| 00:00.22 | rare1980_ | can jump into vnc with same pass and check plz just for 1 min ? |
| 00:00.23 | rare1980_ | plz |
| 00:03.04 | *** join/#asterisk jksM (jks@193.189.93.254) |
| 00:03.18 | Systemt` | [TK]D-Fender: http://pastebin.com/pPNcxBFG |
| 00:04.38 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 00:05.33 | leifmadsen | Asterisk 1.4.30-rc3, 1.6.0.26-rc1, 1.6.1.18-rc1, and 1.6.2.6-rc1 are now available! See the release announcement for more information: http://www.asterisk.org/node/49915 |
| 00:06.31 | ruben23 | <PROTECTED> |
| 00:06.38 | leifmadsen | hi |
| 00:06.52 | rare1980_ | miamiseb: http://www.voip-info.org/wiki/view/port+forwarding |
| 00:07.04 | ruben23 | <PROTECTED> |
| 00:07.07 | rare1980_ | do u think this port forwarding will help me? |
| 00:07.24 | leifmadsen | ruben23: just ask the question and if I have time and know the answer I'll respond - maybe someone else will as well :) |
| 00:07.26 | ruben23 | i already installed im just now with the dial plan |
| 00:07.29 | leifmadsen | I've done VERY LITTLE with gtalk |
| 00:09.10 | ruben23 | i have an existing dial plan for incoming and want to do gtalk to send IM message for incoming calls displaying it number also..my dial plan is this---> http://pastebin.com/KX64x8dR |
| 00:15.55 | *** join/#asterisk Dibri (~gavit@190.98.33.229) |
| 00:17.12 | Systemt` | [TK]D-Fender ? |
| 00:19.29 | miamiseb | Time to go home |
| 00:19.30 | miamiseb | night all |
| 00:19.36 | Systemt` | night |
| 00:21.40 | idespinner | if one has an issue listed in the bug tracker that was labeled as resolved, how does one know what version of asterisk it is fixed in? |
| 00:22.08 | *** join/#asterisk luckyaba (~lucky@ip72-194-218-169.sb.sd.cox.net) |
| 00:29.35 | *** join/#asterisk SaiSoma (~SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |
| 00:35.06 | *** join/#asterisk netpro25_ (~mmanning@64-238-176-105.ksg.apt.gru.net) |
| 00:36.46 | [TK]D-Fender | Systemt`: Contact: <sip:0737000331@192.168.0.102> <--- STILL wrong. |
| 00:37.36 | Systemt` | what i need to do ? |
| 00:37.41 | *** join/#asterisk stevex (~steve@65-120-138-46.dia.static.qwest.net) |
| 00:37.59 | stevex | could someone hel me? I have an asterisk server that has just started causing me problems today. |
| 00:38.06 | stevex | I am not sure if I am dealing with network issues or server issues |
| 00:39.06 | dlynes | stevex, could you perhaps .... elaborate?? |
| 00:39.07 | stevex | When the server starts up all the peers register but when I dial one of the peers I do not hear ringing on my end but the phone I dial rings |
| 00:39.20 | stevex | However, when the other phone is answered there is no audio |
| 00:39.30 | stevex | every few minutes the peers become unreachable |
| 00:39.33 | dlynes | ~nat |
| 00:39.34 | infobot | hmm... nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 00:39.36 | stevex | I am also getting this error |
| 00:39.37 | stevex | [Mar 4 17:37:34] WARNING[3852]: chan_sip.c:1976 retrans_pkt: Maximum retries exceeded on transmission e6d518fb-243e720@10.10.42.102 for seqno 110 (Non-critical Request) -- See doc/sip-retransmit.txt. |
| 00:39.41 | stevex | [Mar 4 17:37:59] WARNING[3852]: chan_sip.c:1976 retrans_pkt: Maximum retries exceeded on transmission 81ec58da-b6f8e6ec@10.10.42.102 for seqno 102 (Non-critical Request) -- See doc/sip-retransmit.txt. |
| 00:39.46 | stevex | [Mar 4 17:37:59] WARNING[3852]: chan_sip.c:1976 retrans_pkt: Maximum retries exceeded on transmission e08f19d2-58ef6b16@10.10.42.102 for seqno 106 (Non-critical Request) -- See doc/sip-retransmit.txt. |
| 00:39.58 | dlynes | stevex, see the part where it says 'WARNING'? That says it's not an error |
| 00:40.27 | dlynes | stevex, anyways...suffice it to say, you've got issues with your nat |
| 00:41.26 | stevex | What might have caused my existing settings to suddenly stop working? This set-up has been working without fail for nearly the last year. |
| 00:41.35 | dlynes | stevex, have you tried looking up the comments in your /etc/asterisk/sip.conf file for externip, localnet, localmask, canreinvite, and nat? |
| 00:41.37 | stevex | I have nat=yes on all sip accounts. |
| 00:41.50 | dlynes | stevex, are you using a low end router? |
| 00:42.01 | stevex | No |
| 00:42.05 | stevex | Juniper SSG-140 |
| 00:42.10 | stevex | This is in an office |
| 00:42.38 | dlynes | stevex, so these sip peers and asterisk server are all on the local network, and none of them are on a vpn? |
| 00:42.44 | stevex | correct |
| 00:42.59 | *** join/#asterisk coppice (~chatzilla@94.201.17.210.dyn.pacific.net.hk) |
| 00:43.18 | dlynes | stevex, and there's absolutely nothing in between your sip peers/sip users and your asterisk box? |
| 00:43.45 | dlynes | stevex, and the sip peers/sip users are on the same subnet as your asterisk box? |
| 00:43.52 | stevex | that is correct |
| 00:44.16 | dlynes | stevex, then none of the issues you've got probably even exist |
| 00:44.51 | dlynes | stevex, those problems are only problems i've seen occur when nat is involved |
| 00:45.07 | dlynes | stevex, you're not running iptables on the asterisk box, either? |
| 00:45.18 | dlynes | stevex, /sbin/iptables -nL to verify |
| 00:45.19 | stevex | IPtables and selinux are turned off |
| 00:45.47 | dlynes | stevex, iptables is turned off? what exactly does that mean? |
| 00:45.56 | stevex | You are right, this is a problem I have seen before with NAT on a different asterisk server that I use for remote users. This is the first time I have seen something like this in the office enviroment. |
| 00:46.09 | dlynes | stevex, you mean default policy on input, output and forward is accept? And there are no other rules? |
| 00:46.49 | stevex | I mean I turned off iptables when I suspected that might be causing issues. |
| 00:47.05 | dlynes | stevex, what does turning off iptables entail? |
| 00:47.33 | dlynes | stevex, to me, you're either filtering traffic, or you're not...there's no on/off switch |
| 00:47.54 | stevex | service iptables stop |
| 00:48.22 | dlynes | stevex, I have no idea what that does, and I don't have a redhatish box to test, either |
| 00:48.34 | stevex | well it stops filtering traffic |
| 00:48.34 | dlynes | stevex, does it mean that all policies are accept, and there's no additional rules? |
| 00:49.00 | dlynes | stevex, when i issue a shorewall stop, it's not the same thing as shorewall clear |
| 00:49.17 | dlynes | stevex, when i do shorewall stop, it's not the same as all policies to accept and no additional rules |
| 00:49.35 | dlynes | stevex, that's why i'm asking you to do an iptables -nL, to verify |
| 00:50.17 | stevex | [root@crasterisk ~]# iptables -nL |
| 00:50.17 | stevex | Chain INPUT (policy ACCEPT) |
| 00:50.17 | stevex | target prot opt source destination |
| 00:50.17 | stevex | Chain FORWARD (policy ACCEPT) |
| 00:50.17 | stevex | target prot opt source destination |
| 00:50.19 | stevex | Chain OUTPUT (policy ACCEPT) |
| 00:50.22 | stevex | target prot opt source destination |
| 00:50.23 | dlynes | Ok, good |
| 00:50.24 | stevex | Chain L (0 references) |
| 00:50.25 | dlynes | thank you |
| 00:50.27 | stevex | target prot opt source destination |
| 00:50.41 | dlynes | You could have done a pastebin, however |
| 00:50.59 | stevex | I am not familiar with pastebin |
| 00:51.06 | dlynes | ~pb |
| 00:51.07 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 00:52.23 | *** join/#asterisk spartan07 (~spartan07@wsip-70-169-241-66.oc.oc.cox.net) |
| 00:52.35 | stevex | ahh nifty |
| 00:52.41 | dlynes | stevex, can you pastebin the following?: /sbin/ifconfig xxx, ethtool -S xxx, and netstat -s, where 'xxx' is your ethernet device that's accepting traffic from the sip peers/users |
| 00:53.35 | dlynes | stevex, fyi, you might not currently have ethtool installed |
| 00:54.03 | spartan07 | Can I run an asterisk server for 20 lines (users) on a P4 or is that low balling too much? |
| 00:55.32 | stevex | http://pastebin.com/cJhdZn2Y |
| 00:57.37 | dlynes | stevex, ummm.....you put in 'ethtool -S eth1, and netstat -s', literally |
| 00:57.42 | stevex | oops |
| 00:57.46 | stevex | just now caught what I did |
| 00:57.50 | stevex | silly copy and paste |
| 00:58.56 | stevex | http://pastebin.com/8Wvqt91C |
| 01:00.12 | Systemt` | [TK]D-Fender: are u still ther? |
| 01:00.31 | dlynes | stevex, Now, did you read any of what you pastebinned? |
| 01:01.16 | dlynes | stevex, take a look at netstat -s, specifically |
| 01:01.46 | dlynes | stevex, also, the 'rx_filtered_packets' in ethtool -S concerns me...I'm not sure if that's something to do with iptables, or not |
| 01:01.58 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 01:02.02 | stevex | "68 dropped because of missing route"? |
| 01:02.06 | stevex | 768* |
| 01:02.13 | dlynes | stevex, that's one concern, yes |
| 01:02.44 | stevex | "1582 packets to unknown port received. 40629 packet receive errors |
| 01:02.53 | dlynes | stevex, that tells you that you were trying to send packets to a machine that is not on your local subnet, and there's no longer a route to that machine |
| 01:03.24 | dlynes | stevex, the 1582 packets to unknown port received' is not really a concern, because the number is so low...you'll always get some of those |
| 01:04.10 | dlynes | stevex, the 40629 packet receive errors is a huge concern though, considering you've only received 7589 udp packets |
| 01:05.18 | stevex | so now if I only knew what might be causing that |
| 01:06.47 | dlynes | stevex, also, you're getting considerably more tcp traffic, than udp traffic |
| 01:07.24 | stevex | I am sshed into it so that may not be surprise |
| 01:07.37 | dlynes | stevex, try checking your asterisk logs, to see if anything looks amiss |
| 01:07.55 | dlynes | stevex, if you're still not seeing anything odd, turn on sip debug |
| 01:08.49 | dlynes | stevex, but based on what I see in those dumps, I'd say that you probably don't know your network topology quite as well as you think you do |
| 01:09.38 | dlynes | stevex, can you try pastebinning a 'sip show peers' dump? |
| 01:10.39 | dlynes | spartan07, it's all going to depend on what call quality you're expecting, and what codecs you're going to use |
| 01:11.25 | stevex | http://pastebin.com/J794x6Ww |
| 01:12.19 | spartan07 | I would like decent call quality and I would use whatever codecs that would work best with counterpath softphones |
| 01:12.21 | Systemt` | ~sipnat |
| 01:12.22 | 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 |
| 01:12.22 | dlynes | stevex, and sip show peer 116-1? |
| 01:12.35 | dlynes | spartan07, ulaw/alaw |
| 01:13.02 | dlynes | Systemt`, his system's acting like it's on nat, but it appears it's not a nat issue...it's just weird |
| 01:13.40 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 01:13.56 | stevex | http://pastebin.com/3L9E4V8C |
| 01:14.02 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 01:14.02 | dlynes | stevex, also, 'Excel' is on a different subnet |
| 01:14.03 | spartan07 | ulaw I believe |
| 01:14.10 | stevex | Excel is yes |
| 01:14.20 | stevex | that is our connection to the world. |
| 01:14.42 | stevex | Excel telecom |
| 01:14.45 | *** join/#asterisk Dibri (~gavit@190.98.33.229) |
| 01:14.51 | dlynes | stevex, ok, now are you getting those one sided conversations when you've got one sip peer talking to another on a local call, and not on an outside call? |
| 01:15.18 | stevex | Happens on local calls |
| 01:15.39 | dlynes | stevex, so happens on say 101-1 talking to 116-1? |
| 01:16.00 | dlynes | stevex, also, is there a reason why you've got six sip peers defined per phone? |
| 01:16.17 | dlynes | stevex, does each peer represent a completely different DID? |
| 01:16.56 | dlynes | stevex, i.e. do you have six different phone numbers? |
| 01:17.13 | stevex | Yes taht is so we can use different DIDs to the phones |
| 01:17.17 | stevex | these are six line business phones |
| 01:17.36 | stevex | each person as their private line and several business lines |
| 01:17.41 | dlynes | stevex, ok...just asking in case you didn't know that you can assign multiple lines to a single sip peer name |
| 01:18.30 | dlynes | stevex, i.e. if they have one private line, and five appearances for the same business phone number, you could assign one sip peer name to the first five appearances, and a second sip peer name to the sixth appearance |
| 01:18.48 | dlynes | stevex, that would cut down on the cpu usage of your phone, which might be causing your phone to flake out |
| 01:19.08 | stevex | Good advice |
| 01:19.17 | stevex | No I wasn't aware I could do that |
| 01:19.17 | dlynes | stevex, especially if you're using blf...cheap phones tend to flake out very well with blf |
| 01:19.30 | stevex | and we are using cheap linksys phones with blf |
| 01:19.53 | dlynes | stevex, test it with one phone first though, in case the linksys 942 or whatever it was is not capable of applying one sip peer name to multiple appearances |
| 01:19.55 | adnc | what is app_morsecode.so? |
| 01:20.47 | dlynes | stevex, linksys phones are actually more expensive than aastra 9143i's, aren't they? |
| 01:21.24 | dlynes | adnc, core show application morsecode |
| 01:21.26 | Systemt` | night bbl |
| 01:21.41 | stevex | have sip debug set for 116-1 |
