00:00.50 | tzanger | I did notice on my own phone that it tries for the better part of an hour to contact the boot server (in a loop) until finally giving up and using whatever config it has on hand |
00:00.55 | tzanger | I'm running OLD ip501 firmware though |
00:01.29 | tzanger | sorry ip601 and sip firmware 1.6.7 |
00:03.02 | [TK]D-Fender | [18:37][TK]D-FenderSo make a new provisioning folder and dump in stock sample configs & firwmware for it to pick up [18:37]Get_The_FishI cant do that. [18:38]Get_The_FishBecause I dont have unfettered access to this entire subnet, and since it's looking for a boot server on a 10.x.x.x network, and is now on a 192.168 subnet, it fails. |
00:03.22 | [TK]D-Fender | He's saying up-front that he doesn't have the means of following advice. |
00:03.30 | tzanger | yeah, same idea as my phone |
00:03.50 | tzanger | it moved from the house to the shop and can only use its internal config until I get off my ass to change things around |
00:03.51 | [TK]D-Fender | [18:41]Get_The_FishNot to mention the fact that it is so old I wouldnt even know which firmware to use on it. |
00:04.28 | [TK]D-Fender | And seems to confirm that he can't read docs on what it even supports. He simply appears to have no clue what it runs, no means of doing anything |
00:04.38 | [TK]D-Fender | And *I* get crap from him. |
00:06.05 | tzanger | you're to blame, don't you know? :-) |
00:06.15 | [TK]D-Fender | naturally. |
00:07.06 | [TK]D-Fender | http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html |
00:07.09 | [TK]D-Fender | ^ |
00:07.21 | [TK]D-Fender | Cause yeah ... RAW-CAT SIGH HENCE |
00:09.11 | [TK]D-Fender | tzanger: 1.6.7 was the one that fixed the presence issue with IP601's and side-cars,e tc.... something to that effect... |
00:09.21 | [TK]D-Fender | I remember that one. |
00:10.31 | tzanger | yeah. I have no sidecar but it was one of the first firmwares that did presence support, which I never did get around to using. :-) |
00:10.43 | [TK]D-Fender | tzanger: This past week brought my IP600's to SIP 3.1.8 (latest they support) which is from Mar 2012. |
00:11.04 | [TK]D-Fender | So I guess :not supported" depends on a certain point of view... |
00:11.25 | [TK]D-Fender | Whci isn't bad for 7 year old phones.... |
00:12.54 | tzanger | hm, maybe I should update |
00:13.13 | tzanger | the only thing I really wish the 501 had was backlight |
00:13.19 | tzanger | it's a pretty nice phone |
00:13.24 | tzanger | er 601 dammit I keep saying that |
00:15.16 | [TK]D-Fender | That's the reason I swapped for the IP335 I run now. |
00:15.41 | [TK]D-Fender | When I change offices in the next 2 months renovations I should end up in a room with good lighting and may switch back |
00:19.31 | rue_work | anyone ever used mgcp or am I the only one? |
00:19.37 | rue_work | ok I have two mgcp gateways that I'm trying to link calls between, but I cant get them both to work at the same time with asterisk |
00:19.46 | rue_work | I dont know if mgcp.conf is limited to one gateway |
00:19.53 | rue_work | I can get one to work |
00:20.02 | rue_work | and it can be either one |
00:20.10 | rue_work | but not both |
00:22.56 | [TK]D-Fender | http://www.quickmeme.com/meme/3sh6mz/ |
00:23.49 | *** join/#asterisk Sinnerman77 (~Sinnerman@174-21-246-21.tukw.qwest.net) |
00:30.32 | Sinnerman77 | Hi there. I'm running Elastix and using Flowroute as my SIP trunk. This is a new setup and I'm pretty much a newbie. I have the trunk setup according to instructions from Flowroute, an outbount route setup using the "local 7/10 digit" wizard and an extension. I have the IP address for the PBX set as allowed on Flowroute's website. When I try to dial out using X-Lite I'm getting "All circuits are busy now." Is there something eas |
00:32.15 | jacekowski | ryan42: i would use show command to make sure it's all loaded correctly |
00:32.27 | jacekowski | but it looks ok |
00:38.28 | [TK]D-Fender | Sinnerman77: PASTEBIN the failed call attempt @ * CLI |
00:38.30 | [TK]D-Fender | ~pb |
00:38.30 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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 install pastebinit with yum or aptitude. |
00:38.32 | [TK]D-Fender | ^^^ |
00:38.34 | [TK]D-Fender | ^^^ |
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00:45.52 | Sinnerman77 | [TK]D-Fender, thank you. I'll do that in a moment if I need to. I'm on the line with Freeroute support. He thinks it's my dial pattern that's the problem. |
00:48.14 | [TK]D-Fender | heads out for a while |
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03:38.04 | macroevolve | hi, i've been trying to find a softphone that supports Silk @ 8kbps. I've already tried Ekiga, Jitsi, Vireo, and Microsip, and none of them work well. Was wondering whether knew of any other softphone I could try? |
03:39.29 | jpsharp | That's not really a standard codec in internet telephony. |
03:39.54 | jpsharp | So you're going to have a PITA time finding something that supports it. |
03:40.11 | macroevolve | jpsharp: yeah - ive alreadd experienced that PITA part unfortunately |
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03:44.53 | Docfxit | I would like to record a voice prompt. I added extension 130 to extensions.conf. The lines I added are at pastebin.com/JJPYZwRx I have restarted Asterisk. It isn't working. Could someone please suggest what I need to do to get it working? |
03:46.43 | *** join/#asterisk gg608f (~Adium@187.207.6.250) |
03:47.05 | pabelanger | Docfxit: you recorded asterisk-Recording, but playback asterisk-recording |
03:47.43 | Docfxit | Great catch |
03:47.49 | Docfxit | I'll test it. |
03:49.00 | asr33 | shakes fist at uppercase |
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03:56.28 | leifmadsen | there's something to be said about using a channel variable there.... |
03:56.59 | leifmadsen | at line 4 I'd have placed a Set(thisRecordedFile=/tmp/asterisk-recording) or something |
03:57.11 | leifmadsen | then used ${thisRecordedFile}.gsm on the next line |
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04:18.14 | Docfxit | When trying to restart asterisk: |
04:18.14 | Docfxit | root@UbuntuAsterisk:~# asterisk -r |
04:18.14 | Docfxit | I get: |
04:18.14 | Docfxit | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
04:18.14 | Docfxit | If I input this: |
04:18.14 | Docfxit | root@UbuntuAsterisk:~# asterisk -vvvc |
04:18.16 | Docfxit | The errors I get are: |
04:18.18 | Docfxit | Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory |
04:18.20 | Docfxit | Unable to bind socket to /var/run/asterisk/asterisk.ctl: No such file or directory |
04:18.23 | Docfxit | [Jan 7 20:33:07] NOTICE[7261]: http.c:611 http_server_start: Unable to bind http server to 0.0.0.0:8088: Address already in use |
04:18.25 | Docfxit | [Jan 7 20:33:07] WARNING[7261]: manager.c:3159 init_manager: Unable to bind socket: Address already in use |
04:19.29 | Docfxit | Any idea what I can do to fix these errors? |
04:21.15 | ChannelZ | does /var/run exist? |
04:21.48 | Docfxit | Yes. |
04:23.52 | [TK]D-Fender | Do you see an Asterisk process actually running? |
04:23.57 | [TK]D-Fender | Can you find the PID somewhere else? |
04:24.12 | [TK]D-Fender | Did you look where whatever you used to init it chooses to put it? |
04:27.25 | Docfxit | Asterisk does run. It answers the phone, plays the prompts, transfers calls. |
04:28.39 | Docfxit | What command would I use to find asterisk.pid? |
04:29.41 | dpilon | do this: ls /var/run/asterisk/*.pid |
04:30.39 | Docfxit | No such file. |
04:31.38 | Docfxit | I found someone else with the same problem and they said SELINUX=disabled did the trick. |
04:31.58 | Docfxit | What is SELINUX=disabled? |
04:33.26 | [TK]D-Fender | Docfxit: look in var/run |
04:35.10 | Docfxit | In var/run/asterisk there is a file asterisk.ctl with zero bytes |
04:36.14 | Docfxit | In var/run there are a number of .pid but no asterisk.pid |
04:36.15 | [TK]D-Fender | I mean the base of it |
04:36.48 | [TK]D-Fender | who owns the PID's in /var/run/asterisk ? |
04:37.24 | [TK]D-Fender | srwxr-xr-x 1 asterisk asterisk 0 Dec 10 20:39 asterisk.ctl |
04:37.26 | [TK]D-Fender | -rw-r--r-- 1 asterisk asterisk 5 Dec 10 20:39 asterisk.pid |
04:37.36 | [TK]D-Fender | that's mine |
04:37.37 | Docfxit | There are no pid's in /var/run/asterisk only one file asterisk.ctl |
04:38.04 | [TK]D-Fender | perhaps you should kill * and restart it |
04:38.45 | Docfxit | /var/run/asterisk/asterisk.ctl -rw-rw-rw- |
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04:41.52 | Docfxit | Is it normal to get 5 lines after the kill * |
04:42.52 | Docfxit | root@UbuntuAsterisk:~# kill * |
04:42.52 | Docfxit | -bash: kill: callfile: arguments must be process or job IDs |
04:42.52 | Docfxit | -bash: kill: configackup.tar.gz: arguments must be process or job IDs |
04:42.52 | Docfxit | -bash: kill: hpec-9.00.007-prescott.tar.gz: arguments must be process or job IDs |
04:42.52 | Docfxit | -bash: kill: register: arguments must be process or job IDs |
04:42.52 | Docfxit | -bash: kill: zaptel-backup.tgz: arguments must be process or job IDs |
04:43.03 | Penguin | ~pb |
04:43.03 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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 install pastebinit with yum or aptitude. |
04:43.35 | Docfxit | Sorry. I will do that. |
04:43.45 | dpilon | ouch...the meant... kill the actual process not a * |
04:44.07 | dpilon | ps axf |grep asterisk get the id |
04:44.22 | dpilon | then: kill -9 <id> |
04:44.36 | Penguin | pkill asterisk <------ |
04:44.49 | dpilon | that works too |
04:45.00 | Penguin | It's a lot less bother. |
04:45.43 | Penguin | Also, pgrep, not ps|grep |
04:45.55 | Penguin | lrn2sysadmin |
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04:48.28 | Docfxit | After pkill asterisk I tried asterisk -r I'm getting the same error |
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04:48.55 | Docfxit | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist? |
04:49.09 | Docfxit | That file is the only file in that folder. |
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04:50.03 | rue_house | anyone have an answer for my mgcp problem? |
05:02.39 | Docfxit | Where is a good place to share a jpg for you? |
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05:04.39 | Scrye | has anyone ever experienced a 7960 that will not attempt dhcp and will not factory reset? |
05:04.49 | Penguin | docfxit: imagebin.org |
05:07.53 | Docfxit | Please see imagebin.org/242086 |
05:08.10 | ryan42 | [6~[6~[6~[6~[6~[6~[6~[6~[6~[6~[6~[6~[6~ |
05:08.20 | ryan42 | err, oops |
05:08.24 | Docfxit | That is the only file in /var/run/asterisk |
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05:34.45 | asteriskmonkey | im having a bit of difficulty with cut, can anyone lend a ahnd? |
05:36.52 | rue_house | I have an mgcp problem, I cant get two gateways to work |
05:37.29 | *** part/#asterisk asteriskmonkey (~philip@206.51.27.151) |
05:39.02 | [TK]D-Fender | patience[-1] |
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05:45.45 | rue_house | [TK]D-Fender, your awake! sweeet |
05:46.00 | rue_house | I couldn't hang by the keyboard at work |
05:46.18 | rue_house | after the first time I asked some guy needed the computer and I had to make an appointment |
05:46.37 | rue_house | the second time I came back and was told the room was being locked up |
05:46.41 | rue_house | so now I'm at home |
05:46.51 | rue_house | nice laaaaaaaaaaaaaaaaaarge scroll buffer |
05:47.06 | rue_house | and a bot I can get kicked for dragging in |
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07:21.26 | schmidts | good morning |
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07:23.13 | bombev | good morning guys :) |
07:25.22 | ChannelZ | Well, average. |
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07:50.31 | gavimobile | I cannot register one of my sip peers, when I try to debug this peers it says Unable to get IP address of peer '0000FFFF0000' |
07:50.38 | gavimobile | from my pc, I can ping this device |
07:54.48 | kaldemar | don't try to enable debug for the device only |
