| 00:00.18 | Evanrude|w | bile_one, running web-server as 'asterisk' fixed the problem |
| 00:00.22 | _Raptor_ | cu |
| 00:00.32 | bile_one | Evanrude sweet. |
| 00:02.00 | *** join/#asterisk mentat (~Mentat@pcp01260498pcs.nhaven01.ct.comcast.net) |
| 00:02.04 | mentat | hi all |
| 00:02.24 | bile_one | Pinhole looks like Weather.php is using text2wav which is festival |
| 00:02.30 | mentat | What should I do if my ixj's volume is just ridiculously low? |
| 00:03.05 | Evanrude|w | well time to call it a day - thanks bile_one |
| 00:03.33 | bile_one | Your're welcome Evanrude |
| 00:03.44 | Evanrude|w | l8r |
| 00:04.08 | sivana | ya.. I just got Linksys approval |
| 00:04.34 | sivana | yay |
| 00:04.40 | *** join/#asterisk __Sparks_ (ringding@bb-195-172-50-212.ukonline.co.uk) |
| 00:04.45 | Pinhole | bile_one, it can use either. You just use a diferent function. |
| 00:05.18 | bile_one | I see than now reading the souce code. Thanx Pinhole |
| 00:05.40 | opus___ | Hmmm |
| 00:05.46 | opus___ | STill problems with broadvoice outgoing |
| 00:06.01 | Pinhole | You can use swift when ever you see text2wav. |
| 00:06.04 | opus___ | Does anyone know if it is neccessary to have a patch for outgoing broadvoice? |
| 00:06.24 | __Sparks_ | Hi, I seem to be havig trouble with having multiple SIP accounts registering together - I get the following errors - chan_sip.c:6801 handle_response: Failed to authenticate on REGISTER to '<sip:SIP ACCOUNT DETAILS HERE> - Do I need to use differnet ports for each one - currently they are all using 5060 |
| 00:06.32 | opus___ | If I have canreinvite=yes for the sip phone that asterisk serves to, it trys to connect to 172.16.203.1 which isn't in my network |
| 00:06.42 | opus___ | If I have canreinvite=no I get a 400 request not found from broadvoice |
| 00:06.51 | DannyF | anyone in here that checked out a clean HEAD and has/had problems with audio not going through even thou all legs gets connected? |
| 00:07.01 | bile_one | later all |
| 00:07.04 | Mw3 | we would like to interconnect with pstn network over ss7. i've read in the wiki that there is some progress (beta version) about it. where can i find that beta ? |
| 00:07.35 | DannyF | Mw3, did u search the lists? was alot of talk about that a while back |
| 00:07.48 | DannyF | seams Digium supports that |
| 00:08.10 | DannyF | someone said they showed it at VON |
| 00:08.39 | mentat | So ideas on the IXJ? I can just barely hear it, and I mean barely |
| 00:08.49 | mentat | Software settings, hardware settings? |
| 00:09.26 | opus___ | mentat sorry i don't know, but you might check to make sure it has a common ground. my wildest guess |
| 00:10.00 | mentat | internet phone jack |
| 00:10.13 | mentat | supported by linux telephony driver |
| 00:10.14 | *** join/#asterisk harryvv (~leonardo@S010600055d210201.vs.shawcable.net) |
| 00:10.23 | jontow | ok |
| 00:10.26 | mentat | It seems to work fine, just really really soft |
| 00:10.31 | jontow | in HEAD .. why must chan_oss.c load even when i use 'noload' on it? :/ |
| 00:11.08 | opus___ | does anyone know how to get a hold of Dimitris Kounalakis |
| 00:11.30 | stevekstevek | opus___: by the neck? |
| 00:11.31 | jontow | also with ICD .. icd_agent_callback() crashes * when you try to log an agent out :( |
| 00:11.39 | bbgun | Does anyone know how to display Caller names instead of their extension number? I can't find the information :) |
| 00:11.44 | jontow | and the call just hangs. |
| 00:12.12 | jontow | :( |
| 00:12.19 | opus___ | stevekstevek -- because of spam protection, the mailing list remove his e-mail and I don't have archives before the 13th |
| 00:13.12 | opus___ | he posted a patch to the mailing list that got destroyed |
| 00:13.14 | opus___ | http://www.mail-archive.com/asterisk-users@lists.digium.com/msg81453.html |
| 00:13.20 | opus___ | I can't extract it :( |
| 00:13.52 | __Sparks_ | When i am registering with multiple SIP accounts - do I need to use different ports foe each? |
| 00:14.16 | Juggie | no |
| 00:15.30 | __Sparks_ | Juggie - any idea why when regerstering with multiple sipgate accounts I get "Failed to authenticate on REGISTER" ? |
| 00:17.07 | __Sparks_ | and also while it is thinking about it, asterisk freezes (If i call my voicemail while it is doing it, the call doesnt connect until after it has finished erroring) |
| 00:18.22 | __Sparks_ | Can anyobe help me with this, it has been a problem for me for ages! |
| 00:21.34 | tuxinator_linux | __Sparks_: Have you checked bugs.digium.com? |
| 00:22.52 | __Sparks_ | tuxinator_linux - Nope, but I am there now! |
| 00:22.55 | tuxinator_linux | __Sparks_: Running stable or HEAD? |
| 00:23.37 | __Sparks_ | tuxinator_linux - Not quite sure what you mean (I am fairly new to * :-) |
| 00:23.49 | tuxinator_linux | How did you install it? |
| 00:24.07 | tuxinator_linux | CVS checkout asterisk? |
| 00:24.17 | __Sparks_ | Xorcom Rapid |
| 00:24.35 | __Sparks_ | downloaded the ISO and istalled it from that - |
| 00:24.54 | tuxinator_linux | Not familiar with that one |
| 00:25.01 | __Sparks_ | (I guess thats cheating :-) |
| 00:25.13 | __Sparks_ | http://www.xorcom.com/ |
| 00:25.17 | tuxinator_linux | what does asterisk -v return? |
| 00:26.55 | brc_ | ~seen slePP |
| 00:26.57 | jbot | slepp is currently on #asterisk (8h 2m 27s) |
| 00:27.02 | __Sparks_ | can I PM you with the results, rather then paste in here? |
| 00:27.04 | *** join/#asterisk bjohnson (~bjohnson@66.11.165.161) |
| 00:27.21 | *** join/#asterisk anthm (~anthm@70.8.109.116) |
| 00:27.21 | *** mode/#asterisk [+o anthm] by ChanServ |
| 00:27.23 | jedaustin | Sparks: asterisk@home? |
| 00:27.47 | __Sparks_ | jedaustin - as in am I using it at home - yes |
| 00:27.55 | *** join/#asterisk r0d3nt|m (anonymous@soveliss.luniac.com) |
| 00:28.17 | tuxinator_linux | he he he |
| 00:28.17 | jedaustin | Sparks: thats what the iso's called that I downloaded |
| 00:28.22 | jedaustin | :) |
| 00:28.47 | __Sparks_ | jedaustin - oh right - no, the one I used was Xorcom Rapid |
| 00:29.25 | DannyF | anyone in here that checked out a clean HEAD and has/had problems with audio not going through even thou all legs gets connected? |
| 00:33.32 | TedC | anybody here know the app_queue code well? |
| 00:34.12 | mentat | arg, now dtmf's are going through on the ixj... |
| 00:34.36 | *** join/#asterisk pcm (~pcm@user-69-73-0-22.knology.net) |
| 00:36.22 | *** join/#asterisk greg_work (~greg@d221-73-198.commercial.cgocable.net) |
| 00:37.18 | sivana | in your fridge?! Brilliant!! |
| 00:38.08 | DannyF | bloody audio is only going TO the * all of a sudden |
| 00:41.41 | *** join/#asterisk rvhi (~rv@66.175.65.89) |
| 00:47.47 | Damin | You ready to rock and roll at 9:00 PM? |
| 00:49.24 | tzanger | who |
| 00:50.06 | *** join/#asterisk peted20 (~chatzilla@24-113-67-25.wavecable.com) |
| 00:52.33 | peted20 | anyone here know how to apply patches to asterisk? |
| 00:52.41 | peted20 | I want to try the new jitterbuffer |
| 00:53.31 | tzanger | peted20: you don't need to |
| 00:53.34 | tzanger | it's in CVS HEAD now |
| 00:53.41 | tzanger | as of like 2.5 hours ago |
| 00:54.09 | peted20 | tzanger: wow, thats awesome! |
| 00:54.15 | tzanger | damn skippy |
| 00:55.06 | peted20 | tzanger: if I just pull from CVS without specifying a branch (i.e 1.0), that is CVS HEAD? |
| 00:55.11 | harryvv | tzanger, I have not seen any issues with jitter what would be a case where the upgrade is needed. |
| 00:55.18 | tzanger | peted20: yes |
| 00:55.30 | tzanger | harryvv: perhaps not for you but it works wonders |
| 00:55.55 | opus___ | <PROTECTED> |
| 00:55.55 | opus___ | *jedaustin* what do you have for your default? |
| 00:55.56 | opus___ | <tzanger> who |
| 00:55.56 | opus___ | *** Signoff: rvhi () <18:57> |
| 00:55.56 | opus___ | *jedaustin* any other vm-'s that you know of other than goodbye? |
| 00:55.57 | opus___ | *** peted20 (~chatzilla@24-113-67-25.wavecable.com) joined #asterisk <18:58> |
| 00:55.58 | opus___ | *jedaustin* Hmm.. cool |
| 00:55.59 | opus___ | *jedaustin* Thanks, I think Im at least talking with broadvoice, thats a plus |
| 00:56.01 | opus___ | *** Signoff: cbachman (Remote closed the connection) <18:59> |
| 00:56.03 | opus___ | *** Signoff: rephorm (Read error: 113 (No route to host)) <19:00> |
| 00:56.05 | opus___ | <peted20> anyone here know how to apply patches to asterisk? |
| 00:56.07 | peted20 | tzanger: thanks! no wonder I was getting "already applied" errors ;-) off to try it now |
| 00:56.08 | opus___ | blah |
| 00:56.10 | harryvv | tzanger mabey the clients will cheer :) |
| 00:56.10 | opus___ | fucking mouse |
| 00:56.39 | tzanger | harryvv: eh? |
| 00:56.53 | *** join/#asterisk bparker (bparker@cable-71-8-65-183.mtv.al.charter.com) |
| 00:57.07 | harryvv | opus mabey its your nerves ;) |
| 00:58.42 | harryvv | talking about shaky and nervousness I have a friend who's going to be on Jeprody tomarro night. The TV game show. The actual filming of the show was a few weeks ago. |
| 00:58.45 | bparker | I am new to Asterisk. I was wondering if the RTP stream of a phone goes through the Asterisk server or if it just does the call setup and tear down? |
| 00:59.16 | tzanger | harryvv: oh yeah? that's cool |
| 00:59.32 | harryvv | yea I read on the site that he would be going against ken jennings. |
| 00:59.32 | tzanger | bparker: depends on how you have it set up, it can go either way |
| 01:00.37 | bparker | cool. What is the limiting factor as to how many devices the server can handle if configured to just just do call setup and tear down? |
| 01:00.41 | harryvv | tzanger when I talked with him on a couple of occations I thought to my self "man this guy is witty he knows details about some subjects that he would whip out a answer so fast" It was actually kind of funny :) |
| 01:00.55 | tzanger | hahaha |
| 01:01.29 | harryvv | He actually needs the money a few months ago he lost his software support position and got back on his feet with another company. |
| 01:02.21 | harryvv | He really needs this. This is his second time on the show. If you watch it tomarro night his name is robert slaven. |
| 01:04.44 | tzanger | right on |
| 01:04.47 | tzanger | why does he really need it? |
| 01:04.55 | harryvv | ohhh very funny i typed in his name and it was the first link on google |
| 01:05.24 | harryvv | what the money? |
| 01:05.33 | tzanger | haha |
| 01:05.41 | tzanger | I was debating whether to watch Frasier or Simpsons |
| 01:05.44 | harryvv | I dont think its for the money. In fact he never told me he was on the show. |
| 01:05.48 | tzanger | and Simpsons has Sideshow Bob on it so I got Frasier too |
| 01:06.35 | *** join/#asterisk Darwin[laptop] (~darwin-la@c-24-3-226-147.client.comcast.net) |
| 01:06.57 | harryvv | :) |
| 01:07.07 | tzanger | hahahaha |
| 01:07.18 | tzanger | "If I wanted to kill you I'd have choked you like a chicken when I got in the door" |
| 01:07.50 | harryvv | dont say that to somone who is in phycosis |
| 01:08.08 | tzanger | eh? |
| 01:08.14 | harryvv | a nutcase |
| 01:08.18 | harryvv | mental problems :) |
| 01:08.48 | tzanger | yeah I just like the choking the chicken reference |
| 01:08.55 | harryvv | :) |
| 01:10.47 | harryvv | btw what the reason voip-info was down most of the day some time ago |
| 01:11.51 | modulus_ | b/c voip sucks |
| 01:13.30 | harryvv | :) |
| 01:13.42 | harryvv | Sounds like the maintainer was on vacation. |
| 01:14.04 | Katty | hmm |
| 01:14.10 | Katty | it's cold |
| 01:14.32 | tzanger | it's not bad |
| 01:15.06 | Katty | i hope that's at least 70F |
| 01:16.53 | *** join/#asterisk sd-tux (sd@2001:6f8:1372:0:0:0:0:2) |
| 01:17.03 | harryvv | I hope when I do some traveling this summer to see moose bear and not Cougar |
| 01:17.34 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
| 01:18.44 | sivana | who wants to finish writing my routing engine? |
| 01:20.00 | Moc | where the register page sivana ? ;) |
| 01:20.42 | sivana | who wants to finish my website too? :) |
| 01:21.05 | sivana | Moc, do you want one now, I can add ya in |
| 01:21.32 | Moc | sure |
| 01:21.43 | sivana | ok |
| 01:23.30 | *** join/#asterisk FryGuy (fryguy@c-24-23-19-33.client.comcast.net) |
| 01:24.59 | FryGuy | Does anyone have any experience with using SetCIDNum to change the caller ID number to an 800 number instead of a local number? I've tried various things and searching on the mailing lists for more information, but nothing has presented itself. |
| 01:25.10 | tzanger | FryGuy: typically you odn't want to do that |
| 01:25.17 | *** join/#asterisk Frantic (~ab@24-193-46-85.nyc.rr.com) |
| 01:25.31 | tzanger | FryGuy: I have personally found that setting your outgoing CID to an 800# and calling certain 800#s caused the call to not complete |
| 01:25.53 | modulus_ | tzanger, yeah ani is a bitch |
| 01:26.15 | FryGuy | MCI claims they have everything set up and we can change the caller id information to anything, but it fails to work. |
| 01:26.18 | tzanger | likely due to the terminating 800# not accepting calls from that area code :-) |
| 01:26.27 | harryvv | If skype has 29 million subscribers and thay do not charge anything how do thay stay in Bussiness? |
| 01:26.40 | Juggie | they dont run the network |
| 01:26.42 | tzanger | they do charge for it |
| 01:26.44 | Juggie | its peer to peer |
| 01:26.56 | Juggie | and they make money by selling skypeout and skype in |
| 01:26.58 | FryGuy | even changing it to another number in the same zip/prefix doesn't work.. so I'm thinking something may be wrong on my end. |
| 01:27.08 | harryvv | was just reading a little on eweek about skype. |
| 01:27.22 | *** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 01:27.52 | FryGuy | Basically, i have an extension that is exten => _9XNNNNNN,1,SetCIDNum(9165551212 |
| 01:27.55 | FryGuy | er |
| 01:28.05 | FryGuy | exten => _9XNNNNNN,1,SetCIDNum(9165551212|a) |
| 01:28.13 | FryGuy | sorry, pipe is in wrong place on this keyboard |
| 01:28.53 | *** join/#asterisk cbachman (~chatzilla@victory.ece.northwestern.edu) |
| 01:28.59 | FryGuy | should this work in normal circumstances? |
| 01:29.10 | *** join/#asterisk shepherd (matt@pcp01541028pcs.huntsv01.al.comcast.net) |
| 01:29.59 | peted20 | :q |
| 01:31.07 | ariel_ | hello everyone |
| 01:31.38 | FryGuy | ^^ |
| 01:38.56 | *** join/#asterisk doughecka_ (~Doug@doughecka.user) |
| 01:39.04 | *** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
| 01:39.13 | doughecka_ | hail ManxPower |
| 01:40.35 | ManxPower | ~docs |
| 01:40.36 | 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 |
| 01:40.38 | ManxPower | ~mailinglist |
| 01:40.39 | jbot | mailinglist is probably Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 01:40.42 | doughecka_ | ok |
| 01:40.50 | ManxPower | Has the flood of n00bs abated any? |
| 01:40.53 | ManxPower | Hello, doughecka |
| 01:40.55 | doughecka_ | not sure |
| 01:40.56 | doughecka_ | <PROTECTED> |
| 01:41.13 | doughecka_ | btw how well does tdmoip work with asterisk? |
| 01:41.16 | doughecka_ | is it stable? |
| 01:41.29 | ManxPower | TDM400P? |
| 01:41.41 | ManxPower | or TDMoverIP? |
| 01:41.52 | doughecka_ | TDMoIP |
| 01:42.17 | ManxPower | As I understand it, TDM over IP stucks and there's little interest in fixing it because IAX2 trunking works so well. |
| 01:42.22 | modulus_ | tdmoip? |
| 01:42.23 | modulus_ | hahahahhaa |
| 01:42.25 | doughecka_ | oh |
| 01:42.30 | doughecka_ | shucks :P |
| 01:42.32 | modulus_ | anoTHER broken protocol for voice? |
| 01:42.33 | modulus_ | hahahahhaa |
| 01:42.45 | tzanger | TDMoIP is useful if you can't get the signaling |
| 01:43.03 | ManxPower | modulus_, Before IAX2 w/trunking, TDMoverIP was the only way to send large numbers of calls over a LAN with low oeverhead. |
| 01:43.10 | doughecka_ | ah |
| 01:43.40 | ManxPower | IAX2 w/trunking has more overhead than TDM over IP, but still much less than any other VoIP protocol. |
| 01:44.31 | harryvv | What low priced poe switch is recomended for a small office of 24 phones. |
| 01:45.00 | hardwire | so |
| 01:45.07 | hardwire | this starband 380 modem has a JTAG interface on it |
| 01:45.08 | hardwire | heh |
| 01:45.17 | tzanger | ManxPower: I disagree |
| 01:45.19 | hardwire | boy wouldn't that piss the FCC off |
| 01:45.23 | ariel_ | ok so does any one have more information on the new soon to be relased Asterisk Enterprise that was talked about in the Von Show? |
| 01:45.37 | tzanger | with a good low bitrate codec and a lot of calls per trunk frame it can be far far far less overhead |
| 01:45.49 | hardwire | wrong channel |
| 01:45.50 | hardwire | heh |
| 01:46.01 | doughecka_ | haha |
| 01:46.13 | doughecka_ | ManxPower: whats another name for a T1/pri splitter? |
| 01:46.16 | harryvv | or wrong network |
| 01:46.20 | modulus_ | channel bank? |
| 01:46.20 | ManxPower | doughecka, no idea. |
| 01:46.28 | harryvv | hardwire this is the only channel your on ;) |
| 01:46.31 | doughecka_ | well, I found things on the web that does it |
| 01:46.37 | ManxPower | Whoo! Whoo! PLC and Jitterbuffer was added for IAX2 and RTP today! |
| 01:46.48 | doughecka_ | PLC? |
| 01:46.50 | harryvv | yea it was |
| 01:47.12 | peted20 | do you have to do anything to turn on PLC? |
| 01:47.13 | tzanger | yup |
| 01:47.17 | tzanger | packet loss concealment |
| 01:47.24 | doughecka_ | cool |
| 01:47.34 | modulus_ | how well does plc work? |
| 01:48.15 | modulus_ | anyone test it out yet? |
| 01:49.12 | ManxPower | modulus_, Both the IAX2 people and the RTP/SIP people have done significant testing and are working todather, as I understand it. |
| 01:49.41 | ManxPower | I think zoa is leasing the SIP/RTP Jitterbuffer and PLC stuff, I think coppice is leading the IAX2 group. |
| 01:49.56 | modulus_ | is it in todays cvs? |
| 01:50.17 | ManxPower | "Anyone that runs CVS-HEAD and is not on the asterisk-cvs mailing list is a moron." --Me |
| 01:50.26 | hardwire | harryvv: you lie |
| 01:50.31 | ManxPower | modulus_, are you a moron? |
| 01:50.49 | modulus_ | manxpower, i just hate mailing lists |
| 01:51.18 | ManxPower | "Anyone that runs CVS-HEAD and is not on the asterisk-cvs mailing list is a moron." --Me |
| 01:51.26 | harryvv | freenode does not :) |
| 01:51.35 | modulus_ | i hate email in general |
| 01:51.42 | modulus_ | the whole of SMTP is braindead |
| 01:52.03 | modulus_ | someone needs to re-write a mail protocol from the ground up |
| 01:52.04 | hardwire | harryvv: just out of curiosity.. what tells you I am only on one channel :) |
| 01:52.21 | ManxPower | I'm honestly not interesed in your excuses, modulus_. |
| 01:52.23 | harryvv | hardwires its a dead subject :) |
| 01:52.35 | hardwire | just like my social life. |
| 01:52.46 | hardwire | you any good with DC/DC power supplies :) |
| 01:52.47 | Nugget | 5 minutes, and 43 seconds until epoch 1111111111 |
| 01:53.03 | tzanger | hehehe |
| 01:53.13 | mikegrb | I know! |
| 01:53.22 | hardwire | heh |
| 01:53.33 | hardwire | I didn't even bake a cake |
| 01:53.35 | hardwire | :( |
| 01:53.40 | tzanger | I'm watching it now |
| 01:53.48 | hardwire | I think everybody is |
| 01:53.52 | tzanger | while [ 1 ]; do date +%s ; sleep 1 ; done |
| 01:53.53 | brc_ | ~seen malcolm |
| 01:53.56 | jbot | malcolm <~x@dsl-212-135-209-195.dsl.easynet.co.uk> was last seen on IRC in channel #gllug, 728d 13h 29m 22s ago, saying: 'Of course I know my own name!'. |
| 01:53.56 | hardwire | anybody with a clock even. |
| 01:55.36 | hardwire | http://216.239.63.104/search?q=cache:gRkIzH86m6gJ:www.1111111111.net/+1111111111&hl=en |
| 01:55.36 | hardwire | had to check |
| 01:55.36 | hardwire | brc_: thats impressive |
| 01:55.36 | brc_ | eh? |
| 01:55.47 | hardwire | he was last seen over 728 days ago |
| 01:55.50 | brc_ | yes |
| 01:56.03 | hardwire | is that a popular ~seen? |
| 01:56.08 | brc_ | I'm still waiting for him to show up too |
| 01:56.14 | brc_ | no that I'm aware of |
| 01:56.22 | *** join/#asterisk Inv_arp (junya@adsl-8-232-188.mia.bellsouth.net) |
| 01:56.46 | doughecka_ | ~date |
| 01:56.47 | jbot | Fri Mar 18 01:56:47 2005 |
| 01:56.47 | hardwire | brc_: are you this patient in the sack? |
| 01:56.48 | doughecka_ | ~time |
| 01:56.49 | jbot | well, time is 1 dimensional, or everlasting |
| 01:56.51 | doughecka_ | :) |
| 01:56.53 | hardwire | cause that would be pretty impressive :) |
| 01:57.00 | brc_ | no |
| 01:57.00 | modulus_ | 1111111023 |
| 01:57.03 | modulus_ | 1111111025 |
| 01:57.03 | modulus_ | 1111111026 |
| 01:57.04 | modulus_ | 1111111027 |
| 01:57.06 | brc_ | honestly people |
| 01:57.11 | brc_ | nobody cares |
| 01:57.24 | modulus_ | sometimes the media does |
| 01:57.34 | hardwire | brc_: let the geeks have their day :) |
| 01:57.35 | brc_ | the media are morons, what's the point? |
| 01:57.43 | modulus_ | no the media is just a business |
| 01:58.41 | modulus_ | freebsd5-stable is temporarily called freebsd5-prerelease wtf? |
| 01:58.48 | hardwire | what |
| 01:58.50 | hardwire | nobody says ra? |
| 01:58.52 | modulus_ | err 5_4 prerelease |
| 01:59.01 | hardwire | is everybody off kissing their pets? |
| 01:59.05 | hardwire | where did all the excitement go? |
| 01:59.10 | harryvv | hehehe |
| 01:59.14 | tzanger | it's passed |
| 01:59.21 | hardwire | well thats enough of that |
| 01:59.23 | tzanger | 1111111110 |
| 01:59.23 | tzanger | 1111111111 |
| 01:59.24 | tzanger | 1111111112 |
| 01:59.27 | hardwire | time to go back to being annoying. |
| 01:59.27 | hardwire | man |
| 01:59.30 | hardwire | we are all so very off |
| 01:59.34 | hardwire | and I thought the atomic clock was accurate |
| 01:59.45 | harryvv | it is |
| 01:59.45 | Nugget | Fri Mar 18 01:59:45 UTC 2005 |
| 01:59.49 | hardwire | tzafrir: you are a minute off from me |
| 01:59.51 | modulus_ | yeah what happened to ntp you people? |
| 01:59.55 | hardwire | and I just set my rtc |
| 01:59.56 | tzanger | yeah I'm not saying I'm 100% accurate |
| 01:59.58 | harryvv | a cesium clock thats been arouns since the 1960s |
| 02:00.04 | mikegrb | 1111111204 |
| 02:00.06 | modulus_ | ntpdate -s -b -p 8 time.nasa.gov |
| 02:00.06 | hardwire | heh |
| 02:00.10 | modulus_ | cron that you people |
| 02:00.17 | Nugget | localhost: stratum 3, offset 0.000022, synch distance 0.08743 |
| 02:00.24 | tzanger | modulus_: you're on crack |
| 02:00.25 | Nugget | no, don't just cron ntpdate. that's lame. |
| 02:00.25 | tzanger | you don't do that |
| 02:00.27 | tzanger | you run ntpd |
| 02:00.27 | Nugget | run ntpd |
| 02:00.30 | hardwire | heh |
| 02:00.35 | tzanger | and I use tic.nrc.ca |
| 02:00.35 | Nugget | cronning ntpdate is a bad solution |
| 02:00.48 | hardwire | Nugget: its the only solutino for me |
| 02:00.51 | hardwire | solution even |
| 02:00.53 | tzanger | remember that asterisk uses gettimeofday() and ntpd fucks with that |
| 02:00.58 | Nugget | why? it's pessimal. |
| 02:01.08 | hardwire | laptopness |
| 02:01.09 | Nugget | ntpd doesn't fuck with that. |
| 02:01.10 | brc_ | ntpd is far far better then ntpdating every once in awhile |
| 02:01.14 | tzanger | it does so |
| 02:01.14 | hardwire | tried using ntpd in set rtc mode |
| 02:01.17 | Nugget | no it doesn't. |
| 02:01.19 | hardwire | it fails |
| 02:01.19 | tzanger | set your time back a minute |
| 02:01.22 | brc_ | tzanger, as in causes problems? |
| 02:01.24 | tzanger | have a call in progress |
| 02:01.26 | tzanger | and run ntpdate |
| 02:01.30 | tzanger | you will get massive jitter |
| 02:01.31 | Nugget | no, ntpd will slew the clock, unlike ntpdate |
| 02:01.33 | tzanger | becaues the itmestamps change |
| 02:01.38 | brc_ | ^^ what Nugget said |
| 02:01.40 | Nugget | that's precisely why ntpd is good. |
| 02:01.42 | yxa | rdate -s works too |
| 02:01.43 | tzanger | yes ntpd does slew it it's not nearly as bad as ntpdate'ing |
| 02:01.45 | hardwire | Nugget: indeed. |
| 02:01.55 | brc_ | and ntpd remembers your skew and corrects for it inbetween updates |
| 02:02.00 | hardwire | howeevr.. it doesn't correct past a certain threshold. |
| 02:02.07 | hardwire | so if the laptop is off quite a bit.. for no damn good reason |
| 02:02.11 | brc_ | which is confirurable in your ntpd.conf |
| 02:02.12 | harryvv | so the new jitter fix is timmed to atomic time |
| 02:02.18 | hardwire | and you have no network when ntpdate is run initially before ntpd |
| 02:02.24 | hardwire | it fails |
| 02:02.26 | hardwire | see.. |
| 02:02.28 | hardwire | oh well |
| 02:02.38 | hardwire | not like I am syncing tcp/ip timestamps here. |
| 02:02.56 | Nugget | hardwire: "ntpd -g" solves that. |
| 02:03.14 | Nugget | it will allow the first sync in ntpd to exceed the thresshold |
| 02:03.33 | hardwire | Nugget: the first sucessfull one.. does it count all the initial timeouts? |
| 02:03.35 | Nugget | and the first sync doesn't have to be at startup. it's just the first one, whenever that becomes possible |
| 02:03.43 | Nugget | the first sucessful one |
| 02:04.04 | dersteer | happy mi11-one-ium everyone |
| 02:04.09 | hardwire | Nugget: you know.. you would think after lokoing through ntpd like crazy.. I would find that option |
| 02:04.14 | mishehu | anybody using an at-320ed phone (atcom) with iax? I'm looking for a user review and to hear how the phone compares to other phones I work with |
| 02:04.17 | hardwire | Nugget: do you have enable ntp in your ntpd.conf? |
| 02:04.32 | Nugget | no |
| 02:04.40 | hardwire | then how do you sync the hwclock? |
| 02:04.51 | hardwire | or even the swclock? |
| 02:05.07 | Nugget | dunno, it just works |
| 02:05.17 | hardwire | that option explicitely resets all of ntpd's logged offsets? |
| 02:05.20 | hardwire | and syncs |
| 02:05.22 | hardwire | I take it |
| 02:05.59 | peted20 | with the new jitterbuffer, how can I tell if PLC is enabled? I have "genericplc => true" in codecs.conf, but I am not sure if it is taking effect |
| 02:07.32 | hardwire | Nugget: ok.. I am giving it a shot |
| 02:07.41 | hardwire | if you lie!!!!!! OOOOH I WILL COME AFTER YOU!! |
| 02:07.45 | hardwire | :) |
| 02:08.12 | modulus_ | FreeBSD 5.4-PRERELEASE (GENERIC) #2: Thu Mar 17 20:12:48 UTC 2005 |
| 02:08.14 | modulus_ | ewww |
| 02:08.20 | modulus_ | damnit i want 5.3-STABLE |
| 02:09.01 | harryvv | looking at ntp.conf interesting. I remember listening to wwv way back in the 80s and has a even broader use today :) |
| 02:09.33 | modulus_ | ntp is the shit |
| 02:09.48 | Darwin[laptop] | ntp? |
| 02:09.58 | Darwin[laptop] | or ntpd |
| 02:10.25 | harryvv | yea atomic time. Based off a Cesium atomic clock in bolder colorado that is accurate to 1 100th of a second every 100 years. |
| 02:10.27 | Nugget | I remember running timesync.exe in dos |
| 02:10.29 | harryvv | Something like that. |
| 02:10.56 | harryvv | good oll wwv :) good to hear it on 10,15,20 meters. |
| 02:11.02 | mikegrb | I prefer nuggetsync.exe |
| 02:11.24 | Darwin[laptop] | yeah |
| 02:12.06 | *** join/#asterisk Mazda-MX5 (~root@220-130-142-43.HINET-IP.hinet.net) |
| 02:12.15 | Darwin[laptop] | I remember it on Ham radio also 15000 18000 22000 mhz |
| 02:13.12 | modulus_ | anyone ever use shortwave? |
| 02:13.19 | harryvv | I do |
| 02:13.25 | harryvv | I have all modes all bands |
| 02:13.30 | modulus_ | my handle was five hotel three delta hotel |
| 02:13.46 | harryvv | handle? |
| 02:14.16 | harryvv | CB was unlicenced 25 years ago :) |
| 02:14.24 | *** join/#asterisk justinnnn (~dsf@solid.mpa.net.au) |
| 02:14.50 | Darwin[laptop] | cb is still open band |
| 02:14.55 | harryvv | yea it is |
| 02:14.59 | harryvv | not used much |
| 02:15.09 | Darwin[laptop] | now ham is starting to open up |
| 02:15.16 | harryvv | its okay if you in boonie land where cell does not work |
| 02:15.27 | harryvv | Darwin open up as in tropo conditions? |
| 02:15.39 | Darwin[laptop] | they did away with novice/ and coded tech |
| 02:16.03 | harryvv | no thay didnt. Its still there. Its just not a req to obtain your licence. |
| 02:16.23 | harryvv | novice is gone but tech plus is still a option |
| 02:16.43 | harryvv | or you can go no code tech just dont have all the priv. |
| 02:17.30 | harryvv | btw, I need to set up another asterisk system to see how well a phone bank will work on one of the digital frequencies. |
| 02:17.32 | Darwin[laptop] | yeah I am working on geting back in to ham radio and sattalite |
| 02:18.23 | harryvv | Calculating the lowest frequency is needed for a 10 phone bank is something I would need to calculate. |
| 02:18.44 | *** join/#asterisk Mazda-MX5 (~root@220-130-142-43.HINET-IP.hinet.net) |
| 02:18.52 | harryvv | That way can get some nice range. wouldntthat be wierd to send a asterisk convo via DX into another country. |
| 02:19.16 | ManxPower | Do european hotels commonly provide any breakfast type of stuff in the morning, as part of the cost of the hotel room. |
| 02:19.34 | harryvv | sip convo :) come to think about it I dont know if there is any studies on which protocol would work best when doing a dx convo. |
| 02:19.56 | harryvv | somone in #europe on efnet may know. |
| 02:20.06 | harryvv | what country? |
| 02:20.36 | modulus_ | try whales |
| 02:20.46 | modulus_ | because that's where catherine zeta jones is from |
| 02:21.11 | modulus_ | and she's just hot |
| 02:21.11 | tzanger | whales? hahahaha |
| 02:21.36 | modulus_ | is anyone here running asterisk on dual xeons? |
| 02:21.59 | harryvv | is there a wiki on how to setup asterisk to call a group of people say 50 and make a general anouncment? |
| 02:22.25 | tzanger | modulus_: single xeon |
| 02:22.27 | harryvv | mod, I am running it on a amd opteron single cpu at the moment. |
| 02:22.47 | modulus_ | opteron ? what's your call volume harryvv? |
| 02:22.55 | modulus_ | tzanger, call volume? |
| 02:23.16 | tzanger | modulus_: not extensive... a dozen max maybe at this time |
| 02:23.24 | harryvv | mod it was purpously built for a heavy load graphics work station. But no load at this moment. Would love to test it though. |
| 02:23.41 | harryvv | Its very fast. |
| 02:23.44 | harryvv | Love it |
| 02:24.04 | harryvv | only bottle neck is the hard drive |
| 02:24.34 | harryvv | why do you ask |
| 02:26.50 | modulus_ | i want to know what to expect from my asterisk box |
| 02:28.15 | modulus_ | i have a dual xeon 1 gig ram |
| 02:28.24 | modulus_ | i think i'll run into memory problems first |
| 02:28.49 | tzanger | I dunno I think I'd make it as spread out as possible |
| 02:29.01 | harryvv | Dont know. |
| 02:29.50 | modulus_ | tzanger, what do you mean spread out? |
| 02:31.36 | harryvv | looks like L3 in sanjose has a bad day today according to the packet drops. Was thinking of using them as a backbone provider. Mabey some construction crew cut into the fiber :) |
| 02:32.21 | modulus_ | harryvv, what's their pricing on b/w? |
| 02:32.56 | *** join/#asterisk sricard (sricard@Toronto-HSE-ppp3740961.sympatico.ca) |
| 02:33.37 | harryvv | Talked to a telus tech the other day he was stating 400-600 for pri but was not sure. |
| 02:34.28 | modulus_ | how much of that is for the loop? |
| 02:34.30 | harryvv | I need to shop around. there is not to many choices up here from what I have heard from people in the industry. Allstream do not own any lines and have the work sourced out to telus. Thats a example there ;) |
| 02:34.55 | modulus_ | voice t1 or voip t1? |
| 02:34.58 | harryvv | you mean pstn data out? |
| 02:35.19 | harryvv | I was talking pstn data out when he came up with those rough figures. |
| 02:35.21 | modulus_ | lotsa data carriers are providing voip now |
| 02:35.26 | bparker | is anyone in here familiar with cisco callmanager |
| 02:35.28 | harryvv | I know. |
| 02:35.32 | modulus_ | kinda nuts |
| 02:35.41 | modulus_ | global crossing sells sip |
| 02:35.51 | harryvv | Its like..I do this yet thay are also so who is reaping the benifits? |
| 02:36.06 | modulus_ | the ceos |
| 02:36.07 | JunK-U | bparker: are we a cisco channel? :P |
| 02:36.16 | harryvv | yea the ceos :) |
| 02:36.27 | modulus_ | everyone else gets shit trickled down |
| 02:36.48 | bparker | no no. I am from that background and I am getting into Asterisk and want to know how it compares to callmanager |
| 02:36.56 | harryvv | I should try and get a hold of my old telco instructor and see if he is still working in the industry he looked like he was heading to retirement. |
| 02:37.40 | modulus_ | i'm trying to get a t1 pri |
| 02:37.48 | modulus_ | but no one wants to pay for it |
| 02:37.49 | modulus_ | haha |
| 02:37.53 | doughecka_ | lol |
| 02:39.13 | harryvv | nifty got a pic of telus backbone |
| 02:39.39 | Mazda-MX5 | Orz |
| 02:39.39 | harryvv | wow even goes down to california. |
| 02:41.48 | modulus_ | damnit |
| 02:41.56 | modulus_ | this compiling is taking too long |
| 02:42.06 | harryvv | on the xeon ? |
| 02:42.06 | Shido6 | then get a faster proc |
| 02:42.21 | modulus_ | harryvv, no freebsd5.4-prerelease on pIII |
| 02:42.26 | harryvv | k |
| 02:42.57 | modulus_ | that's it i'm going home |
| 02:43.08 | modulus_ | gotta hit the bodega for some cigs and some liquor |
| 02:44.31 | modulus_ | <PROTECTED> |
| 02:44.31 | modulus_ | <PROTECTED> |
| 02:44.38 | modulus_ | wow 2% idle cpu |
| 02:45.12 | modulus_ | thank god it's not redhat |
| 02:45.25 | modulus_ | it would've slowed to a crawl |
| 02:52.03 | *** join/#asterisk rvhi (~rv@66.175.65.89) |
| 02:52.18 | *** join/#asterisk hawaiianphoneguy (~mdarnell@66.135.226.125) |
| 02:52.52 | hawaiianphoneguy | anyone know about gr303 in * |
| 02:56.51 | *** join/#asterisk Mazda-MX5 (~root@220-130-142-43.HINET-IP.hinet.net) |
| 02:57.02 | *** join/#asterisk pciccone (~pciccone@24.115.30.135.res-cmts.wb.ptd.net) |
| 02:57.58 | pciccone | Was wondering if someone had some time to answer a question regarding sip.conf and a register command. Not getting expected results with BV proxy |
| 02:58.00 | *** join/#asterisk IQ (~IQ@70-59-160-224.omah.qwest.net) |
| 02:58.29 | Mazda-MX5 | hi , What is the "iax2" channels ? |
| 02:58.37 | *** join/#asterisk heison (~heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 02:59.21 | IQ | Mazda-MX5: IAX ? |
| 02:59.34 | IQ | try iax |
| 03:00.15 | Mazda-MX5 | IQ > yes . it is in asterisk-1.0.6/channels/cha_iax2.so , is it for FAX ? |
| 03:00.59 | IQ | sorry - no idea |
| 03:01.20 | Mazda-MX5 | thanks |
| 03:03.08 | Mazda-MX5 | IQ > what is the "IAX" for ? |
| 03:03.14 | mgth | mazda-mx5 |
| 03:03.19 | mgth | read the voip-info site |
| 03:03.35 | mgth | iax is a voip protocol for asterisk |
| 03:03.35 | Mazda-MX5 | voip-info ? |
| 03:03.58 | mgth | voip-info.org |
| 03:04.00 | *** join/#asterisk DannyF (~wizardone@h67n3c1o848.bredband.skanova.com) |
| 03:04.09 | Mazda-MX5 | thank you. voip-info.org |
| 03:04.56 | Mazda-MX5 | sorry, I am newbis of VoIP |
| 03:05.17 | Cherebrum | IAX is a channel like SIP |
| 03:05.21 | Cherebrum | but it's better than SIP |
| 03:05.27 | Mazda-MX5 | ! |
| 03:05.28 | Cherebrum | it only uses port 4569 UDP |
| 03:05.29 | IQ | no need to be sorry... many of us are new |
| 03:05.37 | Cherebrum | and it can traverse NAT |
| 03:05.45 | Mazda-MX5 | better than SIP !!! |
| 03:05.50 | Cherebrum | yes |
| 03:06.01 | Cherebrum | There are even some ATA's that can do IAX |
| 03:06.20 | Cherebrum | like the IAXy see: www.digitnetworks.com |
| 03:06.25 | Mazda-MX5 | thanks all , I think I must read 'voip-info.org' forst. |
| 03:06.34 | Cherebrum | yea. read that |
| 03:06.35 | Mazda-MX5 | first. |
| 03:06.39 | Cherebrum | it's some good stuff |
| 03:07.01 | Cherebrum | I gotta take a fat shit.. and my girlfriend is in the damn shower |
| 03:07.40 | IQ | Cherebrum: dont lose this mument |
| 03:07.51 | Chuji | ~iax |
| 03:07.54 | jbot | it has been said that iax is 4569 and 5036, or pronounces "Eeks" |
| 03:07.55 | Chuji | ~rtfw |
| 03:07.56 | jbot | i guess rtfw is Read That F*cking Wiki, the one at http://www.voip-info.org/tiki-index.php |
| 03:08.26 | Jer13261 | hey why do i get a beep during a 3 way call? |
| 03:08.29 | Cherebrum | 5036? |
| 03:08.37 | *** join/#asterisk fgravato (5Zeagon@ool-44c02d18.dyn.optonline.net) |
