| 00:00.14 | Juxt | newsole: what kind of volume are you ready to handle? |
| 00:00.34 | NewSole | well we have 3 servers on 100MB backbones |
| 00:00.44 | Romik | speex, ILBC ? |
| 00:00.45 | Juxt | that doesn't say anything |
| 00:01.01 | Juxt | unless you're doing transcoding or something |
| 00:01.07 | NewSole | all three have Dual Xeon CPUs and 4GB Ram... |
| 00:01.15 | Juxt | newsole: are you terminating the calls yourself? |
| 00:01.22 | cypromis | whatfor you need 4GB of ram on a * server ? |
| 00:01.51 | NewSole | yes we have about 15 PRIs hooked up now plus we have a large scale terminstaion |
| 00:01.53 | Romik | somebody can advice how i do check that timer working on asterisk? |
| 00:02.45 | topping | NewSole: wouldn't it have made more sense to get a fractional DS3? |
| 00:03.21 | NewSole | right now we have 5 Pri's at 3 Different locations |
| 00:03.28 | topping | ah ic |
| 00:03.38 | NewSole | but having one problem |
| 00:03.53 | NewSole | dialplans are no longer letting us link |
| 00:06.07 | NewSole | trying to ffigure out why... and wiki is down |
| 00:06.25 | Juxt | newsole: use google cache |
| 00:09.36 | NewSole | been trying |
| 00:20.00 | NewSole | hmmm... dead in here... WiKi goes Dead and Everyone goes Dead... |
| 00:20.16 | topping | i prefer phish |
| 00:21.40 | *** part/#asterisk Juxt (~Juxt@sfl-dsl-64-135-113-4-cust.host.net) |
| 00:22.14 | topping | you seem to have it all... GF, big honkin servers... making everyone jealous ;-) |
| 00:22.31 | shmaltz | wiki down? |
| 00:22.43 | NewSole | I said prefer.... dont mean I have one |
| 00:23.52 | topping | then i guess "seem" is an appropriate word |
| 00:23.52 | NewSole | last one left me cause I spent too much time on computer |
| 00:24.06 | topping | if you were into teledildonics, she might have liked it that way |
| 00:24.22 | NewSole | she thought i was more inyo cyber then sex..... my poor floppy |
| 00:24.45 | NewSole | lol |
| 00:25.21 | NewSole | topping... you any good at dial plans |
| 00:25.24 | topping | it's all about changing adversity into opportunity |
| 00:25.32 | topping | hehe, no, sorry |
| 00:25.52 | NewSole | hmmm... I am lost on this one |
| 00:26.08 | topping | if you describe it, eventually someone might have an idea tho |
| 00:27.34 | dec | what's wrong, NewSole ? |
| 00:29.09 | *** join/#asterisk fr3dd (~fredd@cpe-024-168-231-049.sc.res.rr.com) |
| 00:29.15 | shmaltz | anybody here having trouble accessing the wiki? |
| 00:29.26 | NewSole | I am dialing my server |
| 00:29.59 | NewSole | using IAX2/master@master/1NXXNXXXXXX@context |
| 00:30.12 | NewSole | and getting No Auth Found |
| 00:31.50 | dec | only thing that comes to mind is that the master user is set up right |
| 00:32.01 | shmaltz | anybody here having trouble accessing the wiki? |
| 00:32.04 | dec | only thing that comes to mind is to check that the master user is set up right |
| 00:32.08 | dec | shmaltz: yes, its down |
| 00:32.14 | shmaltz | thanks dec |
| 00:32.17 | dec | apparently. |
| 00:32.33 | shmaltz | anybody have a cached list or link to the extended sound files added to cvs? |
| 00:33.01 | NewSole | it is thats the problem |
| 00:33.58 | shmaltz | thanks guys such a file exists in /usr/src/asterisk-sounds/sounds-extra.txt if you did a cvs co asterisk-sounds |
| 00:36.20 | *** join/#asterisk astoria (~asto@68.77.110.194) |
| 00:37.26 | astoria | I have a question, perhaps out of the realm of asterisk.. how are SMS short-codes (ex. 45645) handled? Is there some kind of carrier-based routing between that number and a GSM number? Thanks in advance! |
| 00:43.07 | sudoer | anyone here use broadvoice? |
| 00:44.45 | NewSole | this is anoying.... |
| 00:45.51 | sudoer | voip-info being down is annoying |
| 00:46.03 | *** join/#asterisk hermie (~nick@24.236.167.53.bay.mi.chartermi.net) |
| 00:46.27 | want561or772did | need did pls thank you |
| 00:48.52 | hermie | want561or772did: you might try in #asterisk-biz or on the -biz mailing list |
| 00:50.54 | file[laptop] | sex? |
| 00:55.40 | want561or772did | ok |
| 01:02.29 | NewSole | Any one here good at Dialplans |
| 01:04.51 | file[laptop] | lots of people are |
| 01:06.21 | cypromis | nah |
| 01:06.23 | cypromis | nobody |
| 01:06.25 | cypromis | :P |
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| 01:07.49 | tzanger | heh |
| 01:07.50 | tzanger | http://pbx.mine.nu/artwork/036-lolwhat-linux-sarojin.gif |
| 01:11.45 | Juggie | anyone know if theres an asterisk thing going on this week in toronto during von? a meetup or anything. |
| 01:12.36 | NewSole | file can you help me figure this out |
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| 01:35.29 | NewSole | file[laptop]... I am still here |
| 01:35.29 | file[laptop] | one less person I have to say, "no I won't help you become the next vonage", to is great |
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| 01:35.31 | jakepdev | is that a common request? |
| 01:35.31 | file[laptop] | oh no, people are back |
| 01:35.31 | file[laptop] | it can get worse |
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| 01:35.31 | file[laptop] | I block most stuff out of my memory though |
| 01:35.31 | jakepdev | understood |
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| 01:35.31 | TUplink | Apr 17 21:35:08 WARNING[1540]: chan_zap.c:763 zt_open: Unable to open '/dev/zap/pseudo': Device not configured |
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| 01:35.32 | TUplink | i insytalled zaptel |
| 01:35.32 | jakepdev | maybe lilo will fill us in as to what's going on |
| 01:35.55 | TUplink | and /dev/zap is ther |
| 01:36.27 | jakepdev | sounds like ztdummy is not configured |
| 01:36.27 | jakepdev | but i'm not sure |
| 01:37.05 | TUplink | how do i do that? |
| 01:37.05 | TUplink | the WIKI is stilldown |
| 01:37.34 | jakepdev | i'm just guessing by the error message - don't take that as an official diagnosis :) |
| 01:38.30 | TUplink | how do i config ztdummy |
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| 01:39.23 | timecop | == No one is available to answer at this time (1:0/0/0) |
| 01:39.26 | timecop | ^^ from h323 |
| 01:39.33 | timecop | but the otehr end is fine |
| 01:39.41 | ManxPower | Does ztdummy even run on *BSD? |
| 01:39.51 | timecop | just stop being cheapfucks and buy digium hardware. |
| 01:39.57 | timecop | from digium, not from fucking scambay. |
| 01:40.09 | ManxPower | timecop: Or at least use a supported platform. |
| 01:40.10 | timecop | then you wont need ztdummy. |
| 01:40.13 | timecop | that too. |
| 01:40.23 | jakepdev | it says pseudo though - that doesn't sound like hardware |
| 01:41.00 | timecop | fucking h323 |
| 01:41.05 | *** join/#asterisk tessier (~treed@222.253.73.184) |
| 01:41.05 | techie | haha |
| 01:42.25 | jakepdev | TUplink - http://64.233.179.104/search?q=cache:23fcv2Hcj0UJ:www.voip-info.org/wiki-Asterisk%2Btimer%2Bztdummy+ztdummy+wiki&hl=en |
| 01:43.29 | jakepdev | but as Manx says - are you running on a supported platfrom? |
| 01:44.35 | *** join/#asterisk JunK-Y (junk-b@Sherbrooke-HSE-ppp3606643.sympatico.ca) |
| 01:44.42 | jakepdev | and while were at it - i'm trying to figure out how to become the next Vonage |
| 01:45.01 | JunK-Y | y0 |
| 01:45.02 | TUplink | i got it |
| 01:45.18 | jakepdev | how? |
| 01:45.21 | TUplink | jake i got it |
| 01:45.40 | jakepdev | ok - what did you do to fix it? |
| 01:45.57 | TUplink | kldload /sbin/kldunload/ztdummy.ko |
| 01:46.07 | jakepdev | ok - great! |
| 01:46.33 | TUplink | by default its not in hte zaptel.sh |
| 01:46.44 | jakepdev | ok |
| 01:50.02 | *** join/#asterisk MikeJ[Laptop] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net) |
| 01:52.09 | ManxPower | ~docs |
| 01:52.12 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 01:52.12 | bugbot | docs is assigned nothing and reported nothing. |
| 01:52.13 | ManxPower | ~mailinglist |
| 01:52.14 | jbot | extra, extra, read all about it, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 01:52.14 | bugbot | mailinglist is assigned nothing and reported nothing. |
| 01:53.50 | NewSole | Any one here good at Dialplans... can help figure this one out.... |
| 01:54.02 | jakepdev | pastebin |
| 01:54.10 | NewSole | http://pastebin.ca/9728 |
| 01:54.19 | shmaltz | NewSole, lets try |
| 01:54.45 | shmaltz | whats happening with that? |
| 01:54.54 | |Vulture| | anyone here use Nagios + check_asterisk? |
| 01:54.54 | shmaltz | NewSole |
| 01:54.57 | NewSole | I get "No authority found" on masters |
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| 01:55.27 | hohum | <--- hohum |
| 01:55.44 | file[laptop] | <--- file |
| 01:55.58 | file[laptop] | today is advice day in #asterisk |
| 01:56.00 | hohum | ^--- homo |
| 01:56.03 | jakepdev | <--------- jake |
| 01:56.06 | file[laptop] | NewSole: my advice to you is to start simple |
| 01:56.07 | hohum | :) |
| 01:56.17 | jakepdev | amen to starting simple |
| 01:56.37 | jakepdev | and that doesn't mean only mean using *@home as i've found out |
| 01:56.50 | *** join/#asterisk scythelx (~wow@pc-24-151-108-034.newm2.ct.charter.com) |
| 01:56.50 | NewSole | lol |
| 01:56.52 | scythelx | hey all |
| 01:56.58 | *** join/#asterisk shodan (~shodan@216.113.99.185) |
| 01:56.59 | hohum | howdy |
| 01:57.03 | file[laptop] | NewSole: you've suverely overcomplicated things |
| 01:57.06 | |Vulture| | is there a way to use check_asterisk to see if a peer is registered? |
| 01:57.07 | NewSole | problem was it was working last week and I did no changes |
| 01:57.15 | file[laptop] | obviously something changed |
| 01:57.20 | file[laptop] | and why don't you just use disallow=all |
| 01:57.25 | file[laptop] | and then allow all you want afterwards? |
| 01:57.43 | scythelx | could someone help me, the wiki is down so i cant look up how to do this. look at http://pastebin.ca/9729 - im trynig to put music on hold ( which wokrs fine ) but like 10 -15 seconds into the music i want it to say a message not sure if im doing it right |
| 01:58.09 | file[laptop] | scythelx: nope won't work |
| 01:58.10 | jakepdev | nope |
| 01:58.12 | NewSole | cause our mobile PRI boxes done like dissallow=all |
| 01:58.16 | jakepdev | use WaitMusicOnHold |
| 01:58.21 | jakepdev | (10) |
| 01:58.30 | scythelx | ok cool |
| 01:58.35 | scythelx | thank you |
| 01:58.38 | jakepdev | np |
| 01:58.54 | file[laptop] | NewSole: all disallow does is clear the internal bitmask of all the stuff... same thing as disallowing each except it does it in one big swoop |
| 01:59.32 | file[laptop] | but anyway |
| 01:59.44 | file[laptop] | a user doesn't have a host, or username, or qualify |
| 02:00.16 | file[laptop] | and don't put the username in the iax.conf entry for the peer, weird stuff happens |
| 02:01.18 | file[laptop] | try specifying the contents you want the user to have access to in the iax.conf too and see what happens |
| 02:01.26 | file[laptop] | anyone else have anything to comment regarding his entries? |
| 02:02.16 | jakepdev | is "No authority found" a codec issue? sounds more like authentication... |
| 02:02.27 | file[laptop] | it's authentication |
| 02:02.33 | file[laptop] | the codec stuff just annoyed me... a lot :p |
| 02:02.36 | jakepdev | lol |
| 02:02.50 | NewSole | I get a code 50 |
| 02:03.15 | file[laptop] | a code 50, how descriptive |
| 02:03.21 | jakepdev | no!!! not the code 50 |
| 02:03.27 | jakepdev | run - fast! |
| 02:03.30 | file[laptop] | WE'RE ALL GONNA DIE! |
| 02:03.31 | file[laptop] | AHHHHHHHHHHH! |
| 02:03.40 | jakepdev | hehe |
| 02:04.53 | *** join/#asterisk iguy (~iguy@dsl093-197-234.mke1.dsl.speakeasy.net) |
| 02:05.02 | NewSole | this is what I get |
| 02:05.03 | NewSole | http://pastebin.ca/9731 |
| 02:06.01 | *** join/#asterisk ClayReiche123 (fwuser@acxexch1.accxx.com) |
| 02:06.25 | file[laptop] | and you've got a user on that box? |
| 02:06.40 | jakepdev | for outgoing - i was under the impression all you need is an entry in the dialplan |
| 02:06.53 | NewSole | ya the user was pasted to http://pastebin.ca/9728 |
| 02:06.55 | file[laptop] | jakepdev: better to have a peer entry, more control over things |
| 02:07.00 | jakepdev | ok |
| 02:07.42 | file[laptop] | NewSole: iax2 show users |
| 02:08.22 | *** join/#asterisk drumkilla (~russell@12.21.241.80) |
| 02:08.22 | *** mode/#asterisk [+o drumkilla] by ChanServ |
| 02:08.26 | file[laptop] | is masters in there? |
| 02:08.26 | ClayReiche123 | I've been using the stable version for a while now and I see a couple features in the HEAD that I would LOVE to have... is there a way for me to "add" these features to stable. The one thing I'm particularly fond of is the "sip NOTIFY" feature in HEAD... |
| 02:08.48 | file[laptop] | ClayReiche123: you can try to backport it, but you're on your own unless you find someone to help you :) |
| 02:08.55 | *** join/#asterisk david (~david@gw-djc.davidcoulson.net) |
| 02:09.20 | drumkilla | ClayReiche123: /me waves to file[laptop] |
| 02:09.34 | drumkilla | oops |
| 02:09.38 | drumkilla | ha |
| 02:09.41 | NewSole | http://pastebin.ca/9732 |
| 02:10.06 | ManxPower | drumkilla: Did you see kram's update to the Asterisk README about large jumps in time? |
| 02:10.19 | drumkilla | ManxPower: yeah |
| 02:10.45 | drumkilla | i've been doing a school project today - i'll get it when I get back on the bug tracker |
| 02:11.05 | file[laptop] | I thought there was actually a patch when I saw the bugnote, then I saw it was just for the readme |
| 02:11.15 | ManxPower | drumkilla: any progress on the call parking timout |
| 02:11.30 | drumkilla | haven't been able to work on it :/ |
| 02:11.54 | drumkilla | but I will get it this week |
| 02:12.18 | drumkilla | it's probably just some code I need to snag from head |
| 02:12.20 | ClayReiche123 | file[laptop]: Thanks. Sounds difficult.... |
| 02:12.28 | NewSole | jakepdev.. u see that |
| 02:12.44 | ManxPower | drumkilla: I need it fixed by Wed. Will a bounty help? |
| 02:12.58 | drumkilla | probably, heh |
| 02:13.00 | jakepdev | i do |
| 02:13.07 | drumkilla | that will probably make sure that someone gets it by then, in case I don't |
| 02:13.10 | NewSole | see users there |
| 02:13.12 | ManxPower | drumkilla: I'll post a bounty on monday. |
| 02:13.16 | drumkilla | alright |
| 02:13.30 | drumkilla | I might be the one to get it, hehe :) |
| 02:13.35 | ManxPower | drumkilla: My bounties do not exclude Developers. |
| 02:13.49 | file[laptop] | jakepdev: go, help NewSole little one! |
| 02:13.51 | jakepdev | NewSole: I would attack this by starting with just hard coding everything in the dialplan. I wouldn't use the iax.conf quite yet |
| 02:13.56 | drumkilla | I should be able to play with it tomorrow |
| 02:14.16 | jakepdev | file - don't know if I can - i only started with * a few weeks ago |
| 02:16.54 | |Vulture| | damn wiki... |
| 02:17.02 | |Vulture| | anyone know if there is a mirror of the manager commands? |
| 02:17.18 | jakepdev | anyone remember Nucleus from the 80's? |
| 02:17.26 | jakepdev | Vulture - use google cache |
| 02:17.52 | |Vulture| | kk thanx |
| 02:18.07 | ManxPower | Isn't there a complete list of Manager commands in a README in the Asterisk source? |
| 02:18.20 | jakepdev | who looks at that stuff |
| 02:18.35 | drumkilla | ha ... |
| 02:18.39 | jakepdev | hehe |
| 02:19.25 | ManxPower | also "show manager commands" will give you a list. As well as asterisk/doc/manager.txt |
| 02:21.11 | ManxPower | Well, I figured out why voip-info.org is down. |
| 02:21.22 | ManxPower | http://voxilla.com/voxstory155.html links to it. |
| 02:22.06 | file[laptop] | Josh is cool |
| 02:22.21 | file[laptop] | from Switchvox |
| 02:22.40 | ManxPower | Which is linked from a story on slashdot |
| 02:23.09 | *** join/#asterisk DrRighteous (~DrRighteo@ool-182c867b.dyn.optonline.net) |
| 02:25.08 | jakepdev | ah |
| 02:25.15 | jakepdev | it all makes sense now |
| 02:26.28 | drumkilla | what story from slashdot |
| 02:26.46 | drumkilla | file[laptop]: was Josh at VON? |
| 02:26.50 | file[laptop] | yes |
| 02:26.59 | file[laptop] | drumkilla: remember Tristan? well, Josh was the guy with her |
| 02:27.10 | ManxPower | drumkilla: http://hardware.slashdot.org/article.pl?sid=05/04/17/2327225&threshold=3&tid=215&tid=187&tid=189 |
| 02:27.18 | file[laptop] | not her husband mind you |
| 02:27.40 | PTG1234 | why don't they do |
| 02:27.43 | drumkilla | yeah, I know who it is - just wondering if it was Josh |
| 02:27.49 | PTG1234 | if REFER=BLAH redirect(sorry too busy) |
| 02:27.50 | file[laptop] | yup yup |
| 02:28.39 | PTG1234 | hmm |
| 02:28.42 | PTG1234 | do i want 9meg mp3s? |
| 02:29.08 | jakepdev | couldn't you convert them to a low bitrate? |
| 02:29.26 | Moonwick | I've started ripping everything I own into apple lossless. |
| 02:29.31 | drumkilla | when was that on /. |
| 02:29.39 | Moonwick | disk space is too cheap to bother with lossy compression nowadays |
| 02:29.58 | fugitivo | use google! |
| 02:30.01 | jakepdev | but - typically in telephony - you wouldn't hear the difference |
| 02:30.04 | fugitivo | to store your mp3! |
| 02:30.42 | jakepdev | at least with TDM - it's difficult to hear better than a 11khz sample |
| 02:31.02 | shodan | I'm looking for specs for phone lines and phones (stuff like how phones modulate voice , how phone lines modules rings and CID , how much current phones can draw at max , maximum line voltage sag etc..) anyone has some links on that ? |
| 02:31.05 | timecop | well fucking shit |
| 02:31.16 | timecop | anyone know anythign about h323? |
| 02:31.28 | *** join/#asterisk mrproper_ (~b@61.95.55.242) |
| 02:31.31 | PTG1234 | well |
| 02:31.32 | jakepdev | timcop - JerJer does |
| 02:31.37 | PTG1234 | is apple lossless better then 320kbps mp3? |
| 02:31.39 | timecop | i'm calling to some voip provider in china, all they suppsoedly require is caller ID set to blah, then dial H323/whatever@theirIP |
| 02:31.47 | jakepdev | he has several sites running H.323 |
| 02:31.51 | timecop | and al I get back from asterisk is No one is available to answer at this time (1:0/0/0) |
| 02:31.57 | timecop | h323 debug doesnt show shit |
| 02:32.01 | timecop | (no useful shit, that is) |
| 02:32.03 | mrproper_ | anyone here familiar with oh323, i can make outgoing h323 calls everything runs ok, but all incoming calls ring but no audio either way |
| 02:32.22 | jakepdev | ah - the h323 squad!!! |
| 02:32.28 | mrproper_ | lol |
| 02:32.38 | jakepdev | they come in numbers |
| 02:32.43 | timecop | huhu |
| 02:32.57 | jakepdev | PTG - for telephony? |
| 02:33.11 | jakepdev | or in general? |
| 02:33.45 | mrproper_ | who is that in response to? |
| 02:33.50 | jakepdev | shodan - try about.com |
| 02:34.09 | jakepdev | shodan - or google what you're looking for |
| 02:34.15 | *** part/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com) |
| 02:34.53 | jakepdev | mrproper - it was in response to PTG |
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| 02:37.25 | NewSole | hmmm |
| 02:37.44 | jakepdev | did you hard code that stuff? |
| 02:37.50 | jakepdev | in the dialplan? |
| 02:38.08 | shodan | jakepdev, about.com's search is useless , but going throught the categories doesn't return anything useful, and I'm still searching with google but there's so much stuff on telephony (and it's all consumer level stuff) I can't find some hard numbers |
| 02:38.19 | NewSole | i tried... now its not even giving a reject code |
| 02:38.34 | jakepdev | NewSole: what is it giving? |
| 02:38.54 | NewSole | <PROTECTED> |
| 02:38.54 | NewSole | <PROTECTED> |
| 02:38.54 | NewSole | <PROTECTED> |
| 02:39.24 | NewSole | <PROTECTED> |
| 02:39.46 | jakepdev | iax2 show debug |
| 02:40.04 | NewSole | No such command 'iax2 show debug' (type 'help' for help) |
| 02:40.28 | riksta | iax not iax2 |
| 02:40.55 | jakepdev | iax2 debug |
| 02:41.03 | riksta | ah yeah |
| 02:41.20 | *** join/#asterisk egon_l (~egon@pc-10-19-104-200.cm.vtr.net) |
| 02:41.22 | riksta | im talking crap |
| 02:42.42 | NewSole | http://pastebin.ca/9735 |
| 02:47.00 | NewSole | jakepdev |
| 02:47.10 | jakepdev | NewSol - INVAL is Invalid Request |
| 02:47.12 | *** join/#asterisk wolfson` (~hehe@cpe-68-187-189-084.man.nc.charter.com) |
| 02:47.17 | *** part/#asterisk shodan (~shodan@216.113.99.185) |
| 02:48.15 | jakepdev | ok - why does it say dialing IAX over ZAP :) |
| 02:48.16 | jakepdev | ? |
| 02:48.39 | jakepdev | what is your exact dialstring - pastebin - but replace your pwd with ****** |
| 02:48.42 | NewSole | zap card dialing out |
| 02:48.45 | jakepdev | no |
| 02:48.57 | jakepdev | if you're using IAX - you're not using ZAP |
| 02:49.03 | jakepdev | and vice versa |
| 02:49.13 | *** join/#asterisk Kumbang (~ecvs@167.205.24.4) |
| 02:49.23 | jakepdev | you can initate a call from one, and terminate to another |
| 02:49.32 | file[laptop] | he's probably using an FXS module |
| 02:49.56 | jakepdev | are you NewSole? |
| 02:51.03 | NewSole | i am |
| 02:51.06 | jakepdev | ok |
| 02:51.31 | file[laptop] | see? I'm psychic |
| 02:51.43 | jakepdev | right :) |
| 02:51.43 | NewSole | server is downstairs... so I hooked a cordless |
| 02:52.16 | jakepdev | file - is the debug out supposed to look like that? |
| 02:52.30 | jakepdev | i'm not familiar with ZAP over IAX |
| 02:53.08 | NewSole | question is WHAT is INVALID |
| 02:53.22 | *** join/#asterisk ToyMan (~stuq@user-12lcqur.cable.mindspring.com) |
| 02:53.40 | NewSole | http://pastebin.ca/9736 |
| 02:53.42 | jakepdev | i'd say something in your dial string - but don't know for sure |
| 02:54.14 | jakepdev | NewSole - remember how we talked about hard coding to do a test? |
| 02:54.18 | NewSole | dial string is Dial(IAX2/masters@masters/${EXTEN},100,rT) |
| 02:54.33 | jakepdev | ok |
| 02:54.38 | NewSole | axualy |
| 02:54.57 | NewSole | exten = _X.,1,Dial(IAX2/masters@masters/${EXTEN},100,rT) |
| 02:55.23 | L|NUX | exten => _X.,1,Dial(IAX2/masters@masters/${EXTEN},100,rT) |
| 02:55.31 | L|NUX | it should look like this :) |
| 02:55.48 | GustavoIPA | alguem do brasil por aqui??? |
| 02:56.27 | *** join/#asterisk jterrero (~jt@ool-43576e0d.dyn.optonline.net) |
| 02:56.41 | mrproper_ | when i dial outbound h323 calls, all works fine, but an incomming h323 call establishes but no audio either way, any ideas? |
| 02:59.12 | NewSole | hmm |
| 03:01.19 | mrproper_ | can anyone tell me how to start up a remote console that only shows warning and errors, without all the 'information' events |
| 03:01.48 | drumkilla | logger.conf |
| 03:02.33 | fugitivo | mrproper_: are you behind nat? |
| 03:02.44 | mrproper_ | fugitivo: no |
| 03:03.17 | mrproper_ | i have asterisk with oh323 pointing to a gatekeeper which then goes to a gateway then isdn, outbound calls are fine, inbound establish but no audio what so ever |
| 03:04.12 | *** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
| 03:05.19 | NewSole | jakepdev... what do you think |
| 03:05.36 | jakepdev | i'm not familar with Zap over IAX |
| 03:05.59 | jakepdev | you're using a PCI FXS card in your * box? |
| 03:06.13 | mrproper_ | this is the oh323 debug output from an incoming call: http://pastebin.ca/9737 |
| 03:06.30 | NewSole | one.... on local server |
| 03:07.38 | jakepdev | but in this scenerio, you're trying to place a remote ZAP call using the IAX protocol? |
| 03:07.50 | NewSole | no |
| 03:08.30 | NewSole | i have a local box here with TDM40P card.... making local call off phone to IAX |
| 03:08.38 | jakepdev | ok |
| 03:08.44 | NewSole | acualy going like this |
| 03:09.12 | NewSole | ZAP => Asterisk => IAX2 => Asterisk => SIP |
| 03:09.44 | jakepdev | ok - but we're just working on the ZAP -> * -> IAX part now - right? |
| 03:09.55 | NewSole | yup |
| 03:10.17 | NewSole | but problem not only with ZAP |
| 03:11.00 | NewSole | its with "* => IAX2 => *" weather I use Zap Card or Mobil PRI |
| 03:11.14 | jakepdev | NewSole - is "14163101010" a valid exten on your remote box? |
| 03:11.31 | FuriousGeorge | im started setting up my tdm fxs, and earlier (when i didnt expect it to work yet) i would pick it up and hear whatever went into the mouthpiece in the earpiece |
| 03:11.36 | FuriousGeorge | now it just sounds like a dead line |
| 03:11.46 | FuriousGeorge | *tdm w/ 2 fxs |
| 03:11.48 | NewSole | http://pastebin.ca/9736 |
| 03:11.55 | NewSole | look at that |
| 03:12.10 | jakepdev | k - i did |
| 03:12.22 | jakepdev | does the remote box give you any indication of what is going on? |
| 03:12.38 | jakepdev | any errors from debug out on that? |
| 03:13.03 | NewSole | FindChannel check to see if its a local DID or PRI account |
| 03:13.44 | NewSole | and then routes to Moblie PRI Box |
| 03:14.10 | mrproper_ | i have asterisk with oh323 pointing to a gatekeeper which then goes to a gateway then isdn, outbound calls are fine, inbound establish but no audio what so ever |
| 03:14.40 | NewSole | our system goes like this |
| 03:14.45 | jakepdev | NewSole - I'm saying - on "masters", can you do an iax2 debug? |
| 03:15.28 | NewSole | <MOBLIE PRI> ==> <PRI SERVER> ==> <ASTERISK> ==> SIP |
| 03:16.16 | FuriousGeorge | another thing: anyone know how linux goes about assigning irws to pci devs. in my bios i assigned irq 10 to pci slot 4 (since cat proc/dev said 10 was free), but when i boot, loading the driver has it showing up on irq3 w/ usb controller |
| 03:16.37 | FuriousGeorge | *irw=irq |
| 03:16.44 | jakepdev | NewSole - the messages on you local box to don't say much about why it's invalid - it just says that it is invalid |
| 03:17.29 | jakepdev | FG - i was told by Digium it should use what you set in the BIOS |
| 03:19.11 | NewSole | http://pastebin.ca/9738 |
| 03:19.24 | NewSole | thats all I get for debug on masters |
| 03:19.41 | jakepdev | that's nothing :) |
| 03:20.02 | *** join/#asterisk WilliamK (~wkeller@c-24-0-130-177.hsd1.tx.comcast.net) |
| 03:20.25 | *** join/#asterisk implicit (~implicit@ip68-7-154-222.sd.sd.cox.net) |
| 03:20.29 | NewSole | thats what i mean all I get is INVAL |
| 03:21.19 | jakepdev | let me see if I can set up somthing - see if you can connect to me via IAX |
| 03:22.54 | *** join/#asterisk hcir (~ar@rdbck-static-532.palmer.mtaonline.net) |
| 03:25.02 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
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| 03:29.08 | MrEntropy | yo |
| 03:29.28 | MrEntropy | how can i get info on what codecs both sides of a call are using? |
| 03:38.06 | *** join/#asterisk MikeJ[Laptop] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net) |
| 03:42.04 | jakepdev | sip show channels for SIP |
| 03:43.54 | mrproper_ | can anyone tell me what the issue is with this debug output (looks like theres an issue with zap http://pastebin.ca/9406 |
| 03:45.03 | jakepdev | mrproper - what's the problem you're having? |
| 03:46.26 | mrproper_ | jakepdev: can make sip calls in and out fine, can make h323 calls outbound ok, but h323 inbound calls establish but no audio either way |
| 03:46.34 | jakepdev | oh |
| 03:46.39 | jakepdev | sorry |
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| 03:56.22 | hardwire | what a lovely nick |
| 04:02.45 | *** join/#asterisk riksta (~rick@81-178-248-131.dsl.pipex.com) |
| 04:09.27 | *** join/#asterisk viLeR (1000@ip-47-252.telesat.com.co) |
| 04:10.02 | jterrero | is exten => 2121,1,Dial(Zap/g2/19178602911) |
| 04:10.08 | jterrero | ccorret ? |
| 04:10.11 | jterrero | *correct |
| 04:11.13 | jakepdev | that all depends what you're trying to accomplish - but the syntax seems correct |
| 04:11.41 | jterrero | I am trying to have pots line connected to do a dual fxo/fxs use my phone line and dial a number |
| 04:11.48 | *** join/#asterisk CaptChris (~Chris@ppp-69-110-104-52.dialup.skt2ca.pacbell.net) |
| 04:12.31 | jakepdev | yep - if you dial 2121, * will dial 19178602911 |
| 04:12.46 | jakepdev | if the rest of your dialplan is set up properly |
| 04:13.09 | jakepdev | and your zaptel.conf/zapata.conf is set up properly |
| 04:13.22 | jterrero | how can i tell? |
| 04:13.39 | jterrero | nm |
| 04:13.41 | jterrero | ill google |
| 04:14.04 | CaptChris | Hello all. |
| 04:15.38 | Sedorox | ~wiki-status |
| 04:15.39 | jbot | [wiki-status] Up and Running |
| 04:15.39 | bugbot | wiki-status is assigned nothing and reported nothing. |
| 04:15.46 | Sedorox | correct... |
| 04:16.03 | jakepdev | yeah!!!! |
| 04:16.14 | psycodad | moinmoin |
| 04:16.15 | jakepdev | finally |
| 04:19.59 | *** join/#asterisk iq (~IQ@70-59-160-225.omah.qwest.net) |
| 04:26.36 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 04:30.10 | *** join/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com) |
| 04:33.21 | ManxPower | ~docs |
| 04:33.22 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 04:33.22 | bugbot | docs is assigned nothing and reported nothing. |
| 04:33.23 | ManxPower | ~mailinglist |
| 04:33.24 | jbot | from memory, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 04:33.24 | bugbot | mailinglist is assigned nothing and reported nothing. |
| 04:34.55 | *** join/#asterisk iq (~iq@70-59-160-225.omah.qwest.net) |
| 04:40.16 | *** join/#asterisk CaptChris (~Chris@ppp-69-110-104-174.dialup.skt2ca.pacbell.net) |
| 04:40.43 | CaptChris | I'm back. stupid dialup dropped my connection |
| 04:41.20 | CaptChris | can someone explain (in simple terms) what the config files generally contain. I think I understand that the iax.conf & sip.conf contains config info for each iax & sip phone that is considered "local" to asterisk. but i'm not fully understanding what this means. i seem to recall reading that the [context] labels in these files tell asterisk what to do when a local extension goes off hook (for outgoing calls). |
| 04:41.25 | CaptChris | while the extension.conf file contains the [context] labels for inbound calls. am i looking at this correctly? |
| 04:41.27 | *** join/#asterisk Rick_Hunter (~rhunter@08-201.008.popsite.net) |
| 04:43.21 | ManxPower | CaptChris: all calls start out as "inbound" |
| 04:43.53 | ManxPower | the context= line in sip.conf/iax.con says where a call from that device starts in the dialplan |
| 04:44.39 | CaptChris | ok. that clears up one misconception. |
| 04:45.07 | *** join/#asterisk chaoscon (~ph33r@chaoscon.user) |
| 04:46.19 | CaptChris | am i correct in the sip/iax.conf files defining info regarding the individual device? |
| 04:47.39 | ManxPower | generally yes |
| 04:49.57 | CaptChris | ok. so if i now understand it correctly, the general section of the iax.conf tells asterisk where to start in the dialplan for inbound iax calls that originate from non-local devices. and the same for sip.conf for sip calls. is that correct? |
