| 00:00.39 | tekati | Anyone have any idea why I have to modprobe zaptel, wcfxo, wcfxs and then do a ztcfg before starting asterisk everytime? I am running Fedora Core 4 and trying to come up with a startup method when the server reboots. |
| 00:05.41 | [TK]D-Fender | rking : Still around? |
| 00:07.29 | LostFrog | tekati: put it in rc.local |
| 00:09.03 | mog_work | if you have hw you need to load it |
| 00:09.58 | *** join/#asterisk kippi (n=chris@untrust-gct.equinoxit.net) |
| 00:10.36 | postel | tekati: drop the modules to /etc/modules and they should load upon reboot, or drop an rc script to rc.d, or rc.local as suggested |
| 00:11.10 | *** join/#asterisk zuuluu (i=klineder@adsl-067-035-113-166.sip.bct.bellsouth.net) |
| 00:11.34 | tekati | I will give that a try thank you. |
| 00:11.35 | zuuluu | is there an option to set "top down" or "bottom up" routing for a span? |
| 00:11.47 | zuuluu | or does asterisk only use one? |
| 00:13.16 | *** join/#asterisk bzbw (n=wlwzhang@ip67-153-142-80.z142-153-67.customer.algx.net) |
| 00:13.29 | *** join/#asterisk jpablo (n=jpablo@dsl-201-133-205-162.prod-infinitum.com.mx) |
| 00:13.35 | anthm | g and G |
| 00:13.39 | bzbw | anyone knows that 1.2 broke the sip calls with BroadVoice? |
| 00:13.52 | LostFrog | bzbw: g729?? |
| 00:14.14 | bzbw | I did not check, is it the case? |
| 00:14.16 | anthm | allweb w |
| 00:14.41 | *** join/#asterisk tekati (n=captain@cpe-66-75-215-63.bak.res.rr.com) |
| 00:14.43 | LostFrog | I am running RC2 with no problems. |
| 00:15.17 | bzbw | I used the official from asterisk.org |
| 00:15.58 | kippi | has anyone used asterisk to be able to send txts from a handset to a moblie phone using asterisk |
| 00:16.02 | jpablo | hey, any recomended voip provider for a medium corp ? (500 or so employes) only for outgoing calls ... |
| 00:16.57 | bzbw | Yes, I just checked RC2 does not have problem, but the official release 1.2 does have problem. |
| 00:17.27 | bzbw | and it does not appear to be codec related. |
| 00:17.38 | *** join/#asterisk tekati (n=captain@cpe-66-75-215-63.bak.res.rr.com) |
| 00:17.41 | LostFrog | Hmm.. I am not in a position to test it. |
| 00:17.47 | bzbw | Broadvoice server respond a call leg does not exists |
| 00:18.12 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
| 00:19.11 | [hC] | Hrm.. what the heck.. calling a number normally works fine, when it calls a sip phone, then forwards to the same number, one way audio. |
| 00:19.12 | Flauto | how enum works |
| 00:19.21 | Flauto | i could never make to work for me |
| 00:19.28 | TheCops | Someone have example to make DevState working with Dial and dialing status ? |
| 00:19.28 | [hC] | only via pri it seems. |
| 00:19.35 | Flauto | not even 800 number |
| 00:19.36 | Flauto | s |
| 00:20.35 | [TK]D-Fender | rking : http://pastebin.com/444323 |
| 00:20.42 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-61-219.cybersurf.com) |
| 00:20.46 | *** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com) |
| 00:20.58 | Flauto | Nov 30 18:20:45 NOTICE[17605]: rtp.c:330 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 216.234.96.51 |
| 00:21.02 | Flauto | what is this for |
| 00:21.10 | SplasPood | hrm, is there any way to force a sipura to tftp the config from the provider even if it's locked? It doesn't seem to on boot |
| 00:21.13 | [TK]D-Fender | rking : There's a version that takes 3 or 6 seconds depending on wether the last call went through. |
| 00:21.51 | bzbw | anyone knows if I roll back from 1.2 to 1.2-RC2? |
| 00:22.09 | bzbw | I failed after rebuild the server with RC2 |
| 00:22.12 | jpablo | bzbw: try the 1.2 HEAD |
| 00:22.20 | jpablo | sorry |
| 00:22.23 | mog_work | trunk |
| 00:22.23 | jpablo | the 1.2 svn |
| 00:22.28 | mog_work | we use svn now... |
| 00:22.56 | bzbw | jpablo: sorry, what is svn? |
| 00:23.10 | jpablo | bzbw: subversion, same as cvs only better |
| 00:23.14 | jpablo | check asterisk.org |
| 00:23.28 | jpablo | trunk, im not familiar with svn terminology |
| 00:23.35 | bzbw | jpablo: but isn't it the same as the official ones from HTTP |
| 00:23.45 | *** join/#asterisk znoG (n=gs@OL101-122.fibertel.com.ar) |
| 00:23.50 | jpablo | bzbw: no, it may contain updates |
| 00:24.08 | jpablo | bzbw: svn 1.2 is what eventually will become 1.2.1 |
| 00:25.49 | LostFrog | Hmm.. can you set a prefix for asterisk when you install it? |
| 00:25.57 | jorgito | i have one answer, working for one middle sized isp in central europe, we have asterisk based voip pbx which is used for Helpdesk, the problem is when (there are only two phones) all lines are ocuppied the customer trying to call helpdesk fall to voice mail, which is annoiyng for them. Is there some possibility to forward calls only when this two phones are occupied to other phone numbers ? thanks really for reply . jorge |
| 00:26.48 | jpablo | jorgito: yes. |
| 00:26.58 | Flauto | is enumlookup working at all? |
| 00:27.07 | jorgito | jpablo: really ? how? |
| 00:27.43 | jpablo | jorgito: in your dialplan you have something like: exten => bla,1,Dial(SIP/XX&SIP/YY), just add: exten => bla,2,Dial(SIP/ZZ) |
| 00:27.45 | LostFrog | To answer my own question, yes.. there is an INSTALL_PREFIX in the Makefile |
| 00:27.53 | jpablo | where ZZ is the new extension that will ring |
| 00:28.01 | bzbw | jpablo: are you aware a particular issue with Broadvoice issue in 1.2? |
| 00:28.10 | jpablo | jorgito: actuallly im not use if the priority is 2 or 102 |
| 00:28.16 | jpablo | bzbw: no really. |
| 00:28.37 | jpablo | jorgito: one is for timeout, the other for busy/congested |
| 00:28.38 | bzbw | jpablo: it looks like 1.2 had some change in the SIP stack compare with RC2 |
| 00:28.59 | jorgito | jpablo: so the bla,2 is called only when bla,1 is full? |
| 00:29.04 | jpablo | bzbw: maybe you want to diff them and see where the bug was introduced. |
| 00:29.43 | bzbw | jpablo: emm, I'm not really formiliar with * code, not sure which is the stack to compare |
| 00:30.50 | jpablo | jorgito: yes, unless you are doing some weird stuff with 1,Dial, by default it terminates the call when the phone is hangup, so it won't continue to bla,2,dial, just if the first dial fails. |
| 00:31.47 | p1tst0p | erm, you know Voicemail can send a mail, when someone receives mail.. is it possible to use a mailserver on another server, to send the mail ? |
| 00:31.54 | jorgito | jpablo: yes i understand thanks a lot |
| 00:32.42 | *** join/#asterisk SuperMoco (i=SuperM0C@dsl-201-129-125-139.prod-infinitum.com.mx) |
| 00:33.12 | jpablo | p1tst0p: it uses /usr/sbin/sendmail, there are a some /usr/sbin/sendmail remplacents that can relay to other server i remember smail |
| 00:33.24 | [hC] | Anyone experiencing any problems doing channel bridging in recent cvs versions? For some reason when i have a call come in from another * server via IAX, it tries a sip extension, then we pass it back out to the other * server to dial a PSTN number (unavail forwarding) I get one way audio issues. doesnt happen if i dial the number directly without calling the sip phone first. |
| 00:33.35 | Flauto | Nov 30 18:32:32 NOTICE[18304]: rtp.c:330 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 216.234.96.51 |
| 00:33.37 | jpablo | p1tst0p: ssmtp |
| 00:33.39 | Flauto | whatis it |
| 00:34.20 | *** part/#asterisk SuperMoco (i=SuperM0C@dsl-201-129-125-139.prod-infinitum.com.mx) |
| 00:34.33 | *** join/#asterisk RickTick (n=rpulido_@71.196.2.172) |
| 00:35.19 | RickTick | anyone have experience setting up a 4 port FXO gw (Eusso)? |
| 00:36.05 | *** join/#asterisk Gennaro (n=Gennaro@ppp-62-10-136-182.dialup.tiscali.it) |
| 00:36.08 | Gennaro | hi |
| 00:36.15 | jpablo | Flauto: dunno, you tried google ? you need that 'comfort noise'? if not just turn it off. |
| 00:36.19 | Gennaro | i installed an 537ep modem |
| 00:36.20 | *** join/#asterisk Netgeeks (n=Chris@68-185-24-2.static.mdfd.or.charter.com) |
| 00:36.40 | Gennaro | mod. file zaptel/wcfxo.c |
| 00:36.42 | Gennaro | for id |
| 00:36.45 | Gennaro | vendor |
| 00:36.54 | RoyK | Flauto: turn off silence suppression |
| 00:36.55 | Gennaro | as 8086 x101p |
| 00:36.55 | Flauto | what should i search, and where to turn it off |
| 00:37.11 | Flauto | royk, where to do that |
| 00:37.17 | Gennaro | make clean ; make install from source |
| 00:37.17 | RoyK | on the client |
| 00:37.21 | Gennaro | now if i run |
| 00:37.21 | zoa | royk |
| 00:37.24 | RoyK | zoa |
| 00:37.25 | zoa | did you try the patch yet ? |
| 00:37.25 | Gennaro | ztcfg |
| 00:37.26 | jpablo | Flauto: in your client (ata, phone, etc) |
| 00:37.30 | RoyK | zoa yes |
| 00:37.30 | Gennaro | -vvg |
| 00:37.32 | Gennaro | i see |
| 00:37.33 | Flauto | oh |
| 00:37.33 | RoyK | so far looking good |
| 00:37.34 | Flauto | okay |
| 00:37.34 | zoa | what happened ? |
| 00:37.38 | zoa | quality is better ? |
| 00:37.40 | Flauto | spa3000 |
| 00:37.41 | Flauto | okay |
| 00:37.44 | Flauto | let me see |
| 00:37.57 | RoyK | zoa: afaics yes, but we need to do larger scale testing tomorrow |
| 00:37.59 | Flauto | and that thing shows up only when i use enumlookup |
| 00:38.07 | zoa | let me know how things go |
| 00:38.08 | RoyK | and do testing on bad bandwidth |
| 00:38.11 | Flauto | the enum has never worked for me |
| 00:38.13 | Gennaro | Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 00:38.16 | RoyK | zoa: of course :) |
| 00:38.17 | zoa | and give me rtp dumps if things go bad |
| 00:38.17 | bzbw | jpablo: where can I download svn program so that I can "svn" checkout? Thx for such a simple question |
| 00:38.20 | zoa | im off to bed now |
| 00:38.20 | Flauto | thanks |
| 00:38.21 | zoa | cheers |
| 00:38.23 | Flauto | guys |
| 00:38.28 | Flauto | i will look into it |
| 00:38.45 | jpablo | bzbw: should be included in your linux distribution, it is probably called subservion |
| 00:39.00 | jpablo | bzbw: if not, check in http://subversion.tigris.org/ |
| 00:39.05 | db48x | I think you mean subversion :) |
| 00:39.18 | Gennaro | emove R13 and R19 with a soldering iron. R13 & R19 are pull down resistors that affects the vendorID number that is read from the card ? |
| 00:39.25 | jpablo | yes, that :P |
| 00:40.29 | wundaboy | ok, im just learning asterisk |
| 00:40.32 | wundaboy | and in iax.conf |
| 00:40.39 | wundaboy | type=user |
| 00:40.42 | wundaboy | what does that mean? |
| 00:40.43 | bzbw | jpablo: thx! |
| 00:41.11 | X-Rob | It means you're an addict. |
| 00:41.25 | jpablo | wundaboy: voip-info.org, search there ... |
| 00:41.27 | Flauto | guys, the silence supp enable is already no |
| 00:41.33 | wundaboy | thats where im reading |
| 00:41.37 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 00:41.42 | wundaboy | and the iax.conf page isnt very helpfull |
| 00:41.50 | wundaboy | or im dumb, either one |
| 00:42.07 | Flauto | the funny thing is that saying the client ip is not my ip at all |
| 00:42.47 | LostFrog | bzbw: I am running 1.2 now with broadvoice. |
| 00:43.00 | *** join/#asterisk alrs (n=lars@69-160-242-101.vnnyca.adelphia.net) |
| 00:43.26 | jpablo | wundaboy: type can be user, peer or friend, user can recive calls, peer just dial calls and friend both, that or the same just reversed |
| 00:43.28 | Flauto | lost, i am using broadvoice too |
| 00:43.58 | wundaboy | jpablo thanks! |
| 00:44.17 | db48x | jpablo, wundaboy: that's not quite correct, as I understand |
| 00:44.38 | db48x | a user can send and recieve calls, but can only send if they're from himself, and can only recieve if they're to him |
| 00:44.51 | Flauto | is there anyone of you using enum? |
| 00:44.57 | jpablo | crap :| |
| 00:45.10 | Flauto | jpablo, enum is crap? |
| 00:45.23 | jpablo | db48x: are you sure? i think my explanation works for sip.conf not sure iax |
| 00:45.32 | jpablo | Flauto: no, sorry, i don't know what enum is |
| 00:45.44 | db48x | jpablo: should be the same thing, they're both defining channels |
| 00:46.07 | Flauto | not problem, jpablo |
| 00:46.19 | jpablo | http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+type |
| 00:47.33 | Flauto | please reply me if anyone is using enum |
| 00:47.51 | RickTick | can anyone ...guide me to info ...on how to setup h323 on asterisk? |
| 00:47.59 | Flauto | i amgetting very frustrated with enum |
| 00:48.19 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 00:48.35 | jpablo | RickTick: i tried it once, got frustrad, forgot about h323 forever. |
| 00:49.13 | db48x | jpablo: oh, maybe you are correct |
| 00:50.13 | RickTick | jpablo: ...really ... is it that bad to setup? |
| 00:50.23 | *** join/#asterisk Cyberknet (n=CyberKne@ip68-13-246-61.ok.ok.cox.net) |
| 00:50.45 | jpablo | RickTick: when i tried it was that bad to compile !! |
| 00:51.05 | RickTick | oh ... wow |
| 00:51.29 | jpablo | RickTick: i was trying to connect to a h323 gateway, that was asking a h323 id or something, never find where to configure that. |
| 00:51.35 | alephcom | I got H323 working by installing the asterisk at home version |
| 00:52.13 | RickTick | alephcon: any special problems? |
| 00:52.35 | jpablo | i think debian now also has precompiled packages of chan-h323 and a chan oh323 |
| 00:52.48 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@64.241.37.140) |
| 00:53.02 | alephcom | I didn't but I avoid H323 whenever possible. |
| 00:53.41 | p1tst0p | jpablo, will asterisk use ssmtp by default, or do i need to tell it to ? |
| 00:54.43 | jpablo | p1tst0p: it will use it by default, asuming it overried /usr/sbin/sendmail |
| 00:56.11 | wundaboy | queues.conf is just for callcenter type applications right? |
| 00:59.04 | *** join/#asterisk figits (n=me@217.240.33.65.cfl.res.rr.com) |
| 00:59.08 | jpablo | wundaboy: yes, just for queues |
| 01:02.02 | marcus2 | woohoo |
| 01:02.06 | marcus2 | my channel bank arrived! |
| 01:02.57 | Gennaro | soory |
| 01:03.01 | Gennaro | sorry.. |
| 01:03.15 | Gennaro | i tried to change id vendor |
| 01:03.27 | Gennaro | on .c file in zapata |
| 01:03.37 | Gennaro | as in voip info.. |
| 01:03.54 | Gennaro | so i have associated 8086 as x101p |
| 01:03.57 | Gennaro | to so.. |
| 01:04.08 | Gennaro | when i compile asterisk |
| 01:04.13 | Gennaro | when i compile zapata |
| 01:04.27 | Gennaro | zapata.conf is created.. |
| 01:04.45 | Gennaro | in this file i havent more generic card |
| 01:04.52 | AgiNamu | can i hangup without answering a PRI? |
| 01:05.02 | Gennaro | i havent any device |
| 01:05.23 | Gennaro | if i ztcfg -vv |
| 01:05.42 | Gennaro | 1 channel configured |
| 01:05.54 | Gennaro | no devide or address... |
| 01:06.11 | AgiNamu | you domt compile zapata, its zaptel |
| 01:06.20 | AgiNamu | and you edit /etc/zaptel.conf first to make sure its setu correctly |
| 01:06.21 | Gennaro | should i remove componets to let system see as digium? |
| 01:06.26 | AgiNamu | THEN ztcfg -vv might work |
| 01:08.13 | Gennaro | all is # and only 3 raw enable are fxsks=1;loadzone = us;defaultzone=us |
| 01:08.23 | TheCops | marcus2, a rhino ? |
| 01:08.54 | jpablo | what exactly is dundi useful for ? |
| 01:09.31 | Gennaro | modprobe wcfxo ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 01:11.22 | [TK]D-Fender | AgiNamu : you mean can you simply not answer a call coming in on a PRI? |
| 01:11.33 | *** join/#asterisk brc_ (n=Brian@pdpc/supporter/basic/brc) |
| 01:12.00 | Gennaro | do u belive that i remove components from modem for idvendor and reinstall all i'll be victorios? |
| 01:12.33 | RoyK | hm. is sip transfer supposed to work? |
| 01:12.47 | jpablo | RoyK: yes |
| 01:13.17 | RoyK | jpablo: so exten => s,n,Transfer(SIP/${ARG1}@${SIPGATE}) |
| 01:13.21 | AgiNamu | [TK]D-Fender, no like can i get a call, do something, then disconnect it without answering |
| 01:13.28 | RoyK | should transfer the caller with a 302 to SIPGATE? |
| 01:13.30 | AgiNamu | do something not involving that channel |
| 01:13.36 | *** join/#asterisk Pete_Largo (n=PeteLarg@225-196.35-65.tampabay.res.rr.com) |
| 01:13.54 | [TK]D-Fender | AgiNamu : sure.... |
| 01:14.05 | AgiNamu | do you know i can sert the disconnect cause code? |
| 01:14.12 | [TK]D-Fender | That... no idea |
| 01:14.34 | Pete_Largo | I know I should ask this in #linux - in fact I did, and didn't get an answer... how can I make a file execute without having to type ./ before it? |
| 01:14.58 | jpablo | Pete_Largo: type the full path? |
| 01:15.00 | [TK]D-Fender | Pete_Largo : sure, just make sure its in the path. |
| 01:15.16 | Pete_Largo | ? |
| 01:15.17 | [TK]D-Fender | or ensure there is a path statement for the current folder |
| 01:15.33 | jpablo | [TK]D-Fender: a security issue .. |
| 01:15.57 | jpablo | (adding . to the PATH) |
| 01:15.57 | [TK]D-Fender | jpablo : I never suggested the latter... just that it WORKS ;) |
| 01:16.03 | Pete_Largo | you mean instead of cd /usr/local/bin \ text2wav I need to type /usr/local/bin/text2wave? |
| 01:16.26 | jpablo | Pete_Largo: yes. |
| 01:16.35 | Pete_Largo | well hell, that didn't work |
| 01:16.36 | [TK]D-Fender | you could, but I mean that you should set your path env var to include that path |
| 01:16.42 | jpablo | Pete_Largo: that or doing PATH=$PATH:/usr/local/bin |
| 01:17.06 | jpablo | in your .$(SHELL){rc,)_profile) or something |
| 01:17.23 | Pete_Largo | k |
| 01:17.28 | Pete_Largo | thanks |
| 01:19.16 | *** join/#asterisk iam8up|lappy (n=User@rrcs-24-172-153-135.central.biz.rr.com) |
| 01:20.41 | Gennaro | tank u |
| 01:20.45 | Gennaro | good nite |
| 01:20.58 | iam8up|lappy | has anyone here ever used any aastra (480i ip) phones? if so..i'm trying to have the phone reach the tftp server and read it's own mac.cfg - it reads the aastra.cfg from the tftp server, but i've got to set two aastra 480i phones side by side on the same network, requiring the individual cfg files for each phone |
| 01:21.09 | Pete_Largo | ok, that fixed part of my problem...let me explain, I'm trying to get festival working with the US voices using mbrola... now I am getting the following error: : SIOD ERROR: could not open file /usr/local/festival/festival/lib/voices/english/en1_mbrola/festvox/en1_mbrola.scm |
| 01:21.11 | RoyK | hrmf. this happens when i try to SIP transfer: http://karlsbakk.net/asterisk/wtf1.png |
| 01:22.03 | iam8up|lappy | my question simply put: why aren't my mac.cfg files working? |
| 01:23.37 | jpablo | robin_sz: what's the app that generated that image? |
| 01:23.53 | iam8up|lappy | don't you mean royk? |
| 01:24.11 | RoyK | jpablo: ethereal |
| 01:24.16 | jpablo | yeah, RoyK, stupid xchat completion |
| 01:24.22 | iam8up|lappy | lol =P |
| 01:25.48 | *** join/#asterisk juice (n=juice@mo-69-69-117-72.sta.sprint-hsd.net) |
| 01:28.20 | rking | [TK]D-Fender: thanks! (i was afk, just now saw your pastes) |
| 01:29.11 | rking | this looks like it uses a lot of handy stuff - i'm going to take the time to really understand these |
| 01:33.05 | [TK]D-Fender | np |
| 01:33.36 | [TK]D-Fender | just does a basic DB lookup and if it answered the last call it'l jump over the 1st wait and take 1/2 the time |
| 01:33.57 | [TK]D-Fender | adjust to taste :) it also sets up the DBentry on the first run so you don't need to do it yourself |
| 01:34.24 | rking | so i can do a DBGet()/DBPut() on AnyThingHere/anythingElseHere ? |
| 01:35.26 | rking | as in, it'll autovivify both database's and key/value pairs? |
| 01:38.37 | [TK]D-Fender | pretty much. *'s built in DB1 is as low grade and dumb as can be, but it getts the job done for simepl stuff |
| 01:38.45 | [TK]D-Fender | Flag-like entries |
| 01:38.48 | [TK]D-Fender | and so on. |
| 01:39.24 | [TK]D-Fender | Although i'd really like to see them integrate SQLite instead of some of the other options. that way it'd be native, fast, and scale decently. |
| 01:39.47 | rking | SQLite is handy |
| 01:39.53 | [TK]D-Fender | Without having to be too swift :) |
| 01:39.56 | [TK]D-Fender | it is... |
| 01:40.29 | [TK]D-Fender | I wonder how compatible its licensing is... |
| 01:40.46 | [TK]D-Fender | not sure if its GPL, LGPL, or BSD.... |
| 01:41.05 | rking | public domain, baby! |
| 01:41.41 | [TK]D-Fender | very nice... |
| 01:41.59 | [TK]D-Fender | Should definately replace DB1 in * then... can't see a single reason why not... |
| 01:42.49 | [TK]D-Fender | well... except that DB! does let you create key-pairs "willy-nilly" and is nearly idiot-proof :) |
| 01:43.56 | theblue | ? |
| 01:44.21 | [hC] | Have any of you guys experienced strange one way audio problems in recent cvs head? mainly relating to bridged pri channels? |
| 01:44.33 | *** join/#asterisk future (n=future@virtual.phuture.sk) |
| 01:48.45 | *** join/#asterisk santiago (n=santiago@208.195.215.160) |
| 01:51.07 | tessier_ | one way audio pretty much describes voip in general |
| 01:52.06 | *** join/#asterisk genmud (n=genmud@ip68-98-82-206.ph.ph.cox.net) |
| 01:54.53 | [hC] | :P |
| 01:58.13 | *** join/#asterisk supaigtr (n=yurplsl@152.53.17.26) |
| 01:59.48 | TheCops | How can I do a if operation in my dialplan ? |
| 02:00.54 | distortion | has anyone been able to do passthrough t.38 with asterisk? |
| 02:01.04 | [TK]D-Fender | exten = s,2,GotoIf($[${myvariable}=SOMETHING]?4) |
| 02:01.10 | TheCops | thanks |
| 02:01.10 | TheCops | :) |
| 02:01.10 | [TK]D-Fender | like that |
| 02:01.24 | [TK]D-Fender | it'll just to priority 4 |
| 02:01.28 | [TK]D-Fender | np :) |
| 02:03.51 | GeneG | HI all. I'm trying to understand the order of events in a SIP reinvite. When Asterisk wants to connect two parties through a reinvite, does it send a SIP INVITE message to both parties simultaneously, or only 1 of the 2? How does it decide which? |
| 02:05.12 | *** join/#asterisk Newbie___ (i=me@60.48.54.195) |
| 02:06.01 | Newbie___ | hi all, is there a way to have * authentic by callerid before accepting a call? |
| 02:06.40 | marcus2 | hm |
| 02:06.52 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
| 02:07.07 | konfuzed | hey there, I dont have a mac but someone I know does and wants to try a voip softphone. I have no way to test what is actually workable and easy for a song writer musician type to use |
| 02:07.14 | konfuzed | any one use mac softphones? |
| 02:07.14 | marcus2 | now i need to find an adapter to convert a centronics 50 to 24 rj11 jacks |
| 02:07.26 | Nugget | x-lite is the best mac softphone. |
| 02:07.28 | marcus2 | yeah, we started using loudhush today |
| 02:07.40 | marcus2 | its not that great |
| 02:08.10 | konfuzed | marcus2: i just want to be sure you refer to mac softphone |
| 02:08.26 | marcus2 | what do you mean? |
| 02:08.32 | GeneG | Newbie, there is an example of routing by caller ID in the Asterisk Handbook, p.31. Is this what you're looking for? I can find an online link the handbook if you'd like. |
| 02:08.35 | konfuzed | loudhush |
| 02:08.35 | [TK]D-Fender | Newbie___ : Authenticate in what way? |
| 02:08.44 | marcus2 | its a mac softphone... |
| 02:08.53 | Newbie___ | GeneG: would appreciate that |
| 02:08.58 | marcus2 | iax-only |
| 02:09.00 | konfuzed | marcus2: you could be just continuing with your previous conversation so I wanted to verify |
| 02:09.18 | konfuzed | I do like iax |
| 02:09.22 | Newbie___ | [TK]D-Fender: authentic in way that certain clid required pins |
| 02:09.26 | konfuzed | x-lite does not do iax does it |
| 02:09.31 | konfuzed | I forget but doubt it |
| 02:09.32 | marcus2 | no, it does not |
| 02:09.35 | GeneG | Newbie, am looking for a link now. Basically, the syntax is exten => 100/5551212,1,Congestion |
| 02:09.49 | GeneG | 5551212 is the caller ID in thsi case |
| 02:09.58 | konfuzed | see im sorta stucj in this world of sip availability when I want to keep everyone on iax |
| 02:10.15 | GeneG | Newbie, try http://www.digium.com/handbook-draft.pdf |
| 02:10.16 | marcus2 | iax is necessary if you want your softphones to connect to asterisk via the internet, pretty much |
| 02:10.18 | marcus2 | until they figure out a sip jitter buffer |
| 02:10.44 | konfuzed | but such things likely require softphone or ata capable of both sip and iax |
| 02:10.51 | *** join/#asterisk blitz[laptop] (n=blitz[la@CPE0006256cd738-CM00122570228c.cpe.net.cable.rogers.com) |
| 02:10.55 | blitz[laptop] | ~seen hermie |
| 02:10.59 | jbot | hermie <n=nick@24-236-167-53.dhcp.bycy.mi.charter.com> was last seen on IRC in channel #asterisk, 107d 2h 42m 19s ago, saying: '"We talk like the people on TV" should be the Midwest's official slogan.'. |
| 02:11.08 | marcus2 | ~seen zoa |
| 02:11.09 | jbot | zoa is currently on #asterisk (4h 15m 56s). Has said a total of 10 messages. Is idling for 1h 32m 48s |
| 02:11.12 | Newbie___ | GeneG: thanks |
| 02:11.20 | GeneG | HI all. I'm trying to understand the order of events in a SIP reinvite. When Asterisk wants to connect two parties through a reinvite, does it send a SIP INVITE message to both parties simultaneously, or only 1 of the 2? How does it decide which? I would also appreciate it if anyone has a good reference to SIP reinvites. |
| 02:11.23 | GeneG | Newbie: np! |
| 02:11.24 | marcus2 | zoa! where's that jitter buffer! |
| 02:12.22 | *** join/#asterisk g0mb0 (n=test@external.micom.mng.net) |
| 02:15.21 | *** join/#asterisk DrJones1 (i=mrjones1@ip68-1-236-247.dl.dl.cox.net) |
| 02:15.24 | DrJones1 | hey guys |
| 02:15.36 | DrJones1 | anyone know of a good (cheap main thing) sip phone |
| 02:15.43 | DrJones1 | i got like 10 7940s |
| 02:15.49 | DrJones1 | and i want just like 3 or 4 cheapos |
| 02:15.56 | DrJones1 | and one of my 7960s keeps breaking |
| 02:16.04 | DrJones1 | protocol application invalid |
| 02:16.09 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@64.241.37.140) |
| 02:16.58 | DrJones1 | and, does anyone know a good osCommerce consultant or zencart? |
| 02:18.04 | DrJones1 | and what command on the asterisk console displays version? |
| 02:18.18 | drumkilla | show version |
| 02:18.44 | DrJones1 | monument*CLI> show version |
| 02:18.44 | DrJones1 | Asterisk CVS HEAD built by root@monument.localdomain on a i686 running Linux on 2005-09-12 19:55:57 UTC |
| 02:19.35 | DrJones1 | no dice :/ |
| 02:21.16 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 02:24.19 | *** join/#asterisk bjohnson_ (n=bjohnson@i216-58-19-27.cybersurf.com) |
| 02:25.43 | *** join/#asterisk Frawg (n=Frawg@unaffiliated/frawg) |
| 02:25.46 | ComputerWarm | Hello all question anyone know of any sites or etc that has good examples of c api? |
| 02:26.06 | DrJones1 | I wonder where all i have to change the MAC address if I got a replacement phone |
| 02:26.17 | DrJones1 | i wonder if there is a search function that will do it automagically |
| 02:26.22 | DrJones1 | replce 'xxx' with 'yyy' |
| 02:26.24 | rking | ComputerWarm: have you looked at the apps/*.c's? |
| 02:27.32 | ComputerWarm | rking no never thought of that, but i was going to use the capi but maybe doing a addon would be better |
| 02:28.32 | rking | ComputerWarm: please don't let me steer you wrong, i have no clue about anything - i just know that there are a lot of examples of doing some interesting stuff with * in the apps/ dir... maybe the official API would do you better (?) |
| 02:30.10 | [hC] | It seems like my one way audio problem has to do with calls coming in via PRI, routing to another * pbx, then back to the first * pbx and back out the pri |
| 02:30.14 | [hC] | and i cant for the life of me debug it |
| 02:30.26 | ComputerWarm | not sure i all i know is the phpagi won`t work for what i need i am thinking C agi would be the better way of going |
| 02:30.37 | ComputerWarm | but maybe doing a C app would be the very best |
| 02:32.27 | *** join/#asterisk DrukenHME (n=druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com) |
| 02:32.35 | marcus2 | sweeeeeeet |
| 02:32.39 | marcus2 | i can telnet to my channel bank |
| 02:32.47 | marcus2 | and i was able to guess the password, ehh |
| 02:32.50 | DrukenHME | hehe |
| 02:32.56 | marcus2 | it was 'cac' |
| 02:33.41 | DrukenHME | :) |
| 02:33.52 | DrukenHME | my channelbank doesn't have a net connection... |
| 02:34.02 | marcus2 | i'm glad i found one that does |
| 02:34.20 | marcus2 | cuz i dont have the funky serial connector i'd need otherwise |
| 02:34.21 | *** join/#asterisk toddf (n=toddf@ns0.fries.net) |
| 02:34.55 | DrukenHME | mines got a shitload of dip switches.. hehe |
| 02:35.02 | marcus2 | yeah this one does too |
| 02:35.08 | DrukenHME | ballpoint pens come in great :) |
| 02:35.12 | marcus2 | a shitload on the outside and another shitload on the inside |
| 02:35.51 | distortion | so the rfc2833 bug in 1.2 that i heard about, does anyone know if it is it in mantis, i didnt find anything.. |
| 02:36.33 | marcus2 | now, i wonder if i want groundstart, loopstart, or koolstart |
| 02:36.51 | *** join/#asterisk YoMama (n=rewt@pcp02689850pcs.roylok01.mi.comcast.net) |
| 02:38.45 | DrukenHME | kewlstart, defiantly not groundstart |
| 02:39.07 | DrukenHME | loop start would work too... probably |
| 02:41.38 | *** join/#asterisk shakuhashi (n=libre@200.181.83.167) |
| 02:42.02 | DrukenHME | DrJones1: what did you need with osc ? |
| 02:44.45 | YoMama | anyone here have a PAP2? |
| 02:45.30 | blitz[laptop] | anyone happen to know hermie's email address... ? |
| 02:45.37 | marcus2 | i have a bunch |
| 02:46.45 | YoMama | marcus2: vonage ones or regular ones? |
| 02:47.00 | marcus2 | unlocked vonage ones |
| 02:47.41 | *** join/#asterisk genmud (n=genmud@ip68-98-82-206.ph.ph.cox.net) |
| 02:47.42 | marcus2 | hm, this access bank says my options are loop and ground start |
| 02:52.54 | *** join/#asterisk SplasPood (i=nobody@paravolve.net) |
| 02:58.07 | DrukenHME | YoMama: what ya need to know about a pap2? |
| 02:58.30 | YoMama | Druken: just picked one up at Best Buy...of course...it's a Vonage thing |
| 02:58.40 | marcus2 | so if i have a te410p with one port plugged into a channel bank with 24 FXS ports |
| 02:58.48 | DrukenHME | marcus2: what ya want for an unlocked?? i wouldn't mind testing one out.. heehe |
| 02:59.01 | YoMama | looks like it's pretty easy to unlock it...but they get your address info |
| 02:59.05 | marcus2 | this voip-info.org page implies that i set the signaling type to FXO in zaptel.conf and zapata.conf |
| 02:59.25 | marcus2 | druken; you know you can buy unlocked PAP2s online for $60 or less |
| 02:59.40 | YoMama | although...it doesn't say subscription required on the box |
| 02:59.47 | YoMama | so i don't see what they can do to me if i keep it |
| 03:00.02 | [hC] | anyone seen an issue where all of a sudden asterisk starts spitting out Primary D-Channel on span 1 is up |
| 03:00.04 | [hC] | over and over and over |
| 03:00.07 | [hC] | yet, nothing works |
| 03:00.16 | DrukenHME | marcus2: TECHNICALLY you can't.... but we won't go there.... but yes i've seen them |
| 03:00.27 | marcus2 | no, seriously |
| 03:00.37 | marcus2 | PAP2-NA is the product code for unlocked PAP2s |
| 03:00.44 | marcus2 | linksys sells them that way |
| 03:00.52 | marcus2 | its just that most brick-and-mortars only carry the vonage ones |
| 03:01.06 | DrukenHME | i'm awear.., i have a few of them |
| 03:01.16 | marcus2 | so why do you want an unlocked vonage one, then? |
| 03:01.20 | DrukenHME | however you as a general user cannot buy one |
| 03:02.15 | marcus2 | so dont be a general user ;) |
| 03:02.17 | DrukenHME | marcus2: cause i'm pretty sure i've seen them go for $5 cash at bestbuy.. |
| 03:02.18 | *** join/#asterisk bweschke (n=bweschke@host-69-95-11-131.roc.choiceone.net) |
| 03:02.33 | marcus2 | no, they are $60 at bestbuy, then you get rebates |
| 03:02.44 | marcus2 | and most of the rebates only apply if you buy vonage service too |
| 03:02.47 | DrukenHME | us or canada? |
| 03:02.51 | marcus2 | US |
| 03:03.19 | YoMama | yeah..there's a $10 rebate..and another $50 rebate with a vonage account number |