| 03:08.37 | Chuji | ports |
| 03:08.38 | *** join/#asterisk goatmilk (~goatmilk@130-127-45-50.chouse.resnet.clemson.edu) |
| 03:08.44 | Chuji | ~iax2 |
| 03:08.50 | Cherebrum | 5036=IAX 4569 = IAX2 right? |
| 03:08.57 | Jer13261 | yep |
| 03:09.03 | Jer13261 | hmm |
| 03:09.04 | Qwell | UDP |
| 03:09.27 | Cherebrum | yea |
| 03:09.28 | Cherebrum | UDP |
| 03:09.39 | Cherebrum | I had to poke all sorts of holes in firewalls for crap like that today |
| 03:09.43 | Chuji | Is it uncommon to only have Jitter one way? Is that because one half of my connection has a jitter buffer? |
| 03:09.53 | Cherebrum | 123 for ntp and 69 for TFTP and 4569 for IAX2 |
| 03:11.34 | *** join/#asterisk Y1 (~Y1@he134.internetdsl.tpnet.pl) |
| 03:13.10 | *** join/#asterisk postel (canonical@postel.user) |
| 03:15.34 | jtodd | anyone had any luck with the bluetooth stuff compiling lately? |
| 03:16.05 | Chuji | jtodd : chan_bluetooth? |
| 03:16.07 | *** join/#asterisk ROM_Man (rom_man@mike.netrom.com) |
| 03:16.20 | Chuji | jtodd : I don't think much development have happened on that in months |
| 03:16.29 | Cherebrum | I just click the little bluetooth icon up by the clock on the powerbook and it turns on |
| 03:16.30 | jtodd | no, the btp stuff from kram |
| 03:16.46 | Chuji | hmm, didn't know he was working on it |
| 03:16.51 | Chuji | on mantis? |
| 03:17.00 | jtodd | it's on the CVS server. |
| 03:17.16 | jtodd | Though, now that I mention it, I see that it actually compiles today. <grumble> |
| 03:17.39 | Cherebrum | Anyone see that article about using the unlimited Mobile-to-Mobile minutes to make unlimited phone calls with asterisk? you just get a second phone and hookers it up to a gateway device and make calls thru it |
| 03:18.01 | *** join/#asterisk knebel (~Gordon@pool-68-160-162-207.bos.east.verizon.net) |
| 03:18.02 | Chuji | Cherebrum : Yeah, but you need a cell dock to do that |
| 03:18.08 | Chuji | bluetooth would be cooler |
| 03:18.11 | jtodd | doesn't have much to do with Asterisk; lots of vendors sell the dock already configured for that kind of hack. |
| 03:18.17 | Chuji | There's been a bounty out there for a year on it |
| 03:18.18 | jtodd | yes, bluetooth would be much more interesting. |
| 03:18.53 | Cherebrum | http://www.voip-info.org/wiki-Asterisk+Connecting+to+the+Cellular+Network |
| 03:18.54 | jtodd | Jer13261: no dock for the 600 or 650. |
| 03:19.00 | Cherebrum | Jer13261: get another phone |
| 03:19.08 | jtodd | that's why I'm interested in bluetooth for Asterisk at the moment; I have the 650. |
| 03:19.11 | Jer13261 | how about the Sanyo 8100? |
| 03:19.39 | Cherebrum | hmm... if I can get bluetooth to work with asterisk maybe I could use my spare Sony T68i as a mobile to asterisk bridge |
| 03:19.46 | Chuji | ~google "sanyo 8100 cell dock" |
| 03:19.58 | *** join/#asterisk MikeJ[Laptop] (~icechat5@pcp02795302pcs.roylok01.mi.comcast.net) |
| 03:20.03 | Jer13261 | cool :) |
| 03:20.11 | Cherebrum | so if I have bluetooth in asterisk and a bluetooth phone.... I don't need the cell dock do i? |
| 03:20.31 | Chuji | Cherebrum : You do. chan_bluetooth is very much infancy |
| 03:20.37 | jtodd | Cherebrum: well, if you can get chan_bluetooth working, you're golden. Doesn't quite make "production" quality yet, though. |
| 03:20.48 | Chuji | Cherebrum : sharpen up your c skillz and make it work! |
| 03:20.54 | Chuji | collect a bounty with it |
| 03:21.25 | Cherebrum | oh wait.. my C skillz are non exsistant.. nothing to sharpen |
| 03:21.26 | Cherebrum | shittyh |
| 03:21.31 | Chuji | jtodd : you make it to San Jose? |
| 03:21.36 | jtodd | yep. |
| 03:21.42 | Cherebrum | but that would be cool |
| 03:21.43 | jtodd | I assume you mean for VON. |
| 03:21.43 | Chuji | Have a good time? |
| 03:21.47 | Chuji | Yeah |
| 03:21.58 | Cherebrum | $20 bluetooth adaptor for linux and then I have a mobile to asterisk bridge! |
| 03:22.00 | Cherebrum | sweet! |
| 03:22.03 | Chuji | I can do Boston, but San Jose is out of the budget |
| 03:22.08 | jtodd | No, it was kind of dull. Spent most of my time in meeting rooms doing nonsense with people wearing expensive suits. |
| 03:22.20 | Cherebrum | too bad it's not cost effective for me to do that.. I allready get 3000 anytime minutes for $50 a month |
| 03:22.37 | Cherebrum | and it would cost me $50 for 2 lines with unlimited mobile 2 mobile |
| 03:22.43 | harryvv | I know the FCC limits the amount of wattage output a wifi based unit can transmit probebly because of prximity to the antenna but what about a high pole mounted isopole style or directional panel antenna mounted 20-40 feet off the ground? |
| 03:22.46 | Cherebrum | 3000 minutes is plenty |
| 03:23.13 | *** join/#asterisk stdio (lynn@pcp09745793pcs.lncstr01.pa.comcast.net) |
| 03:23.18 | stdio | greets all! |
| 03:23.51 | Cherebrum | girlfriends TP is so soft on my ass.. |
| 03:24.33 | Cherebrum | sorry.. was that random? |
| 03:24.37 | *** part/#asterisk Y1 (~Y1@he134.internetdsl.tpnet.pl) |
| 03:24.54 | Chuji | Cherebrum : /join #asstricks They talk about soft things on their ass in there |
| 03:26.05 | Cherebrum | oh... this isn't #asstricks? |
| 03:26.06 | Cherebrum | damn. |
| 03:26.16 | *** part/#asterisk Cherebrum (~jgarland@216.32.77.10) |
| 03:27.17 | *** join/#asterisk Cherebrum (~jgarland@216.32.77.10) |
| 03:27.34 | Chuji | jtodd : What is a locater? in the btp |
| 03:27.47 | harryvv | locator? |
| 03:28.33 | *** join/#asterisk NewSole (david@i216-58-44-245.avalonworks.net) |
| 03:28.38 | Cherebrum | there really is a #asstricks |
| 03:28.58 | Chuji | ~asstricks |
| 03:29.16 | Chuji | ~#asstricks |
| 03:29.22 | stdio | trying to get an fxs module working on a tdm400p .... have zaptel.conf set... and THOUGHT i had zapata.conf set up correctly... but I am specifically telling it a different context, and it still seems to want to send me to default... |
| 03:29.24 | *** join/#asterisk hermie (~nick@24.236.167.53.bay.mi.chartermi.net) |
| 03:29.35 | Chuji | Yeah, there really is |
| 03:32.46 | Chuji | harryvv : In the bluetooth presence stuff |
| 03:33.17 | rvhi | hi |
| 03:33.39 | rvhi | in directory command, does it check first name or last name? |
| 03:36.02 | harryvv | okay |
| 03:38.14 | *** join/#asterisk viLeR (1000@ip-47-252.telesat.com.co) |
| 03:41.08 | *** join/#asterisk kks (~kks@203.115.208.140) |
| 03:45.49 | hermie | .5c/min! I've got a new dumping ground for NECA calls! |
| 03:48.47 | Inv_arp | ok my ivr context "A" includes => internal SIP phone context "B" which includes => pstn "C" ; all works fine ... except during IVR "A" anyone can press 91 and get an outside line ... |
| 03:49.46 | Inv_arp | how can i allow "A" to see "B" but not pstn "C" |
| 03:50.14 | eric_ | make extra contexts |
| 03:50.34 | eric_ | make a D which includes B and C |
| 03:50.49 | eric_ | then make A include B, which will not have C |
| 03:51.17 | eric_ | and switch any references to B into a D |
| 03:51.48 | Inv_arp | eric_: nice thx |
| 03:51.53 | eric_ | np |
| 03:53.05 | Inv_arp | eric_: wasnt ware u can make a context with just includes |
| 03:53.16 | Inv_arp | s/sware/aware |
| 03:53.21 | Inv_arp | bah whatever |
| 03:53.24 | eric_ | haha |
| 03:53.42 | eric_ | sometimes the most obvious thing to do isnt the most obvious |
| 04:00.46 | *** join/#asterisk lilo_ (lilo@levin-pdpc.staff.freenode) |
| 04:09.22 | hawaiianphoneguy | anyone use gr303 with *? |
| 04:14.36 | *** join/#asterisk lilo_ (lilo@levin-pdpc.staff.freenode) |
| 04:23.50 | *** join/#asterisk KirkL (~me@c-24-22-57-111.client.comcast.net) |
| 04:35.19 | stdio | on a very high level, what needs to happen in extensions.conf in order for asterisk to act correctly when it receives a call from a fax machine? |
| 04:40.37 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l02m-4-165.d4.club-internet.fr) |
| 04:41.14 | *** join/#asterisk Mazda-MX5 (~root@220-130-142-43.HINET-IP.hinet.net) |
| 04:41.23 | brc_ | answer, wait, have a fax extension in the current context |
| 04:41.46 | brc_ | call has to be in a state where asterisk can hear fax tones for at least a few seconds |
| 04:41.50 | brc_ | that could be in an ivr menu |
| 04:41.56 | brc_ | or just a rude wait(3) |
| 04:42.12 | brc_ | ST-3, |
| 04:42.15 | brc_ | stdio, |
| 04:47.21 | *** join/#asterisk SPoon_TSX (~SPoon_TSX@d206-116-121-40.bchsia.telus.net) |
| 04:48.09 | SPoon_TSX | Hello everyone, I just got my first TDM400P PCI card with 4 FXO. However, I saw there is a power supply on the card. DO I need to connect a Powercord to there? |
| 04:50.46 | *** join/#asterisk w0w0 (~w0w0@80.26.162.27) |
| 04:52.30 | SPoon_TSX | Anyone?? |
| 04:52.36 | *** join/#asterisk Qorky (~goaway@dsl-202-72-146-104.wa.westnet.com.au) |
| 04:53.21 | SPoon_TSX | Does anyone knows what is that power plug on the TDM400 card for? |
| 04:56.26 | wolfson | to get ringing voltage |
| 04:56.35 | *** join/#asterisk `Kirk (~me@c-24-22-57-111.client.comcast.net) |
| 05:01.06 | rious | wow, weirdest thing ever: when I turn off IAX2 debug, ext 18 rings twice then disconects, when I turn on IAX2 debug, ext 18 rings forever and never connects |
| 05:01.17 | rious | how can that be ? |
| 05:02.45 | *** join/#asterisk yertle (yertle@ip68-6-98-122.sb.sd.cox.net) |
| 05:11.35 | *** join/#asterisk andyjones (~andy@user-12lc8ms.cable.mindspring.com) |
| 05:16.50 | *** part/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net) |
| 05:20.04 | *** join/#asterisk KirkL (~me@c-24-22-57-111.client.comcast.net) |
| 05:20.29 | Beirdo | !seen Shido6 |
| 05:21.27 | sudhir492 | hi all |
| 05:21.35 | sudhir492 | anyone using te110p card here? |
| 05:23.43 | *** join/#asterisk Landrocker (~landrocke@203.152.127.9) |
| 05:26.09 | Landrocker | Hey all, I'm looking for away to dial in to my asterisk box and after dialing a certain extension be connected to a dial in terminal (mgetty, etc). Hunting on google hasn't seemed to have got me anywhere, any ideas? |
| 05:27.25 | Jer13261 | app_pppd |
| 05:27.32 | Landrocker | cheers |
| 05:28.26 | *** part/#asterisk yertle (yertle@ip68-6-98-122.sb.sd.cox.net) |
| 05:28.35 | *** join/#asterisk yertle (yertle@ip68-6-98-122.sb.sd.cox.net) |
| 05:28.52 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 05:29.17 | shmaltz | helo everybody |
| 05:29.23 | shmaltz | sivana, you around? |
| 05:29.28 | shmaltz | ~seen sivana |
| 05:29.33 | jbot | sivana <~sivana@165.154.13.35> was last seen on IRC in channel #asterisk, 4h 7m 50s ago, saying: 'ok'. |
| 05:29.51 | shmaltz | ~sleeping |
| 05:29.52 | jbot | i guess sleeping is the magical thing geeks have forgotten how to do |
| 05:30.17 | *** join/#asterisk xeet2 (~xeet3@gw1.istx.net) |
| 05:31.05 | xeet2 | is there a way to force caller id information in inbound calls from an iax peer, other than using the setcallerid app? |
| 05:31.56 | Landrocker | Jer13261, is their anything similar for analogue lines (for an install with only one line handled by a TDM400P) |
| 05:32.20 | Jer13261 | Landrocker hmm write an agi ????? |
| 05:33.16 | Landrocker | I didn't realise agi can do that - it'd still need a way of somehow hooking the line into mgetty I would imagine |
| 05:33.26 | Jer13261 | hmm true |
| 05:33.29 | Jer13261 | i am not sure if it cant |
| 05:33.30 | Jer13261 | can |
| 05:34.08 | Landrocker | Looks like the best solution might be to write a dummy modem kernel module that emulates a modem on one side and a sip phone on the other - probably a bit over my head though |
| 05:35.32 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 05:36.05 | Landrocker | aha! perhaps one of the fax apps can be modified |
| 05:43.42 | xeet2 | is there a way to force caller id information in inbound calls from an iax peer, other than using the setcallerid app? will setting the callerid field on the peer do this? |
| 05:44.12 | *** join/#asterisk marshall (~test@S0106000f66563988.wp.shawcable.net) |
| 05:47.47 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
| 05:47.47 | *** topic/#asterisk is Asterisk: The Open Source PBX || 1.0.7 RC - bug #3746 || http://www.theregister.co.uk/2005/03/16/asterisk_open_source_pbx/ |
| 05:50.31 | marshall | is it possible for an IAX to IAX trunk to introduce echo into calls? |
| 05:50.36 | marshall | other than just the extra latency |
| 05:51.12 | xeet2 | no, it can just make existing echo more noticable |
| 05:51.38 | marshall | must be these polycom phones, they sound terrible |
| 05:51.41 | marshall | the iaxy sounds good |
| 05:52.08 | xeet2 | mmm, what do you have connected to the iaxy? |
| 05:52.29 | xeet2 | cheapo analog phones can cause echo too |
| 05:52.39 | marshall | a good analog phone |
| 05:52.45 | marshall | but it sounds perfect |
| 05:52.58 | marshall | it goes direct to * then out the pri |
| 05:53.14 | xeet2 | oh so its not between the iaxy and the polycom's? |
| 05:53.36 | marshall | no, polycoms - pri - pots |
| 05:53.40 | xeet2 | you can have a pri but if there's issues at the remote end thats analog, the echo can come from there too |
| 05:53.48 | marshall | thats what I am thinking |
| 05:53.55 | marshall | some calls are perfect |
| 05:53.59 | xeet2 | so when you call the same numbers on the iaxy, you don't get the echo? |
| 05:54.11 | marshall | not quite as bad |
| 05:54.23 | marshall | but if the IAX trunk could amplify an existing problem that makes sense |
| 05:54.36 | marshall | the IAXY doesnt use the trunk |
| 05:54.43 | xeet2 | well, its not so much that, its analog<>digital, the delay caused by that conversion |
| 05:54.53 | xeet2 | or any delay introduced |
| 05:55.34 | marshall | I just need to find the magic configuration for these polycoms, stand on one foot while scratching my head to make the calls sound good |
| 05:55.56 | xeet2 | what codecs are you using to the polycom's? across the iax trunk? and on the iaxy's? |
| 05:56.11 | marshall | polycoms are ulaw to the internal server |
| 05:56.20 | marshall | the trunk I have tested both ulaw and gsm |
| 05:56.26 | xeet2 | ok, and server has the pri? |
| 05:56.32 | marshall | correct |
| 05:56.48 | marshall | the gsm seems a little better for echo avoidance |
| 05:57.09 | Essobi | How does an IAX to IAX trunk amplify echos? Latency? |
| 05:57.31 | marshall | apparently |
| 05:57.34 | xeet2 | its not really an iax trunk specificly, its anything that adds latency |
| 05:57.42 | Essobi | roger that |
| 05:57.53 | marshall | the trunk adds a good 60-70ms to the trip |
| 05:57.58 | Essobi | IE why cellphones + voip = echooooooo |
| 05:58.07 | Essobi | youch.. so untrunk it. |
| 05:58.30 | marshall | I prefer to use SIP internall then IAX to the pri servers in the colo |
| 05:58.33 | Essobi | throw the magic sip stick at it. |
| 05:58.33 | xeet2 | marshall: so whats the network latency? |
| 05:58.47 | marshall | 60-70ms in this case |
| 05:58.55 | Essobi | esh |
| 05:58.58 | xeet2 | ok so its not adding that, thats just what you're seeing |
| 05:59.03 | Essobi | hehe |
| 05:59.08 | xeet2 | thats not *too* bad |
| 05:59.13 | marshall | I can live with that |
| 05:59.16 | Essobi | ping -f BABY |
| 05:59.22 | xeet2 | hehe |
| 05:59.49 | Essobi | run your IAX channel over an stunnel or ssh portfoward. ;) |
| 05:59.50 | xeet2 | are you doing any jitter buffering? |
| 05:59.53 | xeet2 | lol |
| 06:00.02 | marshall | I was but it seemed to make it worse |
| 06:00.15 | Essobi | I made * core 4 times today. :) Repeatable. |
| 06:00.20 | xeet2 | I do that with sip at work, but only because I'm not "allowed" to use sip at work |
| 06:00.28 | Essobi | H323 + conference = not happy. |
| 06:00.35 | xeet2 | (the ssh portforward) |
| 06:00.42 | Essobi | what what? |
| 06:00.49 | Essobi | Not allowed to sip at work? |
| 06:00.53 | Essobi | PSSSSH. |
| 06:01.02 | Essobi | Gheeeeetto |
| 06:01.10 | marshall | xeet2 would you think gsm or ulaw within the trunk? |
| 06:01.23 | marshall | I havent seen a definitive answer either way |
| 06:01.25 | Essobi | transcode as little as possible |
| 06:01.27 | Essobi | :) |
| 06:01.47 | xeet2 | marshall: well adding 60-70 ms latency will definitely make an echo more apparent... have you tried messing with echo cancellation? |
| 06:02.00 | Essobi | I'd say ulaw if you got the bandwidth all the way across so the colo's are not transcoding to PRI. |
| 06:02.09 | xeet2 | essobi: its great, I work at a phone company and I'm not allowed to use sip |
| 06:02.15 | Essobi | HAHAHAHAHAHA. |
| 06:02.19 | Essobi | Are you serious? |
| 06:02.23 | xeet2 | yes |
| 06:02.29 | xeet2 | maybe you've heard of them... mci? |
| 06:02.31 | xeet2 | fuckers |
| 06:02.37 | Essobi | MAha. |
| 06:02.40 | Essobi | That's funny. |
| 06:02.41 | marshall | essobi, we are the colo |
| 06:02.47 | Essobi | WTF you do working for MCI? |
| 06:03.21 | xeet2 | they bought us and they pay a good salary... slowly trying to break way though |
| 06:03.25 | Essobi | that's funny.. no sip. |
| 06:03.45 | Essobi | y'know.. the mob pays good too, but I wouldn't work for them either. |
| 06:03.50 | xeet2 | hehe |
| 06:04.08 | Essobi | time to fire up gdb me thinks |
| 06:04.29 | yertle | ebbbbers |
| 06:05.26 | xeet2 | I met him once, reminded me of a fish out of water |
| 06:06.18 | xeet2 | he was in a hurry to shake everyone's hand and then get out of the "common employee buildings" |
| 06:06.44 | xeet2 | hope he enjoys his common cellmates |
| 06:06.52 | yertle | snicker |
| 06:09.01 | xeet2 | marshall: did you try out echo cancellation any? |
| 06:09.13 | marshall | yes |
| 06:09.21 | xeet2 | no luck? |
| 06:09.21 | marshall | a bunch of different combinations |
| 06:09.26 | marshall | intermittent |
| 06:09.30 | marshall | maybe I am being too picky |
| 06:09.47 | marshall | I've tested other voip solutions that were nearly perfect |
| 06:09.55 | marshall | thats what I am expecting |
| 06:10.00 | xeet2 | if you're even noticing it, thats not really picky at all |
| 06:10.16 | marshall | thats good to know |
| 06:10.17 | xeet2 | well, you mean other solutions outside of *? |
| 06:10.21 | marshall | right |
| 06:10.40 | xeet2 | mmm, alot of vendors actually use g.168 echo cancellation |
| 06:10.51 | xeet2 | which actually works about 99% of the time |
| 06:11.02 | marshall | is that in the handset? |
| 06:11.19 | marshall | Im reading from a google post that the polycom handsets dont actually do cancellation yet |
| 06:11.31 | xeet2 | we wound up replacing a bunch of zaptel + cb solutions with multitech boxes running sip to * because of that |
| 06:11.36 | yertle | which polycom do you have? i have an ip300.. thing is dumb |
| 06:11.47 | marshall | 300 and 500 |
| 06:11.51 | marshall | the 300 is the one giving me grief |
| 06:12.27 | *** join/#asterisk shmooz (~nobody@host6411912762.biz.tor.fcibroadband.com) |
| 06:12.30 | shmooz | hi |
| 06:12.56 | xeet2 | marshall: have you talked to your pri provider? sometimes their gear can do echo cancellation, just have to ask |
| 06:13.10 | marshall | no - I had no idea |
| 06:13.17 | xeet2 | *sometimes* |
| 06:13.24 | marshall | I'll definitely try that tomorrow |
| 06:13.49 | marshall | they correct the echo coming in over the pri? |
| 06:13.57 | marshall | makes sense |
| 06:13.58 | xeet2 | if they're the ilec, usually forget about it, thats a sign that you're doing something outside of what they want you to do, but alot of clecs will do it |
| 06:14.06 | xeet2 | well, it can come from alot of places |
| 06:14.16 | marshall | what are clecs? |
| 06:14.51 | xeet2 | if you're in the us, clecs/ilecs refer to the local incumbent and competitive telco's |
| 06:15.01 | marshall | ahhh |
| 06:15.23 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:15.39 | marshall | Im in Canada, competing with pigeons and party lines |
| 06:15.47 | xeet2 | if you're getting echo on an all voip/digital call, and the only analog point is at the remote end, then I'd say 90% chance thats where its coming from |
| 06:15.53 | xeet2 | hehe |
| 06:16.13 | marshall | thats what I am thinking xeet2 |
| 06:16.18 | marshall | just have to squash it |
| 06:16.24 | xeet2 | er, that sentence was contradictory, but you get what I mean =P |
| 06:17.04 | marshall | I'll give it a try tomorrow, thanks for all your help, you gave me some good new ideas to try |
| 06:17.54 | xeet2 | if that doesn't work, if you have other gear available or some $$, might be worth trying a gateway that does g.168 to get to * instead |
| 06:18.15 | xeet2 | (for instance,if you have a cisco 26xx or 36xx router, you can do it there too) |
| 06:18.31 | marshall | we're setting up a few dozen sites, not cost effective |
| 06:18.48 | xeet2 | few dozen sites all with pri's? |
| 06:19.01 | marshall | all trunking to the pri's |
| 06:19.21 | marshall | oh I see what you mean |
| 06:19.28 | xeet2 | well you don't have to have another gateway at each site, just to talk to the pstn |
| 06:19.38 | xeet2 | just a last resort option though |
| 06:19.58 | xeet2 | alot of people use cisco gear to bring the pstn into * |
| 06:20.00 | *** join/#asterisk peted20 (~chatzilla@d2-168.rb.gh.centurytel.net) |
| 06:20.06 | marshall | interesting |
| 06:20.36 | marshall | so pri-cisco-T100? |
| 06:20.50 | xeet2 | pri-cisco(or other vendor)-sip-* |
| 06:21.23 | marshall | whats the best device going these days? |
| 06:21.36 | xeet2 | depends on what your needs are |
| 06:21.40 | xeet2 | =) |
| 06:21.52 | marshall | I need to not hear everything I say twice :) |
| 06:22.25 | xeet2 | so far we like multitech and cisco, the multitech's can be flaky when not configured right, and the cisco's can be hard to configure if you're not familiar |
| 06:23.24 | xeet2 | but there are quite a number of vendors, might want to ask on the mailing lists if you come down to making that decision |
| 06:23.53 | marshall | sounds like a plan |
| 06:23.58 | marshall | thanks again - time for sleep |
| 06:24.06 | marshall | if your around tomorrow I'll let you know how it went |
| 06:24.16 | *** part/#asterisk marshall (~test@S0106000f66563988.wp.shawcable.net) |
| 06:27.01 | Landrocker | I'm having problems using iaxcomm under linux kernel 2.6.10 - I think the problem is alsa related - the sound device is set to be /dev/dsp and when I place a call the status bar just says "Can't start audio" |
| 06:27.05 | Landrocker | anyone have any ideas? |
| 06:30.25 | *** join/#asterisk pratik (~root@202-149-48-204.broadband.isp.exatt.net) |
| 06:35.06 | rvhi | have some issues with dtmf generating/detection, got this error: Auto-deactivating generator |
| 06:37.48 | opus___ | hey, does anyone use broadvoice here. I get the 'I'm sorry your call can not be completed " from the bv chick, but I can call tech support... is there a reason why I can't call out? |
| 06:38.00 | opus___ | I got the cheapo $5.95 plan |
| 06:38.38 | opus___ | suppose to have minutes... |
| 06:39.12 | xeet2 | are you using the correct password? its not the same one you use to log in to their web interface |
| 06:40.10 | opus___ | yes, i authenicate atleast |
| 06:40.53 | xeet2 | hmmm |
| 06:40.53 | xeet2 | odd |
| 06:40.59 | xeet2 | so, what did bv say? |
| 06:41.40 | opus___ | 'i'm sorry but i can not reach the number |
| 06:41.49 | opus___ | if you want I can give you an extension and you can try yourself |
| 06:41.58 | xeet2 | I mean their tech support |
| 06:42.24 | opus___ | Dunno, I don't think they are up let me ask them. |
| 06:44.53 | *** part/#asterisk yertle (yertle@ip68-6-98-122.sb.sd.cox.net) |
| 06:49.13 | Beirdo | link2voip "support" is really starting to get on my nerves |
| 06:49.21 | Beirdo | answer the emails, dammit! |
| 06:50.18 | opus___ | hmmm |
| 06:50.30 | opus___ | beirdo whats not workign |
| 06:50.58 | opus___ | xeet2 - hmm, from some reason i can't send dtmf tones to their voicetre |
| 06:51.02 | opus___ | voice menu |
| 06:51.03 | Beirdo | still waiting for my US50CA toll free to be properly activated |
| 06:51.31 | xeet2 | opus: bv requires inband dtmf |
| 06:51.38 | Beirdo | and they offered to give my the number of the DID it is supposed to be forwarded to.. in the mean time. |
| 06:51.54 | Beirdo | and they said that a week ago, and I've now sent 2 reminders |
| 06:54.32 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l02m-4-165.d4.club-internet.fr) |
| 07:20.52 | *** join/#asterisk Corydon76-home (brown@pcp08665860pcs.500ash01.tn.comcast.net) |
| 07:22.39 | *** join/#asterisk R3DB0x (nobody@66.142.28.36) |
| 07:25.14 | *** join/#asterisk bjohnson_ (~bjohnson@66.11.165.161) |
| 07:28.05 | rvhi | anyone know how to get rid of this: Mar 17 21:25:03 DEBUG[1720339]: channel.c:470 ast_channel_walk_locked: Avoiding initial deadlock for 'Zap/3-1' |
| 07:34.35 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
| 07:36.16 | *** join/#asterisk booyeah11 (~afdas@cpe-24-175-29-253.houston.res.rr.com) |
| 07:37.33 | booyeah11 | lo |
| 07:41.06 | *** join/#asterisk cjk (~cjk@80.92.75.13) |
| 07:41.22 | cjk | hi, anyone who can tell me if the password in the sip traffic is cleartext? |
| 07:41.29 | cjk | i think it is. anyone can confirm? |
| 07:41.37 | booyeah11 | yes |
| 07:41.39 | Mavvie | cjk: use ngrep to see what goes over the line. |
| 07:42.04 | booyeah11 | tethereal |
| 07:42.09 | booyeah11 | or tcpdump |
| 07:42.20 | Mavvie | I can't see, because my password is "ZAphbHNrZGpsYSBqZGxhIHhqbGFzayB4bGprYXMgeGxhIGRqYXMgamRs" |
| 07:43.02 | cjk | Mavvie, yeah i got similra results using tethereal. |
| 07:43.08 | booyeah11 | did you use tethereal or tcpdump |
| 07:43.10 | cjk | just wanted a confirmation |
| 07:43.29 | booyeah11 | unless it is hashed some how |
| 07:43.33 | booyeah11 | it should be cleartext |
| 07:44.44 | modulus_ | cjk, what does auth=md5 do? |
| 07:45.56 | SwedMiro | Does anyon have a good tip on a program that can check a router and what it does to VoIP traffic? |
| 07:46.06 | SwedMiro | anyone |
| 07:46.19 | cjk | modulus_, i geuss i schould look it up |
| 07:46.24 | booyeah11 | QoS ? |
| 07:46.25 | cjk | maybe creating md5 hashes |
| 07:46.33 | *** join/#asterisk Alexis (~alexis@www.trim.it) |
| 07:47.43 | *** join/#asterisk langals (~icechat5@196.7.14.183) |
| 07:47.59 | cjk | well i sniffed now for my password |
| 07:48.03 | cjk | and i cant really see it |
| 07:48.39 | Alexis | hi all |
| 07:49.22 | booyeah11 | cjk: send me the tethereal or tcpdump output |
| 07:49.26 | booyeah11 | i can find it |
| 07:51.16 | cjk | booyeah11, ok i did tcpdump -vvvvvvvvvvvvv | grep password |
| 07:51.20 | langals | Hi there guys. Someone pointed me to the Asterisk@Home website because I am fairly new to Asterisk. But looking at the installation instructions it seems to suggest that it will reformat my whole harddrive with Asterisk and Linux!! Is this true and why does it doe this? |
| 07:51.21 | cjk | i rebooted my telehphone |
| 07:51.26 | cjk | whcih registered |
| 07:51.37 | cjk | but grep did not find anything |
| 07:52.02 | booyeah11 | you need to use tethereal or tcpdump |
| 07:52.22 | booyeah11 | ngrep might not find it |
| 07:52.30 | cjk | booyeah11, i did use tcpdump |
| 07:52.47 | cjk | booyeah11, i cant send you the whole output. i have 500 users and alot of traffic |
| 07:52.57 | booyeah11 | thats why you filter it |
| 07:53.01 | booyeah11 | for a certain host and port |
| 07:53.16 | booyeah11 | i believe port 5080 |
| 07:53.18 | booyeah11 | *5060 |
| 07:53.41 | cjk | hmmm |
| 07:53.56 | cjk | see above. is that commande not good? i know its basic |
| 07:54.02 | cjk | but it should find something, no? |
| 07:54.03 | booyeah11 | let me make the command |
| 07:54.06 | cjk | ok |
| 07:54.07 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 07:54.55 | booyeah11 | tethereal port 5060 |
| 07:55.11 | booyeah11 | tethereal port 5060 -w foo.dmp |
| 07:55.26 | booyeah11 | send that |
| 07:56.32 | *** join/#asterisk bjohnson (~bjohnson@66.11.165.161) |
| 07:57.40 | *** join/#asterisk tvmkid (~chatzilla@202.88.232.14) |
| 08:01.43 | *** join/#asterisk CoderCR (~creyna@adsl-67-112-135-29.dsl.sndg02.pacbell.net) |
| 08:01.45 | CoderCR | hey guys |
| 08:01.51 | CoderCR | does anyone here have a fax machine? |
| 08:01.54 | booyeah11 | yes |
| 08:02.12 | CoderCR | could you send me a fax to 858 558 1200 for an asterisk fax test |
| 08:02.21 | CoderCR | just seeing if i can receive them properly |
| 08:02.41 | booyeah11 | let me set up my fax machine |
| 08:02.46 | booyeah11 | wait |
| 08:02.47 | CoderCR | cheers |
| 08:02.52 | booyeah11 | i cant send long disntance |
| 08:02.53 | booyeah11 | nm |
| 08:03.01 | CoderCR | o |
| 08:03.04 | CoderCR | well let me call you |
| 08:03.09 | CoderCR | and you send it to me |
| 08:03.12 | CoderCR | would that work |
| 08:03.15 | CoderCR | it still works |
| 08:03.16 | CoderCR | i call you |
| 08:03.19 | booyeah11 | not sure how that would work |
| 08:03.23 | CoderCR | you press send and i press send |
| 08:03.24 | booyeah11 | i have an asterisk system set up |
| 08:03.26 | CoderCR | it works |
| 08:03.29 | booyeah11 | 8662291552 |
| 08:03.31 | Jer13261 | DISA? |
| 08:03.34 | booyeah11 | its a conference system i wrote |
| 08:03.49 | booyeah11 | well not wrote, but extended |
| 08:04.08 | CoderCR | what is the number |
| 08:04.11 | CoderCR | your fax ? |
| 08:04.31 | booyeah11 | 713 842 4251 |
| 08:05.50 | booyeah11 | is there a way to fax inside asterisk? |
| 08:07.09 | booyeah11 | ok its ringing |
| 08:07.56 | CoderCR | yes there is |
| 08:08.00 | *** part/#asterisk shmooz (~nobody@host6411912762.biz.tor.fcibroadband.com) |
| 08:08.17 | CoderCR | that did not work |
| 08:08.23 | CoderCR | it said receiving but that was it |
| 08:08.40 | booyeah11 | it said line error |
| 08:08.48 | booyeah11 | is it going over a voip connection? |
| 08:08.54 | CoderCR | no |
| 08:08.58 | CoderCR | analog lines |
| 08:09.04 | booyeah11 | then it should work fine |
| 08:09.05 | CoderCR | but i am trying to configure the audio for it |
| 08:09.06 | booyeah11 | .... |
| 08:09.13 | CoderCR | lots of line noise in house |
| 08:09.17 | CoderCR | let me try one last thing |
| 08:11.23 | booyeah11 | k |
| 08:11.55 | CoderCR | ok |
| 08:11.59 | CoderCR | i am going to try again |
| 08:12.02 | booyeah11 | ok |
| 08:13.04 | CoderCR | get ready to send when i call |
| 08:13.25 | CoderCR | ok it is calling |
| 08:13.43 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
| 08:14.11 | booyeah11 | ringing |
| 08:15.36 | CoderCR | remember i am trying to receive |
| 08:15.40 | CoderCR | not send a fax |
| 08:15.44 | booyeah11 | its saying recieving |
| 08:15.52 | CoderCR | yours should say sending |
| 08:15.53 | booyeah11 | can you test this out |
| 08:16.00 | booyeah11 | 8662291552 |
| 08:16.01 | CoderCR | put a piece of paper |
| 08:16.03 | booyeah11 | it says recieve |
| 08:16.04 | CoderCR | sure |
| 08:16.15 | booyeah11 | put did nothing |
| 08:16.33 | CoderCR | you need to answer and hit start |
| 08:26.29 | *** join/#asterisk MuppetMaster (~MuppetMas@a82-92-73-185.adsl.xs4all.nl) |
| 08:26.55 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
| 08:27.52 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 08:29.45 | *** join/#asterisk Zgarbi (~my@212.58.125.68) |
| 08:32.18 | Zgarbi | Is anyone here who would like help me with asterisk feature? |
| 08:32.36 | Jer13261 | what doyou need? |
| 08:32.46 | Zgarbi | http://bugs.digium.com/bug_view_page.php?bug_id=0003778 |
| 08:33.05 | Zgarbi | I report as a bug but reveive ansver that is not a bug |
| 08:33.23 | Zgarbi | answer was: This is a known missing feature, not a bug. We match on the IP address of the peer for incoming calls. Find someone on the #irc to help you or on the -users mailing list. |
| 08:34.05 | Jer13261 | i am doing this so maybe i can help you PM me |
| 08:34.58 | *** part/#asterisk tecnico (~tecnico@user-24-236-123-31.knology.net) |
| 08:35.03 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 08:35.28 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@whiteoaks5024-2.clients.easynet.fr) |
| 08:36.10 | *** join/#asterisk leolarrel (~root@220-130-142-43.HINET-IP.hinet.net) |
| 08:36.34 | *** part/#asterisk leolarrel (~root@220-130-142-43.HINET-IP.hinet.net) |
| 08:39.27 | CoderCR | well it did not communicate at all |
| 08:39.56 | booyeah11 | ok it said pages 0, time 40", note cancel |
| 08:40.05 | *** part/#asterisk MuppetMaster (~MuppetMas@a82-92-73-185.adsl.xs4all.nl) |
| 08:40.06 | booyeah11 | in the transaction report |
| 08:40.57 | CoderCR | ok |
| 08:41.03 | CoderCR | i need someone to send me a fax for a test |
| 08:41.18 | booyeah11 | yeah i cant send it from this line |
| 08:41.30 | booyeah11 | i can try to set something up on asterisk i guess |
| 08:41.44 | CoderCR | dont worry about it |
| 08:41.50 | *** join/#asterisk datareactor (datareacto@203.81.192.33) |
| 08:42.18 | datareactor | can i use cisco 7960G with Vonage services |
| 08:42.44 | Jer13261 | NO :P |
| 08:42.58 | Jer13261 | welllllllllllll |
| 08:43.03 | Jer13261 | with a softphone acct yes |
| 08:43.49 | datareactor | Jer13261 are they use SIP ? |
| 08:44.08 | Jer13261 | yes but they wont let you use your own device unless you pay them extra |
| 08:44.32 | booyeah11 | vonage sux0rs |
| 08:44.46 | Jer13261 | well i wanted to say that but I wasnt sure if i could here heheh |
| 08:44.53 | Jer13261 | vonage blows ......................................A#^$)((#&$()#$()#& |
| 08:46.37 | CoderCR | i need to get a fax sent! |
| 08:46.59 | Jer13261 | CoderCR: maybe i can help? |
| 08:47.01 | Jer13261 | what are you trying to do |
| 08:49.04 | CoderCR | get a fax |
| 08:49.07 | CoderCR | receive a fax |
| 08:49.09 | CoderCR | i can send |
| 08:49.18 | CoderCR | but i need to make sure i can receive faxes as well |
| 08:49.21 | Jer13261 | what is your fax did # |
| 08:49.39 | CoderCR | 858 558 1200 |
| 08:49.52 | Jer13261 | dialing |
| 08:49.53 | *** join/#asterisk meppl (~mephisto@pD9E68DDF.dip.t-dialin.net) |
| 08:49.56 | CoderCR | cheers |
| 08:49.58 | booyeah11 | im installing some fax software now |
| 08:50.08 | Jer13261 | no answer |
| 08:50.11 | *** join/#asterisk hanhoong (~hanhoong@218.111.48.15) |
| 08:50.29 | Jer13261 | voice mail box |
| 08:51.38 | Jer13261 | not really useful if your system doesnt answer the line |
| 08:52.30 | CoderCR | hmm it rang but did not detect a fax |
| 08:52.47 | booyeah11 | what do you use to send faxes in asterisk? |
| 08:52.50 | CoderCR | could you try one more time |
| 08:52.52 | Jer13261 | didnt even pickup on this end |
| 08:52.53 | Jer13261 | sure thing |
| 08:53.12 | *** join/#asterisk dg1nsw (~schulte@gate.sympat.de) |
| 08:53.18 | CoderCR | it is not detecting a fax |
| 08:53.21 | datareactor | Jer13161 is there any voip provider which give me US incoming number # |
| 08:53.26 | CoderCR | so it is not been routed |
| 08:53.33 | Jer13261 | hahah |
| 08:53.38 | Jer13261 | there is 1000's of them |
| 08:53.52 | *** join/#asterisk Guybrush|work (~Guybrush|@mail.paneura.com) |
| 08:53.54 | CoderCR | jer: are you using a fax machine to dial it? |
| 08:54.04 | datareactor | can u name a few |
| 08:54.08 | Jer13261 | do you do tone detection |
| 08:54.16 | Guybrush|work | anyone has reports of bristuff causing problems with spandsp ? |
| 08:54.27 | datareactor | no |
| 08:54.40 | Guybrush|work | i'm getting mad about my spandsp not working |
| 08:54.45 | Jer13261 | nufone teliax BV voipjet etc |
| 08:55.06 | Jer13261 | CoderCR you doing fax tone detection? |
| 08:55.16 | *** join/#asterisk hanhoong (~hanhoong@218.111.48.15) |
| 08:55.20 | CoderCR | yes |
| 08:55.33 | Jer13261 | thats why....I am waiting for an answer |
| 08:55.37 | CoderCR | lol |
| 08:55.38 | CoderCR | sorry |
| 08:55.40 | Jer13261 | and i never see one on this end |
| 08:55.44 | CoderCR | forgot to mention that |
| 08:55.44 | Jer13261 | so i dont send tone |
| 08:55.54 | CoderCR | you cannot send tone? |
| 08:55.57 | Jer13261 | how can i send tone to something that isnt answered |
| 08:56.06 | Jer13261 | well i CAN but.....lol |
| 08:56.16 | CoderCR | <PROTECTED> |
| 08:56.16 | CoderCR | <PROTECTED> |
| 08:56.16 | CoderCR | <PROTECTED> |
| 08:56.16 | CoderCR | <PROTECTED> |
| 08:56.16 | CoderCR | <PROTECTED> |
| 08:56.17 | CoderCR | <PROTECTED> |
