |
15:57.50 | heka | file[laptop]: I have mysql-3.23.54a-11 |
15:57.51 | Damin | file: Well.. Not LATE night.. |
15:57.55 | yashax | cool... Would you guys recommend the Cisco 7960 or the other model? |
15:57.59 | Damin | file: Don't know.. Haven't booked my tickets yet.. |
15:58.05 | heka | file[laptop]: Isn`t this? |
15:58.05 | file[laptop] | heka: you don't have the mysql client library installed, believe me |
15:58.22 | Damin | yazhax: I've not had any problems w/ volume on the Cisco speakerphone. It rocks! |
15:58.24 | file[laptop] | Damin: well anyway, 10:43PM for me |
15:58.49 | file[laptop] | taking 14 hours to get there... oh joy |
15:58.53 | Damin | file: Do you have the Tel# for the Hotel? |
15:59.00 | file[laptop] | 408-298-1234 |
15:59.15 | zoa | file what day is the sunday ? |
15:59.18 | zoa | is that the fifth ? |
15:59.20 | zoa | or the 6th ? |
15:59.21 | file[laptop] | 6th |
15:59.24 | zoa | ah damn |
15:59.28 | zoa | so i arrive the 5th |
15:59.29 | file[laptop] | arrive Sunday, leave Saturday |
15:59.33 | zoa | i will be all alone there |
15:59.50 | Damin | zoa: I could fly in early if you have a Hotel room I could crash at! ;) |
15:59.56 | file[laptop] | and if nobody shows to pick me up at the airport I'm disowning all of you :p |
16:00.06 | Damin | file: it's called a "taxi" ;) |
16:00.12 | file[laptop] | I'll still disown you |
16:00.28 | file[laptop] | :p |
16:00.41 | *** join/#asterisk Inv_arp (junya@adsl-3-244-239.mia.bellsouth.net) |
16:00.49 | yashax | would you recommend any other cisco models then 7960??? |
16:00.57 | Damin | yashax: nope. |
16:01.04 | Damin | yashax: well.. 7940 is good too.. |
16:01.23 | yashax | also where would you recommend is the best place to buy them? Also, should I ask any questions to ensure that I get the right the correct one with all the features? |
16:01.48 | Damin | yashax: voipsupply.com has them... Decent people.. good support.. |
16:02.03 | Damin | yashax: Just tell them you want SIP loaded on it.. |
16:02.48 | yashax | got it.... thank you so much.... |
16:04.24 | yashax | can you please tell me if I do not want to use the analog lines and want the straight voip, then I would need PSTN gateway, right? Where can I find someone that will offer me a few local numbers with unlimited local/long distance for a low fee per month? |
16:06.44 | Damin | yashax: Is this for a business? |
16:07.03 | Damin | yashax: If so, you'll want to tie it into the PSTN directly rather than rely on VoIP only.. |
16:07.13 | yashax | yes... there are just two of us for now, but will be adding just a few folks shortly.. |
16:08.09 | yashax | we will in the future, but due to cost and the situation can not get any pots lines right now.... |
16:09.24 | yashax | can you please recommend someone? |
16:10.42 | *** join/#asterisk dca[laptop] (~dca[lapto@sta-207-174-139-178.rockynet.com) |
16:14.46 | yashax | Damin? Still there? |
16:14.47 | sivana | yashax: where are you located? |
16:14.52 | yashax | Atlanta |
16:15.12 | sivana | you will need to find someone with Atlanta DIDs |
16:15.31 | yashax | yeah.... do you know someone? |
16:15.54 | sivana | well.. you can email the mailing list or check out the larger providers like Broadvoice |
16:16.11 | yashax | k.. thanks.. |
16:16.18 | Dr-Linux | to defiene Holdon Music necessary to make entry with users extention in extenstions.conf ? |
16:16.36 | file | Dr-Linux: no. |
16:17.04 | sivana | does Nufone have other DIDs besides Michigan? |
16:17.12 | file | sivana: no. |
16:17.27 | Dr-Linux | file: can you tell me what to do, i'm wanna add a song in holdon music ? |
16:17.47 | yashax | Also, if I buy the phones on ebay, can then load the SIP on them myself? Is it pretty simple? Can you please point me to the right info source? |
16:17.52 | Dr-Linux | i put my song in muhmp3 file |
16:18.03 | file | Dr-Linux: then you have to set up musiconhold.conf... just uncomment default if it's not already uncommented |
16:18.13 | file | Dr-Linux: and make sure you have mpg123 0.59r |
16:18.16 | Dr-Linux | but i'm still listening music which not exist in muhmp3 file |
16:18.21 | sivana | yashax: I think you need to buy license or contract from Cisco if you buy on eBay |
16:18.25 | file | Dr-Linux: please note this has been documented in numerous places |
16:18.39 | file | yashax: as sivana said, you need a support contract to get the firmware to load onto it |
16:19.27 | file | Dr-Linux: there's already music included with asterisk for music-on-hold |
16:19.39 | yashax | so basically, if I make sure that SIP firmware is installed on them, then it is ok?! |
16:19.39 | file | Dr-Linux: you also have to restart (not just reload) asterisk for new music |
16:19.45 | *** join/#asterisk unixgeek (~unixgeek@216-220-234-197.exploremaine.com) |
16:20.26 | *** join/#asterisk easydone_ (~easydone@195-144-092-027.dyn.adsl.xs4all.be) |
16:20.26 | sivana | yashax: I think so |
16:20.37 | file | mmm food |
16:22.37 | yashax | thank you... |
16:24.11 | zoa | damin, Deal! |
16:24.17 | zoa | well still need to check for that hotel room though |
16:24.30 | zoa | as i dont feel like sleeping on the streets either |
16:24.31 | zoa | :) |
16:24.36 | file | Hyatt Sainte Claire, that's the new of her... Google and you shall find it |
16:24.42 | *** join/#asterisk ctooley ([U2FsdGVkX@199.89.146.18) |
16:24.42 | Dr-Linux | file: i'm installing mpg123 0.59r after extracting what should i do with "make" option "make dec" or what, i'm using Redhat ? |
16:25.08 | P-Chan | I've been having this problem for a while now: Executing Dial("SIP/xws103-df0f", "Zap/g1/BYEXTENSION") in new stack |
16:25.08 | P-Chan | Feb 13 02:16:58 NOTICE[6485]: app_dial.c:749 dial_exec: Unable to create channel of type 'Zap' |
16:25.20 | file | Dr-Linux: read it... cause I haven't done an install like that in awhile |
16:25.22 | Dr-Linux | [root@I2C-TEST mpg123-0.59r]# make |
16:25.22 | Dr-Linux | You must specify the system which you want to compile for: |
16:25.27 | file | DO NOT PASTE IT |
16:25.34 | Dr-Linux | ok |
16:25.44 | file | P-Chan: BYEXTENSION is deprecated, use ${EXTEN} - and do you have the group defined? |
16:26.24 | P-Chan | ooooh...I didn't know about that. Ok, where do I check if the groups defined? This config came from an old setup of asterisk |
16:26.44 | P-Chan | I just dumped it onto a new server (old one crapped out) |
16:27.50 | file | zapata.conf in /etc/asterisk |
16:28.00 | *** part/#asterisk sezuan (sezuan@port-212-202-57-119.dynamic.qsc.de) |
16:28.11 | P-Chan | I now get: |
16:28.14 | P-Chan | eb 13 02:24:55 NOTICE[6512]: app_dial.c:749 dial_exec: Unable to create channel of type 'Zap' |
16:28.14 | P-Chan | <PROTECTED> |
16:28.14 | P-Chan | <PROTECTED> |
16:28.14 | P-Chan | <PROTECTED> |
16:28.50 | file | are the drivers installed? do they load fine? is zaptel.conf configured? does asterisk spit out an error when chan_zap is loaded? what does zap show channels show? |
16:30.21 | P-Chan | [chan_zap.so] => (Zapata Telephony w/PRI) |
16:30.21 | P-Chan | <PROTECTED> |
16:30.44 | P-Chan | <PROTECTED> |
16:30.44 | P-Chan | <PROTECTED> |
16:30.44 | P-Chan | <PROTECTED> |
16:30.46 | P-Chan | ... |
16:30.52 | P-Chan | <PROTECTED> |
16:30.53 | file | try Zap/1/ instead |
16:31.05 | P-Chan | so remove the g? |
16:31.06 | P-Chan | ok |
16:31.11 | file | yeah, just try and see... |
16:32.30 | P-Chan | -- Executing Dial("SIP/xws103-4ac0", "Zap/1/4805772665") in new stack |
16:32.30 | P-Chan | Feb 13 02:29:27 NOTICE[6537]: app_dial.c:749 dial_exec: Unable to create channel of type 'Zap' |
16:32.44 | nirs | hey all |
16:32.49 | nirs | anybody home ? |
16:32.51 | Nukemizer | If I am using a TE110P and I have wcte11xp loaded as a module, and it is "Used by 0" indicated in lsmod, would this be fron no PRI channels in use or because I have not configured my PRI correctly ? |
16:33.28 | *** join/#asterisk nepon_ (user50@83.73.5.142.ip.tele2adsl.dk) |
