| 00:01.36 | rudholm | Strom_M: how's that BRI Asterisk integration project going? |
| 00:03.58 | HeyItsMe | If all my trouble has been because of the wrong patch cable, I owe you guys a BUNCH! |
| 00:05.39 | *** join/#asterisk areski (n=areski@121.Red-83-55-102.dynamicIP.rima-tde.net) |
| 00:06.05 | Strom_M | rudholm: waiting for AT&T to call me back |
| 00:08.03 | areski | anyone succeed to stream a mp3 feed into AGI ? my attempts with MP3Player give me a weird sound |
| 00:10.45 | *** part/#asterisk cocomp (n=jeremy@82-43-235-140.cable.ubr02.pres.blueyonder.co.uk) |
| 00:12.56 | CuriosCat | AGI? |
| 00:14.08 | areski | yes |
| 00:14.35 | areski | stream directly a mp3 feed into a AGI |
| 00:14.59 | areski | normally it should work with MP3Player but this fail for me |
| 00:15.22 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 00:15.22 | *** mode/#asterisk [+o anthm] by ChanServ |
| 00:15.40 | HeyItsMe | OK, I have put the T1 crossover in place and the lights on the back of the digium te205p are slowly flashing red |
| 00:15.50 | blitzrage | no sync |
| 00:15.51 | Strom_M | thats no good |
| 00:16.01 | Strom_M | do the lights go green/yellow when using the other cable? |
| 00:17.05 | HeyItsMe | no, the port that I plug the cable into turns green |
| 00:17.34 | HeyItsMe | It is one of these fancy t1s that is capable of pri, fxs, or whatever |
| 00:17.39 | Strom_M | now you're contradicting yourself |
| 00:17.46 | HeyItsMe | Me? |
| 00:18.03 | Strom_M | or you're not being clear |
| 00:18.24 | *** join/#asterisk fab5freddy (n=vmware@bas1-montreal19-1177817971.dsl.bell.ca) |
| 00:18.31 | HeyItsMe | Please tell me where I can do better. |
| 00:18.31 | Strom_M | does the circuit come out of red alarm when using the T1 crossover cable? Does it come out of red alarm when using the cable you were using previously? |
| 00:18.47 | HeyItsMe | exactly like you said |
| 00:18.56 | HeyItsMe | yes,yes |
| 00:19.16 | Strom_M | but earlier, you said the lights are slowly flashing red |
| 00:19.19 | Strom_M | not solid green |
| 00:20.29 | HeyItsMe | Yes, after I plugged in the crossover cable they began to turn flashing red |
| 00:21.06 | Strom_M | see, i ask the question one way, and the circuit is in red alarm. I ask it another way, and the circuit is green. |
| 00:21.08 | fab5freddy | i registered a did and i am now looking to place outbound calls, i put the configurations as per the provider in the proper configuration files. what is my next step? |
| 00:21.36 | jnc | fab5freddy: make some calls? |
| 00:21.37 | jnc | :) |
| 00:22.03 | *** join/#asterisk fender211 (n=Administ@12.171.15.10) |
| 00:22.13 | HeyItsMe | Ooops sorry I see it. I am getting tired I guess. Thanks for your patience Strom |
| 00:22.18 | fender211 | anyone familiar with UnixODBC? |
| 00:22.31 | Strom_M | HeyItsMe: now, please, answer my question definitively |
| 00:22.37 | HeyItsMe | crossover=red, straight=green |
| 00:22.47 | fab5freddy | jnc: yes, but all i get is user not found when i try to make calls |
| 00:22.53 | Strom_M | ok, then your problem is not the cable. do not use the crossover cable. |
| 00:23.04 | HeyItsMe | yes |
| 00:23.15 | Strom_M | what exactly is your problem, anyway |
| 00:23.34 | jnc | fab5freddy: okay, that's not related to the outgoing call part, are you using a pstn or sip phone to make the call? |
| 00:23.46 | fab5freddy | jnc: i have sip client |
| 00:23.47 | HeyItsMe | I have gotten the zaptel to populate the channels, but can't get out. |
| 00:23.56 | Strom_M | "can't get out"? |
| 00:24.17 | fender211 | so I'm working on setting up voicemail ODBC which I've done before.. this time it's on a Centos 4.4 64 bit O/S and while it compiles fine I get a message about a shared object when trying to use isql to my dsn? Anyone familiar with this setup? |
| 00:24.17 | jnc | fab5freddy: you'll need to make sure that there is an extension handy (users.conf?) and that your sip client is authenticating to it properly. |
| 00:24.32 | jnc | fab5freddy: you doing this by hand or with a GUI ? |
| 00:24.37 | fab5freddy | jnc: by hand |
| 00:24.42 | jnc | fabulous |
| 00:24.57 | jnc | sip set debug |
| 00:25.13 | jnc | core set verbose 3 |
| 00:25.20 | jnc | that should give you plenty of output going |
| 00:25.22 | jnc | heh |
| 00:25.27 | HeyItsMe | I can dial extensions and they work fine. When I try to dial out, I get a dead sound connection |
| 00:25.39 | Strom_M | and where is "out"? |
| 00:25.40 | fab5freddy | jnc: says invalid command in the asterisk cli prompt |
| 00:25.59 | jnc | hm. maybe asterisk 1.4 (which I'm messing with) is different |
| 00:25.59 | HeyItsMe | an outside telephone number |
| 00:26.16 | Strom_M | but are you dialing out over a PRI? SIP trunk? thin air? cheesecake? |
| 00:26.19 | jnc | fab5freddy: set debug, set verbose 4 ? |
| 00:26.46 | HeyItsMe | the adtran ta 905 t1 |
| 00:27.15 | fab5freddy | jnc: set debug <level>, what do i use for level? |
| 00:27.24 | jnc | 3 is pretty vocal |
| 00:27.28 | jnc | I'd use that |
| 00:28.06 | jnc | in /etc/asterisk/users.conf there should be an "extension" (user, really) set up for your SIP device/softphone |
| 00:28.23 | Strom_M | HeyItsMe: but what kind of entrance facilities do you have from the telco? |
| 00:28.24 | jnc | [6000] |
| 00:28.27 | jnc | ... |
| 00:28.30 | sevard | Strom_M: TDMCheeseCake |
| 00:28.33 | jnc | secret = cheesecake |
| 00:28.36 | jnc | like that :) |
| 00:28.51 | fab5freddy | jnc: yes, there are a few |
| 00:29.04 | jnc | sip client needs to be configured to register to your asterisk server with username (6000) and secret (cheesecake) |
| 00:29.08 | jnc | yeah |
| 00:29.16 | jnc | is it registering alright? |
| 00:29.24 | fab5freddy | jnc: yes registering no problem |
| 00:29.50 | jnc | can you get to the echo test (600) ? |
| 00:29.51 | sevard | ABFAB |
| 00:30.37 | fab5freddy | jnc: i was able to but i erased the bulk of the contents that came with the original configuration file as i wanted to start from the ground up to better understand what was going omn |
| 00:31.31 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 00:32.10 | fab5freddy | jnc: i am using in extensions.conf exten => 4000,Dial(SIP/provider/${EXTEN}) |
| 00:33.15 | fab5freddy | but when i try to dial 4000 is says user not found |
| 00:33.15 | mrdigital | anyone try Verizon's Voicewing with Ast |
| 00:33.37 | jnc | fab5freddy: if you do 'extensions reload' in asterisk prompt (or whatever the 1.2 equivillent is) you should see the contexts |
| 00:34.54 | fab5freddy | jnc: it says cannot find extension '4000' in context (null) |
| 00:36.10 | fab5freddy | jnc: and it says the same for extension 200.1 (extension 200) |
| 00:36.30 | jnc | okay, so you need to give your extension a context (default maybe?) |
| 00:36.53 | jnc | that's about as much I know on the matter |
| 00:37.04 | fab5freddy | jnc: is this in extensions.conf or sip.conf? |
| 00:39.12 | jnc | fab5freddy: depends on how you want to set this up, you could have sip.conf set so that calls are in context like from-sip-external, then in extensions.conf a default context rule that will change the context. I think, not 100% sure |
| 00:39.58 | jnc | I'm still hacking my way through extensions for understanding |
| 00:40.46 | Vec | Is there any kind of release schedule for asterisk, like new releases come out every Tuesday, just dont want to compile asterisk, and then the next day a new release comes out ? |
| 00:40.55 | Strom_M | Vec: no, not necessarily |
| 00:41.04 | Strom_M | it's pretty much "releases come out as needed" |
| 00:41.25 | *** join/#asterisk etfonhomey (n=etfonhom@74-140-213-69.dhcp.insightbb.com) |
| 00:41.26 | Vec | Strom_M : thanks |
| 00:41.35 | Strom_M | and regardless, SVN 1.2 and 1.4 branches are updated continuously anyway |
| 00:41.45 | Strom_M | so just compile the branch and you're set |
| 00:42.48 | Vec | Strom_M : don't want to use the branches as I'm in a production environment, rather want to wait for the "official" releases. |
| 00:43.00 | Strom_M | SVN branch is fine for production |
| 00:43.05 | Strom_M | it's SVN trunk that you should avoid |
| 00:43.23 | Vec | Strom_M : oh k, thanks |
| 00:43.29 | fender211 | Is anyone running Asterisk on Centos 4.4 64 bit production environment? |
| 00:43.58 | Vec | fender211 : I am running asterisk on RHEL 4 64 bit production environment. |
| 00:44.16 | fender211 | Hey Vec: any chance you've setup Realtime or Voicemail ODBC on that box? |
| 00:44.47 | Vec | fender211 : no sorry, use normal voicemail have not used Realtime yet, but will probably be soon. |
| 00:45.42 | fender211 | Vec: Okay.. I'm having problems getting UnixODBC to work correctly.. had no issues with the 32 bit environment but can't seem to make it work in 64 bit.. just wondering if anyone else ran across that same problem |
| 00:46.56 | jnc | it isn't quite explained in the sample extensions.ael, is '_' a matcher for "begins with" |
| 00:47.04 | *** join/#asterisk jhiver (i=jhiver@165-242.206-83.static-ip.oleane.fr) |
| 00:47.18 | jhiver | hey |
| 00:47.18 | jnc | i.e. _91NXXNXXXXXX |
| 00:47.18 | fender211 | howdy |
| 00:47.25 | jhiver | i was wondering how codec negotiation + canreinvite=yes played together |
| 00:47.32 | Qwell | jnc: _ means that it's a pattern |
| 00:47.39 | jnc | ah oaky |
| 00:48.52 | *** join/#asterisk punk0 (n=Administ@189.146.139.7) |
| 00:48.58 | *** join/#asterisk psycybrfrk (n=psychicc@pool-162-83-180-142.ny5030.east.verizon.net) |
| 00:49.29 | psycybrfrk | can you use a 56k modem to create a voip gateway? |
| 00:49.38 | Strom_M | no |
| 00:49.56 | jnc | psycybrfrk: the asterisk demo suggested that you'd need at least 28.8 modem to connect to their test |
| 00:50.06 | Strom_M | he wants to use the modem as an FXO card |
| 00:50.09 | Strom_M | and that ain't gonna work |
| 00:50.15 | jnc | ohhh |
| 00:50.22 | psycybrfrk | ok so I need an FXO card... k |
| 00:50.29 | Strom_M | get a digium TDM01B |
| 00:50.31 | jnc | it would work, but it would function so poorly you will not be happy |
| 00:50.43 | Qwell | jnc: if he wrote drivers |
| 00:50.48 | jnc | the "clone" cards are hacked modems |
| 00:50.54 | Nugget | hey Qwell, what's the executive summary on chan_cellphone? is it in a state where playing with it might yield a useful system? |
| 00:50.59 | *** join/#asterisk |Johny| (n=gomesper@bacus.corp.fccn.pt) |
| 00:51.10 | Qwell | Nugget: yes, I've used it... it worked reasonably well |
| 00:51.16 | Nugget | cool, thanks. |
| 00:51.25 | Qwell | I was supposed to talk to Kevin about it today, actually |
| 00:51.29 | Qwell | I forgot ;/ |
| 00:51.43 | Nugget | I find the concept very appealing but I don't want to fiddle with it if it's still a total mess. |
| 00:51.47 | Qwell | we may merge it in the almost immediate future |
| 00:51.49 | Nugget | partial mess I can cope with |
| 00:51.51 | Qwell | to -addons |
| 00:51.53 | Nugget | cool |
| 00:52.00 | *** join/#asterisk benjamin7062 (n=bhudgens@64-132-190-102.static.twtelecom.net) |
| 00:52.42 | benjamin7062 | Any experts alive? |
| 00:52.47 | *** part/#asterisk psycybrfrk (n=psychicc@pool-162-83-180-142.ny5030.east.verizon.net) |
| 00:52.56 | Qwell | benjamin7062: no, the only one just left |
| 00:53.05 | benjamin7062 | Bummer |
| 00:53.13 | Nugget | Do you really need an expert, or do you just need someone who knows the answer to your question? |
| 00:53.13 | Strom_M | and the other three contracted food poisoning |
| 00:53.14 | Qwell | something us nubs can help with? |
| 00:53.17 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 00:53.19 | fender211 | what is chan_cellphone? |
| 00:53.31 | Qwell | fender211: bluetooth cell/headset stuffage |
| 00:53.35 | benjamin7062 | It's probably going to end up being a nub question. =) |
| 00:53.36 | Nugget | fender211: http://macnugget.org/photos/nz2007 |
| 00:53.37 | Qwell | pretty friggen neat |
| 00:53.39 | fender211 | wow ;-) |
| 00:53.46 | fender211 | drooling |
| 00:53.52 | Nugget | it looks awesome |
| 00:54.07 | sevard | poop |
| 00:54.09 | Vec | I read somehwere that jumping to n+101 when an error occurs for some apps is depricated in asterisk 1.4, if this is the case how are errors handled now ? |
| 00:54.31 | fender211 | What am I looking at Nugget? |
| 00:54.44 | blitzrage | Vec: usually a STATUS variable, like ${DIALSTATUS} |
| 00:54.47 | Nugget | Vec: use the ${DIALSTATUS} variable in the dialplen |
| 00:54.58 | Nugget | or the dialplan, your choice. |
| 00:55.01 | benjamin7062 | I just upgraded from asterisk 1.4.0.. along with the latest libpri, and zaptel. My upgrades have gone smooth since 1.2.x but after this upgrade I'm having trouble trying to connect to a Zap channel; the complaint is something along the lines of being unable to translate from ulaw to an 'unknown' format. |
| 00:55.23 | benjamin7062 | I'm guessing I haven't specified a format somewhere and got away with it in the past.. perhaps 1.4.2 is more specific. |
| 00:55.32 | Nugget | like shown here: http://www.voip-info.org/wiki-Asterisk+variable+DIALSTATUS |
| 00:55.57 | Nugget | fender211: set of patches to SVN trunk (1.4 dev) that adds chan_cellphone and a messload of comments about how it works |
| 00:56.24 | Vec | Nugget, blitzrage : thanks |
| 00:56.30 | fender211 | Nugget: is that encrypted in your photos ? :-) that was the link you sent http://macnugget.org/photos/nz2007 |
| 00:56.37 | file | I spy... cows |
| 00:56.46 | Nugget | oh, shit. I'm a cut and paste newb. |
| 00:56.48 | Nugget | sec. |
| 00:56.52 | fender211 | heh |
| 00:56.57 | Nugget | http://bugs.digium.com/view.php?id=8919 is what I meant to paste. |
| 00:56.57 | fender211 | nice photos thought |
| 00:57.03 | Strom_M | i spy... file |
| 00:57.03 | Nugget | not my travel photos from last week. :) |
| 00:57.09 | Qwell | Nugget: I've seen people paste worse |
| 00:57.14 | Qwell | much, much, MUCH worse |
| 00:57.16 | Nugget | troodat |
| 00:57.39 | Qwell | file: Things?! |
| 00:57.44 | fender211 | Nugget: oh wow, your up in the ATX.. I'm an Austin native living in San Antone now |
| 00:57.48 | file | Qwell: yes :( |
| 00:57.53 | Qwell | such as? |
| 00:57.55 | Nugget | yay austin. |
| 00:57.58 | file | a sofa! and end tables! |
| 00:58.01 | Qwell | eep |
| 00:58.05 | Qwell | pleather? |
| 00:58.08 | Nugget | file is getting domestic! |
| 00:58.10 | Qwell | (sp) |
| 00:58.15 | jnc | whoa, cellphone as FXO. that is really neat |
| 00:58.15 | file | which will arrive on Saturday... so I will have a day to enjoy it |
| 00:58.25 | file | Nugget: yes :( |
| 00:58.26 | Qwell | jnc: and headset as fxs-like |
| 00:58.40 | Qwell | with a $20 bluetooth adapter |
| 00:58.50 | Qwell | it's really quite impressive |
| 00:58.55 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 00:58.56 | file | Random knowledge: Out of 8 access points around here I am the only one using WPA! |
| 00:59.02 | Nugget | pretty soon file will stop working on asterisk to spend more of his time focused on writing recipe databases and household maintenance scheduling software. |
| 00:59.03 | Qwell | file: yay? |
| 00:59.19 | Qwell | Nugget: and baking muffins, no doubt |
| 00:59.20 | jnc | file: you bastard ;) |
| 00:59.20 | benjamin7062 | pastebin is being mean to me; anyone know an alternative? |
| 00:59.20 | file | Nugget: muffin recipe database |
| 00:59.24 | Nugget | indeed |
| 00:59.28 | Qwell | benjamin7062: uuoc.com |
| 00:59.50 | file | Nugget: attack! |
| 00:59.59 | Nugget | eep! muffin attack |
| 01:00.10 | jnc | Josh, I finally have occasion to get my hands dirty with asterisk configuration. |
| 01:00.17 | file | jnc: uh oh |
| 01:00.25 | jnc | fear for the world, my friend |
| 01:00.31 | *** join/#asterisk csd-199 (n=adsf@189.158.223.247) |
| 01:00.42 | file | jnc: how is fmr these days? or is it something different now |
| 01:01.16 | jnc | it's a comprehensive latency test for freeno... I mean, it's same as it was before. dormant. |
| 01:01.28 | file | ah |
| 01:01.34 | benjamin7062 | Here is the error I'm getting if anyone has any ideas: |
| 01:01.35 | benjamin7062 | http://uuoc.com/1754 |
| 01:02.33 | benjamin7062 | Been going strong for quite some time on 1.4.x and 1.2.x .. so, I'm sure I'm missing something simple |
| 01:03.02 | Qwell | muffin in the middle attack! |
| 01:03.29 | file | *gasp* |
| 01:03.32 | file | Qwell: meanie |
| 01:04.38 | DrukenLPY | file: i'm not using wpa... however i'm also trying to install a captive portal.... |
| 01:05.03 | file | everyone else is WEP |
| 01:05.15 | csd-199 | I have a "canreinvite" question, Can someone help? |
| 01:05.30 | jnc | file: I've read a few bits and pieces posted about asterisk, I think you wrote some of them, very nice and informitive |
| 01:06.15 | Strom_M | woot |
| 01:07.16 | jnc | "_1234 => &ael-std-exten-ael(${EXTEN}, "IAX2");" does this match literally "1234" or "[1-4][1-4][1-4][1-4]" ? |
| 01:07.27 | Qwell | literally 1234 |
| 01:07.52 | jnc | where is EXTEN coming from then |
| 01:07.54 | benjamin7062 | I suppose no one has a simple answer for me: Last question and I'll be on my way; is there a specific place (zapata.conf) or elsewhere that I specify the encapsilation for a T1/Pri card? Something perhaps new to 1.4.2? |
| 01:07.57 | Qwell | 1234 |
| 01:08.11 | jnc | oh |
| 01:09.35 | csd-199 | I have 2 asterisk servers. Server A and Server B, each in a diferent internet connection. All works just fine, but if I login to any of those servers with my laptop from any other internet connection, I can make calls without problem with the server I'm logged, but to the other server the quality is poor... Any idea why or how can I fix it? |
| 01:10.12 | *** join/#asterisk [hC] (n=hardcore@S0106000d8891877c.vc.shawcable.net) |
| 01:12.22 | csd-199 | I have 2 asterisk servers. Server A and Server B, each in a diferent internet connection. All works just fine, but if I login to any of those servers with my laptop from any other internet connection, I can make calls without problem with the server I'm logged, but to the other server the quality is poor... Any idea why or how can I fix it? |
| 01:12.43 | *** join/#asterisk djs_2_6 (n=DJS@cpe-071-077-048-198.nc.res.rr.com) |
| 01:13.02 | Vec | Can someone explain to me: Do the patch files on new releases eg, asterisk-1.4.2.patch.gz modify everything in the previous release source i.e. 1.4.1 to make it the same as 1.4.2 ? |
| 01:13.16 | Qwell | yes |
| 01:13.27 | Vec | So if I want to see all the source code changes between version I can just look at the patch file ? |
| 01:13.34 | Qwell | yes |
| 01:13.44 | DrukenLPY | god damn he's sharp.... |
| 01:13.46 | Vec | Also the patch file for 1.4.2 will not work to update 1.4.0 |
| 01:13.53 | Qwell | correct |
| 01:13.56 | *** join/#asterisk Kumbang (n=macan@167.205.24.67) |
| 01:13.58 | Vec | DrukenLPY : he is |
| 01:14.06 | Vec | Qwell : thanks |
| 01:14.31 | Vec | i.e. it works the same like the linux kernal :) |
| 01:14.51 | Qwell | well, I'm not sure how sub versions work |
| 01:14.52 | Vec | kernel |
| 01:15.12 | Vec | Qwell : does svn generate the patch files ? |
| 01:15.15 | Qwell | like 1.2.9.1 > 1.2.10 |
| 01:15.22 | Qwell | umm, yeah, I think so |
| 01:15.25 | *** join/#asterisk BSD_Tech (n=bsdtech@ppp-69-238-48-150.dsl.irvnca.pacbell.net) |
| 01:18.30 | BSD_Tech | anyone seen l0rdr0ck |
| 01:18.36 | BSD_Tech | he owes me |
| 01:19.06 | BSD_Tech | and I have not heard from him since the port got finished |
| 01:19.09 | BSD_Tech | grrrr |
| 01:20.34 | jnc | are there any contexts besides "default" that are implicit? |
| 01:20.51 | jnc | or does everything kind of spawn off of 'default' context in extensions |
| 01:21.17 | BSD_Tech | you writ your own dialplan you can use default or remove it |
| 01:21.32 | jnc | hmm |
| 01:22.04 | pfn | huh? there are no 'implicit' contexts |
| 01:22.15 | jnc | so you're saying it's more of a convention in the config files than a hardcoded facet of configuration? |
| 01:22.26 | BSD_Tech | yes |
| 01:22.28 | pfn | a convention in the samples |
| 01:22.28 | pfn | yes |
| 01:22.36 | jnc | that makes sense now |
| 01:22.39 | Qwell | well, default is used in voicemail if you don't specify a context |
| 01:23.52 | BSD_Tech | ok I think we need a dialplan project |
| 01:24.10 | BSD_Tech | where everyone can add to it and share |
| 01:24.22 | Qwell | you mean like the wiki? |
| 01:24.27 | BSD_Tech | so we can make 1 full functional dialplan for asterisk |
| 01:24.43 | BSD_Tech | but I never see anyone post a full dial plan |
| 01:24.54 | jnc | it would be at least nice to generate a flowchart of this stuff |
| 01:25.44 | Qwell | BSD_Tech: most dialplans are extremely custom |
| 01:25.56 | Qwell | and some are incredibly complex |
| 01:26.39 | BSD_Tech | well share them so others can learn and grow them |
| 01:26.48 | BSD_Tech | I use to post alot. |
| 01:26.57 | csd-199 | I have 2 asterisk servers. Server A and Server B, each in a diferent internet connection. All works just fine, but if I login to any of those servers with my laptop from any other internet connection, I can make calls without problem with the server I'm logged, but to the other server the quality is poor... Any idea why or how can I fix it? |
| 01:26.59 | BSD_Tech | of min but I am rewriting them for 1.4 |
| 01:27.30 | BSD_Tech | qos/better routing |
| 01:27.32 | jnc | csd-199: save yourself the trouble and read up on IRC ettiquette |
| 01:28.07 | csd-199 | oh come on, I'm just remmembering that I have a question |
| 01:28.38 | csd-199 | if I repeat the question every moment, you will be right |
| 01:28.41 | csd-199 | any way |
| 01:28.47 | BSD_Tech | but the fact is that most pbx systems come with a full dialplan and then new features are added |
| 01:28.54 | csd-199 | I'm waiting for an asnwer |
| 01:29.00 | BSD_Tech | I told you |
| 01:29.11 | BSD_Tech | qos and better routing to your provider |
| 01:29.41 | BSD_Tech | and dont be a smart ass like your last comment it will piss people off |
| 01:29.55 | BSD_Tech | <csd-199> I'm waiting for an asnwer is not a smart thing to say |
| 01:29.58 | pfn | hmm, so why doesn't asterisk using the posix realtime extensions for timing? |
| 01:30.19 | csd-199 | hmm... you are very sensitive |
| 01:30.40 | csd-199 | i'm waiting for an answer means, i'm waiting for help |
| 01:30.50 | BSD_Tech | no its IRC ettiquette |
| 01:30.55 | csd-199 | or, may I can help someone... if I know the answer |
| 01:31.04 | csd-199 | ok, you are right... sorry |
| 01:31.11 | csd-199 | :) |
| 01:31.17 | BSD_Tech | that like calling for tech support and say I am waiting for a answer while they try to help you |
| 01:31.18 | csd-199 | but... Any help? |
| 01:31.37 | BSD_Tech | I have answerd you 2 timies |
| 01:31.42 | BSD_Tech | read back |
| 01:31.49 | csd-199 | ok... let me see |
| 01:32.24 | csd-199 | oh, yes... but what about the option "canreinvite" ? |
| 01:32.36 | csd-199 | can work that way? |
| 01:32.41 | BSD_Tech | that wount help your call quality |
| 01:33.11 | BSD_Tech | the call quality is most likly qos and routing |
| 01:34.05 | BSD_Tech | qwell the idea of a full dialplan that has most basic feathers would make asterisk more out of the box friendly |
| 01:34.38 | BSD_Tech | and hellp more people get started learning by giving the more code to read and learn from |
| 01:34.53 | csd-199 | yes, but this is the fact: If I login to Server A, all the calls to the A network works fine but not to server B, but if I loggin to server B, all the calls to network B will be fine, but not to network A |
| 01:35.20 | BSD_Tech | you have a network issue |
| 01:35.29 | BSD_Tech | go find it and fix it |
| 01:35.52 | csd-199 | well... may be... but Is there a way to not make a triangulation? |
| 01:36.06 | *** join/#asterisk Mavvie (n=edwin@ppp29-32.lns1.syd6.internode.on.net) |
| 01:36.32 | BSD_Tech | use ngrep and trace route and find your net issue |
| 01:36.42 | *** join/#asterisk hardwire (n=bip@rdbck-4271.palmer.mtaonline.net) |
| 01:36.45 | BSD_Tech | and then your problem will go away |
| 01:36.50 | hardwire | sweet |
| 01:36.57 | hardwire | BOLIVIAN is now in my linked in :) |
| 01:36.58 | csd-199 | ok, I'll try something and let you know |
| 01:37.01 | csd-199 | thanks a lot |
| 01:37.53 | CuriosCat | How do I define a channel on the command line? |
| 01:44.48 | |Johny| | hello to all |
| 01:44.59 | |Johny| | did you already had a problem similar to this: |
| 01:45.01 | |Johny| | retrans_pkt: Maximum retries exceeded on transmission |
| 01:45.02 | |Johny| | ? |
| 01:45.17 | |Johny| | for seqno 12282 (Critical Response) |
| 01:45.34 | |Johny| | <PROTECTED> |
| 01:46.08 | *** join/#asterisk bjohnson (n=bjohnson@i209-195-79-216.cia.com) |
| 01:46.28 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 01:47.13 | *** join/#asterisk zepmantra (i=dsadsads@125.212.110.115) |
| 01:47.22 | mrdigital | is there guy who knows alot about ast here? |
| 01:47.30 | *** part/#asterisk zepmantra (i=dsadsads@125.212.110.115) |
| 01:48.13 | *** join/#asterisk illsci (n=illsci@evil.hack3rs.org) |
| 01:49.40 | pfn | no |
| 01:50.45 | JunK-Y | guy? i only know donald :P |
| 01:51.05 | Qwell | Guy is a jerk |
| 01:51.11 | Qwell | darn, he left |
| 01:51.20 | JunK-Y | hehhhee |
| 01:51.27 | JunK-Y | the worse jerk is jason! |
| 01:51.33 | *** join/#asterisk [hC] (n=hardcore@S0106000d8891877c.vc.shawcable.net) |
| 01:51.41 | Qwell | totally |
| 01:51.55 | JunK-Y | Qwell: whatcha think about jason? |
| 01:52.07 | Qwell | I think people should give him lots and lots of money |
| 01:52.19 | JunK-Y | mouhaha ha |
| 01:52.22 | [hC] | Any of you guys know much about polycom error messages that might show up in a log after an IP601 crashes? |
| 01:52.35 | JunK-Y | [hC]: we dont know guy, we only know jason! |
| 01:52.42 | [hC] | Who's jason? :) |
| 01:52.50 | JunK-Y | a son of a bitch! |
| 01:52.56 | [hC] | hahaha |
| 01:53.22 | JunK-Y | [hC]: didnt play with 601, only 501. |
| 01:53.29 | JunK-Y | [hC]: btw, Qwell is jason. |
| 01:53.40 | [hC] | Oh, that jason |
| 01:53.41 | JunK-Y | or jason is qwell, depends how ya see things. |
| 01:54.00 | [hC] | I figured since you asked him, you meant another jason |
| 01:54.05 | [hC] | That explains his answer :) |
| 02:01.26 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-216-188.socal.res.rr.com) |
| 02:02.51 | BSD_Tech | JC what issue you having with polycom |
| 02:02.57 | BSD_Tech | pastebin the error |
| 02:03.01 | BSD_Tech | and point me to it |
| 02:04.02 | [hC] | well, the phone would randomly reboot every couple days. I have a theory though that it might have rebooted because i didnt have "append" enabled on my ftp server, i wonder if the phone rebooted cause it couldnt append to its log file. I'l pastebin the thing anyways, theres a couple things i dont understand in there. |
| 02:04.42 | wunderkin | uh.. doubtful.. were you using the phone? what firmware |
| 02:05.03 | [hC] | http://pastebin.ca/432131 |
| 02:05.14 | [hC] | 1.6.7 and 2.0.3 both seemed to be doing it |
| 02:05.33 | [hC] | one has 2 sidecars, one has 3 |
| 02:05.37 | wunderkin | do you have other phones, poe, .. |
| 02:05.39 | [hC] | it only seems to happen with 2+ sidecars attached |
| 02:06.00 | [hC] | One is PoE one is on an AC Adapter, yes there are other phones on the network, and they arent doing it |
| 02:06.16 | [hC] | however, at one site, their 430s have done it a couple times too |
| 02:06.31 | [hC] | but its very hard to confirm that, cause they were doing some weird crap over there for the first couple days of their install |
| 02:07.16 | wunderkin | i have had problems with 430s yes |
| 02:07.29 | [hC] | wunderkin: did you figure out what was causing it? |
| 02:07.32 | *** join/#asterisk De_Mon (i=de_mon@fl-71-55-184-242.dhcp.embarqhsd.net) |
| 02:07.53 | blitzrage | JunK-Y: !! |
| 02:07.59 | wunderkin | are there 2 phones having the problem or 1 |
| 02:08.10 | JunK-Y | blitzrage: !!! |
| 02:08.12 | wunderkin | bad hardware |
| 02:08.22 | wunderkin | the 501s we got were fine |
| 02:08.27 | blitzrage | hrmmm... I wonder why my Asterisk boxes aren't reinviting... |