| 08:56.19 | CoderCR | <PROTECTED> |
| 08:56.21 | CoderCR | <PROTECTED> |
| 08:56.23 | CoderCR | <PROTECTED> |
| 08:56.25 | CoderCR | <PROTECTED> |
| 08:56.27 | CoderCR | <PROTECTED> |
| 08:56.29 | CoderCR | <PROTECTED> |
| 08:56.31 | CoderCR | <PROTECTED> |
| 08:56.32 | Jer13261 | ok ok ok |
| 08:56.33 | CoderCR | <PROTECTED> |
| 08:56.35 | CoderCR | <PROTECTED> |
| 08:56.37 | CoderCR | <PROTECTED> |
| 08:56.38 | Jer13261 | having fun yet? |
| 08:56.39 | CoderCR | <PROTECTED> |
| 08:56.40 | Jer13261 | lol |
| 08:56.41 | CoderCR | <PROTECTED> |
| 08:56.43 | CoderCR | <PROTECTED> |
| 08:56.45 | CoderCR | <PROTECTED> |
| 08:56.47 | CoderCR | <PROTECTED> |
| 08:56.49 | CoderCR | <PROTECTED> |
| 08:56.51 | CoderCR | <PROTECTED> |
| 08:56.53 | CoderCR | <PROTECTED> |
| 08:56.53 | booyeah11 | hmm |
| 08:56.55 | CoderCR | <PROTECTED> |
| 08:56.55 | RoyK | ~pastebin |
| 08:56.56 | jbot | pastebin is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 08:56.57 | CoderCR | oops |
| 08:56.59 | CoderCR | that was an accident |
| 08:57.01 | CoderCR | sorry |
| 08:57.02 | booyeah11 | im not going to be able to implement a fax transmit tonight |
| 08:57.03 | RoyK | ~lart CoderCR with pastebin |
| 08:57.06 | booyeah11 | the code is too old |
| 08:57.26 | Jer13261 | lol lol |
| 08:57.56 | CoderCR | booyeah what did you try out? |
| 08:58.16 | booyeah11 | app_txfax |
| 08:58.41 | *** join/#asterisk ard (~ard@2001:7b8:32d:0:20c:6eff:fe18:d11f) |
| 08:58.54 | booyeah11 | doesnt like the new asterisk include files |
| 08:59.05 | CoderCR | ftp://ftp.opencall.org/pub/spandsp/spandsp-0.0.1/ |
| 08:59.06 | Jer13261 | ard: ipv6? |
| 08:59.18 | ard | jup... |
| 08:59.21 | booyeah11 | ok ill try that one |
| 08:59.28 | ard | but with a stupid question... |
| 08:59.36 | booyeah11 | im using the latest one |
| 08:59.48 | CoderCR | dont |
| 08:59.52 | CoderCR | latest are unstable |
| 08:59.54 | ard | I've got asterisk up and running with a te110p, I am just wondering if I have to use a twisted cable to connect to the telco or not? |
| 09:00.00 | CoderCR | make sure you have tiff library installed as well |
| 09:00.06 | booyeah11 | yeah its installed |
| 09:00.15 | booyeah11 | <PROTECTED> |
| 09:00.19 | booyeah11 | spandsp compiled |
| 09:00.22 | ard | And if so, they are probably not compatible with plain ethernet twisted cables? |
| 09:00.23 | booyeah11 | but not the tx file |
| 09:00.29 | *** join/#asterisk visik7 (~ciao@visik7.user) |
| 09:00.35 | booyeah11 | im using the latest stable asterisk |
| 09:00.39 | CoderCR | how did you try and compile the tx file? |
| 09:00.50 | booyeah11 | just make |
| 09:00.53 | booyeah11 | without any options |
| 09:00.56 | Jer13261 | what do you mean ard? |
| 09:01.01 | Jer13261 | a cable is a cable :) |
| 09:01.04 | ard | Jup |
| 09:01.48 | ard | But if I am correct it is comparable with BRI. And I don't know if I just connet the telco with a straight cable into the PRI card. |
| 09:01.58 | booyeah11 | libbtiff 3 or 4? |
| 09:02.22 | Jer13261 | i'd use straight |
| 09:02.27 | ard | ah.. |
| 09:02.30 | ard | I guessed to... |
| 09:02.36 | booyeah11 | install newest libtiff ? |
| 09:02.44 | booyeah11 | im using debian woody one |
| 09:02.48 | ard | well I am off making my server reachable, and back again... |
| 09:02.57 | Guybrush|work | i'm trying to setup spandsp on woody too |
| 09:03.27 | rvhi | what the market price for setting a medium size pbx system? say 40 phones |
| 09:03.29 | ard | Jer13261 : The question arose when I pulled the plug from the original PBX out of the telco PRI terminator... |
| 09:03.42 | rvhi | with vm and acd? |
| 09:03.52 | visik7 | why there is FUD on the wiki ? |
| 09:03.55 | booyeah11 | im installing the latest stable tiff |
| 09:03.58 | *** join/#asterisk oej (~oej@apollo.webway.se) |
| 09:04.26 | booyeah11 | then installing the spandsp you sent me |
| 09:05.45 | booyeah11 | getting errors on make |
| 09:06.00 | booyeah11 | t4.c:38:21: tiffiop.h: No such file or directory |
| 09:07.59 | Guybrush|work | booyeah11: |
| 09:08.16 | Guybrush|work | you need to copy the tiffiop.h and tiff_dir.h in /usr/include |
| 09:08.32 | Guybrush|work | you find those on spandsp site, in the folder where you got the source |
| 09:09.14 | booyeah11 | still problems |
| 09:09.18 | booyeah11 | ok |
| 09:09.57 | booyeah11 | ok doing that now |
| 09:10.17 | booyeah11 | ghetto way of installing it |
| 09:12.56 | booyeah11 | In file included from /usr/include/tiffiop.h:45, |
| 09:12.56 | booyeah11 | <PROTECTED> |
| 09:12.56 | booyeah11 | /usr/include/tif_dir.h:240: error: conflicting types for `TIFFFieldInfo' |
| 09:13.22 | booyeah11 | im going to have to mess with this later |
| 09:15.35 | *** join/#asterisk fitzel (~flint@p3EE390BD.dip0.t-ipconnect.de) |
| 09:16.13 | fitzel | Hi, can I write the output of the console (like show dialplan) easily to a file? |
| 09:16.41 | Guybrush|work | fitzel, look at /etc/asterisk/logger.conf |
| 09:18.48 | ard | Sheez... Now I am really getting confused... I got a termination device from the telco, but now I don't really know which is the right cable. One cable comes from the pbx and looks like two coax, ending in an rj45, and another from the pbx with plain cat5. I guess the latter is the one I have to replace with a connection to the te110p |
| 09:19.34 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 09:20.23 | ard | Hmmm, at least one side says S-HDLC-U |
| 09:20.47 | *** join/#asterisk christo (~chris@office.enovi.com) |
| 09:23.07 | christo | morning |
| 09:23.34 | *** join/#asterisk afe ([AfqlG6HLc@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
| 09:23.52 | Alexis | please, is * working fine on a mandrake X ? |
| 09:25.46 | christo | Alexis - I've not tried it on mandrake. sry |
| 09:26.36 | *** join/#asterisk X-Gen (~x-gen@rrba-146-121-118.telkomadsl.co.za) |
| 09:27.54 | CoderCR | ok |
| 09:27.58 | CoderCR | for it working with 1.0.6 |
| 09:28.04 | CoderCR | damn it is out of day |
| 09:28.06 | CoderCR | date |
| 09:30.15 | booyeah11 | hmm |
| 09:30.28 | booyeah11 | the app_faxtx is throwing mad errors |
| 09:30.37 | X-Gen | anyone got pictures of mark and his new tub ? |
| 09:31.11 | booyeah11 | no |
| 09:35.48 | Alexis | christo: wich distrib du you use ? |
| 09:36.02 | *** join/#asterisk pranav (pranav@202.149.48.198) |
| 09:37.07 | pranav | hello everyone |
| 09:38.16 | pranav | i want to record the calls made by asterisk |
| 09:39.03 | CoderCR | booyeah: i think i got it to build. |
| 09:39.35 | pranav | hi booyeah |
| 09:39.49 | CoderCR | pranav: do you have a fax machine? |
| 09:39.57 | pranav | y |
| 09:40.05 | CoderCR | i want to see if i can receive faxes |
| 09:40.24 | pranav | well no |
| 09:40.48 | *** join/#asterisk bjohnson (~bjohnson@66.11.165.161) |
| 09:41.40 | pranav | Codercr:do you know how to record calls using asterisk |
| 09:41.52 | Guybrush|work | i have a segmentation fault if using spandsp 0.0.1 with libtiff 3.7.2 |
| 09:41.57 | Guybrush|work | anyone has a clue ? |
| 09:42.12 | christo | Alexis - Slackware |
| 09:42.14 | Guybrush|work | with 3.5.7 the faxes are completely screwed |
| 09:42.57 | booyeah11 | dunno, the build system seems fuxr0red |
| 09:43.26 | pranav | hi booyeah11 i got my fwd stuff working |
| 09:43.46 | pranav | i guess yesterday it was booyeah7 |
| 09:44.31 | booyeah11 | cool |
| 09:44.36 | booyeah11 | yeah |
| 09:44.49 | pranav | yup |
| 09:47.07 | *** join/#asterisk MuppetMaster (~MuppetMas@a82-92-73-185.adsl.xs4all.nl) |
| 09:49.06 | *** part/#asterisk pranav (pranav@202.149.48.198) |
| 09:49.06 | *** join/#asterisk pranav (pranav@202.149.48.198) |
| 09:52.59 | meppl | guten morgen |
| 09:53.01 | CoderCR | anyone here have a fax machine? |
| 09:53.19 | MuppetMaster | CoderCR: Somewhat. |
| 09:53.36 | CoderCR | could you try and send a fax to 858 558 1200 |
| 09:53.51 | CoderCR | as long as you know you can send faxes i dont care how i get it |
| 09:53.52 | CoderCR | :) |
| 09:54.27 | booyeah11 | i wish i had a line i could go outbound |
| 09:54.28 | MuppetMaster | CoderCR: Give this a try: http://www.tpc.int/ |
| 09:54.38 | booyeah11 | i guess i could hook it up to a sipura device |
| 09:54.44 | MuppetMaster | That way you may run several tests. I am looking now to see if they support your area code, I take it is in the US. |
| 09:54.45 | *** join/#asterisk iamx (~DmD@pppoe59-99-luxdsl-246.pt.lu) |
| 09:54.47 | iamx | Hi |
| 09:55.30 | MuppetMaster | CoderCR: It appears your number is accessible, so try to use the webfrom: http://www.tpc.int/sendfax.html?destnumber=1%20858%20558%201200 |
| 09:56.08 | MuppetMaster | iamx: Hi |
| 09:56.17 | CoderCR | cheers |
| 09:56.25 | booyeah11 | i tried that also |
| 09:56.41 | booyeah11 | not getting in ringing |
| 09:56.43 | MuppetMaster | CoderCR: No worries, good luck! BTW - if you are testing Fax capabilities over VoIP with Asterisk (spanDSP) I have had limited success even with ULAW only. |
| 09:56.53 | iamx | Did anyone of you have the problem that capi doesn't find the capi device with the specified msn ? |
| 09:57.09 | iamx | Mar 18 09:43:24 NOTICE[3970]: didn't find capi device with outgoing msn = 26310274. you should check your config! |
| 09:57.29 | iamx | but the config should be ok because it worked for 1 day or so |
| 09:59.00 | MuppetMaster | iamx: What changed between now and a few days ago? |
| 09:59.16 | Guybrush|work | can anyone give me a working, valid fax tiff file to use for tests ? |
| 09:59.19 | CoderCR | well that fax site does not seem to be working |
| 09:59.29 | iamx | nothing, absolutely nothing thats the problem, the machine had been rebooted |
| 09:59.47 | iamx | after the reboot it didn't work anymore |
| 09:59.56 | MuppetMaster | iamx: Have you tried a power down for a bit and then restarting, maybe the card is just a bit 'fuzzy'... |
| 10:00.10 | iamx | but it works with incoming calls so it can't be a capi issue, i think |
| 10:00.29 | MuppetMaster | CoderCR: Not always 100% reliable as it is a community driven fax service. The other option is to sign up with http://www.efax.com, which is what I use, but of course that costs $$$. |
| 10:01.19 | CoderCR | well |
| 10:01.30 | CoderCR | Muppet send me some text efax |
| 10:01.36 | CoderCR | test efax |
| 10:01.42 | CoderCR | i will send you a 2 dolars over paypal |
| 10:02.27 | MuppetMaster | Just a moment, seeing if my eFax client works under OSX now, or if I need to switch machines. |
| 10:02.35 | MuppetMaster | Also, just a paypal for the amount is fine. |
| 10:02.40 | MuppetMaster | Won't be $2. |
| 10:03.28 | CoderCR | ok |
| 10:05.06 | iamx | http://pastebin.ca/7686 here's the debug output but it doesn't say much |
| 10:05.31 | MuppetMaster | CoderCR: Sending now. |
| 10:06.57 | CoderCR | hmm |
| 10:07.01 | CoderCR | it did not go trhough |
| 10:07.07 | MuppetMaster | Checking the status online, just a moment. |
| 10:07.13 | CoderCR | it hang up before the fax picked up or something |
| 10:09.18 | *** join/#asterisk Inv_arp (junya@adsl-8-232-188.mia.bellsouth.net) |
| 10:10.04 | *** join/#asterisk nicknick (~nicknick@cf1.74899.hso.uk.com) |
| 10:10.07 | MuppetMaster | CoderCR: Did a call attempt occur? As I don't see it in the activity log yet, although eFax has scheduled. |
| 10:10.31 | CoderCR | yes |
| 10:10.34 | CoderCR | it is sending |
| 10:10.38 | Inv_arp | eric_: u around? |
| 10:10.46 | MuppetMaster | CoderCR: Okay, let me know if it comes through. |
| 10:10.58 | Inv_arp | ~seen eric_ |
| 10:11.00 | jbot | eric_ is currently on #asterisk |
| 10:13.23 | CoderCR | did not work |
| 10:14.51 | booyeah11 | that app didnt work for me |
| 10:14.57 | booyeah11 | efax sucks |
| 10:15.05 | MuppetMaster | CoderCR: Looking at the log on my side |
| 10:16.58 | CoderCR | ok |
| 10:17.41 | MuppetMaster | booyeah11: I agree, but not a lot of other reasonable options. Do you have any recommendations? |
| 10:22.37 | fac_ | is it possible to connect with skype by SIP? |
| 10:23.14 | cypromis | no |
| 10:23.48 | fac_ | cypromis hi. ;] |
| 10:24.19 | booyeah11 | MuppetMaster: make your own |
| 10:24.19 | Guybrush|work | could you tell me which versions of spandsp, asterisk and libtiff you use with success ? |
| 10:24.26 | booyeah11 | soon i will have the ability |
| 10:24.26 | Guybrush|work | i cannot get it to work... |
| 10:25.05 | CoderCR | Muppet |
| 10:25.20 | CoderCR | ok i am going to see if i can use rxfax |
| 10:25.25 | MuppetMaster | CoderCR: I saw that it went through. |
| 10:25.36 | MuppetMaster | CoderCR: Just got this email: |
| 10:25.51 | CoderCR | after 3 retrys |
| 10:25.55 | CoderCR | it took too long |
| 10:25.57 | CoderCR | lets try this |
| 10:26.11 | CoderCR | it was sent to a real fax machine no asterisk |
| 10:26.17 | CoderCR | try and send it to asterisk now |
| 10:27.11 | Inv_arp | any good providers that email faxes like efax? |
| 10:27.15 | booyeah11 | email doesnt mean it went through |
| 10:27.39 | MuppetMaster | booyeah11: The contents of the email indicated whether it was sent or not. |
| 10:27.53 | CoderCR | Muppet: could you try 1 more time please :) |
| 10:27.59 | Zeeek | CoderCR for info, I can always get faxes from efax/jfax/j2 to spandsp but many fax machines and even software faxes will not work for me. |
| 10:28.02 | MuppetMaster | CoderCR: Will do, just a moment. |
| 10:29.28 | MuppetMaster | CoderCR: Just launched another test with a single cover page. |
| 10:30.00 | CoderCR | brb |
| 10:30.02 | CoderCR | i need to go pee |
| 10:30.02 | booyeah11 | anyone have a FWD number? |
| 10:30.07 | booyeah11 | i want to test this out |
| 10:30.18 | MuppetMaster | booyeah: 65896 |
| 10:30.26 | *** join/#asterisk Shoragan (~shoragan@d072.apm.etc.tu-bs.de) |
| 10:32.02 | booyeah11 | hmm |
| 10:32.10 | MuppetMaster | booyeah: I can hear you but you can not hear me. |
| 10:32.12 | booyeah11 | can you here me? |
| 10:32.16 | CoderCR | wow |
| 10:32.20 | CoderCR | it went through |
| 10:32.21 | booyeah11 | nat problem? |
| 10:32.31 | MuppetMaster | booyeah11: Well, my Asterisk system is up and running without NAT and that number works. |
| 10:32.39 | *** join/#asterisk pulu (~chatzilla@65.77.78.3) |
| 10:32.41 | MuppetMaster | booyeah11: Take calls on it all the time. |
| 10:32.42 | booyeah11 | im thinking nat problem on my side |
| 10:33.08 | MuppetMaster | booyeah11: Ah, most likely, you are not getting the inbound RTP stream. |
| 10:33.33 | MuppetMaster | booyeah11: Are you redirecting the RTP streams at the gateway as well as adding the [general] options for NAT in your sip.conf? |
| 10:33.45 | MuppetMaster | Unless of course you are using IAX that it should be much easier. |
| 10:33.46 | *** join/#asterisk mbranca (~matteo@81.208.92.210) |
| 10:33.56 | MuppetMaster | CoderCR: Good, so what is your setup. |
| 10:34.36 | booyeah11 | im using using a sip proxy now |
| 10:34.50 | MuppetMaster | booyeah11: I could hear you, but you not me. |
| 10:35.03 | CoderCR | ok |
| 10:35.10 | CoderCR | no it did not work |
| 10:35.12 | CoderCR | Muppet |
| 10:35.19 | CoderCR | could you try one last time please |
| 10:35.27 | MuppetMaster | CoderCR: Okay |
| 10:35.28 | booyeah11 | ok last time |
| 10:35.32 | booyeah11 | i think this will work |
| 10:35.36 | booyeah11 | got the stun settings |
| 10:35.37 | CoderCR | i hope so |
| 10:35.47 | booyeah11 | arg |
| 10:35.51 | CoderCR | i think it will cost like 2 dolars |
| 10:35.51 | booyeah11 | got the stun settings also |
| 10:35.54 | MuppetMaster | booyeah11: Could hear you, but not the otherway around again. |
| 10:35.59 | booyeah11 | damn |
| 10:36.00 | CoderCR | maybe more since the euro is high these days |
| 10:36.05 | booyeah11 | well i got the stun stuff working |
| 10:36.07 | CoderCR | yeah it worked |
| 10:36.21 | booyeah11 | not sure why its not working |
| 10:36.23 | CoderCR | i gues not! |
| 10:36.24 | CoderCR | lol |
| 10:36.24 | MuppetMaster | CoderCR: Okay, so did it work, or no? Do you want me to retry? |
| 10:36.34 | CoderCR | it crashed Asterisk |
| 10:36.34 | CoderCR | lol |
| 10:36.45 | MuppetMaster | CoderCR: Ooops. I do have SpanDSP working without a problem. |
| 10:36.45 | CoderCR | i am going to put it back to the fax machine and try one last thing |
| 10:36.56 | booyeah11 | pissing me off |
| 10:37.00 | MuppetMaster | Just not very reliable over VoIP/ULAW. |
| 10:37.03 | CoderCR | brb |
| 10:39.58 | booyeah11 | echo test works |
| 10:40.06 | CoderCR | ok |
| 10:40.08 | CoderCR | done |
| 10:40.45 | CoderCR | Muppet what version of SpanDSP are you using? |
| 10:40.56 | booyeah11 | arg |
| 10:40.58 | booyeah11 | shitty |
| 10:41.00 | CoderCR | ok |
| 10:41.04 | CoderCR | system is ready for calls now |
| 10:41.08 | MuppetMaster | booyeah11: Crap connection, appears there are some bandwidth issues in between. |
| 10:41.16 | MuppetMaster | CoderCR: So you would like me to try again? |
| 10:41.17 | CoderCR | sorry i went to change settings on the rhino channel bank for the fax machine |
| 10:41.20 | CoderCR | yes |
| 10:41.21 | CoderCR | last one |
| 10:41.22 | booyeah11 | ill try with asterisk |
| 10:41.23 | CoderCR | this will work |
| 10:41.30 | MuppetMaster | CoderCR: Just a moment, I will check my version. |
| 10:41.32 | CoderCR | Muppet what number can i fax you at? |
| 10:41.59 | MuppetMaster | 0.0.2 |
| 10:42.04 | MuppetMaster | For SpanDSP |
| 10:42.47 | CoderCR | o no wonder |
| 10:42.49 | CoderCR | lol |
| 10:42.52 | CoderCR | i was trying 0.0.1 |
| 10:43.11 | MuppetMaster | CoderCR: Alright, just launched a third test. |
| 10:45.15 | MuppetMaster | CoderCR: Alright, just sent a test image. |
| 10:45.19 | MuppetMaster | So 4th fax. |
| 10:46.10 | booyeah11 | pbx_extension_helper: No application 'IAX2/${FWDNUMBER}:${FWDPASSWORD}@iax2.fwdnet.net/65896' for extension (sip-travis, 65896, 2) |
| 10:46.32 | *** join/#asterisk riksta (~rick@81-178-200-105.dsl.pipex.com) |
| 10:47.54 | booyeah11 | my mistake |
| 10:48.32 | CoderCR | i get line errors on images |
| 10:51.05 | MuppetMaster | CoderCR: Hmmmm.... |
| 10:51.09 | booyeah11 | cool |
| 10:52.15 | MuppetMaster | Sorry, but I have to run. Taking my son to http://www.tunfun.nl. |
| 10:52.20 | MuppetMaster | But will be back later today. |
| 10:52.42 | booyeah11 | cool |
| 10:52.52 | CoderCR | later |
| 10:52.55 | booyeah11 | late |
| 10:53.36 | CoderCR | thank you for the help |
| 10:53.49 | *** part/#asterisk MuppetMaster (~MuppetMas@a82-92-73-185.adsl.xs4all.nl) |
| 10:55.02 | *** part/#asterisk X-Gen (~x-gen@rrba-146-121-118.telkomadsl.co.za) |
| 10:57.37 | *** join/#asterisk TheEmperor (TheEmperor@218.111.51.183) |
| 10:59.00 | *** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au) |
| 11:00.01 | *** join/#asterisk HuangDi (TheEmperor@218.111.51.183) |
| 11:02.32 | *** join/#asterisk kumbang (~unknown@167.205.24.4) |
| 11:06.10 | CoderCR | well |
| 11:07.37 | *** join/#asterisk mbranca (~matteo@81.208.92.210) |
| 11:07.55 | christo | I'm trying to get * to accept an incoming call. I have a single E1 with an 0800 number routed to it, but * currently doesn't detect the incoming calls. I'm sure I need something in my dialplan to pick them up. My zapata.conf specifies 'context=default' for all my channels and I have a line like this in my default context in the extensions.conf: exten => _.,1,NoOp(Incoming call on E1 from ${CALLERID} for ${EXTEN}) |
| 11:08.01 | *** join/#asterisk pranav (pranav@202.149.48.209) |
| 11:08.03 | christo | I'm hoping to see something in debug, but nothing comes |
| 11:09.00 | CoderCR | sure i can help |
| 11:09.15 | CoderCR | do you have a PRI E1 |
| 11:10.35 | christo | CoderCR - me? |
| 11:10.56 | pranav | can anyone tell me how to record calls usig asterisk |
| 11:11.06 | christo | Monitor |
| 11:11.11 | christo | pranav - look up the Monitor command |
| 11:11.37 | christo | pramav: http://www.voip-info.org/wiki-Asterisk+cmd+Monitor |
| 11:11.53 | pranav | with monitor command i can see the calls which i have made |
| 11:12.15 | pranav | i mean i can get the details of the call |
| 11:14.46 | Guybrush|work | i failed, i wasted 3 days on spandsp and i retreat in anger |
| 11:21.18 | *** part/#asterisk pranav (pranav@202.149.48.209) |
| 11:21.21 | *** join/#asterisk pranav (pranav@202.149.48.209) |
| 11:23.35 | *** join/#asterisk pranav (pranav@202.149.48.209) |
| 11:23.48 | *** join/#asterisk puppet (puppet@1-1-3-3b.ox.mlm.bostream.se) |
| 11:26.48 | *** join/#asterisk Jas_Williams (~Jason@host81-155-66-178.range81-155.btcentralplus.com) |
| 11:26.53 | puppet | hi all:) |
| 11:27.38 | puppet | 'anyone experince "laggy" phone calls, or choppy after a while when asterisks have been on a while? |
| 11:28.06 | *** join/#asterisk Mazda-MX5 (~leo@220-130-142-43.HINET-IP.hinet.net) |
| 11:34.32 | CoderCR | hello all |
| 11:36.41 | *** join/#asterisk Mother__ (~m@53.Red-217-126-93.pooles.rima-tde.net) |
| 11:36.45 | Mother__ | greetings |
| 11:44.06 | pulu | puppet: i did but i upgraded and it went away |
| 11:44.24 | *** join/#asterisk CosmicRay (~jgoerzen@2002:4545:7206:1:20e:a6ff:fe5c:55e1) |
| 11:44.25 | puppet | pulu: upgraded to what version? |
| 11:44.36 | pulu | puppet: the 1.0 cvs |
| 11:44.40 | puppet | ok |
| 11:44.56 | puppet | 1.0.7 that is? |
| 11:45.27 | pulu | not sure but when i had the problems i was running a really old version, and it didn't do it with sip, just everything else |
| 11:45.48 | puppet | doing it with sip now |
| 11:46.17 | pulu | i still have problems with iax sometimes but it depends entirely on the client i'm using so i'm sure it's that... the iax between my servers works fine |
| 11:46.38 | puppet | this is pure sip |
| 11:46.48 | puppet | inc sip > same sipline' |
| 11:47.17 | pulu | no ideas.. i asked in here and people said upgrade so i did and it was gone, but that was like 6 months ago |
| 11:51.36 | RoyK | 1.0.7 rc |
| 11:52.27 | RoyK | current stable is 1.0.6 |
| 11:52.48 | *** join/#asterisk nextime (~nextime@danex.i-m-c.it) |
| 11:53.35 | Alexis | How can I difference an exten dial by a specific user ? |
| 11:53.55 | Alexis | in local of course |
| 11:54.17 | CoderCR | does anyone here have a fax machine? |
| 11:54.51 | CoderCR | could some one send a fax to 858 558 1200 for a test |
| 11:57.14 | CoderCR | Roy |
| 11:57.27 | Mazda-MX5 | .. |
| 11:57.29 | CoderCR | would you be able to send a fax to 858 558 1200 for a test please |
| 11:57.44 | pulu | i can send one but it'll end up going over ilbc so i don't think that's a very useful test |
| 11:58.00 | CoderCR | pulu. no worries |
| 11:58.59 | dreamcode | is posible to have a call transfer to another number , not extension ? |
| 11:59.30 | CoderCR | yes |
| 11:59.49 | dreamcode | how..? pls |
| 12:00.08 | CoderCR | just like an extension |
| 12:00.37 | CoderCR | eg Dial(Zap/1/18005551212) |
| 12:00.42 | dreamcode | but.. * reads only one digit |
| 12:01.02 | dreamcode | not like that.. i want to be able to dial what number i want |
| 12:01.08 | CoderCR | you have not set it up right then |
| 12:01.16 | CoderCR | i know |
| 12:01.17 | dreamcode | from the phone .. not from dialplan |
| 12:01.28 | CoderCR | read up on your extensions.conf |
| 12:01.47 | CoderCR | read the handbook |
| 12:02.00 | CoderCR | dreamcode |
| 12:02.05 | CoderCR | do you have a fax machine? |
| 12:02.16 | dreamcode | no..sorry |
| 12:02.22 | dreamcode | :( |
| 12:02.38 | Mother__ | fax? we don't need no stinkin' fax! |
| 12:02.47 | CoderCR | i need to test some things |
| 12:02.51 | CoderCR | so i do need a fax |
| 12:03.05 | CoderCR | i am too poor to get an efax account |
| 12:03.06 | Mother__ | just kidding ;) - I don't have one here right now |
| 12:03.10 | Mother__ | lol |
| 12:05.55 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 12:08.24 | *** part/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 12:09.03 | Mother__ | hi Zeeek |
| 12:10.36 | *** join/#asterisk libpcp (libpcp@210.16.20.5) |
| 12:10.46 | libpcp | hi guys |
| 12:12.20 | CoderCR | hey |
| 12:12.28 | CoderCR | libpcp: do you have a fax machine? |
| 12:13.55 | CoderCR | man |
| 12:14.07 | CoderCR | i cannot belive i cannot find someone with a fax machine |
| 12:18.07 | Mother__ | a lot of people use email for most things nowadays |
| 12:18.17 | Mother__ | I have some clients who no longer have fax machines |
| 12:18.40 | Mother__ | they do the banking over internet, their purchases and accounting via email, etc. |
| 12:18.49 | libpcp | nope i dont have CoderCR |
| 12:18.59 | libpcp | CoderCR: why? |
| 12:19.03 | Mother__ | even my hardware wholesaler now doesn't do anything via fax, not even RMAs |
| 12:19.24 | CoderCR | i need to get a test |
| 12:19.34 | CoderCR | i know |
| 12:19.40 | CoderCR | but people still fax in this day and age |
| 12:19.43 | CoderCR | it is amazig |
| 12:19.47 | CoderCR | amazing |
| 12:20.01 | Mother__ | yes, indeed, it's just that the fax population has reduced quite a bit :) |
| 12:22.54 | *** join/#asterisk feral_kid (~not@209.205.207.130) |
| 12:23.44 | feral_kid | Good morning all... |
| 12:24.07 | feral_kid | Jas_Williams: How goes it this morning? |
| 12:25.33 | CoderCR | good morning |
| 12:25.39 | CoderCR | feral: do you have a fax machine/ |
| 12:26.00 | Mother__ | maybe we should put that in the topic :) |
| 12:26.10 | feral_kid | CoderCR: Good morning to you... And no, I don't have a fax that is readily available... |
| 12:26.38 | jontow | email, postal mail, or "damnit, no!" |
| 12:26.38 | jontow | :) |
| 12:26.44 | *** join/#asterisk sysdef (~sysdef@sysdef.admin.debiancenter) |
| 12:26.48 | feral_kid | I am too busy trying to fight my problems with Asterisk@Home |
| 12:27.32 | jontow | i think the @home stuff is way too overcomplicated |
| 12:27.41 | jontow | but i haven't had an extremely in-depth look at it yet |
| 12:27.50 | Mother__ | anyone here tried CentOS 4 with *? |
| 12:28.01 | Mother__ | since it has 2.6 kernel and other trimmings |
| 12:28.15 | iamx | @home is good because of the preinstalled festival, amp, things but it sucks with capi and isdn support... |
| 12:28.15 | feral_kid | jontow: Not overly difficult, but just problematic at times... |
| 12:28.16 | jontow | with so many layers, there is so many ways to be confused.. just straight * on a clean linux install makes way more sense to me :/ |
| 12:28.37 | jontow | but i do see that people want the extras :) |
| 12:29.03 | feral_kid | jontow: You are right there, but a quick deployment with @home for customer types is nice... |
| 12:29.07 | jontow | (damnit, UPS man.. I want my laptop) :) |
| 12:29.20 | jontow | agreed |
| 12:29.22 | libpcp | i encountered a wierd problem with my existing asterisk server right now especially with registration, when i disconnected the endpoints and try to reconnet again, it failed |
| 12:29.37 | libpcp | before i didnt have this kind of problem |
| 12:30.53 | feral_kid | jontow: For instance, I have been having a dog of a problem just getting to iax2 fwdnet trunks to function properly... |
| 12:31.16 | jontow | aha |
| 12:31.41 | CoderCR | does anyone have a fax machine :S |
| 12:31.42 | jontow | i've been doing a lot of potentially ugly sutff with IAX2 for work |
| 12:31.50 | jontow | codercr; i do at the office.. but i've never used it ;) |
| 12:31.58 | CoderCR | lol |
| 12:32.11 | CoderCR | are you at home then |
| 12:32.12 | jontow | and my grandma does.. and i've never used that one too :) |
| 12:32.19 | CoderCR | lol |
| 12:32.20 | jontow | both of which are a few minutes away |
| 12:32.31 | CoderCR | well i have all day |
| 12:32.38 | CoderCR | untill 9am my time |
| 12:32.44 | CoderCR | 4:32am right now |
| 12:32.53 | libpcp | is there any issues with asterisk version 1.0.1 ? |
| 12:32.56 | jontow | damn |
| 12:33.06 | jontow | 07:35 here :) |
| 12:33.14 | CoderCR | i just want to get this faxing working before i leave |
| 12:33.20 | *** join/#asterisk wdatkinson (~wdatkinso@pcp986542pcs.northw01.in.comcast.net) |
| 12:34.35 | CoderCR | what laptop? |
| 12:34.38 | jontow | (this is the replacement for the one that arrived quickly and on time, with a severely evil BIOS password) |
| 12:34.51 | jontow | Dell Latitude C600, 1ghz/256MB RAM/20GB HDD |
| 12:34.52 | CoderCR | lol |
| 12:34.57 | CoderCR | bad idea |
| 12:35.01 | CoderCR | DELLs suck |
| 12:35.06 | CoderCR | they have so many issues |
| 12:35.06 | jontow | i've had 3 |
| 12:35.06 | *** join/#asterisk MikeJ[Laptop] (~icechat5@65.170.43.34) |
| 12:35.11 | CoderCR | well there you go |
| 12:35.14 | jontow | and they are very compatible with freebsd |
| 12:35.15 | jontow | no |
| 12:35.17 | jontow | im using one now :) |
| 12:35.25 | CoderCR | i am on a toshiba |
| 12:35.30 | jontow | a latitude LM p166/72MB RAM/30GB HDD |
| 12:35.30 | jontow | :D |
| 12:36.06 | jontow | my gf has a p133/64MB/10GB that used to be mine.. until i ordered a new one and then she stole it |
| 12:37.51 | CoderCR | help me figure out a solution for my fax please :) |
| 12:38.02 | CoderCR | wait let me change that |
| 12:38.11 | CoderCR | help me figure out if i got the SpanDSP working |
| 12:38.34 | jontow | heheheh |
| 12:38.45 | CoderCR | i just want go to home and sleep |
| 12:38.49 | CoderCR | and i cannot till this is done |
| 12:38.52 | jontow | seems as though its a very unstable codebase :/ |
| 12:39.05 | jontow | and i have no idea how to debug pthreaded code :( |
| 12:39.10 | CoderCR | i cannot even get an account on efax because some stupid hold that should not be there for 895 |
| 12:39.17 | jontow | cvs HEAD + ICD = crash on agent logout |
| 12:40.07 | MikeJ[Laptop] | jontow, ICD has been around for a while, it is very stable with older code |
| 12:40.27 | jontow | i couldnt' even get it to compile with -r v1-0 |
| 12:40.38 | MikeJ[Laptop] | if you like thefuctionality, I can point you towards somone you can hite to deal get it current |
| 12:40.51 | MikeJ[Laptop] | I have it compiled, I don't recall with what version |
| 12:40.56 | jontow | structures have changed enough to break it between 1.0.x and HEAD |
| 12:41.07 | MikeJ[Laptop] | yes, that is the issue, |
| 12:41.20 | jontow | and i see at least 2 versions of the code |
| 12:41.32 | jontow | the ICD module in cvs.digium.com's head, and the app_icd one from the maintainers |
| 12:41.46 | MikeJ[Laptop] | don't use the one from digiums cvs |
| 12:41.51 | MikeJ[Laptop] | the other one is maintained |
| 12:41.52 | jontow | ICD i couldn't get to go anywhere.. but didn't try hard enough, and app_icd was the one that compiled/linked into HEAD with no problems |
| 12:42.05 | MikeJ[Laptop] | y |
| 12:42.37 | MikeJ[Laptop] | but you had a crash on agent logout? |
| 12:42.39 | MikeJ[Laptop] | hmmm |
| 12:43.32 | jontow | icd_agent_callback() |
| 12:43.32 | MikeJ[Laptop] | y |
| 12:43.32 | jontow | i am not a fan of the behavior of the regular icd_agent() where you have to keep the line open |
| 12:43.32 | *** join/#asterisk negativecreep (~yama@202.147.174.98) |
| 12:43.33 | negativecreep | hi all |
| 12:43.33 | MikeJ[Laptop] | ask anthm when he shows up this morning, he should be able to tell you a code date it is stable with |
| 12:43.33 | jontow | thats annoying to the agent (i worked tech support for too long) :) |
| 12:43.36 | MikeJ[Laptop] | jontow, you can to callback agents in regular acd... |
| 12:43.40 | negativecreep | I am having a problem with my X100P. When I hangup the phone, it wont detect that and the channel remains busy. |
| 12:43.59 | jontow | yeah, im looking for something a little more robust, expandable, and stable |
| 12:44.01 | *** join/#asterisk _m_ (~m@fbta199.fbta.uni-karlsruhe.de) |
| 12:44.06 | MikeJ[Laptop] | hell, you don't even need to use chan_acd, you can do it all inthe dialplan and just use app_queue if you want |
| 12:44.23 | MikeJ[Laptop] | y, talk to anthm... |
| 12:44.35 | jontow | will do :D thank you (every bit of insight helps) |
| 12:44.41 | MikeJ[Laptop] | there are people, at least one major call center, that uses icd daily |
| 12:45.08 | jontow | indeed |
| 12:45.20 | MikeJ[Laptop] | ok |
| 12:45.36 | jontow | my place of employment builds tools for call-centers..and they've employed me to work with asterisk and hopefully make it a complete solution |
| 12:46.04 | negativecreep | MikeJ[Laptop]: can you help me with this problem. Zap channel on X100P wont hangup when call ends. |
| 12:46.26 | MikeJ[Laptop] | no thanks. |
| 12:46.29 | jontow | nice guys they are sometimes.. but i have a feeling its just another place that wants to code minimally on top of *, sell it as a service and never contribute anything back.. bothers me. |
| 12:47.14 | jontow | ah well |
| 12:47.17 | MikeJ[Laptop] | that can get tricky with the gpl |
| 12:47.20 | jontow | i suspect its happening an awful lot |
| 12:47.26 | MikeJ[Laptop] | y |
| 12:47.52 | jontow | and yes it can :) |
| 12:48.03 | negativecreep | Anyone . please help me out? |
| 12:48.29 | jontow | nice nirvana reference, ya got there.. :) |
| 12:48.59 | negativecreep | jontow: yeah they are cool |
| 12:49.10 | negativecreep | jontow: i like em still today. |
| 12:49.27 | MikeJ[Laptop] | nega- are you int he us? |
| 12:49.36 | MikeJ[Laptop] | or elsewhere |
| 12:50.13 | jontow | damn i love the local UPS' station's routes.. |
| 12:50.17 | negativecreep | MikeJ[Laptop]: no |
| 12:50.18 | MikeJ[Laptop] | the problem with pots lines is there is no good way to detect hangup most of the time... |
| 12:50.31 | MikeJ[Laptop] | it can be more of a prob in diff countries |
| 12:50.34 | MikeJ[Laptop] | where are you? |
| 12:50.34 | negativecreep | MikeJ[Laptop]: I am using euroisdn as a standard..basically in Pakistan. |
| 12:50.59 | MikeJ[Laptop] | ummm |
| 12:51.10 | MikeJ[Laptop] | how are you doing isdn into an x100p? |
| 12:51.22 | negativecreep | the standard is euroisdn in zapata.conf |
| 12:51.32 | MikeJ[Laptop] | o |
| 12:51.57 | MikeJ[Laptop] | I honestly have never looked at any of the overseas stuff, cuz us settings are all standard |
| 12:52.00 | negativecreep | the switchtype is set to euroisdn. |
| 12:52.07 | iamx | Can't open /dev/isdninfo or /dev/isdn/isdninfo: No such device <- :=( did anyone have the same problem ? |
| 12:52.09 | negativecreep | right. |
| 12:52.21 | MikeJ[Laptop] | switchtype is an isdn setting, will not do anything for an x100p |
| 12:52.26 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 12:52.51 | MikeJ[Laptop] | check out the wiki or config files for how to config correctly for other standards |
| 12:52.53 | negativecreep | MikeJ[Laptop]: right |
| 12:52.54 | MikeJ[Laptop] | ~docs |
| 12:53.08 | 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 |
| 12:54.01 | jontow | bbiaw.. gonna go outside and have a cigarette, then crouch in the bushes at the end of the driveway waiting for the UPS man ;P |
| 12:59.00 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 13:02.28 | Zeeek | that's a long ciggie break |
| 13:05.28 | *** join/#asterisk coolschool (~coolschoo@server1.pointnet.co.uk) |
| 13:07.30 | puppet | == Spawn extension (outgoing, xxxxxxxxxxx, 2) exited non-zero on 'SIP/1001-3cc4 but context=outgoingblock in sip.conf, how come?:o |
| 13:08.32 | Zeeek | context conflict |
| 13:08.45 | puppet | how cna they conflict? |
| 13:08.48 | Zeeek | typo? |
| 13:08.54 | Zeeek | like the one above :) |
| 13:09.09 | puppet | [outgoingblock] |
| 13:09.13 | puppet | is in the extensions |
| 13:09.31 | jontow | :P |
| 13:09.43 | Zeeek | what is in sip.conf? |
| 13:09.44 | jontow | so it was, i stood outside in the sun |
| 13:09.49 | jontow | gonna be another nice day here |
| 13:09.56 | puppet | zeeek: context=outgoingblock |
| 13:10.16 | Zeeek | and the Dial() command is ? |
| 13:10.36 | puppet | in extensions? |
| 13:10.52 | Zeeek | that is a good place for the dial command |
| 13:11.04 | puppet | the tihng is it goes to wrong dial command |
| 13:11.16 | Zeeek | it's the software's fault? |
| 13:11.27 | puppet | what i see i have configured right? |
| 13:11.35 | Zeeek | not. |
| 13:11.45 | Zeeek | is this @home or something? |
| 13:11.49 | puppet | no |
| 13:11.56 | Zeeek | straight * ? |
