| 00:01.11 | *** part/#asterisk swytch (n=ezcall@d83-179-214-255.cust.tele2.fr) |
| 00:08.05 | robin_sz | word |
| 00:08.13 | ManxPower | AMP/FreePBX/Asterisk@Home users should join #freepbx for support |
| 00:09.09 | ManxPower | <PROTECTED> |
| 00:09.40 | drfoomod2 | no vin, nano |
| 00:09.43 | drfoomod2 | vim |
| 00:09.47 | Strom_C | AMP/FreePBX/Asterisk@Home users should join #what-do-you-mean-I-have-to-use-the-keyboard for support |
| 00:09.57 | drfoomod2 | Strom_C: you don;t want it hurt _that_ bad |
| 00:10.05 | ManxPower | ~thebook |
| 00:10.06 | jbot | [thebook] Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 00:10.25 | Strom_C | drfoomod2: ? |
| 00:10.29 | orlok | the thing is |
| 00:10.39 | drfoomod2 | [19:56] <Strom_C> just you, asterisk, and vim |
| 00:11.03 | Strom_C | drfoomod2: hey, thats how I work with asterisk |
| 00:11.03 | orlok | most admins who manage any decent amount pof systems eventually see the good things about automated scripts and smart templating systems rather than making every single change by hand |
| 00:11.25 | robin_sz | the basic problem is that theres a difference between "I ahve aproblem with asterisk" and "I have a problem with the configuration written automagically, by a GUI I dont understand, and I dont understand the configuration either" |
| 00:11.41 | robin_sz | well yes |
| 00:11.43 | robin_sz | but ... |
| 00:11.56 | Strom_C | orlok: sure, but thats why you write custom scripts suited to your exact need |
| 00:11.59 | robin_sz | when you need support, go see the author of the templating system |
| 00:12.20 | orlok | robin_sz: well, the specific templates that manage asterisk |
| 00:12.31 | orlok | as the same templates also handle apache, squid, qmail, dhcp, etc etc |
| 00:12.44 | robin_sz | not so ... |
| 00:12.53 | orlok | it makes it very easy to collect and manage customisations, push them out to lots of servers, etc |
| 00:12.56 | surfdue | i know this isnt the room but i use freepbx, does anyone know off hand why freepbx could not be writing to the configs with no errors? |
| 00:13.03 | orlok | heh |
| 00:13.19 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 00:15.04 | robin_sz | orlok, the problem is that 99% of the problems sre not from asterisk per-se, but from the configuration written by freepbx, a@h etc, we dont know how they manage their configs, the users dont either, the only people they can go to for support is #freepbx or #amp or whatever |
| 00:15.34 | robin_sz | like the guy just then |
| 00:15.43 | orlok | robin_sz: yeah, it would be like trying to diagnose a bug in java from the output of a coldfusion developer |
| 00:16.00 | robin_sz | right |
| 00:16.01 | orlok | fix an engine by looking at the dash lights, etc |
| 00:16.18 | orlok | the bonnet must be open! :) |
| 00:16.44 | robin_sz | worse than that, fix an engine, based on a description of the dash lights from someone who doesnt understand them |
| 00:17.24 | jql | hmm... linux must be easy to install these days with the abundance of gui-addicts using it. I'm heartened |
| 00:17.31 | orlok | "the watering can is red!" |
| 00:17.57 | orlok | jql: yup |
| 00:18.05 | orlok | jql: www.contribs.org |
| 00:18.08 | robin_sz | but ... they need to go see their integrator for support |
| 00:18.56 | robin_sz | much the same way as you wouldnt get the ford engine design team being hassled by old granny smithers about her non-working fiesta |
| 00:19.21 | robin_sz | s/fiesta/some other flavour of ford/ |
| 00:19.27 | wunderkin | orlok, your packet was out of order |
| 00:19.35 | orlok | heh |
| 00:21.35 | orlok | hmm. 41 meg of bzip2'd .sql files |
| 00:21.35 | *** join/#asterisk niZon (n=ilt@S010600080db4ab60.wp.shawcable.net) |
| 00:23.39 | *** join/#asterisk RES2 (n=RES@chello213047231029.tirol.surfer.at) |
| 00:25.25 | tainted- | <Strom_C> orlok: you should try to set up a system without any guis or prototyping tools or anything |
| 00:25.27 | tainted- | idiot |
| 00:26.29 | Strom_C | hey, look, it's the troll |
| 00:26.56 | tainted- | nothing wrong with guis |
| 00:27.07 | tainted- | depends on the needs of the end user |
| 00:27.23 | RES2 | Hi. |
| 00:27.23 | Strom_C | tainted-: he was complaining about being unfamiliar with asterisk, so I suggested that as an exercise to increase familiarity |
| 00:27.23 | tainted- | just empower ppl and get the fuck out of the way |
| 00:27.26 | Strom_C | ifiot |
| 00:27.31 | Strom_C | also, typinh |
| 00:27.34 | RES2 | Can anyone help me with spandsp please? |
| 00:27.36 | Strom_C | blah |
| 00:28.23 | Strom_C | I cut my fingers off |
| 00:28.27 | file | that's unfortunate |
| 00:28.27 | Strom_C | so I can't poke you really |
| 00:28.36 | robin_sz | end users should always be dealt with by the highest level of abstraction |
| 00:28.38 | Strom_C | i can kind of jam my elbow into you, though |
| 00:28.56 | websae | such pleasant thoughts! |
| 00:29.20 | RES2 | "loader.c: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: t30_completion_code_to_str" |
| 00:29.22 | robin_sz | not as good as the book |
| 00:33.04 | *** join/#asterisk mr-russ (n=admin@CPE-60-224-135-193.vic.bigpond.net.au) |
| 00:33.15 | tainted- | robin_sz it was a book? |
| 00:34.15 | *** join/#asterisk Johnnie (n=jdlewis@dynamic-acs-24-154-91-195.zoominternet.net) |
| 00:34.17 | robin_sz | tainted-, yes, a very famous book |
| 00:34.32 | robin_sz | tainted-, won the boardman/tasker prize when it cam out |
| 00:35.24 | robin_sz | tainted-, yes, a very famous book |
| 00:35.26 | robin_sz | tainted-, won the boardman/tasker prize when it cam out |
| 00:38.35 | orlok | Shuld the asterisk registrar also be a peer? |
| 00:44.10 | *** part/#asterisk sandos (n=sandos@me-0-50-da-e0-bb-36.2.cust.bredband2.com) |
| 00:44.55 | *** join/#asterisk TUplink (n=Tommy@68-232-82-147.chvlva.adelphia.net) |
| 00:45.25 | *** join/#asterisk demigod2k (n=joey@cpe-24-210-97-162.twmi.res.rr.com) |
| 00:45.30 | TUplink | i have 2 SIP lines to the PSTN... in my dialplan how can i make it use one or the other... if one is bussy |
| 00:48.51 | orlok | May 1 10:48:25 NOTICE[3945]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 00:49.49 | tainted- | orlok is the SIP client registered? |
| 00:50.19 | tainted- | robin_sz the movie's ending was too rushed |
| 00:51.18 | robin_sz | tainted-, read the book, its totally gripping, the movie left me cold, the people who made it were non climbers really, it meant little to me |
| 00:52.16 | robin_sz | its one of those books that you end up reading cover to cover in one sitting |
| 00:52.58 | orlok | tainted-: sip show registry seems correct |
| 00:53.00 | drfoomod2 | is there any xml/soap shim for the manager api? |
| 00:53.25 | demigod2k | which book? |
| 00:53.25 | orlok | tainted-: should the provider i am registered with show as a sip peer as well? |
| 00:55.31 | tainted- | robin_sz i'll have to check it out |
| 00:55.43 | tainted- | orlok are they your origination/termination provider? |
| 00:55.58 | tainted- | orlok can u pastebin your sip.conf w/o the username/password |
| 00:56.11 | orlok | tainted-: yup they are |
| 00:56.39 | orlok | wtf, i'm getting a 400: Bad Request in response to a NOTIFY sip: |
| 00:56.59 | *** join/#asterisk techman97_andy (n=me@70-98-31-249.dsl1.rsm.mn.frontiernet.net) |
| 00:57.23 | *** join/#asterisk PMantis (n=pmantis@cpe-66-66-115-197.rochester.res.rr.com) |
| 00:57.46 | techman97_andy | hey all - I need to get off of my current VoIP provider, VoiceEclipse, like yesterday. Their reliability is shoddy at best. Any good suggestions for a SIP provider? |
| 00:58.04 | PMantis | techman97_andy, Location ? |
| 00:58.13 | techman97_andy | (there should be no reason that I have to have people calling my number wait 15 seconds before it is delivered to my * box) |
| 00:58.18 | techman97_andy | MN - USA |
| 00:58.57 | orlok | tainted-: http://pastebin.ca/52463 |
| 00:59.58 | PMantis | techman97_andy, For DIDs, you can look at: connect.voicepulse.com, BroadVoice, ViaTalk |
| 01:00.11 | tainted- | drfoomod2 what platform are u using? |
| 01:00.13 | PMantis | techman97_andy, BV and VT also offer good OB dialing plans |
| 01:00.33 | tainted- | broadvoice eats babies |
| 01:00.37 | techman97_andy | wow |
| 01:00.38 | techman97_andy | hehehe |
| 01:00.38 | tainted- | i have a support ticket for that too |
| 01:00.45 | PMantis | lol |
| 01:01.02 | PMantis | I've had BV problems, but attibuted that to latency. |
| 01:01.05 | xachen | ViaTalk is too buddy |
| 01:01.06 | tainted- | <Broadvoice> we have been known to eat the occassionaly baby |
| 01:01.14 | xachen | :P |
| 01:01.28 | xachen | er, buggy |
| 01:01.39 | tainted- | orlok what does your Dial string look like |
| 01:02.41 | PMantis | tainted-, xachen, ok... then from my list, that leaves voicepulse ... which charges by the minute for all OB dialing. |
| 01:02.52 | tekati | tainted: Tell me another VOIP provider besides Broadvoice that gives you unlimited inbound and outbound calling to the US even for a flat rate of 20.00 or close to it. |
| 01:03.17 | tekati | I agree their service leaves a bit to be desired at times. |
| 01:03.52 | tainted- | 1) 78.02 to set up an acct |
| 01:03.52 | PMantis | tekati, and 4 simultaneous calls |
| 01:04.03 | tainted- | 2) unlimited not really unlimited - softcap at 1000 |
| 01:04.44 | tainted- | 3) little asterisk support, weird downtime, random config changes that break your box, no support for anything but inband ulaw |
| 01:05.02 | tainted- | 4) customer service by special olympians |
| 01:05.14 | PMantis | heh |
| 01:06.03 | tekati | Who is that? Broadvoice with Asterisk did not charge me that to start up. They do not softcap me at 1000 I had over 3250 minutes last month. Very weird asterisk support with weird downtimes and random configuration changes I will give you that. LOL on the Customer Service I agree there. |
| 01:06.43 | tainted- | did u sign up for byod? |
| 01:06.52 | tekati | If they could get their stuff together they would have a good service. |
| 01:06.58 | tekati | Yes I did sign up for the BYOD. |
| 01:07.03 | tainted- | try consistently pushing 3250 every month |
| 01:07.24 | tekati | Hmm interesting let me see what my average a month is. |
| 01:07.39 | PMantis | I'm doing between 2 and 2.5k /month with BV |
| 01:07.58 | tainted- | wow must be a new policy then |
| 01:08.07 | tainted- | all US? |
| 01:08.18 | tekati | Yes. for me anyway. |
| 01:08.34 | PMantis | Yes, mostly within the same areacode |
| 01:09.35 | PMantis | But my BIGGEST complaint with them is the fact that the closest DID is located 20 Miles away, and long distance for my neighbors... second biggest probelm is the 10+ seconds is sometimes takes to connect. |
| 01:09.54 | tekati | I did 3100 last month. 3250 the month before that. 3225 the month before that. 1339 the month before that. 1439 the month before that. 2245 the month before that. So consistantly over 1000 for sure. |
| 01:10.20 | tekati | I guess I am lucky they have local numbers for me here. |
| 01:10.31 | tekati | Actually they were the only ones I could find that actually do. |
| 01:10.41 | tainted- | tekati where? |
| 01:10.48 | *** join/#asterisk Ridgeback (n=jircii@180.246.8.67.cfl.res.rr.com) |
| 01:11.03 | tainted- | what are u doing to push 3250/min/month |
| 01:11.05 | tekati | Bakersfield, CA |
| 01:11.23 | tekati | Conference calls unfortunatly. I work from home. |
| 01:11.34 | tainted- | 20.00 is their residential plan |
| 01:11.40 | tainted- | but you're doing business on it |
| 01:11.42 | tekati | Yep this is my house. |
| 01:11.47 | *** join/#asterisk anthm (n=anthm@CPE-69-76-83-52.wi.res.rr.com) |
| 01:11.47 | *** mode/#asterisk [+o anthm] by ChanServ |
| 01:11.47 | tainted- | so technically you're cheating them |
| 01:12.00 | tekati | Its a very gray area. |
| 01:12.01 | tekati | :) |
| 01:12.15 | tainted- | well i would stay with them |
| 01:12.31 | tainted- | b/c the more customer like u that go with BV, the more likely they will go bankrupt |
| 01:12.32 | tekati | In my defense I have 2 numbers with them. one is just a additional number to the main account. |
| 01:13.04 | tekati | I would not mind paying them 45 if their service got a little better. |
| 01:13.17 | tekati | It has got better lately. |
| 01:13.23 | tekati | But still not quite there. |
| 01:13.43 | tekati | I just do not like that I can not send them CID. I have to use someone else for that. |
| 01:13.46 | tainted- | yea when i had them they'd suspend if i pushed too many mins through |
| 01:14.22 | tekati | Really. That is interesting. The first time they do that we are going to have to exchange some words. I have been there since 7/2004 though. |
| 01:14.27 | tekati | So I hope it does not come to that. |
| 01:14.55 | tainted- | one of the olympians told me that there's an algorithm to detect abuses |
| 01:15.05 | tainted- | predictive dialers and etc |
| 01:15.29 | tekati | The reason they probably leave me alone is that I am very rarely on more then 1 call at a time unless the kids are home calling a friend or something. |
| 01:15.46 | tekati | Yea I don't do any of that kind of stuff. |
| 01:15.48 | tainted- | yea don't they offer two channels? |
| 01:15.48 | tekati | Real usage. |
| 01:16.04 | tekati | Up to 4 from what PMantis was saying. |
| 01:16.13 | tainted- | i thought that was VP |
| 01:16.20 | tekati | Ah maybe it is. |
| 01:16.22 | tekati | Could be. |
| 01:16.37 | tainted- | who do u conference call with |
| 01:16.37 | *** join/#asterisk nain (n=nain@202.59.90.180) |
| 01:16.41 | nain | Hi Every body |
| 01:16.45 | tainted- | headquarters? |
| 01:16.57 | tainted- | same folks each time or different |
| 01:17.04 | PMantis | tainted-, Perhaps that is the case... but I *know* we've been on at least 2 before, and I *think* we've gone up to 3 simultaneous calls with them. |
| 01:17.14 | tekati | Different people but it is a headquarters conference bridge I do use. |
| 01:17.20 | nain | I would like to patch this file to asterisk for packetization purpose of frame: http://bugs.digium.com/file_download.php?file_id=9746&type=bug |
| 01:17.32 | tekati | Sometimes the same people. |
| 01:17.42 | tekati | Same number to dial in anyway. |
| 01:17.45 | nain | What is the syntex or where to place a file for patch to be work correctly |
| 01:17.55 | tainted- | ahh |
| 01:18.07 | tainted- | why not set up a conference bridge and then everyone can talk for free |
| 01:18.17 | nain | like can any one guide me how to patch any file ? like "patch -p1 < ...... ?" |
| 01:18.25 | Strom_C | nain: man patch |
| 01:18.39 | Ridgeback | nain -- copy that web page to a text file and use patch -p..... |
| 01:18.44 | Strom_C | PMantis: I just tested - I can only do two concurrent calls on my BV account |
| 01:18.45 | Cybertoy | tainted, try ext 514 in fwd... |
| 01:18.53 | nain | Strom_C: Ok where to place the file and how to execute the patch |
| 01:19.12 | tekati | Strom_C does that include if you do three way calling etc? |
| 01:19.20 | Ridgeback | nain in the same directory as the original source file |
| 01:19.31 | tekati | nain: patch -p1 < patchfile |
| 01:19.44 | Strom_C | tekati: "three-way calling" is just setting up another SIP call and letting the phone switch back and forth |
| 01:20.02 | nain | tekati: I did the same but patch required different file to patch in different directory of asterisk and it ask me for path |
| 01:20.02 | Strom_C | i'm using this with asterisk; dont know about using an ATA |
| 01:20.28 | tekati | nain: patch -p1 filetopatch < /somedir/somepatch |
| 01:20.33 | nain | However i have provided path but some not all of files or Hunked are successfully patched |
| 01:20.42 | PMantis | Does anyone know if a SIP phone in a call center can allow an agent to be set "Not Ready" ? |
| 01:20.59 | tekati | nain: sounds like the patch is out of date. |
| 01:21.26 | Ridgeback | PMantis. perhaps by using the "presence" functionality? |
| 01:21.33 | nain | rtp.packetization-2006-03-30.patch [^] (13,344 bytes) 03-29-06 14:40 |
| 01:21.57 | nain | Tekati: isnn't it, could u plz check it if will it work for current release or not ? |
| 01:22.08 | anthm | cute that was my patch =D |
| 01:22.28 | tekati | nain: sounds like you go the author of the patch to talk to :) |
| 01:22.41 | PMantis | Ridgeback, Hmmm, suggest a SIP phone that can do that? |
| 01:23.00 | PMantis | Ridgeback, Any know of any way to report on how long an agent was "present" ? |
| 01:23.13 | tekati | Strom_C: You appear to be correct. 2 calls maximum at a time. Did not even know that. |
| 01:23.19 | tekati | Learn something new everyday. |
| 01:23.27 | Ridgeback | PMantis, yep! Ploycom IP600. I have one of them. Within a local Asterisk network, the phones can be set for anynumber of "statuses |
| 01:23.29 | Ridgeback | " |
| 01:23.57 | Qwell | anthm: You can help him fix it then. ;) |
| 01:23.59 | Ridgeback | PMantis, the only problem is, it cannot send SIP prescense data across IAX or SIP to other asterisk switch, that I know of... |
| 01:24.23 | tekati | I know the answer to this already but no one by chance has the ADMIN GUIDE for a PAP2 do they? I have the PAP2-NA and need to know what some of the configuration options are. |
| 01:24.54 | anthm | i wrote it in september |
| 01:25.02 | anthm | so now it's kinda out of my hands |
| 01:25.23 | tekati | nain: that is going to be the problem then the patch is way out of date. |
| 01:26.08 | anthm | http://bugs.digium.com/view.php?id=5162 |
| 01:26.10 | tekati | That patch date is 3/30/06 though so it should not be that out of date. |
| 01:26.16 | Ridgeback | time for bed.... later guys. |
| 01:26.30 | nain | aha |
| 01:26.38 | *** part/#asterisk Ridgeback (n=jircii@180.246.8.67.cfl.res.rr.com) |
| 01:27.01 | nain | tekati: it's not too old patch it should work |
| 01:29.00 | tekati | nain: Any changes that have been commited since 3/30 can cause that patch to fail. anthm would be your best bet if he does not want to do anything you will have to go through it yourself and try to correct the imbalance. |
| 01:29.39 | tekati | PayPal him a few bucks. That might get him to work on it for ya. |
| 01:31.20 | tekati | Anyone want to trade a TDM40B for a decent PolyCom or Cisco 79XX phone? |
| 01:31.31 | tekati | Within reason that is. |
| 01:31.43 | Qwell | I'll give you a grandstream |
| 01:31.58 | Strom_C | can I give you the 7960 that has a broken base? |
| 01:32.03 | tekati | Had many offers for Grandstreams. |
| 01:32.21 | Strom_C | wait, I already have two TDM400 cards, what the hell do I need a third for? |
| 01:32.27 | tekati | LOL |
| 01:32.37 | file | you should send me one |
| 01:32.42 | file | :D |
| 01:32.51 | Strom_C | file: well one is in my stable PBX and one is in my dev box |
| 01:33.12 | Strom_C | oh Qwell, can you link me to your patch again? |
| 01:33.23 | Qwell | team/north/chan_skinny-fixup |
| 01:33.43 | nain | well patch is another case: Actually i am getting problem of overhead of IP, with g729 or any other codec that is 4 time to codec bandwidth required |
| 01:34.04 | nain | like 8kbps required by g729 but it is consuming upto 30kbps |
| 01:34.11 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-177-36.lsanca.fios.verizon.net) |
| 01:34.21 | nain | Can any one have idea how to resolve or reduce overheads ? |
| 01:35.17 | Qwell | iax2 trunking |
| 01:35.23 | Strom_C | or just go to ulaw so you have proportionally less overhead ;) |
| 01:36.01 | nain | Strom_C: I can't afford ulaw .... |
| 01:36.07 | Strom_C | it was a joke |
| 01:36.28 | nain | :) |
| 01:37.01 | nain | Qwell: iax2 trunking will only reduce few kbits like maximum 3 to 4 and it's not enough compare to 8kbps |
| 01:37.43 | PMantis | 90k vs 13k packet size helps some... |
| 01:38.05 | *** join/#asterisk ramo (n=ramo@59.92.137.242) |
| 01:38.07 | nain | like 8kbps for g729 and it's going to almost 28kbps up and 28kbps down so it's too much for bandwidth |
| 01:38.29 | Strom_C | nain: what kind of connection are you on where 28kbps is too much? |
| 01:38.54 | PMantis | 19.2kbps modem ;) |
| 01:40.01 | PMantis | Or he's running a torrent on his OC-48. LOL |
| 01:40.10 | orlok | cool, hello-world works |
| 01:40.11 | *** join/#asterisk techie (n=gus@antibala.com) |
| 01:40.12 | orlok | theres a step |
| 01:40.35 | nain | Strom_C: I have 512Kbps connection for 10 channels and it seems not to be good enough. |
| 01:40.49 | tainted- | nain what codec |
| 01:40.55 | Strom_C | nain: whats your latency like to your host? |
| 01:40.57 | nain | tainted: g79 |
| 01:40.59 | nain | g729 |
| 01:41.08 | tainted- | yea ping your provider |
| 01:41.21 | tainted- | could be a crappy router too |
| 01:41.28 | Strom_C | nain: I routinely run six or seven ulaw calls concurrently on a 608kbps uplink using ulaw with no quality problems |
| 01:41.34 | nain | latency is not more then 230ms |
| 01:41.47 | tainted- | nain that's high |
| 01:41.48 | Strom_C | 230ms is almost pushing it |
| 01:42.02 | Strom_C | i think my voip proxies are like 60ms away |
| 01:42.02 | nain | This is the maximum |
| 01:42.12 | tainted- | who cares about maximum |
| 01:42.14 | tainted- | what is avg |
| 01:42.34 | anthm | yah he's probably in pk where that is normal |
| 01:42.51 | nain | anthm: u right |
| 01:42.52 | anthm | which is why he knows he needs to maximize the ratio |
| 01:42.55 | anthm | =D |
| 01:42.58 | tainted- | ah |
| 01:43.16 | tainted- | how far is your provider in relation to u |
| 01:43.21 | PMantis | Anyone here build an inbound call support center with multiple queues that can give a couple pointers? |
| 01:43.28 | nain | Provider is in US |
| 01:43.32 | anthm | 20ms g729 is only 3 bytes bigger that 1 rtp packet header |
| 01:44.02 | tainted- | nain put a box at a good colo and point your users to the colo |
| 01:44.16 | tainted- | then trunk to the us provider from the colo |
| 01:44.16 | anthm | so you can see why one would want to put more data per packet |
| 01:44.37 | *** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com) |
| 01:45.29 | orlok | hey |
| 01:45.36 | orlok | the url for the book is missing from the topic |
| 01:45.42 | Strom_C | ~thebook |
| 01:45.46 | jbot | from memory, thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 01:45.46 | nain | tainted: well, i have my server colocated in well known place and terminated to US, The only thing is that call is orignating from pk and that's why delay is must |
| 01:46.43 | anthm | PMantis run while you still can |
| 01:47.28 | nain | Strom_C: bandwidth issue is on orignation side, Here in Pk bandwidth is too expensive and for 10 channels 512kbps is too much and even with 512kbps i am not getting good quality |
| 01:47.45 | Strom_C | nain: are you getting good quality with just one channel? |
| 01:48.00 | nain | Srom_C: right, |
| 01:48.20 | nain | Strom_C: voice is fine for even 5 or 6 channels |
| 01:48.46 | Strom_C | using IAX or SIP? |
| 01:49.53 | nain | I tried both SIP and IAX, IAX take about 3.2KB up and 3.2KB down per channel, While SIP take almost 3.8KB up and down |
| 01:50.08 | Strom_C | nain: what packet size? |
| 01:50.15 | nain | so 1 channel take almost 8KB overall |
| 01:50.43 | nain | Strom_C: 2 Frame per packet |
| 01:50.47 | tainted- | nain are u transcoding? |
| 01:50.57 | Strom_C | how many ms per packet? |
| 01:51.27 | tainted- | could be cpu |
| 01:51.39 | nain | STrom_C: 20ms default |
| 01:53.02 | nain | Strom_C: Call is orignating from SIP Dialer like Eyebeam and it won't allow to make changes in packet per frame etc.... neither asterisk |
| 01:53.27 | *** part/#asterisk surfdue (n=tyler@unaffiliated/surfdue) |
| 01:54.10 | infinity1 | i'm trying to get a backup service working for outbound calls (voxee) and i'm getting this error which i can't resolve. SIP/2.0 404 Not Found |
| 01:54.29 | infinity1 | anyone have a clue for me? |
| 01:54.45 | Strom_C | infinity1: you have a typo somewhere, methinks |
| 01:54.56 | infinity1 | err |
| 01:54.58 | orlok | infinity1: when i had that issue, i tcpdumped the connection from * to the voip provider, checked that, turns out it was a password error |
| 01:55.25 | orlok | infinity1: are you getting that from the phone -> asterisk, or asterisk -> sip provider? |
| 01:55.36 | infinity1 | asterisk-> sip provider |
| 01:55.44 | orlok | typo/password error i'd say |
| 01:55.51 | orlok | resource not found |
| 01:56.02 | nain | Strom_C: any clue |
| 01:56.41 | Strom_C | nain: try using IAX trunking, if possible...if not, use a bigger pipe or another box on another pipe |
| 01:57.03 | infinity1 | looking... |
| 01:57.33 | PMantis | Can someone recommend a phone that has: one-button status changes, voicemail lights, speed-dial buttons, LCD for CallerID, etc. |
| 01:57.45 | orlok | PMantis: most of them? |
| 01:57.50 | Strom_C | haha |
| 01:57.53 | *** part/#asterisk Isaiah (n=Isaiah@208-187-93-4.br1.hnv.mi.frontiernet.net) |
| 01:57.55 | orlok | though notsure about the one button status changes |
| 01:58.10 | PMantis | orlok, I have little to no experience with SIP hardphones |
| 01:58.12 | orlok | PMantis: sipura does all that i think, |
| 01:58.13 | orlok | ahh |
| 01:58.34 | orlok | sipura/linksys/cisco would be a good bet then |
| 01:58.35 | infinity1 | sheesh . can't find a type |
| 01:58.52 | orlok | <PROTECTED> |
| 01:58.52 | infinity1 | er typo. i set it up with eyebeam and it worked first trt |
| 01:58.54 | infinity1 | er rty |
| 01:59.02 | infinity1 | try |
| 01:59.34 | infinity1 | sorry. just broke my thumb. i'm still getting use to typing with a cast |
| 02:00.23 | PMantis | orlok, What SIP phones do you have experience with? |
| 02:02.51 | *** join/#asterisk bkw_ (n=brian@adsl-70-143-63-171.dsl.tul2ok.sbcglobal.net) |
| 02:03.31 | infinity1 | i'm out of ideas. what else is out there besides voipjet and voxee? |
| 02:03.45 | Strom_C | there's asterlink ;) |
| 02:03.50 | Strom_C | nufone |
| 02:03.54 | Strom_C | voicepulse connect |
| 02:03.59 | Strom_C | what else |
| 02:06.02 | *** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
| 02:09.46 | nain | Strom_C: Well i have tried another box as well and even i tested it between very low latency server but problem is same |
| 02:12.24 | orlok | PMantis: Cisco, Grandstream, Linksys |
| 02:12.45 | orlok | PMantis: Grandstream cheap+nasty feeling, Cisco's are $$$, Linksys are a nice balane between them |
| 02:13.02 | orlok | only thing cisco has that linksys doesnt is a minibrowser |
| 02:13.18 | orlok | got one of each on my desk currently |
| 02:13.59 | PMantis | orlok, I was just looking at some Linksys phones, D-Link phones. |
| 02:15.22 | PMantis | orlok, What do you think of the Cisco CP-7912G |
| 02:16.33 | PMantis | Or snom phones ? |
| 02:18.42 | [TK]D-Fender | Snom = Flakey, Linksys=overpriced, and no speed-dial/presence support, 7912 = way overpriced and sup-functional |
| 02:18.44 | orlok | havent used snom or dlink |
| 02:18.59 | orlok | yeah, i'd avoid the cisco's :) |
| 02:19.23 | PMantis | ok, I'm hearing a Linksys recommendation |
| 02:19.50 | VoicePulse | PMantis: The Linksys phones also have Sipura firmware with is a huge plus in terms of provisioning/features. |
| 02:20.21 | VoicePulse | PMantis: Although the importance of that depends on your deployment size. |
| 02:20.42 | PMantis | VoicePulse, Going to be about 20 Agents to start... |
| 02:20.55 | PMantis | I'm looking at http://www.voiplink.com/Linksys_Voice_Over_IP_Phones_s/35.htm trying to decide on a recommendation. |
| 02:21.54 | VoicePulse | PMantis: Don't use the 841s though. I believe they are based on older hardware that wasn't as refined as the new 941/942s. |
| 02:21.59 | PMantis | Don't care to find the cheapest vendor (yet), just looking for a side-by-side comparison to find the cheapest phone that supports what's needed. |
| 02:24.08 | PMantis | ...and since I've never setup a SIP phone, let alone in a call center, I'm not sure what's needed yet. |
| 02:24.42 | PMantis | I've only used a carded system so far... so dual RJ45's are a must for QOS. |
| 02:28.06 | *** join/#asterisk DoktorGreg (n=Greg@70.91.121.89) |
| 02:28.11 | PMantis | Cool. Linksys SPA 922 supports syslogging |
| 02:29.45 | DoktorGreg | funny sip nat problem |
| 02:30.04 | DoktorGreg | calling in always works fine |
| 02:30.09 | DoktorGreg | however |
| 02:30.28 | DoktorGreg | on the first phonecall out, cant hear on sip phone in question |
| 02:30.47 | DoktorGreg | after that all is fine for about 15 minutes |
| 02:30.56 | DoktorGreg | then first phone call out cant hear |
| 02:31.03 | DoktorGreg | then all is fine again |
| 02:31.22 | DoktorGreg | is that my nat device? |
| 02:33.01 | DoktorGreg | ok question |
| 02:33.05 | DoktorGreg | rather reality check |
| 02:33.19 | DoktorGreg | I just replaced my ups with a bigger ups |
| 02:33.24 | PMantis | Sniff the packets to find out for sure... its the SIP communication that dictates the RTP setup. |
| 02:34.02 | DoktorGreg | with the new ups the lcd screens seem brighter |
| 02:34.11 | DoktorGreg | and the computer seems to be running faster... |
| 02:34.20 | DoktorGreg | am i imagining that??? |
| 02:34.24 | *** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com) |
| 02:35.23 | DoktorGreg | PMantis, since i doubt you will walk me through sniffing packets |
| 02:35.25 | DoktorGreg | on irc |
| 02:35.41 | DoktorGreg | can you point me to a tutorial on sip packet sniffing? |
| 02:35.48 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
| 02:35.55 | orlok | DoktorGreg: ethereal and tcpdump |
| 02:36.35 | *** join/#asterisk d-tech (n=dtc@72.245.233.107) |
| 02:36.40 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
| 02:36.50 | PMantis | Unfortunately, I don't know of a tutorial.. I just felt my way through the packet headers a couple times. |
| 02:37.44 | ManxPower | The SIP RFCs will help you understand the contents of the SIP packets, but really, the RFC is poorly written and confusing |
| 02:39.21 | DoktorGreg | what phones, becaues im having problem with spa 1001 |
| 02:42.30 | [TK]D-Fender | PMantis : Linksys is missing quite a few things. First, 2 lines, no presence, flimsier feel than Polycom / Cisco, poor usability of LCD. |
| 02:42.56 | [TK]D-Fender | linksys does have a backlit screen on the 922 though. |
| 02:43.37 | [TK]D-Fender | I'd rate the Polycom IP 501 as a much better purchase for that range |
| 02:43.59 | PMantis | [TK]D-Fender, Oh really? I was looking at the 942. I need presence (like an agent setting themselves as not avail for queue calls, but still taking direct extension calls). |
| 02:44.05 | *** join/#asterisk IceBerg (n=iceberg@cpe-65-29-93-57.indy.res.rr.com) |
| 02:44.10 | [TK]D-Fender | Sorry, meant to type 942 there :) |
| 02:44.20 | PMantis | heh |
| 02:44.36 | [TK]D-Fender | PMantis : Linksys doesn't do presence. Only SIP-B which * does not support, and only across 2 lines. |
| 02:44.44 | *** join/#asterisk fabiokl (n=fabio@200.175.4.169.adsl.gvt.net.br) |
| 02:44.58 | [TK]D-Fender | Whic means you can only monitor *1* if and when * actually gets around to supporting it anyways |
| 02:45.10 | PMantis | hah |
| 02:45.19 | PMantis | ok, that's not good. |
| 02:45.25 | [TK]D-Fender | Without paying to activate the other "lines |
| 02:45.30 | [TK]D-Fender | No its not.... |
| 02:45.32 | *** part/#asterisk fabiokl (n=fabio@200.175.4.169.adsl.gvt.net.br) |
| 02:45.36 | PMantis | ANy "gotcha's" with Polycom 501 ? |
| 02:45.51 | [TK]D-Fender | If the price point was lower it'd have a place in my suggestion scale. Just not yet. |
| 02:45.54 | tainted- | price |
| 02:46.49 | [TK]D-Fender | PMantis : Yes. You have an initial choice on the 501 for PoE included or wall-wart. switching will cost a little extra. You should also get it from a cert'd dealer to ensure support. |
| 02:46.57 | [TK]D-Fender | Polycom IP 501 = $170 |
| 02:46.57 | PMantis | [TK]D-Fender, I'm not sure I understand your sentence. That mean you don't have a Polycomm phone ? |
| 02:47.24 | [TK]D-Fender | PMantis : I own every model of Polycom desk IP phone. I've also owned an SPA-941 |
| 02:47.39 | PMantis | ok |
| 02:47.43 | [TK]D-Fender | I ditched my 941 to financeits Polycom replacements. |
| 02:47.52 | [TK]D-Fender | And glad of it. |
| 02:47.56 | PMantis | lol |
| 02:48.17 | [TK]D-Fender | I run all 600's at work, and a 501 + 301 at home now. |
| 02:48.46 | PMantis | Ok, what's the difference between the Polycom phones, and will the differences matter at all for an inbound tech cupport call center? |
| 02:49.11 | [TK]D-Fender | Don't get me wrong, the SPA-941/2 are decent phones, just that they don't fit in the right price bracket and are more suited to home hackers |
| 02:49.52 | [TK]D-Fender | PMantis : For inbound typicall agents will only be on 1 call at a time. They also typically won't be speakerphone users and likely on headset. |
| 02:49.57 | PMantis | What I'd *really* like to see is a single button agent logoff (perhaps logon, but what about roaming agents?) |
| 02:50.07 | PMantis | [TK]D-Fender, Agreed |
| 02:50.15 | [TK]D-Fender | PMantis : That in mind I'd strongly suggest the IP 301 + Plantronics M12 amp + headset |
| 02:50.56 | [TK]D-Fender | PMantis : Polycom has some specialized agent functionality that is planned to be merged into 1.4 due this summer. |
| 02:51.03 | [TK]D-Fender | For login/out/pause |
| 02:51.19 | PMantis | OOoooooooooooh |
| 02:51.25 | [TK]D-Fender | PMantis : Naturally as its only supported in a patch I haven't tested it personally. |
| 02:51.35 | PMantis | and the 301 supports all an agent would need ? |
| 02:52.00 | DoktorGreg | ok, have ethereal installed and working |
| 02:52.04 | [TK]D-Fender | PMantis : All the typical SIP stuff (hold, blind/att xfer, conference), and so on |
| 02:52.10 | DoktorGreg | now how do i spy on my ata adaptor? |
| 02:52.30 | PMantis | I guess it would help me to see a side-by-side comparison of features on the phones. |
| 02:52.56 | PMantis | [TK]D-Fender, Here's a provider that's selling a 301 w/ a headset.. |
| 02:52.57 | PMantis | http://www.voipsupply.com/product_info.php?&products_id=1021 |
| 02:53.04 | [TK]D-Fender | PMantis : Special note : RJ12 headset jack so you WILL want to get them a Plantroncs (GN Netcom, etc) amp + headset. It makes all the difference VS el-cheapo 2.55mm ones |
| 02:53.37 | *** join/#asterisk ptiggerdine (n=ptiggerd@203-206-88-247.dyn.iinet.net.au) |
| 02:53.59 | PMantis | [TK]D-Fender, You saying that after opening the link I sent? |
| 02:54.02 | [TK]D-Fender | PMantis : How big a call center, and what kind of cll volume? |
| 02:54.32 | [TK]D-Fender | PMantis : No, I was typing before your line came over |
| 02:55.02 | PMantis | [TK]D-Fender, brand new call center, hasn't opened doors yet. Will be inbound technical support (paid). They're expecting about 15-20 agents to start. |
| 02:55.13 | [TK]D-Fender | I'm looking at that link now.... headset looks cheap and I don't see it AMP'd... straight does not measue up if you're in a potentially noisy environment |
| 02:56.05 | PMantis | Do you know what this means? "Handset operation through independent jack" |
| 02:56.20 | [TK]D-Fender | I don't know the brand of that headset so I distrust it by nature, plus no mention of AMP. |
| 02:56.20 | PMantis | LOL |
| 02:56.51 | [TK]D-Fender | PMantis : that means the headset isn't used in-line with the handset, it has its own RJ12 jack. |
| 02:57.03 | Strom_C | RJ9 |
| 02:57.11 | PMantis | heh |
| 02:57.15 | Strom_C | RJ12 is six-position four-conductor |
| 02:57.17 | [TK]D-Fender | Others use the RJ12 handset to wire in and you need to leave the handset off-hook. |
| 02:57.19 | PMantis | RJ11/12 is the wall jack size |
| 02:57.32 | Strom_C | RJ9 is four-position four-conductor |
| 02:57.42 | [TK]D-Fender | Strom_C : RJ12 = mini handset connector. |
| 02:58.02 | Strom_C | says who? |
| 02:58.13 | DoktorGreg | oh man for headsets use softphone and usb headset |
| 02:58.26 | PMantis | ha |
| 02:58.45 | PMantis | Not when a client says they want hard phones. :) |
| 02:58.51 | [TK]D-Fender | Strom_C : Last I checked. could be mistaken. Either way the headset jack is the same style as the handset jack :) |
| 02:58.56 | DoktorGreg | zactly |
| 02:59.26 | DoktorGreg | i just let people in my office try the usb headsets |
| 02:59.35 | DoktorGreg | and they said they wanted the better isolation |
| 02:59.38 | Strom_C | [TK]D-Fender: yes, they're the same. but the 4-position 4-conductor plug is RJ9. |
| 02:59.48 | Strom_C | RJ-11 and up are six-position |
| 02:59.53 | DoktorGreg | but we have noisy office |
| 02:59.57 | [TK]D-Fender | Strom_C : I'll want to brush up on it to be sure. |
| 03:00.18 | Strom_C | RJ-11 == 2-conductor, RJ12 == 4-conductor, RJ-14 == 6-conductor |
| 03:00.27 | DoktorGreg | next im gonna map a both a hard phone and a soft phone to particular desks |
| 03:00.28 | Strom_C | (all six-position) |
| 03:00.45 | DoktorGreg | so they can use either seamlessly |
| 03:01.09 | MikeJ[Laptop] | rj smar-j |
| 03:01.13 | [TK]D-Fender | Strom_C : Ummm. RJ is the jack type, not jsut the # of conductors....RJ11 = 3 pair, typically 1-2 used |
| 03:01.23 | Strom_C | gah, NO NO NO NO NO. |
| 03:02.03 | DoktorGreg | can anybody tell me how to spy on my ata with ethereal? |
| 03:02.39 | file | give me wireless any day. |
| 03:02.42 | [TK]D-Fender | Strom_C : I'll jsut assume that my numbering is wrong OK? |
| 03:02.49 | MikeJ[Laptop] | http://en.wikipedia.org/wiki/RJ-11 |
| 03:02.52 | file | [TK]D-Fender: Strom is obsessive compulsive |
| 03:03.01 | MikeJ[Laptop] | and wrong I think |
| 03:03.31 | PMantis | Heh, All I know is that RJ11/12 is what's in my wall that I plug y deskphones into. |
| 03:03.32 | *** join/#asterisk coppice (n=chatzill@153.192.17.210.dyn.pacific.net.hk) |
| 03:04.11 | MikeJ[Laptop] | yeah.. rj11 is 1 pair, 14 is 2, 12/25 is 3 |
| 03:04.13 | [TK]D-Fender | Ah whatever! Not something I'd really want to argue over right ro wrong. |
| 03:04.25 | Strom_C | I think I found the requisite document, but it'll take a few minutes to download from the TIA |
| 03:04.42 | PMantis | Heh, if it works.. no discusion needed. |
| 03:05.03 | PMantis | [TK]D-Fender, So whatever the RJ # is, it's a headset connector.... |
| 03:06.54 | PMantis | [TK]D-Fender, Thanks for taking the time to go over phones. I think I'll be suggesting the IP-301's. |
| 03:06.59 | [TK]D-Fender | PMantis : Yeah, that :) |
| 03:07.45 | [TK]D-Fender | PMantis : I have tried a plantronics headset that came with an adapter to it that plugs direct to the phone without an amp, but it sounded weak. Unless your reps are well isolated I wouldn't really suggest it. |
| 03:08.03 | PMantis | [TK]D-Fender, so 1.4 will directly support polycom status updates? Is that officially written anywhere? |
| 03:08.08 | [TK]D-Fender | PMantis : If you follow my lead you'll find you'll actually spend more on headset tech than the phones themselves :) |