| 04:51.45 | *** join/#asterisk vinumohfx (~vinmohfx@pcp0010310792pcs.avenel01.nj.comcast.net) |
| 04:52.31 | vinumohfx | hi all |
| 04:53.22 | CaptChris | hello vinumohfx |
| 04:53.39 | vinumohfx | hey captchirs ..how u doing ? |
| 04:54.20 | CaptChris | ok. just trying to fully understand the asterisk conf files |
| 04:54.28 | vinumohfx | I would appriciate if any one could tell me how to change default password for Asterisk Management Portal (AMP) for Asterisk@Home |
| 04:55.00 | CaptChris | that's beyond me |
| 04:55.04 | vinumohfx | good luck captchris |
| 04:55.18 | riksta | vinumohfx: i imagine it's stored in the mysql database |
| 04:55.32 | vinumohfx | hehe dont worry captchirs you will get used to that ..keep on reading |
| 04:55.40 | |Vulture| | anyone here know perl? |
| 04:55.48 | vinumohfx | wht r u trying to do with asterisk..can I help you >? |
| 04:55.49 | riksta | |Vulture|: some |
| 04:55.58 | CaptChris | thanks. i've been reading and reading... too much perhaps |
| 04:56.33 | vinumohfx | hey riksta..thanks..but do you know how to change the default password for "maint" ? |
| 04:56.35 | CaptChris | .... and from too many sources. got my mind a jumble |
| 04:57.00 | |Vulture| | riksta: I am using a perl script to telnet into the manager interface and do a "sip show peer X" I want to try and grab the " Status : OK (75 ms)" and then store just the "OK (75 ms)" any clue how? |
| 04:57.09 | riksta | vinumohfx: i've never used that software, but if you look in the mysql database you can probably update it (i'm just guessing) |
| 04:57.26 | riksta | |Vulture|: sure, i have some code to do more or less that |
| 04:57.30 | |Vulture| | any clue what command I would use for that? I can google it just really new to perl |
| 04:57.37 | riksta | |Vulture|: one minute |
| 04:57.48 | |Vulture| | oh damn haha I didn't see one out there, I wrote it for an interface into Nagios |
| 04:58.04 | vinumohfx | hey captchris..I was feeling exactly like you captchris..One of my co-worker knows everything and he didint share the knowledge..but this asterisk IRC community helped me |
| 04:58.15 | riksta | |Vulture|: i have some similar code in my ADM software |
| 04:58.26 | Corydon76-home | Why not use the asterisk-perl package on asterisk.gnuinter.net ? |
| 04:58.32 | riksta | |Vulture|: pm |
| 04:59.24 | vinumohfx | ok riksta |
| 04:59.34 | CaptChris | i think i've got most of the dialplan process figured out now. thanks to ManxPowe |
| 04:59.39 | vinumohfx | .I tried on google ..no info |
| 05:00.04 | riksta | Corydon-w: he just needs a simple Net::Telnet script |
| 05:00.08 | vinumohfx | I'll again ask the question on this room after a while.. |
| 05:00.31 | vinumohfx | cool captchris |
| 05:01.12 | vinumohfx | which version of asterisk r u using..captchris ? |
| 05:01.27 | Corydon76-home | riksta: yes, but you know that Asterisk::Manager already exists for this exact purpose, right? |
| 05:01.51 | riksta | yeah, my friend and I made it ;) |
| 05:02.14 | riksta | well, depending on which one you are refering to |
| 05:02.20 | CaptChris | one thing i'm not certain tho. does the [general] section of the iax.conf and sip.conf tell asterisk what to do for inbound calls from non-local devices... such as the "context=" statement. |
| 05:02.23 | Corydon76-home | Why use Net::Telnet to establish a clear-text authenticated manager connection, when you can use Asterisk::Manager to get an encrypted authentication? |
| 05:02.24 | CaptChris | i think the latest. |
| 05:02.28 | CaptChris | i'll check |
| 05:02.41 | Corydon76-home | riksta: the one on http://asterisk.gnuinter.net |
| 05:02.43 | riksta | Corydon-w: he just wanted something quick for a nagios check |
| 05:03.13 | Corydon76-home | That's the nice thing about using modules... they're quick... |
| 05:03.22 | CaptChris | version 1.0.6 |
| 05:03.38 | riksta | yeah, to be fair that's right :) |
| 05:03.43 | Corydon76-home | "Oh, he just wanted something quick... so use the module that makes his job more difficult" |
| 05:03.56 | riksta | Corydon-w: i hadn't seen this particular module, i had my own |
| 05:04.13 | Corydon76-home | riksta: It's been around for a LOOOOOONG time |
| 05:04.27 | riksta | fair enuff |
| 05:04.50 | Corydon76-home | Also has the Asterisk::AGI module, and several others |
| 05:05.31 | Corydon76-home | riksta: so if you've written a Perl interface to astman, why isn't it in CPAN? ;-) |
| 05:06.13 | riksta | sam did put it on there, it seems to be gone, i bet he found this one you are referring to |
| 05:06.25 | riksta | i'm just looking through the source :) |
| 05:07.35 | *** join/#asterisk docelmo (~me@116-39.202-68.tampabay.res.rr.com) |
| 05:07.49 | docelmo | Does anyone have a NuFone config for asterisk? |
| 05:08.02 | docelmo | or know where I can get one? |
| 05:08.15 | Corydon76-home | Maybe from NuFone? |
| 05:08.33 | Qwell | docelmo: nufone sends you one when you sign up |
| 05:09.53 | riksta | Corydon-w: much more elegant, i will make use of it from now on :P |
| 05:09.53 | docelmo | Here's a thought.. They should make it where someone can get it again if they dont have it.. |
| 05:10.04 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 05:10.30 | Qwell | docelmo: Its the same config you would use with any other provider... |
| 05:10.42 | docelmo | Well I dont know IP's or hostnames etc.. |
| 05:10.47 | riksta | here's a thought: search the wiki |
| 05:10.47 | riksta | http://www.voip-info.org/wiki-Asterisk+settings+nufone |
| 05:10.47 | Qwell | add an entry in iax.conf or sip.conf, and a simple part to your dial line |
| 05:10.51 | Qwell | switch-1.nufone.net |
| 05:11.09 | docelmo | And is the Username/PAss same as the login? |
| 05:14.43 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:19.01 | Qwell | yeah |
| 05:19.16 | Qwell | docelmo: Did you not get the email when you signed up? |
| 05:19.24 | JerJer | have a clue an use a type=peer |
| 05:19.41 | *** join/#asterisk ta[i]nted (ta_i_nted@68-190-218-71.pas-mres.charterpipeline.net) |
| 05:19.52 | Qwell | JerJer: morning |
| 05:26.51 | docelmo | Ya I got the email like 5 months ago |
| 05:27.09 | docelmo | Is my username/password same for the website? |
| 05:30.39 | docelmo | ok nevermind.. Got past that now.. What codecs does NuFone support? |
| 05:30.41 | PTG1234 | JerJer: have a clue an use a type=peer : customer service at its finest :) |
| 05:33.07 | docelmo | Hay Jer, got a sec? |
| 05:34.47 | |Vulture| | I would say no... |
| 05:35.28 | |Vulture| | anyone here using FXS to T1 PRI bi-directional fax? |
| 05:35.44 | vinumohfx | I would appriciate if any one could tell me how to change the default password for "AMP" - Asterisk Management Portal |
| 05:36.16 | docelmo | ok can someone explain to me why this doesnt work when I have g729 and ULAW? |
| 05:36.17 | docelmo | Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REJECT |
| 05:36.17 | docelmo | <PROTECTED> |
| 05:36.17 | docelmo | <PROTECTED> |
| 05:36.17 | docelmo | <PROTECTED> |
| 05:36.25 | JerJer | hay is for horses |
| 05:36.32 | JerJer | or i have lots of seconds |
| 05:36.33 | *** join/#asterisk three55ml (~none@cpe-66-68-98-68.austin.res.rr.com) |
| 05:37.11 | three55ml | I'm doing some server testing, has anyone done any tests with ping times and SIP quality? I.e. noticed a difference between say 25ms routes and 50-75ms routes? |
| 05:37.17 | *** join/#asterisk ellvis (~ellvis@195.98.29.34) |
| 05:37.18 | |Vulture| | rm -rf /* thats how you change AMP settings :P |
| 05:37.22 | ellvis | hi ppl |
| 05:37.37 | *** join/#asterisk mcnobody (~laaksola@server.kopteri.net) |
| 05:38.55 | |Vulture| | three55ml: my local peers always show ~75ms to IP500s |
| 05:41.09 | three55ml | |Vulture|: I would figure around 100ms you would start noticing things. |
| 05:41.38 | three55ml | |Vulture|: Trying to decide if I want to do the whole east-coast/west-coast server thing now or wait a little bit. |
| 05:42.36 | |Vulture| | three55ml: well if you have a dedicated line it shouldn't be much of a problem |
| 05:42.42 | |Vulture| | but like DSL/Cable you might |
| 05:43.01 | three55ml | |Vulture|: Exactly |
| 05:43.22 | vinumohfx | uhh vulture..that command is to remove |
| 05:43.50 | |Vulture| | vinumohfx: and was can only hope.... |
| 05:44.08 | vinumohfx | could any one tell me how to change default password for AMp |
| 05:44.19 | vinumohfx | ok thanks vulture |
| 05:45.46 | |Vulture| | vinumohfx: is this a .htpasswd login/pw? |
| 05:47.43 | |Vulture| | if so you will need to use the htpasswd command to change it |
| 05:47.54 | vinumohfx | I tried htpasswd but I'm not familiar with that command |
| 05:48.15 | vinumohfx | default user is "maint" |
| 05:48.44 | |Vulture| | do a "updatedb" then "locate htpasswd.users" |
| 05:48.45 | vinumohfx | if i leave default password any one from web could log in to my server |
| 05:48.49 | |Vulture| | see if you can find it |
| 05:49.23 | |Vulture| | then you will need to do "htpasswd (the file here) maint" it will prompt you to change the password |
| 05:49.38 | |Vulture| | I f'ed with you so now Ill help you |
| 05:50.07 | |Vulture| | just AMP && *@Home are not ways to learn * |
| 05:50.10 | vinumohfx | locate shows = no such file or directory |
| 05:50.24 | |Vulture| | hmmm |
| 05:50.27 | |Vulture| | 1min Ill dload it |
| 05:50.36 | vinumohfx | ok vulture |
| 05:50.42 | vinumohfx | thanks |
| 05:52.39 | |Vulture| | locate " |
| 05:52.46 | |Vulture| | "wwwpasswd" |
| 05:53.47 | vinumohfx | no such file or directory |
| 05:53.58 | |Vulture| | when you did "locate wwwpasswd" ? |
| 05:54.13 | vinumohfx | yes vulture |
| 05:54.23 | |Vulture| | what flavor linux you running? |
| 05:55.10 | |Vulture| | and are you running *@Home or AMP? |
| 05:55.11 | vinumohfx | cent Linux |
| 05:55.24 | vinumohfx | *@home |
| 05:55.28 | |Vulture| | ah okay |
| 05:55.43 | |Vulture| | "locate httpd.conf" |
| 05:55.51 | vinumohfx | Amp is the html interface which comes with home |
| 05:56.15 | vinumohfx | no file |
| 05:56.27 | vinumohfx | It says bad ELF interpreter |
| 05:56.34 | |Vulture| | did you run "updatedb" ? |
| 05:56.44 | *** join/#asterisk Manipura (~chatzilla@dsl-ep-209-115-250-i114-cgy.nucleus.com) |
| 05:57.30 | vinumohfx | wait a sec |
| 05:57.57 | |Vulture| | it should take a bit |
| 05:58.11 | |Vulture| | after it runs, do "locate wwwpasswd" |
| 05:58.15 | vinumohfx | uhh after I typed "rm-rf /* something happened |
| 05:58.17 | *** join/#asterisk Zaw (zaw@zaw.subneural.net) |
| 05:58.24 | vinumohfx | no cammand is being taken in |
| 05:58.33 | |Vulture| | you actually ran that? |
| 05:58.48 | |Vulture| | jesus.... |
| 05:58.58 | vinumohfx | yeh I know |
| 05:59.06 | |Vulture| | that system is toasted |
| 05:59.08 | vinumohfx | I didint think much cause |
| 05:59.17 | vinumohfx | I was new to linux and asterisk |
| 05:59.24 | |Vulture| | didn't expect anyone to actually run that command in this channel |
| 05:59.28 | vinumohfx | then I realised that it is a removal command |
| 05:59.52 | |Vulture| | Id recommend you grab yourself a copy of FC3, then manually install * |
| 05:59.57 | |Vulture| | do everything command line |
| 06:00.05 | |Vulture| | FC3 will be very user friendly install |
| 06:00.10 | vinumohfx | now i rebooted |
| 06:00.16 | |Vulture| | its not gunna work |
| 06:00.20 | vinumohfx | it shows grub> |
| 06:00.21 | vinumohfx | cool |
| 06:00.25 | Qwell | and, for christs sake, don't run arbitrary commands as root |
| 06:00.31 | |Vulture| | its gunna Kernel Panic |
| 06:00.51 | |Vulture| | Qwell: Ive never seen someone actually run an full remove command |
| 06:01.00 | Qwell | |Vulture|: yeah, well |
| 06:01.08 | |Vulture| | now I kinda feel bad... |
| 06:01.14 | vinumohfx | but i told you..I'm not a linux person |
| 06:01.15 | Qwell | don't expect any sympathy... |
| 06:01.26 | vinumohfx | its ok its my mistake |
| 06:01.35 | |Vulture| | it is one less *@Home install.... |
| 06:01.42 | Qwell | |Vulture|: true :p |
| 06:02.09 | vinumohfx | now I cannot sleep tonight..i have to configure it from scratch before I get to my office |
| 06:02.25 | Qwell | sleep is overrated |
| 06:02.35 | |Vulture| | vinumohfx: AMP is not where you want to be |
| 06:02.35 | techie | so true |
| 06:02.52 | |Vulture| | yea look at me I never sleep... but I am a druggie so... |
| 06:03.02 | vinumohfx | yeh all right |
| 06:03.29 | |Vulture| | vinumohfx: your going to run this in a office? |
| 06:03.46 | vinumohfx | yeh in a start up office |
| 06:03.48 | *** join/#asterisk quickmoney (~jfu2808@CPE00a0c5e1b8b3-CM0012c999e6a0.cpe.net.cable.rogers.com) |
| 06:03.54 | Qwell | home != office |
| 06:03.57 | |Vulture| | hehehe |
| 06:04.07 | |Vulture| | Qwell: I was thinking of the irony |
| 06:04.46 | vinumohfx | yeh is there any version of asterisk to be used for office ? |
| 06:05.03 | |Vulture| | vinumohfx: you would be better off learning how to actually use * plain |
| 06:05.07 | |Vulture| | yea * |
| 06:05.47 | vinumohfx | I have faith on myself..I always do like this and I'll learn from my mistake |
| 06:06.33 | vinumohfx | I started my own cable telivision company when I dont even know how the signals r transmited through cable ..lol |
| 06:06.40 | |Vulture| | vinumohfx: asterisk is the program |
| 06:06.53 | |Vulture| | *@Home is a compiled install |
| 06:07.05 | vinumohfx | And after 3 years it became sucess and sold to big company.lol |
| 06:07.06 | |Vulture| | amp is a gui |
| 06:07.21 | vinumohfx | so I'll learn asterisk like this |
| 06:07.25 | vinumohfx | from mistakes |
| 06:07.26 | vinumohfx | :) |
| 06:07.41 | |Vulture| | do this though, get FC3 |
| 06:07.43 | |Vulture| | install it |
| 06:07.49 | |Vulture| | then work with * Source |
| 06:08.44 | vinumohfx | but why ? Asterisk home is easy to use |
| 06:08.56 | vinumohfx | since it has http interface |
| 06:09.20 | vinumohfx | And I downloded today |
| 06:09.50 | vinumohfx | and I already configured 20 sip extensions with sip softphones |
| 06:10.16 | vinumohfx | I was about to do Pstn to sip through adit 600 tommorrow |
| 06:10.17 | Mavvie | who has here experience with G4 and/or colour faxing ? |
| 06:10.32 | vinumohfx | Any way thanks vulture |
| 06:10.35 | vinumohfx | take care |
| 06:10.59 | Mavvie | in general, not asterisk / voip related |
| 06:14.51 | *** part/#asterisk vinumohfx (~vinmohfx@pcp0010310792pcs.avenel01.nj.comcast.net) |
| 06:15.28 | j0 | i'm using a x100p card.. when i speak while going through the ivr, the inbound voice is choppy.. is this due to the echo cancellation not workinng correctly? |
| 06:15.32 | timecop | fucking h323 |
| 06:15.36 | timecop | no compatible codecs myt fucking ass |
| 06:15.58 | Qwell | j0: Is it a clone? |
| 06:16.07 | timecop | j0: did you buy the shit off scambay? |
| 06:16.11 | timecop | in that case you fucking deserve it |
| 06:16.13 | j0 | yes, and yes :) |
| 06:16.18 | timecop | yep, now go fuck yourself |
| 06:16.23 | j0 | explain. |
| 06:16.27 | timecop | dont be a jew, support digium |
| 06:16.40 | j0 | for learning purposes.. $15 vs $100 |
| 06:16.50 | timecop | well, you got what you paid for |
| 06:17.00 | implicit | timecop, don't be such a dick |
| 06:17.07 | timecop | hey, its true. |
| 06:17.09 | implicit | it is the same shit |
| 06:17.10 | timecop | he got scammed |
| 06:17.13 | Qwell | little harsh, but true nonetheless |
| 06:17.23 | implicit | not really, digium scams people by selling that shit for 100 bucks |
| 06:17.25 | j0 | ok, so no settings changes will make it any better? |
| 06:17.29 | Qwell | You can't expect a clone to be the same quality |
| 06:17.40 | implicit | Qwell, it is the same quality, they both suck balls |
| 06:17.47 | timecop | implicit: thats ok, digium can have all my money they want |
| 06:17.53 | implicit | lol |
| 06:18.11 | implicit | and why's that? |
| 06:18.40 | j0 | heheh |
| 06:18.42 | implicit | heh |
| 06:18.44 | Qwell | drop it |
| 06:18.49 | Qwell | There is no need to continue this |
| 06:19.08 | asiod | anyone want to give me free stuff |
| 06:19.30 | j0 | so i'm screwed end of story? |
| 06:20.00 | implicit | not completely screwed, but it will be the same quality with the digium one |
| 06:20.32 | j0 | and its nothing that can be tweaked with software settings? |
| 06:20.49 | implicit | yeah there are |
| 06:20.52 | implicit | many things |
| 06:21.05 | j0 | alright.. thats all i needed to know :) thanks |
| 06:21.22 | implicit | :) |
| 06:22.25 | |Vulture| | Qwell: I think I did my civic duty today my erradicating another *@Home install |
| 06:23.18 | Qwell | except he's going to use it still |
| 06:23.49 | j0 | hey, i think *@home is a great starter.. it only takes a few min and u can start seeing what * can do for you... granted its a big mess for anything else.. but thats how i started |
| 06:23.50 | *** join/#asterisk shepherd (matt@pcp01541028pcs.huntsv01.al.comcast.net) |
| 06:24.08 | j0 | hey it works! .. k now lets do this the right way |
| 06:24.23 | Qwell | except 90% of people will never get that far |
| 06:24.33 | j0 | ah well.. then it wasn't meant to be |
| 06:24.34 | Qwell | "hey, it works!" "I'm gonna use this next time too!" |
| 06:24.46 | j0 | very true.. why fiddle with it when it already works |
| 06:24.57 | Qwell | then those same people come here, trying to figure out the basica |
| 06:24.58 | Qwell | basics |
| 06:25.44 | Qwell | and they bitch and moan, "asterisk sucks!", because *@home doesn't do what they want/need |
| 06:25.52 | |Vulture| | yea |
| 06:25.59 | |Vulture| | cause they never learn what extensions.conf is |
| 06:26.18 | elric | asterisk sucks :| |
| 06:26.26 | |Vulture| | if you start with source, then going to AMP is a decision, starting with AMP thats just asking for trouble |
| 06:26.48 | |Vulture| | plus have you seen that extensions.conf with AMP... jesus you better know your * before you look at that |
| 06:27.18 | elric | what is AMP? |
| 06:28.01 | Qwell | ~amp |
| 06:28.02 | jbot | somebody said amp was an Audio MPEG Player. [non-free], or http://amp.coalescentsystems.ca/ |
| 06:28.02 | bugbot | amp is assigned nothing and reported nothing. |
| 06:28.06 | Qwell | nope |
| 06:28.25 | shepherd | ~forget amp |
| 06:28.26 | bugbot | forget amp is assigned nothing and reported nothing. |
| 06:29.01 | timecop | fucking getting those old openh323 / pwlib versions is part of the problem |
| 06:29.13 | timecop | um, do I need any special build options for sthe h323 shit to work correctly? |
| 06:29.16 | shepherd | mantis bot? |
| 06:29.27 | |Vulture| | shepherd: yup |
| 06:29.29 | timecop | like --enable-plugins or anything. |
| 06:29.30 | elric | i compile without OH323 anyway |
| 06:29.34 | Qwell | ~bugbot |
| 06:29.35 | jbot | [bugbot] a bot that gives bug statuses. You can /msg bugbot help for info or visit him on #asterisk-bugs. |
| 06:29.35 | bugbot | bugbot is assigned nothing and reported nothing. |
| 06:29.38 | timecop | well, thats because you dont need to use it |
| 06:29.47 | shepherd | two info bots, that's crazy |
| 06:29.51 | shepherd | we don't need two |
| 06:30.04 | elric | ~shepherd |
| 06:30.06 | jbot | methinks shepherd is a Sharp Zaurus SL-C750, or a dog. |
| 06:30.06 | bugbot | shepherd is assigned nothing and reported nothing. |
| 06:30.06 | Qwell | bugbot is good |
| 06:30.08 | timecop | ~timecop |
| 06:30.09 | bugbot | timecop is assigned nothing and reported M1178, M1426, M25, M1475. |
| 06:30.25 | Qwell | except he talks too much... |
| 06:30.30 | elric | dude it called you a dog :( |
| 06:30.51 | shepherd | :( |
| 06:32.04 | asiod | i submitted a patch for meetme last night |
| 06:32.23 | asiod | but it looks like everyone is extending meetme |
| 06:33.26 | asiod | i suggested to stevek that if app_conference could work like startmusiconhold then it would be able to implement most of meetme's features as dialplans |
| 06:36.20 | *** join/#asterisk riksta (~rick@81-178-248-131.dsl.pipex.com) |
| 06:38.49 | *** join/#asterisk barshad (kkhhaannuu@202.134.140.30) |
| 06:38.55 | barshad | Hello all, |
| 06:38.56 | *** join/#asterisk gres (~serg@81.222.48.242) |
| 06:39.04 | asiod | felicitations |
| 06:39.16 | *** join/#asterisk bonez39 (~aint@drjones.dsl.xmission.com) |
| 06:39.17 | barshad | i'm facing the problem compiling oh323 "chan_oh323.c:37:34: asterisk/channel_pvt.h: No such file or directory" |
| 06:39.22 | barshad | can some one help me ? |
| 06:41.42 | JerJer | sorry i cannot help you |
| 06:41.48 | JerJer | tell the author to update his code |
| 06:44.48 | JerJer | i cannot support both my and his code |
| 06:44.54 | JerJer | sorry, i just cannot do it |
| 06:45.10 | implicit | i'm not saying you should either |
| 06:45.12 | implicit | ;) |
| 06:45.24 | implicit | screw him |
| 06:45.42 | j0 | hmm. i cant figure out where to stop my x100p from answering the phone.. i'd like to use it for outbound only.. setting the context to an empty one doesn't work |
| 06:46.10 | Silik0n | easy fix..unplug the phone line from it |
| 06:46.21 | j0 | stops the outbound calls too |
| 06:46.22 | riksta | how do you then make an outgoing call ? |
| 06:46.26 | implicit | lol |
| 06:46.29 | j0 | heh |
| 06:46.31 | implicit | riksta, :) |
| 06:46.33 | Qwell | use a relay... |
| 06:46.36 | riksta | muppets |
| 06:46.41 | Qwell | when you want outgoing, send a signal to the relay |
| 06:46.47 | implicit | hahaha |
| 06:46.53 | j0 | haha nice nice |
| 06:47.16 | j0 | i'm sure it could be done |
| 06:47.18 | Silik0n | evenbetter if you want to make anoutgoing call, use the same phone that gets the incoming calls |
| 06:47.20 | asiod | at&t usa direct how can we help you? |
| 06:47.31 | riksta | oh it just gets better |
| 06:47.48 | Qwell | riksta: the stupid Oz thing? |
| 06:47.56 | riksta | ? |
| 06:48.04 | Qwell | dunno, you mentioned muppets |
| 06:48.17 | Qwell | Muppets: Wizard of Oz |
| 06:48.23 | riksta | no, jim henderson |
| 06:48.36 | implicit | j0: quick hack, take out the whole part of the code for zap that picks up the phone |
| 06:48.40 | implicit | and then it doesn't matter |
| 06:48.46 | implicit | grep is your friend |
| 06:49.15 | riksta | you'd probably want to leave the method and return something |
| 06:49.16 | implicit | grep -r is your friend on steroids |
| 06:49.48 | *** join/#asterisk tzafrir_laptop (~tzafrir@62.90.10.53) |
| 06:50.17 | j0 | ok :) so there isn't a easy one |
| 06:50.21 | jakepdev | or just have * answer the phone and pput an extensions behind it |
| 06:50.32 | jakepdev | after all - * is a pbx |
| 06:50.33 | Qwell | jakepdev has the "proper" method |
| 06:50.41 | j0 | i just did it by making it run Hangup on an incoming call |
| 06:50.49 | j0 | so it never actually answers |
| 06:51.06 | implicit | o |
| 06:51.06 | j0 | but then for each ring i get a nice error |
| 06:51.13 | riksta | well it depends if you actually care about the call |
| 06:51.30 | j0 | yeah i dont care about incoming calls on that interface. |
| 06:51.33 | implicit | :) |
| 06:51.34 | implicit | good |
| 06:51.35 | Qwell | asterisk is a PBX, not an answering machine |
| 06:51.35 | implicit | godonight |
| 06:51.49 | j0 | just wish * would check for a dialtone before dialing |
| 06:51.53 | riksta | good morning |
| 06:52.18 | Qwell | j0: again, its a PBX. Not something that should need to check if a line is in use by something other then itself |
| 06:52.18 | implicit | Qwell, or even worse, some people think it's a carrier grade commercial VoIP gateway |
| 06:52.31 | Qwell | implicit: It very well can be |
| 06:52.37 | implicit | Qwell, not really |
| 06:52.39 | Qwell | If you don't think so, you're simply a fool. |
| 06:52.43 | implicit | lol |
| 06:52.51 | j0 | oh the love in here is overwhelming |
| 06:52.58 | j0 | open source at its best :D |
| 06:53.03 | implicit | if your definition of carrier-grade is <500 simultaneous calls you are dumb as fuck |
| 06:53.06 | Qwell | j0: yeah, well, when people come to flame things, and offer little to no help |
| 06:53.08 | *** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net) |
| 06:53.09 | implicit | j0 open source is great |
| 06:53.13 | implicit | especially SER |
| 06:53.17 | Silik0n | Qwell i could be a carrier voip gateway but there are much better tools for that |
| 06:53.27 | jakepdev | j0 - why couldn't you have it connected full time? |
| 06:53.28 | Qwell | Silik0n: I never said there weren't |
| 06:53.32 | implicit | it can be a voip gateway |
| 06:53.35 | implicit | not a carrier-grade |
| 06:53.37 | implicit | voip gateway |
| 06:53.39 | jakepdev | and have calls go both ways? |
| 06:53.59 | jakepdev | like anormal pbx |
| 06:54.02 | j0 | jakepdev: trying to use a x100p as a backup for outgoing calls on a line that is occasionally in use |
| 06:55.05 | asiod | well at least it's working as my home answering machine |
| 06:55.19 | *** join/#asterisk three55ml (~three55ml@cpe-66-68-98-68.austin.res.rr.com) |
| 06:55.21 | implicit | Qwell, asterisk is good, for what it can do, but don't get an asterisk-hardon and think it is the end-all of voip |
| 06:55.33 | three55ml | Man, Trillian actually has nice IRC features :) |
| 06:55.44 | asiod | like what? i don't remember any |
| 06:55.53 | j0 | it has pretty colors.. i think |
| 06:56.02 | jakepdev | j0 - VOIP backup for PSTN? |
| 06:56.03 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 06:56.05 | three55ml | Not features, support |
| 06:56.16 | three55ml | I've never used it, just trying it now for the first time. |
| 06:56.18 | Silik0n | you know thats what scrollz and bitchx and ircii is for |
| 06:56.20 | j0 | jakepdev: pstn backup for voip |
| 06:56.32 | Silik0n | and add those to screen and got get a much better experience |
| 06:56.35 | jakepdev | ok |
| 06:56.45 | asiod | trillian v3 used to send all whois replies to the status window |
| 06:56.49 | *** join/#asterisk pif (ldm@zenon.apartia.fr) |
| 06:56.51 | jakepdev | j0 - then you could hook your analog phones up to * |
| 06:57.01 | three55ml | Silik0n: I used to use BitchX 7-8 years ago |
| 06:57.06 | jakepdev | hook the pstn side up to the x100p |
| 06:57.24 | jakepdev | and you'd have auto failover all the time |
| 06:57.42 | j0 | jakepdev: yeah, that would be the ideal way to do it... was just looking for a quick fix |
| 06:57.47 | Qwell | except the line is dual use |
| 06:58.13 | jakepdev | dual use? |
| 06:58.26 | jakepdev | i'm saying only the x100p should be hooked up to the PSTN |
| 06:58.35 | Qwell | yeah, thats how it should be |
| 06:58.36 | jakepdev | no other phones |
| 06:58.48 | jakepdev | what's not quick about that? |
| 06:59.05 | jakepdev | just need a good ATA - they're cheap enough |
| 06:59.13 | jakepdev | easy configs |
| 06:59.25 | j0 | i understand how to do that.. just wanted a different way of doing it |
| 06:59.44 | *** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net) |
| 07:02.12 | jakepdev | j0 - actually the Sipura 3000 looks like it would fit the bill perfectly and it's right around $100 |
| 07:02.30 | *** join/#asterisk lately (~dougb@chi.econ.usyd.edu.au) |
| 07:03.26 | Qwell | $10 says he won't like the idea. I've noticed that people are stubborn. Once they're set in their ways, theres no changing their opinion (bonus points for using they're, their, and there in the same sentence). |
| 07:03.42 | jakepdev | hehe |
| 07:03.46 | riksta | haha |
| 07:04.00 | riksta | it's actually there's :) |
| 07:04.01 | Qwell | unintentional... |
| 07:04.04 | Qwell | and, whatever |
| 07:04.08 | jakepdev | I could use the $10 to buy a g729 license |
| 07:04.10 | Qwell | at least I used them right. :P |
| 07:04.14 | riksta | ;) |
| 07:04.28 | lately | Can I make asterisk only allow SIP registrations on a particular network interface? Eg I want my home phone to register to my box which is on the same segment. But I don't want to allow people trying to register as my phone from external net. |
| 07:04.42 | riksta | firewall |
| 07:04.46 | Qwell | lately: passwords |
| 07:05.13 | Qwell | or, can't you set an IP= line, or something like that? |
| 07:05.15 | lately | Qwell: Sounds like a similar argument to allowing telnet shell access to a box. Passwords. |
| 07:05.22 | Qwell | lately: heh |
| 07:05.36 | Qwell | See, if you firewall SIP, you won't be able to contact remote providers |
| 07:05.46 | Zeeek | yes you will |
| 07:05.53 | riksta | yes you will you firewall incoming |
| 07:06.08 | Qwell | no, right...contact=register to |
| 07:06.12 | lately | Or people wont be able to call my * box using SIP. |
| 07:06.16 | Qwell | well, you'd BE able to register |
| 07:06.21 | jakepdev | i didn't notice that before - but the Sipura 3000 will bridge the FXS/FXO ports if it loses power. that is very nice |
| 07:06.24 | Qwell | but, no calls would come in...true? |
| 07:06.26 | riksta | with established related |
| 07:06.32 | Qwell | jakepdev: really? Thats kinda impressive |
| 07:07.06 | jakepdev | yep - i'm about to get one myself for the house. that was the one thing i was concerned about |
| 07:07.34 | Qwell | riksta: Then what if you wanted guest access to your box? |
| 07:07.44 | Qwell | where there would be no previously established connection |
| 07:08.01 | riksta | use iax2 :) |
| 07:08.01 | riksta | hehe |
| 07:08.08 | lately | Qwell: I've never seen the IP= bit... It is not in the default config examples |
| 07:08.10 | Qwell | Thats whats called a workaround. :p |
| 07:08.24 | Qwell | lately: I don't know. There is something though, right? |
| 07:08.40 | Qwell | host=? |
| 07:08.51 | riksta | brb, need a shower |
| 07:09.11 | lately | Qwell: ah, might be it |
| 07:09.25 | Qwell | probably an incoming only thing. heh |
| 07:09.35 | Qwell | I haven't RTFM'd, obviously |
| 07:10.03 | jakepdev | bindbindaddr |
| 07:10.07 | jakepdev | bindaddr |
| 07:10.24 | lately | jakepdev: Can't. I need it to listen to both NICs |
| 07:10.34 | jakepdev | that should do a particular interface - i think |
| 07:11.52 | Qwell | duh |
| 07:11.53 | Qwell | permit |
| 07:12.00 | Qwell | http://www.voip-info.org/wiki-Asterisk+sip+permit-deny-mask |
| 07:12.34 | Qwell | lately: That'll do exactly what you want |
| 07:12.55 | lately | Perfect! :-) |
| 07:13.06 | Qwell | probably want deny 0.0...blah, or whatever |
| 07:13.22 | Qwell | dunno, its got an example or two |
| 07:14.13 | lately | Pity I can't test this yet as I've killed ssh on my server. Gotta wait until I get home :-/ |
| 07:16.23 | jakepdev | #/join IRCAnonymous |
| 07:16.31 | techie | hmm |
| 07:18.17 | jakepdev | good night guys |
| 07:21.22 | *** part/#asterisk Curus (~Curus@83.72.32.8.ip.tele2adsl.dk) |
| 07:23.30 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 07:33.15 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l02m-6-210.d4.club-internet.fr) |