| 01:21.44 | stevex | does this tell you anything? |
| 01:21.45 | stevex | http://pastebin.com/rPWuRN36 |
| 01:21.49 | dlynes | stevex, anyways...if i was a betting man, I'd say your issue is completely related to the quality of the phones |
| 01:22.29 | *** part/#asterisk pacmanfan (~pacmanfan@d4-44.rb5.clm.centurytel.net) |
| 01:22.41 | dlynes | stevex, yeah...seems like your blf is flaking everything out |
| 01:23.56 | dlynes | stevex, try disabling blf, to see if your problem magically clears up |
| 01:25.20 | dlynes | stevex, I had the same issue with Aastra 9133i's, when they were overdosing on blf's, and 57i's as well |
| 01:25.50 | dlynes | stevex, Upgrading the 9133's to 9143's solved the problem (same cpu and memory as the 57i, but less blf programmability) |
| 01:26.15 | dlynes | stevex, the 57i's even had weird shit happening with them like random lock ups when too many blf's got activated all at once |
| 01:27.26 | stevex | stupid question but I set this up a year ago and first time I ever set-up blf. What is the quickest way to disable it? |
| 01:27.50 | *** part/#asterisk DMeloUK (~DominicMe@64.129.95.226) |
| 01:32.11 | stevex | so it has to be turned off on every phone? |
| 01:33.57 | spartan07 | dlynes, if Im expecting to do no more than 10 at the most concurrent calls would a P4 >1G ram be alright? |
| 01:34.30 | spartan07 | most if not all phones would be softphones and ip phones |
| 01:44.32 | stevex | I had them unplug the phone at the reception desk that has the sidecar |
| 01:44.34 | stevex | I rebooted the server |
| 01:44.37 | stevex | and all was well |
| 01:44.50 | stevex | the blf on the side car bust have been the issue |
| 01:44.55 | stevex | now have to figure it out tomorrow |
| 01:44.58 | stevex | thank you dlynes for your help |
| 01:45.03 | stevex | and good night |
| 01:45.17 | stevex | exit |
| 01:45.18 | *** join/#asterisk Tech_Travis (~Administr@cpe-76-168-191-127.socal.res.rr.com) |
| 01:51.29 | *** join/#asterisk caim (~user@unaffiliated/caim) |
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| 01:56.58 | *** join/#asterisk Jhirley (~Jhirley@adsl-3-193-8.mia.bellsouth.net) |
| 02:00.19 | *** join/#asterisk WinZ (~winz@82.146.61.218) |
| 02:02.19 | WinZ | hello guys |
| 02:03.43 | WinZ | if an incoming fax call always goes to the fax extension when 'faxdetect=yes' in sip.conf, does it mean, that my SIP provider support T.38? |
| 02:05.32 | *** join/#asterisk Majost (~majost@f00kie-1-pt.tunnel.tserv8.dal1.ipv6.he.net) |
| 02:05.33 | WinZ | I mean, if the provider didn't support T.38, faxdetect wouldn't work, would it? |
| 02:06.14 | coppice | faxdetect is not related to T.38 |
| 02:06.40 | WinZ | hmm |
| 02:06.47 | WinZ | "If 'faxdetect=yes' in sip.conf, switch to a 'fax' extension after T38 is negotiated." |
| 02:06.55 | WinZ | written in sip.conf |
| 02:07.23 | coppice | exactly. faxdetect is something done before T.38 is even negotiated |
| 02:08.47 | SaiSoma | http://news.freeallegiance.org/allegiance-celebrates-ten-years-of-gaming/ |
| 02:08.48 | WinZ | oh, that's where my problem is |
| 02:08.54 | *** join/#asterisk Kumbang (~kumbang@167.205.24.69) |
| 02:08.55 | SaiSoma | ahh .c rap .. .sorry, wrong channel |
| 02:09.16 | WinZ | coppice, so, faxdetect just listens for fax tones? |
| 02:09.36 | coppice | yes |
| 02:09.45 | WinZ | thank you! |
| 02:15.50 | *** join/#asterisk nickaugust (~anonymous@167.83.189.72.cfl.res.rr.com) |
| 02:21.38 | *** join/#asterisk bjhaid (~herbayjha@41.206.15.1) |
| 02:25.57 | Majost | I am having some weird issues getting my simple setup working properly. I have two SIP phones, one internal and one connecting from the outside, and one text extension (700). They both register, but only the external phone seems to be working. |
| 02:27.09 | Majost | When I try to dial either the external (200) or the test extention (700) from the internal sip phone, it just gives me a fast bust and the console says: [Mar 4 18:26:56] NOTICE[3303]: chan_sip.c:19546 handle_request_invite: Call from '100' to extension '200' rejected because extension not found. |
| 02:29.20 | Majost | actually, there is a bit more preceeding that notice message -- http://pastebin.ca/1823430 |
| 02:31.03 | Majost | anyway, I am not sure whats causing it, and I haven't found any information on how to fix it |
| 02:31.24 | *** join/#asterisk steliosk (~Stelios@ipa107.2.tellas.gr) |
| 02:42.11 | Majost | placing them both on the internal network doesn't seem to change the behavior either |
| 02:47.29 | *** join/#asterisk mykhyggz (~col@evolone.org) |
| 02:52.53 | *** join/#asterisk iq (~iq@unaffiliated/iq) |
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| 03:24.46 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 03:35.12 | Katty | i have discovered a new Favorite Toy of Merry's |
| 03:35.16 | Katty | anti-static baggies. |
| 03:35.24 | Katty | like network cards and video cards come in |
| 03:38.45 | *** join/#asterisk thansen (~thansen@c-67-177-0-194.hsd1.ut.comcast.net) |
| 03:39.15 | TJNII | Heh. We have dozens upon dozens in the lab. Merry would have a blast. |
| 03:39.33 | Katty | can i come bring him for the afternoon? |
| 03:39.38 | Katty | i promise he will stash them all for you |
| 03:39.45 | TJNII | works at HP, which is well known for overpackaging. Even screws come in anti-static bags. |
| 03:39.59 | TJNII | As long as he's not a cord chewer. |
| 03:40.03 | Katty | TJNII: if i give you my mailing address, will you send me some? :P |
| 03:40.20 | TJNII | I can smuggle some out. |
| 03:40.27 | Katty | :> |
| 03:40.32 | *** join/#asterisk Dibri (~gavit@190.98.33.229) |
| 03:40.32 | Katty | k, lemme know |
| 03:40.41 | Katty | i will make a paypal donation if it will help |
| 03:41.05 | TJNII | Size preference? |
| 03:41.18 | Katty | well the one he just stashed a video card came in |
| 03:41.35 | TJNII | I can't get too many of the big ones (3' square), but I can get a couple. |
| 03:41.38 | ChannelZ | Can you get me a discount on a Z800? :P |
| 03:41.41 | Katty | a 4870 card |
| 03:41.55 | Katty | 3ft is too big |
| 03:41.58 | Katty | merry could fit in it |
| 03:42.05 | Katty | probably like 6" by uhh |
| 03:42.09 | Katty | 3 or 4" |
| 03:42.10 | TJNII | I'm a contractor. No discount. It sucks. |
| 03:42.35 | ChannelZ | hehe bummer |
| 03:42.39 | TJNII | Yea. |
| 03:42.44 | Katty | http://benchmarkreviews.com/images/reviews/video_cards/Sapphire_100243L/Sapphire-Radeon-HD-4870-Kit.jpg |
| 03:42.48 | Katty | ^- static bag that came in |
| 03:43.04 | Katty | the bag was about as big as merry. |
| 03:43.19 | Katty | ryan keeps ammo to one of his guns in a ziplock bag, in the bedroom closet. |
| 03:43.25 | Katty | Merry likes to stash that bag too |
| 03:43.38 | Katty | i think mostly cause it makes noise tho |
| 03:43.46 | TJNII | With the ammo? THat doesn't sound good. |
| 03:44.06 | Katty | he doesn't much care for regular ziplock bags. tho... i don't think i've tried putting those little ringy cat balls in a ziplock bag for him yet. that might do the trick |
| 03:44.13 | Katty | well that's where ryan keeps it |
| 03:44.19 | Katty | and it usually ends up under the bed. |
| 03:45.02 | Katty | the gun is loaded... |
| 03:45.06 | Katty | but you have to uhh. |
| 03:45.10 | Katty | i forget what it's called. |
| 03:46.57 | TJNII | ChannelZ: Besides, this is closer to what I work on. :P http://kabru.eecs.umich.edu/pub/Main/BladeSystem/c7000_serverblades.jpg |
| 03:47.14 | sbrath | Will Hangup work in the dialplan if it dosen't have the () ? |
| 03:47.26 | TJNII | http://i.brentozar.com/wp-content/uploads/2008/01/hp-c7000-back.JPG |
| 03:47.34 | Katty | Naikrovek: http://www.kittyhell.com/wp-content/uploads/2010/01/hello-kitty-xbox-controller.jpg |
| 03:48.08 | TJNII | Hello Kitty chainsaw is still my favorite. |
| 03:48.36 | Katty | http://www.toplessrobot.com/hello-kitty-beer.gif |
| 03:49.13 | ChannelZ | sbrath: as far as I know yes |
| 03:51.33 | sbrath | I'm trying to figure out why the a short extension plan that sets a DB paramater, and then does Hangup dosen't actually Hangup... |
| 03:51.46 | Katty | why can't i find a cute little pink camera case to put my controller into? :< |
| 03:52.52 | sbrath | Their must be one on Ebay :) |
| 03:53.08 | Katty | yeah but i can't make sure my controller fits if it's on ebay. |
| 03:53.39 | Katty | http://www.photoclasses.co.uk/wp-content/uploads/2009/11/51k2-WbQttL._SS500_-300x300.jpg <- why can't they sell this at target?! )= |
| 03:53.50 | Katty | i think my controller would fit nicely into that one! |
| 03:56.49 | Katty | maybe what i need is a miniture tardis |
| 03:56.51 | Majost | does anyone use sipdroid with their asterisk box? |
| 03:56.56 | Katty | that way i can carry a tripod too |
| 03:58.35 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-jnfzpuobrrivvjsr) |
| 04:01.07 | thansen | is 'wav' no longer supported as voicemail format? |
| 04:01.41 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 04:05.02 | *** join/#asterisk fofware (~chatzilla@190.224.76.151) |
| 04:09.31 | thansen | I clearly have 'wav' in my voicemail.conf but I'm getting this.. |
| 04:09.32 | thansen | [Mar 4 21:08:27] WARNING[6269]: file.c:1160 ast_writefile: No such format 'wa' |
| 04:09.32 | thansen | [Mar 4 21:08:27] WARNING[6269]: app.c:868 __ast_play_and_record: Error creating writestream '/var/spool/asterisk/voicemail/default/100/tmp/IBn9zD', format 'wa' |
| 04:12.02 | TJNII | Have you reloaded the config? |
| 04:15.45 | Katty | http://i458.photobucket.com/albums/qq302/Ghandi_Khan/ferret.jpg |
| 04:20.07 | Majost | So what would cause a phone to be able to hear a person talking, but not be able to speak? |
| 04:21.29 | *** join/#asterisk sip7 (~sip83@69.196.159.201) |
| 04:22.35 | sip7 | Hi, I just have a quick question about Asterisk 1.6.2.5. I am trying "stop now" but the command does not work. Is it deprecated? |
| 04:24.33 | *** join/#asterisk thansen (~thansen@c-67-177-0-194.hsd1.ut.comcast.net) |
| 04:25.12 | thansen | TJNII: yes, I've restarted the whole server trying different things |
| 04:25.53 | TJNII | grep "wa" /etc/asterisk/* | grep -v "wav" |
| 04:26.00 | sip7 | Is anybody here using 1.6.2.5? |
| 04:26.02 | TJNII | My best suggestion. |
| 04:27.26 | TJNII | Majost: Is NAT involved at all? |
| 04:27.40 | Majost | TJNII, yes. |
| 04:27.50 | TJNII | Majost: In what way? |
| 04:28.28 | Majost | The phone which is not able to send audio is outside the nat, the asterisk server is inside |
| 04:28.37 | TJNII | ~sipnat |
| 04:28.38 | 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 |
| 04:33.37 | *** join/#asterisk skymeyer (~skymeyer@91.183.54.9) |
| 04:33.41 | LemensTS | sip7: type 'help' at the cli |
| 04:33.45 | LemensTS | it will show u commands |
| 04:35.30 | Majost | TJNII. yay! Thanks! |
| 04:54.51 | *** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110) |
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| 05:26.45 | *** part/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 05:29.05 | sip7 | LemensTS: I found "core stop now" and "core stop gracefully" in the list. I used "core stop now"; when I restarted asterisk (1.6.2.5), those commands disappeared... |
| 05:29.34 | *** join/#asterisk jermudgeon (~jermudgeo@15-138-42-72.gci.net) |
| 05:39.49 | *** join/#asterisk pyite (~dschreibe@unaffiliated/pyite) |
| 05:45.28 | sip7 | Is 1.6.2.5 stable enough for a production environment? |
| 05:56.00 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
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| 07:24.11 | cvnet | hi all |
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| 08:16.11 | hipitihop | how does one transfer a call from one phone to another |
| 08:16.40 | tuxx- | http://www.voip-info.org/wiki/view/Asterisk+config+features.conf |
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| 08:21.02 | ChannelZ | depends on your phone |
| 08:21.53 | hipitihop | tuxx-, thanks, seems I have standard features.conf so parkext => 700 etc. |
| 08:27.59 | *** join/#asterisk c0rnoTa (~c0rnoTa@178.176.140.137) |
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| 08:43.47 | *** join/#asterisk Polysics (~Luca@host39-66-dynamic.30-79-r.retail.telecomitalia.it) |
| 08:43.49 | Polysics | hello |
| 08:44.12 | Polysics | can i set up * so that if someone is not logged in on SIP, they get called on their cellphone? |
| 08:44.25 | Polysics | i already have a VOIP account set up and can dial directly |
| 08:46.37 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 08:50.15 | kaldemar | sure. as always, many ways to do it. either try to dial the peer and catch the failure, or test the availability first e.g. with func DEVICE_STATE. |
| 08:52.06 | hipitihop | ChannelZ, can you be more explicit ? I have an siemens Gigaset E495 base with matching cordless E49H handset .. looks like I better read its manual |
| 08:54.02 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
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| 09:03.08 | Polysics | hmm, i configured a SIP peer for incoming calls on a VOIP provider |
| 09:03.23 | Polysics | but i keep getting a "User not available" message from the provider |
| 09:03.28 | Polysics | what do I need to check? |
| 09:03.52 | *** join/#asterisk Da-Geek (~Da-Geek@62.189.17.99) |
