| 00:01.32 | sahafeez | in sip.conf can i use wild cards or do i have to define each extendion |
| 00:02.15 | *** join/#asterisk huslage (n=huslage@c-67-169-200-122.hsd1.or.comcast.net) |
| 00:02.47 | lancey`busy | sahafeez: i believe you can't use wildcards |
| 00:02.58 | sahafeez | that such |
| 00:02.59 | sahafeez | thanks |
| 00:03.32 | sahafeez | s/such/sucks |
| 00:04.28 | *** join/#asterisk Kyreeth (n=ashley@aquila.feathers.net) |
| 00:04.38 | *** join/#asterisk DarkShayD (n=dante@66.155.145.194) |
| 00:04.52 | DarkShayD | is there an asterisk@home channel? |
| 00:05.03 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 00:05.26 | DarkShayD | anyone familiar with TFTP here? |
| 00:09.56 | sahafeez | SJphone on the Mac seems to work well, btw. |
| 00:10.06 | sahafeez | DarkShayD: what do you need to knw |
| 00:10.18 | *** join/#asterisk KranZ (i=KranZ@music.queso.net) |
| 00:10.49 | KranZ | real quick, why is GotoIfTime not accurate to the time range |
| 00:11.02 | KranZ | it wont kick in till 2 mins after |
| 00:13.24 | DarkShayD | why doesn't TFTP work behind a firewall/nat? |
| 00:13.44 | sahafeez | because it is stateless UDP |
| 00:13.57 | sahafeez | i assume you mean thru |
| 00:14.00 | *** join/#asterisk cio (n=na@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 00:14.16 | DarkShayD | yeah |
| 00:14.16 | sahafeez | and it is a local broadcast only. |
| 00:14.21 | DarkShayD | what's that mean stateless UDP? |
| 00:14.30 | cio | Anyone here use a headset with a Polycom IP50x? I just plugged one in and the volume is way to low to be useable. I didn't see anything when I googled for an answer. Any suggestions? |
| 00:14.54 | sahafeez | TCP/IP Illistrated Vol. 1 by Steven if you want to know the full bit. |
| 00:15.01 | justinu | good suggestion |
| 00:15.09 | DarkShayD | sahafeez, so there's no way a phone outside your firewall can hit the TFTP server behind your corporate firewall? |
| 00:15.12 | sahafeez | it will not work. there is no way to get it to work. it is not designed to work |
| 00:15.20 | justinu | dark: no |
| 00:15.20 | sahafeez | Yes. |
| 00:15.25 | sahafeez | that is coorect |
| 00:15.25 | DarkShayD | I see. thx |
| 00:15.35 | sahafeez | unless you have a vpn tunnel on the same subnet |
| 00:15.39 | DarkShayD | right |
| 00:15.48 | sahafeez | most phones will do ftp |
| 00:15.56 | DarkShayD | cisco phones will do ftp? |
| 00:16.12 | DarkShayD | i mean, will cisco phones do ftp? |
| 00:17.06 | sahafeez | polycom will, however you have to hardcode the phone, or make sure the local dhcp server that the phone gets it ip from hast the boot server set to your remote ftp |
| 00:17.11 | sahafeez | i do not know about cisco |
| 00:17.22 | DarkShayD | i see |
| 00:17.52 | *** join/#asterisk oej (n=Olle@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 00:17.57 | sahafeez | google for ftp boot cisco model number |
| 00:18.53 | *** join/#asterisk theblue (n=theblue@pcp04402293pcs.nrockv01.md.comcast.net) |
| 00:18.56 | theblue | Hi all. |
| 00:19.22 | sahafeez | global seach and replace in VI? anyone - cannot remember. 1st find is :1,$s/bla/foo |
| 00:19.39 | justinu | yeah |
| 00:19.45 | justinu | or maybe :1,% |
| 00:19.48 | justinu | can't remember |
| 00:20.17 | lancey`busy | xm |
| 00:20.19 | sahafeez | no. $=bottom of file |
| 00:20.23 | justinu | ok |
| 00:20.26 | lancey`busy | 1,$s/oldstring/newstring/g |
| 00:20.31 | lancey`busy | or something like that? |
| 00:20.45 | justinu | g means it replaces all instances of oldstring on the line |
| 00:20.46 | justinu | so yeah |
| 00:21.09 | lancey`busy | sahafeez why don't you get some other editor easier to work with? :) |
| 00:21.22 | justinu | vi is easy |
| 00:21.23 | sahafeez | VI is the only editor! |
| 00:21.34 | sahafeez | i just do not remember that command |
| 00:21.39 | lancey`busy | :) |
| 00:21.47 | lancey`busy | well, if you say that... |
| 00:22.14 | lancey`busy | i remember the first time ending up in vi |
| 00:22.18 | sahafeez | and vi is installed on every unix install so you need to know it |
| 00:22.23 | lancey`busy | had a hard time exiting ;) |
| 00:22.54 | cio | <esc>:<x><enter> |
| 00:23.04 | lancey`busy | yeah, i know |
| 00:23.06 | lancey`busy | now |
| 00:23.07 | lancey`busy | :) |
| 00:23.08 | justinu | http://www.gnu.org/fun/jokes/ed.msg.html |
| 00:23.38 | sahafeez | ah. add /g at the end for global |
| 00:24.12 | lancey`busy | the greatest WYGIWYG editor |
| 00:24.13 | lancey`busy | LOL |
| 00:24.13 | lancey`busy | :) |
| 00:24.24 | justinu | Ed is for those who can *remember* what they are working on. If you |
| 00:24.24 | justinu | are an idiot, you should use Emacs. If you are an Emacs, you should |
| 00:24.24 | justinu | not be vi. If you use ED, you are on THE PATH TO REDEMPTION. THE |
| 00:24.24 | justinu | SO-CALLED "VISUAL" EDITORS HAVE BEEN PLACED HERE BY ED TO TEMPT THE |
| 00:24.24 | justinu | FAITHLESS. DO NOT GIVE IN!!! THE MIGHTY ED HAS SPOKEN!!! |
| 00:24.42 | theblue | Double You Tee Eff, Mate? |
| 00:24.58 | sahafeez | VI the editor of the beast |
| 00:25.18 | lancey`busy | or the beast of the editors |
| 00:25.18 | lancey`busy | ;) |
| 00:25.21 | lancey`busy | the evil one :) |
| 00:25.24 | justinu | When I use an editor, I don't want eight extra KILOBYTES of worthless |
| 00:25.25 | justinu | help screens and cursor positioning code! |
| 00:26.00 | theblue | VI = Vaguely Intelligible |
| 00:26.08 | theblue | EMACS = Eighty Megs And Constantly Swapping |
| 00:26.17 | theblue | ED = Extremely Dumb |
| 00:26.18 | cio | heh |
| 00:26.20 | lancey`busy | hehehe :)))) |
| 00:26.27 | justinu | Of course, on the system *I* administrate, vi is symlinked to ed. |
| 00:26.27 | justinu | Emacs has been replaced by a shell script which 1) Generates a syslog |
| 00:26.27 | justinu | message at level LOG_EMERG; 2) reduces the user's disk quota by 100K; |
| 00:26.27 | justinu | and 3) RUNS ED!!!!!! |
| 00:26.49 | theblue | XD. |
| 00:26.55 | cio | Nice. Bet your users love their computer system. |
| 00:27.04 | cio | wine notepad.exe |
| 00:27.05 | cio | heh |
| 00:27.10 | lancey`busy | :)))) |
| 00:28.07 | theblue | NANO = Not Another Null Operator |
| 00:28.24 | *** part/#asterisk dca[laptop] (n=dca[lapt@sta-206-168-218-202.rockynet.com) |
| 00:28.50 | lancey`busy | someone here using pf/altq to prioritize VoIP traffic? |
| 00:29.00 | cio | How's CVS feeling today? Worth installing? |
| 00:29.09 | lancey`busy | cio: kinda works |
| 00:29.16 | lancey`busy | despite making some people evil |
| 00:29.17 | justinu | heh |
| 00:29.17 | lancey`busy | :) |
| 00:29.23 | cio | kinda in the literal sense or kinda as in it's really broke? |
| 00:29.45 | lancey`busy | depends :) |
| 00:29.50 | cio | heh |
| 00:29.58 | lancey`busy | all the general stuff works flawless |
| 00:29.58 | cio | k, thanks. |
| 00:30.21 | lancey`busy | there's something wrong with the SIP URIs on outgoing channels |
| 00:30.27 | lancey`busy | which i'm trying to investigate |
| 00:30.30 | lancey`busy | *trying :) |
| 00:30.53 | lancey`busy | sometimes chan_sip doesn't honor the callerid |
| 00:30.58 | cio | ahh... |
| 00:31.01 | *** join/#asterisk nomazda (i=nyyankee@user-0c6tnqf.cable.mindspring.com) |
| 00:31.01 | lancey`busy | and doesn't set it as "fromuser" |
| 00:31.01 | cio | good luck! :) |
| 00:31.11 | lancey`busy | heh |
| 00:31.12 | lancey`busy | :) |
| 00:31.32 | cio | If I make asterisk on a 2.6 kernel, do I need to pass any args to make? i.e., make linux26? |
| 00:31.37 | justinu | is the 1.2 beta1 release better than CVS head for weird anomolies like that? |
| 00:31.41 | *** join/#asterisk Rez (i=lorez@freenode/staff/lorez) |
| 00:32.01 | lancey`busy | justinu: i don't have a clue |
| 00:32.10 | lancey`busy | in fact i don't know if this happens only with latest CVS |
| 00:32.18 | lancey`busy | just needed it now, and found it out |
| 00:32.23 | justinu | hm, i use fromuser and it works |
| 00:32.30 | lancey`busy | yes, fromuser works |
| 00:32.31 | justinu | cvs from sometime a few weeks ago |
| 00:32.33 | lancey`busy | if it's in sip.conf |
| 00:32.35 | justinu | oh |
| 00:32.48 | lancey`busy | what i want is not have fromuser in sip.conf |
| 00:32.54 | lancey`busy | and set callerid in the dialplan |
| 00:33.01 | lancey`busy | and then dial out the sip channel |
| 00:33.08 | lancey`busy | it SHOULD use the callerid as "fromuser" |
| 00:33.15 | justinu | ah |
| 00:33.18 | lancey`busy | but it sometimes fills it with "Unknown" |
| 00:33.35 | lancey`busy | which in fact turns out to be the "default" caller id :) |
| 00:34.06 | lancey`busy | http://bugs.digium.com/view.php?id=5405 |
| 00:34.16 | lancey`busy | here's what i'm talking about, a bit more explained |
| 00:35.14 | lancey`busy | the most weird thing is it *does* work as expected, when called from an IAX2 phone |
| 00:35.20 | lancey`busy | but when called from another * box - nada |
| 00:35.28 | lancey`busy | EVIL :) |
| 00:35.39 | justinu | heh |
| 00:36.27 | lancey`busy | #define CALLERID_UNKNOWN "Unknown" |
| 00:36.29 | lancey`busy | :) |
| 00:42.27 | *** join/#asterisk brookshire (n=matt@gateway.digium.com) |
| 00:43.35 | *** join/#asterisk patpatnz (n=pjs@unaffiliated/patpatnz) |
| 00:43.57 | patpatnz | question: is there any way to configure the rtp from ip address in asterisk? |
| 00:44.29 | *** join/#asterisk supaigtr (n=yurplsl@152.53.17.1) |
| 00:44.39 | supaigtr | Hello. |
| 00:45.45 | cio | Hi. |
| 00:45.59 | cio | patpatnz: eh? |
| 00:46.47 | patpatnz | cio, my asterisk is configured to listen to SIP on 127.0.0.1:5060 but I want the RTP connections to come from the eth interface |
| 00:47.29 | supaigtr | Anyone have any recommendations for IAX2 to IAX2 for a link that has some clicking etc? |
| 00:47.36 | patpatnz | outgoing connections |
| 00:48.26 | cio | Maybe if you explain what you're after it will help me (or anyone else here) understand the question a little better? |
| 00:48.37 | lancey`busy | patpatnz you need the SIP port to be accessible too |
| 00:49.02 | patpatnz | lancey`busy, okay, ta |
| 00:49.12 | lancey`busy | patpatnz so bound the sip channel to the eth IP |
| 00:49.18 | lancey`busy | what do you need it for at 127.0.0.1? |
| 00:49.24 | lancey`busy | or just bound it to 0.0.0.0 |
| 00:49.29 | lancey`busy | (as it is by default) |
| 00:49.50 | lancey`busy | anyone here familiar with CallerPres? |
| 00:49.55 | lancey`busy | it seems to be causing my trouble |
| 00:50.44 | patpatnz | I have SER listening to SIP on the public interface |
| 00:50.54 | patpatnz | Asterisk is acting as a SIP->H323 proxy |
| 00:51.19 | lancey`busy | patpatnz u using Asterisk for H323? |
| 00:51.42 | lancey`busy | i wish you good luck |
| 00:51.57 | patpatnz | lancey`busy, we've been using it for a while, seems okay |
| 00:52.00 | lancey`busy | this is beyond my knowledge and willingness to do :) |
| 00:52.15 | lancey`busy | patpatnz if H323 is listening to the eth interface |
| 00:52.19 | lancey`busy | the RTP should flow, too |
| 00:52.39 | patpatnz | no, it always seems to use the SIP address |
| 00:52.50 | patpatnz | I think because we're going SIP->H323 |
| 00:53.11 | lancey`busy | dunno, maybe there's some way to force it to use another |
| 00:53.21 | lancey`busy | there was something like external_addr |
| 00:53.55 | lancey`busy | why not try to bind it's SIP channel to 0.0.0.0 |
| 00:53.58 | lancey`busy | but on different port? |
| 00:54.02 | patpatnz | checked the voip-info wiki and all it listed was start and end ports |
| 00:54.09 | MnxPower | Wow, I've not seen this much traffic on asterisk-biz since...well...since the last flame war! |
| 00:54.14 | patpatnz | no, the sip parts get confused |
| 00:54.29 | patpatnz | because asterisk doesn't put the port into the sip packets |
| 00:54.34 | MnxPower | Why even try to force it to bind to a specific addr? |
| 00:54.51 | lancey`busy | MnxPower it seems it sends out RTP streams out of 127.0.0.1 |
| 00:54.58 | lancey`busy | in his case... |
| 00:55.00 | patpatnz | because I don't want anyone communicating direct to asterisk |
| 00:55.12 | lancey`busy | patpatnz why is that? |
| 00:55.19 | patpatnz | SER is used to authenticate users |
| 00:55.25 | MnxPower | lancey`busy, Ah. He must not have the reverse DNS or /etc/hosts set up correctly. |
| 00:55.29 | lancey`busy | * could do it, too |
| 00:55.37 | MnxPower | patpatnz, use the linux iptables/ipchains, that's what it's there for. |
| 00:55.45 | patpatnz | No, we're authenitcating from radius |
| 00:55.46 | lancey`busy | MnxPower: he also has SER |
| 00:55.54 | *** join/#asterisk P-NuT (n=pnut_@fw.office.unitedip.net.au) |
| 00:55.59 | lancey`busy | patpatnz there was some work on radius authentication for * |
| 00:56.00 | MnxPower | Getting Asterisk to bind to specific ports/addresses sucks big time. |
| 00:56.06 | patpatnz | It ties in with our existing structures |
| 00:56.11 | MnxPower | and it never seemed to work forrectly for 1.0.x |
| 00:56.21 | patpatnz | I'm using CVS-HEAD |
| 00:56.23 | P-NuT | Hi all, can anyone help me with getting an offsite sip extension working? |
| 00:57.10 | MnxPower | patpatnz, I don't know if it was fixed in CVS-HEAD. |
| 00:57.23 | P-NuT | I've got an asterisk box behind a NAT @ home and want to get an extension here at work. |
| 00:57.28 | P-NuT | Is it possible? |
| 00:57.31 | patpatnz | lancey`busy, we're pretty settled on using the SER for the authentication part, it's much lighter weight than * |
| 00:57.42 | MnxPower | Honestly, use iptable/ipchains to keep everyone but those you want from communicating with Asterisk |
| 00:57.48 | lancey`busy | ;externip = 200.201.202.203 ; Address that we're going to put in outbound SIP messages |
| 00:57.49 | lancey`busy | <PROTECTED> |
| 00:57.49 | MnxPower | P-NuT, yes, it's possible, no it's not easy. |
| 00:57.52 | lancey`busy | ? |
| 00:57.54 | patpatnz | MnxPower, seems to work well except for not being able to set the src ip fro rtp |
| 00:57.56 | P-NuT | oh. |
| 00:58.09 | lancey`busy | this is from sip.conf |
| 00:58.09 | lancey`busy | ? |
| 00:58.15 | P-NuT | How would I go about it? Would it be better doing it with IAX? |
| 00:58.16 | MnxPower | patpatnz, Yeah, that's what I mean by " Getting Asterisk to bind to specific ports/addresses sucks big time." |
| 00:58.18 | lancey`busy | wouldn't it work? |
| 00:58.36 | MnxPower | lancey`busy, SIP and RTP are two totally different things. |
| 00:58.44 | MnxPower | SIP == call setup, RTP=audio |
| 00:58.45 | patpatnz | MnxPower, well, I'll just put another IP on that box, no worries, had hoped to be able to use 127 but nm |
| 00:58.47 | lancey`busy | yup, i know |
| 00:59.08 | lancey`busy | just thought this might have an influence on that |
| 00:59.12 | acidfoo | and IAX multiplex both session call and voice data |
| 00:59.39 | patpatnz | can asterisk use rtpproxy ? |
| 00:59.42 | lancey`busy | patpatnz why not add a second IP |
| 00:59.53 | lancey`busy | and not use localhost at all? |
| 01:00.08 | patpatnz | lancey`busy, I didn't want to if I didn't have to, but it looks like I have no choice |
| 01:00.16 | patpatnz | Luckily I have a /19 ;) |
| 01:00.22 | *** join/#asterisk iguy (n=iguy@dsl093-197-234.mke1.dsl.speakeasy.net) |
| 01:00.38 | lancey`busy | MnxPower: doesn't channels control somehow what is the outgoing address of the rtp streams? |
| 01:00.49 | MnxPower | lancey`busy, in theory 8-) |
| 01:01.00 | lancey`busy | beyond my knowledge again :) |
| 01:01.06 | lancey`busy | too many things out there :) |
| 01:01.08 | MnxPower | patpatnz, d0n't count on adding another IP fixing anything. |
| 01:01.15 | MnxPower | See the mailing list archive. |
| 01:01.17 | MnxPower | ~mailinglist |
| 01:01.19 | jbot | i heard mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 01:01.21 | lancey`busy | MnxPower: familiar with CallerPres? |
| 01:01.30 | patpatnz | MnxPower, I've had it working with a seperate IP don't worry |
| 01:01.37 | MnxPower | The thing about Asterisk, is don't fight it. Let it do what it wants to do in the way it wants to do it. |
| 01:01.40 | patpatnz | I thought I'd just try out 127 too |
| 01:02.20 | lancey`busy | well, if * should establish an RTP stream to some IP address which is obviously accessible on some interface |
| 01:02.28 | lancey`busy | why should it still go for 127.0.0.1? |
| 01:02.43 | patpatnz | pass |
| 01:03.00 | lancey`busy | i don't seem to get it all :/ |
| 01:03.10 | *** join/#asterisk Asylum (i=Asylum@CPE-60-226-88-2.qld.bigpond.net.au) |
| 01:03.52 | patpatnz | I'm looking at RTPproxy now |
| 01:04.35 | patpatnz | Is there any decent SIP<->H.323 gateway software around? |
| 01:04.45 | acidfoo | asterisk ;P |
| 01:04.56 | MnxPower | lancey`busy, I don't understand how Alivil works either, but I just accept that it does. 8-) |
| 01:05.25 | patpatnz | acidfoo, not really |
| 01:06.31 | lancey`busy | :P |
| 01:07.05 | lancey | bad SIP URIs fixed |
| 01:07.11 | lancey | *temporarily* though |
| 01:07.17 | lancey | shit, it's 04:07 am... |
| 01:09.27 | patpatnz | no it's not |
| 01:09.45 | Asylum | 11:08:30 |
| 01:09.48 | Asylum | am :P |
| 01:09.53 | cio | 8:09 |
| 01:10.05 | patpatnz | 14:09 |
| 01:10.07 | lancey | hehe :) |
| 01:10.15 | Asylum | Started a trend :P |
| 01:10.16 | lancey | patpatnz was closer :) |
| 01:10.16 | *** part/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 01:10.30 | Asylum | lol. |
| 01:10.33 | lancey | patpatnz: remove the first digit and you get it right :) |
| 01:10.42 | patpatnz | ;) |
| 01:10.52 | patpatnz | I'm 12 hours ahead of you |
| 01:10.59 | patpatnz | I win |
| 01:11.15 | supaigtr | Should jittbuffer and iaxcompat be used to combat some clicking on IAX-IAX? |
| 01:11.15 | patpatnz | ;) |
| 01:11.22 | lancey | patpatnz: what's the date? |
| 01:11.32 | patpatnz | 11oct |
| 01:11.35 | lancey | shit :) |
| 01:11.39 | patpatnz | heh |
| 01:11.50 | patpatnz | I'm right on the dateline (almost) |
| 01:11.53 | lancey | :) |
| 01:12.04 | lancey | you are 10 hours ahead, i think ;) |
| 01:12.11 | lancey | unless 14 means 4 pm :) |
| 01:12.17 | patpatnz | ah, true |
| 01:12.24 | Asylum | haha. |
| 01:12.27 | patpatnz | my maths was never very good ;) |
| 01:12.52 | lancey | we've got a topic to talk about again here :) |
| 01:13.06 | Asylum | hey, talking of topics |
| 01:13.32 | Asylum | When someone dials in.. they get the IVR.. but once they choose a selection that rings a ring group.. |
| 01:13.44 | Asylum | It pauses for about 10 seconds before dialing the ring group? |
| 01:13.53 | lancey | dunno... |
| 01:13.56 | Asylum | Any idea how to get rid of the pause.. it's only just started happening... |
| 01:14.08 | lancey | what do you mean by a ring group |
| 01:14.17 | lancey | Dial(bla-bla&bla-bla&bla-bla) ? |
| 01:14.30 | Asylum | Well, if they choose option 1 it rings.. ext 610 and 611 |
| 01:14.34 | lancey | or queues? |
| 01:14.45 | Asylum | Dial(bla-bla&bla-bla&bla-bla) |
| 01:15.06 | lancey | works okay for me right now |
| 01:15.23 | Asylum | It was working ok for me too up untill a few days ago heh! |
| 01:15.27 | lancey | though with 2 extensions only |
| 01:15.58 | Asylum | *shrug* i'll figure it out! |
| 01:16.03 | lancey | what do those extension ring? |
| 01:16.10 | lancey | could it be some DNS timeouts or something? |
| 01:16.16 | lancey | just improvising... |
| 01:16.36 | lancey | brb |
| 01:17.22 | Asylum | they ring two snom360's |
| 01:17.38 | *** join/#asterisk Veto_laptop (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 01:18.20 | *** join/#asterisk Tili (i=Tili@202-133-65-171-dialup.sat.net.pk) |
| 01:19.13 | Damin | OK... |
| 01:19.16 | Damin | This is unique.. |
| 01:19.33 | Damin | 11 Access points and all of them configured properly... that includes beer.. |
| 01:19.46 | Asylum | .... |
| 01:23.07 | lancey | hehe :) |
| 01:23.45 | lancey | anyone any opinion on iLBC vs. g729 sound quality? |
| 01:23.59 | brookshire | g729 |
| 01:24.10 | lancey | what about jittery links? |
| 01:24.14 | patpatnz | g729 cos nothing supports iLBC |
| 01:24.27 | Nivex | patpatnz: asterisk supports it :) |
| 01:24.35 | lancey | assuming both ends support iLBC? |
| 01:24.40 | lancey | patpatnz: my IP phone does support it |
| 01:25.20 | patpatnz | lancey, none of the sip devices we have support it |
| 01:25.33 | patpatnz | which are cisco and polycom |
| 01:25.35 | Tili | what about Speex. Speex is pretty good specially with latest code from speex.org? |
| 01:25.44 | lancey | as well as iax |
| 01:25.46 | patpatnz | I have a grandstream, does it support it? |
| 01:25.53 | coppice | speex is far better than iLBC |
| 01:26.08 | lancey | :) |
| 01:26.14 | lancey | another codec war formed out here |
| 01:26.15 | Nivex | coppice: even with the ungodly conversion latency? |
| 01:27.20 | coppice | iLBC uses twice the bit rate of g.729 or speex, to get some packet loss tolerance. if out double the data rate in g.729 or speex with redundancy you beat iLBC hands down |
| 01:27.21 | patpatnz | lancey, nothing as enjoyable as a holy way |
| 01:27.22 | patpatnz | er |
| 01:27.24 | patpatnz | war |
| 01:27.36 | coppice | Nivex: what conversion latency? |
| 01:28.11 | Nivex | coppice: the ones listed in "show translation" |
| 01:28.26 | Nivex | ulaw to ilbc is 198ms |
| 01:28.31 | lancey | :) |
| 01:28.41 | lancey | Nivex |
| 01:28.46 | lancey | that depends on the machine used |
| 01:28.54 | coppice | Nivex: something is seriously wrong for that to happen |
| 01:29.03 | Nivex | speex is 382 |
| 01:29.13 | lancey | <PROTECTED> |
| 01:29.17 | lancey | 3 to ulaw |
| 01:29.18 | lancey | 3 to alaw |
| 01:29.24 | Nivex | hmm. |
| 01:29.34 | lancey | <PROTECTED> |
| 01:29.37 | lancey | the last one - 13 |
| 01:29.40 | lancey | is ulaw to ilbc |
| 01:29.57 | lancey | and it's the highest payload at all |
| 01:29.58 | patpatnz | if we're using g729, and we have voicemail with all the prompts in g729 format, do we need licenses for * to do g729? |
| 01:30.14 | lancey | patpatnz |
| 01:30.18 | lancey | what do u use? |
| 01:30.21 | lancey | linux or freebsd? |
| 01:30.24 | patpatnz | linux |
| 01:31.01 | lancey | in this case you wouldn't need codecs, though |
| 01:31.31 | patpatnz | no, I didn't think so, thanks :) |
| 01:34.07 | Tili | so i think if network is developed where you all clients and servers use same codec we can get best results with speex |
| 01:34.53 | coppice | g.729 is necessary of you need to interoperate with g.729 users. otherwise speex is a good choice |
| 01:35.42 | *** join/#asterisk desktophero (n=desktoph@ip24-56-30-250.ph.ph.cox.net) |
| 01:36.07 | cio | g729 = low bandwidth |
| 01:36.32 | cio | Is there an * command that will record directly to a voicemail box without the intro? |
| 01:37.17 | coppice | speex uses the same bandwidth as g.729 |
| 01:37.28 | cio | Does * support speex? |
| 01:37.36 | coppice | yes |
| 01:37.46 | cio | Cool. What hardphones support speex? |
| 01:37.47 | patpatnz | isn't speex lower quality than g729 though? |
| 01:37.50 | lancey | <PROTECTED> |
| 01:37.50 | lancey | * 's' instructions for leaving the message will be skipped. |
| 01:38.05 | cio | Thanks - |
| 01:38.12 | lancey | show application VoiceMail |
| 01:38.13 | lancey | :) |
| 01:38.39 | coppice | speex is comparable in quality to g.729. close enough you would need proper blind tests to find a winner |
| 01:38.53 | cio | Is there any hardphones that support speex? |
| 01:39.04 | coppice | not sure |
| 01:39.26 | cio | But it would be good for say remote offices via IAX? |
| 01:39.48 | lancey | IAX is protocol |
| 01:39.51 | lancey | speex is codec |
| 01:39.57 | coppice | if you want broad support in hardphones and low bit rate g.729 is the obvious choice |
| 01:40.11 | lancey | ah, i got your point, sorry |
| 01:40.18 | lancey | it must be the hour |
| 01:43.20 | patpatnz | lancey, I find it hard to concentrate in the afternoon too ;) |
| 01:43.43 | lancey | :) |