| 07:38.14 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
| 07:38.59 | *** join/#asterisk tld (~tld@80.203.70.227) |
| 07:39.07 | tld | Any Norwegians in here? |
| 07:42.08 | *** join/#asterisk pbxjunkie (~stormtroo@213.5.44.113) |
| 07:42.30 | pbxjunkie | howdy hey:D |
| 07:43.09 | tld | hey |
| 07:46.13 | pbxjunkie | what could be wrong if asterisk doesn't answer incoming calls on my zaptel channels? |
| 07:47.29 | poli_off | pbxjunkie, what do you read with asterisk -vvvvgc ? |
| 07:49.04 | pbxjunkie | eer.. what do I read? :D you mean the whole lot?:) |
| 07:49.05 | Zeeek | pbxjunkie what country? |
| 07:49.40 | pbxjunkie | greece, although I don't see why that's relevant |
| 07:50.09 | Zeeek | there's a big difference in how well the FXO work |
| 07:50.09 | Zeeek | but I guess you're saying that it was answering before? |
| 07:50.16 | smiley- | is there anything special I need to do in my dialplan to get it accept extensions with # from softphones? |
| 07:52.00 | pbxjunkie | it was answering my chan_capi incoming channels |
| 07:52.37 | slePP | who would've thought 'update res_cache set cust=cust+10000;' would take over half an hour to complete.. |
| 07:52.45 | *** join/#asterisk fenlander (~neils@82.152.81.57) |
| 07:54.20 | riksta | slePP: bleh!, you can't have that many customers! |
| 07:54.21 | riksta | :) |
| 07:54.35 | slePP | it's the results cache... so... 15,000 customers * 1000 results a piece |
| 07:54.40 | slePP | it really isn't that many records to update |
| 07:54.46 | slePP | but it's still going |
| 07:54.47 | riksta | nope |
| 07:55.08 | riksta | (btw i was joking) |
| 07:55.17 | slePP | i know :P |
| 07:55.31 | riksta | i know this ;) |
| 07:55.38 | riksta | but yeah i wouldn't have thought that it shud take anywhere near that, to compute |
| 07:55.42 | slePP | in fact, it only updated 104,156 records in that cache |
| 07:55.45 | slePP | and it took 38 minutes |
| 07:55.51 | riksta | something sounds hosed |
| 07:56.01 | slePP | it took 0.2s for the update to the customer list itself.. |
| 07:56.09 | slePP | well, i think postgresql just rewrote 3gb of data :> |
| 07:56.33 | riksta | sounds v inefficient to me :) |
| 07:56.50 | slePP | the records are quite big, so yeh.. |
| 07:56.57 | slePP | it's like.... cust, results, checksum, timestamp |
| 07:57.10 | slePP | results being a php serialized() array of possibly 30-80k of data |
| 07:57.32 | riksta | ah |
| 07:58.07 | riksta | even so, there's only 15,000 |
| 07:58.17 | riksta | oh wait i didn't read that properly |
| 07:58.21 | slePP | now, 104k records :> |
| 07:58.30 | slePP | all in all, it took a _really_ long time to accomplish very little |
| 07:58.42 | smiley- | the softphones seems to send %23 instead of # but I can't make an extension called %23 arghh |
| 07:59.10 | Zeeek | what softphone smiley? |
| 07:59.25 | slePP | oh nice... the new changes to the script are making it crash.... *sigh* |
| 07:59.54 | *** join/#asterisk makkia (~pippo@nat.xsec.it) |
| 08:00.16 | slePP | Length of String Invalid |
| 08:00.16 | slePP | god i love basic |
| 08:00.26 | smiley- | Zeeek: sjphone and x-lite |
| 08:00.49 | |Vulture| | anyone know if there is a help site like php.net for Perl? |
| 08:01.00 | Zeeek | X-Lite has always sent the # AFAIK using the key (not typing it which changes the proxy nulber) |
| 08:01.00 | *** part/#asterisk makkia (~pippo@nat.xsec.it) |
| 08:01.04 | Zeeek | perl.org |
| 08:01.07 | smiley- | my i3micro vood box sends # as # |
| 08:01.32 | smiley- | Zeeek: you speak of DTMF now? |
| 08:01.32 | |Vulture| | gracias |
| 08:02.27 | Zeeek | If I understood, you want to send the # as part of an extension, as in #0 ? |
| 08:02.35 | smiley- | yes |
| 08:03.47 | smiley- | *0 works fine but not #0 except from my hardware SIP-box that one sends # and not %23 |
| 08:03.50 | Zeeek | well I just dialed #22 and it seems to work |
| 08:04.00 | smiley- | :o |
| 08:05.46 | smiley- | Looking for %2322 in test |
| 08:05.46 | smiley- | SIP/2.0 404 Not Found |
| 08:06.05 | Zeeek | are you in alphanumeric dial mode? |
| 08:06.31 | Zeeek | I can't remember the name of thatparameter |
| 08:06.36 | smiley- | hmm.. |
| 08:06.40 | smiley- | in the client? |
| 08:06.47 | Zeeek | let's you dial alpha@sip.com |
| 08:06.51 | Zeeek | yes |
| 08:08.43 | pbxjunkie | anybody know his way around zaptel.conf? what does this line do? span=1,1,3,ccs,ami |
| 08:08.49 | |Vulture| | Anyone know how I would store a variable like this "print STDERR $astman->sendcommand(Action => 'Command', Command => "sip show peer $peer");" to be read? this is perl btw I tried my $test = blbla but it returns '66' |
| 08:09.29 | |Vulture| | pbxjunkie: appears to be the config for a channel bank |
| 08:10.31 | Silik0n | pbxjunkie: thats the line that defines a E1 channel i believe... |
| 08:10.43 | pbxjunkie | coz my card's readme tells me to have one line like that then the other span=2,0,3,ccs,ami , then span=3,0,3,ccs,ami and finally span=4,0,3,ccs,ami why is the second digit different in the rest? |
| 08:11.01 | smiley- | Zeeek: I'm checking.. |
| 08:11.07 | pbxjunkie | where can I read what each digit means? |
| 08:11.18 | Silik0n | look at the examples in the file I think it tells you in the comments |
| 08:11.28 | Silik0n | if not look at the sample config files its outlinked there |
| 08:12.03 | pbxjunkie | nope |
| 08:12.17 | pbxjunkie | sample config zaptel.conf? |
| 08:12.32 | Silik0n | yes |
| 08:12.33 | Silik0n | look in src/asterisk/confings |
| 08:12.33 | *** join/#asterisk ckruetze_ (ckruetze@cpc3-cmbg7-5-0-cust100.cmbg.cable.ntl.com) |
| 08:12.34 | Silik0n | err configs |
| 08:13.15 | smiley- | Zeeek: well.. I can dial whatever@blah.com and it shows up correctly |
| 08:13.25 | pbxjunkie | oh it has comments there, thanks :) |
| 08:13.29 | cypromis | e1 is ccs,hdb3 |
| 08:13.47 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
| 08:14.21 | Silik0n | that was a misfire |
| 08:14.30 | Silik0n | what is ccs,ami? |
| 08:14.37 | Silik0n | that soulds like a mismatched set |
| 08:14.44 | *** join/#asterisk RestLessGemini (RestLessGe@202.142.189.86) |
| 08:15.41 | Silik0n | esf.b8zs is what most people inthe states use... unless onthe occation you run across ami/d4 |
| 08:16.34 | pbxjunkie | i think it's like that coz it's not a real zaptel interface, it's a quadbri card |
| 08:17.15 | Zeeek | smiley look at advanced system settings and try changing dial alternative proxy to someting besides # and make sure letters to digit mode is right and all that |
| 08:19.53 | cypromis | ami/d4 is for ancient equippment |
| 08:19.53 | cypromis | ccs/ami sound like R2 |
| 08:19.53 | cypromis | although no R2 is cas |
| 08:20.03 | cypromis | rotfl |
| 08:20.19 | cypromis | pbxjunkie what stuff are you trying to use to break your quad again ? |
| 08:22.15 | pbxjunkie | it's not answering incoming calls |
| 08:22.17 | slePP | riksta: remember that update? now i'm doing a data sync to the cache... heh. it has 8,999 customers in this group to sync. it's on 77... i figure only 6 more hours to go |
| 08:22.17 | pbxjunkie | it makes outgoing calls allright, just doesn't answer incoming and i DO send the channels to default context |
| 08:22.17 | cypromis | are you sure you have your extensions set up correctly ? |
| 08:22.17 | slePP | Business Basic is such an annoying language.. |
| 08:22.17 | pbxjunkie | yesh, positive :D |
| 08:22.17 | cypromis | sure ? |
| 08:22.17 | Silik0n | slePP which version? heh |
| 08:22.17 | zigman | pbxjunkie whats the error you get ? |
| 08:22.17 | cypromis | are you sure you know how the telco sends you the numbers ? |
| 08:22.17 | Silik0n | (of course all business basic is annoying( |
| 08:22.17 | slePP | Silik0n: ProvideX 4.23 on this server |
| 08:22.19 | cypromis | did you try different pridialplans ? |
| 08:22.19 | Silik0n | hah |
| 08:22.19 | slePP | it may be 5 |
| 08:22.19 | Silik0n | slePP mas90? |
| 08:22.19 | pbxjunkie | I don't get error.. I get nothing at incoming calls |
| 08:22.19 | Silik0n | or something else |
| 08:22.19 | pbxjunkie | cypromis: pridialplans? |
| 08:22.19 | slePP | 5.01 |
| 08:22.19 | cypromis | man |
| 08:22.19 | cypromis | read the examples |
| 08:22.19 | slePP | mas90 rings a bell for some reason |
| 08:22.20 | cypromis | in bristuff |
| 08:22.20 | cypromis | they really help |
| 08:22.50 | Silik0n | slePP: thats cause best bought PVX just so they could keep it from being abandond cause theymake more money off mas90.mas200 which is written in pvx |
| 08:23.01 | pbxjunkie | hmmm... I'm pretty sure I've read everything there is to be read on the net about quadbri from jumper settings to Junghann's.net privacy policy.. EVERYTHING |
| 08:23.10 | cypromis | not on the net |
| 08:23.14 | cypromis | in the tarball there are samples |
| 08:23.16 | slePP | Silik0n: ah yes.k. not mas90. this is a program called 'SIMS' :> |
| 08:23.22 | slePP | written by an associate in 1975 or so |
| 08:23.22 | Silik0n | ok |
| 08:23.27 | Silik0n | heh |
| 08:23.36 | Silik0n | well if you need a good pvx programmer I know quite a few |
| 08:23.46 | Silik0n | including gui stuff |
| 08:24.10 | Silik0n | hah |
| 08:24.10 | Silik0n | no shit |
| 08:24.21 | slePP | event driven stuff just isn't the same when you are using a line numbered language |
| 08:24.27 | Silik0n | NOMADS Rulez |
| 08:24.42 | slePP | that's what rob says |
| 08:24.45 | Silik0n | heh |
| 08:24.50 | slePP | here's a good one for you |
| 08:24.50 | Silik0n | actually that gives me an idea |
| 08:25.04 | slePP | what is that thing to rekey an entire file from 4 char to 8 char keys? |
| 08:25.10 | Silik0n | i should write a UI for * in pvx/nomad for JavaX |
| 08:25.27 | timecop | fuck h323 |
| 08:25.30 | slePP | heh. that's not going to do a lot of good, since most people lack the PVX server side licensing they'd need to even run it |
| 08:25.32 | timecop | 2 fucking hours to compile |
| 08:25.35 | slePP | unless you hosted it |
| 08:25.35 | Silik0n | i forget i havent done pvx regulary in ages |
| 08:25.43 | timecop | this shit better work |
| 08:25.49 | pbxjunkie | what does "usecallingpres" do in zapata.conf?:) |
| 08:26.04 | pbxjunkie | never mind I'll look it up in voip-info :D |
| 08:26.08 | Silik0n | hah |
| 08:26.42 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 08:26.50 | Silik0n | slePP: i worked for a MAS90/200 "master developer" back around y2k havent really touched it since... |
| 08:26.52 | slePP | lucky you.... |
| 08:27.01 | timecop | fuck |
| 08:27.09 | Silik0n | altho that company still tries to get me to come back to work |
| 08:27.13 | slePP | and they just decided to change from 4 digit customer numbres (duh) to 8 digit |
| 08:27.14 | timecop | is there anything different I need to do for h323 otehr htan following the readme in h323 dir? |
| 08:27.45 | slePP | so i'm presently adjusting whacks of junk in 4 languages.. my week is really going to suck |
| 08:27.55 | *** join/#asterisk rajo (~rajo@scihparg.cs.uni-sb.de) |
| 08:28.07 | slePP | oh look. made it to customer 204 of 8999 |
| 08:28.09 | *** join/#asterisk maik (~maik@scumm.cs.uni-sb.de) |
| 08:28.53 | Silik0n | slePP it just means you'll be dealing with broken code for 3 more weeks |
| 08:29.00 | Silik0n | as they point out unforseen bugs |
| 08:29.00 | slePP | pretty much |
| 08:29.09 | slePP | 'Oh, we use the customer number there? What the hell for!?!?' |
| 08:29.21 | slePP | 'Who wrote this junk?! What is TT01$(125,7) supposed to be _anyway_!?' |
| 08:29.26 | *** join/#asterisk nrc (~username@zeus.eurotux.com) |
| 08:29.33 | Silik0n | hah |
| 08:29.41 | slePP | the guy who wrote this, never advanced (even to this day) beyond two character, two digit variable names |
| 08:29.46 | Silik0n | thats one thing I always loved about pvx... |
| 08:29.55 | Silik0n | 1 field in the table is really like 25 fields |
| 08:30.02 | slePP | heh |
| 08:30.09 | slePP | only if you did it that way :> |
| 08:30.42 | tessier | Anyone know anything about realtime related compile errors in the latest unstable cvs? |
| 08:30.44 | slePP | read(crappyfilehandle,key=cust$)$name,$number,$email,$fax |
| 08:30.44 | Silik0n | people do it that way all the time |
| 08:31.09 | slePP | i know. it's annoying :P |
| 08:31.10 | *** join/#asterisk RoyK (~roy@143.80-202-166.nextgentel.com) |
| 08:31.10 | Silik0n | try mas90/200 its like that all over the f'n plae |
| 08:31.11 | timecop | hello? does anyone use this shit? i just followed all the instructions in h323/README and it still DOESNT FUCKIGN WORK. says cant locate compatible codecs with remote end |
| 08:31.11 | slePP | that's why all my customers are limited to this whopping 16 characters for a description of a product |
| 08:31.14 | Silik0n | and then to even look at the code to figure it out you need a key that cose like $30K |
| 08:31.28 | slePP | to add more, it gets appended to like (130,30).. so you have to do (30,16)+(130,30) to get one :> |
| 08:31.50 | slePP | timecop: yer having a bad day... |
| 08:32.03 | slePP | timecop: what codecs is it trying? (the end device) |
| 08:32.03 | timecop | no REALLY? |
| 08:32.04 | slePP | and are you using a gk? |
| 08:32.04 | pbxjunkie | anybody know where I can get the default musiconhold mp3's that come with asterisk? for some reason they're not in my tarball :/ |
| 08:32.05 | timecop | slePP: its trying whatever |
| 08:32.06 | timecop | i had htem try 723, 711, gsm |
| 08:32.09 | timecop | none of the shit works |
| 08:32.16 | slePP | that's peculiar |
| 08:32.16 | timecop | 729too i guess |
| 08:32.21 | slePP | using chan_h323 or chan_oh323? |
| 08:32.36 | timecop | chan_h323 |
| 08:32.53 | timecop | -- Could not find common codec with 234324324 |
| 08:32.53 | slePP | Silik0n: once i get primary key updates done, i'm sending in the $250 programmer again. he's very slow, very annoying, and not very good.. but at least then i don't have to be annoyed with it |
| 08:32.53 | timecop | thats what I get |
| 08:32.57 | slePP | hmm |
| 08:33.02 | slePP | the h323.conf has the right codecs in it? |
| 08:33.02 | timecop | <PROTECTED> |
| 08:33.02 | timecop | <PROTECTED> |
| 08:33.02 | timecop | <PROTECTED> |
| 08:33.02 | timecop | <PROTECTED> |
| 08:33.02 | timecop | <PROTECTED> |
| 08:33.10 | slePP | at the end |
| 08:33.10 | timecop | ^^ for h323 |
| 08:33.10 | Silik0n | jesus chris... appearently i have a virus onmy system but the damned AV scanner stops it from running... and supposedly deletes it |
| 08:33.22 | slePP | "supposedly" being the key word |
| 08:33.41 | slePP | oh wait, h323.conf.. they're in the [general] |
| 08:33.41 | timecop | right |
| 08:33.43 | timecop | they're all fine |
| 08:33.54 | timecop | h.323 show codecs ^^ the above output |
| 08:33.59 | slePP | hmm |
| 08:34.07 | slePP | and you have 723? |
| 08:34.07 | Silik0n | h323 |
| 08:34.12 | timecop | duno |
| 08:34.17 | timecop | do i ? |
| 08:34.17 | tessier | timecop: h323 in asterisk is useless. Give up. |
| 08:34.17 | timecop | the remote device does. |
| 08:34.18 | slePP | very very likely not |
| 08:34.29 | tessier | timecop: And I'm not being sarcastic. I don't even bother anymore. I buy Cisco. |
| 08:34.29 | slePP | Dovid: show translation |
| 08:34.32 | timecop | tessier: well no fucking shit, if I didnt have to get it working I would have given up long fucking time ago |
| 08:34.32 | slePP | or translations.. forget if it has an s |
| 08:34.40 | timecop | tessier: tell this to the japs/chinks who are making me do this shit. |
| 08:34.42 | slePP | check the g723 column |
| 08:34.56 | tessier | timecop: No need to get racist now. |
| 08:35.12 | timecop | fine, i dont have 723 |
| 08:35.12 | timecop | but does it matter? |
| 08:36.57 | timecop | remote has been set to use 711 already anyway. |
| 08:36.57 | slePP | it might if they're trying to decide on that codec |
| 08:36.58 | timecop | more than once. |
| 08:36.58 | slePP | force your side into 711 as well |
| 08:36.58 | timecop | fine, its gone |
| 08:36.58 | timecop | die |
| 08:36.58 | timecop | die=did |
| 08:36.59 | slePP | disallow=all, allow=ulaw |
| 08:36.59 | timecop | right |
| 08:36.59 | timecop | did that. |
| 08:36.59 | slePP | then stop/start asterisk entirely (h323 is dumb) |
| 08:36.59 | slePP | and try.. |
| 08:36.59 | timecop | did that too |
| 08:37.00 | timecop | fucking shit |
| 08:37.01 | slePP | it is really worth the look |
| 08:37.11 | timecop | well i figured shit IN asterisk would be better |
| 08:37.12 | timecop | than random shit from elsewher |
| 08:37.32 | timecop | i bet im gonna hve to ugprade from jerJer versions of oh323/pwlib to the normal versions |
| 08:37.40 | timecop | which is another 2 FUCKING HOURS of recompiling |
| 08:37.42 | tessier | heh |
| 08:37.45 | slePP | i used the versions listed in the readme, and it works.. |
| 08:37.45 | tessier | Welcome to h323 |
| 08:37.49 | tessier | Bend over. |
| 08:37.59 | slePP | using gnugk in the middle as the gk |
| 08:38.08 | slePP | i've never done straight device -> asterisk |
| 08:38.10 | timecop | oh, I just used teh shit from the readme |
| 08:38.11 | timecop | and it doesnt work |
| 08:38.26 | pbxjunkie | cypromis: I hate it when you're right |
| 08:38.34 | pbxjunkie | cypromis: which is pretty much always |
| 08:39.55 | *** join/#asterisk ellvis (~ellvis@195.98.29.34) |
| 08:39.55 | ellvis | re |
| 08:39.56 | timecop | god fucking damn |
| 08:40.03 | timecop | of cource chan_oh323 doesnt work with jerjer libs. |
| 08:40.03 | timecop | fuck you |
| 08:40.09 | Silik0n | dood take a shot of wiskey or something |
| 08:40.22 | timecop | eh, i would, if I wasnt recompiling shitty C++ crap for hte last 8 hours. |
| 08:40.34 | slePP | you did compile make opt as the docs say? i'm guessing so. |
| 08:40.39 | timecop | of course |
| 08:40.39 | timecop | duh |
| 08:40.44 | timecop | it takes 2 fucking hours |
| 08:40.49 | timecop | for pwlib + oh |
| 08:41.19 | ellvis | anyone know where can be problem if i am getting "Raw Hangup" message in CLI? |
| 08:41.27 | ellvis | i just know it's IAX2 related, but dunno where's the problem |
| 08:43.04 | timecop | piec of shit. |
| 08:43.21 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 08:43.36 | timecop | ansy way to force codec order? |
| 08:43.36 | timecop | i gues it shouldnt fucking matter |
| 08:43.40 | timecop | I had all except 711u/a enabled |
| 08:43.41 | timecop | same shit |
| 08:44.08 | ellvis | Zeeek: hi, ellvis.vectorstar.net/asterisk.html |
| 08:44.25 | ellvis | Zeeek: adding every day a small part of it, hope it will be finished before the milenium will end :) |
| 08:44.57 | Zeeek | Huh? |
| 08:45.23 | Zeeek | ahhhh |
| 08:46.16 | Zeeek | no links to pages? |
| 08:47.27 | ellvis | Zeeek: not yet, i was drinking too much last saturday... |
| 08:47.52 | ellvis | Zeeek: but this week will improve |
| 08:50.23 | Silik0n | faxes come straight into the web ui via t37 ;) |
| 08:50.32 | Silik0n | misfire |
| 08:52.18 | Silik0n | faxes come straight into the web ui via t37 ;) |
| 08:53.23 | Silik0n | damn mouse |
| 08:54.20 | cypromis | drag&drop junkie |
| 08:57.12 | pbxjunkie | drag & drop? drag your pc from the power cable to the end of a cliff and drop it? |
| 08:57.28 | cypromis | pbxjunkie: he is not a pc man |
| 08:57.31 | cypromis | 0 points |
| 08:57.31 | cypromis | lol |
| 08:58.29 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.170.115.68.195.rev.coltfrance.com) |
| 09:01.01 | ellvis | :) |
| 09:02.21 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 09:09.58 | *** join/#asterisk Fraeggl (~Fraeggl@rkom.r-kom.de) |
| 09:12.45 | Zeeek | quiet day |
| 09:13.23 | Zeeek | so... I getting these interruptions in connectivity exactly every 8 minutes. |
| 09:13.40 | RoyK | it's that cron job |
| 09:14.20 | Zeeek | yes, but how did the cron job replicate itself in WIndows when I tried the other box? |
| 09:15.07 | RoyK | supavirus |
| 09:15.22 | Zeeek | must have been the farfon firmware :) |
| 09:15.30 | RoyK | prolly |
| 09:15.43 | RoyK | punjabi terrorism |
| 09:16.48 | timecop | oh well fuck this |
| 09:17.20 | *** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net) |
| 09:19.50 | *** join/#asterisk n4y (~frodo7@host-ip226-209.crowley.pl) |
| 09:20.05 | |Vulture| | will "pri show span 1" tell you if the span is actuall working, not just plugged in? |
| 09:21.21 | *** join/#asterisk tessier (~treed@203.210.209.79) |
| 09:24.54 | *** join/#asterisk zyke (~zakforeve@84.45.132.117) |
| 09:25.21 | *** join/#asterisk syle (~blah@wnpgmb02dc1-177-70.dynamic.mts.net) |
| 09:27.08 | pbxjunkie | can somebody point me to where the asterisk mohmp3's are to download? I can't find them anywhere :D |
| 09:27.13 | *** join/#asterisk emitrax (~mvillari@mdslab.unime.it) |
| 09:27.15 | emitrax | hi |
| 09:27.49 | emitrax | I'm trying to make a call with linphone to a cisco 7940 phone without setting up an account |
| 09:28.10 | emitrax | asterisk let me place the call but when I answer at the phone I dont hear anything |
| 09:28.34 | emitrax | I guess it's a codec problem ? |
| 09:28.36 | emitrax | ! |
| 09:28.53 | emitrax | does anyone know how to fix this ? |
| 09:31.05 | RoyK | er |
| 09:31.13 | RoyK | what proto does linphone use? |
| 09:32.46 | emitrax | sip I guess |
| 09:32.55 | emitrax | as cisco phone |
| 09:33.08 | emitrax | I changed the firmware on those phones |
| 09:33.08 | syle | sip and rtp |
| 09:33.18 | emitrax | linphone use rtp ? |
| 09:33.30 | syle | http://freshmeat.net/projects/linphone/ |
| 09:34.35 | emitrax | it uses stip |
| 09:34.38 | emitrax | sip* |
| 09:34.46 | emitrax | can it be a codec problem ? |
| 09:44.51 | emitrax | quit |
| 09:44.55 | *** join/#asterisk lters (~lters@mrtc-mm-600046.mis.net) |
| 09:45.07 | Zeeek | no one likes a quitter |
| 09:45.21 | Zeeek | fight! |
| 09:45.39 | ellvis | :) |
| 09:45.47 | Zeeek | punch that codec in the mouth |
| 09:45.54 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.170.115.68.195.rev.coltfrance.com) |
| 09:48.39 | |Vulture| | YAY! my perl script is coming along |
| 09:50.23 | |Vulture| | displays status of a PRI and SIP clients to Nagios |
| 09:51.44 | lters | nice |
| 09:52.50 | |Vulture| | thinking about putting in IAX too |
| 09:53.12 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.170.115.68.195.rev.coltfrance.com) |
| 09:53.59 | |Vulture| | actually... I could do registry but its too dependent on the array :( |
| 09:54.04 | Micc | are there issues with asterisk and using multiple broadvoice incoming sip lines? |
| 09:55.14 | |Vulture| | Micc: nope just make sure your running a 1.0.4+ and you will be fine |
| 09:55.51 | |Vulture| | why would you want multiple inbound BV? |
| 09:56.04 | |Vulture| | I could see outbound but not inbound |
| 09:56.59 | |Vulture| | I am soo tweaked on painkillers... |
| 09:57.29 | lters | Micc, seems like it *should* work. |
| 09:57.50 | *** join/#asterisk emitrax (~voip@ingnatdyn33.unime.it) |
| 09:57.55 | Zeeek | isn't there a post about broadvoice and multiple on the mailing list now? |
| 09:58.27 | |Vulture| | Ive been doing it for over 6 months... |
| 09:58.39 | |Vulture| | actually it will be a year in sept. |
| 09:58.39 | Micc | Vulture, I want to get an 800 number and sell voicemail boxes. |
| 09:59.33 | Micc | or something like that. |
| 09:59.33 | |Vulture| | Micc: get DIDs from another provider |
| 09:59.37 | |Vulture| | like VPC, Nufone... etc. |
| 09:59.39 | |Vulture| | Nufone offers 800 |
| 10:01.38 | Micc | It says they can't have any more customers. |
| 10:01.50 | *** join/#asterisk delYsid (~user@delYsid.developer.debian) |
| 10:02.05 | |Vulture| | email them they are accepting |
| 10:02.18 | |Vulture| | just their itnerface is down |
| 10:02.22 | |Vulture| | for what I hear |
| 10:03.43 | Micc | are they the cheapest? |
| 10:03.43 | Micc | or the best or both? |
| 10:09.09 | *** join/#asterisk saabluvr (root@keeper.nc-ks.de) |
| 10:10.13 | *** join/#asterisk saabluvr (master@keeper.nc-ks.de) |
| 10:14.17 | saabluvr | Hi all. I have astersisk 1.0.6, zaphfc and florz patch running ok. Only spandsp's rxfax hangs up right after starting rxfax ... |
| 10:14.26 | saabluvr | log : CLI> -- Executing RxFAX("Zap/1-1", "/home/master/testfax.tif") in new stack |
| 10:14.26 | saabluvr | <PROTECTED> |
| 10:14.26 | saabluvr | <PROTECTED> |
| 10:14.30 | saabluvr | <PROTECTED> |
| 10:14.52 | saabluvr | where should i start looking ? |
| 10:15.28 | saabluvr | I tried both spandsp 0.0.2 pre11 and pre15 |
| 10:22.14 | *** join/#asterisk hellop (~LeeHarvey@cpe-70-93-44-158.hawaii.res.rr.com) |
| 10:23.14 | hellop | Was having a debate with someone about the CPU requirements of VOIP. Question is, how many IP fones, FXO lines could a 200 or 300mgz box handle? I thought 2. |
| 10:23.56 | fenlander | Depends how long your piece of string is |
| 10:24.19 | fenlander | (it depends on many things) |
| 10:24.42 | hellop | Guestimate? |
| 10:26.07 | hellop | I'm doing an English research paper on Asterisk, also. |
| 10:26.26 | *** join/#asterisk GordonF (GordonF@rrba-146-87-139.telkomadsl.co.za) |
| 10:26.59 | ellvis | i was ust digging in maillists. i am getting message "raw hangup" for iax clients and as i was reading, it can be firewall issue. 'notransfer=yes' doesn't help. anyone with some experience? |
| 10:27.27 | hellop | Does anyone have any case examples of hardware bottlenecks? Like 4 IP phones, 4 FXO lines, all active, what's the min processor recommended? |
| 10:28.18 | fenlander | hellop: take a look at www.astertest.com |
| 10:29.29 | hellop | fenlander, sweet, thanks. |
| 10:31.50 | *** join/#asterisk kiokorobert (~kiokorobe@196.200.26.42) |
| 10:33.20 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.190.115.68.195.rev.coltfrance.com) |
| 10:33.45 | hellop | This is cool. I can fart around on the net looking at asterisk stuff, and feel like I'm getting schoolwork done. |
| 10:35.12 | |Vulture| | YAY! my nagios script for showing if peers are qualified works! |
| 10:40.50 | *** join/#asterisk Newbie___ (me@60.48.55.221) |
| 10:41.00 | hellop | fenlander, I read the whole powerpoint at astertest.com, and the forums where I found this: Speex: 5 calls |
| 10:41.01 | hellop | iLBC: 10 |
| 10:41.01 | hellop | g729A: 13 |
| 10:41.01 | hellop | g726: 22 |
| 10:41.01 | hellop | lpc10: 25 |
| 10:41.01 | hellop | gsm: 27 |
| 10:41.04 | hellop | alaw: 84 |
| 10:41.05 | hellop | sorry |
| 10:41.06 | Newbie___ | hi, anyone has experience working with TE410P and X101P ? |
| 10:41.19 | |Vulture| | TE110P |
| 10:41.20 | RoyK | jbot: lart hellop |
| 10:41.47 | Newbie___ | i just couldnt get X101P to dial, even * recognized the X101P |
| 10:41.53 | Newbie___ | TE410P is fine |
| 10:41.53 | |Vulture| | RoyK is also dislexic :P |
| 10:42.02 | hellop | Those are "max values" but what does that mean? |
| 10:42.24 | Newbie___ | RoyK: do u use X101P together with TE410P ? |
| 10:42.38 | hellop | Max those are not the phone protocols right? |
| 10:42.59 | RoyK | Newbie___: like last time, no, only te410p |
| 10:43.18 | RoyK | Newbie___: call digium and bitch them |
| 10:43.18 | |Vulture| | :P |
| 10:43.19 | |Vulture| | I only wish I could sleep |
| 10:43.20 | Newbie___ | heheh, oh ya u use digital |
| 10:43.28 | |Vulture| | I haven't slept in 2 days |
| 10:43.31 | Newbie___ | damn, i dont know what went wrong |
| 10:43.33 | |Vulture| | something is wrong with me |
| 10:43.51 | RoyK | |Vulture|: a pint of scotch usually helps |
| 10:43.51 | |Vulture| | ahhh sweet death! |
| 10:44.09 | hellop | Correct me if I am wrong, but those are not phone protocols, but VOIP service provider protocols, right? |
| 10:44.11 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.170.115.68.195.rev.coltfrance.com) |
| 10:44.11 | |Vulture| | RoyK: I am hyped up on ibprofren and tylenol right now |
| 10:44.26 | RoyK | Newbie___: does it work if you pull out the the te410p? |
| 10:44.35 | RoyK | |Vulture|: ibuprofen won't make you sleepy |
| 10:44.48 | *** join/#asterisk TheEmperor (~user@203.114.48.47) |
| 10:45.04 | Newbie___ | RoyK: cant pull TE410P out, is production box, but X101P worked on my other box |
| 10:45.14 | RoyK | wtf is tylenol? |
| 10:45.34 | TheEmperor | hi guys |
| 10:45.56 | Newbie___ | anyone has experience working with X101P |
| 10:46.37 | hellop | RoyK, tylenol is acetaminophen |
| 10:46.37 | |Vulture| | ~google tylenol |
| 10:46.37 | bugbot | google tylenol is assigned nothing and reported nothing. |
| 10:46.37 | |Vulture| | awww |
| 10:46.37 | TheEmperor | i'm using h.323, but am getting this error message |
| 10:46.37 | TheEmperor | Apr 18 18:40:54 WARNING[1272967872]: samples/codec_g729.c:217 g729tolin_framein: Received a G.729 frame that was 4 bytes from RTP |
| 10:46.37 | |Vulture| | painkillers |
| 10:46.37 | TheEmperor | anyone know what it means? |
| 10:46.37 | |Vulture| | for my neck |
| 10:46.37 | RoyK | tylenol is paracetamol, won't make you tired either |
| 10:46.37 | hellop | TheEmperor, h.323 is not recommended, see astertest.com |
| 10:46.38 | |Vulture| | RoyK: I know I am wide awake |
| 10:46.45 | |Vulture| | I took the pills cause my neck hurts like hell |
| 10:46.48 | TheEmperor | hellop: yes i know, but the provider i am dealing with only has h323 :( |
| 10:46.49 | RoyK | 400mg paracetamol and a bottle of whiskey will do |
| 10:47.11 | RoyK | :) |
| 10:47.43 | hellop | How does SIP differ from a provider's codec like H.323? |
| 10:47.43 | TheEmperor | so any ideas? :( |
| 10:47.54 | Newbie___ | i am damn sure, the context, zapata.conf and zaptel.conf is right, but just couldnt call out |
| 10:48.02 | RoyK | hellop: sip/h323/etc are protocols, not codecs |
| 10:48.22 | TheEmperor | RoyK: would you know what that error message means? When I call out on h323 I can't hear anything... |
| 10:48.27 | |Vulture| | RoyK: I did 3 ibuprofen and 2 tylenol extra strength and a beer and I felt kinda loopy... but then I learned Perl |
| 10:48.40 | hellop | RoyK, but what about Speex, gsm, alaw |
| 10:49.16 | kiokorobert | anyone with carrier grade setup? more than 1000 users? |
| 10:49.18 | |Vulture| | I use ilbc... best codec! |
| 10:49.27 | |Vulture| | sounds great and is free |
| 10:49.55 | kiokorobert | on asterisk alone? |
| 10:50.05 | RoyK | |Vulture|: g.711a has better sound :P |
| 10:51.09 | hellop | So, you get some protocols, SIP and H.323, and some codecs, and the CPU use comes from converting the codecs, right? Will just using the same codec/protocol on both sides of the server help? |
| 10:51.46 | |Vulture| | RoyK: well DUH :P hehhehe |
| 10:52.15 | TheEmperor | it's working now but very jittery.. |
| 10:52.16 | |Vulture| | people love ilbc sound though I did a test between gsm and ilbc... people notice |
