| 00:20.29 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 00:23.17 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
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| 00:32.41 | Tozz_ | great :-\ 2 SIP clients hacked |
| 00:32.56 | Tozz_ | lots of money bye bye |
| 00:36.11 | IsUp | Tozz_: what happened? |
| 00:39.50 | Tozz_ | hacked SIP accounts |
| 00:39.51 | p3nguin | SIP clients hacked? Are you sure this is the right description for what you've experienced? I doubt it is. |
| 00:40.19 | Tozz_ | s/clients/accounts/ |
| 00:40.21 | Tozz_ | better?:) |
| 00:40.33 | IsUp | how you lost money? |
| 00:40.36 | *** join/#asterisk sosyopat (~sosyopat@78.180.106.25) |
| 00:40.47 | Tozz_ | well, my customers have lost money |
| 00:41.02 | Tozz_ | because their PBX'es have been calling all over the world |
| 00:41.24 | IsUp | premium numbers? |
| 00:42.16 | Tozz_ | just international |
| 00:42.26 | Tozz_ | Cuba mostly |
| 00:42.36 | IsUp | +53 |
| 00:43.10 | Tozz_ | yes |
| 00:43.13 | Tozz_ | and some other countries too |
| 00:43.21 | p3nguin | I guess someone should have learned how to secure their shit. |
| 00:43.35 | p3nguin | I don't feel bad for anyone who encounters this issue. |
| 00:43.47 | IsUp | so they are premium numbers or what? |
| 00:44.14 | IsUp | because premium numbers are just like usual numbers |
| 00:44.22 | IsUp | i mean international premium numbers |
| 00:44.27 | Tozz_ | but most of the time you cant call premium numbers |
| 00:44.31 | Tozz_ | from outside that country |
| 00:44.57 | Tozz_ | p3nguin: can be, but I'm getting the complaints that their invoice is so high |
| 00:45.01 | p3nguin | I believe if you can't secure your shit because you don't have the skill level, you need to recognize it and hire someone to do it for you. |
| 00:45.10 | Tozz_ | _my_ machine isnt hacked |
| 00:45.14 | Tozz_ | our customer's machine is |
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| 00:45.34 | p3nguin | First of all, nothing was hacked. |
| 00:46.13 | Tozz_ | compromised, guessed accounts |
| 00:46.13 | p3nguin | Someone had some shit passwords and some lamers/skiddies did some fancy password guessing and found a good account. |
| 00:46.14 | Tozz_ | whatever |
| 00:46.25 | Tozz_ | but I've seen hacked machines too |
| 00:46.27 | Tozz_ | mostly Trixbox |
| 00:46.43 | Tozz_ | from what i've heard they have some default account that not everyone knows about? |
| 00:46.59 | IsUp | correct |
| 00:47.01 | p3nguin | That's not hacking. There's almost no skill required to guess passwords. |
| 00:47.14 | IsUp | no |
| 00:47.17 | IsUp | its something different |
| 00:47.26 | IsUp | Trixbox has some "glory" holes |
| 00:47.37 | IsUp | since its running under root, you are able to do anything |
| 00:47.43 | IsUp | including take-over whole box |
| 00:47.50 | Tozz_ | funny how these ppl from cuba keep listening to 'tt-weasels' on repeat |
| 00:47.50 | p3nguin | Even *I* can guess passwords, and I have a pretty low skill level when it comes to that sort of thing. |
| 00:48.25 | Tozz_ | p3nguin: yes yes you made your point ;) |
| 00:48.52 | p3nguin | It just really annoys me when people incorrectly use the term "hacking." |
| 00:49.03 | IsUp | Tozz_: i guess numbers are starting with 53 319? |
| 00:49.11 | Tozz_ | let me check |
| 00:49.37 | p3nguin | How come no one ever makes expensive calls on my system? |
| 00:49.46 | Tozz_ | no, not 319 |
| 00:49.50 | p3nguin | I mean how come they never try. |
| 00:49.56 | IsUp | what is 3 digits after 53? |
| 00:50.02 | Tozz_ | well, its random |
| 00:50.12 | Tozz_ | 005372605556, 005345242328, 005372621126, etc, etc |
| 00:50.30 | Tozz_ | I see mobile destinations too, such as 005352413804 |
| 00:50.45 | p3nguin | I get the occasional attempt to register a device, but I almost never see attempts to make calls. |
| 00:51.06 | Tozz_ | and from what i've heard (i've started a ChanSpy) it seems just like 2 people talking. |
| 00:51.19 | Tozz_ | not some fancy pay per minute service or something |
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| 00:51.30 | sosyopat | maybe they're ordering cuban cigar |
| 00:51.52 | Tozz_ | could be ;) |
| 00:51.54 | IsUp | lol |
| 00:52.20 | p3nguin | It's possible those people could have paid someone for "cheap phone service," and they aren't the ones stealing your service, but the people they paid are the ones stealing it from you. |
| 00:52.31 | Tozz_ | yes I know |
| 00:52.39 | IsUp | route reselling |
| 00:52.41 | Tozz_ | we've seen it before |
| 00:52.50 | p3nguin | I'd interject in their conversation. |
| 00:53.14 | Tozz_ | and then wat? :) I dont speak their language |
| 00:53.17 | *** join/#asterisk ariel_ (~chatzilla@99-1-236-49.lightspeed.miamfl.sbcglobal.net) |
| 00:53.32 | IsUp | play some "ballsofsteel" on channels, http://www.youtube.com/watch?v=vLoxfdpaQ14 |
| 00:54.42 | Tozz_ | yes I was looking for a nice audio to play |
| 00:54.47 | Tozz_ | but all I could find was tt-weasels ;) |
| 00:55.11 | p3nguin | How are you injecting the sounds into the conversation? |
| 00:55.24 | Tozz_ | i'm not, i've changed the dialplan |
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| 00:55.35 | Tozz_ | so all international calls do Playback() |
| 00:55.48 | p3nguin | I'd probably change it to dump them into a MeetMe so I could do interesting things. |
| 00:55.54 | Tozz_ | mm |
| 00:55.58 | Tozz_ | thats also a funny idea;) |
| 00:56.17 | IsUp | put ChanSpy to whisper mode :P |
| 00:56.41 | p3nguin | With a MeetMe running, it's easy to use originate to play sounds and stuff. |
| 00:57.12 | p3nguin | With ChanSpy, you pretty much have to do things with your phone, don't you? |
| 00:57.33 | IsUp | exactly |
| 00:59.16 | alecdavis | Tozz_: in case you weren't aware the default in sip.conf is allowguest=yes there is no password for that, it may not even be in you sip.conf! If your asterisk box is accessible from the public internet, and if your default context allows dialout, then your box is available to anyone. no password required. |
| 00:59.26 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-129-161.lsanca.fios.verizon.net) |
| 00:59.35 | Tozz_ | yeah yeah but as I said.. I'm not the one hacked |
| 01:00.29 | alecdavis | fine. |
| 01:01.04 | Tozz_ | lol |
| 01:01.06 | Tozz_ | meetme is great ;) |
| 01:01.38 | p3nguin | I wouldn't mind connecting to the conference and listening to what you do to them. |
| 01:02.26 | Tozz_ | I do nothing, i'm just listening |
| 01:02.31 | Tozz_ | but I dont understand their language |
| 01:02.41 | Tozz_ | but from what I make of it, some one is looking for Maria |
| 01:02.43 | Tozz_ | but she isnt there! |
| 01:02.56 | alecdavis | record it with mixmonitor and send to translators/authorities |
| 01:03.21 | Tozz_ | authorities here dont do anything with that |
| 01:03.43 | Tozz_ | they probably wont even take my case |
| 01:05.11 | p3nguin | I'd still connect to the conf just to listen. |
| 01:05.26 | IsUp | tell the authorities that they are running drug trafficking on your PBX |
| 01:05.29 | IsUp | so they can take care |
| 01:05.45 | Tozz_ | it looks like they found out whats going on |
| 01:05.55 | Tozz_ | call volume is dropping |
| 01:06.35 | IsUp | '*sssh* pablo, get the cigars tonite, i gotta go' |
| 01:08.10 | Tozz_ | :) |
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| 03:15.11 | phix | ok so, I am trying to route incoming calls from two different account with same VoIP provider. My sip.conf, extensions.conf and a debug output can be found at --> http://bin.cakephp.org/view/891457096, any assistance will be appreciated. |
| 03:16.10 | ChannelZ | still |
| 03:16.16 | ChannelZ | have you posted a SIP debug yet? |
| 03:16.21 | phix | yes |
| 03:16.24 | phix | check out th link |
| 03:16.26 | phix | th = the |
| 03:16.45 | ChannelZ | That would be no then. |
| 03:18.47 | ChannelZ | I'm not sure what you are expecting to have happen just looking at this, without seeing what your ITSP is actually sending you. |
| 03:18.59 | phix | oh sip debug |
| 03:19.08 | phix | yes I have some of that too, I will paste it |
| 03:21.16 | ChannelZ | BBL |
| 03:22.22 | phix | aaaww |
| 03:26.20 | *** join/#asterisk JamesHarrison (~jharr@hometree.mmmetrics.co.uk) |
| 03:36.57 | phix | hmmmm, Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x110c (ulaw|alaw|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) |
| 03:37.16 | phix | They dropped GSM support :\ oh well, I don't notice it on my 6Mbit link |
| 03:37.26 | phix | bastards didn't even tell me |
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| 03:57.10 | FransWillem | Is there any way to force a dahi channel to put the hook back on? FXS channel is off-hook, and it won't reset :-/ |
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| 05:13.30 | p3nguin | This is really weird... I don't remember who I was talking with the other night about using my spare Cisco phone for travel purposes, but I connected it to the network without having the sccp config files and firmware available by tftp, and even though it tried to find them, it still loaded up and can make calls on sccp. |
| 05:15.37 | p3nguin | My 7940 and 7960 phones will never go "live" if the sccp files aren't available by tftp, so I figured I would have to change this spare phone (which is a 7912) over to SIP since the 7940/7960 can use SIP without tftpd. |
| 05:17.25 | p3nguin | I guess the 7912 behaves differently, and may just work on any network on sccp without having the tftp available to provide the files. |