| 09:04.06 | kaldemar | Polysics: sip debug for the call |
| 09:04.35 | Polysics | there is totally no output :-/ |
| 09:04.47 | Polysics | as if something is REALLY wrong :-) |
| 09:05.06 | kaldemar | are you registering to the provider to let them know where you are? |
| 09:08.37 | *** join/#asterisk asteriskuser (~asterisku@195.145.16.195) |
| 09:08.39 | asteriskuser | hi |
| 09:09.02 | Polysics | well, i can do VOIP to cellphone calls on the same account, so I suppose so |
| 09:10.16 | asteriskuser | everybody here? |
| 09:11.05 | Polysics | kaldemar, http://pastebin.com/JzAZYVkw |
| 09:11.08 | Polysics | not much in the debug |
| 09:11.45 | c0rnoTa | asteriskuser: Hi! Your nickname as like login for connecting my asterisk to CDR database ^) |
| 09:11.46 | kaldemar | Polysics: they don't know how to authenticate to you. are they supposed to? |
| 09:12.14 | Polysics | i would say no |
| 09:12.15 | *** join/#asterisk grEvenX (~even@apb9hb.ip.ssc.net) |
| 09:12.40 | kaldemar | Polysics: if not, add insecure=port,invite for the peer |
| 09:13.05 | Polysics | sip show peers shows the peer as registered |
| 09:13.09 | kaldemar | port may not be needed, try with invite only first. |
| 09:13.32 | Polysics | sip.messagenet.it/5379434 212.97.59.76 N 5061 OK (34 ms) |
| 09:13.37 | kaldemar | they still may have some perverted setup that cannot authenticate invites. |
| 09:13.53 | kaldemar | registering is a separate animal from actual calls. |
| 09:14.16 | Polysics | i have no invite option under my sip.conf entry. what is the syntax, please? |
| 09:14.34 | *** join/#asterisk pengo (~xtofu@124-168-115-150.dyn.iinet.net.au) |
| 09:15.09 | kaldemar | Polysics: insecure=invite |
| 09:15.21 | Polysics | oh, i have "very" at the moment there |
| 09:15.31 | Polysics | kaldemar, http://pastebin.com/GynFfNmK is the sip.conf entry |
| 09:15.51 | kaldemar | what version are you using? |
| 09:16.49 | Polysics | 1.6.1.11 |
| 09:18.09 | kaldemar | there is no "very" in 1.6.1. take a look at the sample config. it's either port, invite or both (or no). |
| 09:18.33 | Polysics | so i will stick invite in |
| 09:19.31 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 09:21.14 | Polysics | ok, something is moving... now i get a bunch of incompatible audio format errors :-) |
| 09:21.25 | Polysics | way better than before |
| 09:21.42 | *** join/#asterisk thansen (~thansen@c-67-177-0-194.hsd1.ut.comcast.net) |
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| 09:26.59 | asteriskuser | i have a problem with my asterisk, many threads of asterisk use 100% of cpu power |
| 09:30.35 | asteriskuser | has everybody an idea? |
| 09:31.04 | kaldemar | asteriskuser: are you starting asterisk with option -c? |
| 09:31.20 | Polysics | how do i get a "fallback" option for my simple IVR? |
| 09:31.59 | Polysics | eg. press 1 for X, press 2 for Y, or wait to be connected |
| 09:32.41 | kaldemar | core show function TIMEOUT |
| 09:33.08 | kaldemar | or use waitexten after background and put a goto after waitexten |
| 09:37.45 | Polysics | and in the case i would like to have a "clean" context for incoming calls, do i use GOTO? |
| 09:38.00 | Polysics | i can't find how to set a different context for the incoming peer |
| 09:38.30 | *** join/#asterisk pyite (~dschreibe@unaffiliated/pyite) |
| 09:45.59 | *** join/#asterisk Tim_Toady (~moi@194.219.131.122.dsl.dyn.forthnet.gr) |
| 09:46.32 | c0rnoTa | asteriskuser - looks like deadlocks. Which version do you use? |
| 09:48.18 | Polysics | i am trying to get my IVR menu to work under a different context, but Goto doesn't work :-/ |
| 09:49.31 | kaldemar | Polysics: show how it doesn't work |
| 09:49.33 | Polysics | actually, it is the Goto with an "s" extension that does not work |
| 09:49.52 | Polysics | if i use a numeric extension in my destination context it works - if i use s it doesn't |
| 09:50.03 | Polysics | i still don't get what "s" extension is, probably :-) |
| 09:51.13 | Polysics | wouldn't Goto(emenu,s,1) activate the S extension? |
| 09:51.31 | kaldemar | it is used when there is no information on the called extension. |
| 09:51.49 | kaldemar | yes, it would go to s in emenu. |
| 09:52.16 | *** join/#asterisk Z_God (~julius@wlan228135.mobiel.utwente.nl) |
| 09:52.54 | Polysics | i get "extension not found" |
| 09:53.07 | Polysics | while if i set the extension to any number, it does work |
| 09:54.33 | kaldemar | show it |
| 09:54.59 | Polysics | my extensions.conf: http://pastebin.com/0314yUwq |
| 09:55.24 | Polysics | ignore the few test extensions in incoming |
| 09:55.32 | Polysics | the first couple are what matters |
| 09:57.12 | Polysics | ok, sorted |
| 09:57.19 | Polysics | typo in the file, my bad |
| 09:58.01 | Polysics | is the "Goto" the correct way to handle a menu? |
| 09:58.06 | Polysics | looks like it will work well |
| 09:59.03 | kaldemar | it depends on how the ivr is structured. is is one way and it works. |
| 09:59.48 | *** join/#asterisk KermitTheFragger (~KermitThe@118-197.bbned.dsl.internl.net) |
| 10:00.22 | Polysics | where can i find a complete, simple IVR example? i feel like i am missing a few key pieces |
| 10:00.59 | *** join/#asterisk Slashman (~Slash@ariane.fimasys.com) |
| 10:01.04 | Polysics | working would be 1 for a SIP, 2 for another, wait to be connected, optionally wrong extension |
| 10:09.58 | Polysics | updated extensions.conf, with menu NOT working: http://pastebin.com/zRXHtNnn |
| 10:10.05 | Polysics | i press numbers and nothing happens |
| 10:11.45 | *** join/#asterisk miloux (~KVIrc@milu.rit.se) |
| 10:13.15 | kaldemar | show a cli output too. |
| 10:13.45 | kaldemar | with core debug so that possible DTMF shows on the screen |
| 10:15.34 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 10:17.16 | Polysics | kaldemar, i would say i would blame the client |
| 10:17.21 | Polysics | that is, Zoiper Web |
| 10:17.29 | Polysics | because from a cellphone the menu works properly |
| 10:17.42 | kaldemar | probably a dtmf mode issue |
| 10:19.17 | Polysics | i am using "media inband" mode |
| 10:19.24 | Polysics | will cehck what is wrong |
| 10:19.54 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 10:21.32 | *** join/#asterisk soman (~somnath@212.226.87.141) |
| 10:22.58 | kaldemar | Polysics: then put dtmfmode=inband for the peer in sip.conf or change the mode. if possible, use either rfc2833 or sip info. rfc2833 is the default on asterisk side. |
| 10:23.48 | Polysics | Zoiper calls rfc2833 "media_outband", i'll use that |
| 10:25.42 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 10:26.55 | angryuser | can someone tell me "Eastern time" = gmt ? |
| 10:30.25 | *** join/#asterisk abcmob (~abcmob@unaffiliated/ccbbaa) |
| 10:31.23 | abcmob | What is the "data port" on panasonic kx 580 phones? |
| 10:41.28 | *** join/#asterisk mort_gib (~mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 10:42.48 | *** join/#asterisk rare1980_ (~rare1980@115.186.2.33) |
| 10:42.58 | rare1980_ | iptables -t nat -A prerouting -i eth0 -d 188.138.48.70 -p tcp --dport 80 -j DNAT --to --destination 192.168.56.1 |
| 10:43.07 | rare1980_ | i am tyring to allow port 80 on ubuntu |
| 10:43.14 | rare1980_ | but when as i enter this command i get msg that... iptables v1.3.8 bad ip address |
| 10:43.18 | rare1980_ | any idea? |
| 10:44.44 | kaldemar | dnat params are invalid |
| 10:47.02 | rare1980_ | plz can you correct it? |
| 10:48.33 | kaldemar | --to 192.168.56.1:80 |
| 10:48.38 | kaldemar | iirc |
| 10:50.39 | rare1980_ | kaldemar: please can write down complete synatax ? |
| 10:56.39 | *** join/#asterisk Dibri (~gavit@pop1.isgroup.sr) |
| 10:59.04 | kaldemar | rare1980_: http://lmgtfy.com/?q=iptables+dnat |
| 11:00.39 | *** join/#asterisk gego (~quassel@b238085.customer.hansenet.de) |
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| 11:04.23 | rare1980_ | kaldemar: i have one host computer on ubutnu and i have installed virutalbox on it and i am using elastix .. and i want to access host machine and virtual machine over the vpn .. i can ping host system and virtual system but i can't access virtual system web interface |
| 11:05.08 | rare1980_ | i enterd NAT on iptables but still it is not working :( |
| 11:10.09 | defswork | anyone got a nokia n900 working with asterisk ? I'm only getting one way audio |
| 11:10.58 | Chainsaw | defswork: Sounds like a NAT issue. |
| 11:11.04 | defswork | no nat involved |
| 11:11.14 | defswork | ip -> ip via vpn |
| 11:11.19 | defswork | and direct when at home |
| 11:11.20 | Chainsaw | defswork: There usually is with unidirectional audio problems. |
| 11:11.30 | Chainsaw | defswork: NAT or firewalling. |
| 11:12.20 | kaldemar | rare1980_: check the ip address. unless you configured the nat interface dhcp settings by hand, you're setting a wrong ip address for the guest system. |
| 11:12.36 | defswork | I've got a routed subnet - theres no NAT |
| 11:12.38 | kaldemar | rare1980_: by default they are 192.168.56.100-.XXX |
| 11:13.01 | rare1980_ | if possible can you jump on my vnc and plz check it for me... |
| 11:13.02 | rare1980_ | ? |
| 11:13.04 | defswork | i was thinking it might be a codec issue |
| 11:13.07 | kaldemar | rare1980_: .1 is the host system address, not the guest. |
| 11:13.18 | kaldemar | rare1980_: i'd rather not. |
| 11:13.49 | rare1980_ | kaldemar: can you come to my vnc? plz |
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| 11:40.58 | hackeron | hey, I'm trying to get dahdi working, and I see this when I ring the pbx number from my mobile: http://pastie.org/855368 -- but I don't hear the welcome recording, I just keep hearing the ringing tones on my mobile - any ideas? |
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| 11:57.40 | adnc | hello, in a context i do have several dialing rules which all have there own priority with 1. is it possible to have an expression (like Set() ) to be used for all in context or do i need to make it for every single dialingrule? |
| 11:59.33 | *** join/#asterisk Ender3 (~Ender3@84.123.83.220.dyn.user.ono.com) |
| 12:02.24 | kaldemar | adnc: with patterns it is. make a pattern extension that matches all the others with priority 1 and does the Set. then set all other extensions as priority n or 2. |
| 12:02.38 | *** join/#asterisk garymc (~chatzilla@host81-148-109-86.in-addr.btopenworld.com) |
| 12:04.23 | adnc | kaldemar, thanks, but i did not get that really, could you give me a simple example? |
| 12:04.43 | *** join/#asterisk kruemeltee (~Maddin@port-92-198-62-82.static.qsc.de) |
| 12:04.47 | kruemeltee | hello |
| 12:10.28 | adnc | kaldemar, could i do this outside of a context? or does it need to be in a context (the pattern that matches all) |
| 12:11.49 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
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| 12:15.23 | kaldemar | adnc: all extensions are in a context. you can do it in another context and goto to the one with the actual extensions. |
| 12:17.10 | LemensTS | . |
| 12:17.16 | *** join/#asterisk Systemt` (~lol@89-139-110-13.bb.netvision.net.il) |
| 12:31.19 | *** part/#asterisk abcmob (~abcmob@unaffiliated/ccbbaa) |
| 12:40.39 | adnc | is it possible to switch off the .ael way of configuring, it looks for a extensions.ael which i do not have. which module is doing this? |
| 12:41.09 | *** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 12:42.25 | kaldemar | adnc: put noload => pbx_ael.so into asterisk's modules.conf |
| 12:42.30 | adnc | ahh thanks |
| 12:43.00 | adnc | res_odbc.c: Adding ENV var: INFORMIXSERVER=my_special_database |
| 12:43.15 | adnc | i have this kind of entries in my messages, although i do not use an informixserver |
| 12:43.28 | adnc | is the module called res_odbc.so? |
| 12:44.04 | kaldemar | yes |
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| 12:59.39 | LemensTS | . |
| 13:06.24 | Systemt` | some one can help me with my sip ? |
| 13:06.29 | Systemt` | i have audio problem... |
| 13:06.36 | Systemt` | NO audion |
| 13:06.38 | Systemt` | NO audio |
| 13:06.48 | *** join/#asterisk xpot (~xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
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| 13:34.52 | elred_ | Systemt`, check the SDP to see if ip match same subnet and if rtp port are reachable |
| 13:35.33 | *** join/#asterisk az (~az@carrot.znaider.de) |
| 13:35.39 | Systemt` | how can i check that ? |
| 13:35.46 | Systemt` | i have port forwarding .. |
| 13:35.54 | Systemt` | on |
| 13:36.05 | Systemt` | 5060~5070 |
| 13:36.09 | Systemt` | 10000~20000 |
| 13:36.13 | kaldemar | Systemt`: did you configure your asterisk to work with NAT? |
| 13:36.13 | Systemt` | 4569 |
| 13:36.17 | kaldemar | ~sipnat |
| 13:36.18 | 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 |
| 13:36.21 | Systemt` | yea |
| 13:36.21 | Systemt` | s |
| 13:36.29 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 13:36.42 | kaldemar | show how. pastebin your sip.conf and a sip debug of a call. |
| 13:36.44 | kaldemar | ~pb |
| 13:36.45 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 13:37.46 | kaldemar | if a provider connection is involved, mask passwords in sip.conf |
