| 00:00.20 | neron | ginvent... |
| 00:00.22 | *** join/#asterisk eldu (n=damajor@tuxmania.org) |
| 00:01.10 | ai-a | Freman: http://images.google.com/images?q=powered%20by%20asterisk |
| 00:02.15 | zoa | none of those are the real thing |
| 00:02.27 | zoa | the real thing is pink and says "asstricks inside" |
| 00:03.25 | Freman | none of them are 80x15 |
| 00:03.51 | supaigtr | shmaltz: vpm? |
| 00:04.02 | Katty | hmm. |
| 00:05.03 | *** join/#asterisk Assid (n=assid@203.115.64.60) |
| 00:05.36 | Assid | hi |
| 00:05.38 | Assid | can someone help me |
| 00:05.51 | Assid | i am trying to setup inbound context to handle a DID |
| 00:06.03 | Assid | but.. it keeps saying Rejected connect attempt from 66.234.228.170, who was trying to reach '2122398008@' |
| 00:06.15 | *** join/#asterisk roulduke_ (i=4r7lg3co@p508D3EDD.dip0.t-ipconnect.de) |
| 00:09.39 | bkw_ | I need people to call 8666799920 ext 665 |
| 00:09.48 | bkw_ | just a few.. please |
| 00:10.00 | *** join/#asterisk aminorex (n=aminorex@12-23-137-226.dhcp.dlth.mn.charter.com) |
| 00:10.27 | Assid | can someone help me on this.. thats a live number |
| 00:10.35 | Assid | and it just refuses to pick it up |
| 00:11.20 | supaigtr | Assid: What was the problem? |
| 00:11.33 | Assid | it just doesnt pickup the call |
| 00:11.41 | websae | anyone here use teliax? |
| 00:11.43 | Assid | and it says rejected connect attempt |
| 00:12.01 | supaigtr | Is this a IAX inbound? |
| 00:12.02 | mmlj4 | websae: gimme a few days |
| 00:12.08 | Assid | i have a exten => 2122398008,1,Answer |
| 00:12.14 | Assid | yes its inbound |
| 00:12.28 | criptos | assid, you don have to Answer :) |
| 00:12.45 | Assid | so what am i missing here? |
| 00:12.45 | criptos | you need to dial a channel :) What phones are u using? |
| 00:13.12 | supaigtr | It looks like the call is being rejected from IP. |
| 00:13.43 | Assid | the 66.234 ip ? |
| 00:17.17 | *** join/#asterisk lung (n=lung@24-148-96-186.ip.mhcable.com) |
| 00:20.59 | websae | anyone using gentoo here? |
| 00:23.02 | nick125 | me |
| 00:24.19 | Assid | criptos: are u sure thats not my issue and is a provider issue ? |
| 00:24.33 | paulc | Assid: try insecure=very |
| 00:24.46 | *** part/#asterisk criptos (n=criptos@201.145.229.183) |
| 00:26.41 | *** join/#asterisk Gronker__ (n=Gronker2@70.152.166.254) |
| 00:26.44 | *** join/#asterisk ledoktre (n=jon@209.152.67.6) |
| 00:26.52 | ledoktre | hello, all :-) |
| 00:27.09 | pablasso | hi there |
| 00:27.46 | ledoktre | I have a <hopefully> simple question, regarding a discrepency I am finding with a IAX to SIP migration. |
| 00:28.54 | pablasso | when i try to receive faxes it keeps sending "Training error" what could it be? |
| 00:29.03 | Assid | do i put that in the context in the iax ? |
| 00:30.10 | Assid | nope |
| 00:30.12 | Assid | didnt work |
| 00:30.35 | ledoktre | I am working at migrating an office full of IAXy boxes to the Sipura SPA-1001's. I am noticing on my SPA-841 (A Sipura Sip phone), and on the 1001's, when you dial any extension on the system, sip or iax, there is a good 5-10 second pause. No errors. Pick up an extension still using IAXy, dials immediately. Any ideas? |
| 00:33.01 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:33.55 | *** join/#asterisk NormAst (n=NormAst@toronto-HSE-ppp3971006.sympatico.ca) |
| 00:34.07 | NormAst | Hi All... Long time now type! |
| 00:34.13 | NormAst | now = no |
| 00:34.13 | *** part/#asterisk aminorex (n=aminorex@12-23-137-226.dhcp.dlth.mn.charter.com) |
| 00:34.28 | Assid | paulc: that didnt work |
| 00:34.32 | nick125 | ledoktre: maybe theres some kind of connection issue where it takes a while to connect to the sip phones |
| 00:34.50 | NormAst | Any one able to get NFAS working on a DMS100 switch? |
| 00:35.04 | pr0m | ok. i've narrowed down my broadvoice problem... |
| 00:35.33 | pr0m | it seems that i'm able to register and make a connection using udp port 5060... |
| 00:35.47 | supaigtr | NormAst: Well kinda. |
| 00:35.58 | pr0m | but the rtp stream for audio is only outbound at my gateway. |
| 00:36.12 | pr0m | no inbound rtp packets from broadvoice. |
| 00:36.53 | pr0m | here's the real kicker.... the tech at broadvoice can call me over an ip-to-ip connection. |
| 00:36.59 | *** join/#asterisk JunK-C (i=junky@69.156.121.182) |
| 00:37.11 | pr0m | we think it's the pstn-to-ip translation that's getting mucked up. |
| 00:37.26 | pr0m | do ya'll have any ideas how to fix this or where to look next? |
| 00:37.36 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 00:37.48 | rg1_ | Is there a place I can get some complete dial plans as examples? |
| 00:38.06 | pr0m | rg1_: voip-info.org |
| 00:38.21 | rg1_ | i'm on there - what would be a good search argument? |
| 00:38.25 | *** join/#asterisk JunK-C (n=junky@MTL-HSE-ppp196887.qc.sympatico.ca) |
| 00:38.31 | pr0m | rg1_: dialplan |
| 00:38.39 | rg1_ | thanks |
| 00:38.57 | mmlj4 | phone quote: asterisk server, 4 POTS lines (not provided by me), 6 polycom 501 phones, $600 labor: $3300 (is this a fair price, you think?) |
| 00:39.58 | pr0m | mmlj4: maintenance? |
| 00:42.13 | NormAst | supaigtr: I can't seem to get calls out... |
| 00:42.33 | pr0m | NormAst: join the club. whose your provider? |
| 00:42.41 | NormAst | supaigtr: How do you kinda have it working.. |
| 00:43.05 | *** join/#asterisk Nox_ (n=secret@star.l93.com) |
| 00:43.05 | NormAst | pr0m: I am stilling down at a DMS250 working with some people to get it going.. |
| 00:43.15 | Nox_ | Does asterisk support G726-32 codec? |
| 00:43.15 | supaigtr | I have each and some popping. |
| 00:43.19 | *** join/#asterisk [1]JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net) |
| 00:43.38 | supaigtr | I can dial out and in. |
| 00:43.46 | pr0m | NormAst: why should i know what dms250 is? |
| 00:44.00 | mmlj4 | pr0m: if they break it, i will come |
| 00:44.03 | *** join/#asterisk The_Ball (n=alex@static-227.35.240.220.dsl.comindico.com.au) |
| 00:44.14 | Equinox | My handytone 286 seems to have a codec compatibility issue |
| 00:44.16 | mmlj4 | but yeah, maintenance would be smooth |
| 00:44.43 | pr0m | mmlj4: right. of course. |
| 00:45.01 | Assid | whats it mean with NO AUTHORITY FOUND |
| 00:45.06 | Assid | i tried it with iax2 debug |
| 00:45.25 | mmlj4 | i come and I charge them, dude :-) |
| 00:46.06 | mmlj4 | but yeah, i should get them to agree to a monthly nuisance fee, say a billable hour per month, or something |
| 00:46.15 | pr0m | mmlj4: and you still haven't answered my question. |
| 00:46.29 | mmlj4 | ok, maybe i didn't understand it, then |
| 00:46.56 | pr0m | what is dms250 and why did you assume that i would know? |
| 00:47.17 | greg_work | mmlj4: just don't call it that! |
| 00:47.56 | hardwire | hmm |
| 00:47.58 | hardwire | I am now the companies operator |
| 00:48.00 | hardwire | this sucks |
| 00:48.02 | hardwire | I am being punished for knowing the most about the new system |
| 00:49.01 | mmlj4 | hardwire: lucky you |
| 00:49.08 | hardwire | its sad. |
| 00:49.12 | hardwire | but I also have the sexiest voice. |
| 00:49.16 | hardwire | so I think it works well |
| 00:51.23 | mmlj4 | pr0m: if I understand it right, you're asking me if the hardware has a warranty? yes, i give 1 year labor on everything i sell, plus what ever the hardware has (e.g. a WD hard drive carries a 3-year warranty) |
| 00:51.48 | mmlj4 | does that price include a monthly fee? no, that would be extra, as noted |
| 00:52.08 | *** part/#asterisk GreyFoxx (i=greg@out.of.phaze.org) |
| 00:52.26 | pr0m | nah. but nevermind. it doesn't matter anyway. ;-) |
| 00:52.40 | mmlj4 | so... good price, or bad? |
| 00:53.15 | pr0m | mmlj4: what kind of hardware you getting with the asterisk server? |
| 00:53.44 | mmlj4 | a white-box athlon or something, nothing fancy |
| 00:53.45 | *** join/#asterisk queuetue (n=queuetue@h69-21-252-54.69-21.unk.tds.net) |
| 00:54.09 | mmlj4 | a gig of ram, maybe |
| 00:54.22 | mmlj4 | food & |
| 00:54.50 | queuetue | Is it possible to get IP phone service as good as a POTS line (assuming my bandwidth level isn't the problem)? |
| 00:55.19 | supaigtr | queuetue: Not as good. |
| 00:55.39 | hardwire | is there a semi standard way of calling somebodies voicemail extension.. |
| 00:56.01 | hardwire | I am using _**.,1,VoiceMail(u${EXTEN:2}) |
| 00:56.08 | hardwire | **1234 from the phones |
| 00:56.12 | hardwire | *1234 intercoms |
| 00:56.14 | queuetue | Why can "they" do it, but the VOIP providers can't? Isn't the entire phone system all network packets? |
| 00:56.45 | supaigtr | TDM/POTS is realtime and no its not a packet network. |
| 00:58.06 | queuetue | supaigtr: What about providing SIP service to a remote location to use POTS lines on my asterisk server? Can that be as clear and latency-free as a normal phone call? (Or very close?) Or is providing service to a remote sales force not possible over the Internet? |
| 00:58.35 | queuetue | (or am I tackling this in the wrong way?) |
| 00:59.23 | queuetue | There is a actual physical pair of copper connecting my phone directly to another phone in Brazil, over a POTS Line? |
| 01:00.01 | supaigtr | VOIP can always be made close. But it doesn't beat a can - string - can. |
| 01:00.59 | *** join/#asterisk shift (i=shift@82-39-34-36.cable.ubr01.benw.blueyonder.co.uk) |
| 01:01.50 | queuetue | Well, close enough to not be noticed ... If it walks like a duck, and it quacks like a duck, that's close enough for me. :) Are there recommended books to learn how to squeak out solid VOIP conversations? Because a sipura connecting directly to broadvoice ranges from crystal-clear with a delay to rolling down a hill in an echo chamber... |
| 01:02.29 | queuetue | I assume sticking asterisk in the middle of the transaction can only make it worse - it' can't improve it, can it? |
| 01:03.18 | *** join/#asterisk surfdue (n=surfdue@user-0c6t1g9.cable.mindspring.com) |
| 01:03.38 | supaigtr | queuetue: I haven't had any problems getting good voip from my desk to downstairs. When I try to get it across town over a T1, wireless, or other IP means its definitly not as realible as POTS. Its noticably not a TDM or POTS connection outside of the office at times. |
| 01:04.00 | supaigtr | queuetue: What are you doing? Graphically? |
| 01:04.01 | *** join/#asterisk cjk_ (n=cjk@212.233.32.4) |
| 01:04.56 | queuetue | supaigtr: Main office in Montreal, with an asterisk server in place. Sales people (two in montreal at remote locations, two in midwest, one in new england) use SIP phones to call into and recieve calls from Main server in Montreal. |
| 01:06.29 | supaigtr | Well theres not way to get POTS/TDM quality in that setup but you can try IP. |
| 01:08.45 | websae | anyone here know where i can buy VOIP phones using paypal online? |
| 01:08.48 | Cherbru_ | Anyone here ever use the S100U? |
| 01:08.51 | *** join/#asterisk MrBelvedr (n=tt@ip70-187-237-193.dc.dc.cox.net) |
| 01:09.06 | Cherbru_ | It's the Digium USB FXS adaptor |
| 01:10.18 | Cherbru_ | I'd like to use it as a timing source on a Apple Xserve G5 |
| 01:10.38 | *** join/#asterisk Arnaud (n=StorM@213.161.223.40) |
| 01:11.02 | *** join/#asterisk PBXtech (i=nik@236.sub-70-218-58.myvzw.com) |
| 01:11.13 | Cherbru_ | But I can't fine one anywhere.. CuPhone seems to be selling an OEM version |
| 01:11.33 | PBXtech | isnt today the last day to get the astricon early bird special?? |
| 01:12.53 | insomni_ | supaigtr and queuetue, I think most of our problem is in ethernet and tcp .. many main providers here in europe are running ip services as backbones now, but ATM protokol which is a lot more moderen and supports qos better |
| 01:12.57 | *** join/#asterisk sivana (n=sivana@mixdown.ca) |
| 01:13.50 | Cherbru_ | I need a damn S100U so I can reverse engineer the damn thing |
| 01:15.02 | supaigtr | insomni_: The problem is when you go over a non 1GIG-10GIG link with other traffic. Mostly cause you dont' control those networks and they aren't setup for QOS. Voice wasn't ment to travel over IP without help :) |
| 01:15.08 | *** join/#asterisk Hmmhesays (i=negative@66.173.103.108) |
| 01:15.39 | h3x0r | there aint much of a point in QoS if your network only has voip traffic |
| 01:15.40 | supaigtr | Is there anyone here that just has all sorts of problems getting spandsp, echo can, etc working or am I the only one? |
| 01:15.42 | h3x0r | what would it have priority |
| 01:15.43 | h3x0r | heh |
| 01:15.45 | h3x0r | over |
| 01:16.09 | *** part/#asterisk Cherbru_ (i=jgarland@72.36.136.226) |
| 01:16.28 | *** join/#asterisk Cherbru_ (i=jgarland@72.36.136.226) |
| 01:16.44 | insomni_ | supaigtr, exactly therefore ATM which has I thing 7 classes of trafic specified as standard |
| 01:16.58 | supaigtr | h3x0r: Some qos detects bottlenecks and moves things around or tightens reigns. |
| 01:17.20 | h3x0r | yeah you have QoS to help, and BGP to destroy your voice quality. |
| 01:17.28 | h3x0r | isnt that great |
| 01:17.56 | supaigtr | MPLS |
| 01:18.49 | PBXtech | i cant find the early bird deal grrr |
| 01:22.06 | Insanity4 | early bird deal? |
| 01:22.17 | Insanity4 | Anyways... can anyone help me compile zaptel? I am getting aggrivated :P |
| 01:22.29 | Hmmhesays | only voip traffic over dsl could use qos |
| 01:22.42 | Hmmhesays | make sure your kernel headers are installed |
| 01:24.00 | queuetue | insomni_: I'm not sure what ATM is - can I use that myself, or do I need a special network (IE, not the Internet) to take advantage of it? |
| 01:24.02 | Insanity4 | Hmmhesays - I instaleld the kernel source rpm |
| 01:24.40 | JamesDotCom | insomni_: another protocol, on par with ethernet |
| 01:24.45 | JamesDotCom | generally pretty costly |
| 01:24.49 | Equinox | queuetue- You'd need a special network. |
| 01:25.51 | Insanity4 | Hmmhesays - I do make linux26 and it says error no rule to make target "modules". |
| 01:27.41 | insomni_ | queuetue, that is for special networks only.. it is to "replace" ethernet for example |
| 01:29.00 | insomni_ | my comment was on somenone saying that packetized sound would not work in real world pots .. but it does |
| 01:30.12 | *** join/#asterisk lters (n=lters@mrtcdsl-034.mis.net) |
| 01:32.10 | DarthClue | Insanity4: try just plain make |
| 01:32.27 | *** join/#asterisk NewSole (i=dave@d226-110-153.home.cgocable.net) |
| 01:32.33 | NewSole | Hello folks |
| 01:32.44 | DarthClue | Evening NewSole |
| 01:33.03 | NewSole | Question.... anyone got a 911 Solution yet |
| 01:33.15 | *** join/#asterisk atmel (n=vlad@ruxi.dynamic.ucsd.edu) |
| 01:33.38 | DarthClue | we don't need no stinkin' 911. we take care of our own problems. |
| 01:33.39 | Qwell | isn't the deadline like yesterday? |
| 01:33.54 | lters | Zap/g1/911 :) |
| 01:34.10 | Qwell | Play(no-911) |
| 01:34.37 | lters | system("halt") |
| 01:35.16 | DarthClue | Play(google-gunshotWoundRepair) |
| 01:35.22 | lters | Is there a 911 'plan'? |
| 01:35.32 | *** join/#asterisk shido6 (n=shido6@d57-87-253.home.cgocable.net) |
| 01:35.38 | Insanity4 | DarthClue - plain make gives me the same error. |
| 01:35.38 | DarthClue | Play(google-heartAttackSave) |
| 01:35.41 | Qwell | I always just report my problems at 911.com |
| 01:35.48 | *** join/#asterisk techie (n=gus@70.86.57.50) |
| 01:35.50 | lters | in the cellular world we hear the words - E911 |
| 01:36.13 | NewSole | ok well... if anyone needs 911 let me know.... we bought a large space for 911 and looking to piece it off to make it cheaper if anyone interested |
| 01:36.32 | lters | Enhanced. all 911 calls have to be sent to 911 *provider* |
| 01:36.44 | lters | with the gps info included. |
| 01:37.00 | h3x0r | NewSole: you did what?! bought a new space for 911? |
| 01:37.28 | h3x0r | DarthClue: Yeah dude, all you need is a big hose, a fire extinguisher, and some guns right |
| 01:37.32 | NewSole | ya we bought a live 911 service.... database updated evey 20 min |
| 01:37.51 | lters | do u send the the gps cords? |
| 01:38.07 | h3x0r | where does it work? |
| 01:38.20 | kFuQ | Vonage is planning an initial public offering to raise nearly $600 million. |
| 01:38.21 | NewSole | works though canada and us |
| 01:38.32 | kFuQ | http://www.marketwatch.com/news/story.asp?guid=%7BC013234D-5650-4DE0-AC3D-988C4969034D%7D&siteid=wsj&dist= |
| 01:38.34 | h3x0r | oh so you run the call center |
| 01:38.51 | NewSole | we have a deal with a 911 call center |
| 01:39.06 | greg_work | that should be a universal service.. 911 call centers |
| 01:39.10 | h3x0r | how are they going to dispatch emergency service anywhere? |
| 01:39.19 | Hmmhesays | kernel source isn't the kernel headers Insanity4 |
| 01:39.52 | greg_work | h3x0r: by transferring to a local dispatcher |
| 01:39.56 | queuetue | h3x0r: They don't - they just keep you on the line with a calm voice until the emergency runs it's course.... Kind of 911 light. |
| 01:40.12 | greg_work | or rather, notifying a local dispatch |
| 01:40.47 | NewSole | its a 911 call center that does the 911 for Telus/Fido/Virisin Cells |
| 01:40.47 | greg_work | the call center can handle tons of calls (ie ,when theres a big car accident and 30 people call 911), and don't have to relay duplicates to the local dispatchers who are busy enough already |
| 01:41.04 | greg_work | PSTN should use it too though |
| 01:41.41 | greg_work | oh, and they can have a division where they transfer the calls to where people call asking for stupid non-emergency shit |
| 01:41.42 | *** join/#asterisk JunK-Y (n=junky@MTL-HSE-ppp196887.qc.sympatico.ca) |
| 01:42.14 | greg_work | that division can be responible for hunting them down and doing awful things to them |
| 01:43.20 | *** join/#asterisk nsgn (n=nsgn@cpe-66-69-197-25.austin.res.rr.com) |
| 01:43.42 | nsgn | ok...i'm having a stupid problem |
| 01:44.14 | nsgn | i try to install A@H and the installer finishes copying all files...but freezes every single time on the "installing bootloader" part |
| 01:44.23 | nsgn | so obviously when i restart, its a corrupted install |
| 01:44.29 | nsgn | help! *sniff* |
| 01:44.44 | Qwell | #asterisk@home |
| 01:44.52 | nsgn | oh...why thank you |
| 01:45.11 | nsgn | er...nvm on the thanks |
| 01:45.17 | nsgn | thats not a room |
| 01:45.20 | nsgn | or rather, nobody is in there |
| 01:45.36 | Qwell | then find the right one |
| 01:45.55 | Qwell | this certainly isn't it |
| 01:46.26 | Insanity4 | Hmmhesays - How can I see if I have the neededs kernel headers? where would they be? |
| 01:46.47 | Hmmhesays | i'm sure you can get them in a redhat package just like you can the source |
| 01:46.48 | nsgn | basically i think its a culinux issue |
| 01:46.54 | nsgn | or whatever A@H is based on |
| 01:47.12 | Qwell | nsgn: this still isn't the right channel |
| 01:47.17 | Hmmhesays | centos |
| 01:47.27 | Hmmhesays | later gators i'm going hunting |
| 01:47.38 | lters | Insanity4, apt-get install kernel-header`uname -a` |
| 01:47.46 | Qwell | -r |
| 01:48.03 | nsgn | quell: you made that clear. i was hoping some nice soul who sympathized with me or has a little linux knowledge might help me while i try to stumble around myself. i'm entirely new to both linux and asterisk |
| 01:48.16 | lters | Qwell, thnxs |
| 01:48.19 | Insanity4 | lters - Suse :) |
| 01:48.22 | Qwell | nsgn: it'd be completely offtopic |
| 01:48.32 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 01:48.39 | nsgn | alright, then can somebody give a n00b a total walkthrough of installing real asterisk? |
| 01:48.48 | Qwell | now you're talking |
| 01:48.49 | Qwell | ~docs |
| 01:48.51 | jbot | somebody said docs was Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 01:48.57 | Insanity4 | It was simple, wget, configure, make, make install. From there, start reading. |
| 01:49.06 | lters | Insanity4, /s/Suse/Debian :) |
| 01:49.15 | Insanity4 | lters - apt-get works on suse? |
| 01:49.15 | nsgn | i was using A@H cause i dont know anything about linux. i am computer consultant, so i know computers, but i work Windows/MacOS. i can learn fast, but i just dont know it at this point |
| 01:49.42 | Insanity4 | nsgn - Installing is VERY easy. Configuring it is another story. By linux program standards, asterisk is very well written. |
| 01:49.42 | lters | Insanity4, don't know, I was hinting at a different os tho :) |
| 01:49.47 | Insanity4 | lters - hehe |
| 01:50.01 | nsgn | insanity4: does the full asterisk have that web config interface? |
| 01:50.04 | nsgn | that was nice in A@H |
| 01:50.23 | lters | nsgn, u will learn it better if you do it by hand. |
| 01:50.40 | nsgn | yeah, but is it there for quick stuff like new extensions? |
| 01:50.51 | nsgn | i know for the config i want the web interface isnt extensive enough |
| 01:50.56 | lters | takes all of 2 seconds. |
| 01:51.11 | Qwell | AMP screws your configs |
| 01:51.17 | Insanity4 | nsgn - nope :) |
| 01:51.18 | Insanity4 | hehe |
| 01:51.45 | Insanity4 | It's not that hard to do for basic stuff. |
| 01:51.48 | Insanity4 | But it can get coplicated. |
| 01:51.48 | nsgn | ok |
| 01:51.51 | Insanity4 | complicated :) |
| 01:52.15 | nsgn | my only fear at this point is the installation. i think i just have a bad taste in my mouth with linux installs both from A@H and other stuff i've tried |
| 01:52.23 | Ariel_ | hello all just stopping by to see how folks are doing. |
| 01:52.38 | nsgn | once its installed i'll be alright. i dont mind cong files |
| 01:52.38 | lters | nsgn, do a clean debian install, cvs the * you want. make && make install |
| 01:52.58 | lters | or if you don't need *up to date* you can apt get asterisk... |
| 01:53.12 | nsgn | apt-get sounds easier, doesnt it? |
| 01:53.12 | Qwell | I'd never recommend using a package of asterisk |
| 01:53.39 | lters | Qwell, me neither, but, what is A@h ? |
| 01:53.50 | Qwell | lters: something I'd never recommend using |
| 01:56.03 | lters | nsgn, if your going to do lots of installs, and go debian, you will want a local mirror on your network. |
| 01:56.10 | lters | than installs are a snap. |
| 01:56.28 | nsgn | what do you mean. if i was making multiple * boxes? |
| 01:56.30 | nsgn | i'm just having one |
| 01:56.32 | nsgn | its a home setup |
| 01:56.35 | *** join/#asterisk chendy (n=Alex_Dot@218.1.218.17) |
| 01:56.42 | nsgn | which is why i was toying with A@H...but guess that doesnt feel like working for me |
| 01:56.47 | Ariel_ | nsgn what was your problem with a@H |
| 01:57.06 | nsgn | ariel_: it installed just fine, but right after the install it says "installing bootloader". it freezes there every time |
| 01:57.19 | nsgn | so the machine comes up to the linux bootloader instead of asterisk when it boots |
| 01:57.26 | nsgn | and when you click the only option, it gives a missing file error |
| 01:57.36 | Ariel_ | nsgn, what is the machine |
| 01:57.40 | nsgn | i can go get the specifics of the error if you think you can help. i did the install like...four times |
| 01:57.57 | lters | actually sounds like a hd prob |
| 01:58.00 | nsgn | haha...AMD 233mhz 192mb ram 6gb disk 100mbps ethernet SVGA |
| 01:58.00 | Ariel_ | nsgn what is your computer spec's |
| 01:58.16 | Ariel_ | ok that is not going to work with asterisk@home version 1.5 |
| 01:58.25 | nsgn | well, the same drive was running A@H on another box |
| 01:58.26 | Ariel_ | if you can get version 1.1 it will work on your system. |
| 01:58.28 | nsgn | but i had to take it off that box |
| 01:58.39 | nsgn | ariel_: oh really? thats the problem? |
| 01:58.45 | nsgn | i didnt know that |
| 01:58.55 | Ariel_ | memory and the os that the newer version wants 256mgs |
| 01:59.23 | Ariel_ | But if you get the ISO for 1.1 it will run and it has the current amp 1.10.08 |
| 01:59.34 | nsgn | so its purely a ram issue? |
| 01:59.41 | nsgn | if so i can put 256 on the box |
| 01:59.44 | Ariel_ | also once you get that far you can change the rules to run via your own config files. |
| 01:59.52 | Ariel_ | well that is for starters |
| 02:00.11 | nsgn | i dont see how the ram causes lockup at bootloader installation |
| 02:00.19 | Ariel_ | if you read RH EL 3.5 you need a 500mhz 256mg rams to run it. |
| 02:01.00 | nsgn | hmm. so you're thinking 1.1 is fine for me? i wont loose any totally vital features going to that? |
| 02:01.08 | Ariel_ | not really |
| 02:01.17 | Ariel_ | you don't really need surgarCRM |
| 02:01.56 | nsgn | ok, im going 1.1 hunting... |
| 02:02.48 | *** join/#asterisk Tili (i=Tili@219.136.218.164) |
| 02:03.37 | nsgn | ok, found it at the A@H site |
| 02:03.55 | nsgn | i'll be downloading this and likely trying it tomorrow night. you think this has a good chance of working? |
| 02:04.13 | Ariel_ | yes it should work |
| 02:04.32 | Ariel_ | I have that version running on a P233 with 128 megs of ram. |
| 02:05.16 | nsgn | awesome. thank ya much |
| 02:05.26 | Ariel_ | nsgn, any time. |
| 02:05.33 | nsgn | i'll be around the next few weeks as i'll be getting asterisk running (HOPEFULLY) and then configuring for my use |
| 02:05.34 | nsgn | (home) |
| 02:05.50 | *** join/#asterisk drumkilla (n=russell@12.21.243.167) |
| 02:05.52 | nsgn | well....i guess home is obvious being that i have such an awesome machine to run it on :-D |
| 02:06.18 | Ariel_ | nsgn, there is an area just for amp users which is what a@H comes with it's #amportal |
| 02:06.29 | nsgn | hmmm, ok. |
| 02:06.48 | nsgn | but if i need to do something beyond the capability of AMP i can just go edit configs like i could on asterisk, right? |
| 02:07.01 | Ariel_ | for some reason people here are anti gui and tend not to want to help newbies lately. |
| 02:07.15 | nsgn | cause i have a few things i want this to do that might not be possible by A@H's default setup or AMP |
| 02:07.16 | Ariel_ | nsgn yes and what is it you need beyond |
| 02:07.24 | Ariel_ | like? |
| 02:07.29 | nsgn | well, i'm not sure cause i'm new to asterisk |
| 02:08.05 | Ariel_ | We have people running billing systems off it, we have people running multi company and multi servers with it. |
| 02:08.09 | nsgn | but prolly a message that picks up after all phones ring five times (like a home answering machine) but gives options to go to people's different voicemail boxes by pressing keys |
| 02:08.32 | Ariel_ | nsgn, that is the very basic setup part |
| 02:08.47 | nsgn | ah, well then all the better for me ;-) |
| 02:09.30 | Ariel_ | example I have it ring the house phones 4 times then goes to a digital recp that directs them to any of the mail boxes or even trys my cell phone. |
| 02:09.33 | nsgn | just making sure the phones ring how i want/when i want, having some voicemail boxes for each family member and my home office, and having that menu to go direct to voicemail boxes if nobody answers the phones is all i need really |
| 02:09.44 | nsgn | what u just said; exactly what i want |
| 02:09.45 | *** join/#asterisk snewpy (n=markl@203-217-78-139.dyn.iinet.net.au) |
| 02:09.50 | nsgn | cept no cell phone transfer |
| 02:09.59 | ledoktre | nick125: sorry, stepped out. the connection, is all on a private lan, 10/100 switch. all the same wiring, i pulled out an IAXy, replaced with the 1001 box, updated asterisk, and boom. pause. I can call the 1001 from an IAXY box, rings rfight away. call from it, tis slow. |
| 02:10.06 | Ariel_ | nsgn, it's just depends on what you want to setup. |
| 02:10.39 | Ariel_ | I even have an extension that I can call put in my password and give me access to long distance via the internet. It's call DISA |
| 02:10.46 | nsgn | cool |
| 02:10.49 | nsgn | yeah i'd heard of that |
| 02:10.50 | nsgn | thats pretty cool |
| 02:11.03 | nsgn | i cant think of anything too insane |
| 02:11.07 | nsgn | but who knows, i might have some fun with it :-P |
| 02:11.21 | nsgn | what would be other useful home things for it to do? |
| 02:11.30 | nsgn | might as well brainstorm while i download hehe |
| 02:11.56 | Ariel_ | well I don't know how much longer I will be on. Hurricane is really bad just over head of my house right now. I found a few water leaks |
| 02:11.58 | nsgn | even though easynews Arizona has a GOOD server. i consistently get 600kbps from them |
| 02:12.22 | nsgn | oh dang. what hurricane? i've been so out of it the past week |
| 02:12.24 | nsgn | work has had me swamped |
| 02:12.32 | Ariel_ | Katrina. |
| 02:12.39 | Ariel_ | I am in Miami, Florida |
| 02:12.49 | Ariel_ | It came ashore about 2 hours ago |
| 02:12.49 | Nugget | Home of the 2005 MTV Video Music Awards! |
| 02:12.51 | nsgn | whoa...*goes to read news and get out from under his cave* |
| 02:13.00 | Ariel_ | ~weather KTMB |
| 02:13.30 | nsgn | hmm, yeah. thats a nice little storm |
| 02:13.42 | Ariel_ | and we are 16miles inland |
| 02:13.45 | nsgn | i'm an amateur meteorologist. you know i've been busy if i missed this. |
| 02:13.53 | nsgn | wow, thats smack dab on the coast |
| 02:13.58 | nsgn | fairly typical trajectory though |
| 02:13.59 | Ariel_ | MTV said it's still going to have the show on |
| 02:14.12 | nsgn | windspeeds typical though. nothing too special, especially compared to recent years |
| 02:14.23 | Katty | what? |
| 02:14.24 | Katty | oh |
| 02:14.30 | Katty | silly people setting of my hilight |
| 02:14.35 | Ariel_ | well it's not Andrew but it's really messing things up outside |
| 02:14.37 | Katty | s/of/off/ |
| 02:14.55 | Ariel_ | my BBQ is at the end of the fence and the shead we had well we had one. |
| 02:15.08 | nsgn | hah, yeah i'd bet |
| 02:15.10 | Ariel_ | Katty, sorry |
| 02:15.12 | nsgn | they still pack a punch |
| 02:15.39 | Ariel_ | just stopping by had to get on line and try to fix a customers system out on the west coast. |
| 02:15.47 | *** join/#asterisk Moc_ (n=mochouin@234-199-0-72-ppp.3menatwork.com) |
| 02:16.12 | Katty | mister moc |
| 02:16.18 | Moc_ | Hi katty |
| 02:18.06 | nsgn | ah |
| 02:18.24 | nsgn | ariel_: well i think i'm good for the night. prolly wont get the chance to try to install 1.1 until late tomorrow anyway |
| 02:18.27 | Ariel_ | damm weather.com has so much adds |
| 02:18.37 | nsgn | so good luck in the storm and i'll be back to you for help sometime soon i'm sure ;-P |
| 02:18.43 | Ariel_ | I hope to be online |
| 02:18.54 | hardwire | time to take my sexy little operator ass home |
| 02:21.12 | *** join/#asterisk drumkilla (n=russell@12.21.243.167) |
| 02:27.53 | *** join/#asterisk drumkilla (n=russell@12.21.243.167) |
| 02:29.34 | *** join/#asterisk toddf (n=toddf@ns0.fries.net) |
| 02:30.39 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 02:31.59 | nsgn | kk im out for real. i'll be back tomorrow night. good luck in the storm ariel_ |
| 02:34.52 | Insanity4 | How can I tell if asterisk is using ztdummy? |
| 02:35.38 | *** join/#asterisk spoot_nick (n=julio@CPE-147-10-168-100.nsw.bigpond.net.au) |
| 02:37.33 | Qwell | Insanity4: zap show channels, look for pseudo, I believe |
| 02:37.47 | Qwell | that tells you if you're using a timing source, afaik |
| 02:39.07 | bkw_ | pseudo is there on all types |
| 02:39.09 | bkw_ | lsmod |
| 02:39.10 | bkw_ | is the best way |
| 02:39.15 | bkw_ | or zttool |
| 02:39.22 | JunK-Y | zap show status |
| 02:39.38 | JunK-Y | but lsmod|grep ztdummy is the solution for sure. |
| 02:40.17 | spoot_nick | does anybody know why i would get this message when somebody is trying to dial a number i registered for in an external sip proxy? |
| 02:40.22 | spoot_nick | chan_sip.c:7327 handle_request: Failed to authenticate user "0433198154" <sip:0433198154@202.92.94.27>;tag=as41632e43 |
| 02:40.27 | Insanity4 | bkw_ - lsmod shows if the module is loaded, butn ot if afterisk is successfully using it. |
| 02:40.36 | bkw_ | use count |
| 02:40.40 | bkw_ | sure does |
| 02:41.02 | bkw_ | if you have a use count next to zaptel and ztdummy is loaded |
| 02:41.04 | bkw_ | its using it |
| 02:41.07 | bkw_ | it uses it if it is there |
| 02:41.09 | bkw_ | and nothing else |
| 02:41.15 | bkw_ | stop trying so damn hard boi |
| 02:41.18 | bkw_ | ztdummy 3648 0 zaptel 190468 9 ztdummy |
| 02:41.22 | JunK-Y | NEXT! |
| 02:41.46 | bkw_ | I find that people that are new to things often try way too hard to understand them |
| 02:41.57 | bkw_ | like your parents trying to set the clock on the VCR |
| 02:42.31 | bkw_ | when I was growing up.. I was know as the person that sets the clocks on the VCR's in the family |
| 02:45.19 | Insanity4 | Should zap show channels show the channel while the music on hold is playing? |
| 02:45.39 | Qwell | So, what is the pseudo channel for? |
| 02:46.18 | *** join/#asterisk chendy_ (n=Alex_Dot@web1.ningo.net) |
| 02:46.28 | *** join/#asterisk Urs_ShPo (n=roth@yona.ursus.net) |
| 02:46.32 | Insanity4 | I don't see anything on zap show channels while the music is playing :( |
| 02:46.59 | supaigtr | I hate 83b faxes. |
| 02:47.12 | *** join/#asterisk atmel` (n=vlad@wireless-am6.ucsd.edu) |
| 02:47.22 | Qwell | is there a point in faxing anymore? |
| 02:48.10 | Insanity4 | Simplicity? |
| 02:48.11 | supaigtr | Not for me. But all these users seem to think its the best damn thing since sliced bread. I've been pulling my hair out for days now. |
| 02:48.29 | Insanity4 | Scan, convert to jpeg, etc. |
| 02:48.34 | Insanity4 | or just... fax... its there. |
| 02:49.08 | supaigtr | I just want fax - > email to work. It works for a few days, then bammo. Just dies. |
| 02:49.23 | Qwell | meh |
| 02:49.26 | supaigtr | Among the other problems I have. |
| 02:49.26 | Insanity4 | I want all my incoming calls to work. 1 in 10 drops :( |
| 02:49.26 | Insanity4 | hehe |
| 02:49.34 | Qwell | When I have a business, and I have an office and employees... |
| 02:49.39 | supaigtr | Insanity4: What hardware. |
| 02:49.40 | Qwell | we aren't going to have fax machines or printers |
| 02:49.47 | Insanity4 | supaigtr - incoming iax from nufone |
| 02:49.59 | *** join/#asterisk aminorex (n=tony@12-23-137-226.dhcp.dlth.mn.charter.com) |
| 02:50.21 | supaigtr | I got no probs with nufone cept the echo and jitter. |
| 02:50.49 | carrar | and they bill you for calls even if you are not connected |
| 02:50.51 | supaigtr | I'm about ready to throw this TDM card thru the wall. Unfortually I put the screw in and can't just pull it out. :) |
| 02:51.02 | Insanity4 | How do I tell if I'm running udev on my sysem? |
| 02:51.19 | supaigtr | Do cat /proc/interrupts |
| 02:51.36 | *** mode/#asterisk [+o drumkilla] by ChanServ |
| 02:52.22 | L|NUX | Starting Asterisk PBX: FATAL: Module ixj not found |
| 02:52.28 | L|NUX | how can i fix this error ? |
| 02:52.39 | JamesDotCom | noload => module_blah.so |
| 02:52.41 | JamesDotCom | in modules.conf |
| 02:53.35 | L|NUX | what is the purpose of this mod |
| 02:54.01 | Insanity4 | supaigtr - not there :) |
| 02:54.24 | Insanity4 | but I do have the udev config file that I added the lines too anyways. |
| 02:54.37 | *** join/#asterisk azrishahril (n=azrishah@219.94.108.230) |
| 02:55.01 | supaigtr | etc udev |
| 02:55.05 | Insanity4 | yes |
| 02:55.07 | Insanity4 | that is present |
| 02:55.16 | Insanity4 | I'm assuming ztdummy won't work until a reboot? |
| 02:55.23 | Insanity4 | I wish asterisk... or soemthign would tell me if its using it :P |
| 02:55.24 | Qwell | Insanity4: incorrect |
| 02:55.27 | Insanity4 | zap show channels shows none in use |
| 02:55.35 | Insanity4 | Qwell - ok... the voip wiki said you had to reboot. |
| 02:55.42 | supaigtr | U tring conferencing with TDM card? |
| 02:55.43 | Qwell | bullshit, heh |
| 02:55.54 | Qwell | Insanity4: link? I'm going to fix it |
| 02:55.55 | supaigtr | Its a module isn't it? |
| 02:55.57 | Insanity4 | Trying to get music on hold to stop sounding like crap. |
| 02:56.08 | Insanity4 | http://www.voip-info.org/tiki-index.php?page=Asterisk+Zaptel+Installation |
| 02:56.16 | Insanity4 | If running a Linux 2.6 system, read the file README.udev, and follow the instructions. |
| 02:56.16 | Insanity4 | REBOOT after you make the changes listed in README.udev |
| 02:56.27 | *** join/#asterisk Koshatul (n=evangeli@ip157.net65.ipnetworks.net.au) |
| 02:56.30 | Insanity4 | http://www.voip-info.org/wiki-Asterisk+timer+ztdummy |
| 02:56.34 | Insanity4 | - Reboot to make udev changes take effect |
| 02:57.36 | flewid | sup |
| 02:58.11 | L|NUX | can we connect CDMA phone with Asterisk Box ? |
| 02:58.16 | L|NUX | any one did this ? |
| 02:59.13 | Insanity4 | L|NUX - cell phone to fx0 adapter, yes. |
| 02:59.16 | Insanity4 | <PROTECTED> |
| 02:59.16 | Insanity4 | Aug 25 22:59:47 WARNING[6074]: chan_zap.c:9651 setup_zap: Ignoring switchtype |
| 02:59.16 | Insanity4 | Waht does that mean? |
| 02:59.16 | Qwell | hmm |
| 02:59.16 | Qwell | a cellphone could be both fxo and fxs, couldn't it? |
| 02:59.16 | spoot_nick | have anyone experienced asterisk trying to auth a caller when he dials in from an external sip number? |
| 02:59.16 | L|NUX | hmm |
| 02:59.16 | JunK-Y | it ignores the switchtype??? |
| 02:59.16 | JunK-Y | ! |
| 02:59.30 | flewid | anyone here use polycom ip500's? |
| 03:00.02 | supaigtr | flewid: hundreds of em. |
| 03:00.02 | flewid | supa: got a weird question for you - we're running the phones on two seperate * servers, one local, one remote |
| 03:00.13 | flewid | both line 1 and line 2 share the same extension #, but are registered to the two diff servers |
| 03:00.14 | *** join/#asterisk lot (n=lot@S0106000f6694b86f.ed.shawcable.net) |
| 03:00.27 | flewid | when a voicemail is left on server b, the phone shows the voicemail on server a |
| 03:00.35 | flewid | and if voicemail is left on server a, it shows voicemail on server a |
| 03:00.43 | Insanity4 | flewid - firmware? limitation? Idon't know :) |
| 03:00.46 | Insanity4 | hehe |
| 03:00.55 | flewid | i'm just wondering if that's because they're the same ext #, and the phone's gettin confused |
| 03:00.59 | supaigtr | U mean vm light? Theres just one to light. |
| 03:01.01 | Qwell | why not just link the * boxes? |
| 03:01.19 | flewid | supa: well you know how the little icon shows on the line if you have a vm |
| 03:01.23 | supaigtr | vm is done by sip account. |
| 03:01.23 | flewid | it always shows on the topmost ext |
| 03:01.26 | flewid | yah |
| 03:01.51 | supaigtr | I'm pretty sure thats firmware. We have similar setup but we only lettem have one voicemail box. |
| 03:02.02 | flewid | ah |
| 03:02.06 | flewid | yeah these guys have one local and one remote |
| 03:02.08 | supaigtr | U running 1.5.2? |
| 03:02.20 | flewid | it's cause they had two pbx's before and we're trying to merge them both, but with as little user headache as possible |
| 03:02.24 | flewid | sec |
| 03:02.48 | *** join/#asterisk eldu (n=damajor@tuxmania.org) |
| 03:03.06 | flewid | boot: 2.6.2.0032 |
| 03:03.16 | flewid | sip: 1.5.2.0054 |
| 03:04.39 | *** join/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
| 03:05.57 | supaigtr | Polycom is a bit behind snom on that but the phone sounds better. |
| 03:06.05 | flewid | yeah i like the polycoms |
| 03:06.11 | supaigtr | DSS consoles are shipping on snoms. |
| 03:06.24 | flewid | i'm a big fan of the aastra's too but i haven't had any time to play with the 480i or 9311 |
| 03:06.34 | flewid | i just have the 390 |
| 03:06.51 | flewid | so you think the voicemail appearing on the wrong line is cause of firmware eh? |
| 03:07.54 | Telamon | I have a hardware question: If I want to interface a POTS line with Asterisk so that people can call in and enter some commands in a voice menu (tones only, no voice capture needed) do I just need an FXO port, or also an FXS port? Will the S101i IAXy do the job, so I don't need any internal cards? |
| 03:08.10 | Insanity4 | fx0. |
| 03:08.15 | Insanity4 | use DISA. |
| 03:08.20 | Insanity4 | OR some custom dialplan |
| 03:08.32 | Telamon | DISA? |
| 03:08.45 | Insanity4 | Direct Inward Service Accesss (dial-in to the system) |
| 03:09.23 | Insanity4 | Telamon - http://www.digium.com/index.php?menu=fxsvfxo |
| 03:09.33 | flewid | telamon: for that setup, i'd get an ATA (iaxy or sipura) and an X100P if you can find one, or a tdm400 with 1 fxo module if you plan on going furthur later |
| 03:09.45 | Telamon | Ah, cool. Any recommendations on a single port FX0 board? It's going in a rackmounted server, so I'm worried about the profile of the standard Digium one. |
| 03:09.47 | Insanity4 | Why can't you buy x100p's anymore? |
| 03:09.59 | JunK-Y | Insanity4: its deprecated. |
| 03:10.00 | Insanity4 | Outside of ebay modem knockoffs |
| 03:10.02 | flewid | they're discontinued, you can find them still but mostly the clones |
| 03:10.02 | supaigtr | flewid: We use a CB with FXO for that. |
| 03:10.03 | Insanity4 | is there any difference? |
| 03:10.06 | Telamon | Cool, you guys are fast. :) Thanks! |
| 03:10.18 | Insanity4 | JunK-Y - But why? I don't want some overpriced 4 port bord :P |
| 03:10.18 | Insanity4 | hehe |
| 03:10.25 | *** join/#asterisk Gronker__ (n=Gronker2@70.152.166.254) |
| 03:10.39 | *** join/#asterisk angom_h (n=angom@red-corp-200.76.229.189.telnor.net) |
| 03:11.13 | *** part/#asterisk Urs_ShPo (n=roth@yona.ursus.net) |
| 03:11.33 | Telamon | Do the IAXy's have an fx0 as well, or just an fxs? |
| 03:11.46 | JunK-Y | Insanity4: why horses are deprecated? cars costs 20 000$ |
| 03:12.13 | fugitivo | hello |
| 03:13.56 | Insanity4 | JunK-Y - How can I positively confirm my ztdummy is being used? Anything in the logs, anywhere? |
| 03:14.15 | fugitivo | asterisk console |
| 03:14.21 | JunK-Y | Insanity4: we already answered that question earlier. |
| 03:15.02 | Insanity4 | JunK-Y - zap show channels doesn't show anythign while I'm on hold w/ MOH. |
| 03:15.18 | Insanity4 | JunK-Y - lsmod just shows the mod is loaded, and it doesn't say anything in use other than ztdummy by zaptel |
| 03:15.27 | JunK-Y | zap show status |
| 03:15.31 | Insanity4 | I still see no sign (positive confirmation) that asterisk itself is using it |
| 03:15.45 | Insanity4 | zap show status is not a valid command. |
| 03:16.09 | *** join/#asterisk Kumbang (n=unknown@167.205.24.5) |
| 03:16.18 | JunK-Y | that command is only available on HEAD. |
| 03:16.25 | Insanity4 | head? |
| 03:17.00 | supaigtr | Anyone seen this? Failed to register zone 'United States / North America': No data available |
| 03:17.14 | *** part/#asterisk Gronker__ (n=Gronker2@70.152.166.254) |
| 03:17.56 | *** join/#asterisk zee001 (n=foobar@MTL-HSE-ppp196887.qc.sympatico.ca) |
| 03:18.03 | JunK-Y | Insanity4: cvs-head |
| 03:18.05 | Insanity4 | JunK-Y - what is HEAD? |
| 03:18.09 | Insanity4 | ahh ok |
| 03:18.20 | Insanity4 | What can I do with the versiono n the website to tell? |
| 03:18.47 | JunK-Y | lsmod|grep ztdummy |
| 03:19.13 | Insanity4 | ztdummy 3620 0 |
| 03:19.13 | Insanity4 | zaptel 232964 1 ztdummy |
| 03:19.33 | JunK-Y | ~naze |
| 03:19.34 | jbot | rumour has it, naze is BB naze |
| 03:19.58 | *** join/#asterisk NSGN (n=NSGN@cpe-66-69-197-25.austin.res.rr.com) |
| 03:20.02 | NSGN | well i'm back early |
| 03:20.03 | NSGN | heh |
| 03:20.06 | NSGN | so 1.1 installed! |
| 03:20.09 | Insanity4 | JunK-Y - What does that tell you? |
| 03:20.25 | NSGN | however...on the first boot the only item to fail so far was bringing ethernet0 online |
| 03:20.34 | NSGN | does that mean my network card is not compatable? |
| 03:20.39 | JunK-Y | ztdummy is up maybe? |
| 03:20.56 | Insanity4 | JunK-Y - But it isn't positive confirmation that asterisk is using it. |
| 03:21.01 | Insanity4 | Does that exist at all? |
| 03:21.09 | Insanity4 | Or if my MOH still sounds like crap... |
| 03:21.36 | JunK-Y | it uses it or chan_zap wouldnt load, u need zaptel to load it. |
| 03:21.57 | NSGN | anybody? its not TOO strange of an ethernet card. its a linksys 100mbps PCI card |
| 03:22.15 | Insanity4 | JunK-Y - Well, I loaded asterisk with chan zap before modprobing it. |
| 03:22.20 | Insanity4 | JunK-Y - On accident, and * still worked. |
| 03:22.37 | fugitivo | NSGN: do you have the right modules loaded? |
| 03:23.05 | JunK-Y | it means ? |
| 03:23.25 | NSGN | fugitivo: its A@H...but it worked on my other pc's ethernet card |
| 03:23.36 | NSGN | but this pc just said FAILED when bringing eth0 online |
| 03:23.40 | Insanity4 | JunK-Y - Talking to me? It means that the module simply loading isn't positive confirmation of ztdummys use |
| 03:23.50 | Insanity4 | What does zap show channels do? |
| 03:23.54 | fugitivo | NSGN: try to load the module at hand and check your msg for errors |
| 03:23.56 | JunK-Y | which mod ztdummy? |
| 03:24.12 | NSGN | fugitivo: alright, i'll do that in a sec. its the first boot so it's compiling asterisk |
| 03:24.22 | *** join/#asterisk bubbajohn (i=bubbajoh@adsl-68-91-7-225.dsl.tulsok.swbell.net) |
| 03:24.24 | fugitivo | NSGN: use another console |
| 03:24.34 | bubbajohn | anyone familiaer with mixnetworks? |
| 03:24.36 | NSGN | like, SSH into it? |
| 03:24.39 | Insanity4 | JunK-Y - yes |
| 03:24.46 | NSGN | is that advisable during it building asterisk? |
| 03:24.50 | fugitivo | NSGN: i don't know, just get a shell to the box |
| 03:25.33 | NSGN | well...how do i do that when eth0 is offline? :-D |
| 03:25.49 | fugitivo | NSGN: with eth1? ;) |
| 03:25.53 | NSGN | haha! |
| 03:26.10 | fugitivo | NSGN: don't you have local access to the box? |
| 03:27.05 | NSGN | yeah |
| 03:27.10 | NSGN | but its compiling |
| 03:27.17 | fugitivo | NSGN: alt+f2? |
| 03:27.25 | NSGN | ........i'm a n00b |
| 03:27.29 | NSGN | but i'll just wait a few |
| 03:27.35 | NSGN | its a frickin 233mhz box |
| 03:27.40 | MicC_ | ouch |
| 03:27.52 | NSGN | haha, hey, built it from the closet of dead computers |
| 03:27.54 | NSGN | its free |
| 03:27.56 | NSGN | ittl run me one line |
| 03:27.57 | NSGN | im happy |
| 03:28.01 | NSGN | i have 200mb of ram in it |
| 03:28.05 | NSGN | 6gb drive |
| 03:28.08 | NSGN | should do alright |
| 03:28.19 | mmlj4 | 200 millibits |
| 03:28.19 | NSGN | compiling is just taking a while.... x_X |
| 03:28.24 | Insanity4 | I'll send you a p-3 500 for $25 delivered :P |
| 03:28.24 | Insanity4 | lol |
| 03:28.29 | fugitivo | NSGN: install gentoo on that |
| 03:28.38 | Insanity4 | hehe |
| 03:28.39 | fugitivo | Insanity4: i want it |
| 03:28.43 | Sedorox | lol |
| 03:28.48 | fugitivo | Insanity4: do you deliver it to argentina? |
| 03:28.49 | NSGN | insanity4: really? ;-P |
| 03:28.53 | NSGN | i'm in the us :-P |
| 03:28.59 | Insanity4 | fugitivo - If you pay shipping |
| 03:29.08 | Insanity4 | NSGN - It's a pentium 3 something, don't knoe the speed but it can't be that bad. |
| 03:29.18 | fugitivo | Insanity4: shipping will cost like a new p4, lol |
| 03:29.28 | fugitivo | plus custom taxes |
| 03:29.29 | NSGN | heh, might take you up on that if this 233 doesnt pull through |
| 03:29.41 | NSGN | i'm hoping it will |
| 03:29.42 | Insanity4 | NSGN - If you're in cali, wa, idaho it should cost about $15 to ship. |
| 03:29.43 | NSGN | that'd be easiest |
| 03:29.50 | Insanity4 | If you're in york, well, I can't ship it for that. |
| 03:29.50 | NSGN | i'm in texas |
| 03:30.02 | NSGN | austin |
| 03:30.03 | Insanity4 | prboably cost me $20 to ship |
| 03:30.09 | Insanity4 | So I'd charge... say... 35 :P |
| 03:30.23 | NSGN | haha |
| 03:30.39 | NSGN | if i come back in here crapping about the 233 nuking itself or something, remind me of that ;-) |
| 03:30.56 | NSGN | right now the screen is black and i hear the disk chugging every few seconds |
| 03:30.56 | NSGN | haha |
| 03:31.35 | *** join/#asterisk sword (n=sword@blacksburg-bsr1-69-174-71-191.chvlva.adelphia.net) |
| 03:34.12 | NSGN | ok i'm going to sleep. ittl compile through the night :-P |
| 03:34.14 | NSGN | later |
| 03:36.03 | *** join/#asterisk zee001 (n=foobar@MTL-HSE-ppp196887.qc.sympatico.ca) |
| 03:37.23 | sword | mm |
| 03:37.24 | sword | hmm |
| 03:37.36 | sword | i want to install asterick so i can do voip in my house |
| 03:37.45 | sword | first im setting up openwrt |
| 03:37.58 | sword | has anyone did this successfully in their house and has their own documentation on it? |
| 03:38.05 | sword | ive been reading the manual |
| 03:38.20 | supaigtr | Why not ser? |
| 03:38.29 | sword | why not what? |
| 03:39.13 | supaigtr | http://sipath.sourceforge.net/ |
| 03:39.22 | azrishahril | how to install two h323 module ? is it possible to have chan_h323 (nufone) & chan_oh323 (inaccess) in one box (asterisk) ? |
| 03:39.46 | supaigtr | sword: * will install on openwrt and will work. VM and other things are issues since you don't have alot of storage space. |
| 03:39.59 | bubbajohn | anyone here recieve a voip circuit from mixnet? |
