| 00:04.49 | drmessano | I guess its time to try IMAP again |
| 00:05.20 | nsgn | drmessano: what happened last time you tried it? i enjoy imap |
| 00:05.32 | *** join/#asterisk jtodd (i=q2x66iq3@ns.fox-den.com) |
| 00:05.32 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 00:05.57 | nsgn | though the experience entirely depends upon your setup. mainly if you're doing it on your own server (good) or google's (bad!) |
| 00:06.08 | drmessano | I used an old version of IMAP and I had 40 extensions across 5 states ringing everytime I would reload |
| 00:06.21 | nsgn | wow |
| 00:06.42 | drmessano | Good times |
| 00:06.45 | nsgn | so where does IMAP come into a phone system? i figured you were just making an off topic email conversation |
| 00:07.19 | drmessano | No, I have used IMAP for years.. Im talking about IMAP storage of emails |
| 00:07.22 | drmessano | errr |
| 00:07.24 | drmessano | vmails |
| 00:07.28 | nsgn | ooh, ok |
| 00:07.45 | nsgn | what is the advantage of such a system? what is your current system? |
| 00:08.31 | drmessano | Advantage is unified messaging.. Store the voicemail in the users inbox and they can listen to it right from there, and when they delete it, the MWI goes out on the phone |
| 00:08.42 | drmessano | Or they can do it all from the phone, and it purges from the inbox |
| 00:09.02 | drmessano | Forwarding vmails with SMTP is nice, but then you have two copies of every voiceamil |
| 00:09.08 | drmessano | One on the PBX, one in your inbox |
| 00:09.40 | jaytee | I just get one in Exchange UM but then I don't get MWI :-( |
| 00:10.30 | *** join/#asterisk jo3sm1th (n=email@12.187.138.2) |
| 00:11.54 | drmessano | I like the Exchange UM thing to a point |
| 00:12.16 | drmessano | Just not 100% happy with Ex2007 |
| 00:12.29 | jaytee | not having MWI kind of sucks |
| 00:12.42 | BrianY | nsgn, please, can you help ? |
| 00:12.54 | drmessano | Very much like Ex2000 fixed so much that 5.5 was lacking, and was really nice, it still was buggy and more a beta for Ex2003 |
| 00:12.59 | jaytee | but I'd hate to have to duplicate the setup of mailboxes on each system and keep them synchronized. |
| 00:13.16 | nsgn | drmessano: i see. sounds good |
| 00:13.29 | drmessano | Ex2007 is nice, fixes a lot of complaints from 2003.. but I think its still a beta for Ex2010 |
| 00:13.29 | nsgn | yeah, exchange is an awesome product but certainly not for many situations |
| 00:13.30 | jaytee | or pay money for some stupid ass middleware product like MWI2007 |
| 00:13.47 | nsgn | BrianY: did you tell us what your problem was..? |
| 00:14.33 | nsgn | actually, i'll let others deal with it. i've been doing this for entirely too long |
| 00:14.37 | drmessano | 2007 has a lot of new features that M$ will tell you on its blogs theyve already made 100x better in 2010.. Like they implemented them in 2007, got it out there, realized a lot of it sucked, so retooled |
| 00:14.42 | drmessano | They did that with 2000 |
| 00:14.49 | nsgn | so i'm out. going to make myself take a break. we'll see how long i last |
| 00:14.54 | nsgn | ttyl. thanks to all who helped |
| 00:15.05 | BrianY | nsgn, yes.I tried version 1.4.22 , of asterisk.In full log of asterisk i saw active call line > Channel SIP/w1-08200e18 was answered.Now i installed 1.6.0.3-rc1, but in this version i can't see anywhere the line containing "call/channel ..was answered". What i do wrong?Maybe i didn`t enabled something on /etc/asterisk/logger.conf ? |
| 00:15.07 | drmessano | Later, phone snob |
| 00:15.19 | drmessano | 1.6.0.3 is OLD |
| 00:15.19 | nsgn | ;) lates |
| 00:15.21 | rjune_ | [TK]D-Fender, I was curious what the syntax was to label a single port on the digium card |
| 00:15.29 | drmessano | At least get on newer code |
| 00:15.36 | rjune_ | it'll be ZAP/G1<something> |
| 00:15.43 | BrianY | I got it, its old, but it's not normal to WORK ? |
| 00:16.17 | drmessano | Oh, now youre gonna get testy.. |
| 00:16.35 | drmessano | Well, god bless you |
| 00:17.08 | BrianY | God, i just asked.Why people is so irascible when others request a bit help ? |
| 00:17.33 | jo3sm1th | Its been 2 years since I used asterisk I cant remmeber how to set up Xlite with my teliax account can anyone show me just basically how to set CID/outbound calling |
| 00:18.57 | drmessano | BrianY: if youre gonna yell and ignore that youre using something that may be old and buggy, YOU may actually be the problem, not everyone else |
| 00:19.12 | drmessano | I hope you make peace with yourself |
| 00:26.30 | BrianY | All i can say is, i still hope my little child will became a man, not an idiot |
| 00:26.35 | BrianY | thanks for your support |
| 00:29.14 | drmessano | I will pray for you, and him too |
| 00:29.48 | rob0 | Always pray for people who annoy you. They hate that! |
| 00:30.16 | drmessano | I only pray for the passive agressive |
| 00:30.36 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 00:33.04 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 00:35.31 | DarthPointer | I've got a quick question about the force loading of the wct4xxp and the wctdm24xxp modules in a specific order |
| 00:35.57 | DarthPointer | the problem I'm having is that they will occasionally load in a different order; so the channels get numbered differently |
| 00:36.09 | DarthPointer | is the best way to handle this to use modprobe.d? |
| 00:36.35 | jaytee | I believe there is a switch on the card itself that you can set |
| 00:37.15 | DarthPointer | I've got two cards; one a 4 port pri / t1 the other a 24 port analog |
| 00:37.38 | DarthPointer | hence the two modules :) |
| 00:37.52 | jaytee | which also means you have the manuals for those cards or have downloaded the PDF version from Digium's site? |
| 00:38.27 | DarthPointer | of course; I was just polling for a little best practices, rather than rtfm |
| 00:40.58 | jaytee | well, not sure about the 24 port analog board but I'm pretty certain the 4 port PRI card has a little rotary switch on it to set the card number. |
| 00:59.05 | *** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk) |
| 01:05.56 | *** join/#asterisk matt_keys (n=matt_key@h155.166.18.98.dynamic.ip.windstream.net) |
| 01:09.31 | matt_keys | I just got a SPA-941 phone. I've managed to get it to register but when I make a call I get the following in syslog/debug "### flags = 24" and a fast busy signal |
| 01:11.08 | matt_keys | any ideas wtf that means? |
| 01:17.19 | *** join/#asterisk saftsack (n=saftsack@p57924C5A.dip.t-dialin.net) |
| 01:17.36 | s14ck | hi everybody |
| 01:18.03 | s14ck | how can I spy one call in progress? |
| 01:21.28 | DarthPointer | s14ck- spy at what level? network like intercept G711 packets and reconstruct to a call? Try vomit: http://vomit.xtdnet.nl/ |
| 01:24.28 | DarthPointer | you need to record the packets first; you can use ethereal / wireshark |
| 01:26.23 | DarthPointer | jaytee, u still around? |
| 01:27.40 | DarthPointer | fwiw, I think I solved the load order problem on the system by adding the file Digium to /etc/modprobe.d/ with: "install wctdm24xxp /sbin/modprobe wct4xxp; sbin/modprobe --ignore-install wctdm24xxp" |
| 01:33.07 | *** join/#asterisk DarthPointer (n=no@82.218.68.216.DED-DSL.fuse.net) |
| 01:38.39 | *** join/#asterisk marv (n=Tim@c-68-62-174-138.hsd1.al.comcast.net) |
| 01:39.38 | marv | hmm, is there no analog to importvar? i.e. set a variable on some other channel from the dialplan or agi? kind of sucks to have to open a manager connection for that |
| 02:11.30 | *** join/#asterisk VaGoNeTaS (n=debian@xen.datapartner.cl) |
| 02:11.35 | VaGoNeTaS | hello everybody |
| 02:15.51 | VaGoNeTaS | DarthPointer ? |
| 02:16.41 | VaGoNeTaS | there is somebody alive, i have an issue |
| 02:26.36 | *** join/#asterisk Deeewayne (n=dwayne@213-132.207-68.elmore.res.rr.com) |
| 02:26.36 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 02:32.56 | trentcreek | just ask |
| 02:33.32 | trentcreek | VaGoNeTaS: just ask |
| 02:36.52 | matt_keys | I just got a SPA-941 phone. I've managed to get it to register but when I make a call I get the following in syslog/debug "### flags = 24" and a fast busy signal |
| 02:36.54 | matt_keys | any ideas wtf that means? |
