| 00:30.56 | *** join/#asterisk jmacz (~jmacz@190.25.38.175) |
| 00:35.49 | *** join/#asterisk darkskiez_ (~dz@62-50-230-5.client.stsn.net) |
| 00:43.43 | *** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110) |
| 00:55.21 | *** join/#asterisk cdahmedeh (~cdahmedeh@CPE000c41445085-CM001225409602.cpe.net.cable.rogers.com) |
| 00:56.11 | *** join/#asterisk Mango (~iMango@209.121.85.243) |
| 00:57.51 | Mango | Hi. I'm trying to set up IAX2 and have not been successful. Based on looking at the CLI and a packet sniffer, Asterisk receives something from Zoiper, and then attempts to respond. But, Zoiper never sees the response. |
| 00:58.11 | Mango | I opened port 4569 in both directions on the server but not sure what else to try. |
| 00:58.16 | Mango | Pastebin: |
| 00:58.38 | Trixboxer | Not able to get agi->get_variable("DIAL_NUMBER") in agibin recordingcheck file... I know that it is not set, but how do I find it ? |
| 00:59.01 | Mango | iax.conf: http://pastebin.ca/1912580 |
| 00:59.35 | Mango | CLI when I attempt to make a call: http://pastebin.ca/1912581 |
| 01:06.22 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 01:23.32 | *** join/#asterisk [Outcast] (~anonymous@24-183-177-242.dhcp.oxfr.ma.charter.com) |
| 01:32.43 | *** join/#asterisk bsdfreak (~alex@ninja.paranode.net) |
| 01:32.57 | *** join/#asterisk SaiSoma|AtHome (~SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |
| 01:35.26 | seanjohn | Trixboxer: variables are passed in agi through stdin and stdout; you have to array the stdin |
| 01:35.37 | seanjohn | to get the individual variables |
| 01:36.04 | Trixboxer | seanjohn, Thanks will search on it :) |
| 01:36.24 | seanjohn | it looks like you're using php. I have the required 10 lines to split up the variables |
| 01:37.00 | Trixboxer | actually the recordingcheck file gets executed before the fixlocalprefix |
| 01:37.11 | Trixboxer | so the variable "DIAL_NUMBER" is not set |
| 01:37.14 | seanjohn | is it php, perl or ? |
| 01:37.18 | Trixboxer | php |
| 01:38.22 | seanjohn | $stdin = fopen('php://stdin', 'r'); |
| 01:39.14 | Trixboxer | oh, I just got a simple way from file phpagi.php |
| 01:39.46 | Trixboxer | you can get the dialled number from $dnid = $agi->request['agi_dnid']; |
| 01:39.52 | Trixboxer | it worked :) |
| 01:41.34 | seanjohn | http://pastebin.com/f1PdtUcF |
| 01:42.02 | seanjohn | well, the functions I just pasted you also allow you to stdout |
| 01:42.13 | seanjohn | always do a flush after each stdout |
| 01:43.21 | seanjohn | I use php to write a busyredial.php; it creates a call file retrying every 30 seconds for 15 minutes, after the caller presses one on the busy message. You should try creating one of your own of these. |
| 01:43.51 | seanjohn | it will teach you asterisk fully to accomplish this |
| 01:43.56 | *** join/#asterisk bsdfreak (~alex@ninja.paranode.net) |
| 01:44.37 | seanjohn | its busy redial or bust :) |
| 01:45.15 | seanjohn | every aspect of asterisk is used in this from macros to call files to dial commands separating call legs. |
| 01:46.03 | Trixboxer | thanks a lot for this seanjohn |
| 02:06.13 | *** join/#asterisk coppice (~chatzilla@245.168.17.210.dyn.pacific.net.hk) |
| 02:23.04 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 02:58.19 | *** join/#asterisk acxty (~acxty@200.107.239.55) |
| 02:58.59 | acxty | Hi guys, I am trying to connect a sip account to asterisk but I am getting this error Registration from 'sip:2@192.168.0.45' failed for '192.168.0.100' - Peer is not supposed to register. 45 is the asterisk ip and 100 is the one that is trying to connect |
| 02:59.23 | acxty | I set the host=192.168.0.100 to it. If I set it to dynamic and connects fine |
| 03:00.20 | acxty | never mind it is reachable now |
| 03:02.58 | p3nguin_ | Only dynamic peers are allowed to register. If the host name is set, registration would not be necessary and therefore yields that error. |
| 03:03.19 | p3nguin_ | Registration is to let Asterisk know how to reach a device. |
| 03:03.40 | acxty | p3nguin_, thanks it is working now ;) |
| 03:03.58 | p3nguin_ | ~sipregister |
| 03:03.59 | infobot | [~sipregister] SIP registration is to tell your provider what IP address & EXTEN to send INCOMING calls to. Some ITSPs let you use a fixed address or host rather than registering. Registration is NOT normally needed to PLACE calls, as those are typically auth'ed independently. Others accept unauth'ed calls once you are registered (saves on negotiation BW). |
| 03:05.42 | acxty | One question. I want to make a dial to a cellphone. I am using a sip peer but I need to dial 9 before the number so it allow me to dial. I am doing Dial(SIP/MySip/9${MyNumber}) but the call is never made and it shows me Call failed to go through, reason (8) Congestion (circuits busy) |
| 03:07.11 | acxty | is that the correct way of doing it? |
| 03:07.32 | p3nguin_ | No. You should not need to dial a 9. |
| 03:07.52 | p3nguin_ | Why would a 9 be needed? |