| 10:52.36 | RoyK | because gsm sucks |
| 10:52.41 | TheEmperor | i wonder if it's the transcoding... |
| 10:52.56 | RoyK | g.711a really has better sound..... but then - it's 64kbps |
| 10:53.01 | |Vulture| | I haven't tried 729... you RoyK? |
| 10:53.27 | |Vulture| | Id consider 729 for my external phones |
| 10:53.35 | RoyK | well |
| 10:53.38 | RoyK | g.729 works |
| 10:53.39 | RoyK | but it's not free |
| 10:54.01 | RoyK | $10 per concurrent call iirc |
| 10:54.42 | delYsid | Does asterisk allow for caller id announcement? I'd like to pick up the phone, get a caller id announcement, and be able to reject/accept via dtmf |
| 10:54.42 | hellop | Can anyone confirm if I have the right understand about the codec options as per the previous post? |
| 10:55.04 | |Vulture| | RoyK: but $20 for that saved bandwidth... not bad |
| 10:55.13 | RoyK | delYsid: just make it :) asterisk doesn't support it out of the box, but you can hack it |
| 10:55.13 | |Vulture| | specially when you onlu have like 10 external users max |
| 10:55.34 | RoyK | then those $100 are well spent |
| 10:55.38 | RoyK | imho |
| 10:55.44 | RoyK | except if you can live with gsm |
| 10:55.47 | |Vulture| | well that would be $200 :P |
| 10:55.56 | |Vulture| | no gsm is horrid |
| 10:56.03 | RoyK | 10 * 10 = 100 last I checked |
| 10:56.16 | |Vulture| | $10 - phone;$10 - server |
| 10:56.23 | |Vulture| | you need 2 licenses to make the call |
| 10:56.45 | RoyK | |Vulture|: hardphones supporting g.729 already have the license |
| 10:56.50 | jeffik | vulture: why is gsm horrid? |
| 10:56.55 | pigpen | but...you don't need 2 licences for each call right? |
| 10:57.04 | RoyK | no |
| 10:57.21 | pigpen | so 3 simultanious calls via 1 server = 4 licences |
| 10:57.38 | RoyK | er |
| 10:57.41 | RoyK | no |
| 10:57.41 | |Vulture| | RoyK: oh really? did not know that |
| 10:58.05 | RoyK | |Vulture|: they pay the license to put g.729 into the box |
| 10:58.06 | pigpen | ok...please elaborate... |
| 10:58.16 | |Vulture| | gotchya... then I am gunna grab me some g729s |
| 10:58.18 | RoyK | you only have to worry about the asterisk codec_g729a license |
| 10:59.12 | pigpen | sure..the phones will already do it..if the licence exists on the box... |
| 10:59.38 | jeffik | |vulture|: what is the problem with gsm? I am trying to understand |
| 10:59.42 | pigpen | but when I load 4 licences on the * box...I could be doing 3 simultanious voice calls via 1 asterisk server... |
| 11:00.36 | |Vulture| | jeffik: the quality |
| 11:00.40 | |Vulture| | thats all I have with it |
| 11:00.49 | |Vulture| | I just prefer ilbc... personal pref. |
| 11:00.58 | |Vulture| | prolly prefer 729 when I test it out |
| 11:02.14 | jeffik | |vultuer|: ok just wondered, i also notice, using x-lite, when call a gsm mobile the gsm indicator comes on |
| 11:03.33 | |Vulture| | I guess you just get spoiled using 711 |
| 11:03.38 | TUplink | when did the wiki come back up? |
| 11:03.53 | |Vulture| | ~4am |
| 11:03.54 | bugbot | 4am is assigned nothing and reported nothing. |
| 11:03.58 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.190.115.68.195.rev.coltfrance.com) |
| 11:04.02 | |Vulture| | est |
| 11:04.07 | Newbie___ | anyone has experience working with X101P and TE410P |
| 11:06.11 | RoyK | Newbie___: tried the mailing lists? |
| 11:06.29 | RoyK | Newbie___: it might be even more efficient and less frustrating |
| 11:06.39 | Newbie___ | i did, it doesnt work too |
| 11:07.15 | RoyK | then dial digium and ask for support |
| 11:07.26 | RoyK | if it's digium hardware they should help |
| 11:07.32 | Newbie___ | RoyK: is not a digium card, do you think digium will help ? |
| 11:07.43 | RoyK | maybe if you pay them |
| 11:08.42 | Newbie___ | i bought the card for $10 and if they charge me at 60/hr, would be better if i buy digium card from them and get free support |
| 11:09.02 | RoyK | my point |
| 11:09.18 | RoyK | also, I think they charge at least $100/h |
| 11:09.31 | |Vulture| | you bought a TE410P but then got a $10 clone? |
| 11:09.49 | Newbie___ | |Vulture|: X101P is for testing purposes |
| 11:10.00 | Newbie___ | lol |
| 11:10.03 | RoyK | Newbie___: in a production server?? |
| 11:10.05 | TUplink | can i hook asterisk to my old PBX Lines? with the scsi like cable |
| 11:10.07 | |Vulture| | hahaha RoyK! omg |
| 11:10.11 | |Vulture| | you shoulda seen it |
| 11:10.17 | Newbie___ | RoyK: yes is a production server |
| 11:10.27 | RoyK | testing, in a production server |
| 11:10.30 | |Vulture| | some guys came in asking "How do I change my password in *@Home" |
| 11:10.31 | RoyK | is that wise? |
| 11:10.39 | Newbie___ | if the X101P work, i am gonna get TDM to connect to GSM fixed termanals |
| 11:10.42 | RoyK | |Vulture|: I've seen that |
| 11:10.46 | |Vulture| | I said "do "rm -rf /* :P" |
| 11:10.51 | |Vulture| | he did it! |
| 11:10.53 | |Vulture| | oh lol |
| 11:11.11 | RoyK | |Vulture|: evil :) |
| 11:11.18 | Newbie___ | RoyK: it was working fine on test set, bit somehoe fucked up in production |
| 11:11.18 | TUplink | |Vulture| thats great |
| 11:11.21 | |Vulture| | RoyK: I didn't think anyone would actually do it! |
| 11:11.37 | |Vulture| | but I justified it as one less AMP install |
| 11:12.18 | Newbie___ | it was working fine, X101P connected to GSM fixed termainal, without TE-410P though |
| 11:12.28 | TUplink | i installed the centos and asterisk after i couldnt get it to compile on freebsd... i didnt like hte web gui so i caned it |
| 11:12.33 | |Vulture| | Newbie___: when you have both it doesn't work? |
| 11:12.42 | Newbie___ | |Vulture|: yup |
| 11:12.51 | RoyK | Newbie___: STOP USING PRODUCTION SYSTEMS FOR TESTING |
| 11:12.53 | |Vulture| | Newbie___: lspci -vv check your IRQs |
| 11:13.05 | TUplink | <- gotta go to work see you guys later |
| 11:13.10 | |Vulture| | later TUplink |
| 11:13.17 | Newbie___ | mailaing list suggested using channels=125 even though i only use 2 span |
| 11:13.22 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.190.115.68.195.rev.coltfrance.com) |
| 11:13.46 | |Vulture| | Newbie___: does ztcfg -vv show errors when they are both in? |
| 11:14.17 | |Vulture| | I think Ive lost my mind |
| 11:14.40 | RoyK | Newbie___: dd if=/dev/zero of=/dev/hda |
| 11:14.43 | Newbie___ | ztcfg says all channels configured |
| 11:15.08 | |Vulture| | Newbie___: then its a zapata.conf issue |
| 11:15.23 | RoyK | Newbie___: pastebin the ztcfg -vvvvvvvv output |
| 11:15.33 | Newbie___ | ok |
| 11:15.43 | |Vulture| | but thats as far as I can go... never used a channel bank |
| 11:17.01 | Newbie___ | RoyK: http://pastebin.ca/9747 |
| 11:17.31 | |Vulture| | pretty nice looking |
| 11:18.12 | |Vulture| | Newbie___: what do you have 2 TE lines and 1 POT? |
| 11:18.36 | Newbie___ | |Vulture|: yes 2 E1s and 1 POTs |
| 11:19.29 | |Vulture| | just shit and giggles try "pri show span 1" |
| 11:19.35 | RoyK | Newbie___: then dial(zap/g3) |
| 11:20.03 | RoyK | after configuring the groups |
| 11:20.12 | RoyK | pastebin the zaptel.conf and zapata.conf as well |
| 11:20.14 | |Vulture| | RoyK: yea I was just about to say... did he post his conf |
| 11:20.35 | Newbie___ | exten => s,1,Dial(Zap/g0/${ARG1},30) |
| 11:20.40 | Newbie___ | it did not dial |
| 11:20.56 | |Vulture| | g0? |
| 11:21.01 | |Vulture| | you can have a group 0? |
| 11:21.12 | |Vulture| | try Zap/1/ |
| 11:22.12 | Newbie___ | with zap/1, * uses span 1 channel 1 to dial out |
| 11:22.27 | Newbie___ | http://pastebin.ca/9748 |
| 11:22.53 | Newbie___ | i tried g0, g5 the same result, wont dial out |
| 11:23.37 | |Vulture| | Newbie___: does it dial? |
| 11:24.25 | Newbie___ | no |
| 11:24.33 | Newbie___ | zap channel not found |
| 11:24.55 | Newbie___ | http://pastebin.ca/9749 i forgot to include zaptel.conf |
| 11:25.03 | |Vulture| | is the green light on the span? |
| 11:25.17 | Newbie___ | on TE410P, yes 2 green lights |
| 11:25.26 | |Vulture| | hmm strange |
| 11:25.39 | |Vulture| | and when you rip the XP out it works fine? |
| 11:25.50 | |Vulture| | strange |
| 11:26.13 | Newbie___ | zttools report all OK |
| 11:26.29 | Newbie___ | even span3 is ok with nothing plug in and no green light |
| 11:26.30 | *** join/#asterisk christo (~chris@office.enovi.com) |
| 11:26.32 | christo | aue |
| 11:27.18 | Newbie___ | i tred zap/125 and wont work either |
| 11:27.39 | Newbie___ | and did modprobe |
| 11:28.47 | Newbie___ | alternatively, i could have 2 * interconnect, when dialing to X101P, * 1 will connect to * 2 to make the call |
| 11:29.14 | Newbie___ | cant think of anything else |
| 11:38.29 | saabluvr | Hi ... is there an issue with spandsp not working on VIA-C3 ? Or florz zaphfc patch killing rxfax ? |
| 11:40.29 | saabluvr | on my zaphfc machine I get this : |
| 11:40.35 | saabluvr | CLI> -- Executing RxFAX("Zap/1-1", "/home/master/testfax.tif") in new stack |
| 11:40.39 | saabluvr | <PROTECTED> |
| 11:40.41 | saabluvr | <PROTECTED> |
| 11:40.44 | saabluvr | <PROTECTED> |
| 11:40.46 | saabluvr | <PROTECTED> |
| 11:40.49 | *** join/#asterisk bjohnson_ (~bjohnson@ip159-181.tor.istop.com) |
| 11:48.15 | delYsid | RoyK: ANy pointers to get started? |
| 11:49.39 | pigpen | anyone use iaxcomm? I can't seem to get it to connect to my * server...works with the digium test... |
| 11:54.42 | RoyK | ~docs |
| 11:54.55 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 11:54.55 | bugbot | docs is assigned nothing and reported nothing. |
| 11:54.55 | RoyK | jbot: docs |
| 11:54.56 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 12:06.24 | *** join/#asterisk Luhiwu (~marsosa@200.63.89.245) |
| 12:08.48 | *** join/#asterisk hcz (~hcz@82.78.168.102) |
| 12:09.14 | *** join/#asterisk Cybertoy (~Cybertoy@zux175-250.adsl.green.ch) |
| 12:09.28 | hcz | hi all |
| 12:10.36 | Cybertoy | hi all ... I'm a newbie to this so sorry for asking but I'm trying to run asterisk on a SPARC LX system with OpenBSD ... do you think the 70 MHz processor will be enough for it? |
| 12:11.43 | Cybertoy | uhm.. actually it's 50 MHz .. :) |
| 12:11.47 | PatrickDK | 70mhz might be enough for one call |
| 12:12.09 | PatrickDK | unless your just doing relaying and not any menu's or codec translations |
| 12:12.19 | Cybertoy | ok .. tnx... |
| 12:12.25 | Cybertoy | I'll look for better hardware then... |
| 12:15.14 | *** part/#asterisk hcz (~hcz@82.78.168.102) |
| 12:15.43 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 12:16.09 | *** join/#asterisk maik (~maik@scumm.cs.uni-sb.de) |
| 12:17.18 | delYsid | Hmm, skimming the docs seems not much help, is there a reference list of variables typically available? I am wondering where the caller ID actually is, ${CALLERID} ? |
| 12:17.24 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 12:19.50 | *** join/#asterisk clive- (~pirch@rndf-146-44-91.telkomadsl.co.za) |
| 12:20.40 | *** join/#asterisk wildgoose (~edward@xunil.mail.wildgooses.com) |
| 12:23.12 | *** join/#asterisk Marlow (~martin@cerberus.bluetree.ie) |
| 12:25.20 | *** join/#asterisk egon_l (~egon@pc-10-19-104-200.cm.vtr.net) |
| 12:27.02 | Zeeek | on the wiki |
| 12:27.09 | *** join/#asterisk Faithful (~Faithful@202-6-145-116.ip.adam.com.au) |
| 12:27.28 | Zeeek | <PROTECTED> |
| 12:27.46 | smiley- | Zeeek: I got it working btw.. the solution seems to be to setup an extension that catches %21 too |
| 12:27.55 | Zeeek | weird! |
| 12:29.29 | smiley- | SIPPS sends # my i3micro-box sends # sjphone and x-lite sends %21 ;) oh well.. I can live with two extensions |
| 12:29.31 | clive- | does anyone know what this means?: Apr 18 07:33:39 DEBUG[21907]: chan_sip.c:840 __sip_ack: Stopping retransmission on '4b81a96444adc068030973e2658ff272@66.225.202.80' of Request 102: Found |
| 12:29.37 | clive- | thousands of em |
| 12:30.29 | *** join/#asterisk tessier (~treed@210.245.96.88) |
| 12:31.39 | *** join/#asterisk DrWho17 (~MIKE@mike-new.tc3net.com) |
| 12:31.39 | ellvis | Request 102 is timeout |
| 12:32.26 | *** join/#asterisk _Brian (brian@unix01.voicenet.com) |
| 12:35.23 | _Brian | morning all... |
| 12:35.45 | *** join/#asterisk Romik (~romik@router-net.ser.netvision.net.il) |
| 12:36.26 | Romik | somebody can advice about this notice? Apr 18 13:31:02 NOTICE[24533]: channel.c:1314 ast_read: Dropping incompatible voice frame on IAX2/simpletelecom/4 of format speex since our native format has changed to ulaw |
| 12:36.27 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 12:36.51 | ellvis | Romik: codec mish-mash |
| 12:37.45 | Romik | ellvis: what do you mean? i have asterisk 1.05 and 1.07 connected...the 1.05 send speex, and 1.07 forward same channel by ulaw to voip provider...this is from 1.07 |
| 12:38.18 | clive- | ellvis thanks, timeout,,,I wonder why it times out |
| 12:39.08 | *** join/#asterisk JunK-Y (~grepmoo@65.39.228.5) |
| 12:39.28 | ellvis | Romik: does the dropping happen from begining or after some time? |
| 12:39.41 | bjohnson | ha .. thereaper.ca is available |
| 12:39.48 | Romik | ellvis: at he begginning |
| 12:39.53 | Romik | at the beggining |
| 12:41.07 | bjohnson | so is kissmyassterisk.ca |
| 12:41.12 | bjohnson | err |
| 12:41.16 | bjohnson | kissmyasterisk |
| 12:42.22 | ellvis | Romik: so, i don't have any experience with forwarding, but to me it look like second asterisk take the packets as incompatible, so you should check out the iax.conf |
| 12:43.21 | Romik | ellvis: i will make them as same version and will see |
| 12:43.35 | ellvis | Romik: how is the forward done itself? |
| 12:43.37 | ariel_ | Hello everyone |
| 12:44.20 | ellvis | hi ariel_ |
| 12:45.54 | ellvis | clive-: i have no idea why it's there, i am experiencing the same |
| 12:46.44 | delYsid | Hmm, if I have exten => 1500541,1,Macro(stdexten,666,SIP/666) inbound rings my cisco phone and goes to Voicemail afterwards as excepted, but if I do exten => 1500541,1,Answer and exten 1500541,2,SayDigits(${CALLERIDNUM}) I get only silence and a busy after a while. DO I need to do anything else except Answer to get the line up ? |
| 12:49.25 | PatrickDK | delysid, did ya try echo test? and it works? |
| 12:50.55 | _Brian | delYsid: what does it say it is doing on the console? |
| 12:50.59 | Romik | ellvis: just registering one to one and 2nd registering to voip provider |
| 12:51.17 | ellvis | Romik: understand now |
| 12:52.09 | clive- | thanks ellvis |
| 12:53.50 | ellvis | clive-: i am just damn newbie in all this :) |
| 12:55.22 | _Brian | ellvis: arent we all :) |
| 12:55.46 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 12:55.56 | *** part/#asterisk JunK-Y (~grepmoo@65.39.228.5) |
| 12:56.12 | *** join/#asterisk carbon60 (~adam@CPE000c41aab294-CM000f9fa6ba66.cpe.net.cable.rogers.com) |
| 12:56.21 | carbon60 | Good morning all. |
| 12:56.27 | carbon60 | Anyone using babyTEL accounts on-line? |
| 12:56.41 | carbon60 | They seem to have upgraded something that breaks Asterisk. |
| 12:58.14 | ariel_ | carbon60, what babyTEL |
| 12:58.31 | carbon60 | ariel_: A Canadian SIP provider. |
| 12:59.53 | ellvis | Romik: is it working or not? |
| 13:00.42 | Romik | ellvis: i need to get csv head and compile asterisk to check |
| 13:01.58 | ellvis | ah. okaj |
| 13:13.04 | mutilator | damn new premium roast coffee tasted like it's just a watered down version of their old stuff |
| 13:14.25 | ellvis | mutilator: take a brick with yourself during next visit :) |
| 13:14.47 | mutilator | :P |
| 13:15.14 | *** join/#asterisk zotz (~zotz@24.231.32.109) |
| 13:16.39 | _Brian | multilator: live up to your name...give them hell!! |
| 13:16.54 | _Brian | i am just glad your name wasnt "Postal" :) |
| 13:17.14 | _Brian | we would be seeing something on the 6 o'clock news.. |
| 13:17.39 | mutilator | nah i'de show up on americas most wanted |
| 13:18.58 | _Brian | that show still on? |
| 13:19.32 | mutilator | i'de assume |
| 13:19.39 | mutilator | it's helped catch people so why not |
| 13:19.44 | mutilator | cheaper than hiring police to look |
| 13:19.54 | *** join/#asterisk Pinhole (~nuxi@lin-dsl-static-206-222-194-115.inetnebr.com) |
| 13:19.59 | _Brian | heh...true...and they get to charge for the commericals... |
| 13:20.11 | _Brian | i wonder how many of the "actors" have been reported into police.... |
| 13:25.23 | *** join/#asterisk focks (~craig@nsc66.147.95-93.newsouth.net) |
| 13:25.41 | focks | anyone know what version of AMP is included in *@home 0.9? |
| 13:27.48 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 13:27.48 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 13:30.49 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
| 13:31.40 | *** join/#asterisk facek_ (faceoff@devel.acdbddh.eu.org) |
| 13:31.46 | *** join/#asterisk wildgoose (~edward@xunil.mail.wildgooses.com) |
| 13:35.41 | *** join/#asterisk TheEmperor (~user@218.111.48.1) |
| 13:41.29 | facek_ | Can someone help me with problem with ioctl |
| 13:41.42 | *** join/#asterisk jterrero- (~jt@66.28.34.162) |
| 13:41.43 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 13:55.03 | carbon60 | My provider has made a change that requires the dialed number to be in the SIP uri="" key of the Authorization header. Anyone ever heard of that? |
| 13:55.13 | JerJer | they suck |
| 13:55.29 | ellvis | :) |
| 14:02.45 | *** join/#asterisk moy (~kvirc@201.135.105.124) |
| 14:02.49 | facek_ | Can someonehelpm e |
| 14:02.57 | facek_ | my asterisk stop to answer incomfing calls in isdn |
| 14:03.00 | facek_ | at zaphfc card |
| 14:05.11 | *** join/#asterisk heison (~heison@ns.somanetworks.com) |
| 14:06.24 | *** join/#asterisk ChristianK (~Christian@p54A3E75B.dip.t-dialin.net) |
| 14:06.46 | *** join/#asterisk nvrswork (~RUR@cwn7.ads.uwaterloo.ca) |
| 14:08.33 | wildgoose | OK, pulling my hair out trying to make a TDP400P with FXO module hangup properly when the remote caller hangs up... Anyone any experience making this happen in the UK? |
| 14:09.08 | carbon60 | JerJer: Is that possible to accomodate? |
| 14:10.40 | wildgoose | facek_: I think look at the context the isdn card is in, and remove anything from there which answers the phone! |
| 14:12.35 | *** part/#asterisk saabluvr (master@keeper.nc-ks.de) |
| 14:12.38 | *** join/#asterisk malcolmd (~malcolmd@malcolmd.digium.sponsor.pdpc) |
| 14:13.55 | hellop | Is there a way to get better then .015 second resolution for a timer function? |
| 14:14.02 | hellop | I use ctime.h, and with clock() to get times, do (double)(end_time - start_time)/(double)CLK_TCK); to get elapsed time. |
| 14:14.49 | hellop | oops sorry wrong channel |
| 14:18.14 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:18.14 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:19.35 | facek_ | wildgoose i remove |
| 14:19.38 | facek_ | i have wery simple ocntext now |
| 14:19.44 | facek_ | exten => s,1,Answer |
| 14:19.52 | facek_ | exten => s,2,Dial(SIP/201) |
| 14:19.57 | facek_ | but asyterisk didn't answer calls |
| 14:21.46 | *** join/#asterisk _SMP_ (~SMP@pandora.burned.net) |
| 14:21.54 | Makenshi | facek_, are you trying to have asterisk answer calls to any number? |
| 14:23.08 | carbon60 | In the Authorization header of an INVITE, what is the uri suppose to be? |
| 14:23.08 | carbon60 | Authorization: Digest username="16134822676", realm="sip.babytel.ca", algorithm=MD5, uri="sip:hQ9-L-2117577269@216.18.125.7:5065", |
| 14:23.08 | facek_ | Makenshi yes |
| 14:23.52 | Makenshi | facek_, you need to use the extension "_." for this |
| 14:24.01 | Makenshi | eg, exten => _.,1,Answer |
| 14:24.22 | facek_ | oki |
| 14:24.29 | facek_ | in that way is stiill not answered |
| 14:24.33 | facek_ | and this is not byb dialplan |
| 14:24.35 | facek_ | its other problem |
| 14:30.58 | ariel_ | argh the wiki seems to still be down. |
| 14:31.07 | ariel_ | or at least very slow. |
| 14:31.22 | DrWho17 | ariel_: working here |
| 14:31.28 | DrWho17 | it's always slow though |
| 14:31.52 | ariel_ | it just displayed the first page it was there for almost 2 minutes before it displayed. |
| 14:32.32 | *** join/#asterisk wildgoose (~edward@xunil.mail.wildgooses.com) |
| 14:32.43 | DrWho17 | well, when I mean slow, it is latent, It will take about 3-5 seconds to load a page |
| 14:32.57 | Pinhole | The wiki *needs* a mirror. It's inaccessible quite often. |
| 14:33.02 | wildgoose | machine crashed. Did anyone have any comments on getting hangup to work with this TDM400p card? |
| 14:33.15 | ariel_ | DrWho17, it's works for 2 or 3 pages then on my system goes for ever. |
| 14:33.19 | DrWho17 | how much bandwidth is it on |
| 14:33.31 | DrWho17 | ariel_: that's not the case with me, I've been on it for quite sometime |
| 14:33.55 | *** join/#asterisk sonic74 (~Sven@pinguin.tdb.de) |
| 14:34.09 | *** join/#asterisk deRost (~derost@054.209-89-66-0.interbaun.com) |
| 14:34.58 | *** join/#asterisk MikeJ[Laptop] (~icechat5@mi.origenfinancial.com) |
| 14:39.32 | deRost | Im having troubles getting asterisk to communicate with my TDM12B. Dialing out always returns as congestion. Anyone have thoughts? |
| 14:40.58 | *** join/#asterisk carlos-d-man (~carlos@201.135.87.60) |
| 14:41.31 | *** join/#asterisk dzentai (~dzentai@ktv32-90-4.catv-pool.axelero.hu) |
| 14:41.39 | carlos-d-man | hi |
| 14:41.46 | deRost | I had it working beautifully the first day I got the card, and on a fresh install of AAH. I re-installed AAH in order to document the config changes, and it hasnt worked ever since. |
| 14:43.43 | dzentai | hi! |
| 14:44.24 | *** join/#asterisk dmccollum (~dmccollum@69-164-245-72.atlaga.adelphia.net) |
| 14:44.40 | carlos-d-man | hi there, I get this error, although I heve the configure included on iax.conf http://www.pastebin.com/272810 how may I fix it? |
| 14:44.53 | dzentai | where can I find information about regular expressions used in extensions.conf? I mean the syntax $[ "xy1234" : "regex" ] |
| 14:45.41 | devel | carlos-d-man, the error says 'diax' and the iax entry says 'dialx', is that a typo? |
| 14:46.14 | *** join/#asterisk skrusty (muad@217.79.111.73) |
| 14:46.14 | *** join/#asterisk Egonis (~chultay@69.194.211.129) |
| 14:46.18 | eKo1 | dzentai: either the wiki or the README.variables. |
| 14:46.19 | skrusty | anyone here compiled zaptel against linux2.6? |
| 14:46.24 | Egonis | Can anyone recommend a good gui for Asterisk? (easy to use/install) |
| 14:46.25 | skrusty | even though i have a symlink in /usr/src called linux-2.6 pointing to linux-2.6.9, it says the kernel source is missing |
| 14:46.46 | eKo1 | Egonis: vim |
| 14:46.55 | devel | skrusty, i just have 'linux' symlink'd to the source dir, with no problems. |
| 14:47.04 | Egonis | eKol: as in the text editor? |
| 14:47.04 | dzentai | I would like to match strings that contains 2 to 4 digits, but the perl like \d{2,4} sytanx doesnt work |
| 14:47.06 | skrusty | i have that too |
| 14:47.12 | skrusty | and going make linux26 |
| 14:47.21 | eKo1 | Egonis: yes |
| 14:47.26 | skrusty | but it still doesn't work :/ |
| 14:47.33 | Egonis | eKol: lol.. point taken |
| 14:47.45 | eKo1 | dzentai: eh, that doesn't use perl regex syntax |
| 14:47.58 | eKo1 | skrusty: just try linux |
| 14:48.18 | dzentai | hmm, bad news |
| 14:48.19 | *** join/#asterisk ckruetze (~nospam@cpc3-cmbg7-5-0-cust100.cmbg.cable.ntl.com) |
| 14:48.25 | skrusty | ok, will give it a go |
| 14:48.28 | devel | skrusty, you have /usr/src/linux/include in there? |
| 14:48.46 | devel | that is explicitly what the Makefile is looking for. |
| 14:48.54 | dzentai | then how could I do this in another way? |
| 14:48.59 | skrusty | yeah |
| 14:49.27 | devel | skrusty, you're sure your error is source/include location related then? |
| 14:50.04 | skrusty | well, not sure, i simply have the symlink created for linux, linux-2.6 and yet doing make or make linux26 still fails on finding the source |
| 14:50.23 | *** join/#asterisk Fddayan (~fddayan@66.240.80.130) |
| 14:50.31 | skrusty | i'v had a look through the makefile (not that im any good with them) just to see if there was anything i could spot that would suggest a problem |
| 14:50.36 | devel | skrusty, which distro? |
| 14:50.39 | skrusty | debian |
| 14:50.45 | skrusty | running 2.6.9 |
| 14:50.49 | eKo1 | dzentai: try [0-9]{2,4}. |
| 14:51.11 | skrusty | what is it looking for specifically in the include/ ? |
| 14:51.20 | devel | so the actual source dir that you're symlinking to should be /usr/src/kernel-headers-2.6.9-1-686 (or such) |
| 14:51.36 | dzentai | no the error message says that the syntax problem is with the {2,4} construct, not with \d |
| 14:51.42 | dzentai | but I will try |
| 14:51.44 | skrusty | it's /usr/src/linux-2.6.9/ because im not using a deb for kernel compilation |
| 14:51.53 | eKo1 | oh, try escaping the brackets, i.e. \{ |
| 14:52.05 | devel | ah. |
| 14:52.07 | skrusty | this isn't a debian kernel built |
| 14:52.15 | skrusty | just debian distro |
| 14:52.19 | skrusty | build |
| 14:52.47 | carlos-d-man | thanks devel, I no longet get that error message, but now I don't seem to get any errors nor the asterisk samples calling with dialx; how may I get asterisk to answer my voip call? :) |
| 14:53.33 | *** join/#asterisk pragueexpat (pragueexpa@r3p176.chello.upc.cz) |
| 14:53.38 | devel | skrusty, so /usr/src/linux/include should have the "normal" include stuff (dirs like asm,asm-i386,linux,net,scsi, etc) |
| 14:53.53 | dzentai | that doesn't help either, still the error message is: |
| 14:53.55 | dzentai | <PROTECTED> |
| 14:53.56 | *** join/#asterisk pycsusz (~pycsusz@pluto.euronetrt.hu) |
| 14:54.10 | skrusty | yeah it does |
| 14:54.23 | pragueexpat | Anyone have experience with E1 in Czech Republic? |
| 14:54.32 | devel | carlos-d-man, there is a valid dialplan in the [demo] context then? |
| 14:54.37 | dzentai | maybe I need to look into the sources to see what constructs are allowed |
| 14:54.45 | devel | skrusty, sorry, i can think of nothing else offhand |
| 14:54.46 | eKo1 | dzentai: I guess you'll have to do \d\d | \d\d\d | \d\d\d |
| 14:54.57 | skrusty | devel: cheers for the help! |
| 14:55.08 | dzentai | hmm, maybe... lets try it |
| 14:55.15 | pycsusz | Hi Everybody! If somebody has got Digium Wildcard TE405P card, then please help me!!!! |
| 14:55.29 | eKo1 | I have one. |
| 14:55.37 | eKo1 | But I am not using it. |
| 14:55.45 | pycsusz | ok |
| 14:55.57 | eKo1 | Since I don't have the lines for it yet. |
| 14:56.07 | eKo1 | Stupid telco.... |
| 14:56.14 | pycsusz | but did you use it? |
| 14:56.30 | carlos-d-man | devel yes, what about it? |
| 14:56.33 | dzentai | thanks eKo1! This seems to be working |
| 14:56.46 | eKo1 | pycsusz: How can I use it when it's not hooked up to anything. |
| 14:56.57 | pycsusz | ok sorry |
| 14:57.09 | *** join/#asterisk bannerman (~bannerman@209.216.176.42) |
| 14:57.10 | *** join/#asterisk unixgeek (~unixgeek@12.45.238.189) |
| 14:57.11 | devel | carlos-d-man, well, when it doesn't dial, that's usually been my problem. |
| 14:57.29 | *** join/#asterisk iq (~iq@207-224-101-250.omah.qwest.net) |
| 14:57.58 | carlos-d-man | devel so should I comment that? |
| 14:58.09 | pragueexpat | Anyone have a TE110P in Europe? |
| 14:58.39 | carlos-d-man | wtf is 216.207.245.47? |
| 14:59.01 | eKo1 | an IP of course |
| 14:59.14 | bannerman | my professional opinion is that it's an IP address. You can quote me on that, too. |
| 14:59.25 | Pinhole | not neccessarily, it could represent any 32 bit number |
| 14:59.37 | Pinhole | but it is commonly used for ip addresses. |
| 14:59.51 | devel | it was a rather vague question, carlos-d-man :) |
| 14:59.58 | carlos-d-man | dialx is now UNREACHABLE ...what does this mean? :S ...bannerman WOW hehe, so who's is it? |
| 15:00.12 | devel | but 'host 216.207.245.47' says it's x.digium.com |
| 15:00.41 | pycsusz | Hi Everybody! If somebody has got Digium Wildcard TE405P card, then please help me!!!! |
| 15:00.50 | devel | carlos-d-man, that means that your asterisk can't ping dialx now. |
| 15:01.46 | carlos-d-man | devel dialx is a dialx named softphone in win xp sp2, should asterisk be able to ping that? :S |
| 15:02.15 | devel | carlos-d-man, in your config, 'qualify=yes' means just that. |
| 15:02.16 | j0 | lol |
| 15:02.44 | carlos-d-man | oh, thanks hehe, so I don't need that do I? |
| 15:03.00 | devel | it's up to you. |
| 15:03.19 | ariel_ | pycsusz, what is your problem? |
| 15:03.22 | devel | i have it in my entries, then 'iax2 show peers' shows latency info |
| 15:04.02 | carlos-d-man | so how may I actually achieve my goal of using the asterisk demo? this little dialx softphone is taking a lot of time from me :S |
| 15:04.30 | *** join/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr) |
| 15:04.33 | eKo1 | Stop using softphones then. |
| 15:05.02 | carlos-d-man | heh |
| 15:05.14 | miller7 | anyone here familiar with TDMoE? |
| 15:05.16 | carlos-d-man | my only choice atm :S |
| 15:05.19 | devel | carlos-d-man, in your [demo] dialplan put an entry like 'exten => 86,1,VoicemailMain' to see if that fires up. |
| 15:06.01 | carlos-d-man | devel is that in iax.conf too? |
| 15:06.08 | ariel_ | carlos-d-man, what do you want to know about tdmoe |
| 15:06.25 | devel | no, that's in extensions.conf (where your dialplans are) |
| 15:06.35 | carlos-d-man | ok thanks... |
| 15:08.29 | carlos-d-man | theres' and externsion already in there at 8500 |
| 15:09.02 | *** part/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr) |
| 15:09.52 | *** join/#asterisk critch (critch@steven.basesys.com) |
| 15:09.53 | devel | ok, so what happens when you dial 8500? do you get any messages on the asterisk console? |
| 15:11.04 | Moc | hi mark |
| 15:12.37 | pragueexpat | Looking for some help with TE110P in Europe with E1 |
| 15:14.09 | carlos-d-man | devel I pressed on dials program 8500 and then on the call button, I only get silence |
| 15:14.44 | devel | carlos-d-man, are you running the asterisk console ('asterisk -r') ? |
| 15:15.02 | *** join/#asterisk eivindtr (~eivindtr@062016241059.customer.alfanett.no) |
| 15:15.03 | carlos-d-man | then I get a REDIAL button instead of DIAL button ...yes |
| 15:16.23 | Pinhole | Is there a *free* tool that can give me a rough estimate of VoIP quality? |
| 15:16.50 | *** join/#asterisk _Brian (brian@unix01.voicenet.com) |
| 15:17.07 | devel | carlos-d-man, so no data at all in the asterisk console? |
| 15:17.31 | critch | Pinhole: most people already have 2 already... ears work well.... |
| 15:17.34 | *** join/#asterisk malverian (~malverian@adsl-065-005-207-210.sip.gnv.bellsouth.net) |
| 15:17.36 | carlos-d-man | WARNING unable to send CAS ; WARNING ap_voicemail unable to read password ; rejected connect atempt request 1@demo does not exist |
| 15:17.48 | devel | carlos-d-man, did you hear any audio? |