| 02:08.39 | wunderkin | only 5 out of 21 ip430s were ok |
| 02:08.49 | [hC] | wunderkin: the 601s that im focusing on are in two different sites, the most problematic one has been replaced once, and i switched it off PoE just recently. |
| 02:08.52 | blitzrage | asterisk keeps putting itself as the RTP destination instead of the service provider... |
| 02:09.23 | [hC] | wunderkin: really. hardware huh.. how did you determine it? |
| 02:09.32 | *** join/#asterisk CpuID2 (n=nathan@gentoo/contributor/cpuid) |
| 02:10.25 | wunderkin | i went as bare as possible, multiple firmware, different offices, at home |
| 02:11.05 | [hC] | wunderkin: what was the symptom? just reboot randomly? |
| 02:11.12 | wunderkin | just make sure you are using the right power supply.. i thought i heard there were some bad ones out there... and i saw something about poe and sidecards.. some kind of adjustment... |
| 02:11.21 | wunderkin | when in use yes |
| 02:11.43 | [hC] | wunderkin: and you just replaced the unit and it went away?? |
| 02:11.47 | [hC] | that doesnt make me feel good at all. |
| 02:12.07 | wunderkin | they were replaced with 501s and they were fine, the other 5 430s never had the problem |
| 02:12.09 | [hC] | our 430s that ive seen reboot occasionally.... 10 of them did it all at once. every time. |
| 02:12.33 | wunderkin | if they were all at once i would say your problem is power or something like that in common |
| 02:13.38 | [hC] | yeah. those ones are all PoE powered from a central switch, and no other phones went down. I havent heard of this happening at all though recently, so im not so concerned about it. im more concerned about the 601s.... one of them does it sometimes 4 times a day at a busy law firm |
| 02:14.18 | pfn | hmm, if I use outkey and have the key 3des encrypted, how do I specify the password for it in iax.conf? |
| 02:14.36 | pfn | and even so, if I specify the password in iax.conf, why bother 3des encrypting the privkey |
| 02:14.56 | *** join/#asterisk newsmafia (n=newsmafi@wsip-68-224-174-204.sd.sd.cox.net) |
| 02:15.14 | wunderkin | use 2.1.0 |
| 02:15.47 | wunderkin | i know 2.0.3 had some reboot bugs in it |
| 02:15.59 | [hC] | ahh.. where do you get this info from? changelogs? |
| 02:16.05 | [hC] | I'll go log in and grab 2.1.0 - are you using it? |
| 02:16.10 | wunderkin | yes |
| 02:16.28 | wunderkin | you are a reseller? |
| 02:16.35 | Strom_M | hmm, where in the 2.1.0 xml files is the option to set the messages key extension? :) |
| 02:16.54 | wunderkin | look on the bottom is all i remember, callback contact or something like that |
| 02:17.32 | wunderkin | <mwi msg.mwi.1.subscribe="" msg.mwi.1.callBackMode="contact" msg.mwi.1.callBack="*98" |
| 02:17.53 | Strom_M | that's not in this version of sip.cfg |
| 02:18.09 | wunderkin | phone config |
| 02:18.10 | *** join/#asterisk mrdigital (n=ASFASDF@pool-72-92-103-188.phlapa.east.verizon.net) |
| 02:18.11 | Strom_M | ah ok, it's in the phone |
| 02:18.13 | Strom_M | stupid |
| 02:19.26 | blitzrage | Service Provider (SIP) <--> Asterisk A (external IP) <--> Asterisk B (Linux Router performing NAT; External IP connected to Asterisk A; Internal IP connected to Phone) <--> Phone |
| 02:19.48 | blitzrage | For some reason I can't seem to get audio from Service Provider to Asterisk B -- it keeps going to Asterisk A |
| 02:20.50 | blitzrage | I have canreinvite=yes at Asterisk A, and also for the peer that connects Asterisk B to Asterisk A. I have canreinvite=no set on the peer for Phone connecting to Asterisk B (since I obviously can't reinvite that media path). Ideas? |
| 02:21.44 | blitzrage | The 183 Session Progress I get from Asterisk A on Asterisk B keeps giving me the IP address of Asterisk A as the RTP destination, instead of the IP address of the Service Provider |
| 02:22.26 | blitzrage | When I verify the peers on the boxes with 'sip show peer foo' I see CanReinvite: Yes |
| 02:23.02 | [hC] | wunderkin: yeah. I am a reseler. |
| 02:23.08 | [hC] | reseller, even. |
| 02:23.10 | wunderkin | bugger |
| 02:23.12 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 02:23.24 | [hC] | the forgot my password page seems to be hosed, it locks up safari and displays an xml browser in firefox. |
| 02:23.27 | [hC] | I havent logged in in a while |
| 02:23.31 | [hC] | since they changed to email address logins |
| 02:23.34 | wunderkin | i put up with that bs since november.. we finally now almost have it taken care of, it would have been nice to have direct contact to polycom, i hate that |
| 02:24.08 | wunderkin | polycom was blaming it on asterisk, fuckers |
| 02:24.40 | [hC] | they are actually pretty bad with support unless you're running business edition |
| 02:24.51 | [hC] | when i said i was running the latest stable they refused to help me |
| 02:25.08 | blitzrage | oooooooooooooooooooo |
| 02:25.17 | wunderkin | we were |
| 02:25.18 | blitzrage | I bet its because I have the W flag in Dial() |
| 02:25.24 | wunderkin | unfortunately |
| 02:25.33 | [hC] | doh |
| 02:25.45 | [hC] | I guess im gonna have to redo my sip.cfg and phone config files, arent i... argh. |
| 02:25.50 | [hC] | thats a bit of an undertaking for 150 seats |
| 02:26.17 | blitzrage | shouldn't you be generating that via a script with the data from a DB? |
| 02:26.20 | ChkDigit | sed, perl, or python is your buddy. |
| 02:26.42 | wunderkin | if you do it how polycom suggests it shouldnt be that hard... |
| 02:27.03 | [hC] | oh, i have tools to do it, but its still gonna be a pain in my ass. most notably the sip.cfg file, remembering to turn off the mwi chrip, etc etc |
| 02:27.19 | wunderkin | thats easy |
| 02:27.34 | ChkDigit | Is GotoIf() supposed to jump to the label no matter what the test results are? |
| 02:27.52 | blitzrage | ChkDigit: huh? |
| 02:27.52 | Qwell | no.. |
| 02:27.59 | Qwell | Your test is probably wrong |
| 02:28.00 | [hC] | wunderkin: other than the MWI chirp, and the one touch voicemail... sntp server and default sip server, do you make any other sip.cfg changes? |
| 02:28.04 | Qwell | ie; not wrapped in a $[] |
| 02:28.14 | blitzrage | GotoIf($[...]?my_label) |
| 02:28.18 | ChkDigit | exten => s,6,GotoIf( $[${REPEATED} < 2]?2 ) |
| 02:28.19 | blitzrage | that'll fall through if the test fails |
| 02:28.27 | ChkDigit | I also had spaces... |
| 02:28.41 | blitzrage | ChkDigit: that'll fall through if ${REPEATED} is not less than 2 |
| 02:28.42 | ChkDigit | The console replies: |
| 02:28.52 | wunderkin | a few.. i am on a different config template right now... |
| 02:29.10 | wunderkin | make sure your xml is valid too |
| 02:29.21 | [hC] | wunderkin: any suggestions? or have a url for a guide to a few? |
| 02:30.50 | [hC] | Ooh, one thing.. Can I turn that MyStat/Buddies softkey crap off and still have the 601 do hints properly? People always ask if they can use that and i have to tell them no |
| 02:31.07 | *** join/#asterisk Pettson (i=andreas@seleya.nh.nation.liu.se) |
| 02:31.18 | wunderkin | i dont use those so i dont know |
| 02:31.25 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-216-188.socal.res.rr.com) |
| 02:32.10 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 02:32.15 | ChkDigit | Hmmm, ${REPEATED} < 2 ==> 3 < 2 = false, should fall through then, right? |
| 02:32.51 | ChkDigit | Console says: |
| 02:33.21 | ChkDigit | <PROTECTED> |
| 02:33.22 | ChkDigit | <PROTECTED> |
| 02:34.00 | ChkDigit | <PROTECTED> |
| 02:34.00 | ChkDigit | <PROTECTED> |
| 02:34.00 | ChkDigit | <PROTECTED> |
| 02:34.01 | ChkDigit | <PROTECTED> |
| 02:34.32 | ChkDigit | So, I'm seeing the result being true(1) or false(0), and getting the same results... |
| 02:34.54 | *** join/#asterisk antlers (n=antlers@ip70-173-90-39.lv.lv.cox.net) |
| 02:35.42 | wunderkin | [hC], i can send you the template that fender sent... i dont like his style... but... hey |
| 02:36.10 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 02:39.33 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 02:39.40 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 02:41.07 | csd-199 | I want to buy a card with 4 analog ports, any cheap idea? |
| 02:41.16 | *** join/#asterisk Fieldy (i=aNR4smZh@gentoo/contributor/Fieldy) |
| 02:41.29 | bkruse_home | tdm400p |
| 02:42.32 | wunderkin | [hC], i see you are using an old bootrom too.. dont know if that matters any |
| 02:43.12 | *** join/#asterisk d3wayne (n=dwayne@c-68-62-209-143.hsd1.al.comcast.net) |
| 02:44.05 | [hC] | wunderkin: sure. (re the template) - cant hurt |
| 02:45.23 | [hC] | I need to build a php driven company directory using the microbrowser now.. the directory on the polycoms is just terrible. |
| 02:48.50 | [hC] | Polycom's site is obviously busted as hell right now, login barely worked, and now downloads arent starting.. |
| 02:48.52 | [hC] | argh. |
| 02:54.04 | wunderkin | it has the f/w and bootrom in it |
| 02:54.35 | wunderkin | so it is about 20mb |
| 02:54.51 | [hC] | ahh gotcha. 2.1.0?? |
| 02:54.59 | [hC] | you can either dcc it to me or email it to me if you like |
| 02:55.22 | ChkDigit | Holy crap. The leading space in GotoIf( $[ ] ) was the problem. |
| 02:56.01 | wunderkin | not sure if i have the forwarding setup on here yet |
| 02:57.10 | [hC] | Doesnt seem happy. |
| 02:57.39 | bkruse_home | ChkDigit: welcome to python! |
| 02:57.41 | bkruse_home | oh wait... |
| 02:59.20 | *** join/#asterisk thoughtpolice (n=austin@c75-111-145-138.plaicmtc01.tx.dh.suddenlink.net) |
| 03:00.01 | [hC] | oh wow. syslog support for error messages |
| 03:00.03 | [hC] | thats handy |
| 03:03.53 | [hC] | there we go |
| 03:04.06 | *** join/#asterisk axisys (n=axisys@c-69-143-190-152.hsd1.va.comcast.net) |
| 03:04.07 | [hC] | super slow, but its going. |
| 03:04.11 | [hC] | oops |
| 03:04.13 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:04.34 | wunderkin | you had to put your porn on hold |
| 03:04.53 | [hC] | dohhh... :) |
| 03:05.23 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 03:05.45 | bkruse_home | kill -9 `pidof wget http://azp-pr0n-pwns.com` |
| 03:05.55 | blitzrage | anyone find that tethreal on a local machine can see the ACK to a 200 OK, but Asterisk doesn't see it at all? |
| 03:07.34 | JunK-Y | blitzrage: have ya notice db_exists could cousume so much cpu? |
| 03:07.42 | *** join/#asterisk bawb2 (n=bawb2@ip51051.estcmp.ku.edu) |
| 03:07.46 | JunK-Y | blitzrage: and nope, never see that. |
| 03:08.00 | blitzrage | JunK-Y: I don't use DB_EXISTS() -- I use func_odbc |
| 03:08.32 | blitzrage | JunK-Y: it's soooooooooo wierd -- tethereal sees the packet come in on port 5060, but Asterisk doesn't, and keeps sending the 200 OK up to 6 retransmissions, then drops the call because it doesn't see the ACK |
| 03:08.50 | JunK-Y | i might switch my customer to it, how it goes in prod? |
| 03:09.01 | blitzrage | JunK-Y: so far so good! |
| 03:09.07 | blitzrage | doesn't crash in heavy load testing |
| 03:09.15 | JunK-Y | which * version? |
| 03:09.19 | blitzrage | 1.4 SVN |
| 03:09.39 | JunK-Y | from like 2 months ago or from like today? |
| 03:09.59 | pfn | wow, JunK-Y is still around :p |
| 03:10.21 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 03:10.25 | BSD_Tech | I have 1.4.2 running on bsd and making over 200 calls a day |
| 03:10.33 | BSD_Tech | and i has yet to crash |
| 03:10.41 | JunK-Y | pfn: heeeyyy dude, yep, long time no see. |
| 03:10.49 | blitzrage | JunK-Y: from today |
| 03:10.53 | blitzrage | I update nearly every morning |
| 03:11.07 | JunK-Y | not in crontab i hope :) |
| 03:11.47 | [hC] | I wish i had the ability to stay that up to date :) |
| 03:12.11 | JunK-Y | [hC]: if ya want, ya can! |
| 03:12.33 | [hC] | JunK-Y: i could, i just wouldnt have any time to play with it! :S |
| 03:12.40 | [hC] | I have yet to install 1.4 at ALL yet... thats how backed up i am |
| 03:13.13 | blitzrage | JunK-Y: ummm... of course I run it in a cronjob! |
| 03:13.34 | blitzrage | I rebuilt my entire platform on 1.4 starting about 3-4 months ago |
| 03:13.41 | blitzrage | I've found numerous bugs and had them fixed :) |
| 03:13.55 | [hC] | i gotta do that too... soon.. |
| 03:14.56 | pfn | I think I'm gonna start making the move to * 1.4 this week |
| 03:15.00 | Corydon76-home | Yeah, he bugs the crap out of us. ;-) |
| 03:15.04 | pfn | I just need to rewrite my webapp to review my cdr's |
| 03:15.24 | pfn | and maybe work with ARA instead of the old res_config_odbc stuff |
| 03:15.55 | Corydon76-home | Um, the new realtime stuff is still with res_config_odbc |
| 03:16.06 | pfn | is it? |
| 03:16.16 | Corydon76-home | Yes, it's just a different API |
| 03:16.16 | pfn | I thought res_config_odbc = "realtime static" |
| 03:16.21 | Corydon76-home | Nope |
| 03:16.23 | pfn | and res_config_odbc itself goes away (the so) |
| 03:16.35 | Corydon76-home | Nope, it implements both sides |
| 03:16.52 | pfn | when I say res_config_odbc, I think of it as it existed in 1.0 |
| 03:17.01 | Corydon76-home | It's evolved since |
| 03:17.13 | pfn | a very basic api for retrieving a config-file from db |
| 03:17.23 | Corydon76-home | Nothing really exists the way it was in 1.0 anymore |
| 03:17.36 | pfn | Corydon76-home, I know, hence my desire to upgrade sometime in the near future |
| 03:17.39 | pfn | if only to play with something new |
| 03:17.50 | blitzrage | Corydon76-home: heh :) |
| 03:19.12 | *** join/#asterisk CaCtus491 (n=a@203-166-248-79.dyn.iinet.net.au) |
| 03:23.01 | *** join/#asterisk bawb2 (n=bawb2@ip51051.estcmp.ku.edu) |
| 03:25.48 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 03:26.00 | *** join/#asterisk InHisName (n=Administ@c-68-38-105-1.hsd1.pa.comcast.net) |
| 03:29.16 | bkruse_home | ouch! |
| 03:29.32 | CaCtus491 | Our asterik setup seems to have somewhat broken over the easter break - we have 3 SIP handsets (2x Cisco 7905 and 1x Linksys SPA921). We use a SIP service provider for incoming/outgoing calls. Incomming and out going calls work fine, as do calls between the handsets, however now voicemail audio seems to be broken - and I can't work out why |
| 03:31.04 | CaCtus491 | everything seems to be working, just no audio at the handsets, ie with asterisk -vvvc, I see it the prompts, and it accepts the mailbox, password, etc |
| 03:31.08 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 03:31.11 | CaCtus491 | any ideas? |
| 03:32.04 | bkruse_home | nat? |
| 03:32.11 | bkruse_home | nat problems it sounds like |
| 03:32.36 | bkruse_home | maybe.......... |
| 03:32.44 | CaCtus491 | there is no nat involved - local asterisk server |
| 03:33.02 | CaCtus491 | and it was all working fine the other week for the last 6 months or so |
| 03:37.35 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 03:38.13 | JT | did anything change? |
| 03:39.04 | *** join/#asterisk mrdigital (n=ASFASDF@pool-72-94-125-39.phlapa.east.verizon.net) |
| 03:40.51 | *** join/#asterisk [hC] (n=hardcore@S0106000d8891877c.vc.shawcable.net) |
| 03:41.51 | pfn | <PROTECTED> |
| 03:41.52 | pfn | <PROTECTED> |
| 03:41.57 | pfn | oops |
| 03:42.00 | pfn | nevermind |
| 03:44.47 | [hC] | wunderkin: does this template use the MWI sound chirp that rings every few minutes when you have a message waiting? judging from the way i change mine to shut up, it looks like this one makes noise... |
| 03:45.17 | CaCtus491 | well, I added the linksys phone to sip.conf, but that shouldn't have broken things, and in anycase, I backed up my configs and tried the samples ones and have the same problem |
| 03:45.26 | blitzrage | I typically have to change the audio prompts in the sip.cfg file to be silent |
| 03:47.23 | wunderkin | in the sip.cfg, patterns, misc, change chord to silence on each one |
| 03:48.10 | *** join/#asterisk mrdigital (n=ASFASDF@pool-72-94-124-149.phlapa.east.verizon.net) |
| 03:49.01 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:49.07 | [hC] | wunderkin: yeah, thats what i usually do too.. its set to chord.. do you change his sip.cfg any other ways, too? |
| 03:50.04 | pfn | ugh, why doesn't color console in asterisk support 'screen' |
| 03:53.14 | *** join/#asterisk bmg505 (n=leon@196.209.182.36) |
| 03:54.02 | tzafrir_laptop | run TERM=vt100 asterisk |
| 03:55.13 | *** join/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
| 03:55.26 | xai | does anyone have a list of large companies using * ? |
| 03:55.35 | *** join/#asterisk DrRighteous (n=DrRighte@ool-44c7ad06.dyn.optonline.net) |
| 03:56.09 | Qwell | xai: Fry's does, Vonage does/did for voicemail |
| 03:56.23 | *** join/#asterisk zuez (i=steve@66.103.132.86) |
| 03:59.04 | *** join/#asterisk [hC] (n=hardcore@S0106000d8891877c.vc.shawcable.net) |
| 04:09.06 | xai | Qwell: thanks.. thats a start.. Would be nice to have a big list to point to . |
| 04:09.50 | xai | Qwell: of course non-telphony companies would be better to convince some businesses to keep or install * |
| 04:12.29 | jnc | CaCtus491: codec issues happened to me |
| 04:12.48 | jnc | CaCtus491: I had to explicitly enable GSM etc. |
| 04:14.10 | wunderkin | [hC], hmm i reset some of the stuff to default.. the chip would be at every reg... do you need registrations anyways? we had problems with short registrations... |
| 04:14.21 | BSD_Tech | gasterisk |
| 04:14.21 | wunderkin | fry's.. electronics...? o rly? |
| 04:15.07 | [hC] | wunderkin: really? how so? |
| 04:18.29 | wunderkin | eh |
| 04:18.52 | wunderkin | need more input |
| 04:21.01 | wunderkin | need more ram too, damn |
| 04:21.46 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-185-4.buckeyecom.net) |
| 04:22.08 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 04:24.28 | [hC] | wunderkin: i mean with the short registrations |
| 04:28.35 | wunderkin | high cpu usage apparantly... i forgot there is a load monitor i think.. should have checked that... but i think whenever it sent a reg.. it would lock up for a second |
| 04:28.48 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 04:29.01 | wunderkin | speaking of sucky polycoms |
| 04:29.02 | wunderkin | ha |
| 04:29.03 | wunderkin | :D |
| 04:29.14 | [TK]D-Fender | wunderkin, sup .... BITCH?! ;) |
| 04:29.20 | wunderkin | moo |
| 04:29.44 | [TK]D-Fender | like the t-shirt says "%$#@ milk, got beer?" |
| 04:29.52 | [TK]D-Fender | I'm up and running again :) |
| 04:29.54 | [TK]D-Fender | yay |
| 04:30.03 | [TK]D-Fender | yum update for 180 packages :D |
| 04:30.19 | [TK]D-Fender | but routing, server files, and * functional |
| 04:31.07 | jnc | macro-stdexten appears hardcoded for voicemail :/ |
| 04:31.07 | [hC] | [TK]D-Fender: have YOU ever seen a 601 w/ sidecars (2 or more) reboot occasionally? |
| 04:31.08 | jnc | attempting at the moment to craft from scratch in an extensions.ael |
| 04:31.10 | [TK]D-Fender | jnc, funny... the file its in isn't read-only locked in MY wolrd :) |
| 04:31.31 | [TK]D-Fender | [hC], I've heard of it at 1 clients place, yes |
| 04:32.02 | [TK]D-Fender | [hC], but only that one place. |
| 04:32.08 | [hC] | [TK]D-Fender: any idea on a fix, or where to start looking? |
| 04:32.13 | jnc | [TK]D-Fender: voicemail requests it, according to users.conf comment "hasvoicemail = yes" turns it on or off |
| 04:32.14 | *** join/#asterisk thoughtpolice (n=austin@c75-111-145-138.plaicmtc01.tx.dh.suddenlink.net) |
| 04:32.27 | [TK]D-Fender | [hC], check your power, then try running without consoles. |
| 04:32.39 | [TK]D-Fender | jnc, screw users.conf |
| 04:33.02 | [hC] | [TK]D-Fender: so far have tried switching from PoE to AC power... took one console off and it seemed to help, but it still does it with 2 on there occasionally.. |
| 04:33.10 | jnc | I am glad that you've got your own setup working nicely |
| 04:34.24 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 04:34.24 | *** mode/#asterisk [+o denon] by ChanServ |
| 04:34.24 | [TK]D-Fender | jnc, seriously... users.conf is a mistake. a jumbled mess of stuff that does NOT actually belong together anywhere execpt on a TOASTER. |
| 04:34.24 | jnc | I'm not sure I know enough yet about * to understand your frustration with the config file system |
| 04:34.24 | [TK]D-Fender | jnc, now if a toaster is indeed what you want Trixbox awaits you! |
| 04:34.41 | [TK]D-Fender | jnc, Its not the system I have an issue with, its just that ONE file :) |
| 04:34.50 | jnc | oh, alright |
| 04:35.04 | jnc | where else would you define whether or not extensions have voicemail capabilities then? |
| 04:35.14 | [TK]D-Fender | jnc, extensions.conf |
| 04:35.25 | [TK]D-Fender | jnc, like any sane person would! |
| 04:35.29 | jnc | strange. they're equivillent? |
| 04:35.59 | [hC] | [TK]D-Fender: by the sounds of it, ill have conversations with you when i foray into 1.4 and beat my head against this users.conf you speak of. |
| 04:36.09 | jnc | this is stuff included in the "make samples", i'm taking it for granted |
| 04:36.16 | [TK]D-Fender | jnc, No. Users.conf is a psycho-mess of trying to define too many things in 1 files. Extensions.conf says step-by-step exactly how a call is processed and that INCLUDES voicemail usage. |
| 04:36.37 | [TK]D-Fender | [hC], You can blow it off.... jsut don't USE it |
| 04:36.57 | [TK]D-Fender | jnc, you must ... UNLEARN .... |
| 04:37.21 | jnc | difficult to do when I haven't learned yet heh |
| 04:37.28 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 04:38.13 | jnc | writing this from scratch in AEL is my target, I have slowly picked apart asterisk to figure out what the flowchart(?) is |
| 04:38.25 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 04:39.31 | jnc | [TK]D-Fender: guessing now, the role of users.conf could effectively be made into a context in extensions.ael for inclusion in the default context? does that make sense? |
| 04:39.53 | *** join/#asterisk dos000 (n=ymo@CPE000f66912f92-CM0018c0c6147e.cpe.net.cable.rogers.com) |
| 04:40.45 | dos000 | is it possible to see all the users configured in a * realtime scenario. i am using ast 1.2 and just want to see the users in the db from the ast shell interface |
| 04:40.50 | [TK]D-Fender | jnc, yes it probably does something entirely too smart (aka STUPID). Trash it. |
| 04:41.19 | [TK]D-Fender | jnc, And I can say that AEL is a nifty idea that few in here use or support... not recommended. |
| 04:41.31 | jnc | oh |
| 04:41.58 | *** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net) |
| 04:42.14 | jnc | I'm a real C dork, it looks easier to read to me, will heed your advices though |
| 04:42.21 | *** join/#asterisk SECGOD (i=SECGOD@c-71-57-36-106.hsd1.il.comcast.net) |
| 04:43.48 | [TK]D-Fender | jnc, I admit it loks tempting at times, but rarely do people actually use enough complex functions to warrant that. Secondly all that happens is that AEL(1/2) is parsed back to STANDARD extensions.conf logic on load so you don't actually gain a single thing you can't do the traditional way. |
| 04:44.29 | jnc | [TK]D-Fender: writing text adventure games? :P |
| 04:44.38 | [TK]D-Fender | jnc, Oh... except there is luke-jr here to help you with whatever happs should you continue down that road regardless. |
| 04:44.48 | jnc | =) |
| 04:44.55 | [TK]D-Fender | jnc, don't follow your "joke".... |
| 04:44.58 | *** join/#asterisk aptura (n=lork@S010600a0c93f6f7e.vs.shawcable.net) |
| 04:45.00 | luke-jr | ... |
| 04:45.04 | [TK]D-Fender | :D |
| 04:45.08 | luke-jr | there's nothing "STANDARD" about .conf nor AEL |
| 04:45.40 | *** join/#asterisk i3inary (i=i3inary@ip68-8-91-87.sd.sd.cox.net) |
| 04:46.00 | jnc | the AEL has a concise means of probability based procedurals |
| 04:46.19 | [TK]D-Fender | luke-jr, Feel free to coin a moniker for our historical extensions.conf "format" if you will. Its just a name.... |
| 04:46.24 | jnc | so like say, random(5) { NoOp("You have been eaten by a grue... |
| 04:46.31 | jnc | something roughly like that |
| 04:46.56 | aptura | evening. |
| 04:47.12 | aptura | so what is the topic of discussion today? |
| 04:47.32 | luke-jr | [TK]D-Fender: deprecated :) |
| 04:47.35 | jnc | aptura: I torture [TK]D-Fender with new boot questions |
| 04:48.07 | [TK]D-Fender | luke-jr, lol.... and they say you don't have a sense of humour! Pshaw! |
| 04:48.19 | aptura | jnc thats okay :) I am compiling ast on bsd now. Its been years since using bsd so starting at the bottom of the learning curve again. |
| 04:48.23 | luke-jr | [TK]D-Fender: who says that now? :p |
| 04:48.56 | aptura | I also need a new system. old hardware sucks for compiles takes so long :) |
| 04:49.05 | aptura | hehe |
| 04:49.13 | [TK]D-Fender | bag* |
| 04:49.32 | aptura | fact, a cat can survive a 4 story fall as long as it lands on all of its feet at the same time. |
| 04:49.36 | jnc | aptura: I got exposure to NetBSD on a handheld MIPS device (formerly WinCE 2.11 pro), it trips out my linix-ified mind |
| 04:49.50 | aptura | jnc thats interesting. |
| 04:50.11 | *** part/#asterisk DrRighteous (n=DrRighte@ool-44c7ad06.dyn.optonline.net) |
| 04:50.19 | jnc | aptura: sadly, was not able to get network hardware functioning |
| 04:50.38 | aptura | yea, I tried netbsd years ago had issues with it. |
| 04:52.12 | aptura | I wonder if there is compile times for ast/bsd online. |
| 04:52.47 | [TK]D-Fender | aptura, yeah, my mom proved that when she was a kid dropping her cat out the window on a bet from her sister. Sure it landed on it feet, and BROKE ALL OF THEM. Didn't survive long past a week./ |
| 04:53.11 | aptura | ohh man |
| 04:53.29 | aptura | kids do stupid things :) |
| 04:54.14 | [TK]D-Fender | dear god... 120/358 updates installed.... gonna take a while |
| 04:54.37 | aptura | Sorry for the cat. My mom told me of a story that today would land the neighbor in jail with what he did with two cats. I wont say it in the channel since it is a little disturbing. |
| 04:55.03 | aptura | But anyway. :) |
| 04:55.14 | aptura | TK ever been to vancouver? |
| 04:55.29 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 04:55.50 | [TK]D-Fender | aptura, Nope, but I have friends of family out that way. I should go some time... |
| 04:56.14 | [TK]D-Fender | aptura, but I've got 2 spots I've been eyeing longer : Bahamas, & New Zealand |
| 04:56.24 | *** join/#asterisk stridernzl (n=neville@125-237-110-53.jetstream.xtra.co.nz) |
| 04:56.35 | aptura | Culture here is very diverse vs say seattle. More then half of Vancouvers population "pop 1.2 mil" was born out side Canada. Its also getting very crouded and congested. |
| 04:56.44 | *** join/#asterisk Abydos313 (n=abydos31@ppp-71-137-118-216.dsl.irvnca.pacbell.net) |
| 04:56.54 | aptura | I will take New Zeland ANY day |
| 04:57.16 | ChkDigit | But I'm a snow nut. |
| 04:57.20 | aptura | im not |
| 04:57.21 | Abydos313 | hey guys |
| 04:57.30 | *** join/#asterisk ManxPower (n=manxpowe@210.sub-70-222-6.myvzw.com) |
| 04:57.33 | *** join/#asterisk rrrobert (n=rrobert@58-65-160-140.nayatel.pk) |
| 04:57.47 | aptura | Our winter sucked this past year. IT was a long dark tunnel of dispair :) |
| 04:58.04 | aptura | Ohh and we had 125% snow pack to. |
| 04:58.17 | [TK]D-Fender | Winter was OK this year... not much to speak of, and I got my Green Christmas. |
| 04:59.24 | aptura | BTW this is a funny fact. Mount Baked which is east of Bellingham Washington had a record snow pack of 102 feet for the winter of 1998. IT was so deep the resort staff had to dug the chair lifts out of the snow up the mountain. Imagine chair lifts that are normally 30-40 feet in the air are under snow :) |
| 04:59.35 | aptura | Mount Baker that is. |
| 04:59.41 | *** join/#asterisk moranil (n=moranil@122.162.73.25) |
| 05:00.05 | aptura | IT was a world record snow pack. |
| 05:00.13 | [TK]D-Fender | Real men ski Mt. Logan ;) |
| 05:00.33 | ManxPower | All winters are long dark tunnels of despair. |
| 05:00.38 | aptura | Utah? |
| 05:00.50 | red9012 | how can I have a context extensions accessible within another context? |
| 05:00.55 | aptura | Manx we got lots of rain and crappy weather here :) |
| 05:01.05 | jnc | aptura: good time to learn how to use a chair lift. not much fear of impact from falling off? |