| 02:37.34 | matt_keys | it doesn't matter what i dial |
| 02:41.57 | [TK]D-Fender | s14ck: "core show applications like spy" |
| 02:41.59 | *** join/#asterisk ki4lzk (n=jjones@ip24-255-222-124.ks.ks.cox.net) |
| 02:42.22 | [TK]D-Fender | matt_keys: Enable SIP DEBUG and actually look at the call |
| 02:42.28 | ki4lzk | hello all |
| 02:44.32 | ki4lzk | i am trying to get my inbound sip trunk to work but i only get a busy signal. my firewall is currently off, and it is registering with the did providers server |
| 02:48.00 | [TK]D-Fender | ki4lzk: Same goes for you |
| 02:50.07 | *** join/#asterisk seanmh (n=johndoe@c-69-254-131-168.hsd1.nm.comcast.net) |
| 02:50.10 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 02:51.14 | VaGoNeTaS | trentcreek |
| 02:51.21 | VaGoNeTaS | my issue is: |
| 02:51.44 | VaGoNeTaS | i have an asterisk configured into a private lan with the ip address 192.168.1.27, and we have 6 SIP phones connected to it |
| 02:51.47 | VaGoNeTaS | and working properly |
| 02:52.16 | trentcreek | you better stick with [TK]D-Fender as he is hte resident expert though a bit cantankerous at times |
| 02:52.23 | VaGoNeTaS | and we also have an WI FI Linksys router, with the ip address 192.168.100.1 address and the dhcp is working over 192.168.100.xx range |
| 02:52.57 | VaGoNeTaS | when the laptop users gets connected to that wi fi connection, they can use the softphones and they are able to call and listen, but nobody can listen them |
| 02:53.28 | VaGoNeTaS | i've just enabled the nat into the sip.conf file, so they can login to the * server |
| 02:53.36 | VaGoNeTaS | but they cant listen |
| 02:53.43 | ki4lzk | http://www.pastebin.ca/1432590 |
| 02:53.47 | VaGoNeTaS | any suggestions? |
| 02:54.22 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.36.7) |
| 02:54.53 | trentcreek | VaGoNeTaS: line 43 tells me a lot |
| 02:54.59 | [TK]D-Fender | VaGoNeTaS: Why is your WiFi running on a separate subnet? |
| 02:56.18 | VaGoNeTaS | i've tried to change it to 1.xx |
| 02:56.25 | VaGoNeTaS | but it doesnt allow me to do it |
| 02:56.29 | [TK]D-Fender | ki4lzk: for your inbound peer entry try "insecure=port,invite" |
| 02:57.02 | VaGoNeTaS | its a linksys without atenna |
| 02:57.30 | VaGoNeTaS | trentcreek i didnt pb |
| 02:57.31 | trentcreek | I would try to connect all those goodies to the same subnet...double NATing means double trouble |
| 02:57.35 | [TK]D-Fender | VaGoNeTaS: Disable DHCP on the linksys and plug your local lan onto a LAN switched port, not WAN |
| 02:57.41 | trentcreek | VaGoNeTaS: oh..sorry |
| 02:57.56 | VaGoNeTaS | we have |
| 02:57.59 | [TK]D-Fender | VaGoNeTaS: And as trentcreek warned you you are asking for trouble |
| 02:58.07 | trentcreek | ki4lzk: look at line 43.....check your passwords.. |
| 02:58.21 | VaGoNeTaS | [TK]D-Fender i didnt pb anything |
| 02:58.35 | VaGoNeTaS | it wasnt me, it was ki4lzk |
| 02:58.37 | [TK]D-Fender | VaGoNeTaS: I didn't ask you to, and no need for your description |
| 02:58.55 | [TK]D-Fender | VaGoNeTaS: Your network layout invites trouble and I suggest fixing THAT |
| 02:59.09 | VaGoNeTaS | yup, i'm gonna reset the wi fi router then |
| 02:59.11 | [TK]D-Fender | VaGoNeTaS: This can resolve your isse |
| 02:59.14 | VaGoNeTaS | and set it up again |
| 02:59.25 | VaGoNeTaS | but, if i dont do that, there is any fix ? |
| 02:59.29 | VaGoNeTaS | like an vpn or some=? |
| 02:59.48 | [TK]D-Fender | VaGoNeTaS: ask AFTER when you really need it |
| 03:00.11 | [TK]D-Fender | VaGoNeTaS: option #2 is to set it to ROUTE, not NAT |
| 03:01.20 | VaGoNeTaS | gimme an example of that route |
| 03:01.59 | trentcreek | VaGoNeTaS: There is no example..it's a setup in the web interface |
| 03:02.11 | VaGoNeTaS | trentcreek im talking about the ROUTE |
| 03:02.16 | trentcreek | so am I |
| 03:02.18 | DarthPointer | on your wireless, what firmware are you using? dd-wrt? |
| 03:02.32 | VaGoNeTaS | i dont know, im not in the office right now |
| 03:02.38 | ki4lzk | http://www.pastebin.ca/1432593 |
| 03:02.47 | ki4lzk | it didn't help it any |
| 03:03.05 | VaGoNeTaS | [TK]D-Fender , related to my other issue , i've just fixed it |
| 03:03.06 | DarthPointer | if you are, you can change your wireless to a bridged connection under wireless--> basic settings --> "network Configuration": select Bridged |
| 03:03.21 | DarthPointer | that will put the LAN ports and the wireless on one Collision domain |
| 03:03.38 | VaGoNeTaS | i've added the line alaw=1-124 to the system.conf file |
| 03:03.49 | VaGoNeTaS | that's it |
| 03:03.53 | [TK]D-Fender | ki4lzk: pastebin your SIP config masking only passwords |
| 03:03.55 | VaGoNeTaS | my problem was solved |
| 03:07.27 | ki4lzk | http://www.pastebin.ca/1432597 |
| 03:09.31 | [TK]D-Fender | ki4lzk: the call got matched against [out] , not [in] |
| 03:10.00 | ki4lzk | ok |
| 03:10.31 | [TK]D-Fender | ki4lzk: Not put it in the right entry. And your ITSP entries should be "nat=no" |
| 03:11.02 | ki4lzk | for both? |
| 03:14.01 | trentcreek | ki4lzk: "nat=no" for ALL VOIP providers |
| 03:14.29 | drmessano | Thats not necessary |
| 03:14.57 | matt_keys | [TK]D-Fender : I'm getting an OPTIONS and OK messages, but that's all. it doesn't look like the call ever gets made on the asterisk console |
| 03:15.19 | matt_keys | but the phone is registered... |
| 03:15.43 | [TK]D-Fender | matt_keys: the PHONE?! |
| 03:15.48 | matt_keys | i can't get the damn thing to upgrade firmware though, it keeps telling me no response |
| 03:15.59 | [TK]D-Fender | ah |
| 03:16.20 | [TK]D-Fender | matt_keys: well if the call isn't arriving to * then you've configeured it wrong |
| 03:16.46 | ki4lzk | how do i make the call rouye correctly? |
| 03:16.59 | matt_keys | default settings, just set the proxy, username/pass and it registered |
| 03:17.20 | matt_keys | i've factory reset this thing probably 4 times now |
| 03:17.22 | *** join/#asterisk stijnbe (n=stijnbe@d54C16246.access.telenet.be) |
| 03:17.29 | trentcreek | matt_keys: have you tried dialing the echo test? |
| 03:17.29 | [TK]D-Fender | drmessano: Generally it is actually... tells * to ignore the RTP server it specifies as wel which in the case of ITSP can't be differnt if they use 1 server for signalling, and another for media |
| 03:17.42 | [TK]D-Fender | trentcreek: No call arrives at * at all |
| 03:17.55 | matt_keys | trentcreek nope not yet good idea though |
| 03:18.15 | trentcreek | yes, but I though he wrote when he dials...only busy signal |
| 03:18.31 | [TK]D-Fender | trentcreek: ON the phone, not back from * |
| 03:18.44 | trentcreek | oh..ok |
| 03:19.02 | drmessano | [TK]D-Fender: never had a problem with it.. Asterisk never seems to have a problem determining what is and what isn't NAT if set up correctly.. and if I was going to force the issue, I would use nat=never.. |
| 03:19.45 | [TK]D-Fender | drmessano: True you might never be impacted by it, but its pretty much a 100% safe and recommendable practice |
| 03:19.49 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 03:20.10 | matt_keys | yeah thats true, if * isn't doing anything when i call i don't guess the echo would work |
| 03:20.27 | matt_keys | so probably something with the nat |
| 03:20.37 | matt_keys | or the phone |
| 03:21.01 | [TK]D-Fender | matt_keys: So your * is behind NAT? |
| 03:21.11 | trentcreek | ki4lzk: your provider does not have any examples for setup? |
| 03:21.29 | matt_keys | phone is, * on line one is not behind nat, * on line 2 is |
| 03:21.30 | ki4lzk | i set it up using their examples |
| 03:21.41 | trentcreek | If it a reall good you, you only need COPY/PASTE |
| 03:21.48 | [TK]D-Fender | matt_keys: 2 x *? |
| 03:22.03 | matt_keys | two different * boxes yes |
| 03:22.17 | [TK]D-Fender | matt_keys: Where does this other box come into play here? |
| 03:22.40 | [TK]D-Fender | matt_keys: first we were just dealing with 1 phone trying to talk to *. Where does this other server come in? |
| 03:22.59 | matt_keys | it's on line two like i said... neither one are getting there though |
| 03:23.30 | [TK]D-Fender | matt_keys: Line 2 of your PHONE? |
| 03:23.37 | [TK]D-Fender | matt_keys: Don't throw vague terms around like that |