| 05:18.14 | p3nguin | I just have to figure out how to tell it where my Asterisk server is. The network configuration settings on the phone do not allow me to change the call manager 1 setting, which shows my local asterisk server IP address. |
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| 06:55.09 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.3 (2011/02/28), 1.6.2.17 (2011/02/28), 1.4.40 (2011/02/28), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 06:57.56 | *** join/#asterisk Tim_Toady (~moi@77.49.3.6.dsl.dyn.forthnet.gr) |
| 07:04.22 | p3nguin | Well... that's interesting. |
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| 07:11.08 | ssh-ad | good morning |
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| 07:17.04 | p3nguin | hi |
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| 08:29.40 | scruz | heloo |
| 08:30.11 | tzafrir | hi |
| 08:30.45 | scruz | tzafrir: hi |
| 08:30.54 | scruz | uno momento. switching clients |
| 08:36.58 | *** join/#asterisk scruz (~scruz@41.139.95.10) |
| 08:37.30 | scruz | Hello again. |
| 08:40.08 | scruz | Someone just asked me if there's a way to mask passwords in Asterisk. While I told him there isn't that I know of, is there any web interface that does that? They use freepbx and it displays the full config in the clear |
| 08:44.41 | schmidts | scruz you can use md5secret instead of secret |
| 08:45.06 | schmidts | or something like this, i am not sure how this parameter relly is called but you should find it |
| 08:46.07 | Corydon76-home | scruz: No. Passwords have to be stored in the clear in order to allow the greatest security over the wire |
| 08:46.33 | Corydon76-home | If you use md5secret, you are subject to a replay attack |
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| 08:53.42 | scruz | Okay, but what about a gui that doesn't display the passwords in the clear but can manage the * config? |
| 08:54.37 | kaldemar | shouldn't be much of a stab to make freepbx mask them. |
| 08:55.26 | scruz | Freepbx displays the entire peer config in a text box |
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| 08:57.05 | *** join/#asterisk garymc (~chatzilla@host81-148-15-59.in-addr.btopenworld.com) |
| 08:57.31 | scruz | Including the secret parameter. Personally I don't care, but like the person who pays $100 extra for extra airbags shows, peace of mind comes at a premium. |
| 08:59.34 | garymc | anyone help me out. My system time is correct on my asterisk server, but today all my polycom phones are one hour ahead??? |
| 09:02.09 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 09:02.10 | kaldemar | check that the phones are getting the time from your server, the distributed time is correct and time zones are correct. |
| 09:03.24 | *** join/#asterisk killdashnine (~matthias@82.161.138.210) |
| 09:04.54 | SiNGLer | garymc: maybe wrong daylight saving setting? |
| 09:05.15 | garymc | but where would that be set? |
| 09:05.48 | SiNGLer | in phone's settings |
| 09:05.55 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 09:06.05 | SiNGLer | I never had polycoms, so can't consult |
| 09:10.51 | kaldemar | garymc: see tcpIpApp.sntp.* in the phone configuration file. |
| 09:11.03 | garymc | in all phones? |
| 09:11.29 | garymc | surely this has a setting in the tftp boot files somewhere? |
| 09:12.40 | kaldemar | a polycom configuration file is something you have on the boot server. |
| 09:13.49 | kaldemar | you can group the config files how you wish, using a single template for all phones or have all phones use their own set of configs. see what your setup is like. |
| 09:13.59 | justdave | usually you've got two files for each phone, typically one will be sip.cfg which is used by all phones, and the other will be specific to each phone. |
| 09:14.19 | justdave | doesn't have to be that way, as kaldemar said you can arrange them how you want, but the sample configs that polycom ships put them that way |
| 09:14.39 | kaldemar | two or three usually. |
| 09:15.06 | justdave | the file that's named for the MAC address of the phone will have a list of the other files in it |
| 09:15.34 | kaldemar | the MAC.cfg points to two files by default, of which sip.cfg has <SNTP/> which defines NTP related settings. |
| 09:17.29 | *** join/#asterisk Amnesia (~Amnesia@unaffiliated/amnesia) |
| 09:17.50 | Amnesia | anyone here known with portech gsm gateways? |
| 09:18.14 | *** join/#asterisk coppice (~chatzilla@m121-202-91-124.smartone-vodafone.com) |
| 09:18.37 | SiNGLer | Amnesia: I can try to help you |
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| 09:18.48 | Amnesia | ah Singler that was your nick:p |
| 09:18.57 | Amnesia | SiNGLer: I keep getting the busy message back |
| 09:19.13 | Amnesia | tried +316xxxx 00316xxxx 06xxxxx all with no success |
| 09:19.33 | Amnesia | and I am pretty sure the gateway is succesfully registered at asterisk |
| 09:21.15 | SiNGLer | can you connect to it now? |
| 09:21.24 | Amnesia | nope |
| 09:21.37 | Amnesia | - Got SIP response 486 "Busy Here" back from 194.1.100.246 |
| 09:21.54 | SiNGLer | I mean connect to it's web interface |
| 09:22.11 | Amnesia | yep |
| 09:22.14 | Amnesia | sorry, yep I can |
| 09:22.51 | p3nguin | its |
| 09:23.00 | SiNGLer | SIP settings ->service domain does show registered? |
| 09:23.18 | Amnesia | yep |
| 09:23.47 | SiNGLer | and is mobile -> status state standby? |
| 09:24.24 | Amnesia | hm no such thing |
| 09:24.29 | Amnesia | everythings blank except for |
| 09:24.50 | Amnesia | sim card id, signal quality and GSM S/N: |
| 09:25.11 | Amnesia | sim card id = 0.0 |
| 09:25.20 | Amnesia | signal quality = 27 |
| 09:25.31 | Amnesia | and GSM S/N: doesnt matter afaik:P |
| 09:26.01 | SiNGLer | well motion state should not be blank |
| 09:26.23 | SiNGLer | and operator field is blank too? |
| 09:26.28 | Amnesia | there isnt a box with motion state |
| 09:27.17 | SiNGLer | what is your firmware version? |
| 09:27.29 | Amnesia | http://pastebin.com/Atdk1xmK |
| 09:27.43 | Amnesia | Firmware Version:Fri Nov 2 09:50:56 2007. |
| 09:27.59 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 09:28.55 | SiNGLer | I think you should try upgrading it. Lan to mobile settings are url=* and callnum=# |
| 09:28.56 | SiNGLer | ? |
| 09:29.04 | Amnesia | hm yep |
| 09:29.20 | Amnesia | strangly enough there are also 2 Lan to mobile setting pages |
| 09:29.49 | Amnesia | tem CID URL Select |
| 09:29.50 | Amnesia | 0 * 194.1.100.214 |
| 09:29.52 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 09:30.11 | Amnesia | ah wait thats speed dialing |
| 09:30.54 | SiNGLer | try deleting speeddial for now |
| 09:31.07 | Amnesia | yep did that |
| 09:31.16 | Amnesia | still getting busy as a response:) |
| 09:31.28 | garymc | Yes that worked. Thanks peeps |
| 09:31.36 | Amnesia | item: 0 CID: * url: $asteriskip |
| 09:31.40 | Amnesia | thats correct right? |
| 09:31.41 | garymc | SNTP in SIP.cfg |
| 09:32.07 | SiNGLer | Amnesia: which menu setting? |
| 09:32.16 | Amnesia | lan to mobile |
| 09:32.39 | Amnesia | btw perhaps its also useful for you to know, that when I call the number of the sim thats in it, it tells me the number isnt reachable |
| 09:32.56 | SiNGLer | I guess that sim card is not up |
| 09:33.34 | Amnesia | how can I get it up? |
| 09:33.39 | Amnesia | its in for sure |
| 09:33.45 | SiNGLer | and lan to mobile I have two columns URL, which is set to * and Call Num is set to # |
| 10:11.08 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 10:11.08 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.3 (2011/02/28), 1.6.2.17 (2011/02/28), 1.4.40 (2011/02/28), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 10:12.14 | Amnesia | wdoekes2: and remove username=125 ? |
| 10:12.34 | wdoekes2 | is that the user configured in the portech? |
| 10:12.45 | wdoekes2 | you can safely leave it there |
| 10:13.20 | Amnesia | yep it is |
| 10:13.21 | Amnesia | okay |
| 10:13.31 | Amnesia | ah wait |
| 10:13.35 | Amnesia | already had fromuser=125 |
| 10:13.56 | wdoekes2 | I don't think that did what it's supposed to.. do a sendrpid=yes |
| 10:14.29 | Amnesia | hm just resetted it |
| 10:14.33 | wdoekes2 | sip reload, and do another outbound call (INVITE).. you should see From: 125@... in the outgoing invite |
| 10:14.35 | Amnesia | got to get its new ip now:p |
| 10:15.30 | Amnesia | ugh how gay is that |
| 10:15.40 | Amnesia | it doesnt act as a dhcp client by default:P |
| 10:16.16 | *** join/#asterisk actzipild (~actzipild@postfix.eureko.ro) |
| 10:16.50 | *** join/#asterisk IsUp (IsUp@unaffiliated/isup) |
| 10:16.52 | actzipild | Hi all. Any clue why a user would authenticate via sip account and after 30 seconds the status would change to registering instead of registered? (Using Zoiper) |
| 10:19.18 | *** join/#asterisk jkroon (~jkroon@wbs-41-208-211-232.wbs.co.za) |
| 10:22.10 | IsUp | actzipild: whats your console says on Asterisk? |
| 10:22.13 | IsUp | enable debugging |
| 10:22.22 | IsUp | 'core set verbose 10', 'core set debug 10' |
| 10:25.22 | actzipild | absolutely nothing |
| 10:27.25 | Amnesia | SiNGLer: now I've updated it in ie it has updated its fw:D |
| 10:29.35 | actzipild | IsUp: I have restarted Zoiper (the client) and I got Registered SIP for user, then handle_response_peerpoke: Peer is now reachable, then twice handle_request_subscribe: Received SIP Subscribe from peer without mailbox.. and now I am back to registering as status in Zoiper. |
| 10:30.52 | Amnesia | SiNGLer: mobile state = initing |