16:34.02 | P-Chan | should I add the g back in? |
16:34.04 | file | Nukemizer: lsmod has no meaning, it's strictly for modules... means that it's not being used by another module |
16:34.09 | file | P-Chan: I'm not a zaptel person... |
16:34.28 | Nukemizer | thank you :) |
16:34.31 | file | P-Chan: sounds funky though |
16:34.56 | file | P-Chan: did you upgrade asterisk? |
16:35.10 | file | or just stick with an old one... |
16:35.39 | file | ANYWAY - that's it for me for now |
16:36.28 | ariel_ | ok folks anyone got a way to do an easy way to do rollover on sip devices. |
16:37.16 | dca[laptop] | anyone know if app_dbodbc will work with idodbc? |
16:38.33 | nirs | anyone has an idea what these mean ? |
16:38.35 | nirs | Feb 13 11:39:30 WARNING[31861]: PRI: !! Got S-frame while link down |
16:38.35 | nirs | Feb 13 11:41:33 WARNING[31861]: PRI: XXX Invalid Progress indicator value received: 04 |
16:38.35 | nirs | Feb 13 11:41:35 WARNING[31861]: PRI: XXX Invalid Progress indicator value received: 04 |
16:38.35 | nirs | Feb 13 11:42:46 WARNING[31861]: PRI: XXX Invalid Progress indicator value received: 04 |
16:39.36 | *** join/#asterisk jgaviria (~icom@201.245.167.66) |
16:41.08 | unixgeek | I have a couple of Cisco 12SP+ phones laying around. One has v3.0 firmware and tthe other is is like a v2.0 firmware. Anyone know where I can find the newer firmware image? |
16:44.36 | ManxPower | unixgeek, Uh, you havee to purchase the firmware from cisco. However, Cisco stoped making the 12SP+ years ago. |
16:44.43 | P-Chan | Any changes to the configuration files like BYEXTENSION now being ${BYEXTEN} from CVS from 12/10/04 until now that I need to know about? |
16:44.51 | P-Chan | configuration file changes I mean |
16:45.02 | P-Chan | until now = 1.0.5 |
16:45.04 | ManxPower | P-Chan, BYEXTEN is not used anymore. It's ${EXTEN} |
16:45.12 | P-Chan | oh |
16:45.55 | unixgeek | manxpower: Yes, I know that Cisco EOL these phones. I just figured that someone would have the firmware out there somewhere. |
16:46.43 | *** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net) |
16:48.13 | file[laptop] | kpfleming: beep beep |
16:48.25 | kpfleming | beep beep |
16:48.29 | ManxPower | unixgeek, *shrug* I'm sure people also have Microsoft Office ISOs too, but it's still copyright infringement. |
16:49.41 | file[laptop] | I swear I don't. |
16:49.44 | file[laptop] | HONEST |
16:53.23 | *** join/#asterisk Total-Net (~me@user-1121s6v.dsl.mindspring.com) |
16:55.50 | Nukemizer | This would be a question for any PRI guru's - IT appears that I have some sort of connection to my PBX with my PIR, because I will see the * reset the B channels and see an accknowledment on the PBX I will alsosee from, the PBX about 15 or so Tx and Rx looking at the D channel message information. Then all of a sudden i get a SYNC LOSS message. this is a cyle that just repeats. I have not yet been able to use any b channels for a test call |
16:56.06 | Nukemizer | this is a new turnup - so it has never been working |
16:56.22 | Nukemizer | any help is appriciated :( |
16:56.49 | *** join/#asterisk Wi_Fi (~OUT@c-24-127-12-85.we.client2.attbi.com) |
16:56.54 | Wi_Fi | hey guys |
16:57.24 | Wi_Fi | when setting up iaxy the user name is also the accountcode? |
16:57.26 | Corydon76-home | unixgeek: you could wait the requisite 95 years for Cisco's copyright to expire... |
16:57.28 | Dr-Linux | file: thanks i can hear my own songs on holdon music :) |
16:58.38 | Wi_Fi | [205] |
16:58.38 | Wi_Fi | username=main |
16:58.38 | Wi_Fi | type=friend |
16:58.38 | Wi_Fi | accountcode=iaxy |
16:58.38 | Wi_Fi | secret=line |
16:58.38 | Wi_Fi | qualify=no |
16:58.40 | Wi_Fi | port= |
16:58.42 | Wi_Fi | notransfer=yes |
16:58.51 | Total-Net | anyone know if http://asterisk.xvoip.com/ is comming back or not? |
17:00.47 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
17:03.14 | tangel | anyone know of a way to playback audio streams using *? |
17:03.32 | Qwell | like Playback()? |
17:03.42 | tangel | like network stream |
17:03.43 | tangel | s |
17:03.49 | *** join/#asterisk [alex] (~[alex]@201.133.98.147) |
17:03.53 | tangel | such as shoutcast or live audio |
17:04.00 | Total-Net | anyone know if http://asterisk.xvoip.com/ is comming back or not? |
17:04.00 | zoa | i know its possible |
17:04.04 | zoa | just dont know how |
17:04.11 | zoa | Total-Net: it seems no one knows |
17:04.15 | jgaviria | hi, i have a problem, i have a music on hold, it seems to work, but i cant hear anything, somebody could hlpme?, can i use music on hold with gsm? |
17:04.26 | tangel | well, i'm looking more for tips on tools.. i'm sure it could be done |
17:04.54 | tangel | does mpg123 support network streams? |
17:04.55 | file | jgaviria: make sure you use mpg123 0.59r |
17:05.28 | Total-Net | zoa: just checking.. |
17:05.45 | jgaviria | file: i think im using an old version, then this is the problem? |
17:05.47 | tangel | mpg123 man (8): Read filenames and/or URLs of MPEG audio streams |
17:05.49 | zoa | i think the mpg123 does it yes |
17:05.51 | tangel | sweet. |
17:05.55 | zoa | jgaviria: read the documentation |
17:06.00 | zoa | it says on millions of places |
17:06.04 | zoa | o.59r only |
17:06.13 | jgaviria | zoa: ok thanks a lot |
17:07.54 | tangel | should i not monitor quality on iaxtel? |
17:08.00 | tangel | it seems to bounce up/down a lot |
17:08.39 | tangel | what number do most people set their qualify parameter for their staple sip provider? |
17:10.09 | tangel | Feb 13 12:09:15 NOTICE[24585]: chan_iax2.c:6188 iax2_poke_noanswer: Peer 'iaxtel' is now UNREACHABLE! |
17:10.13 | tangel | Feb 13 12:09:25 NOTICE[24585]: chan_iax2.c:5668 socket_read: Peer 'iaxtel' is now REACHABLE! |
17:10.43 | Qwell | How low is quality? |
17:10.51 | zoa | its not quality |
17:10.55 | zoa | its qualiFy |
17:10.56 | tangel | qualify i mean |
17:11.01 | Qwell | right, heh |
17:11.08 | Qwell | typo on my part |
17:11.14 | tangel | which monitors latency quality sort of |
17:11.15 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
17:11.15 | zoa | well type on his part first :) |
17:11.33 | *** join/#asterisk Inv_arp (junya@adsl-3-246-242.mia.bellsouth.net) |
17:11.44 | Qwell | subliminal typo messages |
17:12.14 | tangel | i don't want to create unnecessary network trafffic |
17:12.35 | Inv_arp | ok my * is from CVS 9/14/04 .. werks perfect... should I upgrade... or "if not broke dont fix" |
17:12.56 | Qwell | Inv_arp: I'd say upgrade if you need some of the new changes |
17:13.02 | Qwell | otherwise, why bother? |
17:13.38 | tangel | so no advice on what qualify should be for iaxtel and my major sip provider? |
17:13.43 | tangel | or should i not set it at all? |
17:13.47 | *** join/#asterisk PBXtech (~nik@67.107.241.9.ptr.us.xo.net) |
17:14.08 | *** join/#asterisk markit (~marco@host119-245.pool8172.interbusiness.it) |
17:14.13 | Inv_arp | Qwell: ahh k .... iax is remarkable.... there need to be more iax providers |
17:14.36 | PBXtech | i have a quad FXO card and only port 1 id detecting a ring, how do i fix that. |
17:14.50 | markit | hi :) what is "slin" sound file format, and how can it be played/converted? |
17:15.09 | Wi_Fi | guys .. i need help |
17:15.30 | Wi_Fi | gettin no registration from IaxY |
17:15.30 | *** join/#asterisk jayden (~jayden@pcp02795302pcs.roylok01.mi.comcast.net) |
17:15.36 | PBXtech | slin is the lowest common denominator of codec |
17:16.12 | markit | PBXtech: what do you mean? how can I play a "greetings.slin" file? or how to convert with sox? |
17:16.44 | PBXtech | dont think * will play slin files. conver it.. the command is in the wiki |
17:16.55 | Inv_arp | Wi_Fi: err that doesnt explain your issue , what provider, paste iax.conf a pastbin sit etc etc..... |
17:17.42 | markit | PBXtech: I've read about slin in asterisk, don't have it by hand. Was about some anthm patch for avoiding mpg123, AFAIR |