07:55.06 | *** join/#asterisk Invader (~Invader@unaffiliated/invader) |
07:56.39 | gavimobile | kaldemar: that what im doing now |
07:57.10 | gavimobile | kaldemar: I still don't see anything though |
07:59.34 | kaldemar | did you successfully enable sip debug? |
08:00.09 | gavimobile | kaldemar: sip set debug on is set |
08:00.30 | gavimobile | and im monitoring it to see if I see anything related to the ip or the peer name of this peer |
08:00.32 | gavimobile | but nothing |
08:01.36 | *** join/#asterisk Neptu (~Neptu@mail.avtech.aero) |
08:02.11 | gavimobile | I did run sip unregister 0000FFFF0000 when my peer was still unregistered |
08:02.46 | kaldemar | suprprise, unregister does nothing then the peer is not registered. |
08:02.46 | gavimobile | however I did sip quality peer 0000FFFF0000 after, also restarted asterisk |
08:03.00 | gavimobile | I also restarted my pap device |
08:03.19 | kaldemar | if you're not seeing messages from the device, then asterisk is not getting any. |
08:03.31 | gavimobile | I can ping the device though |
08:03.52 | kaldemar | ping won't help you if the device is misconfigured. |
08:04.41 | gavimobile | kaldemar: ok, so ill re-configure the device but I would like to know what caused the problem if possible.. it was working fine, and today all I did was turn it on after a power outage |
08:06.10 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:f949:40f4:6f06:2fea) |
08:06.28 | kaldemar | if the device does not send messages to asterisk, you won't find a reason for it by digging around on your asterisk server. |
08:08.05 | gavimobile | kaldemar: so I should be looking on google for reasons why my spa2102 won't registrer |
08:10.18 | ChannelZ | Which side, the FXO or FXS? Have you made sure there's a proxy typed in for either/both, that it's Line Enabled? |
08:10.43 | schmidts | gavimobile you should first take a look at the status page what it says there about the state of line 1, if it says something like unreachable you know the pap cant reach your asterisk |
08:11.05 | gavimobile | ChannelZ: my device doesn't have an fxo port only 2 fxs ports |
08:11.30 | gavimobile | schmidts: first page of the pap2 device settings? |
08:11.42 | kaldemar | gavimobile: you should look at the device itself. |
08:11.55 | gavimobile | once again there was no change in the device. im guessing it needs a firmware update |
08:12.07 | kaldemar | and not waste time googling for possible reasons. |
08:12.31 | *** join/#asterisk kleszcz (tick@linuxmafia.pl) |
08:12.40 | ChannelZ | oh right, mixing up the 3102. Same basic question applied anyway. |
08:12.47 | ChannelZ | applies |
08:15.40 | gavimobile | kaldemar: im looking at the device, there's only 2 lights on it.. in the settings page I do see that it says under status that registration failed |
08:16.49 | kaldemar | check addresses. where is it supposed to register itself? |
08:17.58 | gavimobile | kaldemar: you mean the domain names? sip.myserver.com |
08:18.20 | gavimobile | ill change it to a local address just for testing |
08:18.26 | gavimobile | should have thought of that |
08:18.42 | ChannelZ | shakes his head |
08:19.23 | gavimobile | Registered SIP '0000FFFF0000' at 192.168.0.102:5060 |
08:19.43 | gavimobile | but why won't the remote address work? |
08:19.53 | ChannelZ | Ask your DNS? |
08:19.53 | kaldemar | what "remote address"? |
08:20.34 | ChannelZ | Or perhaps the SPA doesn't even know a DNS server to ask. (psst: *something* changed) |
08:21.58 | gavimobile | ChannelZ: dns isn't configured on the device, its set to dhcp |
08:22.31 | ChannelZ | keep going |
08:22.37 | kaldemar | those are not mutually exclusive. |
08:23.25 | gavimobile | ? |
08:23.28 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.144) |
08:23.49 | ChannelZ | It gets -- or should get -- one from the DHCP server. It'd be on the Status tab as well. |
08:23.57 | kaldemar | something using dhcp does not mean that it does not have dns configured. |
08:24.15 | kaldemar | if it does not, your network is dumb. |
08:25.37 | gavimobile | kaldemar: I just looked and the dns is set to the default gateway from my router which should be fine |
08:26.42 | ChannelZ | 'should be'? Is the DNS server running, and/or returning the right IP for whatever hostname was in there? |
08:26.56 | kaldemar | and what address (if any) does the default gateway give for the domain name you had set? |
08:28.22 | Maliuta | depends on how you conf your dhcp server. There is no reason you _have_ to have the dhcp server actually provide a DNS server (or two), it could use a static ... but then it's more sensible to provide the servers via DHCP/BOOTP |
08:28.31 | gavimobile | here is the status information from my default gateway (my router) http://pastebin.com/tukryXGy |
08:28.52 | Maliuta | Is the port on the switch configured to the right vlan? |
08:29.06 | kaldemar | gavimobile: irrelevant |
08:29.12 | Maliuta | (mow I'm just being difficult) |
08:29.43 | Maliuta | you assigned the device a /32? |
08:29.51 | gavimobile | Maliuta: yes, the switch connects to the router directly |
08:30.05 | gavimobile | Maliuta: these are the default settings |
08:30.15 | ChannelZ | Well you've answered your own question insomuch as putting in the IP worked. So it's a bit up to you to figure out why it doesn't otherwise based on your network and servers and what is what |
08:30.29 | Maliuta | I'd say that something is wrong right there, unless the device has a decent routing table it won't know where to route packets to the gateway. |
08:31.08 | bulkorok | are there any good (voicemail-)prompts ins spanish out there!? |
08:31.25 | gavimobile | Maliuta: so what would you recommend? |
08:31.32 | Maliuta | bulkorok: the standard * ones aren't good enough |
08:31.51 | bulkorok | spanish ones?! I'll check that |
08:32.17 | Maliuta | gavimobile: well I don't know what your network looks like. But I'd suggest it should at least be on a /30 or /29 that the gateway also has an IP on |
08:32.22 | bulkorok | where to download!? |
08:32.45 | ChannelZ | http://downloads.asterisk.org/pub/telephony/sounds/ ? |
08:32.46 | gavimobile | Maliuta: that's a static ip we have |
08:33.01 | bulkorok | great :-) |
08:33.09 | kaldemar | gavimobile: start by finding the problem. try a DNS query for the value you had configured in the pap. |
08:34.36 | gavimobile | kaldemar: http://network-tools.com/default.asp?prog=dnsrec&host=sip.shn.co.il |
08:34.51 | Maliuta | gavimobile: unless there is point-to-point link on that device it's not going to be routing properly. Are the DNS servers your upstreams? or in house? |
08:35.33 | gavimobile | my dns server points to two different locations. to a web server and to my network where my sip server is located... |
08:35.46 | Maliuta | gavimobile: can you ping the device and get a response? if you can't then your problem is it's network config (and yes you can supply a borked network config via DHCP/BOOTP) |
08:35.47 | kaldemar | gavimobile: from your network of course, where the device is. and using the dns server that the pap has in its config. |
08:35.56 | gavimobile | Maliuta: yes I can ping the device |
08:36.49 | Maliuta | gavimobile: can you telnet/ssh into the device? |
08:37.01 | gavimobile | Maliuta: never tried |
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08:37.22 | Maliuta | gavimobile: it's a Cisco/Linksys right? |
08:37.40 | gavimobile | Maliuta: today its cicso it was once linksys |
08:37.40 | gavimobile | right |
08:38.58 | Maliuta | gavimobile: same diff. There should be a telnet port open (or you can open it with a tftp config). I'd be going in and trying a bunch of stuff from the console |
08:39.15 | Maliuta | gavimobile: it's standard cisco stuff, the commands are all self help |
08:39.46 | ChannelZ | this is off track |
08:40.03 | Maliuta | gavimobile: it might give you a better feel for where the problem is. I know that using telnet into my Cisco handset has helped in the past |
08:40.29 | Maliuta | ChannelZ: I'm trying to give him all the diagnosis tools for the device |
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08:41.42 | Maliuta | ChannelZ: to me the network config looks borked. That would stop it talking to the DNS and SIP servers, and anything else in the world |
08:44.18 | gavimobile | apparently I cannot ssh or telnet inside |
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08:45.04 | gavimobile | Maliuta: what about other devices that are behind the same nat and DO work |
08:45.20 | Maliuta | gavimobile: it is a feature that can be turned on and off. I use TFTP to configure my devices. |
08:45.21 | gavimobile | if the network config was faulty, other local peers shouldn't be working either |
08:45.35 | gavimobile | Maliuta: yea, I scanned the settings for this feature |
08:45.58 | Maliuta | gavimobile: do they have the same configs? i.e. /32 addresses? do they have routing tables? too many variables |
08:46.34 | Maliuta | I'm only going to be here until I finish eating, then I'm off to play Go for a couple of hours. |
08:46.43 | gavimobile | :-/ |
08:47.11 | Maliuta | gavimobile: are you assigning this device an address out of the same pool as the other devices? |
08:47.26 | gavimobile | Maliuta: yes |
08:47.38 | Maliuta | good start |
08:47.40 | gavimobile | dhcp gives it that address anyways |
08:47.59 | Maliuta | so I'm assuming the address you pastebin'd is not the address on the device |
08:48.31 | Maliuta | gavimobile: you're assigning out of what? a /24 a /29? |
08:48.48 | gavimobile | I don't know cidr blocks :-( |
08:48.48 | Maliuta | and then doing NAT somewhere. |
08:49.04 | Maliuta | what is the last 3 digits in the subnet mask? |
08:49.05 | gavimobile | but the same thing that any basic router will do 192.168.x.x |
08:49.09 | gavimobile | 0 |
08:49.18 | gavimobile | in the lan its 0 |
08:49.22 | gavimobile | in the wan its 255 |
08:49.40 | Maliuta | so the subnet mask on your lan is 255.255.255.0? |
08:50.40 | Maliuta | if the device is behind a working nat, forget the WAN side of things and look at the LAN side |
08:51.41 | Maliuta | it is not unusual for ISP's to assign a /32 (which is a single IP address on a PTP link) to a device, then if you have any other static nets they route them down that |
08:52.00 | Maliuta | or you use NAT for all devices behind it |
08:52.17 | gavimobile | Maliuta: not sure what that all means |
08:52.30 | Maliuta | gavimobile: so you don't know networking then? |
08:52.55 | gavimobile | Maliuta: not enough :-( |
08:53.42 | Maliuta | if you didn't understand that I'd say none, butI have fairly high standards for what constitutes "knowledge" :) |
08:54.09 | Maliuta | the device that is handing out the dhcp lease this device is getting. Is it a commodity router? |
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08:55.12 | Maliuta | gavimobile: you're not assigning it some sort of odd static address? And you're configuring the device via a web interface? |
08:55.54 | gavimobile | Maliuta: no odd static address and configuring it through web interface. correct |
08:58.25 | Maliuta | gavimobile: does the web interface have a network test function somewhere? |
08:58.43 | gavimobile | Maliuta: no network testing function |
08:58.48 | Maliuta | most of them do. |
08:59.07 | Maliuta | normally somewhere in the admin or status areas |
08:59.23 | Maliuta | that makes it harder to diagnose |
08:59.50 | gavimobile | there is only 2 buttons |
08:59.55 | gavimobile | the rest are fields |
09:01.00 | Maliuta | gavimobile: what is the device type _exactly_ (i.e. down to hardware revision)? |
09:01.52 | Maliuta | I'm about to go out, but I'll take my laptop and I can do some looking while I'm out. If I come up with something I can just jump on IRC and tell you. |
09:01.54 | gavimobile | spa 2102. doesn't have a rev but it does say near the fcc logo 1.1.1 |
09:02.17 | gavimobile | Maliuta: I appreciate your help! thanks man |
09:02.54 | Maliuta | gavimobile: I try ... when I'm not being trying. |
09:03.52 | Maliuta | gavimobile: tip. either stay in channel, or re-join when ever your infront of a machine. I'm always here, but not always in front of my machine |