| 13:12.09 | puppet | context=outgoingblock should make outcoming calls to go threw [outgoingblock] section in extensions.conf |
| 13:12.12 | puppet | right? |
| 13:12.22 | Zeeek | no |
| 13:12.26 | *** join/#asterisk isam (~isam@213.186.190.122) |
| 13:12.26 | Zeeek | wrong |
| 13:12.35 | puppet | oh |
| 13:12.40 | Zeeek | The dialplan is the heart of an Asterisk system, as it defines how Asterisk should handle calls. |
| 13:12.40 | Zeeek | http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN650 |
| 13:13.33 | Zeeek | the [blaockname] syntax is confusing in that sometimes two [names] don't mean the same thing, as you have just seen |
| 13:13.45 | *** join/#asterisk dubphil (~dubphil@80.124.137.201) |
| 13:14.08 | Zeeek | context= in sip.conf or iax.conf is for INcoming calls |
| 13:14.43 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 13:14.45 | Zeeek | to call out, you need to use the dial application in the extensions.conf dialplan |
| 13:15.37 | puppet | Yeah, it works standard, but i wanne createa friend that cant call some numbers, with ctrict rules what he can call |
| 13:15.48 | *** join/#asterisk Chuji (Chuji@pcp09930052pcs.tulipgrove.tn.nash.comcast.net) |
| 13:16.18 | Zeeek | you do that with contexts - which are explained in the link |
| 13:18.01 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 13:21.44 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 13:23.05 | Zeeek | hey bjohnson |
| 13:23.14 | bjohnson | hi |
| 13:24.14 | iamx | Can't open /dev/isdninfo or /dev/isdn/isdninfo: No such device <- :=( did anyone have the same problem ? |
| 13:25.15 | Jas_Williams | iamx: NO what ISDN stack are you trying to use ? |
| 13:26.22 | iamx | capi, incoming calls work in asterisk, but outgoing don't because * doesen't find the device with msn ... |
| 13:27.03 | Jas_Williams | iamx: what is your dial line ? |
| 13:27.05 | iamx | all modules are properly loaded |
| 13:27.19 | iamx | hmm what do you mean by dial line ? |
| 13:28.20 | Jas_Williams | to call an external number i use exten = _3XXXXX,1,Dial(CAPI/330417:b${EXTEN}) |
| 13:28.20 | Jas_Williams | <PROTECTED> |
| 13:28.29 | coolschool | need help with sip express router |
| 13:28.31 | coolschool | anyone used it? |
| 13:29.24 | iamx | i use the extension 9 to dial out with CAPI/26310274 |
| 13:30.14 | Jas_Williams | iamx: can you copy the line from extensions.conf so I can check for errors also post capi.conf to pastebin.ca |
| 13:30.28 | iamx | ok |
| 13:30.39 | bjohnson | coolschool: some do, most don't. google would be your best bet |
| 13:31.07 | coolschool | damn it i have a nat 2 nat situation i need to solve |
| 13:31.55 | coolschool | asterisk behind a nat |
| 13:32.03 | coolschool | phone behind another nat |
| 13:32.07 | coolschool | net in the middle |
| 13:32.14 | coolschool | anyone done it? |
| 13:34.17 | puppet | Mar 18 14:33:59 NOTICE[23325]: chan_sip.c:7681 handle_request: Registration from 'PUPPETs proxy <sip:avatar@sip.puppet.nu>' failed for '213.114.142.176' |
| 13:34.18 | MikeJ[Laptop] | who's using PRI's? |
| 13:34.20 | puppet | oops |
| 13:34.23 | puppet | wrong chan :) |
| 13:35.06 | bjohnson | coolschool: yes |
| 13:35.09 | Jas_Williams | MikeJ[Laptop]: I don't have one on my sytem but I have configures a few |
| 13:35.16 | iamx | Jas_Williams: http://pastebin.ca/7693 |
| 13:35.21 | bjohnson | coolschool: but not me. I use FWD as a go between |
| 13:35.54 | bjohnson | MikeJ[Laptop]: a lot of people .. but not here right now .. try again in couple of hours when N.A. wakes up |
| 13:36.15 | coolschool | whats FWD how do i use it? |
| 13:36.23 | bjohnson | ~FWD |
| 13:36.36 | jbot | hmm... fwd is Free World Dialup: Brainchild of Jeff Pulver. URL: http://www.pulver.com/fwd/ |
| 13:36.36 | bjohnson | hmm |
| 13:36.36 | Dandan | ~fwd |
| 13:36.38 | jbot | well, fwd is Free World Dialup: Brainchild of Jeff Pulver. URL: http://www.pulver.com/fwd/ |
| 13:36.38 | bjohnson | free world dialup |
| 13:36.41 | *** join/#asterisk Darwin[laptop] (~darwin-la@c-24-3-226-147.client.comcast.net) |
| 13:36.55 | coolschool | sorry, yeah |
| 13:37.19 | Jas_Williams | iamx: OUT_1 = CAPI/26310274 |
| 13:37.20 | Jas_Williams | <PROTECTED> |
| 13:37.30 | *** part/#asterisk coolschool (~coolschoo@server1.pointnet.co.uk) |
| 13:37.43 | iamx | no, that0s ok OUT points to OUT_1 |
| 13:37.46 | bjohnson | I had a couple of phones that I wanted to be able to load to others and not reconfig my system (I wouldn't know the remote subnet addresses). So I signed up for a free account at FWD for my * server and another account for each phone. Then the external phones could call my * through FWD |
| 13:38.01 | bjohnson | loan to others |
| 13:38.04 | bjohnson | not load |
| 13:38.11 | iamx | if you look to the log entry on the bottom you see that it correctly tries to dial out with CAPI/26310274 |
| 13:38.50 | bjohnson | then in my dial plan I do authbyCID so when the phone accounts call in, they get access to internal extensions |
| 13:39.23 | bjohnson | I don't know about production systems, but it works well enough for occasioanl use |
| 13:39.41 | Jer13261 | bjohnson: have you ever beenable to get FWD peering to work? |
| 13:40.19 | bjohnson | don't think I've tried. Is that with the other voip networks? |
| 13:40.30 | Jer13261 | yes |
| 13:41.49 | Jas_Williams | iamx: the log entry shows something else CAPI/26310274:bs the number to call from the macro is s not a vaild extension ? |
| 13:42.24 | iamx | it's a valid extension, i have no idea why it shows "bs" |
| 13:42.58 | *** join/#asterisk miltux (~miltux@ppp18-adsl-129.ath.forthnet.gr) |
| 13:43.21 | Jas_Williams | iamx: ok whynot just create a test dial in the extensions.conf |
| 13:43.49 | iamx | ok |
| 13:44.08 | iamx | with Dial() ? |
| 13:44.49 | Jas_Williams | exten => numberyouwishto call,1,dial(CAPI/b${EXTEN}) |
| 13:45.56 | *** join/#asterisk florz (~florz@2001:1a50:503c:0:0:0:0:1) |
| 13:46.04 | Jas_Williams | iamx: also what are you passing into the dialout macro ? |
| 13:46.10 | *** part/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 13:49.14 | iamx | exten => _${DIAL_OUT_1}.,1,Macro(dialout,1,${EXTEN}) |
| 13:53.08 | *** join/#asterisk kcir (~kcir@ariadne.sanguinary.net) |
| 13:53.23 | iamx | http://pastebin.ca/7694 |
| 13:53.31 | kcir | so my fax machine has been triping the power alarms on my tdm400 |
| 13:53.36 | iamx | same thing with the test extension |
| 13:54.03 | kcir | is there some sort of line condidtioning for phone lines? |
| 13:54.13 | Jas_Williams | iamx: I think the line should read exten => _${DIAL_OUT_1}.,1,Macro(dialout,${EXTEN}) |
| 13:54.50 | iamx | i'll try that |
| 13:57.21 | iamx | it still sets "bs" for the extension, but i don't think that's the real problem because also the test extension doesn't work |
| 13:59.25 | Jas_Williams | iamx: I see your test extensione does not work hmmm |
| 13:59.28 | *** join/#asterisk niZon (~ilt@S0106deadbeef6977.wp.shawcable.net) |
| 14:00.10 | *** join/#asterisk didz_ (didz_@200.218.192.52) |
| 14:01.00 | *** join/#asterisk negativecreep (~yama@202.147.174.98) |
| 14:01.05 | negativecreep | hi all |
| 14:01.06 | Jas_Williams | iamx: your capi.conf looks fine however the msn is not forund, the value looks very large, the msn normaly does not have the area code in it ? |
| 14:01.09 | negativecreep | hey MikeJ[Laptop] |
| 14:01.23 | negativecreep | hi Jas_Williams |
| 14:01.24 | *** join/#asterisk jmacz (~jmacz@63.245.86.116) |
| 14:02.04 | negativecreep | Is it possible to execute some asterisk macro|command during a call when the caller presses some specified digit? |
| 14:02.13 | Jas_Williams | iamx: can you turn on capi debug and call into the system and then post the results so I can see the inbound msn |
| 14:02.47 | iamx | mmh no, the msn is ok, in luxembourg all msn's start with 26 and 4 numbers |
| 14:02.57 | iamx | ok |
| 14:03.40 | iamx | err 6 numbers |
| 14:04.11 | *** join/#asterisk kensuke (~bryan@rrba-146-82-252.telkomadsl.co.za) |
| 14:04.22 | kensuke | greetings !! |
| 14:04.27 | negativecreep | hi kensuke |
| 14:04.30 | *** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net) |
| 14:04.37 | kensuke | Ok, ive wikkied to my wits end ! :-) |
| 14:04.53 | kensuke | please tell me ... can asterisk route outbound calls via a sip soft phone ? |
| 14:05.08 | kensuke | hmmm ... maybe Im vague |
| 14:05.29 | kensuke | I mean ... from kphone ... can I call out via my asterisk server ? can it be done ? |
| 14:05.43 | kensuke | to a pstn network |
| 14:06.05 | negativecreep | kensuke yes |
| 14:06.26 | kensuke | sweet ! ... um ... how |
| 14:06.28 | kensuke | :-) |
| 14:06.33 | negativecreep | you need to plug in a pstn line to your asterisk box |
| 14:06.48 | kensuke | assume I can call other sip phones on my network ... |
| 14:06.53 | negativecreep | set up an extension to dial the Zap channel with the string you specify. |
| 14:06.59 | kensuke | and incomming calls make the soft phones ring |
| 14:07.02 | dreamcode | are there any posibilities to create a call conference without using ztdummy ? |
| 14:07.12 | *** join/#asterisk fugitivo (~ajf@201.255.100.126) |
| 14:07.36 | negativecreep | kensuke: exten => _77.,1,Dial(Zap/1/${EXTEN:2}) |
| 14:07.47 | negativecreep | use something like this in your sip client's context |
| 14:07.53 | Jer13261 | dreamcode:no |
| 14:08.06 | Jer13261 | whats wrong with ztdummy |
| 14:08.12 | negativecreep | when you press 77, it shall give you the tone and then dial the number you wish to dial..sip -> pstn call |
| 14:08.15 | Jer13261 | well yes hardware card |
| 14:08.27 | *** join/#asterisk iguy (~iguy@dsl093-197-234.mke1.dsl.speakeasy.net) |
| 14:08.31 | kensuke | ahh ... |
| 14:08.33 | kensuke | interesting ... |
| 14:08.49 | dreamcode | i don't have any hardware card, and i have some problems when compiling ZAp |
| 14:08.50 | *** join/#asterisk CosmicRay (~jgoerzen@2002:4463:7269:1:20e:a6ff:fe66:c5a3) |
| 14:08.59 | kensuke | :-) |
| 14:09.12 | kensuke | thanx negativecreep ... |
| 14:09.13 | Jer13261 | would you like a hand compiling Zap? |
| 14:09.36 | negativecreep | you are welcome kensuke |
| 14:09.40 | dreamcode | if you can.. :) |
| 14:09.44 | Jer13261 | sure just PM me |
| 14:09.54 | kensuke | Im gonna go vi some stuff ... shot alot ! |
| 14:10.03 | negativecreep | good luck kensuke |
| 14:10.35 | *** join/#asterisk Nix (~Nix@dsl81-214-9283.adsl.ttnet.net.tr) |
| 14:10.39 | negativecreep | Jer13261: is it possible to execute a macro|command during a call? |
| 14:10.43 | iamx | Jas_Williams: http://pastebin.ca/7695 |
| 14:11.09 | Jer13261 | i know dial lets you do that after the call is complete....during dont know |
| 14:11.30 | kensuke | negativecreep: do I call just 77 ... or 77<number to dial> ? |
| 14:12.45 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 14:12.45 | *** mode/#asterisk [+o denon] by ChanServ |
| 14:13.45 | negativecreep | kensuke: when you press 77, you shall hear the tone. |
| 14:14.20 | negativecreep | kensuke: asterisk will create a connection to the Zap Channel for you..then you will just dial the number like a normal phone call |
| 14:14.33 | kensuke | cool |
| 14:14.47 | negativecreep | Jer13261: the problem is that on a X100P, asterisk is not detecting call hangup.. |
| 14:14.56 | negativecreep | Zap channel stays busy. |
| 14:15.25 | negativecreep | I would like that the user can execute softhangup or hangup when he is done talking by pressing some specific keys. |
| 14:16.17 | Jer13261 | negativecreep how about callprogress? |
| 14:17.34 | kensuke | negativecreep: is there a way that I can dial 0<number> striaght into kphone ? without dialing the extra stuff ? |
| 14:19.03 | negativecreep | Jer13261: callprogress? |
| 14:19.08 | negativecreep | kensuke: explain! |
| 14:19.45 | Jer13261 | http://www.voip-info.org/tiki-index.php?page=Asterisk+Disconnect+Supervision |
| 14:20.12 | negativecreep | Jer13261: nah not that |
| 14:20.22 | kensuke | I want to dial the number 0833091643 ... thats the number ... at the moment I connect via sip by dialing 0 ... then enter DTMF 833091643 |
| 14:20.43 | negativecreep | ok |
| 14:20.44 | jmacz | Hi everyone. I have a PBX with an Initial voice menu and have a problem with contexts inclusion for DID. |
| 14:21.16 | Jas_Williams | iamx: Hmm MSN is correct then bah I wonder why the msn is not being forund for outbound calls |
| 14:22.31 | negativecreep | kensuke: so what do you want to do? |
| 14:23.12 | jmacz | I got an Incoming-calls context that processes 3-digit ids from two E1s and need to config Direct Inward Dialing in some of them. Anyone can help me? |
| 14:26.29 | *** join/#asterisk ckruetze (ckruetze@cpc1-cmbg7-5-0-cust31.cmbg.cable.ntl.com) |
| 14:28.39 | kensuke | I want to type in 0833091654 and I want it to dial the number ... without me having to dmf its ass :-) |
| 14:29.41 | Jas_Williams | kensuke: That should not be a problem post your extension.conf to pastebin.ca and I'm sure we can advise |
| 14:30.27 | negativecreep | kensuke: its fairly easy. |
| 14:30.59 | negativecreep | Jas_Williams: can you help me with an issue? Executing commands DURING a call? |
| 14:31.25 | *** part/#asterisk dubphil (~dubphil@80.124.137.201) |
| 14:31.38 | kensuke | negativecreep... ok ? |
| 14:31.42 | kensuke | enlighten me ... |
| 14:31.45 | kensuke | if you will ? ;-) |
| 14:32.24 | iamx | Jas_Williams i don't know, i'm searching sice 2 weeks for that problem, recompiles chan_capi etc multiple times, i had many problems witch have been solved but this one kills me lol |
| 14:32.29 | *** join/#asterisk mogorman (~mogorman@dhcp-162.digium.com) |
| 14:32.55 | Jas_Williams | negativecreep: I do not belive it is possible, to collect DTMF during talking what do you wnat to allow the caller to do ? |
| 14:33.10 | negativecreep | exten => _08XXXXXX, 1, Dial(Zap/1/0833091654) |
| 14:33.48 | negativecreep | Jas_Williams: I would like the caller to press say 09 during a call to disconnect the call...now that sounds stupid but I donot have remote disconnect supervision. |
| 14:33.55 | negativecreep | so i need a solution for that. |
| 14:34.03 | Jer13261 | there is a feature for that |
| 14:34.10 | negativecreep | Jer13261: like? |
| 14:34.12 | Jas_Williams | Ah there is a feature you could use |
| 14:34.21 | negativecreep | :) |
| 14:34.37 | Jer13261 | check features.conf i thikn its called disconnect :) |
| 14:34.54 | Jas_Williams | yes follow Jer13261 advice |
| 14:35.03 | Jer13261 | then modidy your dial() command...and your set |
| 14:35.05 | *** join/#asterisk cbachman (~chatzilla@129.105.7.250) |
| 14:35.12 | *** join/#asterisk Wonka (produziert@wonka.support.madwifi) |
| 14:35.12 | negativecreep | Jas_Williams: is there anything on the wiki related to it? |
| 14:35.22 | Jer13261 | 1 sec i'll grab it for ya |
| 14:35.24 | Wonka | mornin... |
| 14:35.34 | Alexis | How can i run a cmd just when i take my phone ? |
| 14:35.39 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 14:35.44 | kensuke | negative creep ... but what If I want all outgoing calls that look like 08XXXXXXXX to dial the relative number ? |
| 14:35.47 | Nugget | "take"? |
| 14:35.57 | Wonka | which libpri version is the best to take for cvs? |
| 14:36.14 | Alexis | Nugget: décrocher in french |
| 14:36.22 | Jer13261 | http://www.voip-info.org/wiki-Asterisk+cmd+Dial |
| 14:36.29 | Zeeek | décrocher |
| 14:36.31 | negativecreep | kensuke: exten => _08XXXXXX, 1, Dial(Zap/1/${EXTEN:2}) |
| 14:36.32 | Alexis | Nugget: I think is the contrary of hangup |
| 14:36.35 | Jas_Williams | kensuke: exten => _08XXXXXX, 1, Dial(Zap/1/${EXTEN}) |
| 14:36.54 | Zeeek | lift the receiver off the hook |
| 14:37.16 | Nugget | "pick up" the phone in "american" |
| 14:37.19 | Jas_Williams | negativecreep: No :2 required as 08 is the full nuber |
| 14:37.22 | Zeeek | raccrocher hangup |
| 14:37.26 | Jer13261 | lol |
| 14:37.34 | negativecreep | oh yeah |
| 14:37.38 | negativecreep | thnx Jas_Williams |
| 14:37.49 | Alexis | ok |
| 14:38.14 | jmacz | Hi. I would like to know wich is the priority regarding context inclusion when I have a condition like _XXX in the container context, and a number in the contained. Wich one has more priority? |
| 14:38.15 | Zeeek | only recently |
| 14:38.19 | Alexis | so i want to Playback just when i pick up my phone |
| 14:38.47 | Jas_Williams | _XXX normaly do a show dialplan and you will see the evaluation order |
| 14:38.55 | Zeeek | jmacz this comes from the evaluation of the included contexts in order AFAIK |
| 14:39.17 | Zeeek | there is a wiki page but I never remember the name or keyword |
| 14:40.08 | jmacz | Jas_Williams, yeah, but I need to jump the dial plan in order to make Direct Inward Dialing |
| 14:40.50 | Alexis | if i put exten => s,1,Playback(transfer) it does not playback when i pick up my phone |
| 14:40.53 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com) |
| 14:40.53 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:40.55 | jmacz | Zeeek excuse me, what is the AFAIK order? |
| 14:41.03 | Zeeek | AFIK as far as I know |
| 14:41.09 | CosmicRay | jmacz: AFAIK means as fas as I know |
| 14:41.09 | Zeeek | find the wiki page |
| 14:41.31 | Zeeek | what would that be dialplan evaluation order? |
| 14:41.41 | jmacz | CosmicRay, oh ok, thanks |
| 14:41.56 | Jas_Williams | jmacz: If you put _XXX in its own context say [catch_all] then include catch_all the extensions will be evaluated first and the the catch all |
| 14:42.10 | Alexis | :( |
| 14:42.53 | *** join/#asterisk MattH (~matth@noc-wireless.chilitech.net) |
| 14:42.56 | Jas_Williams | Alexis: have an immediate=yes in zapata.conf for the phone |
| 14:43.01 | Zeeek | <PROTECTED> |
| 14:43.08 | MattH | Hi, does anyone know in sipura devices.. is there a way to set the packetization? |
| 14:43.22 | Zeeek | Need something catchier than extsnion matching, but once you remember it, it works |
| 14:43.53 | jmacz | Jas_Williams, mmm ok, I guess that's the problem. Thank you. I'l try it out inverting the inclusion |
| 14:44.37 | jmacz | Zeeek, thanks for the link, I'm checking it |
| 14:44.45 | Zeeek | It's the gospel :) |
| 14:44.47 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-201-225.dsl.scarlet.be) |
| 14:44.54 | Zeeek | (I hope it's accurate) |
| 14:45.01 | Zeeek | the gopel prolly isn't |
| 14:45.19 | *** join/#asterisk zno (~chatzilla@ip-160-79-174-101.autorev.intellispace.net) |
| 14:46.52 | *** join/#asterisk Weezey (Weezey@lan6.LO.iasl.com) |
| 14:51.33 | Essobi | What IAX softphones support URL pushing? |
| 14:51.41 | Essobi | It's IAX the only one that does thus far? |
| 14:51.48 | Essobi | s/it's/isn't/ |
| 14:54.49 | Wonka | may it be that www.voip-info.org is a little slow at the moment? |
| 14:55.06 | *** join/#asterisk coppice (~chatzilla@111.196.17.210.dyn.pacific.net.hk) |
| 14:56.13 | ard | Hehh... what do you call slow... It's like a gprs line :-( |
| 14:56.40 | *** join/#asterisk rephorm (~rephorm@ip67-95-13-60.z13-95-67.customer.algx.net) |
| 14:56.53 | Wonka | "slow" as in "nothing gets through" |
| 14:57.16 | Wonka | i watched my line with tethereal... and saw resend on resend |
| 14:57.21 | Wonka | but seldom a reply |
| 14:57.29 | *** join/#asterisk bile_one (~bile_one@pcp03281999pcs.gillst01.ar.comcast.net) |
| 14:58.02 | ard | Cannot open the HTTP connection to www.voip-info.org port 80; [Connection timed out]. |
| 14:58.08 | ard | aaaghh... |
| 14:58.18 | bile_one | ManxPower, looks like I need help. |
| 14:58.20 | Wonka | 19: commp-2.border17.lax.pnap.net (216.52.253.50) asymm 17 416.927ms |
| 14:58.21 | Wonka | 20: las-66-151-54-101.commpartners.us (66.151.54.101) asymm 18 518.134ms reached |
| 14:58.33 | ard | need... voip-info.org ... need.. to... config... asterisk... urgent... aaagh :-( |
| 14:58.39 | Wonka | quite much... |
| 14:59.04 | Jas_Williams | bile_one: what sort of help do you need |
| 14:59.13 | Jas_Williams | ard use google cache |
| 14:59.19 | ard | :-) |
| 14:59.23 | bjohnson | ard: and show application |
| 14:59.29 | bjohnson | ard: and the tutorials |
| 14:59.31 | bjohnson | ~doca |
| 14:59.33 | bjohnson | ~docs |
| 14:59.35 | 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 |
| 15:00.11 | *** join/#asterisk Logan (~logan@planetmath.cc.vt.edu) |
| 15:00.35 | Logan | My wcfxs module stopped detecting inbound calls after about 25 days. |
| 15:00.50 | Logan | Unloading and reloading the module seemed to fix the problem. |
| 15:00.50 | *** join/#asterisk lespiggot (~les@217.206.141.131) |
| 15:04.03 | *** join/#asterisk Alexis (~alexis@www.trim.it) |
| 15:04.28 | Weezey | how do I compile an asterisk application? |
| 15:05.02 | lespiggot | hi, I'm runing asterisk on openwrt (Linksys WRT54) but its having difficulty in identifing its IP address for SIP, Can I use the bind statement (in sip.conf) for explicitly stating its internal and external IP addresses? |
| 15:06.12 | Logan | Hmm, I found a bug report on bugs.digium.com, where the bug was both acknowledged and set as "not a bug" at the same time. |
| 15:06.32 | Logan | Which is really confusing because it doesn't give me any clue whether this bug is actually fixed in any particular version. |
| 15:06.54 | *** join/#asterisk epoch (epoch@octane.breakbeats.org) |
| 15:08.12 | *** join/#asterisk weezey (Weezey@lan6.LO.iasl.com) |
| 15:08.47 | Jas_Williams | Logan: there have been some changes in the latest zaptel drivers, for this type of problem |
| 15:09.22 | Logan | Jas_Williams: Is there a particular version that would be good to be running? |
| 15:09.51 | *** join/#asterisk unixgeek (~unixgeek@12.45.238.189) |
| 15:10.02 | *** join/#asterisk TheBear (~brif8@lazyjtrainingcenter.com) |
| 15:10.35 | *** join/#asterisk _wintermute_ (~wintermut@56.speede.mycybernet.net) |
| 15:11.18 | TheBear | hi All. I'm having problems with my * config, http://pastebin.ca/7697 holds my incoming extensions definition. When a call comes in it rings, during the wait(10), but then goes to musiconhold, when it diverts depending on the time of day. How do I have musiconhold just when I transfer a call not during calling in ? |
| 15:12.06 | CosmicRay | argh, voip-info seems to be having troubles again |
| 15:12.19 | bjohnson | lespiggot: out of interest, the G or the GS model? and which version? |
| 15:12.19 | TheBear | I did have musiconhold = default in zapata.conf but took this out and yet it still has musiconhold when a call comes in |
| 15:12.42 | bjohnson | lespiggot: yes you can use the bind argument |
| 15:12.49 | eKo1 | Hmm...these fucking ghost channels are eating up my memory. |
| 15:12.53 | lespiggot | bjohnson: Its a GS 1.1 |
| 15:13.03 | eKo1 | Time for some toilet cleaning. |
| 15:13.06 | bjohnson | experimental openwrt? |
| 15:13.38 | bjohnson | I've tried using * on my Gv2.2 but not enough RAM when using experimental |
| 15:13.52 | lespiggot | bind was set originally to bind= 0.0.0.0 and get the issue, yes, experimental |
| 15:15.24 | bile_one | TheBear you have your gotoiftime wrong |
| 15:15.42 | TheBear | bile_one: what? |
| 15:15.51 | lespiggot | bjohnson: if I explicitly set bind=external IP the interface will start, but not on the internal iface |
| 15:15.57 | cbachman | lespiggot, I'm running openwrt and on mine bind=0.0.0.0, of course that isn't helping me because it seg-faults attempting to bridge the audio |
| 15:16.10 | bile_one | Your gotoiftime is not in the correct sysntax |
| 15:16.46 | lespiggot | cbachman: still 1 step in front of me lol |
| 15:16.58 | TheBear | bile_one: how should it look, I used one that I found online |
| 15:17.18 | cbachman | notes that that core dump is with openwrt non-experimental (latest cvs) asterisk stable (latest cvs from yesterday afternoon) and SIP |
| 15:18.43 | TheBear | bile_one: if it's between 9am-5pm then it must ring the SIP phones (line 7) otherwise ring the ZAP phones (line 4). It works fine, but it could be my problem on music on hold, so I would like to get it right |
| 15:19.30 | bile_one | TheBear you have to send the call somewhere so it should be: exten => s,3,GotoifTime(9:00-17:00|sun-sat|*|*? somecontext,s,4) |
| 15:19.39 | greg_work | bjohnson: you around? |
| 15:20.19 | bile_one | TheBear or exten => s,3,GotoifTime(9:00-17:00|sun-sat|*|*? somecontext,7:4) |
| 15:20.38 | lespiggot | cbachman: the Gv2.X and GS1.X have different ethernet drivers hence I need to use experimental :o( |
| 15:21.28 | cbachman | lespiggot, ah. Mine's actually a motorola wr850g, which is very similar to a Gv1 |
| 15:21.30 | bile_one | TheBear send your MOH.conf to pastebin |
| 15:21.31 | TheBear | bile_one: what about the :4 ?daytime,1:nightime,1) would that work if I had a [daytime] and a [nightime] context |
| 15:21.46 | *** join/#asterisk sivana (~sivana@165.154.13.35) |
| 15:21.56 | bile_one | TheBear yes extactly. I will send you one in pastebin |
| 15:22.23 | TheBear | bile_one: http://pastebin.ca/7698 |
| 15:23.24 | bile_one | TheBear here is an IVR I did http://pastebin.ca/7699 for you |
| 15:24.04 | *** join/#asterisk Zgarbi (~my@212.58.125.68) |
| 15:24.12 | bile_one | It checks for Holidays too! |
| 15:24.54 | Zgarbi | hi. is somebody here who can help me with compile astersik? while compile I receive /usr/bin/ld: cannot find -lssl |
| 15:25.15 | Zgarbi | what it can be? |
| 15:25.37 | bile_one | TheBear, that one handles SIP too. |
| 15:25.56 | bile_one | TheBear do you have all ther mpg123 installed? |
| 15:27.16 | TheBear | bile_one: yes it plays the music on hold fine. It just rings, and then goes to music while the SIP or ZAP is ringing, instead of ringing until a phone is answered |
| 15:28.47 | bile_one | I see, get rid of the wait(10) |
| 15:29.01 | Zeeek | TheBear what doesz the 'm' stand for in the Dial app again? |
| 15:29.13 | Zeeek | show applications dial |
| 15:29.29 | TheBear | Zeeek: ooooHH yeah, music oops thanks |
| 15:29.40 | TheBear | bile_one: http://pastebin.ca/7701 is this better? |
| 15:30.22 | *** part/#asterisk Nix (~Nix@dsl81-214-9283.adsl.ttnet.net.tr) |
| 15:31.08 | bile_one | TheBear, yes very clean |
| 15:31.27 | TheBear | ok thanks |
| 15:31.36 | bile_one | TheBear, does it work on SIP only? |
| 15:32.10 | TheBear | it worked on both fine. during the day my sip phones rang, during the night my ZAP phone rang. the process worked, just the music was a problem to me |
| 15:32.12 | TheBear | thanks |
| 15:32.15 | *** join/#asterisk negativecreep (~yama@202.147.174.98) |
| 15:32.38 | negativecreep | hey Jer13261 |
| 15:33.36 | bile_one | TheBear, why do you want the caller to be able to transfer the call them selves? |
| 15:34.42 | bile_one | TheBear, the capital T means that. |
| 15:35.06 | TheBear | I do have another funny issue though. I'm still getting a delay or reverb on my ZAP phone? I answer calls on a X100p ZAP/1 and have a std. phone connected to a TDM10B FXS module. The std. phone only accepts 2-wire. http://pastebin.ca/7703 holds my zapata.conf where I have tried to cancel the echo but had minimal sucess |
| 15:35.17 | TheBear | bile_one: not sure, I'll take it out |
| 15:35.53 | bjohnson | greg_work: off and on |
| 15:36.34 | Zeeek | note that if you are the caller, and think you want to transfer with # and use T, wait til you need to hit the # key in an IVR :) |
| 15:36.34 | bile_one | TheBear, did you catch my question on mpg123? Is it installed for Music-On-Hold? |
| 15:37.10 | bile_one | # means hang-up |
| 15:37.50 | TheBear | bile_one: yes I have mpg123 the latest built from source installed. Yes it is installed. and works. I can hear music. |
| 15:37.52 | Zeeek | or park... pointis, if you are caller and have # enabled, if you need for someone else's menu you're in a bad way |
| 15:38.05 | TheBear | bile_one: it was just at the wrong time in the call process |
| 15:38.15 | *** join/#asterisk lespiggot (~les@217.206.141.131) |
| 15:38.18 | *** join/#asterisk malcolmd (~malcolmd@malcolmd.digium.sponsor.pdpc) |
| 15:38.25 | *** join/#asterisk coolschool (~coolschoo@server1.pointnet.co.uk) |
| 15:38.33 | *** part/#asterisk coolschool (~coolschoo@server1.pointnet.co.uk) |
| 15:39.27 | negativecreep | Jer13261: u there? |
| 15:40.25 | bile_one | TheBear, is it working now? |
| 15:41.06 | Jer13261 | yep i am |
| 15:41.25 | TheBear | bile_one: was about to try.... |
| 15:41.48 | negativecreep | Jer13261: X100P is an FXO? |
| 15:42.52 | epoch | yes it is |
| 15:43.39 | negativecreep | I need to connect approximately 8 extensions to my * server |
| 15:43.44 | negativecreep | 8 analog extensions? |
| 15:43.49 | TheBear | bile_one: was about to try.... YEs it works now You hear it ring, then a very quick pause then rings again, I guess it poauses when it changes over |
| 15:43.51 | negativecreep | X100P supports 1 |
| 15:43.52 | TheBear | thank you very much |
| 15:44.09 | epoch | negativecreep: you need FXSes for that |
| 15:44.16 | Alexis | GGRRR ! |
| 15:44.19 | negativecreep | epoch will you explain? |
| 15:44.34 | epoch | negativecreep: by "analog extensions" you mean phones, right? |
| 15:44.37 | Alexis | I can't use the monitoring extension states |
| 15:44.49 | negativecreep | epoch right! |
| 15:44.49 | Alexis | with my snom |
| 15:44.49 | TheBear | bile_one: any ideas on my echo/delay problem ? |
| 15:44.54 | Alexis | can someone help me please |
| 15:45.06 | epoch | negativecreep: an FXS provides dial-tone, so you need FXSes to hook those phones up to, not FXOs |
| 15:45.07 | *** join/#asterisk outsidefactor (barf@203-173-32-225.dyn.iinet.net.au) |
| 15:45.11 | Alexis | i do not understand the help from |
| 15:45.12 | Alexis | http://voip-info.org/tiki-index.php?page=Asterisk%20phone%20snom |
| 15:45.29 | epoch | negativecreep: you'll need a pair of Digium TDM400P cards with 4 FXS modules each |
| 15:45.34 | Alexis | when they put |
| 15:45.35 | Alexis | <PROTECTED> |
| 15:45.44 | negativecreep | epoch the phones are already connected to a panasonic pbx |
| 15:45.45 | Alexis | wich one is the snom ? |
| 15:45.50 | Alexis | the 2000 |
| 15:45.54 | *** join/#asterisk outsidefactor (barf@203-173-32-225.dyn.iinet.net.au) |
| 15:46.00 | Alexis | or Phone/phone0 ? |
| 15:46.11 | epoch | negativecreep: do the panasonic has 8 outgoing lines? |
| 15:46.23 | negativecreep | I would like to take 8 connections from the pbx and hook em upto the * server. That way i will have 8 lines to my asterisk server which can be used for inbound/outbound calling to other * servers |
| 15:46.27 | yashax | Help!! Guys, I have 3 Polycom IP500 phones. Is it possible on my phone to see if the other 2 parties are on the phone? |
| 15:46.34 | negativecreep | epoch its an interoffice exchange |
| 15:46.51 | negativecreep | no pstn lines involved. |
| 15:47.07 | Alexis | Is someone seeing that ? |
| 15:47.11 | epoch | negativecreep: ok well, same difference -- you'll need 8 FXS ports |
| 15:47.22 | Alexis | Can someone read THAT ? |
| 15:47.27 | Alexis | :( |
| 15:47.38 | negativecreep | epoch didnt get your point! |
| 15:47.45 | *** join/#asterisk fugitivo (~ajf@201.255.100.126) |
| 15:47.52 | *** join/#asterisk JerJer[mobile] (~jj@feth100-fw.fament.net) |
| 15:47.55 | negativecreep | epoch let me explain |
| 15:47.57 | epoch | negativecreep: look up the difference between FXO and FXS ports on voip-info.org |
| 15:47.59 | eKo1 | OK. Looks like * is leaking memory. |
| 15:48.03 | JerJer[mobile] | mooo |
| 15:48.06 | negativecreep | epoch ok |
| 15:48.08 | epoch | negativecreep: no, I know what you're trying to do ;) |
| 15:48.34 | Wonka | aaaaaaaaaaaaaargh. since when does ast_channel_register take only one argument? and why does chan_misdn not know about that? |
| 15:48.40 | epoch | negativecreep: btw, I wouldn't recommend that setup either, you might find you have issues with audio quality |
| 15:48.40 | unixgeek | Anyone know of a SIP client that supports multiple SIP proxies for Mac OSX that does not keep crashing like X-lite? |
| 15:48.41 | *** part/#asterisk Alexis (~alexis@www.trim.it) |
| 15:48.59 | TheBear | I do have another funny issue though. I'm still getting a delay or reverb on my ZAP phone? I answer calls on a X100p ZAP/1 and have a std. phone connected to a TDM10B FXS module. The std. phone only accepts 2-wire. http://pastebin.ca/7703 holds my zapata.conf where I have tried to cancel the echo but had minimal sucess |
| 15:49.11 | fugitivo | anyone knows if there is support for a fxo clone on openbsd? |
| 15:49.27 | Nugget | unixgeek: I run a local asterisk install on my powerbook, to do all the heavy lifting, then connect to it with x-lite. |
| 15:49.39 | Nugget | seems pretty non-crashy and lets me use iax2 instead of sip |
| 15:49.44 | epoch | haha |
| 15:49.50 | epoch | htat's pretty heavy duty nugget |
| 15:49.54 | ManxPower | ~docs |
| 15:49.55 | 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 |
| 15:49.57 | ManxPower | ~mailinglist |
| 15:49.58 | 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/ |
| 15:50.02 | Nugget | nah, not really |
| 15:50.16 | JerJer[mobile] | http://tinyurl.com/4pq6u |
| 15:50.18 | epoch | Nugget: x-lite works just fine on my ibook |
| 15:50.20 | Zeeek | hows your polycomm today ManxPower ? |
| 15:50.31 | negativecreep | epoch: we have two * servers in two different cities and we have an existing analog pbx setup. What i have done is that I hve plugged in 1 X100P in each server and the servers are connected via broadband. So at any time 1 person can talk to the other city. I would like to expand it to 8 lines |
| 15:50.33 | *** join/#asterisk jsolares (~jsolares@200.30.141.85) |
| 15:50.37 | Nugget | epoch: yeah, I haven't had much trouble with x-lite either. but I have had problems getting it to work through hotel nat and the like. |
| 15:50.49 | Nugget | being able to use iax2 instead can often mean the difference between connecting or not |
| 15:51.26 | epoch | oh, well NAT is a whore |
| 15:51.29 | Nugget | yup. |
| 15:51.31 | Nugget | nat blows goats |
| 15:51.35 | yashax | Help!! I have 3 Polycom IP500 phones. Is it possible on my phone to see if the other 2 parties are on the phone? |
| 15:51.54 | epoch | yashax: sure, call them and see if you get a busy signal :) |
| 15:52.22 | epoch | yashax: the phones themselves support a "buddywatch" extension, which isn't supported by asterisk |
| 15:52.22 | yashax | great advise... but please be serious... |
| 15:52.53 | ManxPower | ~astcvs |
| 15:52.54 | jbot | i guess astcvs is echo "CVS-HEAD:"; cvs co asterisk asterisk-sounds libpri zaptel; echo "CVS 1.0.x:"; cvs co -r v1-0 asterisk asterisk-sounds libpri zaptel; echo "Anyone that uses CVS and is not on asterisk-cvs mailing list, is a moron!" |
| 15:52.55 | yashax | can that be done? is it possible? |
| 15:53.21 | yashax | epoch: can that be setup on the phones? |
| 15:53.24 | negativecreep | epoch read the difference. |
| 15:53.30 | epoch | ManxPower: jbot's grammar is horrible! there's a comma splice in there! ;) |
| 15:53.38 | negativecreep | and I think I still need FXO and not FXS |
| 15:53.47 | negativecreep | the pbx is generating the tone already |
| 15:54.21 | epoch | yashax: I think it's part of the SIMPLE extensions, which are supported by SER, but not asterisk... so in theory you could set up SER as a front-end to asterisk for the SIP clients |
| 15:54.25 | epoch | (i.e. the IP500s) |
| 15:54.37 | ManxPower | yashax, I don't think Asterisk currently supports remote extension status information. If you want to investigate the issue, search the asterisk mailing lists for talk about SNOM and the hint priority. |
| 15:54.38 | epoch | negativecreep: man, your setup makes no sense to me :) |
| 15:55.01 | iamx | Jas_Williams it works now, just by unloading and reloading modules 3 or 4 times, don't ask me why, but many many thanks for your help ! |
| 15:55.25 | yashax | thanks guys |
| 15:55.38 | rious | is anybody else having trouble w/ asterisk cvs running, I get an error about ast_monitor_stop |
| 15:56.27 | negativecreep | epoch..i have found the solution |
| 15:56.29 | ManxPower | rious, There were MASSIVE updates to CVS-HEAD, as I'm sure you know, since you are on the asterisk-cvs mailing list. |