17:18.15 | Wi_Fi | k |
17:18.34 | markit | but I don't know how to convert it, since sox seems not to have an explicit "slin" file type supported |
17:18.52 | *** join/#asterisk PTG123 (~PTG123@ip68-106-19-249.ph.ph.cox.net) |
17:19.15 | PBXtech | dunno |
17:19.37 | markit | PBXtech: ok, thanks anyway :) |
17:19.56 | PTG123 | hey kram |
17:20.13 | tzanger | werd to the goatherd |
17:20.36 | bkw_ | yo yo yo yo |
17:20.38 | bkw_ | its kram da man |
17:20.47 | PBXtech | i have a quad FXO card and only port 1 id detecting a ring, how do i fix that. |
17:21.08 | Wi_Fi | guys .. i need help http://www.pastebin.com/241427 |
17:21.12 | Wi_Fi | gettin no registration from IaxY |
17:21.13 | tzanger | PBXtech: all four ports are being detected? |
17:21.27 | PBXtech | yes outbound calls seem ok |
17:22.24 | tzanger | PBXtech: just for shits and giggles, swap the module that is working with another module... see if the ring detect stays with the port or with the module |
17:22.44 | PBXtech | good idea, ill have to do that on monday thou |
17:23.01 | tzanger | PBXtech: which port is ring being detected on anyway? |
17:23.06 | PBXtech | 1 |
17:23.16 | tzanger | and you *are* ringing port 2/3/4 ? |
17:23.22 | PBXtech | yup |
17:23.23 | tzanger | dumb question but you never know |
17:23.24 | tzanger | :-) |
17:23.39 | tzanger | before you swap ports, I'd swap lines in port 1 and 2 just to see |
17:23.45 | tzanger | and then swap modules |
17:23.50 | tzanger | just try to isolate what's going on |
17:24.01 | PBXtech | did that, all lines are ok. only port 1 sees the incoming call |
17:24.16 | tzanger | there's no protection on the TDM400P card base so it's not a matter of the card itself being obstinate but I'm just trying to narrow it down to a driver or module(s) problem |
17:24.35 | tzanger | ok so when you put line 1 in port 2 and ring line 1, port 2 does not see it |
17:24.42 | tzanger | but if line 1 is in port 1 and rings, port 1 sees it just fine |
17:25.44 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-40-134.cust.bezeqint.net) |
17:26.02 | Wi_Fi | Inv_arp? |
17:26.27 | Inv_arp | Wi_Fi: looking |
17:26.35 | Wi_Fi | thnx |
17:27.26 | PBXtech | [tzafrir]: yes |
17:30.57 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l01v-27-233.d4.club-internet.fr) |
17:35.41 | jgaviria | hi, i have a pri conected throw a TE110P, when i call to this, i got busy, but just from 2 hours ago, i think that the cable was disconnected, somebody knows how can i check it?, the server is in another city |
17:36.10 | tzanger | jgaviria: what's cat /proc/zap/1 say |
17:36.22 | tzanger | er /proc/zaptel/1 |
17:36.37 | tzanger | don't paste it here, just tell me if anything looks out of place |
17:36.44 | tzanger | as in anything about "red alarms" |
17:37.32 | jgaviria | tzanger: ok thanks, im gonna check it, right now i lost connection with the server |
17:37.42 | tzanger | jgaviria: sounds like you have bigger issues :-) |
17:38.18 | *** join/#asterisk fafnir (~hello@tdds-gw.Moscow.gldn.net) |
17:39.13 | zoa | KRAM! |
17:39.15 | zoa | BKW! |
17:39.23 | fafnir | OMFG! |
17:39.31 | tzanger | LOL!!!1!!11 |
17:41.42 | Total-Net | anyone know how to get into a telport-16? |
17:42.00 | jgaviria | tzanger: yes, some weird happens, if i reboot the server, the wcte11xp module doesnt load, but if i power down and then poweron again, the module load ok |
17:42.20 | tzanger | I never trust autoloading modules |
17:43.47 | jgaviria | tzanger: if a reboot the system, this doesnt load wcte11xp, even if i try to do it manually, always i need to poweroff the server and then the modules load ok |
17:44.09 | tzanger | jgaviria: odd |
17:44.16 | tzanger | I think you have bigger issues |
17:44.17 | Mw3 | P-Chan: any success ? |
17:44.35 | tzanger | jgaviria: try another card first (or another mobo, whichever you have around faster) and see if the problem goes away |
17:45.23 | jgaviria | tzanger ok, thanks. I got connection: |
17:45.37 | tzanger | just give me the top line |
17:45.37 | jgaviria | Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" HDB3/CCS/CRC4 RED |
17:45.44 | tzanger | red alarm = it can't see the other side |
17:45.46 | jgaviria | is a telco problem ok? |
17:45.58 | *** join/#asterisk netsurfer (~netsurfer@dreambox.myvnc.com) |
17:46.06 | tzanger | see if the telco has a red alarm |
17:46.10 | tzanger | if so, you've got a cabling problem |
17:46.35 | jgaviria | tzanger:thanks |
17:47.06 | Damin | Morning.. |
17:48.27 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
17:51.26 | file | |
17:51.28 | file | gah |
17:52.41 | *** join/#asterisk ToyMan (~konversat@user-12lcqq2.cable.mindspring.com) |
17:56.03 | *** join/#asterisk BoRiS (~boris@S01060040ca1e5b54.wp.shawcable.net) |
17:59.59 | nestAr | hrmm |
18:00.24 | nestAr | i'm having trouble with my SPA-3000 and my Polycom doing a native bridge |
18:00.30 | nestAr | anyone ever have trouble with that? |
18:01.17 | Essobi | is there a keypad sequence to get a 7960sip 7.X to reboot? |
18:01.18 | bjohnson | anyone here know logrotate & asterisk. I'm having trouble getting them to cooperate |
18:02.00 | Essobi | I thought it was **# or somesuch. |
18:02.49 | *** join/#asterisk mr_zack (zack@ppp-70-243-32-231.dsl.hstntx.swbell.net) |
18:05.11 | djin | Essobi, telnet is an option. |
18:05.12 | tangel | * doesn't register flash on my motorola pots phone... what can i do to correct? |
18:14.12 | Essobi | djin True true.. |
18:14.23 | Essobi | In some cases telnet isn't an option thou. |
18:14.33 | Essobi | Nat. |
18:14.43 | Moc | Essobi, yea * 6 settings |
18:14.46 | tangel | bjohnson, what's the problem? |
18:15.06 | Essobi | Moc at the same time? |
18:15.14 | Moc | i think so |
18:15.23 | Moc | i havent plug in my 7960 for a while |
18:15.41 | Essobi | man, it's like the vulcan death grip or something. |
18:15.48 | Moc | lol |
18:15.55 | tzanger | jgaviria: red alarm = I can't see the other side (i.e. you transmit RA) |
18:15.55 | Essobi | But thanks that worked |
18:16.04 | tzanger | jgaviria: yellow alarm = other side can't see me (i.e. you receive YA) |
18:16.06 | Moc | Essobi, get Polycom phone instead, I made a feature to reboot it via the CLI |
18:16.14 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
18:16.14 | Essobi | Umm. |
18:16.15 | tzanger | there's a blue alarm too but I don't recall what that is for |
18:16.22 | Moc | maybe sip notify could be hacked to work with 7960 too but.. |
18:16.35 | Essobi | I'm not replacing 3K 7960's with polycoms. :) |
18:16.48 | Moc | your screwed then ;) |
18:16.52 | Essobi | Heh. |
18:16.57 | Nugget | heh |
18:17.02 | Essobi | Nah, the vulcan death grip worked. |
18:17.05 | tzanger | hmm |
18:17.07 | tzanger | RAI = yellow |
18:17.11 | tzanger | AIS = blue but I knew it as red |
18:17.16 | Nugget | I have a perl script that will reboot a cisco 7960 remotely |
18:17.33 | Essobi | Nugget telnet probably. |
18:17.44 | Essobi | I got some endpoints that are behind a nat thou. |
18:17.47 | Nugget | yeah |
18:18.04 | Essobi | Mmm. There should be a sip notify for reboot. I wonder if there is. |
18:18.29 | Nugget | one of my 7960s is flaky and locks up after a while, so I reboot it every night. |
18:18.41 | Essobi | I need to dig into the TFTPs and see if there's a way to disable |
18:18.47 | *** join/#asterisk IQ (~IQ@70-59-163-130.omah.qwest.net) |
18:18.47 | Essobi | Warranty? |
18:18.49 | Essobi | Heh. |
18:18.53 | Nugget | nah, I picked it up used off ebay |
18:18.56 | Essobi | I need to dig into the TFTPs and see if there's a way to disable URL dialing.. |
18:19.04 | Moc | Essobi, disable what ? |
18:19.04 | Essobi | s/TFTPs/configs |
18:19.09 | Essobi | URL dialing. |
18:19.10 | Moc | ha.. |
18:19.17 | tzanger | ahhh |
18:19.24 | Moc | Essobi, that another thing you wish had polycom ;) |
18:19.29 | Moc | xml config is so extensive |
18:19.35 | Essobi | *SHRUG* |