09:04.13 | gavimobile | Maliuta: will do! thanks |
09:06.14 | Maliuta | there are a lot of helpful people around in here, we just don't always agree on how to track down a problem :) Or the best way to do a certain task. |
09:06.45 | Maliuta | Off to get my butt handed too me. |
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09:26.10 | k3asd` | hi |
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10:40.50 | gavimobile | my main trunk incomming doesn't seem to be working properly. |
10:41.21 | gavimobile | running sip show peers says netvision-trunk-66 82.166.66.43 5060 UNREACHABLE netvision-trunk-67 82.166.67.43 5060 UNREACHABLE |
10:43.48 | kaldemar | more network issues |
10:44.16 | gavimobile | I am getting the call now but no sound |
10:45.28 | gavimobile | kaldemar: before I upgraded the firmware an immeditly my pap device worked |
10:45.52 | kaldemar | gavimobile: i don't understand what you mean. |
10:46.39 | gavimobile | kaldemar: my current problem is when people in the outside call they don't hear the ivr but my phones ring |
10:47.26 | kaldemar | sip debug will most likely tell the reason. |
10:49.25 | gavimobile | kaldemar: here's my sip debug http://pastebin.com/aL7saHtf |
10:51.00 | gavimobile | it amazes me that only 1 of my istp's won't connect |
10:51.04 | gavimobile | and they say its not them |
10:53.34 | kaldemar | don't limit the debug |
10:54.54 | kaldemar | and naturally you need to look at a sip debug of a call. |
10:56.11 | kaldemar | if they don't answer to your OPTIONS messages (which are sent because of qualify=yes), don't send them. disably qualify for the peer. then you won't see UNREACHABLE in sip show peers. |
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10:57.26 | gavimobile | kaldemar: this is what I was able to log. http://pastebin.com/znEL4k0b |
10:57.49 | gavimobile | kaldemar: qualify=yes and they still show unreachable |
10:59.19 | gavimobile | on-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) |
10:59.26 | gavimobile | Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) |
10:59.28 | gavimobile | line 129 |
10:59.28 | kaldemar | re-read what i just wrote about the unreachable. |
11:00.29 | gavimobile | I disabled qualify and now they show as unmonitored |
11:00.48 | gavimobile | should I make another test call? |
11:01.00 | gavimobile | with qualify disabled? |
11:01.03 | kaldemar | no. |
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11:01.33 | gavimobile | is there a problem with codec or permissions? |
11:01.42 | gavimobile | cause last week I acidently deleted the sounds folder |
11:01.59 | gavimobile | but all was working find till this morning |
11:02.23 | kaldemar | does not seem to be. both have ulaw and asterisk says it plays the sound file. |
11:02.39 | gavimobile | kaldemar: in addition if I call the extention locally I hear sound |
11:02.46 | gavimobile | but not from calling using my cellphone |
11:03.25 | kaldemar | do you see rtp going out with rtp debug? |
11:04.18 | gavimobile | no |
11:04.36 | gavimobile | <PROTECTED> |
11:04.39 | gavimobile | then call |
11:05.51 | gavimobile | what am I looking for in the debug |
11:05.58 | gavimobile | its all the same two lines over and over again |
11:07.29 | kaldemar | and they are... |
11:08.51 | gavimobile | http://pastebin.com/ECydMgSZ |
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11:13.09 | kaldemar | looks normal. |
11:15.40 | gavimobile | kaldemar: so is this a problem with my provider? if so how can I prove it and what do I tell them |
11:17.29 | kaldemar | not necessarily. might be a firewall that does not let the audio go through or whatever. |
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11:21.51 | gavimobile | kaldemar: I am not sure about that.. why was until yesterday my trunks from this provider always listed as ok and not they are not |
11:22.14 | gavimobile | qualify was always set to yes. and the show peer results was ok |
11:23.20 | gavimobile | what's going on here |
11:23.44 | kaldemar | your router is one point of failure. see that you don't have any ALG (application level gateway) for SIP enabled at all. |
11:25.01 | gavimobile | so I need to change my router? |
11:25.10 | kaldemar | did i say that? |
11:25.22 | kaldemar | you need to make sure that it does not interfere. |
11:25.52 | gavimobile | im not ever sure where to start |
11:26.15 | gavimobile | opening dmz's for testing? |
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11:28.34 | kaldemar | verywiseman: don't target people with private messages like that. |
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11:28.58 | verywiseman | kaldemar, why ? |
11:29.17 | gavimobile | making outgoing calls works find on both sides |
11:30.10 | kaldemar | ~ask |
11:30.10 | 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. |
11:30.35 | kaldemar | verywiseman: and because i am not private consultant. |
11:31.11 | wdoekes | you could be.. for the right amount ;) |
11:31.33 | schmidts | kaldemar you are NOT :( |
11:33.13 | kaldemar | s/not/not your/ |
11:33.40 | kaldemar | that was what i really meant, a word was missing. :P |
11:34.52 | wdoekes | s/your/your free/ |
11:35.20 | kaldemar | getting better |
11:36.24 | verywiseman | kaldemar, you must say that when i talk to you in private channel , it is not needing to tell me that in public :( |
11:37.11 | wdoekes | but the ~ask doesn't work in private |
11:38.22 | verywiseman | ok , it was enough to inform me that when i asked him in private channel , i was not knowing that |
11:41.38 | wdoekes | but now you do; we won't hold it against you |
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12:50.23 | Greenlight | Hi all. Just wondering if anyone has a UK ISDN30 (Virgin Media) setup and presenting outbound callerid. Virgin assure us that we've "type 5" CLI and we can present ANYTHING, but the only number that seems to present is the main bearer number (or withheld)? |
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13:51.24 | BorjaGVO | Hi people. Something strange happened some days ago. I had in my full log the following message 245 times in minutes intervals: "NOTICE[25529] chan_sip.c: Sending fake auth rejection for device 1000<sip:1000@IP.IP.IP.IP>;tag=f2a400as". The wird thing is that the IP.IP.IP.IP is my server own IP address. It's a CentOS running just Asterisk, FreePBX and some other Asterisk-related tools. I don't understand how can this happen... |
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13:52.12 | BorjaGVO | I have a many operators calling and receiving calls through asterisk but none of them making calls from that box (IP.IP.IP.IP) |
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13:57.03 | kaldemar | BorjaGVO: that is not the address where the request is coming from. |
14:03.23 | BorjaGVO | well, when I make calls without registering with domain with random extension, it logs the same message with IP address where request is coming from |
14:03.32 | BorjaGVO | (kaldemar) |
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14:06.47 | kaldemar | BorjaGVO: care to pastebin the output? |
14:07.03 | BorjaGVO | kaldemar: sure |
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14:15.53 | BorjaGVO | kaldemar: http://pastebin.com/Rjy44usg. I see now that the output from one case to the other vary. In one there is not port specification. |
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14:18.54 | raden | Katty, !!!! :) |
14:19.21 | Katty | hi raden. |
14:19.27 | raden | Hiya !!!!!!!!!!! |
14:19.32 | raden | I need a Katty hug |
14:19.40 | Katty | why's that? |
14:20.22 | raden | the girl I was / am into has went from sweet , caring , awesome to total ice queen and like yea omg |
14:20.47 | raden | cause some dude at work is filling her head with a bunch of shit |
14:21.04 | Katty | i bet it's not like that. |
14:21.14 | raden | ooooo tell me what its like |
14:21.30 | Katty | i would imagine that the doubts have been there all along. |
14:21.36 | BorjaGVO | kaldemar: brb, going out to lunch (10 minutes) |
14:21.46 | Katty | else she'd have no reason to listen to him |
14:22.11 | Katty | that or you've done something to make an ass of yourself. |
14:22.37 | raden | nope she tells her friends how sweet i am and all this , even her sister is like she is so into you .... |
14:22.55 | raden | yet I have got stood up last 2 weekends |
14:23.00 | raden | and she always says how sweet i am |
14:23.13 | raden | my head starting to hurt from this shit |
14:23.27 | Katty | if you're not happy, move on. |
14:23.36 | Katty | don't waste your energy. |
14:23.39 | raden | yea , thats what im doing |
14:23.45 | Katty | good (= |
14:23.48 | Katty | hugs raden |
14:24.31 | raden | I have no problem attracting women , not like im desperate , just shes awesome and fun and things were easy until dude at work started talking shit ..... |
14:25.05 | Katty | don't blame some guy. |
14:25.11 | Katty | guys do what guys do. |
14:25.34 | raden | lol |
14:25.45 | raden | that is true |
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14:25.56 | raden | but at the same time shes " trusts his udgement " |
14:26.06 | raden | and the dude fucking hates me |
14:26.13 | raden | there a lot to it ... |
14:26.29 | raden | yet she texts me every freaking day ....... |
14:26.54 | raden | Katty, why cant these things ever just be like boy meets girl simple ? |
14:27.09 | Katty | because humans aren't simple. |
14:27.23 | Katty | and having a heart broken makes the process even more complicated. |
14:27.37 | Katty | because then you have worry and fear running rampant on an otherwise perfectly healthy relationship. |
14:27.48 | raden | I showed interest to early and gave her to much control that was my bad , I know how to control women and I did not want to play games with her that was my stupid mistake .... |
14:28.02 | Katty | not everyone can be controlled. |
14:28.10 | raden | I shouldnt say control |
14:28.12 | Katty | some it just pisses off, and they will send you home talking to yourself. |
14:28.12 | raden | attract |
14:28.46 | raden | I dont try to control people , i just know the right things to do to attract them .... and I did none of that with her we just hit it off .... |
14:28.57 | raden | lol , send you home talking to yourself :) I like that |
14:29.33 | raden | I feel as though I have been put in the imaginary friend zone .... LOL |
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14:29.36 | Katty | i don't think you should have to do anything to attract the right person. just be yourself. that should be enough to attract someone. |
14:29.40 | raden | who knows I should move on |
14:29.53 | Katty | maybe you have. |
14:30.00 | Katty | maybe he is better suited to her than you are. |
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14:30.09 | raden | hes older than her by far |
14:30.14 | Katty | so? |
14:30.19 | raden | good point |
14:30.20 | Katty | maybe she wants someone to take care of her. |
14:30.29 | Katty | i know someone like that....on both sides. |
14:30.33 | raden | I can take care of her way more than he can :P |
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14:30.58 | Katty | some people like that. some people's inner jew bothers them if they don't hold up their half financially. |
14:31.09 | Katty | all types of people in the world. |
14:31.15 | raden | I totally agreee |
14:31.34 | raden | i never put to much forward |
14:32.18 | raden | Just frustrating , 2 months invested in this BS |
14:32.51 | raden | known her for 20 years |
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14:33.00 | raden | gah im getting old LOL |
14:34.22 | Katty | 2 months is nothin |
14:34.29 | Katty | let's take this to private lol |
14:34.30 | raden | I know , I know |
14:34.43 | Katty | no one wants to hear this conversation :P |
14:35.06 | schmidts | no one else is saying something, maybe everybody only listens ;) |
14:36.03 | kaldemar | was that about a service level agreement with a client? |
14:36.48 | Katty | totally. |
14:37.15 | kaldemar | the client did not want to pay for it. |