| 03:23.40 | trentcreek | ki4lzk: you may want to paste your sanitized settings |
| 03:25.32 | matt_keys | the phone has two lines configured for user 201 and 0154. 0154 is a DID off of a PRI, which gets broke out from an AudioCodes box behind a NAT to an * box. 201 is an extension I set up on an * box that has a static public IP. |
| 03:26.04 | matt_keys | 'nuff said? |
| 03:26.57 | matt_keys | my phone is behind a NAT here at home |
| 03:27.24 | [TK]D-Fender | matt_keys: Clearer... the other line isn't so relevent, but * simply isn't getting packets... |
| 03:27.33 | matt_keys | right |
| 03:27.49 | [TK]D-Fender | matt_keys: So its either the phone, or networking equipment on one side or the other |
| 03:28.06 | ki4lzk | trentcreek: this is the link for their settings http://www.pastebin.ca/1432597 |
| 03:28.35 | matt_keys | i've got 5060-5063 tcp/udp forwarded to the phone, and 16384-16482 udp forwarded to the phone |
| 03:28.42 | matt_keys | here at home |
| 03:28.58 | trentcreek | matt_keys: maybe it is time for a softphone so you can actually see what is happening live on your computer. |
| 03:29.01 | [TK]D-Fender | matt_keys: Remote phones should not need forwarding |
| 03:29.04 | trentcreek | maybe x-lite |
| 03:29.26 | matt_keys | just trying anything to make it work |
| 03:29.33 | matt_keys | i figure it's a nat prob somewheres |
| 03:29.54 | trentcreek | with the softphone..you can usually turn on a debug , or log feature to see what it is doing |
| 03:30.14 | matt_keys | i'll fire up ekiga then |
| 03:30.43 | [TK]D-Fender | matt_keys: For all we know the ISP its behind is screwing with SIP behind your back |
| 03:32.45 | matt_keys | i seen the register on asterisk using softphone |
| 03:33.12 | trentcreek | ki4lzk: line 23..erase it, or set it to NO |
| 03:33.16 | matt_keys | and it dials |
| 03:33.47 | matt_keys | dials with the one behind nat too |
| 03:34.03 | ki4lzk | ok |
| 03:34.22 | matt_keys | shitty phone |
| 03:34.43 | trentcreek | ki4lzk: line 41...same thing |
| 03:36.01 | ki4lzk | ok, and this is the lines from there example config or from my sip.conf? |
| 03:36.30 | trentcreek | ki4lzk: you set it up---you dont know? |
| 03:36.53 | ki4lzk | i'm just making sure i am on track with you |
| 03:38.17 | jaytee | http://www.eatmedaily.com/2009/05/towards-a-grand-unification-of-cutlery/ |
| 03:41.06 | trentcreek | ki4lzk: reload and try it |
| 03:41.20 | ki4lzk | i am |
| 03:42.09 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.36.3) |
| 03:42.29 | [TK]D-Fender | matt_keys: Softphone is on the same subet as the SPA? |
| 03:43.35 | [TK]D-Fender | jaytee: splayd doesn'r look sharp enough :) |
| 03:44.06 | matt_keys | yeah and same switch |
| 03:44.12 | matt_keys | this desktop in fact :) |
| 03:44.30 | jaytee | [TK]D-Fender, and spife looks totally impractical for any purpose involving dining |
| 03:45.05 | matt_keys | i plug my laptop into it with a crossover, set it to the same subnet and try to update the flash and it still tells me no response |
| 03:45.22 | matt_keys | the allow flash upgrade thing is checked yes |
| 03:45.31 | matt_keys | firewall is off on the laptop |
| 03:45.32 | trentcreek | ki4lzk: Last Line: srvlookup=no |
| 03:45.40 | [TK]D-Fender | jaytee: frankly you are probably pinning your target with something in the "fork" sphere so combining knife & fork is kinda redundant |
| 03:45.42 | trentcreek | ki4lzk: you have that in wrong place |
| 03:45.50 | ki4lzk | yeah i just realized that |
| 03:46.01 | [TK]D-Fender | ki4lzk: And REGISTER has to come after EVERYTHIGN else under [general] |
| 03:46.09 | matt_keys | but i can ping the phone |
| 03:46.23 | [TK]D-Fender | matt_keys: SPA's are generally very good with being behind NAT |
| 03:46.46 | matt_keys | have you ever encountered one that did this? |
| 03:47.21 | ki4lzk | trentcreek: i also didn't have the correct username in my register string |
| 03:47.44 | trentcreek | ki4lzk: yeah that would help |
| 03:47.59 | drmessano | why would you set srvlookup=no? |
| 03:48.10 | [TK]D-Fender | matt_keys: You mean NOTHING? No... they've all worked |
| 03:48.42 | [TK]D-Fender | ki4lzk: your register has no impact on the processing of the inbound call |
| 03:48.43 | matt_keys | well it registers and responds to pings so it's not a brick, and the web interface works on it |
| 03:49.17 | [TK]D-Fender | jaytee: Oh, and a NEW surprise this week at martial arts... |
| 03:49.27 | ki4lzk | ok it actually connects now before it goes to the busy signal. going to restart the server |
| 03:49.28 | jaytee | heh? |
| 03:49.51 | matt_keys | I get a dial tone and all the lights are green |
| 03:50.12 | [TK]D-Fender | jaytee: in the last 2 months I did 5&4 on the same day, did 3 about 2 weeks ago and completed it across 2 classes and was never actually told that I passed. Its all feels so much like a foregone conclusion :) |
| 03:50.15 | trentcreek | Asterisk does not support DNS SERVER lookups for inbound calls. |
| 03:50.44 | drmessano | uh what? |
| 03:50.57 | [TK]D-Fender | jaytee: New development is that my Sensei told me my test for 2nd Kyu is supposed to be TOMORROW... he did this of course only on THURSDAY when I had booked up the day :) |
| 03:51.23 | jaytee | [TK]D-Fender, so you're not ready? |
| 03:51.36 | trentcreek | Asterisk does not support DNS SERVER lookups for inbound calls. So if have a SIP number bound to the SIP Trunk add it to the sip_general_custom.conf file as well |
| 03:51.36 | [TK]D-Fender | jaytee: Thats besides the point :) |
| 03:52.03 | [TK]D-Fender | jaytee: Compounded by the fact I never got papers for the CONTENT of my LAST test let alone this NEXT one and I like KNOWING what I'm being tested on :) |
| 03:52.07 | matt_keys | <PROTECTED> |
| 03:52.07 | matt_keys | <PROTECTED> |
| 03:52.25 | [TK]D-Fender | matt_keys: About f-ing time :) |
| 03:52.34 | matt_keys | like i said, it registers |
| 03:52.40 | drmessano | 4.1.8? wow |
| 03:52.41 | [TK]D-Fender | matt_keys: Ok, so it reg's... wait... it always did that, didn't it? |
| 03:52.55 | jaytee | perhaps the next test is all about how you respond and react to the unexpected :-) |
| 03:53.08 | [TK]D-Fender | matt_keys: I still think you have a proxy setup issue in the phone and did something screwy.... |
| 03:53.16 | [TK]D-Fender | jaytee: DUCK!!!!!!!! |
| 03:53.19 | [TK]D-Fender | *whack* |
| 03:53.37 | drmessano | factory reset FTW |
| 03:53.49 | matt_keys | drmessano already did that |
| 03:53.57 | matt_keys | like 5 f'n times.. |
| 03:53.59 | [TK]D-Fender | jaytee: I've always realized just how much I put into what I do as compared to just about everybody else. |
| 03:54.15 | [TK]D-Fender | matt_keys: Let one of use look at it perhaps |
| 03:54.24 | matt_keys | sure |
| 03:54.40 | matt_keys | i'm factory resetting it htough |
| 03:54.47 | matt_keys | you can play with it then |
| 03:54.51 | drmessano | I would suggest doing a reset and updating the firmware before going further |
| 03:55.03 | drmessano | 4.1.8 is teh old |
| 03:55.04 | jaytee | hehehe, my mom sent me a birthday card and on the front a porcupine, a chicken and a giraffe are riding in a car, the porcupine is driving. They are quickly approacing a bridge overpass and the porcupine and the chicken yell, "Duck!!!" and the giraffe is looking up at the sky saying, "Where?" |
| 03:55.08 | matt_keys | drmessano i've been trying but it keeps saying "No repsonse" |
| 03:55.15 | matt_keys | that f'n utility sucks balls |
| 03:55.26 | drmessano | I've never had a problem with it |
| 03:55.49 | matt_keys | i wish it would tell me why there's no response or some sorta verbose output |
| 03:55.51 | ki4lzk | where should srvlookup be? |
| 03:55.55 | ki4lzk | in the general? |
| 03:56.11 | drmessano | Why are you setting it to NO? |
| 03:57.48 | drmessano | srvlookup=no was recommended for 1.2 as it broke some things, but you need srvlookup nowadays.. Well, you dont NEED it, but providers are using SRV records more and more |
| 03:59.21 | matt_keys | http://mattkeys.blogsite.org |
| 03:59.25 | matt_keys | that should point to the phone |
| 04:00.14 | ki4lzk | drmessano according to the to the providers site they recommend setting it |
| 04:00.31 | [TK]D-Fender | matt_keys: You flushed and didn't reconfig the lines? |