18:19.40 | Moc | ;) |
18:19.42 | Essobi | That's a good thing. (TM) |
18:19.53 | Moc | anyone got Nortel i2004 ? |
18:19.57 | tzanger | red alarm = local alarm (corrupt/bad signal) while blue alarm is actually what you transmit |
18:19.59 | Nugget | I just with cisco would make a sip phone with a backlit display. I want one in my bedroom that would be visible at night |
18:20.01 | bjohnson | http://www.voip-info.org/tiki-index.php?page=logrotate |
18:20.05 | Essobi | Woetel? Oh NO! |
18:20.06 | tzanger | so red/blue are almost identica |
18:20.08 | Dr-Linux | whats difference between JVM and JDK ? |
18:20.18 | Essobi | Mmm. |
18:20.19 | Nugget | JVM is the virtual machine you need to run code. |
18:20.27 | Nugget | JDK is the developer's kit you need to write code |
18:20.32 | shido6 | la la la |
18:20.35 | Moc | Essobi, it will be soon * compatible ;) |
18:20.49 | Essobi | I'd almost consider pulling a graveyard and seeing how hard a backlight mod would be. |
18:20.49 | ctooley | Nugget, Polycom and Cisco work on their phones together, if one would do it they both would have it. |
18:21.05 | Essobi | For all you haxors out there. |
18:21.09 | Moc | ctooley, not really.. |
18:21.34 | Essobi | You ever talk to Cisco before? |
18:21.44 | Moc | cisco use polycom mic/speaker technology |
18:21.46 | tzanger | Essobi: several times, yes :-) |
18:21.50 | Moc | but that about it |
18:21.56 | Essobi | "have" usually isn't in their vocabulary, unless they are talking to you about licensing. ;) |
18:22.46 | Moc | polycom is missing a IP console and a wireless phone (and a ata too) |
18:22.58 | tzanger | ahhh there is an abort halt command |
18:23.12 | Moc | hehe yes' |
18:23.21 | Essobi | Does polycom push any feature controllers? If no, it's in their best interest to make their phones compat with as much as possible. Where as in cisco's case.. I'm surprised the 60's support anything other then SCCP. |
18:23.22 | Wi_Fi | Inv_arp any good? |
18:28.36 | Essobi | Hhhmm.. I got a neet idea.. |
18:29.05 | Darien | can anyone suggest any prerecorded standard prompts/ |
18:29.06 | Darien | ? |
18:29.21 | tzanger | Darien: for what |
18:29.31 | tzanger | asterisk has a ton of them already |
18:29.31 | Essobi | A program that sniffs (or proxies) sip and uses telnet to perform call testing and record the results.. |
18:29.35 | tzanger | and even more in asterisk-addons |
18:29.46 | Nugget | telnet? ewwww. |
18:29.58 | Wi_Fi | any Iaxy pros here |
18:30.01 | Essobi | I get people that say, oh when I dial blah blah blah 1/99 times it's busy, but it's not supposed to be. |
18:30.24 | Essobi | Well.. the only reason I suggest that is, I bang up the speaker phone, and record the RTP too. |
18:30.33 | tzanger | Wi_Fi: never used one, sorry |
18:31.56 | Essobi | Hmm.. anyone know of a cli SIP client that can record RTP steams? That'd be just as easy. ;) |
18:32.29 | tzanger | Essobi: what's wrong with tcpdump or tethereal? |
18:32.39 | Essobi | But the shock value of a wall of 7960s mass dialing would be pretty damn neat too thou. Heh. |
18:33.11 | Essobi | I need to fire of 30 dials at a time.. you want to wade through that shite? |
18:33.29 | tzanger | Essobi: ? I'm not sure I understand |
18:33.41 | tzanger | what is your magical program going to do differently? |
18:34.19 | Essobi | It's mostly for a testing a suite. I want to be able to pickup, 30 exts and dial. |
18:34.32 | Essobi | and watch for sip results.. like reorder or whatnot. |
18:35.04 | tzanger | Essobi: tcpdump -s0 -w somefile with your filter and then a perl script which sorts through the recorded stream would work quite well |
18:35.10 | tzanger | I think |
18:35.13 | Essobi | Baah, typos and misplaced comma. I'm tired. |
18:35.14 | tzanger | I may be misunderstanding though |
18:35.22 | Essobi | Automated dialing. |
18:35.59 | Essobi | I won't have 30 people sitting on 7960s.. So either an automated 7960 telnet scripts, or just a CLI sip program would fit the bill. |
18:36.18 | bjohnson | http://www.voip-info.org/tiki-index.php?page=Asterisk+user+authentication |
18:37.34 | tzanger | Essobi: any reason why you need to use physical phones then? |
18:38.19 | Essobi | No, other then I have them ;) |
18:38.28 | Essobi | That's why I said a CLI sip program. |
18:38.44 | tzanger | ... why not just use .call files? I am not getting why these phones need to be used |
18:39.02 | Essobi | Well.. Maybe dual use as an automated 7960 testing program. We have people that return phones that are perfectly fine. |
18:39.28 | tzanger | Essobi: ok but if nobody's at the phone how are you going to test them (display, keypad, ear&mouthpiece? |
18:39.36 | Essobi | I need the calls to originate outside of the * box.. I guess I could setup another one. |
18:40.05 | Essobi | ehh |
18:41.25 | Wi_Fi | yeghaaaa |
18:41.28 | Wi_Fi | got it workin |
18:41.37 | Wi_Fi | now i get now incoming from pstn |
18:41.38 | *** join/#asterisk lyroy (~lyroy@modemcable117.123-202-24.mc.videotron.ca) |
18:41.53 | Wi_Fi | i can dial out but no incoming |
18:42.02 | tzanger | kram: got a moment or two to comment on iax bridge optimization? |
18:42.19 | kram | tzanger: sure, it's basically obsolete. |
18:42.35 | *** join/#asterisk denon (denon@synapse.subneural.net) |
18:42.35 | *** mode/#asterisk [+o denon] by ChanServ |
18:43.14 | Essobi | lol |
18:43.30 | tzanger | ok so commenting out #define BRIDGE_OPTIMIZATION won't cause odd problems |
18:43.53 | tzanger | I've been testing it with that #define commented out and it seems fine but I dind't know if there were any subtle problems that might crop in |
18:43.57 | tzanger | er up |
18:43.58 | Essobi | Kram's the man. |
18:44.38 | Essobi | kram We ever gonig to have a G729 binary for FreeBSD? |
18:44.54 | Essobi | s/gonig/going |
18:45.16 | tzanger | kram: it's the calculate_fakestamps() that is causing those fucked-up IAX2 timestamps... |
18:45.36 | tzanger | basically it's calculating that the fakestamp is earlier than the last sent packet so it sets it to lastsent+1 |
18:47.02 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@d01m-12-200.d4.club-internet.fr) |
18:48.42 | tzanger | kram: I haven't been able to figure out what BRIDGE_OPTIMIZATION does to optimize anything, as it seems to only fuck with the timestamps |
18:48.47 | *** join/#asterisk zd_eyez (~zd_eyez@Ottawa-HSE-ppp259713.sympatico.ca) |
18:50.49 | jgaviria | tzanger: thanks, another question, when i have an incoming call, this have a did 000-999, then in the dial plan i need to put exten=000,... exten=001,... exten=999.. this is a lot of lines, there exists another way to do that?, i probed XXX, but it doesnt work |
18:51.10 | tzanger | jgaviria: _XXX will match any 3 digit number not already matched |
18:51.18 | tzanger | for example |
18:51.21 | tzanger | what I have in my own dialplan |
18:51.46 | tzanger | <PROTECTED> |
18:51.46 | tzanger | <PROTECTED> |
18:51.46 | tzanger | <PROTECTED> |
18:51.46 | tzanger | <PROTECTED> |
18:51.46 | tzanger | <PROTECTED> |
18:52.00 | tzanger | anything I want to specifically match I have before that entry |
18:52.12 | tzanger | so _29220XX matches the rest of my DIDs that aren't in use |
18:52.21 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
18:52.39 | jgaviria | tzanger: THANKS a lot!!! |
18:53.11 | tzanger | with our Bell Canada PRI it's neat -- if you do not accept a call Bell will play back the "not in service" message for you |
18:53.19 | tzanger | so I can make any DID "disappear" by not matching it |
18:53.29 | Wi_Fi | damn |
18:53.41 | Wi_Fi | zap is not being built on incoming |
18:53.51 | tzanger | huh? |
18:54.16 | Wi_Fi | when outside call incoming to pstn it just rings and rings |
18:54.28 | Wi_Fi | no autoattendant |
18:54.56 | tzanger | Wi_Fi: do you have an 's' exten in your zap context? |
18:55.17 | file | Windows, she is degrading |
18:55.18 | file | I must... restart |