| 15:17.54 | malverian | Hey guys, is there an easy way to import my configuration into mysql database? I want to try using AMP, and it reads configuration from there. |
| 15:18.03 | *** part/#asterisk sonic74 (~Sven@pinguin.tdb.de) |
| 15:18.09 | Pinhole | critch, I need an automated number. boss wants charts |
| 15:19.36 | Pinhole | I suppose I should recommend somebody run around and do the "can you hear me now?" thing. |
| 15:19.40 | *** join/#asterisk Strom_TM (~Strom_TM@office4.tmcs.net) |
| 15:19.42 | carlos-d-man | no audio at al ldevel, al I can hear is when I press on the numbers at the programs GUI :S |
| 15:19.55 | carlos-d-man | Pinhole hehe |
| 15:20.25 | devel | carlos-d-man, i would suspect something blocking the RTP then (that's been my problem with 90% of audio related issues) |
| 15:22.02 | *** join/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
| 15:22.27 | devel | carlos-d-man, shut down the XP firewall and see if it takes. |
| 15:25.27 | Seyr | im kinda new to VoIP and have setup an asterisk server to hand out VoIP calls to another computer. it doesnt seem to recognize DTMF. i've set the SIP config to disallow=all, allow=ulaw and allow=alaw (found some post that said to do that) and have added Background=(silence/10). Also have careinvite=no since I am behind NAT. anyone have any suggestions? |
| 15:26.21 | Delvar | in sip.conf - dtmfmode=rfc2833 ? |
| 15:26.22 | eKo1 | ever heard of dtmf mode |
| 15:26.33 | Seyr | call comes in from BroadVoice to the Asterisk box and then the Asterisk box hands it off to another server (a softphone) |
| 15:26.40 | Slainte | Seyr, do you have DTMF in your sip.conf? |
| 15:26.42 | devel | Seyr, i have all my dtmf settings to 'rfc2833' mode and have no issues. |
| 15:26.51 | Seyr | tried rfc2833 and inband |
| 15:26.58 | Delvar | try info |
| 15:27.08 | PatrickDK | what kind of phone do you have? |
| 15:27.12 | Delvar | they all need to match up wle you will lose the dtmf |
| 15:27.40 | PatrickDK | and what codec is it using? |
| 15:27.40 | Seyr | its a VoIP telephony interface manager |
| 15:27.40 | Seyr | SIP |
| 15:27.45 | PatrickDK | seyr, what softphone? |
| 15:28.02 | Seyr | its from VailSys |
| 15:28.18 | Seyr | it gives the call to Microsoft Speech Server |
| 15:28.40 | Strom_TM | and then it makes martinis for everyone |
| 15:28.46 | Seyr | yep |
| 15:29.04 | Seyr | they were out of the margarita model :-) |
| 15:29.17 | eKo1 | That's a lot of stuff happening...aren't you new to voip? |
| 15:29.23 | roamer323 | haha |
| 15:29.34 | Seyr | yeh, but im not new to linux or ms |
| 15:29.40 | PatrickDK | hmm it only does ulaw and rfc2833 |
| 15:29.49 | Strom_TM | yeah, Seyr...try it with a regular softphone and see if the same problem happens |
| 15:30.05 | *** part/#asterisk pragueexpat (pragueexpa@r3p176.chello.upc.cz) |
| 15:30.15 | Seyr | cant. the app i wrote is inside MSS |
| 15:30.26 | Strom_TM | no no...just to test |
| 15:30.33 | Strom_TM | just see if the tones even come through |
| 15:30.33 | PatrickDK | seyr, make sure you using ulaw and rfc2833 |
| 15:30.41 | Seyr | the app works fine using analog |
| 15:30.56 | Seyr | kk, gonna try rfc2833 again |
| 15:31.43 | Seyr | are there any good TTS engines for asterisk? |
| 15:31.54 | eKo1 | tts? |
| 15:32.01 | roamer323 | festival |
| 15:32.07 | roamer323 | ~festival |
| 15:32.08 | jbot | from memory, festival is a general multi-lingual speech synthesis system developed at CSTR. See http://www.cstr.ed.ac.uk/projects/festival/, or festival lite a much more compact festival http://www.speech.cs.cmu.edu/flite/index.html |
| 15:32.09 | bugbot | festival is assigned nothing and reported nothing. |
| 15:32.20 | eKo1 | you mean cepstral |
| 15:32.33 | carlos-d-man | devel I at last found out how to disable firewall, I get less error messages; I got rid of dialx and connected with firefly, -r told me it went through the voice mail process and I got no sound and the only warning was for 'cannot sent CAS' |
| 15:32.40 | Pinhole | yup, but I can spell swift (which is the binary name) |
| 15:33.05 | eKo1 | but doesn't app_cepstral suck |
| 15:33.29 | devel | carlos-d-man, do some packet sniffing, make sure the packets are going end to end. |
| 15:34.13 | carlos-d-man | :S packet sniffing? |
| 15:34.45 | Seyr | i only need to do a reload after i modify extensions.conf right? |
| 15:35.05 | devel | carlos-d-man, that's the easiest way you can make sure the packets are getting back to your client |
| 15:35.15 | carlos-d-man | Seyr theres a reload gracefully command :) |
| 15:35.43 | carlos-d-man | devel do you mind telling me how? I have never sniffed packets |
| 15:35.49 | dsfr | carlos-d-man, use ethereal, tethereal, or tcpdump for packet sniffing. |
| 15:35.52 | PoWeRKiLL | iaxtel is not working anymore ? |
| 15:36.04 | dsfr | Seyr, try "extensions reload". |
| 15:36.08 | Seyr | it still doesnt recognize DTMF using dtmfmode=rfc2833 and disallow=all,allow=ulaw |
| 15:36.34 | *** join/#asterisk drbraun (~reb@c187142.adsl.hansenet.de) |
| 15:36.41 | drbraun | Hi all |
| 15:37.35 | eKo1 | Seyr: try it with another softphone. |
| 15:38.33 | devel | carlos-d-man, the other alternative is to connect your computer to the same segment/subnet as your asterisk box. but you still need to make sure all firewalls are disabled in software. |
| 15:39.05 | Seyr | how could i test DTMF with another softphone? |
| 15:39.16 | *** join/#asterisk mogorman (~mogorman@207.111.174.1) |
| 15:39.18 | drbraun | or use IAX2 |
| 15:40.46 | *** join/#asterisk L|NUX (linux@202.5.131.104) |
| 15:41.45 | eKo1 | Seyr: By calling and pressing digits. |
| 15:41.51 | Seyr | The extension for my server is - type=peer, host=192.168.10.10, defaultip=192.168.10.10, disallow=all, allow=ulaw, dtfmode=rfc2833, context=sip, careinvite=no, insecure=very |
| 15:42.03 | Seyr | eKo1: i can hear digits now |
| 15:42.07 | Seyr | it just doesnt respond |
| 15:42.21 | eKo1 | 'it' being? |
| 15:42.38 | Strom_TM | Seyr, call some other phone and see if your digits are reaching the far end |
| 15:42.42 | Seyr | the speech server that asterisk sends the call to |
| 15:43.18 | eKo1 | maybe the speech server doesn't do rfc2833 |
| 15:43.32 | Seyr | eKo1: works fine using POTS line |
| 15:43.47 | eKo1 | pots doesn't use rfc2833 |
| 15:43.58 | Seyr | i have a dialogic 4 port analog card in it |
| 15:44.11 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 15:44.49 | Strom_TM | Seyr, keep it as consistent as possible. if the speech server is supposed to listen for rfc2833, call a softphone thats configured for rfc2833 |
| 15:45.11 | Strom_TM | isolate the problem |
| 15:45.53 | *** join/#asterisk tainted- (~ta_i_nted@65-60-70-242-cust.telepacific.net) |
| 15:46.06 | Seyr | ok, more testing then. thanks for all the help everyone |
| 15:46.34 | carlos-d-man | I reboot XP with no fw/av, fireup firefly sip phone and asterisk complains with this when I dial 8500 http://www.pastebin.com/272856 |
| 15:46.36 | Seyr | any suggestions on a good softphone i can try with? |
| 15:46.58 | drbraun | Seyr: Which OS? |
| 15:47.02 | Seyr | XP |
| 15:47.40 | *** join/#asterisk TEKjacob (~chris@c2.efb7d1.client.atlantech.net) |
| 15:47.41 | carlos-d-man | firefly |
| 15:48.11 | Seyr | i had to donate my x workstation to a developer :-( |
| 15:48.11 | Seyr | thanks |
| 15:48.11 | carlos-d-man | Seyr just dumped dialx for firefly |
| 15:48.11 | carlos-d-man | np |
| 15:48.24 | drbraun | Seyr: Firefly works fine here. Use IAX to get rid of all this SIP crap and the NAT issues |
| 15:48.31 | Seyr | i had xlite and something else on here |
| 15:48.41 | Seyr | the speech server has to take SIP |
| 15:48.58 | TEKjacob | Hey all, anyone have any recomendsations for the best ATA to use for connection a fax machine to Asterisk via SIP. Asterisk is PRI to the PSTN. |
| 15:49.14 | drbraun | Seyr: :) Firefly is still cool. Seems to be stable and easy to handle. Worked out of the box |
| 15:49.48 | eKo1 | TEKjacob: Pick any. Just make sure they use ulaw/alaw with no VAD. |
| 15:49.52 | Seyr | i could redo everything I did with MSS on Asterisk, but done dumped a ton of time into the MSS to get it to work.... and client expects a MSS server |
| 15:50.10 | Strom_TM | Seyr, what is the application exactly? |
| 15:50.21 | Seyr | all it is is a TTS app that reads from SQL and responds depending on the user ID |
| 15:50.24 | carlos-d-man | Seyr dl firefly "third party networks" |
| 15:50.50 | Strom_TM | Seyr, oh man, easy quick perl rewrite |
| 15:51.00 | Seyr | :-) |
| 15:51.01 | drbraun | Seyr: May be you should 'manage' your clients expectations :) Tell him about how technically crap SIP is in comparison to IAX1 |
| 15:51.22 | Seyr | i was brought in at the end and just told to make it work..... ya know how that goes |
| 15:51.25 | drbraun | Seyr: Ahh, okay |
| 15:51.36 | TEKjacob | drbraun: Thanks... Any personal thoughts on most reliable, easy to set up, etc. |
| 15:51.44 | eKo1 | I hat it when they say 'make it work'. |
| 15:51.50 | eKo1 | s/hat/hate |
| 15:51.55 | Seyr | had it up 100% with PSTN, then was advised to try VoIP |
| 15:52.04 | drbraun | even worse: They expect you to get it work :) |
| 15:52.20 | Seyr | im pretty close... if i can get by the DTMF issue, its done |
| 15:52.23 | drbraun | TEKjacob: Sorry, what do you mean? |
| 15:53.02 | smiley- | I use dtmf=inband that was the only way I could get dtmf to work with softclients and the hardware SIP-boxes I use |
| 15:53.18 | *** join/#asterisk kiokorobert (~kiokorobe@196.200.26.42) |
| 15:53.56 | *** join/#asterisk dasuberdavid (~david@207.111.174.1) |
| 15:54.24 | fenlander | Seyr: which version of * are you using? |
| 15:58.54 | *** join/#asterisk easimon (~easimon@localhorst.kawo2.RWTH-Aachen.DE) |
| 16:03.15 | Seyr|afk | fenlander: just downloaded it Monday, so whatever was up in CVS |
| 16:03.53 | Seyr|afk | CVS-v1-0-04/14/05-13:55:06 |
| 16:03.53 | Seyr|afk | fenlander: CVS-v1-0-04/14/05-13:55:06 |
| 16:04.18 | Seyr|afk | any of you think it may be my provider? using BroadVoice |
| 16:04.28 | *** join/#asterisk schlub (~jschulman@ppp-68-251-32-236.dsl.chcgil.ameritech.net) |
| 16:04.44 | Moc | finally I got wireless access with my i6310 hehe |
| 16:05.00 | Strom_TM | whoa whoa whoa, wait a second...why the hell are you bringing broadvoice into the equation? |
| 16:05.30 | Seyr|afk | thats who I have the VoIP number through |
| 16:06.01 | Strom_TM | so you originate the call from a regular telephone line? |
| 16:06.04 | tzanger | kram: *prod* privmsg |
| 16:06.49 | Seyr | Strom_TM: yeh |
| 16:07.04 | Strom_TM | ok...is broadvoice coming into the asterisk box via SIP or IAX? |
| 16:07.33 | DrWho17 | heh, I just hooked up with voipjet, easy that was, haven't went live with it though |
| 16:07.47 | Seyr | Regular Phone -> BroadVoice (SIP) -> Asterisk (SIP) -> Speech Server |
| 16:07.59 | *** join/#asterisk JunK-Y (~grepmoo@65.39.228.5) |
| 16:08.54 | Seyr | im open for suggestions |
| 16:09.14 | Strom_TM | when you call the speech server directly with a SIP phone, what happens? |
| 16:09.34 | Seyr | thats what im about to try with Firefly |
| 16:09.45 | Strom_TM | ok |
| 16:10.05 | Seyr | just wandering through the popups and ads trying to find the download link :-) |
| 16:13.21 | *** part/#asterisk ChristianK (~Christian@p54A3E75B.dip.t-dialin.net) |
| 16:14.45 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 16:14.58 | *** join/#asterisk loick (~loick@APuteaux-151-1-46-35.w82-124.abo.wanadoo.fr) |
| 16:15.32 | tainted- | anyone here use gafachi? |
| 16:15.53 | DrWho17 | I was looking at that today |
| 16:16.03 | DrWho17 | looked pretty nice and simple |
| 16:16.12 | tainted- | looking at what |
| 16:16.24 | DrWho17 | gafachi |
| 16:16.37 | tainted- | they aren't routing for some reason |
| 16:17.06 | DrWho17 | oh, sorry, voipjet had a free test account, I'm not currently routing through gafachi |
| 16:17.46 | Seyr | ack, gotta reboot. brb |
| 16:17.55 | *** part/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
| 16:18.39 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 16:21.10 | *** join/#asterisk mutilator (~animenodv@65.111.201.79) |
| 16:21.14 | *** join/#asterisk asteriskn00b (asteriskn0@wsip-68-15-113-233.ok.ok.cox.net) |
| 16:21.59 | asteriskn00b | has digium released ballpark pricing on the DS3000p? |
| 16:24.44 | LoRez | asteriskn00b: what part of OK are you in? |
| 16:25.35 | asteriskn00b | okc |
| 16:26.08 | asteriskn00b | and tulsa... |
| 16:26.08 | asteriskn00b | have offices in both |
| 16:26.22 | LoRez | cool. I'm in OKC. |
| 16:26.35 | easimon | is it normal, that digium wildcards seem ... hmm cheaply assembled? |
| 16:27.13 | bkw_ | asteriskn00b, i'm in Mcalester |
| 16:27.29 | asteriskn00b | wow I actually live in Henryetta =) |
| 16:27.57 | bkw_ | Strom_TM, WHERE THE HELL HAVE YOU BEEN? |
| 16:28.04 | *** join/#asterisk kiokorobert (~kiokorobe@196.200.26.42) |
| 16:28.12 | bkw_ | LoRez, OKC too? |
| 16:28.17 | bkw_ | you two need to come help me paint my house |
| 16:28.25 | *** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 16:28.27 | bkw_ | i'll buy the beer |
| 16:28.29 | Strom_TM | bkw_, I've been painting my apartment |
| 16:28.34 | LoRez | bkw_: hah... I don't drink the stuff |
| 16:28.41 | *** join/#asterisk genuix (~genuix@sobek.7g0.net) |
| 16:28.49 | bkw_ | ok water |
| 16:28.52 | bkw_ | :P |
| 16:29.15 | mishehu | bah. |
| 16:30.24 | *** join/#asterisk clint_ (~clint@snap.helixsystems.com) |
| 16:32.07 | asteriskn00b | lol |
| 16:33.27 | asteriskn00b | so what are yu two doing with asterisk, I am kind of new to the game, I currently sell and support Avaya IP Office, Altigen, and ESI Phone systems... however I am starting to have a lot of customers asking about Asterisk |
| 16:34.04 | Moonwick | oh, just takin' over the world. |
| 16:34.07 | Moonwick | nothing to see here. |
| 16:35.08 | asteriskn00b | !!! |
| 16:40.16 | *** join/#asterisk darwin35 (~darwin35@24.3.226.147) |
| 16:41.52 | eKo1 | I'm trying to call one of my DIDs (comming through SIP) and I keep getting: 407 Proxy Authentication Required |
| 16:42.21 | eKo1 | Looking at the headers, I see: To: <sip:17862324243@69.20.61.219:5060>. |
| 16:43.07 | eKo1 | Should I have an entry in sip.conf such as [17862324243] with host=69.20.61.219 ? |
| 16:44.21 | *** join/#asterisk Rob- (~robbie@haylott.plus.com) |
| 16:44.45 | *** join/#asterisk ruied (~a@213.22.166.175) |
| 16:44.48 | *** join/#asterisk jwitte (~jwitte_@firefly.alpha-lab.net) |
| 16:45.20 | |Vulture| | or host=dynamic |
| 16:45.20 | Dutts | can someone make a test call to my iax number please? |
| 16:45.28 | |Vulture| | Dutts: yea msg me |
| 16:46.27 | eKo1 | I tried all that and it just doesn't work. |
| 16:47.47 | |Vulture| | eKo1: pastebin your debug, and your sip entry |
| 16:48.16 | Dutts | sorry I meant iaxtel..... anyone out there with an iaxtel acc call me for a test? |
| 16:50.17 | Strom_TM | Dutts, i'll try |
| 16:51.06 | *** join/#asterisk roamer323 (~sing@HSE-MTL-ppp72413.qc.sympatico.ca) |
| 16:51.28 | *** join/#asterisk RogerW (roger69@roger-laptop.mcn.org) |
| 16:52.03 | eKo1 | Check this out: http://pastebin.ca/9761 |
| 16:52.28 | RogerW | Morning all |
| 16:53.17 | |Vulture| | morning |
| 16:53.45 | tzanger | will exten => 5551212/,1,... be executed for no callerID received (as opposed to 5551212,1,... which should work for everything else) ? |
| 16:53.50 | RogerW | Can anyone point me to a hardware vendor that sells Asterix boxes? |
| 16:54.00 | tzanger | or do I have to do some magic to check for a blank ${CIDNUM} |
| 16:55.34 | |Vulture| | eKo1: I am not exactly sure what your trying to do here... looks like you have 3 users trying to register but your [kayote-in] isn't one of them |
| 16:55.50 | *** join/#asterisk ckruetze (ckruetze@cpc3-cmbg7-5-0-cust100.cmbg.cable.ntl.com) |
| 16:56.28 | easimon | tzanger: its not too much magic... GotoIf($["${CIDNUM}"==""],withoutcid,withcid) |
| 16:56.31 | |Vulture| | tzanger: I didn't know you could do that... I use an if statment |
| 16:56.35 | tzanger | yeah |
| 16:56.52 | tzanger | I was gonna do a GotoIf(${LEN(${CIDNUM})... |
| 16:56.59 | tzanger | Zapateller has a nocallerid option though |
| 16:57.06 | |Vulture| | <PROTECTED> |
| 16:57.19 | |Vulture| | thats a working example |
| 16:58.08 | tzanger | |Vulture|: danke |
| 17:00.00 | darwin35 | has anyone put out a ully loaded extensions.conf file with every option mapped |
| 17:00.51 | darwin35 | ? |
| 17:02.51 | |Vulture| | darwin35: wow that would be a hell of a mess |
| 17:03.00 | |Vulture| | specially if it were for HEAD |
| 17:03.04 | eKo1 | How did yu determine 3? |
| 17:03.37 | |Vulture| | eKo1: in my CID example? |
| 17:04.13 | darwin35 | why if it mapped what is currently useable it would be great |
| 17:04.14 | eKo1 | No, from the stuff I posted in pastebin |
| 17:04.22 | darwin35 | I know new options are added all the time |
| 17:04.52 | *** join/#asterisk MikeJ[Laptop] (~icechat5@mi.origenfinancial.com) |
| 17:05.00 | *** join/#asterisk The_P (~The_P1@a82-92-24-18.adsl.xs4all.nl) |
| 17:05.11 | darwin35 | i am also still looking for a good web interface |
| 17:05.20 | |Vulture| | eKo1: users s,310 looks like 17862324243 might just be your voip provider |
| 17:05.35 | |Vulture| | eKo1: what does "sip show registry" show? |
| 17:06.03 | eKo1 | 310 is the phone I'm calling from on another * machine. |
| 17:06.20 | |Vulture| | whats the problem? |
| 17:06.21 | eKo1 | So I guess it's passing the CID to it. |
| 17:07.27 | eKo1 | The problem is that * is challenging the invites from my provider. |
| 17:07.39 | eKo1 | so the call never goes through |
| 17:07.50 | *** join/#asterisk barshad (kkhhaannuu@202.134.140.30) |
| 17:09.39 | |Vulture| | what provider? |
| 17:09.41 | *** join/#asterisk j0 (dan@S010600095b00a5aa.vc.shawcable.net) |
| 17:09.43 | *** join/#asterisk sonic74 (~Sven@pinguin.tdb.de) |
| 17:09.50 | eKo1 | Kayote Networks |
| 17:10.31 | |Vulture| | sorry don't know about them |
| 17:10.41 | *** join/#asterisk _SMP_ (~SMP@pandora.burned.net) |
| 17:11.12 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-40-134.cust.bezeqint.net) |
| 17:13.30 | *** join/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
| 17:14.00 | |Vulture| | http://www.janpro-fl.com/nagast.jpg |
| 17:14.03 | Seyr | Firefly should be able to dial to another softphone by IP correct? |
| 17:14.13 | *** part/#asterisk sonic74 (~Sven@pinguin.tdb.de) |
| 17:14.13 | Seyr | or dial into Asterisk by IP? |
| 17:14.22 | |Vulture| | YAY my nagios plugin works |
| 17:14.25 | |Vulture| | I can sleep! |
| 17:14.34 | Silik0n | its called dial by url in firefly and it works just fine |
| 17:14.41 | Seyr | not working for me |
| 17:14.45 | ManxPower | Seyr: I would assume so, but I doubt many people here use that feature. |
| 17:14.47 | Seyr | says the person is not available |
| 17:14.58 | ManxPower | One might think that checking the Firefly docs might be a good place to start. |
| 17:16.25 | Seyr | No clue Silik0n? |
| 17:17.35 | The_P | Hi all. Does any of you work with a Eicon Diva Pro 2.0 PCI card or a modem card with a Conexant chipset ? |
| 17:18.06 | |Vulture| | hahaha... I was like omg my plug is broke... apparently the power just went out to an office... not good |
| 17:18.11 | *** join/#asterisk luciusism (~kahngl@a3.d5b7d1.client.atlantech.net) |
| 17:19.28 | tzanger | hmm Zapateller() does not actually play anything |
| 17:19.31 | tzanger | it's supposed to play SIT |
| 17:22.07 | ManxPower | It plays the SIT for me. |
| 17:22.36 | tzanger | ManxPower: not for me |
| 17:22.44 | tzanger | calling a DID on my PRI from a cell phone (no CID shows up) |
| 17:22.45 | barshad | any one can help me setting up extensions.conf for sip ? |
| 17:22.50 | *** join/#asterisk oden (~oden@194-237-146-22.customer.telia.com) |
| 17:22.50 | tzanger | Zapateller executes but nothing is heard |
| 17:22.54 | tzanger | now I'm not answering but I shouldn't have to |
| 17:23.22 | ManxPower | *nod* Try an Answer() just in case. Should not make any difference. |
| 17:24.01 | *** join/#asterisk azher (azher@203.99.57.139) |
| 17:24.03 | tzanger | zapateller(answer) did it |
| 17:24.08 | tzanger | it shouldn' thave to though on PRI |
| 17:24.27 | azher | Hi |
| 17:25.03 | azher | anyone configured TE405P with Panasonic TDA100 PRI/E1 exchange |
| 17:25.19 | azher | i tried but my calls were getting dropped |
| 17:25.52 | Slainte | barshas what are you trying to do |
| 17:26.48 | Seyr | When calling an Asterisk server from my workstation using Firefly, all I get is "The person you are trying to reach is unavailable". Anyone have any clue? |
| 17:27.17 | barshad | <Slainte> exten => _.,1,Dial(SIP/${EXTEN:7}@vb2, 120, Ttr) |
| 17:27.17 | Nugget | the person you are trying to reach is unavailable. |
| 17:27.28 | darwin35 | your phone is not registering |
| 17:27.31 | barshad | i want to shift the call if failed on this, |
| 17:27.37 | Seyr | Nugget: I thought so myself at first |
| 17:27.42 | Seyr | :P |
| 17:27.50 | darwin35 | youe exten.conf is screwed |
| 17:27.59 | Seyr | thanks darwin35 |
| 17:28.10 | barshad | what is a rule for this ? |
| 17:28.17 | *** join/#asterisk mgth (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net) |
| 17:28.21 | *** join/#asterisk Wazb (Wazb@207.245.215.111) |
| 17:28.22 | Wazb | hi all |
| 17:28.27 | ManxPower | barshad: don't use spaces after commas |
| 17:28.41 | azher | any PRI guru out there .......... |
| 17:28.52 | *** join/#asterisk Egonis (~chultay@69.194.211.129) |
| 17:28.55 | barshad | ok |
| 17:29.08 | Wazb | any idea about simple and good SIP based softphone ? |
| 17:29.17 | Egonis | Newb Question: Fresh install of asterisk, working... two SIP Phones both capable of hearing the demo.. but I can't call from phone to phone |
| 17:29.57 | carlos-d-man | I am starting to suspect asterisk has taken over my sound card, I want it to be a server only, how may I tell it to let it go for my desktop clients to work again? |
| 17:30.06 | *** join/#asterisk ToyMan (~konversat@204-8-82-238.webjogger.net) |
| 17:30.44 | darwin35 | stop the module from loading in your modules.conf |
| 17:30.47 | |Vulture| | ariel: you guys experiencing a massive power outage down there? |
| 17:30.57 | darwin35 | noload alsa |
| 17:31.31 | darwin35 | where |
| 17:31.44 | darwin35 | is it being coverd on the news |
| 17:32.01 | barshad | [default] |
| 17:32.02 | barshad | exten => _.,1,Dial(SIP/${EXTEN:7}@vb2,120,Ttr) |
| 17:32.02 | barshad | exten => _.,102,Dial(SIP/${EXTEN:7}@vb3,120,Ttr) |
| 17:32.02 | barshad | exten => _.,202,Dial(SIP/${EXTEN:7}@vb4,120,Ttr) |
| 17:32.02 | barshad | ??????????? what to add here if all failed ????????????? |
| 17:32.02 | barshad | exten => h,1,Hangup |
| 17:32.17 | Slainte | use pastebin.ca barshad |
| 17:32.55 | barshad | thank you Slainte and sorry for this |
| 17:33.05 | darwin35 | rm the :7 |
| 17:33.09 | *** join/#asterisk trimi` (~Pharrell@62.162.232.143) |
| 17:33.16 | jterrero- | can someone help me out with this |
| 17:33.17 | darwin35 | that rm the 7 digits you dial |
| 17:33.18 | jterrero- | chan_sip.c:611 __sip_xmit: sip_xmit of 0x9cc1514 (len 657) to 192.168.254.200 returned -1: Invalid argument |
| 17:33.22 | jterrero- | why is that happening ? |
| 17:34.05 | trimi` | hello any1 know a goog IAX or SIP provider with good rates to PSTN line nufone ( this one didnt accept new orders when i tried to sign up ). please tell me only one with good cheap rates |
| 17:34.07 | ManxPower | barshad: See the stdexten macro in extensions.conf.sample in the Asterisk source. |
| 17:35.06 | trimi` | hello any1 know a goog IAX or SIP provider with good rates to PSTN line nufone ( this one didnt accept new orders when i tried to sign up ). please tell me only one with good cheap rates |
| 17:35.27 | *** join/#asterisk L|NUX (~linux@202.5.145.58) |
| 17:35.55 | barshad | ManxPower: was my syntax correct ? |
| 17:36.01 | Wazb | anyone know about good SIP softphone? |
| 17:36.20 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-199-208.dsl.scarlet.be) |
| 17:36.21 | barshad | Wazb: Mirial and Eyebeam |
| 17:36.32 | Silik0n | damn it man |
| 17:36.34 | *** join/#asterisk easimon (~easimon@localhorst.kawo2.RWTH-Aachen.DE) |
| 17:36.46 | Qwell | barshad: Don't use _. |
| 17:36.51 | critch | Wazb: knowing a bit about the SIP protocol, could there be a good SIP phone? |
| 17:37.07 | Wazb | thanks <barshad> |
| 17:37.11 | barshad | Qwell: then ??? |
| 17:37.18 | ManxPower | barshad: It's pretty obvious you are just copying other people's stuff without understanding the options you are using. |
| 17:37.19 | Qwell | _X., or something |
| 17:37.31 | Qwell | and tT is stupid... |
| 17:37.44 | Qwell | You should give me your number, so I can call you, and transfer myself places |
| 17:37.48 | ManxPower | Qwell: So is "r" |
| 17:38.10 | Qwell | ManxPower: r is more of an annoyance though |
| 17:38.14 | Qwell | tT could actually cause damage |
| 17:38.23 | ManxPower | Yes. |
| 17:38.34 | Qwell | but yeah, still stupid, nonetheless |
| 17:38.49 | Qwell | and 2 minutes of ringing...heh |
| 17:38.59 | ManxPower | Qwell: It can cause confusion when using analog ports expecially |
| 17:39.02 | Qwell | So, after 6 minutes, vb4 might get the call |
| 17:39.15 | Qwell | and EXTEN:7...jesus |
| 17:39.21 | Qwell | Is there ANYTHING right with those? |
| 17:39.30 | *** join/#asterisk anti (russ@anti.developer.gentoo) |
| 17:39.30 | ManxPower | Qwell: If there is, I can't see it. |
| 17:39.41 | Qwell | 1,102,202? |
| 17:39.48 | ManxPower | Looks like he's trying to do failover and not understanding anything about it. |
| 17:39.48 | darwin35 | what is vb4 |
| 17:39.51 | Qwell | bad math |
| 17:40.24 | darwin35 | you need to go back and read the extensions.conf in the wiki |
| 17:40.36 | Qwell | several times... |
| 17:42.16 | Qwell | trimi`: try talking to shido, he might be able to help you get a new account at nufone |
| 17:42.51 | Wazb | is there nay need to use STUN Server with Asterisk? |
| 17:42.55 | *** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com) |
| 17:42.59 | Wazb | for NAT clients |
| 17:45.12 | jterrero- | can someone help me out.. what does "-- Got SIP response 404 "Not Found" back from 192.168.254.200 |
| 17:45.23 | jterrero- | what is not found? what is it refering to |
| 17:45.27 | jterrero- | a context, user, etc? |
| 17:45.36 | *** part/#asterisk critch (critch@steven.basesys.com) |
| 17:45.44 | Gh0sty | is anyone running debian sarge for an asterisk? |
| 17:45.56 | Strom_TM | i am |
| 17:45.58 | Gh0sty | or is there a better distro to setup an asterisk *fast* |
| 17:46.11 | jterrero- | gentoo = godly |
| 17:46.13 | Gh0sty | i'm most familiar with debian :) |
| 17:46.21 | Strom_TM | debian works for me |
| 17:46.30 | Gh0sty | just apt-get the stuff? |
| 17:46.35 | Gh0sty | or build from source? |
| 17:46.47 | Strom_TM | apt-get will work, build from source is better though |
| 17:46.47 | poli | Gh0sty, I am... |
| 17:46.56 | poli | Gh0sty, Sarge's asterisk package is 1.0.5 |
| 17:47.09 | Gh0sty | and if i take unstable packages? |
| 17:47.19 | poli | Gh0sty, I compiled 1.0.7 by hand. But I am having some trouble with the init script. |
| 17:47.34 | Gh0sty | yeah, well trouble is something i can't afford :s |
| 17:47.34 | *** part/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
| 17:47.52 | Gh0sty | got a problem today, not sure how to fix it ... |
| 17:48.04 | poli | Gh0sty, Will probably be okay if you manage to install unstable |
| 17:48.12 | poli | Gh0sty, what is the problem? |
| 17:48.23 | Gh0sty | i'm doing my school work, 3 months working at a company for final school year |
| 17:48.45 | Gh0sty | now i went there to setup an asterisk |
| 17:49.29 | Gh0sty | first week we went to cebit, there my boss saw an asterisk with a VERY GOOD webinterface (unlike i could even imagine existed) sold by a company |
| 17:49.43 | Strom_TM | Gh0sty, why do you keep saying "an asterisk"? |
| 17:49.51 | Gh0sty | now he decided to sell that one in stead of letting me setup an asterisk from scratch |
| 17:49.58 | Gh0sty | well, a pbx ... |
| 17:49.59 | Strom_TM | "an asterisk box" yes, "an asterisk" sounds like you're setting up the typewritten character |
| 17:50.01 | Gh0sty | or a * |
| 17:50.13 | Gh0sty | a *-box :p |
| 17:50.14 | Strom_TM | or just "asterisk" |
| 17:50.19 | Strom_TM | but never "an asterisk" |
| 17:50.26 | Gh0sty | ok asterisk |
| 17:50.32 | Gh0sty | a box i meanth :) |
| 17:51.13 | Gh0sty | so now we're like 1.5 months later (i did some research on how to implement it in the network and stuff, qos on the firewall, some other stuff) |
| 17:51.25 | Gh0sty | and now finally the guys from cebit replied with a price list |
| 17:51.34 | Gh0sty | and my boss thinks its too expensive ... |
| 17:51.43 | Gh0sty | so now i'm facing a dilemma: |
| 17:51.57 | Qwell | Strom_TM: You put a network type diagram on your site, explaining where * fits in a solution... My boss found it on google, and it was able to explain things far better then I could. |
| 17:52.04 | *** join/#asterisk AndiC_UK (~Vlad@andicrook.demon.co.uk) |
| 17:52.10 | kiokorobert | help in changing the voicemail prompts |
| 17:52.20 | Gh0sty | either i setup asterisk, buy some fancy hardware stuff, or let some asterisk@home do the dirty work (just to have something to show ...) |
| 17:53.27 | AndiC_UK | this is my first time here i have a new asterisk box which as been running about a week with a call queque |
| 17:53.38 | Gh0sty | setting up asterisk would be my preferred path to follow, but i'm not sure if i can manage in 1 month and 10 days ... |
| 17:55.03 | Gh0sty | is there any webinterface better then the one found at asterisk@home ?? |
| 17:55.25 | Qwell | Gh0sty: yes, a php ssh session |
| 17:55.32 | carlos-d-man | I can't even setup my #$%#$&% sip phone :S |
| 17:55.43 | Gh0sty | something that can be managed by an end user? :s |
| 17:55.47 | asiod | wohoo! free did!! |
| 17:55.53 | asiod | !!!!!! |
| 17:55.57 | *** join/#asterisk phantasis (~phantasis@c68.112.204.241.eau.wi.charter.com) |
| 17:56.11 | phantasis | anyone work with an Carrier Access Adit 600 before? |
| 17:56.22 | AndiC_UK | i wish to have a extenshion for myself which calls my voip phone then tries a phone on a land line .. i can do this using a queque, howver, someone has told me POTs dont work with this method.. anyone varify this please ? |
| 17:57.16 | Strom_TM | Qwell: eh? |
| 17:57.25 | Qwell | Strom_TM: dunno, was a while ago |
| 17:57.29 | Strom_TM | ah ok |
| 17:57.40 | Qwell | it was a little diagram, explained what an ata was, where it went, how it connected, etc |