| 01:45.20 | Tili | is there anyone who knows C a bit. can someone look at chan_iax2.c for asterisk-1.0.9 on line 5762. I am confused in mutex locking code in this function |
| 01:46.48 | *** join/#asterisk Uberbot (n=Uberbot@69.252.219.76) |
| 01:48.03 | *** join/#asterisk pifiu (n=myassisb@208.205.181.170) |
| 01:49.04 | Uberbot | Hi all. |
| 01:49.10 | lancey | i'm gonna take a nap |
| 01:49.13 | lancey | good luck all, guyz |
| 01:49.30 | *** part/#asterisk patpatnz (n=pjs@unaffiliated/patpatnz) |
| 01:49.56 | Uberbot | If you had the option of buying a Sipura SPA-2002 or a Linksys PAP2-NA, which would you buy? |
| 01:50.19 | pifiu | is there a generic wav file by allyson that says "If you know your party's extension, you may dial it at any time" |
| 01:50.53 | Uberbot | ... or for that matter, "What are you wearing tonight?" :-D |
| 01:51.17 | pifiu | ? |
| 01:52.44 | Uberbot | Wow! Slow night. |
| 01:57.17 | *** join/#asterisk gniretar (n=gniretar@66-227-204-109.dhcp.bycy.mi.charter.com) |
| 01:57.34 | gniretar | hi everyone |
| 01:57.43 | gniretar | greetings from my new PowerBook |
| 01:57.53 | gniretar | and it has xchat :-D |
| 01:57.58 | gniretar | gotta ove the fink project |
| 01:58.10 | gniretar | so has anyone ever used LoudHush? |
| 01:58.29 | blitzrage | apple and powerbooks suck |
| 01:59.17 | gniretar | oh shut your ignorant trap |
| 01:59.21 | *** join/#asterisk SplasPood (n=sp@brooklyn.paravolve.net) |
| 01:59.32 | gniretar | PowerBooks kick a$$ |
| 02:00.25 | brookshire | epic 4 life! |
| 02:01.33 | pifiu | does anyone know what allison smith looks like? |
| 02:01.41 | pifiu | is www.allisonsmith.org her site? |
| 02:05.30 | jake1932 | no - http://www.theivrvoice.com/ is |
| 02:07.07 | cio | Anyone know how to 'list tables' in sqlite? |
| 02:07.21 | blitzrage | pifiu: yep... met her last year |
| 02:08.41 | jake1932 | cio - did you check here? http://www.sqlite.org/faq.html#q9 |
| 02:09.20 | pifiu | did she look nice blitz? |
| 02:09.29 | pifiu | worth the trip to cali? |
| 02:09.34 | pifiu | to see her in astricon |
| 02:09.39 | cio | rtfm? NEVER! |
| 02:09.50 | MnxPower | pifiu, Um, NO woman is worth a trip to Calif |
| 02:10.29 | pifiu | gotcha |
| 02:10.29 | pifiu | lol |
| 02:10.59 | MnxPower | But I did hear that someone saw minnie mouse coming out of blitzrage's hotel room this morning. |
| 02:11.31 | Uberbot | If you had the option of buying a Sipura SPA-2002 or a Linksys PAP2-NA, which would you buy? |
| 02:11.56 | MnxPower | Uberbot, They are pretty much the same thing, but I'd go with the SPA-2002 |
| 02:11.58 | cio | If I'm using SQLITE for my CDR logs, is there a way to disable the CSV logs? I don't see the point in having two sets of logs... |
| 02:11.58 | brookshire | same thing right? |
| 02:12.14 | Uberbot | Any reason, MnxPower? |
| 02:12.24 | MnxPower | cio, did you try a noload => cdr_csv.so? |
| 02:12.27 | cio | heh |
| 02:12.34 | *** join/#asterisk Inv_arp (i=junya@adsl-156-145-65.mia.bellsouth.net) |
| 02:12.35 | cio | As soon as I hit enter. |
| 02:12.41 | MnxPower | Uberbot, Becuase the SPA is officially supported and more people have them |
| 02:12.42 | cio | Sorry, I'm out, too tired. |
| 02:12.50 | *** join/#asterisk lbow (n=lbow@206.165.75.199) |
| 02:13.19 | MnxPower | If Linksys didn't own Sipura, I'd say "Because Linksys have been such bastards about the PAP2-NA" |
| 02:14.49 | Uberbot | MnxPower, Good answers. Thanx. |
| 02:15.29 | *** join/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 02:17.05 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivflkc.dialup.mindspring.com) |
| 02:20.29 | *** join/#asterisk Rubble (n=Rubble@206.165.75.199) |
| 02:22.45 | Rubble | rubble is here from South Africa - 22 hours of flights later! |
| 02:23.43 | lbow | rubble: you are too keen |
| 02:24.43 | *** join/#asterisk mcunixjr (i=mcunixjr@pdpc/supporter/gold/McUnixJr) |
| 02:24.46 | mcunixjr | ola |
| 02:26.57 | *** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 02:28.16 | pifiu | how do I play .gsm files? |
| 02:29.06 | mcf3782 | install the sox package and use the 'play' command that comes with sox. "play my-gsm-file-name.gsm" |
| 02:29.26 | pifiu | yeah but i have the files here in my windows machine |
| 02:29.37 | pifiu | winamp wont play the |
| 02:29.39 | pifiu | m |
| 02:30.08 | jake1932 | sound forge will |
| 02:30.14 | mcf3782 | sorry.. can't help you with windows. |
| 02:30.43 | pifiu | ok thanks |
| 02:30.50 | mcunixjr | i have a AMP question - but it seems the amportal channel is deadly quiet - i am getting an error when trying to install AMP on redhat EL3. I have PHP installed with mysql option and have mysql installed. i can login into mysql via PHP with no problem. but when i run ./install (for AMP) it says [FATAL mysql php libraries not installed |
| 02:31.40 | lbow | apple quicktime plays .gsm files too |
| 02:32.15 | Nivex | http://pastebin.ca/25184 |
| 02:37.06 | mcunixjr | anyone? |
| 02:37.12 | Nivex | Are the translation times in my pastebin sane, or do I have more problems? |
| 02:38.44 | *** join/#asterisk tainted_ (n=identd@ppp-71-137-169-240.dsl.irvnca.pacbell.net) |
| 02:38.51 | tainted_ | hello |
| 02:39.41 | tainted_ | let's say i have two users connected to my asterisk box via SIP |
| 02:39.47 | tainted_ | how can i connect them together? |
| 02:40.15 | tzanger | tainted_: have one dial the ohter |
| 02:40.42 | tainted_ | would they have to go offline first? |
| 02:40.45 | tainted_ | or can that happen in realtime |
| 02:40.47 | tzanger | huh? |
| 02:41.07 | tainted_ | what do u mean dial the other? |
| 02:41.19 | tzanger | you have exten => 100,1,Dial(SIP/100) |
| 02:41.25 | tzanger | and exten => 101,1,Dial(SIP/101) |
| 02:41.41 | tzanger | then when user 100 picks up the phone and dials 101 it rings 101's phone |
| 02:41.52 | tzanger | if 101 picks up, they're connected |
| 02:42.00 | tainted_ | oops |
| 02:42.03 | tainted_ | i think i confused u |
| 02:42.17 | tzanger | yes you did |
| 02:42.17 | tainted_ | let's say both are outside callers |
| 02:42.19 | tainted_ | dialing my IVR |
| 02:42.26 | tainted_ | and i want to connect the two of them up |
| 02:42.51 | tzanger | drop them both into a meetme then |
| 02:43.09 | tainted_ | can that happen transparently? |
| 02:43.18 | tainted_ | w/o their input? |
| 02:43.34 | tzanger | how could it? |
| 02:43.58 | tzanger | how do you get htem to signal "I want to be connected with some other caller I have no idea who they are or even if they're online" ?? |
| 02:44.04 | tzanger | what is it you want to do, exactly |
| 02:44.41 | *** join/#asterisk PhreeStyle (n=PhreeSty@cpe-24-221-52-165.az.sprintbbd.net) |
| 02:48.20 | Vco | huh... |
| 02:48.33 | Vco | i didn't know allison smith was from alberta.. |
| 02:54.32 | *** join/#asterisk Teeli (i=Tili@202-133-65-171-dialup.sat.net.pk) |
| 02:57.38 | *** join/#asterisk wolfson (n=hehe@usr-kdh-208-6-58-26.beachlink.com) |
| 02:58.02 | *** join/#asterisk loud (n=ariel@cypher.punk.net) |
| 03:10.19 | *** join/#asterisk shmaltz (n=chatzill@69.28.255.210) |
| 03:10.32 | shmaltz | stupid verizon |
| 03:11.59 | Vco | employmeny apathy? |
| 03:12.05 | Vco | emplyment rather |
| 03:12.09 | Vco | fucing hell |
| 03:12.19 | Vco | ahh.. |
| 03:12.25 | Vco | cookie crumbs |
| 03:15.41 | *** join/#asterisk Ambrose (n=ambrose@we-dont.gotdns.org) |
| 03:16.41 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 03:17.03 | *** join/#asterisk rubble (n=netclass@206.165.75.199) |
| 03:17.26 | Ambrose | Can anyone link me to a good document about building an IVR? |
| 03:21.07 | rubble | http://www.voip-info.org/wiki-Asterisk+tips+IVR+menu |
| 03:21.40 | Ambrose | That's the one I've been looking at :-( |
| 03:21.49 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 03:22.36 | Ambrose | I want to build a menu that starts off with "Press 1 for English 2 for French" And then once the user presses 1 a new menu starts that says "press 1 for support 2 for sales" etc, but I can't figure out how to seperate the english/support (ext 1) from each other |
| 03:26.57 | loud | Wait(1) ? |
| 03:32.07 | *** join/#asterisk nexis (n=nexis@12-219-60-252.client.mchsi.com) |
| 03:32.17 | Teeli | Ambrose: its bad to say this, but there is no substitute for trying it out and playing things yourself. anyway for a start do this |
| 03:32.27 | Teeli | [default] |
| 03:32.44 | Teeli | exten => s,1,Wait(1) |
| 03:32.50 | Teeli | exten => s,2,Answer |
| 03:33.39 | Teeli | exten => s,3,Background(your_prompt_for_fr_and_en) |
| 03:33.46 | Teeli | exten => s,4,Hangup |
| 03:34.01 | Teeli | oops |
| 03:34.04 | Teeli | mistake |
| 03:34.32 | *** join/#asterisk Veto_laptop (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 03:34.33 | Teeli | big mistake |
| 03:35.40 | Ambrose | Teeli : Gotcha, thx :-) |
| 03:36.46 | nexis | im having a problem, when i call somone, or somone calls me, asterisk routes the calls to a ATA, but when the other person hangs up, it fails to term the call on the ATA, at the end of the exten calls, i have a hangup clause, but its not killing the connection. |
| 03:39.29 | Teeli | Ambrose: in continuation to that |
| 03:39.36 | Teeli | exten => 1,1,Background(This_is_ur_english_menu) |
| 03:39.38 | Teeli | exten => 2,1,Background(this_is_ur_french_extension) |
| 03:39.40 | Teeli | so add more priorities to extension 1 or 2 for your IVR |
| 03:41.56 | *** join/#asterisk Corndawg_ (n=notreall@c-66-176-249-51.hsd1.fl.comcast.net) |
| 03:44.05 | *** join/#asterisk Asylum (i=Asylum@CPE-60-226-88-2.qld.bigpond.net.au) |
| 03:44.34 | *** join/#asterisk ZX81 (n=ZX81@222-153-100-242.jetstream.xtra.co.nz) |
| 03:45.02 | *** join/#asterisk oej (n=Olle@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 03:47.54 | *** join/#asterisk Los415 (n=los415@64.201.109.62) |
| 03:48.04 | *** join/#asterisk marc324 (n=marc3234@206-248-159-56.dsl.teksavvy.com) |
| 03:50.24 | *** join/#asterisk newmember (n=newmembe@S010600a0c93dce87.cg.shawcable.net) |
| 03:50.46 | *** join/#asterisk bmg505 (n=leon@rndf-146-27-248.telkomadsl.co.za) |
| 03:51.10 | *** part/#asterisk nexis (n=nexis@12-219-60-252.client.mchsi.com) |
| 03:51.37 | *** join/#asterisk jdv79 (n=jdv79@u1057064.ul.warwick.net) |
| 03:52.13 | jdv79 | does anyone use a DB cdr setup to a seperate box? |
| 03:52.43 | jdv79 | i'm curious what happens if that DB box disappears? |
| 03:54.29 | wasim | you get an err on CDR write |
| 03:54.51 | ZX81 | jdv79 unless you are writing to 2 cdrs |
| 03:54.55 | ZX81 | which is what we do |
| 03:54.55 | jdv79 | asterisk will continue processing and just unfatally error on the CDR ops? |
| 03:55.06 | ZX81 | we use the Master.csv file as backup |
| 03:55.19 | ZX81 | so if the db goes down we still have another copy |
| 03:55.24 | jdv79 | you can load 2 up at once?/ |
| 03:55.39 | *** join/#asterisk sycofly (n=syco@sycofly.com) |
| 03:55.40 | ZX81 | yep |
| 03:55.43 | jdv79 | that's actually what i was thinking about doing |
| 03:55.46 | ZX81 | you can have 10 |
| 03:55.47 | ZX81 | :D |
| 03:55.55 | jdv79 | thanks |
| 03:55.58 | ZX81 | np |
| 03:56.02 | ZX81 | ~adn |
| 03:56.04 | jbot | i heard adn is the Asterisk Daily News - http://www.sineapps.com/news.php for HTML and http://www.sineapps.com/rssfeed.php for RSS |
| 03:56.06 | ZX81 | :) |
| 03:56.10 | jdv79 | do you periodcially dump the CSV? |
| 03:56.20 | ZX81 | we do billparts |
| 03:56.25 | jdv79 | ok |
| 03:56.29 | ZX81 | so every day we build part of the bill |
| 03:56.39 | ZX81 | then at the end of the billing period we add them all together |
| 03:56.47 | *** part/#asterisk sycofly (n=syco@sycofly.com) |
| 03:56.55 | ZX81 | that way if a rate changes during the month, the customer is billed at that day's rates |
| 03:57.07 | ZX81 | :) |
| 03:57.10 | jdv79 | i just want the CSV around if the DB cdr breaks so we can cobble it together afterwards |
| 03:57.16 | jdv79 | otherwise its useless to me |
| 03:57.22 | ZX81 | yeah |
| 03:57.31 | ZX81 | type locate Master.csv |
| 03:57.36 | marc324 | I get : Loading module cdr_pgsql.so failed! |
| 03:57.43 | marc324 | why? |
| 03:57.46 | jdv79 | ouch marc324 |
| 03:57.57 | jdv79 | that's the one i was gonna use |
| 03:59.59 | *** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1) |
| 04:00.23 | wasim | ZX81: Farfon today formally annouced ceasure of further development on their IAX2 platform due to the availability of more cost effective alternates such as PA168. Farfon would like to thank the * community for its valuable support and the team had a great learning experience. Farfon intends to utilize the development for other efforts, such as platforms for teaching DSP and VoIP device development. |
| 04:00.45 | *** join/#asterisk Jzalae (n=sk@216-220-248-175.midmaine.com) |
| 04:01.50 | *** join/#asterisk gambolputty (n=gambolpu@72.240.241.108) |
| 04:02.02 | ZX81 | :( boo hoo |
| 04:02.11 | ZX81 | you gonna start doing 1688 stuff? |
| 04:04.36 | wasim | well, we've been using it and recommending it for some time now, but we aren't doing any dev on it |
| 04:04.50 | *** part/#asterisk Uberbot (n=Uberbot@69.252.219.76) |
| 04:05.25 | ZX81 | ok |
| 04:05.30 | ZX81 | posting news article |
| 04:05.54 | wasim | yes, we don't want people to wonder what happened to us, all hue and cry and then nothing |
| 04:06.13 | *** join/#asterisk AlexCTI (n=alex@weston-69.65.86.197.myacc.net) |
| 04:06.38 | *** part/#asterisk jdv79 (n=jdv79@u1057064.ul.warwick.net) |
| 04:06.57 | ZX81 | :D |
| 04:07.01 | ZX81 | understood |
| 04:07.04 | AlexCTI | Some has a good mp3 file to MOH? and the wait to set up it? |
| 04:07.16 | ZX81 | I don't use mp3s |
| 04:07.23 | ZX81 | transcode files to the right codec |
| 04:07.26 | wasim | danke ZX81 |
| 04:07.27 | AlexCTI | The one that i have sound not good |
| 04:07.32 | *** part/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 04:07.36 | Nugget | I made some music using apple soundtrack. royalty-free and it's nice and mellow. :) |
| 04:08.08 | Nugget | I need a telemarketer moh queue, though, that's got death metal. That would be handy |
| 04:08.23 | Nugget | or maybe the hampsterdance song |
| 04:08.30 | AlexCTI | that my config is queue1english => custom:/var/lib/asterisk/mohmp3/queue1english,/usr/local/bin/mpg123 -q -r 8000 -f 8192 -b 2048 --mono -s |
| 04:08.41 | tuppa | Nugget: or the badgerbadgerbadger song |
| 04:08.59 | wasim | tuppa: is that a univ of wisconsin song? |
| 04:09.05 | file | ZX81 has lost it |
| 04:09.11 | tuppa | wasim: I know nothing about univ of wisconsin |
| 04:09.27 | wasim | tuppa: they have an affinity for badgers |
| 04:10.32 | tuppa | wasim: I'm thinking about this http://www.badgerbadgerbadger.com/ |
| 04:10.40 | tuppa | warning, flash required, etc |
| 04:13.17 | *** join/#asterisk count (n=adam@corp.alanne.com) |
| 04:13.30 | wasim | you think we could do an Asterisk for Earthquake Relief? |
| 04:14.02 | wasim | the situation is pretty grim for over a million homeless here |
| 04:14.48 | wasim | if the VoIP terminators gave say .01c per minute and possibly other donations from *masters we might be able to help a few folks |
| 04:15.25 | *** join/#asterisk shido6 (n=shido6@d221-68-210.commercial.cgocable.net) |
| 04:15.41 | *** join/#asterisk almafuerte (i=0@200-70-113-55.mrse.com.ar) |
| 04:16.01 | wasim | we could make a difference, a direct, visible difference |
| 04:18.29 | AlexCTI | Some has a good MOH file in MP3? i'm converting a u-law to MP3 but i don't get a good quality.. |
| 04:18.52 | wasim | AlexCTI: is the ulaw in good quality? |
| 04:19.04 | AlexCTI | yes |
| 04:19.14 | wasim | what are you using to convert? |
| 04:19.25 | AlexCTI | i'm using gold wave |
| 04:21.29 | wasim | hmm ... perhaps thats the problem |
| 04:22.06 | AlexCTI | i think so, it is not making a good job |
| 04:22.31 | marc324 | what is unixodbc-dev? |
| 04:24.50 | AlexCTI | wasim, do you have good quality on MOH? |
| 04:25.27 | almafuerte | <PROTECTED> |
| 04:26.04 | wasim | AlexCTI: yes, its ok quality for phone work |
| 04:26.22 | marc324 | i get: WARNING[23183]: loader.c:554 load_modules: Loading module cdr_pgsql.so failed! |
| 04:29.09 | almafuerte | <PROTECTED> |
| 04:29.39 | almafuerte | <PROTECTED> |
| 04:29.54 | almafuerte | at least you have a good excuse ;-) |
| 04:30.08 | marc324 | I just installed psql,unixodbc |
| 04:30.24 | marc324 | after reinstalling asterisk... i get that error |
| 04:30.26 | *** join/#asterisk JerJer[mobile] (n=jj@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 04:33.44 | JerJer[mobile] | man don't everyone type at once - i cannot keep up |
| 04:33.46 | *** join/#asterisk tin2 (n=abc@203.143.170.32) |
| 04:35.58 | wasim | we've already collected over a few $k for the relief operation, and every bit helps |
| 04:36.36 | marc324 | what package is need for asterisk realtime -- UNIXODBC, POSTGRESQL, |
| 04:36.49 | marc324 | anything else before installing asterisk? |
| 04:37.25 | *** join/#asterisk mog_home (n=mogorman@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 04:37.40 | file | Mattttttttttt |
| 04:38.00 | mog_home | Fileeee |
| 04:38.05 | file | hi. |
| 04:38.07 | marc324 | what package is needed before installing asterisk with RT? UNIXODBC, POSTGRESQL. |
| 04:38.14 | marc324 | anything else? |
| 04:39.15 | Damin | Juggie: Me and Blitzrage.. |
| 04:39.21 | Damin | mog: There we goo.. can you see the access point now? |
| 04:39.41 | *** join/#asterisk drumkilla_laptop (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 04:39.41 | *** mode/#asterisk [+o drumkilla_laptop] by ChanServ |
| 04:39.53 | Damin | drumkilla: What up dog? |
| 04:39.56 | *** join/#asterisk KaBewM (n=kabewm@66-215-7-106.dhcp.psdn.ca.charter.com) |
| 04:40.11 | Damin | drumkilla: You chillin in the Laguna? |
| 04:41.06 | Damin | mog? |
| 04:41.08 | Juggie | damin, not doing anything intreasting? |
| 04:41.17 | Damin | Juggie: Twisted just showed up.. |
| 04:41.24 | Damin | Juggie: We're going to to drinking! |
| 04:41.46 | *** join/#asterisk brookshire (n=matt@esbrooks3.traveller.com) |
| 04:42.10 | Damin | mog? |
| 04:42.13 | Damin | MOG!! |
| 04:42.24 | Juggie | where? |
| 04:42.30 | Damin | Juggie: Basement? |
| 04:42.38 | Juggie | basement? |
| 04:42.40 | drumkilla_laptop | Damin: yeah, your AP is working now :) |
| 04:42.53 | brookshire | russell! |
| 04:43.02 | file | Mattttttttt |
| 04:43.06 | brookshire | file! |
| 04:43.07 | Damin | drumkilla: Cool.. someoone ripped out the cord.. |
| 04:44.27 | *** join/#asterisk mog_home (n=mogorman@206.165.75.198) |
| 04:44.36 | Juggie | beer, where |
| 04:44.44 | brookshire | there is mog |
| 04:44.56 | mog_home | hey |
| 04:46.44 | drumkilla_laptop | Damin: wtf, how did that happen |
| 04:47.15 | Juggie | where is the beer? |
| 04:47.15 | *** join/#asterisk planetjk (n=planetjk@69.255.45.187) |
| 04:48.41 | file[laptop] | I should go to sleep |
| 04:49.24 | file[laptop] | since I've been coding since... this morning... I should check all my regular sites |
| 04:49.26 | Corydon76-home | Might I just say that Mark is very generous... |
| 04:49.39 | Juggie | why what happened? |
| 04:49.57 | Moc | Just read the news about farfon |
| 04:50.03 | Corydon76-home | I could tell you, but then I'd have to kill you... ;-) |
| 04:50.08 | file[laptop] | hi Moc |
| 04:50.10 | Juggie | hahaha..... |
| 04:50.13 | Moc | hi file |
| 04:50.25 | wasim | Moc: :( |
| 04:50.26 | Juggie | now your making it sound gay Corydon :P |
| 04:50.35 | *** part/#asterisk almafuerte (i=0@200-70-113-55.mrse.com.ar) |
| 04:50.37 | Moc | wasim, yea |
| 04:51.08 | Moc | any channel bank or something on the way still ? |
| 04:51.41 | wasim | Moc: no, Adit has a product out that does 24 SIP-FXS in the same price range |
| 04:51.42 | Corydon76-home | Juggie: who said it wasn't? |
| 04:51.49 | Moc | ha |
| 04:51.50 | Juggie | hhaha |
| 04:51.56 | *** join/#asterisk Cresl1n (n=matt@206.165.75.198) |
| 04:52.04 | Juggie | remind me not to stand back on to you |
| 04:52.05 | Moc | what is a platform for teaching dsp ? |
| 04:52.11 | Juggie | matt, did the payment go through after? |
| 04:52.13 | *** join/#asterisk JerJer[mobile] (n=jj@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 04:52.25 | Corydon76-home | Juggie: nah, you have to consent first |
| 04:52.33 | Juggie | haha |
| 04:52.35 | brookshire | JerJer[mobile]: are you at lax? |
| 04:52.54 | Corydon76-home | Juggie: did you just say, "please"? |
| 04:52.57 | wasim | Moc: we are working on a framework to allow the hw to be used to teach TI c54x DSP code and implementation |
| 04:53.06 | wasim | Moc: and release it to universities etc |
| 04:53.19 | Corydon76-home | I coulda sworn I heard you say "please" |
| 04:53.27 | Corydon76-home | ;-) |
| 04:53.29 | Juggie | no |
| 04:53.30 | Moc | interesting |
| 04:53.31 | Juggie | no please :P |
| 04:53.51 | Corydon76-home | No or please? Damn, you're confusing? |
| 04:53.52 | wasim | Moc: and also allow students to develop other applications etc |
| 04:54.42 | Damin | drumkilla: Don't know.. |
| 04:56.03 | Moc | gota check it |
| 04:56.41 | planetjk | Evening... I had AAH working on a test machine, swapped it out with a good machine. Just installed. Ran netconfig. I can ping the IP but can't browse. So I ran [[/etc/rc.d/init.d/httpd restart]] and I get [[Stopping httpd: Failed |
| 04:57.57 | *** join/#asterisk twisted|astricon (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 04:57.58 | *** mode/#asterisk [+o twisted|astricon] by ChanServ |
| 04:59.52 | planetjk | the forums suggest it's a hosts problem... but I think my hosts file is ok. what's the default in hosts? |
| 05:00.59 | Corydon76-home | You using VirtualHosts in Apache? |
| 05:01.51 | planetjk | that one's over my head. It's a fresh install, haven't done anything but run netconfig |
| 05:02.03 | *** join/#asterisk alphaque (n=alphaque@218.111.204.40) |
| 05:02.09 | *** join/#asterisk Eight (n=blake@12-227-171-175.client.mchsi.com) |
| 05:02.36 | Corydon76-home | Your IP likely changed with your transplant, and all the config files are now set up with the old IP |
| 05:02.55 | Corydon76-home | Hence, major brokenness |
| 05:03.20 | planetjk | no transplant, though. two different PC's |
| 05:03.33 | Corydon76-home | Two different network cards? |
| 05:03.39 | planetjk | the other one's shutdown and off the LAN |
| 05:03.52 | Corydon76-home | Ah, so new install? |
| 05:03.56 | planetjk | yup. the test box was a desktop. I'm using AAH on a laptop now |
| 05:03.58 | planetjk | yes |
| 05:04.17 | Corydon76-home | No idea. Best to just learn how to use the machine |
| 05:05.08 | planetjk | I know there's only one line in the initial hosts file... where can I find what it's supposed to be? |
| 05:05.46 | Corydon76-home | Trying to use a point-and-click install on an OS which wasn't designed to be point-and-click can have weird side effects |
| 05:06.15 | planetjk | wow, that's deep. :) except I have to be able to use it before I can learn to use it. or maybe it was supposed to be a short lesson. lol |
| 05:06.43 | planetjk | so... the problem is with AAH, then? you're losing me. which, of course, is obviously not difficult at all |
| 05:07.44 | Corydon76-home | This is why the experts don't use AAH |
| 05:08.30 | planetjk | ah. I missed that "not the choice of any experts. whatsoever" disclaimer |
| 05:09.07 | planetjk | good thing I'm a beginner, and therefore overly qualified |
| 05:09.14 | *** join/#asterisk ZX81 (n=ZX81@222-153-100-242.jetstream.xtra.co.nz) |
| 05:10.59 | planetjk | so your suggestion is not to do anything until I'm an expert? |
| 05:11.01 | *** join/#asterisk Pikoro (n=webmaste@db.sunny-net.ne.jp) |
| 05:11.03 | *** join/#asterisk newmember (i=user@S010600036d1139fb.cg.shawcable.net) |
| 05:12.00 | alphaque | planetjk: it's about flexibility. the GUI based model, which AAH follows does limit you somewhat, but is easier for newbies to grasp |