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| 13:41.45 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:43.11 | Systemt` | my sip is invite |
| 13:43.15 | Systemt` | with out password |
| 13:52.39 | Polysics | argh |
| 13:52.50 | Polysics | Zoiper does not send proper DTMF toens :-/ |
| 13:53.20 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
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| 13:54.32 | *** join/#asterisk CoolCat2012 (~IceChat7@189.100.217.254) |
| 13:54.35 | CoolCat2012 | hi all! |
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| 13:56.07 | CoolCat2012 | does anyone here use skype for asterisk? |
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| 14:10.04 | Polysics | anyone uses Adhearsion here? the channel isn't very lively |
| 14:10.17 | CoolCat2012 | i wonder if skype for asterik allow place call to skype users, and how. |
| 14:11.36 | adnc | http://pastebin.com/4gUN1JVN here i have the same behaviour for different pattern, is there a way to simplify this? |
| 14:14.12 | [TK]D-Fender | adnc: "core show application goto" |
| 14:14.39 | [TK]D-Fender | adnc: "core show application macro" |
| 14:14.43 | [TK]D-Fender | adnc: "core show application gosub" |
| 14:14.46 | adnc | i see |
| 14:14.47 | [TK]D-Fender | adnc: take your pick |
| 14:16.46 | [TK]D-Fender | adnc: I'd use a macro, and apss it the sound file to play as well to abstract it for multiple uses |
| 14:17.49 | adnc | [TK]D-Fender, i understand, if i would use goto, using goto is no problem but the playback how would i adress it exten => tobeexecuted,1,Playback(privacy-incorrect) |
| 14:18.18 | [TK]D-Fender | adnc: Just jump to the exten that does all the stuff you want to do |
| 14:18.37 | adnc | with which number |
| 14:18.47 | [TK]D-Fender | AndyGraybeal: MAKE ONE |
| 14:18.51 | [TK]D-Fender | adnc: rather |
| 14:19.06 | adnc | exten => 000000,1,Playback(...) |
| 14:19.52 | adnc | if it matches at exten => _0900.,1,Goto(jumppoint) how does the jumppoint look like? |
| 14:20.05 | adnc | there is no extension i can write |
| 14:20.10 | [TK]D-Fender | adnc: There is |
| 14:20.20 | [TK]D-Fender | adcTry treading the app's instructions |
| 14:21.11 | Katty | ohai |
| 14:21.21 | adnc | Katty, hi |
| 14:21.35 | Katty | what is happenings |
| 14:21.48 | Katty | on this gorgeous FRIDAY morning |
| 14:21.59 | [TK]D-Fender | Katty: U CAN HAZ GRAMMER?! |
| 14:22.07 | Katty | no |
| 14:22.16 | Katty | grammar would defeat the cuteness |
| 14:22.17 | [TK]D-Fender | Katty: .... Mew :) |
| 14:22.25 | Katty | [TK]D-Fender: mew. |
| 14:23.35 | *** part/#asterisk CoolCat2012 (~IceChat7@189.100.217.254) |
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| 14:28.44 | Naikrovek | grammAr |
| 14:28.49 | sip7 | Is 1.6.2.5 stable enough for a production environment? |
| 14:29.09 | Naikrovek | Katty: nice hello kitty controller you got there |
| 14:29.26 | Naikrovek | likes going through X-Chat's URL grabber in the morning. |
| 14:29.28 | Katty | Naikrovek: that's not mine. |
| 14:29.33 | Katty | Naikrovek: but i thought it was cute |
| 14:29.34 | Naikrovek | Katty: uh huh |
| 14:29.45 | [TK]D-Fender | [09:28]<Naikrovek>grammAr <- meme FAIL |
| 14:29.55 | Naikrovek | that's a meme? |
| 14:29.56 | Naikrovek | oh |
| 14:30.00 | Katty | your mom's a meme |
| 14:30.03 | Naikrovek | will have to look that up |
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| 14:30.25 | adnc | Katty, ;) |
| 14:30.55 | [TK]D-Fender | Naikrovek: .... Dear God my yogurt has more culture than you this morning...even if it's only BACTERIAL :p |
| 14:31.07 | Naikrovek | [TK]D-Fender: that's probably accurate. |
| 14:31.37 | coppice | [TK]D-Fender: if you want culture what are you doing on the internet? |
| 14:32.46 | Polysics | wow, i don't look at the window for 15 minutes and everything goes ape in here :-) |
| 14:40.19 | hackeron | hey, I'm trying to get dahdi working, and I see this when I ring the pbx number from my mobile: http://pastie.org/855588 and the call never goes to my dial plan :( - any ideas? |
| 14:43.02 | Katty | dances with file |
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| 14:43.10 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:43.16 | file | falls over |
| 14:43.28 | Katty | :< |
| 14:43.33 | Naikrovek | why would you fall over |
| 14:43.48 | leifmadsen | does anyone know if Asterisk will try to take dtmf from a phone (such as via rfc2833) and convert it to inband on an outgoing channel if the peer is setup for dtmfmode=inband? |
| 14:43.51 | leifmadsen | Asterisk 1.6.2.2 |
| 14:44.03 | Naikrovek | leifmadsen: one would think so, but i've not tested |
| 14:44.07 | file | I'm a cardboard cutout |
| 14:44.31 | Naikrovek | file: you're like a lot of people i work with. look at it face on, they appear to have all the skill and personality of a real person |
| 14:44.40 | Naikrovek | file: but when you ask them to do something, they fall flat on their face |
| 14:44.49 | Naikrovek | and are as thin as a playing card |
| 14:45.10 | Naikrovek | like a baloon painted like a rock or something |
| 14:45.12 | leifmadsen | Naikrovek: ya, I'm testing with someone, and I don't think I'm seeing that. Having DTMF issues where a phone can't handle inband (because they want to use g729) but the provider only supports inband |
| 14:45.25 | Naikrovek | you think you have something with substance, but when you need it, it floats away or pops or whatever |
| 14:45.29 | Naikrovek | useless |
| 14:45.58 | Naikrovek | provider only supports inband? the heck? |
| 14:46.19 | leifmadsen | Naikrovek: ya don't ask -- not my box :) |
| 14:46.34 | leifmadsen | and not my choice of provider |
| 14:46.43 | Naikrovek | well i would think taht asterisk would do that. if i have a dahdi card, and analog lines, but voip phones, it does the conversion to inband then |
| 14:46.54 | sip7 | Does G.722 support inband? |
| 14:46.54 | leifmadsen | Naikrovek: ya that would make sense eh |
| 14:47.15 | Naikrovek | sip7: they all "support" it, whether or not it actually works is another matter |
| 14:47.22 | Naikrovek | sip7: but g722 probably would be just fine |
| 14:47.26 | Naikrovek | since it's a wideband codec |
| 14:47.41 | sip7 | Ahhh.. |
| 14:49.49 | hackeron | anyone? - I'm trying to get dahdi working, and I see this when I ring the pbx number from my mobile: http://pastie.org/855588 and the call never goes to my dial plan :( - any ideas? |
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| 14:51.41 | Naikrovek | hackeron: someone will step up if they know the answer. don't leave or anything. |
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| 15:08.18 | Geminizer | Hello all. I have a dialplan context which makes use of Dial(). This is meant to access an extension I have created using asterisk. If that extension is not available, and VM is enabled for that extension, should the caller be automatically taken to that extension's VM ? |
| 15:09.38 | Geminizer | if the extension doesn't answer, that is |
| 15:13.53 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 15:15.34 | leifmadsen | Geminizer: if there is no answer due to a rejection or timeout, then execution of the dialplan continues to the next line |
| 15:15.49 | leifmadsen | Geminizer: at which point, it could do anything, such as utilize the Voicemail() application |
| 15:16.12 | leifmadsen | if the call is answered, execution of the dialplan stops at call completion, and the call is hung up |
| 15:18.50 | Geminizer | ok... so it's not a feature of the extension properties which dictates whether to go to that extensions voicemail, but rather the dialplan |
| 15:20.06 | leifmadsen | the dialplan is the end-all-be-all of call handling |
| 15:20.58 | [TK]D-Fender | Geminizer: Every action * takes for a call = diallpan |
| 15:21.25 | [TK]D-Fender | Geminizer: An extension is a number you dial. What it does is your job |
| 15:21.44 | Kobaz | leifmadsen: well... hehe... there's always AMI |
| 15:21.51 | [TK]D-Fender | Geminizer: And never call a "phone" an "extension". A phone is a device that can place a call, or be called |
| 15:21.59 | leifmadsen | Kobaz: lets not confuse the poor lad :) |
| 15:23.14 | Katty | yay, i beat dungeon 4! |
| 15:23.31 | Geminizer | I don't think I ever I established that equivalency |
| 15:24.06 | Katty | i don't recall where number 5 is |
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| 15:27.20 | Katty | did they make more than 1 zelda for NES? |
| 15:35.15 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
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| 15:41.09 | smooth_penguin | Katty, !! |
| 15:42.15 | Katty | smooth_penguin: :> |
| 15:42.21 | Katty | Naikrovek: i got my blue ring! |
| 15:42.24 | Katty | Naikrovek: :>>> |
| 15:42.27 | Katty | hugs smooth_penguin |
| 15:44.03 | hackeron | I get error: [Mar 5 15:43:19] ERROR[2755]: ais/clm.c:141 ast_ais_clm_load_module: Could not initialize cluster membership service: Try Again -- what is causing this? |
| 15:44.14 | smooth_penguin | hey Katty, whats cooking |
| 15:44.27 | Katty | nuffin |
| 15:44.34 | Katty | just sitting on hold with symantec and playin zelda |
| 15:44.37 | smooth_penguin | btw my weekend has started :P :P |
| 15:44.38 | *** join/#asterisk yoda1410 (~yoda@aether.hipocoon.be) |
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| 15:44.44 | Katty | :< |
| 15:44.47 | Katty | i guess mine started this morning |
| 15:44.50 | Katty | but i gots to work some |
| 15:44.54 | smooth_penguin | kk |
| 15:45.04 | Katty | if they will answer the phone |
| 15:45.46 | smooth_penguin | Katty, hows the critters |
| 15:46.42 | Katty | excellent :> their playpen came in |
| 15:46.54 | Katty | http://www.youtube.com/watch?v=jyQgIbEqc1U |
| 15:48.14 | hackeron | Anyone has an analog interface card and can help me? -- I set chan_dahdi.conf with context.default, default context in extensions.conf is set to answer the call and play a message, dahdi show status and dahdi show channels seems to show everything correctly (but I see a red alarm for some reason on asterisk start) - I see cid errors when I ring the PBX number but the call never makes it to the dial plan -- anyone at all??? |
| 15:52.17 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 15:55.24 | raden_work | morning Katty |
| 15:57.54 | *** join/#asterisk Akiraa (~Akiraaaa@79.112.13.42) |
| 15:57.59 | hackeron | I'm getting a red alarm on my analog interface card (wctdm24xxp+ d161:8006 Wildcard AEX410P) when I start asterisk, any ideas why? -- chan_dahdi.c:5691 handle_alarms: Detected alarm on channel 1: Red Alarm |
| 15:58.03 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:59.17 | Naikrovek | blue ring, nice |
| 16:01.08 | Katty | hi raden_work (= |
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| 16:05.52 | *** join/#asterisk wcselby (~wcselby@216.110.88.194) |
| 16:05.58 | wcselby | o/ |
| 16:06.38 | *** join/#asterisk nightrid3r (kvirc@41.214.156.133) |
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| 16:08.39 | wcselby | question - how does one setup g729 passthrough, and also t38 fax passthrough? I've setup g729 and t38, but never in passthrough mode. |
| 16:08.59 | *** join/#asterisk sourcode (~code@ppp-58-8-115-128.revip2.asianet.co.th) |
| 16:09.41 | ChannelZ | any codec will pass through so long as Asterisk doesn't need to be in the media stream and convert it |
| 16:09.48 | wcselby | always set it up with asterisk being inside the call loop with g729 (i.e not passthrough), and I've setup FFA with asterisk as the t38 endpoint |
| 16:09.58 | *** part/#asterisk lupine_85 (~lupine_85@unaffiliated/lupine-85/x-7392152) |
| 16:10.05 | wcselby | hmmm, so there's no special config? |
| 16:10.06 | ChannelZ | IE both ends of the call are g729, you aren't using MeetMe, any prompts * might play are g729.. |
| 16:10.23 | *** join/#asterisk nightrid3r (kvirc@41.214.156.133) |
| 16:11.31 | Katty | gets bored of zelda |
| 16:11.38 | Katty | so who can recommend me a new NES game |
| 16:12.05 | Akiraa | It turns out modem-over-IP is no joke as I thought... |
| 16:12.48 | wcselby | Katty - castelvania? |
| 16:12.53 | Akiraa | http://www.callcentermagazine.com/shared/article/showArticle.jhtml?articleId=8706907 |
| 16:13.12 | leifmadsen | Katty: RadRacer! |
| 16:13.25 | leifmadsen | Katty: even has a 3D version :) |
| 16:13.34 | Katty | k, i'll try out both of those |
| 16:13.56 | Katty | wcselby: number 1, 2, or 3? |
| 16:14.34 | wcselby | Katty - i've always enjoyed the original, but to be honest it's been a long time, so I don't remember the difference. |
| 16:14.41 | Katty | k |
| 16:14.57 | Katty | hmm |
| 16:15.01 | Katty | looks like that kung foo game |
| 16:15.08 | coppice | Akiraa: No? look at the date on that article, and look at the deployment level 8 years later |
| 16:15.15 | leifmadsen | Katty: Contra! |
| 16:15.18 | wcselby | anyone know where I can get a sidecar for a polycom 601 in Houston, Tx - today? |
| 16:15.31 | Katty | i gots contra |
| 16:15.36 | Katty | wcselby: hrmm |
| 16:15.45 | Katty | wcselby: not /today/ no |
| 16:15.47 | Naikrovek | man contra wasted many an afternoon in my childhood |
| 16:16.08 | Naikrovek | my father could beat contra and super mario both straight through without dying once |
| 16:16.43 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 16:16.47 | Naikrovek | no konami code, no warp pipes |