| 05:01.35 | jnc | note... I dislocated my left shoulder in a snowboard related accident this year |
| 05:01.49 | ChkDigit | red9012: You mean like using include => extensions? |
| 05:01.50 | aptura | jnc dont know. I used to ski in the early 80s and it was okay. |
| 05:02.10 | jnc | hmm. |
| 05:02.35 | ChkDigit | Not too deep, but not too shallow. |
| 05:02.45 | jnc | well... my accident was because I fell on ice. you know, not much snow in the midwest north of chicago |
| 05:02.58 | aptura | love huskies and malamutes :) btw met one of the Disney Snow Dogs once. Striking eyes :) |
| 05:03.00 | jnc | ChkDigit: cool beans |
| 05:04.43 | aptura | btw I can probebly append a shutdown now after make so it shuts off ? |
| 05:05.09 | aptura | for any bsd fans who may know. |
| 05:05.13 | ManxPower | Snow is uncivilized. People only like winter because they don't know any better. |
| 05:05.28 | ChkDigit | >=) |
| 05:06.19 | [TK]D-Fender | "Skiing has made a lot of people happy, most of them doctorrs.." |
| 05:06.31 | aptura | we had lots of really bad accidents this winter. Alex frasier bridge was a total nightmare one day. Two jackknifed semis and smashed up cars shut the bridge down for 6 hours. Wife was stick on anaciss island for 5 hours. It takes perhaps 5 min to get off the island :) |
| 05:06.39 | *** join/#asterisk hrmphh (i=patrick@notchill.com) |
| 05:06.48 | aptura | im tired cannot spell right :) |
| 05:07.42 | aptura | well I am just going to sign off this compile is taking to long and need to head off to bed. |
| 05:07.50 | aptura | night |
| 05:07.55 | ChkDigit | That is the nice thing about Saskatchewan. no islands to get stuck on, no hills to slide down... |
| 05:11.57 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 05:12.29 | [TK]D-Fender | ChkDigit, where the land is so flat you can see your dog running away from you for DAYS... and the population density is so low that its a fine line between camping and HOMELESSNESS! |
| 05:13.33 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 05:14.36 | hrmphh | can someone take a look at this log? not sure why using Directory() tries to call w-40-100 (extension 100 being the one i want to dial) instead of just "100" |
| 05:14.40 | hrmphh | http://pastebin.ca/432322 |
| 05:16.25 | [TK]D-Fender | hrmphh, we'd have to see your extensions.conf |
| 05:16.31 | [TK]D-Fender | er... voicemail.conf |
| 05:16.45 | hrmphh | voicemail.conf just has extension 100 |
| 05:16.52 | hrmphh | its trying to find w-40-100 for some reason |
| 05:16.52 | hrmphh | sec |
| 05:18.01 | hrmphh | 100 => 12345,My Name,myemail@address.com |
| 05:18.13 | [TK]D-Fender | -- Executing [100@ael-internal:1] Goto("Zap/2-1", "sw-40-100|1") in new stack |
| 05:18.20 | [TK]D-Fender | this looks like its your dialplan doing the Goto |
| 05:18.30 | hrmphh | sec |
| 05:19.25 | pfn | ael results in a messy * console.... |
| 05:19.33 | hrmphh | http://pastebin.ca/432326 |
| 05:19.42 | hrmphh | non-ael results is ugly syntax |
| 05:20.26 | *** join/#asterisk tenzind (n=tenzind@202.144.144.21) |
| 05:20.37 | [TK]D-Fender | hrmphh, And that isn't the context brb |
| 05:20.44 | *** join/#asterisk boch (n=fran@190.48.203.133) |
| 05:24.37 | dos000 | anyone i am trying to run asterisk with realtime conf from a db. is there a way i can check if a user has vmail configured from the * shell promt ? |
| 05:25.18 | boch | could you help me with this: http://pastebin.ca/432328 ? |
| 05:25.19 | dos000 | or even ponters to realtime conf will be fine |
| 05:26.41 | dos000 | boch: try dig nueva_pbx. it looks like the server nueva_pbx is not in your dns .. or dns is not properly configured |
| 05:27.21 | BSD_Tech | man 1.4.2 rocks on bsd |
| 05:28.27 | boch | dos000, but this line: register => 556:viejapbxpwdd34@192.168.0.250:5060/nueva_pbx in the sip.conf file shouldnt bind 'nueva_pbx' to that registry record ? |
| 05:29.48 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 05:29.52 | Mahmoud | hello |
| 05:30.09 | dos000 | i am not really an expert on * syntax ... i just guessed because it tried connecting to that host and it failed |
| 05:30.43 | Mahmoud | i have SPA-3102, and have my POTS interface (FXO) registered to asterisk. How to place calls to POTS network from asterisk? |
| 05:30.53 | *** join/#asterisk saftsack (n=oliver@p54a7ee81.dip.t-dialin.net) |
| 05:31.24 | hrmphh | somehow its inserting sw-XX before my extension |
| 05:31.25 | hrmphh | any idea why? |
| 05:36.01 | hrmphh | http://www.voip-info.org/wiki/index.php?page=Asterisk+AEL2 |
| 05:36.02 | hrmphh | there |
| 05:36.05 | hrmphh | saerch for "sw-" |
| 05:36.08 | hrmphh | wtf is it doing that? |
| 05:36.43 | BSD_Tech | ael is a pain |
| 05:37.09 | [hC] | any of you guys do faxing from a Pri into a SIP ATA with an analog fax machine plugged in? |
| 05:39.10 | hrmphh | here: http://pastebin.ca/432339 |
| 05:39.14 | hrmphh | thats the problem |
| 05:39.24 | hrmphh | see ael2 code and generated dialplan and wtf |
| 05:43.16 | jnc | hrmphh: perhaps it's specific to that ? no idea about the tech involved |
| 05:43.18 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:43.44 | *** join/#asterisk zeeesh (i=zeeesh@202.38.55.125) |
| 05:43.51 | zeeesh | hi |
| 05:44.01 | hrmphh | its some stupid ael2 "feature" |
| 05:44.43 | Mahmoud | i'm using sipura as my voice gateway. it has two phone ports, an FXS and an FXO. both are registered as SIP accounts |
| 05:44.50 | Mahmoud | I can use the analog phone connected to the FXS port and call any SIP phone easily |
| 05:44.57 | Mahmoud | but the problem is, how can I use the FXO port via SIP phones, and call phones located in the POTS network? |
| 05:46.18 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 05:46.23 | mcab | wunderkin: sorry this is going a while back, but do you have any app logs of the 430s you had problems with that you can pastebin? |
| 05:47.28 | wunderkin | maybe... are you having the same problem or are you with polycom? sometimes you make me wonder :D |
| 05:47.34 | hrmphh | can someone remind me how you take the last x characters of ${EXTEN}? |
| 05:48.03 | Mahmoud | :3 |
| 05:48.08 | Mahmoud | hmm or :-3, not sure |
| 05:48.15 | hrmphh | dont think its :x |
| 05:48.23 | hrmphh | think that takes off front |
| 05:48.30 | Mahmoud | try :-3 |
| 05:48.35 | hrmphh | k |
| 05:49.53 | mcab | wunderkin: I have a hunch I want to confirm... |
| 05:51.11 | *** join/#asterisk saftsack (n=oliver@p54a7e510.dip.t-dialin.net) |
| 05:51.23 | *** join/#asterisk af_ (n=getsmart@81-174-45-50.f5.ngi.it) |
| 05:53.23 | wunderkin | mcab, http://www.pastebin.ca/432348 |
| 05:53.33 | mcab | thanks |
| 05:53.33 | dos000 | anyone: whic sql statement is executed when i call MailboxExists ? ist the re a way (in debug) mode to see the actual sql ? |
| 05:55.19 | wunderkin | not sure if that is the only type of crash.. but several of the phones do it.. this is just 1 of them |
| 05:57.00 | mcab | wunderkin: yeah, I've seen that issue before. Looks like someone forgot an assert in the 430 DSP code :-) |
| 05:57.22 | wunderkin | kjfsakjfadf |
| 05:57.27 | mcab | but the 501s use an older processor, so wouldn't trip over it |
| 05:57.45 | wunderkin | yes... so it can happen on any 430? |
| 05:58.02 | mcab | given the right conditions, yeah |
| 05:58.07 | wunderkin | we have 5 that have never had the problem... but the rest have all of the time from 1.6.7 to 2.1.0 |
| 05:58.08 | mcab | I don't know what triggers it |
| 05:58.21 | wunderkin | they are probably not very high usage phones though.. the 'good' ones |
| 05:58.37 | wunderkin | we have been battling polycom since november on it and they blamed asterisk, i told them that it wasn't |
| 05:59.13 | wunderkin | i figured it was a software thing |
| 05:59.36 | mcab | heh, the phone should be able to handle pretty much anything that get's thrown at it... |
| 06:00.00 | hrmphh | what must be installed in order to get voicemails sent via email |
| 06:00.39 | wunderkin | yes... we also had a problem doing 60 sec regs w/ 30 sec nat keepalives.. it seems like everytime it goes to register, the cpu is too busy, so take for instance the cursor.. it stops blinking so you cant press any keys.. a couple seconds later it is ok.. we changed to 1 hour regs and it is fine |
| 06:00.40 | *** join/#asterisk sumasuma (n=kurukko@61.14.86.23) |
| 06:00.59 | wunderkin | that was on a 430 and 501, cant tell me that the cpu cant handle a reg.. come on :D |
| 06:01.52 | wunderkin | if they are dialing on hook, they would hear a dtmf sound for a long period of time, it would not register further keypresses.. everything points to high cpu usage |
| 06:01.56 | jql | if it wasn't a firmware app, that behavior "feels" like a DNS hangup |
| 06:02.09 | jql | but, no idea what happens on a phone |
| 06:02.24 | mcab | wunderkin: heh |
| 06:02.25 | wunderkin | well, i'm using an ip address.. no hostnames... |
| 06:02.41 | mcab | jql: you'd think the u/i would be a seperate thread/process/whatever |
| 06:03.43 | wunderkin | we have to switch to using a wrt and ser because they need failover on a secondary connection... |
| 06:03.55 | wunderkin | regarding the reboots i remember seeing assert in dsp messages before too... |
| 06:04.06 | ChkDigit | hrmphh: app_voicemail and a functioning MTA. |
| 06:05.29 | mcab | wunderkin: before when? before 430s? |
| 06:05.50 | wunderkin | no on the bad ip430s.. it could have been on a previous firmware i dont remember.. we have went through 1.6.7 all the way up |
| 06:06.31 | wunderkin | is there any way to avoid this problem other than not use a 430? i keep joking telling people on here not to get a 430 :P |
| 06:06.38 | mcab | *nod* |
| 06:06.41 | hrmphh | any way to test sendmail mail |
| 06:06.43 | hrmphh | from the console? |
| 06:06.48 | hrmphh | the asterisk console |
| 06:08.22 | sumasuma | hrmphh: yes, type mail subject contents body signature toaddress fromaddress drop/deliver |
| 06:08.34 | sumasuma | it will deliver email to asterisk system |
| 06:08.52 | sumasuma | hrmphh: you mean you want to control asterisk with email ? |
| 06:09.14 | wunderkin | mcab, umm... just dont use a 430 is that what you are saying? does it affect any other models? |
| 06:09.26 | hrmphh | umm |
| 06:09.32 | hrmphh | i want to test that asterisk can send mail |
| 06:09.34 | hrmphh | and use sendmail |
| 06:09.40 | hrmphh | i know the system itself can |
| 06:09.51 | hrmphh | echo blah | mail user@address.com works |
| 06:10.08 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:10.12 | sumasuma | oh ok |
| 06:10.14 | mcab | wunderkin: there's a case open on it with polycom, there's supposed to be a fix in new software |
| 06:10.36 | wunderkin | blah and they look at me funny why |
| 06:10.38 | mcab | AFAIK it only hits 430s |
| 06:10.41 | sumasuma | hrmphh: you can try with Systemcall in asterisk, that wil work fine |
| 06:10.43 | wunderkin | thats good |
| 06:10.45 | *** join/#asterisk saftsack (n=oliver@p54A7D74B.dip.t-dialin.net) |
| 06:10.53 | hrmphh | suma; example please? new to me |
| 06:11.19 | mcab | wunderkin: Polycom has an odd relationship with Asterisk... |
| 06:11.25 | wunderkin | mcab, would you happen to know the number, so i can watch out for it on the release notes? |
| 06:11.29 | wunderkin | heh yeah tell me about it |
| 06:12.21 | mcab | not off-hand, I'd have to root around my work e-mail, I think it might be there |
| 06:12.29 | wunderkin | i'll need to let [tk]d-fender know.. he has a client with that problem too |
| 06:12.38 | wunderkin | work=polycom? |
| 06:13.23 | wunderkin | [names have been changed to protect the innocent] |
| 06:13.43 | wunderkin | aha dragnet, thats right, i was trying to think of the name earlier |
| 06:14.06 | *** join/#asterisk darken_darken (n=marco@10.154.77.83.cust.bluewin.ch) |
| 06:14.15 | dos000 | how does one check the voicemail enteries .. i keep getting "No entry in voicemail config file for ..." |
| 06:14.24 | mcab | just the facts ma'am... |
| 06:14.32 | wunderkin | dos000: make sure you specify the vm context... |
| 06:14.46 | wunderkin | voicemail.conf |
| 06:15.33 | dos000 | wunderkin: i am running 1.2 and i made sure the extconfig.conf has ... voicemail => odbc,asterisk,vmusers .. i still cant get it to accept voicemail :( |
| 06:15.51 | wunderkin | nfi on rt.. lmnop |
| 06:16.04 | dos000 | nfi ... ??? |
| 06:16.11 | wunderkin | hmm.. |
| 06:16.12 | wunderkin | ~nfi |
| 06:16.23 | jbot | nfi is probably No Fucking Idea |
| 06:16.23 | mcab | No F..... Idea :-) |
| 06:16.30 | dos000 | tow! |
| 06:16.31 | wunderkin | :D |
| 06:16.51 | dos000 | and lmnop :8- |
| 06:16.59 | wunderkin | it sounded cool |
| 06:17.04 | wunderkin | all of the abbrevs |
| 06:17.13 | dos000 | heh |
| 06:17.32 | wunderkin | from the excitement ADD kicking in because of mcab |
| 06:17.42 | wunderkin | you dont want to know the rest |
| 06:18.11 | dos000 | i ve been trying to get this going for a while now ... |
| 06:18.24 | jnc | is the "default" context a real default? it conflicts with the AEL keyword 'default' |
| 06:18.35 | dos000 | i came back to * after a long absence |
| 06:18.42 | wunderkin | dos000, well, are you specifying the vm context when you call voicemail? |
| 06:19.11 | dos000 | wunderkin: hmm .. no .. i dont specify any context. what should i specify ? |
| 06:19.12 | jnc | are all contexts specified from config file syntax or are some contexts defauling to "default" context if not specified |
| 06:19.26 | wunderkin | dos000, ... the voicemail context it is defined in... |
| 06:20.15 | wunderkin | [vmcontext] 1 => stuff; Voicemail(1@vmcontext) |
| 06:21.04 | *** part/#asterisk newsmafia (n=newsmafi@wsip-68-224-174-204.sd.sd.cox.net) |
| 06:21.31 | dos000 | wunderkin: here is my extconf http://pastebin.ca/432364 |
| 06:22.32 | dos000 | wunderkin: i am tryng to follow this http://voip-info.org/wiki/view/Realtime%20Integration%20Of%20Asterisk%20With%20OpenSER |
| 06:22.35 | wunderkin | ... yeah which shows that you are not specifying the vmcontext... and also putting the options in the wrong spot |
| 06:23.10 | wunderkin | voicemail(${EXTEN:1}@vmcontext|u) |
| 06:23.24 | dos000 | ah ! |
| 06:23.29 | codefreeze | jnc: "default" for a context name has a special grammar rule to allow it in that spot. Too many people use it. |
| 06:23.46 | jnc | codefreeze: okay, no sense avoiding it? |
| 06:24.02 | dos000 | wunderkin: now where does vmcontext comes from ? where is it specified ? |
| 06:24.31 | codefreeze | jnc: Let me know if you have trouble because of it. |
| 06:24.39 | wunderkin | mcab, so is the only way to avoid that problem is to not ever make any calls on the ip430? hehe.. |
| 06:24.54 | wunderkin | dos000, your db |
| 06:25.05 | wunderkin | in the voicemail config... context |
| 06:25.11 | mcab | wunderkin: sure! you can use it as a wonderfully expensive desk clock :-) |
| 06:25.14 | jnc | codefreeze: no worries yet, I am still beginning to wrap my head around all the config files initially. I have recently begun to use asterisk and configure it in depth |
| 06:25.27 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 06:25.58 | dos000 | wunderkin: i only have a general context .. i dont have a specific context called vmcontext |
| 06:26.09 | jnc | wunderkin: "sip.conf:;vmexten=voicemail"... |
| 06:26.11 | wunderkin | whatever you call it |
| 06:26.25 | jnc | is that specifying that context 'vmexten' is the context for voicemail? |
| 06:26.43 | jnc | or is 'vmexten' a built-in keyword of some kind |
| 06:27.01 | wunderkin | never used it |
| 06:27.07 | jnc | oh |
| 06:27.42 | jnc | you said that dos000 had the settings in the wrong places, but what makes it incorrect? wouldn't these settings work in any of the config files? |
| 06:28.44 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 06:29.13 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 06:29.37 | dos000 | i saw on google that there is an option to show the actual sql query happening to look for the actual vm box. i just dont know which flag i have to turn on to see that sql query happening |
| 06:30.15 | jnc | dos000: might be a debug feature at the asterisk prompt |
| 06:30.36 | dos000 | i tried -gcvvvv .. no help |
| 06:31.12 | dos000 | Apr 10 10:23:56 WARNING[14527]: app_voicemail.c:2461 leave_voicemail: No entry in voicemail config file for 'test31003' |
| 06:31.22 | dos000 | tow! |
| 06:31.34 | wunderkin | jnc, it was deprecated... |
| 06:31.47 | dos000 | it just cant go to the db for the vm config |
| 06:33.06 | jnc | wish I had an overview of any special handling precautions to keep in mind with the asterisk config files |
| 06:33.08 | dos000 | i keep saying show voicemail users it comes back empty |
| 06:33.29 | dos000 | man is this thing obscure to debug |
| 06:34.07 | wunderkin | i doubt that you can use the cli to view rt users... but i dont know.. because.. i dont use it |
| 06:38.03 | *** join/#asterisk saftsack (n=oliver@p54a7f9bd.dip.t-dialin.net) |
| 06:38.31 | dos000 | more help .. google again .. http://astbill.com/node/389 |
| 06:39.28 | dos000 | not sure if this searchcontext is fixed in the latest builds |
| 06:42.26 | wunderkin | mcab, alright... nite..thanks man |
| 06:42.41 | mcab | wunderkin: g'night, no worries |
| 06:46.35 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 06:54.22 | *** join/#asterisk maverickbna (i=sentinel@wikipedia/Shadowhntr) |
| 07:02.18 | *** join/#asterisk saftsack (n=oliver@p54a7f22b.dip.t-dialin.net) |
| 07:07.43 | *** join/#asterisk shay|work (n=shay@unaffiliated/shay) |
| 07:07.45 | shay|work | hello |
| 07:16.29 | *** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140) |
| 07:16.41 | dos000 | yay .. vmail finally works ! |
| 07:21.36 | *** join/#asterisk saftsack (n=oliver@p54A7E207.dip.t-dialin.net) |
| 07:23.31 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 07:23.35 | Mahmoud | hello |
| 07:24.13 | Mahmoud | whenever I attempt to stablish a voip-to-pstn call, i hear the tone "tooooo..etc" |
| 07:24.20 | Mahmoud | I'm using Asterisk with SPA-3102 |
| 07:25.20 | Mahmoud | my dial plan looks like: exten => _X.,1,dial(SIP/spaAccount:${EXTEN}) |
| 07:29.36 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
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| 07:51.26 | *** join/#asterisk matt_ (n=matt@mattyspc.ipv6.mattstone.net) |
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| 08:01.40 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-221315.home.otenet.gr) |
| 08:02.20 | nasls_lsa | goodmorning ! |
| 08:07.25 | Chicago | Guten Morgen! |
| 08:07.46 | Chicago | Dobre Utra |
| 08:13.12 | *** join/#asterisk donkilla (n=rogers@196.200.26.174) |
| 08:13.20 | donkilla | Hi Everyone |
| 08:13.51 | donkilla | Have anyone had experiance installing on Fedora Core 6? |
| 08:14.20 | *** part/#asterisk donkilla (n=rogers@196.200.26.174) |
| 08:14.25 | *** join/#asterisk donkilla (n=rogers@196.200.26.174) |
| 08:14.44 | donkilla | I get this error |
| 08:14.46 | donkilla | make[3]: *** [/usr/local/zaptel-1.2.16/xpp/xbus-core.o] Error 1 |
| 08:14.46 | donkilla | make[2]: *** [/usr/local/zaptel-1.2.16/xpp] Error 2 |
| 08:14.46 | donkilla | make[1]: *** [_module_/usr/local/zaptel-1.2.16] Error 2 |
| 08:14.46 | donkilla | make[1]: Leaving directory `/usr/src/kernels/2.6.18-1.2798.fc6-i586' |
| 08:14.46 | donkilla | make: *** [all] Error 2 |
| 08:14.50 | *** join/#asterisk darkskiez (n=mbryars@195-11-205-216.suip.mezzonet.net) |
| 08:15.14 | nasls_lsa | I installed the BeroNet BNS40, and now I want to get incoming calls to my DialPlan .. any ideas ? |
| 08:15.31 | nasls_lsa | I did the con figuration in midsn.conf and extensions.conf but I am not sure if it is right ... |
| 08:18.58 | donkilla | Any idea/? |
| 08:19.08 | *** join/#asterisk vgster (n=vgster@h14658.navonline.net) |
| 08:26.07 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
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| 08:31.24 | Mahmoud | any SPA-3000 or 3102 users here? |
| 08:31.46 | Mahmoud | calls comming from PSTN are directed into Line 1 (the analog phone attached to SPA) |
| 08:32.01 | Mahmoud | I don't want this, I want SPA to send calls from PSTN to Asterisk |
| 08:32.07 | Mahmoud | is it possible? |
| 08:32.34 | darkskiez | yes, i dont remember what box u tick tho |
| 08:32.37 | *** join/#asterisk mquin (n=mike@pdpc/supporter/active/mquin) |
| 08:33.15 | darkskiez | I had that setup, however, routing the call back to it if you did want it, keeping asterisk in the media path did produce nasty latencies. |
| 08:33.50 | Mahmoud | it's for home use |
| 08:35.41 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 08:36.07 | Mahmoud | darkskiez, how will spa send the call to asterisk? |
| 08:36.23 | Mahmoud | darkskiez, i mean, how will the dial plan look like? |
| 08:36.44 | darkskiez | think u put it in the call forwarding |
| 08:37.39 | Mahmoud | i see, makes sense |
| 08:41.54 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 08:43.37 | Mahmoud | thanks, works :) |
| 08:44.42 | darkskiez | if only someone could suggest a fix for my pri line :) |
| 08:44.56 | *** join/#asterisk equinox0r (n=equinox@et-1-16.gw-nat.bs.ka.oneandone.net) |
| 08:45.17 | *** part/#asterisk donkilla (n=rogers@196.200.26.174) |
| 08:49.29 | equinox0r | hi there.. |
| 08:49.48 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 08:50.27 | equinox0r | i try to use asterisk with a dialplan in the mysql db .. i've read that i have to switch from the textfile based extensions to the extmysql thingies .. but i dont want to swith everytime i get a call, i want asterisk to use the mysql table *always* .. |
| 08:51.41 | putzz | hmmm |
| 08:51.48 | putzz | so keep all your stuff in realtime |
| 08:52.00 | equinox0r | yep .. but how? |
| 08:52.13 | putzz | put your data in sql only |
| 08:52.20 | equinox0r | sip and iax work already |
| 08:52.28 | equinox0r | you mean sql statements ? |
| 08:52.58 | putzz | sip and iax works so what are your trying to do? |
| 08:54.09 | equinox0r | the dialplan |
| 08:54.28 | equinox0r | i want this to have completly in the database |
| 08:54.48 | putzz | did u add your extensions table yet? |
| 08:55.16 | equinox0r | yep |
| 08:55.37 | equinox0r | extconfig.conf -> extensions => mysql,asterisk,extensions |
| 08:56.14 | putzz | ok so it should work already if u configured it properly |
| 08:57.27 | nasls_lsa | I installed the BeroNet BNS40, and now I want to get incoming calls to my DialPlan .. any ideas ? |
| 08:58.31 | equinox0r | i tried again, but the command show dialplan does only show the default settings that are creating by pbx_ael for example .. |
| 09:00.51 | putzz | equinox0r: switch => Realtime/mycontext@realtime_ext |
| 09:01.04 | equinox0r | putzz, where to put that? |
| 09:01.22 | equinox0r | and why mycontext? i want asterisk to use the realtime extensions always |
| 09:01.39 | putzz | thats what the switch is for |
| 09:01.50 | putzz | extensions.conf |
| 09:01.59 | equinox0r | in the default context or the global thing? |
| 09:03.19 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
| 09:03.50 | Mahmoud | darkskiez, 2nd problem, can you help? |
| 09:03.59 | darkskiez | no |
| 09:04.17 | darkskiez | for i have no idea what it is. |
| 09:04.42 | Mahmoud | when i call from my mobile phone, the call is forwarded to asterisk which is fine |
| 09:05.06 | Mahmoud | but when my mobile goes on-hook, asterisk still doesn't detect his, and still sends RTP packets |
| 09:05.31 | Mahmoud | could be that SPA is not sending proper call ternimation messages |
| 09:05.34 | darkskiez | thats your SPA not detecting it, not asterisk, you've got to configure it |
| 09:05.51 | Mahmoud | where to configure this thing? |
| 09:06.08 | darkskiez | there is stuff like reverse polarity detection and silence detection stuff, its specific to your telephone company, you'll need to google for them |
| 09:06.29 | Mahmoud | i'll try them |
| 09:06.29 | nasls_lsa | well , I didn't make it to take calls yet from my misdn card , but I did calls out. From a SIP phone, I dialed a number and he couldn't hear me .. any ideas ? |
| 09:07.11 | Mahmoud | darkskiez, found it, it's in "PSTN Disconnect Detection" section |
| 09:07.33 | darkskiez | obvious now isnt it |
| 09:07.34 | darkskiez | :) |
| 09:08.39 | putzz | heh |
| 09:11.39 | nasls_lsa | there is a small delay in my calls ... |
| 09:15.50 | *** part/#asterisk gordonjcp (n=gordonjc@cpc1-broo2-0-0-cust991.renf.cable.ntl.com) |
| 09:19.00 | *** join/#asterisk MrChimpy (n=MrChimpy@212.158.8.162) |
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| 09:22.52 | *** part/#asterisk moranil (n=moranil@122.162.73.25) |
| 09:23.18 | *** join/#asterisk Zefk (n=Zefk@wsc-fo.b.astral.ro) |
| 09:23.59 | nasls_lsa | I got at console the message : P[ 0] maxnum:3P[ 1] GOT IGNORE SETUP ... what does that mean ? |
| 09:27.02 | *** join/#asterisk AF-Slash (n=AF-Slash@63-227-170-144.hlna.qwest.net) |
| 09:29.45 | equinox0r | Apr 10 13:33:07 NOTICE[3494]: pbx.c:1741 pbx_extension_helper: Cannot find extension context 'default' |
| 09:30.14 | equinox0r | why does asterisk cant find this context? i have the switch statement in my extensions.conf and there is a default context in my realtime mysql database |
| 09:35.04 | equinox0r | please help :( |
| 09:36.34 | nasls_lsa | ok , finaly I got incoming call ! but I can't hear anything :( |
| 09:37.38 | *** join/#asterisk vgster (n=vgster@h14658.navonline.net) |
| 09:54.54 | Mahmoud | why can't i dial numbers like "*33*4444#" ?? |
| 09:55.17 | Mahmoud | dial(SIP/foo/*33*4444#) |
| 09:55.23 | nasls_lsa | do you have any extension like _X33X ? |
| 09:55.36 | nasls_lsa | no ideas |
| 09:55.56 | Mahmoud | exten => 111,1,dial(SIP/foo/*33*4444#) |
| 09:56.19 | Mahmoud | forgot to say, SIP/foo is connected to SPA |
| 10:02.35 | *** join/#asterisk tenzind (n=tenzind@202.144.144.21) |
| 10:05.12 | equinox0r | i have now "installed" the asterisk gui as described on the channel topic from #asterisk-gui .. now i try to login via ( /setup/install.html ), before i added a new user to manager.conf but the login does not work (invalid password) .. |
| 10:05.52 | equinox0r | an idea why it does not work? |
| 10:06.11 | nasls_lsa | how do I set the CallerID number to that who calling me at ISDN channel ? |
| 10:09.31 | *** join/#asterisk MRH2 (n=Mr_happy@62.49.242.3) |
| 10:15.49 | *** join/#asterisk SoftIce (n=phil@vc-196-207-45-253.3g.vodacom.co.za) |
| 10:16.05 | nasls_lsa | exten => s,1,Dial(SIP/snom/CallerID(${EXTEN})) <- why I don't get the calledID ? |
| 10:16.53 | nasls_lsa | I tried : exten => s,1,Dial(SIP/snom/Set(CallerID(${EXTEN}))) that too .. :/ |
| 10:17.12 | SoftIce | hm, how does asterisk-gui work with 1.2 ? does it or doesn't it / |
| 10:20.45 | *** join/#asterisk threat (n=phix@60-240-43-214.static.tpgi.com.au) |
| 10:20.47 | threat | hey |
| 10:22.43 | *** join/#asterisk voltagex (n=voltagex@124-254-104-78-dsl.ispone.net.au) |
| 10:22.49 | voltagex | anyone awake? |
| 10:25.12 | voltagex | hi stkn |
| 10:49.06 | *** join/#asterisk grEvenX (n=even@ti500720a080-3765.bb.online.no) |
| 10:53.28 | *** join/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net) |
| 10:56.49 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 11:01.30 | threat | hi |
| 11:02.06 | threat | What would cause intermediate line drop outs and bad quality? |
| 11:02.27 | threat | how would I go about debugging / troubleshooting this? |
| 11:03.12 | voltagex | bad latency |
| 11:03.16 | voltagex | lack of bandwidth |
| 11:06.19 | *** join/#asterisk friedrich| (n=friedric@e177240229.adsl.alicedsl.de) |
| 11:06.23 | threat | 512/128 ADSL |
| 11:06.31 | voltagex | yep |
| 11:06.35 | voltagex | which codec? |
| 11:06.42 | voltagex | and which voip provider? |
| 11:07.12 | SoftIce | yukkk 512/128 ADSL I hope you have some serious QoS pols setup |
| 11:07.22 | SoftIce | diffserv, rsvp, etc |
| 11:07.41 | SoftIce | otherwise tcp will hop in the queue and udp packets will start dropping |