| 15:56.33 | negativecreep | epoch thnx |
| 15:56.47 | *** join/#asterisk spackle (~spackle@209.234.83.19) |
| 15:57.31 | epoch | np |
| 15:58.49 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivflhb.dialup.mindspring.com) |
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| 16:00.06 | fugitivo | does the zaptel driver work for openbsd? |
| 16:00.15 | Nugget | no |
| 16:00.24 | Nugget | it barely works for freebsd, for that matter |
| 16:00.38 | Nugget | if you need zaptel it's best to stick with linux, sadly |
| 16:01.12 | Jas_Williams | iamx: Glad you got it working You need to reload after any changes to capi.conf |
| 16:01.31 | *** join/#asterisk jeffik (~jeffik@node-423a160a.mdw.onnet.us.uu.net) |
| 16:01.54 | fugitivo | Nugget: is any plan to make it work for *BSD? |
| 16:02.11 | BrianR___ | hey folks.. |
| 16:02.28 | negativecreep | quit |
| 16:02.51 | BrianR___ | Does anyone know why Asterisk might not be sending the callerid name out a T1 configured as pri_net? |
| 16:03.13 | TheBear | Could anyone suggest some tricks to get rid of echo/delay. I have read the manuals tried different values to echocancel, yes/no/16/32/64..... What helped the most was rxgain and txgain currently set to -2.0 It helped greatly, but it is still there and can be rather sickening after awhile |
| 16:03.21 | Nugget | fugitivo: the freebsd zaptel work is fairly active, I don't know about for open, net, or dragonfly. |
| 16:03.30 | Nugget | but it's still not what I'd consider production-quality |
| 16:03.46 | *** join/#asterisk feral_kid (~not@209.205.207.130) |
| 16:03.48 | Nugget | to be fair, though, that's due in large part to the general flakiness of zaptel, even in linux |
| 16:04.19 | BrianR___ | I managed to get zaptel 1.0.6 to crash my box earlier today after changing zaptel.conf and rerunning ztcfg ;( |
| 16:04.41 | Nugget | my freebsd asterisk server pretty consistently locks up the whole machine whenever I stop asterisk. |
| 16:04.45 | Nugget | (thanks to zaptel) |
| 16:04.55 | fugitivo | Nugget: ok, thanks for the info, i'm going to use it just for testing, so i'll try openbsd first and if it doesn't work, i'll stay with linux |
| 16:05.04 | Nugget | zaptel will not work at all with openbsd. |
| 16:05.13 | Darwin[laptop] | if you are using mpg123 this will happen change over to madplayer |
| 16:05.14 | Nugget | asterisk will work just great |
| 16:05.26 | Darwin[laptop] | on fbsd |
| 16:05.34 | fugitivo | Nugget: huh, ok, no openbsd then :) |
| 16:05.42 | Nugget | even in linux, though zaptel is kinda wonky. I can't reboot my linux machin, I have to powercycle it. |
| 16:05.49 | Darwin[laptop] | zaptel works on fbsd fine |
| 16:05.56 | Nugget | if I warmstart, the zaptel drivers don't see the card |
| 16:06.05 | Nugget | Darwin[laptop]: I disagree |
| 16:06.09 | Wonka | i don't like zaptel |
| 16:06.25 | fugitivo | anyone using *bsd and zaptel here? :) |
| 16:06.26 | Wonka | because of the necessity to patch around everywhere |
| 16:06.31 | jontow | Wonka; does it make you itchy? :) |
| 16:06.32 | Darwin[laptop] | I have zaptel on mine and it workd fine |
| 16:06.35 | Wonka | i like chan_mISDN |
| 16:06.36 | Nugget | fugitivo: I am |
| 16:06.45 | Wonka | but that's some weeks behind at least |
| 16:06.50 | jontow | i am testing netbsd 2.0+ drivers for wcfxo cards |
| 16:06.55 | fugitivo | Darwin[laptop]: with freebsd? |
| 16:06.57 | Darwin[laptop] | I have a tdm with 4 fxs ports |
| 16:07.01 | Darwin[laptop] | yes |
| 16:07.26 | fugitivo | Nugget: and you don't recommend it |
| 16:07.27 | *** join/#asterisk rontecxt44 (~rontecxt4@dsl9-173.rb.comporium.net) |
| 16:07.28 | Darwin[laptop] | it works fine |
| 16:07.32 | Nugget | no, I don't recommend it |
| 16:07.47 | fugitivo | Darwin[laptop]: you recommend it |
| 16:07.48 | fugitivo | ? |
| 16:08.10 | Darwin[laptop] | they dont recomend anything but linux |
| 16:08.20 | Darwin[laptop] | I am fine with it |
| 16:08.30 | Darwin[laptop] | I am happy on fbsd |
| 16:08.41 | fugitivo | I need a third person :) |
| 16:08.57 | fugitivo | i'd like to use it with openbsd, but it doesn't work at all |
| 16:09.05 | TheBear | Could anyone suggest some tricks to get rid of echo/delay. I have read the manuals tried different values to echocancel, yes/no/16/32/64..... What helped the most was rxgain and txgain currently set to -2.0 It helped greatly, but it is still there and can be rather sickening after awhile |
| 16:09.56 | rontecxt44 | has anyone experienced problems with calls randomly dropping after only a brief moment of silence on the line? |
| 16:10.19 | *** join/#asterisk fitzel (~flint@p3EE390BD.dip0.t-ipconnect.de) |
| 16:10.31 | rontecxt44 | I cannot reproduce the problem consistently.... |
| 16:10.49 | *** part/#asterisk CoderCR (~creyna@adsl-67-112-135-29.dsl.sndg02.pacbell.net) |
| 16:10.51 | CosmicRay | TheBear: I believe there is quite a bit of content about echo cancellation on the wiki |
| 16:11.06 | Rival | feti u ever generate pdf's using php? |
| 16:11.12 | fitzel | Hi, is anybody using zetebee together with iaxcomm on windoze? |
| 16:11.13 | Rival | err shit |
| 16:11.52 | Pinhole | Is it possible for an agi to change the extension that gets logged in the CDR? |
| 16:12.20 | eKo1 | Pinhole: Not that I know of. |
| 16:12.35 | fitzel | I try to build up a tunnelconnection, but when I try to set up iaxcomm, it rings only itself. |
| 16:12.39 | *** join/#asterisk rephorm (~rephorm@ip67-95-13-60.z13-95-67.customer.algx.net) |
| 16:13.21 | kcir | i keep getting |
| 16:13.21 | kcir | Ouch, part reset, quickly restoring reality (3) |
| 16:13.21 | kcir | Power alarm on module 4, resetting! |
| 16:13.30 | kcir | that's the fax machine... |
| 16:13.35 | rontecxt44 | my setup is pretty simple and everything has been working fine...sometimes when someone calls from a landline...if I just hold the phone for a sec before handing it to someone...the call just disconnects... |
| 16:13.36 | *** join/#asterisk DannyF (~wizardone@h186n11c1o848.bredband.skanova.com) |
| 16:13.37 | jontow | ick |
| 16:13.39 | *** join/#asterisk logicalonline (~logicalon@border.logicalonline.com) |
| 16:13.41 | kcir | is there some way i can prevent that? |
| 16:14.08 | rontecxt44 | a couple of folks reported hearing squealing on the other end like fax or dsl before getting disconnected |
| 16:14.19 | bannerman | that sounds fun |
| 16:14.27 | rontecxt44 | but we only have one line...no call waiting... |
| 16:14.34 | rontecxt44 | so it can't be fax |
| 16:14.36 | CosmicRay | now that sounds weird |
| 16:15.02 | rontecxt44 | my biggest concern is that....we have one analog line... |
| 16:15.16 | rontecxt44 | i had the telco phisically split the one line into two jacks |
| 16:15.30 | rontecxt44 | dsl is on one....asterisk on the other |
| 16:15.47 | rontecxt44 | so i'm wonder if there is some kind of interference |
| 16:15.52 | rontecxt44 | even though there shouldn't be |
| 16:15.54 | bannerman | rontecxt44: it's your background. You have a picture of Bush on your background. |
| 16:16.04 | bannerman | rontecxt44: that's what I'd do if I was the computer, too. |
| 16:16.15 | CosmicRay | bannerman: oh crap, his asterisk is trying to bring freedom to the telephone network! |
| 16:16.27 | bannerman | CosmicRay: freedom from ugly people! |
| 16:16.35 | *** part/#asterisk Jones (Jack@ool-18bc8151.dyn.optonline.net) |
| 16:16.46 | fitzel | rontex: Maybe a codec problem? |
| 16:16.58 | rontecxt44 | i'm open to any idea... |
| 16:16.59 | CosmicRay | bannerman: ah, we went to war in iraq for less :-) |
| 16:17.09 | rontecxt44 | had an important client call dropeed yesterday |
| 16:17.10 | CosmicRay | rontecxt44: it could be a hardware issue, too |
| 16:17.19 | fitzel | Decoding alaw with ulaw sounds for untechnical people like "faxnoise". |
| 16:17.22 | CosmicRay | rontecxt44: are you using the x100p? |
| 16:17.29 | bannerman | CosmicRay: lol, I shouldn't get into that with you, we'll fight n'stuff qq |
| 16:17.36 | CosmicRay | bannerman: heh |
| 16:17.45 | rontecxt44 | CosmicRay: TDM400 |
| 16:17.53 | rontecxt44 | one fxs and one fxo |
| 16:17.55 | BrianR___ | Does anyone know why Asterisk might not be sending the callerid name out a T1 configured as pri_net with isdn national 2 signalling? |
| 16:18.09 | CosmicRay | bannerman: I can just say that "Bush reminds me of Nixon" and you can feel a healthy sense of pride while I feel a healthy sense of loathing :-) |
| 16:18.23 | bannerman | CosmicRay: ouch, that was below the belt! |
| 16:18.33 | CosmicRay | heh |
| 16:19.09 | rontecxt44 | CosmicRay: there is one more quirk |
| 16:19.10 | *** join/#asterisk quigleymd (~quigleymd@24-53-142-5.chvlva.adelphia.net) |
| 16:19.10 | *** part/#asterisk MikeJ[Laptop] (~icechat5@65.170.43.34) |
| 16:19.14 | *** join/#asterisk ckruetze (ckruetze@cpc1-cmbg7-5-0-cust31.cmbg.cable.ntl.com) |
| 16:19.20 | pigpen | Does anyone know if asterisk will compile and work on 64bit linux? |
| 16:19.23 | rontecxt44 | I have our analog line passing through a fax machine |
| 16:19.33 | CosmicRay | pigpen: I run it on my alpha, no trouble |
| 16:19.37 | rontecxt44 | but it only answers on a distinctive ring |
| 16:19.40 | pigpen | cool. |
| 16:19.51 | CosmicRay | pigpen: what platform are you thinking of? |
| 16:19.51 | rontecxt44 | it allows normal passthrough to asterisk on regular calls |
| 16:19.54 | Darwin[laptop] | dec alphas rock |
| 16:20.05 | Darwin[laptop] | I love my dual 21264 600 |
| 16:20.08 | CosmicRay | pigpen: I haven't tried any zaptel hardware yet, but I've got an x100p on its way from ebay |
| 16:20.13 | CosmicRay | Darwin[laptop]: nice nice |
| 16:20.14 | elriah | Is there a way to pass a fax call in asterisk to a fax service on that same box? |
| 16:20.19 | CosmicRay | Darwin[laptop]: I have a single 21164a 600MHz |
| 16:20.49 | CosmicRay | Darwin[laptop]: I have two complaints about it. 1) it's noisy, and 2) it doesn't have enough pci slots. |
| 16:21.06 | *** join/#asterisk yertle (yertle@ip68-6-98-122.sb.sd.cox.net) |
| 16:21.13 | fitzel | elriah, try faxrc |
| 16:21.18 | Darwin[laptop] | I have 2 1 is a asterisk server the other is a x/kde/app server |
| 16:21.28 | elriah | Thanks. |
| 16:21.32 | Darwin[laptop] | for 55 xterminals |
| 16:21.50 | CosmicRay | nice |
| 16:21.57 | fitzel | elria, spandsp is the package name. |
| 16:21.58 | CosmicRay | I had really bad performance with X on my alpha |
| 16:21.59 | *** join/#asterisk Dandan (dandan@234.88.149.195.in-addr.arpa.virt-ix.net) |
| 16:22.03 | *** part/#asterisk yertle (yertle@ip68-6-98-122.sb.sd.cox.net) |
| 16:22.06 | CosmicRay | but that could be due to lack of an AGP slot |
| 16:22.14 | CosmicRay | didn't run much X remotely from it |
| 16:22.25 | Dandan | re |
| 16:22.36 | Darwin[laptop] | ahh I dont run video localy on that box |
| 16:22.36 | christo | how do I register a new channel type? |
| 16:22.37 | fitzel | I used it on a P3-600 laptop with pcmcia-isdncard and I was able to get some pages through. |
| 16:22.53 | Darwin[laptop] | its just a server |
| 16:23.10 | CosmicRay | Darwin[laptop]: yup, they are great at that. |
| 16:23.14 | CosmicRay | Darwin[laptop]: they just sit there and work. |
| 16:23.15 | Darwin[laptop] | the asterisk has a pci video and its all it neds for he cli |
| 16:23.39 | CosmicRay | Darwin[laptop]: I'm going to go headless once I get my x100p in. completely out of pci slots and I rarely use the console anyway. |
| 16:24.02 | Darwin[laptop] | yeah I ssh in more then anything |
| 16:24.33 | cbachman | whooo!!! found a version of asterisk that actually appears to work with sip on openwrt |
| 16:24.35 | Darwin[laptop] | I just updated both mine so they should be fine for awhile |
| 16:24.58 | *** join/#asterisk bile_one (~bile_one@pcp03281999pcs.gillst01.ar.comcast.net) |
| 16:25.02 | Darwin[laptop] | now have to work on my amd 64 get it built and loaded |
| 16:25.28 | bile_one | TheBear has anyone helped on your dely/echo problem? |
| 16:25.50 | TheBear | bile_one: no |
| 16:25.53 | CosmicRay | Darwin[laptop]: and that is my new favorite platform :-) |
| 16:26.14 | CosmicRay | Darwin[laptop]: I've got two amd64 boxen around here. one of them is still running i386 debian, but the other is running amd64 debian |
| 16:26.20 | CosmicRay | very nice platform, that. |
| 16:26.42 | bile_one | TheBear how much echo? |
| 16:27.04 | fugitivo | CosmicRay: try gentoo :) |
| 16:27.16 | CosmicRay | fugitivo: I do once a year or so |
| 16:27.30 | CosmicRay | fugitivo: gentoo and netbsd I keep an eye on. keep going back to debian though. |
| 16:28.05 | CosmicRay | fugitivo: you may be interested in http://people.debian.org/~jgoerzen/dfs/. a gentoo-like install/rescue cd for debian. |
| 16:28.06 | TheBear | bile_one: currently you can hear the end of the sentence repeat over and over and over again. Like "How are you, you, you , you..." It also sounds like a submarine movie with the odd "ping and sratch noise". |
| 16:28.22 | Darwin[laptop] | I am going fbsd |
| 16:28.28 | Darwin[laptop] | my favorit os |
| 16:28.38 | bile_one | TheBear hang on a sec |
| 16:28.45 | CosmicRay | my main gripe about all non-debian OSs is the lack of a package system as nice as debian's |
| 16:29.06 | CosmicRay | and that is especially true on the BSDs, where the base system is distinct from the packaging system |
| 16:29.10 | mogorman | ports and portage are niec |
| 16:29.11 | Darwin[laptop] | but I am going to be building a debian box soon |
| 16:29.24 | mogorman | er nice |
| 16:30.00 | CosmicRay | mogorman: I am no fan of the ports system. upgrading software on production machines is nasty. I remember one update, libpng or something, that uninstalled php, apache, etc, etc, before spending several hours compiling stuf, then crapping out |
| 16:30.17 | ManxPower | ARRRRGH!!!!!! The opening date of the office were I will be installing a 60 phone (polycom) / 2 T-1 Asterisk system, has been postponed. The opening date is now EXACTLY in the middle of my 6 week trip to Europe. |
| 16:30.21 | Darwin[laptop] | thats why I love freebsd |
| 16:30.24 | mogorman | what |
| 16:30.28 | CosmicRay | Darwin[laptop]: heh |
| 16:30.32 | mogorman | freebsd has never done that to me with ports |
| 16:30.37 | bile_one | TheBear, check this out: http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.html |
| 16:30.41 | Darwin[laptop] | we have portupgrade util that updates all your installed ports for you |
| 16:30.47 | ManxPower | They were supposed to open that office on Jan 1 |
| 16:30.59 | CosmicRay | I will never deny that the *BSDs, especially netbsd, have a much cleaner development model and kernel than linux |
| 16:31.06 | CosmicRay | which has a great deal of appeal to me |
| 16:31.09 | Darwin[laptop] | and cvsup on fbsd is your friend |
| 16:31.11 | CosmicRay | but I just can't get past this |
| 16:31.24 | CosmicRay | Darwin[laptop]: this *was* freebsd |
| 16:31.32 | CosmicRay | I used to have a jail from johncompanies.com |
| 16:31.53 | bile_one | TheBear, also this: http://www.voip-info.org/wiki-Asterisk+echo+cancellation |
| 16:31.53 | Darwin[laptop] | I have never had a problem updating fbsd boxes and having them crash |
| 16:32.06 | CosmicRay | Darwin[laptop]: well, the box didn't crash, the port compile did |
| 16:32.13 | zno | in my experience the ports system on freebsd is much more mature than debian |
| 16:32.17 | CosmicRay | Darwin[laptop]: the kernel, etc. stayed up. there was just some bug in a build that day. |
| 16:32.18 | bjohnson | ManxPower: tell them you'll do it but the price doubles |
| 16:32.31 | Darwin[laptop] | where you using portupgrade ? |
| 16:32.40 | bjohnson | ManxPower: then hire a subcontractor to do it |
| 16:32.40 | CosmicRay | Darwin[laptop]: I can't remember, I'm afraid |
| 16:32.44 | zno | portupgrade is the cleanest way to go |
| 16:32.49 | zno | you can even do just binary upgrades |
| 16:32.55 | Darwin[laptop] | yeah |
| 16:33.01 | Darwin[laptop] | if there are packages |
| 16:33.13 | CosmicRay | I have used portupgrade, but I don't know if I did for this particular case or not. |
| 16:33.16 | mogorman | which there almost always are |
| 16:33.22 | zno | well typically I'd do it for a REL version |
| 16:33.47 | Darwin[laptop] | for rel ver pkgupgrade is the way to go |
| 16:33.58 | Darwin[laptop] | for stable have them build the pkgs |
| 16:33.59 | bile_one | TheBear, since you are using an X100P try this too! http://www.voip-info.org/wiki-Asterisk+x100p+echotraining |
| 16:34.26 | CosmicRay | zno: out of curiousity, where do you think the freebsd ports system has an edge over debian? |
| 16:35.23 | bile_one | ManxPower, I can't get David to talk at all. |
| 16:35.37 | bile_one | ManxPower well from the command line he talks |
| 16:35.51 | zno | CosmicRay: much simpler. Instead of apt-get install or apt-cache search or dpkg -l etc, there's pkg_* |
| 16:35.53 | Darwin[laptop] | the Job market in PA stinks |
| 16:36.19 | zno | CosmicRay: and also, it's easier to fix the package database |
| 16:36.36 | Darwin[laptop] | on fbsd you have pkg_delete pkg_add mae deinstall and make install |
| 16:36.41 | Darwin[laptop] | mae/make |
| 16:36.51 | CosmicRay | zno: I dunno, I've never had the package database break in debian. but it is all flat files if it does. |
| 16:36.52 | Darwin[laptop] | those are your most needed to know |
| 16:37.18 | Darwin[laptop] | and how to use cvsup and the /usr/src and /usr/ports dir |
| 16:37.19 | CosmicRay | zno: I have had weird things happen to the package database ni freebsd, like it thinking I have two different versions of something installed |
| 16:37.32 | zno | CosmicRay: then you fix it via pkgdb -F |
| 16:37.35 | CosmicRay | Darwin[laptop]: apt-get install, apt-get remove :-) |
| 16:37.47 | Darwin[laptop] | that also |
| 16:38.01 | Darwin[laptop] | apt-get -r installs the package |
| 16:38.19 | Darwin[laptop] | and fetches it |
| 16:38.26 | bile_one | Darwin[laptop] the Job market in whole sucks. |
| 16:38.44 | zno | just think about the syntax: apt-get remove ... get and remove? |
| 16:39.00 | Darwin[laptop] | I have 11 job offers but all are from up north Milwalkie and Minn/STPaul |
| 16:39.13 | *** join/#asterisk Aze` (~aze@host229-162.pool80105.interbusiness.it) |
| 16:39.13 | elriah | More than 100K? |
| 16:39.15 | zno | it's like Steve Balmer's mother who complained: "So you have to press "Start" to shut down and stop windows?" |
| 16:39.18 | *** join/#asterisk NetOfSickCoder (~um@200.121.129.178) |
| 16:39.22 | bile_one | Darwin[laptop] Beer! need I say more? |
| 16:39.29 | Aze` | hi all |
| 16:39.39 | NetOfSickCoder | hi friends |
| 16:39.40 | Darwin[laptop] | hehhee |
| 16:39.44 | Aze` | Anyone have experience about I4L ? |
| 16:39.59 | NetOfSickCoder | i've a problem with my asterisk :( |
| 16:40.16 | ManxPower | Aze`, almost everyone that uses I4L switches to ZapBRI or CAPI |
| 16:40.17 | Pinhole | clean your keyboard. :) |
| 16:40.19 | *** join/#asterisk Uther_P (~uther_p@66.180.120.83) |
| 16:40.25 | Darwin[laptop] | dont state you have a problem |
| 16:40.33 | Darwin[laptop] | use pastebin post the error |
| 16:40.43 | Darwin[laptop] | and then past the link in the channel |
| 16:40.57 | Darwin[laptop] | it does no good to use if we dont see the issue |
| 16:40.58 | elriah | Serious question - is there a text to speech option in asterisk? I called Digium and was navigating around in there system and ran into a robotic sounding synthisized auto attendent ... |
| 16:41.00 | NetOfSickCoder | Mar 18 11:34:02 NOTICE[18042]: chan_sip.c:7681 handle_request: Registration from '200 <sip:200@192.168.1.12>' failed for '192.168.1.61' |
| 16:41.16 | Darwin[laptop] | ok you fone did not register |
| 16:41.21 | bile_one | elriah yes there are two opitons for TTS. |
| 16:41.30 | *** part/#asterisk lespiggot (~les@217.206.141.131) |
| 16:41.36 | Aze` | MancPower i cant switch to CAPI because my kernel is 2.4 and BAD QUALITY with ZAPBRI (my problem is only with PTP MODE) |
| 16:41.40 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 16:41.43 | Darwin[laptop] | make sure you have sip.conf setup right and the phone matches |
| 16:41.56 | ManxPower | Aze`, Give up then. |
| 16:42.11 | TheBear | bile_one: whenever np. |
| 16:42.14 | NetOfSickCoder | wel i use 2 sip user agent's D-Link |
| 16:42.19 | elriah | bile_one: A little hint? On the wiki maybe? |
| 16:42.47 | elriah | festival? |
| 16:43.02 | Pinhole | elriah, look for festival or swift on voip-info.org |
| 16:43.04 | Aze` | ManxPower have any experience about PTP mode ? |
| 16:43.16 | NetOfSickCoder | Mar 18 11:32:42 NOTICE[18042]: chan_sip.c:7681 handle_request: Registration from '600 <sip:600@192.168.1.12;user=phone>' failed for '192.168.1.60' |
| 16:43.19 | ManxPower | Aze`, no. |
| 16:43.19 | Darwin[laptop] | but still no go |
| 16:43.32 | elriah | Thanks. |
| 16:43.44 | Darwin[laptop] | net check you have your sip setup right |
| 16:43.52 | bile_one | elriah, yes. There is a good setup for weather using festival and Aterisk@home, it work in all asterisk boxes as long as festival is installed. However Cepstral has cleaner voices, but costs a measly 30.00 bucks! |
| 16:43.57 | Darwin[laptop] | this is a sip.conf issue |
| 16:44.03 | Darwin[laptop] | go read |
| 16:44.22 | Darwin[laptop] | what type of phone |
| 16:44.57 | NetOfSickCoder | analogic phone |
| 16:45.04 | bile_one | elriah, the setup of swift aka Theta is not easy. Festival is way easier, but sounds like crap. |
| 16:45.20 | Darwin[laptop] | whNet what device are yo using to connect the phones |
| 16:45.32 | Pinhole | cepstral (swift) works well from agi. |
| 16:45.40 | Darwin[laptop] | a sipra ? |
| 16:45.46 | Darwin[laptop] | a ata ot what |
| 16:45.52 | *** part/#asterisk sysdef (~sysdef@sysdef.admin.debiancenter) |
| 16:45.56 | *** part/#asterisk rontecxt44 (~rontecxt4@dsl9-173.rb.comporium.net) |
| 16:46.04 | bile_one | elriah, ManxPower is responsible for workout a lot of the agi with TTS using both festival and Theta, but he is also responsilbe for my current headaches |
| 16:46.13 | Darwin[laptop] | yes but cepstral is not open src or I would port it |
| 16:46.17 | fitzel | whats "better" festival or mbrola? |
| 16:46.24 | NetOfSickCoder | voIP gateway sip |
| 16:46.28 | *** join/#asterisk rephorm (~rephorm@ip67-95-13-60.z13-95-67.customer.algx.net) |
| 16:46.51 | Darwin[laptop] | net what hardware are you using to connect the phones to the box |
| 16:46.56 | *** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 16:47.19 | bile_one | Pinhole, proveit! |
| 16:47.21 | ManxPower | fitzel, mbrola is a set of voices for Festival |
| 16:47.59 | ManxPower | bile_one, Remember, I never streamed the tts to the caller, I always rendered it to a file and then used Playback or Background |
| 16:48.36 | Pinhole | bile_one. download the phpagi version 2 from cvs. take any example that has text2wav and replace it with swift. |
| 16:48.37 | NetOfSickCoder | phone ----> gateway sip --->(eth0)linuxbox with asterisk |
| 16:48.55 | bile_one | ManxPower, thanks for clearing that up for me. I want to do both? I need to archive the file and the text, and send the text out 4 ports |
| 16:48.59 | fitzel | manx, ah, whow. I played around with mbrola some month ago. |
| 16:49.03 | Darwin[laptop] | net your not listening |
| 16:49.18 | Darwin[laptop] | what hard ware are you connecting the phones |
| 16:49.37 | Darwin[laptop] | is it a sipra or a ata 286 or are these sip phones ? |
| 16:50.06 | *** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
| 16:50.46 | NetOfSickCoder | FXS D-link similar to ata |
| 16:51.05 | FuriousGeorge | hi, all. i was about to dive into asterisk and set up a small test system using two voip lines. im just wondering which voip dialtone provider is most compatible w/ asterisk (less of a hassle to set up) breadvoice? sip phone? |
| 16:51.18 | *** part/#asterisk GMsoft (~r0_ot@gmsoft.developer.gentoo) |
| 16:51.31 | FuriousGeorge | broadvoice* |
| 16:51.37 | FuriousGeorge | lol |
| 16:51.52 | Darwin[laptop] | ok you need to check your sip.conf and you dlink unit and make sure they are matched for info |
| 16:52.07 | bjohnson | FuriousGeorge: nufone |
| 16:52.27 | CosmicRay | I have heard a lot of bad things about both broadvoice and nufone on the list in the past week |
| 16:52.32 | Darwin[laptop] | broadvoice is fine in 1.0.6 |
| 16:52.41 | Darwin[laptop] | it was fixed |
| 16:53.02 | FuriousGeorge | nufone, huh. never heard of them. any thing that stands out about nufone over say broadvoice |
| 16:53.32 | bjohnson | they are here |
| 16:54.13 | *** join/#asterisk tuxinator_linuxM (~tuxinator@m410e36d0.tmodns.net) |
| 16:54.41 | *** join/#asterisk Grooby (~Grooby@12.22.232.212) |
| 16:54.50 | JerJer[mobile] | CosmicRay: when people get pissed off they tend to stretch the truth or simply lie just to make people feel sorry for themselves |
| 16:55.02 | CosmicRay | sigh |
| 16:55.15 | tuxinator_linuxM | Morning Gents |
| 16:57.45 | Eight | CosmicRay: BV is fine. |
| 16:57.51 | Eight | CosmicRay: 1.0.6 works unpatched. |
| 16:58.10 | Eight | CosmicRay: and the 'second example' on the BV settings wiki page has worked every time I"m aware of. |
| 16:58.24 | CosmicRay | ok, that is good to hear. |
| 16:58.41 | NetOfSickCoder | a question, when a config the agents in asterisk can be |
| 16:58.52 | NetOfSickCoder | agent => 500,500,eder |
| 16:59.08 | NetOfSickCoder | agent => 600.600,chris |
| 16:59.40 | FuriousGeorge | darwin: this nufone is scarce on the info. they say on the main page its 2C a minute for calling, but how much do they charge for a phone number for incomming calls? do they have any unlimited callingpackages? do they have numbers local to me? |
| 16:59.57 | Eight | FuriousGeorge: call them. |
| 17:00.03 | Eight | post what you get on the wiki |
| 17:00.18 | NetOfSickCoder | or should be: agent => 500,500,eder 600,600,chris |
| 17:00.39 | Uther_P | voicepulse has a nice package called connect |
| 17:01.00 | ManxPower | ~actcvs |
| 17:01.10 | ManxPower | ~astcvs |
| 17:01.11 | jbot | somebody said astcvs was echo "CVS-HEAD:"; cvs co asterisk asterisk-sounds libpri zaptel; echo "CVS 1.0.x:"; cvs co -r v1-0 asterisk asterisk-sounds libpri zaptel; echo "Anyone that uses CVS and is not on asterisk-cvs mailing list is an idiot! See http://lists.digium.com/mailman/listinfo/asterisk-cvs" |
| 17:01.13 | ManxPower | ~astdoc |
| 17:01.15 | Uther_P | $8 per phone number, free incomming calls, 3c per minute outgoing, unlimited simultanious connections |
| 17:01.26 | ManxPower | ~mailinglist |
| 17:01.27 | jbot | [mailinglist] Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 17:01.48 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 17:02.21 | *** join/#asterisk dumbo (~dambodog@xd8ad3284.ip.e-nt.net) |
| 17:03.55 | Eight | Uther_P: 8/mo, or just setup? |
| 17:04.06 | Uther_P | $8 per month per phone number |
| 17:04.12 | dumbo | hi all. i am interested in setting up a small call center (4 ppl) and have the rest of my office on the same pbx box. i need to be able to setup confrence calls, caller id, vm and about 30 ext.. what are the basic things that i would need. I currently have 4 pstn lines coming in. ideas? |
| 17:04.14 | FuriousGeorge | be nice if some1 answered the nufone |
| 17:04.38 | Eight | Asterisk. TDM400P, IP phones. |
| 17:04.42 | CosmicRay | Uther_P: I think I am going to try livevoip |
| 17:04.44 | Eight | dumbo: Asterisk. TDM400P, IP phones. |
| 17:04.45 | fac_ | Can somebody look. and maybe help me http://pastebin.com/259733 |
| 17:04.48 | CosmicRay | Uther_P: 1.2 cents per minute outgoing |
| 17:04.53 | FuriousGeorge | uther_p: isnt it 2 cents a min to us and 48 |
| 17:04.54 | fac_ | that i received when i try to compile asterisk |
| 17:05.04 | CosmicRay | Uther_P: incoming optional, and they are $4 to $6/mo, with $6/mo being unlimited incoming up to 2 channels |
| 17:05.13 | Uther_P | FuriousGeorge: 3c is what voicepulse was last time I checked |
| 17:05.13 | FuriousGeorge | i mean inside the continental 48, isnt it 2 centds a min |
| 17:05.17 | CosmicRay | Uther_P: the other cool thing is that an 800 number is $1/mo with incoming bilad at 1.2 cpm |
| 17:05.33 | Darwin[laptop] | wow I just sold 5 soho units |
| 17:05.45 | Uther_P | CosmicRay: thats cool if you are only going to use 2 channels, voicepulse has no limit on channels |
| 17:05.48 | Darwin[laptop] | with the spa phones |
| 17:05.48 | FuriousGeorge | soho units of what? asterisk servers? |
| 17:05.49 | PTG123 | something to mention about livevoip, the other day they just decided to cancel us dids, and cancled everyones accounts on the spot |
| 17:05.59 | CosmicRay | Uther_P: true |
| 17:06.02 | Darwin[laptop] | my asterisk box I developed |
| 17:06.17 | FuriousGeorge | you preconfiure em and sendf them out |
| 17:06.22 | ManxPower | "All VoIP service providers suck!" "All softphones suck!" |
| 17:06.22 | CosmicRay | PTG123: huh, they're still selling them |
| 17:06.23 | *** join/#asterisk mutilator (~animenodv@65.111.201.79) |
| 17:06.23 | Darwin[laptop] | this rocks |
| 17:06.27 | Uther_P | voicepulse is freakin reseller's prices... which is what I'm going to be doing with it |
| 17:06.34 | PTG123 | cosmic: not last night when i checked they weren't |
| 17:07.01 | PTG123 | i talked to someone on an account on there and he told me |
| 17:07.07 | PTG123 | then i went to support/network status on their site |
| 17:07.11 | PTG123 | and it says that on there |
| 17:07.11 | Darwin[laptop] | its for a inhouse service Manx |
| 17:07.16 | CosmicRay | wow |
| 17:07.18 | CosmicRay | that sucks. |
| 17:08.06 | Darwin[laptop] | 5 phones on each unit |
| 17:08.37 | Darwin[laptop] | but they are for some new Houses going up for students |
| 17:08.39 | FuriousGeorge | why would live voice have an asterisk specific plan |
| 17:08.44 | FuriousGeorge | i mean livevoip |
| 17:08.50 | dumbo | Eight, what about redundency? if i want to make sure that my users will always be "on"? |
| 17:08.53 | Darwin[laptop] | they want a inhouse phone stup |
| 17:08.58 | Nugget | LookupCIDName makes my life easier. |
| 17:09.34 | PTG123 | Nugget: what do you look it up against? |
| 17:09.43 | FuriousGeorge | darwin: thats just the DIY plan, in other words |
| 17:09.47 | Nugget | my os x address book |
| 17:10.01 | JerJer[mobile] | so slePP where is my money? I knew that moron would post his bullshit to one of the asterisk lists |
| 17:10.04 | Uther_P | PTG123: it looks it up in a local db |
| 17:10.16 | JerJer[mobile] | :) |
| 17:10.44 | Uther_P | would be nice if someone had a complete db for cid names that we could look numbers up in |
| 17:10.55 | JerJer[mobile] | dumbo: then add in redunancy... it is trivial |
| 17:11.19 | NetOfSickCoder | muy fxs gateway sip unit, show " Get_error_code" |
| 17:13.08 | FuriousGeorge | <PROTECTED> |
| 17:13.21 | Darwin[laptop] | Net was this device setup for a diff company like vonage at 1 point |
| 17:13.28 | Shido6 | err |
| 17:13.31 | Shido6 | NuFone is answering |
| 17:13.58 | *** join/#asterisk Remowylliams (~Mare@168.215.138.106) |
| 17:14.32 | NetOfSickCoder | for register the users should be add in the agent.conf |
| 17:15.24 | TheBear | bile_one: any ideas yet ? |
| 17:16.08 | bile_one | TheBear you should set the txgain and the rxgain to 4.5. Any change you make to a zaptel device requires a full restart of aster. |
| 17:17.09 | ManxPower | actually it requires stoping and then starting asterisk. |
| 17:17.52 | ManxPower | Shido6, What phone number is NuFone answering? |
| 17:18.30 | *** join/#asterisk caesar2 (~igerl@ppp-82-135-65-72.mnet-online.de) |
| 17:19.00 | ManxPower | I hear users saying "Provider X is not answering", but never provide the number they are dialing to reach support. Then I see provider X say "Yes, we are answering!", but they never say what number they are answering. |
| 17:19.49 | ManxPower | Adn then the user quits IRC before even waiting for any answer. |
| 17:19.52 | bile_one | ManxPower haa haa haa, 2 funny |
| 17:20.08 | NetOfSickCoder | jeje when i call to other phone, asterisk response with a operator |
| 17:20.51 | ManxPower | bile_one, Other people say things like "asterisk response with a operator" and then never say what the "operator" is saying. |
| 17:21.05 | *** part/#asterisk Dandan (dandan@234.88.149.195.in-addr.arpa.virt-ix.net) |
| 17:21.27 | bannerman | Is there a cheap device I can use for very very very rare PSTN calls? I just want 911 support |
| 17:21.35 | bannerman | everything else I have is pure voip |
| 17:21.50 | ManxPower | bannerman, It's called "an analog line and an analog phone" |
| 17:21.59 | bannerman | manxpower: that's the best idea I've heard yet. |
| 17:22.00 | bile_one | get a generic x100p and put a phone on it. |
| 17:22.16 | bannerman | x100p? |
| 17:22.16 | bannerman | k |
| 17:22.18 | Darwin[laptop] | othe then for the sale he made |
| 17:22.18 | bannerman | thanks |
| 17:22.22 | ManxPower | bannerman, I have a nice bright red phone in my apartment that is hooked directly into the local PSTN line. |
| 17:22.53 | bile_one | ManxPower does the pres call you on it. |
| 17:23.13 | ManxPower | bile_one, Only if he wants better advice than he's getting now. |
| 17:23.17 | Uther_P | no bile, that phone is directly to the commisioner |
| 17:23.24 | TheBear | bile_one: as much as 4.5 won't that effect the volume you hear ? |
| 17:23.29 | *** join/#asterisk sysdebug (~jonasgoes@200.163.193.247) |
| 17:23.41 | ManxPower | Of course, my advice would be "Shoot yourself in the head." but it's still much better advice than he's getting now. |
| 17:23.45 | TheBear | and that would be -4.5 right |
| 17:23.56 | Uther_P | manx: hah, yea... that would be good advice |
| 17:23.57 | bile_one | Echo is hit and miss. I have terrible luck with IAX to ZAP so I use only SIP now. |
| 17:24.23 | TheBear | ok thanks |
| 17:24.36 | bile_one | Pinhole thanks. I assumen the docs are in this to help install and configure? |
| 17:25.24 | Pinhole | The docs could be better. You basically just put the script in your agi-bin and call them from your dial plan. |
| 17:25.44 | bile_one | TheBear a lot of echo problems are really bandwidth problems on a non QoS network too!, for example if you are downloading 200 gigs of mp3's and talking on a softphone from your computer. |
| 17:25.45 | bjohnson | bannerman: a cell phone is another option |
| 17:25.59 | *** join/#asterisk _NaNDao_ (~my@c9066570.virtua.com.br) |
| 17:26.05 | _NaNDao_ | hello |
| 17:26.37 | ManxPower | _NaNDao_, The answers you seek are at http://www.voip-info.org/ |
| 17:26.47 | bannerman | yeah, cell phones are also all over here |
| 17:26.50 | _NaNDao_ | I have a AudioCodes (FXS-2 Rev.B & AVP-04 Rev.c). Is it works with asterisk ? |
| 17:27.09 | bannerman | but I don't want to have some emergency, have some guy frmo the shop run in here and grab a phone and dial 911 and get a fast busy signal. |
| 17:27.11 | ManxPower | _NaNDao_, The answers you seek are are in the Asterisk mailing list archives |
| 17:27.19 | bannerman | if someone gets run over by a forklift, I'd like any phone be able to call 911 |
| 17:27.20 | ManxPower | ~google site:lists.digium.com audiocodes |
| 17:27.28 | TheBear | bile_one: ok I'll try -4.5 will that effect the volume that you hear the person also or not ? |
| 17:27.47 | ManxPower | ~google site:lists.digium.com echo gain |
| 17:27.52 | bannerman | why are there "Digium Wildcard X100P OEM FXO PCI Card for Asterisk PBX" on ebay for $6.95 brand new buy it now when they cost $60-100? |
| 17:28.00 | bile_one | TheBear, again not sure. It will depend on your certain conditions |
| 17:28.08 | ManxPower | bannerman, The X100P is no longer sold. |
| 17:28.09 | Uther_P | bannerman: if someone gets run over by a forklift, I don't think time would be an issue anymore ;) |
| 17:28.19 | bannerman | Uther_P: No, but it's the concept! |
| 17:28.30 | Uther_P | heh |
| 17:28.46 | NetOfSickCoder | Asterisk have a GUI interface for managment? |
| 17:29.04 | NetOfSickCoder | in a forum i see a asterisk manager |