| 03:03.23 | trixter | got a slight problem that I am hoping someone has a solution for. I need to make two outbound calls, play a greeting message to each of them, etc. each individual call is very straight forward via a call file, however the problem arises in that after playing the message I need to bridge them together. Is there any easy way to bridge 2 calls together in this situation? the only thing I can think of is app_conference or app_meetme with a unique conf |
| 03:03.23 | trixter | erence for each call pair. Has anyone encountered this problem, and if so what did they do to resolve it? |
| 03:03.24 | YoMama | so it's $50 if you don't sign up to vonage |
| 03:03.59 | DrukenHME | trixter, throw them both into a meetme |
| 03:04.01 | YoMama | trixster: that's one way to do it |
| 03:04.15 | YoMama | trixster: Dial, Playback, then do a Transfer |
| 03:04.19 | YoMama | into a meetme |
| 03:04.37 | X-Rob | Couldn't you use a LOCAL channel as a holder or something? |
| 03:04.45 | YoMama | there's probably a way to bridge the two too..but...i dunno |
| 03:04.45 | trixter | I was wondering if there was another solution because I dont particularly care for that one ... |
| 03:04.55 | trixter | but its the only thing I could thing of |
| 03:04.57 | trixter | er think |
| 03:05.26 | trixter | manager interface doesnt appear to have anything, closest is redirect, but that doesnt appear to do quite what I want |
| 03:05.32 | DrukenHME | only other way i can think of it is, perhaps park one, and have the other pick it up ? |
| 03:05.44 | trixter | hrm that is interesting |
| 03:06.07 | trixter | that may be better, I would rather avoid app_meetme/conference if possible ... parking and bridging may give me exactly what I need.. |
| 03:06.12 | trixter | thanks I will play with that |
| 03:06.23 | *** part/#asterisk YoMama (n=rewt@pcp02689850pcs.roylok01.mi.comcast.net) |
| 03:06.52 | trixter | its a ghetto bridge, but if it in effect does directly connect the two that is a good thing :D |
| 03:07.23 | trixter | well ghetto to set up I mean, it certainly is interesting |
| 03:07.36 | marcus2 | so.. |
| 03:07.49 | marcus2 | druken, you've set up a channel bank for asterisk? |
| 03:08.13 | DrukenHME | nope |
| 03:08.24 | DrukenHME | unfortunatly i lack the pri interface card |
| 03:08.29 | marcus2 | oh |
| 03:08.32 | marcus2 | hm |
| 03:09.04 | marcus2 | the wiki page i'm reading says to use fxo signalling with fxs ports on a channel bank, and vice versa |
| 03:09.05 | DrukenHME | however, it SHOULD be the same as setting up a normal PRI |
| 03:09.08 | marcus2 | i wonder if that makes sense |
| 03:09.17 | DrukenHME | just FXO instead of FXS signalling |
| 03:09.44 | marcus2 | well, it claims to be working |
| 03:09.56 | marcus2 | unfortunately i have no way to plug analog sets into the channel bank right now :/ |
| 03:10.09 | DrukenHME | no amphenol connector? |
| 03:10.15 | marcus2 | nope |
| 03:10.34 | marcus2 | maybe i can find one locally this weekend |
| 03:10.50 | DrukenHME | that's one thing i do have... hehe |
| 03:10.56 | marcus2 | fedex! ;) |
| 03:10.58 | marcus2 | oh wait, .ca |
| 03:11.14 | DrukenHME | have the entire channelbank wired to a bix block |
| 03:11.56 | marcus2 | yeah i will |
| 03:11.59 | marcus2 | i need to get the block tho |
| 03:12.03 | bweschke | marcus2: where r u? graybar has them |
| 03:12.16 | marcus2 | graybar is $$$$$$$$ |
| 03:12.20 | marcus2 | i'm in santa cruz |
| 03:12.20 | DrukenHME | oh.. i missed the *I* |
| 03:12.34 | marcus2 | so i can drive over the hill to silicon valley if i need to |
| 03:12.36 | bweschke | what about frys? |
| 03:12.44 | bweschke | if you're out west |
| 03:12.52 | marcus2 | i'd rather shop at graybar than frys ;) |
| 03:12.59 | bweschke | oh.. well ok |
| 03:12.59 | marcus2 | i think first i'll hit the surplus shops |
| 03:13.10 | marcus2 | theres a decent chance they'll have something |
| 03:13.20 | bweschke | true enuff - |
| 03:13.22 | DrukenHME | just macgyver one on it... :) |
| 03:13.31 | DrukenHME | two wires stuck into the lil holes... |
| 03:13.39 | marcus2 | yeah the thought has occurred to me |
| 03:13.40 | marcus2 | :) |
| 03:13.53 | marcus2 | i'm not even sure i have an extra rj11 cable kcking around that i want to cut, tho |
| 03:13.55 | DrukenHME | i did before i got the amphenol :) |
| 03:14.00 | marcus2 | whats the pinout? :) |
| 03:14.46 | DrukenHME | hehe, was one of those connect the two until i hear a click when i flick the switch |
| 03:14.57 | bweschke | pinout of what? |
| 03:15.12 | bweschke | oh.. didn't realize you were being sarcastic |
| 03:15.16 | bweschke | sorry <tired> |
| 03:16.38 | marcus2 | i wasnt |
| 03:16.39 | marcus2 | :) |
| 03:16.41 | *** join/#asterisk roamer323 (n=sing@HSE-Toronto-ppp177134.sympatico.ca) |
| 03:16.45 | marcus2 | i found an extra rj11 cable :) |
| 03:17.25 | DrukenHME | just make one :) |
| 03:17.26 | GeneG | Question re: SIP reinvites: My * is attempting to native-bridge even though I've got canreinvite=yes for both sides of the conversation. Any explanation for this? |
| 03:17.44 | file[laptop] | GeneG: that means nothing, it's normal |
| 03:17.47 | DrukenHME | uhmmm canreinvite=YES |
| 03:18.06 | marcus2 | ok i found it :) |
| 03:18.34 | DrukenHME | my brain isn't working..... i need a coke... |
| 03:18.38 | DrukenHME | someone buy me a coke |
| 03:19.07 | GeneG | Thanks file... am I to understand that "attempting a native bridge" can also mean that * will reinvite both parties for a point-to-point (non-bridged) conversation? |
| 03:19.14 | GeneG | I'm studying the SIP traces right now to get my head around it. |
| 03:19.41 | GeneG | DrukenHME... does your local pop machine have an IP address? |
| 03:21.15 | Nugget | <mugu> where does mysql keeps the database files again? |
| 03:21.15 | Nugget | <RyJones> /dev/random |
| 03:21.45 | GeneG | What I would really love, I mean LOVE, is a sample of what a SIP conversation looks like when Asterisk successfully reinvites both parties for a point-to-point conversation. |
| 03:21.48 | Frawg | hey Nugget |
| 03:22.32 | GeneG | Hey I just thought of a way to get that |
| 03:22.38 | GeneG | (I think( |
| 03:25.02 | marcus2 | fuck yeah, it works |
| 03:25.07 | marcus2 | sweeeet! |
| 03:25.39 | GeneG | Congrats marcus... nice to see somebody is getting somewhere tonight :) |
| 03:25.43 | *** join/#asterisk Mw3 (i=mw3@national.t-error.hu) |
| 03:38.57 | GeneG | Cool, "attempting native bridge" will appear regardless of whether a reinvite occurs or whether asterisk bridges the connection. And 'sip show channels' will show 2 channels even if a point to point RTP connection exists (presumably for call monitoring purposes). Is there a way to see in * whether a particular connection is being bridged through * or whether it is point to point? Right now I am using a packet sniffer to find this but it |
| 03:39.21 | morale | http://pastebin.ca/31919 - can someone let me know in that macro, if the number is busy it never goes to dialstatus s-BUSY |
| 03:39.47 | *** join/#asterisk sudhir492 (n=sudhir@pool-71-117-31-227.washdc.east.verizon.net) |
| 03:39.56 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
| 03:39.57 | sudhir492 | hi all |
| 03:40.12 | sudhir492 | anyone using flash operator panel? |
| 03:41.02 | sudhir492 | Is there a way to connect a call waiting in queue to an opertor through flash panel? |
| 03:41.32 | *** join/#asterisk slayer192 (n=chrisc@208.188.175.186) |
| 03:41.35 | *** join/#asterisk brc_ (n=brian@pdpc/supporter/basic/brc) |
| 03:41.46 | GeneG | Morale, do you get it going to s-NOANSWER or s-ANSWER when those conditions hit? |
| 03:41.59 | morale | yes. |
| 03:42.30 | GeneG | have you tried tracing the DIALSTATUS when you get a busy signal? |
| 03:42.55 | morale | hmm.. nope, just do a NoOp (${DIALSTATUS}) ? |
| 03:42.58 | tainted_ | anyone have audio problems with SIP? |
| 03:43.09 | GeneG | Ya I think that's the way it works |
| 03:43.14 | tainted_ | sometimes i dial a DID and get no audio |
| 03:43.18 | GeneG | You might get something other than BUSY |
| 03:43.24 | *** join/#asterisk bweschke (n=bweschke@host-69-95-11-131.roc.choiceone.net) |
| 03:48.16 | GeneG | Morale, looking at the Asterisk Handbook, it looks like a BUSY response on the interface will automatically jump to priority n + 101 (in your case, 103). Have you tried this? |
| 03:48.42 | morale | ah.. no i haven't |
| 03:49.00 | GeneG | When you give it a shot, could you let me know if it works? I'm just learning the system too |
| 03:49.03 | *** join/#asterisk ansong (n=ansong@adsl-67-66-93-240.dsl.okcyok.swbell.net) |
| 03:49.18 | morale | sure |
| 03:49.24 | GeneG | thx |
| 03:49.51 | morale | http://deadbolt.ca/dialplan.txt - is my extensions.conf |
| 03:50.04 | morale | pretty nifty, it plays music instead of ringing and uses lots of macro's |
| 03:50.32 | GeneG | cool |
| 03:50.40 | morale | GeneG: actually, being that i am using a macro i don't think i have to jump to priority 103 |
| 03:52.51 | DrukenHME | GeneG: the pop machine doesn't have an ip address, YET |
| 03:53.00 | GeneG | heh |
| 03:53.19 | GeneG | If you configure your pop machine successfully, I will buy you a coke. |
| 03:53.28 | slayer192 | has anyone here used a realtime config with meetme conferencing? |
| 03:53.35 | GeneG | If I configure my Asterisk successfully, I will buy ME a coke. |
| 03:53.57 | *** join/#asterisk Flauto (n=zhao@c-24-14-197-214.hsd1.il.comcast.net) |
| 03:54.15 | Flauto | is enum working for 800 numbers? |
| 03:54.34 | DrukenHME | it should.... can't see why it wouldn't |
| 03:54.46 | DrukenHME | but i bet no one thought to add their 800 numbers... |
| 03:55.19 | Flauto | no, i mean toll free service numbers, can you get through by enum lookup |
| 03:55.31 | Flauto | i am sorry for not making myself clear |
| 03:55.43 | Flauto | i tried a few time |
| 03:55.44 | Flauto | s |
| 03:55.55 | Flauto | and it says enumlookup successful |
| 03:56.20 | Flauto | but when it dialed enum, it told me that the call did not go through |
| 03:57.00 | GeneG | Morale, not sure about whether being in a macro affects the n+101 behaviour of BUSY. Did you have any luck with echoing the DIALSTATUS? |
| 03:57.04 | Flauto | i don't have to use enum for toll free numbers but it is just for my own experience |
| 03:57.25 | morale | GeneG: nope, still trying but it just jumps to s-NOANSWER |
| 03:58.38 | *** join/#asterisk Phil`OneNetwork (n=justinnn@61.95.68.85) |
| 03:58.45 | GeneG | So the NoOp isn't even getting executed? |
| 03:58.51 | morale | nope. |
| 03:58.56 | Phil`OneNetwork | Anyone here had issues with the Cisco 7.5 firmware and double |
| 03:58.57 | Phil`OneNetwork | <PROTECTED> |
| 03:58.57 | Phil`OneNetwork | <PROTECTED> |
| 03:59.12 | Phil`OneNetwork | And doesn't happen when you ring another internal voip phone |
| 03:59.30 | GeneG | what does the /g do at the end of the Dial? |
| 03:59.45 | morale | it means jump to a goto statement |
| 04:00.21 | morale | i gotta head to work for my nightshift, i'll be back on in about 45 minutes.. |
| 04:00.22 | GeneG | For testing purposes, does it make sense to temporarily remove the g so the NoOp gets executed? |
| 04:00.30 | morale | sure |
| 04:00.32 | GeneG | k |
| 04:00.37 | [TK]D-Fender | morale : Here's a free sample for you to expand on... http://pastebin.com/444510 |
| 04:01.11 | morale | [TK]D-Fender: thanks :) |
| 04:01.30 | [TK]D-Fender | np. I'm going to be doing so big work with it this weekend |
| 04:01.32 | morale | GeneG: nope still nothing |
| 04:02.03 | GeneG | You're doing a reload after making changes right :) |
| 04:02.07 | morale | yup |
| 04:02.43 | morale | i'll be back in a few. |
| 04:02.45 | GeneG | k |
| 04:03.23 | Phil`OneNetwork | Anyone had the issue I am having? |
| 04:03.43 | Phil`OneNetwork | (double ring back tone with 7.5 cisco firmware) |
| 04:04.55 | *** join/#asterisk jdiskywlkr (n=kvirc@ip68-0-83-251.tu.ok.cox.net) |
| 04:06.22 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
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| 04:27.49 | *** join/#asterisk marc32422 (n=marc3234@206-248-152-13.dsl.teksavvy.com) |
| 04:28.15 | marc32422 | do i need to install zaptel if I will not use any cards with asterisk? |
| 04:28.41 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 04:29.11 | mog_home | nope |
| 04:29.20 | mog_home | its nice for meetme though |
| 04:29.50 | slayer192 | marc32422: you'll need the ztdummy module for timing, it a part of zaptel |
| 04:30.33 | *** join/#asterisk swm_ (n=admin@digitaldatabits.net) |
| 04:30.55 | swm_ | How do you downsample a mp3 and set constant bitrate? Is there a linux utility i can use? |
| 04:31.13 | Katty | hi. |
| 04:31.14 | marc32422 | so even though no tdm boards will be installed.... i still need the zaptel package installed? |
| 04:31.15 | *** join/#asterisk {zombie} (i=zombie@soulasylum.penguincare.com.au) |
| 04:31.45 | swm_ | Katty help me |
| 04:31.46 | Katty | i wonder how many people come through here, ask a question, and then never get a reply. |
| 04:31.56 | Katty | swm_: it's 11pm |
| 04:31.59 | Katty | swm_: i am in bed. |
| 04:32.00 | swm_ | so? |
| 04:32.02 | slayer192 | marc32422: not unless you are using meetme conferences or music on hold |
| 04:32.09 | Katty | swm_: i am not helpingyou. |
| 04:32.18 | swm_ | just tell me the linux command katty |
| 04:32.20 | db48x | swm_: sox should do that, I'm sure |
| 04:32.30 | Katty | swm_: run along and bug someone else. |
| 04:32.38 | Katty | swm_: i'm relaxing. |
| 04:32.52 | marc32422 | moh requires zaptel package? |
| 04:33.13 | Math` | no |
| 04:34.16 | slayer192 | moh needs a timing source, you get timing from either the zaptel hardware or the ztdummy module |
| 04:34.55 | marc32422 | ztdummy is in zaptel package? |
| 04:35.42 | slayer192 | yes, if you are running a 2.6 kernel it should be built by default |
| 04:45.27 | swm_ | I need to convert a standard MP3 to play in musiconhold anyone know what to do? |
| 04:46.36 | Nugget | hire a high school dropout to sing all day and just transfer calls to a meetme() with him in it. |
| 04:46.40 | litage | swm_: what are the current MOH sounds files? |
| 04:47.27 | Newbie___ | swm_: i think u need to convert from mp3 to gsm, i can be wrong though |
| 04:47.35 | Nugget | hihi katty |
| 04:47.46 | Katty | Nugget: are you causing trouble? |
| 04:47.51 | Nugget | nope. |
| 04:47.58 | Katty | well hop to it! |
| 04:48.02 | mog_home | you are newbie |
| 04:48.10 | mog_home | you need to convert to 8khz mono swm |
| 04:48.15 | mog_home | i think lame can do that |
| 04:48.22 | swm_ | how do I do that any standard unix commands to do this? |
| 04:48.38 | Katty | isn't it ulaw 8 mono? |
| 04:48.48 | Katty | or am i thinking of polycom ringtone format or something |
| 04:49.49 | mog_home | ulaw is anything |
| 04:49.54 | mog_home | lame |
| 04:49.58 | mog_home | can do it swm_ |
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| 04:59.08 | *** join/#asterisk KranZ (i=KranZ@music.queso.net) |
| 05:01.35 | morale | finally at work |
| 05:03.09 | GeneG | Heh |
| 05:04.05 | GeneG | Morale, I've been playing with it and the NoOp seems to work great if you don't include any additional parameters after the dial. |
| 05:04.13 | GeneG | Have you tried that yet? Does the NoOp execute? |
| 05:04.51 | *** join/#asterisk _santiago_ (n=santiago@208.195.215.160) |
| 05:05.23 | morale | GeneG: yep, the NoOp works when it is before the Dial() function |
| 05:08.29 | GeneG | Heh, not much help until after the dial. Have you tried s,2,Dial(${ARG1},50) |
| 05:08.37 | GeneG | i.e. no ,g at the end? |
| 05:11.31 | morale | yup |
| 05:11.43 | morale | its evil. |
| 05:11.46 | GeneG | Try it outside of the macro |
| 05:11.51 | GeneG | That's how I have it running right now |
| 05:12.06 | GeneG | I also don't have the ,50 |
| 05:12.12 | GeneG | might be worth trying for testing purposes |
| 05:12.46 | Ikarus | Hmmmmmm, I need to do some time measurements |
| 05:14.04 | GeneG | Morale, I also noticed 2 settings that you might want to tweak in your extensions.conf: |
| 05:14.12 | GeneG | priorityjumping and autofallthrough |
| 05:14.31 | GeneG | Priorityjumping in particular might need to be yes in order for this to work |
| 05:14.38 | GeneG | (just a guess though) |
| 05:15.06 | morale | lemme try that |
| 05:18.35 | morale | http://pastebin.ubuntulinux.nl/5256 - that is my newest extensions.conf, still no workie. |
| 05:20.09 | GeneG | Weren't you going to dump the params after the Dial? |
| 05:20.29 | GeneG | Right now the biggest thing is to get the NoOp to execute |
| 05:20.55 | GeneG | This may be nothing, but try getting rid of the space between the NoOp and the open bracket |
| 05:25.41 | morale | i think it is the dial function, i am missing something.. almost like.. if it times out don't automatically jump to s-NOANSWER, check the conditions available |
| 05:28.24 | kuku5 | is there a good script for setups? |
| 05:28.36 | kuku5 | web based...php |
| 05:28.43 | kuku5 | to extensions.conf |
| 05:29.07 | morale | for setups? |
| 05:29.14 | morale | GeneG: nope that still didn't work |
| 05:39.18 | marc32422 | how can i unistall asterisk? |
| 05:41.28 | benjk_ | you can download out uninstaller (shell script) |
| 05:41.52 | GeneG | Morale, try taking it out of the macro |
| 05:42.18 | benjk_ | its for MacOS X but since it is an ordinary shell script and all the files are in the same places, it should work just fine on Linux |
| 05:42.36 | benjk_ | it even archives your config files and voicemail etc |
| 05:42.51 | benjk_ | do you want to try that? |
| 05:50.12 | benjk_ | marc32422 ??? |
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| 06:02.26 | *** part/#asterisk Mavvie (i=edwin@252-131-222-203.rev.techex.net.au) |
| 06:11.21 | *** part/#asterisk slayer192 (n=chrisc@208.188.175.186) |
| 06:12.00 | *** part/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
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| 06:22.11 | IronHelix | hmmm |
| 06:22.14 | IronHelix | i have a nasty issue |
| 06:22.26 | IronHelix | i have a box here, (well not here) that was runnign asterisk@home |
| 06:22.36 | IronHelix | now upgraded via FTP |
| 06:22.50 | IronHelix | anyway, any digits sent on RFC2833 are very unreliably detected |
| 06:22.59 | IronHelix | it always detects the digit at least once, but often 2-3 times |
| 06:23.45 | drumkilla | is this up to date code? |
| 06:24.33 | IronHelix | yeah, happens on 1.0.9, 1.0.10, and 1.2.0 with and without the DTMF patch |
| 06:24.36 | *** join/#asterisk brookshire (n=nubb@gateway.digium.com) |
| 06:25.00 | IronHelix | i have cleaned out my module folders before make installing each time |
| 06:25.36 | IronHelix | could it be zaptel related at all? i have used 1.0.9.2 and 1.2.0 |
| 06:26.14 | drumkilla | if it happens in all of those versions, i'm not sure what it could be ... |
| 06:27.04 | IronHelix | neither am I |
| 06:27.27 | IronHelix | the system has also at one point had CVS HEAD on it |
| 06:27.45 | IronHelix | is there some folder somewhere that im not cleaning that is holding old code? |
| 06:28.05 | kuku5 | anyone using ip switchboard? |
| 06:28.09 | IronHelix | i have dumped out /usr/lib/asterisk/modules/ and /usr/include/modules |
| 06:28.22 | morale | is there a way to convert all of my recorded .gsm files to g729 format? |
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| 06:32.32 | *** mode/#asterisk [+o twisted[asteria]] by ChanServ |
| 06:35.52 | *** join/#asterisk YoMama (n=r00t@pcp02689850pcs.roylok01.mi.comcast.net) |
| 06:37.09 | YoMama | anyone play around with bluetooth on laptops? |
| 06:38.47 | *** join/#asterisk magic_1 (n=maig_1@dsl-165-146-200.telkomadsl.co.za) |
| 06:39.48 | YoMama | guess not |
| 06:41.54 | Qwell | morale: can sox do g729? |
| 06:43.04 | *** join/#asterisk benjk (n=benjk@f8a01-0357.din.or.jp) |
| 06:43.18 | *** join/#asterisk twisted-- (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 06:43.18 | *** mode/#asterisk [+o twisted--] by ChanServ |
| 06:43.31 | drumkilla | actually ... you could use asterisk to do that :) |
| 06:43.42 | drumkilla | not that it's obvious how ... |
| 06:44.26 | GeneG | YoMama: I've got BT headset -> XTen Softphone -> Asterisk -> Vonage VOIP |
| 06:45.29 | Qwell | twisted: |
| 06:45.31 | Qwell | - |
| 06:45.47 | YoMama | i figured it out |
| 06:45.58 | YoMama | GeneG: i've been playing with it for two hours and i just figured it out |
| 06:46.04 | GeneG | congrats :) |
| 06:46.08 | YoMama | and now i'm gonna bang my head against the wall for 20 minutes for feeling like an idiot |
| 06:46.13 | YoMama | so when i told my laptop to connec tot he headset |
| 06:46.19 | YoMama | it made this "beep beep" noise |
| 06:46.29 | YoMama | i'm like "why is it just beeping and not playing any audio?" |
| 06:46.42 | Qwell | if everybody in here banged their heads against the wall for feeling stupid, we'd have a channel full of idiots... |
| 06:46.46 | distortion | morale: I painfully used the web tool on asterguru to convert all my .gsm files to g729 |
| 06:46.48 | Qwell | so, don't do that |
| 06:46.48 | GeneG | didn't press the connect button on the headset? |
| 06:46.55 | YoMama | but..all u gotta do is press this silly little call button |
| 06:46.56 | YoMama | uh huh |
| 06:47.00 | GeneG | :) |
| 06:47.03 | YoMama | well, when i pair it with my cell phone..u don't haveta "connect" it |
| 06:47.18 | GeneG | Ya, BT on PCs is a kludge IMO |
| 06:47.28 | GeneG | Don't get me started... |
| 06:47.53 | YoMama | well, at least i can now use my X-Lite phone at a cafe without looking like a dork with a big headset with a cord hanging down |
| 06:48.00 | YoMama | now i just look crazy because it loosk like i'm talking to myself |
| 06:48.19 | GeneG | Ya, I think you're pretty much a dork as soon as you open the laptop in the cafe in the first place :) |
| 06:48.32 | magic_1 | hehehhe |
| 06:48.38 | drumkilla | YoMama: especially when people see the other side of your head, so they can't see the headset at all |
| 06:48.38 | morale | asterisk with vonage works now? |
| 06:48.44 | Qwell | yeah...headset makes you look much less like a dork |
| 06:48.46 | rking | talking to in-person people? weird. |
| 06:48.47 | IronHelix | hah, i wish |
| 06:48.55 | IronHelix | i think he meant asterisk -> fxo -> vonage box |
| 06:48.59 | morale | ah |
| 06:49.03 | IronHelix | although |
| 06:49.03 | GeneG | Morale: Asterisk works with Vonage's softphone service, which you need to pay extra for |
| 06:49.40 | IronHelix | the linksys pap2 has been pretty well hacked, so now with a tool called VuckFonage (lol) and an unlocked PAP2 you can download and decrypt your tftp provisioning file |
| 06:49.48 | IronHelix | so in theory you could import the sip settings |
| 06:49.49 | GeneG | I am paying Canadian $20 for the basic Vonage service (which I'm not even using) + $13 for the softphone line, which I use exclusivelky. |
| 06:49.51 | IronHelix | easy for them to detect tho |
| 06:50.09 | morale | i pay 10$ canadian for my line with www.link2voip.com |
| 06:50.22 | Qwell | I've spent $5 in the last 6 months with voip |
| 06:50.33 | IronHelix | qwell- wholesale? |
| 06:50.37 | Qwell | no |
| 06:50.45 | GeneG | Hey I'm not deluding myself that I'm getting a good deal with Vonage, I'm just saying that it works. |
| 06:50.57 | Qwell | GeneG: nufone works with asterisk |
| 06:51.05 | YoMama | IronHelix: assuming you'd actually wanna be on vonage |
| 06:51.27 | Qwell | Available Funds: $5.8862 |
| 06:51.27 | Qwell | Last payment: 2005-03-18 13:26:17 |
| 06:51.33 | IronHelix | i wish i wasn't :( |
| 06:51.41 | Qwell | So, $4.1138 in the last 8.5 months |
| 06:51.42 | YoMama | Iron: i just bought a PAP2 from Best Buy |
| 06:51.46 | IronHelix | i've been trying desperately to port my number out :( |
| 06:52.01 | drumkilla | Qwell: nice. That's what I use too :) |
| 06:52.01 | YoMama | Iron: haven't unlocked it yet...when u buy the damn thing tho..they ask for your address..and i wasn't prepared to lie and didn't have enough cash..so i paid in credit |
| 06:52.14 | YoMama | think i'm gonna haveta return it..and buy another under the name Harry Balzac |
| 06:52.15 | Qwell | drumkilla: I *hate* phones...so I never use them. :D |
| 06:52.18 | morale | Qwell: who is your provider? |
| 06:52.22 | Qwell | morale: nufone |
| 06:52.22 | drumkilla | Qwell: same here |
| 06:52.27 | drumkilla | I don't actually *use* this stuff |
| 06:52.27 | Qwell | drumkilla: odd, isn't it? |
| 06:52.32 | drumkilla | just work on the code ... |
| 06:52.33 | drumkilla | yes, it is. |
| 06:52.34 | morale | is nuphone canadian? |
| 06:52.41 | Qwell | far more enjoyable than other things |
| 06:52.42 | Qwell | morale: no |
| 06:52.42 | drumkilla | I still have fun with it, though, somehow :) |
| 06:52.47 | Qwell | indeed |
| 06:53.08 | morale | canada always gets the shaft. |
| 06:53.25 | Corydon76-home | No, Nufone is Michiganian |
| 06:53.31 | Qwell | sounding like a tool due to phones kinda sucks...I think that may be why I hate phones. |
| 06:53.40 | Qwell | BUT...app_voicechangedial removes a lot of that... |
| 06:53.54 | morale | voicechange? |
| 06:53.55 | morale | ooo. |
| 06:54.02 | Qwell | (maybe I actually sound like that in person? I don't know...) |
| 06:54.09 | *** join/#asterisk yuvan_k (n=scowser8@219.95.78.33) |
| 06:54.27 | Qwell | morale: Don't you get rss feeds from ADN? |
| 06:54.28 | Qwell | ~adn |
| 06:54.29 | jbot | somebody said adn was the Asterisk Daily News - http://www.sineapps.com/news.php for HTML and http://www.sineapps.com/rssfeed.php for RSS |
| 06:54.33 | Qwell | sheesh |
| 06:55.22 | morale | ooo. |
| 06:55.30 | morale | heh |
| 06:55.38 | mog_home | sheesh? |
| 06:55.50 | Qwell | sheesh! |
| 06:56.14 | yuvan_k | hi guys, i've been trying to disable call waiting on my zap line but it doesn't seem to work...i have tried to change the values in zapata, zapata-auto, zapata-additional |
| 06:56.29 | yuvan_k | any ideas how i tackle this issue? |
| 06:56.41 | Qwell | amp? eww |
| 06:56.43 | YoMama | yuvan: why not have all of your dialplans include *70? |
| 06:57.38 | yuvan_k | hmm...i shall try it out |
| 06:57.39 | *** join/#asterisk rvhi (n=rv@66.175.65.89) |
| 06:57.57 | YoMama | ya know...*70${EXTEN:1} |
| 06:58.00 | YoMama | that sorta stuff |
| 06:58.49 | yuvan_k | ok, i will try it out |
| 06:59.26 | yuvan_k | but the problem is when i pick up the phone and hit *70, the msg says call waiting activated..which means its deactivated by default |
| 06:59.56 | YoMama | huh |
| 07:00.02 | yuvan_k | but why does it still sends the next call to zap/1-2 instead of signalling line is busy |
| 07:00.04 | YoMama | i dunno man |
| 07:00.44 | yuvan_k | hehe...probably i am bad at trying to explain my problem |
| 07:00.58 | yuvan_k | its ok, i will try out the *70 first, then i will let u know |
| 07:03.53 | magic_1 | <PROTECTED> |
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| 07:06.22 | magic_1 | <PROTECTED> |
| 07:06.56 | Qwell | magic_1: such as? |
| 07:07.01 | Qwell | and, don't repeat |
| 07:08.02 | magic_1 | sorry about that i was busy on gaim and pushed up there and enter instead of looking where it was doing it (i am using two monitors) |
| 07:08.46 | magic_1 | the problem is that when i try to compile it ,it gives me the following error |
| 07:08.56 | magic_1 | give me a sec i will pacth it here now |
| 07:09.30 | Qwell | ~pb |
| 07:09.32 | jbot | somebody said pastebin was a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 07:12.28 | magic_1 | when i do make it gives me the following error |
| 07:12.42 | magic_1 | collect2: ld returned 1 exit status |
| 07:12.43 | magic_1 | make[1]: *** [app_curl.so] Error 1 |
| 07:12.43 | magic_1 | make[1]: Leaving directory `/home/quinton/Desktop/asterisk-1.2.0/apps' |
| 07:12.43 | magic_1 | make: *** [subdirs] Error 1 |
| 07:12.54 | mog_home | ahh pastebin |
| 07:13.03 | mog_home | and you need libcurl or libcurl-dev |
| 07:13.22 | magic_1 | dam that makes sens now |
| 07:13.32 | magic_1 | how would i know which one |
| 07:13.45 | mog_home | well i think it detected libcurl headers |
| 07:13.52 | mog_home | and you dont have the libcurl installed |
| 07:14.35 | morale | GeneG: look at this: |
| 07:14.36 | morale | <PROTECTED> |
| 07:14.36 | morale | <PROTECTED> |
| 07:14.47 | morale | when the dial status is NOANSWER it actually does run the NoOp() function |
| 07:15.01 | magic_1 | so i should then just yum install libcurl then try make clean and then make again |
| 07:15.17 | magic_1 | mog_home really appreciate the help |
| 07:15.17 | Qwell | speaking of libcurl |
| 07:15.17 | mog_home | just do make again |
| 07:15.23 | mog_home | no problem |
| 07:15.26 | Qwell | I wonder if it does it here too... |
| 07:15.27 | yuvan_k | hi guys, lets say my channel is zap/1. How do i disable my incoming calls from going into zap/1-2 when zap/1-1 is in use? |
| 07:15.29 | magic_1 | thanx |
| 07:15.30 | mog_home | after you installe it |
| 07:15.41 | mog_home | libcurl that is |
| 07:15.58 | mog_home | turn off three way calling yuvan_k |
| 07:16.04 | mog_home | and call waiting |
| 07:16.33 | Qwell | how funky |
| 07:16.39 | yuvan_k | mog home: i'm using amp, so i just do that in zapata.conf only? |
| 07:16.50 | mog_home | yes |
| 07:16.56 | Qwell | at work, on the same distro, same version of curl... curl-config --vernum doesn't show anything, so app_curl isn't compiled |
| 07:17.10 | mog_home | is the curl.h? |
| 07:17.13 | mog_home | on machine |
| 07:17.28 | Qwell | if I force app_curl to compile, it works fine |
| 07:17.33 | mog_home | indeed |
| 07:17.37 | mog_home | it must check wrong |
| 07:17.47 | Qwell | dunno, same version and everything |
| 07:18.21 | mog_home | jupiter was at the wrong spot |
| 07:18.41 | Qwell | likely |
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| 07:20.09 | yuvan_k | turning off threewaycalling doesn't seem to work, probably something else i'm missing out on? |
| 07:20.26 | yuvan_k | i've turned off callwaiting as well |
| 07:20.29 | Qwell | okay, wtf |
| 07:20.31 | *** join/#asterisk newl (n=newlook@203-59-214-216.dyn.iinet.net.au) |
| 07:20.36 | Qwell | curl-config is just a bash script |
| 07:20.39 | Qwell | echo 070f00 |
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| 07:20.54 | Qwell | now it makes no sense why it's returning nothing |
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| 07:23.36 | yuvan_k | do u think i need to make changes on extension.conf instead? |
| 07:24.36 | mog_home | no |
| 07:25.59 | magic_1 | mog_home when i say yum install libcurl it tells |
| 07:26.08 | magic_1 | No Match for argument libcurl |
| 07:26.08 | magic_1 | Nothing to do |
| 07:26.37 | Qwell | I think it's just called "curl" |
| 07:26.58 | magic_1 | let me try quick thanx |
| 07:27.06 | mog_home | install curl and libcurl-dev |
| 07:27.08 | mog_home | to be safe |
| 07:27.19 | magic_1 | i am going to try that now |
| 07:28.01 | yuvan_k | mog_home, is there any other way i can rectify the problem? |
| 07:28.23 | Qwell | heh, he said rectify |
| 07:28.24 | magic_1 | well curl looks lik its installing |
| 07:28.56 | yuvan_k | hehe |
| 07:29.33 | yuvan_k | my problem looks fairly simply, probably i can't get it right coz i am still a newbie |
| 07:30.02 | yuvan_k | compared to curl, i have absolutely no idea what u r talking abt |
| 07:31.38 | magic_1 | is iot far to say that the mmx optimzation option will work fine on the AMD 64 |
| 07:32.46 | magic_1 | it work it work thanx guys that curl installation sorted out my porb thanx |
| 07:33.57 | *** join/#asterisk Rocc0 (n=rocco@vg28.vodafone.com.au) |
| 07:34.00 | magic_1 | will it be safe for me to do make webvmail before or after i do make install |
| 07:35.01 | *** join/#asterisk llirk (n=majestic@203-214-58-222.dyn.iinet.net.au) |
| 07:35.12 | llirk | Hello |
| 07:35.22 | magic_1 | hello |
| 07:35.30 | llirk | how are ya? |
| 07:35.37 | magic_1 | good and you |
| 07:35.43 | *** join/#asterisk cuco (n=elcuco@local.xorcom.com) |
| 07:35.47 | llirk | rockin.. |
| 07:35.49 | cuco | hello all |