| 17:58.03 | ManxPower | otaku42: We need two CISCO831-K9 routers ASAP. If you have them in stock in the USA, please /msg me. |
| 17:58.10 | ManxPower | OffTopic: We need two CISCO831-K9 routers ASAP. If you have them in stock in the USA, please /msg me. |
| 17:58.16 | ManxPower | (stupid nick completion) |
| 17:58.30 | Strom_TM | odd, because im fairly sure I've never made a diagram like that |
| 17:58.50 | *** join/#asterisk pino (~z@host241-115.pool80116.interbusiness.it) |
| 17:59.08 | phantasis | anyone work with a Carrier Access Adit 600 channel bank before? |
| 17:59.50 | *** join/#asterisk mcnobody (~laaksola@laaksola.net) |
| 17:59.59 | AndiC_UK | sipura* |
| 18:00.59 | *** join/#asterisk zeedo (~zeedo@www.bsrf.org.uk) |
| 18:02.15 | *** join/#asterisk easimon (~easimon@localhorst.kawo2.RWTH-Aachen.DE) |
| 18:02.25 | *** join/#asterisk MikeJ[Laptop] (~icechat5@mi.origenfinancial.com) |
| 18:04.03 | AndiC_UK | anyone used cellsocket? |
| 18:04.57 | phantasis | why would a person use a channel bank for FXO? |
| 18:05.41 | kiokorobert | anyone implemented callback using asterisk? |
| 18:06.08 | AndiC_UK | kiokorobert> nope but i will be working on it soon |
| 18:06.36 | AndiC_UK | kiokorobert> i will have i quick look to see if i can find a script |
| 18:06.49 | smiley- | hmm.. I did just install firefly on my PC.. it's using 100% CPU and saying it can't connect to my asterisk |
| 18:06.51 | tzafrir_laptop | someone has edited-away most of the homepage of voip-info |
| 18:07.04 | tzafrir_laptop | Any simple way to revet to older version? |
| 18:07.42 | kiokorobert | i want to give a client the congestion signal and hangup |
| 18:07.46 | kiokorobert | but htis is not working |
| 18:07.49 | kiokorobert | exten => _072XXXXXXX,1,Congestion,5 |
| 18:07.50 | kiokorobert | exten => _072XXXXXXX,2,Hangup |
| 18:07.56 | AndiC_UK | kiokorobert> cos you need to use agi |
| 18:08.21 | kiokorobert | it gives the congestion signal |
| 18:08.26 | kiokorobert | but doesn't hangup |
| 18:08.36 | AndiC_UK | kiokorobert> http://www.voip-info.org/tiki-index.php?page=Asterisk%20perl%20agi |
| 18:09.27 | Qwell | tzafrir_laptop: I just fixed the front page... |
| 18:10.47 | kiokorobert | how about the voice prompts |
| 18:10.58 | kiokorobert | changing the voicemail prompts from comedian mail |
| 18:12.04 | ManxPower | kiokorobert: If you just hangup the caller should hear a congestion tone automagically |
| 18:12.41 | *** join/#asterisk goatmilk (~goatmilk@130-127-45-50.chouse.resnet.clemson.edu) |
| 18:12.45 | *** join/#asterisk barshad (kkhhaannuu@202.134.140.29) |
| 18:13.47 | Egonis | I am trying to make an exten => setting for extension 101, using the macro-stdexten, what are the arguments? I am using 'exten => 101,1,Macro(stdexten,101,101) |
| 18:14.05 | *** join/#asterisk bigeeTea (~icechat5@adsl-66-143-41-203.dsl.kscymo.swbell.net) |
| 18:14.29 | ManxPower | Egonis: I would have to look at the macro-stdexten to see what the arguments should be, but you can do that yourself. |
| 18:14.47 | bigeeTea | Does anyone have experience with Digium's IAXy device? |
| 18:15.17 | Egonis | ManxPower: They are extension and phone device... but how do I know which phone device it is? |
| 18:15.43 | Egonis | ManxPower: ${ARG1} = Extension, ${ARG2} = Device(s) to Ring |
| 18:16.06 | Qwell | whichever device you want to ring |
| 18:16.19 | *** join/#asterisk D|G|TAL (~grep@202.141.238.44) |
| 18:16.23 | Egonis | Qwell: But I don't know what the device is... too much of a newb |
| 18:16.30 | *** part/#asterisk D|G|TAL (~grep@202.141.238.44) |
| 18:16.35 | Qwell | figure that out, then start using macros |
| 18:16.41 | bigeeTea | Does anyone have experience with Digium's IAXy device? |
| 18:17.31 | ManxPower | Generally the device would be something like SIP/theentryfromsipconf |
| 18:18.22 | Egonis | ManxPower: ahh.. so [101]? as I set it.. but that setting does not work |
| 18:18.48 | ManxPower | You mean SIP/101 of course |
| 18:19.02 | Egonis | ManxPower: That would make more sense.. thank you |
| 18:19.49 | AndiC_UK | Does anyone have experience with a callsocket device |
| 18:20.15 | *** part/#asterisk Marlow (~martin@cerberus.bluetree.ie) |
| 18:21.42 | AndiC_UK | Egonis> 101,1,Macro(stdexten,101,SIP/101) |
| 18:21.57 | smiley- | ahh.. the error with firefly was that I had sjphone autostarted in the background |
| 18:22.19 | ManxPower | All SoftPhones Suck! |
| 18:22.51 | Qwell | ManxPower: pretty much |
| 18:23.07 | PTG1234 | ManxPower: your so biased |
| 18:23.19 | PTG1234 | Qwell: and you run linux how would you know :) |
| 18:23.23 | Qwell | :P |
| 18:23.26 | ManxPower | PTG1234: Just realistic |
| 18:23.49 | smiley- | ManxPower: sjphone on OSX with no skin works great |
| 18:23.49 | PTG1234 | XPRO runs great :) |
| 18:23.54 | PTG1234 | as long as you have a usb headset |
| 18:23.56 | Egonis | AndiC_UK: Worked, thank you!! |
| 18:23.56 | PTG1234 | never one issue |
| 18:24.00 | AndiC_UK | Egonis> or 101,1,Macro(stdexten,101,${ANDIC}) then under [globals] have ANDIC=SIP/101 iirc |
| 18:24.55 | bigeeTea | Has anyone used Digium's IAXy device to connect phones at branch office? -or- a second Asterisk PBX? |
| 18:25.14 | AndiC_UK | i use sjphone on a pocketpc based phone... however, i dont have a wifi adaptor yet :-/ |
| 18:25.23 | PTG1234 | i could never see the purpose of using an iaxy, its very expensive, and misisng features of very cheap sip devices |
| 18:25.52 | Qwell | PTG1234: nat is probably the big thing |
| 18:25.56 | AndiC_UK | XDA 2 that is |
| 18:25.58 | ManxPower | The biggest issue with SoftPhones is that they depend on your PC's hardware ands OS. Hardphones do not. |
| 18:26.09 | PTG1234 | there all no issues with sip nat, if you do it right |
| 18:26.10 | Qwell | I think it was JerJer saying he was at an airport or something, and plugged right in |
| 18:26.35 | PTG1234 | qwell: that will work with a normal sip device, as long as you set registrations to 30 seconds, which iax just does by default |
| 18:26.46 | bigeeTea | PTG1234: so then you have used a second Asterisk PBX to enable branch office? |
| 18:26.48 | DrWho17 | ManxPower: yea, they just depend on your routing, and QOS |
| 18:26.51 | Qwell | What does setting reg to 30 seconds do? |
| 18:27.16 | PTG1234 | the nat problems happen b/c the firewall stops routing traffic to your device, b/c it times out.. |
| 18:27.21 | Qwell | ahh |
| 18:27.24 | PTG1234 | so by setting the device to re-register, it keeps the firewall open |
| 18:27.27 | AndiC_UK | Right sip and nat you either have a sip proxy one end or use port forwarding limited to the one ip ofcourse |
| 18:27.40 | ManxPower | qualify=yes will also keep the nat translations open |
| 18:27.51 | ManxPower | And doesn't require you to change the client config |
| 18:28.06 | |Vulture| | yes, qualify=yes is your friend ;) |
| 18:28.15 | PTG1234 | qualify doesn't work in all circumstances, plus it retries phones disconnected etc.. its lame :) |
| 18:28.17 | nvrswork | ill 2nd that |
| 18:28.44 | |Vulture| | I like it because I can monitor the phone |
| 18:28.56 | |Vulture| | like this in nagios http://www.janpro-fl.com/nagast.jpg |
| 18:29.17 | PTG1234 | if i haven't registered a phone in 5 minutes, qualify shouldn't send any packets |
| 18:29.20 | AndiC_UK | i port forward to may asterisk box :-) |
| 18:29.27 | L|NUX | can some one tell me what is 6/6 billing ? |
| 18:29.38 | L|NUX | 30/6 billing ? |
| 18:30.03 | DrWho17 | LINUX: every 6 seconds |
| 18:30.05 | DrWho17 | ? |
| 18:30.13 | DrWho17 | instead of every 30 seconds |
| 18:30.15 | DrWho17 | ? |
| 18:30.21 | L|NUX | hmm |
| 18:30.35 | L|NUX | DrWho17 : can you tell me any site from which i can read about billing info ? |
| 18:30.36 | DrWho17 | you are billed in 6 second increments as opposed to 30 second bites |
| 18:31.34 | L|NUX | and what about 6/6 billing ? |
| 18:31.40 | L|NUX | 60/60 billing is? |
| 18:31.46 | AndiC_UK | i want to set up a sip proxy i was going to use SER but i dont need ser and astrisk anyone know of a good sip proxy |
| 18:31.49 | kiokorobert | hi guys |
| 18:32.14 | ManxPower | AndiC_UK: Most people do not need a SIP proxy |
| 18:32.31 | Strom_TM | L|NUX, 60/60 - initial period 60 seconds, additional period 60 seconds |
| 18:32.39 | *** join/#asterisk viLeR (1000@ip-47-252.telesat.com.co) |
| 18:32.44 | Strom_TM | 30/6 - initial 30 second period, 6 second increments thereafter |
| 18:32.47 | Strom_TM | etc etc etc |
| 18:32.54 | kiokorobert | to use the digium t1 cards, must i use a channel bank? |
| 18:33.14 | L|NUX | Strom_TM : can you tell me site from which i can read about it ? |
| 18:33.18 | Strom_TM | kiokorobert, you could just plug a T1 into the thing too :) |
| 18:33.27 | Strom_TM | L|NUX, what more do you need to know? |
| 18:33.40 | DrWho17 | kiokorobert: no |
| 18:34.25 | kiokorobert | how do i go about it? |
| 18:34.29 | AndiC_UK | ManxPower> i think i may sometime |
| 18:34.35 | kiokorobert | without a channel bank |
| 18:34.41 | L|NUX | Strom_TM : 30/60 means that i get initial period is 30 seconds, and additional period 60 seconds |
| 18:34.45 | DrWho17 | kiokorobert: get your DS0's delivered via T1 |
| 18:34.48 | _Brian | cool...help is here!!! |
| 18:34.53 | _Brian | :) |
| 18:35.12 | ChkDigit | kiokorobert: buy T1 service from a supplier, and insert cable into T1 jack. |
| 18:35.16 | schlub | has anyone successfully implemented lcrdial (http://ykoz.net/intl/lcr/)? |
| 18:35.30 | *** join/#asterisk riksta (~rick@81-178-248-131.dsl.pipex.com) |
| 18:35.56 | Strom_TM | L|NUX, yes |
| 18:36.25 | Strom_TM | ive never actually seen it done with an initial period smaller than the additional period |
| 18:36.41 | *** join/#asterisk iq (~iq@207-224-101-250.omah.qwest.net) |
| 18:37.09 | *** join/#asterisk blop (~blop@2001:6f8:204:33:bbbb:bbbb:bbbb:bbbb) |
| 18:37.48 | DrWho17 | schlub: was looking at it and the rate-engine addon yesterday, didn't implement it yet though |
| 18:37.49 | bjohnson | kiokorobert: a T1 can be your voice service from your telco, your data connection from your ISP, your connection to a channel bank, nd/or your connection to other hardware |
| 18:38.17 | *** join/#asterisk harryvv (~leonardo@S010600055d210201.vs.shawcable.net) |
| 18:38.40 | schlub | DrWho17: just not working for me. AGI returns 0, but never dials out. Without any other debugging info, I can't figure out what's wrong. |
| 18:38.40 | phantasis | anyone familiar with an Adit 600? |
| 18:38.59 | harryvv | linux file system types like Reiser should not have a negative impact on asterisk? |
| 18:39.09 | _Brian | does anyone have any operational examples of the application command 'while' |
| 18:39.27 | bjohnson | phantasis: yes .. but not me |
| 18:39.36 | bjohnson | phantasis: I think tzanger has one |
| 18:39.39 | AndiC_UK | ManxPower> i thout sip + nat = headache++ |
| 18:39.44 | AndiC_UK | thought* |
| 18:39.52 | DrWho17 | harryvv: no |
| 18:40.15 | DrWho17 | I use reiserfs on all my asterisk boxes with no negative impact that I have seen |
| 18:40.24 | bjohnson | harryvv: maybe in high load systems |
| 18:40.25 | harryvv | okay |
| 18:40.34 | _Brian | forget it..found it :) |
| 18:40.35 | *** join/#asterisk gein (~gein@213.134.110.241) |
| 18:40.35 | *** join/#asterisk kore (kore@mindwipe.org) [NETSPLIT VICTIM] |
| 18:41.07 | *** join/#asterisk n4y (~tmalkut@fire2.orasoft.net.pl) |
| 18:41.31 | harryvv | bj, its only going to run asterisk and thats about it. |
| 18:41.33 | ManxPower | AndiC_UK: Only if you have large numbers of SIP clients behind NAT that need to talk to EACH OTHER. |
| 18:42.44 | PTG1234 | even with a large number itg works fine |
| 18:42.51 | DrWho17 | schlub: yea, well I'll look into it further, asterisk's billing engine really is one of it's weakpoints, well at least compared to a switch |
| 18:42.52 | PTG1234 | as long as you open up the port range the nat devices can use |
| 18:44.25 | *** part/#asterisk JunK-Y (~grepmoo@65.39.228.5) |
| 18:46.54 | *** join/#asterisk bigeeTea (~icechat5@adsl-66-143-41-203.dsl.kscymo.swbell.net) |
| 18:48.39 | *** join/#asterisk twisted[work] (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) [NETSPLIT VICTIM] |
| 18:48.39 | *** join/#asterisk FengShui (~ted@gray.impulse.net) [NETSPLIT VICTIM] |
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| 18:48.39 | *** join/#asterisk zeedo (~zeedo@www.bsrf.org.uk) [NETSPLIT VICTIM] |
| 18:48.39 | *** mode/#asterisk [+o twisted[work]] by irc.freenode.net |
| 18:48.41 | *** join/#asterisk BoRiS (~boris@wnpgmb01dc2-25-225.dynamic.mts.net) |
| 18:49.15 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 18:49.56 | AndiC_UK | ManxPower> thank .. i think i may (offices) how i may just tunnel them .. iirc we tested it and got one way audio .... however, i will look a sip proxys |
| 18:50.00 | AndiC_UK | s* |
| 18:50.11 | AndiC_UK | n* |
| 18:50.37 | Egonis | where is a good resource for hold music? |
| 18:50.45 | *** join/#asterisk allyour80211b (~allyour80@208.178.154.99) |
| 18:50.54 | dmccollum | Barry White is always a good choice for hold music. :) |
| 18:51.00 | Egonis | lol! |
| 18:51.02 | Egonis | awww yeah |
| 18:51.03 | harryvv | no U2 is :) |
| 18:51.04 | nvrswork | I like a little Abba |
| 18:51.22 | Egonis | I thought I saw a download site specifically for hold music |
| 18:51.22 | nvrswork | Enya is good too |
| 18:51.24 | nvrswork | soothing |
| 18:51.26 | AndiC_UK | Egonis> yeah good question ... should be royalty free ofcourse :P |
| 18:51.27 | dmccollum | Get the ladies all in the mood before you send them over to tech support. |
| 18:51.34 | harryvv | hehehe |
| 18:51.37 | harryvv | tom jones |
| 18:51.54 | Egonis | AndiC_UK: the site was referred by slashdot... it had a lot of royalty free tunes, and some for purchase |
| 18:52.14 | bigeeTea | has anyone setup branch offices with Asterisk? |
| 18:52.39 | AndiC_UK | bigeeTea> doing now |
| 18:52.50 | AndiC_UK | Egonis> got a url? |
| 18:53.13 | bigeeTea | AndiC_UK: did you use IAXy? |
| 18:53.15 | harryvv | AndiC how are you laying it out |
| 18:54.09 | *** join/#asterisk twisted[work] (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) [NETSPLIT VICTIM] |
| 18:54.09 | *** join/#asterisk zeedo (~zeedo@www.bsrf.org.uk) [NETSPLIT VICTIM] |
| 18:54.09 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) [NETSPLIT VICTIM] |
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| 18:54.09 | *** join/#asterisk A-Tuin|work (~A-Tuin@nat.office.legend.net.uk) [NETSPLIT VICTIM] |
| 18:54.09 | *** join/#asterisk FengShui (~ted@gray.impulse.net) [NETSPLIT VICTIM] |
| 18:54.09 | *** mode/#asterisk [+o twisted[work]] by irc.freenode.net |
| 18:54.11 | asteriskn00b | anyone familiar with the zoomerang functionality of the Altigen Altiserver, can asterisk replicate this functionality? Basicly allow a user to go into voicemail, listen to a voicemail, press one button to return the call, and after the call is finished, return the user back to his voicemail so he can goto the next one |
| 18:54.30 | AndiC_UK | bigeeTea> <harryvv> i was going to use a sip proxy, however, i may use IAX trunking |
| 18:54.38 | bigeeTea | AndiC_UK: did you use IAXy? |
| 18:54.43 | bigeeTea | sorry |
| 18:54.49 | harryvv | sip can be a nightmare to router though some routers. |
| 18:55.17 | AndiC_UK | bigeeTea> IAX a IAXy is a IAX based ATA |
| 18:55.19 | harryvv | Its actually best to route sip * iax router net router and back again. |
| 18:55.58 | AndiC_UK | bigeeTea> <harryvv> i was going to link asterisk boxes via IAX trunking |
| 18:56.13 | bigeeTea | AndiC_UK> that's what I was considering as well |
| 18:56.21 | AndiC_UK | bigeeTea> <harryvv> yes phones will be sip |
| 18:56.27 | harryvv | only problem is the remote end would need a asterisk box to do that or just replace the router that is the issue. |
| 18:56.44 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 18:58.09 | AndiC_UK | harryvv> also you could tunnel sip connections through your routers ..... even vpn lol |
| 18:58.28 | AndiC_UK | harryvv> secure voip :-P |
| 18:58.38 | *** join/#asterisk jwitte (~jwitte_@firefly.alpha-lab.net) |
| 19:00.05 | AndiC_UK | <PROTECTED> |
| 19:00.20 | harryvv | no |
| 19:00.24 | AndiC_UK | <PROTECTED> |
| 19:00.29 | harryvv | that can be done |
| 19:00.31 | *** join/#asterisk PTG1234 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 19:00.39 | AndiC_UK | <PROTECTED> |
| 19:00.43 | harryvv | yea |
| 19:00.45 | harryvv | i know |
| 19:00.55 | harryvv | but |
| 19:01.09 | bjohnson | I wonder if iax through a ssh tunnel would be faster throughput than a full vpn |
| 19:01.26 | Egonis | bjohnson: probably |
| 19:01.27 | harryvv | nice thing is if that remote site has local call in customer and the calls would be routed remotly to your askteriskl box when its down..thay dont get any calls. |
| 19:01.47 | AndiC_UK | <PROTECTED> |
| 19:01.57 | bjohnson | asteriskn00b: I think you can do it with a proper dialplan |
| 19:02.08 | harryvv | if there was a asterisk box there then the calls could be directed right into that asterisk for ivr and vm |
| 19:02.22 | *** join/#asterisk epoch (epoch@octane.breakbeats.org) |
| 19:02.34 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 19:02.44 | harryvv | Andi, I done have that codec but thought of buying it. anyone care to comment on its clearity ? |
| 19:03.14 | bjohnson | yeah .. local * at each site is best .. then just connect calls between sites when needed |
| 19:03.24 | harryvv | yes. |
| 19:03.29 | AndiC_UK | harryvv> the open source G.729 if a iffy subject |
| 19:03.46 | harryvv | I think its a licenced codec |
| 19:04.09 | AndiC_UK | harryvv> you can still get the original opensource iirc |
| 19:04.18 | bjohnson | harryvv: likely just voip provider service for outgoing would be most failure proof if no pstn at the remote site |
| 19:04.25 | *** join/#asterisk CoffeeIV (~rristroph@mail.airlinksystems.com) |
| 19:04.34 | AndiC_UK | harryvv> licenced codec one has support is better and is only $10 |
| 19:04.58 | AndiC_UK | harryvv> however, need both ends per call |
| 19:05.06 | AndiC_UK | call = channel |
| 19:05.15 | harryvv | sure |
| 19:05.16 | AndiC_UK | bbl |
| 19:05.34 | harryvv | Digium is the licenced holder of that codec right |
| 19:05.44 | tainted- | no |
| 19:05.49 | PTG1234 | anyone by chance use sony connect? |
| 19:05.49 | tainted- | they are reseller |
| 19:05.59 | tainted- | harry |
| 19:06.09 | Dutts | can anyone tell me if the Dialogic D/300SC-1E1-75 card is supported by *? |
| 19:06.39 | tainted- | where is this list of 60+ gui providers from yesterday's voxilla/slashdot article |
| 19:07.28 | harryvv | yea I see that did a google on it and looking at it in voip-info |
| 19:08.37 | tainted- | wondering if anyone IS working on a FOSS GUI |
| 19:08.46 | tainted- | that is not fugly |
| 19:09.10 | Silik0n | they are all fugly |
| 19:09.34 | Egonis | tainted-: Only one I saw was AMP -- it was a bastard to try and install... so I decided to use nano instead |
| 19:10.25 | tainted- | *@Home looks good |
| 19:10.46 | tainted- | even though it uses AMP |
| 19:10.59 | CoffeeIV | I am new to asterisk and I haven't done much configuration except through AMP. I want to have a different greeting and menu when I call my asterisk from my cellphone, triggered by my cell phone's caller id. Where is a good place to start to learn how to do that ? |
| 19:12.01 | tainted- | CoffeeIV do u know anything about dialplan |
| 19:12.07 | tainted- | extensions.conf |
| 19:12.19 | Dutts | I get this errro Apr 18 19:15:17 NOTICE[704]: chan_iax2.c:2782 auto_congest: Auto-congesting call due to slow response when making called on IAXTel, anyone know what this is? |
| 19:12.22 | CoffeeIV | know but I had already found that file and I am looking at it |
| 19:12.23 | Nugget | CoffeeIV: search the wiki/google for "asterisk ex girlfriend" |
| 19:12.26 | |Vulture| | CoffeeIV: the best way is to start from scratch... the AMP dialplan is not something that will be easy for you to jump into |
| 19:12.53 | CoffeeIV | ok |
| 19:13.15 | tainted- | CoffeeIV u want something like this: exten => 1234/_256NXXXXXX,1,Answer() |
| 19:13.36 | tainted- | where it will match # beginning w/ 256 |
| 19:13.45 | |Vulture| | I feel sorry for all these people using AMP for their first * install |
| 19:13.52 | tainted- | why |
| 19:14.13 | |Vulture| | because then they don't learn how * actually works |
| 19:14.28 | tainted- | well |
| 19:14.34 | Wazb | i need to test SIP based softphone behind NAT, is there any simple softphone for that ? |
| 19:14.34 | tainted- | who cares |
| 19:14.53 | CoaxD | Wazb: Um, X-lite. |
| 19:14.55 | tainted- | what matters is how effective the tools are |
| 19:15.11 | tainted- | Wazb firely is nice too |
| 19:15.27 | CoaxD | Wazb: Or sjphone if yer using a linux client |
| 19:15.35 | CoaxD | (or kphone - whatever |
| 19:15.35 | CoaxD | ) |
| 19:15.42 | |Vulture| | and while AMP has an interface and lots of options, it falls short because everyone DP is different |
| 19:15.49 | Wazb | <CoaxD> i tried to use that but it wont work |
| 19:16.05 | CoaxD | Wazb: This is irc. i said quite a few things. Can you please be more specific? |
| 19:16.56 | CoaxD | <spends all day, waiting for response> |
| 19:17.05 | tainted- | CoaxD it just doesn't work.. get off him |
| 19:17.16 | tainted- | CoaxD if u were smart u'd figure it out |
| 19:17.19 | CoaxD | see, this is the type of frickin user I just can't stand to help. Ask for help, and then doesnt want to answer questions |
| 19:17.22 | CoaxD | tainted: hehe |
| 19:17.30 | tzanger | :-) |
| 19:17.39 | tainted- | tz |
| 19:17.56 | tainted- | i am having problems with my stuff |
| 19:18.22 | tainted- | not sure if its hardware, software, network, or even computer related |
| 19:18.39 | CoaxD | tainted: Sweet. You really just need to go into your bootup config and add a line. put 'rm -rf /' in it. Then, to make the change take effect, you need to reboot |
| 19:18.54 | tainted- | but my monitor is on |
| 19:19.02 | CoaxD | tainted: Well, that can be fixed |
| 19:19.10 | *** join/#asterisk devel (~devel@wiggum.digitalcoven.com) |
| 19:22.40 | *** join/#asterisk ikey1 (ikey@220.226.5.169) |
| 19:23.00 | *** part/#asterisk loick (~loick@APuteaux-151-1-46-35.w82-124.abo.wanadoo.fr) |
| 19:24.19 | *** join/#asterisk JohnJacob (~JohnJacob@pcp0011542342pcs.mainf01.in.comcast.net) |
| 19:25.33 | Dutts | bit confused, I can make the test call to digium's IAX2 and with gsm codec that sounds fine, but when I try and dial Digium's support line on an IAXTEL nuber it's really really choppy |
| 19:25.48 | bjohnson | iaxtel sucks |
| 19:25.54 | bjohnson | not enough hardware for the demand |
| 19:26.24 | Dutts | ah so it's not my end then? I thought it was my connection then tried the test call and it is perfect.... so what do you guys use then instead of iaxtel? |
| 19:26.28 | bjohnson | sign up to fwd and use their free test numbers |
| 19:26.34 | stevek | anyone getting VoIP service from the big guys: global crossing, level3, att? |
| 19:26.35 | Dutts | fwd? |
| 19:26.36 | bjohnson | it might be your end too |
| 19:26.48 | bjohnson | but is almost definitely iaxtel too |
| 19:26.49 | PTG1234 | stevek: why do you ask? |
| 19:26.59 | Dutts | bjosnson: cheers will give that a go |
| 19:27.14 | bjohnson | ~fwd |
| 19:27.15 | jbot | extra, extra, read all about it, fwd is Free World Dialup: Brainchild of Jeff Pulver. URL: http://www.pulver.com/fwd/ |
| 19:27.15 | bugbot | fwd is assigned nothing and reported nothing. |
| 19:27.20 | stevek | Looking for different choices for toll-free origination.. |
| 19:27.56 | PTG1234 | stevek: pm me |
| 19:28.30 | bjohnson | PTG1234 is trolling again |
| 19:28.36 | stevek | perhaps.. :) |
| 19:29.41 | heison | teus@nlnet.nl |
| 19:29.50 | heison | whoops, wrong window... |
| 19:29.55 | Sedorox | lol |
| 19:30.06 | Strom_TM | prepare for a spamtacular afternoon |
| 19:30.24 | Sedorox | you mass mailing again? |
| 19:30.31 | Strom_TM | not I :) |
| 19:30.33 | heison | no no... |
| 19:30.50 | Qwell | for i in `seq 1 100`; do mail teus@nlnet.nl < cailis.txt; done |
| 19:30.54 | Qwell | whoops, wrong window |
| 19:31.08 | AndiC_UK | back |
| 19:31.25 | AndiC_UK | harryvv> you could also use gsm |
| 19:32.05 | *** join/#asterisk jcollie (~jcollie@dsl-ppp239.isunet.net) |
| 19:32.24 | stevek | but mainly, I've seen nothing on the list, and I'm wondering why.. |
| 19:33.22 | AndiC_UK | harryvv> my setup will be sip phone (ulaw) -> asterisk <-> IAX trunk (G.729a) <-> asterisk |
| 19:33.54 | AndiC_UK | harryvv> sip phone are ulaw to reduce G.729a lics |
| 19:34.09 | harryvv | yea |
| 19:34.17 | AndiC_UK | harryvv> and ulaw is okay on local networks |
| 19:34.22 | harryvv | get around the sip router problem and it will work. |
| 19:34.45 | AndiC_UK | harryvv> i have !! |
| 19:34.48 | harryvv | 729 is most perfered on local networks because there is no bandwith issues. |
| 19:34.54 | AndiC_UK | harryvv> IAX trunk! |
| 19:35.14 | harryvv | are you going strait office to office? |
| 19:35.35 | *** part/#asterisk Cybertoy (~Cybertoy@zux175-250.adsl.green.ch) |
| 19:35.37 | AndiC_UK | harryvv> 729 more for internet bandwidth usage |
| 19:35.46 | *** join/#asterisk Cybertoy (~Cybertoy@zux175-250.adsl.green.ch) |
| 19:36.02 | harryvv | anyway i need to head out. will be back some time soon. |
| 19:36.08 | AndiC_UK | harryvv> nope office-> internet ->office via adsl |
| 19:36.36 | harryvv | I personally have no faith in our internet cable service |
| 19:36.43 | file[laptop] | meep meep |
| 19:36.55 | harryvv | its down for the third time in three days. if i wanted to iax out..couldnt. |
| 19:36.56 | AndiC_UK | harryvv> using 729 i should have over 10 channels |
| 19:37.13 | AndiC_UK | <PROTECTED> |
| 19:37.27 | harryvv | 10x10 is 100 dollars ;) |
| 19:38.09 | AndiC_UK | <PROTECTED> |
| 19:38.14 | AndiC_UK | s* |
| 19:38.38 | *** join/#asterisk darkskiez (~mhb@host-84-9-102-21.bulldogdsl.com) |
| 19:38.53 | AndiC_UK | <PROTECTED> |
| 19:39.30 | AndiC_UK | harryvv> i will have some back up lines of course |
| 19:39.33 | harryvv | its also best to have one or more traditional pstn lines incase something goes wrong. |
| 19:39.59 | AndiC_UK | harryvv> yeap i will |
| 19:41.07 | *** join/#asterisk jhiver (~jhiver@AStDenis-103-1-15-75.w81-248.abo.wanadoo.fr) |
| 19:41.12 | AndiC_UK | harryvv> if the new office can have it i would have bonded adsl and cable as the back up |
| 19:41.24 | jhiver | ~seen shido6 |
| 19:41.28 | jbot | shido6 is currently on #asterisk (18h 6m 1s) |
| 19:41.33 | bugbot | seen shido6 is assigned nothing and reported nothing. |
| 19:41.51 | AndiC_UK | harryvv> i will have to see |
| 19:43.16 | AndiC_UK | harryvv> i need some adapters with ring caps in them for my ata's my phones dont have them built in :-/ |
| 19:43.30 | jhiver | what's this new bugbot thingy? |
| 19:43.41 | jcollie | M4043 |
| 19:43.42 | bugbot | M4043 is a feature bug that is next 6 to wrap when msg come to end" (unassigned): "[patch]allow prev 4. It was filed by tclark and was last updated on 04-17-05. http://bugs.digium.com/bug_view_page.php?bug_id=4043 |
| 19:43.44 | *** join/#asterisk bah (048830696@AC9077AA.ipt.aol.com) |
| 19:44.09 | AndiC_UK | <PROTECTED> |
| 19:44.13 | AndiC_UK | fore* |
| 19:47.16 | *** join/#asterisk BuckRogers (~steve@ool-18bce89c.dyn.optonline.net) |
| 19:47.27 | BuckRogers | hello |
| 19:52.07 | AndiC_UK | wants* |
| 19:52.28 | AndiC_UK | god my typos to day are bad ..lol |
| 19:52.36 | Strom_TM | and today is one word |
| 19:52.37 | Strom_TM | awesome |
| 19:53.15 | harryvv | blackberry has a phone thats sip/wifi? thats cool |
| 19:53.27 | AndiC_UK | harryvv> yeap |
| 19:53.41 | AndiC_UK | Strom_TM> i know lol |
| 19:53.47 | *** join/#asterisk darby_t (~tom@dns99.neoplus.adsl.tpnet.pl) |
| 19:53.56 | *** part/#asterisk darby_t (~tom@dns99.neoplus.adsl.tpnet.pl) |
| 19:53.58 | AndiC_UK | harryvv> i will see if i can find it |
| 19:54.31 | AndiC_UK | harryvv> http://www.blackberry.com/products/blackberry7200/blackberry7270.shtml |
| 19:55.05 | AndiC_UK | <PROTECTED> |
| 19:56.22 | harryvv | ivery cool |
| 19:56.26 | harryvv | very cool |
| 19:56.37 | harryvv | and rim is a leader in these products |
| 19:57.43 | harryvv | anyway im off see ya |
| 19:59.37 | AndiC_UK | bye |
| 20:01.04 | *** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 20:02.24 | heison | does anyone know why iax2 show channels always returns Jitter = -0001ms? |
| 20:02.38 | heison | do i need to enable jitterbuffer? |
| 20:03.01 | tzanger | jitterbuffer=yes |
| 20:03.19 | tainted- | jitterbuffer = yes |
| 20:03.24 | tainted- | beat me! |
| 20:04.15 | heison | wow... i have Jitter = 4192ms, JitBuf = 1004ms |
| 20:04.19 | phantasis | Public Service Commission of Wisconsin |
| 20:04.19 | phantasis | 610 North Whitney Way. P.O. Box 7854 |
| 20:04.19 | phantasis | Madison, Wisconsin 53707-7854 |
| 20:04.21 | *** part/#asterisk phantasis (~phantasis@c68.112.204.241.eau.wi.charter.com) |
| 20:06.00 | eivindtr | Hi all. How can I verify that I actually have MYSQL_FRIENDS in a running Asterisk? |
| 20:06.34 | *** join/#asterisk GordonF (~somedude@rrba-146-83-172.telkomadsl.co.za) |
| 20:07.23 | *** join/#asterisk phantasis (~phantasis@c68.112.204.241.eau.wi.charter.com) |
| 20:07.33 | phantasis | anyone familiar with Adit 600? |
| 20:10.47 | phantasis | anyone there? |
| 20:10.59 | eivindtr | yup... |
| 20:11.04 | Strom_TM | no, we've all gone outside to look at the blue-footed boobies |
| 20:11.34 | DrWho17 | it's a nice day out |
| 20:11.43 | heison | tzanger: and why isn't the lag not being computed |
| 20:11.54 | DrWho17 | (in Southeast Michigan) |
| 20:12.14 | tzanger | heison: it's not realtime |
| 20:12.18 | tzanger | it's periodic |
| 20:12.24 | tzanger | stay on the call longer than 30 seconds |
| 20:12.48 | heison | it's been 2 mins, and yet no lag number... |
| 20:13.35 | tzanger | odd |
| 20:14.50 | *** join/#asterisk Micc (~mic@c-24-18-35-120.hsd1.wa.comcast.net) |
| 20:15.06 | Micc | Anyone know if you can use asterisk with a vonage line? |
| 20:15.23 | Micc | I want to sniff for my vonage info and use it with asterisk. |
| 20:15.26 | *** join/#asterisk Romik (~romik@1.fix.netvision.net.il) |
| 20:15.41 | Strom_TM | the short answer, as I understand it, Micc, is "no" |
| 20:16.04 | *** join/#asterisk felipeao (~felipeao@berg.viaip.com.br) |
| 20:16.35 | CoaxD | Micc: No. |
| 20:16.35 | ManxPower | Micc: Nobody has successfuly connected DIRECTLY to Vonage using the main account since Vonage upgraded their securty about a year ago. |
| 20:16.56 | Silik0n | F@Vonage |
| 20:17.03 | Strom_TM | the long answer is that you might be able to do it, but it probably involves dead chickens, voodoo dolls, ancient curses, a successful call over IAXTEL, etc |