| 03:08.44 | PMantis | Yeah, I've seen the prices of good headsets... |
| 03:08.45 | [TK]D-Fender | PMantis : It's planned. What will actually happen is anyones guess, but its in the mailing lists. Look up "polycom ACD" |
| 03:09.09 | coppice | PMantis: no. you've seen the prices of "professional" headset :-) |
| 03:09.09 | [TK]D-Fender | PMantis : And they're worth it. I didn't believe so at first... but sometimes you jsut have to do a job right..... |
| 03:09.20 | *** join/#asterisk kainam (n=Jake@202.137.160.110) |
| 03:10.04 | coppice | the prices of things like plantronics are totally unreasonable, but they are tough enough to stand life in a call-centre |
| 03:10.24 | PMantis | GN-Netcoms are good, too.... |
| 03:10.42 | [TK]D-Fender | PMantis : Both are good... GN is only nominally cheaper |
| 03:10.46 | coppice | and Hello Voice. those 3 pretty much are the market |
| 03:11.43 | [TK]D-Fender | coppice : Hadn't heard of them... |
| 03:12.10 | [TK]D-Fender | coppice : Of course I learned what I had to to equip my team when their manager started whining and might have sooner than later :) |
| 03:12.36 | coppice | they are kinda weird, because their of their supply chain. you have to mail order them or something. a lot of call centres use them, though, with good results |
| 03:13.49 | MikeJ[Laptop] | coppice, you dropped your unicall! |
| 03:13.55 | MikeJ[Laptop] | over there --->> |
| 03:13.59 | file | oh god |
| 03:14.04 | MikeJ[Laptop] | :P |
| 03:14.14 | [TK]D-Fender | coppice : I only find the in-your-face advertised stuff and read up.... anything "behind the scenes" is out of my scope. |
| 03:14.15 | file | in my highly hyper caffeinated state, I laughed out loud at that |
| 03:14.23 | MikeJ[Laptop] | it's late |
| 03:14.33 | PMantis | hehe |
| 03:14.34 | file | yeah, it's late... |
| 03:14.44 | [TK]D-Fender | file : no more Red Bull for you! |
| 03:14.52 | PMantis | It is here, too... but *somewhere* it's early. |
| 03:14.57 | file | I don't drink red bull, I tried it when I was at Digium HQ and it gave me a horrible headache |
| 03:15.03 | xachen | when electricity pulses through my body I laugh for no reason. does that count? :P |
| 03:15.13 | MikeJ[Laptop] | russellb, so autoconf done yet? |
| 03:15.29 | russellb | ? |
| 03:15.34 | russellb | yes, autoconf is complete. |
| 03:15.36 | PMantis | file, red bull sucks... XS Energy is *tons* better |
| 03:15.41 | russellb | took me 30 years to write it |
| 03:15.42 | MikeJ[Laptop] | LIAR! |
| 03:15.55 | coppice | PCP is more effective |
| 03:16.02 | MikeJ[Laptop] | hehe |
| 03:16.14 | file | I like Bawls |
| 03:16.24 | coppice | so does my wife |
| 03:16.31 | MikeJ[Laptop] | ummmmmmm |
| 03:16.35 | PMantis | ROFL |
| 03:16.35 | MikeJ[Laptop] | wow.... |
| 03:16.48 | file | that's hot |
| 03:16.58 | ManxPower | #Asterisk: After Dark |
| 03:17.06 | file | after hours |
| 03:17.18 | PMantis | I've seen #ltsp do that, but not #asterisk |
| 03:17.33 | PMantis | of course, I'm in #ltsp a TON more |
| 03:19.13 | MikeJ[Laptop] | file, what has Corydon done to you!! |
| 03:19.20 | file | no no |
| 03:19.28 | file | it's "what has Corydon tried to do to you!!" |
| 03:19.35 | russellb | everything? |
| 03:19.36 | MikeJ[Laptop] | phew..... |
| 03:19.42 | MikeJ[Laptop] | I was worried for a second |
| 03:19.50 | MikeJ[Laptop] | so you had mace? |
| 03:20.07 | russellb | he has the whole being really far away thing on his side |
| 03:20.11 | file | if by mace you mean muffins |
| 03:20.12 | file | then yes |
| 03:20.29 | MikeJ[Laptop] | were they old muffins with spikes coming out of them? |
| 03:20.36 | file | yes |
| 03:20.47 | MikeJ[Laptop] | well that works |
| 03:20.58 | file | baked right in even |
| 03:21.45 | MikeJ[Laptop] | so who did the scary kp as uncle sam picutre |
| 03:21.48 | MikeJ[Laptop] | picture |
| 03:21.57 | file | MikeJ[Laptop]: are YOU on the DTMF task force? |
| 03:22.03 | russellb | i haven't seen it |
| 03:22.10 | MikeJ[Laptop] | no, the picture scared me away |
| 03:22.17 | MikeJ[Laptop] | :P |
| 03:22.19 | file | makes sense |
| 03:22.34 | orlok | hey, in the asterisk book, they are using a Zaptel card for outgoing |
| 03:22.39 | orlok | specified as Zap/4 |
| 03:22.40 | MikeJ[Laptop] | but I am a solid supporter of proper dtmf support |
| 03:22.50 | MikeJ[Laptop] | orlok, one sec |
| 03:22.50 | orlok | what would i do for a sip registrar? |
| 03:22.58 | file | ah yes, solid DTMF support |
| 03:23.08 | *** join/#asterisk jsmith (n=jsmith@smithfam.dsl.xmission.com) |
| 03:23.19 | russellb | file: where's that pic |
| 03:23.31 | MikeJ[Laptop] | orlok, oh look.. theres jsmith.. I think he wrote that book |
| 03:23.37 | russellb | lol |
| 03:23.42 | *** join/#asterisk trbldwine (n=trbldwin@71.194.161.170) |
| 03:23.45 | russellb | jsmith: run! |
| 03:23.55 | jsmith | Documentation? On Asterisk? Get out... |
| 03:23.57 | MikeJ[Laptop] | russellb, and he fell for it |
| 03:23.59 | coppice | if a VoIP platform can't get something basic liek DTMF right, there doesn't seem a lot of hope for it :-) |
| 03:24.18 | jsmith | coppice: You should know ... DTMF is overrated |
| 03:24.28 | file | I said we should just get rid of DTMF support altogether |
| 03:24.34 | MikeJ[Laptop] | coppice, TROLL! |
| 03:24.36 | russellb | file: i second that |
| 03:24.38 | MikeJ[Laptop] | :P |
| 03:24.38 | file | something can't be broken if it doesn't exist |
| 03:24.46 | Strom_C | no no no no, then how will I decode what people are dialing just by listening? |
| 03:25.02 | MikeJ[Laptop] | orlok, SIP/ |
| 03:25.03 | jsmith | file: But we need to keep support for the ABCD digits in DTMF... otherwise, only the government will have control of them |
| 03:25.16 | orlok | MikeJ[Laptop]: SIP/username? |
| 03:25.17 | coppice | file: so if a bridge falls down becaus a bolt is missing, rather than broken, its OK? :-) |
| 03:25.23 | file | coppice: yes |
| 03:25.36 | file | cause like - who needs bridges |
| 03:25.40 | file | we have legs... we can jump! |
| 03:25.44 | MikeJ[Laptop] | well... SIP/somthing |
| 03:25.53 | MikeJ[Laptop] | it could be a friend! |
| 03:25.57 | MikeJ[Laptop] | or an ip |
| 03:26.11 | *** join/#asterisk trbldwine (i=trbldwin@71.194.161.170) |
| 03:26.13 | MikeJ[Laptop] | or a dns name |
| 03:26.19 | MikeJ[Laptop] | file, what else can you use |
| 03:26.21 | coppice | file: I'll consider that in an hour or so when crossing the worlds largest road+rail bridge :-) |
| 03:26.30 | file | coppice: excellent |
| 03:26.38 | file | MikeJ[Laptop]: or... a peer |
| 03:26.43 | MikeJ[Laptop] | user? |
| 03:26.47 | file | can't call a user |
| 03:27.06 | MikeJ[Laptop] | why not :P |
| 03:27.11 | jsmith | Users, Peers, and Friends, oh my! |
| 03:27.26 | file | becuz Uncle File says so |
| 03:27.37 | MikeJ[Laptop] | jsmith, be my friend? |
| 03:27.37 | file | and you know better then to disobey me! |
| 03:27.48 | MikeJ[Laptop] | what else could you SIP/ to? |
| 03:28.16 | coppice | isn't VoIP wonderful. "how will we convey DTMF across channels which corrupt the voice". "why, every possible way anyone can think of, of course" :-) |
| 03:29.12 | MikeJ[Laptop] | ok.. here is a fun game.. how many sip dtmf methods can you name |
| 03:29.24 | file | specific to SIP? |
| 03:29.27 | coppice | dumb and dumber? |
| 03:29.28 | russellb | inband, RFC2833, INFO, NOTIFY |
| 03:29.31 | MikeJ[Laptop] | inband, 2833, notify, info, |
| 03:29.34 | MikeJ[Laptop] | theres more... |
| 03:29.34 | russellb | i win |
| 03:29.36 | MikeJ[Laptop] | ummmm |
| 03:29.38 | jsmith | MikeJ[Laptop]: More? |
| 03:29.39 | file | it's a trick question |
| 03:29.41 | MikeJ[Laptop] | yeah |
| 03:29.53 | file | inband isn't a SIP DTMF method, neither is RFC2833 |
| 03:29.56 | MikeJ[Laptop] | file, what are the others |
| 03:30.01 | russellb | file: i know that. |
| 03:30.01 | MikeJ[Laptop] | blah.... |
| 03:30.06 | MikeJ[Laptop] | you knew what I mean |
| 03:30.07 | file | I just know the 4 |
| 03:30.09 | russellb | file: but it's still used in conjunction with SIP, you goof |
| 03:30.10 | MikeJ[Laptop] | there are at least 2 more |
| 03:30.29 | coppice | why are they not SIP methods? SIP says the media will use RTP |
| 03:30.31 | file | how many other ways could you convey it... |
| 03:30.50 | file | coppice: I was picking on his use of words |
| 03:31.04 | orlok | hmm |
| 03:31.06 | file | you could encode it via XML and send it along |
| 03:31.15 | orlok | whats the dissalow= for in sip.conf? |
| 03:31.15 | russellb | file: that's notify! |
| 03:31.23 | file | russellb: do they really do it via XML? |
| 03:31.25 | orlok | disallow, even |
| 03:31.29 | russellb | file: i believe so, yes. |
| 03:31.32 | file | dear god |
| 03:31.36 | russellb | exactly. |
| 03:31.55 | russellb | it's quite a fsck up, if you ask me. |
| 03:32.20 | russellb | oh noes! |
| 03:32.29 | jsmith | orlok: That's to disallow certain codecs from being used |
| 03:32.59 | russellb | file: at least there is only ***ONE*** way to send DTMF with IAX2 |
| 03:33.26 | file | well |
| 03:33.32 | russellb | one correct way. |
| 03:33.35 | *** join/#asterisk Ixthod (n=Ixthod@198.174.206.41) |
| 03:33.39 | file | yes |
| 03:33.46 | file | that chinese chip supports inband though... |
| 03:33.53 | russellb | well that's just stupid |
| 03:34.04 | orlok | jsmith: ahh, so that stops all by default, then i've got the allow ulaw and alaw |
| 03:34.05 | orlok | cool |
| 03:34.15 | russellb | file: i hope asterisk refuses to support it |
| 03:34.29 | file | we don't fire up a DSP on the channel so it never gets recognized internally |
| 03:34.31 | russellb | file: "crappy IAX2 implementation detected, immediately closing connection" |
| 03:34.47 | file | INVAL! INVAL! INVAL! |
| 03:35.01 | russellb | you're speaking in IAX2! |
| 03:35.05 | file | ACK |
| 03:35.15 | russellb | REGREQ |
| 03:35.22 | file | AUTHREQ |
| 03:35.31 | russellb | AUTHREP |
| 03:35.32 | russellb | :/ |
| 03:35.42 | russellb | i haven't looked at it in quite a while |
| 03:35.53 | file | I'm focusing on something else... |
| 03:36.04 | MikeJ[Laptop] | his muffin |
| 03:36.05 | russellb | i am ... too ... |
| 03:36.07 | file | but it's |
| 03:36.40 | file | REGREQ, REGAUTH, REGREQ, REGACK |
| 03:36.40 | PMantis | Can * send a text message to a Polycom IP-301 for things like # callers in queue, # agents available, etc? |
| 03:37.04 | PMantis | ...for display on the LCD, of course. |
| 03:37.09 | jsmith | PMantis: Good question... I have no clue |
| 03:37.25 | PMantis | heh, but at least you admit it. |
| 03:37.35 | file | PMantis: theoretically yes |
| 03:37.42 | orlok | jsmith: uhh, is it bad if i just printed out the whole PDF? ;0 |
| 03:37.45 | file | you'd just have to code it |
| 03:37.57 | jsmith | orlok: Nope... that's what it's there for. |
| 03:38.12 | MikeJ[Laptop] | I just found somthing funny: "Cisco Unity treats any incoming RTP payload above 90 as a DTMF event. |
| 03:38.12 | MikeJ[Laptop] | " |
| 03:38.20 | MikeJ[Laptop] | heheh |
| 03:38.21 | file | MikeJ[Laptop]: seriously? |
| 03:38.26 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-152-241-16.red.bezeqint.net) |
| 03:38.34 | jsmith | MikeJ[Laptop]: You've got to be kidding |
| 03:38.42 | MikeJ[Laptop] | http://doc.trecom.tomsk.su/cisco/cc/td/doc/product/voice/c_unity/whitpapr/sipcomp.htm#1047608 |
| 03:39.04 | coppice | orlok: yes, its bad. you shouldn't consume wood to make paper. you should consume it in a furnace making electricity :-) |
| 03:39.09 | file | ha |
| 03:39.28 | orlok | heh |
| 03:39.34 | orlok | dont let our clients hear that :) |
| 03:39.42 | orlok | we are in the printing industry |
| 03:39.56 | jsmith | MikeJ[Laptop]: |
| 03:40.07 | jsmith | MikeJ[Laptop]: That' looks like an old doc -- 2002? |
| 03:40.29 | jsmith | MikeJ[Laptop]: That's before Cisco even acknowledged that SIP existed in the real world |
| 03:41.05 | file | hello world! |
| 03:41.09 | PMantis | file, jsmith, others, How does an agent normally monitor a queue, to see how many people are waiting, etc? |
| 03:41.30 | r0d3nt|m | !dlrow olleh |
| 03:41.36 | orlok | exten => _0XXXXXXXXX,1,agi(selintra,OutRoute,Outgoing) |
| 03:41.42 | MikeJ[Laptop] | jsmith, maybe.. still scary |
| 03:41.51 | orlok | i should have more entries in my config file for OutRoute and Outgoing, correct? |
| 03:42.16 | mds2 | anyone know if you can change the default dialtone on Cisco 79xx phones? |
| 03:42.21 | *** join/#asterisk mog_home (n=achika54@68.62.237.103) |
| 03:43.36 | jsmith | PMantis: Using an app that talks to Asterisk via the Manager Interface |
| 03:43.46 | jsmith | mds2: Yes, slightly, see dialplan.xml |
| 03:43.54 | PMantis | jsmith, Recommend one? |
| 03:44.08 | PMantis | jsmith, For a Linux desktop... |
| 03:44.25 | mds2 | jsmith: thanks, have been playing with that. I'm curious about changing the default primary dial, busy and reorder tones to something more.. local |
| 03:44.49 | jsmith | mds2: Good luck :-( |
| 03:44.53 | mds2 | jsmith: from what I've read dialplan.xml will let you change secondary dialtone only. can it do more than that? |
| 03:45.05 | jsmith | PMantis: Nope... most are custom apps |
| 03:45.07 | mds2 | hum |
| 03:45.12 | jsmith | mds2: Not that I know of |
| 03:45.19 | mds2 | righto, thanks anyway |
| 03:46.19 | MikeJ[Laptop] | file, found another: http://www.ietf.org/internet-drafts/draft-ietf-sipping-kpml-07.txt |
| 03:46.40 | mog_home | file! |
| 03:46.51 | file | mog! |
| 03:47.02 | jsmith | MikeJ[Laptop]: Using SUBSCRIBE, huh... that's new... |
| 03:48.53 | *** join/#asterisk websae (n=websae@CPE-24-167-206-22.wi.res.rr.com) |
| 03:49.00 | file | let's all just pretend that doesn't exist, mmmk? |
| 03:49.03 | MikeJ[Laptop] | oh wait... then there is 2833 in info.. forgot about that one |
| 03:49.25 | websae | file: back when you did work for asterlink, how long did it take to get a 800 number without it being rushed? |
| 03:49.53 | xachen | Ported or Vanity? |
| 03:50.03 | websae | vanity |
| 03:50.32 | *** join/#asterisk bmg505 (n=leon@c1-177-3.rndf.isadsl.co.za) |
| 03:50.55 | xachen | 7-14 days |
| 03:51.07 | orlok | Do i have the syntax for this right? |
| 03:51.07 | websae | do you work for asterlink? |
| 03:51.10 | orlok | exten => 123,1,Dial(SIP/0385121550,0438015327,1) |
| 03:51.28 | xachen | yes |
| 03:51.31 | file | hahaha |
| 03:51.38 | MikeJ[Laptop] | there are lots... |
| 03:51.43 | file | chan_iax2.c:7700 socket_process: Immediately destroying 666, having received INVAL |
| 03:51.49 | file | 666! |
| 03:51.59 | MikeJ[Laptop] | yes |
| 03:52.10 | orlok | channel of the beast |
| 03:52.23 | coppice | Defining a KPML that can't replace MIDI is stupid :-) |
| 03:52.46 | jsmith | coppice: chan_midi.so? |
| 03:52.46 | brookshire | does sip keep getting more and more bloated everyday or what? |
| 03:52.56 | jsmith | brookshire: No kidding... |
| 03:52.58 | file | brookshire: just like... er nevermind |
| 03:53.11 | brookshire | file!!!!! !! !!!!!!!!!!!!!!!!!!! |
| 03:53.11 | orlok | hmm |
| 03:53.14 | brookshire | !!!!!!!!!!!!!!!!!!!!!! !! |
| 03:53.15 | brookshire | ! |
| 03:53.17 | brookshire | !! |
| 03:53.20 | orlok | when i dial 123, i get a message "Caller ID is Blocked" |
| 03:53.24 | brookshire | oh.. and !!!! ! |
| 03:53.57 | coppice | brookshire: SIP was developed by people who thought telephony didn't have to be as complex as H.323, while H.323 was developed by people who had spent years finding it does. |
| 03:54.14 | MikeJ[Laptop] | ok.. to some up, dtmf methods: inband, 2833, info, 2833 in info, megaco based event detection in INFO, NOTIFY, and kpml.. |
| 03:54.36 | brookshire | why does there need to be 90 billion ways to detect dtmf too? |
| 03:54.42 | coppice | if you look at SIP and H.323 today, which looks cleaner? |
| 03:54.54 | jsmith | coppice: None of the above... |
| 03:55.10 | MikeJ[Laptop] | coppice, of the 2? |
| 03:55.11 | coppice | jsmith: yeah, it a close call :-) |
| 03:55.17 | MikeJ[Laptop] | h323 |
| 03:55.20 | brookshire | sip happens |
| 03:55.22 | MikeJ[Laptop] | hands down |
| 03:55.22 | brookshire | though |
| 03:56.01 | MikeJ[Laptop] | number of lines of rfc is always a fun one to compare |
| 03:56.14 | *** join/#asterisk downunder33 (n=downunde@219.95.158.235) |
| 03:56.31 | jsmith | Why'd you guys talk me into hanging out in here anyway? |
| 03:56.39 | coppice | comparing spec lines says nothing really. |
| 03:56.42 | file | wasn't me. |
| 03:56.46 | mog_home | #asterisk is the coolest jsmith... |
| 03:56.57 | jsmith | Yeah, if I were smart like y'all |
| 03:57.11 | coppice | the real fun specs leave out most of the detail (e.g. T.38). of course, that limits their length |
| 03:57.40 | jsmith | Exactly... specs are what people use to distract suits from the details |
| 03:58.01 | file | I like a protocol that I can just change on a whim... |
| 03:58.01 | coppice | spec == fly in my soup |
| 03:58.25 | MikeJ[Laptop] | file, heh |
| 03:58.40 | MikeJ[Laptop] | DO IT |
| 03:58.41 | file | that nooooobody has to be compatible with! |
| 03:58.45 | file | because it's MINE! |
| 03:58.50 | jsmith | chan_file.so |
| 03:58.53 | file | and that protocol would be, FTP2! |
| 03:58.54 | coppice | file: XML is ideal for that. a language with no semantics :-) |
| 03:58.55 | MikeJ[Laptop] | so you can talk to yourself |
| 03:58.58 | file | File Transfer Protocol... 2! |
| 03:59.10 | *** join/#asterisk redondos_ (n=redondos@190.48.46.149) |
| 03:59.20 | coppice | is FTP2 anything like MSFTP? |
| 03:59.28 | MikeJ[Laptop] | are you saying file has no semantics? |
| 03:59.29 | file | complete opposite |
| 03:59.42 | MikeJ[Laptop] | so youll need to make it IAX3 |
| 03:59.42 | file | FTP2 encodes everything in pig latin |
| 03:59.44 | MikeJ[Laptop] | errrr |
| 03:59.46 | MikeJ[Laptop] | FTP3 |
| 04:00.10 | brookshire | oh why oh why is myspace sooo slow |
| 04:00.16 | PMantis | lol |
| 04:00.22 | russellb | oh why oh why are you using myspace? |
| 04:00.32 | brookshire | because myspace > #asterisk |
| 04:00.39 | MikeJ[Laptop] | ummmm |
| 04:00.42 | MikeJ[Laptop] | your lame |
| 04:00.44 | MikeJ[Laptop] | :P |
| 04:00.45 | file | MikeJ[Laptop]: I refuse to make IAX3 ... yet |
| 04:00.46 | brookshire | haha |
| 04:00.48 | *** kick/#asterisk [brookshire!n=russell@asterisk/developer-and-stable-maintainer/drumkilla] by russellb (off to myspace, then!) |
| 04:00.54 | *** join/#asterisk brookshire (i=mbrooks@hijacked.us) |
| 04:00.59 | brookshire | russell: boo! |
| 04:00.59 | russellb | :-p |
| 04:01.07 | file | russellb: you're supposed to be asleep |
| 04:01.13 | russellb | file: oh yeah ... |
| 04:01.16 | brookshire | don't make me drive to the carolinas |
| 04:01.23 | russellb | brookshire: you wouldn't |
| 04:01.28 | brookshire | you're right |
| 04:01.30 | coppice | compatibility is for wimps. real mean reverse engineer all protocols on the fly |
| 04:01.51 | file | coppice: you rock! :P |
| 04:01.53 | brookshire | actually.. i should convience some people to take a road trip |
| 04:02.25 | brookshire | convince also |
| 04:02.33 | coppice | i saw a road trip one. a volvo with a blowout did 3 sommersauts |
| 04:03.00 | russellb | that's not cool. |
| 04:03.12 | brookshire | i've seen a wreck worse than that :( |
| 04:03.17 | coppice | i was amazes to find a 2 tone car could do that |
| 04:03.19 | russellb | well let's not talk about it! |
| 04:03.38 | russellb | or i guess you can if you want, i'm going to sleep now :-p |
| 04:03.44 | file | yay |
| 04:03.47 | file | bye bye russell! |
| 04:03.48 | orlok | i'm suprised a volvo had the velocity to flip three times |
| 04:03.51 | coppice | this wasn't a bad wreck. it was a volvo - they all walked away :-) |
| 04:03.57 | brookshire | yay! i agree with file! |
| 04:04.16 | file | yay |
| 04:04.47 | coppice | orlok: you obviously haven't driven a high end volvo. they aren't slow. heck, they made The Saint's car :-) |
| 04:05.47 | orlok | coppice: farkin hemi moite |
| 04:06.59 | jsmith | orlok: I don't know what you're trying to do with this syntax, but it's wrong: exten => 123,1,Dial(SIP/0385121550,0438015327,1) |
| 04:07.15 | orlok | jsmith: thought it might be |
| 04:07.21 | MikeJ[Laptop] | once upon a time there was the little protocol that could... and it got bigger, and bigger, and bigger, and it became SIP! |
| 04:07.26 | orlok | jsmith: i've got two sip channels set up i want to use for outbound dialling |
| 04:07.49 | orlok | i was trying to set it up so 123 would dial 0438015327 via the 0385121550 sip account |
| 04:08.09 | orlok | jsmith: asterisk book is great, but the fact that the start focuses on zaptel stuff is annoying |
| 04:08.13 | orlok | but i didn tpay, so meh :) |
| 04:08.19 | file | MikeJ[Laptop]: it growed up?!? |
| 04:08.28 | MikeJ[Laptop] | yes |
| 04:08.30 | jsmith | SIP/0385121550/0438015327 or SIP/0438015327@ 0385121550 |
| 04:08.32 | file | how cute |
| 04:08.34 | MikeJ[Laptop] | well.. it's a teenager |
| 04:08.35 | orlok | ahh, cool |
| 04:08.42 | file | entered puberty yet? |
| 04:08.47 | MikeJ[Laptop] | use the # |
| 04:08.49 | coppice | why would anyone sane want to filter entered DTMF at source? you can only have 10 digits per second. Its hardly going to be an issue to send anything that is entered |
| 04:09.03 | MikeJ[Laptop] | who filters |
| 04:09.04 | MikeJ[Laptop] | ?? |
| 04:09.28 | coppice | file: well, its certainly already been f**ked up a lot |
| 04:09.38 | file | coppice: very true |
| 04:10.14 | *** join/#asterisk esculapio_ (n=ESCulapi@145stb68.codetel.net.do) |
| 04:10.14 | MikeJ[Laptop] | ererr @ |
| 04:10.45 | esculapio_ | tecnico, hola esta hay |
| 04:10.45 | MikeJ[Laptop] | naptime, or finish the coding I need to do? |
| 04:10.48 | jsmith | orlok: The next edition of the book will have more SIP coverage |
| 04:11.28 | jsmith | orlok: At the time it was written, the SIP support in Asterisk was... well... let's just say it wasn't my favorite channel type. |
| 04:11.39 | esculapio_ | websae, no soy dominicano |
| 04:11.43 | orlok | heh |
| 04:11.49 | websae | que pasa? |
| 04:12.00 | websae | no hablas ingleis? |
| 04:12.06 | orlok | jsmith: we sell high quality broadband, and our broadband supplier is owned by one of the arger pabx (an dslam) manufacturers |
| 04:12.29 | orlok | jsmith: so we have high quality, low latency broadband, plus sip/voip services provided to us by the same company |
| 04:12.29 | esculapio_ | websae, no muy poco |
| 04:12.41 | websae | que necesitas esculapio_? |
| 04:12.46 | orlok | so theres not that much reason to be using ata's |
| 04:13.02 | orlok | hmm. |
| 04:13.09 | *** join/#asterisk shmxtra (n=shmxtra@219.95.158.235) |
| 04:13.09 | orlok | i'm getting "caller id is blocked" |
| 04:13.13 | esculapio_ | websae, mi llamadas en mi asterisk no salen ni entran solo suena ocupado |
| 04:13.28 | esculapio_ | websae, pero tengo comunicacion interna |
| 04:13.42 | esculapio_ | websae, me puedes ayudar |
| 04:13.56 | websae | yo lo veo |
| 04:14.21 | websae | solomente hablo espanol un poco, pero yo atare ayudarte |
| 04:14.23 | orlok | wtf does "caller id is blocked" mean |
| 04:14.26 | Strom_C | por favor, habla ingles aqui o usa los mensajes privitos |
| 04:14.31 | orlok | upstream doesnt like my cid or lack of it? |
| 04:14.39 | orlok | donde esta la pollo |
| 04:14.46 | orlok | me no hablo espanol |
| 04:14.58 | Strom_C | es como afeitando con un gato en tu grabadora |
| 04:15.00 | websae | Strom_C: porque no hablas espanol aqui? |
| 04:15.26 | Strom_C | websae: because most in here don't speak spanish and it's mainly spam and clutter for the majority of users |
| 04:15.40 | websae | Strom_C: necesitamos ayudarlos que necesitan ayudar |
| 04:15.55 | file | it looks like random characters put together to me... ha |
| 04:15.56 | Strom_C | websae: use private messages in that case |
| 04:16.07 | jsmith | esculapio_: Tal vez puede encontrar mas ayuda en #asterisk-es |
| 04:16.19 | Strom_C | or that |
| 04:16.30 | MrDigital | no espanol hablo engles |
| 04:16.56 | jsmith | Strom_C: Neither did I, until about 30 seconds ago |
| 04:17.04 | Strom_C | hah |
| 04:17.17 | asterboy | hablo un poco |
| 04:17.24 | coppice | ah, maybe this KPML isn't so dumb. it allows things like "A # of at least 3 seconds" to be defined. that means effective digit length related input methods can be made useful in limited contexts. |
| 04:18.28 | websae | Well I am glad we can segregat in here :) |
| 04:18.31 | websae | that's good |
| 04:18.38 | MikeJ[Laptop] | what's goto line number in emacs? |
| 04:18.52 | websae | *segregate |
| 04:18.55 | jsmith | MikeJ[Laptop]: CTRL-ALT-OpenApple-Meta-ALSKDFJA |
| 04:19.00 | file | Esc+X goto-line |
| 04:19.10 | coppice | MikeJ: something just as obscure as in vi, of course |
| 04:19.24 | jsmith | coppice: Naw, it's easier in vi |
| 04:19.28 | MikeJ[Laptop] | thank you |
| 04:19.40 | *** part/#asterisk downunder33 (n=downunde@219.95.158.235) |
| 04:20.05 | coppice | I love these vi and emacs wars. the participants seem blissfully unaware they suck equally :-) |
| 04:20.10 | MikeJ[Laptop] | not working |
| 04:20.21 | file | hit Esc, then X, then type in goto-line and hit enter |
| 04:20.23 | coppice | you didn't add pixie dust |
| 04:20.25 | file | then enter the line number and hit enter |
| 04:20.56 | tainted- | coppice how dare u reveal the truth |
| 04:21.05 | sevard | i can't stand emacs at all and vi hobbels along, i'd much rather use pico or nano. |
| 04:21.12 | file | I don't care what people use for a text editor... use what works |
| 04:21.25 | MikeJ[Laptop] | not working :( |
| 04:21.30 | Strom_C | sevard: but but but vi is the lubeless anal rape you grow to love! |
| 04:21.32 | sevard | i use pico and nano because i don't want to do fancy fucking things with my editor, i just want to edit a file. |
| 04:21.48 | file | MikeJ[Laptop]: :( |
| 04:21.56 | MikeJ[Laptop] | just says no match |
| 04:22.00 | tainted- | u text editor trolls |
| 04:22.01 | sevard | Strom_C: That's a whole other ballgame. |
| 04:22.03 | file | for goto-line? |
| 04:22.05 | coppice | vi is far more useful than emacs, only because its the editor that's always there. however "its always there" is also the only thing Windows has going for it |
| 04:22.20 | Corydon76-home | Emacs is a nice operating system, but it lacks a decent editor. |
| 04:22.29 | file | the hate, the hate! |
| 04:22.36 | MikeJ[Laptop] | I hit escape and x, it says M-x 123 |
| 04:22.37 | sevard | Corydon76-home: i've heard that one, it's great. |
| 04:22.39 | file | next you'll be telling me Linux sucks |
| 04:22.43 | MikeJ[Laptop] | I hit enter, it says no match |
| 04:22.51 | file | MikeJ[Laptop]: clear the 123 |
| 04:22.56 | file | and type in goto-line |
| 04:23.07 | coppice | I wish someone would add key stroke recording to SciTE. |
| 04:23.22 | Corydon76-home | I tend to threaten Emacs users with making it their shell. |
| 04:23.25 | sevard | I use pico and nano becaues software should be easy to run. I don't want to open a emacs or vi reference book that weighs more than a small child to find the command to jump to a line. i just want to edit a fucking file. |
| 04:23.27 | MikeJ[Laptop] | oh that's dumb |
| 04:23.44 | Qwell | Corydon76-home: threaten them with making it their editor |
| 04:24.51 | coppice | the sentence for copyright infringement should be something like entering the bible from a printed copy by emacs. |
| 04:25.05 | Corydon76-home | I would use emacs, as it was the first editor on Unix I was ever taught. However, I prefer having enough memory left over that my kernel doesn't need to be swapped out of memory. |
| 04:25.39 | rdgzt | Corydon-w: So, how's life with a computer with 16 megs of mem these days? |
| 04:25.39 | PMantis | file, Unanymous vote? |
| 04:25.46 | file | darn right |
| 04:25.55 | coppice | slipping into bed sounds very keystone cops |
| 04:26.09 | file | coppice: wha? |
| 04:26.28 | PMantis | heh, speled with an 'i'... I knew that didn't look right. |
| 04:27.03 | file | yay bed |
| 04:27.08 | coppice | when emacs was written, 16 megs was a luxury very few had |
| 04:29.29 | sevard | 16 megs still is. |
| 04:29.57 | sevard | OMFG HI |
| 04:31.05 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:34.51 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 04:35.05 | *** join/#asterisk melte (n=melisate@219.95.158.235) |
| 04:35.17 | brookshire | HIHIHIH! |
| 04:37.08 | coppice | sevard: yeah. if you want anything as small as 16M, its a specialist product at a high price :-) |
| 04:37.11 | MikeJ[Laptop] | kram, how many sip dtmf methods can YOU name! |
| 04:37.12 | *** part/#asterisk KaBewM (n=DA-MAN@66-215-7-106.dhcp.psdn.ca.charter.com) |
| 04:39.07 | *** join/#asterisk melt1 (n=melisa@219.95.158.235) |
| 04:41.33 | *** part/#asterisk melt1 (n=melisa@219.95.158.235) |
| 04:41.36 | coppice | MikeJ: do combinations count? like one digit by RC2833, the next by NOTIFY, etc? :-) |
| 04:42.37 | MikeJ[Laptop] | heh |
| 04:43.14 | MikeJ[Laptop] | I wonder how bad you could confuse a UA by sending overlapping 2833 digits |
| 04:43.56 | MikeJ[Laptop] | ok.. sleepy time for me.. |
| 04:44.24 | coppice | have you seen how easy it is to confuse most ATAs? hard lockup seems their usual way to get pissed off :-) |
| 04:47.03 | Corydon76-home | They must not be coded in ADA... |
| 04:47.51 | coppice | that's right. they absolutely must not under any circumstances be coded in ADA |
| 04:57.49 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 04:58.40 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 05:00.36 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 05:09.14 | websae | it always gets quiet right around now |
| 05:09.28 | Zeeek | shhhhhhh! |
| 05:09.30 | websae | I suppose everyone needs a nap once in awhile |
| 05:09.38 | Zeeek | keep it down! |
| 05:09.47 | websae | I'll keep |
| 05:09.48 | websae | my |
| 05:09.48 | Zeeek | or get it up, if that's your thing ;) |
| 05:09.50 | websae | sentences |
| 05:09.51 | websae | short |
| 05:10.08 | websae | lol |
| 05:10.15 | websae | how's it going Zeeek? |
| 05:10.17 | websae | where you from? |
| 05:10.38 | coppice | lunch seems just the thing right now |
| 05:10.38 | Zeeek | Minneapolis, MN, USA |
| 05:10.52 | Zeeek | lunch? I just had my first coffee |
| 05:14.11 | Corydon76-home | So, is it cold or hot in Mpls? |
| 05:15.12 | websae | Milwaukee, WI here |
| 05:15.39 | Zeeek | Actually I'm in Paris now. But my son says it never gets cold in the winter anymore |
| 05:16.05 | websae | it's about 50ish here |
| 05:16.07 | websae | in Wisconsin |
| 05:16.16 | websae | Zeeek, what are you doing in Paris? |
| 05:16.33 | Zeeek | I've lived here for the last 25 years working on asterisk :) |
| 05:16.36 | CunningPike | Last tango |
| 05:16.43 | CunningPike | Sorry - couldn't resist |
| 05:16.45 | *** join/#asterisk wenko (n=wenko@142.232.8.200) |
| 05:16.46 | websae | we just started offering DIDs from Paris, France |
| 05:16.55 | websae | oooo |
| 05:16.59 | Zeeek | I invented asterisk and gave the idea to Mark in 1980 |
| 05:17.08 | websae | haha |
| 05:17.22 | Zeeek | oh, what's your Paris offer? |
| 05:17.31 | Zeeek | I have a few already (testing) |
| 05:17.52 | Zeeek | I have an interesting logistic problem I'm working on now |
| 05:17.56 | websae | $13 USD/month DID and unlimited |
| 05:18.11 | Zeeek | You want to kknow what the competition does? |
| 05:18.19 | websae | they blow it away :) |
| 05:18.26 | Zeeek | naw, not really |
| 05:18.32 | websae | good thing I don't compete with them in the U.S. |
| 05:18.32 | coppice | websae: 50ish? wow, that's desert temperature :-) |
| 05:18.36 | Zeeek | Wengo €7 unlimited |
| 05:18.44 | websae | coppice, where are you from? |
| 05:19.31 | coppice | from? from about 10,000km from here :-) |
| 05:20.45 | websae | oh |
| 05:20.47 | websae | fantastic |
| 05:21.17 | websae | 12:21AM Monday, May 01, 2006 |
| 05:21.26 | Zeeek | this is like ham radio all over again |
| 05:21.43 | websae | that it is |
| 05:21.46 | coppice | 01:21PM Monday, May 01, 2006 |
| 05:21.59 | websae | you're only a timezone away hehehe |
| 05:22.09 | websae | well quite a few timezones away |
| 05:22.11 | websae | are you in AU? |
| 05:22.20 | Cardoe | So what's the advantage of AEL over the conf files? |
| 05:22.28 | coppice | nope. .au would be 03:21PM |
| 05:22.31 | Cardoe | are the conf files going away? should I go with AEL? |
| 05:22.34 | Zeeek | you can pretend you're programming in C |
| 05:23.03 | coppice | lots of people pretend they are programming in C |
| 05:23.05 | websae | and be really cool then |
| 05:23.14 | jsmith | Cardoe: No, the conf files aren't going away |
| 05:23.29 | Zeeek | I never pretended to program in C. I kludge a lot though |
| 05:25.22 | Cardoe | so basically don't bother with ael |
| 05:26.50 | jsmith | Cardoe: You have your choice on whether to create your dialplan in AEL or the old style. Internally, AEL gets translated back to the old style, so it's no going away. |
| 05:26.57 | jsmith | Cardoe: In short, it's about choice. |
| 05:27.57 | coppice | time for lunch, then off to disneyland. turn our noses up an disneyland, and play on the lake next door to it :-) |
| 05:28.10 | *** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 05:29.55 | websae | I want lunch |
| 05:34.13 | Zeeek | hey kram, randy here |
| 05:35.05 | *** join/#asterisk Ciber311 (n=Ciber@user-1087e94.cable.mindspring.com) |
| 05:35.59 | Strom_C | nothing makes your mind work hard quite like speaking a language you learned in school but never spoke conversationally in the first place |
| 05:38.00 | Zeeek | what mind? What is work? What is school? |
| 05:38.32 | coppice | Strom_C actually, that just makes my mind switch off |
| 05:38.54 | Strom_C | haha |
| 05:38.57 | Strom_C | yeah, I know what you mean |
| 05:39.01 | Strom_C | you forget simple things |
| 05:39.08 | Zeeek | the first time I actually had a conversation in a foreign language, I was nearly drunk |
| 05:39.17 | Zeeek | it flowed better that way |
| 05:39.33 | websae | Zeeek: they always do |
| 05:39.48 | websae | Even when you speak your native language as well sometimes |
| 05:40.03 | Zeeek | in fact I had a very funny experience related to that night that I think I shall regale you all with |
| 05:40.37 | coppice | the first time I ever needed a foreign language in earnest, it was one my school would never have considered teaching |
| 05:41.07 | Zeeek | My "business" colleague, a technician, accepted the late night hotel clerk's offer of finding "a girl" (at like 2AM) |
| 05:41.40 | Zeeek | We were sitting in the bar and she shows up and sits down next to me and says "so what's your sign?" |
| 05:42.05 | Zeeek | I had to say "Oh it wasn't me that called, it was my friend" |
| 05:42.48 | Zeeek | She: "so um, what do we do?" Me: I'm going to get up and you'll take my seat next to him." And that's how my French career started |
| 05:53.06 | *** join/#asterisk naS_- (n=andrew@182.136.233.220.exetel.com.au) |
| 05:53.25 | *** join/#asterisk codebreaker (n=codebrea@xserver.flexserv.de) |
| 05:56.07 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
| 05:56.08 | rdgzt | Typical reason for ztcfg saying no such device or address? |
| 05:56.26 | Zeeek | driver not loaded |
| 05:56.33 | rdgzt | Using a TDM400P with 4 FXOs. |
| 05:56.40 | rdgzt | That should be the tor2 driver, right? |
| 05:56.47 | wasim | wctdm |
| 05:56.53 | Zeeek | wasim, my man |
| 05:56.59 | wasim | tor2 is the old isa card |
| 05:57.05 | wasim | bonjour monsieur Zeeek |
| 05:57.10 | Zeeek | comment va? |
| 05:57.14 | wasim | bien |
| 05:57.45 | rdgzt | Ok, modprobing wctdm didn't give any error messages, but neither did it seem to change anything. |
| 05:57.57 | wasim | ztcfg -vvvvvvvvvvvv |
| 05:58.41 | rdgzt | Tells me the channel map with four channels, that look ok, and then says |
| 05:58.42 | rdgzt | 4 channels configured. |
| 05:58.42 | rdgzt | ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 05:59.02 | wasim | what does dmesg show |
| 05:59.26 | rdgzt | Am I looking for anything specific here? |
| 05:59.52 | rdgzt | [ 45.812423] Zapata Telephony Interface Registered on major 196 |
| 05:59.53 | rdgzt | [ 45.812510] Zaptel Version: 1.2.5 Echo Canceller: KB1 |
| 05:59.53 | rdgzt | [ 46.839809] Registered Tormenta2 PCI |
| 05:59.53 | rdgzt | [ 47.130273] usbcore: registered new driver wcusb |
| 05:59.53 | rdgzt | [ 47.130344] Wildcard USB FXS Interface driver registered |
| 06:00.02 | wasim | and your zaptel.conf should have fxsks=1-4 |
| 06:00.02 | L|NUX | wasim : bro i need your little help :) |
| 06:00.03 | rdgzt | Is the only thing Zaptel-related I can see. |
| 06:00.09 | rdgzt | wasim: That it does. |
| 06:00.11 | wasim | L|NUX: what happened |
| 06:00.36 | L|NUX | wasim : bro see i have senrio i hope you can understand me better :) |
| 06:01.19 | L|NUX | wasim : suppose if some one call me on extension e.g i have this in my extensions.conf exten => 1,1,Dial(SIP/1,20,tr) |
| 06:01.55 | L|NUX | wasim : when its time out then it will Goto new context [email] and execute systemcommand :) |
| 06:02.26 | wasim | L|NUX: only if 1,2,Goto(email|exten|prio) |
| 06:02.31 | L|NUX | i tried |
| 06:02.35 | L|NUX | but in system |
| 06:02.55 | L|NUX | wait |
| 06:03.44 | wasim | rdgzt: you have neither usb nor tor2 |
| 06:03.49 | L|NUX | i am using this in [email] context |
| 06:03.49 | L|NUX | exten => s,1,SetVar(CALLFILENAME=/var/spool/asterisk/monitor/${TIMESTAMP}) |
| 06:03.49 | L|NUX | exten => s,2,System(/bin/echo -e "'Incoming call from : ${CALLERID} \\r Received: ${DATETIME}'" | /bin/mail -s "'Phone call'" f4fahmed@gmail.com) |
| 06:03.54 | L|NUX | and its not working :( |
| 06:04.13 | rdgzt | So that still doesn't work. |
| 06:04.15 | rdgzt | That's really weird. |
| 06:05.45 | L|NUX | any idea wasi |
| 06:05.48 | L|NUX | wasim bro |
| 06:06.07 | wasim | L|NUX: build the command a single step at a time |
| 06:07.05 | L|NUX | well this thing is working |
| 06:07.05 | L|NUX | like |
| 06:07.17 | L|NUX | if put it in [sip] context like this |
| 06:07.21 | L|NUX | i remove s |
| 06:07.24 | L|NUX | and add only 1 |
| 06:07.27 | L|NUX | and its working |
| 06:07.51 | wasim | ofcourse, 1,2 != s,1 |
| 06:08.18 | L|NUX | O_o |
| 06:08.32 | L|NUX | so i will have to use s,3,setVar.... |
| 06:09.38 | rdgzt | Modules seem to be loaded correctly, but ztcfg still tells me the device doesn't exist. |
| 06:09.41 | rdgzt | Any other ideas? |
| 06:09.53 | wasim | L|NUX: no, 1,2 ... where is CALLFILENAME being used? |
| 06:10.40 | L|NUX | humm |
| 06:10.50 | wasim | [4294681.327000] Zapata Telephony Interface Registered on major 196 |
| 06:10.50 | wasim | [4294681.351000] ACPI: PCI Interrupt 0000:06:02.0[A] -> GSI 18 (level, low) -> IRQ 18 |
| 06:10.53 | wasim | [4294681.351000] Freshmaker version: 71 |
| 06:10.56 | wasim | [4294681.352000] Freshmaker passed register test |
| 06:10.58 | wasim | [4294682.052000] Module 0: Installed -- AUTO FXO (FCC mode) |