| 03:08.59 | acxty | I am connected to a pbx. If I want to dial to a cellphone I need t press 9 before the number. Like that is how it is configure |
| 03:09.36 | p3nguin_ | So you connected Asterisk with another SIP-enabled PBX? |
| 03:09.41 | acxty | yes |
| 03:10.39 | p3nguin_ | When you were dialing through that PBX directly, did you have to dial 9, then wait a second or so for a new dial tone, then dial the real number? |
| 03:11.41 | acxty | yes. Yo dial 9 and wait for the tone |
| 03:11.46 | acxty | then you dial the cellphone |
| 03:11.48 | p3nguin_ | If you dialed 9-real-num without waiting between the 9 and the real num, would it fail? |
| 03:13.00 | p3nguin_ | I'm thinking you might be able to add some wait time between your 9 and the real number in the Dial() command and solve the issue. |
| 03:13.56 | acxty | how can I do that? that is what I have been looking |
| 03:15.19 | p3nguin_ | I'm guessing Dial's D option will be used along with a couple w's. I'm not completely sure. |
| 03:15.46 | p3nguin_ | w provides .5 second pause in dialing. |
| 03:17.36 | p3nguin_ | Something like Dial(SIP/MySip/9,30,D(www${EXTEN})) |
| 03:18.19 | p3nguin_ | exten => 3149691077,1,Dial(SIP/MySip/9,30,D(www${EXTEN})) |
| 03:18.50 | p3nguin_ | I think this would call 9 on the PBX and then wait 1.5 seconds, then send 3149691077 as the phone number. |
| 03:19.16 | p3nguin_ | You could obviously use patterns rather than fixed extension numbers. |
| 03:19.46 | p3nguin_ | exten => _NXXNXXXXXX,1,Dial(SIP/MySip/9,30,D(www${EXTEN})) |
| 03:21.55 | p3nguin_ | That's where I would start, anyway. |
| 03:25.35 | acxty | invalid timeout specified |
| 03:27.40 | acxty | it works only with one w |
| 03:29.16 | p3nguin_ | Invalid timeout? What generated that message? |
| 03:36.44 | acxty | p3nguin_, I did it that wait now when you dial it doesn't give tone and then it cut the call |
| 03:39.28 | acxty | and said status 'NO ANSWER' |
| 04:08.01 | *** join/#asterisk Goshen (~Goshen@c-98-202-22-89.hsd1.ut.comcast.net) |
| 04:38.26 | *** join/#asterisk Pk408 (~PsychoKlo@unaffiliated/pk408) |
| 04:39.48 | Pk408 | Hey Everyone. Was wondering if someone could help me with a sound playback issue. I'm getting "Unable to find a codec translation path from ulaw to gsm" when sounds should be playing. |
| 04:53.27 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 05:09.44 | *** join/#asterisk gamedna (~Adium@cpe-70-125-155-74.satx.res.rr.com) |
| 05:11.42 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 05:13.02 | gamedna | i am playing w/ the latest SVN asterisk 1.8 on OS X and am having problems w/ SIP Devices not having audio. Call is established, but no audio is coming inbound or outbound (have not done packet logging yet). IAX devices / connections do not have any problems, and audio plays fine w/ no jitter or lag. Any ideas? |
| 05:31.42 | *** join/#asterisk netmax (~netmax@is.linux-administrator.com) |
| 05:32.10 | ChannelZ | routing/nat/firewall issues |
| 05:33.30 | *** join/#asterisk coppice (~chatzilla@245.168.17.210.dyn.pacific.net.hk) |
| 05:33.56 | ChannelZ | Pk408: what does 'core show translation' say |
| 05:34.48 | Pk408 | For gsm it just shows -'s |
| 05:44.05 | *** join/#asterisk StompeR (~stomper@75-175-226-180.hlna.qwest.net) |
| 05:45.10 | StompeR | Good Evening, quick question - is there a dialplan command that i can use to detect asterisk version so i can have one dialplan with the correct syntax to call a agi script - ex | vs , |
| 05:50.38 | ChannelZ | Pk408: strange.. |
| 05:50.44 | kaldemar | StompeR: core show function VERSION |
| 05:51.13 | StompeR | k |
| 05:51.24 | ChannelZ | Pk408: I think you need libgsm |
| 05:51.27 | Pk408 | ChannelZ: Is there a way to see if the codec is installed? |
| 05:51.43 | Pk408 | ChannelZ: Let me see if thats installed. |
| 05:52.19 | ChannelZ | perhaps not. hmm |
| 05:54.17 | Pk408 | ChannelZ: I think it might not be installed. Gotta ask the OSH team. |
| 05:55.43 | ChannelZ | It's interesting, I have libgsm installed but not the -dev package |
| 05:58.29 | *** join/#asterisk russ (foobar@ip70-176-251-1.ph.ph.cox.net) |
| 06:03.59 | *** join/#asterisk shamelessn00b (~chatzilla@58-65-172-114.nayatel.pk) |
| 06:06.25 | shamelessn00b | Hi, I wanted to use asterisk as a media gateway, I wanted to ask if I should set up the interconnecting asterisk boxes using IAX or SIP |
| 06:35.17 | gamedna | i use both, but for different applications⦠each has its advantages / dissadvantages. I find that IAX handles NAT slightly better when dealing w/ only a few channels over a trunk. Sip generally has less overhead. |
| 06:36.02 | gamedna | shamelessn00b are you on a local lan? |
| 06:36.10 | gamedna | for the gateways? |
| 06:40.16 | shamelessn00b | gamedna: local lan |
| 06:41.37 | shamelessn00b | We will terminate E1s on one of our servers and then based on what service was requested, we would be forwarding the request to the server(s) the requested service is hosted on, so if one of the services goes down the others are not effected |