| 10:31.55 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 10:31.57 | IsUp | actzipild: do you have 'qualify=yes' on your SIP peer? |
| 10:32.43 | actzipild | IsUp: I have |
| 10:32.52 | actzipild | sorry, I have qualify=300 |
| 10:33.04 | IsUp | actzipild: and your SIP peer behind NAT? |
| 10:33.11 | actzipild | no, it's on LAN |
| 10:33.17 | IsUp | it sounds like "zoiper" issue |
| 10:33.26 | IsUp | did you try with eyeBeam or something different? |
| 10:33.52 | actzipild | I thought the same, but the thing is that I have the same version on different computers and they work. |
| 10:34.43 | actzipild | only for two users is not working: me and another one. there are 16 other people in the callcenter and for them is working flawlessly. |
| 10:36.04 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 10:39.44 | SiNGLer | Amnesia: check now mobile state, at first it is initing, and after initialization it should be standby |
| 10:39.44 | *** join/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110) |
| 10:39.52 | *** join/#asterisk Fatalfury1981 (~chatzilla@host25-96-static.48-79-b.business.telecomitalia.it) |
| 10:39.58 | Amnesia | SiNGLer: facepalm |
| 10:40.09 | Amnesia | I had the sim card inserted wrong-.- |
| 10:40.34 | SiNGLer | :) |
| 10:41.09 | SiNGLer | Amnesia: http://www.awesometoast.com/wp-content/uploads/2010/03/tactical_facepalm.jpg |
| 10:41.26 | Amnesia | haha |
| 10:41.52 | SiNGLer | now it works? |
| 10:42.29 | Amnesia | yep |
| 10:42.41 | Amnesia | got to fine tune my extensions.conf now |
| 10:44.30 | Amnesia | but ehm |
| 10:44.38 | Amnesia | I've got the rule |
| 10:44.48 | Amnesia | exten => _0X.,1,Answer() |
| 10:45.01 | Amnesia | but now what if I want to dial 0612345678 ? |
| 10:45.16 | Amnesia | then it'll dial 612345678 |
| 10:45.41 | SiNGLer | Dial(SIP/125/${EXTEN}) ? |
| 10:46.25 | Amnesia | http://pastebin.com/EN48Xeet |
| 10:48.14 | Amnesia | what should I dial or do to get into the gsm context? |
| 10:48.24 | SiNGLer | you should not answer the line at the beggining (unless you have a reason for it) if in outgoing extension you want to dial gsm gw, just insert Dial(SIP/125/${EXTEN}) |
| 10:48.24 | Amnesia | exten => 0653228835,1,Dial(SIP/125/0653228835,60,tTr) |
| 10:49.14 | Amnesia | hm well incoming could be incoming from the dahdi card or the voip gw |
| 10:50.03 | SiNGLer | let's begging from the start, what do you want to do? |
| 10:50.09 | SiNGLer | *begin |
| 10:51.52 | Amnesia | well ehm |
| 10:52.04 | Amnesia | calls directed to exten => 0653228835,1,Dial(SIP/125/0653228835,60,tTr) |
| 10:52.11 | Amnesia | should go trough the gw |
| 10:52.29 | Amnesia | and calls that arent in the list should go through the pstn line |
| 10:54.25 | SiNGLer | rename outgoing to eg dahdi_c and use http://pastebin.com/eFQ4DSXJ |
| 10:54.46 | SiNGLer | or something like that |
| 10:54.58 | Amnesia | hmkay thx:) |
| 10:55.10 | Fatalfury1981 | how can i show a message on display of my snom phone when i digit some number for activate/deactivate some function managed by asterisk? |
| 10:57.17 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
| 11:03.00 | *** join/#asterisk sourcode (~code@ppp-58-8-159-250.revip2.asianet.co.th) |
| 11:10.33 | *** join/#asterisk ruyo (~psantos@a83-132-152-91.cpe.netcabo.pt) |
| 11:17.42 | *** join/#asterisk qjb (~qjb@2001:888:17b4:0:21f:5bff:fef1:f005) |
| 11:27.01 | *** join/#asterisk iulhk (iulhk@175.110.36.67) |
| 11:28.09 | iulhk | i hv configured realtime asterisk 1.6 , when any sip peer registering by using any sip client its not updating lastms field, any idea ? |
| 11:30.00 | schmidts | Fatalfury1981 take a look at custome device state handling. you can configure one key of your snom phone as blf key and then just change the state inside asterisk |
| 11:39.54 | WIMPy | Or SentText() |
| 11:45.15 | kaldemar | Fatalfury1981: it used to be possible to show arbitrary messages on snom screens, i used sipsak to do it. it's been years since though. |
| 11:46.04 | kaldemar | Fatalfury1981: http://wiki.snom.com/FAQ/How_to_display_a_text_message_to_the_phone_in_order_to_appear_on_the_display |
| 12:06.54 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 12:06.54 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.3 (2011/02/28), 1.6.2.17 (2011/02/28), 1.4.40 (2011/02/28), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 12:09.43 | *** join/#asterisk blinky42 (~quassel@66.54.228.71) |
| 12:38.52 | *** join/#asterisk tomaw (tom@freenode/staff/tomaw) |
| 12:51.25 | *** join/#asterisk kleszcz (tick@80.54.23.253) |
| 12:54.54 | *** join/#asterisk reber (~reber@ns1.sylogix.net) |
| 12:55.39 | *** join/#asterisk wonderworld (~ww@port-92-201-38-241.dynamic.qsc.de) |
| 12:58.11 | *** join/#asterisk garymc (~chatzilla@host81-148-15-59.in-addr.btopenworld.com) |
| 13:03.15 | iulhk | anybody there ? |
| 13:03.21 | iulhk | i hv configured realtime asterisk 1.6 , when any sip peer registering by using any sip client its not updating lastms field, any idea ? |
| 13:09.33 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 13:11.28 | schmidts | i need an idea how i can solve a problem. i have a server running asterisk on it and also a webpage. both (* and web) are reachable over the same URL and i want to move the website to another server. i cannnot change the ip of the domain entry cause many phones also use the url as proxy. i know i can set up a simple apache to redirect it to another server and i can use iptables for port redirection but this only works with nat. Any further ide |
| 13:11.28 | schmidts | as anybody? |
| 13:12.36 | actzipild | try redirect from apache using the .htaccess file |
| 13:13.57 | schmidts | actzipild thats one of the options i allready know but i am not really happy with this solution |
| 13:15.18 | AviMarcus | can you use SRV in the dns for the domain? |
| 13:15.30 | AviMarcus | SRV sip to one place, SRV :80 web to another |
| 13:15.43 | schmidts | AviMarcus will this be supported from every device? |
| 13:15.53 | Chainsaw | schmidts: Absolutely not. |
| 13:16.14 | schmidts | or atleast from every browser? |
| 13:16.16 | Chainsaw | schmidts: If you want a solution that always works, it's pretty much an Apache-based redirect or... an Apache-based redirect. |
| 13:16.24 | AviMarcus | :) |
| 13:16.51 | AviMarcus | and next time, split everything to different subdomains? even if they happen to be hosted at the same place? |
| 13:17.00 | schmidts | chainsaw ;) i love to have different options |
| 13:17.12 | AviMarcus | I should probably make my panel "account.domain.com" in preparation for that. |
| 13:17.19 | schmidts | AviMarcus please tell this to the guy who built this system several years ago |
| 13:17.43 | AviMarcus | how big is the website you have on it? you need to switch it out? |
| 13:18.15 | AviMarcus | considered using something like nginx + php5-fpm (if you have php) to standardize the load? or to reverse-proxy to the new machine if it's in the same data center |
| 13:18.31 | *** join/#asterisk Fatalfury1981 (~chatzilla@host25-96-static.48-79-b.business.telecomitalia.it) |
| 13:18.43 | schmidts | the point is, i move the ip from this server to another with a sip proxy and i didnt want to have the website running on the same server |
| 13:19.13 | AviMarcus | well, if everyone is using the same dns entry (and not the SIP IP) then I suppose you don't have very many options |
| 13:19.21 | AviMarcus | unless you have provisioning set up for ALL the phones? |
| 13:19.21 | Amnesia | what's wrong with the GOTO in #62: http://pastebin.com/iEGTDjNT |
| 13:20.00 | schmidts | AviMarcus definitly no;) more than 3700 phones at the moment |
| 13:20.35 | AviMarcus | number of phones doesn't mean you don't have auto provisioning |
| 13:20.38 | schmidts | Amnesia its a loop, you goto to yourself |
| 13:20.48 | schmidts | AviMarcus i know ;) |
| 13:20.59 | Amnesia | ah:P |
| 13:21.48 | AviMarcus | so.. how much load is on that web server? |
| 13:21.53 | Amnesia | fixed:P |
| 13:22.34 | schmidts | AviMarcus i will post the output of an ps faux | grep apache i am not sure how to trace this things down |
| 13:22.58 | schmidts | http://pastebin.com/9tgq0L0q |
| 13:23.20 | schmidts | a lot of memory usage but not much cpu consumption |
| 13:24.50 | AviMarcus | maybe just reduce the workers? |
| 13:25.02 | AviMarcus | or try nginx if you don't have much .htaccess stuff going on |
| 13:25.06 | AviMarcus | also mod_php ? |
| 13:26.57 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 13:27.33 | AviMarcus | schmidts, I don't have much volume, so I set down to a few: http://pastebin.com/kmpXPQfk |
| 13:28.30 | AviMarcus | I can probably go lower. heh. it's just the GUI for the sip server |
| 13:28.57 | schmidts | AviMarcus for me too but i wont say "just" ;) |
| 13:29.08 | schmidts | but i like the idea of SRV entries for web |
| 13:29.42 | AviMarcus | are you moving the web to the same datacenter? |
| 13:30.00 | schmidts | same rack but different server |
| 13:30.11 | AviMarcus | so maybe use an ngingx reverse proxy? |
| 13:30.25 | schmidts | no sorry not really when its finished the proxy and the webpage are running on the same host server but virtualised |
| 13:33.18 | schmidts | ok SRV isnt really supported by browsers so this will not help anything. |
| 13:33.46 | schmidts | i will take a look at ngingx and php_fmp |
| 13:34.14 | AviMarcus | php5-fpm |
| 13:34.16 | *** join/#asterisk hairyraven (~nobody@95.72.26.165) |
| 13:34.23 | schmidts | ;) |
| 13:34.52 | AviMarcus | it's a replacement for apache. If you have fancy .htaccess stuff, you'd have to rewrite them.. |