| 06:11.02 | wasim | rdgzt: you should get 0,1,2,3 moudles like that |
| 06:11.03 | L|NUX | so its means i have to logged call file in [sip] context |
| 06:11.35 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 06:11.37 | rdgzt | wasim: That's weird, so what does it mean if I don't get that? |
| 06:11.49 | wasim | busted card, bad PCI slot, lots of possibilities |
| 06:12.20 | rdgzt | Nah, this card worked earlier today, but I uninstalled zaptel and asterisk stuff that I had gotten from Ubuntu packages, and decided to build from source, to solve another problem. |
| 06:12.28 | rdgzt | So I'm pretty sure the hardware's fine. |
| 06:12.47 | rdgzt | I haven't changed anything hw-related. |
| 06:12.52 | wasim | ok, did you make install |
| 06:12.55 | rdgzt | Yes. |
| 06:13.21 | wasim | and lspci shows 06:03.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
| 06:13.37 | rdgzt | Yup. |
| 06:13.54 | wasim | <PROTECTED> |
| 06:13.54 | wasim | zaptel 187140 59 wctdm |
| 06:13.58 | rdgzt | Well, it says "communication controller", strictly speaking, but apart from that, yes. |
| 06:14.30 | rdgzt | It seems it loaded all the modules on boot, actually, for some reason, so yes, zaptel, wctdm, and many more. |
| 06:15.23 | rdgzt | rmmodding all the modules and then just modprobing zaptel and wctdm and leaving out all the others makes no difference. |
| 06:18.43 | *** join/#asterisk Assid (n=assid@203.115.64.12) |
| 06:19.04 | Assid | yello |
| 06:21.40 | codebreaker | if my $database-server is offline. will asterisk then cache the cdr-logs until this server is back online or will they got lost? |
| 06:21.40 | Zeeek | exit |
| 06:21.43 | Zeeek | su |
| 06:21.46 | Zeeek | when4fsre6 |
| 06:21.56 | Zeeek | Oh, nooooooo |
| 06:22.17 | codebreaker | rooted :) |
| 06:22.26 | Assid | hahh |
| 06:23.14 | jql | well, it'll be on google by tomorrow |
| 06:23.19 | jql | chop, chop |
| 06:26.23 | *** join/#asterisk BugKham (n=BugKham@125.24.6.174) |
| 06:27.24 | Assid | anyone managed to get their sipbroker working |
| 06:27.30 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 06:31.16 | Assid | isnt that message supposed to come when someone messages him ? |
| 06:32.20 | wasim | Assid: no, it won't cache the logs for the db |
| 06:32.57 | Assid | huh? |
| 06:33.06 | Assid | no no.. i was talking about krams' away |
| 06:35.08 | wasim | oh sorry, s/Assid/codebreaker |
| 06:36.01 | Assid | you ever used sipbroker? |
| 06:41.47 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
| 06:44.15 | Assid | seriously.. no one? |
| 06:46.53 | tainted- | Assid everyone's asleep |
| 06:46.59 | tainted- | try again in a few hours |
| 06:47.34 | *** join/#asterisk k-man (n=jason@unaffiliated/k-man) |
| 06:47.38 | k-man | hello |
| 06:48.00 | codebreaker | wasim: thanks. so its better to have a local postgres running and feed my "important" master via fetch :) then i will nly loose logs when the local db is down. but htne i shutdown the hole node. |
| 06:48.12 | Zeeek | "I'm sorry, your answer is unavailable at this time. Please try again later" Dahp. Dahp. Dahp. |
| 06:48.19 | k-man | how does one ensure availability of ADSL so that you don't have phone lines going down when using voip? |
| 06:48.36 | tainted- | k-man are u having downtime? |
| 06:48.42 | k-man | no |
| 06:48.47 | k-man | well |
| 06:48.48 | tainted- | Zeeek why are u getting that? |
| 06:48.50 | Zeeek | k-man we have two ADSL lines |
| 06:48.51 | k-man | not yet |
| 06:48.58 | k-man | i want to get some voip lines |
| 06:49.11 | tainted- | this must be 2 adsl line night |
| 06:49.14 | k-man | so i want to upgrade our ADSL to ensure it never goes down |
| 06:49.15 | Zeeek | really |
| 06:49.23 | tainted- | u can't |
| 06:49.24 | k-man | currently we get downtime every so often |
| 06:49.26 | k-man | not sure why |
| 06:49.34 | Zeeek | well SDSL is a step in that direction |
| 06:49.40 | tainted- | what sort of downtime |
| 06:49.42 | k-man | oh? |
| 06:49.45 | k-man | sounds expensive |
| 06:49.47 | tainted- | power outage |
| 06:49.53 | k-man | no |
| 06:49.56 | Zeeek | you need a "professional" contract. Way more expensive |
| 06:50.05 | tainted- | lol |
| 06:50.10 | k-man | i just know that about 1-2 times a week our network goes down |
| 06:50.17 | tainted- | sounds like a secret handshake |
| 06:50.18 | k-man | i have no idea if its isp or hardware related |
| 06:50.20 | Zeeek | network or connection? |
| 06:50.25 | codebreaker | k-man: but this wont help if your adsl-hardware-providing place goes down. you have two providers but they both use the same hardware |
| 06:50.26 | tainted- | k-man how far from central office are u |
| 06:50.45 | k-man | um.... 1-2kms |
| 06:50.55 | tainted- | what country |
| 06:51.01 | k-man | australia |
| 06:51.14 | Zeeek | koalas on the line |
| 06:52.04 | Zeeek | actually it depends on the cause, codebreaker. When our line goes off, the other is still on |
| 06:52.22 | Zeeek | I chose to put astrisk on the one that doesn't go down often |
| 06:52.33 | Zeeek | ironically, it's the cheaper consumer line |
| 06:54.08 | Zeeek | so... nufone... |
| 06:58.06 | *** join/#asterisk Defraz (n=t0tal@24-119-94-19.cpe.cableone.net) |
| 06:59.15 | k-man | ok |
| 06:59.37 | k-man | so do isps have some sort of rating in terms of reliability and also, what about modem reliability? |
| 06:59.43 | k-man | are some ADSL modems cheaper than others? |
| 06:59.49 | k-man | cheaper? |
| 06:59.52 | k-man | i meant better |
| 07:00.03 | k-man | in terms of their reliability |
| 07:00.16 | wasim | but ofcourse |
| 07:00.18 | dlynes_ | k-man: 3com's are good |
| 07:00.29 | dlynes_ | k-man: 3com office connect |
| 07:01.35 | Zeeek | http://www.dslreports.com/forum/voip |
| 07:01.50 | Zeeek | k-man you know about that site? |
| 07:02.01 | dlynes_ | Also, there's a difference in the dsl providers...there's the telcos that don't do anything about all the viruses, spyware, worms, spambots, ... on their networks...then there's other dsl providers that'll shut customers off with that kinda crap abusing their network |
| 07:02.10 | k-man | nothing |
| 07:03.12 | k-man | oh, no, i don't know about it |
| 07:03.38 | Zeeek | it has a lot of consumer comments about all DSL (in the uS) |
| 07:03.44 | k-man | oh |
| 07:03.45 | k-man | ok |
| 07:03.48 | k-man | we hvae something like that here |
| 07:03.53 | k-man | whirlpool.net.ayu |
| 07:03.57 | k-man | .au |
| 07:04.05 | Zeeek | but it might have useful info even outside the us. It has speed test for example |
| 07:04.21 | Zeeek | yeah whirlpool is actually pretty good |
| 07:04.58 | dlynes_ | yeah...telstra probably blows though |
| 07:05.04 | dlynes_ | it's the national telco |
| 07:05.11 | dlynes_ | optus probably blows too |
| 07:07.29 | Zeeek | hey check this out: http://www.dslreports.com/shownews/74008 |
| 07:07.38 | Zeeek | about the king of spammers |
| 07:08.58 | k-man | yeah, telstra is crap |
| 07:09.07 | k-man | and they are expensive |
| 07:09.14 | k-man | and have a monopoly on the local loop |
| 07:14.22 | Assid | damn |
| 07:14.51 | hads|home | k-man: I thought au had unbundled? |
| 07:14.59 | Assid | i dunno what extension should i make to handle this incoming sip uri |
| 07:15.00 | k-man | nafaik |
| 07:15.37 | Assid | if i have a assid@blah.org .... what kinda extension do you need for handlng that? |
| 07:16.18 | Zeeek | how about exten => assid,1,Dial(yer_phone) |
| 07:16.38 | Zeeek | in a guest or default context |
| 07:17.47 | Assid | i gotta go that for every sip extension ?!?!? |
| 07:17.51 | Assid | err.. do.. |
| 07:17.58 | orlok | k-man: upgrading your dsl doesnt protect against backhoe fade |
| 07:18.09 | k-man | backhoe fade? |
| 07:18.11 | k-man | whats that? |
| 07:18.14 | orlok | k-man: oh, you know telstra. |
| 07:18.16 | orlok | an aussie |
| 07:18.24 | Zeeek | Assid no you could just accept all names |
| 07:18.26 | orlok | backhoe fade.. no sync/dialtone |
| 07:19.02 | k-man | oh |
| 07:19.03 | orlok | k-man: you need redundant connections basically |
| 07:19.09 | k-man | oh |
| 07:19.12 | k-man | i see |
| 07:19.25 | k-man | i was thinking that it might be worth getting a seperate ADSL connection just for voip |
| 07:19.26 | orlok | yeah, preferably of different types, |
| 07:19.33 | k-man | and keeping other traffic off it |
| 07:19.36 | orlok | theres lots of different faults |
| 07:19.37 | Assid | Zeeek: so how would that be ? s,1,Dial(s/Local) ? |
| 07:19.44 | k-man | yeah |
| 07:20.14 | orlok | k-man: like, dslam line cards can die, somebody can use the pair for something else, the isp can have an upstream fault |
| 07:20.21 | Zeeek | Assid you could do something like _XXXXX. to keep it a minimum number of characters |
| 07:20.30 | k-man | yeah |
| 07:20.34 | orlok | so you want a different sort of media through another provider, ideally |
| 07:20.37 | Zeeek | otherwise anything at all will ring phones |
| 07:21.01 | Zeeek | but if you wanted assid@ to ring one phone and farida@ another, you'd need the actual extensions |
| 07:21.03 | Assid | damn.. my user is spherelinx-satish |
| 07:22.50 | *** join/#asterisk BoRiS (i=boris@S010600112f38a61e.wp.shawcable.net) |
| 07:23.19 | Zeeek | and? |
| 07:23.46 | Zeeek | guys I had the worst and wierdest problem last year with one of our DSL lines |
| 07:24.10 | Zeeek | it would retain sync but pass no traffic for 10-20 seconds *exactly* evey 8 minutes |
| 07:24.38 | Zeeek | Every time I made a call I had to tell people if I disappear wait 2à seconds and I'll be back! |
| 07:24.54 | Zeeek | that went on for 3 weeks before they fixed it |
| 07:25.04 | Zeeek | I'm positive it was in the dslam |
| 07:25.14 | *** join/#asterisk bzbw (n=wlwzhang@68-190-223-129.dhcp.mtpk.ca.charter.com) |
| 07:27.02 | BoRiS | Anyone has an updated IPP g729 patch that works with the latest cvs changes? |
| 07:27.18 | Assid | Zeeek: i have it wanting to go to default context for the calls.. anyway to force it to use another context? |
| 07:32.03 | Assid | Zeeek: also.. how would you handle different context (vhost) |
| 07:33.22 | *** join/#asterisk ramo (n=ramo@59.92.137.242) |
| 07:38.41 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 07:50.14 | Zeeek | Assid I wonder what ${EXTEN} contains when it arrives at the _XX. extension? I think I'll try that when I get a second |
| 07:51.17 | codebreaker | can i do something if phone 123 connects tell it to use serverB instead to connect. also something like a redirect? |
| 07:51.54 | *** join/#asterisk key2 (n=key2@gob75-2-81-56-64-17.fbx.proxad.net) |
| 07:52.17 | key2 | !seen kram |
| 08:02.44 | Zeeek | see 'switch' |
| 08:04.20 | *** join/#asterisk kamileon (n=kamileon@68.62.190.253) |
| 08:04.48 | Assid | Zeeek: it takes the extension from before the @ |
| 08:04.59 | Assid | or... if you use sipbroker.. its the one after the broker code |
| 08:06.27 | Assid | brb |
| 08:08.18 | *** part/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 08:20.39 | Zeeek | yes it works fine |
| 08:21.10 | Zeeek | you'd need to have an address like user_domain@domain.tld |
| 08:23.13 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:27.01 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 08:30.32 | *** join/#asterisk alexbr (n=alexbr@adsl-ull-83-239.47-151.net24.it) |
| 08:31.17 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 08:33.28 | *** join/#asterisk Alystair (n=bob@CPE001109c15241-CM00407b8794db.cpe.net.cable.rogers.com) |
| 08:33.47 | *** part/#asterisk alexbr (n=alexbr@adsl-ull-83-239.47-151.net24.it) |
| 08:36.59 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 08:37.14 | *** join/#asterisk UlbabraB (n=caplaz@host241-43.pool8172.interbusiness.it) |
| 08:46.54 | *** join/#asterisk naS_- (n=andrew@182.136.233.220.exetel.com.au) |
| 08:47.31 | *** join/#asterisk RoyK (n=roy@ti211210a080-0268.bb.online.no) |
| 08:57.20 | naS_- | anyone able to help with g729 pass through trunking using IAX2? |
| 09:04.59 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 09:15.17 | Assid | Zeeek: supose i have assid@abc.com and assid@xyz.com .. but i want 2 different phones to ring.. both on the same * box |
| 09:16.13 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 09:27.49 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 09:28.59 | *** join/#asterisk alexbr (n=alexbr@adsl-ull-83-239.47-151.net24.it) |
| 09:29.09 | alexbr | hi all |
| 09:30.17 | alexbr | is it possible to use asterisk with a standard isdn card? |
| 09:31.13 | jql | yes |
| 09:31.28 | alexbr | and with an analogic modem? |
| 09:32.40 | *** join/#asterisk r0d3nt|m (i=r0d3nt@tinfoilhat.net) |
| 09:33.41 | jql | no clue. it's supposed to work with anything supported by the standard linux isdn drivers |
| 09:33.58 | alexbr | ok, thanks a lot |
| 09:34.28 | alexbr | and do you know about any localized documentation? |
| 09:34.57 | jql | I wouldn't know where to find that |
| 09:35.39 | *** join/#asterisk RoyK (n=roy@ti211210a080-0268.bb.online.no) |
| 09:38.24 | *** join/#asterisk Tili (i=Tili@202.133.67.36) |
| 09:39.04 | *** join/#asterisk ToTo (n=ToTo@host187-131.pool872.interbusiness.it) |
| 09:40.49 | *** join/#asterisk ghost99 (n=neville@222-152-219-77.jetstream.xtra.co.nz) |
| 09:46.19 | *** join/#asterisk ghenry (n=ghenry@195.38.86.72) |
| 09:47.20 | ghenry | for sip and NAT, I only need to have ports 5060 and 5061 UDP forwarded to a particular client? I can't get it to work any other way |
| 09:48.54 | *** part/#asterisk alexbr (n=alexbr@adsl-ull-83-239.47-151.net24.it) |
| 09:55.16 | tainted- | what do u mean |
| 09:55.19 | tainted- | was that a question? |
| 10:00.45 | mut | does the zaptel echo can control hw echo can on the sangoma cards |
| 10:00.45 | mut | ? |
| 10:01.35 | ghenry | tainted-: sorry. that was a question |
| 10:03.59 | ghenry | everywhere I have read says onlt port UDP 5060 |
| 10:04.07 | ghenry | but it doesn't work without 5061 too |
| 10:04.16 | ghenry | or is that wrong? |
| 10:07.49 | hwt | does astertest work with more recent versions (1.2.x) of *? |
| 10:08.09 | hwt | or are there other, perhaps more advanced tools, available? |
| 10:08.41 | tainted- | ghenry it's wrong |
| 10:08.57 | ghenry | I don't understand then |
| 10:09.03 | tainted- | mut yes sangoma has echo contol |
| 10:09.15 | tainted- | hwt what is astertest |
| 10:09.29 | tainted- | ghenry udp works on whatever port u want it to |
| 10:09.30 | ghenry | I have opened up ports 5060 UDP, and could reg, but asterisk couldn't sip_poke |
| 10:09.34 | mut | ... |
| 10:09.42 | ghenry | needed to open up 5061 |
| 10:09.44 | mut | yes, does zaptel control the echo cancel on it tho |
| 10:09.45 | tainted- | ghenry u want 5060 TCP |
| 10:09.53 | mut | or is there a seperate sangoma util for it |
| 10:09.59 | ghenry | asterisk can't do TCP for SIP |
| 10:10.00 | tainted- | and a range of ports in UDP for rtp to go through |
| 10:10.09 | tainted- | ghenry wrong |
| 10:10.16 | tainted- | where the fuck are u reading that |
| 10:10.20 | ghenry | so the book and every other docs I read |
| 10:10.30 | *** join/#asterisk rkr245 (n=ravi@office.callsat-telecom.com) |
| 10:10.36 | ghenry | O'Reilly, voip-info.org etc. |
| 10:10.37 | tainted- | u mean the SIP signalling or the audio |
| 10:10.47 | mut | asterisk only does udp sip.. |
| 10:10.52 | mut | for signalling |
| 10:10.54 | tainted- | u don't want the audio stream in tcp |
| 10:11.08 | mut | and rtp is udp as well |
| 10:11.10 | ghenry | I don't know. That's what I am asking. Newbie ;-) |
| 10:11.11 | tainted- | signalling = tcp, audio = udp |
| 10:11.15 | ghenry | ah |
| 10:11.15 | mut | no |
| 10:11.17 | mut | it's udp |
| 10:11.18 | hwt | tainted-: http://www.asteriskguru.com/tutorials/astertest.html |
| 10:11.29 | mut | asterisk doesn't do tcp SIP |
| 10:11.35 | ghenry | Couldn't find a Firewall hint sheet for ports etc. |
| 10:11.38 | ghenry | Thought so |
| 10:11.39 | tainted- | mut block tcp 5060 and see if u can receive calls |
| 10:11.47 | mut | it is |
| 10:11.50 | mut | and i do |
| 10:11.53 | hwt | mut: it will soon, though. |
| 10:11.58 | ghenry | 1.4 will |
| 10:12.00 | ghenry | I read |
| 10:12.07 | hwt | ghenry: yup. |
| 10:12.11 | hwt | ghenry: to support sip-tls. |
| 10:12.17 | naS_- | does anybody know much about IAX2 trunking? |
| 10:12.18 | ghenry | cool |
| 10:12.28 | ghenry | So, firewall guide/howto? |
| 10:12.53 | mut | all you should need open is 5060 |
| 10:12.55 | mut | udp |
| 10:13.09 | mut | and whatever ports you're using for audio/rtp |
| 10:13.10 | ghenry | If every client softphone needs port 5060, how can you have more than 1 client behind a NAT router? |
| 10:13.17 | tainted- | where does it say asterisk SIP signalling is 5060 UDP |
| 10:13.27 | Zeeek | you use 5061 for the second client |
| 10:13.29 | ghenry | Not sure mut. I think sip.conf says 8000 |
| 10:13.36 | ghenry | ah Zeeek |
| 10:13.41 | ghenry | So, start again. |
| 10:13.53 | Zeeek | no change to asterisk, just the client |
| 10:14.08 | Zeeek | domain.name:5061 |
| 10:14.13 | ghenry | I can only register my client sooftphone when ports 5060 and 5061 UDP point to the same client |
| 10:14.24 | ghenry | that's not right |
| 10:14.29 | Zeeek | not normal |
| 10:14.42 | ghenry | that's the port it sends the OPTIONS back on |
| 10:15.00 | Zeeek | oh, I just remembered why my laptop IAX wasn't working |
| 10:15.25 | Zeeek | 4569 is forwarded - what an idiot |
| 10:16.02 | tainted- | http://www.voip-info.org/tiki-index.php?page=Asterisk+firewall+rules |
| 10:16.07 | tainted- | i stand corrected |
| 10:16.13 | tainted- | 5060 udp |
| 10:16.27 | tzanger | tainted-: jeez you're up early |
| 10:16.51 | tainted- | tzanger coding the night away |
| 10:17.04 | tainted- | tzanger must be 5-6am where u are |
| 10:17.07 | tzanger | yep |
| 10:17.07 | mut | tzanger: you have any idea? sangoma hw echo can controlled by zaptel |
| 10:17.09 | ghenry | Thanks tainted- |
| 10:17.10 | tzanger | 6:17am |
| 10:17.14 | mut | or is there a sang util |
| 10:17.23 | tzanger | mut: I don't think so, I tihnk it's handled by the hwectools |
| 10:17.28 | ghenry | anyone integrated Asterisk with a Avaya VoIP existing PBX? |
| 10:17.52 | mut | know what they are? |
| 10:18.09 | tzanger | it's a tarball that gets applied ot the sangoma drivers from the sangoma ftp site |
| 10:18.26 | ghenry | hmm tainted- doesn't mention UDP port 5061 |
| 10:18.35 | ghenry | so not sure why it doesn't work without that open |
| 10:18.57 | tainted- | ghenry what do u need it to work on? |
| 10:19.07 | tainted- | what clients are u using |
| 10:19.11 | tainted- | softphones, hardphones? |
| 10:19.14 | *** join/#asterisk kavit (n=kavit@210-84-40-39.dyn.iinet.net.au) |
| 10:19.17 | ghenry | I just testing a basic sip.conf with ekiga |
| 10:19.47 | ghenry | Can only register and make a call when 5061 UDP and 5060 UDP is open |
| 10:21.08 | tainted- | what is ekiga |
| 10:21.21 | ghenry | gnomemeeting renamed |
| 10:21.40 | ghenry | very nice |
| 10:22.00 | *** join/#asterisk AsteriskAlbania (n=info@217.24.244.130) |
| 10:22.28 | AsteriskAlbania | ASTERISK & RADIUS any information will be appriciated |
| 10:23.10 | tainted- | ghenry can u configure ekiga to different audio ports? |
| 10:23.19 | ghenry | aye |
| 10:23.26 | ghenry | anything you like |
| 10:24.19 | *** join/#asterisk lucifr (n=chatzill@c-24-126-108-87.hsd1.ca.comcast.net) |
| 10:24.26 | lucifr | Hello.. |
| 10:24.53 | Zeeek | should be e-geeka ! |
| 10:25.07 | tainted- | try setting to something else |
| 10:25.11 | ghenry | tainted-: Ah, no you can't . sorry Might be able to, but can't see where |
| 10:25.12 | tainted- | and then calling |
| 10:25.29 | ghenry | it won't register without UDP 5061 open |
| 10:26.18 | lucifr | I'm trying my new Asterisk installation and I wan to have PSTN termination... Does anyone know who provides it, at least for testing? |
| 10:26.37 | Zeeek | look on the wiki |
| 10:26.40 | tainted- | it won't register? |
| 10:26.46 | *** join/#asterisk saftsack (n=saftsack@p54A7F622.dip.t-dialin.net) |
| 10:26.49 | lucifr | mmmmmm.... |
| 10:26.54 | tainted- | lucifr what sort of testing |
| 10:27.01 | Zeeek | lucifr in fact you can get 0.25 free account to test voipjet |
| 10:27.07 | lucifr | Zeeek, what just I look for? |
| 10:27.12 | Zeeek | the wiki has a liong list though |
| 10:27.18 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 10:27.20 | tainted- | ghenry how do u know? 'sip show peers'? |
| 10:27.23 | lucifr | Zeeek, where? |
| 10:27.26 | Zeeek | providers |
| 10:27.37 | Zeeek | ok wait a second, here take my hand. |
| 10:27.37 | ghenry | how do I know? |
| 10:27.55 | Assid | hey zeek: how do you set it up such that zeek@abc.com and zeek@xyz.com ring on 2 different sip channels.. using the same * box ? |
| 10:28.10 | Zeeek | http://www.voip-info.org/ |
| 10:28.29 | Zeeek | Assid I was just reading about that |
| 10:28.30 | lucifr | tainted-, I don't have a phone line here at home, so I would like my friends calling from a regular phone and ring on my computer... Can it be done? |
| 10:28.33 | ghenry | tainted-: D N 5061 UNREACHABLE |
| 10:28.49 | Assid | cause when you make exten => zeek,1,Dial(SIP/zeek-abc) |
| 10:28.53 | Zeeek | lucifr this is the last time: GO LOOK AROUND - see the site I wrote above |
| 10:29.01 | Assid | that would be like universal for all zeeks |
| 10:29.09 | tainted- | lucifr yes, with a soft client like firefly or x-ten |
| 10:29.28 | Zeeek | Asid it has to do with SIP_HEADER(TO) and/or SIPDOMAIN |
| 10:29.34 | tainted- | lucifr u wouldn't need an asterisk server in that case, only a pstn->voip provider |
| 10:29.37 | ghenry | strange tainted- now it says port 5067 |
| 10:29.38 | lucifr | Zeeek, I got it... |
| 10:29.55 | Zeeek | lucifr yes in fact you can buy a phone and use a provider |
| 10:30.12 | tainted- | ghenry u have serious firewall issues |
| 10:30.19 | tainted- | put the thing in a dmz and figure it out |
| 10:30.24 | ghenry | yup |
| 10:30.27 | tainted- | then plug it back behind the nat |
| 10:30.31 | Assid | * should actually let you setup this on context based.. so suppose you want abc.com it should be like [domain-abc_com] |
| 10:30.40 | Zeeek | lucifr here is the exact sopt: http://www.voip-info.org/wiki-VOIP+Service+Providers |
| 10:30.44 | tainted- | it's not limiting to 5061, ur just diagnosing incorrectly |
| 10:30.53 | lucifr | well, I want to do more than just that... I want to play with Asterisk... |
| 10:30.54 | Assid | like how you do for voicemail |
| 10:31.01 | Zeeek | ok, see above |
| 10:31.08 | ghenry | thanks. later all |
| 10:31.14 | lucifr | Zeeek, thanks |
| 10:31.40 | Zeeek | or google for voip, voipjet, voicepulse, wengo, voiptalk, sipgate, nufone, junction networks, euriax |
| 10:32.00 | Zeeek | lucifr it depends on what country you want the number in |
| 10:32.05 | Assid | Zeeek: but is it possible to catch everythig and then handle based on the sipdomain.. and thereby throwing it to the context? |
| 10:32.20 | Zeeek | Assid accornding to what I was just reading it is |
| 10:32.28 | lucifr | Zeeek, I see |
| 10:32.36 | Assid | you got the urls' open with you ? i can refer to? |
| 10:32.50 | Zeeek | looking cause I already forgot about that |
| 10:33.39 | Zeeek | here's something interesting but doesn't give the answer: http://slacker.com/~nugget/projects/asterisk/page7 |
| 10:33.53 | Zeeek | Our very own Nugget |
| 10:38.39 | Zeeek | I can't find that page for some reason |
| 10:39.06 | Assid | weell ${SIPDOMAIN} does exist |
| 10:39.54 | Zeeek | the actual method I saw was exten => zeeek,1,GoToIf |
| 10:40.41 | Zeeek | $[${SIP_HEADER(TO)}=domain1]] |
| 10:40.54 | Assid | so.. i think what we can do .. is .. exten _.,1,Macro(handledomain,${SIPDOMAIN}) |
| 10:40.55 | Zeeek | with one less ] :) |
| 10:41.08 | Zeeek | that looks reasonable enough |
| 10:41.18 | Zeeek | or look it up in the astdb |
| 10:41.35 | Zeeek | domain = family and user = user |
| 10:42.11 | Assid | and then handledomain would have something like s,1,Dial(SIP/${arg1}-user) |
| 10:42.13 | Assid | or something |
| 10:42.45 | Assid | my astdb knowledge sucks |
| 10:43.44 | RoyK | methinks astdb should be replaced by an sql solution |
| 10:44.35 | Assid | isnt sip_header for the whole sip uri ? |
| 10:45.17 | Assid | http://www.voip-info.org/wiki-Asterisk+variables -- doesnt really document that variable |
| 10:47.08 | *** join/#asterisk ringe (n=runar@ti531210a080-6380.bb.online.no) |
| 10:50.04 | Assid | Zeeek: you tried wengo |
| 10:50.05 | Assid | ? |
| 10:52.13 | Alystair | are snom phones really good? |
| 10:52.23 | Zeeek | wengo is pretty good |
| 10:52.28 | Assid | they are decent Alystair |
| 10:52.48 | Assid | 1.0c/min right? |
| 10:53.50 | RoyK | Alystair: imho snom phones are good and expensive, but if you or your customers can afford them, it's very good indeed |
| 10:54.09 | ringe | Say I've got a Via EPIA based Asterisk server. Now I'd like to just connect a USB phone and place a call directly from that. Is it supported? |
| 10:54.39 | Assid | RoyK: just curious.. how would you rate them against polycom? |
| 10:54.46 | Assid | ringe: you would need a softphone |
| 10:54.53 | RoyK | Assid: i don't know polycom |
| 10:55.09 | Zeeek | PolCom is very good |
| 10:55.09 | RoyK | ringe: there've been talk about chan_usb, but only talk |
| 10:55.29 | Zeeek | plus, it keeps your dentures tight |
| 10:55.56 | Zeeek | Assid where are you |
| 10:56.01 | Assid | hrmm.. always wondered how they compared to one another |
| 10:56.03 | RoyK | ringe: http://www.voip-info.org/wiki/view/USB+Phone |
| 10:56.03 | Assid | Zeeek: india |
| 10:56.15 | Zeeek | you want to call france? |
| 10:56.36 | Zeeek | or be called from it ? |
| 10:56.47 | Zeeek | wengo is france |
| 10:56.59 | Assid | me?nah.. i do play with * boxes for my friends.. they call europe often.. but i wouldnt mind a US terminator too |
| 10:57.04 | Zeeek | there is a new provider called euriax for europe |
| 10:57.19 | Zeeek | euriax is very cheap |
| 10:57.29 | Zeeek | but I'm wondering about their routing |
| 10:58.10 | Ahrimanes | hey Zeeek :) |
| 10:58.11 | ringe | RoyK: That's clarifying. So a better approach would be to just connect a hardware phone to the same ethernet. Maybe using a crossed cable to connect directly to eth0. |
| 10:58.14 | Assid | what about us-pstn calls? |
| 10:58.19 | Zeeek | Ahrimanes - beer? |
| 10:58.30 | Ahrimanes | Zeeek: nah, not right now, hows life? |
| 10:58.31 | Zeeek | Assid there are a million providers for that |
| 10:58.32 | Assid | ringe: definately |
| 10:58.52 | Assid | yeah.. but not all million are good.. and nicely priced |
| 10:58.54 | RoyK | ringe: some client on ethernet, it doesn't really matter an ATA, hardphone or softphone |
| 10:59.04 | Zeeek | voipjet is good for price:quality |
| 10:59.26 | Assid | yeah. didnt have a problem with them.. BUT now.. they need verified paypal accounts |
| 10:59.29 | Zeeek | the best networks are relativel expensive, retail like 3-4 cents/min |
| 10:59.45 | Zeeek | yeah paypal sucks but it's better than nothing |
| 11:00.07 | Zeeek | you want to pay 4c/minute? |
| 11:00.13 | Assid | they dont have a verified account.. and paypal doesnt verify on CC, they need bank account to cverify |
| 11:00.18 | Assid | nah.. 1.5 odd |
| 11:00.40 | Zeeek | retail-wise not many will do less than 2c |
| 11:00.41 | Assid | voicepulse is 2.4.. im trying to find a replacement for voipjet |
| 11:00.57 | Zeeek | voicepulse is good but expensive on the penny level |
| 11:01.14 | Zeeek | and what about lag times for India? |
| 11:01.34 | Assid | not that bad really.. i make quite a few calls.. dont see any reall issues |
| 11:01.42 | Zeeek | using voipjet? |
| 11:01.45 | Assid | i dont use as much as those guys.. they use it in their office |
| 11:01.51 | Zeeek | what's the time in ms usually? |
| 11:01.54 | Assid | me personally? my usage is on sipdiscount |
| 11:02.15 | Assid | from my asterisk box.. to me? or.. asterisk box to provider ? |
| 11:02.25 | Zeeek | both, what the heck |
| 11:02.36 | Zeeek | I show you mine if you show me yours ;) |
| 11:02.59 | Assid | hehe |
| 11:03.07 | Zeeek | just curious |
| 11:03.17 | Assid | around 321ms me to asterisk box.. |
| 11:03.19 | Zeeek | I'm keeping a chart of all providers , lag and unreachable times |
| 11:03.25 | Zeeek | WHoA! |
| 11:03.33 | Zeeek | talking to the moon! |
| 11:03.56 | Zeeek | where's the box? |
| 11:03.59 | Assid | this box to voicepulse -- 59ms |
| 11:04.00 | Zeeek | Jupiter? |
| 11:04.06 | Assid | err.. mercury |
| 11:04.27 | Zeeek | funny, my lag to most providers is 50-100ms |
| 11:04.33 | Zeeek | 30-40 to my box |
| 11:04.49 | Zeeek | the French providers are 20-30ms from the box |
| 11:05.04 | Zeeek | so the echo is much shorter :) |
| 11:05.05 | Assid | 60 ms.. from this box to voicepulse |
| 11:05.17 | mut | anyone ever used a sangoma a104d? the wac_ec_client command dies when i try to stat it |
| 11:05.37 | mut | <PROTECTED> |
| 11:05.37 | mut | > syntax error (argv=(null),offset=0) |
| 11:05.44 | Zeeek | my best US providers are around 100ms |
| 11:05.48 | Assid | round-trip min/avg/max = 23.9/24.5/25.8 ms -- from my friends box -- to voicepulse |
| 11:06.05 | Zeeek | where is it though, in the US? |
| 11:06.14 | Assid | 1 in texas.. 1 in ny |
| 11:06.32 | Zeeek | yeah ok. My box in the US is 3ms away from one of the providers! |
| 11:06.42 | Assid | his box is in ny.. mine is in datacenter in texas |
| 11:06.48 | Zeeek | gotcha |
| 11:06.59 | Zeeek | makes sense the 30ms to vpulse then |
| 11:08.41 | Zeeek | Assid that's interesting, those times. I thought I was bad off with 125ms to my second box |
| 11:08.43 | lucifr | Does any of the provider support more than 1-channel? I mean, I want to be able to setup multiple phones at home and have several people make calls! |
| 11:09.01 | Zeeek | lucifr if you read their sites they tell you |
| 11:09.02 | kamileon | where do you guys colocate ? |
| 11:09.03 | Assid | 320ms is from my in india to houston /texas |
| 11:09.14 | Zeeek | I understood that Assid |
| 11:09.23 | Zeeek | and mine is Paris-> Virginia |
| 11:09.24 | lucifr | Zeeek, yes, I'm going through the list! |
| 11:09.28 | Assid | hosted the box with ev1 |
| 11:09.42 | Zeeek | lucifr they usually specify it in the FAQ |
| 11:09.50 | lucifr | ok |
| 11:09.57 | lucifr | tnx |
| 11:10.06 | Zeeek | or write them. It's very revealing to call or write before you buy |
| 11:10.20 | Zeeek | try calling cust "care" before you sign up |
| 11:12.17 | Zeeek | lucifr where are you located? That will also determine who you want to go with |
| 11:12.24 | lucifr | Zeeek, thanks... you're my heroe! |
| 11:12.27 | lucifr | :) |
| 11:12.42 | Zeeek | I've been thru the shooping experience for two years |
| 11:12.45 | Assid | Zeeek: most providers for me is 50-70ms |
| 11:12.52 | lucifr | Zeeek, I'm in the West side of the states |
| 11:12.58 | Zeeek | you mean for you box? |
| 11:13.02 | Assid | yeah |
| 11:13.07 | Zeeek | then you add the 300ms to you |
| 11:13.11 | Assid | yep |
| 11:13.16 | Assid | but frankly |
| 11:13.18 | Assid | its nto that bad |
| 11:13.32 | Zeeek | lucifr look up junction ( jnctn.net ) |
| 11:13.43 | nettie | Hi guys, I noticed that when we receive a call from overseas the + or the double 0 is omitted in the CID |
| 11:13.44 | Zeeek | or voipjet.com |
| 11:13.49 | lucifr | Zeeek, thanks |
| 11:13.54 | Zeeek | just a thought |
| 11:13.58 | nettie | any of your guys know hot to tell asterisk to add it please? |
| 11:14.39 | Zeeek | nettie I believe all that can vary from country to country so there's no perfect way |
| 11:14.59 | Zeeek | unless maybe detecting the lack of a '1' in the beginning |
| 11:15.21 | Assid | Zeeek: you connected to sip broker? |
| 11:15.35 | Zeeek | no sir i am not. Never heard of them til today |
| 11:16.13 | Zeeek | is it free? |
| 11:16.17 | Assid | its like a dns based service exchange.. |
| 11:16.52 | jql | 00 is dropped in some pstn calls as well. it's irritating... |
| 11:16.53 | Zeeek | i'm looking at the site now |
| 11:16.59 | Assid | like i have black.abc.com .. and when i signup. i get something like *746 for example as a number for my box |
| 11:17.19 | Assid | and if i am on extension 201.. then you just dial *746201 |
| 11:17.36 | Assid | it iwll make a sip/uri call to 201@black.abc.com |
| 11:17.48 | Zeeek | ok I see the point now |
| 11:18.41 | Zeeek | is there a service that allows you to make a SIP URI call from a site? |
| 11:18.52 | Zeeek | for testing |
| 11:19.03 | Assid | well.. it has a test call thing |
| 11:19.07 | Assid | you cant hear nothing |
| 11:19.12 | Assid | but atleast you know if its working |
| 11:19.13 | Zeeek | good enuf |
| 11:19.21 | Zeeek | test the SIP URI? |
| 11:19.24 | Assid | yep |
| 11:19.35 | Zeeek | is it free before I fill out 50 forms? |
| 11:19.57 | Assid | no forms.. user/pass/uri/email |
| 11:19.58 | Assid | thats it |
| 11:20.03 | Assid | and then you get confirmation email |
| 11:20.10 | Assid | to activate |
| 11:20.17 | Zeeek | and pay $100 ? |
| 11:20.20 | lucifr | Zeeek, I hope you can help me with this.... At the site you mentioned earlier it says on their website "Up to 25 Simultaneous calls - FREE". Does that mean 25 simultaneous channels open to make phone calls? |
| 11:20.51 | Assid | pay $100 for a free service? you feeling rich ? |
| 11:20.52 | Zeeek | I don't know but I *assume* they mean you are billed for the minutes of *each* call |
| 11:21.07 | Zeeek | lucifr was that junction? |
| 11:21.13 | lucifr | yes |
| 11:21.33 | nettie | Zeeek well all country doesnt have the + ore the 00 |
| 11:21.34 | Zeeek | I think they bill by the MINUTE, so 25 1 second calls would be billed as 25 minutes |
| 11:21.56 | Zeeek | nettie but you see the 00 always? In that case it's easy |
| 11:21.56 | nettie | Zeeek if it's france I get 330498784533 |
| 11:22.02 | nettie | no |
| 11:22.06 | nettie | never |
| 11:22.10 | Zeeek | ok you're screwed then |
| 11:22.26 | nettie | youthink it's a problem of my carrier? |
| 11:22.33 | lucifr | I wish they (Junction Networks) had unlimited outbound calls! |
| 11:22.37 | Assid | Zeeek: that was why i wanted to know how to capture the domain.. |
| 11:22.40 | Zeeek | how about "if you are calling froma foreign country, please hit the pound key" |
| 11:22.47 | nettie | which doesnt send me the correct cid? |
| 11:23.00 | Assid | i got 2-3 vhosts on the same box |
| 11:23.17 | Zeeek | Assid yeah. If I can test mine I'd have the answer |
| 11:23.21 | nettie | Zeeek eheh nah, this just to have the correct cid? |
| 11:23.22 | nettie | noway |
| 11:23.23 | nettie | ehehe |
| 11:23.49 | Zeeek | yeah I know but I don't see how to determine it because of the variation in number lengths |
| 11:24.14 | Assid | nettie: just consider all the numbers as starting with 1 |
| 11:24.16 | Assid | err |
| 11:24.17 | Assid | + |
| 11:24.25 | nettie | no |
| 11:24.28 | nettie | that's not good |
| 11:24.32 | Assid | so.. if you get from 3304....... the number is +3304 |
| 11:24.42 | nettie | because cells are 338-4567898 |
| 11:24.53 | nettie | so it wont work |
| 11:25.03 | Assid | yes.. and when a cell calls you.. it will be 13384567898 |
| 11:25.14 | nettie | nah |
| 11:25.18 | nettie | it's paing |
| 11:25.19 | nettie | ehehe |
| 11:25.25 | Assid | sure it is |
| 11:25.26 | Assid | try it |
| 11:25.46 | nettie | I maybe try to ask my carrier to send me the proper cid for int number and see what they say me |
| 11:26.10 | Assid | okay one more development change and then i can keep this aside |
| 11:26.20 | Assid | i wish i had more time.. i wanna play with that sip uri |
| 11:26.27 | Assid | unfortunately.. i got work to do |
| 11:26.38 | Zeeek | so how do I test the number Assid? |
| 11:26.41 | Assid | Zeeek: http://www.astmasters.net/howtos.html incase you wanna read |
| 11:26.53 | Zeeek | I saw that one but I don't think it was the best |
| 11:27.17 | Assid | Zeeek: right side. there is a ezdial |
| 11:27.23 | Assid | else call *75493001 |
| 11:27.34 | Assid | provided you set up your dial plan and sip already |
| 11:27.45 | Assid | thats for outgoing |
| 11:27.54 | Assid | did you get a sip broker number? |
| 11:28.11 | Assid | http://faq.sipbroker.com/tiki-index.php?page=Asterisk+Configuration <-- |
| 11:30.35 | Assid | i cant seem to add a dyndns.org account |
| 11:36.41 | Zeeek | yes |
| 11:36.48 | Zeeek | im on the phone now |
| 11:37.25 | *** join/#asterisk ghost99 (n=neville@222-152-219-77.jetstream.xtra.co.nz) |
| 11:39.32 | Assid | works? |
| 11:40.13 | Zeeek | no it was my wife calling on DID from New York |
| 11:40.21 | Assid | oh |
| 11:40.23 | Zeeek | excellent quality call by the way |
| 11:40.34 | Assid | through which network? |
| 11:40.35 | Zeeek | Junction Networks |
| 11:40.45 | Zeeek | but 0.04c/min :( |
| 11:41.01 | Zeeek | too bad voipjet doesn't have tollfree DID |
| 11:41.07 | Zeeek | and nufone is down |
| 11:41.19 | Zeeek | and teliax is too far away |
| 11:41.21 | Assid | err.. 2.9c/min |
| 11:41.23 | Assid | thats what it says |
| 11:41.29 | Assid | oh wait |
| 11:41.31 | Assid | tollfree |
| 11:41.33 | Zeeek | not on TOLLRFREE incoming my friend |
| 11:41.46 | Zeeek | prolly 3.9 |
| 11:41.58 | Zeeek | but like I said I think they round up to the minute too |
| 11:42.04 | Assid | junction charging for incoming too!! normal numnber |
| 11:42.06 | *** join/#asterisk Itburnz (n=UNITY@itburnz.de) |
| 11:42.11 | Zeeek | so getting back to the url test |
| 11:42.26 | Itburnz | good afternoon everyone |
| 11:42.27 | Assid | shit.. i gotta get back to finishing this code |
| 11:42.38 | Assid | Zeeek: let me know how it works out |
| 11:42.40 | Zeeek | too bad all I want to do is ring the sip url |
| 11:42.43 | Assid | tyr and macronise it |
| 11:42.43 | Zeeek | ok |
| 11:42.53 | Itburnz | hey does anyone of you know about a problem regarding asterisk 1.2.7.1 & spanDSP ? i get the error "app_rxfax.so: undefined symbol: t30_get_far_ident" - was wondering if anyone here knows a solution |
| 11:43.04 | Assid | ringing sip url aint a problem |
| 11:43.50 | Assid | the bad thing is.. you need to have the dialplan defined into [default] |
| 11:44.56 | Assid | 8610 ? yours? |
| 11:45.10 | Zeeek | can you dial sip url? |
| 11:45.16 | Assid | yep |
| 11:45.24 | Assid | you signup already? |
| 11:45.34 | Zeeek | I already have code in extensions for it |
| 11:45.47 | Zeeek | I didn'yt set up sipbroker lookup though |
| 11:45.57 | Assid | in default?!?!? |
| 11:45.58 | Zeeek | I don't want or need that service |
| 11:46.19 | Zeeek | yeah as I said I have code in the default section to receive calls for certain aliases |
| 11:46.27 | Assid | well. its just that people can call you for free.. throough a 'central' location |
| 11:47.01 | Assid | gimme your sip uri.. |
| 11:47.12 | Zeeek | thry this: |
| 11:47.31 | Zeeek | tsturl@r.declic.com |