| 03:40.00 | supaigtr | sipath is much easier imo |
| 03:40.04 | sword | I can install it on an separate box supaigtr |
| 03:40.19 | sword | provide it w/ 40 gigs if need be |
| 03:40.22 | sword | hmm okay supaigtr |
| 03:40.32 | pygrammer | what is openwrt? i've heard of it, but i forget |
| 03:40.39 | sword | i just think it'd be interesting to call anywhere in the country for free :D |
| 03:40.41 | supaigtr | why not just use that box? |
| 03:40.58 | sword | pygrammer, a firmware for certain wireless routers |
| 03:41.04 | supaigtr | free??? |
| 03:41.10 | sword | supaigtr, i wanted multiple people to use it |
| 03:41.15 | sword | supaigtr, www.openwrt.org |
| 03:41.19 | supaigtr | SIP to SIP works. |
| 03:41.20 | newl | wouldn't it be easier to nfs mount some space to the router? :) |
| 03:41.49 | pygrammer | sword, it's not free... |
| 03:41.56 | pygrammer | unless these people you're talking about are using sip as well |
| 03:42.02 | pygrammer | in which case something like skype would work just as well |
| 03:42.05 | sword | my ISP is adelphia |
| 03:42.07 | sword | hmm |
| 03:42.07 | supaigtr | Right. |
| 03:42.10 | *** join/#asterisk [1]JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net) |
| 03:42.29 | newl | even then, it isn't free. Your net connection usually costs something, as does the power, etc. |
| 03:42.31 | pygrammer | pstn termination costs -- don't kid yourself :) |
| 03:42.38 | sword | newl, of course |
| 03:42.41 | sword | that isn't a problem |
| 03:42.51 | *** join/#asterisk mosty (i=mostynm@laptop-mostyn.csse.monash.edu.au) |
| 03:42.53 | pygrammer | sippath.sf.net isn't working for me |
| 03:42.59 | sword | what i meant it isnt significantly extra |
| 03:43.05 | supaigtr | termination = money nobodys giving it away :) |
| 03:43.10 | *** join/#asterisk Ahewes (n=rsb@dsl092-048-248.sfo4.dsl.speakeasy.net) |
| 03:43.22 | sword | hehe |
| 03:43.23 | pygrammer | supaigtr, plus, there are federal taxes |
| 03:43.30 | supaigtr | Yep. |
| 03:44.05 | pygrammer | damn firefox is being slow |
| 03:44.58 | bubbajohn | i just got a voip trunk delivered to me via mixnetworks, it is a test account and they send me a ip address of their server, and told me i did not need a username or password because my * box has a static ip. I created the trunk, assigned it to the outbound routing, but when i dial the number i just get dead air and eventually a all circuits are busy error, any ideas? |
| 03:45.18 | supaigtr | Has anyone seen this???? ioctl(ZT_LOADZONE) failed: Inappropriate ioctl for device |
| 03:45.18 | nick125 | manually configured? |
| 03:45.46 | Insanity4 | The ever elusive DID problem in a rural area... will there ever be a solution? |
| 03:47.37 | supaigtr | Its got my whole server screwed up. |
| 03:48.26 | nick125 | bubbajohn: i know that sometimes, if you use amp, it will do that (/me knows this from experence) |
| 03:50.02 | pygrammer | so, like, how much do yall pay for DIDs? |
| 03:50.04 | pygrammer | and how many do you hvae |
| 03:50.07 | pygrammer | *have |
| 03:51.04 | tris | I think I pay around $300/mo for 200 DIDs from XO |
| 03:51.38 | Qwell | for just the DIDs? |
| 03:51.48 | bubbajohn | $.20 ea from nuvox |
| 03:52.08 | *** join/#asterisk bmg505 (n=leon@rndf-165-85-43.telkomadsl.co.za) |
| 03:53.44 | pygrammer | tris, voip or via a |
| 03:53.47 | pygrammer | *PRI |
| 03:55.12 | *** join/#asterisk los415 (n=los415@c-24-126-63-233.hsd1.ca.comcast.net) |
| 03:55.55 | pygrammer | and bubbajohn, is that monthly? do you have to pay per minute? what about termination? |
| 03:57.16 | los415 | hey guys i have a question i have a asterisk box with a t1 card in it. on that t1 i have a channel bank hanging off of it broken down as normal zap ports. on zap/1 i have a computer with a usr 56k fax modem on zap/2 i have some telentry system. i have zap/1 call zap/2 i'm bridgeing the calls and i keep getting a NO CARRIER error on the usr modem side i have echocancelwhenbridged=no and echotraning=no set on the ports in zapata.conf |
| 04:01.51 | *** join/#asterisk santiago (n=santiago@63.245.86.163) |
| 04:03.09 | *** part/#asterisk santiago (n=santiago@63.245.86.163) |
| 04:03.12 | Ahewes | los415 I'm crappy at debugging zap, but I would plug a regular phone in place of the modem and see if you are actually getting anything. |
| 04:03.45 | Ahewes | Actually, my question would be: How do you see what is really going on with a zap channel in the CLI. |
| 04:03.53 | Ahewes | Been meaning to ask that for a while. |
| 04:04.30 | mosty | "zap show channel 1" ? |
| 04:05.24 | Insanity4 | How bad is a did from south africa to the usa? quality wise? |
| 04:06.05 | *** join/#asterisk Uberbot (n=Uberbot@pcp01880954pcs.sandia01.nm.comcast.net) |
| 04:06.07 | Ahewes | mosty : thanks. |
| 04:06.15 | los415 | well |
| 04:06.25 | los415 | i have plugged a normal phone into it |
| 04:06.26 | *** join/#asterisk ryansc (n=ryansc@c-67-165-228-253.hsd1.co.comcast.net) |
| 04:06.33 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 04:06.39 | los415 | and works fine |
| 04:06.45 | los415 | and yea i'm just looking at the call in the cli |
| 04:06.50 | los415 | i can see the call bridges |
| 04:06.53 | Uberbot | Hi all. |
| 04:07.04 | los415 | they actully look like they sync up data is sent then get a no carrier error |
| 04:08.59 | JunK-Y | ou est passé la mere noel. |
| 04:13.18 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 04:14.44 | mosty | i'm trying to figure out how to setup a TE110P card, i have the kernel module loaded, i've setup the span and channels in zaptel.conf, and corresponding settings in zapata.conf, i ran ztcfg and created an extension that answers then plays demo-congrats then hangs up, and restarted asterisk. but when i dial a number in our DID range, i get "network busy" on my (cell) phone. what could be wrong? |
| 04:16.13 | mosty | i can paste my config files on a pastebin website somewhere if anyone is willing to look through it with me |
| 04:17.49 | Ahewes | mosty: can you dial out? |
| 04:18.01 | *** join/#asterisk Eminence (n=achin@cpe-24-198-66-186.maine.res.rr.com) |
| 04:18.27 | *** join/#asterisk Maveric (n=maveric@ip68-3-248-136.ph.ph.cox.net) |
| 04:18.27 | mosty | i don't have the correct hardware with me right now to test that |
| 04:18.51 | mosty | all i have is a cell phone and a plain old telephone |
| 04:19.31 | Insanity4 | Anyone else here use nufone? |
| 04:19.42 | Ahewes | mosty: I'm not the greatest at this, but there could be a lot going on. You might want to get a softphone set up and try dialing out. |
| 04:19.56 | Ahewes | mosty: you should first, of course, make sure your T1 is synced. |
| 04:20.22 | mosty | i'll see if i can get a softphone installed. i have no mic here, but i have headphones at least |
| 04:20.51 | mosty | i'm in australia btw, so i assume we have an E1 instead of a T1. how do i check if it's synced? |
| 04:21.56 | *** join/#asterisk hellagony (n=hellagon@200.121.153.177) |
| 04:22.39 | *** join/#asterisk devonst17|bed (n=devonst1@dsl092-032-215.lax1.dsl.speakeasy.net) |
| 04:23.00 | Ahewes | mosty: there is a little green light on the back of the card that should be lit green if it's synced. |
| 04:23.17 | Ahewes | mosty: I'm reading about this stuff right now myself. |
| 04:24.37 | mosty | ahh :) another trip to the colo room then |
| 04:25.02 | Ahewes | mosty: there may be another way to figure that out, but I don't know it. |
| 04:25.08 | Ahewes | so keep reading. |
| 04:26.18 | Ahewes | right now I'm trying to debug a functioning system with a T1 and a channel bank, that just doesn't start properly |
| 04:27.11 | Ahewes | It doesn't sync all the time. It's difficult to tell when it's synced or not, and when the channel bank is in a funny state instead of the T1 card. |
| 04:27.23 | Ahewes | I'm having to log into the channel bank quite often. |
| 04:31.10 | *** join/#asterisk blessen (n=blessen@140.99.23.26) |
| 04:31.23 | blessen | hi |
| 04:31.37 | blessen | i have an issue with asterisk and kphone.. |
| 04:31.51 | mosty | ahewes: so if you remove the channel bank from the machine, the t1 works, and vice versa? |
| 04:32.25 | blessen | i cannot make calls ...using my kphone but i can register the kphone user but cannot make calls..i get the error. Call Failed : Not founf |
| 04:32.31 | supaigtr | Anyone have problems with setting debug flag on RxFAX? If I set debug it won't connect at all. |
| 04:32.36 | blessen | can anyone help with this |
| 04:32.38 | blessen | please |
| 04:32.53 | Ahewes | mosty: I don't have a T1, just a T1 card and a channel bank. |
| 04:32.54 | mosty | blessen: what do you see in the asterisk logs? |
| 04:33.11 | Ahewes | I have 8 analog lines |
| 04:33.47 | blessen | where is it |
| 04:33.49 | blessen | located |
| 04:33.59 | blessen | in the console i do not get any error... |
| 04:34.00 | Ahewes | mosty: not much in the logs |
| 04:34.26 | Qwell | blessen: turn verbosity up in the CLI |
| 04:34.58 | mosty | ahewes: i'm confused... you have a t1 card that also has a channel bank, but no t1 line? |
| 04:35.25 | Qwell | mosty: Some channel banks have T1 connectors. T1 is just an interface |
| 04:35.33 | Qwell | interface may be the wrong word... |
| 04:35.47 | Ahewes | It looks like this [PC with T1 card in it] ---crossover cable---[channel bank]---PSTN |
| 04:36.05 | supaigtr | RxFAX(${fax}||debug) seems to really screw up faxing. |
| 04:36.16 | Qwell | a channel bank is say 25 FXO or FXS ports |
| 04:36.42 | Ahewes | mosty: I need all 24 analog ports, 8 for outside lines, 16 for fxs, and then the rest of the stations are SIP |
| 04:37.40 | mosty | qwell: oh ok |
| 04:37.55 | *** join/#asterisk viLeR (i=1000@ip-47-252.telesat.com.co) |
| 04:37.57 | *** join/#asterisk NSGN (n=NSGN@cpe-66-69-197-25.austin.res.rr.com) |
| 04:38.05 | NSGN | hello. i'm back YET again |
| 04:38.05 | blessen | i did that |
| 04:38.15 | NSGN | i cant get my stupid PCI ethernet card to work in asterisk. any hints? |
| 04:38.24 | NSGN | i give it an IP but i cant connect to it remotely in any way |
| 04:38.53 | mosty | nsgn: are you running a firewall? |
| 04:39.04 | NSGN | no. im local with it |
| 04:39.16 | NSGN | i think it literally dislikes the hardware |
| 04:39.44 | blessen | qwell |
| 04:39.55 | mosty | nsgn, can you ssh into that machine remotely? |
| 04:40.00 | NSGN | nope |
| 04:40.07 | blessen | u there...i cannot register the user |
| 04:40.07 | NSGN | my router shows it is not connecting to the network |
| 04:40.09 | blessen | it shows error |
| 04:40.15 | blessen | and cannot make calls. |
| 04:40.21 | mosty | nsgn: then the problem is nothing to do with asterisk, it's an OS level problem |
| 04:40.24 | NSGN | the card lights and the lights blink, but it wont grab its own IP nor does it work when i give it a static one |
| 04:40.30 | Qwell | ~pastebin |
| 04:40.31 | jbot | it has been said that pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 04:40.34 | Qwell | blessen: put your errors there |
| 04:40.37 | NSGN | ok, makes sense. any pointers anyone can give me? |
| 04:40.46 | mosty | nsgn, depends what OS you're using i guess |
| 04:40.53 | NSGN | whatever comes with A@H |
| 04:40.56 | NSGN | culinux? |
| 04:41.28 | mosty | linux? ok then did you set the ip/netmask/default gateway with ifconfig properly? |
| 04:41.41 | mosty | er, default gateway is set with route, not ipconfig |
| 04:41.44 | NSGN | yes |
| 04:41.46 | NSGN | its all set fine |
| 04:41.56 | mosty | can you ping other machines on your lan? |
| 04:42.07 | NSGN | i used netconfig |
| 04:42.12 | NSGN | yes |
| 04:42.17 | NSGN | i'm on the internet with u on my lan |
| 04:42.19 | NSGN | and i print on the lan |
| 04:42.22 | NSGN | and many things |
| 04:42.25 | NSGN | the lan works :-P |
| 04:42.41 | mosty | yes but can the a@h machine ping other machines on your lan, is what i'm asking |
| 04:43.01 | NSGN | whoa. i just looked at the screen of the A@H machine |
| 04:43.07 | supaigtr | Anyone seen this with RxFAX? http://pastebin.ca/21200 |
| 04:43.15 | NSGN | its flooding with errors about ethernet. lemme look at em and type a few just a sec |
| 04:43.41 | mosty | use a paste site if it's a large amount of text |
| 04:43.52 | NSGN | i cant copy from it... |
| 04:43.55 | NSGN | its another computer |
| 04:44.03 | NSGN | and im not gonna type a whole screen :-P |
| 04:44.05 | NSGN | just a sec |
| 04:44.47 | *** join/#asterisk file[laptop] (n=jcolp@mctnnbsa30w-156034035250.nb.aliant.net) |
| 04:44.58 | NSGN | eth0: transmit error |
| 04:45.13 | mosty | what kind of NIC is it? (brand/model) |
| 04:45.15 | NSGN | then a whole bunch of chars trying to tell me something about it |
| 04:45.17 | blessen | check this http://pastebin.ca/21201 |
| 04:45.32 | NSGN | it is a linksys 100mbit pci card |
| 04:45.35 | NSGN | dont know exact number |
| 04:45.38 | NSGN | prolly could look at it |
| 04:46.09 | blessen | Can you tell me...what i have to do to get my kphone and my asterisk to work together..So that i can call and make calls from my system |
| 04:47.23 | newl | 1.21 GIGAWATTS!?! 1.21 gigawatts! Great Scott!!! |
| 04:47.27 | mosty | nsgn: try googling the error message :/ |
| 04:47.34 | NSGN | hmmm. ok |
| 04:47.42 | mosty | nsgn, or put another card in there if you have a spare |
| 04:48.02 | NSGN | oh wait my bad |
| 04:48.04 | NSGN | its a 3com card |
| 04:48.05 | NSGN | x_X |
| 04:48.06 | NSGN | sorry |
| 04:50.47 | *** join/#asterisk spoot_nick (n=julio@CPE-147-10-168-100.nsw.bigpond.net.au) |
| 04:51.22 | spoot_nick | how does Linksys PAP2 gets incoming calls behind nat with no redir rules, or stun, or an outbound proxy? |
| 04:52.01 | blessen | please help me....i would like to get my kphone to connect to my asterisk server and allow me to receive and make calls |
| 04:52.01 | spoot_nick | i mean, unless it's forcing one of them without me knowing that. i have all these options set to no, and the calls can still come through |
| 04:52.12 | mosty | with great difficulty ;) you need to setup asterisk to ignore the ip it gets from the SIP protocol |
| 04:52.37 | snewpy | spoot_nick: symmetrical ports for SIP and RTP, and a smart SIP proxy at the other end that works out the address inside the SIP message ain't really the address |
| 04:53.10 | snewpy | spoot_nick: which could be asterisk with nat=yes |
| 04:53.25 | spoot_nick | snewpy: the other end you mean, my provider? all they have is an asterisk box in their end |
| 04:53.29 | NSGN | mosty: the only thing i can find out about this is that its a kernel driver issue not feeding data into the PCI card fast enough? lol |
| 04:53.44 | mosty | blessen, kphone seems to segfault on me, you're better off with an ATA or physical SIP phone |
| 04:53.53 | spoot_nick | i didn't think it was enough to put nat=yes and have it all done |
| 04:54.01 | mosty | nsgn, do you have a different nic you can try? |
| 04:54.03 | Uberbot | kphone never did work for me. |
| 04:54.12 | blessen | ooooh... |
| 04:54.17 | snewpy | spoot_nick: nat=yes for the peer in sip.conf will likely make it work, along with having the registration interval lower than the NAT port allocation timeout on your gateway |
| 04:54.24 | blessen | any soft phones which i can use to get it work with asterisk |
| 04:54.24 | NSGN | mosty: right away on hand all i have is a 10megabit ISA card...would my setup suffer badly from that? x_X |
| 04:55.02 | mosty | nsgn: probably not. 10Mbit is plenty |
| 04:55.03 | spoot_nick | snewpy: hmm, wouldn't know the second one. it's linux 2.6.11 |
| 04:55.10 | spoot_nick | snewpy: do you know the default for it? |
| 04:55.11 | NSGN | mosty: for real? |
| 04:55.27 | NSGN | dang....voice really takes less than i expected. ok lemme go pop in the ISA card and hope for the best |
| 04:55.56 | Insanity4 | If you call a number and after 4 rings it says all circuits are busy should I look through my config, blame my provider, or what? |
| 04:55.56 | BhaalWK | Anyone got any ideas why asterisk might be crashing when I try and make a call? It sends the call request through (to another asterisk server via iax2) and then crashes... |
| 04:55.58 | Insanity4 | It's nufone |
| 04:56.04 | snewpy | spoot_nick: not off the top of my head, but try setting to to 60 seconds.. you'll generate more sip messages, but you won't time out |
| 04:56.05 | Insanity4 | I can call out just fine. |
| 04:56.12 | *** join/#asterisk af_ (n=af@ip-142-5.sn1.eutelia.it) |
| 04:56.17 | supaigtr | How do i get the color highlights and debug messages in CLI with commandline? |
| 04:56.19 | mosty | nsgn: remember that voip can work over 256K dsl, 10Mbit is a lot more than that |
| 04:56.27 | NSGN | kk |
| 04:57.03 | spoot_nick | probably i'll lower the registration intervals... easier |
| 04:57.22 | hmodes | brah! |
| 04:57.48 | *** part/#asterisk Uberbot (n=Uberbot@pcp01880954pcs.sandia01.nm.comcast.net) |
| 04:58.17 | Optic | moo |
| 04:58.42 | Optic | what's reasonable latency for a sip->pri->pstn voip call? |
| 04:58.44 | hmodes | so i don't suppose anyone has noticed problems with call signalling to tmobile through nufone lately? :) |
| 04:59.30 | blessen | What are the things which i will have to do to get my asterisk and kphone talk to each other in fedora 4 |
| 04:59.31 | Insanity4 | hmodes - I can't dial my nufone # |
| 04:59.36 | Insanity4 | hmodes - all cricuits are busy |
| 04:59.41 | hmodes | Optic: 150ms starts to get dicey |
| 04:59.57 | hmodes | hrmm |
| 05:00.00 | Qwell | ok, thats weird... |
| 05:00.08 | hmodes | don't have any dids, but calls complete perfectly outbound |
| 05:00.17 | mosty | blessen, looks like kphone is buggy. ask the author for help |
| 05:00.23 | hmodes | just certain numbers answering gets translated to 'ringing' |
| 05:00.24 | Insanity4 | Well they just started having other problems :P |
| 05:00.26 | Insanity4 | hehe |
| 05:00.39 | Qwell | When I call my cellphone, it'll ring my phone and the cellphone. If I don't answer the cellphone, my phone continues ringing, and thats all I hear. If I wait long enough, it sends my cell a message with whatever I say |
| 05:00.58 | Insanity4 | hmodes - Try calling 1-866-249-2403 and tell me if you can get through =) |
| 05:01.08 | Optic | hmodes: what's the best case? |
| 05:01.09 | JamesDotCom | heh, it amazes me the amount of people who always have trouble with what seem to be the bigger providers |
| 05:01.13 | Optic | and what's the best way to test? |
| 05:01.15 | JamesDotCom | is it generally a server-side issue? |
| 05:01.35 | JamesDotCom | everyone in the voip world seems mighty tolerant of downtime |
| 05:01.39 | hmodes | Insanity4: allison saying congrats ;p |
| 05:01.46 | Insanity4 | hmodes - It worked? |
| 05:01.48 | hmodes | but then i think i route 8xx to nufone |
| 05:01.53 | hmodes | so it was probably switch to switch |
| 05:02.05 | Qwell | Insanity4: You're on nufone, right? |
| 05:02.08 | Insanity4 | hmodes - Yeah... use your cell |
| 05:02.08 | Insanity4 | yes |
| 05:02.10 | hmodes | Optic: lan? |
| 05:02.12 | Insanity4 | Qwell - DID is DEAD |
| 05:02.13 | NSGN | mosty: i found a 10megabit PCI im tryin before the ISA |
| 05:02.14 | Optic | yes |
| 05:02.18 | NSGN | i have a lot of crap in my closet... |
| 05:02.19 | Qwell | Can I have you call a number for me, from your nufone account? |
| 05:02.23 | Optic | 20ms? |
| 05:02.33 | hmodes | 50ms is fine |
| 05:02.35 | Insanity4 | Qwell - I can call, but won't be able to talk (but can hear) -- nop microphone |
| 05:02.37 | hmodes | 80 is good |
| 05:02.40 | hmodes | 100 is noticeable |
| 05:02.46 | hmodes | 150 gets annoying |
| 05:02.47 | Optic | we're having more echo problems than i'd expect from our pri setup |
| 05:02.50 | hmodes | from my experience, anyway |
| 05:02.57 | Qwell | Insanity4: I wasn't going to answer anyhow. I just want to see if you can actually hear my voicemail on my cell |
| 05:02.57 | Insanity4 | Is 100 with no jitter ok? |
| 05:03.01 | Optic | mostly from remote hybrid imbalance |
| 05:03.02 | Insanity4 | ok |
| 05:03.03 | Insanity4 | #? |
| 05:03.06 | Qwell | or if it just keeps ringing |
| 05:03.10 | Insanity4 | hehe |
| 05:03.24 | Optic | not that I have much experience in that sort of thing :) |
| 05:03.27 | niZon | anyone here running xorcom rapid? |
| 05:03.37 | hmodes | yean, circuits busy from cell |
| 05:03.42 | Optic | i upgrade to HEAD tonight and went to MARK2 with AGREESIVE_SUPRESSOR |
| 05:03.43 | hmodes | that's ghey |
| 05:03.52 | Optic | it seems a bit better |
| 05:05.41 | Nugget | I cvs upped today but forgot to build. :) |
| 05:06.35 | *** join/#asterisk cp5 (n=samy@108.sub-70-219-235.myvzw.com) |
| 05:06.50 | cp5 | hola |
| 05:08.05 | blessen | is there anything i have to do to get asterisk to connect to my head phones ..so that i can use them. |
| 05:11.47 | *** join/#asterisk litage_ (n=nick@203.201.97.50) |
| 05:12.25 | twisted | wheee |
| 05:12.33 | cp5 | what's up twisted |
| 05:12.37 | twisted | not much |
| 05:12.46 | twisted | just updated and fixed patch 2471 to work on current head |
| 05:12.52 | twisted | now supplying us with full rpid support ;) |
| 05:12.59 | cp5 | nice |
| 05:13.04 | Qwell | ~ripd |
| 05:13.08 | Qwell | ~rpid |
| 05:13.20 | twisted | jbot: rpid is Remote-Party-ID |
| 05:13.21 | jbot | twisted: okay |
| 05:13.24 | twisted | lol |
| 05:13.30 | twisted | very descriptive, aren't i? |
| 05:13.58 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 05:15.09 | Qwell | Whats to describe? |
| 05:15.23 | Qwell | Its the ID of the party for remotes |
| 05:16.33 | *** join/#asterisk ugenn (n=ugenn@cm188.omega71.maxonline.com.sg) |
| 05:16.51 | ugenn | hi, anyone? |
| 05:16.56 | Qwell | noone, sorry |
| 05:17.36 | ugenn | damn |
| 05:18.41 | ugenn | is it possible to turn a regular PC and phoneline into a simple automated answering system or do i need special hardware? |
| 05:21.35 | samy | question for some of you...i have a PRI in an asterisk box. an apartment buzzer system dials into asterisk, asterisk does a SendDTMF(9) (and i've tried SIPdtmfmode() before the SendDTMF() with all three types). the buzzer should hear the dtmf 9 and open but it doesn't. if i then trunk the call out to my cell phone and hit 9 from the cell (buzzer -> * -> cell), the buzzer system hears it properly |
| 05:22.14 | samy | any ideas on why it would not be getting it from the SendDTMF()? i've even produced a gsm file with a dtmf 9 and play it with no success |
| 05:22.55 | supaigtr | Bingo. Ecan is staying on when a fax comes in. Anyone know how to disable ecan from dialplan? |
| 05:25.11 | *** part/#asterisk Ahewes (n=rsb@dsl092-048-248.sfo4.dsl.speakeasy.net) |
| 05:27.49 | Qwell | ugenn: You need something to plug the line into |
| 05:27.50 | Qwell | ~fxo |
| 05:27.51 | jbot | foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo |
| 05:28.26 | Qwell | samy: try doing a Wait(2) or something, before the senddtmf and after the answer |
| 05:29.23 | samy | qwell, i'm already doing: Answer(), WaitExten(2), SendDTMF(9) |
| 05:29.37 | samy | exactly like that |
| 05:31.35 | NSGN | none of my NICs wanna work with this |
| 05:31.43 | NSGN | i'll have to steal one from a computer that is in use right now tomorrow |
| 05:31.56 | Qwell | NSGN: almost all NICs are supported in Linux. |
| 05:32.05 | Qwell | You probably just need to load the drivers for them |
| 05:32.31 | NSGN | ok, that would be good. i'm a total linux n00b, so where would i find them for A@H's flavor of linux, and how do i install them? |
| 05:32.48 | Qwell | out of scope of this channel... |
| 05:32.49 | *** join/#asterisk astadmin (n=shafqat@pk-isb-trg-sc01-001.speedcast.com) |
| 05:32.59 | *** join/#asterisk zoo (i=nobody@ip-193-16.travedsl.de) |
| 05:33.03 | astadmin | hi everybody |
| 05:33.05 | hmodes | hah! |
| 05:33.05 | Qwell | try #centos or something |
| 05:33.12 | hmodes | that's cold |
| 05:33.33 | hmodes | but warranted |
| 05:33.34 | Qwell | hmodes: I'll help him with his issue, if you help me with my taxes. |
| 05:33.50 | Qwell | gotta draw the line somewhere, right? |
| 05:33.50 | astadmin | i have problem " chan_zap.c:3445 zt_handle_event: Ring/Off-hook in strange state 6 on channel 1" |
| 05:34.08 | hmodes | you owe the government a ton, do not pass go, do not collect shit |
| 05:34.20 | hmodes | there, your taxes are done ;p |
| 05:34.55 | *** join/#asterisk Defraz_ (n=t0tal@24-119-12-238.cpe.cableone.net) |
| 05:34.58 | astadmin | anyhelp for " chan_zap.c:3445 zt_handle_event: Ring/Off-hook in strange state 6 on channel 1" |
| 05:35.16 | Qwell | astadmin: wait more then 60 seconds between asking |
| 05:35.43 | hmodes | lest you invoke various wraths |
| 05:36.13 | hmodes | and have you bothered to look at the code indicated by the error? |
| 05:37.01 | hmodes | why do _you_ think you have a ringing channel that should be free? ;p |
| 05:37.04 | NSGN | well, can someone just point me in the right direction. qwell doesnt have to be you :-P |
| 05:37.55 | hmodes | pfft |
| 05:38.35 | hmodes | the error shows where it croaks, in great detail |
| 05:38.42 | hmodes | what more could you want? |
| 05:40.14 | Insanity4 | nufone back up! |
| 05:40.59 | Qwell | Insanity4: outgoing still funky? |
| 05:41.05 | Insanity4 | dunno |
| 05:41.05 | ugenn | Qwell: i have a regular modem? Does that count as an FXO? |
| 05:41.09 | Qwell | ugenn: no |
| 05:41.26 | Qwell | yep, outgoing still funky |
| 05:41.38 | Qwell | *ring*ring*ring*"y..*ring*ring*ring |
| 05:41.43 | Insanity4 | lol |
| 05:41.53 | hmodes | has jerjer been around to bitch at recently? |
| 05:42.04 | Qwell | dunno, he needs to come around though |
| 05:42.20 | hmodes | yeah... something is very obviously wrong. |
| 05:42.36 | Qwell | the issue I'm seeing has been happening since at least 2 nights ago |
| 05:42.53 | hmodes | it appears to have been weeks for me |
| 05:42.55 | BhaalWK | Argh... Im getting permission denied when asterisk tries to open the iax timer |
| 05:43.00 | hmodes | people just now started complaining :) |
| 05:43.05 | BhaalWK | Anyone know what needs permissions changed? |
| 05:43.20 | Qwell | hmodes: when a nufone account calls my cell, it'll ring over the voicemail, and I can't hear anything but ringing. it WILL leave a message though |
| 05:43.26 | Insanity4 | I can't figure out why I have to dial my stanaphone number lik 5 times before saterisk picks the damn thing up |
| 05:43.48 | hmodes | yeah, i don't even get messages |
| 05:43.56 | hmodes | just endless ringing if i pick up the cell |
| 05:44.01 | Qwell | hmodes: a message on my cell I mena |
| 05:44.08 | hmodes | it appears when it goes to voicemail nether end picks up |
| 05:44.10 | Qwell | it'll keep ringing even if you answer? |
| 05:44.12 | hmodes | and it just rings forever |
| 05:44.19 | hmodes | yeah |
| 05:44.20 | Qwell | see, if I answer, it works fine |
| 05:44.26 | Qwell | what provider? |
| 05:44.42 | hmodes | nufone/tmobile 732688 |
| 05:44.49 | Qwell | odd |
| 05:44.50 | hmodes | ring ring ring ring ring ad nauseum |
| 05:44.51 | ugenn | Qwell: i'm lost. according http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html/x522.html, "a modem is a classic example of an FXO card". |
| 05:44.52 | Qwell | nufone/sprint pcs |
| 05:44.52 | hmodes | in all cases |
| 05:45.28 | Qwell | ugenn: a modem isn't an fxo |
| 05:45.39 | hmodes | i'm using iax.cc for forwarding presently |
| 05:45.55 | hmodes | their cid restrictions are beat tho' |
| 05:46.03 | hmodes | i really want to get nufone fixed |
| 05:46.04 | Insanity4 | The one prblem with a coneference server... someone pushes hold with MOH enabled. god. |
| 05:46.43 | Insanity4 | There should be a "detect music or excessive bitching" script which automatically mutes that channel... lol |
| 05:46.54 | hmodes | moderator good ;p |
| 05:47.03 | Insanity4 | hehe |
| 05:47.13 | Insanity4 | How can the moderator tell whos on hold though? |
| 05:47.14 | hmodes | with appropriate manager interface integration the moderator can drop people ;p |
| 05:47.34 | *** join/#asterisk r0d3nt (i=r0d3nt@tinfoilhat.net) |
| 05:47.55 | supaigtr | Is there any reason the new kb1 echo can wound't work with faxdetect. It doesn't seem to be turning off anymore on a fax call. |
| 05:47.58 | Insanity4 | Ok, I swear gsm is backhauling those 800#'s over gsm or something. ulaw to stanaphone is clearly better quaity. |
| 05:47.59 | hmodes | tho knowing who is sending moh to a conference would be amazingly tricky |
| 05:48.01 | hmodes | it's possible |
| 05:48.02 | drray | I'm playing with zapbarge to create something where only the speaker can speak |
| 05:48.25 | Insanity4 | drray - You can moderate, but finding the MOH bastard is hard. |
| 05:48.33 | drray | zapbarge is muted |
| 05:49.36 | Insanity4 | Is there a way to show all concurent calls and codecs in use? |
| 05:49.50 | hmodes | concurrent calls is easy |
| 05:49.58 | hmodes | transmitting calls, not so much |
| 05:50.24 | hmodes | but still, the dsp interface is there |
| 05:50.48 | hmodes | a simple counter could axe excessive transmitters |
| 05:51.56 | Insanity4 | hmodes - Talking about conference? Yes, I suppose. either moh folks or bitching/moaning folk hehe. |
| 05:52.08 | hmodes | if !moderator and transmitcount > $x.... fuckem! |
| 05:52.17 | hmodes | :) |
| 05:52.22 | Insanity4 | Yup |
| 05:52.31 | Insanity4 | To kick anyone hogging more than 40% of the dead air |
| 05:52.32 | Insanity4 | lol |
| 05:52.40 | Insanity4 | What's the largest conference line you've had going? |
| 05:52.41 | Insanity4 | hehe |
| 05:52.59 | Insanity4 | uh oh |
| 05:52.59 | drray | I think it would be great if there was a token and you could pass speaking rights to speakers |
| 05:53.01 | *** join/#asterisk ptiggerdine (n=ptiggerd@c210-49-98-194.rochd1.qld.optusnet.com.au) [NETSPLIT VICTIM] |
| 05:53.01 | Insanity4 | freenode going down for upgrades |
| 05:53.02 | Insanity4 | hehe |
| 05:53.03 | *** join/#asterisk CdtDelta (n=CdtDelta@dsl081-225-161.chi1.dsl.speakeasy.net) [NETSPLIT VICTIM] |
| 05:53.07 | *** join/#asterisk The_Ball (n=alex@static-227.35.240.220.dsl.comindico.com.au) [NETSPLIT VICTIM] |
| 05:53.08 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) [NETSPLIT VICTIM] |
| 05:53.14 | Nugget | "upgrades" indeed. |
| 05:53.16 | *** join/#asterisk smurfix (n=smurf@run.smurf.noris.de) [NETSPLIT VICTIM] |
| 05:53.18 | Insanity4 | drray - We did that shit in grammer school. The last one talking would pass it off. |
| 05:53.25 | Insanity4 | And only that person could talk |
| 05:53.30 | Insanity4 | how agrivating it was, hehe. |
| 05:53.32 | *** join/#asterisk ptblank (n=MURDER1@ca-yorbalnd-cuda2-c1a-124.anhmca.adelphia.net) [NETSPLIT VICTIM] |
| 05:53.35 | *** join/#asterisk newl (n=newlook@203-59-187-240.dyn.iinet.net.au) [NETSPLIT VICTIM] |
| 05:53.37 | drray | yes, it's called token ring |
| 05:53.44 | *** join/#asterisk mithro (n=tim@c213-100-42-188.swipnet.se) |
| 05:53.44 | drray | as opposed to collision avoidance |
| 05:53.46 | *** join/#asterisk mogorman (n=mogorman@digium.com) [NETSPLIT VICTIM] |
| 05:53.46 | *** join/#asterisk crash3m (i=crash3m@unaffiliated/crash3m) [NETSPLIT VICTIM] |
| 05:53.47 | cochi | token-ring-over-voip? cooooool |
| 05:53.54 | *** join/#asterisk juice (n=juice@mo-67-77-177-215.dyn.sprint-hsd.net) [NETSPLIT VICTIM] |
| 05:53.54 | cochi | ;) |
| 05:53.56 | Insanity4 | hehe |
| 05:53.57 | *** join/#asterisk nvrs (i=RUR@Toronto-HSE-ppp3867418.sympatico.ca) |
| 05:54.10 | drray | I'm thinking of setting up a zapbarge for people who just have listen rights |
| 05:54.15 | hmodes | <null> |
| 05:54.25 | drray | and the speakers will be able to be qued in |
| 05:54.30 | cochi | so who's talking now, huh ? *burps* |
| 05:54.44 | hmodes | i don't respond to consumed tokens, the protocol says so! |
| 05:54.47 | cochi | also called "round robin conference" ;) |
| 05:55.17 | *** join/#asterisk ugenn (n=ugenn@cm188.omega71.maxonline.com.sg) [NETSPLIT VICTIM] |
| 05:55.18 | cochi | oh or shortest remaining next... "yo" "hi" "what the" "just short" *all leave* "hello, somebody there? i wanna talk. DAMN" |
| 05:55.19 | Insanity4 | Ok |
| 05:55.28 | drray | my favorite was when someone put the phone down to tell someone that he was forced to sit through and listen to these idiots talk |
| 05:55.32 | Insanity4 | now guys -- how come I have to call myself like 3 times to get it to pick up. |
| 05:55.56 | Insanity4 | hehe |
| 05:56.05 | Insanity4 | sip with stanaphone takes a few tries to get a pickup. |
| 05:56.12 | Insanity4 | nufone = 1 in 5 don't connect. |
| 05:56.16 | harryvv | token ring over voip? |
| 05:56.29 | *** join/#asterisk Uberbot (n=Uberbot@pcp01880954pcs.sandia01.nm.comcast.net) |
| 05:56.35 | hmodes | that's one heck of a broken srv ;p |
| 05:56.37 | Uberbot | Hi all. |
| 05:56.47 | Insanity4 | I dont know why |
| 05:56.56 | harryvv | so i guess this 911 voip debacle is comming to a head on tuesday |
| 05:56.59 | Insanity4 | when it doesnt connect its like the providers aren't even trying to contact my server |
| 05:57.33 | drray | Harryvv.. I've set it up here that when an extension makes a 911 call, the operator panel is notified.. |
| 05:58.03 | harryvv | operator panel? |
| 05:58.04 | *** join/#asterisk grimse (n=grimse@p5481D7CC.dip.t-dialin.net) |
| 05:58.16 | harryvv | I hold a FCC licence and its nessesary. |
| 05:58.18 | Insanity4 | head on teusdaY? operator panel? |
| 05:58.20 | drray | I talked with someone at the seattle police department and tehy said that would be fine for them.. they are sending a letter |
| 05:59.00 | harryvv | Insanity, thats when most voip local/long distance carriers are required to shut off there customers calls if thay dont have 911 service. |
| 05:59.15 | Insanity4 | harryvv - Can a customer "opt out"? |
| 05:59.17 | harryvv | drray, are you in seattle? |
| 05:59.20 | drray | yes |
| 05:59.22 | harryvv | Insanity dont know |
| 05:59.36 | harryvv | drray, lived in washington most of my life and even seattle. |
| 05:59.39 | Insanity4 | harryvv - How does that effect terminators like nufone? Are htey classified as somethign else? |
| 05:59.56 | drray | harryvv - had tog ive it up? |
| 06:02.37 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 06:02.37 | *** topic/#asterisk is Preview the new website! http://beta.asterisk.org || Asterisk: The Open Source PBX || Asterisk 1.2 Janitors Needed! http://dev.asteriskdocs.org || Astricon 2005 in Anaheim Oct 12-14 http://www.astricon.net/2005/ - speakers wanted |
| 06:02.40 | twisted | the net didn't split, the servers all bounce3d |
| 06:02.47 | *** join/#asterisk _[MUPPETS]Gonzo (n=gonzo@80.69.47.16) |
| 06:02.51 | NSGN | so nobody knows where i can get some drivers for my NICs? |
| 06:02.55 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 06:02.59 | twisted | NSGN, google |
| 06:03.02 | *** part/#asterisk twisted (n=twisted@24.96.140.230) |
| 06:03.02 | *** join/#asterisk twisted (n=twisted@24.96.140.230) |
| 06:03.05 | twisted | hrm. |
| 06:03.08 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 06:03.13 | Qwell | twisted: eveninoon |
| 06:03.15 | twisted | *serv doesn't appear to be up yet |
| 06:03.21 | NSGN | ok, i'm a real linux noob. what would i search for? do they have to be specific to the flavor of linux? |
| 06:03.21 | cochi | GDrivers? ;))) google should search for software indeed |
| 06:03.23 | Qwell | twisted: he authed me |
| 06:03.29 | *** join/#asterisk mCherebru (i=jgarland@72.36.136.226) |
| 06:03.32 | *** join/#asterisk shido6 (n=shido6@d57-87-253.home.cgocable.net) |
| 06:03.34 | Qwell | NSGN: try #centos... |
| 06:03.36 | *** join/#asterisk sfarquhar (n=sfarquha@b128D.static.pacific.net.au) |
| 06:03.37 | *** join/#asterisk clint_ (n=clint@snap.helixsystems.com) |
| 06:03.43 | *** mode/#asterisk [+o twisted] by ChanServ |
| 06:03.45 | twisted | hah. |
| 06:03.52 | *** join/#asterisk Wi_Fi (n=OUT@c-24-127-12-85.hsd1.ca.comcast.net) |
| 06:04.01 | NSGN | qwell: i did. they said my NIC is broken.... |
| 06:04.03 | NSGN | wrong |
| 06:04.06 | NSGN | works in a winxp machine |
| 06:04.10 | Corydon76-home | Okay, so 911 is emergency service, and if a provider can't provide good emergency service, they're going to prevent you from calling during an emergency... |
| 06:04.12 | Qwell | well, when thousands of people try to msg nickserv at once... |
| 06:04.26 | *** join/#asterisk Rez (i=lorez@freenode/staff/lorez) |
| 06:04.30 | *** join/#asterisk twisted[work] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 06:04.31 | *** mode/#asterisk [+o twisted[work]] by ChanServ |
| 06:04.36 | cochi | 911 doesnt help with nomadic anyway does it |
| 06:04.43 | *** join/#asterisk bonez41 (n=aint@c-67-166-77-14.hsd1.ut.comcast.net) |
| 06:04.45 | Corydon76-home | Anybody else see the minor flaw in that logic? |
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| 06:05.22 | *** join/#asterisk J[SS] (i=ph33r@smartserv.ipv6.smart-serv.net) |
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| 06:05.29 | *** join/#asterisk Wonka (i=produzie@chaos.in-kiel.de) |
| 06:05.37 | *** join/#asterisk psycodad (n=obiwan@2001:4060:4419:b1:0:0:0:2) |
| 06:05.41 | *** join/#asterisk shift (i=shift@82-39-34-36.cable.ubr01.benw.blueyonder.co.uk) |
| 06:05.49 | *** join/#asterisk skrusty (i=muad@xdev.net) |
| 06:05.50 | *** join/#asterisk festr_ (n=festr@ns.regnet.cz) |
| 06:05.53 | *** join/#asterisk jontow (i=jontow@ws.woflsys.net) |
| 06:05.56 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 06:05.58 | *** join/#asterisk valence (n=valence@Quebec-HSE-ppp230300.qc.sympatico.ca) |
| 06:05.59 | *** join/#asterisk drumkilla (n=russell@12.21.243.167) |
| 06:06.01 | *** join/#asterisk bendy24 (n=slb@50.tender.1meg.golden.net) |
| 06:06.02 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 06:06.02 | twisted | Corydon76-home, the fcc doesn't make flaws. |
| 06:06.05 | *** join/#asterisk sivana (n=sivana@mixdown.ca) |
| 06:06.09 | twisted | and on that note, i'm out. |
| 06:06.17 | Corydon76-home | G'night, twisted |
| 06:06.22 | *** join/#asterisk Pj (n=pj@fernande.happycoders.org) |
| 06:06.22 | *** join/#asterisk kaldemar (n=kalde@vipunen.hut.fi) |
| 06:06.26 | *** join/#asterisk kajtzu (n=kajtzu@shell1.fi.basen.net) |
| 06:06.26 | *** join/#asterisk pjz (n=pj@place.org) |
| 06:06.34 | *** join/#asterisk Syncros (n=sysop@noc.routermonkey.net) |
| 06:06.48 | *** join/#asterisk rajo (n=rajo@bfs.cs.uni-sb.de) |
| 06:06.57 | *** join/#asterisk switch (n=switch@61.206.115.5.user.ad.il24.net) |
| 06:07.25 | *** join/#asterisk denon23532 (i=denon@synapse.subneural.net) |
| 06:07.46 | *** join/#asterisk bmg505 (n=leon@rndf-146-25-170.telkomadsl.co.za) |
| 06:09.14 | *** join/#asterisk criptos (n=criptos@201.145.229.183) |
| 06:09.27 | *** join/#asterisk Uberbot (n=Uberbot@pcp01880954pcs.sandia01.nm.comcast.net) |
| 06:09.33 | Uberbot | Hi agian. |
| 06:09.44 | criptos | zaptel kernel modules manage the ringing and the hangup of the channel rigth? |
| 06:09.51 | criptos | hi again |
| 06:10.10 | Uberbot | Does anyone know what this error message means? |
| 06:10.14 | Uberbot | <PROTECTED> |
| 06:10.20 | Uberbot | From the * coonsle. |
| 06:10.21 | *** join/#asterisk cj (n=cjcollie@216.39.139.201) |
| 06:10.23 | *** join/#asterisk faa_ (i=faceoff@devel.acdbddh.eu.org) |
| 06:11.14 | *** join/#asterisk Sgt_Stedenko (n=piespy@c-67-185-117-105.hsd1.wa.comcast.net) |
| 06:11.55 | criptos | sip or iax? Mostly, a networked channel is trying to send a frame, and the "client" is unable to acknowledge the frame... |
| 06:12.11 | Uberbot | Sip. It's a pingtel. |
| 06:12.14 | cochi | happened to me in a box with two nics last time. preceeded by "call we dont know" |
| 06:13.08 | criptos | I belive or they will be networking issues about latency, or nat, use sip debug and check the ip of the incoming or outgoing stream |
| 06:13.20 | Uberbot | sip debug ip 10.0.1.60 is giving me more info... |
| 06:13.24 | Uberbot | SIP/2.0 403 Forbidden |
| 06:13.39 | Uberbot | That can't be good. :-D |
| 06:14.26 | Uberbot | Is this a context issue? |
| 06:15.11 | criptos | maybe is an autentification issue :) |
| 06:15.30 | Uberbot | The pingtel doesn't seem to have a password config option. |
| 06:15.40 | Uberbot | So I auth by IPaddress.....? |
| 06:15.40 | gordonjcp | it must do |
| 06:15.48 | gordonjcp | it's kind of required |
| 06:16.34 | *** join/#asterisk anonobomber (n=anonobom@c-67-170-91-29.hsd1.wa.comcast.net) |
| 06:16.34 | Uberbot | The device shows up in sip show peers.... |
| 06:16.37 | harryvv | k |
| 06:16.45 | harryvv | im so beat |
| 06:16.51 | Uberbot | I can call it. |
| 06:16.58 | Uberbot | Tough day? |
| 06:18.56 | *** join/#asterisk WilliamK (n=wkeller@c-67-172-202-228.hsd1.tx.comcast.net) |
| 06:21.02 | drray | uberbot - I have a similar problem, I can call an extension when I dial from outside, or from a sip/iaxy phone.. but none of my internal zap phones can call my sip phone |
| 06:21.35 | drray | no nat, and all in teh same context |
| 06:23.02 | *** join/#asterisk insomni_ (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 06:24.02 | *** join/#asterisk Ahrimanes (n=aron@hobbes.bsd-dk.dk) |
| 06:24.17 | harryvv | well this is fustrating. my email to list.digium.com did not go though. Anyone know a way to make asterisk start dial plan listen to belcore ring signal and then divert it to a extention? outside audio control panel sends these signals when somone want to ring our phone with that uniqe ring sequence but in my case, thay hear my ivr and then press the extention killing the panel. that is for safety so the passcode to enter the building i |
| 06:32.06 | *** join/#asterisk Tili (i=Tili@219.136.218.164) |
| 06:35.21 | *** join/#asterisk Gronker__ (n=Gronker2@70.152.166.254) |
| 06:37.52 | *** join/#asterisk Saaib (n=nabudoco@ns1.ensenada.gob.mx) |
| 06:40.03 | *** join/#asterisk PakiPenguin (i=uppal@unaffiliated/pakipenguin) |
| 06:43.56 | *** join/#asterisk shido6 (n=shido6@d57-87-253.home.cgocable.net) |
| 06:44.31 | Qwell | shido6: That really you? |
| 06:44.35 | shido6 | ZzzZz |
| 06:44.49 | shido6 | whats up? |
| 06:45.07 | Qwell | nufone to my cell is b0rked |
| 06:45.15 | shido6 | whats the # |
| 06:45.35 | Qwell | shido6: let it ring at least 8 times. if voicemail answers, its better |
| 06:45.49 | Qwell | voicemail should pickup in like 5 rings, but...its weird |
| 06:46.32 | Qwell | should have just kicked to vm |
| 06:46.38 | Qwell | but, its probably still ringing |
| 06:47.03 | shido6 | we dont do that number |
| 06:47.06 | shido6 | we have michigan and 8xx |
| 06:49.34 | *** join/#asterisk jayk- (i=jayk@vapid.reprehensible.net) |
| 06:51.01 | pygrammer | anyone here have the motorola e815? |
| 06:51.12 | jayk- | i'm trying to set up call parking. i defined extension 600 in features.conf, but it doesn't seem to work. |
| 06:51.33 | jayk- | asterisk says that it loaded the parking module and registered extension 600..anybody have any ideas? |
| 06:51.52 | *** join/#asterisk websae (n=websae@207-118-143-134.dyn.centurytel.net) |
| 06:52.07 | websae | anyone running freebsd here? |
| 06:52.58 | jayk- | i am |
| 06:53.05 | websae | mind if i msg you for a sec? |
| 06:54.39 | websae | looking for some help with asterisk install on FreeBSD, just having a ton of problems... |
| 06:58.21 | criptos | what is the use of CONFIG_CAC_GROUNDSTART at zconfig.h from zaptel ? |
| 06:59.06 | websae | criptos do you use FreeBSD? |
| 07:00.17 | criptos | Nope, I use mostly linux, but what is your question? I used to be a seasoned 2.0/3.0 freebsd user.. |
| 07:00.53 | websae | can't get it to install |
| 07:03.52 | blitzrage | yo yo |
| 07:04.16 | criptos | from source or from ports? |
| 07:04.16 | supaigtr | Anyone have any ideas on killing echo can when a fax comes in? |
| 07:04.59 | websae | from ports |
| 07:06.10 | criptos | newly sourceode or good know when sourcecode? |
| 07:06.46 | litage | since you can authenticate with ldap, why would you want to use radius to authenticate, along with ldap? |
| 07:10.26 | blitzrage | supaigtr: in zconfig.h, uncomment /* #define NO_ECHOCAN_DISABLE */ |
| 07:10.28 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 07:10.38 | websae | Anyone successfully using FreeBSD and asterisk? |
| 07:11.21 | *** part/#asterisk criptos (n=criptos@201.145.229.183) |
| 07:12.10 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 07:12.20 | supaigtr | blitzrage: I only want to disable on a fax call. It would seem faxdetect doesn't always works. Besides it would be nice to just say echo=disable and rxfax the call. |
| 07:14.20 | shido6 | anyone with a mac |
| 07:14.25 | shido6 | using something other than toast to burn |
| 07:14.26 | shido6 | ? |
| 07:15.00 | dersteer | you running the new os X ? |
| 07:18.12 | websae | really looking for some FreeBSD and asterisk setup help...running into a few issues |
| 07:19.34 | *** join/#asterisk DannyF (n=dannyf@c-6d4fe353.24-0099-74657210.cust.bredbandsbolaget.se) |
| 07:21.27 | *** join/#asterisk darkskiez (n=darkskie@host86-133-147-187.range86-133.btcentralplus.com) |
| 07:22.23 | *** join/#asterisk zigman (n=zigman@irc.zigman.de) |
| 07:23.51 | *** join/#asterisk wzl (n=Miranda@82-169-62-42-mx.xdsl.tiscali.nl) |
| 07:23.57 | wzl | good mornin |
| 07:23.59 | *** join/#asterisk websae2k (n=websae@207-118-143-134.dyn.centurytel.net) |
| 07:24.02 | *** join/#asterisk FITA1 (n=m_ahmed@202.5.145.50) |
| 07:24.25 | websae2k | any FreeBSD users running asterisk in here? looking for some help please!? |
| 07:24.39 | FITA1 | hi all |
| 07:24.50 | websae2k | FITA1: do you use FreeBSD? |
| 07:25.12 | *** join/#asterisk Uther_P (n=uther_p@66.180.120.82) |
| 07:25.48 | *** join/#asterisk rvhi (n=rv@66.175.65.89) |
| 07:25.49 | *** join/#asterisk Newbie___ (n=me@espeed24-74.brunet.bn) |
| 07:26.52 | Newbie___ | hi all, asterisk is great, but where can i find billing software that work with asterisk |
| 07:27.26 | Qwell | ~google asterisk billing site:voip-info.org |
| 07:27.37 | *** join/#asterisk colombus (n=colombus@193.190.210.151) |
| 07:27.55 | Qwell | well, bot doesn't like googling it seems, but you get the picture |
| 07:28.06 | Newbie___ | Qwell: i did that |
| 07:28.09 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 07:28.20 | jayk- | is anybody here using valet parking or super valet parking? |
| 07:28.20 | Qwell | and? |
| 07:29.37 | Newbie___ | so many of them, need advice on which is commonly use |
| 07:29.44 | FITA1 | we are using asterisk-1.0.7 and have 2 fxo cards on the system. The problem is we cannot use both cards for incoming calls at a time. |
| 07:30.28 | *** join/#asterisk BharatS (n=bharatsa@210.211.246.47) |
| 07:30.35 | BharatS | hi there |
| 07:30.59 | BharatS | which is the latest version of Asterisk availble? |
| 07:31.06 | BharatS | can anybody please tell me |
| 07:31.08 | Qwell | BharatS: cvs head |
| 07:31.15 | Qwell | or do you mean stable? |
| 07:31.16 | *** join/#asterisk pigpigpig (n=pig@203.125.63.150) |
| 07:31.20 | BharatS | yes |
| 07:31.23 | Qwell | 1.0.9 |
| 07:31.37 | BharatS | the one which is used at the production level |
| 07:31.52 | Qwell | alot of people use cvs head in production |
| 07:31.58 | BharatS | ok |
| 07:32.02 | Qwell | whether you should or not, is your choice |
| 07:32.10 | BharatS | oh |
| 07:32.14 | *** part/#asterisk pigpigpig (n=pig@203.125.63.150) |
| 07:32.14 | Qwell | depends on if you need some of the features in head or not |
| 07:32.39 | websae2k | FreeBSD users anyone? |
| 07:32.50 | BharatS | ok where will i get the details about the structure of the latest version of the Asterisk |
| 07:32.54 | Qwell | websae: What is your problem? |
| 07:32.55 | BharatS | that is the 1.0.9 |
| 07:33.00 | drray | there are times when head is broken, but usually head is as good or better than stable |
| 07:33.03 | Qwell | BharatS: asterisk.org? |
| 07:33.06 | websae2k | not installing |
| 07:33.14 | Qwell | websae2k: doesn't help |
| 07:33.19 | BharatS | alright <websae2k>? |
| 07:33.21 | BharatS | thanks |
| 07:33.24 | *** join/#asterisk evilteddyrux (n=ted@pcp01539543pcs.huntsv01.al.comcast.net) |
| 07:33.41 | jayk- | anybody here use valet parking or super valet parking? |
| 07:33.46 | FITA1 | each card is working when we attach only one telephone line to one of them, but when we want to connect to telephone lines the call on the second chennel is not answered by asterisk |
| 07:33.55 | *** join/#asterisk pr0 (n=pr0@ndn-165-144-110.telkomadsl.co.za) |
| 07:33.58 | pr0 | lo all |