| 11:07.50 | voltagex | SoftIce: I use that kind of connection |
| 11:07.54 | equinox0r | can someone help me with my realtime extensions? i dont know how/where to use the switch statement |
| 11:08.05 | voltagex | SoftIce: it's one of the better ones available to Australians |
| 11:08.26 | SoftIce | voltagex: and you not using QoS and you using it for browsing,etc? |
| 11:08.39 | threat | I am using the wondershaper too (traffic shaping script) :) |
| 11:08.43 | threat | ummm |
| 11:08.45 | threat | codec, hold on |
| 11:08.47 | voltagex | SoftIce: no, I have qos set up |
| 11:08.53 | SoftIce | thats why |
| 11:09.04 | SoftIce | if you give udp its own queue and traffic priority its another story |
| 11:09.30 | voltagex | SoftIce: FreeWorldDialup in particular will drop your call in an instant if the ping/latency is too high |
| 11:09.32 | threat | codecs are defined in codec.conf? |
| 11:09.49 | SoftIce | threat: codecs are defined by you |
| 11:09.54 | threat | speex |
| 11:10.18 | SoftIce | voltagex: so prevent jitterbuffer |
| 11:10.19 | threat | that is what is defined in codecs.conf |
| 11:10.45 | threat | SoftIce, you have links to rsvp howtos / information? |
| 11:10.46 | voltagex | threat: no, defined by allow= in sip.conf or iax.conf |
| 11:10.51 | SoftIce | threat: no, in your sip.conf, iax.conf, etc what codecs are you using? |
| 11:10.58 | threat | voltagex, ok |
| 11:11.09 | SoftIce | or what codecs are your phones using to pass out through your pbx on |
| 11:11.20 | *** join/#asterisk heart (n=zippetto@lugbari/people/heart) |
| 11:11.26 | voltagex | threat: try allow=gsm or allow=g726 not many providers support speex |
| 11:11.28 | threat | gsm, alaw, ulaw |
| 11:11.34 | voltagex | in that order? |
| 11:11.37 | threat | yes |
| 11:11.48 | voltagex | alaw and ulaw wil MAX! your upstream bandwith |
| 11:11.57 | threat | should I piss them off? |
| 11:12.01 | voltagex | which voip provider are you using? |
| 11:12.13 | threat | ummm, I am moving to go talk |
| 11:12.17 | voltagex | threat: no, you will find providers like FreeWorldDialup force ulaw |
| 11:12.23 | SoftIce | threat: hmf, ive done extensive research into QoS, not much i dont know about QoS with VOIP, but no im not sure where to find good documentations, ive only found small pieces everywhere, try getting the book, 'switching to voip' |
| 11:12.28 | SoftIce | I must go anyway, & |
| 11:12.42 | threat | I was on engin |
| 11:13.16 | voltagex | threat, we need to do some testing, are you on MSN or AIM? |
| 11:13.41 | threat | voltagex, yeah, MSN and ICQ, although IRC is great :) |
| 11:13.45 | equinox0r | can someone please help me with my realtime extensions? i dont know how/where to use the switch statement |
| 11:21.22 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 11:22.57 | zeeesh | i registered 2 peers at 2 different asterisk server .. like peer 100 registered at asterisk1 and peer 200 registered at asterisk2 .. how can both peers can comunicate with each other by using SIP????? |
| 11:23.55 | SoftIce | setup a dialplan |
| 11:24.05 | SoftIce | and setup a trust relationship |
| 11:25.07 | equinox0r | ok another problem .. i have a default context in my extensions.conf .. when i try to call a extension in that context everything is fine .. but how can i call extensions in another context? do i have to add a new switch statement for the other context in my extensions.conf ? |
| 11:26.28 | equinox0r | or do i have to make a new extension in the default context that filters out the dialed number (extension) and redirects to the other extension@context ? |
| 11:26.45 | *** join/#asterisk tenzind (n=tenzind@202.144.144.21) |
| 11:27.18 | SoftIce | no you can include contexts or you can goto a context |
| 11:27.26 | SoftIce | depending on what you want to do |
| 11:27.28 | equinox0r | hum .. |
| 11:27.29 | equinox0r | ok |
| 11:27.43 | equinox0r | i have the following extensions in default: 3201 to 3204 |
| 11:27.59 | equinox0r | and i have 11 to 14 in context "menu" |
| 11:28.21 | *** join/#asterisk Dovid (n=Dovid@l192-117-114-1.broadband.actcom.net.il) |
| 11:28.22 | equinox0r | every call that is established goes to default |
| 11:28.27 | equinox0r | and i cant reach any other context |
| 11:28.37 | SoftIce | well use a gotoif statment |
| 11:28.45 | equinox0r | how would that look like? |
| 11:29.00 | SoftIce | or in sip/iax define what context you want each registration to go to |
| 11:29.11 | *** join/#asterisk uski (n=uski@bre01-1-88-162-0-210.fbx.proxad.net) |
| 11:29.16 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:29.19 | equinox0r | my sip phone should reach both contexts |
| 11:29.26 | equinox0r | default context is default |
| 11:29.42 | uski | hi :) |
| 11:29.42 | equinox0r | for the sip account (softphone) |
| 11:30.25 | SoftIce | so use an include statment |
| 11:30.38 | equinox0r | in the context (extensions.conf) ? |
| 11:30.44 | uski | does anyone knows if Cingular sends caller ID on internationnal calls ? i.e. if someone in the USA with a Cingular cellphone calls me here, outside the USA, will i see his number ? i ask this because i'll move to the usa and i'd like to setup a callback to save my euros, but i need the caller id to work |
| 11:31.10 | uski | if someone who has a cingular cellphone is kind enough to call me so i can see if it works... it'd be much appreciated |
| 11:31.31 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 11:31.46 | equinox0r | sorry for these questions, but i'm very new to asterisk :/ |
| 11:32.35 | threat | equinox0r, ditto |
| 11:32.42 | equinox0r | ;) |
| 11:32.49 | threat | equinox0r, help me out when you find out how to set it up :P |
| 11:33.00 | equinox0r | threat, what u mean especially? |
| 11:33.56 | threat | equinox0r, well lets start with softphones :) what softphone do you recommend? |
| 11:34.01 | equinox0r | x-lite |
| 11:34.02 | threat | (for linux) |
| 11:34.04 | equinox0r | hm |
| 11:34.04 | threat | ok |
| 11:34.10 | equinox0r | x-lite is windows-only i think |
| 11:34.18 | equinox0r | but you can use .. gna... forgotten name .. |
| 11:34.19 | equinox0r | erm wait .. |
| 11:34.38 | threat | there is a linux version of x-lite |
| 11:34.42 | *** join/#asterisk friedrich| (n=friedric@e177240229.adsl.alicedsl.de) |
| 11:34.56 | equinox0r | i just saw it ;) |
| 11:35.22 | equinox0r | x-lite-2.0.1105d.ebuild |
| 11:35.28 | equinox0r | (if you use gentoo =P) |
| 11:35.52 | threat | (I am a Debian man) |
| 11:36.04 | equinox0r | ok dont start that distribution things ;) |
| 11:38.25 | equinox0r | i now have the include => menu in my default context, but when i try to call the extension 11 the logs tell my that there is no extension 11 in the default context .. .oO |
| 11:41.23 | *** join/#asterisk kumbalae (n=sarisdjk@cm115.omega176.maxonline.com.sg) |
| 11:44.27 | kumbalae | is there is any software for WebMeetme ? |
| 11:44.37 | *** join/#asterisk darken_darken (n=marco@246.176.76.83.cust.bluewin.ch) |
| 11:45.13 | threat | ok |
| 11:45.15 | threat | I have x-lite |
| 11:45.18 | *** join/#asterisk friedrich| (n=friedric@e177240229.adsl.alicedsl.de) |
| 11:47.33 | threat | hmmm |
| 11:47.43 | threat | ok now I need to add in a user to asterisks |
| 11:47.54 | equinox0r | a sip account |
| 11:47.55 | threat | I have a LDAP directory, can I get asterisks to use this? |
| 11:48.00 | equinox0r | dunno |
| 11:48.07 | threat | yes a sip account |
| 11:48.15 | threat | ok what is the usually way to create a sip account? |
| 11:48.29 | threat | links to docs anyone> |
| 11:48.29 | threat | ? |
| 11:48.50 | equinox0r | http://www.voip-info.org/wiki-Asterisk+config+sip.conf |
| 11:49.04 | equinox0r | to have the sip accounts in the sip.conf fil |
| 11:49.06 | equinox0r | file* |
| 11:49.16 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-185-4.buckeyecom.net) |
| 11:49.33 | threat | thnx |
| 11:50.03 | *** join/#asterisk Fieldy (i=TaERKDPK@gentoo/contributor/Fieldy) |
| 11:50.18 | threat | equinox0r, I would like to eventually be able to add in an entry to LDAP to create a new extention for my users, but I guess that is for the future :) |
| 11:50.32 | equinox0r | i dont know, sorry ;) |
| 11:51.14 | equinox0r | but seems to work |
| 11:51.14 | equinox0r | http://www.voip-info.org/wiki/index.php?page=Asterisk+LDAP |
| 11:52.02 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
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| 11:57.24 | appelza | Is there any way Asterisk would send voicemail notifications to only certain domains? |
| 12:02.25 | putzz | if you add the emails of the certain domains under the users yes |
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| 12:20.55 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
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| 12:20.58 | *** mode/#asterisk [+o Qwell] by ChanServ |
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| 12:29.33 | appelza | putzz: thanks, I mean..would asterisk have any 'say' in which email addresses I use? like a list of allowed-domains or something |
| 12:29.46 | appelza | (dont see any reason why it would, but just curious) |
| 12:29.54 | putzz | nop |
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| 12:32.55 | appelza | sweet |
| 12:32.56 | appelza | tnx |
| 12:33.57 | voltagex | can I get festival to read from an asterisk variable? Festival() causes an error that the argument must be text if I try Festival(${var}) |
| 12:35.44 | *** join/#asterisk EzWay (n=ez@c66.110.149-45.clta.globetrotter.net) |
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| 12:38.42 | putzz | voltagex did u put it in quotes? |
| 12:38.57 | voltagex | putzz: doesn't seem to read anything |
| 12:40.35 | putzz | have u tried: Festival('${var}') ? |
| 12:40.35 | putzz | hmm |
| 12:40.35 | putzz | so it doesn read anything not even plain text? |
| 12:40.35 | *** join/#asterisk af_ (n=getsmart@81-174-45-50.f5.ngi.it) |
| 12:40.47 | threat | G'day |
| 12:40.59 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
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| 12:41.56 | threat | i am having problems dialing a iax extension, I am getting a 603 declined message from my asterisks box |
| 12:42.47 | voltagex | putzz: just reads ${var} out |
| 12:42.51 | bkw_ | I wasn't aware IAX gave 603's back |
| 12:42.58 | voltagex | putzz: not the contents of {var} |
| 12:43.05 | bkw_ | putzz, try app_ceptral much better |
| 12:43.05 | voltagex | bkw_: he means his softphone |
| 12:43.17 | threat | bkw_, yes my softphone using SIP :) |
| 12:43.20 | bkw_ | what softphone? |
| 12:43.23 | threat | x-lite |
| 12:43.24 | bkw_ | and what does the debug say? |
| 12:43.26 | voltagex | bkw_: you mean me, I have no money for cepstrel |
| 12:43.30 | bkw_ | x-lite and eyebeam are pure crap |
| 12:43.45 | threat | bkw_, what do you suggest? |
| 12:43.48 | bkw_ | I can't believe they actually try to sell that crap |
| 12:44.02 | putzz | cepstral is free if u know how to use it!! ;-) |
| 12:44.03 | bkw_ | threat, what OS? |
| 12:44.08 | putzz | I use cepstral for everything |
| 12:44.21 | bkw_ | putzz, you just dot out the text in the .so with blanks |
| 12:44.27 | threat | bkw_, linux |
| 12:44.33 | voltagex | putzz: yes I could generate the wav on the website, but that defeats the purpose |
| 12:44.45 | bkw_ | threat, you like command line voip clients or gui? |
| 12:46.57 | putzz | voltagex: http://www.oldskoolphreak.com/tfiles/voip/installing_app_cepstral.txt |
| 12:47.12 | voltagex | haha osp |
| 12:47.13 | bkw_ | that app_cepstral is ass |
| 12:47.30 | voltagex | all I want is damn TTS |
| 12:47.30 | bkw_ | http://www.freeswitch.org/node/50 |
| 12:47.38 | bkw_ | that one on that page uses the streaming API |
| 12:47.40 | voltagex | and I don't want to have to mortgage my house to pay for it |
| 12:47.44 | bkw_ | and doesn't require you to write temp files |
| 12:47.55 | threat | bkw_, gui |
| 12:48.14 | *** join/#asterisk navigo (n=navigo@adsl-074-239-053-186.sip.gnv.bellsouth.net) |
| 12:48.24 | bkw_ | threat, can't tell you what your choices are.. I use freeswitch as my VoIP client daily |
| 12:48.36 | Dovid | ,. |
| 12:48.53 | voltagex | bkw_: I doubt it's his client as I have a similar setup with no trouble |
| 12:49.17 | voltagex | bkw_: he's mostly got the default configs from 1.2 except for a few changes like adding users and extensions |
| 12:49.22 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 12:49.27 | bkw_ | well go fix him.. |
| 12:49.39 | bkw_ | :P |
| 12:50.06 | bkw_ | the MWI message can crash both xlite and eyebeam :P |
| 12:50.07 | voltagex | bkw_: typo! |
| 12:50.39 | bkw_ | it has to do with \n vs. \r\n in the packet |
| 12:50.53 | voltagex | bkw_: I have cepstral_lawrence and that one you linked me to |
| 12:51.00 | voltagex | now what? |
| 12:51.33 | bkw_ | it has a make file |
| 12:51.35 | voltagex | oh, and I'm using 1.4 |
| 12:51.46 | bkw_ | oh it'll have to be updated for 1.4 |
| 12:51.48 | bkw_ | its on 1.2 now |
| 12:52.05 | voltagex | :/ |
| 12:52.07 | voltagex | f*** |
| 12:52.18 | bkw_ | funny part about that module is its BSD |
| 12:52.29 | *** join/#asterisk mquin (n=mike@pdpc/supporter/active/mquin) |
| 12:52.40 | bkw_ | I think it was the first BSD module for asterisk |
| 12:53.10 | voltagex | wtf, what is the Dial() line for 1.2? |
| 12:53.19 | bkw_ | voltagex, I might get that updated for 1.4 today and posted |
| 12:53.29 | voltagex | bkw_: exten => 500,1,Dial(IAX/home.voltagex.org) |
| 12:53.32 | voltagex | doesn't work in 1.2 |
| 12:53.36 | voltagex | ok thanks |
| 12:54.26 | voltagex | in 1.2 you get Apr 10 22:52:22 WARNING[31588]: channel.c:2597 ast_request: No channel type registered for 'IAX' |
| 12:55.01 | putzz | IAX2 |
| 12:55.04 | putzz | ? |
| 12:55.16 | voltagex | is it? |
| 12:55.17 | zeeesh | what does it means .. "WARNING[5494]: codec_ilbc.c:175 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)?"???? |
| 12:55.23 | putzz | yes |
| 12:55.30 | voltagex | putzz: that changed in 1.4? |
| 12:56.00 | voltagex | or not |
| 12:56.00 | putzz | I dont use 1.4 |
| 12:56.06 | voltagex | don't drink and drive asteriks |
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| 12:56.29 | putzz | IAX2 is the only one |
| 12:56.45 | voltagex | yeah, I just can't think today |
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| 13:07.14 | docelmo | I allready fix app_cepstral for 1.4 |
| 13:07.19 | docelmo | its on the wiki.. |
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| 13:12.28 | equinox0r | how can i do this in mysql realtime ? exten => _32[59]X|1|ip.ad.re.ss|{EXTEN} |
| 13:13.05 | equinox0r | because there is no command in this extension .. leave it empty? |
| 13:14.25 | [TK]D-Fender | What is the point of an entry with no valid app? |
| 13:14.48 | equinox0r | i dont know which app i should use |
| 13:15.08 | equinox0r | i want all numbers from 3250 to 3299 to go to another asterisk-server |
| 13:15.24 | voltagex | I need to get Festival to read text from a variable, but it just reads the variable name |
| 13:15.28 | voltagex | how can I get it to work? |
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| 13:24.17 | [TK]D-Fender | equinox0r: And how do you propose the call get there? |
| 13:24.47 | [TK]D-Fender | voltagex: Perhaps you can show us how you're attempting to do this... |
| 13:24.49 | [TK]D-Fender | ~pb |
| 13:24.50 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 13:25.45 | voltagex | [TK]D-Fender: exten => 5,1,Backticks(python /root/netspeed.py|netspeed) |
| 13:25.46 | voltagex | exten => 5,2,Festival('{$netspeed}') |
| 13:27.07 | [TK]D-Fender | voltagex: Sorry but that's not how you reference a variable. http://www.voip-info.org/wiki-Asterisk+variables |
| 13:27.23 | voltagex | [TK]D-Fender: I am teh noob |
| 13:27.47 | [TK]D-Fender | voltagex: Forget "newb"... you're jsut not paying attention or reading the big print :) |
| 13:27.55 | *** join/#asterisk kovger (n=kovger@217.27.219.154) |
| 13:28.01 | voltagex | err |
| 13:28.09 | voltagex | I can't see what I'm missing |
| 13:28.22 | voltagex | oh |
| 13:28.23 | voltagex | shite |
| 13:28.48 | anonymouz666 | If I have a background() inside a loop, after enter a digit, this is will cause to jump to some extensions outside loop but in the same context? |
| 13:29.25 | [TK]D-Fender | anonymouz666: I guess we'd have to see exactly how you're doing this "loop". |
| 13:29.36 | [TK]D-Fender | anonymouz666: Pastebin is your friend... |
| 13:31.26 | voltagex | [TK]D-Fender: for the output of backticks to be passed to a variable, does the program have to print to stdout or stderr? |
| 13:31.44 | anonymouz666 | [context] while(), background(), endwhile, extension1, extension2, ... |
| 13:31.51 | equinox0r | [TK]D-Fender, this i dont know .. im a n00b to asterisk and the dialplan confuses me more than java programming |
| 13:32.35 | [TK]D-Fender | voltagex: PASTEBIN. |
| 13:32.40 | [TK]D-Fender | equinox0r: http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers |
| 13:33.11 | [TK]D-Fender | equinox0r: Try not to invent syntax out of thin air and go read up on that link to see how you can send calls between servers. |
| 13:33.27 | anonymouz666 | [TK]D-Fender, the answer is... if I type something in background() this will cause to jump to extension1 if it is the case |
| 13:33.30 | [TK]D-Fender | equinox0r: When you're done with that consider this as well : "show application transfer" |
| 13:33.32 | threat | ok, so a ATA links normal telephones to a VoIP service? |
| 13:33.54 | [TK]D-Fender | anonymouz666: I want to see EXACTLY how you're doing this. Pastebin it. |
| 13:33.55 | equinox0r | [TK]D-Fender, i've read this twice now and i have both peers and users configured on both systems |
| 13:34.54 | *** join/#asterisk friedrich| (n=friedric@e177240229.adsl.alicedsl.de) |
| 13:35.02 | [TK]D-Fender | equinox0r: Good, then you should be ready to do the last little step or two to make use of what you say you've done. |
| 13:36.23 | voltagex | [TK]D-Fender: http://pastebin.ca/432819 |
| 13:36.52 | equinox0r | [TK]D-Fender, whats the last step(s)? |
| 13:37.24 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 13:38.00 | [TK]D-Fender | equinox0r: it would help if you **DIAL** to the other server... |
| 13:38.17 | equinox0r | ok .. ill give it a try |
| 13:38.32 | [TK]D-Fender | voltagex: That is not an AGI from what I can tell, and it does NOT set a variable that can be returned to * |
| 13:38.48 | voltagex | [TK]D-Fender: I never said it was an AGI |
| 13:39.07 | *** join/#asterisk Waverly360 (n=irc@209.12.249.243) |
| 13:39.12 | [TK]D-Fender | voltagex: And what makes you think it can set an * variable then? |
| 13:39.39 | voltagex | errr because it isn't, backticks should do it |
| 13:40.00 | equinox0r | [TK]D-Fender, exten => _32[59]X,1,dial(ip.ad.re.ss),${EXTEN} ? |
| 13:40.50 | [TK]D-Fender | equinox0r: "show application dial" and go read that guide again |
| 13:41.11 | [TK]D-Fender | voltagex: Please pastebin the instructions for that app. |
| 13:41.42 | voltagex | [TK]D-Fender: typo in the arguments for backticks |
| 13:41.54 | [TK]D-Fender | voltagex: Congratulations. |
| 13:42.07 | [TK]D-Fender | voltagex: New hope arises |
| 13:42.12 | voltagex | it does |
| 13:42.33 | voltagex | except backticks isn't actually available for 1.4 |
| 13:42.35 | voltagex | :/ |
| 13:42.48 | [TK]D-Fender | voltagex: I don't have it in my 1.2 install either... |
| 13:43.02 | voltagex | [TK]D-Fender: addon |
| 13:43.10 | [TK]D-Fender | voltagex: And look at this : |
| 13:43.24 | [TK]D-Fender | -- Executing [5@menu:1] Answer("SIP/0-081e2788", "") in new stack |
| 13:43.26 | [TK]D-Fender | -- Executing [5@menu:2] Festival("SIP/0-081e2788", """") in new stack |
| 13:43.27 | equinox0r | [TK]D-Fender, umm .. so i take the iax2-type/name for the server like this? Dial(IAX2/1u1-MTB),${EXTEN} ? |
| 13:43.38 | [TK]D-Fender | voltagex: Funny its not even CALLING BACKTICKS |
| 13:43.50 | [TK]D-Fender | voltagex: Having 2 extens with the same priority is BAD <- |
| 13:43.54 | voltagex | [TK]D-Fender: yes, I completely messed that up |
| 13:43.56 | [TK]D-Fender | voltagex: Go get some coffee |
| 13:44.00 | voltagex | yeah |
| 13:44.03 | voltagex | or sleep |
| 13:44.13 | voltagex | bye, thanks |
| 13:45.04 | [TK]D-Fender | equinox0r: Go read Dial's instructions again, you are clearly not getting how the line is supposed to be formated |
| 13:46.05 | anonymouz666 | ~pb |
| 13:46.18 | jbot | i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 13:46.18 | anonymouz666 | I forgot the damn URL |
| 13:46.50 | equinox0r | [TK]D-Fender, i meant Dial(IAX2/1u1-MTB,${EXTEN}) ^^ but i see that the only thing you can tell me is rtfm and that is what i do the whole day long .. ^^ |
| 13:49.05 | anonymouz666 | [TK]D-Fender here it goes... http://pastebin.ca/432841 |
| 13:51.22 | anonymouz666 | the loop populate the data and is used by mysql() to insert into db |
| 13:52.19 | *** join/#asterisk punjab (n=punjab@233-9.in-addr.fiber.cz) |
| 13:52.53 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:53.07 | punjab | hi |
| 13:53.47 | punjab | I try make IVR menu in asterisk with examples from google and voip info |
| 13:54.07 | punjab | I get my welcome message |
| 13:54.33 | punjab | but seems like asterisk dont detect if i pres 1 on phone |
| 13:54.42 | punjab | i get timeout |
| 13:54.54 | [TK]D-Fender | equinox0r: the second parameter of dial is TIMOUT. that should be a "/", not a "," |
| 13:54.55 | *** join/#asterisk mathai (n=root@dvere.psg.sk) |
| 13:55.39 | [TK]D-Fender | equinox0r: the target # is part of the tech string and appeared as such in all of the examples |
| 13:57.23 | *** join/#asterisk DarKnesS_WolF (n=sherif@unaffiliated/sherif) |
| 13:57.45 | *** join/#asterisk MuteThis (i=bdill@cpe-24-209-209-97.cinci.res.rr.com) |
| 13:57.59 | *** join/#asterisk littleball (n=littleba@bb220-255-71-61.singnet.com.sg) |
| 13:58.52 | [TK]D-Fender | equinox0r: " exten => _7XXX,1,Dial(IAX2/serverB/${EXTEN:1},30,r) " <- right off Example 2 on the link I gave you |
| 13:59.24 | MuteThis | i was just shown the Asterisk project and I have a couple of capabilities questions. Does asterisk support multiple users while doing outbound predictive dialing? |
| 13:59.31 | tzanger | MuteThis: yep |
| 13:59.43 | tzanger | MuteThis: predictive dialing is handled "externally" though |
| 13:59.47 | littleball | hello, is it possible to get DTMF tone before answer the call? |
| 13:59.49 | tzanger | vicidial is one, but there are others as well |
| 13:59.55 | [TK]D-Fender | anonymouz666: So what is this SUPPOSED to do? |
| 14:00.16 | anonymouz666 | [TK]D-Fender: get the data through this loop and build a list inserting into db |
| 14:00.22 | MuteThis | tzanger: can you be more specific about handled externally? Do i write a custom module or something? |
| 14:00.25 | tzanger | littleball: nope. one-way audio (early audio) is supported on most digital circuits but that's from Asterisk out, not the other way |
| 14:00.31 | tzanger | MuteThis: check out vicidial |
| 14:00.34 | anonymouz666 | [TK]D-Fender: get and background(data) |
| 14:01.10 | [TK]D-Fender | anonymouz666: that should be PLAYBACK, not BACKGROUND. But that aside, what part isn't working? |
| 14:01.19 | littleball | tzanger, i am using ISDN/PRI |
| 14:01.21 | MuteThis | tzanger: thanks, will do |
| 14:01.37 | littleball | it is digital |
| 14:01.45 | tzanger | littleball: well, as I said, for an incoming call to asterisk, you (asterisk) can send audio, but not receive it |
| 14:01.55 | anonymouz666 | [TK]D-Fender: if i use playback how can I will handle the 'i' extension? |
| 14:02.05 | [TK]D-Fender | anonymouz666: exten => s,n,Background(${CUT(stage5_states,\,,${j})}.gsm) <- and you never ever put the file type extension in playback/background |
| 14:02.30 | anonymouz666 | ok, fixed. |
| 14:02.38 | tzanger | littleball: I do it all the time; exten => _X.,1,Playback(num-i-have), exten => _X.,n,SayDigits(${EXTEN}) exten => _X.,n,Playback(vm-goodbye) exten => _X.,n,Hangup |
| 14:02.43 | [TK]D-Fender | anonymouz666: You aren't even LOOKING for input. Look at that pattern match down below. All you do is jump OUT after 2 digits. on "i" you just HANGUP. there IS no "i" |
| 14:02.59 | tzanger | wtf |
| 14:03.02 | tzanger | stage5_states? |
| 14:03.08 | blitzrage | its a variable name |
| 14:03.09 | [TK]D-Fender | anonymouz666: Right now it just loops and does its thing, then sits around and does nothing |
| 14:03.23 | anonymouz666 | its not ready yet |
| 14:03.35 | anonymouz666 | but I can't use playback() I think |
| 14:03.56 | *** join/#asterisk MikHell (n=michel@c-71-56-231-165.hsd1.co.comcast.net) |
| 14:04.12 | MikHell | Hi |
| 14:04.29 | anonymouz666 | but this background will work? backgroud(press-1) and outside loop exten => 1,1,blah() |
| 14:04.30 | [TK]D-Fender | anonymouz666: what does this succeed in doing so far? |
| 14:04.49 | [TK]D-Fender | anonymouz666: You can't do ANYTHING until the loop has ended. |
| 14:05.07 | [TK]D-Fender | MikHell: Not the place to ask. We don't do GUI;s here. |
| 14:05.09 | anonymouz666 | so the logic is totally wrong |
| 14:05.35 | [TK]D-Fender | anonymouz666: Well your "ivr" options are totally useless. |
| 14:05.41 | MikHell | [TK]D-Fender: OK :) That's a valid answer to the second part. Now for the first part? |
| 14:05.43 | [TK]D-Fender | anonymouz666: all you have is : exten => _XX.,1,Goto(dynamic_extensions,s,1) |
| 14:05.48 | MikHell | 1.2 or 1.4? |
| 14:05.51 | *** join/#asterisk hijacked (i=0SLL@66.255.220.17) |
| 14:05.58 | [TK]D-Fender | anonymouz666: and you set no timeouts, ahave no invalid handler or anything. |
| 14:05.58 | MuteThis | right now, we have a call center that is using a cati dialer package from spss, sms, for sample management, will that function with the asterisk & vicidial? (i know a left field question, sorry) |
| 14:06.29 | [TK]D-Fender | MikHell: Most people who actually depend on * are still using 1.2 |
| 14:06.39 | MikHell | [TK]D-Fender: Which one do YOU use? |
| 14:07.16 | threat | I am happy |
| 14:07.20 | anonymouz666 | [TK]D-Fender: that is my big challange. to build a dynamic list, read the value and jump to the digit typed |
| 14:07.21 | threat | asterisk is working fine |
| 14:07.23 | [TK]D-Fender | MikHell: at home I just upgraded yesterday to 1.4 because of the new devstate and SLA stuff I'm looking to test and help debug. At work and all but one of my clients, 1.2 |
| 14:07.29 | threat | now, what else should I add in? ) |
| 14:07.31 | threat | :) |
| 14:07.58 | [TK]D-Fender | anonymouz666: Well you have no extens to dial in there. All you do is jump out after a 3+digit entry. |
| 14:08.32 | MikHell | [TK]D-Fender: I am starting a new inst at home. Played with 1.4 for a while but now I am doing a fresh inst. I am using a VoIP Grandstream phone and a Linksys ATA. What would you recommend? |
| 14:08.57 | [TK]D-Fender | MikHell: Burn the GS, bury it, and salt the earth. |
| 14:09.15 | [TK]D-Fender | MikHell: Everything else is secondary. |
| 14:09.31 | MikHell | [TK]D-Fender: Why? I've played with it and it is pretty good so far. |
| 14:09.38 | [TK]D-Fender | ~gs |
| 14:09.40 | jbot | extra, extra, read all about it, gs is South Georgia and the South Sandwich islands, or ghostscript. GrandSuck phones are cheap junk which should be avoided with extreme prejudice |
| 14:10.43 | MikHell | [TK]D-Fender: So 1.2 or 1.4 does not matter? |
| 14:11.58 | *** join/#asterisk iCEBrkr (i=icebrkr@cyberdyne.org) |
| 14:12.50 | [TK]D-Fender | MikHell: dEPENDS IF YOU NEED ANYTHING IN 1.4 SPECIFICALLY. |
| 14:12.58 | [TK]D-Fender | MikHell: I'm using 1.4 ok so for but I wouldn't put a business on it yet. |
| 14:13.18 | *** join/#asterisk dakmatt (n=dakmatt@60.50.59.82) |
| 14:13.37 | threat | hi |
| 14:13.43 | dakmatt | hi |
| 14:13.52 | MikHell | [TK]D-Fender: I don't know. It's just for my small home phone system. What's in 1.4 that's not in 1.2? |
| 14:14.30 | [TK]D-Fender | MikHell: You'll just have to read the articles and changelogs for all that.... |