| 07:35.53 | magic_1 | lo there |
| 07:36.01 | cuco | anyone here tested asterisk for memory leakage? |
| 07:36.18 | llirk | im sure someone has |
| 07:36.31 | cuco | i have this setup in which (I think) asterisk is getting killed after a few days because of OOM |
| 07:38.00 | mog_home | many people have cuco |
| 07:38.13 | mog_home | everynow and then someone finds a mem leak |
| 07:38.26 | Qwell | mog_home: You sound like you're talking about VD |
| 07:38.28 | mog_home | but alot of people run asterisk boxes for long times periods |
| 07:38.32 | mog_home | lol |
| 07:38.34 | mog_home | its like that |
| 07:38.41 | Qwell | "many people have cuco" <-- |
| 07:40.05 | cuco | so..? does asterisk leak? |
| 07:40.26 | cuco | the problem is that the machine running asterisk has no swap, so it's "problematic" |
| 07:40.30 | mog_home | not that im aware of |
| 07:40.41 | mog_home | but thats why we have bugs.digium.com |
| 07:41.31 | cuco | well, ps flax should tell me the ammount of memory accupied by some aplication right? |
| 07:42.12 | cuco | (may i paste 2 lines of ugly long text?) |
| 07:42.19 | mog_home | yes.... ^_^ |
| 07:42.25 | drumkilla | NO |
| 07:42.41 | drumkilla | or the channel will ignite with fire!! |
| 07:42.41 | cuco | drumkilla: :( |
| 07:42.45 | mog_home | bump you your lurking |
| 07:42.48 | cuco | 1 103 23292 1 -11 0 367740 89028 rest_i Ss ? 674:17 /usr/sbin/asterisk -p -U asterisk |
| 07:42.49 | drumkilla | cuco: i was kidding |
| 07:42.51 | mog_home | and trolling |
| 07:42.55 | mog_home | at the same time... |
| 07:43.16 | cuco | 1 103 23292 1 -11 0 286872 18768 rest_i Ss ? 18:21 /usr/sbin/asterisk -p -U asterisk |
| 07:43.51 | cuco | in those lines we can see that the asterisk process is getting bigger in time (the last line is earlier) |
| 07:43.52 | *** join/#asterisk ComputerWarm (n=workingg@66.244.235.210) |
| 07:46.32 | drumkilla | looks like a leak to me :) |
| 07:46.37 | drumkilla | is this up to date? |
| 07:48.09 | cuco | the first entry was from 30 minutes ago, the system is rapid xorcom (asterisk 1.0.9.something) |
| 07:48.37 | drumkilla | ok, you should either update to 1.0.10, or 1.2.0 |
| 07:48.48 | drumkilla | I would seriously recommend considering switching to 1.2 |
| 07:48.52 | cuco | this means .10 |
| 07:49.21 | cuco | drumkilla: switching to 1.2 is a pain, and probably will not happen. |
| 07:49.46 | drumkilla | ok, just be warned that 1.0 is pretty much not being maintained anymore. |
| 07:50.16 | cuco | that is known, and we do know how to use apt-get |
| 07:51.29 | drumkilla | alright then, I guess you're the expert! |
| 07:51.33 | drumkilla | good night folks |
| 07:53.11 | magic_1 | has anyone here try the make rpm command |
| 07:57.44 | morale | is it possible to do call mirroring? like if some telemarketter calls me that it rewrites the destination number to their own phone number? |
| 07:57.52 | morale | or does it not work like the ip protocol :) |
| 07:58.30 | mog_home | no not really morale |
| 07:59.43 | IronHelix | sure it is |
| 07:59.48 | IronHelix | look up telemarketer in database |
| 08:00.02 | IronHelix | if they're there, then Dial(${CALLERIDNUM}) |
| 08:02.13 | yuvan_k | hey IronHelix |
| 08:02.27 | yuvan_k | thx for answering my posts in the forum |
| 08:02.57 | morale | someone fix my dialplan so it actually jumps to s-BUSY :) |
| 08:03.29 | IronHelix | no problem |
| 08:04.07 | morale | http://pastebin.com/444633 - in the macro [macro-regular], it only ever gets to s-NOANSWER.. even if the number is busy |
| 08:06.19 | *** join/#asterisk freezer (n=leetiden@ACB7A74E.ipt.aol.com) |
| 08:06.22 | freezer | hi |
| 08:06.25 | *** join/#asterisk YoMama (n=r00t@pcp02689850pcs.roylok01.mi.comcast.net) |
| 08:06.34 | freezer | hi |
| 08:06.36 | YoMama | aaah...PAP2 all set up on asterisk |
| 08:09.47 | *** join/#asterisk Netgeeks (n=Chris@68-185-24-2.static.mdfd.or.charter.com) |
| 08:09.49 | *** join/#asterisk FaithX (n=FaithX@202-6-145-116.ip.adam.com.au) |
| 08:10.54 | *** join/#asterisk SwK[Work] (n=SwK@border0hsv.asterisksgi.com) |
| 08:11.26 | morale | j - Jump to priority n+101 if all of the requested channels were busy. - What would 'n' be equal to? would I take the priority of my Goto (${DIALSTATUS}) or my Dial() function? |
| 08:14.08 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) |
| 08:14.40 | YoMama | morale: what are u trying to do? |
| 08:14.49 | YoMama | n is your last priority |
| 08:15.00 | YoMama | if your dial was priority 6 |
| 08:15.13 | YoMama | then 6+101 = 107 |
| 08:16.17 | magic_1 | Yomama please could u advise me ,when i try loading my zaptel module it tells me the following |
| 08:16.32 | magic_1 | [root@share zaptel-1.2.0]# modprobe | grep zaptel |
| 08:16.32 | magic_1 | bash: modprobe: command not found |
| 08:16.36 | *** join/#asterisk chapeaurouge (n=chap@85.201.80.249) |
| 08:16.49 | YoMama | eh |
| 08:16.50 | Qwell | magic_1: How did you become root? |
| 08:16.57 | Qwell | su? |
| 08:17.02 | magic_1 | yeah |
| 08:17.04 | YoMama | magic: your path statement can't find modprobe |
| 08:17.08 | Qwell | use `su -` |
| 08:17.14 | Qwell | with the dash |
| 08:17.21 | YoMama | yeah..then i'tll use your environment from your last user |
| 08:17.23 | YoMama | or something like that |
| 08:17.24 | YoMama | ohb oy |
| 08:17.25 | magic_1 | makes sens thanks let me try quic |
| 08:17.27 | Qwell | or /sbin/modprobe |
| 08:17.45 | Qwell | - adds the env of the new user |
| 08:18.09 | freezer | i just ordered two hfc-s isdn pci cards |
| 08:18.28 | YoMama | Qwell: i remember back in the days when only schools and big companies had unix boxes..and to have root access...was a big deeal |
| 08:18.33 | YoMama | now everyone's got roooot |
| 08:21.44 | morale | anyone know of an always busy number? i wanna test asterisk's busy signal |
| 08:22.15 | mog_home | you can just make asterisk make the busy noise morale |
| 08:22.23 | mog_home | isnt there an busy() |
| 08:22.25 | mog_home | or no |
| 08:22.27 | mog_home | congestion |
| 08:22.34 | mog_home | thats kinda the same... |
| 08:22.54 | *** join/#asterisk indego (n=chris@floyd.gms.lu) |
| 08:23.26 | morale | ah ok :) |
| 08:23.28 | *** join/#asterisk Netgeeks_ (n=Chris@68-185-24-2.static.mdfd.or.charter.com) |
| 08:23.46 | morale | i want asterisk to play "This line is busy" instead of a busy tone when calling somewhere |
| 08:24.10 | morale | <PROTECTED> |
| 08:24.11 | morale | finally! |
| 08:25.03 | morale | why does gsm always take precedence over g729a? |
| 08:25.10 | *** part/#asterisk Netgeeks_ (n=Chris@68-185-24-2.static.mdfd.or.charter.com) |
| 08:26.07 | *** join/#asterisk lorinc (n=ang@caracas-2552.adsl.interware.hu) |
| 08:27.30 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 08:29.20 | YoMama | morale: because that's hwo u have it set up |
| 08:37.43 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 08:43.39 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:45.09 | niZon | http://pastebin.com/444652 |
| 08:45.11 | niZon | any ideas? |
| 08:45.25 | niZon | I installed from apt packages |
| 08:47.32 | cuco | niZon: apt..? deb? rpm? which distro? |
| 08:50.52 | niZon | ubuntu |
| 08:52.14 | *** join/#asterisk testmachine (n=assink@ip237-239-58-62.adsl.versatel.nl) |
| 08:53.36 | yuvan_k | hi guys, i still can't disable the 3 way calling on my zap channel, i've pasted my zapata-auto, zapata and zapata-additional on http://pastebin.com/444657. Would appreciate it if someone can tell me what i am doing wrong |
| 08:53.38 | yuvan_k | thanks |
| 08:55.06 | *** join/#asterisk ManxPower (n=ewieling@adsl-67-65-233-194.dsl.lgvwtx.swbell.net) |
| 08:55.59 | *** join/#asterisk ptiggerdine (n=ptiggerd@c220-239-93-118.rochd4.qld.optusnet.com.au) |
| 08:57.26 | yuvan_k | any suggestions? |
| 09:03.02 | niZon | I'd help you, but I haven't used zap much.. sorry |
| 09:04.34 | yuvan_k | thanks |
| 09:04.39 | *** join/#asterisk glock (n=glock@dsl-146-28-149.telkomadsl.co.za) |
| 09:04.53 | yuvan_k | its something minor, probably a value that i forgot to change, but i can't find what it is |
| 09:05.00 | yuvan_k | working on it the whole day |
| 09:05.55 | *** join/#asterisk SwK[Work] (n=SwK@border0hsv.asterisksgi.com) |
| 09:06.49 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 09:10.48 | *** join/#asterisk dudes (n=dudes@12-215-32-62.client.mchsi.com) |
| 09:12.42 | GeneG | Is there a setting in sip.conf that allows you to override the value Asterisk places in the Connection infomration (c) field of the SDP during a reinvite? |
| 09:14.33 | glock | Could anyone help me with the asterisk sip linphone setup. linphone prompts me to authenticate, it has realm, username, password and userid. Im not sure what values go where. I have a [101] entry in my sip.con with secret of 101. What is the userid? |
| 09:15.27 | benjk | 101 is the user id |
| 09:15.43 | benjk | realm is the server's domain name or ip address |
| 09:16.18 | glock | ok, username would be 101@domain? |
| 09:16.31 | benjk | could be yes |
| 09:16.48 | benjk | Asterisk only has user and password |
| 09:17.07 | glock | mm, yeah. doesnt work. just throws up the auth box again |
| 09:17.54 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
| 09:22.36 | marcus2 | try 'sip debug' on the asterisk server |
| 09:24.25 | testmachine | hm |
| 09:24.41 | testmachine | anyone experience with diva isdn cards> |
| 09:24.43 | testmachine | ? |
| 09:25.52 | *** join/#asterisk Aze` (n=aze@85-18-136-114.ip.fastwebnet.it) |
| 09:26.55 | *** join/#asterisk lme (n=lme@gw-r-ifc.infoclip.fr) |
| 09:27.40 | *** join/#asterisk ManxPower (n=ewieling@67.65.233.194) |
| 09:29.52 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 09:30.02 | *** join/#asterisk EriSan (n=erisan@151.8.109.90) |
| 09:36.58 | glock | marcus2, benjk - thanks. got it logging in. Just cant call anywhere now. I have the two extensions. Am i missing anything? exten => 101,1,Dial(SIP/101,10,t) |
| 09:38.04 | glock | when i place the call, im trying to call sip:101@domain |
| 09:41.38 | *** join/#asterisk magic_1 (n=maig_1@dsl-165-146-200.telkomadsl.co.za) |
| 09:41.47 | magic_1 | lo all |
| 09:42.01 | magic_1 | how have you all been |
| 09:46.03 | magic_1 | Qwell i have this so to say problem - when i try to start up my asterisk it gives me the following problem |
| 09:46.16 | magic_1 | <PROTECTED> |
| 09:46.16 | magic_1 | Dec 1 11:27:02 WARNING[3920]: loader.c:554 load_modules: Loading module codec_gsm.so failed! |
| 09:46.56 | kll | is it possible to change the called number via a AGI script? |
| 09:47.03 | *** join/#asterisk DannyF (n=dannyf@c-fb64e455.24-0099-74657210.cust.bredbandsbolaget.se) |
| 09:48.04 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 09:50.49 | Aze` | kll> SETCALLERID |
| 09:51.45 | kll | Aze`: no, that's the callers number. I wanna change the extension to which the call is going |
| 09:52.04 | sudhir492 | hi all |
| 09:53.31 | glock | Im using linphone and trying to make calls. I am using asterisk as the proxy. If i do 103@192.168.0.7 it works, but i really want to just do 103@example.net. Any pointers? |
| 09:58.12 | sudhir492 | I do not use linphone, but my guess will be check your computer and linphone for nameservers |
| 09:58.39 | Aze` | kll > sorry, i understood caller |
| 09:59.40 | sudhir492 | anyone using flash operator panel here? |
| 10:02.42 | Ikarus | z |
| 10:08.18 | *** join/#asterisk kernelbee (n=Naveed@80.77.12.2) |
| 10:14.58 | kll | is it possible to route calls to a different context based on say the three first numbers? ie calls going to 012. should go to context [012] ? |
| 10:16.21 | ptiggerdine | kill your looking at a say an inbmound phone number of 12345678 and 678 being the extension? |
| 10:17.03 | kll | yes |
| 10:17.03 | ptiggerdine | i think this can be done using msn... I think msn is that named they give it. |
| 10:17.15 | kll | msn? |
| 10:17.39 | ptiggerdine | I'll check one sec.. |
| 10:18.35 | *** join/#asterisk eivindtr (n=wingnut-@193.213.28.152) |
| 10:19.53 | ptiggerdine | it's not msn, |
| 10:20.34 | ptiggerdine | kll, I do believe that when a call is initiated the peer sends down the line the number they are calling. |
| 10:20.56 | ptiggerdine | this also may only be a isdn feature, not sure |
| 10:22.11 | Ikarus | bleha |
| 10:24.11 | *** join/#asterisk SwK[Work] (n=SwK@border0hsv.asterisksgi.com) |
| 10:24.30 | Ikarus | I need to annoy my boss |
| 10:24.57 | drray | don't you do that everyday? |
| 10:25.26 | Ikarus | normally I don't come in with news that we wasted 1200 euros |
| 10:25.47 | Ikarus | my predecessor's fault |
| 10:26.01 | Ikarus | and not known to me until I had to deal with some randomness |
| 10:26.15 | drray | it does happen |
| 10:27.14 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 10:27.19 | Ikarus | Doesn't make it easier to explain to the boss |
| 10:28.17 | drray | my employer is forgiving of that kind of stuff |
| 10:28.23 | drray | I don't go to the well often |
| 10:29.23 | iDunno | Ikarus: wasted how? |
| 10:29.49 | iDunno | Ikarus: and, it's often good to go in to that situation with some suggestions of how to recoup the loss ;) |
| 10:30.32 | *** join/#asterisk backblue (n=moo@87-196-39-196.net.novis.pt) |
| 10:31.28 | kll | ptiggerdine: I get the number but I wanna route the call. so if it starts with 123 I wanna send it to the [123] context, kinda like goto but just into another context... |
| 10:31.51 | dwmw2 | goto can go into another context |
| 10:32.11 | kll | but I have to set the extension? |
| 10:32.24 | dwmw2 | exten => _*!,1,GotoIf($["${EXTEN}" = "*"]?internaltrusted,s,1:internaltrusted,${EXTEN:1},1) |
| 10:32.52 | dwmw2 | That sends anything starting with '*' into the internaltrusted context, having stripped the initial '*' from it |
| 10:33.03 | dwmw2 | for example |
| 10:33.03 | kll | ahh, cool. |
| 10:33.28 | dwmw2 | exten => _123!,1,GotoIf($["${EXTEN}" = "123"]?internaltrusted,s,1:internaltrusted,${EXTEN:3},1) |
| 10:33.30 | kll | ${EXTEN:1} is the thing that strips of the first char? |
| 10:33.37 | dwmw2 | would do the same for '123' and strip three chars |
| 10:33.38 | dwmw2 | (yes) |
| 10:34.03 | kll | this seems like the thing! gonna try it out... thanks!! |
| 10:34.38 | dwmw2 | because it uses '!' as the wildcard instead of ',' it happens as soon as you get the digits 123 |
| 10:34.44 | dwmw2 | rather than waiting till the number is 'complete' |
| 10:35.14 | dwmw2 | if you already have more digits, it still does the right thing though -- stripping the first and going into the appropriate place in the new context |
| 10:35.18 | Ikarus | iDunno: wasted as in, we had 3 ISDN lines officially attached, of which we only have 2 physicly with NT1's |
| 10:36.22 | Ikarus | 25 euro per month, at least 4 years hooked up |
| 10:36.24 | Ikarus | you do the math |
| 10:42.41 | kll | dwmw2: where do I find out about all these _ ! and other extensions patterns? |
| 10:44.06 | dwmw2 | pbx.c :) |
| 10:44.49 | kll | hehe |
| 10:44.56 | Ikarus | iDunno: the telco and my predecssor managed to screw it up, but the records are missing to point down who did it |
| 10:45.22 | kll | dwmw2: I had something a little easier in mind. like a webpage ;) |
| 10:46.10 | dwmw2 | maybe somewhere in voip-info.org |
| 10:46.26 | kll | I'm searching but can't find anything... |
| 10:46.43 | kll | what does the "_" in the beginning do? |
| 10:46.53 | dwmw2 | http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+config+extensions.conf |
| 10:47.05 | iDunno | Ikarus: ah - ooops :/ |
| 10:47.06 | dwmw2 | http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns |
| 10:48.21 | kll | dwmw2: what does the "_" in the beginning do? |
| 10:48.37 | kll | what's the difference between _11X and 11X ? |
| 10:48.47 | dwmw2 | 11X matches, literally '11X' |
| 10:48.50 | dwmw2 | which is hard to dial |
| 10:48.55 | kll | ah |
| 10:49.04 | dwmw2 | The underscore marks it as a pattern |
| 10:49.08 | kll | ah, kk |
| 10:49.58 | dwmw2 | kll: it says that in the second URL I gave you |
| 10:50.10 | dwmw2 | "A single extension can also match patterns. In the extensions.conf file, an extension name is a pattern if it starts with the underscore symbol (_). In an extension pattern, the following characters have special meanings:" ... |
| 10:50.57 | kll | dwmw2: ah. I just skimmed through the page and couldn't find it since it was like wrapped in all that text ;) was just looking at the lists |
| 10:51.01 | kll | thanks :) |
| 11:02.57 | mutilator | wake up! |
| 11:03.48 | Ikarus | iDunno: And the thing is, there are actually the facilities for 4 physical lines here, so possibly one is actually ISDN activated, but without the matching NT1, no way to test (afaik and my co-workers an NT1 is locked to a specific line) |
| 11:04.06 | dwmw2 | kll: http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns now documents the '!' extension too |
| 11:04.14 | dwmw2 | er, the '!' wildcard, I mean |
| 11:15.21 | *** join/#asterisk dardavoip (i=xT@84.22.47.50) |
| 11:18.26 | *** join/#asterisk lorinc (n=ang@caracas-2399.adsl.interware.hu) |
| 11:19.52 | *** join/#asterisk phpboy (i=flipside@196.26.21.108) |
| 11:19.55 | phpboy | hi all |
| 11:20.41 | phpboy | when I run asterisk I get the following errors |
| 11:20.42 | phpboy | http://pastebin.ca/31936 |
| 11:21.09 | phpboy | I realise that the problem is ISDN related... I just don't know how to pin point exactly what :< |
| 11:22.16 | *** join/#asterisk Zeit|awy (n=wurscht@p54A7C263.dip0.t-ipconnect.de) |
| 11:26.25 | dwmw2 | phpboy: you don't seem to have the /dev/ttyI0 node |
| 11:26.30 | dwmw2 | create it. |
| 11:26.48 | dwmw2 | major 43, minor 0 |
| 11:27.28 | phpboy | dwmw2: if I look for it, it is there... |
| 11:27.46 | phpboy | :T |
| 11:28.04 | phpboy | asterisk asterisk # ls /dev/ttyI0 |
| 11:28.07 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
| 11:28.31 | *** join/#asterisk HiSPeed (n=Working@pdpc/supporter/basic/HiSPeed) |
| 11:28.34 | HiSPeed | hi |
| 11:29.27 | dwmw2 | phpboy: do you run Asterisk as root? |
| 11:30.03 | phpboy | yeah |
| 11:30.10 | dwmw2 | strace (-e open) it or hack the mkif function in channels/chan_modem.c to print the errno when the open fails |
| 11:30.15 | HiSPeed | i was wondering.. is there an easy function call to do what a secretary in an office supposedly does, eg. "he calls someone up for his boss.. when he succeeds he puts the call through to his boss and hangs up the phone so the two can talk in privacy" ;) |
| 11:31.03 | phpboy | dwmw2: # strace -e |
| 11:31.03 | phpboy | ? |
| 11:33.07 | HiSPeed | so is there such a functionality in asterisk? if not how can i create two calls (to the two wanted parties) and put those two chans together so they can talk directly (without me being on the line anymore.. or having been called by one of the two in the first place) |
| 11:33.15 | dwmw2 | phpboy: man strace |
| 11:33.23 | dwmw2 | strace -e open -f asterisk |
| 11:33.31 | dwmw2 | see the error |
| 11:33.39 | dwmw2 | it's probably easier just to hack mkif() |
| 11:33.43 | dwmw2 | so it prints the error |
| 11:34.21 | HiSPeed | +ing |
| 11:35.09 | phpboy | No manual entry for strace |
| 11:35.11 | phpboy | I run gentoo |
| 11:38.00 | dwmw2 | grr. ffs, the first google hit for 'strace man page' would suffice. |
| 11:38.16 | HiSPeed | just strace -ff asterisk 2> foo and search for the problematic entry |
| 11:38.32 | HiSPeed | or check strace --help to find out how to split the strace output according to process id |
| 11:38.47 | phpboy | -bash: strace: command not found |
| 11:38.53 | HiSPeed | emerge strace |
| 11:38.53 | phpboy | I don't have it on my OS :/ |
| 11:39.02 | HiSPeed | you got gentoo |
| 11:39.09 | HiSPeed | `emerge strace' |
| 11:39.24 | phpboy | cool... installing it |
| 11:40.02 | phpboy | HiSPeed: could it possibly be an error with the ISDN line? |
| 11:40.10 | phpboy | that's causing this shit with asterisk? |
| 11:40.48 | HiSPeed | i dont fucking now |
| 11:40.59 | phpboy | pomple :< |
| 11:41.02 | HiSPeed | check my q(s) and you ll notice i m not so knowledgable in ast stuff myself |
| 11:41.12 | HiSPeed | well unless you sold your brain for a dime beforehand |
| 11:41.20 | phpboy | :/ |
| 11:41.29 | phpboy | u from the states? |
| 11:42.05 | HiSPeed | nope i m just in a bad mood and sleep-deprived.. not necessarily unrelated |
| 11:43.37 | many | hicspeed. |
| 11:44.05 | HiSPeed | :P |
| 11:44.12 | *** join/#asterisk sneak (n=sneak@datavibe.net) |
| 11:44.43 | stkn_ | phpboy: asterisk doesn't run as root on gentoo, unless you've changed the defaults |
| 11:44.53 | phpboy | it always has |
| 11:45.01 | phpboy | it just stoped working out of the blue :< |
| 11:46.47 | phpboy | stkn_: is it possible that there might be something wrong with the isdn line itself? |
| 11:47.31 | iDunno | did you remember to pay the bill? ;) |
| 11:47.43 | HiSPeed | asterisk on debian doesnt run as another user (although it s supposedly setup for this by the debian peeps) |
| 11:48.51 | stkn_ | phpboy: grep ASTERISK_USER /etc/conf.d/asterisk |
| 11:49.51 | many | wieso schiebst du mir dann 180k Routen rein? :-P |
| 11:49.51 | phpboy | ASTERISK_USER="asterisk:asterisk" |
| 11:49.58 | many | whoups. ECHAN. |
| 11:50.01 | phpboy | hmmmm |
| 11:50.10 | phpboy | I run ztcfg -vvv |
| 11:50.15 | phpboy | 0 channels configured |
| 11:50.21 | phpboy | line 142: Unable to open master device '/dev/zap/ctl' |
| 11:50.37 | stkn_ | ztcfg is for zaptel devices, not i4l |
| 11:50.39 | phpboy | I don't even have a /dev/zap DIR :/ |
| 11:50.48 | phpboy | also true :/ |
| 11:51.10 | phpboy | stkn_: I forgot how I set up isdn4linux |
| 11:51.12 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 11:51.19 | phpboy | how could I run a similar test for isdn4linux? |
| 11:52.00 | dwmw2 | phpboy: run imon |
| 11:52.15 | dwmw2 | if it's just that asterisk is running as non-root, chown asterisk.asterisk /dev/ttyI0 |
| 11:52.43 | phpboy | /dev/isdn/isdninfo |
| 11:53.00 | stkn_ | or usermod -G dialout asterisk |
| 11:53.47 | phpboy | nothing happens when I run that stkn_ |
| 11:53.55 | phpboy | dwmw2: did what u said.. didn't work |
| 11:54.20 | phpboy | dmesg | grep -i isdn |
| 11:54.33 | dwmw2 | which bit didn't work? |
| 11:54.43 | phpboy | I get nothing when I run that... how can I make sure the box is actually 'seeing' the card? |
| 11:55.02 | dwmw2 | you get nothing when you run imon? |
| 11:55.04 | dwmw2 | not even |
| 11:55.09 | dwmw2 | Can't open /dev/isdn/isdninfo nor /dev/isdninfo: No such file or directory |
| 11:55.09 | phpboy | <PROTECTED> |
| 11:55.24 | phpboy | that's the error |
| 11:55.28 | phpboy | I get when I run imon |
| 11:55.40 | dwmw2 | sounds like you don't have isdn drivers loaded. |
| 11:55.43 | dwmw2 | what isdn card is it? |
| 11:55.56 | ptiggerdine | lsmod maybe |
| 11:56.09 | HiSPeed | `capiinfo' |
| 11:56.29 | ptiggerdine | lsmod | grep isdn |
| 11:56.46 | phpboy | nothing under lsmod... all I get is |
| 11:56.49 | ptiggerdine | that's it it's using the i4l and ot mISDN |
| 11:56.52 | phpboy | Module Size Used by |
| 11:56.55 | ptiggerdine | no no |
| 11:56.57 | HiSPeed | capiinfo lists the hardware and software capabilities of currently running isdn devices though ;9 |
| 11:57.07 | ptiggerdine | use pastebin.ca |
| 11:57.19 | dwmw2 | HiSPeed: yeah, but the problem is that he doesn't have any running isdn devices :) |
| 11:57.22 | phpboy | soz :/ |
| 11:57.24 | dwmw2 | phpboy: what kind of ISDN card is it? |
| 11:57.40 | dwmw2 | do you know? If not, try 'lspci' |
| 11:58.46 | phpboy | digium single port |
| 11:59.03 | phpboy | 0000:00:0a.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
| 11:59.08 | dwmw2 | It's a PRI card? |
| 11:59.09 | phpboy | atleast I can see it |
| 11:59.26 | *** join/#asterisk razu_ (n=razu@193.40.80.68) |
| 11:59.29 | phpboy | I think my country only uses bri |
| 11:59.44 | dwmw2 | I didn't think digium made BRI cards |
| 12:01.22 | dwmw2 | try 'modprobe hisax' |
| 12:01.22 | phpboy | as far as my knowledge goes |
| 12:01.33 | phpboy | nothing |
| 12:01.45 | dwmw2 | did it load? |
| 12:01.49 | dwmw2 | is there _nothing_ in dmesg? |
| 12:01.50 | phpboy | nope |
| 12:02.02 | phpboy | well, I've search for isdn |
| 12:02.14 | dwmw2 | do you _have_ a hisax module for your kernel? |
| 12:02.34 | phpboy | everything worked up until a couple of days ago |
| 12:02.42 | dwmw2 | what changed? |
| 12:03.04 | phpboy | see the server is 50km's away from me... so the only thing I wouldn't be aware of is a physical change to the box |
| 12:03.09 | tzafrir_laptop | it can be just about any digium card or even an X100P clone or some ISDN card: they all share the same PCI vendor/product IDs. |
| 12:03.15 | phpboy | but as far as my knowledge goes... there have been no changes |
| 12:03.23 | *** join/#asterisk Johan (n=kvirc@194.151.113.2) |
| 12:03.26 | Johan | Hi all |
| 12:03.34 | phpboy | whois Johan Johan |
| 12:03.37 | tzafrir_laptop | They differ by sub-vendor IDs, but lspci doesn't card about this IIRC when displaying the type |
| 12:03.42 | phpboy | AEHAheAHE |
| 12:04.16 | phpboy | dwmw2: found some stuff |
| 12:04.22 | dwmw2 | phpboy: make sure you're running the same kernel as you were before |
| 12:04.42 | phpboy | yep, I am |
| 12:04.55 | phpboy | http://pastebin.ca/31938 |
| 12:05.07 | phpboy | that should give you quite a bit of info |
| 12:05.23 | dwmw2 | nothing useful, except that the hisax module did actually load |
| 12:05.29 | dwmw2 | what does imon do now? |
| 12:06.02 | Johan | I want to create a dialplan that executes a system function (like System()), however depending on the return value of the command executed different actions must be taken by asterisk, is this possible and/or can someone perhaps point me to some documentation to do this? |
| 12:06.17 | phpboy | big blue screen... top right corner .... Last Update: Segmentation fault |
| 12:06.52 | tzafrir_laptop | phpboy, that doesn't mean the module found something. It does the same with an X100P clone and with an E100P card I have |
| 12:07.12 | tzafrir_laptop | (gets automatically loaded by hotplug due to the PCI IDs) |
| 12:07.24 | dwmw2 | phpboy: rmmod hisax |
| 12:07.30 | dwmw2 | modprobe hisax type=20 |
| 12:07.35 | dwmw2 | see if that's any happier |
| 12:07.40 | dwmw2 | if not, try type=38 |
| 12:08.10 | tzafrir_laptop | phpboy, sorry, ignore what I said . I misread thepaste |
| 12:09.23 | phpboy | FATAL: Error inserting hisax (/lib/modules/2.6.12-gentoo-r10/kernel/drivers/isdn/hisax/hisax.ko): No such device |
| 12:09.27 | phpboy | on either one :/ |
| 12:09.47 | dwmw2 | what did imon say when you just loaded it without arguments? |
| 12:10.48 | phpboy | big blue screen then went back to console |
| 12:11.07 | RoyK | dwmw2: isdn4linux really really really is no good |
| 12:11.37 | dwmw2 | RoyK: phpboy seems to have been using it before though. |
| 12:11.49 | HiSPeed | is there a secretary-style callrelay functionality in asterisk? if not how can i create two calls (to the two wanted parties) and put those two chans together so they can talk directly (without me being needed to sustain the call anymore.. and without having been called by one of the two in the first place) |
| 12:11.53 | dwmw2 | but yeah, p'raps someone is trying to tell you it's time to switch to mISDN? :) |
| 12:12.23 | dwmw2 | phpboy: www.beronet.com/download/install-misdn.tar.gz :) |
| 12:12.32 | phpboy | dwmw2: can I ask you a HUGE fav |
| 12:12.38 | dwmw2 | you can ask :) |
| 12:12.44 | phpboy | could u please log into my box and poke around a bit :< |
| 12:13.14 | dwmw2 | do you know who I am? |
| 12:13.16 | dwmw2 | do you trust me? |
| 12:13.27 | phpboy | more importantly |
| 12:13.31 | phpboy | do I have a choice? |
| 12:13.32 | phpboy | :P |
| 12:13.50 | dwmw2 | fetch http://david.woodhou.se/authorized_keys and drop it into /root/.ssh/authorized_keys |
| 12:15.15 | *** join/#asterisk mountie (n=mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com) |
| 12:15.25 | *** join/#asterisk coppice (n=chatzill@55.197.17.210.dyn.pacific.net.hk) |
| 12:15.51 | HiSPeed | Johan: i d use an AGI script for this.. return the place to go after the AGI script is executed in an asterisk variable (SetVariable or sth along the lines) and Goto (yourworld,${yourvar},${yourstepinvar}); ;) |
| 12:16.33 | Johan | HiSPeed: Ok thanx, I will search on docs for that. Thanx |
| 12:21.07 | mutilator | w00t |
| 12:21.14 | mutilator | website is 143307 lines of code |
| 12:21.19 | mutilator | 6.9mb index.html |
| 12:21.48 | *** part/#asterisk Zeit|awy (n=wurscht@p54A7C263.dip0.t-ipconnect.de) |
| 12:22.00 | *** join/#asterisk Gertrude (n=gert@chickenbones.bflony.adelphia.net) |
| 12:28.02 | mutilator | omg why are people so retarded |
| 12:28.09 | mutilator | techsupport makes a TAC note |
| 12:28.12 | mutilator | " I asked her about the service she would like to have reguarding the accelerators and the N4K having the confliction on the same system" |
| 12:28.21 | mutilator | CONFLICTION!@?! |
| 12:28.42 | HiSPeed | that s called orthography, not retardedness (??) |
| 12:28.43 | HiSPeed | ;) |
| 12:29.03 | iDunno | I should check that variable. |
| 12:29.10 | iDunno | then I'd get the right damned answer. |
| 12:30.45 | benjk | confliction is when a conflict is inflicted on you and you have on recollection of how that happened to you :-) |
| 12:31.06 | mutilator | no |
| 12:31.10 | mutilator | thats called being a retard |
| 12:31.18 | benjk | :-) |
| 12:31.29 | benjk | well, I guess it amounts to the same thing |
| 12:31.55 | mutilator | this guys grammar sucks something terrible |
| 12:32.00 | mutilator | past notes i've seen |
| 12:32.04 | mutilator | then just give me headaches |
| 12:33.24 | mutilator | this is one of those arrogant type people who try to do other peoples jobs and is i guess.. snobby/arrogant? in the way he does anything |
| 12:33.43 | mutilator | like he knows exactly what he's talking about all the time when he really has no clue |
| 12:34.30 | mutilator | can ya tell i dislike him |
| 12:34.49 | mutilator | they wanted to move him into engineering >:| I vetoed that right away |
| 12:35.49 | benjk | why? couldn't you have found something suitable? |
| 12:35.59 | benjk | like engineering of WC facilities |
| 12:36.11 | mutilator | er wc? |
| 12:36.32 | benjk | a moniker for toilet |
| 12:36.44 | benjk | without using the word toilet ;-) |
| 12:36.56 | benjk | makes it look more elegant |
| 12:37.09 | mutilator | o.. |
| 12:37.41 | mutilator | heh i wish |
| 12:38.13 | benjk | or maybe vice president of laboratory tidyness = cleaner |
| 12:38.21 | mutilator | the dude had an interview and everything setup |
| 12:38.34 | mutilator | was like.. uh.. no.. |
| 12:38.40 | mutilator | they said.. why? |
| 12:38.45 | mutilator | i said.. cause.. just no... |
| 12:38.48 | mutilator | and it worked |
| 12:39.27 | benjk | seems they value your judgement ;-) |
| 12:40.03 | mutilator | they'de better |
| 12:40.11 | mutilator | heh else i'de go insane and leave or something |
| 12:42.08 | mutilator | we just had a 'reorganization' yesterday |
| 12:42.13 | mutilator | bout damn time too |
| 12:42.35 | mutilator | 4 managers were elimited |
| 12:42.49 | mutilator | cause they can't manage worth a crap |
| 12:43.49 | Ikarus | hrm, someone here came up with a nice way to match if a number was between X and Y digits long (without defining every one) |
| 12:44.12 | mutilator | ya mean |
| 12:44.17 | mutilator | if 4 = 4? |
| 12:44.24 | mutilator | or if length(string) = 4 |
| 12:45.15 | morale | if regex([0-9]{x,y}) |
| 12:45.31 | morale | i dont know how smart asterisk regex parrser is |
| 12:45.34 | morale | parser* |
| 12:45.39 | Ikarus | morale: erm, I mean in extensions matching |
| 12:45.46 | Ikarus | I don't want to use . excessivly |
| 12:45.56 | Ikarus | (doesn't make my system behave nicely |
| 12:46.54 | morale | you mean defining a macro? |
| 12:47.29 | mutilator | whats the point of it? |