18:56.17 | *** part/#asterisk P-Chan (~pchan@68.142.66.200) |
18:57.23 | _Vile | blah blah blah |
18:58.46 | *** join/#asterisk pkoegel (~pkoegel@68.142.66.200) |
18:58.52 | Wi_Fi | chekin tzanger |
18:59.27 | Wi_Fi | [from-pstn] |
18:59.27 | Wi_Fi | include => ext-did |
18:59.27 | Wi_Fi | exten => s,1,GotoIf($[${IN_OVERRIDE} = forcereghours]?from-pstn-reghours,s,1:) |
18:59.27 | Wi_Fi | exten => s,2,GotoIf($[${IN_OVERRIDE} = forceafthours]?from-pstn-afthours,s,1:) |
18:59.27 | Wi_Fi | exten => s,3,GotoIfTime(${REGTIME}|${REGDAYS}|*|*?from-pstn-reghours,s,1:) |
18:59.27 | Wi_Fi | exten => s,4,Goto(from-pstn-afthours,s,1) |
18:59.30 | Wi_Fi | crap |
18:59.33 | Wi_Fi | sorry guys |
19:01.31 | Moc | hi kram |
19:02.05 | tzanger | Wi_Fi: simplify that temporarily |
19:02.15 | tzanger | exten => s,1,NoOp(got a call) |
19:02.28 | tzanger | exten => s,2,Answer |
19:02.33 | tzanger | exten => s,3,Echo |
19:02.40 | tzanger | exten => s,4,Hangup |
19:02.55 | Wi_Fi | will do |
19:06.46 | Wi_Fi | nuthin |
19:06.48 | Wi_Fi | same thing |
19:06.56 | Wi_Fi | nothing in console either |
19:07.34 | tzanger | then it is not seeing the ring at all |
19:07.39 | Wi_Fi | yah |
19:07.43 | Wi_Fi | i think |
19:07.55 | tzanger | someone else said they weren't seeing ring on anything but port 1 on a TDM404P |
19:08.02 | Wi_Fi | zapata conf then? |
19:08.35 | tzanger | perhaps but theother person wasn't an asterisk newb |
19:08.44 | tzanger | not sure if you are or not :-) |
19:12.04 | pkoegel | I'am new > problem getting channels up |
19:13.23 | Wi_Fi | im soso |
19:13.24 | Wi_Fi | hehe |
19:14.02 | *** join/#asterisk waddy (waddy@66.90.92.190) |
19:17.35 | Moc | oh well, gota get back to reality, gota go do the grocery |
19:21.25 | Silik0n | *YAWN* |
19:23.25 | *** part/#asterisk gbs (~GBS@adsl-5063187d.monradsl.monornet.hu) |
19:27.04 | PTG123 | o did |
19:31.11 | *** join/#asterisk mrempire (~user1@h71032.upc-h.chello.nl) |
19:36.12 | *** join/#asterisk Himeko (~himeko@S01060040ca128fc3.ed.shawcable.net) |
19:38.37 | *** join/#asterisk Juggie (agony@CPE00c049d9f271-CM014270110981.cpe.net.cable.rogers.com) |
19:39.20 | mrempire | Does anyone know why i get the following when i dial 551: Requested device 'ttyI0/0235407616' does not exist |
19:41.10 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@d01m-21-177.d4.club-internet.fr) |
19:43.59 | djin | mrempire, you linked a 'Haarlem' number to 551 in you extensions.conf? |
19:44.00 | Darien | is AMP the best (free) management interface out there? |
19:44.22 | Darien | neither do I |
19:44.35 | Darien | but it's better than editing config files every time I need to make a change |
19:44.38 | mrempire | yes, i have posted it to pastebin http://pastebin.ca/5768 |
19:44.51 | djin | management interfaces like AMP take full control of your extensions.cong |
19:44.54 | djin | .conf. |
19:46.00 | Darien | well I'm interested in playing around with Asterisk and getting it working, but I don't want to have to constantly edit the config files and restart |
19:46.13 | djin | restart? |
19:46.28 | Darien | hmm |
19:46.40 | Darien | if I edit a config file, does it take effect immediately? |
19:46.49 | Darien | or is it just a simple HUP ? |
19:47.05 | djin | hopefully you know the command 'reload' from the CLI? |
19:47.15 | Darien | I know literally nothing |
19:47.21 | djin | or 'asterisk -rx reload' from the command. |
19:47.25 | Darien | ah |
19:47.37 | djin | it reload the config without restart. |
19:47.41 | Darien | alright |
19:47.50 | Darien | I'd still like a graphical interface off some kind |
19:47.52 | djin | damn, you're life you got way easier ;) |
19:47.53 | Darien | personal preference |
19:47.57 | Darien | haha yeah |
19:48.15 | djin | abandon asterisk, go for windows >:) |
19:48.25 | Darien | don't tempt me :p |
19:48.38 | Darien | it seems like every GUI out there has something wrong with it though |
19:48.54 | djin | invest some time, you'll be rewarded. |
19:49.09 | mrempire | I love asterisk now ;) |
19:49.22 | Darien | I do need a billing solution though |
19:49.40 | djin | mrempire, checking you config now. |
19:50.04 | djin | exten => 551,1,Dial(Modem/ttyI0/0235407616:0650506652) |
19:50.06 | *** join/#asterisk Nohair (~jt@srscomp.demon.co.uk) |
19:50.15 | mrempire | thanks djin I also posted my modem.conf |
19:50.33 | djin | I don't know much about modem, but something wrong there. |
19:50.46 | Nohair | Hi any 1 know why the startup script for redhat fails it moans about ixj what ever that is |
19:51.11 | djin | Nohair, please be more specific. |
19:51.35 | mrempire | djin: do you mean in the extensions.conf |
19:52.08 | *** join/#asterisk Mother__ (~m@53.Red-217-126-93.pooles.rima-tde.net) |
19:52.12 | Mother__ | hi all |
19:52.36 | djin | yes mrempire,you're connecting to a modem Asterisk doesn't know. |
19:53.11 | Mother__ | I am trying to route a couple of PSTN channels via IAX2 to a remote *, can I set it up to call more than one remote extension? |
19:53.31 | Mother__ | i.e. I want that when a PSTN call comes in, five SIP extensions on the remote * server ring |
19:53.35 | mrempire | I allso configured it in modem.conf |
19:54.05 | Wi_Fi | oh big crap |
19:54.07 | djin | Mother, sure, route multiple extensions to the remote * and then divide them from there. |
19:54.14 | mrempire | So shouldn't asterisk know about it |
19:54.28 | Wi_Fi | my internal ext cant dial out |
19:54.38 | djin | mrempire, could you pastebin your modem.conf? |
19:55.21 | Mother__ | djin: thanks, I'll do that - so it would be such as local zap/1 -> IAX2 Context on remote * -> exten => Dial(whatever extensions) |
19:55.24 | mrempire | djin: http://pastebin.ca/5767 |
19:55.31 | Wi_Fi | getting all circuits are busy now |
19:55.35 | Nohair | Djin There is a startup script in the asterisk install directory that you place into etc/init.d this reports on start up the following : - Module ixj not found |
19:55.50 | ariel_ | Mother__, when the call comes into your remote box. put an exten that rings the 5 sip exten => s,1,Dial(Sip/1&Sip/2&Sip/3) |
19:56.14 | djin | Nohair, you can use ' make config' to arrange startup scripts. |
19:56.39 | Mother__ | ariel: thanks, that's what I thought, I have a 30 minute walk between sites, so I wanted to have a confirmation :) |
19:57.17 | ariel_ | Mother__, ssh with putty works like being there... |
19:57.29 | djin | Mrempire, you're from Haarlem NL? |
19:57.40 | mrempire | Yes |
19:57.47 | djin | Damn, me2. |
19:58.18 | mrempire | djin: I'm tring to phone my mobile from home |
19:58.45 | mrempire | djin: hahahahahaha close by?? |
19:59.20 | Wi_Fi | wtf all outbound to pstn are all busy |
19:59.23 | Wi_Fi | grrrr |
19:59.31 | mrempire | djin: schalkwijk/europawijk |
20:00.00 | djin | Transvaal/Haarlem-Noord. |
20:00.12 | djin | did modem ever work? |
20:00.38 | Nohair | djin what is ixj |
20:00.42 | mrempire | I got the ttyI0 devices, i know what the problem was |
20:00.53 | flewid | sup |
20:01.07 | mrempire | in kernel 2.6 the ttyI0 devices got depriciaated |
20:01.09 | flewid | anyone notice in the newest HEAD that when calling *xx with zap channels it brings you to another dialtone |
20:01.15 | flewid | (i have *99 and *98 for checking voicemail) |
20:01.31 | flewid | when using sip or iax, works fine, but zap just delivers another stutter tone after completing dialed |
20:01.58 | flewid | and i can't dial anything while in that second dialtone |
20:02.01 | djin | Nohair, I think it's support for a Quicknet Internet Linejack, an ISA Phone board. |
20:02.13 | djin | Actually, I'm pretty sure. |
20:03.13 | djin | mrempire, kernel 2.6 uses udev to lighten the /dev directory. I only have bad experience with this and only ron 2.4 on production for now. |
20:03.49 | mrempire | djin: that's what I have done, for the third time installed linux in a week |
20:04.33 | djin | mrempire, did any modem extension work? |