| 17:29.05 | Nugget | NetOfSickCoder: if you want to automate or delegate some of the more boring routine maintenance tasks, then yes. |
| 17:29.07 | bannerman | So because it's outdated, it's cheap.. I like it! |
| 17:29.25 | Nugget | if you think that installing a GUI will allow you to avoid learning how the config files work, then no, there is no solution you will be happy with |
| 17:29.38 | NetOfSickCoder | no |
| 17:29.39 | Pinhole | GUI == GIANT UGLY INTERFACE? :) |
| 17:30.02 | NetOfSickCoder | i think same |
| 17:30.11 | NetOfSickCoder | the GUI is for my BOSS :/ |
| 17:30.24 | Nugget | in that case, then yes, there might be one which you will be happy with |
| 17:30.29 | NetOfSickCoder | he don't understand the beautiful line commands |
| 17:30.33 | Uther_P | I looked at asterisk gui's for about 2 minutes |
| 17:30.48 | NetOfSickCoder | jeje |
| 17:31.01 | Nugget | I like asternic.org, but it's not a management interface, it's an operator console. |
| 17:31.12 | Nugget | it's flashy (no pun intended) and makes BOSSes happy. |
| 17:31.19 | *** join/#asterisk steve_murphy (murf@wyoming.e-tools.com) |
| 17:31.24 | *** join/#asterisk NewSole (david@i216-58-44-245.avalonworks.net) |
| 17:32.41 | NetOfSickCoder | i work with my slackware 10, full console |
| 17:33.02 | *** join/#asterisk angler_ (~angler@suid.digium.com) |
| 17:33.07 | bile_one | bannerman, you can get an IA92 modem to work as an alternative, but no call processing will work. |
| 17:33.09 | *** join/#asterisk Feral_Kid (~me@209.205.207.130) |
| 17:33.16 | *** join/#asterisk widowlicker (~Naturalbl@62.77.178.121) |
| 17:33.17 | sudhir492 | Anyone here has Asterisk with E1 card? |
| 17:33.17 | NetOfSickCoder | but in this moment i'm over winbugs :/, omg all for my BOSS |
| 17:33.21 | widowlicker | hi there |
| 17:33.28 | widowlicker | i have an issue with iax connections |
| 17:33.47 | widowlicker | when i connect up and check the iax shoe registry |
| 17:33.55 | widowlicker | it keeps givin a diddferent port |
| 17:34.01 | widowlicker | instead of 4569 |
| 17:34.05 | widowlicker | any idea why |
| 17:34.13 | widowlicker | it is set to bind to it in iax.conf |
| 17:35.36 | steve_murphy | _bkw... you there? Markster? |
| 17:35.37 | NewSole | the port its showing is just the port its connected to on other server not your server |
| 17:36.03 | widowlicker | no this is under the perceived tab |
| 17:36.16 | widowlicker | im showing the right port on there side |
| 17:36.55 | NewSole | you have more then one registry |
| 17:37.29 | NewSole | mine shows up different |
| 17:38.14 | Feral_Kid | NewSole: I have all the proper ports, but I can't get incoming calls... |
| 17:38.35 | *** join/#asterisk mkhan (~mkhan@ip66-105-190-122.z190-105-66.customer.algx.net) |
| 17:39.14 | mkhan | hello |
| 17:40.09 | *** join/#asterisk Madounet (~Madounet@ASt-Lambert-103-1-4-33.w80-15.abo.wanadoo.fr) |
| 17:40.11 | mkhan | I just bought a TDM400P with 1 FXO and 1 FXS .. the 4 ports it have on the card ..looks like RJ45 .. is this port indendent for telephone line ? |
| 17:40.32 | Uther_P | the middle pair |
| 17:40.50 | Jer13261 | blue/bluewhite |
| 17:41.07 | Uther_P | the fxo is for the line from the telco, the fxs goes to the telephone or phone device |
| 17:41.43 | NewSole | Feral_Kid... sorry I am not sure why yours without looking at it... |
| 17:42.56 | *** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
| 17:43.45 | mkhan | will I have to make a telephone cable using RJ45 clip ? !! |
| 17:43.48 | FuriousGeorge | one of the most disconcerrting things to me about picking a voip dialtone provider is that every time i speak to thm on the phone, im unimpressed with the voice quality on their end, to put it nicely |
| 17:44.17 | Jer13261 | who have you tried |
| 17:45.16 | Uther_P | has anyone ever consitered making an mp3 codec for VoIP? |
| 17:45.23 | FuriousGeorge | nufone and livevoip today |
| 17:45.30 | *** join/#asterisk NetOfSickCoder (~um@200.121.129.178) |
| 17:45.35 | NewSole | mkhan... your RJ11 with fit right in |
| 17:45.44 | FuriousGeorge | i tried braodvoice a few months back, they were better, but it sounded packet-lossy from time to time |
| 17:45.49 | `Sauron | Uther_P: mp3 codec would probably be pointless in terms of bang-for-the-buck |
| 17:45.57 | Uther_P | why? |
| 17:46.11 | Uther_P | it would be very compact |
| 17:46.22 | `Sauron | But at what cost? |
| 17:46.26 | coppice | compact in what sense? |
| 17:46.52 | `Sauron | mp3 de/encoding is cpu intensive |
| 17:46.52 | mkhan | NewSole, but isn't it RJ45 on the card or I saw wrong? |
| 17:47.25 | NewSole | ya I have same card... my RJ11 is pluged into it |
| 17:47.47 | `Sauron | You can plug an rj11 connector into a rj45/rj48 jack |
| 17:47.50 | NewSole | they are RJ45 but RJ11 will fit in... |
| 17:47.59 | `Sauron | They don't fit well, and will bend the pins in the jack |
| 17:48.10 | NewSole | and the pins are right where they should be |
| 17:48.23 | CoolAcid | Trying to debug an incoming general SIP call w/o ext. When dialing directly to PBX gives 404. Any thoughts? |
| 17:48.35 | coppice | nonsense. RJ11 is designed to plug into the middle of RJ45 |
| 17:48.41 | *** join/#asterisk twisted[work] (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) |
| 17:48.41 | *** mode/#asterisk [+o twisted[work]] by ChanServ |
| 17:49.04 | `Sauron | coppice: Yes, and you'll trash pin 1 and 8 of the rj45/48 connector. |
| 17:49.13 | NewSole | yes and on the TDM400P the mane lines are in pin 4 &5 |
| 17:49.25 | `Sauron | "designed" and "works well long term" are two completely different things. |
| 17:49.52 | TheBear | in the DialPlan. I call a std. phone Dial(ZAP/2,90,tr) the phone has an answering machine. How do I tell * that even if no-one answers let the answering machine on the phone being dialed take the call. At the moment. The answer message starts and then hangs up ? |
| 17:50.25 | *** join/#asterisk doug (~doug@h-67-102-173-11.sttnwaho.covad.net) |
| 17:50.25 | *** join/#asterisk Elshar (~Elshar@ip205-68.oregonfast.net) [NETSPLIT VICTIM] |
| 17:50.31 | ManxPower | TheBear, you ask more questions than anyone I have ever seen on this channe. |
| 17:50.42 | ManxPower | TheBear, Asterisk WILL do that, by default, unless you break it. |
| 17:50.49 | doug | wow, a few people are here. |
| 17:50.53 | coppice | Uther_P: MP3 is not a voice optimised codec. it has a rather high bit rate for the same quality as a voice optimised codec. of course, if you want music over IP it is better. :-) |
| 17:50.57 | doug | anyone seen brettnem? |
| 17:51.09 | ManxPower | doug, last night or yesterday |
| 17:51.11 | TheBear | ManxPower: sorry, how am I breaking it ? |
| 17:51.47 | ManxPower | TheBear, I don't know. Are you using randomlydisconnectmycalls=yes. Oh! Sorry, the option is called busydetect=yes or callprogress=yes |
| 17:52.05 | *** join/#asterisk ClayReich (fwuser@acxexch1.accxx.com) |
| 17:52.09 | TheBear | I'm not using any of those |
| 17:52.44 | ManxPower | Perhaps a pastebin of the CLI output for the failed call would be a good start..... |
| 17:53.25 | TheBear | Zap2 ringing, Zap2 Answered, Zap2 Hangup |
| 17:53.41 | FuriousGeorge | the nufone guy is gonna call me back and im gonna exprexss my concern to him |
| 17:55.38 | ClayReich | Can someone tell me why this is happening? I have several DIDs in asterisk with areacode 813. phone number 1 is 8138644411 phone number 2 is 8138644422. If I pick up phone 1 and dial 18138644422 I get dead air. When I hang up, asterisk writes about 50 CDRs.... |
| 17:55.55 | ClayReich | The call never connects. |
| 17:56.01 | *** join/#asterisk SPoon_TSX (~SPoon_TSX@d206-116-121-40.bchsia.telus.net) |
| 17:56.18 | SPoon_TSX | hello everyone. Just wondering how can I dial a number from the CLI?? |
| 17:56.27 | ClayReich | My dial plan points all 1+ calls to a cisco gateway. |
| 17:57.42 | doug | thanks, ManxPower |
| 17:58.55 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 17:59.00 | SPoon_TSX | I got a very weird problem on my ASterisk. For some reason I cannot go into a VoiceMail when the extension is not avariable. |
| 18:00.41 | Zeeek | who can tell me the input voltage is for polycom ip500? SInce the power supply is 110v, I'll need my own. |
| 18:01.28 | ClayReich | I get this message------Mar 18 12:39:25 WARNING[16465]: channel.c:311 ast_channel_alloc: Alert pipe creation failed! |
| 18:01.47 | ClayReich | Followed by this---------Mar 18 12:39:25 WARNING[16465]: chan_sip.c:2074 sip_new: Unable to allocate channel structure |
| 18:01.48 | ManxPower | FuriousGeorge, That telephone number were you CALLING? |
| 18:02.09 | Zeeek | ManxPower can you tell me the DC input voltage to the ip500? |
| 18:02.10 | ClayReich | Then this--------Mar 18 12:39:25 NOTICE[16465]: app_dial.c:746 dial_exec: Unable to create channel of type 'SIP' |
| 18:02.23 | ManxPower | Zeeek, no. |
| 18:02.34 | Zeeek | it's not written on the little hole? |
| 18:02.43 | TheBear | ManxPower: http://pastebin.ca/7705 is the CLI You hear the answering machine kick in on line 7 and disconnect on line 9 |
| 18:02.48 | Zeeek | well, above or below |
| 18:02.49 | ManxPower | ClayReich, That message should read "I cannot connect to the destination device you specified" |
| 18:03.35 | ManxPower | Zeeek, All the polycom phones are over 50 miles away from me. |
| 18:03.50 | Zeeek | funny that this data is not on the datasheet, even the supply itself repeats "10w" |
| 18:03.55 | ManxPower | I use a cordless phone on a SIPura ATA at home. |
| 18:04.14 | Zeeek | understood - I'll ask again later if I haven't found it next week :) |
| 18:04.18 | ClayReich | ManxPower: so, that means my Cisco? |
| 18:04.45 | ClayReich | Am I attempting a hairpin call? |
| 18:05.29 | ManxPower | ClayReich, I have no idea what you just said. |
| 18:05.43 | Zeeek | calling out of therouter and back in |
| 18:06.00 | *** join/#asterisk kore (kore@rosa.st) [NETSPLIT VICTIM] |
| 18:06.02 | ClayReich | Zeek: yes... is that a no no? |
| 18:06.10 | Zeeek | most routers won't do it |
| 18:06.17 | Zeeek | or so I've read |
| 18:06.18 | Darwin[laptop] | kram around ? |
| 18:06.25 | Zeeek | and why would you ever want to? |
| 18:06.39 | Zeeek | if you can talk to 10.0.0.10 |
| 18:07.28 | SPoon_TSX | Can someone help me out? I am using X-LIte and try to call my own extension but I cannot hear anything from my voicemail, even the promte... would it because I don't have the sound card installed? |
| 18:07.58 | Zeeek | on the box running X-Lite? |
| 18:08.16 | SPoon_TSX | Yes. |
| 18:08.24 | ClayReich | Zeeek: I have several DIDs with different area codes. My customers won't neccessarily know they are calling a number that is on the same asterisk system. They would just dial 1-925-XXXX unaware that that is an "ON-NET" call.... |
| 18:08.33 | Zeeek | X-Lite wants to talk thru your sound card |
| 18:08.43 | Nugget | without a sound card how did you *expect* to hear anything? |
| 18:08.45 | Nugget | seriously |
| 18:08.47 | Eight | where the heck were you expecting the sound to come out of? |
| 18:08.48 | Nugget | I want to know |
| 18:08.51 | Zeeek | ClayReich I'm betting that YOPU will need to find out :) |
| 18:08.52 | mogorman | esp? |
| 18:08.54 | SPoon_TSX | O, no no. I got a sound card on my computer but not on the server. |
| 18:09.03 | stevekstevek | My car seems to be very rough and noisy when I drive. There's also lots of sparks. Could this be because I have no tires? |
| 18:09.12 | Nugget | asterisk does not require a sound card |
| 18:09.14 | mogorman | tires?!?!?! |
| 18:09.30 | Nugget | in fact, the things it can do with a sound card are more just novel curioisities than anything useful |
| 18:09.54 | Zeeek | ClayReich I think you'll need to know in the dialplan |
| 18:10.09 | SPoon_TSX | THen it killing me now. I tried to call a Non-registered extension, I do saw the asterisk try to play back the voicemail message but it just hung my call up in no time. |
| 18:10.32 | Zeeek | SPoon_TSX make sure Transmit Silence is set to YES |
| 18:10.39 | *** join/#asterisk [cc]smart (~smart@62.65.149.158) |
| 18:10.41 | Zeeek | in X-Lite |
| 18:11.04 | ClayReich | Zeeek: you mean I won't have the luxury of saying exten => _1.,Dial(SIP/gateway) |
| 18:11.18 | Zeeek | in one of the 25 menus maybe under system, audio |
| 18:11.25 | JerJer[mobile] | ClayReich: that is just evil |
| 18:11.33 | SPoon_TSX | Zeeek: I am using eyeBeam, the most update one but seems there is no such settings. |
| 18:11.36 | Zeeek | ClayReich well you haven't yet proved it's the hairpin that's causing problems |
| 18:11.53 | Zeeek | SPoon_TSX why do say I'm using X-Lite and then change it? |
| 18:12.02 | Zeeek | Is that supposed to make it easier to help? |
| 18:12.14 | ManxPower | Many audio problems are caused by allow=all or a bandwidth= line in iax.conf or sip.conf |
| 18:12.16 | JerJer[mobile] | _1. is not valid |
| 18:12.22 | SPoon_TSX | X-Lite eyeBeam. |
| 18:12.25 | Zeeek | good call Manx |
| 18:12.26 | JerJer[mobile] | _1X. is valid but not proper |
| 18:12.33 | Darwin[laptop] | shit shabang |
| 18:12.35 | JerJer[mobile] | _1NXXNXXXXXX is proper |
| 18:12.44 | JerJer[mobile] | for nanpa dialing |
| 18:12.44 | Zeeek | SPoon_TSX there may be that setting, you'll never know until you go look |
| 18:13.17 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
| 18:13.18 | *** topic/#asterisk is Asterisk: The Open Source PBX || 1.0.7 RC - bug #3746 || http://www.theregister.co.uk/2005/03/16/asterisk_open_source_pbx/ |
| 18:13.24 | JerJer[mobile] | ManxPower: correct.... my theory is to pick ONE AND ONLY ONE codec and use it everywhere |
| 18:13.36 | Zeeek | and o,ne and only one provider? |
| 18:13.50 | Pinhole | _. is always fun! |
| 18:13.53 | Jer13261 | g729 :) |
| 18:14.01 | JerJer[mobile] | Zeeek: i cannot answer that question as I am biased |
| 18:14.06 | Zeeek | nah! |
| 18:14.12 | *** join/#asterisk Trepalium (~chadk@wnpgmb02dc1-60-221.dynamic.mts.net) |
| 18:14.22 | Zeeek | My answer is, no that'd be pretty sillly for general use |
| 18:14.27 | JerJer[mobile] | i happen to like Speex |
| 18:14.33 | Zeeek | because if the one is down, your quiet |
| 18:14.38 | ClayReich | JerJer: I think you get my meaning though.... my memory fails....:) |
| 18:14.41 | Zeeek | (1 provider) |
| 18:15.00 | Trepalium | Is it possible to hook a channel bank up to one port on a T410P, and a telco provided T1 to another port of the same card? |
| 18:15.10 | SPoon_TSX | How do I setup my SIP client talk via G711u? |
| 18:15.16 | SPoon_TSX | I mean sip.conf |
| 18:15.21 | jontow | disallow=all |
| 18:15.23 | jontow | allow=ulaw |
| 18:15.24 | Darwin[laptop] | open skull insert new memory module screw top back on |
| 18:15.28 | JerJer[mobile] | what he sad |
| 18:15.29 | JerJer[mobile] | said |
| 18:15.42 | *** join/#asterisk didz_ (didz_@200.218.192.52) |
| 18:15.48 | Zeeek | or read one of these: |
| 18:15.49 | Zeeek | Starter tutorial: |
| 18:15.49 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 18:15.49 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 18:15.49 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 18:15.49 | Zeeek | THE reference of the moment: |
| 18:15.51 | Zeeek | http://www.asteriskdocs.org |
| 18:15.58 | Zeeek | any one has the answer |
| 18:16.14 | ManxPower | Trepalium, 400,000 asterisk users are doing exactly that! |
| 18:16.25 | Trepalium | Okay, I thought so. Just wanted to be absolutely sure. |
| 18:16.52 | Trepalium | Digium's restocking fee is too high for me to make silly mistakes. |
| 18:17.00 | SPoon_TSX | F, I narrow it down to the one of my SIP phone problem. |
| 18:17.02 | ClayReich | Zeek: What is the best way for me to approach? Is this a job for agi? |
| 18:17.30 | Zeeek | anyone here would know better than I ! I just happen to know what hairpin is |
| 18:17.52 | Zeeek | and only from seeing it in a discussion about NAT which I was troubleshooting at the time |
| 18:17.58 | *** join/#asterisk Mcwebtree (~ha@82-69-199-97.dsl.in-addr.zen.co.uk) |
| 18:18.22 | *** join/#asterisk algorithmn (~na@ool-18bce89c.dyn.optonline.net) |
| 18:18.41 | Zeeek | ClayReich but surely you have a database of what domains are locally hosted? |
| 18:19.20 | ClayReich | Zeeek: I have a database of my numbers, yes. |
| 18:19.28 | Zeeek | how many numbers? |
| 18:19.33 | NetOfSickCoder | Zeeethank you for the doc's |
| 18:19.39 | ClayReich | 500 |
| 18:19.45 | NetOfSickCoder | Zeeek, thank you fot the doc's |
| 18:19.56 | Zeeek | that's the only reason we keep posting em : so eventually one person will take a look :) |
| 18:20.06 | *** join/#asterisk SexyKen (~sexyken@c-67-161-5-149.client.comcast.net) |
| 18:20.08 | SexyKen | Mar 18 12:25:20 WARNING[30522]: chan_sip.c:739 retrans_pkt: Maximum retries exceeded on call fc5f56c2-a1f933bc-bdeb6963@192.168.1.125 for seqno 112 (Non-critical Request) |
| 18:20.15 | SexyKen | Anyone know why I'd get this over and over and over? |
| 18:20.25 | SexyKen | My phone isn't even making or recieving calls when the message appears. |
| 18:20.34 | ClayReich | Sounds like I need to strip the 1, check the 10 digits for a match in my database and tailor the call... |
| 18:20.43 | Zeeek | somthin like that |
| 18:20.45 | *** join/#asterisk EvlHimeko (~himeko@S01060040ca128fc3.ed.shawcable.net) |
| 18:20.52 | Zeeek | Perl |
| 18:21.00 | ClayReich | right... |
| 18:21.07 | ClayReich | agi and perl? |
| 18:21.07 | Zeeek | or whatever your fav flavor is |
| 18:21.11 | Zeeek | yeah |
| 18:21.22 | Pinhole | or php or python or java or c or BASH!!!! |
| 18:21.22 | Zeeek | or a module if you like c |
| 18:21.31 | ClayReich | ok thanks! I was hoping I could get around that some way.... |
| 18:21.56 | JerJer[mobile] | or a module if u want the application to scale |
| 18:21.57 | Zeeek | move all possible receiving domains somewhere else :) |
| 18:22.17 | Zeeek | I wrote a module just for fun, and it worked afetr a few segfaults |
| 18:22.29 | doug | anyone ever set up a unix system to *receive* SMS messages? |
| 18:22.46 | Zeeek | doug yes, and then the application changed the directories suddenly |
| 18:22.57 | Zeeek | where it stoers them |
| 18:22.59 | Pinhole | So can you dump a core into the audio stream for debuging by the user? lol ;) |
| 18:23.10 | *** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || 1.0.7 RC - bug #3746 || http://www.theregister.co.uk/2005/03/16/asterisk_open_source_pbx/ || Dev Conf 1PM CST APRIL 1st -> IAX2/guest@switch-3.asterlink.com/996 |
| 18:23.32 | Inv_arp | can faxes be done thru voip? |
| 18:24.04 | JerJer[mobile] | Inv_arp: via LAN, sure |
| 18:24.08 | JerJer[mobile] | WAN, don't count on it |
| 18:24.12 | JerJer[mobile] | wait for T.38 |
| 18:24.12 | *** join/#asterisk BuckRogers (~steve@ool-18bce89c.dyn.optonline.net) |
| 18:24.24 | JerJer[mobile] | or roll your own with app_rx and tx fax |
| 18:24.32 | BuckRogers | hello mr nufone |
| 18:24.36 | JerJer[mobile] | moo |
| 18:24.42 | BuckRogers | hows it chillin |
| 18:24.58 | JerJer[mobile] | im alive, so I cannot complain |
| 18:25.03 | JerJer[mobile] | too much |
| 18:25.10 | BuckRogers | word-up |
| 18:25.12 | BuckRogers | same here |
| 18:25.19 | *** join/#asterisk didz_ (didz_@200.218.192.52) |
| 18:25.35 | *** join/#asterisk afe ([mL3Jk5T2D@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
| 18:25.36 | Inv_arp | JerJer[mobile]: no any el cheapo fax2email guys? |
| 18:25.42 | doug | zeeek: the application change the directories...? |
| 18:25.47 | Mcwebtree | how do I check and see if my IAX is conected? |
| 18:25.58 | doug | setting up to send SMS is dirt easy, but receiving SMS is a very different story. |
| 18:26.03 | BuckRogers | has anyone encountered asterisk issues with FC3 and SE linux |
| 18:26.11 | Zeeek | doug ya it saves the sms messages in a directory and that moved with no notice - or I didn't notice |
| 18:26.23 | doug | the only services i've found that do SMS receive online do it from the UK |
| 18:26.41 | Zeeek | doug I can receive them in France, but it really isn't that great a need |
| 18:26.46 | Inv_arp | Mcwebtree: iax2 show peers |
| 18:26.47 | *** join/#asterisk bsd3 (~bsd@203.145.128.5) |
| 18:26.55 | ManxPower | Mcwebtree, iax2 show users|peers|registry |
| 18:27.01 | Zeeek | the only use would be to send an SLMS to asterisk to tell it to call me or something |
| 18:27.27 | ManxPower | I think SMS -> Asterisk is called "E-Mail" |
| 18:27.28 | Mcwebtree | hi Inv_arp I get iax2 command not found when run at cli over ssh |
| 18:27.33 | Zeeek | SMS: rm -r / |
| 18:28.09 | Zeeek | well if you have SMS which we do, free reception, it can be useful (and discreet) |
| 18:28.18 | Inv_arp | Mcwebtree: type "ia" press <tab><tab> to autocomplete |
| 18:28.21 | Zeeek | plus you could send whole commmand sequences |
| 18:28.34 | ManxPower | or just type "help" at the asterisk cli |
| 18:28.35 | BuckRogers | has anyone encountered asterisk issues with FC3 and SE linux |
| 18:28.46 | Mcwebtree | inv_arp: it just beeps at me :( |
| 18:28.56 | doug | yeah, receiving in france isn't going to help me much. sorry, thanks for trying tho. |
| 18:28.58 | ManxPower | ~google site:lists.digium.com problem OR issue AND FC3 |
| 18:29.01 | Inv_arp | Mcwebtree: u sure ure in cli? |
| 18:29.18 | Inv_arp | BuckRogers: what knd of probs? |
| 18:29.27 | ManxPower | ~google site:lists.digium.com problem OR issue AND "se linux" |
| 18:29.45 | *** part/#asterisk bsd3 (~bsd@203.145.128.5) |
| 18:29.57 | Mcwebtree | inv_arp: I've logged into the box over ssh, what do I have to do to be in the cli? |
| 18:30.11 | Inv_arp | Mcwebtree: asterisk -rvvv |
| 18:30.11 | *** part/#asterisk sysdebug (~jonasgoes@200.163.193.247) |
| 18:30.12 | Zeeek | god I wish all mfrs would just provide 100-240VAC supplies and be dine with it! |
| 18:30.23 | *** part/#asterisk doug (~doug@h-67-102-173-11.sttnwaho.covad.net) |
| 18:30.40 | coppice | Zeeek: outside the US they generally do |
| 18:30.46 | BuckRogers | Modules and file permissions denied |
| 18:30.55 | BuckRogers | need to creat a security polocy |
| 18:30.58 | ManxPower | BuckRogers, AS ROOT! |
| 18:31.00 | Mcwebtree | inv_arp: I definately wasn't in the asterisk cli, but now am, and not I know my setup is definately wrong :S |
| 18:31.06 | BuckRogers | as user asterisk |
| 18:31.07 | ClayReich | Wow... don't know how I missed this.... chan_sip.c:6912 handle_response: Hairpin detected, setting up call forward for what it's worth |
| 18:31.07 | ClayReich | <PROTECTED> |
| 18:31.10 | Zeeek | yeah but the products are more attractive with the euro at $1.30+ |
| 18:31.23 | Zeeek | coppice^^ |
| 18:31.27 | Jer13261 | whats Hairpin? |
| 18:31.48 | ManxPower | ~google site:lists.digium.com hairpin |
| 18:32.00 | Zeeek | http://www.cisco.com/univercd/cc/td/doc/product/access/ip_ph/ip_ks/cme32/cme32sa/cme32bsc.htm |
| 18:32.34 | Zeeek | AND..... |
| 18:32.35 | Zeeek | Hairpin call routing uses the Cisco CME router to reoriginate a terminated call and route it as appropriate to complete a transfer or forward generated by a phone or other application attached to the router. There was still no way to automatically identify which endpoints supported H.450 standards, and hairpin call routing has the disadvantage of using two calls' worth of bandwidth for the duration of the transferred or forwarded call. |
| 18:32.52 | Zeeek | just ignore the cisco part |
| 18:33.00 | Jer13261 | k |
| 18:33.03 | Jer13261 | thankyou |
| 18:33.07 | Zeeek | np |
| 18:33.35 | Jer13261 | hey have anyone got FWD to work? i cant get my @#*(&#@ to work right....and i wanna use there peering stuff |
| 18:34.09 | Zeeek | did you get it to work first with ust a SIP client? |
| 18:34.11 | BuckRogers | maxpower i need it as user asterisk |
| 18:34.19 | BuckRogers | not root |
| 18:34.29 | Jer13261 | nope...should i? |
| 18:35.04 | Zeeek | It would seem be a wise path since that would check that a) FWD is up and b) you can reach it thru your net |
| 18:35.07 | Inv_arp | BuckRogers: http://voip-info.org/tiki-index.php?page=Asterisk%20non-root |
| 18:35.30 | BuckRogers | inv_arp with se linux support? |
| 18:35.42 | Zeeek | and c) you won't need asterisk to talk to people |
| 18:35.43 | Jer13261 | i can reach 615 for example |
| 18:35.57 | Jer13261 | on fwd with * |
| 18:36.07 | Zeeek | thazt's nice |
| 18:36.36 | Jer13261 | i am trying to reach vonage but without luck |
| 18:36.44 | Jer13261 | i dot know if they sitll have the peering or not |
| 18:37.05 | Mcwebtree | I have set up my * with 2 extensions. I log into one with X-Lite and it shows Logged In - Your number is 200 (the extension I have set) whenever I try to dial any of the internal extensions or other numbers I just get Call Not Approved. Any Suggestions? |
| 18:37.10 | Inv_arp | BuckRogers: no idea, dont use se linux... but first get it to run as normal user |
| 18:37.12 | Zeeek | normally I'd say go talk to the FWD forum, lot's of competent folks here, but it's been blown up |
| 18:37.27 | *** join/#asterisk Pinhole (~nuxi@lin-dsl-static-206-222-194-115.inetnebr.com) |
| 18:37.29 | Jer13261 | blown up? |
| 18:37.50 | Jer13261 | i tried to search and it did nto work one single bit |
| 18:37.51 | Zeeek | yeah most of the sections have been emptied - I suspect hw failure or lack of sleep |
| 18:37.58 | Zeeek | it does |
| 18:38.09 | Zeeek | you have to change the default from 7 days to 100 days |
| 18:38.24 | Zeeek | but there's not much to search there now |
| 18:38.39 | Zeeek | Hairpin call routing uses the Cisco CME router to reoriginate a terminated call and route it as appropriate to complete a transfer or forward generated by a phone or other application attached to the router. There was still no way to automatically identify which endpoints supported H.450 standards, and hairpin call routing has the disadvantage of using two calls' worth of bandwidth for the duration of the transferred or forwarded call. |
| 18:38.42 | Zeeek | ooops |
| 18:39.00 | Zeeek | http://yabb.pulver.com/cgi-bin/yabb/YaBB.cgi#general_cat |
| 18:43.05 | Mcwebtree | :) |
| 18:43.29 | Zeeek | well, I never did get theCID issue settled on my Siemens C200 on FXS. Maybe the phone is not so great |
| 18:44.52 | *** join/#asterisk ennuyeux72 (~ennuyeux7@83.146.53.34) |
| 18:45.03 | Inv_arp | bah i need a cheap fax2email solution... efax is rathr pricey for what i need |
| 18:45.45 | Zeeek | it's only really handy if you need to send faxes online AND need voicemail too |
| 18:46.09 | Zeeek | I'm trying to get out from under the yoke of a jfazx (same co) payment each month |
| 18:46.17 | Zeeek | jfax=j2.com |
| 18:47.21 | Zeeek | Need DID in Russia? http://www.telphin.com/contact.php |
| 18:47.32 | Zeeek | they happened to mention hairpin in their instructions |
| 18:47.42 | *** part/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 18:47.49 | Mcwebtree | <PROTECTED> |
| 18:48.31 | *** join/#asterisk buddah (~hnic@208.179.86.5) |
| 18:49.00 | Inv_arp | Mcwebtree: paste your extension.conf on pastebin.ca |
| 18:52.13 | Mcwebtree | inv_arp added http://pastebin.ca/7711 I created the extensions using the *@home and the web management thing |
| 18:52.57 | *** join/#asterisk Drel (~drel@dsl254-029-130.sea1.dsl.speakeasy.net) |
| 18:53.54 | *** join/#asterisk Mike (~mike@201.135.48.217) |
| 18:55.20 | Mike | hi guys anyone knows what where this file should goo |
| 18:55.21 | Mike | [chan_h323.so]Mar 18 06:53:48 WARNING[491]: loader.c:258 ast_load_resource: libh323_linux_x86_r.so.1.12.2: cannot open shared object file: No such file or directory |
| 18:55.21 | Mike | Mar 18 06:53:48 WARNING[491]: loader.c:391 load_modules: Loading module chan_h323.so failed! |
| 18:55.22 | *** join/#asterisk SPoon_TSX (~SPoon_TSX@d206-116-121-40.bchsia.telus.net) |
| 18:55.23 | buddah | ok, we have a quintum gateway in bangledesh, and when calls go through it, the CLI gets flooded with this message |
| 18:55.24 | buddah | Mar 18 10:59:24 WARNING[20606]: codec_g729.c:196 g729tolin_framein: Invalid data (4 bytes at the end) |
| 18:55.28 | buddah | any clue as to why? |
| 18:55.32 | buddah | is it a VAD thing? |
| 18:55.33 | BrianR___ | aah.. I got bidirectional caller id with name over PRI working by switching from national to dms100.. |
| 18:55.50 | buddah | i'm getting it 4-10 times per second |
| 18:55.54 | buddah | until the call is done |
| 18:56.29 | Drel | Hello all! I am using Asterisk@Home on reasonably fast hardware, and am experiencing approx .5 - .7 second (by ear; I'm not sure how to measure this using software tools) of latency on a call that's between a SIP softphone (Xten Lite) -> 100P FXO -> PSTN. I'm not going out on the net at all. Local network between softphone and X100P is 802.11g. I'm curious what is causing this latency,... |
| 18:56.31 | Drel | ...and how to lower it? |
| 18:56.40 | SPoon_TSX | Hi there, got a quick questions. I got my TDM400 Card installed, but when I try to call via the Zap Channel, it always say |
| 18:56.49 | SPoon_TSX | NOTICE[-1344656464]: app_dial.c:554 dial_exec: Unable to create channel of type 'Zap' |
| 18:57.06 | SPoon_TSX | I have a regular phone line hooked up with the card now. |
| 18:57.54 | SPoon_TSX | Any idea why it happen like that?? |
| 18:58.20 | Drel | Spoon_TSX: Zap modules loaded? |
| 18:58.44 | SPoon_TSX | I loaded it as mobprobe zaptel, then mobprobe wcfxs |
| 18:58.59 | SPoon_TSX | I am using the Stable release of Asterisk. |
| 18:59.00 | Drel | I'm assuming you saw this? -- http://lists.digium.com/pipermail/asterisk-users/2004-November/070952.html |
| 19:00.09 | buddah | any idea on the codec_g729.c:196 error? |
| 19:00.35 | *** join/#asterisk tzanger (~tzanger@165.154.13.35) |
| 19:01.01 | tzanger | I fucking hate that +e |
| 19:01.16 | Qwell | tzanger: xchat is retarded about it |
| 19:01.21 | tzanger | I don't use xchat |
| 19:01.23 | Qwell | it tries to join, then it sends the password |
| 19:01.31 | Qwell | so I have to /j #asterisk manually anyways |
| 19:02.05 | zno | Qwell: I had the same problem, but I thought it was something I did. |
| 19:02.05 | Inv_arp | Qwell: u too? man i thought it was just me |
| 19:02.11 | zno | haha |
| 19:02.11 | Qwell | heh |
| 19:02.14 | Inv_arp | heh |
| 19:02.19 | Qwell | xchat is just stupid anyways |
| 19:02.38 | zno | chatzilla plugin on firefox seems best |
| 19:02.47 | Drel | Anyone experienced similar latency using the Digium X100P? I'm surprised to have it be so noticeable and think it must be a misconfiguration. I'd expect some latency on a net call, but this is just local network to PSTN. |
| 19:03.07 | JerJer[mobile] | digium's X100P or a clone? |
| 19:03.16 | SPoon_TSX | Try to recomplie the asterisk now to see if it works. |
| 19:03.26 | SPoon_TSX | Will stable release run on 2.6.10 kernel?? |
| 19:03.31 | Qwell | SPoon_TSX: sure |
| 19:03.41 | NewSole | Got a question... is there a good accounting package for asterisk out there.... |
| 19:03.57 | SPoon_TSX | Very very weird. But there is no such stable release of Zaptel, right? |
| 19:03.58 | JerJer[mobile] | NewSole: write your own |
| 19:04.01 | Drel | JerJer[mobile]: A clone, I believe, though it was advertised as a "Digium X100P" when I bought it. Are there any known differences with the clones? |
| 19:04.10 | JerJer[mobile] | Drel: most certianly |
| 19:04.16 | JerJer[mobile] | demand a refund |
| 19:04.18 | Qwell | JerJer[mobile]: Can I get a quick clarification on nufone tollfree's? |
| 19:04.42 | JerJer[mobile] | Drel: because you were ripped off |
| 19:04.46 | Qwell | I know its $0.02/minute for incoming/outgoing on a regular line, and it says $0.02 for tollfree...is that combined? |
| 19:04.51 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 19:04.54 | JerJer[mobile] | no |
| 19:05.14 | Qwell | hmm |
| 19:05.19 | JerJer[mobile] | where does it say incomng/outgoing on a regular line? |
| 19:05.20 | Drel | JerJer[mobile]: What is the difference? It was advertised as "Digium X100P OEM", which to me means a Digium card with no retail packaging. |
| 19:05.30 | Qwell | umm, maybe I made that up? lemme refresh my memory |
| 19:05.39 | JerJer[mobile] | no it means they lied to you and sold you an inferior knock off |
| 19:05.44 | SPoon_TSX | -- Executing Dial("SIP/2000-eeeb", "Zap/g1/97788961678|20|t") in new stack |
| 19:05.54 | JerJer[mobile] | Digium has not sold X100Ps in quite a while |
| 19:05.55 | Qwell | oh, it doesn't...I see |
| 19:06.00 | SPoon_TSX | Mar 18 11:06:14 NOTICE[-1344500816]: app_dial.c:554 dial_exec: Unable to create channel of type 'Zap' |
| 19:06.04 | tzanger | Drel: Digium doesn't sell OEM cards. Digium *is* an OEM. I highly doubt they're allowing others to sell using their name for so little |
| 19:06.11 | SPoon_TSX | == Everyone is busy at this time |
| 19:06.15 | SPoon_TSX | Same thing... |
| 19:06.32 | tzanger | Drel: now granted teh X100P was OEMed by Digium but they no longer sell it and htere are countless variants of that specific modem |
| 19:06.33 | JerJer[mobile] | SPoon_TSX: then either that zap channel is in use or not configured properly |
| 19:06.41 | JerJer[mobile] | or the telco is broken |
| 19:07.19 | SPoon_TSX | JerJer: I for sure Telco is not broken because I can hook up a same phone line with my regular phone and dial a number no problem. |
| 19:07.28 | SPoon_TSX | In terms of Zap config.... |
| 19:07.37 | Qwell | JerJer[mobile]: "We can provide you with 2.0 cents per minute for US48" I assume thats outgoing, and that incoming is the same? "We can also provide [...] US48 Toll-Free numbers for only 2.0 cents per minute" |
| 19:07.49 | SPoon_TSX | I got 4 FXO on my TDM400 Card. and my config are like this: |
| 19:07.51 | Drel | tzanger: So, you know that the latency I am experiencing (.5 second or so) is due to using a X100P variant, and nothing else could be the problem? |
| 19:08.00 | tzanger | Drel: it could be any number of things |
| 19:08.03 | tzanger | Drel: it's impossible to tell |
| 19:08.03 | SPoon_TSX | Zaptel.conf: fxsks=1-4 |
| 19:08.16 | doughecka | whats the licensing required for cisco phones? |
| 19:08.20 | Qwell | To me, that reads if I get a call on my tollfree did, it'd cost me $0.04...or something? |
| 19:08.24 | doughecka | do I need a license for each phone to be legal? |
| 19:08.27 | tzanger | easy way out -- try the same card in a different system. Pull the HDD and card out and put them in anothe rsystem. Linux kicks ass for that. If the delay is stranger or gone or osmething it's the interaction of the system |
| 19:08.30 | SPoon_TSX | Zapata.conf: [channels] |
| 19:08.30 | SPoon_TSX | signalling=fxs_ks |
| 19:08.30 | SPoon_TSX | echocancel=yes ; You can set this to 32, 64, or 128, tweak to your needs. |
| 19:08.30 | SPoon_TSX | echocancelwhenbridged=yes |
| 19:08.30 | SPoon_TSX | echotraining=400 ; Asterisk trains to the beginning of the call, number is in milliseconds |
| 19:08.30 | SPoon_TSX | callerid=asreceived |
| 19:08.32 | SPoon_TSX | group=1 |
| 19:08.34 | SPoon_TSX | context=pstn ; Points to the default context of your extensions.conf |
| 19:08.36 | SPoon_TSX | channel => 1-4 ; Again X is the number of FXO modules you have |
| 19:08.54 | SPoon_TSX | Any one can tell me if I am doing something wrong on those file? |
| 19:08.54 | Drel | tzanger: Any tips for trying these other things? Something tells me it will be extremely difficult to get a refund without the shipping, restocking, whatever fees costing more than the card. |
| 19:09.18 | JerJer[mobile] | Qwell: reload |
| 19:09.38 | JerJer[mobile] | Drel: demand one |
| 19:09.41 | SPoon_TSX | btw, when I typ Zap Show Channels, I got no channel showing on my screen... is that normal? |
| 19:09.50 | JerJer[mobile] | you were lied to |
| 19:09.50 | Qwell | now, I saw it change...but what changed exactly? |
| 19:09.58 | JerJer[mobile] | blantely |
| 19:10.04 | JerJer[mobile] | without a doubt |
| 19:10.20 | JerJer[mobile] | Qwell: the statement u were questioning |
| 19:10.37 | Qwell | ahh, "for calls to" |