| 13:35.37 | *** join/#asterisk ruchir (~ruchir@122.169.95.126) |
| 13:35.41 | ruchir | hi all |
| 13:35.47 | ruchir | i'm having strange issue |
| 13:35.58 | ruchir | sip user is connected to meetme conf |
| 13:36.18 | ruchir | if i try to redirect the channel to other dialplan which eventually goes to other meetme conf room |
| 13:36.20 | ruchir | the call drops |
| 13:36.25 | ruchir | any idea what might be causing this? |
| 13:39.35 | ruchir | anybody there? |
| 13:47.34 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 13:51.19 | *** join/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 13:53.51 | creativx | are there any good tools to parse cdr files from asterrrrr |
| 13:56.25 | AviMarcus | creativx, specifically? or just general.. pcre / regexp stuff does a good job |
| 13:57.06 | *** join/#asterisk cyborg-one (1000@212-178-13-243.broadband.tenet.odessa.ua) |
| 13:57.13 | creativx | well thats a good question |
| 13:57.18 | creativx | hehe.. looking at our csv files now |
| 13:57.44 | creativx | basically trying to get stats for number of calls answered, avg wait time |
| 13:58.01 | *** join/#asterisk andygraybeal (~andy.gray@obsidian.casanueva.com) |
| 13:58.24 | schmidts | creativx take a look at asterstats from asternic.biz the guy who wrote the flashoperationpanel |
| 13:58.39 | creativx | k thanks |
| 13:58.56 | schmidts | or astercdr |
| 14:00.12 | creativx | when I come to think of it I'm looking at measuring at queue level |
| 14:00.17 | creativx | might be some queue stats inside asterisk for that :> |
| 14:00.20 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 14:00.21 | *** join/#asterisk espiceland (~erin@nat/digium/x-tfdcldturcjxntch) |
| 14:02.03 | schmidts | creativx asternic.biz also have something for this, but its not free |
| 14:02.17 | schmidts | we use it cause we also use FOP2 and so everything looks like the same ;) |
| 14:03.02 | creativx | hehe okay |
| 14:03.47 | creativx | thanks for the tips, ill look into that |
| 14:03.54 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-129-161.lsanca.fios.verizon.net) |
| 14:07.31 | p3nguin | I can't stand apps like that because they take over control of the Asterisk configs most of the time. That totally and completely ruins a good Asterisk system. |
| 14:09.29 | schmidts | p3nguin what do you mean FOP or the statistic thing? |
| 14:10.18 | p3nguin | I'm talking about whichever one takes over Asterisk's configuration. |
| 14:10.31 | schmidts | p3nguin the static programs they offer only runs a python daemon which checks the queue_log file every second or uses the cdr stored in a database, you will not have to change everything on asterisk config to use this |
| 14:10.38 | p3nguin | If a stats app does that, it's a piece of shit. |
| 14:10.57 | schmidts | p3nguin thats true ;) |
| 14:11.04 | p3nguin | s/does that/takes control of configs/ |
| 14:11.05 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-arsyyyulwwvvsxbn) |
| 14:11.21 | p3nguin | (just to clarify) |
| 14:11.31 | schmidts | ;) |
| 14:11.50 | creativx | most of these stat builders seems to just parse the cdr file |
| 14:11.59 | creativx | and build indexed views in a separate db somewhere |
| 14:12.22 | p3nguin | THat's what they should do... read the cdr and display what it finds. |
| 14:12.28 | creativx | yeh |
| 14:12.34 | creativx | im looking at our cdr and it looks like a mess to parse |
| 14:12.35 | creativx | hehe |
| 14:14.24 | Amnesia | hm |
| 14:14.38 | Amnesia | how can I make calls to 0646392820, go through gsm_out and not pstn_out? |
| 14:14.40 | Amnesia | http://pastebin.com/4pFmQWfV |
| 14:17.24 | p3nguin | Why do you use tTr dial options? |
| 14:17.49 | Amnesia | hm was a copy paste:P |
| 14:17.56 | Amnesia | but ehm got any idea how I could achieve this? |
| 14:18.01 | p3nguin | Take them out if you don't know what they do. |
| 14:18.29 | p3nguin | On every single line of that dialplan. |
| 14:18.46 | p3nguin | If you need the options, you'll know why you have them on there. |
| 14:19.21 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 14:19.24 | p3nguin | What is the peer name for your gsm_out entity? |
| 14:20.01 | p3nguin | Is it really SIP/125? Can you show me the [125] definition from sip.conf? |
| 14:20.21 | p3nguin | C'mon, now... stay with me. |
| 14:20.52 | Amnesia | call transferring |
| 14:20.52 | Amnesia | right? |
| 14:22.02 | p3nguin | t and T do DTMF transfers for the calling party as well as the called party. Is that really what you intend to provide? |
| 14:22.26 | *** join/#asterisk Amnesia (~Amnesia@unaffiliated/amnesia) |
| 14:22.30 | p3nguin | t and T do DTMF transfers for the calling party as well as the called party. Is that really what you intend to provide? |
| 14:22.47 | Amnesia | p3nguin: hm nope |
| 14:22.49 | Amnesia | I removed them |
| 14:23.06 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:23.09 | Amnesia | but ehm, got any idea how I can get 0646392820 and the other to be run from gsm_out and not pstn_out |
| 14:23.11 | p3nguin | Take out the r, too, unless you're sure you need it. You probably don't need it. |
| 14:23.15 | Amnesia | (since there's a leading 0 in front of them) |
| 14:23.20 | p3nguin | (0819.24) <p3nguin> What is the peer name for your gsm_out entity? |
| 14:23.21 | p3nguin | (0819.57) <p3nguin> Is it really SIP/125? Can you show me the [125] definition from sip.conf? |
| 14:23.27 | p3nguin | Stay with me. |
| 14:23.47 | p3nguin | If I'm going to take my time to help you, at least you can pay attention to what I am asking you and answer my questions. |
| 14:24.14 | Amnesia | my net failed sorry |
| 14:24.27 | Amnesia | 125 works, it's a gsm gateway |
| 14:24.30 | p3nguin | I'll overlook it this time. ;) |
| 14:24.31 | Amnesia | but it seems to work now |
| 14:24.34 | Amnesia | lol |
| 14:25.56 | p3nguin | Every time you change extensions.conf, make sure you save the file and run "dialplan reload" to make the changes take effect. Similarly, every time you change sip.conf, save the changes and run "sip reload" to make the changes take effect. |
| 14:26.36 | Amnesia | yeah I know:) |
| 14:27.14 | p3nguin | When the correct extensions exist and aren't working, those are often the reason it doesn't work. |
| 14:27.24 | Amnesia | heh:P |
| 14:27.26 | Amnesia | faq^^ |
| 14:35.38 | *** join/#asterisk marienz (~marienz@freenode/staff/marienz) |
| 14:37.56 | *** join/#asterisk cnu (cnu@2001:470:28:1fe::10) |
| 14:40.03 | *** join/#asterisk dancarlson (~dancarlso@S0106687f74d1e881.va.shawcable.net) |
| 14:45.06 | *** join/#asterisk ks3 (~ksandy@74.203.195.1) |
| 14:45.39 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
| 14:46.28 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:46.28 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:52.28 | *** join/#asterisk hairyraven (~nobody@95.72.32.72) |
| 14:55.51 | ks3 | Is there a way to find the current SIP client's IP address via the dialplan? I haven't been able to find one so far... |
| 14:58.39 | p3nguin | The phone making the call? |
| 14:59.20 | ks3 | More or less... in this instance it'a actually the IP address of a proxy |
| 14:59.23 | *** part/#asterisk benngard (~mabe@213.88.138.230) |
| 14:59.41 | leifmadsen | probably the CHANNEL() function |
| 15:00.32 | p3nguin | DumpChan() has a line showing my own phone's IP address when I call another phone. |
| 15:01.15 | dancarlson | If I have a DAHDI channel DIALing a SIP channel, how can I get the name of the SIP Channel when I'm back in the DAHDI channel (for reading the sip response code from the hangupcause)? |
| 15:01.19 | ks3 | Aha! There appear to be a few items in CHANNEL that may give what I need. Thanks. |
| 15:04.16 | leifmadsen | dancarlson: before you call the other channel, set the name of the current channel and make it inherited |
| 15:04.21 | leifmadsen | Set(__CurrentChannel=...) |
| 15:04.38 | leifmadsen | (the double underscores makes it inherity infinite, a single is inherit a single time) |
| 15:05.42 | leifmadsen | Set(_thisChannel=${CHANNEL(...)}) or whatever you want to save |
| 15:09.23 | *** part/#asterisk AviMarcus (~avi@bzq-79-181-184-242.red.bezeqint.net) |
| 15:10.35 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 15:17.16 | Amnesia | hm question here |
| 15:17.35 | Amnesia | I've got a intercom which is registered to asterisk |
| 15:17.59 | Amnesia | but how can I define a context to it when someone pushes the button |
| 15:18.02 | Amnesia | throuhg sip.conf? |
| 15:18.48 | p3nguin | If it has a peer entry in sip.conf, that's where you are required to assign the context. |
| 15:19.02 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:19.47 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 15:20.38 | Amnesia | thx |
| 15:20.50 | Amnesia | should've thought about that myself...:p |
| 15:20.59 | dancarlson | leifmadsen: Before I do my Dial, if I set the name of the current channel (the DAHDI channel) into a variable. However, when the Dial Macro option puts me into a macro, it seems I'm operating in the SIP Channel's space, and I don't have a way to set the variable in the original DAHDI channel. Perhaps I'm misunderstanding your last instruction. |
| 15:21.44 | leifmadsen | dancarlson: yes because the Dial M() option is run on the *called* channel (per the documentation) |
| 15:23.55 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 15:24.53 | *** join/#asterisk youngproguru (~youngprog@208.69.84.100) |