| 11:48.29 | Zeeek | I used to use e143.org |
| 11:48.33 | Zeeek | oops, not exactly |
| 11:50.07 | lucifr | Zeeek, how about just getting IP phones for my mom overseas and configure that phone to call me over in the states? |
| 11:50.26 | Zeeek | that's the best solution for what you want I'd say |
| 11:50.57 | lucifr | Zeeep, good... now we are getting somewhere... |
| 11:51.25 | Zeeek | you might want to get a cheap IAX hardphone for her |
| 11:51.36 | Zeeek | that way there is zero setup problems |
| 11:51.48 | Zeeek | of course she needs a high speed internet connection |
| 11:52.05 | Zeeek | Assid, I am not getting any calls |
| 11:52.13 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 11:52.13 | lucifr | Zeeep, how about a "device" that I can configure and also convert signals from a regular phone (cutting cost here) :P |
| 11:52.15 | Assid | sip debug |
| 11:52.17 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 11:52.18 | Assid | pastebin |
| 11:52.37 | Zeeek | lucifr yes, that'll work well, it's called an ATA |
| 11:52.41 | lucifr | Zeep, she has DSL overseas |
| 11:52.48 | Zeeek | but again, requires Internet access |
| 11:52.51 | Zeeek | ok |
| 11:53.14 | Zeeek | Would she have others she'd like to call in the US? |
| 11:53.20 | lucifr | Zeeep, computer required for ATA? |
| 11:53.28 | Zeeek | no that's the point |
| 11:53.37 | lucifr | Zeeep, not necesarely.. |
| 11:53.38 | Zeeek | I hate being tied to a PC phone |
| 11:53.56 | Zeeek | and you want to set up an asterisk box to play? |
| 11:54.29 | Zeeek | Assid try again, I'll turn debug on |
| 11:55.12 | lucifr | "and you want to set up an asterisk box to play?" Are you asking me? |
| 11:55.18 | Zeeek | yes |
| 11:55.23 | Zeeek | I'm trying to help |
| 11:55.28 | lucifr | I'm not following you.. |
| 11:55.32 | Zeeek | determine what the best choices are |
| 11:55.48 | RoyK | <PROTECTED> |
| 11:56.06 | Assid | did you get it? |
| 11:56.11 | *** join/#asterisk v3rmap (n=puser@unaffiliated/v3rmap) |
| 11:56.16 | Zeeek | looking |
| 11:57.03 | Zeeek | not getting anything even remotely suspicious |
| 11:57.14 | Zeeek | like 'tst' or 'declic' |
| 11:57.20 | v3rmap | Hi, I've installed asterisk on Ubuntu, and started it. But I see thru nmap that the port no. 5060 is closed. An x-lite phone is also not able to connect and times out. What could be wrong? |
| 11:57.55 | lucifr | Zeeek, what I really want is for her to call me from a IP phone and Asterisk redirect the call to my wireless IP phone and reach me anywhere... Now, she is not computer savvy and I'm trying to cut cost... |
| 11:57.59 | v3rmap | I've modified extensions.conf and sip.conf to add users and extensions. |
| 11:58.06 | *** join/#asterisk homac (i=holle@strace.org) |
| 11:58.15 | Zeeek | lucifr I understand, millions do this daily :) |
| 11:58.33 | jql | v3rmap: did you try running it manually as asterisk -f -vvvvvv or some such? |
| 11:58.44 | *** part/#asterisk homac (i=holle@strace.org) |
| 11:58.45 | lucifr | Zeeep, I only need a ATA and that's it? |
| 11:58.53 | v3rmap | jql: I just started it manually as "asterisk" |
| 11:59.00 | Zeeek | by wireless, you mean cellphone? |
| 11:59.15 | lucifr | who me? |
| 11:59.24 | jql | v3rmap: adding -f keeps it in the foreground, and -vvvvv makes it print lots of verbose trace info |
| 11:59.25 | Zeeek | lucifr wireless ip phone? What is it? |
| 11:59.34 | v3rmap | jql: asterisk is running and I can go to the asterisk command interpreter by using "asterisk -r" |
| 11:59.48 | v3rmap | jql, thanks I'll try -vvvv |
| 12:00.04 | jql | rather than nmap, does netstat -a show udp:5060 open? |
| 12:00.33 | lucifr | it's a wi-fi phone that register itself to a Asterisk box through a open wireless network so Asterisk can redirect calls.. (inbound and outbound(\\\) |
| 12:00.44 | v3rmap | jql, netstat -a does not show any line that has 5060 in it. |
| 12:00.53 | jql | well, that's bothersome |
| 12:01.06 | Zeeek | lucifr oh so you'd wander around town and talk thru hotspots? |
| 12:01.40 | *** join/#asterisk MoutaPT (n=MoutaPT@85.139.183.56) |
| 12:02.14 | lucifr | Zeeep, not that popular now.... but since Google will be rolling out free wi-fi across the US... it will be nice... |
| 12:02.35 | Zeeek | ok, I'm not folliwng that too closely at the moment |
| 12:02.43 | MoutaPT | Hi, any one could advice me how do i check what went wrong in my production server that just went down? |
| 12:02.44 | lucifr | :) |
| 12:02.48 | Assid | Zeeek: did you get it? |
| 12:02.57 | Zeeek | anyway,n yeah an ATA and a normal phone plugged in to it will do the trick |
| 12:03.06 | Zeeek | Assid, nothing at all |
| 12:03.12 | Assid | dude.. it has to work |
| 12:03.20 | Zeeek | What is this "debug server" they mention? |
| 12:03.24 | Assid | try calling this.. 3001@mercury.sphrerelinx.com |
| 12:03.36 | MoutaPT | I've started it now, and also didn't get the msg like your pc has been incorrectly shut down... |
| 12:03.42 | Assid | ignore that |
| 12:03.48 | Assid | just sip debug your * box |
| 12:04.01 | MoutaPT | Assid talking to me? |
| 12:04.02 | Zeeek | ok but anyway I haven't put ANY code in extensions |
| 12:04.15 | Zeeek | Assid I did that and there was nothing trying to come in |
| 12:04.15 | Assid | MoutaPT: nope |
| 12:04.20 | lucifr | Zeeep, what a good and reliable ATA device? |
| 12:04.40 | Assid | sip:tsturl@r.declic.com |
| 12:04.45 | Zeeek | you'll have to do research on that. Sipura has a lot of happy users |
| 12:04.54 | jql | <-- sipura user |
| 12:05.07 | *** join/#asterisk madounet (n=madounet@juv34-2-82-226-155-19.fbx.proxad.net) |
| 12:05.15 | coppice | good *and* reliable. gee, some people want *everything* :-) |
| 12:05.18 | Zeeek | exten => tsturl,1,NoOp(${EXTEN} ${CONTEXT} ${SIPDOMAIN}) |
| 12:05.20 | lucifr | Zeeep, beautiful.... |
| 12:05.28 | jql | I actually shopped for cheap |
| 12:05.32 | jql | that it worked was a bonus |
| 12:05.41 | Zeeek | lucifr watch out for power supply voltages if you buy it! |
| 12:05.49 | Assid | r.declic.com --- is that your box |
| 12:06.01 | Zeeek | it's an alias for now, yeah |
| 12:06.16 | lucifr | ok, thx |
| 12:06.23 | lucifr | brb |
| 12:06.42 | *** join/#asterisk tdonahue (n=tdonahue@www.vonworldwide.com) |
| 12:08.38 | Zeeek | Assid, actually I see what I wanted to know because Dial, while complaining of a loop condition when I dial myself, also says "Thank to ..." and gives the domain name |
| 12:09.07 | Zeeek | that same uri is likely available in the SIP_HEADER(TO) variable |
| 12:09.37 | *** join/#asterisk the_magic_bean (n=the_magi@209.43.15.211) |
| 12:09.45 | *** part/#asterisk the_magic_bean (n=the_magi@209.43.15.211) |
| 12:10.18 | *** join/#asterisk ghost99 (n=neville@222-152-219-77.jetstream.xtra.co.nz) |
| 12:10.49 | *** join/#asterisk the_magic_bean (n=the_magi@209.43.15.211) |
| 12:10.56 | mut | man o mn |
| 12:11.08 | mut | why does mcdonalds breakfast have to be so expensive |
| 12:11.26 | lunk | because you bought too much stuff that will kEEL YOU |
| 12:11.27 | nextime | bleah |
| 12:12.07 | lunk | i'm having a bfast of champions right now, redbull and poptarts |
| 12:12.12 | lunk | fear the sugar rush |
| 12:13.08 | nextime | cappuccino and croassant, and a black strong expresso coffe, is all what you need for a good bfast :) |
| 12:13.23 | mut | i got bad acid reflux |
| 12:13.27 | lunk | cappuccino and expresso? |
| 12:13.29 | mut | i can't drive coffee/cappa |
| 12:14.06 | mut | drive = drin |
| 12:14.07 | mut | k |
| 12:14.08 | mut | :P |
| 12:14.17 | nextime | lunk yep, cappuccino as a real bfast, and coffee for your brain wakeup |
| 12:14.19 | lunk | obviously you need to |
| 12:14.36 | lunk | ah |
| 12:14.54 | nextime | lunk : i'm italian, expresso is a must for me :) |
| 12:14.59 | v3rmap | folks, can anyone tell me why asterisk is not listening at port 60 on my system? |
| 12:15.17 | lunk | ah, someday i will visit italy |
| 12:15.24 | v3rmap | Here is the sip.conf: http://pastebin.com/692092 |
| 12:16.12 | nextime | lunk : you will probably try the best food in the world here :) |
| 12:16.37 | lunk | nextime: millions of fat americans would argue with thtat, but i'll take your word for it |
| 12:16.59 | *** join/#asterisk Mavvie (n=edwin@252-131-222-203.rev.techex.net.au) |
| 12:16.59 | Assid | damn.. my noop didnt get executed |
| 12:17.09 | lunk | my great grandmother on my mom's side came from sicily around 1910 or so |
| 12:17.24 | v3rmap | Oh, sorry, I meant asterisk is not listening on port 5060. And the sip.conf is: http://pastebin.com/692092 |
| 12:17.39 | nextime | lunk : yeah, so you have some italian blood in your body :) |
| 12:18.04 | lunk | oh yea, but my brother got all the outward genes, i just got the rage \o/ |
| 12:19.15 | lunk | anyway, when dealing with .call files, if your trunk is in use and a second call file is added to the outbound spool |
| 12:19.25 | lunk | does it fail when the trunk is in use (or at capaciy)? |
| 12:19.34 | nextime | anyway, you can find good food in any part of the world, the real problem of fat americans is the food education, not the food itself |
| 12:20.11 | lunk | i want to be able to drop like 10 .call files and not worry about them |
| 12:21.14 | nextime | lunk, you can set retry timeout 1, 2 3 time or so... |
| 12:21.47 | nextime | and when it exceed it disappear from your spool dir |
| 12:21.58 | lunk | yea, that seems kinda lame |
| 12:22.11 | lunk | i'd like to know that every .call file is actually routed outbound at least once |
| 12:22.25 | Greek-Boy | if a call comes in on a zap interface can caller id show on the IP phone? |
| 12:22.34 | lunk | Greek-Boy: yes |
| 12:22.37 | mut | would anyone consider 15 min breaks in an 8 hr work day a lot? |
| 12:22.44 | mut | 3 15 min breaks* |
| 12:22.56 | lunk | mut: no, but it depends on the situation |
| 12:23.10 | mut | in an accounting office |
| 12:23.17 | lunk | oh hell no |
| 12:23.24 | lunk | that crap is boring enough as it is ;) |
| 12:23.26 | mut | no? |
| 12:23.27 | Greek-Boy | nice |
| 12:24.28 | nextime | lunk : i think that you can set an extension that do Dial and some other work instead of dialout directly from the .call file, no? |
| 12:24.43 | lunk | probably |
| 12:24.51 | lunk | ideally i'd like to use Queues in a backwards way |
| 12:24.57 | nextime | ( maybe i'm saying something stupid, but it can work ) |
| 12:25.20 | lunk | nextime: if i want it to show up in the log at all i have to route it to an extension |
| 12:25.23 | Assid | err |
| 12:25.27 | Assid | something is totally wrong here |
| 12:25.48 | nextime | Assid : probably my synapses |
| 12:25.51 | Zeeek | Assid so you are supposed to be able to dial these numbers as *12345 ? |
| 12:26.21 | Zeeek | is srvlookup needed? |
| 12:27.01 | Assid | yes |
| 12:27.13 | Assid | err.. am having very very weird issue |
| 12:27.25 | Assid | i disabled my extension from default |
| 12:27.28 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 12:27.30 | Assid | and yes.. reloaded it |
| 12:27.37 | Assid | still the calls are coming on my extension |
| 12:27.57 | Zeeek | Assid do you have any really wide expressions like _. ? |
| 12:28.12 | Zeeek | I had this happen yesterday: |
| 12:28.14 | Assid | in default.. NOTHING |
| 12:28.37 | Zeeek | exten => _${SOMEVAR}.,1,do something |
| 12:28.40 | Assid | and i know for a fact it tries to match with default cotext |
| 12:28.49 | Zeeek | well, if SOMEVAR happens to be empty.... |
| 12:29.00 | Zeeek | that will talke ALL calls |
| 12:29.10 | Assid | nothing |
| 12:30.06 | Assid | im removing the default context |
| 12:30.09 | Assid | lets see what it does |
| 12:30.19 | Assid | !@#!#@@#$#@ its still calling |
| 12:30.21 | Assid | wtf |
| 12:30.33 | Assid | sip cached?!?!? |
| 12:30.37 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:30.53 | orlok | Assid: reload or restart? |
| 12:31.13 | Assid | tried both |
| 12:31.21 | Assid | i even shut down asterisk for 10 seconds |
| 12:32.27 | Zeeek | did you kill a chicken during a full moon, though? |
| 12:32.34 | Assid | nah |
| 12:32.38 | Zeeek | because without that, no dialplan is safe |
| 12:32.45 | Assid | there was chicken flu |
| 12:32.48 | Assid | H5N1 |
| 12:33.02 | Zeeek | so I got sipbroker working. Now what? |
| 12:33.11 | Assid | yuou did? |
| 12:33.17 | Assid | minne just went kaboom |
| 12:33.20 | Zeeek | well I got the test messages |
| 12:33.27 | Assid | i removed the extension .. it still doesnt work |
| 12:33.35 | Assid | ewrr..it still works |
| 12:33.37 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:34.25 | Assid | finally dies |
| 12:35.13 | kavit | video conferening and oh323, stuff nightmares are made of |
| 12:35.27 | kavit | no, oh323, stuff nightmares are made of |
| 12:35.28 | kavit | :( |
| 12:35.56 | lucifr | Anybody familiar with the Linksys PAP2 Phone Adapter (ATA)? Is it locked to "Vonage"? |
| 12:36.11 | Zeeek | the models ending in NA are unlocked |
| 12:36.17 | Zeeek | (I theeeeeenk) |
| 12:36.45 | lucifr | hmmm... |
| 12:36.46 | Assid | umm.. it doesnt wanna check in the default context no more |
| 12:36.47 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
| 12:36.56 | Assid | Zeeek: is yours checking in the default context? |
| 12:37.00 | lucifr | ok, thanks Zeep |
| 12:37.03 | Assid | May 1 18:06:29 NOTICE[1004]: pbx.c:1738 pbx_extension_helper: Cannot find extension context 'spherelinx' |
| 12:37.17 | pigpen2 | Hi all, would I assume correctly that Asterisk does not have an ACD (Automatic Call Distribution) solution yet? |
| 12:38.41 | Assid | oh wait |
| 12:38.41 | Assid | i think i know why |
| 12:39.06 | Zeeek | Assid, these international number I see: you dial *code*number or what? |
| 12:39.24 | Zeeek | on this page: http://www.sipbroker.com/sipbroker/action/providerWhitePages |
| 12:40.43 | [TK]D-Fender | pigpen : No you wouldn't. |
| 12:41.12 | [TK]D-Fender | pigpen : I take it you looked REALLY hard :) |
| 12:41.29 | [TK]D-Fender | (#2) that is... |
| 12:42.02 | Assid | yes yes.. i got it |
| 12:42.05 | coppice | one of the greatest cons of the telecoms world was how PBX vendors sold customers on the idea that ACD is a really really hard problem. |
| 12:42.06 | [TK]D-Fender | Wish I could make autocomplete do a best case match agianst "last spoke" |
| 12:42.15 | kavit | pigpen2: asterisk is really virtual monkeys on a virtual switchboard jacking random lines |
| 12:42.15 | Assid | i understood how to do vhosting on this based |
| 12:42.21 | Assid | soo freaking easy |
| 12:42.51 | [TK]D-Fender | kavit : Virtual monkeys were too expensive to develop so we outsourced it to Indian monkeys :) |
| 12:43.01 | pigpen2 | [TK]D-Fender, well, after searching the wiki and googling, all I could find is a page referencing commercial products.... |
| 12:43.14 | pigpen2 | so my assumption was based on research. |
| 12:43.27 | pigpen2 | personally, ACD sounds like a well built queue. |
| 12:43.27 | [TK]D-Fender | pigpen2 : Dear God you really didn't look to well... try "asterisk queues" on the wiki..... |
| 12:43.38 | coppice | if an ACD solution requires monkeys, i doubt typical call centre staff would really be up to the job |
| 12:43.40 | lunk | how hard is it to install the PGSQL application after-the-fact? |
| 12:43.54 | kavit | [TK]D-Fender: hahaha saved some money I bet |
| 12:43.57 | [TK]D-Fender | pigpen2 : There is this miraculous module called "app_queue" thats been around for a great many years.... |
| 12:44.12 | pigpen2 | yeah...as long as you don't use chan_agent |
| 12:44.15 | [TK]D-Fender | coppice : Note we didn't say TRAINED monkeys :) |
| 12:44.26 | [TK]D-Fender | coppice : These are RANDOM connections after all! |
| 12:44.45 | coppice | simply saying monkeys implies a certain basic IQ |
| 12:45.00 | pigpen2 | [TK]D-Fender, sorry to get you on the wrong side of the bed....I was unsure if ACD was like queues..... |
| 12:45.07 | [TK]D-Fender | pigpen2 : What about chan_agent? Can work depending on your needs. You can also to static agents, etc. |
| 12:45.16 | [TK]D-Fender | ACD = Queues |
| 12:45.18 | pigpen2 | yes...static works fine... |
| 12:45.30 | kavit | http://www.voip-info.org/wiki/view/Agents+without+agent+channel |
| 12:45.38 | kavit | pigpen2: there you go |
| 12:45.48 | Zeeek | Assid, it does work |
| 12:45.51 | [TK]D-Fender | pigpen2 : There is also an add-on package calld ICD you can lookup on the WIKI which takes a different approach. |
| 12:45.58 | Assid | yeh |
| 12:46.03 | coppice | even if during the daytime the agents might appear to outwit a monkey, in the middle of the night on the graveyard shift high on crack could they still manage it? |
| 12:46.05 | Assid | you just use simple goto |
| 12:46.14 | [TK]D-Fender | kavit : Thats pre 1.2.x code and basically all stupid dial-plan..... |
| 12:46.43 | kavit | [TK]D-Fender: don't shoot the messenger :( |
| 12:47.03 | [TK]D-Fender | kavit : Sorry, you're all I've got :) |
| 12:47.15 | pigpen2 | [TK]D-Fender, thanks...sorry about the confusion...I was just unfamiliar with the term "ACD"...queues, well...lets just say, chan_agent made my life hell for about 2 weeks. |
| 12:47.32 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 12:47.33 | [TK]D-Fender | pigpen2 : What was your probelm, and what do you need? |
| 12:47.38 | kavit | [TK]D-Fender: alright I will just start bothering you with oh323 video conferencing questions |
| 12:48.24 | pigpen2 | [TK]D-Fender, well, using chan_agent was causing -some- sort of an issue that was causing a dead lock of asterisk. Going to static agents took care of it. |
| 12:48.34 | [TK]D-Fender | kavit : Sorry, I have successfully avoided H.323 since discovering * and SIP :) |
| 12:49.06 | the_magic_bean | anyone have any idea how to backup/restore voicemail? I have the audio files but im not sure how asterisk stores the 'database' connecting messages to users to files. |
| 12:49.17 | [TK]D-Fender | pigpen2 : Could be that it was a bug that has since been resolved.... What kind of dead-lock. Like hard-code or something making the dial-plan stick? |
| 12:49.56 | [TK]D-Fender | the_magic_bean : Depends, sounds like your using a database to store it as opposed to the native mailbox method |
| 12:50.06 | pigpen2 | [TK]D-Fender, well, this was about 2 weeks ago...maybe 3. I opened a ticket with Digium, and they said "yeah...use static's....it is a know issue" |
| 12:50.37 | kavit | [TK]D-Fender: yeah but people want everything, I told them if you want a video conferencing bridge go Polycom, but curiosity got the better of me and here I am trying to see if it can be done with Open source products |
| 12:50.47 | codebreaker | what is the first place to look when "res_odbc.c:563 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno= 0 [unixODBC][Driver ManagerData source name not found, and no default driver specified" |
| 12:51.15 | the_magic_bean | [TK]D-Fender: well not really sure about that, I set the box up, and i did not do anything with a pgSQL or mySQL, so i would assume it is whatever native mailbox method is |
| 12:51.32 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 12:51.40 | [TK]D-Fender | kavit : You could use Ekiga for video.... |
| 12:51.41 | kavit | codebreaker: unixODBC cannot find a data source and a driver? |
| 12:52.18 | kavit | [TK]D-Fender: no I want to see if I can get it running over ISDN or play nice with another polycom av bridge |
| 12:52.36 | [TK]D-Fender | the_magic_bean : Then you just need to back up the voicemail foder (by default /var/spool/asterisk/voicemail |
| 12:52.53 | [TK]D-Fender | kavit : Oh... its too late :) hehe sorry! |
| 12:53.03 | kavit | codebreaker: what DB are you using? |
| 12:53.09 | the_magic_bean | [TK]D-Fender : thank you sir |
| 12:53.37 | kavit | codebreaker: install a driver for that db and if you want ease of configuration use ODBCConfig, its a gui tool that is packaged with unixODBC |
| 12:54.25 | kavit | [TK]D-Fender: yeah there is not one decent open source solution that can do video conferencing with h323 properly |
| 12:54.33 | codebreaker | kavit: i use postgres and tried http://www.asteriskguru.com/tutorials/realtime_pgsql.html |
| 12:55.11 | kavit | codebreaker: did you check if the driver was installed? |
| 12:57.33 | codebreaker | kavit: jupp is installed. i now go to increase ma logs on the postgres. |
| 12:58.55 | Assid | Zeeek: are you setting up the srv? |
| 12:59.10 | Zeeek | I used to but it's off now |
| 12:59.22 | Zeeek | if you mean srvlookup |
| 13:00.06 | Assid | yeah |
| 13:00.12 | Zeeek | so people using anyone those services can call me free? I can't get that to work yet |
| 13:00.15 | Assid | im trying to setup the dns to use it |
| 13:00.28 | Assid | err.. well.. did you setup your sip broker? |
| 13:00.30 | Zeeek | dyndns allows all those records |
| 13:00.49 | Assid | err.. im doing it frommy own dns |
| 13:01.04 | Zeeek | I don't see what the alias is |
| 13:02.53 | Zeeek | is alias the same as member number? |
| 13:03.25 | Assid | no no.. whats your broker number |
| 13:03.29 | Assid | its as soon as you login |
| 13:03.37 | Zeeek | that say member number |
| 13:03.46 | *** join/#asterisk zaf (n=zaf@65.255.203.114) |
| 13:03.52 | Assid | yeah |
| 13:03.53 | Zeeek | oh that was a number I made up^ |
| 13:03.54 | Assid | whats that |
| 13:04.01 | Assid | its part of the sip uri |
| 13:04.08 | Assid | *xxx-sipuser |
| 13:07.43 | *** join/#asterisk DoktorGreg (n=Greg@70.91.121.89) |
| 13:08.06 | Zeeek | Assid if the code given for wengo is *248 what do you dial after? |
| 13:08.17 | Zeeek | the whole *011blahblah? |
| 13:10.34 | Greek-Boy | with IP phones that have two RJ45 ports for passing one link to the PC, does the PC and the phone have to be on the same network? |
| 13:10.37 | [TK]D-Fender | kavit : Have you tried Ekiga? Does H.323 & SIP |
| 13:10.37 | Assid | err.. |
| 13:10.42 | Greek-Boy | i mean same subnet |
| 13:10.45 | Assid | *248 and your DID |
| 13:11.00 | [TK]D-Fender | Greek-Boy : Not if your switch has vlan support. |
| 13:11.40 | Greek-Boy | is it better to put the phones on a seperate unmanaged switch or two use vlans? |
| 13:11.41 | [TK]D-Fender | Greek-Boy : Not that it would save you from bandwidth issues, and assumes the phone supports it as well. |
| 13:11.51 | [TK]D-Fender | Greek-Boy : Seperate LAN naturally. |
| 13:12.11 | Greek-Boy | ok but are unmanaged switches better? |
| 13:12.20 | Greek-Boy | coz all bandwidth is dedicated to the phones |
| 13:12.51 | [TK]D-Fender | Greek-Boy : depends on your tastes & needs. Technically better, yes, worth it? Depends on you. I personally don't care for it. |
| 13:13.04 | [TK]D-Fender | Greek-Boy : I wouldn't think twice if you're getting a seperate lan. |
| 13:13.10 | Greek-Boy | i see |
| 13:13.10 | Greek-Boy | :) |
| 13:13.13 | [TK]D-Fender | Greek-Boy : Describe your deployment |
| 13:13.52 | Assid | YES... it works |
| 13:14.12 | Assid | Zeeek: i got this to work.. you can have virtual hosting this |
| 13:14.15 | Assid | very very easy |
| 13:14.18 | Assid | simple macro |
| 13:14.23 | kavit | [TK]D-Fender: yeah but asterisk doesnt do conference rooms with Video and h323 methinks |
| 13:15.24 | Zeeek | Assid so let's see it! |
| 13:15.25 | Greek-Boy | [TK]D-Fender 122 phones that I need to connect to asterisk, I dont have wiring in place but I have time to get the wiring done. But I was thinking if I should put phones and PC's on same network as CAT6 wiring for PC's is already in place but I think i'll do it seperate |
| 13:15.30 | techman97_andy | good morning all - has anyone setup a faxing module in * where I can accept faxes and convert them to a TIF/PDF/JPG/whatever? |
| 13:15.51 | coppice | nope. nobody ever did that |
| 13:15.56 | techman97_andy | =P |
| 13:16.03 | Zeeek | hahaa |
| 13:16.13 | kavit | Greek-Boy: with some phones you can piggy back the computer off the back of the phone |
| 13:16.19 | Zeeek | they did but it's not perfect :) |
| 13:16.22 | kavit | Greek-Boy: look into 802.1Q |
| 13:16.26 | kavit | vlan |
| 13:16.42 | [TK]D-Fender | kavit : No... it wouldn't manage it, it'd only allow the endpoints to call each other. |
| 13:16.43 | Assid | gimme a min.. setting up my line |
| 13:17.00 | [TK]D-Fender | Greek-Boy : are your PC really network heavy? |
| 13:17.04 | kavit | so you can use the same physical network but seperate vlan segments |
| 13:17.26 | Greek-Boy | [TK]D-Fender, not really heavy. It's a gigabit network |
| 13:17.31 | kavit | [TK]D-Fender: yeah I was trying to get asterisk to pass ISDN video calls to gnugk and openmcu |
| 13:17.33 | kavit | a lot of issues |
| 13:17.51 | kavit | i am just going to give up or wait until someone develops something |
| 13:17.54 | [TK]D-Fender | Greek-Boy : I mean are your PC's going to be constantly working on huge files? because pluggin in-line with your phone will drop everything to 10/100 |
| 13:18.18 | [TK]D-Fender | kavit : Well you are definately working outside my field of experience.... |
| 13:18.26 | codebreaker | kavit: fixed. a typo in /etc/odbc.ini "only" one s to much in [asterissk] |
| 13:18.35 | Greek-Boy | not huge files all the time, just database stuff |
| 13:18.49 | kavit | [TK]D-Fender: i am working outside my own field of experience as well |
| 13:19.30 | kavit | [TK]D-Fender: I though I would get back into hardcore programming and write a few enhancements to openmcu, I looked at the oh323 code |
| 13:19.41 | kavit | and just shook my head and gave up |
| 13:19.43 | [TK]D-Fender | Greek-Boy : Typically I'd trust a decent phone to take care of you just fine. |
| 13:19.51 | [TK]D-Fender | Greek-Boy : What models are you considering? |
| 13:19.53 | kavit | codebreaker: ah simple error yet a show stopper |
| 13:20.31 | Greek-Boy | ST-302 IAX2 IP Phone |
| 13:20.37 | Greek-Boy | but also polycom |
| 13:20.56 | kavit | Greek-Boy: I did an Cisco VoIP roll out for a major client and we used vlans, really for normal workstation stuff piggybacking off the phone is good enough |
| 13:21.03 | Greek-Boy | i also looked at linksys sipura phones |
| 13:21.09 | Assid | Zeeek: satish@spherelinx.com |
| 13:21.10 | kavit | polycom are generally good |
| 13:21.11 | Assid | call that |
| 13:21.17 | Assid | sip uri |
| 13:21.23 | Greek-Boy | i'll consider vlans kavit |
| 13:21.38 | Greek-Boy | but its generally best practice to seperate voip network, right? |
| 13:21.59 | kavit | Greek-Boy: what for? unless you are hosting stuff no |
| 13:22.01 | [TK]D-Fender | Hahah, craptastic SIP hardphone suggestion for * article on NewsForge :D |
| 13:22.23 | coppice | page filler of the week |
| 13:22.33 | [TK]D-Fender | Greek-Boy : Ummm... Don't know that phone and trust it even less... Go with Polycom or Cisco for sure if you're going in-line. |
| 13:22.44 | Greek-Boy | what wiring works better? CAT5 or CAT6? I know this might sound like a stupid question but a few network components out there work better with CAT5 and this is why i'm asking even though CAT6 can handle higher throughput |
| 13:22.51 | Assid | Zeeek: alternatively 3001@spherelinx.com |
| 13:22.56 | Assid | or |
| 13:23.00 | [TK]D-Fender | Greek-Boy : Irrelevent since they're stuck on 10/100. Cat5 = fine |
| 13:23.07 | Assid | *75493001 |
| 13:23.09 | kavit | Greek-Boy: also a word of advice, make sure all the workstations and the network card drivers support 802,1Q or you will find vlans will mess everything up |
| 13:23.21 | Greek-Boy | hmmm, ok |
| 13:23.22 | Assid | they all will work |
| 13:23.34 | Assid | Zeeek ? |
| 13:23.36 | Greek-Boy | which phones are better between cisco and polycom? |
| 13:23.58 | kavit | I had to mess around for 3 days rechecking all my cisco switch config and turns out the customers ARCHAIC SOE was to blame |
| 13:24.02 | kavit | i was not happy |
| 13:24.20 | kavit | Greek-Boy: CIsco if you have the cash |
| 13:24.34 | kavit | Greek-Boy: their phones are generally very reliable. |
| 13:24.45 | Greek-Boy | even better voice quality then polycom? |
| 13:25.11 | Alystair | hmmm |
| 13:25.29 | kavit | Greek-Boy: voice quality has a lot of factors that come into play like bandwidth, codecs etc but on an internal network |
| 13:25.33 | Greek-Boy | I also need to make sure that the phones support PoE as i want power to be distributed from a central point. I will have a 10KVA UPS to supply power to all phones, hope that will keep them going for atleast an hour or two |
| 13:25.40 | kavit | Greek-Boy: it shouldnt matter |
| 13:25.49 | kavit | Greek-Boy: Cisco supports PoE |
| 13:26.04 | kavit | Greek-Boy: even their Linksys models do |
| 13:26.14 | [TK]D-Fender | Greek-Boy : I'd say they're about equal, but Polycom being a far better value. |
| 13:26.24 | *** join/#asterisk robin_sz (n=robin@adsl.redpoint.org.uk) |
| 13:26.28 | kavit | Alystair: whats your budget? I found Grandstream GXP 2000 to be ok for a low priced phone |
| 13:26.33 | Assid | i have only one issue |
| 13:26.37 | Assid | i cant hear crap |
| 13:26.44 | Zeeek | Assid sorry I was feeding my face |
| 13:26.45 | Assid | prolly cause of routing |
| 13:26.51 | Assid | but more or less works |
| 13:26.53 | kavit | Assid: go to #windows :P crap talks there |
| 13:26.54 | [TK]D-Fender | GS = Plagued. A "to be avoided" brand onmy list... |
| 13:27.02 | Zeeek | hahha |
| 13:27.13 | Greek-Boy | i was thinking of buying d-link PoE 48-port switches but I might as well as get cisco for that too since i'm going to be buying cisco phones |
| 13:27.29 | kavit | Greek-Boy: cisco are very expensive |
| 13:27.35 | robin_sz | ohh goody, new GXP 2000 firmware at last, and it fixes my GUI problem |
| 13:27.52 | kavit | Greek-Boy: esp 7940 and upwards |
| 13:28.01 | [TK]D-Fender | Greek-Boy : Save a bundle of cash and get a D-Link DES-1526 24 port PoE switch and some Polycom PoE phones.... |
| 13:28.18 | robin_sz | and Ive won the lotto, and been appointed chief shower-keeper for the female showers at the next olympics |
| 13:28.44 | Zeeek | robin_sz better the next Vegas revue, doncha think? |
| 13:28.49 | mut | why, theres no boobs at the olympics |
| 13:28.50 | kavit | Greek-Boy: if you really want to go the cisco path cisco switches work really well with Vlans |
| 13:29.08 | Zeeek | go rent "Showgirls" again :) |
| 13:29.09 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
| 13:29.10 | kavit | [TK]D-Fender: i wouldnt trust dlink with proper vlan and qos |
| 13:29.11 | Greek-Boy | [TK]D-Fender, then I'll have to buy 5 switches for all the phones, lol |
| 13:29.12 | robin_sz | Vegas? no . thats a clean living mormon town ... |
| 13:29.29 | Zeeek | Watch the Sopranos |
| 13:30.53 | DoktorGreg | Sopranos has gotten boring |
| 13:30.53 | [TK]D-Fender | kavit : They support it and we are talking a minimal install here. |
| 13:30.53 | Zeeek | is anyone actually using e164 ? |
| 13:30.54 | DoktorGreg | its time for them to have a masacre and kill off everyone |
| 13:30.54 | Assid | Zeeek: did you try |
| 13:30.54 | Zeeek | What season you on? |
| 13:30.54 | robin_sz | actually, I think there is more chance of either if those things happening than my GXP getting fixed :( |
| 13:30.54 | [TK]D-Fender | Greek-Boy : True, but look at the COST of those switches... $400 USD |
| 13:30.54 | Hmmhesays | so i specifically arrange for a ride to work this morning |
| 13:30.54 | Zeeek | DoktorGreg Tony already is bleeding to death in the kitchen |
| 13:30.54 | Hmmhesays | and she doesn't show up |
| 13:30.54 | kavit | [TK]D-Fender: ah ok, well I have just had a lot of issues with D-Link hardware |
| 13:30.55 | DoktorGreg | shut up! |
| 13:30.55 | DoktorGreg | omg |
| 13:30.55 | coppice | then they'll rehash all the stories as the Tenors, and everyone will think its fresh and new |
| 13:30.55 | [TK]D-Fender | kavit : I've been trouble free..... |
| 13:30.55 | Greek-Boy | [TK]D-Fender, they dont come in 48-port? |
| 13:30.55 | Zeeek | DoktorGreg the last season first epsode is free on the net by the way |
| 13:31.01 | kavit | Greek-Boy: is it your money? or is it your companys money? |
| 13:31.04 | Zeeek | legally thur google video |
| 13:31.05 | DoktorGreg | thanks |
| 13:31.08 | *** join/#asterisk C4T3l (n=rcall01@216.54.143.2) |
| 13:31.09 | Greek-Boy | company money |
| 13:31.12 | Greek-Boy | lol |
| 13:31.12 | Greek-Boy | :) |
| 13:31.19 | robin_sz | spend it like rain then |
| 13:31.25 | [TK]D-Fender | Greek-Boy : Not yet... but we're talking minimal patching here... how much are 48 port Cisco-capable PoE switches going to cost you? |
| 13:31.38 | Zeeek | Assid I haven't tried anything but I can see the routing works because asterisk detects the loop |
| 13:31.50 | Assid | cool |
| 13:31.52 | Zeeek | I never talk on the phone anyway, I just TEST them ;) |
| 13:31.57 | Assid | haha |
| 13:32.02 | Assid | well |
| 13:32.05 | coppice | do cisco have 48 port switches with 48 ports of PoE now? |
| 13:32.05 | Assid | it works nice |
| 13:32.18 | Zeeek | unlike faxes which I wouldn't send for fun |
| 13:32.25 | kavit | Greek-Boy: if you work for someone else, i suggest you bulk buy cisco hardware and talk your account manager into getting you signed up for a few discounted courses so you get paid to go and sit in a course for a week or two and your boos gets jealous |
| 13:32.38 | DoktorGreg | I need like a 100 person conference call for a day |
| 13:32.40 | kavit | boss even |
| 13:32.48 | Zeeek | DoktorGreg see junction networks |
| 13:32.50 | Greek-Boy | lol kavit |
| 13:32.53 | Greek-Boy | nice thinking |
| 13:32.53 | Greek-Boy | :) |
| 13:33.02 | DoktorGreg | Zeeek, thx |
| 13:33.25 | Greek-Boy | [TK]D-Fender i know what u mean, its all about cost-effectiveness. If the company pays for cisco then I'll consider that route |
| 13:33.26 | Zeeek | there is a company that does just conferences too but I can't remember their name |
| 13:33.37 | *** join/#asterisk ComputerWarm (n=donc@HS196-230-97.nt.net) |
| 13:34.01 | kavit | [TK]D-Fender: they cost a lot cisco switches, we had a catalyst 6500 switch chassis |
| 13:34.10 | *** part/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 13:34.31 | kavit | Greek-Boy: i have been contracting for the last few years and you learn how to milk the clients who make life hard for you |
| 13:34.34 | lucifr | nite every1 |
| 13:34.49 | codebreaker | somebody a pointing link for voicemail in database without *local* wav files? |
| 13:34.52 | lucifr | wait... good morning.... lol |
| 13:35.06 | lucifr | let me say.... goodbye |
| 13:35.15 | *** part/#asterisk lucifr (n=chatzill@c-24-126-108-87.hsd1.ca.comcast.net) |
| 13:35.45 | kavit | i feel like a dark evil cloud of misspelling has been lifted off this channel |
| 13:36.18 | [TK]D-Fender | Greek-Boy : Yeah, typically lets say that PoE Polycom IP 501's = $200 USD and the Swich = $400 (16$/port) = $216 / ext. |
| 13:36.27 | Assid | whats better.. making a new extension.. or a goto ? |
| 13:36.28 | DoktorGreg | omg charge PITA premiums |
| 13:36.37 | DoktorGreg | PITA premiums are the norm |
| 13:36.59 | [TK]D-Fender | Assid : That makes no sense... |
| 13:38.27 | Assid | okay i have exten => assid,1,Dial....... and i have exten => 3001,1,Dial....... both have the same dial |
| 13:39.11 | Assid | just wondering if it makes any difference to the processor |
| 13:40.28 | lunk | how can i pass in an arbitrary dial-plan variable with a Call file? |
| 13:41.15 | C4T3l | hello world |
| 13:42.02 | Assid | anyone behind a nat? |
| 13:42.11 | Assid | and can help me test a sip uri call? |
| 13:42.45 | Zeeek | how long do these ezdial calls take to happen? |
| 13:42.52 | [TK]D-Fender | Assid : Goto or Macro...... |
| 13:43.31 | Zeeek | Assid, by the way if you don't hear anything, notice the canreinvitie=yes in their sip entries? If yiou're behind nat that may be why |
| 13:44.06 | Hmmhesays | or if you have a t1/e1 card in with nothing plugged into it |
| 13:44.16 | the_magic_bean | when using RealTime for configuration and whatnot with a sql db, does asterisk only store things like users in the sql db or would voicemail files get stored in sql also? |
| 13:44.19 | Assid | Zeeek: i only get that problem with sip uri direct calling |
| 13:44.37 | *** part/#asterisk ComputerWarm (n=donc@HS196-230-97.nt.net) |
| 13:45.48 | Zeeek | so I should be able to call FWD and use the code to call back to asterisk. I think I did that once and there was like 50000ms of lag, no kidding |
| 13:45.59 | Hmmhesays | 50 seconds? |
| 13:45.59 | Zeeek | 5000 |
| 13:46.10 | Zeeek | no 5-10 seconds |
| 13:46.24 | Hmmhesays | nice math |
| 13:46.26 | Zeeek | and it was an octave lower than my voice! |
| 13:46.31 | file | wow, my bank actually has Euros |
| 13:46.52 | Zeeek | what? here a ms is 1/10,000 of a second! |
| 13:47.01 | Hmmhesays | haha |
| 13:47.08 | Zeeek | on national holidays especially |
| 13:47.11 | Hmmhesays | sorry I missed a booty call this weekend, i'm a little uptight |
| 13:47.21 | file | poor Hmmhesays |
| 13:47.32 | Zeeek | missed because? |
| 13:47.40 | Hmmhesays | I was incarcerated |
| 13:47.41 | kavit | [TK]D-Fender: i like the new ekiga, it is good |
| 13:47.53 | Zeeek | e-geek-a |
| 13:48.00 | [TK]D-Fender | kavit : I"m still waiting for a Win32 build as I don't use LInux as a desktop platform... |
| 13:48.18 | Dr-Linux | what is Ekiga? |
| 13:48.42 | kavit | [TK]D-Fender: maybe its time to stop playing Battlefield or world of war craft :P |
| 13:48.53 | codebreaker | the_magic_bean: there a two ways i ve heared. one ist storing only users and an other approuch saves also all files( this one i am also looking for) when i find the link i will you inform |
| 13:49.51 | Hmmhesays | jail, any type of jail sucks |
| 13:50.15 | file | Hmmhesays: didn't get no lovin' in jail? :P |
| 13:50.42 | Hmmhesays | well, it was min security, but no conjecalK(sp?) visits |
| 13:51.09 | vader-- | morning |
| 13:51.17 | Hmmhesays | seriously if you live in fargo and haven't been to jail at least once... my god you must be bored |
| 13:51.28 | vader-- | hi d-fender, hmm, dr |
| 13:51.30 | [TK]D-Fender | kavit : I only have 2 games, HL2 & Diablo2. And I harly ever play either :) |
| 13:51.58 | [TK]D-Fender | Hmmhesays : this is for that car trashing your GF gave you? |
| 13:52.12 | Hmmhesays | [TK]D-Fender: X girlfriend yes |
| 13:52.27 | kavit | [TK]D-Fender: last game i REALLY got into was Carmageddon 2 |
| 13:52.28 | [TK]D-Fender | Hmmhesays : Avoid drunks..... |
| 13:52.33 | kavit | AGES ago |
| 13:52.42 | codebreaker | the_magic_bean: http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage |
| 13:52.51 | Hmmhesays | [TK]D-Fender: no, now when I got out there is a 20 hidden in my wallet for a cab ride no matter what |
| 13:53.15 | lunk | what is the syntax for the Data: prompt in a .call file? |
| 13:53.19 | ghost99 | hi all Wondering if someone can help, I don't seem to be able to dial out, I have the following exten => _9.,1,Dial(Zap/1-1/${EXTEN:1}) in the context [sip] in extensons.cof wher should it be ? |