| 07:34.39 | websae2k | it won't compile...errors |
| 07:35.03 | *** join/#asterisk mamcinty (n=mamcinty@adsl-068-209-174-113.sip.int.bellsouth.net) |
| 07:35.28 | drray | websae - why not get it working under linux first, BSD is difficult to get asterisk working under |
| 07:35.32 | BharatS | stability of an Asterisk server is judged on what basis? |
| 07:35.52 | BharatS | I mean what factors are considered |
| 07:35.59 | pr0 | uptime/throughput |
| 07:36.07 | BharatS | ok |
| 07:36.15 | BharatS | anything else |
| 07:36.18 | drray | I think stable means the code does not change day by day |
| 07:36.24 | *** join/#asterisk W|NGNUT (n=wingnut-@207.80-203-25.nextgentel.com) |
| 07:36.45 | drray | there have ben times when cvs head has been broken, only to be fixed a few minutes/hours later |
| 07:36.57 | drray | been |
| 07:37.00 | pr0 | so if a server stays on for 5 months and serves 5 milion minutes a day, thats the kind of stabilaty were getting |
| 07:37.20 | drray | once you get asterisk working with head, it tends to keep working |
| 07:37.33 | websae2k | what is head? |
| 07:37.38 | pr0 | true true |
| 07:37.39 | drray | I'm running head from 4 months ago in a production system |
| 07:39.21 | drray | head is the bleeding edge current development version of asterisk |
| 07:40.12 | *** join/#asterisk RoyK (n=roy@80.239.107.80) |
| 07:40.32 | *** join/#asterisk Doofmannsgehilfe (n=chatzill@dsl-084-060-061-212.arcor-ip.net) |
| 07:41.06 | *** join/#asterisk oden (n=oden@194-237-146-22.customer.telia.com) |
| 07:42.17 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 07:43.17 | *** join/#asterisk Kumbang (n=unknown@167.205.24.5) |
| 07:45.19 | RoyK | ~seen zoa |
| 07:45.29 | jbot | zoa <n=k@213.91.216.136> was last seen on IRC in channel #asterisk, 7h 43m 2s ago, saying: 'the real thing is pink and says "asstricks inside"'. |
| 07:45.29 | RoyK | ~seen zoa2 |
| 07:45.30 | jbot | zoa2 <n=kkk@83.228.8.96> was last seen on IRC in channel #asterisk, 11h 57m 52s ago, saying: 'yeah head'. |
| 07:45.30 | pr0 | anyone has experience with the swissvoice phones? |
| 07:46.55 | *** join/#asterisk meppl (n=mephisto@p54AAEF6B.dip.t-dialin.net) |
| 07:47.14 | pr0 | stupid question... but how would I go about making a bootable cd/dvd with the slackware cd's boot loader and my own files (especially bare.i) |
| 07:47.29 | pr0 | I need to make a 2.6.12 bare.i |
| 07:47.42 | pr0 | er wrong chan |
| 07:48.16 | *** join/#asterisk OgunWork (n=johangri@h236n2fls34o865.telia.com) |
| 07:48.18 | Doofmannsgehilfe | which the best isdn-cards ? |
| 07:48.43 | websae2k | how do i tell what version asterisk i am running? |
| 07:49.16 | kajtzu | sh ver |
| 07:49.30 | Doofmannsgehilfe | I'm will start and must buy new cards |
| 07:49.52 | opus__ | hi |
| 07:50.28 | FITA1 | we are using asterisk-1.0.7 and have 2 fxo cards on the system. The problem is we cannot use both cards for incoming calls at a time. |
| 07:50.48 | FITA1 | <PROTECTED> |
| 07:51.01 | opus__ | read the manual for /etc/zap*.conf |
| 07:51.20 | *** join/#asterisk MmmmToop (n=chatzill@c1-50-2.rndf.isadsl.co.za) |
| 07:52.57 | OgunWork | Doofmannsgehilfe: PRI or BRI? |
| 07:53.08 | *** part/#asterisk oden (n=oden@194-237-146-22.customer.telia.com) |
| 07:53.15 | FITA1 | opus__: etc/zaptel.conf |
| 07:53.29 | Doofmannsgehilfe | BRi |
| 07:53.31 | Doofmannsgehilfe | the popular card is AVM but the card dont work in NT-Modus |
| 07:53.42 | OgunWork | Doofmannsgehilfe: HFC-S cards. |
| 07:54.08 | OgunWork | Doofmannsgehilfe: They work in both NT and TE. |
| 07:58.38 | rabelais | what does a "user license" for a cisco phone do? |
| 07:59.21 | drray | to get my cisco 7960 working, I had to buy a service contract, which allowed me to download the sip firmware from the webpage |
| 07:59.33 | Doofmannsgehilfe | i would buy Billion Bipac PCI V3.0 is for an TEST not for |
| 07:59.48 | rabelais | you're kidding me... |
| 07:59.54 | drray | no |
| 08:00.00 | drray | and it was a pain in the arse |
| 08:00.01 | rabelais | so it's not enough just to by the phone, you also have to bye the firmware for it? |
| 08:00.06 | rabelais | *buy |
| 08:00.15 | drray | cisco is not interested in selling joe schmoe a phone |
| 08:00.23 | rabelais | apparently not |
| 08:00.27 | drray | BUT |
| 08:00.41 | drray | the cisco 7960 is a really7 nice sip phone |
| 08:00.50 | *** part/#asterisk xylome (n=asterisk@hg-msq-hol.levigo.de) |
| 08:01.15 | rabelais | ya...perhaps, but I'm not about to shell out over $500 for a bloody phone |
| 08:01.29 | drray | we use it for an operator console |
| 08:01.33 | drray | to replace a mitel one |
| 08:01.45 | drray | otherwise buy a budgetone or otehr chaper sip phone |
| 08:01.50 | rabelais | ya...my phone usage is limited to...just me |
| 08:01.52 | drray | ,er cheaper |
| 08:02.02 | rabelais | I think I'll just wait until prices drop quite some |
| 08:02.07 | drray | or get an iaxy |
| 08:02.16 | rabelais | I have a sipura2000 |
| 08:02.19 | rabelais | it does the job |
| 08:02.23 | drray | yeah |
| 08:02.31 | rabelais | hmm |
| 08:02.48 | rabelais | so is 285 per phone+license a good deal? for a 7960? |
| 08:02.49 | drray | I mean, the cisco is great for transfering and putting calls on hold and teh xml menuing stuff works for us |
| 08:03.14 | drray | I don't know if that gets you access to the web site or not |
| 08:03.42 | websae2k | anyone here use asterlink@home? |
| 08:04.04 | drray | we paid $400 for our first 7960/power cube/ smartnet contract |
| 08:04.30 | rabelais | smartnet? is that the website access that you're talking about? where you get the firmware? |
| 08:04.37 | drray | yes |
| 08:05.03 | drray | or you could buy a bootleg cd off of ebay if that does not bother you |
| 08:05.24 | rabelais | well, of course...getting access to the software is never an issue |
| 08:05.29 | rabelais | but if I wanted to be _legal_ |
| 08:05.42 | drray | well, the sip image is not on a p2p network |
| 08:05.55 | drray | or the one I found was not |
| 08:06.26 | *** join/#asterisk harlequin516 (n=sham@dsl01-ppp-4444.fastq.com) |
| 08:06.34 | drray | but they are nice nice phones |
| 08:07.00 | rabelais | I believe you |
| 08:07.09 | drray | and the word cisco on the console made selling asterisk to bossman that much more palletable |
| 08:07.15 | drray | for him |
| 08:07.19 | *** join/#asterisk shadebob (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma) |
| 08:07.21 | shadebob | hi |
| 08:07.22 | drray | "oh, cisco does networking" |
| 08:07.25 | harlequin516 | OKay I have compiled asterisk, but I'm afraid to install on my perfectly rpm based SUSE system. Can I do this neatly in a directory like /opt? |
| 08:07.29 | drray | yes, corky that's right |
| 08:07.32 | Doofmannsgehilfe | a very god ip-phone is the snop look at snom.com |
| 08:07.55 | BharatS | Any links for understand the LCR in Asterisk... |
| 08:08.05 | drray | the corky coment was in reply to my comment |
| 08:08.08 | rabelais | harlequin516: yes, of course, but you'll have to recompile and use the prefix= directive in your ./configure |
| 08:08.09 | drray | not anyone elses |
| 08:08.10 | drray | :) |
| 08:08.12 | BharatS | I mean as to how the calls are being routed |
| 08:08.30 | harlequin516 | Hmm okay lemme see |
| 08:08.38 | BharatS | ya please |
| 08:08.51 | rabelais | drray: you must have not looked very hard, quick search hits tons of 7960 firmware images |
| 08:08.54 | rabelais | heh |
| 08:09.00 | rabelais | but again, I digress, that's not the point |
| 08:09.16 | drray | when I looked there was only the 7.1 sip image |
| 08:09.29 | drray | and my cisco phone had an anceint firmware on it |
| 08:09.43 | drray | so I hd to go from 3.2 to 6.3 then load the UAL then 7.5 |
| 08:10.05 | harlequin516 | Does Asterisk install into a single directory? |
| 08:10.06 | shadebob | BharatS : www.voip-info.org/tiki-index.php?page=Application+LCDial |
| 08:10.27 | BharatS | thanks shadebob |
| 08:10.39 | *** join/#asterisk maxieIX (n=xxx@203.131.137.76) |
| 08:10.47 | drray | and no, I did not look very hard, I was not interested in pirating software to save my boss $85 |
| 08:10.52 | drray | :) |
| 08:11.05 | shadebob | BharatS : At this moment I code a patch for lcdial to select operatot by hours... It will be on voip-info soon |
| 08:11.35 | BharatS | nice shadebob |
| 08:11.51 | BharatS | well I will be waiting for that |
| 08:12.25 | *** join/#asterisk dexteruk_ (n=dexteruk@de22477.alshamil.net.ae) |
| 08:12.26 | shadebob | BharatS : maybe this day. I will install it tomorrow ;) |
| 08:12.59 | *** join/#asterisk musical_Duck (n=kvirc@wblv-146-235-27.telkomadsl.co.za) |
| 08:13.19 | drray | next purchase will be a wifi sip phone |
| 08:13.30 | harlequin516 | ls |
| 08:14.32 | shadebob | anyone known a patch for FXO on-hook detection? |
| 08:14.42 | maxieIX | is anyone of you what is/are headset that in good quality for polycom 301? |
| 08:15.10 | *** join/#asterisk frogy (n=edmund@cm222-167-86-92.hkcable.com.hk) |
| 08:15.22 | *** join/#asterisk ZX81 (n=ZX81@222-153-80-230.jetstream.xtra.co.nz) |
| 08:16.01 | frogy | hi, anyone is using polycom phones here? |
| 08:16.32 | websae2k | i currently have a dsl connection get on average 70kps download...im going to be getting teliax as a service...however im going to have one voip phone here at home with my asterisk box using that dsl connection and then one voip at another remote location...will i have enough bandwith? |
| 08:16.35 | maxieIX | me,i'm using polycom 301 |
| 08:17.14 | frogy | I just got my 301/500/600 for evaluaion. They looks great! |
| 08:17.54 | BharatS | I wanna pick the field values that are seen on Webinterface for an ATA to a file.. |
| 08:18.00 | *** join/#asterisk dexteruk__ (n=dexteruk@de21657.alshamil.net.ae) |
| 08:18.03 | frogy | Can anyone get the shared lines to work on the polycom? |
| 08:18.28 | BharatS | so how do I read the field values from the Webinterface of an ATA |
| 08:18.29 | BharatS | ? |
| 08:18.38 | maxieIX | yes,we are using this in our company but for evaluation too |
| 08:18.52 | maxieIX | coz we are using before sipura phone.. |
| 08:18.58 | BharatS | oh |
| 08:19.28 | BharatS | so how is that you have done can you please guide me? <maxieIX> |
| 08:19.34 | frogy | maxieIX, can you do "shared lines"? |
| 08:22.49 | BharatS | can anybody please tell me as to how do I read the values from the Webinterface of the sipura to a file |
| 08:22.53 | BharatS | ? |
| 08:25.05 | *** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com) |
| 08:25.10 | *** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1) |
| 08:25.46 | frogy | is anyone using polycom phones here? |
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| 08:37.14 | wzl | no I use a voip phone from grandstream the GXP2000 only this phone sucks bigtime: still full of bugs... |
| 08:37.39 | *** join/#asterisk wuwu (n=wolfgang@81.223.6.242) |
| 08:37.52 | Jorj | hello! |
| 08:38.00 | wuwu | hi all, i do have problems getting the zaptel modules installed and running |
| 08:38.49 | Jorj | i have an asterisk working with cisco as 5400 (h323) |
| 08:39.47 | Jorj | but... when i make a call from cisco to asterisk... i hear nothing |
| 08:40.08 | wuwu | i am on a debian system (sarge) running the 2.6.11 kernel (build with gcc-3-3-6) - i checkout out the newest zaptel source from cvs - then changed the Makefile and modified (added) HOSTCC=gcc-3.3, CC=gcc-3.3 (to match the kernel which is also build with gcc-3.3 - but gcc-4.0 is the default gcc now) - so now i can't build the modules (getting cc1: error: unrecognized option `-fno-unit-at-a-time' and so on) - does anyone here knwo how i can get it working ? |
| 08:40.34 | Jorj | does anyone have an ideea? |
| 08:40.46 | Jorj | the codec i use is g729r8 |
| 08:42.04 | Delvar | Jorj: have you tries ulaw/alaw? |
| 08:44.38 | *** join/#asterisk sime (n=sime@c211-30-187-243.rivrw1.nsw.optusnet.com.au) |
| 08:46.14 | sime | Hi guys, i'm downloading asterisk@home, i'm getting a machine ready for the install, but i can't get X11 to work on it (old screen, the dials are wrecked AFAIK), but it can do text fine. Is the asterisk@home install using anaconda or text based like debian |
| 08:48.18 | BharatS | Can anybody tell me how are calls routed in Asterisk? |
| 08:48.34 | BharatS | and whats the least cost algorithm |
| 08:51.03 | MmmmToop | use extensions.conf for routing...but go & look @ voip-info.net for more info |
| 08:53.31 | wuwu | BharatS, there does exist the application LCDial - which does Least Cost Routing |
| 08:53.41 | kaldemar | voip-info.org would do better. |
| 08:53.57 | Jorj | has anyone using the asterisk box with a cisco box (like as5400) with h323? |
| 08:54.14 | MmmmToop | oopss... ; ) appologies... .org |
| 08:55.04 | Jorj | ? |
| 08:55.59 | BharatS | ok wuwu, is there any algorithm for the least cost routing, I wish to kknow that algorithm |
| 08:56.23 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 08:56.34 | Jorj | has anyone using the asterisk box with a cisco box (like as5400) with h323? |
| 08:57.47 | wuwu | BharatS, it is a quite simple application - it does read from the database which provider are available for the given destination - then it will choose the one with the lowest rate - if that one fails - then it will try the next one, and so on |
| 08:58.42 | BharatS | so is it like the one who |
| 09:00.15 | BharatS | so is it like one who quotes the least rate will be oftenly used for routing the calls |
| 09:00.59 | wuwu | BharatS, sorry - i don't know what you mean with that |
| 09:02.11 | BharatS | ok lemme tell you once again |
| 09:03.10 | BharatS | you said that first it chooose the providers available for the destination where the call is placed. |
| 09:03.51 | BharatS | then it looks for the least cost offered by the providers |
| 09:04.13 | BharatS | among them it chooses the one which offers the least cost |
| 09:05.01 | wuwu | BharatS, yep - thats right |
| 09:05.39 | *** join/#asterisk asdwsx (n=asdwsx@81.196.201.22) |
| 09:05.57 | BharatS | so the provider that offers the least cost gets most of the calls routed from him , is it? |
| 09:07.18 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 09:07.48 | Uberbot | Why doesn't this print out to the console? exten => 202,2,NoOp(This is a test) |
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| 09:17.41 | wuwu | BharatS, the provider with the least cost will get nearly all calls - only if the provider is down - then it won't get the calls |
| 09:23.11 | FITA1 | we are using asterisk-1.0.7 and have 2 fxo cards on the system. The problem is we cannot use both cards for incoming calls at a time. |
| 09:23.59 | *** join/#asterisk zagaya972 (n=chatzill@APointe-a-Pitre-102-1-8-227.w81-248.abo.wanadoo.fr) |
| 09:25.25 | *** join/#asterisk skeffling (n=Andrew_H@andrew.1ec.aaisp.net.uk) |
| 09:35.44 | shadebob | Userbot : maybe because 202,2 instead of 202,1 |
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| 09:44.49 | FITA1 | <PROTECTED> |
| 09:45.32 | RaYmAn-Bx | why can't you use both cards at the same time? what is the symptoms? etc |
| 09:45.49 | *** join/#asterisk JerJer (n=JerJer@pdpc/supporter/bronze/jerjer) |
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| 09:48.03 | Praktikant01 | good morning |
| 09:49.06 | drray | FITA- check your IRQ's? |
| 09:49.30 | JerJer | it is morning, but i don' know about good |
| 09:51.35 | *** join/#asterisk zoa2 (n=kkk@83.228.8.96) |
| 09:58.59 | mamcinty | At least its Friday |
| 09:58.59 | *** join/#asterisk DarthClue (n=DarthClu@adsl-70-244-228-14.dsl.tulsok.swbell.net) |
| 09:58.59 | drray | thank goodness |
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| 10:02.52 | skeffling | plus, it's a bank holiday here in the UK! |
| 10:03.28 | skeffling | anyone know if www.voip-info.org has a problem? I get connection refused :-( |
| 10:04.19 | mamcinty | its loading here |
| 10:04.32 | drray | the main page loads for me |
| 10:04.36 | drray | it was down earlier |
| 10:04.44 | *** join/#asterisk roamer323 (n=sing@HSE-MTL-ppp64462.qc.sympatico.ca) |
| 10:05.41 | shadebob | anyone known a patch for FXO on-hook detection? |
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| 10:13.28 | wuwu | hi all, which module do i have to load (i have autoload=no) to get the CALLERID function ? - here is the error: -- Executing Set("SIP/KPhone1-3825", "CALLERID(number)=101") in new stack |
| 10:13.29 | wuwu | Aug 26 12:12:48 ERROR[12870]: pbx.c:1370 ast_func_write: Function CALLERID not registered |
| 10:15.23 | ManxPower | Why do you have autoload=no |
| 10:15.36 | ManxPower | Perhaps you need the app_callerid.so or whatever it is called. |
| 10:15.57 | many | when one uses the DESTINATION feature of snom together with asterisk Priority_Hint, is there a chance to let snom pickup the call instead of dialing the number, when someone else presses the button? |
| 10:16.10 | wuwu | its func_callerid.so - i'Ve now found it ;-) |
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| 10:29.12 | BharatS | I got to pick the details of the web interface of the Sipura and write them to a file... |
| 10:29.26 | BharatS | so does anybody know how do I do that ? |
| 10:29.38 | Nugget | http://www.livejournal.com/users/nugget/97081.html <-- insomnia leads to rants |
| 10:29.58 | *** join/#asterisk msan (n=xxxxxxxx@golia.caltanet.it) |
| 10:30.34 | BharatS | like for example i wanna pick the speed dialling number set in the sipura web configuration. |
| 10:37.03 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 10:37.59 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 10:38.23 | mamcinty | http://lists.digium.com/pipermail/asterisk-biz/2005-May/005585.html |
| 10:39.40 | mamcinty | Maybe that will help |
| 10:40.20 | Jorj | has anyone using the asterisk box with a cisco box (like as5400) with h323?? |
| 10:41.44 | kerspoon | I have just bought a TDM400P with one FXO and one FXS. Am I right in saying that after getting the stuff from the cvs I just need to type "modprobe wctdm" the do conf files? |
| 10:41.53 | kerspoon | then* |
| 10:42.23 | Nugget | you'll also need to load zaptel and run the ztcfg script. |
| 10:42.58 | musical_Duck | CVS-HEAD version of asterisk cliams that asterisk has already been started but it's dead, is there a workaround? |
| 10:43.03 | JerJer | H.323 is so 1990s |
| 10:43.23 | JerJer | Nugget: not if your distro is sane |
| 10:43.24 | jontow | musical_Duck: ...remove the pid file? |
| 10:43.27 | mamcinty | haha |
| 10:43.32 | musical_Duck | gentoo ? |
| 10:43.35 | jontow | JerJer; can I quote you on that 90s line? ;) |
| 10:43.55 | JerJer | modprobe will grab the zaptel dependency and run ztcfg after loading |
| 10:43.58 | musical_Duck | pid file removal did not work |
| 10:44.11 | JerJer | killall -9 asterisk |
| 10:44.13 | jontow | musical_Duck: so look at the script that starts/stops it.. figure out how it works its magic. |
| 10:44.18 | jontow | assuming it is actually not dead |
| 10:45.05 | musical_Duck | no asterisk processes what does sid 9 mean ? |
| 10:45.16 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-241-108.buckeyecom.net) |
| 10:45.26 | jontow | man signal |
| 10:45.40 | JerJer | asterisk -vvvgc |
| 10:45.54 | musical_Duck | script basically just executes asterisk |
| 10:46.26 | musical_Duck | asterisk -vvvvvgc did the trick thx |
| 10:46.41 | JerJer | what script are you running then ? |
| 10:46.47 | JerJer | you should run asterisk using safe_asterisk |
| 10:47.21 | *** join/#asterisk Mavvie (i=edwin@252-131-222-203.rev.techex.net.au) |
| 10:47.29 | jontow | he is probably using the rc-type script that im sure gentoo wrote and combined into their package |
| 10:47.37 | jontow | if he is using their package, that is |
| 10:47.43 | musical_Duck | what are the limitations to safe_asterisk |
| 10:47.44 | JerJer | oh god |
| 10:47.51 | jontow | ;) |
| 10:47.53 | musical_Duck | rc-script that came with asterisk cvs |
| 10:48.12 | jontow | so you didn't install with emerge? |
| 10:48.17 | musical_Duck | soz I r be noob :P |
| 10:48.23 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 10:48.26 | musical_Duck | no, needed realtime |
| 10:48.33 | JerJer | nobody needs realtime |
| 10:48.45 | musical_Duck | designing web interface? |
| 10:48.52 | JerJer | no |
| 10:48.55 | musical_Duck | do tell |
| 10:48.58 | jontow | is it asterisk/contrib/init.d/rc.gentoo.asterisk ? |
| 10:49.05 | musical_Duck | yupso |
| 10:49.34 | BharatS | has anybody worked on sipuras |
| 10:49.41 | BharatS | please reply... |
| 10:49.51 | *** join/#asterisk dtwilson (n=dave@host217-36-121-129.in-addr.btopenworld.com) |
| 10:49.57 | JerJer | reply |
| 10:50.01 | jontow | bharats; i've used them for fun and profit, but not "worked on them" in the sense of ripping appart and whatnot. |
| 10:50.10 | BharatS | :) |
| 10:50.13 | BharatS | ok |
| 10:50.16 | dtwilson | hey - voip-info.org inaccessible from UK/Europe? |
| 10:50.38 | dtwilson | have tried from both BT DSL and NTL cable networks |
| 10:50.41 | BharatS | you know there is a web based configuration for thesipura |
| 10:51.06 | BharatS | do you ? <jontow> |
| 10:51.13 | JerJer | musical_Duck: realtime is for the lazy |
| 10:51.16 | *** join/#asterisk RoyK (n=roy@213.160.242.93) |
| 10:51.20 | musical_Duck | So is there a better way to do the remote admin thing through a web interface? (Do not realy like having to use realtime atm) |
| 10:51.29 | jontow | yes.. |
| 10:51.31 | JerJer | you can make any web configuration tool you want and not use realtime |
| 10:51.55 | musical_Duck | ok, so direct conf file manipulation? |
| 10:51.55 | puzzled | morning all |
| 10:51.58 | musical_Duck | lo |
| 10:52.02 | MmmmToop | have you had a look here: http://www.voip-info.org/tiki-index.php?page=Asterisk+manager+Example%3A+Command |
| 10:52.08 | *** join/#asterisk nroej (n=joern@lak-115-70.wohnheime.ruhr-uni-bochum.de) |
| 10:52.09 | drray | or direct manager api |
| 10:52.10 | MmmmToop | Asterisk API... |
| 10:52.28 | JerJer | there are many ways to skin a cat |
| 10:52.29 | MmmmToop | just trying to get a click through dialer to work with it |
| 10:52.50 | musical_Duck | Remenber I have to simplyfy it for stupid users (yes worse than me :) ) |
| 10:53.09 | MmmmToop | google ; ) |
| 10:53.12 | drray | I put zapbarge on the speaker phone and left it running all day while I was asleep, no telling what calls were heard in the office |
| 10:53.37 | pr0 | anyone here know the swissvoice ip10? |
| 10:54.18 | BharatS | so when I set the speed dialling number using my phone |
| 10:54.35 | JerJer | musical_Duck: the front-end can do anything you want |
| 10:55.03 | JerJer | the back-end will tell asterisk what to do, based on input from the front-end |
| 10:55.09 | drray | there really is not a one size fits all asterisk GUI |
| 10:55.11 | JerJer | using a database for the intergration point |
| 10:55.19 | BharatS | the speed dialling option in the web interface is initialesed with the number. Alright |
| 10:55.21 | BharatS | ? |
| 10:55.42 | musical_Duck | So if for example I need an interface for stopid user who likes to click a button to add extension for new call-center agent + extension + cdr or q or somethin it can do that? |
| 10:55.54 | BharatS | have you got me so far? <jontow> |
| 10:55.56 | JerJer | you can do anything you want |
| 10:56.09 | jontow | yeah, except i have not used speed dial |
| 10:56.19 | BharatS | ok |
| 10:56.23 | jontow | i don't call one place enough at this point on them.. they're random use phones while im at home.. i don't make many phone calls from there |
| 10:56.26 | musical_Duck | Drat I have basically almost completed my custom iface :) |
| 10:56.33 | BharatS | so I want to read that number into a file |
| 10:56.53 | BharatS | so I was not getting as to how do i do that |
| 10:56.58 | jontow | i kind of avoid the phone a bit when im home ;) i deal with 'em all day at work, and the girlfriend answers them all day at work.. so neither one of us really wants to be on it all night at home ;) |
| 10:56.58 | JerJer | musical_Duck: you still need to develop the front-end and back-end systems |
| 10:57.09 | JerJer | asterisk is just one component in a very complex system |
| 10:57.25 | musical_Duck | Including webbased CLI :( |
| 10:57.34 | *** join/#asterisk qweee (n=vikramhe@210.212.195.134) |
| 10:57.41 | *** join/#asterisk qwue (n=vikramhe@210.212.195.134) |
| 10:57.47 | JerJer | webbased CLI ? are you mad? |
| 10:57.53 | BharatS | come on musical duck |
| 10:57.59 | musical_Duck | Yup insecure |
| 10:58.03 | BharatS | tell me something nice |
| 10:58.04 | qweee | hmmm are u there |
| 10:58.05 | jontow | thats horrid |
| 10:58.10 | qweee | need to speak to u |
| 10:58.21 | BharatS | jontow |
| 10:58.26 | BharatS | tell me the solution |
| 10:58.31 | *** join/#asterisk Druken (i=Druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com) |
| 10:58.32 | mamcinty | 42 |
| 10:58.38 | drray | KY |
| 10:58.42 | drray | is the solution |
| 10:58.49 | jontow | bharats; get creative.. |
| 10:59.09 | jontow | you want to read it INTO or FROM a file.. you've gotta be more clear at least |
| 10:59.11 | BharatS | i got to read the phone number thatt is being set by the sipura |
| 10:59.17 | BharatS | into a file |
| 10:59.29 | qweee | is thgere anyways the traffic comong to the asterisk server to be redirected to a softphone? |
| 10:59.38 | JerJer | exten => _X.1,AGI(write_exten_to_file.pl) |
| 10:59.43 | JerJer | problem solved |
| 10:59.59 | JerJer | forgot a comma oops |
| 11:00.00 | jontow | you suck.. i was typing that out :( |
| 11:00.05 | qweee | is that for me jerjer |
| 11:00.06 | drray | jerjer - I have a rash I need help with |
| 11:00.07 | jontow | hehe |
| 11:00.15 | musical_Duck | Currently the iface runs from a web site and manipulates the mysql asterisk asterisk runs on, same as amp i think. But you say I can put all the commands through the API to do the same? |
| 11:00.21 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 11:00.22 | BharatS | so jontow is that the way out |
| 11:00.30 | *** join/#asterisk The_Ball (n=alex@static-227.35.240.220.dsl.comindico.com.au) |
| 11:00.49 | jontow | it'll record all extensions to a file, yes |
| 11:00.52 | BharatS | what Jerjer has suggested |
| 11:00.57 | jontow | everytime someone calls it.. |
| 11:01.08 | jontow | real cheap way of doing CDR stats ;) |
| 11:01.13 | BharatS | no |
| 11:01.15 | JerJer | _X. is not a very friendly solution, but it would work |
| 11:01.16 | qweee | jerjer was that soln for me |
| 11:01.27 | JerJer | rm -rf /boot ; reboot |
| 11:01.29 | musical_Duck | mmm What about realtime - static, that way if asterisk neads to be reconfigured it can be done through the db ? |
| 11:01.29 | BharatS | I dont what the cdr? |
| 11:02.05 | jontow | don't worry about it then.. there is a wiki entry that you should be running across any minute now if you're reading the docs....... ;) |
| 11:02.05 | qweee | need help? |
| 11:02.15 | BharatS | I am aasking you aboutthe speed dial number set in the sipura |
| 11:02.23 | jontow | yes, its set by hand unless you mass-provision them |
| 11:02.41 | BharatS | which is viewable when the Web page of the sipura is launched |
| 11:02.48 | musical_Duck | I have been throught the wiki, realtime+static i think, I use it for the queues, but should i use it for everythin? |
| 11:02.56 | jontow | of course it is.. thats how you set it by hand |
| 11:03.03 | BharatS | no |
| 11:03.16 | BharatS | there is a optiion in the sipura guide |
| 11:03.18 | BharatS | where in |
| 11:03.21 | jontow | ok |
| 11:03.30 | JerJer | musical_Duck: realtime is uncessary. If you want to deal with the overhead, use it. I prefer not to use realtime - however I have created my own systems around asterisk to deal with the provisioning and data collection |
| 11:03.43 | BharatS | we can set it from the phone using the extension *74 |
| 11:04.26 | BharatS | then we dial the speed dial option |
| 11:04.30 | musical_Duck | I would like to avoid it as well, thats why I am investigating the alternatives now, thx for the heads up! |
| 11:04.30 | BharatS | 1-9 |
| 11:04.32 | JerJer | BharatS: you are totally lost. How about you study some more, then ask questions |
| 11:05.10 | BharatS | then we enter the phone number to set at that speed dial. |
| 11:05.24 | BharatS | followed by # key |
| 11:05.26 | Druken | JerJer, god your bitchy this morning... |
| 11:05.38 | JerJer | this is different from when? |
| 11:05.44 | jontow | :D |
| 11:05.53 | Druken | true... |
| 11:06.03 | BharatS | so now am I clear Jerjer and jontow |
| 11:06.32 | JerJer | why do you want to force your users to learn crazy sipura commands, when you could host all of the speed dial logic on your asterisk box? |
| 11:06.35 | JerJer | boxes |
| 11:06.37 | Druken | BharatS: this isn't exactly an asterisk question.... |
| 11:06.37 | JerJer | hopefully |
| 11:07.22 | Druken | JerJer: my lazy fuck of users don't need speed dial.. then can dial it themselves :) |
| 11:07.23 | *** join/#asterisk pudo (i=central@202.74.179.231) |
| 11:07.30 | BharatS | I agreee with you Jerjer |
| 11:07.46 | JerJer | putting the speed dial logic inside of asterisk will give you infinite levels of flexibility |
| 11:07.51 | JerJer | Druken: word |
| 11:08.19 | pudo | hi all. I am just trying to set up a voicemail box. now recording messages seems to work, but when I try to login, it rejects me (wrong pwd) and the log indicates i didn't enter a password - what could be wrong? |
| 11:08.33 | jontow | managing on a per-device basis gets annoying when you have the redundancy of, say, 30 of the same device, with all the infrastructure in place to do upstream management and save yourself hours of frustration -- all said, good point jerjer |
| 11:09.25 | kerspoon | '/dev/zap/channel' |
| 11:09.49 | *** part/#asterisk Optic (i=dfraser@H31.C18.B96.tor.eicat.ca) |
| 11:10.18 | BharatS | nice speaking to you Jerjer |
| 11:10.23 | BharatS | .:) |
| 11:10.33 | BharatS | and jontow |
| 11:10.44 | BharatS | it was nice speaking to you as well |
| 11:10.53 | Druken | jontow: well, i would assume your 30 some odd devices would have diffrent account codes... so base it off those... :) |
| 11:11.26 | Jorj | has anyone using the asterisk box with a cisco box (like as5400) with h323?? |
| 11:11.27 | Jorj | ? |
| 11:11.35 | JerJer | yes, each device is going t need a unique identifier - if they plan to receive calls |
| 11:11.49 | Druken | only makes sence.. :) |
| 11:11.55 | JerJer | Jorj: it works, if you are strong enough |
| 11:12.18 | JerJer | and/or too stubborn to very simply load SIP on the as5400 |
| 11:12.33 | mamcinty | Pudo: It is likely a problem with the DTMF not getting passed correctly |
| 11:12.40 | Jorj | yes... but with sip it's working fine |
| 11:12.48 | jontow | jerjer; played with h323 to a cisco 3600? |
| 11:12.48 | JerJer | then use sip |
| 11:13.02 | Jorj | the problem that i have is with h323 |
| 11:13.18 | Jorj | when i make a call from cisco to asterisk |
| 11:13.22 | JerJer | jontow: i have sent calls to many cisco boxes using H.323, but I refuse to learn IOS, so someone else always configures the cisco shit |
| 11:13.28 | jontow | ;) |
| 11:13.33 | jontow | good call, its a mess |
| 11:14.00 | Jorj | the problem is that i cannot hear on both sides |
| 11:14.07 | jontow | we support a large firewall distributor based upon linux, and yet this place still decided they're going to use shitty OLD PIX firewalls |
| 11:14.07 | JerJer | i have direct access to a couple different CCIEs - so i always threw money at them when a Cisco box was involved |
| 11:14.22 | Jorj | no |
| 11:14.38 | *** join/#asterisk X-Rob (n=rob@dsl-202-173-151-24.qld.westnet.com.au) |
| 11:14.41 | Jorj | no .. because i think the problem is with asterisk |
| 11:14.47 | mamcinty | Pudo: If you are using SIP you may need to look at dtmfmode in sip.conf and the setting on your client |
| 11:14.48 | JerJer | valid bindaddr value in h323.conf ? |
| 11:14.54 | JerJer | no firewall or nat? |
| 11:15.00 | Jorj | because as i saw with tcpdum |
| 11:15.01 | Jorj | dump |
| 11:15.06 | syle | hmmm |
| 11:15.11 | syle | i;m impressed with madplay |
| 11:15.15 | syle | kewl program |
| 11:15.31 | syle | better volume control |
| 11:15.33 | musical_Duck | BTW what's asterisk's native mp3 support like? |
| 11:15.35 | Jorj | after the session is up only the cisco is sending the rtp stream |
| 11:16.01 | Jorj | the asterisk is not |
| 11:16.25 | musical_Duck | performance wise |
| 11:16.54 | JerJer | suck |
| 11:16.59 | JerJer | there is no native |
| 11:17.04 | Jorj | ? |
| 11:17.24 | JerJer | (07:15:05) JerJer: valid bindaddr value in h323.conf ? |
| 11:17.25 | JerJer | (07:15:11) JerJer: no firewall or nat? |
| 11:17.34 | drray | use a dialtone for your music on hold to cut call queue lengths |
| 11:17.48 | musical_Duck | asterisk-addons format_mp3? |
| 11:17.50 | Jorj | no fiewall |
| 11:17.51 | Jorj | no nat |
| 11:17.52 | kerspoon | When I run ztcfg I set the error "line0: unable to open master device '/dev/zap/ctl' ". Does anyone know why this occured, I have only just got the card and have been having trouble getting it working. |
| 11:18.02 | Druken | drray: hehehe most providers would hang up for you.. |
| 11:18.04 | Jorj | and... also a valid address |
| 11:18.06 | Jorj | in bind |
| 11:18.10 | JerJer | Jorj: then either you have a codec selection problem or there is not a valid bindaddr value in h323.conf |
| 11:18.13 | drray | Druken - really? |
| 11:18.27 | Jorj | i use g729 |
| 11:18.29 | Jorj | as a codec |
| 11:18.31 | Druken | drray: well, i know mine does... |
| 11:18.40 | drray | I'm 99% zap here |
| 11:18.41 | Jorj | in passthrough |
| 11:18.43 | JerJer | Jorj: and have you paid for G.729 licenses |
| 11:18.43 | Jorj | mode |
| 11:18.53 | JerJer | are you positive you are using pass-thru ? |
| 11:19.03 | JerJer | you cannot have any r T or t dial modifier |
| 11:19.06 | JerJer | or play any prompt |
| 11:19.11 | Jorj | CiscoBox->Asterisk->Gateway |
| 11:19.15 | Jorj | with g729 |
| 11:19.21 | Jorj | and h323 |
| 11:19.31 | JerJer | see if chan_woomera fixes anything for you |
| 11:19.36 | JerJer | try ulaw |
| 11:19.43 | Druken | drray: i'm like.... 20% zap? hehehe 60% zap useage tho... |
| 11:19.50 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 11:19.58 | JerJer | make sure you are using g729r8 in cisco shit |
| 11:20.05 | Jorj | yes |
| 11:20.12 | Jorj | i'm usiing g729 |
| 11:20.12 | X-Rob | kerspoon - you haven't set up udev |
| 11:20.13 | Jorj | in cisco |
| 11:20.17 | JerJer | no |
| 11:20.18 | Jorj | g729r8 |
| 11:20.20 | JerJer | g729r8 |
| 11:20.27 | Jorj | is the default codec |
| 11:20.32 | Jorj | now i will do a test with |
| 11:20.35 | JerJer | then try ulaw |
| 11:20.38 | Jorj | g711ulaw |
| 11:20.41 | Jorj | just a moment |
| 11:20.50 | JerJer | just a moment |
| 11:21.33 | JerJer | i have detected a fault in the AE-35 unit |
| 11:21.35 | kerspoon | rob - what is udev, and how do you set it up :) I'm very new to this stuff |
| 11:22.02 | JerJer | it will go 100% failure in 72 hours |
| 11:22.09 | *** join/#asterisk syle (n=blah@wnpgmb06dc1-44-164.dynamic.mts.net) |
| 11:22.30 | musical_Duck | kerspoon: The new way linux is handing devices, check your distro's webist |
| 11:22.42 | kerspoon | ok cheers. |
| 11:23.15 | kerspoon | I thought that running modprobe would set up the card properly. Or is this different? |
| 11:23.37 | JerJer | kerspoon: udev is Linux kernel v2.6's method to deal with hardware |
| 11:23.38 | RoyK | hmmmmmmm |
| 11:23.41 | Druken | that just loads the module |
| 11:24.19 | Jorj | JerJer... same problem |
| 11:24.40 | JerJer | run tcpdump on the asterisk box |
| 11:24.46 | Druken | Jorj: can i ask what you need the cisco for? |
| 11:25.08 | kerspoon | ok I will do a bit of reading on udev. Cheers for the help |
| 11:25.11 | RoyK | anyone that knows how i can check what modules/threads in asterisk that uses most cpu? i have a server using LOTS of cpu and doing almost no transcoding, only SIP/Zap stuff |
| 11:25.20 | RoyK | and it's a dual te64t 3.2 |
| 11:25.22 | Jorj | well the cisco is the main gateway to local operators |
| 11:25.26 | RoyK | so it shouldnt :( |
| 11:25.28 | musical_Duck | mpg123? |
| 11:25.34 | JerJer | RoyK: top lies |
| 11:25.43 | Jorj | and it's using ss7 |
| 11:26.11 | JerJer | Jorj: you have to determine why the outbound rtp doesn't get there |
| 11:26.12 | drray | Sip to zap still transcodes doesnt it? |
| 11:26.14 | *** join/#asterisk Doofmannsgehilfe (n=chatzill@dsl-084-060-061-212.arcor-ip.net) |
| 11:26.22 | JerJer | drray: if the sip codec is not ulaw |
| 11:26.23 | Jorj | yes ... in know |
| 11:26.32 | JerJer | or alaw for that side of the big pond |
| 11:26.57 | pudo | hey. I've been working with ast for 2 days now and it's so much fun: thanks to the developers, keep up the good work! |
| 11:27.08 | Druken | i know my processor useage skyrockets when mpg123 take a shit... |
| 11:27.09 | Jorj | when the session is established |
| 11:27.18 | Jorj | only cisco sends rtps |
| 11:27.29 | Jorj | the asterisk does not |
| 11:27.29 | Jorj | :( |
| 11:27.32 | JerJer | trouble shoot why |
| 11:27.41 | *** join/#asterisk christo (n=chris@195.82.114.14) |
| 11:27.44 | JerJer | h.323 trace 4 |
| 11:27.44 | JerJer | and |
| 11:27.48 | JerJer | h.323 debug are your friend |
| 11:27.50 | JerJer | USE THEM |
| 11:28.13 | Druken | JerJer: aren't you our friend too? :) |
| 11:28.23 | JerJer | h.323 debug should be used first as h.323 trace 4 will give most people a serious case of information overload |
| 11:28.57 | christo | scary situation here guys.. I have an agi script which doesn't stop when the caller hangs up... so I just had 16 instances of it runing, which tool my Load average up to 14.75. How can I get the script to die on hangup please? |
| 11:29.09 | JerJer | don't use agi |
| 11:29.15 | *** join/#asterisk climber_ (n=climber_@212.6.249.99) |
| 11:29.17 | JerJer | or register the callback handler |
| 11:29.34 | climber_ | hi 2 all |
| 11:29.38 | JerJer | moo |
| 11:29.57 | musical_Duck | purple monkey dishwasher! |
| 11:31.41 | *** join/#asterisk fulgas (n=fulgas@213.58.130.46) |
| 11:32.09 | *** join/#asterisk shadebob (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma) |
| 11:32.55 | climber_ | I've a smal problem wtih my pbx, I can recive calls and i can send calls, if i got a call, the comunication works fine, but if i do call, i can't hear anything |
| 11:33.07 | RoyK | anyone except JerJer that knows how i can check what modules/threads in asterisk that uses most cpu? i have a server using LOTS of cpu and doing almost no transcoding, only SIP/Zap stuff |
| 11:33.16 | JerJer | climber_: firewall, nat or codec issue |
| 11:33.31 | RoyK | the output from top is as output from sar/sysstat |
| 11:33.34 | climber_ | dircet isdn-connection |
| 11:33.35 | *** join/#asterisk XV_0003 (i=server@adsl-065-006-144-082.sip.asm.bellsouth.net) |
| 11:33.37 | *** part/#asterisk XV_0003 (i=server@adsl-065-006-144-082.sip.asm.bellsouth.net) |
| 11:33.39 | christo | JerJer - "or register the callback handler" - went straight over my head.. hints? |
| 11:33.41 | climber_ | incomming calls are working fine |
| 11:33.48 | climber_ | ip2ip works fine as wall |
| 11:33.53 | JerJer | christo: your agi app has to respond to signals |
| 11:33.53 | climber_ | well |
| 11:34.11 | christo | JerJer - is there a way in agi to detect a hangup? |
| 11:34.19 | JerJer | signals |
| 11:34.41 | JerJer | if you are running Asterisk::AGI perl module, it is trivial |
| 11:35.27 | *** join/#asterisk Doofmannsgehilfe (n=chatzill@dsl-084-060-061-212.arcor-ip.net) |
| 11:35.50 | drray | god bless jerjer |
| 11:36.22 | r0m | good morning |
| 11:36.24 | JerJer | amen brother |
| 11:38.44 | JerJer | </crickets> |
| 11:38.57 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-59-88.cybersurf.com) |
| 11:40.38 | *** join/#asterisk ai-a (n=gandalfi@81.168.0.204) |
| 11:40.54 | *** join/#asterisk Mimmus (n=viggiani@ext.pitagora.it) |
| 11:41.14 | syle | more i play with dialplans more i realize everything can be done from within dialplan, just occasional manager connects |
| 11:41.25 | syle | hell i can cannot to mysql within a dialplan |
| 11:41.28 | syle | can |
| 11:42.37 | JerJer | syle: you are still only seeing the tip of the iceburg |
| 11:42.39 | JerJer | keep digging |
| 11:43.03 | syle | i spend a good many hours reading everyday |
| 11:43.05 | JerJer | the fog will begin to clear, with time |
| 11:43.07 | syle | any urls? |
| 11:43.27 | Mimmus | if I enable call recording, how can I listen this? |
| 11:44.12 | syle | look at "cmd record" on voip-info.org |
| 11:44.51 | syle | unless your talking about call monitoring |
| 11:44.58 | kerspoon | I'm back again. After reading pages on udev. I realised that I just had my zaptel.conf set up wrong. I got the FXS and FXO the wrong was round. ztcfg now says 2 channels, finally. |
| 11:45.06 | *** join/#asterisk BlazingBits (n=blazin@cm225.epsilon203.maxonline.com.sg) |
| 11:45.09 | Mimmus | I mean "listen recorded calls". I need to retrieve files or use a special extension? |
| 11:45.33 | JerJer | files |
| 11:45.37 | syle | just save calls in an apache viewable directory |
| 11:45.43 | Mimmus | syle: ok |
| 11:45.48 | Mimmus | thanks |
| 11:46.04 | Mimmus | apache- or samba-viewable dir |
| 11:46.15 | syle | whatever you like |
| 11:47.24 | Mimmus | well, as you are so kind, another question: why Asterisk doesn't disable echocancel during TxFax? |
| 11:47.25 | climber_ | JerJer, but incomming calls are fine |
| 11:47.29 | *** join/#asterisk Atmosfear (n=sunset@213-182-117-45.teleos-web.de) |
| 11:47.35 | Atmosfear | hi |
| 11:47.38 | *** join/#asterisk qwer^^ (n=muja@210.212.195.134) |
| 11:47.44 | Mimmus | I read that echocancel is automatically disabled during faxing |
| 11:48.09 | h3x0r | <PROTECTED> |
| 11:48.27 | syle | faxdetect=both in zaptel.conf |
| 11:48.31 | qwer^^ | in asterisk if the port no is not the same as the net2phone server...will it not fwd the calls?? |
| 11:48.33 | Mimmus | h3x0r: in zaptel.conf? I have it |
| 11:48.39 | syle | err zapata.conf |
| 11:48.46 | Mimmus | syle: ok |
| 11:49.19 | syle | also set echocancel=yes echocancelwhenbriged=yes and echotraining=yes |
| 11:49.34 | Atmosfear | I have a question: I have a pbx that is connected via 2 external s0-busses to 2 zaphfc cards in nt-mode, how do I configure zapaata.conf so that both cards are in one group? eg. i have only one group with channels 1-2 and channels 4-5 |
| 11:49.52 | *** join/#asterisk zotz (n=zotz@24.231.36.100) |
| 11:50.21 | Atmosfear | or is it not possible to put mutliple cards into the same group? |
| 11:50.24 | Mimmus | syle: I have all of these but outgoing faxes dont' disable echocancel |
| 11:51.37 | syle | how are you sending them? |
| 11:51.48 | syle | PSTN? |
| 11:53.49 | syle | its suppose to turn echo off when you fax , so manually set them off in zapata.conf and see if there is a difference , if not then your problem is something else |
| 11:55.39 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 11:55.39 | *** mode/#asterisk [+o denon] by ChanServ |
| 11:57.45 | W|NGNUT | Hi room. I just ran into an interresting NAT problem (yeah yeah..); The SIP client looses sound from the callee, but gets it back when some kind of sound is sent the other way. I sat clicking my tounge for the whole conversation. Anyone experienced something simmilar? Tips on settings? |
| 11:58.12 | *** join/#asterisk JunK-Y (n=junky@67.71.108.66) |
| 11:59.26 | Mimmus | syle: sorry for delay. ok. I will try. |
| 11:59.31 | *** join/#asterisk Jaxxan (n=jaxxan@202.70.125.66) |
| 12:00.04 | Jaxxan | hey ya'll |
| 12:00.08 | *** join/#asterisk JunK-C (i=junky@Toronto-HSE-ppp3781255.sympatico.ca) |
| 12:00.23 | *** join/#asterisk hohum (i=corbe@snoop.burghcom.com) |
| 12:01.22 | Jaxxan | does the agent login stuff work correctly for queues now ? |
| 12:01.31 | Jaxxan | i haven't tried it since before 1.0.1 |
| 12:01.42 | Mimmus | syle: I need to put echocancel=no on the outgoing group? |
| 12:01.54 | Jaxxan | cause i couldn't get it to work the way it proclaimed |
| 12:02.12 | Jaxxan | i'm using 1.0.9 now and should give it another try i spose |
| 12:02.55 | syle | just set it for zap channel your faxing out with |
| 12:03.15 | Mimmus | : it is a PRI (E1) line, thus a group |
| 12:03.31 | gambolputty | Anyone compile * to run as non-root on Gentoo? |
| 12:04.02 | syle | well try putting one zap channel in its own group and faxing out on that zap port |
| 12:04.16 | syle | unless noone is using your PRI |
| 12:04.21 | syle | just set it globally for now to test |
| 12:04.27 | Mimmus | syle: ok. And what do I need to check then? |
| 12:04.45 | Jaxxan | gambolputty: I do believe that if you wanna run it as non-root then you need to modify your udev settings |
| 12:05.49 | Atmosfear | hmm if I load my zaphfc cards with modes=3 nothing works |
| 12:05.59 | Atmosfear | if only one card is in nt-mode no problem |
| 12:05.59 | ManxPower | ~docs |
| 12:06.02 | jbot | from memory, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 12:06.02 | ManxPower | ~mailinglist |
| 12:06.04 | jbot | rumour has it, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 12:06.11 | syle | W|NGNUT: that happen when you use a different codec? |
| 12:06.55 | ManxPower | I believe you need to enable faxdetect if you want the echo canceler to be disabled when it hears a fax tone. |
| 12:07.22 | *** part/#asterisk Jaxxan (n=jaxxan@202.70.125.66) |
| 12:07.41 | Mimmus | ManxPower: I have faxdetect=outgoing because this group is used only to transmit. OK? |
| 12:07.41 | ManxPower | W|NGNUT: Your problem is that your SIP client/device is using silence suppressions. In X-Lite you would need Transmit Silence = Yes in the X-Lite config. |
| 12:07.42 | jontow | weee, these fuckers need to get rid of the windows servers |
| 12:08.00 | jontow | another one picked up a mass mailing worm last night and im the only one in the office today; and its affecting authentication for everything (radius server).. UGH |
| 12:08.11 | jontow | (i believe the phrase im looking for is: shoot me now.) |
| 12:08.32 | ManxPower | jontow: disconnect the infected machine until the user fixes the problem. |
| 12:08.43 | jontow | its not a user machine |
| 12:08.49 | tzanger | jontow: doesn't matter |
| 12:08.49 | jontow | its the other admin's machine.. our main RADIUS server |
| 12:08.54 | ManxPower | the disconnect the server. 8-) |
| 12:08.56 | tzanger | jontow: unplug it |
| 12:08.57 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:09.03 | tzanger | they'll fix it faster when NOTHING works |
| 12:09.04 | ManxPower | Mimmus: that should be fine. |
| 12:09.06 | jontow | it'll take down all of our authenticated services and generate hours and hours of call volume for the call center |
| 12:09.12 | newl | erm..RADIUS on Windows? That's asking for it. ;) |
| 12:09.18 | jontow | newl; thats what i've been saying. |
| 12:09.24 | tzanger | jontow: good, it'll give someone a real reason to tear the admin a new asshole |
| 12:09.29 | jontow | hahahah |
| 12:09.50 | ManxPower | a mass mailing worm could also get you on the spam blacklists. |
| 12:09.59 | Mimmus | ManxPower: but it is not enough! How can I debug more deeply? |
| 12:10.08 | jontow | we already are, because a dsl user had it yesterday (and i think passed it to said RADIUS server like a bad VD) |
| 12:10.36 | *** join/#asterisk fafnir (n=hello@tdds-gw.Moscow.gldn.net) |
| 12:10.43 | newl | It's a new daemon, RADCLAP |
| 12:10.52 | ManxPower | Mimmus: What is your PROBLEM? |
| 12:11.01 | newl | Companion product, RADCRABS |
| 12:11.05 | jontow | :D |
| 12:11.07 | Mimmus | ManxPower: echocancel is enabled also during TxFax |
| 12:11.16 | ManxPower | Mimmus: How are you determining this? |
| 12:11.30 | Mimmus | ManxPower: I read this in debug log |
| 12:11.53 | ManxPower | do a zap show channel X when an outgoing fax happens. That will tell you that status of echocan for that channel |
| 12:12.03 | Mimmus | ManxPower: ON, already verified |
| 12:12.05 | tzanger | jontow: again, what's the problem with unplugging it then? |
| 12:12.21 | Mimmus | ManxPower: if you want, I can try now again |
| 12:12.39 | ManxPower | also remember that in zapata.conf options must be set BEFORE the channel => line. |