| 05:12.16 | alphaque | planetjk: editting .conf files directly is harder, but lets u do more outside the constrains of the gui |
| 05:12.28 | Pikoro | what could possibly allow incoming SIP calls but not outgoing SIP calls? |
| 05:12.32 | Pikoro | trunking? |
| 05:12.50 | alphaque | Pikoro: wrong auth credentials ? |
| 05:12.58 | Pikoro | incoming works... |
| 05:13.08 | Pikoro | and the credentials are correct, i'm sure of that |
| 05:13.28 | planetjk | I understand that... I'm not really addressing the issue from a method perspective... Just troubleshooting. Per the docs, should be able to config via web. I can't... so I'm trying to figure it out |
| 05:13.49 | Pikoro | i get the operator for the voip provider... a japanese version of "All circuts are busy" |
| 05:14.00 | *** join/#asterisk docE (n=docE@206.165.75.198) |
| 05:14.04 | docE | SUP SUP!?!?!?! |
| 05:14.41 | Ambrose | There is no seperator between the Voicemail flags and boxnumber@context ? |
| 05:14.49 | docE | Who is here @ astricon? |
| 05:15.00 | alphaque | Pikoro: what does sip debug say ? |
| 05:15.05 | brookshire | root! |
| 05:15.14 | brookshire | oops |
| 05:15.14 | drumkilla_laptop | docE: #astricon |
| 05:15.24 | *** join/#asterisk mog_home (n=root@user-24-236-84-48.knology.net) |
| 05:15.35 | docE | kewl |
| 05:15.49 | mog_home | woot |
| 05:15.54 | Pikoro | well.. |
| 05:16.11 | Pikoro | it says alot :D |
| 05:16.16 | Pikoro | lemme go look it up... |
| 05:16.50 | alphaque | Pikoro: use pastebin.ca |
| 05:17.31 | Pikoro | k |
| 05:18.25 | *** join/#asterisk Qwell (n=chatzill@pool-71-108-241-125.lsanca.dsl-w.verizon.net) |
| 05:20.17 | *** part/#asterisk aciddemon (i=aciddemo@freeforge.net) |
| 05:23.24 | *** join/#asterisk ubergoober (n=ubergoob@c-24-16-108-200.hsd1.ca.comcast.net) |
| 05:24.48 | *** join/#asterisk justinnnnn (n=justinnn@61.95.68.85) |
| 05:25.10 | justinnnnn | hey ppls :) |
| 05:27.16 | JamesDotCom | yo justinnnnn |
| 05:27.17 | *** join/#asterisk pooh_ (n=hfwang@cust.15.241.adsl.cistron.nl) |
| 05:27.56 | pooh_ | Hi, anybody knows why I have no sound on my * box when /dev/zap devices are PRESENT, if I remove those, I have sound again |
| 05:28.20 | ZX81 | maybe cos when they are present you are running asterisk |
| 05:28.27 | ZX81 | and asterisk is loading sound card drivers |
| 05:28.40 | ZX81 | which may steal the soundcard |
| 05:28.45 | ZX81 | dunno just a guess |
| 05:28.49 | pooh_ | hmmm |
| 05:28.54 | pooh_ | let me check modules.conf |
| 05:28.54 | Pikoro | Oct 11 14:25:30 DEBUG[2090]: Ooh, format changed from unknown to ulaw |
| 05:28.56 | Qwell | If you want sound outside of asterisk, you can noload => chan_oss.so and chan_alsa.so in /etc/asterisk/modules.conf |
| 05:28.58 | Pikoro | what do those mean? |
| 05:29.03 | ZX81 | pooh_ yeah |
| 05:29.13 | Corydon76-home | Pikoro: they are DEBUG statements |
| 05:29.13 | Qwell | That will disable the ability to dial from the console though...which isn't that important anyhow |
| 05:29.16 | ZX81 | Pikoro that you have allow=all maybe |
| 05:29.30 | Pikoro | i have allow=alaw&ulaw |
| 05:29.31 | ZX81 | Qwell: does it even work? |
| 05:29.35 | ZX81 | & |
| 05:29.37 | Corydon76-home | They are not errors, they aren't even warnings... |
| 05:29.42 | ZX81 | allow=alaw |
| 05:29.43 | Qwell | got me... |
| 05:29.44 | ZX81 | allow=ulaw |
| 05:29.46 | Pikoro | i know they're debug statements |
| 05:30.01 | Pikoro | i think perhaps my sip provider doesn't support that? |
| 05:30.01 | Corydon76-home | Turn... off... the debug |
| 05:30.13 | Pikoro | that will make the call go through? |
| 05:30.17 | *** join/#asterisk jets (n=jets@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 05:30.18 | Pikoro | turning off debug? |
| 05:30.32 | Qwell | Pikoro: turning off debug will make debug messages go away |
| 05:30.38 | Corydon76-home | Are you intentionally ignoring the statement in logger.conf concerning debug messages? |
| 05:30.41 | jets | mmmmmm! |
| 05:30.42 | Pikoro | my outbound SIP calls don't work but i can receive calls |
| 05:30.49 | jets | Gorgeous little devils! |
| 05:30.50 | *** join/#asterisk JonR800 (i=jon@p1mp.org) |
| 05:30.53 | *** part/#asterisk JerJer[mobile] (n=jj@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 05:30.54 | Pikoro | so i turned on debug to try to figure out why |
| 05:30.55 | jets | Who's at the hyatt already HMMMM? |
| 05:31.07 | ZX81 | jets: try #astricon |
| 05:31.13 | Corydon76-home | Pikoro: are you a coder? |
| 05:31.23 | pooh_ | ZX81: alsa and oss are NOT loaded |
| 05:31.37 | Pikoro | yah.. but not an asterisk coder |
| 05:32.14 | marc324 | how do you manage large sip users, without a database |
| 05:32.16 | Corydon76-home | The debug messages are meant for programmers. If we thought it was a serious problem, it would be an ERROR or a WARNING, not a debug. |
| 05:32.22 | Corydon76-home | Hence, ignore it. |
| 05:32.28 | Pikoro | ok |
| 05:32.31 | *** join/#asterisk docE (n=docE@206.165.75.198) |
| 05:32.44 | Qwell | pooh_: I think you can do show modules, or something, to see what modules are loaded |
| 05:32.54 | Pikoro | ok, then any suggestions on why inbound sip would work but not outbound? |
| 05:32.56 | Qwell | I'm not at an asterisk box right now, so I can't check the exact command |
| 05:32.58 | Pikoro | a first place to look perhaps? |
| 05:33.03 | Corydon76-home | Are you behind a NAT? |
| 05:33.11 | Pikoro | my client is, the asterisk box is not |
| 05:33.11 | pooh_ | Nope, local LAN setup |
| 05:33.27 | Corydon76-home | What are you trying to contact? |
| 05:33.37 | Corydon76-home | when you make an 'outbound' call? |
| 05:33.50 | Pikoro | anything. another sip number, a pstn number... |
| 05:33.53 | Pikoro | internal works great |
| 05:33.57 | Pikoro | inbound sip works great |
| 05:34.13 | Corydon76-home | Define inbound |
| 05:34.17 | Pikoro | outbound gets me the sip provider operator recording saying all circuts are busy |
| 05:34.26 | Pikoro | someone calls my sip number from the pstn |
| 05:34.33 | Corydon76-home | Every call is composed of both an inbound and an outbound leg |
| 05:34.33 | pooh_ | Qwell: show modules shows no oss or alsa |
| 05:34.54 | Corydon76-home | Then you probably have an error in your Dial line |
| 05:35.06 | Pikoro | ok.. so outbound routing? |
| 05:35.17 | Pikoro | would be a good place to start |
| 05:35.38 | Corydon76-home | Does your provider have a suggestion for Asterisk configuration? |
| 05:35.52 | Pikoro | no, they "do not support" anything but their hardware |
| 05:36.09 | Pikoro | i'm in japan.. they're funny like that |
| 05:36.23 | Corydon76-home | Then you're going to have to try several variations of the SIP Dial line |
| 05:36.37 | Corydon76-home | Tweak it until it works |
| 05:36.46 | Pikoro | like i had to do with the register line :D |
| 05:36.49 | Pikoro | k |
| 05:36.57 | Pikoro | i appreciate a good starting place :D |
| 05:37.14 | Pikoro | you mean dial patterns right? |
| 05:37.18 | Corydon76-home | You might have to run a 'sip debug' to find out where in the SIP conversation the problem arises |
| 05:37.25 | Pikoro | the whole NXXXX stuff? |
| 05:37.49 | Corydon76-home | No, the arguments to Dial |
| 05:37.54 | Ambrose | Anybody know if one can limit the voicemail message length? My FXO isn't detecting hangups and I keep getting 15 minute voicemails (14.5 minutes of silence :p) |
| 05:38.06 | Corydon76-home | The NXXXX stuff is all local to your machine |
| 05:38.26 | Corydon76-home | Ambrose: maxmessage in voicemail.conf |
| 05:39.04 | Ambrose | Corydon-w: Ok thanks |
| 05:39.42 | Corydon76-home | and second, when you're done with VoiceMail(), do a Hangup() |
| 05:39.55 | Ambrose | Yeah I've already go the Hangup in there |
| 05:40.00 | Corydon76-home | otherwise with an i extension, you might create an infinite loop |
| 05:40.17 | Corydon76-home | Or t extension, rather |
| 05:40.22 | Corydon76-home | Damn, it's late |
| 05:40.32 | Ambrose | Yeah I've had the looping problem before :p |
| 05:41.10 | *** join/#asterisk IPmonger (n=ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net) |
| 05:44.35 | Pikoro | arguments to dial? |
| 05:45.05 | Pikoro | ok, which conf file and i'll figure it out from there.. don't wanna keep buggin you |
| 05:47.53 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
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| 05:58.23 | marc324 | ~docs |
| 05:58.25 | jbot | methinks docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 06:12.31 | kaldemar | hello. |
| 06:13.08 | kaldemar | would anyone know a way to continue dialplan processing after app dial? |
| 06:13.21 | kaldemar | ,,g won't do because it only continues if the remote side hangs up. |
| 06:13.33 | wunderkin | m runs a macro on answer |
| 06:14.33 | kaldemar | and that macro continues to run after the call has been hung up? |
| 06:15.41 | wunderkin | g is for hangup |
| 06:15.45 | wunderkin | what do you want? |
| 06:15.55 | *** join/#asterisk oej (n=Olle@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 06:16.58 | kaldemar | i want to remove a variable from asterisk's database after the call is done. |
| 06:17.23 | wunderkin | why dont you make an h exten then |
| 06:18.29 | kaldemar | have to take a look at that. |
| 06:21.53 | kaldemar | i think i can pull it off with the h extension. a million thanks for that. |
| 06:25.19 | wunderkin | sure |
| 06:25.31 | *** join/#asterisk citats (n=james@bgp925576bgs.brghtn01.mi.comcast.net) |
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| 06:44.47 | *** join/#asterisk invi_ (n=invi_@dsl-crow-209-5-162-155-cgy.nucleus.com) |
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| 06:51.28 | invi_ | latest * source & 2.6.12-1.1378_FC3 > make install gives 10 warnings on "Input/output error" |
| 06:51.43 | invi_ | any idea? |
| 06:55.05 | *** part/#asterisk rajiv (n=irc@gentoo/developer/rajiv) |
| 07:04.24 | *** join/#asterisk MuppetMaster (n=MuppetMa@81.184.73.169) |
| 07:05.26 | MuppetMaster | Hello |
| 07:09.36 | *** join/#asterisk uter (n=fn@213.178.78.120) |
| 07:10.02 | uter | moin |
| 07:10.31 | MuppetMaster | Got my book yesterday: http://www.oreilly.com/catalog/asterisk/index.html |
| 07:10.35 | MuppetMaster | It seems to be quite good. |
| 07:10.44 | MuppetMaster | Both for newbies and as a reference manual for the more experienced. |
| 07:10.51 | uter | 9:10am here, so midnight over there? |
| 07:11.51 | *** join/#asterisk _omer (i=p@203.215.180.250) |
| 07:11.53 | _omer | hi |
| 07:12.48 | MuppetMaster | Where is over there? |
| 07:12.53 | MuppetMaster | California? |
| 07:13.17 | Juggie | its 12:10yes |
| 07:13.49 | wunderkin | sync your clock juggie |
| 07:13.55 | wunderkin | :P |
| 07:13.55 | _omer | anyone who could tell me ..where do I get "YUM" from ? |
| 07:15.30 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 07:15.38 | puzzled | morning |
| 07:16.23 | invi_ | anybody from dev here? |
| 07:17.32 | invi_ | or does anybody know if * runs on 2.6.12-1.1378_FC3 ? |
| 07:18.10 | pooh_ | _omer: waht distro? |
| 07:18.36 | _omer | RH9 |
| 07:19.53 | pooh_ | _omer: rpmfind.net |
| 07:20.03 | _omer | thanks.... |
| 07:20.06 | pooh_ | np |
| 07:21.15 | pooh_ | _omer: http://dag.wieers.com/packages/yum/yum-2.0.7-3.0.rh9.test.noarch.rpm |
| 07:21.24 | pooh_ | more easy |
| 07:21.50 | Qwell | invi_: it'll run on just about anything as long as it's 2.4 or above |
| 07:21.54 | *** part/#asterisk MuppetMaster (n=MuppetMa@81.184.73.169) |
| 07:22.06 | _omer | pooh_: thanks...:) |
| 07:33.03 | pooh_ | Hi, anybody knows why I have no sound on my * box when /dev/zap devices are PRESENT, if I remove those, I have sound again |
| 07:33.18 | *** join/#asterisk tin2 (n=abc@203.143.170.32) |
| 07:34.24 | *** join/#asterisk djin (n=djin@213-132-172-4.multikabel.nl) |
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| 07:58.43 | _omer | I tried to compile Asterisk ...zaptel and libpri were successful but go this error when I tried make install in "asterisk" |
| 07:58.44 | _omer | configure: error: termcap support not found |
| 07:58.45 | _omer | make: *** [editline/libedit.a] Error 1 |
| 07:58.48 | _omer | anyhelp??? |
| 07:59.25 | *** join/#asterisk christo (n=chris@office.enovi.com) |
| 08:11.55 | *** join/#asterisk tobiasWolf (n=konversa@195.162.255.10) |
| 08:12.04 | *** join/#asterisk pwnobody (n=chatzill@p54A7EBF8.dip.t-dialin.net) |
| 08:13.15 | *** join/#asterisk Laibsch (n=Laibsch@p54B98501.dip0.t-ipconnect.de) |
| 08:13.49 | pwnobody | Hello, I need some help, I have it set up to always record incoming and outgoing calls on my extensions, how do I access these recordings |
| 08:14.24 | Laibsch | Hi, I was wondering if I can set up * for conference call via VoIP only. The information I found on the web regarding meetme was inconsistent. |
| 08:15.14 | Laibsch | I want to use * as a server for hosting conference calls on the Intranet. |
| 08:15.49 | Delvar | Laibsch: yes you can |
| 08:16.36 | Delvar | Laibsch: you will need a server with either a digium card or usb controler (i cant remember which type) to get timing |
| 08:18.41 | pooh_ | _omer: http://snowbird-linux.com/rh9/RPMS/termcap-11.0.1-16.noarch.rpm |
| 08:20.43 | uter | i have problems with my zaptel. insmod fails: |
| 08:20.46 | uter | insmod: error inserting '../zaptel-1.0.9.2/zaptel.ko': -1 Invalid module format |
| 08:21.05 | uter | and dmesg says the following: |
| 08:21.09 | uter | zaptel: version magic '2.6.11 686 gcc-4.0' should be '2.6.11-1-686 686 gcc-3.3' |
| 08:21.46 | uter | i found a lot on this at google, but no solution |
| 08:22.44 | *** join/#asterisk deployer (n=ice_d@h10-n63-91-81.zra.ru) |
| 08:24.38 | *** join/#asterisk folsson (n=filip@h147n1fls32o985.telia.com) |
| 08:26.17 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 08:26.19 | *** join/#asterisk contrabanda (n=G@213.131.37.202) |
| 08:26.23 | contrabanda | hi all |
| 08:26.59 | contrabanda | is it good to use sangoma cards in asterisk? |
| 08:27.07 | contrabanda | or its better to use digium? |
| 08:27.20 | Laibsch | Delvar: Thank you for your quick response. I will try to get this going as a test case. Still very new to VoIP. |
| 08:27.20 | X-Rob | sangoma cards will be fully supported in openpbx |
| 08:27.43 | X-Rob | but for asterisk, use a digium card. |
| 08:28.46 | contrabanda | X-ROb: in sangoma documentation, they write that it can work with asterisk |
| 08:29.00 | *** join/#asterisk isam (n=isam@213.186.188.175) |
| 08:29.04 | isam | hi all.. |
| 08:29.12 | deployer | hi |
| 08:29.17 | isam | in extentions.conf.. how can I tell asterisk to take a literal @ |
| 08:29.31 | isam | and not parse it as a host@ |
| 08:29.37 | alphaque | contrabanda: sangoma cards work with asterisk as well |
| 08:30.29 | isam | I have a sip username what comes with an @.. and I am not able to test it as it can't take numer@something@SIP_PROXY/${EXTEN} |
| 08:30.48 | isam | it is trying to connect to @something@SIP_PROXY |
| 08:30.55 | contrabanda | alphaque: but why its better to use digium andd not sangoma? |
| 08:31.03 | isam | while I want it to take the first @ as part of the username |
| 08:31.38 | alphaque | contrabanda: i didnt say one was better than the other, or otherwise. all i said was that /both/ could be used. |
| 08:31.50 | isam | talking about the fork |
| 08:32.43 | contrabanda | alphaque: so its a good solution to use sangoma? |
| 08:32.51 | joelsolanki | Hello all: we need to setup inbound calls. means i want to provide usa local number to my country on cisco ata box. currently for demo purpose it working. service provider is using cisco call manager and we have ata registered to call manager. |
| 08:33.23 | alphaque | contrabanda: i really wouldnt know, given that i dont know what u're using it for. and even if i did, i'd still think u'd need to get that evaluation done urself. |
| 08:33.44 | joelsolanki | I want to know is that possible that my provider routes the thing on my asterisk box and then ata box connects to asterisk and incoming works? |
| 08:33.54 | joelsolanki | any body have this type of setup ? |
| 08:34.19 | *** join/#asterisk x-router (i=richard@uberpussy.net) |
| 08:34.56 | *** join/#asterisk \PsyKo\ (n=xxxxxxxx@golia.caltanet.it) |
| 08:36.06 | contrabanda | ok 10xs |
| 08:37.04 | joelsolanki | Any hints ? |
| 08:40.51 | *** join/#asterisk newmember (n=newmembe@S010600a0c93dce87.cg.shawcable.net) |
| 08:41.21 | meppl | guten morgen |
| 08:42.06 | johnm | Moghrey mie |
| 08:42.19 | pwnobody | I have my * extensions set up to record all incoming and outgoing calls, How do I access these recordings? |
| 08:48.17 | _omer | default folder is /var/spool/asterisk/monitor for recorded calls |
| 08:48.57 | *** part/#asterisk KaBewM (n=kabewm@66-215-7-106.dhcp.psdn.ca.charter.com) |
| 08:53.03 | pwnobody | _omer, thank you, I am using putty to access my * box and can see the wav files. Now what would be the easiest way to transfer them to my workstation? I am using *@home |
| 08:53.38 | pwnobody | I have never used linux before, and I am just starting out, so please excuse me if it is super simple. |
| 08:54.36 | pwnobody | ls | less |
| 08:54.39 | pwnobody | oops |
| 08:54.54 | *** join/#asterisk etiennesw (n=etienne@194.204.96.50) |
| 08:55.06 | etiennesw | hello |
| 08:55.49 | etiennesw | Can anyone help me with Asterisk queues |
| 08:56.26 | contrabanda | whats better for asterisk with havy load, Intel processor or AMD 64? |
| 08:57.28 | uter | pwnobody does putty support scp? |
| 08:58.03 | kaldemar | putty scp, aka pscp.exe |
| 08:58.17 | johnm | contrabanda: moreso than anything else... lots of RAM and decent kit. But if your talking something like dual-core amd vs. say, Dual Xeons fex, then go for the Multi-Proc dual-core stuff. |
| 08:58.34 | *** join/#asterisk LANmower (n=LANmower@ndn-165-136-69.telkomadsl.co.za) |
| 08:58.39 | LANmower | lo all |
| 08:59.25 | LANmower | the asterisk-addons via cvs looks different than the downloaded 1.0.9 one, for instance the downloaded one fails its compile complaining about logger.h |
| 08:59.40 | LANmower | any tips on getting it compiled? |
| 08:59.42 | pwnobody | Ok, it seems yes, it is another part, pscp like kaldemar said |
| 09:00.31 | contrabanda | ok |
| 09:01.19 | pwnobody | uter, though I have no clue how to use that |
| 09:01.24 | etiennesw | I have setup queues and agents, however, when a new call comes in, the agents who are alreading on a call get their phone ringing again. How can I limit this. They are using softphones (Iaxcomm) |
| 09:01.39 | uter | pwnobody so try to copy the files with scp, i dont really know the exact syntax |
| 09:02.20 | *** join/#asterisk Eight (n=blake@12-227-171-175.client.mchsi.com) |
| 09:02.21 | pwnobody | uter, ok, I just tried to run it from windows, but it just closed, so going to cmd now, see what it says |
| 09:02.27 | uter | it should be something like pscp root@asteriskbox:/var/spool/asterisk/monitor/* . |
| 09:03.06 | LANmower | er |
| 09:03.18 | LANmower | how can I make asterisk addons see the asterisk directory? |
| 09:03.23 | *** join/#asterisk etiennesw (n=etienne@194.204.96.50) |
| 09:03.24 | LANmower | trying to make it |
| 09:03.39 | LANmower | oh found it |
| 09:05.25 | pwnobody | uter, Thank you, it seems that would be simple, though what would I use as the target? |
| 09:06.13 | pwnobody | everything that I have tried results in a "local to local copy not supported" |
| 09:06.14 | uter | . |
| 09:06.31 | *** join/#asterisk vandien (i=sted@aditu.dahltronics.de) |
| 09:06.56 | uter | try "." as target |
| 09:07.10 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 09:07.39 | deployer | тут есть русско-говорящие? |
| 09:07.58 | pwnobody | uter, Same result |
| 09:10.10 | etiennesw | Any help with the queues..pls |
| 09:12.53 | uter | pwnobody hmm |
| 09:13.34 | uter | does pscp --help provide something useful? |
| 09:13.49 | LANmower | I'm having trouble compiling asterisk-addons ;( |
| 09:13.55 | LANmower | anyone wanna help me? |
| 09:15.13 | etiennesw | ..or maybe does anyone know how to use the SetVar within the queues.conf ?? |
| 09:16.31 | deployer | have a problem with adjustment H323 on asterisk |
| 09:16.31 | deployer | dlink fxo h323 is not registerred |
| 09:16.52 | LANmower | anyone? |
| 09:16.54 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 09:17.09 | deployer | dlik sip - orderly works |
| 09:17.36 | pwnobody | uter, I didn't figure that out, though I did get filezilla to connect to it via SFTP using SSH2 |
| 09:17.58 | LANmower | I see there is an include path in the makefile... can someone tell me where it should point? |
| 09:18.23 | pwnobody | It seems that with the pscp that the only problem that I was having is the target, I will have to look this up for future reference |
| 09:18.31 | pwnobody | Thank you for your help |
| 09:19.44 | iDunno | if you're windows bound, then you might want to get winscp |
| 09:19.54 | iDunno | which is graphical and lets you browse around |
| 09:19.57 | *** join/#asterisk eivindtr (n=wingnut-@static-213-115-144-116.sme.bredbandsbolaget.se) |
| 09:19.58 | LANmower | ... anyone? |
| 09:19.59 | uter | np |
| 09:20.53 | *** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
| 09:21.11 | Zeeek | morning |
| 09:22.33 | LANmower | as expected from irc... no help |
| 09:22.34 | pwnobody | iDunno, sounds good, that's why I tried to get Filezilla to work as I use it all the time, I just didn't have the servertype set correctly, I was trying to use quickconnect |
| 09:23.07 | iDunno | ahh - I think filezilla will do sftp, won't it? |
| 09:23.23 | Zeeek | yes |
| 09:23.25 | pwnobody | yes |
| 09:23.43 | pwnobody | that's how I set it up now, but using the quickconnect feature it doesn't automatically try that |
| 09:24.44 | deployer | can real asterisk to work as gateway between H323 and SIP |
| 09:24.54 | deployer | ? |
| 09:24.56 | invi_ | how do i check zaptel version? |
| 09:26.58 | deployer | It to me? |
| 09:29.30 | LANmower | er |
| 09:29.32 | LANmower | yeah |
| 09:29.43 | LANmower | thanx for nothing, dunno why I even tried |
| 09:30.09 | Zeeek | lancey`away repeat the question |
| 09:30.17 | *** join/#asterisk fulgas (n=fulgas@213.58.130.46) |
| 09:30.26 | etiennesw | Please, can anyone help with the queues.conf so that bust agents won't be dialed twice?? |
| 09:30.31 | *** join/#asterisk planet_guru (n=chris@195.82.114.14) |
| 09:30.55 | *** join/#asterisk burton (i=mimx@w201.ljudmila.org) |
| 09:35.47 | riksta | is there a way to find the amount of total calls via the manager interface easily? |
| 09:36.46 | Zeeek | I usually use the cdr for that |
| 09:37.02 | riksta | how do you determine it from that? |
| 09:37.15 | riksta | from the ZapX ? |
| 09:37.26 | Zeeek | wait you mean total calls for one day, a period of time or one instant now? |
| 09:37.34 | riksta | no no live at this instant |
| 09:37.40 | riksta | obviously i use the CDR for the rest |
| 09:37.46 | Zeeek | something like show channels? |
| 09:38.04 | riksta | yeah but i just wanted to return a number really, ill just iterate through that lot |
| 09:38.06 | riksta | ta |
| 09:38.25 | Zeeek | I sometimes use the commands and parse the output |
| 09:38.31 | riksta | thats what ill do too |
| 09:38.40 | Zeeek | like for peers, I like to seethem listed in order of lag, low to high |
| 09:39.06 | Zeeek | and show the unreachables in grey |
| 09:39.12 | Zeeek | etc etc |
| 09:39.13 | *** join/#asterisk nextime (n=nextime@213-140-6-96.ip.fastwebnet.it) |
| 09:39.19 | riksta | sure |
| 09:40.09 | *** join/#asterisk macTijn (i=martijn@linda.net.insecure.nl) |
| 09:40.14 | invi_ | how do i check zaptel version? |
| 09:41.16 | Zeeek | let me know if you find out |
| 09:41.20 | *** join/#asterisk w14 (n=asterisk@62.140.193.212) |
| 09:43.09 | invi_ | cmon u useless kunts, talk to me |
| 09:43.35 | Zeeek | more flies with honey and all that |
| 09:43.45 | *** join/#asterisk folsson (n=filip@h147n1fls32o985.telia.com) |