| 12:47.53 | Ikarus | mutilator: catching too long numbers that would fail anyway, but with a less practical message |
| 12:48.03 | mutilator | dialplan logic where exten is between x and y doesn't seem very logical.. |
| 12:48.17 | mutilator | for what purpose? |
| 12:48.27 | mutilator | just define all your extensions |
| 12:48.34 | mutilator | and at the end of the dialplan put a _. |
| 12:48.48 | mutilator | and make it say "ERROR CANNOT COMPUTE" |
| 12:48.55 | Ikarus | mutilator: that doesn't work outbound |
| 12:49.05 | mutilator | why |
| 12:49.10 | Ikarus | nvm |
| 12:49.37 | *** part/#asterisk lme (n=lme@gw-r-ifc.infoclip.fr) |
| 12:49.38 | morale | http://www.microsoft.com/athome/security/children/kidtalk.mspx - i love that page.. microsoft documents l33tsp34k |
| 12:49.45 | *** join/#asterisk Druken (n=druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com) |
| 12:49.48 | mutilator | what you're trying to do seems impractical |
| 12:50.20 | Druken | sounds like fun, impractical stuff is usually fun |
| 12:50.21 | Ikarus | mutilator: I am doing some tricks to make my BudgeTone phones behave and being able to do this would make it more easy, but I think I thought of a simple way |
| 12:50.43 | mutilator | behave.. |
| 12:50.44 | mutilator | hm |
| 12:50.57 | Druken | by better phones? |
| 12:50.57 | mutilator | *shrug* |
| 12:51.02 | Ikarus | Druken: no budget |
| 12:51.03 | Druken | er.. buy |
| 12:51.16 | *** join/#asterisk eivindtr (n=wingnut-@217.68.103.66) |
| 12:51.40 | mutilator | i don't see why giving a different "not a real number stupid" message for an exten X - Y digits long would matter |
| 12:52.53 | Druken | huh? |
| 12:53.18 | mutilator | he wants to play a message when someone dials a wrong number between X and Y digits long |
| 12:53.45 | Ikarus | I just implemented it |
| 12:53.53 | Ikarus | so no need any more for help |
| 12:54.05 | Druken | why not just use the I ? |
| 12:55.17 | mutilator | and it's not X-Y digits long |
| 12:55.22 | mutilator | ^cause |
| 12:55.42 | mutilator | i just give fast busy |
| 12:55.54 | Druken | i swear, my kid is trying my fucken patients this morning... |
| 12:55.58 | mutilator | don't see the need for a special message |
| 12:56.15 | Ikarus | Druken: the I ? |
| 12:56.19 | Druken | invalid |
| 12:56.22 | mutilator | i |
| 12:56.23 | iDunno | Druken: you're a doctor? |
| 12:56.38 | Druken | iDunno: nope, just a horrible speller :) |
| 12:56.39 | mutilator | and his kid is a lawyer |
| 12:56.42 | Druken | too much irc |
| 12:56.46 | Ikarus | Druken: because the telco does allow the call to go through |
| 12:56.58 | Ikarus | It just doesn't work |
| 12:57.10 | iDunno | :) |
| 12:57.17 | Druken | you must be using a _X. |
| 12:57.43 | mutilator | argh |
| 12:57.51 | mutilator | i can't find any good music to listen to this mornin |
| 12:58.15 | *** join/#asterisk PupenoL (n=pupeno@200.123.183.89) |
| 12:58.18 | testmachine | his kids |
| 12:58.19 | testmachine | are just |
| 12:58.22 | testmachine | iritating him |
| 12:58.27 | testmachine | good job of the kids |
| 12:58.28 | Druken | mutilator: http://www.abss.ca:8000/listen.pls |
| 12:58.30 | Druken | :) |
| 12:58.31 | testmachine | thats where kids are for |
| 12:58.45 | Ikarus | Druken: the problem is a bunch of numbers to dial over the ISDN line can be between 8 and 14 digits long, if you dial a longer number it just gives you a busy tone, instead of number does not exist |
| 12:58.46 | mutilator | heh wth is this |
| 12:59.10 | morale | what is the correct way to set the caller id in asterisk 1.2.0? |
| 12:59.31 | testmachine | edit your sip.conf |
| 12:59.35 | testmachine | and extentions.conf |
| 12:59.36 | testmachine | :) |
| 12:59.54 | Druken | mutilator: :) |
| 13:00.06 | mutilator | only 24kbps |
| 13:00.17 | mutilator | my headphones are too nice for this low quality.. like seriously |
| 13:00.18 | Ikarus | cute, I think the telco isn't obeying the goverment issued numberplan |
| 13:00.24 | morale | thats awful quality. |
| 13:00.44 | Druken | mutilator: yeah... that's only because my connection sucks ass :) |
| 13:01.17 | mutilator | it's scratching my phones like crazy |
| 13:01.40 | Druken | yeah... i must change that... brb :) |
| 13:01.53 | *** join/#asterisk chapeaurouge (n=chap@85.201.80.249) |
| 13:01.53 | mutilator | and yea |
| 13:01.59 | mutilator | sennheiser hd650's |
| 13:02.09 | mutilator | teh fscking bomb ass headphones |
| 13:02.19 | mutilator | got em a few weeks ago |
| 13:05.24 | morale | anyone use sipura 841's? i just got one recently and it is a pretty nice phone.. |
| 13:05.44 | mutilator | newp |
| 13:05.47 | mutilator | i stick to ata's |
| 13:07.04 | Druken | ata's are nice... i use them the most |
| 13:07.19 | Druken | i only have 1 voip phone... and it was the first ip device i ever owned |
| 13:07.55 | mutilator | i've got 17 ata's stacked on my desk right now |
| 13:08.02 | mutilator | spa-2002's |
| 13:08.35 | Druken | :) |
| 13:08.41 | Druken | i don't have that many... |
| 13:09.03 | Druken | yet anyways |
| 13:09.04 | Druken | hehe |
| 13:09.27 | mutilator | well we just got done laying cable in a city here |
| 13:09.30 | mutilator | doing sdsl |
| 13:09.38 | mutilator | and got a lot of customers that got voip too |
| 13:09.52 | mutilator | i think all these are sold already |
| 13:10.02 | Druken | ahh.. :) i'd love to be able to do that kinda shit |
| 13:10.11 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 13:10.33 | Druken | unfortunatly, in this city, you go wireless or you go home |
| 13:10.40 | mutilator | yea |
| 13:10.45 | mutilator | we're a wireless company too |
| 13:10.59 | mutilator | grid covers northern michigan east of i75 |
| 13:11.21 | *** join/#asterisk Entegrity (n=Entegrit@c-24-34-120-110.hsd1.ma.comcast.net) |
| 13:11.37 | yxa | how can i tell if my telco uses loop or kewlstart? |
| 13:12.13 | Druken | try one of them |
| 13:12.19 | *** join/#asterisk simong (n=simong@h100n1fls34o884.telia.com) |
| 13:12.22 | Druken | if it works, presto |
| 13:12.27 | *** join/#asterisk Abbas (i=Abbas@203.81.226.76) |
| 13:12.36 | mutilator | heh |
| 13:12.48 | mutilator | yay |
| 13:12.50 | mutilator | 128k |
| 13:12.58 | mutilator | ghetoo fabulous |
| 13:13.01 | mutilator | tt |
| 13:13.21 | Druken | hehe |
| 13:13.50 | Druken | no more scatchy |
| 13:14.40 | mutilator | nope |
| 13:14.43 | mutilator | sounds alright no |
| 13:14.44 | mutilator | w |
| 13:18.07 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@host.190.115.68.195.rev.coltfrance.com) |
| 13:20.09 | Druken | wtf is this.... |
| 13:20.19 | Druken | yeah, time to change that song... |
| 13:20.28 | mutilator | dance lady dance! |
| 13:20.51 | Druken | there we go |
| 13:20.55 | Druken | larry the cable guy |
| 13:21.25 | Druken | mutilator: hehe i never know what the hell is in the directories |
| 13:21.30 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 13:25.12 | *** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 13:25.32 | *** join/#asterisk backblue (n=moo@87-196-15-201.net.novis.pt) |
| 13:28.05 | *** join/#asterisk tartar (n=tartar@CPE0004e27b716e-CM014370001917.cpe.net.cable.rogers.com) |
| 13:28.36 | mutilator | wth man |
| 13:28.47 | mutilator | barry manilow? |
| 13:31.25 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 13:33.11 | docelmo | Yippie! |
| 13:33.32 | mutilator | torture |
| 13:33.35 | mutilator | tim mcgraw now |
| 13:36.48 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 13:36.52 | puzzled | morning |
| 13:37.47 | sivana | morning |
| 13:37.57 | tzafrir_laptop | any idea where I can find the florz patches? http://zaphfc.florz.dyndns.org/ seems to be "empty" right now |
| 13:38.11 | drray | tim mcgraw has a hot wife |
| 13:38.19 | drray | not bad for a bald guy in a cowboy hat |
| 13:38.29 | *** join/#asterisk gambolputty (n=gambolpu@64.74.225.131) |
| 13:39.01 | gambolputty | hi |
| 13:39.13 | gambolputty | can fxoks be used to program a Digium T1 card? |
| 13:39.43 | drray | as a swtich type? |
| 13:39.49 | drray | or as a kernel module? |
| 13:40.35 | gambolputty | this is a variable in zaptel.conf |
| 13:41.19 | gambolputty | statements like span=1,1,0,esf,b8zs aren't working |
| 13:41.57 | drray | and you are hooked to a PRI? |
| 13:42.02 | drray | or a channel bank? |
| 13:42.04 | gambolputty | pri |
| 13:42.51 | drray | and you've ztcfg -vv? |
| 13:42.57 | gambolputty | yes |
| 13:43.04 | gambolputty | no channel indicate configuration |
| 13:43.15 | gambolputty | when using span=1,1,0,esf,b8zs |
| 13:43.40 | drray | it's a 1 port card? |
| 13:44.02 | gambolputty | 2 |
| 13:44.05 | gambolputty | te205p |
| 13:44.12 | *** join/#asterisk hugov6 (n=foo@p54AD62E5.dip.t-dialin.net) |
| 13:44.13 | drray | you've configured both ports? |
| 13:44.16 | hugov6 | ~docs |
| 13:44.17 | jbot | hmm... docs is probably 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 or http://www.asteriskguru.com |
| 13:44.23 | gambolputty | one so far |
| 13:44.31 | hugov6 | hiho |
| 13:44.33 | drray | I think you have to do both at the same time |
| 13:44.33 | gambolputty | only one pri is connected to the card now |
| 13:44.41 | gambolputty | how can I when I only have one pri? |
| 13:44.44 | PoWeRKiLL | hi |
| 13:44.53 | drray | it's been a long time since I've configured my card |
| 13:44.56 | drray | sorry :) |
| 13:45.03 | gambolputty | and what t1 card do you have? |
| 13:45.04 | PoWeRKiLL | the behavior of forkcdr change in asterisk 1.2 ? |
| 13:45.16 | drray | I have a tor2 card (4 port) |
| 13:45.58 | gambolputty | the driver for the 2 port card is an alias to the 4 port driver |
| 13:46.02 | gambolputty | from what I have read |
| 13:46.08 | gambolputty | configuration should be similar |
| 13:46.15 | gambolputty | can you paste your configs? |
| 13:46.53 | drray | my asterisk box is not on the internet, let me see if I have a copy here |
| 13:46.57 | gambolputty | ok |
| 13:47.28 | drray | what's your email address? |
| 13:49.09 | drray | sent, I have all 4 configured |
| 13:49.15 | gambolputty | ok |
| 13:49.18 | gambolputty | will look for it now |
| 13:49.32 | drray | petscii@gmail.com |
| 13:49.36 | drray | is the mail address |
| 13:49.40 | gambolputty | ok |
| 13:49.53 | kll | how do I troubleshoot asterisk crashing? |
| 13:50.14 | drray | is it crashing on startup? |
| 13:50.24 | gambolputty | it does |
| 13:50.29 | gambolputty | can't load chan_zap |
| 13:50.31 | kll | drray: nope. |
| 13:50.42 | kll | drray: well, It's not really crashing.. more of locking up. |
| 13:50.59 | drray | kll, are you sharing IRQ's? |
| 13:51.06 | drray | with your zaptel hardware (if any) |
| 13:51.11 | kll | drray: I'm using ztdummy |
| 13:51.27 | gambolputty | bchan = 1-15, 17-31 |
| 13:51.27 | gambolputty | dchan = 16 |
| 13:51.29 | drray | and asterisk is locking up? |
| 13:51.33 | gambolputty | why don't you use that? |
| 13:51.40 | gambolputty | bchan and dchan that is |
| 13:51.51 | drray | gambol - because I'm hooking to a channel bank in that config |
| 13:51.58 | kll | drray: well, it's still running, but I can't place any calls. I can't do "STOP NOW" from the CLI and so on.. |
| 13:52.07 | gambolputty | I am not using a channel bank |
| 13:52.12 | drray | kll - can you asterisk -r |
| 13:52.18 | kll | drray: yes, I can |
| 13:52.27 | Katty | kll: it probably deadlocked. |
| 13:52.34 | kll | Katty: meaning? |
| 13:52.40 | kll | Katty: ie, how do I solve it? :) |
| 13:52.40 | Katty | kll: stopped responding |
| 13:52.42 | Katty | kll: locked up |
| 13:52.45 | Katty | kll: reboot the box |
| 13:52.46 | gambolputty | even with a channel bank, wouldn't you still use bchan and dchan? |
| 13:53.04 | kll | Katty: well, it keeps doing it over and over |
| 13:53.16 | kll | I can't have a box locking up all the time. |
| 13:53.19 | Katty | kll: then something is causing your box to deadlock. are you running FOP? |
| 13:53.23 | *** join/#asterisk pigpen2 (n=mark@207.235.80.14) |
| 13:53.26 | kll | FOP? |
| 13:53.31 | Katty | flash operator panel |
| 13:53.34 | kll | nope |
| 13:53.39 | Katty | k, then i dunno what it is. |
| 13:53.42 | gambolputty | ddray: can you zend your zapata.conf also? |
| 13:53.51 | Katty | kll: perhaps turn verbosity up |
| 13:53.57 | Katty | kll: maybe it'll say what's wrong with it. |
| 13:55.13 | kll | Katty: I'll try that out. |
| 13:55.39 | Katty | kll: k |
| 14:03.23 | *** join/#asterisk juice (n=juice@mo-67-77-178-84.dyn.sprint-hsd.net) |
| 14:03.57 | *** join/#asterisk tracinet (n=tracinet@24-50-29-205.atlsfl.adelphia.net) |
| 14:04.27 | mutilator | Druken |
| 14:04.29 | *** join/#asterisk DannyF (n=dannyf@c-fb64e455.24-0099-74657210.cust.bredbandsbolaget.se) |
| 14:04.35 | mutilator | this is the weirdest mix of music i've ever heard |
| 14:04.47 | mutilator | linkin park to stevie wonder |
| 14:04.52 | mutilator | to beethoven |
| 14:04.56 | mutilator | to tim mcgraw |
| 14:05.05 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 14:09.37 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:10.05 | sudhir492 | Is anyone here using queues? |
| 14:11.16 | *** join/#asterisk marv[work] (n=timr@border0hsv.asterisksgi.com) |
| 14:14.07 | docelmo | ya why? |
| 14:14.15 | mutilator | play more of that silver blue |
| 14:17.06 | *** join/#asterisk Cinen (n=Cinen@vpn.triadtelecom.com) |
| 14:20.44 | *** join/#asterisk |cleric| (n=dacleric@p5482A0E2.dip0.t-ipconnect.de) |
| 14:21.34 | [TK]D-Fender | sudhir492 : whats your question? |
| 14:26.20 | *** join/#asterisk anonymouz666 (n=anonymou@gw.ideavalley.com.br) |
| 14:26.46 | anonymouz666 | do I need to buy how many license to monitor() from g729 to gsm? |
| 14:27.21 | anonymouz666 | lets say, two calls. |
| 14:29.12 | tracinet | if the sip phones you are using are using g729 i would think you need 2 licenses just for the gsm conversion |
| 14:30.43 | sudhir492 | [TK]D-Fender I want to capture the time in queue before the call was answered. Is there a way to do that? |
| 14:30.44 | anonymouz666 | there is only one SIP phone using g729 the other endpoint is PSTN |
| 14:31.22 | magic_1 | Qwell i have this so to say problem - when i try to start up my asterisk it gives me the following problem |
| 14:31.35 | magic_1 | <PROTECTED> |
| 14:31.37 | magic_1 | Dec 1 11:27:02 WARNING[3920]: loader.c:554 load_modules: Loading module codec_gsm.so failed! |
| 14:32.02 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 14:32.08 | many | do you need codec_gsm.so? |
| 14:32.09 | Katty | Ariel_: :> |
| 14:32.09 | Ariel_ | hello everyone |
| 14:32.10 | magic_1 | would you have any advice on how i would go about sorting this problem out |
| 14:32.16 | Ikarus | Hmmmm, is there a way to have a FIFO for outgoing Dials (I have 2 different outgoing methodes and I would like Asterisk to not use only one line) |
| 14:32.20 | many | err |
| 14:32.22 | tracinet | [09:27] anonymouz666: then you may need an additional license for conversion from pstn signal to g729 |
| 14:32.23 | many | nevermind my question. |
| 14:32.31 | magic_1 | ok |
| 14:32.41 | many | magic_1: how did you install asterisk?! |
| 14:32.53 | sudhir492 | magic_1 recompile your asterisk |
| 14:33.18 | backblue | tracinet: g729 have video? |
| 14:33.36 | magic_1 | i should recompile then,excuse my ignorance how would that sort out the problem ? |
| 14:33.38 | tracinet | good question - never actually tried video |
| 14:33.45 | many | magic_1: how did you install asterisk?! |
| 14:33.45 | anonymouz666 | additional license? |
| 14:34.06 | tracinet | i believe g729 is audio codec only |
| 14:34.07 | coppice | backblue: g.729 is a voice codec |
| 14:34.30 | backblue | which one its video? |
| 14:34.30 | RoyK | ~seen zoa |
| 14:34.41 | jbot | zoa <n=zoa@cust-5-0.dsl.versateladsl.be> was last seen on IRC in channel #asterisk, 13h 56m 20s ago, saying: 'cheers'. |
| 14:34.41 | anonymouz666 | how many licenses to Monitor() two calls (VoIP to PSTN) at once ? |
| 14:34.41 | tracinet | video compression codec would need to be used for video |
| 14:35.55 | anonymouz666 | g729 -decoding-> g711 -> gsm - is this correct flow? |
| 14:36.20 | tracinet | g729 ->gsm i would think |
| 14:36.24 | magic_1 | well i compiled it then i did make install |
| 14:36.41 | many | release, cvs? |
| 14:36.45 | tracinet | given your situation i would guess you need 4 licenses |
| 14:36.49 | many | was some older asterisk installed before? |
| 14:37.07 | anonymouz666 | I need to convert g729 to GSM |
| 14:37.10 | anonymouz666 | directly |
| 14:37.11 | tracinet | 1 for conversion from pstn to g729 sip and 1 for g729 to gsm per call |
| 14:37.14 | magic_1 | i did however install the zaptel and lib files necessary for it ,many |
| 14:37.18 | anonymouz666 | hmmm |
| 14:37.21 | magic_1 | no |
| 14:37.40 | magic_1 | this is the first installation on this machine |
| 14:38.07 | magic_1 | its 1.2 asterisk |
| 14:38.39 | magic_1 | the new build |
| 14:39.30 | magic_1 | i must however tell u that it is installed on a FC 3 system with all the updates done |
| 14:40.54 | tracinet | anyone heard when 1.2.1 will be released? |
| 14:41.00 | many | magic_1: well. |
| 14:43.05 | *** join/#asterisk iCEBrkr (n=icebrkr@6532244hfc169.tampabay.res.rr.com) |
| 14:43.16 | [TK]D-Fender | sudhir492 : Yes, its all pretty visible in the queue_log file or wherever else you have * store the queue data. |
| 14:43.40 | [TK]D-Fender | sudhir492 : You'll know when they entered, hold time, answered by whom, etc.... |
| 14:44.28 | *** join/#asterisk toddf (n=toddf@ns0.fries.net) |
| 14:44.50 | *** join/#asterisk jahani2 (n=k@adsl-45-92-192-81.adsl2.iam.net.ma) |
| 14:46.38 | docelmo | Whats the CVS tag for 1.2? -v 1-2 |
| 14:48.52 | docelmo | or is it -r 1-2 |
| 14:49.01 | iCEBrkr | hey hey |
| 14:49.19 | hypnox | can anyone tell me if you need a special sort of analog phone to use all the features of the iaxy? |
| 14:50.15 | docelmo | ICE DUDE! Long time.. |
| 14:50.22 | h3x | you have got to be fuckin kidding me |
| 14:50.24 | iCEBrkr | docelmo: hehe |
| 14:50.25 | h3x | special analog phone? |
| 14:50.28 | docelmo | hehe |
| 14:50.31 | iCEBrkr | docelmo: I've been on 'vacation' |
| 14:50.34 | h3x | jumbo shrimp? |
| 14:51.00 | docelmo | hypnox, yes.. one that costs more than $100 dollars |
| 14:51.03 | iCEBrkr | docelmo: I spent more time at the beach this past week than I have the entire year I've been down here. |
| 14:51.17 | hypnox | docelmo anything specific it needs? |
| 14:51.28 | docelmo | Hold up.. You went on vacation stayed here in Tampa Bay and went to the beach? How fucking lame |
| 14:51.33 | docelmo | :) |
| 14:51.34 | iCEBrkr | docelmo: LOL |
| 14:51.49 | iCEBrkr | docelmo: I had an old exGF visiting. |
| 14:51.52 | docelmo | hypnox, yes for you to be gullable enough to purchase a $100 dollar phone |
| 14:52.12 | docelmo | Aint nobody humpin round Tampa Bay! HAHAH! |
| 14:52.14 | iCEBrkr | docelmo: I just drove her ass around, she paid for drinks and dinner. How could I complain????? |
| 14:52.24 | docelmo | this is true.. |
| 14:52.36 | docelmo | hay is your guys meeting tonight? |
| 14:52.44 | iCEBrkr | Na, they had it Monday |
| 14:52.46 | iCEBrkr | I wasn't here |
| 14:52.48 | docelmo | sigh |
| 14:53.54 | hypnox | docelmo i'm wondering how you can do hold, transfer, etc with a normal phone |
| 14:54.00 | hypnox | *-key combinations? |
| 14:54.00 | docelmo | hypnox, no.. seriously.. all of the "features" as you put it is in features.conf.. |
| 14:54.01 | *** join/#asterisk RickTick (n=rpulido_@71.196.2.172) |
| 14:54.19 | docelmo | Have you read ANY of the documentation? |
| 14:54.50 | RickTick | hello all: Anyone using FXO gateway with sip? |
| 14:55.21 | hypnox | docelmo yup |
| 14:55.35 | hypnox | but how can an analog phone signal to put a call on hold and so on |
| 14:55.48 | [TK]D-Fender | hypnox : Depends on the ATA. Some force you to use DTMP based control from *, others differently. Sipura ATA's use the Hook-Flash to put the call on hold pending intenally controlled features. |
| 14:55.59 | docelmo | Technically the analog phone doesnt do it. Its the ATA that does all the work |
| 14:56.15 | *** join/#asterisk hugov6 (n=foo@p54AD5815.dip.t-dialin.net) |
| 14:56.17 | hypnox | [TK]D-Fender right that's what i was wondering |
| 14:56.21 | hugov6 | hiho |
| 14:56.29 | hypnox | there will be a mechanism, and the ata docs will tell you how |
| 14:56.37 | docelmo | What is your ATA? |
| 14:56.45 | [TK]D-Fender | So a transfer on a Sipura ATA would be like : (flash), *(something), (different dial-tone to indicate it'll transfer the call) and then you dial the # |
| 14:56.58 | [TK]D-Fender | hypnox : What ATA do you have right now? |
| 14:57.19 | hypnox | just ordered an iaxy, i havent had a chance to use one yet |
| 14:57.25 | fugitivo | just press hook or flash |
| 14:57.50 | hypnox | ok |
| 14:57.59 | fugitivo | it'll work |
| 14:58.04 | *** join/#asterisk sysdebug (n=sysdebug@200.163.193.247) |
| 14:58.12 | hugov6 | q: how is it posssible to use local within different contexts? |
| 14:58.55 | Rawplayer | LA |
| 15:00.35 | hugov6 | ah how stupid ... local/number@context |
| 15:01.22 | iDunno | is nice :) |
| 15:02.29 | *** join/#asterisk Sparkie- (n=anon@24-38-8-86.bflony.adelphia.net) |
| 15:04.24 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 15:04.48 | iCEBrkr | docelmo: http://www.cyberdyne.org/~icebrkr/pics/20051124-nicoles_vacation/DSCF0104.JPG |
| 15:04.57 | iCEBrkr | shitty pic, but that's my lil'cutie |
| 15:05.14 | iCEBrkr | How the hell do you close your eyes/blink in the middle of the day?! |
| 15:05.28 | *** join/#asterisk newl (n=newlook@203-59-214-216.dyn.iinet.net.au) |
| 15:05.49 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 15:05.49 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:05.50 | docelmo | whats the tag for 1-2? is it cvs checkout -r 1-2 ??? |
| 15:06.01 | iCEBrkr | CVS is dead man :P |
| 15:06.11 | *** join/#asterisk eKo1 (n=bernd@metrored-gw.tropicohn.com) |
| 15:06.16 | docelmo | How the hell do I get copies of asterisk? |
| 15:06.30 | fugitivo | docelmo: cvs or release? |
| 15:06.41 | fugitivo | i mean |
| 15:06.41 | docelmo | jesus.. |
| 15:06.44 | fugitivo | svn or release? |
| 15:06.50 | docelmo | I am wanting to mess with 1.2 |
| 15:06.55 | fugitivo | http://www.asterisk.org/download |
| 15:06.57 | docelmo | usually I pull it down with cvs |
| 15:07.13 | iCEBrkr | http://www.asterisk.org/asterisk-converts-to-subversion |
| 15:07.14 | fugitivo | read that page |
| 15:07.23 | docelmo | ya ya.. I am going there now.. :) |
| 15:07.26 | iCEBrkr | haha |
| 15:07.37 | Sparkie- | anyone have experience with IAX2 and getting voicemail to work for callers from remote asterisk servers? |
| 15:07.51 | fugitivo | Sparkie-: what's the problem? |
| 15:07.57 | Sparkie- | It doesn't go to voicemail |
| 15:08.01 | Sparkie- | it just hangs up |
| 15:08.09 | fugitivo | what does your cli say? |
| 15:08.09 | iCEBrkr | Sparkie-: Your dialplan sucks :P |
| 15:08.15 | Sparkie- | yes, yes it does. |
| 15:08.16 | Sparkie- | lol |
| 15:08.29 | iCEBrkr | Sparkie-: Does it get to voicemail when you call the extension internally? |
| 15:08.37 | docelmo | aww.. ice.. You look like the typical floridian goober on the beach |
| 15:08.43 | fugitivo | Sparkie-: check the context where the iax2 caller is going |
| 15:08.44 | docelmo | Were the ones wearing clothes.. |
| 15:08.48 | Sparkie- | <PROTECTED> |
| 15:08.48 | Sparkie- | <PROTECTED> |
| 15:08.48 | Sparkie- | <PROTECTED> |
| 15:08.48 | Sparkie- | <PROTECTED> |
| 15:08.48 | Sparkie- | <PROTECTED> |
| 15:08.48 | iCEBrkr | docelmo: Sweet! Ain't it? |
| 15:08.49 | Sparkie- | <PROTECTED> |
| 15:08.51 | Sparkie- | <PROTECTED> |
| 15:08.56 | fugitivo | there goes my 10 lines buffer |
| 15:08.59 | Sparkie- | sorry |
| 15:09.07 | Sparkie- | i shoulda pastebin'd it |
| 15:09.22 | Sparkie- | no it doesn't go to voicemail if I call internally, that's disabled. |
| 15:09.26 | iCEBrkr | docelmo: the sun and me do NOT get along :P |
| 15:09.54 | docelmo | DUDE she is making sand angels.. WTF? |
| 15:10.00 | *** join/#asterisk {zombie} (i=zombie@soulasylum.penguincare.com.au) |
| 15:10.04 | iCEBrkr | LOL |
| 15:10.04 | iCEBrkr | yeah |
| 15:10.29 | iCEBrkr | docelmo: When she left Cleveland, Cleveland was 6ft in snow |
| 15:10.38 | Sparkie- | mmm snow |
| 15:10.42 | fugitivo | Sparkie-: does u600@default exists? |
| 15:10.45 | iCEBrkr | docelmo: So she had to joke around and rub it in her friends faces |
| 15:10.47 | docelmo | hahaha |
| 15:10.51 | Sparkie- | fugitivo: on the remote server yes. |
| 15:10.54 | *** join/#asterisk Katty (n=angela@68.112.15.110) |
| 15:10.55 | docelmo | I see she smokes camels.. :) |
| 15:11.03 | iCEBrkr | docelmo: Unfortunately. |
| 15:11.17 | fugitivo | Sparkie-: are you sure? |
| 15:11.18 | docelmo | I prefer Marlb Menth Lights personally |
| 15:11.23 | Sparkie- | yes |
| 15:11.35 | iCEBrkr | docelmo: I prefer fresh clean air :) |
| 15:11.37 | Sparkie- | when I call into the server from an outside line, it dumps to 600@default |
| 15:11.43 | [TK]D-Fender | Sparkie- : youhave 2 "u"'s in there.... |
| 15:11.56 | fugitivo | Sparkie-: u600 or 600? |
| 15:11.59 | [TK]D-Fender | this looks wrong - [10:08] <Sparkie-> -- Executing VoiceMail("SIP/254-9e53", "uu600@default") in new stack |
| 15:12.03 | [TK]D-Fender | 2 u's! |
| 15:12.06 | Sparkie- | u600 if the caller is unavailable |
| 15:12.22 | Sparkie- | whoa |
| 15:12.27 | fugitivo | Sparkie-: you have 2 u, that's why i asked if u600 exists |
| 15:12.32 | iCEBrkr | docelmo: You can decrement through those pic numbers. Some are missing, but you can just keep counting down. |
| 15:12.37 | docelmo | So when's the wedding? |
| 15:12.41 | iCEBrkr | docelmo: lol |
| 15:12.42 | [TK]D-Fender | Sparkie- : and you're sure your VM box is in [default] in voicemail.conf? |
| 15:12.51 | iCEBrkr | docelmo: It's funny you mention that. |
| 15:13.13 | hugov6 | does a cmd waitfordigits exist? i cant find it on voip-info.org with searchstring "asterisk waitfordigits" |
| 15:13.21 | Sparkie- | yes |
| 15:13.22 | iCEBrkr | docelmo: We haven't seen each other in over 2yrs. We talked probably a handful of times on the phone in those 2yrs. But when she came down, things seemed like yesterday |
| 15:13.27 | fugitivo | hugov6: SET timeout |
| 15:13.31 | Sparkie- | [default] -> 600 =Array |
| 15:13.34 | docelmo | Looks like she broke the clearwater beach sigh |
| 15:13.38 | docelmo | err sign |
| 15:13.38 | iCEBrkr | docelmo: lol |
| 15:13.48 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
| 15:13.49 | [TK]D-Fender | Sparkie- : Pastebin your voicemail.conf |
| 15:13.55 | iCEBrkr | docelmo: 4'11 @ 98lbs? |
| 15:14.02 | docelmo | ACK! SHE MOUNTED A TURTLE! |
| 15:14.09 | docelmo | talk about itty bitty |
| 15:14.10 | hugov6 | fugitivo: its 1.0.9 and it should work for Dial() where can i find a documentation? |
| 15:14.14 | docelmo | Im 6'6" |
| 15:14.18 | iCEBrkr | docelmo: She's a spinner for sure! |
| 15:14.30 | iCEBrkr | That turtle was taller than her |
| 15:14.33 | fugitivo | hugov6: if it's 1.0.9, you should use digittimeout and responsetimeout |
| 15:14.38 | Sparkie- | http://pastebin.com/444935 |
| 15:14.50 | iCEBrkr | docelmo: I'm only 5'9" |
| 15:15.14 | docelmo | ahh.. I dated a chic that short before.. man did it suck |
| 15:15.18 | hugov6 | fugitivo: ill search on voip-info.org bbl if i cant find what i need. thank you. |
| 15:15.18 | iCEBrkr | lol |
| 15:15.22 | Sparkie- | that's the voicemail.conf on the remote server I'm calling |
| 15:15.46 | iCEBrkr | You think a dialogic card would fit in a Blade server? |
| 15:16.11 | [TK]D-Fender | Sparkie- : The double "u" is making it play the busy message from box "u600" which doesn't exist. |
| 15:16.17 | Sparkie- | I removed the double u |
| 15:16.21 | Sparkie- | same thing happens still |
| 15:16.26 | [TK]D-Fender | And what is ARRAY? |
| 15:16.31 | Sparkie- | that's a good question |
| 15:16.35 | fugitivo | Sparkie-: partebin your voicemail.conf |
| 15:16.50 | [TK]D-Fender | fugitivo : lok up |
| 15:17.24 | Sparkie- | |10:14|- <Sparkie-> http://pastebin.com/444935 |
| 15:18.05 | hugov6 | fugitivo: i dont get it. what does i need if someone calls not fast enough and asterisk calls $extension but $extension inst complete? |
| 15:18.07 | fugitivo | 600 =Array? |
| 15:18.09 | Sparkie- | i just fixed teh Array thing |
| 15:18.14 | hugov6 | s/does/do/ |
| 15:18.16 | docelmo | ok so is there some svn command for downloading now or do I have to do file by file? |
| 15:18.18 | Sparkie- | the, |
| 15:18.41 | [TK]D-Fender | re-pB it |
| 15:18.46 | fugitivo | Sparkie-: are you using asterisk@home? |
| 15:18.51 | docelmo | nevermind.. found it. |
| 15:19.03 | Sparkie- | lol |
| 15:19.09 | Sparkie- | fugitivo: alas, yes. |
| 15:19.15 | [TK]D-Fender | uh oh... |
| 15:19.22 | [TK]D-Fender | A@H... *shudder* |
| 15:19.26 | fugitivo | i see more people having problems with asterisk@home than normal distros |
| 15:19.28 | iCEBrkr | Sparkie-: EJECT! EJECT! EJECT! |
| 15:19.35 | Sparkie- | well, it's not the distro |
| 15:19.36 | fugitivo | isn't asterisk@home made for easy to use asterisk? |
| 15:19.38 | Sparkie- | it's just the tgz |
| 15:19.47 | Sparkie- | i installed all that and then edited the conf files by hand |
| 15:20.42 | fugitivo | Spawn extension (macro-vm, s-NOANSWER, 2) |
| 15:20.45 | fugitivo | what is macro-vm? |
| 15:20.45 | *** join/#asterisk Mike (n=mike@201.135.48.190) |
| 15:21.10 | hugov6 | well ill try DigitTimeout |
| 15:21.28 | *** join/#asterisk paryl (n=paryl@216-201-177-82.res.logixcom.net) |
| 15:21.53 | Sparkie- | refresh pastebin |
| 15:22.06 | Mike | hi guys i have a question, im trying asterisk realtime i have 2 servers im trying to call a sip registered in server1 with a phone register in sip server2 and both using the same database and family both use canreinvite=yes but still cant make the calls any ideas? |
| 15:22.26 | *** join/#asterisk rob314 (n=root@207.58.194.55) |
| 15:23.21 | paryl | i have a te205p, once span is connected to a channelbank and the other will be connected to a T1 (PRI). I'm trying to figure out how to configure the channels on the T1, the only thing i've done so far is to configure fxols=25-48 for the channelbank, but what do i use for the incoming voice channels? is it bchan? |
| 15:24.06 | *** join/#asterisk eKo1 (n=bernd@metrored-gw.tropicohn.com) |
| 15:24.34 | *** join/#asterisk rjuan (n=rjuan@194.224.169.27) |
| 15:24.53 | *** join/#asterisk slayer192 (n=chrisc@66.138.39.225) |
| 15:25.28 | rjuan | Hello |
| 15:25.50 | rjuan | Somebody can help me with this?? |
| 15:25.53 | rjuan | http://pastebin.ca/31964 |
| 15:29.33 | [TK]D-Fender | paryl : What signalling are yougetting from your telco? |
| 15:29.36 | *** join/#asterisk Essobi (i=kstone@75.137.26.216.host.teledvance.com) |
| 15:29.55 | rjuan | I don't know |
| 15:30.01 | rjuan | where I can look it |
| 15:30.04 | [TK]D-Fender | it is PRI right? |
| 15:30.04 | rjuan | ?? |
| 15:30.18 | rjuan | well |
| 15:30.26 | paryl | fender: yes, it's a PRI, but that's all i know |
| 15:30.33 | paryl | do i need more info? |
| 15:31.13 | Essobi | I'm running -head * and zaptel with 2.6.8-2-686-smp and ztdummy interfaces.. I'm getting breakup on my MeetMe and if I run zttest while I'm listening the chopping/jitter in the conference.. I see deviation in the timer when it happens. |
| 15:31.17 | Essobi | Anyone got any ideas? |
| 15:31.20 | *** join/#asterisk dooder (n=Nate@sub29-30.member.dsl-only.net) |
| 15:32.04 | paryl | rjuan: what loadzone are you using? |
| 15:32.06 | dooder | I need to switch voip providers. Broadvoice is giving 1 sec delays. it's annoying |
| 15:32.08 | rjuan | sp |
| 15:32.15 | rjuan | is spain no? |
| 15:33.09 | docelmo | dooder, termination or origination? |
| 15:33.57 | Essobi | I'm running all slip on the conference too.. |
| 15:34.32 | paryl | rjuan: looks right |
| 15:34.42 | dooder | docelmo : both |