| 19:10.49 | Drel | JerJer[mobile]: Well, I will try. :-) What SHOULD I be looking for, for a cheap home FXO solution? I basically followed the advice/document of a guy who posted a Asterisk Howto, using Asterisk@Home + a Digitum OEM card off ebay. |
| 19:10.50 | eKo1 | SPoon_TSX: No it isn't. |
| 19:11.15 | Qwell | JerJer[mobile]: So basically, if I don't have a tollfree did, outgoing will be $0.02, and incoming will be...$0, because its impossible for me to get calls? |
| 19:11.15 | tzanger | cheap home FXO? Sipura perhaps? TDM11P isn't bad for price either |
| 19:11.21 | JerJer[mobile] | Qwell: reload agan |
| 19:11.36 | Qwell | I'm slow this morning. |
| 19:11.36 | SPoon_TSX | eKo1: You mean I SHOULD see all 4 channels even they are no in use right? |
| 19:11.58 | eKo1 | SPoon_TSX: Yes. |
| 19:11.58 | JerJer[mobile] | Qwell: correct, if you do not have a DID from us nobody can ever call you, thus we can never bill you |
| 19:12.07 | Drel | Crap, I keep typing 'digitum' instead of 'digium', not sure why, my fingers do it by themselves :) |
| 19:12.08 | SPoon_TSX | Damn. What could be wrong then?? |
| 19:12.20 | Qwell | and with a tollfree DID, I wouldn't get charged (more) for outgoing, just incoming... |
| 19:12.24 | Drel | This is the howto I followed: http://geekgazette.com/index.php?option=com_content&task=view&id=2&Itemid=26 |
| 19:12.33 | Qwell | ok, I'll sign up right now. Gonna yell at Verizon in a few hours. |
| 19:12.50 | Drel | "the most recommended card is the Digium Wildcard X100P FXO card which can be purchased brand new on eBay for $6.95 each" |
| 19:13.07 | tzanger | Drel: that is old documentation and blatantly wrong |
| 19:13.12 | Drel | So, you can see why I'm a little frustrated :) |
| 19:13.15 | tzanger | Drel: geekgazette should be shot |
| 19:13.16 | Qwell | JerJer[mobile]: I must say, from what I've seen elsewhere, thats a pretty good rate. |
| 19:13.31 | Drel | tzanger: What model Sipura should I be looking for? |
| 19:13.40 | tzanger | Drel: I don't know offhand, I don't touch anything that talks SIP |
| 19:13.41 | Qwell | JerJer[mobile]: Which is the only reason why I was questioning it...I figured I couldn't have been reading it right. |
| 19:13.56 | Beirdo | Drel: if you want FXO, then you want SPA-3000 |
| 19:14.07 | Beirdo | it has one FXO, one FXS |
| 19:14.45 | Drel | Gah! I just ordered yesterday (and it's in the mail already) a SPA-1001 from Voxilla store. :-( |
| 19:15.08 | Beirdo | that's a single port FXS, no? |
| 19:15.16 | Drel | Yeah. :( |
| 19:15.24 | Beirdo | not that there's anything wrong with that |
| 19:15.34 | Beirdo | it'll be good to hook up another analog phone |
| 19:15.48 | Beirdo | but if you are looking for FXO as well, the -3000 is your toy |
| 19:15.55 | Drel | Well, it's $80 hard earned low-wage dollars :-) I can't be spending another $100 on another piece of hardware. |
| 19:16.11 | Drel | Er, $70, sorry. |
| 19:16.15 | Beirdo | ah. well that's a shame |
| 19:16.45 | *** join/#asterisk brettnem (~mive29@user-0ccsr2l.cable.mindspring.com) |
| 19:16.57 | brettnem | hello all |
| 19:17.01 | Drel | is there anything else that could be the problem with the fxo I currently use? silence detection on the sip softphone? echo cancellation? lack of chicken sacrifice? |
| 19:17.28 | Beirdo | what's the issue? |
| 19:17.51 | Beirdo | other than getting ripped off and getting a knockoff... what are the symptoms of the problem? |
| 19:18.14 | Qwell | Drel: Might I ask how much you paid for the clone? |
| 19:18.16 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
| 19:18.19 | Drel | Beirdo: I currently have a very simple setup. Asterisk@Home with a "OEM" Digium X100P FXO. When I place a call from Xten Lite SIP softphone to a PSTN number across the FXO, there's a .5 second delay. |
| 19:18.40 | Drel | Qwell: $13 including shipping. |
| 19:18.51 | Qwell | yeah... |
| 19:18.52 | JerJer[mobile] | Drel: its not an OEM Digium X100P...it is a clone and a poor one at that |
| 19:19.03 | Qwell | The Digium ones are more like $100 |
| 19:19.07 | JerJer[mobile] | Cavaet Emptor |
| 19:19.09 | Beirdo | Hmm. |
| 19:19.11 | Drel | JerJer[mobile]: See the quotes? ;) |
| 19:19.24 | JerJer[mobile] | but call it really s |
| 19:19.25 | JerJer[mobile] | is |
| 19:19.27 | Beirdo | it's a knockoff :) |
| 19:19.30 | Drel | I guess I should have said "OEM Digium" rather than "OEM" Digium ;) |
| 19:19.42 | JerJer[mobile] | no "CLONE" X100P |
| 19:19.43 | Drel | Beirdo: Yeah, I figured that out :( |
| 19:19.44 | Qwell | OEM "Digium" |
| 19:19.49 | Beirdo | just like mine is a knockoff and I make no bones about it. |
| 19:20.00 | Qwell | I bought mine because it was a knockoff ;/ |
| 19:20.04 | SPoon_TSX | When I try to install the Asterisk, should I download the -r stable_1_0 or just go to the most updated snap shot? |
| 19:20.23 | Drel | Beirdo: I didn't realize it was a knockoff when I ordered it. I thought it was a new OEM card. Live and learn. |
| 19:20.35 | Beirdo | well, you were lied to then |
| 19:20.42 | Nugget | stable_1_0 is not a valid cvs tag. |
| 19:20.44 | Juggie | how hard is it to build a modem |
| 19:20.46 | Juggie | it cant be that bad |
| 19:21.08 | Qwell | JerJer[mobile]: I assume I'll get instructions on how to order a DID after I click submit here? |
| 19:21.16 | SPoon_TSX | sorry, I mean -r v1-0_stable |
| 19:21.18 | Beirdo | anyways, the 0.5s delay may be a codec translation thing |
| 19:21.26 | Beirdo | !seen Shido6 |
| 19:22.02 | sivana | ~seen shido6 |
| 19:22.15 | jbot | shido6 is currently on #asterisk |
| 19:22.15 | Drel | Beirdo: I guess so. I found a howto that got me excited about trying Asterisk (http://geekgazette.com/index.php?option=com_content&task=view&id=2&Itemid=26), that said that new oem cards were avialable on ebay for $7 or so, I looked on ebay, and sure enough, "oem digium x100p" cards were listed for that price. I'm not complaining, because I probably wouldn't have tried playing around with... |
| 19:22.15 | Drel | ...this if the barrier to entry was much higher. |
| 19:22.26 | Drel | Beirdo: Codec translation thing? Please tell me more... |
| 19:22.52 | Beirdo | well, if your asterisk box needs to translate from one codec to another, it takes finite time |
| 19:22.57 | Beirdo | show translation |
| 19:23.13 | Beirdo | will show you the amount of time from each codec to each of the others |
| 19:23.21 | stdio | can someone take a look at my zapata.conf file, and tell me why my fxs module isn't working correctly? |
| 19:23.24 | brettnem | hey anyone know about any problems getting SNOM's to keep their registrations? |
| 19:23.28 | Drel | Beirdo: Ah, I see. What should I be looking for, and any tips on what I can change it to? |
| 19:23.32 | *** join/#asterisk peted20 (~chatzilla@d2-168.rb.gh.centurytel.net) |
| 19:24.09 | Drel | Beirdo: Do you think this would be an issue on an Athlon 1800+ processor + Linux 2.4? |
| 19:24.14 | Beirdo | well, figure out what codec your SIP phone is using, and find out how long it takes to translate to/from that and ulaw |
| 19:24.26 | jesster | to add another class into musiconhold.conf do I just do a: name => mp3:/path/to/mp3/dir ? |
| 19:24.39 | Beirdo | there is a certain amount that is inevitable and you can't get rid of |
| 19:24.47 | brettnem | argh.. my snom 190 isn't re-registering with authentication! |
| 19:25.00 | jesster | i have a 'default' in there, i want a second, seperate mp3 dir |
| 19:25.21 | Drel | I'm basically trying this out at home with an aim to convince my boss to use Asterisk + VoIP for the small business I work for. |
| 19:26.05 | Drel | I know he won't go for noticeable latency, so I'm definitely crossing X100P clone off the list of acceptable hardware ;) |
| 19:26.53 | jesster | quit |
| 19:26.55 | jesster | oops ;) |
| 19:27.13 | *** join/#asterisk sleepy_one (~chatzilla@dhcp16632045.neo.rr.com) |
| 19:27.14 | Beirdo | for sure, but don't blame the hardware yet |
| 19:27.17 | sleepy_one | hey all |
| 19:27.24 | Drel | Beirdo: I appreciate the tips, I will definitely check out the translation settings. |
| 19:27.38 | Beirdo | no problem |
| 19:28.07 | brettnem | interesting.. the 401 isn't making it back to my phone.. * is using the rport to send the packet back.. |
| 19:29.53 | SPoon_TSX | Do I need an extension for my Zap channel in order to make a out going call? |
| 19:30.15 | Drel | We are currently using analog Polycom "SoundPoint Pro" telephones here at work, that kind of simulate a PBX (badly). Does anyone have any experience using these phones in an Asterisk setup? Is it asking for trouble? |
| 19:31.16 | sleepy_one | I get "ZT_SPANCONFIG failed on span 1: No such device or address (6) |
| 19:31.18 | sleepy_one | FATAL: Error running install command for wct1xxp" |
| 19:31.32 | sleepy_one | when modprobe wct1xxp on FC3_64 running 2.6.9.1 |
| 19:32.27 | sleepy_one | any ideas? I already added lines to /etc/udev/......./ etc and even mknod'ed /dev/zap/1 - 256 by hand |
| 19:33.50 | *** join/#asterisk jtodd (~jtodd@garthim.fox-den.com) |
| 19:34.18 | *** join/#asterisk frizlfry (~blah@adsl-66-120-140-19.dsl.lsan03.pacbell.net) |
| 19:34.33 | frizlfry | anyone here? quick question... is there a way to use G.728 w/ the asterisk server? |
| 19:35.19 | Qwell | ack |
| 19:35.33 | Qwell | JerJer[mobile]: "ERROR: No Toll-Free numbers available. Please try again later." |
| 19:35.38 | tzanger | frizlfry: of course... write the g728 codec .so |
| 19:36.40 | frizlfry | well i'm not familiar w/ asterisk and how all that works. a customer of ours wants us to change our codec to G.729 or GSM but it's already made w/ G.728. what do you mean by write the codec? how is it integrated with * |
| 19:37.08 | *** join/#asterisk Goshen (~Goshen@70-57-80-147.slkc.qwest.net) |
| 19:37.29 | *** join/#asterisk AvengerX (~h_avenger@200.216.189.251) |
| 19:38.14 | *** join/#asterisk johnnyb (~johnnyb@sdsl-38-17-139.tulsaconnect.com) |
| 19:39.53 | *** part/#asterisk mkhan (~mkhan@ip66-105-190-122.z190-105-66.customer.algx.net) |
| 19:40.17 | JerJer[mobile] | Qwell: yep |
| 19:40.31 | Qwell | I guess I should be looking at vanity numbers then? |
| 19:40.51 | *** join/#asterisk brettnem (~mive29@user-0ccsr2l.cable.mindspring.com) |
| 19:40.52 | brettnem | grr |
| 19:40.53 | JerJer[mobile] | or just be patient |
| 19:41.10 | Qwell | yeah, I don't even have my hardware yet...I can wait I guess |
| 19:41.36 | JerJer[mobile] | people keep provisioning numbers from us then never using them |
| 19:41.43 | JerJer[mobile] | guess its time to deploy the monthly fee |
| 19:41.46 | Qwell | Thats dumb |
| 19:41.55 | Goshen | Or just a setup fee |
| 19:42.07 | Goshen | that discourages people from signing up for the number if they are not serious |
| 19:42.19 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
| 19:42.19 | Qwell | setup fee unless xyz minutes used per month? |
| 19:42.23 | Qwell | dunno, just throwing out ideas. :p |
| 19:42.52 | Qwell | JerJer[mobile]: Just out of curiousity, how many customers do you guys have? |
| 19:43.08 | tzanger | ONE POINT TWENTY ONE GIGGAWATTS!! |
| 19:43.09 | tzanger | oh |
| 19:43.11 | tzanger | I mean... |
| 19:43.12 | tzanger | :-) |
| 19:43.18 | Qwell | jiggawatts :p |
| 19:43.29 | PBXtech | does this packet8 video phone stream the video through packet8? |
| 19:43.40 | jontow | oh yeah? :/ i have a servor at my house! |
| 19:43.56 | *** join/#asterisk sysdebug (~jonasgoes@200.163.193.247) |
| 19:44.23 | JerJer[mobile] | Qwell: over 3,000 |
| 19:44.27 | Qwell | nice |
| 19:44.30 | JerJer[mobile] | haven't counted lately |
| 19:44.39 | PBXtech | jerjer what mobile phone you using? |
| 19:45.08 | *** part/#asterisk JerJer[mobile] (~jj@feth100-fw.fament.net) |
| 19:45.13 | *** join/#asterisk JerJer[mobile] (~jj@feth100-fw.fament.net) |
| 19:45.26 | JerJer[mobile] | 6620 nokia |
| 19:45.31 | JerJer[mobile] | i'm on wifi at the moment |
| 19:46.15 | PBXtech | thats got to be hard. i use treo650 and thats slow enough |
| 19:46.36 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 19:48.04 | PBXtech | who is running video through *? |
| 19:49.00 | johnnyb | Who is the best provider to get (1) 800 numbers through, or (2) regular phone numbers, of which you can receive an unlimited number of calls at a time? |
| 19:49.30 | JerJer[mobile] | best is subjective |
| 19:49.41 | greg_work | johnnyb: look on voip-info.org, theres some pages on voip providers |
| 19:49.47 | Jer13261 | fioding that kind of provider is going to be hard |
| 19:50.00 | JerJer[mobile] | PBXtech: if i am in an EDGE area its not too bad... lke 200ms latency to my switch-1 |
| 19:50.08 | CosmicRay | johnnyb: livevoip.com looks to be the cheapest for 800 numbers; I'll probably try to hook up with them for that and outgoing termination this weekend |
| 19:50.23 | CosmicRay | johnnyb: their 800 rates are something like $1/mo and 1.27 cents per minute |
| 19:50.44 | greg_work | CosmicRay: i JUST setup an iax termination account with them |
| 19:50.48 | Beirdo | I'm sure there's a catch though, there always is |
| 19:51.00 | brc_ | ~tell johnnyb about nufone |
| 19:51.01 | greg_work | it works, but so far i've only used it for one call so :) |
| 19:51.12 | johnnyb | CosmicRay: I'm curious -- has anyone ever tried using VOIP for doing simple calling card mechanisms? |
| 19:51.14 | Beirdo | as JerJer[mobile] said... best is subjective :) |
| 19:51.15 | greg_work | Beirdo: yes, $29 minimum purchase |
| 19:51.16 | tzanger | johnnyb: I use jerjer's company, nufone. it's per-minute so he don't care whether you take 1 call or a thousand calls simultaneously |
| 19:51.18 | greg_work | per DID |
| 19:51.31 | CosmicRay | JerJer[mobile]: nufone is your company? |
| 19:51.31 | greg_work | as in, if you want 3 800 #'s, you have to pay $29 * 3 |
| 19:51.35 | JerJer[mobile] | yes |
| 19:51.42 | Beirdo | greg_work: and if you want outbound calling, gotta put in another $29 minimum, IIRC |
| 19:51.47 | CosmicRay | JerJer[mobile]: it sounds great *but* there have been quite a fwe complaints about hard to reach support lately |
| 19:51.57 | Qwell | JerJer[mobile]: I just checked...I get some pretty good latency to both of your switches. nice |
| 19:52.14 | CosmicRay | JerJer[mobile]: anything to that? |
| 19:52.30 | greg_work | JerJer[mobile]: you should put a couple of minutes into putting some info on your website ;P |
| 19:52.34 | JerJer[mobile] | CosmicRay: says who? a pissed off ex-customer that didn't bother to read the website before ordernig a number? |
| 19:52.39 | greg_work | Beirdo: yes |
| 19:52.49 | JerJer[mobile] | greg_work: there is a lot of info on there |
| 19:52.58 | Beirdo | that there is :) |
| 19:53.00 | greg_work | Beirdo: i think they just don't want people to sign up unless they actually want service |
| 19:53.00 | JerJer[mobile] | more than enough for informed consumers |
| 19:53.03 | *** join/#asterisk husher (~andrew@68.143.92.130.nw.nuvox.net) |
| 19:53.17 | Qwell | greg_work: jerjer was kind enough to update his site a few moments ago, for me...heh |
| 19:53.26 | Qwell | greg_work: But I'm an idiot though, so... |
| 19:53.34 | greg_work | JerJer[mobile]: where? :P theres one page and a signup page |
| 19:53.46 | JerJer[mobile] | what more do you need? |
| 19:54.02 | Beirdo | :) |
| 19:54.06 | JerJer[mobile] | plus don't forget the crappy Let me in pages |
| 19:54.20 | JerJer[mobile] | if u want pretty look here: http://ww2.nufone.net |
| 19:54.22 | mikegrb | LET ME IN! |
| 19:54.25 | JerJer[mobile] | but don't expect much |
| 19:54.34 | *** join/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.rr.com) |
| 19:54.37 | Qwell | ooo, nice |
| 19:55.16 | greg_work | lol.. i cant think of what i was looking for, but a while ago i didnt sign up because it was asking me to enter my credit card # before I even found what i was looking for |
| 19:55.26 | greg_work | i realize thats not very helpful though :p |
| 19:55.43 | CosmicRay | JerJer[mobile]: http://lists.digium.com/pipermail/asterisk-users/2005-March/094749.html was the thread that concerned me most, I guess |
| 19:56.13 | JerJer[mobile] | ahh yes... slepp gave me 10:1 odds that he wouldn't post somethng |
| 19:56.15 | JerJer[mobile] | he lots |
| 19:56.17 | JerJer[mobile] | lost |
| 19:56.26 | Darwin[laptop] | is there going to be a Asterisk Certification class |
| 19:56.26 | bjohnson | slepp? |
| 19:56.36 | brc_ | Darwin[laptop], hahahahah |
| 19:56.38 | brc_ | no. |
| 19:56.43 | *** join/#asterisk jgaviria (~jgaviria@201.245.164.174) |
| 19:56.58 | Darwin[laptop] | I want to be asterisk certified rep/installer/repairtech |
| 19:57.06 | CosmicRay | bah. |
| 19:57.07 | brc_ | okay |
| 19:57.09 | brc_ | step 1 |
| 19:57.13 | brc_ | go to office max |
| 19:57.17 | Mcwebtree | inv_arp I take it you didn't get a chance to look at my conf file? |
| 19:57.20 | bjohnson | Darwin[laptop]: hang around here for a while .. you'll get learned. And then you can stick a star to your forehead |
| 19:57.23 | greg_work | JerJer[mobile]: if its any consolation though, there isnt a single voip provider that has a decent website :p |
| 19:57.26 | brc_ | step two buy some certificate stock |
| 19:57.30 | jgaviria | hi, there is a way to get logs for entrys in a dinamic meetme?, for example, get logs for each dinamically room created |
| 19:57.37 | brc_ | step 3 go to kinkos and print yourself a certification |
| 19:57.43 | brc_ | step 4 ??? |
| 19:57.44 | CosmicRay | JerJer[mobile]: I mean seriously, is that kind of response time typical? That's all I want to know. I'm not trying to assert that it is or attack you or anything |
| 19:57.46 | JerJer[mobile] | no its |
| 19:57.46 | brc_ | step 5 PROFIT!!! |
| 19:57.52 | JerJer[mobile] | 1. Sell T-1 boards |
| 19:57.53 | greg_work | hehe |
| 19:57.54 | JerJer[mobile] | 2. ???? |
| 19:57.59 | greg_work | ww2 looks nice, just needs content ;) |
| 19:58.00 | JerJer[mobile] | 3. Take over the world |
| 19:58.06 | brc_ | ah yes, my bad |
| 19:58.06 | CosmicRay | JerJer[mobile]: ooo, digium's business model! |
| 19:58.09 | Darwin[laptop] | but it has to have Marks john handcock on it |
| 19:58.12 | JerJer[mobile] | ?! there is content |
| 19:58.13 | CosmicRay | JerJer[mobile]: ahh, they haven't figured out step 3 yet. |
| 19:58.16 | Darwin[laptop] | and a official seal |
| 19:58.20 | brc_ | <PROTECTED> |
| 19:58.22 | *** join/#asterisk elriah (~jfulcrum@adsl-068-209-198-242.sip.bhm.bellsouth.net) |
| 19:58.45 | Darwin[laptop] | Iaxy needs work |
| 19:58.54 | Darwin[laptop] | I want to see rev 2 of it |
| 19:59.04 | brc_ | hold your horses |
| 19:59.09 | bjohnson | he's whining about 4 days? |
| 19:59.16 | Qwell | JerJer[mobile]: wtf, did he photoshop the screenshot of your page? |
| 19:59.31 | *** join/#asterisk r0d3nt|m (anonymous@soveliss.luniac.com) |
| 19:59.41 | CosmicRay | bjohnson: about 7 days to get a reply to an e-mail, I think |
| 19:59.44 | JerJer[mobile] | Qwell: look for yourself |
| 19:59.49 | Qwell | I am...its hilarious |
| 19:59.53 | elriah | Greets, all. I just received my 25 polycom phones (ip500) and I'm about to build a testing asterisk system. My pbx is built, I can get a softphone connected, but I need to find some documentation on the polycom xml files. I assume all I need is an entry in my sip.conf (per wiki) and then modify one of the xml files for the phone on my ftp server. I have successfully upgraded the phone's bootrom and flash via ftp. Any help |
| 20:00.05 | Qwell | look how its at the EXACT same position. Your text is right-aligned, I'd wager |
| 20:00.07 | JerJer[mobile] | CosmicRay: he neglected to mention that he found email from us in his spam folder |
| 20:00.20 | CosmicRay | JerJer[mobile]: ah. that would explain quite a lot. |
| 20:00.24 | Qwell | but that period isn't anywhere near aligned to the "?" |
| 20:00.36 | CosmicRay | JerJer[mobile]: I'm surprised he wasn't publically larted :-) |
| 20:00.40 | JerJer[mobile] | of course he didn't bother to mention that little tidbit |
| 20:00.48 | Darwin[laptop] | back in a bit have to go pick up the roomie from his work |
| 20:00.57 | AvengerX | what's the name they give for the 'phone numbers' bound to an E1 circuit? |
| 20:01.02 | JerJer[mobile] | people are tired of off topic flame wars on asterisk-users |
| 20:01.16 | JerJer[mobile] | AvengerX: DIDs? |
| 20:01.28 | JerJer[mobile] | or DDIs depending on which side of the big pond you happen to be |
| 20:01.29 | PBXtech | its funny to me to see all this nufone bashing, and yet nufone is here in this channel all the time. they are not the ones hiding. i dont use them as a provider just though it be interesting |
| 20:01.33 | Qwell | I hear ManxPower say something about DDI |
| 20:02.15 | Qwell | PBXtech: well, when I get hardware, I'll be sure to report my findings. I'm sure they'll be great |
| 20:02.22 | CosmicRay | well there is a lot of bashing of everyone. |
| 20:02.26 | *** join/#asterisk chetan (freetibet@24-193-188-21.nyc.rr.com) |
| 20:02.38 | CosmicRay | PBXtech: personally, I love to bash voipjet, because it is clear that they suck jsut by reading their terms of service. |
| 20:02.43 | CosmicRay | I don't even have to create an account. :-) |
| 20:02.50 | JerJer[mobile] | people think that flaming providers is going to stop people from signing up |
| 20:02.56 | Qwell | vonage bashing is fun too |
| 20:03.04 | PBXtech | i have an account with them just because i get 30ms ping to them |
| 20:03.05 | Beirdo | vonage deserves bashing |
| 20:03.14 | JerJer[mobile] | but what they don't realize is they actually push more people to risk 5 or 10 bucks to see how it goes |
| 20:03.23 | CosmicRay | PBXtech: this is why I won't sign up with them: http://lists.digium.com/pipermail/asterisk-users/2005-March/094229.html |
| 20:03.28 | PBXtech | personally i feel you should have a couple IP providers |
| 20:03.31 | Qwell | JerJer[mobile]: "oh, never heard of nufone...wonder how good they are really?" |
| 20:03.58 | JerJer[mobile] | nufone would be cool if the cocksuckers would call me back |
| 20:04.08 | Qwell | heh |
| 20:04.11 | brc_ | you got it man |
| 20:04.11 | buddah | no shit |
| 20:04.17 | buddah | nufone is horrible at CS |
| 20:04.30 | buddah | took me 5 days to get someone |
| 20:04.37 | buddah | err 4 i think |
| 20:05.06 | Qwell | buddah: 3 day weekend? ;] |
| 20:05.33 | buddah | hmm |
| 20:05.42 | buddah | tuesday wednesday thursday count as weekend now? |
| 20:05.46 | buddah | and friday |
| 20:05.47 | JerJer[mobile] | sure |
| 20:05.57 | buddah | i want those weekends ;) |
| 20:06.07 | buddah | i could deal working monday |
| 20:06.10 | buddah | and just monday |
| 20:06.48 | jgaviria | i have a problem, i have random disconnections in a meetme room, any suggestion? |
| 20:07.06 | PBXtech | does anyone know the number of outgoing lines broadvoice limits an account to? |
| 20:07.18 | CosmicRay | PBXtech: I have to keep my landline for my dsl, so I figure to just use it for backup |
| 20:07.28 | CosmicRay | PBXtech: just a hunch here, but have you tried asking broadvoice? :-) |
| 20:07.43 | JerJer[mobile] | buddah: contrary to popular belief you are not the only nufone customer |
| 20:07.48 | buddah | i know this |
| 20:07.49 | PBXtech | yea im on hold, talk about CS issues :) |
| 20:07.57 | CosmicRay | JerJer[mobile]: are you the one that runs that FWD or IAXtel inward call thing throughout michigan? |
| 20:08.28 | JerJer[mobile] | CosmicRay: the clec side of our operation does, yes |
| 20:08.33 | CosmicRay | ah, nice. |
| 20:08.59 | bjohnson | buddah: you're talking to them now |
| 20:09.06 | JerJer[mobile] | buddah: and when random people wire a large sum of cash to us, we tend to assist them first |
| 20:09.11 | JerJer[mobile] | sorry, but that's how it goes |
| 20:09.11 | buddah | i figured |
| 20:09.16 | buddah | yeah i would agree |
| 20:09.26 | buddah | i'd take large accounts WAY before my $5 test |
| 20:09.31 | buddah | WAY WAY |
| 20:09.31 | tzanger | JerJer[mobile]: switch-3's running with trunktimestamps=yes? |
| 20:09.33 | AvengerX | JerJer[mobile]: Thanks! it is! :) |
| 20:09.42 | JerJer[mobile] | tzanger: um |
| 20:09.52 | JerJer[mobile] | no |
| 20:09.53 | tzanger | I'm not trunking to it YET |
| 20:09.55 | tzanger | but should try that soon |
| 20:09.58 | JerJer[mobile] | want it to be yes? |
| 20:10.03 | tzanger | see if that weirdass bug is fixed |
| 20:10.05 | buddah | but it sucks when you are told we'll call you right back about the fix then no call back at all that day or the next |
| 20:10.07 | bjohnson | PBXtech: I think I read something in their terms of service about charging per minute for more than one concurrent call |
| 20:10.23 | Secretive | Hey guys, when I Try to fax something into Asterisk, I get a few errors. |
| 20:10.25 | tzanger | JerJer[mobile]: well it's the last part of the new code :-) it caused issue with me back in the day though, I haven't tried it yet |
| 20:10.27 | Secretive | I mean, no errors. |
| 20:10.30 | bjohnson | but I've lost track of which conditions came from which ToS |
| 20:10.38 | Secretive | But the fax that I'm sending from says 'No fax answered' |
| 20:10.38 | Secretive | -- Executing Macro("IAX2/teliax@teliax/1", "faxreceive") in new stack |
| 20:10.39 | Secretive | -- Executing SetVar("IAX2/teliax@teliax/1", "FAXFILE=/var/spool/asterisk-fax/1111176877.2139.tif") in new stack |
| 20:10.39 | Secretive | -- Executing SetVar("IAX2/teliax@teliax/1", "EMAILADDR=faxteam@successfulhosting.com") in new stack |
| 20:10.40 | Secretive | -- Executing RxFAX("IAX2/teliax@teliax/1", "/var/spool/asterisk-fax/1111176877.2139.tif") in new stack |
| 20:10.46 | Secretive | But that's what Asterisk says. |
| 20:10.52 | Qwell | hmm |
| 20:11.01 | JerJer[mobile] | tzanger: ok its =yes now |
| 20:11.02 | JerJer[mobile] | we' |
| 20:11.03 | JerJer[mobile] | ll |
| 20:11.05 | JerJer[mobile] | need a peer |
| 20:11.07 | JerJer[mobile] | for you |
| 20:11.37 | PBXtech | speaking of spandsp is it possible to when it gets a bad fax to delete the tif? |
| 20:11.37 | tzanger | JerJer[mobile]: 165.154.13.13 |
| 20:11.39 | Qwell | If I have two wildcard extensions that would match a given number... How does * know which one to pick? They're both in different contexts, included from default |
| 20:11.42 | JerJer[mobile] | ok just reg - it was already setup :) |
| 20:11.57 | CosmicRay | Qwell: the one that is included first, I believe. |
| 20:12.08 | tzanger | JerJer[mobile]: ok |
| 20:12.10 | Qwell | perfect, thanks |
| 20:12.17 | JerJer[mobile] | CosmicRay: no - it is all loaded into memory |
| 20:12.58 | Qwell | oh, not perfect then? |
| 20:13.16 | CosmicRay | JerJer[mobile]: am I missing something? this page appears to disagree with you, to me at least: http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf%20sorting |
| 20:13.34 | Qwell | basically, say I have a normal provider, and iaxtel...if I dial 1700, I don't want my provider to pick it up |
| 20:13.55 | tzanger | Qwell: so have the 1700 stuff in a context and include= it before your normal telco stuff |
| 20:14.16 | Qwell | tzanger: I'm getting conflicting answers. :p |
| 20:14.32 | PBXtech | is there a cheep place to get a wisip phone? |
| 20:14.33 | bjohnson | Qwell: try both |
| 20:14.38 | bjohnson | PBXtech: no |
| 20:14.41 | tzanger | PBXtech: you don't want a wisip |
| 20:14.46 | PBXtech | why not |
| 20:14.56 | tzanger | they suck diseased goat ass |
| 20:15.00 | *** join/#asterisk StealthMethod (~nelsonx@adsl-070-148-141-009.sip.mia.bellsouth.net) |
| 20:15.01 | bjohnson | nice |
| 20:15.05 | StealthMethod | hello |
| 20:15.10 | Qwell | I could just show dialplan, and see what it thinks |
| 20:15.24 | CosmicRay | the fact that there are only about 3 models on the market and they start at $250 should be an indication to you :-) |
| 20:15.28 | PBXtech | phone is just crap? is there any other wifi phone out? |
| 20:15.31 | bjohnson | Qwell: it doesn't delve into the includes .. but go ahead and try it |
| 20:15.39 | Qwell | oh |
| 20:15.46 | bjohnson | try it |
| 20:15.49 | bjohnson | I dare you |
| 20:15.49 | Qwell | did |
| 20:15.54 | CosmicRay | heh |
| 20:16.13 | StealthMethod | farely new to asterisk, looking for help with extensions.conf file, i need to record outbound calls somehow, can any1 help |
| 20:16.14 | bjohnson | goaded |
| 20:16.22 | Qwell | bjohnson: cannot |
| 20:16.25 | Beirdo | Mmm. wisip would be cool if it didn't cost so damn much |
| 20:17.08 | PBXtech | that cheep for a phone |
| 20:17.09 | CosmicRay | StealthMethod: there are recipes for that in the wiki |
| 20:17.17 | StealthMethod | kewl |
| 20:17.18 | StealthMethod | ill look |
| 20:17.19 | StealthMethod | thanx |
| 20:17.25 | CosmicRay | no prob |
| 20:18.16 | tzanger | Beirdo: no, it would be cool if a) the display weren't crap b) the battery life weren't crap c) the phone could handle encryption without bogging down d) the configuration interface were better e) the charging station had SOME kind of positive feedback so you could tell it was securely connected f) the keylock and most functions didn't have HUGE wait times associated with them g) you had ANY control over the soft buttons (MGCP-like?) h) the phone didn't g |
| 20:18.23 | tzanger | Beirdo: need I go on? |
| 20:18.34 | Beirdo | hehe |
| 20:18.43 | Beirdo | the *concept* is cool |
| 20:18.44 | bjohnson | I think it would be cool if 17" LCD monitors were < $50 |
| 20:18.56 | Beirdo | and it would be cool if it used 802.11g |
| 20:19.18 | bjohnson | and exactly where did those parachuting ladies go? Now THAT was cool !! |
| 20:19.28 | PBXtech | samsung made a g version |
| 20:20.01 | NetOfSickCoder | a domain is necesary for the asterisk work? |
| 20:20.20 | husher | PBXtech: stay the hell away from the ZyXel wifi phones |
| 20:20.23 | PBXtech | his g spot? |
| 20:20.40 | *** join/#asterisk riksta (~rick@81-178-200-105.dsl.pipex.com) |
| 20:20.48 | Beirdo | heya, riksta |
| 20:22.09 | Qwell | yeah, looks like it works on the order of the includes |
| 20:24.03 | slePP | JerJer[mobile]: no no. 10:1 against. i don't think anyone bet on those odds. |
| 20:24.43 | riksta | hey Beirdo |
| 20:25.35 | Pinhole | When an absolute timeout happens, why does the channel remain open? |
| 20:25.39 | Qwell | 7 digit dialing within a specific area code could be as easy as - exten => _NXXXXXX,1,Dial,IAX2/something/1234${EXTEN} - right? |
| 20:25.58 | *** join/#asterisk jaxxan (~jaxxan@202.70.125.109) |
| 20:26.39 | Weezey | N? |
| 20:27.04 | Weezey | Qwell, what's N do? |
| 20:27.05 | Nugget | Qwell: correct. |
| 20:27.12 | Nugget | Weezey: what's google do? :) |
| 20:27.26 | PBXtech | N is 2-9 isnt it? |
| 20:27.35 | *** join/#asterisk p0lar (~p0lar@64.254.225.62) |
| 20:27.48 | Nugget | N and X are adequately documented. surely there's no room for speculation on the subject. |
| 20:27.51 | Qwell | I should 911'ify my dialplan |
| 20:28.13 | PBXtech | speculation hmm |
| 20:28.15 | p0lar | If I've got a soekris VPN4801 with a TDM400 and 3 FXO ports, can I use one of the FX0s as a modem interface for dial-up inet access? |
| 20:28.29 | Nugget | p0lar: no |
| 20:28.32 | p0lar | d'oh |
| 20:28.35 | p0lar | Nugget: thanks |
| 20:28.50 | p0lar | But.. I could use one of the com ports for it. ;) |
| 20:28.56 | p0lar | Thanks even more. :D |
| 20:29.15 | PBXtech | someone wrote a fax driver, maybe someday there will be a modem driver :) |
| 20:29.32 | Nugget | bear in mind that you won't be able to run FXS in that setup, unless you have some other plan for getting power in there. |
| 20:29.56 | Nugget | dunno what power supply you were planning to use. I've only used soekris with cf card storage |
| 20:30.16 | Nugget | the tdm400p card will want a molex power connector plugged into it, I think just for fxs support though |
| 20:30.27 | tzanger | Nugget: correct, FXS only |
| 20:32.49 | BuckRogers | does any one have any companies they could recommend for online credit card acceptance? |
| 20:33.35 | StealthMethod | CosmicRay: found something in wiki about monitor application, but seems like i would have to do every time, can it be automated to record outbound everytime... |
| 20:34.02 | StealthMethod | without having to execute command for each channel |
| 20:35.50 | *** part/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.rr.com) |
| 20:35.58 | p0lar | nah, I wno't need FXS, strictly FXO |
| 20:36.04 | p0lar | or are you talking FXS signalling? |
| 20:36.15 | p0lar | No, the power comes on the line with FXS signalling |
| 20:36.17 | Nugget | no, I mean the FXS modules. |
| 20:36.19 | p0lar | yeah |
| 20:36.23 | p0lar | I should be good |
| 20:36.44 | p0lar | console on com1, external modem on com2, 2 fxo ports via TDM400p |
| 20:36.46 | p0lar | yeah, all set.. :D |
| 20:36.54 | p0lar | time to order..hehe |
| 20:37.08 | *** join/#asterisk bile_one (~bile_one@pcp03281999pcs.gillst01.ar.comcast.net) |
| 20:37.15 | *** join/#asterisk Milligan (~noc@nas-05.mtbg-noc.valuelinx.net) |
| 20:37.30 | *** part/#asterisk Milligan (~noc@nas-05.mtbg-noc.valuelinx.net) |
| 20:37.40 | *** join/#asterisk eKo1 (~bernd@63.245.57.70) |
| 20:37.51 | *** join/#asterisk file[laptop] (~file@mctn1-7919.nb.aliant.net) |
| 20:42.21 | buddah | hah |
| 20:45.35 | heison | StealthMethod: can you not do it in a macro? |
| 20:46.20 | StealthMethod | found script |
| 20:46.24 | StealthMethod | wiki |
| 20:48.16 | CosmicRay | StealthMethod: thought so :-) |
| 20:48.46 | StealthMethod | yeah |
| 20:48.50 | StealthMethod | thanx |
| 20:49.49 | CosmicRay | it really is an excellent wiki |
| 20:50.05 | tzanger | dammit there seems to perpetually be 2 calls on this systme |
| 20:50.29 | *** part/#asterisk didz_ (didz_@200.218.192.52) |
| 20:51.32 | doughecka | ~seen atacomm |
| 20:51.35 | jbot | atacomm <~dan@69.54.45.98> was last seen on IRC in channel #asterisk, 43d 18h 54m 42s ago, saying: 'anyone want a IP 3000 conference phone? looking to replace ours with a IP 4000 model. Barely been used, in great condition.... looking for around $500'. |
| 20:52.09 | *** join/#asterisk criptos (~criptos@201.129.126.24) |
| 20:52.24 | *** join/#asterisk DrRighteous (~DrRighteo@ool-182c867b.dyn.optonline.net) |
| 20:52.56 | DrRighteous | Anyone around who is running * into Cisco 53/54/58 xx series ?? |
| 20:54.37 | bannerman | uh.. any voipjet problems today? |
| 20:54.58 | bannerman | when I try to dial out I get a sound best described as "reeeeeeEEEEEEEEEEEEEN <chka chka chka chka chka> reEEEEEEEEEEEEEN" |
| 20:54.59 | tzanger | bannerman: well if you're asking I'd suspect the answer is 'yes' |
| 20:55.27 | bannerman | well, *I* am having voipjet problems today.. the question is whether I screwed something up, or they did :-P |
| 20:55.31 | Essobi | sounds like feedback |
| 20:55.38 | Essobi | bannerman speakerphone? :) |
| 20:55.43 | bannerman | Essobi: nah |
| 20:55.50 | Essobi | DrRighteous I am. |
| 20:55.57 | bannerman | I haven't made any configuratino changes, and it's not a feedback sound, its' electronically generated sounding |
| 20:56.03 | bannerman | with like.. warbles and stuff |
| 20:56.06 | Essobi | nice |
| 20:56.09 | bannerman | it'd be cool if I wasn't trying to make a phone call |
| 20:56.27 | Essobi | Makes you all nostalgic for jimmi hendrix ehh? |
| 20:56.43 | DrRighteous | Essobi: what model, what kind of TDM interconnect? |
| 20:57.02 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
| 20:57.29 | PBXtech | can i send voicemail attachments to 2 emails? instead of one? the second entry is for a pager |
| 20:57.30 | *** join/#asterisk marc32344 (~marc32344@69-90-241-15.dsl.teksavvy.com) |
| 20:57.58 | mikegrb | send it to an alias which forwards to the two addresses |
| 20:58.21 | PBXtech | awe good idea |
| 21:00.14 | *** join/#asterisk bah (048830696@AC95E1AF.ipt.aol.com) |
| 21:00.34 | bjohnson | can the pager read the attachment? |
| 21:00.42 | Inv_arp | ~seen eric_ |
| 21:00.44 | jbot | eric_ is currently on #asterisk |
| 21:00.48 | bannerman | I guess it's just one phone doing that.. odd |
| 21:02.42 | *** join/#asterisk GBAGAMEST (~PIMPER@69-168-111-27.sbtnvt.adelphia.net) |