| 15:25.05 | dancarlson | yes, that has become clear to me. : - ). I don't care whether I use a Macro or not, but if I don't use a Macro I'm afraid I don't know where I would set the _thisChannel=${CHANNEL(...)} you suggested in your previous post. |
| 15:25.06 | *** join/#asterisk lordvadr (~something@jose-tc.ctc.biz) |
| 15:26.49 | lordvadr | Anybody here ever use voip cpe's by a company called EdgeWater? They make these linux\asterisk cpe devices, t1, ethernet, dsl wan options. |
| 15:27.35 | *** join/#asterisk LemensTS (~matthew@adsl-70-238-136-43.dsl.stlsmo.sbcglobal.net) |
| 15:27.45 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-136-43.dsl.stlsmo.sbcglobal.net) |
| 15:28.59 | *** join/#asterisk strehlow (~strehlow@24-247-41-88.dhcp.mrqt.mi.charter.com) |
| 15:29.28 | ks3 | dancarlson, The few times I've needed something similar, I've used SHARED |
| 15:32.26 | *** join/#asterisk kriegerod (~krieger@chaf.tenet.odessa.ua) |
| 15:37.51 | dancarlson | ks3: that looks perfect, thanks for your suggestion! |
| 15:39.56 | Amnesia | hm, anyone willing to help me out with my intercom? |
| 15:40.33 | Amnesia | http://pastebin.com/ZY7MmSXj |
| 15:41.26 | Amnesia | I arent getting any output at all in asterisk while I've ran core set verbose/debug 10 |
| 15:42.48 | brainiac | Would anyone know where I'd start looking to solve a pickup mark loop problem? |
| 15:44.10 | ks3 | Amnesia, so the intercom is a SIP device? If so, you should include the relevant parts of your sip.conf as well. |
| 15:44.31 | Amnesia | ks3: http://pastebin.com/7HmpQLa5 |
| 15:45.03 | *** join/#asterisk kjs (~kjs@fedora/kjs) |
| 15:45.07 | dancarlson | amnesia: what kind of intercom is it? |
| 15:45.46 | Amnesia | ehm a pcb on a piece of iron connectec through an papt2 (internet phone adapter) |
| 15:45.52 | Amnesia | it is registered |
| 15:47.10 | kjs | Hi guys, weird problem with my asterisk box. Everything has been working fine for months... Now when I attempt to place an outbound call it hangs and does not work. If I hang up and redial it works most of the time. I am using a SIP trunk from Gradwell (in the UK) to access the PSTN, here is my pastebin logs from the console: http://pastebin.com/q2442wsi |
| 15:47.29 | kjs | Could this be my provider? |
| 15:48.54 | *** join/#asterisk timahvo1 (~rogue@41.215.1.35) |
| 15:49.47 | p3nguin | amnesia: Be sure to delete that duplicate for sip peer name 10. |
| 15:50.06 | p3nguin | and 19. |
| 15:50.26 | p3nguin | and 15. |
| 15:50.38 | p3nguin | WTF? Why do you have duplicates? You can't do that. |
| 15:50.50 | Amnesia | <PROTECTED> |
| 15:50.52 | Amnesia | my bad:p |
| 15:50.53 | p3nguin | You can't have two 10s, two 15s, and two 19s. |
| 15:51.12 | Amnesia | hm my bad |
| 15:51.14 | Amnesia | fixed it |
| 15:51.22 | p3nguin | save, sip reload |
| 15:51.26 | Amnesia | http://pastebin.com/rrT6LTv4 |
| 15:51.28 | Amnesia | also did that |
| 15:51.34 | p3nguin | Now show me a problem with your device. |
| 15:52.10 | Amnesia | well nothing gets logged when I press the intercom button |
| 15:52.32 | Amnesia | it has been connected to asterisk though afaik |
| 15:52.40 | Amnesia | Registration State:Online |
| 15:52.49 | p3nguin | Show me the line that starts with 10 when you run sip show peers. |
| 15:53.26 | p3nguin | Wait. The intercom device is 10, isn't it? |
| 15:54.16 | Amnesia | no 9999 |
| 15:54.33 | p3nguin | Oh, then show me the line starting with 9999 when you run sip show peers. |
| 15:55.10 | Amnesia | 9999/9999 194.1.100.247 D N 5060 OK (12 ms) |
| 15:55.44 | p3nguin | Okay, so it's registered, as you said. You press a button on the device, and nothing happens? |
| 15:55.54 | *** join/#asterisk cyborg-one (1000@212-178-8-45.broadband.tenet.odessa.ua) |
| 15:56.02 | Amnesia | yeah basically |
| 15:56.48 | *** join/#asterisk Skrusty (~ben@93-97-20-22.zone5.bethere.co.uk) |
| 15:56.58 | Amnesia | when I temporarily add it to the users context and then dial to it it works |
| 15:57.01 | p3nguin | Run "sip set debug peer 9999" and then try to use your intercom device. You should see something on the cli. |
| 15:57.04 | Amnesia | so the connection and the device are fine |
| 15:57.14 | p3nguin | erm |
| 15:57.22 | p3nguin | So what is the problem if it works? |
| 15:57.24 | *** join/#asterisk path (~luis@64.76.149.84) |
| 15:57.25 | p3nguin | You said it didn't work. |
| 15:57.32 | Skrusty | afternoon al |
| 15:57.33 | Skrusty | l |
| 15:57.34 | p3nguin | Now it does. Make up your mind. |
| 15:58.12 | Amnesia | p3nguin: thats calling to the intercom from a telephone |
| 15:58.27 | Amnesia | I need to make calls from the intercom to the phones inside ofc |
| 15:59.10 | p3nguin | inside office? |
| 15:59.28 | Amnesia | yep |
| 15:59.40 | p3nguin | Back to the sip debug, then. sip set debug peer 9999 |
| 15:59.54 | p3nguin | Make a call. Copy EVERYTHING and pastebin it. |
| 16:00.04 | p3nguin | I need to see the entire failed call. |
| 16:00.46 | Amnesia | http://pastie.org/1629066 |
| 16:01.18 | Amnesia | not sure thats the intercom... |
| 16:01.27 | Amnesia | cause now when I press the button no output is given |
| 16:01.53 | Amnesia | When I press the button I getting a sound like someone has hanged up the phone and then a beep (builtin the intercom I guess) |
| 16:02.28 | Amnesia | http://pastie.org/1629072 |
| 16:03.02 | p3nguin | This isn't a debug of a call. |
| 16:03.10 | p3nguin | A phone call will start with an INVITE. |
| 16:04.14 | Amnesia | I've seriously done what you said |
| 16:04.56 | p3nguin | I believe you. There's just something else wrong that I'm not able to think of right now. |
| 16:05.17 | Amnesia | probably something with the pap2t device:/ |
| 16:05.30 | *** join/#asterisk nicola_pav (~nicola_pa@mail2.tikalnetworks.com) |
| 16:05.54 | p3nguin | You connect the intercom device to the PAP2T, and you configure the PAP2T with the peer information for 9999, correct? |
| 16:06.14 | *** join/#asterisk benngard (~mabe@h30n2-g-ml-a11.ias.bredband.telia.com) |
| 16:06.27 | Amnesia | Call 1 State:Invalid |
| 16:06.32 | Amnesia | Call 1 Tone:Ring Back |
| 16:06.45 | Amnesia | correct |
| 16:07.27 | Amnesia | the device has two "line ins" and one ethernet output |
| 16:08.01 | p3nguin | Yeah, but you can connect to line 1 and configure line1 in the web interface. |
| 16:08.15 | Amnesia | yep |
| 16:08.36 | nicola_pav | hello. i have asterisk 1.6.2.13. I have problem in receiving faxes. |
| 16:08.49 | nicola_pav | here is what i get from the cli: [Mar 3 17:55:28] WARNING[17826] app_fax.c: Error transmitting fax. result=13: Unexpected message received. |
| 16:08.58 | nicola_pav | and [Mar 3 17:55:28] WARNING[17826] app_fax.c: Transmission failed |
| 16:09.03 | nicola_pav | any hints please? |
| 16:09.48 | *** join/#asterisk tgrahamcapital (~trevorfra@ip68-12-146-104.ok.ok.cox.net) |
| 16:10.21 | tgrahamcapital | ~take-a-number Does anyone recommend a specific SS7 implementation for Asterisk? |
| 16:11.36 | *** join/#asterisk mechbangirc (~mechbangi@115-186-140-193.nayatel.pk) |
| 16:12.27 | *** join/#asterisk fofware (~fofware@wdctf.siup.gov.ar) |
| 16:12.35 | Amnesia | it sounds like it rings once and then does nothing |
| 16:12.48 | mechbangirc | hi using cdr_adaptive_odbc to log my cdr, no matter what i do it wont insert calldate column rest of the columns are ok |
| 16:13.10 | tgrahamcapital | is lib_ss7 or chan_ss7 better? |
| 16:13.23 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 16:13.33 | Amnesia | Call 1 Type:Inbound while it should be outbound I guess |
| 16:17.01 | p3nguin | Does your intercom device connected to the ATA get a dial tone? |
| 16:19.11 | Amnesia | what does a dialing tone sound like? |
| 16:19.30 | p3nguin | Have you ever picked up a phone and put it to your ear without dialing any numbers on the keypad? |
| 16:19.50 | Amnesia | ah |
| 16:19.56 | p3nguin | That's a dial tone. |
| 16:20.09 | p3nguin | It's a tone that indicates to you that you may dial a number. |
| 16:20.21 | Amnesia | don't think so |
| 16:21.02 | *** join/#asterisk BezNalogov (~nameless@46.47.91.113) |
| 16:21.08 | p3nguin | If there's no dial tone on the device, assume it doesn't work. Connect a regular bell telephone to line1 port on the ATA instead of the intercom device. |
| 16:21.37 | Amnesia | that works |
| 16:21.41 | BezNalogov | Hi people. I recently upgraded to asterisk 1.6. Unfortunately it seems that the queues aren't working anymore. I get this: [Mar 3 17:17:02] WARNING[4046]: pbx.c:3675 pbx_extension_helper: No application 'Queue' for extension (mainmenu-nl2, 2, 3) |
| 16:21.59 | BezNalogov | How do the queues work in asterisk 1.6? |
| 16:22.04 | leifmadsen | 1.6 is not a version |
| 16:22.11 | leifmadsen | (or even a valid branch) |
| 16:22.12 | p3nguin | Using a phone instead of the intercom device is working fully? You can send and receive calls? |
| 16:22.23 | Amnesia | it just sounds like it rings for one time and then it does nothing anymore |
| 16:22.25 | benngard | how do i write a ";" set db, for example exten => 11/0317998985,1,SET(DB(CFC/0317998975)=Gått for dagen) |
| 16:22.28 | leifmadsen | BezNalogov: although the problem appears as if app_queue.so is not loaded/compiled |
| 16:22.34 | leifmadsen | benngard: \; |
| 16:22.40 | Amnesia | p3nguin: to the pap2t yes, to other phones no |
| 16:22.43 | benngard | tried that |
| 16:22.45 | BezNalogov | Asterisk 1.6.2.5-0ubuntu1.3 built by buildd @ allspice on a x86_64 running Linux on 2011-01-21 15:10:32 UTC |
| 16:22.48 | leifmadsen | benngard: what version of asterisk? |
| 16:22.58 | leifmadsen | 1.6.2.5? that's crazy old |