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14:43.06 | pbxMan | hello |
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14:44.10 | *** mode/#asterisk [+o sruffell] by ChanServ |
14:54.21 | BorjaGVO | anyone knows why this kind of messages happen to log? It's strange because the IP address is the server address. It's like the own server made calls. Actually, one of our operators recevied a call from this extension (1000) some hours after. No one was on the other side (no audio stream): http://pastebin.com/Rjy44usg |
14:54.47 | Katty | infobot: crittercam |
14:54.47 | infobot | [crittercam] Katty's Critter Cam http://tinyurl.com/b5k3lt4 |
15:01.22 | *** join/#asterisk MarKsaitis (~MarKsaiti@81.101.81.114) |
15:08.37 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
15:08.37 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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15:25.34 | *** mode/#asterisk [+o putnopvut] by ChanServ |
15:27.47 | wdoekes | BorjaGVO: the log message is confusing. it shows the To: field in the header, which usually contains the server address, not the IP you're interested in |
15:29.04 | BorjaGVO | How do you know is the To: field? |
15:29.11 | BorjaGVO | (wdoekes) |
15:34.57 | fullstop | I'm watching a squirrel eat birdseed. |
15:35.15 | Katty | and peanuts. |
15:35.49 | Katty | tho i guess you're right. he's mostly going for the sunflowers. |
15:35.58 | fullstop | nice snowman |
15:36.04 | Katty | thanks |
15:36.07 | Katty | https://sphotos-a.xx.fbcdn.net/hphotos-prn1/154579_10100452825052137_2118825807_n.jpg <- snowman |
15:36.20 | fullstop | I'll be honest, though. It has seen better days. |
15:36.32 | Katty | yesh |
15:36.53 | fullstop | almost looks like a snow-cat. |
15:37.10 | Katty | he was made to be put on reddit. |
15:37.16 | fullstop | I see |
15:37.17 | carrar | HELLO SNOW KITTY |
15:37.22 | Katty | hi carrar |
15:37.34 | carrar | Hi Katty! |
15:38.03 | Katty | hugs carrar |
15:38.09 | carrar | Woo Woo!! |
15:38.16 | Katty | carrar: i got my stegosaurus done. i named him Steve, the chronically depressed Rainbow Stegosaurus |
15:38.18 | carrar | spins Katty around |
15:38.30 | *** join/#asterisk UQlev (~yuriy@46.251.117.25) |
15:38.37 | carrar | heh |
15:38.41 | Katty | carrar: http://42ndknitstreet.blogspot.com/2012/11/stegasaurus.html <- steve. |
15:38.44 | *** part/#asterisk UQlev (~yuriy@46.251.117.25) |
15:38.51 | carrar | <PROTECTED> |
15:39.04 | Katty | he's already built! |
15:39.19 | carrar | Woah that looks danergously green |
15:39.21 | Katty | (scroll to the bottom to see friends) |
15:39.30 | *** join/#asterisk navaismo (~navaismo@189.191.2.44) |
15:39.34 | wdoekes | BorjaGVO: enable: sip set debug on |
15:39.48 | carrar | dinasur and bunnies |
15:40.02 | wdoekes | then you'll get the entire sip mesage, and you'll see the source IP that's causing the "Fake auth reject" log messages |
15:40.06 | Katty | Bun, Ham, and Doc Oct |
15:40.16 | Katty | they were trying to console steve. |
15:40.21 | carrar | heh |
15:40.48 | BorjaGVO | wdoekes: I cannot do that since I don't know when the attepmt is going to happen. It only happened once... |
15:41.00 | Katty | now i'm finishign some garland for the xmas tree i already took down. heh. what are you up to carrar? |
15:41.23 | carrar | I just woke up! |
15:41.41 | Katty | sounds like it's time for Red Bull in the shower. |
15:42.08 | carrar | I have to still make my espresso! |
15:42.28 | Katty | did i tell you my gentlemen friend bought me an espresso machine for xmas? |
15:42.39 | carrar | gentlemen friend!! |
15:42.42 | carrar | PICS |
15:42.46 | carrar | of the espresso machine |
15:42.57 | Katty | http://www.amazon.com/DeLonghi-EC155-Espresso-Cappuccino-Maker/dp/B000F49XXG/ref=sr_1_1?ie=UTF8&qid=1357659756&sr=8-1&keywords=espresso <- that one |
15:43.01 | carrar | I have pics of mine too! |
15:43.16 | carrar | Oh nice |
15:43.17 | Katty | it's a dream. |
15:43.28 | carrar | Espresso Machines really are a dream come true |
15:43.33 | Katty | i got him a burr grinder and an electric tea kettle for xmas. |
15:43.42 | Katty | he already had a french press |
15:43.48 | Katty | so now he have a coffee bar. |
15:43.49 | fullstop | because of wifestop's occupation, we have random stuff sent to our house all the time. |
15:43.54 | Katty | finds photo of coffee bar |
15:44.09 | fullstop | Apparently there is a blendtec blender on the way... and we get to keep it! |
15:44.17 | carrar | Mine: http://pics.osburn.com/album/1230 |
15:44.32 | carrar | burr grinders are the best! |
15:44.38 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
15:44.40 | Katty | https://sphotos-b.xx.fbcdn.net/hphotos-prn1/68603_10100447794433537_1245549956_n.jpg <- coffee bar |
15:44.57 | Katty | there's a coffee pot too, but it wouldn't fit |
15:45.00 | carrar | holy cafe! |
15:45.10 | Katty | we take our coffee very seriously. |
15:45.14 | carrar | It's like a cafeteria |
15:45.18 | Katty | yesh. |
15:45.35 | carrar | Were is the food dispensing machine! |
15:45.52 | Katty | behind the photographer |
15:45.54 | Katty | in the fridge hehe |
15:45.58 | carrar | heh |
15:46.28 | Katty | the local grocery store puts their 'old' meat on sale on tuesdays to clear it out |
15:46.44 | Katty | they've been having lots of steak recently. |
15:47.01 | Katty | so the Food Despensing Machine has lots of frozen steaks to pick from |
15:47.25 | Katty | last week they had a package of ribeyes for 6 bucks. 2 of them. |
15:47.27 | carrar | heh |
15:48.46 | carrar | It's too danergous to eat meat anymore |
15:49.00 | carrar | dangerous |
15:49.03 | Katty | cause of all the stuff theyput in it? |
15:49.07 | carrar | yeah |
15:49.09 | Katty | well. into the cows. and chickens. and what not |
15:49.15 | carrar | or things they didn;t know got in it |
15:49.19 | Katty | yes :< |
15:49.21 | carrar | like crazy cow |
15:49.22 | BorjaGVO | wdoekes: I didn't know that letting enabled sip debugging got it on logs. Thanks. |
15:49.30 | Katty | carrar: i've had to switch to organic eggs. |
15:49.37 | Katty | carrar: something about the regular ones makes me nauseous. |
15:49.54 | Katty | carrar: dunno what. my guess is the chemicals in the chicken feed. |
15:50.08 | carrar | biochemically engineered eggs! |
15:50.30 | carrar | or something like that |
15:50.32 | carrar | heh |
15:50.41 | Katty | well they have special feed for chickens. |
15:50.47 | Katty | it has growth hormones and antibiotics in it |
15:51.03 | carrar | All makes for a better super bug! |
15:51.07 | Katty | there's a documentary on netflix where they talk about Tyson chicken. |
15:51.45 | Katty | i should just get a few chickens. |
15:51.51 | Katty | very spoiled fat little chickens. |
15:51.58 | carrar | yeah |
15:51.58 | fullstop | We used to get free-range chicken eggs from a friend that had a farm. |
15:52.19 | Katty | fullstop: those are the best. |
15:52.19 | fullstop | The shells were crazy thick and the yolk was bright orange. |
15:52.20 | carrar | make sure to wash them! |
15:52.25 | fullstop | Naturally |
15:52.33 | Katty | fullstop: my parents used to have chickens. |
15:52.39 | fullstop | They were awesome. But then they moved. |
15:52.43 | Katty | they had a big yard too...and just meandered about looking for bugs. |
15:53.05 | fullstop | Some other friends had two chickens: Original Recipe and Extra Crispy. |
15:53.25 | Katty | awww now that would be a wonderful name for a couple of hens. |
15:53.27 | Katty | Original and Crispy |
15:53.29 | carrar | You could raise angus cows too! |
15:53.31 | fullstop | One was eaten by a fox, and the other wasn't much for laying eggs. They'd get one or two a week. |
15:53.35 | *** join/#asterisk MarKsaitis (~MarKsaiti@81.101.81.114) |
15:53.39 | carrar | feed them grain |
15:53.41 | Katty | carrar: for cheese and milks? |
15:53.48 | carrar | for dah beef! |
15:53.48 | Katty | carrar: why not grasses |
15:53.54 | Katty | oh i wouldn't have the heart to kill it |
15:54.02 | fullstop | That's why you take it to a butcher. |
15:54.02 | carrar | grain feed cows the meat taste better |
15:54.06 | fullstop | no sir |
15:54.09 | fullstop | grass all the way |
15:54.12 | Katty | i disagree. |
15:54.16 | Katty | i'm a fan of grass fed beefs. |
15:54.28 | carrar | we've done it |
15:54.31 | carrar | for several years |
15:54.50 | carrar | we've (as in my parents) |
15:54.51 | Katty | guess we have different tastes then (= |
15:54.58 | carrar | yeah |
15:55.03 | coppice | Katty how do you feed it once it has become beef? :-\ |
15:55.05 | Katty | more for me. more for you! |
15:55.30 | carrar | coppice, feed it buy opening wide and inserting said meat into mouth! |
15:55.39 | dr0ck | the way you feed it when its still a cow. cut a hole in the side and pour feed inside |
15:55.40 | Katty | yes. then proceed to omnomnom. |
15:55.43 | carrar | haha |
15:55.48 | *** join/#asterisk wonderworld (~w@dsdf-4db541ae.pool.mediaWays.net) |
15:55.56 | Katty | let's not talk about cows. |
15:56.02 | Katty | i feel bad for them. |
15:56.03 | carrar | moo |
15:56.15 | fullstop | ಠ_ಠ |
15:56.24 | carrar | There is always kale |
15:56.38 | Katty | mmm kale chips. |
15:56.44 | Katty | sprinkled with sea salt |
15:56.44 | coppice | Katty: do you feel sorry for the ones fed on beer, and massaged every day? |
15:56.52 | fullstop | I have two raised garden beds now, 4x8 each. |
15:57.00 | fullstop | Hopefully expanding to 4 next year. |
15:57.02 | Katty | fullstop: i've been thinking about doing a raised garden |
15:57.08 | Katty | fullstop: what do you grow? |
15:57.12 | fullstop | Where are you, geographically? |
15:57.19 | Katty | southern missouri. |
15:57.24 | fullstop | I made mine from cedar, but it is kind of hard to find here. |
15:57.32 | fullstop | east coast |
15:57.33 | Katty | oh well. i can't do that anyway. |
15:57.36 | Katty | i'm allergic to cedar. |
15:57.40 | fullstop | !! |
15:57.43 | coppice | my wife's cousin has a raised garden. it covers the roof of a 30 floor tower |
15:57.54 | Katty | coppice: wow nice. |
15:58.05 | fullstop | Hopefully they have a long hose. |
15:58.18 | coppice | they grow fruit and veg, and raise chickens |
15:58.35 | Katty | chickens? on a roof? |
15:58.40 | Katty | seems dangerious. |
15:58.49 | coppice | why? |
15:59.05 | Katty | i'm afraid the chickens would fly off. and crash land somewhere...30 stories down. |
16:00.14 | coppice | they have an avery sized cage for the chickens, although chickens aren't major flyers anyway |
16:00.31 | Katty | ah. well at least they're caged. |
16:00.43 | Katty | and no, they're not major flyers. but they're not exactly graceful when they try either |
16:02.50 | coppice | its kinda weird standing in a market garden, looking down at the streets of GuangZhou 30 floors below |
16:03.13 | navaismo | BorjaGVO: enable the sip debug then you can check it in the full log(enable the full log too) and secure your asterisk, allowguest=no, dont use weak passwords, close ports if you dont use it |
16:03.15 | Katty | i imagine. |
16:03.27 | Katty | i love the idea that someone is doing it tho. |
16:03.32 | Katty | i'd gladly buy from them. |
16:04.35 | fullstop | coppice: can you read Cantonese? |
16:05.00 | coppice | 是的 |
16:05.51 | fullstop | I'll have to see if I still have it, but my wife received a camera bag for christmas with a chinese newspaper stuffed inside. There's a comic in it that I was wondering if it would be funny when translated. |
16:06.19 | fullstop | hopefully she did not pitch it |
16:08.43 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
16:21.14 | Kobaz | anyone have a problem where you have a loaded system and completion of an attended transfer is very delayed |
16:21.57 | Kobaz | ie: A calls B, B calls C... B hits transfer in order to connect A and C together. And it takes 5-10 seconds for the audio stream to start from A to C |
16:27.11 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
16:27.57 | carrar | ok, getting properly caffinated |
16:28.17 | mjordan | Kobaz: does the log show RTP in probation? |
16:30.54 | Kobaz | mm, what would probation look like? |