| 14:16.27 | anonymouz666 | [TK]D-Fender I can do this loop easily if I use playback and after a background() but if I use playback() the customer will need to listen all the options before type the choosen one |
| 14:16.34 | blitzrage | I'm building our new distributed vPBX platform on 1.4, but I've been working on it for 4 months and have 5 years of experience with Asterisk |
| 14:17.07 | blitzrage | for a home system, there probably isn't really anything in 1.4 that you *really* need |
| 14:17.14 | blitzrage | especially if you're new to Asterisk |
| 14:17.29 | MuteThis | what sample management are you guys using? |
| 14:18.19 | blitzrage | we built our own |
| 14:18.54 | blitzrage | oh... you really meant sample management... and not system management :) |
| 14:20.00 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 14:20.59 | xheliox | WARNING[9155]: chan_sip.c:3089 update_call_counter: Inringing for peer '6337' < 0 ---- I'm using realtime, I presume there's a call count field I'm probably missing in the database? Anyone know what it is (before I start guessing or looking at code)... |
| 14:21.22 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-221315.home.otenet.gr) |
| 14:21.37 | *** join/#asterisk KryoStoffer (n=kri@helium.kri.dk) |
| 14:21.39 | blitzrage | call-limit maybe? |
| 14:22.01 | [TK]D-Fender | anonymouz666: well right now they don't even HAVE options as to what to dial. |
| 14:22.58 | xheliox | blitzrage: I'm using call-limit, I suspect that's why it's looking to update the # of calls? |
| 14:23.09 | blitzrage | possibly? :) |
| 14:23.14 | blitzrage | I've avoided it honestly |
| 14:23.20 | blitzrage | I use GROUP() and GROUP_COUNT() |
| 14:23.32 | blitzrage | I don't like not having control over separate incoming and outgoing call limits |
| 14:23.55 | xheliox | Yeah, ring-inuse = yes in queues.conf for SIP devices requires call-limit... |
| 14:24.04 | dakmatt | guys, i got this error while compiling, make[1]: Entering directory `/usr/src/asterisk/sounds' |
| 14:24.04 | dakmatt | gzip: stdin: unexpected end of file |
| 14:24.04 | dakmatt | tar: Unexpected EOF in archive |
| 14:24.04 | dakmatt | tar: Error is not recoverable: exiting now |
| 14:24.05 | dakmatt | make[1]: *** [/var/lib/asterisk/moh/.asterisk-moh-freeplay-wav] Error 2 |
| 14:24.07 | dakmatt | make[1]: Leaving directory `/usr/src/asterisk/sounds' |
| 14:24.09 | dakmatt | make: *** [datafiles] Error 2 |
| 14:24.10 | xheliox | It's not arbitrary. :) |
| 14:24.38 | dakmatt | pwd |
| 14:24.44 | xheliox | erm, ringinuse = no, I mean. :) |
| 14:24.51 | anonymouz666 | [TK]D-Fender: yeap but when doing the loop if the option 1 is the choice, i will have to listen all the others options |
| 14:24.51 | blitzrage | dakmatt: yah -- the whole .tar.gz didn't download |
| 14:24.52 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:24.57 | *** join/#asterisk CosmicRay (n=jgoerzen@gatekeeper.excelhustler.com) |
| 14:25.07 | dakmatt | ok...thank you. I will download them |
| 14:25.27 | CosmicRay | does anybody know of an ATA other than the SPA-3000/SPA-3102 that has FXS and FXO ports, and will automatically bridge them together when the power is out or when the asterisk server is down? do the tdm400p cards do that? |
| 14:26.26 | kumbalae | how will i found a difference between spoofed callerid and the original one ? |
| 14:26.42 | *** join/#asterisk Ebola (n=Ebola@host86-136-130-202.range86-136.btcentralplus.com) |
| 14:26.42 | blitzrage | very carefully |
| 14:26.43 | Mercestes | ANI should tell you |
| 14:27.26 | hijacked | CosmicRay: the ISP I work for uses a zoom x5v IAD that does that. |
| 14:27.26 | blitzrage | assuming its coming from the PSTN |
| 14:28.04 | CosmicRay | mihinomenest: does that seem to work reliably? |
| 14:28.29 | CosmicRay | mihinomenest: and with no echo problems? |
| 14:28.39 | mihinomenest | no echo problems. |
| 14:28.56 | mihinomenest | but, it doesn't support DTMF via SIP INFO, so none of our customers can check voicemail. |
| 14:29.06 | mihinomenest | it also has serious problems with call waiting. |
| 14:29.17 | mihinomenest | ...and it requires a DSL circuit. |
| 14:29.22 | CosmicRay | hrm. |
| 14:29.35 | CosmicRay | I really want it to talk to the asterisk server on my lan |
| 14:29.40 | [TK]D-Fender | anonymouz666: Well go make your menu choices and come back |
| 14:29.46 | CosmicRay | I do have dsl but probably would want to use my existing modem |
| 14:30.28 | mihinomenest | it doesn't really work well unless you're using it as a NAT Router and it's NAT options are arcaic and assinine. |
| 14:30.37 | mihinomenest | so, you probably don't want to use it. |
| 14:31.34 | [TK]D-Fender | CosmicRay: Shats your problem with the SPA-3XXX series? |
| 14:31.39 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:34.47 | anonymouz666 | [TK]D-Fender: http://pastebin.ca/432909 |
| 14:34.49 | CosmicRay | [TK]D-Fender: I haven't been able to make the echo cancellation work right, ever. |
| 14:34.51 | anonymouz666 | new sutff |
| 14:35.39 | CosmicRay | [TK]D-Fender: I've got the spa-3000 but the spa-3102 is said to be the same. apparently they just have really crappy echo cancelers. other than that, I'm happy with 'em. they have all the features I want. config is a little weird and finicky, but it works. |
| 14:36.28 | CosmicRay | mihinomenest: it looks like it has an FXS port only? or can it use the FXO port for both the DSL and voice tie-in? |
| 14:37.05 | [TK]D-Fender | CosmicRay: It can be a little hi-or-miss, but there are ways to work on the echo. Go check out the forums at www.voxilla.com . They have a LOT of articles on how to tweak these 2 units |
| 14:37.23 | [TK]D-Fender | CosmicRay: And they are the only show in town for anything decent that doesn't cost a fortune. |
| 14:37.35 | CosmicRay | [TK]D-Fender: I have. I have played with gain, various combinations of cancellation and suppression, impedance, etc. |
| 14:38.40 | CosmicRay | it either does too little cancellation, or does too much (essentially making the line half-duplex), or one side or the other is too quiet. |
| 14:38.50 | CosmicRay | there seems to be no happy medium where it just sounds good to everyone. |
| 14:39.13 | [TK]D-Fender | CosmicRay: I guess you can hit a certain point where you are just in the worst state to try and work through and for your failover options there aren't a lot of choices. |
| 14:39.14 | *** join/#asterisk yenno (i=yunien@84-72-188-127.dclient.hispeed.ch) |
| 14:39.29 | *** join/#asterisk powerwade (n=wade@ix.wade.hu) |
| 14:39.37 | [TK]D-Fender | CosmicRay: I hate to sat it but maybe the GrandSuck HT-488 might do the job.... |
| 14:39.40 | powerwade | hi |
| 14:40.01 | CosmicRay | [TK]D-Fender: I wondered about that. but yes, their reputation appears to be about what you are saying ;-) |
| 14:40.25 | CosmicRay | I'm willing to spring for a TDM400p if it has failover. |
| 14:40.33 | CosmicRay | but I can't find anything that says that it does |
| 14:40.57 | powerwade | here's a quick and simple question from a newbie: how to make asterisk running from cron (everymorning i have to restart it:) to use colors? the cron file also has TERM var exported... |
| 14:41.06 | [TK]D-Fender | CosmicRay: Nope, NO failover on the TDM |
| 14:41.09 | anonymouz666 | [TK]D-Fender: I think the problem is what I am saying the user will have to wait all the options to choose one |
| 14:41.23 | anonymouz666 | until finish the loop and reachs background() |
| 14:41.25 | [TK]D-Fender | anonymouz666: background may wqork then, switch it back |
| 14:41.40 | yenno | hi, asterisk gives me a "488 not acceptable here" (Insufficient information for SDP (m = 'audio 5061 RTP/AVP 8 0', c = '') -- so whats wrong with this session descriptor? http://pastebin.ca/432926 |
| 14:41.40 | littleball | tzanger, why asterisk cannot receive DTMF before anser the call? I found Read() cmd has one option "noanswer" |
| 14:42.05 | littleball | <PROTECTED> |
| 14:42.09 | tzanger | littleball: well since you seem to know what you're doing, why not try it? |
| 14:42.20 | littleball | tzanger, i tried |
| 14:42.37 | threat | hmmm |
| 14:42.42 | threat | so slow! |
| 14:42.45 | littleball | but why Read() has one option "noanswer'? |
| 14:43.24 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 14:43.38 | *** join/#asterisk wunderkin (n=kev@dslstat-ppp-95.fastq.com) |
| 14:44.13 | threat | does real time priority make a difference? |
| 14:44.19 | *** part/#asterisk d3wayne (n=dwayne@c-68-62-209-143.hsd1.al.comcast.net) |
| 14:45.11 | irule | what is TRUNKMSD? |
| 14:46.41 | [TK]D-Fender | irule: inherently NOTHING. |
| 14:48.06 | *** join/#asterisk djs_2_6 (n=DJS@cpe-071-077-048-198.nc.res.rr.com) |
| 14:48.12 | irule | >s |
| 14:48.57 | irule | I searched it on voip-info and found nothhing and I dont understand it |
| 14:50.49 | threat | wow, asterisk is very overwhelming, where can I find good howtos / reference books from |
| 14:50.56 | blitzrage | ~docs |
| 14:51.27 | jbot | well, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
| 14:52.02 | irule | threat http://www.google.com/custom?hl=en&ie=ISO-8859-1&oe=ISO-8859-1&client=pub-6210650267389726&cof=FORID%3A1%3BGL%3A1%3BS%3Ahttp%3A%2F%2Fwww.voip-info.org%3BL%3Ahttp%3A%2F%2Fwww.voip-info.org%2Fimages%2FVOIP-info.jpg%3BLH%3A20%3BLW%3A100%3BLBGC%3AFFFFFF%3BLP%3A1%3BBGC%3A%23e9ecef%3BLC%3A%230000ff%3BVLC%3A%23663399%3BGFNT%3A%230000ff%3BGIMP%3A%230000ff%3BDIV%3A%23336699%3B&domains=www.voip-info.org&q=*&btnG=Search&sitesearch=www.voip-info.org |
| 14:52.02 | Mercestes | ...holy shit |
| 14:52.19 | threat | insane |
| 14:52.51 | irule | I did not create the voip-info search page lol |
| 14:53.04 | irule | re-type THAT lol |
| 14:53.09 | irule | cc |
| 14:54.27 | Mercestes | what was the point of that, Irule? |
| 14:55.23 | threat | irule, you are crazy |
| 14:55.27 | threat | well I am off to bed |
| 14:55.48 | threat | my extension is now 600 if anyone wants to talk to me :P :P |
| 14:55.53 | irule | good question, I just wanted to help, I did not notice before that the URL was so long :S |
| 14:56.09 | [TK]D-Fender | irule: Perhaps you can SHOW us how it is being used.... |
| 14:56.57 | irule | TRUNKMSD? |
| 14:57.17 | Mercestes | maybe you should post a link referencing it. :P |
| 14:58.07 | [TK]D-Fender | maybe even a PASTEBIN with your code in it! |
| 14:58.34 | [TK]D-Fender | on noes! (c) file |
| 14:58.47 | irule | well this is an unformatted version I guess :D http://google.com/search?q=site%3Avoip-info.org+*&btnG=Google+Search |
| 14:58.55 | Mercestes | irule: that was anticlimactic. |
| 14:58.58 | *** join/#asterisk ReD-MaN (n=redman@CPE0002b38bce8b-CM0018c0b357cc.cpe.net.cable.rogers.com) |
| 14:59.39 | [TK]D-Fender | Mercestes: Was it good for you? |
| 14:59.55 | Mercestes | [TK]D-Fender, oh yea, baby. |
| 15:01.47 | Mercestes | oh, *your* happy. |
| 15:02.10 | wunderkin | [TK]D-Fender, yo, someone confirmed my suspicions last night... there is a bug in the software for the ip430s.. hopefully it will be fixed in the next release.. nothing you can do to workaround except.. not make a call... |
| 15:02.49 | irule | I am happy because I know that you are happy with my fix |
| 15:02.50 | [TK]D-Fender | wunderkin: Go play some Simon & Garfunkel tunes ;) |
| 15:02.59 | wunderkin | heh |
| 15:05.24 | coppice | or the Kink's Tired of Waiting |
| 15:05.39 | coppice | or Blondie's Hanging on the Telephone |
| 15:06.02 | coppice | what did Simon and Garfunkel offer the pissed off telephone user? |
| 15:06.20 | [TK]D-Fender | coppice: "Dound of SILENCE" ;) |
| 15:06.21 | [TK]D-Fender | Sound*( |
| 15:06.27 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 15:06.36 | coppice | Oh, yeah. forgot that one |
| 15:06.42 | [TK]D-Fender | AKA don't touch that phone :) |
| 15:07.02 | *** join/#asterisk Fieldy (i=I6XAGpEO@gentoo/contributor/Fieldy) |
| 15:08.49 | [TK]D-Fender | wunderkin: Got a link confirming the nature of the bug? |
| 15:09.35 | wunderkin | no... |
| 15:10.46 | wunderkin | i guess you would say it is an unofficial confirmation |
| 15:11.00 | wunderkin | from someone that goes bump in the night |
| 15:11.39 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 15:13.18 | docelmo | oi! |
| 15:14.06 | threat | hmmm |
| 15:15.05 | [TK]D-Fender | You know, yesterday's "tragic server death" was really the kick in the ass I needed to upgrade... now all I need to do is figure out my HDTV's ModeLine's |
| 15:15.21 | [TK]D-Fender | threat: Yeah.... thats really going to hurt..... YESTERDAY ;) |
| 15:15.35 | [TK]D-Fender | threat: Adn that'd be tachions ;) |
| 15:15.57 | [TK]D-Fender | threat: Hopefully you are capable of ID-ing my earlier reference... |
| 15:16.51 | [TK]D-Fender | "tackions" <- particles emitted by whatever crime-against-fashion Lindsay Lohan is sporting? |
| 15:17.59 | Mercestes | ROFL! Tackions. |
| 15:19.25 | threat | [TK]D-Fender, :) |
| 15:19.45 | threat | [TK]D-Fender, I forgot |
| 15:19.50 | [TK]D-Fender | i r gud :) |
| 15:20.12 | threat | [TK]D-Fender, or next week! :) |
| 15:20.13 | *** part/#asterisk littleball (n=littleba@bb220-255-71-61.singnet.com.sg) |
| 15:20.45 | threat | [TK]D-Fender, it sounds like a reference from a cartoon |
| 15:21.12 | [TK]D-Fender | threat: Indeed there, but movies prior... |
| 15:21.56 | threat | [TK]D-Fender, hmmm ok |
| 15:22.17 | threat | [TK]D-Fender, too old for me :) |
| 15:22.45 | threat | neither |
| 15:22.52 | threat | I seriously dont remember |
| 15:23.00 | Mercestes | and i'm relatively old |
| 15:24.16 | Mercestes | is that like a "Buck Rogers and the 24 and a half century." |
| 15:24.44 | coppice | that should be Duck Dodgers in the 24th and a half century |
| 15:25.20 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 15:25.21 | *** mode/#asterisk [+o denon] by ChanServ |
| 15:27.00 | *** join/#asterisk icel (n=dan@65.200.26.49) |
| 15:27.06 | mkl1525 | Hi, trying to hide my phone number when using the "Hide clid" feature of my snom360 - but I always get my real name and number so does anybody know how to do this with snoms? |
| 15:27.37 | drako | can you put on voicemail more than one email for an account? |
| 15:28.34 | [TK]D-Fender | Mercestes: hint : it was in reference to "bump in the night". |
| 15:29.01 | [TK]D-Fender | mkl1525: Are you setting CLID in sip.conf? |
| 15:29.11 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 15:29.59 | mkl1525 | [TK]D-Fender, yes, I'm setting the callerid in sip.conf |
| 15:30.20 | [TK]D-Fender | mkl1525: that overrides anything the phone feels like telling * and is why you can't block it. |
| 15:30.45 | mkl1525 | [TK]D-Fender, thanks for the hint will try without it |
| 15:32.08 | *** join/#asterisk `p4r14h`work (n=josh@24-119-48-78.cpe.cableone.net) |
| 15:32.19 | icel | how do you register zap channels? I have a voice T-1, ztcfg sees 24 channels (23 voice) but When I try to dial I get 'No channel type registered for 'Zap' ' |
| 15:32.42 | mkl1525 | [TK]D-Fender, thanks - it's the solution! |
| 15:33.52 | [TK]D-Fender | icel: Sounds like you compiled Zaptel without recompiling * after |
| 15:34.01 | [TK]D-Fender | mkl1525: ywc |
| 15:34.24 | icel | D-Fender: so I need to recompile asterisk and it may magically work? |
| 15:35.43 | [TK]D-Fender | icel: Its not "magic", but it appears * knows nothing of Zaptel and thats the common reason. |
| 15:36.00 | *** join/#asterisk _VoicemeUpDotCom (n=Voice2@145-27.mc.cite.net) |
| 15:36.00 | icel | thx,thx, I will give it a try |
| 15:36.16 | punjab | hello. Can somebody help me with IVR script? I am trying script from this page: http://users.pandora.be/Asterisk-PBX/IVR.htm. I get welcome message. When I pres number 1 on cell phone i dont get to secr message. |
| 15:39.33 | *** join/#asterisk ManxPower (n=manxpowe@210.sub-70-222-6.myvzw.com) |
| 15:42.56 | Mercestes | [TK]D-Fender, I still don't get it. I guess I'm just off today |
| 15:43.24 | ManxPower | Bell says I have a groundfault on in the CPE side. |
| 15:43.57 | [TK]D-Fender | Mercestes: Ghostbusters <- |
| 15:44.12 | Mercestes | those weren't proton packs. |
| 15:44.35 | coppice | isn't groundfault day in february? |
| 15:44.45 | _VoicemeUpDotCom | lol coppice |
| 15:44.50 | [TK]D-Fender | punjab: that is * 1.0 compatable code, and is bad on 1.2, and completely defective on 1.4 Chances are autofallthrough is killing it |
| 15:45.04 | [TK]D-Fender | Mercestes: Yes they were... |
| 15:45.25 | ManxPower | coppice: I admit it is possible. I can't imagine where unless my adtran is bad |
| 15:45.25 | [TK]D-Fender | Mercestes: http://en.wikipedia.org/wiki/Proton_pack |
| 15:46.15 | ManxPower | I wonder if an off-by-one error on the punchdown block could cause that. |
| 15:47.14 | ManxPower | I hate 25 pair cables |
| 15:47.18 | Mercestes | oh, I guess it is |
| 15:47.32 | *** join/#asterisk Fieldy (i=TvNj2E4P@gentoo/contributor/Fieldy) |
| 15:47.49 | punjab | [TK]D-Fender: I try remove SetMusicOnHold. Replace timeouts with Set(TIMEOUT(digit)=5) and Set(TIMEOUT(response)=10) |
| 15:48.02 | punjab | [TK]D-Fender: but still nothing |
| 15:48.16 | [TK]D-Fender | punjab: You need to make sure "autofallthrough=no" is under [globals] |
| 15:48.38 | [TK]D-Fender | punjab: And of course make sure you are using the right dtmfmode for your device you are testing with. |
| 15:48.53 | ManxPower | [TK]D-Fender: why not just add a WaitExten at the end? |
| 15:49.04 | punjab | [TK]D-Fender: I have "yes" on this. Thanks for tips. I try it |
| 15:49.47 | [TK]D-Fender | punjab: OH.. and you can't run IVR's of anything but "s" unless you use the WaitExten app. Even then its not recommended as you can dial the "40" that that sample is running on. |
| 15:49.58 | [TK]D-Fender | punjab: Its author should be dragged out and shot. |
| 15:52.54 | *** join/#asterisk wyoming (n=steve_mu@216.166.159.235) |
| 15:52.54 | *** join/#asterisk kn0x (n=pinochle@c-67-176-194-29.hsd1.il.comcast.net) |
| 15:53.02 | kn0x | goodmorning gents |
| 15:53.16 | kn0x | any one familair with manager api? |
| 15:53.20 | Waverly360 | Does anyone see a problem with this line? I simply want to dial into my pbx with 5551212 and have it dial another number for me. exten => 5551212,1,Dial(Zap/g2/5557098) |
| 15:53.24 | kn0x | im having some trouble passing variables |
| 15:53.45 | [TK]D-Fender | Waverly360: LOOKS fine, but thats dependant on your groups setup and channel definitions |
| 15:53.53 | kn0x | Waverly360- you need to set the context for you incomming line |
| 15:53.56 | ManxPower | Waverly360: You are picking up a phone on your PBX and dialing 5551212? |
| 15:54.02 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 15:54.13 | Waverly360 | ManxPower: No, this is from an outside line |
| 15:54.22 | *** join/#asterisk hal2k (n=am@2002:5470:9fd9:0:0:0:0:1) |
| 15:54.35 | ManxPower | Waverly360: It might help for us to know how Asterisk is connect to the PSTN |
| 15:54.38 | [TK]D-Fender | Waverly360: And what kind of interface is it coming in under? |
| 15:54.53 | Waverly360 | coming in on a PRI, and calling back out on a PRI |
| 15:55.31 | Waverly360 | Here's my problem. |
| 15:55.33 | [TK]D-Fender | Waverly360: Looks fine and it should literally forward out including call-progress. |
| 15:55.34 | ManxPower | Waverly360: That should work as long as g2 is correct, and assuming the telco hands you the 7-digit DID and not a 10-digit DID |
| 15:55.37 | Waverly360 | when I dial 5551212, I get dialtone |
| 15:55.49 | Waverly360 | the setup is this. |
| 15:56.04 | Waverly360 | it's a dual PRI card, with two separate PRIs connected. Both are in g2 |
| 15:56.05 | [TK]D-Fender | Waverly360: pastebin all of your configs and the CLI output of your failed attempt at verbose 10 |
| 15:56.35 | [TK]D-Fender | Waverly360: zaptel, zapata, and the relevent context(s) from extensions.conf |
| 15:57.16 | Waverly360 | [TK]D-Fender: I'll do that if I can't figure it out in the next couple of minutes. I was just curious if anything seemed blatantly obvious. This box *should* be identical to others that we have. So that's why I'm a bit lost... |
| 15:57.40 | ManxPower | Waverly360: Um, if you get dialtone then you are NOT doing what you pasted. |
| 15:57.47 | [TK]D-Fender | Waverly360: You've shown us *1* line of dialplan which I aleady said could be perfectly fine..... |
| 15:57.59 | [TK]D-Fender | Waverly360: So clearly we need MORE |
| 15:58.02 | ManxPower | Waverly360: get it working the way you showed us, then you can make it complicated |
| 15:58.35 | *** join/#asterisk b11d (n=no@234-200-29-134.hcc.mnscu.edu) |
| 15:58.47 | b11d | says hello |
| 15:59.14 | ManxPower | [TK]D-Fender: I may have a bad Adtran |
| 15:59.25 | [TK]D-Fender | ManxPower: z0mg! |
| 15:59.41 | [TK]D-Fender | ManxPower: Emergency eBay session must be called! |
| 15:59.41 | ManxPower | [TK]D-Fender: It would totally suck |
| 16:00.13 | ManxPower | [TK]D-Fender: HA! Emergency paycheck first, then emergency car payoff, then emergency rent payment, then..., then.... |
| 16:00.17 | Waverly360 | [TK]D-Fender: I know you need more configs, and I'll get them if I can't figure it out. I just wanted a quick sanity check to make sure I wasn't doing something blatantly obvious. |
| 16:00.20 | [TK]D-Fender | Would probably be cheaper & easier to just get a TDM card.... |
| 16:00.45 | ManxPower | [TK]D-Fender: I have like 8 TDM cards laying around. |
| 16:00.46 | [TK]D-Fender | ManxPower: You forgot about the part of solving world hunger, and peace :) |
| 16:01.12 | ManxPower | Waverly360: exten => 5551212,1,Dial(Zap/g2/5557098) should work. |
| 16:01.13 | Waverly360 | ManxPower: and yeah, I'm doing exactly what I posted..which is what's weird. I've set up dozens of boxes the same way..this one is different..I can only assume it's because of the weird way their breaking out their PRI. |
| 16:01.15 | b11d | the worst part is, world hunger could actually BE solved. Easily. |
| 16:01.17 | ManxPower | I hate TDM cards |
| 16:01.18 | [TK]D-Fender | Waverly360: well... we have no idea if that LINE is itn the right place or if its contexts are at all valid. |
| 16:01.19 | b11d | but no, we dont. |
| 16:01.32 | [TK]D-Fender | contents* |
| 16:01.35 | kn0x | so anyone know how to send variables in asterisk manager? |
| 16:01.41 | _VoicemeUpDotCom | emergency , poker hand of AA, AKK26 at 20/40$ stakes with all in with 900$ and 9 folloing in |
| 16:01.47 | ManxPower | Waverly360: if you get dialtone with that line you have something seriously screwed up |
| 16:01.58 | [TK]D-Fender | Waverly360: So I guess we'll see you if you can't figure it out on your own.... |
| 16:02.19 | ManxPower | [TK]D-Fender: I don't use TDM cards because they have been so unreliable for me in the past |
| 16:02.21 | [TK]D-Fender | kn0x: Send a variable WHERE? |
| 16:02.25 | Waverly360 | ManxPower: Tell me about it. I didn't actually put the box in, so I haven't physically seen the pri setup..I was just told yesterday that it was pretty different. |
| 16:02.29 | kn0x | on an originate TK |
| 16:02.37 | _VoicemeUpDotCom | Btw... im dialin over TDM400 and get nothing .. ztcvf says connected.. but nothing no audio... SIP has audio on trunk |
| 16:02.46 | ManxPower | Waverly360: watching the CLI output is needed |
| 16:02.49 | Waverly360 | [TK]D-Fender: I'll let you know what was screwed up at any rate... |
| 16:02.51 | [TK]D-Fender | kn0x: I seem to recal a SetVar option.... |
| 16:02.54 | _VoicemeUpDotCom | any way to see packets ? or errors ? full debug shows it does niet |
| 16:03.12 | _VoicemeUpDotCom | no.. SetVar = Set(VAR=BLAH) |
| 16:03.14 | _VoicemeUpDotCom | now |
| 16:03.21 | ManxPower | [TK]D-Fender: IT does not help that the telco routes my line several miles out of the way |
| 16:03.35 | kn0x | [TK]D-Fender: it says the command is| Variable: <Variable>:<Value> |
| 16:03.45 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:03.53 | [TK]D-Fender | ManxPower: Nope... |
| 16:03.56 | kn0x | but ive tried ${VARIABLE} VARIABLE and $VARIABLE |
| 16:03.58 | [TK]D-Fender | BBIAB.. lunch time |
| 16:04.06 | kn0x | but it doesnt make it to the dialplan |
| 16:05.04 | kn0x | anyone? |
| 16:05.12 | punjab | [TK]D-Fender: many thanks. I set dtmfmode = inband ang working |
| 16:05.57 | _VoicemeUpDotCom | oh |
| 16:06.00 | _VoicemeUpDotCom | knox... |
| 16:06.10 | kn0x | yeah? |
| 16:06.12 | _VoicemeUpDotCom | Set(__VAR=BLAH) |
| 16:06.20 | kn0x | from manager? |
| 16:06.31 | _VoicemeUpDotCom | underscore once is per context and 2 __ = global i think |
| 16:06.43 | _VoicemeUpDotCom | when you set it.. then you call as usual ${BLAH} |
| 16:06.45 | ManxPower | kn0x: your searched the mailing list |
| 16:06.51 | _VoicemeUpDotCom | i mean ${VAR} |
| 16:06.53 | kn0x | so its Set: |
| 16:07.05 | ManxPower | kn0x: I highly doubt that Set(__VAR=BLAH) will work in manager |
| 16:07.06 | Corydon-w | No, single underscore is single level of inheritance, double underscore is infinite inheritance |
| 16:07.15 | _VoicemeUpDotCom | hmm |
| 16:07.24 | kn0x | well thats wat im asking |
| 16:07.29 | _VoicemeUpDotCom | ok you want to set in variable ? |
| 16:07.36 | kn0x | i know how to set it in the dialplan |
| 16:07.37 | _VoicemeUpDotCom | in mamanger i mean |
| 16:07.42 | _VoicemeUpDotCom | ok hold ill show you mine |
| 16:07.56 | ManxPower | _VoicemeUpDotCom: He wants to do it using the AMI via a TCP socket into the manager interface. |
| 16:08.27 | kn0x | right |
| 16:08.35 | _VoicemeUpDotCom | kk |
| 16:09.17 | kn0x | http://www.voip-info.org/wiki-Asterisk+manager+API |
| 16:09.45 | kn0x | according to that im supposed to do: Variable: <Variable 1>=<Value 1><CRLF> |
| 16:09.46 | kn0x | <PROTECTED> |
| 16:09.58 | kn0x | so in php, this is what im doing |
| 16:10.28 | kn0x | fputs($socket, "Variable: \${FPHONE}="."123 \r\n"); |
| 16:10.33 | kn0x | ive also tried: |
| 16:10.39 | _VoicemeUpDotCom | <PROTECTED> |
| 16:10.39 | kn0x | fputs($socket, "Variable: {FPHONE}="."123 \r\n"); |
| 16:10.40 | _VoicemeUpDotCom | they said |
| 16:10.47 | kn0x | fputs($socket, "Variable: FPHONE="."123 \r\n"); |
| 16:10.48 | _VoicemeUpDotCom | so replace = by : |
| 16:10.55 | kn0x | ohh shit |
| 16:10.59 | _VoicemeUpDotCom | fputs($socket, "Variable: FPHONE:"."123 \r\n"); |
| 16:11.00 | kn0x | i am stuipid |
| 16:11.05 | _VoicemeUpDotCom | heeh... |
| 16:11.14 | _VoicemeUpDotCom | just a bad reader. for tech subtilities ;) |
| 16:11.22 | kn0x | no wait |
| 16:11.23 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 16:11.26 | kn0x | it says <Variable 2>=<Value 2><CRLF> |
| 16:11.40 | kn0x | <PROTECTED> |
| 16:11.49 | kn0x | @! |
| 16:13.15 | *** join/#asterisk OneWhoKnows (n=OneWhoKn@rh-la-32-99.rhythm.com) |
| 16:13.30 | *** join/#asterisk Jon335 (i=jon335@unaffiliated/jon335) |
| 16:14.15 | *** join/#asterisk GiantPickle (n=GiantPic@S01060016b600537f.gv.shawcable.net) |
| 16:16.56 | kn0x | could i use: ttp://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+SetVar |
| 16:17.00 | kn0x | instead? |
| 16:17.06 | kn0x | what channel would i put it in? |
| 16:17.29 | OneWhoKnows | for the TE412P, is there anything additional that you need to do to get HWEC to work other than echocancel=yes? |
| 16:17.41 | OneWhoKnows | or should it actually be no? |
| 16:20.39 | [TK]D-Fender | OneWhoKnows: that should do it |
| 16:23.28 | ManxPower | OneWhoKnows: if you read the readme it told you |
| 16:23.29 | ManxPower | OneWhoKnows: you need to run the zaphpec_enable every time you load zaptel, as it says in the readme as well |
| 16:23.48 | ManxPower | the readme also tells you how to confirm the HWEC is built and active. |
| 16:24.18 | ManxPower | OneWhoKnows: what specific issue are you having with the HPEC |
| 16:24.32 | wunderkin | not hpec.. HW EC |
| 16:24.48 | *** join/#asterisk Juggie (i=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 16:24.52 | ManxPower | I know, I know |
| 16:25.01 | wunderkin | :-D |
| 16:26.31 | OneWhoKnows | 09:23 < ManxPower> OneWhoKnows: you need to run the zaphpec_enable every time |