| 04:00.33 | drmessano | Who is the provider? |
| 04:00.46 | matt_keys | woudln't want to give out the info for that |
| 04:01.04 | [TK]D-Fender | matt_keys: set temp PW's and point to a safe context |
| 04:01.23 | matt_keys | do you have one to test with? |
| 04:01.27 | [TK]D-Fender | matt_keys: FFS we're trying to help... don't handicap the process any further |
| 04:01.46 | matt_keys | just a sec |
| 04:01.53 | ki4lzk | voipvoip |
| 04:02.31 | drmessano | ki4lzk: your provider is clueless.. or they think a 1.2 era config recommendation is good forever |
| 04:02.40 | drmessano | Which I guess is the same |
| 04:03.04 | ki4lzk | so try it without it? |
| 04:05.50 | matt_keys | ok look now, i've got extension 666 registered |
| 04:09.08 | matt_keys | [TK]D-Fender : I created two, i can either register the other or give you the credentials to put in your phone |
| 04:09.35 | matt_keys | or can call the softphone |
| 04:09.40 | matt_keys | i'll do that first.. |
| 04:13.30 | matt_keys | softphone won't let me register the second line |
| 04:13.46 | matt_keys | but i can the first one |
| 04:16.28 | [TK]D-Fender | matt_keys: try with "make call without reg=yes" |
| 04:16.42 | carrar | matt_keys, you should put the current firmware on that wide open to the internet linksys 941 of yours |
| 04:16.49 | carrar | heh |
| 04:16.54 | carrar | ponders |
| 04:17.00 | [TK]D-Fender | matt_keys: then tell * that the phone is not behind NAT and enable the keep-alives on the phone itself |
| 04:20.59 | *** join/#asterisk MaliutaLap (n=biteme@kiev.lusan.id.au) |
| 04:24.19 | matt_keys | carrar keep up man, i've been talking about that for hours |
| 04:24.24 | matt_keys | it won't take a firmware upgrade |
| 04:25.22 | drmessano | matt_keys: You want to know why? |
| 04:25.30 | matt_keys | i'd love to know |
| 04:25.58 | carrar | perhaps you are just doing it wrong |
| 04:26.13 | carrar | I could upgrade it from here |
| 04:26.28 | drmessano | Google a bit and you'll find you can't just jump from 4.1.8 to 5.1.7 |
| 04:26.52 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 04:26.53 | drmessano | You need to upgrade to something like 5.0.1 .. or some firmware revision I cant remember |
| 04:27.15 | matt_keys | gimme a sec.. |
| 04:28.23 | [TK]D-Fender | drmessano: OMG, SPA's firmware is like a gateway drup... next thing you know you'll replace all your gear with Linksys, and then find yourself hooked on Cisco and unable to escape! JUST SAY NO! |
| 04:28.28 | [TK]D-Fender | DRUG* |
| 04:28.36 | drmessano | LOL |
| 04:29.10 | drmessano | For the life of me I cant remember which version it was.. I thought I had saved it |
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| 04:36.12 | drmessano | matt_keys |
| 04:36.15 | drmessano | http://firmware.linksys-cisco.cz/SPA941/?C=M;O=D |
| 04:36.35 | carrar | haha |
| 04:36.35 | drmessano | Grab the 4.1.15, see if you can update |
| 04:36.38 | carrar | <PROTECTED> |
| 04:38.34 | drmessano | Hang on |
| 04:38.44 | drmessano | http://www.vcommassist.com/vendor/linksys/firmware/spa941-4.1.15.zip |
| 04:38.47 | drmessano | There you go |
| 04:40.36 | carrar | I don't ever recall having upgrade issues with 94x |
| 04:40.54 | drmessano | I did.. with very old firmware |
| 04:41.02 | carrar | I know with cisco's you have too |
| 04:47.31 | carrar | oldest source I have is spa941-04-01-12-a.bin |
| 04:47.37 | carrar | was ginna downgrade and try it |
| 04:49.43 | *** join/#asterisk davevg (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
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| 05:00.13 | carrar | won't let downgrade to 4.1.9 |
| 05:00.14 | carrar | heh |
| 05:03.45 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 05:06.17 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 05:06.22 | drmessano | I think somewhere in there, something changed |
| 05:06.51 | drmessano | I was thinking I had to hit an early 5.x first, but very well could have been a later 4.x |
| 05:08.23 | carrar | be nice if could bring the 941 config to the same level as the 942 |
| 05:08.34 | drmessano | Yes |
| 05:08.40 | drmessano | I would love that |
| 05:08.58 | carrar | 6-1-3a is actually usable |
| 05:09.46 | coppice | linksys's firmware update policy is total chaos |
| 05:09.58 | carrar | As is there directory |
| 05:12.07 | *** join/#asterisk Optic (n=dfraser@miso.capybara.org) |
| 05:12.11 | coppice | dunno about the phones, but the spa2102 and spa3102 are a mess. they behave quite differently, although they are built on the same platform. fixes have been arbitrarily applied to each, without being carried over to the other |
| 05:12.39 | *** part/#asterisk Optic (n=dfraser@miso.capybara.org) |
| 05:12.44 | coppice | and they still ship with 2 year old firmware, possibly indicating the faith linksys have in newer versions |
| 05:13.00 | carrar | Thats Sipura's for ua |
| 05:13.03 | carrar | ya |
| 05:14.48 | drmessano | I dont think its a matter of faith |
| 05:15.46 | drmessano | I really think they pick some firmware revision that doesnt kill a baby seal, and just make that whats ships |
| 05:15.57 | drmessano | and never really update |
| 05:15.58 | [TK]D-Fender | "If it ain't broke(n badly enough) don't fix it" |
| 05:16.14 | rob0 | Did you hear the one about the baby seal who walked into a club? |
| 05:16.41 | [TK]D-Fender | revokes rob0's comedic license |
| 05:16.46 | drmessano | lol |
| 05:17.04 | drmessano | Just like the SPA-941s shipping with firmware licensed for 2 lines |
| 05:17.30 | drmessano | Surely they would want them to come out of the box with 4 lines.. since they're now sold that way |
| 05:17.32 | coppice | rob0: did you hear about the guy who walked into a bar? someone called an ambulance. it was an iron bar |
| 05:17.33 | drmessano | But noooo |
| 05:18.02 | coppice | i thought they charged extra for the firmware update |
| 05:18.12 | drmessano | Havent in some time |
| 05:18.41 | drmessano | Not sure what the milestone release was, but all recent firmware has enabled 4 lines without the license |
| 05:21.54 | drmessano | Early 2007 |
| 05:22.17 | drmessano | Sorry, mid 2007 |
| 05:22.57 | drmessano | Found a forum post from Nov 2007 where someone found the upgrade was free now with latest firmware.. someone else noted they had upgraded a few months back |
| 05:24.48 | tzafrir_laptop | BTW: Rony Ron is also the name of an aledged murderer in a high-profile murder case in Israel |
| 05:26.32 | rob0 | They don't have baby seals in Israel, do they? |
| 05:28.24 | coppice | they probably have some navy seals |
| 05:29.47 | rob0 | hmm, that sounds plausible |
| 05:39.31 | sfire | I need help.. I installed asterisk now tonight.. I can log in via the console.. I cannot get in at all via the web interface.. it won't accept my username and password |
| 05:39.31 | sfire | I have googled the question.. (all different ways) how do I get access to the website?? |
| 05:40.12 | rob0 | ... web interface ... ? |
| 05:40.23 | sfire | yea.. asterisknow install |
| 05:40.48 | sfire | I banged my head on the monitor for 3 hours trying to fix it.. looked like a fool :( |
| 05:41.02 | sfire | what am I missing? |
| 05:41.14 | drmessano | AsteriskNOW requires entering password on the keyboard, not the monitor |
| 05:41.21 | drmessano | So yes, you look like a fool |
| 05:41.32 | sfire | I did enter it on the keyboard |
| 05:41.49 | sfire | tried the setup password.. tried the "secret password" listed in the configuration files |
| 05:41.50 | *** join/#asterisk chendy (n=chatzill@119.139.82.18) |
| 05:42.09 | sfire | username 'admin' 'root' I tried it all :( |
| 05:42.58 | sfire | I have used it in the past and the website worked easily |
| 05:43.08 | sfire | (previous installation with an older version) |
| 05:44.06 | *** join/#asterisk MrNaz (n=mrnaz@ppp118-208-203-152.lns10.mel6.internode.on.net) |
| 06:19.05 | sfire | nevermind all ... I just got it admin/admin |
| 06:19.25 | sfire | I think I might have to volunteer for the documentation project :( |
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| 12:08.26 | IPGHOST | hi |
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| 12:24.44 | matt_keys | yawns. |