| 21:03.33 | GBAGAMEST | Can I use 2 Digium Digium Wildcard X100P modems for 2 phone lines |
| 21:03.46 | Qwell | sure, why not? |
| 21:04.15 | *** join/#asterisk JerJer[mobile] (~jj@feth100-fw.fament.net) |
| 21:04.17 | GBAGAMEST | Will it work ok with Asterisk@home? |
| 21:04.32 | Qwell | if asterisk@home is worth its salt, sure |
| 21:04.48 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 21:04.52 | JerJer[mobile] | why not learn asterisk yourself? |
| 21:04.57 | Qwell | I think I just mixed two things there... |
| 21:05.03 | JerJer[mobile] | don't let someone elses crappy packaging fool you |
| 21:05.15 | Essobi | lol |
| 21:05.16 | ariel_ | Good afternoon everyone. |
| 21:05.16 | GBAGAMEST | I want to learn the basics and see what it can do |
| 21:05.29 | GBAGAMEST | it is only for my brothers office |
| 21:05.40 | GBAGAMEST | for a small business |
| 21:05.40 | Qwell | if my brother said that, I'd smack him :p |
| 21:05.42 | Essobi | JerJer[mobile] I got 4 cores from H323 and confereces.. and they are repeatable on my install. ;) |
| 21:05.52 | JerJer[mobile] | good for you |
| 21:06.01 | JerJer[mobile] | want a cookie? |
| 21:06.05 | Essobi | Yup. |
| 21:06.10 | GBAGAMEST | I want to keep it simple for him |
| 21:06.47 | tzanger | hash brownie |
| 21:07.14 | GBAGAMEST | what is good inexpensive sip phone wired/wireless |
| 21:07.42 | GBAGAMEST | or would I be better off using softphones |
| 21:08.21 | Qwell | I need to get me an FXS or something. This echo from iaxcomm is crap |
| 21:08.37 | Qwell | I call my cell, I hear myself like 1/2 second later |
| 21:08.48 | Inv_arp | GBAGAMEST: i prefer hardphone i use handytone 486 to connect to my normal cordless, i hear supuras are good ... |
| 21:09.03 | GBAGAMEST | thanks Inv_arp |
| 21:09.35 | Darwin[laptop] | ok bsd needs some nettv streaming apps |
| 21:10.12 | GBAGAMEST | how about using a softphone client on a wireless Pocket PC |
| 21:10.19 | GBAGAMEST | anyone try that? |
| 21:10.22 | *** join/#asterisk lespiggot (~les@cpc1-ersk1-5-1-cust28.renf.cable.ntl.com) |
| 21:10.45 | Inv_arp | GBAGAMEST: dont see why it shouldnt work... if the software is avail |
| 21:12.02 | GBAGAMEST | can I program asterisk to detect a incoming fax is their a good FAQ on this |
| 21:12.10 | GBAGAMEST | sorry to sound like a noob |
| 21:13.08 | bannerman | I changed the default codec on my phone to ulaw and it works fine ow |
| 21:13.20 | bannerman | default used to be gsm, thought it would automagically switch |
| 21:13.20 | Weezey | I'm having trouble with codecs. 711u, should sound perfect, right? callers complain about echoey sound. I've narrowed it down to either being the codec or my SPA3000 endpoint just isn't cutting it. |
| 21:13.59 | *** join/#asterisk neopher (~crazy@mail.techhelpresources.com) |
| 21:14.43 | tzanger | you can start by getting a partition ready with xfs filesystem |
| 21:14.48 | Inv_arp | Weezey: if the bandwidth is there, pstn quality |
| 21:14.50 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
| 21:14.52 | tzanger | oops |
| 21:15.18 | Weezey | inv: hrmm: okay, so then it must be the endpoints then. |
| 21:15.26 | Weezey | then. |
| 21:15.48 | Weezey | I suppose it could be the phones too. |
| 21:17.51 | lespiggot | hi, can anyone explain to me how the sip interface works out the IP addresses it binds to please, on openwrt (experimental) SIP is disabled as it cannot verifty IP addresses of its interfaces |
| 21:19.17 | Inv_arp | lespiggot: ?works out the IP addresses it binds |
| 21:19.19 | brc_ | Four words: Jar-Jar in 3D |
| 21:19.19 | brc_ | "Oh the humanity!" |
| 21:19.32 | *** join/#asterisk mkhan (~mkhan@ip66-105-190-122.z190-105-66.customer.algx.net) |
| 21:19.55 | doughecka | does anyone know what the licenses 'required' by cisco when buying cisco ip phones |
| 21:20.06 | mkhan | hello.. how can learn about telephone wiring on RJ45 .. ? |
| 21:20.20 | lespiggot | Inv_arp, yeah at the moment, asterisk disables the sip interface on my wrt54g |
| 21:20.32 | *** join/#asterisk tessier (~treed@210.245.98.227) |
| 21:20.49 | Qwell | mkhan: google |
| 21:21.12 | neopher | lespiggot: did you update the wrt54g firmware, there was a fix for that |
| 21:21.16 | lespiggot | Inv_arp: not sure how it determines what interfaces to use and what IP addresses that are assigned |
| 21:21.51 | lespiggot | neopher: was there? I'm on the openwrt experimetal build from the 15th |
| 21:22.02 | *** part/#asterisk DrRighteous (~DrRighteo@ool-182c867b.dyn.optonline.net) |
| 21:22.04 | doughecka | password |
| 21:22.18 | doughecka | oh no |
| 21:22.25 | doughecka | gotta change it now |
| 21:23.15 | mkhan | Qwell, didn't have good luck ..or I didn't understand well !! |
| 21:23.27 | neopher | lespiggot: are you talking about the fact the voip data is not passing through the wrt54g? |
| 21:24.46 | neopher | Firmware 3.01.3 |
| 21:24.56 | neopher | Resolves issue with VoIP adapters |
| 21:25.01 | mkhan | Is the color code for telephone wiring on RJ45 same as the color code of network cable wiring with RJ45? |
| 21:25.16 | lespiggot | neopher: no, basically the SIP subsystem cannot initilise as it cannot bind to any IP addresses/interfaces |
| 21:25.49 | lespiggot | neopher: In running asterisk on the wrt54 :) |
| 21:26.01 | Hmmhesays | haha another customer won't talk to me anymore |
| 21:26.02 | Hmmhesays | nice |
| 21:26.04 | *** join/#asterisk buleeahn (~buleeahn@199.89.146.53) |
| 21:26.08 | Weezey | wh-blue blue wh-or or wg-gr gr wh-br br |
| 21:26.08 | neopher | lespiggot: interesting |
| 21:26.15 | lespiggot | sorry typo: I'm running asterisk on the wrt54 |
| 21:26.19 | bjohnson | lespiggot: where did you find an experimental build from the 15? I only see the 13th |
| 21:26.41 | bjohnson | doughecka: try something a little harder to crack .. like 1234 |
| 21:26.50 | lespiggot | bjohnson: 2 secs I'll check, thought it was the 15th |
| 21:27.10 | Weezey | who's running the dev call? |
| 21:27.27 | bjohnson | dev call? |
| 21:28.02 | Weezey | that IAX2 conference yesterday? (that's still going on today) |
| 21:28.13 | bjohnson | man .. those guys are dedicated |
| 21:28.16 | cbachman | lespiggot, I finally got mine to work. I'm using both the internal and external interfaces to pass sip calls back and forth |
| 21:28.43 | buleeahn | Where are good docs about which sound file gets played when someone is transferred to voicemail? |
| 21:28.44 | bjohnson | they're not very active today .. but I guess they must be tired |
| 21:28.59 | Weezey | bjohnson: do you know how they've got that conference configured? |
| 21:29.05 | bjohnson | nope |
| 21:29.21 | Weezey | bummer, sounds great, I'd like to set something like that up to demo at the office. |
| 21:29.46 | Weezey | you listening to him fiddle with his microwave now? |
| 21:30.19 | *** join/#asterisk Juxt (~Juxt@adsl-068-213-216-087.sip.bct.bellsouth.net) |
| 21:30.29 | Juxt | hi |
| 21:30.43 | Juxt | can anyone refer me to a voip company that offers nice toll free origination rates? |
| 21:30.54 | *** join/#asterisk criptos (~criptos@201.129.126.24) |
| 21:30.55 | *** part/#asterisk criptos (~criptos@201.129.126.24) |
| 21:31.08 | lespiggot | bjohnson: youre right, 13th, sorry though it was the 15th :( my mistake |
| 21:31.29 | bjohnson | I'm waiting for a new version. |
| 21:31.39 | *** join/#asterisk criptos (~criptos@201.129.126.24) |
| 21:31.42 | bjohnson | pppoe is supposed to be fixed in the next experimental |
| 21:32.05 | jesster | I have an acd queue setup to have music = 007 where 007 is defined in musiconhold.com as: 007 => mp3:/path/to/dept7/mp3s and when a call comes into that queue, * gives errors that the class 007 doesn't exist, any ideas? |
| 21:33.07 | mkhan | 1 wh/or 2 TxData + |
| 21:33.07 | mkhan | 2 or 2 TxData - |
| 21:33.07 | mkhan | 3 wh/grn 3 RecvData+ |
| 21:33.07 | mkhan | 4 blu 1 |
| 21:33.07 | mkhan | 5 wh/blu 1 |
| 21:33.08 | mkhan | 6 grn 3 RecvData- |
| 21:33.10 | mkhan | 7 wh/brn 4 |
| 21:33.12 | mkhan | 8 brn 4 |
| 21:33.22 | lespiggot | brb |
| 21:33.51 | *** join/#asterisk val_0 (~xxx@69-175-3-68.ventca.adelphia.net) |
| 21:34.16 | spackle | juxt: liveLOIP seems decent for 800 |
| 21:34.33 | *** join/#asterisk kuj (~kuj@c-67-165-241-16.client.comcast.net) |
| 21:34.51 | Inv_arp | woa livevoip finally has miami DID's :) |
| 21:35.38 | Uther_P | anyone know of a program that will redirect udp traffic? |
| 21:35.40 | *** part/#asterisk Remowylliams (~Mare@168.215.138.106) |
| 21:35.48 | *** join/#asterisk newpers (newpers@ip68-3-127-190.ph.ph.cox.net) |
| 21:36.15 | Uther_P | I wanna keep my asterisk box nat'ed, but I need the redirect the sip traffic from the firewall |
| 21:36.17 | bjohnson | iptables |
| 21:36.23 | *** part/#asterisk logicalonline (~logicalon@border.logicalonline.com) |
| 21:36.28 | Uther_P | my firewall uses ipfw |
| 21:36.50 | Uther_P | the closest thing i've seen is the FWD rule, but it doesn't change the destination ip |
| 21:37.28 | bjohnson | with iptbles it is -t nat |
| 21:37.41 | bjohnson | google it |
| 21:37.55 | Uther_P | my firewall doesn't use iptables |
| 21:37.58 | Uther_P | it uses ipfw |
| 21:38.00 | Inv_arp | Uther_P: freshmeat has plently uredir is popular |
| 21:38.07 | Uther_P | cool, thanks |
| 21:38.18 | bjohnson | Uther_P: I don't know ipfw .. I know iptables |
| 21:38.24 | val_0 | do any of you set up PBX commercially? seriouse inquiry |
| 21:38.30 | bjohnson | yes |
| 21:39.25 | criptos | many? :) |
| 21:39.45 | bjohnson | many |
| 21:40.12 | *** join/#asterisk pgray007 (~patgrayjr@ool-43571666.dyn.optonline.net) |
| 21:40.49 | Trepalium | Uther_P: I'm not familiar with BSD firewalling, but I think you need to use natd to configure something like that, not ipfw. |
| 21:41.04 | GBAGAMEST | so does the wrt54g have a GUI interface when using the Asterisk firmware |
| 21:41.06 | sleepy_one | c'yall later :) |
| 21:41.13 | bjohnson | GBAGAMEST: no |
| 21:41.55 | *** join/#asterisk bile_one (~bile_one@pcp03281999pcs.gillst01.ar.comcast.net) |
| 21:42.30 | *** join/#asterisk r0d3nt (anonymous@soveliss.luniac.com) |
| 21:42.56 | jesster | I have an acd queue setup to have music = 007 where 007 is defined in musiconhold.com as: 007 => mp3:/path/to/dept7/mp3s and when a call comes into that queue, * gives errors that the class 007 doesn't exist, any ideas? |
| 21:43.06 | *** join/#asterisk FryGuy (fryguy@c-24-23-19-33.client.comcast.net) |
| 21:45.57 | rephorm | can you have a variable in a switch statement? |
| 21:46.03 | Jer13261 | <PROTECTED> |
| 21:46.05 | Uther_P | bunk |
| 21:46.12 | Uther_P | uredir doesn't seem to work under FreeBSD |
| 21:46.15 | rephorm | e.g. switch => Realtime/${CONTEXT}@extensions |
| 21:46.24 | pgray007 | hello everyone! i am losing my mind trying to find SIP firmware for a cisco 7960... cisco is no help.. anyone have any ideas? Thanks! |
| 21:46.25 | Jer13261 | uredir????? |
| 21:46.28 | rephorm | Jer13261: ok. thanks |
| 21:46.31 | chetan | anyone do h323 to h323? |
| 21:46.37 | Uther_P | I'm sure there is a kernel option for it... ELF binary type "0" |
| 21:47.16 | Uther_P | Jer13261: yes, a udp redirector |
| 21:47.59 | Jer13261 | nad ti doesnt work becuase? |
| 21:48.09 | Uther_P | perhaps "options COMPAT_LINUX" |
| 21:48.21 | Uther_P | Jer13261: because its a linux binary |
| 21:48.31 | Jer13261 | then enabled linux compat |
| 21:48.34 | Uther_P | but I know I can run them... I just need to find the right kernel option |
| 21:48.42 | Uther_P | I just said that.. jeez, you are so helpful |
| 21:49.39 | Weezey | I know I'm missing something really stupid here, I've got app_conference built, but how do I make it part of my asterisk config? |
| 21:49.57 | Jer13261 | put it into your modules dir |
| 21:50.18 | Weezey | even though it's an app, not a module? |
| 21:50.25 | Weezey | or are they all just modules? |
| 21:50.46 | *** join/#asterisk Geo- (~no@h-66-134-200-254.snvacaid.covad.net) |
| 21:50.49 | rephorm | Jer13261: hmm. isn't working. were you responding to me when you said "yes"? |
| 21:51.10 | Jer13261 | yes i was |
| 21:51.26 | rephorm | Jer13261: hmm. ok. let me try one more thing real quick then :) |
| 21:51.55 | *** join/#asterisk malcolmd (~malcolmd@malcolmd.digium.sponsor.pdpc) |
| 21:52.37 | Geo- | has anyone ever been able to get the asterisk portaOne radius authentication working properly for them? |
| 21:53.36 | buddah | ok, so i have 2 quintum gateways here, both passing calls to pakistan, is there a way to set it so that calls from quintum A and B go through one carrier, but if its specifically a 9221 number, AND from gateway B, then it goes to a different carrier? |
| 21:53.49 | buddah | only way i could think is to have gateway B register in with sip |
| 21:53.53 | buddah | and made a context to direct it |
| 21:53.58 | buddah | is there another way to do that? |
| 21:54.46 | *** join/#asterisk agave-txlink (phanop@216.81.43.75) |
| 21:57.42 | bjohnson | don't you already have it registering with sip and connected to a context? |
| 21:57.50 | buddah | yes |
| 21:57.59 | buddah | just curious if that was the best way to do it or not |
| 21:58.06 | *** join/#asterisk madounet (~mad|net@juvenal-3-82-226-155-19.fbx.proxad.net) |
| 21:58.07 | bjohnson | just make a new context and point it at that |
| 21:58.11 | buddah | i have 2 gateways, and one isnt registered, one is |
| 21:58.21 | buddah | and the one that is registered is giving me stupid errors |
| 21:58.25 | buddah | about VAD i think |
| 21:58.32 | buddah | and the configs on both gateways are the same |
| 21:58.34 | bjohnson | if that is your goal, having a separate context is the easiest way to separate calls from sip devices |
| 21:58.35 | buddah | so i dont get why its doing it |
| 21:59.11 | buddah | so i thought i'd try to set up this one the same way in sip, without it registering, but then i didnt know how to direct the 9221 calls |
| 21:59.15 | *** part/#asterisk pgray007 (~patgrayjr@ool-43571666.dyn.optonline.net) |
| 21:59.17 | *** join/#asterisk cripito (~ncripito@68.216.32.186) |
| 22:01.05 | *** join/#asterisk bile_one (~bile_one@pcp03281999pcs.gillst01.ar.comcast.net) |
| 22:01.21 | bjohnson | I suppose you could use exten pattern matching and CID matching .. but having a separate context is easier |
| 22:02.01 | *** join/#asterisk ctooley (~ctooley@rrcs-24-153-228-2.sw.biz.rr.com) |
| 22:02.14 | ctooley | We're having some problems getting voicemail prompts to play correctly. |
| 22:02.19 | *** part/#asterisk buleeahn (~buleeahn@199.89.146.53) |
| 22:02.33 | *** join/#asterisk buleeahn (~buleeahn@199.89.146.53) |
| 22:03.58 | Goshen | How do you control the ring to your sip phone? I want to have one ring for some calls into * and another ring for other calls |
| 22:04.16 | Goshen | the only info I found on the wiki was about incoming distrincive ring on zap |
| 22:05.41 | cripito | hi |
| 22:05.53 | Goshen | hi |
| 22:05.56 | Geo- | hewwo |
| 22:05.59 | cripito | anyone having issues with firefly and the new version of head? |
| 22:06.12 | Juxt | nope mine works great |
| 22:07.40 | *** part/#asterisk Juxt (~Juxt@adsl-068-213-216-087.sip.bct.bellsouth.net) |
| 22:07.56 | spackle | Goshen, some phones and ATA's notable, the Sipura's allow you to set distinctive ring up for certain phone numbers. |
| 22:08.06 | PTG123 | Hey anyone in here have pris in california? |
| 22:08.08 | spackle | er, notably. |
| 22:08.26 | buddah | yeah ptg |
| 22:08.40 | PTG123 | buddah: what part, and you want to sell some channels? |
| 22:08.43 | Goshen | Spackle: I have a sipura, and a grandstream phone |
| 22:08.51 | buddah | PTG123: long beach |
| 22:09.17 | PTG123 | You have local outbound calling on them unlimited? |
| 22:09.51 | Geo- | Has anyone in here been able to work with asterisk to succeed in authenticating sip users via radius? |
| 22:09.56 | sivana | how do I send traffic to a gateway with just an IP? |
| 22:10.04 | sivana | is it exten@ip? |
| 22:10.26 | ctooley | <PROTECTED> |
| 22:10.28 | *** join/#asterisk husher (~andrew@68.143.92.130.nw.nuvox.net) |
| 22:10.37 | ctooley | Why is it playing that instead of unavail or busy? |
| 22:10.48 | spackle | Goshen, there may be another way, hold on a minute. |
| 22:11.00 | *** join/#asterisk Matt-E- (~Matt-E-@66-224-125-137.atgi.net) |
| 22:12.33 | Wonka | something's stuttering in my local s0 bus... |
| 22:12.33 | Goshen | spackle: great |
| 22:13.25 | Goshen | I would like to have one set of phones in the whole house, then when line one rings, it does a normal ring |
| 22:13.46 | Goshen | then if line two rings, it does a ring beep ring or some type of distrinctive ring to let you know it is line 2 |
| 22:13.56 | Goshen | lines being different providers |
| 22:17.03 | modulus_ | goshen are you from indiana? |
| 22:17.41 | Goshen | lol nope |
| 22:17.43 | Goshen | SLC UT |
| 22:18.10 | spackle | Goshen: sorry i can't find the reference now. There was a setvar in the dial plan. I think it set the phone up for a special ring. Anybody else know? |
| 22:19.09 | ariel_ | sivana, all you do is exten => X.,Dial(Sip/IPAddress/${EXTEN}) |
| 22:20.04 | terrapen | SLC |
| 22:20.07 | terrapen | i miss that place |
| 22:20.10 | *** part/#asterisk mkhan (~mkhan@ip66-105-190-122.z190-105-66.customer.algx.net) |
| 22:20.28 | ariel_ | Goshen, how are your phones connected via sip or zap ports? |
| 22:23.16 | spackle | Goshen: here it is: http://www.voip-info.org/tiki-index.php?page=Asterisk%20SIP%20channels see "distinctive ring styles" |
| 22:24.35 | *** join/#asterisk greg_work (~greg@d221-73-198.commercial.cgocable.net) |
| 22:25.36 | Goshen | ariel: I have two cordless phones connected via seperate sipura FXS ports |
| 22:25.39 | johnnyb | It seems that whenever we have a new call come in to asterisk, for a second or two the server loses all of its idle time. Could this be caused by low memory? We're not using any swap, but we're using all but 32M of memory. |
| 22:25.40 | Goshen | and a grandstream sip phone |
| 22:26.11 | mogorman | hey is there away to automatically delete silent messages in asterisk? |
| 22:26.16 | mogorman | voicemail that is |
| 22:26.28 | Goshen | I an thinking once I get my numbers ported over to broadvoice I will disconnect from the telco network |
| 22:26.36 | Goshen | and plug the entire house network into the sipura |
| 22:26.52 | madounet | Hi, I dev an app (.so not AGI) for HTTP request (in case of incoming call) with libcurl and i search a way to have a kind of HTTP persistent connections manager, anyidea? |
| 22:27.28 | Goshen | spackle: I will check it out, thank you |
| 22:29.25 | Goshen | time to run, good luck guys |
| 22:29.33 | Goshen | thanks for the help, I will keep digging |
| 22:31.00 | *** join/#asterisk fugitivo (~ajf@201.255.100.126) |
| 22:46.47 | Grooby | hey guys..what you all think of IP500 phone? |
| 22:46.55 | Grooby | finally got an go-ahead to setup * here at work |
| 22:47.03 | Grooby | and thinking about getting that for the office here |
| 22:48.34 | *** join/#asterisk MikeJ[Laptop] (~icechat5@65.170.43.34) |
| 22:50.28 | spackle | OT: anybody here use Scalix for e-mail? |
| 22:51.44 | spackle | Anybody using the IAXy know if the heartbeat setting is useful? |
| 22:51.56 | johnnyb | What is the minimum memory requirements for asterisk? |
| 22:52.37 | Eight | johnnyb: unimportant? |
| 22:52.38 | spackle | johnnyb, probably 128MB on a system only used and only running Asterisk. |
| 22:53.40 | *** join/#asterisk Y1 (~Y1@he134.internetdsl.tpnet.pl) |
| 22:53.48 | *** join/#asterisk landrocker (~landrocke@port-222-152-54-115.fastadsl.net.nz) |
| 22:54.10 | spackle | JohnnyB: how many people do you intend to use it? |
| 22:54.18 | spackle | simulataneously? |
| 22:55.00 | spackle | Grooby: still here? |
| 22:55.43 | johnnyb | spackle: I've got 4 zap lines and 16 office users |
| 22:55.49 | johnnyb | Never more than 3 people at once. |
| 22:56.10 | *** part/#asterisk ctooley (~ctooley@rrcs-24-153-228-2.sw.biz.rr.com) |
| 22:56.17 | johnnyb | I'm running into a problem where whenever I have an incoming or outgoing call, my system idle time drops to near zero. |
| 22:56.25 | johnnyb | This is causing problems for the zap card. |
| 22:56.32 | spackle | Johnnyb, throughput is the most important thing. You don't want Asterisk Swapping. |
| 22:56.40 | Moonwick | oof |
| 22:56.54 | *** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
| 22:56.56 | johnnyb | spackle: top only reports 156K of swap being used. |
| 22:57.10 | spackle | Johnnyb: how much memory in the box? |
| 22:57.11 | Grooby | still here |
| 22:57.14 | johnnyb | 128M |
| 22:57.21 | spackle | Johnnyb, what else is it doing? |
| 22:57.30 | johnnyb | Just asterisk and mail to mail out voicemails |
| 22:57.56 | spackle | johnnyb, sorry if this is obvious - not X running, right? |
| 22:58.02 | johnnyb | nope. |
| 22:58.08 | hardwire | and the other way around |
| 22:58.09 | ManxPower | jonas, add another 128M to the box |
| 22:58.09 | spackle | what about httpd? |
| 22:58.17 | johnnyb | nope. |
| 22:58.39 | hardwire | X? |
| 23:00.02 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 23:00.24 | jesster | I have an acd queue setup to have music = 007 where 007 is defined in musiconhold.conf as: 007 => mp3:/path/to/dept7/mp3s and when a call comes into that queue, * gives errors that the class 007 doesn't exist, any ideas? |
| 23:00.25 | *** join/#asterisk bparker (bparker@cable-71-8-65-183.mtv.al.charter.com) |
| 23:00.58 | johnnyb | Asterisk, postfix, three user shells, crond, and syslog |
| 23:01.20 | johnnyb | asterisk is by far the biggest memory use. |
| 23:01.25 | johnnyb | s/use/user/ |
| 23:01.29 | hardwire | its a piiiiiiigy wiggy |
| 23:01.48 | spackle | I'm with Manx, My home machine has 512 in it. Just because it can run in 128 doesn't mean it should |
| 23:01.59 | *** part/#asterisk MikeJ[Laptop] (~icechat5@65.170.43.34) |
| 23:02.00 | Nugget | ram is cheap. |
| 23:02.04 | johnnyb | spackle: the wierd thing is, though, is that it's not swapping. |
| 23:02.09 | hardwire | well |
| 23:02.10 | hardwire | hmm |
| 23:02.22 | hardwire | does it require a huge surplus of memory to handle zaptel channels? |
| 23:02.25 | spackle | it's still constrained |
| 23:02.29 | johnnyb | Nugget: not for this machine. It's a Vaio that used specialized RAM that isn't available anymore :( |
| 23:02.42 | hardwire | because I can pass quite a few ulaw channels on a 64 meg machine. |
| 23:02.48 | hardwire | w/o taxing it too heavily. |
| 23:02.51 | johnnyb | Nugget: adding memory means I have to replace the box. |
| 23:02.58 | Nugget | boxes are cheap :) |
| 23:03.07 | hardwire | johnnyb: are you using a super cpu intensive codec? |
| 23:03.15 | johnnyb | hardwire: I'm using iLBC. |
| 23:03.23 | hardwire | tried just using ulaw? |
| 23:03.24 | Eight | johnnyb: just give it a shot, and if you have problems, upgrade then. |
| 23:03.31 | spackle | Aha! |
| 23:03.39 | johnnyb | aha what? |
| 23:04.35 | johnnyb | Eight: give what a shot? |
| 23:04.50 | johnnyb | Nugget: so are we :) |
| 23:05.07 | Nugget | heh |
| 23:05.34 | ManxPower | I wonder if I could get fired for having submited the name of my largest customer to the Guinness Book of World Records |
| 23:06.03 | Nugget | if you get fired, maybe it'll be just the incentive you need to get off your ass and move to holland. :) |
| 23:06.21 | ManxPower | Nugget, the "incentive" is called "a job" |
| 23:06.40 | twisted[work] | lol |
| 23:06.41 | landrocker | anyone know if it's possible to trigger an agi script after a call hangs up? |
| 23:06.43 | Nugget | pompiedom |
| 23:06.44 | ManxPower | Nugget, I got the first draft of my resume back from the resume writing company. |
| 23:06.57 | ManxPower | I felt slightly sick reading it. |
| 23:06.59 | cripito | jhonnyb: i have an small laptop with 64mb .. with a few ulaw codecs without problems |
| 23:07.10 | brc_ | ~seen kpfleming |
| 23:07.12 | jbot | kpfleming <~chatzilla@ip68-3-230-141.ph.ph.cox.net> was last seen on IRC in channel #asterisk, 14d 21h 57m 57s ago, saying: 'add -g'. |
| 23:07.23 | spackle | JohnnyB: are the calls interoffice our outside the office through a provider? |
| 23:07.26 | twisted[work] | ManxPower, you could get fired for telling your customer that he should pay for an extra seat license; Hey, if delta can do it, you can too |
| 23:07.41 | Nugget | hah |
| 23:08.21 | ManxPower | Since nobody ASKED....I submited them for the world record of the lowest number of support staff at a US$550 million/year company. |
| 23:08.43 | twisted[work] | how many support staffers? |
| 23:08.53 | ManxPower | Not that I expect GBoWR to accept it. |
| 23:09.08 | ManxPower | twisted, a manager, a help desk person, and a part time consultant. |
| 23:09.17 | twisted[work] | lol |
| 23:09.42 | ManxPower | Shouldn't they be recognized for that accomplishment? |
| 23:09.43 | johnnyb | spackle: interoffice |
| 23:09.48 | twisted[work] | They should |
| 23:10.06 | johnnyb | spackle: that is, they are interoffice, unless going through the zap lines. But it is going through the zap liens that are killing us. |
| 23:10.21 | johnnyb | spackle: but, we always keep Asterisk in the media path no matter what. |
| 23:10.29 | ManxPower | twisted, Why are you happy? |
| 23:10.31 | ManxPower | Ah. |
| 23:10.36 | spackle | Because of the codec translation no doubt. |
| 23:10.44 | twisted[work] | Mexico, too. |
| 23:11.05 | criptos | mexioco where? |
| 23:11.07 | johnnyb | spackle: it's actually in the media path for attended transfers/call parking |
| 23:11.09 | twisted[work] | Tulum |
| 23:11.17 | spackle | JohnnyB, you should switch to ulaw and see if that makes a difference. |
| 23:11.20 | criptos | huu,.. nice... nice weather, not to hot rigth now.. |
| 23:11.25 | twisted[work] | yep |
| 23:11.28 | twisted[work] | perfect time of year ;) |
| 23:11.29 | *** join/#asterisk dlemire (~denis@68.148.230.233) |
| 23:11.32 | ManxPower | twisted, The same company told me today that the office that was supposed to open Jan 1 2005, then postponed to April 1 2005, will now open on May 31 2005. This is the office I'll be installing 60+ phones, 2 x T-1 PRI lines. I'll be in Europe on May 31. |
| 23:11.32 | johnnyb | spackle: I'll give it a whirl |
| 23:11.45 | twisted[work] | ManxPower, is this microsoft? |
| 23:11.56 | ManxPower | twisted, real estate company,. |
| 23:11.59 | twisted[work] | ManxPower, ah. |
| 23:12.16 | ManxPower | twisted, supporting 350+ end users. |
| 23:12.23 | twisted[work] | oh wow |
| 23:13.09 | ManxPower | And people wonder WHY I'm sometimes very grumpy. |
| 23:14.14 | *** part/#asterisk Matt-E- (~Matt-E-@66-224-125-137.atgi.net) |
| 23:14.14 | *** join/#asterisk spackle (~spackle@209.234.83.19) |
| 23:14.24 | dlemire | Got a few general telco questions if someone would like to point me in the right direction. |
| 23:15.25 | terrapen | wooo hooo |
| 23:16.00 | *** part/#asterisk Grooby (~Grooby@12.22.232.212) |
| 23:16.27 | johnnyb | spackle: Tried it. No luck. |
| 23:16.50 | criptos | dlemire, humm... google? |
| 23:16.55 | terrapen | Guten Tag, New Braunfels. How are you today? Hola, old San Antone', please stand out of my way |
| 23:17.24 | dlemire | criptos: Ah yes, good ol' google. |
| 23:17.52 | terrapen | ~jbot google |
| 23:17.53 | jbot | i guess google is a search engine found at http://www.google.com/ |
| 23:17.58 | terrapen | ~jbot google test |
| 23:18.04 | terrapen | nice |
| 23:18.11 | terrapen | ~jbot google Max Stalling |
| 23:18.26 | terrapen | sweet |
| 23:18.28 | Jer13261 | ~jbot google asterisk |
| 23:18.29 | terrapen | a googling bot |
| 23:19.05 | dlemire | ~jbot google help for wannabe pbx geek asterisk setup |
| 23:19.33 | terrapen | i wonder how hard it would be to install this stereo in my truck |
| 23:19.57 | terrapen | i really am sick of AM radio |
| 23:20.38 | *** join/#asterisk pbxman (~clec@rdu57-93-009.nc.rr.com) |
| 23:21.17 | terrapen | i need a wiring harness i think |
| 23:21.30 | terrapen | and probably a beefy positive lead from the battery |
| 23:21.47 | pbxman | I am having a problem with getting asterisk cvs-head to compile. It goes through 75% and errs out with a [PBX] error 1... Any suggestions? |
| 23:24.49 | pbxman | It appears this has something to do with the DUNDI protocol being added... |
| 23:24.59 | terrapen | well, there's your answer |
| 23:25.41 | pbxman | anyone there? |
| 23:26.36 | Mavvie | nope |
| 23:27.00 | *** join/#asterisk pbxman (~clec@rdu57-93-009.nc.rr.com) |
| 23:27.04 | Mcwebtree | strange how a room with dozens of people can be so quiet |
| 23:27.35 | pbxman | sorry about that.. I lost connection... Did anyone reply to my compile problem? |
| 23:27.48 | *** join/#asterisk kant (~bernd@63.245.57.70) |
| 23:28.01 | Mcwebtree | nope, no answer. |
| 23:28.18 | pbxman | OK... Is anyone else having this same problem? |
| 23:28.33 | Mcwebtree | what, lack of compile, or lack of answers ;) |
| 23:28.55 | pbxman | Both :-) |
| 23:30.00 | pbxman | make[1]: *** [pbx_dundi.o] Error 1 |
| 23:30.31 | pbxman | make: *** [subdirs] Error 1 |
| 23:31.04 | pbxman | anyone? |
| 23:31.09 | Mcwebtree | I haven't a clue. I'm still trying to set up my sip softphone to my * |
| 23:32.43 | Mcwebtree | and not having any success! |
| 23:34.33 | slePP | has anyone found a PRI w/ asterisk sending fast busies for no reason? |
| 23:34.42 | slePP | like. call, get a fast busy (this is incoming, btw).. |
| 23:34.47 | slePP | and then call again a moment later, and it rings through |
| 23:35.00 | hardwire | anybody need a nuera FXO _> HDLC box? |
| 23:35.19 | hardwire | or something screwy like that? |
| 23:35.24 | Wonka | ~jbot kernel tuning |
| 23:35.38 | *** join/#asterisk t3t (~t3t@bar.pangalacticgargleblaster.com) |
| 23:35.43 | Wonka | mh |
| 23:36.00 | Wonka | is there any linux kernel tuning howto centered on asterisks needs? |
| 23:36.12 | hardwire | no.. but you should write one :) |
| 23:36.28 | Wonka | O.o |
| 23:36.35 | hardwire | o.O |
| 23:36.40 | Wonka | ;) |
| 23:36.51 | hardwire | you know what I always immedietly think when I see o.O |
| 23:36.56 | hardwire | Totoro. |
| 23:38.11 | Wonka | i don't know any animes and related stuff... |
| 23:38.32 | hardwire | boy are you missing out |
| 23:39.09 | Wonka | i'm more into ircing with friends, hacking stuff... |
| 23:39.34 | Wonka | and ATM I'm determined to get asterisk to do what i want |
| 23:41.17 | landrocker | grar, should the h extension work inside a macro? |
| 23:41.40 | machinehd | hey all, should a cisco 7960 be configured to use inband? It has a setting "Out of Band DTMF" which is set to AVT. Not sure if I should set that to NONE. |
| 23:42.41 | *** part/#asterisk Y1 (~Y1@he134.internetdsl.tpnet.pl) |
| 23:43.03 | t3t | machinehd: I have dtmf_inband: 1 and dtmf_outofband: avt set |
| 23:43.11 | Druken | slePP: i got that calling today... |
| 23:43.24 | Druken | first was fast busy, second went threw |
| 23:43.31 | t3t | machinehd: That's "dtmf_outofband: avt" |
| 23:43.34 | tzanger | slePP: that is unusual |
| 23:43.37 | tzanger | pri debug span 1 |
| 23:43.41 | machinehd | t3t, thanks, and in your sip.conf what did you set it to? |
| 23:43.46 | tzanger | and post one of the congestions |
| 23:43.54 | *** join/#asterisk cbachman (~chatzilla@129.105.7.250) |
| 23:43.56 | *** join/#asterisk neopher (~crazy@mail.techhelpresources.com) |
| 23:44.16 | t3t | machinehd: that's what it is in SIP<MAC>.cnf |
| 23:44.23 | sivana | 289259 |
| 23:44.34 | sivana | bah |
| 23:45.01 | t3t | machinehd: same in SIPDefault.cnf |
| 23:45.13 | machinehd | t3t, right, but I'm just curious what you've set in your sip.conf for the extension? dtmfmode= |
| 23:45.24 | t3t | oh |
| 23:45.51 | t3t | dtmfmode=info |
| 23:46.07 | machinehd | oh |
| 23:46.15 | t3t | <PROTECTED> |
| 23:46.40 | machinehd | haha, yeah. I'd think the phoen should have dtmf_inband: 0 |
| 23:47.13 | t3t | machinehd: if we're using info DTMF then it shouldn't do it inband too :) |
| 23:47.50 | t3t | I ASSume that the setting is negotiated between * and the 7960 and they both use the same, but that's just a WAG |
| 23:48.42 | machinehd | t3t, great thanks for the help. I'll test it out right now |
| 23:48.46 | t3t | sure thing |
| 23:49.15 | slePP | tzanger: riiiiight |
| 23:49.32 | slePP | tzanger: 15 channels going nuts, and hit'n'miss of about 1 in 25-50 calls that do it |
| 23:49.43 | slePP | tzanger: i've been trying for 20 minutes to get it to congest again |
| 23:49.46 | ManxPower | machinehd, Inband DTMF only works with the ulaw and alaw codecs. |
| 23:50.05 | slePP | tzanger: but channel usage never breaks 18, so we should always have about 5 sitting around empty |
| 23:50.32 | tzanger | slePP: well what do you expect me to tell you |
| 23:50.41 | tzanger | get yourself to a point where you can debug |
| 23:51.59 | machinehd | ManxPower, I am using ulaw. I just wansn't sure if the cisco should be set to inband/rfc2833/info. |
| 23:52.13 | tzanger | slePP: you want help but are unable to provide the data necessary to fix it... it's gonna be a lot of data to sift though but you should be able to get what you want out of it, I didn't say it'd be easy |
| 23:52.14 | *** part/#asterisk mozilla (sabre@69.149.209.83) |
| 23:52.46 | slePP | tzafrir: :P |
| 23:52.51 | slePP | tzanger, too |
| 23:53.00 | slePP | tzanger: yeh, i know. i'm working on it, it may jsut take me the next 3 hours |
| 23:53.04 | tzanger | I'm willing ot try and help but I need data :-) |
| 23:53.09 | *** join/#asterisk voip123 (~root@200.121.129.178) |
| 23:53.13 | slePP | because as far as i can tell, asterisk isn't rejecting anything.. not that it logs to console, anyway |
| 23:53.30 | tzanger | well like I said the q931 logs will really help |
| 23:53.39 | Druken | slePP: could it be a bad channel on the pri ? |
| 23:53.41 | slePP | i managed to a congestion once, but i didn't have pri debugging on at that point. i had turned it off, and went to call someone to say i was working on it, and sure enough, it congests. |
| 23:53.42 | slePP | stupid thing |
| 23:53.50 | slePP | Druken: i'd wonder how that happens :> |
| 23:53.50 | machinehd | hmm, just tried with "info" and couldn't navigate the voicemail menu |
| 23:54.01 | slePP | tzanger: i know. i'm watching and dialing again and again from my cell phone |
| 23:54.28 | Druken | dunno, was an idea i figured i'd throw out |
| 23:54.42 | Druken | i'm sure telco's can have equipment break just as we can |
| 23:55.14 | slePP | i'd think the entire PRI would just die |
| 23:55.19 | slePP | as opposed to a specific channel |
| 23:55.39 | slePP | tzanger: my basic question was, does it sound like something you've heard of before? |
| 23:55.41 | *** join/#asterisk YoYo (YoYo@dilbert.psknet.com) |
| 23:55.41 | Druken | perhaps, but exactly how is the pri constructed on the telco side? |
| 23:55.52 | slePP | Druken: on some wires :P |
| 23:56.16 | YoYo | how do I view misc call stats on a cisco phone? |
| 23:56.34 | slePP | Druken: gonna start on your server this weekend. you want gentoo or slack? |
| 23:56.37 | slePP | or rh? |
| 23:56.50 | Wonka | debian! |
| 23:56.55 | slePP | god no |
| 23:57.03 | slePP | but, oh well :P |
| 23:57.13 | Druken | i was gonna say.. uhmm.. rh? wtf? |
| 23:57.14 | Druken | hehe |
| 23:57.16 | Mcwebtree | where can I get a quick guide to setting up my xlite - * - voipuser system. xlite - voipuser is fine. I get hundreds of errors and nothing works having followed the guides I can find. |
| 23:57.20 | Wonka | testing, and good |
| 23:57.31 | slePP | tzanger: i'll let you know when i get a useful log |
| 23:57.52 | Wonka | i don't know gentoo very much, but the concept of compiling everything myself is broken... |
| 23:57.58 | Wonka | much pain nearly no gain |
| 23:58.14 | Wonka | so, i'd stay with stage3 |
| 23:58.45 | Wonka | and where's the big difference to debian then? |
| 23:59.02 | slePP | so stage3's don't work out |
| 23:59.10 | tzanger | I'd say fuck all of those other distros and admit that Slackware 0wnz you |
| 23:59.34 | *** join/#asterisk _sam- (sam@ns2.kneedraggers.com) |