| 16:26.31 | OneWhoKnows | ah, i wasn't aware of that |
| 16:26.31 | wunderkin | no no |
| 16:26.36 | *** join/#asterisk [hC] (n=hardcore@S0106000d8891877c.vc.shawcable.net) |
| 16:27.58 | ManxPower | wunderkin: no? |
| 16:28.44 | wunderkin | manx was thinking of something else.. |
| 16:29.11 | ManxPower | wunderkin: what was I thinking of? |
| 16:31.55 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 16:31.55 | ManxPower | gawd I hate punching down 25 pairs |
| 16:31.55 | b11d | haha.. same.. i remember the old days when I loved it though |
| 16:31.55 | OneWhoKnows | hm, zaptel 1.2.12 not support the TE412P's HWEC? |
| 16:31.55 | ManxPower | OneWhoKnows: Um, HPEC software EC, or the hardware EC on a card? |
| 16:31.57 | [TK]D-Fender | ManxPower: HWEC. |
| 16:32.00 | OneWhoKnows | hardware echo cancellation, sorry |
| 16:32.06 | ManxPower | OneWhoKnows: ignore everything I said then |
| 16:32.21 | OneWhoKnows | gotcha |
| 16:32.23 | ManxPower | [TK]D-Fender: I do all my EC outside of Asterisk. |
| 16:32.41 | [TK]D-Fender | ManxPower: You are losing coherence... go fetch some corn-starch fast! |
| 16:32.55 | OneWhoKnows | lol |
| 16:33.10 | OneWhoKnows | should echocancelwhenbridged=yes or no? |
| 16:33.12 | [TK]D-Fender | ManxPower: I know.....I'm Mr. Polycom / Sangoma, you are Mr. Tellabls. Our roles are well defined ;) |
| 16:33.21 | OneWhoKnows | i know that the sangoma cards say to set it to yes |
| 16:33.24 | ManxPower | [TK]D-Fender: Just a bit rattled by being woken up by someone hollering thru my door "Bellsouth is here!" |
| 16:33.40 | [TK]D-Fender | ManxPower: You must have been quaking in disbelief ;) |
| 16:33.44 | ManxPower | [TK]D-Fender: I switched to Sangoma too |
| 16:33.49 | [TK]D-Fender | :O |
| 16:33.51 | JunK-Y | Juggie: hey! |
| 16:33.58 | [TK]D-Fender | ManxPower: the non-ec variety I take it... |
| 16:34.14 | ManxPower | OneWhoKnows: if you get echo on non-VoIP calls set it to yes, otherwise set it to no |
| 16:34.22 | ManxPower | [TK]D-Fender: yeah |
| 16:34.26 | [TK]D-Fender | ManxPower: I <3 my Sangoma S518 ADSL card |
| 16:34.29 | OneWhoKnows | ManxPower: thanks |
| 16:35.02 | b11d | Sangoma rocks |
| 16:35.07 | [TK]D-Fender | I also <# my new CentOS server setup..... |
| 16:35.15 | [TK]D-Fender | <3* |
| 16:35.23 | b11d | i love that they have people keeping WANPIPE up for the BSDs |
| 16:35.30 | [TK]D-Fender | And its newfound ZTDUMMY support! |
| 16:35.36 | ManxPower | [TK]D-Fender: all new installs and any time we upgrade the hardware of an asterisk server we use a 2-port sangoma card |
| 16:35.48 | [TK]D-Fender | now I have to set up some semi-public convference rooms. |
| 16:35.58 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 16:35.58 | *** mode/#asterisk [+o mog] by ChanServ |
| 16:36.20 | [TK]D-Fender | ManxPower: Have you actually tried a HWEC model of theirs? |
| 16:36.47 | ManxPower | [TK]D-Fender: nope. |
| 16:37.30 | b11d | I do! |
| 16:37.31 | [TK]D-Fender | ManxPower: Well worth it, and they now have HWEC on even the 1-ports models. |
| 16:37.32 | ManxPower | [TK]D-Fender: With tellabs we get HW EC for under $100 for up to about 8 T-1s |
| 16:37.33 | b11d | i use an a104d |
| 16:37.34 | b11d | HWEC |
| 16:37.35 | b11d | :) |
| 16:37.46 | b11d | it works without flaw |
| 16:37.49 | [TK]D-Fender | ManxPower: Can you give me a model # for reference? |
| 16:38.00 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 16:38.03 | OneWhoKnows | b11d: what version of zaptel are you using? |
| 16:38.17 | ManxPower | [TK]D-Fender: whatever is talked about on the Wiki |
| 16:38.20 | b11d | i run on freebsd, so i use the svn of zaptel-bsd-trunk |
| 16:38.22 | irule | can someone explain TRUNKMSD to me? thanks |
| 16:38.24 | ManxPower | they are a bitch to set up the first time or two |
| 16:38.48 | [TK]D-Fender | ManxPower: Same can be said for Polycom..... is the documentation up to the task? |
| 16:38.51 | pigpen | What would the simplest agi language to write a simple script to grab some values from postgres, live in the dialplan? I know many like perl... |
| 16:38.56 | ManxPower | I sent doughecka a couple of them once. |
| 16:38.59 | OneWhoKnows | b11d: does freebsd have the lspci command? if so, does it show up as the TE410P? |
| 16:39.06 | [TK]D-Fender | pigpen: probably all equally as easy. |
| 16:39.06 | b11d | no |
| 16:39.07 | Mercestes | b11d!!!! |
| 16:39.08 | b11d | it doesnt |
| 16:39.11 | b11d | Mercestes!@!@!@! |
| 16:39.18 | b11d | turns out that girl I liked is hardcore prego.. |
| 16:39.19 | Mercestes | hi!\ |
| 16:39.20 | ManxPower | [TK]D-Fender: Yes, but the docs are really for CO people |
| 16:39.23 | pigpen | [TK]D-Fender, thanks...I know my business partner likes ruby. |
| 16:39.23 | Mercestes | prego? |
| 16:39.26 | Mercestes | like, expecting? |
| 16:39.28 | b11d | aye |
| 16:39.30 | OneWhoKnows | b11d: it doesn't have the command or it doesn't show up as a TE410P? =D |
| 16:39.32 | b11d | so that was like "uhh.. cya" |
| 16:39.39 | Mercestes | Oh dear. |
| 16:39.40 | Mercestes | Good job tho |
| 16:39.42 | [TK]D-Fender | b11d: And here I was thinking ITALIAN ;) |
| 16:39.43 | b11d | it was a relief |
| 16:39.44 | b11d | :) |
| 16:39.50 | b11d | TK :) |
| 16:39.50 | Mercestes | I'm banned from ...well, that other channel. STILL |
| 16:39.55 | b11d | you're totally in there |
| 16:39.59 | OneWhoKnows | pregnant girls are bad news |
| 16:40.03 | b11d | aye |
| 16:40.04 | b11d | they are!! |
| 16:40.30 | b11d | i so wish I was the father though :) |
| 16:40.36 | [TK]D-Fender | ManxPower: Ugh... we should do something about that. Like a "Tellabs from A-Z for Asterisk" guide |
| 16:40.42 | OneWhoKnows | well, when you're the father it's not as bad haha |
| 16:41.14 | b11d | hey |
| 16:41.18 | ManxPower | [TK]D-Fender: the wiki page is MASSIVLY better then 4 years ago when we first put them in |
| 16:41.20 | b11d | is anyone in here a "beard masters" ? |
| 16:41.27 | b11d | who here wears a full beard.. in the garibaldi style? |
| 16:41.39 | OneWhoKnows | i can hardly grow a mustache >>> asian |
| 16:41.54 | Mercestes | yea, the bad thing about preggy girls is there's a jealous someone out there somewhere. |
| 16:41.58 | OneWhoKnows | b11d: did you have to do anything special to get the HWEC on your card working or was it fairly plug and play? |
| 16:42.12 | Mercestes | yay for azn |
| 16:42.25 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 16:42.41 | b11d | i enabled int in "wancfg" |
| 16:42.47 | b11d | which is part of WANPIPE |
| 16:42.49 | b11d | from Sangoma |
| 16:42.58 | [TK]D-Fender | b11d: Women tend to dislike facial hair..... |
| 16:43.00 | pigpen | Will asterisk run on a Wang VS5000? |
| 16:43.03 | pigpen | heh... |
| 16:43.06 | b11d | no, it actually fits me well. |
| 16:43.06 | OneWhoKnows | it worked on a digium card? interesting |
| 16:43.30 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 16:43.32 | b11d | no |
| 16:43.36 | b11d | i never once said it was digium |
| 16:43.37 | [TK]D-Fender | pigpen: I do not want to hear ANYTHING about your "wang" |
| 16:43.39 | b11d | its a SANGOMA A104D |
| 16:43.49 | pigpen | [TK]D-Fender, yeah...I had to post that.... |
| 16:44.09 | OneWhoKnows | .... wow, how did i read it as a te410p |
| 16:44.09 | b11d | i need to learn how to tame a beard, mine just goes wild.. |
| 16:44.12 | OneWhoKnows | i need coffee |
| 16:44.17 | b11d | come on.. there are NO bearded men in here? |
| 16:44.17 | [TK]D-Fender | I have just enough Metis in me to stop at a respectable stubble :) |
| 16:44.21 | b11d | christ almighty. |
| 16:44.24 | pigpen | OneWhoKnows, count me in. |
| 16:44.34 | b11d | Metis rock! infinity symbol all the way! |
| 16:44.42 | [TK]D-Fender | b11d: You loking for tips? the John Deere look is back in ;) |
| 16:44.54 | OneWhoKnows | pigpen: black or cream and sugar? |
| 16:44.57 | b11d | i dont know what that is |
| 16:45.13 | pigpen | Carmel Machiatto for me... |
| 16:45.14 | *** join/#asterisk lunaphyte (n=lunaphyt@207.106.12.202) |
| 16:45.20 | OneWhoKnows | classy |
| 16:45.25 | Mercestes | Triple Shot Cinnamon Dolce' latte. |
| 16:45.26 | [hC] | I have a beard, but not a full thick ass beard, just that 'unshaven' look |
| 16:45.37 | Mercestes | I have a goatee |
| 16:45.38 | ManxPower | b11d: In the winter I usually have a beard. Go back to a gotee in the summers |
| 16:45.39 | pigpen | tripple shot, vinte (sp?), extra hot. |
| 16:45.46 | b11d | im growing mine out, just need to learn how to "tame" and "control" it.. |
| 16:45.46 | [TK]D-Fender | b11d: I am about 1/16 Micmac which defeats the facial hair gene a great deal.... now if I could get it to stop altogether, but head out & prep time would be greatly benifited :) |
| 16:45.47 | Mercestes | Aye. |
| 16:45.49 | Mercestes | Vente |
| 16:45.51 | b11d | it just grows wild all over the neck.. |
| 16:46.06 | b11d | nice TK :) |
| 16:46.10 | ManxPower | b11d: I never bother with that |
| 16:46.11 | b11d | i'd kill to be naturally hairless :) |
| 16:46.18 | Mercestes | s/ 1\/16 micmac/1\/16 melmacian |
| 16:46.27 | b11d | from the jawline up, it's tamed nicely.. but the neck is all haggard |
| 16:46.33 | Mercestes | s/defeats/promotes/ |
| 16:46.51 | *** join/#asterisk lunaphyte (n=lunaphyt@207.106.12.202) |
| 16:46.52 | [TK]D-Fender | Mercestes: You = ALF (Anal-retentive Life Form) |
| 16:47.11 | Mercestes | ROFL |
| 16:47.24 | b11d | Mercestes? Anal? I knew it! |
| 16:47.24 | b11d | :) |
| 16:47.25 | [TK]D-Fender | Mercestes: pwned |
| 16:47.29 | ManxPower | I want to go back to bed. |
| 16:47.43 | b11d | you run your life. go. |
| 16:47.45 | Mercestes | Yes, I would like to retain my anus, thank you. |
| 16:47.48 | Mercestes | brar |
| 16:48.12 | b11d | yeah so now i'm going to go after this new bartender.. |
| 16:48.19 | b11d | she's only like 20 though.. which is shitty.. also good. |
| 16:49.37 | b11d | is that appropriate here in the USA? 26 going out with 20? |
| 16:49.44 | b11d | ouch :) that sounds wrong as hell. |
| 16:50.19 | coppice | >16 going out with >16 is always OK |
| 16:50.27 | b11d | :) |
| 16:50.28 | b11d | nice thinking |
| 16:50.57 | *** join/#asterisk ToyMan (n=Stuart@74-32-9-170.dsl1.mdl.ny.frontiernet.net) |
| 16:50.58 | [TK]D-Fender | b11d: thats not so huge a difference |
| 16:51.05 | _VoicemeUpDotCom | wahts easiest way to .. if caller = called then vm |
| 16:51.13 | [TK]D-Fender | b11d: I had a LTR with a woman 6 years my senior.... |
| 16:51.22 | coppice | if you get into a relationship, you no longer even think about age |
| 16:51.58 | b11d | yeah, i dont think about age as it is.. its those around me :) |
| 16:52.02 | b11d | also im good friends with her father |
| 16:52.20 | *** join/#asterisk ming_zym (n=ming_zym@124.254.53.139) |
| 16:52.50 | [TK]D-Fender | _VoicemeUpDotCom: assuming a var like ARG1 holds the # dialed : exten => (whatever),1,GotoIf($["${ARG1}"="${CALLERID(num)}"]?10) |
| 16:53.08 | _VoicemeUpDotCom | hmm |
| 16:53.15 | coppice | I'm half way between my wife and her faster's age, but i don't relate to hime at all :-) |
| 16:53.34 | b11d | :) |
| 16:53.43 | b11d | well im going to proceed anwyays |
| 16:53.46 | b11d | we'll see how it goes |
| 16:54.00 | [TK]D-Fender | b11d: Do your tastes and life experiences fit? |
| 16:54.07 | _VoicemeUpDotCom | then i need a macro.. since i cant really use callerid |
| 16:54.10 | _VoicemeUpDotCom | maybe chanel |
| 16:54.14 | b11d | dont know.. we've always been "friends" but never really hung out.. cant tell.. |
| 16:54.39 | coppice | he has a taste for 20 year olds, and she is one. sounds like a fit :-) |
| 16:54.44 | [TK]D-Fender | b11d: Try, find out, do (her) ;) |
| 16:54.46 | b11d | :) |
| 16:54.51 | b11d | i will do that |
| 16:55.07 | [TK]D-Fender | coppice: Shortest path wins again. |
| 16:56.32 | Waverly360 | ok guys, I give.... http://pastebin.ca/433150 |
| 16:58.41 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 17:01.33 | Waverly360 | crap..I forgot cli output |
| 17:02.57 | Defraz | Does anyone have an example config file to change the softbuttons on a cisco 7940 |
| 17:03.00 | Defraz | I can't seem to find any info on it. |
| 17:03.41 | Waverly360 | http://pastebin.ca/433163 |
| 17:04.25 | [TK]D-Fender | Waverly360: You deserately need to start learning to use macro's |
| 17:04.41 | Jon335 | Does anyone have a termination provider that allows you to set the Caller ID Name? |
| 17:04.50 | [TK]D-Fender | Waverly360: and that one huge context is a psychotic mess :) |
| 17:05.02 | Waverly360 | [TK]D-Fender: ok ok :P |
| 17:05.17 | *** join/#asterisk tclark (n=TC@24.69.13.51) |
| 17:05.23 | *** join/#asterisk X-Rob_ (n=Rob@ppp214-210.static.internode.on.net) |
| 17:05.24 | Waverly360 | [TK]D-Fender: are you saying it's a mess just because it's big? |
| 17:06.12 | [TK]D-Fender | Waverly360: massive overkill on whitespace and repetative code |
| 17:06.28 | [TK]D-Fender | Waverly360: a little effort could chop that to 25% of its current size |
| 17:06.47 | [TK]D-Fender | Waverly360: And you're using 1.0 deprecated stuff... |
| 17:07.16 | Waverly360 | [TK]D-Fender: Well..to my credit, I inherited a lot of this from someone else. I didn't design the config files initially. |
| 17:07.34 | Waverly360 | [TK]D-Fender: I'll do my best to redesign them when I have the time..but that's just not in the cards right now. |
| 17:08.00 | [TK]D-Fender | Waverly360: Ok, not sure why you'd get tone at that point. Are you on a channel-bank on Zap/22? |
| 17:08.35 | Waverly360 | [TK]D-Fender: I hate to say it, but I really don't know. I have two people telling me different things about the setup there. I'm probably going to have to drive there myself. |
| 17:09.15 | [TK]D-Fender | Waverly360: thats about the only circumstance I can think of. |
| 17:09.40 | Waverly360 | [TK]D-Fender: If it helps any, dialing any number during that dialtone does nothing. |
| 17:10.04 | Waverly360 | [TK]D-Fender: I'll just take a trip out there myself. *sigh* There isn't enough of me to go around this place, I swear. |
| 17:10.33 | [TK]D-Fender | Waverly360: have you tried dialing the target # by hand and seeing what happens? |
| 17:10.38 | ManxPower | Waverly360: the CLI output should tell you everything you need to know. |
| 17:10.40 | ManxPower | Waverly360: are you testing this from your cell phone? |
| 17:11.05 | Waverly360 | ManxPower: Well, I'm dialing from a PBX here, to the PBX there. |
| 17:11.19 | Waverly360 | ManxPower: though I have tried with my cell phone, and got the same result. |
| 17:11.26 | ManxPower | Waverly360: try it from your cell phone. |
| 17:11.40 | Waverly360 | ManxPower: The CLI output is in that last pastebin...do you see anything weird? |
| 17:12.03 | [TK]D-Fender | Waverly360: If you get the same result, then its the number you are dialing, not your * setup (duh) :) |
| 17:12.13 | ManxPower | I didn't see the last pastebin |
| 17:12.22 | ManxPower | there it is |
| 17:12.54 | Waverly360 | [TK]D-Fender: I think you misunderstood me. |
| 17:12.59 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 17:12.59 | *** mode/#asterisk [+o mog] by ChanServ |
| 17:13.13 | [TK]D-Fender | Waverly360: Just following what you wrote... |
| 17:13.36 | [TK]D-Fender | Waverly360>ManxPower: though I have tried with my cell phone, and got the same result. |
| 17:13.37 | Waverly360 | [TK]D-Fender: it doesn't matter what number I put in the Dial, I always get dialtone |
| 17:13.45 | [TK]D-Fender | Waverly360: Hrm. |
| 17:14.16 | ManxPower | put a Noop(HANGUPCAUSE is ${HANGUPCAUSE} as the priority after the dial and add the "g" option to Dial |
| 17:14.34 | Waverly360 | [TK]D-Fender: Oh...I was playing around earlier, and I tried putting a few w's in front of the number I wanted to dial...I didn't get dialtone then..it rang once, and then just sat there. |
| 17:14.48 | kn0x | so anyone familair with AMI Originate, Variable: |
| 17:14.52 | kn0x | ? |
| 17:14.59 | ManxPower | Waverly360: "w" only works on non-pri non-voip |
| 17:15.15 | Waverly360 | ManxPower: That's what I figured...I was just shooting into the dark. |
| 17:15.41 | Waverly360 | Crap..I can't mess around with it anymore right now..I have to run. Thanks for the help..I'll check into it further later. |
| 17:16.19 | [TK]D-Fender | Waverly360: You may need to do some real tests... |
| 17:16.37 | Waverly360 | [TK]D-Fender: what I really need is to be on site. |
| 17:16.47 | Waverly360 | [TK]D-Fender: I can only do so much from here. |
| 17:16.58 | [TK]D-Fender | Waverly360: Yup... god only knows what they plugged that port into or what else they could havs screwed up |
| 17:17.10 | Waverly360 | [TK]D-Fender: Yeah. |
| 17:17.30 | Waverly360 | anyways..later guys. Thanks |
| 17:19.20 | Defraz | I guess on a cisco you can't program the softkeys in SIP mode. |
| 17:20.55 | *** join/#asterisk opioid (n=karlhain@209.12.254.75) |
| 17:21.29 | *** join/#asterisk shinux__ (n=shinux@208.70.5.150) |
| 17:22.02 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 17:29.10 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 17:32.50 | *** join/#asterisk Ebola (n=Ebola@host86-136-130-202.range86-136.btcentralplus.com) |
| 17:33.07 | *** join/#asterisk saftsack (n=saftsack@pD9E07871.dip.t-dialin.net) |
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| 17:36.00 | anonymouz666 | Can I use after a background() a switch => Realtime/@blah statement for background() look into realtime extension table if exist the digit ? |
| 17:36.49 | anonymouz666 | oops I don't think so. If i type something there is no way to call switch |
| 17:37.07 | anonymouz666 | i am stuck |
| 17:41.17 | *** join/#asterisk wo-man (n=fdsfsd@ip70-189-120-242.ok.ok.cox.net) |
| 17:41.44 | wo-man | <PROTECTED> |
| 17:42.35 | *** join/#asterisk shinux__ (n=shinux@208.70.5.150) |
| 17:42.38 | Strom_M | omg - you want to Be The Next Vonage? |
| 17:43.31 | wo-man | hahahahaha |
| 17:43.31 | wo-man | no |
| 17:43.32 | wo-man | I just wanna provide for a about 400 cutomers |
| 17:43.32 | anonymouz666 | wo-man: use openser |
| 17:43.53 | anonymouz666 | solve the problem by reading the background() help |
| 17:43.53 | anonymouz666 | lol |
| 17:44.09 | Jon335 | Does anyone have a termination provider that allows you to set the Caller ID Name? |
| 17:44.30 | Strom_M | Jon335: you don't understand how the PSTN works, apparently : |
| 17:44.31 | Strom_M | er :) |
| 17:44.41 | Strom_M | CNAM is looked up on the terminating end of the call |
| 17:45.15 | Jon335 | Strom_M, that's weird as when I set a perfectly valid Caller ID number it shows up as Long Distance or Unknown Name |
| 17:45.40 | Strom_M | is there a name associated with that number in the telco's CNAM database? |
| 17:45.50 | Jon335 | Strom_M, yes |
| 17:46.16 | Strom_M | perhaps you need to set the country code as well |
| 17:46.21 | Strom_M | fiddle with it a bit |
| 17:46.45 | Jon335 | Strom_M, would it be different in Canada (where I am) |
| 17:46.56 | Strom_M | canada uses country code "1" |
| 17:47.42 | [TK]D-Fender | Strom_M: I have ILEC's here that let you set CNAM.... |
| 17:49.03 | Strom_M | odd |
| 17:49.09 | Strom_M | ive never seen that |
| 17:49.36 | *** join/#asterisk S2AnGeL (n=S2AnGeL@CPE0014bf103d31-CM000039529869.cpe.net.cable.rogers.com) |
| 17:49.48 | [TK]D-Fender | Strom_M: You need to trade in those smokey shades for rose ;) |
| 17:50.19 | S2AnGeL | is there a way to call people sorta like ring groups .. need a way for a temp employment office to call employees when a job is available |
| 17:50.55 | S2AnGeL | sorta it dials through a list |
| 17:51.12 | Strom_M | [TK]D-Fender: never! I love my ray-bans |
| 17:51.54 | S2AnGeL | it would save valubel time to have it all in a data base.. and who ever is most recent it dials through and they get a call with a msg and press one if you want to hear the job.. or call us back sorta thing |
| 17:51.56 | [TK]D-Fender | S2AnGeL: "show application queue" |
| 17:52.34 | [TK]D-Fender | S2AnGeL: Or just code something yourself. |
| 17:52.39 | docelmo | YAY! |
| 17:52.53 | *** join/#asterisk groogs (n=greg@d38-54-164.commercial1.cgocable.net) |
| 17:53.21 | S2AnGeL | I am looking into ti |
| 17:53.23 | S2AnGeL | it |
| 17:53.28 | S2AnGeL | thanks for the tip |
| 17:53.29 | docelmo | wo-man that can be done from 2 asterisk servers.. |
| 17:55.55 | *** join/#asterisk murdmath (n=vircuser@mail.kimballequipment.com) |
| 17:56.46 | murdmath | Is there a way to continue my dial plan after a Page has finished? |
| 17:57.32 | *** join/#asterisk MACscr (n=MACscr@adsl-75-23-80-143.dsl.peoril.sbcglobal.net) |
| 17:57.47 | murdmath | The problem I'm having is that if someone does a page to quickly after another page has finished it rings the phones instead of paging them. |
| 17:57.51 | MACscr | <PROTECTED> |
| 17:58.52 | [TK]D-Fender | murdmath: Shouldn't happen. We'd have to see your full dialplan setting up the page |
| 17:59.29 | *** join/#asterisk darken_darken (n=marco@246.176.76.83.cust.bluewin.ch) |
| 18:01.22 | *** part/#asterisk S2AnGeL (n=S2AnGeL@CPE0014bf103d31-CM000039529869.cpe.net.cable.rogers.com) |
| 18:01.34 | murdmath | [TK]D-Fender: I don't think it's an Asterisk issue, I think it is a Snom issue. I think the phone is not hanging up fast enrough. |
| 18:01.58 | [TK]D-Fender | murdmath: Quite possible. That would indeed do it. |
| 18:02.16 | b11d | hey.. does a telco vendor have the option to disregard LNP here in the USA? |
| 18:02.18 | [TK]D-Fender | murdmath: And that would ahve nothing to do with being able to continue running dialplan (which you CAN). |
| 18:03.06 | murdmath | [TK]D-Fender: Well what I was going to do is set a flag in the astdb that would check to see if the page was in use. After the page was done it would wait a few seconds and then clear the flag alowing someone else to page. |
| 18:03.32 | [TK]D-Fender | murdmath: What you can do is add a "lastended" check to the end/start of your script to see if enough time has passed, if not wait(3) or something |
| 18:03.48 | [TK]D-Fender | murdmath: Feel the synchronicity :) |
| 18:04.35 | [TK]D-Fender | murdmath: Right up the same alley... oh and BTW this mental frequency has been reserved under the FCC and you're infringing on my bandwidth! Back off! ;) |
| 18:04.54 | murdmath | [TK]D-Fender: :) |
| 18:05.53 | *** join/#asterisk MRH2 (n=Mr_happy@62.49.242.3) |
| 18:06.03 | murdmath | [TK]D-Fender: What you are saying is put the time of the page in a astdb variable and when the next page comes in check the time and make sure it enought time has passed. |
| 18:06.17 | [TK]D-Fender | murdmath: Correct |
| 18:06.35 | murdmath | [TK]D-Fender: I assume there is not a way to find out when the page ended is there? |
| 18:06.50 | *** join/#asterisk fluffyfluffy (n=fluffyfl@h69-130-215-2.69-130.unk.tds.net) |
| 18:07.09 | [TK]D-Fender | murdmath: At the end of a page push currtime into astdb. on start check if ABS(timestamp - lastended) > 3s |
| 18:07.48 | murdmath | [TK]D-Fender: The problem is my dial plan seems to end after the Page is started. |
| 18:07.50 | [TK]D-Fender | murdmath: Sure there is. Either grab it after the page has finished with a timeout, or "h" |
| 18:08.04 | [TK]D-Fender | murdmath: "h" <- |
| 18:08.44 | *** join/#asterisk cuco (n=elcuco@bzq-88-154-161-123.red.bezeqint.net) |
| 18:08.50 | cuco | tzafrir: ping |
| 18:08.52 | murdmath | [TK]D-Fender: I tried the h with a verbose and I never saw anything come up in the consol. So I thought nothing was happing. |
| 18:09.01 | tzafrir | cuco, pong |
| 18:09.08 | [TK]D-Fender | murdmath: Got a priority after your call to Page? |
| 18:09.31 | [TK]D-Fender | murdmath: because ONE of the 2 HAS to pick it up.... |
| 18:09.46 | murdmath | [TK]D-Fender: This is what I have: exten => 82,6,Page(SIP/128|s) |
| 18:10.02 | [TK]D-Fender | murdmath: pastebin the whole mess :) |
| 18:10.17 | [TK]D-Fender | ~pb |
| 18:10.30 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 18:10.31 | murdmath | [TK]D-Fender: Ok. |
| 18:10.31 | BSD_Tech | gasterisk the phone pbx to get gassy by |
| 18:10.31 | murdmath | ~pb |
| 18:10.32 | jbot | i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 18:11.43 | *** join/#asterisk Juggie (i=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 18:11.49 | murdmath | [TK]D-Fender: http://pastebin.ca/433263 |
| 18:11.54 | jnc | [TK]D-Fender: what makes a user from users.conf a valid extension? my sip softphone complains "no such user" when dialing, yet it is able to register as a sip client okay. I've looked at the dialing rules and I don't yet understand if this is a pattern match or some other part of the extensions spec? |
| 18:12.22 | jnc | w/ the default demo extensions.conf rules it does work |
| 18:12.33 | jnc | trying to go from scratch so I understand more about the system |
| 18:13.23 | murdmath | [TK]D-Fender: It's a bit messy, had some debugging stuff in there. |
| 18:14.29 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 18:14.54 | [TK]D-Fender | murdmath: "h" is not a PRIORITY, it is a STANDARD EXTENSION <- |
| 18:16.43 | *** join/#asterisk shinux__ (n=shinux@208.70.5.150) |
| 18:17.36 | *** join/#asterisk murdmath (n=vircuser@mail.kimballequipment.com) |
| 18:19.00 | *** join/#asterisk shinux__ (n=shinux@208.70.5.150) |
| 18:19.12 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:19.32 | *** join/#asterisk gbodemantv (n=gbodeman@corpex.pivotal.televerde.com) |
| 18:19.32 | anonymouz666 | how do I can save a value typed in background into a var? |
| 18:19.39 | gbodemantv | hey all |
| 18:20.11 | [TK]D-Fender | anonymouz666: you have the EXTEN due to pattern-matching. Do the math... |
| 18:20.18 | BSD_Tech | save it where |
| 18:20.58 | murdmath | [TK]D-Fender: Internet hiccuped, did I miss something. |
| 18:22.14 | [TK]D-Fender | [14:14]<[TK]D-Fender>murdmath: "h" is not a PRIORITY, it is a STANDARD EXTENSION <- |
| 18:22.35 | murdmath | [TK]D-Fender: Ya got that one. let work on it for a bit. |
| 18:22.52 | [TK]D-Fender | murdmath: You should be inches away... this is not a hard one to do. |
| 18:26.33 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 18:27.06 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 18:28.28 | anonymouz666 | [context] exten => switch => exten => works ? |
| 18:28.32 | anonymouz666 | this order |
| 18:30.52 | *** join/#asterisk kannan (n=kannan@210.211.178.211) |
| 18:30.53 | Defraz | I have a cisco 7940, I am having trouble changing the SoftKeys |
| 18:30.53 | Defraz | I can't seem to find anything out there to do it. |
| 18:30.53 | Nugget | You can't change the soft keys if you're using the sip firmware. |
| 18:30.59 | anonymouz666 | switch statement must be the first in a context? |
| 18:31.39 | anonymouz666 | exten => 1,1,NoOp() switch => Realtime exten => ... |
| 18:31.44 | anonymouz666 | works ? |
| 18:32.10 | [TK]D-Fender | anonymouz666: What are you trying to capture that you'll have fixed AND variable extens? |
| 18:32.45 | [TK]D-Fender | anonymouz666: in the same context.... I presume you can INCLUDE 2 eachw ith their own rules... |
| 18:32.49 | *** join/#asterisk hrmphh (i=patrick@notchill.com) |
| 18:33.05 | anonymouz666 | remember that background() you told me to use. I need to save the ${exten} before sending it to match through realtime |
| 18:33.58 | anonymouz666 | Or is better to set it through realtime extensions too ? |
| 18:34.06 | hrmphh | do you guys recommend enabling aggressive suppression? |
| 18:34.13 | hrmphh | im trying to get rid of the initial echo on PSTN calls |
| 18:34.21 | Corydon-w | I don't recommend realtime extensions |
| 18:34.24 | *** join/#asterisk opioid (n=karl@207.191.91.242) |
| 18:34.33 | hrmphh | why not |
| 18:34.44 | Corydon-w | Because they're slow and inflexible |
| 18:35.03 | Corydon-w | If you want database driven extensions, there are better ways to do that |
| 18:35.06 | anonymouz666 | Corydon-w: what I should use then? lots of global vars into dialplan? |
| 18:35.13 | [TK]D-Fender | anonymouz666: just do your accumulating first. You are getting your head turned around in a mess and are not dealing with the basics first. |