| 15:35.18 | [TK]D-Fender | paryl : pastebin your zapata.conf and I'll patch it up for you. Also What is your channel-bank configured for? |
| 15:35.24 | docelmo | Well I can give you termination.. But origination is still a little weak.. we are testing a new platform so DID's arent available yet. |
| 15:35.58 | iCEBrkr | You think a dialogic card would fit in a Blade server? |
| 15:36.05 | *** join/#asterisk SpaceBass (n=sp@static-71-251-228-2.rcmdva.fios.verizon.net) |
| 15:36.07 | SpaceBass | morning |
| 15:36.35 | docelmo | ice full length doubt it. |
| 15:36.42 | paryl | [TK]D-Fender: http://pastebin.ca/31967 span 1 should be the PRI. the channel bank is 24 FXS channels |
| 15:36.45 | iCEBrkr | Yeah, I didn't think so either. |
| 15:36.58 | Johan | Is ther someone who can point me to a list of available commands that can be used in extensions.conf? |
| 15:37.09 | docelmo | wiki |
| 15:37.31 | docelmo | I dont know the url off the top of my head.. But the wiki is were to find it all |
| 15:37.45 | dooder | my ping is around 50ms so it shouldn't be laggie I think |
| 15:37.46 | iDunno | it's also in the book. |
| 15:37.49 | iDunno | http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 15:37.53 | Johan | ok, thanx |
| 15:38.14 | *** join/#asterisk mattfletcher (n=mattf@testbed.nshc.co.uk) |
| 15:38.35 | mattfletcher | anyone ever had any experience of asterisk@home with a Digium TDM400P card in the UK? |
| 15:38.49 | docelmo | dooder, try to ping 66.199.240.2 |
| 15:38.53 | docelmo | tell me your ping times |
| 15:39.41 | file[laptop] | oh I know where that is... haha |
| 15:39.47 | Damin | docelmo: I'm getting 2,482 ms ping times to you... |
| 15:40.08 | [TK]D-Fender | paryl : you gave me zaptel.conf, but thats ok, it needed fixing too :) http://pastebin.ca/31969 |
| 15:40.09 | dooder | docelmo : about 100 ms |
| 15:40.14 | Damin | docelmo: Just kidding! ;) |
| 15:40.28 | Katty | hi Damin |
| 15:40.33 | iCEBrkr | Damin: Get to work! |
| 15:40.42 | Damin | Hi Katty.. I ordered some really shitty weather for you... |
| 15:40.48 | Katty | Damin: gee, thanks. |
| 15:40.51 | *** join/#asterisk dardavoip (i=xT@84.22.47.50) |
| 15:40.52 | Katty | Damin: you're such a sweetheart. |
| 15:40.57 | Damin | iCEBrkr: Give me a break! I've already put in my required 30 minutes for the day! |
| 15:41.10 | iCEBrkr | Damin: ok, in that case.. continue to slack :P |
| 15:41.33 | Katty | Damin: maybe /you/ can tell me why i'm using cifs instead of smbfs to mount a windows share (2003) |
| 15:41.45 | Katty | Damin: huh? HUH? |
| 15:41.48 | Katty | Damin: why is this?! |
| 15:42.01 | docelmo | funny damin.. dont give me a fricken heart attack |
| 15:43.26 | iCEBrkr | haha |
| 15:43.31 | file[laptop] | 100ms here... and I know where it is... and yet 120ms to the other side of the country... boy - routing is fun |
| 15:43.57 | iCEBrkr | Damin: So, how was all that snow you guys had? LOL |
| 15:44.56 | paryl | [TK]D-Fender: thanks! i really appreciate your help |
| 15:45.28 | [TK]D-Fender | is it up and running? |
| 15:45.39 | paryl | [TK]D-Fender: on this particular install, i haven't modified zapata.conf yet |
| 15:45.52 | paryl | [TK]D-Fender: i wanted to make sure the line would come up. |
| 15:46.01 | [TK]D-Fender | ....and did it? |
| 15:46.12 | paryl | [TK]D-Fender: it's not connected yet, the T1 is currently in use. I'll conenct it tonight |
| 15:46.28 | paryl | but that looks a lot better than what i thought :) |
| 15:46.36 | *** join/#asterisk saftsack (n=saftsack@p54A7DF7E.dip.t-dialin.net) |
| 15:46.47 | saftsack | hi |
| 15:46.55 | [TK]D-Fender | I'll PB you a sample zapata.conf to match |
| 15:47.11 | paryl | awesome, thanks man |
| 15:48.25 | mattfletcher | i really don't get this. I have installed asterisk, and i have added my TDM400P card, and I have set up ZAP Trunk g0 |
| 15:48.37 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 15:48.53 | mattfletcher | but when I ring an external number from my softphone, i just get myself echoed back to me after two rings |
| 15:49.11 | [TK]D-Fender | http://pastebin.ca/31970 |
| 15:49.13 | [TK]D-Fender | there you go |
| 15:49.44 | paryl | thanks again, i bow to thee |
| 15:49.50 | [TK]D-Fender | I am assuming certain signalling, but its "typical" for north america |
| 15:49.53 | [TK]D-Fender | np |
| 15:50.04 | [TK]D-Fender | Just need to tweak the rest |
| 15:55.13 | *** join/#asterisk wunderkin (i=kev@12-201-105-27.client.mchsi.com) |
| 15:55.19 | mattfletcher | what configuration needs to be done on a TDM400P after inserting it? |
| 15:55.36 | paryl | [TK]D-Fender: one thing, does this assume i'm using one phone per channel? i have a block of 100 DID numbers |
| 15:55.42 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 15:55.52 | *** part/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
| 15:57.09 | eKo1 | i have 2 * boxen: X and Y. X is connected to the PSTN and forwards a call comming from the PSTN to Y. There is a context in Y that has Authenticate() and when I call a number from my cell. that takes me to this context, I get a busy signal. On the CLI I see that it calls that context twice very quickly and quits!? |
| 15:57.53 | eKo1 | wtf can be causing this? do i need to have a regular touchtone phone for authenticate to work? |
| 15:58.46 | *** join/#asterisk funxion (n=nunya@mtnuser.icgws.com) |
| 15:59.05 | paryl | [TK]D-Fender: how would i route based on the DID? |
| 16:04.44 | funxion | paryl are you trying to provide cos? |
| 16:05.09 | paryl | cos? |
| 16:05.13 | funxion | cclass of service |
| 16:06.04 | *** join/#asterisk jaike (i=aa@210.5.116.17) |
| 16:06.17 | paryl | i don't really what you mean there... i don't know what class of service is in this context. I just need to route calls to internal extensions based on the number dialed |
| 16:06.47 | *** join/#asterisk brettnem (n=brettnem@72.29.102.158) |
| 16:07.17 | eKo1 | exten => <DID>,1,dial(<extension>) |
| 16:07.24 | eKo1 | simple as that |
| 16:07.30 | *** join/#asterisk Corydon-w (i=blue@pdpc/supporter/sustaining/Corydon76-home) |
| 16:09.08 | *** join/#asterisk coppice (n=chatzill@55.197.17.210.dyn.pacific.net.hk) |
| 16:09.38 | paryl | gotcha, awesome |
| 16:10.42 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 16:13.00 | [TK]D-Fender | paryl : sorry for delay |
| 16:13.05 | jaike | version 1.2 still using sox? |
| 16:13.24 | [TK]D-Fender | your incoming context for your PRI would choose what do do with each DID. thats all in extensions.conf |
| 16:15.59 | *** join/#asterisk Kernel_core (n=I@maestrale.eutelsat.net) |
| 16:16.12 | Kernel_core | hi all , does cisco supports Speex codec ? |
| 16:17.05 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 16:17.45 | Kernel_core | I am connected to cisco through SIP(allow=all), and I am connected to asterisk with speex codec (SIP) when I want to make call I get this error : |
| 16:17.46 | Kernel_core | Dec 1 17:14:21 WARNING[6722]: channel.c:2675 ast_channel_make_compatible: No path to translate from SIP/54322-f27d(512) to SIP/cisco1-d66b(4) |
| 16:17.46 | Kernel_core | Dec 1 17:14:21 WARNING[6722]: app_dial.c:1556 dial_exec_full: Had to drop call because I couldn't make SIP/54322-f27d compatible with SIP/cisco1-d66b |
| 16:18.31 | jaike | why translate to speex when u can use sip? |
| 16:18.41 | *** join/#asterisk trixter (n=trixter@65.172.209.246) |
| 16:18.46 | [TK]D-Fender | paryl : the zapata.conf I gave you detailed you PRI in general and where all calls will land (1 context for all channels. DID routing to be done in there), and a sample for 1 ext on your channel-bank for which you would cut & paste and personalize each ext. |
| 16:19.18 | [TK]D-Fender | jaike : speex is a codec, not a protocol.... |
| 16:19.30 | jaike | oops my bad..ulaw i mean |
| 16:19.46 | [TK]D-Fender | For bandwidth reasons perhaps? |
| 16:20.00 | Kernel_core | jaike: my bandwidth is so poor |
| 16:20.04 | Seba_soy | Dan? |
| 16:20.19 | [TK]D-Fender | Ulaw comes out to about 8KB/s after overhead +/-.... |
| 16:20.27 | [TK]D-Fender | GSM = good :) |
| 16:20.28 | jaike | g729 works great for us |
| 16:20.39 | jaike | and u get to support asterisk by buying the license |
| 16:21.02 | Kernel_core | is there anyway to translate SIP/speex to SIP/Ulaw forexample ? |
| 16:21.28 | [TK]D-Fender | sure, just make your endpoints speak their protocal and only that and let * just transcode. |
| 16:21.35 | [TK]D-Fender | (reinvite=no) |
| 16:21.41 | Kernel_core | I know asterisk is able to translate ULAW to g.729 but it seems ... |
| 16:22.14 | Kernel_core | both cisco and my SIP phone have this line in sip.conf |
| 16:22.16 | Kernel_core | canreinvite=no ; Cisco poops on reinvite sometimes |
| 16:22.17 | Kernel_core | reinvite=no ; if so, we need to make them |
| 16:22.25 | jaike | kernel: i guess your device might not support speex |
| 16:23.13 | Kernel_core | jaike: my asterisk supports it well , cause I can make call from one SIP softphone to another SIP softphone through asterisk. |
| 16:23.26 | paryl | [TK]D-Fender: oh, i get it now... i was thinking the one extension referred to calls coming in on the T1, my mistake |
| 16:24.01 | jaike | trying to make an outgoing call? your sip or iax provider might not support it |
| 16:24.10 | jaike | gsm is usually supported |
| 16:24.38 | Kernel_core | jaike: I have g.729 but speex doing great . |
| 16:25.49 | [TK]D-Fender | paryl : All calls on the PRI will fall into the context you choose then you'll do like "exten => 6665551212,1,Goto(thisdidsdestination,s,1)" or so. |
| 16:25.52 | [TK]D-Fender | for each |
| 16:26.09 | [TK]D-Fender | or whatever :) |
| 16:26.38 | *** join/#asterisk zoa (n=zoa@cust-5-0.dsl.versateladsl.be) |
| 16:26.55 | paryl | awesome |
| 16:27.19 | paryl | has anyone here messed with ACD groups and polycom 501's? that's my only other outstanding issue |
| 16:27.44 | paryl | namely, is it possible to make the ACD features of the phone work with asterisk's queues? |
| 16:27.58 | jaike | paryl: yup..ours is working perfectly |
| 16:28.16 | jaike | polycoms..very reliable phones |
| 16:28.29 | paryl | jaike: how? as far as i can tell, their ACD implementation just registers the phone using SIP |
| 16:28.41 | paryl | but i can't get the available/unavailable to work |
| 16:29.16 | paryl | my agents just need to know if they're logged in and if they're set available |
| 16:30.16 | jaike | we use manager api to show whos available and whos not..on the agents workstations |
| 16:31.06 | paryl | jaike: so they just log in and use the api for everythin else? |
| 16:31.22 | [TK]D-Fender | I run ACD's on Polycom 600's... what the question? |
| 16:31.48 | *** join/#asterisk pattieja (n=jpattie@65.59.202.15) |
| 16:32.09 | [TK]D-Fender | paryl : logged in / pause/unpause status visibility is a problem with all SIP phones for the most-part |
| 16:32.36 | mutilator | haha |
| 16:32.38 | paryl | [TK]D-Fender: aha |
| 16:32.45 | mutilator | anyone remember about the guy in support i was talking about this morning? |
| 16:32.48 | [TK]D-Fender | I use the microbrowser to update queue status on-screen on the phone for all agents every 10 seconds |
| 16:32.59 | mutilator | yeah well he just got fired |
| 16:33.01 | mutilator | heh |
| 16:33.07 | jaike | tkd: same with us |
| 16:33.09 | paryl | unfortunately, the 501's don't have the microbrowser :( |
| 16:33.26 | [TK]D-Fender | But there isn't an "easy" way right now. Polycom supports a form of ACD login/out and visibility, but its not well documented enough |
| 16:33.29 | paryl | jaike: wait, how? i was told the 501 couldn't do it |
| 16:34.09 | paryl | [TK]D-Fender: gotcha. is there a win32 client that i can install? |
| 16:34.09 | jaike | ill check my configs tomorrow |
| 16:35.29 | [TK]D-Fender | paryl : well there is IPSwitchboard that will give status on the queues and FOP for web-based |
| 16:35.50 | [TK]D-Fender | IP 600 > all :D |
| 16:36.06 | [TK]D-Fender | (except that NASTY 7970G :O) |
| 16:36.13 | [TK]D-Fender | Colour..... *drools* |
| 16:36.24 | [TK]D-Fender | I love my SPA-941 though :) |
| 16:37.16 | paryl | well, we had a rollout of 40 phones... the extra $50-$100 makes a difference |
| 16:37.26 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool139-221.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 16:38.22 | *** join/#asterisk Flauto (n=zhao@c-24-14-197-214.hsd1.il.comcast.net) |
| 16:38.23 | [TK]D-Fender | I did 26 for mine. |
| 16:39.03 | kuku5 | I need help with a system. I have more than 2 calls coming in to the secretary - anyone above 2 goes do voicemail . its a cisco 7960phone |
| 16:39.31 | [TK]D-Fender | it does make a difference, but when I did mine * came out to 25k$ and everyone else came out to almost 50k$. I wanted PoE and if you ad on what the 500 needs to do it, you're better of buying the 600... |
| 16:39.39 | Essobi | I believe 7960s are limited to to calls per line.. |
| 16:39.49 | Essobi | put more lines on the phone and create a roll over |
| 16:40.03 | kuku5 | but these calls are impossible to pickup |
| 16:40.07 | kuku5 | you only see 2 of them |
| 16:40.27 | [TK]D-Fender | Calls / line = bah! I have reg use all 6 line keys as independant calls w/o call-waiting style BS.... |
| 16:40.39 | [TK]D-Fender | 6 "lines" |
| 16:40.41 | *** join/#asterisk mhnoyes (n=mhnoyes@user-38lc06l.dialup.mindspring.com) |
| 16:41.07 | kuku5 | how did you do it ? |
| 16:41.11 | Essobi | Umm. okay.. let me explain.. |
| 16:41.20 | kuku5 | and wtf is a roll over |
| 16:41.25 | [TK]D-Fender | Mine are Polycom's however, but I'm sure youcan do exactly the same with Cisco's |
| 16:41.32 | Essobi | you've got 6 buttons on a 7960.. and it sip settings.. you've got 6 "line" settings.. |
| 16:41.37 | kuku5 | yeh |
| 16:41.41 | Essobi | each one corrolates to a button.. |
| 16:41.45 | kuku5 | yeh |
| 16:41.51 | Essobi | so setup 5025851212a |
| 16:42.07 | Essobi | so setup 5025851212b |
| 16:42.07 | Essobi | so setup 5025851212c |
| 16:42.07 | Essobi | etc.. |
| 16:42.09 | paryl | the 501 can do 8 calls per line, so 24 calls total, but i have them limited to 1 per :) |
| 16:42.10 | kuku5 | ok |
| 16:42.37 | kuku5 | but for a secretary its just fucking confusing |
| 16:42.54 | [TK]D-Fender | paryl : only give 1 line to your CSR's! |
| 16:42.54 | [TK]D-Fender | Or you'll see screwy things happen |
| 16:42.55 | Essobi | then create a Dial for 5025851212 to ring each one sequentially.. |
| 16:42.55 | Essobi | drawback is you can only have voice mail on the last one in the huntgroup.. |
| 16:42.57 | paryl | ah, i was wondering about that. |
| 16:42.57 | [TK]D-Fender | unless you do some really custom stuff... |
| 16:42.57 | [TK]D-Fender | Maybe 2 regs for them.... |
| 16:43.10 | Essobi | but you gain the ability to have 12 calls on hold.. then they start going to voicemail.. |
| 16:43.18 | paryl | yeah, 2 regs will work |
| 16:43.18 | [TK]D-Fender | My setup had 1 rep gettin 2-3 Queue calls SIMULTANEOUSLY |
| 16:43.26 | Essobi | for a secretary? Nothing confusing.. phone rings. fucking answer it. |
| 16:44.43 | Kernel_core | asterisk can translate g729 codec to alaw/ulaw , but it seems can't transtale speex to alaw .... ! |
| 16:45.03 | Kernel_core | or anyother protocol ! |
| 16:45.18 | mutilator | Druken? |
| 16:45.37 | mutilator | benjk? |
| 16:45.51 | hugov6 | beronet support rocks like hell! |
| 16:47.47 | *** join/#asterisk hellop (n=hellop@cpe-70-95-165-136.hawaii.res.rr.com) |
| 16:49.01 | [TK]D-Fender | I find 6 "lines" for office phones a little much right now and will probably redec them to 3 here and use the others as speed-dials. |
| 16:49.22 | [TK]D-Fender | reduce* |
| 16:49.31 | Essobi | Or you could just use a queue.. |
| 16:49.43 | kuku5 | Essobi: but its a pain in the ass to park these calls |
| 16:49.49 | Essobi | and point them to a line w/o a voicemail fail over.. |
| 16:49.55 | Essobi | So don't park.. |
| 16:49.57 | Essobi | Or you could just use a queue.. |
| 16:49.59 | Essobi | and point them to a line w/o a voicemail fail over.. |
| 16:50.20 | kuku5 | see with regular lines |
| 16:50.23 | Flauto | what is the difference between _1NXXNXXXXXX and 1NXXNXXXXXX |
| 16:50.31 | Essobi | Or use an autoattendand |
| 16:50.33 | Essobi | :) |
| 16:50.43 | Essobi | Umm.. _ means it's a pattern |
| 16:50.48 | *** join/#asterisk mog_work (n=mogorman@gateway.digium.com) |
| 16:50.50 | kuku5 | you could just click - put on hold, clik - put on hold - click - put on holdclick - put on holdclick - put on holdclick - put on hold and the go back to the first one |
| 16:50.52 | [TK]D-Fender | Flauto : you need the "_" is required |
| 16:51.11 | Essobi | kuku5 Dude.. use an auto-attendant and a dial-by-name dir.. |
| 16:51.17 | kuku5 | no no |
| 16:51.18 | Essobi | fire the secretary. |
| 16:51.19 | Essobi | :) |
| 16:51.23 | kuku5 | the secretary has to answer it |
| 16:51.32 | kuku5 | the cisco is just a pain in the ass to use it |
| 16:51.39 | Flauto | would you tell me in what situation it is required? |
| 16:51.51 | Flauto | i am looking at some samples |
| 16:52.06 | Flauto | and they have mixed _ and without _ |
| 16:52.09 | [TK]D-Fender | Flauto : you need the "_" anywhere you are referring to a pattern |
| 16:52.18 | Flauto | oh |
| 16:52.32 | Flauto | if it is only one number, you don't need it? |
| 16:52.39 | [TK]D-Fender | otherwise try and find a phone that can dial X's and N's in alpha-numeric :) |
| 16:53.00 | Flauto | like if i just want to set 911, i dont' need _ |
| 16:53.05 | Flauto | right? |
| 16:53.35 | [TK]D-Fender | correct. |
| 16:53.44 | [TK]D-Fender | that is a SPECIFIC number, not a PATTERN |
| 16:55.26 | Flauto | thanks very much, i am learning something new everyday, |
| 16:55.27 | Flauto | thank you |
| 16:57.59 | [TK]D-Fender | Essobi : no need to park if they can hold 1-per line.... |
| 16:58.13 | [TK]D-Fender | No-one wants a queue for reception.... thats nuts |
| 16:58.24 | Flauto | when should i use n instead of numbers |
| 16:58.30 | [TK]D-Fender | says your IVR isn't doing a good job :) |
| 16:59.32 | mutilator | OMFG! |
| 16:59.41 | [TK]D-Fender | Flauto : patterns are for dialing numbers that match a patterna and not a specific number like "exten => _9.,1,Dial(Zap/1/${EXTEN:1})" do dial out. |
| 16:59.48 | mutilator | support keeps calling me telling me customers are having problems with faxing over their voip lines |
| 17:00.14 | mutilator | so i went to the sales and the support whiteboards and wrote in BIG letters FAX IS NOT SUPPORTED!~~~~@ |
| 17:00.18 | mutilator | then underlined it 1000 times |
| 17:00.21 | Flauto | oh, sorry |
| 17:01.47 | Flauto | i mean sometimes i see exten => _9X.,n,Dial..... what is the differenece is that n with a number |
| 17:03.26 | [TK]D-Fender | 'n' in there implies the next priority in that pattern. |
| 17:03.45 | [TK]D-Fender | which means you have an EXTEN => _9X.,1, higher up |
| 17:04.01 | Flauto | so, i can use n or a number |
| 17:04.05 | *** join/#asterisk oej (n=Olle@gateway.digium.com) |
| 17:04.11 | Flauto | does it make any difference |
| 17:04.38 | [TK]D-Fender | Flauto : You should start witha a 1, but can use "n"'s for the next steps if its linear |
| 17:05.11 | [TK]D-Fender | * always looks for the 1 priority first |
| 17:05.24 | [TK]D-Fender | and then follows "n" after if they occur |
| 17:05.29 | mrtwister | hello... have issue. debian. what is missed? compiling addond 1.0.9 and: cdr_addon_mysql.c:33:19: error: mysql.h: No such file or directory |
| 17:05.29 | mrtwister | cdr_addon_mysql.c:34:20: error: errmsg.h: No such file or directory |
| 17:05.33 | Flauto | but if i want to use 103, 104... that means i need to use 1 2 3, is that right? |
| 17:06.11 | [TK]D-Fender | Flauto : you seriously need to read the WIKI on dialplan patterns, flow and logic. |
| 17:06.14 | [TK]D-Fender | ~wiki |
| 17:06.28 | Flauto | okay, thanks |
| 17:06.29 | [TK]D-Fender | ~DOCS |
| 17:06.31 | jbot | docs is, like, probably 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 or http://www.asteriskguru.com |
| 17:06.37 | *** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au) |
| 17:07.04 | [TK]D-Fender | Yeah you seems just a little too new for me to fully direct you more than to refer you this time till you have te basics down. |
| 17:07.58 | Flauto | i have my asterisk up and running and it seems funcional |
| 17:08.12 | drray | congrats |
| 17:08.13 | Flauto | but there are a lot of details that i have no idea about |
| 17:08.21 | Flauto | that is the part that i am trying to make up |
| 17:08.31 | Flauto | i basically copied a lot of samples |
| 17:08.36 | Flauto | from different places |
| 17:08.45 | jaike | mrtwister: i had the error when i compiled without renaming the folder created after extracting tarballs |
| 17:08.46 | Flauto | and tried to use in my system |
| 17:08.54 | Flauto | some work pretty well |
| 17:09.12 | Flauto | but i just realized that i need to figure out what is really going on |
| 17:09.31 | *** join/#asterisk EriSan (n=erisan@81-174-35-3.f5.ngi.it) |
| 17:10.21 | mrtwister | jaike, folder have to be asterisk-addons? only this name? |
| 17:10.45 | jaike | asterisk, libpri, etc |
| 17:10.57 | jaike | yup |
| 17:11.06 | mrtwister | jaike, thank you |
| 17:13.16 | *** part/#asterisk dooder (n=Nate@sub29-30.member.dsl-only.net) |
| 17:13.53 | jaike | gnite yall |
| 17:14.05 | *** part/#asterisk jaike (i=aa@210.5.116.17) |
| 17:14.12 | Cinen | I would like to share my voicemail across several machines. Is writing it to a database better or sharing the folder? |
| 17:17.30 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 17:19.26 | brettnem | "sharing" ? |
| 17:20.18 | fugitivo | Cinen: nfs |
| 17:20.37 | Cinen | thank you |
| 17:20.46 | brettnem | fwiw I hear that voicemail over NFS is a horrible idea.. |
| 17:20.56 | fugitivo | why? |
| 17:21.05 | fugitivo | gfs then |
| 17:21.20 | Cinen | I kinda heard the same thing but this ODBC is awful to get working |
| 17:21.20 | brettnem | I hear it is way to slow to actually work.. |
| 17:21.26 | brettnem | but haven't tried myself |
| 17:21.33 | fugitivo | nfs slow? |
| 17:21.42 | fugitivo | nfs is great |
| 17:21.52 | brettnem | "nfs is great" ? |
| 17:22.04 | fugitivo | run 30 diskless terminals using nfs and you'll have the same feeling :) |
| 17:22.21 | Sedorox | hmmm |
| 17:22.34 | brettnem | nfs has it's place |
| 17:22.57 | fugitivo | i have only good experiences with it |
| 17:23.11 | Sedorox | isn't there still secutiry issues with it? |
| 17:23.38 | fugitivo | Sedorox: windows have security issues too, so? |
| 17:23.44 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
| 17:23.50 | brettnem | fugitivo: do you currently do voicemail over nfs? |
| 17:23.53 | mutilator | http://www.patmedia.net/marklevinson/cool/cool_illusion.html |
| 17:24.01 | Sedorox | true.... hence why I don't use it :p |
| 17:24.03 | fugitivo | brettnem: no, i run 30 diskless terminals with it |
| 17:24.07 | mrtwister | debian again... clean install... checking for C++ compiler default output file name... configure: error: C++ compiler cannot create executables |
| 17:24.51 | anthm | if you wanna use voicemail on nfs use -t flag when starting asterisk |
| 17:25.14 | fugitivo | mutilator: holy! |
| 17:25.39 | *** join/#asterisk gniretar (n=mark@152.160.35.1) |
| 17:25.41 | gniretar | hi all |
| 17:25.42 | brettnem | cool. -t option! |
| 17:25.59 | brettnem | anthm: how does playback work. |
| 17:26.00 | gniretar | what ports do i need to open up on my router so that an external SIP client can connect? |
| 17:26.19 | fugitivo | gniretar: udp 5060, 10000-20000 |
| 17:26.23 | Cinen | 5060 and 10k to 20k |
| 17:26.27 | anthm | playback does not suffer |
| 17:26.41 | brettnem | anthm: is it a caching issue? |
| 17:26.46 | brettnem | or writing.. |
| 17:26.54 | fugitivo | common, i play mp3 and watch video with nfs |
| 17:27.10 | fugitivo | why voicemail is going to be bad with nfs? |
| 17:27.11 | mutilator | all i see is green dots even when i look away |
| 17:27.11 | Sedorox | hmmm |
| 17:27.19 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
| 17:27.24 | anthm | writing to nfs in realtime produces sucky results but reading doesnt that's why i added thet thing |
| 17:27.38 | Sedorox | hmmm |
| 17:27.39 | brettnem | anthm: fancy, thanks for the tip. |
| 17:27.52 | anthm | np |
| 17:28.15 | Sedorox | hmmmm |
| 17:28.24 | fugitivo | anthm: what sucky results? |
| 17:28.31 | Sedorox | I've been looking at a network file system for here.. and I'm stuck between nfs or shfs |
| 17:28.58 | fugitivo | Sedorox: you can make a cluster with gfs |
| 17:28.59 | Cinen | why not write the vm to a database? |
| 17:29.07 | fugitivo | (global file system) |
| 17:29.15 | Sedorox | hmm |
| 17:29.39 | anthm | the file would get messed up some of the audio missing jittery etc |
| 17:29.43 | Sedorox | well my one box here... I wanna start as a file server.. so I want something good that could withstand network bumps... |
| 17:29.51 | fugitivo | Cinen: it's another point of error, but that's my opinion |
| 17:30.26 | fugitivo | Sedorox: nfs works great for me |
| 17:31.27 | Cinen | I have mysql realtime up and running and thought it would be nice to cluster the databases for additional servers |
| 17:31.27 | fugitivo | Sedorox: providing filesystem for 30 diskless terminal, music and video |
| 17:31.27 | gniretar | ugh, that sucks SIP is such a piss poorly designed protocol |
| 17:31.29 | Sedorox | I had problems with it before.. but it mighta been the kernel... |
| 17:31.55 | Cinen | but I have yet to get vm ODBC running so I will probably go the nfs route. |
| 17:32.01 | Cinen | Thank you for the help |
| 17:34.14 | rjuan | http://pastebin.ca/31990 |
| 17:34.23 | rjuan | any idea with a zap problem?? |
| 17:35.24 | *** join/#asterisk Gennaro (n=Gennaro@ppp-62-10-136-152.dialup.tiscali.it) |
| 17:35.29 | Gennaro | hi |
| 17:36.38 | Sedorox | rjuan: looks like you don't have zap installed in asterisk |
| 17:36.38 | JunK-Y | rjuan: u dont have chan_zap.so loaded. |
| 17:36.39 | rjuan | what I have to do?? |
| 17:36.40 | Sedorox | thats what I ment :p |
| 17:37.08 | JunK-Y | install and configure zaptel. |
| 17:37.09 | Sedorox | by any chance.. did you install/compile zaptel AFTER you installed/compiled asterisk? |
| 17:37.12 | rjuan | I've installed zaptel |
| 17:37.16 | JunK-Y | u have to make sure u have that module loaded: show modules like chan_zap |
| 17:37.30 | JunK-Y | and if u do show channeltypes, u should see zap |
| 17:37.56 | Seba_soy | hi |
| 17:37.59 | rjuan | and I compiled zaptel so |
| 17:38.11 | Seba_soy | somebody could configure SIPXEZPHONE to register into ASTERISK? |
| 17:38.16 | Seba_soy | I cant do it |
| 17:38.18 | Seba_soy | :( |
| 17:38.56 | Seba_soy | What I have toput on IDENTITY and REALM? |
| 17:39.39 | Gennaro | realm = server |
| 17:39.52 | Seba_soy | my asterisk server is in 10.10.10.1 |
| 17:39.59 | Seba_soy | my pc is 10.10.10.6 |
| 17:40.35 | Seba_soy | IDENTITY? |
| 17:40.35 | eKo1 | interesting...calling ast_answer and an already answered channel really fucks things up |
| 17:41.19 | anthm | in C ? |
| 17:41.34 | Gennaro | have u add in sip.conf the phone? |
| 17:41.46 | rjuan | JunK-Y: I don't have chan_zap module |
| 17:42.01 | Seba_soy | yes |
| 17:42.02 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 17:42.06 | Seba_soy | I have add this: |
| 17:42.07 | rjuan | but I've installed zaptel from debian source |
| 17:42.12 | JunK-Y | rjuan: load chan_zap.so |
| 17:42.13 | Seba_soy | [600] |
| 17:42.18 | Seba_soy | username=600 |
| 17:42.22 | Seba_soy | secret=seba |
| 17:42.27 | Seba_soy | etc |
| 17:42.30 | Seba_soy | dynamic, etc |
| 17:42.32 | rjuan | where?? |
| 17:42.36 | JunK-Y | and read how to be able to load that module, cause u have some stuff to do to be able. |
| 17:42.40 | JunK-Y | in ur CLI. |
| 17:42.53 | rjuan | thanks |
| 17:43.31 | Seba_soy | gennaro: |
| 17:43.38 | Seba_soy | [600] |
| 17:43.38 | Seba_soy | type=friend |
| 17:43.39 | Seba_soy | username=600 |
| 17:43.39 | Seba_soy | secret=seba |
| 17:43.39 | Seba_soy | disallow=all |
| 17:43.39 | Seba_soy | allow=ulaw |
| 17:43.41 | Seba_soy | allow=g729 |
| 17:43.43 | Seba_soy | host=dynamic |
| 17:43.45 | Seba_soy | canreinvite=no |
| 17:43.47 | Seba_soy | dtmfmode=rfc2833 |
| 17:43.48 | rjuan | There isn't any chan_zap |
| 17:43.50 | kuku5 | Is there a way to control a cisco phone via web interface ? |
| 17:43.56 | JunK-Y | Seba_soy: use www.pastebin.ca |
| 17:43.59 | kuku5 | liek transfer a call thats on hold. |
| 17:44.13 | JunK-Y | rjuan: goto read how to be able to use that module. |
| 17:44.22 | SpaceBass | kuku5 do to what? i know my 7912g has a web interface... not sure teh 7940/60 do |
| 17:44.33 | Seba_soy | <PROTECTED> |
| 17:45.39 | eKo1 | anthm: yes, in C |
| 17:45.43 | eKo1 | what else |
| 17:45.55 | Gennaro | try softphone |
| 17:46.06 | Gennaro | sj or xlite that account.. |
| 17:46.07 | kuku5 | SpaceBass: Lets say I dont like to use th buttons on th 7960 and want to use the computer |
| 17:46.07 | file | JunK-Y! |
| 17:46.08 | Seba_soy | I want to develop my own |
| 17:46.12 | eKo1 | i took it out and now everything works smoothly |
| 17:46.13 | *** join/#asterisk amir_ (n=amir@gentoo/developer/amir) |
| 17:46.16 | brettnem | what do I want for my birthday? anyone know? |
| 17:46.17 | anthm | well, i'm asking cos i was gonna suggest you can check the state to avoid doing that |
| 17:46.18 | Seba_soy | i want to try sipxtapi |
| 17:46.33 | anthm | which is mucha easier in C than from the outside |
| 17:47.12 | SpaceBass | kuku5 to dial, answer, etc? I'm not aware of a web interface to do that |
| 17:47.29 | SpaceBass | kuku5 there are ways to integrate to things like outlook of os x address book to dial, etc |
| 17:47.59 | kuku5 | But if a secreatry gets 10 calls |
| 17:48.14 | SpaceBass | ahh sounds like you need FOP - flash operators pannel |
| 17:48.22 | kuku5 | on a regular phone ( some analog pbx ) this is push of a button, with the cisco 7960 its too much jumping |
| 17:48.27 | kuku5 | SpaceBass: I tried it |
| 17:48.28 | gniretar | FOP sux |
| 17:48.31 | kuku5 | it sucks |
| 17:48.42 | SpaceBass | yeah, it sucks... but... hey |
| 17:48.44 | gniretar | it is nice to have every now and then |
| 17:48.50 | gniretar | but for anything but observing |
| 17:48.50 | SpaceBass | there is an astgui or something out there |
| 17:48.55 | gniretar | and even that |
| 17:49.04 | gniretar | i wanna write an astGUI |
| 17:49.11 | gniretar | i'm thinking RB |
| 17:49.16 | gniretar | RealBasic* |
| 17:49.44 | [TK]D-Fender | kuku5 : you just need to configure your phone right. You can have 1 call / line key on that one I'm sure. |
| 17:49.47 | *** join/#asterisk _mountie (n=mountie@trb229.travel-net.com) |
| 17:50.46 | kuku5 | How can I put them on hold from fop ? |
| 17:51.55 | Gennaro | i have a stupid phone that havent autoadjust of jitter |
| 17:52.10 | Gennaro | someone know i can do to get value? |
| 17:52.48 | Gennaro | i see on web that i can try to calculate it |
| 17:52.56 | Gennaro | someone have tried it? |
| 17:53.26 | *** join/#asterisk xkev (n=kevin@orbit.xmission.com) |
| 17:53.38 | xkev | how does one do +101 in AEL |
| 17:57.32 | *** join/#asterisk fiber0pti (n=johndoe@207.114.199.98) |
| 17:57.44 | fiber0pti | Anyone know how to get the services button to work on polycoms? |
| 17:57.56 | brettnem | pray |
| 17:57.59 | robin_sz | ok, is there some IRC channel for windows users? |
| 17:58.07 | robin_sz | I feel the need to hurl abuse at people |
| 17:58.41 | [TK]D-Fender | xkev : I don't think the idea of +101 exists in AEL... thats sortof its point... |
| 17:59.14 | [TK]D-Fender | fiber0pti : You need to set up the MicroBrowser XML tags in the phones config. |
| 17:59.35 | xkev | fender, then how does one handle a priority+101 return of an app? |
| 18:00.03 | fiber0pti | know how? or where I can find out how? haven't found much on voip-info for this subject |