| 16:16.50 | Naikrovek | perfect games all |
| 16:17.02 | Naikrovek | "when you and your brohers can do this, i'll buy you a new game." |
| 16:18.02 | leifmadsen | lol |
| 16:19.06 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 16:19.35 | Katty | i don't get how to get through a wall :/ |
| 16:19.54 | Naikrovek | Katty: shoot the glowy red things all over it |
| 16:20.40 | Katty | shoot? |
| 16:20.44 | Katty | all i got is this..whip..thing |
| 16:20.49 | hackeron | what module is causing this error? < "ais/clm.c:141 ast_ais_clm_load_module: Could not initialize cluster membership service: Try Again" |
| 16:21.27 | Naikrovek | oh i thought you were playing contra |
| 16:21.29 | Naikrovek | what game? zelda? |
| 16:21.37 | Katty | castlevania |
| 16:21.40 | Naikrovek | oh |
| 16:21.46 | Naikrovek | memories... |
| 16:21.56 | wcselby | you use the whip to kill things, and swing I think? |
| 16:22.00 | Naikrovek | why can't there be more great 2d games |
| 16:22.01 | wcselby | can you catch a lamp or something katty? |
| 16:22.06 | Naikrovek | yes you can swing with it |
| 16:22.16 | Naikrovek | bionic commando was also one of my faves |
| 16:24.17 | *** join/#asterisk Deeewayne (~dwayne@75.76.254.162) |
| 16:24.17 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 16:25.11 | wcselby | hackeron - are you trying to run some kind of clustering? |
| 16:28.45 | *** join/#asterisk cguerrero (~cuauhtemo@200.79.231.94) |
| 16:30.25 | *** join/#asterisk cesar_CR (~cesar@201.192.86.30) |
| 16:33.51 | hackeron | wcselby: no |
| 16:34.40 | hackeron | wcselby: just a default asterisk configuration, I changed chan_dahdi.conf and extensions.conf -- dahdi show channels shows everything correctly but I'm not able to get incoming calls :( -- only error I can see is that |
| 16:37.05 | *** join/#asterisk nightrid3r (~kvirc@41.214.217.251) |
| 16:38.19 | hackeron | Anyone? - I'm having trouble accepting incoming calls, when I call the PBX, this is what I see: |
| 16:38.22 | hackeron | <PROTECTED> |
| 16:38.25 | hackeron | <PROTECTED> |
| 16:38.27 | hackeron | [Mar 5 16:37:32] NOTICE[1549]: chan_dahdi.c:8422 ss_thread: Got event 17 (Polarity Reversal)... |
| 16:38.31 | hackeron | [Mar 5 16:37:34] WARNING[1549]: chan_dahdi.c:8480 ss_thread: CID timed out waiting for ring. Exiting simple switch |
| 16:38.33 | Kobaz | stop |
| 16:38.34 | hackeron | <PROTECTED> |
| 16:38.36 | Kobaz | pastebin.ca |
| 16:38.37 | Kobaz | STOP |
| 16:38.39 | hackeron | and it just keeps showing this over and over while I hear a ringing sound on my phone :( |
| 16:38.43 | *** join/#asterisk ruben23 (~ITadmin@122.55.48.243) |
| 16:38.49 | *** join/#asterisk Systemt` (~lol@89-138-251-111.bb.netvision.net.il) |
| 16:38.58 | hackeron | Kobaz: it's 4 lines... |
| 16:39.04 | Kobaz | ~pastebin |
| 16:39.05 | infobot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 16:39.06 | Systemt` | Hello |
| 16:39.13 | Katty | HELLO THAR |
| 16:39.39 | Systemt` | Katty we spoke yesterday right ? |
| 16:40.07 | p3nguin_ | How can I get VoiceMail() to play the person's name but NOT play busy or unavailable messages. If I use ,u it says "Jan Sanders is unavailable. Please leave a message..." and ,b says "Jan Sanders is on the phone. Please leave..." I just want it to say "Jan Sanders. Please leave your message after..." Possible? |
| 16:40.11 | Kobaz | hackeron: turn off callerid, and retry |
| 16:40.13 | wcselby | sorry hackeron i was looking but I'm also doing some work |
| 16:40.32 | hackeron | Kobaz: I did, nothing :( |
| 16:40.34 | wcselby | p3nguin_, try the s switch? |
| 16:40.51 | Kobaz | p3nguin_: there's an option to turn off the auto built-in messages |
| 16:40.52 | hackeron | Kobaz: the warning disappears, but the call is still not answered |
| 16:40.59 | Kobaz | hackeron: what card |
| 16:41.09 | Katty | Systemt`: i don't really remember. |
| 16:41.12 | Katty | Systemt`: we may have tho |
| 16:41.13 | hackeron | Kobaz: pci:0000:04:08.0 wctdm24xxp+ d161:8006 Wildcard AEX410P |
| 16:41.15 | p3nguin_ | wcselby: Yeah, it just plays the beep with no instructions at all. |
| 16:41.19 | Katty | HAI PENGUIN |
| 16:41.30 | p3nguin_ | Hello, Katty. |
| 16:41.37 | Katty | well don't get all excited or anything |
| 16:41.43 | Kobaz | p3nguin_: write a macro that does a playback of the name stored in the mailbox |
| 16:42.11 | Systemt` | hey i have problem with audio when the person is answer the call ther is no audio |
| 16:42.17 | Kobaz | hackeron: pastebin your dahdi configs |
| 16:42.25 | Katty | Systemt`: i am sorry to hear that. |
| 16:42.38 | Systemt` | what cani do ? |
| 16:42.46 | Kobaz | Systemt`: rest, ice, and ibuprofin... you'll be tip top in the morning |
| 16:42.47 | wcselby | Systemt` - check your nat |
| 16:42.50 | Systemt` | i was making port forwarfind |
| 16:42.56 | Katty | Systemt`: well, firstly... |
| 16:43.01 | Katty | Systemt`: i would start by having lunch. |
| 16:43.11 | Systemt` | ? |
| 16:43.12 | Katty | Systemt`: and then i would google my issue and see what there is to see out there |
| 16:43.26 | wcselby | Systemt` - it's a nat issue |
| 16:43.27 | Katty | Systemt`: then, if googling failed me, i would pastebin all relevant information for the channel. |
| 16:43.28 | wcselby | ~sipnat |
| 16:43.30 | 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 |
| 16:43.41 | hackeron | Kobaz: one sec |
| 16:43.42 | *** join/#asterisk ccesario_ (~ccesario@189.88.3.2) |
| 16:43.43 | Systemt` | yea but i making port forwarding to this ports |
| 16:43.48 | Katty | Systemt`: then ask very specific questions. |
| 16:43.51 | Systemt` | 5060~5070 |
| 16:43.53 | wcselby | Systemt` - it's a nat issue |
| 16:43.58 | Systemt` | and |
| 16:44.05 | Systemt` | 10000~350000 |
| 16:44.08 | wcselby | Systemt` - read the links I just provided |
| 16:44.08 | Systemt` | 10000~35000 |
| 16:44.15 | Katty | NEXT!!! |
| 16:44.16 | wcselby | you're forwarding udp, right? |
| 16:44.24 | Katty | wcselby: let him read. it wlil do him good. |
| 16:44.26 | wcselby | it's a nat issue, check the links |
| 16:44.27 | Systemt` | tcp\udp |
| 16:44.33 | Systemt` | i was reding that |
| 16:45.19 | hackeron | Kobaz: http://pastie.org/855791 |
| 16:45.29 | hackeron | Kobaz: thanks for looking at this :) - much appreciated |
| 16:45.35 | wcselby | Systemt` - read it some more |
| 16:45.54 | wcselby | damnit |
| 16:45.57 | Katty | so i was thinking Mexican for lunch |
| 16:46.00 | wcselby | my headache is making me all bitchy |
| 16:46.03 | Katty | maybe some vegetarian nachos |
| 16:46.09 | Katty | wcselby: take midol. |
| 16:46.15 | Katty | wcselby: it works wonders for the crankies. |
| 16:46.17 | wcselby | i've taken it before |
| 16:46.23 | Katty | wcselby: pain killer, muscle relaxant, caffeine |
| 16:46.34 | wcselby | had someone give it to me at an old job, I was like, wtf is this? |
| 16:46.44 | Katty | it's awesome |
| 16:46.46 | Katty | that's what it is |
| 16:46.50 | Katty | they shouldn't just market it to women. |
| 16:46.55 | wcselby | laughed a bit, took it, didn't help one bit |
| 16:46.57 | Kobaz | hackeron: and what did you change when you tried without callerid |
| 16:47.06 | Katty | wcselby: ^_- |
| 16:47.16 | Katty | wcselby: i don't know what to say about that |
| 16:47.20 | Katty | wcselby: works wonders for me |
| 16:47.24 | hackeron | Kobaz: set usecallerid = no and commented out cidsignalling and cidstart |
| 16:47.41 | Kobaz | and paste your console with callerid disabled |
| 16:47.50 | Systemt` | wcselby --> pm plz |
| 16:48.08 | Katty | Systemt`: i think you two should talk here |
| 16:48.16 | Katty | Systemt`: that way the whole channel can benefit from it |
| 16:48.41 | wcselby | Systemt` - I can check your box, but that's not free. |
| 16:48.44 | hackeron | Kobaz: LeoPBX2*CLI> |
| 16:48.49 | hackeron | Kobaz: that's it, nothing appears when I call the line |
| 16:48.54 | Systemt` | how much u want? |
| 16:49.12 | wcselby | wow |
| 16:49.17 | hackeron | Kobaz: I only see the callerid warning and notice when callerid is enabled, otherwise, nothing :( |
| 16:49.17 | Systemt` | ? |
| 16:49.21 | wcselby | okay, i'll chat in the pm, one sec |
| 16:50.08 | Kobaz | hackeron: add answeronpolarityswitch=yes and keep callerid disabled... see what you get |
| 16:51.45 | hackeron | Kobaz: no change :( |
| 16:52.50 | Kobaz | hackeron: are you sure fxsks is the proper signalling for your line? |
| 16:52.56 | *** join/#asterisk afink (~chatzilla@204.26.87.226) |
| 16:53.11 | hackeron | Kobaz: not at all - but it's a standard BT line and that's what I used in the past |
| 16:53.31 | hackeron | Kobaz: will it make a difference to try loop start? |
| 16:53.31 | Kobaz | hackeron: have you tried a different card? |
| 16:53.40 | hackeron | Kobaz: nope, I don't have a different card :( |
| 16:53.47 | Kobaz | hackeron: give it a shot... you might have a different type of line then you've used before |
| 16:53.57 | Kobaz | and there's also ground start |
| 16:54.20 | *** join/#asterisk RobH (~robh@216.38.133.254) |
| 16:55.00 | hackeron | Kobaz: it says in the documentation that it's only used by some rare PBXs and not to worry about it generally? |
| 16:55.15 | *** join/#asterisk rgsteele (~rgsteele@207.106.239.81) |
| 16:56.18 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 16:56.21 | hackeron | Kobaz: and dahdi_monitor 2 -v shows RX spikes btw, so something's coming trhough |
| 16:56.31 | hackeron | Kobaz: no change with loop start - trying ground start |
| 16:58.17 | Systemt` | http://pastie.org/855806 <-- My Sip.conf |
| 16:59.20 | Katty | http://www.youtube.com/watch?v=mUCRZzhbHH0 |
| 16:59.23 | hackeron | Kobaz: hmm, I put fxsgs = 1-4 in /etc/dahdi/system.conf and signalling = fxs_gs -- but asterisk is saying: [Mar 5 16:59:13] ERROR[1475]: chan_dahdi.c:10050 mkintf: Signalling requested on channel 1 is FXS Groundstart but line is in FXS Kewlstart signalling |
| 16:59.24 | Katty | ^- just watch it. |
| 16:59.59 | *** join/#asterisk HorizonXP (~xitij@76-10-162-208.dsl.teksavvy.com) |
| 17:00.05 | wcselby | Systemt` - you're running freepbx |
| 17:00.08 | Systemt` | http://pastie.org/855808 |
| 17:00.13 | Systemt` | yes |
| 17:00.14 | wcselby | there's a support channel for that - #freepbx |
| 17:00.22 | Systemt` | this is my sip configuration |
| 17:00.43 | hackeron | Kobaz: any ideas? |
| 17:00.46 | Kobaz | hackeron: chan_dahdi and system/dahdi need to match |
| 17:00.54 | Systemt` | but i dont have freepbx problems |
| 17:00.59 | Systemt` | i have sip roblems |
| 17:01.14 | Systemt` | *problems |
| 17:01.16 | hackeron | Kobaz: Changing signalling on channel 1 from FXS Kewlstart to FXS Groundstart |
| 17:01.17 | hackeron | DAHDI_CHANCONFIG failed on channel 1: Invalid argument (22) |
| 17:01.17 | hackeron | Did you forget that FXS interfaces are configured with FXO signalling |
| 17:01.40 | hackeron | Kobaz: so dahdi refuses to set it to groundstart - loopstart and kewlstart work though |
| 17:01.42 | Kobaz | okay.. so your card doesn't support it |
| 17:02.01 | Kobaz | i would call up digium hardware support |
| 17:02.16 | hackeron | Kobaz: ok, I'll try that, thanks |
| 17:02.28 | wcselby | ~freepbx |
| 17:02.29 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 17:02.33 | wcselby | Systemt` ^^^^ |
| 17:02.45 | hackeron | Kobaz: I'm not next to the PBX at the moment though - so I can't give them the serial number - will they reduce to speak to me without it do you think? |
| 17:03.15 | hackeron | will they refuse I mean* |
| 17:03.19 | Kobaz | hackeron: i've never dealt with digium support, but it's their hardware, they should support it no matter when/where/who you bought it from |
| 17:03.31 | hackeron | Kobaz: ok, thanks, I'll try |
| 17:03.31 | Kobaz | if i call up sangoma, they just say... give us your logs |
| 17:03.32 | Systemt` | wcselby: i dont use him ... |
| 17:04.25 | hackeron | Kobaz: oh and last thing, so any ideas what's causing this? < ERROR[1528]: ais/clm.c:141 ast_ais_clm_load_module: Could not initialize cluster membership service: Try Again |
| 17:04.36 | Kobaz | hackeron: that looks bad... no idea |
| 17:04.59 | hackeron | Kobaz: it's a vanilla asterisk compiled manually with make samples :( |
| 17:05.32 | Kobaz | you could go and disable modules you're not using |
| 17:05.40 | Kobaz | delete out config files for stuff you dont need |
| 17:05.47 | wcselby | Systemt` - the point is, freepbx uses a whole lot of special files that are different from a standard asterisk install. there's sip.conf, and sip_custom.conf, and sip_general.conf, and sip_general_custom.conf, and sip.............etc. There's a whole lot of files, and their organized a special way by freepbx. you should find someone that does a lot of freepbx support to help you with your nat issues. |
| 17:06.23 | hackeron | Kobaz: yeah, that's my question, what module do I disable for that to go away? -- also, how do I get a list of loaded modules? |
| 17:06.29 | Kobaz | basically all you need for a base system is asterisk.conf, dahdi, sip/iax, cdr, and logging, indications, features.conf manager.conf modules.conf, voicemail.conf and that's about it |
| 17:06.47 | Kobaz | module show will show you everything |
| 17:08.08 | Kobaz | and musiconhold if you wan that too |