| 09:44.02 | etiennesw | hey Zeeek, can you help me out?? |
| 09:44.15 | Zeeek | I dunnon, what's the problem? |
| 09:45.27 | etiennesw | I am using the queues.conf to call 4 agents, however, when a new call come in, it rings all 4 agents, even those that a currently on a call....can I limit this? |
| 09:45.45 | etiennesw | ..to the free agents only? |
| 09:56.23 | *** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 10:00.30 | Zeeek | isn't that part of the definiton of the queue itself? |
| 10:01.19 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
| 10:06.16 | etiennesw | the queue defines the agents etc......but I dont want the queue to call an agent that is busy |
| 10:07.14 | *** join/#asterisk mithro (n=tim@tagung-233-167.tagung.uni-hamburg.de) |
| 10:07.18 | etiennesw | in the extensions.conf you can do that by using SetVar and mark the extension as busy.........but how can u do that in the queues.conf?? |
| 10:08.15 | deployer | Prompt. Whether there is what that subtleties in adjustment H323 in asterisk. I can not achieve, that dlisk H323 it would be registered. With SIP problems have not arisen |
| 10:09.54 | deployer | I am sorry for bad English. |
| 10:10.40 | deployer | dlisk - read - d-link :) |
| 10:18.00 | *** join/#asterisk newl (n=newlook@203-59-214-216.dyn.iinet.net.au) |
| 10:24.17 | *** join/#asterisk frenzy (n=frenzy@193.220.82.108) |
| 10:26.23 | Zeeek | etiennesw what about just turning off callwaiting? |
| 10:34.37 | *** join/#asterisk oob (n=oob@219-89-58-172.dialup.xtra.co.nz) |
| 10:37.43 | *** join/#asterisk venkat (n=venkat@212.159.2.3) |
| 10:42.56 | *** join/#asterisk [Jedi] (n=hhgds4@213.162.200.226) |
| 10:42.57 | [Jedi] | Hello |
| 10:43.08 | [Jedi] | Anyone could help me with TxFax please? :( |
| 10:43.20 | [Jedi] | anyone uses spandsp here? |
| 10:43.56 | Zeeek | only for receiveing and then onluy for spam faxes |
| 10:45.59 | *** join/#asterisk Akelavlk (n=jansun@82.119.239.141) |
| 10:47.36 | *** join/#asterisk skrusty (i=muad@xdev.net) |
| 10:48.01 | skrusty | can anyone tell me what the problem was in this example in pastebin (it's exactly the same error as im getting): http://de.pastebin.ca/15588 |
| 10:48.12 | skrusty | i have freetds-dev installed |
| 10:48.24 | Akelavlk | Hello, is there any good ISDN hardware for Asterisk? |
| 10:49.16 | johnm | Akelavlk: the digium wildcards are very good for PRI |
| 10:49.44 | Akelavlk | You mean TDM400P for example? |
| 10:50.19 | johnm | I use the te405p, te205p etc. but yes |
| 10:50.40 | *** join/#asterisk ZX81 (n=ZX81@222-153-100-242.jetstream.xtra.co.nz) |
| 10:51.04 | ZX81 | what's up with this: Restricting registration for peer '6838601' to 60 seconds (requested 30) |
| 10:51.12 | ZX81 | anyway to remove it? |
| 10:51.25 | Akelavlk | johnm, So I just plug ISDN cable into TDM400P? Are you sure. I thought that analog phone has diffrent things than ISDN |
| 10:52.34 | *** join/#asterisk venkat (n=venkat@212.159.2.3) |
| 10:52.53 | skrusty | anyone here know anything about cdr_tds? :) |
| 10:53.19 | johnm | Akelavlk: im not sure about the TDM400P, is it an ISDN2 or an ISDN30? |
| 10:53.37 | [Jedi] | TDM400P is not ISDN I think |
| 10:54.00 | Akelavlk | johnm, I am talking about this card http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TDM400P |
| 10:54.20 | johnm | Akelavlk: what do you have ISDN wise which you need to support? |
| 10:55.40 | Akelavlk | johnm, I need buy hardware for asterisk.. Because comapny is switching to ISDN from analog. |
| 10:55.55 | johnm | Akelavlk: but what kind of ISDN connection are they getting? |
| 10:56.19 | Akelavlk | johnm, ISDN2. |
| 10:56.44 | johnm | Akelavlk: then you want a BRI card. check out Junghanns.net |
| 10:57.08 | Akelavlk | johnm, I know about their products.. |
| 10:57.37 | Akelavlk | But, do you recommend that hardware? May be it's peace of sht. |
| 10:58.27 | johnm | Nah, the junghanns stuff is pretty good. And on top of that I dont know of anything which digium produce which is capable of BRI |
| 10:58.36 | johnm | at least I've only ever used Junghanns for BRI |
| 10:58.46 | johnm | if you're worried, get it on a 30 day evaluation |
| 10:59.19 | Akelavlk | Great.. |
| 10:59.27 | Akelavlk | It sounds good.. |
| 11:01.27 | [Jedi] | I'm lost with spandsp |
| 11:01.36 | [Jedi] | Anyone ever used brooktrout software? |
| 11:01.39 | [Jedi] | sorry hardware? |
| 11:04.20 | Akelavlk | johnm, thanks for help.. |
| 11:05.56 | *** join/#asterisk shido6 (n=curtis@d221-68-210.commercial.cgocable.net) |
| 11:07.13 | *** join/#asterisk Asylum (i=Asylum@CPE-60-226-88-2.qld.bigpond.net.au) |
| 11:09.45 | *** part/#asterisk Asylum (i=Asylum@CPE-60-226-88-2.qld.bigpond.net.au) |
| 11:17.09 | *** join/#asterisk MuppetMaster (n=MuppetMa@81.184.73.169) |
| 11:17.13 | MuppetMaster | Hello |
| 11:17.52 | *** join/#asterisk shanky (n=shanky@238.Red-80-33-29.staticIP.rima-tde.net) |
| 11:17.59 | shanky | hi, good afternoon |
| 11:18.09 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
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| 11:21.21 | *** part/#asterisk MuppetMaster (n=MuppetMa@81.184.73.169) |
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| 11:30.33 | *** join/#asterisk zotz (n=zotz@24.231.36.100) |
| 11:34.48 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
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| 11:37.47 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
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| 11:41.59 | *** part/#asterisk shanky (n=shanky@238.Red-80-33-29.staticIP.rima-tde.net) |
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| 12:02.36 | *** join/#asterisk ManxPower (n=eric@adsl-70-247-221-174.dsl.lgvwtx.swbell.net) |
| 12:03.27 | ManxPower | ~mailinglist |
| 12:03.29 | jbot | rumour has it, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 12:03.42 | *** join/#asterisk Nix (n=Nix@81.214.255.57) |
| 12:03.55 | RoyK | ~lart himself |
| 12:07.11 | *** join/#asterisk tatuman (n=Miranda@joltid-gw.joltid.org) |
| 12:07.30 | tatuman | hi |
| 12:07.33 | *** join/#asterisk Tili (n=Tili@202-133-65-156-dialup.sat.net.pk) |
| 12:07.48 | tatuman | anyone knows where i can read more about meetme app design? |
| 12:07.58 | tatuman | the code is not so comented |
| 12:08.14 | ManxPower | tatuman, I thnik the code is the only place. |
| 12:08.23 | tatuman | :S |
| 12:08.35 | ManxPower | But you might consider looking at the asterisk-dev mailing list archive. |
| 12:08.58 | tatuman | anyone have looked extensively to this code? |
| 12:08.59 | etiennesw | PLEASE HELP ----- HOW CAN I MARK AGENTS WHO ARE BUSY SO THAT THE QUEUE DOES NOT SEND THEM NEW CALLS?? |
| 12:09.35 | ManxPower | etiennesw, You can't, as far as I know. The agent has to log off the queue if they don't want calls. |
| 12:09.42 | tatuman | ManxPower: hav u ever worked with meet me ? |
| 12:09.54 | ManxPower | tatuman, not on the code. |
| 12:10.12 | tatuman | do u know anyone who done it? |
| 12:10.35 | ManxPower | tatuman, bugs.digium.com would have bug reports and patches for MeetMe. |
| 12:10.49 | tatuman | ok, i will give it a look |
| 12:10.51 | ManxPower | anyone that has submitted a patch has looked at the code. |
| 12:10.51 | tatuman | thanks |
| 12:12.03 | tatuman | yeah :), i wanted to see how good it scales and to have multiple asterisk providing conferences, but with a central control of all the servers |
| 12:12.27 | tatuman | do u know if is there any work in this area? |
| 12:12.45 | ManxPower | tatuman, I don't know. |
| 12:12.54 | ManxPower | You can ask on the -users or -dev mailing lists |
| 12:13.11 | tatuman | ok,thanks |
| 12:13.24 | kaldemar | etiennesw: maybe you could use RemoveQueueMember and AddQueueMember? |
| 12:13.50 | etiennesw | ManxPower, but if he is busy handling a call, I want the queue to dial the other 3 agents who are free |
| 12:15.07 | ManxPower | etiennesw, Um, that is the way queues work |
| 12:16.10 | etiennesw | Kaldemar, I looked into that but do you know to to put it together since once the busy agent is off the line I want him to be back available in the queue |
| 12:16.50 | *** join/#asterisk mmmToop (n=chatzill@196.31.11.194) |
| 12:17.12 | ManxPower | etiennesw, It sounds like you have call waiting enabled on your phones. |
| 12:17.24 | etiennesw | ManxPower, maybe you know a way to limit the max number of simultanious calls from the iax.conf or extensions.conf? |
| 12:17.27 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
| 12:17.33 | ManxPower | IF that is the case, then the queue doesn't know the person is busy and will try to send calls to the person |
| 12:18.18 | etiennesw | the phones are softphone (Iaxcomm).....but there doesn't seem to be a way to limit from the client side |
| 12:19.04 | *** join/#asterisk Jzalae (n=sk@216-220-248-166.midmaine.com) |
| 12:25.35 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:27.48 | *** part/#asterisk Akelavlk (n=jansun@82.119.239.141) |
| 12:28.07 | *** join/#asterisk Mimmus (n=viggiani@ext.pitagora.it) |
| 12:29.45 | Mimmus | hi, can someone help me with MeetMe? |
| 12:30.40 | *** join/#asterisk RandomAndy (n=randoman@adsl-64-161-212-104.dsl.snfc21.pacbell.net) |
| 12:40.40 | *** join/#asterisk ^X-works (n=drttrtr@81-208-62-98.ip.fastwebnet.it) |
| 12:40.57 | Zeeek | ManxPower longtime no C |
| 12:41.15 | Zeeek | glad you survived |
| 12:44.16 | *** join/#asterisk lehel (n=asd@82.79.20.17) |
| 12:45.02 | *** join/#asterisk Laibsch (n=Laibsch@p54B98501.dip0.t-ipconnect.de) |
| 12:45.11 | *** part/#asterisk w14 (n=asterisk@62.140.193.212) |
| 12:46.26 | lehel | hello |
| 12:52.27 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 12:53.40 | *** join/#asterisk Tili (i=Tili@202-133-65-162-dialup.sat.net.pk) |
| 12:57.35 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 12:58.35 | *** join/#asterisk folsson (n=filip@h147n1fls32o985.telia.com) |
| 12:58.59 | Laibsch | I hope this question is not off-topic, but what is a good client (soft phone) for VoIP for a Windows client? |
| 12:59.16 | Zeeek | X-Ten X-Lite |
| 12:59.20 | Laibsch | SIP preferred but IAX possible |
| 12:59.27 | Laibsch | Zeeek: OK, thank you. |
| 12:59.32 | Zeeek | google asterisk iax windows client |
| 12:59.40 | Zeeek | wiki |
| 12:59.47 | Zeeek | IAXPhone |
| 12:59.51 | RoyK | ~zeeek? |
| 12:59.52 | jbot | zeeek is probably someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 12:59.52 | Zeeek | DIAX |
| 12:59.54 | Zeeek | Firefly |
| 13:00.08 | Zeeek | very true |
| 13:00.18 | Laibsch | I found X-Ten a bit hard to configure. |
| 13:00.24 | RoyK | rotf |
| 13:00.48 | Laibsch | It is not the most intuitive if you are new to VoIP. |
| 13:00.51 | Zeeek | what's good about X-Ten is youlearn something from it |
| 13:01.16 | Laibsch | Good, I am always eager to learn. |
| 13:01.31 | Zeeek | they used to have an extensive manual that explained every setting |
| 13:01.33 | *** join/#asterisk Laerte (n=io@195.47.232.200) |
| 13:01.36 | Laerte | hy |
| 13:02.24 | Laibsch | Zeeek: That is what I mean. At first, I do not want to know everything. I want to get it up and running quickly, play around and then tweak it. |
| 13:02.49 | Laibsch | So a good, comprehensive manual is VERY nice |
| 13:03.16 | Laibsch | but a "how to VoIP with $SW in 5 minutes" is also very important to have. |
| 13:03.37 | Zeeek | there are a millions pages on the web written about this |
| 13:03.57 | Zeeek | here are four of them |
| 13:03.58 | Zeeek | Starter tutorial: |
| 13:03.58 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 13:03.58 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 13:03.59 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 13:03.59 | Zeeek | THE reference of the moment: |
| 13:04.00 | Zeeek | http://www.asteriskdocs.org |
| 13:04.06 | InfraRed | you want support, pay for support |
| 13:04.14 | Laibsch | Zeek: Thank you. |
| 13:04.15 | InfraRed | you want to google, it's free |
| 13:04.19 | Laibsch | I read those page. |
| 13:04.31 | Zeeek | if you read those pages, you already know the answers |
| 13:04.40 | Laibsch | InfraRed: I am considering payment seriously. |
| 13:04.42 | RoyK | ~docs |
| 13:04.44 | jbot | i guess docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 13:05.30 | Zeeek | why would a BT100 not answer OPTIONS? or did they never do so? |
| 13:05.57 | *** join/#asterisk AsterNov (i=richard_@82-45-239-141.cable.ubr01.enfi.blueyonder.co.uk) |
| 13:06.07 | pooh_ | RoyK: can you check my * server pls? |
| 13:06.14 | pooh_ | I think it is ok now |
| 13:06.36 | *** part/#asterisk Craziman2 (n=Craziman@63.108.128.250) |
| 13:06.45 | Laibsch | But I need to get up a test case to play around and find if it can get the job done. I am sure * is fine. But today I was amazed to see that most Windows VoIP softphones (I am on Debian myself) are only capable of connecting to the service it was designed for (pulver.communicator, for example cannot connect to Asterisk, if I am not mistaken). |
| 13:07.08 | Zeeek | you are mistaken |
| 13:07.29 | Laibsch | Wow, glad to hear. |
| 13:07.32 | Zeeek | someone has posted a full photo tutorial on FWD forum about configuring it |
| 13:07.46 | Zeeek | it seems you haven't looked around much though |
| 13:07.53 | Laibsch | I could not find the option where to configure it for anything but FWD. |
| 13:07.58 | Laibsch | I'll look in the forum. |
| 13:08.14 | Zeeek | all you do is replace the server name |
| 13:08.18 | Laibsch | Believe me, I spent most of yesterday night and all morning. |
| 13:08.22 | Zeeek | and user, password. |
| 13:08.31 | InfraRed | i think i found a cool hotel in nyc |
| 13:08.33 | InfraRed | http://www.gershwinhotel.com/ |
| 13:08.34 | Zeeek | is ENglish your native toungue? |
| 13:08.54 | Zeeek | did you get X-Lite working with FWD? |
| 13:09.44 | *** join/#asterisk jeffik (n=Jeff@CPE0020ed8494b8-CM0012c999ca4e.cpe.net.cable.rogers.com) |
| 13:10.08 | Laibsch | No, english is not my native tongue. |
| 13:10.15 | Laibsch | I got X-Lite working with FWD. |
| 13:10.18 | Zeeek | well that explains part of it |
| 13:10.26 | Laibsch | I found said option now. |
| 13:10.41 | Zeeek | sop what's the big deal? Youc an see the FWD specific stuff I mentioned above, then change it to what you need |
| 13:10.53 | Laibsch | It is greyed out and I missed the online doc part OR maybe I thought this makes it payware and did not proceed further. |
| 13:11.00 | Zeeek | if you don't know how to set up asterisk for a user account, you have not read the docs I posted |
| 13:11.25 | Zeeek | look at this |
| 13:11.26 | Laibsch | I think I have two user accounts running. |
| 13:11.28 | Zeeek | http://www.plus.net/support/adsl/plustalk_support/plustalk_guides/plustalk_setup_guide.shtml?pn_session=4fab12ee27d08743084f97862af7086f |
| 13:11.51 | Laibsch | Zeeek: Thank you. I think I will manage from here. |
| 13:11.52 | Zeeek | substitue "plustalk" for "myserver" |
| 13:12.06 | Laibsch | Thanks for all the references. |
| 13:12.12 | pooh_ | damn: Unable to open channel 1: Device or resource busy |
| 13:12.31 | Zeeek | Laibsch np, you just need to pour over all that stuff |
| 13:13.13 | *** join/#asterisk _omer (i=p@203.215.180.250) |
| 13:13.37 | pooh_ | quozap guru around pls ? |
| 13:13.43 | pooh_ | qozap |
| 13:14.12 | *** join/#asterisk eldu (i=damajor@tuxmania.org) |
| 13:15.59 | Zeeek | Anyone using this: http://www.voxbone.com/ |
| 13:18.29 | *** join/#asterisk Cadu20 (n=Cadu20@200.102.53.174) |
| 13:18.53 | lehel | Zeeek: no but tell me about it.. should i use? |
| 13:20.19 | eldu | Their services seems good, but I think that their DID are too expensive. |
| 13:20.44 | AsterNov | I've got asterisk@home running with extensions, but the voicemail says it's writing to the directory, except when I check it's empty. The directory permissions seem ok and belong to asterisk. Anybody got any ideas? |
| 13:22.53 | Zeeek | eldu you used them? |
| 13:23.08 | eldu | no i just contact them for special prices |
| 13:23.10 | Zeeek | I can't find a per minute rate, only setup and monthly |
| 13:23.16 | eldu | but no success :) |
| 13:23.29 | *** join/#asterisk AlexCTI (n=alex@weston-69.65.86.197.myacc.net) |
| 13:23.31 | Zeeek | It would appear that the network infrastructure is very good, this is why I was asking |
| 13:23.44 | eldu | Zeeek: u got inbound with voxbone, not outbound |
| 13:23.52 | Zeeek | can't have interruptions and noise and echo on business phones |
| 13:24.01 | AlexCTI | Hey, some has to use well the raw player to MOH? |
| 13:24.03 | Zeeek | yes, only inbound |
| 13:24.30 | Mimmus | I'm not able to hear any tone when a user join/leave a MeetMe conference even if I use 'i' option |
| 13:24.34 | Mimmus | any help? |
| 13:28.27 | *** join/#asterisk ozant (n=ozan@85.96.199.40) |
| 13:28.36 | skrusty | dont know if anyone can help, just added realtime support to asterisk (and built from cvs) and now i keep getting 'Not a local SIP domain' on all registration requests. Why is that? in the bug tracker, it says this is disabled unless otherwise specified |
| 13:29.48 | ozant | hi all, Can asterisk make difference with a dual CPU? or it just use one? |
| 13:30.52 | johnm | ozant: the way linux works means that applications which aren't written for dual-proc should still function across multi-proc machines and reap the benefits. |
| 13:31.04 | johnm | ozant: generally speaking at least ;) |
| 13:31.13 | johnm | ozant: so yes. |
| 13:32.00 | ozant | johnm umm thanks... |
| 13:33.42 | skrusty | anyone seen that error before? :( |
| 13:34.02 | *** join/#asterisk kippi (n=chrisfro@untrust-gct.equinoxit.net) |
| 13:34.04 | kippi | hey |
| 13:34.21 | skrusty | hi |
| 13:36.05 | *** part/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
| 13:36.20 | Cadu20 | Can't change the transfer digit... no matter what does features.conf says, Asterisk aways assume # as transfer. Why is that? |
| 13:39.01 | kippi | if I add a DID Route with a idsn card, will asterisk work out the DID? |
| 13:41.27 | *** join/#asterisk |cleric| (n=dacleric@p548281C3.dip0.t-ipconnect.de) |
| 13:43.11 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 13:43.30 | *** join/#asterisk gambolputty (n=gambolpu@72.240.241.108) |
| 13:46.58 | Faithful | Anyone in AU got a Sun Ultra 10 they want to get rid of? |
| 13:47.11 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:47.50 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 13:52.34 | Nix | Faithful: I used to have one but I gave it away when I moved from AU :-) |
| 13:53.00 | *** join/#asterisk oej (n=Olle@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 13:53.30 | Lathos42 | I have an Ultra 5, but i'm in the US :) |
| 13:54.46 | *** join/#asterisk shadebob (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma) |
| 13:54.59 | [Jedi] | I have a pair of Ultra 5 and a few Netra's |
| 13:55.04 | [Jedi] | but in .es |
| 13:55.20 | [Jedi] | =D |
| 13:55.33 | file[laptop] | :| |
| 13:56.04 | *** join/#asterisk nomazda (i=nyyankee@user-0c6tnqf.cable.mindspring.com) |
| 13:56.15 | shadebob | Hi, I have a problem with FXO and billing. It seem Asterisk doesn't detect when callee take call... Any ideas? |
| 13:56.18 | skrusty | anyone know why the new domain patch for asterisk would be preventing me from accepting sip registrations? |
| 13:56.25 | shadebob | billing start when phone ring |
| 13:56.30 | skrusty | assuming that's in cvs head now |
| 13:57.11 | *** join/#asterisk Weezey (i=Weezey@206.210.109.232) |
| 13:57.13 | file[laptop] | shadebob: you can't do accurate call progress on analog |
| 13:57.28 | file[laptop] | shadebob: there is an option for it, but your calls might randomly disconnect |
| 13:58.00 | shadebob | file[laptop] : wich option? Silence detection? |
| 13:58.21 | eldu | how can i see the packets lost ? |
| 13:58.24 | eldu | per call |
| 13:58.38 | file[laptop] | shadebob: uh no |
| 13:58.45 | file[laptop] | shadebob: callprogress |
| 13:59.22 | shadebob | file[laptop] : ok I will see |
| 13:59.35 | shadebob | file[laptop] ; thanks |
| 13:59.50 | kippi | i am better off installing asterisk and then installing asterisk at home? or should i just install asterisk? |
| 14:00.01 | kippi | install asterisk at home |
| 14:00.15 | kippi | opps |
| 14:00.21 | kippi | sorry i'll start again |
| 14:00.27 | lehel | kippi: asterisk@home includes asterisk [zzz] |
| 14:00.41 | kippi | i am better off installing asterisk and then installing AMP? or should i just install asterisk at home? |
| 14:01.08 | eldu | i bet is easier to just install A@H |
| 14:01.30 | kippi | A@H will work just aswell? |
| 14:01.30 | lunk | A@H is the easiest |
| 14:01.32 | eldu | AMP needs some skills to be installed smoothly |
| 14:02.13 | kippi | is there a channel for a@h? |
| 14:02.34 | eldu | yes look for #amportal |
| 14:02.52 | eldu | be patient after asking ;) |
| 14:03.05 | lehel | :D |
| 14:03.20 | Faithful | Nix who did you give it to? |
| 14:03.46 | Ariel_ | strange things are happening. A tropical storm with winds of 45 mph has hit the cost of Portugal and Spain. Just strange. |
| 14:04.19 | eldu | any packet lost ? :) |
| 14:05.08 | johnm | Ariel_: not too surprising following on from the earthquake. but in general bad weather and stuff is happening a lot at present |
| 14:05.21 | johnm | Ariel_: equalibrium has to happen I guess. |
| 14:05.25 | Ariel_ | johnm, yes it sure is |
| 14:05.29 | johnm | 45mph is nothing anyways :) |
| 14:06.10 | Ariel_ | johnm, it's not the wind speed but the fact that instead of going accross the pond it just travel north into the EU |
| 14:06.29 | ful^work | is it on the opossite direction |
| 14:06.42 | johnm | Ariel_: all depends on pressure cycles I guess. |
| 14:06.46 | ful^work | well so far it didn't done much damage... |
| 14:06.52 | AlexCTI | that's true.. ask in Fl, USA.. winds more than 90 mph |
| 14:07.05 | Faithful | There will be great earthquakes, famines, and plagues in various places. There will be terrors and great signs... |
| 14:07.37 | Ariel_ | Faithful, I guess you reading our doom and gloom messages. |
| 14:08.18 | Nix | Faithful: to my old flatmate |
| 14:08.29 | *** join/#asterisk mkrufky (n=mk@68.160.103.77) |
| 14:08.51 | Faithful | Nix: got his email... I will buy it off him if it is still going... which city ? |
| 14:08.55 | Nix | Qantas seemed to think that a Sun would put me over my baggage limit :-) |
| 14:09.01 | Faithful | I need one bad |
| 14:09.09 | Nix | I think he gave it away to someone else |
| 14:09.19 | Nix | should be easy to get one on ebay.com.au |
| 14:09.28 | Faithful | Not so... |
| 14:09.33 | shadebob | file[laptop] : for my answering line question it seem to have an answeronpolarityswitch option. Maybe it will be work in my country |
| 14:09.34 | Faithful | But I am trying |
| 14:09.52 | Faithful | I got one already but it was a dud so I sent it back |
| 14:10.06 | johnm | Faithful: Might be an idea to get in touch with the local telco and see if they have any going for sale. or a bank. |
| 14:10.41 | Faithful | There is a co. in Sydney selling them for about $220 which might be the go. |
| 14:11.26 | Faithful | johnm: yeah, I am thinking in that direction... I know the technology park near by must have storerooms full of them. |
| 14:12.22 | Faithful | Nix: Do they need a keyboard to powerup? |
| 14:13.30 | Nix | Faithful: shouldn't |
| 14:16.11 | Laibsch | Do I have to configure extensions.conf before asterisk will finally start up? Right now, I fail to get the process to run. |
| 14:16.18 | kippi | when you get the link from the voicemail to say you have message, how can you change the link that it sends you? |
| 14:16.34 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 14:17.18 | lehel | kippi: in voicemail.conf or smwhere there |
| 14:17.26 | kippi | found it |
| 14:23.11 | *** join/#asterisk paryl (n=paryl@209.236.78.59) |