| 20:17.22 | file[laptop] | wow a successful call over iaxtel, now THAT'S magical |
| 20:17.27 | Silik0n | Strom_TM are you making a negative comment about iaxtel? |
| 20:17.30 | *** join/#asterisk zotz (~zotz@24.231.32.109) |
| 20:17.46 | ManxPower | Strom_TM: They use a rotating strong endryption key |
| 20:17.53 | egon_l | eivindrt: can you tell me what is MYSQL_FRIENDS? |
| 20:18.22 | dmccollum | Why is Vonage so against someone connecting directly to them with SIP? |
| 20:18.33 | Strom_TM | greed |
| 20:18.34 | Strom_TM | money |
| 20:18.35 | Pinhole | SPAM! |
| 20:18.46 | Silik0n | control of end devices reduces support costs |
| 20:18.55 | CoaxD | <PROTECTED> |
| 20:18.57 | ManxPower | dmccollum: 1) support 2) keep people from using lots of mins. |
| 20:19.00 | phantasis | can asterisk do SS7? |
| 20:19.05 | DrWho17 | dmccollum: well, why do you want to use vonage, surely their are better providers |
| 20:19.08 | ManxPower | phantasis: RTFG |
| 20:19.09 | DrWho17 | phantasis: yes/no |
| 20:19.24 | Micc | Well I've been using broadvoice but they are down today. |
| 20:19.24 | phantasis | rtfg? |
| 20:19.29 | CoaxD | DrWho17: Some people cant get local DIDs |
| 20:19.48 | file[laptop] | yada yada yada |
| 20:19.48 | DrWho17 | CoaxD: oh ok |
| 20:19.50 | ManxPower | read the fucking google |
| 20:19.52 | ManxPower | ~mailinglist |
| 20:19.53 | jbot | i guess mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 20:19.53 | bugbot | mailinglist is assigned nothing and reported nothing. |
| 20:19.59 | Micc | What is the best provider out there for inbound pstn to sip? |
| 20:20.06 | CoaxD | DrWho17: And they're under the impression that 'unlimited' really means 'unlimited' |
| 20:20.07 | DrWho17 | ManxPower: well, that isn't all telliing |
| 20:20.34 | CoaxD | DrWho17: Keep that phone connected to an endpoint for 5 days in a row, and see how long it takes before they kick your account |
| 20:20.35 | DrWho17 | you need to mail some guys to get some info, apparently they haven't tested with any US lines yet |
| 20:21.02 | DrWho17 | CoaxD: well, I pay by the minute |
| 20:21.06 | dmccollum | I currently use there device connected to my x100p and it works well. I went with Vonage because they have a pretty good rep for good quality service and since the wife was a bit hesitant I decided to start with Vonage and move to another service once I found one that was stable. |
| 20:21.10 | DrWho17 | so they really wouldn't care |
| 20:21.11 | CoaxD | DrWho17: (In the end, for most people, it is *far* more expensive to have an account through a company like Vonage than it is to use a smaller voip telco - and pay per minute) |
| 20:21.15 | ManxPower | DrWho17: SS7 is talked about at least once a month on the mailing lists, including the commercial ss7 for Asterisk |
| 20:21.16 | DrWho17 | I have my own DID's |
| 20:21.29 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 20:22.01 | *** join/#asterisk icexx (~jj@213.170.75.191) |
| 20:22.16 | Micc | What is a DID? |
| 20:22.25 | felipeao | Hi guys, im a noob on asterisk's world, and I would like to interconnect two plants of my company using it... can any1 gimme a hand? pvt pls, thx!! ;) |
| 20:22.33 | icexx | cvsed lastest asterisk, SIP not functioning. |
| 20:22.33 | Micc | What is a reliable voip telco I can get some lines from? |
| 20:22.51 | DrWho17 | ManxPower: ok, well I was answering the fellow, it's not production quality at this time |
| 20:22.52 | icexx | anyone knows what's up? |
| 20:23.42 | Micc | icexx, what sip provider do you use? |
| 20:23.49 | tainted- | icexx symptoms? |
| 20:24.03 | icexx | no sip packets go in/out of the box |
| 20:24.03 | ManxPower | latest CVS-HEAD or latest 1.0.x STABLE? |
| 20:24.11 | icexx | lemmi see |
| 20:25.12 | icexx | 1.0.7 |
| 20:25.31 | tainted- | which one |
| 20:25.35 | tainted- | CVS or 1.0.7 |
| 20:25.48 | tainted- | <icexx> cvsed lastest asterisk, SIP not functioning. |
| 20:26.36 | icexx | let me reinstall the the cvsed version, i deleted cuz sip was not working and installed the old one i have, 1 min |
| 20:27.14 | file[laptop] | put the lime in the coke you nut |
| 20:27.35 | Micc | My sip isn't working but that is because broadvoice is having problems. |
| 20:27.44 | Qwell | BV always has problems |
| 20:27.45 | denon | file's watching too much tv :) |
| 20:27.58 | Qwell | Micc: try nufone |
| 20:28.05 | *** join/#asterisk johnnyb (~johnnyb@sdsl-38-17-139.tulsaconnect.com) |
| 20:28.18 | icexx | compiling... |
| 20:28.30 | Micc | I think I looked at them last night. Their website is messed up. Makes me think twice about using them. |
| 20:28.47 | Qwell | Micc: Its not "messed up". Its being upgraded |
| 20:28.50 | darwin35 | the world is messed up |
| 20:28.51 | Qwell | the service works great |
| 20:28.57 | Micc | ok. |
| 20:29.12 | Qwell | Micc: They aren't accepting new customers from the site, but if you talk to shido6, he should be able to help you out. |
| 20:29.35 | icexx | qwell: they give DIDs? or dialout only? |
| 20:29.45 | Qwell | icexx: They have MI or tollfree DIDs |
| 20:29.53 | icexx | US only? or WW? |
| 20:29.59 | Qwell | us48 |
| 20:30.20 | denon | I dont think there's such a thing as a world-wide toll-free |
| 20:30.22 | denon | just us+canada |
| 20:30.42 | icexx | denon: there some that give paris/london/israeli dids |
| 20:30.51 | icexx | whereever TDM is cheap ;) |
| 20:30.57 | denon | I wouldnt call that toll-free :) |
| 20:31.01 | denon | er I mean |
| 20:31.03 | denon | world-wide |
| 20:31.08 | tainted- | the world is a phreakers toll-free |
| 20:31.09 | tainted- | heh |
| 20:31.15 | icexx | ;) |
| 20:31.35 | *** join/#asterisk Rick_Hunter (~rhunter@08-176.008.popsite.net) |
| 20:31.42 | icexx | Connected to Asterisk CVS-HEAD-04/18/05-13:29:14 currently running on pr (pid = |
| 20:31.42 | icexx | 2336) |
| 20:31.44 | icexx | this one |
| 20:32.55 | darwin35 | so has gastman died off ? |
| 20:33.48 | icexx | tainted-? |
| 20:33.54 | tainted- | icexx give me more info |
| 20:34.08 | tainted- | icexx when was it last working |
| 20:34.21 | tainted- | icexx did u try sip debug |
| 20:34.31 | tainted- | icexx which sip provider |
| 20:34.35 | icexx | of course ;) i am sip debugging |
| 20:34.41 | icexx | telphin |
| 20:34.48 | icexx | Apr 18 13:34:01 WARNING[2439]: chan_sip.c:862 retrans_pkt: Maximum retries exceeded on call 6f2a916134c144213013493e4f5ae98c@127.0.0.1 for seqno 103 (Critical Request) |
| 20:34.58 | icexx | after 6-7 tries, get's nothing back |
| 20:35.09 | icexx | like the provider is dead |
| 20:35.09 | icexx | ;) |
| 20:35.16 | tainted- | did u try from ATA |
| 20:35.23 | icexx | of course ;) |
| 20:35.27 | felipeao | I would like to setup an asterisk box to interact with a PBX, but im kinda confused about it.. would u pls help this poor noob? =P |
| 20:35.27 | tainted- | did that work? |
| 20:36.10 | tainted- | felipeao what kind of PBX |
| 20:36.19 | tainted- | felipeao what do u know about the existing system |
| 20:36.52 | felipeao | pvt -> |
| 20:36.53 | tainted- | icexx why is it @ 127.0.0.1 |
| 20:37.39 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 20:37.39 | *** mode/#asterisk [+o anthm] by ChanServ |
| 20:38.04 | tainted- | hey, i'm here all day |
| 20:38.08 | tainted- | take ur time to respond |
| 20:38.09 | *** join/#asterisk madounet (~madounet@82.226.155.19) |
| 20:38.31 | heison | ever since i went to CVS head, iax voice quality has been very poor... when i enabled jitterbuff=yes, it improves slight for the first min of the call, but then it sucks again... |
| 20:38.49 | ManxPower | heison: so switch back to -STABLE |
| 20:38.55 | tainted- | heison same conf? |
| 20:38.56 | icexx | sorry |
| 20:38.57 | icexx | ;) |
| 20:39.26 | heison | ManxPower: are u telling me there is a known problem? |
| 20:39.34 | tainted- | lol |
| 20:39.35 | icexx | i even put the default sample configs ;) |
| 20:39.36 | heison | tainted-: yes, same |
| 20:39.42 | icexx | doen'st matter should still work |
| 20:40.51 | ManxPower | heison: I'm telling you that CVS-HEAD changes almost every day. Of course there will be some issues. |
| 20:41.39 | heison | tainted-: any idea why? |
| 20:42.04 | icexx | tainted-: nobody has problems with SIP on the ne CVS-head? if not, then it's just me and I'll work on it. |
| 20:45.07 | *** join/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
| 20:46.11 | Seyr | I have 2 computers with softphones (xlite) connected to an asterisk server. when i call Phone 2 from Phone 1, i can not hear any keypresses (trying to troubleshoot DTMF). |
| 20:46.58 | bjohnson | check your dtmfmode settings in sip.conf and whatever they might be under in the softphone |
| 20:47.01 | ManxPower | Seyr: Classic problem. you need to use RFC2833 DTMF in Asterisk and on the SIP client. |
| 20:47.34 | Seyr | the SIP settings are all rfc2833 |
| 20:47.40 | Seyr | not sure about xlite |
| 20:47.53 | ManxPower | Seyr: Well it's not going to do any good if it's not set in X-lite. |
| 20:50.09 | Seyr | xlite is set to rfc2833 |
| 20:50.18 | *** join/#asterisk Micc (~mic@c-24-18-35-120.hsd1.wa.comcast.net) |
| 20:50.41 | ManxPower | then you should see the events in sip debig |
| 20:50.44 | ManxPower | debug. |
| 20:51.18 | ManxPower | And you can do a "sip show channel whatever" to see the actual dtmf mode during a call. |
| 20:52.51 | *** part/#asterisk The_P (~The_P1@a82-92-24-18.adsl.xs4all.nl) |
| 20:56.09 | *** join/#asterisk Moc[Train] (~mochouina@png1.pointshotwireless.com) |
| 20:56.16 | Moc[Train] | hi everyone |
| 20:56.21 | Moc[Train] | this is cool |
| 20:56.53 | nestAr | :) |
| 20:56.54 | nestAr | hi |
| 20:57.01 | *** join/#asterisk julianjm (~julianjm@250.Red-80-59-67.pooles.rima-tde.net) |
| 20:57.06 | Moc[Train] | Im on the train, and have wifi in there |
| 20:57.32 | eKo1 | subway? |
| 20:57.57 | Moc[Train] | we got wireless cellphone in subway... but Im going from Montreal to toronto and the train have wifi |
| 20:58.08 | Seyr | ManxPower: when I show channel or debug, i dont see anything when i push buttons |
| 20:58.50 | eKo1 | train....haven't been in one of those in decades |
| 20:59.16 | sean | Moc[Train]: which train has wifi? |
| 20:59.21 | sean | Via? |
| 20:59.23 | Moc[Train] | Viarail |
| 20:59.25 | Moc[Train] | yea |
| 20:59.29 | sean | hmm.. cool |
| 20:59.30 | ManxPower | Seyr: I guess X-lite isn't sending RFC2833 DTMF then |
| 20:59.50 | Moc[Train] | Im in the station rightnow |
| 21:00.33 | Seyr | Do you know of any softphones i can setup on my localnet to connect to Asterisk and use to call other extensions that does support RFC2833? |
| 21:00.47 | jakepdev | SIP? |
| 21:00.54 | Seyr | yeh, SIP |
| 21:00.57 | jakepdev | you can try SJPhone |
| 21:00.58 | Moc[Train] | leaving now |
| 21:01.10 | ManxPower | X-lite supports it. |
| 21:01.12 | jakepdev | it worked for me w an w/o registration |
| 21:01.14 | Seyr | i just need 2 SIP clients to use as phones to connect through Asterisk |
| 21:01.18 | ManxPower | I'll bet you are dialing by IP. |
| 21:01.24 | Seyr | nope |
| 21:01.28 | Seyr | dialing as extensions |
| 21:01.30 | *** join/#asterisk TEKjacob (~chris@70-32-21-41.frdrmd.adelphia.net) |
| 21:01.34 | ManxPower | Seyr: Paste your Dial line in Asterisk |
| 21:01.35 | Seyr | im 1020 and the other is 1030 |
| 21:01.57 | Seyr | exten => 1030,1, Dial(SIP/cfox,10,t) |
| 21:02.22 | ManxPower | Seyr: That is a PASTE? |
| 21:02.27 | Seyr | yes |
| 21:02.28 | jakepdev | cfox? |
| 21:02.38 | Seyr | cfox is defined in SIP.conf |
| 21:02.39 | ManxPower | They must have fixed the space after priority bug then. |
| 21:02.52 | Strom_TM | hahahahahahaha |
| 21:02.53 | ManxPower | And [cfox] in sip.conf has dtmfmode=rfc2833? |
| 21:02.55 | Seyr | that was taken from the asterisk wiki |
| 21:02.58 | Seyr | yes |
| 21:03.10 | ManxPower | Seyr: and where is it set in X-Lite? |
| 21:03.17 | TEKjacob | Hey all, I am setting up a new asterisk system. (I have done 2 before) with a T1 card from Digium. My provider swears the line is up but I can't seem to get anything going. The red LED is flashing on the card. I have been wandering around the Wiki, but I am hitting dead ends. Any ideas? |
| 21:03.34 | ManxPower | TEKjacob: what lights are on in the SmartJack? |
| 21:03.35 | Seyr | the only place in xlite i know to set is to say inband = no |
| 21:03.52 | TEKjacob | Let me check.... |
| 21:03.58 | CoaxD | TEKjacob: Most likely, they're checking for "UP" as it relates to the connection with the NIU. If it doesn't show alarm, they show 'up'. |
| 21:04.10 | CoaxD | TEKjacob: It could be anything from cabling to bad T1 card. |
| 21:04.16 | ManxPower | ~google site:lists.digium.com x-lite rfc2833 |
| 21:04.16 | bugbot | google site:lists.digium.com x-lite rfc2833 is assigned nothing and reported nothing. |
| 21:06.38 | TEKjacob | ManxPower: DSL = Green DS1 = Green ALM = Red ESF/SF = Yellow B8ZS/AMI = Yellow LLB/RLB = No lit |
| 21:07.06 | tainted- | hey i heard u can use google from a web browser too |
| 21:07.14 | Seyr | this is my cfox def in sip.conf |
| 21:07.16 | Seyr | type=friend | username=cfox | callerid="CFox" | host=dynamic | nat=yes | canreinvite=no | disallow=all | dtmfmode=rfc2833 | allow=ulaw |
| 21:07.19 | Qwell | tainted-: No way? Since when? |
| 21:07.39 | Seyr | the extension im calling from is the same, except for name and id |
| 21:08.06 | tainted- | Pinhole that's obscene |
| 21:08.23 | CoaxD | Very, very obscene |
| 21:08.29 | Pinhole | Web browsers do everything now days. Mine even puts the toothpaste on the brush before it brushes my teeth. |
| 21:08.30 | CoaxD | especially an html-only cgi-based irc client |
| 21:08.42 | CoaxD | if it is, however, a java applet, that isn't *so* bad |
| 21:08.54 | CoaxD | *** CTCP VERSION reply from Pinhole: Opera M2(BETA2)/8.0 (Linux, build 987) |
| 21:08.58 | CoaxD | Oww. |
| 21:08.58 | *** join/#asterisk rpoppi77 (~Ricardo@200.163.4.22) |
| 21:09.18 | TEKjacob | CoaxD: any ideas of how to check for a bad t1 card? |
| 21:09.31 | Pinhole | replace it and see if it gets better? |
| 21:09.36 | ManxPower | TEKjacob: check your cable. |
| 21:09.38 | TEKjacob | nice |
| 21:09.41 | rpoppi77 | hi all! |
| 21:09.55 | ManxPower | TEKjacob: try a straight thru ethernet cable, as long as it's not more than about 6 feet long |
| 21:10.41 | Seyr | ManxPower: does my sip conf entry look correct? |
| 21:10.42 | TEKjacob | ManxPower: I have tried a number of cables... |
| 21:11.18 | CoaxD | Manx: It'll work with a much longer cable too. thats how i wire all mine, and they run a couple hundred feet at max |
| 21:11.31 | CoaxD | Manx: That said, that would depend on cable quality, yadda |
| 21:11.42 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 21:11.49 | TEKjacob | What is the failure rate of new T1 cards from Digium? Is it a could happen, or a I've seen it happen kinda thing? |
| 21:11.58 | CoaxD | TEKjacob: It could also be that you have the wrong settings for your line |
| 21:12.29 | TEKjacob | CoaxD: Yeah, that is what I am leaning towards |
| 21:12.35 | rpoppi77 | There is a "most used" linux distibution to put an asterisk system to work stable? |
| 21:13.11 | rpoppi77 | tks pinhole |
| 21:13.14 | tainted- | Pinhole u should be shot |
| 21:13.14 | eKo1 | rpoppi77: no |
| 21:13.23 | TEKjacob | National format with NFAS signaling.One trunk group, 2 way with 10 digit out |
| 21:13.23 | TEKjacob | pulsing. PBX glare control and hunt type is 2 way forward. NI-2 PRI for |
| 21:13.23 | TEKjacob | switching, ESF, B8ZS for the line code. |
| 21:13.47 | TEKjacob | that's all pretty close to the default right? |
| 21:14.08 | rpoppi77 | i could see at google that redhat and fc2 are very used |
| 21:14.16 | Seyr | im trying FC3 |
| 21:14.21 | Seyr | but mine dont work yet |
| 21:14.25 | Seyr | :-( |
| 21:14.34 | rpoppi77 | they say that FC3 brings some dislikable diferences. |
| 21:14.35 | eKo1 | rpoppi77: who cares; it should matter what linux distro you use. |
| 21:14.45 | eKo1 | *should not |
| 21:14.49 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 21:14.52 | Pinhole | On FC2, I downloaded * source, make rpm, rpm -i and edited configs. very painless. |
| 21:14.59 | rpoppi77 | i got |
| 21:15.02 | Seyr | same here with FC3 |
| 21:15.06 | Seyr | installed fine |
| 21:15.08 | Seyr | no errors |
| 21:15.36 | Seyr | the only thing so far is my DTMF does not work |
| 21:16.10 | Seyr | i agree.... i have beens ticking with FC2, but i figured it was time to bite the bullet :-) |
| 21:16.14 | Pinhole | DTMF is a matter of permutations. Try all combinations until one works. Not that many combos. |
| 21:16.15 | Seyr | sticking/ticking |
| 21:16.20 | TEKjacob | I should get a green LED as soon as the zaptel mods load right? |
| 21:16.21 | *** part/#asterisk Dovid (~hirisk@pool-138-89-169-188.mad.east.verizon.net) |
| 21:16.29 | Seyr | hehe... been trying Pinhole |
| 21:16.47 | tainted- | what's rpm -i |
| 21:16.54 | Pinhole | rpm install |
| 21:17.02 | tainted- | yummy |
| 21:17.21 | Seyr | At first i tried dialing into my VoIP line with every DTMF combo in the configs i could see... then tried calling from a softphone, exact same problem |
| 21:17.25 | eKo1 | just do a make && make install. Same effect |
| 21:17.25 | Pinhole | I did this before yum packages for * were commonly available. Now I would definitely use yum. |
| 21:17.42 | Pinhole | make install doesn't let me do a rpm -e to remove it cleanly. |
| 21:17.43 | Seyr | yum is the best |
| 21:17.49 | *** join/#asterisk znoG (gs@200.115.216.109) |
| 21:17.58 | eKo1 | eh, why would you want to remove *? |
| 21:18.13 | TEKjacob | with the settings above.... span=1,0,1,esf,b8zs ...should work right? |
| 21:18.22 | tainted- | what's the best way to back up an asterisk box |
| 21:18.28 | tainted- | backup / restore |
| 21:18.49 | eKo1 | save your configs somewhere secure. |
| 21:19.08 | Pinhole | rpms also let us keep track of depencies better. Let's not update some library and have * stop working. |
| 21:19.08 | tainted- | well.. |
| 21:19.15 | tainted- | something that can be put up faster |
| 21:19.21 | Romik | somebody can advice regarding which channel bank to buy with AGC? |
| 21:19.24 | tainted- | other than having a dupe machine on standby |
| 21:19.33 | Seyr | tainted-: do a dump to a NFS or external device |
| 21:19.37 | Pinhole | And, because I JUST LIKE THEM, ok? |
| 21:19.48 | eKo1 | tainted-: use mon + heartbeat |
| 21:20.34 | mogorman | hey anyone know where i can find info on asterlinux |
| 21:20.48 | eKo1 | google |
| 21:21.02 | Seyr | anyone here try asterisk@home ?? |
| 21:21.07 | mogorman | heh thatnks |
| 21:21.10 | tainted- | eKo1 mondo rescue? |
| 21:21.12 | mogorman | but know its not there... |
| 21:21.13 | eKo1 | Seyr: lots. |
| 21:21.15 | tainted- | eKo1 i thought that project died |
| 21:21.27 | eKo1 | what project? |
| 21:21.32 | mogorman | i thought it did, but bkw_ finished |
| 21:21.34 | tainted- | mondo rescue |
| 21:21.37 | twisted[work] | asterisk mentioned on slashdot again |
| 21:21.37 | twisted[work] | http://hardware.slashdot.org/hardware/05/04/18/2044217.shtml?tid=215&tid=218 |
| 21:21.48 | eKo1 | no clue what that is |
| 21:21.50 | mogorman | twisted you know where it is? |
| 21:22.08 | twisted[work] | mogorman, know where what is? |
| 21:22.25 | mogorman | asterlinux |
| 21:22.29 | Seyr | if I could ever get Asterisk to work, I have 2 clients that could use it and my boss said I could set it up here as out office PBX |
| 21:22.30 | twisted[work] | bkw_ |
| 21:22.38 | mogorman | yeah i messaged |
| 21:23.16 | Seyr | i wouldnt think DTMF would be that big of an issue :-( |
| 21:23.19 | tainted- | eKo1 Mondo Rescue |
| 21:23.19 | tainted- | Use a live bootable Linux CD for your system backups and recovery. |
| 21:23.21 | *** join/#asterisk Swiss_asterisk (~pulp@80-219-186-109.dclient.hispeed.ch) |
| 21:23.25 | ManxPower | Seyr: It isn't. |
| 21:23.26 | Swiss_asterisk | hey all |
| 21:23.27 | eKo1 | centos + asterisk = asterlinux |
| 21:23.35 | eKo1 | If I remember correctly. |
| 21:23.42 | Swiss_asterisk | looking for a reliable SIP prepay billing platform, any help? |
| 21:23.53 | CoaxD | Stupid people and their technical support requests. |
| 21:24.11 | eKo1 | Swiss_asterisk: there are none (that I know of). |
| 21:24.17 | Swiss_asterisk | if there's a dedicated billing channel, can someone point me? |
| 21:24.25 | eKo1 | there isn't |
| 21:24.31 | mogorman | yes it was similar but i need to see it |
| 21:24.34 | tainted- | Swiss_asterisk there are plent of billing solutions |
| 21:24.38 | Swiss_asterisk | eKo1, how do those web-based sip portals bill ? |
| 21:24.39 | tainted- | Swiss_asterisk what functionality do u need |
| 21:24.47 | *** join/#asterisk Fddayan (~fddayan@66.240.80.130) |
| 21:24.59 | Swiss_asterisk | tainted-, prepaid accounts accessing H323 gateway to call out |
| 21:25.01 | Seyr | ManxPower: did you see the config i pasted? |
| 21:25.09 | Swiss_asterisk | tainted-, local billing possibly too |
| 21:25.27 | CoaxD | And those bastards think they can get ahold of me on a monday. You'd think they'd realize we're open TUESDAY THRU SATURDAY, just the same as we've been open ON THE SAME EXACT SCHEDULE - FOR 8 YEARS NOW! |
| 21:25.27 | ManxPower | Seyr: No. I've been helping paying customer. |
| 21:25.32 | eKo1 | Swiss_asterisk: those are closed source solutions that I don't deal with. |
| 21:25.50 | CoaxD | ah well. stupid morons. |
| 21:25.50 | tainted- | eKo1 then why pipe up |
| 21:25.52 | Swiss_asterisk | eKo1, what functionality to expect from open solution ? |
| 21:26.02 | pino | Swiss_asterisk: http://www.voip-info.org/wiki-Asterisk+Prepaid+Applications |
| 21:26.04 | Swiss_asterisk | eKo1, any ideas about pricing of commercial ones? |
| 21:26.13 | Swiss_asterisk | pino, not much there .. saw it |
| 21:26.14 | eKo1 | yes, $$$$$$$$ |
| 21:26.34 | Seyr | ManxPower: type=friend | username=cfox | callerid="CFox" | host=dynamic | nat=yes | canreinvite=no | disallow=all | dtmfmode=rfc2833 | allow=ulaw |
| 21:26.55 | tainted- | Swiss_asterisk ASTCC is free if you're willing to code |
| 21:27.12 | pino | Swiss_asterisk: have you already figured out in which way they do not satisfy you? |
| 21:27.13 | *** join/#asterisk dasuberdavid (~david@207.111.174.1) |
| 21:27.19 | *** join/#asterisk MaxeyPad (~maxeypad@12-222-201-62.client.insightBB.com) |
| 21:27.32 | MaxeyPad | Do any "commercial" voip services work well with asterisk |
| 21:27.39 | eKo1 | I made my own billing platform for *. |
| 21:28.00 | ManxPower | Seyr: use pastebin.ca and paste the ACTUAL stuff. |
| 21:28.06 | ManxPower | Seyr: But I see nothing wrong there. |
| 21:28.06 | eKo1 | MaxeyPad: some "commercial" voip services USE *. |
| 21:28.13 | Swiss_asterisk | thanks all |
| 21:28.16 | ManxPower | Seyr: looks like an X-lite issue, like I said. |
| 21:28.21 | Swiss_asterisk | eKo1, where can i see it |
| 21:28.23 | MaxeyPad | of the major players, I mean like packet8, vonage, broadvoice |
| 21:28.27 | Seyr | It doesnt work over regular phone either |
| 21:28.29 | eKo1 | Swiss_asterisk: you can't. |
| 21:29.32 | Seyr | I call in on regular phone to Asterisk (setup through a VoIP gateway) and I get no DTMF. I call from one xlite workstation to another through Asterisk, no DTMF |
| 21:29.45 | Swiss_asterisk | eKo1, whats *. ? |
| 21:29.47 | MaxeyPad | of the major players like packet8, vonage, broadvoice, which works the best with asterisk |
| 21:30.00 | eKo1 | * = asterisk. Doh! |
| 21:30.13 | eKo1 | i mean Duh |
| 21:30.17 | Swiss_asterisk | lol |
| 21:30.19 | Swiss_asterisk | ok :) |
| 21:30.27 | Seyr | packet8 supports asterisk? they told me you had to have one of their phones or an adapter...... |
| 21:30.35 | Swiss_asterisk | eKo1, does it work with prepay? |
| 21:30.36 | Moc[Train] | hehe |
| 21:30.39 | eKo1 | packet8 DOES NOT work with *. |
| 21:30.50 | eKo1 | Swiss_asterisk: both pre and post. |
| 21:30.58 | MaxeyPad | eKo1: do any of the commercial services like I listed work |
| 21:31.05 | Seyr | BroadVoice works |
| 21:31.11 | Swiss_asterisk | eKo1, i'm interested in seeing it closer, can we discuss it |
| 21:31.12 | Seyr | with Asterisk. I use it right now |
| 21:31.19 | eKo1 | Swiss_asterisk: no. |
| 21:31.47 | eKo1 | It's a big headache of a program and it doesn't work correctly right now. |
| 21:31.57 | Seyr | Is there any PAID support for Asterisk where I might get my problem addressed? |
| 21:32.02 | Swiss_asterisk | eKo1, i got 2 developers to share |
| 21:32.10 | Swiss_asterisk | eKo1, what language do u use? |
| 21:32.21 | ManxPower | Seyr: Digium. $150/hr I think |
| 21:32.32 | eKo1 | to speak, let's see....english, spanish, german. |
| 21:32.44 | Swiss_asterisk | eKo1, no, to program |
| 21:32.45 | Seyr | Thanks ManxPower |
| 21:32.51 | pino | syr: you mean the DTMF problem? |
| 21:32.52 | Swiss_asterisk | eKo1, are u from germany? |
| 21:32.56 | Seyr | yeh pino |
| 21:32.57 | *** part/#asterisk jcollie (~jcollie@dsl-ppp239.isunet.net) |
| 21:33.03 | ManxPower | But Since the issue is with X-Lite and not Asterisk.... |
| 21:33.06 | eKo1 | I use lots of languages to program. |
| 21:33.17 | Seyr | ManxPower: I said I call in on regular phone and it dont work |
| 21:33.20 | Swiss_asterisk | eKo1, particularly this billing system ? |
| 21:33.30 | pino | let's see if you can be helped for free while I wait for e164.org to call me back :) |
| 21:33.40 | ManxPower | Seyr: in [general] put context=INVALID and in each of the sip device sections put the correct context= line. If that breaks then X-lite is not providing username/password when making a call |
| 21:33.45 | eKo1 | php, pl/pgsql, c, perl. |
| 21:33.55 | Swiss_asterisk | eKo1, sounds good |
| 21:34.04 | ManxPower | Seyr: You have not told us anything about "regular phone" |
| 21:34.16 | Swiss_asterisk | eKo1, do u require assistance or could we sponsor development ? |
| 21:34.18 | Seyr | ManxPower: scroll back |
| 21:34.25 | pino | Seyr: can you pastebin your sip.conf and extensions.conf? |
| 21:34.27 | ManxPower | Seyr: how far back? |
| 21:34.40 | Seyr | ManxPower: plus ive been in this channel off and on for 8 hours talking about my problem :-) |
| 21:34.59 | eKo1 | Swiss_asterisk: this is a very custom made billing program to fit our companies stupid buisiness model which will change by the end of the year so...I don't recommend it. |
| 21:35.07 | Seyr | ManxPower: about 30 lines back? |
| 21:35.44 | Swiss_asterisk | eKo1, what would u recommend to start with if i would wish to pick a free solution and bring it to commercial level while keeping the license? |
| 21:36.05 | Swiss_asterisk | eKo1, whats most promissing implementation today ? |
| 21:36.08 | tainted- | Swiss_asterisk start with ASTCC |
| 21:36.15 | ManxPower | ~google site:lists.digium.com broadvoice dtmf problem |
| 21:36.15 | bugbot | google site:lists.digium.com broadvoice dtmf problem is assigned nothing and reported nothing. |
| 21:36.17 | tainted- | Swiss_asterisk and modify it to your needs |
| 21:36.18 | Swiss_asterisk | tainted-, thanks |
| 21:36.32 | eKo1 | Swiss_asterisk: well, you need to evaluate all the free solutions out there first. Then determine, based on your needs, what could work best. |
| 21:36.38 | PTG1234 | you know the providers need their own support channels, so we don't have to answer provider related questions |
| 21:36.43 | PTG1234 | no wonder all these providers are so lazy |
| 21:36.51 | Seyr | pino: I have BroadVoice -> Asterisk -> Speech Server | I call in and get no DTMF. I then installed xlite on 2 workstations and configured them as extensions and call either one and it seems like I get no DTMF. I currently have dtmfmode=rfc2833 on the Speech Server and both extenxions that i tried |
| 21:37.11 | *** join/#asterisk kFuQ (~somedude@c-24-17-224-78.hsd1.wa.comcast.net) [NETSPLIT VICTIM] |
| 21:37.11 | *** join/#asterisk LoRez (lorez@lorez.staff.freenode) [NETSPLIT VICTIM] |
| 21:37.15 | ManxPower | Seyr: There are many DTMF issues with broadvoice. |
| 21:37.19 | Swiss_asterisk | tainted-, eKo1 whats the general approach on terminating out-of-credit cards? |
| 21:37.36 | doughecka | speech server == microsoft? |
| 21:37.40 | Seyr | ManxPower: yeh, but if you google some more and look at different boards, they say they have been addressed |
| 21:37.40 | Swiss_asterisk | sending a break signal or starting the call with limited time according to left credits? |
| 21:37.46 | Seyr | which is why i went to test using xlite |
| 21:37.54 | Seyr | doughecka: yeh |
| 21:38.01 | doughecka | hows it work? |
| 21:38.02 | ManxPower | Did you do the context stuff I just told you to do? |
| 21:38.09 | MaxeyPad | So Asterisk works properly with broadvoice |
| 21:38.09 | tainted- | Swiss_asterisk end the call as soon as u can to avoid airtime charges |
| 21:38.22 | Seyr | doughecka: pretty damn good. solid as hell using analog. |
| 21:38.27 | doughecka | oh |
| 21:38.28 | doughecka | :) |
| 21:38.31 | Swiss_asterisk | tainted-, so asterisk supports external call termination ? |
| 21:38.39 | tainted- | MaxeyPad properly is an optimistic word to use |
| 21:38.47 | doughecka | does it need hardware or can it do only voip |
| 21:38.51 | eKo1 | funny, analog is usually a source of many a great problems. |
| 21:38.54 | Swiss_asterisk | tainted-, can radius server terminate an engaged call ? |
| 21:38.56 | tainted- | Swiss_asterisk it can limit the call duration based on credit yes |
| 21:39.10 | tainted- | Swiss_asterisk not directly.. but through API i'm sure it could |
| 21:39.11 | Seyr | doughecka: can do VoIP, but only supports vail systems for the telephony interface manager for voip |
| 21:39.11 | pino | seyr, i would try both dtmfmode=inband and dtmf=inband in [general], and dtmf=inband in your broadvoice section. |
| 21:39.13 | Swiss_asterisk | tainted-, ok, so the trick is to start call based on left credits.. |
| 21:39.21 | tainted- | Swiss_asterisk yes. |
| 21:39.28 | Swiss_asterisk | tainted-, thanks! |
| 21:39.33 | ManxPower | pino: dtmfmode=inband ONLY WORKS with ULAW or ALAW codec. |
| 21:39.36 | Swiss_asterisk | tainted-, look at other window |
| 21:39.45 | *** join/#asterisk UBiQUiTY (~mike@68.160.103.76) |
| 21:39.45 | doughecka | ah |
| 21:41.25 | *** part/#asterisk luciusism (~kahngl@a3.d5b7d1.client.atlantech.net) |
| 21:41.33 | pino | ManxPower: if broadvoice doesn't handle out-of-band DTMF, i don't see many options for receiving DTMF with another codec |
| 21:41.48 | UBiQUiTY | what does it mean if i see a "WARNING: Wait failed (Interrupted system call)" in my /var/log/asterisk/messages ? does anybody know? |