| 18:35.30 | Corydon-w | func_odbc, for example |
| 18:35.36 | [TK]D-Fender | hrmphh: Depends what hardware you are using |
| 18:37.00 | hrmphh | TK; Digium TDM13B |
| 18:37.06 | hrmphh | 3 fxo and 1 fxs |
| 18:37.13 | anonymouz666 | oh my its a very complex IVR |
| 18:37.14 | hrmphh | or do you mean the pc itself? |
| 18:37.26 | anonymouz666 | my braincells are burning lol |
| 18:37.27 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 18:37.27 | *** mode/#asterisk [+o mog] by ChanServ |
| 18:37.31 | [TK]D-Fender | hrmphh: Have you gone through the usuals of fxotune, gain settings, echocancel=yes, etc? |
| 18:37.44 | gerphimum | does anyone know if the new nokia n95 cell phone can connect to asterisk and be used as its own channel |
| 18:37.48 | hrmphh | echocancel=yes echotraining=yes in zapata.conf |
| 18:37.52 | hrmphh | i dont know about fxotune and gain settings |
| 18:37.54 | hrmphh | have an url? |
| 18:37.54 | [TK]D-Fender | anonymouz666: No... its NOT. You just need to think things through 1 step at a time. |
| 18:38.01 | [TK]D-Fender | ~wikis |
| 18:38.03 | jbot | from memory, wikis is http://www.voip-info.org |
| 18:38.10 | [TK]D-Fender | hrmphh: Lookup echo cancellation in there |
| 18:38.14 | hrmphh | k thnx |
| 18:38.19 | [TK]D-Fender | hrmphh: Don't have the specific link offhand |
| 18:38.21 | Corydon-w | anonymouz666: all realtime extensions gets you is a load of extensions.conf into a database. It is no more advantageous than static realtime. |
| 18:38.51 | Corydon-w | In fact, in many ways, realtime extensions is actually a step backwards from static realtime |
| 18:38.53 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 18:39.03 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 18:39.19 | irule | I am not doing anythingm and suddenly the phone rings shortly and this appears on the CLI, what is it? " -- Saved useragent "Sipura/SPA2000-2.0.10(e)" for peer sip603" |
| 18:41.21 | jnc | irule: completely guessing here, maybe the device rebooted |
| 18:41.53 | [TK]D-Fender | irule: Means the phone just registered |
| 18:42.08 | anonymouz666 | i need water |
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| 18:47.37 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
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| 18:50.24 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 18:53.12 | irule | where exactly should I put this? thanks! Set(CHANNEL(language)=es) |
| 18:53.17 | jnc | irule: some ATAs ring all the connected lines when they connect |
| 18:53.18 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 18:53.18 | *** mode/#asterisk [+o denon] by ChanServ |
| 18:53.18 | *** join/#asterisk newmember (n=newmembe@static-66-11-81-65.ptr.terago.ca) |
| 18:53.52 | jnc | irule: it's useful for diagnosing (to common people) whether or not the device is working |
| 18:54.05 | jnc | i.e. tell your customer to reboot their junk, wait for a ring. |
| 18:54.20 | Mercestes | irule: That looks like an extensions.conf assignment to me. |
| 18:54.44 | [TK]D-Fender | Mercestes: irule>I am not doing anythingm and suddenly the phone rings shortly and this appears on the CLI, what is it? " -- Saved useragent "Sipura/SPA2000-2.0.10(e)" for peer sip603" <- No |
| 18:54.57 | murdmath | [TK]D-Fender: Well I'm getting closer. I can't use the wait() function in h extenstion, I'm going to have to go the time route. How can i pull the time current time into a dial plan? |
| 18:55.02 | [TK]D-Fender | Mercestes: Registration notice... |
| 18:55.43 | [TK]D-Fender | murdmath: You don't do the Wait in "h", you simply set the "last ended" time there. You CHECK for it at tthe START, and THEN wait if needed |
| 18:56.13 | J4k3 | wow, don't bother doing business with voipsupply... they'll sell you something that has no hope of ever working correctly then refuse to give you a refund... just claim they are, then send you another busted-ass worthless piece of crap. Glee. |
| 18:56.44 | newmember | is there a current LDAP how to somewhere? |
| 18:57.13 | J4k3 | is there by chance a class action lawsuit or anything against utstarcom? they certianly deserve it. |
| 18:58.41 | Mercestes | [TK]D-Fender, I was referring to irule> where exactly should I put this? thanks! Set(CHANNEL(language)=es) |
| 18:58.49 | murdmath | [TK]D-Fender: Well I was trying to just use flags first and just wait to set the the flag three seconds after it hit the H context. But that didn't work so now I'm going to do it the way you suggested. Just found the time variable. |
| 18:59.53 | *** join/#asterisk saftsack (n=saftsack@pD9E07871.dip.t-dialin.net) |
| 19:00.03 | Mercestes | J4k3, The problem is ...utstarcom does technically work. |
| 19:02.32 | J4k3 | Mercestes: yes and no... I can't MAKE it give me decent audio |
| 19:02.58 | [TK]D-Fender | J4k3: I do believe you've been warned here.... |
| 19:03.10 | GreyFoxx | Anyone using any sort of webbased interface for managing conferences? |
| 19:03.20 | J4k3 | I think I'll make a nice youtube video about the situation then send the piece of crap back again, this time with a proper "take this thing and massage your prostate with it" |
| 19:03.28 | Mercestes | J4k3, But it does do what it says, I'm pretty sure they never guarantee the quality of delivery, only the delivery |
| 19:04.06 | [TK]D-Fender | ~wifisip |
| 19:04.07 | jbot | Wi-Fi SIP phones suck. All of them. HARD. Some only slightly less than others... |
| 19:04.33 | uski | (doh) |
| 19:05.08 | Mercestes | sucks like a gay man at a frostee eating contest. |
| 19:05.09 | [TK]D-Fender | GreyFoxx: Sure, Some people use Thirdlane, FreePBX, the Asterisk GUI, and so on.... |
| 19:05.12 | J4k3 | sucks pretty bad considering xlite on my wifi-equipped laptop works great on bluetooth... :| |
| 19:05.36 | Mercestes | GreyFoxx, and other assorted crap |
| 19:05.44 | J4k3 | unluckily bt earpiece + computer connectivity = dead bt battery real quick. |
| 19:05.56 | J4k3 | and/or, you get to fiddle with it on every phone call. |
| 19:07.01 | denon | J4k3: take a look at idefisk |
| 19:07.09 | denon | you'll be much happier with it I think |
| 19:07.32 | GreyFoxx | [TK]D-Fender: Know of any other that just "Web-meetme" for just controlling conferences? Those others seem to be overall asterisk management, which we most definately do not want |
| 19:07.44 | GreyFoxx | Otherwise I'll end up writing something |
| 19:08.42 | Mercestes | J4k3, and don't forget randomly reboot it |
| 19:09.30 | J4k3 | randomly reboot what? |
| 19:09.36 | J4k3 | the f1000g? it reboots itself :P |
| 19:09.37 | [TK]D-Fender | GreyFoxx: .... |
| 19:09.41 | [TK]D-Fender | ~toywy |
| 19:09.43 | jbot | well, toywy is The one you write yourself. |
| 19:09.48 | *** join/#asterisk bkw_ (i=brian@ppp-70-128-123-137.dsl.tulsok.swbell.net) |
| 19:09.52 | GreyFoxx | Figured as much |
| 19:10.21 | [TK]D-Fender | J4k3: Oh, and don't forget the nasty slow qualify response timeouts that cause them to blink in and out of contactability ;) |
| 19:10.59 | J4k3 | haha |
| 19:11.42 | Mercestes | and the fact that it's usability range is about 50' |
| 19:15.55 | anonymouz666 | I have another loop now using saynumber() to say values like 343, 423... with an indice digit using saydigits()... 1 for 343... etc. how can I read the choosen option? a read() inside the loop? it's very difficult to do anything without background() |
| 19:16.00 | *** join/#asterisk saftsack (n=saftsack@pd9e07871.dip.t-dialin.net) |
| 19:16.21 | *** join/#asterisk tonyb2006 (n=tonyb@2002:4571:29c2:0:0:0:0:1) |
| 19:16.32 | tzafrir | anybody here using a suse zaptel rpm package? |
| 19:16.36 | tonyb2006 | how can I tell if a modem is a so-called "winmodem", I have 4 PCI modems lying on my desk |
| 19:17.19 | Corydon-w | tonyb2006: uh, why? |
| 19:17.36 | tzafrir | well, it is a "winmodem" in the same way zaptel cards and such are "linmodems". |
| 19:17.46 | tzafrir | Astrerisk is about host processing... |
| 19:17.56 | tonyb2006 | Because I've been told a non-WinModem could be used with asterisk, or somthing along those lines |
| 19:18.06 | Corydon-w | A better question is, is this modem half-duplex or full-duplex? |
| 19:18.33 | p0g0 | uhmm..there are funky intels that you_can_ use, but they sound like shit and have serious echo problems |
| 19:18.35 | Corydon-w | Because if it's half-duplex, it's completely useless for use as an FXO devices (most are) |
| 19:18.57 | J4k3 | hrm... the f1000g doesn't bother to ack packets at the 802.11 level. Thats the 'big problem' with it, and why it ends up getting SIP packets all out of order. |
| 19:19.01 | tonyb2006 | well how could I figure that out than |
| 19:19.17 | Corydon-w | tonyb2006: you read the manufacturer's specs |
| 19:19.29 | Mercestes | tonyb2006, You don't try to use a modem as an FXO card. |
| 19:19.35 | p0g0 | google- the ones I tried had to have pull up resistors de-soldered. |
| 19:19.35 | Corydon-w | There is no way to look at a board and "know" |
| 19:19.35 | tzafrir | tonyb2006, look for "x100p" for relevant information. Generally only a few select modems could be used that way |
| 19:19.45 | [TK]D-Fender | anonymouz666: Not sure you can interrupt or intermix saydigits/daynumber with background.... |
| 19:19.55 | p0g0 | hang on, I'll give you an FCC# |
| 19:20.06 | Mercestes | lol. you look for the words "digium", "Sangoma", or "FXO" stamped on it |
| 19:20.27 | [TK]D-Fender | pogo : Just plug the darn thing in and compile zaptel and see if its recognized. |
| 19:20.27 | Corydon-w | X100P worked, but it did not have Digium stamped on it |
| 19:20.55 | p0g0 | the worthless, don't waste your time on these intel winmodems are the IA92 series |
| 19:21.29 | p0g0 | they have an ambient chipset ffy55-000 0241 |
| 19:21.51 | Corydon-w | I dunno, X100P worked for me; my home setup still runs it |
| 19:22.37 | p0g0 | resistors r13, r19 and r17 needed to be removed to spoof the card ID |
| 19:22.44 | Corydon-w | It's been running since before 0.20 |
| 19:22.54 | *** join/#asterisk shinux__ (n=shinux@208.70.5.150) |
| 19:23.04 | J4k3 | I have no idea what intel and ambient's relationship is, but ambient = cirrus logic, and they've had a long legacy of sucking ass. |
| 19:23.34 | denon | you know, it baffles me why people continue to try to use the x100 |
| 19:23.44 | denon | its really not designed to be very robust |
| 19:23.52 | p0g0 | you can edit the ID list in, iirc, the zaptel module, to accept your cards ID...but they are not worth it- buy a real zaptel card. |
| 19:24.00 | Corydon-w | denon: because it's cheap and it works fine for answering machine setups? |
| 19:24.09 | J4k3 | denon: its all crap, so its hard to get excited about using something 'better' |
| 19:24.12 | anonymouz666 | [TK]D-Fender I would say that I didn't see nothing related to intermix saydigits with background |
| 19:24.14 | J4k3 | I mean |
| 19:24.25 | denon | J4k3: dunno, real tdm cards tend to be a little more solid |
| 19:24.31 | J4k3 | if you're hooking POTS up to anything except a 1972 IBM Desk phone, you're already performing "The Great Suck" |
| 19:24.45 | p0g0 | I have these striclty to use to test a machine & the zaptel module- I wouldn't use them in any environment like home or production..but they were $7 |
| 19:25.24 | anonymouz666 | I need something like Background(SayNumber(${var})) - hehe |
| 19:25.44 | J4k3 | then again, I have the worst POTS lines in north america, I think. |
| 19:25.56 | [TK]D-Fender | anonymouz666: I think those 2 apps act like playback ad wait till they can seize the channel for audio and therefor only the backgrounds AFTER it will count. |
| 19:26.12 | p0g0 | for a laugh- the FCC # is us:56JFBOOBAMI |
| 19:26.23 | J4k3 | hahaha |
| 19:26.25 | J4k3 | BOOBAMI! |
| 19:27.26 | p0g0 | J4k3: I was (and am nearly still) the longest wire in my telco's entire system... I had to learn a lot about line conditioning and lightening protection over the last 40 years |
| 19:27.28 | anonymouz666 | [TK]D-Fender: i don't understand |
| 19:28.03 | J4k3 | my straight-wired lines (32kft) actually worked pretty well... unluckily they were all turned into T1 loops. |
| 19:28.53 | J4k3 | theres an early 70s remote, analog at the CO, 8 kft in the other direction |
| 19:29.24 | J4k3 | I'm suprised nobody has taken shots at the remote yet.. Telco's so worthless they won't even replace the batteries in it. |
| 19:30.09 | *** part/#asterisk fluffyfluffy (n=fluffyfl@h69-130-215-2.69-130.unk.tds.net) |
| 19:32.48 | p0g0 | J4k3: after 30 years of bitching, I got a MUX only 3.5 miles off- I even get DSL now...at rapacious prices. One of the most interesting bits of hardware I own is a Talking Technologies BigMouth card- that thing is So Sensitive to capacitance changes on the line that I know wen any of the physics changes...and you'd never be able to tap the analog without it noticing... |
| 19:34.25 | J4k3 | hmm interesting |
| 19:34.50 | J4k3 | I took a lot of the uglyness out of the f1000g by dropping my AP's beacon time from 100ms to 50ms |
| 19:35.19 | J4k3 | beacons have *something* to do with 802.11 power saving, so that makes a bit of sense. |
| 19:36.12 | *** join/#asterisk Hmmhesays (n=Neg@24-117-131-41.cpe.cableone.net) |
| 19:36.13 | J4k3 | haha, Windstream attempts to install DSL on 25k+ ft loops all the time |
| 19:36.26 | J4k3 | basically they send a tech out to look busy until you run him off |
| 19:36.57 | Hmmhesays | fun |
| 19:37.01 | J4k3 | one guy, OBVIOUSLY too far out of town for ADSL, ordered it.. they had two "techs" there from 10am til 11pm trying to make it work |
| 19:37.10 | J4k3 | of course it never did |
| 19:37.19 | J4k3 | I can't imagine what that cost the telco in labor time |
| 19:37.21 | p0g0 | J4k3: I've got a VOIP over wifi link here- is that what's giving you trouble? Once I "balanced the hybrid" with the real zaptel card, and set up decent antennas on the wrt54g's it's been pretty much fine |
| 19:37.53 | J4k3 | p0g0: the problem is one end is a f1000g... I've tested from my laptop and it works great |
| 19:37.58 | [hC] | J4k3: so your wifi phone gets a lot more battery life now that you reduced the beacon timing? |
| 19:38.19 | J4k3 | [hC]: dunno... we'll find that out in a few days |
| 19:38.20 | BSD_Tech | I have a gift for you all |
| 19:38.24 | BSD_Tech | http://pastebin.ca/433400 |
| 19:38.43 | BSD_Tech | it will grow and in time I will post on the wiki |
| 19:38.54 | BSD_Tech | but I have put alot of work into it for you all |
| 19:39.14 | jm|laptop | BSD_Tech: thx |
| 19:39.44 | BSD_Tech | I will be better documenting as I go and I am working on a full dial plan |
| 19:40.02 | BSD_Tech | most I hope will go ingot asterisk and asterisk-now |
| 19:40.12 | BSD_Tech | to make it more usable out of the box |
| 19:40.32 | *** part/#asterisk tonyb2006 (n=tonyb@2002:4571:29c2:0:0:0:0:1) |
| 19:40.58 | BSD_Tech | I put it in a file called nanpa.conf and add a #include = nanpa.conf line |
| 19:41.14 | ManxPower | How do you balance the hybrid, p0g0 |
| 19:41.46 | BSD_Tech | ? |
| 19:42.10 | p0g0 | I read a howto and ran a routine...I'll dig up the name, but it's in the zaptel subdir, iirc. |
| 19:42.12 | ManxPower | (14:36:45) p0g0: J4k3: I've got a VOIP over wifi link here- is that what's giving you trouble? Once I "balanced the hybrid" with the real zaptel card, and set up decent antennas on the wrt54g's it's been pretty much fine |
| 19:42.22 | *** part/#asterisk zapp-branigan (n=zapp-bra@81.202.214.78.dyn.user.ono.com) |
| 19:42.24 | ManxPower | p0g0: you mean fxotune |
| 19:42.48 | p0g0 | sounds like that might be it, it was a 1 time deal, and I did it a while back |
| 19:42.51 | ManxPower | that just preloads some settings onto the board, IITRC |
| 19:43.05 | [hC] | Should there be an issue with a digium t1 pri card and sending faxes to a SIP ATA with an analog fax machine plugged in? faxes keep coming out blank or corrupted most of the time... ive made sure echo cancellation is off on the ATA.. do i need to use faxdetect= in zapata.conf for it to turn off echo cancellation on the PRI rather than just sending the call to the ATA? |
| 19:43.18 | p0g0 | no this routine did a bunch of sampling, then set up to load the parms, iirc. |
| 19:43.27 | anonymouz666 | how can I set a read() var to be global ? |
| 19:43.37 | ManxPower | [hC]: don't expect fax to work well over VoIP |
| 19:43.49 | ManxPower | [hC]: What codec are you using? |
| 19:43.52 | [hC] | ManxPower: er.. its coming in a PRI and going over a LAN using ulaw to an ATA. |
| 19:44.09 | [hC] | the only "VoIP" is the SIP link on the LAN between the ATA and the PBX. |
| 19:44.09 | BSD_Tech | manax if you get time lookat the pastebin and give me feed back |
| 19:44.13 | ManxPower | [hC]: Still. don't expect fax to work well over ATAs. |
| 19:44.27 | ManxPower | I solved all my fax problems with Asterisk and ATAs early on. |
| 19:44.28 | [hC] | ManxPower: what is the best way to do this then, if my fax number is on this PRI, and the PRI goes into asterisk? |
| 19:44.39 | [hC] | ManxPower: usually i tell people to just keep an analog line. |
| 19:44.57 | [hC] | ManxPower: otherwise i use IAXmodem and Hylafax, and that has about a 99% success rate. some people dont like eFax tho |
| 19:44.58 | ManxPower | [hC]: that is how I solved all my fax problems. |
| 19:45.14 | J4k3 | I fixed all my fax problems 11 years ago |
| 19:45.16 | ManxPower | Once I moved the fax machine and fax number to a standard analog line I got ZERO complaints about faxing |
| 19:45.23 | J4k3 | "if you can't email it, I don't need it" |
| 19:45.50 | ManxPower | J4k3: that doesn't work for my clients |
| 19:45.56 | p0g0 | fax: the Helen Keller of telecommunications technology |
| 19:46.03 | J4k3 | well, for those that it doesn't work on, tell them the US mail works well. |
| 19:46.10 | J4k3 | (the US mail also works as a nonsense filter, too) |
| 19:46.30 | J4k3 | if someone refuses to send something USPS, I figure its illegal and I don't want it anyways ;) |
| 19:46.39 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 19:46.41 | ManxPower | *nod* Postal mailing prescription refil authorizations works just peachy |
| 19:46.49 | *** join/#asterisk GNU\colossus (i=nobody@85-124-31-18.dynamic.xdsl-line.inode.at) |
| 19:46.54 | ManxPower | same with real estate contracts |
| 19:47.12 | J4k3 | real estate contracts through a fax machine are about as good as a dirty piece of toilet paper. |
| 19:47.25 | *** join/#asterisk shinux__ (n=shinux@208.70.5.150) |
| 19:47.32 | p0g0 | ManxPower: the only fax card I've found worth a darn is the USR 2977, class 2 fax, full PCI, runs great under Linux... |
| 19:47.48 | ManxPower | J4k3: once the agent gets the comission they don't really care how binding the contract is |
| 19:48.11 | ManxPower | p0g0: fax card? My users have enough trouble turning on their computer |
| 19:48.12 | J4k3 | ManxPower: yeah I learned that the hard way (Couch Mortgage/Stewart title company...) |
| 19:48.25 | GNU\colossus | our largest customer is seeking commercial support for asterisk (and the process of migrating its current telephony-solution to it) in switzerland, anyone able to give me direction on where to look for a provider? |
| 19:48.37 | ManxPower | J4k3: most of the faxing is for changes to contracts before the actual signature |
| 19:48.39 | Mercestes | Oooo! Mee! Mee! Meeee! |
| 19:48.43 | Mercestes | I'm a consultant! Pay MEEEEE! |
| 19:48.50 | ManxPower | GNU\colossus: where are you located? |
| 19:49.13 | GNU\colossus | ManxPower: Austria, EU. Our customer is located in Switzerland. |
| 19:49.23 | ManxPower | I knows switzerland is not all THAT big, but the city might be helpful to people |
| 19:49.24 | Mercestes | GNU\colossus, I dunno where to get sweedish numbers. Let me ask. |
| 19:49.51 | GNU\colossus | the city would be Bern |
| 19:50.01 | ManxPower | GNU\colossus: if you don't find anything helpful here, try the asterisk-biz mailing list. many providers subscribe to that list |
| 19:50.02 | Mercestes | I thought switzerland was a city in Europe |
| 19:50.23 | ManxPower | Mercestes: no, no, no! It is a private bank! |
| 19:50.27 | Mercestes | ah, crap, my switzerland contact doesn't know of anythign. Knows some austrian providers tho |
| 19:50.32 | Mercestes | Ahhh.. |
| 19:51.04 | hrmphh | does wcfxo need to be loaded in addition to wctdm? |
| 19:51.12 | *** join/#asterisk sevard (n=sev@c-76-102-2-4.hsd1.ca.comcast.net) |
| 19:51.18 | ManxPower | hrmphh: you did not read the Zaptel readme, did you? |
| 19:51.41 | ManxPower | it tells you exactly what kernel modules are requred for which cards. |
| 19:52.25 | hrmphh | k just checked, so dont need wcfxo |
| 19:52.28 | hrmphh | w/my TDM400P |
| 19:52.30 | hrmphh | thnx |
| 19:53.09 | p0g0 | ManxPower: yeah, fxotune, run once with the -i to run the sampling, thenafter with -s |
| 19:54.31 | ManxPower | p0g0: *nod* I use T-1 cards w/channel bank |
| 19:55.32 | *** join/#asterisk shinux__ (n=shinux@208.70.5.150) |
| 19:57.21 | ManxPower | Apparently Verizon is canceling my cell data account because I use too much bandwidth |
| 19:57.31 | hrmphh | >5GB/mo? |
| 19:57.38 | ManxPower | hrmphh: Yup |
| 19:57.57 | ManxPower | But I on'y exceeded that for 1 month. |
| 19:58.00 | ManxPower | or at least it should have only been 1 month |
| 19:58.10 | hrmphh | yeah they suck |
| 19:58.15 | hrmphh | see the /. article on them recently |
| 19:58.19 | ManxPower | looks like I will have to go back to dialup |
| 19:58.27 | hrmphh | where can i get zaptel from cvs? |
| 19:58.33 | hrmphh | manx; just get a cingular card |
| 19:58.48 | ManxPower | hrmphh: Verizon is the ONLY cell company that works where I live |
| 19:58.51 | hrmphh | s/cvs/current ftp/ |
| 19:58.51 | sevard | how do you use more than 5GB/mo in data on your freaking cell? |
| 19:58.53 | hrmphh | yeah not surprising |
| 19:59.00 | hrmphh | sevard; if you use it for your primary connection |
| 19:59.09 | sevard | ahhh |
| 19:59.11 | ManxPower | sevard: no, on my pcmcia laptop internet card |
| 19:59.13 | hrmphh | or not hard to do other ways |
| 19:59.41 | sevard | you should have installed a counter if you have that low of a cap |
| 19:59.47 | J4k3 | ManxPower: were you paying the $60 account level, or just the $10 unlimited vcast? |
| 19:59.58 | ManxPower | sevard: I might have if the service was not "unlimited" |
| 19:59.59 | J4k3 | oh, you used a pccard. |
| 20:00.08 | ManxPower | J4k3: $59.95/month |
| 20:00.13 | sevard | ManxPower: thou shalt not confuse unlimited with infinate. |
| 20:00.13 | J4k3 | ugh |
| 20:00.15 | J4k3 | ripoff. |
| 20:00.16 | ManxPower | since about a week after katrina |
| 20:00.29 | sevard | You're in the phone business, you should know that. |
| 20:00.46 | ManxPower | they sent me the "Welcome to Verizon" letter 1 YEAR after I got the service |
| 20:00.50 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 20:00.50 | sevard | haha |
| 20:00.58 | *** join/#asterisk SECGOD (n=traderz@65.114.86.29) |
| 20:00.59 | J4k3 | Verizon should not sell UNLIMITED service that is actually limited. |
| 20:01.01 | sevard | good management in that company. |
| 20:01.03 | hrmphh | erm why does zaptel put in all these install commands to /etc/modprobe.d/zaptel for things like "wcfxs" and "wctdm8xxp" when i only tell it to build wctdm?? |
| 20:01.24 | ManxPower | hrmphh: see /etc/sysconfig/zaptel |
| 20:01.27 | *** part/#asterisk SECGOD (n=traderz@65.114.86.29) |
| 20:01.31 | sevard | J4k3: I can rattle off about a thousand companies that sell "unlimited" service. THOU SHALT NOT confuse unlimited with infinate! |
| 20:01.38 | hrmphh | uhh |
| 20:01.41 | hrmphh | no sysconfig on debian |
| 20:01.49 | hrmphh | you thought i ran redhat :( |
| 20:01.55 | J4k3 | Verizon's pretty much screwed up their service here (canceled one of their original tower colo agreements, decided to replace that 450' mount with a 180' tower... now there are massive holes on the highway. |
| 20:02.01 | ManxPower | hrmphh: no, I thought you ran an RPM based distro |
| 20:02.07 | hrmphh | nah sorry |
| 20:02.07 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-153.district.north-van.bc.ca) |
| 20:02.08 | hrmphh | from src |
| 20:02.11 | hrmphh | zaptel 1.4.1 |
| 20:02.18 | hrmphh | which apparently has buggy menuselect |
| 20:02.25 | J4k3 | now the competition isn't any better... Cingular's combination of awful tower placement and suckarse technology makes them STILL inferior to Verizon's service... :P |
| 20:02.35 | CunningPike | Has anyone gotten IMAP voicemail working on an MS Exchange server? |
| 20:02.39 | ManxPower | J4k3: Fortunatly my cell PHONE is verizon prepay and I know from personal experience just how seperate prepay and postpay is with them |
| 20:02.56 | J4k3 | ManxPower: heh, the only reason why I maintain a voice account is for data service. |
| 20:03.07 | sevard | J4k3: I live in bumf#*% minnesota and if I drive 10 miles out of town I lose cell. That's why I got rid of it. |
| 20:03.17 | ManxPower | J4k3: If I put an antenna on MY "tower" I could prolly get cingular internet acces |
| 20:03.50 | ManxPower | where tower = 28 ft telephone pole with a big ass TV antenna, two wifi access points and 3 satelllite dishes |
| 20:04.07 | J4k3 | ManxPower: yeah... I'm only 6 miles or so from a cingular tower... I'm also only about 26 miles from an alltel evdo tower (and like 12 miles from an alltel 1xRTT-capable tower) |
| 20:04.14 | J4k3 | with a 150' tower in the side yard. |
| 20:04.34 | ManxPower | Hmmm...I'll bet Alltell would use Verizon's towers in my area |
| 20:04.56 | J4k3 | yeah, I'm near the dividing line... Verizon owns 800B here, Alltel owns 800B to the north |
| 20:05.03 | ManxPower | anyway I'll call verizon up, ask for information about the class action lawsuit so I can join it. |
| 20:05.10 | J4k3 | Cingular owns 800A here, "Dobson Wireless" (Cellular One) owns 800A north of here. |
| 20:05.17 | *** join/#asterisk yakkop (n=yakkop@c-69-181-237-92.hsd1.ca.comcast.net) |
| 20:05.45 | ManxPower | then let offer to stay with them if they do not cancel my account if I am more careful about my usage. |
| 20:05.45 | J4k3 | http://www.intrastar.net/~jsuter/stuff/finishedtowerpics/ <- my tower |
| 20:05.57 | yakkop | hi... anyone here doing sms with asterisk in the US? can you recommend a gateway? |
| 20:06.28 | hrmphh | erm |
| 20:06.28 | ManxPower | yakkop: nobody does SMS in the USA |
| 20:06.32 | hrmphh | where did cvs.digium.com go? |
| 20:06.46 | ManxPower | hrmphh: they don't use CVS anymore |
| 20:06.54 | hrmphh | what do they use |
| 20:06.56 | ManxPower | they use subversion aka svn |
| 20:06.59 | dwmw2_BOS | they changed to some other already-obsolete system |
| 20:06.59 | hrmphh | ah |
| 20:07.01 | hrmphh | svn |
| 20:07.02 | hrmphh | word |
| 20:07.07 | bulle | ManxPower: why doesnt usa people use sms ? |
| 20:07.08 | dwmw2_BOS | hg or arch or svn or something |
| 20:07.33 | ManxPower | bulle: maybe because carriers do not want to deal with end users running their own gateways |
| 20:07.47 | Hmmhesays | some carriers do |
| 20:07.55 | hrmphh | hmm not seeing on wiki |
| 20:08.13 | ManxPower | hrmphh: The Wiki: |
| 20:08.14 | ManxPower | ?? |
| 20:08.21 | yakkop | ManxPower: hum... are you aware of any 3rd party gateways.... via http or email? |
| 20:08.23 | J4k3 | americans don't have a lot of time to stand around and tap on their phones |
| 20:08.31 | J4k3 | the SMS target market in the USA is like 12-16 years old |
| 20:08.31 | ManxPower | yakkop: oh zillions of them |
| 20:08.45 | J4k3 | the SMS target market in asia is a LOOOOT wider than that. |
| 20:09.00 | ManxPower | yakkop: pretty much all carriers (that use the SMS protocol or not) have an e-mail gateway |
| 20:09.03 | J4k3 | reason? Americans drive, Asians sit on a bus/train. |
| 20:09.11 | bulle | J4k3: guess europe is more like asia then, also, mms is pretty darn popular nowadays |
| 20:09.12 | hrmphh | why is digium.com so slow? |
| 20:09.14 | ManxPower | there are many commercial companies that have http or other interfaces |
