| 00:01.23 | *** join/#asterisk bsaxon (~bsaxon@233.sub-97-190-72.myvzw.com) |
| 00:10.55 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 00:21.49 | *** join/#asterisk Zfield (~z@CPE00134936dbc6-CM0011ae922d14.cpe.net.cable.rogers.com) |
| 00:21.59 | *** join/#asterisk garymc (~chatzilla@host81-151-189-22.range81-151.btcentralplus.com) |
| 00:22.22 | Zfield | would anyone be able to help me to get asterisk working with google voice? |
| 00:25.22 | Zfield | how about upgrading from asterisk 1.7.* to 1.8? I'm not sure which exact version i have |
| 00:25.35 | Zfield | I just have the Asterisk Now package installed (which I did a couple days ago) |
| 00:28.43 | [TK]D-Fender | Zfield: AsteriskNOW has its own support channel, as does FreePBX. Neither are supported here |
| 00:28.52 | [TK]D-Fender | Zfield: #asterisknow , #freepbx |
| 00:28.59 | ChannelZ | they should be put to sleep |
| 00:29.34 | Zfield | thanks =) |
| 00:29.57 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-123-250.lsanca.fios.verizon.net) |
| 00:31.35 | Zfield | whats the difference b/w asterisk and asterisk now - i thought it was just the fact that asterisk now comes bundled w/ linux? |
| 00:33.40 | Zfield | ahh, right... the webgui also |
| 00:33.49 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 00:33.53 | [TK]D-Fender | Zfield: AsteriskNOW uses RPM packages which is a repo issue, and also bundles FreePBX which is a 3rd party GUI which completely owns your configs |
| 00:34.03 | [TK]D-Fender | Zfield: You aren't configuring *... IT is. |
| 00:41.08 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 00:46.01 | vinhdizzo | I was here last week and was referred to http://www.davidvossel.com/?p=28 where I could pick up some pointers on using google voice and asterisk. I successfully reproduced the example (but still don't understand much). Does anyone have some references for me to do a few things? 1) Set up an auto-attendant (someone calls google voice, asterisk picks up): play certain messages and transferring calls to different numbers. 2) |
| 00:48.49 | [TK]D-Fender | ~book |
| 00:48.49 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 00:48.52 | [TK]D-Fender | ^^ |
| 00:49.24 | vinhdizzo | [TK]D-Fender: still reading the book |
| 00:49.40 | vinhdizzo | im looking at "Building an Interactive Dialplan" |
| 00:51.32 | vinhdizzo | [TK]D-Fender: would it possible for you to explain to me what the setup in each of the .conf file does in the davidvossel site? If you have time. I think it'll help if I understand on how google voice is set up in my asterisk |
| 00:52.00 | vinhdizzo | the only thing I kind of know from VoIP is connecting to a SIP server from a softphone. |
| 00:55.26 | *** join/#asterisk eharris (~eharris@99-179-7-82.lightspeed.austtx.sbcglobal.net) |
| 01:03.04 | [TK]D-Fender | vinhdizzo: That site is talking about AMI scripting and a lot of other complicated things that require real programming knowledge. this is NOT for beginners, and is completely unnecessary for your goals from what I can tell |
| 01:03.28 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 01:03.39 | vinhdizzo | [TK]D-Fender: thanks. I know the AMI is advance stuff. I was more interested in the .conf files |
| 01:04.57 | [TK]D-Fender | zinIf you have a specific question I may be able to help but the "explain everything in there to me" bit is not. |
| 01:05.43 | vinhdizzo | ok |
| 01:05.52 | vinhdizzo | let me look it over and ask some specific questions then |
| 01:06.22 | vinhdizzo | manager.conf: this defines how the user access the asterisk server right? |
| 01:07.25 | vinhdizzo | jabber.conf: this is connecting the asterisk server to google voice? I've illustrated dialing out in the example. Does this have anything to do with receiving calls? |
| 01:07.52 | vinhdizzo | gtalk.conf: what does this do in addition to jabber.conf? |
| 01:08.47 | vinhdizzo | extensions.conf: this is where all the dialplan stuff go in right? tells asterisk exactly what to do when dialing and receiving calls? |
| 01:08.58 | vinhdizzo | many thanks! |
| 01:11.04 | [TK]D-Fender | [21:06]<vinhdizzo>manager.conf: this defines how the user access the asterisk server right? <- what "user"? |
| 01:11.15 | [TK]D-Fender | vinhdizzo: that is for AMI connections. |
| 01:11.24 | vinhdizzo | [TK]D-Fender: OK, for AMI |
| 01:11.43 | vinhdizzo | so it isn't needed for what I want to do, got it |
| 01:11.44 | [TK]D-Fender | [21:08]<vinhdizzo>extensions.conf: this is where all the dialplan stuff go in right? tells asterisk exactly what to do when dialing and receiving calls? <- yes |
| 01:12.11 | [TK]D-Fender | vinhdizzo: I suggest you do a basic GOOGLE on the other 2. |
| 01:12.42 | vinhdizzo | cool |
| 01:13.16 | vinhdizzo | So I narrowed down that I need to work on jabber.conf and gtalk.conf to set up google voice, and extensions.conf to set up auto-attendant |
| 01:13.26 | vinhdizzo | now i know where to look |
| 01:21.28 | vinhdizzo | [TK]D-Fender: a quick question, is it possible to send/receive fax with asterisk + voip provider such as google voice? yes/no is fine |
| 01:24.26 | [TK]D-Fender | vinhdizzo: GV no, I doubt they support T.38. Other providers may |
| 01:25.30 | vinhdizzo | [TK]D-Fender: So I just need a provider with T.38? Then faxing is possible? |
| 01:26.17 | [TK]D-Fender | vinhdizzo: yes |
| 01:51.36 | *** join/#asterisk Juggie (~Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 01:55.53 | ManxPower | many providers claim fax support, but they don't use T.38 |
| 01:59.20 | vinhdizzo | ManxPower: any suggestions on a provider that supports T.38 thats affordable? |
| 02:00.49 | ManxPower | vinhdizzo, no. |
| 02:00.57 | vinhdizzo | ok |
| 02:01.07 | ManxPower | we only do fax over pri or pots |
| 02:04.00 | *** join/#asterisk radic (~radic@178.2.210.197) |
| 02:04.42 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 02:32.10 | titter | ManxPower: Can you explain what the dahdi buffers mean, or point me somewhere I can read up on it. Specifically what immediate, full, or half mean. |
| 02:32.37 | titter | Been having issues with remote_disconnects and T2_timeouts with my fax for asterisk over pri. |
| 02:32.56 | titter | Only from one or two of our branch offices, but I it happens nearly 95% of the time |
| 02:34.20 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 02:48.09 | mmlj4 | well, there's the dahdi buffers, the mommy buffers, and the baby buffers |
| 02:48.22 | pabelanger | vinhdizzo: I did a blog post last week about Google Voice and Asterisk 1.8 |
| 02:48.24 | pabelanger | vinhdizzo: http://blog.polybeacon.com/2010/10/17/asterisk-1-8-and-google-voice/ |
| 02:51.20 | sshock | flowroute question: Can I test inbound calls on the $0.25 test credit, or only outbound calls? |
| 02:57.03 | p3nguin | sshock: Did you get a DID already? No DID, no inbound calls. |
| 02:58.27 | sshock | ok, that was my next question |
| 02:59.12 | sshock | hmm, but does that mean there is no way to make a SIP-only call to flowroute? |
| 02:59.25 | sshock | e.g., if my prefix number is 12345678 |
| 02:59.47 | sshock | can't I call 12345678@sip.flowroute.com from any sip phone? |
| 02:59.58 | ManxPower | sshock, that would be a question for flowroute tech support. Generally providers don't let you do that |
| 03:00.13 | ManxPower | sshock, um, why not call direct from the sip phone to your PBX. |
| 03:00.15 | sshock | really? that surprises me |
| 03:00.41 | sshock | I guess I was spoiled with gizmo... |
| 03:01.05 | sshock | ManxPower: actually you are right if I want to I will just call directly to * |
| 03:01.18 | sshock | but I was just curious about it |
| 03:01.45 | sshock | what about with my DID number once I get it? Will I be able to make a SIP call to DIDNUMBER@sip.flowroute.com ? |
| 03:01.50 | ManxPower | I suspect there is not enough demand to justify billing for those calls, setc. |
| 03:02.12 | ManxPower | sshock, providers are for PSTN connectivity. If you are not connecting to/from the PSTN you don't need a provider. |
| 03:02.39 | sshock | yeah, good point; I guess it's mainly just curiosity |
| 03:02.53 | ManxPower | providers don't spend money to make curious stuff work. |
| 03:03.22 | ManxPower | hell, our customers are stupid enough when they get PRI handoff from us. I'd hate to let them have direct SIP handoff. |
| 03:03.30 | sshock | well, Gizmo worked that way so I guess that's the only reason I was thinking about it |
| 03:03.53 | ManxPower | shudders in horror at the thought |
| 03:06.36 | *** join/#asterisk Mhaddog (~Mhaddog@99-207-66-217.pools.spcsdns.net) |
| 03:19.18 | sshock | any sip numbers out there that will help me do a reverse echo test? |
| 03:19.29 | sshock | i.e., they will call me back? |
| 03:22.39 | *** join/#asterisk BugKhaM (~BugKhaM@113.53.213.104) |
| 03:23.42 | BugKhaM | which SIP parameter is set to allow a SIP user to make an outgoing call without registration? |
| 03:25.03 | BugKhaM | for me If they are not registered, they will be sent to a default context |
| 03:25.21 | *** join/#asterisk GameGamer43|Mac (~GameGamer@CPE-65-27-76-78.new.res.rr.com) |
| 03:26.58 | p3nguin | bugkham: You've confused SIP REGISTER with AUTH to make a call. |
| 03:33.19 | *** join/#asterisk madduck (~madduck@debian/developer/madduck) |
| 03:33.54 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 03:33.57 | BugKhaM | p3nguin: Yeah, aren't they related? |
| 03:34.33 | ManxPower | BugKhaM, they are not related |
| 03:34.38 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 03:35.13 | ManxPower | registration tells the remote server what ip address is associated with which user/password. Mainly used with clients with dynamic ip address. |
| 03:35.16 | BugKhaM | ManxPower: I've noticed that if my sip client is not registered they'll be sent to my default context providing that they are using correct username/password |
| 03:35.49 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 03:35.49 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 03:35.54 | ManxPower | BugKhaM, you are incorrect. |
| 03:36.08 | BugKhaM | ManxPower: so sending to a default context meaning that there's something wrong with AUTH? |
| 03:36.17 | ManxPower | registration allows a server to send calls to a client that has a non-static IP address. It has nothing to do with calls from the client ot the server. |
| 03:36.31 | ManxPower | BugKhaM, asterisk accepts unauthenticated calls unless you have allowguest=no |
| 03:38.10 | BugKhaM | ManxPower: I have |
| 03:38.11 | BugKhaM | allowguest=yes and alwaysauthreject=yes |
| 03:38.13 | ManxPower | I normally set allowguest=no and put context=INVALID (that context just hangs up the call) |
| 03:38.22 | ManxPower | that's pretty useless. |
| 03:38.36 | *** join/#asterisk cnu (cnu@2001:470:28:1fe::10) |
| 03:39.05 | ManxPower | what did you say the ip address of your server is? I'll add it to my outbound routes. |
| 03:39.23 | BugKhaM | ManxPower: so, is that why my Unauthorized sip clients go to my default SIP context |
| 03:40.08 | BugKhaM | ManxPower: just sent you the private msg |
| 03:40.20 | ManxPower | I was joking. |
| 03:40.37 | BugKhaM | ManxPower: ohh, sorry |
| 03:40.45 | ManxPower | you mean your default dialplan context. there is no such thing as a "sip context" |
| 03:41.20 | BugKhaM | ManxPower: yes, that's right |
| 03:41.44 | BugKhaM | ManxPower: sorry to confuse you |
| 03:41.53 | ManxPower | I hope your default context is secure. |
| 03:42.52 | BugKhaM | ManxPower: I have a few extensions there but only for internal use |
| 03:43.47 | BugKhaM | ManxPower: because I thought that unregistered clients will be sent to the default context |
| 03:44.13 | BugKhaM | ManxPower: so I plaed a few extensions there for them to reach me |
| 03:45.52 | BugKhaM | ManxPower: Do you have any ideas on sip users with correct AUTH will be sent to the default context? |
| 03:46.33 | ManxPower | BugKhaM, they won't. |
| 03:46.52 | ManxPower | if they are authenticated they will go to the context you specify in their sip.conf [username] |
| 03:47.53 | BugKhaM | ManxPower: even with a bad network connection? |
| 03:49.16 | ManxPower | BugKhaM, authentication is sent with EVERY call. |
| 03:49.46 | ManxPower | maybe you should read up on how SIP works? |
| 03:51.15 | BugKhaM | ManxPower: OK, thanks for your advice, I better read more on that |
| 03:51.22 | ManxPower | client tries to send a call to the server, server rejects the call and hands back a "nonce", client uses the nonce + their password (MD5 Auth IIRC), and retries the call, server accepts the call if the auth is correct. |
| 03:51.53 | ManxPower | It works surprisingly like HTTP Digest Auth. |
| 03:52.13 | BugKhaM | ManxPower: OK |
| 03:53.23 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 03:56.42 | *** join/#asterisk sshock (~sshock@63.248.133.83) |
| 03:57.32 | *** join/#asterisk b0gatyr_ (~b0gatyr_@adsl-2-204-24.mia.bellsouth.net) |
| 04:05.07 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 04:05.46 | *** join/#asterisk felipe_ (~felipe@my.nada.kth.se) |
| 04:14.50 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 04:17.54 | p3nguin | bugkham: No, register tells Asterisk how to reach a device... auth tells asterisk that a device is allowed to make a call. |
| 04:18.58 | *** join/#asterisk coldsteal (~Administr@unaffiliated/coldsteal) |
| 04:20.50 | coldsteal | is there a way to send a notice or echo something to the console? |
| 04:21.35 | mmlj4 | write |
| 04:22.06 | mmlj4 | oh, with asterisk? |
| 04:22.15 | coldsteal | yes |
| 04:22.35 | coldsteal | from within a dial plan |
| 04:23.00 | mmlj4 | will NoOp work? I'd have to check |
| 04:23.23 | titter | mmlj4: nice answer lol |
| 04:25.28 | titter | So does anyone have any documentation on the dahdi buffer policy implementation? |
| 04:25.35 | mmlj4 | I get this: -- Executing [3000@chalupa-in:9] NoOp("SIP/deskphone-0000002a", "5042722060 authenticated") in new stack |
| 04:25.38 | mmlj4 | so yeah |
| 04:26.40 | mmlj4 | titter: I'm still waiting for someone to laugh at my baby bear joke |
| 04:37.56 | coldsteal | how can i make sure a call is reaching pbx |
| 04:38.24 | mmlj4 | watch the logs or console on the pbx while you make a call? |
| 04:38.33 | pabelanger | titter: http://blogs.digium.com/2010/04/21/more-fax-more-features/ |
| 04:39.05 | titter | pabelanger: I saw that, that's why I am asking |
| 04:39.20 | pabelanger | titter: What version of Asterisk you using? |
| 04:39.52 | titter | pabelanger: The buffers and the ability to set them in dialplan was backported to 1.6.2 as of 1.6.2.9, but it was undocumented. I am on 1.6.2.13 |
| 04:40.21 | titter | I was more or less curious if there is more documentation on the full, half, and intermediate settings |
| 04:47.04 | coldsteal | call isnt making it to pbx and im getting a "all circuits are busy now" |
| 04:47.19 | *** join/#asterisk mog (~mog@c-68-62-169-225.hsd1.al.comcast.net) |
| 04:50.14 | ChannelZ | calling from what to what? you haven't described your problem that well |
| 04:50.56 | leifmadsen | titter: Verbose() |
| 04:51.18 | pabelanger | titter: Not sure, but you can check the source for DAHDI and see what DAHDI_POLICY_WHEN_FULL, DAHDI_POLICY_IMMEDIATE and DAHDI_POLICY_HALF_FULL represent. All I know is they are for faxing |
| 04:51.31 | leifmadsen | just saw The Social Network -- good movie |
| 04:51.43 | leifmadsen | I was hesitant to go see it, but ended up being pretty entertaining |
| 04:51.52 | leifmadsen | makes the inventor of Napster look like a douchebag though :) |
| 04:52.16 | pabelanger | leifmadsen: I'm the Napster |
| 04:52.25 | leifmadsen | pabelanger: lol, also a good movie |
| 04:52.43 | leifmadsen | movie makes me want to invent something and stay up coding for 3 days ;) |
| 04:52.49 | ChannelZ | I'll wait for netflix |
| 04:53.26 | titter | leifmadsen: So Verbose(buffers) ? |
| 04:53.53 | leifmadsen | titter: I totally misunderstood what you were talking about then -- I was referring to the question about outputting information to the console from the dialplan |
| 04:54.13 | titter | haha I was confused .. thinking I missed something in the past three years |
| 04:54.29 | leifmadsen | beyond that I have no idea what you're talking about ;) |
| 04:54.36 | titter | lies! |
| 04:54.45 | leifmadsen | I haven't been drinking tonight, that's my problem |
| 04:54.53 | leifmadsen | thinking about going and getting my groove on with ableton live... |
| 04:55.17 | leifmadsen | I was flattered in the car when my neighbour asked if the Chemical Brothers I was playing was a song I wrote :) |
| 04:55.17 | titter | Basically, Digium support told me to Set(CHANNEL(buffers)=12,half) for my fax for asterisk problems ... just trying to understand this command a bit more as it hasn't helped |
| 04:59.33 | titter | leifmadsen: how do you like abelton? i've only used fruity loops |
| 05:00.02 | leifmadsen | I like ableton a lot. I have a launchpad as well. |
| 05:00.46 | titter | I might have to give it a whirl ... need to dust off my tascam |
| 05:00.54 | leifmadsen | haven't used it in a few months since I was in the process of moving, but I setup my studio/jam space in my new house |
| 05:01.13 | leifmadsen | got the electronic drums setup, and a desk with some gear |
| 05:01.24 | leifmadsen | just need to run some more power cords and bring the laptop downstairs |
| 05:01.53 | titter | I haven't used my stuff since moving into my condo ... neighbors complained about my alarm clock the first week, and have bitched about anything and everything |
| 05:02.04 | leifmadsen | need to find the rest of my audio cords so I can do some routing between the external sound card and my headphone amp |
| 05:02.43 | leifmadsen | that can be the problem with condos. I had my electronic drums there and never had a problem with complaints luckily enough. I had them setup on a couple of those foam mats that you use for standing on |
| 05:02.55 | leifmadsen | just doubled them up to reduce the vibrations |
| 05:04.19 | leifmadsen | ok, so I'm going to go do that I guess... either that or I'm going to decide to go to bed, and I'm too tired to start writing dialplan code for some new fun thing :) |
| 05:04.22 | leifmadsen | lates |
| 05:04.37 | leifmadsen | I'll leave you all with a treat though :) http://ofps.oreilly.com |
| 05:04.45 | titter | I have tried that, no dice. I got hold of a digital yahama piano last year, and have heard them grumble about that ... |
| 05:24.41 | *** join/#asterisk erinspice (~erin@207.98.195.107) |
| 05:41.27 | *** part/#asterisk MACscr (~macscr@c-98-214-103-147.hsd1.il.comcast.net) |
| 05:50.32 | *** part/#asterisk GameGamer43|Mac (~GameGamer@CPE-65-27-76-78.new.res.rr.com) |
| 06:38.48 | *** join/#asterisk sshock (~sshock@63.248.133.83) |
| 06:39.45 | sshock | Hi, can anyone tell me how can I debug MOH? I had it working earlier but now it is no longer working and I don't remember changing anything. |
| 06:41.28 | sshock | I tried core set verbose 10 and there are no error messages. It shows a "Started music on hold, class 'default'" message |
| 06:46.50 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 06:52.33 | titter | Whats the default look like in your moh conf? |
| 06:56.28 | sshock | mode=files |
| 06:56.34 | sshock | directory=/var/lib/asterisk/moh |
| 06:56.40 | sshock | random=yes |
| 06:56.49 | sshock | I would have used pastebin, but only 3 lines :) |
| 06:57.08 | sshock | the files I have in /var/lib/asterisk/moh are mp3 files |
| 06:57.43 | sshock | but that should be ok because I have the format_mp3 module loaded; and this was working the other day... |
| 06:58.10 | *** join/#asterisk jkroon (~jkroon@dsl-241-237-36.telkomadsl.co.za) |
| 07:04.15 | *** join/#asterisk Polysics (~Luca@95.236.24.31) |
| 07:04.56 | Polysics | hello! anyone has a link to a mirror for the changelog, pleasE? |
| 07:05.09 | Polysics | it looks like too many people were interested :-) |
| 07:06.09 | drmessano | http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0 ???? |
| 07:06.15 | drmessano | Works here |
| 07:07.11 | Polysics | sorry, i mis-expressed myself |
| 07:07.23 | Polysics | i would like to read CHANGES, not the changelog |
| 07:08.57 | *** join/#asterisk gavimobile (~user@bzq-84-108-105-6.cablep.bezeqint.net) |
| 07:10.16 | Polysics | http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=markup is not loading for me |
| 07:15.41 | *** part/#asterisk Tech_Travis (~Administr@cpe-76-168-191-127.socal.res.rr.com) |
| 07:17.19 | ChannelZ | yeah it was broken yesterday too |
| 07:17.49 | Polysics | i will just get the source package |
| 07:18.07 | Polysics | when will the addons for 1.8 be available, please? |
| 07:18.12 | Polysics | i use mysql with * a lot |
| 07:19.50 | *** join/#asterisk ritzt3ch (~ritzt3ch@ip70-189-221-21.lv.lv.cox.net) |
| 07:20.08 | ritzt3ch | how do you know in your sip.conf What context to use for dialing OUTBOUND ? |
| 07:20.23 | ritzt3ch | from siptrunk ASTERISK -> Mitel ? |
| 07:21.29 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 07:21.29 | [TK]D-Fender | ritzt3ch: HUH? |
| 07:21.47 | [TK]D-Fender | ritzt3ch: sip.conf does not use contexts for OUTBOUND |
| 07:21.50 | *** part/#asterisk sshock (~sshock@63.248.133.83) |
| 07:21.56 | Polysics | i am afraid i didn't understand either :-) |
| 07:22.03 | ritzt3ch | i know i never make sence |
| 07:22.05 | ritzt3ch | haha |
| 07:22.59 | ritzt3ch | i have a sip trunk Mitel - Asterisk and i have to set somehow to go to a context to where my extensions/Disa When dialing 8xxx -> the mitel what context shall i use |
| 07:23.06 | ritzt3ch | i put from-internal in the sip trunk |
| 07:23.42 | ritzt3ch | Executing [8905@disa-dial: |
| 07:24.04 | ritzt3ch | DISA Dial failed due to NOANSWER - returning to dial tone |
| 07:24.28 | ritzt3ch | im trying somehow to deliver those digits 8905 to route to my Trunk mitel-sip |
| 07:26.36 | titter | sshock: have you tried unloading and reloading the module? module unload format_mp3.so |
| 07:27.09 | ritzt3ch | nm i think i found it |
| 07:31.12 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 07:32.16 | ChannelZ | bollocks |
| 07:39.48 | ritzt3ch | damm thats why im getting a 120 Rejected |
| 08:04.58 | *** join/#asterisk Dovid (~Dovid@213.8.118.62) |
| 08:09.28 | *** join/#asterisk Marini (~maringjon@95.107.216.228) |
| 08:10.16 | Marini | hello i need some help, i desperately need to make a patch on asterisk and i don't know how to do it |
| 08:10.38 | Marini | patch is here: https://issues.asterisk.org/view.php?id=13243 |
| 08:11.04 | Marini | i want to know the steps to apply this patch |
| 08:11.11 | Marini | any help? |
| 08:14.25 | Marini | i am having asterisk svn 1.4 branch |
| 08:17.39 | Marini | anybody can help |
| 08:17.41 | Marini | ?? |
| 08:20.32 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 08:21.51 | *** join/#asterisk ferdna (~yup@cpe-24-92-114-97.elp.res.rr.com) |
| 08:40.41 | *** join/#asterisk Obeliks (obeliks@gentoo/contributor/Obeliks) |
| 09:02.32 | *** join/#asterisk mesfet (~mesfet@host165-3-static.25-87-b.business.telecomitalia.it) |
| 09:02.48 | *** part/#asterisk mesfet (~mesfet@host165-3-static.25-87-b.business.telecomitalia.it) |
| 09:09.55 | Marini | hello |
| 09:13.18 | heffer | Marini: patch -p0 < foo.patch |
| 09:13.31 | heffer | the -p0 may vary depending on how the patch was made |
| 09:20.43 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 09:32.02 | *** join/#asterisk imox1234 (~imox1234@p4FC5C356.dip0.t-ipconnect.de) |
| 09:51.20 | *** join/#asterisk ik_5 (~ik@109.226.17.43) |
| 09:51.26 | ik_5 | hello |
| 09:52.56 | ik_5 | I have a need to allow one person to talk on the phone any many others (it can be many people more then 100) just to listen on the phone. any idea what type of approach to take with this ? |
| 09:57.34 | ectospasm | ik_5: you can designate one channel with two-way audio, and other channels with one-way, see options to MeetMe |
| 09:58.08 | ik_5 | ectospasm, how many people can be in a meetme room ? |
| 09:59.00 | ectospasm | ik_5: depends on many factors. |
| 09:59.25 | ectospasm | ...network connection bandwidth, host system CPU power, RAM, etc. |
| 09:59.46 | ectospasm | ...codecs involved for each channel, etc. |
| 10:00.33 | ik_5 | ectospasm, i understand, thanks |
| 10:02.21 | ectospasm | ik_5: may want to google Asterisk dimensioning on voip-info.org, and look up documentatio for MeetMe (or ConfBridge) |
| 10:03.16 | ik_5 | ectospasm, already on that :) thanks |
| 10:15.40 | *** join/#asterisk Tim_Toady (~moi@178.128.8.62.dsl.dyn.forthnet.gr) |
| 10:18.12 | Marini | hello |
| 10:18.37 | Marini | i want help |
| 10:19.22 | Marini | i want to aplly a patch for asterisk, that i need, i haven't done it before |
| 10:19.27 | Marini | can somebody help |
| 10:19.42 | Marini | patch is here: https://issues.asterisk.org/view.php?id=13243 |
| 10:21.22 | heffer | Marini: patch -p0 < foo.patch |
| 10:21.24 | heffer | the -p0 may vary depending on how the patch was made |
| 10:22.26 | Marini | heffer: thanks, patch -p0 will make the differences, waht i do after that? compile all asterisk again, or only the the modified c file |
| 10:22.27 | Marini | ? |
| 10:22.38 | heffer | all of asterisk |
| 10:22.45 | heffer | usually is better to do so |
| 10:23.13 | *** join/#asterisk tengulre (~tengulre@221.182.46.142) |
| 10:23.27 | Marini | Heffer: ok, but my asterisk is with SVN branch, will it delete dhe patch later if i do svn update? |
| 10:23.53 | heffer | Marini: yes that is possible |
| 10:26.25 | Marini | thanks heffer, let me try it |
| 10:27.20 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 10:31.48 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 10:35.21 | titter | Anyone use dns srv records for iax? I added srvlookup=yes and it works, but doesn't failover |
| 10:36.37 | Marini | heffer: i executed 'wget 'https://issues.asterisk.org/file_download.php?file_id=19783&type=bug' -O - | patch -p0' |
| 10:36.53 | Marini | while being in /usr/src/asterisk |
| 10:37.17 | Marini | Resolving issues.asterisk.org... 76.164.171.231, 2001:470:e0d4::e7 |
| 10:37.18 | Marini | Connecting to issues.asterisk.org|76.164.171.231|:443... connected. |
| 10:37.18 | Marini | HTTP request sent, awaiting response... 200 OK |
| 10:37.18 | Marini | Length: 912 [text/plain] |
| 10:37.18 | Marini | Saving to: `STDOUT' |
| 10:37.18 | Marini | <PROTECTED> |
| 10:37.18 | Marini | Cannot write to `-' (Success). |
| 10:37.26 | Marini | what is this? |
| 10:37.59 | heffer | obviously the redirection to STDOUT doesn't work |
| 10:38.11 | heffer | just download the file to somewhere and try it from there |
| 10:41.51 | Marini | heffer: i didn't have patch app installed, i did it and the result now is |
| 10:42.25 | Marini | 100%[======================================================================================================================================================================>] 912 --.-K/s in 0s |
| 10:42.25 | Marini | 2010-10-24 12:38:17 (10.8 MB/s) - `-' saved [912/912] |
| 10:42.25 | Marini | (Stripping trailing CRs from patch.) |
| 10:42.25 | Marini | patching file channels/chan_sip.c |
| 10:42.25 | Marini | Hunk #1 FAILED at 6667. |
| 10:42.25 | Marini | patch unexpectedly ends in middle of line |
| 10:42.25 | Marini | Hunk #2 FAILED at 7022. |
| 10:42.26 | Marini | 2 out of 2 hunks FAILED -- saving rejects to file channels/chan_sip.c.rej |
| 10:43.56 | Marini | heffer: what you think i have done wrong? |
| 10:44.59 | heffer | Marini: first, you should use a pastebin service for pasting that amount of lines, second, the SVN copy probably is too new for the patch |
| 10:46.10 | Marini | sorry for my ignorance, but was is a pastebin? then i think that too, svn is newer file, but the bug is still present and need to solve it |
| 10:46.24 | heffer | Marini: other than that if you're really using the most recent 1.4 SVN the patch should already be in there |
| 10:46.44 | heffer | Marini: this is a pastebin: http://pastebin.com/ |
| 10:47.14 | heffer | Marini: i mean you can try applying this patch but i'm pretty confident that it doesn't fix your problem |
| 10:47.19 | Marini | let me explain shortly the bug, is about the functionality of SIP_codec variable to set preferred codec of incoming and outgoing leg of a call in the dialplan |
| 10:47.45 | Marini | and sip_codec changes codec only for for incoming leg and not for outgoing |
| 10:47.53 | Marini | the patch makes it possible for outgoing |
| 10:48.06 | Marini | and i need it in order to avoid transcoding |
| 10:48.13 | heffer | Marini: did you notice that the variable for outgoing calls is SIP_CODEC_OUTBOUND and not SIP_CODEC? |
| 10:48.26 | Marini | yes i tried it, is not working |
| 10:51.06 | *** join/#asterisk marksaitis (~Mk@78-61-148-80.static.zebra.lt) |
| 10:51.36 | heffer | i'm pretty sure that the error is in your configuration not the code |
| 10:57.53 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 11:09.47 | *** part/#asterisk Obeliks (obeliks@gentoo/contributor/Obeliks) |
| 11:11.34 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 11:17.23 | *** join/#asterisk marksaitis (~Mk@78-61-148-80.static.zebra.lt) |
| 11:18.00 | *** join/#asterisk nik_ (c4d28d92@gateway/web/freenode/ip.196.210.141.146) |
| 11:33.47 | marksaitis | does latest freepbx work on latest asterisk 1.8? |
| 11:42.06 | *** join/#asterisk jblack (~jblack@71.181.209.104) |
| 11:49.22 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 11:53.55 | *** join/#asterisk bjornts (~Adium@247.62-97-195.bkkb.no) |
| 11:58.41 | *** join/#asterisk visik7 (~Adium@unaffiliated/visik7) |
| 12:00.16 | *** join/#asterisk visik7 (~Adium@unaffiliated/visik7) |
| 12:07.57 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 12:23.31 | *** join/#asterisk lauris (~la@unaffiliated/lauris) |
| 12:28.18 | *** join/#asterisk hehol (~Adium@ip-78-94-0-76.unitymediagroup.de) |
| 12:30.56 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 12:36.41 | *** join/#asterisk tamiel (~tamiel@ip-81.net-89-3-209.rev.numericable.fr) |
| 12:54.13 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:01.25 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 13:02.05 | *** join/#asterisk timahvo1 (~rogue@41.223.57.75) |
| 13:07.35 | nik_ | hi |
| 13:07.55 | nik_ | any asterisk gurus online? WIMPy? |
| 13:10.12 | marksaitis | ;] |
| 13:10.35 | marksaitis | if asterisk 1.8 comes with tls srtp, where do I enable it? |
| 13:12.06 | pabelanger | marksaitis: if you have libstrp-dev installed, menuselect should enable res_srtp.so |
| 13:12.18 | pabelanger | ~ask |
| 13:12.18 | 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. |
| 13:12.21 | pabelanger | nik_: ^ |
| 13:12.33 | nik_ | I am rewriting my extensions.conf |
| 13:12.52 | nik_ | used the one installed by asteriskNOW, but there is too much stuff in it |
| 13:13.06 | nik_ | so I want to start clean and better understand how it works |
| 13:13.35 | nik_ | I've got incoming calls and outgoing calls working, but strangely the internal calls are failing |
| 13:13.50 | marksaitis | pabelanger, well, asterisk 1.8 is already installed on my system |
| 13:13.58 | marksaitis | how do I check for that do u know? |
| 13:14.06 | nik_ | exten = 3000,n,Dial(SIP/3000,40,${DIALOPTIONS}) |
| 13:14.29 | pabelanger | marksaitis: *CLI> module load res_srtp.so |
| 13:14.31 | nik_ | what is wrong with the above? I was expecting that if the extension dialaed is 3000, it will ring SIP/3000? |
| 13:15.11 | pabelanger | <PROTECTED> |
| 13:15.17 | marksaitis | pabelanger, it says module load failed |
| 13:15.32 | pabelanger | Then you didn't compile it |
| 13:15.38 | marksaitis | OMG |
| 13:15.42 | pabelanger | missing libstrp |
| 13:16.06 | marksaitis | I just tried yum install libsrtp-devel - nothing |
| 13:16.15 | marksaitis | libsrtp-dev and libsrtp |
| 13:16.16 | marksaitis | nothing |
| 13:16.38 | nik_ | pabelanger: chan_sip.c:20200 handle_request_invite: Call from '3021' to extension '3000' rejected because extension not found in context 'DLPN_local' |
| 13:16.46 | nik_ | pabelanger: same as before |
| 13:16.51 | pabelanger | Marini: Likely not in your repo, manually download and compile it |
| 13:17.00 | *** part/#asterisk rasengan (~rasengan@96.43.136.14) |
| 13:17.01 | *** join/#asterisk Jasnejac (kvirc@81.91.107.235) |
| 13:17.16 | nik_ | pabelanger: exten = 4999,1,VoiceMailMain(${CALLERID(num)}@default) |
| 13:17.19 | pabelanger | nik_: Looks like a context issue. |
| 13:17.34 | nik_ | pabelanger: this works just fine and it is the line above the exten = 3000 .... |
| 13:17.38 | marksaitis | pabelanger okay I will try that. How do I fully stick it in to my asterisk after that? |
| 13:17.40 | pabelanger | nik_: *CLI> dialplan show 3000@DLPN_local |
| 13:18.22 | pabelanger | marksaitis: you need to rebuild and install asterisk to detect the new libraries. ./configure && make && make install |
| 13:18.30 | nik_ | pabelanger: '3000' => 2. Dial(SIP/3000,40) [pbx_config] |
| 13:19.15 | marksaitis | pabelanger, really, I need to reinstall asterisk? will I loose my conffigs? |
| 13:19.40 | nik_ | pabelanger: hmm I think I got it |
| 13:19.53 | pabelanger | marksaitis: Yes you need to reinstall, no you won't loose your .configs. Do a backup before hand |
| 13:20.04 | nik_ | pabelanger: the sequence is 2, should be 1? So each extension needs to start from 1, sorry missed this |
| 13:20.23 | marksaitis | pabelanger, I will give it a spin |
| 13:21.32 | marksaitis | pabelanger, when u build srtp, do you know if there is a need to do "make install"? as it doesnt say so in install doc |
| 13:22.03 | pabelanger | Not sure, I would assume so |
| 13:22.25 | marksaitis | so I will need to find srtp somewhere in menuselect? |
| 13:22.34 | titter | marksaitis: As long as you don't run make samples it will not overwrite your configs. Hoever do a backup first. |
| 13:22.47 | marksaitis | titter, ok :) |
| 13:22.57 | marksaitis | thanx tit |
| 13:23.12 | pabelanger | marksaitis: After libstrp is installed, you can confirm within menuselect res_srtp is selected |
| 13:23.45 | marksaitis | im in makemenuselect now, which section? |
| 13:24.04 | marksaitis | yeah it seems it is in there |
| 13:24.16 | marksaitis | it has XXX kinda forced selection |
| 13:24.21 | marksaitis | im not sure, mabye I had it before |
| 13:24.33 | titter | XXX means it is missing a lib |
| 13:24.41 | marksaitis | really |
| 13:24.44 | pabelanger | yup |
| 13:24.49 | marksaitis | omg |
| 13:25.04 | titter | Look at the bottom of menuselect, it will say what it depends on |
| 13:25.09 | pabelanger | make sure you re-run ./configure |
| 13:25.15 | marksaitis | I just did |
| 13:25.23 | marksaitis | after configing and making libsrtp |
| 13:26.06 | pabelanger | try |
| 13:26.08 | pabelanger | ./configure --with-srtp=PATH |
| 13:26.18 | pabelanger | PATH = the path to libstrp |
| 13:26.55 | marksaitis | im not sure where libsrtp is. I think I didnt installed libsrtp correctly |
| 13:26.59 | marksaitis | il try make install |
| 13:27.01 | marksaitis | on it |
| 13:27.16 | titter | pabelanger: you ever configure iax with dns srv records? seems like it isn't failing over like I am expecting. It registers, but when I take down the first, it will just go unreachable. |
| 13:27.47 | titter | marksaitis: what build of linux are you using? |
| 13:28.02 | marksaitis | it seems it coppied to usr/local/lib |
| 13:28.08 | marksaitis | im using centos 5.5 |
| 13:28.17 | marksaitis | im just reconfing asterisk now, mabye it will find it |
| 13:28.19 | marksaitis | now |
| 13:28.38 | pabelanger | titter: not recently. Is dnsmgr enabled? |
| 13:29.30 | titter | pabelanger: I thought I did, but I might have missed it |
| 13:29.56 | marksaitis | ok, it seems ./configure on asterisk18 got stuck now on srtp section |
| 13:30.04 | marksaitis | it says it could not be linked |
| 13:30.07 | marksaitis | but it found it |
| 13:30.54 | titter | marksaitis: ftp://ftp.owlriver.com/pub/local/ORC/srtp/ wget it, and rpm -ivh it |
| 13:31.09 | marksaitis | ok I will give that a spin |
| 13:31.22 | *** join/#asterisk __derek__ (bc871549@gateway/web/freenode/ip.188.135.21.73) |
| 13:31.43 | pabelanger | marksaitis: pb your config.log |
| 13:31.44 | __derek__ | Hi... |
| 13:32.11 | marksaitis | [root@redc RPMs]# rpm -ivh srtp-1.4.4-1orc.src.rpm |
| 13:32.11 | marksaitis | <PROTECTED> |
| 13:32.11 | marksaitis | warning: group herrold does not exist - using root |
| 13:32.11 | marksaitis | warning: user herrold does not exist - using root) |
| 13:32.12 | marksaitis | ? |
| 13:33.39 | __derek__ | I have asterisk configured at 2 locations and an IAX trunk between them... We use a prefix to call the other location but the caller ID does not show the prefix of the first location so I cannot dialback, how do I resolve this? |
| 13:33.56 | marksaitis | titter, whats user herrold in this rpm? |
| 13:34.39 | __derek__ | For example I dial 464XXX from location 1 to location 2 |
| 13:34.53 | __derek__ | And 464XXX the other way round |
| 13:35.19 | __derek__ | oops I mean 736XXX the other way round |
| 13:35.39 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 13:36.29 | __derek__ | anyone? |
| 13:38.05 | marksaitis | titter ! |
| 13:39.01 | __derek__ | marksaitis: why don't u just create the user and group herrold before installing the RPM |
| 13:39.18 | cusco | __derek__: set the caller id var to a new value? |
| 13:39.27 | marksaitis | __derek__, thats not the point. Why the fuck would it give these stupid messages? |
| 13:39.44 | __derek__ | well who created the RPM? |
| 13:39.56 | cusco | yep, ask to the RPM's creator |
| 13:40.01 | __derek__ | cusco: but that would change the CID internally too.... |
| 13:40.02 | marksaitis | __derek__, dunno |
| 13:40.08 | __derek__ | internally we use just 3 digits |
| 13:40.35 | cusco | __derek__: so...? what is your point? |
| 13:40.46 | titter | marksaitis: because it is an rpm from a forum because it seems you were unable to build the source, trying to make it easier. |
| 13:41.03 | cusco | __derek__: perhaps you should use DUNDI, and not need them prefixes... |
| 13:41.05 | __derek__ | Internally I want to see only 3 digits |
| 13:41.30 | cusco | __derek__: well the internally set cid, is what will show on the softphone |
| 13:41.31 | __derek__ | externally I'd like to know that a call is from the other location |
| 13:41.40 | cusco | what do you mean externally? |
| 13:41.47 | cusco | set a new var then! |
| 13:41.49 | marksaitis | I see |
| 13:42.12 | __derek__ | marksaitis: it appears tha ta guy named herrold created the rpm.... |
| 13:42.19 | __derek__ | thats why ur facing the problem :) |
| 13:42.56 | marksaitis | nah, damn that rpm, I want a normal install |
| 13:42.57 | marksaitis | ;] |
| 13:43.13 | cusco | compile from source |
| 13:43.19 | marksaitis | im trying |
| 13:43.37 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 13:43.57 | marksaitis | I do ./configure, make and make install |
| 13:44.02 | marksaitis | it says it copies files |
| 13:44.08 | marksaitis | but asterisk install wont find it |
| 13:44.16 | __derek__ | maybe the prefix |
| 13:44.32 | marksaitis | checking for the ability of -lsrtp to be linked in a shared object... no |
| 13:44.32 | marksaitis | configure: *** |
| 13:44.32 | marksaitis | configure: *** libsrtp could not be linked as a shared object |
| 13:44.32 | marksaitis | configure: *** try compiling libsrtp manually and configuring with |
| 13:44.32 | marksaitis | configure: *** ./configure CFLAGS=-fPIC --prefix=/usr |
| 13:44.32 | marksaitis | configure: *** replacing /usr with the prefix of your choice |
| 13:44.32 | marksaitis | configure: *** |
| 13:44.38 | marksaitis | thats what asterisk ./configure says |
| 13:44.56 | cusco | that is wat we call dependencys |
| 13:45.07 | marksaitis | ? |
| 13:45.11 | cusco | you are missing some library |
| 13:45.13 | __derek__ | so did u try to compile srtp with --prefix=/usr ? |
| 13:45.17 | marksaitis | another one? |
| 13:45.19 | marksaitis | cusco |
| 13:45.42 | cusco | asterisk will use many libraries, and those libraries will use many others... |
| 13:45.53 | marksaitis | __derek__, I wouldnt know what should I give to --prefix=??? |
| 13:45.54 | cusco | you already installed one? good luck with the next 50 |
| 13:46.14 | __derek__ | marksaitis: do a ./configure --prefix=/usr for srtp |
| 13:46.17 | marksaitis | cusco, well, all I want is that damn srtp support |
| 13:46.25 | __derek__ | make and make install |
| 13:46.30 | cusco | ok |
| 13:46.39 | marksaitis | __derek__, I did so in libsrtp |
| 13:46.52 | __derek__ | with the prefix? |
| 13:46.55 | marksaitis | and thats the message it gives when I ./configure asterisk |
| 13:47.13 | marksaitis | as I said, I do not know what should I point --prefix= to |
| 13:47.26 | __derek__ | set it to --prefix=/usr |
| 13:47.47 | __derek__ | thats the default for asterisk but most probably not the default for srtp |
| 13:50.37 | __derek__ | cusco: the issue I'm facing is that I won't be managin the asterisk system... and freepbx is installed... so not sure about the "set a variable" option |
| 13:52.08 | __derek__ | marksaitis: any luck? |
| 13:52.10 | marksaitis | nah, fuckit, no luck |
| 13:52.30 | __derek__ | hmmm |
| 13:53.11 | marksaitis | that stupid message asks me to compile it manually... thats exactly what im doing |
| 13:53.35 | __derek__ | marksaitis: u extracted the sources of srtp |
| 13:53.43 | __derek__ | cd srtp-1.4.4/ |
| 13:53.51 | __derek__ | ./configure --prefix=/usr |
| 13:53.53 | __derek__ | make |
| 13:53.56 | __derek__ | make install |
| 13:54.02 | marksaitis | thats what I did :) |
| 13:54.03 | __derek__ | cd ../asterisk-XXX |
| 13:54.12 | __derek__ | strange |
| 13:54.28 | marksaitis | and then I been doing ./configure inside asterisk18 |
| 13:55.03 | __derek__ | yeh I installed asterisk 1.8 yesteday.... and it installed into prefix /usr |
| 13:55.15 | __derek__ | so I don't see why its complaining |
| 13:55.42 | marksaitis | u installed asterisk or libsrtp to /usr? |
| 13:56.21 | __derek__ | I installed asterisk without setting a prefix |
| 13:56.26 | __derek__ | and it chose /usr |
| 13:57.01 | marksaitis | well it worked for me. But when you try to add srtp support, it doesnt work this way |
| 13:57.14 | __derek__ | hmmm ok |
| 13:57.22 | __derek__ | lemme try |
| 13:57.46 | marksaitis | what distro do u have? and where r u getting srtp for urself? |
| 13:58.12 | __derek__ | I've installed without srtp |
| 13:58.18 | __derek__ | am trying with srtp now |
| 13:58.40 | *** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 13:58.41 | marksaitis | well, u need to compile srtp. where r u getting urs from? |
| 13:58.51 | __derek__ | its ubuntu |
| 13:58.54 | __derek__ | its there in the repos |
| 13:58.59 | marksaitis | is it |
| 13:59.01 | marksaitis | u lucky |
| 13:59.24 | __derek__ | yeh, ur on fedora? |
| 13:59.31 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:59.31 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:59.31 | marksaitis | im on centos 5.5. |
| 14:00.17 | __derek__ | the asterisk rpms don't have srtp I guess... they have some rpms for asterisk 1.8 tho |
| 14:01.05 | marksaitis | dunno what u mean. i am trying to libsrtp from src |
| 14:02.15 | __derek__ | marksaitis: did u try using the ./configure --with-srtp= |
| 14:02.34 | __derek__ | when running configure for asterisk18 |
| 14:02.47 | FiReSTaRT | out of curiosity, is any one here running * on arch? |
| 14:04.12 | marksaitis | __derek__, im done now. It seems I just did need to compile libsrtp with a flag option and no prefix options at all |
| 14:04.23 | __derek__ | oh |
| 14:04.25 | __derek__ | cool |
| 14:04.47 | __derek__ | I thought thats what u tried in the first place :) |
| 14:05.10 | marksaitis | no, I dont know why did it want that flag... |
| 14:05.40 | __derek__ | ok |
| 14:08.35 | *** part/#asterisk __derek__ (bc871549@gateway/web/freenode/ip.188.135.21.73) |
| 14:09.59 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 14:17.56 | p3nguin | firestart: I do |
| 14:18.33 | leifmadsen | marksaitis: what command did you run to get it to work? |
| 14:18.46 | leifmadsen | marksaitis: just curious, because I'm going ot try here in a moment for documentation purposes |
| 14:19.05 | FiReSTaRT | p3nguin: i'm installing arch for the first time.. any advice on which packages to preselect.. i probably wanna run it with xfce while learning asterisk, but eventually ill run it without a gui for production |
| 14:20.05 | p3nguin | firestart: Well, you certainly don't need Xfce to run Asterisk... pick what you need, then install Asterisk. |
| 14:21.14 | p3nguin | firestart: The nice thing about Arch is that even things which are not packages are going to end up being packages with dependency resolution. See the wiki on using AUR if you don't know about it. |
| 14:21.54 | FiReSTaRT | p3nguin: i don't need it, but it will make initial editing a lot easier.. i never truly warmed up to to vi and i like to visually see where my files are going until i get a better lay of the land |
| 14:21.57 | p3nguin | firestart: Which branch/version of Asterisk do you want to install? |
| 14:22.39 | FiReSTaRT | p3nguin: the latest stable release... i heard something about 1.8 already being out (haven't seen it as stable on friday), if not... 1.6 |
| 14:22.47 | leifmadsen | 1.8.0 |
| 14:22.48 | marksaitis | leifmadsen, it was ./configure -CFLAGS=-fPIC on srtp |
| 14:23.01 | leifmadsen | marksaitis: hmmm, ya I just did that too and it still says it can't link as a shared library |
| 14:23.10 | leifmadsen | I'll have to figure out what I'm doing wrong. Have to get ready for astricon! |
| 14:23.31 | marksaitis | leifmadsen, also, in my case, I tried to select all in section module embedding using menuselect, but if u do so, it does not compile ;] I just deselected all of them, and it compiled perfectly |
| 14:23.41 | *** join/#asterisk deonv (~adium@196.1.28.226) |
| 14:23.57 | leifmadsen | marksaitis: ah ya, I don't module embed (I use external files per the default) |
| 14:24.03 | leifmadsen | I don't even get past the ./configure part though :) |
| 14:24.10 | leifmadsen | I'll just ask twilson at astricon |
| 14:24.29 | p3nguin | firestart: You'll need to install a suitable AUR helper (the last recommendation was to use clyde), but Asterisk 1.8.0 is available. |
| 14:24.44 | marksaitis | leifmadsen, ok |
| 14:24.46 | FiReSTaRT | p3nguin: i have no issues with installing from source :) |
| 14:25.08 | p3nguin | firestart: Right. Installing via AUR _is_ installing from source. |
| 14:25.22 | FiReSTaRT | p3nguin: i mean manually compiling |
| 14:25.43 | marksaitis | does anyone know if there is an easy way turning tls on in freepbx? |
| 14:25.57 | p3nguin | firestart: Please learn how to use pacman and AUR if you're going to use Arch. |
| 14:25.58 | leifmadsen | unless the gui supports it, then probably not |
| 14:26.40 | FiReSTaRT | p3nguin: that i will, of course :) package managers are just too useful not to know when you run a new distro |
| 14:27.14 | p3nguin | firestart: If it's too terribly difficult to run "clyde -S asterisk", you're in the wrong distro. |
| 14:28.33 | cusco | what is aur? |
| 14:28.36 | FiReSTaRT | p3nguin: wait... don't i just browse to asterisk.com, download the exe to my desktop, find it among other installers and doubleclick? :O |
| 14:28.55 | p3nguin | cusco: It's the Arch Users' Repository. |
| 14:29.11 | cusco | that is not mainstream I would guess |
| 14:29.30 | marksaitis | does anybody know what is the maximum rsa key size asterisk supports? |
| 14:29.40 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 14:29.42 | FiReSTaRT | cusco: arch is a linux distro for those that aren't afraid of getting their hands dirty... if you want easy, just run ubuntu :) |
| 14:29.57 | cusco | cool FiReSTaRT, I wasn't aware |
| 14:30.06 | leifmadsen | I like Ubuntu and CentOS :) |
| 14:30.09 | titter | marksaitis: I just tried and had the same results ... tried to compile it a few ways |
| 14:30.14 | leifmadsen | and now I must leave to pack |
| 14:30.15 | leifmadsen | lates! |
| 14:30.30 | cusco | I actually thougt arch was a ubuntu-like distro, and slackware or gentto was something you used to "get your hands dirty" |
| 14:30.39 | p3nguin | cusco: It's where we store unsupported pkgbuilds for software for Arch. The pkgbuild is like a makefile for a source build, but it rolls the build into a package so that all software gets managed appropriately. |
| 14:30.58 | cusco | whatever |
| 14:31.14 | FiReSTaRT | leifmadsen: i gave cent a quick try but i accidently destroyed a dep for gnome while trimming it down, so i figured i better run arch, start from the bottom and work my way up instead of trimming the dead branches |
| 14:31.15 | p3nguin | Arch is like the new Gentoo. |
| 14:31.23 | cusco | ok |
| 14:32.17 | p3nguin | Coming from years of Slackware use, Arch was a nice change of scenery. |
| 14:32.23 | titter | So has anyone used dns srv records with iax ... doesn't want to failover |
| 14:40.12 | *** join/#asterisk timahvo1 (~rogue@41.223.57.75) |
| 14:44.06 | jamko | Morning. I have a second parking lot setup, and seems to be functioning properly so far. My questions are, is it "proper" for asterisk to be using "Local" when parking the call to the secondary parking lot? Also, I am having to use attended xfer with the 2nd lot, but with the default I can just use #1. Is this normal? |
| 14:48.16 | jamko | I am using: exten = > 600,1, Set(CHANNEL(parkinglot)=parkinglot_new) ; exten => 600,n,Park ---- to use the second lot. |
| 14:51.55 | [TK]D-Fender | jamko: Well Park DOES need to know WHICH to park to. This is specific to the person parking, so features.conf does not know any better. |
| 14:52.15 | [TK]D-Fender | jamko: there is a CHANNEL VARIABLE which chooses the lot if you dont pass it manually to Pak() |
| 14:52.43 | [TK]D-Fender | jamko: Go set it different if you want each device to point to a specific lot for your #1 |
| 15:04.18 | *** join/#asterisk Benwa (~azerty@unaffiliated/benwa) |
| 15:04.26 | Benwa | hi |
| 15:10.36 | jamko | [TK]D-Fender: Okay.. but why does it seem certain features are automatically built into the dialplan from the [general] of features.conf ? For example the default pickupexten *8 just "works" , but if I add an alternate pickupexten in my second lot, it does not |
| 15:11.08 | jamko | register with asterisk unless I use the parkedcall app in the dialplan? |
| 15:11.34 | jamko | I don't have *8 anywhere in my extensions.conf, but yet it still works. |
| 15:15.41 | marksaitis | Seriusly, has anybody had REGISTRATION ERROR fORBIDDEN 403? I mean my sip client was working fine, but after I added tls support this is what happens. Even on new users not configured for transport=tls |
| 15:15.54 | *** join/#asterisk JuStIcIa_ (~justicia@190.52.236.133) |
| 15:21.34 | marksaitis | ok I know why that happens, once u put transport=tls, it only allows tls I suppose |
| 15:21.53 | WIMPy | That's the idea. |
| 15:22.03 | WIMPy | You can put a list there. |
| 15:22.07 | marksaitis | but even though, after configuring certificates and all the other stuff, I cant connect using tls anyway - error 503 |
| 15:22.17 | marksaitis | service unavailable |
| 15:23.10 | WIMPy | Is your phone able to verify the certificate? |
| 15:23.16 | marksaitis | WIMPy, have u ever managed to get tls working? |
| 15:23.53 | WIMPy | yes |
| 15:24.12 | marksaitis | and u never had error 503 on client? |
| 15:25.12 | WIMPy | Can't remember. But in case you used a self-signed certificate, you will have to upload that to your phone(s). |
| 15:25.38 | marksaitis | WIMPy, really? Yeah it is self signed. |
| 15:26.12 | marksaitis | I dont think CounterPath Bria softphone version on iphone 4 has such a capability |
| 15:26.26 | WIMPy | Depends on your phone, off course, but if it can't veryfy it, it might refuse to connect. |
| 15:26.29 | marksaitis | WIMPy, and thats a 100% must do? |
| 15:27.01 | marksaitis | WIMPy, strange thing is that it drops me same erro no matter if I close or open 5061 tcp on the server |
| 15:27.08 | WIMPy | Does the phone have a log? |
| 15:27.21 | marksaitis | no it has nothing |
| 15:27.30 | marksaitis | at least not in the GUI |
| 15:28.18 | WIMPy | Ok, sounds like something else is going on. |
| 15:28.36 | marksaitis | it kills me |
| 15:28.54 | marksaitis | I followed all the documentation, completely everything |
| 15:29.00 | marksaitis | simple sip accounts work fine |
| 15:29.16 | marksaitis | srtp support is compiled in, asterisk listens on sip-tls port |
| 15:29.37 | marksaitis | I have enited one user for tls support as asked by documentation |
| 15:29.43 | marksaitis | but it just wont work |
| 15:29.49 | marksaitis | dear oh dear |
| 15:33.26 | marksaitis | somebody,any sugestions? |
| 15:33.47 | marksaitis | any commands on how could I check if asterisk gets any kind of connection over sip-tls? |
| 15:33.51 | marksaitis | please |
| 15:33.52 | marksaitis | :) |
| 15:34.21 | WIMPy | tcpdump? |
| 15:34.46 | marksaitis | I know there is something inside asterisk cli |
| 15:34.57 | marksaitis | I dont know how to use tcpdump |
| 15:35.02 | marksaitis | in my case |
| 15:35.10 | titter | core set verbose 10 might help |
| 15:35.13 | WIMPy | sip debug. |
| 15:35.21 | titter | sip debug as well |
| 15:35.32 | WIMPy | But I have no idea how much information you get on tls connects there. |
| 15:35.43 | marksaitis | titter - good sugestion |
| 15:35.52 | marksaitis | == Problem setting up ssl connection: error:00000000:lib(0):func(0):reason(0) |
| 15:36.03 | marksaitis | thats what it gives me when i trie to connect with tls enabled client |
| 15:36.12 | marksaitis | any clues what should I do next? |
| 15:36.17 | titter | google |
| 15:36.26 | marksaitis | lets give it a go |
| 15:36.54 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 15:38.10 | marksaitis | not much luck |
| 15:39.25 | marksaitis | anything else I could try? |
| 15:42.11 | *** join/#asterisk imox1234 (~imox1234@p4FC5C356.dip0.t-ipconnect.de) |
| 15:43.12 | marksaitis | im trying tcpdump now, gosh, what a wonderful utility |
| 15:45.01 | marksaitis | but not much use in my case |
| 15:47.46 | marksaitis | :( |
| 15:48.19 | *** join/#asterisk nunners (~chatzilla@host86-134-211-99.range86-134.btcentralplus.com) |
| 15:49.12 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 15:49.14 | p3nguin | What are you looking for? |
| 15:49.23 | nunners | Evening all... I'm having to reinstall my asterisk pbx server, due to a hardware/os problem, and am wondering whether to install 1.8 instead of 1.6. I know it's been formally released, but is 1.8 robust enough to use in a live environment, or should it still be considered for testing only? |
| 15:49.57 | marksaitis | p3nguin, on asterisk cli I get error - == Problem setting up ssl connection: error:00000000:lib(0):func(0):reason(0) |
| 15:50.10 | marksaitis | when I trie to connect over tls with a softphone |
| 15:51.42 | p3nguin | tcpdump and wireshark aren't helping? |
| 15:52.25 | marksaitis | well, tcpdump doesnt show much |
| 15:52.45 | marksaitis | it just show that ther has been a lil bit of communication between the two |
| 15:53.01 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 15:53.31 | marksaitis | is tha tfucking ssl so fucking rocket science... |
| 15:54.08 | marksaitis | its 3rd day im playing with it |
| 15:55.06 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 15:55.16 | *** part/#asterisk nunners (~chatzilla@host86-134-211-99.range86-134.btcentralplus.com) |
| 15:55.24 | WIMPy | What? Cryprography is NOT rocket science? |
| 15:55.44 | marksaitis | cryptography is. But I mean adding its support in here |
| 15:55.45 | marksaitis | ;] |
| 15:56.24 | WIMPy | Well, you have a connect, now comes the crypto part. |
| 15:57.15 | marksaitis | well, I have absolutely no idea on what should I do next |
| 15:59.41 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 16:00.07 | marksaitis | WIMPy |
| 16:00.24 | marksaitis | do you think thers a chance that it has to do something with this? |
| 16:00.25 | marksaitis | http://ejd021-linux.blogspot.com/2010/05/ssl-error00000000lib0func0reason0.html |
| 16:00.58 | WIMPy | Hard to say. Maybe worth a try. |
| 16:02.19 | marksaitis | nah, centos 5.5 doesnt want to downgrade it |
| 16:03.47 | marksaitis | dear oh dear, damn who invented those errors |
| 16:06.25 | marksaitis | WIMPy, do u know if thers a log I could check? |
| 16:07.26 | WIMPy | sorry |
| 16:07.57 | marksaitis | does anybody know if asterisk tls supports 2048 bit certificates? |
| 16:08.11 | marksaitis | where the hell I can find such info |
| 16:08.59 | bougyman | that would depend on the lib it uses for ssl, i'd think. |
| 16:09.02 | bougyman | not on asterisk itself. |
| 16:09.35 | marksaitis | I see |
| 16:09.57 | marksaitis | I dont know what should I do |
| 16:14.15 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 16:14.19 | [sr] | 1.8 is out, cool! |
| 16:15.03 | marksaitis | yeah it is |
| 16:15.03 | bougyman | 1.9.2 is out |
| 16:15.09 | bougyman | oh |
| 16:15.19 | bougyman | wrong software, heh |
| 16:15.35 | *** join/#asterisk timahvo1 (~rogue@41.223.57.73) |
| 16:16.34 | marksaitis | WIMPy, u said u got it working. |
| 16:16.49 | marksaitis | what softphone have u tested it with? have u been using selfsigned certs? |
| 16:17.24 | WIMPy | I'm using a SNOM 360 and that doesn't care about the self signed cert. |
| 16:17.51 | marksaitis | ok |
| 16:18.05 | WIMPy | It just logs a warning. |
| 16:18.09 | marksaitis | and it just works for u doesnt it? for incoming and outgoing tls srtp calls? |
| 16:18.34 | WIMPy | I'm still on 1.6, so not srtp yet. |
| 16:18.44 | marksaitis | dear oh dear, why does it not work for me |
| 16:19.17 | jamko | I am having an issue with re-parking a call via DTMF, after it has been picked up. The problem being that when dialing the auto created extension of the parked call, asterisk does not specify "T" option in the DIAL , so therefore DTMF tones are not recognized. |
| 16:20.10 | jamko | Is there a way to get asterisk to specify the T optionwhen dialing the auto created extensions? |
| 16:21.48 | *** join/#asterisk felipe_ (~felipe@my.nada.kth.se) |
| 16:22.09 | marksaitis | Following the forums I read, the error I got is very common. I think centos 5.5 have a bad openssl..... I dont know how things like that can happen on server distros... just magical |
| 16:22.39 | jamko | I know I can create an alternate extension to point to the parked extensions, but that just seems over complicated for users to have to dial, ie - 800 to access 700 .. |
| 16:22.43 | WIMPy | Nothing special. |
| 16:22.52 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 16:23.03 | marksaitis | it looks special, its ssl, the world depends on it |
| 16:23.09 | WIMPy | The have been at least two dostros with broken compilers. |
| 16:23.22 | marksaitis | how do these things happen |
| 16:23.27 | WIMPy | And actually from the same source. |
| 16:23.39 | marksaitis | dear oh dear |
| 16:23.58 | marksaitis | anybody knows any good yum repo for centos 5.5 with the newest openssl? |
| 16:24.16 | WIMPy | Well, some distros want to be ahead of others. Often ending up, being adhead on bugs. |
| 16:24.33 | marksaitis | omg, I typed in yum remove openssl, and it wants to remove 829 other packages.... |
| 16:24.50 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-139-208.ks.ks.cox.net) |
| 16:25.16 | marksaitis | strangely enough, centos 5.5 wants to be below others like at least 10 years ;] |
| 16:26.11 | [sr] | ei WIMPy |
| 16:28.32 | WIMPy | [sr]: Already compiling 1.8? ;-) |
| 16:29.08 | [sr] | WIMPy: na... for now i have a testing machine with asterisknow... still dont have nothing in production |
| 16:29.32 | WIMPy | You downgraded to GUI? |
| 16:29.51 | [sr] | well...using freepbx... yes :p |
| 16:31.39 | marksaitis | WIMPy, what version of openssl do u have? |
| 16:31.55 | *** join/#asterisk Tim_Toady (~moi@178.128.8.62.dsl.dyn.forthnet.gr) |
| 16:32.20 | WIMPy | 0.9.8n |
| 16:32.46 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 16:32.51 | marksaitis | ok |
| 16:33.53 | ManxPower | ~freepbx |
| 16:33.53 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 16:33.55 | marksaitis | WIMPy, do u know how to remove it with yum but leave all other packages it wants to remove? |
| 16:34.00 | ManxPower | ~asterisknow |
| 16:34.00 | infobot | hmm... asterisknow is based on Asterisk, but is difficult to support in #asterisk for a number of reasons. Please seek support in #asterisknow instead. |
| 16:34.11 | ManxPower | ~gui |
| 16:34.11 | infobot | gui is, like, (Graphical User Interface) Any system that uses graphics to represent the functions of a program. All Windows operating systems are GUI. or point-and-drool, double-click, drag-and-drop, such as http://toastytech.com/guis/miscwin1xp.png. Of course Real Programmers use the command line interface. See cli |
| 16:34.20 | *** join/#asterisk b0gatyr_ (~b0gatyr_@adsl-2-204-24.mia.bellsouth.net) |
| 16:34.27 | WIMPy | Sorry, no idea about yum. |
| 16:36.55 | vinhdizzo | pabelanger: thanks for sharing the link the asterisk + google voice link. going to read it now. |
| 16:37.58 | *** join/#asterisk quenenni (~quenenni@85.234.197.125) |
| 16:43.18 | *** join/#asterisk kami` (~user@unaffiliated/kami-) |
| 16:43.47 | *** join/#asterisk jblack (~jblack@71.181.209.104) |
| 16:43.50 | *** part/#asterisk nik_ (c4d28d92@gateway/web/freenode/ip.196.210.141.146) |
| 16:44.08 | kami` | Hi, I have problems with mISDN (hfcsusb). Is this the right place to ask questions? |
| 16:44.54 | WIMPy | ~ask |
| 16:44.54 | 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. |
| 16:46.15 | WIMPy | misdn1 or misdn2? |
| 16:46.24 | kami` | WIMPy: sorry. Wanted to know whether mISDN is on-topic, as the first sentence of your infobot suggests. |
| 16:47.09 | kami` | WIMPy: misdn2 |
| 16:47.37 | WIMPy | still waits for the question. |
| 16:48.11 | kami` | When hfcsusb is loaded, the message "HFC-S_USB: probe of 4-2:1.0 failed with error -5" is logged. |
| 16:49.07 | kami` | Increasing debug level doesn't give more info |
| 16:49.27 | WIMPy | dmesg? |
| 16:50.09 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 16:52.00 | kami` | WIMPy: http://paste.lisp.org/+2HEC |
| 16:54.28 | WIMPy | Do you have a patched version? My X-Tensions thing wasn't recognized. What's your USB ID? |
| 16:54.34 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 16:56.53 | quenenni | Hello. I'm trying to have the meetme app work but I have an error: "app_meetme.c:1097 build_conf: Unable to open pseudo device". A lsmod | grep zap shows me "zaptel 184068 5 ztdummy,wcusb,wctdm,wcfxo,wctdm24xxp" and this "crc_ccitt 2080 1 zaptel" but the cmd in the Cli shows me no zaptel module loaded (cmd: module show like zaptel). I can do a /etc/init.d/zaptel start and there is no error.. Any idea where to look? |
| 16:58.03 | kami` | WIMPy: in a German forum, a user reports that it works for him like a charm with this patch: http://www.ip-phone-forum.de/showthread.php?p=1611166 |
| 17:00.01 | WIMPy | Ok, so then it's the same as the one I've got, and yes, that should work. |
| 17:00.30 | WIMPy | But it looks like something is going wrong there. |
| 17:00.51 | ManxPower | quenenni, you have to rebuild Asterisk after you install zaptel or dahdi. you should be using dahdi, but that is not likely the issue |
| 17:01.47 | quenenni | ManxPower: ok.. I didn't rebuild Asterisk, so, I'm going to try that. Thanks |
| 17:06.59 | quenenni | Maybe someone has a link for how to rebuild Asterisk (1.6) with Zaptel integrated. I installed Asterisk thanks to apt-get from Debian and I'm not so good with the rebuild thingy :-[ |
| 17:08.55 | WIMPy | kami`: Where did you find revision 0.3.4? Linux 2.6.36 still has 0.3.3. |
| 17:10.06 | *** join/#asterisk timahvo1 (~rogue@41.223.57.76) |
| 17:13.46 | ManxPower | Asterisk 1.6 does NOT support Zaptel, it is DAHDI only |
| 17:14.37 | quenenni | oups.. yes.. true.. sorry.. |
| 17:15.24 | *** join/#asterisk jkroon (~jkroon@dsl-241-237-36.telkomadsl.co.za) |
| 17:16.55 | *** join/#asterisk visik7 (~Adium@unaffiliated/visik7) |
| 17:25.51 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 17:27.09 | *** join/#asterisk GameGamer43|Mac (~GameGamer@CPE-65-27-76-78.new.res.rr.com) |
| 17:44.48 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 17:49.29 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 17:51.16 | marksaitis | hey asterisk guys, does this strace ring any bells to u |
| 17:51.17 | marksaitis | read(0, 0x7fff6db41f4f, 1) = ? ERESTARTSYS (To be restarted) |
| 17:51.17 | marksaitis | --- SIGURG (Urgent I/O condition) @ 0 (0) --- |
| 17:51.17 | marksaitis | rt_sigreturn(0x17) = 0 |
| 17:55.58 | vinhdizzo | so i have asterisk connected to google voice via jabber.conf and gtalk.conf. Can anyone point me to a reference that shows me how to use a sip client to connect to the asterisk and use that google voice account to dial out? |
| 17:56.21 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 17:56.43 | vinhdizzo | I believe I have to create a "user" in SIP.conf, but how do I specify to use the google voice account? |
| 17:57.03 | drmessano | ~book |
| 17:57.03 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 17:58.30 | tzafrir_laptop | quenenni, there's no asterisk module called "zaptel". The zap one was called "chan_zap". The newer one is called chan_dahdi |
| 18:04.08 | quenenni | thanks tzafrir & ManxPower.. I knew it but I forgot when we upate to Astersik 1.6 few times ago. Dahdi seems well present and loaded. Now, my cmd in the Cli (module show like dah) shows me 6 loaded modules: app_dahdiras.so, codec_dahdi.so, app_dahdiscan.so, app_dahdibarge.so, chan_dahdi.so and res_timing_dahdi.so.. But I still have my problem when using meetme appl. When I start Astersik, I have this error in the log: "ERROR |
| 18:06.13 | quenenni | I don't understand what it means this error. I didn't find lots of infos on it. |
| 18:07.43 | *** join/#asterisk jblack (~jblack@71.181.209.104) |
| 18:08.31 | pabelanger | vinhdizzo: You can look in sip.conf.samples for some examples on how to register your SIP client to asterisk. Once registered you'll just make you should SIP client can access your dialplan where you defined: exten => _91NXXNXXXXXX,1,Dial(Gtalk/gmail/+${EXTEN:1}@voice.google.com) |
| 18:13.31 | *** join/#asterisk ariel_ (~chatzilla@c-98-254-138-7.hsd1.fl.comcast.net) |
| 18:14.27 | vinhdizzo | pabelanger: based on the example i've been googling, i think i see how to set up something in asterisk where a user on a sip client could log on to asterisk, but don't see where or how in the config does one set that user to point to the google voice. |
| 18:15.53 | pabelanger | vinhdizzo: It is handled by your dialplan (extensions.conf), not a .config file. |
| 18:15.59 | pabelanger | that is where the Dial command comes into play |
| 18:16.11 | vinhdizzo | pabelanger: |
| 18:16.12 | vinhdizzo | ok |
| 18:16.18 | pabelanger | So, exten => _91NXXNXXXXXX,1,Dial(Gtalk/gmail/+${EXTEN:1}@voice.google.com) |
| 18:16.35 | vinhdizzo | ahhh i see |
| 18:16.46 | vinhdizzo | then context= in sip.conf determines which dial plan |
| 18:16.46 | vinhdizzo | got it |
| 18:17.00 | pabelanger | any 91 + 10 digit number will be routed to google voice |
| 18:17.14 | pabelanger | vinhdizzo: Yes, |
| 18:17.31 | vinhdizzo | makes a lot more sense now |
| 18:17.40 | vinhdizzo | one last question |
| 18:18.08 | vinhdizzo | well not last, just a simpel one, what port does asterisk use by default for sip clients to connect to? |
| 18:18.11 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 18:18.15 | vinhdizzo | will need to request admin to open up that port |
| 18:19.41 | Jasnejac | vinhdizzo: one I can answer :D 5060 for SIP, 5061 for TLS if you are using it |
| 18:20.12 | vinhdizzo | Jasnejac: thanks. don't know tls |
| 18:21.09 | *** join/#asterisk [canniballllera] (~cannibale@201-35-197-39.fnsce703.dsl.brasiltelecom.net.br) |
| 18:25.00 | pabelanger | vinhdizzo: 5060/UDP |
| 18:25.45 | vinhdizzo | pabelanger: thanks. since ur here, i want to ask u another question about your post |
| 18:26.32 | vinhdizzo | I have in my extensions include => incoming under [default]. incoming does answer, play hello world, and hangup |
| 18:26.33 | pabelanger | shoot |
| 18:26.52 | vinhdizzo | however, when i call the gv #, gv picks up |
| 18:26.58 | vinhdizzo | and not asterisk |
| 18:27.19 | quenenni | we tried the meetme on another server and we have the same error: "WARNING[5672]: app_meetme.c:1097 build_conf: Unable to open pseudo device".. I don't know where to look.. everything seems good.. |
| 18:28.52 | pabelanger | vinhdizzo: Did you enable allowguest within gtalk.conf? What is the context you setup? Can you see the incoming call in the CLI? |
| 18:29.14 | pabelanger | We don't have access to inbound google voice numbers in Canada, so I actually have not been able to test it |
| 18:29.38 | vinhdizzo | i have allowguest=yes under [general] in gtalk.conf |
| 18:30.00 | vinhdizzo | let me pastebin u what i have |
| 18:30.46 | pabelanger | quenenni: you have DAHDI installed? |
| 18:31.10 | marksaitis | is there a version of asterisknow with asterisk 1.8, srtp and tls all in one ready to use? |
| 18:31.28 | marksaitis | Im tired of shit getting tls working on a normal distro |
| 18:31.38 | pabelanger | vinhdizzo: And does /dev/dahdi/pseudo exists? |
| 18:31.49 | pabelanger | vinhdizzo: Sorry, wrong person |
| 18:31.51 | quenenni | pabelanger: yep.. well, the "module show like dah" command in the Cli shows me 6 loaded modules |
| 18:32.00 | pabelanger | quenenni: does /dev/dahdi/pseudo exist? |
| 18:32.33 | pabelanger | marksaitis: I don't believe so, 1.8.0 was just release 2 days ago |
| 18:32.48 | vinhdizzo | pabelanger: http://pastebin.com/1VpiRGrt |
| 18:32.51 | quenenni | pabelanger: no.. no /dev/dahdi folder |
| 18:33.18 | quenenni | on both servers |
| 18:33.24 | pabelanger | quenenni: That is your issue, is DAHDI loaded? |
| 18:33.38 | quenenni | with lsmod? |
| 18:34.20 | pabelanger | quenenni: sudo /etc/init.d/dahdi start |
| 18:35.00 | pabelanger | vinhdizzo: looks good. Is you google voice number forward to your gmail.com account? |
| 18:35.13 | vinhdizzo | pabelanger: let me check |
| 18:35.57 | quenenni | hmm.. I think you pointed in the right direction.. /etc/init.d/start -> /etc/dahdi/system.conf not found. Nothing to do (in /etC/dahdi, I have 1 file: xpp.conf). |
| 18:35.58 | marksaitis | shit happens |
| 18:36.02 | marksaitis | complete shit |
| 18:36.04 | vinhdizzo | pabelanger: yes forward to google chat |
| 18:36.19 | titter | quenenni: did you run make config when installing dahdi? |
| 18:36.24 | marksaitis | im lost |
| 18:37.04 | marksaitis | where is the info bit which says whats in asterisknow? |
| 18:37.16 | quenenni | I'm going to do it again I think.. do I have to remove the zaptel module, or it doesn't matter? |
| 18:37.33 | *** join/#asterisk deonv (~adium@196.1.28.226) |
| 18:37.35 | titter | quenenni: if this is a 1.6.x install, remove zaptel |
| 18:37.38 | vinhdizzo | pabelanger: i called it and picked up in gmail |
| 18:37.50 | quenenni | titter: ok.. thanks |
| 18:37.51 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 18:38.12 | pabelanger | vinhdizzo: Have you tried calling your gmail.com address directly? Does asterisk pickup? |
| 18:38.30 | vinhdizzo | pabelanger: does it have anything to do with ports? my testing server is at school and most ports are closed. |
| 18:38.45 | vinhdizzo | pabelanger: how do i call gmail.com address directly? |
| 18:39.31 | pabelanger | vinhdizzo: PM it to me, I can call from my google account |
| 18:39.50 | pabelanger | also: *CLI> jabber show connected |
| 18:39.53 | pabelanger | to see if you are online |
| 18:41.01 | titter | quenenni: If dahdi installs successfully, and you have no hardware cards ... dahdi_dummy will load which is the timing source required for meetme(). If dahdi was improperly configured when installing 1.6, meetme might not have been included. I would suggest rebuilding Asterisk if you run into problems once dahdi is installed. |
| 18:41.34 | pabelanger | vinhdizzo: See anything on your CLI? |
| 18:42.15 | marksaitis | Is there anybody smart enough on this channel to tell me why my tls does not work? :) |
| 18:42.29 | *** join/#asterisk nicoAMG (~nicoAMG@201.237.49.131) |
| 18:43.17 | marksaitis | in asterisk18 |
| 18:44.16 | pabelanger | ~collectdebug |
| 18:44.16 | infobot | i heard collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to doc/HOWTO_collect_debug_information.txt or http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt |
| 18:44.20 | pabelanger | vinhdizzo: ^^ |
| 18:47.38 | marksaitis | pabelanger, thanx, I will give it a spin |
| 18:56.06 | kami` | WIMPy: ping |
| 18:56.21 | WIMPy | pong |
| 18:56.38 | kami` | sorry, had to bring my kids to bed. Took a bit longer. |
| 18:56.59 | kami` | I downloaded the current version from misdn.org. |
| 18:57.26 | WIMPy | There is a corrent version of misdn2 as well? |
| 18:57.40 | kami` | Wait. |
| 18:58.17 | kami` | http://misdn.org/index.php/MISDN_v2_Download |
| 18:58.21 | WIMPy | Since it's in the kernel, I only took the misdnuser from there. |
| 18:58.47 | quenenni | titter: excellent.. thanks for the tips.. I'm still in the process of removing completely the zaptel module.. I removed the source I used to build it (hehe).. |
| 18:59.05 | kami` | OK. I thought the kernel version might be too old and wanted to try the current version. |
| 18:59.36 | WIMPy | The standard kernel has been fine for me. |
| 19:01.42 | kami` | WIMPy: so you have an x-tension working? Or another usb adapter with HFC chip? |
| 19:01.55 | WIMPy | Both |
| 19:02.14 | WIMPy | Even at the same time :-) |
| 19:03.07 | kami` | Encouraging! When compiling mISDNuser, I get "error: AF_ISDN undefined and need to be set with --with-AF_ISDN=PROTOCOLNUMBER" |
| 19:03.19 | kami` | What should I set AF_ISDN to? |
| 19:04.08 | WIMPy | Looks like either a kernel header problem or a kernel without mISDN enabled. |
| 19:05.16 | *** join/#asterisk leif[mobile] (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:05.16 | *** mode/#asterisk [+o leif[mobile]] by ChanServ |
| 19:06.40 | marksaitis | pabelanger I did what u sugested |
| 19:06.40 | marksaitis | http://pastebin.com/wiq6iiQT |
| 19:07.34 | ChannelZ | http://www.youtube.com/watch?v=99FDDLWz15Y |
| 19:07.55 | marksaitis | nah, dont think thats much in that log |
| 19:08.04 | marksaitis | it doesnt even displau tls error |
| 19:08.18 | pabelanger | marksaitis: So, looks like an issue in tcptls.c. Not sure why though |
| 19:08.31 | *** join/#asterisk fofware (~Fabian@host176.190-31-223.telecom.net.ar) |
| 19:08.54 | marksaitis | oh, yeah I just found it too |
| 19:09.05 | marksaitis | it says file open failed |
| 19:09.10 | marksaitis | how d oI know what file ;] |
| 19:09.22 | marksaitis | now thats something to get going with |
| 19:09.26 | marksaitis | after half a day |
| 19:09.32 | *** join/#asterisk doolittlework (~d@41-134-22-10.dsl.mweb.co.za) |
| 19:10.18 | *** join/#asterisk riddlebox (~james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 19:10.22 | doolittlework | hi there is the a way to get the string lenght of a variale say $test=123456 must be 6? |
| 19:10.53 | WIMPy | core show function len |
| 19:10.53 | pabelanger | marksaitis: Maybe a missing cert of something, permission issue? |
| 19:10.59 | doolittlework | thx |
| 19:13.59 | marksaitis | pabelanger, well I made asterisk user to be an owner of the whole certs dir |
| 19:20.05 | marksaitis | WIMPy |
| 19:20.49 | WIMPy | here! |
| 19:20.57 | WIMPy | Wassup? |
| 19:20.59 | marksaitis | u said u have tls working, and if you use freepbx, can u tell me where did u put tlsenable=yes |
| 19:21.00 | marksaitis | ? |
| 19:21.01 | marksaitis | ;] |
| 19:21.23 | WIMPy | I'm not using any GUI. |
| 19:21.32 | marksaitis | I see |
| 19:21.50 | marksaitis | so u just sticked that little tls settings stack in sip.conf file? |
| 19:22.14 | WIMPy | yes |
| 19:22.38 | marksaitis | is it simmilar to the one in here? http://www.voip-info.org/wiki/view/SIP+TLS |
| 19:22.49 | marksaitis | at the end |
| 19:23.52 | WIMPy | Do you have tcp enabled? |
| 19:24.03 | marksaitis | tcp enabled? |
| 19:24.11 | marksaitis | not sure what u mean |
| 19:24.15 | WIMPy | Well, you wouldn't see much if you hadn't. |
| 19:24.40 | WIMPy | tcpenable=yes |
| 19:25.10 | doolittlework | WIMPy: what is wrong with this exten => _X.,1,Set(dst_len = ${LEN(${CDR(dst)})}) |
| 19:25.14 | marksaitis | so we need tcpenable=yes and tlsenable=yes |
| 19:25.16 | marksaitis | right? |
| 19:26.00 | WIMPy | yes |
| 19:29.26 | marksaitis | nah, doesnt make a difference |
| 19:29.31 | marksaitis | sadly enough |
| 19:29.36 | doolittlework | never WIMPy forgot to do reload |
| 19:29.49 | doolittlework | lol sleeping |
| 19:29.53 | marksaitis | from the logs I know that when I try to use tls, system is unable to open some kind of file, but it wont say which |
| 19:31.59 | kami` | WIMPy: just for the records, I managed to compile mISDNuser with './configure --with-AF_ISDN=34' (which is the definition from socket.h kernel header) |
| 19:33.06 | kami` | and it seems like the message 'failed with error -5' is just a warning |
| 19:33.37 | kami` | and misdn_info shows: Port 0 'HFC-S_USB.1': TE/NT-mode BRI S/T (for phone lines & phones) |
| 19:34.43 | kami` | Now, I'm a little bit confused about the config. http://misdn.org/index.php/Configuring_mISDN says, I need a /etc/mISDN.conf xml file. |
| 19:35.25 | *** join/#asterisk sshock (~sshock@63.248.133.83) |
| 19:35.40 | marksaitis | does anybody know any good softphone for windows? |
| 19:35.49 | kami` | I should be able to create a default one with 'mISDN config', but in mISDNuser, there is no mISDN. |
| 19:38.21 | ChannelZ | marksaitis: I like Zoiper Classic |
| 19:41.18 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 19:42.35 | WIMPy | kami`: You are right. I just tried it. Mine also throws that error. So it obviousely doesn't matter. |
| 19:42.51 | marksaitis | ChannelZ, it doesnt support tls |
| 19:44.14 | WIMPy | kami`: There is no config file any more. |
| 19:44.26 | ChannelZ | well not for free |
| 19:45.35 | ChannelZ | (and actually you'd need Communicator not Classic in that case too) |
| 19:47.56 | marksaitis | yeah |
| 19:48.03 | marksaitis | any free ones? |
| 19:48.13 | marksaitis | is sip market dead? I really cant find anything normal |
| 19:48.21 | marksaitis | just crap all around |
| 19:48.38 | *** join/#asterisk [canniballllera] (~cannibale@201-35-197-39.fnsce703.dsl.brasiltelecom.net.br) |
| 19:50.53 | p3nguin | Is there a simple way to recompile just one module? |
| 19:51.09 | ChannelZ | make |
| 19:51.31 | ChannelZ | unless you didn't keep your previous build |
| 19:52.15 | p3nguin | I'm in $src/$ver/res and I see a Makefile along with all the modules. I just want to rebuild one of them in the list. |
| 19:53.26 | *** part/#asterisk [canniballllera] (~cannibale@201-35-197-39.fnsce703.dsl.brasiltelecom.net.br) |
| 19:53.45 | *** join/#asterisk fofware (~Fabian@host124.200-117-29.telecom.net.ar) |
| 19:53.58 | leif[mobile] | just run make and it will only rebuild what has been changed since the last build |
| 19:56.41 | ChannelZ | remove the .o file of the module and run 'make' from the top level |
| 19:59.29 | ChannelZ | actually I'd guess you want to remove the .so as that's the target |
| 20:00.02 | ChannelZ | or really both so it recompiles :) |
| 20:05.56 | quenenni | pabelanger, titter: thanks a lot.. it works :) It was well a problem in the installation of dahdi.. |
| 20:06.43 | titter | quenenni: no problem |
| 20:09.47 | doolittlework | http://pastebin.com/sgU796K7 can someone please help me i am doing something wrong |
| 20:10.18 | doolittlework | call goes to exten => h,n(int),Hangup() |
| 20:10.40 | doolittlework | should go to exten => h,n(int),NoOp(********************WORKS**********************) |
| 20:13.55 | drmessano | Anyone using google voice with 1.8? |
| 20:14.29 | pabelanger | drmessano: yes |
| 20:15.20 | drmessano | pabelanger, I can't seem to make an outbound call, inbound are not coming in (nothing on the CLI), and I also noticed if I don't specify the account correctly in my Dial(), it segfault asterisk |
| 20:15.26 | drmessano | segfaults* |
| 20:15.48 | pabelanger | drmessano: check out http://blog.polybeacon.com/2010/10/17/asterisk-1-8-and-google-voice/ |
| 20:15.51 | drmessano | I tend to think these are all related |
| 20:16.15 | pabelanger | drmessano: Is this 1.8.0? |
| 20:16.20 | ChannelZ | doolittlework: Remove the (int) off n(int) for all but the first priority you're jumping to |
| 20:16.55 | drmessano | pabelanger, of course |
| 20:17.10 | ChannelZ | A priority label should only be on one priority for the extension; It's like a marker. You're jumping to your Hangup step because each time you say n(int) it 'overwrites' the previous marker. |
| 20:17.35 | pabelanger | drmessano: I suspect this is your problem: https://issues.asterisk.org/view.php?id=18149 |
| 20:18.02 | doolittlework | thank you ChannelZ you r a * |
| 20:18.18 | pabelanger | drmessano: Also, enable debugging and PB your log |
| 20:18.31 | drmessano | pabelanger, yeah, that looks like part of it |
| 20:18.39 | drmessano | The crashing bit anyway |
| 20:19.58 | pabelanger | drmessano: So, I was say you have a typo in your dialplan or your gtalk peer does not exist |
| 20:21.06 | drmessano | pabelanger, we're talking about two different issues.. The issue with the non-existant peer was something I noticed when I indeed had the incorrect peer and was segfaulting. I have the correct peer now, and I get nothing but ringing. No calls coming in either |
| 20:22.10 | *** join/#asterisk bjornts (~Adium@247.62-97-195.bkkb.no) |
| 20:22.12 | pabelanger | drmessano: pb your result |
| 20:25.37 | *** join/#asterisk Sheath (Smegma@unaffiliated/coil) |
| 20:25.48 | sshock | how can I debug why MOH is not working? |
| 20:26.26 | ChannelZ | Be more vague. |
| 20:26.54 | sshock | ChannelZ: I can't hear the music and there are no errors |
| 20:27.12 | sshock | turning on verbose shows "Started music on hold, class 'default', on SIP/tammy-081d1500" |
| 20:27.22 | drmessano | pabelanger, I resolved my outbound issue. Let me try inbound and see where I am going wrong here |
| 20:27.24 | sshock | so it should be playing but I can't hear it on the phone |
| 20:27.50 | ChannelZ | sshock: are you doing this by answering a call and placing it on hold, or calling MusicOnHold() ? |
| 20:28.20 | sshock | at the moment by just calling MusicOnHold() |
| 20:28.26 | drmessano | pabelanger, I noticed on your blog post you address nothing more than [guest] in the gtalk.conf. Why that and not specifying the connections themselves? |
| 20:28.32 | sshock | I set up an extension that goes straight to it |
| 20:28.44 | ChannelZ | Answer() the channel first |
| 20:28.51 | sshock | here's the conf (not much to it): http://pastebin.com/7S0xaz7s |
| 20:28.57 | sshock | oh; umm... |
| 20:29.15 | sshock | yeah, it has an Answer() first already |
| 20:29.55 | sshock | actually let me double check that... |
| 20:30.46 | sshock | oh my heck, I can't believe it |
| 20:30.47 | pabelanger | drmessano: Because I wanted to allow any incoming connection to my asterisk box. Defining a specific user will only allow them access |
| 20:31.10 | sshock | ok, so now back to my normal dialplan; why is it not working there? |
| 20:31.23 | sshock | hmmm, do I need an Answer() before calling Dial() from an internal context? |
| 20:31.34 | ChannelZ | no |
| 20:31.44 | ChannelZ | well I should say not necessarily |
| 20:31.45 | sshock | hmm... |
| 20:32.20 | sshock | can you expound? |
| 20:32.21 | drmessano | Hmmm |
| 20:32.23 | ChannelZ | What is the problem with your 'normal dialplan'? |
| 20:32.37 | *** join/#asterisk Sheath (Smegma@unaffiliated/coil) |
| 20:33.05 | sshock | when I call an extension that doesn't answer, I can't hear the MOH |
| 20:33.23 | ChannelZ | uhm... why would you? |
| 20:33.23 | sshock | if I take of the ",,m" from the Dial() then it does ring just fine |
| 20:33.40 | sshock | Dial(SIP/tammy,,m) |
| 20:33.54 | sshock | a call from internal to internal |
| 20:34.13 | ChannelZ | In that case yes I imagine you wouldn't hear it. |
| 20:34.22 | ChannelZ | without answering the channel first. |
| 20:34.35 | sshock | ok, so I do need Answer() before Dial() |
| 20:34.57 | ChannelZ | in this case yes |
| 20:35.01 | sshock | I can't believe I missed that, but it's weird you don't need it when using normal ring. |
| 20:35.06 | marksaitis | has anybody in the whole wolrd managed to get 1.8 asterisk with tls running? |
| 20:35.20 | ChannelZ | It's hard to explain |
| 20:35.34 | sshock | I think I can almost, kinda, guess |
| 20:35.56 | sshock | btw, I assume if Answer() has been done previously it won't hurt to do it again |
| 20:36.01 | ChannelZ | If you enter the dialplan and never answer the channel, Asterisk isn't sending/receiving any media from the device |
| 20:36.19 | pabelanger | sshock: You could also use Progress(), if you want early media |
| 20:36.24 | sshock | ok, so MOH requires media but ring doesn't? |
| 20:36.34 | sshock | hmm, Progress()... |
| 20:36.49 | ChannelZ | Asterisk isn't usually generating the ringing sound.. the device does |
| 20:36.53 | pabelanger | sshock: MOH, yes. |
| 20:37.15 | pabelanger | unless you are providing ringback from Asterisk, there is not RTP involved. |
| 20:37.15 | ChannelZ | Asterisk tells the device the call is 'making progress' and the device makes a ringing sound. Once the other end picks up the call bridges and the media streams are setup |
| 20:37.16 | sshock | ok, now that explains it well |
| 20:37.33 | pabelanger | EG: Ringing() |
| 20:38.10 | pabelanger | marksaitis: Likely not. It's only been out for 3 days. |
| 20:38.17 | pabelanger | on my TODO list |
| 20:39.26 | sshock | it's working great now, thanks! |
| 20:39.35 | marksaitis | SOMEBODY KILL ME# |
| 20:39.45 | marksaitis | I cant get this bastard to work for me |
| 20:39.47 | drmessano | ~kill marksaitis |
| 20:39.47 | infobot | ACTION shoots a charged anti-positrino gun at marksaitis |
| 20:39.53 | marksaitis | yeeeh |
| 20:39.54 | marksaitis | ;] |
| 20:45.18 | marksaitis | I just tried bria with tls on windows, the error is different on asterisk cli: |
| 20:45.19 | marksaitis | <PROTECTED> |
| 20:46.10 | vinhdizzo | in extensions.conf, if I have in [incoming], exten => 2,1,Goto(tnttsp-forward,9,1), does pressing 2 take me into the tnttsp-forward context? That is, if I have 1 defined in both contexts, pressing 1 after 2 will yield the extension 1 defined in tnttsp-forward? |
| 20:50.29 | ChannelZ | yes.. to extension 9 priority 1 though in your example |
| 20:59.16 | *** join/#asterisk xzenon (~xzenon@173-165-132-189-utah.hfc.comcastbusiness.net) |
| 21:00.14 | xzenon | Can someone direct me to the documentation for using google voice with asterisk 1.8? |
| 21:04.43 | titter | marksaitis: I just tried this on my test box and see similar errors, trying to get my Polycom registered now |
| 21:05.45 | marksaitis | titter really |
| 21:06.04 | marksaitis | what softphone? or just polycom? |
| 21:06.07 | titter | SSL certificate ok |
| 21:06.07 | titter | <PROTECTED> |
| 21:06.07 | titter | [Oct 24 17:01:19] WARNING[3601]: tcptls.c:218 handle_tcptls_connection: FILE * open failed! |
| 21:06.24 | marksaitis | titter, yeah, same shit |
| 21:06.42 | marksaitis | but I didnt get SSL certificate ok anywhere |
| 21:07.03 | *** join/#asterisk ukine (~ukine@14-145.97-97.tampabay.res.rr.com) |
| 21:07.48 | marksaitis | I made it to give a different error by using sofphone on windows machine. My sip client from phone gave me the same error ur having! |
| 21:08.26 | titter | marksaitis: I never used tls before, so I am working on figuring out what needs to be set on the Polycom side |
| 21:10.26 | vinhdizzo | ChannelZ: I know it takes me to extension 9 in my example, but after this is pressed, do i stay in the tnttsp-forward context, or still in the incoming context |
| 21:10.50 | *** join/#asterisk ferdna (~yup@cpe-24-92-114-97.elp.res.rr.com) |
| 21:13.07 | marksaitis | titter, what do u think about that - http://support.xten.net/viewtopic.php?f=1&t=14004 |
| 21:13.17 | marksaitis | regarding the fields of certificates |
| 21:18.38 | *** join/#asterisk Husky (Smegma@unaffiliated/coil) |
| 21:23.25 | marksaitis | titter, any luck? |
| 21:30.50 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:31.06 | titter | marksaitis: trying again, stepped away for a bit |
| 21:35.39 | marksaitis | ;] |
| 21:35.46 | marksaitis | im trying again and again |
| 21:35.48 | marksaitis | as well |
| 21:39.57 | marksaitis | fucking son of a bitch, doesnt work whatever u do. I tried setting CN to local ip or to FQDN, as sugested bu guides, still the same! |
| 21:41.37 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
| 21:43.03 | ChannelZ | vinhdizzo: yes |
| 21:43.17 | ChannelZ | vinhdizzo: you stay in the new context |
| 21:43.28 | vinhdizzo | ChannelZ: thanks |
| 21:45.21 | titter | Polycom requires that the IP or Hostname you are using match the certificate ... I can add the DNS, but I am trying one other thing first |
| 21:45.41 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:45.42 | marksaitis | titter, I know, according to counterpath, they require it too |
| 21:46.02 | marksaitis | I have tried local IP and FQDN, no luck |
| 21:52.53 | ChannelZ | Are you really saying anything that important that needs to be encrypted? |
| 21:53.50 | marksaitis | sure |
| 21:59.51 | *** join/#asterisk sshock (~sshock@63.248.133.83) |
| 22:06.29 | *** join/#asterisk marksaitis (~Mk@78-61-148-80.static.zebra.lt) |
| 22:06.34 | marksaitis | titter, any luck? |
| 22:06.38 | titter | :tlsv1 alert unknown ca |
| 22:06.49 | marksaitis | yes, thats the same I get |
| 22:07.09 | marksaitis | I have just tried recreating selfsigned cert with FQDN in CN field |
| 22:07.14 | marksaitis | restarted asterisk |
| 22:07.23 | marksaitis | disconnected from my local internet |
| 22:07.28 | marksaitis | and connected using mobile one |
| 22:07.38 | marksaitis | and pointed sip-tls client to that FQDN |
| 22:07.51 | marksaitis | it still talks the same shit :tlsv1 aler unknown ca |
| 22:08.33 | marksaitis | I suppose its client saying unknown ca isnt it? |
| 22:08.40 | WIMPy | So that looks like a problem with the certificate chain, I'd guess. |
| 22:09.11 | marksaitis | titter, do u have CA flag set? |
| 22:09.41 | marksaitis | WIMPy, well it looks like a problem with cert yeah..... but whoo knows what problem as I tried so many different choices |
| 22:10.18 | titter | marksaitis: nope, it's an issue with chain ... certs are something I don't familiarize myself with too much |
| 22:10.34 | WIMPy | Isn't CA the pointer upwards? |
| 22:11.07 | ChannelZ | Sounds like the same reason your browser yells when you go to an internal site with a self-signed cert. |
| 22:11.22 | WIMPy | So it's what I envisaged many hours ago, that it can't verify the cert. |
| 22:11.29 | marksaitis | I dont know |
| 22:11.38 | WIMPy | Indeed. |
| 22:12.19 | marksaitis | how to se CA flag? |
| 22:12.23 | ChannelZ | The browser can't authenticate who signed it. So you need to be your own root authority or to be able to otherwise tell the client "yeah yeah, just accept this certificate anyway" |
| 22:15.35 | marksaitis | I think I will need to try a free star certificate with real CA |
| 22:15.40 | marksaitis | mabye that will help |
| 22:16.07 | WIMPy | Take a look at what root CAs your phone knows. |
| 22:16.24 | WIMPy | But there must be a possibility to add others. |
| 22:16.41 | marksaitis | who knows what it knows ;] |
| 22:16.44 | marksaitis | its iphone 4 |
| 22:18.35 | marksaitis | I have bria sip client on win too |
| 22:18.59 | marksaitis | I suppose these things must know as much as normal IE |
| 22:19.56 | marksaitis | titter, u been followin this one? http://www.voip-info.org/wiki/view/SIP+TLS |
| 22:20.01 | marksaitis | what ur up to? |
| 22:23.41 | marksaitis | titter, I noticed another strange thing. Unknown ca is only an alert isnt it, I suppose this whole thing should work with an alert, its not an error |
| 22:24.20 | marksaitis | I heard that polycom phones log some kind of alert but they do actually work |
| 22:24.40 | jdoe | marksaitis: if you're trying to do sip/tls with bria on an iphone you need to tell the phone to trust the cert. |
| 22:25.04 | jdoe | marksaitis: easiest way is to email yourself the appropriate key (or I think you can get it from safari too) |
| 22:25.29 | marksaitis | jdoe, what if I have another bria client on windows |
| 22:25.33 | marksaitis | what do I do then? |
| 22:25.44 | jdoe | I assume you need to tell it to trust the cert too. |
| 22:25.50 | jdoe | I dunno how that works. Through IE maybe? |
| 22:25.52 | marksaitis | hmzzzz |
| 22:25.57 | marksaitis | lets give it a spin |
| 22:25.58 | jdoe | gotta run. |
| 22:26.36 | pabelanger | marksaitis: titter: Have you checked out http://svn.digium.com/svn/asterisk/branches/1.8/contrib/scripts/ast_tls_cert |
| 22:27.11 | marksaitis | no |
| 22:27.15 | marksaitis | what the hell is dat |