20:04.45 | Mother__ | ariel: I know, but I have to go and set it up first :D |
20:04.58 | Mother__ | and I want to get as much stuff right the first time as I can |
20:05.05 | mrempire | I don't know how to test? |
20:05.06 | Mother__ | thanks for your help :) |
20:06.44 | mrempire | djin: at first i thought, lets test by dialing out |
20:08.59 | djin | mrempire, it looks ok from here :? |
20:09.48 | mrempire | djin: the modem.conf and extensions.conf? |
20:11.46 | djin | actually, both . . . |
20:12.28 | mrempire | djin: oooh my god ;( nooooooooooooo |
20:13.01 | mrempire | It's like hitting my head against a concrete wall |
20:14.38 | djin | You ISDN modem is working properly on other applications? |
20:14.56 | mrempire | I have no other applications ;( |
20:15.15 | djin | ok. |
20:15.23 | mrempire | Do you know with what i can test it? |
20:15.39 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
20:16.32 | djin | no, but did you fully restart aterisk (stop now -> safe_asterisk)? |
20:16.33 | mrempire | I have two isdn cards: an eicon and an hfs one, both give the same results |
20:17.15 | *** join/#asterisk Darwin35 (~Darin@c-24-3-241-22.client.comcast.net) |
20:18.34 | mrempire | djin: safe_asterisk , where is my CLI now, how can i stop asterisk now |
20:19.21 | djin | asterisk -r |
20:19.24 | Darwin35 | that or on tty9 |
20:19.37 | Darwin35 | safe asterisk puts the cli on tty9 |
20:19.54 | Darwin35 | unless you edited the safeasterisk file |
20:20.07 | mrempire | Thanks, I haven't edited it |
20:20.31 | djin | any changes? |
20:20.50 | mrempire | no, same result |
20:21.23 | mrempire | ;((((( |
20:22.07 | flewid | w t f |
20:22.18 | flewid | i think one of my fxs daughterboards might have died |
20:22.25 | nirs | hey all |
20:22.30 | nirs | how are we all doing ? |
20:22.41 | *** join/#asterisk Ad-Hoc (~ad-hoc@ppp4-adsl-193.ath.forthnet.gr) |
20:22.46 | Darien | unwell |
20:22.47 | Ad-Hoc | hello ppl |
20:22.49 | djin | mrempire, I'm out of ideas for now. |
20:22.49 | Darien | or at least, clueless |
20:23.15 | nirs | what's the matter darien ? |
20:23.15 | nirs | maybe I can help out |
20:23.18 | djin | I only have CAPI experience on ISDN. |
20:23.41 | nirs | btw, has anyone used the inAccessNetworks chan_oh323 ? |
20:25.28 | nirs | silence, dead air, just like an H323 VoIP call in here tonight |
20:25.31 | nirs | :)) |
20:27.15 | mrempire | djin:thanks |
20:27.31 | mrempire | whicj isdn card do you have |
20:29.01 | djin | Eicon Server boards. |
20:31.42 | *** join/#asterisk zd_eyez (~zd_eyez@Ottawa-HSE-ppp257967.sympatico.ca) |
20:32.18 | Luke-Jr | Is it possible to wait one second, but if the person starts dialing, let them finish? |
20:33.45 | flewid | hmm |
20:34.01 | flewid | okay so on one zap channel if i dial anything it just presents another dialtone |
20:34.07 | flewid | but other zap channels work as expected |
20:34.12 | flewid | configuration is the same |
20:34.17 | flewid | safe to assume the fxs card died? |
20:35.45 | mrempire | eicon server boards?? cooooool |
20:38.05 | mrempire | how can I disconnect from the CLI |
20:39.32 | *** join/#asterisk tsetane (~tsetane@pppoecl74115.minlos.no) |
20:45.28 | *** join/#asterisk kietlak (~kietlak@11-mo3-6.acn.waw.pl) |
20:51.25 | djin | mrempire, 'quit' |
20:52.17 | *** join/#asterisk jetscreamer (~jetscream@adsl-64-219-216-41.dsl.hstntx.swbell.net) |
20:52.33 | djin | Yeah Eicon server are pretty cool. Got 5xPRI, 1x8BRI and 3x1BRI, so I guess I'm pretty addicted to them ;) |
20:55.14 | mrempire | djin: i'm drooling |
20:56.18 | djin | most of them running other software though. |
20:58.35 | mrempire | djin: do you know some thing about capi? If I want to use capi for my eicon home isdn card |
21:00.20 | Luke-Jr | my * works when I dial from a softphone, but how can I get it to recognise DTMF? |
21:02.06 | djin | mrempire, yes. |
21:02.19 | djin | Got chan_capi running flawlessly. |
21:02.58 | Damin | I think I might be screwed.. |
21:03.12 | Damin | I can't get a hotel room for VON.. |
21:03.24 | Damin | Anyone intersted in sharing a room? |
21:03.36 | mrempire | djin: do i only have to compile chan_capi or do I allso have to load the driver or so?? |
21:03.39 | flewid | hmm |
21:03.46 | djin | Wow Damin, are you able to pay for access? |
21:03.51 | *** join/#asterisk afrosheen (afrosheen@txprotoa8.august.net) |
21:03.59 | afrosheen | hello |
21:04.11 | afrosheen | !seen bentley |
21:04.11 | djin | mrempire, chan_capi is quite simple to run. |
21:04.14 | WifiFred | bentley was last seen in #asterisk 4 days, 27 minutes and 10 seconds ago saying: kiso79: you'll get good support using the AMP mail-list or forums |
21:04.19 | Damin | djin: What do you mean? Access to VON? Sure.. |
21:04.26 | djin | as long as you have CAPI. |
21:04.36 | Damin | djin: It's not a question of cost.. it is a question of room availability.. |
21:04.39 | djin | Damin, I saw the prices and . . . |
21:04.46 | mrempire | hahaha there comes the rabbit out of the hat ;) |
21:05.06 | flewid | damin |
21:05.10 | flewid | are there no rooms left? |
21:05.13 | djin | what rabbit? where? kill, kill! |
21:05.20 | mrempire | So I need CAPI, is that just an rpm?? |
21:05.20 | flewid | i priced out my tickets and hotel about a month ago and gave it to my boss |
21:05.23 | Damin | flewid: Not anywhere close.. |
21:05.28 | flewid | but he's lallygagging around and hasn't ordered them yet |
21:05.29 | flewid | oh well |
21:05.33 | flewid | looks like he has to rent me a car now too |
21:05.34 | flewid | the fuck |
21:05.35 | flewid | okay thanks |
21:06.03 | *** join/#asterisk yashax (~yasha_x@69.15.218.218) |
21:06.55 | *** join/#asterisk Ogun (~kvirc@h127n2fls34o865.telia.com) |
21:07.27 | djin | mrempire, for Server boards and with the right kernel, yes. |
21:08.08 | mrempire | djin:kernel 2.4 , I don't have server board just plain home use |
21:08.20 | mrempire | djin: is that oke? |
21:08.48 | djin | mrempire, not sure. |
21:09.03 | Luke-Jr | my * works when I dial from a softphone, but how can I get it to recognise DTMF? |
21:11.34 | porkchop | * and the softphone have to agree on a way to carry DTMF info between them |
21:11.48 | porkchop | options are inband, info or rfc2something |
21:11.59 | *** join/#asterisk wasim (~wasim@203.81.213.118) |
21:12.00 | porkchop | keep it simple, use info |
21:12.24 | Luke-Jr | I think I have that setup properly |
21:12.32 | Luke-Jr | I just don't know how to configure * to listen for it |
21:12.56 | porkchop | it listens |
21:14.28 | afrosheen | anyone using amp |
21:14.32 | Luke-Jr | If I have a Wait(1), can I tell it to wait longer if the user starts dialing? |
21:15.45 | Luke-Jr | porkchop: also, voip-info says 'info' doesn't work w/ voicemail... |
21:16.02 | porkchop | hrm. thought I used it vm before. |
21:16.41 | *** join/#asterisk tecnico (~tecnico@user-24-236-123-31.knology.net) |
21:16.48 | Total-Net | anyone know how to get into a telport-16? |
21:17.10 | porkchop | it waits until timeout airc. |
21:18.19 | Luke-Jr | I'm trying to use 'inband' but nothing happens |
21:18.37 | porkchop | is your phone set to it as well as its entry in sip? |
21:19.02 | Luke-Jr | err... forgot to define the DTMF stuff in extensions |
21:19.20 | Luke-Jr | if exten 2900 is what has the menu, how do I handle menu options? |
21:20.19 | *** join/#asterisk SeaForth (~SeaForth@c-24-1-126-202.client.comcast.net) |
21:21.25 | porkchop | have 2900 Goto() another context. That context can then have extensions that corospond to your options |
21:22.21 | *** join/#asterisk Mother__ (~m@53.Red-217-126-93.pooles.rima-tde.net) |
21:22.35 | *** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com) |
21:23.13 | *** join/#asterisk expressfone1 (~expressfo@62-15-97-163.inversas.jazztel.es) |
21:23.22 | expressfone1 | Hi all |
21:23.28 | *** join/#asterisk zoa (zoa@82.103.76.147) |