| 13:54.08 | [TK]D-Fender | Hmmhesays : Seriously blows... whats the insurance situation? |
| 13:54.23 | [TK]D-Fender | ghost99 : Bad Zap format |
| 13:54.24 | Hmmhesays | insurance paid for it, I have no car now |
| 13:54.27 | Hmmhesays | no license |
| 13:54.38 | [TK]D-Fender | ghost99 : exten => _9.,1,Dial(Zap/1/${EXTEN:1}) |
| 13:54.39 | Hmmhesays | $1000 dollar fine, 5 days in jail |
| 13:54.53 | file | Hmmhesays: but now you're back! |
| 13:55.01 | Hmmhesays | file: i go back tonight at 6 |
| 13:55.03 | [TK]D-Fender | Hmmhesays : Wait, what did you hit, and WHO was covered? |
| 13:55.14 | file | well, back for a bit |
| 13:55.21 | DoktorGreg | does anybody have a fax retreive script? |
| 13:55.23 | [TK]D-Fender | Hmmhesays : And you killed your points on this did you? |
| 13:55.27 | Hmmhesays | I was covered, and she pulled the steering wheel hard to the right on a residential street |
| 13:55.32 | DoktorGreg | go up to fax machine |
| 13:55.40 | DoktorGreg | dial into an extension |
| 13:55.50 | Hmmhesays | I was just over the limit, passed my field sobriety test, failed breath |
| 13:55.54 | DoktorGreg | extension starts transmitting stored faxes |
| 13:56.01 | Hmmhesays | so... I got busted for dui |
| 13:56.08 | Hmmhesays | that's why all the fines |
| 13:56.14 | DoktorGreg | Hmmhesays, just get a lawyer |
| 13:56.25 | Hmmhesays | city prosecutor wouldn't budge |
| 13:56.40 | Hmmhesays | I have some other stuff on my record from my mischevious years |
| 13:56.51 | Hmmhesays | like I said, this is fargo... you'd know if you lived here |
| 13:56.57 | DoktorGreg | lol |
| 13:57.00 | [TK]D-Fender | Hmmhesays : Not your first DUI I take it having lost your license, or are you too new a driver? |
| 13:57.06 | DoktorGreg | Im originally from vally city |
| 13:57.12 | MikeJ[Laptop] | hmmmm he says |
| 13:57.25 | file | uh oh it's MikeJ |
| 13:57.26 | Hmmhesays | then you know DoktorGreg: there is nothing to do around here except work, and get in trouble |
| 13:57.27 | file | everyone hide |
| 13:57.38 | DoktorGreg | cookie salad! |
| 13:57.48 | Hmmhesays | [TK]D-Fender: no, had one way back when I actually deserved it |
| 13:57.52 | C4T3l | i'm running * 1.2.7.1 and having problems with one way communications using SIP i'm assuming its a firewall issue. As far as I know both ends are free of firewalling. The far end can recieve calls and listen to pre-recorded messages on my *. But I can't hear anything on my end. Any suggestions? |
| 13:57.56 | Hmmhesays | I mean, really deserved it |
| 13:57.57 | DoktorGreg | I get about 20 new cookie salad recipes every time i go back |
| 13:58.23 | DoktorGreg | sheesh, move to mineapolis |
| 13:58.35 | DoktorGreg | North Dakota is a dead end |
| 13:58.41 | ghost99 | [tk]D-fender: I tried it that way also took out of sip context and put into global context in extensions.conf and still no luck .. any more cluses ? |
| 13:58.41 | Hmmhesays | DoktorGreg: actually i was on my way out of here when jamie helped my car into the junkyard |
| 13:59.13 | [TK]D-Fender | Hmmhesays : Next time plow her back to her side.... |
| 13:59.26 | [TK]D-Fender | ghost99 : HUH!? |
| 13:59.29 | Hmmhesays | [TK]D-Fender: there will never be a next time |
| 13:59.37 | Hmmhesays | plus the girl i'm seeing now is pretty cool |
| 13:59.38 | [TK]D-Fender | Hmmhesays : Not with HER at least... |
| 14:00.07 | Hmmhesays | [TK]D-Fender my life is fscked enough becuase of this, it will not happen, EVER again |
| 14:00.38 | DoktorGreg | the girl also owes you a sympath fuck! |
| 14:00.44 | jake1932 | C4T3l: if you run sip debug, or rtp debug, you can see the ip addresses and ports of the endpoints |
| 14:00.57 | Hmmhesays | I'd rather she pay half the fines |
| 14:00.57 | [TK]D-Fender | DoktorGreg : Oh I think she fucked him pretty good already |
| 14:01.04 | Hmmhesays | could pay for a much hotter hooker |
| 14:01.06 | Hmmhesays | LOL |
| 14:01.25 | Hmmhesays | [TK]D-Fender: in every way conceivable |
| 14:01.31 | C4T3l | jake1932: thanks. I've only been using asterisk for a month so theres alot i'm not aware of |
| 14:01.33 | Hmmhesays | more bad than good |
| 14:02.22 | jake1932 | C4T3l: are both enpoints (your asterisk box and the device) both on the same network? |
| 14:02.31 | *** join/#asterisk Lino` (n=Lino@i577BD559.versanet.de) |
| 14:02.36 | C4T3l | jake1932: No. diff nets |
| 14:02.46 | jake1932 | C4T3l: both have public IPs? |
| 14:02.48 | Hmmhesays | on the upside, I think i'm going to make it to cluecon this year |
| 14:02.53 | C4T3l | jake1932: yes |
| 14:03.20 | jake1932 | C4T3l: maybe you can pastebin a sip debug. - i'll take a quick look |
| 14:04.19 | Hmmhesays | although it has been interesting, I had to make someone my bitch and shank a guy the first day |
| 14:04.29 | Itburnz | hey does anyone of you know about a problem regarding asterisk 1.2.7.1 & spanDSP ? i get the error "app_rxfax.so: undefined symbol: t30_get_far_ident" - was wondering if anyone here knows a solution |
| 14:04.40 | Hmmhesays | missing some header |
| 14:04.54 | Hmmhesays | or the name of that function changed |
| 14:04.56 | C4T3l | jake1932: i wont be able to do that right away. I'm at work right now (this is an unrelated work prob. ) |
| 14:05.15 | jake1932 | oh |
| 14:05.21 | C4T3l | jake1932: It's really just for fun at home for this point |
| 14:05.24 | Itburnz | hrm, it's a fresh asterisk installation |
| 14:05.32 | Hmmhesays | ahh tech for fun at home |
| 14:05.59 | Itburnz | at least that explains why recompiling didnt do the job |
| 14:06.01 | C4T3l | I guess its not everyone's idea of fun |
| 14:06.17 | C4T3l | lol |
| 14:07.09 | [TK]D-Fender | same damn thing! |
| 14:07.18 | Hmmhesays | fantastic |
| 14:07.25 | Hmmhesays | although i'd probably fail my ua tonight |
| 14:07.35 | file | Hmmhesays: so who is your bitch? |
| 14:07.38 | [TK]D-Fender | UA? |
| 14:07.43 | Hmmhesays | piss test |
| 14:07.45 | [TK]D-Fender | Ah |
| 14:08.00 | Hmmhesays | although it is min security, they frown upon that type of thing |
| 14:08.05 | [TK]D-Fender | file : Careful with words like that.... he just got out of jail.... |
| 14:08.20 | file | [TK]D-Fender: true |
| 14:08.28 | Hmmhesays | [TK]D-Fender i got 3 days left |
| 14:08.32 | Hmmhesays | i'm out on work release |
| 14:08.43 | [TK]D-Fender | Hmmhesays : like house arrest, only let out to work? |
| 14:09.02 | [TK]D-Fender | Gotcha |
| 14:09.02 | Hmmhesays | kind of yeah |
| 14:09.02 | Hmmhesays | I have them convinced i'm self employed and on call 24/7 though |
| 14:09.06 | [TK]D-Fender | Hmmhesays : What did you hit exactly? |
| 14:09.12 | file | need us to call you randomly? |
| 14:09.18 | Hmmhesays | file: that's what cron is for |
| 14:09.18 | eric_hill | :) |
| 14:09.26 | *** join/#asterisk IceManRISK (n=kart@201.15.214.182) |
| 14:09.28 | file | :D |
| 14:09.45 | Hmmhesays | it's actually pretty funny, they come running with my phone when it rings |
| 14:09.51 | kavit | [TK]D-Fender: yeah I just gave up on the whole h323 thing for now |
| 14:09.52 | [TK]D-Fender | Yeah you're chronic SOMETHING all right... |
| 14:09.58 | kavit | too much pain |
| 14:09.59 | Hmmhesays | a 1991 volvo |
| 14:10.37 | [TK]D-Fender | Hmmhesays : Parked or mobile? |
| 14:10.40 | Itburnz | hrm i installed spandsp in /usr/include/spandsp... how i can check where the app_rxfax.so tries to access the t30 header ? |
| 14:10.52 | Hmmhesays | Parked: we were driving down a residential street at 2:30am |
| 14:11.10 | *** join/#asterisk austinnichols102 (n=austinni@70.46.69.131) |
| 14:11.11 | Hmmhesays | I gave her a pretty good shove off me and she grabbed the steering wheel |
| 14:12.41 | *** join/#asterisk azzie (n=az@azzie.net) |
| 14:13.18 | Hmmhesays | I missed that reference |
| 14:15.43 | Hmmhesays | was it funny or something else |
| 14:16.51 | [TK]D-Fender | Hmmhesays "Stupid girls" |
| 14:17.04 | Hmmhesays | ahhh |
| 14:17.22 | *** part/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 14:17.34 | *** join/#asterisk LoRez (i=lorez@freenode/staff/lorez) |
| 14:17.34 | Hmmhesays | no guitar for 5 days, i'm going to have to learn to play all over again, lol |
| 14:18.33 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 14:18.35 | [TK]D-Fender | I think it fitting you start out with some Elvis :) |
| 14:18.50 | *** join/#asterisk BadPacket (n=root@unaffiliated/badpacket) |
| 14:18.55 | Hmmhesays | I was thinking "old time rock n roll" by bob seger |
| 14:19.05 | file | pesky kids |
| 14:19.07 | [TK]D-Fender | I was thinking "Jail House Rock" :) |
| 14:19.12 | Hmmhesays | LOL |
| 14:19.18 | Hmmhesays | i think i could play that lefty |
| 14:20.03 | [TK]D-Fender | You're not "over the hill yet".... if only she hit the accelerator at the same time you'd have gained sufficient velocity :D |
| 14:20.34 | Hmmhesays | ok stop making references I don't get |
| 14:20.53 | Hmmhesays | jail is making me dumberer |
| 14:20.54 | [TK]D-Fender | Going airborne instead of stopping cold on the crash. |
| 14:20.59 | Hmmhesays | ahh lol |
| 14:21.06 | file | airborne express! |
| 14:21.43 | [TK]D-Fender | I am tending to speak in the "implied" tense. Works good with those that handle context & stream of consciousness well.... |
| 14:21.45 | kavit | is asterisk business edition released yet? |
| 14:21.56 | [TK]D-Fender | Not to be confused with "Field & Stream" ;) |
| 14:22.01 | [TK]D-Fender | kavit : Serveral |
| 14:22.04 | Hmmhesays | I don't fish or hunt |
| 14:22.05 | kavit | i read Marks interview somewhere |
| 14:22.07 | file | kavit: it's been released for some time.. |
| 14:22.23 | kavit | file: i dont get out much :( |
| 14:22.25 | MikeJ[Laptop] | does ABE have version numbers?? |
| 14:22.31 | MikeJ[Laptop] | kavit, get out more |
| 14:22.33 | file | it's got letters |
| 14:22.38 | [TK]D-Fender | F |
| 14:22.39 | MikeJ[Laptop] | what letter are they on? |
| 14:22.40 | [TK]D-Fender | :O |
| 14:22.48 | file | ... A |
| 14:23.10 | file | I wonder what happens when it reaches Z |
| 14:23.16 | [TK]D-Fender | file : What is the approximate std release equivalent for it? |
| 14:23.16 | MikeJ[Laptop] | like hurricanes? |
| 14:23.20 | Hmmhesays | AA |
| 14:23.22 | Hmmhesays | BB |
| 14:23.23 | [TK]D-Fender | file : LIke Excel.... AA :D |
| 14:23.31 | [TK]D-Fender | Hmmhesays : no, AB :) |
| 14:23.34 | Hmmhesays | sometime in 2050 |
| 14:23.40 | file | [TK]D-Fender: of A? I have no clue |
| 14:23.42 | Hmmhesays | when hopefully i'll be long dead |
| 14:23.48 | Hmmhesays | or living on a beach somewhere |
| 14:23.50 | [TK]D-Fender | Believe me Asterisk will not last that long. |
| 14:24.08 | [TK]D-Fender | Hmmhesays : Hefner seems to be doing OK for his age... |
| 14:24.15 | Hmmhesays | i'm not that rich |
| 14:24.24 | Hmmhesays | 1 he can pay for good medicine |
| 14:24.25 | MikeJ[Laptop] | there may not even be phones in 50 years :P |
| 14:24.34 | Hmmhesays | 2 surrounded by beautiful women constantly |
| 14:24.35 | C4T3l | just a quick question: What's an acceptable load avg for a Xeon (dual core) running * with 200+ users? |
| 14:24.48 | C4T3l | sip users |
| 14:24.56 | file | C4T31: that depends |
| 14:24.58 | C4T3l | er peers |
| 14:25.03 | [TK]D-Fender | C4T3l : Are they oall on coffee break? |
| 14:25.07 | MikeJ[Laptop] | C4T3l, till it makes stuff break it's good, once it starts breaking, it's bad |
| 14:25.18 | C4T3l | good one |
| 14:25.19 | Hmmhesays | you should be able to cook an egg on your processor(s) |
| 14:25.21 | file | SO - IAXtel survived the night with my latest mods... |
| 14:25.30 | MikeJ[Laptop] | file, I can fix that |
| 14:25.46 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 14:25.48 | file | I bet |
| 14:26.07 | MikeJ[Laptop] | it's on how many boxes? |
| 14:26.09 | MikeJ[Laptop] | just 1? |
| 14:26.14 | C4T3l | just 1 |
| 14:26.26 | MikeJ[Laptop] | C4T3l, not you, iaxtel |
| 14:26.35 | C4T3l | sorry :) |
| 14:26.43 | file | yessir, just 1 |
| 14:26.52 | MikeJ[Laptop] | I wonder how many calls I can send to file at once |
| 14:26.55 | MikeJ[Laptop] | hmmmm |
| 14:27.03 | kavit | file what sort of mods? |
| 14:27.04 | Hmmhesays | gotta be at least 5 or 6 |
| 14:27.08 | BugKham | the debug shows chan_sip.c: Failed to authenticate user "asterisk" when calling from another server |
| 14:27.16 | BugKham | iax calls work fine |
| 14:27.19 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 14:27.30 | file | kavit: some updates... that'll get merged into trunk once they get tested by someone else who has been having troubles as well |
| 14:28.12 | BugKham | insecure=port,invite on the [general] section, what else should I look at? |
| 14:28.14 | file | whomever coco_tseng is, you're not registered |
| 14:28.15 | kavit | ah, file do you work for digium? |
| 14:28.24 | MikeJ[Laptop] | file broke it! |
| 14:28.26 | file | kavit: yes |
| 14:28.32 | kavit | or do you commit out of the kindness of your heart |
| 14:28.50 | file | I'm commited! |
| 14:28.52 | kavit | heh file, planning to ask for a raise now that the business grade enterprise version is out? |
| 14:28.58 | MikeJ[Laptop] | kavit, he has no kindness inhis heart.. only bitterness |
| 14:29.05 | ringe | I have to open the firewall to be able to place two-way calls between two astersisk servers, right? |
| 14:29.06 | file | kavit: it's been out for awhile |
| 14:29.10 | file | like, a year |
| 14:29.30 | *** join/#asterisk littleball (n=littleba@cm55.epsilon171.maxonline.com.sg) |
| 14:29.42 | MikeJ[Laptop] | there is now a business grade enterprise version of file? |
| 14:29.50 | kavit | MikeJ[Laptop]: he must have tried to fix h323, enough to make anyone bitter |
| 14:29.55 | file | isn't he talking about BE? |
| 14:29.57 | *** join/#asterisk jahani (n=k@adsl196-213-242-217-196.adsl196-16.iam.net.ma) |
| 14:30.00 | Hmmhesays | who can we get on the case we need .... |
| 14:30.04 | file | if not then I have no clue what you're talking about |
| 14:30.05 | MikeJ[Laptop] | fix h323? |
| 14:30.22 | kavit | file: I didnt see the year on the article, need sleep |
| 14:30.30 | MikeJ[Laptop] | go sleep |
| 14:31.14 | *** part/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 14:32.41 | file | sleep is evil, don't listen to him |
| 14:33.04 | ManxPower | Cell phone companies suck |
| 14:33.10 | Hmmhesays | yes |
| 14:33.19 | file | yes, they do |
| 14:33.23 | kavit | certainly not, I just need to replenish the redbull content of my blood stream |
| 14:33.34 | ManxPower | Verizon expects me to pat a $583 bill, which I didn't even know I had until a week ago, without any copy of the charges. |
| 14:33.58 | mut | too many 1-900 numbers eh |
| 14:33.59 | file | they won't send you the stuff explaining how it's $583? |
| 14:34.02 | kavit | ManxPower: tell them very politely where they can stick it |
| 14:34.03 | file | nice. |
| 14:34.05 | Hmmhesays | where the hell did alltel wireless come from |
| 14:34.16 | ManxPower | kavit, I'll send them a letter by certified mail. |
| 14:34.42 | ManxPower | Hmmhesays, they are a regional cell company in the southeast (maybe other places). |
| 14:34.56 | Hmmhesays | their commercials claim they have the largest network in the US |
| 14:35.05 | ManxPower | kavit, they said that they already send it when they sent out the original bill who knows how long ago. |
| 14:35.07 | Hmmhesays | and even use characters from other cell companies |
| 14:35.08 | kavit | "Customers are not supposed to be scared of their companies, Companies are supposed to be scared of their customers" |
| 14:35.58 | kavit | ManxPower: thats a standard reply, their default stance is "We are always right, we can never be wrong, we have more money than you we will do what we want" |
| 14:36.41 | kavit | ManxPower: atleast they arent partially government owned like telstra here... probably the worst telco on this planet |
| 14:37.34 | mut | heh |
| 14:37.40 | mut | centurytel is the worst telco on the planet |
| 14:38.16 | kavit | which leads me to believe telcos suck as a rule. |
| 14:38.47 | Hmmhesays | it'll be cool when I can download full length dvd's to my cell phone |
| 14:38.57 | austinnichols102 | telcos suck (tm) |
| 14:39.19 | file | Hmmhesays: using multiplexing teletectonic data transfer?!? |
| 14:39.27 | Hmmhesays | um sure |
| 14:39.31 | Hmmhesays | whatever the fark that is |
| 14:39.44 | file | it's from ImaginaryTech(tm) |
| 14:39.45 | mut | tap your foot on the ground, morse code man! |
| 14:39.57 | Ikarus | file: shaking the earth in different directions ? |
| 14:39.58 | Hmmhesays | file: cool, sounds like it has something to do with smashing your phone against a rock |
| 14:39.59 | *** join/#asterisk coppice (n=chatzill@153.192.17.210.dyn.pacific.net.hk) |
| 14:40.01 | Ikarus | AM of FM ? |
| 14:40.51 | Hmmhesays | you think you'd have better luck on am with a rock |
| 14:41.47 | Hmmhesays | but what do I know |
| 14:42.30 | kavit | hey Tesla wanted to distribute power like that |
| 14:42.47 | Hmmhesays | they also wanted to rock your faces off |
| 14:43.00 | Alystair | haha |
| 14:43.01 | *** join/#asterisk PakiPenguin (i=uppal@linuxpakistan/admin/pakipenguin) |
| 14:43.11 | Alystair | "multiplexing teletectonic data transfer" |
| 14:43.22 | I-MOD | mut: whats up with centurytel? |
| 14:43.23 | Hmmhesays | so that would be hitting a rock with two phones at once |
| 14:43.35 | brettnem | wait a minute.. telcos don't suck.. |
| 14:43.39 | Alystair | you send faxes via earthquake |
| 14:43.55 | brettnem | ITSPs suck |
| 14:43.57 | kavit | Hmmhesays: i dont know which Tesla you are talking about, but Tesla had no time for music |
| 14:44.27 | Hmmhesays | go download "modern day cowboy" |
| 14:44.41 | kavit | ah |
| 14:44.53 | kavit | I meant Tesla the inventor of AC current |
| 14:45.01 | Hmmhesays | kavit: you have no sense of humour |
| 14:45.29 | brettnem | you long hair hippie people |
| 14:45.39 | kavit | Hmmhesays: actually I have never heard of Tesla in context of music before |
| 14:45.47 | Hmmhesays | kavit: you're missing out |
| 14:46.00 | brettnem | I like tesla and not just for the coil |
| 14:46.06 | Hmmhesays | the screaming vocals of jeff keith and guitar of frank hannon and tommy scheoch |
| 14:46.10 | kavit | hahaha |
| 14:46.10 | coppice | brettnem: I used to be a long haired hippie, but my hair is shorter these days |
| 14:46.33 | Hmmhesays | they had some of the better dual lead solos of the late 80's and early 90's |
| 14:46.47 | brettnem | coppice: heh, me too.. until I realized I was that "dorky long hair computer guy" and I seriously needed a change.. so I cut my hair got a wife and 2.5 kids.. |
| 14:46.51 | wasim | b |
| 14:46.58 | Hmmhesays | where do you buy one of those |
| 14:47.05 | Hmmhesays | does it come in a package? |
| 14:47.10 | brettnem | Hmmhesays: Edmond scientific!! |
| 14:47.47 | Hmmhesays | hmm |
| 14:48.00 | kavit | brettnem: did you feel like RMS ? |
| 14:48.13 | LostFrog | shop, even. |
| 14:48.23 | brettnem | Here you go |
| 14:48.26 | brettnem | http://scientificsonline.com/product.asp_Q_pn_E_3070301 |
| 14:48.32 | Hmmhesays | they're so cute, little legs walking around by themselves |
| 14:49.05 | Hmmhesays | 50,000V what a rip off |
| 14:49.06 | brettnem | kavit: I may have been root, and maybe even square back then, but never mean |
| 14:49.10 | brettnem | haha |
| 14:49.17 | brettnem | "Watts up!" |
| 14:49.24 | brettnem | pff |
| 14:49.28 | Hmmhesays | go pick up an ingnition coil out of the junkyard for 10 bucks |
| 14:49.49 | Hmmhesays | newer ones put out about 100Kv |
| 14:50.22 | *** join/#asterisk robin_sz (n=robin@adsl.redpoint.org.uk) |
| 14:50.22 | LostFrog | Buy that and your amateur radio neighbors will hate you! |
| 14:50.38 | brettnem | whoo http://bellsouthpwp.net/B/u/BunnyKiller/tcoil.htm |
| 14:51.29 | brettnem | awesome |
| 14:51.36 | coppice | the latest trick in India to avoid paying for electricity is a little box. in it is a line output stage from a TV, about to generate about 30W at 25kV. They link this to the live wire of the mains, putting massive noise on it. This blows up most of the electronics in the house, including the electricity meter. |
| 14:52.09 | Ikarus | coppice: here they then proceed to bill you at the high end of "average" for the size house |
| 14:52.36 | Ikarus | Without giving you any oppertunity to prove you didn't do it on purpose (unless you reported it within a month or so) |
| 14:52.46 | coppice | how? you have a dead electricity meter. there's no way to prove how it got like that |
| 14:52.47 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 14:53.24 | Hmmhesays | unless you hit it with a 20lb sledge |
| 14:53.36 | coppice | the real downside is you can't keep doing it. the occassional dead meter is one thing, but failing between every reading it quite another |
| 14:53.53 | brettnem | heh |
| 14:54.02 | brettnem | this thing is cool |
| 14:54.03 | brettnem | http://bellsouthpwp.net/B/u/BunnyKiller/bigpig.html |
| 14:55.05 | DoktorGreg | slash dot and mac evangalists are a funny mix |
| 14:55.18 | *** join/#asterisk brif8 (n=Techno@lazyjtrainingcenter.com) |
| 14:55.21 | DoktorGreg | "independance day was dumbest movie ever" |
| 14:55.29 | Hmmhesays | I liked that movie |
| 14:55.32 | DoktorGreg | "But it had a mac in it" |
| 14:55.37 | Hmmhesays | back when Will Smith was all the rage |
| 14:55.53 | DoktorGreg | "like you can write an alien virus on a mac" |
| 14:56.08 | DoktorGreg | "But you CAN write an Alien virus on a max!" |
| 14:56.19 | DoktorGreg | lol |
| 14:56.28 | DoktorGreg | the gist of a thread on slashdot right now |
| 14:56.42 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 14:57.04 | *** join/#asterisk mosty (i=mostynm@adsl-137-244.swiftdsl.com.au) |
| 14:57.06 | DoktorGreg | The the post that explains how you write alien viruses on Mac gets modded +4 insiteful |
| 14:57.06 | Hmmhesays | I go to fark for my down time |
| 14:58.33 | *** join/#asterisk kink0 (n=k@62.37.205.161) |
| 14:59.10 | kink0 | hello |
| 14:59.43 | kink0 | I still suffering memory leakages problems with Asterisk 1.2.7 + h323, anybody can help ? |
| 15:00.52 | trelane_ | docelmo, what's the first rule of slashdot? |
| 15:00.58 | *** join/#asterisk nite (n=nite@gateway.digium.com) |
| 15:01.30 | LostFrog | trelane: You don't talk about /.? |
| 15:01.50 | Alystair | So, what are the nicest phones to use? I'm thinking Polycom? |
| 15:01.55 | trelane_ | LostFrog, THATS RIGHT! |
| 15:02.07 | eric_hill | LostFrog: Funny, that's the same as the second rule! |
| 15:02.10 | *** join/#asterisk GerbilWrk (i=GerbilNu@65.88.144.41) |
| 15:02.12 | *** join/#asterisk bweschke (n=bweschke@66.152.225.74) |
| 15:02.25 | trelane_ | eric_hill, amazing how that works :) |
| 15:02.36 | kavit | First Rule of Slashdot is Do not talk about slashdot? |
| 15:02.42 | eric_hill | Alystair: The ones that work for what you need. |
| 15:02.45 | trelane_ | kavit, right. |
| 15:02.54 | mosty | i thought the first rule of slashdot was check your brain at the door |
| 15:03.01 | trelane_ | mosty, wrong |
| 15:03.02 | Hmmhesays | does anyone else in here hate myspace.com |
| 15:03.06 | LostFrog | kavit: Watch "Fight Club." |
| 15:03.18 | trelane_ | Hmmhesays, yes, not that that's on topic |
| 15:03.27 | chapeaurouge | Alystair, using polycom here too.. very nice |
| 15:03.36 | Hmmhesays | trelane_: so? |
| 15:03.51 | mosty | i'm trying to figure out how to barge in on calls, is that possible in asterisk? |
| 15:03.56 | Hmmhesays | chanspy |
| 15:04.06 | Hmmhesays | or a manager redirect to a conference room |
| 15:04.11 | Hmmhesays | either will get you there |
| 15:04.39 | kavit | LostFrog: I have read it |
| 15:04.42 | GerbilWrk | I have an issue where two asterisk boxes are calling eachother via sip, and each have at least one g729 codec available. Their context is set to disallow all, and allow only g729. The call goes through, but not using g729 |
| 15:04.43 | kavit | Books are better |
| 15:04.44 | mosty | Hmmhesays: thanks |
| 15:05.06 | Hmmhesays | np: i had to say something on topic to keep trelane_ from skewering me |
| 15:05.16 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.66.Dial1.SanJose1.Level3.net) |
| 15:06.05 | trelane_ | what skewer? |
| 15:07.25 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.66.Dial1.SanJose1.Level3.net) |
| 15:11.53 | GerbilWrk | anyone have any ideas about the codec issue? |
| 15:11.55 | C4T3l | GerbilWrk: what codec is used instead? |
| 15:12.21 | *** join/#asterisk zaf (n=zaf@dsl081-237-122.lax1.dsl.speakeasy.net) |
| 15:12.25 | *** join/#asterisk Splas (n=jwb@206.252.198.101) |
| 15:12.40 | GerbilWrk | according to sip show channels, the form is g729 to the phones |
| 15:12.57 | C4T3l | GerbilWrk: have you attempted sip debug? |
| 15:13.09 | GerbilWrk | but, show g729 shows 0/0 inuse |
| 15:13.42 | mosty | gerbilwrk: you have registered the g729 licences with asterisk? |
| 15:13.52 | *** join/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 15:14.10 | ghost99 | could i ask if anyone could take a look at my dialplan etc and fix a big i have , I keep getting error 404 on my softphone and can't dial out .. would be nice if someone could help me out |
| 15:14.26 | GerbilWrk | i registered the codecs per digiums instructions, and asterisk does show there are 3 available on one server, and 1 on the other |
| 15:16.24 | mosty | gerbilwrk: if a call is g729 at both ends, asterisk doesn't need to use any g729 licences, only the phones do. perhaps if you set one of the phones to use a different codec, you would see asterisk using a licence to transcode to and from g729 |
| 15:16.41 | *** join/#asterisk DrDeke (i=Rusty@causticsoda.engin.umich.edu) |
| 15:17.44 | GerbilWrk | so the g729 codecs are only to the phones? not server to server? |
| 15:18.20 | mosty | if both ends of a call talk the same codec, asterisk does not interfere |
| 15:18.43 | mosty | however if they are both talking in a different codec, asterisk will translate |
| 15:19.04 | DrDeke | Can anyone point me in the direction of a SIP or IAX softphone that can use G.722? |
| 15:19.05 | GerbilWrk | so, in reality, once the calls connected, it's just phone to phone right? |
| 15:20.13 | mosty | gerbilwrk: depends how asterisk is setup |
| 15:20.46 | noname32 | hey question in the log does this mean it is using codec 2? Oooh, format changed to 2 |
| 15:21.49 | *** join/#asterisk brodiem (i=1000@cpe-66-69-222-36.austin.res.rr.com) |
| 15:22.47 | vader-- | have any of you guys posted anything to www.voip-info.org? |
| 15:23.10 | DrDeke | Ever? Yes, I have. |
| 15:25.27 | vader-- | i wanna post some info on a part number for a dell poweredge 2800 to make the digium TDM analog cards work with it |
| 15:25.39 | vader-- | the dell poweredge 2800 and 2850 don't ship with molex connectors |
| 15:26.01 | vader-- | so for you to be able to use FXS modules ya need this part |
| 15:26.09 | vader-- | not sure where to put i on the site though |
| 15:26.20 | DrDeke | hmm |
| 15:27.03 | Hmmhesays | wow, this itsp doesn't send a phone number lol |
| 15:27.06 | Hmmhesays | fantastic |
| 15:27.53 | DrDeke | which ITSP is that? |
| 15:28.29 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 15:28.29 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:29.24 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-2.rcmdva.fios.verizon.net) |
| 15:29.39 | *** part/#asterisk austinnichols102 (n=austinni@70.46.69.131) |
| 15:32.58 | Hmmhesays | axvoice |
| 15:33.18 | Hmmhesays | one of the eleventy billion generic ones out there |
| 15:33.26 | DrDeke | ahh, yeah |
| 15:33.30 | Hmmhesays | level 19 |
| 15:33.38 | Hmmhesays | and we're not talking D&D |
| 15:33.54 | DrDeke | hahaha |
| 15:34.04 | brettnem | AX voice? |
| 15:34.25 | brettnem | "pay your bill, or you'll get the AX!" |
| 15:34.32 | DrDeke | I find that my "customers" (family & friends) could almost not care less about voice quality so long as I give them free telephone calls. |
| 15:34.38 | DrDeke | Nevermind that I'm about tier-5 ;) |
| 15:34.40 | Hmmhesays | yeah pretty much |
| 15:34.43 | mut | in iax |
| 15:34.50 | mut | if i do bandwidth=high |
| 15:34.57 | mut | does that do ulaw codec or what? |
| 15:35.01 | Hmmhesays | A@H doesn't deal well with names for incoming calls |
| 15:35.21 | DrDeke | Yeah, it allows ulaw/alaw (the 64kbps payload codecs). You could just as well leave out the bandwidth line and set "disallow=all allow=ulaw allow=alaw" and whatever else you want |
| 15:35.47 | *** join/#asterisk pythos (i=pythos@unaffiliated/pythos) |
| 15:35.52 | pythos | mornin! |
| 15:35.59 | DrDeke | mornin |
| 15:36.00 | DrDeke | ' |
| 15:36.19 | Hmmhesays | boobies |
| 15:36.26 | Hmmhesays | (.)(.) |
| 15:36.32 | vader-- | do they just let anyone modify anything on voip-info.org's wiki? |
| 15:36.38 | Hmmhesays | if you're signed up |
| 15:36.47 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 15:36.49 | vader-- | what happens if someone delete's something? |
| 15:37.00 | VoicePulse | It keeps a history of changes that can be undone. |
| 15:37.07 | vader-- | gotcha |
| 15:37.08 | DrDeke | vader-- The wiki software keeps track of all changes that people make, and it is trivially easy to undo vandalism. |
| 15:37.32 | vader-- | i found a section for me to put that info |
| 15:37.36 | vader-- | http://www.voip-info.org/wiki/view/Asterisk+hardware |
| 15:37.43 | pythos | ok, so I got my TDM400P with 2 fxo and 2 fxs's installed, and such, as per report by DMESG. Now what do I do? <ok, Im lost in terms of what to do next, but I DID at least get the card recognized, etc.> |
| 15:37.46 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 15:38.22 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 15:38.29 | vader-- | did you modify your zaptel.conf to setup the ports? |
| 15:38.52 | vader-- | ya have to tell zaptel.conf what ports are fxs and what ports are fxo |
| 15:38.57 | pythos | nope. I just barely got the hardware stuff figured out. |
| 15:39.31 | pythos | Is there a good primmer, not to techie? Im kind of slow, as its Monday here all week long |
| 15:39.43 | Hmmhesays | I'll tell you my dirty little secret |
| 15:39.50 | Hmmhesays | don't tell anyone or you'll be just another regret |
| 15:40.15 | pythos | I got fxs on moduels 0 and 1, and fxo on 2,3 |
| 15:40.23 | *** join/#asterisk techie (n=gus@antibala.com) |
| 15:40.25 | puzzled | hi |
| 15:41.06 | jake1932 | dirty little secret |
| 15:41.18 | SpaceBass | pythos, nerdvittles.com and www.archatechs.com both have good primers up currently... focused on asterisk@home but the concepts are the same |
| 15:41.34 | Hmmhesays | ok there is something confusin in this freepbx dp |
| 15:41.44 | Hmmhesays | exten => s,1,Set(FROM_DID=s) |
| 15:41.45 | Hmmhesays | exten => s,n,Goto() |
| 15:41.45 | Hmmhesays | exten => _X.,1,Goto(ext-did,s,1) |
| 15:41.48 | Hmmhesays | why does that work |
| 15:42.06 | Hmmhesays | and the first person to give me shit about a 3 line paste gets their foot nailed to the floor |
| 15:42.30 | Hmmhesays | j/k //not really ///but kinda |
| 15:42.37 | jake1932 | what's the purpose of 2? |
| 15:43.06 | Hmmhesays | it gets longer hold on |
| 15:43.39 | Hmmhesays | http://pastebin.ca/52584 |
| 15:44.11 | Hmmhesays | the call comes in _X. catches it, sends it to s,1 |
| 15:44.42 | Hmmhesays | nevermind I figured it out |
| 15:45.53 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 15:46.11 | *** join/#asterisk Tagor (n=Tagor@s55928c6d.adsl.wanadoo.nl) |
| 15:46.14 | Tagor | Hi |
| 15:46.31 | Tagor | I'm using asterisk (with an external SIP provider) and grandstream phones |
| 15:46.53 | Tagor | Now I would like to make a script that first calls the grandstream and then onces it's picked up, it should call an external number |
| 15:47.04 | Tagor | Is there an easy way to do that? |
| 15:47.36 | wasim | call files baby |
| 15:47.59 | jake1932 | exten => s,n,Goto()? |
| 15:50.01 | Tagor | Thanks wasim |
| 15:50.27 | *** join/#asterisk shaynes (n=shayne@c-67-161-190-26.hsd1.ca.comcast.net) |
| 15:51.10 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 15:51.27 | shaynes | Can anyone help? I am setting up an IVR using Asterisk@Home (latest-ver) and I am having trouble using a WAV as the announcement. Are there any special requirements of the WAV file? |
| 15:51.42 | wasim | shaynes: mono, 8khz, 8 bit |
| 15:51.56 | CunningPike | shaynes: What he said |
| 15:52.05 | CunningPike | Just beat me to it lol |
| 15:52.48 | shaynes | wasim: no way. |
| 15:52.50 | CunningPike | Wow |
| 15:52.50 | [TK]D-Fender | shaynes : Please read the channel topic. |
| 15:53.22 | shaynes | [TK]D-Fender: you're too kind. |
| 15:53.32 | [TK]D-Fender | shaynes : Not possible :) |
| 15:53.38 | CunningPike | Is that channel new? |
| 15:53.46 | CunningPike | I hadn't noticed it before |
| 15:53.47 | shaynes | [TK]D-Fender: figured. ;) |
| 15:54.05 | Katty | [TK]D-Fender: we should talk :P |
| 15:54.18 | Katty | [TK]D-Fender: this isn't asterisk related. |
| 15:54.33 | [TK]D-Fender | shaynes : Sorry if I sounbd abrupt... you did come in kinda cold without so mcuh as a "hello all", and flew right into the "grey" topics... |
| 15:54.45 | [TK]D-Fender | Katty : :O. Always open. |
| 15:54.46 | Hmmhesays | wow this version of freepbx is retarded |
| 15:54.56 | [TK]D-Fender | Katty : and "mew", and pm if you like |
| 15:54.56 | shaynes | [TK]D-Fender: "grey?" -- open source, open mic? |
| 15:54.59 | Katty | [TK]D-Fender: yeah well it'll take a bit. |
| 15:55.05 | [TK]D-Fender | Katty : Sure |
| 15:55.07 | Katty | k |
| 15:55.10 | Katty | also! mew |
| 15:55.25 | shaynes | Hello All! |
| 15:55.45 | [TK]D-Fender | shaynes : Grey in the sens that all things AMP related tend to lead towards being shuffled off. |
| 15:56.17 | blitzrage | has anyone else been experiencing VM in 1.2.7 deleting the audio files, but leaving the .txt file behind? |
| 15:56.30 | pythos | Hmm, anyuone help me with auto-ignore problem in bitchx? |
| 15:57.17 | *** part/#asterisk shaynes (n=shayne@c-67-161-190-26.hsd1.ca.comcast.net) |
| 16:01.23 | sevard | I'm having a little issue with DTMF on my IVR. It seems I have to dial extensions at the prompt very slowly |
| 16:01.32 | *** join/#asterisk miguel3239 (n=miguel32@ns1.nashuacs.com) |
| 16:01.34 | CunningPike | blitzrage: Very occasionally, and I can't remember what caused it |
| 16:01.53 | CunningPike | It hasn't happened for ages |
| 16:02.00 | sevard | I have |
| 16:02.02 | sevard | exten = s,4,Set(TIMEOUT(digit)=3) |
| 16:02.03 | sevard | exten = s,5,Set(TIMEOUT(response)=15) |
| 16:02.08 | sevard | Is that not suggested? |
| 16:02.09 | blitzrage | CunningPike: yah, we're talking about it in dev -- seems to be in 1.2.7 -- is that what you're running? |
| 16:02.26 | CunningPike | We're running 1.2.1 in production right now |
| 16:02.35 | CunningPike | Test is 1.2.7.1 |
| 16:03.14 | CunningPike | This would have been quite a while back for us - maybe pre-1.2 - and I think it was something we were doing rather than something in * |
| 16:03.33 | CunningPike | We have our vm files on a mounted share |
| 16:04.00 | CunningPike | sevard: We use timeouts of 2 |
| 16:04.10 | CunningPike | Otherwise you have to wait or press # |
| 16:04.24 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 16:04.55 | sevard | CunningPike: digit and response? |
| 16:05.01 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 16:05.10 | CunningPike | sevard: Yes |
| 16:05.20 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 16:05.56 | CunningPike | 15 is way long |
| 16:06.15 | CunningPike | Morning wunderkin |
| 16:06.50 | sevard | CunningPike: I get the issue, even with the 2 time out when I enter in extension 2003 I get invalid extension 203, or invalid extension 00, or invalid extension 003, etc. |
| 16:07.01 | pythos | k |
| 16:07.05 | CunningPike | Ah - that's a DTMF problem then |
| 16:07.15 | *** part/#asterisk mosty (i=mostynm@adsl-137-244.swiftdsl.com.au) |
| 16:07.18 | sevard | CunningPike: right :) |
| 16:07.21 | CunningPike | What type of DTMF signaling are you using |
| 16:07.29 | sevard | CunningPike: I am unsure how to find out |
| 16:07.33 | Hmmhesays | this freepbx seems to be using a query string that I cannot find in any of the source files |
| 16:07.41 | wunderkin | hey |
| 16:07.42 | CunningPike | What phones are you using |
| 16:07.52 | sevard | CunningPike: I have some calls originating from ATAs, from some SIP phones, some cell phones and some land line phones from the pstn |
| 16:08.15 | CunningPike | sevard: And they're all exhibiting problems? |
| 16:08.26 | sevard | CunningPike: mostly the ones from the cellphones |
| 16:09.13 | CunningPike | sevard: We've noticed that a couple of times as well - not very often though.... |
| 16:09.18 | *** join/#asterisk CoffeeIV_ (n=CoffeeIV@64.149.168.97) |
| 16:09.33 | CunningPike | sevard: Let me check something |
| 16:09.40 | sevard | CunningPike: it happens very often with the cellphone, when a cell dials in 90% of the time they can't ring the extension because of this problem |
| 16:10.06 | CoffeeIV_ | I have an asterisk built from source from CVS a while ago,and I am trying to figure out exactly which verison it is -- include/asterisk/version.h doesn't seem to have anything useful in it, how can I tell ? |
| 16:10.44 | sevard | asterisk -V |
| 16:10.45 | CunningPike | sevard: In your sip.conf, what codecs are enabled? |
| 16:10.56 | sevard | CunningPike: only ulaw |
| 16:11.06 | CunningPike | Ah - try allowing gsm as well |
| 16:11.09 | sevard | CoffeeIV_: asterisk -v |
| 16:11.12 | CoffeeIV_ | asterisk -V gives "Asterisk CVS-HEAD" which isn't reall helpful |
| 16:11.13 | sevard | CoffeeIV_: asterisk -V |
| 16:11.14 | *** join/#asterisk lzhang (n=lewiszha@67.95.13.46) |
| 16:11.19 | sevard | CunningPike: why gsm? |
| 16:11.31 | lzhang | is sip.conf the only place where I specify codecs? |
| 16:12.07 | CunningPike | sevard: Cell phone calls are gsm? Not sure of the exact reasoning, but it works for us :D |
| 16:12.21 | dlynes | CunningPike: cell phone calls are gsm v4 though |
| 16:12.27 | dlynes | CunningPike: asterisk i think uses gsm v2 |
| 16:12.47 | CunningPike | dlynes: Would that cause transcoding though? |
| 16:12.55 | dlynes | no idea |
| 16:13.05 | dlynes | I just know asterisk doesn't use gsm v4 because it's patented |
| 16:13.29 | CunningPike | OK |
| 16:13.37 | sevard | CunningPike: i'm not sure if that's helping ;/ |
| 16:13.40 | dlynes | The compression on gsm v4 is significantly greater, while still allowing better quality |
| 16:14.15 | sevard | CoffeeIV_: I'm not sure, try xdd asterisk | less and see if you can find anything |
| 16:14.18 | sevard | CoffeeIV_: or strings |
| 16:14.22 | sevard | erm, xxd |
| 16:14.25 | dlynes | CoffeeIV_: how about show version in the CLI? |
| 16:14.27 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 16:14.55 | dlynes | Just realized |
| 16:14.57 | CoffeeIV_ | show version at the *CLI> gives "CVS-HEAD" again |