| 06:48.42 | *** join/#asterisk raden (~J@66-168-15-100.dhcp.stpt.wi.charter.com) |
| 07:05.13 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 07:07.49 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 07:09.16 | *** join/#asterisk nextime (~nextime@unaffiliated/nextime) |
| 07:11.14 | nextime | hello. I'm experimenting some trubble in dtmf detection with an incoming SIP call from my voip provider in a Read() application. every 10 digits i "lose" 1 or 2 digits, using both g729 and g711 and dtmf mode rfc8733 |
| 07:12.10 | nextime | using wireshark i can see that all digits are coming right from my provider, and they are coming out of band in rfc8733 as expected |
| 07:12.21 | nextime | i'm using latest asterisk 1.6.10 |
| 07:12.43 | nextime | any idea on what i can do to better match dtmf digits? |
| 07:55.23 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 08:00.25 | russ | is there any plan to turn off the dahdi timing source while no calls are in progress? |
| 08:02.41 | russ | it be nice for those of us running smaller services, it'd allow the system to save power |
| 08:03.21 | russ | ...er...wtf, right now I'm getting 1000 interrupts per/sec from the dahdi card...but powertop is telling me that asterisk is waking up 18690 times per second |
| 08:03.26 | russ | I don't even have any active channels |
| 08:05.24 | *** join/#asterisk g_r_eek (~g_r_eek@ppp-94-67-66-10.home.otenet.gr) |
| 08:06.15 | *** join/#asterisk FILLVAIO3 (~v_agarkov@79.165.89.20) |
| 08:09.52 | russ | I think powertop is on crack on that asterisk thing |
| 08:10.32 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 08:11.24 | FILLVAIO3 | Hello guys! Is there possible using "Fax For Asterisk" for sending/receiving faxes via simple fax phone connected throught AudioCodes MP-202 VoIP gateway? |
| 08:12.06 | russ | I'm pretty sure I can tell you the answer is no, even though I don't know what it is |
| 08:12.20 | russ | its one of those, if you have to ask, it doesn't work type of scenerios |
| 08:17.20 | russ | ok, yes, asterisk is going nuts on timer interrupts and is related to chan_alsa somehow interacting with the alsa pc speaker driver |
| 08:18.03 | russ | *sigh* 'default:CARD=pcsp' |
| 08:24.12 | *** part/#asterisk gamedna (~Adium@cpe-70-125-155-74.satx.res.rr.com) |
| 08:37.26 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 08:40.52 | *** join/#asterisk Shlomo1632 (~roger@70.94.30.225) |
| 08:41.17 | Shlomo1632 | how is it going everyone? |
| 08:42.07 | ChannelZ | Fantasmically |
| 08:43.15 | Shlomo1632 | I have a asterisk prob that is kicking my butt..... I'm using an AsteriskNow platform with FreePBX 2.8...... inbound works great.... but everytime I dial outside on my sip trunk, it seems to bounce back to my local system |
| 08:43.38 | Shlomo1632 | == Using UDPTL CoS mark 5 |
| 08:43.38 | Shlomo1632 | [Feb 26 05:25:39] VERBOSE[2952] app_dial.c: -- Called MAIN/8162556823 |
| 08:43.38 | Shlomo1632 | [Feb 26 05:25:39] VERBOSE[2952] app_dial.c: -- Now forwarding SIP/101-0000000d to 'Local/8162556823@from-trunk-sip-MAIN' (thanks to SIP/MAIN-0000000e) |
| 08:43.38 | Shlomo1632 | [Feb 26 05:25:39] VERBOSE[2953] pbx.c: -- Executing [8162556823@from-trunk-sip-MAIN:1] Set("Local/8162556823@from-trunk-sip-MAIN-09a9;2", "GROUP()=OUT_1") in new stack |
| 08:43.38 | Shlomo1632 | [Feb 26 05:25:39] VERBOSE[2953] pbx.c: -- Executing [8162556823@from-trunk-sip-MAIN:2] Goto("Local/8162556823@from-trunk-sip-MAIN-09a9;2", "from-trunk,8162556823,1") in new stack |
| 08:44.00 | Shlomo1632 | why is it calling an external number (LOCAL) ? |
| 08:44.42 | ChannelZ | #freepbx |
| 08:44.56 | Shlomo1632 | think its a freepbx issue? |
| 08:45.02 | Shlomo1632 | config thing? |
| 08:45.11 | ChannelZ | well it's obviously setup some wierd route |
| 08:45.28 | Shlomo1632 | yea i never seen this before |
| 08:46.00 | Shlomo1632 | it the most basic setup ever.... I am converting a load of thin clients into soho voip servers |
| 08:46.56 | ChannelZ | I don't know what 'MAIN' is |
| 08:47.09 | Shlomo1632 | main is the only SIP trunk |
| 08:48.06 | Shlomo1632 | it comes back and the any DID would catch it ( when i had it enabled ) and ring the phone i was calling on |
| 08:48.12 | ChannelZ | so you dial any number out and it dials back into your incoming? Is that what 8162556823 is? |
| 08:48.54 | Shlomo1632 | yea the 816255623 is just a outside number.. but it shoots it right back to the local asterisk |
| 08:49.55 | Shlomo1632 | asterisk doesn't match the DID..so it shoots it to the any did, rings back on the originating phone... strange stuff |
| 08:50.10 | Shlomo1632 | like it hits a mirror |
| 08:50.15 | Shlomo1632 | incoming calls are fine |
| 08:50.23 | Shlomo1632 | using VOIPINNOVATIONS |
| 08:51.03 | ChannelZ | No idea, something is configged wrong but I don't use freepbx |
| 08:51.16 | Shlomo1632 | the (LOCAL) prefix tells it to route locally right? |
| 08:51.44 | Shlomo1632 | 'Local/8162556823@from-trunk-sip-MAIN' |
| 08:52.34 | ChannelZ | Local channels are virtual channels that let you dial back into the dialplan |