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| 12:38.43 | matt_keys | if anybody was curious as to what fixed the firmware update problem on teh SPA-941, it was Windows Vista causing the prob. Once I moved to an XP machine and tried it took the 5.1.8 upgrade first time |
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| 13:57.43 | *** mode/#asterisk [+o jtodd] by ChanServ |
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| 15:44.49 | mkillebrew | how can I view SIP invites? I'm trying to see if voicepulse sends ANIs when call blocked numbers call my toll free |
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| 15:53.53 | [TK]D-Fender | mkillebrew: "sip set debug on" |
| 16:08.25 | mkillebrew | hm, I'm seeing nothing |
| 16:20.26 | [TK]D-Fender | mkillebrew: then either you're not getting ANY traffic or you haven't enabled it. Didi it say its enabled? |
| 16:34.44 | mkillebrew | no I mean, I'm seeing no ANI or CPN when I call block and call my 800 |
| 16:35.23 | mkillebrew | just tcpdump'd udp 5060 as well in case asterisk wasn't catching all the fields |
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| 16:49.29 | [TK]D-Fender | mkillebrew: No, its a raw test dump... |
| 16:49.32 | [TK]D-Fender | text* |
| 16:49.41 | [TK]D-Fender | mkillebrew: You don't see it then they didn't send it |
| 17:01.07 | mkillebrew | I thought they were required to send ANI with toll-free numbers |
| 17:02.08 | [TK]D-Fender | mkillebrew: Maybe from their telco to your ITSP, not sure about your ITSP to you though |
| 17:02.31 | mkillebrew | ah |
| 17:02.37 | mkillebrew | yea that makes sense. |
| 17:02.57 | *** join/#asterisk thomasrr (n=scroogey@cp811981-a.mill1.nb.home.nl) |
| 17:02.58 | thomasrr | hello |
| 17:03.16 | mkillebrew | ohai thar |
| 17:03.32 | thomasrr | i am having trouble with my asterisk installation in combination with voipbuster |
| 17:04.27 | thomasrr | the problem is that i dont get incoming calls |
| 17:06.45 | mchou | most likely that's your firewall |
| 17:11.14 | *** join/#asterisk thomasrr (n=scroogey@195-240-213-212.ip.telfort.nl) |
| 17:11.16 | thomasrr | hello |
| 17:11.19 | thomasrr | back |
| 17:11.26 | thomasrr | yes, but i am having some odd issues |
| 17:11.33 | thomasrr | for example when i call a full number it doesn't work |
| 17:11.59 | thomasrr | but when i call to an extension and then say like _600X => { Dial(SIP/voipbuster/1234567890); }; |
| 17:12.03 | thomasrr | it works |
| 17:12.08 | thomasrr | but not when i call to the same number directly |
| 17:14.49 | *** join/#asterisk apocn (n=apo@unaffiliated/apocn) |
| 17:15.39 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.229.58) |
| 17:16.26 | apocn | Hello all, I've configured asterisk behind an ADSL Router (with 2 agents and 1 queue). The caller can hear the agent, but the agent cant hear the caller, then the connection falls after 15 seconds. I have setup nat=yes, but its not working :'(, any hints? |
| 17:16.33 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
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| 17:16.45 | apocn | I also set rtp from 50000 through 51000 and I configured the NAPT on the router, etc. |
| 17:17.29 | thomasrr | sometimes you need to disable some feature on the router |
| 17:17.33 | thomasrr | to improve support for SIP |
| 17:17.37 | thomasrr | sometimes they are mangling with packets |
| 17:17.46 | thomasrr | but i am also having issues :> |
| 17:17.53 | apocn | :\ |
| 17:18.04 | thomasrr | cant get voipbuster voip-in number to work :( |
| 17:20.10 | thomasrr | back to recovering the ip of my siemens sip phone :) |
| 17:20.22 | [TK]D-Fender | apocn: Takes a hell of a lot more settings than just that... |
| 17:20.25 | [TK]D-Fender | ~sipnat |
| 17:20.26 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 17:20.31 | [TK]D-Fender | apocn: ^^^^ read the guide |
| 17:20.44 | apocn | thanks [TK]D-Fender |
| 17:21.13 | thomasrr | damn Siemens S450IP :) |
| 17:21.21 | [TK]D-Fender | [13:12]<thomasrr>but not when i call to the same number directly <- pardon? |
| 17:21.44 | thomasrr | i found the issue :) |
| 17:21.49 | thomasrr | {$EXTEN} ;) |
| 17:21.51 | thomasrr | ${EXTEN} |
| 17:22.03 | thomasrr | now only the incoming calls |
| 17:24.47 | thomasrr | [TK]D-Fender: do you know if you need to do something special for the trunk sip config? |
| 17:25.07 | [TK]D-Fender | thomasrr: That is a uselessly vague question you know.... |
| 17:25.08 | thomasrr | i can receive calls like extension@myip from outside |
| 17:25.21 | thomasrr | to get voipbuster voip-in to work i mean |
| 17:25.41 | [TK]D-Fender | thomasrr: Set up a peer for them like they probably even give you a sample for |
| 17:27.21 | apocn | [TK]D-Fender: In the document you sent me, the #5 is my scenario (Asterisk as a SIP server behind nat, clients on the inside connecting to Asterisk). And the solution #5 Works - no NAT in between |
| 17:29.35 | thomasrr | [TK]D-Fender: i havent found any info about it for asterisk :/ |
| 17:29.41 | thomasrr | i got the calling out going for voipbuster |
| 17:29.49 | thomasrr | but do i need to connect a second time for the incoming calls? |
| 17:30.16 | *** join/#asterisk vgster (n=vgster@94-194-190-189.zone8.bethere.co.uk) |
| 17:31.08 | [TK]D-Fender | thomasrr: http://www.google.ca/search?hl=en&q=voipbuster+asterisk+howto&btnG=Google+Search&meta=&aq=f&oq= |
| 17:31.13 | [TK]D-Fender | thomasrr: JFGI |
| 17:31.56 | [TK]D-Fender | thomasrr: In fact the peer you set up for your outbound might very well work for incoming calls. So far I don't see you showing us SIP debug for when a call DOES arrive at your box... |
| 17:33.27 | thomasrr | yes, right. let me check the debug line again :) |
| 17:36.37 | thomasrr | [TK]D-Fender: what was the line again for sip debug? |
| 17:37.30 | [TK]D-Fender | thomasrr: "sip set debug on" |
| 17:38.37 | thomasrr | thanks let me call again |
| 17:42.16 | thomasrr | hmm |
| 17:43.31 | apocn | [TK]D-Fender: ahhhh! the problem was in my SBC |
| 17:43.51 | thomasrr | maybe the issue is with the trunk type |
| 17:43.53 | thomasrr | which is now peer |
| 17:43.57 | thomasrr | might need to be friend |
| 17:44.00 | apocn | now that I made a dump, I saw the problem. That usually happens when there are 2 laidies waiting for you to go to the beach... |
| 17:44.36 | apocn | :] |
| 17:45.17 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 17:51.48 | [TK]D-Fender | thomasrr: No the problem is your not showing anything :) |
| 17:51.56 | [TK]D-Fender | thamAnd I'm out of time to help right now... |
| 17:51.58 | [TK]D-Fender | BBIAB |
| 17:54.04 | thomasrr | oki |
| 17:54.15 | thomasrr | keep getting all kind of stuff passing by |
| 17:55.08 | thomasrr | hmm |
| 17:57.02 | *** join/#asterisk MrNaz (n=mrnaz@ppp118-208-203-152.lns10.mel6.internode.on.net) |
| 17:58.22 | hardwire | does a boogie |
| 17:58.36 | hardwire | [TK]D-Fender: you are relieved of duty for the rest of the day.. go have fun in the sun. |
| 17:58.44 | hardwire | I'LL HANDLE THE CHANNEL FOR A WHILE |
| 17:58.47 | hardwire | woooohoooo |
| 17:59.07 | hardwire | wheres the steering wheel? |
| 18:02.44 | tzafrir_laptop | eyes the lifeboat |
| 18:04.02 | jaytee | grabs a lifejacket |
| 18:04.17 | thomasrr | hmm |
| 18:04.19 | thomasrr | i am getting this now: |
| 18:04.20 | thomasrr | [May 16 15:41:01] NOTICE[3681]: chan_sip.c:18160 handle_request_invite: Call from 'voipbusterusername' to extension 's' rejected because extension not found. |
| 18:04.23 | thomasrr | Scheduling destruction of SIP dialog 'dc90461cdd40452ab7895edfeafb1ca2' in 6400 ms (Method: INVITE) |
| 18:04.27 | thomasrr | asterisk*CLI> |
| 18:04.38 | thomasrr | is s some mandatory extension? |
| 18:05.35 | jaytee | thomasrr, http://www.voip-info.org/wiki/view/Asterisk+standard+extensions |
| 18:06.57 | thomasrr | so i need to make that extension in my context? |
| 18:07.44 | thomasrr | uh, i think i mean an extension voipbusterusername |
| 18:07.59 | *** join/#asterisk MrNaz (n=mrnaz@ppp118-208-203-152.lns10.mel6.internode.on.net) |
| 18:09.10 | rob0 | Forget the steering wheel. Where's the accellerator? |
| 18:09.20 | rob0 | Let's see what this baby can do!! |