| 16:15.08 | sevard | CoffeeIV_: strings asterisk | less, scroll down, see if you get what i get |
| 16:15.09 | dlynes | If it's CVS HEAD, he must be running 1.0.9.2 or earlier |
| 16:15.17 | CunningPike | sevard: Are these all on a PRI? |
| 16:15.20 | sevard | CoffeeIV_: or just download a new version |
| 16:15.27 | sevard | CunningPike: is what on a PRI? |
| 16:15.30 | wunderkin | yeah, does it really matter? it is going to be old anyway |
| 16:15.57 | CunningPike | sevard: Your troublesome DTMF calls - I'm wondering is the gain screwing you up |
| 16:16.36 | sevard | CunningPike: It's a bit more complicated than that :) I have sipura ATAs coming into to a TDM400P |
| 16:16.39 | dlynes | sevard: is your dtmf problems occurring with at least one leg on pstn, pri, or t1? |
| 16:16.51 | dlynes | sevard: yeah, i guess it is :) |
| 16:17.11 | sevard | dlynes: I'm not sure what you're asking or how to give you those answers. school me :) |
| 16:17.14 | dlynes | sevard: try cunningpike's suggestion then...it's probably your gain control |
| 16:17.19 | CoffeeIV_ | strings can't give me anything that isn't in the #defines in the code, which seems to "CVS-HEAD" . . .. I found a reference to version 1.2 in UPGRADE.txt, I guess that is best I can do |
| 16:17.21 | sevard | Where is my gain? |
| 16:17.29 | dlynes | sevard: zapata.conf |
| 16:17.31 | CunningPike | In zapata.conf |
| 16:17.37 | dlynes | sevard: that's for your tdm400p |
| 16:17.46 | sevard | rxgain=12.0 |
| 16:17.46 | sevard | txgain=3.0 |
| 16:17.46 | CunningPike | Damn I gotta drink more coffee |
| 16:17.47 | dlynes | sevard: your sipura also has a gain control, I think |
| 16:17.52 | brodiem | what is the recommended codec for SIP over broadband? |
| 16:18.03 | dlynes | brodiem: g729 |
| 16:18.13 | sevard | it matters how much broadband you have |
| 16:18.15 | CunningPike | sevard: Also try dtmfmode=rfc2833 for your ATA entries in sip.conf |
| 16:18.22 | wunderkin | CoffeeIV_: what is your reason for finding out? |
| 16:18.25 | brodiem | sevard well just as a generalization |
| 16:18.32 | CunningPike | And make sure the ATA is set appropriately also |
| 16:18.51 | dlynes | sevard: oh yeah...one other thing, too...if you're using sipuras, make sure you disable auto dtmf detection on them |
| 16:18.55 | brodiem | dlynes, is that the codec thatrequires a paid license? |
| 16:19.00 | CunningPike | dlynes: Good point |
| 16:19.02 | sevard | Will do. |
| 16:19.23 | dlynes | sevard: they have a major problem with correctly detecting dtmf...you often get dtmf codes in your conversation cause it thinks it hears dtmf |
| 16:19.31 | sevard | hahaha |
| 16:19.37 | viperdudeuk | how do i stop outgoing calling from voicemail? |
| 16:19.40 | dlynes | sevard: and even the newest firmware doesn't fix it |
| 16:19.49 | sevard | the sipuras that come into the tdm400p aren't defined in sip.conf |
| 16:19.56 | dlynes | sevard: it occurs on both sipura 2000's and sipura 3000's |
| 16:20.05 | CunningPike | sevard: Right - sorry |
| 16:20.06 | sevard | Yeah, I have 2002s |
| 16:20.07 | dlynes | sevard: not to mention sipura 2002's and pap-2's |
| 16:20.24 | Hmmhesays | wow that version of freepbx had gone retarded |
| 16:20.50 | Hmmhesays | missing a freaking column in the database |
| 16:20.52 | dlynes | sevard: 2002 and pap-2's are just 2000's with a different cover and newer default firmware |
| 16:20.54 | Hmmhesays | i had to manuall add it |
| 16:21.16 | sevard | Hmmhesays: why do you use that crap |
| 16:22.06 | dlynes | brodiem: yes...if you want to avoid codec licensing, try ilbc, but it requires more cpu power, albeit, it consumes less bandwidth than g729 |
| 16:22.23 | CoffeeIV_ | wunderkin: my reason for finding out is that someone asked me . . . they seemed statisfied when I told them "probably 1.2", and I'll be installing a newer version in a few weeks anyway |
| 16:22.37 | brodiem | dlynes, thanks |
| 16:22.46 | *** join/#asterisk Tier_1 (n=Tier@c-24-9-75-234.hsd1.co.comcast.net) |
| 16:23.06 | CoffeeIV_ | but some process should put a version number, or at least a date, in version.h when you check into CVS, or something |
| 16:23.59 | CunningPike | brodiem: You could _try_ ulaw - I'm having success with ulaw on IAX - ymmv |
| 16:24.15 | *** join/#asterisk SparFux (n=player@e182016205.adsl.alicedsl.de) |
| 16:24.22 | sevard | does anyone know off hand how to tell ZAP trunk to give you a dialtone on one line? |
| 16:24.30 | Hmmhesays | nothing wrong with freepbx |
| 16:24.39 | Hmmhesays | well.. not nothing, but it is useful |
| 16:25.14 | sevard | like exten = 400,1,(ZAP/1) |
| 16:27.21 | brodiem | CunningPike, yeah it just seems ulaw has some shakiness sometimes from periodic laginess |
| 16:27.42 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 16:27.49 | sevard | exten = 400,1,DIAL(ZAP/1) |
| 16:27.51 | brodiem | CunningPike where dropped frames cause a bit of broken speech |
| 16:27.51 | CunningPike | Yes - it does sometimes, but I find it quite useable |
| 16:27.53 | sevard | rock |
| 16:28.30 | CunningPike | brodiem: It depends on your connection - no QoS here :D |
| 16:28.48 | brodiem | CunningPike, I just figured it would be better off to use something that has a bit more overhead but shorter intervals and packet sizes to make up for it |
| 16:29.46 | CunningPike | What UA are you using? |
| 16:30.28 | brodiem | CunningPike I find ulaw quite usable too just picky :) |
| 16:30.36 | *** join/#asterisk copland (n=stonecol@209.216.65.10) |
| 16:30.38 | CunningPike | :) |
| 16:30.39 | brodiem | just annoys me when I hear any little breakups |
| 16:30.51 | *** join/#asterisk PakiPenguin (i=uppal@linuxpakistan/admin/pakipenguin) |
| 16:30.51 | copland | hello is there a room for asterisk at home on freenode |
| 16:30.59 | CunningPike | Ya - I can live with it. |
| 16:31.03 | brodiem | copland, look at topic |
| 16:31.24 | CunningPike | brodiem: I work from home one day a week and use an IAXy with ulaw |
| 16:31.30 | CunningPike | It works |
| 16:31.31 | copland | sorry irrsi burried the topic I see it now |
| 16:31.53 | CunningPike | IAXy doesn't do gsm, and I've only enabled ulaw and gsm on our sever |
| 16:32.36 | mut | ther any way to bridge a call on my pri and free both channels it's using? |
| 16:32.42 | mut | say someone calls in on 1234 |
| 16:32.45 | mut | i call out on 4321 |
| 16:32.51 | mut | can i bridge that and drop those channels? |
| 16:33.05 | mut | or will * need to stay in between |
| 16:33.41 | CunningPike | mut: Can't see how - your PRI channels are your connection to the POTS - unless I misunderstand the question |
| 16:33.42 | jsharp | There's some basic support for that in libpri. Dunno how well its tested, though. |
| 16:34.14 | Hmmhesays | carlos mencia is hilarious |
| 16:34.20 | SparFux | How can I run a script or something whenever a special user logs in to asterisk via sip? |
| 16:34.48 | copland | has anyone managed to get stanaphone service to work with asterisk? |
| 16:35.17 | wunderkin | mut, 2 b channel transfer? |
| 16:35.52 | Hmmhesays | doesn't look very complicated copland |
| 16:36.10 | brodiem | SparFux I would think you'd need something listening for manager events to trigger a script, otherwise a system call when a dial plan gets used for that SIP ext |
| 16:36.47 | SparFux | brodiem: So far no hook available for that, I suppose? :-( |
| 16:41.22 | lunk | is there a return type command that will send you back to the calling context? |
| 16:41.42 | ManxPower | Yay! I'm getting 3,000 ft of water pipe for $250! |
| 16:41.46 | lunk | a goto from ctxt1 sends you to ctxt2 where you do something and want to return |
| 16:41.51 | trelane_ | ManxPower, why? |
| 16:42.00 | mut | wunderkin: yae |
| 16:42.05 | ManxPower | trelane, conduit to bury cable |
| 16:42.18 | trelane_ | ManxPower, a valid use of your time, I approve, carry on. |
| 16:42.19 | websae | CVC? |
| 16:42.28 | websae | 1/8inch? |
| 16:42.56 | websae | whoops...mean 1'' 1/4 |
| 16:43.27 | mut | 1/8... YEA! |
| 16:43.51 | websae | haha...find like maybe a twisted pair in that |
| 16:43.52 | websae | haha |
| 16:43.56 | mut | few strands of fiber |
| 16:44.00 | mut | plenty enuf |
| 16:44.10 | websae | haha |
| 16:45.49 | *** join/#asterisk key2 (n=key2@gob75-2-81-56-64-17.fbx.proxad.net) |
| 16:46.16 | techman97_andy | hey all - I'm trying to put a "ring" tone across incoming lines - I'm doing "exten => (my phone #),1,PlayTones(ring)", the CLI shows the command executing, but I do not hear the ring sound in the phone - what am I doing wrong? |
| 16:47.05 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 16:47.27 | viperdudeuk | techman97_andy: you need to answer the channel first |
| 16:47.33 | techman97_andy | lemme try that quick |
| 16:47.50 | pythos | anyone care to comment on using Debian for platform to run asterisk? I am trying to decide what release to use <stable, testing, unstable> ?? |
| 16:48.19 | viperdudeuk | i use debian stable without a problem |
| 16:48.35 | pythos | and version 1.0.7 then? |
| 16:48.45 | pythos | <asterisk> |
| 16:48.47 | techman97_andy | viperdudeuk: I did the answer command, but I still do not hear my ringtones...=( |
| 16:49.24 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 16:50.34 | viperdudeuk | so you have answer as priority 1 and playtones as priority 2? |
| 16:50.52 | techman97_andy | answer = 1, wait = 2 (just to test), playtones(ring) = 3 |
| 16:51.02 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 16:51.13 | sevard | What sort of PRI signaling does * do? Ground/wink/mediate |
| 16:51.14 | viperdudeuk | anything after 3? |
| 16:51.19 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
| 16:51.30 | sevard | or is that card dependant |
| 16:51.33 | techman97_andy | yes, I have a background menu sound play |
| 16:51.35 | techman97_andy | that works - |
| 16:51.42 | techman97_andy | press 1 for...2 for... |
| 16:51.44 | bkw__ | sevard, um |
| 16:51.46 | viperdudeuk | pythos: I use 1.2.7.1 but all versions I have tried are ok |
| 16:51.49 | bkw__ | sevard, those are not PRI |
| 16:51.58 | sevard | bkw__: what are they |
| 16:51.58 | techman97_andy | the call gets picked up...the commands fly by in the CLI, and then my menu plays |
| 16:52.12 | bkw__ | sevard, CT1 signalling, Inband |
| 16:52.24 | viperdudeuk | so the playtones dont fire because it immediate goes to the background command |
| 16:52.43 | techman97_andy | no, priority 4 is the background command |
| 16:53.06 | pythos | viperdudeuk: 1.2.7.1 is the latest, I think. Not sure of which Debian release it might be in yet, or if that release would be more/less desireable than stable/with 1.0.7 |
| 16:53.28 | viperdudeuk | yes i know but the playtones will only work until another command plays audio, the background plays straightway... try putting a wait after the playtones |
| 16:53.31 | sevard | bkw__: I'm looking at getting a PRI and that's what my telco asked me what my * box does |
| 16:53.32 | techman97_andy | viper: I can put wait commands between the playtones, but still nothing plays. |
| 16:53.48 | bkw__ | sevard, no what switch types do they support first |
| 16:53.49 | bkw__ | then you can pick one |
| 16:53.51 | viperdudeuk | do you have a ring tone defined? |
| 16:53.55 | bkw__ | chances are asterisk does it |
| 16:54.08 | viperdudeuk | pythos: i always compile from source |
| 16:54.10 | bkw__ | technically its not asterisk its libpri that does it but we aren't splitting hairs here |
| 16:54.21 | pythos | what is the GNU GUI for asterixk? |
| 16:54.28 | *** join/#asterisk tekati (n=captain@cpe-66-75-215-63.bak.res.rr.com) |
| 16:54.28 | techman97_andy | how do you define a ring tone? I thought was just a sound file * played? |
| 16:54.49 | viperdudeuk | indications.conf |
| 16:54.51 | *** join/#asterisk MooingLemur (n=troy@shells200.pinchaser.com) |
| 16:55.08 | viperdudeuk | there is no gui for * |
| 16:57.17 | Damin | Is there an agenda for tommorow's con call yet? |
| 16:57.32 | lzhang | guys, any ideas about what this debug message means: channel.c: Didn't get a frame from channel: SIP/8113-9e81 |
| 16:57.35 | pythos | huh, thought I had read of a bunch... |
| 16:57.42 | lzhang | I'm getting one way audio problems and I think it may be related |
| 16:57.53 | viperdudeuk | thre are some web interfaces |
| 16:58.01 | pythos | oh, ok |
| 16:58.41 | *** join/#asterisk ramo (n=ramo@59.92.137.242) |
| 16:59.02 | viperdudeuk | lzhang: firewall / NAT issue? one way audio is usually one of those |
| 16:59.04 | pythos | viperdudeuk: thus requiring something like apache? Does it call for https: ? |
| 16:59.41 | viperdudeuk | pythos: not if you dont require it.... Asterisk @ Home is a full asterisk based distro with a gui |
| 17:00.56 | pythos | viperdudeuk: oh, so, getting debian all set is not required I guess... Wonder why I didn't go that route in the first place, now that I have debian/sarge all configured for my tdm400p |
| 17:01.42 | viperdudeuk | pythos: lol ok, not sure if A@H handles the tdm400p, you would need to check first |
| 17:03.17 | pythos | viperdudeuk: the package said digium tdm22b, but it is recognized by my kernel dmesg as tdm400p 2fxs/2fxo |
| 17:03.29 | *** join/#asterisk MstlyHrmls (n=mh@melbourne.mostly-harmless.ca) |
| 17:03.33 | [TK]D-Fender | pythos : thats exactly right. |
| 17:04.46 | CunningPike | sevard: Your telco may be talking about ESF/b8zs signalling |
| 17:04.54 | CunningPike | That's the norm in NA |
| 17:05.08 | sevard | CunningPike: actually EFF/B8ZF |
| 17:05.25 | techman97_andy | viperdudeuk: ok - I'm back. If I dial in on my ZAP line, I get the ring tone. Calls that come across my SIP peer don't. |
| 17:05.35 | sevard | oh, she said EFF, made her repeat, echo foxtrot foxtrot, it's ESF? |
| 17:05.47 | justinu|laptop | ESF is correct |
| 17:05.47 | CunningPike | sevard: OK - ours is what I said :) |
| 17:06.09 | viperdudeuk | techman97_andy: the SIP calls hit the same exten? |
| 17:06.41 | techman97_andy | correct - the only difference between the two is that the SIP calls come in a seperate context, but I execute a GoTo(s,1) in that context |
| 17:06.50 | CunningPike | sevard: No-one ever went broke underestimating the intelligence of telco customer service agents |
| 17:06.56 | rajiv|work | on my PRI when i call out to my cell phone i see my calling #, but not when i call an analog line that has working caller id. anyone know why? |
| 17:07.09 | sevard | CunningPike: haha |
| 17:07.12 | CunningPike | :D |
| 17:07.18 | lzhang | viperdudeuk: it's not firewall/NAT, no NAT and I turned off the firewall temporarily |
| 17:07.47 | viperdudeuk | you are specifying the context in the goto? |
| 17:07.47 | lzhang | I think it may be a g729 issue, how do I turn this codec off? just commenting out the allow line in sip.conf is not working |
| 17:07.55 | techman97_andy | yes I am - sorry 'bout that. |
| 17:08.17 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 17:08.23 | viperdudeuk | lzhang: disallow = all allow=ulaw or what ever codec you want |
| 17:08.33 | CunningPike | rajiv|work: We are continually amazed by the inconsistency of CID presentation - inbound as well as outbound. Our telco hasn't been able to explain it either |
| 17:08.37 | viperdudeuk | do the sip channels hear the background after? |
| 17:09.00 | techman97_andy | viper: SIP channels hear the background, yes. |
| 17:09.06 | lzhang | viperdudeuk: yeah that's what I tried but for some reason I'm still getting no compatible codecs with Xlite |
| 17:09.22 | brodiem | any recommendations on an origination provider (sip and/or iax) with good toll free inbound rates? |
| 17:09.25 | viperdudeuk | i use ulaw ok with x-lute |
| 17:09.31 | viperdudeuk | i use ulaw ok with x-lite even |
| 17:09.31 | techman97_andy | viper: the only thing that is different to the end user is that SIP calls cannot hear the ring tone. |
| 17:09.32 | *** join/#asterisk trbldwine (i=trbldwin@vpn163245.vpn.northwestern.edu) |
| 17:09.43 | viperdudeuk | techman97_andy: weird |
| 17:09.45 | brif8 | how can I disconnect from the console a call on SCCP which is hung |
| 17:09.49 | techman97_andy | tell me about it! =) |
| 17:10.15 | viperdudeuk | brif8: soft hangup SCCP/callid |
| 17:10.22 | brif8 | sccp show lines shows that the phone is connected yet the phone is off |
| 17:10.26 | poisoner | hmmm |
| 17:10.39 | poisoner | brif8 reload sccp |
| 17:11.33 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 17:11.52 | viperdudeuk | no reload on chan_sccp have to unload then load again which cuts off all sccp channels |
| 17:12.23 | *** join/#asterisk MacDome (n=eseidel@A17-255-100-181.apple.com) |
| 17:12.38 | ManxPower | perhaps you are using bindaddr? |
| 17:13.07 | brif8 | viperdudeuk: I get SCCP/100-0000003 is not a known channel |
| 17:13.13 | SpaceBass | I have a cisco 7940 that is behaving strangely |
| 17:13.26 | brif8 | poisoner: sccp reload is not yet implemented |
| 17:13.35 | brodiem | VoicePulse no I'm in the US, but a lot of calls come from canada |
| 17:13.51 | viperdudeuk | brif8: looks like chan_sccp is crashed, need to unload chan_sccp followed by load chan_sccp |
| 17:14.03 | SpaceBass | I'm using a modified POE cable from the wiki and it powers on fine, but says "ethernet disconnected" ... i get the same result if i use a standard able and a cisco power cube...however I have ONE port on an older switch on which it does work |
| 17:14.11 | SpaceBass | and I cannot for the life of me figure out what is different about that port |
| 17:14.16 | viperdudeuk | chan_sccp crashes a lot which is why i ditched it |
| 17:14.19 | brif8 | unload sccp |
| 17:14.19 | brif8 | Unable to unload resource sccp |
| 17:14.23 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 17:14.30 | *** join/#asterisk trbldwine (n=trbldwin@vpn163245.vpn.northwestern.edu) |
| 17:14.30 | viperdudeuk | brif8: ouch |
| 17:15.23 | harryvv | I have mailed the digium news groups for a question about the messages button on the ip500 and got no responce. Anyone here own a ip500 and have been able to configure the xml part of it to make this feature work? |
| 17:15.32 | harryvv | Same with the conferance button. |
| 17:15.57 | brif8 | viperdudeuk: do I have to reload * |
| 17:16.21 | ManxPower | harryvv, you read the admin buide? |
| 17:16.24 | CunningPike | harryvv: Polycom IP500? |
| 17:16.34 | SpaceBass | harryvv, what does the XML part do? |
| 17:16.35 | ManxPower | buide == guide |
| 17:16.45 | *** join/#asterisk TripleFFFFFFFFFF (n=TripleFF@147-102.mc.cite.net) |
| 17:16.45 | jaiger | harryvv, I have one |
| 17:16.50 | ManxPower | SpaceBass, Polycoms use XML config files |
| 17:16.55 | TripleFFFFFFFFFF | can one help me with realtime ? |
| 17:17.04 | TripleFFFFFFFFFF | seems it doesnt read it or something |
| 17:17.09 | TripleFFFFFFFFFF | i get 404 from the debug |
| 17:17.18 | SpaceBass | ah |
| 17:17.20 | *** join/#asterisk Ixthod (n=Ixthod@intellop.static.iaxs.net) |
| 17:17.34 | harryvv | jaiger cool. did you get the conferance a messages button to work? |
| 17:17.35 | *** join/#asterisk wulfstan1 (i=nclarey@cpc3-cmbg1-0-0-cust887.cmbg.cable.ntl.com) |
| 17:17.54 | [TK]D-Fender | harryvv : Read the admin guide... they spell out how to do it pretty clearly. |
| 17:17.58 | jaiger | conference works out of the box |
| 17:18.00 | viperdudeuk | brif8: you can try that or you might have to kill asterisk and restart |
| 17:18.02 | jaiger | messages... |
| 17:18.10 | [TK]D-Fender | Yup.... |
| 17:18.17 | brif8 | viperdudeuk: okay |
| 17:18.19 | harryvv | I press conferance button and no reply |
| 17:18.23 | harryvv | no responce that is |
| 17:18.30 | ManxPower | BTW, I finally got pricing info from my reseller for polycoms. /msg me if you were looking for that info |
| 17:19.01 | jaiger | harryvv, msg.mwi.1.callBack="*86" msg.mwi.1.subscribe="1" msg.mwi.1.callBackMode="contact" |
| 17:19.01 | *** join/#asterisk MacDome (n=eseidel@A17-255-100-181.apple.com) |
| 17:19.10 | ghost99 | deos the centos ISO have an FTP server ? |
| 17:19.14 | SpaceBass | Voipsupply.com still hasn't shipped my wip330 and they had the nerve to call me about volume pricing |
| 17:19.20 | [TK]D-Fender | harryvv : to conference > be on a call > press "conference" > call the 2nd party. > they answer >press conference again > Done |
| 17:19.24 | jaiger | harryvv, where callBack is the extension to call to check voicemail |
| 17:19.32 | harryvv | I see |
| 17:19.40 | wulfstan1 | My TDM400P won't detect hangups on my NTL line in the UK. Has anyone had any success getting this to work? |
| 17:19.50 | SpaceBass | ghost99, not sure its enabled by default, but it has ssh so you can use sftp and scp |
| 17:19.51 | justinu|laptop | ntl? |
| 17:19.52 | jaiger | harryvv, I also set bypassInstantMessage="1" to skip a menu |
| 17:19.57 | [TK]D-Fender | harryvv : And the default callbackmode is "Registration", not "contact" |
| 17:19.58 | wulfstan1 | NTL is a provider in the UK |
| 17:20.02 | justinu|laptop | ah |
| 17:20.02 | [TK]D-Fender | yup |
| 17:20.04 | wulfstan1 | NTL and BT are the two major telcos |
| 17:20.11 | justinu|laptop | bt i know about |
| 17:20.15 | [TK]D-Fender | ghost99 :PM |
| 17:20.18 | wulfstan1 | BT apparently works fine |
| 17:20.22 | wulfstan1 | NTL does not |
| 17:20.23 | justinu|laptop | do you know how your telco signals disco? |
| 17:20.24 | CunningPike | Can someone run slabtop on their * box and see if they have really high numbers for size-32 |
| 17:20.30 | wulfstan1 | Nope |
| 17:20.36 | justinu|laptop | is it a drop in loop current, or possibly a polarity reversal? |
| 17:20.36 | wulfstan1 | How can I find out |
| 17:20.38 | viperdudeuk | wulfstan1: is it NTL ISDN? |
| 17:20.41 | wulfstan1 | Nope |
| 17:20.44 | wulfstan1 | It is NTL analog |
| 17:20.58 | justinu|laptop | if you could get in touch with one of the switch techs perhaps |
| 17:20.59 | viperdudeuk | ok ok |
| 17:21.05 | wasim | CunningPike: 140k |
| 17:21.14 | justinu|laptop | perhaps call the trouble reporting number, make up some problem to get a knowledgeable tech on the line |
| 17:21.17 | justinu|laptop | then ask |
| 17:21.24 | viperdudeuk | lol you have never tried NTL customer services |
| 17:21.26 | CunningPike | Thanks, wasim - how many objects |
| 17:21.29 | wulfstan1 | I could always just use a multimeter |
| 17:21.39 | fourcheez-away | don't both with ntl customer services |
| 17:21.40 | wulfstan1 | It's probably more likely to tell me what I need to know than NTL customer service |
| 17:21.43 | justinu|laptop | you might be able too, but it could happen fast enough you won't see it |
| 17:21.49 | wulfstan1 | Ahh |
| 17:21.50 | wulfstan1 | Right |
| 17:22.01 | fourcheez-away | you need an oscilloscope |
| 17:22.05 | justinu|laptop | or they may not be sending any signalling |
| 17:22.05 | wulfstan1 | Eeep |
| 17:22.05 | wasim | CunningPike: 113 |
| 17:22.05 | harryvv | Man, polycom came up with some kind of wireless communicator that links with skype. |
| 17:22.08 | viperdudeuk | wulfstan1: it will have more of a clue than CS |
| 17:22.17 | justinu|laptop | wulfstan1: call someone, then ask them to hangup... what happens? |
| 17:22.19 | harryvv | I wonder if asterisk can link with skype |
| 17:22.23 | justinu|laptop | (use a test phone) |
| 17:22.33 | wulfstan1 | Well, I'll try it with my mobile |
| 17:22.34 | CunningPike | wasim: So, I guess 2241365 is a little unusual, then |
| 17:22.34 | wulfstan1 | Two secs |
| 17:22.36 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 17:22.37 | CunningPike | :D |
| 17:22.40 | generalhan | Whats up all !! |
| 17:22.41 | viperdudeuk | harryvv: i think there are some SIP gateways to skype around |
| 17:22.41 | wasim | CunningPike: indeed |
| 17:22.43 | sevard | O cam |
| 17:22.46 | techman97_andy | hey all: new issue. I just installed a GrandStream GXP as a SIP client on my * server. With the xLite softphone, we can do multi-line conferencing just fine. If the Grandstream says, "no compatible codecs". The Grandstream can make and receive calls without any other issues...Any ideas? |
| 17:22.53 | sevard | I can't find this. What does ESF stand for? |
| 17:23.14 | fourcheez-away | eat some figs |
| 17:23.28 | sevard | Oh, that makes sense. Yes. My PRI wants figs! |
| 17:23.28 | CunningPike | techman97_andy: Using g729? |
| 17:23.34 | techman97_andy | g729 end to end |
| 17:23.47 | wasim | mmmh ... figs and mozarella cheese, baked in the oven |
| 17:24.03 | justinu|laptop | interesting combo |
| 17:24.18 | CunningPike | techman97_andy: I believe that * Meetme can't do g729 because * doesn't have the necessary licensing |
| 17:24.21 | ManxPower | sevard, eXTENDED sUPER fRAME |
| 17:24.21 | sevard | wasim: that sounds lime some nasty styff |
| 17:24.42 | generalhan | Does anyone know of a way to test a SIP channel to see if it is active BEFORE sending a call to it? |
| 17:25.06 | jsharp | * meetme can do g729 IF you have a G729 codec license. |
| 17:25.10 | [TK]D-Fender | generalhan : "show application chanisavail" |
| 17:25.12 | fourcheez-away | sevard: http://www.google.com/search?q=define%3A+esf&start=0&ie=utf-8&oe=utf-8&client=firefox&rls=org.mozilla:en-GB:unofficial |
| 17:25.14 | wasim | sevard: try a search on google, lots of places are now doing it |
| 17:25.20 | viperdudeuk | generalhan: a sip peer on the * box or in general? |
| 17:25.23 | wulfstan1 | Ok |
| 17:25.27 | generalhan | [TK]D-Fender: well ... i mean in extensions.conf |
| 17:25.27 | brif8 | has anyone else experienced SCCP hanging I've even tried sccp reset DEVICEID it also did not work |
| 17:25.33 | wulfstan1 | So I just tried calling my line from my mobile |
| 17:25.40 | [TK]D-Fender | generalhan : Thats the command |
| 17:25.43 | wulfstan1 | I get nothing for 5 seconds when I hang up |
| 17:25.43 | sevard | wikipedia failed me on the acroynm but wins all typed out |
| 17:25.44 | [TK]D-Fender | generalhan : Look it up |
| 17:25.49 | wulfstan1 | and then I get a continuous tone |
| 17:25.57 | viperdudeuk | brif8: we had chan_sccp crashing * |
| 17:26.00 | generalhan | Im still trying to find a way to turn off call waiting on these stupid aastra phones. and this is my next solution to try |
| 17:26.08 | justinu|laptop | wulfstan1: ok... you can probably program asterisk to listen for that tone, and detect a hangup |
| 17:26.12 | brif8 | viperdudeuk: what was your solution ? |
| 17:26.20 | justinu|laptop | wulfstan1: not the best way to do it, but it maybe the only choice you have |
| 17:26.24 | viperdudeuk | i switched from SCCP to SIP lol |
| 17:26.27 | wulfstan1 | Yeah, it's like 9 seconds |
| 17:26.31 | wulfstan1 | Before it starts the tone |
| 17:26.41 | wulfstan1 | Ok, so what do I need to do |
| 17:26.56 | justinu|laptop | there's something called "callprogress=yes" that might do it |
| 17:27.15 | wulfstan1 | I tried that and it didn't seem to do anything |
| 17:27.32 | wulfstan1 | Although to be fair it may have just been waiting for the 9 seconds to timeout before it hung up |
| 17:27.33 | justinu|laptop | maybe your locale needs to be modified |
| 17:27.40 | wulfstan1 | I have set it to UK |
| 17:27.41 | justinu|laptop | asterisk needs to know what tone to listen for |
| 17:27.51 | justinu|laptop | maybe that tone isn't defined in UK |
| 17:27.57 | wulfstan1 | Entirely possible |
| 17:28.00 | justinu|laptop | i'm not an expert on this type of disconnect supervision |
| 17:28.14 | brif8 | viperdudeuk: but the Cisco 7920 is not SIP capable yet or is it ? |
| 17:28.17 | wulfstan1 | So let me get this straight |
| 17:28.22 | justinu|laptop | i usually use ISDN, CAS trunks, SS7 or SIP |
| 17:28.36 | *** part/#asterisk TripleFFFFFFFFFF (n=TripleFF@147-102.mc.cite.net) |
| 17:28.37 | rajiv|work | CunningPike: i wonder if my cell carrier is using ANI and not CID |
| 17:28.37 | justinu|laptop | disconnect and answer supervision sucks on analog lines |
| 17:28.39 | wulfstan1 | Usually a terminal can figure out that the line has hung up because of some voltage condition |
| 17:28.46 | viperdudeuk | brif8: not sure i use 7940's and 7960's which have SIP firmware |
| 17:28.53 | justinu|laptop | here in the US, we use "kewlstart" |
| 17:29.01 | justinu|laptop | which is a 500ms drop in loop current |
| 17:29.02 | wulfstan1 | But you can fall back on tone detection |
| 17:29.19 | CunningPike | rajiv|work: It's possible..... who is your carrier? |
| 17:29.20 | wulfstan1 | Right |
| 17:29.40 | viperdudeuk | BT signalling for analog is signalling=fxs_ks |
| 17:29.46 | justinu|laptop | i remember reading that some UK providers use polarity reversal |
| 17:29.48 | rajiv|work | CunningPike: conversent on the PRI, tmobile for cell, RCN for analog phone (all in boston, ma, usa) |
| 17:29.50 | wulfstan1 | Yup, I realise that works for BT |
| 17:30.04 | wulfstan1 | I was going to experiment with that on a friend's line tonight |
| 17:30.16 | wulfstan1 | So if it's polarity reversal, how would I enable that |
| 17:30.53 | CunningPike | rajiv|work: Is your CID consistent on incoming calls? We were told by our telcos that CID handling from one network to another is inconsistent, but I'm not really buying that |
| 17:30.55 | justinu|laptop | http://www.asteriskguru.com/tutorials/resolving_hangup_detection_problems_fxo_tdm_voicemail.html |
| 17:30.58 | justinu|laptop | check that link wulfstan |
| 17:31.23 | Ikarus | CunningPike: CID can be inconsistent, with the leading 0 being dropped by some, but not others here |
| 17:31.30 | rajiv|work | CunningPike: incoming to the PRI works fine. incoming to the analog i've never had problems until i started calling from my own pri |
| 17:32.07 | rajiv|work | should i be sending 10 or 11 digits for CID on the pri ? |
| 17:32.23 | ManxPower | rajiv|work, In the USA 10 digits. |
| 17:33.26 | ManxPower | The leading 1 is NOT part of the callerid or phone number, it's a toll access code. |
| 17:33.26 | CunningPike | Ikarus: We get inconsistencies around whether we get CID name at all or not - sometimes we get name, the majority of the time we don't |
| 17:33.26 | MooingLemur | it's a country code |
| 17:33.26 | DrDeke | ManxPower: Although that's the way I see it as well, many ITSPs do send the "1" as part of the caller ID number when the caller is from the US. It's very annoying. |
| 17:33.26 | ManxPower | CunningPike, put a wait(1) before anything else. |
| 17:33.27 | Ikarus | CunningPike: ah |
| 17:33.27 | CunningPike | We put a Wait(2) in - no joy |
| 17:33.32 | Ikarus | CunningPike: quite different for the issue here |
| 17:33.36 | ManxPower | DrDeke, caller from us and calling a non-usa number? |
| 17:33.39 | CunningPike | Ikarus: True :) |
| 17:33.44 | DrDeke | ManxPower: Caller from US calling a US number |
| 17:33.46 | rajiv|work | ManxPower: so my equip should send 10 digits... but what if i am calling non-usa? then dont they need to have 11 ? |
| 17:33.49 | ManxPower | dr0ck, ah. |
| 17:34.13 | DrDeke | ManxPower: yeah, it's a pain that they aren't all consistent. But I do think you're right; you really "should" send 10 digits |
| 17:34.14 | wulfstan1 | Right, thanks I will give it a try and see if I have any luck |
| 17:34.16 | ManxPower | rajiv|work, I think the carrier will add the country code of 1 at the beginning, not sure. |
| 17:34.17 | wulfstan1 | Will be back if I don't |
| 17:34.19 | generalhan | [TK]D-Fender: i dont know if that will help me accomplish my goal ... i have 20 phones in a call queue and i cant turn the callwaiting off in the config or on the phone itself (i called the manufacturer) so what i want to do is dial to a context from the queue that will check to see if the person is on a call and only ring to the 15 of the 20 that are not on the phone. |
| 17:34.20 | wulfstan1 | :-) |
| 17:34.20 | CunningPike | rajiv|work: Your CID is unlikely to survive the journey anyway |
| 17:34.39 | ManxPower | generalhan, what phone. |
| 17:34.40 | [TK]D-Fender | generalhan : Yes this command IS what you need. |
| 17:34.42 | MooingLemur | we get UK callerid from time to time at work |
| 17:34.43 | rajiv|work | CunningPike: really? inbound caller id from germany to my analog lines in boston works |
| 17:34.48 | generalhan | Aastra 9112i |
| 17:34.58 | brif8 | anyone working on getting a Cisco 7920 to be SIP capable, or what is the process ? |
| 17:34.59 | CunningPike | rajiv|work: Really - interesting |
| 17:35.00 | [TK]D-Fender | generalhan : And I have used it for EXACTLY that reason. |
| 17:35.23 | Ikarus | CunningPike: CID in the form of a name never survives internationally with current protocols |
| 17:35.27 | CunningPike | We usually get the number for our nearest access point - but I'm in Canada, so it may be different here |
| 17:35.31 | Ikarus | well, possible exception, USA and Canada |
| 17:35.43 | generalhan | [TK]D-Fender: in the configs it talks about trying one line and if its not avail going to the next line ... but i want it to test several of the lines, and then dial ALL that arent on a call |
| 17:35.45 | ManxPower | brif8, you don't "work on it" Cisco either has SIP firmware for the phone or they do not. |
| 17:35.51 | viperdudeuk | brif8: you need the SIP firmware from cisco |
| 17:35.51 | MooingLemur | the name seems to be a lookup by your local phone company |
| 17:36.21 | brif8 | any ideas if or when they may get SIP firmware for the 7920 ? |
| 17:36.37 | ManxPower | brif8, Cisco seldom announces such things. |
| 17:36.45 | brif8 | ok |
| 17:37.01 | ManxPower | Didn't you confirm that the phone supported SIP before you bought it? |
| 17:37.02 | Ikarus | rajiv|work, MooingLemur: number based Caller ID is highly reliably internationally in my experience |
| 17:37.18 | viperdudeuk | check the cisco site... you will probably need a cisco support contract to get the sip image |
| 17:37.21 | ManxPower | brif8, MANY Cisco phones never get SIP firmware |
| 17:37.31 | eric_hill | Cisco doesn't offer a SIP firmware for the 7920 at the present time. |
| 17:37.44 | Nugget | Cisco has difficulty spelling SIP. |
| 17:37.46 | eric_hill | SCCP only, and you have to have crypto enabled. |
| 17:38.05 | brif8 | ManxPower: my hope is that the 7920 isn't one |
| 17:38.07 | viperdudeuk | cisco want you to by ccm |
| 17:38.10 | viperdudeuk | buy |
| 17:38.31 | [TK]D-Fender | generalhan : You don't ring multiple phones on a queue pas normall, jsut an individual agent through "agentcallbacklogin" |
| 17:38.33 | brif8 | eric_hill: I realize presently it is only SCCP, I'm hopes is they change as they did with the 7960 |
| 17:38.58 | Qwell[] | 7960 has pretty much always had sip firmware |
| 17:39.00 | generalhan | [TK]D-Fender: i dont use agents at all ... |
| 17:39.02 | Qwell[] | 7970 however... |
| 17:39.26 | Qwell[] | but, what's wrong with sccp? |
| 17:39.41 | viperdudeuk | chan_sccp crashes a lo |
| 17:39.42 | viperdudeuk | lot |
| 17:39.49 | Qwell[] | yeah, but what |
| 17:39.54 | Qwell[] | 's wrong with sccp itself? |
| 17:40.07 | generalhan | [TK]D-Fender: i just have each SIP line that i want to dial to in the queues.conf file and they ring that way. but if i just put one single member in there i can have that one be a different context in extensions.conf that will test all the lines and them ring them all |
| 17:40.30 | viperdudeuk | nothing wrong with it just not stable on * |
| 17:40.31 | brif8 | Qwell[]: it would appear mine just crashed. I started a call and now have the phone off, yet sccp show lines still shows it connected. sccp reload doesn't work nor has sccp reset DEVICEID or sccp restart DEVICEID |
| 17:40.58 | Qwell[] | yes, chan_sccp is buggy...but that doesn't mean sccp is a bad protocol |
| 17:41.09 | generalhan | [TK]D-Fender: so ChanIsAvail will work if ${AVAILCHAN} can hold multiple lines |
| 17:41.11 | [TK]D-Fender | generalhan : Either way it will test a given line and you can chain them up any way you likel |
| 17:41.18 | viperdudeuk | thats why i said chan_sccp crashes not sccp crashes |
| 17:41.28 | brif8 | Qwell[]: I'm not calling it a good or bad protocol. just SIP seems alot more stable. |
| 17:42.18 | *** join/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu) |
| 17:42.43 | [TK]D-Fender | generalhan : Read the instructions CLOSER. You're not paying attention... |
| 17:43.01 | [TK]D-Fender | generalhan : There is an option to se if they are on ANY CALL AT ALL. |
| 17:43.26 | generalhan | [TK]D-Fender: lol ... ok ok im still looking into it |
| 17:44.31 | generalhan | [TK]D-Fender: i just get discouraged when i see things like this in the wiki "If you want to use it for limiting simultaneous calls to the peer, it will not work reliably for you. |
| 17:45.06 | [TK]D-Fender | <PROTECTED> |
| 17:45.06 | [TK]D-Fender | <PROTECTED> |
| 17:45.20 | [TK]D-Fender | generalhan : Like Nike says "Just Do It" |
| 17:45.36 | harryvv | btw, how do i get the ip500 to switch over to the caller waiting caller if I am already talking to one party. |
| 17:48.12 | harryvv | I have missed a few calls trying to press the reciver button and ended up disconecting the caller. |
| 17:48.55 | Hmmhesays | yay, the gf is bringing me to lunch |
| 17:49.10 | Qwell[] | Hmmhesays: would be better if she was bringing you two lunch |
| 17:49.27 | Hmmhesays | it'd be better if she was bringing me over to the warehouse |
| 17:49.41 | Hmmhesays | cause then I could get some and play guitar |
| 17:50.55 | *** join/#asterisk DeV-rAd (n=jesse@fl-69-69-130-197.sta.sprint-hsd.net) |
| 17:51.09 | Hmmhesays | and that my friends would be fantastic |
| 17:51.30 | DeV-rAd | can someone help, i need help setting up 2 fxo cards |
| 17:51.48 | Hmmhesays | $75/hour |
| 17:51.49 | [TK]D-Fender | harryvv : depends how you set up your line keys. if the cascade w/ 1 line/call then you jst need to hit the next line key, otherwise you use the cursor keys to pick the call on the keeys thats ringing |
| 17:52.16 | [TK]D-Fender | Hmmhesays : The one who got you into that mess? |
| 17:52.20 | harryvv | tk, this is a single line pstn phone system. |
| 17:52.24 | Hmmhesays | [TK]D-Fender she is long gone |
| 17:52.33 | Hmmhesays | i met a new one about a month ago |
| 17:52.53 | Hmmhesays | 6ft 135lbs |
| 17:52.57 | Hmmhesays | all kinds of yummy |
| 17:53.02 | sevard | Hmmhesays: always cat > /dev/null before cleaning your keyboard |
| 17:53.15 | Hmmhesays | hahahaa |
| 17:53.35 | *** join/#asterisk IceManRISK (n=kart@200.138.147.142) |
| 17:53.51 | *** part/#asterisk pythos (i=pythos@unaffiliated/pythos) |
| 17:53.52 | [TK]D-Fender | Hmmhesays : Always dodge left on ice-picks.... remember it.... |
| 17:53.52 | IceManRISK | hi |
| 17:54.01 | Hmmhesays | lol |
| 17:54.19 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
| 17:54.31 | brif8 | which a more stable version of chan_sccp berlios or sourceforge and why the two ? |
| 17:54.35 | sevard | [TK]D-Fender: always tip chinese hookers an extra dime if it was worth it |