| 08:53.13 | ChannelZ | so it's trying extension 8162556823 in the context 'from-trunk-sip-MAIN' |
| 09:30.27 | *** join/#asterisk Z_God (~julius@2001:888:141f:0:221:5dff:fe2a:6806) |
| 09:47.49 | *** join/#asterisk shamelessn00b (~chatzilla@58-65-172-114.nayatel.pk) |
| 09:47.53 | shamelessn00b | Can anyone help me out on whether I should go with SIP trunk or IAX2? |
| 10:05.30 | ChannelZ | iax trunking is apparently better with bandwidth with lots of calls than SIP |
| 10:07.24 | shamelessn00b | hmm, but I'm reading on various places on the internet, its mentioned that under heavy call loads, IAX trunks experiance congestion |
| 10:08.29 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 10:12.28 | ChannelZ | fine use SIP then! jesus |
| 10:12.48 | ChannelZ | see if it works for your configuration. |
| 10:12.59 | shamelessn00b | so is it a matter of trial and error |
| 10:13.17 | ChannelZ | well everything you read on the internet is true you know |
| 10:13.27 | shamelessn00b | lol |
| 10:13.35 | shamelessn00b | Thats why I'm asking |
| 10:13.47 | shamelessn00b | otherwise I'd just go with whatever was written |
| 10:14.37 | xheliox | Ask 5 people, you'll probably get 5 opinions. SIP is going to be more compatible long term with other clients/phone systems. My preference would be SIP. |
| 10:15.14 | shamelessn00b | Well the scenario is something like this |
| 10:15.34 | shamelessn00b | I'm using SS7, we have 48 E-1s terminating on our asterisk servers |
| 10:16.12 | shamelessn00b | multiple short codes for various services, right now, whats happening is that if 1 service goes down, it effects the rest too |
| 10:17.06 | shamelessn00b | so I want to isolate the services from each other, I want to accomplish this by terminating all E1s on a single asterisk box, and then have independant servers connected to that box using SIP/IAX trunks |
| 10:17.42 | shamelessn00b | call lands on the media gateway and it forwards it to the server thats hosting the service, they are all going to be on the same network, connected using the same switch |
| 10:29.35 | *** join/#asterisk cdahmedeh (~cdahmedeh@CPE000c41445085-CM001225409602.cpe.net.cable.rogers.com) |
| 10:40.44 | *** join/#asterisk Beltechs (~Beltechs@208.127.3.20) |
| 10:41.00 | Beltechs | Hello, ive google and cant seem where to find chan_sip.c so I can edit it. Using AsteriskNow |
| 10:42.00 | shamelessn00b | Beltechs: you'll need to download the asterisk source |
| 10:43.05 | *** join/#asterisk defsdoor (~andy@plingit.gotadsl.co.uk) |
| 10:43.40 | Beltechs | like svn or something? |
| 10:43.43 | *** join/#asterisk kuttan_ (~ops@114.79.141.38) |
| 10:43.56 | *** join/#asterisk cdahmedeh (~cdahmedeh@CPE000c41445085-CM001225409602.cpe.net.cable.rogers.com) |
| 10:44.22 | Beltechs | so when I installed my *now the installation did not download the source? |
| 10:44.28 | Beltechs | is that correct? |
| 10:45.43 | shamelessn00b | Beltechs: get the source tarball from the asterisk download page |
| 10:47.14 | Beltechs | im looking there now |
| 11:02.30 | *** join/#asterisk cdahmedeh (~cdahmedeh@99.241.40.92) |
| 11:04.06 | Beltechs | shamelessn00b: downloaded the tarball and I found the file in question but how or where does it belong in asterisk? |
| 11:04.45 | shamelessn00b | you make the chages in the file |
| 11:04.52 | shamelessn00b | then you run the configure shell script |
| 11:05.00 | shamelessn00b | followed by make, make install |
| 11:05.36 | shamelessn00b | an object file will be created in the asterisk/modules directory |
| 11:05.56 | shamelessn00b | chan_sip.so I guess |
| 11:11.52 | *** join/#asterisk Tim_Toady (~moi@77.49.3.102.dsl.dyn.forthnet.gr) |
| 11:11.57 | Beltechs | wowsers let me see if I can figure that out |
| 11:14.20 | Beltechs | do I place the file in the * directory after I make the changes? |
| 11:14.34 | shamelessn00b | after you run make install |
| 11:14.52 | shamelessn00b | the directory will automatically be created and files will automatically be placed there |
| 11:15.10 | shamelessn00b | have you never built * from source? |
| 11:15.19 | Beltechs | never |
| 11:15.24 | Beltechs | im lost |
| 11:15.37 | shamelessn00b | did you make the chage |
| 11:15.55 | shamelessn00b | change? |
| 11:16.07 | Beltechs | not yet not sure what I have to change yet still parsing the filr |
| 11:16.10 | Beltechs | file |
| 11:16.31 | Beltechs | got a registration error |
| 11:17.17 | shamelessn00b | registration error? |
| 11:17.38 | shamelessn00b | what were you intending to do |
| 11:18.05 | shamelessn00b | were you trying to register a sip client with * |
| 11:18.07 | shamelessn00b | ? |
| 11:18.15 | Beltechs | NOTICE[7508] chan_sip.c: -- Registration for 'EXXXXXXXXXXX01@proxy01.losangeles.talk4free.com' timed out, trying again |
| 11:19.05 | Beltechs | http://www.digitaloffensive.com/2010/03/hacking-the-magic-jack-in-2010-for-use-on-trixbox-or-any-other-sip-device/ |
| 11:19.28 | xheliox | Oy vey. |