| 16:23.03 | leifmadsen | I just released 1.6.2.17 |
| 16:23.10 | p3nguin | amnesia: Connect the phone to the ATA line1. Make that work fully before trying anything else. |
| 16:23.20 | BezNalogov | This is the version that ubuntu has in the repositories I think |
| 16:23.26 | Amnesia | I can already confirm that works |
| 16:23.34 | leifmadsen | benngard: try \\\; |
| 16:23.39 | benngard | leifmadsen: Asterisk 1.8.2.3 |
| 16:23.46 | leifmadsen | oh then \; should work |
| 16:24.01 | Amnesia | <PROTECTED> |
| 16:24.18 | p3nguin | "on a x86_64 running Linux" <--- Do you realize I had to make my own patch to fix this grammatical error? |
| 16:24.37 | benngard | leifmadsen: /CFC/0317998975 : Gå\\\;tt for dagen |
| 16:24.45 | benngard | i try with 2 |
| 16:24.52 | p3nguin | I mean, how hard should it be for a dev to throw an 'n' on that a? |
| 16:25.30 | benngard | leifmadsen: with \\; /CFC/0317998975 : Gå\\;tt for dagen |
| 16:25.51 | leifmadsen | p3nguin: not hard, not high prioritiy |
| 16:26.03 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:26.06 | p3nguin | It's been fucked up for YEARS. |
| 16:26.41 | benngard | leifmadsen: it stores the \ in the databse :( |
| 16:26.47 | leifmadsen | benngard: it sure does |
| 16:26.55 | leifmadsen | p3nguin: yep |
| 16:27.05 | p3nguin | I would have thought that a 60 second time period would have opened up in the past eight years where it could have been corrected. |
| 16:27.36 | benngard | i want it to be: Gått for dagen |
| 16:27.48 | benngard | i can do it from cli |
| 16:27.50 | Amnesia | has no one else ever used an intercom here with asterisk? |
| 16:27.50 | tgrahamcapital | Does anyone have a recommendation on using lib_ss7 or chan_ss7? |
| 16:28.02 | p3nguin | amnesia: I use phones for intercom. |
| 16:28.39 | leifmadsen | p3nguin: perhaps no one noticed it, or filed an issue, or cared enough to fix it |
| 16:28.48 | leifmadsen | it takes more than 30 seconds because you then have to merge it across 4 branches |
| 16:29.21 | Amnesia | hm okay |
| 16:29.22 | p3nguin | I don't know anything about that. |
| 16:30.56 | benngard | i can ofc hard code the swedish caracters, but ugly as hell |
| 16:31.14 | p3nguin | office hard code? |
| 16:31.26 | p3nguin | Weird people and their funny dialects. |
| 16:32.17 | leifmadsen | ofc == of course |
| 16:32.23 | p3nguin | Huh? |
| 16:32.37 | leifmadsen | nevermind, I misinterpreted what you were confused about |
| 16:32.39 | p3nguin | There's no c is of. There's no of in course. |
| 16:32.40 | Amnesia | p3nguin: think it's a problem with the intercom or the connection between the intercom and the pap2t device? |
| 16:32.48 | p3nguin | s/is/in/ |
| 16:33.06 | leifmadsen | it doesn't make sense, but I presume 'ofc' == 'of course' |
| 16:33.24 | *** join/#asterisk Failrar (~Failrar@5ED66E6D.cm-7-7b.dynamic.ziggo.nl) |
| 16:33.47 | *** part/#asterisk tgrahamcapital (~trevorfra@ip68-12-146-104.ok.ok.cox.net) |
| 16:34.11 | BezNalogov | leifmadsen, in /usr/lib/asterisk/modules I do have app_queue.so. See: 10623416 -rw-r--r-- 1 root root 135632 2011-01-21 16:13 app_queue.so |
| 16:35.24 | p3nguin | I would never arrive at that conclusion because of how stupid it is. |
| 16:36.48 | p3nguin | A couple more funny ones that I see from some weird internet dialects: l8r and w8 |
| 16:36.54 | p3nguin | l8r = leightr |
| 16:37.02 | p3nguin | w8 = weight |
| 16:37.11 | p3nguin | They try to use if for later and wait. |
| 16:37.56 | *** join/#asterisk Tim_Toady (~moi@77.49.3.6.dsl.dyn.forthnet.gr) |
| 16:38.33 | p3nguin | amnesia: Until you connect a phone to the ATA and can both get calls and make calls, you haven't ruled out any one piece of the puzzle. |
| 16:39.38 | p3nguin | If you connect the phone and can send calls as well as receive calls, but when you connect the intercom device and it falls apart, the problem is with the intercom device and/or its connection to the ATA. |
| 16:39.56 | benngard | ugly way to hard code: SET(DB(CFC/0317998975)=GÃtt fÃr dagen), that works |
| 16:41.19 | Amnesia | p3nguin: already done that |
| 16:41.20 | Amnesia | and it works |
| 16:41.32 | p3nguin | both ways? in and out? |
| 16:41.36 | Amnesia | the intercom itself also dials a number when the button is pressed right? |
| 16:41.37 | Amnesia | yep |
| 16:41.50 | p3nguin | Just a minute ago you said one direction worked and the other failed. |
| 16:42.13 | Amnesia | well I can't try to make a call with the intercom since I havent got a dialpad:p |
| 16:42.23 | p3nguin | I know nothing about the intercom device. I doubt it dials anything just by pressing a button. |
| 16:42.59 | Amnesia | well then what happens when the button gets pressed? |
| 16:43.30 | p3nguin | Like I just said, I know nothing about the device. |
| 16:43.34 | Amnesia | when I simply pick up the horn when no one calls also nothing gets sent out |
| 16:43.39 | Amnesia | hm ok |
| 16:43.52 | p3nguin | Is it homemade? |
| 16:44.11 | p3nguin | If it is commercial, what make and model is it? |
| 16:44.55 | Amnesia | well I dunno |
| 16:44.59 | Amnesia | I think commercial |
| 16:45.12 | p3nguin | Does it have a brand on it? |
| 16:45.34 | p3nguin | sticker, decal, etching, engraving? |
| 16:46.58 | russellb | p3nguin: btw, regarding your rant about "a" vs. "an". It's not guaranteed to be x86 there. It could be powerpc. Writing code to try to figure out whether 'a' or 'an' is right seems like a pretty huge waste of time. |
| 16:47.24 | russellb | It could be a lot of different things. It's not as obvious as you acted like. |
| 16:49.20 | Amnesia | but there isnt any documentation/model number |
| 16:49.32 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 16:49.56 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 16:50.26 | Amnesia | p3nguin: nope |
| 16:50.29 | Amnesia | nothing |
| 16:50.36 | Amnesia | anyway thx for your help so far |
| 16:50.39 | Amnesia | I;m going home |
| 16:50.41 | Amnesia | cheerio:) |
| 16:51.14 | benngard | leifmadsen: u mean u cant put in a ; without the \ in the databse with SET(DB from dialplan? |
| 16:51.28 | leifmadsen | eh? |
| 16:52.22 | benngard | leifmadsen: i would like to have for example å in the database but it doesnt seems to work |
| 16:53.37 | leifmadsen | it sure doesn't |
| 16:53.41 | leifmadsen | I have no idea how to get around that |
| 16:54.01 | *** join/#asterisk navaismo (~navaismo@189.146.47.108) |
| 16:54.14 | benngard | oki, but shouldnt put(db remove the \ |
| 16:54.20 | leifmadsen | shrugs |
| 16:54.36 | benngard | and just place ; in the database? |
| 16:56.08 | *** join/#asterisk Iamnach0 (~Iamnacho@ip174-70-138-161.ks.ks.cox.net) |
| 16:56.13 | p3nguin | if [ `arch` = i686 ];then echo you are running an i686;else if [ `arch` = x86_64 ];then echo you are running an x86_64;else echo "you are apparently running something else";fi;fi |
| 16:56.40 | leifmadsen | how do you guarantee you've done it right for every architecture? |
| 16:56.42 | leifmadsen | you don't |
| 16:56.49 | p3nguin | hmm? |
| 16:57.00 | *** join/#asterisk GreatSUN (~greatsun@88-117-3-230.adsl.highway.telekom.at) |
| 16:57.48 | p3nguin | I just did it for two explicit architectures, and an all inclusive for everything that isn't one of the first two. |
| 16:58.44 | leifmadsen | that is an incredibly narrow focus |
| 16:59.16 | lordvadr | Has anybody used any of the voip cpe devices from EdgeWater? |
| 16:59.21 | p3nguin | Doesn't matter, though. I don't expect my one-liner shell script to be implemented as a check for Asterisk. I'm just talkin' about it. |
| 17:00.36 | *** join/#asterisk zerohalo (~zerohalo@173-13-92-27-NewEngland.hfc.comcastbusiness.net) |
| 17:00.55 | *** join/#asterisk Sipster_ (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 17:01.03 | *** join/#asterisk beardy_ (~beardy@unaffiliated/beardy) |
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| 17:02.20 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 17:02.34 | p3nguin | It could actually be broken into a group of architectures which start with a vowel sound and a group of those that don't. if group A, then "an"; else group B. |
| 17:02.54 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 17:03.24 | p3nguin | I'm sure there aren't too terribly many architectures that I wouldn't be able to compose a list of them all. |
| 17:04.23 | leifmadsen | you're welcome to provide a patch but I'm going to spend absolutely no time creating that patch |
| 17:04.53 | *** join/#asterisk zerohalo (~zerohalo@173-13-92-27-NewEngland.hfc.comcastbusiness.net) |
| 17:05.05 | p3nguin | I do want to reiterate, I'm not suggesting that my one line script be employed. |
| 17:09.56 | *** join/#asterisk phix (~threat@123-243-44-131.static.tpgi.com.au) |
| 17:10.55 | *** join/#asterisk fofware (~fofware@wdctf.siup.gov.ar) |
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| 17:26.50 | *** join/#asterisk esaym (~esaym153@cpe-72-181-109-14.satx.res.rr.com) |
| 17:28.25 | BezNalogov | leifmadsen, I have reinstalled asterisk completely, the directory modules has been regenerated. app_queue.so is there now, still asterisk doesn't know the command Queue |
| 17:28.41 | BezNalogov | Is there a way that I can see if the module is loaded |
| 17:28.55 | BezNalogov | or to try to load it via the console, so I might see what causes this |
| 17:37.30 | *** join/#asterisk MrWallace (~none_of@d5152C9CB.static.telenet.be) |
| 17:37.44 | *** join/#asterisk Joe_CoT (~joecot@pdpc/supporter/active/joe-cot) |
| 17:38.52 | Joe_CoT | Hi guys. When going to voicemail, it gives the message "The person at extension [] is unavailable". It should be reading an extension there, but it's not. Anyone know why that could be? I'm passing in the mailbox number, and $EXTEN is set |
| 17:39.31 | p3nguin | joe_cot: What "extension" do you want it to read, and what does it currently say when you reach the voicemail? |
| 17:40.11 | Joe_CoT | It currently says "The person at extension is unavailable". Either the voicemail box number, or the destination extension in $EXTEN, they're the same |
| 17:41.39 | p3nguin | So it's just a blank spot? |
| 17:41.49 | Joe_CoT | correct |
| 17:42.25 | *** join/#asterisk chasing`Sol (~cS@smtp.master-zone.net) |
| 17:43.19 | Joe_CoT | p3nguin, http://pastebin.com/ezwDFd35 |
| 17:43.27 | Joe_CoT | I'm quite confused by it |
| 17:46.53 | paulc | Joe_CoT: No prompts on the console about missing speech files etc (thinking maybe some of the vocab prompts are missing)... what are your verbose and debug levels set to? |
| 17:46.53 | Joe_CoT | Verbosity is at least 26 |
| 17:47.16 | paulc | Pretty high then :) How about debug? And what version of Asterisk? |
| 17:47.39 | Joe_CoT | Asterisk 1.6.2.9 |
| 17:47.49 | Joe_CoT | (trying to figure out how to get or set debug currently) |
| 17:47.54 | p3nguin | Nothing changes above verbose 4, just so you know. verbose 4 is exactly the same as verbose 215753. |
| 17:47.56 | krion | hi |
| 17:48.16 | Joe_CoT | p3nguin, I realize, I just add v's until my index finger gets tired |
| 17:48.18 | krion | is it a normal behaviour to get a "Didn't get a frame from channel" |
| 17:48.33 | krion | in debug enabled mode |
| 17:48.53 | krion | for channel.c, looks like it's ok because there is no warning, but juste to be sure |
| 17:49.30 | p3nguin | You can set the debug level with "core set debug <value>" |
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| 17:52.35 | p3nguin | joe_cot: I've never seen that problem before, but since your version is quite outdated, consider upgrading to the current version in your branch and see if you still have the problem. |
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| 17:54.25 | Joe_CoT | set debugging to 13, no extra messages there |
| 17:54.26 | paulc | Joe_CoT: Weird behaviour, it doesn't make sense. That said, it's early and I'm still waking up... can you pastebin the fragment from your dialplan, and a full console capture of the call from start to end? |
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| 17:55.50 | Joe_CoT | p3nguin, http://pastebin.com/ikLb9FxP updated |
| 17:56.06 | Joe_CoT | sorry, paulc, http://pastebin.com/ikLb9FxP |
| 17:56.11 | Joe_CoT | I'll try updating it |
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| 18:08.12 | Joe_CoT | compiling asterisk takes a while on an atom processor :( |
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| 18:18.27 | paulc | Joe_CoT: Hmm.. dunno - it all looks right to me, it should be playing the digits. I've got a 1.6.2.9 box here, maybe I'll have a play.. in the mean time, does it make any difference if you use a different context for your voicemail? 1000@vm instead of 1000@default (and setup in voicemail.conf right) |
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| 18:21.08 | p3nguin | joe_cot: I guess that depends on what Atom processor it is. My office PBX has a Pentium III (Coppermine) 930.354 MHz processor. |
| 18:21.54 | pabelanger | Joe_CoT: I _just_ picked up an Intel Core i7 CPU and mobo, installing them in 15mins |
| 18:21.56 | pabelanger | :) |
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| 18:27.10 | joachim_- | Hi |
| 18:27.38 | joachim_- | I get this error every 3 seconds... WARNING[1341]: file.c:751 ast_readaudio_callback: Failed to write frame |
| 18:27.44 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
| 18:27.49 | joachim_- | Is there any way i can stop this without restarting asterisk? |
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| 18:28.18 | p3nguin | If I can get everything in order, I'm wanting to build an embedded version of my existing PBX on a Via 800 MHz system. |
| 18:28.19 | joachim_- | Its kinda stuck. |
| 18:29.03 | p3nguin | It's stuck in a rut? |
| 18:29.11 | joachim_- | p3nguin: yup! |
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| 18:32.52 | GreatSUN | re |
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| 18:33.52 | GreatSUN | you know what is the best? |
| 18:34.06 | GreatSUN | if you have a working pbx :-D |
| 18:34.37 | *** join/#asterisk Poincare (~jefffnode@2001:5c0:150f:1704:230:48ff:fe86:1622) |
| 18:35.33 | GreatSUN | I love asterisk 1.8.x |
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| 18:40.59 | aut | can anyone recommend a good provider to host my 800 number? |
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| 18:50.30 | Joe_CoT | p3nguin, paulc, I updated to the latest 1.6.2, works now. wtf |
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| 19:11.27 | p3nguin | aut: Are you looking for a hosted PBX or looking only for a provider of a phone number for you own PBX? |
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| 19:22.32 | wasabi | Is anybody aware of a decent T.38 provider, that can sell a single trunk, for US calls? |
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| 19:22.46 | GreatSUN_ | re |
| 19:23.01 | GreatSUN | better like that |
| 19:23.05 | GreatSUN | :o) |
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| 19:56.46 | BlackBishop | any ideas on how could I make asterisk execute something upon .. for example .. a message is successfully sent through a datacard ? |
| 19:57.13 | blee_ | a script outside of asterisk probably |
| 19:57.21 | blee_ | asterisk -rx 'dothis' |
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| 19:59.06 | BlackBishop | yeah but asterisk only replies with the queue id |
| 19:59.23 | BlackBishop | I want something to happen when asterisk successfully sends it ! |
| 19:59.30 | WIMPy | Maybe you shouldn't use asterisk but a real SMS application? |
| 19:59.49 | BlackBishop | well, I can't use asterisk and an application the port is used by asterisk |
| 20:00.15 | BlackBishop | doesn't asterisk have some kind of events manager ? |
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| 20:00.41 | WIMPy | You have only one port? |
| 20:01.40 | BlackBishop | well, yeah .. /dev/ttyUSBx and it's open by asterisk to talk to the modem |
| 20:01.50 | BlackBishop | it's a datacard .. |
| 20:02.48 | WIMPy | Yes, but they usually have at least two ports. Often they have two ports with command interpreter so you can use them with two applications at the same time. |
| 20:04.49 | BlackBishop | one is audio .. one is data .. |
| 20:05.08 | WIMPy | Write a proxy :-) |
| 20:05.18 | BlackBishop | I wish I were that good |
| 20:06.06 | BlackBishop | googles some asterisk events stuff |
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| 20:13.32 | *** join/#asterisk pcangel (~yoink@www.vehiclestars.com) |
| 20:13.56 | pcangel | Hi I have dynamic members in queue 1000 with strategy=rrmemory and ringinuse=no |
| 20:14.11 | pcangel | all the members are using SIP - and the queue rings in on their second line if they are on the phone |
| 20:14.22 | pcangel | how can I make sure the SIP driver is properly reporting that they are on the phone? |
| 20:15.01 | WIMPy | do you have call counters enabled? |
| 20:15.34 | pcangel | I don't know what that is so I'm guessing no :p |
| 20:15.40 | pcangel | I'm searching for it now |
| 20:15.43 | wasabi | Revising questions: Anybody know any providers that support T.38, and are pay as you go? =) |
| 20:15.53 | WIMPy | sip.conf |
| 20:16.12 | Tozz_ | we support T38 |
| 20:16.14 | Tozz_ | but we are in .NL |
| 20:16.31 | pcangel | ok I've enabled it, I'll check if it's working how I expect it now |
| 20:17.09 | Katty | where can i temporarily host a file? |
| 20:17.14 | Katty | something small |
| 20:17.15 | Katty | under 10mb |
| 20:17.21 | Katty | just to share it |
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| 20:17.46 | wasabi | Revision #2: and terminates in the US |
| 20:19.14 | BlackBishop | Katty: rapidshare ? a filebox ? |
| 20:19.20 | BlackBishop | I have a filebox if interested |
| 20:19.40 | leifmadsen | dropbox? |
| 20:20.00 | BlackBishop | something like that |
| 20:20.19 | *** join/#asterisk Charrit (~Zairus@226.109.165.83.dynamic.mundo-r.com) |
| 20:20.24 | wasabi | This is amazingly hard to find a solution for. |
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| 20:50.35 | DDRP | Can I post a link to a job spec here? |
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| 20:52.25 | leifmadsen | #asterisk-biz and asterisk-biz mailing list |
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| 20:56.00 | benngard | i am tired atm, can any anser me how i add 1 hour to the result of ${STRFTIME(,,%H:%M)}? |
| 20:56.03 | DDRP | @leifmadsen Thanks, I shall investigate |
| 20:56.51 | leifmadsen | what does the output look like |
| 20:57.12 | benngard | 21:59 |
| 20:57.31 | benngard | but i would like tit to be 22:59 |
| 20:57.52 | lanning | change the timezone? |
| 20:58.06 | benngard | thought about it;) |
| 20:59.00 | benngard | i will ofc work, but i thought mabe it was a more pracical way to do it |
| 20:59.06 | benngard | it* |
| 21:02.51 | beek | would like tit now please. |
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| 21:18.19 | leifmadsen | benngard: Set(Result=${INC(STRFTIME(,,%H))}:${STRFTIME(,,%M)}) |
| 21:18.26 | leifmadsen | untested of course |