21:23.56 | expressfone1 | have capacity for sell termination to Cuba, +53 |
21:24.56 | *** part/#asterisk djin (~djin@gridfox.xs4all.nl) |
21:29.22 | Luke-Jr | porkchop: it doesn't see DTMF while doing Playback, only after everything plays :/ |
21:30.58 | *** join/#asterisk ]data[ (~data@213.221.191.134) |
21:31.14 | porkchop | oh |
21:31.18 | porkchop | Background() |
21:31.22 | porkchop | not Playback() |
21:32.40 | Luke-Jr | ah |
21:33.06 | Luke-Jr | what about during Wait |
21:33.17 | porkchop | I don't use Wait anywhere. |
21:33.19 | denon | use background with a silent file |
21:33.22 | denon | instead of wait |
21:33.32 | porkchop | thats a reasonable idea too |
21:33.56 | denon | I think the asterisk-sounds even has some n-seconds files too |
21:34.05 | denon | otherwise its easy to make some |
21:34.18 | porkchop | or just get the jepordy music |
21:34.20 | denon | yeah, silence/1.gsm |
21:34.23 | denon | yeah, silence/2.gsm |
21:34.24 | denon | etc |
21:34.32 | Luke-Jr | hrm |
21:34.33 | denon | 0-9 |
21:34.33 | Luke-Jr | weird |
21:35.00 | porkchop | I'd still get the jepordy music. |
21:35.18 | Luke-Jr | too bad that workaround doesn't work w/ eg 2.5 seconds |
21:35.34 | denon | sure its does |
21:35.34 | porkchop | make a 2.5 second file |
21:35.37 | denon | make a file 2.5 seconds long |
21:35.49 | denon | or better still, make a .5 second one, so you can use it in conjunction with others |
21:37.00 | Luke-Jr | still a bad workaround =p |
21:37.11 | denon | its an excellent workaround |
21:37.49 | Damin | ~seen jerjer |
21:37.51 | jbot | jerjer is currently on #asterisk. Has said a total of 60 messages. Is idling for 16h 51m 30s |
21:37.52 | porkchop | expically since you could have jepordy music with a sinlge mv command |
21:39.35 | Luke-Jr | denon: Is it so complex to support extens in Wait? |
21:40.05 | *** join/#asterisk dsmouse (~mouse@rrcs-24-199-146-243.midsouth.biz.rr.com) |
21:43.10 | ]data[ | jerjer idling? nothing changes then :-) |
21:43.55 | *** part/#asterisk kietlak (~kietlak@11-mo3-6.acn.waw.pl) |
21:45.06 | *** join/#asterisk stepcut (~redlion@ip68-107-21-88.sd.sd.cox.net) |
21:46.22 | *** join/#asterisk kietlak (~kietlak@11-mo3-6.acn.waw.pl) |
21:47.35 | Luke-Jr | hmm |
21:47.49 | Moc | what up |
21:47.50 | bjohnson | anyone know SPA 2000 dial plans? I want to take one number and dial a completely different number |
21:47.53 | Luke-Jr | I have # setup to play the menu again... the first time I press it, it works, but the second time, it hangs up |
21:52.30 | *** join/#asterisk file (~symlink@mctn1-1056.nb.aliant.net) |
21:53.37 | Luke-Jr | any ideas what is happening? |
21:54.49 | *** join/#asterisk Firestrm (F1r3570rm@S010600047577bccd.gv.shawcable.net) |
21:55.02 | Firestrm | hi :) |
21:55.30 | *** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net) |
21:56.25 | Firestrm | dsmouse, goodluck |
21:56.32 | dsmouse | :) |
21:57.03 | Firestrm | dsmouse, its not that hard to install.. setup is the hard part :) |
21:57.46 | Sedorox | si |
21:58.41 | bjohnson | nevermind .. found it easier than I thought .. (<replacethis:withthis>|nextpattern) |
22:00.32 | jets | Any o1 developers? |
22:01.28 | dsmouse | whats o1? |
22:01.50 | dsmouse | (obviously, not me but...) |
22:02.14 | *** join/#asterisk bly1 (~bly@adsl-68-75-197-196.dsl.akrnoh.ameritech.net) |
22:03.35 | bly1 | Hello all |
22:03.46 | bly1 | been checking google on this particular error. |
22:04.05 | bly1 | just compiled * on a new box and when I start *, I get the following |
22:04.07 | bly1 | monmp3thread: Request to schedule in the past?!?! |
22:04.13 | bly1 | about 3 times a second. |
22:05.22 | bly1 | fyi: i do not have any zaptel hardware in the box |
22:06.39 | *** join/#asterisk Ogun (~kvirc@h127n2fls34o865.telia.com) |
22:10.47 | nestAr | ugh.. echos with pots lines suck. |
22:16.04 | bly1 | just compiled * on a new box and when I start *, I get the following |
22:16.06 | bly1 | monmp3thread: Request to schedule in the past?!?! |
22:17.11 | dsmouse | bly1: what version of *, btw? |
22:17.32 | dsmouse | (oh yea, I'm *just* starting to learn about this) |
22:17.51 | bly1 | Asterisk 1.0.5, Copyright (C) 1999-2004 Digium. |
22:17.51 | bly1 | Written by Mark Spencer <markster@digium.com> |
22:18.19 | bly1 | fyi: i do not have any zaptel hardware in the box |
22:18.24 | *** join/#asterisk eivindtr (~Eivind@062016241059.customer.alfanett.no) |
22:18.43 | bly1 | root@pb:~# lsmod |
22:18.43 | bly1 | Module Size Used by |
22:18.43 | bly1 | ztdummy 2468 0 |
22:18.43 | bly1 | zaptel 218500 1 ztdummy |
22:18.43 | bly1 | crc_ccitt 1408 1 zaptel |
22:18.43 | bly1 | ipv6 183940 10 |
22:19.28 | dsmouse | bly1: did you try http://lists.digium.com/pipermail/asterisk-bsd/2004-June/000046.html ? |
22:20.53 | *** join/#asterisk cc (~cc@byte.fedora) |
22:20.54 | bly1 | not using FBSD, think it would still apply to me? |
22:20.58 | jets | is rasterisk in use yet? |
22:21.04 | dsmouse | or is that the other problem that's addressing...... |
22:21.27 | dsmouse | I'm not really sure :( Like I said, I'm just now compiling and installing it myself |
22:22.28 | dsmouse | I think pthreads are a libc thing, so it /might/ |
22:23.59 | *** join/#asterisk lagCisco (~Francisco@adsl-69-111-235-196.dsl.irvnca.pacbell.net) |
22:24.51 | *** join/#asterisk ZeroXeal (~zeroxeal@ool-44c166d7.dyn.optonline.net) |
22:26.27 | stepcut | argh, my freebsd install is so old, make does not understand -C :) |
22:29.43 | Nugget | FreeBSD moo.distributed.net 2.2.8-STABLE FreeBSD 2.2.8-STABLE #0: Wed Dec 9 18:06:05 GMT 1998 |
22:29.59 | Nugget | <PROTECTED> |
22:30.23 | dsmouse | how many proc's? |
22:30.27 | Nugget | just one |
22:30.32 | dsmouse | ah |
22:30.38 | Nugget | freebsd 2.2.8 can't do SMP, silly. |
22:31.09 | dsmouse | oh |
22:33.15 | Nugget | stepcut: install the gmake port. |
22:33.22 | dsmouse | uugh, |
22:35.59 | *** join/#asterisk cryz (cryzeck@sms.trangest.no) |
22:37.02 | *** part/#asterisk lagCisco (~Francisco@adsl-69-111-235-196.dsl.irvnca.pacbell.net) |
22:41.11 | stepcut | Nugget: I have gmake, but the makefile does not seem to work with it ... |
22:41.38 | cryz | Which ports does asterisk run on? |
22:42.54 | dsmouse | the example config listeds on udp: 2727 4569 5060, right? |
22:45.13 | *** join/#asterisk mikegrb (~michael@thegrebs.com) |
22:48.40 | stepcut | Am I correct in thinking that I can use asterisk with a service like teliax and run my own toll-free number for $5/month + $0.02/min, or am I misunderstanding something? |
22:49.59 | Mother__ | hi all |
22:51.32 | Nugget | stepcut: I'm not familiar with teliax, but you are correct. I have two toll-free numbers from nufone for $0/month and $0.02/min. |
22:52.10 | stepcut | Would you recommend nufone ? |
22:52.47 | stepcut | ie, do you find them reliable ? |
22:53.08 | Nugget | I haven't used them enough to have formed an opinion. |
22:53.58 | Luke-Jr | stepcut: I have Teliax |
22:54.44 | stepcut | Luke-Jr: Do you find the service to be reliable for non-critical business use ? |
22:55.17 | Luke-Jr | stepcut: So far, it's been choppy, but they seem to be understaffed at the moment so I'm hoping it will improve |
22:55.31 | Luke-Jr | stepcut: the $5/mo is only for a non-toll-free DID, note |
22:55.37 | stepcut | ah |
22:55.39 | Luke-Jr | I think toll-free cost more |
22:56.14 | Luke-Jr | Tech support is good tho :) |
22:56.23 | stepcut | *Optional $5/mth local or toll-free telephone number |
22:56.29 | Luke-Jr | They got me up and running in an hour |
22:56.47 | Luke-Jr | stepcut: really? maybe the difference is that toll-free is 2c/min incoming then |
22:56.55 | stepcut | teliax provides both incoming and outgoing then ? |
22:56.59 | Luke-Jr | stepcut: yes |
22:57.15 | stepcut | Luke-Jr: who do you use for outgoing ? |
22:57.17 | Luke-Jr | Voipjet is cheaper |
22:57.19 | *** join/#asterisk twisted (~twisted@twisted-active-pdpc.developer.and.friend.of.asterisk) |