| 17:55.20 | *** join/#asterisk bzbw (i=bwz@ip67-153-142-109.z142-153-67.customer.algx.net) |
| 17:56.25 | [TK]D-Fender | sevard : Not my lifestyle... |
| 17:56.36 | sevard | [TK]D-Fender: i've seen your clsoets |
| 17:56.38 | sevard | closets* |
| 17:57.01 | [TK]D-Fender | Yeah... I have the body of a Chippendale..... its buried in my backyard :D |
| 17:57.59 | *** join/#asterisk timscott (n=a@d198-53-19-216.abhsia.telus.net) |
| 17:58.23 | lunk | any ideas why an internal sip phone can respond to a message by pressing 1, but an externally called person can not? |
| 17:58.56 | jake1932 | how are you getting out? |
| 17:59.10 | *** join/#asterisk timscott (n=a@d198-53-19-216.abhsia.telus.net) |
| 17:59.20 | lunk | it's a broadvoice trunk |
| 17:59.27 | justinu|laptop | dtmf modes |
| 17:59.30 | jake1932 | check compatible dtmf modes |
| 17:59.46 | lunk | will do |
| 18:00.39 | sevard | Do Digium PRI cards do ESF, b8zf and NI2? |
| 18:00.43 | sevard | I would assume so. |
| 18:01.15 | *** join/#asterisk IOscanner (n=IOscanne@c-67-164-154-209.hsd1.tx.comcast.net) |
| 18:01.52 | Hmmhesays | sevard: you could do the unthinkable |
| 18:02.06 | sevard | Hmmhesays: superman powers? |
| 18:02.11 | Hmmhesays | look on the digium website |
| 18:02.14 | sevard | OMG |
| 18:02.18 | jake1932 | no! |
| 18:02.27 | sevard | the devil's handbook, lad. |
| 18:02.47 | justinu|laptop | sevard: yes, they do |
| 18:03.25 | sevard | justinu|laptop: just wading through all of this tdm crap and trying to soak it up, do you know if the TE205P does ground/wink/mediate |
| 18:03.37 | Hmmhesays | anyone ever read "saucer" by stephen coonts? |
| 18:04.56 | bzbw | strange, my * is sending DNS query to a none exist DNS name where part of it is the context name, and I did not define any domain for *, anyone has any clue? |
| 18:05.07 | *** join/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net) |
| 18:06.14 | justinu|laptop | sevard: afaik, the digium cards support all common T1 signalling methods, including FXS_GS, E&M wink/immediate |
| 18:06.26 | mog_work | true true |
| 18:06.31 | justinu|laptop | sevard: correction, there may be issues with wink start |
| 18:06.37 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-152-241-16.red.bezeqint.net) |
| 18:06.39 | justinu|laptop | myself and others havent been able to make it work right |
| 18:07.14 | sevard | justinu|laptop: alright, i just found the TE405P signaling page |
| 18:07.28 | sevard | it includes Wink (E&M) but for nubsake i should stray from that? |
| 18:07.34 | jake1932 | bzbw: if it's failing, it should should which module is trying the lookup |
| 18:07.41 | jake1932 | should say |
| 18:07.50 | *** join/#asterisk inv_arp[work] (i=junya@adsl-10-153-159.mia.bellsouth.net) |
| 18:08.00 | justinu|laptop | sevard: what do you need to do with it? |
| 18:08.06 | bzbw | I have a BroadVoice context in sip.conf for dialing out using my BV account, and the * box itself has a dns name called asterisk.mydomain.com, and it is sending BroadVoice.mydomain.com DNS query, why? |
| 18:08.21 | sevard | justinu|laptop: i'm buying pri service from my telco |
| 18:08.36 | sevard | justinu|laptop: they asked if my card does mediated start, wink start, or ground start |
| 18:08.49 | bzbw | jake1932: it should be sip.conf, right? |
| 18:08.53 | *** join/#asterisk Mike (n=mike@dsl-201-129-119-118.prod-infinitum.com.mx) |
| 18:08.56 | *** part/#asterisk Alystair (n=bob@CPE001109c15241-CM00407b8794db.cpe.net.cable.rogers.com) |
| 18:08.56 | jake1932 | bzbw: looks like you missed ".com" or ".net" |
| 18:08.56 | justinu|laptop | sevard: ok, they're on crack... tell them that PRI is mutually exclusive from wink/immediate start. |
| 18:09.00 | sevard | it's looking like i should do ground start |
| 18:09.02 | justinu|laptop | sevard: stupid telco morons |
| 18:09.08 | sevard | justinu|laptop: what do you mean? |
| 18:09.09 | jake1932 | bzbw: it's using the default dns suffix |
| 18:09.10 | justinu|laptop | PRI is common channel signalling |
| 18:09.16 | Mike | how can i make a channel that always gives me all circuits busy? like a sip channel incominglimit=0? |
| 18:09.38 | justinu|laptop | all those other signalling protocols aren't used with PRI... it's either or. |
| 18:10.05 | bzbw | jake1932: how do I get rid of it? I don't want * to use the suffix for my * box and instead, use broadvoice.com for the suffix. |
| 18:10.35 | bzbw | jake1932: for dialing out using BV accounts. |
| 18:10.39 | justinu|laptop | sevard: it's unfortunate that your order entry people are so clueless, but the provisioners who configure their switches will understand. |
| 18:10.50 | justinu|laptop | s/your/your telco's/ |
| 18:11.04 | jake1932 | bzbw: most likely it's somewhere in your sip.conf file that says "broadvoice" instead of "broadvoice.com" |
| 18:11.13 | CunningPike | justinu|laptop, sevard: It's frightening how moronic some telcos can be |
| 18:11.24 | jake1932 | either in the register line or somewhere in the BV context |
| 18:11.40 | sevard | justinu|laptop: so PRI is a signaling method and wink/ground/immediate start are not encapsulated by PRI |
| 18:11.44 | bzbw | jake1932: yes, I define the context as BroadVoice instead of BroadVoice.com. |
| 18:11.54 | justinu|laptop | sevard: that's right. |
| 18:12.07 | justinu|laptop | wink/ground/immediate start are NOT applicable on a PRI |
| 18:12.20 | bzbw | Jake1932: but inside the context, it defines host=sip.broadvoice.com and fromdomain=sip.broadvoice.com |
| 18:12.32 | mut | immediate! |
| 18:12.40 | mut | like a partyline! |
| 18:12.42 | mut | woohoo |
| 18:12.45 | jake1932 | what about the register line? |
| 18:12.55 | sevard | justinu|laptop: that's where the confusion lies, where are they applicable? |
| 18:12.59 | bzbw | jake1932: and in extensions.conf, I use Dial(SIP/${EXTEN}@BroadVoice) |
| 18:13.10 | LostFrog | Party??? Where? |
| 18:13.28 | justinu|laptop | sevard: those signalling methods apply to a channel associated signalling t1 |
| 18:13.32 | justinu|laptop | older technology |
| 18:13.34 | bzbw | Jake1932: I thought it will resolve BroadVoice to the right peer for calling out:) |
| 18:13.46 | jake1932 | <PROTECTED> |
| 18:13.51 | justinu|laptop | sevard: those things are commonly used when talking to a channel bank connected to analog phones. |
| 18:14.15 | bzbw | jake1932: the strange thing is, it was working till a week ago:( |
| 18:14.21 | jake1932 | <PROTECTED> |
| 18:14.36 | bzbw | jake1932: yes |
| 18:15.33 | jake1932 | it clearly thinks broadvoice is a host name and not a peer entry |
| 18:15.45 | *** join/#asterisk ToTo (n=ToTo@host187-131.pool872.interbusiness.it) |
| 18:15.55 | bzbw | why??? |
| 18:16.07 | jake1932 | can you pastebin both? |
| 18:16.17 | jake1932 | (just the relevant portions) |
| 18:16.21 | blitzrage | bzbw: that is the wrong formatting -- SIP/BroadVoice/${EXTEN} is what you want |
| 18:16.44 | bzbw | blitzrage: I tried that format, same thing. |
| 18:16.46 | jake1932 | blitzrage: that format works for me |
| 18:16.47 | blitzrage | if you use an @, its going to assume a hostname |
| 18:17.07 | jake1932 | maybe it's being phased out though |
| 18:17.08 | blitzrage | bzbw: what does your sip.conf entry look like? (pastebin) |
| 18:17.22 | bzbw | blitzrage: let me get them pastebin |
| 18:19.01 | sevard | justinu|laptop: alright, the problem wasn't with the telco but the person who gave me the notes from the telco lady was retarded |
| 18:19.06 | sevard | i called her directly and sorted it out |
| 18:19.08 | sevard | thank you |
| 18:19.15 | mut | anyone know any settings that might help fax over voip? |
| 18:19.18 | *** join/#asterisk a1fa (n=a1fa@207.210.210.202) |
| 18:19.21 | mut | like more fxs impedence |
| 18:19.26 | mut | or no echo cancel on the ata |
| 18:19.27 | mut | or something |
| 18:19.37 | a1fa | what wa s the guys name that owns kneedraggers.com? |
| 18:19.49 | a1fa | -- |
| 18:20.01 | mut | anyone |
| 18:21.35 | a1fa | no |
| 18:21.37 | bzbw | here is the related info, I hide the real configuration:) http://pastebin.ca/52629 |
| 18:22.01 | justinu|laptop | sam |
| 18:22.15 | bzbw | btw: I'm using 1.2.7.1 |
| 18:24.26 | bzbw | does network setting in the * box affect the DNS query from *? |
| 18:25.33 | lunk | jake1932: hey man, changing the dtfm mode to rfc2833 solved the problem, thanks a bunch |
| 18:25.39 | jake1932 | np |
| 18:26.22 | bzbw | blitzrage: any clue? |
| 18:27.24 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 18:28.17 | jake1932 | <PROTECTED> |
| 18:28.19 | jake1932 | ? |
| 18:28.25 | blitzrage | bzbw: need to see what the CLI is doing |
| 18:28.38 | blitzrage | bzbw: and get rid of dtmf=inband, thats an invalid option |
| 18:29.40 | bzbw | k. there is no error, the sip invite is sent to the wrong dns name with BroadVoice.mydomain.com:( |
| 18:31.01 | SpaceBass | I have a cisco 7940 that is behaving strangely |
| 18:31.05 | SpaceBass | I'm using a modified POE cable from the wiki and it powers on fine, but says "ethernet disconnected" ... i get the same result if i use a standard able and a cisco power cube...however I have ONE port on an older switch on which it does work |
| 18:31.10 | SpaceBass | anyone seen anything like that |
| 18:31.21 | jake1932 | unless BroadVoice.mydomain.com is a valid host, there should be an error |
| 18:31.24 | Qwell[] | SpaceBass: wrong port? |
| 18:31.39 | Qwell[] | ie; the "working" switch port is auto-sensing? |
| 18:32.01 | SpaceBass | Qwell[], the older swich is retired due to a VERY loud sound it makes constently.... the new one is an unmanaged autosensing switch |
| 18:32.05 | *** join/#asterisk saftsack (n=saftsack@p54A7FC22.dip.t-dialin.net) |
| 18:32.30 | justinu|laptop | maybe the speed negotiation isn't working right |
| 18:32.38 | SpaceBass | Qwell[], I checked the old managed switch carefully and could not find a thing different about that port...it was set to full duplex, autosensing, 100mbs... |
| 18:32.58 | SpaceBass | justinu|laptop, could be....thought about checking the settings in the phone to see if its dialed down to 10mbs or something |
| 18:33.02 | Qwell[] | which port are you using on the phone? |
| 18:33.24 | SpaceBass | 10/100sw |
| 18:33.26 | SpaceBass | :) |
| 18:33.42 | Qwell[] | use the middle one...whatever it says :p |
| 18:33.53 | Qwell[] | rs323, lan, switch...something like that |
| 18:34.25 | SpaceBass | settings on the phone are media type: auto, |
| 18:34.48 | Qwell[] | iirc, the switch port is the far left one... |
| 18:34.52 | SpaceBass | the odd thing is that its not the crazy modified POE cable I made |
| 18:35.01 | SpaceBass | Qwell[], far left, yeah |
| 18:35.06 | Qwell[] | yeah, use the middle one |
| 18:35.11 | SpaceBass | (well, between aux on the left and the PC port on the right) |
| 18:35.16 | SpaceBass | using the middle one |
| 18:35.30 | Qwell[] | it can't be far left AND middle :P |
| 18:35.36 | Strom_C | did somebody say Cisco?! |
| 18:35.39 | bzbw | jake1932: http://pastebin.ca/52636 |
| 18:36.01 | SpaceBass | using the middle one |
| 18:36.08 | Qwell[] | okay then |
| 18:36.28 | Qwell[] | straight through cable doesn't work? |
| 18:36.40 | SpaceBass | the Network port 2 device type is set to hub/switch and I can change it to PC |
| 18:36.49 | SpaceBass | not sure what the diference is...guess slip cable vs straight through |
| 18:37.04 | SpaceBass | Qwell[], yeah, if I use the power supply and a regular straight cable, i still get the error |
| 18:37.12 | bzbw | jake1932: the dns did respond, but the ip is not bv proxy:( |
| 18:37.13 | Qwell[] | then it's busted :p |
| 18:37.20 | SpaceBass | d'oj! |
| 18:37.22 | SpaceBass | d'oh |
| 18:37.27 | SpaceBass | no smartnet |
| 18:37.53 | Qwell[] | make a crazy two-connector crossover/poe cable |
| 18:37.56 | SpaceBass | that means I have to take the phone out of my master bathroom in the mean time |
| 18:37.56 | Qwell[] | :P |
| 18:38.04 | SpaceBass | :) |
| 18:38.11 | Qwell[] | Or just a regular one...I imagine the pc port can take poe |
| 18:38.51 | SpaceBass | regular what? phone? Don't have any analogue phones or a spare ATA....i took great happiness in smashing my old POS cordless when I went all VoIP |
| 18:38.51 | SpaceBass | :) |
| 18:39.07 | Qwell[] | no, regular crossover/poe cable |
| 18:39.22 | Strom_C | jeez, and here I still use a rotary phone made in 1948 |
| 18:39.53 | jake1932 | bzbw: i tried to replicate what you're doing without an error. every time i put in an invalid host, it comes up with an error. |
| 18:40.00 | Qwell[] | Strom_C: Do you have to crank it? |
| 18:40.03 | Qwell[] | if not...pfft |
| 18:40.12 | Strom_C | Qwell[]: I said rotary, not magneto |
| 18:40.17 | Strom_C | sheesh |
| 18:40.32 | SpaceBass | HEY HEY! I changed it to 10mb full duplex and it worked! |
| 18:40.35 | SpaceBass | eurika! |
| 18:40.45 | SpaceBass | nope...false alarm |
| 18:41.00 | SpaceBass | just pretended to work until I tried to make a call |
| 18:42.11 | bzbw | jake1932: no, the BroadVoice.mydomain.com does resolve to multiple A record, which is pointing to multiple mydomain.com's IP, but they are not valid sip proxy, not BV proxy:( |
| 18:42.13 | LostFrog | Damn faker phones. :) |
| 18:42.29 | SpaceBass | hosting your own BV proxy? |
| 18:42.51 | jake1932 | bzbw: maybe backup your sip.conf and create one with just the bv register and peer entry. the err seems to have to do with not using BroadVoice as a peer entry |
| 18:42.53 | bzbw | jake1932: the issue is, why does it put "mydomain.com" suffix after the context "BroadVoice"? |
| 18:43.29 | SpaceBass | have a DNS search suffix on that box? |
| 18:43.29 | Mike | can i create a zap channel with out a zap card? just so i can create a zap channel that always returns busy? |
| 18:43.29 | jake1932 | because it doesn't think BroadVoice is a peer entry - it thinks it's a host |
| 18:43.41 | SpaceBass | Mike check out zapdummy on the wiki |
| 18:43.43 | bzbw | SpaceBass: I'm just trying to make a peer entry works, which is using BV accont |
| 18:44.00 | SpaceBass | bzbw gotchat...have you tried using the ip or putting an entry in the /etc/host file? |
| 18:44.44 | bzbw | SpaceBass: I did, it does not work either, looks like BV proxy want From, To header with "sip.broadvoice.com" instead of ip address:( |
| 18:45.01 | bzbw | SpaceBass: i mean in the SIP INVITE. |
| 18:45.42 | SpaceBass | bzbw, gotcha.... I have sip.broadvoice.com in my sip.conf and then i have a line in /etc/host for the Washington DC server |
| 18:45.47 | Mike | SpaceBass, no zapdummy just ztdummy |
| 18:45.58 | SpaceBass | Mike oops...good catch |
| 18:46.22 | Mike | SpaceBass, thats just a timer |
| 18:46.37 | Mike | SpaceBass, i need a zap channel that always returns all circuits busy to get a 34 message |
| 18:46.43 | Mike | on my hangucause |
| 18:46.45 | SpaceBass | Mike thought that might do it....if you want something that will always return busy, why does it have to be ZAP? |
| 18:46.50 | SpaceBass | ah |
| 18:46.52 | bzbw | SpaceBass: I have this: "147.135.20.128 sip.broadvoice.com" in /etc/hosts |
| 18:47.02 | Mike | SpaceBass, hangupcause |
| 18:47.48 | bzbw | jake1932: u r right, why does it think it is NOT a peer:( |
| 18:48.37 | jake1932 | maybe your sip.conf may has an issue - check my earlier suggestion |
| 18:48.47 | jake1932 | ooh - horible english |
| 18:48.49 | bzbw | jake1932: and funny things is, how does the * knows the suffix of my linux box and attach it to the context name? |
| 18:49.07 | jake1932 | it doesn't |
| 18:49.17 | jake1932 | it's appending the default suffix |
| 18:49.34 | bzbw | jake1932: where does default suffix defined in *? |
| 18:49.41 | jake1932 | it's not |
| 18:49.52 | jake1932 | it's in your dns file - resolv.conf (i think) |
| 18:49.54 | bzbw | then where does it gotten this? |
| 18:50.25 | bzbw | I don't have domain defined in the /etc/resolv.conf. |
| 18:50.34 | SpaceBass | DHCP? |
| 18:51.15 | jake1932 | resolver.conf |
| 18:51.25 | bzbw | no static ip |
| 18:51.48 | SpaceBass | hummmm |
| 18:52.23 | bzbw | jake1932: there is no resolver.conf in my linux:( |
| 18:53.12 | jake1932 | i had it right the first time |
| 18:53.21 | jake1932 | -- /etc/resolv.conf. |
| 18:53.21 | nextime | bzbw : /etc/resolv.conf, not resolver |
| 18:53.55 | bzbw | that's why I meant, it's just a couple name server with ip address. |
| 18:54.43 | nextime | bzbw : search domain.tld at the first line of your resolv.conf plus a configured zone in your dns server is what you need. |
| 18:55.18 | jake1932 | bzbw - did you try backing up your sip.conf and creating a new one with just the broadvoice (and one phone)? |
| 18:55.22 | nextime | ( if i understand right what you need ) |
| 18:55.56 | *** join/#asterisk mercestes (n=merceste@69.15.174.114) |
| 18:56.02 | bzbw | nextime: I don't control the DNS server:(, but what do u mean "domain.tld"? |
| 18:56.41 | bzbw | jake1932: this is a production server, I will have to have a sleepless night again to do so. |
| 18:56.45 | SpaceBass | ARRRUUGGGG this crazy ass cisco phone just started working...and i didnt change a thing |
| 18:56.54 | SpaceBass | do I dare move it back to the kitchen where it lives? |
| 18:57.33 | jake1932 | bzbw: what did you change before it stopped working? |
| 18:58.42 | bzbw | as far as I know, not a thing, but if I do remember, I might have already solved this puzzle:). |
| 18:59.16 | jake1932 | did it create a backup file for you? |
| 18:59.25 | nextime | bzbw : google.com, google is a domain, .com is a tld :) |
| 18:59.26 | bzbw | I did upgraded it to 1.2.7.1, but that's after it was not working, so I don't think 1.2.7.1 is the reason. |
| 18:59.59 | bzbw | nextime: thx. |
| 19:00.00 | jake1932 | i believe your problem to be in sip.conf |
| 19:00.16 | nextime | if you have your own domain to resolve in your network, you will use search domain.tld in resolv.conf, it's simple |
| 19:00.19 | bzbw | but what u see is what i have:) |
| 19:00.25 | jake1932 | well |
| 19:00.29 | jake1932 | that's part of it |
| 19:01.09 | jake1932 | for instance - i'm using 1.2.7.1 and doing sip lookups just fine |
| 19:01.15 | *** join/#asterisk tdonahue-laptop (n=tdonahue@www.vonworldwide.com) |
| 19:01.19 | nextime | startrek enterprise is starting, bye bye |
| 19:02.17 | jake1932 | is BV your only peer? |
| 19:02.30 | jake1932 | (besides your phones)? |
| 19:02.59 | bzbw | jake1932: I have another peer with is using SER, same thing:( |
| 19:03.31 | *** join/#asterisk TripleFFFFFFFFFF (n=TripleFF@147-102.mc.cite.net) |
| 19:03.40 | *** part/#asterisk TripleFFFFFFFFFF (n=TripleFF@147-102.mc.cite.net) |
| 19:03.47 | *** join/#asterisk TripleFFFFFFFFFF (n=TripleFF@147-102.mc.cite.net) |
| 19:03.53 | jake1932 | the plot thickens |
| 19:03.57 | TripleFFFFFFFFFF | _1NXXNXXXXXX |
| 19:04.05 | TripleFFFFFFFFFF | how i make this with or without the 1 |
| 19:04.26 | TripleFFFFFFFFFF | to match 123-123-1234 and 1-123-123-1234 |
| 19:04.38 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
| 19:04.43 | jake1932 | _NXXNXXXXXX |
| 19:04.49 | [TK]D-Fender | TripleFFFFFFFFFF : Add another for _NXXNXXXXXX |
| 19:04.59 | [TK]D-Fender | jake1932 : Ok, you win this one :) |
| 19:05.22 | TheCops | Hello Fender |
| 19:05.22 | TheCops | :) |
| 19:05.33 | jake1932 | 1 for me, 500 for [TK]D-Fender |
| 19:06.26 | TripleFFFFFFFFFF | ?? |
| 19:06.29 | TripleFFFFFFFFFF | AS IN |
| 19:07.12 | [TK]D-Fender | :D |
| 19:07.17 | [TK]D-Fender | TripleFFFFFFFFFF : ADD ANOTHER PATTERN |
| 19:07.21 | TripleFFFFFFFFFF | <PROTECTED> |
| 19:07.23 | [TK]D-Fender | TheCops : y0 |
| 19:07.29 | brodiem | Is there a difference between the open source G729 and the licensed G729 w/ *? |
| 19:07.49 | bzbw | jake1932: is there a cmd that allows me see how * execute dialing for a context? |
| 19:07.49 | TheCops | [TK]D-Fender. how are you |
| 19:07.55 | [TK]D-Fender | TripleFFFFFFFFFF : You DON'T make one that will work with both patterns, you make 1 EACH to account for EACH pattern |
| 19:08.01 | [TK]D-Fender | TheCops : Still breathing. |
| 19:08.08 | TripleFFFFFFFFFF | k |
| 19:08.20 | TripleFFFFFFFFFF | i know i tought we could make as one |
| 19:08.21 | TripleFFFFFFFFFF | sucks |
| 19:08.28 | jake1932 | bzbw: set verbose 30 |
| 19:08.29 | *** join/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 19:08.32 | TripleFFFFFFFFFF | shoul have {!} |
| 19:08.37 | TripleFFFFFFFFFF | [1] |
| 19:08.39 | TripleFFFFFFFFFF | I MEAN |
| 19:08.43 | TripleFFFFFFFFFF | i mean |
| 19:08.43 | [TK]D-Fender | TripleFFFFFFFFFF : nope |
| 19:08.48 | bzbw | jake1932: I set it to more than 37:( |
| 19:10.30 | *** join/#asterisk willt (i=wt@wifi-napanet-static-206-81-99-68.napanet.net) |
| 19:11.10 | willt | Does anyone know if I can disable the speakerphone on a cisco 7960 without using callmanager? |
| 19:11.33 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 19:11.39 | jake1932 | bzbw: did you try pinging BroadVoice |
| 19:11.42 | SpaceBass | willt, I'm not aware of a setting in the sip config to do that |
| 19:11.50 | jake1932 | bzbw: does it append the suffix? |
| 19:12.21 | TripleFFFFFFFFFF | http://pastebin.ca/52650 |
| 19:12.24 | willt | I know callmanager can do it but I can't find anything in the sip config.. |
| 19:12.25 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 19:12.26 | TripleFFFFFFFFFF | so this would make sense ? |
| 19:12.33 | TripleFFFFFFFFFF | to make group res have only 4 channels ? |
| 19:12.35 | bzbw | jake1932: from console? |
| 19:12.38 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 19:13.01 | TripleFFFFFFFFFF | so in extensions i would macro res.. then dial whereever |
| 19:13.24 | brodiem | is anyone using the intel based g729 implemntation? |
| 19:13.25 | jake1932 | from a cmd line |
| 19:13.49 | [TK]D-Fender | TripleFFFFFFFFFF : res? |
| 19:13.56 | TripleFFFFFFFFFF | residential |
| 19:13.57 | TripleFFFFFFFFFF | lol |
| 19:14.00 | TripleFFFFFFFFFF | like a service |
| 19:14.11 | bzbw | jake1932: hey, u are good, it did responded, just as I described, it tried BroadVoice.mydomain.com and the server responded |
| 19:14.11 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 19:14.19 | [TK]D-Fender | TripleFFFFFFFFFF : I would basically manipulate the # as required to format it so the dial doesn't have to do the thinking. |
| 19:14.21 | TripleFFFFFFFFFF | so my users before lettimg them receive the call i would do that |
| 19:14.31 | bzbw | jake1932: how do I get rid of this? |
| 19:14.47 | bzbw | I did not specify anywhere for this |
| 19:15.24 | TripleFFFFFFFFFF | so in extensions.. => context incoming-user ... prio 1 would be set account =blah , prio 2 , macro res, prio 3 dial wherever |
| 19:16.47 | jake1932 | bzbw: try pinging something else |
| 19:17.02 | jake1932 | sounds like 2 issues |
| 19:17.30 | jake1932 | how come BroadVoice.mydomain.com is responding? |
| 19:17.46 | TripleFFFFFFFFFF | would it work or not |
| 19:17.49 | TripleFFFFFFFFFF | how would one do it ? |
| 19:18.02 | bzbw | jake1932: I found something, let me delete that file first, it is in networking/profiles/default/resolv.conf: |
| 19:18.57 | generalhan | [TK]D-Fender: ok so i got it to work with one channel for the ChanIsAvail cmd. now my question is can i have the queue, call on that entry that i made in extensions.conf ? |
| 19:19.06 | *** join/#asterisk Weezey (n=ohno@lo20.loit.ca) |
| 19:19.31 | bzbw | jake1932: does not work. I try to delete that file:( |
| 19:19.34 | [TK]D-Fender | generalhan : I would have though you already had it doing that... how is it calling them NOW? |
| 19:19.38 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:19.53 | *** part/#asterisk terr_ (n=terr_@dsl-cap-66-18-218-43-cgy.nucleus.com) |
| 19:20.31 | jake1932 | bzbw: not sure if you have to restart networking to reread resolv.conf |
| 19:20.56 | TripleFFFFFFFFFF | cro_exec: Context 'macro-res' for macro 'res' lacks 's' extension, priority 1 |
| 19:21.01 | TripleFFFFFFFFFF | oh well |
| 19:21.11 | generalhan | [TK]D-Fender: well i just set up a context in extensions.conf and called it to make sure that it worked... and i got it to work so that if i call an extension it goes to a macro with that stuff in it to decide if they are avail. or not. now i want to set that context to the queue so that all of the members that are dialed have to do through that macro to see if they are available |
| 19:21.50 | Weezey | anyone use realtime? I added an IAX2 peer and it answers with congestion when I dial IAX2/user@context/${EXTEN} I had this problem before with another one and I can't remember how I ended up fixing it. |
| 19:22.08 | Weezey | If I dial user:secret@host it goes just fine. |
| 19:22.19 | Weezey | but I'd like to keep the secret out of the CDR |
| 19:24.02 | bzbw | jake1932: I did restart the networking many times, not working. |
| 19:24.22 | jake1932 | my asterisk experience is better than my linux |
| 19:24.34 | jake1932 | you gots some DNS issues |
| 19:25.35 | bzbw | jake1932: yes, why does this damn thing automatically attach a domain, where I can not find it:(( |
| 19:26.01 | Qwell[] | Did you do what blitzrage suggested, and change the Dial string? |
| 19:26.16 | Qwell[] | Dial(SIP/BroadVoice/${EXTEN}) |
| 19:26.30 | *** join/#asterisk Druken (n=Druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com) |
| 19:26.43 | *** part/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 19:27.44 | Hmmhesays | well that was a pleasant lunch |
| 19:28.00 | Druken | was it? |
| 19:28.08 | Hmmhesays | yes |
| 19:34.02 | bzbw | jake1932: Thx for the help, I somehow work around the problem and it is now working:) |
| 19:34.43 | hwt | are there any good asterisk stress testing tools out there? |
| 19:34.47 | bzbw | jake1932: your tip on pinging BroadVoice from cmd helped me:) |
| 19:34.58 | jake1932 | very good |
| 19:34.58 | TripleFFFFFFFFFF | oh god |
| 19:35.02 | hwt | or SIP initiators that i can use to stress test. |
| 19:35.06 | bzbw | bye now |
| 19:35.08 | *** part/#asterisk bzbw (i=bwz@ip67-153-142-109.z142-153-67.customer.algx.net) |
| 19:35.13 | TripleFFFFFFFFFF | http://pastebin.ca/52662 |
| 19:35.18 | TripleFFFFFFFFFF | doesnt work |
| 19:35.26 | trelane_ | hwt, astertest |
| 19:35.27 | TripleFFFFFFFFFF | just passes by it even when i called from 3 phones at same time |
| 19:35.32 | TripleFFFFFFFFFF | i see the noop etc |
| 19:35.38 | TripleFFFFFFFFFF | and it rigns my local cisco |
| 19:35.56 | hwt | trelane_: it doesn't seem to be working properly on 1.2.x. |
| 19:36.19 | hwt | noop() is just for debugging, right? |
| 19:36.38 | Qwell[] | hwt: sipp |
| 19:37.12 | TripleFFFFFFFFFF | y |
| 19:37.28 | trelane_ | hwt, works for me? |
| 19:37.41 | trelane_ | hwt, I use astertest here at work |
| 19:37.48 | TripleFFFFFFFFFF | god |
| 19:37.56 | trelane_ | TripleFFFFFFFFFF, yes? |
| 19:38.02 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 19:38.02 | TripleFFFFFFFFFF | anyway to limit # of channel incoming + outgoing for a user part of a group ? |
| 19:38.05 | trelane_ | though that might be overly presumptious of me, there may be others |
| 19:38.07 | eric_hill | trelane_: lol! |
| 19:38.15 | TripleFFFFFFFFFF | like .. DUMMIES could have all 2 channels each |
| 19:38.20 | TripleFFFFFFFFFF | and GURUS 100 |
| 19:38.33 | TripleFFFFFFFFFF | but you could have if we want 500 GURUS.. each with 100 channels |
| 19:38.41 | hwt | Qwell[]: intriguing, but i would prefer something that also establishes RTP-channels. |
| 19:38.46 | Qwell[] | yeah, sipp |
| 19:38.48 | TripleFFFFFFFFFF | i DONT mean 100 total .. but 100 for that user |
| 19:38.54 | TripleFFFFFFFFFF | ?? |
| 19:39.13 | trelane_ | Qwell, sipp took effort to understand, I had astertest up in minutes and it draws nifty graphs |
| 19:39.13 | TripleFFFFFFFFFF | so lets say we had 100 dummies thats a max of 200 channels |
| 19:39.14 | hwt | Qwell[]: ah, "SIPp can also send media (RTP) traffic through RTP echo and RTP / pcap replay." |
| 19:39.18 | hwt | Qwell[]: missed a line. |
| 19:39.40 | hwt | trelane_: well, you have to recompile asterisk. that kinda sucks. |
| 19:40.02 | trelane_ | hwt, what makes you recompile asterisk for astertest? |
| 19:40.44 | *** join/#asterisk SuperLag (n=aaron@gentoo/developer/SuperLag) |
| 19:40.52 | hwt | trelane_: i just looked at this one: http://www.asteriskguru.com/tutorials/astertest.html |
| 19:41.08 | trelane_ | hwt, you know how to use managers.conf? |
| 19:41.50 | TripleFFFFFFFFFF | darn |
| 19:41.52 | trelane_ | you dont' *NEED* those modules |
| 19:41.56 | TripleFFFFFFFFFF | ok i need group = account number lol |
| 19:42.33 | hwt | trelane_: yeah? |
| 19:42.50 | hwt | trelane_: really? what are they for, then? |
| 19:42.57 | hwt | trelane_: i should probably rtfm, though. :) |
| 19:44.55 | trelane_ | hwt: they can return cpu/memory load metrics from the two asterisk systems |
| 19:45.10 | *** join/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 19:46.48 | TripleFFFFFFFFFF | works |
| 19:47.00 | Zodiacal | anyone know if theres a way to make the button press tones quieter? |
| 19:47.12 | hwt | trelane_: ah, but i feel that's a pretty nice feature. |
| 19:47.46 | TripleFFFFFFFFFF | ok thanks for chatting lol actually put my mind aside and helped me out |
| 19:48.06 | Hmmhesays | Zodiacal: ear muffs |
| 19:48.13 | Zodiacal | :P |
| 19:48.31 | Zodiacal | might not be a bad idea, the a/c here is allways on full blast. |
| 19:49.15 | Zodiacal | i guess i would have to increase the volume of the audio and lower the handsets volume... |
| 19:49.26 | Zodiacal | but if i increase the gain's will that increase the tones too? |
| 19:49.57 | Hmmhesays | on what hardware? |
| 19:50.17 | Zodiacal | cisco 7960 phones with digim tdm400p fxo's |
| 19:50.25 | Hmmhesays | which tones are loud? |
| 19:50.38 | Zodiacal | when i press a button on the phone, i.e. 1,2,3, etc |
| 19:51.03 | Zodiacal | it hurts when calling automated systems that require button presses |
| 19:51.11 | Hmmhesays | on the 7960? |
| 19:51.16 | Zodiacal | yep |
| 19:51.22 | Hmmhesays | that is really odd |
| 19:51.24 | Hmmhesays | mine doesn't do that |
| 19:51.29 | Zodiacal | sccp |
| 19:51.32 | Hmmhesays | oh |
| 19:51.56 | Zodiacal | ya sip was softer |
| 19:52.06 | Zodiacal | i remember now |
| 19:52.14 | Zodiacal | small draw back |
| 19:52.54 | *** join/#asterisk IceManRISK (n=kart@200.138.147.142) |
| 19:53.01 | Hmmhesays | he's on teh dance floor yelling freebird |
| 19:56.55 | TripleFFFFFFFFFF | at least not yelling freeewilly |
| 19:57.08 | noname32 | lol |
| 19:57.10 | Hmmhesays | that comes later in the night |
| 19:57.38 | TripleFFFFFFFFFF | my 7960 firmware is fucked i think |
| 19:57.49 | TripleFFFFFFFFFF | each time i go adimin on it ti fizzles ,m wiggles and reboots on me |
| 19:58.02 | TripleFFFFFFFFFF | and htat with only 1 finger |
| 19:58.22 | TripleFFFFFFFFFF | might be that the phone heard too much crazy shit and went nuts |
| 19:58.31 | TripleFFFFFFFFFF | bah cisco not wht they used to be |
| 19:58.35 | Hmmhesays | how do I use place holders in a bash script |
| 19:58.44 | TripleFFFFFFFFFF | place holders ? |
| 19:58.47 | TripleFFFFFFFFFF | as in ?\ |
| 19:58.47 | C4T3l | PAGING MERCESTES |
| 19:58.49 | TripleFFFFFFFFFF | ARGS ? |
| 19:58.52 | Hmmhesays | yes |
| 19:58.58 | TripleFFFFFFFFFF | ${1} |
| 19:59.25 | Hmmhesays | ./my_script boobies weee ${1} is boobies and ${2} is weeee? |
| 19:59.42 | Hmmhesays | minus the typo |
| 19:59.55 | TripleFFFFFFFFFF | ./test.sh 1 |
| 19:59.55 | TripleFFFFFFFFFF | 1 |
| 20:00.07 | TripleFFFFFFFFFF | echo "${1}"; |
| 20:01.00 | TripleFFFFFFFFFF | so echo "${1} is my first arg and ${2} is my second" my third is in a manual , and my whole is where ? |
| 20:01.10 | Qwell[] | $0 |
| 20:01.13 | TripleFFFFFFFFFF | try |
| 20:01.31 | Hmmhesays | got it |
| 20:01.33 | Hmmhesays | danke |
| 20:02.04 | SpaceBass | $0 would be the command (script name) itself, right? |
| 20:02.08 | TripleFFFFFFFFFF | echo "${1}ead{$2}he${3}ucking${4}anual" |
| 20:02.16 | TripleFFFFFFFFFF | and parse it Rtfm |
| 20:02.16 | TripleFFFFFFFFFF | lol |
| 20:02.17 | Qwell[] | maybe |
| 20:02.20 | TripleFFFFFFFFFF | no prob |
| 20:02.23 | TripleFFFFFFFFFF | just having fun with you |
| 20:02.27 | *** join/#asterisk darylp (n=daryl_ju@63-208-162-59.digitalrealm.net) |
| 20:02.33 | TripleFFFFFFFFFF | ${0 is fielname |
| 20:02.57 | SpaceBass | i gotta get around to writing a few bash scripts later today....change some firewall settings |
| 20:02.59 | *** join/#asterisk Samoied (n=Samoied@200-193-14-53.fnsce7003.dsl.brasiltelecom.net.br) |
| 20:03.15 | SpaceBass | I have one I wrote to control iTunes through asterisk....never got around to installing it on my current * box though |
| 20:03.24 | *** join/#asterisk nagl (n=nagl@86.59.54.237) |
| 20:03.39 | SpaceBass | and it had no real point, except that I could call my DISA while I was traveling and crank Guns n Roses at 2 am to scare the shit out of my wife |
| 20:04.18 | darylp | I have a slightly off topic phone related question regarding fwd |
| 20:05.02 | TripleFFFFFFFFFF | lol |
| 20:05.08 | TripleFFFFFFFFFF | sapcebass too neat |
| 20:05.23 | TripleFFFFFFFFFF | had a guy fuck me over 10k |
| 20:05.28 | SpaceBass | im gonna blog the script soon if I get time |
| 20:05.49 | TripleFFFFFFFFFF | i called his GF at 4 am till 5 am randomly every night and him the same but with each others numbers on callerid |
| 20:05.49 | TripleFFFFFFFFFF | lol |
| 20:06.23 | TripleFFFFFFFFFF | what ? who you think you are ? yeah you did call me .. i got your digitts on da phone lol |
| 20:06.28 | TripleFFFFFFFFFF | and vice verca |
| 20:06.29 | SpaceBass | lol! |
| 20:06.37 | *** join/#asterisk PakiPenguin (i=uppal@linuxpakistan/admin/pakipenguin) |
| 20:06.38 | SpaceBass | cid spoofing is the greatest thing since sliced bread |
| 20:06.53 | pigpen | Does Asterisk support Faxes? <<<< Ha Ha! just kidding... |
| 20:06.54 | TripleFFFFFFFFFF | so i guess im now only -9960.89$ |
| 20:07.13 | TripleFFFFFFFFFF | i guess 1 monh of fon costes me like ...39$ for around 5000 calls |
| 20:07.14 | TripleFFFFFFFFFF | not bad |
| 20:07.30 | TripleFFFFFFFFFF | now what do do fo r9960$ |
| 20:07.40 | Qwell[] | TripleFFFFFFFFFF: What's his car worth? :P |
| 20:07.44 | [TK]D-Fender | SpaceBass : Sorry, I've already claimed that one :) |
| 20:07.57 | TripleFFFFFFFFFF | qwell not sure |
| 20:08.05 | TripleFFFFFFFFFF | unpaid |
| 20:08.07 | TripleFFFFFFFFFF | why |
| 20:08.07 | TripleFFFFFFFFFF | lol |
| 20:09.08 | SpaceBass | arrrruuuuggg nerdvittles always locks up firefox on os x |
| 20:09.21 | Qwell[] | SpaceBass: Don't do that then |
| 20:09.39 | darylp | Oval telecom wants to know the sip or iax destination of my uk phone number, how can I send that into a fwd number? |
| 20:09.41 | SpaceBass | well I wanted to read about nvfaxdetect in @home 2.8 :( |
| 20:09.54 | *** join/#asterisk MacDome (n=eseidel@A17-255-100-181.apple.com) |
| 20:10.16 | TripleFFFFFFFFFF | isnt fax in addons yet ? |
| 20:10.20 | TripleFFFFFFFFFF | wth they waiting if not |
| 20:10.43 | darylp | basically I'm trying to understand how one specifies an iax desination |
| 20:10.48 | darylp | destination |
| 20:11.23 | TripleFFFFFFFFFF | user:pass@ip |
| 20:12.58 | drfoomod2 | has anyone mucked around w/ sipX? |
| 20:13.27 | *** part/#asterisk lzhang (n=lewiszha@67.95.13.46) |
| 20:13.31 | *** join/#asterisk lzhang (n=lewiszha@67.95.13.46) |
| 20:13.45 | *** part/#asterisk ringe (n=runar@ti531210a080-6380.bb.online.no) |
| 20:13.54 | a1fa | anybody know where to download agi scripts, weather, wake-up? etc/ |
| 20:14.12 | lzhang | hi guys, so I'm having this problem with Polycom phones and Asterisk, I'm using sip to dial out and the calls turn into one way sound after the third ring |
| 20:14.14 | TripleFFFFFFFFFF | my brain has alot of them.. just need to compile them |
| 20:14.27 | TripleFFFFFFFFFF | prob i got no ports left |
| 20:14.42 | TripleFFFFFFFFFF | lzhang NAT |
| 20:14.46 | TripleFFFFFFFFFF | rtp |
| 20:14.52 | TripleFFFFFFFFFF | connect dircet and test |
| 20:14.57 | TripleFFFFFFFFFF | you are double natted |
| 20:15.09 | TripleFFFFFFFFFF | SERVER -> NAT -> NET -> NAT -> PHONE |
| 20:15.22 | TripleFFFFFFFFFF | make sure nat=yes in sip.conf peer |
| 20:15.24 | lzhang | the asterisk box is the gateway and has a public ip, and the phones are on an internal network getting dhcp from the asterisk box |
| 20:15.37 | TripleFFFFFFFFFF | make sure nat=yes in sip.conf peer |
| 20:15.55 | TripleFFFFFFFFFF | and insecure=port,invite |
| 20:16.07 | darylp | I know, for example, that ipkall doesn't have my password, only my fwd number. How do they route incoming calls to fwd? |
| 20:16.17 | TripleFFFFFFFFFF | look in rtp.conf an dforward that rang e from your router to your phone as udp |
| 20:16.20 | TripleFFFFFFFFFF | also |
| 20:16.33 | TripleFFFFFFFFFF | ip ? |
| 20:16.44 | lzhang | TripleFFFFFFFFFF: are you sure this is a NAT situation? ip is 66.76.53.31 |
| 20:17.17 | TripleFFFFFFFFFF | checkign oyu |
| 20:17.51 | TripleFFFFFFFFFF | hm,mm |
| 20:17.56 | TripleFFFFFFFFFF | 5060 closed from outside |
| 20:18.27 | lzhang | try again |
| 20:19.02 | TripleFFFFFFFFFF | nope |
| 20:19.03 | TripleFFFFFFFFFF | sorry |
| 20:19.07 | lzhang | hmm |
| 20:19.18 | TripleFFFFFFFFFF | ok what peer name ? |
| 20:19.28 | TripleFFFFFFFFFF | CLI> debug peer pernamehere |
| 20:19.33 | TripleFFFFFFFFFF | and see what it does.. |
| 20:19.46 | TripleFFFFFFFFFF | youll see hwen it stops the rtp stream its because its gonna use internal ips |
| 20:19.50 | TripleFFFFFFFFFF | like 192.168.1.x |
| 20:20.06 | TripleFFFFFFFFFF | on you r asterisk.. now obviousely it cant forwards that to your phone location |
| 20:20.09 | TripleFFFFFFFFFF | thats NAT |
| 20:22.51 | lzhang | k hmmm |
| 20:23.08 | *** join/#asterisk gbodemantv (n=gbodeman@216.142.38.154) |
| 20:23.10 | gbodemantv | hey all |
| 20:23.30 | gbodemantv | anypne ever get May 1 13:16:32 WARNING[14610]: format_wav.c:247 update_header: Unable to find our position |
| 20:23.35 | gbodemantv | repeating over and over |
| 20:23.55 | *** join/#asterisk Abydos313 (i=abydos31@adsl-71-129-63-236.dsl.irvnca.pacbell.net) |
| 20:24.16 | Qwell[] | gbodemantv: What version of *? |
| 20:24.18 | TripleFFFFFFFFFF | no |
| 20:24.21 | lzhang | TripleFFFFFFFFFF: how do you check the port from the outside |
| 20:24.22 | gbodemantv | 1.2.4 |
| 20:24.25 | Qwell[] | upgrade |
| 20:24.35 | Qwell[] | lzhang: nmap |
| 20:24.36 | TripleFFFFFFFFFF | hehehe |
| 20:24.38 | TripleFFFFFFFFFF | nmap |
| 20:26.39 | Druken | what do i want for dinner? |
| 20:26.47 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 20:27.30 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
| 20:27.38 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 20:28.25 | *** part/#asterisk a1fa (n=a1fa@207.210.210.202) |
| 20:30.26 | Weezey | Druken: Beer? |
| 20:30.37 | Druken | nah.... |
| 20:30.43 | Weezey | Wine. |