16:31.06 | Kobaz | i dont like any rtp, but i can log it for an experimental session |
16:31.11 | Kobaz | s/like/log/ |
16:33.22 | mjordan | Kobaz: it should be just a normal debug message - you don't have to have rtp debug on |
16:33.46 | Kobaz | ah |
16:33.50 | Kobaz | lemme search |
16:34.20 | mjordan | strictrtp requires a sequential number of RTP packets to be received from a particular source before it trusts that it should send RTP to that source. Basically prevents someone from sneaking into the audio stream and snagging it away from an endpoint. |
16:34.44 | Kobaz | i dont have strictrpt set |
16:34.58 | Kobaz | no logs matching *prob* for today |
16:35.16 | Kobaz | or yesterday |
16:36.37 | *** part/#asterisk rkeene (1011@oc9.org) |
16:38.40 | *** join/#asterisk spriggan (~ircap@excsupercol.supercable.net.co) |
16:39.58 | Kobaz | this is on 1.8 |
16:40.13 | spriggan | hi every one ,, some one can help me ? im using the manager api to get the QueueStatusComplete info.. i need to know if the wait time is the time since the call gets into the queue or if it's since the call gets into the ivr |
16:43.46 | mjordan | Kobaz: not sure then. I haven't heard of that happening. |
16:44.57 | *** join/#asterisk shadar (~eugene@37.113.202.81) |
16:55.29 | *** join/#asterisk Mon|A|rch (~SBean@72.29.180.35) |
16:56.09 | Mon|A|rch | hey |
16:56.15 | Mon|A|rch | anyone around that can help me with some issues? |
16:56.36 | WIMPy | ~ask |
16:56.36 | 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. |
16:57.30 | Mon|A|rch | lol, fair enough |
16:58.24 | Mon|A|rch | so, I'm setting up an IVR, and within the local network, I can call the asterisk server's extension (502) and it'll route me through wherever I want to go |
16:58.45 | Mon|A|rch | the issue I'm having is that I can't originate calls with the ami, because my sip channel status is 4 |
16:58.50 | Mon|A|rch | or unavailable i suppose |
16:59.02 | Mon|A|rch | sip show peers shows me my sip channels |
16:59.05 | Mon|A|rch | sip show registry does not |
16:59.30 | Mon|A|rch | I'm taking over someone elses project, so their code is a little wonky in extensions.conf |
16:59.39 | Mon|A|rch | I'm thinking I might need to just go ahead and replace the whole file |
16:59.39 | Kobaz | sip show registry will show you registrations that asterisk is set up to make with other devices |
16:59.50 | Mon|A|rch | okay |
17:00.27 | Mon|A|rch | I was a little confused on the difference |
17:00.29 | Mon|A|rch | anyway |
17:00.38 | Mon|A|rch | I think I set up my incoming trunk incorrectly |
17:00.38 | Kobaz | like, say an itsp requires registration, you'll add the registration in your configs and then it'll show up in sip show registry |
17:01.02 | Mon|A|rch | okay |
17:01.35 | Kobaz | sip show peers will show configured sip peer devices and their status |
17:01.53 | Kobaz | not necessarily all available sip channels because there is sip show users as well |
17:02.42 | Mon|A|rch | i see |
17:02.49 | Mon|A|rch | so I've got one more user than i have peers |
17:03.04 | Mon|A|rch | the extensions i want to access are listed in sip show users/peers |
17:03.05 | Kobaz | if you do type=friend in the sip.conf then it will add a peer, and a user |
17:03.14 | Mon|A|rch | k |
17:03.14 | Kobaz | if you do type=user, then it's just a user |
17:03.24 | Mon|A|rch | I've been using type=friend |
17:03.45 | Kobaz | there's generally very little need to specifically do type=user |
17:04.21 | Mon|A|rch | i would assume so |
17:04.45 | Mon|A|rch | so, I'm assuming I need to set up a user, or a sip channel that represents my incoming/outgoing trunks? |
17:04.48 | Kobaz | you can think of users as only inbound |
17:04.56 | Mon|A|rch | okay |
17:04.58 | *** join/#asterisk ujjain (ujjain@unaffiliated/ujjain) |
17:05.09 | Kobaz | friend as inbound/outbound |
17:05.43 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
17:07.34 | Kobaz | and it's better to refer to your setup as sip peers and users, not really channels |
17:07.41 | Kobaz | because channels have a specific meaning |
17:07.50 | Kobaz | it's more like available to use channels, but not really |
17:08.06 | Kobaz | a channel is a leg of a call, (a call in progress) |
17:08.17 | Kobaz | sip show peers, is more like, showing you available devices |
17:10.40 | Mon|A|rch | so, are there any configs, aside from sip.conf and extensions.conf that actually determine whether or not a sip friend can be accessed? |
17:10.57 | *** join/#asterisk retentiveboy (~retentive@74-95-28-34-Atlanta.hfc.comcastbusiness.net) |
17:11.26 | Mon|A|rch | I've set the default context for my trunk and the friends associated with the extensions I'm using to the context that the extensions are in |
17:11.48 | Mon|A|rch | do i need to declare my trunk within sip.conf? |
17:20.33 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
17:24.34 | Mon|A|rch | from what i can tell, I need to register my sip provider (which is a cisco server we're using) in sip.conf? |
17:29.10 | *** join/#asterisk qakhan (~qakhan@208.253.91.58) |
17:29.17 | qakhan | hi all |
17:29.33 | qakhan | anyone has polycom .cfg files |
17:29.44 | qakhan | tamplates |
17:37.19 | Mon|A|rch | *sigh* so much lurking |
17:37.27 | Mon|A|rch | well, if anyone has a suggestion, let me know |
17:37.48 | Mon|A|rch | I'm just unsure how to properly set up outbound and inbound |
17:43.53 | kaldemar | Mon|A|rch: you need to edit both sip.conf and extensions.conf for inbound and outbound: http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/OutsideConnectivity_id291235.html#OutsideConnectivity_id291268 |
17:45.05 | Mon|A|rch | kaldemar, thanks |
17:45.07 | Mon|A|rch | i appreciate it |
17:48.09 | *** join/#asterisk Dibbler (~Dibbler@host109-148-34-244.range109-148.btcentralplus.com) |
17:48.25 | Mon|A|rch | kaldemar, so, if i were to set a friend to host=dynamic, it would receive all calls? |
17:49.32 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
17:50.26 | *** join/#asterisk citrusfizz (~chatzilla@70.184.40.66) |
17:51.11 | citrusfizz | I setup a conference room on Asterisknow, with out fail, everyone gets kicked off after 30 minutes exactly. what could be the cause of this? |
17:51.42 | Qwell | a timer on the room? |
17:55.03 | kaldemar | Mon|A|rch: setting something host=dynamic has little to do with receiving calls. it only tells asterisk that the device should register to it. after registering, asterisk knows an ip address and a port to reach the device. |
17:55.32 | Mon|A|rch | okay |
17:56.09 | Mon|A|rch | do i need to actually use users.conf? |
17:57.29 | Mon|A|rch | sorry about my confusion, I've had to figure this all out pretty quickly |
17:58.51 | *** join/#asterisk JustinAiken (~JustinAik@justinaiken.com) |
17:59.18 | JustinAiken | Hi all, having an issue with DTMF detection on ast 1.8 i was looking for some help on |
17:59.31 | JustinAiken | sometimes a digit is doubled; pressing it once receives it twice |
17:59.42 | JustinAiken | if i look at the dtmf logging, i see stuff like: |
17:59.52 | JustinAiken | '9' has duration 39 but want minimum 80, emulating on SIP/vitel-inbound2-0000014a |
18:00.06 | JustinAiken | i do i stop it from emulating if it's below the minimum? |
18:00.21 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
18:01.05 | [TK]D-Fender | <Mon|A|rch> do i need to actually use users.conf? <- no |
18:01.39 | [TK]D-Fender | Mon|A|rch, Only AsteriskGUI really needs it. Aside from that I wouldn't touch it ever. |
18:02.02 | Mon|A|rch | alright |
18:02.58 | Mon|A|rch | so, i want to originate calls to outside numbers, which are passed in the originate command from some php code, to an extension in my incoming context |
18:03.18 | [TK]D-Fender | <qakhan> anyone has polycom .cfg files <---- polycom.com. It comes with the firmware |
18:03.36 | [TK]D-Fender | Mon|A|rch, cool. |
18:03.46 | Mon|A|rch | do i need to set up a SIP friend for that extension specifically? i figure i need to set up a trunk for my provider |
18:04.26 | [TK]D-Fender | Mon|A|rch, for the actuall calls.. yes |
18:04.59 | *** join/#asterisk retentiveboy (~retentive@74-95-28-34-Atlanta.hfc.comcastbusiness.net) |
18:05.28 | [TK]D-Fender | Mon|A|rch, What "extension"? |
18:05.46 | [TK]D-Fender | Mon|A|rch, you set up SIP entires to devices and providers you need to call/get calls from |
18:06.16 | Mon|A|rch | okay |
18:06.52 | *** join/#asterisk slidesinger (~slidesing@c-69-141-208-250.hsd1.nj.comcast.net) |
18:07.13 | Mon|A|rch | so what i need to do is set up a single sip entry for my provider server, to route calls to my incoming extension |
18:07.36 | [TK]D-Fender | What is this "incoming extension" you are referring to? |
18:08.54 | Mon|A|rch | sorry, incoming is just a context in extensions.conf, and there's an extension in there that is the IVR |
18:09.51 | [TK]D-Fender | Mon|A|rch, then that is just dialplan.... so just the SIP entry to match the incoming call. |
18:12.02 | Mon|A|rch | alright |
18:13.31 | Mon|A|rch | will i need to change the entry for outgoing? will type=friend cover that? |
18:19.58 | gusto | so |
18:20.12 | Mon|A|rch | I'm getting a -1 status when i use extensionstate |
18:20.19 | Mon|A|rch | does that generally mean it can't find the extension? |
18:20.30 | gusto | no idea |
18:20.36 | gusto | what extension? |
18:20.41 | gusto | are you in right context |
18:20.43 | gusto | ha? |
18:21.05 | [TK]D-Fender | ~pb |
18:21.05 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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 install pastebinit with yum or aptitude. |
18:21.07 | [TK]D-Fender | Mon|A|rch, ^^^ |
18:21.21 | [TK]D-Fender | Show us precisely what you're doing and what's happening... |
18:21.49 | Mon|A|rch | alright |
18:27.32 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
18:27.59 | Mon|A|rch | [TK]D-Fender, http://pastebin.com/nZhaSCqg |
18:28.25 | Mon|A|rch | I'll paste the originate command in a sec |
18:28.36 | *** join/#asterisk Defraz (~Defraz@168.103.142.217) |
18:28.43 | [TK]D-Fender | Mon|A|rch, exten => 502,1, goto(incoming,500,1) <- you have no 500,1 |
18:28.57 | [TK]D-Fender | Mon|A|rch, You started with "n" and have no step 1 |
18:29.25 | [TK]D-Fender | exten => 500,n, Goto(diab-exh-are-you,s,1) <- I'd avoid the space there too... |
18:29.49 | Mon|A|rch | oh, there was a 1 priority, it just wasn't necessary |
18:29.51 | Mon|A|rch | deleted it for the paste |
18:29.52 | Mon|A|rch | sorry |
18:30.13 | Mon|A|rch | been amended |
18:30.23 | Mon|A|rch | I've tested that extension by dialing internally |
18:30.28 | Mon|A|rch | and it works pretty much as advertised |
18:30.29 | [TK]D-Fender | Mon|A|rch, Show a new PB with the relevant bits as well as precisely what you're checking for and status dumps to match |
18:30.32 | Mon|A|rch | I'll avoid the spacing |
18:30.41 | [TK]D-Fender | we need to see the error |
18:31.03 | Mon|A|rch | alright |
18:31.28 | Mon|A|rch | also, i don't know what a PB is |
18:32.17 | [TK]D-Fender | PASTEBIN |
18:32.30 | Mon|A|rch | sorry |
18:34.30 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
18:34.56 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
18:35.21 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
18:36.28 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
18:36.55 | Mon|A|rch | http://pastebin.com/e6TbwnRK |
18:37.28 | Mon|A|rch | [TK]D-Fender, let me know if that's not enough information |
18:38.59 | jacekowski | ehh, i've got a sip provider that uses multiple ip addresses for incoming calls |
18:39.12 | jacekowski | as in, i may register with .65 and then have stuff coming from .66 |
18:39.14 | [TK]D-Fender | Mon|A|rch, What's the actual goal? |
18:39.35 | jacekowski | and asterisk is rejecting packets from that provider |
18:40.11 | qakhan | [TK]D-Fender web.cfg and license.cfg is not included in their package |
18:40.57 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
18:41.26 | [TK]D-Fender | qakhan, Who said it was supposed to? |
18:41.30 | Mon|A|rch | the eventual goal is to originate a call from asterisk to a phone outside the network, when they pickup, they get led through the IVR, and enter answers to questions n' things |