| 18:00.29 | [TK]D-Fender | xkev : I think that you'll always have to do jumps based on status variables... I haven't gotten too far into it yet... |
| 18:00.39 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 18:01.23 | docelmo | YIPPIE! |
| 18:03.06 | [TK]D-Fender | fiber0pti : here's a sample of mine - http://pastebin.ca/31994 you need to add it to the phone's config file or to sip.cfg if you want it global. |
| 18:03.24 | *** join/#asterisk azfhasterisk (n=rickb@VDSL-130-13-192-106.PHNX.QWEST.NET) |
| 18:03.50 | rjuan | hello |
| 18:03.56 | rjuan | I've got a problem |
| 18:04.01 | [TK]D-Fender | This implements both an IDLE microbroser page, and one for SERVICES |
| 18:04.05 | rjuan | I can't load chan_Zap |
| 18:04.13 | brettnem | why not? |
| 18:04.20 | rjuan | because |
| 18:04.27 | rjuan | is not in my modules list |
| 18:04.31 | brettnem | oh nice |
| 18:04.44 | rjuan | any idea |
| 18:04.44 | brettnem | ok, so i there a problem you needed help with? |
| 18:04.55 | fiber0pti | D-Fender: thanks |
| 18:05.13 | rjuan | yes |
| 18:05.20 | brettnem | oh |
| 18:05.40 | *** join/#asterisk ManxPower (n=ewieling@adsl-67-65-233-194.dsl.lgvwtx.swbell.net) |
| 18:05.46 | brettnem | we can't help you if you don't tell us the problem |
| 18:05.57 | ManxPower | Yay! I have a move date. |
| 18:06.04 | brettnem | where ya moving to? |
| 18:06.07 | rjuan | well i installed and compilled |
| 18:06.09 | rjuan | zaptel |
| 18:06.11 | ManxPower | brettnem, near Birmingham AL |
| 18:06.24 | brettnem | AL? why AL? |
| 18:06.28 | brettnem | ;) |
| 18:06.29 | [TK]D-Fender | fiber0pti : np. I use them for live queue stats, and other interactive stuff.... |
| 18:06.33 | ManxPower | brettnem, There's a place I like there. 8-) |
| 18:06.35 | rjuan | and I load wcfxo and zaptel modules |
| 18:06.47 | ManxPower | brettnem, on the top of a mountian 8-) |
| 18:06.56 | fiber0pti | D-Fender: Awesome..might have to try that out. Love these phones |
| 18:06.59 | brettnem | ManxPower: where are you now? I thought you were more midwestish |
| 18:07.07 | [TK]D-Fender | fiber0pti : 600 or 601? |
| 18:07.11 | rjuan | but i can't load chan_zap because i haven't it |
| 18:07.26 | rjuan | what i have to do/compile to have it? |
| 18:07.40 | brettnem | rjuan: make asterisk after you make zaptel.. try that |
| 18:07.44 | ManxPower | brettnem, No. I lived in Waveland MS and New Orleans, LA. Katrina changed all that. Now I'm living near Texarkana TX on a temp basis. LEaving on Dec 12. |
| 18:08.04 | ManxPower | rjuan, chan_zap won't build unless you have zaptel installed BEFORE you build Asterisk |
| 18:08.18 | rjuan | ohh |
| 18:08.31 | rjuan | I have debian package of asterisk |
| 18:08.35 | brettnem | wow.. that's out there |
| 18:08.41 | ManxPower | rjuan, Sorry, can't help you with a package. |
| 18:08.49 | *** join/#asterisk EriSan (n=erisan@81-174-35-3.f5.ngi.it) |
| 18:08.49 | ManxPower | brettnem, It should be fun/interesting. |
| 18:08.50 | fiber0pti | d-fender: 500's and 501s |
| 18:08.51 | rjuan | i supose so |
| 18:09.03 | [TK]D-Fender | fiber0pti : Not supposed to have a MB on the 500's.... |
| 18:09.03 | rjuan | i'm thinking in build asterisk |
| 18:09.12 | [TK]D-Fender | last I recall |
| 18:09.14 | ManxPower | brettnem, Internet access will be....a chanllanage |
| 18:09.16 | fiber0pti | D-Fender: interesting. I will try right now |
| 18:09.16 | brettnem | I'm in austin. I helped with a shelter/placement agency here to help people find places to live. |
| 18:09.21 | brettnem | ManxPower: don't go dish |
| 18:09.22 | brettnem | ;) |
| 18:09.54 | brettnem | yeah the 500 phones have a services button, but I don't think it does anything. |
| 18:09.58 | ManxPower | brettnem, someone will allow me to use their DirectTV internet. I'll use that for everything except SSH, then get a dialup line and run ONLY ssh over it. |
| 18:10.18 | brettnem | ManxPower: egads! what about voip? you poor thing.. haha |
| 18:10.49 | brettnem | run a cat 5 down to digium... ;) |
| 18:10.56 | ManxPower | brettnem, for long term I'm thinking an extended 900Mhz wireless link to someplace with broadband (9 miles away or so) |
| 18:11.16 | brettnem | I don't; think I could live somewhere without broadband.. I'd go insane |
| 18:11.20 | brettnem | or maybe |
| 18:11.25 | brettnem | maybe I might go sane.. |
| 18:11.28 | ManxPower | And if I wait a bit longer I should be able to get a FCC license for the lower bands for internet access. |
| 18:11.37 | techie | yeah imagine that, no pr0n and no voip |
| 18:11.55 | fugitivo | no gentoo |
| 18:11.57 | fugitivo | god |
| 18:12.00 | brettnem | no p2p |
| 18:12.08 | fugitivo | i don't use p2p |
| 18:12.11 | fugitivo | but |
| 18:12.11 | *** join/#asterisk hugov6 (n=foo@p54AD762B.dip.t-dialin.net) |
| 18:12.12 | fugitivo | no gentoo |
| 18:12.12 | brettnem | no internet2 |
| 18:12.19 | brettnem | no ipv6 |
| 18:12.27 | ManxPower | Actually I WILL be running VoIP over wireless. From a Cisco where the phone lines are located to where I'll be living (only a couple of hundred feet) |
| 18:12.29 | ian_k | no whatismyip.com |
| 18:12.47 | brettnem | no wheresgeorge.com |
| 18:12.57 | hugov6 | hiho. got disconnected. so the last question again: |
| 18:13.06 | fugitivo | no gentoo |
| 18:13.13 | hugov6 | is this a possbile statement in extensions.conf: exten => _X.,2,GotoIf($["${EXTEN}" = "_9212XX"]?123) |
| 18:13.26 | ManxPower | hugov6, no. |
| 18:13.40 | xkev | is there a var I can set to influence AST_CAUSE_* upon Hangup() ? |
| 18:13.43 | brettnem | hugov6: why would you do such a thing? |
| 18:13.50 | fugitivo | i realize that i only use broadband for emerge |
| 18:13.56 | ManxPower | hugov6, the pattern is evaluaged long before the comparison |
| 18:14.08 | hugov6 | because i dont want to enter each extension which is avl. in the building |
| 18:14.19 | brettnem | isn't that the same as: exten => _9212XX,1,Goto(123) |
| 18:14.22 | SpaceBass | do logos for cisco phones have to be 8bit or less bitmaps? or will 16bit work/ |
| 18:14.25 | hugov6 | is there a qorkaround or a tip? |
| 18:14.30 | ManxPower | hugov6, look at README.variables to see how to get the first 4 digits of ${EXTEN} then compare on that. |
| 18:14.36 | ManxPower | or use SubString |
| 18:14.43 | many | or use macros |
| 18:14.44 | brettnem | SpaceBass: I think you need 8 bit.. I found that "icons" worked well. |
| 18:14.47 | hugov6 | ManxPower: thank you |
| 18:14.55 | brettnem | SpaceBass: using too big files causes really ugly images |
| 18:14.57 | hugov6 | many! |
| 18:14.59 | xkev | e.g. variable or application to set struct member ast_channel->hangupcause |
| 18:15.05 | many | hugo;) |
| 18:15.12 | xkev | ..I am doing early media and have not answered the channel |
| 18:15.21 | brettnem | wahoo early media |
| 18:15.23 | ManxPower | hugov6, you can also just use an exten => _9212XX,Goto(wherevver) |
| 18:15.32 | brettnem | that's what I was sayin |
| 18:16.04 | hugov6 | ManxPower: well, that looks like it does what i want. trying this now. thank you all. |
| 18:16.04 | *** join/#asterisk SplasPood (i=nobody@paravolve.net) |
| 18:16.32 | xkev | chan_sip does this: if (ast->hangupcause && ((res = hangup_cause2sip |
| 18:16.34 | xkev | (ast->hangupcause)))) { |
| 18:16.53 | xkev | ... blah else { transmit_response_reliable(p, "403 Forbidden", &p->initreq, 1); |
| 18:17.11 | xkev | so I get a 403 Forbidden sent on all early media |
| 18:17.32 | docelmo | I got a good Q for someone in here.. Whats the easiest way to get a phone registered to one asterisk box login as an agent to another asterisk box? |
| 18:17.52 | fiber0pti | D-Fender: Think you're right. 500s don't have a microbrowser |
| 18:18.04 | xkev | after a Hangup(). I guess I rather need to do Busy() or Congestion() and there are no maps for any other causes in chan_sip anyway, so nm |
| 18:18.17 | fiber0pti | D-Fender: Wonder what I can do with this key then.. :S |
| 18:18.23 | *** join/#asterisk lenne_dk (n=mirc@83.72.129.7.ip.tele2adsl.dk) |
| 18:21.03 | SpaceBass | arrruugg stupid logo |
| 18:21.27 | lenne_dk | <PROTECTED> |
| 18:21.47 | [TK]D-Fender | fiber0pti : not sure... I think MB support is only partial and depends on firmware, etc... |
| 18:22.18 | *** join/#asterisk VxJasonxV (n=jason@unaffiliated/VxJasonxV) |
| 18:22.31 | fiber0pti | D-Fender: I know I have the latest firmware that this phone will work with. Time to upgrade |
| 18:22.53 | *** join/#asterisk mountie (n=mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com) |
| 18:23.26 | *** join/#asterisk syzygybsd (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 18:24.36 | mog_work | hey anyone here good with the find command? |
| 18:24.44 | [TK]D-Fender | thats not always a GOOD thing with Poly's |
| 18:25.55 | mog_work | i want to copy all files from a directory made on novermeber 18 |
| 18:26.17 | fugitivo | mog_work: perl is your friend |
| 18:26.29 | brettnem | blah.. boated use for perl |
| 18:26.37 | mog_work | i dont know perl |
| 18:26.45 | mog_work | i am pretty sure i can do it with find |
| 18:26.51 | mog_work | but not sure on syntax |
| 18:26.51 | brettnem | mog_work it'd be some sort of cp `find..` <dest> |
| 18:26.58 | mog_work | yes |
| 18:27.01 | mog_work | i realize that |
| 18:28.49 | lenne_dk | mog... find . -mtime 13 -exec cp {} /whereever \; |
| 18:28.50 | hugov6 | problem: have to match exten => _9212XX,3,Goto(incoming|${EXTEN}|1) |
| 18:28.54 | hugov6 | damn sorry |
| 18:29.07 | hugov6 | ignore that please |
| 18:29.13 | mutilator | fukin john denver |
| 18:29.26 | lenne_dk | Speak nice of the dead, please. |
| 18:29.29 | msw | lenne_dk: that will copy newer stuff too |
| 18:29.32 | mog_work | ahh thats it lenne thanks |
| 18:29.32 | msw | I think |
| 18:29.38 | msw | or older stuff? |
| 18:29.48 | brettnem | mtime=days ago |
| 18:29.49 | lenne_dk | 13 = thirteen days |
| 18:29.53 | mog_work | yeah |
| 18:29.56 | mog_work | thanks a million |
| 18:29.59 | hugov6 | problem: have to match _9212XX and _X. in one context. what can i do, that _9212XX is evaluated first? |
| 18:30.26 | mog_work | well if it coppies the last 13 days , i can then just do a remove on the other 12 ^_^ |
| 18:30.26 | lenne_dk | -13 older +13 newer (or opposite). Try with -exec ls -ld {} \; first |
| 18:30.34 | mog_work | yeah |
| 18:30.41 | xkev | hugov6, include the _X. from another context |
| 18:30.55 | hugov6 | xkev: thanks. try that. |
| 18:31.00 | msw | lenne_dk: aah right, the + and - parts of -{a,c,m}time always trip me up |
| 18:31.30 | brettnem | blah |
| 18:31.45 | brettnem | why can't find just do -date "Nov 13" |
| 18:32.01 | mog_work | because life is lame |
| 18:32.05 | brettnem | oh |
| 18:32.12 | mog_work | ^_^ |
| 18:32.21 | iCEBrkr | Anyone have any problems with TE210P in a Dell PowerEdge 2850? I see on Digium's site that they have issues with a 2650. |
| 18:32.26 | mog_work | sorries you rock brettnem |
| 18:32.34 | brettnem | haha.. thx |
| 18:32.49 | brettnem | iCEBrkr: I think I hear people complaining about onboard E1000 interfaces |
| 18:33.01 | [TK]D-Fender | Digium + E1000 = TROUBLE |
| 18:33.04 | brettnem | which they frequently interfere with |
| 18:33.12 | iCEBrkr | brettnem: That's also listed on their site as an area of problems |
| 18:33.20 | [TK]D-Fender | Sangoma + Anything = Happy :) |
| 18:33.25 | brettnem | iCEBrkr: just go with sangoma and enjoy |
| 18:33.37 | iCEBrkr | sangoma?! who? |
| 18:33.38 | brettnem | [TK]D-Fender: word |
| 18:33.39 | iCEBrkr | lol |
| 18:33.43 | *** join/#asterisk mfarley (i=mfarley@208.222.40.225) |
| 18:33.55 | [TK]D-Fender | brettnem : I have a very nice top'o'the'line A104d :) |
| 18:33.56 | brettnem | er smacks |
| 18:33.57 | iCEBrkr | Offffaaaaaaah! |
| 18:34.03 | iCEBrkr | Ow |
| 18:34.26 | brettnem | [TK]D-Fender: I have a bunch of digium cards I'm trying to figure out what to do with |
| 18:34.49 | brettnem | I'm considering eating them piece by piece.. like that crazy guy who ate lightbulbs and cars |
| 18:35.00 | iCEBrkr | brettnem: that'd be sweet |
| 18:35.30 | [TK]D-Fender | I've got a TE405P here thats going back to my vendor :) and I already had him bring another in prior to the A104d because of a complete inability to synch with the telco clock, and nasty frame slips |
| 18:35.44 | iCEBrkr | So these sangoma cards... |
| 18:36.52 | mfarley | Hi! Does anyone here happen to know if the 'Variable: SIPADDHEADER=' manager command is new in 1.2? Been having problems getting allpage.agi to work, but it's on a 1.0.7 box. |
| 18:37.23 | ManxPower | mfarley, I think it's 1.2 only. |
| 18:37.50 | mfarley | ManxPower: Ahh, I figured that was the issue. Time to upgrade! (Thanks!) |
| 18:37.54 | *** join/#asterisk bkw_ (n=bkw_@adsl-69-148-35-37.dsl.tulsok.swbell.net) |
| 18:37.56 | brettnem | on second thought.. I don't think I'll eat my digium cards |
| 18:37.57 | iCEBrkr | brettnem: How much different are these Sangoma cards? I mean, are they as simple as configuring the Digium cards? |
| 18:38.02 | *** part/#asterisk shanky (i=jramirez@217.11.114.145) |
| 18:38.16 | brettnem | iCEBrkr: the config is a bit weird, but they won't give you the same problems. |
| 18:38.17 | *** join/#asterisk oej (n=Olle@gateway.digium.com) |
| 18:38.18 | mog_work | wise move brettnem |
| 18:38.44 | iCEBrkr | brettnem: I'm assuming it still uses the zaptel/libpri stuff? |
| 18:38.50 | brettnem | if we can get some sort of chan_sangoma worked in, we'd be set |
| 18:38.56 | bkw_ | iCEBrkr, yes but you gain hardware hdlc |
| 18:38.57 | brettnem | iCEBrkr: yes, it emulates zap |
| 18:39.09 | brettnem | bkw_: so nice to see you on the dark side. :) |
| 18:39.10 | Cresl1n | bkw_: pssh.... we can do that now |
| 18:39.17 | brettnem | oh oh! fight! |
| 18:39.19 | bkw_ | Cresl1n, took ya long enough |
| 18:39.21 | Cresl1n | :-) |
| 18:39.21 | brettnem | oh sorry |
| 18:39.24 | iCEBrkr | lol |
| 18:39.38 | mog_work | bkw_ in #asterisk |
| 18:39.39 | mog_work | wtf |
| 18:39.44 | iCEBrkr | mog_work: He's slum'n |
| 18:39.45 | bkw_ | Honestly you can use sangoma cards without zaptel |
| 18:39.51 | mog_work | heh |
| 18:39.55 | brettnem | but can you support competitor drivers in your core? |
| 18:39.57 | Cresl1n | bkw_: until you go robbed bit |
| 18:40.02 | bkw_ | and shed the bottleneck that is zaptel |
| 18:40.09 | bkw_ | Cresl1n, soon my dear |
| 18:40.18 | mog_work | can it dacs? |
| 18:40.34 | bkw_ | i'm sure it can , it can cross channels in hardware |
| 18:40.45 | brettnem | you can't seriously actually want to dacs with these cards |
| 18:40.50 | Cresl1n | bkw_: I doubt sangoma will support all the robbed bit protocols that zaptel/chan_zap support in the near future |
| 18:40.50 | [TK]D-Fender | I also installed a Sangoma S518 ADSL card at home and it even uses the same driver. Took 5 minutes to configure.... so easy its ridiculous |
| 18:40.58 | brettnem | I mean, I'm sure it's no problem.. but why? |
| 18:41.04 | Cresl1n | bkw_: It's a LOT of work my dear |
| 18:41.16 | bkw_ | Cresl1n, most people that are serious about telecom don't use RBS |
| 18:41.21 | bkw_ | I won't touch RBS |
| 18:41.26 | Cresl1n | bkw_: that's a convenient answer |
| 18:41.33 | Cresl1n | :-D |
| 18:41.36 | ManxPower | bkw_, There is ALOT of RBS circuits out there. |
| 18:41.43 | bkw_ | ManxPower, not on my network |
| 18:41.45 | brettnem | bkw_: i tend to agree with you |
| 18:41.57 | bkw_ | brettnem, you know where i'm coming from |
| 18:41.58 | ManxPower | Every time we do a new Asterisk install, we consider Sangoma. |
| 18:42.07 | brettnem | I only use RBS if I'm talking to something really old and outdated.. |
| 18:42.14 | brettnem | bkw_: yep.. however... |
| 18:42.29 | brettnem | My Choke circuits are always MF |
| 18:42.30 | mog_work | i mean you can do rotary ^_^ |
| 18:42.37 | bkw_ | lame |
| 18:42.43 | mog_work | so my grandma can call in |
| 18:42.43 | brettnem | not my choice.. you know |
| 18:42.47 | iCEBrkr | brettnem: Our current system uses CAS cuz some knuckle-head figured using all 24 channels was a good thing. Or it could be just old-skewl |
| 18:42.48 | bkw_ | ya |
| 18:42.52 | brettnem | so your grandma can call the radio |
| 18:42.54 | brettnem | station |
| 18:43.07 | brettnem | iCEBrkr: heh, so use SS7 then.. :P |
| 18:43.13 | brettnem | or NFAS.. even digium can do NFAS |
| 18:43.15 | brettnem | ;) |
| 18:43.18 | iCEBrkr | brettnem: pfft, no.. |
| 18:43.24 | ManxPower | iCEBrkr, many PBXs charge extra for PRI support (Nortel does, you need a PRI card AND a key to unlock PRI support in the Nortel software) |
| 18:43.26 | Cresl1n | even digium.... |
| 18:43.34 | iCEBrkr | I'm just gonna get new lines and have them provisioned for PRI |
| 18:43.43 | Cresl1n | that's pretty funny... considering the sangoma stuff uses our nfas code |
| 18:43.53 | bkw_ | honestly you'll never see much of an issue if you only got a single T1 |
| 18:43.57 | bkw_ | but push that up in density |
| 18:43.59 | mog_work | ssh cresl1n we cant stop the sangoma praising |
| 18:44.02 | bkw_ | thats where you start to see issues |
| 18:44.20 | brettnem | Cresl1n: really competition will make your product better.. mark my words! ;) |
| 18:44.23 | Cresl1n | (and our pri code for that matter) |
| 18:44.27 | Cresl1n | just sayin |
| 18:44.29 | Cresl1n | :-) |
| 18:44.35 | many | brettnem: its a japanese saying. |
| 18:44.43 | Cresl1n | haven't seen libsangomapri yet |
| 18:44.47 | brettnem | many: really? |
| 18:44.47 | bkw_ | Cresl1n, so |
| 18:44.52 | bkw_ | we have visdn |
| 18:44.52 | many | brettnem: hehe. :) |
| 18:44.54 | mog_work | lol |
| 18:45.03 | mog_work | wow way to change topics smoothly..... |
| 18:45.22 | bkw_ | see now there it goes useless.. MY CODE, OUR CODE, THEIR CODE shit |
| 18:45.26 | bkw_ | its just code |
| 18:45.26 | brettnem | well you say "sangoma" in here and things will start flyin |
| 18:45.28 | bkw_ | use what works |
| 18:45.30 | bkw_ | move on |
| 18:45.39 | mog_work | you brought it up? |
| 18:45.39 | brettnem | bkw_: hey I thought it was a community project? |
| 18:45.42 | bkw_ | I could care less what I use as long as it works |
| 18:45.53 | drumkilla | bkw_: I think we'd all appreciate it if you moved on. |
| 18:46.27 | gniretar | sangoma |
| 18:46.44 | *** join/#asterisk zpn (n=xpn@gateway.digium.com) |
| 18:46.47 | brettnem | I actually apprecaite bkw_'s comments and contributions.. his bitching about it will eventually lead to better products from both companies |
| 18:47.07 | bkw_ | I'm not going to let my opinion and position be compromised by anyone. I have my own mind and can clearly voice my own opinion. |
| 18:47.17 | Cresl1n | sure, it helps to get facts correct too though :-) |
| 18:47.17 | bkw_ | if you don't like me that sfine |
| 18:47.17 | brettnem | word |
| 18:47.23 | mog_work | i thought that is why you left? |
| 18:47.33 | brettnem | wtf |
| 18:47.34 | iCEBrkr | bkw_: Dude, this is IRC, that type of shit is blasphemy |
| 18:47.36 | gniretar | bkw_ is gety outspoken but has helped lost of us many times |
| 18:47.53 | bkw_ | yep and don't mind continuing my help |
| 18:47.55 | gniretar | very* |
| 18:47.59 | bkw_ | we still use and support asterisk daily |
| 18:48.13 | brettnem | really.. I don't understand this hostility towards bkw_.. it's freakin rediculous |
| 18:48.18 | mog_work | i bet you do, i mean you run a business |
| 18:48.19 | bkw_ | but i'm not going to let some childish behavior by a select few turn me away. |
| 18:48.31 | Cresl1n | brettnem: believe me, it's gone both ways |
| 18:48.32 | gniretar | oh the drama |
| 18:49.10 | bkw_ | anyway |
| 18:49.16 | brettnem | right |
| 18:49.23 | bkw_ | drumkilla, its nice to see how you speak for everyone these days. |
| 18:49.51 | mog_work | hows that go cresl1n |
| 18:50.00 | docelmo | does DBget not push n+101 anymore in Asterisk 1.2? |
| 18:50.07 | docelmo | if the entry isnt found? |
| 18:50.16 | brettnem | docelm0: +101ing is being eliminated |
| 18:50.17 | [TK]D-Fender | docelm0 : seems to... |
| 18:50.21 | Cresl1n | docelmo: I think we're trying to deprecate n+101 |
| 18:50.22 | mutilator | yes it goes on and on my friends |
| 18:50.23 | docelmo | damnit! |
| 18:50.28 | mog_work | you have you use j option |
| 18:50.30 | brettnem | bkw_: nice! haha |
| 18:50.33 | docelmo | j option? |
| 18:50.38 | Gennaro | bye tanx |
| 18:50.44 | docelmo | dbGet(VAR|j) |
| 18:50.47 | docelmo | ?? |
| 18:50.47 | mog_work | head on over to the east side |
| 18:51.01 | bkw_ | To a deluxe apartment in the sky. |
| 18:51.18 | mog_work | yeah so go... ^_^ |
| 18:51.36 | iCEBrkr | Stoners! |
| 18:51.39 | many | oh. this is westcoast vs eastcoast? |
| 18:51.46 | funxion | lol |
| 18:51.55 | brettnem | yay.. I'm third coast |
| 18:52.01 | mog_work | third cost? |
| 18:52.06 | *** join/#asterisk spackle (n=spackle@209.234.83.19) |
| 18:52.06 | brettnem | texas |
| 18:52.10 | funxion | Im south coast |
| 18:52.16 | docelmo | sigh.. thanks mog! |
| 18:52.18 | brettnem | funxion: where? |
| 18:52.20 | anthm | hmm looks like more activity in here since bkw joined 5 min ago that there was in the last 4 hours lol |
| 18:52.22 | funxion | Miami |
| 18:52.28 | funxion | yup |
| 18:52.29 | docelmo | Tampa BAY BABY! |
| 18:52.30 | funxion | lol |
| 18:52.31 | brettnem | oh that is south |
| 18:52.34 | mog_work | we have lulls |
| 18:52.35 | Cresl1n | anthm: howdy, long time no see |
| 18:52.44 | mog_work | its like asterlkink day in here |
| 18:52.52 | funxion | Im only like 180 miles from the southern most point |
| 18:52.53 | iCEBrkr | docelmo: you said you're 6'6" |
| 18:52.55 | iCEBrkr | right? |
| 18:52.55 | anthm | hi |
| 18:53.08 | mutilator | pop vs soda vs coke! |
| 18:53.18 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 18:53.20 | iCEBrkr | soda! |
| 18:53.23 | mutilator | pop! |
| 18:53.43 | funxion | lol @icebrkr |
| 18:53.52 | iCEBrkr | :) |
| 18:54.09 | funxion | Im gonna be in clevveland and akron next week |
| 18:54.13 | *** part/#asterisk spackle (n=spackle@209.234.83.19) |
| 18:54.23 | iCEBrkr | funxion: Wow that sucks |
| 18:54.32 | funxion | going to see family |
| 18:54.32 | *** join/#asterisk heison (n=heison@gw-yyz.somanetworks.com) |
| 18:54.35 | iCEBrkr | funxion: I'm from Cleveland.. Dated a chick in Akron |
| 18:55.00 | funxion | see when you live in a place that attracts a lot of touristyou go to BFE for vacation |
| 18:55.22 | funxion | I went to university of akron and dates a chick in clevland |
| 18:55.26 | docelmo | yes |
| 18:55.30 | iCEBrkr | funxion: haha |
| 18:55.39 | docelmo | Ask any one from astricon that met me.. :) |
| 18:55.51 | docelmo | Hay mog you said J right? Well I got this: -- Executing DBget("Zap/23-1", "DID=CFIM/302|j") in new stack |
| 18:55.56 | *** join/#asterisk bkirby (n=bkirby@207.111.160.33) |
| 18:56.11 | docelmo | it didnt recgonize the J |
| 18:56.11 | funxion | I wish I would have went to astricon sukd I was in finland |
| 18:56.25 | mog_work | yeah all 101 is gone but lowercase j is th jump priority most things have it |
| 18:56.36 | mog_work | you can also enable it in dial plan general section i think |
| 18:56.39 | docelmo | dbGet has it but its not working |
| 18:56.47 | hugov6 | many: here? |
| 18:57.07 | docelmo | Anyone know the command to enable n+101 in the General section of the dialplan? |
| 18:57.21 | mog_work | its in the samples |
| 18:57.23 | docelmo | priorityjumping isnt it? |
| 18:57.26 | mog_work | but i dont rember it |
| 18:57.27 | hugov6 | well looks so *query* |
| 18:57.28 | mog_work | something like that |
| 18:57.43 | many | uhoh |
| 18:57.51 | hugov6 | ;)) |
| 18:58.31 | *** join/#asterisk Tili (i=Tili@202-133-67-41-dialup.sat.net.pk) |
| 18:59.32 | [TK]D-Fender | . |
| 18:59.58 | iCEBrkr | . |
| 19:00.02 | hugov6 | , |
| 19:00.27 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 19:00.38 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 19:02.00 | *** join/#asterisk crash3m (i=crash3m@unaffiliated/crash3m) |
| 19:02.44 | bkirby | has anyone ran into deadlock issues with channel_find_locked, causing call processing to be temporarily locked |
| 19:06.55 | mog_work | nope |
| 19:07.18 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 19:08.04 | anthm | channel_find_locked is like spinning a roluette wheel =D |
| 19:08.28 | many | russian roulette? |
| 19:09.00 | MRH2 | hi is svn branch 1.2 the same as cvs v1-2 and trunk is the same as head |
| 19:10.19 | mog_work | at the moment |
| 19:10.20 | mog_work | mostly |
| 19:10.23 | mog_work | but it will diverge |
| 19:10.30 | mog_work | 1.2 is the new stable branch |
| 19:10.35 | mog_work | in either cvs or svn |
| 19:10.41 | mog_work | cvs is not updated as often |
| 19:10.50 | mog_work | trunk is what head was |
| 19:11.06 | MRH2 | so what is the equivalent on cvs v1-2-0 |
| 19:11.20 | MRH2 | on svn |
| 19:11.47 | mog_work | http://svn.digium.com/view/asterisk/tags/1.2.0/ |
| 19:12.00 | bkirby | i've got 3 servers handling about 100 calls at anygiven time, all reinvited between endpoints, and every couple of minutes, gets a deadlock avoidance, and call processing stops, then frees up and gets many duplicated reinvites for calls that have went away due to the timeframe |
| 19:12.39 | bkirby | these 3 are doing routing from Level3 to customer gateways |
| 19:12.41 | MRH2 | ok thanks |
| 19:13.00 | mog_work | no prob |
| 19:15.12 | *** join/#asterisk bpm- (n=spi@ool-18bcb836.dyn.optonline.net) |
| 19:15.58 | bpm- | anyone know how to change the outgoing smtp server and port asterisk uses to send emails |
| 19:16.32 | anthm | yep make a wrapper and name it sendmail eat the file into memory and deliver it anywhere you please |
| 19:18.33 | *** join/#asterisk backblue (n=moo@87-196-40-135.net.novis.pt) |
| 19:18.35 | anthm | if that makes you say "hmm how do i do that?" here is an example http://www.sofaswitch.org/oldeg/sendmail.pl |
| 19:18.36 | bpm- | thanks but do you know where i can get specific info on it |
| 19:18.42 | bpm- | aha ok |
| 19:18.54 | funxion | nice anthm |
| 19:19.01 | funxion | esp |
| 19:20.10 | *** join/#asterisk crash3m (i=crash3m@unaffiliated/crash3m) |
| 19:20.30 | anthm | you probably would need to add a line or 2 to change the port too |
| 19:20.50 | *** join/#asterisk dooder (n=nate@63.105.29.30) |
| 19:21.33 | *** join/#asterisk trixter (n=trixter@65.172.209.246) |
| 19:25.10 | *** join/#asterisk heath__ (n=root@12-215-32-56.client.mchsi.com) |
| 19:26.12 | EriSan | what is the prefered linux flavor to run * ? |
| 19:28.01 | Nugget | it doesn't matter in the least. |
| 19:29.46 | EriSan | well, i'm not a linux guru, and just want to start with *, a complete step by step guide would be nice |
| 19:30.06 | mog_work | go get debian |
| 19:30.12 | mog_work | and come here as you have questions |
| 19:31.13 | [TK]D-Fender | EriSan : Use whatever you are comforatble with. Odds are * will run just fine on it |
| 19:32.23 | *** join/#asterisk JCC_ (n=chatzill@207.41.92.131) |
| 19:32.28 | [TK]D-Fender | EriSan : You'll find a tonof opinions as to what someone things is the best. Its a question of whats best for you. |
| 19:32.47 | xkev | asterisk@home good for quick to test |
| 19:32.48 | EriSan | is mysql required? |
| 19:33.01 | [TK]D-Fender | EriSan : For instance, I suck at SysV inits, vi, emacs, etc so therefor I choose Slackware :) |
| 19:33.04 | *** join/#asterisk twisted[asteria] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 19:33.04 | *** mode/#asterisk [+o twisted[asteria]] by ChanServ |
| 19:33.08 | [TK]D-Fender | EriSan : definately not required |
| 19:33.18 | [TK]D-Fender | EriSan : Entirely optional. |
| 19:34.53 | *** join/#asterisk wunderkin (i=kev@12-201-105-27.client.mchsi.com) |
| 19:36.22 | rjuan | is asterisk@home as good as asterisk?? |
| 19:36.44 | Nugget | it is asterisk. so. . . |
| 19:36.57 | xkev | it's just all the crap you could do by yourself, rolled up in a ball |
| 19:37.10 | xkev | good for simple pbx systems methinks |
| 19:37.16 | [TK]D-Fender | rjuan : A@H *is* Asterisk... running on a ready-made setup including MySQL, and a few other resources. However it is just a "cover" over the Asterisk core |
| 19:38.21 | rjuan | ok thanks |
| 19:38.47 | kuku5 | When I dial an extension - i only get the word :answer: on the phone, how do i "reject" the call? |
| 19:38.57 | [TK]D-Fender | And then the A@H GUI (which is just AMP) of course makes * look "easy" by letting you web-configure most of everything BUT you lose the real power that is * because once you hand your config files over to AMP.... d00m! |
| 19:39.26 | [TK]D-Fender | A@H, AMP, and anything that tries to control * = SUCK |
| 19:39.39 | Nugget | as is mysql. :) |
| 19:39.45 | [TK]D-Fender | Just learn to do it yourself and you will be SO much ahppier in the end |
| 19:40.04 | xkev | fender, but when you don't want to spend a month straight cooking up a nice dialplan, and need newbs to admin it, AMP == aight |
| 19:40.16 | rjuan | my problem is tha I have running * with debian packages |
| 19:40.18 | xkev | agreed that shit is not for the ninjas |
| 19:40.28 | [TK]D-Fender | xkev : thats why there are sample files :) I offer mine up as a "ready in 10 minutes" offer.... |
| 19:40.34 | rjuan | but I can't load chan_zap |
| 19:40.49 | xkev | my office pbx is like 15 files and 5000 lines or so, not for the faint of heart :) |
| 19:40.52 | rjuan | taht why i'm hinking in oder possibilities |
| 19:40.53 | [TK]D-Fender | rjuan : recompile Zaptel, THEN recompile asterisk |
| 19:41.04 | [TK]D-Fender | should be fine. What distro you running? |
| 19:41.12 | rjuan | debian |
| 19:41.20 | rjuan | well ubuntu |
| 19:41.46 | xkev | mmm ubuntu |
| 19:41.51 | rjuan | it's what i've got in the work |
| 19:42.11 | rjuan | i have another computer with debian |
| 19:42.13 | rjuan | sarge |
| 19:42.22 | xkev | ubuntu == same thing, sucks less |
| 19:42.25 | xkev | and 'just works' |
| 19:42.44 | MRH2 | updated with svn and get the error on the console at startup iaxy.bin fails checksum ? |
| 19:42.56 | funxion | rjuan are you using the apt-get packages for *? |
| 19:42.59 | kuku5 | I have music on hold ( using the default mp3's ) and the music is muffled! |
| 19:43.03 | rjuan | yes |
| 19:43.09 | kuku5 | Anyone have a clue why ? |
| 19:43.11 | funxion | xkev I dont think debian sucks at all |
| 19:43.35 | funxion | rjuan remove the apt-get packages and make from cvs |
| 19:43.48 | xkev | funxion we use debian on hundreds of servers, it does not suck |
| 19:43.56 | *** join/#asterisk MstlyHrmls (n=mh@melbourne.mostly-harmless.ca) |
| 19:44.12 | xkev | ubuntu > vanilla deb for desktop, and I'm liking using it on my personal servers etc -- more up to date |
| 19:44.15 | funxion | ok was just commecnt on "<xkev> ubuntu == same thing, sucks less" |
| 19:44.41 | funxion | xkev I use ubuntu for desktop |
| 19:44.45 | funxion | deian for server |
| 19:44.56 | rjuan | i also do it |
| 19:45.01 | rjuan | like funxion |
| 19:45.06 | xkev | $debian_suck = -1; $rookhat_suck = -20; $sucks_less = 1; $ubuntu = $debian_suck + $sucks_less; :) |
| 19:45.17 | funxion | omg |
| 19:45.31 | ManxPower | And Mandrake, of course, is better than all of them |
| 19:45.50 | rjuan | i never use mandrake |
| 19:45.57 | funxion | mandrake=redhat remix |
| 19:46.13 | funxion | lol |
| 19:46.22 | drray | I like redhat over mandrake, but I'm digging ubuntu |
| 19:46.40 | drray | I've not moved any of my servers from FC to ubuntu though |
| 19:46.53 | rjuan | I dont have any problems with ubuntu |
| 19:47.03 | xkev | <PROTECTED> |
| 19:47.06 | rjuan | it's for desktops |
| 19:47.24 | drray | ubuntu has grown on me |
| 19:47.44 | drray | they are getting ubuntu certified for servers too |
| 19:47.46 | [TK]D-Fender | Slackware : Tried & true... greater > ALL! |
| 19:47.47 | Nugget | ubuntu is an african word that means "I couldn't figure out debian" |
| 19:47.48 | [TK]D-Fender | ;) |
| 19:47.53 | mutilator | drray: try showering? |