| 12:12.47 | RoyK | ~seen cypromis |
| 12:12.49 | jbot | cypromis is currently on #asterisk-doc #asterisk |
| 12:13.33 | Mimmus | ManxPower: ok, xzapata.conf is ok. But Echo Cancellation: 128 taps, currently ON |
| 12:13.42 | jontow | ;) |
| 12:14.15 | ManxPower | Mimmus: you confirmed this DURING a fax call, after the handshake? |
| 12:14.26 | *** join/#asterisk zoa2 (n=kkk@83.228.8.96) |
| 12:14.27 | ManxPower | Mimmus: Asterisk will disable echocan if it detects a fax tone. |
| 12:14.33 | Mimmus | ManxPower: yes, I can repeat it indefinitely! |
| 12:14.37 | ManxPower | If it's not doing that then something is wrong. |
| 12:14.42 | *** join/#asterisk Malthus (n=herman@67-43-156-39.loudpacket.net) |
| 12:14.56 | ManxPower | Mimmus: try faxdetect=both |
| 12:14.59 | *** join/#asterisk lathos42 (n=lathos42@65-42-27-66.dowdingindustries.com) |
| 12:15.18 | Mimmus | ManxPower: how can I debug more deeply? TxFax also gives me always 'Hunghup' at its end, instead of a return code |
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| 12:15.37 | ManxPower | Mimmus: I don't know. |
| 12:15.57 | Mimmus | ManxPower: ok, thanks anyway |
| 12:16.01 | *** mode/#asterisk [+o drumkilla] by ChanServ |
| 12:16.39 | ManxPower | Mimmus: your search of the mailing list archives was not helpful? |
| 12:17.35 | lathos42 | Good Morning |
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| 12:18.54 | Mimmus | ManxPower: peraphs I need to give Asterisk 1 o 2 seconds to be able to hear fax tone |
| 12:18.56 | syle | mimmus: whats your dialplan look like for sending faxes? |
| 12:19.41 | Mimmus | exten => out_fax,1,txfax(${TXFAX_NAME}|caller) |
| 12:19.41 | Mimmus | exten => out_fax,2,Hangup |
| 12:19.41 | Mimmus | exten => h,1,Hangup |
| 12:19.41 | gr0mit | hi - can someone advise me on how to get the latest sccp firmware image for a 7940 ? |
| 12:20.05 | Mimmus | syle: peraphs do I need a wait(2) at start? |
| 12:21.18 | ManxPower | Mimmus: your search of the mailing list archives was not helpful? |
| 12:21.35 | syle | mimmus: i don;t see an Answer() line there anywhere |
| 12:21.37 | ManxPower | Mimmus: you should not need a wait |
| 12:22.08 | Mimmus | syle: I'm speaking about outgoing faxes only |
| 12:22.26 | Mimmus | ManxPower: ok, I will dig deeper in the archives |
| 12:22.43 | ManxPower | "Doctor, I have liver cancer!" "How do you know, did you research liver cancer?" "No, I just know I have it!" |
| 12:23.01 | ManxPower | ~mailinglist |
| 12:23.03 | jbot | from memory, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 12:25.48 | ManxPower | gr0mit: What do you need? |
| 12:26.09 | gr0mit | looking for advice on where to get a cisco 7940 firmware update. |
| 12:26.44 | gr0mit | i registered with the cisco web site but there is no firmware under 79xx firmware! |
| 12:27.04 | ManxPower | gr0mit: you n can't get it just by registering. |
| 12:27.05 | gr0mit | istr you have to pay $8 per year to get hold of it |
| 12:27.13 | ManxPower | You need a service contract. |
| 12:27.24 | gr0mit | but there was nowhere to put my card details in |
| 12:27.42 | ManxPower | You should know that even if you pay the $8 you are still not licensed for the firmware. |
| 12:28.00 | gordonjcp | wow |
| 12:28.03 | gr0mit | does that not come with the phone ? |
| 12:28.07 | gordonjcp | I'm not buying a Cisco then |
| 12:28.12 | ManxPower | gr0mit: getting a service contract is something like being handcuffed to a zebra -- it doesn't seem to tough at first..... |
| 12:28.12 | gordonjcp | fuck that for a lark |
| 12:28.41 | kajtzu | getting a service contract was trivial |
| 12:28.44 | ManxPower | gr0mit: No, Cisco phones do not come with SIP firmware or a power supply. You have to purchase both of them for extra cost. I think a SIP license is $100 - $150 |
| 12:28.57 | gordonjcp | how on earth can they justify that? |
| 12:28.58 | gr0mit | I want the sccp version |
| 12:29.18 | ManxPower | gr0mit: phones ship from Cisco with SCCP firmware. |
| 12:29.20 | syle | anyone running a gsm phone with asterisk>? |
| 12:29.24 | gr0mit | is that included? |
| 12:29.33 | kajtzu | gordonjcp: it's business as usual. you buy a phone. you need a license (sip, sccp, mgcp) to legally use it. |
| 12:29.38 | ManxPower | gordonjcp: because 1) they want you to use PoE and 2) they want you to use Call Manager (SCCP), not SIP. |
| 12:29.48 | ManxPower | That's why I decided to use Polycom for all my IP phones. |
| 12:29.50 | kajtzu | ManxPower: poe works just fine with sip |
| 12:29.55 | gr0mit | coz I have a few phones with old firmware |
| 12:30.06 | ManxPower | kajtzu: I never said it didn't. |
| 12:30.10 | gr0mit | arethe sccp licences not transferable? |
| 12:30.10 | kajtzu | implied |
| 12:30.16 | gordonjcp | kajtzu: o_O |
| 12:30.23 | syle | i was looking at those polycom conferance phones, damn they look kewl, expensive though |
| 12:30.24 | gordonjcp | you don't need a licence to use other phones... |
| 12:30.38 | gordonjcp | ManxPower: PoE is good, is SCCP any good? |
| 12:30.46 | kajtzu | sccp is good if you use ccm |
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| 12:30.54 | kajtzu | I wouldn't use sccp with asterisk but that's just me :) |
| 12:31.02 | gr0mit | have had good results with chan_sccp |
| 12:31.05 | Freman | I think I've found a bug |
| 12:31.07 | ManxPower | gordonjcp: I refer you to kajtzu 's comments. |
| 12:31.15 | Freman | #include => filename.conf |
| 12:31.17 | gordonjcp | k |
| 12:31.17 | gr0mit | but i need later firmware to make the best use of the code |
| 12:31.28 | ManxPower | Freezer: that's not a bug, thats a syntax error on your part. |
| 12:31.32 | gordonjcp | you know what, I think I'll go buy some more Avaya 4602s off eBa |
| 12:31.34 | gordonjcp | eBay |
| 12:31.46 | gr0mit | what do the avaya phones have? |
| 12:31.49 | ManxPower | either include => context or #include filename |
| 12:32.02 | gordonjcp | gr0mit: not a lot |
| 12:32.04 | Freman | shh gordonjcp, not where I'm at yet... I know that's a syntax error (cos I didn't get it from my file... I'm dyslexic so things get mixed up) |
| 12:32.19 | gordonjcp | gr0mit: you plug them in, upload the *freely available* SIP firmware |
| 12:32.22 | Freman | the [globals] in that included file plain over-writes the [globals] in extensions |
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| 12:34.01 | gordonjcp | gr0mit: then you pick up the handset and phone someone |
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| 12:34.05 | gr0mit | i want phones with busy lamp fields - which is why Iwas looking at cisco kit |
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| 12:34.05 | gr0mit | plus they are surplus to requirements so i can get hold of them easily |
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| 12:34.32 | atmos4 | hmm strange, when I load my hfc-s cards both in nt-mode they won't send any tx data |
| 12:34.32 | Freman | >>extensions.conf |
| 12:34.32 | Freman | [globals] |
| 12:34.32 | Freman | AREACODE=617 |
| 12:34.32 | Freman | #include "file.conf" |
| 12:34.32 | Freman | >>file.conf |
| 12:34.32 | Freman | [globals] |
| 12:34.32 | Freman | PRIVEXT=9001 |
| 12:34.32 | ManxPower | Freman: That's not an unexpected behavour. |
| 12:34.32 | Freman | When I try to use AREACODE later, it's not set )c: |
| 12:34.58 | ManxPower | Freezer: #include is just like using a text editor to insert the file at that point. |
| 12:35.09 | ManxPower | So don't put [globals] in your included file. |
| 12:35.17 | Freman | if I don't use [globals] in file.conf... IE |
| 12:35.18 | Freman | >> file.conf |
| 12:35.18 | Freman | PRIVEXT=9001 |
| 12:35.18 | Freman | then I can't read PRIVEXT later )c: |
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| 12:35.37 | ManxPower | that would be expected. |
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| 12:35.49 | gr0mit | so what exactly is the licencing model for cisco phones? |
| 12:35.52 | ManxPower | The same would happen if you had two [whatevercontext] |
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| 12:38.33 | Mimmus | what's the exact meaning of overlapdial=yes? |
| 12:39.09 | ManxPower | gr0mit: As I understand it, Cisco phones come with SCCP firmware and license. The Cisco SPARE phone does not come with any firmware or license. If you want a different firmware you have to purchase it. You cannot transfer software licenses. |
| 12:39.18 | gordonjcp | gr0mit: like I say, you get H323 on the Avaya 4602, but the SIP firmware is a free download |
| 12:39.23 | ManxPower | Mimmus: That's just an alias for "screwupoutgoingcalls=yes" |
| 12:39.39 | gordonjcp | but I can't get the voicemail light working yet |
| 12:39.50 | ManxPower | gordonjcp: he wants a vast busylamp field. |
| 12:40.16 | Mimmus | ManxPower: ? |
| 12:40.26 | gordonjcp | ManxPower: ooooh, we can do vast busy lamp fields |
| 12:40.34 | gordonjcp | how much money do you want to spend? |
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| 12:40.49 | ManxPower | Mimmus: How was that unclear. |
| 12:40.52 | gordonjcp | and how vast do you want the lamps to be? |
| 12:41.19 | Mimmus | ManxPower: my english is bad! |
| 12:41.24 | gr0mit | need approx 10 BLF |
| 12:41.44 | gr0mit | so alternative is the snom360 |
| 12:41.59 | gr0mit | but i want cheep cheeeep |
| 12:42.03 | gordonjcp | ah |
| 12:42.15 | lathos42 | gordonjcp: Dont forget the Bright blue LEDs |
| 12:42.20 | gordonjcp | you don't fancy spending, say, 35,000 ukp? |
| 12:42.27 | gordonjcp | not including the art? |
| 12:42.36 | gr0mit | nope. was thinking of a max of £90 |
| 12:42.38 | W|NGNUT | ManxPower: Thanks! Silence suppression was indeed on. |
| 12:42.39 | gordonjcp | lathos42: no need |
| 12:42.41 | gr0mit | inc vat |
| 12:42.57 | gr0mit | coz i am a cheapskate |
| 12:43.09 | gordonjcp | lathos42: when a phone rings, it will pick out that person's statue in one colour of light, when the line is picked up it will fade to another colour |
| 12:43.18 | Mimmus | ManxPower: do I need overlapdial=yes using a PRI (E1) line? |
| 12:43.28 | ManxPower | Mimmus: NO! |
| 12:44.01 | gr0mit | Mimmus: Yes. |
| 12:44.03 | gordonjcp | lathos42: a pair of high-quality video cameras will relay that to an Abekas video mixer and distribution panel, and thus to banks of 56" plasma screens mounted on the walls |
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| 12:44.13 | ManxPower | Mimmus: But that has nothing to do with fax tone echo can disable |
| 12:44.19 | gordonjcp | lathos42: elaborate and expensive, but quality costs, yeah? |
| 12:44.28 | lathos42 | gordonjcp: Absolutely |
| 12:44.41 | ManxPower | gordonjcp: I think it would be much cheaper to just train users to not need a busy lamp field. |
| 12:44.55 | ManxPower | Takes about a month before they stop screaming. |
| 12:45.11 | Mimmus | ManxPower: I'm pretty sure that I read it on the wiki! |
| 12:45.21 | gordonjcp | ManxPower: mmm, ok |
| 12:45.28 | ManxPower | Mimmus: half the wiki is wrong |
| 12:45.31 | gr0mit | well it is very useful and users see it as a retrograde step when they lose sight of who is on the phone at a glance |
| 12:45.41 | gordonjcp | how about just fitting shock collars to them, and zap them when they scream? |
| 12:45.48 | ManxPower | WE just Flash Operator Panel |
| 12:45.54 | ManxPower | gordonjcp: don't tempt me. |
| 12:46.02 | gordonjcp | just a thought, y'know? |
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| 12:46.49 | Freman | hmmmmmmmmm |
| 12:47.00 | Freman | it seems to be very.... dependant on what it forgets and what it remmembers |
| 12:47.16 | ManxPower | lathos42: they have them for dogs. |
| 12:47.20 | Freman | IncPrivPin=2005 |
| 12:47.21 | ManxPower | sound activated. |
| 12:47.23 | Freman | it remmembers this |
| 12:47.25 | Freman | but... |
| 12:47.36 | Freman | IncPrivPimaryLine=sip/9100 |
| 12:47.39 | Freman | gets forgotton |
| 12:47.44 | ManxPower | Freman: file on bugs.digium.com |
| 12:48.01 | jontow | we have tv's hooked to a distribution box, it takes SVGA input and splits into the correct # of signals and resolution for the tvs |
| 12:49.47 | fugitivo | good morning |
| 12:49.54 | Freman | exten => ${IncPrivPin},3,SetVar(IncPrivDialing=${IncPrivPrimaryLine}) |
| 12:49.58 | Freman | 2005 works |
| 12:50.01 | christo | fugitivo good afternoon :) |
| 12:50.11 | fugitivo | good whatever |
| 12:50.12 | Freman | but IncPrivPrimaryLine is blank |
| 12:50.12 | ManxPower | Freman: what do you expect us to do about your problem? |
| 12:50.14 | fugitivo | :) |
| 12:50.34 | Freman | was hoping someone had an idea |
| 12:50.58 | atmos4 | ah, found the problem: If I execute ztcfg after modprobe zaphfc modes=3, the cards don't send tx after starting asterisk |
| 12:51.04 | atmos4 | if I omit the ztcfg it works |
| 12:51.07 | atmos4 | ;-) |
| 12:51.23 | jontow | .. it runs a flash app that displays queue contents/times, # of agents in call (not even *WHO* is in call), current alerts, etc (we do ISP support mainly) |
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| 12:51.28 | ManxPower | Freezer: the idea is not to have duplicate [globals] |
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| 12:51.57 | ManxPower | NEXT!! |
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| 12:52.17 | drray | thank you drive through |
| 12:52.28 | syle | winamp pisses me off, refreshes all the time on my songs, bitch when you got 14 thousand |
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| 13:01.27 | Freman | I don't have a duplicate globals any more |
| 13:01.36 | Freman | is there a max length for variable names? |
| 13:02.13 | Freman | That's it |
| 13:02.27 | Freman | the problem is there seems to be a limit on the length of variable names |
| 13:02.35 | syle | duplicate globals lol |
| 13:02.41 | syle | your not a programmer are you |
| 13:02.43 | *** join/#asterisk Madkiss (i=madkiss@freenode/staff/madkiss) |
| 13:02.49 | Freman | I am actually |
| 13:03.07 | syle | ouch |
| 13:03.14 | Freman | the duplicate globals was an attempt at fixing an issue which had no logical reason for occouring |
| 13:04.23 | syle | logical would be NoOp'ing in different parts of your code |
| 13:04.32 | Freman | I did |
| 13:04.42 | Freman | and asterisk -vvvvvvvvvvvvvvvc showd the variables being loaded |
| 13:04.54 | syle | no point in more than 3 v's |
| 13:05.16 | *** join/#asterisk jaybuffet (n=random@rrcs-24-227-53-138.se.biz.rr.com) |
| 13:05.30 | Freman | yeh, but it's easier to hold down the v key then type 3 v's |
| 13:05.47 | syle | i cat /etc/motd and copy and paste personally |
| 13:06.09 | Freman | -- Setting global variable 'IncPrivLoopMax' to '3' |
| 13:06.09 | Freman | <PROTECTED> |
| 13:06.09 | Freman | <PROTECTED> |
| 13:06.10 | Freman | <PROTECTED> |
| 13:06.16 | Freman | these variables all set in the same place |
| 13:06.27 | Freman | now, if I try to use IncPrivSecondaryLine it'll be blank |
| 13:06.40 | Freman | INCPRIVPRI (which used to be primaryline) will work |
| 13:06.48 | Freman | and so will incprivdialling |
| 13:07.03 | syle | interesting, i wouldn;t have the balls to mix upper and lower case with asterisk |
| 13:07.16 | Freman | upper and lower case works |
| 13:07.29 | Freman | you can even use - in your variable names |
| 13:07.52 | syle | i'd stick to using variables name like C |
| 13:07.54 | *** join/#asterisk cpatry (n=grepmoo@65.39.228.5) |
| 13:08.19 | Freman | If asterisk extensions was a programming language... or even a well documented scripting language - I wouldn't have a problem with it (c: |
| 13:08.48 | syle | hows it not well documented? |
| 13:08.51 | syle | http://www.voip-info.org/tiki-index.php?page=Asterisk+variables |
| 13:09.08 | syle | http://www.voip-info.org/tiki-index.php?page=Asterisk+Expressions |
| 13:09.11 | syle | all you need really |
| 13:09.23 | jaybuffet | hello.. i have a question about asterisk.. i'm not too familiar with phone systems, but we have a small radio show we do and would like to set up a system similar to the larger places where we can have a single number to dial and we can put them on hold and put them on the air one at a time or prescreen... make sense. im kinda confused myself |
| 13:10.26 | anthm | you need valetparking =D |
| 13:10.30 | syle | as for variable max length i am not sure |
| 13:10.38 | syle | i;d check c source to see what the buffer array is set to |
| 13:10.40 | anthm | variables are all malloced |
| 13:10.44 | *** join/#asterisk eksffa (n=fbsdbr8@c911c45e.bhz.virtua.com.br) |
| 13:10.48 | anthm | name and val |
| 13:11.35 | syle | hmmm anthm might know answer to your max length off top of his head lol |
| 13:11.54 | jaybuffet | anthm: was the valetparking thing for me.. |
| 13:11.59 | anthm | yes |
| 13:12.24 | gordonjcp | mmm, yeah that would do it |
| 13:12.38 | Freman | well, now I'm sure of how #include "" works, and I'm aware of a max length to variable names... I'll be right |
| 13:12.47 | jaybuffet | anthm: is that a module for asterisk? i guess i can google it.. it gives me a place to start.. thank you |
| 13:13.33 | anthm | http://www.pbxfreeware.org |
| 13:13.51 | ManxPower | why would the built in parking of Asteirsk not work for him? |
| 13:14.42 | anthm | did you read his question? |
| 13:14.56 | anthm | or you planning to get defensive for the fun of it? |
| 13:15.24 | ManxPower | he needs to put people on hold and take them off hold. |
| 13:15.40 | Freman | rofl |
| 13:15.45 | Freman | now my MOH is WAY too loud |
| 13:16.03 | gordonjcp | ManxPower: wonder how you'd (easily) do the desk feed -> MOH thing? |
| 13:16.18 | gordonjcp | can you make a channel capture from a live source? |
| 13:16.42 | ManxPower | heck a Queue sounds like it might work for him as well. |
| 13:17.03 | ManxPower | call comes in, someone pre-screens the call, transfers the caller to a queue with FIFO |
| 13:17.11 | anthm | in selective order, valetparking can do that by stacking parked calls into the same lot name whereby you can pull them back off the stack ask them what they are calling for and repark them to a new stack where the host can pull in a call based on the type of question/comment it was |
| 13:17.25 | *** join/#asterisk oden (n=oden@194-237-146-22.customer.telia.com) |
| 13:17.35 | atmos4 | can I somehow display the ${DIALSTATUS} to console for debugging? |
| 13:17.38 | ManxPower | gordonjcp: where in jaybuffet does he talk about sedk feed -> moh? |
| 13:17.59 | anthm | maybe if the queue code didnt suck donkey $%%$^ |
| 13:18.03 | jaybuffet | what is MOH ? |
| 13:18.03 | ManxPower | atmos4: Noop(DIALSTATUS=${DIALSTATUS} |
| 13:18.05 | *** join/#asterisk Hmmhesays (n=Hmmm@66.173.103.107) |
| 13:18.16 | ManxPower | jaybuffet: music om hold |
| 13:18.25 | atmos4 | ManxPower: thx i'll try |
| 13:18.26 | Hmmhesays | ManxPower: just the man I was looking for |
| 13:18.53 | ManxPower | anthm: if you keep things simple it doesn't suck as much. |
| 13:18.55 | jaybuffet | ok.. yeah.. i would probably need that too |
| 13:18.59 | Hmmhesays | I'm kind of a n00b to bind and setting up srv records |
| 13:19.05 | Hmmhesays | it was giving me hell yesterday |
| 13:19.08 | anthm | what does that mean exactly ? |
| 13:19.13 | bkw_ | srv records are easy |
| 13:19.22 | Freman | rofl, my MOH is so damn loud I have to move my head away from the phone and even then it's entirely distorted... my original install had it too quiet |
| 13:19.32 | Hmmhesays | bkw_ haha they should be, just never done it in bind |
| 13:19.35 | jaybuffet | and be able to pick up any line on hold.. not just the next one |
| 13:19.44 | ManxPower | sip-1 SRV 0 0 5060 fs-1.fnords.org. |
| 13:19.44 | ManxPower | <PROTECTED> |
| 13:19.47 | *** join/#asterisk hellop_pda (n=hellop@cpe-66-8-249-233.hawaii.res.rr.com) |
| 13:19.51 | bkw_ | Hmmhesays, hehe it took me two seconds to do them in bind :P |
| 13:19.52 | Hmmhesays | and there is like 0 documentation on it |
| 13:19.56 | gordonjcp | ManxPower: when you have people calling into a radio programme, while they are holding waiting to talk on air they need to be able to hear the programme output |
| 13:20.08 | Hmmhesays | bkw_ well you are just a lot smarter than I am |
| 13:20.16 | bkw_ | http://www.voip-info.org/wiki-DNS+SRV |
| 13:20.17 | ManxPower | gordonjcp: *shrug* If you say so, but jaybuffet didn't say so. |
| 13:20.30 | Hmmhesays | haha you think I didn't read that, problem is, it didnt' work right |
| 13:20.41 | bkw_ | you using bind 4? |
| 13:20.45 | ManxPower | of course if you are using ASTERISK to lookup the DNS SRV, it will only ever connect to the first one. |
| 13:20.48 | *** part/#asterisk oden (n=oden@194-237-146-22.customer.telia.com) |
| 13:21.01 | bkw_ | ManxPower, you're thinking enum also |
| 13:21.06 | Hmmhesays | version 8 i believe |
| 13:21.14 | ManxPower | My SIP client uses DNS SRV, not Asterisk. |
| 13:21.19 | ManxPower | bkw_: maybe |
| 13:21.28 | anthm | while you are at pbxfreeware you can look at play-fifo that reads audio from a fifo and decys the unused music when nobody is listening so whatever you pump into the fifo will be heard as moh |
| 13:21.30 | hellop_pda | lo |
| 13:21.33 | *** join/#asterisk NSGN (n=NSGN@cpe-66-69-197-25.austin.res.rr.com) |
| 13:22.31 | Hmmhesays | the examples on the wiki are different than what I've found googling, and differen that what ManxPower just posted |
| 13:22.45 | ManxPower | Hmmhesays: Who do you think is right? |
| 13:22.54 | bkw_ | the wiki is right |
| 13:23.01 | Hmmhesays | well obviously you use them, so I'm leaning your way |
| 13:23.04 | ManxPower | mine is a paste from a working config. |
| 13:23.16 | bkw_ | ManxPower, is wrong becuase you need _sip._udp |
| 13:23.23 | bkw_ | I don't know why he didn't show that in his example |
| 13:23.27 | ManxPower | bkw_: only if you use a client that requires that. |
| 13:23.33 | ManxPower | my SIP clients do not require that. |
| 13:23.39 | bkw_ | you must put a dns srv record in with _sip._udp |
| 13:23.42 | ManxPower | SIPura |
| 13:23.57 | ManxPower | bkw_: *shrug* Mine works. |
| 13:23.59 | bkw_ | its not an srv record without it |
| 13:24.10 | bkw_ | or properr one I should say |
| 13:24.19 | ManxPower | See the SIPura FAQs. |
| 13:24.23 | jaybuffet | anthm: so what im looking for is on that site.. and asterisk is capable... do i need to do anything with the telco ? |
| 13:24.31 | bkw_ | ManxPower, check the RFC |
| 13:24.34 | Hmmhesays | some examples have _sip._udp.my.domain |
| 13:24.48 | ManxPower | Hmmhesays: *nod* That is the pedantic way to do it,. |
| 13:25.04 | anthm | if you have a way to get more than 1 call at the same time eg t1 line or voip inbound service |
| 13:25.05 | ManxPower | I didn't say mine is correct, only that it works in my production enviroment. |
| 13:25.07 | bkw_ | check chan_sip |
| 13:25.13 | bkw_ | how it looks for srv records |
| 13:25.16 | bkw_ | it looks for _sip._udp |
| 13:25.23 | bkw_ | channels/chan_sip.c: snprintf(service, sizeof(service), "_sip._udp.%s", peer); |
| 13:25.23 | bkw_ | channels/chan_sip.c: if (ast_get_ip_or_srv(&peer->addr, v->value, "_sip._udp")) { |
| 13:25.23 | bkw_ | channels/chan_sip.c: if (ast_get_ip_or_srv(&outboundproxyip, v->value, "_sip._udp") < 0) |
| 13:25.26 | ManxPower | bkw_: Um, I don't use DNS SRV with chan_sip. |
| 13:25.33 | bkw_ | doesn't matter its how its down |
| 13:25.39 | jaybuffet | anthm: ok thank you |
| 13:25.39 | *** join/#asterisk daXas (n=on@85.96.199.40) |
| 13:25.48 | ManxPower | bkw_: do you currently run DNS SRV records? |
| 13:25.56 | bkw_ | I will in a few hours... |
| 13:25.56 | *** join/#asterisk Hopper (n=MattH@63.174.244.174) |
| 13:26.05 | Hmmhesays | so I was doing it right, it was probably firefly doing something wrong |
| 13:26.06 | bkw_ | I'm actually setting this up today and have in the past |
| 13:26.13 | bkw_ | Hmmhesays, bet so |
| 13:26.15 | Hopper | Quick question.... does anyone here know does the mysql_vm routine addon still work in the current CVS-HEAD? |
| 13:26.29 | anthm | np |
| 13:26.30 | Hmmhesays | <PROTECTED> |
| 13:26.31 | daXas | hi all, anyone using app_switch ? ... i think it is not for CVS-HEAD .. i can not compile it .. |
| 13:26.36 | ManxPower | Hmmhesays: I suspect that, like Asterisk, firefly requires the RFC pedantic format. |
| 13:27.10 | ManxPower | The SIPura even has a config option for "require _sip._udp in DNS SRV = Yes/No" |
| 13:27.20 | Hmmhesays | yeah I'll look at the docs if they exist |
| 13:27.26 | shadebob | is it a solution to desactivate a port on a TDM400? |
| 13:27.28 | bkw_ | lame if you ask me |
| 13:27.43 | ManxPower | shadebob: zap destroy channel X |
| 13:27.54 | ManxPower | or remove the port from your config. |
| 13:28.07 | shadebob | zap destroy ... thanks |
| 13:28.22 | hellop_pda | /msg nickserv |
| 13:28.23 | ManxPower | shadebob: ya know how I found out about the options? |
| 13:28.35 | ManxPower | "help zap" |
| 13:29.02 | *** join/#asterisk squirrelv5 (n=squirrel@202.57.81.146) |
| 13:29.11 | Hopper | never mind stupid question.. blah as always docs answer question |
| 13:29.25 | Hmmhesays | bkw_ you were right about the button up too |
| 13:29.38 | bkw_ | Hmmhesays, ;) |
| 13:29.41 | ManxPower | http://www.sipura.com/Documents/faq/Section_2.html Question 12 |
| 13:30.12 | NSGN | ahh, well i finally got A@H to install on my 233 mhz machine ^_^ |
| 13:30.21 | NSGN | however i'm having NIC issues :-/ |
| 13:30.22 | syle | lol |
| 13:30.24 | bkw_ | ManxPower, thats because Sipura does it wrong. You will need those to work with everything else out there. |
| 13:30.34 | bkw_ | go read the rfc |
| 13:31.00 | ManxPower | bkw_: I didn't say it was right. I said it works in my production enviroment. |
| 13:31.13 | bkw_ | only if all clients are broken like SIPura |
| 13:31.21 | zoa2 | ah brian |
| 13:31.25 | zoa2 | you silly bastard |
| 13:31.33 | zoa2 | you didnt post something on mantis! |
| 13:31.54 | bkw_ | zoa2, I will in a few |
| 13:31.57 | bkw_ | been kinda buys boi |
| 13:32.00 | zoa2 | hehe |
| 13:32.01 | zoa2 | i can imagine |
| 13:32.07 | zoa2 | any traces of a memory leak ? |
| 13:32.21 | bkw_ | one single call.. none that I can tell |
| 13:32.28 | zoa2 | ah k |
| 13:33.06 | NSGN | whoa...i just found a crazy fix for the 3com NIC issue under A@H |
| 13:33.11 | Hmmhesays | well firefly seems to have 0 documentation |
| 13:33.25 | NSGN | disable kudzu at runtime and it works |
| 13:33.34 | bkw_ | Hmmhesays, firefly sucks in the first place |
| 13:34.49 | zoa2 | brian, did you ever try idefisk ? |
| 13:34.55 | bkw_ | for? |
| 13:35.04 | zoa2 | softphone, but windows only |
| 13:35.05 | *** join/#asterisk Nobbie1 (n=ppan@mail.allwilliams.com) |
| 13:35.10 | NSGN | eexxcellent now i can SSH into my A@H box. i'm good to go! now i just need some hardware |
| 13:35.13 | zoa2 | (but working hard on a linux native version) |
| 13:35.16 | Hmmhesays | well i'd like a client to test this out |
| 13:35.22 | NSGN | PSTN interface |
| 13:35.27 | zoa2 | Hmmhesays: go for idefisk :) |
| 13:35.33 | zoa2 | like thousands did before |
| 13:35.37 | Hmmhesays | <chuckle> |
| 13:35.41 | zoa2 | :) |
| 13:36.42 | Hmmhesays | I will |
| 13:37.43 | Nobbie1 | Hello all |
| 13:37.56 | Nobbie1 | need help with Asterisk and Broadvoice |
| 13:38.20 | Nobbie1 | have been able to get outbound calling working but incoming keeps getting a busy tone |
| 13:38.35 | Freman | I fixed my sound problem |
| 13:38.39 | Nobbie1 | I have no more hair left trying to figure this one out |
| 13:38.45 | Freman | http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20MusicOnHold&comments_threshold=0&comments_offset=0&comments_sort_mode=commentDate_desc&comments_maxComments=10&comments_parentId=202#threadId352 |
| 13:38.47 | Freman | that's rude |
| 13:39.25 | ManxPower | Freman: why is that rude? |
| 13:39.55 | NSGN | ok this is stupid. AMP wont let me log in with the default password..... |
| 13:40.04 | Freman | well, asterisk on my prototype box used mpg321 it uses mpg123 on the production box.... |
| 13:40.12 | Nobbie1 | try maint and password |
| 13:40.30 | bkw_ | mpg321 don't work right |
| 13:40.35 | ManxPower | Freman: You could have just done "make mpg123" in the source directory. |
| 13:40.48 | Freman | tis gentoo |
| 13:40.50 | Freman | you emerge (c: |
| 13:40.51 | puzzled | or use madplay |
| 13:41.01 | bkw_ | Freman, um still 0.59r is what you use |
| 13:41.02 | bkw_ | even on gentoo |
| 13:41.03 | *** join/#asterisk grolloj (n=grolloj@slim-eth0.horizonlive.net) |
| 13:41.15 | Nobbie1 | anyone running with Broadvoice? |
| 13:41.19 | bkw_ | if you use the wrong one you'll get some strange behavior |
| 13:41.22 | *** join/#asterisk Blissex (n=Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
| 13:41.33 | Freman | modifying the default moh class : default => custom:/var/lib/asterisk/mohmp3,/usr/bin/mpg123 -q -r 8000 -f 1 -b 2048 --mono -s |
| 13:41.57 | bkw_ | you can take out the -b 2048 |
| 13:42.00 | bkw_ | its not needed |
| 13:42.02 | SwK | what would be nice for MOH would be a nif |
| 13:42.07 | Freman | emerge -vp mpg123 |
| 13:42.08 | Freman | [ebuild R ] media-sound/mpg123-0.59s-r9 -3dnow -esd +mmx -nas +oss 0 kB |
| 13:42.21 | bkw_ | i'm telling you 0.59s has issues |
| 13:42.34 | Freman | of this... my ear drums are painfully aware |
| 13:42.45 | bkw_ | very loud I take it |
| 13:42.46 | bkw_ | haha |
| 13:42.56 | bkw_ | -f 1? |
| 13:43.01 | bkw_ | funny stuff |
| 13:43.04 | Freman | vailable versions: 0.59s-r8 0.59s-r9 ~0.59s-r10 |
| 13:43.10 | Freman | I don't have an r availible |
| 13:43.20 | Freman | anyway... -f1 works nice (c: |
| 13:43.22 | bkw_ | go build it |
| 13:43.25 | bkw_ | its just 1 binary |
| 13:43.33 | bkw_ | or convert your files to ulaw |
| 13:43.36 | bkw_ | like i'm gonna do |
| 13:43.41 | bkw_ | :P |
| 13:43.46 | bkw_ | and use native music on hold |
| 13:44.10 | jontow | hooray for rawplayer |
| 13:44.12 | jontow | ;) |
| 13:44.23 | bkw_ | don't even need that if they are in a native format |
| 13:44.24 | bkw_ | ;) |
| 13:45.01 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 13:46.58 | *** join/#asterisk keef (n=btktech@205.242.191.115) |
| 13:47.40 | mishehu | bah. |
| 13:47.43 | Freman | well, I got my g729 codec working... happy about that |
| 13:47.45 | mishehu | I hate mornings. |
| 13:47.52 | bkw_ | mishehu, root beer? |
| 13:48.04 | syle | put on some linken park music, then mornings are bareable |
| 13:48.04 | bkw_ | that should wake you up yo |
| 13:48.15 | mishehu | bkw_: you root your beer? wow, you must be bored. |
| 13:48.28 | bkw_ | haha |
| 13:48.30 | bkw_ | lame |
| 13:49.03 | mishehu | so according to UPGRADE.TXT in cvs head (as of last nighit), the new IAX2 dial string should be DIAL(IAX2/user@host-exten@context) ? |
| 13:49.30 | Nugget | freaky |
| 13:49.51 | Mimmus | with PRI and DID, do I need overlapdial=yes? |
| 13:49.57 | hellop_pda | freeman, notice a difference? |
| 13:49.58 | zoa2 | no you dont need it |
| 13:50.19 | Mimmus | zoa2: wiki tells yes! |
| 13:50.30 | bkw_ | mishehu, you're shitting me right? |
| 13:50.31 | Mimmus | what's the exact meaning of overlapdial? |
| 13:51.11 | *** join/#asterisk spackle (n=spackle@209.234.83.19) |
| 13:51.24 | bkw_ | mishehu, what line in upgrade.txt? |
| 13:51.35 | mishehu | sec, let me pull it up |
| 13:51.47 | mishehu | I was have zombified last n ight when I was doing this work |
| 13:52.18 | ManxPower | Mimmus: Perhaps you didn't believe me when I said you should not use overlapdial? |
| 13:52.23 | bkw_ | because if its that dumb ass format i'm going to raise a bitch fit |
| 13:52.51 | mishehu | * The naming convention for IAX channels has changed in a minor way such that the call number follows a "-" rather than a "/" character. |
| 13:53.03 | Mimmus | ManxPower: no! I leave my desk due to a technician in machine room and lost any reference |
| 13:53.13 | musical_Duck | New CVS? as of last night? Is it worth it? |
| 13:53.17 | mishehu | if I remembered how to get the current line number to display in vim, I'd let you know exact line |
| 13:53.27 | Mimmus | ManxPower: thank you for your continous help! |
| 13:53.33 | zoa2 | the wiki is wrong |
| 13:53.44 | Mimmus | zoa2: can I correct it? |
| 13:54.01 | Mimmus | the wiki is wrong and now is down |
| 13:54.06 | drray | :set number |
| 13:54.18 | bkw_ | mishehu, oh thats the inbound channel names... |
| 13:54.18 | spackle | bkw_: do you have a set of best practices for eliminating echo? I have a PRI terminating to a digium card and the remote end hears echo. |
| 13:54.22 | musical_Duck | Ja, that wiki had me runnin in circles for a while grrrr |
| 13:54.23 | bkw_ | that was done months ago |
| 13:54.46 | bkw_ | spackle, thats a good question |
| 13:55.06 | ManxPower | spackle: other than the extensive thread over the past few days on the asterisk-users mailing list? No. Pretty much everything is covered in that thread. |
| 13:55.09 | spackle | bkw_: I just wondered if you had settings you turned off or on by default. |
| 13:55.34 | spackle | Manx, I saw that, some good stuff showing up there. |
| 13:56.23 | spackle | bkw_: because you knew they do nothing or cause problems. |
| 13:57.30 | musical_Duck | So looks like I am stuck with a realtime asterisk system ATM due to time constraints. Is there a way to make asterisk cache all realtime configs? |
| 13:58.26 | DrWho17 | musical_Duck: you can cache sip, check rtcachefriends=yes |
| 13:58.41 | DrWho17 | there are some other rtcache settings as well |
| 13:59.57 | musical_Duck | thx, everybit helps |
| 14:00.26 | atmos4 | ManxPower: btw. I have some strange behaviour: if call an ext from a sip client, then redirect the call to a zap ext, then answer the call on zap ext, then redirect callk to sip client and hangup zap phone, the zap channel doesn't get freed until i also hangup the sip phone |
| 14:00.41 | *** join/#asterisk RoyK (n=roy@213.160.242.93) |
| 14:00.47 | Hmmhesays | good lord outbound proxy setups are shit |
| 14:00.54 | spackle | bkw_: if only I'd waited until after cluecon. 8-( |
| 14:00.56 | ManxPower | atmos4: I'm sorry to hear that. I can't help. |
| 14:01.26 | atmos4 | ManxPower: ok |
| 14:02.44 | *** join/#asterisk Doofmannsgehilfe (n=chatzill@dsl-084-060-061-212.arcor-ip.net) |
| 14:03.43 | musical_Duck | Anyone ever build a web-interface for asterisk using php's SWFMovie->streamMp3 thinger? |
| 14:04.03 | musical_Duck | To listen to call recordings that is? |
| 14:04.43 | NSGN | ok, i have a small problem. i cant get my softphone (x-lite) to log onto my SIP server |
| 14:05.09 | musical_Duck | some info plz |
| 14:05.28 | NSGN | i'm using sip debug on the terminal of the * machine |
| 14:05.34 | NSGN | x-lite sends register |
| 14:05.41 | NSGN | with all applicable info |
| 14:05.49 | NSGN | and it sends 403 forbidden right back at it :-P |
| 14:05.50 | ManxPower | NSGN: WHAT is the error message? |
| 14:05.59 | ManxPower | i.e. Registration refused.... |
| 14:06.03 | NSGN | x-lite says login failed! contact administrator |
| 14:06.10 | *** join/#asterisk \etc\bin (n=squirrel@202.57.81.146) |
| 14:06.10 | musical_Duck | gtg quittin time cheers all |
| 14:06.11 | NSGN | however i'm using the right password..... |
| 14:06.23 | NSGN | i just remade the extension several times to be sure |
| 14:06.24 | ManxPower | NSGN: I don't care. I care about what's on the Asterisk CLI. The Registration refused message. |
| 14:06.43 | *** part/#asterisk \etc\bin (n=squirrel@202.57.81.146) |
| 14:06.58 | NSGN | i dont see registration refused |
| 14:07.29 | ManxPower | NSGN: maybe you need to turn off sip debug |
| 14:07.29 | NSGN | i just see it receives the whole request, then sends back the info, but at the top of it it attaches 403 forbidden |
| 14:07.35 | NSGN | ok, its off |
| 14:07.38 | NSGN | shall i relogin> |
| 14:07.44 | NSGN | *? from xlite? |
| 14:07.55 | *** join/#asterisk huehner (n=huehner@kt-gmbh.de) |
| 14:08.06 | ManxPower | No idea. We need to see the registration refused message that will print on the Asterisk console. |
| 14:08.18 | ManxPower | Now, if you are trying to make a CALL then that's not a registration. |
| 14:08.23 | NSGN | well, the asterisk console prints nothing when i try to log in |
| 14:08.31 | NSGN | im not calling, i cant even get x-lite to say logged in |
| 14:08.32 | ManxPower | define "log in" |
| 14:08.44 | NSGN | opening the app with the proper extension number and password in it's settings |
| 14:08.54 | NSGN | and waiting for it to say logged in in it's main window |
| 14:08.57 | ManxPower | NSGN: That's unfortunate, since the Registration refused message will contain the exact information to fix your problem. |
| 14:09.18 | NSGN | well...why the heck wont it display? i kicked up the verbos mode to 4 |
| 14:09.33 | ManxPower | NSGN: I don't know. |
| 14:09.39 | NSGN | will it not show up on a SSH connection? should i go look at the actual screen of the machine? |
| 14:09.43 | ManxPower | I start use "asterisk -rvvv" and it works just fine. |
| 14:10.04 | *** join/#asterisk Cheetah (i=bbense@secure.bense.de) |
| 14:10.06 | Cheetah | heya |
| 14:10.25 | ManxPower | NSGN: *sigh* Put the output of the sip debug on pastebin.ca |
| 14:10.33 | NSGN | ok |
| 14:10.37 | Nobbie1 | anyone running with Broadvoice? |
| 14:10.44 | Nobbie1 | I have no more hair left trying to figure this one out |
| 14:10.48 | Nobbie1 | have been able to get outbound calling working but incoming keeps getting a busy tone |
| 14:10.50 | ManxPower | then paste the section of sip.conf for the sevice. |
| 14:10.52 | ManxPower | device |
| 14:10.54 | bkw_ | FYI asterisk fuckign blowws |
| 14:10.57 | bkw_ | GOD DAMN PILE OF SHIT |
| 14:11.08 | ManxPower | Nobbie1: Well without the console error messages nobody can help you |
| 14:11.11 | bkw_ | no audio passing pile of shit |
| 14:11.17 | NSGN | http://pastebin.ca/21229 |
| 14:12.03 | fugitivo | anyone is using the atcom at-320 ip phone? |
| 14:12.21 | Cheetah | i've got another question :D |
| 14:12.22 | Nobbie1 | ok thanx for the response ManxPower...wanted a response before i entered that |
| 14:12.40 | Mimmus | can I define a voice mailbox without defining an extension? |
| 14:12.45 | Cheetah | is it possible to "break" into an existing channel/conversation with another phone? |
| 14:12.47 | NSGN | manxpower: you'll have to help me get to the sip.conf file. i'm an A@H user, a linux newbie |
| 14:12.50 | NSGN | where is it located/ |
| 14:12.51 | NSGN | *? |
| 14:12.54 | Cheetah | like two people are talking and someone wants to join in |
| 14:13.01 | ManxPower | /etc/asterisk/sip.conf |
| 14:13.05 | Cheetah | i dont want to do this with conferences room |
| 14:13.08 | Cheetah | +s |
| 14:13.17 | NSGN | ok, just a moment |
| 14:13.40 | NSGN | what is a text editor i can open it in that is built into centos? |
| 14:14.06 | ManxPower | NSGN: I don't provide OS help. |
| 14:14.12 | NSGN | (sorry for my raw n00bness...i'm a unix guy, but am new to all this linux stuff) |
| 14:14.18 | ManxPower | NSGN: If you have not set up the device in sip.conf then it will not work. |
| 14:14.32 | *** join/#asterisk marv[work] (n=timr@border0hsv.asterisksgi.com) |
| 14:14.38 | ManxPower | NSGN: what would you use to edit a text file on Unix? |
| 14:14.40 | NSGN | i used AMP to set it up...but regardless i need to learn to edit the files by hand |
| 14:15.07 | ManxPower | NSGN: Sorry, I can't help with AMP problems. |
| 14:15.10 | NSGN | pico maybe |
| 14:15.26 | NSGN | but pico doesnt seem to be on this linux build |
| 14:15.32 | ManxPower | NSGN: how about vi. |
| 14:15.39 | ManxPower | That's what unix guys use. |
| 14:15.39 | NSGN | hm, that worked |
| 14:15.42 | ManxPower | or "less" |
| 14:15.43 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfil2.dialup.mindspring.com) |
| 14:15.48 | ManxPower | or "cat" |
| 14:16.04 | NSGN | here comes that file |
| 14:16.08 | spackle | .... waiting for emacs..... |
| 14:16.21 | ManxPower | NSGN: if you flood the channel with the info I will put you on /ignore. |
| 14:16.21 | NSGN | http://pastebin.ca/21230 |
| 14:16.38 | NSGN | ...thus my pastebin link |
| 14:16.39 | ManxPower | NSGN: Sorry, I cannot help you with AMP configs. |
| 14:16.50 | ManxPower | Best of luck getting it fixed. |
| 14:16.58 | NSGN | then help me config it for real. i dont care about amp, i just didnt know any other way. i'll do it normally. i need to learn to anyway |
| 14:17.10 | ManxPower | NSGN: you need to learn Linix first. |
| 14:18.09 | NSGN | i can stumble around decently enough (especially with google standing by, heh) if you dont wish to help just say so. i'm sorry, i'm new and i came here to get a little help. i had a rough ride even getting this stuff installed because of hardware issues |
| 14:19.35 | Nobbie1 | ok everyone |
| 14:20.40 | *** join/#asterisk santiago (n=santiago@63.245.86.163) |
| 14:21.08 | *** join/#asterisk edwin_ (i=edwin@252-131-222-203.rev.techex.net.au) |
| 14:21.10 | Nobbie1 | here's the debug maybe you can see what i am missing |
| 14:21.19 | Nobbie1 | Sip read: |
| 14:21.19 | Nobbie1 | INVITE sip:3473420008@192.168.1.200:5060 SIP/2.0 |
| 14:21.19 | Nobbie1 | Call-ID: ff027d-4b@192.168.1.200 |
| 14:21.19 | Nobbie1 | CSeq: 1 INVITE |
| 14:21.19 | Nobbie1 | From: "New York NY"<sip:2126555478@192.168.1.200;user=phone>;tag=pstv |
| 14:23.04 | fugitivo | flash doesn't work in my atcom 320 |
| 14:24.13 | Mimmus | ManxPower: if I comment out overlapdial=yes, outgoing calls dont't work anymore! |
| 14:24.20 | *** part/#asterisk santiago (n=santiago@63.245.86.163) |
| 14:24.28 | *** join/#asterisk endom0rph (n=endom0rp@zerogravity.plus.com) |
| 14:24.35 | ManxPower | Mimmus: you prolly have pridialplan= something other than unknown |
| 14:25.07 | Mimmus | ManxPower: yes, I have pridialplan=local |
| 14:25.16 | Mimmus | ManxPower: wrong? |
| 14:25.47 | atmos4 | or try switching immediate |
| 14:26.02 | ManxPower | atmos4: that will make everything fail. |
| 14:26.10 | ManxPower | Mimmus: Well it might be correct if you only ever made local calls. |
| 14:26.57 | ManxPower | What part of "; PRI Dialplan: Only RARELY used for PRI." in zapata.conf.sample was unclear? |
| 14:27.16 | Mimmus | ManxPower: this is waht happens when you collect info from many sources... noone is authoritative |
| 14:27.21 | ManxPower | comment out PRI dialplan. |
| 14:27.41 | ManxPower | Mimmus: generally the *.sample files that come with your verison of Asterisk are where you want to look. |
| 14:28.18 | ManxPower | atmos4: Have you read the zapata.conf.sample comments for immediate=? |
| 14:28.31 | Mimmus | ManxPower: I think always that they are tailored for USA (I'm in Italy) |
| 14:28.52 | ManxPower | Mimmus: You can argue, or you can try it. |
| 14:29.20 | ManxPower | And yes, sometimes you need prididialplan and prilocaldialplan for non-USA, but it still seems fairly uncommon. |
| 14:29.51 | ManxPower | and immediate=yes will break incoming PRI calls. |
| 14:30.02 | Mimmus | ManxPower: I will try when current users hanghup... and I can restart Asterisk |
| 14:30.06 | ManxPower | overlapdial can break outgoing calls (but not all them) |
| 14:33.03 | Cheetah | hmm |
| 14:34.40 | ManxPower | lathos42: money |
| 14:34.45 | *** join/#asterisk Defraz_ (n=t0tal@24-119-12-238.cpe.cableone.net) |
| 14:35.12 | *** join/#asterisk johnm-work (n=johnm@wormhole.domicilium.com) |
| 14:35.15 | *** part/#asterisk johnm-work (n=johnm@wormhole.domicilium.com) |
| 14:35.36 | ManxPower | I want a nap |
| 14:35.38 | lathos42 | ManxPower: Naah, Money doesnt bleed enough to be a good sacrifice |
| 14:36.48 | ManxPower | lathos42: it's always worked for me. Post bounty, get code, pay money. |
| 14:36.53 | ManxPower | Quite simple, really. |
| 14:38.01 | Hmmhesays | I got my first real 6 string, bought it at the 5 and dime, played it till my fingers bled, was the summer of 69 |
| 14:38.27 | ManxPower | Apparently Hmmhesays is bored too. |
| 14:38.32 | spackle | make it stop |
| 14:38.38 | ManxPower | I'll be bored until my two cisco routers arrive this afternoon |
| 14:38.52 | *** join/#asterisk file[lap1op] (n=jcolp@mctnnbsa30w-156034035250.nb.aliant.net) |
| 14:38.53 | Hmmhesays | naw just waiting for a unit to reboot |
| 14:38.59 | Hmmhesays | and grab a new config file |
| 14:39.03 | spackle | manx: and then you will be nickled and dimed-to-death |
| 14:39.15 | ManxPower | spackle: I doubt it. |
| 14:39.19 | *** join/#asterisk mariogamboa (n=sudaikdd@201.138.152.159) |
| 14:39.28 | mariogamboa | hi all |
| 14:39.32 | ManxPower | spackle: I already manage about 15 Cisco routers. |
| 14:39.38 | mariogamboa | i have a little question |
| 14:40.15 | ManxPower | Ugh. I have to go to New Orleans on monday. |
| 14:40.19 | zoa2 | i got my first real sex dream, ... |
| 14:40.34 | mariogamboa | in avaya i can make a cor for restrict the users to make calls with account code in asterisk how i can assign the account code to make only local or local cellular o local cellullar and international |
| 14:40.36 | Hmmhesays | congrats |
| 14:40.45 | spackle | zoa: wrong channel |
| 14:41.29 | ManxPower | mariogamboa: you would do that using contexts, Authenticate, and your dialplan. |
| 14:41.38 | mariogamboa | yep |
| 14:41.47 | ManxPower | Or you can just have your LD carrier require an auth/account code and make your life much simplier. |
| 14:41.57 | mariogamboa | nop |
| 14:42.09 | *** join/#asterisk file[lap1op] (n=jcolp@mctnnbsa30w-156034035250.nb.aliant.net) |
| 14:42.25 | *** join/#asterisk jero (n=jero@savoirfairelinux.net) |
| 14:42.30 | mariogamboa | i need the account code for example jim have permision to make call only local susan have permision to make local and cellular |
| 14:43.04 | mariogamboa | but when jim put your account code he can make too cellular call |
| 14:43.41 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
| 14:44.03 | ManxPower | mariogamboa: you have a dialplan problem. |
| 14:44.14 | mariogamboa | maybe |
| 14:44.35 | fulgas | anyone knows if ser supports clustering ? |
| 14:44.52 | crash3m | define 'supports clustering' |
| 14:44.55 | mog_home | i dont know fulgas, but ser can support like 10,000 calls |
| 14:45.00 | mog_home | you need more? |
| 14:45.22 | *** join/#asterisk NSGN (n=NSGN@cpe-66-69-197-25.austin.res.rr.com) |
| 14:45.39 | fulgas | yes |
| 14:45.46 | fulgas | otherwise i wont ask :) |
| 14:45.51 | NSGN | god i'm a noob. ok, so i've set up two extensions on my test machine. i called one from the other and left some voicemail. now how in the heck do i check the voicemail?! |
| 14:46.04 | *** join/#asterisk edwin__ (i=edwin@252-131-222-203.rev.techex.net.au) |
| 14:46.06 | mog_home | well maybe, but a lot of people come in here askign for things they dont need |
| 14:46.09 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-218-206.rockynet.com) |
| 14:46.13 | mog_home | you need voicemailmain nsgn |
| 14:46.22 | fulgas | oks :) |
| 14:46.23 | mog_home | exten => 1000,1,voicemailmain |
| 14:46.24 | drray | NSGN extension.conf |
| 14:46.32 | mog_home | is an example |
| 14:46.40 | mog_home | put it in extension.conf |
| 14:46.50 | fulgas | need it to a new voip provider |
| 14:47.02 | NSGN | ok, so with that in there one would dial 1000 to check their own voicemail? |
| 14:47.10 | mog_home | right |
| 14:47.15 | mog_home | or any ones |
| 14:47.19 | mog_home | if they knew the pass |
| 14:47.23 | NSGN | ok |
| 14:47.26 | mog_home | do a show application voicemailmain |
| 14:47.29 | mog_home | in asterisk |
| 14:47.30 | NSGN | hmm...i wonder why A@H doesnt come with that line in the file |
| 14:47.33 | NSGN | seems like they should have |
| 14:47.34 | NSGN | *shrug* |
| 14:47.35 | mog_home | probably does |
| 14:47.37 | drray | or caller ID |
| 14:47.44 | mog_home | do a grep for voicemailmain |
| 14:47.46 | *** join/#asterisk Doofmannsgehilfe (n=chatzill@dsl-084-060-061-212.arcor-ip.net) |
| 14:47.47 | NSGN | well dialing 1000 is not a valid extension |
| 14:47.49 | NSGN | ok |
| 14:47.49 | mog_home | youll probably find it |
| 14:47.53 | drray | you have to add it |
| 14:47.53 | mog_home | 1000 was an example |
| 14:47.54 | mariogamboa | mm |
| 14:47.57 | mog_home | you can make it any number |
| 14:47.58 | NSGN | got ya |
| 14:48.01 | NSGN | *goes grepping* |
| 14:48.01 | drray | on mine it's 6 |
| 14:48.09 | mog_home | and at work its 8500 |
| 14:48.17 | mariogamboa | how i can make a good dialplan for restrict users to make calls |
| 14:48.19 | mariogamboa | any info |