| 17:11.04 | hackeron | Kobaz: thanks |
| 17:11.46 | Kobaz | you can look at your initial console output for seeing what modules are loaded |
| 17:12.15 | p3nguin_ | How the hell am I supposed to troubleshoot a T1 channel bank? Doesn't it either WORK or NOT WORK? If in condition of NOT WORK, you hire someone to repair it? |
| 17:12.28 | Kobaz | p3nguin_: you might have signalling wrong |
| 17:12.53 | Kobaz | and if it doesn't work... that's probably what it is |
| 17:12.57 | p3nguin_ | I'm going to make the assumption that it has been working and now does not work. |
| 17:13.14 | Kobaz | are you sure *nothing* has changed |
| 17:13.33 | p3nguin_ | Nope. I didn't touch it... that's the only guarantee I can make. |
| 17:13.40 | Kobaz | what if someone else did |
| 17:13.56 | p3nguin_ | It's possible. Maybe there is a loose cable or something. |
| 17:14.27 | *** join/#asterisk wcselby (~wcselby@216.110.88.194) |
| 17:14.28 | wcselby | o/ |
| 17:14.46 | Kobaz | order of operations... reboot... check hardware (cabling/power/internal modules), configuration, reboot some more... call support.... rma |
| 17:14.52 | p3nguin_ | Here's the message I got: Network is currently down. T-1 splits into two networks in a channel bank, one channel bank is down. Go on site to trouble shoot. |
| 17:15.24 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.165.106.dsl.dyn.forthnet.gr) |
| 17:16.01 | Kobaz | at least it wasn't |
| 17:16.07 | Kobaz | "office down, pls fix... tks" |
| 17:18.08 | p3nguin_ | Might as well have been that general. |
| 17:18.43 | wcselby | at least you didn't get "hey dude, the website's down, can you reboot it?" |
| 17:18.52 | p3nguin_ | hahahahaha |
| 17:18.59 | wcselby | ala webdude vs. salesguy |
| 17:19.30 | p3nguin_ | I would have replied with the "reboot your computer three times" instruction. |
| 17:20.49 | wcselby | "dude I already did that, three times like you always say to" |
| 17:21.45 | Kobaz | i remember reading an it horror story... some dept head was like... reboot the webserver, i cant log in |
| 17:21.59 | Kobaz | and the guy was like, the daemon was crashed out, i restarted it... it works now |
| 17:22.16 | Kobaz | and the dept head was still saying, reboot the server... but there were other services on the server... and active connections |
| 17:22.41 | Kobaz | and he kept getitng pestered and pestered... so he's like okay... i rebooted it... and then the dept head was like... my email is down!!!! |
| 17:22.54 | wcselby | Kobaz - http://www.thewebsiteisdown.com/ |
| 17:23.10 | Kobaz | heh |
| 17:23.20 | p3nguin_ | People are funny. |
| 17:26.35 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 17:28.10 | Kobaz | haha |
| 17:28.14 | Kobaz | the one at the bottom os great |
| 17:29.25 | Kobaz | oh wait no... i'm watching the second one.. |
| 17:30.39 | *** join/#asterisk Sidnicious (~Sidney@pdpc/supporter/professional/sidney) |
| 17:30.41 | p3nguin_ | Here's another funny story. "Go to building X and install this router." "How far away is the demarc?" "Not sure, better get some cable." So I call someone for a box of bulk cable. $70 for the cable, $127 to overnight it to me so I'll have it on time. |
| 17:31.01 | p3nguin_ | NO THANKS! |
| 17:33.54 | Sidnicious | So, my company's moving into a space with an old intercom system. Push to talk, push to listen, push to open door. A tone comes in when our button is pressed downstairs. Is anyone familiar with a way to adapt that to our phone system? |
| 17:36.32 | Sidnicious | I'm envisioning some piece of hardware that generates a call when it sees voltage on the line, uses VOX to switch between talk and listen, and closes the door switch for a few seconds when a key is hit |
| 17:37.26 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 17:37.31 | Kobaz | Sidnicious: there are sip activated door plates |
| 17:37.42 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 17:38.53 | ariel_ | Hello folks |
| 17:39.05 | Sidnicious | Kobaz: That'd be nice, but we only control what's inside our space. |
| 17:39.44 | Kobaz | mmm |
| 17:40.41 | Sidnicious | OK, failing the existence of this thing, are there any barebones, hackable sip boxes that I could use as a starting point to build one? |
| 17:40.54 | Kobaz | buy a cheap sip phone and just wire it up to the intercom system |
| 17:40.59 | Kobaz | that's what we did |
| 17:41.14 | Kobaz | we took apart a polycom, set it to autoanswer, soldered on the paging system leads to the speakerphone leads |
| 17:43.02 | Sidnicious | Do the push-to-talk and push-to-listen buttons not actually control anything in the rest of the intercom system? |
| 17:43.09 | Sidnicious | That's pretty awesome. |
| 17:43.52 | Kobaz | well the intercom stuff is only used over the ip system now |
| 17:44.03 | Sidnicious | i.e. does it work properly without them? |
| 17:45.42 | *** join/#asterisk upb (cmpxchg@88.80.13.92) |
| 17:52.25 | *** join/#asterisk dymaxion (~dymaxion_@host217-40-240-249.in-addr.btopenworld.com) |
| 17:53.18 | paulc | Definition of pain: co workers on a conference call, over the cubicle wall, speakerphone LOUD, trying to understand what a 404 is.. "uh, guys - they ARE hitting our server, but the path is wrong/not found" ARRGGGHHHH |
| 17:53.28 | paulc | dreams of a job doing Asterisk all day |
| 17:53.33 | paulc | TGIF! |
| 17:54.26 | Naikrovek | yes polycoms can intercom without them |
| 17:54.37 | Naikrovek | i have a 3 digit extension that pages (full duplex) all phones in the facility |
| 17:54.44 | Naikrovek | so i can hear people respond |
| 17:54.56 | *** part/#asterisk mykhyggz (~col@evolone.org) |
| 17:55.02 | Naikrovek | and polycom or asterisk is smart enough to mute phones where people aren't talking |
| 17:55.25 | Naikrovek | as soon as someone talks though i hear them, and them only |
| 17:55.40 | Sidnicious | cool, cool |
| 17:56.12 | Naikrovek | but kobaz is doing something kinda cool there |
| 17:56.15 | *** join/#asterisk mykhyggz (~col@evolone.org) |
| 17:56.19 | Naikrovek | he's using a phone as the interface to his paging system |
| 17:56.22 | Naikrovek | which is neat |
| 17:56.53 | Naikrovek | i woudln't have thought that would have worked by just straight soldering the speaker leads to the paging system input |
| 17:58.21 | Naikrovek | hrm. |
| 17:58.33 | wcselby | oh dear lord |
| 17:58.34 | Naikrovek | these layer3 hp gigabit poe switches are interesting |
| 17:58.39 | Naikrovek | soooo much cheaper than a cisco |
| 17:58.49 | wcselby | i rickrolled someone using asterisk, and now they're doing it to everyone on our helpdesk |
| 17:58.56 | Naikrovek | oh man |
| 17:59.18 | wcselby | using this link - http://unf.net/2009/12/asterisk-rick-roll.php |
| 17:59.21 | wcselby | dialing the phone number |
| 17:59.31 | wcselby | then turning on call forward on my phone |
| 18:00.10 | Naikrovek | finally watched district 9 last night |
| 18:00.13 | Naikrovek | wow |
| 18:00.53 | [TK]D-Fender | Naikrovek: It was kinda touching... nothing amazing, just a good story that feels like it matters at least a little |
| 18:01.01 | Naikrovek | yeah |
| 18:01.15 | Naikrovek | well it was highly reflective for the south african folks i know |
| 18:01.15 | *** join/#asterisk jermudgeon (~jermudgeo@15-138-42-72.gci.net) |
| 18:01.40 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 18:01.47 | *** part/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 18:01.58 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 18:02.14 | *** part/#asterisk jermudgeon (~jermudgeo@15-138-42-72.gci.net) |
| 18:02.16 | Katty | sprawls |
| 18:02.28 | Katty | uggggg. too much nachos |
| 18:02.51 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 18:03.26 | Naikrovek | i've been watching calorie intake starting this week |
| 18:03.30 | Naikrovek | not going over 1200/day |
| 18:03.35 | Naikrovek | it's been remarkably rewarding |
| 18:03.47 | Naikrovek | a bit hungry all the time but nothing a glass of water doesn't fix |
| 18:03.54 | paulc | Wow.. isn't that a bit low? |
| 18:04.03 | Naikrovek | if you're trying to lose 100lbs it's probably fine |
| 18:04.07 | Naikrovek | doctor says it's fine |
| 18:04.08 | Katty | that's not low |
| 18:04.11 | Katty | that's about what i eat |
| 18:04.11 | Naikrovek | as long as i get nutrition |
| 18:04.28 | Katty | 300-400 per meal, and 100-150 for snacks, twice a day |
| 18:05.35 | ariel_ | hummm but what about my once or twice a week 1200 cal Ice Cream from ColdStone???? |
| 18:05.39 | lanning | I need to get that control... |
| 18:06.00 | Katty | ariel_: you can still have it, just eat less of it |
| 18:06.57 | ariel_ | I bbl I just ate lunch and need to walk. (I promised my wife I would walk at least 30 minutes after lunch)..... |
| 18:08.07 | *** join/#asterisk jermudgeon (~jermudgeo@15-138-42-72.gci.net) |
| 18:08.15 | *** part/#asterisk jermudgeon (~jermudgeo@15-138-42-72.gci.net) |
| 18:08.35 | ruben23 | hi |
| 18:08.49 | *** join/#asterisk saghul (~saghul@ip3e830637.speed.planet.nl) |
| 18:08.57 | Naikrovek | lanning: as soon as you start, self control is not a problem |
| 18:09.27 | lanning | I need to loose about 200... :( |
| 18:09.28 | Naikrovek | lanning: that was my concern as well; but once i did it one time i gained self control, just like i lost self control every time i ate something i knew i shouldn't |
| 18:10.07 | Naikrovek | every time you make a decision taht you know is good for yourself, even though you want to do what's bad for you, you get a little piece of your soul back |
| 18:10.14 | Naikrovek | and that extra piece makes it easier next time |
| 18:10.21 | Naikrovek | it spirals, but it spirals up |
| 18:10.26 | Katty | so what was better on n64, zelda majora's mask or zelda ocarina of time |
| 18:10.27 | Naikrovek | instead of down |
| 18:10.40 | Naikrovek | Katty: i like OoT but many love MM |
| 18:10.45 | Katty | k |
| 18:10.58 | Naikrovek | OoT is widely regarded (by some) as the best game ever |
| 18:11.04 | Katty | hmm. |
| 18:11.05 | lanning | Katty: I was addicted to Mario Cart! :P |
| 18:11.09 | Katty | meh. |
| 18:11.13 | Katty | lanning: i'm more of an RPG person |
| 18:11.27 | Katty | tho mario RPG legend of the seven starts on snes was awesome. |
| 18:11.38 | Katty | really super awesome. |
| 18:11.41 | lanning | Rocket Propelled Grenade? :P |
| 18:11.42 | *** join/#asterisk clintc (~clintc@n128-227-15-193.xlate.ufl.edu) |
| 18:11.54 | Katty | don't remember anything about a rocket propelled grenade |
| 18:11.55 | Naikrovek | Katty: the Mario and Luigi: Bowser's Inside Story for NDS is 10/10 in my book |
| 18:12.07 | Katty | makes note to download NDS emulator |
| 18:12.08 | Katty | oh |
| 18:12.17 | Katty | do they make emulators for that? |
| 18:12.26 | Katty | seems like they're still selling Nintendo DS at the store |
| 18:12.57 | Naikrovek | yes there are emulators |
| 18:13.03 | Naikrovek | but i don't think piracy talk is condoned in here |
| 18:13.04 | Katty | k |
| 18:13.15 | Naikrovek | not judging though |
| 18:13.17 | Naikrovek | just saying |
| 18:13.20 | Katty | well i would never do anything illegal. |
| 18:13.21 | Katty | (= |
| 18:13.34 | Katty | naturally. |
| 18:13.53 | Naikrovek | you know |
| 18:14.11 | Naikrovek | my daughter (5yo) is absoutely in love with LittleBigPlanet on PS3 |
| 18:14.16 | *** join/#asterisk hfb (~hfb@pool-96-247-114-78.lsanca.dsl-w.verizon.net) |
| 18:14.27 | Katty | i've been considering getting a ps3 |
| 18:14.30 | Naikrovek | I've never heard her laugh so hard as when she lays down some cloth, tacks a rocket on it, then grabs it |
| 18:14.40 | Katty | but it seems that the majority of games released for xbox, ps3, etc are first person shooters. |
| 18:14.43 | Naikrovek | rocket takes off with her attached and spirals out of control |
| 18:14.49 | Naikrovek | her laughing hysterically for 10 minutes |
| 18:14.54 | Naikrovek | Katty: |
| 18:14.55 | Katty | cute |
| 18:15.06 | Naikrovek | final fantasy XIII comes out tuesday, for ps3 and x360 |
| 18:15.13 | Naikrovek | and ffxiii is why i bought my ps3 |
| 18:15.17 | Katty | i haven't played any of the ff games |
| 18:15.22 | Naikrovek | oh they are awesome |
| 18:15.30 | Katty | what did Final Fantasy I come out on? |
| 18:15.35 | Naikrovek | awe...SOME! |
| 18:15.37 | Naikrovek | NES I think |
| 18:16.12 | Naikrovek | final fantasy 12 was pretty close to 10/10 for me |
| 18:16.14 | Katty | hmm |
| 18:16.18 | Naikrovek | ff 13 is HD |
| 18:16.47 | Naikrovek | one can pick up a ps2 from craigslist for $40 often, and ff 12 used from gamestop for $12 |
| 18:16.52 | wcselby | how can I disconnect a dahdi channel in progress? |
| 18:16.54 | wcselby | from the cli |
| 18:16.59 | wcselby | using asterisk 1.4.x |
| 18:17.32 | Katty | wonders how to map her controller to n64 |
| 18:17.58 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 18:18.11 | Naikrovek | wcselby: that may be an AMI only kind of operation, disconnecting |
| 18:18.14 | Naikrovek | i'm not sure though |
| 18:18.17 | Naikrovek | digging through CLI now |
| 18:19.33 | wcselby | dahdi destroy channel xx-x |
| 18:19.37 | wcselby | got it |
| 18:19.44 | Naikrovek | nice |
| 18:22.38 | [TK]D-Fender | DON'T |
| 18:22.49 | [TK]D-Fender | Naikrovek: that will KILL the entire channel, nt jsut a call |
| 18:22.52 | [TK]D-Fender | not* |
| 18:23.13 | [TK]D-Fender | Naikrovek: And then you'll start failing things like group dials, etc |
| 18:23.21 | Naikrovek | it wasn't me asking |
| 18:23.24 | Naikrovek | but thank you |
| 18:23.32 | Naikrovek | wcselby: read what [TK]D-Fender said |
| 18:25.13 | wcselby | [TK]D-Fender - i did dahdi destroy call 74-4 |
| 18:25.23 | wcselby | erm |
| 18:25.28 | wcselby | dahdi destroy channel 74-4 |
| 18:25.49 | wcselby | i see what you're saying |
| 18:25.54 | wcselby | so now I'm short one channel |
| 18:25.59 | wcselby | gotcha |
| 18:26.06 | Katty | the fate of the world DEPENDS ON THEE |
| 18:26.40 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 18:33.17 | *** join/#asterisk trentcreek (~kvirc@129.113.44.94) |
| 18:33.52 | ruben23 | are there any chance on asterisk i can capture the inbound number coming to my asterisk. |
| 18:34.27 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 18:34.29 | trentcreek | How are we suppose to identify the new scam from the Ma Bells, and block asterisk from calling one of those "Pay-per-Call" numbers? |
| 18:38.35 | bmoraca_work | ruben23: if you have digital trunking or DID lines from the telco, they will send DNIS information |
| 18:40.22 | *** join/#asterisk TheNoOne (~thenoone@91.114.16.34) |
| 18:40.58 | TheNoOne | hi .. i'm trying to get a beronet BN8S0 running under dahdi |
| 18:41.14 | TheNoOne | so far everthing loaded |
| 18:41.21 | TheNoOne | but all ports come up as TE |
| 18:41.30 | ruben23 | bmoraca_work:im having voip trunk |
| 18:41.43 | TheNoOne | some are jumpered as NT |
| 18:41.56 | TheNoOne | is there a module parameter |
| 18:42.10 | TheNoOne | to tell wcb4xxp which ports are NT |
| 18:45.18 | *** join/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 18:46.10 | TheNoOne | my beronet is rev.1 .. so there are only 4 TE NT jumpers not 5 |
| 18:47.43 | TheNoOne | the card is like a junghanns.net octoBRI rev.1 |
| 18:48.00 | *** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri) |
| 18:48.28 | *** part/#asterisk Sidnicious (~Sidney@pdpc/supporter/professional/sidney) |
| 18:50.48 | *** join/#asterisk Failrar (~Failrar@5ED66E6D.cable.ziggo.nl) |
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| 18:55.54 | *** join/#asterisk drako (~luisjose@nelug/coreteam/luisjose) |
| 18:58.39 | ruben23 | hi |
| 19:07.05 | *** join/#asterisk voipmonk (~shido6@adsl-696402-76.lou.bluegrass.net) |
| 19:08.21 | RobH | ruben23: are you not getting the callerid info from your termination/origination provider? |
| 19:09.24 | RobH | ruben23: in your dialplan, if you are not sure, you can toss in something like: exten => s,n,NoOp(THE CALLERID FOR THIS CALL IS: ${CALLERID(all)}) |
| 19:09.35 | RobH | and it will output it to the CLI when folks call. |
| 19:10.44 | RobH | all incoming calls to asterisk will store to the callerid variable if its passed on by the provider. (even if its not, then its just a blank variable afaik, but i always get UNKNOWN from my provider when its blocked or not available. |
| 19:11.58 | *** join/#asterisk korihor (~korihor@201.210.226.98) |
| 19:12.45 | TheNoOne | where can i find a list of wcb4xxp module parameters? |
| 19:15.57 | TheNoOne | ok found it in base.c |
| 19:16.18 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 19:16.37 | TheNoOne | hm ... doesnt seem to be support for manually setting NT modes |
| 19:16.46 | TheNoOne | this is bad :( |
| 19:17.56 | *** join/#asterisk Talirk81 (~tt@rrcs-67-78-39-22.sw.biz.rr.com) |
| 19:19.12 | Talirk81 | I added a "friend" into my sip.conf , but when i try to dial out it fails. Registering works fine, how can i trouble shoot why its failing to dial out. Dial() from my agi scripts works fine from inbound did contexts. |
| 19:19.21 | Talirk81 | btw running 1.4 |
| 19:20.41 | *** join/#asterisk etfonhomey (~etfonhome@74-143-192-74.static.insightbb.com) |
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| 19:21.29 | *** join/#asterisk sulex (~sulex@host-78-14-173-189.cust-adsl.tiscali.it) |
| 19:22.39 | ruben23 | <PROTECTED> |
| 19:22.55 | *** join/#asterisk corretico (~laguilar@201.201.46.106) |
| 19:23.17 | ruben23 | i got the message to my gtalk is calling you NOW but the callerID is not been send. |
| 19:23.17 | trentcreek | Talirk81: That is not enough, you should always pastebin your config in so people can see what you are doing |
| 19:23.44 | Talirk81 | well i dont know what part of the many files you need , ive not tried to use a softphoen before |
| 19:24.29 | *** join/#asterisk Tim_Toady (~moi@194.219.131.122) |
| 19:25.07 | trentcreek | Talirk81: You just wrote: " I added a "friend" into my sip.conf " |
| 19:25.25 | Talirk81 | right and that registers fine but it cant dial out |
| 19:25.37 | Talirk81 | so i dont know if you need excepts from extensions or other sip files |
| 19:25.49 | *** join/#asterisk correcaminos (~luis.agui@201.199.12.190) |
| 19:25.51 | trentcreek | how about everything you did to make it |
| 19:25.55 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 19:27.05 | Talirk81 | how about this then, what is required to allow a softphone to dial out, once its registered, do you need to do something to point that trunk to dial out to another trunk from your sipprovider. |
| 19:27.29 | RobH | ahh, i have not messed with gtalk on asterisk sorr =[ |
| 19:28.44 | *** join/#asterisk friartuck (~pmccary@66.162.90.56) |
| 19:28.50 | trentcreek | Talirk81: A proper setup in extensions.conf |
| 19:30.13 | RobH | Talirk81: you need to have some pattern match in the dialplan, with the dial application being called. |
| 19:30.56 | RobH | hrmm, the book has excellent section walking you through that |
| 19:30.57 | RobH | ~book |
| 19:30.58 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 19:30.59 | RobH | ? |
| 19:31.00 | *** join/#asterisk ChrisWi (~admin@mx2.wwserver.net) |
| 19:31.02 | RobH | there we go |
| 19:31.08 | Talirk81 | RobH, wouldnt this be the same pattern that my normal Dial() commands from AGI use forexampe Dial(SIP/trunkname/number) |
| 19:31.18 | RobH | yep |
| 19:31.27 | Talirk81 | because those work perfectly but the softphone fails to dial |
| 19:31.28 | RobH | you need to have your sip phone having access to the context |
| 19:31.53 | RobH | makesure whatever context your sip phone is in either directly hits that context, or hits it with include => dial_context_name |
| 19:31.56 | trentcreek | Thus why we need to see how oyu set it up.. Show us or read the book |
| 19:32.09 | Katty | ohai |
| 19:32.12 | RobH | Talirk81: we need to see your sip.conf and your extentensions.conf |
| 19:32.20 | RobH | but wipe the secret data before pastebinning it. |
| 19:32.21 | Katty | i got bored with ocanara of time |
| 19:32.43 | Talirk81 | trentcreek: showing you my sip.conf wouldnt have helped you as you said, olny my extensions.conf would have which is exactly why i asked would you would need. |
| 19:33.10 | trentcreek | Talirk81: Then go read gthe book, see now e got more people wanting to see what you go in thtem |
| 19:33.33 | RobH | your sip conf shows the context of your sip phone |
| 19:33.36 | RobH | so we really need both. |
| 19:33.43 | friartuck | slightly off topic but may I ask for input on hosted conference call service? Anyone have good luck with one? We use Intercall and they drop our calls a lot. |
| 19:33.50 | RobH | plus just wipe the secret= line and its safe to show. |
| 19:33.51 | RobH | oh |
| 19:33.57 | RobH | wipe any registration => lines too. |
| 19:34.03 | RobH | or just clear out the password data. |
| 19:34.16 | *** join/#asterisk ruchir (~ruchir@67.41.130.152) |
| 19:34.21 | ruchir | hi all |
| 19:34.32 | RobH | friartuck: I used both teliax and vitality, and disliked both enough to just use t1s and pots. |
| 19:34.34 | ruchir | is it possible to add prefix to voicemail files |
| 19:34.40 | trentcreek | friartuck: That is WAY off topic. Why not set your own up on asterisk? |
| 19:34.45 | RobH | teliax is super cheap though |
| 19:34.53 | RobH | but i wouldnt ever use it for a corporate phone service again |
| 19:34.58 | RobH | but i use it for my personal development use. |
| 19:35.14 | friartuck | trentcreek because we host large conferences and don't have the lines |
| 19:35.21 | ruchir | i need to sync 2 voicemails on 2 asterisk servers running as load balancers so need to make sure files are not overwritten |
| 19:35.35 | trentcreek | friartuck: you dont need "lines" using VOIP |
| 19:36.11 | russellb | ruchir: why not use a database to store voicemail, instead? |
| 19:36.36 | friartuck | Robh thx. We are considering just that |
| 19:37.19 | ruchir | russellb: to avoid overhead and complexity |
| 19:37.39 | ruchir | as db will be shared and it'll introduce unnecessary network overhead |
| 19:37.44 | trentcreek | friartuck: Setup a number people can call and have all the callers you want |
| 19:38.13 | russellb | as you wish, but that's the solution for sharing voicemail between servers like that. |
| 19:38.24 | russellb | unless you want to use NFS or something |
| 19:39.00 | trentcreek | How are we suppose to identify the new scam from the Ma Bells, and block asterisk from calling one of those "Pay-per-Call" numbers? |
| 19:39.04 | RobH | i would do the database. |
| 19:39.07 | RobH | its better supported. |
| 19:39.10 | RobH | for voicemail storage. |
| 19:40.38 | friartuck | trentcreek sorry, I'm not following. How can I have all the callers I want with one phone number? Given that people will from mobiles, home, and different places? Seems like we need a bunch of lines/PRI/channels...what am I missing? |
| 19:41.23 | RobH | if you are using an online origination/termination |
| 19:41.33 | RobH | any new calls simply will generate another channel to teliax in this example. |
| 19:41.52 | RobH | (unless it restricts you to a # of concurrent channels, which depends on your provider and your plan with the provider) |
| 19:42.00 | trentcreek | friartuck: Buy one DID with as many channels as you need |
| 19:42.04 | RobH | thats the nice part about third party origination/termination. |
| 19:42.14 | friartuck | <PROTECTED> |
| 19:42.24 | Talirk81 | http://pastebin.ca/1824127 |
| 19:42.26 | RobH | look specifically at the pay as you go plan with teliax. |
| 19:42.39 | *** join/#asterisk IBC_JKENNEY (~jkenney@ip65-44-169-66.z169-44-65.customer.algx.net) |
| 19:42.55 | RobH | (as an example, i cannot endorse teliax in any way or form, as they annoyed my business to the pont that I installed plain old telephone lines) |
| 19:42.55 | IBC_JKENNEY | I know this is not the hylafax channel but nobody is ever in there |
| 19:43.05 | *** join/#asterisk corretico (~laguilar@201.201.46.106) |
| 19:43.06 | trentcreek | friartuck: Or everyone use ATA devices, VOIP phones, or Softphones and have them log into the conference directrly |
| 19:43.12 | IBC_JKENNEY | does anyone here have experience with doing fax blasting |
| 19:43.23 | friartuck | RobH of course. thx again. Just looking for direction. |
| 19:43.30 | RobH | glad to give it =] |
| 19:44.32 | friartuck | trentcreek that's the issue. I can't require everyone to be on VOIP/softphones |
| 19:45.01 | trentcreek | friartuck: then get a DID with as many channels as you need. |
| 19:45.59 | Katty | wow. |
| 19:46.05 | Katty | there's a game called Barbie Super Model for snes. |
| 19:46.10 | friartuck | trentcreek that is an option we are considering. I think with that we would simply have less head-aches, but maybe some more expenses. They may swing for it though. We've had a lot of issues with conference providers |
| 19:46.50 | Talirk81 | RobH: I also added "outbound-allroutes" to my include for test but that didnt help at all since the pasebin of http://pastebin.ca/1824127 |
| 19:47.10 | RobH | sorry, staff meeting resuming, i rather be tinkering in asterisk and in here =P |
| 19:47.25 | RobH | asterisk = fun, meetings = the opposite of fun. |
| 19:48.05 | *** join/#asterisk lesouvage (~lesouvage@82.73.69.76) |
| 19:55.22 | *** join/#asterisk xpot-mobile (~xpot@66.60.101.91) |
| 19:56.45 | paulc | agrees with RobH - I'm in the same boat mate |
| 19:58.46 | *** join/#asterisk correcaminos (~luis.agui@201.199.12.190) |
| 20:05.22 | *** join/#asterisk Z_God (~julius@2001:888:141f:0:221:5dff:fe2a:6806) |
| 20:05.45 | Z_God | is anyone here using asterisk with chan_jingle? |
| 20:08.00 | Z_God | with chan_gtalk I was able to differential calls to different addresses, but now I am having trouble with this, everything is coming in on the same context with s |
| 20:10.54 | *** join/#asterisk RobH (~robh@216.38.133.254) |
| 20:11.39 | *** join/#asterisk K3rN3L (~s@189.133.72.193) |
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| 20:19.32 | *** join/#asterisk hfb (~hfb@pool-96-247-114-78.lsanca.dsl-w.verizon.net) |
| 20:20.03 | Katty | Lamp draws near!!! |
| 20:21.54 | *** join/#asterisk TheDavidFactor (~chatzilla@nc-71-0-16-133.dhcp.embarqhsd.net) |
| 20:23.06 | *** join/#asterisk trentcreek (~kvirc@red1.cs.panam.edu) |
| 20:26.37 | *** join/#asterisk tkrn (~tkrn@216.196.213.32) |
| 20:27.57 | keith4 | is Polycom's "HD Voice" worth the price bump? |
| 20:28.08 | Naikrovek | i've heard it is |
| 20:29.43 | keith4 | any magic needed to make it work with asterisk? |
| 20:29.57 | Gugge | if you only use it to call the PSTN network, its not :) |
| 20:30.19 | keith4 | well, yah. obviously |
| 20:30.23 | Kobaz | yes, some voodoo is required |
| 20:31.09 | [TK]D-Fender | nope |
| 20:31.26 | Kobaz | voodoo, some black magic, and some pagan rituals |
| 20:31.32 | Kobaz | three chickens, and nine goats |
| 20:31.32 | [TK]D-Fender | And the HD phones have been reported to sound better even at G.711 than their non HD counterparts |
| 20:31.55 | keith4 | [TK]D-Fender: really now. interesting |
| 20:32.14 | keith4 | ah, it wants G.722 for "HD voice"? |
| 20:32.24 | [TK]D-Fender | keith4: Clearly |
| 20:35.56 | Qwell | keith4: It's recommended that you use 1.6 for G.722 transcoding support, but 1.4 can do it in passthrough. |
| 20:36.24 | Qwell | but, it should "Just Work" (you may need to fiddle with codecs settings, depending, but that's trivial) |
| 20:36.56 | Qwell | and yes, even in G.711, they sound amazing. *Far* higher quality speakers/mic |
| 20:37.19 | Chainsaw | [TK]D-Fender: I support that notion actually. I have mine on G.711 and the difference with say... a Cisco 7960 is amazing. |
| 20:38.10 | [TK]D-Fender | Chainsaw: Another to the "yeah, we mean it" category |
| 20:38.29 | keith4 | oh, hmm. might stick with G.711 then |
| 20:38.43 | Qwell | Why? O.o |
| 20:39.11 | Chainsaw | 99% of our calls go to Patton gateways which don't do better than G.711 |
| 20:39.23 | Chainsaw | So I have no plans to roll out the "HD" codec itself. |
| 20:39.25 | carrar | Holy Cow I love this web site |
| 20:39.26 | carrar | http://bacolicio.us/http://www.asterisk.org/downloads |
| 20:39.35 | keith4 | Qwell: system in question is 1.4 |
| 20:39.55 | Qwell | upgrade :p |
| 20:39.57 | keith4 | for now, anyway. hopefully upgrading in the summer? not sure |
| 20:39.59 | keith4 | not my call |
| 20:40.04 | carrar | Does that not want to make you use asterisk or what |
| 20:40.26 | Chainsaw | is on 1.6.2.5 |
| 20:42.31 | friartuck | chan_bacon |
| 20:42.56 | Qwell | No, it would definitely be func_bacon, with audiohooks |
| 20:43.10 | netpro25_ | Anyone successfully using g726 with a pap2 or spa9xx? |
| 20:43.35 | Qwell | The_Boy_Wonder: there's an idea for your next func_funkeffects. SIZZLE() |
| 20:43.36 | netpro25_ | I am getting beeps and poor audio quality |
| 20:43.47 | carrar | heh |
| 20:44.05 | *** join/#asterisk fofware (~chatzilla@190.30.113.46) |
| 20:49.29 | *** join/#asterisk aidinb (~Aidin@wsip-98-190-45-225.oc.oc.cox.net) |
| 20:49.38 | trentcreek | dear god, * just started rejecting ALL calls. "chan_sip.c: Sending fake auth rejection for user " |
| 20:50.02 | trentcreek | well ones coming from Google Voice |
| 20:51.14 | *** join/#asterisk andres833 (~andres833@190.144.139.78) |
| 20:53.00 | trentcreek | So how can that be changed? |
| 20:53.14 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:53.14 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 20:55.39 | [TK]D-Fender | trentcreek: maybe you should look for a blatant setting in sip.conf <- |
| 20:57.28 | netpro25_ | [TK]D-Fender, do you know of any issues with g726 and linksys devices? In 1.6.2 I am getting static when I call an extension using the g726 codec. |
| 20:58.24 | *** join/#asterisk rttrey (~trey@209.208.18.121) |
| 20:58.49 | *** join/#asterisk Warp4 (~robert.wo@firewall-a.buf.ny.i-evolve.net) |
| 20:59.34 | upb | hi. can anyone advise about the asterisk build system? Where should i add libraries to be linked and include paths for a res module? |
| 20:59.54 | [TK]D-Fender | netpro25_: Chich codecs DON'T give you "static"? |
| 21:00.43 | netpro25_ | well it sounds like static, but its just really poor quality, and then there are what sound like beeps every few seconds |
| 21:01.51 | *** join/#asterisk TheDavidFactor (~chatzilla@nc-71-0-16-133.dhcp.embarqhsd.net) |
| 21:03.06 | trentcreek | [TK]D-Fender: Could that be something like "call-limit"? |
| 21:04.28 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 21:08.03 | [TK]D-Fender | trentcreek: No. |
| 21:08.37 | trentcreek | [TK]D-Fender: I don't see anything else that could be stopping calls in SIP.CONF |
| 21:09.19 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 21:09.33 | [TK]D-Fender | trentcreek: Neither do I |
| 21:09.54 | trentcreek | [TK]D-Fender: Then it must be elsewhere in * |
| 21:10.06 | [TK]D-Fender | trentcreek: I never said that |
| 21:10.51 | *** join/#asterisk voipmonk (~shido6@m485e36d0.tmodns.net) |
| 21:11.31 | trentcreek | [TK]D-Fender: Well I don't see how this setup that has been used for a year, all of a sudden, starts rejecting Google Voice |
| 21:11.40 | [TK]D-Fender | trentcreek: and I see virtually nothing |
| 21:11.51 | Qwell | trentcreek: maybe Google changed something? |
| 21:11.58 | [TK]D-Fender | Qwell: nope |
| 21:12.08 | Qwell | then what did you change? |
| 21:14.06 | trentcreek | I changed nothing. Actually I send the Google Voice calls to a DID, , but for some reason the CID number goes back to google, i.e. 2125551234@66.54.140.46 |
| 21:14.55 | trentcreek | but that is IPKALL |
| 21:15.38 | trentcreek | Oh..I see why now... |
| 21:15.45 | trentcreek | I did make a change |
| 21:16.00 | [TK]D-Fender | "authreject" sure sounds like a "suspect" |
| 21:16.07 | *** join/#asterisk bakermd (~bakermd@38.104.0.102) |
| 21:16.27 | trentcreek | I hade my server offline for a month, and thought I switched my IPKALL number to another because they are suppose to cut you off after 30 days of non use |
| 21:17.27 | bakermd | I need a dialplan entry to post some data to a URL - what is the best option for this? I can only see a method of shelling out to the OS and executing commands there, and I want to keep everything contained within the dialplan. Thanks |
| 21:17.52 | Qwell | bakermd: func_curl |
| 21:18.00 | trentcreek | Yeah that was it, I thought I switched service |
| 21:18.05 | bakermd | Qwell: aah.. I was looking at applications |
| 21:18.45 | *** join/#asterisk Dibri (~gavit@190.98.33.38) |
| 21:22.05 | trentcreek | Thanks for the enlightenment |
| 21:24.14 | *** join/#asterisk voipmonk (~shido6@m085e36d0.tmodns.net) |
| 21:24.37 | *** join/#asterisk Katty (~Angela@mail.copi-rite.com) |
| 21:24.40 | Katty | ohaider |
| 21:24.59 | Katty | so. i bought a A101D, and an A101. |
| 21:25.07 | Katty | but the A101 box has a A102 in it ^_- |
| 21:25.26 | Katty | IT"S CREEPY I TELL YOU |
| 21:30.15 | russellb | Katty: that makes me sad |
| 21:32.20 | Katty | russellb: am i sorries. |
| 21:32.23 | Katty | russellb: would you like a hug? |
| 21:32.45 | russellb | no. |
| 21:32.48 | russellb | pouts some more |
| 21:32.55 | Katty | also! can i swap a a101d around with this a102 without many issues? |
| 21:32.57 | Katty | or recompiling |
| 21:32.57 | netpro25_ | Should RTP packet size on linksys devices still be set to .02, or has this been resolved. I am seeing a lot of old posts about it |
| 21:33.01 | Katty | or other ....assorted...things |
| 21:34.58 | Katty | i bet sangoma would know! |
| 21:38.46 | Katty | they did know :> |
| 21:38.48 | Katty | <3 sangoma |
| 21:40.51 | Kobaz | Katty: basically run wancfg_dahdi... and you're good to go |
| 21:41.08 | Katty | yes'r, that's what mister sangoma guy said too |
| 21:41.49 | Katty | and i'm sure the nice folks at sangoma will help me if i get my pantines in a wad over it. |
| 21:42.12 | Kobaz | panteen prov |
| 21:45.21 | Katty | that stuff sucks |
| 21:45.27 | Katty | i'd recommend giovanni |
| 21:45.33 | Kobaz | heh |
| 21:46.36 | Katty | the smooth as silk one is paritcularly nice |
| 21:47.02 | Katty | unfortunately the smooth as silk conditioner by giovanni is a bunch of crap. |
| 21:48.18 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:48.53 | ellisdee | never a dull moment in here. |
| 21:51.34 | *** join/#asterisk Tim_Toady (~moi@188.4.62.188.dsl.dyn.forthnet.gr) |
| 21:52.46 | *** join/#asterisk TimeRider (steve@5ac7b34d.bb.sky.com) |
| 21:53.26 | Katty | well we wouldn't want that |
| 21:53.30 | Katty | someone might fall asleep |
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| 22:02.24 | sahafeez | dumb question |
| 22:02.30 | sahafeez | why does it say not in uses |
| 22:02.32 | sahafeez | Local/5029@from-internal/n with penalty 15 (Not in use) has taken no calls yet |
| 22:02.50 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 22:04.48 | russellb | that's the device state |
| 22:05.00 | russellb | if it was on a call, it would say something else |
| 22:05.02 | russellb | like ... In Use |
| 22:05.03 | *** join/#asterisk hfb (~hfb@pool-98-112-220-104.lsanca.dsl-w.verizon.net) |
| 22:05.42 | sahafeez | ah, got it. thanks |
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| 22:07.03 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 22:11.25 | sahafeez | when i add wait to a queue |
| 22:11.34 | sahafeez | the ext,xx |
| 22:16.06 | *** part/#asterisk rttrey (~trey@209.208.18.121) |
| 22:19.37 | hardwire | ok.. anybody know if you can pass ENV to AGI? I seem to have lost the ability or forgotten how |
| 22:19.52 | hardwire | I'm using Set(ENV(variable)=data) before calling AGI(...) |
| 22:22.58 | *** join/#asterisk capitan (~alalalal@fw-0.jm811.aerioconnect.net) |
| 22:23.08 | capitan | hellloooo :) |
| 22:24.13 | capitan | hmmm... i 1.4.23.1 |
| 22:24.25 | [TK]D-Fender | hardwire: why would you need to? |
| 22:24.38 | capitan | hmmm... i'm running 1.4.23.1... and my voicemails are getting written out with really messed up permissions, and so i can't listen to them: |
| 22:24.52 | capitan | -------rw- 1 asterisk asterisk 280 2010-03-05 13:38 msg0000.txt |
| 22:25.00 | upb | wrong umask ? |
| 22:25.49 | [TK]D-Fender | capitan: Upgrade. You're already 6 behind |
| 22:26.01 | [TK]D-Fender | capitan: and verify your voicemail.conf settings for permissions |
| 22:26.07 | *** join/#asterisk mnick86 (~mnick86@95-90-248-233-dynip.superkabel.de) |
| 22:26.10 | capitan | upb: oddly enough, the wav files are correct |
| 22:26.12 | capitan | -rwx------ 1 asterisk asterisk 131884 2010-03-05 14:02 msg0000.wav |
| 22:26.46 | upb | i really know 0 about asterisk but i would strace the process which writes them :) |
| 22:26.59 | capitan | [TK]D-Fender, i'd like to verify that it's not a local setup issue before trying that... |
| 22:27.20 | *** join/#asterisk voipmonk (~shido6@m485e36d0.tmodns.net) |
| 22:28.30 | idespinner | a long shot but... any shoretel guys in here? |
| 22:28.50 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 22:33.37 | capitan | [TK]D-Fender, plus, i don't see any bug reports related to this... so i'm suspecting my local install :( |
| 22:34.36 | *** join/#asterisk defsdoor (~andy@defsdoor.gotadsl.co.uk) |
| 22:35.51 | hardwire | [TK]D-Fender: nevermind.. I found the problem with the agi file... apparently there was a mispelling. |
| 22:35.56 | capitan | hmmm... i see someone reporting the same thing in #asterisk logs... |
| 22:39.08 | capitan | damn... the guy did a workaround :( adding a externnotify script to fix up the permissions |
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| 22:45.11 | Qwell | capitan: upgrade.. |
| 22:45.16 | Katty | SO |
| 22:45.22 | Katty | i was thinking BLTs for dinner |
| 22:45.28 | Qwell | Katty: BLF for dinner |
| 22:45.31 | Katty | who wants to come over to my house for dinner and some drinks. |
| 22:45.45 | Katty | Qwell: i'm afeared that went over my head, perhaps |
| 22:45.54 | Qwell | ~blf |
| 22:45.54 | infobot | extra, extra, read all about it, blf is Busy Lamp Field, aka little lights next to speed dials that light up when the person is on the phone and blink when that line is ringing. hint extensions are static mapped to SIP or other channels. |
| 22:46.09 | Katty | ohthat |
| 22:47.19 | Katty | yeah i think that BLTs sound better. |
| 22:47.35 | raden_work | heya Katty :) |
| 22:47.39 | Katty | hi raden |
| 22:50.45 | Katty | should get a pair of Genets |
| 22:57.37 | Katty | k,time to go home |
| 22:57.40 | Katty | later gators! |
| 23:00.13 | hardwire | does anybody ever see funky jitter wheny ou leave a pri to the telco and come straight back in another? |
| 23:00.21 | hardwire | it's the strangest thing to me. |
| 23:01.22 | hardwire | I have two boxes.. each has a PRI to the local telco on the same telco switch in a "group" so that one fails over to the other. |
| 23:01.26 | hardwire | for inbound |
| 23:01.49 | hardwire | anyways.. I can eliminate this as a problem, just thought it was strange |
| 23:01.59 | *** join/#asterisk korihor (~korihor@201.210.226.98) |
| 23:03.03 | hardwire | nevermind |
| 23:03.48 | capitan | Qwell and [TK]D-Fender, would it be safe to just install from source over my old version, since it's just a few minor revisions? or do i have to uninstall first? |
| 23:04.46 | *** join/#asterisk Dibri (~gavit@190.98.33.38) |
| 23:10.03 | hardwire | jitter due to low disk space. |
| 23:10.04 | hardwire | sigh |
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| 23:51.44 | hipitihop | do I assume correctly that the skype client cannot be used as a general sofphone on asterisk without purchasing licensed add-on ? |
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| 23:57.24 | hipitihop | is there a way to query my voip provider to see which codecs they support |