| 18:09.33 | thomasrr | oh wait i dont have a context for my voipbuster trunk specified |
| 18:10.27 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-198-94-rbry-bb-1.dynamic.isadsl.co.za) |
| 18:13.51 | drmessano | Anyone know how to jump to a line number while in nano? |
| 18:14.01 | thomasrr | exit |
| 18:16.08 | drmessano | Ah got it |
| 18:16.49 | *** join/#asterisk thomasrr (n=scroogey@195-240-213-212.ip.telfort.nl) |
| 18:17.05 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-198-94-rbry-bb-1.dynamic.isadsl.co.za) |
| 18:18.00 | thomasrr | time to experiment with the s-extension |
| 18:21.02 | jaytee | remember, when experimenting, to always wear your safety glasses |
| 18:24.01 | tzafrir_laptop | drmessano, <NUM>G, isn't it? |
| 18:24.09 | tzafrir_laptop | :-) |
| 18:24.26 | drmessano | Ctrl-Shift-_ or ESC-G |
| 18:24.40 | thomasrr | i keep getting pokeanswer notices |
| 18:24.41 | drmessano | As I just found a few mins ago |
| 18:24.52 | thomasrr | chan_sip.c:9923 sip_reg_timeout for example |
| 18:28.21 | MaliutaLap | anyone know why, when * received a correct incoming CID, doing a DIAL() to a dahdi channel results in "Didn't finish Caller-ID spill. Cancelling."? |
| 18:40.04 | *** join/#asterisk beniwtv (n=beniwtv@87.111.61.140) |
| 18:42.10 | beniwtv | hi all... In most call centers in my country, if you are on hold in a queue, just before the agent picks up you hear ringing tones. So, while the next available agent is located, you hear music. When he is located, you hear rings until he picks up. Any way to replicate this with asterisk queues? |
| 18:42.47 | hardwire | you can use a 'transfer on pickup' macro |
| 18:42.48 | thomasrr | what does regex in sip.conf exactly do for users? |
| 18:43.21 | thomasrr | http://www.voip-info.org/wiki/view/Asterisk+sip+regexten |
| 18:43.23 | thomasrr | is a bit empty :+ |
| 18:43.55 | beniwtv | hardwire: but transfer to what? to an extension that DIAL's SIP/<agent>? |
| 18:44.48 | hardwire | that rings first.. waits.. then dials him again in a different way. |
| 18:45.09 | beniwtv | I see... |
| 18:45.10 | beniwtv | thanks |
| 18:45.14 | *** part/#asterisk beniwtv (n=beniwtv@87.111.61.140) |
| 18:45.16 | hardwire | beniwtv: here's a hint tho |
| 18:45.17 | hardwire | damnit |
| 18:45.22 | hardwire | I was GOOONNNAA say.. |
| 18:45.34 | hardwire | that the ring you hear is probably because the agent is in an immediate pickup scenario |
| 18:45.58 | hardwire | and that he needs to press a key to answer anyways. So when he becomes available you hear a ring and he has to hit his "don't fire me" button. |
| 18:47.45 | thomasrr | hmm password of gigaset S450IP :o |
| 18:57.08 | *** join/#asterisk ORD3R (n=ORD3R@adsl-68-77-5-187.dsl.emhril.ameritech.net) |
| 18:57.34 | *** part/#asterisk ORD3R (n=ORD3R@adsl-68-77-5-187.dsl.emhril.ameritech.net) |
| 19:00.00 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 19:01.10 | thomasrr | cool it all works now |
| 19:01.15 | thomasrr | but i am getting all kind of timeouts |
| 19:01.17 | thomasrr | is that normal? |
| 19:01.41 | thomasrr | for example: [May 16 16:37:27] NOTICE[3667]: chan_sip.c:9923 sip_reg_timeout: -- Registration for 'XXXX@sip.voipbuster.com' timed out, trying again (Attempt #2) > doing dnsmgr_lookup for 'sip.voipbuster.com' |
| 19:05.50 | *** join/#asterisk bbryant (n=brett@c-76-26-221-76.hsd1.sc.comcast.net) |
| 19:16.06 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 19:18.13 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
| 19:41.39 | *** join/#asterisk x1nux (n=x1nux@unaffiliated/x1nux) |
| 19:41.43 | x1nux | hi |
| 19:41.59 | x1nux | i need help about asterisk |
| 19:43.25 | mmlj4 | x1nux: not unless you ask helpful questions |
| 19:43.38 | x1nux | I want to know how to authenticate with the function Autenticate in a mysql database |
| 19:43.40 | x1nux | ? |
| 19:44.04 | mmlj4 | have you looked at the various pages on the wiki? |
| 19:44.09 | mmlj4 | ~wiki |
| 19:44.43 | x1nux | I have inside my mSQL 2 tables |
| 19:44.50 | x1nux | names, pins |
| 19:44.56 | mmlj4 | ok, let me ask differently |
| 19:44.57 | x1nux | pins = numeric |
| 19:45.38 | mmlj4 | have you looked up "authenticate" and "database" on http://www.voip-info.org ? |
| 19:45.45 | mmlj4 | also, you might need AGI |
| 19:46.00 | x1nux | yes |
| 19:46.06 | x1nux | but i don't know |
| 19:46.25 | x1nux | how this work ! |
| 19:49.06 | mmlj4 | can you explain in english what you are trying to do? |
| 19:49.36 | x1nux | ok |
| 19:50.55 | x1nux | I make a call, then prints a welcome message, then asks for a PIN number, the PIN is in the BD. |
| 19:51.14 | mmlj4 | in mysql, you mean? |
| 19:51.19 | x1nux | yeap |
| 19:51.44 | x1nux | exten => 100,3,Authenticate(/etc/asterisk/pwd) => is OK! |
| 19:51.51 | x1nux | but with Mysql ? |
| 19:52.27 | mmlj4 | one way you can make this work is to use AGI, which can fetch the password from the DB |
| 19:52.41 | x1nux | ok |
| 19:52.48 | x1nux | i have the script en AGI ... |
| 19:52.50 | x1nux | so |
| 19:52.55 | mmlj4 | I am doing the same thing, actually |
| 19:53.17 | x1nux | ok |
| 19:53.20 | x1nux | i wait |
| 19:53.21 | mmlj4 | I am using perl, but you can use any programming language you wish |
| 19:53.30 | x1nux | php or perl |
| 19:53.33 | x1nux | no problem |
| 19:53.35 | mmlj4 | either |
| 19:54.07 | x1nux | iegther |
| 19:54.57 | mmlj4 | I will not write the script for you, but here is what you need to do: |
| 19:55.43 | x1nux | hey i have the script |
| 19:56.11 | x1nux | but don'w know, how put the line in the extension.conf .. |
| 19:56.11 | mmlj4 | pass caller ID to the script, which then retrieves the password, then pass the PIN back to the script, which you then use in the autheniticate command |
| 19:57.08 | x1nux | so |
| 19:57.39 | x1nux | exten => 100,3,AGI(script.agi) ? |
| 19:57.40 | mmlj4 | er, let me fix that |
| 19:57.48 | mmlj4 | pass caller ID to the script, which then retrieves the password, then pass the PIN back to the DIALPLAN, which you then use in the autheniticate command |
| 19:58.07 | mmlj4 | yes |
| 19:58.10 | mmlj4 | like that |
| 19:58.15 | x1nux | ok |
| 19:58.17 | x1nux | umm |
| 19:58.36 | x1nux | how fix the callerid in the script, is my problem ... |
| 19:58.37 | x1nux | :s |
| 19:59.36 | mmlj4 | yes, it's a problem |
| 20:00.06 | x1nux | so the line in the extension.conf is: exten => 100,3,AGI(script.agi) ? |
| 20:00.08 | x1nux | or |
| 20:00.39 | x1nux | exten => 100,3,Authenticate(script.agi) ? |
| 20:00.48 | mmlj4 | read the AGI chapter in the book, and look at the PHP AGI page on the wiki |
| 20:01.06 | mmlj4 | no, not that way, that won't work |
| 20:01.59 | x1nux | ok |
| 20:02.29 | thomasrr | i am keeping all kinds of crazie notices |
| 20:02.39 | thomasrr | like:chan_sip.c:20761 sip_poke_noanswer: Peer 'kantoor2' is now UNREACHABLE! Last qualify: 18 |
| 20:02.42 | thomasrr | and |
| 20:03.01 | thomasrr | chan_sip.c:9923 sip_reg_timeout: -- Registration for 'XXX@sip.voipbuster.com' timed out, trying again (Attempt #2) > doing dnsmgr_lookup for 'sip.voipbuster.com' |
| 20:03.06 | thomasrr | how can i solve these issues? |
| 20:03.20 | thomasrr | because when it's not registered i can't make outgoing calls |
| 20:03.27 | thomasrr | then i hear like bliep bliep bliep |
| 20:10.44 | *** join/#asterisk Greek-B0y (n=greek@41.222.89.77) |
| 20:10.58 | *** part/#asterisk x1nux (n=x1nux@unaffiliated/x1nux) |
| 20:12.53 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 20:13.10 | *** part/#asterisk mkillebrew (n=mkillebr@edge.50ae.net) |
| 20:15.45 | *** join/#asterisk lost_soul (n=shawn@cpe-67-241-67-197.twcny.res.rr.com) |
| 20:19.05 | *** join/#asterisk Gremlin (n=MadMoney@unaffiliated/gremlin) |
| 20:19.41 | *** join/#asterisk telecos (n=sergio@210.167.219.87.dynamic.jazztel.es) |
| 20:21.20 | Gremlin | Hi, I'm trying to save money on a 20 seat ATA. |
| 20:21.59 | drmessano | ~save money |
| 20:22.00 | infobot | ACTION runs into a burning building and saves money from certain death. Is that enough? |
| 20:22.02 | drmessano | ~savemoney |
| 20:22.03 | infobot | <Gremlin> It's okay, but I was hoping to save some money with a soldering iron and a bunch of sound cards. |
| 20:22.18 | Gremlin | I come on, that was like two months ago. |