| 14:48.19 | mog_home | you can have up to like a 2000 line extension |
| 14:48.28 | mog_home | mario contexts are your friend |
| 14:48.53 | mariogamboa | yep i know |
| 14:49.11 | mog_home | just set up a context for each level of user |
| 14:49.18 | mog_home | then default those users to those levels |
| 14:49.21 | mariogamboa | but i don't know how assign to context the level |
| 14:49.31 | mog_home | the level is in your mind |
| 14:49.33 | mariogamboa | or if i can with callgroup |
| 14:49.35 | mog_home | not a real thing |
| 14:49.47 | mog_home | have context for each type of dialing right |
| 14:49.53 | mariogamboa | yep |
| 14:49.55 | mog_home | then create contexts for each group |
| 14:49.57 | mariogamboa | is right |
| 14:50.09 | drray | then you can include them in the default to nest them if needed |
| 14:50.10 | mog_home | then include those contexts for each group accordingly |
| 14:50.25 | NSGN | hey where is that extensions file? |
| 14:50.30 | mog_home | and just tell each sip, zap ,iax channel |
| 14:50.32 | NSGN | like, where is it located i mean |
| 14:50.35 | mog_home | /etc/asterisk |
| 14:50.38 | NSGN | ah, k |
| 14:50.40 | drray | /etc/asterisk/extension.conf |
| 14:50.41 | mog_home | in asterisk@home you have 3 |
| 14:50.48 | mariogamboa | but the group how i can make |
| 14:50.50 | mog_home | as they include things on top of things |
| 14:51.00 | mariogamboa | i need some example |
| 14:51.02 | mog_home | what do you mean grouip mario |
| 14:51.12 | mog_home | well for that you would have to call me at digium |
| 14:51.16 | drray | do a google for hitchhikers guide to asterisk |
| 14:51.18 | ManxPower | Wow, lots of people are /msg'ing me today and getting automatically put on /ignore. |
| 14:51.21 | mog_home | and go through our express support |
| 14:51.28 | mog_home | system |
| 14:52.14 | mog_home | happy to give advice, but doing it for you |
| 14:52.30 | mog_home | i just cant do |
| 14:54.51 | syle | hey mog |
| 14:54.54 | mog_home | hey |
| 14:54.56 | syle | you play final fantasy 11? |
| 14:54.56 | mog_home | syle |
| 14:55.05 | mog_home | nope |
| 14:55.13 | mog_home | pso fan myself |
| 14:55.24 | mog_home | never liked any ff games after 3 |
| 14:55.28 | NSGN | ok i found a line referring to voicemailmain.....exten => a,1,VoiceMailMain(${ARG1}) |
| 14:55.34 | syle | just wondering, i visited my mog home so many times in that game hehe |
| 14:55.54 | mog_home | heh |
| 14:56.00 | mog_home | nsgn thats not it |
| 14:56.08 | syle | i had to purposely get myself banned online so i could have a life again |
| 14:56.15 | mog_home | my friends used to give me mogs |
| 14:56.24 | mog_home | i had a big fluffy doll at one point |
| 14:56.30 | NSGN | what part of the file should it be under? there are some headings/subheadings in here |
| 14:56.58 | mog_home | to be honest nsgn this is why i stay out of asterisk@home, but here is what i would do if i were you |
| 14:57.08 | ManxPower | mog_home: These people are wanting private Asterisk consulting for free. |
| 14:57.09 | mog_home | grep VoiceMailMain * |
| 14:57.14 | mog_home | inside of asterisk@home |
| 14:57.17 | ManxPower | I laugh at them and then put them on /ignore. |
| 14:57.37 | Hmmhesays | I tell them pm's are for paid support only |
| 14:57.41 | NSGN | ok. *goes grepping..again* |
| 14:57.47 | mog_home | my rule of thumb is if i dont have to login or think i am happy to do it for free |
| 14:57.47 | Hmmhesays | then if they insist they get on ignore |
| 14:58.09 | mog_home | but anything more they should head to digium |
| 14:58.14 | zoa2 | i get people asking for free support all the time too |
| 14:58.16 | mog_home | unless its interesting then i do it |
| 14:58.18 | ManxPower | Hmmhesays: I used to do that. |
| 14:58.20 | zoa2 | they think its only 30 minutes |
| 14:58.22 | ManxPower | Now I'm just sick of it. |
| 14:58.33 | zoa2 | they forget that there are 20 people asking it at the same time |
| 14:58.40 | Hmmhesays | everyone's problem is a 'quick one' |
| 14:59.01 | mog_home | not mine, they are always huge, but motly interesting |
| 14:59.13 | mishehu | and everybody's problem is the most important |
| 14:59.27 | Hmmhesays | let me rephrase everyone says their problem is a quick one so they shouldn't have to pay |
| 14:59.36 | ManxPower | I get almost as mad when people call my cell phone. |
| 14:59.42 | mog_home | heh obviously my 1 line pbx is more important than zoas multi 1000 lined pbx |
| 14:59.49 | mog_home | mine is down NOW |
| 14:59.57 | ManxPower | Fortunatly I leave it powered off most of the time. |
| 15:00.06 | mog_home | manxpower thats why i dont have one |
| 15:00.19 | mishehu | ManxPower: my mistake was giving out my cell # when I first started consulting... I should have coughed up for the landline from the get-go. |
| 15:00.22 | ManxPower | mog_home: People are supposed to call my extension and then hit 0 if they want my cell phone. |
| 15:00.24 | *** join/#asterisk CoffeeIV_ (n=CoffeeIV@mail.airlinksystems.com) |
| 15:00.27 | mog_home | i am inredibly hard to reach if you dotn have the right channel |
| 15:00.43 | ManxPower | mishehu: get a new cell number. |
| 15:00.57 | ManxPower | my cell phone also doesn't have voicemail. |
| 15:00.59 | mishehu | ManxPower: I still get idiots calling my cell, and I let them go to the message that says "if this is a business related matter, please call XXX-XXXX", and they still leave the message on the cell! |
| 15:01.14 | *** join/#asterisk edwin_ (i=edwin@252-131-222-203.rev.techex.net.au) |
| 15:01.22 | NSGN | ok i guess i'll have to be adding an extension for voicemail. that grep string just returned nothing |
| 15:01.26 | mishehu | ManxPower: I might. am thinking of getting one of those treos, so I might tell sprint to fork over a new number. |
| 15:01.27 | Hmmhesays | some people get in a technophobe frenzy and they just can't think |
| 15:01.32 | mog_home | nsgn can i login for a sec |
| 15:01.35 | mog_home | ill show it to you |
| 15:01.44 | zoa2 | i never want to talk to people on the phone :) |
| 15:01.50 | Hmmhesays | me either, i hate it |
| 15:01.54 | zoa2 | they could hunt you down forever |
| 15:01.56 | mishehu | yeah, nsgn, how about free root for everyone? ;-) |
| 15:02.01 | zoa2 | especially since most people are from the states here |
| 15:02.03 | zoa2 | and im not |
| 15:02.09 | zoa2 | and then they start calling you in the middle of the night |
| 15:02.13 | zoa2 | for free support |
| 15:02.15 | zoa2 | no thanks |
| 15:02.16 | Hmmhesays | i don't want to talk to people in the states either |
| 15:02.16 | mog_home | welll nsgn you could message me... |
| 15:02.18 | ManxPower | I did tech support at major software vendors for about 6 years. I hate the phone. |
| 15:02.23 | mishehu | zoa2: isn't that what IVR's are for? |
| 15:02.24 | mog_home | as that would be less public |
| 15:02.25 | *** join/#asterisk rking (n=rking@ip68-105-231-56.lu.dl.cox.net) |
| 15:02.27 | NSGN | heh, i dont think my router is set up to let external SSH come in ;-P |
| 15:02.44 | mog_home | well how about you post your extensions some where |
| 15:02.46 | ManxPower | Get your average call time down! Get the queue wait times down! |
| 15:02.48 | mog_home | ill show it to you |
| 15:03.08 | mog_home | heh /me is glad digium is not like that |
| 15:03.09 | mishehu | ManxPower: never "Help them fix their problem, even though we know they're dumb as mud!" |
| 15:03.10 | NSGN | ok, you want the whole extensions file or just a part or what? |
| 15:03.14 | ManxPower | Finally one April Fools day I hacked the reports and made my average call time 10 seconds and too 20x more calls than all the other techs. |
| 15:03.20 | mog_home | all 3 or 4 of them |
| 15:03.29 | mog_home | heh |
| 15:03.29 | ManxPower | They didn't whine quite as much after that. |
| 15:03.45 | mishehu | ManxPower: they feared you after that |
| 15:03.46 | mishehu | heh |
| 15:03.49 | mog_home | reminds of me of summer job at mcdonalds |
| 15:03.57 | Hmmhesays | fsck phone support, unless you are a friend I don't want to talk to you |
| 15:03.57 | mog_home | avg serving time is 56 seconds |
| 15:04.05 | mog_home | by end of summer 15 seconds |
| 15:04.12 | *** join/#asterisk pussfeller (n=todd@12.150.129.171) |
| 15:04.16 | mog_home | such a lame job |
| 15:04.30 | mishehu | Hmmhesays: and if your favorite actor/actress suddenly calls up to ask you out on a date? |
| 15:04.35 | ManxPower | mog_home: You started each conversation with "Welcome to McDonalds, geeze you're fat! |
| 15:04.44 | CoffeeIV_ | Is the asterisk database -- not the mysql used by AMP, but the one whose contents are shown by "database show" at the CLI> prompt -- built from a text file that I could have scripts edit ? |
| 15:05.00 | ManxPower | CoffeeIV_: no it's a Berkley DB |
| 15:05.10 | mog_home | no "welcometomcdonaldshowcanihelpyoutodaywouldyoulikefrieswiththatthanksgoodbye" < 5seconds |
| 15:05.26 | Nugget | ding fries are done |
| 15:05.47 | mishehu | mog_home: if you were in chicago still, you'd have to use a mexican accent for that |
| 15:05.58 | *** join/#asterisk criptos (n=criptos@201.145.229.183) |
| 15:06.04 | criptos | hi |
| 15:06.05 | mog_home | heh |
| 15:06.11 | CoffeeIV_ | ManxPower: thanks -- any idea what file holds the Berkley DB ? I know how to handle those -- I probably should interface to it through asterisk, but I'd like to at least make sure it is backed up |
| 15:06.16 | mog_home | i learned maybe 30 languages working there |
| 15:06.33 | mog_home | it amazes me how few people speak english in the south |
| 15:06.35 | Hmmhesays | mishehu: is she naked? |
| 15:06.39 | mishehu | mog_home: I'm sure one of those languages was bad english |
| 15:06.51 | mog_home | 20 of them where southern one word |
| 15:06.58 | mishehu | Hmmhesays: she's on the phone, unless it's a vidphone, would it matter if she's nekkid? |
| 15:07.02 | mog_home | you take all words and make them one long long word |
| 15:07.27 | mishehu | hell, if natali portman called me up to ask me out on a date, I don't think I'd ask her "btw, are you naked?" |
| 15:07.40 | mog_home | heh |
| 15:07.49 | mishehu | i'd probably just say "betaH" (means "of course") |
| 15:08.09 | mishehu | mog_home: did you go down to the 'hood to learn ebonics? |
| 15:08.17 | ManxPower | CoffeeIV_: look in /var/lib/asterisk |
| 15:08.20 | mog_home | i got that to |
| 15:08.23 | mog_home | and spanglish |
| 15:08.26 | mog_home | ukranish |
| 15:08.35 | NSGN | mog_home: i'll get them files in a few. client on the phone |
| 15:08.39 | mog_home | and many other english odd language i dont speak combos |
| 15:08.45 | mog_home | okies |
| 15:08.52 | criptos | Hi, I Have random hangups, only in long distance incoming calls, any ideas? |
| 15:09.06 | CoffeeIV_ | ManxPower: /var/lib/asterisk/astdb sounds very suspiciously named -- thanks a million man |
| 15:09.07 | *** join/#asterisk pc4 (n=pc@209.151.52.81) |
| 15:09.11 | ManxPower | criptos: set busydetect=no and callprogress=no |
| 15:09.12 | eldu | hello is there a way to force an incoming comm to use a specific codec ? |
| 15:09.24 | mog_home | "looks foggy ask again later" |
| 15:09.29 | ManxPower | eldu: yes. |
| 15:09.35 | mog_home | can you give more info criptos |
| 15:09.41 | pc4 | What other provides provide 800 # incoming DIDs, unlimited simultanous calls (per minute), iax, and let you set your own CID? |
| 15:09.47 | Hmmhesays | mishehu: i have an active imagination |
| 15:09.50 | criptos | What more info you whant mog_home? |
| 15:10.01 | ManxPower | pc4: almost all carriers. |
| 15:10.01 | rking | criptos: i think they're saying you're bying cryptic |
| 15:10.13 | mishehu | Hmmhesays: so do I, but isn't that what the internet is for - dl'ing nekkid pics of your favorite star? |
| 15:10.22 | pc4 | ManxPower - Well, most consumer oriented ones like broadvoice don't. |
| 15:10.30 | *** join/#asterisk newl (n=newlook@203-59-187-240.dyn.iinet.net.au) |
| 15:10.37 | mog_home | what kind of lines you are running, how these calls come in, when they disconnect, are you sure its only long distance etc |
| 15:10.38 | mishehu | I couldn't find much else useful on this new fangled information superhighway |
| 15:10.38 | ManxPower | But why do you want to set your own caller id for INCOMING calls? |
| 15:10.45 | eldu | ManxPower: :) i mean for exemple 10 telco numbers on a sip trunk, can i force a codec to be used by only one of my telco number ? |
| 15:10.49 | mog_home | saying my lines get hangups is increadibly vague |
| 15:10.54 | pc4 | ManxPower - For outgoing calls. |
| 15:10.54 | ManxPower | eldu: yes. |
| 15:11.00 | pc4 | ManxPower - I'm trying to find somethign with decent voice quality... nufone stinks there. my stanaphone account sounds 10 times better. |
| 15:11.06 | ManxPower | pc4: you only specified incoming 800 DID. |
| 15:11.08 | eldu | ManxPower: any tips to do that ? |
| 15:11.09 | criptos | ManxPower, I`m using groundstart fxo signalling, I have 8 analog ptsn lines... And have no disconection announcement, why should be work better without busydetect instead using it? |
| 15:11.16 | ManxPower | pc4: I use NuFone and Teliax |
| 15:11.28 | mog_home | you in the states criptos |
| 15:11.40 | ManxPower | criptos: busydetect and callprogress don't work well and will cause random hangups. |
| 15:11.42 | pc4 | many - Is the quality on nufone marginal for you? |
| 15:11.51 | ManxPower | criptos: are you SURE you have groundstart lines and not loopstart lines? |
| 15:11.55 | mog_home | meh busydetect isnt bad |
| 15:11.56 | ManxPower | pc4: no. |
| 15:12.00 | *** join/#asterisk libpcp (n=libpcp@cm15.omega16.maxonline.com.sg) |
| 15:12.01 | *** join/#asterisk Roots (i=olsen@adsl-66-143-177-197.dsl.austtx.swbell.net) |
| 15:12.04 | mog_home | and manx is right on the groundstart part |
| 15:12.06 | libpcp | hi all |
| 15:12.14 | Hmmhesays | misheshu: my favorite stars are already nekkid in movies |
| 15:12.19 | libpcp | is anyone using sangoma card connected to an e1? |
| 15:12.21 | criptos | No, I in mexico... I have loopstart, not groundstart |
| 15:12.22 | ManxPower | eldu: well disallow=all and allow=thecodecyou want for the [provider] section of sip.conf. |
| 15:12.23 | mog_home | what kind of car are you using? |
| 15:12.31 | mog_home | criptos what card ? |
| 15:12.33 | criptos | tdp400p |
| 15:12.38 | blitzrage | reminder: TAUG meeting tonight at 7:30pm at Toby's Good Eats |
| 15:12.40 | ManxPower | <criptos> ManxPower, I`m using groundstart fxo signalling, |
| 15:12.40 | criptos | tdm400p |
| 15:12.41 | mog_home | tormenta2? |
| 15:12.42 | blitzrage | ~taug |
| 15:12.43 | jbot | hmm... taug is The Toronto Asterisk Users group. The website can be found at http://www.taug.ca |
| 15:12.50 | mishehu | Hmmhesays: ah, those kind of stars |
| 15:12.50 | ManxPower | If you have loopstart lines, why are you using groundstart signaling in Asterisk? |
| 15:12.50 | criptos | sorry, is loopstart :) |
| 15:12.51 | pc4 | ManxPower - call 1-347-534-2667 and dial extension 1001... listen to the quality. then call 866-249-2403 and do the same. 866 is nufone #. It sounds like crap. |
| 15:12.53 | mog_home | or tdm400p |
| 15:12.57 | zoa2 | bkw, if you set the jitter buffer to a negative size, maybe it will go faster :) |
| 15:12.58 | mishehu | stars of moan |
| 15:12.58 | criptos | My mistake... |
| 15:12.59 | ManxPower | try using fxs_ks |
| 15:13.05 | zoa2 | you need to use a negative buffer |
| 15:13.06 | *** join/#asterisk edwin_ (i=edwin@252-131-222-203.rev.techex.net.au) |
| 15:13.06 | mog_home | yeah bingo |
| 15:13.13 | libpcp | is sangoma uses zap channel ? |
| 15:13.14 | ManxPower | pc4: maybe it's your codec? |
| 15:13.14 | bkw_ | haha |
| 15:13.16 | zoa2 | than it will send you the packets earlier than they arrived |
| 15:13.21 | eldu | ManxPower: yup i did that, it works fine, but with this [provider] i got many telco num, i want them all use one codec and only one or two of them to use another codec |
| 15:13.22 | zoa2 | but dont overdo it |
| 15:13.29 | zoa2 | or you will know what the guy says before he said it |
| 15:13.32 | pc4 | ManxPower - g711u -- but it sounds like gsm. |
| 15:13.37 | criptos | So, I try to use fks_ks and without busydetect and callprogress? |
| 15:13.38 | ManxPower | eldu: you can't set the codec on a per incoming call basis |
| 15:13.40 | pc4 | ManxPower - Unless nufone is getting those dids via gsm |
| 15:13.45 | ManxPower | criptos: yes. |
| 15:13.56 | ManxPower | pc4: no, NuFone's DIDs come in via PRI |
| 15:13.57 | criptos | Well, lets try that... |
| 15:13.59 | eldu | ManxPower: exactly what i wanted |
| 15:14.01 | mog_home | never ever use busydetect and callprogress |
| 15:14.01 | mishehu | bah. need to put things back togthere here... rearranged everything... blah. |
| 15:14.02 | eldu | :( |
| 15:14.03 | mog_home | its either or |
| 15:15.05 | ManxPower | eldu: get 2 accounts with different userids and passwords. But some DIDs on one account and some on the other account. |
| 15:15.30 | eldu | ManxPower: yes i thought i will do like that |
| 15:15.41 | ManxPower | you CAN specific the codec on a per accuont basis. This assumes that your provider authenticates as a different user for different accounts when they send calls to you. Not all of them do. |
| 15:16.27 | criptos | Done. |
| 15:16.51 | criptos | channles are now fxs_ks at zaptel and zapata.conf and there is no support for busydetect or callprogress... |
| 15:18.15 | eldu | ManxPower: interesting, ill ask them asap, coz atm i dont use any kind of auth |
| 15:18.26 | *** join/#asterisk |cleric| (n=dacleric@p54828E6F.dip0.t-ipconnect.de) |
| 15:18.27 | eldu | just ip to ip sip trunk |
| 15:19.06 | libpcp | maybe someone can answer my question. i just want to know if sangoma card is using zaptel channel in asterisk, because show channels doesnt show zaptel channels |
| 15:19.21 | mog_home | it would show it as taht |
| 15:19.31 | mog_home | sangoma piggybacks off the zaptel stuff |
| 15:19.37 | mog_home | and zapata |
| 15:19.43 | mog_home | you probably didnt config it |
| 15:19.57 | ManxPower | libpcp: the drivers for sangoma are zaptel compatable. |
| 15:20.06 | libpcp | so how can i use the sangoma card to dialout? |
| 15:20.14 | mog_home | you have to configure it |
| 15:20.16 | mog_home | for asterisk |
| 15:20.22 | mog_home | you havent |
| 15:20.22 | ManxPower | libpcp: In Asterisk it's exactly the same as a Digium card. |
| 15:20.28 | mog_home | exactly |
| 15:20.33 | ManxPower | No idea how you set up the drivers, you would have to talk to Sangoma about that. |
| 15:20.37 | mog_home | but if you dont configure zaptel or zapata |
| 15:20.44 | libpcp | yeah i configured it but what about in the exten part. should it be Zap/1-x ? |
| 15:20.53 | mog_home | zap/g1 |
| 15:20.56 | mog_home | or whatevre |
| 15:21.01 | mog_home | but if you did zap show channels |
| 15:21.03 | ManxPower | libpcp: Dial(Zap/g1/5551212) just like ANY OTHER zap dial |
| 15:21.06 | mog_home | and nothing showed up |
| 15:21.10 | mog_home | asterisk doesnt know about it |
| 15:21.42 | ManxPower | libpcp: you did not configure /etc/zaptel.conf or /etc/asterisk/zapata.conf |
| 15:22.12 | libpcp | if i do zap show channels it says no such command zap |
| 15:22.20 | mog_home | then you dont have chan_zap |
| 15:22.23 | mog_home | in asterisk |
| 15:22.28 | ManxPower | libpcp: then you did not install zaptel before building Asterisk |
| 15:22.34 | *** join/#asterisk DirtyD (n=rob@ool-18bce078.dyn.optonline.net) |
| 15:22.40 | mog_home | make sure you built zaptel, libpri,asterisk in that order |
| 15:22.43 | DirtyD | hihihi |
| 15:22.47 | mog_home | hi |
| 15:22.52 | libpcp | ManxPower: i did. that the first thing i did before installing asterisk from source |
| 15:22.55 | ManxPower | mog_home: WOW! You mean just like the docs?!?! |
| 15:23.03 | DirtyD | how is the world of opensource voice recognition for Asterisk coming along? |
| 15:23.06 | mog_home | omg yes, arent they wonderful |
| 15:23.10 | ManxPower | libpcp: then asterisk did not see zaptel.. |
| 15:23.21 | mog_home | good dirtyd |
| 15:23.30 | ManxPower | If "show modules" does not show chan_zap.so then asterisk is not built for zap |
| 15:23.37 | libpcp | well i should try to recompile the asterisk again |
| 15:23.38 | greg_work | quick poll: TIA 568A or B? |
| 15:23.44 | mog_home | could these docs be at http://www.asterisk.org/download |
| 15:23.51 | mog_home | answers point to yes |
| 15:24.06 | tzanger | greg_work: if you're in Canada? A. If you're in US: B |
| 15:24.22 | DirtyD | mog_home: what should I be researching to get voice recognition up and running with Asterisk. |
| 15:24.28 | criptos | is there any document, about the architecture of the iax_channel? I need callpickup at iax, after searching, now I belive that the easiest way is to code it.. |
| 15:24.34 | mog_home | it already is there dirtyd |
| 15:24.36 | ManxPower | greg_work: for what? |
| 15:24.37 | mog_home | go to voipinfo |
| 15:24.43 | DirtyD | ok thanks. |
| 15:24.43 | mog_home | search for sphynx |
| 15:24.44 | greg_work | ethernet |
| 15:24.52 | mog_home | or is it sphinx |
| 15:24.55 | mog_home | i cant remeber |
| 15:24.55 | bweschke | Olle Johansson? are you on this channel? |
| 15:24.59 | criptos | but I have found chan_iax2.c quite "dark" and undocumentend... |
| 15:25.05 | mog_home | bah |
| 15:25.09 | DirtyD | thanks, man. |
| 15:25.10 | mog_home | code is self documenting |
| 15:25.12 | zoa2 | olle is not here i think |
| 15:25.16 | zoa2 | his name is oej here |
| 15:25.21 | mog_home | but he is not... |
| 15:25.31 | tzanger | no code is self-documenting, don't kid yourself |
| 15:25.32 | bweschke | k. thanks. trying to test out patch 3644 with a snom 360 here |
| 15:25.39 | mog_home | have you read asteirsk |
| 15:25.45 | ManxPower | greg_work: B is for Ethernet |
| 15:25.49 | ManxPower | A is common for telecom |
| 15:25.50 | mog_home | asterisk* its as easy as quantum physics |
| 15:26.02 | criptos | Yeap, but it takes long to understand pure code, that documented code or a reference... |
| 15:26.20 | bweschke | sip show subscriptions is showing the subscription as active |
| 15:26.31 | mog_home | yeah, i can read all the chans now just fine, except chan_zap which is a blackhole to me |
| 15:26.34 | *** join/#asterisk zhamrockzz (n=libpcp@cm10.omega20.maxonline.com.sg) |
| 15:26.41 | zhamrockzz | sorry guys i got disconnected |
| 15:26.47 | criptos | well, so, I have a few questions. So far, I`m able to know if I pressed the pickupexten... |
| 15:26.59 | zhamrockzz | its weird because chan_zap is not showing in my modules |
| 15:27.00 | *** join/#asterisk hotgrits (n=hotgrits@192.160.238.156) |
| 15:27.07 | zhamrockzz | ill try to recompile asterisk again |
| 15:27.16 | criptos | And there is a ast_pickup_call(ast_peer *c) function.. |
| 15:27.16 | mog_home | eep class... |
| 15:27.25 | bweschke | but when I go offhook on the subscribed extension - I don't see any sip msgs in sip debug going to the snom phone |
| 15:27.53 | criptos | But, at iax2 everythings seems to be controlled by a ** to chan_iax2_pvt called iaxs |
| 15:27.54 | bweschke | related to the state change |
| 15:27.59 | ManxPower | bweschke: you won't until it makes a call |
| 15:28.11 | bweschke | right.. |
| 15:28.15 | ManxPower | bweschke: SIP phones collect the digits, THEN talk to the server. |
| 15:28.27 | bweschke | understand... |
| 15:28.29 | *** join/#asterisk lot (n=lot@68.148.192.184) |
| 15:28.33 | criptos | I have modified chan_iax2 to accept callpickup pickupgroup at peer definition in iax.conf.. |
| 15:28.39 | bweschke | but if my snom is 1000 |
| 15:28.50 | bweschke | and I've subscribed to 1001 with 1000 |
| 15:28.58 | criptos | but, the thing I`m been unable to do, is to find that info at the iaxs structure :( |
| 15:29.01 | criptos | any help? |
| 15:29.28 | bweschke | and then I INVITE 8500 with 1001, should I see a msg going to 1000 after the rtp streams get nailed up? |
| 15:29.28 | ManxPower | criptos: perhaps #asterisk-dev? or asterisk-dev mailing list? |
| 15:29.31 | *** join/#asterisk file (n=jcolp@mctnnbsa30w-156034035250.nb.aliant.net) |
| 15:29.52 | *** join/#asterisk Dark_ (n=humpf@200-204-38-187.dsl.telesp.net.br) |
| 15:30.31 | criptos | I will go there :) |
| 15:30.36 | *** part/#asterisk criptos (n=criptos@201.145.229.183) |
| 15:31.35 | *** join/#asterisk criptos (n=criptos@201.145.229.183) |
| 15:32.07 | criptos | evreyone seems to be dead at #asterisk-dev, has it been takeover by Umbrella corporation? |
| 15:32.34 | spackle | criptos: yeah, and what happened to the 996 conference? |
| 15:32.38 | ManxPower | criptos: if you want immediate help expect to pay for it. Otherwise be patient. |
| 15:33.26 | Mimmus | ManxPower: finally I tried without overlapdial=yes. It doesn't work. |
| 15:33.41 | criptos | manxpower, no, I dont wat immediate help, I just was expecting a sign of life :) |
| 15:33.45 | anthm | what's your question? |
| 15:34.30 | ManxPower | criptos: it's before noon EDT, they are all still asleep |
| 15:35.03 | criptos | <PROTECTED> |
| 15:35.54 | file | I'm trying to write out documentation but noooooooooo, uunet and cogent have to be bitches |
| 15:36.13 | nick125 | lol |
| 15:36.26 | nick125 | that sucks |
| 15:36.33 | ManxPower | "Closing Ticket because of no response for 3 days." |
| 15:36.56 | Juggie | dear god, they finally commited my patch :) |
| 15:37.15 | ManxPower | Juggie: how many weeks did it take? |
| 15:37.16 | nick125 | ManxPower: lol |
| 15:37.17 | file | latency here, latency there! |
| 15:37.44 | ManxPower | nick125: I'm getting tired of users having "urgent problems" and never responding to requests for more information. |
| 15:37.45 | Juggie | 5 or 6 |
| 15:37.50 | criptos | Every thing is about latency :) |
| 15:38.04 | Hmmhesays | lol |
| 15:38.22 | criptos | ManxPower, I tougth that if you help other, other will help you, and well, you need to wait :) |
| 15:38.32 | Juggie | manx, only 5-6 weeks hah |
| 15:38.38 | Juggie | it was a greuling process :) |
| 15:38.42 | eldu | ManxPower: are you still there ? |
| 15:38.43 | nick125 | ManxPower: yeah, i get what you are saying |
| 15:39.00 | ManxPower | criptos: I assume you are not longer needing developement help? anthm is trying to help, but you are ignoreing him. |
| 15:39.34 | *** part/#asterisk ManxPower (n=eric@dsl-209-16-67-160.datasync.com) |
| 15:39.51 | Mimmus | ManxPower leaved! |
| 15:39.59 | file | leaved? |
| 15:40.01 | file | left. |
| 15:40.07 | *** part/#asterisk NSGN (n=NSGN@cpe-66-69-197-25.austin.res.rr.com) |
| 15:40.07 | jontow | loved! |
| 15:40.10 | jontow | wait.. wrong movie. |
| 15:40.10 | Mimmus | ManxPower: I said that may english is very bad! |
| 15:40.25 | nick125 | :P |
| 15:40.50 | Mimmus | and here in Italy it's 17:40 friday |
| 15:41.08 | nick125 | its only 9:41am here lol |
| 15:41.16 | MmmmToop_ | <PROTECTED> |
| 15:41.21 | nick125 | it feels like 17:40 though :P |
| 15:41.22 | Mimmus | my eyes are crossed |
| 15:41.54 | MmmmToop_ | guess it it time to go home to wife....mmm |
| 15:41.57 | criptos | So, I have "issues" with chan_iax and ast_pickup_call... |
| 15:41.58 | MmmmToop_ | chat tomorrow... |
| 15:42.12 | *** part/#asterisk MmmmToop_ (n=chatzill@196.14.216.119) |
| 15:42.35 | *** join/#asterisk _T3_ (n=rposada@35.229.uio.satnet.net) |
| 15:42.38 | Mimmus | -2, -3 diopters due to Asterisk |
| 15:42.52 | criptos | As far I understand, chan_iax manages almos everything in a ** chan_iax2_pvt structure called iaxs |
| 15:43.20 | _T3_ | good morning guys |
| 15:43.29 | anthm | an array of pvts |
| 15:43.32 | anthm | yes |
| 15:43.35 | criptos | ast_pickup_call(ast_channel *c) expect ast_channel.. |
| 15:43.40 | Mimmus | _T3_: morning? |
| 15:43.52 | criptos | and the obly ast_channel structure I see at chan_iax2_pvt is owner... |
| 15:43.53 | _T3_ | sorry i just wake up |
| 15:44.33 | Mimmus | _T3_: ahhh, 17:45 in Italy: I'm dead tired |
| 15:44.51 | *** join/#asterisk Doofmannsgehilfe (n=chatzill@dsl-084-060-061-212.arcor-ip.net) |
| 15:44.58 | Hmmhesays | 10:45am in fargo and i'm not doing so hot |
| 15:45.01 | _T3_ | is friday there ? |
| 15:45.02 | anthm | owner is a pointer to the channel that the pvt is joined to |
| 15:45.13 | criptos | To make callpickup and pickupgroup, I also modifed read then from the iax.conf file, and those info is stored at a ** iax2_peer structure called peerl |
| 15:45.33 | file | peers don't use the system, users do |
| 15:45.47 | anthm | peerl is a global linked list of the peers read from the config file or realtime |
| 15:45.52 | criptos | So, Mi big big questions, is how Do i get for iaxs[] to peel... |
| 15:46.10 | anthm | that are used per call to configure newly invented pvts |
| 15:46.18 | _T3_ | ok, who use REALTIME? |
| 15:46.25 | *** join/#asterisk jets (n=b@guardian.pmt.org) |
| 15:46.47 | *** join/#asterisk bvane (n=bvane@131.93.21.11) |
| 15:47.43 | anthm | you need to add it to the code that sets up new pvt |
| 15:47.48 | *** join/#asterisk edwin__ (i=edwin@252-131-222-203.rev.techex.net.au) |
| 15:48.04 | anthm | so when one is spawned it has the peer used to create it and the new pvt in the same scope |
| 15:48.09 | criptos | thumm.... |
| 15:48.12 | anthm | you then cp it into the pvt |
| 15:48.17 | anthm | so you have it when you need it |
| 15:48.52 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 15:49.27 | criptos | at new_iax I suppose.. |
| 15:50.16 | *** join/#asterisk NSGN (n=NSGN@cpe-66-69-197-25.austin.res.rr.com) |
| 15:50.34 | NSGN | OK i'm back, and whoever was gonna help me with the extensions.conf file, i'm ready now |
| 15:50.38 | criptos | new_iax returns apointer to a chan_iax2_pvt structure,it does create a tmp, but no further info is provided about the peer, creating it. |
| 15:51.00 | bvane | NSGIN what do you need help with |
| 15:51.16 | anthm | moreso you need to deal with users |
| 15:51.21 | criptos | And there is find_callno, wow! NOW I`m getting.. |
| 15:51.22 | anthm | look at check_access |
| 15:51.36 | *** join/#asterisk Matt-E- (n=Matt-E-@66-224-125-137.atgi.net) |
| 15:51.38 | anthm | 4716 |
| 15:52.02 | anthm | that is where it copies everything from users to the new pvt |
| 15:52.16 | *** part/#asterisk Matt-E- (n=Matt-E-@66-224-125-137.atgi.net) |
| 15:52.59 | criptos | ies is the "native channel" who will get copied to the iaxs structure? |
| 15:53.07 | *** join/#asterisk aminorex (n=tony@12-23-137-226.dhcp.dlth.mn.charter.com) |
| 15:53.26 | NSGN | bvane: well i was having the darndest time figuring out what extension my voicemail was on, but i found out now it is *98 |
| 15:53.27 | _T3_ | nobody |
| 15:53.29 | NSGN | yippie |
| 15:53.29 | anthm | iaxs is an array of pbt |
| 15:53.32 | anthm | pvt |
| 15:53.42 | _T3_ | i have this error: chan_sip.c:5782 register_verify: Peer '501' is trying to register, but not configured as host=dynamic |
| 15:54.03 | _T3_ | i'm using realtime |
| 15:54.07 | _T3_ | static |
| 15:54.13 | anthm | iaxs[callno] is a struct iax2_pvt |
| 15:54.20 | criptos | yeap... |
| 15:54.42 | Juggie | realtime static is such a waste of code |
| 15:54.42 | file | database, database, fetch me a row |
| 15:54.47 | bvane | NSGN good.. |
| 15:54.48 | anthm | when a new one is being setup you have that 1 chance to take any data you can from the user or peer it's being configured by |
| 15:54.50 | Juggie | all the static could should be entirely removed |
| 15:54.52 | lathos42 | anthm: Is changrab written so that if the I did a changrab(SIP/399) and SIP/399 was ringing, it would connect me with the phone that was doing the ringing? |
| 15:55.06 | criptos | and is build from a creating call + user info, for what check_access shows me. |
| 15:55.09 | Juggie | i have no idea why there are TWO database formats |
| 15:55.11 | NSGN | bvane: now i'm toying with voicemail. i think i'm actually in decent shape now. i'm sure i'll have more questions soon though |
| 15:55.11 | *** join/#asterisk pfn (n=pfnguyen@netblock-66-245-252-239.dslextreme.com) |
| 15:55.13 | Juggie | one for dynamic & one for static |
| 15:55.14 | anthm | you have both obj in scope and you must cp the data onto the pvt because they will never meet again |
| 15:55.38 | *** join/#asterisk CoaxD (i=coax@shell1.cornernet.com) |
| 15:55.41 | CoaxD | Blah |
| 15:55.46 | CoaxD | network stupidness |
| 15:55.52 | *** join/#asterisk xADDY (n=xAD_nFL@host144-199.pool8290.interbusiness.it) |
| 15:55.53 | _T3_ | i dont know relly |
| 15:55.58 | criptos | but the data copied if from the user structure ant not from the peer structure rigth? |
| 15:56.00 | Juggie | anthm, is it just me or is that stupid, realtime static vs dynamic have dif table formats |
| 15:56.07 | _T3_ | the only thing that i want is manage everything from database |
| 15:56.18 | Juggie | i have no problem with static & dynamic config, but i'm not sure why they cant share a format. |
| 15:56.23 | anthm | lathos42, it will take that exact channel, if you add a |-b it will take who it's bridged to |
| 15:56.40 | anthm | waht is realtime static |
| 15:56.52 | Juggie | hahaha |
| 15:56.55 | Juggie | oh you should see this |
| 15:57.12 | Juggie | what a hunk of crap code this is... <goes to get a link> |
| 15:57.18 | NSGN | ok i have one already |
| 15:57.33 | _T3_ | any guide |
| 15:57.34 | Juggie | anthm, see http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime+Static |
| 15:57.39 | Juggie | look at the dable structure |
| 15:57.43 | _T3_ | i read that already |
| 15:57.46 | _T3_ | <PROTECTED> |
| 15:57.56 | Juggie | _T3_ is your name anthm? |
| 15:57.58 | _T3_ | but this is the message that i obtain |
| 15:58.09 | Juggie | the message you obtain is self explanatory |
| 15:58.12 | NSGN | how do i make a custom voicemail greeting? |
| 15:58.24 | NSGN | nevermind...i found it. geez this is confusing sometimes |
| 15:58.26 | *** join/#asterisk denon23532 (i=denon@synapse.subneural.net) |
| 15:58.35 | Juggie | the host is trying to register to you, and it can go one of two ways |
| 15:58.36 | anthm | oh that, oh ya i made that table =p |
| 15:58.43 | *** join/#asterisk Santiago (n=santiago@200.68.84.75) |
| 15:58.44 | Juggie | 1) the host is not supposed to register, fix them. |
| 15:58.48 | lathos42 | anthm: Hmm.. The guy who did the patch for 1.0.x must have broken something.. it works as designed when the call has been answered, but while its ringing it connects me to the phone that's ringing, not the phone that originated the call.. |
| 15:58.48 | _T3_ | i know that, but iun the database exits a record with host=dynamic |
| 15:58.55 | Santiago | hello. |
| 15:58.55 | criptos | If sip is trying to register, then sip configuration expects, that only register sip phones will be using dynamic IP, so, or you put host=dynamic at your sip.conf entry or you tell the phone not to register. |
| 15:59.00 | *** join/#asterisk drumkilla_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:59.00 | *** mode/#asterisk [+o drumkilla_] by ChanServ |
| 15:59.09 | Juggie | 2) they are and you havnt set it right in the database |
| 15:59.16 | Santiago | can anyone give some information about asrisk experiences..? |
| 15:59.20 | Juggie | _T3_, do a 'sip show peers' |
| 15:59.28 | _T3_ | Asterisk CVS-NHEAD-08/24/05-20:08:55 built by root@voipserver on a i686 running Linux on 2005-08-25 01:05:25 UTC |
| 15:59.32 | Juggie | umm |
| 15:59.35 | Juggie | thats show version |
| 15:59.38 | _T3_ | i download a few days ago |
| 15:59.38 | Juggie | i asked for sip show peers |
| 15:59.44 | _T3_ | ok ok ok |
| 15:59.50 | Santiago | i have a hotel... |
| 16:00.01 | mog_home | DRUMKILLA |
| 16:00.02 | _T3_ | Name/username Host Dyn Nat ACL Mask Port Status |
| 16:00.02 | _T3_ | 502 (Unspecified) 0.0.0.0 5060 Unmonitored |
| 16:00.02 | _T3_ | 501/501 (Unspecified) 0.0.0.0 5060 Unmonitored |
| 16:00.02 | _T3_ | 333 (Unspecified) 0.0.0.0 5060 Unmonitored |
| 16:00.02 | _T3_ | 123 (Unspecified) 0.0.0.0 5060 Unmonitored |
| 16:00.02 | _T3_ | 4 sip peers [4 online , 0 offline] |
| 16:00.07 | mog_home | yuck |
| 16:00.08 | Juggie | ok |
| 16:00.12 | Juggie | see how DYN is there |
| 16:00.15 | *** join/#asterisk edwin_ (i=edwin@252-131-222-203.rev.techex.net.au) |
| 16:00.17 | Juggie | (and dont paste in the chan BTW) |
| 16:00.20 | anthm | ok so what was the question about the table? |
| 16:00.21 | Juggie | use www.pastebin.ca |
| 16:00.23 | _T3_ | yes but is empty |
| 16:00.27 | Juggie | right, |
| 16:00.33 | Juggie | thus the table is not configured properly |
| 16:00.37 | _T3_ | when i use the sip.conf |
| 16:00.49 | Juggie | anthm, no question about the table, it just sucks :) |
| 16:01.00 | _T3_ | in the Dyn column is "D" |
| 16:01.07 | _T3_ | hahaha |
| 16:01.08 | Juggie | it was a great idea @ the time, but with the realtime dynamic tables, static should just use the same table design |
| 16:01.31 | anthm | what is a realtime dynamic table ? |
| 16:01.32 | Juggie | which is considerabally easier to edit, etc. |
| 16:01.42 | _T3_ | ok, some help about the table |
| 16:01.43 | NSGN | ok here's a real one. so i record a busy message, but after it that lady says the deal about recording after the tone. how can i make * not play that, just my busy message then the tone? |
| 16:02.01 | Juggie | umm like the realtime sip/iax table |
| 16:02.03 | *** part/#asterisk gr0mit (n=w10277@fw.mot-tools.co.uk) |
| 16:02.05 | Juggie | http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime+Sip |
| 16:02.27 | _T3_ | from there i take the example |
| 16:02.40 | Juggie | if you used that format, and told it to be static, it would be great. no need to have code in * to parse two dif formats. |
| 16:02.49 | mog_home | drumkilla_!!! |
| 16:02.59 | CoaxD | NSGN: I'm not entirely sure its possible |
| 16:03.10 | CoaxD | NSGN: You could, however, record a very short gsm file and replace the prompt with it |
| 16:03.14 | *** join/#asterisk Essobi (i=kstone@75.137.26.216.host.teledvance.com) |
| 16:03.16 | NSGN | coaxd: eh? that seems odd, being how flexable * is |
| 16:03.26 | CoaxD | NSGN: The voicemail system is hardcoded in a lot of ways |
| 16:03.28 | Juggie | NSGN |
| 16:03.33 | mog_home | or you could pay for a nice developer to add that option |
| 16:03.34 | NSGN | couldnt i just change the pattern it runs through for taking a voicemail in extensions.config, or is it hardcode? |
| 16:03.36 | mog_home | ^_^ ^_^ |
| 16:03.37 | _T3_ | sorry |
| 16:03.37 | Juggie | look at the parameters for Voicemail() |
| 16:03.40 | anthm | keep in mind you need a table strucure system that covers the entire config format |
| 16:03.45 | _T3_ | i dont use that structure |
| 16:03.47 | CoaxD | NSGN: No |
| 16:03.53 | CoaxD | NSGN: You're only calling Voicemail() |
| 16:03.57 | NSGN | :-/ darn |
| 16:04.00 | CoaxD | NSGN: The pattern in question isnt in extensions.conf |
| 16:04.05 | CoaxD | NSGN: It is in Voicemail() |
| 16:04.07 | mog_home | but for cash |
| 16:04.10 | anthm | if you went to implement what you are talking about you'd get stuck in the middle with the patient's chest cracked open and say "ohhh oops" |
| 16:04.12 | _T3_ | mysql> desc ast_config |
| 16:04.13 | _T3_ | <PROTECTED> |
| 16:04.13 | _T3_ | +------------+--------------+------+-----+---------+----------------+ |
| 16:04.13 | _T3_ | | Field | Type | Null | Key | Default | Extra | |
| 16:04.13 | _T3_ | +------------+--------------+------+-----+---------+----------------+ |
| 16:04.13 | _T3_ | | id | int(11) | | PRI | NULL | auto_increment | |
| 16:04.15 | _T3_ | | cat_metric | int(11) | | | 0 | | |
| 16:04.16 | CoaxD | NSGN: Really, you probably could eliminate 2 lines of C code... |
| 16:04.17 | _T3_ | | var_metric | int(11) | | | 0 | | |
| 16:04.19 | NSGN | heh, well if it comes down to it i'll replace the stupid file it plays |
| 16:04.19 | _T3_ | | commented | int(11) | | | 0 | | |
| 16:04.21 | _T3_ | | filename | varchar(128) | | MUL | | | |
| 16:04.23 | _T3_ | | category | varchar(128) | | | default | | |
| 16:04.24 | Juggie | stop stop |
| 16:04.25 | _T3_ | | var_name | varchar(128) | | | | | |
| 16:04.27 | mog_home | ewwww |
| 16:04.27 | _T3_ | | var_val | varchar(128) | | | | | |
| 16:04.29 | _T3_ | +------------+--------------+------+-----+---------+----------------+ |
| 16:04.29 | CoaxD | ./k _t3_ |
| 16:04.31 | _T3_ | 8 rows in set (0.00 sec) |
| 16:04.31 | mog_home | big table... |
| 16:04.32 | NSGN | ....what is going on? |
| 16:04.33 | criptos | anthm, a user and a peer are built at the same time? So, callpickup and pickupgroup configurationshould be added to the user.. becose this info goes to the pvt rigth? |
| 16:04.38 | *** join/#asterisk azrishahril (n=azrishah@60.50.204.87) |
| 16:04.45 | mishehu | thank you for not using a pastebin |
| 16:04.48 | anthm | yes |
| 16:04.49 | NSGN | :-D |
| 16:04.51 | anthm | add it to the user |
| 16:04.55 | CoaxD | NSGN: I suggest replacing the prompt in question - in the voicemail prompt directory |
| 16:04.58 | *** join/#asterisk samy (n=samy@226.sub-70-209-168.myvzw.com) |
| 16:04.58 | anthm | then cp it to the pvt in check_access |
| 16:05.04 | NSGN | alright |
| 16:05.05 | CoaxD | NSGN: With like a 1/4 second prompt |
| 16:05.09 | NSGN | ok |
| 16:05.10 | Juggie | NSGN, see http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+VoiceMail |
| 16:05.17 | Juggie | look specifically at the voicemail options |
| 16:05.36 | NSGN | next question: wtf does the "digital receptionist" thing in AMP even do? i make one...and its like..ok, what does it do? |
| 16:05.46 | NSGN | either its kinda dumb or i'm missing it's point |
| 16:05.50 | NSGN | latter more likely |
| 16:05.57 | mog_home | is there #amp? |
| 16:06.03 | anthm | lathos did you try app_intercept ? |
| 16:06.07 | mog_home | if so we should send people there... |
| 16:06.13 | CoaxD | NSGN: Bwahahaha. i love how you put that |
| 16:06.13 | *** join/#asterisk christian[asgi] (n=christia@border0hsv.asterisksgi.com) |
| 16:06.20 | NSGN | heh |
| 16:06.22 | CoaxD | NSGN: It is *exactly* what i say, when i've got an issue like that |
| 16:06.26 | *** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com) |
| 16:06.36 | CoaxD | NSGN: (I dont know the answer to your question) |
| 16:06.40 | NSGN | ok |
| 16:06.43 | Juggie | so, Voicemail(sb${EXTEN}@default) |
| 16:07.10 | NSGN | so if i made a voice menu like that ( i think thats what it tries to make ) then how would i get someone to it? |
| 16:07.12 | Juggie | sb=play busy, but not introduction |
| 16:07.14 | criptos | Ok, now, I have the callpickup info at the chan_iax2_pvt, but ast_pickup_chan at res_features.c expects a ast_channel *c to answer the phone, now from chan_iax2_pvt how do I get that info? and also, ast_channel has callpikup and picupgroup, so when does the info at iax2_pvt structure get passed to a ast_channel structure? |
| 16:07.26 | NSGN | can you give it an extension? |
| 16:07.32 | Juggie | nsgn, your question does not compute. |
| 16:07.43 | lathos42 | anthm: I havent yet.. will it play nice with app_changrab so I can have 1 extension that will either grab the ringing call, or grab the on-hold call? |
| 16:08.09 | CoaxD | NSGN: Read up. |
| 16:08.15 | CoaxD | NSGN: Thats what the docs are for. |
| 16:08.19 | CoaxD | NSGN: And start playing with it. and testing. |
| 16:08.21 | NSGN | juggie: ok, so say i wanted all phones in the house to ring for 15 seconds, then for some voice message to pick up giving options to go to different voicemail boxes for diff family members |
| 16:08.35 | CoaxD | NSGN: You have all the tools. (A softphone client is all you need. hell, dont even need a mic to test IVR stuff.) |
| 16:08.56 | anthm | app_intercept is virtual *8 |
| 16:09.04 | NSGN | coaxd: ok. then here is my final question for a while. i dont have a PSTN line or a VoIP provider connected to this * setup at the moment. how would i test what an incoming caller would hear? |
| 16:09.16 | criptos | where is that app_intercept? |
| 16:09.17 | anthm | its entire mission is for what you want |
| 16:09.19 | mog_home | bye |
| 16:09.22 | Juggie | NSGN, that is very easy, i suggest you read up on dialplan logic. |
| 16:09.23 | Nugget | MSGN: download a soft phone and call yourself. |
| 16:09.34 | Nugget | like x-lite or something |
| 16:09.40 | NSGN | i have xlite...its one of my extensions |
| 16:09.45 | NSGN | i dont wanna test my own voicemail |
| 16:09.47 | Nugget | call yourself with it. |
| 16:09.59 | NSGN | i wanna test what an incoming PSTN line would hear when they call and asterisk handles it |
| 16:09.59 | *** join/#asterisk PBXtech (i=nik@209-181-147-50.slkc.qwest.net) |
| 16:10.02 | Nugget | so call yourself with it. |
| 16:10.02 | NSGN | but i dont have the PSTN hardware right now |
| 16:10.03 | Juggie | then use a cell phone |
| 16:10.08 | lathos42 | anthm: Ok, so as long as it has a intercept value to grab onto, it doesnt care what state the call is in? |
| 16:10.12 | NSGN | what number do i dial? |
| 16:10.13 | NSGN | heh |
| 16:10.17 | Nugget | what number did you set up? |
| 16:10.20 | Juggie | get the hardware then :) |
| 16:10.28 | Juggie | what you want is very simple |
| 16:10.30 | Juggie | keep reading |
| 16:10.39 | Nugget | asterisk won't do diddly squat for a caller unless you tell it what to do in the dialplan. |
| 16:10.41 | *** join/#asterisk MuppetMaster (n=MuppetMa@62.57.152.193) |
| 16:10.42 | anthm | http://www.pbxfreeware.org has app_intercept |
| 16:10.43 | Juggie | i could do it for you but then i'd hjave to charge you |
| 16:10.55 | Nugget | if you haven't done that, then there is no "what a caller would hear" |
| 16:10.55 | Juggie | so you may as well learn yourself |
| 16:11.06 | CoaxD | NSGN: You really need to read the documentation |
| 16:11.12 | CoaxD | NSGN: And stop relying on IRC for your information. |
| 16:11.18 | NSGN | nugget: understood. so how do i test that dialplan how it would sound for an external user without the PSTN hardware. is there a way to call from my softphone and have it be like an external incoming call? |
| 16:11.24 | CoaxD | NSGN: Asterisk is very flexible. it will do *anything*. but you have to TELL it what to do. |
| 16:11.30 | Nugget | call it from your softphone. |
| 16:11.31 | NSGN | coaxd: i have my head in docs :-P dont think i'm not reading |
| 16:11.33 | Juggie | NSGN, absolutely |
| 16:11.38 | CoaxD | NSGN: Everything has a dialplan. You need to figure out how to configure dialplans and contexts. |
| 16:11.58 | CoaxD | NSGN: Once you try it and figure it OUT, you will become VERY proficient at asterisk very quickly |
| 16:12.05 | Juggie | we are here to help with complex problems, what you are doing is very simple |
| 16:12.10 | Juggie | go read some more, play around |
| 16:12.16 | Juggie | and if you are still having trouble, come back. |
| 16:12.37 | criptos | thanks antm, I will verify that ... |
| 16:12.42 | anthm | Intercept([<Channel Name>|<varmatch>|auto]) |
| 16:12.43 | anthm | Intercept an unanswered channel: |
| 16:12.43 | anthm | <PROTECTED> |
| 16:12.43 | anthm | <PROTECTED> |
| 16:12.43 | anthm | <PROTECTED> |
| 16:12.54 | Hmmhesays | heh bind responds with a no host found when it gets sent _sip._upd.hot1.booty.call |
| 16:12.58 | Juggie | on the other hand, if you want instant gratifaction, send me money and i'll configure it all for you :) |
| 16:13.02 | criptos | /leave |
| 16:13.05 | NSGN | ok, well thanks all |
| 16:13.07 | NSGN | lunch then work |
| 16:13.08 | criptos | thanks :) |
| 16:13.08 | NSGN | later |
| 16:13.11 | CoaxD | NSGN: Welcome |
| 16:13.14 | NSGN | :-) |
| 16:13.18 | *** part/#asterisk criptos (n=criptos@201.145.229.183) |
| 16:13.23 | Juggie | its funny |
| 16:13.29 | Juggie | people feel so overwealmed at first |
| 16:13.34 | Juggie | and then they realise how simple it is |
| 16:13.38 | Hmmhesays | people don't ever want to pay anything |
| 16:13.40 | Hmmhesays | <chuckle> |
| 16:13.48 | *** join/#asterisk asteriskDOTbz (n=logger@pbxtech.com) |
| 16:14.02 | CoaxD | Hmmhesays: Do YOU want to pay anything? |
| 16:14.12 | asteriskDOTbz | <PROTECTED> |
| 16:14.15 | Hmmhesays | nope, but I don't want people to fix anything for free either ;) |
| 16:14.45 | CoaxD | What does a * develoepr want to charge me to write up a nice * record-on macro that'll store everything neatly by year, month, day ? :) |
| 16:14.47 | *** join/#asterisk jtodd (n=jtodd@ti.fox-den.com) |
| 16:14.55 | CoaxD | I'd do it, but.. I dont have time :) |