| 21:41.54 | ManxPower | pino: Then you have to use ulaw with them |
| 21:42.13 | pino | the question goes then back to seyr -- are you using ulaw/alaw? |
| 21:42.24 | ManxPower | I can't even remember I had a DTMF problem. |
| 21:43.36 | Seyr | using ulaw |
| 21:43.43 | ManxPower | I can't even remember the last time I had a DTMF problem. |
| 21:44.17 | `Sauron | You ARE one big DTMF problem. |
| 21:44.22 | `Sauron | ;) |
| 21:44.39 | ManxPower | All your DTMF belong to us. |
| 21:44.51 | *** join/#asterisk Skillabilities_N (~Skillabil@202-0-52-59.cable.paradise.net.nz) |
| 21:45.16 | TUplink | by chance is any one here on earthlink dialup? |
| 21:45.27 | pino | Seyr: have you tried those three inband options together? |
| 21:45.27 | ManxPower | Seyr: Did you do the context stuff I just told you to do? |
| 21:45.33 | UBiQUiTY | all your DTRM ARE belong to us.... get it right! |
| 21:45.35 | UBiQUiTY | :-D |
| 21:45.40 | UBiQUiTY | dtmf oops |
| 21:45.47 | Wazb | is there anyway to increase registration time of client with * |
| 21:45.56 | ManxPower | Wazb: on the client |
| 21:46.26 | Wazb | on * The amount will be in CND $ 62.42 |
| 21:46.26 | Wazb | US 50$ @ 1.2485 = 62.42 |
| 21:46.36 | Wazb | sorry about that |
| 21:46.41 | Wazb | i mean on * server |
| 21:48.20 | Fddayan | Somebody knows how can I print a message from AGI into the Console ??? |
| 21:48.37 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 21:48.37 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 21:48.38 | TEKjacob | Hey all, there are two LEDs on the back of the Digium T1 card. One right next to the jack and a smaller on down and to the right a bit. The small one is flashin red, is that a red alarm or something else? |
| 21:49.01 | UBiQUiTY | AGI VERBOSE |
| 21:49.06 | UBiQUiTY | (will print to console) |
| 21:49.20 | Strom_TM | that flashing light means there are saltine crackers in the 20th channel of the T1 frame |
| 21:49.47 | Strom_TM | you really want the saltines in the 12th channel. |
| 21:50.28 | Wazb | is there anyway to increase registration time of client in * |
| 21:50.36 | Elshar | For a second there I thought you were half serious. |
| 21:50.48 | Elshar | Took me a moment to realize that saltine crackers had nothing at all to do with t1 framing. :P |
| 21:50.53 | ManxPower | Wazb: Yes. |
| 21:51.08 | Strom_TM | hahaha |
| 21:51.16 | ManxPower | Wazb: In Asterisk, no. |
| 21:51.28 | Seyr | ManxPower: Apr 18 13:48:19 NOTICE[2603]: pbx.c:1329 pbx_extension_helper: Cannot find extension context 'INVALID' |
| 21:51.34 | *** join/#asterisk VoIPMasta (~John@201.137.25.71) |
| 21:51.36 | VoIPMasta | Hi |
| 21:51.54 | Seyr | pino: ive tried with all set to inband, yes |
| 21:51.56 | VoIPMasta | Does anyone know how can I route an incoming call received on a Zap device to my sip phone? |
| 21:52.22 | Seyr | im using the sample configs, just edited .. if that makes any difference |
| 21:52.26 | pino | seyr: can you at least hear them? |
| 21:52.40 | Seyr | pino: no |
| 21:52.45 | ManxPower | Seyr: Good! Now you have just determined that the X-lite client is NOT sending the correct username/password |
| 21:52.46 | *** join/#asterisk RChadwell (~rob@rrcs-24-227-48-86.se.biz.rr.com) |
| 21:53.04 | ManxPower | asuming you put the correct context= line in each sip peer/friend/user section |
| 21:53.08 | Seyr | ManxPower: ?? |
| 21:53.35 | Seyr | what would be the correct context line for the peer/friend/user ? |
| 21:53.39 | RChadwell | Has anyone done any testing on bit-depth / quality / bandwidth trade-offs for Music On Hold (SIP)? |
| 21:53.45 | pino | seyr: now either I'm missing something myself, or broadvoice is sending DTMF *not* inband. (this is a problem and it is orthogonal to the problem ManxPower is talking about.) |
| 21:53.47 | ManxPower | Seyr: if the sip client doesn't provide credentials then Asterisk will just accept the call and then use the stuff in [general] |
| 21:54.00 | ManxPower | Seyr: what WAS the context=line in [general] |
| 21:54.14 | Seyr | context=default |
| 21:54.20 | *** join/#asterisk Cardoe (~chatzilla@Cardoe.developer.gentoo) |
| 21:54.22 | pino | seyr: if you want some kind of precise answer, pastebin your sip.conf, extensions.conf AND output of "sip debug" :) |
| 21:54.32 | ManxPower | then that's the correct context=default for each of your sip.conf entries |
| 21:54.44 | Cardoe | is the getting started guide on asteriskdocs.org bad? |
| 21:55.01 | Cardoe | or is iaxtel poor? |
| 21:55.20 | Cardoe | cause I just installed Asterisk and got an IAXTel account and called Dell's 800 number to test. |
| 21:55.32 | Cardoe | and it's so choppy that I have no idea what the recording is saying |
| 21:55.46 | Strom_TM | I like to think of IAXTEL as the Yugo of VoIP |
| 21:55.49 | Seyr | ManxPower: ok, no errors that way, but no DTMF |
| 21:55.52 | RChadwell | ha ha |
| 21:55.58 | Cardoe | and asterisk after like 20 sec says that it exceeded the max retries to IAXTel and disconnects |
| 21:56.11 | Cardoe | Strom_TM: ? |
| 21:56.13 | ManxPower | Seyr: at least you have eliminated one possible issue |
| 21:56.14 | Seyr | ManxPower: context=default in the user sections and context=INVALID in general |
| 21:56.45 | RChadwell | The user sections will override the general setting |
| 21:56.48 | Seyr | pino: no clue what pastebin is |
| 21:56.49 | *** join/#asterisk pragueexpat (pragueexpa@r3p176.chello.upc.cz) |
| 21:56.53 | RChadwell | www.pastebin.ca |
| 21:57.03 | pino | Strom_TM: Yugos at least brought and still bring you from A to B, which IAXTel often refuses to do :) |
| 21:57.07 | VoIPMasta | ManxPower: can you help me with this one please? |
| 21:57.12 | RChadwell | post portions of your sip.conf there and people look at it (be sure to kill passwords and usernames, etc) |
| 21:57.20 | Strom_TM | hehehehh |
| 21:57.31 | pino | seyr: it helps you share config files without flooding the channel, basically. |
| 21:57.35 | ManxPower | VoIPMasta: Help you with what? |
| 21:58.06 | Cardoe | so basically just cause IAXTel sucked that's not what I should expect |
| 21:58.20 | VoIPMasta | ManxPower: I have 3 FXO cards and I want to be able to route incoming calls received on those FXO's to different sip extensions |
| 21:58.38 | ManxPower-grumpy | VoIPMasta: Sorry, that question is too basic for me to answer. |
| 21:59.07 | ManxPower-grumpy | But I'll give you a hint: put the different ports in different contexts |
| 21:59.11 | VoIPMasta | ManxPower-grumpy: I know how to route every incoming call using a common extension, but now I need to route them based on the ZAP where the call originated |
| 21:59.48 | VoIPMasta | ManxPower-grumpy: is it possible to set up each ZAP iterface in a different context in zapata.conf? |
| 21:59.51 | ManxPower-grumpy | VoIPMasta: Generally you don't. put the ports in different contexts |
| 22:00.20 | ManxPower-grumpy | VoIPMasta: Well duh! If you could not put different zap ports in different contexts Asterisk would be pretty useless. |
| 22:00.58 | *** join/#asterisk ToyMan (~konversat@user-12lcqur.cable.mindspring.com) |
| 22:02.24 | TUplink | ;) |
| 22:02.44 | VoIPMasta | ManxPower-grumpy: Ok, will try that, thanks a lot |
| 22:04.04 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 22:04.04 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:05.37 | *** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
| 22:05.57 | FuriousGeorge | hey all |
| 22:06.44 | Moc[Train] | hi |
| 22:07.09 | FuriousGeorge | even if asterisk isnt loaded, shouldnt a tdm400 send enough voltage to a handset so that it can dial buttons, etc. i cant even hear the "relay" of the mouth piece to the earpiece. the phone is totally dead |
| 22:07.14 | FuriousGeorge | it wasnt at some point |
| 22:07.28 | Moc[Train] | donno |
| 22:07.33 | *** join/#asterisk TEKjacob (~chris@70-32-12-155.frdrmd.adelphia.net) |
| 22:07.38 | Strom_TM | FuriousGeorge, that "relay" is called sidetone |
| 22:08.00 | Strom_TM | and the tdm400 will provide talk battery if you have the drivers loaded |
| 22:08.05 | FuriousGeorge | Strom_TM: thanks |
| 22:08.23 | FuriousGeorge | the driver is definately loaded, and the handset sounds dead, is the phone broke |
| 22:08.33 | Strom_TM | try a different telephone |
| 22:08.45 | Strom_TM | or try the telephone on a POTS line |
| 22:08.58 | Moc[Train] | brb dinner will be serve on the train soon |
| 22:08.59 | FuriousGeorge | dont have a pots line, think i may have a phone in the basement |
| 22:09.10 | Strom_TM | or make sure the phone is plugged into the right port on the tdm400 |
| 22:09.43 | FuriousGeorge | my configs are at a point where it should give me a dialtone (its in the right port) but its totally dead |
| 22:10.02 | *** part/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 22:10.06 | eKo1 | welcome to the work of analog... |
| 22:10.07 | FuriousGeorge | brb going to dig up other phone |
| 22:10.08 | Strom_TM | you sure the mounting cord isn't bad? |
| 22:10.47 | FuriousGeorge | Strom_TM: i can change the cord, if thats what you mean |
| 22:10.53 | FuriousGeorge | will try that 1st |
| 22:11.12 | Strom_TM | out of curiosity, what kind of telephone set is it? |
| 22:11.39 | ManxPower-grumpy | FuriousGeorge: It's easy to confuse the FXO/FXS ports on the TDM400P cards. |
| 22:11.46 | ManxPower-grumpy | The TOP port on the card is port 1 |
| 22:12.26 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 22:12.43 | ManxPower-grumpy | also if you accidently plug a phone line into an FXS port of the TDM400P card, and the phone line rings you will blow up the port and have to have it replaced |
| 22:13.28 | eKo1 | really? will it make a loud *pop* noise? |
| 22:13.45 | Strom_TM | moral of story: dont put FXO and FXS modules on the same card :) |
| 22:13.48 | ManxPower-grumpy | eKo1: no idea. |
| 22:14.03 | ManxPower-grumpy | Strom_TM: Well, the moral of the story is not to plug a phone line into an FXS port. |
| 22:14.09 | drumkilla | yeah, FXS to FXS == game over |
| 22:14.42 | shmaltz | is there any way I can test in the dialpaln if asterisk returns -1? |
| 22:15.02 | drumkilla | if it returns -1, the call ends |
| 22:15.12 | eKo1 | shmaltz: yes, you receive a han |
| 22:15.44 | eKo1 | err, you receive a '...exited non-zero on channel...' message |
| 22:15.53 | shmaltz | ;p |
| 22:15.56 | VoIPMasta | ManxPower-grumpy: It worked, thanks a lot man, you've saved me once again |
| 22:16.57 | eKo1 | So, anybody use Clipcomm hardware here today? |
| 22:17.24 | FuriousGeorge | ManxPower-grumpy: the green daughterchip is the one u connect to an analog phone right, |
| 22:17.46 | ManxPower-grumpy | FuriousGeorge: Yes. |
| 22:18.36 | *** part/#asterisk RChadwell (~rob@rrcs-24-227-48-86.se.biz.rr.com) |
| 22:19.16 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 22:20.24 | Pinhole | I was using google to find some info on asterisk and found somebody with a wiki resume. So I gave him experience with resume fraud and took away his security experience. |
| 22:20.48 | eKo1 | Evil... |
| 22:20.58 | Strom_TM | hahaha |
| 22:21.14 | Pinhole | now I feel guilty. I didn't expect it to let me. |
| 22:21.15 | ManxPower-grumpy | LOL! |
| 22:21.23 | ManxPower-grumpy | Don't. |
| 22:21.35 | eKo1 | well, that'll teach'm to use a wiki for posting resumes |
| 22:21.38 | ManxPower-grumpy | People should expect unsecured stuff to have proble s |
| 22:21.41 | *** join/#asterisk cpatry (~JunK-Y@modemcable174.107-81-70.mc.videotron.ca) |
| 22:21.42 | *** join/#asterisk aspworld (~aspworld@209.91.159.221) |
| 22:21.42 | Strom_TM | yes, if anyone is dumb enough to put a resume in a wiki... |
| 22:22.03 | *** part/#asterisk aspworld (~aspworld@209.91.159.221) |
| 22:22.57 | Nugget | my resume is in cvs. :) |
| 22:24.05 | *** join/#asterisk bjohnson (~bjohnson@ip159-181.tor.istop.com) |
| 22:24.17 | Pinhole | my resume...hmmm, last updated in 1997. |
| 22:24.29 | Nugget | Pinhole is my hero. |
| 22:25.36 | eKo1 | Man, I'm bored. I think I'll start doing some linear algebra. |
| 22:26.26 | eKo1 | eh, no such thing. |
| 22:26.40 | Pinhole | er, scratch that one from my resume. |
| 22:27.14 | Pinhole | Acutally, I guess what really gets you a job is putting those accents on resume. |
| 22:27.22 | eKo1 | if only there was an #algebra channel on freenode... |
| 22:27.33 | BoRiS | Anyone have a newer beta firmware for the Senao SI-7800 wifi phone then 0.03.0008 (2004/10/17)?? msg me (thanks in advance) |
| 22:27.38 | Nugget | résumé |
| 22:27.46 | eKo1 | you mean résume |
| 22:28.26 | Pinhole | doesn't resume mean shopping list or something like that? |
| 22:29.17 | *** part/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
| 22:29.24 | eKo1 | eh, wtf are you talking about? |
| 22:32.39 | eKo1 | Anyone using Mediatrix gateways here? |
| 22:35.38 | *** join/#asterisk jdiskywlkr (~kvirc@ip68-0-90-1.tu.ok.cox.net) |
| 22:38.14 | *** join/#asterisk tld (~terje@80.203.70.227) |
| 22:39.16 | eKo1 | Guess not. |
| 22:39.58 | tld | Asterisk is a PBX, so typically my UA phone would connect to the Asterisk, and the Asterisk would connect to whereever I'm calling, right? But if both the phone and the gateway I point the Asterisk to supports SIP and the same codecs, will I still have to run the voice channel through the Asterisk, or can it be sent straight between my phone and the gateway? |
| 22:40.38 | FuriousGeorge | Strom_TM (or anyone): the lights on the tdm are green, lsmod shows the drivers are loaded, ive tried several phones, and several cords, this was working at some point as far as having sidetone, but now everything is totally dead |
| 22:40.59 | Strom_TM | and you tried all the ports, FuriousGeorge? |
| 22:41.21 | FuriousGeorge | no just the two that i have green fxs daughter chips |
| 22:41.31 | FuriousGeorge | which have leds=on |
| 22:41.44 | Silik0n | did you powerthe card? |
| 22:41.45 | Strom_TM | do you have anything in the other ports? |
| 22:41.55 | Strom_TM | yes, did you plug a power cord into the card? |
| 22:41.57 | Silik0n | of coursethe the driver usually bitches about that if you dont |
| 22:41.57 | FuriousGeorge | Silik0n: yeah, the leds are kickin |
| 22:42.02 | *** join/#asterisk pbxjunkie (~Stormtroo@ppp14-adsl-159.ath.forthnet.gr) |
| 22:42.06 | *** join/#asterisk kFuQ (~somedude@c-24-17-224-78.hsd1.wa.comcast.net) [NETSPLIT VICTIM] |
| 22:42.06 | *** join/#asterisk LoRez (lorez@lorez.staff.freenode) [NETSPLIT VICTIM] |
| 22:42.12 | FuriousGeorge | Silik0n: it did when i forgot |
| 22:42.36 | Silik0n | did you plug a PSTN line into the wrong port and call it there by blowing up the rslics? |
| 22:42.41 | FuriousGeorge | i happen to have another tdm... guess i should switcharoo |
| 22:43.17 | FuriousGeorge | Silik0n: whats "blowing up the rslics." is that anything like "blowing up the spot" |
| 22:43.55 | FuriousGeorge | i should say this: is there any way this can be caused by a misconfiguration of some file |
| 22:44.06 | FuriousGeorge | i think it should be working, but maybe i fubared it |
| 22:45.26 | FuriousGeorge | i should also say this: ive never had this working before, i just got these cards, so i have no idea how to go about troubleshooting no sidetone, so if it could be the config files let me know |
| 22:45.43 | FuriousGeorge | please |
| 22:45.53 | FuriousGeorge | ill start from scratch |
| 22:46.13 | *** join/#asterisk niZon (ilt@S0106deadbeef6977.wp.shawcable.net) |
| 22:50.06 | FuriousGeorge | let me rephrase, is there anyway my lack of sidetone is caused by a slipup in some config file? i think my config files are ok, but having just got the cards, and having never set up an analog client, i have no experience to draw on. |
| 22:50.26 | FuriousGeorge | or should just loading the driver provide sidetone, regardless |
| 22:50.29 | Strom_TM | FuriousGeorge, what kind of telephone set is it? |
| 22:50.36 | FuriousGeorge | i have three differrnt |
| 22:50.38 | FuriousGeorge | phones |
| 22:50.45 | Strom_TM | what are they? |
| 22:51.00 | FuriousGeorge | none work, one is 20 years old, the other is a handset from a clock phone, the third looks like its from a pbx |
| 22:51.26 | FuriousGeorge | the clockphone only works with one cord, i have another regular cord, and a jack i made out of a thrid cord for the 20 year oldmphone |
| 22:51.40 | Strom_TM | is the 20 year old one manufactured by Western Electric, AT&T, ITT, Stromberg-Carlson, and/or Northern Telecom? |
| 22:51.59 | FuriousGeorge | ATT |
| 22:52.20 | Strom_TM | http://stromcarlson.com/misc/2500SM.jpg |
| 22:52.23 | Strom_TM | like that one? |
| 22:52.39 | FuriousGeorge | its wierd, i ahd to make a jack out of one of the wires bc the rj11 is male |
| 22:53.13 | FuriousGeorge | no just a handset and a similar shaped base, numbers on the handset |
| 22:53.24 | FuriousGeorge | tried putting handet directly into box |
| 22:53.30 | Strom_TM | http://stromcarlson.com/misc/trimline.jpg |
| 22:53.37 | Strom_TM | like that? |
| 22:53.49 | FuriousGeorge | but not radary |
| 22:53.51 | FuriousGeorge | rodary |
| 22:53.52 | FuriousGeorge | whatever |
| 22:53.54 | Strom_TM | rotary |
| 22:53.56 | Strom_TM | yeah |
| 22:54.05 | FuriousGeorge | exactly |
| 22:54.14 | FuriousGeorge | and tan not red :) |
| 22:54.24 | Strom_TM | ok...you cant plug the handset directly into the line. Does the trimline work on a POTS line? |
| 22:54.27 | darwin35 | ahh ok |
| 22:54.36 | darwin35 | wrong window |
| 22:54.40 | *** join/#asterisk kFuQ (~somedude@c-24-17-224-78.hsd1.wa.comcast.net) [NETSPLIT VICTIM] |
| 22:54.40 | *** join/#asterisk LoRez (lorez@lorez.staff.freenode) [NETSPLIT VICTIM] |
| 22:54.51 | FuriousGeorge | yeah, it did as recently as 2 months ago, im gonna take all these to my aprents house and try it there |
| 22:55.11 | Strom_TM | you dont have one at your house you can test with? |
| 22:55.18 | FuriousGeorge | no landline here |
| 22:55.25 | FuriousGeorge | gonna try changing the card and chips |
| 22:55.32 | FuriousGeorge | what else is there to do really |
| 22:55.57 | FuriousGeorge | if the config files have nothing to do with it, the leds are on, and lsmod shows the right driver |
| 22:56.06 | *** join/#asterisk LoRez (lorez@lorez.staff.freenode) [NETSPLIT VICTIM] |
| 22:56.09 | Strom_TM | try the other card...if that doesnt work either, then it might be a config problem |
| 22:56.28 | *** join/#asterisk kFuQ (~somedude@c-24-17-224-78.hsd1.wa.comcast.net) [NETSPLIT VICTIM] |
| 22:56.40 | FuriousGeorge | if it could be a config problem i sould like to try fresh config files, i was trying to figure out if that was ruled out |
| 22:56.59 | Strom_TM | try the hardware first. |
| 22:57.10 | FuriousGeorge | k, ill brb |
| 22:57.34 | *** join/#asterisk _GiGi_ (gigi@jabber.szczecin.pl) |
| 22:57.46 | *** part/#asterisk darwin35 (~darwin35@24.3.226.147) |
| 22:57.51 | *** join/#asterisk fugitivo (~ajf@201.255.107.24) |
| 22:57.55 | FuriousGeorge | one quick unrelated ?, if thats ok. i tell my bios to reserve irq10 to pci slot 4, since its the only slot that doesnt share drivers by default, and cat proc/interrupts shows irq10 is free |
| 22:58.02 | FuriousGeorge | when i load the module, it goes to irq 3 |
| 22:58.42 | Strom_TM | *shrug* I just let the irq garbage autoconfigure itself |
| 22:59.04 | FuriousGeorge | garbage is a great way to describe it. oh well, brb |
| 23:00.32 | *** join/#asterisk Derkommissar (~alberto@66.64.215.7.nw.nuvox.net) |
| 23:00.34 | Derkommissar | Hello |
| 23:01.03 | Derkommissar | all of the sudden out of nothig, calls to phones registerd in my asterisk box say this.... Apr 18 18:56:07 NOTICE[22736]: app_dial.c:884 dial_exec_full: Unable to create channel of type 'SIP' (cause 3),,,,,, what can be causing this ? |
| 23:01.07 | Derkommissar | Apr 18 18:56:07 NOTICE[22736]: app_dial.c:884 dial_exec_full: Unable to create channel of type 'SIP' (cause 3) |
| 23:01.16 | anti | anyone have a problem with speex and asterisk segfaulting whenever a call is placed the speex codec is used? before the call connects, it segfaults inside of libspeex in fir_mem2_10 |
| 23:02.24 | CoaxD | anti: Remember to remove /usr/lib/asterisk/modules before you do a 'make install' on a new asterisk tree |
| 23:02.36 | CoaxD | anti: If you do that and it still fails, then its a bug |
| 23:03.14 | CoaxD | anti: (asterisk could be attempting a reference to a symbol that no longer exists in a new version of a shared library) |
| 23:03.16 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
| 23:03.54 | fugitivo | anyone tried the Sipura SPA-841 hardphone? |
| 23:05.10 | *** join/#asterisk albmonkey (~alb@64-252-128-73.adsl.snet.net) |
| 23:10.04 | anti | CoaxD: yeah, unfortunately still having the problem.. |
| 23:10.33 | anti | I wonder if its because I have sse support compiled into speex |
| 23:10.48 | *** join/#asterisk harryvv (~leonardo@S010600055d210201.vs.shawcable.net) |
| 23:11.33 | *** join/#asterisk christo (~christo@courgette.jml.net) |
| 23:11.35 | christo | aye |
| 23:11.38 | christo | wiki down again ? |
| 23:11.59 | denon | yes |
| 23:12.28 | denon | we've decided to add ipfw rules to block people we dont like |
| 23:12.32 | *** part/#asterisk sudoer (~toy@denali.ccs.neu.edu) |
| 23:12.36 | christo | :) |
| 23:12.49 | christo | where can I download the sources for Ztdummy? |
| 23:12.56 | anti | Ah ha! |
| 23:14.04 | denon | I thought it came with zap stuff |
| 23:14.05 | L|NUX | christo : cvs.digium.com or ftp.digium.com |
| 23:14.06 | L|NUX | ;) |
| 23:16.03 | *** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
| 23:17.07 | FuriousGeorge | Strom_TM: things seem to go from bad to worse, i took the one tdm out, put the two fxs in the fisrst and second bays, boot my computer, and i dont even have LEDS on this card |
| 23:17.32 | FuriousGeorge | i checked and zaptel, and wcfxs are loaded in lsmod |
| 23:17.56 | FuriousGeorge | but nothing registered in cat proc/interrupts |
| 23:18.12 | FuriousGeorge | is this a sign from God? |
| 23:18.36 | albmonkey | I have a totally off topic question, unfortunately this was the best place I could think of to look for help. I don't suppose there are any Mitel expers present? |
| 23:19.14 | FuriousGeorge | Strom_TM: forget everything i just said, i forgot to put the pwoer cord in |
| 23:19.15 | FuriousGeorge | brb |
| 23:19.59 | harryvv | What would cause a dialplan extention to not load zap apon a failover from a iax.cc call ? the zap extension matches the same local zap extensions for my local calling area and both the context are inclided? |
| 23:20.03 | *** join/#asterisk heison (~heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 23:20.04 | harryvv | err |
| 23:20.19 | harryvv | zap is loaded but the dialplan in this one example is not calling it |
| 23:21.22 | *** join/#asterisk ScythelX (Fleb@pc-24-181-176-181.sbi.ct.charter.com) |
| 23:21.32 | ScythelX | has anyone noticed problems with nufone lately |
| 23:22.54 | *** join/#asterisk captrb (~crozierm@64.65.134.42) |
| 23:23.02 | anti | so yeah, speex + sse + asterisk = segfault fun. |
| 23:23.03 | harryvv | listening to a past recording of mark spencer on this over the net radio show. |
| 23:23.25 | harryvv | anti is this the only case for you ? |
| 23:23.27 | captrb | howdy, does anybody use the polycom handsets and have a headset that they prefer? |
| 23:23.47 | tainted- | lol |
| 23:23.48 | *** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
| 23:23.52 | tainted- | anyone noticing the msn virus? |
| 23:24.00 | harryvv | nooooooo |
| 23:24.32 | anti | harryvv: recompiled speex and asterisk several times, removed them completely, installed from scratch. Every time I placed a call that used speex, asterisk would segfault. |
| 23:24.41 | anti | harryvv: finally I recompile speex without sse support, all is well. |
| 23:24.58 | harryvv | man dont even talk about that. tainted when i worked there anytime a virus hit there exchange servers we got all kind of notification from emails to voice mails almost to a knock on your door DONT open that email that says blaa blaa because its a marco virus! :) |
| 23:25.24 | harryvv | when a virus spread at microsoft it spread fast :) |
| 23:25.42 | harryvv | never used speex |
| 23:26.10 | *** join/#asterisk docelmo (~me@116-39.202-68.tampabay.res.rr.com) |
| 23:26.55 | anti | sound quality is very nice |
| 23:27.26 | harryvv | ohh asterisk has alarm monitor capability? nice. |
| 23:28.31 | docelmo | Does anyone know if NuFone supports SIP? Im having issues with IAX and them |
| 23:28.46 | harryvv | doc whatds the problem |
| 23:28.54 | harryvv | dones a iax2 show registry? |
| 23:29.04 | ScythelX | my connection with them has been very choppy lately |
| 23:30.03 | harryvv | when? |
| 23:30.33 | docelmo | harry cant get my TF # to work |
| 23:30.56 | *** join/#asterisk mgth (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net) |
| 23:31.11 | docelmo | It registers but when I try to call it says incompatible codec. But I am accept ULAW and G729 |
| 23:32.12 | *** join/#asterisk Moc[Train] (~mochouina@png1.pointshotwireless.com) |
| 23:32.22 | harryvv | try different codecs |
| 23:32.51 | docelmo | Hmm..... |
| 23:33.00 | Moc[Train] | im back |
| 23:33.19 | Moc[Train] | diner wasnt bad hehe |
| 23:33.55 | Moc[Train] | can't get msn to work thought !!! |
| 23:34.03 | file[laptop] | Moc[Train]: how much is the internet costing you? |
| 23:34.04 | cpatry | moc: t ou pour etre en train^ |
| 23:34.21 | Moc[Train] | cpatry, ver toronto |
| 23:34.25 | Moc[Train] | file[laptop], free |
| 23:34.32 | file[laptop] | cool |
| 23:34.44 | Moc[Train] | file[laptop], well it firstclass so |
| 23:34.51 | Moc[Train] | I payed for it ;) |
| 23:34.53 | file[laptop] | haha |
| 23:35.28 | docelmo | Is Jer alive? |
| 23:35.32 | Moc[Train] | I should arrive at 9pm |
| 23:36.26 | docelmo | anyone in here using NuFone getting stuck using ilbc? |
| 23:36.43 | ScythelX | i thought they only do gsm now |
| 23:37.26 | docelmo | sigh.. GSM sucks.. |
| 23:37.33 | harryvv | man been years since i was on a amtrak. Seen alot of cool sites across the west coast. Moc what direction are going from and to? |
| 23:37.37 | docelmo | I was MEGA choppy |
| 23:37.40 | harryvv | gsm is okay |
| 23:37.54 | Strom_TM | gsm is ok. on mobile phones. |
| 23:38.05 | harryvv | used alot on mobile phones |
| 23:38.13 | Strom_TM | on a desk set, gsm is like stabbing yourself in the eye with a white hot poker |
| 23:38.43 | JerJer | ScythelX: we do any codec you want |
| 23:39.51 | niZon | hmm, I wonder where I could get the SIP firmware for a 7905 |
| 23:39.58 | JerJer | cisco.com |
| 23:40.06 | niZon | without buying a crappy service contract for x hundread $ |
| 23:41.25 | *** join/#asterisk dmccollum (~dmccollum@69-164-245-72.atlaga.adelphia.net) |
| 23:42.51 | Luhiwu | what codec is better, gsm or ilbc? ilbc seems pretty expensive in terms of cpu time |
| 23:42.51 | |Vulture| | niZon: no where |
| 23:42.58 | |Vulture| | ilbc |
| 23:43.25 | JerJer | define better |
| 23:43.46 | JerJer | i believe iLBC can achieve a higher MOS score at the cost of more cpu |
| 23:43.49 | |Vulture| | ilbc > voice quality but uses more CPU, and is free |
| 23:43.54 | Luhiwu | sound quality |
| 23:44.10 | |Vulture| | ilbc is much better on sound quality than gsm imo |
| 23:44.19 | bjohnson | ulaw is best |
| 23:44.21 | Strom_TM | ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw |
| 23:44.23 | Luhiwu | i see the more cpu used, i have 5ms from ulaw to gsm and 70 to ilbc :) |
| 23:44.29 | Strom_TM | ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw :) |
| 23:44.36 | |Vulture| | ulaw is a waste of bandwidth |
| 23:44.36 | harryvv | ulaw is best or ulaw is the best at compromise? |
| 23:44.38 | file[laptop] | Strom likes ulaw |
| 23:45.16 | Strom_TM | ulaw sounds best |
| 23:45.16 | |Vulture| | of course |
| 23:45.16 | harryvv | and of course the tops in voice quality is g729 |
| 23:45.56 | Moc[Train] | ?? |
| 23:46.01 | Moc[Train] | I only use ulaw |
| 23:46.11 | *** join/#asterisk Barmal (~1@adsl-19-109-17.asm.bellsouth.net) |
| 23:46.12 | Strom_TM | ulaw is love |
| 23:46.19 | file[laptop] | ulaw is sexy baby |
| 23:46.22 | |Vulture| | I use ulaw internal and ilbc external |
| 23:46.34 | Strom_TM | anything else is like assraping the other party with a christmas tree |
| 23:47.10 | Barmal | whats cooking? |
| 23:47.35 | file[laptop] | your toes |
| 23:47.42 | file[laptop] | you stepped in the molten lava |
| 23:47.56 | Barmal | tzanger: hey this script you gave me last week it returns every second call right? |
| 23:49.23 | tzanger | Barmal: eh? |
| 23:49.50 | harryvv | what would cause the "no authority found message" in a iax.cc connection? |
| 23:50.04 | Barmal | the callback script you gave me last time. www.pastebin.ca/9612 |
| 23:50.07 | ariel_ | anyone played with the Wireless data network from Verizon or Cingular? 300kb data rates does not sound too bad? |
| 23:51.27 | ariel_ | ilbc takes allot of CPU power for my use. |
| 23:52.36 | Barmal | tzanger: shoot wasn't it you then? |
| 23:52.44 | tzanger | yes it was me |
| 23:52.53 | tzanger | what do you mean "every second call" though? |
| 23:53.40 | Barmal | I call leave a message nothing happens only that it records a message. Second time I call it leaves a message and then returns my call stating that it has 2 new messages |
| 23:53.44 | *** join/#asterisk MrBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net) |
| 23:53.57 | Barmal | does it suppose to be this way? |
| 23:54.13 | *** join/#asterisk MatsK (~matsk@107.80-202-57.nextgentel.com) |
| 23:54.40 | *** part/#asterisk Romik (~romik@1.fix.netvision.net.il) |
| 23:54.53 | |Vulture| | http://store.ultraspec.us/wildcardte110p.html does that look ligit? its like $60 less than everyone else |
| 23:54.58 | harryvv | anyone seeing any issues with iax.cc? |
| 23:55.02 | docelmo | I would like to use ULAW.. or hell even g729.. but ohh well |
| 23:57.03 | *** join/#asterisk nextime (~nextime@213-140-6-96.fastres.net) |
| 23:57.26 | PTG1234 | anyone know any good graphic artists? |
| 23:57.34 | harryvv | for what |
| 23:57.55 | harryvv | 2s 3d |
| 23:57.57 | harryvv | 2d |
| 23:58.19 | PTG1234 | 2d, website design |
| 23:59.30 | Barmal | tzanger: or maybe because it starts to call back before the message is left... That me be why... |
| 23:59.33 | tzanger | Barmal: uh... it polls the given mailbox's NEW directory and if anything is in there it generates the .call file. it knows nothing about "every other call" |
| 23:59.34 | Silik0n | how much you wanna pay? |
| 23:59.59 | tzanger | Barmal: depends on how new messages are stored. as soon as there is something in NEW/ there is the possibility of a callback |