18:41.37 | [TK]D-Fender | qakhan, How do I know which one you even grabbed? |
18:41.45 | Mon|A|rch | i can dial into the extension from outside the network successfully |
18:41.56 | [TK]D-Fender | Mon|A|rch, Ok, and the "extension_state" bit |
18:41.57 | [TK]D-Fender | ? |
18:42.13 | Mon|A|rch | making sure the channel isn't congested or unavailable |
18:43.10 | qakhan | [TK]D-Fender do you have? |
18:43.32 | Mon|A|rch | am I being dumb here? |
18:43.32 | [TK]D-Fender | qakhan, You are not being informative at all.... |
18:43.46 | [TK]D-Fender | Mon|A|rch, It's always available ... it's just dialplan.... |
18:43.58 | [TK]D-Fender | Mon|A|rch, You can 1000 calls there if you want as far as I can see... |
18:44.04 | [TK]D-Fender | Mon|A|rch, What do you see that should limit you? |
18:45.06 | [TK]D-Fender | jacekowski, Add another peer or match by name for type=user |
18:45.14 | Mon|A|rch | it's only asterisk informing me that the extension is unavailable |
18:45.23 | Mon|A|rch | I'll remove that check, and see if the calls work |
18:45.38 | Kobaz | do de do... so i'm back |
18:46.42 | Kobaz | configuring service pack, do not turn off your computer |
18:49.21 | Mon|A|rch | [TK]D-Fender |
18:49.35 | Mon|A|rch | had some errors, will pastebin |
18:53.16 | Mon|A|rch | http://pastebin.com/rsevQ8jr |
18:53.33 | Mon|A|rch | the calls are falling through |
18:54.43 | [TK]D-Fender | Mon|A|rch, and everything around it please... I see a call landing on s@default... not the "500" you showed earlier |
18:54.49 | *** join/#asterisk camelCase (~camelCase@unaffiliated/camelcase) |
18:54.56 | Mon|A|rch | oh, sorry |
18:55.38 | camelCase | anyone see this: http://blog.exodusintel.com/2013/01/07/who-was-phone/ |
18:56.41 | Qwell | camelCase: It was sent to the asterisk-announce mailing list, so yes, everybody using Asterisk saw it. Unless they're bad and not subscribed to that list. |
18:56.45 | Mon|A|rch | http://pastebin.com/sSJxR9Ww |
18:56.49 | Mon|A|rch | [TK]D-Fender, there you go |
18:56.52 | qakhan | [TK]D-Fender i downloaded UC_Software_4_0_3F_release_sig_split from Polycom |
18:56.56 | Mon|A|rch | i appreciate your patience |
18:57.11 | Mon|A|rch | i realize this is probably nerve-grinding |
18:57.45 | qakhan | then i made copy of phone1.cfg from internet |
18:57.50 | [TK]D-Fender | Mon|A|rch, Only for you... np here... |
18:58.01 | Mon|A|rch | fair enough |
18:58.06 | Mon|A|rch | thanks anyway though |
18:58.11 | [TK]D-Fender | == Starting SIP/10.3.1.1-00000000 at default,,1 failed so falling back to exte n 's' |
18:58.19 | [TK]D-Fender | that's clearly not good... |
18:58.25 | [TK]D-Fender | I need to see your originate, etc... |
18:58.33 | Mon|A|rch | okay |
18:58.45 | [TK]D-Fender | ALL the backup for what started this call and everything you think it should be doing... |
18:59.00 | qakhan | not when i upload to phone this require MAC-web.cfg and MAC-license.cfg |
19:01.04 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
19:05.08 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
19:10.43 | Mon|A|rch | [TK]D-Fender, http://pastebin.com/pkUjrF2s |
19:10.50 | spriggan | hi every one ,, some one can help me ? im using the manager api to get the QueueStatusComplete info.. i need to know if the wait time is the time since the call gets into the queue or if it's since the call gets into the ivr |
19:11.06 | Mon|A|rch | I can paste the entire method, but that's the meat of it |
19:11.24 | Mon|A|rch | later on it'll take different extensions and contexts for different functionality |
19:11.30 | Mon|A|rch | but i need to get originating working first |
19:12.12 | Mon|A|rch | oh |
19:12.12 | Mon|A|rch | whoa |
19:12.14 | Mon|A|rch | captcha |
19:12.17 | Mon|A|rch | wasn't paying attention |
19:12.18 | Mon|A|rch | sorry |
19:12.30 | Mon|A|rch | http://pastebin.com/pkUjrF2s |
19:12.31 | Mon|A|rch | there you go |
19:13.55 | Mon|A|rch | users.conf was modified by the guy who started the project, i can rename the file for now if you think it's messing with the sip config |
19:14.54 | [TK]D-Fender | Mon|A|rch, Your original dialplan showed me extension 500 |
19:15.03 | [TK]D-Fender | ttp://192.168.3.103:8088/asterisk/rawman?action=originate&channel=sip/10.3.1.1/918059941511&extension=500&context=default&priority=1 |
19:15.06 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
19:15.19 | [TK]D-Fender | Mon|A|rch, And this isn't pointing to the same context at all |
19:15.33 | [TK]D-Fender | Mon|A|rch, "default" != "incoming" |
19:15.48 | [TK]D-Fender | Mon|A|rch, Which is probably the mistake right of the bat |
19:15.56 | Mon|A|rch | facepalms to death |
19:17.14 | Mon|A|rch | hm |
19:17.18 | Mon|A|rch | still getting the same error |
19:17.19 | Mon|A|rch | hold on |
19:17.24 | *** join/#asterisk FLeiXiuS (~FLeiXiuS@unaffiliated/fleixius) |
19:17.51 | fullstop | Mon|A|rch: This is your next option: http://i.imgur.com/3L3r2.gif |
19:18.57 | danfromuk | Hi, Ive got a client that received fax calls over ISDN (ports 1 and 2 on an ISDN card) and routes them out of port 4 to a fax server running on a windows machine. Its not working. How can I go about diagnosing it? Any tips? I've only ever dealt with SIP. |
19:19.10 | danfromuk | Normal calls are coming in fine. |
19:19.44 | Mon|A|rch | it's still trying to work with the default context >< |
19:19.56 | [TK]D-Fender | PB |
19:20.22 | Mon|A|rch | k |
19:20.58 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
19:22.40 | *** join/#asterisk lifeforms (~walter@tau.lfms.nl) |
19:22.49 | Mon|A|rch | http://pastebin.com/QFLxLZ22 |
19:22.53 | Mon|A|rch | [TK]D-Fender, k |
19:23.18 | [TK]D-Fender | qakhan, It doesn't come with the provisioning. It comes when you LICENSE the phones. |
19:23.42 | Mon|A|rch | it's like it doesn't see my extension |
19:23.50 | Mon|A|rch | "failed at incoming,,1" |
19:24.06 | [TK]D-Fender | <PROTECTED> |
19:24.13 | [TK]D-Fender | now no EXTENSION |
19:24.28 | Mon|A|rch | hm |
19:24.35 | [TK]D-Fender | incoming (comma) (comma) 1 |
19:24.44 | [TK]D-Fender | Mon|A|rch, Watch for more typos |
19:25.06 | Mon|A|rch | it all looks good, vardumping, hold on |
19:25.23 | [TK]D-Fender | Mon|A|rch, I see no reason for it to be variable so far at all... |
19:25.30 | [TK]D-Fender | Mon|A|rch, You have a fixed target it seems... |
19:26.06 | *** join/#asterisk Russ (~russ@md20536d0.tmodns.net) |
19:26.15 | Mon|A|rch | it'll be variable later |
19:26.22 | Mon|A|rch | but for now I've set it static at 500 |
19:27.01 | [TK]D-Fender | Apparently you "oopsed" in there somewhere... |
19:27.33 | Mon|A|rch | apparently |
19:27.35 | lifeforms | I'm having trouble with iax.conf, a new section seems to 'inherit' settings from my VOIP provider.. I have a [providername] sort of 'template' as received from my VOIP provider, I'm using it for redundancy as in [prov01](providername) and [prov02](providername)... now my new [office] section doesn't work because when the office connects, I am getting: 'chan_iax2.c:11015 socket_process: Host 192.0.2.1 failed to authenticate as providername' even though that |
19:27.53 | *** join/#asterisk Docfxit (~Docfxit@netblock-75-79-6-10.dslextreme.com) |
19:27.58 | Qwell | lifeforms: pastebin your config |
19:28.07 | [TK]D-Fender | ~pb |
19:28.07 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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 install pastebinit with yum or aptitude. |
19:28.11 | [TK]D-Fender | lifeforms, ^^^ |
19:28.13 | lifeforms | ok hold on :) |
19:28.44 | Mon|A|rch | [TK]D-Fender, http://pastebin.com/hgWgJvj7 |
19:28.49 | Mon|A|rch | that's all i have for the originate string |
19:28.56 | Mon|A|rch | after that it's all CURL |
19:29.45 | [TK]D-Fender | Mon|A|rch, more complete code, more complete call please. |
19:29.54 | Mon|A|rch | alright |
19:30.27 | lifeforms | http://pastebin.com/urcUyadL there goes! |
19:31.19 | lifeforms | so my problem is, 'office' connects great but my home asterisk says: Host <ip> failed to authenticate as speakup-lifeforms |
19:31.47 | lifeforms | and I am a bit puzzled why it would even expect that :) but I don't know that "(speakup-lifeforms)" notation at all, my iax provider gave it to me, and it's very hard to google on ( ) :) |
19:31.51 | Qwell | lifeforms: (!) makes something a template |
19:32.03 | Qwell | [speakup-lifeforms](!) |
19:32.08 | lifeforms | (!) |
19:32.14 | lifeforms | aha... |
19:32.18 | Qwell | I doubt it'll actually fix your problem, but it's b0rked |
19:32.29 | Mon|A|rch | http://pastebin.com/3ijP9si6 |
19:32.33 | FLeiXiuS | Will events after my dialplan hangup occur? |
19:32.43 | lifeforms | let's see what happens if I make that a proper template then |
19:32.58 | Mon|A|rch | [TK]D-Fender, I'm sure this is going to be some horrible typo somewhere ;_; |
19:33.49 | lifeforms | okay, something did change :) |
19:33.55 | [TK]D-Fender | Mon|A|rch, extension=500 <--- IIRC this is supposed to be "exten", not "extension" |
19:34.09 | Mon|A|rch | lol |
19:34.20 | *** join/#asterisk elico (~Thunderbi@109.67.209.186) |
19:34.29 | Mon|A|rch | rofl |
19:34.34 | lifeforms | I did [speakup-lifeforms](!) now, this seems better. but now, when 'office' tries to connect, it says: Host <ip> failed to authenticate as speakup01 :') |
19:34.48 | lifeforms | the sarcastic smiley was not part of the log message |
19:35.06 | lifeforms | so [office] seems to reuse some part of the earlier section |
19:35.35 | Mon|A|rch | [TK]D-Fender, it all works no |
19:35.48 | [TK]D-Fender | Mon|A|rch, Please go stand on that large plastic sheet in the corner while I grab my pistol..... |
19:35.50 | [TK]D-Fender | :p |
19:36.02 | Mon|A|rch | rofl |
19:36.12 | Mon|A|rch | seriously |
19:36.13 | [TK]D-Fender | hated the cleaning bill from the last time he forgot the sheet |
19:36.20 | [TK]D-Fender | Mon|A|rch, You're welcome. |
19:36.45 | Qwell | lifeforms: should office maybe be a peer, and not a user? |
19:37.28 | lifeforms | Qwell: could be nice in the future, but for now, I just want office to reach me at home |
19:38.12 | Docfxit | I would like to record a voice prompt. I added extension 130 to extensions.conf. The lines I added are at pastebin.com/t3Yqgqzp I have restarted Asterisk. It isn't working. Could someone please suggest what I need to do to get it working? |
19:38.43 | Docfxit | It isn't recording the file. |
19:39.01 | citrusfizz | Qwell: you asked if i had a timer on the room, but i don't see any place to set that in the asterisknow web interface |
19:40.07 | navaismo | Docfxit: did you mis the SET in the pastebin? |
19:40.13 | navaismo | or is actuallñy your diaplan |
19:41.02 | Docfxit | navaismo: on the recordedfile line? |
19:41.07 | navaismo | yes |
19:41.22 | [TK]D-Fender | Docfxit, RecordedFile=/tmp/asterisk-recording <---- this is not a dialplan extension calling Set() ...... |
19:41.43 | lifeforms | I could get rid of the templates altogether, I guess, let's try that first |
19:42.55 | ghost75 | is it not possible to hold call on specific channels over ami ? |
19:43.39 | ghost75 | found only mute |
19:44.23 | Docfxit | Does this look better? pastebin.com/7rJ4fkKG |
19:45.13 | navaismo | same => n,Set(RecordedFile=/tmp/asterisk-recording) |
19:45.17 | [TK]D-Fender | Docfxit, NO |
19:45.33 | [TK]D-Fender | Docfxit, EXTEN => ............ |
19:45.59 | [TK]D-Fender | Docfxit, It is a dialplan app. It requires all the same basic formatting as every other line of dialplan |
19:49.49 | lifeforms | okay, removing the templates didn't help, still my new iax2 connection 'failed to authenticate as speakup01' (the topmost section).. but if I remove [speakup01] and [speakup02] sections from iax.conf, office can connect! |
19:49.57 | lifeforms | very interesting |
19:56.28 | *** join/#asterisk Penguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
19:58.09 | Qwell | lifeforms: should office maybe be a peer, and not a user? |
20:01.31 | qakhan | [TK]D-Fender can you tell me how to setup auto answer on exts |
20:01.52 | lifeforms | Qwell: if I remember correctly, the receiving machine would call the caller a 'user', right? |
20:02.03 | lifeforms | but let's try something else, yeah |