| 16:15.05 | CoaxD | (Note: Directory creation needs to work.) |
| 16:16.15 | Hmmhesays | hey file, whats up |
| 16:16.16 | CoaxD | see? When you actually WANT to do business with a * developer, they're never around. |
| 16:16.25 | *** part/#asterisk MuppetMaster (n=MuppetMa@62.57.152.193) |
| 16:16.25 | file | CoaxD: toast. |
| 16:16.34 | file | Hmmhesays: not toooo much, you? |
| 16:16.38 | *** join/#asterisk hellagony (n=hellagon@200.121.129.178) |
| 16:16.42 | Hmmhesays | tinkering with some hardware |
| 16:16.47 | anthm | $100 |
| 16:16.49 | file | cool cool |
| 16:17.25 | christian[asgi] | anthm: that's up front right? |
| 16:17.37 | jontow | uh-oh.. zap issues.. wonder what i broke :( |
| 16:18.18 | jontow | cool, i think my boss disabled interlata on the PRI |
| 16:18.36 | jontow | :( |
| 16:19.36 | endom0rph | hi, is it possible to have an extension reroute on busy/noanswer to a oh323 external voice processing system passing the original called extension number and reason(busy/noanswer). I get the extension number of the h323 when I just add a Dial on busy no-answer. Do I need a different command/macro? |
| 16:19.51 | Santiago | hello.. |
| 16:20.05 | _T3_ | help! |
| 16:20.15 | Santiago | anyone can introduce me to * |
| 16:20.20 | _T3_ | i change to new table structure |
| 16:20.28 | Santiago | i have many hotel clients interested. |
| 16:20.35 | *** join/#asterisk ]bodhi[ (n=not@h48n2fls302o1034.telia.com) |
| 16:20.36 | Beirdo | there, did my part, he shouldn't drown now |
| 16:20.44 | _T3_ | F1 F1 F1 |
| 16:20.47 | jontow | santiago; http://www.voip-info.org/ can |
| 16:20.49 | file | Santiago: we're not here to sell you asterisk/etc, Google/websites do that |
| 16:20.54 | file | sell you on asterisk rather |
| 16:21.11 | Beirdo | asterisk... Santiago... |
| 16:21.18 | Beirdo | Santiago.. asterisk.. |
| 16:21.22 | Beirdo | there, you're introduced |
| 16:21.24 | Beirdo | have fun :) |
| 16:21.26 | file | awww how cute |
| 16:21.32 | _T3_ | hahaha |
| 16:21.50 | *** join/#asterisk shadebob (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma) |
| 16:21.51 | shadebob | hi |
| 16:22.15 | _T3_ | Juggie |
| 16:22.27 | _T3_ | J u g g i e |
| 16:23.53 | Santiago | argentina |
| 16:24.13 | Santiago | Beirdo funny... thanks... |
| 16:25.33 | lathos42 | Is there a Dialplan app that will check the status of a channel, ie whether its ringing or answered? |
| 16:25.49 | _T3_ | FOP |
| 16:26.17 | _T3_ | flash operator panel |
| 16:26.27 | cpatry | lathos42: type show application dial and see variable dialstatus |
| 16:26.31 | *** join/#asterisk pbxbart__ (n=pbxbart@p54B032BC.dip0.t-ipconnect.de) |
| 16:26.36 | cpatry | ~fop |
| 16:26.37 | jbot | An XSL formatter written in Java that outputs PDF. URL: http://www.jtauber.com/fop/ |
| 16:26.37 | *** part/#asterisk pbxbart__ (n=pbxbart@p54B032BC.dip0.t-ipconnect.de) |
| 16:27.32 | cpatry | ~fop |
| 16:27.33 | jbot | An XSL formatter written in Java that outputs PDF. URL: http://www.jtauber.com/fop/, or the Flash Operator Panel |
| 16:27.50 | *** join/#asterisk Al_Pocke (n=al_pocke@dsl-084-060-061-212.arcor-ip.net) |
| 16:28.17 | cpatry | lathos42: work a bit, i wont do it for ya. |
| 16:28.20 | _T3_ | ok, i do everything about config realtime mysql |
| 16:28.38 | _T3_ | everything is working but i dont have any peer |
| 16:28.40 | _T3_ | loaded |
| 16:29.19 | *** join/#asterisk Defraz (n=t0tal@tim.ibccom.net) |
| 16:30.05 | *** join/#asterisk edwin__ (i=edwin@252-131-222-203.rev.techex.net.au) |
| 16:30.39 | Cherebrum | I just posted a bounty on voip-info. see: http://tinyurl.com/cxuww |
| 16:30.44 | _T3_ | Atention.....realtime users... please talk with _T3_ now.... |
| 16:30.56 | tzanger | _T3_: that's a quick way to an /ignore |
| 16:30.57 | _T3_ | thank you |
| 16:31.03 | _T3_ | ups |
| 16:31.10 | *** join/#asterisk santiago (n=santiago@200.68.84.75) |
| 16:31.21 | *** join/#asterisk chendy_idle___ (n=Alex_Dot@218.1.218.17) |
| 16:31.47 | *** join/#asterisk tartar (n=tartar@CPE0004e27b716e-CM014370001917.cpe.net.cable.rogers.com) |
| 16:32.00 | _T3_ | never happend again |
| 16:33.51 | *** join/#asterisk SimonR (n=SimonR@CPE001310092352-CM001371142e78.cpe.net.cable.rogers.com) |
| 16:34.22 | lathos42 | anthm: How much would you want to write an app that combined app_intercept and the answered pickup of app_changrab? I'm going to attempt to put together an AGI to do it myself, but i'd like an option to give my company if they Really want call pickup functionality |
| 16:34.38 | anthm | $325 |
| 16:35.08 | SimonR | Anyone know who does free origination? |
| 16:35.11 | anthm | but are you sure app_intercept is not all you need? |
| 16:35.23 | SimonR | As in a CLEC that gets paid for the inbound minutes and will give you 0 cents per minute? |
| 16:35.58 | lathos42 | anthm: Our current phone system has the ability to either pickup a ringing call, or an answered call from one button on the phone.. I need to emulate that behavior for the new phone system |
| 16:36.15 | bkw_ | The public internet SUCKS!!!! |
| 16:36.19 | CoaxD | SimonR: Everybody does |
| 16:36.31 | CoaxD | SimonR: NuFone, Broadvoice, VoicePulse, etc |
| 16:36.33 | *** join/#asterisk myiagy (n=myiagy@200.138.215.78) |
| 16:36.40 | *** join/#asterisk yogurt2ungue (n=yogurt2u@44-170-114-200.fibertel.com.ar) |
| 16:37.18 | SimonR | most of them won't give you a large number of phone lines, however. |
| 16:39.44 | anthm | how do you pick up an answered call with 1 button ? |
| 16:40.52 | lathos42 | anthm: well, its not technically one button.. there's a Pick-up button on the phone, and it asks for the extension you want to pick up.. it then either grabs the call that is ringing into that extension, or the call that extension has on hold |
| 16:41.54 | anthm | ok, so yeah let me know I can do it in 1 day if need be |
| 16:43.10 | cpatry | lathos42: what kind of phone is that exactly? |
| 16:43.21 | lathos42 | anthm: Ok, cool.. I'll let you know one way or the other |
| 16:43.23 | *** join/#asterisk Nukemizer (n=Nuke@67.137.28.163) |
| 16:44.30 | lathos42 | cpatry: Its a Merlin Magix phone system |
| 16:45.07 | *** join/#asterisk dasuberdavid (n=david@digium.com) |
| 16:45.23 | *** join/#asterisk jief- (n=jief@modemcable214.181-80-70.mc.videotron.ca) |
| 16:46.02 | jief- | hello, i was wondering if any of you use Eicon 4 FXO cards? If yes, how do they compare to Digium cards? |
| 16:46.05 | lathos42 | Ok, its lunchtime for me |
| 16:47.07 | *** join/#asterisk NSGN (n=NSGN@cpe-66-69-197-25.austin.res.rr.com) |
| 16:47.16 | *** join/#asterisk cjk (n=cjk@80.92.64.103) |
| 16:47.37 | NSGN | -_- so i suppose the fact that my on-hold music plays in three second increments is because my * server is 233mhz? -_- |
| 16:47.38 | *** join/#asterisk gooagle (n=goldenol@ns2.xoasisnetworks.com) |
| 16:47.44 | gooagle | hi |
| 16:47.56 | gooagle | anyone had any luck with hardware echo canceler devices? |
| 16:48.34 | rking | NSGN: what file type is your music? |
| 16:48.39 | NSGN | mp3 |
| 16:48.43 | NSGN | just some defaults |
| 16:49.07 | rking | NSGN: you should probably use the native format, so you don't have to process as much, no? |
| 16:49.19 | NSGN | that'd make sense. what IS the native format? heh |
| 16:49.26 | NSGN | a wav/aiff? |
| 16:50.56 | rking | oh, hrmm.. i have a doc here for the iaxcomm format... but yeah, i'd think either wav/aiff would help |
| 16:50.59 | *** join/#asterisk crich1999 (n=crich@p54BFAC2D.dip0.t-ipconnect.de) |
| 16:51.18 | NSGN | iaxcomm? how the heck would you encode a song into that? :-P |
| 16:51.24 | rking | mp3 decoding is pretty processor-intensive |
| 16:51.26 | NSGN | yeah |
| 16:51.35 | rking | NSGN: ringtones is what the iaxcomm doc is talking about |
| 16:51.36 | NSGN | i'm about to load up a wav |
| 16:51.43 | NSGN | oh, ok |
| 16:52.32 | NSGN | argh, what folder do you put hold music in? |
| 16:53.14 | *** join/#asterisk _Kris_ (n=kris@lnx001.nat.hst.tmcsolutions.net) |
| 16:53.37 | *** join/#asterisk allanon (i=allanon@netblock-66-245-233-131.dslextreme.com) |
| 16:55.59 | jief- | <PROTECTED> |
| 16:56.01 | *** join/#asterisk criptos (n=criptos@201.145.229.183) |
| 16:56.09 | criptos | Hi again... |
| 16:56.40 | criptos | about the hanging up, I set busydetect=no and callprogrees=no using ks singalling, and now. * nevers hangup a call... |
| 16:57.09 | NSGN | well this is lovely. apparently i dont have privileges to access that folder via FTP |
| 16:57.24 | criptos | and also, pbxfreeware.com is down rigth? |
| 16:57.39 | cpatry | pbxfreeware.org |
| 16:57.51 | jief- | NSGN: it totally makes sense to not have access there |
| 16:57.58 | jief- | as only root should have access to /var/lib |
| 16:58.03 | NSGN | well.. how do i put a stupid song in there? :-D |
| 16:58.23 | jief- | you upload it to your ~, then you move it where it belongs with the root account |
| 16:58.28 | jief- | and change permissions and so forthj |
| 16:58.43 | NSGN | heh, guess so |
| 16:58.52 | criptos | pbxfreeware.org is down? or it´s me :) |
| 17:00.07 | file | criptos: just you. |
| 17:00.10 | *** join/#asterisk atmos4 (n=sunset@213-182-117-217.teleos-web.de) |
| 17:00.13 | atmos4 | re |
| 17:01.11 | puowvip | I wonder how many concurrent calls does a DS3 handle |
| 17:01.12 | mariogamboa | wow i resolve the context in my asterisk |
| 17:01.15 | mariogamboa | :) |
| 17:01.29 | cpatry | puowvip: 672 |
| 17:01.32 | mariogamboa | but i have a doubt about access code |
| 17:01.42 | cpatry | at24 chan per t1 |
| 17:01.45 | criptos | file, can you emailme app_intercept? I`m really unable to get to pbxfreeware.org |
| 17:02.01 | mariogamboa | how i can implement access code not for extension if for users |
| 17:02.03 | mariogamboa | ? |
| 17:02.07 | mariogamboa | any ideas |
| 17:02.10 | file | criptos: sure, what's your email? |
| 17:03.01 | jief- | puowvip: a DS3 is 32 PRI, and there's 24 lines available per PRI as far as i know |
| 17:03.39 | cpatry | its 28, not 32, iirc |
| 17:05.03 | *** join/#asterisk myiagy (n=myiagy@200.138.215.78) |
| 17:05.44 | gooagle | I keep reading about turning down the jitterbuffer to decrease echo... Where do I find this setting? |
| 17:05.56 | *** join/#asterisk _T3_ (n=rposada@35.229.uio.satnet.net) |
| 17:06.59 | _T3_ | one question? |
| 17:07.19 | *** join/#asterisk dsalama (n=dsalama@adsl-9-47-42.mia.bellsouth.net) |
| 17:07.36 | _T3_ | realtime is working, but i cant see my peers using sip show peers...why? |
| 17:10.01 | jief- | cpatry: you were right, its 28, not 32 |
| 17:10.27 | Hmmhesays | ack this is not workikng right |
| 17:10.29 | Hmmhesays | *working |
| 17:10.48 | *** join/#asterisk fulgas (n=fulgas@213.58.130.46) |
| 17:11.16 | samy | any ideas on why it would not be getting it from the SendDTMF()? i've even produced a gsm file with a dtmf 9 and play it with no success |
| 17:11.22 | samy | question for some of you...i have a PRI in an asterisk box. an apartment buzzer system dials into asterisk, asterisk does a SendDTMF(9) (and i've tried SIPdtmfmode() before the SendDTMF() with all three types). the buzzer should hear the dtmf 9 and open but it doesn't. if i then trunk the call out to my cell phone and hit 9 from the cell (buzzer -> * -> cell), the buzzer system hears it properly |
| 17:11.30 | samy | wrong order for messages, heh |
| 17:11.36 | NSGN | alright...i just did a test call through my * server between two software phones...and the audio quality was total crap. what the heck is wrong? |
| 17:12.38 | atmos4 | NSGN: change the codec config |
| 17:12.48 | NSGN | on the softphones or *? |
| 17:12.56 | NSGN | and what is a good codec? |
| 17:12.56 | atmos4 | if you have lan ulaw and alaw should be fine |
| 17:13.32 | atmos4 | I use something like disallow=all\nallow=ulaw\nallow=alaw |
| 17:13.41 | atmos4 | where \n is a linebreak |
| 17:14.15 | *** join/#asterisk blessen (n=blessen@140.99.23.26) |
| 17:14.30 | atmos4 | besides you should make sure the soundcards of the pcs at least have a microphone boost option |
| 17:14.50 | atmos4 | because without it passive microphones from headset are about unusable |
| 17:15.01 | NSGN | these are mac laptops, good mics and speakers. i use ichat on em all the time |
| 17:15.14 | NSGN | and skype |
| 17:15.27 | NSGN | so where would i make those codec changes? |
| 17:15.35 | blessen | Hi spackel : let me first of all thank you for what you have done for me by helping me configure asterisk with kphone:-)...it was really nice of u... |
| 17:15.41 | atmos4 | skype has an inbuilt ampflification, it works even when I use my onboard sound which has no mic boost |
| 17:15.57 | atmos4 | but for everything else I needed to switch the mic to my audigy |
| 17:16.04 | NSGN | well this wasnt quiet sound, this was buzzing and corruption |
| 17:16.08 | blessen | i have one more issue ...spackle |
| 17:16.13 | atmos4 | NSGN: in the sip peer |
| 17:16.30 | atmos4 | eg. your friend entry in sip.conf |
| 17:16.38 | NSGN | hm...ok |
| 17:16.40 | atmos4 | just look at the comments and you'll get the idea |
| 17:16.48 | blessen | spaclkle may i chat with you directly |
| 17:17.37 | atmos4 | also I had quite a difference in quality with different softphones, sipps from ahead gives me the best quality and surrounding noise suppression so far |
| 17:17.37 | *** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1) |
| 17:17.42 | atmos4 | but it's win only |
| 17:17.45 | *** join/#asterisk angler (n=angler@digium.com) |
| 17:18.20 | NSGN | ah |
| 17:18.26 | atmos4 | besides the fact that their installer is so stupid it deletes the nero install ;-) |
| 17:18.57 | NSGN | haha |
| 17:19.00 | NSGN | ok i'm in sip.conf |
| 17:19.08 | NSGN | i see the codec stuff |
| 17:19.11 | *** join/#asterisk ManxPower (n=eric@dsl-209-16-67-160.datasync.com) |
| 17:19.58 | NSGN | disallow=all |
| 17:19.58 | NSGN | allow=ulaw |
| 17:19.59 | NSGN | allow=alaw |
| 17:20.04 | NSGN | thats what its set to right now |
| 17:20.07 | NSGN | doesnt that seem right? |
| 17:21.38 | gooagle | in zapata.conf is jitterbuffers is not defined, does it default to 0 or ? |
| 17:21.39 | NSGN | !ping |
| 17:23.15 | atmos4 | NSGN: looks ok |
| 17:23.25 | NSGN | hmm, then what is with my audio quality? |
| 17:23.46 | NSGN | there was a digital sounding pulse/buzzing that made the person almost inaudable on the other end |
| 17:24.06 | mariogamboa | how i can make call with accountcode |
| 17:24.21 | mariogamboa | depend of accound code is the line the user can take |
| 17:24.45 | atmos4 | NSGN: have you reloaded asterisk and reconnected the sip clients? |
| 17:25.10 | NSGN | no. this setup is pretty fresh. as of last night |
| 17:25.21 | ManxPower | NSGN: don't allow both ulaw and alaw |
| 17:25.28 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 17:25.52 | ManxPower | If you are in the USA/Canada/Japan(?) then allow ulaw. If you are anywhere else, allow alaw. |
| 17:26.01 | NSGN | but not both? |
| 17:26.08 | bjohnson | no point |
| 17:26.11 | ManxPower | NSGN: correct. |
| 17:26.21 | NSGN | but would that screw my call quality? |
| 17:26.25 | ManxPower | Asterisk has a history of poor transcoding between ulaw and alaw |
| 17:26.41 | bjohnson | and is usually not even needed |
| 17:26.46 | *** join/#asterisk zloc (n=zloc@zm1.tribunemedia.net) |
| 17:26.46 | ManxPower | bjohnson: *nod* |
| 17:26.52 | blessen | can anyone help me with getting asterisk configured for vonage along with kphones ...my complete details and configurations are pasted here |
| 17:26.55 | blessen | http://www.voipuser.org/forum_topic_2179.html |
| 17:27.01 | bjohnson | ewww |
| 17:27.05 | bjohnson | CC info? |
| 17:27.09 | NSGN | manxpower: hm, the corruption did sound like when i mixed up codecs in video editing once |
| 17:27.14 | NSGN | and quicktime was playing in an odd one |
| 17:27.14 | Nukemizer | Does or can Asterisk support Megaco + protocol ? |
| 17:27.20 | atmos4 | NSGN: try sip show channels to see what codecs it's using (console) |
| 17:27.29 | NSGN | ok |
| 17:27.42 | syle | ok i give up , whoever wrote cmd MYSQL should be shot |
| 17:27.44 | NSGN | i'm disabling alow right now |
| 17:27.59 | ManxPower | A duplex-mismatch could also cause audio issues. |
| 17:28.41 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 17:28.55 | NSGN | what would cause that/what is that |
| 17:29.06 | NSGN | argh, i cant edit sip.conf cause its in use |
| 17:29.18 | ManxPower | blessen: you cannot connect Asterisk to Vonage unless you have a BYOD account. |
| 17:29.31 | ManxPower | NSGN: of course you can. |
| 17:29.39 | NSGN | well, vi isnt too happy about it :-P |
| 17:29.54 | NSGN | gives me a page of complaining then wont let me |
| 17:29.55 | ManxPower | I think Vonage calls that "softphone account", but you don't get unlimited calling on it. |
| 17:30.01 | syzygyBSD | any idea why asterisk console would tell me it is running soxmix but never does? I copy the command from the console into a shell and run it and it works though. |
| 17:30.17 | ManxPower | sycofly: path? permissions? |
| 17:30.23 | *** join/#asterisk Ahmuck (n=chatzill@24.225.23.67) |
| 17:30.31 | ManxPower | NSGN: it's in use by another VI session. |
| 17:30.37 | *** part/#asterisk criptos (n=criptos@201.145.229.183) |
| 17:30.40 | NSGN | ..i'm pretty sure it's not |
| 17:30.41 | *** join/#asterisk frogy (n=edmund@cm222-167-86-92.hkcable.com.hk) |
| 17:30.41 | Ahmuck | good morning. what are the min requirements for asterisk |
| 17:30.52 | NSGN | at least not one I started |
| 17:31.04 | Hmmhesays | ok the example right out of the damn RFC2782 doesn't work |
| 17:31.08 | Hmmhesays | argh |
| 17:31.37 | *** join/#asterisk Lex (n=icechat5@62.249.217.145) |
| 17:32.25 | Lex | Hi everyone, Is there a channel for the 1.2 RC? |
| 17:32.34 | ManxPower | NSGN: "killall -9 vi" |
| 17:32.54 | ManxPower | then run it again just to be sure. |
| 17:32.58 | NSGN | vi: no processes killed |
| 17:33.08 | NSGN | again, same reply |
| 17:34.48 | blitzrage | question: which alarm systems work with the alarmreceiver app? Any experiences? |
| 17:34.56 | NSGN | oh well, i've gotta go to work now |
| 17:35.04 | NSGN | a client way out in nowhere land.. x_X |
| 17:35.13 | ManxPower | NSGN: do you have permission? |
| 17:35.15 | NSGN | i'll be back this evening to address whatever the heck this quality issue is |
| 17:35.16 | *** join/#asterisk psycodad (n=obiwan@2001:4060:4419:b1:0:0:0:2) |
| 17:35.18 | shadebob | ManxPower : after a Zap Destroy can I reload channel without reload zaptel driver ? |
| 17:35.25 | NSGN | manxpower: i'm SSH as root |
| 17:35.27 | ManxPower | shadebob: no. |
| 17:35.39 | ManxPower | shadebob: you said you wanted to kill a channel, not that you wanted to resurect it. |
| 17:36.02 | ManxPower | in CVS-HEAD you MIGHT be able to to a reload chan_zap.so |
| 17:36.15 | ManxPower | If not you would have to unload chan_zap.so and then load chan_zap.so |
| 17:36.30 | ManxPower | shadebob: maybe you really just wanted to hangup an active channel? |
| 17:37.17 | shadebob | ManxPower : no. I want to stop an FXS channel for block phone on certain condition. If condition is OK i need to resurect it |
| 17:37.20 | NSGN | manxpower: but i'll be back later with a few hours to mess with it |
| 17:37.29 | ManxPower | shadebob: you can't do that. |
| 17:37.35 | *** join/#asterisk Moc (n=mochouin@h66-201-214-109.gtconnect.net) |
| 17:37.39 | NSGN | thanks all, later |
| 17:38.21 | ManxPower | I guess you could do something like dial Zap/5/ in a .call file. that would pick up the fxo port and not actually dial and never timeout. |
| 17:38.34 | ManxPower | might piss off your telco, however. |
| 17:38.38 | shadebob | ManxPower : with an unload - load on stable it's ok.... |
| 17:38.52 | shadebob | ManxPower : thanks |
| 17:39.14 | ManxPower | shadebob: when you cannot unload chan_zap.so if you have any active zap calls. |
| 17:39.27 | endom0rph | is it possible to have an extension reroute on busy/noanswer to a oh323 external voice processing system passing the original called extension number and reason(busy/noanswer). I get the extension number of the h323 when I just add a Dial on busy no-answer. Do I need a different command/macro? thanks |
| 17:40.05 | shadebob | ManxPower : * wait for hangup all active zap channel and after unload chan_zap maybe? |
| 17:40.22 | shadebob | ManxPower : I haven't an * near me |
| 17:41.53 | *** join/#asterisk eKo1 (n=bernd@metrored-gw.tropicohn.com) |
| 17:42.20 | *** join/#asterisk jeremywhiting (n=jeremy@71-37-79-70.slkc.qwest.net) |
| 17:42.29 | *** join/#asterisk BrianR (i=brianr@c-24-61-206-174.hsd1.ma.comcast.net) |
| 17:42.53 | syzygyBSD | is there a program to append two gsm files rather then mix them with soxmix? |
| 17:43.45 | zoa2 | its called cp |
| 17:43.57 | zoa2 | just copy file1 + file2 to file3 |
| 17:44.06 | nick125 | no, its called cat |
| 17:44.12 | zoa2 | or cat |
| 17:44.16 | zoa2 | does cat do binary well ? |
| 17:44.23 | nick125 | cat file2 >> file1 |
| 17:44.24 | nick125 | IIRC |
| 17:44.31 | zoa2 | thats for ascii iirc |
| 17:44.37 | zoa2 | on windows it works with copy |
| 17:44.43 | zoa2 | and i think on linux the cp might do it |
| 17:45.07 | syzygyBSD | I will try cp |
| 17:45.08 | jarrod | how do i remove the last digit from the dialplan .. similar to EXTEN:1 for popping off the first |
| 17:46.30 | cochi | uhm tried :-1 maybe? |
| 17:47.34 | *** join/#asterisk drumkilla_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 17:47.34 | *** mode/#asterisk [+o drumkilla_] by ChanServ |
| 17:48.21 | netsurfer | hi guys, from time to time i'm hearing broken up audio on my * box, its 1.8ghz with 512mb/80gb - its basically just running * and has at most 2 - 3 calls going thru it (all using GSM) at any one time. Is it likely to be a bandwidth issue or cpu/ram ? |
| 17:48.54 | BrianR | netsurfer: bandwidth - the cpu is way overkill |
| 17:48.55 | *** join/#asterisk Romik (n=romik_@1.fix.netvision.net.il) |
| 17:49.02 | BrianR | netsurfer: also make sure you have realtime enabled |
| 17:49.16 | frogy | is anyone know if the "shared lines" features available in the asterisk-cvs? |
| 17:49.22 | netsurfer | BrianR - that will help with the broken up audio ? |
| 17:49.51 | bendy24 | netsurfer: sounds like a bw issue |
| 17:49.59 | bendy24 | try some QOS goodness |
| 17:50.06 | *** join/#asterisk Cresl1n (n=Cresl1n@digium.com) |
| 17:50.25 | netsurfer | bendy24 - ok.. thanks for the tip |
| 17:50.27 | BrianR | netsurfer: If it's related tointermittant load from other processes on the machine, yes. Otherwise you probably have a bandwidth problem. |
| 17:50.39 | *** join/#asterisk [hC] (n=hardcore@8.10.2.5) |
| 17:50.56 | netsurfer | BrianR - all its running is * mysql and apache |
| 17:51.04 | *** join/#asterisk The_Ball (n=alex@static-227.35.240.220.dsl.comindico.com.au) |
| 17:51.09 | netsurfer | surely 1.8ghz and 512mb is enough for all that ? |
| 17:51.21 | bendy24 | yeah, im runniong 800mhz |
| 17:51.24 | bendy24 | and its fine |
| 17:51.29 | [hC] | is there a way to get a better organized version of 'show channels' so i can see current in/outgoing calls? I can piece them together as is using show channels but i usually have to follow thru a couple channels if they're bridged to see who's calling what |
| 17:51.42 | netsurfer | im not choosy, but the users might be ;) |
| 17:51.59 | spackle | maybe a better network card? |
| 17:52.28 | bendy24 | netsurfer: hows the cpu load? |
| 17:52.42 | bendy24 | try running top and watch it for a while |
| 17:52.57 | netsurfer | bendy24 - have done, it seems fine |
| 17:53.05 | bendy24 | bw issue for sure then |
| 17:53.11 | dudes | if you're running top remotely account for that the bandwith it uses |
| 17:54.01 | bendy24 | anyone have any luck compiling h323 on tiger? |
| 17:54.01 | syzygyBSD | cat works to append bianary files too |
| 17:54.14 | *** join/#asterisk shmaltz (n=chatzill@69.28.255.210) |
| 17:54.19 | *** join/#asterisk cisco-mex (n=fco_varg@146.82.251.88) |
| 17:54.47 | bendy24 | damn pwlib refuses to build |
| 17:54.49 | *** join/#asterisk meppl (n=mephisto@p54AAF097.dip.t-dialin.net) |
| 17:54.51 | shmaltz | anybody here know how to use the onboard echo can with the TE406? |
| 17:55.04 | bendy24 | i think its the openssh libraries |
| 17:55.56 | [hC] | Anyone ever come across a CDR record storage issue where the proper incoming caller ID is detected as far as asterisk is concerned (NoOp ${CALLIERIDNUM} shows the right data) yet when it logs to CDR it sometimes shows a previous caller's callerID? |
| 17:56.08 | eKo1 | Dang it, why am I getting "Unable to create channel of type 'zap'". zttool shows no alarm on the channel I'm trying to dial through and ztcfg isn't complaining either. |
| 17:56.19 | ManxPower | [hC]: never in my wildest dreams. |
| 17:56.32 | ManxPower | eKo1: what does "zap show channels" say? |
| 17:56.53 | [hC] | ManxPower: Hm. Im trying to figure out where the error may be, im running CVS head. If asterisk outputs the correct caller ID, i would assume its something in the cdr code. |
| 17:56.54 | eKo1 | <PROTECTED> |
| 17:56.55 | eKo1 | <PROTECTED> |
| 17:56.55 | eKo1 | <PROTECTED> |
| 17:56.57 | eKo1 | etc.. |
| 17:56.59 | *** join/#asterisk dswillia (i=dswillia@wsip-68-15-113-253.ok.ok.cox.net) |
| 17:57.04 | [hC] | ManxPower: trying to narrow it down |
| 17:57.04 | ManxPower | eKo1: what card do you have? |
| 17:57.12 | eKo1 | te410p |
| 17:57.20 | ManxPower | eKo1: pri? |
| 17:57.22 | eKo1 | yep |
| 17:57.37 | ManxPower | pri debug span 1 Also check the value of HANGUPCAUSE after the dial. |
| 17:57.54 | ManxPower | you can look up the cause codes on the web. google for isdn pri cause code |
| 17:57.56 | eKo1 | I'm noticing that InAlarm: 1 when I do 'zap show channel 1'. That's bad right? |
| 17:58.14 | ManxPower | eKo1: dunno. zttool says "OK"? |
| 17:58.21 | *** join/#asterisk adjacent_ (n=scott@cpe-024-168-216-235.sc.res.rr.com) |
| 17:58.30 | dswillia | i just had a sip trunk delivered to me, and they are telling me that I need to insert a pin # they have given me before the number i.e. pin#+1+areacode+7digit number. Which conf file would I need to configure to obtain this? |
| 17:58.40 | eKo1 | ManxPower: on span 1 it does. |
| 17:58.44 | ManxPower | all ztcfg does is apply your config to the card, it doesn't verify the line is working. |
| 17:58.47 | eKo1 | which is the one i'm using. |
| 17:58.53 | ManxPower | pri debug span 1 Also check the value of HANGUPCAUSE after the dial. |
| 17:59.09 | mariogamboa | how i can define password for my user to make call via password |
| 17:59.09 | ManxPower | eKo1: can you receive calls? Is this a new PRI? |
| 17:59.10 | spackle | eKo1: my card gets stubborn after I have monkeyed with the settings and I actually have to power cycle the machine to get it back. |
| 17:59.27 | jarrod | :-1 then only gives the last digit |
| 17:59.33 | *** join/#asterisk myiagy (n=myiagy@200.138.215.78) |
| 17:59.35 | jarrod | any other way to pop the last digit off an extension? |
| 17:59.43 | ManxPower | jarrod: README.variables |
| 18:00.04 | eKo1 | ManxPower: yes, it is a new PRI |
| 18:00.15 | ManxPower | eKo1: your telco didn't finish turning it up |
| 18:00.19 | blessen | can anyone help me send and receive calls using asterrisk with vonage ...using kphone as soft phones |
| 18:00.32 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 18:00.34 | ManxPower | blessen: do you have a softphone or BYOD account? |
| 18:00.38 | *** join/#asterisk clint_ (n=clint@snap.helixsystems.com) |
| 18:00.43 | blessen | i use softphones |
| 18:00.56 | eKo1 | ManxPower: Actually, the PRI line is connected to a PRI<->SS7 switch that I have and I don't see any ISDN frames being sent to it. |
| 18:01.13 | ManxPower | blessen: Vonage will not let you connect with ANYTHING except the device they provide unless you have a special "softphone account" |
| 18:01.16 | spackle | blessen: manx wants to know if you have a softphone account with Vonage. You need one to use it with Asterisk IIRC |
| 18:01.28 | blessen | ManxPower: have a vonage account |
| 18:01.36 | blessen | yes i have a softphone account |
| 18:01.55 | ManxPower | Of course, conage does not provide unlimited calls with the softphone account. |
| 18:01.55 | jarrod | manx: perhaps you didnt know my question was not covered in that README? |
| 18:01.59 | riksta | are sangoma e1 cards supposed to show as "0000:03:03.0 Network controller: Unknown device 1923:0300" in lspci ?? |
| 18:02.00 | *** join/#asterisk PakiPenguin (i=uppal@unaffiliated/pakipenguin) |
| 18:02.01 | blessen | spackle.: you back..i thought you where busy |
| 18:02.19 | eKo1 | hmm, the hangupcause is AST_CAUSE_NOTDEFINED |
| 18:02.50 | blessen | ManxPower: Yes, i have a softphone account with vonage |
| 18:02.56 | ManxPower | jarrod: it's a documentation bug |
| 18:03.03 | ManxPower | eKo1: Um, it's a number, not a string. |
| 18:03.03 | jarrod | hah |
| 18:03.17 | spackle | blessen: just at lunch. And running out of ideas. everything *looks* good. |
| 18:03.28 | blessen | spackel : :-( |
| 18:03.39 | eKo1 | yes, the number is 0 and #define AST_CAUSE_NOTDEFINED 0 is what's in causes.h |
| 18:03.48 | blessen | Spackel: you tried your best and i did a lot..thanks a lot boss... |
| 18:04.03 | blessen | spackle: sorry you did a lot :-) |
| 18:04.19 | eKo1 | I don't see any debug on span 1 |
| 18:04.31 | eKo1 | weird. |
| 18:04.39 | blessen | spackle: i thank you for that..but i got to get this working soon.. |
| 18:04.40 | clint_ | Anyone off the top of their head know how to do 'account codes'? |
| 18:04.50 | harryvv | account codes? |
| 18:04.50 | spackle | blessen: you are very close. I to wonder are people using kphone successfully with Asterisk. |
| 18:05.09 | harryvv | http://edition.cnn.com/2005/TECH/internet/08/26/phones.e911.ap/index.html |
| 18:05.14 | eKo1 | pri show span 1 says "Status: Provisioned, In Alarm, Down, Active" |
| 18:05.17 | clint_ | account codes: dial number, second dialtone, dial account code used for billing sorting.. |
| 18:05.21 | harryvv | FCC extends deadline for VoIP providers |
| 18:05.47 | ManxPower | eKo1: never heard of that. |
| 18:05.49 | blessen | ManxPower: can you help me...my complete configuration is at http://www.voipuser.org/forum_topic_2005.html |
| 18:06.05 | ManxPower | blessen: I CANNOT help you. |
| 18:06.13 | jarrod | eKo: check linecode/fram and signaling |
| 18:06.17 | ManxPower | eKo1: what about pri show span 2 |
| 18:06.30 | ManxPower | jarrod: he has no alarms in zttool |
| 18:06.35 | eKo1 | span 2, 3, and 4 aren't there since I'm not using them |
| 18:06.37 | jarrod | oh |
| 18:06.47 | *** join/#asterisk |Barcode (n=barcode@h-68-165-204-41.chcgilgm.covad.net) |
| 18:06.56 | ManxPower | eKo1: unless, of course, you or asterisk is confused about which port is which span.... |
| 18:07.10 | eKo1 | OK, now zttool is show a red alarm. |
| 18:07.12 | eKo1 | wtf. |
| 18:07.29 | eKo1 | argh... |
| 18:07.31 | jarrod | here we go down a rabbit trail |
| 18:07.31 | jarrod | hah |
| 18:07.35 | ManxPower | I'll bet you have a bad crimp on the cable. |
| 18:07.39 | eKo1 | it was OK just a while ago. |
| 18:07.51 | eKo1 | I crimped that stupid cable with both hands very tight. |
| 18:08.15 | harryvv | manx, bandwith problems? |
| 18:08.25 | ManxPower | harryvv: HUH???? |
| 18:08.29 | eKo1 | OK, I guess I have to go over there and check that dang cable. |
| 18:08.51 | harryvv | a crimp in cat5 can upset the impedence of the cable and bottle neck the cable bandwith |
| 18:09.19 | spackle | haryvv: so can overweight electrons. |
| 18:09.23 | harryvv | twist or crimp some where on the cable can do that. |
| 18:09.24 | *** join/#asterisk groogs (n=greg@d221-73-237.commercial.cgocable.net) |
| 18:09.34 | ManxPower | harryvv: his line in bouncing between RED and OK |
| 18:09.44 | harryvv | ahh |
| 18:09.49 | harryvv | connection problem |
| 18:09.56 | harryvv | do a wiggle test |
| 18:11.43 | *** join/#asterisk jeobjeobjeob (i=jeobjeob@66.246.162.202) |
| 18:11.53 | *** join/#asterisk cinix (n=beecards@tor/session/x-81938a2c32a07fcf) |
| 18:12.23 | jeobjeobjeob | where would i get sounds from the chick who made the recordings used in asterisk? |
| 18:12.35 | blitzrage | ~thevoice |
| 18:12.40 | spackle | jeobjeob: thevoice.digium.com |
| 18:12.42 | nick125 | thevoice.digium.com |
| 18:12.49 | blitzrage | www.theivrvoice.com :) |
| 18:14.46 | ManxPower | jeobjeobjeob: her name is Allison |
| 18:15.24 | *** join/#asterisk Guggemand (n=irc@0x55513327.adsl.cybercity.dk) |
| 18:15.34 | BrianR | I'm thinking Polycom IP501's for the offices and Budgetones for the cube farm. Any thoughts? |
| 18:16.10 | ManxPower | BrianR: do you really hate the pepople in the cube farm THAT much? |
| 18:16.12 | spackle | BrianR: save yourself the headache and go all 501's or get 301's for the cubes. |
| 18:16.34 | ManxPower | I would not give my ex-wife a Budgetone. I'd give her a cup of cyanide, but not a Budgetone. |
| 18:17.04 | BrianR | spackle: 301's run around $140? |
| 18:17.19 | BrianR | Has anyone found a cheap source for the POE adaptors? Aparently the 501 and 501 don't have built in POE? |
| 18:17.27 | spackle | BrianR less if you buy bulk |
| 18:17.34 | ManxPower | BrianR: correct. you need the special polycom cable for PoE |
| 18:17.44 | BrianR | It's like $40 from voipsupply :( |
| 18:18.17 | ManxPower | BrianR: At least one company will sell you polycoms with PoE cable, without wall power adapter for the same price. |
| 18:18.26 | spackle | BrianR, the Unidens and the Sipura phone are in the $100 ballpark too, but the polycoms are so much nicer. |
| 18:18.30 | ManxPower | contact VoipSupply and/or Voxzilla. |
| 18:18.31 | riksta | are sangoma e1 cards supposed to show as "0000:03:03.0 Network controller: Unknown device 1923:0300" in lspci ?? |
| 18:18.45 | BrianR | spackle: Yes. They seem to be the nicest phones in the under $200 price range. |
| 18:19.16 | spackle | brianr: ask voipsupply for their quantity pricing. |
| 18:19.19 | frogy | can anyone get the "shared lines" feature on polycom working with asterisk? |
| 18:19.36 | BrianR | spackle: Not sure if I qualify - I only plan to buy about 30 phones total.. |
| 18:19.51 | *** join/#asterisk mago (n=maxgluck@200.109.166.172) |
| 18:19.55 | ManxPower | frogy: never tried. I beat my users until they admitted they really didn't need that feature. Only 1 of them died in the process. |
| 18:19.58 | BrianR | frogy: What does the share dlines feature do? |
| 18:20.53 | mago | hello, i'm looking for someone to install a calling card application on an asterisk server, any volunteer? |
| 18:21.07 | shmaltz | why would I get echo only when calling a pots? |
| 18:21.22 | ManxPower | shmaltz: Um, because only POTS can cause echo. |
| 18:21.23 | BrianR | ManxPower: what's that feature do anyway? |
| 18:21.24 | frogy | BrianR shared line is kind of one extension shared by multiple phone sets. |
| 18:21.27 | shmaltz | mago, yeah, here just paypal me the money first :)P |
| 18:21.43 | BrianR | frogy: Aah. Can't one play games with the dialplan to get the same effect? |
| 18:21.47 | ManxPower | BrianR: google the mailing list for "busy lamp field" or "BLF" |
| 18:21.49 | ManxPower | ~mailinglist |
| 18:21.50 | jbot | it has been said that mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 18:22.00 | mago | shmaltz, shure, first we have to agree on the terms |
| 18:22.14 | shmaltz | ManxPower, that much I know, but it only happnes when going SIP < > Asterisk <> PRI <> POTS and not when PRI <> Asterisk <> PRI <> POTS |
| 18:22.29 | BrianR | ManxPower: the polycoms have semi-working BLF support. If you use the buddy list feature it does a SIP subscribe and shows which extensions are in use. |
| 18:22.31 | shmaltz | mago, you said volunteer |
| 18:22.41 | ManxPower | shmaltz: No, the echo is still there, but it's SO short it sounds like "sidetone" when VoIP isn't involved. |
| 18:23.07 | harryvv | broanr worked with the 500s |
| 18:23.26 | shmaltz | ManxPower, when calling SIP <> Asterisk <> PRI <> Cell Phon (digital) No echo only when calling POTS |
| 18:23.37 | ManxPower | shmaltz: That would be expected. |
| 18:23.44 | harryvv | BrianR I mean have you worked with the 500s also? |
| 18:23.56 | shmaltz | ManxPower, as always thanks |
| 18:24.05 | BrianR | shmaltz: turn up the echo cancellation stuff in the zapata.conf file |
| 18:24.08 | jeobjeobjeob | when you call a goto, and it finds an invalid extension, does it return to the previous place? |
| 18:24.12 | frogy | BrainR, not quite, it because if someone seizes the lines, the rest cannot use it until it is free again. |
| 18:24.22 | shmaltz | BTW, did you get the setvar for sip.conf backed rolled at the end? |
| 18:24.22 | BrianR | harryvv: Yes. I have 4 of the IP500... |
| 18:24.22 | ManxPower | shmaltz: see http://www.cisco.com/univercd/cc/td/doc/cisintwk/intsolns/voipsol/ea_isd.htm |
| 18:24.28 | mago | seriously, anyone willing to install areski or astcc? let me know how much you charge... also appradius to work with phone extensions calling out... |
| 18:24.50 | shmaltz | BrianR, I did, and I have a TE406 with the onboard echo can and still |
| 18:24.57 | BrianR | frogy: Aah.. Maybe that makes more of a difference if you're trying to simulate a key system with pots lines or something. |
| 18:25.10 | ManxPower | shmaltz: Asterisk SHOULD be canceling out the echo. |
| 18:25.20 | harryvv | BrianR get all the buttons to work? |
| 18:25.26 | shmaltz | ManxPower, meaning? |
| 18:25.30 | BrianR | frogy: At least our key system has a PRI, so the concept of two sets being able to seize the same extension number won't seem so foriegn to users. |
| 18:25.43 | ManxPower | shmaltz: meaning that if you have the echocan daughter card and still have echo, call digium. |
| 18:26.19 | BrianR | harryvv: There was some stuff I couldn't make work - but aparetnly the IP500 doesn't actually have a web browser. |
| 18:26.25 | frogy | BrianR, exactly, and that's been required by customers all the time. |
| 18:26.34 | shmaltz | ManxPower, I emailed them already will wait for their response. Did you ever get that setvar patch for sip.conf to work with 1.0-9? |
| 18:27.06 | *** join/#asterisk cinix (n=beecards@tor/session/x-c37076dee6c25432) |
| 18:27.14 | *** join/#asterisk ttyp0 (n=ttyp0@112.Red-83-53-241.pooles.rima-tde.net) |
| 18:27.14 | cinix | Has anyone else had the problem where you can call out fine, and you can always hear the other end fine, but sometimes the other end can't hear you? And it changes from one phone call to the next, everything will work great, i'll hang up and dial again and they can't hear me. |
| 18:27.20 | BrianR | frogy: Maybe it's desirable for keeping the "feel" of a key system when replacing one with VOIP sets - but i'm not convinced it's got any other practical use. |
| 18:27.39 | shmaltz | cinix, Nat nat nat nat |
| 18:28.02 | shmaltz | frogy, use FOP |
| 18:28.04 | ManxPower | shmaltz: ended up not needing it. I upgraded my polycoms to 1.5.2 |
| 18:28.08 | cinix | oh hmmm. |
| 18:29.00 | shmaltz | ManxPower, I need ${BLINDTRANSFER} to work in 1.0-9 do you know who can help me? |
| 18:29.16 | *** join/#asterisk veteran (n=veteran@cpe-66-25-30-132.houston.res.rr.com) |
| 18:29.26 | *** join/#asterisk |Barcode (n=barcode@h-68-165-204-41.chcgilgm.covad.net) |
| 18:29.43 | *** join/#asterisk cynfab (n=cynfab@dm211.ipmvs.com) |
| 18:29.50 | veteran | is "leavewhenempty" in queues.conf only in cvs? |
| 18:29.58 | frogy | BrainR, it is more than just the feel. It's such a common feature that's in the PBX that users like to see. |
| 18:30.03 | cinix | If I use port forwarding on the IAX and SIP ports do you think it will still happen? |
| 18:30.11 | shmaltz | anybody know how to figure out in the dialplan if 302 redirect has been given by a SIP client? |
| 18:30.23 | frogy | shmaltz, I'm using FOP already. |
| 18:30.27 | *** part/#asterisk mago (n=maxgluck@200.109.166.172) |
| 18:30.51 | ManxPower | shmaltz: ${RDNIS} should be poulated. |
| 18:30.58 | shmaltz | frogy, its not a PBX feature, its a key feature |
| 18:31.01 | BrianR | frogy: The whole concept of lamps and line numbers is not a PBX thing - it's a key system thing |
| 18:31.29 | shmaltz | ManxPower, you mean when using 302?, hm, makes sense, thanks |
| 18:31.47 | ManxPower | shmaltz: see the pastebin I'm about to give you |
| 18:31.56 | shmaltz | I will |
| 18:32.11 | ManxPower | http://pastebin.ca/21252 |
| 18:33.00 | BrianR | On a PBX, it's definantly a "feel" thing. Key system style exclusive buttons are actually a feature-reduced version of the busy-lamp-field & speed dial you'd get on a hybrid key system or PBX with SIP SUBSCRIBE support. |
| 18:33.08 | *** join/#asterisk PupenoL (n=pupeno@200.123.183.89) |
| 18:33.25 | shmaltz | ManxPower, thanks for that I like the way you check for a blank var (always had this problem) I used to use ${LEN()} thants for that tip |
| 18:34.26 | *** join/#asterisk cinix (n=beecards@tor/session/x-d46c70502847868d) |
| 18:34.56 | cinix | I read that there are NAT options I can use in the configure files, what's my best course of action? |
| 18:35.00 | *** join/#asterisk zoo (i=nobody@ip-193-16.travedsl.de) |
| 18:35.05 | ManxPower | shmaltz: I think that example is from before $LEN existed. |
| 18:35.18 | shmaltz | I see |
| 18:35.27 | ManxPower | but it still works.... |
| 18:35.53 | shmaltz | ManxPower, have you ever tried using promiscrdir in sip.conf? |
| 18:36.00 | cinix | but it's not SIP I'm using by IAX, and I only am affected by outgoing voice, exact opposite as I'd expect in a NAT problem. hmm |
| 18:36.28 | shmaltz | cinix, what technology you using end to end? |
| 18:36.30 | ManxPower | shmaltz: no. I feel that is is an evil option |
| 18:36.48 | ManxPower | I don't want want users to be able to get out of the PBX |
| 18:37.07 | shmaltz | ManxPower, I agree with you, unless it would be configurable, right now I'm using the DP for all |
| 18:37.31 | cinix | soft sip client -> asterisk server -> iax pstn provider |
| 18:37.53 | ManxPower | OK. Pizza arrived. Once UPS arrives I'll officially be having a "good day" |
| 18:37.55 | shmaltz | cinix, beween sip and * you have nat? |
| 18:38.02 | shmaltz | LOL |
| 18:38.02 | cinix | no |
| 18:38.03 | cinix | lan |
| 18:38.20 | ManxPower | cinix: then you don't need anything special |
| 18:38.31 | shmaltz | cinix, then get a different provider, who is the provider? |
| 18:38.34 | cinix | voicepulse |
| 18:38.38 | shmaltz | lol |
| 18:38.45 | cinix | :-P okay why are you laughing |
| 18:39.09 | ManxPower | cinix: you're not doing something silly like running asterisk on your NAT router, are you? |
| 18:39.10 | shmaltz | because my wife just joked with me :) |
| 18:39.37 | cinix | ManxPower, yes, but it connects with IAX which I thought was fine. |
| 18:39.48 | ManxPower | cinix: so what is your specific problem? |
| 18:40.04 | cinix | I call and talk to someone works great, hang up, call someone else I hear them great, they can't hear me. |
| 18:40.20 | shmaltz | ManxPower, never realized its on the wiki |
| 18:40.22 | shmaltz | http://www.voip-info.org/wiki-asterisk+cmd+gotoif |
| 18:40.23 | shmaltz | scroll down to HINT |
| 18:40.31 | cinix | It's very irregular, I can go days with no problems, then days where no one can hear me, without any intervention on my part. |
| 18:40.40 | ManxPower | A huspand and wife are watching the news on television. There's a story about gay marriage on. The husband turns to his wife and asks "Haven't they suffered enough?" |
| 18:41.02 | shmaltz | lol |
| 18:41.22 | shmaltz | cinix, its a providers issue |
| 18:41.25 | ManxPower | men usually laugh at that joke, women usually just glare at me. |
| 18:41.27 | shmaltz | try nufone |
| 18:41.35 | harryvv | hehe |
| 18:41.43 | ManxPower | cinix: I strongly doubt changing providers will make any different |
| 18:41.43 | shmaltz | heh |
| 18:42.00 | *** join/#asterisk toddf (n=toddf@net-66-210-104-252.theshop.net) |
| 18:42.15 | shmaltz | ManxPower, I do think so based on experience with them |
| 18:42.33 | harryvv | gay marrige is something that should have never happened. |
| 18:42.50 | ManxPower | harryvv: I feel the same about straight marriage. |
| 18:43.13 | cinix | when I call fwd's echo server it always works |
| 18:43.31 | cinix | used to do that as soon as it wasn't working because I thought it was my mic or something. without fail never had a problem |
| 18:44.13 | harryvv | manx, marrige is of a religions meaning |
| 18:44.15 | harryvv | brb |
| 18:45.12 | ManxPower | harryvv: It SHOULD be a religions meaning, but is IT a religions and a govt meaning. |
| 18:45.31 | shmaltz | I think marrige is a commitment just like * is, there fore gay marrige is silly, who takes out the garbage in a gay marrrige? |
| 18:46.04 | ManxPower | I've been with my boyfriend for 6 years I think. |
| 18:46.05 | MikeJ[Laptop] | The Kids! |
| 18:46.16 | ManxPower | MikeJ[Laptop]: We finally agree on something! *grin* |
| 18:46.21 | shmaltz | lol |
| 18:46.24 | syle | its not a commitment, guys marry for passport or taxwriteoff and women for money like it always has been |
| 18:46.30 | harryvv | Some study showed that unmarried couples that live together break up far mor often then those who marry. |
| 18:46.30 | *** join/#asterisk myiagy (n=myiagy@200.138.215.78) |
| 18:46.37 | MikeJ[Laptop] | same ones who mow the lawn and pick up the house |
| 18:46.46 | harryvv | i want a house |
| 18:46.53 | *** join/#asterisk Shizzla (i=steve@d081054.adsl.hansenet.de) |
| 18:46.53 | harryvv | on the oregon coast :) |
| 18:46.59 | MikeJ[Laptop] | harryvv, you are making arguements for gay marriage now? |
| 18:47.01 | shmaltz | harryvv, of corse since married ppl are commited |
| 18:47.07 | harryvv | mike, nops |
| 18:47.09 | Delvar | im off home! |
| 18:47.11 | ManxPower | harryvv: I think that's because it's such a hassle to get divorced. |
| 18:47.20 | harryvv | mabey :) |
| 18:47.28 | *** join/#asterisk crich1999 (n=crich@port-212-202-0-21.dynamic.qsc.de) |
| 18:47.34 | MikeJ[Laptop] | harryvv, but you said married couples stay together... commitment is good right? |
| 18:47.50 | riksta | anyone got a sangoma t1/e1 card that can lend me a hand for a minute please? |
| 18:48.05 | MikeJ[Laptop] | sangoma does ;) |
| 18:48.06 | ManxPower | I'll bet the correlation is more of couples w/kids staying togather more than couples without kids, regardless of marriage status. |
| 18:48.19 | riksta | MikeJ[Laptop]: :) |
| 18:48.26 | ManxPower | riksta: Not many Asterisk users run Sangoma, so there are not a lot of people to help you. |
| 18:48.41 | spackle | yet..... |
| 18:48.42 | riksta | ManxPower: oh really? i was advised to get one from here :) |
| 18:48.51 | ManxPower | riksta: not from me. |
| 18:48.56 | MikeJ[Laptop] | studies also show that long term gay couples stary together more often than married strait couples... |
| 18:48.58 | ManxPower | I hear they are good cards. |
| 18:49.02 | ManxPower | Just not many people use them |
| 18:49.13 | riksta | i'm having trouble with the kernel patches it applys, missing symbols and such |
| 18:49.14 | MikeJ[Laptop] | that being said, I am happily married to a woman.. so what do I know ;) |
| 18:49.25 | Beirdo | MikeJ[Laptop]: studies can say whatever people want them to say, unfortunately |
| 18:49.33 | ManxPower | I was married once, a long time ago. |
| 18:49.35 | Beirdo | hard to find the truth |
| 18:49.41 | riksta | hey Beave |
| 18:49.42 | MikeJ[Laptop] | hmmmmm |
| 18:49.44 | riksta | Beirdo: even |
| 18:49.47 | MikeJ[Laptop] | kids are fun.. |
| 18:49.48 | Beirdo | hehe |
| 18:49.54 | harryvv | mike, yea..it makes the couple think more seriosly what thay are getting into when thay tie the knot. To often young couples dont look down the road and are only foccused on the getting married :) |
| 18:49.54 | MikeJ[Laptop] | I need to go hug mine |
| 18:49.55 | hardwire | snom is going to hate me |
| 18:49.55 | Beirdo | hey, riksta |
| 18:49.57 | hardwire | feature request after feature request |
| 18:49.59 | hardwire | just give me the damn code |
| 18:50.01 | hardwire | :) |
| 18:50.04 | Beirdo | MikeJ[Laptop]: good idea :) |
| 18:50.05 | spackle | I like women so much, I think I would still like them if I was one. |
| 18:50.14 | ManxPower | kids should be locked away until they turn 18 |
| 18:50.20 | spackle | LOL |