| 21:19.19 | LemensTS | i have a voip provider with user/secrete authentication. If I want to monitor r its registration, i did qualify=yes. Is this just seeing if that ip is pingable, or is it actually testing the registration between me and the voip provider? |
| 21:20.15 | WIMPy | somewhere in between |
| 21:20.23 | WIMPy | It's a ping but via SIP |
| 21:20.56 | LemensTS | If i changed the secret to a bad password, will the sip peer quit showing "OK" |
| 21:21.23 | WIMPy | no |
| 21:21.32 | WIMPy | spi show registry |
| 21:21.37 | WIMPy | sip |
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| 21:28.33 | LemensTS | Thanks WIMPy, when I do sip show registry it is blank tho. I have qualify=yes on that sip provider |
| 21:28.39 | *** join/#asterisk calmh (~jb@acro.nym.se) |
| 21:29.29 | WIMPy | Then you don't seem to register anywhere. |
| 21:29.30 | saxa | p3nguin: strangely now without touching anything the home phone works in both directions, calling it from my office, and calling from my home phone to the office. |
| 21:32.04 | pcangel | Thanks for the help WIMPy, turning on call counters solved my problem |
| 21:32.39 | pcangel | asterisk -rx reload crashed the server after enabling it, but it was a small price to pay |
| 21:33.54 | saxa | i have also an asterisk crash , with version 1.8.3 but i dont know why it crashed, i just saw it was not running anymore, and console was disconnected. |
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| 21:40.22 | LemensTS | WIMPy: is there a sip.conf setting that could prevent it from showing in 'sip show registry' ? I can make calls out, but it doesn't show up |
| 21:42.52 | WIMPy | You don't need to register to make calls. |
| 21:43.23 | WIMPy | That's only about telling your provider where to send calls. |
| 21:47.38 | LemensTS | WIMPy: yea, I am just curious on why it is not showing. I am writing a program that tells the user if the they are registered/unregistered to the voip provider |
| 21:48.58 | LemensTS | I mean it is working, so obviously it is registered. But im not sure how to display if ir is registered or not if it is not in sip show registry |
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| 21:50.10 | saxa | anybody knows to explain me why when I try to login into my voicemail, it seems that it doesnt receive any digit from my phone ? |
| 21:50.28 | nestAr | saxa: DTMF not in the right mode? |
| 21:50.30 | saxa | i use sip, and its dtmfmode=rfc |
| 21:50.59 | saxa | i have this setting, I think its dtmfmode=auto |
| 21:51.04 | WIMPy | LemensTS: Can you receive calls? |
| 21:51.07 | saxa | let me check to see what i have |
| 21:51.28 | nestAr | might try swapping it around |
| 21:52.06 | saxa | yes, will try to play with it, its dtmfmode=info |
| 21:52.19 | saxa | but in my sip phones i set it as rfc2833 |
| 21:52.41 | *** part/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 21:52.43 | LemensTS | WIMPy: I do not have a DID on that voip provider. I can set one up. If I cant receive a call, than i will fix it and try again. Thanks |
| 21:52.48 | saxa | since i have three options, in audio, rfc2833 and via sip info. |
| 21:53.35 | l0st-soul | hi there. could anyone point me to a technique could i use to keep original callerid after a '302 moved' forward ? |
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| 21:54.45 | GreatSUN | re |
| 21:55.18 | nestAr | saxa: make sure they match |
| 21:58.14 | l0st-soul | like, is there a way to set a variable in db when doing the original call, and reading it back to override callerid when initiating the call to the forwarded destination ? |
| 21:58.44 | l0st-soul | i would need a common value to both calls to be able to point at the same variable for both calls |
| 21:58.50 | l0st-soul | or something |
| 21:59.07 | saxa | nestAr: yeah, i got it working, I had a wrong setting in sip.conf, i had dtmfmode=info and on the phones i had via RTP. So now its working. Thanks! |
| 21:59.12 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-136-43.dsl.stlsmo.sbcglobal.net) |
| 22:00.02 | saxa | by the way I was reading there is a file one can configure the voicemail buttons for the listening of a message, is this to be done in a voicemail.conf in general context ? |
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| 22:03.04 | justsomedood | I'm having a problem after moving form asterisk 1.6 to 1.8.3 with a Polycom phone. When it gets a SIP INVITE (UDP) it sends the TRYING response, but usually asterisk retransmits another INVITE (probably because of a timeout) and the phone gets a second INVITE after it sent the TRYING and now gives an error 400 and stops ringing. This all happens under one second. |
| 22:04.51 | justsomedood | I don't know if I call tell asterisk to wait longer for a retransmit for this SIP device |
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| 22:11.06 | *** join/#asterisk demiurgeinc (~aaron@75-144-102-250-Miami.hfc.comcastbusiness.net) |
| 22:13.50 | demiurgeinc | Good Afternoon... I've been spending some time lately trying to get asterisk up and running to manage a small call center with deep integration into our web based administration system with AGI. I have most of all that figured out, but I wanted to integrate the full suite of communications.. namely IM and Presence. There seems to be a great divide with different technologies and ways to accomplish it. I initially tried it |
| 22:13.50 | demiurgeinc | Jabber, but connecting to both a jabber server and a sip server using SIP communicator.. its possible to voice and video chat over the jabber connection.. i want to have the user login just to the asterisk SIP account and be able to text as well. |
| 22:14.45 | demiurgeinc | Seems asterisk isn't quite built to support Instant Messaging on its own? I need Kamailio or OpenSIPs? In hopes that I dont have to reinstall asterisk again.. im hoping someone here can shed some light? |
| 22:16.22 | demiurgeinc | Is it possible to have instant message with Asterisk 1.6 by itself? Do I need to also install Kamailio? |
| 22:17.55 | *** join/#asterisk kuku (~kuku@c-24-13-139-34.hsd1.il.comcast.net) |
| 22:17.58 | demiurgeinc | Sip2Sip.info works the way I want, but there configuration is quite hefty.. what's the best way to achieve single server approach for Voice, Video, Instant Messaging, and Presence; using Asterisk as the core? |
| 22:18.44 | demiurgeinc | If I already have asterisk 1.6, is there simply configuration I'm missing to enable instant messaging / presence or do I need another component? |
| 22:19.45 | *** join/#asterisk fofware (~Fabian@host63.190-31-62.telecom.net.ar) |
| 22:23.54 | demiurgeinc | Anyone here? |
| 22:24.14 | IsUp | me. |
| 22:24.50 | demiurgeinc | Awesome. :) |
| 22:25.12 | demiurgeinc | Any ideas about my question? hehe |
| 22:25.52 | IsUp | no, not a clue |
| 22:27.01 | *** join/#asterisk chasing`Sol (~cS@smtp.master-zone.net) |
| 22:27.07 | demiurgeinc | haha awesome |
| 22:27.26 | demiurgeinc | Ive been building and rebuilding asterisk for the last bunch of days... |
| 22:27.50 | demiurgeinc | just want to figure out how to get Instant Messaging and Presence working with SIP. |
| 22:27.56 | demiurgeinc | using Asterisk as the core media server. |
| 22:27.58 | demiurgeinc | :( |
| 22:33.18 | leifmadsen | there is no instant messaging engine with asterisk (yet) |
| 22:33.21 | leifmadsen | looks at russellb |
| 22:34.08 | demiurgeinc | so if i want to add that on top, is kamailio the way to go? |
| 22:34.30 | demiurgeinc | seems to be the suggestion in information of pieced together, that or opensips? |
| 22:36.19 | demiurgeinc | the sip2sip.info seems to achieve exactly what id like to do.. but again they have servers for everything, mediaproxy, mrtp relay, opensips, asterisk, etc |
| 22:38.18 | demiurgeinc | im setting up a 10 person (20 person max) call center.. a single server option is what im after.. but not sure what other element of the sip2sip.info configuration is enabling the instant messaging capability when connected through sip communicator.. i believe its opensips.. and then kamailio seems to be easier to setup (they have a guide that shows setting it up on the same server).. just hoping to get some insight b4 i t |
| 22:38.19 | demiurgeinc | install and setup asterisk with kamailio. |
| 22:38.46 | demiurgeinc | kamailio and opensips were once one project that forked right? |
| 22:45.06 | *** join/#asterisk Joe_CoT (~joecot@pdpc/supporter/active/joe-cot) |
| 22:45.19 | Joe_CoT | so has anyone seen asterisk just dump dtmf packets before? |
| 22:46.02 | Joe_CoT | http://pastebin.com/Ymxgv4fK |
| 22:46.41 | Joe_CoT | 3 separate spurts of RFC2833 packets from my phone (10.10.10.53). The first 2 are relayed to 10.10.10.103, the last one isn't |
| 22:47.23 | Joe_CoT | as well, is it normal for packets to asterisk to say RFC2833, and packets from asterisk to another device to say DTMF? Both are have RFC2833 set in their SIP peer settings |
| 22:49.24 | *** join/#asterisk dancarlson (~dancarlso@S0106687f74d1e881.va.shawcable.net) |
| 22:52.48 | wasabi | Cool. Faxes working. |
| 23:02.34 | *** part/#asterisk demiurgeinc (~aaron@75-144-102-250-Miami.hfc.comcastbusiness.net) |
| 23:07.31 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 23:19.38 | *** join/#asterisk lost_soul (shawn@cpe-74-78-191-114.twcny.res.rr.com) |
| 23:31.46 | *** join/#asterisk seraphie (~erin@207.98.195.107) |
| 23:35.14 | *** join/#asterisk marienz (~marienz@freenode/staff/marienz) |
| 23:52.33 | *** join/#asterisk csnook (~chris@c-76-19-64-161.hsd1.ma.comcast.net) |