| 11:19.33 | *** join/#asterisk RypPn (~TuMbL@rosscom.co.uk) |
| 11:20.30 | Beltechs | is there and existing chan_sip.c file already in my system? |
| 11:22.44 | Beltechs | what a way to freak everybody out |
| 11:42.27 | *** part/#asterisk kuttan_ (~ops@114.79.141.38) |
| 11:59.57 | *** join/#asterisk Thorn (~thorn@unaffiliated/thorn) |
| 12:23.45 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 12:54.46 | *** join/#asterisk riddlebox (~james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 12:58.32 | riddlebox | does anyone use the linksys spa921 phones, and if so, do you like them? |
| 13:06.03 | *** join/#asterisk nettie (~nettie@stewie.freax.it) |
| 13:08.14 | nettie | Hi guys, anyone using sangoma cards please? I can only hear noise instead of clean audio. |
| 13:11.23 | *** join/#asterisk Trixboxer (~Trixboxer@office.supportdepartment.net) |
| 13:14.13 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
| 13:31.12 | *** join/#asterisk guilhermebr (~Guilherme@189.27.62.155.dynamic.adsl.gvt.net.br) |
| 13:43.16 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 14:02.19 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
| 14:30.13 | drmessano | HAH, wow |
| 14:30.16 | drmessano | http://www.magicjacksupport.com/post52672.html#52672 |
| 14:30.44 | drmessano | Extensions.conf deprecated? lol |
| 14:41.31 | p3nguin_ | "Use extensions.ael and extensions.lua. Delete extenisons.conf. DON'T USE IT!" How is there any functional difference between using one extensions file format and another? |
| 14:42.03 | drmessano | IIRC, extensions.ael doesn't support everything extensions.conf does. |
| 14:42.12 | drmessano | It's just different |
| 14:42.21 | drmessano | and this guy sounds like he's just being a major blowhard |
| 14:42.37 | p3nguin_ | So on a technical level, the ael format is actually worse? |
| 14:43.08 | p3nguin_ | I've never had a need to learn how to use the ael stuff. |
| 14:43.46 | drmessano | From the comments I have seen on extensions.ael, it serves its purpose, but is not a replacement |
| 14:43.58 | drmessano | I've been doing this a long time, and I know exactly what I'm doing and I am exactly right. You won't and can't prove me wrong. Forgive me for trying to steer some of you into smarter not harder thinking. |
| 14:44.03 | drmessano | ^^^ lol |
| 14:48.17 | p3nguin_ | There's another one saying the same stuff, too. "extensions.conf has been deprecated since asterisk 1.4.x It has been retained for backward compatibility only. Extensions.ael is the preferred dialplan with extensions.lua as an adjunct where asterisk is compiled with lua." |
| 14:57.02 | drmessano | Yeah |
| 14:57.17 | drmessano | Some poeple should be shot |
| 14:58.49 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 15:13.38 | *** join/#asterisk fenlander (~fenlander@82.152.81.57) |
| 15:15.17 | *** join/#asterisk fenlander (~fenlander@82.152.81.57) |
| 15:19.06 | TJNII | "Dan isn't smart enough to hire me" <-- What a dbag. |
| 15:29.16 | *** join/#asterisk Mhaddog_ (~Mhaddog@z65-50-118-232.ips.direcpath.com) |
| 15:36.20 | *** join/#asterisk nicoAMG (~nicoamg@201.237.49.131) |
| 15:41.30 | *** join/#asterisk lhz (~shrekz@c-dba672d5.021-158-73746f34.cust.bredbandsbolaget.se) |
| 15:42.30 | drmessano | TJNII: I almost want to register there and put him in his place |
| 15:45.06 | *** join/#asterisk andresmujica (~andresmuj@ubuntu/member/andresmujica) |
| 15:46.10 | drmessano | http://xkcd.com/386/ <-- Someone is wrong on the internet! |
| 15:53.33 | TJNII | Eh, wrong on the internet doesn't bug me as much as when I meet them in real life. Like when a coworker tried to tell me you could run a 120V incandescent lamp off 220V and how it was perfectly safe to run a clothes dryer on a standard US 120V outlet. |
| 15:53.40 | TJNII | "hahaha...ha.......What?" |
| 15:57.19 | *** join/#asterisk andresmujica (~andresmuj@ubuntu/member/andresmujica) |
| 16:06.37 | *** join/#asterisk BugKhaM (~BugKhaM@125.27.44.109.adsl.dynamic.totbb.net) |
| 16:07.59 | BugKhaM | I currently have 2052 active SIP channels from an IP 10.1.1.118 which isn't in my local network |
| 16:08.23 | BugKhaM | anyway to get block these sip requests? |
| 16:09.21 | TJNII | Open ... as in active calls? |
| 16:09.45 | TJNII | That's not a routable IP (RFC 1918) Is it just on a nother subnet of your private network? |
| 16:12.14 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 16:13.01 | BugKhaM | TJNII, I only have 1 network interface and it's set to dhcp |
| 16:13.27 | TJNII | Home user? |
| 16:13.31 | BugKhaM | ifconfig -a doesn't show an ip in that range either |
| 16:13.56 | TJNII | What is the IP address and netmask of the server? |
| 16:13.56 | BugKhaM | nope the IP I got is a public IP |
| 16:14.27 | BugKhaM | it's 255.255.254.0 |
| 16:14.51 | BugKhaM | and my ip is in the 119.171.x.x range |
| 16:15.35 | TJNII | So you have 2052 active calls from that IP? |
| 16:15.38 | p3nguin_ | It's not really that unusual to see 10/8 addresses on your WAN interface. |
| 16:16.32 | BugKhaM | p3nguin_, meaning it's common? |