| 18:50.25 | hardwire | spackle: after the years it would take to get over liking yourself. |
| 18:50.25 | harryvv | manx, we were kids |
| 18:50.26 | MikeJ[Laptop] | I could never personally marry a guy... they are too difficult ;) |
| 18:50.29 | ManxPower | But a lot of people love their pet humans. |
| 18:50.29 | shmaltz | I believe that the longest living person was married to the oposite sex |
| 18:51.00 | spackle | schmaltz, usually their spouse is long dead though. |
| 18:51.09 | hardwire | so |
| 18:51.20 | hardwire | gay/lesbians live shorter lifespans? |
| 18:51.30 | shmaltz | also, that study is simply not true, since there is far more straight cpls that lived far longer than the longest gay couple did |
| 18:52.07 | ManxPower | If I cared that much about living longer I'd stop smoking, not get married. |
| 18:52.09 | harryvv | manx, kids are NOTHING compares to a kid that has autism. |
| 18:52.12 | shmaltz | just like it must be that there are 2 trees with the same amount of leafs on it |
| 18:52.13 | spackle | yeah, that's like saying grandstreams have more problems than polycoms. . . |
| 18:52.55 | *** join/#asterisk taz-^ (i=demon@195.38.75.178) |
| 18:53.36 | taz-^ | hi |
| 18:54.07 | clint_ | Ok, so now that we're done with the important stuff :) |
| 18:54.07 | shmaltz | taz-^, hello |
| 18:54.09 | Cresl1n | hey!? |
| 18:54.10 | Cresl1n | :-) |
| 18:54.18 | *** join/#asterisk rvhi (n=rv@66.175.65.89) |
| 18:54.21 | clint_ | Is voip-info.org just down, or is my ISP retarded? |
| 18:54.31 | *** join/#asterisk fugitivo (n=ajf@201.255.105.43) |
| 18:54.37 | ManxPower | clint_: Yes. |
| 18:54.39 | harryvv | manx, my step dad had to have his chest opened up and have 1/3 his lungs cut out from years of smoking. |
| 18:54.52 | harryvv | clint its probebly down |
| 18:55.04 | clint_ | ManxPower: I knew they were retarded... :) |
| 18:55.07 | ManxPower | harryvv: and all he had to do was get married! |
| 18:55.24 | harryvv | :) |
| 18:55.27 | harryvv | clint try this |
| 18:55.29 | harryvv | http://66.151.54.101/tiki-index.php |
| 18:55.36 | harryvv | mabey there dns is down |
| 18:55.40 | shmaltz | clnt_ you ISPs fault |
| 18:55.44 | ManxPower | If you think I'm an asshole online, wait until you see me when I'm quitting smoking. |
| 18:55.51 | Beirdo | hehehe |
| 18:55.58 | Beirdo | that does many people in, ManxPower |
| 18:56.07 | *** join/#asterisk Charlie88 (i=Charlie@client-82-2-46-113.manc.adsl.tesco.net) |
| 18:56.08 | hardwire | I just registered (206) 965-9506 with my voip provider |
| 18:56.09 | *** part/#asterisk Moc (n=mochouin@h66-201-214-109.gtconnect.net) |
| 18:56.13 | clint_ | ISP between me and them... Looping route... Ah well. |
| 18:56.16 | hardwire | and.. so.. I called it.. to see if I get a not found error |
| 18:56.21 | ManxPower | Beirdo: I need to get Welbutrin |
| 18:56.22 | *** join/#asterisk Moc (n=mochouin@h66-201-214-109.gtconnect.net) |
| 18:56.22 | harryvv | ohh hardware forgot your in seattle |
| 18:56.25 | hardwire | its the worlds largest adult entertainment network |
| 18:56.26 | hardwire | how nice |
| 18:56.31 | hardwire | harryvv: I am in Alaska |
| 18:56.45 | harryvv | i see thats a did |
| 18:56.45 | Beirdo | my friend in TN just went on Welbutrin... he was at 3-4 packs of Marlboro a day |
| 18:56.46 | hardwire | 9508 |
| 18:56.48 | hardwire | not 9506 |
| 18:56.49 | Beirdo | it will be rough |
| 18:56.53 | *** part/#asterisk Charlie88 (i=Charlie@client-82-2-46-113.manc.adsl.tesco.net) |
| 18:56.57 | ManxPower | hardwire: Alaska? That explains alot. |
| 18:57.08 | hardwire | ManxPower: i'm not from here you jerk. |
| 18:57.14 | *** join/#asterisk Bile_One (n=bile_one@pcp454527pcs.gillst01.ar.comcast.net) |
| 18:57.16 | ManxPower | Beirdo: it helped the last time I quit. |
| 18:57.17 | Beirdo | heh |
| 18:57.22 | shmaltz | I used to smoe 2-3 packs marlboro a day I quit Oct 2 years ago |
| 18:57.43 | cinix | od I just started talking to someone wasn't working at first, and it started working mid call. first time that happened. i'll couldn't make it even flicker and I was dancing all over the room thinking it was the headset |
| 18:57.49 | ManxPower | Beirdo: Of course at $120/month welbutrin isn't any cheaper than smoking |
| 18:57.54 | harryvv | shmaltz good for you get alot of your taste buds back? |
| 18:57.57 | Beirdo | heh, true |
| 18:58.01 | *** join/#asterisk zoa (n=k@213.91.216.136) |
| 18:58.01 | Beirdo | but a bit better for ya |
| 18:58.08 | shmaltz | my friends would laguh at me with my cig in my mouthe and both hads on teh keyboard my nose was always burned :) |
| 18:58.32 | shmaltz | harryvv, it's over now, don't want it, and I'm starting to hate the smell |
| 18:58.57 | harryvv | btw, one person who quite smoking back in the 30s? he name is lamay and while he died along ago, he took all his cigarett and alcohol money to buy cars. Today the lamay car collection is one of the largest in the world. |
| 18:59.30 | syle | what a bunch of bs |
| 18:59.32 | Beirdo | mmmm, booze |
| 18:59.48 | hardwire | I will destroy the old pbx.. call me if you have issues |
| 18:59.53 | hardwire | almost all my line indicators lit up |
| 18:59.57 | Beirdo | hehehe |
| 19:00.02 | ManxPower | I'll be back after I upgrade my cisco |
| 19:00.17 | shmaltz | ManxPower to what version? |
| 19:00.19 | jeobjeobjeob | what makes cisco better than pcom? |
| 19:00.27 | hardwire | its.. cisco. |
| 19:00.33 | shmaltz | anybody know if Uniden UIP200 supports autoanswer? |
| 19:00.40 | jeobjeobjeob | so its only brand name |
| 19:00.42 | hardwire | what makes nike better than reebook? |
| 19:00.45 | hardwire | bok :) |
| 19:00.49 | jeobjeobjeob | heh |
| 19:00.52 | hardwire | bok bok bok |
| 19:00.53 | harryvv | syle excuse me? thats a well known fact in Tacoma where I am from. |
| 19:00.56 | spackle | jeobjeobjeob: I don't think it does. |
| 19:01.05 | jeobjeobjeob | i meant functionally |
| 19:01.06 | puowvip | Uniden makes ip phones? |
| 19:01.16 | jeobjeobjeob | like, i can think of a few reasons why pcom is better than csco |
| 19:01.16 | puowvip | (Uniden isn't out of business yet?) |
| 19:01.27 | *** join/#asterisk earthsnd (n=phish@digium.com) |
| 19:01.28 | shmaltz | puowvip, yep, and the best ones for the price |
| 19:01.28 | syle | you a hick? |
| 19:01.40 | spackle | puoivip: yeah, much better than some of their other phones. |
| 19:01.57 | puowvip | Huh. ok |
| 19:02.08 | Beirdo | hardwire: cheaper slave labor of course |
| 19:02.24 | Bile_One | Does any one use SER for NAT traversal? |
| 19:02.38 | Beirdo | I use IAX for NAT traversal :) |
| 19:02.51 | Bile_One | I would like to, but have to use SIP. |
| 19:03.04 | shmaltz | imagine me giving you an emila address you can put into google talk, and when you make a call to it, it's like calling my main 800 number |
| 19:03.05 | shmaltz | well, it's going to happen soon |
| 19:03.12 | Beirdo | I'm SER is the best tool to help, but I don't use it |
| 19:03.20 | shmaltz | or my tech support hot line |
| 19:03.21 | bjohnson | Beirdo: thought you had to toss the cigars? |
| 19:03.29 | Beirdo | yeah, I did |
| 19:03.47 | Beirdo | last one I had was in TN almost 3 months ago |
| 19:03.54 | riksta | he still loves the scotch tho ;) |
| 19:03.59 | Beirdo | mmmm. |
| 19:04.01 | Beirdo | scotch |
| 19:04.05 | Bile_One | haa haa haa |
| 19:04.07 | riksta | hey bjohnson |
| 19:04.20 | bjohnson | got any left overs |
| 19:04.27 | Beirdo | nope, smoked em all |
| 19:04.30 | Beirdo | sorry |
| 19:04.42 | *** join/#asterisk kothog (n=kothog@S010600500480af12.gv.shawcable.net) |
| 19:04.47 | Beirdo | you want a decent small humidor, let me know |
| 19:04.48 | *** join/#asterisk dwmw2_gone (i=ctrlprox@baythorne.infradead.org) |
| 19:04.51 | queuetue | What's the default password on a grandstream 100? |
| 19:04.51 | *** part/#asterisk earthsnd (n=phish@digium.com) |
| 19:04.55 | Beirdo | holds about 20 |
| 19:05.28 | spackle | quetue: admin |
| 19:05.47 | queuetue | spackle: Thank you. I don't think I got a book. :) |
| 19:06.13 | Bile_One | You download the book at their site. |
| 19:06.18 | shmaltz | anybody here using zap show status? |
| 19:06.24 | *** join/#asterisk Darwin35 (n=kvirc@ip70-179-214-245.dl.dl.cox.net) |
| 19:06.38 | *** join/#asterisk cp5 (n=samy@128.sub-70-209-69.myvzw.com) |
| 19:06.39 | bjohnson | Beirdo: got a lawyer friend who could use one |
| 19:06.43 | cpatry | shmaltz: sure. |
| 19:06.47 | bjohnson | he's a cigar newbie |
| 19:06.56 | shmaltz | cpatry, what card do you have? |
| 19:07.07 | Beirdo | bjohnson: ahh, cool, it has a dried out humidifier, and a working humidity meter |
| 19:07.10 | Beirdo | :) |
| 19:07.18 | cpatry | X100P, TE410P, ZTDUMMY |
| 19:07.22 | cpatry | and T100P |
| 19:07.27 | Beirdo | so it will take a few days to get back to proper moisture |
| 19:07.31 | cpatry | why, is there any problem ? |
| 19:07.32 | Beirdo | but I have no need for it |
| 19:07.45 | *** join/#asterisk dwmw2_gone (i=ctrlprox@baythorne.infradead.org) |
| 19:08.05 | shmaltz | cpatry, on the one with the TE410, what does it show? what card is in? |
| 19:08.40 | cpatry | ? what card is in? huh? |
| 19:08.46 | shmaltz | mine shows just t4xxp |
| 19:08.50 | *** join/#asterisk jorgito (n=jorgito@snat2.arachne.czfree.net) |
| 19:08.54 | jorgito | hi everybody |
| 19:08.58 | bjohnson | Beirdo: where are you these days? TO or Ottawa? |
| 19:09.08 | Beirdo | bjohnson: Toronto. |
| 19:09.19 | shmaltz | cpatry this is my output: |
| 19:09.20 | shmaltz | Description Alarms IRQ bpviol CRC4 |
| 19:09.21 | shmaltz | T4XXP (PCI) Card 0 Span 1 OK 0 0 0 |
| 19:09.25 | Beirdo | until tomorrow, then I'll be away for a week in Puerto Rico |
| 19:09.28 | cpatry | ~pastebin |
| 19:09.29 | jbot | i heard pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 19:09.39 | bjohnson | Beirdo: I'll meet you there for the pickup |
| 19:09.45 | Beirdo | heh |
| 19:09.48 | bjohnson | whua ha ha |
| 19:09.50 | *** join/#asterisk dwmw2_gone (i=ctrlprox@baythorne.infradead.org) |
| 19:09.50 | Beirdo | I'm not bringing it to PR |
| 19:09.51 | Beirdo | hehe |
| 19:09.54 | Beirdo | she'd kill me |
| 19:10.16 | bjohnson | tell her it's your toiletries luggage |
| 19:10.27 | Beirdo | silly |
| 19:10.52 | harryvv | I say have the next voip convention at the oregon coast. Good write off :) |
| 19:12.20 | *** join/#asterisk DirtyD (n=rob@ool-18bce078.dyn.optonline.net) |
| 19:12.47 | *** join/#asterisk fiXXXerMe (n=kvirc@ip67-154-236-201.z236-154-67.customer.algx.net) |
| 19:13.21 | fiXXXerMe | Hi everyone. Is this the right channel to ask a few questions about asterisk? |
| 19:13.57 | DirtyD | Hi. I'm looking to create a more advanced IVR for asterisk, one that I could use to do automated credit card payment, etc. The basic extensions.conf files seems to lack key features. Is there another API that I could use? Has there been much work in this area? Perhaps a particular keyword I can google? |
| 19:13.59 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 19:14.24 | shmaltz | crazy ppl: |
| 19:14.25 | shmaltz | http://news.yahoo.com/s/ap/20050826/ap_on_fe_st/britain_human_zoo_3 |
| 19:15.29 | *** join/#asterisk earthsnd (n=phish@digium.com) |
| 19:15.42 | DannyF | DirtyD, you'd wanna look into AGI |
| 19:16.15 | DannyF | DirtyD, there is nothing you can't do |
| 19:16.34 | [hC] | its so tempting to try AMP, even though ive already gone and almost implemented everything they have |
| 19:16.47 | [hC] | the interface is just so handy |
| 19:16.55 | Bile_One | If one is trying to do nat traversal do I need to use a SIP proxy, or a STUN server? |
| 19:16.57 | DirtyD | ahh wonderful.. AGI, that's what I needed, kinda like a CGI ;) |
| 19:16.58 | DannyF | [hC], it's a nice app |
| 19:17.09 | DannyF | DirtyD, yupp ,) |
| 19:17.22 | syle | thats funny shmaltz, all they need to do is have live sex shows now and they'd made tons of money |
| 19:18.03 | fiXXXerMe | Would asterisk@home be a good choice for a business with about 35 employees? |
| 19:18.11 | [hC] | DannyF: just that i do so many things "my way" i fear it wont let me be creative enough |
| 19:18.46 | DannyF | [hC], well nothing says your installation have to be compatible with anyone elses :) |
| 19:19.36 | harryvv | fiXXXerMe ask Ariel |
| 19:19.59 | fiXXXerMe | Who is that? |
| 19:20.04 | harryvv | he has done 100 with not at home but with asterisk with asterisk portal manager |
| 19:20.22 | DirtyD | DannyF: Is there a way to include Voice recognition using AGI? |
| 19:20.31 | harryvv | amp is what he uses |
| 19:20.49 | Hmmhesays | i hate phone support omg |
| 19:20.53 | DannyF | Dibbler, saw someone working on it, but havent seen anything published |
| 19:21.01 | DannyF | DirtyD, * |
| 19:23.28 | mutilator | i still get PCI Master Abort errors |
| 19:23.44 | mutilator | my irq on the te110p isn't shared either |
| 19:23.57 | shmaltz | where in usr/src/asterisk is the source for the cli commands? |
| 19:24.13 | cpatry | cli.c |
| 19:24.29 | *** join/#asterisk gdsm (n=gdsm@e1-1.ns500-1.ts.milt.as9105.net) |
| 19:24.30 | cpatry | but theres a lot in almost each files |
| 19:24.38 | mutilator | running zttest tests and i get 99.98%-100% all the time |
| 19:25.22 | shmaltz | cpatry, where is the zap show status? |
| 19:25.35 | *** part/#asterisk Uberbot (n=Uberbot@pcp01880954pcs.sandia01.nm.comcast.net) |
| 19:25.38 | cpatry | chan_zap.c |
| 19:25.53 | cpatry | handle_zap_show_status |
| 19:26.07 | shmaltz | cpatry, thanks |
| 19:26.16 | *** join/#asterisk taz-^ (i=demon@195.38.75.178) |
| 19:26.29 | queuetue | On that same budgetone 100, It is reporting 403 - forbidden... Does that imply bad password, or something else? |
| 19:27.13 | harryvv | queuetue what extention is it assigned? |
| 19:27.29 | queuetue | harryvv: I assigned it 307 |
| 19:29.09 | queuetue | The sipura on 305 works fine... I can get to the web interface ... I'm not sur ehow much could go wrong with it at this point. :) |
| 19:30.41 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 19:31.49 | *** part/#asterisk zoo (i=nobody@ip-193-16.travedsl.de) |
| 19:34.12 | *** join/#asterisk slak- (n=alex@dsl093-236-129.hfd1.dsl.speakeasy.net) |
| 19:34.27 | queuetue | Wrong password - it works! :) |
| 19:34.38 | queuetue | Does it flash blue all the time? |
| 19:34.47 | slak- | hi, all of a sudden 4 of my sipura ata's deregistered froma asterisk, when i call their extensions it says the number is busy leave vm, and this is the log in asterisk console: Aug 26 16:26:21 NOTICE[5926]: app_dial.c:764 dial_exec: Unable to create channel of type 'SIP' |
| 19:34.52 | slak- | any idea wtf? |
| 19:35.12 | slak- | i tried to powercycle the ata units they fail to authenticate with asterisk |
| 19:35.12 | Netgeeks | what version of asterisk? |
| 19:35.22 | slak- | both places can ping eachother' |
| 19:35.27 | slak- | Netgeeks stable |
| 19:35.34 | Netgeeks | 1.0.9? |
| 19:35.45 | riksta | why can't i get to voip-info.org any more? |
| 19:35.47 | riksta | can anyone else |
| 19:35.48 | slak- | Asterisk CVS-v1-0-08/17/05-10:50:16 |
| 19:35.53 | Netgeeks | at the CLI, try the command 'sip reload' |
| 19:35.59 | Netgeeks | see if that helps |
| 19:36.11 | slak- | i tried restarting and reloading *' |
| 19:36.29 | pc4 | Ok guys... no bs... what is the best ata to buy for most practicle home uses? |
| 19:36.35 | pc4 | Counting price/value/firmware/whatever. |
| 19:36.37 | slak- | sipura |
| 19:36.47 | pc4 | slak- - Which model? |
| 19:36.48 | slak- | Netgeeks still cant register no idea why |
| 19:36.48 | Netgeeks | slak: no luck after restarting and reloading asterisk? |
| 19:37.03 | *** join/#asterisk PupenoL (n=pupeno@200.123.183.89) |
| 19:37.22 | Netgeeks | hrm, okay, sorry i was shooting for a known bug issue I thought it might have been. I have to run, doc appointment in 20 mins |
| 19:37.49 | queuetue | I have my first digium card. I'm about to power down and pop it in - what do I need to do after boot to set it up? (I am an Asterisk@home user.) |
| 19:38.09 | pc4 | slak- - Which model? And ebay? or some website? Where should I buy it? |
| 19:38.38 | Micro | Hi, I'm trying to use /var/spool/asterisk/outgoing, and when I move my call file in, I get "Unable to request channel, call failed reason 0"? |
| 19:39.17 | *** join/#asterisk rob314 (n=rob314@207.58.194.2) |
| 19:40.02 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-241-108.buckeyecom.net) |
| 19:41.01 | syle | interesting |
| 19:41.11 | syle | my cat likes celine dion |
| 19:41.15 | syle | hes falling asleep |
| 19:41.16 | *** join/#asterisk ryansc (n=ryansc@c-67-164-188-180.hsd1.co.comcast.net) |
| 19:41.24 | cinix | you consider that liking? |
| 19:41.38 | cinix | probably boring it into a sleep |
| 19:41.48 | syle | lol |
| 19:42.07 | hardwire | file: the spa-841 was good to you right? |
| 19:43.46 | shmaltz | thats what happens if you piggyback of someone elses wifi |
| 19:43.53 | Hmmhesays | anyone got their resume on monster.com on here |
| 19:43.57 | Hmmhesays | *in here |
| 19:44.12 | *** join/#asterisk paulc (n=Paul@216.187.75.190.novuscom.net) |
| 19:44.15 | Hmmhesays | get any job offers from it? |
| 19:44.26 | mutilator | no |
| 19:44.29 | Hmmhesays | <chuckle> |
| 19:44.30 | mutilator | i have no degree |
| 19:44.39 | Hmmhesays | <shrug> i've only got an aas in networking |
| 19:44.44 | Hmmhesays | what a piece of shit that thing is |
| 19:44.48 | paulc | nor me.. and look how I did.. |
| 19:44.50 | Beirdo | an ass? |
| 19:44.52 | paulc | oh wait.. yeah.. look :( |
| 19:45.05 | Hmmhesays | I could have taught every IT class I had to take to get that |
| 19:45.08 | mutilator | any degree is gold to empoloyers for some reason |
| 19:45.14 | paulc | what's your degree in? |
| 19:45.17 | mutilator | employers even |
| 19:45.20 | puowvip | I don't know *anyone* who has *ever* found a job with monster, dice, yahoo-hotjobs, etc. EVER. |
| 19:45.29 | puowvip | it's all bunch of BULL SHIT. |
| 19:45.39 | Beirdo | it's the man... |
| 19:45.43 | mutilator | well actaully |
| 19:45.43 | mutilator | come to think of it |
| 19:45.44 | Beirdo | homey don't play that |
| 19:45.51 | Hmmhesays | my official aas is for networking & microcomputer operating system |
| 19:45.51 | mutilator | i have gotten offers for parttime/contract things |
| 19:46.01 | mutilator | where a school or company needed like 100 pc's installed |
| 19:46.03 | mutilator | but thats about it |
| 19:46.08 | CoffeeIV_ | I know people who have gotten a job by posting their resume on a simple homepage -- make sure you submit it to google |
| 19:46.10 | Hmmhesays | I wouldn't do bitch work like that |
| 19:46.27 | mutilator | neither do i |
| 19:46.35 | paulc | Yeah - I've had a couple of email me after seeing my CV online.. but most of my stuff has come from referrals and happy customers etc :) |
| 19:46.38 | Delta34 | can someone answer some questions regarding a t1 setup vs a pri setup on *? |
| 19:47.08 | cinix | Google alreayd knows who you work for, and who you have worked for, and where you live, and your phone number |
| 19:47.20 | cinix | why submit more info to them? they should just auto create it for you |
| 19:47.23 | mutilator | heh i apply for jobs like crazy tho |
| 19:47.23 | Hmmhesays | and about that hefty chick you took home last night |
| 19:47.29 | mutilator | my app history in 10 pages |
| 19:47.32 | Delta34 | for a pri i only need to set 11 digits to my carrier |
| 19:47.44 | mutilator | 20 per page |
| 19:48.10 | Delta34 | for a t1 i hav to send 21 digits to the carrier, 11 digits for dnis and 10 digits for ani |
| 19:48.23 | CoffeeIV_ | I have no problem with google knowing everyhting as long as it stays this easy to put false information in it |
| 19:48.27 | paulc | Delta34: The difference between in band and out of band signalling? |
| 19:49.10 | *** join/#asterisk ManxPower (n=eric@dsl-209-16-67-160.datasync.com) |
| 19:49.55 | Delta34 | so for t1 u always need to send and receive extra digits? |
| 19:49.55 | Darwin35 | there are no jobs to be had in the USA that are helpdesk/tech Releated anymore. they have all been outsourced to asia |
| 19:50.15 | mutilator | heh |
| 19:50.19 | Darwin35 | at 1/16 the cost |
| 19:50.27 | mutilator | who wants to be a helpdesk tech tho |
| 19:51.03 | paulc | Delta34: No - the difference is in the signalling.. with PRI you send a call setup message that contains the presentation caller ID as well as number to dial.. T1 is more like a handshaking thing.. |
| 19:52.26 | Hmmhesays | no one in their right mind wants helpdesk |
| 19:53.00 | Darwin35 | I enjoyed being a helpdask tech and listening to people make ass of them selves |
| 19:53.10 | *** join/#asterisk jero (n=jero@savoirfairelinux.net) |
| 19:53.15 | jero | hi folks |
| 19:53.49 | Darwin35 | welcome jero |
| 19:53.58 | *** join/#asterisk Zeut (n=zeutzeut@ool-4353a8a5.dyn.optonline.net) |
| 19:54.04 | |Barcode | Darwin35: I hear you on that one. Yesterday one guy was trying to tell me that the RJ-45 connector was the wrong size for the NIX. |
| 19:54.07 | |Barcode | er.. NIC |
| 19:54.31 | Zeut | hello all, I have a question I hope is easy to answer. |
| 19:54.34 | Hmmhesays | speak of the devil |
| 19:54.39 | Hmmhesays | help desk call |
| 19:54.41 | ManxPower | shmaltz: I was upgrading the flash chip |
| 19:54.56 | Zeut | I have an AAH box setup and working, I 3 polycom 500 phones with latest SIP working. |
| 19:55.31 | Zeut | The phones can call each other, and used to be able to call Asterisk to check their voicemail, but now when ever * would play a recording to the phone, the phone doesn't hear anything. |
| 19:55.34 | Zeut | Any ideas? |
| 19:56.07 | Darwin35 | brb call |
| 19:56.38 | mutilator | god damn mother fscking pice of shit digium card! |
| 19:56.52 | Zeut | it's very weird, all this was working recently, but just stopped working, and I don't know why. |
| 19:58.45 | *** join/#asterisk mago (n=maxgluck@200.109.166.172) |
| 19:58.50 | Zeut | Aug 26 15:58:18 DEBUG[2856]: Call from user '201' is 1 out of 0 |
| 19:58.50 | Zeut | Aug 26 15:58:18 DEBUG[2856]: build_route: Contact hop: <sip:201@192.168.3.201> |
| 19:58.50 | Zeut | Aug 26 15:58:18 VERBOSE[2856]: -- Executing Answer("SIP/201-ec07", "") in new stack |
| 19:58.50 | Zeut | Aug 26 15:58:18 VERBOSE[2856]: -- Executing Wait("SIP/201-ec07", "1") in new stack |
| 19:58.51 | Zeut | Aug 26 15:58:18 DEBUG[2856]: Stopping retransmission on '5a108cbb-4ad81ef5-bf876d0a@192.168.3.201' of Response 2: Found |
| 19:58.51 | Zeut | Aug 26 15:58:19 VERBOSE[2856]: -- Executing VoiceMailMain("SIP/201-ec07", "default") in new stack |
| 19:58.53 | Zeut | Aug 26 15:58:19 DEBUG[2856]: Ooh, format changed from unknown to ulaw |
| 19:58.55 | Zeut | Aug 26 15:58:19 DEBUG[2856]: Scheduling timer at 160 sample intervals |
| 19:58.57 | Zeut | Aug 26 15:58:19 VERBOSE[2856]: -- Playing 'vm-login' (language 'en') |
| 19:58.58 | *** join/#asterisk fwae (n=pc@209.151.52.81) |
| 19:59.13 | Nivex | ~pastebin |
| 19:59.15 | jbot | [pastebin] a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 19:59.15 | Nivex | o.O |
| 19:59.20 | Nivex | oh, slow bot today |
| 19:59.30 | Zeut | oh sorry.... |
| 19:59.31 | Zeut | ok |
| 20:01.45 | Zeut | cool, I didn't know about pastebin.ca, very nice, anyway I have now posted my log to http://pastebin.ca/21261 if anyone would care to take a look. |
| 20:01.58 | Zeut | It's as though someone hit a mute button on the * server. Very weird... |
| 20:02.10 | riksta | anyone using a sangoma a101 |
| 20:02.52 | Cresl1n | not here |
| 20:02.53 | Cresl1n | :-P |
| 20:03.17 | *** join/#asterisk shido6 (n=shido@d57-87-253.home.cgocable.net) |
| 20:03.48 | riksta | Error: Device wanpipe1 is not supported by kernel << that really can't be good. Seeing as I compiled all the modules to the letter from the instructions :) |
| 20:04.06 | *** join/#asterisk queuetue (n=queuetue@h69-21-252-54.69-21.unk.tds.net) |
| 20:04.17 | shido6 | what were the kernel paramaters to disable acpi/apci ? |
| 20:04.24 | *** part/#asterisk DirtyD (n=rob@ool-18bce078.dyn.optonline.net) |
| 20:04.30 | riksta | acpi=off |
| 20:04.50 | riksta | pci=noacpi too |
| 20:04.52 | shido6 | its the line you put on the knerle |
| 20:04.57 | riksta | yeah |
| 20:04.59 | shido6 | at boot |
| 20:05.08 | *** join/#asterisk vagonetta (n=f@host194.200-43-12.telecom.net.ar) |
| 20:05.19 | riksta | ^^ |
| 20:05.32 | Hmmhesays | wahoo i've been waiting for these guys to call |
| 20:06.00 | vagonetta | opays y cafe 150 sobn lo pero que puede haber no lo vayan a usar nunca |
| 20:06.22 | shido6 | ~twisted |
| 20:06.27 | jbot | extra, extra, read all about it, twisted is twisted@indigent-networks.com, but you can paypal him at toastido@toastido.net |
| 20:06.51 | Zeut | ok, I have some more information. I just rebooted my server to different kernel that doens't have the zaptel modules built for, now the sound works fine to my digium phones. |
| 20:07.00 | Zeut | I would think that these issues are unrelated but apparently not. |
| 20:07.07 | *** join/#asterisk methos (n=lot@68.148.192.184) |
| 20:07.12 | *** part/#asterisk vagonetta (n=f@host194.200-43-12.telecom.net.ar) |
| 20:08.33 | *** join/#asterisk cianhughes (n=cian@cian.ws) |
| 20:09.05 | *** join/#asterisk MattH (n=MattH@63.174.244.174) |
| 20:09.05 | *** join/#asterisk Defraz (n=t0tal@tim.ibccom.net) |
| 20:09.16 | MattH | Hi... is there an example someplace of how one would go about pointing a DID number down an IAX pipe to another server? |
| 20:09.49 | Hmmhesays | and in english? |
| 20:10.23 | shido6 | dislplan logic |
| 20:10.24 | Sedorox | exten => <didnumber>,1,Dial(IAX/user@host/${EXTEN]) |
| 20:10.26 | Sedorox | exten => <didnumber>,1,Dial(IAX/user@host/${EXTEN}) |
| 20:10.48 | MattH | where user is what's defined in [] or the user registered on that machine? |
| 20:10.54 | shido6 | then give user at that host a context where that ${EXTEN} actually exists |
| 20:11.29 | Zeut | Ok, I Have reposted my asterisk log, this time once where it fails and once where it works, anyone see what might be going on here? http://pastebin.ca/21262 |
| 20:11.33 | Sedorox | user would be the [] in iax.conf |
| 20:11.45 | Sedorox | or... the username= part.. I foget.. I think the [] for remotes connecting |
| 20:12.23 | MattH | k thanks |
| 20:12.27 | Sedorox | yup |
| 20:12.31 | Sedorox | think of it this way |
| 20:12.36 | Sedorox | its like dialing a phone... |
| 20:12.43 | Sedorox | 'cept your pointing to a extention... |
| 20:12.53 | Sedorox | and the remote machine needs that extention in the dialplain... |
| 20:13.14 | ManxPower | riksta: I have a suggestion for you about your Sangoma card. |
| 20:14.07 | ManxPower | riksta: CONTACT SANGOMA TECH SUPPORT |
| 20:14.18 | Hmmhesays | the sangoma guys are pretty nice |
| 20:14.51 | *** join/#asterisk viLeR (i=1000@ip-47-252.telesat.com.co) |
| 20:15.04 | Hmmhesays | they won't bite your head off like I will if you call me |
| 20:15.07 | *** join/#asterisk lot (n=lot@68.148.192.184) |
| 20:15.16 | paulc | Hmmhesays is the new BOFH? |
| 20:15.25 | paulc | "Hmmhesays Helpdesk - please hold..." |
| 20:15.29 | paulc | (for eternity) |
| 20:15.33 | *** part/#asterisk rob314 (n=rob314@207.58.194.2) |
| 20:15.51 | ManxPower | "To continue in english enter the score for the 2006 World Series." |
| 20:15.53 | *** join/#asterisk Doofmannsgehilfe (n=chatzill@dsl-084-060-052-015.arcor-ip.net) |
| 20:15.56 | riksta | ManxPower: I am doing no need for caps |
| 20:15.57 | ManxPower | "Please wait...." |
| 20:16.00 | Hmmhesays | paulc <chuckle> |
| 20:16.04 | mago | hello, anyone knows a good postgre db client for xp? |
| 20:16.16 | Hmmhesays | I would put them in queue with yanni and michael bolton live |
| 20:16.26 | crash3m | I'd say phppgsql, but it sucks horribly IMHO |
| 20:16.34 | ManxPower | Hmmhesays: You ARE evil. |
| 20:16.52 | Hmmhesays | haha damn straight |
| 20:17.15 | paulc | LMAO :D YANNI! |
| 20:17.21 | Hmmhesays | and I like long walks on the beach, and a girl who can kick my ass at pool |
| 20:17.30 | Hmmhesays | or at least come close |
| 20:17.57 | paulc | oh dear.. only one inflatable life saving ring.. |
| 20:18.00 | paulc | which way to throw it? |
| 20:18.26 | mutilator | everyone needs a terrawind! |
| 20:18.29 | Hmmhesays | file is there a module for ser to send register requests round robin style a few different registrars? |
| 20:18.32 | mutilator | http://www.terrawind.com/terrawind.htm Terra Wind Prices from the $850,000's as shown $1.2M |
| 20:18.36 | mutilator | thats pretty badass tho |
| 20:18.38 | Hmmhesays | i'm not seeing anything like that |
| 20:18.55 | Hmmhesays | I want this www.sunnyleone.com (NOT SAFE FOR WORK) |
| 20:19.06 | Hmmhesays | or those offended by nudity |
| 20:19.21 | file | Hmmhesays: WELL... no |
| 20:19.35 | *** join/#asterisk Exstatica (i=exstatic@static-71-116-196-11.lsanca.dsl-w.verizon.net) |
| 20:19.37 | paulc | hmm.. sunny leone.. versus www.ghoststalks.com - she's got nothing on Da Juana! :) |
| 20:19.45 | paulc | uh.. www.ghoststalk.com even |
| 20:19.58 | MattH | hrmm well the call is going |
| 20:19.59 | Hmmhesays | LOL |
| 20:19.59 | paulc | Terra Wind looks cool |
| 20:20.01 | Hmmhesays | um... |
| 20:20.02 | MattH | called voipswitch2/200 |
| 20:20.09 | MattH | but it just sits there... odd no errors on the receiving side |
| 20:20.33 | jeobjeobjeob | hey |
| 20:20.43 | jeobjeobjeob | why does directory die on me? |
| 20:20.46 | jeobjeobjeob | i tried using director |
| 20:20.48 | jeobjeobjeob | y |
| 20:20.49 | jeobjeobjeob | it finds the names |
| 20:20.52 | *** join/#asterisk zeedo (n=zeedo@www.reboot-robot.net) |
| 20:21.01 | jeobjeobjeob | but when i ask it to dial the number (using 1) it plays dir-nomore |
| 20:21.05 | Hmmhesays | are you using welsh names in it? |
| 20:21.13 | *** join/#asterisk cekc (n=sean@rrcs-24-199-36-210.west.biz.rr.com) |
| 20:21.14 | paulc | LOL |
| 20:21.19 | paulc | those bloody welsh - they get everywhere! |
| 20:21.20 | jeobjeobjeob | heh |
| 20:21.37 | paulc | clanfairpwl*mumble*agogagogagoch |
| 20:21.51 | Hmmhesays | jeobjeobjeob, the extension doesn't exist in the context your are in |
| 20:21.51 | jeobjeobjeob | it finds names |
| 20:21.57 | Hmmhesays | lol paulc |
| 20:22.04 | Hmmhesays | to many vowels in there though |
| 20:22.11 | jeobjeobjeob | wait, i call directory(vm) |
| 20:22.12 | cekc | I have asterisk running now with the sample configuration, where do I go to configure it so that I can dial it from a SIP phone? |
| 20:22.21 | jeobjeobjeob | and it finds the names |
| 20:22.24 | Hmmhesays | extensions.conf/sip.conf |
| 20:22.33 | jeobjeobjeob | but voicemail works fine |
| 20:23.24 | Hmmhesays | your point being? |
| 20:23.45 | harryvv | saw a hydrogen fuel cell car yesterday. it was # 2 V-FC for the city of vancover. that thing was fast :) |
| 20:24.05 | Hmmhesays | i saw a '69 chevelle yesterday that was pretty quick |
| 20:24.20 | *** join/#asterisk MikeJ__ (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 20:24.25 | harryvv | Hmmhesays what makes this interestng is the amount of steam that comes out the tail pipe |
| 20:24.34 | jeobjeobjeob | hmm |
| 20:24.38 | jeobjeobjeob | well |
| 20:24.44 | paulc | #2 V-FC? |
| 20:24.47 | harryvv | yea |
| 20:24.48 | jeobjeobjeob | exten => #,1,Directory(cust-u1005) |
| 20:24.51 | Hmmhesays | paste the relevent portions of your dialplan dude |
| 20:24.55 | Hmmhesays | on www.pastebin.ca |
| 20:25.00 | Hmmhesays | and fedex me a sammich |
| 20:25.16 | harryvv | I think V stands for vancouver and FC stands for fuel cell. So this is vancouver cities # 2 fuel cell car. A ford focus. |
| 20:25.16 | Hmmhesays | while I crank up the will smith and take my pants off |
| 20:25.26 | harryvv | Car probebly cost $150,000 |
| 20:26.09 | *** part/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 20:26.15 | harryvv | Paul, yea it was in front of me on highway 99 and turned off onto the knight street bridge in a pretty fast clip of 120 kph |
| 20:26.29 | Hmmhesays | whats that like 65mph? |
| 20:26.45 | harryvv | 100 kph is 62 mph |
| 20:26.51 | jeobjeobjeob | http://pastebin.ca/21265 |
| 20:26.53 | Hmmhesays | so closer to 75mph |
| 20:27.01 | jeobjeobjeob | those are the relevant entries |
| 20:27.05 | *** join/#asterisk apeman (n=cmurray@zoidberg.digital-storm.net) |
| 20:27.16 | *** join/#asterisk [1]JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net) |
| 20:27.24 | harryvv | my next car will be a older toyota diesel |
| 20:27.25 | harryvv | :) |
| 20:27.36 | bendy24 | toyata mkes diesel? |
| 20:27.51 | *** join/#asterisk lot (n=lot@68.148.192.184) |
| 20:28.05 | Hmmhesays | does extension 136 exist in context cust-u1005 ? |
| 20:28.08 | harryvv | the toyota imported diesel landcruisers and trucks up to 1989 |
| 20:28.18 | bendy24 | bah, just get a vw tdi |
| 20:28.25 | harryvv | there are alot of right hand diesels made after that period. |
| 20:28.27 | Hmmhesays | I want a shopping card with an oldsmobile small block 350 |
| 20:28.31 | harryvv | tdi does not have a bed. |
| 20:28.32 | Hmmhesays | *shopping cart |
| 20:28.54 | bendy24 | true, no bed for vw's |
| 20:29.08 | harryvv | perfer it be a truck |
| 20:29.17 | jeobjeobjeob | yeah |
| 20:29.23 | jeobjeobjeob | wait |
| 20:29.31 | jeobjeobjeob | context in the extensions.conf or voicemail |
| 20:29.36 | *** part/#asterisk earthsnd (n=phish@digium.com) |
| 20:29.36 | bendy24 | the vw toureg has a 10 cylender diesel |
| 20:29.45 | bendy24 | its truckish |
| 20:29.55 | harryvv | and toyotas are known to be tough and reliable...ask a Cia agent or Taliban members in Afganastan. Thay drive them. |
| 20:29.55 | Hmmhesays | jeobjeobjeob, in extensions.conf |
| 20:29.56 | harryvv | :) |
| 20:30.04 | jeobjeobjeob | why do i need one in extensions.conf? |
| 20:30.23 | Hmmhesays | iirc thats where your call is going to go |
| 20:31.03 | Hmmhesays | exten => 136,1,DIAL(SIP/BOBBARKER) |
| 20:32.16 | *** join/#asterisk xtrvd (n=test@s207-6-25-182.bc.hsia.telus.net) |
| 20:32.23 | Hmmhesays | da juana byrd isn't nearly as hot as sunny leone |
| 20:32.57 | Hmmhesays | although I haven't seen her nekkid |
| 20:33.17 | paulc | LMAO :D |
| 20:33.18 | paulc | ewww |
| 20:33.19 | paulc | don't go there! |
| 20:33.32 | paulc | she's crazy but nice.. always makes me laugh when she phones me up.. strong texas accent.. |
| 20:33.57 | *** join/#asterisk newmember (n=newmembe@S010600a0c93dce87.cg.shawcable.net) |
| 20:33.57 | xtrvd | I currently have a configuration for an IAX service provider on my asterisk box, but I've signed up for a free Italian number from mytcom.it and I was wondering how I go about setting up an incoming SIP account on Asterisk in conjunction with my current IAX provider. |
| 20:34.24 | Hmmhesays | theres a loaded question with not enough info |
| 20:34.44 | jeobjeobjeob | but ... |
| 20:35.02 | Hmmhesays | xtrvd: do you want to register with a sip provider? |
| 20:35.02 | riksta | xtrvd: you can just copy a configuration of how everyone else does it, eg FAQ on sipgate.co.uk or voipuser.org |
| 20:35.22 | Hmmhesays | my nose is running |
| 20:35.30 | riksta | run after it! |
| 20:35.39 | xtrvd | Hmmhesays: I want to register with the SIP Provider, yes, but I don't want to change how my current IAX system is running. |
| 20:35.41 | Hmmhesays | whoa, Male stripper kills drag queen with samurai sword |
| 20:35.50 | Hmmhesays | xtrvd: why would you have to do that |
| 20:35.53 | xtrvd | I just want the SIP to add onto my current service, |
| 20:35.59 | Hmmhesays | ok, whats the problem? |
| 20:36.13 | Hmmhesays | http://www.newsnet5.com/news/4901720/detail.html |
| 20:37.00 | xtrvd | I'm trying to figure out how this configuration is going to look like. How does one get the SIP account to register to the sip.mytcom.it servers? |
| 20:37.19 | xtrvd | Hmmhesays: That's not right... |
| 20:37.22 | xtrvd | =) |
| 20:37.24 | *** join/#asterisk RoldyxRoot (n=root@201.255.104.166) |
| 20:37.28 | RoldyxRoot | hello |
| 20:37.52 | Hmmhesays | xtrvd: in sip.conf |
| 20:38.06 | xtrvd | Adding the user in sip.conf registers with the provider? |
| 20:38.08 | Hmmhesays | look at the config for broadvoice or fwd one of the million other providers out there |
| 20:38.14 | Hmmhesays | no.. |
| 20:38.18 | Hmmhesays | ~fwd |
| 20:38.19 | jbot | well, fwd is Free World Dialup: Brainchild of Jeff Pulver. URL: http://www.pulver.com/fwd/ |
| 20:38.29 | Hmmhesays | ~wiki |
| 20:38.36 | Hmmhesays | damnit |
| 20:38.36 | cpatry | ~wikis |
| 20:38.38 | jbot | wikis is probably http://www.voip-info.org |
| 20:38.44 | Hmmhesays | thanks junk-Y |
| 20:38.50 | cpatry | np |
| 20:38.50 | Hmmhesays | er... cpatry |
| 20:38.54 | cpatry | :) |
| 20:39.03 | Hmmhesays | xtrvd: check out the sample for free world dialup |
| 20:39.03 | riksta | xtrvd: you can just copy a configuration of how everyone else does it, eg FAQ on sipgate.co.uk or voipuser.org |
| 20:39.22 | xtrvd | *sigh* I've been looking at the FWD setup and numerous others for the past hour but haven't figured it out; that's why I came here, |
| 20:39.30 | xtrvd | But I'll see if i can find a more specific question. |
| 20:39.31 | Hmmhesays | what can't j00 figure out? |
| 20:40.33 | xtrvd | After I've added the entry into the sip.conf complete with the 'general' context, how does one specify how to route the incoming call> |
| 20:40.35 | xtrvd | ?* |
| 20:40.38 | *** join/#asterisk websae (i=websae@207-118-143-134.dyn.centurytel.net) |
| 20:40.45 | Hmmhesays | with a peer/user entry |
| 20:40.46 | RoldyxRoot | i need make a throbleshooting about it |
| 20:40.57 | Hmmhesays | just like it shows in the fwd config on voip-info |
| 20:41.30 | RoldyxRoot | i ve a lot of problem with asterisk |
| 20:41.42 | Hmmhesays | heh |
| 20:41.48 | Hmmhesays | is there an echo in here? |
| 20:42.00 | RoldyxRoot | s???? |
| 20:42.12 | Hmmhesays | <chuckle> we've all probably said that at one point or another |
| 20:42.18 | Hmmhesays | whats your major malfunction son |
| 20:42.28 | *** join/#asterisk zotz (n=zotz@24.231.36.100) |
| 20:42.35 | file[lap1op] | MUFFIN |
| 20:42.42 | Hmmhesays | rmmod full metal jacket |
| 20:42.45 | *** join/#asterisk cjk (n=cjk@212.233.32.4) |
| 20:42.49 | RoldyxRoot | Hmmhesays, sorry i speak spanish |
| 20:42.51 | Hmmhesays | rmmod, wtf am I thinking |
| 20:42.51 | file[laptop] | oh god the phone |
| 20:43.26 | Hmmhesays | RoldyxRoot, what issue are you having |
| 20:44.22 | RoldyxRoot | dial 113 |
| 20:44.22 | RoldyxRoot | <PROTECTED> |
| 20:44.22 | RoldyxRoot | <PROTECTED> |
| 20:44.22 | RoldyxRoot | <PROTECTED> |
| 20:44.27 | mago | any user of appradius? |
| 20:44.32 | RoldyxRoot | .............. |
| 20:44.46 | RoldyxRoot | -- Executing Hangup("ALSA/default", "") in new stack |
| 20:44.46 | RoldyxRoot | <PROTECTED> |
| 20:44.46 | RoldyxRoot | <PROTECTED> |
| 20:45.02 | RoldyxRoot | but i dont ear nothing |
| 20:45.09 | websae | anyone running CENTOS in here? |
| 20:45.40 | Hmmhesays | every aah user is running centos |
| 20:46.03 | xtrvd | 'aah' user? |
| 20:46.13 | Hmmhesays | asterisk at home |
| 20:46.18 | xtrvd | ahh |
| 20:46.31 | Hmmhesays | indeed |
| 20:46.31 | RoldyxRoot | Hmmhesays, tell me what do you think? |
| 20:46.54 | Hmmhesays | RoldyxRoot, not a clue sorry |
| 20:47.03 | websae | anyone using the AAH? |
| 20:47.19 | Hmmhesays | you seem to have a question websae what is it? |
| 20:47.49 | *** join/#asterisk Uberbot (n=Uberbot@pcp01880954pcs.sandia01.nm.comcast.net) |
| 20:47.52 | RoldyxRoot | somebody speak spanish? |
| 20:47.57 | *** join/#asterisk johnASG (n=jon@border0hsv.asterisksgi.com) |
| 20:48.02 | *** join/#asterisk ManxPower (n=eric@dsl-209-16-67-160.datasync.com) |
| 20:48.03 | RoldyxRoot | alguien habla español? |
| 20:48.06 | Uberbot | Hi all. |
| 20:48.16 | fugitivo | RoldyxRoot: yo |
| 20:48.45 | websae | hola me llamo brandon |
| 20:48.53 | websae | como estats RoldyxRoot |
| 20:48.58 | websae | Que necesitas? |
| 20:49.58 | bjohnson | un cervesas |
| 20:50.01 | RoldyxRoot | websae, no puedo escuchar absolutamente nada |
| 20:50.18 | RoldyxRoot | websae, skype funciona bien |
| 20:51.08 | websae | tu solomente quieres hablar sobre asterisk |
| 20:51.18 | eKo1 | what the hell, since when are we speaking español here. |
| 20:51.19 | websae | estas usando asterisk ? |
| 20:51.21 | xtrvd | If I do a 'register => username:password@sip.myTCom.it/116' in my sip.conf, that will forward the calls to extension 116 in my extensions.conf ? |
| 20:51.56 | eKo1 | ¿quien, yo? |
| 20:53.11 | RoldyxRoot | websae, sip, pero no puedo escuchar mi contestador |
| 20:53.55 | xtrvd | Does anybody have an answer for that last question regarding the end of the register command in the sip.conf? |
| 20:53.59 | eKo1 | *contestadora |
| 20:54.04 | ManxPower | xtrvd: that will request the provider send the calls to extension 116 on your server. Many providers ignore the request. |
| 20:56.13 | xtrvd | Hmm, I see. I'm trying to figure out how I can have my newly created SIP account actually connect 'into' my * box for redirection. I can't seem to figure out where to go after the registration in sip.conf |
| 20:56.48 | cekc | this sip stuff has me stumped too |
| 20:56.59 | cekc | i've only had asterisk for a few hours though |
| 20:57.04 | xtrvd | hehe |
| 20:57.28 | xtrvd | I've found the wiki has been a godsend, but I'm trying to route incoming calls only, and I can't figure it out |
| 20:58.13 | Hmmhesays | register, put an extension in your default context for the incoming did |
| 20:58.23 | Hmmhesays | wham bam, quick and dirty way |
| 21:01.34 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 21:01.41 | *** join/#asterisk roulduke_ (i=0nx36t89@p508D0C3B.dip0.t-ipconnect.de) |
| 21:01.56 | endom0rph | is it possible to have an extension reroute on busy/noanswer to a oh323 external voice processing system passing the original called extension number and reason(busy/noanswer). I get the extension number of the h323 when I just add a Dial on busy no-answer. Do I need a different command/macro? thanks |
| 21:02.10 | *** join/#asterisk mhnoyes_ (n=mhnoyes@user-2ivflc1.dialup.mindspring.com) |
| 21:02.16 | *** join/#asterisk surfdue (n=surfdue@unaffiliated/surfdue) |
| 21:04.06 | Druken | endom0rph: i'm not sure what your asking... |
| 21:04.14 | xtrvd | endom0rph: I don't quite understand your question either... |
| 21:04.42 | endom0rph | ok basically trying to use external voicemail and voice recognition |
| 21:04.47 | Hmmhesays | well folks i'm out of here for awhile |
| 21:04.59 | endom0rph | I need to know which extension the call came from |
| 21:05.36 | fugitivo | endom0rph: ${CALLERID} ? |
| 21:05.37 | Druken | well, your dialing an oh323 right? why not just dial the extension ? |
| 21:06.01 | endom0rph | I tried CALLERID but that didn't work |
| 21:06.43 | endom0rph | I dial the extension with a 5 sec time out then dial the external system, but it doesn't automatically add the CALLERID |
| 21:06.55 | ManxPower | If there is callerid information it will be in ${CALLERID} |
| 21:06.56 | fugitivo | endom0rph: what external system? |
| 21:07.06 | websae | anyone here running CENTOS? |
| 21:07.09 | endom0rph | external voicemail |
| 21:07.11 | websae | the at home asterisk? |
| 21:07.18 | fugitivo | why are you using external voicemail? |
| 21:07.33 | harryvv | web i am |
| 21:07.45 | Druken | endom0rph: can't you dial the entension on the remote system ?? oh323/host/extension ? |
| 21:07.54 | endom0rph | I am testing the external voicemail and wanted to use a software pbx |
| 21:08.07 | ManxPower | ender: Um, then nothing is sending the callerid info along with the call |
| 21:08.50 | endom0rph | ManxPower- not sure I am explaining it well :) |
| 21:09.08 | endom0rph | Druken - yes I can dial direct and that works |
| 21:09.11 | fugitivo | endom0rph: how are you dialing the extension and then the external system? |
| 21:09.20 | ManxPower | endom0rph: that's obvious |
| 21:09.27 | RoldyxRoot | fugitivo, me has abandonado |
| 21:09.33 | RoldyxRoot | :( |
| 21:09.57 | fugitivo | RoldyxRoot: el cliente de irc te bloqueo por 10 minutos porque me enviaste demasiados mensajes seguidos, perdon |
| 21:10.06 | Druken | endom0rph: so i'm missing the problem then.... is it a CID problem ? |
| 21:10.20 | RoldyxRoot | fugitivo, ok |
| 21:10.36 | fugitivo | RoldyxRoot: entra a #asterisk-es |
| 21:10.43 | endom0rph | fugitivo - I have a exten Dial to the phone with a 5 sec time out then a dial to the oh323 number after it times out |
| 21:11.39 | ManxPower | ender: then you do not have a callerid= line in zapata.conf for that extension |
| 21:11.40 | fugitivo | the oh323 number is the external system? |
| 21:11.48 | *** join/#asterisk jeffik (i=jeffik@CPE00c049565af7-CM0012256ead9e.cpe.net.cable.rogers.com) |
| 21:12.03 | ManxPower | ender: What DEVICE are you dialing from? |
| 21:12.04 | endom0rph | scenario 201 dials 200..... 200 doesn't answer.... the exten then dials 555 (the oh323 external system) |
| 21:12.06 | Druken | ManxPower: why must people complicate things? |
| 21:12.06 | ManxPower | i.e. the phone |
| 21:12.13 | endom0rph | ManxPower x-lite |
| 21:12.19 | *** join/#asterisk dasuberdavid (n=david@digium.com) |
| 21:12.28 | ManxPower | endom0rph: and what is the callerid= line for the sip.conf entry for the x-lite? |
| 21:12.35 | ManxPower | Druken: 'cause they are lazy. |
| 21:12.54 | Druken | ManxPower: i usually simplify things because i'm lazy.... |
| 21:13.04 | fugitivo | lol |
| 21:13.05 | ManxPower | Druken: that's the way normal people do it. |
| 21:13.22 | endom0rph | ManxPower - Callerid= unknown |
| 21:13.34 | ManxPower | endom0rph: why? |
| 21:13.44 | ManxPower | endom0rph: no wonder nothing is getting the correct callerid. |
| 21:13.53 | ManxPower | callerid=Robert Dobbs <666> |
| 21:14.10 | endom0rph | this is under general |
| 21:14.12 | fugitivo | endom0rph: what is the result of ${CALLERID} when calling the phone, and what is the result when calling the external system? |
| 21:14.17 | ManxPower | ender: That is not what I asked. |
| 21:14.26 | ManxPower | I said "the sip.conf entry for x-lite" |
| 21:14.45 | endom0rph | ManxPower - sorry I am new to it :) |
| 21:14.50 | ManxPower | fugitivo: you skipped a LOT of troubleshooting. |
| 21:15.02 | ManxPower | endom0rph: I'm waiting for the answer. |
| 21:15.13 | endom0rph | ok it says callerid="Chris"<200> |
| 21:15.56 | ManxPower | try callerid=Chris <200> |
| 21:16.16 | ManxPower | you need a space before the < and some SIP devices refuse to accept the quotes. |
| 21:16.26 | riksta | when i ping voip-info.org i get time to live exceeded, and i cant view the website!!! |
| 21:16.33 | ManxPower | <ManxPower> callerid=Robert Dobbs <666> <-- correct, pedantic format |
| 21:16.35 | *** join/#asterisk zeedo (n=zeedo@www.reboot-robot.net) |
| 21:17.04 | *** part/#asterisk xtrvd (n=test@s207-6-25-182.bc.hsia.telus.net) |
| 21:17.30 | ManxPower | endom0rph: once you set that and issue a "reload" define an extension that just says something like exten => 299,1,Noop(CALLERID=${CALLERID}) |
| 21:17.43 | endom0rph | ManxPower - that had no change |
| 21:17.53 | RoldyxRoot | fugitivo, me volvieorn a baniar? |
| 21:17.53 | ManxPower | then dial 299 x-lite and watch the Asterisk console for the informative noop message with the current callerid info |
| 21:17.57 | mago | can anyone tell me how to change a postgresql user password? |
| 21:18.06 | endom0rph | ok |
| 21:18.10 | ManxPower | mago: I'll bet someone on the postgress channel can |
| 21:18.33 | Druken | postgres is a cunt for permissions |
| 21:18.34 | mago | thx |
| 21:18.35 | ManxPower | paste the one noop line, as printed on the asterisk console |
| 21:19.11 | ManxPower | I need a car in my driveway. Apparently many people think nobody is home if there's no car in the driveway. |
| 21:19.47 | Druken | ManxPower: make one out of cardboard :) |
| 21:20.07 | Druken | or put a sign up, "Yes, Were Home!" |
| 21:20.38 | Druken | make it look like those yes were open signs :) hehe |
| 21:22.00 | ryansc | has anyone used the x100p clone cards from ebay? |
| 21:22.07 | Druken | i do |
| 21:22.18 | ryansc | have you had people tell you the audio is really low? |
| 21:22.30 | ryansc | by probably 50 percent? |
| 21:22.32 | ManxPower | Druken: I suppose I could just buy a car. |
| 21:22.33 | Druken | yes, i did.. |
| 21:22.44 | ManxPower | I bought mine from newegg |
| 21:22.47 | Druken | ManxPower: do you need one? |