| 14:25.06 | paryl | i'm having an issue with a gxp-2000 in a server room. it acts like silence suppression is on, the noise from the fans goes in and out, but it's turned off. is there a setting in asterisk that could be making it do that, or is it just the phone? |
| 14:28.11 | AsterNov | my voicemail is not actually writing any files to the voicemail directory |
| 14:30.23 | jake1932 | AsterNov: perimssions issues? |
| 14:31.47 | AsterNov | I checked the directory and it actually belongs to asterisk with permission to write to it |
| 14:33.06 | jake1932 | AsterNov: any errors in the CLI? |
| 14:33.45 | AsterNov | none at all! |
| 14:33.58 | AsterNov | that's what baffles me |
| 14:35.18 | lehel | AsterNov: set verbose 15 |
| 14:35.49 | AsterNov | ok @lehel |
| 14:37.11 | *** join/#asterisk Katty (n=angela@68-112-15-110.dhcp.cpgr.mo.charter.com) |
| 14:37.18 | Katty | hihi |
| 14:38.12 | AsterNov | it gives no more additional output, as far as asterisk is concerned, all went ok @lehel |
| 14:39.25 | jake1932 | AsterNov: what version? |
| 14:39.50 | AsterNov | asterisk@home 1.2.0 Beta1 |
| 14:41.05 | jake1932 | hmm. is that the version number from "show version" in the CLI? |
| 14:42.33 | AsterNov | it doesn't actually tell me the version # @jake1932 |
| 14:42.34 | AsterNov | asterisk1*CLI> show version |
| 14:42.35 | AsterNov | Asterisk built by root@asterisk1.local on a i686 running Linux on 2005-09-15 20:19:20 UTC |
| 14:43.05 | jake1932 | must be a custom build |
| 14:44.14 | desktophero | hello everyone |
| 14:44.21 | AsterNov | so yours actually shows the version # @jake1932? |
| 14:44.26 | jake1932 | AsterNov: yes |
| 14:44.48 | jake1932 | AsterNov: i'm not using @home though. (abandoned that a while ago) |
| 14:45.20 | lehel | jake1932: what ver do you use?.. me: CVS |
| 14:45.36 | AsterNov | so jake1932, is your asterisk build quite stable then? |
| 14:45.45 | jake1932 | i'm using CVS HEAD as of 9/15 |
| 14:45.51 | jake1932 | seems pretty stable |
| 14:47.06 | jake1932 | although I'm not pushing it - just a couple of sip and IAX connection + voicemail |
| 14:47.25 | AsterNov | jake1932: what about the attended transfer function? |
| 14:48.26 | *** join/#asterisk tzanger (n=tzanger@mixdown.ca) |
| 14:48.35 | jake1932 | what about it? |
| 14:49.30 | Katty | mew? |
| 14:51.39 | AsterNov | when I do a attended transfer, after person B & C finish their conversation, asterisk reloads. |
| 14:52.07 | lehel | wow;) |
| 14:52.13 | jake1932 | ok - just tried it - seems to work fine - no reload |
| 14:53.52 | AsterNov | basically you get to hear person B & C before transferring them using the hook flash button? @jake1932 |
| 14:54.12 | *** join/#asterisk obsidian-studios (n=obsidian@c-66-177-188-197.hsd1.fl.comcast.net) |
| 14:54.46 | jake1932 | AsterNov: i just call A - press hold - call B - press Transfer |
| 14:54.46 | obsidian-studios | are polycom soundpoint IP 500 phones decent? |
| 14:55.25 | lehel | jake1932: which is ur :hold" button? |
| 14:56.05 | jake1932 | lehel: the one that says "Hold" :) I have a cisco 7960 with several soft buttons |
| 14:56.29 | AsterNov | jake32: that works fine for me using a softphone or any phone with a transfer button, but not using the hook flash button (R) |
| 14:56.35 | lehel | ok jake :D, i didn't know about ur cisco |
| 14:57.02 | jake1932 | AsterNov: what ATA? |
| 14:57.15 | AsterNov | GS 486 |
| 14:57.30 | Laibsch | I have installed asterisk on a Debian testing system. I made the changes to /etc/default/asterisk so that * should start. I have added two users to /etc/asterisk/sip.conf as per http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html?page=3#config and restarted *. "nmap localhost" does not show any new ports running, especially not 5600. I have no firewall. I tried this both on a Debian and a SUSE system. I increased verbosage f |
| 14:57.59 | Laibsch | Sorry for the long text but I tried to cover everything I did to get this running. |
| 14:58.07 | lehel | jake1932: why i'm not able to transfer calls with "#" in Firefly? |
| 14:58.39 | Katty | :< |
| 14:58.53 | jake1932 | lehel: in order to # xfer, asterisk needs to be in the stream, and you need to have the option turnder on in features.conf |
| 14:59.16 | Laibsch | ps faux|grep asterisk does not show a running asterisk process even after restarting asterisk. |
| 14:59.27 | Katty | :<<< |
| 15:00.12 | jake1932 | Laibsch: did you try running it asterisk -vvvvvgcd |
| 15:00.34 | *** join/#asterisk Veto_laptop (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 15:00.42 | *** join/#asterisk Inkubot (n=inkubot@200.75.4.7) |
| 15:01.06 | Inkubot | how can i change the language of the mail when i got a message in the voicemail ? |
| 15:01.10 | jake1932 | AsterNov: not familiar with that ATA |
| 15:01.26 | Laibsch | jake1932: Thank you for the reply. No, I did not turn up verbosage that much. I can do it, though. |
| 15:01.31 | *** join/#asterisk dalabera (n=Dalabera@pmr.pmrtechnologies.com) |
| 15:01.31 | AsterNov | grandstream |
| 15:02.45 | Laibsch | jake1932: I have it set to "-g -v". |
| 15:03.46 | jake1932 | Laibsch: if it's not starting chances are it caught an error - just need to find out what the error is |
| 15:04.51 | jake1932 | Laibsch: i've always been able to locate the error when running it asterisk -vvvvvgcd |
| 15:04.59 | Laibsch | Is there a paste service for a file instead of the clipboard somewhere? |
| 15:05.22 | Laibsch | I ran it like that. Now I need to get you the output. |
| 15:05.24 | Katty | why is it that no one ever says hi to me :P |
| 15:05.31 | Katty | it |
| 15:05.32 | jake1932 | hi Katty |
| 15:05.33 | Katty | 's just silly |
| 15:05.36 | Laibsch | Hi Katty ;-) |
| 15:05.41 | Katty | k, all better. |
| 15:05.47 | jake1932 | ~pastebin |
| 15:05.53 | jbot | [pastebin] a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 15:05.59 | jake1932 | Laibsch: use pastebin |
| 15:06.08 | Laibsch | Thanks jake1932 |
| 15:06.25 | lehel | jake1932: are you experienced in faxing? |
| 15:06.57 | jake1932 | lehel: i am - however not in troubleshooting fax issues in asterisk - think that's what yuou're asking |
| 15:07.52 | [Jedi] | anyone experienced with asterisk+faxing here? :(( |
| 15:07.58 | [Jedi] | Anyone made TxFax work? |
| 15:08.17 | [Jedi] | I'd like to ***COPY*** the configuration and distro and everything from anyone who has a working TxFax |
| 15:08.21 | obsidian-studios | [Jedi]: I have it working here at my office, tried to replicate for client, and nothing but problems |
| 15:08.23 | AsterNov | thanks for the help jake1932 & lehel, got to dash now. |
| 15:08.37 | [Jedi] | obsidian-studios: uhm |
| 15:08.57 | [Jedi] | obsidian-studios: I have tried with TDMoE, with direct Digium E1 channels, with BRIstuff channels... and *nothing* |
| 15:09.19 | [Jedi] | I have tried with 1.1 CVS from july, with 1.2 beta1, with 1.0.9, with 1.0.9-bristuff'ed |
| 15:09.47 | obsidian-studios | [Jedi]: I got it working via el cheapo X100p clone card here, tried to do it at a clients with a TDM04B, 4 fxs modules, and nothing |
| 15:10.15 | [Jedi] | we're going to buy a brooktrout client |
| 15:10.16 | obsidian-studios | [Jedi]: I did nothing special and I can pastebin the extensions in the context |
| 15:10.20 | [Jedi] | a brooktrout fax board |
| 15:10.31 | [Jedi] | and buy a fax server for windows |
| 15:10.53 | obsidian-studios | [Jedi]: I have other clients faxing and processing credit cards all day long via fxs ports in their cisco 827-4v router, but the router has t.38 support in it |
| 15:11.05 | [Jedi] | I have lost a lot of time with TxFax. I'm sure it's a great piece of software |
| 15:11.33 | *** part/#asterisk Inkubot (n=inkubot@200.75.4.7) |
| 15:11.47 | lehel | [Jedi]: in other words you want to send faxes, right? |
| 15:11.53 | obsidian-studios | [Jedi]: you look at hylafax? It was recommended to me when I had problems with txfax. Really I would love a generic fxs device with t.38 support |
| 15:11.55 | [Jedi] | that's it |
| 15:12.05 | [Jedi] | lehel: I need to send a whole lot of faxes |
| 15:12.08 | *** join/#asterisk ManxPower (n=eric@adsl-70-247-221-174.dsl.lgvwtx.swbell.net) |
| 15:12.13 | [Jedi] | I need a full E1 for faxing |
| 15:12.29 | lehel | [Jedi]: you can receive at the moment? |
| 15:12.36 | [Jedi] | haven't tried |
| 15:12.40 | [Jedi] | I just need sending |
| 15:12.44 | *** join/#asterisk fall0ut (n=tim@216.106.191.1) |
| 15:13.09 | [Jedi] | lehel: I get TxFax executed, and if I try to send a fax to my own phone, I can hear the "beep" |
| 15:13.17 | Laibsch | jake1932: stdout is at http://pastebin.ca/25227, stderr is at http://pastebin.ca/25228 |
| 15:13.31 | [Jedi] | but NOTHING ELSE gets done by TxFa |
| 15:13.31 | [Jedi] | x |
| 15:13.40 | ManxPower | do you use the "caller" option to TxFax? |
| 15:13.47 | [Jedi] | I've tried with and without caller |
| 15:13.58 | [Jedi] | I've even forced the caller option in the app_txfax.c file |
| 15:14.25 | lehel | [Jedi]: could you pastebin your txfax command, from the extensions? |
| 15:14.49 | [Jedi] | sure |
| 15:16.15 | *** join/#asterisk jets (n=jets@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 15:16.25 | jake1932 | Laibsch: are you trying to do faxing? |
| 15:16.38 | jake1932 | scratch that |
| 15:16.38 | fall0ut | so, how do I disable comfort noise generation on asterisk? |
| 15:16.40 | fall0ut | on g711ulaw? |
| 15:16.59 | [Jedi] | http://pastebin.ca/index.php |
| 15:17.11 | RoyK | ~pb |
| 15:17.12 | jbot | i guess pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 15:17.16 | Laibsch | jake1932: No. just VoIP conferencing. |
| 15:17.17 | jake1932 | Laibsch: it's trying to load app_dtmftotext.so |
| 15:17.23 | RoyK | ~nickometer [Jedi] |
| 15:17.24 | jake1932 | Laibsch: and it's failing |
| 15:17.34 | RoyK | ~nickometer jake1932 |
| 15:18.10 | Laibsch | I remember that being a separate package in Debian. I'll check up on it. |
| 15:18.38 | RoyK | app_musictodtmf |
| 15:18.57 | [Jedi] | ї?ї?ї? why? |
| 15:19.03 | jake1932 | Laibsch: app_RoyKtodtmf |
| 15:19.04 | lehel | ЙP |
| 15:19.08 | [Jedi] | wop sorry xDDD |
| 15:19.24 | [Jedi] | http://pastebin.ca/25230 |
| 15:19.26 | [Jedi] | I'm sorry xD |
| 15:19.31 | ManxPower | fall0ut, Asterisk does not generate comfort noise |
| 15:19.31 | lehel | [Jedi] .. don't woory.. it's a bot;) |
| 15:20.21 | ManxPower | fall0ut, Asterisk CANNOT generate comfort noise |
| 15:20.50 | jake1932 | prob generated by his provider |
| 15:21.10 | fall0ut | ManxPower: going from gw -> asterisk having it answer generates noise |
| 15:21.25 | ManxPower | fall0ut, Define "noise" |
| 15:21.37 | fall0ut | its generating comfort noise |
| 15:21.41 | ManxPower | fall0ut, Your noise is caused or genrated by something ELSE. |
| 15:21.43 | kippi | if i install asterisk at home, can i upgrate askterisk to the pro version is i have a copy? |
| 15:22.02 | ManxPower | fall0ut, Perhaps you did not hear me. Asterisk CANNOT generate comfort noise. |
| 15:22.09 | [Jedi] | My .extension is fine right? |
| 15:22.19 | [Jedi] | my call file is this one: http://pastebin.ca/25233 |
| 15:22.20 | Laibsch | Well, in Debian it seems to be in the package "asterisk-app-dtmftotext" which I had installed. I'll try and check permissions and deinstall the package if nothing helps. I can live without that functionality. |
| 15:22.21 | jake1932 | kippi: pro version? |
| 15:22.55 | [Jedi] | lehel: did you take a look at my extensions? is it fine? |
| 15:23.22 | ManxPower | kippi, There is no "pro version" of Asterisk |
| 15:23.28 | jake1932 | kippi: you mean "Asterisk Business Edition"? |
| 15:23.36 | kippi | yep |
| 15:24.01 | ManxPower | kippi, ABE is based on CVS-HEAD. |
| 15:24.23 | ManxPower | kippi, Why do you want to use Business Edition? |
| 15:24.32 | kippi | because it has the support? |
| 15:24.57 | ManxPower | kippi, Um, regular Asterisk has support too. In fact, from the same people |
| 15:25.06 | kippi | but you have to pay? |
| 15:25.08 | jake1932 | install support |
| 15:25.26 | Katty | hmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmm. |
| 15:25.37 | ManxPower | kippi, You pay for support one way or another. |
| 15:25.38 | jake1932 | or support in here for everything else |
| 15:25.47 | jake1932 | moral support |
| 15:25.53 | [Jedi] | hello? anyone has taken a look at my .ext and .call? :)))) |
| 15:26.04 | kippi | is there a good how to install AMP |
| 15:26.14 | lehel | [Jedi]: looks good! |
| 15:26.50 | lehel | [Jedi]: you use capi? |
| 15:27.07 | [Jedi] | lehel: no, a TE405P |
| 15:27.08 | ManxPower | kippi, the primary reason to buy ABE is because of short sighted bean counters that require a "?boxed product" |
| 15:27.14 | [Jedi] | I have a full E1 dedicated to faxing |
| 15:27.38 | *** join/#asterisk Ayano (n=erik_lee@adsl-70-245-190-90.dsl.spfdmo.swbell.net) |
| 15:27.56 | *** join/#asterisk brc_ (n=Brian@pdpc/supporter/basic/brc) |
| 15:28.26 | Chuji | Is the iax extention to the dev meeting open? |
| 15:28.40 | Laibsch | -rw-are--are-- 1 root root 22404 2005-05-10 22:45 /usr/lib/asterisk/modules/app_dtmftotext.so so it should be OK permission-wise, right? |
| 15:28.47 | lehel | [Jedi]: did you tried instead of "n" using numbers? [i'm not sure if matters] |
| 15:28.55 | [Jedi] | lehel: yes, I've tried |
| 15:29.00 | Laibsch | s/are/are |
| 15:29.00 | [Jedi] | lehel: I've tried almost everything |
| 15:29.05 | [Jedi] | lehel: do you have that working? |
| 15:29.13 | kippi | ManxPower: boxed product? this come with built in web interface? |
| 15:29.15 | lehel | [Jedi]: what your CLI shows? |
| 15:29.20 | [Jedi] | lehel: nothing |
| 15:29.22 | ManxPower | kippi, I doubt it. |
| 15:29.25 | [Jedi] | lehel: I'll pastebin it |
| 15:29.32 | Laibsch | Please disregard "are". My client insists on replacing an "R" with are |
| 15:29.34 | lehel | nothing?.. impossible |
| 15:29.43 | lehel | [Jedi] increase the verbosity level |
| 15:29.43 | jake1932 | kippi: are you new to asterisk? |
| 15:29.47 | kippi | yeah |
| 15:29.58 | jake1932 | kippi: are you on a tight deadline? |
| 15:30.01 | [Jedi] | lehel: just this: -- Executing TxFAX("Zap/32-1", "/faximage.tif|caller|debug") in new stack |
| 15:30.05 | kippi | yep |
| 15:30.09 | [Jedi] | lehel: *nothing* else |
| 15:30.15 | *** join/#asterisk Ganlron (n=Ganlron@omega.csolve.net) |
| 15:30.18 | ManxPower | [Jedi], you have the tiff file in the root directoryu |
| 15:30.28 | ManxPower | kippi, if you are on a tight deadline then don't use Asterisk |
| 15:30.29 | [Jedi] | ManxPower: yes, it's for testing |
| 15:30.42 | [Jedi] | [root@ccard01 fax]# ls -la /faximage.tif |
| 15:30.42 | [Jedi] | -rwxr-xr-x 1 root root 5032 oct 7 19:27 /faximage.tif |
| 15:30.53 | Ganlron | Odd question, has anyone managed to get a Toshiba StrataIP Phone connected to Asterisk? |
| 15:30.59 | kippi | great, well i need to use aterisk |
| 15:31.04 | kippi | asterisk |
| 15:31.24 | ManxPower | kippi, then be prepared to spend a few weeks or months builting test systems to get it working the way you want |
| 15:31.27 | jake1932 | kippi: when I first started, i used a consultant |
| 15:31.45 | jake1932 | deadline |
| 15:31.53 | [Jedi] | lehel: how can I increase the verbose level ? I have it already at full logging and I still don't get any message from TxFax |
| 15:32.15 | Ganlron | Toshiba's documentation is crap. not even 100% sure what protocol these phones use |
| 15:32.44 | jake1932 | kippi: does the computer you're installing it on have an internet connection? |
| 15:32.45 | ManxPower | [Jedi], see logger.conf |
| 15:32.56 | [Jedi] | ManxPower: I have it at full |
| 15:33.08 | ManxPower | Ganlron, your search of the mailing list archive didn't come up with anything. |
| 15:33.15 | [Jedi] | full => notice,warning,error,debug,verbose |
| 15:33.25 | *** join/#asterisk contrabanda (n=G@213.131.37.202) |
| 15:33.27 | ManxPower | [Jedi], so you have console => notice,warning,error,verbose,debug |
| 15:33.28 | lehel | [Jedi]: *CLI> set verbose 20 |
| 15:33.44 | contrabanda | i have bought g729 license. can i use it several times? |
| 15:33.45 | ManxPower | [Jedi], that line would send the messages to /var/log/asterisk/full |
| 15:33.47 | ManxPower | not to the console |
| 15:33.54 | ManxPower | contrabanda, no |
| 15:34.12 | jake1932 | contrabanda: one box, 1 connection per license |
| 15:34.30 | [Jedi] | ManxPower: the full log doesn't show anything from TxFax, also |
| 15:34.56 | ManxPower | [Jedi], tx/rx fax mat send to stdout. stop asterisk, start is as "asterisk -cvvvddd" |
| 15:35.25 | lehel | even more "v"s [Jedi] |
| 15:35.32 | contrabanda | if something will happen with asterisk and ill reinstall it |
| 15:35.38 | contrabanda | i have to buy new license? |
| 15:35.42 | [Jedi] | ManxPower: really??? wow |
| 15:36.12 | *** join/#asterisk sambal (n=sambal@213.148.236.189) |
| 15:36.13 | ManxPower | [Jedi], it's one of the Annoying Things About Asterisk |
| 15:36.32 | ManxPower | stderr/stdout is only sent to whatever tty is the Asterisk console. |
| 15:36.52 | [Jedi] | great |
| 15:36.55 | [Jedi] | now txfax |
| 15:36.58 | [Jedi] | is showing things |
| 15:36.59 | [Jedi] | in my screen |
| 15:37.04 | [Jedi] | success - delivered 1 pages |
| 15:37.06 | ManxPower | contrabanda, I think you can reinstall up to 3 times. |
| 15:37.07 | [Jedi] | AOIHDFsadf0'asdf`hiasdfap9sgdfasdтfb |
| 15:37.21 | [Jedi] | I WAS GETTING CRAZY |
| 15:38.39 | *** join/#asterisk JerJer[mobile] (n=jj@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 15:39.22 | JerJer[mobile] | anyone ever get chan_oss or chan_alsa working on a VIA Mini-itx crap? |
| 15:39.22 | contrabanda | so if ill install it in 3 different asterisks is it a problem? how can they check it? |
| 15:39.29 | [Jedi] | incredible |
| 15:39.30 | *** join/#asterisk wrmem (n=monnin@monnin-win.cso.uiuc.edu) |
| 15:39.38 | [Jedi] | txfax works |
| 15:39.40 | ManxPower | contrabanda, it connects back to Digium |
| 15:40.04 | JerJer[mobile] | [Jedi]: only 9600 |
| 15:40.08 | JerJer[mobile] | and no error correction |
| 15:40.29 | contrabanda | hor every g729 connection? |
| 15:40.46 | contrabanda | i'll register it with 1 ip then change this ip |
| 15:40.47 | lehel | [Jedi] bravo!;) |
| 15:40.57 | ManxPower | contrabanda, we are not going to help you get around the license. |
| 15:41.10 | contrabanda | loooooooooool |
| 15:41.17 | file | the g729 license is based on the MAC address of the NICs in your computer :P |
| 15:41.17 | Ganlron | Manx nothing particularily useful.. some information about putting Asterisk in front of the toshiba switch, but couldn't find anything about the phones connecting to asterisk themselves |
| 15:41.20 | contrabanda | i just wanderrrr |
| 15:41.30 | contrabanda | how does all this procedures work |
| 15:41.36 | file | well. |
| 15:42.06 | jake1932 | contrabanda: it's simple - you buy the number of licenses you need 1 for each connection |
| 15:43.36 | *** join/#asterisk andrewct (n=root@jumbo.ntplx.net) |
| 15:43.42 | jake1932 | here's some more info: http://www.voip-info.org/wiki-Asterisk+G.729+Licensing |
| 15:44.00 | contrabanda | thanks |
| 15:45.28 | BrianR___ | Yes.. The g729 license thing is a big pain the ass - there's a high risk of the thing not working if you have to switch servers... :( |
| 15:46.04 | jake1932 | BrianR___: I'm surprised Digium wouldn't help with the install of the licenses on teh new machine |
| 15:46.09 | jake1932 | the |
| 15:46.25 | BrianR___ | jake1932: Sure they will - but it's an extra step that adds to your downtime. |
| 15:46.30 | *** join/#asterisk TrNv2 (n=trn@dsl093-048-180.mia1.dsl.speakeasy.net) |
| 15:47.05 | jake1932 | BrianR___: there's always redundancy |
| 15:47.26 | jake1932 | BrianR___: why not have a hot spare waiting? |
| 15:47.35 | BrianR___ | jake1932: Yeah. It's just a cost thing. |
| 15:47.43 | ManxPower | It's only an issue if you have to replace your eithernet card more than 3 times, and if you have to do that, you have bigger problems than a G729 license issue |
| 15:48.01 | BrianR___ | jake1932: It works out to be a lot more than $5/channel or whatever they charge. Still not so bad. |
| 15:48.05 | TrNv2 | Is IAXTEl currently working. :p |
| 15:48.15 | TrNv2 | ? |
| 15:48.15 | ManxPower | And it's not Digium's fault, it's the patent holders of G729 |
| 15:48.28 | contrabanda | soo it means that its depend on mac address and not on Ip addresss? |
| 15:48.38 | BrianR___ | In the end, it's probably cheaper than direct licensing of g729 for all but the largest users. |
| 15:48.40 | ManxPower | contrabanda, correct. |
| 15:48.58 | contrabanda | sooooo, its possible to change mac address |
| 15:49.04 | contrabanda | on lan card |
| 15:49.10 | *** join/#asterisk fugitivo (n=ajf@201.255.105.36) |
| 15:49.11 | fugitivo | hello |
| 15:49.22 | contrabanda | whats happen in this case? |
| 15:49.24 | jake1932 | BrianR___: if your users are that dependant upon 100% uptime - hot spares are usually an option |
| 15:49.30 | ManxPower | contrabanda, try it and see. |
| 15:49.34 | contrabanda | so i can use it in 3 different sites? |
| 15:49.41 | BrianR___ | contrabanda: Yes. That's one workaround. But aparently the g729 license check stuff is based on _all_ of the NIC's in the system... So adding a NIC also invalidates your key. |
| 15:49.42 | [Jedi] | is there any way to set a maximum concurrency of outgoing .call-generated phonecalls? |
| 15:49.51 | ManxPower | contrabanda, only if you want to be arrested for license violation and patent violations |
| 15:50.07 | contrabanda | :) |
| 15:50.33 | BrianR___ | I don't know if patent infringement has been criminalized yet. The music mafia and movie/software cartels are trying their best with copyright infringement. |
| 15:50.55 | contrabanda | i got stollen license |
| 15:50.55 | *** join/#asterisk Laibsch (n=Laibsch@p54B98501.dip0.t-ipconnect.de) |
| 15:51.03 | ManxPower | [Jedi], Channels can be in more than one group. For example: |
| 15:51.04 | [Jedi] | ManxPower: I really love you. We were going to buy an ***EXPENSIVE*** brooktrout card just because we weren't able to make TxFax work |
| 15:51.04 | Nugget | between the pirates and the media congolomorates, I blame the pirates more for the state we're in right now. |
| 15:51.05 | ManxPower | group=1 |
| 15:51.06 | ManxPower | channel => 1-10 |
| 15:51.06 | ManxPower | group=1,2 |
| 15:51.06 | ManxPower | channel => 11-16 |
| 15:51.12 | contrabanda | so they will arest me :))) |
| 15:51.16 | ManxPower | Then send your faxes out G2 |
| 15:51.23 | [Jedi] | ManxPower: ok, so I create a 15-channel group and use it for faxing |
| 15:51.25 | [Jedi] | great |
| 15:51.51 | ManxPower | [Jedi], love me with a paypal donation |
| 15:52.01 | tzanger | BrianR___: that's untrue -- we added/removed NICs without issue. It's the first detected NIC that seems to be used |
| 15:52.04 | [Jedi] | ManxPower: I'll suggest it in my company ;) |
| 15:52.09 | Laibsch | OK, * seems to be able to start now that I have deinstalled asterisk-app-dtmftotext. Still port 5060 is not open. What is also weird is that /etc/init.d/asterisk start will not return me to the bash prompt but to the asterisk prompt. |
| 15:52.31 | jake1932 | Laibsch: did SIP start? |
| 15:52.48 | ManxPower | Laibsch, We reallyl can't help you unless you install from source. |
| 15:52.56 | Laibsch | I am not sure. * started. |
| 15:53.04 | ManxPower | "show modules" will show you what modules are installed and running in Asterisk |
| 15:53.12 | TrNv2 | Hey, I'm having problems getting connected to IAXtel. I keep getting registration refused messages, but I can use the same username and password to log into the iaxtel web interface. |
| 15:53.28 | TrNv2 | Also, this is a system that is already in production and working - I just haven't tried IAXtel until today. |
| 15:53.40 | Laibsch | Is there a pager in *, the messages scroll off the screen. |
| 15:54.06 | jake1932 | Laibsch: log it on your client |
| 15:54.38 | jake1932 | Laibsch: or configure logger.conf |
| 15:55.13 | Laibsch | jake1932: OK, I will go for logger.conf since I do not think I will be able to connect. There is no port open. |
| 15:55.39 | Laibsch | Output from "show modules" that fit on the screen is available at http://pastebin.ca/25237 |
| 15:55.47 | andrewct | Any digium hardware people here? I have a TDM400 hardware question... |
| 15:56.24 | ManxPower | trig_hm, If you change your password in the web interface it does NOT change it for your iaxtel acocunt, it only changes it for the web interface. |
| 15:58.18 | TrNv2 | ManxPower: Hrrm. I get the same registration refused with the original password sent via e-mail too. |
| 15:58.33 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-218-202.rockynet.com) |
| 16:00.13 | ManxPower | TrNv2, maybe it's just down. It always seems to be down |
| 16:00.55 | TrNv2 | When attempting to make a call, I get an error also. :p |
| 16:00.57 | TrNv2 | Registration of 'afh' rejected: Registration Refused |
| 16:01.07 | TrNv2 | I guess it just might be down. |
| 16:02.17 | jake1932 | Laibsch: chan_sip is not loaded |
| 16:03.20 | *** join/#asterisk RoyK (n=roy@100.80-203-27.nextgentel.com) |
| 16:04.43 | *** join/#asterisk oej (n=Olle@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 16:06.21 | AlexCTI | Hi everyone, how can i get better quality on MOH using RAW, i have a little static, but the file is perfect.. |
| 16:07.07 | Laibsch | jake1932: Maybe it is. Maybe the portion relating to it scrolled off the screen. I am fiddling with logger.conf and will report back. |
| 16:08.25 | *** join/#asterisk TedC (n=ted@gray.impulse.net) |
| 16:09.02 | *** join/#asterisk drumkilla_laptop (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:09.02 | *** mode/#asterisk [+o drumkilla_laptop] by ChanServ |
| 16:09.03 | *** join/#asterisk Cresl1n (n=matt@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 16:09.40 | *** join/#asterisk grumpie (n=vijay@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 16:10.11 | *** part/#asterisk grumpie (n=vijay@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 16:10.28 | Cresl1n | meow? |
| 16:12.00 | sigterm | woof |
| 16:12.02 | *** join/#asterisk jero (n=sflphone@savoirfairelinux.net) |
| 16:12.07 | jero | hi |
| 16:12.36 | sigterm | hello jero |
| 16:13.07 | AlexCTI | I have a little noise on the music on hold, it is like static, anyone knows how can i remove it? |
| 16:13.10 | *** part/#asterisk TedC (n=ted@gray.impulse.net) |
| 16:14.14 | Laibsch | jake1932: First part of debug output is at http://pastebin.ca/25238, second part at http://pastebin.ca/25239. Seems to me that sip is running. |
| 16:15.29 | jake1932 | Laibsch: 'show modules' should show chan_sip |
| 16:15.51 | RoyK | anyone here using some queue stats tools? |
| 16:15.55 | Laibsch | jake1932: I guess it does but there are so many modules that they scroll off the screen. |
| 16:16.09 | Laibsch | Can I limit the output of modules? |
| 16:16.19 | *** join/#asterisk Nix (n=Nix@81.213.125.220) |
| 16:16.44 | RoyK | Laibsch: what modules? |
| 16:16.55 | tatuman | hi, is anyone here an expert in meetme app? |
| 16:17.27 | RoyK | yeah. doctor of science with the meetme subject |
| 16:17.30 | Laibsch | RoyK: The one for SIP. |
| 16:17.40 | *** join/#asterisk lbow (n=lbow@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 16:17.40 | RoyK | chan_sip.so.... |
| 16:17.55 | RoyK | Laibsch: how do you want to limit its output? |
| 16:18.11 | Laibsch | I need to know if chan_sip.so is running. |
| 16:18.20 | RoyK | show modules |
| 16:18.26 | Laibsch | RoyK: A pager or a grep would do. |
| 16:18.36 | ManxPower | Laibsch, "sip show peers" If you get an error, it's not loaded |
| 16:18.43 | *** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com) |
| 16:18.43 | RoyK | asterisk -rx 'show modules' | grep chan_sip |
| 16:19.00 | ManxPower | RoyK's suggestion is better |
| 16:19.03 | *** join/#asterisk Jas_Williams (n=Jason@host86-130-0-82.range86-130.btcentralplus.com) |
| 16:19.08 | Laibsch | I get something looking like the two users I configured. |
| 16:19.12 | ManxPower | RoyK, he installed from a package, not from source |
| 16:19.33 | RoyK | ManxPower: whatever. -rx should work anyway... |
| 16:19.43 | *** join/#asterisk tgrman (n=tgrman@63.77.68.10) |
| 16:20.38 | Laibsch | yes, it did. And it shows chan_sip.o to be there. |
| 16:20.53 | Laibsch | chan_sip.so Session Initiation Protocol (SIP) 0 |
| 16:21.20 | Laibsch | Still, no port 5060 open even for localhost. |
| 16:21.34 | ManxPower | Laibsch, How can you tell? |
| 16:21.54 | ian_k | Laibsch: Did you bind the port in sip.conf? |
| 16:21.56 | Laibsch | "nmap localhost" does not show 5060 in the list. |
| 16:22.26 | ManxPower | Try this: netstat -an | grep 5060 |
| 16:22.37 | Laibsch | yes, 5060 in sip.conf. |
| 16:23.13 | Laibsch | # netstat -an | grep 5060 |
| 16:23.13 | Laibsch | udp 0 0 0.0.0.0:5060 0.0.0.0:* |
| 16:23.22 | Laibsch | So maybe the port is open? |
| 16:23.30 | ian_k | Laibsch: Do a "nmap -sU localhost" and it'll probably show up |
| 16:23.37 | Laibsch | I'll try with a client to connect again. |
| 16:24.01 | Laibsch | Indeed, it does. Although as filtered. |
| 16:24.08 | Laibsch | Thanks, guys. |
| 16:24.11 | ian_k | DO you have a kernel firewall? |
| 16:24.24 | Laibsch | no, no firewall. |
| 16:24.30 | Laibsch | no iptables, etc. |
| 16:24.36 | ian_k | do a "iptables -L" to make sure |
| 16:24.42 | ian_k | some distros have one enabled by default |
| 16:26.44 | RoyK | Laibsch: nmap -sU -p 5060 localhost |
| 16:27.28 | RoyK | Laibsch: but I think I've seen that error before... not listening to localhost, only the main ip, but netstat telling me otherwise |
| 16:27.31 | RoyK | although that was os x |
| 16:27.39 | RoyK | iirc |
| 16:28.47 | *** join/#asterisk AsteriskNoob (i=BoredBoz@207-114-232-10.gen.twtelecom.net) |
| 16:28.56 | AsteriskNoob | goooooood morning everyone! |
| 16:30.02 | *** join/#asterisk rubble (n=netclass@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 16:30.38 | *** join/#asterisk drumkilla_laptop (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:30.38 | *** mode/#asterisk [+o drumkilla_laptop] by ChanServ |
| 16:30.42 | AsteriskNoob | RoyK must be on the other side of the big sphere |
| 16:30.55 | RoyK | prolly |
| 16:30.56 | AsteriskNoob | its 10:30AM here |
| 16:30.57 | RoyK | .no |
| 16:31.03 | AsteriskNoob | boise idaho |
| 16:31.07 | RoyK | ahki |
| 16:31.20 | RoyK | the USAnians joins the channel |
| 16:31.48 | AsteriskNoob | lol, well yeah, alot of us USAnians are just waking up |
| 16:31.59 | AsteriskNoob | where are you? Europe somewhere obviously |
| 16:32.24 | jake1932 | only about half |
| 16:32.33 | jake1932 | in EST it's lunch time |
| 16:32.49 | AsteriskNoob | true jake |
| 16:33.03 | AsteriskNoob | but then again, its insanity time everywhere! |
| 16:33.10 | *** join/#asterisk shido6 (n=curtis@d221-68-210.commercial.cgocable.net) |
| 16:33.11 | jake1932 | :) right |
| 16:34.25 | *** join/#asterisk bkuhn (n=bkuhn@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 16:34.48 | *** join/#asterisk Godsey (i=lanny@pdpc/supporter/sustaining/Godsey) |
| 16:35.54 | Laibsch | ian_k: iptables -L does not show any targets und INPUT, FORWARD and OUTPUT. |
| 16:38.00 | *** part/#asterisk bkuhn (n=bkuhn@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 16:41.07 | *** join/#asterisk [ViRii] (n=virii@68-186-170-246.dhcp.smrt.tn.charter.com) |
| 16:42.10 | [ViRii] | anyone installed asterisk on ubuntu distro? |
| 16:42.46 | Laibsch | [ViRii]: Not ubuntun. But I am in the process of installing it on Debian testing. |
| 16:44.21 | *** join/#asterisk AlexCCCC (n=www@200.180.50.200) |
| 16:44.37 | RoyK | [ViRii]: asterisk runs on any distro i've tried |
| 16:44.45 | RoyK | including freebsd and os x |
| 16:44.52 | RoyK | and colinux |
| 16:44.56 | ian_k | Asterisk runs fine on Ubuntu.. |
| 16:45.15 | AlexCCCC | My x-lite just works inside of my network, how can I do this form outside? |
| 16:45.16 | *** part/#asterisk tgrman (n=tgrman@63.77.68.10) |
| 16:45.17 | RoyK | just use your favourite distro |
| 16:45.23 | *** join/#asterisk tgrman (n=tgrman@63.77.68.10) |
| 16:46.01 | *** join/#asterisk marc324 (n=marc3234@206-248-159-56.dsl.teksavvy.com) |
| 16:46.23 | *** join/#asterisk myiagy (n=myiagy@200.138.215.78) |
| 16:48.44 | *** join/#asterisk twilson (n=twilson@69.17.122.227) |
| 16:50.10 | AlexCCCC | any one? |
| 16:53.36 | *** join/#asterisk Damin_PDA (n=pocketir@158.sub-70-209-180.myvzw.com) |
| 16:57.00 | Katty | hmm |
| 16:57.11 | ManxPower | does anyone know of you can set a per peer jitterbuffer in 1.0.x? |
| 16:57.14 | *** join/#asterisk dalfry (n=dalfry@gateway.ishisystems.com) |
| 17:01.25 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 17:03.38 | *** join/#asterisk XTR-II (n=xtr@staff-nat.netnation.com) |
| 17:04.49 | synthetiq | what the default lien build out on the t1s |
| 17:06.48 | ManxPower | synthetiq, you can usually assume a line build out (LBO) of 0 |
| 17:07.14 | loud | Does anyone have the link to the codec_g723.so file ? |
| 17:07.33 | [ViRii] | where best place to d/l asterisk |
| 17:08.05 | synthetiq | 0-133ft? |
| 17:08.07 | loud | digium ftp server. |
| 17:08.17 | TrNv2 | Hrm. |
| 17:08.28 | TrNv2 | FWD seems to work just fine, IAXtel does not. |
| 17:08.33 | TrNv2 | I guess it really is just down. |
| 17:09.59 | ManxPower | So the servers survived the roof blowing off, two hurricanes, the cealing falling in, only to be done in by the contracters when they sanded the walls in the computer room with the equipment still running |
| 17:10.32 | ManxPower | loud, there is no codec_g723.so codec file, since it's patended. |
| 17:11.08 | *** part/#asterisk obsidian-studios (n=obsidian@c-66-177-188-197.hsd1.fl.comcast.net) |
| 17:11.51 | loud | im willing to buy it, have you ever done this implementation ? without hacking it ? |
| 17:12.59 | *** join/#asterisk lancey (i=Shady@support.net1.cc) |
| 17:13.03 | *** join/#asterisk PupenoL (n=pupeno@200.123.183.89) |
| 17:17.24 | RoyK | loud: licensing g.723.1 will prolly cost you at least $250k |
| 17:18.15 | loud | i see. |
| 17:19.09 | gordonjcp | ManxPower: lol, kind of |
| 17:20.13 | *** join/#asterisk CrazyYoss (n=nobody@c-24-5-170-39.hsd1.ca.comcast.net) |
| 17:21.36 | RoyK | loud: so perhaps g.729 might do? |
| 17:23.53 | loud | n/m some person in here sent me that file i was looking for. |
| 17:24.05 | CrazyYoss | can Voicemail() record in mp3 format if you specify format=mp3? Or am I going to need to create a shell script to use an encoder? |
| 17:25.38 | hardwire | suddenly.. wanting to go see a movie about lindsay lohan driving around .. just doesn't do it for me |
| 17:29.21 | *** join/#asterisk drumkilla_laptop (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 17:29.21 | *** mode/#asterisk [+o drumkilla_laptop] by ChanServ |
| 17:29.45 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 17:30.16 | generalhan | whats up everyone ?! |
| 17:31.19 | CrazyYoss | its awfully quiet |
| 17:32.22 | *** join/#asterisk greekman (n=alex@host254.209.113.199.conversent.net) |
| 17:33.48 | hardwire | grr |
| 17:36.27 | CrazyYoss | can Voicemail() record in mp3 format if you specify format=mp3? Or am I going to need to create a shell script to use an encoder? |
| 17:37.48 | *** join/#asterisk ManxPower (n=eric@slip-12-65-24-211.mis.prserv.net) |
| 17:38.15 | Damin_PDA | get on #astricon |
| 17:38.48 | generalhan | oh yeah thats right |
| 17:40.05 | generalhan | CrazyYoss: you can make it record in mp3 |
| 17:40.15 | *** join/#asterisk MuppetMaster (n=MuppetMa@81.184.73.169) |
| 17:40.21 | MuppetMaster | Hello. |
| 17:40.29 | tzanger | I think my posts to the lists are being moderated |
| 17:40.29 | MuppetMaster | Anyone here use the Asterisk-IM plugin from Jive Messenger? |
| 17:40.38 | generalhan | nope |
| 17:40.58 | *** join/#asterisk devel (n=devel@wiggum.digitalcoven.com) |
| 17:41.05 | CrazyYoss | generalhan: how so? when i specify format=mp3 it doesnt work |
| 17:41.14 | generalhan | Anyone ever used the asternic FOP ?? i need some assitance |
| 17:41.40 | MuppetMaster | This one: http://www.jivesoftware.org/asterisk-im/ |
| 17:42.05 | tzanger | I've posted a number of messages and the politically charged ones have not made it public |
| 17:42.09 | tzanger | yet the ones afterward have |
| 17:42.10 | loud | RoyK, ive already bought 200 licences from digium (g729). |
| 17:42.42 | tzanger | that's a lot of g729 |
| 17:43.00 | loud | which i only use 30%. |
| 17:43.22 | Damin_PDA | 8 |
| 17:43.53 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool139-69.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 17:43.55 | *** part/#asterisk TrNv2 (n=trn@dsl093-048-180.mia1.dsl.speakeasy.net) |
| 17:44.07 | *** part/#asterisk MuppetMaster (n=MuppetMa@81.184.73.169) |
| 17:44.39 | *** join/#asterisk ManxPower (n=eric@slip-12-65-24-211.mis.prserv.net) |
| 17:46.33 | *** join/#asterisk silug (n=steve@206.80.72.34) |
| 17:47.08 | InfraRed | tzanger: what's the quality like on g729 ? |
| 17:47.42 | InfraRed | i am using 711 atm and quality is good but i need more from the bandwidth |
| 17:48.19 | *** join/#asterisk zobia (n=laura_sh@218.6.242.212) |
| 17:48.50 | zobia | Hello every one. my phone always got cut . is this hardware problem or asterisk problem? |
| 17:49.14 | Damin_PDA | get on #astricon |
| 17:49.16 | Beirdo | keep the machete away from the phone |
| 17:49.41 | InfraRed | zobia: connect to asterisk then run sip debug |
| 17:50.09 | zobia | the debug just said it hangup |
| 17:50.15 | zobia | but i did not hangup |
| 17:50.16 | InfraRed | yes |
| 17:50.20 | InfraRed | who sent the hangup |
| 17:50.22 | *** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl) |
| 17:50.38 | zobia | the caller |
| 17:50.52 | CrazyYoss | can Voicemail() record in mp3 format if you specify format=mp3? Or am I going to need to create a shell script to use an encoder? |
| 17:51.15 | InfraRed | are you calling or receiving |
| 17:51.22 | zobia | i am calling |
| 17:51.24 | ManxPower | InfraRed, G729 sounds close to G711 |
| 17:51.51 | InfraRed | ManxPower: so no real noticeable difference? |
| 17:52.02 | ManxPower | zobia, are you using Zap? |
| 17:52.10 | zobia | yes , i am using Zap |
| 17:52.13 | ManxPower | InfraRed, Try it an see, it's cheap enough. |
| 17:52.37 | InfraRed | ManxPower: i plan to, waiting for teleco to deliver the E1 |
| 17:53.19 | tzanger | I don't mind gsm quality at all. g729's a little "harsher" but still good |
| 17:53.22 | tzanger | think cell phone calls |
| 17:53.26 | ManxPower | InfraRed, Um, what does the telco have to do with it? |
| 17:53.54 | InfraRed | ManxPower: how am i supposed to make the calls :) |
| 17:53.54 | zobia | InfraRed and ManxPower . do u have any idea with my problem? |
| 17:54.09 | InfraRed | phones -> asterisk -> E1 (30 channels) |
| 17:54.13 | InfraRed | new asterisk install |
| 17:54.16 | InfraRed | will try it on that |
| 17:54.27 | ManxPower | zobia, if you are using Zap, then busydetect=yes or callprogress=yes will cause false hangups |
| 17:54.40 | ManxPower | InfraRed, Are the phones on the local lan? |
| 17:55.36 | zobia | ManxPower , even i was calling successfully for some while then cut ? also becoz of that? |
| 17:55.58 | ManxPower | zobia, if you are using Zap, then busydetect=yes or callprogress=yes will cause random hangups |
| 17:56.26 | zobia | okay. let me try. |
| 17:56.42 | ManxPower | set them to no. |
| 17:57.01 | ManxPower | In fact, I think it even tells you in the sample config files that they will cause false hangups |
| 17:57.25 | *** join/#asterisk PBXtech (n=nik@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 17:58.39 | BrianR___ | hmm.. .Maybe I should turn the echocancel value down on this T1 in hopes of making it converge faster... |
| 17:59.52 | tzanger | man sip is convoluted |
| 17:59.56 | tzanger | 403 forbidden for what reason? |
| 18:00.25 | file | authentication failed |
| 18:00.56 | tzanger | ok but why |
| 18:01.06 | tzanger | kphone and * are on the same network (no nat) |
| 18:01.26 | *** join/#asterisk terrapen (n=cjs@fw-01.satx.bikeworld.net) |
| 18:01.27 | tzanger | [akohlsmith] type friend, host dynamic, secret 12345 |
| 18:01.37 | file | do I look like I'm psychic? |
| 18:01.53 | tzanger | kphone user part of SIP URL is akohlsmith, host part is the * IP, authentication username is akohlsmith... wtf |
| 18:03.12 | *** join/#asterisk mtaht4 (n=user@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 18:03.17 | wundaboy | if my ip is 10.1.1.69 and my subnet mask is 255.255.255.192 what is my 10.1.1.0/8 (i dont know what that is called) |
| 18:03.19 | *** part/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 18:03.22 | wundaboy | did that make any sense? |
| 18:03.45 | *** join/#asterisk diclophis (n=diclophi@adsl-69-238-124-226.dsl.pltn13.pacbell.net) |
| 18:04.00 | diclophis | hello everyone |
| 18:04.04 | CrazyYoss | can Voicemail() record in mp3 format if you specify format=mp3? Or am I going to need to create a shell script to use an encoder? |
| 18:04.33 | terrapen | why would you want to record VM in mp3? |
| 18:04.41 | lancey | CrazyYoss if somebody knew, he would have already told ya, stop asking the same question plz |
| 18:04.46 | CrazyYoss | why not? |
| 18:05.02 | terrapen | because wav49 is more than suitable? |
| 18:05.04 | diclophis | does anyone have experience with asterisk clusters? |
| 18:05.12 | terrapen | and can be played on most any computer? |
| 18:05.28 | CrazyYoss | so can mp3s now and reduces storage needs |
| 18:05.50 | lancey | CrazyYoss you might use .gsm as well |
| 18:05.53 | terrapen | uhm, don't think mp3 is smaller than wav49 |
| 18:06.02 | lancey | and mp3 is not that small at all |
| 18:06.13 | terrapen | and mp3 is designed for music, not voice |
| 18:06.22 | diclophis | just get more disk-space |
| 18:06.24 | zobia | ManxPower , if i turn them to no. what else it will effect me? |
| 18:06.30 | diclophis | if you buy in bulk its cheap enough |
| 18:07.04 | *** join/#asterisk Rubble (n=netclass@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 18:07.06 | CrazyYoss | true, but I want to be able to throw the vm's onto an mp3 player. I could convert them all at 1 point in time |
| 18:07.20 | *** join/#asterisk PakiPenguin (n=pingu@linuxpakistan/admin/pakipenguin) |
| 18:07.40 | lancey | vms on mp3 player? |
| 18:07.44 | lancey | what's that for :) |
| 18:07.52 | terrapen | your wav49's should play on the mp3 player |
| 18:08.02 | *** join/#asterisk Defraz (n=t0tal@72.24.26.215) |
| 18:08.42 | CrazyYoss | random project i want to do to have random people call into a voicemail box leave a message and then ill serve them up so others can listen to them |
| 18:09.22 | shido6 | are u charging, CrazyYoss ? |
| 18:10.42 | CrazyYoss | nope, right now its one of those "I want to see if I can do it" |
| 18:10.53 | diclophis | sounds like a dating service to me |
| 18:10.58 | CrazyYoss | haha |
| 18:11.51 | *** join/#asterisk Corydon-w (i=pink@pdpc/supporter/sustaining/Corydon76-home) |
| 18:12.04 | diclophis | so nobody has any ideas about an asterisk cluster? |
| 18:12.11 | CrazyYoss | tell you what if I can ever figure this out ill send you the number and you can record your stats and see if people respond |
| 18:12.24 | tgrman | asterisk can be clustered but it's not easy |
| 18:12.39 | diclophis | howso? |
| 18:13.09 | CrazyYoss | although it might be weird if it went something like "I just scratched my balls" "Hi my name is diclophis, Im a male, age 37 looking for a good time, come see me in IRC" "I just shat my pants" |
| 18:13.13 | twilson | it would be nice to have the ability to add contexts in realtime extensions w/o having to do reloads... |
| 18:13.15 | loud | im using esx from vmware. |
| 18:13.30 | tgrman | you'd have to have a proxy out front, at least with SIP, such as SER since there is no global registration DB that can be shared betwen servers |
| 18:13.31 | diclophis | CrazyYoss, pfft the only stats women care about can be stored using numbers |
| 18:13.34 | diclophis | income and length |
| 18:13.45 | Damin_PDA | <PROTECTED> |
| 18:13.52 | CrazyYoss | hehe |
| 18:14.01 | tgrman | reSIProcate |
| 18:14.23 | *** join/#asterisk justinu (n=j2@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 18:14.29 | diclophis | hmm |
| 18:14.42 | diclophis | any links with howtos or anything? |
| 18:14.44 | justinu | hello |
| 18:15.16 | justinu | is this the channel for astricon, or is it on a seperate channel? |
| 18:15.22 | file | #astricon |
| 18:15.25 | justinu | thanks |
| 18:15.28 | [ViRii] | phones that respond to |
| 18:15.28 | [ViRii] | 'Alert_Info' via the dial plan to autoanswer. |
| 18:15.40 | [ViRii] | i cant get mine to do that on my polycoms |
| 18:19.28 | tzanger | ok so kphone doesn't work worth a shit |
| 18:19.37 | *** join/#asterisk Corydon-w (i=violet@pdpc/supporter/sustaining/Corydon76-home) |
| 18:19.41 | tzanger | a KDE app and it's still asking if I want an OSS or ALSA device, why not go through arts like hte rest of KDE? |
| 18:20.31 | ManxPower | tzanger, artsd adds lots of latency |
| 18:21.14 | tzanger | ManxPower: ok, so how do I use audio for the rest of my system ? |
| 18:22.15 | tzanger | I mean my system's all ALSA but when I select ALSA kphone hangs (presumably because ARTS is running) |
| 18:26.06 | tzanger | yup it was because ARTS is there |
| 18:26.20 | tzanger | so how the hell do you get regular apps and kphone to work on the same system... yikes |
| 18:26.45 | *** join/#asterisk darwin35 (n=darwin35@208.139.193.178) |
| 18:27.38 | *** join/#asterisk buddah (n=djbrianc@67.110.253.129) |
| 18:27.46 | buddah | is there a default limit for mailboxes in * |
| 18:27.52 | buddah | like a message limit, or size limit? |
| 18:27.59 | Corydon-w | 100 per folder |
| 18:28.08 | buddah | 100 messages? |
| 18:28.22 | buddah | how about a limit on length per message? |
| 18:28.29 | Corydon-w | maxmessages= |
| 18:28.37 | buddah | ahh nice |
| 18:28.37 | *** join/#asterisk nomazda (i=nyyankee@user-0c6tnqf.cable.mindspring.com) |
| 18:29.18 | buddah | where is that at? |
| 18:29.21 | buddah | voicemail.conf? |
| 18:29.25 | *** join/#asterisk groogs (n=greg@d221-73-237.commercial.cgocable.net) |
| 18:29.57 | buddah | i got it, thanks |
| 18:38.00 | *** join/#asterisk vandien (i=sted@aditu.dahltronics.de) |
| 18:38.11 | *** join/#asterisk epoch (n=epoch@octane.breakbeats.org) |
| 18:43.15 | *** part/#asterisk Laibsch (n=Laibsch@p54B98501.dip0.t-ipconnect.de) |
| 18:47.54 | *** join/#asterisk zeedo (n=zeedo@80.68.92.188) |
| 18:50.49 | *** join/#asterisk kippi (n=chrisfro@cpc4-hatf3-6-0-cust228.lutn.cable.ntl.com) |
| 18:50.50 | kippi | hey |
| 18:51.02 | kippi | is there a howto manual for how to install AMP |
| 18:52.36 | *** join/#asterisk syle2 (n=blag@unaffiliated/syle) |
| 18:53.36 | Damin_PDA | get on #astricon |
| 18:53.36 | *** join/#asterisk clive- (n=pirch@ndn-165-135-236.telkomadsl.co.za) |
| 18:54.30 | clive- | has anyone here used sirrix cards before?....having a little trouble configuring here |
| 19:00.37 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 19:04.30 | *** part/#asterisk diclophis (n=diclophi@adsl-69-238-124-226.dsl.pltn13.pacbell.net) |
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| 19:26.13 | *** join/#asterisk SplasPood (i=jwb@ludicrous.paravolve.net) |
| 19:28.59 | ender | When dealing w/ Polycoms and FTP/TFTP provisioning, what do I need to put in dhcpd.conf to make the phones automagically find the provisioning server? |
| 19:32.49 | CrazyYoss | is there a list of formats Voicemail() supports? |
| 19:34.37 | kshumard | CrazyYoss, gsm, wav, WAV |
| 19:34.52 | *** join/#asterisk darkskiez (n=darkskie@host86-138-169-183.range86-138.btcentralplus.com) |
| 19:35.16 | CrazyYoss | kshumard: thank you! |