| 16:16.35 | TJNII | p3nguin_: Oh? If he has a public IP from his ISP why would he be seeing requests from RFC 1918 addresses? |
| 16:17.19 | p3nguin_ | Cable networks have those addresses floating around at all times. |
| 16:18.10 | TJNII | BugKhaM: Did you buy a public, static IP or is that just the address you get? |
| 16:18.17 | BugKhaM | p3nguin_, can I block them using iptables? |
| 16:18.23 | p3nguin_ | sure you can. |
| 16:18.37 | TJNII | BugKhaM: So you have 2052 active calls from that IP? <- Please answer this question |
| 16:20.03 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 16:20.48 | BugKhaM | TJNI, it's 2003 currently |
| 16:20.53 | *** join/#asterisk fenlander (~fenlander@82.152.81.57) |
| 16:21.18 | TJNII | BugKhaM: But active, open, calls, correct? |
| 16:21.35 | BugKhaM | another 49 belongs to other IPs |
| 16:21.45 | TJNII | If so, then you need to secure your asterisk setup. |
| 16:22.22 | BugKhaM | they are all like "10.1.1.118 (None) 251541611 00101/00001 0x0 (nothing) No Rx: REGISTER" |
| 16:22.23 | TJNII | Since if anyone can make calls through your box you have a gaping security hole where IPtables will just ba a band-aid. |
| 16:22.34 | *** join/#asterisk Tim_Toady (~moi@77.49.3.102.dsl.dyn.forthnet.gr) |
| 16:22.40 | BugKhaM | I mean all -> Rx: REGISTER |
| 16:22.41 | TJNII | Oh, so its just registration attempts? |
| 16:22.57 | BugKhaM | yes, it looks like that |
| 16:23.52 | TJNII | Well, that's not a cause for panic then. Black hats scan for open VoIP systems frequently. What I do is restrict my VoIP ports to only allowed subnets vis IPtables. |
| 16:23.54 | BugKhaM | or I better run sip debug to see |
| 16:24.08 | BugKhaM | but that's too many |
| 16:25.08 | BugKhaM | TJNII, OK, will try that with iptables first |
| 16:25.16 | TJNII | You can use fail2ban, too. |
| 16:25.51 | TJNII | goes to the bank |
| 16:26.19 | p3nguin_ | Remember, I said small unmarked bills. |
| 16:27.06 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 16:28.28 | BugKhaM | TJNII, i see |
| 16:30.04 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 16:31.09 | TJNII | p3nguin_: Oh, right. Harder to track when I buy the claw hammer with fingerprint resistant grip, spade, quicklime, and roll of heavy plastic. You said to meet in the dark alley behind where you work, right? |
| 16:37.22 | BugKhaM | TJNII, i did restart iptables, sip reload but they are still active |
| 16:37.38 | BugKhaM | TJNII, sip debug shows SIP/2.0 401 Unauthorized |
| 16:37.38 | BugKhaM | Via: SIP/2.0/UDP 10.1.1.118:5060;branch=z9hG4bK-369678036;received=209.251.21.170;rport=32399 |
| 16:38.45 | BugKhaM | TJNII, 209.251.21.170 is the actual source address? |
| 16:39.05 | [TK]D-Fender | BugKhaM: You're missing the big print before that |
| 16:39.18 | [TK]D-Fender | BugKhaM: <----- SIP read from |
| 16:41.04 | BugKhaM | [TK]D-Fender, yeah it's <--- SIP read from 209.251.21.170:32399 ---> |
| 16:42.54 | *** join/#asterisk mlarsen (~mlarsen@0x57370318.esnqu1.dynamic.dsl.tele.dk) |
| 16:44.21 | BugKhaM | dropping 209.251.21.170's request in iptables seems to help |
| 16:45.27 | BugKhaM | and my cpu usage for asterisk dropped from 80% to 20% |
| 16:48.11 | BugKhaM | 209.251.21.170 is not from my ISP actually, it's from the US |
| 17:00.07 | *** join/#asterisk devmod (~devmod@c-76-100-208-204.hsd1.md.comcast.net) |
| 17:08.01 | *** join/#asterisk cesar_CR (~cesar@201.201.41.242) |
| 17:20.31 | *** join/#asterisk Alton35 (~alton@69.45.116.128) |
| 17:55.17 | *** join/#asterisk cdahmedeh (~cdahmedeh@CPE001cdfab341a-CM001225409602.cpe.net.cable.rogers.com) |
| 18:06.19 | *** join/#asterisk cdahmedeh (~cdahmedeh@CPE001cdfab341a-CM001225409602.cpe.net.cable.rogers.com) |
| 18:14.40 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 18:52.00 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 18:54.55 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 18:58.14 | *** join/#asterisk ChannelZ (~bobm@burner.com) |
| 18:58.43 | *** join/#asterisk [netman] (~netman@83.55.242.113) |
| 18:58.53 | *** join/#asterisk guilhermebr (~Guilherme@189.114.45.141) |
| 19:06.59 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 19:15.35 | *** join/#asterisk devmod (~devmod@c-76-100-208-204.hsd1.md.comcast.net) |
| 19:31.12 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-139-208.ks.ks.cox.net) |
| 19:38.37 | *** join/#asterisk flimmern (~se@f054151171.adsl.alicedsl.de) |
| 19:38.39 | *** join/#asterisk Goshen (~Goshen@c-98-202-22-89.hsd1.ut.comcast.net) |
| 19:39.54 | *** join/#asterisk Goshen (~Goshen@c-98-202-22-89.hsd1.ut.comcast.net) |
| 19:45.26 | *** join/#asterisk shido6 (~shido6@c-24-4-217-126.hsd1.ca.comcast.net) |
| 19:49.37 | *** join/#asterisk astrutt (astrutt@pinky.ratman.org) |
| 19:50.00 | *** part/#asterisk flimmern (~se@f054151171.adsl.alicedsl.de) |
| 19:55.03 | *** join/#asterisk andresmujica (~andresmuj@ubuntu/member/andresmujica) |
| 20:02.02 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 20:04.24 | *** join/#asterisk irule (~irule@187.140.137.110) |