| 21:22.48 | ManxPower | $p |
| 21:22.49 | ryansc | is there a way around that problem? |
| 21:22.55 | ManxPower | Druken: Yes, but I don't have a license. |
| 21:23.07 | ManxPower | Having a car in the driveway might motivate me to get one. |
| 21:23.10 | Druken | ryansc: yeah, boost the outgoing audio |
| 21:23.21 | Druken | ManxPower: too true |
| 21:23.21 | clint_ | Allright, I asked this before, but now that I've spent an hour with google, perhaps someone else's here that knows... Any ideas on long distance account codes like those used in law offices, etc.? i.e. lift handset -> dial LD number -> second dialtone -> enter account code -> code appears in CDR with call... |
| 21:23.22 | ryansc | that will do it? I must have been changing the wrong file then |
| 21:23.31 | Druken | i should get mine back... since i drive for a living... |
| 21:23.40 | ryansc | what file is the audio control in? |
| 21:23.45 | ManxPower | Druken: Greyhound closed the local bus station so my transportation options are VERY limited. |
| 21:24.02 | Druken | ouch... |
| 21:24.07 | Druken | where do you live again ? |
| 21:24.51 | endom0rph | ManxPower - --Executing NoOp("SIP/201-61c5", ""Chris External" <201>") in new stack |
| 21:25.17 | Druken | take those quotes out |
| 21:25.23 | ryansc | Drunken: where was the audio control section located? |
| 21:25.38 | Druken | zapata.conf |
| 21:26.54 | endom0rph | ManxPower - --Executing NoOp("SIP/201-a082", "Chris External <201>") in new stack |
| 21:27.32 | lathos42 | anthm: Well, I successfully wrote an AGI, albeit ugly, that combines app_intercept and app_changrab |
| 21:28.05 | anthm | aww |
| 21:28.13 | anthm | well congrats =D |
| 21:28.41 | lathos42 | anthm: Thanks.. I'm kinda proud of myself.. its my first AGI, and my second Bash script |
| 21:28.45 | riksta | ManxPower: i know you probably don't care...but i worked out all the sangoma trouble myself, it's working fine ;) |
| 21:29.33 | ManxPower | endom0rph: the 2nd one you pasted indicates the callerid is correct inside of asterisk. |
| 21:29.57 | ManxPower | endom0rph: Now, does the callerid show up correctly on the phone you are CALLING? |
| 21:30.09 | ManxPower | riksta: Great! |
| 21:30.24 | Uberbot | I just had a iax2 to zap phone call just cut out and dissappear. Is this common? |
| 21:30.45 | riksta | it seems that the beta software is the one to use, the previous one doesn't like 2.6.12 anyway :) (if anyone ever asks) |
| 21:30.51 | ManxPower | riksta: As I've said before, lots of people say the Sangoma cards are good cards. However, Sangoma does not appear to contribute much to Asterisk's source tree, not many people are using it and so support on the channel will be tough. |
| 21:31.08 | riksta | ManxPower: well now i can help anyone else :) |
| 21:31.27 | h3x0r | shya |
| 21:31.34 | ManxPower | Uberbot: not that I know of |
| 21:31.44 | h3x0r | like as if digium has a bunch of people hanging out here 24/7 or something |
| 21:31.45 | h3x0r | heh |
| 21:31.58 | ManxPower | Uberbot: unless you are doing something stupid like set callprogress=yes or busydetect=yes for the zap port. |
| 21:32.34 | ManxPower | I don't have any of their cards, I just scammed a t-shirt at a trage show. |
| 21:32.50 | netsurfer | heh.. been there got the t-shirt |
| 21:32.55 | netsurfer | :P |
| 21:33.12 | Druken | sounds like me and my AMD shirts... |
| 21:33.21 | Druken | got like 3-4 of them... don't own a single AMD processor |
| 21:34.51 | *** join/#asterisk Hmmhesays (n=Hmmm@24-117-213-113.cpe.cableone.net) |
| 21:35.06 | Hmmhesays | everyone breathe a sigh of a relief i'm back |
| 21:35.28 | Druken | s'cuse me... |
| 21:35.59 | Hmmhesays | ha |
| 21:36.06 | *** part/#asterisk lathos42 (n=lathos42@65-42-27-66.dowdingindustries.com) |
| 21:36.10 | Hmmhesays | hopefully it wasn't beer and chili night last night |
| 21:37.29 | endom0rph | ManxPower - I'll do some more work on this rather than waste your time, you've made me look at some different things. I'll come back another time if I still need help. Thanks |
| 21:38.09 | Druken | endom0rph: why not just use asterisk's voicemail? it's a nice system... :) |
| 21:38.28 | Hmmhesays | haha this company just called cause they need a sip/t1 pbx setup by the 9th of september, I just kind of chuckled and said it would be ok |
| 21:39.19 | Druken | as long as the hardware is available... that's an overnight jobbie |
| 21:39.37 | endom0rph | Druken - because my job is working on the other voicemail but a normal pbx is big to carry around! I am running 3-4 voicemails and asterisk under VMWare, plus a complete voice access system |
| 21:40.11 | Druken | sounds nasty.... |
| 21:40.58 | *** join/#asterisk Atannia (n=Atannia@ti531210a080-7046.bb.online.no) |
| 21:41.03 | endom0rph | druken - you bet and learning asterisk isn't helping but it is a great system! I will get it working somehow |
| 21:41.07 | *** part/#asterisk johnASG (n=jon@border0hsv.asterisksgi.com) |
| 21:41.20 | Hmmhesays | yeap |
| 21:41.23 | *** join/#asterisk ryansc_ (n=ryansc@c-67-164-188-180.hsd1.co.comcast.net) |
| 21:41.27 | *** part/#asterisk Atannia (n=Atannia@ti531210a080-7046.bb.online.no) |
| 21:41.28 | Druken | endom0rph: you have access to the h323 system right? |
| 21:41.50 | endom0rph | yes it is running as a vmware virtual system |
| 21:43.00 | endom0rph | druken - I can dial into the vm and all works but I need to know who's voicemail the diverted call is from that is the sticking point |
| 21:43.28 | *** join/#asterisk papsow (n=pape@modemcable224.217-130-66.mc.videotron.ca) |
| 21:43.38 | Druken | endom0rph: so is your problem with callerid not being right on the h323 or you can't get the right voicemail to answer after it calls your phone ? |
| 21:44.28 | *** join/#asterisk kg (n=kg@host-81-190-173-57.lomza.mm.pl) |
| 21:45.00 | endom0rph | druken - callerid not getting through on the h323... voicemail shows the called extension as 555 (h323 vm number) when it should be 200 (the x-lite I called) |
| 21:45.44 | ManxPower | endom0rph: now is the time to search the mailing lists. |
| 21:45.49 | ManxPower | ~mailinglist |
| 21:45.50 | jbot | from memory, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 21:46.20 | ManxPower | endom0rph: you can also try a different H323 driver for Asterisk (there are 4 of them) |
| 21:47.40 | endom0rph | ManxPower - Thanks I have done a lot of googling but getting the right search term is proving problem. especially when everone says why don't you use asterisk vm lol |
| 21:47.52 | endom0rph | I'll check the other drivers though |
| 21:48.24 | *** join/#asterisk taz-^ (i=demon@195.38.75.178) |
| 21:48.50 | ManxPower | endom0rph: chan_h323 (included with Asterisk), asterisk-oh323 (Objective Systems?), chan_oh323 (in the asterisk-addons tree), and chan_wombat...er...chan_woomers (pbxfreeware.org ?) |
| 21:48.58 | ManxPower | ..er..chan_woomera |
| 21:49.32 | ManxPower | If *I* needed H323 I would try chan_ooh323 from asterisk-addons first. |
| 21:49.56 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-16-118.cybersurf.com) |
| 21:50.14 | endom0rph | ManxPower I have the asterisk-addons one downloaded from sourceforge |
| 21:50.41 | Hmmhesays | inaccess networks i think is oh323 |
| 21:52.25 | ManxPower | endom0rph: The correct place to get is the asterisk-addons tree from Digiums servers. |
| 21:52.31 | ManxPower | Hmmhesays: you are correct. |
| 21:52.48 | *** part/#asterisk kg (n=kg@host-81-190-173-57.lomza.mm.pl) |
| 21:53.17 | Hmmhesays | i think i'm going to vomit |
| 21:53.52 | Hmmhesays | clay aikin is singing some horrible song on oprah |
| 21:54.29 | eksffa | anyone can help me making an AGI call run twice in the same dial plan? |
| 21:54.42 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 21:54.44 | Hmmhesays | exten => _X,1,agi(my.agi) |
| 21:54.51 | Hmmhesays | exten => _X,2,agi(myagi) |
| 21:54.53 | ManxPower | exten => _X,2,agi(my.agi) |
| 21:54.58 | eksffa | Hmmhesays: does not work after a dial call |
| 21:55.04 | Hmmhesays | haha |
| 21:55.10 | eksffa | exten => _17XXXXXXXX,1,AGI(before.agi) |
| 21:55.11 | eksffa | exten => _17XXXXXXXX,2,dial(SIP/${EXTEN}@sip.freebsdbrasil.com.br,20) |
| 21:55.11 | eksffa | exten => _17XXXXXXXX,3,AGI(after.agi) |
| 21:55.15 | Hmmhesays | i typo'd, i'm eating pizza and laying in my bed |
| 21:55.16 | eksffa | wasnt it supposed to work? |
| 21:55.26 | ManxPower | eksffa: see the "g" option to Dial and the exten => h,1,Blah as (hopefully) documented in extensions.txt |
| 21:55.41 | eksffa | ManxPower: tried it later, see this |
| 21:55.56 | ManxPower | exten => _17XXXXXXXX,2,dial(SIP/${EXTEN}@sip.freebsdbrasil.com.br,20,g) |
| 21:56.02 | Hmmhesays | ok i'm going to go wash the motorcycle now |
| 21:56.09 | brettnem | cause I think.. you know.. a bunch of o's are a good thing.. |
| 21:56.14 | ManxPower | eksffa: Doing stuff after a dial happens is...complicated. |
| 21:56.42 | eksffa | ManxPower: I tried with the "g"... still the same behavior |
| 21:56.55 | eksffa | I really needed it to work after the dial |
| 21:57.05 | ManxPower | eksffa: different things happen dependind on if the caller or the callee hangs up first. |
| 21:57.19 | eksffa | ManxPower: like what? |
| 21:57.46 | ManxPower | If the destination (called person) hangs up then using "g" on Dial will make it go to the next priority. |
| 21:58.09 | ManxPower | If the Caller (person that dialed) hangs up, exten => h,1,Whatever will be called if it exists. |
| 21:59.10 | riksta | is there a voip-info.org mirror? all i get is "From wvfiber.ge2-2.br01.atl01.pccwbtn.net (63.216.31.138) icmp_seq=1 Time to live exceeded" when pinging |
| 21:59.11 | ManxPower | eksffa: sounds like you have the exact same hangup processing needs as CallingCard apps. |
| 21:59.28 | ManxPower | And you can read the docs from the billion or so calling card apps for Asterisk to see how they handle it in the dialplan. |
| 22:00.25 | *** join/#asterisk criptos (n=criptos@201.145.229.183) |
| 22:00.41 | criptos | anyone using app_intercept? |
| 22:00.46 | eksffa | hmm, where can I find this doc? |
| 22:00.52 | criptos | good afternoon |
| 22:01.39 | *** join/#asterisk _unnamed_ (n=pape@modemcable224.217-130-66.mc.videotron.ca) |
| 22:01.45 | ManxPower | eksffa: I don't know, I never need calling card applications. There's astcc in digium's CVS, asteki or something like that. Do a mailing list search for "calling card" |
| 22:01.48 | ManxPower | ~mailinglist |
| 22:01.50 | jbot | from memory, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 22:02.23 | eksffa | ManxPower: right, Ill search in the asterisk-users ML |
| 22:02.28 | eksffa | thank you |
| 22:02.41 | *** part/#asterisk websae (i=websae@207-118-143-134.dyn.centurytel.net) |
| 22:02.53 | ManxPower | eksffa: I'm a CorporateGeek, not a ServiceProviderGeek |
| 22:03.04 | eksffa | ;) |
| 22:03.09 | *** join/#asterisk NewSole (i=dave@d226-110-153.home.cgocable.net) |
| 22:03.26 | ManxPower | eksffa: so I don't need things like billing users for calls, calling card stuff, web interfaces, etc. |
| 22:03.43 | ManxPower | MY users have enough trouble dialing an analog phone. |
| 22:04.01 | ManxPower | "I don't use text messaging on my cell phone. It's too complicated." <-- actual user comment |
| 22:04.14 | criptos | thats a good definition Corporative Geek vs SPGeek |
| 22:04.39 | ManxPower | Actual conversation - Me: You do not need to dial "9" when using the fax machine. User: So, do I need to dial "9" when I send a fax? |
| 22:04.42 | fugitivo | ManxPower: are you kidding? in argentina all the people uses sms because it's cheaper than a regular mobile call |
| 22:05.09 | h3x0r | this country is ass backwards |
| 22:05.10 | ManxPower | fugitivo: in the usa you do NOT get free incoming calls or sms to cell phones. You pay for incoming calls/sms |
| 22:05.18 | h3x0r | ManxPower: depends on the carrier actually |
| 22:05.22 | h3x0r | some of them dont charge for incoming |
| 22:05.31 | h3x0r | legacy at&t wireless dosent |
| 22:05.31 | fugitivo | ManxPower: here you pay too, but it's cheaper |
| 22:05.34 | ManxPower | h3x0r: *nod* A few of the smaller ones don't, if you have the right plan. |
| 22:05.42 | h3x0r | at&t isnt "small" :P |
| 22:05.57 | wunderkin | only one i know of that doesnt charge for incoming would be cricket.. who else? |
| 22:05.57 | ManxPower | h3x0r: Um, AT&T doesn't have mobile service anymore. |
| 22:05.59 | h3x0r | and its like that on all plans |
| 22:06.01 | h3x0r | yes they do |
| 22:06.05 | h3x0r | they just dont market new phones |
| 22:06.06 | wunderkin | they are cingular now |
| 22:06.10 | ManxPower | they were bought by Congular |
| 22:06.12 | fugitivo | ManxPower: did you try MetroPCS? |
| 22:06.13 | puowvip | cingular |
| 22:06.18 | *** join/#asterisk Jaxxan (n=jaxxan@202.70.125.66) |
| 22:06.19 | h3x0r | its called the "blue" and "orange" networks. |
| 22:06.25 | ManxPower | fugitivo: I stick to major carriers. |
| 22:06.33 | ManxPower | Cingular, Verizon, Sprint, etc. |
| 22:06.35 | *** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1) |
| 22:06.43 | fugitivo | ManxPower: that's expensive |
| 22:06.51 | ManxPower | fugitivo: but more reliable. |
| 22:06.54 | h3x0r | I don't think sprint charges for incoming sms either. |
| 22:06.57 | h3x0r | probably do on MMS though |
| 22:06.59 | ManxPower | and customer service is not too bad. |
| 22:07.01 | fugitivo | ManxPower: i know |
| 22:07.03 | Jaxxan | hey ya'll |
| 22:07.11 | h3x0r | they never did when i had sprint PCS |
| 22:07.16 | wunderkin | oh, i meant incoming calls |
| 22:07.40 | ManxPower | Verizon charges 2 cents per incoming text message and 5 cents per outgoing text message, if you do not have a text messaging plan. |
| 22:07.53 | h3x0r | I'm pretty sure its a minority of carriers that charge for incoming sms |
| 22:07.57 | ManxPower | h3x0r: URL for free incoming service with AT&T? |
| 22:08.11 | h3x0r | http://localhost/mybill.pdf |
| 22:08.19 | ManxPower | h3x0r: Cingular charges 5 cents per incoming text message if you don't have a plan. |
| 22:08.25 | ManxPower | I don't know what sprint charges. |
| 22:08.32 | h3x0r | Yeah but its like $5 a month for a zillion sms's |
| 22:08.47 | ManxPower | h3x0r: what makes you think they still offer that plan to new customers. |
| 22:08.57 | *** join/#asterisk jayk- (i=jayk@vapid.reprehensible.net) |
| 22:09.10 | h3x0r | they still have decent sms plans |
| 22:09.13 | ManxPower | almost all my calls are incoming to my cell phone and almost all text messages are incoming |
| 22:09.14 | h3x0r | on the orange network |
| 22:09.25 | jayk- | i'm trying to get my voicemail message light indicator to work on my cisco 7960 phone. it works fine inside our office, but at home over my cable modem, the light won't come on. does anybody have any ideas? |
| 22:09.34 | h3x0r | verizon rapes everybody |
| 22:09.37 | ManxPower | h3x0r: blue is TDMA and orange is GS<? |
| 22:09.42 | h3x0r | no they are both GSM |
| 22:09.42 | fugitivo | Sprint charges _everything_ |
| 22:09.53 | h3x0r | and they also have the blue TDMA network still |
| 22:09.57 | ManxPower | jayk-: add qualify=yes to the sip.conf entry for your home phone |
| 22:10.07 | h3x0r | cingluar blue still has a shitload of TDMA traffic from wholesale customers that sell prepaid phones |
| 22:10.29 | ManxPower | h3x0r: less and less. They keep decommissioning TDMA towers. |
| 22:10.41 | h3x0r | http://www.wirelesswholesale.net/ |
| 22:10.50 | *** join/#asterisk l-fy (n=diana@yate/developer/l-fy) |
| 22:10.50 | *** join/#asterisk malcolmd (n=malcolmd@digium.com) |
| 22:10.51 | l-fy | hello |
| 22:10.56 | h3x0r | They are selling NEW phones that do TDMA |
| 22:10.57 | l-fy | is there anyone in .kr? |
| 22:10.59 | h3x0r | even at walmart |
| 22:11.04 | ManxPower | one of my clients finally switched to Verizon after Cingular decommissioned so many TDMA towers he was getting signal about %50 the time. |
| 22:11.12 | l-fy | i really need a firmware for a vizufon cip 4500 and one 5500 |
| 22:11.13 | jayk- | what does qualify do? |
| 22:11.15 | criptos | humm... any one working with app_intercept nicely? |
| 22:11.28 | h3x0r | you seen Trac Fone? |
| 22:11.28 | *** join/#asterisk riksta (n=rick@84.93.141.188.plusnet.pte-ag1.dyn.plus.net) |
| 22:11.32 | h3x0r | That uses AT&T TDMA |
| 22:11.51 | h3x0r | the GSM wholesale product sucks ass. |
| 22:11.52 | ManxPower | TracPhone? Is that one of those "rape me in the ass" prepaid things? |
| 22:12.07 | h3x0r | actually tracfone is better than virgin mobile or anything else |
| 22:12.11 | criptos | jayk helps to deter the latency of a connection to a phone/peer and in some channels, that info is used to configure jitterbuffers properly. |
| 22:12.13 | h3x0r | you can buy a card that has minutes that take a year to expire |
| 22:12.21 | ManxPower | h3x0r: cost per min? |
| 22:12.28 | h3x0r | 10c |
| 22:12.33 | h3x0r | which if you work out what a monthly plan costs |
| 22:12.36 | h3x0r | is about the same |
| 22:12.38 | criptos | Mostly, the recomendation should be no to use qualify |
| 22:12.39 | ManxPower | not terrible, I guess. |
| 22:12.52 | ManxPower | can you refill without speaking to a human? |
| 22:12.55 | h3x0r | yes |
| 22:13.07 | h3x0r | you can link it to a credit card if you want for auto recharge |
| 22:13.14 | ManxPower | h3x0r: that's handy. |
| 22:13.25 | ManxPower | Maybe I'll consider it when my CONTRACT runs out. |
| 22:13.26 | h3x0r | this is the other thing, the system on the orange side of cingular isn't worthy of setting up a prepaid phone thing |
| 22:13.48 | ManxPower | h3x0r: I don't care about colors, I care about GSM and TDMA 8-) |
| 22:13.49 | h3x0r | it lacks the features to show this stuff to the user in real time |
| 22:13.58 | h3x0r | actually its delayed 24 hrs or osmething but still |
| 22:14.21 | r0m | good evening |
| 22:14.32 | h3x0r | i guess they expect you to set up a WAP interface but tahts stupid coz usually youd ont want data enabled on a prepaid phone |
| 22:14.33 | *** part/#asterisk l-fy (n=diana@yate/developer/l-fy) |
| 22:14.34 | jayk- | ManxPower: didn't seem to work. i set qualify=yes, reloaded and rebooted my phone. it's on a public IP address, no nat traversal |
| 22:14.40 | h3x0r | maybe they have fixed this by now |
| 22:14.54 | h3x0r | there are more features in the blue HLR than the orange one |
| 22:15.06 | h3x0r | both gsm and tdma |
| 22:15.18 | ManxPower | jayk-: Ah. |
| 22:15.27 | h3x0r | its going to be many years before they actually phase out the blue networks |
| 22:15.46 | ManxPower | h3x0r: I still don't see unlimited incoming anywhere. |
| 22:15.49 | h3x0r | in fact i bet sprint will sooner phase out iDEN than blue |
| 22:16.04 | Uberbot | ManxPower. Thanx for the pointer on my dropped call. I'll investigate. It could also have been a network issue...... |
| 22:16.06 | h3x0r | It isn't advertised that way, its just how their billing system works |
| 22:16.11 | ManxPower | Sprint usees CDMA. |
| 22:16.15 | ManxPower | Nextel uses iDEN |
| 22:16.20 | jayk- | ManxPower: any other ideas on how to get the message light indicator to work? |
| 22:16.22 | h3x0r | Sprint bought Nextel |
| 22:16.27 | criptos | My grandmother uses Wheels... |
| 22:16.29 | h3x0r | Sprint is decommissioning iDEN |
| 22:16.42 | ManxPower | h3x0r: I had forgotten about that. It was approved? |
| 22:16.49 | ManxPower | jayk-: sip debug |
| 22:16.49 | h3x0r | yes |
| 22:16.54 | h3x0r | www.nextel.com |
| 22:17.08 | ManxPower | make sure it's mailbox=vmbox@vmcontext not @extensionscontext |
| 22:17.11 | h3x0r | News |
| 22:17.24 | h3x0r | haha they have a new logo |
| 22:17.25 | ManxPower | h3x0r: good. I never liked nextel. |
| 22:17.34 | h3x0r | me either. |
| 22:17.37 | ManxPower | Well, I never liked iDEN. |
| 22:17.42 | h3x0r | free mobile to mobile killed push to talk |
| 22:18.06 | h3x0r | but sprint's CDMA sucks a lot more. |
| 22:18.15 | h3x0r | if i was them, id do the opposite |
| 22:19.08 | Uberbot | ManxPower, I was doing callprogress=yes on the zap channel. Can you tell me why this was "stupid" beyond the fact that it was experimental? Thanx. |
| 22:19.17 | h3x0r | you know whats funny though |
| 22:19.24 | h3x0r | at&t is going to reemerge and resell sprint pcs |
| 22:19.36 | h3x0r | once their non compete runs out |
| 22:20.02 | h3x0r | Did you know that almost 50% of the minutes of traffic on Sprint PCS's network is wholesale ? |
| 22:20.09 | criptos | Uberbot: callprogress, at source code states that only works for usa, and isn`t reliable... |
| 22:20.22 | Uberbot | Good enough for me. Thanx. |
| 22:20.28 | h3x0r | such as virgin mobile, etc. |
| 22:20.35 | ManxPower | h3x0r: Yeah. I read an article about it. |
| 22:20.36 | h3x0r | prepaid phone vendors |
| 22:20.36 | Uberbot | "Use The Source...." |
| 22:20.52 | ManxPower | My next cell phone will be a quad-band GSM phone. |
| 22:20.53 | criptos | Uberbot: are u having issues with random hangups? |
| 22:20.59 | h3x0r | hahaha |
| 22:21.03 | *** part/#asterisk criptos (n=criptos@201.145.229.183) |
| 22:21.11 | ManxPower | But that won't be for at least a year. |
| 22:21.16 | Uberbot | I'm only doing initial testing. Most of my calls had been short. |
| 22:21.21 | h3x0r | tdma is so fucking crisp and clear now if you are in an area with lots of towers |
| 22:21.29 | h3x0r | coz they have been moving everybody to gsm |
| 22:21.34 | h3x0r | theres like nobody on the tdma towers |
| 22:21.35 | Uberbot | Then I called a co-worker and it dropped the call after about 20 minutes. |
| 22:21.45 | *** join/#asterisk Rav1974 (n=r@ool-457a17a9.dyn.optonline.net) |
| 22:21.51 | Uberbot | He was still quite impressed, though. |
| 22:21.57 | h3x0r | hell i wanna switch back to AMPS |
| 22:21.59 | ManxPower | Uberbot: callprogress randomly disconnects calls. |
| 22:22.10 | h3x0r | id have a whole tower to myself. |
| 22:22.23 | ManxPower | h3x0r: my phone lets me force analog. |
| 22:22.27 | *** join/#asterisk tessier (n=treed@wsip-68-224-172-77.sd.sd.cox.net) |
| 22:22.35 | h3x0r | but paying for roaming sucks |
| 22:22.43 | *** join/#asterisk criptos (n=criptos@201.145.229.183) |
| 22:23.12 | Nugget | I miss having analog cellular to fall back on, but being able to take my phone overseas offsets the loss. |
| 22:23.13 | criptos | <PROTECTED> |
| 22:23.16 | Uberbot | So it seems. Thanx. |
| 22:23.37 | h3x0r | nugget thats what triband GSM is for |
| 22:23.49 | Nugget | triband GSM doesn't let you fall back to analog cellular. |
| 22:23.56 | h3x0r | but everybody else has real GSM |
| 22:23.57 | Nugget | I *have* triband gsm, silly. |
| 22:24.21 | Nugget | not japan. I still can't take my phone to japan. |
| 22:24.25 | Nugget | they're nutty. |
| 22:24.27 | ManxPower | hmm...not bad call quality |
| 22:24.39 | Nugget | but all I was saying is that I miss having analog cellular. sometimes you just need it. |
| 22:24.49 | h3x0r | id buy a japanese phone |
| 22:24.52 | h3x0r | they have better shit than us |
| 22:25.18 | Nugget | eh |
| 22:25.24 | ManxPower | I was pretty disapointed with the cost of pre-pay cell service in europe |
| 22:25.32 | Rav1974 | hello guys |
| 22:25.36 | h3x0r | dude its europe |
| 22:25.38 | ManxPower | I must have spent $200 on it. |
| 22:25.49 | h3x0r | the ghetto costs as much as living in LA |
| 22:25.50 | *** join/#asterisk DoctorWho (n=lot@68.148.192.184) |
| 22:25.51 | Rav1974 | hoping to get some support with TE110P |
| 22:25.52 | h3x0r | or new york |
| 22:25.59 | ManxPower | Fortunatly I used up most of my mins before leaving each country. |
| 22:26.17 | Rav1974 | i just plugged in the TE110P, and I don't know how to configure it with the ADIT 600 |
| 22:26.28 | Rav1974 | is there some docs you could point me to? |
| 22:26.51 | ManxPower | Rav1974: your extensive search of the mailing list archive and wiki (which is down at the moment) didn't help? |
| 22:27.06 | ManxPower | ooo.... /umode +E |
| 22:27.22 | ManxPower | ~docs |
| 22:27.24 | jbot | hmm... docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 22:27.24 | ManxPower | ~mailinglist |
| 22:27.26 | jbot | i guess mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 22:27.35 | Rav1974 | thanks |
| 22:27.46 | ManxPower | h3x0r: not when they automagically generate nicks to get around bans |
| 22:28.02 | h3x0r | heh |
| 22:28.32 | Rav1974 | ManxPower: no it is a little confusing for me since I didn't do any configuration. I used asterisk@home |
| 22:28.45 | ManxPower | I can't help you with Asterisk@Home |
| 22:29.06 | *** part/#asterisk apeman (n=cmurray@zoidberg.digital-storm.net) |
| 22:29.14 | h3x0r | meanwhile |
| 22:29.16 | h3x0r | lilo is spamming |
| 22:29.17 | h3x0r | hahah |
| 22:29.21 | criptos | I not using asterisk@home can u help me.. ha ha ha :) |
| 22:29.23 | [hC] | Do i have to start privmsg lilo and spam him back again |
| 22:29.24 | [hC] | jesus |
| 22:29.38 | drray | I've blocked more people in the last two days |
| 22:29.50 | ManxPower | criptos: |
| 22:29.51 | ManxPower | span=1,1,0,esf,b8zs |
| 22:29.53 | CoaxD | hC: Guess so |
| 22:29.55 | h3x0r | hc, send him some porn |
| 22:30.05 | ManxPower | fxoks=1-24 |
| 22:30.06 | [hC] | i wonder if the lilo pr0n we made is still up |
| 22:30.11 | CoaxD | oh, yeah |
| 22:30.12 | CoaxD | it is |
| 22:30.18 | CoaxD | http://www.lilofree.net |
| 22:30.29 | ManxPower | that's /etc/zapatel.conf |
| 22:30.32 | [hC] | Haha |
| 22:30.32 | [hC] | http://tinyurl.com/9sy4x |
| 22:30.35 | [hC] | >:) |
| 22:30.41 | criptos | dammnn.. app_intercep seems to work fine, but when a call is intercepted iax2 protocol, the call get lost... |
| 22:30.57 | ManxPower | /etc/asterisk/zapata.conf would be: |
| 22:31.01 | ManxPower | signalling=fxo_ks |
| 22:31.07 | [hC] | bahaha |
| 22:31.10 | [hC] | i just msged him |
| 22:31.11 | ManxPower | channel => 1-24 |
| 22:31.18 | criptos | evetn app_intercep says: -- Releasing IAX2/compras/17 and IAX2/asis2@asis2/23 |
| 22:31.23 | [hC] | [18:30] >lilo< Hi, i pasted some spam reports up on http://tinyurl.com/9sy4x, maybe it will help you figure the problem out |
| 22:31.30 | [hC] | <lilo> thanks |
| 22:31.34 | ManxPower | criptos: try notransfer=yes in iax.conf |
| 22:31.39 | [hC] | im going to be k-lined in a second. |
| 22:31.43 | criptos | what seems weird to me, is IAX2/asis2@asis2/23 |
| 22:35.56 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 22:35.56 | *** topic/#asterisk is Preview the new website! http://beta.asterisk.org || Asterisk: The Open Source PBX || Asterisk 1.2 Janitors Needed! http://dev.asteriskdocs.org || Astricon 2005 in Anaheim Oct 12-14 http://www.astricon.net/2005/ - speakers wanted |
| 22:36.10 | Delta34 | am i suppose to store the ani as a callerid |
| 22:36.22 | eisman | I just got ANI/DNIS working over T1 to my asterisk box. |
| 22:36.39 | eisman | asterisk wants it in *ANI*DNIS format |
| 22:37.01 | eisman | are you getting 20 digits as DTMF? |
| 22:37.06 | *** join/#asterisk websae (n=websae@207-118-143-134.dyn.centurytel.net) |
| 22:37.17 | websae | anyone here running ASTERISK@HOME? |
| 22:37.24 | ManxPower | eisman: You don't have a PRI? |
| 22:37.34 | Delta34 | so i need to tell carrier to change from *DNIS*ANI to *ANI*DNIS |
| 22:37.35 | eisman | don't have the card available |
| 22:37.47 | *** join/#asterisk pfn_ (n=82416d67@pfnguyen.best.vwh.net) |
| 22:37.51 | ManxPower | Delta34: I think he's on a PRI. |
| 22:37.52 | criptos | manxpower, notransfer=yes at iax.conf, stoped an reloades, didn`t work |
| 22:38.06 | eisman | I am running D4/AMI E&M wink to a TE110P |
| 22:38.09 | ManxPower | Delta34: use pri debug span x to see if you are ACTUALLY getting the ANI and stuff. |
| 22:38.27 | eisman | Delta, are you getting the DTMF right now? |
| 22:38.36 | Delta34 | yeh |
| 22:38.38 | Delta34 | number works |
| 22:38.44 | ManxPower | eisman: Um, this isn't 1975 any more. Why not use ESF/B8ZS? |
| 22:38.45 | Delta34 | i get call to come in correctly |
| 22:38.51 | Delta34 | just caller id all screwed =) |
| 22:38.59 | ManxPower | you don't get DTMF on PRIs |
| 22:39.04 | ManxPower | not for callerid stuff. |
| 22:39.04 | eisman | ManxPower, you tell that to the Mitel system I have sitting here. |
| 22:39.48 | eisman | Delta34, is this a T1 or PRI? |
| 22:39.52 | Delta34 | t1 |
| 22:40.07 | ManxPower | Delta34: Why did I think you had a PRI? |
| 22:40.24 | eisman | ManxPower, yea we are $250k+ into this phone system and are going to have to fork lift it out of here in teh next year. |
| 22:40.24 | Delta34 | i am using same as u eisman d4/ami e&m wink to dual port digium card |
| 22:40.39 | [hC] | Is there anyone here that really understands what happens in a native bridge handoff procesS? |
| 22:40.45 | ManxPower | Delta34: then he is the world expert right now. |
| 22:41.03 | ManxPower | [hC]: Only mark I think, and only after he's had a few drinks. |
| 22:41.05 | [hC] | im having nothing but problems trying to figure out why calls act funny after a native bridge. |
| 22:41.11 | [hC] | I think i read an article once about it |
| 22:41.14 | [hC] | on -dev |
| 22:41.17 | *** join/#asterisk zotz (n=zotz@24.231.36.100) |
| 22:41.21 | [hC] | i wish i had a better rss feed for -users and -dev |
| 22:41.22 | *** join/#asterisk riksta (n=rick@84.93.141.188.plusnet.pte-ag1.dyn.plus.net) |
| 22:41.24 | [hC] | all i get is subjects |
| 22:41.28 | Delta34 | so how should i do this? *ANI*DNIS |
| 22:41.49 | eisman | Delta, if you can get your provider to change the DNIS/ANI string, do so. If not I'm trying to find the code where * looks at it, a code change may fix it quick. |
| 22:41.50 | Delta34 | set ani as a callerid variable? |
| 22:42.18 | eisman | if you get it *ANI*DNIS, asterisk will do it for you. |
| 22:42.48 | eisman | and $CALLERID, $CALLERIDNUM will be ANI |
| 22:43.49 | criptos | maxp no, didn`t work... I have this now: -- Attempting native bridge of IAX2/asis2@asis2/14 and IAX2/compras/16 |
| 22:43.49 | criptos | <PROTECTED> |
| 22:43.53 | criptos | :( |
| 22:44.11 | Delta34 | so does your match look like this _NXXNXXXXXX,DNIS |
| 22:45.45 | *** join/#asterisk insomni_ (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 22:45.52 | websae | what is a trunk? |
| 22:46.12 | criptos | websae: a group of lines. |
| 22:46.35 | websae | teliax...would that be a trunk if i have service witht hem? |
| 22:46.45 | drray | if you had a connection to them |
| 22:47.00 | criptos | In telephony speaking, you now, how each "world" has it`s owns terminology: Trunk = Group. |
| 22:47.29 | *** join/#asterisk pablasso (n=pablasso@201.138.255.59) |
| 22:47.43 | websae | hmm okay |
| 22:47.44 | websae | thank you |
| 22:47.53 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 22:48.08 | jeffgus | is anyone else around here getting /msg's from random people on freenode? |
| 22:48.32 | cochi | jeff, known problem |
| 22:48.38 | cochi | there was some notice about /mode +E or so |
| 22:49.02 | cochi | [00:26] -lilo- [Global Notice] Hi all. Apparently we're having problems with message spam. One good interim solution might be: "/umode +E" .... no messages other than from registered users. Please consider registering and autoidentifying if you haven't already done so. |
| 22:49.18 | [hC] | When is the next big asterisk con? |
| 22:49.32 | [hC] | I really want to be in a like minded area for a while and be able to trade tricks with people :P |
| 22:49.40 | jeffgus | cochi, thank you |
| 22:50.00 | jeffgus | cochi, i wanted to ask that question in a more appropriate channel, but #freenode didn't give me a voice |
| 22:50.28 | criptos | Astricon Anaheim |
| 22:50.29 | criptos | Anaheim, California, USA |
| 22:50.29 | criptos | 12 October?14 October 2005 |
| 22:51.00 | [hC] | Cool |
| 22:51.04 | [hC] | Maybe i'll try to go to that |
| 22:52.13 | ManxPower | Wow! I may have just found a piece of telecom equipment I've been looking for off and on for 6 months. |
| 22:52.18 | ManxPower | on eBay. |
| 22:52.22 | pygrammer | hey, can one transfer a call via Manager? |
| 22:52.26 | eisman | Delta, it looks like it would be a non-trivial code change to make asterisk understand to blocks of 10 digits instead of the delimited string *ANI*DNIS |
| 22:52.26 | [hC] | ManxPower: whats that? |
| 22:52.42 | ManxPower | [hC]: I'll tell you on monday when I've won the auction. |
| 22:53.01 | pablasso | hi guys |
| 22:53.13 | drray | ManxPower just bought a beige box |
| 22:53.16 | [hC] | ManxPower: lol. :) I'm in Florida. We were just hit by a hurricane, all the power is out for at least 2more days, I cant get to a bank anyways. :) |
| 22:53.37 | pablasso | you think is possible sending a fax via one iax server to another iax server with ulaw as codec? |
| 22:53.44 | cochi | .oO(That's american. no power, hazards but chatting here) |
| 22:53.44 | ManxPower | drray: I prefer purple. |
| 22:53.49 | drray | manx :) |
| 22:53.51 | [hC] | This might be a stupid question |
| 22:54.04 | [hC] | But If i do g729 from a phone to *, then * does ULAW from it to a voip peer, |
| 22:54.13 | [hC] | will it sound any better than if i did g729 instead of ulaw in the second part of the call? |
| 22:54.25 | pygrammer | can i transfer a CALL via the Manager interface??????? |
| 22:54.30 | Hmmhesays | yes |
| 22:54.32 | pygrammer | i.e. with a web application |
| 22:54.38 | Hmmhesays | redirect |
| 22:54.40 | pygrammer | Blind and/or supervised |
| 22:54.45 | eisman | hC: I doubt it, but you won't need to transcode. |
| 22:54.50 | drray | Flash oeprator panel |
| 22:54.59 | drray | can do transfers |
| 22:55.04 | [hC] | eisman: well right now i do g729 -> * g729 -> |
| 22:55.08 | Hmmhesays | it's like 20 lines of code in perl or php |
| 22:55.09 | pygrammer | ya |
| 22:55.09 | cochi | wasn't g729 compressed and ulaw not? |
| 22:55.10 | drray | although I hate flash operator panel |
| 22:55.11 | criptos | When changing from one codec to other codec, there is a reformating, and yes, u waste time, becose you need to decode/re encode. So using the same format, you save time and cpu.. |
| 22:55.15 | [hC] | i was just curious if i would gain anything by switching the second part over ulaw |
| 22:55.17 | pygrammer | Hmmhesays, not using either one :( |
| 22:55.27 | Hmmhesays | probably pretty easy in python too |
| 22:55.29 | Hmmhesays | ;) |
| 22:55.31 | [hC] | im having a lot of call quality issues when native bridging occurs, and i might just disable it |
| 22:55.31 | eisman | doubtful |
| 22:55.36 | pygrammer | using Python |
| 22:55.37 | pablasso | yeah g729 is compressed and ulaw is not |
| 22:55.38 | pygrammer | yep |
| 22:55.38 | pygrammer | :D |
| 22:55.55 | cochi | so how could quality improve if you exchange ulaw by g729? |
| 22:55.59 | pygrammer | now, i'm wondering what the best python interface to manager and/or AGI (the dialplan) is |
| 22:56.00 | pygrammer | pyast? |
| 22:56.01 | eisman | it may be compressed, but is ulaw lossy? |
| 22:56.03 | pygrammer | pyst* |
| 22:56.05 | Hmmhesays | examples of connecting to the manager with python on the wiki i believe.... could be mistaken |
| 22:56.09 | cochi | compression is either lossless or lossy. so it can at best stay same eh? ;) |
| 22:56.19 | [hC] | cochi: cause the second part of the call is not subject to codec manipulation based on lag,etc. |
| 22:56.24 | eisman | err g729 rather |
| 22:56.33 | Hmmhesays | you will not get decent lossless compress in realtime on anything |
| 22:56.34 | pygrammer | Hmmhesays, didn't see any exampes |
| 22:56.35 | cochi | mh ah now i see your point. ok. still nooby :) |
| 22:56.36 | [hC] | cochi: it wouldnt get any better sounding, but it could potentially suffer from less jitter. |
| 22:56.37 | pygrammer | *examples |
| 22:56.39 | pygrammer | which page? |
| 22:56.45 | Hmmhesays | python use sockets? |
| 22:56.48 | Jaxxan | instead of playing hold music when a call is in a queue, is there a way to just do "ringing" ? |
| 22:57.04 | pygrammer | Hmmhesays, yes |
| 22:57.12 | eisman | put an mp3 of ringing in the moh folder? |
| 22:57.12 | pygrammer | Jaxxan, yes |
| 22:57.17 | Hmmhesays | check the php or perl example they, they are straight forward |
| 22:57.31 | Hmmhesays | s/they/then |
| 22:57.33 | pygrammer | but it's a different interface with python |
| 22:57.39 | criptos | pythin uses sockets, also php, even shellscript can use sockets |
| 22:57.50 | Hmmhesays | can't be that i different |
| 22:57.55 | criptos | and the use of sockets at python is pretty easy.... |
| 22:58.15 | Jaxxan | so in the queues.conf what would i use for musiconhold = ? |
| 22:58.19 | Hmmhesays | basically you are just flushing text throuh a socket to the manager |
| 22:58.26 | Jaxxan | or do i just comment that line out ? |
| 22:58.30 | pygrammer | yes, i've used sockets in python, but i was asking what's the best module for creating python-asterisk apps... i.e. pyst or something? |
| 22:58.33 | pygrammer | or pyastre |
| 23:01.06 | pygrammer | anyone? |
| 23:01.25 | Jaxxan | is there a ringing sound file included ? |
| 23:01.38 | criptos | you can generate it with tones... |
| 23:01.53 | criptos | there is a app to generate tones :) playtones(ring) will do the trick . |
| 23:02.08 | Jaxxan | can i do that in a queue though / |
| 23:02.16 | Jaxxan | oh |
| 23:02.18 | Jaxxan | i see |
| 23:02.28 | *** join/#asterisk fatalis_ (i=unyanya@customer-192.143.livas.lv) |
| 23:02.39 | Jaxxan | not sure how to do that though )= |
| 23:03.12 | *** part/#asterisk fatalis_ (i=unyanya@customer-192.143.livas.lv) |
| 23:04.32 | pygrammer | ok, so no one here does python/asterisk work? |
| 23:04.35 | pygrammer | i know there are modules to interface with asterisk manager/agi |
| 23:04.38 | pygrammer | just not which is bets |
| 23:04.41 | pygrammer | *best |
| 23:05.17 | ManxPower | [hC]: http://tinyurl.com/7nc4s |
| 23:05.23 | criptos | I decide based on documentantion and code understanding... |
| 23:05.39 | pygrammer | do you use python with asterisk? |
| 23:05.48 | ManxPower | [hC]: It doesn't support PRI, but we don't really NEED PRI support. |
| 23:07.05 | ManxPower | Jaxxan: um....use the "r" option to Queue |
| 23:08.33 | Jaxxan | Queue(operator|r) |
| 23:08.34 | Jaxxan | ? |
| 23:08.39 | Jaxxan | like that ? |
| 23:10.07 | *** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com) |
| 23:10.19 | Jaxxan | sweet |
| 23:10.26 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 23:10.26 | *** mode/#asterisk [+o denon] by ChanServ |
| 23:10.53 | [hC] | ManxPower: ah, cool :) I dont really have a use for something like that yet. |
| 23:10.54 | [hC] | :) |
| 23:11.07 | criptos | any one using app_intercept? |
| 23:11.38 | ManxPower | [hC]: I strongly suspect many of our Asterisk issues are problems with messed up gains. |
| 23:11.55 | [hC] | ManxPower: really.. hmm |
| 23:11.57 | ManxPower | but since we can ONLY measure the gains on Zap interfaces with Asterisk.... |
| 23:12.07 | ManxPower | [hC]: fax problems, mostly |
| 23:12.07 | [hC] | ManxPower: do share your findings :) |
| 23:12.12 | ManxPower | but some modem issues |
| 23:12.49 | ManxPower | [hC]: we'll be installing a hardware echocanceler (tellabs) on tuesday on our largest asterisk install. |
| 23:13.01 | ManxPower | then we can up the txgain on the pri and maybe fix the fax problems. |
| 23:13.10 | Jaxxan | hrm |
| 23:13.30 | [hC] | ManxPower: what termination do you use mostly? and by largest install, do you mean most PRI channels, or sip/iax trunks, or what? (and how many?) |
| 23:13.48 | ManxPower | [hC]: 20 channel PRI. |
| 23:13.55 | ManxPower | 60 polycom phones |
| 23:13.59 | Jaxxan | if i use Queue(operator) the call isn't answered (which is what i want), but when i do Queue(operator|r) the call is answered but it plays ring tones they way i want. )= |
| 23:15.17 | ManxPower | [hC]: we'll be doing a full PRI into Asterisk and a channel bank into the corporate PBX soon as well. |
| 23:15.30 | ManxPower | currently it's CT1 to carrier's channel bank into the corporate PBX |
| 23:15.44 | paulc | Jaxxan: what's the strategy for the operator queue though? |
| 23:15.56 | Jaxxan | hey there you are |
| 23:16.06 | Jaxxan | rrmemory |
| 23:16.11 | [hC] | ManxPower: cool. |
| 23:16.22 | ManxPower | [hC]: It sucks. |
| 23:16.42 | [hC] | ManxPower: haha :) |
| 23:17.02 | websae | I am setting up X-LIte....and i entered my server info for domain/realm and sip proxy and phone just isn't connecting |
| 23:17.02 | [hC] | ManxPower: right now i have two 23 channel PRI's lit up, and about 65 cisco 7960's deployed |
| 23:17.07 | websae | how do i get it to connect |
| 23:17.17 | ManxPower | I have like fxo_ks, fxo_ls, e&m/wink and DACS channels going thru 1 asterisk box on 2 CT1s |
| 23:17.22 | [hC] | ManxPower: and about 4 * boxes, spread out in various places |
| 23:17.27 | [hC] | Full PoE, etc. |
| 23:17.50 | ManxPower | [hC]: you a corporate geek or a service provider geek? |
| 23:18.08 | websae | anyone here familiar with Xten and the X-Lite phone? |
| 23:18.22 | criptos | Me, a little bit |
| 23:19.13 | pygrammer | I am |
| 23:19.17 | ManxPower | [hC]: I manage 9 Asterisk systems, but most of them are small. |
| 23:19.25 | pygrammer | I use it on both winblows and linux |
| 23:20.03 | *** join/#asterisk Damin_ (n=damin@nucleus.nacs.net) |
| 23:20.31 | Damin | Howdy.. |
| 23:21.08 | websae | anyone here use Asterisk@home? |
| 23:22.05 | christian[asgi] | heh |
| 23:24.33 | niZon | hmm |
| 23:24.39 | niZon | i can't compile zaptel :( |
| 23:24.42 | niZon | many many errors |
| 23:24.45 | christian[asgi] | why not? |
| 23:25.11 | niZon | It doesn't seem to like the kernel header files |
| 23:25.25 | niZon | many many errors |
| 23:25.28 | christian[asgi] | ic |
| 23:25.28 | criptos | humm.. you have the HOLE kernel installed? |
| 23:25.32 | christian[asgi] | doesn't sound fun |
| 23:25.32 | niZon | /usr/include/linux/device.h:183: error: syntax error before 'ssize_t' |
| 23:25.35 | criptos | kernel----src bla bla.. |
| 23:25.41 | niZon | yeah i have the source installed |
| 23:25.44 | niZon | 2.4.27 |
| 23:26.04 | criptos | where do u haveit installed? is it at /usr/src/linux-2.4.27? |
| 23:28.09 | niZon | yeah |
| 23:28.17 | niZon | hmm |
| 23:31.07 | christian[asgi] | recent kernel upgrade? |
| 23:31.12 | niZon | nope |
| 23:31.20 | niZon | debian netinstall default |
| 23:31.29 | christian[asgi] | oh.. debian |
| 23:31.46 | christian[asgi] | Wish I knew something about Debian to help you out. |
| 23:31.55 | niZon | i think i might just do the lazy thing and use zorcom rapit |
| 23:31.57 | niZon | bleh |
| 23:31.58 | niZon | *rapid |
| 23:32.12 | christian[asgi] | oh rapid |
| 23:32.14 | niZon | i just hope it included ztdummy/zaprtc |
| 23:32.23 | niZon | er includes |
| 23:32.23 | christian[asgi] | you can't compile anything on rapid |
| 23:32.26 | christian[asgi] | no compiler |
| 23:32.30 | christian[asgi] | http://www.voip-info.org/tiki-index.php?page=Running+Asterisk+on+Debian |
| 23:32.35 | niZon | yeah i know |
| 23:32.44 | christian[asgi] | did you read that link already? |
| 23:34.32 | Wi_Fi | hey |
| 23:34.39 | *** join/#asterisk Lathos42 (n=Lathos42@68.77.108.51) |
| 23:34.54 | Wi_Fi | does aah have an irc channel? |
| 23:35.09 | niZon | looking at it now |
| 23:35.34 | christian[asgi] | it should Wi_Fi |
| 23:35.40 | Romik | what this means? Aug 26 23:35:31 NOTICE[28763]: chan_iax2.c:5773 socket_read: Rejected connect attempt from 208.8.11.119, requested/capability 0x400/0x600 incompatible with our capability 0xf900. |
| 23:35.55 | christian[asgi] | codec mismatch |
| 23:36.00 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-241-108.buckeyecom.net) |
| 23:36.16 | Romik | christian: how i can know which requested and which need? |
| 23:36.35 | christian[asgi] | Romik, both your boxes? |
| 23:36.43 | Romik | christian: yes |
| 23:36.49 | christian[asgi] | in iax.conf |
| 23:37.10 | christian[asgi] | make sure the peers both have allow=<codec> for the same codecs |
| 23:37.58 | Romik | disallow=all and allow=g729 |
| 23:38.12 | ManxPower | Romik: go buy some G729 licenses. |
| 23:38.16 | christian[asgi] | :) |
| 23:38.28 | Romik | manxpower: why? ;) |
| 23:38.48 | Romik | also anybody can tell me why i see negative jitter? http://pastebin.ca/21273 |
| 23:38.51 | christian[asgi] | Romik: try with gsm first |
| 23:38.58 | ManxPower | Romik: because if you want to use G729 you want G729 licenses |
| 23:39.38 | Romik | manxpower: are you from Digium?:) |
| 23:40.05 | christian[asgi] | MaxPower: I am in a lot of the rooms you're in.. But I was thinking of buying that shirt for my brother. |
| 23:40.25 | ManxPower | Romik: no. |
| 23:41.15 | Romik | manxpower: do you know why i have negative jitter? and how it can be? |
| 23:42.20 | *** join/#asterisk Ash (i=aaron@outofband.org) |
| 23:42.27 | Ash | holy lots of people batman |
| 23:42.46 | ManxPower | Romik: the version of Asterisk I run doesn't have a jitterbuffer. |
| 23:42.58 | ManxPower | christian[asgi]: See #*tricks channel |
| 23:43.01 | websae | anyone know anything about XTEN XLITE phone? |
| 23:43.07 | websae | looking for some help please :) |
| 23:43.23 | Ash | has anybody ever seen ztmontior report audio on a channel when that line is not connected anywhere? |
| 23:43.44 | christian[asgi] | Ash: tdm400? |
| 23:43.45 | Romik | manxpower: i do not undestand how jitter can be negative |
| 23:44.03 | Ash | christian[asgi]: a TE110P running PRI |
| 23:44.46 | christian[asgi] | Ash: I don't know why on a t/e |
| 23:45.03 | *** join/#asterisk r0d3nt (i=r0d3nt@tinfoilhat.net) |
| 23:45.04 | ManxPower | http://www.tshirthell.com/hell.shtml They have some REALLY HORRIBLE designs, but they have a few good ones. |
| 23:45.04 | Ash | christian[asgi]: do you know why on a tdm400? |
| 23:45.06 | Ash | :) |
| 23:45.07 | ManxPower | eek! |
| 23:45.30 | christian[asgi] | Ash: Noisy modules.. noisy card.. bad either... |
| 23:45.36 | Ash | ah |
| 23:46.05 | websae | anyone here used the XLITE softphone still looking for some help connecting |
| 23:46.06 | websae | please |
| 23:46.14 | Ash | ok, another question then; has anybody dealt with echo cancellation on polycom 301s or 501s? |
| 23:46.28 | Ash | :D |
| 23:46.39 | niZon | yum |
| 23:46.44 | christian[asgi] | Ash: Do you get it from phone to phone, or only through tdm/asterisk? |
| 23:46.47 | niZon | 250MB in like 5 minutes |
| 23:46.59 | Ash | christian[asgi]: only when they're heading through the PRI |
| 23:47.27 | Ash | ztmonitor shows their levels as ridiculously high, so we tuned the gain on the phones down |
| 23:47.32 | Ash | but that causes other fun problems |
| 23:47.35 | fwae | Can ou buy a sipura adapter at any retail stores (compusa/bestbuy), or do they offer any unlocked selection? |
| 23:47.53 | christian[asgi] | Ash: are you running echo can in zap? |
| 23:48.07 | Ash | christian[asgi]: yep |
| 23:48.19 | Ash | turning up the echocancel/training settings helped a lot |
| 23:48.35 | Ash | but we are still seeing some echo |
| 23:48.38 | christian[asgi] | Ash: have you tried with Aggressive? |
| 23:48.52 | Ash | not yet |
| 23:48.59 | Ash | is it a whole lot better than the default? |
| 23:49.18 | christian[asgi] | Ash: Aggressive silences the line when you're not talking |
| 23:49.28 | Ash | mm, that might help |
| 23:49.37 | fwae | Can you unlock something like this for use on any serveR? http://www.bestbuy.com/site/olspage.jsp?skuId=6778365&type=product&id=1089889717675 |
| 23:49.39 | christian[asgi] | Ash.. it will still have to train a bit for sidetones I believe |
| 23:49.47 | Ash | because I think most of the problem is delayed sidetone audio |
| 23:49.48 | Ash | ah |
| 23:50.23 | Ash | yeah, but that's not too huge a deal |
| 23:50.32 | Ash | I will try building it with the aggressive option |
| 23:50.38 | christian[asgi] | Ash: good luck |
| 23:50.46 | Ash | thanks for the suggestions! |
| 23:51.09 | christian[asgi] | Ash: yw.. set the training=yes to something rediculously low like training=200 |
| 23:51.28 | Ash | okay |
| 23:53.38 | fwae | Is there ANY sip ATA that is at retial stores and unlockable? |
| 23:54.37 | christian[asgi] | fwae: haven't seen any. |
| 23:54.51 | christian[asgi] | fwae: where are you located? |
| 23:55.53 | *** join/#asterisk neron (n=chatzill@adsl-67-119-149-94.dsl.lsan03.pacbell.net) |
| 23:56.51 | Wi_Fi | does aah have an irc channel? |
| 23:57.38 | christian[asgi] | the a@h affirmation channel. Where the blind lead the blind. |