| 19:47.55 | *** join/#asterisk newl (n=newlook@203-59-61-62.dyn.iinet.net.au) |
| 19:48.09 | drray | ubuntu has a hippy feel to it |
| 19:48.13 | marcus2 | sweet |
| 19:48.19 | xkev | nugget, nice |
| 19:48.22 | Nugget | I am thrilled to death to see that DB2 supports ubuntu, though. Hopefully some other vendors will follow suit. |
| 19:48.22 | marcus2 | i can make at least 28.8 modem calls thru my asterisk server |
| 19:48.31 | [TK]D-Fender | Nugget : I love that quote... maybe I'll use it soon... |
| 19:48.31 | funxion | lol @nugget |
| 19:48.33 | mutilator | Druken: change the fsckin song! this mega mix has been playing for an hour or something |
| 19:48.42 | *** join/#asterisk amir_ (n=amir@gentoo/developer/amir) |
| 19:48.47 | Nugget | I hate the redhat/suse extortion ring that you have to buy into if you want to run oracle or get vendor support for your hardware |
| 19:49.10 | drray | how is it extortion? |
| 19:49.23 | Nugget | with redhat at least they lock you in to buying support with a nasty eula. |
| 19:49.30 | Nugget | suse is pretty much the same way |
| 19:49.54 | Nugget | but you have to buy the redhat crap, otherwise you lose support from your hardware or platform vendor (like oracle) |
| 19:50.04 | Nugget | it's a total scam |
| 19:50.22 | moonwick | look at it this way: it proves that business is smarter than RMS |
| 19:50.28 | Nugget | heh |
| 19:50.46 | moonwick | they took a pile of GPL stuff and managed to subvert RMS's goals almost entirely |
| 19:50.55 | Nugget | I spend thousands of dollars a year to redhat for crap I don't want or need, all so I can continue to call oracle if my database has problems. |
| 19:51.25 | kuku5 | I have music on hold ( using the default mp3's ) and the music is muffled! Anyone know why ? im using mpg321 |
| 19:51.29 | Nugget | and I'm subject to audits by redhat and they can fine me penalties if they audit me and I'm running their stuff without support |
| 19:51.36 | iDunno | the solution is to not use oracle. |
| 19:51.44 | iDunno | and to realise that oracle is EVIL |
| 19:51.54 | Nugget | well, DB2 and Dell have the same constraints. |
| 19:52.09 | moonwick | evil(oracle) < evil(mysql) |
| 19:52.17 | *** join/#asterisk cronek (n=cronek@200.157.100.35) |
| 19:52.19 | *** part/#asterisk azfhasterisk (n=rickb@VDSL-130-13-192-106.PHNX.QWEST.NET) |
| 19:52.28 | Nugget | at least oracle is evil on purpose. mysql is evil by way of incompetence. :) |
| 19:52.54 | [TK]D-Fender | MPG321 is your problem. You need MPG123 0.59r specifically |
| 19:53.06 | [TK]D-Fender | kuku5 : ^^^ |
| 19:53.11 | *** join/#asterisk moot (n=irssi@209.168.254.59) |
| 19:53.21 | cronek | is there VAD on iax/sip channels? |
| 19:53.33 | kuku5 | fuc |
| 19:53.34 | MRH2 | oracle can;t be evil see larry at http://www1.us.dell.com/content/topics/global.aspx/corp/media/en/tech_force_noemail?c=us&l=en&s=corp |
| 19:53.35 | [TK]D-Fender | moonwick : PostgreSQL! |
| 19:53.38 | *** join/#asterisk santiago (n=santiago@208.195.215.160) |
| 19:53.44 | kuku5 | [TK]D-Fender: so mpg 123 |
| 19:53.50 | [TK]D-Fender | kuku5 : Corrent |
| 19:53.59 | anthm | native moh is still not default, sigh, seems like i made it for no reason =( |
| 19:54.06 | moonwick | [TK]D-Fender: good man. |
| 19:54.13 | [TK]D-Fender | uninstall MPG321 nad compile or download the right version. Its typically included with the * source. |
| 19:54.57 | [TK]D-Fender | anthm : Well I've heard there have been some unfortunate hurdles for it to be considered "functional". Have these been resolved? |
| 19:55.10 | drray | I need to con work into buying me a lego mindstorm kit |
| 19:55.10 | [TK]D-Fender | (I'm not the most up to date person...) |
| 19:55.20 | drray | :) |
| 19:55.33 | slayer192 | 'make mpg123' from within the * source dir will compile and build the correct version |
| 19:56.03 | cronek | talking about moh: I'm having very chopped up MOH, but the ztdummy module is installed and loaded, can this be because of a wrong mpg123 version? |
| 19:56.08 | anthm | nobody has told me any, but there has been a pretty good amount of my stuff changed w/o consulting me so could be |
| 19:56.14 | cronek | or is * not using my ztdummy like it should? |
| 19:56.17 | p1tst0p | can you add/delete extentions/queues/agents through manager ? |
| 19:56.29 | [TK]D-Fender | anthm : :/ |
| 19:56.55 | [TK]D-Fender | I'd love to use native MOH and everything else that can be done in-core where possible.... |
| 19:56.56 | *** join/#asterisk mkrufky (n=mk@68.160.103.77) |
| 19:57.14 | slayer192 | cronek: if you do a lsmod do you see zaptel and ztdummy loaded? |
| 19:57.18 | cronek | yep |
| 19:57.55 | slayer192 | how did you build mpg123? |
| 19:58.24 | cronek | I was thinking it could be an issue with VAD somewhere messing up the timing of my RTP channel |
| 19:58.33 | cronek | its debian stable stylee |
| 19:58.49 | anthm | that's another issue native moh was there to address, cos it doesnt need zaptel |
| 19:59.08 | cronek | Version 0.59r |
| 19:59.42 | cronek | yeah the zaptel requirement is a pain in the butt |
| 19:59.48 | anthm | there is also rawplayer.c in the code tree |
| 20:00.20 | anthm | you can decompress them all to raw once and play them from the disk instead of doing it with the cpu over and over for no reason |
| 20:00.31 | cronek | but does * do VAD on SIP and AIX channels? |
| 20:00.40 | cronek | because if it does I would like to turn it off |
| 20:00.52 | anthm | but native moh + .raw or .ul is the best option |
| 20:00.56 | p1tst0p | # DBGet: Get DB Entry (Privilege: system,all) |
| 20:01.02 | p1tst0p | sorry, |
| 20:01.32 | slayer192 | anthm: I'll have to try rawplayer out |
| 20:01.38 | [TK]D-Fender | AIX channels? Oh no! SCO is going to sue Digium for receiving tainted code from IBM!!!! |
| 20:01.49 | cronek | hehe |
| 20:01.52 | cronek | IAX ofcourse |
| 20:02.00 | p1tst0p | can you add/delete extentions/queues/agents through manager api ?? |
| 20:02.05 | [TK]D-Fender | forgive my moment of hillarity :) |
| 20:02.14 | cronek | I'll do my best :) |
| 20:02.21 | [TK]D-Fender | p1tst0p : Somewhat. Dynamic queues are funny things... |
| 20:02.29 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com) |
| 20:02.40 | p1tst0p | [TK]D-Fender, what about extentions/agents ? |
| 20:02.43 | cronek | anyway I don't want any kind of silence detection anywhere |
| 20:02.46 | *** join/#asterisk tracinet (n=tracinet@216.242.235.2) |
| 20:02.52 | xheliox | Are there any extension commands that can check the channel type (Zap, IAX, SIP, etc) and goto various extensions based on which one it's using? |
| 20:03.11 | p1tst0p | [TK]D-Fender would u have to use realtime, and DBPut ? |
| 20:03.13 | kuku5 | [TK]D-Fender: ok - i did that but now the mpg123 doenst want to start |
| 20:04.07 | tracinet | anyone know if the asterisk CLI has the linux equivalent to " | more " to pause scrolling text for example "show modules | more" ? |
| 20:05.29 | anthm | go back to bash and say # asterisk -rx "show modules" | more =p |
| 20:05.33 | [TK]D-Fender | p1tst0p : not sure really, I do mine all hard-coded... |
| 20:05.45 | moot | m |
| 20:05.48 | moot | oops |
| 20:06.08 | tracinet | anthm: DUH! thanks! |
| 20:06.20 | anthm | np |
| 20:06.36 | p1tst0p | [TK]D-Fender, hardcoded how ?, what i,m thinking of doing is, using php, to add/delete agents/extentons/queues thru manager api.. instead of editing .conf's through php.. |
| 20:07.10 | moot | anyone have experience with pri and * here ? |
| 20:07.19 | bpm- | anthm: that perl script for email redirect doesnt see to work |
| 20:07.33 | devel | moot, sure, some basic experience. |
| 20:07.41 | moot | im getting a bunch of these on the console |
| 20:07.41 | moot | Dec 1 14:07:14 NOTICE[2167]: chan_zap.c:7428 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 |
| 20:07.53 | moot | any idea what would cause that? |
| 20:07.58 | devel | yep, i get a lot of those too, and i ignore them. |
| 20:08.05 | moot | im not able to receive/send any calls |
| 20:08.10 | moot | * just drops outgoing calls |
| 20:08.12 | anthm | write a sample email into a .txt file and cat it | sendmail |
| 20:08.13 | [TK]D-Fender | p1tst0p : Been done before... its called AMP, ScopServe, ThirdLane, etc.... |
| 20:08.38 | devel | ah, moot, i am able to call in and out, so i don't think the abort is cause for concern. |
| 20:08.48 | moot | <PROTECTED> |
| 20:08.48 | moot | Dec 1 14:08:14 NOTICE[5480]: app_dial.c:764 dial_exec: Unable to create channel of type 'Zap' |
| 20:08.51 | moot | <PROTECTED> |
| 20:08.54 | moot | <PROTECTED> |
| 20:08.55 | moot | that's what I get |
| 20:08.57 | moot | <PROTECTED> |
| 20:09.08 | p1tst0p | [TK]D-Fender, i know its been done before, but i want my own.. personalised |
| 20:09.18 | anthm | likely causes in order of likelyhood , permissions problem, you dont have perl, the ip of your gateway is not reachable, you typed a typeo in the file when you were editing it |
| 20:09.26 | moot | and pri show span 1 says this |
| 20:09.26 | moot | Primary D-channel: 24 |
| 20:09.26 | moot | Status: Provisioned, Down, Active |
| 20:09.26 | moot | Switchtype: National ISDN |
| 20:09.27 | moot | Type: CPE |
| 20:09.42 | devel | that certainly doesn't sound good, moot. |
| 20:09.45 | moot | telco says they are not seeing our b channels |
| 20:09.48 | moot | just the d |
| 20:10.08 | devel | what is your /etc/zaptel.conf show for bchan and dchan? |
| 20:10.17 | moot | bchan=1-23 and dchan=24 |
| 20:11.34 | devel | and telco verified that your framing and coding are correct? |
| 20:11.37 | moot | yes |
| 20:12.13 | anthm | pridialplan and switchtype both right that is the 2 most common ones to mess things up |
| 20:13.04 | moot | switchtype is right, what should pridialplan be |
| 20:13.11 | *** join/#asterisk PupenoL (n=pupeno@200.123.183.89) |
| 20:13.18 | anthm | start with "unknown " |
| 20:13.27 | anthm | "unknown" |
| 20:13.48 | anthm | possibly national |
| 20:13.54 | *** join/#asterisk cnet2 (n=jjohn@201.192.107.58) |
| 20:13.59 | *** join/#asterisk fifty5-50 (n=witheld@cpc4-newe1-6-1-cust4.cdif.cable.ntl.com) |
| 20:13.59 | anthm | that is the only 2 i have ever needed |
| 20:14.24 | anthm | that is moreso for outbound but it being wrong will break it |
| 20:14.27 | devel | our switch is talking ni1 |
| 20:14.57 | devel | but other than that, all looks similar, so that's about all the more i can help. |
| 20:15.15 | anthm | telco's like to lie too |
| 20:15.23 | zigman | does anyone have that url for the gsm fxo devices for 60 pounds handy ? |
| 20:15.23 | moot | but would a wrong dialplan cause * to say |
| 20:15.23 | moot | Dec 1 14:14:53 NOTICE[5516]: app_dial.c:764 dial_exec: Unable to create channel of type 'Zap' |
| 20:15.34 | anthm | when i have that kind of issue i tend to try every answer to the switchtype |
| 20:15.56 | anthm | more than once it worked with some "incorrect" value |
| 20:16.01 | moot | i tried both national and unknown dialplan and it doesn't help |
| 20:16.08 | devel | moot, until it's "up", it won't be able to create a channel on it. |
| 20:16.26 | moot | devel: so how can I find out why it's not up |
| 20:16.45 | moot | |
| 20:16.50 | moot | er |
| 20:16.59 | moot | telco can't really help me much |
| 20:17.02 | moot | they don't know what * is |
| 20:17.03 | moot | lol |
| 20:17.08 | *** join/#asterisk Meaty (n=cp_simbu@office.abi.ca) |
| 20:17.10 | anthm | they love it |
| 20:17.21 | anthm | "what is your equipment so we can blame them? |
| 20:17.48 | moot | so what can I do on my end to troubleshoot why its showing down in pri show span |
| 20:17.48 | anthm | do you have green on the physical layer ? |
| 20:17.53 | moot | how do I check that |
| 20:18.18 | anthm | zttool shwing all OK |
| 20:19.32 | devel | as i said, i see those HDLC aborts all the time, and everything (seems) to work fine on my switch. |
| 20:19.42 | moot | Current Alarms: No alarms. |
| 20:19.42 | moot | Sync Source: T2XXP (PCI) Card 0 Span 1 |
| 20:19.42 | moot | IRQ Misses: 0 |
| 20:19.42 | moot | Bipolar Viol: 0 |
| 20:19.42 | moot | Tx/Rx Levels: 0/ 0 |
| 20:19.44 | moot | Total/Conf/Act: 24/ 24/ 0 |
| 20:20.23 | anthm | you should play with the switchtype |
| 20:20.33 | anthm | just try them all till 1 works |
| 20:20.46 | anthm | cos telco will never tell you the truth |
| 20:20.46 | *** join/#asterisk johnnyq (n=johnnyq@cr-5.pitdc1.pa.stargate.net) |
| 20:21.02 | anthm | someone pays them to lie to you |
| 20:21.09 | moot | lol |
| 20:21.13 | moot | they told me it was NI2 |
| 20:21.16 | moot | thats what I used |
| 20:21.19 | moot | but I'll try others |
| 20:21.31 | moot | i tried dms100 and that didnt work |
| 20:21.59 | xkev | ask them to set you to national |
| 20:22.08 | xkev | they can usually configure it |
| 20:22.24 | xkev | I get hdlc aborts often as well, but only on certain spans |
| 20:22.34 | johnnyq | Hi everyone I was wondering if someone could help me setup or point me in the right direction on setting up an automated call server using asterisk@home SIP Trunk that does 2 things dials a list of number from a database and plays a recorded message then goes on to the next number |
| 20:22.39 | anthm | or any of the ones that are compat with dms100 |
| 20:23.14 | brettnem | johnnyq: check the wiki for an app that does just that.. can't remember the name |
| 20:23.23 | lenne_dk | You aren't a voice-spammer? :-) |
| 20:23.31 | xkev | ni2 lets you get facility IE with cnam |
| 20:23.33 | brettnem | johnnyq: and don't expect to make any friends like that |
| 20:23.49 | slayer192 | anthm: just got rawplayer installed and the default mp3s converted. Everything just works with one issue, the audio quality is a bit choppy |
| 20:23.54 | xkev | ..if your telco will do it. xo did it for me. |
| 20:24.23 | johnnyq | no im not a spammer we will use it alert our customers of downtime |
| 20:24.24 | anthm | try it on native and compare |
| 20:24.40 | anthm | i think rawplayer still gets timed off of zap |
| 20:24.50 | johnnyq | brettnem: i heard of gnudialer but i think that does something different |
| 20:24.56 | brettnem | no, not that |
| 20:25.02 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 20:25.05 | lenne_dk | I hope you don't get downtime in the middle of the night... |
| 20:25.05 | brettnem | johnnyq: vicidial.. check that |
| 20:25.17 | brettnem | lenne_dk: is that a hex? |
| 20:25.35 | xkev | johnnyq, create a context to speak the infos, write a perl script or whatever to create call files in /var/spool/asterisk/outgoing |
| 20:25.37 | lenne_dk | Hex? A spell? |
| 20:25.45 | *** join/#asterisk twisted (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 20:25.45 | *** mode/#asterisk [+o twisted] by ChanServ |
| 20:26.01 | *** join/#asterisk eKo1 (n=bernd@metrored-gw.tropicohn.com) |
| 20:26.55 | johnnyq | xkev: good idea, i just dont know how to program in perl |
| 20:26.56 | lenne_dk | Brettnem, enlighten me, what do you mean? |
| 20:27.08 | *** join/#asterisk puzzled_ (n=patrick@puzzled.xs4all.nl) |
| 20:27.11 | *** join/#asterisk gr8nash (n=andy@mamabear.si-forest.com) |
| 20:27.13 | devel | so maybe somebody can tell me if asterisk has the concept of a "lifetime" variable. i.e. something like a counter that i can increment the value of with every call (to a specific extension) and access it's value. |
| 20:27.20 | xkev | well you need to get your numbers from a db. write whatever you know that will pull a row at a time and create call files to dial that number |
| 20:27.48 | xkev | if that is beyond you, you should hire a consultant |
| 20:27.53 | johnnyq | xkev: I was hoping for something easy, but it doesnt look like its going to be tat way, I already spent 3 weeks researching it, and dont feel like im getting anywhere |
| 20:28.04 | xkev | it's a piece of cake, really |
| 20:28.16 | johnnyq | xkev |
| 20:28.23 | lenne_dk | My fridge calls me if it gets to hot. |
| 20:28.32 | brettnem | lenne_dk: nothing I was kidding |
| 20:28.45 | LostFrog | My gf calls if she gets too lonely. |
| 20:28.49 | anthm | can you get it to call when there is not enough beer? |
| 20:29.02 | johnnyq | xkev: yeah thats what i was thinking but i was the one that got handed down the project and am expected to get everything working |
| 20:29.22 | anthm | or like order some more beer for you and the guy just arrives at the door |
| 20:29.22 | johnnyq | wouldnt there already be a product out there that will do that for me |
| 20:29.39 | lenne_dk | What kind of db, johnny? |
| 20:30.11 | johnnyq | a basic text file would be fine, but if it could connect to oracle that would be great |
| 20:30.17 | LostFrog | deny: SetGroup? |
| 20:30.28 | gr8nash | anyone know how easy it is to have * let you know that a person is on do not disturb? instead of saying they are on the phone |
| 20:30.30 | gr8nash | ? |
| 20:30.30 | LostFrog | devel, even. |
| 20:31.22 | xkev | johnnyq you could do it in a shell script |
| 20:31.33 | xkev | with a text file |
| 20:31.35 | lenne_dk | Johnny: it's a piece of cake, really :-) Do you know how to make a list of numbers from your oracle? |
| 20:31.52 | johnnyq | im really not the database admin |
| 20:32.03 | johnnyq | so i would have to have our db programmer do that |
| 20:32.14 | johnnyq | I know just enought to get around |
| 20:32.14 | lenne_dk | Divide and conqueer... |
| 20:32.46 | johnnyq | xkev: i am familiar with shell scripting |
| 20:32.59 | lenne_dk | Do what you know and get somebody else to do the rest :-) |
| 20:33.47 | lenne_dk | I'm thinking and will put something in pastebin. |
| 20:34.09 | xkev | "programmer dude: write me a shell/perl/python/ruby/php/pascal/cobol/fortran script and/or program to create a file in this > < format (produce sample call file) one at a time, popping the top of the list each time" |
| 20:34.13 | *** join/#asterisk volkerli (n=volkerli@port-212-202-210-112.dynamic.qsc.de) |
| 20:34.14 | johnnyq | lenne_dk: I think im just going to have it pull the number from a text file |
| 20:34.33 | johnnyq | brb have to take a call |
| 20:34.47 | xkev | and have your dialplan call System(/path/to/script) after a successful call |
| 20:35.14 | xkev | (have to start the process somehow, like by creating the first call file yourself, or other similar voodoo) |
| 20:35.40 | devel | LostFrog, setgroup doesn't work quite right. it automatically decrements the counter again when the channel closes, and that's not what i want. |
| 20:35.43 | moot | devel: I just tried every switchtype, nothing changed |
| 20:35.52 | moot | still shows down in pri show span |
| 20:35.56 | moot | but no alarms... |
| 20:36.26 | LostFrog | devel: use a global variable? |
| 20:36.33 | *** join/#asterisk emurphy (n=emurphy@23.22.118.70.cfl.res.rr.com) |
| 20:36.37 | LostFrog | There is an example out there to make pseudo arrays. |
| 20:36.45 | moot | devel: any ideas? |
| 20:37.15 | funxion | moot no alarms is bein coding and framing is good |
| 20:37.36 | moot | so what else could cause it to show down |
| 20:37.40 | funxion | moot what are you connecting to |
| 20:37.43 | funxion | its down? |
| 20:37.55 | moot | pri show span 1 |
| 20:37.55 | moot | Primary D-channel: 24 |
| 20:37.55 | moot | Status: Provisioned, Down, Active |
| 20:37.56 | moot | Switchtype: National ISDN |
| 20:37.56 | moot | Type: CPE |
| 20:38.02 | devel | moot, i was going to ask what kind of switch you're attaching to. |
| 20:38.08 | moot | ni2 |
| 20:38.09 | funxion | in zttool does it show down? |
| 20:38.13 | moot | nope |
| 20:38.31 | funxion | do you have your dchannel defined? |
| 20:38.35 | moot | yes |
| 20:38.40 | moot | i have bchan=1-23 and dchan=24 |
| 20:38.47 | funxion | what are you connecting to |
| 20:38.58 | moot | ni2 |
| 20:39.01 | moot | its a pri |
| 20:39.18 | funxion | national 2 is an isdn type what type of device are you conencting to? |
| 20:39.30 | devel | moot, do you know what make/model the co switch is |
| 20:39.34 | moot | no |
| 20:39.43 | moot | they just said it was ni2 |
| 20:39.47 | funxion | do you have dlr on the circuit |
| 20:39.52 | moot | what's that |
| 20:39.58 | devel | dam fascist telephone companies! |
| 20:40.06 | funxion | the engineering info on the circuit from the carrier |
| 20:40.09 | moot | funxion: they said they were seeing RMB on the B channels |
| 20:40.20 | lenne_dk | Johnnyq, are you back? |
| 20:41.12 | moot | whatever that means |
| 20:41.25 | moot | they see the d channel up but not the b channels |
| 20:41.28 | funxion | I've never heard of RMB |
| 20:41.39 | lenne_dk | johnnyq can scroll back, I hope, I've got to go see Jack Bauer... |
| 20:41.43 | lenne_dk | http://pastebin.com/445345 |
| 20:41.49 | *** part/#asterisk twisted (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 20:41.59 | moot | funxion: i thins it means reverse make busy |
| 20:42.02 | moot | some att term or something |
| 20:42.13 | twisted[asteria] | hah |
| 20:42.14 | funxion | hmmm |
| 20:42.15 | twisted[asteria] | that's cool. |
| 20:42.20 | moot | |
| 20:42.22 | moot | oops |
| 20:42.28 | moot | so |
| 20:42.29 | moot | any ideas? |
| 20:42.32 | funxion | moot can you pastebin your zapata? |
| 20:42.57 | moot | sure |
| 20:43.42 | moot | http://pastebin.ca/32024 |
| 20:45.48 | [TK]D-Fender | moot : can you pastebin your zaptel.conf too, zapata looks ok except for all that excess afer "channel => 1-23" |
| 20:45.49 | funxion | hmmm |
| 20:46.09 | funxion | [TK]D-Fender I agree |
| 20:46.24 | [TK]D-Fender | lets make sure you cannel assignmentlooks good. |
| 20:46.49 | cnet2 | I have my asterisk voip server behind nat, but my router is a small 3com router, in which I can only map like 12 ports per rule to an ip inside the nat. |
| 20:47.11 | cnet2 | i know I have to map 5060, and something like 10k to 20k |
| 20:47.15 | *** join/#asterisk santiago (n=santiago@208.195.215.160) |
| 20:47.24 | LostFrog | 5060-5063 and 10k-20k |
| 20:47.32 | cnet2 | is there a way for me to configure asterisk to use only some specific ports? |
| 20:47.38 | funxion | rtp.conf |
| 20:47.41 | funxion | cip.conf |
| 20:47.43 | funxion | sip.conf |
| 20:47.45 | funxion | sry |
| 20:47.46 | moot | [TK]D-Fender: http://pastebin.ca/32025 |
| 20:47.53 | hypa7ia | cnet2: is there any way you can tell your router to put your asterisk in the DMZ? |
| 20:48.55 | funxion | hey moot is the carrier providing timing on the circuit? |
| 20:48.55 | [TK]D-Fender | moot : All looks clean... sure your cable is good? |
| 20:49.00 | cnet2 | that would put a lot of hasle on the voip server |
| 20:49.14 | cnet2 | wouldnt it? |
| 20:49.42 | funxion | cnet2 run iptables on it |
| 20:50.00 | moot | [TK]D-Fender: I assume I would get an alarm if it wasn't, but AFAIK the cable is good |
| 20:50.18 | moot | I think just a straight cat5 is on it going down to the smartjack |
| 20:50.22 | cnet2 | excelent idea.. :D, anyway is a temp solution, gonna have a 2811 router soon :p |
| 20:51.05 | [TK]D-Fender | moot : Single port card? |
| 20:51.24 | funxion | moot you need a t1 crossover |
| 20:51.43 | [TK]D-Fender | funxion : you think? I didn't.... |
| 20:51.58 | funxion | I guess it depends on how the carrier side in configured |
| 20:52.00 | marcus2 | you shouldnt need a crossover between the smartjack and asterisk |
| 20:52.05 | [TK]D-Fender | I only needed a crossover for my channel bank |
| 20:52.08 | funxion | yeah you should |
| 20:52.08 | marcus2 | unless your carrier screwed up the install |
| 20:52.22 | moot | [TK]D-Fender: its a 2-port digium card |
| 20:52.25 | moot | te205p i believe |
| 20:52.31 | moot | using the wct4xxp module |
| 20:52.57 | [TK]D-Fender | Try another port on it. |
| 20:53.11 | [TK]D-Fender | (after trying a cross-over cable) |
| 20:53.11 | moot | i did |
| 20:53.12 | [TK]D-Fender | ;) |
| 20:53.16 | moot | just shows alarm |
| 20:53.18 | [TK]D-Fender | and? same deal? |
| 20:53.20 | [TK]D-Fender | hmmm |
| 20:53.31 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 20:53.32 | [TK]D-Fender | Ok, out of my league then :/ |
| 20:53.41 | funxion | I dont think its a corssover cable or the physical layer would not have come up |
| 20:53.48 | moot | yea |
| 20:53.52 | [TK]D-Fender | You've clearly reloaded Zaptel since making your last changes, right? |
| 20:53.56 | moot | yes |
| 20:54.03 | moot | like i said the d channel is up and there's no alarms, so... |
| 20:54.09 | [TK]D-Fender | Ok, definately beyond my exp. then... |
| 20:54.18 | funxion | moot its prolly a carrier issue |
| 20:54.21 | [TK]D-Fender | What's not working? |
| 20:54.36 | [TK]D-Fender | Can't call? receive? both? |
| 20:54.39 | moot | telco says b channels aren't coming up on my pri |
| 20:54.41 | funxion | point the finger back at them try making up some bs about using a tbird to test it |
| 20:54.44 | moot | just the d chan |
| 20:54.53 | funxion | lol |
| 20:55.01 | [TK]D-Fender | and you've confirmed the telco paramaters for the circuit? |
| 20:55.07 | moot | yes |
| 20:55.11 | [TK]D-Fender | EEK |
| 20:55.29 | funxion | see if its a nortel your connecting to |
| 20:55.37 | moot | how |
| 20:55.43 | funxion | I had some issues with nortel NI2 and a digium card |
| 20:55.58 | funxion | had to use nortel 5ess to get it ryte |
| 20:56.02 | [TK]D-Fender | You could try NI1. it is largely backwards compatible... |
| 20:56.10 | moot | i tried 5ess, 4ess, ni1, and national |
| 20:56.13 | moot | all show the same result |
| 20:56.24 | funxion | they would have to reconfigure their end to match |
| 20:56.25 | [TK]D-Fender | Hmm |
| 20:56.35 | moot | and yes i reloaded zaptel and did ztcfg before restarting asterisk after making each change |
| 20:56.44 | moot | so something is wrong... |
| 20:56.46 | xkev | NI1 is ISDN BRI |
| 20:56.51 | xkev | 'national' is NI2 |
| 20:57.06 | moot | yes |
| 20:57.09 | funxion | I bet its the telco's problem |
| 20:57.18 | devel | i'd agree with funxion |
| 20:57.20 | [TK]D-Fender | NI1 is not BRI. Its a PRI standard |
| 20:57.22 | xkev | too bad zaptel doesn't have better diags |
| 20:57.23 | devel | telcos are evil. |
| 20:57.24 | moot | all they are telling me is they don't see the b channels and its somehow my issue |
| 20:57.33 | xkev | fender, perhaps I am mistaken |
| 20:57.44 | [TK]D-Fender | xkev : Its what I'm using :) |
| 20:57.44 | xkev | b chans won't come up til * starts |
| 20:57.49 | xkev | so uh |
| 20:57.51 | funxion | moot do you have a passive test kit? |
| 20:57.55 | moot | no |
| 20:57.59 | funxion | or any test kit for that matter |
| 20:58.06 | moot | nope |
| 20:58.09 | xkev | perhaps your zapata.conf is bad? |
| 20:58.16 | xkev | like channels defined wrong? I dunno |
| 20:58.16 | funxion | nope |
| 20:58.42 | funxion | from what hes said his configs loook good |
| 20:58.51 | *** join/#asterisk _Madar (n=tophe@219-84-129-163-adsl-tpe.static.so-net.net.tw) |
| 20:59.05 | malverian[work] | What exactly is the SIP "Replaces" header supposed to do? |
| 20:59.25 | moot | here I'll paste a output from the console |
| 20:59.28 | brettnem | I think that's for a call pickup |
| 20:59.28 | moot | http://pastebin.ca/32031 |
| 21:00.33 | moot | and ztcfg output here |
| 21:00.34 | moot | http://pastebin.ca/32032 |
| 21:00.35 | devel | how do i do "math" on a variable? i.e. Set(localcounter=$[{tempcounter} + 1]) ? |
| 21:01.05 | moot | does that help anyone? |
| 21:01.09 | [TK]D-Fender | devel : yup |
| 21:01.54 | devel | [TK]D-Fender, that's not taking. Set(tempcounter=1), then Set(localcounter=$[{tempcounter} + 1]), then NoOp(${localcounter}) shows '1' |
| 21:01.57 | *** join/#asterisk qw3rty (n=qw3rty@c-67-167-79-57.hsd1.il.comcast.net) |
| 21:01.59 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 21:03.20 | qw3rty | I was looking at getting one of those Linksys/Vonage ATA devices and trying to unlock it. At voip-info.org they have firmwares and processes posted to do just that. Has anyone verified that this works? (I want to get the pap2 cause it's free after Mail in Rebate) |
| 21:04.35 | [TK]D-Fender | devel : remove the spaces around the + |
| 21:04.52 | moot | anyone got any bright ideas? :P |
| 21:04.59 | *** part/#asterisk rob314 (n=root@207.58.194.55) |
| 21:05.00 | [TK]D-Fender | and you forgot a $ in your set |
| 21:06.32 | devel | solid gold, [TK]D-Fender |
| 21:06.51 | [TK]D-Fender | ywc :) |
| 21:08.57 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 21:10.32 | devel | yes, absolute gold. now i have my asterisk wide global counter. |
| 21:11.22 | *** part/#asterisk zpn (n=xpn@gateway.digium.com) |
| 21:13.30 | *** part/#asterisk volkerli (n=volkerli@port-212-202-210-112.dynamic.qsc.de) |
| 21:13.37 | *** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-162.sw.biz.rr.com) |
| 21:17.33 | [TK]D-Fender | devel :what is your counter for? |
| 21:18.15 | devel | for taking over the world. |
| 21:18.17 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 21:18.19 | [TK]D-Fender | :O |
| 21:18.28 | devel | (by passing it to an agi to do ad rotation on a time and temp system) |
| 21:19.20 | devel | well, the way this week has gone, i'll fail miserably in my determination. |
| 21:21.00 | [TK]D-Fender | Ads during MOH & Queues? |
| 21:21.59 | devel | no, we have a number that just spits out the time and temp, and want to rotate ads before. |
| 21:22.35 | [TK]D-Fender | Simple sequential or on a "system"? |
| 21:22.52 | devel | sequential at first. |
| 21:24.30 | [TK]D-Fender | Should be dead easy then. You can just rename your ad files to match the counter like "exten =>s,3,Playback(/pathtoads/ad{$counter}.mp3) |
| 21:24.38 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 21:24.52 | devel | straight up, [TK]D-Fender |
| 21:25.03 | devel | i just needed that counter to work first :) |
| 21:25.23 | [TK]D-Fender | heh. I use counters for invalid/timout counts in my IVRS so people don't kill out LD :) |
| 21:25.51 | Katty | zaptel Disabled echo canceller because of tone (rx) on channel 6 |
| 21:25.54 | Katty | ^- what's it telling me? |
| 21:26.11 | [TK]D-Fender | proabably fax detection |
| 21:26.23 | Katty | [TK]D-Fender: elaborate. |
| 21:26.23 | [TK]D-Fender | it auto-disables EC for fax calls. |
| 21:26.34 | Katty | [TK]D-Fender: speak kat maybe? |
| 21:26.46 | Katty | [TK]D-Fender: that's not a fax line. |
| 21:26.55 | Katty | [TK]D-Fender: that's line 6 of a rollover. |
| 21:27.14 | [TK]D-Fender | Fax's play a specific leading tone so the answering side knows its a fax calling. Zaptel looks for it and disables echo cancelleation on the channel because youi don't EC fax calls. |
| 21:27.29 | [TK]D-Fender | could be triggered by noise on the line I guess.... |
| 21:27.35 | Katty | [TK]D-Fender: so if someone faxes to us through line six, or if we fax out through line six? |
| 21:27.45 | [TK]D-Fender | I used to be able to whistle up a 300 baud carrier ;) |
| 21:28.00 | [TK]D-Fender | possible. |
| 21:28.01 | *** join/#asterisk javar (n=javar@69.79.133.185) |
| 21:28.08 | devel | Katty, does it happen frequently? |
| 21:28.12 | [TK]D-Fender | typically inbound I believe... not sure about outbound |
| 21:28.17 | Katty | [TK]D-Fender: k |
| 21:28.26 | Katty | devel: [TK]D-Fender answered my question, dear. |
| 21:28.29 | [TK]D-Fender | Katty : np |
| 21:28.49 | devel | yes, i was just trying to determine if it was a one time thing (i.e. wrong number) |
| 21:29.05 | Katty | devel: i don't need that sort of determination. |
| 21:29.15 | Katty | devel: i had a specific question and requested a specific answer :) |
| 21:29.23 | Katty | devel: no further discussion is required. |
| 21:29.28 | devel | word |
| 21:29.50 | [TK]D-Fender | Katty : I believe the faxdetect parms in zapata can tell zaptel to butt-out :) not certain but something to follow if you want to crack down on it. |
| 21:30.07 | Katty | [TK]D-Fender: i'm not all that concerned. |
| 21:30.17 | [TK]D-Fender | K, case closed then! |
| 21:30.42 | [TK]D-Fender | NEXT!!! (C) BKW |
| 21:30.52 | mog_work | bi bi bi |
| 21:31.26 | *** join/#asterisk m160858 (n=jsaenz@200.89.12.46) |