| 20:07.16 | irule | I used voxilla's configuration wizzard for spa3102, fxo is registered, but fxs is not registered, how may I troubleshoot it? |
| 20:10.43 | ChannelZ | Well it's everything on the Line 1 tab |
| 20:20.28 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 20:28.39 | *** join/#asterisk irule (~irule@187.140.137.110) |
| 20:36.45 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 20:51.57 | *** join/#asterisk chazzam (~chazz@173-24-237-15.client.mchsi.com) |
| 20:57.25 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 21:07.18 | *** join/#asterisk Aaron___ (~chatzilla@cpe-66-74-202-161.san.res.rr.com) |
| 21:07.32 | Aaron___ | hi guys |
| 21:08.13 | Aaron___ | I just bought a Digium 400 card and I have a single FXO module installed. I'm not sure what I need to do to get it running. |
| 21:10.32 | Aaron___ | anybody home? |
| 21:22.07 | *** join/#asterisk grungies1138 (~grungies1@cpe-66-74-202-161.san.res.rr.com) |
| 21:22.21 | grungies1138 | hey guys, long time no chat |
| 21:23.32 | *** join/#asterisk Gershwin (~noone@unaffiliated/gershwin) |
| 21:28.45 | *** join/#asterisk unspin (~unspin@S0106001451226d9c.vc.shawcable.net) |
| 21:38.28 | [TK]D-Fender | grungies1138: Feels you only came & went 25 minutes ago |
| 21:48.26 | grungies1138 | hmm? |
| 21:54.28 | [TK]D-Fender | [17:07]-->|Aaron___ (~chatzilla@cpe-66-74-202-161.san.res.rr.com) has joined #asterisk |
| 21:54.35 | [TK]D-Fender | [17:22]-->|grungies1138 (~grungies1@cpe-66-74-202-161.san.res.rr.com) has joined #asterisk |
| 21:54.50 | [TK]D-Fender | No, we don't notice hostnames here |
| 21:55.13 | grungies1138 | yeah. heh. not like i was hiding. just fixed it |
| 21:55.58 | Gershwin | anyone here have experience w/screenos (i.e., juniper)? |
| 21:56.00 | *** join/#asterisk ChannelZ (~bobm@burner.com) |
| 21:59.57 | grungies1138 | so anyway how do i set up a digium analog card? |
| 22:00.35 | grungies1138 | what file do I need to look at? |
| 22:00.47 | ChannelZ | did you build dahdi? |
| 22:01.12 | grungies1138 | build? |
| 22:01.18 | ChannelZ | compile |
| 22:01.19 | ChannelZ | install |
| 22:01.35 | grungies1138 | hmm. i used apt-get |
| 22:01.56 | ChannelZ | well that's probably fine |
| 22:02.07 | ChannelZ | so you need to config the hardware in /etc/dahdi/system.conf |
| 22:02.46 | ChannelZ | basically to tell it which channels are fxo, which are fxs.. |
| 22:02.52 | raden | I have 2 asterisk boxes in round robin for our customers to register to sharing mysql database...... when our ITSP sends our servers calls is there a way to tell ITSP hey send to box 1 thats where the phone registered ? |
| 22:03.03 | ChannelZ | then you can configure /etc/asterisk/chan_dahdi.conf to consume those channels in Asterisk |
| 22:04.01 | raden | cause right now I have to do this customer 1,3,5,7 calls always get sent to box 1 where customer 2,4,6,8 calls always go to box 2 anf for failover if 2 not available them calls go to 1 1 not available them calls go to 2 |
| 22:05.03 | *** join/#asterisk russ (foobar@ip70-176-251-1.ph.ph.cox.net) |
| 22:10.41 | raden | ???? |
| 22:10.43 | raden | no one ? |
| 22:11.06 | raden | does someone sit on the internet all freaking day and buy all the domain names that are decent ? |
| 22:11.52 | WIMPy | has heard of a company, trying to register an entire dictionary. |
| 22:11.52 | ChannelZ | yes |
| 22:12.20 | ChannelZ | pissguzzlinggrannies.com is still open |
| 22:16.38 | *** join/#asterisk xheliox (jeff@heliosj.iddings.us) |
| 22:20.43 | raden | trying to get a name for ITSP and everything is literally taken |
| 22:20.52 | raden | even when i mispell shit it taken |
| 22:21.00 | raden | any answers to my other question ? |
| 22:22.23 | [TK]D-Fender | raden: The answer is "no". |
| 22:22.57 | nextime | Hello. Anyone have an idea on what to do with this issue: asterisk latest 1.6.10 from sources. SIP channel, incoming call, with both g729 or g711 codecs, rfc2833 dtmftype, no fax detection, not jbenable on chan_sip. Sometime i lose some dtmf digits ( about 10% of digits ) |
| 22:23.19 | nextime | looking with wireshark i can see all dtmf events coming in the right way from the voip provider |
| 22:23.40 | nextime | what can cause to * to not match all dtmf in the right way and lose some digits? |
| 22:24.17 | nextime | ( i get digits with Read() on my dialplan ) |
| 22:24.41 | jql | well, the words "tele" and "voip" and "phone" and "fone" are right out, when it comes to domain names |
| 22:25.49 | jql | nextime: define "lose" |
| 22:26.07 | raden | [TK]D-Fender, what do companies do that need redundancy ? |
| 22:26.07 | [TK]D-Fender | nextime: So You can see the RFC2833 packets in but not ack'd by Read? |
| 22:26.13 | jql | nextime: when you turn on debug and verbose logging, do they not even show up in the asterisk/full log? |
| 22:26.24 | raden | do what im doing ? |
| 22:26.30 | nextime | jql : if the remote part send, for example, 345678, from Read i see only 35678 coming |