| 19:35.35 | *** join/#asterisk Tarox (i=user@e178022183.adsl.alicedsl.de) |
| 19:35.48 | Corydon-w | ender: you gotta put it in the tftp-server setting |
| 19:36.03 | *** join/#asterisk burton (i=mimx@w201.ljudmila.org) |
| 19:36.09 | Corydon-w | but it will use the ftp server preferentially |
| 19:36.23 | ender | Corydon-w: I have 'option tftp-server-name "10.0.2.1"; |
| 19:36.35 | Corydon-w | Yep, that should work |
| 19:36.38 | ender | Corydon-w: phone still just says 'can't find provisioning server'. I'm using tftp as the service. |
| 19:36.45 | ender | I think maybe I should move to ftp. |
| 19:36.53 | Corydon-w | Is tftpd actually running? |
| 19:36.58 | Corydon-w | Yeah, you should use FTP |
| 19:37.55 | ender | yes, tftpd is running. If I interrupt the phone and manually change the server type from ftp to tftp and the address, then it works. |
| 19:38.40 | Corydon-w | Huh, we always use FTP, because we don't even have to unpack the phone from the box... |
| 19:39.31 | Corydon-w | We provision it, and when the phone is assembled and plugged in, it just works |
| 19:39.36 | ender | wait. |
| 19:39.37 | ender | n/m |
| 19:39.53 | Corydon-w | Are you using PlcmSpIp/PlcmSpIp as the user/pass ? |
| 19:39.53 | ender | co-worker told me the wrong port to move to the phone network, phones weren't hitting my dhcp server correctly. |
| 19:40.03 | ender | Corydon-w: tftp doesn't really have usernames |
| 19:40.06 | *** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1) |
| 19:40.12 | Corydon-w | no, but ftp does |
| 19:40.27 | Corydon-w | ender: ah, that'll do it |
| 19:41.21 | ender | heh |
| 19:41.23 | ender | trying again. |
| 19:41.34 | ender | MUCH better |
| 19:42.15 | ender | another user was complaining that they coudln't get to the internet... it was her port I switched. |
| 19:42.35 | *** join/#asterisk dalabera (n=Dalabera@pmr.pmrtechnologies.com) |
| 19:43.12 | darkskiez | anyone used asterisk on OSX in the last week or two ? |
| 19:43.31 | justinu | i run it on osx |
| 19:43.32 | jake1932 | has anyone used alternate voicemail prompts? I'm pretty sure for instance that asterisk is capable of speaking the numbers faster |
| 19:43.45 | r0d3nt | any ideas as to why a softphone wouldnt take a call from asterisk ?? |
| 19:43.46 | darkskiez | justinu: have you run cvs head in the past week or so ? |
| 19:43.51 | justinu | yeah |
| 19:43.56 | r0d3nt | it's registered, no firewall, it's local to asterisk, it can make calls... |
| 19:43.57 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfnd5.dialup.mindspring.com) |
| 19:44.09 | r0d3nt | but no recieving calls |
| 19:44.16 | darkskiez | justinu: i can issue commands via asterisk -r, but i dont see any output, even 'help' doesnt come back with anything :( |
| 19:44.33 | darkskiez | justinu: it was working maybe a month ago |
| 19:44.43 | darkskiez | justinu: works ok for you? |
| 19:52.12 | *** join/#asterisk ian_k (n=ian@gateway.digium.com) |
| 19:52.28 | ian_k | ig nickserv |
| 19:52.59 | *** join/#asterisk Laibsch (n=Laibsch@p54B98501.dip0.t-ipconnect.de) |
| 19:53.00 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 19:53.20 | *** join/#asterisk Veto_laptop (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 19:56.30 | *** join/#asterisk netrin0 (n=neutrin0@CPE000d8802a707-CM0011e6c7edb1.cpe.net.cable.rogers.com) |
| 19:56.47 | netrin0 | greetings |
| 19:59.04 | lancey | darkskiez: how do you call it? |
| 19:59.43 | darkskiez | call what? |
| 20:00.23 | *** join/#asterisk Qwell (n=chatzill@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 20:01.07 | lancey | [22:44] <darkskiez> justinu: i can issue commands via asterisk -r |
| 20:01.15 | BrianR___ | Gah. I've been getting reports of gxp-2000 phones in my deployment which are heating up and crashing :( |
| 20:01.23 | darkskiez | lancey: i type them at the prompt that comes up |
| 20:01.40 | lancey | i does work ok |
| 20:01.52 | lancey | Asterisk CVS HEAD built by root@insomnia.net1.cc on a i386 running FreeBSD on 2005-10-10 23:32:05 UTC |
| 20:02.01 | darkskiez | lancey: the commands i type do operate, but nothing gives feedback |
| 20:02.15 | lancey | yeah, i read that |
| 20:02.16 | lancey | it IS ok |
| 20:02.22 | lancey | insomnia*CLI> show version |
| 20:02.23 | lancey | Asterisk CVS HEAD built by root@insomnia.net1.cc on a i386 running FreeBSD on 2005-10-10 23:32:05 UTC |
| 20:02.28 | lancey | for e.g. |
| 20:02.34 | lancey | 'help' also works |
| 20:02.38 | lancey | as anything else |
| 20:02.52 | darkskiez | show version doesnt work, help doesnt work |
| 20:03.12 | darkskiez | i just get a prompt back |
| 20:03.15 | darkskiez | no error |
| 20:03.22 | lancey | it should be some issue at your side |
| 20:03.41 | lancey | tried restarting * ? |
| 20:03.49 | darkskiez | lancey: lots of times |
| 20:03.58 | lancey | dunno then... |
| 20:04.18 | darkskiez | asterisk -c work |
| 20:04.33 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 20:04.35 | darkskiez | but without -c, i cant connect and issue commands. |
| 20:05.51 | *** join/#asterisk Assid (n=assid@203.115.64.57) |
| 20:05.54 | Assid | umm |
| 20:06.01 | Assid | anyone's HEAD version working? |
| 20:06.17 | Assid | i cant compile |
| 20:06.18 | Assid | ast_expr2f.c:1784: warning: no previous prototype for `ast_yyget_column' |
| 20:06.19 | Assid | ast_expr2f.c:1860: warning: no previous prototype for `ast_yyset_column' |
| 20:06.19 | Assid | ast_expr2.fl:95: error: conflicting types for `ast_expr' |
| 20:06.19 | Assid | include/asterisk/ast_expr.h:26: error: previous declaration of `ast_expr' |
| 20:06.43 | *** join/#asterisk Tili (i=Tili@202-133-65-122-dialup.sat.net.pk) |
| 20:06.50 | netrin0 | greets |
| 20:07.30 | netrin0 | i'm getting an unknown symbol err but the docs don't mention anything about add_preemp_count |
| 20:07.54 | netrin0 | *add_preempt_count |
| 20:08.02 | netrin0 | any suggestions? |
| 20:08.04 | Assid | anyone got head working? |
| 20:08.36 | darkskiez | upgrade yacc i think |
| 20:08.50 | netrin0 | yacc? |
| 20:08.55 | Qwell | yet another compiler compiler |
| 20:08.56 | netrin0 | ok tks |
| 20:09.12 | Qwell | or something |
| 20:09.13 | Assid | anyone know about my issue |
| 20:09.15 | Assid | with head? |
| 20:09.26 | Assid | first off.. * just stops working |
| 20:09.29 | Assid | so i said okay.. |
| 20:09.30 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 20:12.04 | Corydon-w | Those of us on Linux don't use yacc... we use bison... |
| 20:12.34 | darkskiez | thats the one i was looking for |
| 20:12.48 | Corydon-w | yacc/bison... get it? |
| 20:12.58 | darkskiez | yes, of course |
| 20:13.06 | Corydon-w | Okay, just checking |
| 20:13.36 | darkskiez | i was thinking moose and buffalo, and then i lost track and started watching tv. |
| 20:13.38 | *** join/#asterisk nagl (n=nagl@213.235.241.6) |
| 20:14.17 | darkskiez | and getting frustrated at the now longstanding bug i've encountered, but dont want to file a bug report for coz I dont want bad karma. |
| 20:15.34 | *** join/#asterisk Rusty5000 (i=user@ip70-187-228-86.dc.dc.cox.net) |
| 20:16.16 | Rusty5000 | Hey guys, anyone know whate happened to the asterisk 1.1 tree? I was running 1.1, accidentally blew it away ... 1.2 is supposed to support the same feature set, but it broke a few things |
| 20:16.24 | *** join/#asterisk Bentley (n=Bentley@S0106000f3d016dd2.cg.shawcable.net) |
| 20:17.07 | Bentley | hey all, does anyone know what this means when I access DISA? "Generator got voice, switching to phase locked mode" |
| 20:20.38 | *** join/#asterisk Corydon-w (i=brown@pdpc/supporter/sustaining/Corydon76-home) |
| 20:23.29 | *** join/#asterisk marc324 (n=marc3234@206-248-159-56.dsl.teksavvy.com) |
| 20:25.15 | *** part/#asterisk Bentley (n=Bentley@S0106000f3d016dd2.cg.shawcable.net) |
| 20:26.09 | Assid | okay weird error |
| 20:26.14 | Assid | timing is totally messed up |
| 20:26.19 | Assid | sometimes i dont hear the playback |
| 20:26.38 | Assid | then sometimes i hear it just as soon as it reaches the end of the voicemail greeting |
| 20:26.43 | *** part/#asterisk clive- (n=pirch@ndn-165-135-236.telkomadsl.co.za) |
| 20:26.48 | Rusty5000 | Hey guys, anyone know whatever happened to the asterisk 1.1 tree? |
| 20:26.53 | *** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net) |
| 20:27.42 | Assid | this is weird |
| 20:27.50 | Assid | all ym asterisk boxes decided to die today |
| 20:28.31 | wunderkin | 1.1 was skipped |
| 20:29.21 | Assid | anyone know what could be the issue here? |
| 20:29.42 | Rusty5000 | wunderkin: there used to be a 1.1 branch under the CVS tree |
| 20:29.56 | lancey | it's now 1.2 i believe |
| 20:30.15 | Rusty5000 | yes, problem is, I was running 1.1, accidentally blew it away ... 1.2 is supposed to support the same feature set, but it broke a few things |
| 20:30.24 | Tili | has anybody tried any voip over GPRS. does it work or its not worth the effort |
| 20:32.34 | Assid | wow,.. i got too many problems toda |
| 20:32.53 | file | Tili: too much latency, not enough bandwidth |
| 20:34.38 | Tili | file: I think it can work on 3G though |
| 20:38.34 | jake1932 | someone has tested it on 3g |
| 20:38.38 | BrianR___ | voice will work over evdo or gprs+edge |
| 20:38.41 | jake1932 | let me see if i can find the article |
| 20:38.50 | BrianR___ | I've even got voice to work over 1xRTT, which is technically a 2.5G feature. |
| 20:39.41 | jake1932 | it was EVDO: http://www.evdoinfo.com/Tips/PC_5220/Using_VOIP_over_EVDO_-_Testing_Vonage_over_EV-DO_20050321138 |
| 20:40.14 | BrianR___ | you could prolly run g711 over evdo :) |
| 20:41.11 | nestAr | anyone know where I can get firmware for the UTStarCom F1000 phone? Doesn't look like i can download it from their site. |
| 20:41.14 | *** join/#asterisk tainted_ (n=identd@ppp-71-137-169-240.dsl.irvnca.pacbell.net) |
| 20:41.25 | Assid | umm |
| 20:41.29 | *** join/#asterisk KranZ (n=user@sme.bestline.net) |
| 20:41.34 | Assid | my asterisk box doesnt work anymore |
| 20:41.40 | Assid | on its own |
| 20:41.51 | tainted_ | "on its own"? |
| 20:41.55 | Assid | yeah |
| 20:41.56 | *** join/#asterisk MuppetMaster (n=MuppetMa@169.red-81-184-73.user.auna.net) |
| 20:41.57 | Assid | i didnt do anything |
| 20:41.57 | KranZ | my asterisk box left me for another admin |
| 20:42.02 | Assid | it just died |
| 20:42.03 | KranZ | the slut |
| 20:42.03 | *** part/#asterisk Rusty5000 (i=user@ip70-187-228-86.dc.dc.cox.net) |
| 20:42.06 | Assid | now refuses to loadup |
| 20:42.09 | Assid | so.. |
| 20:42.13 | Assid | i got the latest CVS.. that didnt help |
| 20:42.14 | tainted_ | hardware problem? |
| 20:42.18 | *** join/#asterisk DukeOfURL (n=chatzill@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 20:42.19 | Assid | the timing just went bad |
| 20:42.23 | Assid | nah.. |
| 20:42.28 | Assid | pretty new machine |
| 20:42.41 | KranZ | was it after you did a cvs update? |
| 20:42.47 | Assid | nope |
| 20:42.55 | MuppetMaster | Has anyone here had any luck getting app_notify.so to work? http://forums.digium.com/viewtopic.php?p=5530#5530 |
| 20:42.58 | Assid | before.. it died.. after cvs update.. timing went weird |
| 20:43.03 | Assid | so i said okay.. lets use the beta |
| 20:43.12 | Assid | beta keeps giving me weird issues |
| 20:43.24 | KranZ | MuppetMaster: what r u using it for? |
| 20:43.39 | MuppetMaster | KranZ: To send notifications to Growl on an OSX machine of incoming calls. |
| 20:44.05 | KranZ | growl looks cool |
| 20:44.12 | Assid | also.. qualify always shows unreachable |
| 20:44.34 | PupenoL | What does the message "zaptel Disabled echo canceller because of tone (rx) on channel N" mean exactly ? what is that tone ? |
| 20:44.46 | MuppetMaster | KranZ: Growl is great, use it with all sorts of apps, would be great to use it with Asterisk too. |
| 20:45.55 | *** part/#asterisk MuppetMaster (n=MuppetMa@169.red-81-184-73.user.auna.net) |
| 20:46.36 | Assid | anyone know what i can do ? |
| 20:47.16 | SwK[Work] | is ztdummy b0rked? |
| 20:47.40 | Nugget | you mean more than usual? :) |
| 20:47.43 | SwK[Work] | yeah |
| 20:47.44 | SwK[Work] | ;) |
| 20:47.53 | SwK[Work] | FATAL: Error inserting ztdummy (/lib/modules/2.6.13-gentoo-r3/misc/ztdummy.ko): Input/output error |
| 20:47.56 | SwK[Work] | FATAL: Error running install command for ztdummy |
| 20:48.16 | Assid | for cvs.. yes |
| 20:48.16 | Assid | i think so atleast |
| 20:48.16 | Assid | like when i call my own extension |
| 20:48.16 | Assid | i cant hear anything |
| 20:48.17 | Assid | then.. when it starts going to voicemail |
| 20:48.20 | Assid | i start hearing |
| 20:48.21 | SwK[Work] | when all else fails blow it away and recheckout |
| 20:48.47 | *** join/#asterisk oej (n=Olle@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 20:49.03 | *** part/#asterisk Utah_Dave (n=boucha@0-1pool139-69.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 20:50.46 | *** join/#asterisk Ayano (n=erik_lee@adsl-70-245-190-90.dsl.spfdmo.swbell.net) |
| 20:51.00 | Assid | do i delete the modules ? |
| 20:51.15 | Ayano | how hard is it to bring a data and a voice t1 strait into an asterisk box? |
| 20:52.10 | Assid | SwK[Work]: are you using CVS? |
| 20:52.45 | SwK[Work] | yeah I'm using cvs |
| 20:52.55 | SwK[Work] | Ayano: not to hard at all |
| 20:53.05 | Assid | and voipbuster isnt working for me either today |
| 20:53.07 | Assid | wow |
| 20:53.09 | Assid | this day sucks |
| 20:53.09 | SwK[Work] | Ayano: theres examples of doing that |
| 20:54.42 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 20:55.51 | *** join/#asterisk justinu (n=j2@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 20:58.51 | Assid | this isnt working right |
| 20:59.05 | *** join/#asterisk Ayano (n=Ayano_@adsl-70-245-190-90.dsl.spfdmo.swbell.net) |
| 20:59.59 | Tili | what is the advantage of trunks on IAX2 |
| 21:00.30 | Assid | SwK[Work]: it doesnt work |
| 21:01.31 | *** join/#asterisk MnDBnDr (n=MnDBnDr@cpe-24-93-166-19.neo.res.rr.com) |
| 21:02.23 | Ayano | SwK, you still around? |
| 21:03.54 | SwK[Work] | Assid: what doesnt work? zaptel from CVS HEAD? |
| 21:06.31 | *** join/#asterisk mcn (n=mcn@ext-gw.newtoncomputing.co.uk) |
| 21:06.36 | *** join/#asterisk deexm (n=deexm@scrot.us) |
| 21:06.38 | MnDBnDr | can ne1 help me with setting up oh323 extensions for Netvision phones? |
| 21:09.09 | Assid | SwK[Work]: not sure |
| 21:09.13 | Assid | its like this |
| 21:09.20 | Assid | wehn i call an extension |
| 21:09.32 | Assid | first i get Oct 11 17:08:52 NOTICE[9387]: app_dial.c:1109 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 21:09.49 | Assid | also i get ..... "hangup or press the pound key" for the voicemail |
| 21:09.54 | Assid | the beginning part |
| 21:09.54 | Assid | i cant hear |
| 21:10.36 | *** join/#asterisk kiwnix (n=egarcia@82.158.159.104) |
| 21:14.00 | SwK[Work] | that looks like you are trying to call a sip device that you dont have a route to |
| 21:14.39 | Assid | 2 issues |
| 21:14.42 | Assid | when i call myself |
| 21:14.48 | Assid | i can get the route |
| 21:14.50 | SwK[Work] | No Route to Destination means the SIP device most likely isnt registered or you are sending the call to an invalid IP address and telling me "it doesnt work" has not thing to do with zap |
| 21:14.53 | Assid | butr.. can never hear the first part of the call |
| 21:15.06 | SwK[Work] | which is what I asked about |
| 21:16.28 | Assid | okay in the the one that the route doesn towrk.. yeah.. it seems that box cant access the internal phones.. |
| 21:16.36 | Assid | BUT.. the other box can.. so i gotta see why not there |
| 21:16.58 | Assid | but.. as of the one where i cant hear the beginning part of the conversation. I dont know.. why |
| 21:17.03 | *** join/#asterisk clyrrad (n=ddd@CPE001195f553c7-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 21:18.15 | *** join/#asterisk Katty (n=katrina@68-112-15-110.dhcp.cpgr.mo.charter.com) |
| 21:18.49 | *** join/#asterisk zotz (n=zotz@24.231.36.100) |
| 21:19.01 | Assid | SwK[Work] : im confused arent i |
| 21:19.56 | *** join/#asterisk ManxPower (n=eric@adsl-70-247-221-174.dsl.lgvwtx.swbell.net) |
| 21:21.31 | *** join/#asterisk mflorell (n=astmattf@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 21:21.33 | *** join/#asterisk shmaltz (n=chatzill@69.28.255.210) |
| 21:21.37 | shmaltz | helo every1 |
| 21:21.52 | MnDBnDr | hi schmaltz |
| 21:22.41 | *** join/#asterisk tombone (n=tombone@c-67-180-181-221.hsd1.ca.comcast.net) |
| 21:22.45 | MnDBnDr | I have 3 Symbol Netvision H.323 phone new in the box to setup on my * box |
| 21:23.02 | MnDBnDr | need a little help with oh323 |
| 21:23.23 | MnDBnDr | I have 3 Zyxel SIP phone working great. |
| 21:23.38 | MnDBnDr | just need a little help with oh323 |
| 21:23.41 | MnDBnDr | it is installed |
| 21:23.53 | tombone | Hi, I'm running CVS Head on Linx 2.6.13 -- after modprobe zaptel and ztdummy, sound playbak doesn't seem to work. |
| 21:23.59 | *** part/#asterisk mflorell (n=astmattf@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 21:23.59 | tombone | Any ideas? |
| 21:24.12 | MnDBnDr | now to configure oh323 extensions |
| 21:24.13 | Ayano | which digium cards are possible to use for both data and voice? |
| 21:26.11 | tombone | Any clues on why ztdummy seems to interfere with sound playback? |
| 21:26.16 | tombone | I'm also getting :chan_iax2.c:3035 iax2_read: I should never be called! |
| 21:26.29 | tombone | whereas I wasn't before modprobing ztdummy |
| 21:28.34 | tombone | hello? anyone here? |
| 21:29.03 | darkskiez | they are all at astricon |
| 21:29.34 | tombone | no laptops? wireless? |
| 21:29.39 | MnDBnDr | anyone use oh323 extensions |
| 21:29.43 | MnDBnDr | hehe |
| 21:29.44 | MnDBnDr | yea |
| 21:29.55 | tombone | OpenPBX is going to take over |
| 21:30.05 | darkskiez | yeh, but they are in the conference meetings. |
| 21:30.14 | tombone | if they keep this up |
| 21:30.55 | tombone | damn... are there reported issues with running asterisk on SMP kernels? |
| 21:31.37 | *** join/#asterisk wundaboy (n=asdf@c-67-164-107-68.hsd1.or.comcast.net) |
| 21:32.05 | tombone | what time is astricon over? |
| 21:32.27 | darkskiez | a quarter past saturday |
| 21:33.05 | sahafeez | interstering question |
| 21:33.41 | sahafeez | can i dial a remote sip thur my asterisk server. so my buddy had an address at www.voipformycompany.com/1111 |
| 21:35.01 | *** join/#asterisk shido6 (n=shido6@d221-68-210.commercial.cgocable.net) |
| 21:35.31 | MnDBnDr | sahafeez check PM |
| 21:35.51 | sahafeez | PM? |
| 21:36.03 | Assid | SwK[Work]: any clue why it doesnt start playing the audio when the call starts |
| 21:36.04 | MnDBnDr | the private message i sent you |
| 21:36.06 | Assid | it used to work before |
| 21:36.53 | MnDBnDr | see it now sahafeez |
| 21:37.55 | Assid | hey shido6 |
| 21:38.11 | *** part/#asterisk Ayano (n=Ayano_@adsl-70-245-190-90.dsl.spfdmo.swbell.net) |
| 21:38.58 | *** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net) |
| 21:39.59 | shido6 | ? |
| 21:40.00 | shido6 | hey |
| 21:40.05 | shido6 | yes assid whats up? |
| 21:40.11 | *** join/#asterisk Malthus (n=admin@port0129-abn-s-adsl.cwjamaica.com) |
| 21:40.19 | Hmmhesays | shido6 long time no argue |
| 21:40.20 | Assid | screwed |
| 21:40.27 | Assid | my boxes refuse to work |
| 21:40.39 | Assid | the main one which is live.. refuses to work correctly |
| 21:41.02 | Assid | i make a call.. first thing.. i cant hear the first part of the call/conversation |
| 21:41.06 | Assid | first 5-10 seconds |
| 21:41.22 | shido6 | stop mistreating your boxes |
| 21:41.32 | shido6 | molest them gently |
| 21:41.48 | Hmmhesays | "you don't always have t fsck her hard, in fact sometimes thats not right to do" |
| 21:42.11 | *** part/#asterisk Malthus (n=admin@port0129-abn-s-adsl.cwjamaica.com) |
| 21:42.26 | MnDBnDr | does anyone have a good doc or wiki link for oh323 extensions? |
| 21:42.41 | shido6 | yeah but its $85 |
| 21:42.42 | *** join/#asterisk starman (n=inshift@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 21:42.49 | MnDBnDr | hehe |
| 21:42.57 | Assid | ??????? |
| 21:43.08 | Assid | me? |
| 21:43.35 | MnDBnDr | I am just about to sell these 3 new Netvision phones and get the Zyxel SIP phones that I already have. They work great |
| 21:44.18 | Assid | okay heres something weird |
| 21:44.32 | Assid | i can ping from a winbox to the voip linux box |
| 21:44.44 | Assid | and from the winbox to the ip phones |
| 21:44.56 | Assid | but.. frm the voip box.. cant ping the ip phones |
| 21:45.38 | groogs | Assid: have a firewall? |
| 21:45.42 | Assid | nope |
| 21:45.45 | InfraRed | are the phones on a different subnet |
| 21:45.45 | Assid | no firewall |
| 21:45.55 | *** join/#asterisk phpboy (i=flipside@tbnb-165-223-63.telkomadsl.co.za) |
| 21:45.57 | nomazda | phones have a gateway? |
| 21:45.58 | InfraRed | are they all in the same IP range |
| 21:46.01 | Assid | yeah |
| 21:46.09 | Assid | they all were working before |
| 21:46.13 | Assid | and yes.. same ip range |
| 21:46.22 | InfraRed | what os on the linux box |
| 21:46.27 | phpboy | hey guys, I can't seem to find nice documention asterisk... regarding... 'registration tokens' does anybody have nice docs for me on that? |
| 21:46.28 | InfraRed | erm distro |
| 21:46.40 | Assid | debia |
| 21:46.44 | Assid | debian |
| 21:47.21 | Assid | 1 have 2 ip ranges.. but i have the windows andlinux box sharing both the ip ranges |
| 21:47.22 | InfraRed | after you ping the phones from the windows box |
| 21:47.25 | InfraRed | type arp -a |
| 21:47.30 | InfraRed | whats the mac address of the phone |
| 21:47.44 | InfraRed | stop |
| 21:47.48 | InfraRed | 2 ip ranges? |
| 21:47.52 | Assid | yes.. |
| 21:47.54 | Assid | worked fine till now |
| 21:48.02 | InfraRed | that's your problem |
| 21:48.12 | InfraRed | check your network settings and configuration and wiring |
| 21:48.27 | Assid | arp -a on linux shows it as well |
| 21:48.43 | InfraRed | i bet you miswired your network |
| 21:48.45 | InfraRed | check that |
| 21:48.48 | shido6 | :) |
| 21:48.49 | Assid | nah.. nothing changed |
| 21:48.58 | Assid | oh yeah.. i saw the registration take place |
| 21:49.03 | Assid | of the phone |
| 21:49.11 | Assid | 1 of them.. that i played with |
| 21:49.29 | InfraRed | obviously something changed |
| 21:49.41 | InfraRed | check first |
| 21:51.10 | phpboy | I'm dealing with a company who's pbx can only pass me a registration token... but I dunno how to do annonymous SIP logins :< |
| 21:51.22 | Assid | what i dont understand is how 1/2 works.. 1/2 doesnt |
| 21:51.32 | *** part/#asterisk starman (n=inshift@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 21:52.36 | *** join/#asterisk phpboy (n=shane@c1-154-14.tbnb.isadsl.co.za) |
| 21:52.46 | phpboy | can anybody help me with anonymous SIP logins? |
| 21:53.24 | InfraRed | phpboy: did you add the registeration line? |
| 21:53.35 | phpboy | I did |
| 21:53.40 | clyrrad | I have been following this http://www.voip-info.org/wiki-Asterisk+cdr+mysql for MYSQL CDRs and for some reason all goes well except a cdr_addon_mysql.so is never created, any idea why that is? |
| 21:53.57 | InfraRed | clyrrad: it's in asterisk-addons |
| 21:54.01 | InfraRed | download and compile that |
| 21:54.07 | clyrrad | InfaRed ... I did that |
| 21:54.22 | InfraRed | did compile work ? |
| 21:54.27 | clyrrad | I donwload as instructed, when into asterisk-addons folder make and make install |
| 21:54.29 | clyrrad | no errors |
| 21:54.29 | terrapen | http://tinyurl.com/a5mm3 |
| 21:54.32 | terrapen | *lust* |
| 21:54.40 | phpboy | InfraRed: any suggestions? |
| 21:55.09 | InfraRed | phpboy: and what happens when you 'sip show registry |
| 21:55.27 | InfraRed | clyrrad: did you 'make install' ? |
| 21:55.36 | fugitivo | clyrrad: mysql is evil |
| 21:55.38 | clyrrad | InfraRed yes i did |