22:57.19 | *** mode/#asterisk [+o twisted] by ChanServ |
22:57.20 | Luke-Jr | 1.3c/min |
22:57.30 | Mother__ | hmm this IAX call setup returned "Call rejected by x.x.x.x no authority found" on the originator |
22:58.24 | Luke-Jr | stepcut: Though Voipjet charges for 'toll-free' calls... I use FWD+SIPPhone for toll-free to bypass that |
22:58.35 | Luke-Jr | and Teliax has cheaper rates for India and Israel, IIRC |
22:59.16 | Luke-Jr | same here, but I did a bit of research on it anyway ;) |
22:59.17 | *** join/#asterisk heath__ (~heath@12-215-32-191.client.mchsi.com) |
22:59.44 | Mother__ | what could cause this "no authority" problem? |
23:00.11 | stepcut | Unfortunately sipphone won't connect to the toll free number I need to call right now :) I should try fwd... |
23:00.34 | *** join/#asterisk OzJames79 (~James@203.208.64.29) |
23:00.39 | Luke-Jr | SIPPhone only does 888 and 800 I think |
23:00.45 | Luke-Jr | which is why I fallback to FWD |
23:00.46 | OzJames79 | hi all anyone here using Fwdout? |
23:00.47 | stepcut | ahh. I need an 877 :) |
23:00.54 | Luke-Jr | FWD is *often* quite unusable quality, though |
23:01.15 | Qwell | I use the free trial at simpletelecom for my toll free calls ;/ |
23:01.25 | Luke-Jr | Qwell: trial? |
23:01.40 | Qwell | Luke-Jr: yeah, found a link off the wiki somewhere |
23:01.49 | Luke-Jr | time limited? |
23:02.00 | Qwell | It didn't say. I was wondering about that myself. |
23:02.01 | heath__ | having a problem using WaitForSilence app http://pastebin.ca/5774 |
23:02.08 | Qwell | I've been using it for like...almost 60 days now? |
23:02.28 | *** join/#asterisk GodThor (~ninja@212.110.67.6) |
23:04.12 | OzJames79 | can anyone help me with the Rejected connect attempt i am getting http://www.pastebin.com/241545 |
23:04.41 | GodThor | can i connect two quintums with asterisk machine? |
23:05.01 | *** join/#asterisk Mother_ (~m@53.Red-217-126-93.pooles.rima-tde.net) |
23:05.07 | Mother_ | grrrr |
23:05.39 | Mother_ | should I have friend or user in iax.conf? some list posts I've seen point in that direction |
23:05.48 | *** part/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
23:07.08 | *** join/#asterisk florz (nobody@odnb-d9baa501.pool.mediaWays.net) |
23:07.18 | stepcut | hoo-ray! I have a make that supports -C now :) |
23:07.29 | Qwell | stepcut: welcome to 1990 :p |
23:08.10 | stepcut | I am only running FreeBSD 4.7 -- it's not *THAT* old |
23:09.00 | stepcut | but I need to be compatible with my isp (pair.com) so I can build binary cgi programs, and they are only on 4.8 :-/ |
23:10.26 | *** part/#asterisk GodThor (~ninja@212.110.67.6) |
23:12.16 | Mother_ | well the authority thing is sorted now, but it's strange that on the originating * I can see the call progressing, but on the receiving * there is absolutely no trace of the dial attempt |
23:12.24 | stepcut | ===> Registering installation for asterisk-1.0.3_1 :) |
23:17.55 | Mother_ | no matter how I format the IAX2 Dial command on the originating *, it doesn't work |
23:18.26 | Mother_ | anyone can give me a hand with this IAX problem? |
23:18.38 | *** join/#asterisk harryvv (~comming@S010600055d210201.vs.shawcable.net) |
23:18.54 | Darwin35 | mother its just like dialing sip but you put iax2 as the protocal |
23:19.04 | Ogun | Mother_: Trying to link up two * to each other? |
23:19.09 | Darwin35 | the wiki shows you |
23:19.23 | Darwin35 | linking is easy |
23:19.26 | Mother_ | Darwin35: it's what I'm doing, but nothing |
23:19.30 | Darwin35 | I have 45 boxes linked |
23:19.31 | Mother_ | the link is OK |
23:19.39 | Mother_ | Ogun: yep |
23:19.40 | harryvv | 45 * boxes linked? |
23:19.46 | Darwin35 | yes |
23:19.49 | harryvv | nice |
23:19.50 | *** join/#asterisk iguy (~iguy@dsl093-197-234.mke1.dsl.speakeasy.net) |
23:19.53 | Darwin35 | all trunked via iax |
23:20.08 | harryvv | are you trying to create a voip network for paying end users? |
23:20.24 | OzJames79 | <PROTECTED> |
23:20.29 | Darwin35 | just me and friends linking our systems and testing things |
23:20.40 | harryvv | I see |
23:20.46 | Mother_ | I have exten => s,3,Dial(IAX2/user:pass@x.x.x.x/${EXTEN}/IAX) |
23:20.50 | Darwin35 | the plan is to setup a fully linked network in the next few months |
23:21.14 | Darwin35 | take the iax2 off the end and put the provider |
23:21.16 | Mother_ | ${EXTEN} becomes s as it's a PSTN inbound call |
23:21.19 | hellop | hello Darwin35 |
23:21.32 | Darwin35 | hello |
23:21.49 | Mother_ | Darwin35: the IAX at the end is the context in which the command is executed no? |
23:21.49 | hellop | * is working for me now! |
23:21.54 | harryvv | Darwin, cool locating a backbone carrier? |
23:21.57 | Darwin35 | good to hear |
23:22.02 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
23:22.02 | *** mode/#asterisk [+o bkw_] by ChanServ |
23:22.13 | Mother_ | in the receiving *, there is an [IAX] context in extensions.conf which should dial the local SIP phones |
23:22.53 | harryvv | bkw, would you know how many iax voip to pstn channels would thay allow on account "me" allow at one time? Want to try linking people in one call. |
23:23.07 | Ogun | Mother_: http://pastebin.ca/5777 |
23:23.45 | Ogun | Mother_: Config snippet that works for me over here |
23:24.01 | *** join/#asterisk _daver_ (~daver@ns1.tmok.com) |
23:25.03 | Mother_ | Ogun: thanks for that - thing is I'm trying to route a PSTN call inbound to the remote * over to the local * which has the SIP phones connected to it |
23:25.08 | Darwin35 | hold a min |
23:25.18 | Mother_ | what does the :2 after EXTEN do? |
23:26.53 | Qwell | Mother_: remove the first two digits from whats passed in |
23:26.54 | Darwin35 | it drops the first 2 digits dialed |
23:27.01 | Qwell | 918675309 becomes 8675309 |
23:27.02 | *** join/#asterisk mrproper_ (~mrproper_@61.95.55.242) |
23:27.13 | mrproper_ | anyone here using pwlib/openh323? |
23:27.14 | Mother_ | Qwell: thanks :) |
23:31.17 | *** part/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
23:33.18 | Mother_ | this is what I have in the extensions |
23:33.21 | Mother_ | http://pastebin.ca/5779 |
23:34.05 | Mother_ | the Wait(3) is to allow a fax machine to do it's tone detection |
23:37.56 | *** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com) |
23:42.17 | Mother_ | I still cannot get the local * to see the incoming call request |
23:43.51 | *** join/#asterisk rene- (~rene-@201.137.86.219) |
23:44.21 | *** part/#asterisk rene- (~rene-@201.137.86.219) |
23:45.29 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
23:45.29 | *** mode/#asterisk [+o bkw_] by ChanServ |
23:48.32 | Total-Net | anyone know how to get into a telport-16? |
23:51.43 | *** join/#asterisk sudoer (~sudoer@65.75.148.190) |
23:52.12 | Luke-Jr | http://pastebin.ca/5780 |
23:52.26 | Luke-Jr | Incoming calls are dropped if they dial something invalid, or if they dial * |
23:52.28 | Luke-Jr | any ideas? |
23:52.39 | sudoer | has anyone setup asterisk so you call into it, then it calls you back and setups disa/etc I cant find anythign on voip-info.org |
23:55.29 | Qwell | Luke-Jr: without looking at it, are you missing an i exten? |
23:56.03 | Luke-Jr | Qwell: shouldn't be |
23:57.00 | netsurfer | lo Qwell |
23:57.01 | Luke-Jr | It's also dropping when * is dialed, which is a valid exten |
23:57.14 | Qwell | netsurfer: anything yet? |
23:57.16 | sudoer | anyone? |
23:57.37 | netsurfer | no.. not a thing, and he wont send me tracking numbers, so i've reported him to paypal |
23:57.44 | Qwell | hmm |
23:58.17 | sudoer | would I need to use agi/code to do that or can I do this just with extensionx.conf? |
23:59.29 | netsurfer | Qwell - it just doesn't happen.. two parcels sent by royal mail special delivery getting lost - I have been using that service for years and never had one lost ever |
23:59.55 | Qwell | netsurfer: usps loses stuff all the time though, heh |