| 20:30.58 | Druken | i was thinking kfc or something :) |
| 20:31.04 | Druken | not sure tho... |
| 20:31.06 | Weezey | eww |
| 20:31.25 | Weezey | too much fast food has made me hate it all. |
| 20:31.44 | Druken | i should have hit the point, yet i havent... |
| 20:32.04 | Druken | last none fast food meal i had was like a week ago? |
| 20:32.14 | Druken | er.. non-fast food |
| 20:33.19 | *** join/#asterisk BrianR___ (i=brianr@setient-sucks.978.org) |
| 20:33.56 | Weezey | You live near Oakville? |
| 20:34.03 | Weezey | Panago makes a mean pie. |
| 20:34.11 | Druken | barrie |
| 20:34.21 | Weezey | hmm, not close enough to Oakville. |
| 20:34.30 | *** part/#asterisk BrianR___ (i=brianr@setient-sucks.978.org) |
| 20:34.31 | Druken | nope.... |
| 20:34.50 | Weezey | Who's providing dial tone in Barrie these days/ |
| 20:34.51 | Weezey | ? |
| 20:34.58 | Druken | bell ? |
| 20:35.08 | Weezey | via IAX or SIP. |
| 20:35.30 | Druken | sprint/rogers has ilec, but bell is still the clec |
| 20:35.51 | Druken | oh... who's providing did's.... not many |
| 20:35.59 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 20:36.45 | *** join/#asterisk dlynes (n=dlynes@216.251.149.66) |
| 20:38.27 | Druken | Weezey: actually.... i had that backwards, bell is the ILEC, rogers has CLEC |
| 20:39.01 | Druken | bell is the incumbent carrier, rogers is the competitive carrier |
| 20:39.56 | *** join/#asterisk philippel (n=p_lindhe@c-24-19-186-72.hsd1.wa.comcast.net) |
| 20:40.54 | pigpen | In regards to faxing, Fax - Digium FXS--(Asterisk)--Digium FXO/PRI---PSTN works pretty good..... |
| 20:41.01 | pigpen | but it gets screwy when sip or iax is in the middle somewhere.... |
| 20:41.31 | Hmmhesays | ok my GET VARIABLE agi isn't working |
| 20:41.41 | pigpen | or can faxing get screwed even in the first one? |
| 20:42.08 | poisoner | hmm... can anyone tell me a good GUI for asterisk which is free? |
| 20:42.21 | pigpen | for config or stats? |
| 20:42.51 | poisoner | both, if possible |
| 20:42.52 | SpaceBass | anyone know if Broadvoice or Telasip will do a real hard bill rather than charging a credit card? |
| 20:43.05 | pigpen | poisoner, good luck. I wrote my own. |
| 20:43.40 | pigpen | I'm sure lots of people would not like it...but getting me happy was #1..... |
| 20:43.55 | pigpen | poisoner, new to asterisk? |
| 20:45.08 | SpaceBass | poisoner, check out www.archatechs.com and then www.nerdvittles.com |
| 20:45.41 | SpaceBass | start there and then you;ll get a sense of the big idea and the gui stuff is mentioned |
| 20:47.08 | pigpen | it was told to me "start command line" then experiment with gui's after you have an idea what is happening under the hood. |
| 20:47.34 | pigpen | if you are anal regarding "doing it your way" then you will end up back at the beginning. |
| 20:48.37 | poisoner | pigpen: At this moment I'm writing config files by hand. This is good enough for me, I think. |
| 20:48.50 | poisoner | But some GUI will come along with nice scripts |
| 20:49.13 | pigpen | sure...I understand... |
| 20:49.27 | poisoner | And perhaps I will manage a coworker to exchange Ciscos CM against * |
| 20:49.59 | poisoner | But He seems to be one which has Problems with a cli or writing Cisco Configs with vi... |
| 20:50.14 | poisoner | So I want too test some things |
| 20:54.12 | noname32 | vi :< |
| 20:54.18 | noname32 | emacs is the best |
| 20:54.19 | SpaceBass | there is a GUI with some nice scripts :D |
| 20:54.23 | SpaceBass | vim |
| 20:54.37 | noname32 | hehe waits for riot |
| 20:54.38 | robin_sz | quick, lets start an editor war! |
| 20:54.40 | SpaceBass | ln -s vi vim |
| 20:54.47 | noname32 | i like freepbx |
| 20:54.49 | drfoomod2 | nano |
| 20:54.52 | drfoomod2 | be-atch! |
| 20:54.55 | noname32 | nano is just pico ;p |
| 20:55.07 | SpaceBass | and I like to save them in word's native format...you know, since its so interoperable |
| 20:55.15 | TripleFFFFFFFFFF | man i type shitdown so often i symlinked to shitdown |
| 20:55.17 | noname32 | i am die hard pico but for coding emacs is so much nicer |
| 20:55.23 | mut | faxing over voip, i used to have people connected via fax machine -> sip -> asterisk -> sip -> cisco as5350 -> pri |
| 20:55.24 | mut | now i have |
| 20:55.41 | mut | fax machine -> sip -> asterisk -> pri via sangoma a104d |
| 20:55.46 | mut | people can' |
| 20:55.48 | mut | t fax anymore |
| 20:55.51 | mut | ideas why? |
| 20:55.55 | drfoomod2 | nano -w |
| 20:56.02 | drfoomod2 | gotta go wide |
| 20:56.22 | noname32 | poisoner, check out #freepbx see if there gui will do what you want |
| 20:56.26 | robin_sz | emacs is without doubt the best mp3 player I have tried |
| 20:56.31 | noname32 | lol |
| 20:56.32 | robin_sz | apparently, it can edit as well, but thats just a rumour |
| 20:56.43 | mut | oh |
| 20:56.46 | sevard | does anyone know how to use ChanSpy |
| 20:56.49 | mut | fax machine -> spa-2002 ata -> sip |
| 20:57.08 | noname32 | some one needs to make a java aplet that will alow editing dynamic pages stored in db so i can code in emacs on the web :) |
| 20:57.47 | brookshire | mut: i can tell you one thing.. the pci card has nothing to do with it |
| 20:58.06 | *** join/#asterisk mikefoo (n=mikefoo@64.124.169.2) |
| 20:58.07 | mut | thats all that changed |
| 20:58.15 | noname32 | fax + voip = troublesome i though |
| 20:58.16 | mut | so how could it have nothing to do with it |
| 20:58.16 | noname32 | lol |
| 20:58.24 | mut | these people have been working for a year |
| 20:58.25 | mut | or more |
| 20:58.26 | *** join/#asterisk abatista (n=Ariel@70.46.87.158) |
| 20:58.47 | brookshire | mut: fax over sip causes problems |
| 20:58.48 | mut | ok |
| 20:58.51 | brookshire | end of story |
| 20:58.52 | mut | i realize this |
| 20:58.58 | brookshire | that's where you're problem is |
| 20:58.58 | mut | but i'm telling you |
| 20:59.00 | mut | it's worked for a year |
| 20:59.05 | mikefoo | Anyone NOT in the USA, and is able to perform a test call to me in the US? |
| 20:59.07 | MikeJ[Laptop] | still work to the 5300? |
| 20:59.08 | mut | i put the cisco back in |
| 20:59.11 | mut | and it works fine again |
| 20:59.12 | *** join/#asterisk bkw_ (n=brian@adsl-70-143-63-171.dsl.tul2ok.sbcglobal.net) |
| 20:59.14 | mikefoo | I need it to originate from outside of the US |
| 20:59.20 | copland | has anyone used stanaphone with asterisk and had it work |
| 20:59.30 | Druken | mikefoo: yeah |
| 20:59.39 | mut | it's not just because it's 'troublesome' |
| 20:59.57 | brookshire | it's because asterisk does not support t38 |
| 21:00.05 | mut | neither does the cisco |
| 21:00.10 | mut | it's just sending it via sip to the cisco |
| 21:00.17 | mut | no special encodings |
| 21:00.21 | mut | just ulaw |
| 21:00.32 | poisoner | mikefoo: give number |
| 21:00.38 | Druken | might help.. hehe |
| 21:00.54 | brookshire | mut: the problem still exists with voip |
| 21:01.05 | *** join/#asterisk wunderkin (i=kev@69.26.192.234) |
| 21:01.05 | mut | heh |
| 21:01.06 | drfoomod2 | robin_sz: that;s like the old joke the emacs is a great OS, it just lacks a good editor |
| 21:01.11 | mut | i'm deoubling the voip and it works better |
| 21:01.14 | mut | doubling* |
| 21:01.24 | mut | if i remove one of the VoIP hops then the problem starts |
| 21:01.36 | mut | how can 'voip/sip is troublesome |
| 21:01.39 | mut | <PROTECTED> |
| 21:01.44 | Druken | mut: doesn't the need for a foip machine? :) i'd like to see a sip/t38 compliant fax machine |
| 21:02.04 | brookshire | mut: jitter messes up faxes |
| 21:02.19 | mut | so theres more jitter by taking out a voip hop? |
| 21:02.25 | justinu|laptop | packet loss messes up faxes |
| 21:02.33 | justinu|laptop | since there's no PLC |
| 21:02.34 | mut | so theres more packet loss by taking out a voip hop? |
| 21:02.51 | MikeJ[Laptop] | is sip jb in tree now? |
| 21:02.56 | brookshire | mut: with sip, unless you are doing a passthrough, it's just going to talk directly endpoint to endpoint |
| 21:03.19 | mut | asterisk stays in the call |
| 21:03.27 | mut | it doesn't bridge it |
| 21:04.02 | Druken | asterisk doesn't handle sip handoff very well, imo |
| 21:04.19 | mut | or does it and somehow |
| 21:04.24 | *** join/#asterisk Disgrntld (n=ahahah@CPE-65-30-153-8.wi.res.rr.com) |
| 21:04.28 | mut | hm |
| 21:04.54 | *** part/#asterisk mikefoo (n=mikefoo@64.124.169.2) |
| 21:04.59 | Disgrntld | im tryng to make a call from extension 5000 and i get a recording saying that extension 5000 is unavailable? what could be wrong? |
| 21:06.02 | VoicePulse | mut: Why has the PCI card been ruled out as the cause of the problem? |
| 21:06.20 | mut | i didn't |
| 21:06.21 | mut | [16:57:47] <brookshire> mut: i can tell you one thing.. the pci card has nothing to do with it |
| 21:07.03 | mut | VoicePulse: i also used to have a te405 with a tellab echo can |
| 21:07.06 | mut | and it had the same problem |
| 21:07.14 | sevard | Disgrntld: Sounds like you're trying to call yourself. |
| 21:07.15 | mut | but that setup sucked period, |
| 21:07.26 | Disgrntld | sevard: i know its odd |
| 21:07.35 | Disgrntld | sevard: i swear i am not though =) |
| 21:07.42 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 21:07.48 | Druken | Disgrntld: check your phone is in the correct context? |
| 21:08.03 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 21:08.11 | Disgrntld | druken: ok |
| 21:08.22 | VoicePulse | mut: So is echo cancellation on or off right now? |
| 21:08.36 | mut | it's on |
| 21:08.59 | Druken | echo can should be off for faxing |
| 21:09.02 | VoicePulse | mut: Try turning it off |
| 21:09.03 | mut | yea |
| 21:09.10 | mut | zaptel should disable it no? |
| 21:09.13 | mut | on fax tone |
| 21:09.18 | *** join/#asterisk jsaunders (n=root@216.86.121.58) |
| 21:09.24 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 21:09.24 | brookshire | heh.. |
| 21:09.27 | VoicePulse | mut: Who knows... that may be the problem. |
| 21:09.42 | mut | well i can't turn it off, thats the whole reason for buying the card |
| 21:09.55 | MRH2 | hi anyway to continue monitoring a call in a single file post call transfer (SIP) |
| 21:09.57 | jsaunders | need urgent help: Can anyone tell me how to get around SIP "407 Proxy Authentication Required" problem w/ inbound calls w/ asterisk? |
| 21:10.41 | Druken | try authenticating to the poxy ? |
| 21:10.44 | *** part/#asterisk TripleFFFFFFFFFF (n=TripleFF@147-102.mc.cite.net) |
| 21:10.46 | justinu|laptop | try setting insecure=very in your peer entry |
| 21:10.51 | mut | i dunno |
| 21:10.53 | jsaunders | Testing.... |
| 21:10.59 | VoicePulse | mut: You can also try messing with the volume/gain settings, especially output gain. But echo cancellation has to be off. |
| 21:11.44 | jsaunders | insecure=very does not help :( |
| 21:11.57 | jsaunders | Could it have anything to do w/ the fact that pbx is behind router and set to dmz? |
| 21:12.02 | mut | i've been trying to mess with gains |
| 21:12.08 | jsaunders | ie, switching address. I am thinking of this because it says "proxy" |
| 21:12.12 | mut | but * needs restarted after i do that doesn't it? |
| 21:12.18 | Druken | jsaunders: did you restart asterisk ? |
| 21:12.26 | jsaunders | Druken: I did a "reload". |
| 21:12.39 | Druken | try a restart, just incase :) |
| 21:12.46 | jsaunders | Druken: I wish. Live system. |
| 21:12.57 | Druken | restart when convient |
| 21:13.01 | jsaunders | A reload should suffice in this instance, only context changes. |
| 21:13.18 | jsaunders | ugh |
| 21:13.27 | jsaunders | I am completely baffled by this one. |
| 21:13.28 | mercestes | If you changed addy's your NAT/ARP translations are screwed. |
| 21:13.37 | VoicePulse | mut: Yes, and the drivers might have to be unloaded/reloaded too. |
| 21:13.47 | mercestes | restart fixes it. |
| 21:14.00 | mut | i'm going to have to take a fax machine to my house or something |
| 21:14.05 | mut | test it at 5am |
| 21:14.06 | jsaunders | The call's making it to the pbx properly as it is set as dmz in router. All ports makes it fine. |
| 21:14.09 | justinu|laptop | your asterisk system is behind a NAT? |
| 21:14.13 | *** part/#asterisk BadPacket (n=root@unaffiliated/badpacket) |
| 21:14.25 | jsaunders | The SIP INVITE is hitting pbx just fine, pbx is returning first a SIP 407 followed by a SIP 403 to provider. |
| 21:14.26 | justinu|laptop | does asterisk use a non-routable ip? |
| 21:14.28 | mut | cuz i can dialup thru my voip in the office |
| 21:14.33 | mut | 31.2k connections every time |
| 21:14.42 | mut | last all day lon |
| 21:15.25 | *** join/#asterisk darylp (n=daryl_ju@63-208-162-59.digitalrealm.net) |
| 21:17.19 | jsaunders | justinu: Yes, behind router (as dmz). I know. :( Temporary. Outbound works great. First attempt at receive sip inbound, failing miserably. |
| 21:19.38 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 21:20.21 | justinu|laptop | jsaunders: did you set "externip" and "localnet" in sip.conf? |
| 21:20.50 | Druken | i don't think that is his problem.... |
| 21:21.09 | justinu|laptop | probably not, since it works one way |
| 21:21.12 | Druken | it's an auth issue, which is usually insecure=very, but he already did that |
| 21:21.29 | justinu|laptop | then his sip peer entry must not be "just right" |
| 21:21.33 | jsaunders | yessir, insecure=very |
| 21:21.51 | Druken | jsaunders: do you register to the proxy ? |
| 21:21.58 | jsaunders | For outbound, yes. |
| 21:22.11 | jsaunders | Err, I register for outbound to provider. |
| 21:22.11 | Druken | what about inbound? |
| 21:22.20 | jsaunders | Inbound, the provider does not register w/ us, no. |
| 21:22.29 | jsaunders | Nor should he have to, as far as I know. |
| 21:22.42 | justinu|laptop | you have to register for inbound calls usually |
| 21:22.46 | jsaunders | context=from-trunk |
| 21:22.48 | Druken | generally.. hehe |
| 21:22.50 | poisoner | anyone here who has some experinces with chan_sccp ? |
| 21:22.51 | jsaunders | freepbx (amportal) |
| 21:22.55 | justinu|laptop | that's how the ITSP knows whre to find you |
| 21:23.08 | Druken | ~amp |
| 21:23.09 | jbot | amp is probably "amp is, like, NOT supported here! people using it should join #freepbx (the new name of amp)" |
| 21:23.28 | jsaunders | Druken: Heheh. amp is just a shiney wrap around for *. |
| 21:23.31 | jsaunders | Same config files. |
| 21:24.04 | Druken | yeah, amp is the condom for asterisk, i know, but i still like trojan |
| 21:24.20 | jsaunders | Heh |
| 21:24.32 | Hmmhesays | they fit better then durex |
| 21:24.39 | Druken | #freepbx supports it |
| 21:25.15 | Druken | Hmmhesays: wouldn't know... haven't had to use one in 4 years... |
| 21:25.28 | generalhan | is there a way to have one of my queues directed to a different context in extensions.conf? i want to specify some rules for the members that are being dialed in this queue, and i cant do it in queues.conf |
| 21:25.30 | Hmmhesays | you shouldn't be barebacking |
| 21:25.47 | Hmmhesays | send them into a different queue? |
| 21:25.55 | Druken | Hmmhesays: when she's living with me, and fixed... there was no need :) |
| 21:26.21 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 21:26.25 | generalhan | Hmmhesays: well ... what i mean is ... i want to use ChanIsAvail before ringing the phones in the queue, and i cant do that once im in there, so i need to find a new way |
| 21:26.41 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 21:26.47 | Hmmhesays | why do you want to do such a thing? |
| 21:26.53 | sevard | Hmmhesays: do you know jack about custom apps in musiconhold.conf |
| 21:27.01 | generalhan | Hmmhesays: lol, [TK]D-Fender was asking me the same thing ! |
| 21:27.14 | Hmmhesays | define custom apps |
| 21:27.20 | sevard | mplayer => custom:/usr/local/bin/mplayer -dumpfile - -dumpaudio - http://64.236.34.97:80/stream/1005 |
| 21:27.27 | sevard | that's what i'm attempting |
| 21:27.49 | generalhan | cause i cant turn the call waiting off on the aastra phones that i have; and my reps are bitching at me about the beeping in their ear. so im trying to get around that by using the chanisavail cmd |
| 21:27.53 | Druken | streaming audio for moh is a bad idea.... |
| 21:28.02 | sevard | Druken: blah blah blah |
| 21:28.29 | Druken | generalhan: so limit the phones to 1 incoming channels |
| 21:30.43 | generalhan | Druken: i WAS using incominglimit=1 ... but its depriciated |
| 21:30.44 | [TK]D-Fender | generalhan : Start setting you agents as loca/1234@context, and set up a macro accordingly. |
| 21:30.44 | Druken | as far as i'm concerned, office phones shouldn't be able to receive call waiting.. that's what voicemail is for |
| 21:30.44 | dlynes | Is there any way to tell whether asterisk dropped a call, or if the pri dropped the call? |
| 21:30.44 | generalhan | but i dont even use agents |
| 21:30.46 | Druken | when was it depricated? |
| 21:30.52 | generalhan | 1.2.6 i think |
| 21:31.03 | justinu|laptop | dlynes: without looking at the protocol traces, i don't think so |
| 21:31.20 | dlynes | justinu|laptop: ah...so no post-mortem method then, eh? |
| 21:31.29 | [TK]D-Fender | generalhan : You do now if you want to use dial-plan logic to limit your calls... |
| 21:31.42 | justinu|laptop | dlynes: unfrotunately, not with the stock CDRs |
| 21:32.04 | generalhan | [TK]D-Fender: great ... does that also mean that i need to have my users "login" and "logout" everyday ? |
| 21:32.22 | dlynes | justinu|laptop: no way to tell by looking at the log, either? |
| 21:32.29 | justinu|laptop | dlynes: full log? |
| 21:32.35 | [TK]D-Fender | generalhan : Not necessarily. Try using Local to define them statically, not as Agent/ |
| 21:32.45 | dlynes | justinu|laptop: mostly full log, yeah |
| 21:32.58 | justinu|laptop | dlynes: iirc, no... it won't log that info |
| 21:32.59 | dlynes | justinu|laptop: I don't have all the debug crap turned on, but other than htat |
| 21:33.08 | justinu|laptop | yeah, you need the pri debug or sip debug |
| 21:33.12 | dlynes | ah |
| 21:33.14 | dlynes | suckage |
| 21:33.21 | justinu|laptop | can you code? it would be pretty easy to put that in, i think |
| 21:33.52 | dlynes | justinu|laptop: Yeah, I can code |
| 21:34.04 | justinu|laptop | actually, you might be able to tell if asterisk hit a "hangup" in the dialplan, but that wouldn't tell you who released if a call was bridged with dial |
| 21:34.10 | dlynes | I wouldn't have a clue about where to start, though |
| 21:34.24 | justinu|laptop | chan_zap.so for the pri side |
| 21:35.18 | dlynes | Ok, so what would I be looking for to signal a drop on the telco's part? |
| 21:35.39 | justinu|laptop | you'd receive a q931 DISCONNECT from telco |
| 21:35.55 | justinu|laptop | if you drop the call, you SEND a DISCONNECT to telco |
| 21:36.09 | dlynes | ah...and that's an error message I'd be able to detect easily? i.e. asterisk has a predefine for that? |
| 21:36.24 | justinu|laptop | lemme take a quick look |
| 21:36.32 | dlynes | I've only glanced at the code...never actually gone in depth on it |
| 21:37.19 | *** join/#asterisk GolobTGG (n=GolobTGG@BSN-77-78-87.dsl.siol.net) |
| 21:41.16 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-152-241-16.red.bezeqint.net) |
| 21:42.14 | justinu|laptop | christ it's a mess |
| 21:42.21 | jsaunders | So does the SIP 407 being sent back mean Asterisk is challenging the INVITE? I found somone saying a permit= should solve this but alas, it did not help. |
| 21:42.41 | vader-- | is there any gui type switchboard software that works with asterisk? |
| 21:42.41 | justinu|laptop | check like 8812 in chan_zap.so |
| 21:42.41 | dlynes | justinu|laptop: exactly why i haven't spent much time looking at it :) |
| 21:42.56 | justinu|laptop | that's where it reads the PRI events off the dchannel |
| 21:43.03 | vader-- | im looking for something our secretaries can use to transfer calls and see what lines are in use and all that jazz |
| 21:43.13 | vader-- | also to see if someone is on a phone call? |
| 21:43.23 | *** join/#asterisk linlin (n=linlin@c-67-184-230-198.hsd1.il.comcast.net) |
| 21:43.24 | brodiem | vader--, try FOP but I personally don't like it |
| 21:43.27 | Hmmhesays | fop can do it, kind of clunky but it works |
| 21:43.28 | vader-- | fop? |
| 21:43.32 | Hmmhesays | flash operator panel |
| 21:43.36 | justinu|laptop | jsaunders: i think asterisk isn't looking at the correct peer entry for inbound calls |
| 21:43.37 | brodiem | vader--, asternic.org |
| 21:43.56 | vader-- | how does it tie in? |
| 21:44.07 | jsaunders | Thanks justinu, I'll look at that closer. |
| 21:44.10 | Hmmhesays | look at the demo on asternic.org |
| 21:44.12 | brodiem | vader--, it uses the manager API |
| 21:44.29 | justinu|laptop | jsaunders: turn on sip debug, capture an inbound call and pastebin it if you want |
| 21:44.49 | justinu|laptop | jsaunders: strike that |
| 21:44.56 | dlynes | justinu|laptop: ah...looks like if it got hung up, it'll say "got hangup"; if asterisk hangs it up, it'll say "got hangup request" |
| 21:45.05 | justinu|laptop | jsaunders: turn on sip debug /and/ full logging, paste that full log |
| 21:45.06 | vader-- | so you guys don't like fop? |
| 21:45.19 | brodiem | vader-- I find it too buggy |
| 21:45.24 | justinu|laptop | dlynes: cool, is that in your log? |
| 21:45.31 | vader-- | he just released a new version the site said |
| 21:45.35 | vader-- | have you tried that? |
| 21:46.02 | brodiem | vader-- notice it's dated 3/13.. but yeah I've used .25 |
| 21:46.13 | vader-- | gotcha |
| 21:46.29 | vader-- | ya seems like it would be kinda hard to use for a larger organization |
| 21:48.48 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 21:49.31 | *** part/#asterisk Ixthod (n=Ixthod@intellop.static.iaxs.net) |
| 21:49.34 | brodiem | vader-- it has some cool features but to me it just seems kind of sloppy |
| 21:50.32 | brodiem | vader-- I definitely wouldn't rely on it for a business to redirect calls though |
| 21:50.44 | *** join/#asterisk Tornad (n=Tornad@LAubervilliers-151-12-105-110.w193-252.abo.wanadoo.fr) |
| 21:51.36 | *** join/#asterisk iCEBrkr (n=icebrkr@6532244hfc169.tampabay.res.rr.com) |
| 21:53.01 | Druken | brodiem: it's nice to watch your extensions tho... i find it works good for that |
| 21:53.28 | brodiem | Druken yeah I agree, but it seems to sometimes show ghost calls |
| 21:53.47 | mercestes | or die for no known reason and require a restart |
| 21:54.00 | brodiem | and when it does you can't disconnect the imaginary call from fop |
| 21:54.02 | Druken | can't say i've seen any of those... but i have seen it not see a call terminate... |
| 21:54.20 | brodiem | then you need to restart it, and it loses all of its idle times, DND states, etc |
| 21:54.21 | dlynes | justinu|laptop: It's in chan_zap.c, line 8847, 8912 |
| 21:54.29 | mercestes | that would be a ghsot call, Druken. |
| 21:54.38 | jsaunders | justinu: will do, uno momento |
| 21:55.12 | Druken | mercestes: yeah i guess.... |
| 21:56.00 | brodiem | i wish there was something better without having to buy an entire pbx from a vendor who developed their own |
| 21:56.15 | Hmmhesays | better? |
| 21:56.17 | Hmmhesays | than what? |
| 21:56.22 | brodiem | than fop |
| 21:56.30 | Hmmhesays | write something op |
| 21:56.32 | Hmmhesays | *up |
| 21:56.49 | brodiem | it's about the only option.. |
| 21:56.51 | Hmmhesays | hell you could do something in *.net easily |
| 21:57.00 | justinu|laptop | dlynes: so there ya go... |
| 21:57.05 | Hmmhesays | you can mash your face on the keyboard and come out with a working program with visual studio |
| 21:57.13 | brodiem | lol |
| 21:57.38 | Druken | printf "hello world"; |
| 21:57.47 | brodiem | i don't use any win machines and don't ever plan to:) |
| 21:57.56 | Hmmhesays | yeah but I bet your secretary does |
| 21:57.59 | dlynes | justinu|laptop: would you happen to know if you can determine if the remote end hung up, or if the telco's switching eq screwed up, though? |
| 21:58.02 | brodiem | yup |
| 21:58.13 | Hmmhesays | brodiem: exactly |
| 21:58.25 | Hmmhesays | wasn't there something awhile back some dude wrote in vb |
| 21:58.36 | brodiem | ip switchboard?> |
| 21:58.59 | dlynes | justinu|laptop: sorry to be a pest...I know pretty much zero about pri signalling |
| 21:59.29 | justinu|laptop | dlynes: the best you can do is look for an abnormal cause code |
| 21:59.36 | justinu|laptop | dlynes: meaning something other than 16 |
| 22:00.07 | Druken | dlynes: seems like your looking for a way to detect if the telco's equipment screws up, do you think they will give you a break on the cost if a screw up happens once in a while ?? hehe |
| 22:00.48 | Hmmhesays | brodiem yeah that was it |
| 22:01.33 | brodiem | Hmmhesays, seems to run better but with the features you'd may as well just run astman |
| 22:02.59 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 22:02.59 | *** mode/#asterisk [+o denon] by ChanServ |
| 22:03.44 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 22:07.00 | distortion | is there a good way to hunt down the cause of "*** glibc detected ***" while using the safe startup script? |
| 22:07.51 | dlynes | Druken: No, but I would like to determine if the cause is the remote end hanging up, or Group Telecom screwing up |
| 22:08.09 | dlynes | Druken: if it's Group Telecom, I can threaten to move my business to Telus |
| 22:08.25 | dlynes | Druken: Or Allstream, for that matter |
| 22:10.56 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 22:11.17 | dlynes | distortion: Are you using any of the gtk or kde modules for asterisk? |
| 22:11.43 | dlynes | distortion: i'm guessing you're using the gtk module |
| 22:12.06 | distortion | gtk? no i dont htink so, my modules.conf is failly stripped down |
| 22:16.29 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 22:19.18 | *** join/#asterisk wunderkin (i=kev@69.26.192.234) |
| 22:19.45 | distortion | hmm, lcdial module seems like its the culprit guess it cant handle it when you're sending ~8-10 calls a second. |
| 22:20.26 | websae | distortion: do you know anything about predicitve dialers for asterisk? |
| 22:20.55 | distortion | no |
| 22:21.42 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 22:21.46 | dlynes | distortion: reason i was asking is because gtk applications will typically output that to the x console when they start up |
| 22:22.35 | dlynes | distortion: it's not an error; it's just merely a notice to let you know it was able to find your glib library (general utility routines) |
| 22:22.40 | *** join/#asterisk CpuID (n=none@gentoo/contributor/cpuid) |
| 22:22.41 | distortion | that was the error im getting when i put asterisk under a heavy load while using lcdial. |
| 22:22.52 | distortion | well, the asterisk app dies when i see that error |
| 22:23.06 | distortion | magically stops with no .core file |
| 22:23.27 | distortion | and hundreds of calls get dropped :( |
| 22:24.10 | dlynes | distortion: maybe libmysqlclient.so uses glib |
| 22:24.16 | *** join/#asterisk g__ (n=g@itd01fw-fibre.itdepartment.com) |
| 22:24.17 | dlynes | distortion: lcdial uses that shared object |
| 22:24.51 | g__ | Question: is it safe to use '+' as a dialplan extension? |
| 22:26.06 | g__ | ie, exten => +16135551234,1,Dial(Zap/g1/${EXTEN:2}) |
| 22:26.11 | tainted- | distortion dump lcdial and get a real lcr |
| 22:26.19 | tainted- | jk |
| 22:27.13 | CunningPike | g__: Try it and see |
| 22:27.25 | dlynes | btw, does anyone know how to dial a couple of digits, do a pause, and then dial some more digits on a sip channel? |
| 22:27.26 | g__ | CunningPike: it appears to work. |
| 22:27.33 | CunningPike | There you are, then |
| 22:27.36 | CunningPike | :) |
| 22:27.42 | tainted- | dlynes live channel? |
| 22:28.11 | dlynes | tainted-: Say like Dial(Zap/g1/*70w6041234567), but on a sip channel |
| 22:28.29 | tainted- | yea |
| 22:28.40 | g__ | CunningPike:I was kinda hoping someone would say "oh, we do it all the time and it's fine" or "don't go there! There are bugs all over the place! Beware! Spoooky!" |
| 22:28.42 | tainted- | it's i think the D() option in dial |
| 22:28.54 | dlynes | tainted-: that only works after the call's been answered, though |
| 22:29.02 | tainted- | oh u need before? |
| 22:29.05 | dlynes | tainted-: i need this to happen after i get a second dial tone |
| 22:29.19 | CunningPike | g__: I understand - but, if you were to wait for that all the time, you'd never get anything done ;) |
| 22:29.20 | g__ | Is that a sip provider you have? |
| 22:29.32 | generalhan | [TK]D-Fender: you still here ? |
| 22:29.33 | tainted- | well technically the channel is answered at that point |
| 22:29.38 | fourcheez-away | dlynes It must be answered to get a dial tone |
| 22:29.43 | g__ | CunningPike: that's a good point. So.. anyone know the answer? :) |
| 22:29.45 | dlynes | fourcheez-away: oh |
| 22:30.00 | dlynes | fourcheez-away: ok, I figured someone talking on the other end was considered to be an answer |
| 22:30.15 | tainted- | no |
| 22:30.18 | fourcheez-away | no, ther'es a very presise definition of answered in SIP |
| 22:30.21 | CunningPike | g__: afaik, '+' isn't any kind of reserved character for an extension name, so I think you're good |
| 22:30.35 | g__ | That's not always the case.. but if you don't have to worry about that consider yourself lucky, dlynes. |
| 22:30.36 | distortion | tainted: real lcr?? ;) |
| 22:30.58 | dlynes | ah |
| 22:31.08 | fourcheez-away | dlynes, think about it - there's no real ringing going on is there? |
| 22:31.10 | tainted- | distortion yea .. something mysql driven |
| 22:31.17 | distortion | lcdial is mysql |
| 22:31.21 | tainted- | oh |
| 22:31.25 | distortion | yup. |
| 22:31.33 | dlynes | fourcheez-away: nope...it's just a voltage issue, and a q931 response sent back |
| 22:31.42 | g__ | CunningPike: I suppose to test this theory, I should try it out and run a large test case that somehow involes a multi-play network game.. it would certainly waste enough time, right? |
| 22:31.51 | CunningPike | lol - sure would |
| 22:31.55 | CunningPike | I like your style |
| 22:32.00 | tainted- | dlynes try the D(435454545wwwwfafe) which each 'w' representing a wait 1 second |
| 22:32.05 | tainted- | u'll like it |
| 22:32.14 | g__ | system-administration-by-doom |
| 22:32.24 | fourcheez-away | dlynes, no in sip there isn't even a voltage |
| 22:32.35 | g__ | unless you lick the wires |
| 22:32.40 | fourcheez-away | only the fact that the client has said that it's answered |
| 22:32.42 | tainted- | nope |
| 22:32.45 | tainted- | not even if u lick wires |
| 22:33.04 | g__ | I'm sure it could happen.. |
| 22:33.10 | g__ | you can't be too careful. |
| 22:33.14 | tainted- | nope |
| 22:33.15 | tainted- | it can't |
| 22:33.35 | anthm | i think w is 1/2 second |
| 22:34.27 | tainted- | really? |
| 22:34.33 | tainted- | i wish it was documented someplace |
| 22:34.41 | tainted- | voip-info's got nothing |
| 22:34.48 | anthm | yah i only know cos i made that function |
| 22:35.11 | tainted- | the 'send dtmf' in live sip channel? |
| 22:35.14 | tainted- | or dial() |
| 22:35.44 | anthm | it's in app.c as a public func called ast_dtmf_stream |
| 22:35.56 | tainted- | wow |
| 22:36.03 | anthm | then used in mods like dial and senddigits |
| 22:36.05 | tainted- | you've really contributed a lot |
| 22:36.12 | g__ | Someone have time to add it to voip-info? |
| 22:37.44 | g__ | Nevermind, I'll add it. |
| 22:39.28 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 22:40.07 | g__ | done. |
| 22:41.53 | *** part/#asterisk CpuID (n=none@gentoo/contributor/cpuid) |
| 22:44.04 | *** part/#asterisk jake1932 (n=Administ@68.236.22.143) |
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| 22:53.05 | dlynes | fourcheez-away: yep |
| 22:53.35 | dlynes | on another note, I'm having the following problem when trying to load wcfxo or ztdummy: |
| 22:53.38 | dlynes | ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 22:53.38 | dlynes | FATAL: Error running install command for ztdummy |
| 22:53.40 | RoyK | <PROTECTED> |
| 22:54.05 | dlynes | I've got a cheapo x100p card in the machine |
| 22:54.39 | dlynes | And the same error when I try to modprobe wcfxo: |
| 22:54.39 | Qwell[] | So why are you using ztdummy? |
| 22:54.47 | dlynes | ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 22:54.47 | dlynes | FATAL: Error running install command for wcfxo |
| 22:54.51 | Dr-Linux | hey dlynes |
| 22:54.53 | dlynes | Because wcfxo isn't working, either |
| 22:54.55 | Qwell[] | check dmesg |
| 22:55.49 | dlynes | waiting for pastebin to become available |
| 22:57.42 | dlynes | http://pastebin.com/693270 |
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| 22:59.43 | dlynes | It's got some definite errors there, but I have no idea what the real error is |
| 23:01.26 | dlynes | Could it just be having an issue because it's sharing an irq? |
| 23:01.39 | Manipura | I have a IAX phone and a DID.. But no asterisk server anymore.. Are there any services out there that allow you to point a DID to them and route it to your phone? |
| 23:01.49 | tainted- | probably |
| 23:01.54 | tainted- | dlynes what kind of card |
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| 23:01.57 | tainted- | digium? |
| 23:01.58 | dlynes | tainted-: x100p.com |
| 23:02.05 | tainted- | gag |
| 23:02.27 | dlynes | tainted-: well, seeing as how digium doesn't make the x100p card anymore, i didn't have a choice, but to buy a clone |
| 23:02.55 | tainted- | it's basically an analog modem |
| 23:03.00 | dlynes | yep |
| 23:03.05 | dlynes | and it's a pci device |
| 23:03.12 | tainted- | i had great luck with a clone |
| 23:03.13 | dlynes | which, technically should be capable of sharing interrupts |
| 23:03.16 | tainted- | forget which one |
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| 23:04.21 | generalhan | Can some one please take a look at my recording setup and advise me on a way to get this to work properly ?? http://generalhan.pastebin.ca/52716 |
| 23:04.57 | generalhan | the calls are not recorded i think becuase * is fighting over where to save the recording ... so i need to figure out a bitter way to do this to make it work correctly |
| 23:06.04 | [TK]D-Fender | generalhan : I don't think you can ask * to make a full pasth like that on demand. |
| 23:06.21 | [TK]D-Fender | generalhan : May all your time/date vars part of the FILENAME, not the PATH |
| 23:06.47 | [TK]D-Fender | generalhan : Its bad enough it hopes to have a folder ready by SIP account |
| 23:06.55 | generalhan | [TK]D-Fender: well i do it on an idividual basis cause thats the same macro i use for regular calls. i just need to know how to make it work for multiple calls |
| 23:08.02 | generalhan | [TK]D-Fender: i got the queue to dial out to the people and do the ChanIsAvail first ... so thats taken care of .. if i can get it to record correctly (or record AT ALL i guess) then ill be almost done |
| 23:08.03 | [TK]D-Fender | generalhan : Follow my ealier advice about removing all other vars from the path than the EXT |
| 23:08.15 | generalhan | [TK]D-Fender: ok ill try that |
| 23:10.54 | generalhan | [TK]D-Fender: still no dice ... on the CLI i see it come up and say that its starting the recording for all the extensions that it is dialing... and i think that is where my problem lies. its like * is fighting over which directory to store this in and it just gives up |
| 23:13.16 | [TK]D-Fender | generalhan :PAstbin your new macro |
| 23:13.27 | [TK]D-Fender | And VERIFY your path for your last attempt. |
| 23:14.08 | generalhan | [TK]D-Fender: i tried it this way too |
| 23:14.09 | generalhan | http://generalhan.pastebin.ca/52717 |
| 23:14.24 | generalhan | to specify only one place to record it to ... still no go on that either |
| 23:15.19 | *** part/#asterisk philippel (n=p_lindhe@c-24-19-186-72.hsd1.wa.comcast.net) |
| 23:15.30 | [TK]D-Fender | generalhan : Stop using the date for the folder! |
| 23:15.37 | generalhan | [TK]D-Fender: i cant |
| 23:15.48 | [TK]D-Fender | yes you can, and you WILL. |
| 23:15.59 | [TK]D-Fender | Are you going to premake EVERY FUTURE VALUE right now?! |
| 23:16.31 | generalhan | [TK]D-Fender: i work in a Law Office and it is the attorney's policy to record all calls to be EASILY referenced .... so they need to be in folders by user and by date |
| 23:17.10 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 23:17.30 | [TK]D-Fender | generalhan : Parse it out late. use a script, whatever, but nothin in there is guranteein your path! |
| 23:18.06 | generalhan | [TK]D-Fender: http://generalhan.pastebin.ca/52718 that is how i have been doing it for regular calls and all incoming calls have been recorded just fine ( IF they are called directly from the autoattendant or transferred from inside the office) |
| 23:18.38 | generalhan | i also have one for outgoing calls as well ... all that works perfectly ... until i try to use it the same way to record out of the queues |
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| 23:19.52 | [TK]D-Fender | I dunno... |
| 23:20.17 | generalhan | [TK]D-Fender: when i specify the date as the folder, as soon as the call starts it automagically creates that folder for me to put the call into |
| 23:21.03 | generalhan | that way when my boss says that an attorney says we DID NOT call him on this date .. i can go right to the date pull up his phone number and play back the conversation for him on the spot |
| 23:21.08 | [TK]D-Fender | generalhan : Never seena system that would invent a whole path jsut to support a new file. |
| 23:21.15 | [TK]D-Fender | generalhan : But hey, whatever |
| 23:21.19 | tainted- | does anyone have the SNL clip of the IVR lady |
| 23:21.20 | generalhan | [TK]D-Fender: really ? |
| 23:21.32 | websae | tainted- that was hilarious from SNL haha |
| 23:21.35 | [TK]D-Fender | generalhan : Just try to seperate HOW the system works, from the fact of it working. |
| 23:21.47 | generalhan | [TK]D-Fender: what do you mean ? |
| 23:21.55 | [TK]D-Fender | generalhan : You could put it all in the name for all it matters as log as its standardized |
| 23:22.32 | generalhan | well if im using a macro for all incoming and outgoing calls i would say thats standardized |
| 23:22.46 | generalhan | nothing gets to an extension without passing though those rules |
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| 23:23.11 | generalhan | its just that when the queue is trying to save 1 call to 15 different places it doesnt seem to want to work. |
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| 23:54.40 | philippel | Corydon: are you there? |
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| 23:56.17 | MRH2 | hi just upgraded to latest svn1.2 stable and it ain't compiling?? |
| 23:58.03 | MRH2 | any known issues atm? |
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