| 21:55.42 | *** join/#asterisk outsidefactor (n=blah@203-217-71-90.dyn.iinet.net.au) |
| 21:55.43 | phpboy | InfraRed: if I run asterisk -vvvvvvr I should see any connection coming in... right? even if it's refused... |
| 21:56.02 | InfraRed | phpboy: you forgot c |
| 21:56.07 | InfraRed | -rcvvvvvvvvvvvvvvvvvv |
| 21:56.20 | InfraRed | as many v's as you can be arsed typibng |
| 21:57.10 | clyrrad | InfaRed whats odd is I have the C file, but not SO after the compile finishes |
| 21:57.14 | InfraRed | terrapen: i bet it doesnt have broadband coverage |
| 21:57.26 | Assid | InfraRed: okay.. but the first 5 seconds.. its just blank |
| 21:57.30 | Assid | thats anotherissue |
| 21:57.35 | InfraRed | clyrrad: then it doesnt compile |
| 21:57.43 | InfraRed | Assid: ? |
| 21:57.53 | Assid | like if i call you |
| 21:57.56 | terrapen | infrared, well, the first place will |
| 21:57.59 | Assid | its like a delay |
| 21:58.05 | clyrrad | InfarRed what is your suggestion? I did not have any errors when I compiled |
| 21:58.06 | terrapen | Sundance is about 15mi from Provo, IIRC |
| 21:58.08 | Assid | its yet ringing.. and i know the other line has already picku |
| 21:58.13 | terrapen | you could easily run a T1 up there |
| 21:58.15 | Assid | themn.. another 3 seconds later |
| 21:58.18 | Assid | i can start hearing |
| 21:58.26 | terrapen | may be even close than that to Provo |
| 21:58.27 | Assid | BUT.. when the conversation is on. then its fine |
| 21:58.37 | InfraRed | Assid: check your network |
| 21:58.41 | terrapen | the second place would work with satellite internet |
| 21:59.04 | terrapen | i'd give up broadband to own one of those |
| 21:59.18 | InfraRed | well you need 1.6mil too |
| 21:59.29 | terrapen | it would be my vacation home. i'd go up there to ski, snowboard, and drive and work on my truck |
| 21:59.54 | Katty | hi. |
| 22:00.51 | syle2 | i think 4 v's to start asterisk will do the same as if you had 100000 v's |
| 22:00.55 | phpboy | InfraRed: I'm running the daemon live... |
| 22:01.52 | InfraRed | phpboy: ok, then sip debug |
| 22:02.07 | InfraRed | and turn logging on, loads of crap will pass through |
| 22:02.19 | lancey | syle2 i believe there are verbose levels higher than 4, though |
| 22:02.56 | phpboy | InfraRed: yeah... what I'm looking at doing is seeing what data get's passed to the box and then I'll beable to figure out how to traslate it... but now I have nothing :/ |
| 22:05.11 | syle2 | lacey prove it |
| 22:05.30 | *** join/#asterisk netsurfer (n=bbjunkie@i-83-67-48-18.freedom2surf.net) |
| 22:06.25 | lancey | syle2: i can't right now, but i recall seeing verbose levels 5 and 6 in the sources |
| 22:06.44 | lancey | though it could have been long ago |
| 22:06.53 | *** part/#asterisk wrmem (n=monnin@monnin-win.cso.uiuc.edu) |
| 22:08.31 | lancey | ot could be bad memory too ;) |
| 22:08.33 | lancey | *it |
| 22:09.40 | *** part/#asterisk mkrufky (n=mk@68.160.103.77) |
| 22:10.10 | *** part/#asterisk pooh_ (n=hfwang@cust.15.241.adsl.cistron.nl) |
| 22:10.24 | terrapen | I wonder if the OpenBSD package for Asterisk is decent |
| 22:11.31 | phpboy | BSD still sucks at the mo |
| 22:11.32 | phpboy | :/ |
| 22:11.45 | terrapen | uhh, i've never had a problem with it |
| 22:11.47 | lancey | phpboy: i'm using * on FreeBSD |
| 22:11.51 | *** join/#asterisk ian_k (n=ian@gateway.digium.com) |
| 22:11.53 | terrapen | Ran * on Mac OS X for quite a while |
| 22:11.59 | lancey | no problems at all |
| 22:12.07 | terrapen | I don't think you have a clue, phpboy |
| 22:12.08 | phpboy | lancey: it works nicely |
| 22:12.22 | phpboy | isdn4bsd isn't widly supported though :/ |
| 22:12.25 | phpboy | not yet at least |
| 22:12.26 | terrapen | well, yeah |
| 22:12.33 | ian_k | ls |
| 22:12.36 | lancey | well, this does not mean BSD sucks :) |
| 22:12.38 | terrapen | but who uses isdn? :P |
| 22:12.39 | phpboy | I need support for the ISDN QUAD card before I'll move to BSD (again) |
| 22:12.50 | phpboy | lancey: I love BSD |
| 22:13.00 | phpboy | terrapen: my country sucks :/ |
| 22:13.07 | lancey | phpboy so does mine :) |
| 22:13.18 | phpboy | nice |
| 22:13.23 | lancey | that's my we run everything over IP |
| 22:13.26 | lancey | *why |
| 22:13.32 | netsurfer | all countries suck, just some more than others ;) |
| 22:13.38 | phpboy | but I'll def move to BSD when they have support for the quad card |
| 22:14.20 | terrapen | I'm going to run Asterisk on OpenBSD/macppc |
| 22:14.25 | shido6 | it works |
| 22:14.34 | shido6 | terrapen |
| 22:14.37 | terrapen | If I had an SGI O2, I'd run it on OpenBSD/sgi |
| 22:14.39 | terrapen | :P |
| 22:14.48 | terrapen | shido, cool |
| 22:14.53 | shido6 | it works on sgi hardware, too |
| 22:15.13 | terrapen | is anybody running Asterisk on IRIX? ;) |
| 22:15.41 | phpboy | Was running it on BSD for a while... you gotta LOVE ports :D |
| 22:16.28 | phpboy | FreeBSD though |
| 22:16.50 | *** join/#asterisk justinu (n=j2@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 22:18.13 | *** join/#asterisk jeremywhiting (n=jeremy@71-37-101-103.slkc.qwest.net) |
| 22:19.09 | justinu | hey asterisknoob |
| 22:19.10 | justinu | you there? |
| 22:21.57 | phpboy | how can I force asterisk to use a codec of my choice? |
| 22:23.44 | *** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca) |
| 22:25.51 | tombone | Anyone have a clue about ztdummy? |
| 22:26.57 | tombone | Anyone know what this means? tvfix: warning too large timestamp? |
| 22:31.32 | ManxPower | phpboy, you force it with disallow=all allow=thecodecyouwant in each section if (sip|iax|mgcp).conf |
| 22:32.15 | *** join/#asterisk shidan (n=shidan@CPE0013107d30c4-CM001371871af0.cpe.net.cable.rogers.com) |
| 22:34.50 | ManxPower | tombone, sounds like an issue with the jitterbuffer |
| 22:38.59 | *** join/#asterisk evoinc (n=sponger@66-193-153-253.gen.twtelecom.net) |
| 22:39.18 | evoinc | hello? |
| 22:40.15 | *** join/#asterisk darwin35 (n=darwin35@208.139.193.178) |
| 22:40.18 | evoinc | hello |
| 22:40.38 | tuppa | ?olleh |
| 22:41.00 | darwin35 | you are here |
| 22:41.04 | evoinc | thanks |
| 22:41.09 | darwin35 | the channel might be quiet |
| 22:41.09 | *** join/#asterisk alohatone (n=anakaoka@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 22:41.29 | darwin35 | they can all be napping |
| 22:41.37 | evoinc | i guess so |
| 22:41.40 | darwin35 | at thier ages they need it |
| 22:41.43 | evoinc | I was wondering if anyone was seeing me |
| 22:42.01 | tuppa | I just came in to work |
| 22:42.30 | phpboy | how do I install a new codec... eg G711? |
| 22:45.53 | evoinc | this room is dead |
| 22:46.45 | phpboy | ManxPower: how can I change what the SIP client in asterisk uses? |
| 22:47.10 | evoinc | edit the sip.conf and then disallow=akk |
| 22:47.23 | evoinc | allow=$<CODEC> |
| 22:47.27 | evoinc | oops |
| 22:47.29 | evoinc | disallow=all |
| 22:47.55 | phpboy | exten => _12345678 ,1,Dial(IAX2/${IAXINFO}@10.10.10.1/${EXTEN:1}@testphone) |
| 22:48.02 | phpboy | for something like that... |
| 22:48.17 | phpboy | must I add those rules per context? |
| 22:48.27 | evoinc | no |
| 22:48.33 | evoinc | you add them per "channel" |
| 22:48.38 | *** join/#asterisk Carp1 (n=c@cpe-69-205-42-57.nycap.res.rr.com) |
| 22:48.46 | evoinc | so in your sip.conf you have a channel setup for each client |
| 22:49.08 | phpboy | ok |
| 22:50.23 | evoinc | you can actually setup globally your codecs in the sip.conf so you do not have to do it for each connection.. but if your sip accounts connect from places of varying bandwidth it may benifit you to specify them per account |
| 22:52.22 | evoinc | hello |
| 22:52.24 | evoinc | oops |
| 22:52.27 | evoinc | hahahah sorry |
| 22:53.25 | *** join/#asterisk xyharley (n=harley@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 22:53.45 | evoinc | '/quyit |
| 22:56.43 | phpboy | I need to set the codec for the outgoing channel |
| 22:58.12 | niZon | anyone know where i can find the dial failure reason codes? |
| 22:58.14 | niZon | for example.. |
| 22:58.39 | niZon | blah blah blah call failed to go through, reason X |
| 22:58.44 | *** part/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net) |
| 23:00.13 | *** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net) |
| 23:01.44 | niZon | hm |
| 23:03.04 | *** join/#asterisk scubasteve (n=steve@cpe-071-065-212-199.nc.res.rr.com) |
| 23:03.19 | scubasteve | Has anyone set up MOH to pull audio from the soundcard? |
| 23:03.39 | phpboy | how can I see which codec my client is trying to use? |
| 23:03.39 | niZon | i tried, and failed |
| 23:03.42 | niZon | then gave up |
| 23:03.47 | phpboy | where in the logs atleast |
| 23:04.42 | niZon | it'll say it in the CLI |
| 23:04.52 | niZon | if you have verbose set to some high number |
| 23:05.02 | niZon | -- Call accepted by 205.234.133.203 (format ulaw) |
| 23:05.07 | scubasteve | Yep |
| 23:05.08 | niZon | like that |
| 23:05.27 | *** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8) |
| 23:06.02 | *** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net) |
| 23:06.37 | phpboy | may I paste 3 lines please? |
| 23:06.49 | denon | yes |
| 23:06.54 | phpboy | Capabilities: us - 0x101 (g723|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x0 (nothing) |
| 23:06.55 | phpboy | Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 (nothing) |
| 23:06.55 | phpboy | Oct 11 21:02:52 NOTICE[8317]: chan_sip.c:2815 process_sdp: No compatible codecs! |
| 23:07.03 | phpboy | that's the 'error' that I get |
| 23:07.20 | phpboy | but I've got allow=g729 |
| 23:07.29 | phpboy | and it doesn't work |
| 23:07.38 | phpboy | but if I change it to allow=all then it works :< |
| 23:08.14 | nomazda | hmm.. still cant figure out how to get vbuzzer working .. silly proxies |
| 23:08.14 | lancey | phpboy it seems the peer support alaw only |
| 23:08.23 | lancey | i mean ulaw |
| 23:09.04 | ManxPower | phpboy, you force it with disallow=all allow=thecodecyouwant in each section if (sip|iax|mgcp).conf |
| 23:09.26 | ManxPower | phpboy, The client will use whatever codec you force in Asterisk, as long as the client supports that codec |
| 23:11.07 | ManxPower | phpboy, Do you have a licensed G729 codec? |
| 23:11.22 | ManxPower | Also the PEER is not saying that it supports G729 |
| 23:11.44 | ManxPower | niZon, I believe the code numbers are the same as the ISDN PRI cause codes. |
| 23:12.11 | phpboy | hmmm.... the I should use g723? it supports that from what I can see |
| 23:12.25 | lancey | peer - audio=0x4 (ulaw)/video=0x0 (nothing) |
| 23:12.29 | ManxPower | But you can see the specific ones in asterisk/include/asterisk/causes.h |
| 23:12.35 | lancey | your peer support ulaw *only* |
| 23:12.49 | ManxPower | phpboy, Asterisk does not support G723 and it only supports G729 if you have a license |
| 23:13.08 | ManxPower | looks like someone screwed up the config on the SIP device |
| 23:13.20 | phpboy | I see... |
| 23:13.25 | phpboy | ManxPower: on the client side? |
| 23:13.30 | lancey | ManxPower it could be made to support g723, though not officially |
| 23:13.36 | ManxPower | The SIP device, as lancy pointed out, says it only supports ULAW |
| 23:13.50 | ManxPower | lancey, I did not see you type that. |
| 23:13.55 | *** part/#asterisk darkskiez (n=darkskie@host86-138-169-183.range86-138.btcentralplus.com) |
| 23:14.03 | lancey | :) |
| 23:14.36 | lancey | phpboy have access to the peer configuration? |
| 23:15.01 | ManxPower | Asterisk can be made to do many things that are illegal. |
| 23:15.03 | phpboy | I do not |
| 23:15.05 | ManxPower | So can a crowbar |
| 23:15.26 | *** join/#asterisk tainted_ (n=somewher@mail.k2usa.com) |
| 23:15.26 | *** join/#asterisk Knight_DKN (n=knight_d@61.95.68.85) |
| 23:15.32 | ManxPower | phpboy, well, if the SIP device is only claiming to support ulaw, then that's the codec you have to use. |
| 23:15.35 | lancey | ManxPower i don't need g.723 at all, just pointing it out |
| 23:15.44 | lancey | btw, is there a way to get a licensed g.723 codec? |
| 23:15.50 | Knight_DKN | Goodmorning boys and girls |
| 23:16.27 | phpboy | I just broke something in my sip.conf |
| 23:16.27 | phpboy | :/ |
| 23:17.25 | Knight_DKN | Does anyoner have problems with their Cisco 79xx's losing thier clocks? |
| 23:17.48 | denon | Knight_DKN: yes |
| 23:17.55 | Nugget | Knight_DKN: are you using ntp? |
| 23:17.55 | lancey | Knight_DKN they must support ntp, though |
| 23:18.05 | phpboy | exten => _1234567,1,Dial(IAX2/${IAXINFO}@10.10.10.1/${EXTEN:1}@testphone) is no longer working after I pushed the wrong button in sip.conf :/ |
| 23:18.11 | tombone | has anyone here had any luck installing sphinx? |
| 23:18.13 | phpboy | What could I have broken? |
| 23:18.43 | lancey | phpboy this used to work?! |
| 23:18.54 | phpboy | here's the errors... Oct 11 21:14:21 WARNING[14400]: channel.c:2189 ast_channel_make_compatible: No path to translate from Modem[i4l]/ttyI0(64) to SIP/10.10.10.1:5060-5000(256) |
| 23:19.14 | lancey | this is codec issue again |
| 23:19.28 | lancey | trying to transcode without having licenses |
| 23:19.31 | phpboy | Oct 11 21:14:21 WARNING[14400]: app_dial.c:1070 dial_exec: Had to drop call because I couldn't make Modem[i4l]/ttyI0 compatible with SIP/10.10.10.1:5060-5000 |
| 23:19.51 | phpboy | I gathered as much |
| 23:20.03 | phpboy | but where in my sip.conf would I effect this extention? |
| 23:20.28 | lancey | phpboy do you have g.729 licenses? |
| 23:20.52 | lancey | it seems like you are trying to do g.729-to-pcm call |
| 23:20.57 | lancey | withouth having a codec |
| 23:21.03 | lancey | *without |
| 23:21.18 | phpboy | well it did work... I changed something |
| 23:21.34 | lancey | is this the same SIP device you talking about some lines up? |
| 23:21.37 | phpboy | and now all of a sudden it doesn't work.. but I can't think for the life of me what I changed :/ |
| 23:21.49 | phpboy | yep... this is the server side though |
| 23:22.01 | lancey | well, it supported alaw only, right? |
| 23:22.04 | lancey | *ulaw |
| 23:22.33 | phpboy | quick picture... I call a number in my country... it then calls a number in the states which comes back to my server then calls through my ISDN line back to my mobile phone |
| 23:22.38 | phpboy | it was working a couple of mins ago :< |
| 23:22.49 | lancey | you have codec issues |
| 23:22.53 | lancey | look again in sip.conf |
| 23:23.02 | lancey | for the codecs configured for 10.10.010.1 |
| 23:23.03 | riksta | Hey, could someone please help, I seem to have a major problem with regards to the logging of the duration of some of my calls from ZAP channels, it seems like someof the calls are not being detected that it was hung up properly, and the telco's logs detect a much smaller duration of the particular call. this happens about 5% of the time. I have asterisk 1.09 and zaptel 1.09 and sangoma wanpipe beta11-2.3.3 |
| 23:23.20 | phpboy | I realise this |
| 23:23.25 | Carp1 | Does anyone have the Efficient Networks sb510FXS Voice Gateway? |
| 23:23.34 | lancey | i also can't work out how a dial cmd with an IAX2 parameter could call a SIP channel?! |
| 23:23.38 | phpboy | but where would the codec for that specific extention lie? |
| 23:23.53 | phpboy | well it 'worked' at a stage :/ |
| 23:24.11 | lancey | what's the contents of $IAXINFO? |
| 23:24.49 | phpboy | IAXINFO=guest |
| 23:24.58 | lancey | со |
| 23:25.00 | *** join/#asterisk darkskiez (n=darkskie@host86-138-169-183.range86-138.btcentralplus.com) |
| 23:25.06 | lancey | so exten => _1234567,1,Dial(IAX2/${IAXINFO}@10.10.10.1/${EXTEN:1}@testphone) |
| 23:25.13 | lancey | calls a SIP device?! |
| 23:25.17 | lancey | no way |
| 23:25.20 | phpboy | affirmative |
| 23:25.24 | phpboy | I swear on my life |
| 23:25.40 | *** join/#asterisk gniretar (n=gniretar@66-227-204-109.dhcp.bycy.mi.charter.com) |
| 23:25.44 | phpboy | took it from an example in the sample files |
| 23:25.45 | gniretar | hi all |
| 23:25.50 | gniretar | anyone familiar with Vonage? |
| 23:26.11 | phpboy | I just need to figure out how to change the codec that string uses |
| 23:26.11 | phpboy | u can help? |
| 23:26.47 | lancey | phpboy codec can't be changed per extension |
| 23:26.52 | lancey | or i'm not aware of that |
| 23:26.57 | gniretar | as in how i could use Asterisk to connect to it directly opposed using a analogue card. |
| 23:27.01 | gniretar | i'm assuming it is SIP |
| 23:27.18 | lancey | the right way is to define the codecs in sip.conf/iax.conf/whatever |
| 23:27.24 | lancey | and let asterisk choose the best one |
| 23:27.29 | phpboy | lancey: I changed it in [general] |
| 23:27.32 | phpboy | allow=all |
| 23:27.35 | phpboy | and now it's working |
| 23:27.44 | phpboy | just need to fine tune it from here |
| 23:27.47 | lancey | so what's that dial command doing? |
| 23:27.56 | lancey | can you paste the log |
| 23:28.00 | lancey | when you execute it? |
| 23:29.09 | phpboy | I can... let me just edit out the particulars "Confidential" |
| 23:29.15 | phpboy | u know how it goes |
| 23:29.19 | phpboy | gimme a sec |
| 23:29.24 | *** join/#asterisk _DAW (n=bob@adsl-150-43-153.msy.bellsouth.net) |
| 23:29.31 | lancey | yup |
| 23:30.09 | *** join/#asterisk acidfoo (n=acidfoo@66.11.160.156) |
| 23:30.39 | rayvd | If I use restrictcid=yes in my sip.conf, and make an outbound call through another SIP gateway... how does the other SIP gateway know who is calling? |
| 23:30.39 | phpboy | -- Executing Dial("Modem[i4l]/ttyI0", "SIP/1234567@10.10.10.1:5060") in new stack |
| 23:30.44 | rayvd | Is their a concept of ANI in a SIP header? |
| 23:30.47 | rayvd | maybe RPID? |
| 23:31.12 | lancey | phpboy you sure this is the same line you posted?! |
| 23:31.17 | lancey | exten => _1234567,1,Dial(IAX2/${IAXINFO}@10.10.10.1/${EXTEN:1}@testphone) ? |
| 23:31.22 | phpboy | affirmative |
| 23:31.25 | lancey | goes into this: |
| 23:31.25 | lancey | -- Executing Dial("Modem[i4l]/ttyI0", "SIP/1234567@10.10.10.1:5060") in new stack |
| 23:31.31 | lancey | it couldn't be |
| 23:31.43 | phpboy | d00d, I swear |
| 23:31.50 | phpboy | that's what it executes |
| 23:31.56 | distortion | is there a way to change rfc2833 payload in asterisk? |
| 23:32.18 | lancey | i can't explain that to myself.... |
| 23:32.47 | phpboy | I'd be amazed if I actually knew what I was doing |
| 23:32.57 | Knight_DKN | Sorry ppl, got distracted |
| 23:33.02 | phpboy | but most of my asterisk configs is hacked up from the sample files |
| 23:33.16 | lancey | phpboy there must be another dial line in your extensions.conf |
| 23:33.18 | phpboy | I'm amazed that I've gotten half the shit right that I've gotten right |
| 23:33.19 | lancey | which is doing that |
| 23:33.31 | phpboy | lancey: nope... it's that line |
| 23:33.36 | phpboy | allow me to confirm |
| 23:33.36 | lancey | it couldn't be a dial command with an IAX2 channel calls a SIP device |
| 23:33.37 | lancey | no way |
| 23:33.44 | lancey | :) |
| 23:34.12 | lancey | phpboy if you are right, we should file a major bug :) |
| 23:35.04 | phpboy | lancey: I've got good news... ur not losing your mind.. I'm just too lame to see it... I found the line that's prolly doing it |
| 23:35.08 | phpboy | exten => s,1,Dial(SIP/1234567@10.10.10.1:5060) |
| 23:35.12 | phpboy | does that look right |
| 23:35.13 | phpboy | ? |
| 23:35.19 | phpboy | comes in on ISDN then dials SIP |
| 23:37.13 | *** join/#asterisk pooh_ (n=hfwang@cust.15.241.adsl.cistron.nl) |
| 23:37.25 | pooh_ | Hi, anybody listening into astricon ? |
| 23:38.17 | pooh_ | if so, tell them to boost the conference microphone please |
| 23:39.15 | InfraRed | tell them yourslef |
| 23:39.22 | InfraRed | yell louder! \o/ |
| 23:39.31 | pooh_ | Channel is muted for listeners ;-) |
| 23:39.38 | lancey | phpboy |
| 23:39.40 | lancey | thank you |
| 23:39.40 | lancey | :) |
| 23:39.58 | lancey | just thought i'm going nuts :) |
| 23:40.15 | pooh_ | Is this Olee speaking atm? the guy with the accent ? |
| 23:40.20 | pooh_ | Olle(sorry) |
| 23:40.21 | phpboy | I need to learn asterisk a little better |
| 23:40.24 | lancey | well, you can replace @10.10.10.1:5060 |
| 23:40.26 | gniretar | . |
| 23:40.28 | lancey | with @sip-peer-name |
| 23:40.39 | *** join/#asterisk christo (n=chris@brezhnev.spiration.co.uk) |
| 23:40.43 | lancey | and make [sip-peer-name] section in your sip.conf |
| 23:40.44 | christo | evening all |
| 23:40.55 | phpboy | ah... now I understand how u make peers or 'why' rather |
| 23:40.56 | lancey | there you can define what codecs to use |
| 23:41.22 | phpboy | hmmmm |
| 23:41.23 | lancey | you can define peer more than once |
| 23:41.23 | lancey | e.g. with different codec preferences |
| 23:41.23 | lancey | another thing i'd like to point out |
| 23:41.32 | lancey | if it`s IP really is 10.10.10.1 |
| 23:41.38 | lancey | it's prolly located on the LAN |
| 23:41.42 | phpboy | (which it's not) |
| 23:41.52 | phpboy | yes, had to change it to something obvious |
| 23:41.53 | phpboy | :P |
| 23:42.02 | lancey | well, if it's on the LAN |
| 23:42.10 | lancey | you have no good reason not to use alaw/ulaw |
| 23:42.25 | phpboy | I understand not the concept behind ulaw/alaw |
| 23:42.26 | phpboy | :/ |
| 23:42.37 | lancey | alaw/ulaw is uncompressed sound |
| 23:42.43 | lancey | => best quality |
| 23:42.53 | lancey | but takes 64 Kbits bandwidth |
| 23:43.18 | christo | is there any such thing as an open source speech to text engine? I've googled and sf.net'd, but can't really see anything mature out there |
| 23:43.24 | lancey | it also doesn't consume any CPU time to encode/decode |
| 23:43.25 | distortion | is rfc2833 payload configurable? |
| 23:43.33 | lancey | => better latency, lower CPU usage |
| 23:43.50 | phpboy | lancey: what's a good 'free' asterisk standard codec to use? |
| 23:43.50 | lancey | christo: i don't think so |
| 23:43.57 | InfraRed | lancey: not counting overheads |
| 23:44.06 | lancey | phpboy everything but g.729 and g.723 is free |
| 23:44.08 | InfraRed | lancey: it's eating 85Kbps here |
| 23:44.15 | InfraRed | (ADSL) |
| 23:44.18 | lancey | InfraRed yes |
| 23:44.26 | lancey | each codec has overheads |
| 23:44.45 | InfraRed | ya you have the media overhead |
| 23:44.46 | christo | r |
| 23:44.55 | lancey | my point was to explain that if it is a high-bandwidth link, there's no point in not using alaw/ulaw |
| 23:45.10 | lancey | overhead also depends on trunking |
| 23:45.39 | InfraRed | r:) |
| 23:46.14 | lancey | phpboy if you are trying just to divert a call from the modem to the SIP server on the lan |
| 23:46.18 | lancey | alaw/ulaw is just fine |
| 23:46.48 | lancey | also, your sip peer doesn't appear to support anything else, so you don't have a choice anyways ;) |
| 23:47.03 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 23:47.19 | *** join/#asterisk zamsler (n=zamsler@c-67-184-232-149.hsd1.il.comcast.net) |
| 23:47.43 | Ariel_ | hello everyone |
| 23:48.09 | zamsler | hey Ariel_ |
| 23:48.53 | lancey | yo Ariel_ |
| 23:50.31 | *** join/#asterisk Dr_Ray (i=drray@dsl254-011-243.sea1.dsl.speakeasy.net) |
| 23:50.59 | Dr_Ray | has there been any talk of getting asterisk to deliver iptv? or am I barking up the wrong tree? |
| 23:53.31 | lancey | Dr_Ray :) |
| 23:54.25 | pauldy | Dr_Ray, first I've hear of it but I'm all ears now |
| 23:54.39 | Dr_Ray | well, I was thinking |
| 23:54.46 | lancey | :) |
| 23:54.59 | Dr_Ray | there is nothing stopping asterisk from doing it |
| 23:56.01 | Dr_Ray | would you cobble up your own client, or try to expand IAX/SIp to do it? |
| 23:57.26 | pauldy | there is one message on the digium list |
| 23:57.38 | pauldy | but it doesn't mention doing it with asterisk |
| 23:58.13 | pauldy | hell I can't even get the eybeam clients to do their video with an asterisk server in the middle |
| 23:58.43 | Dr_Ray | the h.264 client |
| 23:59.43 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |