| 00:03.54 | *** join/#asterisk Maliuta (~nobusines@kiev.lusan.id.au) |
| 00:08.11 | *** join/#asterisk zyphlar (~zyphlar@wsip-68-14-229-148.ph.ph.cox.net) |
| 00:23.57 | *** join/#asterisk sterne (~user@75.99.222.123) |
| 00:26.41 | *** join/#asterisk Ionic (ionic@ionic.de) |
| 00:27.15 | p3nguin | Is there any temporary method to increase verbosity and/or debug via dial plan? |
| 00:28.25 | WIMPy | system()? |
| 00:29.18 | p3nguin | I suppose that'll work okay before a Dial, but that means I have to use System() in extension h to turn verbose back down again? |
| 00:30.14 | WIMPy | It's certainly not a nice thing, but it should work. |
| 00:33.17 | p3nguin | I couldn't think of any other ways. |
| 00:33.55 | WIMPy | neither |
| 00:35.38 | p3nguin | Maybe using Dial's g option would allow me to run System again after the Dial ends. |
| 00:37.01 | WIMPy | Only if the callee hangs up. |
| 00:38.48 | *** join/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 00:53.23 | saxa | p3nguin: i think yes, i'm not sure since the other day i was playing also with the phone settings and i don't remember if i disable it or not. |
| 00:53.30 | saxa | p3nguin: let me check |
| 00:55.02 | *** part/#asterisk rdm (rdm@unaffiliated/qubix) |
| 00:58.11 | saxa | p3nguin: yes it is enabled |
| 00:58.14 | saxa | http://oi41.tinypic.com/2dgl6ki.jpg |
| 00:58.19 | saxa | see the screenshot |
| 00:59.41 | saxa | i have removed the ip from the setting you see marked in here http://oi40.tinypic.com/rrpflv.jpg and it stopped working |
| 01:01.03 | p3nguin | I told you three days ago to disable nat traversal on the phone. It's no wonder it wasn't working. |
| 01:02.03 | p3nguin | Normally, you disable nat traversal on the phone and let asterisk handle the translations. |
| 01:02.24 | p3nguin | If you enabled nat traversal on the phone but did not otherwise configure it correctly, it's no wonder it failed. |
| 01:12.21 | saxa | i tried to disable it 3 days ago, but it was not working |
| 01:12.31 | saxa | so then i enable it again. |
| 01:12.54 | saxa | ok let me try to disable it. |
| 01:13.54 | saxa | do i disable it completely or do i leave the send keep-alive signal ? |
| 01:13.57 | WIMPy | If you've got Asterisk on the other side, never enable STUN or any other nat helper options, unless you know very exactely what you're doing. |
| 01:14.19 | WIMPy | keep-alive can't do any harm. |
| 01:14.20 | saxa | thats not my case :) |
| 01:14.38 | saxa | i do not know 100% what am i doing |
| 01:15.15 | saxa | i'm sorry but i'm kind of new to this sip thing |
| 01:17.10 | saxa | ok but now it does not work anymore |
| 01:17.38 | saxa | i was able to hear the voicemail login only when i set the ip in the phone |
| 01:17.59 | saxa | stun is disabled |
| 01:18.41 | WIMPy | Is nat enabled for the peer on Asterisk? |
| 01:18.50 | saxa | yes |
| 01:19.07 | saxa | in the peer section and in [general] |
| 01:19.28 | saxa | nat=yes |
| 01:19.35 | saxa | directmedia=no |
| 01:19.36 | WIMPy | And the router you use now has not SIP nat support? |
| 01:19.44 | saxa | no |
| 01:19.48 | saxa | afaik |
| 01:20.09 | saxa | on the asterisk side for sure not |
| 01:20.12 | WIMPy | That was a linux thing? |
| 01:20.37 | saxa | on the client side i will check it to be 100% sure |
| 01:20.49 | saxa | WIMPy: how you mean that ? |
| 01:21.17 | WIMPy | The router |
| 01:22.11 | saxa | unfortunately on the aterisk side i have a huawei hg521 modem/router |
| 01:22.31 | WIMPy | On the client side |
| 01:22.37 | saxa | so now i added the ip on my phone, and with stun disabled it does not work either |
| 01:22.45 | WIMPy | Do you have other clients connected? |
| 01:23.17 | saxa | on the client side i have a computer, few wifi clients and this sip phone |
| 01:23.42 | WIMPy | Clients at other locations |
| 01:23.50 | saxa | all are connected to the linksys |
| 01:23.57 | saxa | no |
| 01:24.05 | saxa | no clients at other locations |
| 01:24.17 | saxa | i can add your phone if you want to test |
| 01:24.26 | WIMPy | So we can't even be sure on which side the trouble happens.. |
| 01:24.41 | saxa | do you wanna test ? |
| 01:24.59 | saxa | give me a sec. I set up an account for you |
| 01:25.40 | WIMPy | Not sure, when. I'm busy writing some important shit ATM. |
| 01:25.48 | WIMPy | But I might be half done. |
| 01:29.52 | saxa | ok seems now i can again here the voicemail. |
| 01:33.02 | p3nguin | I'll register to your system if you give me an account. |
| 01:37.22 | saxa | ok as said, if i put the ip in my phone then i can hear the sounds, otherwise is muted |
| 01:39.41 | saxa | p3nguin: let me know when are you ready. |
| 01:40.10 | saxa | p3nguin: in case please try to ring 1005 which is my phone here |
| 02:04.53 | p3nguin | saxa: I'm ready as soon as I get the credentials/info for your system. |
| 02:05.21 | saxa | p3nguin: right now WIMPy is registered |
| 02:08.41 | *** join/#asterisk willbradley (~wbradley@unaffiliated/willbradley) |
| 02:15.16 | *** join/#asterisk angryuser_laptop (~angryuser@ps444-1-78-224-152-133.fbx.proxad.net) |
| 02:23.22 | WIMPy | Erm, is register => no longer supported? |
| 02:23.33 | p3nguin | haha |
| 02:23.36 | p3nguin | Of course it is! |
| 02:23.41 | p3nguin | Why wouldn't it be? |
| 02:24.23 | WIMPy | Well, I've go two ITSPs configured, one with callbackextension and one with a seperate register line, and the later is no longer shown on 'sip show registry'. |
| 02:25.26 | p3nguin | Do you have the register statement in the correct place? It must be in the general section, before the authentication section if it exists. |
| 02:25.45 | WIMPy | The same place it has always been. |
| 02:26.06 | WIMPy | Damn. |
| 02:27.07 | WIMPy | wonders if the web I/F tells me since when I'm offline. |
| 02:29.58 | WIMPy | Hmm. Mine tells me I registered 35 minutes ago. |
| 02:30.02 | WIMPy | That came from AMI. |
| 02:30.07 | WIMPy | gives it a kick. |
| 02:30.46 | saxa | is doing a factory reset of the phone to see if it helps. |
| 02:30.59 | WIMPy | He, test1 is still reachable. |
| 02:31.14 | WIMPy | Oh, it's back. |
| 02:31.18 | WIMPy | Interesting. |
| 02:31.33 | WIMPy | must have happened while testing. |
| 02:32.04 | WIMPy | But sip reload didn't help, but after core restart now it's back. |
| 02:32.14 | WIMPy | I hate this shit. |
| 02:35.11 | saxa | factory reset of the phone has not helped :( |
| 02:36.53 | WIMPy | Do you have the possibility to restart the router on the server side? |
| 02:37.25 | saxa | yes |
| 02:37.47 | saxa | but will then need to wait about 10 min to reupdate the IP |
| 02:38.00 | WIMPy | Have you tried that? |
| 02:38.16 | saxa | of course , many times |
| 02:38.28 | WIMPy | Sometimes routers keep some information that confuses themselves. |
| 02:38.31 | WIMPy | ok |
| 02:38.34 | saxa | but in my opinion something is wrong somewhere in the middle |
| 02:38.50 | WIMPy | That would surely be an explanation. |
| 02:39.00 | saxa | it worked like a charm on your box |
| 02:39.11 | WIMPy | But it's the least likely place. |
| 02:39.12 | saxa | for you worked on my box |
| 02:39.36 | saxa | its just my phone that doesnt work with my box :) |
| 02:39.54 | saxa | no longer , because it was working :) |
| 02:41.05 | saxa | see, if i put my external ip in the phone to use it , the thing start to work |
| 02:41.07 | WIMPy | Very strange. |
| 02:41.31 | saxa | so something for some reason is not comunicating to asterisk where my phone is |
| 02:41.53 | saxa | also if it registers with the correct ip |
| 02:42.18 | saxa | i mean the phone registers itself on the box with the correct ip |
| 02:42.51 | saxa | but as [TK]D-Fender said, for some reason the rtp trafic goes to my internal ip |
| 02:43.01 | saxa | the privat one |
| 02:43.27 | WIMPy | Do you have localnet configured on Asterisk? |
| 02:44.00 | WIMPy | Is the Asterisk on a public IP directly? |
| 02:45.35 | saxa | http://pastebin.ca/2109839 |
| 02:46.05 | saxa | this is how it looks rtp packets when i use the wan ip in this phone setting and it works. |
| 02:46.24 | saxa | i have localnet setting in sip.conf |
| 02:46.44 | saxa | WIMPy: no asterisk is behind a nat |
| 02:47.05 | saxa | i have opened the 5060 port and the 10000 to 10200 |
| 02:47.39 | WIMPy | wonders if Aasterisk somehow gets confused into thinking the phone is local. |
| 02:48.40 | saxa | maybe |
| 02:49.00 | saxa | but it sents the packets to the correct local ip |
| 02:49.16 | saxa | it sends packets to 192.168.12.x |
| 02:49.37 | saxa | but asterisk itself its on a 192.168.0.1 |
| 02:49.49 | saxa | so there is no confusion |
| 02:50.02 | WIMPy | I still wonder why I couldn't hear you, either. |
| 02:50.26 | saxa | but i heard you perfectly |
| 02:50.40 | WIMPy | on your server. |
| 02:50.47 | saxa | oh yes |
| 02:50.58 | saxa | have no idea |
| 02:51.09 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 02:51.30 | saxa | anyway i will leave the phone like it is now since it seem to work , at least partially it works |
| 02:52.07 | saxa | then i will try to buy a new phone and will see if this one will work |
| 02:52.25 | saxa | any suggestion of the new phone i should buy ? |
| 02:52.34 | saxa | i have to buy it in any case |
| 02:52.49 | saxa | will put that one i have to use it on a local net |
| 02:53.03 | saxa | and will use the new one in my home |
| 02:53.11 | WIMPy | I like the old Snoms, but everyone will have a differen opinion there. |
| 02:53.14 | saxa | cisco ? |
| 02:53.33 | saxa | i bought the grandstream because they were cheap |
| 02:53.43 | saxa | about 35 euros |
| 02:53.50 | saxa | iirc |
| 02:54.58 | saxa | ok sleep time. |
| 02:55.10 | saxa | thanks for all your help and time |
| 02:55.16 | WIMPy | Good night |
| 02:55.23 | saxa | c u , night |
| 02:58.21 | *** join/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 03:17.44 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 03:28.28 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 03:38.03 | *** part/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 03:50.11 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 03:55.28 | *** join/#asterisk arnotixe (~arnotixe@cl-205.udi-01.br.sixxs.net) |
| 03:56.01 | arnotixe | hi all I have this issue from time to time (weeks between): Iax2 kind of "gives up". http://paste.pound-python.org/show/16578/ |
| 03:56.02 | *** join/#asterisk D-Boy (~Geek@unaffiliated/cain) |
| 03:57.18 | WIMPy | Is that one of the early 1.8 versions? |
| 03:57.22 | arnotixe | iax2 reload usually fixes it in a matter of mSeconds. Is this something related to dynamic DNS on the remote end maybe? If so, can I tell asterisk to never give up, and to lookup the remote IP somehow? |
| 03:57.47 | WIMPy | Do you use dnsmgr? If no, turn it on. |
| 03:58.33 | arnotixe | it's 1.8.8.0 |
| 03:58.46 | arnotixe | googling dnsmgi |
| 03:58.49 | arnotixe | gr |
| 03:59.06 | WIMPy | dnsmgr in Asterisk |
| 03:59.31 | arnotixe | hmm at least there's no /etc/asterisk/dnsmgr.conf... |
| 03:59.43 | arnotixe | should I just copy http://www.voip-info.org/wiki/view/Asterisk+config+dnsmgr.conf |
| 03:59.53 | arnotixe | seems quite simple that file |
| 04:00.32 | WIMPy | Yes, but uncomment the two options. |
| 04:00.45 | arnotixe | ok. then restart asterisk maybe? |
| 04:01.15 | WIMPy | Not sure how much needs a restart for a new option. |
| 04:02.01 | arnotixe | ok I restarted all anyways, it's just my home pbx. seems to have come up again. How do I check if it's loaded? |
| 04:02.52 | WIMPy | 'dnsmgr status' |
| 04:03.32 | arnotixe | ah I tried that but before reloading... Now it's much better, says "enabled" :D |
| 04:03.48 | arnotixe | I'll leave it like this for a while to see if it hangs ever again. |
| 04:04.45 | arnotixe | Would this be necessary on both ends? The server I'm connecting to has a dynamic DNS name, but this here end has no name actually. |
| 04:05.22 | WIMPy | Only where you use the dynamic name. |
| 04:05.36 | p3nguin | You can restart dnsmgr with dnsmgr reload. |
| 04:05.42 | p3nguin | No sense in restarting the entire thing. |
| 04:06.01 | arnotixe | oh so the dnsmgr must be running on the machine with the dynamic dns? |
| 04:06.02 | arnotixe | hehe |
| 04:06.45 | p3nguin | Names just make things easier to remember. |
| 04:07.06 | WIMPy | No. use as in where it's used in the configuration. |
| 04:07.48 | WIMPy | Is it possible to send another connected line update during a call? |
| 04:09.56 | arnotixe | ok maybe that same issue causes the SIP subscriptions to hang as well, I've turned on dnsmgr on both sides to see what happens.. thank |
| 04:09.58 | arnotixe | s |
| 04:10.31 | WIMPy | Yes, that's where I found I needed it. |
| 04:10.56 | WIMPy | Otherwise it wouldn't recover after an DNS issue. |
| 04:18.39 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 04:29.57 | *** part/#asterisk sterne (~user@75.99.222.123) |
| 04:37.55 | *** join/#asterisk pdtpatr1ck (~pdtpatric@ip72-211-207-15.oc.oc.cox.net) |
| 05:15.53 | *** join/#asterisk Sorcier_FXK (~nsystem@unaffiliated/sorcierfxk) |
| 05:16.14 | Sorcier_FXK | !rss |
| 05:16.16 | Sorcier_FXK | oops |
| 05:16.18 | Sorcier_FXK | sry |
| 05:46.27 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 05:52.00 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 05:54.57 | WIMPy | Hmm. My outbound registration vanishes every time I do a 'sip reload'. |
| 05:54.59 | WIMPy | nice |
| 06:14.17 | *** join/#asterisk bipul (~root@unaffiliated/bipul/x-4918593) |
| 06:14.40 | p3nguin | Sounds broken. |
| 06:14.49 | p3nguin | It should re-register, but not vanish. |
| 06:15.41 | WIMPy | Yes, that's the way I remembered it. But somethign has changed, it seems. |
| 06:15.50 | p3nguin | Works correctly for me. |
| 06:16.04 | p3nguin | Asterisk 1.8.8.0 built by rob @ cpe-e650 on an i686 running Linux on 2011-12-20 23:27:42 UTC |
| 06:16.46 | WIMPy | trunk from yesterday |
| 06:17.15 | p3nguin | That's why I only use stable software. |
| 06:17.51 | WIMPy | Not something I'd use with Asterisk. |
| 06:23.50 | *** join/#asterisk Russ (foobar@ip68-106-254-4.ph.ph.cox.net) |
| 06:26.58 | WIMPy | Ah. It's dundi that throws the tons of mutex errors. |
| 06:29.40 | *** join/#asterisk zyphlar (~zyphlar@wsip-68-14-229-148.ph.ph.cox.net) |
| 06:31.23 | *** join/#asterisk roham (~ali@31.184.187.2) |
| 06:46.22 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 06:55.59 | *** join/#asterisk ccesario (~ccesario@177.35.100.170) |
| 07:30.09 | *** join/#asterisk tm1000 (~tm1000@provisioner.net) |
| 07:58.36 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 08:09.20 | *** join/#asterisk singler (~singler@84.15.129.49) |
| 09:15.05 | *** join/#asterisk [ALT][F4] (~altf4@ppp-58-11-84-79.revip2.asianet.co.th) |
| 09:15.24 | [ALT][F4] | afternoon .. |
| 09:16.10 | [ALT][F4] | guess that's another dump questions .. but does anyone have a FAQ or somethings regarding voice-drops while active SIP calls in the intranet ? |
| 09:16.25 | [ALT][F4] | like every 3 minutes i have total silence for about 5 sec. |
| 09:21.05 | kaldemar | sounds like network issues. |
| 09:21.15 | [ALT][F4] | network issues excluded since it's intranet (phone - switch - asterisk) |
| 09:21.43 | [ALT][F4] | yeah i was thinking the same … but its quite straight forward setup |
| 09:21.56 | kaldemar | do you see anything in asterisk when that happens? |
| 09:22.07 | [ALT][F4] | no zero log entries |
| 09:22.15 | kaldemar | not log, but CLI. |
| 09:22.50 | [ALT][F4] | have not try CLI yet. |
| 09:23.07 | kaldemar | that's what you should do. is RTP going directly between phones? |
| 09:23.36 | [ALT][F4] | this may be related: if i just pickup the SIP phone … and hang up .. the phone shows "no service" for about 10 sec. |
| 09:23.48 | [ALT][F4] | also sometimes in idle it just drop the registration |
| 09:24.05 | [ALT][F4] | thats why i as suspecting its some asterisk setting |
| 09:25.05 | [ALT][F4] | 90% of calls are placed trough SIP trunks or Analog trunks .. so RTP is unlikely a issue .. |
| 09:25.07 | [ALT][F4] | is it ? |
| 09:25.31 | kaldemar | you're ruling out possible issues quite lightly. |
| 09:26.59 | kaldemar | especially if you're using SIP providers, do not rule out network issues since there is other network besides your own switch involved. |
| 09:27.49 | [ALT][F4] | oh yes .. absolutly agree … but its happening in the same fashion if i use analog trunks |
| 09:28.03 | kaldemar | SIP debug or RTP debug during an incident may tell you something useful. |
| 09:28.57 | [ALT][F4] | i may just call the remote mailbox for a test … with debugging on .. |
| 09:29.22 | kaldemar | what is the phone you're using? |
| 09:29.50 | [ALT][F4] | Funkwerk IP50 .. on Atcom IP04 PBX |
| 09:30.32 | [ALT][F4] | but i have the same issue on another phone (analog with Linksys PAP2) |
| 09:30.34 | kaldemar | so it's not a phone-switch-asterisk setup but phone-pbx-switch-asterisk at least? |
| 09:31.15 | [ALT][F4] | no its Phone - switch - Asterisk |
| 09:31.38 | [ALT][F4] | the ip04 is a pbx with asterisk running on it .. |
| 09:32.10 | [ALT][F4] | so the ip phone connects via UDP directly to asterisk |
| 09:34.45 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:21f:d0ff:fea1:568e) |
| 09:35.04 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 09:38.32 | *** join/#asterisk plundra (1000@v0.article.se) |
| 09:42.08 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 09:43.30 | [ALT][F4] | loging on … talk since 15 minutes with the remote voicemail .. but no drop :( |
| 09:43.39 | [ALT][F4] | demonstration effect i guess |
| 09:47.12 | *** join/#asterisk bn-7bc (bjarne@pdpc/supporter/active/bn-7bc) |
| 09:54.56 | *** join/#asterisk whateverany (~yaaic@182-239-213-216.ip.adam.com.au) |
| 10:00.55 | *** join/#asterisk [ProB]CrazyMan (~chatzilla@mx40.roterschnee.com) |
| 10:01.48 | *** part/#asterisk whateverany (~yaaic@182-239-213-216.ip.adam.com.au) |
| 10:01.58 | *** join/#asterisk jzaw (~jzaw@2001:8b0:7:0:216:cbff:febe:f5dd) |
| 10:02.13 | *** join/#asterisk khem_ (~ws@64.85.163.32) |
| 10:02.22 | *** join/#asterisk mjordan (~mjordan@conference/fosdem/x-bdjdessqmivtmaog) |
| 10:02.22 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 10:13.02 | *** join/#asterisk jsjc (~Adium@160.Red-88-6-109.staticIP.rima-tde.net) |
| 10:34.27 | *** join/#asterisk Rich[ard] (admin@ppp203-122-217-95.static.internode.on.net) |
| 10:36.45 | Rich[ard] | Hey all, i have just setup my own AsteriskNow server, and have come to a halt because of outgoing calling plans, for an example, in the GUI i have setup a rule so i can dial 111 for my ISP's voicemail, so i added rule _XXX! x being 0-9, or am i doing this completely wrong? |
| 10:37.35 | *** part/#asterisk khem_ (~ws@64.85.163.32) |
| 10:39.52 | *** join/#asterisk Charrit (~Zairus@30.109.165.83.dynamic.mundo-r.com) |
| 10:39.59 | Rich[ard] | Could anyone please assist me |
| 10:44.08 | *** join/#asterisk pnunn (~pnunn@58.108.220.221) |
| 10:53.00 | *** join/#asterisk dandate2 (~dan@180.190.172.44) |
| 10:53.16 | dandate2 | if i'm just going to setup a bunch of ip phones is the catalyst 3550 preferable to the 2950? |
| 11:04.16 | Rich[ard] | Anyone available |
| 11:04.27 | Rich[ard] | ? |
| 11:05.20 | Rich[ard] | Seriously this many people in the channel and no one free to provide assistance? |
| 11:10.51 | *** join/#asterisk Dibbler (~Dibbler@87-194-103-72.bethere.co.uk) |
| 11:12.50 | Charrit | 3550 is always preferable to the 2950, but 2950 is a very good choice, you don't need to go further |
| 11:13.14 | Rich[ard] | * dandate2 (~dan@180.190.172.44) Quit (Ping timeout: 252 seconds) |
| 11:14.12 | Charrit | Rich[ard], what's your problem? |
| 11:14.55 | Rich[ard] | Just requiring assistance with AsteriskNow in regards to Outgoing Calling Rules |
| 11:15.46 | *** join/#asterisk dandate2 (~dan@180.190.172.44) |
| 11:16.02 | Charrit | I don't use AsteriskNow, but what's the problem? perhaps may I can help |
| 11:16.26 | Rich[ard] | dandate2, I believe Charrit had a reply for you |
| 11:16.36 | Rich[ard] | <Charrit> 3550 is always preferable to the 2950, but 2950 is a very good choice, you don't need to go further |
| 11:16.44 | dandate2 | oh |
| 11:17.05 | dandate2 | any performance difference or just crazy features i might need if i also hook up a bunch of servers to the network? |
| 11:17.55 | Charrit | cat 2950 has a good performance for servers too |
| 11:18.33 | dandate2 | oh heh see i'm only trying to run ip phones and nothing else |
| 11:18.48 | Charrit | my recomendation for your setting up is to put telephony in a separate vlan to the network servers using the switch capabilities not the phone tagging |
| 11:18.51 | dandate2 | thats why i'm wondering if the 3550 would make any difference for me, the 2950 is available locally |
| 11:18.59 | dandate2 | available for sale local i mean |
| 11:19.39 | dandate2 | yeah our pbx is located in the united states and i'm trying to hook up all these ip phones in the philippines |
| 11:20.18 | dandate2 | i could order a 3550 from texas but it might not arrive = |
| 11:20.18 | Nugget | Don't mess with Texas. |
| 11:20.28 | Charrit | 3550 is a level 3 switch and if you're asking about what to do with it... probably you don't need it |
| 11:20.36 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 11:20.42 | dandate2 | LOL good advice! |
| 11:20.44 | [sr] | yellow |
| 11:22.21 | Charrit | I wanted to write layer 3, not level 3 |
| 11:34.33 | *** join/#asterisk luke0512 (~eric@HSI-KBW-091-089-189-101.hsi2.kabel-badenwuerttemberg.de) |
| 11:34.53 | luke0512 | hello |
| 11:35.27 | *** join/#asterisk mjordan (~mjordan@conference/fosdem/x-ynekpnooltjqkkuv) |
| 11:35.27 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 11:35.57 | Rich[ard] | Is there any dial plans so i can dial 0 to get out and then dial what ever i want exept adding more dial plans? |
| 11:40.49 | Rich[ard] | http://i41.tinypic.com/2co6a2h.png |
| 11:41.17 | Rich[ard] | Can someone help me, those are the rules i am using, but unable to make phone calls |
| 11:44.07 | dandate2 | i think you need to make a speed dial or a combination that will never be used like *0 or #0 |
| 11:45.37 | Rich[ard] | Sorry dandate2, did quite get you, so i would put *0 or #0 into the pattern? |
| 11:45.53 | Rich[ard] | sorry setting up a pbx for the first time |
| 11:46.13 | Rich[ard] | and thanks for the reply |
| 11:53.55 | luke0512 | does someone know a useable howto to configure a hfc card with cologne chip connected to an intern S0 bus |
| 12:03.09 | luke0512 | http://s14.directupload.net/file/d/2791/pl4angdf_jpg.htm <-- my network/multimedia structure |
| 12:05.02 | luke0512 | from modem there are 3 different kinds of connection |
| 12:05.25 | *** join/#asterisk xertres (~xertres@h-136-1-238.a176.priv.bahnhof.se) |
| 12:05.38 | luke0512 | one is ethernet to fli4l one is cable TV and the third is ISDN |
| 12:19.19 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 12:39.39 | dandate2 | does the catalyst 2950 not POE my ip phones? |
| 12:42.28 | dandate2 | i mean to ask, does it come with inline power? |
| 12:48.05 | luke0512 | joins the sunshine outside and bbl |
| 13:04.34 | *** join/#asterisk darkbasic (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 13:10.32 | *** part/#asterisk bbourdage (~bbourdage@75-150-221-205-Illinois.hfc.comcastbusiness.net) |
| 13:16.25 | *** join/#asterisk angryuser_laptop (~angryuser@2a01:e34:ee09:8850:2cb7:4419:8b13:7c21) |
| 13:17.12 | *** join/#asterisk bbourdage (~bbourdage@75-150-221-205-Illinois.hfc.comcastbusiness.net) |
| 13:24.52 | *** join/#asterisk binbash_ (~peter@server.digitog.nl) |
| 13:25.19 | *** join/#asterisk binbash_ (~peter@server.digitog.nl) |
| 13:26.18 | *** join/#asterisk binbash_ (~peter@server.digitog.nl) |
| 13:45.00 | *** join/#asterisk mjordan (~mjordan@conference/fosdem/x-dzkqumqhiksdohyq) |
| 13:45.00 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:51.32 | *** join/#asterisk Srini (~Srinivasa@219.91.145.198) |
| 13:51.43 | Srini | Hi Room |
| 13:52.23 | Srini | I am trying a digium te220 on centos (vicidial distro) and seeing DAHDI_SPANCONFIG failed on span 1: Invalid argument (22) |
| 13:52.29 | Srini | Can someone help please? |
| 13:55.11 | Srini | My configurations are at : http://pastie.org/3321393 |
| 15:01.16 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 15:12.53 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 15:32.25 | *** join/#asterisk jsjc (~Adium@160.Red-88-6-109.staticIP.rima-tde.net) |
| 15:39.21 | *** join/#asterisk zyphlar (~zyphlar@wsip-68-14-229-148.ph.ph.cox.net) |
| 15:41.14 | *** join/#asterisk mjordan (~mjordan@193.191.35.20) |
| 15:41.15 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 16:04.25 | *** join/#asterisk chasing`Sol (~cS@197.132.196.205) |
| 16:30.51 | *** join/#asterisk serafie (~erin@75.76.38.159) |
| 16:35.07 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 16:38.21 | *** join/#asterisk mjordan (~mjordan@28.254-240-81.adsl-static.isp.belgacom.be) |
| 16:38.22 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 16:52.31 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 17:07.53 | *** join/#asterisk linuxviewer (~test@ip70-162-145-222.ph.ph.cox.net) |
| 17:15.08 | *** join/#asterisk mjordan (~mjordan@28.254-240-81.adsl-static.isp.belgacom.be) |
| 17:15.08 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 17:18.42 | luke0512 | bye |
| 17:57.40 | *** join/#asterisk saftsack (~saftsackl@bautzen.weh.RWTH-Aachen.DE) |
| 18:00.06 | *** join/#asterisk jsjc (~Adium@160.Red-88-6-109.staticIP.rima-tde.net) |
| 18:06.01 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 18:21.07 | *** part/#asterisk bbourdage (~bbourdage@75-150-221-205-Illinois.hfc.comcastbusiness.net) |
| 19:08.32 | *** join/#asterisk angryuser_laptop (~angryuser@ps444-1-78-224-152-133.fbx.proxad.net) |
| 19:31.08 | *** join/#asterisk Korolev (~Korolev@nmd.sbx08290.northfl.wayport.net) |
| 19:38.12 | *** join/#asterisk nedimzecic (~nedimzeci@178.77.21.47) |
| 19:40.43 | *** join/#asterisk mautas (~saturno@186.23.83.25) |
| 19:41.07 | *** part/#asterisk nedimzecic (~nedimzeci@178.77.21.47) |
| 19:44.12 | mautas | hi, anybody in here? |
| 19:44.23 | *** join/#asterisk twanny796 (~twanny@46.11.4.100) |
| 19:50.26 | *** join/#asterisk tamiel (~tamiel@85-171-171-179.rev.numericable.fr) |
| 19:52.09 | p3nguin | ~ask |
| 19:52.09 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 19:56.58 | *** join/#asterisk xertres (~xertres@h-136-1-238.a176.priv.bahnhof.se) |
| 20:05.51 | *** join/#asterisk mjordan (~mjordan@28.254-240-81.adsl-static.isp.belgacom.be) |
| 20:05.51 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 20:10.10 | *** join/#asterisk pdtpatr1ck (~pdtpatric@ip72-211-207-15.oc.oc.cox.net) |
| 20:57.58 | ChrisInSydney | p3nguin: Mornin' |
| 20:59.06 | p3nguin | Hi. |
| 20:59.32 | ChrisInSydney | Got 1.8 installed and running on a virtual host. |
| 20:59.44 | ChrisInSydney | Managed to get the P-Asserted Identity working on transfers too |
| 20:59.48 | ChrisInSydney | on Snoms |
| 21:00.59 | WIMPy | Including the new destination? |
| 21:01.18 | ChrisInSydney | Both sides |
| 21:01.21 | ChrisInSydney | :-) |
| 21:01.43 | ChrisInSydney | unless I was haloucenating |
| 21:01.54 | WIMPy | I only see the information that the call is diverted, but the new destination is not displayed. |
| 21:02.03 | ChrisInSydney | What handsetd |
| 21:02.06 | ChrisInSydney | handsets |
| 21:02.08 | WIMPy | Not before it answers anyway. |
| 21:02.13 | ChrisInSydney | (need coffee) |
| 21:02.18 | WIMPy | Snom 360 |
| 21:02.25 | ChrisInSydney | once it answers, that when it happens |
| 21:02.34 | ChrisInSydney | get v8 firmware on it |
| 21:02.40 | ChrisInSydney | latest |
| 21:02.42 | WIMPy | Ok, so not more than I've got. |
| 21:02.48 | ChrisInSydney | 7.3.30 doesnt cut it |
| 21:02.54 | WIMPy | I've got v8 for ages. |
| 21:03.12 | ChrisInSydney | I've been holding back on the 300 series |
| 21:03.23 | WIMPy | In v7 the mini browser is too slow to be usefull. |
| 21:03.29 | ChrisInSydney | happy enough now though |
| 21:03.31 | ChrisInSydney | true |
| 21:04.15 | WIMPy | Fast approaching the nineties... |
| 21:04.39 | ChrisInSydney | Has its advantages, you still made mix tapes |
| 21:05.00 | ChrisInSydney | You had Nirvana and Soundgarden to look firward to :-) |
| 21:05.05 | WIMPy | Not sure that was an advantage. |
| 21:05.14 | ChrisInSydney | not for Kurt it wasnt |
| 21:05.29 | ChrisInSydney | Then we got stuck with Courtney |
| 21:05.31 | WIMPy | true |
| 21:06.17 | ChrisInSydney | Actually, the Ninetees was a pretty cool reneaisance in the music business, lots of cool stuff |
| 21:06.43 | ChrisInSydney | back to the...whats this decade called ? |
| 21:07.32 | ChrisInSydney | I works for the Cisco SPA525G2 without the rpid in sip.conf |
| 21:07.45 | ChrisInSydney | The Aastra 6731 is not behaving |
| 21:08.17 | ChrisInSydney | I think that there is a setting burried in there somewhere. only via TFTP though |
| 21:15.12 | *** join/#asterisk pdtpatr1ck (~pdtpatric@ip72-211-207-15.oc.oc.cox.net) |
| 21:51.02 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 22:10.31 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 22:11.08 | *** part/#asterisk Cain (~Geek@unaffiliated/cain) |
| 22:11.40 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 22:17.43 | *** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com) |
| 22:19.19 | *** join/#asterisk TimeRider (steve@188-220-34-144.dsl.cnl.uk.net) |
| 22:23.18 | *** join/#asterisk TimeRider (steve@188-220-34-144.dsl.cnl.uk.net) |
| 22:32.59 | *** join/#asterisk zyphlar (~zyphlar@wsip-68-14-229-148.ph.ph.cox.net) |
| 22:46.44 | *** join/#asterisk zyphlar (~zyphlar@wsip-68-14-229-148.ph.ph.cox.net) |
| 22:53.56 | *** join/#asterisk mjordan (~mjordan@28.254-240-81.adsl-static.isp.belgacom.be) |
| 22:53.56 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 22:54.53 | *** join/#asterisk mpe (~mpe@gate.ipvision.dk) |
| 22:55.50 | *** part/#asterisk mjordan (~mjordan@28.254-240-81.adsl-static.isp.belgacom.be) |
| 23:23.14 | *** join/#asterisk Korolev (~Korolev@c-98-254-233-27.hsd1.fl.comcast.net) |