| 22:27.53 | titter | pabelanger: that looks good, but I still need to install my CA on the client (Polycom) ... so I am looking up that now. |
| 22:28.05 | titter | pabelanger: since my CA is self signed |
| 22:28.24 | pabelanger | A script recently added to source. I haven't tried it, but know it is related to TLS |
| 22:28.28 | pabelanger | and certs |
| 22:28.47 | ChannelZ | https://support.quovadisglobal.com/KB/a64/how-do-i-install-a-digital-certificate-onto-an-iphone.aspx |
| 22:28.47 | marksaitis | il give it a spin |
| 22:30.35 | titter | I found how to do it on the Polycom |
| 22:30.48 | titter | I am going to regen my certs with that script |
| 22:31.26 | doolittlework | can one do a gotoif with multiple instances must match like say ${EXTEN}=1000 gotoif($[${EXTEN:-4:1} = 1] and $[${dst_len} = 4]?int) |
| 22:32.00 | doolittlework | sorry like can one do a gotoif with multiple instances must match like say ${EXTEN}=1000 gotoif($[${EXTEN:-4:1} = 1] and $[${EXTEN} = 4]?int) |
| 22:34.08 | marksaitis | lol - 1.Open the .p12 or .pfx certificate file using your iPhone. ....... epic fail |
| 22:34.20 | marksaitis | how the hell does one open a file on iphone ;]]]] |
| 22:34.57 | ChannelZ | Email it to yourself? |
| 22:35.14 | ChannelZ | I don't use one and never will but that's just what i found Googling for 3 seconds |
| 22:36.15 | ChannelZ | doolittlework: see the 'expressions' section of the asterisk reference guide |
| 22:36.37 | *** join/#asterisk marksaitis (~Mk@78-61-148-80.static.zebra.lt) |
| 22:36.38 | marksaitis | ;] |
| 22:36.55 | ChannelZ | see asterisk.pdf in doc/tex |
| 22:37.11 | marksaitis | did I miss anythin important |
| 22:38.31 | doolittlework | thanks again |
| 22:38.43 | marksaitis | wow |
| 22:38.47 | marksaitis | I think it works now |
| 22:39.42 | marksaitis | ouh yes |
| 22:39.45 | marksaitis | it really does |
| 22:40.03 | WIMPy | \o/ |
| 22:40.26 | titter | [Oct 24 18:39:22] NOTICE[4309]: chan_sip.c:19523 handle_response_peerpoke: Peer '200' is now Reachable. (25ms / 2000ms) |
| 22:40.29 | titter | yay |
| 22:40.47 | titter | pabelanger: thanks for the script |
| 22:41.02 | marksaitis | it worked for me even without a script :) |
| 22:41.15 | marksaitis | titter, have u installed that cert on polycom? |
| 22:41.33 | titter | marksaitis: yes I had to install the ca to the Polycom and set the Polycom to use All Certs |
| 22:41.56 | marksaitis | cert install helped me out too! |
| 22:42.06 | marksaitis | however, on my phone only, not on windows machine ;] |
| 22:42.18 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 22:42.25 | marksaitis | but that is still progress =) |
| 22:42.48 | titter | I could do it on my Evo if I wanted ... just don't like using the SIP client on my cell phone, gets annoying |
| 22:43.03 | titter | So I just setup a queue to dial it on incoming numbers to my extension |
| 22:43.21 | titter | Intrastate calls are free with my SIP trunk provider |
| 22:43.40 | marksaitis | good for u in usa |
| 22:43.40 | marksaitis | ;] |
| 22:44.22 | marksaitis | is there a test number I could call on fresh asterisk and freepbx install? |
| 22:44.37 | titter | create one in extensions.conf |
| 22:45.12 | ChannelZ | freepbx.. vomit |
| 22:46.08 | marksaitis | ChanServ? |
| 22:46.20 | marksaitis | dunno how to create a test one ;] |
| 22:47.26 | titter | exten => 100,1,Playback(tt-weasels) |
| 22:47.29 | *** join/#asterisk [canniballllera] (~cannibale@201-35-197-39.fnsce703.dsl.brasiltelecom.net.br) |
| 22:48.24 | titter | place it in whatever context your sip user is going to |
| 22:48.29 | titter | then dial it |
| 22:49.05 | marksaitis | brrr, im just thinking how to do when I have freepbx |
| 22:49.38 | ChannelZ | I thought the GUI was supposed to make things easier? Huh. |
| 22:49.38 | titter | no clue |
| 22:49.50 | ChannelZ | Seems a bit ironic. |
| 22:49.50 | titter | I would imagine the GUI has a spot for extensions |
| 22:49.52 | marksaitis | ChanServ, yeah :) |
| 22:50.12 | marksaitis | titter, do I put that somewhere in users context? |
| 22:50.17 | ChannelZ | <-- not ChanServ |
| 22:50.19 | marksaitis | or just anywhere in lets say sip file |
| 22:50.26 | titter | ChannelZ: It could be worse ... he could be combining Asterisk and Shitel ... I mean Shoretel |
| 22:50.35 | marksaitis | :D |
| 22:51.41 | titter | marksaitis: In your sip user add context=whatever, then in extensions.conf create [whatever] and place the code in the context. |
| 22:52.37 | *** join/#asterisk Juggie (~Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 22:53.12 | titter | marksaitis: this is assuming you are using Asterisk not some silly gui |
| 22:53.47 | marksaitis | nah, il be fine with gui afterwards, it will save me some time |
| 22:53.54 | marksaitis | in most cases |
| 22:53.55 | titter | Now back to my regular scheduled program of trying to get my iax to failover with dns srv |
| 22:54.06 | titter | Don't see how |
| 22:54.35 | titter | Freepbx takes control of your dialplan and does who knows what with it |
| 22:55.53 | ChannelZ | screws it up, that's what |
| 22:56.19 | marksaitis | well, we still have custom.conf files in there ;] |
| 22:57.15 | marksaitis | I added ur code, called 100 - couldnt hear a thing |
| 22:57.23 | titter | What did the CLI say |
| 22:58.20 | marksaitis | <PROTECTED> |
| 22:58.20 | marksaitis | <PROTECTED> |
| 22:58.20 | marksaitis | <PROTECTED> |
| 22:58.20 | marksaitis | <PROTECTED> |
| 22:58.20 | marksaitis | <PROTECTED> |
| 22:59.38 | *** join/#asterisk Juggie (~Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 23:00.09 | titter | It worked, looks like there is another issue at play |
| 23:00.14 | marksaitis | but after that it says its playing silence |
| 23:00.15 | *** join/#asterisk Juggie (~Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 23:00.43 | titter | Does your SIP client support gsm? |
| 23:00.50 | marksaitis | <PROTECTED> |
| 23:00.50 | marksaitis | <PROTECTED> |
| 23:00.50 | marksaitis | <PROTECTED> |
| 23:00.50 | marksaitis | <PROTECTED> |
| 23:00.50 | marksaitis | <PROTECTED> |
| 23:00.51 | marksaitis | <PROTECTED> |
| 23:01.17 | marksaitis | well it says media active g711u |
| 23:01.21 | WIMPy | marksaitis: Stop junping into my bin. |
| 23:01.22 | marksaitis | call media encrypted |
| 23:01.30 | marksaitis | WIMPy? |
| 23:01.39 | ChannelZ | now you probably have port/firewall problems |
| 23:01.45 | titter | Yup |
| 23:01.47 | marksaitis | ouh, yeah |
| 23:01.51 | marksaitis | ChanServ! |
| 23:01.51 | titter | Looks like a NAT issue |
| 23:01.52 | marksaitis | ;] |
| 23:01.53 | WIMPy | read "don't flood" |
| 23:02.03 | marksaitis | what is that huge port range I need to open? |
| 23:02.07 | titter | RTP by default is 10k-20k |
| 23:02.32 | titter | UDP |
| 23:02.39 | marksaitis | WIMPy, whats the difference? its 2010 now isnt it ;] everybody here probably has fiberoptic internet... not slow modem anymore |
| 23:03.12 | WIMPy | It's not about bandwidth it's about spam. |
| 23:03.19 | marksaitis | is it 10-20 exactly? |
| 23:03.29 | titter | look at rtp.conf |
| 23:03.35 | titter | default is 10000-20000 |
| 23:03.38 | marksaitis | WIMPy, well, it wasnt spam ;] |
| 23:03.39 | ChannelZ | and note that's only one-way.. incoming audio |
| 23:03.58 | titter | I use Echo() to test two way audio personally |
| 23:04.05 | WIMPy | My client thought otherwise. |
| 23:04.34 | marksaitis | yes |
| 23:04.35 | marksaitis | it works now |
| 23:04.41 | marksaitis | after that range is open |
| 23:04.45 | marksaitis | works well |
| 23:04.46 | titter | It should |
| 23:04.48 | *** join/#asterisk mr_ian (~mr_ian@S0106001b63f49383.du.shawcable.net) |
| 23:05.18 | titter | now add exten => 100,n,Echo() after the first line |
| 23:05.35 | titter | It will play back the file, then enter echo which should echo everything you say |
| 23:05.42 | *** join/#asterisk marksaitis (~Mk@78-61-148-80.static.zebra.lt) |
| 23:05.58 | marksaitis | what the hell is wrong with me ;] I keep disconnecting |
| 23:06.17 | titter | You can get more info on echo by changing playback to Playback(demo-echotest) |
| 23:06.42 | marksaitis | will try ;] |
| 23:06.54 | marksaitis | I think im happy today |
| 23:08.04 | marksaitis | smashing |
| 23:08.25 | titter | Also just remember you are using tls, but rtp is still not secure |
| 23:08.34 | titter | Unless you are using srtp |
| 23:08.41 | marksaitis | im using srtp as well |
| 23:08.46 | ChannelZ | I thought that was the point of this whole exercise |
| 23:09.04 | marksaitis | ur right ChanServ |
| 23:09.09 | ChannelZ | so no one could hear his phone sexings |
| 23:09.20 | marksaitis | :D |
| 23:09.46 | marksaitis | im just wondering where does it put recorded calls now |
| 23:20.25 | marksaitis | where would be the best place to keep recorded asterisk messages? |
| 23:20.36 | marksaitis | and where does aterisk puts voicemail does anybody know? |
| 23:21.11 | [TK]D-Fender | marksaitis: asterisk.conf <-------- |
| 23:29.20 | drmessano | Isn't all this in the newb.conf file? |
| 23:29.28 | marksaitis | ;] |
| 23:30.28 | titter | marksaitis: did you get srtp built on centos 5.5? |
| 23:30.37 | marksaitis | yeah |
| 23:30.45 | marksaitis | u? |
| 23:31.21 | marksaitis | I know it wanted a flag somewhere |
| 23:31.48 | titter | I didn't try, what flag did you set |
| 23:32.25 | marksaitis | I cant remember. Asterisk configure will tell you what flag it wants. it will also offer u to put preffix, but dont put that one |
| 23:34.09 | titter | ./configure CFLAGS=-fPIC |
| 23:34.48 | marksaitis | yeah |
| 23:37.33 | marksaitis | im just wondering if my call recording works |
| 23:37.36 | titter | worked, but still not selectable in make menuselect |
| 23:37.54 | marksaitis | have u tried make install? |
| 23:38.00 | marksaitis | autoconf ? |
| 23:38.09 | titter | for srtp? |
| 23:38.13 | marksaitis | yeah |
| 23:38.39 | marksaitis | its strange that in manual it doesnt say make install, but its a must do |
| 23:38.53 | titter | just did it, trying now |
| 23:39.33 | *** join/#asterisk mifadir (~mifadir@dynamic.casap1-164-30-137-41.wanamaroc.com) |
| 23:40.00 | mifadir | i can't found sip firmware for my cpva503w |
| 23:40.16 | mifadir | any one know any info |
| 23:40.17 | mifadir | ?? |
| 23:40.38 | florz | firmware doesn't need a foundation |
| 23:40.52 | mifadir | this one yes |
| 23:40.57 | mifadir | i can't found it |
| 23:41.06 | florz | yeah, and you don't need to |
| 23:41.10 | mifadir | there"s three version sip/h323/mgcp |
| 23:41.18 | mifadir | i can't find sip one |
| 23:41.19 | florz | it will stand upright without a foundation |
| 23:41.37 | mifadir | what you mean |
| 23:41.42 | mifadir | i don't understand |
| 23:41.54 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 23:42.28 | mifadir | Hello |
| 23:42.34 | mifadir | i don't understand |
| 23:43.30 | florz | see, "to found" means "to create a foundation", which is quite different from "to find" which means "to locate" |
| 23:44.45 | mifadir | thanks for this cours |
| 23:45.13 | mifadir | find is the right word |
| 23:45.24 | jamko | evening... Could someone help me figure out what I am doing wrong with this dialplan logic: http://pastebin.com/ER6VpCZU |
| 23:45.28 | drmessano | mifadir, your cpva503w is a gateway device that a provider would likely need to change the firmware on |
| 23:45.38 | drmessano | mifadir, contact your ISP |
| 23:45.48 | mifadir | blah, blah |
| 23:45.55 | drmessano | blah blah? |
| 23:46.31 | mifadir | it work fine with h323 |
| 23:46.44 | *** part/#asterisk quenenni (~quenenni@85.234.197.125) |
| 23:46.47 | mifadir | who need help can't help |
| 23:47.04 | *** part/#asterisk mifadir (~mifadir@dynamic.casap1-164-30-137-41.wanamaroc.com) |
| 23:47.24 | drmessano | Idiot.. and here I checked the manufacturer website for him and everything |
| 23:47.31 | drmessano | No "Thanks for the google proxy" |
| 23:47.33 | drmessano | Nada |
| 23:48.27 | Chainsaw | thinks drmessano needs a cookie |
| 23:49.02 | WIMPy | He probably already got one on that website. |
| 23:49.26 | jamko | http://pastebin.com/ER6VpCZU I want asterisk to go to ?donedial if dialstatus returns ANSWER ... If dialstatus returns anything other than ANSWER, I want asterisk to go to ?dialnext .. However, it is going to ?donedial every time. What am I doing wrong?? |
| 23:50.12 | Chainsaw | jamko: Missing a terminating " on the first GotoIf. |
| 23:50.16 | Chainsaw | jamko: Compare the two. |
| 23:50.57 | [TK]D-Fender | ChaActually, he doesn't have a STARTING one on the proper side and added one at a place that broke this even being anexpression at all |
| 23:51.25 | jamko | damn it all... |
| 23:52.27 | [TK]D-Fender | exten => _NXXXXXXXXX,n,Dial(SIP/1${EXTEN}@my_service_provider,120,T,t,K,k) <-- WTF is with the extra half-dozen ","? |
| 23:52.38 | jamko | TK: Would you mind putting them in the right spot, and submitting the correction? |
| 23:52.39 | *** join/#asterisk Juggie (~Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 23:52.48 | [TK]D-Fender | jamOption is single PILE of letters, not each separated by commas |
| 23:53.03 | jamko | oh |
| 23:53.09 | Chainsaw | hands Fender a new tab key |
| 23:53.12 | jamko | that would explain certain dtmf issue. |
| 23:55.16 | [TK]D-Fender | Chainsaw: tab works fine.... it wasn't a good enough match so it didn't. |