20:02.03 | Penguin | You don't want auto-answer on extensions. |
20:02.05 | Penguin | You want auto-answer on PHONES. |
20:02.29 | Penguin | The PHONE is the important part. |
20:02.31 | [TK]D-Fender | qakhan, http://www.voip-info.org/wiki/view/Polycom+auto-answer+config |
20:02.43 | [TK]D-Fender | qakhan, for POLYCOM's only |
20:02.46 | FLeiXiuS | What would be the cause of my extension going from a Hangup to the beginning of an extension? It goes into the extension name H then back into S after it hangs up. |
20:03.03 | [TK]D-Fender | qakhan, Every phone maker that even offers the feature does it differently. |
20:03.19 | Penguin | fleixius: What does h have in it? What else is in the context? |
20:03.20 | [TK]D-Fender | FLeiXiuS, Show us |
20:03.38 | Penguin | Extension patterns like _. are BAD. |
20:05.04 | FLeiXiuS | [TK]D-Fender, Penguin http://pastie.org/5650059#18 |
20:05.27 | [TK]D-Fender | FLeiXiuS, and the call....... |
20:06.45 | FLeiXiuS | [TK]D-Fender, http://pastie.org/5650068 |
20:07.05 | [TK]D-Fender | FLeiXiuS, the COMPLETE call |
20:07.18 | FLeiXiuS | Bah! Woops |
20:08.48 | Penguin | I'd also like to know the context of those extensions that you showed. |
20:10.32 | FLeiXiuS | [TK]D-Fender, http://pastie.org/5650092 |
20:13.55 | [TK]D-Fender | FLeiXiuS, Apparently calling "hangup" in "h" = bad |
20:14.50 | FLeiXiuS | [TK]D-Fender, That what I was going too..but how else would I execute a quit..or a stop? Not putting anything at all. |
20:14.55 | FLeiXiuS | ? * |
20:15.15 | [TK]D-Fender | FLeiXiuS, Run out . |
20:15.16 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
20:15.26 | lifeforms | Qwell: okay, apparently the problem goes away when each section has a 'username='... I am guessing when it's not set, asterisk (incorrectly?!) uses the username= setting of a former iax.conf section... pretty strange! |
20:15.29 | FLeiXiuS | Penguin, The context of this call is tricky ;-P |
20:15.44 | Penguin | No it isn't. |
20:15.59 | FLeiXiuS | Penguin, It's doing some silly things, whcih makes it unlike a phone call |
20:16.25 | Penguin | The call can only have one context per extension involved. |
20:16.32 | Penguin | There are two extensions. |
20:16.32 | FLeiXiuS | [TK]D-Fender, I'll try without the hangups within the 'h' |
20:17.32 | FLeiXiuS | Penguin, how so? |
20:17.58 | Mon|A|rch | [TK]D-Fender, any idea how to improve call quality on outbound calls? I'm in the us, so I'd imagine i should use ulaw, and disable alaw |
20:18.00 | Penguin | While I can see the context of 'control' being asserted, I doubt that is the context within which the extension actually resides. I feel there is an include involved. |
20:18.10 | Mon|A|rch | from what i understand they're meant for us and european connections respectively |
20:18.24 | [TK]D-Fender | Mon|A|rch, ULAW vs ALAW should have no real quality difference at all. |
20:18.38 | Mon|A|rch | hm |
20:18.41 | Mon|A|rch | gsm? |
20:19.01 | Mon|A|rch | do i need to poke around in my provider server? |
20:19.04 | [TK]D-Fender | Mon|A|rch, If it's stuttery it could be network jitter or bandwidth. May sure other things don't cut in on that (QoS), etc |
20:19.09 | Mon|A|rch | I just sort of assumed this was an asterisk issue |
20:19.13 | FLeiXiuS | Penguin, The call is made through AMI, directly to this context / 's' extension. |
20:19.25 | Mon|A|rch | good call |
20:19.29 | FLeiXiuS | I believe removing the Hangup() from the 'h' exten worked. |
20:19.36 | Penguin | The main difference between ulaw and alaw in the US is that almost no one in the US uses alaw. |
20:20.00 | Penguin | fleixius: So the call goes to s@control, then. |
20:20.09 | [TK]D-Fender | the companding between ULAW and ALAW is almost identical and transcoding shows hardly nay quality loss at all. |
20:20.20 | FLeiXiuS | Penguin, Correct. |
20:20.31 | Mon|A|rch | good to know |
20:20.42 | Penguin | That broke a theory I was trying to work on. |
20:21.07 | Mon|A|rch | btw, when I'm passing in my originate string, can i set channel variables? I'd like to be able to get transfers to whomever opens a session on the front-end |
20:21.17 | Mon|A|rch | their sessions have their particular extension |
20:21.32 | [TK]D-Fender | Mon|A|rch, &setvar=var=value |
20:21.32 | Mon|A|rch | is it something like: "variable=var1=5|var2=4"? |
20:21.40 | Mon|A|rch | k |
20:22.15 | Mon|A|rch | then access it like s,1,goto(incoming,${var},1)? |
20:22.24 | Mon|A|rch | or dial or w/e |
20:22.26 | FLeiXiuS | Penguin, [TK]D-Fender Removing the Hangup from the 'h' exten worked. I guess it was acting as it was intended too. |
20:23.06 | [TK]D-Fender | Mon|A|rch, yup |
20:23.27 | Mon|A|rch | cool beans |
20:23.55 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
20:24.59 | Docfxit | I am still having trouble getting it to record a prompt the latest version is at pastebin.com/ukN4WAv2 |
20:25.58 | navaismo | why? what sows the cli |
20:26.12 | [TK]D-Fender | Docfxit, Why are you not showing us the FAILURE? |
20:26.28 | Docfxit | I'd be happy to. |
20:26.43 | Docfxit | Let me see if I can get that. |
20:26.48 | [TK]D-Fender | "if"? |
20:27.27 | [TK]D-Fender | You could be hitting a failure than isn't directly related to your code, but rather the working environment. |
20:27.48 | [TK]D-Fender | Which makes this the equivalent of showing a brochure for a car ... and then why asking why YOURS crashed. |
20:28.35 | Mon|A|rch | [TK]D-Fender, after a call is originated, do i need to use answer() before dialing someone to put on the other end? |
20:28.56 | Mon|A|rch | I'm not entirely sure what answer does, guessing just general config for the call |
20:29.03 | [TK]D-Fender | Mon|A|rch, Depends exactly where, and exactly why. |
20:29.13 | [TK]D-Fender | Mon|A|rch, There are times you'll want to, others you specifcally don't |
20:29.18 | Penguin | It answers the line, as strange as that may sound. |
20:30.04 | Mon|A|rch | well, i suppose since i don't have a specific reason not to use it, I'll use it. |
20:30.18 | [TK]D-Fender | Usually one would do the REVERSE |
20:30.22 | Penguin | Your thinking is backward. |
20:30.24 | Mon|A|rch | all this extension will do is call someone out in the world, then patch them through to a CSR that will try to sell them crap |
20:30.26 | [TK]D-Fender | You do things because you need to.... |
20:30.35 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
20:30.42 | Mon|A|rch | well, generally i assume i need to answer the line |
20:30.48 | Penguin | Let the callee answer it. |
20:31.06 | Mon|A|rch | makes sense |
20:33.12 | [TK]D-Fender | Mon|A|rch, It can affect billing, success codes, retries, etc... |
20:34.09 | Docfxit | As I am sure you can tell I don't work with Asterisk every day. (Actually very rarely) It's been very solid and I don't make changes very often. I'm sorry I'm not more experienced with asterisk. I am in root now. I entered asterisk -vvvvvvr. I made the call. It doesn't show in the terminal window. Isn't that the correct way to show cLi? |
20:34.34 | Mon|A|rch | good to know tk |
20:37.12 | [TK]D-Fender | Docfxit, If you connect via that command as listed and see nothing then your call isn't even being processed |
20:37.27 | navaismo | Docfxit: try with the cmd: core set verbose 3 |
20:38.13 | [TK]D-Fender | "I entered asterisk -vvvvvvr" <- that's already *6* |
20:40.28 | navaismo | LOL |
20:40.34 | navaismo | missed that |
20:46.43 | Docfxit | navaismo: I'm getting command not found. I am running an old version 1.4.22. I do have a new computer that I am almost ready to load the latest version on. |
20:47.38 | Penguin | Disregard his suggestion anyway. |
20:47.45 | navaismo | yep ^ |
20:49.12 | [TK]D-Fender | Docfxit, "set verbose 10 |
20:52.11 | Penguin | Set it to 9000, if you want. |
20:53.18 | FLeiXiuS | [TK]D-Fender, Thanks btw. |
20:53.30 | [TK]D-Fender | FLeiXiuS, np |
20:57.37 | JustinAiken | Anyone know how to sort out occasional DTMF-doubled tones? |
20:57.56 | JustinAiken | Looking at a tcp dump in wireshark, it looks like we occasonally get a DTMF end packet that shouldn't be there, |
20:58.04 | JustinAiken | giving us really short tones (30ms) or so |
20:58.10 | JustinAiken | is there a way to reject those? |
21:01.27 | Docfxit | I'm at root@ubuntuAsterisk:~# set verbose 9000 I made the call. Nothing is showing in terminal. |
21:01.57 | Penguin | That is not the asterisk cli. |
21:02.56 | Docfxit | Oh good. How can I get to the cli? |
21:03.07 | Penguin | asterisk -r |
21:03.31 | navaismo | you said "asterisk -vvvvvvr" |
21:03.39 | navaismo | do it again |
21:05.09 | Docfxit | When I put in asterisk -r I get what is in pastebin.com/cJFVcHVR |
21:05.33 | Penguin | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
21:05.46 | Penguin | That means asterisk might not be running! |
21:05.55 | Docfxit | There is a file in /var/run/asterisk/asterisk.ctl |
21:06.08 | Docfxit | That is the only file in that folder. |
21:06.35 | Penguin | pgrep asterisk |
21:06.41 | Penguin | Does it return the PID? |
21:07.11 | Docfxit | Yes. It returns two pid's. |
21:07.29 | Docfxit | 5367 and 5377 |
21:08.03 | Penguin | ps -C asterisk u |
21:08.09 | Penguin | Which user/uid is asterisk running under? |
21:09.05 | Docfxit | root 5377 |
21:09.16 | Penguin | Root? Good grief. |
21:09.38 | Penguin | Who is the owner of /var/run/asterisk/asterisk.ctl? |
21:11.01 | Docfxit | The /var/run/asterisk folder was there last night. It isn't there today. |
21:11.26 | Penguin | Create it and then restart asterisk. |
21:12.19 | Docfxit | I am currently signed in as root. Is it ok to create it with the user root? |
21:12.32 | Penguin | Yes. |
21:12.56 | Penguin | I'm not a big fan of your running asterisk with root, but that's another lesson. |
21:13.42 | *** join/#asterisk ThomasLocke (~ThomasLoc@pdpc/supporter/active/thomaslocke) |
21:16.01 | Docfxit | Hopefully soon I will find someone to help me build another box correctly and these problems will go away. |
21:16.16 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
21:16.45 | Penguin | For an easy deployment, I would use AsteriskNOW with the "No GUI" install option. |
21:17.12 | Penguin | It's been a while since I did it, but it went rather well that last time I did it. |
21:17.45 | Docfxit | Why the "No GUI" |
21:18.06 | *** join/#asterisk pbxbrian (~pbxbrian@79.97.2.26) |
21:18.23 | Penguin | GUIs are junk, they break things, make it impossible to work out some problems, and we don't support them here. |
21:18.37 | Docfxit | ok. |
21:18.59 | Docfxit | Someone else suggested I go with FreePBX |
21:19.09 | Penguin | I'm sure they did. |
21:19.18 | carrar | FreeSwitch! |
21:19.25 | Chainsaw | Docfxit: Then it's up to the someone else to do your FreePBX support work really. |
21:19.50 | Docfxit | I'd rather be here. |
21:20.14 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
21:23.44 | Docfxit | I created the folder /var/run/asterisk I re-booted, The folder isn't there now. |
21:24.04 | Penguin | I don't remember advising you to reboot. |
21:24.04 | [TK]D-Fender | I think you should look at what you need out of yoru system, who'll maintain, it and prioritize accordingly/. |
21:25.26 | *** join/#asterisk Russ (~russ@md20536d0.tmodns.net) |
21:27.02 | Docfxit | Penguin: I'm sorry. You asked me to restart asterisk and since asterisk -r gives me an error the only way I know how is to re-boot. I will re-create the folder now. |
21:28.10 | Docfxit | I currently have the folder /var/run/asterisk. How should I restart asterisk? |
21:30.06 | Penguin | With the directory existing, start or restart asterisk. |
21:32.26 | Docfxit | The directory does exist now. I tried sudo asterisk -vvvvvvr It didn't create a file in /var/run/asterisk It did give me the error saying asterisk.ctl doesn't exist. |
21:32.52 | Penguin | When you start or restart asterisk, it will output information. |
21:33.09 | Penguin | If it cannot create the ctl file, it should say so. |
21:33.45 | Penguin | How are you starting and/or restarting asterisk? |
21:34.33 | Docfxit | The information it outputs is in pastebin.com/jz4pxsTv |
21:35.04 | Docfxit | I'm restarting it with sudo asterisk -vvvvvvr |