| 20:22.52 | drmessano | or last week |
| 20:23.27 | Gremlin | Actually, I found out that idea would work if I soldered wires to each phone's mic and speaker and then to the sound cards, bypassing the DTMF stuff (no dial tone). |
| 20:23.35 | thomasrr | anyone? |
| 20:24.42 | Gremlin | I've actually abandoned all soldering ideas. |
| 20:25.12 | drmessano | Actually, we told you why your idea wouldnt work.. but I guess you remember it differently |
| 20:25.55 | Gremlin | I remember perfectly... RJ11 phone lines are fundamentally different. |
| 20:26.24 | MaliutaLap | you were going to power these things how? |
| 20:26.28 | Gremlin | The voltage is totally different, the phone pulls more current, et cetera. |
| 20:26.48 | MaliutaLap | for a start |
| 20:28.34 | Gremlin | Basically, the only way my idea would have worked would be hooking the speaker in the phone to the audio line out on the sound card and the putting a compatible microphone in the receiver (and hooking it to the line in on the sound card). At that point, it wouldn't be feasible. |
| 20:28.44 | Gremlin | So while my idea would work, it wouldn't actually save any money. |
| 20:30.08 | drmessano | How would the extensions ring and go on/off hook? |
| 20:30.10 | MaliutaLap | we really should give infobot a "urmom" feature |
| 20:30.35 | *** join/#asterisk stijnbe (n=stijnbe@d54C16246.access.telenet.be) |
| 20:30.56 | Gremlin | Simple $1.00 head phones with a mic are the proper solution (tied to a soft SIP phone). I just need to figure out how to run 30 soft phones on one computer (which is easy). |
| 20:30.57 | thomasrr | how can i solve timeout issues? |
| 20:31.18 | thomasrr | Gremlin: you can use Telephone.app :+ |
| 20:31.22 | thomasrr | and then 30 accounts |
| 20:31.31 | MaliutaLap | thomasrr: wait? |
| 20:31.43 | MaliutaLap | thomasrr: that solves most timeout issues ;) |
| 20:31.48 | Gremlin | Simple: use a CLI Linux soft phone app. |
| 20:31.55 | Gremlin | With many consoles |
| 20:32.14 | MaliutaLap | hands Gremlin a fork() |
| 20:32.34 | Gremlin | To pick up a line, do Alt-F6 or whatever key combo to get to the console with the CLI softphone app running in the shell. |
| 20:32.41 | thomasrr | MaliutaLap: yes, indeed but when its not connected you can't call :) |
| 20:33.12 | Gremlin | It could support many phones simultaneously with one computer as long as the keyboard is centrally located. |
| 20:33.26 | MaliutaLap | thomasrr: the problem is network related |
| 20:33.29 | drmessano | So all 20 agents need to be near the button? |
| 20:33.47 | drmessano | What if two calls come in at once? |
| 20:33.54 | MaliutaLap | thomasrr: you're seeing a timeout because you can't reach the host within the TTL of the packets |
| 20:34.10 | thomasrr | MaliutaLap: ooh |
| 20:34.11 | MaliutaLap | thomasrr: so get a better net connection for your SIP stuff :) |
| 20:34.22 | Gremlin | drmessano: That problem is solved, too. A PS/2 keyboard PCB with wired soldered to the contacts going to switched to activate the key combinations. |
| 20:34.28 | MaliutaLap | thomasrr: are you QoSing this link? |
| 20:34.37 | thomasrr | MaliutaLap: probalby not |
| 20:34.57 | Gremlin | thomasrr: Don't do VOIP on Dial-Up unless you're trying to save money. |
| 20:35.09 | thomasrr | i am having cable and ADSL :P |
| 20:35.14 | MaliutaLap | thomasrr: because that might make a difference, give the SIP/RTP packets a higher priority and guaranteed bandwidth |
| 20:35.17 | thomasrr | getting fibre |
| 20:35.41 | thomasrr | http://www.voip-info.org/wiki/view/Draytek,+SIP,+and+QoS :) |
| 20:35.49 | MaliutaLap | thomasrr: QoS is still a sane thing with any connection not dedicated to SIP/RTP traffic |
| 20:36.09 | thomasrr | oh it will be dedicated later when i am getting fibre |
| 20:36.23 | thomasrr | ADSL will then be backupline + SIP |
| 20:36.32 | thomasrr | only those slackers at the fibre isp... |
| 20:36.40 | MaliutaLap | thomasrr: there are other ways ... terminate your dsl/cable into an openbsd box it's QoS is good |
| 20:36.57 | Gremlin | thomasrr: Will this be for a business? |
| 20:37.21 | MaliutaLap | urges the .au NBN forward ... I need fibre to this place |
| 20:38.42 | Gremlin | Sewer systems would be great for broadband. |
| 20:39.26 | MaliutaLap | IP over TURD |
| 20:40.09 | MaliutaLap | recommends IETF establish the new protocol |
| 20:40.25 | thomasrr | difficult stuff |
| 20:40.47 | thomasrr | Gremlin: yes, my business :P |
| 20:40.48 | MaliutaLap | thomasrr: you mean constipated? |
| 20:41.17 | thomasrr | hehe |
| 20:44.18 | MaliutaLap | has a suggestion for dealing with Gremlin/MadMoney -> http://www.questionablecontent.net/view.php?comic=1410 |
| 20:44.20 | thomasrr | i will dig up the manual of the draytek |
| 20:45.03 | MadMoney | Hi, I'm Jim Cramer. Today, I'm going to tell you how to save money on capital gains taxes by buying high dividend yield stocks that are converting market capitalization into dividend payouts. |
| 20:47.11 | *** join/#asterisk killfill (n=killfill@200.63.96.244) |
| 20:48.55 | thomasrr | MadMoney: |
| 20:49.05 | thomasrr | enough money here ;) |
| 20:49.35 | thomasrr | problem is that it's invested in stuff :) |
| 20:50.27 | thomasrr | even clients would pay :P |
| 20:50.43 | thomasrr | now they all pay after 100 days :( :( |
| 20:50.54 | MadMoney | You aren't trying to start a VoIP company off of ADSL are you? |
| 20:51.10 | thomasrr | no :) |
| 20:51.17 | thomasrr | just want to get rid of this siemens pbx :P |
| 20:52.49 | killfill | hi. im having problems with analog TDM410P with fxo modules. When the person calling form PSTN hangs up, asterisk takes too long to know that. Do you guys recomend me something to read about it? any tips welcome |
| 20:53.51 | MadMoney | fxo meaning Foreign Exchange Organization |
| 20:55.06 | MaliutaLap | thomasrr: ewww, I hate those things ... we looked at a way of doing that at one place I was at, was going to cost us a fortune to either do ISDN trunking to * or to put in a SIP card |
| 20:55.07 | MadMoney | How long does it take? |
| 20:55.38 | MaliutaLap | killfill: do you have your zone settings correct? |
| 20:55.38 | thomasrr | MadMoney: it's a BizIP thing |
| 20:55.45 | thomasrr | we got some fucked in the ass hard by Siemens |
| 20:56.04 | MaliutaLap | thomasrr: is there any other way to get it from them? |
| 20:56.06 | thomasrr | phones of 250-300$USD per piece :+ |
| 20:56.07 | MadMoney | SO many interpretations |
| 20:56.17 | killfill | MaliutaLap, yup, defaultzone and loadzone in zaptel.conf are just fine. |
| 20:56.41 | thomasrr | MadMoney: you can use the resellers but if the phones ain't SIP compatibile |
| 20:56.47 | killfill | i guess that values, what they do is tell asterisk to ge tthe info from indications.cnf right? |
| 20:56.49 | thomasrr | and the PBX sucks balls so that it hangs everyday... |
| 20:56.54 | MaliutaLap | killfill: and you have the signalling right? |
| 20:57.02 | thomasrr | never again Siemens :P |
| 20:57.26 | killfill | MaliutaLap: got them with fxoks |
| 20:57.30 | MaliutaLap | thomasrr: the only siemens kit I like is the medical imaging stuff I get thrown into all the time |
| 20:58.06 | MaliutaLap | thomasrr: and I've even thrown up on that :) |
| 20:58.16 | thomasrr | lol |
| 20:58.24 | thomasrr | its nice |
| 20:58.33 | thomasrr | but it's weekend no medical stuff now :) |
| 20:58.47 | MaliutaLap | killfill: sounds odd. there are other ways that you could be detecting a remote PSTN hangup |
| 20:58.47 | killfill | MaliutaLap: i mean with fxsks, sorry... |
| 20:58.55 | thomasrr | tomorrow it's time again to play with stem cells again :> |
| 20:59.11 | killfill | MaliutaLap: other ways? |
| 20:59.13 | MaliutaLap | killfill: not knowing where you are makes it a little difficult |
| 20:59.21 | thomasrr | cutting mice tomorrow :+ |
| 20:59.38 | MaliutaLap | killfill: the hangup could be a voltage change on the line, or other signalling |
| 20:59.45 | killfill | where am i?.. well im in Chile, sudamerica. [cl] exists in indications.conf |
| 20:59.49 | MaliutaLap | killfill: it varies from country to country |
| 21:00.15 | killfill | yup, i know... |
| 21:01.47 | killfill | and more so, there seem to be difference between telcos in here.. :S |
| 21:02.15 | thomasrr | h:) |
| 21:02.21 | thomasrr | lovely |
| 21:02.37 | thomasrr | 17 minutes to burn a DVD :( |
| 21:03.52 | MaliutaLap | killfill: have you tried using a different signalling method? ls or gs? |
| 21:04.30 | killfill | MaliutaLap: not really. i have read ks is better in all cases than the others? |
| 21:05.00 | thomasrr | MaliutaLap: maybe the timeouts are also caused by my vmware machine |
| 21:08.03 | MaliutaLap | thomasrr: generally you're seeing an actual packet/session timeout |
| 21:11.38 | MaliutaLap | killfill: I'm just digging for some info I know I've read |
| 21:19.05 | thomasrr | :) |
| 21:20.46 | thomasrr | bye |
| 21:20.50 | thomasrr | thanks all |
| 21:20.55 | thomasrr | i will experiment with QoS tomorrow |
| 21:21.18 | thomasrr | might will call the manufacturer of the router/modem thingy |
| 21:28.37 | TheKmartTroll | The easiest way to fix problems with the connection is to threaten to cancel your service. |
| 21:33.26 | *** join/#asterisk wierdo (n=jimmy@wifi-traf5.networx-bg.com) |
| 21:35.37 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 21:37.28 | MaliutaLap | killfill: all the stuff I remember and am reading says it's in the zone info and/or the signal type |
| 21:42.26 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:59.38 | *** join/#asterisk phurl (n=mdupont@62.103.98.54) |
| 22:11.35 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.229.58) |
| 22:13.56 | killfill | hm... |
| 22:32.48 | *** join/#asterisk saftsack (n=saftsack@p579246C0.dip.t-dialin.net) |
| 22:35.32 | MaliutaLap | anyone from .au know about making a direct connection to an iinet SIP account? |
| 22:40.44 | killfill | MaliutaLap, ive try with loop start and the problem persist. :S, could not make ztcfg pass with ground start |
| 22:43.01 | killfill | asterisk detects the pstn pary haged up after like 2.5 rings.. :S |
| 22:43.47 | *** join/#asterisk apeiron (n=Chris@c-76-124-252-61.hsd1.pa.comcast.net) |
| 22:53.17 | *** join/#asterisk doolittlework (n=f@196.211.34.2) |
| 22:53.26 | doolittlework | hi there anyone home? |
| 22:54.39 | doolittlework | voip is cahnging the world and then u get customers that is stuck in the past, does anyone know how one sets asterisk to dump cdr records to a com port? |
| 22:55.13 | doolittlework | i have a customer that wants his call reports in his old management system? |
| 22:55.31 | killfill | when asterisk (or zaptel) thinks the line is still ringing (and making the SIP phones ring), and its not, then i can hear that linea is not busy, and has the typical dial tone. |
| 22:55.51 | killfill | so io guess busydetect is not the option im looking for.. :P |
| 22:56.13 | killfill | is there something similar that could help asterisk better detect when a caller has already hanged up? |
| 22:56.42 | doolittlework | channel status |
| 22:57.15 | killfill | channel status? |
| 22:57.45 | killfill | doolittlework: that do you mean |
| 22:58.17 | doolittlework | check if "chanIsAvailable" |
| 22:59.27 | killfill | hm.. |
| 22:59.33 | doolittlework | killfill: check this out http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS |
| 22:59.46 | doolittlework | what do yo want to do killfill? |
| 23:00.20 | doolittlework | are u using zap or sip? |
| 23:00.34 | killfill | hm.. |
| 23:00.40 | killfill | zap. |
| 23:01.13 | killfill | when someone calls from pstn, and i dont answear any sip phone. Asterisk takes like 3 rings to detect that the caller has hangup. |
| 23:01.46 | killfill | and it happends bad thing when i get a new call within thouse 3 rings. (like callerId problem) |
| 23:01.54 | doolittlework | no way of getting around that, belive me i tried |
| 23:02.18 | killfill | I know one of thouse linksys SPA works quite fine |
| 23:02.31 | doolittlework | u using diguim? |
| 23:02.41 | killfill | yup |
| 23:02.51 | killfill | i tries with a sangoma and clone too.. |
| 23:02.54 | killfill | tried |
| 23:03.25 | doolittlework | yip nalog sucks, get a bri and your problem will go away |
| 23:03.31 | doolittlework | analog |
| 23:03.53 | killfill | yeah i know.. i got analog or pri here. i really need analog.. :S |
| 23:04.18 | doolittlework | to ry and speed it up you can disable cli, and put immediate=yes |
| 23:04.30 | doolittlework | try |
| 23:04.39 | killfill | disable cli? |
| 23:04.45 | doolittlework | callerid |
| 23:05.27 | killfill | Ah, but i guess that will help on make it faster to ring. I dont really have problems with initial ringing (maybe its loosing 1 ring) |
| 23:05.38 | killfill | but 1 is ok. |
| 23:06.02 | killfill | half really.. |
| 23:06.13 | doolittlework | <PROTECTED> |
| 23:06.38 | killfill | nothing special, im putting the calls throught a queue |
| 23:07.35 | doolittlework | busydetect=? in zapata.conf? |
| 23:08.14 | killfill | http://www.pastebin.ca/1433375 |
| 23:08.23 | doolittlework | are u from US? |
| 23:08.27 | killfill | nope |
| 23:08.32 | killfill | cl.. |
| 23:08.33 | killfill | :) |
| 23:09.05 | doolittlework | use pastebin.com for some reason i can't viewc.ca |
| 23:10.00 | killfill | http://pastebin.com/m6bf5d5bf |
| 23:10.44 | killfill | hanguponpolarityswitch=yes is getting ignored tho. i think asterisk 1.4.4 does not understand that. |
| 23:10.55 | doolittlework | answeronpolarityswitch=yes what does this do never seen it before |
| 23:10.56 | killfill | (next week ill upgrade) |
| 23:11.05 | killfill | its ignored.. :) |
| 23:11.11 | killfill | readed it somewhere |
| 23:11.41 | doolittlework | where is cl |
| 23:12.16 | killfill | Chile, Sudamerica |
| 23:12.26 | doolittlework | i see |
| 23:13.25 | doolittlework | immediate is more for outgoing calls, so it's not going to help your incoming calls |
| 23:15.33 | doolittlework | i know i had this problem before and gave up in the long run, bri or pri, best bet. There is a anlog gateway that i had that worked like a charm, i think it way megatech |
| 23:15.43 | doolittlework | no sry multitech |
| 23:16.23 | killfill | yeah, actually i loose the faith, time ago too. But i really need it now. ill keep searching :) |
| 23:16.44 | doolittlework | good luck |
| 23:17.14 | killfill | thanks.. :P |
| 23:17.59 | doolittlework | you are not using cellrouters on these channels? |
| 23:18.49 | killfill | nope, plain line |
| 23:20.28 | *** join/#asterisk sah-work (n=Bawbatos@p29158-adsau12honb7-acca.tokyo.ocn.ne.jp) |
| 23:20.52 | doolittlework | good some cellrouter voltage drops on disconnects is so poor that it takes the card forever to realise that the chANNEL HAS BEEN DROPED |
| 23:21.41 | *** join/#asterisk timeshell (n=chatzill@206.248.136.108) |
| 23:23.20 | killfill | what is zap show cadences for?.. |
| 23:23.38 | MikhailGorbachev | Is Asterisk the most efficient way to go from SIP trunk to FXO lines? |
| 23:24.06 | MikhailGorbachev | Maybe I mean FXS |
| 23:24.07 | killfill | does it have something to do with indications.conf?.. |
| 23:26.19 | *** join/#asterisk maximo (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 23:28.08 | doolittlework | indication is more tone related, ring tone, here in RSA its linke rrrrrrrrrrrrrrrring rrrrrrrrrrrrring, whereas in US its ring ring |
| 23:28.40 | doolittlework | waht do u mean MikhailGorbachev? |
| 23:29.30 | killfill | Ah, nothing to do with detection of events on a line. |
| 23:30.23 | doolittlework | i dont think so , might be wrong |
| 23:32.54 | doolittlework | you might be onto something killfill |
| 23:33.05 | doolittlework | check http://www.voip-info.org/wiki/view/Asterisk+config+indications.conf |
| 23:34.45 | Maximo | hi my people, can you give me the instruc for setting up asterisk using Ubuntu? |
| 23:34.56 | Maximo | a desktop |
| 23:35.57 | doolittlework | Maximo https://wiki.ubuntu.com/AsteriskOnUbuntu |
| 23:36.11 | Maximo | doolittlework: thanks alot |
| 23:45.15 | *** join/#asterisk s14ck (n=s14ck@190-76-124-192.dyn.movilnet.com.ve) |
| 23:46.05 | killfill | woot |
| 23:46.11 | killfill | [May 24 19:45:14] NOTICE[2710]: indications.c:505 ast_unregister_indication_country: Removed default indication country 'cl' |
| 23:46.13 | doolittlework | sounds good |
| 23:46.21 | killfill | hm... unregister?.. |