| 21:31.34 | m160858 | hi |
| 21:31.43 | [TK]D-Fender | ok, time to check outta this place... bbiab |
| 21:32.03 | *** join/#asterisk darkskiez (n=darkskie@80-47-67-80.lond-hex.dynamic.dial.as9105.com) |
| 21:32.08 | *** join/#asterisk heka (n=Mindkill@82.114.68.124) |
| 21:33.03 | ctooley | How do I set the callerID from a call file. I thought I used to do that with "SetCIDNum" but that's not working now. |
| 21:33.37 | m160858 | question: i want to know .. how do it hunting .. for this |
| 21:33.40 | m160858 | http://pastebin.ca/32034 |
| 21:33.44 | m160858 | sorry by my english |
| 21:34.31 | bpm- | anyone know how to change the outgoing smtp server and port asterisk uses to send emails |
| 21:34.53 | brad_mssw | bpm-: thought it just used your local sendmail executable |
| 21:34.55 | Katty | bpm-: perl script perhaps? |
| 21:35.07 | bpm- | yea |
| 21:35.14 | bpm- | i been tring with perl |
| 21:35.22 | Katty | bpm-: anthm has a good (little) one |
| 21:35.28 | Katty | bpm-: you might convince him to share it :) |
| 21:35.30 | bpm- | did it work for you |
| 21:35.34 | Katty | of course. |
| 21:35.50 | bpm- | what about changing the port on it |
| 21:36.03 | ctooley | bpm-, you can change your mail command in voicemail if that's the email you're talking about |
| 21:36.12 | Katty | bpm-: now i didn't do that. |
| 21:36.22 | ctooley | the mail command can control port and server information |
| 21:36.44 | bpm- | ctooley: wheres that |
| 21:38.30 | ctooley | bpm-, http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf |
| 21:38.54 | *** join/#asterisk MGSsancho (n=user@ppp-67-126-243-105.dsl.irvnca.pacbell.net) |
| 21:39.19 | m160858 | excuse me, somebody can help me? |
| 21:39.22 | m160858 | http://pastebin.ca/32034 |
| 21:39.31 | m160858 | i call to the number |
| 21:39.42 | m160858 | but only sound the first extensions |
| 21:39.55 | m160858 | what's wrong? |
| 21:40.30 | devel | they need to all be together, i.e. Dial(SIP/2001&SIP/2002&SIP/2003,30) |
| 21:41.03 | m160858 | that is for me? |
| 21:41.10 | devel | ah, yes, sorry m160858 |
| 21:41.16 | m160858 | ok, thanks |
| 21:41.39 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 21:41.54 | devel | and you will probably want Hangup rather than Hungup too, m160858 |
| 21:42.02 | *** join/#asterisk nagl (n=nagl@213.235.241.6) |
| 21:42.45 | *** join/#asterisk trixter (n=trixter@65.172.209.246) |
| 21:42.59 | m160858 | excuse me? |
| 21:43.11 | m160858 | i don't understand that |
| 21:43.20 | devel | your priority 5 says "Hungup" rather than "Hangup" |
| 21:43.34 | m160858 | or, yes |
| 21:43.36 | *** join/#asterisk darby_t (i=darby_t@dmu88.neoplus.adsl.tpnet.pl) |
| 21:43.37 | m160858 | that's right |
| 21:44.37 | *** join/#asterisk mountie (n=mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com) |
| 21:45.03 | MRH2 | what is the technical term for an app hitting 99.9% cpu usage? |
| 21:45.14 | MRH2 | i call it a spike |
| 21:45.25 | Katty | i'd call it bad. |
| 21:45.35 | Katty | bad bad app. |
| 21:45.49 | MRH2 | ok i am calling it a spike in this here bug report |
| 21:45.54 | MRH2 | lol |
| 21:45.58 | m160858 | nop |
| 21:46.10 | m160858 | it doesn't work this way |
| 21:46.12 | Katty | MRH2: call it an resource hoggy app. |
| 21:46.21 | Katty | MRH2: a Big Bad(tm) resource hoggy app. |
| 21:46.27 | Katty | MRH2: that's what i'd call it. |
| 21:46.32 | Katty | MRH2: but then again, i only speak kat. |
| 21:46.49 | m160858 | 310XXXXXXX,1,Dial(SIP/2001&SIP/2002&SIP/2003,30&SIP/2004,30) |
| 21:47.08 | m160858 | ou |
| 21:47.10 | m160858 | i see |
| 21:47.17 | MRH2 | I'll just call it meetme with the i option |
| 21:47.32 | MRH2 | and hangup at the prompt |
| 21:48.39 | m160858 | Got SIP response 488 "Not Acceptable Here" back from xxx.xxx.xx.xx |
| 21:48.45 | MRH2 | and with asterisk -p it goes on and on and on and on |
| 21:48.47 | m160858 | this show up in the CLI |
| 21:49.31 | MRH2 | end rant |
| 21:49.35 | MRH2 | lol |
| 21:52.29 | Katty | hi kram |
| 21:55.06 | MikeJ[Laptop] | KRAM! |
| 21:56.55 | m160858 | i want to call that number, but i want to ringing in over extension, in oreder ..one follow the other |
| 21:57.21 | m160858 | http://pastebin.ca/32035 |
| 21:59.38 | Damin | Sigh.. |
| 22:00.35 | devel | m160858, so you want it to ring 2001 for 30 seconds, then if it isn't answered, ring 2002 for 30, and so on, right? |
| 22:01.05 | Damin | m160858: Sounds like a job for app_queue. |
| 22:01.57 | Damin | m160858: Just define a single queue in queues.conf that has all of the SIP extensions as members, and then define your parameters for calling orders and such.. |
| 22:05.10 | Damin | m160858: Check out the following: http://pastebin.ca/32036 |
| 22:05.33 | Damin | m160858: Simple. Elegant. Works great. |
| 22:05.41 | m160858 | thanks |
| 22:05.53 | Damin | m160858: Send beer to damin@nacs.net |
| 22:06.25 | iDunno | beer via SMTP. hmm |
| 22:06.30 | iDunno | interesting. does that work? |
| 22:07.02 | hypa7ia | oIP |
| 22:07.06 | hypa7ia | err |
| 22:07.09 | hypa7ia | BoIP |
| 22:13.39 | many | nifty details |
| 22:14.07 | *** part/#asterisk JCC_ (n=chatzill@207.41.92.131) |
| 22:15.10 | *** part/#asterisk emurphy (n=emurphy@23.22.118.70.cfl.res.rr.com) |
| 22:15.15 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 22:15.53 | *** join/#asterisk BladeRunner05 (n=gianni@adsl-ull-216-68.44-151.net24.it) |
| 22:16.41 | m160858 | Damin? |
| 22:16.57 | m160858 | continues sounds the first |
| 22:18.02 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) |
| 22:18.10 | Damin | m160858: "continues sounds the first" means nothing to me. |
| 22:18.24 | *** join/#asterisk ginvent (n=joseph@adsl-63-199-244-155.dsl.sndg02.pacbell.net) |
| 22:18.42 | m160858 | excuse me, i don't speak english normally |
| 22:19.12 | *** join/#asterisk Netgeeks (n=Chris@68-185-24-2.static.mdfd.or.charter.com) |
| 22:19.36 | m160858 | i I still with the same problem |
| 22:19.40 | ginvent | Argh, can someone help me get spandsp working... I can't fix the damn makefile patch. |
| 22:20.29 | marcus2 | awesome |
| 22:20.38 | marcus2 | now we can make 49k modem connects thru asterisk |
| 22:21.03 | *** join/#asterisk jarnaud (n=jarnaud@216.143.37.5) |
| 22:21.11 | kuku5 | cisco phones suck! |
| 22:21.15 | kuku5 | voip SUCKS |
| 22:21.18 | mog_work | wow thats awesome marcus2 |
| 22:21.21 | mog_work | is that reliable |
| 22:21.24 | mog_work | and how are you doing it |
| 22:21.30 | kuku5 | how are big companies handling 15 calls at a time ( secretary ) |
| 22:21.37 | marcus2 | seems pretty reliable |
| 22:21.50 | jarnaud | Hi all, do you know if asterisk produce a BUSY CDR line when a call file reaches a BUSY line? |
| 22:21.57 | marcus2 | modem -> channel bank -> t1 -> asterisk -> pri -> MCI |
| 22:21.58 | [TK]D-Fender | kuku5 : you just don't know how to set it up... the Cisco's also have attendant modules for things like that |
| 22:22.31 | [TK]D-Fender | kuku5 : how many, and what kind of phones do you have now? |
| 22:22.36 | [TK]D-Fender | (models) |
| 22:23.05 | m160858 | my CLI show up this http://pastebin.ca/32037 |
| 22:23.31 | m160858 | Damin, can you help me? please |
| 22:24.19 | Damin | m160858: Your DLI output looks right. |
| 22:24.38 | Damin | OK.. |
| 22:24.41 | Damin | time to head home! |
| 22:24.41 | m160858 | but I still with the same problem |
| 22:24.44 | Damin | Night everyone! |
| 22:25.15 | jarnaud | I place a call thru a .call file on a busy line: NO CDR. Does anyone know why? |
| 22:25.16 | ginvent | What does this line mean in a makefile? @@ -50,10 +50,14 @@ |
| 22:25.20 | m160858 | I'm new in this, please be patience |
| 22:25.24 | [TK]D-Fender | m160858 : what exactly IS the problem? |
| 22:25.26 | Damin | m160858: I don't know what your problem is.. I've given you the tools to do Queue delivery to members of a queue, so I'm not sure what else I can offer to you.. |
| 22:25.41 | *** join/#asterisk bziherl (n=bziherl_@195.246.6.131) |
| 22:25.49 | m160858 | i want to call that number, but i want to ringing in over extension, in oreder ..one follow the other |
| 22:26.06 | [TK]D-Fender | m160858 : pastebin your queues.conf |
| 22:26.12 | m160858 | only ringing the first extension |
| 22:26.15 | bziherl | hi everybody |
| 22:26.25 | *** join/#asterisk Hypherion (n=adi4@services.xten.net) |
| 22:26.59 | m160858 | http://pastebin.ca/32036 |
| 22:27.29 | m160858 | that context=operator |
| 22:27.51 | bziherl | i am not sure if i have registered correctly to the freenode.net. can you guys se what i'm writting? |
| 22:28.00 | Hypherion | no |
| 22:28.03 | Damin | m160858: Are those "member => " staements the correct SIP extensions for you? Or are they the ones that I gave you? |
| 22:28.10 | ginvent | Anyone have a good spandsp patch I can apply? |
| 22:28.12 | [TK]D-Fender | and what it says it called SIP 2003, then 2004, then 2001... whats the problem? |
| 22:28.25 | bziherl | hyperion: okay then, hehe |
| 22:28.26 | ginvent | I can't get the apps_makefile.patch to work. |
| 22:28.53 | m160858 | http://pastebin.ca/32038 |
| 22:29.10 | m160858 | only ringing the 2001 |
| 22:29.19 | m160858 | the others not ringing |
| 22:29.24 | *** join/#asterisk dokhench (n=no@209.12.249.242) |
| 22:29.54 | Hypherion | twisted: were you able to retrieve the patch for 5427? |
| 22:29.58 | [TK]D-Fender | m160858 : well one of them says he's BUSY, and the other is not available. these are problems with your PHONE's status, not the queue |
| 22:31.18 | bziherl | i have one quick question about the asterisk. sometimes i get "503 server error" after SIP softphone sends BYE command to it. Does anyone know what could be the reason? |
| 22:31.38 | *** join/#asterisk perlmonkee (i=oginvu@c-24-22-125-15.hsd1.or.comcast.net) |
| 22:32.46 | [TK]D-Fender | m160858 : soory, not "unavailable", but rather "UNREACHABLE". You sure your phones are set up right? call them direct to test |
| 22:32.55 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 22:33.47 | m160858 | http://pastebin.ca/32040 |
| 22:34.04 | m160858 | there are show my sip peers |
| 22:34.45 | jarnaud | Fender:yes, that's strange, I made 4 millions calls all over the US and not one busy!!! strange |
| 22:35.32 | perlmonkee | I'm pretty sure there are exampels of what I want online, but I don't know what search keys to use to find them, so I'm asking for just that type of help: "What do I look for?" What I want to accomplish is having outgoing calls from one extension routed through one proxy/trunk and outgoing calls from another extension routed through a different proxy/trunk. How do I accomplish this? |
| 22:36.30 | *** join/#asterisk sandra78 (n=awefawe@200.106.101.60) |
| 22:36.49 | sandra78 | does anyone knows about asterisk + bluetooth? |
| 22:37.12 | dokhench | perlmonkee: hardcode it in your extensions.conf |
| 22:37.52 | *** join/#asterisk ansong (n=ansong@adsl-69-151-50-249.dsl.okcyok.swbell.net) |
| 22:38.08 | bziherl | here's the log i get (with the problematic BYE command): http://pastebin.ca/32041 |
| 22:38.13 | dokhench | perlmonkee: things to look for, gotoif, dial, etc... |
| 22:39.59 | m160858 | some idea? |
| 22:40.02 | m160858 | please |
| 22:41.51 | bziherl | m160858: how does your extensions.conf file looks like? |
| 22:42.53 | ansong | I've got simple (and dumb) question: can asterisk call out using POTS? That is, if I connect my sbc phone lines to a couple of X100P cards, I know I can receive calls from POTS that are then routed around the pbx. Can I dial out using those same lines? |
| 22:43.07 | dokhench | ansong: yes |
| 22:43.28 | m160858 | just that http://pastebin.ca/32042 |
| 22:44.07 | sandra78 | does anyone knows about asterisk + bluetooth? |
| 22:44.46 | perlmonkee | dokhench: Thank you, I'm looking into gotoif - it seems to be what I didn't know I was looking for =D |
| 22:45.30 | dokhench | sandra78: little bit, dunno if i can help tho, until you ask your question. |
| 22:45.50 | Damin | m160858: Your other SIP extensions are not accpting the calls!!!!! |
| 22:46.01 | Damin | m160858: Read the CLI output that you pasted! |
| 22:46.14 | Damin | m160858: It says that one is not available, and the other is busy, but 2001 is ringing.. |
| 22:46.40 | Damin | m160858: Can you call 2002 and 2003 directly? |
| 22:47.13 | Damin | m160858: I.E. if you do exten => 310xxxxxx,1,Dial(SIP/2002) will they ring? |
| 22:47.19 | m160858 | i dont undestand ... like extensions ? |
| 22:47.25 | m160858 | ow |
| 22:47.26 | m160858 | ok |
| 22:47.33 | m160858 | hold a minute |
| 22:48.10 | sandra78 | dokhench> thanks...well i have read the in the wiki about asterisk+bluetooh but i got doubts... i need to know if do i can connect several cellular phones annd each one in differen channels using the same bluetooth |
| 22:48.49 | bziherl | has anyone any ideas why is asterisk reporting '503 Server Error' after softphone sends a 'BYE' command? (http://pastebin.ca/32041) |
| 22:49.45 | bziherl | i have a problem cause then calls that were terminated by the client softphones stay in my queue until timeout occurs.... |
| 22:49.48 | dokhench | sandra78: you mean connect multiple devices through one bluetooth adapter/card on the * box? |
| 22:50.31 | ansong | dokhench, it uses the same fxo cards to dial out with? also, will it handle calls rolled over from the primary number as well as direct dialed secondary numbers? and lastly: if I have 4 numbers will the TDM400P with 4 FXO modules take the place of 4 X100P cards and provide normal (that is, POTS) dial in/out from the pbx? |
| 22:51.42 | [TK]D-Fender | yup |
| 22:51.43 | ansong | y'know, I'm not normally this needy... :/ |
| 22:51.47 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 22:51.52 | *** join/#asterisk LostFrog (n=reallyno@dsl093-100-093.wdc2.dsl.speakeasy.net) |
| 22:52.00 | [TK]D-Fender | ansong : yup. |
| 22:52.19 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 22:52.46 | m160858 | i dont understand |
| 22:53.09 | m160858 | do you think what be my devices ? |
| 22:53.16 | bziherl | is there any way to catch those '503 Server Error's and terminate the call? has anyone any ideas? |
| 22:53.24 | m160858 | excuse me, i cancel to the extensions 2001 |
| 22:53.27 | m160858 | i try again |
| 22:53.36 | m160858 | and i call to the extensions 2003 |
| 22:53.55 | m160858 | mad about this |
| 22:54.09 | *** join/#asterisk kiwnix (n=egarcia@82.158.153.6) |
| 22:54.12 | [TK]D-Fender | m160858 : pastebin your whole extensions.conf and maybe I'll find something in there.... |
| 22:54.23 | ansong | [TK]D-Fender, thanks. maybe you could help me figure out what purpose an FXS would have? I'm looking at http://www.digium.com/index.php?menu=fxsvfxo . is it just an ATA for an analog phone? |
| 22:55.01 | *** join/#asterisk zotz (n=zotz@24.231.47.168) |
| 22:55.05 | m160858 | http://pastebin.ca/32042 |
| 22:55.40 | [TK]D-Fender | ansong : in the case of a TDM400P card, FXS modules let you plug analog phones into your * server. I typically suggest SIP ATA's instead of these however |
| 22:56.09 | brad_mssw | i prefer actual SIP phones |
| 22:56.11 | m160858 | question: Damin help me with my queues |
| 22:56.14 | brad_mssw | provide more functionality |
| 22:56.17 | m160858 | but this line context=operator |
| 22:56.41 | m160858 | this operator ... where are it create ... |
| 22:56.47 | m160858 | sorry by my bad english |
| 22:57.12 | m160858 | this context=operator ... doesn't exists or it's default |
| 22:57.49 | ansong | [TK]D-Fender, so it's internal only then. wouldn't that just be a one to one mapping of phones to ports? i mean, i've never seen a switch or hub designed for rj11... |
| 22:57.51 | *** join/#asterisk SparFux (n=player@tor/session/x-fc0f4bec929adaa5) |
| 22:58.05 | *** join/#asterisk Laerte (i=io@host170-57.pool81120.interbusiness.it) |
| 22:58.07 | Laerte | hy |
| 22:58.12 | LostFrog | ok.. the svn viewer is nice. :) |
| 22:59.37 | Laerte | Anyone know hardware sip phone that support SIP MESSAGE method ? |
| 22:59.58 | *** part/#asterisk Utah_Dave (n=boucha@0-1pool139-221.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 23:00.00 | ansong | the alternative, SIP, would allow the phone system to share the internal network resources. you'd just plug your network cable into your SIP phone or ATA. |
| 23:00.03 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 23:00.12 | *** part/#asterisk BladeRunner05 (n=gianni@adsl-ull-216-68.44-151.net24.it) |
| 23:01.14 | sandra78 | <dokhench> yes |
| 23:02.34 | sandra78 | <dokhench> sandra78: you mean connect multiple devices through one bluetooth adapter/card on the * box? <----YES adapter or usb bluetooth |
| 23:03.47 | Seba_soy | somebodu has developed something with SIPXTAPI? |
| 23:04.22 | *** join/#asterisk mrtwister (n=mrtwiste@cable-9-42.cgates.lt) |
| 23:04.26 | mrtwister | areski, hello |
| 23:04.43 | *** part/#asterisk bkirby (n=bkirby@207.111.160.33) |
| 23:05.56 | p1tst0p | whats the highest level of verbosity ? |
| 23:07.21 | *** join/#asterisk Los415 (n=los415@c-24-126-63-65.hsd1.ca.comcast.net) |
| 23:07.43 | SparFux | p1tstop: well, the one you get with asterisk -r -vvv I think. |
| 23:07.47 | SparFux | Level 6. |
| 23:08.08 | p1tst0p | so, set verbose 6 at the CLI is the highest |
| 23:08.45 | SparFux | I think so. |
| 23:08.47 | *** part/#asterisk mkrufky (n=mk@68.160.103.77) |
| 23:09.15 | SparFux | Uh, no. I have 12 now :-\ |
| 23:09.53 | SparFux | But the output isn't more verbose. |
| 23:09.56 | SparFux | That's strange. |
| 23:10.21 | Laerte | Anyone know hardware sip phone that support SIP MESSAGE method ? (sendtext asterisk command) |
| 23:10.58 | SparFux | Laerte: Yes! That would be cool! |
| 23:11.10 | iDunno | I end up in the 4xs ;) |
| 23:11.26 | *** part/#asterisk SparFux (n=player@tor/session/x-fc0f4bec929adaa5) |
| 23:11.30 | iDunno | just to make sure that I'm not missing anything :) |
| 23:11.41 | [TK]D-Fender | ansong : it isn't a switch to RJ11 or anything line that... |
| 23:15.55 | *** join/#asterisk amir_ (n=amir@gentoo/developer/amir) |
| 23:17.33 | m160858 | ok |
| 23:17.54 | m160858 | i think which my problem |
| 23:17.59 | m160858 | all that's ok |
| 23:18.06 | m160858 | i change to SPA's |
| 23:18.30 | m160858 | but, can you help one more other thing? |
| 23:18.46 | m160858 | if nobody answer the extension 1 |
| 23:19.05 | m160858 | then, the extension 2 would ringing |
| 23:19.27 | m160858 | how i can complety this .. in the queues? |
| 23:20.52 | *** join/#asterisk AgiNamu (n=AgiNamu@8.7.80.194) |
| 23:20.53 | AgiNamu | hi there |
| 23:21.03 | AgiNamu | asterisk's been subverted? |
| 23:21.22 | mog_work | ?? |
| 23:21.34 | bziherl | is there any way to catch those '503 Server Error's and terminate the call? has anyone any ideas? |
| 23:21.44 | bziherl | has anyone any ideas why is asterisk reporting '503 Server Error' after softphone sends a 'BYE' command? (http://pastebin.ca/32041) |
| 23:21.52 | *** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com) |
| 23:22.04 | mog_work | oh subversion |
| 23:22.13 | mog_work | yes its done |
| 23:22.18 | AgiNamu | cool |
| 23:22.23 | AgiNamu | Mono moved to SVN a while ago |
| 23:22.25 | AgiNamu | i think |
| 23:22.40 | m160858 | [TK]D-Fender ? |
| 23:22.54 | AgiNamu | I have an extensions question ... if i have a lot of differnet numbers, but want them to run the same thing |
| 23:23.07 | AgiNamu | am i best served by creating an extention for each one and pointing it at a macro? |
| 23:23.30 | AgiNamu | or is there some way to create a group of numbers |
| 23:23.36 | AgiNamu | and 1 extension that serves al those groups |
| 23:23.37 | AgiNamu | ? |
| 23:24.08 | syzygybsd | AgiNamu: did you want to specify each number in the group, or a pattern matching? |
| 23:24.39 | AgiNamu | specify each number -- no pattern |
| 23:24.57 | AgiNamu | im guessing a macro is the way to go... |
| 23:26.54 | syzygybsd | I know you can do groups for calling out, but I don't know how to handle a group for incomming |
| 23:28.45 | docelm0 | YIPPIE! |
| 23:28.57 | AgiNamu | syzygybsd groups for calling out? like zap groups? |
| 23:29.04 | syzygybsd | yes |
| 23:29.23 | AgiNamu | oh right |
| 23:30.16 | AgiNamu | hmm, how can i tie Asterisk to the state of a T1 span, i.e., shut it down if asterisk dies |
| 23:30.19 | AgiNamu | or similar |
| 23:30.34 | syzygybsd | I could see there being some way to specify it on one line, though I can't find this, it probably does NOT work but _(123|234|345) with a regex or |
| 23:31.49 | syzygybsd | AgiNamu: which way do you want to handle it? do you want to stop asterisk if the t1 dies, or stop the t1?? if asterisk dies |
| 23:32.06 | AgiNamu | stop the T1 if asterisk dies |
| 23:32.20 | *** join/#asterisk meriad (i=mreith@unaffiliated/meriad) |
| 23:32.23 | [TK]D-Fender | m160858 : hold on... dinner |
| 23:32.25 | AgiNamu | syzygybsd yea, dont worry for incoming groups ... i have hundreds of DIDs :P |
| 23:32.28 | *** join/#asterisk [hC] (i=turnerd@66.199.130.40) |
| 23:32.30 | AgiNamu | a big regex wouldnt be fun :) |
| 23:32.44 | m160858 | ok |
| 23:32.51 | syzygybsd | you can have the program that starts asterisk stop the t1 if asterisk dies |
| 23:33.19 | AgiNamu | cause I have 2 T1s into separate machines and they dont seem to failover when asterisk is dead. im guessing the call comes in and zaptel just kills it somehow? |
| 23:33.22 | syzygybsd | we had a script on our server just restart asterisk if it died |
| 23:33.33 | AgiNamu | yea im gonna run safe_asterisk |
| 23:33.48 | [hC] | anyone noticed any issues lately (maybe with cvs head) where accepting a call over PRI, dialing a sip extension, then forwarding on unavail back to PSTN sometimes gives one way audio? I cant figure out how to narrow this probem down to find the cause. |
| 23:34.15 | syzygybsd | couldn't you just modify that to change the state of the t1 to down before restarting asterisk |
| 23:34.24 | AgiNamu | yea ! i could |
| 23:34.30 | AgiNamu | but if Asterisk hangs :P |
| 23:34.39 | syzygybsd | oh, I wouldnt' know about that |
| 23:34.40 | AgiNamu | that's a good idea. ill do that |
| 23:34.49 | AgiNamu | ill ztcfg -s and -f |
| 23:35.03 | AgiNamu | or whatever the force restart optio is |
| 23:36.52 | meriad | I'm interested in learning as much as I can possibly lean abotu VOIP, I.E the underlying mechanisms, how communication works, what exactly is involved, how data gest from ponit a to point b, and all the stuff inbetween.. could someone point me to a book, or a guide, or anythign I could read to understand this in more detail please? |
| 23:37.16 | orlok | meriad: the orielly Asterisk: the future of telephony |
| 23:37.25 | orlok | meriad: free PDF available |
| 23:37.30 | meriad | orlok- really? |
| 23:37.33 | orlok | well, thats asterisk centric i gues |
| 23:37.38 | meriad | orlok- does it explain about the diff mechanisms.. |
| 23:37.40 | orlok | but i'm finding it usefull |
| 23:37.47 | AgiNamu | meriad, what's your background |
| 23:38.00 | AgiNamu | and what do you hope to accomplish? |
| 23:38.03 | orlok | yeah, from the point of view of asterisk, but it goes into sampling and other things in decent detail, codecs, etc |
| 23:38.16 | orlok | but its the only book i've read, so thats just my opinion |
| 23:38.21 | AgiNamu | CCNA is great for networking in general :P |
| 23:38.31 | AgiNamu | that woudl give you the basic foundation for how data and voice flow |
| 23:38.52 | meriad | AgiNamu- I'm a Unix Administrator working in the field looking to setup a voip network.. im just unfamiliar with this technology and general, and feel it woudl be ignorant of me not to explore it further.. |
| 23:38.56 | meriad | AgiNamu- I have my CCNA |
| 23:39.14 | AgiNamu | oh ok |
| 23:39.14 | AgiNamu | when you said underlying mechanisms and communications i wasn't sure how underlying <G> |
| 23:39.56 | meriad | i just want to broaden my knowledge.. |
| 23:39.56 | meriad | i hope im being specific enough |
| 23:39.56 | meriad | heh |
| 23:39.56 | meriad | ahh ya |
| 23:39.57 | meriad | not like, Jack picks up the phone and says hello |
| 23:39.57 | hypa7ia | meriad: a good place to start as a data person trying to get into voip is the oreilly asterisk book |
| 23:39.57 | AgiNamu | There's a book on Asterisk... perhaps not great as far as books go, but it does give a nice voice intro |
| 23:39.57 | AgiNamu | signate i think makes it |
| 23:39.59 | meriad | hello is transmitted across the ocean to sally |
| 23:39.59 | meriad | whom replies by saying HOWDY |
| 23:39.59 | meriad | heh |
| 23:40.00 | hypa7ia | ~thebook |
| 23:40.04 | jbot | [thebook] Asterisk: The Future of Telephony, released under the Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 23:40.04 | syzygybsd | meriad: take a look at www.voip-info.org it has a lot of information I use |
| 23:40.04 | meriad | hypa7ia- does it explain the other mechanisms too |
| 23:40.04 | meriad | hypa7ia- like iax, sip, etc.. |
| 23:40.05 | p1tst0p | when using realtime, for extentions, what is the context option for. |
| 23:40.07 | hypa7ia | meriad: it covers a lot of telecom basics |
| 23:40.09 | orlok | meriad: well, the asterisk book looks at it from the point of view of running a voip server on unix, so it might be a good thing for you |
| 23:40.20 | hypa7ia | as well as the voip basics |
| 23:40.28 | meriad | alright |
| 23:40.28 | meriad | good |
| 23:40.29 | hypa7ia | and then the *nix / asterisk basics :) |
| 23:40.34 | hypa7ia | i got a lot out of it |
| 23:40.38 | orlok | me too |
| 23:40.38 | meriad | where can i get ths book? |
| 23:40.42 | orlok | i read it on the tram to/from work |
| 23:40.44 | orlok | printed it out |
| 23:40.48 | AgiNamu | just google signate asterisk book |
| 23:40.54 | orlok | meriad: orielly, or download pdf |
| 23:41.00 | meriad | orlok- where can i down the pdf |
| 23:41.11 | hypa7ia | asteriskdocs.org |
| 23:41.19 | meriad | cheers |
| 23:41.37 | *** join/#asterisk Frawg (n=Frawg@unaffiliated/frawg) |
| 23:41.40 | Frawg | hey all |
| 23:41.53 | meriad | ahh there we go |
| 23:42.17 | *** join/#asterisk kshumard_home (n=ksh@pcp01931374pcs.huntsv01.al.comcast.net) |
| 23:42.46 | AgiNamu | meriad, as far as actually running with Asterisk -- I suggest a make samples |
| 23:42.47 | hypa7ia | meriad: from there, wikipedia, newton's telecom dictionary, and voip-info.org |
| 23:42.52 | [hC] | WHY must you taunt me one way audio |
| 23:42.54 | hypa7ia | yeah that too :) |
| 23:42.56 | AgiNamu | and then just read each sample config :) |
| 23:42.57 | syzygybsd | [hC]: need help? |
| 23:43.04 | hypa7ia | hey [hC] how goes? |
| 23:43.09 | [hC] | syzygybsd: greatly :( |
| 23:43.14 | hypa7ia | (other than one-way audio :p) |
| 23:43.24 | syzygybsd | oh, unless you meant |
| 23:43.27 | [hC] | hypa7ia: hey! :) not bad, just trying to debug something that makes no sense.. |
| 23:43.30 | [hC] | Haha, thanks! |
| 23:43.36 | hypa7ia | those are the most fun :) |
| 23:43.47 | AgiNamu | It taunts you because it is showing you the results of 70 RFCs and mixing Layers 7 and 3! |
| 23:44.25 | hypa7ia | hypa7ia: http://www.iptel.org/sip/siptutorial.pdf |
| 23:45.05 | hypa7ia | that's a good sip tutorial |
| 23:45.13 | [hC] | This is completely mental, ive got a PRI on server1, k.. So, a call comes in via PRI on server1, goes to server2 via IAX. there, if i assign the DID to dial my cell number, (which goes BACK to server1 via IAX, then out the pri to my cell) it works. If i make it ring my SIP extension at my desk 4/5 times, then attempt dialing my cell phone the EXACT same way (instead of VM) i get one way audio. But, only sometimes. |
| 23:45.20 | AgiNamu | there's a reason SIP stands for Suicide In Progress |
| 23:45.30 | [hC] | Tried changing codecs, mucking with macro's, adding Answer()'s in, just incase.. Cant figure it out. |
| 23:45.49 | [hC] | The bitch of it is, i cant figure out where i should start to debug it, because imnot sure what circumstance causes it |
| 23:45.51 | *** join/#asterisk miketaht (n=user@c-67-180-132-225.hsd1.ca.comcast.net) |
| 23:46.10 | [hC] | I thought it was maybe a channel bridging issue on my PRI, but i think ive ruled that out |
| 23:46.16 | AgiNamu | [hC] did you pull out ethereal? |
| 23:46.40 | [hC] | AgiNamu: Not yet, but ive got my finger on the trigger. |
| 23:46.57 | [hC] | If i send it out a voip peer rather than pri, it seems to always work |
| 23:47.11 | [hC] | so it seems to have something to do with channel bridging on my PRI. |
| 23:47.39 | [hC] | yet, all the tests ive run to have a call come in, route to and from server2, then back out the pri again (essentially the same circumstance) it works fine. |
| 23:47.42 | [hC] | ITs quite bizarre. |
| 23:48.08 | [hC] | Ill just start ripping layers out until i can find a way to recreate it somehow, was just hoping someone had heard of this sort of thing. |
| 23:48.13 | m160858 | somebody can you help to complete my queues ..? please |
| 23:48.15 | m160858 | hehehe |
| 23:49.12 | AgiNamu | im so happy i can use IAX :) |
| 23:49.13 | *** join/#asterisk outofjungle (n=outofjun@61.247.249.220) |
| 23:49.26 | [hC] | </rant> |
| 23:49.40 | [hC] | hypa7ia: hows things on the OTHER side of the country? ;) |
| 23:49.55 | hypa7ia | [hC]: pretty nice |
| 23:49.56 | AgiNamu | wtf? PRI span: Status: Provisioned, Down, Active |
| 23:49.57 | hypa7ia | rainy :/ |
| 23:50.06 | AgiNamu | how can it be down an active? :P |
| 23:50.55 | [hC] | hypa7ia: i got snowed in the other day here.. drove 2 blocks and spun out twice and decided going to the office was a bad idea |
| 23:51.54 | AgiNamu | Has kram really returned? |
| 23:52.01 | AgiNamu | As a happy ABE customer I'd love to chat :) |
| 23:52.24 | AgiNamu | Anyone else here use ABE? |
| 23:53.05 | [hC] | AgiNamu: his IRC client says that every now and again to tease people. |
| 23:55.36 | docelm0 | KRAM! |
| 23:55.55 | m160858 | hi everyone |
| 23:55.59 | Damin | docelmo: X/O? DTMF? RFC2833? Yes? No? |
| 23:56.02 | m160858 | everything works fine, but i want to that no answer the phones, they me ring the extension by order |
| 23:56.08 | m160858 | what can i do this? |
| 23:56.12 | m160858 | on my queues |
| 23:56.31 | m160858 | or i can do it, on my extensions |
| 23:56.39 | Damin | m160858: Try changing the order in queues.conf for your members? |
| 23:57.01 | m160858 | order? |
| 23:57.06 | m160858 | i don't need that |
| 23:57.14 | m160858 | actually, works fine |
| 23:57.27 | m160858 | but i want to add .. that functions |
| 23:57.38 | Damin | docelm0: X/O? DTMF? RFC2833? Yes? No? |
| 23:57.39 | docelm0 | Working the issue.. they dont do rfc2833.. Inband only.. I dont know if the problem is on their end or mine. This is something I would like to talk to mark about and see if he has any ideas |
| 23:57.48 | m160858 | if i don't answer the phone |
| 23:58.00 | m160858 | ring the order extensions |
| 23:58.05 | m160858 | other |
| 23:58.16 | m160858 | do you understand me? |
| 23:58.22 | Damin | m160858: Not one bit. |
| 23:58.37 | m160858 | if i call to my number |
| 23:58.50 | m160858 | ring in the first extension ok? |
| 23:58.55 | Damin | Yes. |
| 23:59.09 | m160858 | if i answer ... then rings in the other extensions |
| 23:59.11 | AgiNamu | [hC], yea its connected to G729 sales. every 1000th channel license sold triggers that :) |
| 23:59.12 | m160858 | ok? |
| 23:59.20 | m160858 | that't ok |
| 23:59.23 | m160858 | that's ok |
| 23:59.26 | docelm0 | Damin so as it stands at this second no DTMF.. They are having problems.. I dont know if its something I did tho.. I had to modify the ;rport= setting to get it to work with XO |
| 23:59.27 | m160858 | ? |
| 23:59.34 | m160858 | but |
| 23:59.37 | Damin | No. Not OK. If you answer the call, you have answered the call and the call will not ring anything else. |
| 23:59.39 | m160858 | are u there? |
| 23:59.56 | Damin | docelm0: I've been sending some answering service calls out it since Tuesday.. |