| 20:09.16 | hrmphh | they need more bw or what? |
| 20:09.22 | J4k3 | and cellphones are a bit too much for americans to use... americans REALLY can't handle smsing and driving. |
| 20:09.24 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 20:09.38 | hrmphh | k |
| 20:09.46 | J4k3 | bulle: yeah... americans are a radically different market. |
| 20:10.01 | bulle | J4k3: okies |
| 20:10.05 | [TK]D-Fender | J4k3: Which doesn't stop them one bit! Nor does it prevent application of makeup, or reading! |
| 20:10.16 | J4k3 | bulle: which is why I scratch my head when cellular carriers make (IMHO stupid) moves toward super-uber-capacity and not super-uber-coverage. |
| 20:10.18 | hrmphh | heh ive read entire books while driving |
| 20:10.36 | bulle | hrmphh: now thats pretty darn scary |
| 20:10.43 | bulle | hrmphh: around here if anyone saw you, they would phone the police immediately |
| 20:10.45 | J4k3 | [TK]D-Fender: maybe if they'd read the first page of their SUV's owners manual... ;) |
| 20:10.57 | J4k3 | "don't read, apply makeup, or dick with your cellphone while operating this top heavy beast" |
| 20:11.30 | J4k3 | hey, better a book than a beer! |
| 20:11.43 | hrmphh | bulle; here is ? |
| 20:11.46 | J4k3 | judging by the beer cans in the ditches around here, drinking and driving is a local sport. |
| 20:12.34 | J4k3 | I maintain 150 meters of rural 2-lane-road-that-goes-nowhere ditch. I'd say I remove 6 12 ounce (400mL?) beer cans from the ditch every month. |
| 20:12.42 | J4k3 | and thats just a measly 150 meters on ONE SIDE of the road. |
| 20:12.51 | red9012 | does the 'exten' field in dialplan need to be a number, or can it be as 'abc' ? |
| 20:13.47 | ManxPower | red9012: almost anything |
| 20:18.19 | p0g0 | J4k3: you'll like my story- I have about 3/m of road frontage (land was cheap in E. KY in the '60s)...there's cliffs along part of it. I've had to haul out two cars from over the cliff in the last 2 years...drinking +pharmacuticals + cliffs=oops |
| 20:18.33 | p0g0 | *3/4 mile |
| 20:19.11 | *** join/#asterisk Assid (n=assid@203.212.204.107) |
| 20:19.47 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 20:22.33 | *** join/#asterisk X-Rob (n=Rob@ppp214-210.static.internode.on.net) |
| 20:23.08 | hrmphh | hrm |
| 20:23.38 | hrmphh | should i be dl'ing from asterisk/branches/1.4/ on csv or asterisk/trunk? |
| 20:25.22 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 20:25.59 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 20:27.37 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 20:29.36 | irule | can I plesae get some help? I see an error message http://www.pastebin.ca/433464 |
| 20:31.23 | Mercestes | irule: Your pastebin is heinously wrong. |
| 20:31.23 | Mercestes | yo uhave your extensions as part of a macro and you didn't even bother to remove the comments from your sample config. You need to read the book or hire a consultant. |
| 20:31.23 | Mercestes | ~book |
| 20:31.39 | jbot | from memory, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 20:32.49 | Mercestes | irule, 2: you have a warning, u601 is deprecated. use 601,u or 601|u. |
| 20:33.06 | Mercestes | 3: I'm *looking* for your error in your pastebin, I have no clue what's wrong. next time, specify the error you are getting first. |
| 20:33.15 | irule | yes, this is how I dial them! exten => 610,1,Macro(stdexten,610,SIP/sip610) |
| 20:33.25 | irule | isn't that cool? |
| 20:33.53 | hrmphh | on make menuconfig for zaptel, do i really need anything besides wctdm? what does zttranscode do? |
| 20:34.03 | Mercestes | irule, fix your macro |
| 20:34.06 | *** join/#asterisk mvanbaak (n=mafkees@vanbaak.xs4all.nl) |
| 20:34.42 | Mercestes | The macro in make samples should use the new syntax tho. Are these the samples from this distro or did you upgrade? or did you copy this from somewhere else?? |
| 20:35.32 | irule | Mercestes if you don't mind, can you tell me what 601,u or 601|u actually means_ of at least please tell me what to search for, I am really doing everything I can to understand clearly what I am doing, thanks a lot |
| 20:36.05 | irule | yes I copied from a working pre 1 version heh |
| 20:36.14 | Mercestes | that explaisn why it's broken. |
| 20:36.34 | Mercestes | maybe you should try "make samples" in your source directory to get the samples taht go with your version of asterisk so this breakage does not occur. |
| 20:36.35 | Mercestes | aye? |
| 20:37.12 | *** join/#asterisk davidcsi (n=davidcsi@53.red-82-158-35.user.auna.net) |
| 20:37.52 | hrmphh | erm |
| 20:37.55 | davidcsi | question guys: I want extract the ip from ${CHANNEL} into a variable... how would i do that? REGEX doesn't seems to work... |
| 20:38.01 | irule | yes I have the samples backed up, I just have another question, where is the documentation that I created with make doc{insert correct there} that needed the doxygen to work correctly? I recall that is info from the source files |
| 20:38.14 | hrmphh | why does zaptel install an /etc/default/zaptel with all sorts of crazy modules enabled when all i did was enable "wctdm" using make menuselect? |
| 20:39.56 | *** join/#asterisk Darastacat (n=Darastac@APuteaux-152-1-43-230.w82-120.abo.wanadoo.fr) |
| 20:39.58 | Qwell[] | ManxPower: ping |
| 20:41.59 | Darastacat | hello I have a very easy question... but I can't find the answer on the asterisk website... it's not a technical question, I'd just like to know under which license is distributed Asterisk |
| 20:42.16 | Qwell[] | Darastacat: GPL |
| 20:42.16 | davidcsi | is Asterisk able to discriminate routing based on incomming ip address |
| 20:42.29 | Darastacat | thx Qwell |
| 20:42.32 | *** join/#asterisk Mavvie (n=edwin@ppp16-31.lns2.syd7.internode.on.net) |
| 20:43.07 | Assid | _VoicePulse: you there |
| 20:43.08 | Darastacat | have a nice evening |
| 20:43.11 | *** part/#asterisk Darastacat (n=Darastac@APuteaux-152-1-43-230.w82-120.abo.wanadoo.fr) |
| 20:44.55 | Assid | does voicepulse have known issues for DTMF |
| 20:44.55 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 20:45.50 | *** join/#asterisk Strom_M (n=strom@70.141.71.195) |
| 20:46.35 | Mercestes | Assid, dtmfmode = auto and canreinvite=yes |
| 20:47.45 | *** join/#asterisk bkw_ (i=brian@ppp-70-128-123-137.dsl.tulsok.swbell.net) |
| 20:49.28 | Assid | canreinvite affects dtmf? |
| 20:49.31 | irule | I just hang up and the phone rang shortly and CLI says this whats up? == Spawn extension (default, s, 5) exited non-zero on 'SIP/sip603-081bbae0' |
| 20:51.48 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 20:52.13 | Assid | Mercestes: ? |
| 20:52.24 | Mercestes | Assid yes |
| 20:53.25 | *** join/#asterisk ucfMethod (n=ucfmetho@office.eyestreet.com) |
| 20:53.41 | Assid | why should it affect dtmf? |
| 20:53.47 | Mercestes | You sure you want to know? |
| 20:53.58 | Assid | if its small and sweet sure |
| 20:54.04 | Mercestes | It's not. |
| 20:54.18 | Mercestes | it's a large, complex multistep screw up |
| 20:54.53 | Assid | h then no |
| 20:54.58 | Mercestes | :) |
| 20:55.03 | Assid | i wouldnt mind a gist / compressed version tho |
| 20:55.24 | Mercestes | Ok. |
| 20:55.54 | Mercestes | DTMF is *supposed* to be Step 1: Event started, and then a series of "DTMF continued" with a duration, and when you realease the button another DTMF event with a total duration. |
| 20:56.22 | Assid | right which has been enhanced with the new 1.4 |
| 20:56.27 | Mercestes | The "dtmf continues" are supposed to be transmitted as logn as yo uhold the button. |
| 20:56.28 | Assid | for dtmf time pressed |
| 20:56.33 | Mercestes | I dunno. |
| 20:56.44 | Mercestes | if you rhaving DTMF issues then....*shrugs* I haven't read the source. |
| 20:56.59 | russellb | </3 dtmf |
| 20:57.00 | russellb | that is all |
| 20:57.00 | Mercestes | Asterisk, (atleast previously) collects *ALL* those events, and blasts them all otu at once when you release the button. |
| 20:57.16 | russellb | that was 1.2 style, heh |
| 20:57.41 | Mercestes | Some Cisco switches completely ignore all the status messages and simply transmit DTMF upon recieving events because.....everyone transmits events while the event is happenign right? |
| 20:57.48 | Assid | err doesnt 1.4 send out the time at the end of how long it was pressed ? |
| 20:58.01 | Mercestes | so when Cisco meets asterisk cisco sees one big blast of DTMFs which results in about a 40 ms tone. |
| 20:58.18 | Mercestes | RFC says 30ms should be able to be read, but, on some IVRs this is not true. |
| 20:58.27 | russellb | 1.4 sends rfc2833 properly, meaning that it sends a begin event, continuation frames, and an end at the real length of the digit |
| 20:58.50 | Mercestes | russellb, sweet. |
| 20:58.53 | Mercestes | assid: There you go. |
| 20:59.05 | Assid | hrmm |
| 20:59.08 | Mercestes | assid: canreinvite =yes should *not* alter DTMF behavior in 1.4 then. |
| 20:59.24 | Mercestes | canreinvite=yes just caused asterisk to handoff the call handling (and thus dtmf) to the cisco switches. |
| 20:59.35 | Assid | but that depends on what voicepulse uses |
| 20:59.42 | *** join/#asterisk MrTelephone (n=test@bas13-toronto63-1178012833.dsl.bell.ca) |
| 20:59.45 | Mercestes | Assid: well if your using 1.4 it shouldn't matter. |
| 21:00.04 | Assid | well 3 people earlier today couldnt negotiate the IVR dtmf |
| 21:00.07 | MrTelephone | is there a way to get 2 ata186 to act as an fxs etender over IP? Like a hotline.. without a sip server? |
| 21:00.15 | Mercestes | It is funny tho that "voice pulse" is having DTMF problems tho. maybe you should try humming DTMF into the phone. >.> |
| 21:00.21 | hrmphh | anyone have a nice file to use for "tone on hold"? |
| 21:00.53 | Mercestes | MrTelephone, I'm giong to go out on a limb and say "no." |
| 21:01.13 | MrTelephone | im looking for a device that will go FXS -> ip <- FXS |
| 21:01.43 | Assid | MrTelephone: get a sip server then |
| 21:01.45 | Mercestes | MrTelephone, two quintums |
| 21:02.32 | MrTelephone | too bad ciscoata186s needs a sip? i guess there is no internal dialplan |
| 21:02.42 | navigo | anyone here help with bridging calls? |
| 21:03.45 | navigo | better still, can anyone tell me how to make the Answer time in the CDR record be the time the remote party answers? |
| 21:03.50 | CunningPike | Has anyone here successfully gotten IMAP voicemail working on an MS Exchange server? |
| 21:05.17 | MrTelephone | navigo it does... but it probably only works over pri |
| 21:05.25 | MrTelephone | properly |
| 21:05.33 | MrTelephone | there is answer time in the cdr |
| 21:07.59 | b11d | ohhh.. let's bring back "/<-r4d" |
| 21:08.01 | Mercestes | MrTelephone, the quintum FXS things can do it but it's nearly impossible to setup |
| 21:08.44 | MrTelephone | impossible to setup? |
| 21:08.58 | hrmphh | yeah ./configure is non-deterministic |
| 21:11.51 | Mercestes | MrTelephone, yea. we called Quintum support and their solution was to log in and do it and not give us directions |
| 21:12.17 | MrTelephone | oh |
| 21:12.21 | MrTelephone | maybe mediatrix |
| 21:12.44 | *** join/#asterisk `p4r14h`work (n=josh@72.22.238.36) |
| 21:12.47 | Mercestes | you should be able to do it with a Cisco IAD sip router |
| 21:12.59 | Mercestes | but honestly.... |
| 21:13.04 | Mercestes | what your asking for is a phone switch really |
| 21:13.11 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:13.20 | Mercestes | a *cheap* phone switch |
| 21:13.58 | MrTelephone | they have an nec phone system and i was just trying to help a guy find a cheaper solution |
| 21:14.02 | MrTelephone | nec's fxs over ip unit is like 1500 |
| 21:16.57 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
| 21:17.26 | MrTelephone | oh well cna't find it |
| 21:17.27 | MrTelephone | shitty |
| 21:17.40 | Mercestes | So use a T1 card or an FXS card. |
| 21:17.42 | Mercestes | ...gah. |
| 21:17.50 | *** join/#asterisk dc3aes (n=matt@S01060001023fe8ca.no.shawcable.net) |
| 21:20.51 | *** join/#asterisk Vec (n=Vec@dsl-244-208-173.telkomadsl.co.za) |
| 21:21.50 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
| 21:23.05 | murdmath | [TK]D-Fender: Thanks for your help. I got it working. |
| 21:23.30 | [TK]D-Fender | murdmath, good to hear.... |
| 21:26.29 | murdmath | ~pb |
| 21:26.38 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 21:27.46 | hrmphh | only really random problem |
| 21:27.51 | hrmphh | just upgraded to latest cvs for branch/1.4 |
| 21:28.00 | hrmphh | and now my analog phone connected to fxs port is actually super strange |
| 21:28.12 | murdmath | [TK]D-Fender: Here was the final result. http://pastebin.ca/433558 |
| 21:28.15 | b11d | well im off for the night.. cya lads |
| 21:28.23 | hrmphh | if i dial more than a couple digits, i get fast busy?? |
| 21:28.28 | hrmphh | softphones work phone |
| 21:28.32 | hrmphh | incoming calls work fine |
| 21:29.31 | [TK]D-Fender | murdmath, looks about right :) |
| 21:29.54 | [TK]D-Fender | murdmath, I might have jsut done a raw calc on epoch thogh. SIMPLIFY! |
| 21:29.56 | hrmphh | this problem is driving me crazy |
| 21:30.06 | hrmphh | -vvvvv is showing nothing |
| 21:31.07 | hrmphh | i get fast busy and the console shows Hungup 'Zap/1-1' |
| 21:31.55 | murdmath | [TK]D-Fender: :) |
| 21:32.37 | hrmphh | any ideas? :( |
| 21:32.55 | yenno | if my clients use nat - who must keep the connection open? asterisk or the clients? |
| 21:33.09 | [TK]D-Fender | yenno, typcally * |
| 21:33.13 | [TK]D-Fender | ~sipnat |
| 21:33.20 | jbot | sipnat is, like, for for more information about configurtion of Asterisk with SIP behind NAT, see http://voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 21:33.34 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 21:33.39 | yenno | but what messages does asterisk send to keep it open, [TK]D-Fender? |
| 21:33.51 | yenno | thanks |
| 21:36.08 | [TK]D-Fender | yenno, "qualify=yes" SIP Options packets |
| 21:36.26 | [TK]D-Fender | yenno, And typically "nat=yes" for their entries as well |
| 21:48.18 | *** join/#asterisk thoughtpolice (n=austin@c75-111-145-138.plaicmtc01.tx.dh.suddenlink.net) |
| 21:49.05 | hrmphh | tk |
| 21:49.09 | hrmphh | any idea why im getting fast busy |
| 21:49.17 | hrmphh | dialing on my fxs port (zap/1-1)? |
| 21:49.22 | hrmphh | after upgrading asterisk and zaptel? |
| 21:49.29 | [TK]D-Fender | hrmphh, Depends if you are going to give us more that a 1-line pasted error... |
| 21:50.33 | hrmphh | ok what do you want |
| 21:50.40 | hrmphh | thats honestly all console is showing |
| 21:51.12 | *** join/#asterisk bawb2 (n=bawb2@ip51051.estcmp.ku.edu) |
| 21:51.12 | [TK]D-Fender | hrmphh, I want full CLI output, your zaptel & zapata setup, and dialplan contexts where applicable |
| 21:51.18 | [TK]D-Fender | ~pb |
| 21:51.25 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 21:51.44 | hrmphh | k |
| 21:55.18 | hrmphh | trying to recompile zaptel before pasting all that |
| 21:55.24 | hrmphh | i should only need wctdm module correct? |
| 21:55.45 | hrmphh | for tdm13b card |
| 21:56.56 | *** join/#asterisk ppyy (i=ppyy@58.216.30.140) |
| 21:58.14 | [TK]D-Fender | hrmphh, please provide the information I have just asked for... |
| 21:58.29 | hrmphh | k |
| 21:58.53 | *** part/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
| 21:59.11 | *** join/#asterisk IOscanner (n=IOscanne@cpe-76-187-194-128.tx.res.rr.com) |
| 22:01.03 | hrmphh | box is rebooting, will gather all up |
| 22:04.20 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 22:07.05 | *** join/#asterisk wunderkin (n=kev@dslstat-ppp-95.fastq.com) |
| 22:08.13 | hrmphh | christ i put in a single extension '6' to dial and sometimes it work, sometimes it doesnt |
| 22:08.18 | hrmphh | sometimes you hit 6 and it does nothing |
| 22:08.23 | hrmphh | then wait 5 seconds and works |
| 22:08.37 | hrmphh | tk, you want /etc/zaptel.conf and /etc/asterisk/zapata.conf? |
| 22:08.38 | danp | analog phone? |
| 22:08.42 | hrmphh | danp; yes |
| 22:08.46 | danp | relaxdtmf maybe |
| 22:08.51 | danp | or turn up your rxgain |
| 22:09.05 | hrmphh | what is relaxdtmf and where do i set that? |
| 22:09.05 | danp | i just had that problem the other day...relaxdtmf fixed it. |
| 22:09.14 | danp | google it, and zapata.conf |
| 22:09.42 | danp | it was funny...i was trying to dial 101. it would miss the 1 but pick up the 0 |
| 22:09.54 | [TK]D-Fender | hrmphh, Ok, this has taken too long and I have to go. I'll be back in a few hours though someone else may be able to help you with what you should have provided. |
| 22:10.00 | danp | then i tried dialing out...a number with digits all >= 6 and it worked |
| 22:10.06 | danp | it was basically 1-5 that had problems |
| 22:10.39 | hrmphh | weird |
| 22:10.39 | hrmphh | did you recently update src? |
| 22:12.10 | danp | it is a recent svn checkout but i'm working with all new hardware...new machine, new T1 card and channel banks |
| 22:12.11 | danp | so i don't have anything to compare this exact setup to |
| 22:12.13 | hrmphh | was your machine just hanging up? |
| 22:12.13 | hrmphh | soon as i dial a couple of #s it goes |
| 22:12.22 | hrmphh | HUngup 'Zap/1-1' |
| 22:12.22 | danp | yeah |
| 22:12.31 | hrmphh | seriously |
| 22:12.35 | danp | if you have full logging turned on you can see which digits it's grabbing or not |
| 22:12.40 | BSD_Tech | man I forgot what its like to write dialplan |
| 22:12.44 | hrmphh | how do you turn that on? |
| 22:12.54 | danp | in my case it missed the 1 and then got the 0 so it thought i was dialing the operator |
| 22:13.11 | hrmphh | where do i set full logging? |
| 22:13.36 | danp | there's an example entry for it in logger.conf |
| 22:13.43 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 22:15.44 | *** join/#asterisk Modcuts (n=Moducts@88-111-107-180.dynamic.dsl.as9105.com) |
| 22:15.48 | hrmphh | you turned debug on? |
| 22:15.51 | hrmphh | cause verbose doesnt seem to show it |
| 22:16.02 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-82-81-107-14.red.bezeqint.net) |
| 22:17.03 | Modcuts | Does anybody have any ideas on working out what is crashing asterisk if you get Asterisk ended with exit status 139 |
| 22:17.03 | Modcuts | Asterisk exited on signal 11. |
| 22:17.11 | _VoicemeUpDotCom | interface.c:215 decodeMP3: Junk at the beginning of frame 00000000 |
| 22:17.16 | _VoicemeUpDotCom | what does that mean ? is it bad ? |
| 22:18.42 | *** part/#asterisk benjamin7062 (n=bhudgens@64-132-190-102.static.twtelecom.net) |
| 22:18.43 | Modcuts | I have checked the full log and there is no warnings showing up before the crash, and asterisk is failing to start everytime? |
| 22:19.05 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 22:19.28 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 22:19.38 | Mercestes | _VoicemeUpDotCom, It can or can not be bad. It's basically, well, junk at the beginning of the voice frame. |
| 22:19.49 | Mercestes | It was unintelligable binary crap |
| 22:27.37 | *** join/#asterisk Faquin_ (n=Juan@168.226.113.124) |
| 22:28.50 | tzafrir_laptop | hi Faquin_ |
| 22:29.23 | hrmphh | hrm for everyone listening to my spam earlier |
| 22:30.26 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 22:30.52 | Faquin_ | hi all |
| 22:31.59 | _VoicemeUpDotCom | emergency |
| 22:32.00 | _VoicemeUpDotCom | zap down |
| 22:32.01 | _VoicemeUpDotCom | chan_zap.c:900 zt_open: Unable to specify channel 1: No such device or address |
| 22:32.08 | _VoicemeUpDotCom | it disappeared after reboot i cant run |
| 22:32.11 | _VoicemeUpDotCom | any idea ? please |
| 22:32.28 | _VoicemeUpDotCom | wanrouter status all ok |
| 22:34.07 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 22:34.07 | Faquin_ | sorry, no idea for zap |
| 22:34.28 | Faquin_ | i am just starting to use and configure asterisk |
| 22:34.35 | _VoicemeUpDotCom | wanpipe1: Wanpipe device is registered to Zaptel span # 1! |
| 22:34.39 | _VoicemeUpDotCom | a;; good |
| 22:34.43 | *** join/#asterisk Cybertoy (n=cybertoy@dsl254-123-112.nyc1.dsl.speakeasy.net) |
| 22:36.58 | Cybertoy | uh .. question regarding function ENUMLOOKUP ... according to doc/enum.txt there's no longer a reference to enum.conf. .. is that config file still needed? |
| 22:37.02 | Cybertoy | and if yes: why? |
| 22:37.04 | *** join/#asterisk Jon335_ (i=jon335@unaffiliated/jon335) |
| 22:37.11 | Cybertoy | since the domain is an argument of the function? |
| 22:39.15 | tzafrir_laptop | _VoicemeUpDotCom, look at /proc/zaptel |
| 22:39.45 | tzafrir_laptop | Faquin_, you wanted to ask something? |
| 22:40.17 | Faquin_ | not for now |
| 22:40.26 | _VoicemeUpDotCom | yes it was there |
| 22:40.42 | _VoicemeUpDotCom | it seems while i updated.. sangomas poorly written scripts overwrote mine |
| 22:40.51 | Faquin_ | i am haveing troubles with sip registration with asterisk |
| 22:40.53 | _VoicemeUpDotCom | and no ztcfg -vvv so imanually added |
| 22:41.02 | Faquin_ | for register to sip providers |
| 22:41.55 | Faquin_ | i can register with two sip diferente providers, but after a pair of hours |
| 22:42.18 | Faquin_ | it starts to fail in "registration timed out" |
| 22:43.19 | Faquin_ | Apr 10 12:28:54 NOTICE[6695] chan_sip.c: -- Registration for '2939267@sip.megavox.com.ar' timed out, trying again (Attempt #4) |
| 22:43.20 | Faquin_ | Apr 10 12:29:07 NOTICE[6695] chan_sip.c: -- Registration for '63850071@sip.bbtel.net' timed out, trying again (Attempt #8) |
| 22:43.26 | Faquin_ | this is the error message |
| 22:43.41 | Mercestes | What version of wanrouter VoicemeUpDotCom? |
| 22:44.18 | Faquin_ | i have set to full to log system, ans waiting to it fails again to see anything "new" to get more help |
| 22:44.56 | hrmphh | how do you resolve the fxotune error "could not fill input buffer" |
| 22:44.57 | hrmphh | ? |
| 22:45.08 | Faquin_ | sorry, my native language is not english, i hope it is enough clear to you for understan my problem |
| 22:45.10 | Faquin_ | :) |
| 22:45.27 | hrmphh | seeing some mailing list entires on it but no resolution |
| 22:45.34 | _VoicemeUpDotCom | wanrouter version |
| 22:45.34 | _VoicemeUpDotCom | WANPIPE Release: 2.3.4-4 |
| 22:45.38 | _VoicemeUpDotCom | my bad i rebooted a blade.. |
| 22:45.47 | _VoicemeUpDotCom | and didnt think a script could be fuck..ing me up |
| 22:45.54 | _VoicemeUpDotCom | oups.. the f word again.. |
| 22:46.10 | Mercestes | nice that you dotted out whatever came between fuck and ing. thank you for that. :P |
| 22:46.28 | Mercestes | and in my experience sangoma scripts have been fairly helpful |
| 22:46.56 | Mercestes | any reason your not using 2.3.4-7? |
| 22:47.15 | Mercestes | latest at the time I'm guessing? |
| 22:50.12 | *** part/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 22:51.29 | _VoicemeUpDotCom | hmm lazy to upgrade and no problem in 60 days |
| 22:51.44 | _VoicemeUpDotCom | except my dumb..ass rebooting a blade to clear out a pri bug.. |
| 22:51.55 | Mercestes | heh |
| 22:52.06 | Mercestes | /dev/zap is owned by whom? |
| 22:52.27 | *** part/#asterisk ucfMethod (n=ucfmetho@office.eyestreet.com) |
| 22:52.43 | tzafrir_laptop | hrmphh, can you call out though the same fxo channel? |
| 22:52.54 | tzafrir_laptop | (when Asterisk is running, that is) |
| 22:53.09 | _VoicemeUpDotCom | i fixed |
| 22:53.09 | _VoicemeUpDotCom | lol |
| 22:53.30 | _VoicemeUpDotCom | like i said.. asterisk coudnt find channel 1 but wanrouter was saying all ok hence ztcfg -vvv didn run first |
| 22:53.49 | Mercestes | Nice |
| 22:53.55 | _VoicemeUpDotCom | so all ok after that and i MOTD a big banner to remind me in case it rhappens |
| 22:53.57 | hrmphh | tza; oh yeah of course |
| 22:54.02 | Mercestes | lol |
| 22:54.03 | hrmphh | all the fxo and fxs chans work fine |
| 22:54.05 | _VoicemeUpDotCom | /etc/motd is a nice sticky not per box |
| 22:54.12 | hrmphh | and /dev/zap is owned by root:dialout |
| 22:54.16 | hrmphh | im running fxotune as root |
| 22:54.20 | Mercestes | I just edited my /etc/init.d to run wanrouter start and wanrouter stop before and after it's scripted ztcfg vvvvvv |
| 22:54.31 | _VoicemeUpDotCom | yes didnt work |
| 22:54.42 | Mercestes | worked for me. (tm) |
| 22:54.52 | _VoicemeUpDotCom | for you it did (tm) (c) 2007 |
| 22:54.53 | _VoicemeUpDotCom | ;) |
| 22:54.56 | Mercestes | :) |
| 22:55.16 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:56.32 | Mercestes | bye |
| 23:01.23 | hrmphh | tza; any idea/ |
| 23:05.45 | Faquin_ | anyone knows about any softphone in linux? |
| 23:05.48 | *** join/#asterisk Qwell[] (i=qwell@pdpc/sponsor/digium/Qwell) |
| 23:05.48 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 23:05.50 | Faquin_ | i am using gentoo linux |
| 23:06.35 | Faquin_ | exit |
| 23:06.37 | Faquin_ | quit |
| 23:06.38 | Faquin_ | :quit |
| 23:07.32 | *** join/#asterisk yenno (i=yunien@84-72-188-64.dclient.hispeed.ch) |
| 23:07.52 | yenno | sorry, got disconnected. thanks [TK]D-Fender :) |
| 23:08.52 | Nugget | ^ and this is your brain on linux. any questions? :) |
| 23:08.52 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 23:09.51 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 23:09.59 | hrmphh | can someone who is familiar with fxotune take a look at an strace at http://www.pastebin.ca/433685? looks like fd 4 is /dev/zap/2 and fxotune is aborting because its getting ERRNO_500 when trying to read that fd? |
| 23:11.44 | *** join/#asterisk CrashHD (n=crashhd@c-67-166-155-233.hsd1.ca.comcast.net) |
| 23:24.05 | *** join/#asterisk `p4r14h`work (n=josh@72.22.238.36) |
| 23:24.13 | *** join/#asterisk goozbach (n=goozbach@brooks.netradius.com) |
| 23:24.42 | goozbach | ~seen in-pt |
| 23:25.54 | jbot | in-pt <n=lokesh@estrela.nortenet.pt> was last seen on IRC in channel #asterisk, 61d 7h 34m 48s ago, saying: 'and use cisco password'. |
| 23:25.56 | *** join/#asterisk [-Quasar-] (n=jelkj@h8441149158.dsl.speedlinq.nl) |
| 23:27.30 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 23:27.57 | hrmphh | wooo figured out my problem |
| 23:28.09 | hrmphh | stupid telco goes to non-silence after like 2 seconds |
| 23:28.14 | hrmphh | when using default of '5' to clear line |
| 23:28.22 | hrmphh | changed it to 1 and i get like 10 secs |
| 23:28.23 | hrmphh | at most |
| 23:34.09 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 23:37.24 | *** join/#asterisk pfn (n=pfnguyen@netblock-66-245-252-239.dslextreme.com) |
| 23:37.52 | goozbach | anyone seen issues with a snom phone, when placing someone on hold causes asterisk to crash? |
| 23:38.12 | goozbach | I saw in the logs that in-pt was having the same problem, but it didn't seem to get solved |
| 23:38.56 | goozbach | I've checked moh.conf, tried the various ways of canreinvite in sip.conf, and have googled the error message I'm getting |
| 23:39.27 | *** join/#asterisk saftsack (n=saftsack@pD9E07871.dip.t-dialin.net) |
| 23:41.11 | goozbach | this is the error I'm seeing: http://rafb.net/p/23578218.html |
| 23:47.14 | *** join/#asterisk Plecebo (n=larry@D-128-208-60-94.dhcp4.washington.edu) |
| 23:51.21 | BSD_Tech | http://pastebin.ca/433721 have fun more to come |
| 23:51.27 | BSD_Tech | still documenting |
| 23:51.34 | BSD_Tech | and fixing |
| 23:51.52 | *** join/#asterisk Faquin_ (n=root@168.226.113.124) |
| 23:52.01 | BSD_Tech | if you find braks let me know |
| 23:57.56 | demlak | whats the best softphone, protokoll and codec to connect a laptop from anywhere in the world over an openvpn connection to my asterisk? itīs just for me, to use my home isdn line anywhere in the world |
| 23:58.24 | *** part/#asterisk Cybertoy (n=cybertoy@dsl254-123-112.nyc1.dsl.speakeasy.net) |
| 23:58.33 | demlak | and i want to use a bluetooth headset under windows xp |
| 23:59.11 | demlak | on client side =) |
| 23:59.13 | GreyFoxx | eyebeam/xlite is pretty good I use it both direct to the asterisk and over openvpn when I'm remote |
| 23:59.52 | demlak | xlite with SIP is what i confugured yet.. but i thaught there might be somehing better |