21:35.10 | Penguin | That isn't starting OR restarting asterisk. |
21:35.19 | Penguin | And you're running that command as a user anyway. |
21:35.33 | Penguin | asterisk -r connects to a RUNNING ASTERISK. |
21:35.40 | Penguin | You have to actually be running asterisk first. |
21:36.48 | Docfxit | I believe asterisk is running. When I call it answers the phone, plays the prompts and lets me connect to an extension. |
21:36.52 | Penguin | sudo -i |
21:36.54 | Penguin | /etc/init.d/asterisk restart |
21:36.56 | Penguin | asterisk -r |
21:37.34 | Penguin | You've failed to restart asterisk. That was key in the instructions. |
21:38.32 | Docfxit | I'm sorry. |
21:38.44 | Docfxit | I just ran the commands you gave me. |
21:39.51 | Docfxit | The results are in pastebin.com/FyVrudeu |
21:40.12 | Penguin | Does the file exist? |
21:40.31 | Penguin | Does the directory still exist? |
21:41.10 | Docfxit | Nothing is in /var/run/asterisk |
21:41.41 | Penguin | Pastebin the contents of your asterisk.conf. |
21:42.13 | Docfxit | Yes the directory does exist. |
21:43.57 | Docfxit | It's in pastebin.com/myXN6QLc |
21:45.40 | Docfxit | Is the astrundir => supposed to be /var/run/asterisk or is it supposed to be /var/run? |
21:46.17 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-mangkwmxpwbejrkx) |
21:51.58 | *** join/#asterisk Docfxit (~Docfxit@netblock-75-79-6-10.dslextreme.com) |
21:54.31 | *** join/#asterisk k1920 (~k1920@160-6.5-85.cust.bluewin.ch) |
21:55.43 | *** part/#asterisk navaismo (~navaismo@189.191.2.44) |
21:57.29 | JustinAiken | anyone know a fix for occasional doubled DTMF tones? |
22:01.31 | k1920 | more informations about "occasional"? always same phone/number? always same time? do you have a single line? did you try a verbose debug? |
22:01.46 | *** join/#asterisk navaismo (~navaismo@189.191.2.44) |
22:02.27 | JustinAiken | about 1 in 10 digits will be doubled, any number |
22:02.35 | JustinAiken | so you type 1234 into the phone, get 122345 |
22:02.50 | JustinAiken | seems nearly completely random |
22:03.22 | JustinAiken | like in this: https://gist.github.com/c7b483f9cd5dc37e7a33 |
22:04.30 | Penguin | /var/run/asterisk would be good. |
22:04.39 | lifeforms | byeee! |
22:04.50 | Docfxit | Penguin: great. |
22:05.00 | Docfxit | That's where it is. |
22:06.33 | Docfxit | Penguin: It sounds like it should be writing the file asterisk.ctl to that folder? |
22:07.23 | Penguin | well, directory, but yes. |
22:08.18 | Penguin | Go ahead and kill off any asterisk processes you have. Kill them forcefully if necessary. |
22:09.20 | Docfxit | With pkill asterisk ? |
22:09.34 | Penguin | That should be fine, yes. |
22:10.00 | Penguin | Be sure they are gone using pgrep or any other way you know how to ensure asterisk is dead. |
22:12.13 | Docfxit | pgrep asterisk doesn't return anything and the phone doesn't answer. So it's dead. |
22:12.54 | k1920 | Well done Penguin.. little tired.. bye all |
22:13.48 | Penguin | Now start asterisk using the following command: asterisk -vvvvddddc |
22:13.53 | Docfxit | Penguin: Thank you for sticking with me. |
22:14.36 | Docfxit | I am now into cli. Yey!!! |
22:14.46 | Penguin | I don't know about that. |
22:14.54 | *** join/#asterisk charley (charley@epicboise.com) |
22:14.58 | Penguin | You should have just started asterisk and that's all. |
22:16.02 | Docfxit | There is a lot scrolling on the screen. |
22:16.07 | Penguin | Once that is running, then, in another terminal, find out if the asterisk.ctl file exists. |
22:16.47 | charley | I know this isn't the right channel.. but I can't find any Broadsoft users. My VoIP engineer needs to convert a *.vml file to .wav, or something that doesn't need a 3rd party player to listen to it. Anyone know where I can get one? He's scoured the web and hasn't had any luck. |
22:17.30 | Penguin | I recall running asterisk 1.4.22 and there was no problem with the socket file. |
22:17.44 | Docfxit | Penguin: asterisk.ctl is in there and asterisk.pid |
22:17.51 | Penguin | Great! |
22:17.58 | Penguin | So asterisk isn't broken. |
22:18.11 | Docfxit | It's been a while since I've seen them. |
22:18.13 | Penguin | It's something else. Could be something in the init fiel. |
22:18.22 | Penguin | s/fiel/file/ |
22:19.25 | navaismo | or selinux |
22:19.32 | Docfxit | What directory is the init file in? |
22:19.35 | Penguin | Not on ubuntu. |
22:19.42 | Penguin | /etc/init.d/ |
22:20.42 | Penguin | Ubuntu does have some security thing, though. I think it's called App Armor. |
22:20.54 | Penguin | doesn't use ubuntu. |
22:21.47 | Docfxit | What do you use? |
22:22.00 | Penguin | Primarily Arch Linux. |
22:22.07 | Penguin | But I don't mind CentOS. |
22:22.45 | charley | anything is better than ubuntu, really. |
22:23.34 | Penguin | Amen. |
22:24.58 | charley | Was it Ubuntu's logo that was so appealing when it blew up? I don't understand. Was it firefox fanboys that were on Ubuntu's nuts? |
22:26.08 | sruffell | I think it was it's willingness to make proprietary software available…especially when the open source video drivers were not existent or lacking. |
22:26.20 | Penguin | its |
22:26.28 | sruffell | thanks |
22:26.46 | charley | lol |
22:27.03 | Docfxit | There are a few warnings in the cli. pastebin.com./iwPUuFYT |
22:27.17 | Penguin | Warnings are usually okay. Errors are bad. |
22:27.40 | Docfxit | I will call the extension now to see what is happening. |
22:28.43 | Docfxit | Asterisk isn't answering the phone. |
22:32.07 | *** join/#asterisk navaismo (~navaismo@189.191.2.44) |
22:32.22 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
22:33.57 | ghost75 | did you know there will be ubuntu phone? |
22:34.08 | *** join/#asterisk Russ (~russ@md20536d0.tmodns.net) |
22:35.19 | Docfxit | Now I really have a problem because the phone isn't answering. |
22:47.57 | Docfxit | I restarted asterisk. I called the extension 130. I found a warning. You can see it at pastebin.com./pGbVpGC7 |
22:48.50 | [TK]D-Fender | Docfxit: paste FULL URLS's |
22:49.01 | [TK]D-Fender | we have to copy/paste it manually all the time |
22:49.07 | navaismo | you dont have an extension 13 in your context voicemenu-custom-5 |
22:49.09 | Qwell | ^^^^^^ |
22:49.15 | navaismo | not mine |
22:49.37 | Docfxit | I have included the new Voice_Pronpt_That_I_Recorded in voicemenu-custom-5 |
22:49.43 | [TK]D-Fender | docfxit: [Jan 8 15:02:11] WARNING[6134]: pbx.c:2514 __ast_pbx_run: Invalid extension '13', but no rule 'i' in context 'voicemenu-custom-5' -- Hungup 'DAHDI/7-1' |
22:49.47 | Qwell | also since when do browsers allow a trailing . in the domain? That's funky. |
22:49.53 | [TK]D-Fender | Docfxit: that is 13 ... NOT 130 |
22:50.09 | Docfxit | It should be 130 |
22:50.40 | [TK]D-Fender | Docfxit: It isn't Pastebin entire calls, not tiny broken little snippets |
22:50.54 | [TK]D-Fender | Docfxit: And we are now in completely different contexts than before. |
22:51.05 | [TK]D-Fender | Docfxit: Be thorough in the configs and debug you show us. |
22:52.46 | Docfxit | Ok. |
22:53.41 | Docfxit | I tried starting asterisk with asterisk -vvvvvvr It removed the files from /var/run/asterisk. |
23:01.19 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
23:02.06 | Docfxit | When I use asterisk -vvvvddddc Asterisk will start and the files come back |
23:03.56 | Docfxit | I tried a new call to extension 130. Pastebin.com says I reached my limit of 10 pastes per 24 hr. I guess I can sign up? |
23:04.47 | JustinAiken | gist.github.com |
23:14.46 | Docfxit | JustinAiken: Thank you. |
23:15.25 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
23:15.36 | [TK]D-Fender | ~pb |
23:15.36 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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 install pastebinit with yum or aptitude. |
23:15.46 | [TK]D-Fender | 4 more right there |
23:18.22 | Docfxit | I started Asterisk with asterisk -vvvvddddc I called x130 I pasted the results at https://gist.github.com/4488923 You will see the problem with x130 and asterisk is shutting down.. Why? |
23:19.24 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
23:23.13 | kaldemar | Docfxit: what problem? |
23:24.07 | [TK]D-Fender | Docfxit: Executing [130@voicemenu-custom-5:1] Answer("DAHDI/7-1", "") in new stack |
23:24.12 | [TK]D-Fender | Docfxit: It answers.... |
23:24.24 | [TK]D-Fender | Docfxit: and you have NOT shown us your dialplan. We have no idea what's in there now. |
23:24.35 | kaldemar | and falls through. |
23:24.36 | [TK]D-Fender | Docfxit: what part about "show us complete configs and debug" was unclear? |
23:24.53 | Docfxit | When I dial x130 asterisk should record a file in /tmp |
23:24.58 | [TK]D-Fender | Docfxit: How can we tell you what's wrong with your code ro call without seeing them both? |
23:25.01 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
23:25.07 | [TK]D-Fender | Docfxit: WHERE IS THE DIALPLAN? |
23:25.08 | *** join/#asterisk SeRi (~wtf@pdpc/supporter/professional/seri) |
23:26.26 | Docfxit | The dial plan in at http://pastebin.com/JJPYZwRx |
23:27.34 | [TK]D-Fender | [Voice_Prompt_That_I_Recorded] |
23:27.39 | [TK]D-Fender | <PROTECTED> |
23:27.42 | kaldemar | wrong context. |
23:27.54 | [TK]D-Fender | the context you are in has nothing to do with the dialplan you created in that OTHER context |
23:28.54 | Docfxit | I don't understand. What do I need to do to fix it? |
23:28.58 | [TK]D-Fender | voicemenu-custom-5 <---- YOU ARE HERE |
23:29.05 | Docfxit | Yes. |
23:29.10 | [TK]D-Fender | [Voice_Prompt_That_I_Recorded] <--- your CODE is here |
23:29.31 | [TK]D-Fender | If I tell you the papers are in the top shelf... and they AREN'T there. Your search is going to FAIL |
23:29.41 | Docfxit | I included Voice_Prompt_That_I_Recorded in voicemenu-custom-5 |
23:29.48 | [TK]D-Fender | Where do WE see this? |
23:30.03 | Docfxit | I will paste it. |
23:30.07 | [TK]D-Fender | \docYou seem to have a very large problem with the idea of showing us COMPLETE information |
23:30.17 | kaldemar | a misspelled include was mentioned earlier. |
23:31.04 | kaldemar | pron pt |
23:34.25 | Docfxit | http://bin.cakephp.org/view/1230663065 |
23:35.30 | kaldemar | and what do you have in [default]? |
23:35.52 | Docfxit | I will paste that for you. |
23:38.14 | kaldemar | if it has a matching extension, it will always be used first because you include it first. besides using default in the first place is often bad practice. |
23:39.20 | Docfxit | That is the way the first person that set it for me did it. |
23:39.39 | Docfxit | I have updated http://bin.cakephp.org/view/1230663065 to include default |
23:40.06 | [TK]D-Fender | Docfxit: .... |
23:40.13 | [TK]D-Fender | Docfxit: You're running 1.4.2 ... right? |
23:40.19 | Docfxit | Yes. |
23:40.23 | [TK]D-Fender | Docfxit: "same" did not even EXIST THEN |
23:40.25 | [TK]D-Fender | ^^^^^^^^^6 |
23:40.37 | [TK]D-Fender | You are using NEW config options thta do not exist at all |
23:40.57 | [TK]D-Fender | Docfxit: All of those lines are BROKEN |
23:40.59 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
23:41.49 | Docfxit | I was just following what someone told me to do yesterday. I will change it back now. Thanks for catching that. |
23:45.45 | *** join/#asterisk doctorray (~ray@72.26.99.19) |
23:47.30 | [TK]D-Fender | heads out for a few hours |
23:50.13 | Docfxit | [TK]D-Fender: Thank you for your help. |
23:52.16 | Docfxit | I have updated http://bin.cakephp.org/view/1230663065 to show the dial plan and I have the cli error here http://bin.cakephp.org/view/675566775 |
23:53.43 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
23:53.43 | *** mode/#asterisk [+o pabelanger] by ChanServ |
23:57.21 | navaismo | [TK]D-Fender, said you need to remove same and use exten |
23:57.42 | Docfxit | navaismo: I did that. |
23:58.04 | Docfxit | You can see that in my paste. |
23:58.12 | navaismo | here http://bin.cakephp.org/view/1230663065?? |
23:58.30 | navaismo | cuase im seein same string there |
23:58.42 | navaismo | seeing* |
23:59.03 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
23:59.45 | Docfxit | Sorry. It's in http://bin.cakephp.org/save/85620 |