| 22:26.33 | jql | cause that's what I would categorize as really "lost" |
| 22:26.51 | nextime | jql : yep, they not even show up |
| 22:26.58 | nextime | [TK]D-Fender : exactly |
| 22:27.00 | [TK]D-Fender | raden: You aren't doing "redundency", you want your ITSP to send calls to the server your CLIENT is reg'd to. You are sitting in between. You reg to the provider. ther is no "send calls there later. |
| 22:27.13 | [TK]D-Fender | raden: the * that reg's is the one that gets the calls. |
| 22:27.13 | jql | is an rfc2833 dtmf digit is not even logged under the debug log, you're f'd |
| 22:27.29 | jql | if you do a wireshark trace, is it in there? |
| 22:27.31 | [TK]D-Fender | raden: And you aren't even dealing with redundency, you are looking to selective route. |
| 22:27.35 | nextime | jql : yes |
| 22:27.45 | jql | okay, you're double f'd |
| 22:28.30 | nextime | jql : ok, now i know that i'm double f'd. Now, what can i try to solve? |
| 22:28.31 | nextime | :D |
| 22:28.57 | [TK]D-Fender | nextime: change providers |
| 22:28.57 | jql | reproduce the issue with sip trace, debug 10, and verbose 10, and a full wireshark capture |
| 22:29.02 | jql | send the results to digium |
| 22:29.05 | jql | cause that ain't right |
| 22:29.06 | nextime | [TK]D-Fender : this isn't an option... |
| 22:29.13 | [TK]D-Fender | nextime: THEY are the ones not sending them to you. You can do nothing about this |
| 22:29.38 | jql | if the digit shows up in wireshark but not in the asterisk log, that's bugworthy |
| 22:29.39 | nextime | [TK]D-Fender : they ARE sending it as i can see it in the wireshark trace |
| 22:30.01 | [TK]D-Fender | [18:26]<nextime>jql : yep, they not even show up |
| 22:30.17 | nextime | [TK]D-Fender : it was referred to the debug log of * |
| 22:30.38 | [TK]D-Fender | nextime: Enable core debug and look closer |
| 22:31.21 | nextime | [TK]D-Fender : i can't see the "lost" digits at all in the debug of * |
| 22:32.17 | nextime | anyway, in 10 mins i will produce a log to better explain |
| 22:35.52 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 22:38.34 | *** join/#asterisk aurix (~aurelio@81.174.13.196) |
| 22:39.01 | nextime | uhmm |
| 22:39.07 | nextime | core debug say |
| 22:39.08 | nextime | Aug 8 00:38:24] DEBUG[20839]: rtp.c:876 create_dtmf_frame: Ignore potential DTMF echo from '82.145.170.244' |
| 22:39.14 | nextime | when it lose digits |
| 22:44.52 | raden | [TK]D-Fender, so whatever box gets the call should just let it send to the phone ? |
| 22:45.34 | nextime | mumble |
| 22:46.03 | nextime | it seem that i can try to solve by set rtp->dtmfmute to a smaller amount of time in rtp.c and recompiling |
| 22:46.08 | nextime | try |
| 22:47.25 | jql | umm... how are you getting two digits so close together? |
| 22:47.37 | jql | is the clock on your machine stable? |
| 22:47.57 | nextime | jql : yes, it is |
| 22:48.30 | nextime | 500 ms isn't so short, it's 1/2 second! |
| 23:11.23 | nextime | ok |
| 23:11.30 | nextime | with 300ms instead of 500 |
| 23:11.33 | nextime | it work good |
| 23:14.06 | WIMPy | I'm pretty sure it's easy to type more than three digits per seconds. |
| 23:17.21 | nextime | WIMPy : the default time is 500ms, i think that reducing it to 300ms is ok |
| 23:20.06 | WIMPy | Got that, but looks like broken design to me. |
| 23:20.41 | nextime | WIMPy: for sure i was more happy to have this timing configurable in a config file rather than need to recompile |
| 23:20.55 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 23:21.33 | nextime | now i have another issue on dahdi, but i will look at it tomorrow, now are 1:21am here |
| 23:21.36 | nextime | :) |
| 23:23.09 | WIMPy | Same TZ then :-) |
| 23:23.35 | nextime | yep, you are german, i'm italian :) |
| 23:27.49 | *** join/#asterisk cdahmedeh (~cdahmedeh@CPE001cdfab341a-CM001225409602.cpe.net.cable.rogers.com) |
| 23:31.22 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 23:37.29 | raden | Im really beginning to believe this is useless finding a .com domain thats not 90 charachters long |
| 23:37.47 | nextime | isn't capable to leave the * server not fully working |
| 23:37.50 | nextime | i'm watching the debug in dmesg of the dahdi modules |
| 23:38.55 | nextime | mumble |
| 23:39.15 | nextime | when i load dahdi and wct4xxp modules |
| 23:39.41 | nextime | i soon get an wct4xxp 0000:01:09.0: Detected loss of E1 alignment on span N! debug |
| 23:40.06 | nextime | after that, no more interrupts on the pri card |
| 23:40.21 | nextime | so, my 4 pri aren't working |
| 23:40.38 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 23:45.44 | nextime | mumble |
| 23:46.05 | nextime | when dahdi detect a loss of E1 alignment call t4_reset_dma |
| 23:46.13 | nextime | it should stop and restart dma |
| 23:46.56 | nextime | but it seem than it stop it and doesn't turn it on as i get no more interrupts |
| 23:49.09 | *** join/#asterisk rdahlin_1 (~rdahlin_1@78-73-19-238-no168.tbcn.telia.com) |
| 23:51.25 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |