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| 00:01.49 | *** mode/#asterisk [+o mog] by ChanServ |
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| 00:12.09 | JunK-Y | yo yo yo |
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| 00:20.32 | roo9 | I have an asterisk setup i'm playing around with, right now I have it connected to my vonage account (via their softphone service), i'm able to get calls in (connecting to x-lite) however i'm not able to dial out (I get a 404 error), any idea where I should be looking at? |
| 00:21.01 | roo9 | I would at least like to isolate that it's not an issue with x-lite dialing out, so is there a way to initiate from inside asterisk |
| 00:21.21 | *** join/#asterisk RoyK (n=roy@ti211110a081-6523.bb.online.no) |
| 00:21.46 | *** join/#asterisk joako (n=joako@64.238.175.93) |
| 00:24.29 | joako | Does anyone have a clue on how to generate config files for provisioning for the Linksys phones? |
| 00:25.05 | fafnir | I do not, but I would bet that someone knows a really easy shortcut, and I will start the betting with 5 dollars. |
| 00:25.43 | joako | Well so far I searched the Voip-info wiki and the Linksys site and turned up nothing... |
| 00:31.15 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:33.06 | nentis | curious if anyone has experience with the UTstarcom or Linksys wifi voip phone. |
| 00:33.21 | nentis | Are the wireless phones worth it yet? |
| 00:33.26 | cekc | <3 digium support |
| 00:33.58 | nentis | it would be cool if they could scan open SSID's for the best connection to a voip gateway or asterisk server. |
| 00:35.18 | *** part/#asterisk diclophis (n=diclophi@65.203.37.58) |
| 00:37.06 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 00:39.51 | aptura | you mean test for robustness of a connection |
| 00:52.26 | ManxPower | If they did that, they would never connect to ANYTHING. |
| 00:52.51 | nentis | maybe per call? |
| 00:52.54 | nentis | :) |
| 00:53.05 | ManxPower | There are four things that always suck: blackholes, microsoft products, softphones, and wifi phones. |
| 00:53.30 | ManxPower | oh, and Grandstream products |
| 00:53.30 | MikeJ | no comment |
| 00:53.36 | nentis | heh. |
| 00:54.00 | nentis | I have three BT-100's. They're ok for a single-line cheap phone. |
| 00:54.30 | MikeJ | I have a bunch of 102's downstairs new in box.... |
| 00:54.54 | MikeJ | somone should buy them... |
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| 00:56.19 | nentis | what I would like to see if a wifi-gsm hybrid that will attempt to use VoIP with standard cell service as a secondary. |
| 00:56.19 | nentis | It will be a long time before that happens in the US. |
| 00:57.44 | MikeJ | nentis, truphone |
| 00:57.47 | *** join/#asterisk dofear (n=dofear@146-142-222-203.static.techex.net.au) |
| 00:58.35 | joako | nentis: the UTStarComm F3000 is crap, the Linksys WIP300 isn't bad... but I've only had it for today |
| 00:58.48 | nentis | interesting (truphone) |
| 00:59.21 | nentis | I used to work for a semiconductor company that would work with UTStarComm. Glancing at their build quality, all their equipment seems like crap. |
| 00:59.36 | nentis | But it was for digital set top boxes, which is all about lowering the cost. |
| 00:59.44 | MikeJ | they are using nokia E series phones.. |
| 01:00.00 | nentis | sort of how Qwest moved to Actiontek away from Cisco for DSL CPE's. |
| 01:00.05 | MikeJ | but they are already doing it now.. |
| 01:00.19 | joako | Well the phone itself was nice and seemed solid, I think the issue is just the software |
| 01:01.11 | joako | And the Linksys phone has way better RF or just can roam between AP better |
| 01:01.46 | joako | The UTStarcom phone I couldnt use outside my office, the Linksys phone I can almost walk around the entire building |
| 01:02.26 | nentis | does it support WPA/2? |
| 01:02.34 | nentis | hm. I can look that up myself. |
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| 01:04.17 | joako | F3000 I think only supports WPA |
| 01:04.25 | joako | WIP300 supports WPA and WPA2 |
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| 01:04.49 | joako | yea... F3000 only support WPA-PSK..... WIP also supports the corprate WPA |
| 01:06.38 | heison | ~seen bkw_ |
| 01:06.40 | jbot | bkw_ is currently on #asterisk, last said: '8ball says don't use asterisk'. |
| 01:07.53 | *** join/#asterisk __robby (n=robby@203.63.126.9) |
| 01:08.20 | pigpen2 | anyone know anything about faxgetty? < yes...very off topic... |
| 01:08.22 | __robby | hey all, i was wondering if anyone knows the manager interface syntax for the queuepause command? |
| 01:09.17 | pigpen2 | __robby, and sorry...not I. |
| 01:10.24 | __robby | ive used faxgetty a bit, whats up? |
| 01:11.30 | pigpen2 | yeah..and no luck for the syntax on the queuepause... |
| 01:11.42 | pigpen2 | Well, I am attempting to get iaxmodem setup with hylafax. |
| 01:11.58 | __robby | where have you gotten to with it? |
| 01:12.04 | pigpen2 | I have iaxmodem ready...but when I run faxgetty /dev/ttyIAX nothing happens. |
| 01:12.18 | pigpen2 | no hint, no bitches.... |
| 01:12.25 | *** join/#asterisk joako (n=joako@64.238.175.93) |
| 01:12.56 | pigpen2 | I may have to drop back and get it to work with a serial port modem....then try this again. |
| 01:13.11 | __robby | so it didnt throw any errors at all? |
| 01:13.27 | pigpen2 | none....just a carriage return. |
| 01:13.38 | pigpen2 | no info in dmesg, syslog, etc.... |
| 01:14.08 | __robby | paste your ttyIAX config file? |
| 01:14.14 | pigpen2 | k |
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| 01:17.42 | Star568 | hi all, anybody show me a sip channel setting for a cisco GW? ( prefix 3344, cisco GW ip 222:222:222:111 ) |
| 01:17.57 | Star568 | out going only |
| 01:18.14 | Star568 | i try send my call to that GW |
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| 01:34.19 | dan42 | ive noticed that when a sip client is unavailable but still has a registration that hasnt timed out, the calling party hears no ring until it goes to voicemail.. it works ok if i force ringback on the dial.. am i conceptually missing something, or should i be doing that for local dials |
| 01:42.02 | [TK]D-Fender | dan42 : thats because * is waiting for the phone you are calling to report back "SIP 180 ringing" |
| 01:42.25 | [TK]D-Fender | dan42 : * only provides indications for status' it knows about. |
| 01:42.45 | dan42 | right, i figured that, but i wasnt sure what the solution would be. giving the calling party dead air isnt so hot |
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| 01:43.29 | dan42 | i would have thought * would figure out that the end point wasnt there instead of ringing for max rings |
| 01:43.36 | [TK]D-Fender | dan42 : unfortunate, but necessary. Forcing indication is a bad thing |
| 01:43.59 | [TK]D-Fender | dan42 : You should fix the fact your peer is becoming unreachable :) |
| 01:44.00 | dan42 | in most cases it seems the calling party switch ends up hanging up the call and giving the calling party some kid of silly message |
| 01:44.30 | dan42 | [TK]D-Fender: cable modems go down.. tell me how i can stop cable modems from going down and we'll be rich |
| 01:44.31 | Flauto | hi all |
| 01:44.40 | Flauto | 1.4 is coming out |
| 01:44.46 | dan42 | [TK]D-Fender: im talking about end user devices here |
| 01:44.50 | Flauto | is the installation the same? |
| 01:45.03 | [TK]D-Fender | dan42 : Set the timeout faster so it jsut boms out immediately instead :) |
| 01:45.05 | Flauto | is there anything to prepare for the installation |
| 01:45.22 | symlink | qualify=yes will send an OPTIONS and measure the response and if there is none it'll be unreachable... but if you catch it at the right time, there will still be a delay |
| 01:45.24 | [TK]D-Fender | Flauto : Yes, the requisite Hokey-pokey! |
| 01:45.41 | Flauto | haha |
| 01:45.42 | Flauto | okay |
| 01:45.50 | Flauto | anything else? |
| 01:46.08 | [TK]D-Fender | Flauto : Don't forget the popcorn! |
| 01:46.09 | dan42 | symlink: right now if you call from nextel in this situation you get "the nextel subscriber you are calling is unavailable" .. not helpful :p |
| 01:46.19 | Flauto | hehe |
| 01:46.30 | symlink | dan42: ...? |
| 01:47.02 | Flauto | are you using 1.4 now? d-fender |
| 01:47.13 | dan42 | symlink: thats what nextel plays after a short bit of the dead air |
| 01:47.18 | symlink | dan42: cool |
| 01:47.40 | symlink | dan42: well, if you use qualify chan_sip will try it's best to determine whether phone is up or not... |
| 01:48.27 | dan42 | symlink: thanks.. ill check into that.. i inherited much of this setup, so im not sure off the top of my head |
| 01:49.41 | [TK]D-Fender | Flauto : Nope. I only use the major "normal" FTP releases |
| 01:49.50 | [TK]D-Fender | Flauto : I can aford to wait. |
| 01:50.19 | [TK]D-Fender | COOOOOKKKKKKIIIIIEEEESSSSS |
| 01:50.27 | joako | dan: and if anything just have asterisk answer the call and indicate ringing.... if its going to go to voicemail anyways it wont really matter |
| 01:50.28 | dan42 | symlink: actually, that wont work im going to assume since we're using realtime |
| 01:50.32 | [TK]D-Fender | munchMUNCHmunchMUNCHmunchMUNCHmunchMUNCHmunchMUNCH |
| 01:50.55 | symlink | unless you're doing caching... then it won't work aye |
| 01:51.13 | symlink | let me rephrase |
| 01:51.25 | symlink | if you are doing caching it should work permitted it is properly setup and the entry is in memory |
| 01:51.36 | symlink | if you are not doing caching, then throw the above out the window |
| 01:52.01 | dan42 | joako: i added a ",r" to the dialing to our local clients and it seems to work as one would expect, but im not sure what side effects making that change might have |
| 01:52.22 | symlink | it'll always return ringing, even in some cases where it is not |
| 01:52.26 | symlink | hides things |
| 01:52.55 | joako | Worst that will happen is the remote end will clip the first 1sec of the call |
| 01:53.47 | joako | usually mobile phones..... |
| 01:54.27 | dan42 | this is only a problem where it sees a registration that hasnt expired for an ATA that isnt actually online anymore |
| 01:54.45 | dan42 | once the registration expires, it drops right to voicemail unavailable |
| 01:55.09 | type0 | I'm trying to compile zaptel, and I'm getting this .. You do not appear to have the sources for the 2.6.9-34.EL kernel installed. |
| 01:55.14 | type0 | im running CentOS |
| 01:55.14 | Flauto | anyone here is using 1.4 now |
| 01:55.23 | symlink | that's because it is still reachable, it doesn't know it's not there... it still has to send out an INVITE and wait for it to time out |
| 01:56.03 | dan42 | symlink: i figured as much |
| 01:56.24 | joako | type0: from my understanding CentOS and RHEL are the same/similar. Install the kernel-sources package.... |
| 01:56.25 | Flauto | type0, get kernel source |
| 01:57.02 | joako | sort of related to the registration issue, right now I am using a SIP phone and sip show peers says UNREACHABLE even when I'm on the phone and I cant get inbound calls.... how can this be? |
| 01:57.36 | Flauto | anyone here is using 1.4? |
| 01:57.50 | symlink | placing calls and receiving calls are two separate things, a device does not have to be able to receive calls in order to place them |
| 01:57.55 | symlink | Flauto: perhaps, why? |
| 01:58.15 | Flauto | symlin, i want to know if the installation is the same |
| 01:58.31 | Flauto | so, i would see if i want to try that |
| 01:58.40 | Flauto | dont' want to run into too much trouble |
| 01:58.42 | SomeJ | joako: iv had that issue before, usualy that happens when a nat/firewall closes the pinhole to fast. Set the registration time on the phone to something low and that usualy helps the issue |
| 01:58.47 | symlink | the process? in order to build 1.4 you have to do more steps... but besides that not really |
| 01:58.50 | SomeJ | atleast it has for us |
| 01:59.13 | [TK]D-Fender | symlink : How much more really? |
| 01:59.15 | Flauto | symlink, what steps |
| 01:59.16 | joako | symlink: Ok., i've noticed when I can recieve inbound calls asterisk sip show peers says its at port 5060, when it cannot its something else like 1024 |
| 01:59.25 | dan42 | symlink: thanks for the help.. appreciate it.. gives me some stuff to look at |
| 01:59.30 | symlink | well you have to do ./configure now |
| 01:59.42 | symlink | to pick up what your system has so it can figure out what can be built |
| 01:59.54 | symlink | joako: NAT? |
| 02:00.43 | Flauto | so, no more make clean, make, make install? |
| 02:00.45 | joako | Yes.... But I swear everything else has always worked here! I've been using Linksys/Sipura this week and random issues... I was using an SPA3000 and I would have to rest the router AND spa to get it to work a few times.... |
| 02:00.59 | symlink | Flauto: yes you still have to do those |
| 02:01.08 | symlink | normally it's ./configure, make, make install |
| 02:01.12 | Flauto | configure first |
| 02:01.19 | Flauto | and then make ..... |
| 02:01.24 | symlink | but once you run ./configure you shouldn't have to do it again unless you install something and want Asterisk to use it |
| 02:01.44 | symlink | or the script gets regenerated in SVN... but that doesn't happen often |
| 02:02.24 | type0 | anyone ever get this? |
| 02:02.25 | type0 | [root@localhost zaptel]# modprobe ztdummy |
| 02:02.25 | type0 | Notice: Configuration file is /etc/zaptel.conf |
| 02:02.25 | type0 | line 0: Unable to open master device '/dev/zap/ctl' |
| 02:02.25 | type0 | 1 error(s) detected |
| 02:02.25 | type0 | FATAL: Error running install command for ztdummy |
| 02:02.27 | Flauto | okay |
| 02:02.36 | Flauto | you are using it? symlink? |
| 02:02.42 | symlink | type0: do it again... and see if it works |
| 02:02.56 | type0 | now I dont get anything |
| 02:02.58 | symlink | Flauto: I'm a developer, so of course lol |
| 02:03.04 | type0 | just back to # |
| 02:03.07 | Flauto | great |
| 02:03.20 | Flauto | it is working with google talk? |
| 02:03.32 | type0 | [root@localhost zaptel]# modprobe ztdummy |
| 02:03.32 | Flauto | you guys are great |
| 02:03.32 | type0 | [root@localhost zaptel]# |
| 02:03.35 | symlink | type0: thought so... /dev/zap/ctl wasn't created fast enough so it freaked out |
| 02:03.40 | type0 | oh alright |
| 02:03.42 | Flauto | i wish i would know more about this kind of stuff |
| 02:03.44 | symlink | Flauto: I don't use it with Google Talk, can't comment |
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| 02:07.39 | Flauto | symlink is the configuration very different in 1.4? |
| 02:08.02 | Flauto | like in extensions.conf, sip.conf........ |
| 02:08.09 | symlink | no... just new options |
| 02:08.15 | Flauto | okay |
| 02:08.20 | Flauto | i read a little bit about it |
| 02:08.29 | Flauto | it has fax function now |
| 02:08.49 | Flauto | does it still come with samples of the configs? |
| 02:08.57 | symlink | 1.4 has T.38 passthrough... and yes, of course there's samples |
| 02:09.24 | Flauto | okay |
| 02:09.28 | Flauto | i will install it |
| 02:09.40 | Flauto | t.38 is for fax? |
| 02:10.04 | symlink | sure. |
| 02:10.18 | dan42 | Flauto: dont abuse symlinks help :p go read :) |
| 02:10.39 | Flauto | yes, sir, would you give me a link |
| 02:10.43 | Flauto | i will read |
| 02:12.37 | Qwell | wow, I'm slow today |
| 02:12.43 | dan42 | well, you have the asterisk tarball.. theres also googe.com and voip-info.org |
| 02:12.48 | Qwell | "Who's symlink? I recognize that name from somewhere.." |
| 02:13.20 | symlink | Qwell: I am not the file you are looking for... |
| 02:13.28 | Qwell | gotcha |
| 02:14.57 | [TK]D-Fender | TK421 why aren't you at you at your post? |
| 02:15.40 | symlink | [TK]D-Fender: ! ! ! |
| 02:16.12 | [TK]D-Fender | symlink : I don't want relationship! |
| 02:16.25 | *** join/#asterisk jtexter3 (n=jtexter3@ip68-97-73-114.ok.ok.cox.net) |
| 02:17.48 | pigpen2 | I know this isn't related, but how do I fix this: with Faxgetty: faxgetty could not create FIFO permission denied |
| 02:18.04 | pigpen2 | I have been racking my brain....it hurts now. |
| 02:22.12 | dan42 | i could stab you in the eye |
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| 02:22.48 | pigpen2 | that would probably help. |
| 02:23.00 | pigpen2 | hmm...my dog just ate a golf ball. |
| 02:23.18 | dan42 | at least he's trying to help |
| 02:24.07 | pigpen2 | well, he is a she...and she is....well....ready....so this is going to be messy. |
| 02:24.34 | pigpen2 | oops...it is coming up. |
| 02:24.44 | X-Rob_ | or, it's choking. |
| 02:24.48 | X-Rob_ | which is more likely. |
| 02:26.06 | pigpen2 | yep...1 golf ball...and 1 sock. |
| 02:26.14 | pigpen2 | hmm..I guess I should feed her more.... |
| 02:26.21 | *** part/#asterisk hyphen (n=hyphen@71.224.213.97) |
| 02:27.26 | jtexter3 | just curious, is anyone here in or near Oklahoma? |
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| 02:27.35 | *** mode/#asterisk [+o mog] by ChanServ |
| 02:27.44 | pigpen2 | FaxGetty[30529]: Could not create FIFO.ttyIAX: Permission denied. |
| 02:27.46 | pigpen2 | geesh. |
| 02:28.00 | pigpen2 | Near Oklahoma? Texas close enough? |
| 02:28.27 | jtexter3 | What part of Texas? |
| 02:28.32 | pigpen2 | middle. |
| 02:28.42 | pigpen2 | San Antonio |
| 02:28.53 | Qwell | jtexter3: I'm not from OK, but I did stay at a Holiday Inn Express last night |
| 02:29.05 | dan42 | 'SA is hte city of the week.. 4th time its come up for me in a weeks time |
| 02:29.14 | jtexter3 | Qwell: LMAO |
| 02:29.21 | axscode | if i have a green module what should be in my zaptel.conf? |
| 02:29.30 | pigpen2 | yeah...I like it in my rear view mirror. |
| 02:29.37 | axscode | fxsks or fxoks |
| 02:29.48 | pigpen2 | fxs if I remember. |
| 02:30.06 | axscode | if i have RED RED GREEN GREEN... |
| 02:30.11 | [TK]D-Fender | FXOKS |
| 02:30.19 | axscode | ok fxoks |
| 02:30.21 | pigpen2 | but remember...I am distracted...my dog just barfed a sock. |
| 02:30.53 | axscode | so it should be [R][R][G][G] ---> [fxsks][fxsks][fxoks][fxoks] ? |
| 02:31.56 | axscode | if i gut my zaptel.conf wrongly assigned.. the ztcfg will give me error right? |
| 02:32.56 | pigpen2 | you would hope. |
| 02:33.27 | [TK]D-Fender | axscode : http://www.voip-info.org/wiki/view/TDM400P |
| 02:33.31 | [TK]D-Fender | READ |
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| 02:33.33 | *** mode/#asterisk [+o file2] by ChanServ |
| 02:33.56 | axscode | one more question.. if i have no error with ztcfg.. in my zapata.conf if i got it wrongly assigned is there an error? |
| 02:34.00 | [TK]D-Fender | This stuff is all in BIG PRINT. What with the guessing?! |
| 02:34.36 | axscode | nope TK.. ive been trying to confirm what ive done... coz ive been reading things and i cant get it right.. |
| 02:35.26 | axscode | when i try to call to pstn line... i have a problem.. the phone rings.. but when its picked up.. its still ringing.. |
| 02:36.09 | pigpen2 | do you see it ring in the cli? |
| 02:36.16 | axscode | yes.. |
| 02:36.22 | axscode | zap/3-1 ringing |
| 02:36.29 | pigpen2 | ok..this is a good sign. |
| 02:36.51 | pigpen2 | ok..so what phone is ringing? sip/iax/zap? |
| 02:36.55 | [TK]D-Fender | Zap/3 is an FXS port...... you can ring it as long as you want.... |
| 02:37.39 | axscode | TK... im dialing Dial(Zap/3/${EXTEN},20,t) |
| 02:38.09 | axscode | the phone really rings at the end.. but when its picked up.. it is still ringing.. |
| 02:38.11 | pigpen2 | axscode, zap/3 should be going to an internal analog phone. |
| 02:38.26 | pigpen2 | ie: zap/3 is providing dialtone |
| 02:38.53 | fafnir | wait wait |
| 02:39.03 | fafnir | are you the pigpen from charlie brown? |
| 02:39.13 | pigpen2 | do you see dust? |
| 02:39.23 | fafnir | this is the internet |
| 02:39.30 | fafnir | dust does not travel well over the internet |
| 02:39.32 | pigpen2 | then sure...that is me. |
| 02:39.42 | [TK]D-Fender | fafnir : now you kick it! |
| 02:39.44 | pigpen2 | actually, it does...but it is called spam. |
| 02:39.49 | fafnir | how did you know that i was charlie brown? o.0 |
| 02:39.57 | fafnir | oh |
| 02:39.59 | fafnir | that was you? |
| 02:39.59 | axscode | sorry... please help me maybe im confused.. |RRGG <-- this is my TDM22B |
| 02:40.33 | axscode | if im getting a dialtone in a port what port is that? |
| 02:40.59 | [TK]D-Fender | axscode : RRG means Zap/3 is a PHONE jack (FXS), not a LINE jack (FXO). |
| 02:41.01 | pigpen2 | axscode, fxs ports provide dialtone...but it uses fxo signaling. |
| 02:41.26 | fafnir | ARRRRRGGGGGHHHHH |
| 02:42.03 | [TK]D-Fender | Schultz = eternal |
| 02:42.05 | pigpen2 | fafnir, I used to be "drsperm" but no one would talk to me. |
| 02:42.12 | fafnir | i would :( |
| 02:42.38 | pigpen2 | : |
| 02:42.45 | axscode | <PROTECTED> |
| 02:42.47 | axscode | confusing |
| 02:43.06 | *** join/#asterisk xpato (n=pato@pc-33-21-104-200.cm.vtr.net) |
| 02:43.10 | pigpen2 | axscode, look at it this way. 1/2 is fxs - 1/2 is fxs... |
| 02:43.16 | xpato | i have a very newbie question. |
| 02:43.17 | pigpen2 | you only really have 4 combinations. |
| 02:43.22 | axscode | Sep 24 15:16:11 sip1 kernel: Module 0: Installed -- AUTO FXO (FCC mode) |
| 02:43.23 | axscode | Sep 24 15:16:11 sip1 kernel: Module 1: Installed -- AUTO FXO (FCC mode) |
| 02:43.23 | pigpen2 | oh crap. |
| 02:43.42 | axscode | Sep 24 15:16:11 sip1 kernel: Module 2: Installed -- AUTO FXS/DPO |
| 02:43.45 | axscode | Sep 24 15:16:11 sip1 kernel: Module 3: Installed -- AUTO FXS/DPO |
| 02:43.47 | xpato | if i have a hardware central->PSTN, and 500 internal extensions |
| 02:43.58 | axscode | thats what in my cat /var/log/messages |
| 02:44.03 | pigpen2 | right. |
| 02:44.25 | pigpen2 | axscode, mod 0 & 1 (zap 1 & 2 ) are fxo's.... |
| 02:44.28 | xpato | i should asterisk to make this 500 exts voip and pass the calls through the hardware central? |
| 02:44.42 | xpato | i should use |
| 02:45.07 | axscode | pigpen2: ok with zap/1 and zap/2 i cant get a dialtone if a put a PHONE right? |
| 02:45.16 | pigpen2 | right. |
| 02:45.31 | *** join/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 02:45.33 | axscode | ok |
| 02:45.33 | pigpen2 | they are rec'ing the voltage from the pstn. |
| 02:45.58 | pigpen2 | xpato, errr....hardware central? |
| 02:46.09 | axscode | signalling=fxs_ks |
| 02:46.10 | pigpen2 | meaning pri's and such? |
| 02:46.11 | axscode | group=1 |
| 02:46.11 | axscode | context=axscode |
| 02:46.11 | axscode | channel => 1 |
| 02:46.22 | axscode | is this right? pigpen??? for my channel => 1 |
| 02:46.34 | axscode | thats in my zapata.conf |
| 02:46.38 | xpato | pigpen2: ericcson md110 |
| 02:47.03 | xpato | pigpen2: im thinking something like this |
| 02:47.32 | xpato | pstn <-> ericcson_md110 <-> asterisk <-> internal voip phones |
| 02:47.38 | pigpen2 | axscode, yeah...looks good. |
| 02:47.48 | pigpen2 | mine are backwards..but I was too lazy to swap the modules. |
| 02:48.00 | axscode | ok.. so i should be dialing to zap/1 or zap/2 not zap/3 right? |
| 02:49.06 | pigpen2 | the telco is connected to 1 & 2 |
| 02:49.19 | pigpen2 | your cheap $5 walmart phone is on 3 & 4 |
| 02:49.36 | axscode | hmmm ok thanks.. so zap/1 and zap/2 in my dial plan. |
| 02:49.40 | pigpen2 | xpato, why not connect the pstn directly to asterisk? |
| 02:49.55 | pigpen2 | ie: through digium or other? |
| 02:51.36 | xpato | we are thinking on that too, but i want to know if what i said works? |
| 02:51.51 | pigpen2 | depends how you deliver it to the * box. |
| 02:51.53 | xpato | i know what you say works, and how |
| 02:52.08 | xpato | pigpen2: what you mean? |
| 02:52.37 | pigpen2 | does the md110 deliver the calls to/from via PRI's, SIP, ?? |
| 02:53.13 | pigpen2 | if SIP, is it standard? I am not familiar with this hardware. |
| 02:53.54 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
| 02:54.07 | [TK]D-Fender | pigpen : http://www.ericsson.com/solutions/enterprise/products/md110.shtml |
| 02:54.19 | [TK]D-Fender | pigpen : Whole damn PBX |
| 02:54.21 | pigpen2 | But with 500 exten's, I imagine that no more than 4 pri's are needed...unless there is allot of call volume...like a call center. |
| 02:54.34 | pigpen2 | [TK]D-Fender, yeah..why asterisk then? |
| 02:55.01 | pigpen2 | seems like too much work for a simple net result. |
| 02:55.20 | teknoprep | having an echo problem with SIP over the INET to a PBX? what is a very good ATA to fix this problem? |
| 02:55.44 | pigpen2 | try a better ISP. |
| 02:55.49 | xpato | i make a mistake, no very good english, we are comparing md110 to asterisk |
| 02:56.10 | pigpen2 | ah..well, my english isn't good either...I live in Texas. |
| 02:56.10 | teknoprep | pigpen2 that is a shitty response... and completely rediculous |
| 02:56.13 | teknoprep | this is for home fool |
| 02:57.03 | pigpen2 | fool...yeah..that will get you help. |
| 02:57.17 | pigpen2 | [TK]D-Fender, can you boot him? |
| 02:57.52 | teknoprep | with the repsonse you gave me... i doubt i need your help |
| 02:58.18 | pigpen2 | xpato, well, like I said, I am not familar with the md110, and I realy don't feel like reading up on it. But I can say for the several 250+ deployments using * is working very nice. |
| 02:59.16 | pigpen2 | teknoprep, dude...you said you had echo issues with sip over the internet. Most of the echo people get are ISP related. |
| 02:59.16 | pigpen2 | You popped off for no reason. |
| 02:59.34 | [TK]D-Fender | teknoprep : Describe the full call path./ |
| 02:59.55 | [TK]D-Fender | pigpen : Cool off okay? |
| 03:00.05 | [TK]D-Fender | All of you..... |
| 03:00.11 | pigpen2 | Yes dad. |
| 03:00.19 | pigpen2 | :) |
| 03:00.19 | justinu|laptop | settle down, beavis |
| 03:00.31 | teknoprep | no i am the great cornholio |
| 03:00.39 | xpato | pigpen2: ok, fair answer, and last question, if the md110 (or another that you know) has pri support, i should connect this via an E1 with *? |
| 03:00.56 | xpato | teknoprep has the period |
| 03:01.02 | [TK]D-Fender | xpato : I'm a little unclear. Do you already hav an MD110? |
| 03:01.21 | pigpen2 | xpato, yeah..I am abit unclear too. |
| 03:01.53 | pigpen2 | teknoprep, so how is your setup configured. |
| 03:02.16 | [TK]D-Fender | teknoprep : So are you going to describe the call path the you receive echo on for us? |
| 03:02.30 | teknoprep | i am just going to setup an IAX trunk from a server here at home to the office |
| 03:02.39 | xpato | no, like i firs say, im new to voip :). we are looking for a PBX. so my boss think about a hardware PBX (md110) to connect to the PSTN and asterisk to manage the voip extens |
| 03:02.52 | xpato | the people of md110 says it can make all the work |
| 03:03.01 | xpato | i said asterisk can too |
| 03:03.02 | [TK]D-Fender | xpato : Oh what are you using right now? |
| 03:03.24 | pigpen2 | teknoprep, I am running the same here. |
| 03:03.35 | xpato | nothing, because if for a new deployment |
| 03:03.40 | xpato | is |
| 03:03.53 | pigpen2 | IAX trunk between the office and my home system....I see my echo's when my isp gets nailed from home users. |
| 03:03.59 | [TK]D-Fender | xpato : Ok, what kind of wiring do you have in the building for this new deployment? |
| 03:04.00 | pigpen2 | gsm codec helps.... |
| 03:04.22 | xpato | so i need to answer, what would we need to connect the asterisk with the md110 |
| 03:04.34 | xpato | cat5 |
| 03:04.46 | pigpen2 | teknoprep, was that helpful? |
| 03:04.54 | teknoprep | lol |
| 03:05.03 | pigpen2 | teknoprep, in regards to ATA's Sipura makes very good equipment. |
| 03:05.09 | teknoprep | ty |
| 03:05.12 | teknoprep | thats all i asked |
| 03:06.41 | pigpen2 | xpato, it just seems that you will have two systems when all you need is one. |
| 03:07.00 | xpato | pigpen2: i really know that |
| 03:07.10 | [TK]D-Fender | xpato : How many phones/etc are you planning on running on the * server? |
| 03:07.13 | xpato | but i really need to answer the question :) |
| 03:07.22 | xpato | 500 extensions |
| 03:07.51 | xpato | 4/4 lines |
| 03:07.52 | [TK]D-Fender | xpato : And what is the MD110 going to be doing then? |
| 03:08.41 | pigpen2 | [TK]D-Fender, does the md110 run sip? |
| 03:08.54 | xpato | ok, i know its very irratating :) but keeping aside that having the md110 and asterisk is stupid, what would i need |
| 03:09.01 | teknoprep | how do i route an inbound route connection to a trunk for another pbx to handle it? |
| 03:09.01 | [TK]D-Fender | pigpen2 : Dunno, didn't read up on it in any detail. |
| 03:09.26 | axscode | TK ----- if a have a panasonic pbx... i will consider that pbx as my pstn? |
| 03:09.38 | pigpen2 | [TK]D-Fender, k, I thought you might be familiar...I am looking into it... |
| 03:09.54 | [TK]D-Fender | xpato : DEPENDS what you are going to do with the MD110. I can't suggest a good car for you if I don't know what you are going to use it for! Would you suggest a Ferrari to do groceries with?! |
| 03:10.06 | symlink | [TK]D-Fender: yes. |
| 03:10.07 | *** join/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone) |
| 03:10.14 | [TK]D-Fender | pigpen2 : No... I just googled it in 5 seconds flat and linked :) |
| 03:10.32 | [TK]D-Fender | symlink : And thats why you're not even a HARD link! ;) |
| 03:10.45 | symlink | eep |
| 03:11.13 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-246-145.buckeyecom.net) |
| 03:11.14 | pigpen2 | I think it does support sip... |
| 03:11.32 | xpato | it does support sip |
| 03:11.47 | gambolputty | Hi. In * 1.4.beta2, voicemails are being recorded with a time of UTC instead of my local time. Any ideas on how to fix this? |
| 03:11.58 | [TK]D-Fender | xpato : if it does SIP why do you need * for that then? |
| 03:12.34 | xpato | because is does not do voicemail and billing |
| 03:12.52 | xpato | well it does, but cost a lot extra |
| 03:13.09 | pigpen2 | [TK]D-Fender, the few deployments I found, use the md110 to connect to the phones, and asterisk for the sip proxy. |
| 03:14.01 | *** join/#asterisk Katty (n=Administ@dialup-4.244.180.239.Dial1.StLouis1.Level3.net) |
| 03:14.55 | xpato | pigpen2: and how they make the md110 to talk with asterisk |
| 03:14.55 | Katty | hey hun (= |
| 03:15.27 | [TK]D-Fender | Katty : Mew. |
| 03:15.42 | [TK]D-Fender | Katty : So got your auto-answer working to your satisfaction? |
| 03:15.50 | Katty | haha, i wish. |
| 03:15.59 | [TK]D-Fender | Katty : I did it myself that very weekend. |
| 03:16.04 | Katty | maybe if i had 10 minutes to sit down at my desk and actually get some work done ;) |
| 03:16.07 | Katty | awesome! |
| 03:16.14 | Katty | that means you can hold my hand if i get lost (= |
| 03:16.20 | Katty | or at least i can hope, haha |
| 03:16.24 | [TK]D-Fender | Katty : I'll keep an eye out for you and help you get this ironed out. |
| 03:16.34 | Katty | whoo! |
| 03:16.36 | pigpen2 | xpato, I am not sure..I just googled "md110 asterisk sip" |
| 03:16.38 | Katty | you're the best (= |
| 03:16.50 | [TK]D-Fender | Or at least a reasonable facsimile ;) |
| 03:17.05 | Katty | very punny. |
| 03:17.52 | Katty | twisted[work]: are you /really/ at work still? |
| 03:17.57 | Katty | twisted[work]: i think you're fibbing. |
| 03:21.37 | teknoprep | how is the IAXY... does it have the same problems with echo as SIP? |
| 03:21.42 | teknoprep | the iaxy ata |
| 03:22.29 | Katty | i don't even know what iaxy ata is. |
| 03:22.34 | Katty | i know that iax doesn't have echo tho. |
| 03:23.02 | Katty | well... |
| 03:23.13 | Katty | analog lines have echo on them, that's a factof life...unfortunately |
| 03:23.22 | teknoprep | hmm |
| 03:23.24 | teknoprep | that sucks |
| 03:23.41 | Katty | so i guess if an analog line was going across the iax protocol, it'd still have that echo. |
| 03:23.56 | Katty | best you can do for that, is port it down a t1 to a channel bank, which turns itinto analog lines |
| 03:23.58 | [TK]D-Fender | teknoprep : Are you ready to answer my question now? |
| 03:24.00 | Katty | tho it really isn't analog. |
| 03:24.05 | teknoprep | [TK]D-Fender, sure whats up |
| 03:24.27 | [TK]D-Fender | teknoprep : I asked you twice to describe the full call path where you experience echo.... |
| 03:25.00 | *** join/#asterisk Ox0F0-0FF (n=pierre@200.216.238.226) |
| 03:25.17 | Katty | fender: down mainstreet, across town, under themississippi river ;) |
| 03:25.43 | teknoprep | analouge phone - voip ata (sipura spa1001) - Firewall (QoS) - Comcast Internet (home) - Comcast Ineternet (office) - Asterisk PBX |
| 03:25.54 | Katty | i do wonder how they got phone cally and internet stuffs under the mississippi |
| 03:25.58 | Katty | how'd they do that? |
| 03:26.11 | Katty | burried fiber? |
| 03:26.21 | Katty | in conduit or something |
| 03:26.28 | pigpen2 | you would hope it is burried. |
| 03:26.40 | Katty | well maybe it's /not/ burried |
| 03:26.45 | Katty | maybe it's transmitted via satelite |
| 03:26.56 | justinu|laptop | satellite sucks... high latency |
| 03:27.05 | Katty | what about shiny red lights. |
| 03:27.05 | justinu|laptop | 750+ms |
| 03:27.09 | pigpen2 | I am in Texas. Most the rivers are dry now. |
| 03:27.09 | [TK]D-Fender | teknoprep : I'm suspecting you are leaving part of this picture out...... |
| 03:27.12 | Katty | like microwaves. |
| 03:27.27 | Katty | nukerwaved internet! |
| 03:27.28 | teknoprep | [TK]D-Fender, nope... what part |
| 03:27.43 | justinu|laptop | it's an underwater cable, there's signs telling boats not to anchor or dredge near them |
| 03:27.47 | teknoprep | [TK]D-Fender, there is 0 NAT at the office between the Office Modem and the PBX |
| 03:27.53 | [TK]D-Fender | teknoprep : You are talking directly and ONLY to * (doing what, voicemailmain?) and getting echo? |
| 03:28.00 | *** join/#asterisk pdt (n=pdthome@c-68-53-40-50.hsd1.tn.comcast.net) |
| 03:28.04 | Katty | justinu|laptop: oh, ah. k |
| 03:28.13 | *** join/#asterisk nortex (n=nortex@64.136.89.54) |
| 03:28.15 | teknoprep | [TK]D-Fender, oh then asterisk would be going out like this |
| 03:28.19 | Katty | what about oceans? |
| 03:28.37 | justinu|laptop | same thing, just longer and there's repeater stations anchored on the ocean floor |
| 03:28.39 | teknoprep | Asterisk PBX - Comcast (office) - internet - VoicePulse |
| 03:28.42 | Katty | woah. |
| 03:28.43 | [TK]D-Fender | teknoprep : Please revise your presentation of "FULL call path" please.... |
| 03:28.44 | pigpen2 | Katty, sat & trans atlantic fiber. |
| 03:28.47 | Katty | that's awesome (= |
| 03:28.52 | justinu|laptop | yeah, it is cool |
| 03:28.55 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 03:29.00 | Katty | marinekatologist |
| 03:29.12 | pigpen2 | justinu|laptop, dam ...you are full of info tongiht. |
| 03:29.18 | justinu|laptop | ;) |
| 03:29.21 | [TK]D-Fender | teknoprep : BETTER. Guess what, VoicePulse could have shitt EC on their end. I've seen it before. |
| 03:29.30 | Katty | what about sharks? |
| 03:29.37 | Katty | and other creature features, that have teeth. |
| 03:29.43 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 03:29.43 | teknoprep | analouge phone - voip ata (sipura spa1001) - Firewall (QoS) - Comcast Internet (home) - Comcast Ineternet (office) - Asterisk PBX - Comcast (office) - internet - VoicePulse |
| 03:29.45 | Katty | or earthquakes. |
| 03:29.54 | Katty | or....or...or... things. |
| 03:30.06 | Katty | under the ocean. that move. haha, i'm such a dork sometimes (= |
| 03:30.09 | pigpen2 | Katty, they have crews that are equiped to fix these...but it does happen. |
| 03:30.21 | Katty | i'm surprised the internet doesn't go down more. |
| 03:30.28 | [TK]D-Fender | teknoprep : I've heard echo on calls straight off an ATA to Vonage, etc. it happens |
| 03:30.32 | orlock | pigpen2: them guys are hardcore |
| 03:30.47 | pigpen2 | orlock, yeah...it was a very cool show. |
| 03:30.56 | pigpen2 | my wife thought it was stupid. |
| 03:31.09 | pigpen2 | I wouln't let her use her laptop for a week. |
| 03:31.11 | orlock | pigpen2: read "Mother Board, Mother Earth" |
| 03:31.19 | justinu|laptop | Katty: this cable is huge, and eqs can really screw them up |
| 03:31.33 | Katty | is it bigger than me? |
| 03:31.49 | pigpen2 | errr...how big /are/ you? |
| 03:31.54 | justinu|laptop | probably at least a 1 foot diameter cable |
| 03:32.02 | Katty | wowser. |
| 03:32.13 | Katty | that's pretty big. |
| 03:32.19 | justinu|laptop | it took them something like 40 years to learn how to lay a cable that would actually work |
| 03:32.31 | justinu|laptop | they tried all sorts of things that failed the first time they used them, or over a short period of time |
| 03:32.56 | Katty | i guess it had to withstand a lot of pressure |
| 03:33.02 | justinu|laptop | i guess a big part of understanding how we transmit stuff across wires comes form that knowledge |
| 03:33.13 | orlock | justinu|laptop: yeah |
| 03:33.32 | orlock | justinu|laptop: read "Mother Board, Mother Earth" |
| 03:33.39 | orlock | neal stephenson goes into it all in detail |
| 03:33.47 | Katty | i bet he's doing good to read the cereal box in the morning. |
| 03:33.52 | orlock | history of undersea cables to global fibre networks |
| 03:33.58 | Katty | we should start putting useful computery tips on cereal boxes. |
| 03:34.04 | Katty | maybe i'd get less tech support calls that way |
| 03:34.11 | teknoprep | [TK]D-Fender, we have 0 echo at the office |
| 03:34.21 | teknoprep | [TK]D-Fender, it NEVER echo's at the office |
| 03:34.39 | Katty | tip 1: if you're computer doesn't work, reboot and try again BEFORE contacting ANYONE |
| 03:34.52 | Katty | there'd go half my issues (= |
| 03:35.09 | justinu|laptop | hmm, i can't find it on amazon |
| 03:35.14 | justinu|laptop | this guy wrote a lot of stuff |
| 03:35.35 | Katty | so how do we get internet to the moon? |
| 03:35.38 | [TK]D-Fender | teknoprep : Could be exacerbated by the extra hops and equipment.... |
| 03:35.47 | [TK]D-Fender | Katty : LASERS! |
| 03:36.02 | Katty | cause polycom is what nasa uses, right? |
| 03:36.15 | Katty | i seem ot remember something about that...once upon a time. |
| 03:36.22 | [TK]D-Fender | Yes... this is an absolute farce! |
| 03:36.27 | Katty | mew? |
| 03:36.29 | Katty | you do not parse. |
| 03:36.36 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 03:36.41 | justinu|laptop | yeah, unless we figure out how to break the speed of light, internet on the moon is gonna suck |
| 03:36.58 | justinu|laptop | 4 second latency, iirc |
| 03:37.02 | Katty | spooky entanglement |
| 03:37.13 | Katty | that's where it's at. |
| 03:37.16 | justinu|laptop | probably |
| 03:37.23 | justinu|laptop | make us a working prototype, katty ;) |
| 03:37.24 | teknoprep | [TK]D-Fender, i am just hooking up an asterisk box at his home |
| 03:37.28 | Katty | okay! |
| 03:37.30 | justinu|laptop | kewl |
| 03:37.37 | Katty | i'll do it overlunch tomorrow |
| 03:37.44 | teknoprep | [TK]D-Fender, he likes the idea... also i will be just setting up one at my home... works better |
| 03:37.55 | teknoprep | later all |
| 03:38.15 | [TK]D-Fender | justinu|laptop : Chan_fluxcapacitor! |
| 03:38.32 | justinu|laptop | oscillator overthruster :) |
| 03:38.41 | justinu|laptop | oscillation? |
| 03:38.41 | Katty | you guys are nutty. |
| 03:38.46 | *** join/#asterisk SubWolf (n=rob@69.92.38.7) |
| 03:38.52 | Katty | what's that? |
| 03:38.58 | justinu|laptop | don't tell me you've never seen Buckaroo Banzai in the 9th dimension |
| 03:38.59 | Katty | a horny oscilliscope? |
| 03:39.10 | *** part/#asterisk SubWolf (n=rob@69.92.38.7) |
| 03:39.14 | Katty | can't say i have |
| 03:39.19 | justinu|laptop | er 8th dimension |
| 03:39.25 | justinu|laptop | it's an awesome movie |
| 03:39.35 | Katty | so is my twirly video |
| 03:39.48 | [TK]D-Fender | Duck Dodgers...... in the 24th and half CENTURY!@!@!@!@ |
| 03:39.49 | Katty | we were testing motion detect, so i went out and did a littl twirly dance ot make sure it worked right |
| 03:39.58 | Katty | url available upon request, etc. |
| 03:40.07 | justinu|laptop | btw, nasa uses cisco call manager for ip calls from the space shuttle |
| 03:40.09 | justinu|laptop | and ISS |
| 03:40.18 | Katty | ISS? |
| 03:40.21 | justinu|laptop | they should be using asterisk, much cheaper |
| 03:40.26 | Katty | oh god. |
| 03:40.29 | Katty | god no |
| 03:40.31 | Katty | PLEASE GOD NO |
| 03:40.33 | justinu|laptop | international space station |
| 03:40.34 | Katty | i mean. |
| 03:40.37 | Katty | yeah, cheaper... |
| 03:40.54 | Katty | symlink: nini. |
| 03:41.01 | justinu|laptop | no trolling, katty ;) |
| 03:41.02 | Katty | symlink: did you ever see the twirly? |
| 03:41.06 | symlink | Katty: no :( |
| 03:41.11 | Katty | symlink: aww? |
| 03:41.19 | symlink | very sad :( |
| 03:41.23 | jtexter3 | Okay Katty, now you have to share the URL |
| 03:41.31 | Katty | justinu|laptop: i'm not a troll, kthx. |
| 03:41.38 | justinu|laptop | :P |
| 03:41.40 | jtexter3 | Anything titled "twirly video" has to be worth a few laughs :D |
| 03:41.50 | Katty | jtexter3: sorry, i only share things like that with my friends (= |
| 03:42.10 | Katty | jtexter3: i wouldn't want it to leak into the wrong hands. and then a be a troll. |
| 03:42.21 | jtexter3 | hahaha |
| 03:42.28 | Katty | symlink: what's wrong? :< |
| 03:42.40 | symlink | no moooooovie |
| 03:42.51 | Katty | well go get the geox codec. |
| 03:44.24 | Katty | i never could find a converter )= |
| 03:46.45 | justinu|laptop | you mean a transcoder? :> |
| 03:47.33 | Katty | transcodecer. |
| 03:47.45 | Katty | executive transcodecer. |
| 03:48.34 | Katty | that's a new word, i just made up. |
| 03:48.46 | Katty | and i'm backing it up, with this codec developed forme by the CCTV organization. |
| 03:49.51 | *** part/#asterisk jtexter3 (n=jtexter3@ip68-97-73-114.ok.ok.cox.net) |
| 03:49.56 | symlink | Katty: you should become a professional pillow |
| 03:50.19 | Katty | that'd be a horrible waist of my brain. |
| 03:50.31 | Katty | and my typing skills, apparently. |
| 03:50.39 | symlink | Katty: :( |
| 03:51.16 | *** join/#asterisk BhaalWK (i=bhaal@freenode/unconfirmed/bhaal) |
| 03:51.37 | symlink | Katty: have you been up to no good? |
| 03:52.01 | Katty | yes, i admit. i helped university students write on the sidewalks with chalk today. |
| 03:52.10 | symlink | you rebel you |
| 03:52.14 | Katty | i know. |
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| 04:01.08 | xpato | back again |
| 04:01.47 | xpato | asterisk would be used to mailvoice and billing |
| 04:01.59 | xpato | and md110 for the phones |
| 04:02.26 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
| 04:03.14 | [TK]D-Fender | xpato : the only way * could do billing is if all calls passed through it. How do you plan on doing that? How many lines are you going to have coming in? |
| 04:03.36 | xpato | i did deduce tha |
| 04:03.54 | xpato | think in 32 lines in and out |
| 04:04.06 | xpato | what should i have |
| 04:04.07 | [TK]D-Fender | xpato : And if all of your calls are going through * then your MD110 serves no purpose. You'd be better off using normal SIP phones. |
| 04:04.10 | *** join/#asterisk aptura (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 04:04.21 | [TK]D-Fender | xpato : 32 lines for 500 extensions? |
| 04:04.32 | Katty | that's a lot of extensions. |
| 04:04.46 | Katty | we have 8 for 14 |
| 04:05.02 | xpato | we have the md110 |
| 04:05.08 | xpato | they have |
| 04:05.22 | xpato | and they are not going to change the phones |
| 04:05.35 | [TK]D-Fender | xpato : you just said they DIDN'T. Can you at least be consistent? |
| 04:05.47 | xpato | its for a callcenter (spamcenter) company |
| 04:06.11 | xpato | [TK]D-Fender: sory, but i just get the correct info |
| 04:06.16 | xpato | they do have the md110 |
| 04:06.23 | JT | callcenters usually need almost as many lines as callcentre extensions |
| 04:06.41 | xpato | and want asterisk to do the billing and voicemail |
| 04:06.49 | Katty | and equal ammounts of pain killers. |
| 04:07.13 | *** join/#asterisk Schulich (n=Jazba@165.154.103.158) |
| 04:07.28 | [TK]D-Fender | Katty : Fruit flavoured Tums are just like Sweet-Tarts you know..... |
| 04:07.38 | Katty | i don't eat tums. |
| 04:07.46 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 04:07.51 | Katty | maybe it's cause i'm not as old as you are ;) |
| 04:08.14 | xpato | [TK]D-Fender: so for this, would be something like this: pstn <-> asterisk <-> md110 <-> phones? |
| 04:08.32 | Katty | what /is/ an md110? |
| 04:08.47 | Katty | some sort of analogy converter |
| 04:09.02 | aptura | katty what ver of asterisk are you running |
| 04:09.44 | Katty | the kind that makes you ask questions. |
| 04:09.49 | Katty | see! it does an awesome job. |
| 04:09.51 | xpato | md110 is a ericcson pbx |
| 04:10.04 | Katty | oh ah. |
| 04:10.07 | aptura | this is not good my host.conf is empty. |
| 04:10.18 | Katty | google it. |
| 04:10.28 | symlink | Katty: mashed potatoes! |
| 04:10.30 | Katty | or voip-info a demo copy of it |
| 04:10.37 | Juggie | aptura, thats because its hosts.conf :) |
| 04:10.38 | Katty | symlink: with bbq sauce! |
| 04:12.08 | aptura | Juggie yea btw what does your say |
| 04:12.11 | [TK]D-Fender | xpato : Well I'm pretty supre you'll want a 4 port E1 card in that box. |
| 04:12.22 | [TK]D-Fender | xpato : 2 for inbound, to for out-bound |
| 04:15.01 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 04:17.01 | xpato | [TK]D-Fender: processor, for 64 concurrent conversations? |
| 04:17.54 | *** join/#asterisk r0d3nt (n=RatMan@69.31.131.44) |
| 04:18.02 | *** join/#asterisk michaelo (n=michaelo@adsl-4-148-78.gsp.bellsouth.net) |
| 04:18.06 | [TK]D-Fender | xpato : For passthrough E1, it doesn't mattet that much. P4 -3Ghz w/1 gig ram should be nice |
| 04:19.13 | JT | i doubt they'd be 32 channels either |
| 04:19.16 | JT | more like 30 each |
| 04:19.23 | blitzrage | 30B+2D |
| 04:19.39 | JT | 1 channel is used by E1 framing, another is usually used by D, if it's a PRI |
| 04:19.45 | JT | blitzrage: bzzt |
| 04:19.56 | blitzrage | *snap* |
| 04:19.59 | blitzrage | *crackle* |
| 04:20.01 | blitzrage | *pop* |
| 04:20.12 | JT | an E1 does not have two D channels |
| 04:20.22 | blitzrage | ? |
| 04:20.30 | blitzrage | show's what the stupid north american knows |
| 04:20.52 | JT | it has a single D channel if it's PRI/ISDN |
| 04:21.01 | JT | the other channel is used for framing |
| 04:28.19 | axscode | TK i got it working.... my TDM22B.. i can now call outside pstn... i got question though.. how to manage the incomming call or how to detect? |
| 04:30.07 | [TK]D-Fender | axscode : ... |
| 04:30.08 | [TK]D-Fender | ~book |
| 04:30.15 | jbot | methinks book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 04:36.02 | axscode | where the ivr of asterisk resides? |
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| 04:46.02 | BlepsoaF | hello all, does anyone know what I have to do, to have perl print to asterisk's console - I'm printing to STDERR |
| 04:46.50 | axscode | how can i wait for 3 rings before auto-answering a call? |
| 04:47.11 | tuck3r | axscode: Wait(X) |
| 04:47.27 | JT | you have to work out what period of time 3 rings is |
| 04:48.00 | axscode | where X = seconds? or number of rings? |
| 04:48.08 | tuck3r | x is seconds |
| 04:48.12 | axscode | ty |
| 04:50.00 | BlepsoaF | anyone know what this doesnt work foreach my $i ( keys %input ) { print STDERR " -- $i = $input{$i} ";} |
| 04:50.25 | BlepsoaF | im setting $|=1 & pulling data like my %input = $AGI->ReadParse(); |
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| 04:54.14 | BlepsoaF | anyone? |
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| 05:26.10 | w32 | if they can build iecho cancellation into a device,and asterisk acts as a device to a provider would it be possible to develop/implment a module for echo cacellation WITHIN asterisk ? |
| 05:27.07 | axscode | nothing is impossible. |
| 05:27.37 | w32 | how far from seeing such a thing are we ? |
| 05:27.57 | JT | err asterisk i thought already had software echo cancellation |
| 05:28.55 | w32 | Perhaps it does ? I'm interested in it as it relates to Faxing |
| 05:29.18 | JT | oh fun |
| 05:29.50 | w32 | yeah, doesn't seem to be very reliable from what I have read |
| 05:29.50 | JT | echo cancellation tends to screw up faxing |
| 05:30.07 | JT | yeah you should be looking at a FoIP solution if you must |
| 05:30.23 | JT | but trying to push fax over VoIP will only end in tears |
| 05:30.48 | JT | you can terminate fax calls to the pstn on an asterisk box |
| 05:31.24 | w32 | I had been doing some reading on it, but foip doesn't seem to be completely compatible with every fax machine..unless I misunderstood what I read |
| 05:31.42 | w32 | terminate fax calls to the pstn ? Explain ? |
| 05:31.58 | axscode | how to create a dialplan everything that starts with 2 and 4? |
| 05:32.01 | JT | maybe explain what you are trying to do |
| 05:32.07 | w32 | ok |
| 05:32.23 | JT | send and receive faxes from the traditional pstn, that can be done without too much trouble i'm led to believe |
| 05:32.37 | w32 | I'd simply like to receive & send |
| 05:32.39 | w32 | yea |
| 05:32.53 | JT | and you have a pstn connection for faxes? |
| 05:33.47 | w32 | I have termination and origination if that's what u mean |
| 05:33.51 | *** join/#asterisk pnlarsson (n=niklas@c83-248-0-248.bredband.comhem.se) |
| 05:34.13 | JT | via the PSTN? |
| 05:34.16 | JT | not voip |
| 05:35.08 | w32 | All I have is VOIP |
| 05:35.34 | w32 | I use viopstreet for termination and origination |
| 05:35.43 | JT | well you're screwed then, to put it nicely |
| 05:35.52 | JT | easiest to outsource the faxing to a fax provider |
| 05:36.13 | w32 | So it seems, any recommendations ? |
| 05:36.35 | w32 | cheap |
| 05:36.36 | JT | i just made one |
| 05:36.47 | w32 | made one ? |
| 05:36.49 | JT | eh, i'm probably not even in the same country as you |
| 05:36.54 | JT | find a fax provider |
| 05:36.57 | w32 | US |
| 05:37.04 | w32 | USA I meant |
| 05:37.04 | JT | or get a PSTN line |
| 05:37.09 | JT | yeah i'm in Australia |
| 05:37.23 | JT | you don't even have a single PSTN line at all?? |
| 05:37.40 | w32 | no, I have no conventional landline |
| 05:37.54 | w32 | the telco will not get a dime from me |
| 05:38.00 | JT | how does Internet come to the site then? |
| 05:38.09 | w32 | cable |
| 05:38.13 | JT | ok |
| 05:38.30 | JT | well that's really pretty perilous not to have a single PSTN line |
| 05:38.53 | orlock | JT: i dont have one at home :) |
| 05:38.55 | orlock | no dialtone, just a ULL |
| 05:39.05 | orlock | is it a phone line if it doesnt have dialtone? :) |
| 05:39.07 | w32 | yeah so I am finding out, but I do have two seperate connections |
| 05:39.20 | JT | voip providers/Internet aren't that reliable, but more importantly, in an emergency, a pstn line is the most reliable way to contact emergency services/others |
| 05:39.33 | JT | orlock: not really :) |
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| 05:57.35 | Dave|id | with the grandstream GXP-2000 phone, when you get a voicemail, does a notification come up on the phone at all? like a flashing light, a little icon on the display.... currently using trixbox and curious if this is the case with the phones |
| 06:02.05 | `Tingles` | what software SIP client would you recommend for winxp |
| 06:02.06 | `Tingles` | ? |
| 06:02.17 | w32 | xlite |
| 06:02.25 | Dave|id | i use that dosgy xen one |
| 06:02.31 | Dave|id | yeah |
| 06:02.33 | Dave|id | does the job |
| 06:02.44 | Dave|id | also works perfectly on my pocket pc |
| 06:03.07 | `Tingles` | hmm.. using xlite right now.. but looking for more.. |
| 06:03.14 | `Tingles` | i guess specifically a push to talk funtion |
| 06:03.17 | `Tingles` | function |
| 06:03.29 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 06:03.35 | Dave|id | spend $100 on a budgetone 102 phone |
| 06:04.03 | `Tingles` | i have 2 GXP-2000's however no headset.. and i want to use my wireless bluetooth headset with the software :) |
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| 06:06.06 | `Tingles` | specifically something with push to talk functionality |
| 06:16.18 | stephane_ | jour |
| 06:19.13 | *** join/#asterisk creativx (n=creadure@196.82-134-19.bkkb.no) |
| 06:21.32 | predder | `Tingles`, try expresstalk |
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| 06:41.22 | axscode | hi guyz... i got problem... when i try to call.... a long beep instead of ring... then the other end answer it... then we talk for about 5-10 seconds then it starts to ring again... any help? |
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| 06:58.31 | DarKnesS_WolF | iaxtel is cool |
| 07:01.27 | axscode | why u use voip to connect to irc? |
| 07:02.07 | *** join/#asterisk oej (n=oej@191.Red-88-9-110.dynamicIP.rima-tde.net) |
| 07:02.11 | JT | what the hell are you talking about axscode ? |
| 07:02.25 | axscode | nothing sir.. forgive me |
| 07:03.22 | *** join/#asterisk oej (n=oej@191.Red-88-9-110.dynamicIP.rima-tde.net) |
| 07:04.03 | kaldemar | someone with a little too much time could cook up an IVR irc client. |
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| 07:15.23 | FlatFoot | anybody there ? |
| 07:15.37 | Dave|id | no |
| 07:15.54 | FlatFoot | aha is there a prob with the latest trunk zaptel ? |
| 07:16.31 | Dave|id | i don't know |
| 07:16.33 | FlatFoot | get a lot of these |
| 07:16.35 | FlatFoot | pciradio.c:1381: error: dereferencing pointer to incomplete type |
| 07:16.54 | Qwell | Those should be warnings |
| 07:17.06 | benjk | do you need pciradio? |
| 07:17.13 | benjk | you can probably remove that |
| 07:17.16 | FlatFoot | not that im' aware |
| 07:17.38 | FlatFoot | just had to re install debian cos the box got screwed yesterday |
| 07:18.18 | benjk | I think this is for Jim Dixon's amateur radio telephony relay boards |
| 07:18.42 | benjk | so unless you are a radio ham and want to use those, you wouldn't need it |
| 07:18.46 | Qwell | sounds like you're setting -Wall or something silly |
| 07:19.11 | benjk | see if you can remove all the stuff related to rpt |
| 07:19.19 | benjk | not to be confused with rtp |
| 07:19.39 | FlatFoot | i shall investigate |
| 07:20.05 | Qwell | -Werror is what I was thinking |
| 07:20.25 | benjk | even then, its unlikely he will need pciradio |
| 07:20.37 | benjk | so if you don't need it you may as well remove it |
| 07:20.48 | Qwell | remove it from make menuselect |
| 07:20.51 | benjk | what's not there can't break |
| 07:20.57 | FlatFoot | sounds like i need another big hammer |
| 07:21.01 | FlatFoot | for my head |
| 07:21.22 | FlatFoot | make install |
| 07:21.47 | FlatFoot | oops wrong screen |
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| 07:37.54 | stoffell | hm, is there anything special as to why a polycom suddenly stops registering with asterisk? |
| 07:45.10 | creativx | probably the same reason my ip10s hangs after reboots |
| 07:45.19 | creativx | magic code |
| 07:47.11 | merlinn | stops registering? |
| 07:47.24 | merlinn | as in it flunks out a couple of times then ceases? |
| 07:47.36 | merlinn | or something else? |
| 07:47.56 | stoffell | merlinn: it keeps trying but gets an unauthorized request.. (happens on +10 phones) |
| 07:48.11 | stoffell | though user/pass hasn't changed and is correct |
| 07:48.49 | merlinn | ah, okay so it's not the new feature that allows a sip registration to cease and desist |
| 07:48.54 | merlinn | after N failures |
| 07:48.59 | *** join/#asterisk qdk (n=qdk@213.150.50.29) |
| 07:49.05 | merlinn | have you got any logs? |
| 07:49.09 | stoffell | hm merlinn .. is that documented somewhere ? :) |
| 07:49.15 | merlinn | yeah it is |
| 07:49.16 | stoffell | merlinn: yeah, logs of ... phone.cfg ? |
| 07:49.22 | merlinn | hang on, I only just saw it |
| 07:50.19 | merlinn | http://www.voip-forum.com/news.php?p=181 |
| 07:50.23 | merlinn | it's not that new |
| 07:50.49 | merlinn | I change versions less frequently than I change cars |
| 07:51.39 | stoffell | merlinn: okay, i will try registerattempts .. |
| 07:52.20 | stoffell | merlinn: that's only for connecting to sip providers it seems |
| 07:52.55 | merlinn | I'd imagine it's for any sip service, surely |
| 07:53.18 | merlinn | although I could be wrong |
| 07:53.29 | merlinn | I'm just reading the docs atm |
| 07:53.46 | stoffell | ok, i'm trying it.. |
| 07:56.05 | stoffell | hm, doesn't seem to help |
| 07:57.50 | stoffell | this is the sip debug log: http://pastebin.ca/182682 |
| 08:05.11 | stoffell | ouch, looks like these errors in this thread: http://threebit.net/mail-archive/asterisk-users/msg07122.html |
| 08:14.17 | CtRiX | nounoursfr, |
| 08:14.23 | CtRiX | you have no nat |
| 08:14.29 | CtRiX | that's not that case. |
| 08:14.34 | CtRiX | (or stoffell ) |
| 08:15.09 | stoffell | CtRiX: no nat, but teh "show channels" does stay like Rx: REGISTER |
| 08:19.51 | *** join/#asterisk Newbie___ (n=me@211.24.146.11) |
| 08:21.12 | Newbie___ | hi all, when a incoming call from POTs and is directed to extension 2000, how do i send to another extension 2002 when extension 2000 is busy? |
| 08:23.15 | creativx | ${dialstatus} |
| 08:24.17 | Newbie___ | creativx: thanks i will google around for instruction |
| 08:30.55 | *** join/#asterisk darkskiez (n=mbryars@195.11.205.216) |
| 08:32.37 | *** part/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 08:36.53 | *** join/#asterisk jeremy_g (n=jerms@static-213-115-44-90.sme.bredbandsbolaget.se) |
| 08:36.55 | jeremy_g | hi |
| 08:37.28 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 08:37.35 | jeremy_g | hello inspired |
| 08:37.51 | inspired | hi |
| 08:38.03 | jeremy_g | tjena |
| 08:39.06 | *** join/#asterisk oQPa (i=Spain@178.Red-83-40-182.dynamicIP.rima-tde.net) |
| 08:39.40 | jeremy_g | i have a question, when * dials out to a phone that has lost its IP connection, then what happens to the dial plan, will asterisk hangup or go to the next priority in that extension |
| 08:40.31 | jeremy_g | dial(sip/phone-that-is-out-of-reach) |
| 08:40.39 | inspired | it will continue |
| 08:41.15 | jeremy_g | thanks |
| 08:41.43 | inspired | you can choose what to do based on the DIALSTATUS variable |
| 08:42.15 | *** join/#asterisk jmls (n=asterisk@host81-159-167-80.range81-159.btcentralplus.com) |
| 08:42.19 | inspired | GotoIf(${DIALSTATUS} = CHANUNAVAIL?do_stuff:or_other_stuffer) |
| 08:42.31 | inspired | might not work. add a few " " and so on |
| 08:43.48 | creativx | or goto(s-${DIALSTATUS}) and make extensions for s-BUSY, s-CONGESTION etc |
| 08:44.19 | benjk | you'll aslo need an i extension then for fallback |
| 08:44.54 | *** join/#asterisk freebsd_fan (n=ebola@catagiuri305.giuri.unige.it) |
| 08:46.15 | darkskiez | the i exten doesnt work like that |
| 08:46.27 | creativx | i for invalid |
| 08:46.58 | *** join/#asterisk Sir_Diddymus (n=doe@pd95b08e3.dip0.t-ipconnect.de) |
| 08:47.04 | darkskiez | you'd need an _s-. exten for fallback |
| 08:47.30 | benjk | no i extension will do |
| 08:47.39 | benjk | anyhing that does not fit will hit i |
| 08:51.25 | creativx | yups |
| 08:51.30 | creativx | i is utter failover |
| 08:57.10 | E-bola | Morning |
| 08:57.20 | *** part/#asterisk Sir_Diddymus (n=doe@pd95b08e3.dip0.t-ipconnect.de) |
| 09:01.38 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) [NETSPLIT VICTIM] |
| 09:01.39 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) [NETSPLIT VICTIM] |
| 09:02.24 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 09:03.40 | stoffell | okay, my polycom issue is resolved, it seems i can't register to "dnsname" but only to "ipaddress", resolving is fine though.. (on polycoms and on * server!) |
| 09:06.01 | *** join/#asterisk ahewitt (n=ahewitt@203-59-103-45.dyn.iinet.net.au) |
| 09:09.05 | *** join/#asterisk thdei (n=thdei@d07v-213-44-66-130.d4.club-internet.fr) |
| 09:10.32 | jeremy_g | inspired:dialstatus works!!! thanks man |
| 09:10.32 | inspired | :) |
| 09:10.36 | jeremy_g | thanks creativx, the dial plan looks so simple and sleek now! |
| 09:11.11 | jeremy_g | :) |
| 09:12.49 | creativx | np |
| 09:12.54 | creativx | it will get messier, trust me. hehe |
| 09:13.06 | thdei | Hi everybody, I have a problem and maybe you have a idea for me. I want to connect a asterisk and a alcatel 4400 and I use this: http://www.voip-info.org/wiki/view/Alcatel+4400+via+PRI, but randomly, after few calls |
| 09:13.24 | thdei | Astersik go in congestion (34) |
| 09:13.32 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 09:13.40 | *** join/#asterisk Sir_Diddymus (n=doe@pd95b08e3.dip0.t-ipconnect.de) |
| 09:14.09 | thdei | But I think there is a link with the fact that the digium card (110) is always yellow |
| 09:14.25 | thdei | Do you have a idea for me ? |
| 09:14.31 | jeremy_g | what does this ./implies e.g. as in exten=_X./77777 ? what is this / and 7777 mean |
| 09:15.35 | *** join/#asterisk nfi|ermes (n=nfi_erme@217.220.121.62) |
| 09:15.47 | thdei | the /777777 say that you phone from the 777777 |
| 09:15.48 | creativx | number matching isnt it |
| 09:15.55 | thdei | to the _.X. |
| 09:18.16 | *** join/#asterisk ahewitt (n=ahewitt@203-59-103-45.dyn.iinet.net.au) |
| 09:18.21 | ahewitt | Hi all....I am *very* new to asterisk, so please excuse the ignorance. Could someone please tell me, if I am making a call to an external number via a service provider, is the from sip address supposed to be <phone_number>@asterisk_server or <extension>@asterisk_server?? |
| 09:18.28 | thdei | for example: exten = _0041X./1000, Dial(CAPI/...) allow the phone 1000 to call a number like 0041.... but not the other |
| 09:22.18 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 09:22.32 | thdei | ahewitt, can you make a example for me ? what is for you a extension ? |
| 09:22.50 | *** join/#asterisk ghenry (n=ghenry@82-69-192-46.dsl.in-addr.zen.co.uk) |
| 09:24.38 | jeremy_g | thdei: you phone from the 777777=> does that mean the callerid should be 77777 ? |
| 09:25.02 | jeremy_g | jester |
| 09:25.04 | jeremy_g | :p |
| 09:25.52 | stoffell | hm, how can i enable the status (xx ms) in ship show peers? |
| 09:26.22 | *** join/#asterisk speekac (n=alwin@60.51.217.61) |
| 09:26.49 | stoffell | okay, never mind, got it.. qualify=yes.. |
| 09:26.59 | ahewitt | I am simply trying to call an outbound number and I keep getting unauthorised....my extention is 3128 and I can see that the from sip address is 3128@10.1.1.1, but I was wondering if it is supposed to be 0134567889@10.1.1.1 |
| 09:27.22 | speekac | did you guys ever try to pull the voicemail data from asterisk ? |
| 09:27.37 | creativx | the wav files? |
| 09:27.49 | speekac | besides of wav? |
| 09:28.09 | speekac | is there any other relevant informations can be found in /var/spool/asterisk/voicemail ? |
| 09:28.19 | *** join/#asterisk grexk (n=grexk@124.107.72.45) |
| 09:32.17 | stoffell | speekac: try it out.. there's an info file for each voicemail (txt) |
| 09:37.13 | klem_ | stoffell: hi, how about w9962? i tried last weekend too and ended up with "tei lapd 1 assign req failed" in my dmesg |
| 09:37.33 | stoffell | klem_: hehe.. :) well, i'm not near the machine right now, but I also had error messages.. |
| 09:37.41 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 09:37.55 | stoffell | klem_: i couldn't dial out, it gave an error, it could well be the same! (i can check it in a few hours when near the machine) |
| 09:38.05 | klem_ | ok |
| 09:38.18 | stoffell | klem_: the hfc chipset worked well though.. |
| 09:38.54 | *** join/#asterisk thdei (n=thdei@d07v-213-44-66-130.d4.club-internet.fr) |
| 09:39.19 | thdei | jeremy_g: Yes, it's the caller id |
| 09:40.10 | klem_ | jep, i switched to hfc also |
| 09:42.27 | thdei | I ask again... maybe somebody come and have the solution: Hi everybody, |
| 09:42.27 | thdei | I have a problem and maybe you have a idea for me. I want to connect a asterisk and a alcatel 4400 and I use this: http://www.voip-info.org/wiki/view/Alcatel+4400+via+PRI, but randomly, after few calls Astersik go in congestion (34). |
| 09:42.28 | thdei | But I think there is a link with the fact that the digium card (110) is always yellow |
| 09:42.30 | thdei | Do you have a idea for me ? |
| 09:42.32 | DarKnesS_WolF | anyone using IAX or SIP java applet ? |
| 09:46.27 | *** join/#asterisk w32 (n=w32@c-71-193-124-77.hsd1.il.comcast.net) |
| 09:48.04 | speekac | stoffell: unfortunately, my voicemail are not working, no .wav or .txt files found eventhough i'd recorded the voicemail |
| 09:49.40 | thdei | speekac: Check the rights on the folder to be sure that asterisk user can write |
| 09:51.08 | *** join/#asterisk baconbuttie_uk (n=bob@host-84-9-143-249.bulldogdsl.com) |
| 09:55.50 | *** join/#asterisk linuxbangalore (n=karsansu@59.92.129.147) |
| 09:57.30 | linuxbangalore | Is there a channel for asterisk beginner's? as I have some very basic questions.. and want to discuss with someone to know more about asterisk.. |
| 10:00.17 | stoffell | linuxbangalore: it might be a good idea to read "the book" |
| 10:00.22 | stoffell | ~docs |
| 10:00.24 | jbot | hmm... docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 10:00.36 | stoffell | linuxbangalore: it's on asteriskdocs.org |
| 10:01.26 | speekac | files was written in ../context/extension/tmp |
| 10:01.58 | speekac | but after I end up the call, those files just disappeared |
| 10:02.37 | *** join/#asterisk phearless (n=phear@host81-138-68-106.in-addr.btopenworld.com) |
| 10:04.34 | *** join/#asterisk flightlinux (n=mustafa@202.5.145.13) |
| 10:04.49 | flightlinux | how can i check if cdr_mysql is loaded or not |
| 10:05.12 | grexk | load cdr_mysql.so |
| 10:05.29 | *** join/#asterisk Kuto (n=kuto@125.60.241.24) |
| 10:07.40 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
| 10:07.52 | *** join/#asterisk Weezey (n=weezey@206.210.109.233) |
| 10:08.23 | flightlinux | its cdr_addon_mysql.so in my system |
| 10:08.31 | flightlinux | i dont know how it was unloaded |
| 10:08.37 | flightlinux | do i always need to load it manually |
| 10:08.51 | flightlinux | how can i make it load automatically |
| 10:10.02 | dezent | flightlinux: /etc/asterisk/modules.conf |
| 10:10.08 | dezent | should be the right place |
| 10:10.34 | dezent | load => cdr_addon_mysql.so |
| 10:13.05 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 10:13.07 | *** join/#asterisk kristalino (i=kristali@gateway/tor/x-af61a381cf395e27) |
| 10:13.30 | flightlinux | thanks |
| 10:15.23 | flightlinux | if autoload = yes then i dont need to do load => cdr_addon_mysql.so |
| 10:15.47 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 10:15.54 | *** join/#asterisk olivier__ (n=olivier@obs92-4-82-239-116-113.fbx.proxad.net) |
| 10:17.57 | *** join/#asterisk vgster (n=vgster@170.252.64.1) |
| 10:18.13 | *** join/#asterisk razu (n=rasmus@tln-kontor.norby.ee) |
| 10:19.49 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
| 10:24.00 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 10:24.56 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) |
| 10:29.35 | *** join/#asterisk ThaZZa (n=me@229.9.233.220.exetel.com.au) |
| 10:29.55 | ThaZZa | Hey All |
| 10:31.23 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 10:38.24 | *** join/#asterisk qdk (n=qdk@213.150.50.29) |
| 10:39.22 | roo9 | anyone know why i am seeing this error: |
| 10:39.23 | roo9 | <PROTECTED> |
| 10:46.21 | ThaZZa | is anyone else having issues with Asterisk 1.4 and sip registration? It works fine on 1.2 |
| 10:56.56 | *** join/#asterisk razu (n=rasmus@tln-kontor.norby.ee) |
| 11:04.48 | backblue | roo9: because it's busy. |
| 11:06.10 | benjk | not necessarily |
| 11:06.26 | benjk | Asterisk always says busy, even when the cause is something else |
| 11:06.41 | benjk | you cannot really trust the console messages |
| 11:06.58 | benjk | only debugging will tell you the real cause, only then can you be sure that its actually busy |
| 11:09.17 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 11:10.45 | *** join/#asterisk ahewitt (n=ahewitt@203-166-225-64.dyn.iinet.net.au) |
| 11:12.15 | ThaZZa | is anyone else having issues with Asterisk 1.4 and sip registration? It works fine on 1.2 |
| 11:13.09 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 11:20.07 | RoyK | hm. there seems to be a rather nasty leak in 1.2.12.1 |
| 11:20.42 | RoyK | ThaZZa: do a sip debug and check if you can find it. post bug on bugs.digium.com if it really is a bug |
| 11:23.29 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 11:23.49 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 11:25.18 | Druken | benjk: you sound like a matrix fanatic up there..... |
| 11:25.45 | benjk | I am not sure I follow you |
| 11:25.57 | Druken | [07:06] <benjk> only debugging will tell you the real cause, only then can you be sure that its actually busy |
| 11:26.06 | Druken | :) |
| 11:26.40 | benjk | I don't quite understand why this is related to the matrix, but I take your word for it ;) |
| 11:27.27 | benjk | the point is that asterisk munches just about everything into congested or busy |
| 11:28.06 | Druken | never mind.... i was getting at the real cause, and be sure it's ACTUALLY busy... you know, morphous talk |
| 11:28.12 | Druken | he he, ha ha... |
| 11:29.26 | ThaZZa | RoyK: Can't see it even trying. |
| 11:30.26 | RoyK | with sip debug ip .... _ |
| 11:30.27 | RoyK | ? |
| 11:32.34 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 11:35.17 | ThaZZa | RoyK: Nope.. |
| 11:35.30 | *** join/#asterisk Cyt (n=danielcy@athedsl-162092.otenet.gr) |
| 11:36.37 | ThaZZa | RoyK: Think i might downgrade to 1.2. at least it works with the same config. |
| 11:36.38 | RoyK | ThaZZa: what does ethereal say? |
| 11:37.01 | RoyK | if it is a bug, file it on bugs.digium.com, please |
| 11:39.28 | merlinn | are there any good books on sip |
| 11:39.37 | merlinn | the RFC is pretty grim reading |
| 11:40.58 | ThaZZa | RoyK: It doesn't show any outbound traffic.. It is like it is not trying to reg |
| 11:41.30 | RoyK | ThaZZa: lol. asterisk 1.4pre-alpha-0.0.2 rocks! |
| 11:41.47 | ThaZZa | RoyK: I am running beta |
| 11:41.58 | RoyK | ThaZZa: yeah - "beta" |
| 11:42.16 | RoyK | ThaZZa: anyway - file a bug |
| 11:42.42 | ThaZZa | RoyK: Where can i get alpha |
| 11:43.34 | RoyK | what I meant was that the 1.4 beta is closer to a pre-alpha than a true beta |
| 11:43.41 | RoyK | s/than/than it is to/ |
| 11:45.06 | ThaZZa | Bugger |
| 11:46.23 | *** join/#asterisk ratz_001 (n=ratz_001@dsl-146-156-241.telkomadsl.co.za) |
| 11:46.28 | ratz_001 | hey all |
| 11:53.11 | *** join/#asterisk razu (n=rasmus@tln-kontor.norby.ee) |
| 11:55.45 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 11:55.50 | RoyK | coppice: evening |
| 11:56.06 | coppice | evening |
| 12:01.41 | *** join/#asterisk DoDaT69 (n=DoDaT69@internal.digitalson.com) |
| 12:02.04 | DoDaT69 | does anyone know if asterisk supports an integrated T1 that uses dynamic voice channel allocation? |
| 12:02.12 | DoDaT69 | and what hardware is compatible with it? |
| 12:02.44 | benjk | its called a PRI |
| 12:03.20 | benjk | Sangoma A10x or Digium Tx0x cards |
| 12:03.26 | DoDaT69 | I already have this integrated T1.. its actually frame relay b8zs |
| 12:03.27 | benjk | also OpenVox D110 |
| 12:03.54 | Druken | openvox? |
| 12:03.56 | DoDaT69 | digium told me yesterday they dont have anythign that would support dynamic voice channel allocation |
| 12:04.00 | benjk | what vendor? |
| 12:04.10 | DoDaT69 | FDN Communications |
| 12:04.15 | DoDaT69 | is who my provider is |
| 12:04.32 | DoDaT69 | I am trying to find somethign that will let me implement an asterisk pbx with the current line I have |
| 12:04.33 | benjk | not the provider, the vendor of your equipment, whatever you have |
| 12:04.40 | DoDaT69 | I dont have anything yet. |
| 12:04.52 | DoDaT69 | I am looking for something that will work with this type of setup |
| 12:04.55 | benjk | Druken, OpenVox sell the original Zapata card |
| 12:04.56 | *** join/#asterisk Cyt (n=danielcy@athedsl-162092.otenet.gr) |
| 12:05.04 | backblue | what it's dinamic alocation in a t1 line, i dont get it. |
| 12:05.07 | benjk | the single port one |
| 12:05.08 | DoDaT69 | I can always go analog, but would rather stay digital |
| 12:05.20 | DoDaT69 | its an integrated T1 package |
| 12:05.33 | benjk | then you're screwed |
| 12:05.35 | DoDaT69 | I ahve 6 lines that when in use, will take away the bandwidth form the data |
| 12:05.44 | DoDaT69 | its already voip |
| 12:05.56 | DoDaT69 | and the other thing is they use mgcp |
| 12:06.04 | Druken | dynamic data/voice switching |
| 12:06.05 | *** join/#asterisk GaryH (n=GaryH@host217-37-44-41.in-addr.btopenworld.com) |
| 12:06.06 | benjk | that's too exotic to be supported by open source |
| 12:06.15 | backblue | hoo, i'm seeying, here we normaly statically alocate the channels for data and for voice. |
| 12:06.19 | DoDaT69 | I figured |
| 12:06.20 | DoDaT69 | heh |
| 12:06.22 | RoyK | ThaZZa: you might consider using a true opensource pbx |
| 12:06.55 | DoDaT69 | yes, thats what digium sales told me yesterday, is I need to have something that is statically allocated |
| 12:07.03 | DoDaT69 | there has to be something out there somewhere tho |
| 12:07.10 | backblue | DoDaT69: well you dont, use a shapper |
| 12:07.16 | backblue | and pass the voice over ip. |
| 12:07.50 | DoDaT69 | I am new to voip, what is a shapper? something that will convert the proto? |
| 12:08.22 | backblue | DoDaT69: well use all your channels for data |
| 12:08.36 | backblue | and you have the max bandwitht you can have with the T1 link |
| 12:08.41 | DoDaT69 | right |
| 12:08.47 | backblue | and then put a linux shapper on each side |
| 12:08.58 | benjk | shaper? |
| 12:09.01 | backblue | and give more priority to voice traffic |
| 12:09.12 | backblue | instted of data |
| 12:09.26 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:09.27 | DoDaT69 | well yea |
| 12:09.33 | backblue | but this is done in l3 |
| 12:09.44 | backblue | what you were speaking, was l2. |
| 12:09.59 | DoDaT69 | but whats going on here is I have carrier --> smartjack --> adtran router |
| 12:10.11 | DoDaT69 | the adtran router splits the voice and data dynamically |
| 12:10.21 | backblue | that it's transparent |
| 12:10.22 | DoDaT69 | it has an amphernol connector that breaks out to a 66 block |
| 12:10.32 | backblue | for the router, it's all ip |
| 12:10.36 | DoDaT69 | right |
| 12:10.37 | fafnir | thats hot |
| 12:10.43 | backblue | so it will think it's allways data |
| 12:10.52 | backblue | and will use the max bandwith |
| 12:11.08 | backblue | indeed you need a good shapper on each side. |
| 12:11.30 | backblue | just make a iax trunk, for each side, and you have your dinamic channel done :P |
| 12:11.31 | DoDaT69 | thats already going on... my carrier network does that.. and I already have that setup on my foundry switch |
| 12:11.53 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 12:12.01 | DoDaT69 | what I want is something that will allow me to interface with my carrier's voip system and oust this adtran |
| 12:12.05 | *** join/#asterisk razu (n=rasmus@tln-kontor.norby.ee) |
| 12:12.18 | DoDaT69 | instead of breaking out to something that is analog, I want to stay all digital and virtualize my support team |
| 12:12.34 | backblue | do what i have sayed |
| 12:12.45 | backblue | use all channels for data, and put a iax trunk on top of it. |
| 12:13.25 | backblue | you can use a digium cards, with all channels for data. |
| 12:13.36 | DoDaT69 | are you saying to covert the iax to mgcp so I can talk to my carrier? |
| 12:14.13 | DoDaT69 | I think we are talking abotu 2 different things |
| 12:14.25 | DoDaT69 | I already have 6 phone lines that are native voip from my carrier |
| 12:14.55 | DoDaT69 | they are using the mgcp protocol. The adtran I have already converts from digital to analog, with the proper fxs and fxo functionality |
| 12:15.11 | DoDaT69 | I want to get rid of that adtran, and put an asterisk based system in its place |
| 12:15.52 | benjk | in that case all you need is a router for the T1 and chan_mgcp to let Asterisk hook into the MGCP based service of your provider |
| 12:16.15 | DoDaT69 | what is this chan_mgcp you speak of? |
| 12:17.06 | benjk | its a plugin for Astrerisk that makes Asterisk speak MGCP |
| 12:17.15 | DoDaT69 | really?!? suweet |
| 12:17.27 | DoDaT69 | yup thats what I need |
| 12:17.31 | benjk | pretty much everything is a plugin with Asterisk |
| 12:17.48 | benjk | without the plugins Asterisk doesn't really do anything |
| 12:17.52 | DoDaT69 | Okay, so when they told me asterisk doesnt support mgcp, thats just out of the box on a default system |
| 12:17.58 | DoDaT69 | right |
| 12:18.11 | benjk | there is one for SIP, one for H323, one for MGCP, one for voicemail etc etc etc etc |
| 12:18.20 | DoDaT69 | I figured there has to be a way to do this.. thats why I came here |
| 12:18.34 | benjk | some of the plugins are more frequently used than others |
| 12:18.39 | DoDaT69 | right |
| 12:18.49 | benjk | the ones that are more frequently used are better supported |
| 12:18.57 | benjk | MGCP happens to be one of the lesser used ones |
| 12:18.58 | DoDaT69 | makes sense |
| 12:19.03 | DoDaT69 | yea.. |
| 12:19.08 | benjk | so there are some things that it cant handle |
| 12:19.13 | DoDaT69 | oh? |
| 12:19.28 | DoDaT69 | so I suppose the next thing I would need on top of that is a csu/dsu card |
| 12:19.32 | DoDaT69 | and I should be in business |
| 12:19.37 | benjk | I think it can't be a gatekeeper or it can only be a gatekeeper or something like that |
| 12:20.05 | DoDaT69 | oh yea.. thats exactly what sales said, asterisk cannot function as an mgcp endpoint |
| 12:20.16 | DoDaT69 | well shit |
| 12:20.22 | DoDaT69 | :( |
| 12:21.36 | DoDaT69 | well hell.. I guess its analog conversion then.. |
| 12:21.45 | DoDaT69 | thanks for the input you all ;) |
| 12:21.52 | benjk | why not find a provider that uses a more sane protocol? |
| 12:22.03 | DoDaT69 | cause I am in a contract |
| 12:22.08 | benjk | even the folks who designed MGCP say that it was a mistake |
| 12:22.15 | DoDaT69 | I already found the correct protocol with another provider.. I will use that for my clients |
| 12:22.24 | DoDaT69 | I am stuck for another year and a half with this provider |
| 12:22.31 | DoDaT69 | Hahah |
| 12:22.48 | benjk | tell them that they are nuts to use this protocol and it can only hurt their business in the long run |
| 12:23.10 | DoDaT69 | Deltacom has a product called Simplici-T that will allow carrier --> smartjack --> adtran |
| 12:23.17 | benjk | tell them you are happy to help them set up an Asterisk box with IAX or SIP and be their laboratory rat |
| 12:23.17 | DoDaT69 | then from teh adtran you have ethernet out |
| 12:23.22 | DoDaT69 | and digital T1 out |
| 12:23.29 | DoDaT69 | with groundstart or loopstart signaling |
| 12:23.47 | benjk | that's pretty silly really |
| 12:23.56 | DoDaT69 | and its statically allocated, so you just plug that into the digium t1 card |
| 12:23.57 | benjk | cause you will be doing digital-analog-digital |
| 12:24.11 | DoDaT69 | it doesnt go analog, it keeps a digital signal |
| 12:24.15 | DoDaT69 | that was my main concern |
| 12:24.20 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 12:24.24 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 12:24.31 | benjk | not if it does groundstart or loopstart |
| 12:24.35 | benjk | those are analog protocols |
| 12:24.40 | DoDaT69 | really? |
| 12:24.55 | benjk | and the adtran is a channel bank as far as I recall |
| 12:25.06 | DoDaT69 | shows how much I know.. I still have a lot to learn in this part of techie |
| 12:25.06 | benjk | that is a box which has T1 in and analog out |
| 12:25.29 | benjk | usually 1xT1 in and 24 analog out |
| 12:25.44 | DoDaT69 | I will have to ask the engineer when I talk to him today.. I might be just going with damn analog cards then... |
| 12:26.16 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:26.54 | benjk | analog is not a good idea either |
| 12:26.59 | DoDaT69 | really? |
| 12:27.06 | DoDaT69 | I hear its got more echo |
| 12:27.07 | benjk | its hit and miss |
| 12:27.19 | benjk | and the quality is solala |
| 12:27.28 | DoDaT69 | Hmm |
| 12:27.37 | DoDaT69 | shit.. I gotta run.. I will bbiab |
| 12:27.42 | DoDaT69 | gotta take the kitty to the vet :-D |
| 12:27.47 | benjk | try to talk your provider into a standard setup |
| 12:27.57 | DoDaT69 | yea.. I will get on the phone with them again today |
| 12:28.07 | DoDaT69 | took me a month to find out what I did yesterday |
| 12:28.08 | benjk | it can only be a benefit to them |
| 12:28.16 | DoDaT69 | how my shit is delivered and all that mess |
| 12:28.21 | benjk | because in the long term they stand to lose customers |
| 12:28.24 | DoDaT69 | anywho, thanks man |
| 12:28.39 | benjk | welcome |
| 12:28.39 | DoDaT69 | I will be around in a few ;) |
| 12:28.39 | ahewitt | could someone please have a look at my config here....http://pastebin.ca/182854.....and see if you can find why its not working? I can see the handset has registered with the asterisk server, however I can't dial in or out |
| 12:28.49 | *** join/#asterisk GaryH (n=GaryH@host217-37-44-41.in-addr.btopenworld.com) |
| 12:38.30 | *** join/#asterisk Cyt (n=danielcy@athedsl-162092.otenet.gr) |
| 12:39.00 | *** join/#asterisk af_ (n=af@ip-171-49.sn1.eutelia.it) |
| 12:40.37 | RoyK | <PROTECTED> |
| 12:40.37 | RoyK | 10545609 bytes allocated 12881 units total |
| 12:41.31 | *** join/#asterisk Twister (n=chrissye@host166.sparenet.ncn.net) |
| 12:41.35 | RoyK | hm. there's a nasty leak in 1.2.12.1, but it doesn't show up in memdebug |
| 12:41.56 | *** join/#asterisk dusan2 (i=dusan@209-223-47-160-static.oplink.net) |
| 12:42.07 | Twister | is there anywhere i can get a list of * error codes? Im getting error 29 from iax when trying to register to another machine |
| 12:42.36 | benjk | hehe, you believe in Santa Claus eh? |
| 12:43.47 | benjk | you can read the iaxlib sources and create such a list though ;) |
| 12:44.34 | Twister | yes i do! because if someone took enough effort to get that fucked up to be able to make up a story about a fat man that comes down a skinny chimney with a small bag that gets bigger when you put stuff in it and gives out presents to kids that are nice |
| 12:44.49 | Twister | then im gonna take the time to believe it |
| 12:46.20 | Twister | thanks for the tip benjk |
| 12:46.29 | *** join/#asterisk jmacz (n=jmacz@190.24.96.103) |
| 12:47.13 | benjk | it would be nice if * devs used a feature in C called "enum", but for some reason they don't |
| 12:47.47 | benjk | so more often than not, you have to pull the possible return codes (including errors) out of your nose |
| 12:48.36 | *** join/#asterisk murf (n=steve_mu@216.166.159.235) |
| 12:49.16 | Twister | ever call microsoft and ask them what an error code means? |
| 12:49.38 | *** part/#asterisk GaryH (n=GaryH@host217-37-44-41.in-addr.btopenworld.com) |
| 12:49.39 | benjk | no, I don't use microsoft |
| 12:49.54 | Twister | unfortuinatly i have to |
| 12:50.11 | Twister | but my point is, they cant tell you, theres no meaning behind any of their errors |
| 12:50.38 | Twister | and for the record, im not comparing the devs to microsoft |
| 12:51.04 | coppice | other people have more obscure errors than anything MS can come up with. they can't even excel at that :-) |
| 12:51.16 | Twister | HAHHAHAHAHHAHAHHAHHAH! |
| 12:51.24 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
| 12:51.40 | coppice | "File no open, other than for open or close" is my favourite |
| 12:51.54 | benjk | I remember to have seen a humorous site that specialised ion Windows errors |
| 12:52.20 | benjk | the one I remember was a dialog box that said "Error: No keyboard. Press any key to continue" |
| 12:52.34 | RoyK | the usual POST error, you meen? |
| 12:53.00 | benjk | I used to work for a company that wanted to make "friendly software" and their CEO had his own ideas of error messages |
| 12:53.06 | Twister | HA ya they still have that one in certain dell systems |
| 12:53.18 | benjk | we had things like this ... |
| 12:53.29 | benjk | "Now there's three of us waiting ..." |
| 12:53.43 | benjk | "You are waiting for me, I am waiting for the printer and the printer is waiting for paper" |
| 12:53.47 | Twister | lol |
| 12:53.58 | benjk | people loved it |
| 12:54.15 | benjk | they tried to create errors just to see what message would pop up |
| 12:54.17 | RoyK | benjk: rotfl |
| 12:54.45 | Twister | my uncle puts in messages like :"dumbass! why did you do that? now the whole thing is messed up and you have to call someone to fix it wasting my time, your time, you bosses time and everyone in between!" |
| 12:55.35 | benjk | the original Mac had a thing that when you pressed a key that had no meaning in the current context, it would do a noise like "boioioioioiiing" |
| 12:55.39 | *** join/#asterisk luite (i=luite@belphegor.deadlysins.nl) |
| 12:56.00 | benjk | people got embarrased, because everybody in the office could here that you are a complete retard |
| 12:56.11 | benjk | s/here/hear |
| 12:56.16 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 12:56.27 | Makenshi | surely the fact that they're using a mac gives it away? |
| 12:56.34 | Makenshi | :p |
| 12:56.44 | *** join/#asterisk angryuser (n=aster@i03v-213-44-169-43.d4.club-internet.fr) |
| 12:56.45 | Makenshi | sorry, couldn't resist |
| 12:56.55 | luite | hello, I was wondering which version of asterisk I would need for a wildcard b410p (4x BRI), and whether it is compatible with european (specifically: dutch) ISDN networks? |
| 12:57.20 | benjk | well, this was in a company where the company decided what computers would be purchased and they chose Macs for everybody |
| 12:57.33 | benjk | mostly because other computers didn't have any networking |
| 12:57.33 | RoyK | luite: isdn is isdn. but do you need four ports? |
| 12:57.47 | benjk | none whatsoever |
| 12:57.47 | Makenshi | ahh well back then that was a sensible choice |
| 12:58.00 | Makenshi | and these days macs are pretty decent too |
| 12:58.07 | Makenshi | they just had a bit of a rough patch |
| 12:58.07 | benjk | today it is a sensible choice again |
| 12:58.16 | benjk | its BSD with a fancy GUI |
| 12:58.35 | Makenshi | yeah osx is allright, and you can run windows or linux if you want to |
| 12:59.05 | RoyK | does anyone else see a nasty memleak in 1.2.12.1? |
| 12:59.07 | luite | RoyK: I thought there were some differences, just wanted to make sure. I do not need 4 ports at the moment, but more than one, and I'm not sure if I want to have multiple hfc pci cards in one server |
| 12:59.45 | benjk | luite, the single port HFC cards are cheap enough to throw away when you need to upgrade |
| 12:59.47 | RoyK | luite: junghanns.net have some rather nice cards, but I don't know if they are any cheaper |
| 12:59.48 | benjk | less than 50 USD |
| 12:59.58 | tzafrir | luite, if you need 2 ports and don't want to pay much, consider 2 hfc cards and the florz patch... |
| 13:00.27 | RoyK | florz patch? |
| 13:00.27 | tzafrir | zaphfc.florz.dyndns.org |
| 13:00.29 | benjk | that's if you want to stop smoking RoyK |
| 13:00.47 | benjk | and helps against tooth decay too |
| 13:01.00 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.160) |
| 13:01.09 | *** join/#asterisk vgster (n=vgster@170.252.64.1) |
| 13:01.10 | luite | RoyK: yes their multi bri cards are a little cheaper, but I think they don't have hardware echo cancellation. and their bristuff drivers seem to be a little 'hackish |
| 13:01.34 | RoyK | luite: erm. does digium's BRI card have EC? |
| 13:01.38 | tzafrir | luite, why do you need EC with ISDN? |
| 13:02.11 | RoyK | tzafrir: because isdn creates echo? |
| 13:02.17 | luite | RoyK: yes the specs say it does |
| 13:02.21 | benjk | luite, they are Zaptel drivers, what else do you need to know |
| 13:02.22 | coppice | why shouldn't you need EC with ISDN? |
| 13:02.37 | RoyK | coppice: because isdn is magic |
| 13:02.43 | luite | but there hardly is any more information about the card than there is on digium.com, and that is not much |
| 13:03.08 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 13:03.11 | coppice | RoyK: rather becasue ISDN isn't universal |
| 13:03.11 | luite | it is not mentioned on the asterisk.org site yet, possibly because it's rather new |
| 13:03.21 | benjk | well the Junghanns stuff works well and has been around for ages long before anybody at Digium even know that Europe as using BRI |
| 13:03.38 | RoyK | tzafrir: is florz' stuff the new bristuff? |
| 13:03.39 | benjk | knew |
| 13:03.47 | benjk | no, its just a patch |
| 13:03.52 | RoyK | k |
| 13:04.12 | *** part/#asterisk hank (n=hank@netwichtig.de) |
| 13:04.19 | tzafrir | It's a patch to the zaphfc driver from bristuff |
| 13:04.28 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 13:04.50 | tzafrir | RoyK, ISDN is digital. echo is generated in the border of analog and digital |
| 13:05.02 | benjk | not only |
| 13:05.36 | RoyK | tzafrir: so sangoma and digium creating PRI boards with HWEC is just for fun? |
| 13:05.54 | coppice | tzafrir: and what relevance does that have to the need for EC? |
| 13:06.40 | tzafrir | I was wondering |
| 13:07.57 | benjk | anyway, if you want to bring in two BRI circuits, two cheap single port HFC cards and BRIstuff will be fine, then buy the quad card when you need more circuits |
| 13:08.35 | luite | benjk: well, they are zaptel, but they do break pri and modify a lot of stuff in asterisk. I was wondering if an 'original' digium card would be better, especially for support in the future |
| 13:09.17 | luite | but I guess I'll try with hfc-pci first then |
| 13:09.30 | RoyK | asterisk uptime: 2 hours. memory allocated by asterisk process: virt/res 746448/127544kB |
| 13:10.34 | coppice | tzafrir: what you said is not actually true, anyway. echo comes from acoustic connections in phones and from 2 to 4 wire hybrids. its nothing specifically to do with analogue/digital interconnect |
| 13:10.49 | benjk | they don't actually break PRI |
| 13:11.04 | benjk | you only have to make sure the drivers load in a particular order |
| 13:11.40 | luite | ah ok, I'll need to look into that then |
| 13:11.45 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 13:12.22 | benjk | and like I said, the single port HFC cards are so cheap, that it is good for starters |
| 13:12.30 | tzafrir | benjk, BTW: I noticed that at least on one of our systems qozap badly lcocks the system up on rmmod. I still have not had a chance to report this to kapejod, and that is 0.3q |
| 13:12.37 | benjk | and BRIstuff is the most straightforward way to get BRI working |
| 13:13.17 | benjk | tzafrir, I use those single port HFC cards (up to 2 in a single system) and it works totally flawlessly |
| 13:15.01 | CtRiX | <RoyK> asterisk uptime: 2 hours. memory allocated by asterisk process: virt/res 746448/127544kB |
| 13:15.10 | CtRiX | RoyK, this is a configuration issue |
| 13:16.06 | benjk | yeah, RoyK, redo your configs |
| 13:16.20 | benjk | dontleakmemory=yes |
| 13:16.54 | CtRiX | RoyK, do you use chan_sip ? |
| 13:17.07 | RoyK | http://bugs.digium.com/view.php?id=8032 |
| 13:17.11 | RoyK | CtRiX: yes |
| 13:19.04 | CtRiX | ok i have the solution |
| 13:19.10 | CtRiX | it's quite trivial. |
| 13:19.17 | RoyK | install openpbx? |
| 13:19.21 | CtRiX | open modules.conf and insert noload => chan_sip.so |
| 13:19.46 | CtRiX | if you don't use chan_isip, you'll ahve less leacks. |
| 13:19.47 | benjk | sounds like a good workaround |
| 13:20.04 | CtRiX | that's a mater of configuration indeed. |
| 13:20.09 | CtRiX | that's a matter of configuration indeed. |
| 13:20.48 | benjk | chan_sip needs to go back to the sipyard for refitting |
| 13:22.11 | RoyK | http://karlsbakk.net/memleak.png |
| 13:22.13 | benjk | then again, they could just use an existing stack and save us all the trouble |
| 13:22.29 | tzanger | RoyK: chan_iax2 has a nasty memleak |
| 13:22.43 | RoyK | i'm not using iax2 |
| 13:22.47 | CtRiX | RoyK, another workaround is restarting * every 5 mins. |
| 13:23.24 | CtRiX | * should be restarted daily to avoid configuration problems and maybe version 1.4 needs less time. |
| 13:23.44 | benjk | every hour on the hour :) |
| 13:23.59 | benjk | from the CNN news centre in Alabama |
| 13:24.20 | *** join/#asterisk ManxPower (n=ManxPwer@71-8-11-111.dhcp.leds.al.charter.com) |
| 13:24.47 | CtRiX | RoyK, it's your fault. you should configure * better. |
| 13:24.56 | RoyK | yes... |
| 13:25.21 | RoyK | that is - it's my fault. i'm using asterisk. and I upgraded to 1.2.12.1. the problem wasn't there in 1.2.10 |
| 13:25.39 | benjk | upgrade to 1.2.10 then |
| 13:25.55 | benjk | I think you downgraded from 1.2.10 to 1.2.11 |
| 13:25.57 | RoyK | in 1.2.10, app_queue keeps crashing |
| 13:26.04 | benjk | so you need to upgrade from 1.2.11 to 1.2.10 |
| 13:26.09 | RoyK | waygrading |
| 13:26.16 | benjk | side grading then |
| 13:27.47 | benjk | RoyK how about running one box with 1.2.10 for SIP and another with 1.2.11 for queues, then link them up via IAX2 :P |
| 13:30.32 | RoyK | rotfl |
| 13:31.47 | benjk | RoyK, you shouldn't be running beta software |
| 13:31.55 | benjk | or is this 1.2.11 official? |
| 13:32.10 | RoyK | 1.2.12.1 'official' yes |
| 13:32.17 | benjk | ah |
| 13:32.22 | benjk | take that comment back then |
| 13:32.54 | benjk | 12 even |
| 13:33.10 | benjk | maybe .13 will fix it |
| 13:33.25 | benjk | and only introduce a bug in some part you don't use |
| 13:33.44 | RoyK | :) |
| 13:33.54 | DarKnesS_WolF | RoyK: what is the problem ? |
| 13:34.03 | benjk | too little RAM |
| 13:34.08 | DarKnesS_WolF | i just upgraded to 1.2.12.1 yesterday |
| 13:34.09 | RoyK | DarKnesS_WolF: memleak from the midst of hell |
| 13:34.09 | benjk | to run 1.12 |
| 13:34.30 | DarKnesS_WolF | what is midst? |
| 13:34.34 | benjk | RoyK, you need to buy more RAM |
| 13:34.45 | RoyK | DarKnesS_WolF: center |
| 13:34.50 | benjk | buy 1G every two hours and it will be ok |
| 13:35.22 | *** join/#asterisk Op3r (n=Op3r@61.28.130.145) |
| 13:36.04 | DarKnesS_WolF | RoyK: i don't use it :-) |
| 13:36.24 | inspired | too bad his ram slots probably are full already ;) |
| 13:36.47 | benjk | take out the first bank then and fill in the new RAM |
| 13:36.49 | coppice | if he rams hard enough he'll get some more in |
| 13:36.56 | inspired | :D |
| 13:37.12 | inspired | good one |
| 13:37.16 | benjk | chance is one of the banks is totally leaked memory only |
| 13:37.16 | Druken | ram riser cards baby :) |
| 13:37.28 | benjk | so just take that out |
| 13:37.34 | RoyK | eh |
| 13:37.34 | Druken | dairy chain them, risers, inside risers :) |
| 13:37.41 | RoyK | from 1.2.12.1-patch |
| 13:37.41 | RoyK | -static int conf_run(struct ast_channel *chan, struct ast_conference *conf, int confflags) |
| 13:37.41 | RoyK | +static int conf_run(struct ast_channel *chan, struct ast_conference *conf, int confflags, char *optargs[]) |
| 13:37.48 | RoyK | changing APIs in stable? |
| 13:37.51 | Druken | s/dairy/daisy/ |
| 13:37.53 | inspired | Druken, doesn't work in the long run. he would have to get a bigger housing |
| 13:37.54 | inspired | ;) |
| 13:37.57 | RoyK | s/stable/release/ |
| 13:38.10 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 13:38.10 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:38.14 | *** join/#asterisk ajedwards (n=justacha@unaffiliated/ajedwards) |
| 13:38.18 | benjk | anthm can help |
| 13:38.19 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 13:38.53 | benjk | he can write you a nice little garbage collector for chan_sip |
| 13:39.05 | RoyK | benjk: what'll be left? |
| 13:39.25 | Druken | chan_garbage_man |
| 13:40.11 | benjk | 8731 bits, RoyK |
| 13:42.14 | *** join/#asterisk zouzou (n=test@mail.splendor.net) |
| 13:43.01 | *** join/#asterisk razu (n=rasmus@tln-kontor.norby.ee) |
| 13:43.17 | zouzou | i already have asterisk installed on red hat |
| 13:43.39 | zouzou | can i installed asterisk@home with it |
| 13:43.46 | zouzou | or it will overwrite it? |
| 13:46.10 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 13:50.24 | *** join/#asterisk Crescendo (n=martinda@adsl-144-167-184.rmo.bellsouth.net) |
| 13:50.54 | Crescendo | I just compiled and installed Asterisk 1.2 on a test box - what do I need to do now? |
| 13:54.52 | rob0 | Crescendo: Celebrate? Throw a party for your 200 closest friends. |
| 13:54.56 | [Airwolf] | Is it possible to have one SIP account and login with two client simultanously and when called both clients wil ring ? |
| 13:54.57 | zouzou | Crescendo:http://www.voip-info.org/wiki-Asterisk+quickstart |
| 13:55.07 | rob0 | ah, that too :) |
| 13:56.22 | [Airwolf] | Because if I login with the second client, that client gets all the calls. |
| 13:57.11 | trelane_ | [Airwolf], right, instead have a Dial(sip/exten&sip/exten) |
| 13:57.16 | trelane_ | and use two different accounts |
| 13:57.55 | *** join/#asterisk lorinc (n=ang@caracas-0685.adsl.interware.hu) |
| 13:59.27 | [Airwolf] | trelane, that is a better solution indeed. |
| 14:00.00 | *** join/#asterisk pbx1 (n=pbx1@124.106.141.64) |
| 14:00.41 | [Airwolf] | I just wanted to have one account for both. :) |
| 14:02.15 | Crescendo | Thanks, guys. |
| 14:02.16 | Crescendo | :) |
| 14:04.51 | dorel__ | sheesh |
| 14:04.57 | dorel__ | zaptel stopped working for some reason |
| 14:05.14 | benjk | kebab |
| 14:05.19 | dorel__ | its a tdm400p 4 slots fxo card and on zttool it shows as UNCONFIGURED |
| 14:06.42 | *** join/#asterisk viler (i=1000@ip-70-228.telesat.com.co) |
| 14:07.10 | *** join/#asterisk dsfr (i=dsfr@pdpc/sponsor/digium/dsfr) |
| 14:07.31 | dorel__ | how do i get it configured? |
| 14:08.48 | RoyK | http://cheekys.net/funnypics/pics/illiterate.jpg |
| 14:09.14 | DoDaT69 | http://www.digitalson.com/pictures/stripper.jpg |
| 14:10.09 | *** join/#asterisk Ahrimanes (n=michael@81.7.159.2) |
| 14:10.28 | Ahrimanes | mornign |
| 14:12.28 | ManxPower | dorel__, you set up /etc/zapata.conf |
| 14:13.12 | ManxPower | I recommend that you use the MAC of the device as it's SIP Account ID |
| 14:14.07 | Crescendo | What's the best client to test with? |
| 14:14.22 | RoyK | ManxPower: why? |
| 14:14.38 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:14.51 | ManxPower | RoyK, It forces us to remember that DEVICE IS NOT EXTENSION |
| 14:15.03 | DoDaT69 | so is digium the "best" hardware to go with? |
| 14:15.06 | ManxPower | It is also an easy to have all unique ids |
| 14:15.22 | CtRiX | DoDaT69 if you want Echo, yes |
| 14:15.26 | DoDaT69 | Hahah |
| 14:15.32 | DoDaT69 | so whats the census |
| 14:15.34 | dorel__ | ManxPower: I setup /etc/zaptel.conf and /etc/asterisk/zapata.conf |
| 14:15.48 | tzafrir | dorel__, "stopped working" as in? what does happen? |
| 14:15.51 | jbalcomb | DoDaT69 I have Digium cards and I feel fine |
| 14:15.51 | DoDaT69 | I am looking at setting up 3-4 of these systems in the next frew months, I want to make sure I get the right stuff |
| 14:15.55 | ManxPower | dorel__, you obviously didn't set it up right. What does ztcfg -vvv say? put the output on pastebin |
| 14:16.07 | DoDaT69 | jbalcomb --> are you analog or all digital? |
| 14:16.09 | jbalcomb | DoDaT69 I have heard the Sangoma cards are better |
| 14:16.17 | ManxPower | DoDaT69, I would use Digium or Sangoma card with external tellabs echo cancel and T-1/PRI |
| 14:16.19 | jbalcomb | DoDaT69: digital |
| 14:16.34 | ManxPower | DoDaT69, anyone that has a choice is all digital |
| 14:16.35 | DoDaT69 | I already have the service.. I have to go analog due to the nature of it |
| 14:16.36 | acrg | is it possible to perform a nested CUT ? |
| 14:16.37 | DoDaT69 | yea |
| 14:16.48 | acrg | ie. a cut inside another cut |
| 14:17.07 | DoDaT69 | my provider is using mgcp :( so I cant tie directly into their system since asterisk is not an endpoint for mgcpo |
| 14:17.11 | ManxPower | acrg, if cut is an app, no. if cut is a function, yes (I think it became a function in 1.2) |
| 14:17.21 | acrg | it's a function, yes |
| 14:17.36 | ManxPower | acrg, at least you SHOULD be able to if it's a function |
| 14:18.08 | acrg | ${CUT(CUT(var,delim,field),delim,field)} - would that syntax be correct ? |
| 14:18.12 | CtRiX | ah DoDaT69 |
| 14:18.24 | CtRiX | forget to use digium on HP/IBM/Dell systems |
| 14:18.32 | dorel__ | ManxPower: thanks, with that im seeing that im using already configured channels... kinda weird |
| 14:18.55 | *** join/#asterisk anthonyl (i=anthony@nat/digium/x-5f42ceaeff47a2ef) |
| 14:19.04 | merlinn | CtRiX: why is that? |
| 14:19.32 | dorel__ | ManxPower: I have a pri card which takes channels 1-15,16, and 17-31 and another 4 ports slot tdm400p fxo card... which channels should i assign to the fxo card then? |
| 14:19.40 | ManxPower | acrg, Maybe ${CUT(${CUT(var,delim,field)},delim,field)} |
| 14:19.52 | CtRiX | http://www.digium.com/en/docs/misc/compatibility_notes.php |
| 14:20.04 | ManxPower | dorel__, that would depend on what driver loads first. |
| 14:20.21 | ManxPower | assuming the PRI loads first, the FXO would be 32-35 |
| 14:20.30 | *** join/#asterisk Ox0F0-0FF (n=pierre@200.216.238.226) |
| 14:20.34 | dorel__ | ManxPower: ahh i see. thanks im checking it out |
| 14:20.42 | CtRiX | merlinn, got the link ? |
| 14:20.57 | ThaZZa | RoyK: You still here? |
| 14:20.57 | ManxPower | I ALWAYS load the T-1 stuff first, since my goal is to get rid of all the analog cards 8-) |
| 14:21.19 | merlinn | CtRiX: yes, thanks |
| 14:21.36 | CtRiX | merlinn, i would buy something compatible to all chipsets/machines |
| 14:21.48 | *** join/#asterisk a1fa (n=a1fa@unaffiliated/a1fa) |
| 14:21.50 | a1fa | hello |
| 14:21.57 | CtRiX | i have always feared "worksforme(tm)" sentences.... |
| 14:22.01 | a1fa | anybody else having problems with BroadVoice |
| 14:22.14 | DoDaT69 | these sangoma cards look pretty fly |
| 14:22.15 | a1fa | "doesnotworkforme(c) |
| 14:22.25 | CtRiX | and what has problems with some chipsets or cards may have other problems as weel with difeerent hardware not listed. |
| 14:22.34 | dorel__ | ManxPower: i tried setting the channels to any of the 32-35 possible options but im still stuck with asterisk not starting up, WARNING[11447] loader.c: Loading module chan_zap.so failed! |
| 14:23.07 | ManxPower | dorel__, I said put the output of ztcfg -vvv on pastebin.ca |
| 14:23.31 | *** join/#asterisk droops (n=droops@adsl-074-245-001-031.sip.jan.bellsouth.net) |
| 14:23.40 | a1fa | hehe |
| 14:23.50 | a1fa | anybody having problems with broadvoice |
| 14:24.24 | *** join/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net) |
| 14:24.31 | *** join/#asterisk fourcheeze (n=rich@office.callmaster.co.uk) |
| 14:25.35 | a1fa | any one in here use bv? |
| 14:25.38 | RoyK | ThaZZa: yes |
| 14:25.42 | fourcheeze | anyone got a nokia e61 to work with *? I've been following some instructions here: http://www.newlc.com/Using-SIP-with-Nokia-Series60-and.html |
| 14:25.49 | fourcheeze | registered once |
| 14:25.58 | *** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net) |
| 14:26.01 | fourcheeze | tried to call out but it didn't and then went back to "registration failed" |
| 14:26.08 | RoyK | asterisk uptime: 3:10, memory usage: 1.1GB |
| 14:26.13 | RoyK | still climbing |
| 14:26.27 | Juggie | which version? |
| 14:26.31 | CtRiX | RoyK, |
| 14:26.32 | RoyK | 1.2.12.1 |
| 14:26.38 | dorel__ | ManxPower: there we go, http://pastebin.ca/182956 |
| 14:26.49 | Juggie | RoyK, core dump it |
| 14:26.56 | CtRiX | you have to restart * every hour or so if you cannot find your configuration problems ! :-) |
| 14:27.15 | RoyK | Juggie: perhaps a good idea..... |
| 14:27.23 | bbz_ | anyone know if any of the /dev/dsp or audio can playback audio on a single mono left or right channel? |
| 14:27.32 | Juggie | RoyK, in scripts theres a script to do it. |
| 14:27.34 | RoyK | Juggie: but then, how do I check where the leak is? |
| 14:27.48 | CtRiX | RoyK, it will coredump on it's own ... |
| 14:27.53 | Juggie | RoyK, thats beyond my expertize... |
| 14:27.55 | ManxPower | dorel__, I suspect that the modules in your analog card do not start at 1 or 0 |
| 14:28.08 | RoyK | Juggie: then it won't really help to coredump it..... |
| 14:28.18 | ManxPower | dorel__, pastebin the dmesg output for the card |
| 14:28.24 | Juggie | are you compiled w/ debug mode? |
| 14:28.50 | Juggie | heh |
| 14:29.01 | Juggie | you need to recompile * in debug mode, and try again :) |
| 14:29.11 | RoyK | no bloody well do not |
| 14:29.14 | dorel__ | ManxPower: im grepping -i zap and tdm, anything else important to find there? |
| 14:29.33 | Juggie | ? |
| 14:29.34 | *** part/#asterisk Ahrimanes (n=michael@81.7.159.2) |
| 14:29.35 | RoyK | I've been debugging this shite for years, and I don't need anymore recompiling |
| 14:30.02 | ManxPower | dorel__, no idea. |
| 14:30.23 | Juggie | well, recompiling it in debug is the only way someone will be able to help you track down your leak. |
| 14:30.36 | *** join/#asterisk }btorch{ (n=kvirc@adelphi.geofocus.com) |
| 14:30.48 | CtRiX | the other way is looking at a diff between 1.2.10 and 1.2.11 |
| 14:30.53 | CtRiX | the other way is looking at a diff between 1.2.10 and 1.2.12.1 |
| 14:30.58 | CtRiX | the leack is there |
| 14:31.00 | *** join/#asterisk Sasch (n=Admin@host102-30-static.107-82-b.business.telecomitalia.it) |
| 14:31.06 | bbz_ | are there any parameters that you can pass to /dev/dsp ? |
| 14:31.12 | Juggie | you can also try using 1.2 trunk |
| 14:31.16 | Juggie | as it may already be fixed |
| 14:31.32 | Sasch | where i can download Spandsp for asterisk ?? |
| 14:31.47 | RoyK | CtRiX: i know. i just don't want to restart this box all the time, being in production |
| 14:32.39 | dorel__ | ManxPower: ok, check the comments on that original post. |
| 14:32.45 | Juggie | RoyK, upgrade the box to 1.2 trunk, see if the leak still happens.. what is the load on the box? |
| 14:34.02 | dorel__ | ManxPower: nm, i think its lost somewhere i cant even find it, ill post it on a new thread |
| 14:34.26 | dorel__ | ManxPower: http://pastebin.ca/182961 |
| 14:35.43 | ManxPower | dorel__, did you give me your /etc/zapata.conf on pastebin? |
| 14:35.58 | dorel__ | ManxPower: yeah, its the post before that |
| 14:36.22 | dorel__ | ManxPower: : http://pastebin.ca/182956 |
| 14:36.23 | RoyK | juanjoc: the load is average, and it worked flawlessly on 1.2.10 |
| 14:36.25 | tzanger | ok that's fucked |
| 14:36.31 | tzanger | sip user will not accept calls |
| 14:36.37 | tzanger | change nothing but type=peer and it works |
| 14:36.41 | tzanger | wtf |
| 14:36.56 | tzanger | users call the PBX, peers get called by the PBX, that's been the rule |
| 14:37.31 | *** join/#asterisk shodan- (n=shodan@ip078.99-113-216.pppoe4.joliette.intermonde.net) |
| 14:37.33 | *** join/#asterisk ambriento (n=melcon@200-158-14-51.dsl.telesp.net.br) |
| 14:38.14 | Juggie | as i've said 5 times |
| 14:38.22 | Juggie | upgrade to 1.2 trunk, and see if you can reproduce your problem |
| 14:38.58 | Juggie | then you can procede from there to figuring out this problem, optionally you can also just use the last working version and forget about it. |
| 14:39.01 | ThaZZa | RoyK: Sorry.. Yeah i fixed the problem.. Running beta again.. I feel so dumb now. |
| 14:39.08 | shodan- | any linksys spa-3*02 users here ? I'd like to know what are the common problems with the fxo line on this thing if any (things like echo,incorrect amplitude,erroneous hangups etc..) |
| 14:39.59 | Sasch | but if i want to receive fax with my tdm400p and asterisk i install Spandsp or another program ?? |
| 14:40.39 | RoyK | Juggie: can't use 1.2.10, since app_queue crashes there every now and then |
| 14:41.39 | tzafrir | dorel__, chan_zap will fail to load if you define a channel in zapata.conf which doesn't exist or is of the worng type |
| 14:42.31 | tzafrir | In zapata.conf you defines channels 32-35 to be FXO channels. But in zaptel.conf there are only two of them |
| 14:42.32 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 14:43.17 | Juggie | RoyK, then upgrade to 1.2 trunk. |
| 14:47.29 | dorel__ | tzafrir: maybe in the posts they dont match but i tried defining in both 32-35 and 33-34 though it still fails |
| 14:47.49 | RoyK | Juggie: you need to try to understand that this stuff is in production and can't be rebooted all the time..... |
| 14:48.05 | *** join/#asterisk amdtech (n=amd011@ss-5-100.shsu.edu) |
| 14:48.15 | dorel__ | tzafrir: in this post http://pastebin.ca/182961 you can see that the 2 channels 33 and 34 are configured both in zapata.conf and zaptel.conf |
| 14:48.55 | Juggie | RoyK, you dont have to reboot your server to change the * version |
| 14:49.18 | RoyK | I need to reboot asterisk |
| 14:49.20 | Juggie | just, make; make install and then @ your asterisk console 'restart when convenient' |
| 14:49.42 | RoyK | convenient is like 03:00 or never |
| 14:50.01 | Juggie | well, your option is to continue to not have a solution |
| 14:50.04 | RoyK | and before that, asterisk has prolly crashed due to no memory left |
| 14:50.13 | Juggie | or, do something, your pick. |
| 14:50.18 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 14:50.22 | tzafrir | ztcfg has no problem configuring a non-existing module to whatever configu you'd like |
| 14:50.24 | RoyK | no... I'm just trying to find a way to find memory leaks from a core file |
| 14:50.51 | Juggie | i'm hardly a gdb expert, and since your * isnt compiled in debug it will likely be impossible to tell |
| 14:51.31 | dorel__ | tzafrir: like i said, i configured 33-34 on both files |
| 14:51.44 | RoyK | it's compiled with bloody well all the debug flags I could find. having used asterisk since 0.6 or something teaches you to use them since you will eventually need them anyway |
| 14:51.52 | tzafrir | and asterisk fails to load? what's the error message? |
| 14:54.03 | *** join/#asterisk linuxbangalore (n=karsansu@59.92.137.210) |
| 14:55.23 | *** join/#asterisk blaylock (n=seth@snap.helixsystems.com) |
| 14:55.33 | Juggie | RoyK, then, you can wait a couple of hours till there some more developers here and see what they can do |
| 14:55.55 | Juggie | since * hasnt cored yet you can either force it to dump (which will disrupt service) or wait |
| 14:56.12 | stoffell | klem_: i also get an error with w6692pci, being: MDL_ERROR|REQ (tei_l2) (and calls don't work) |
| 14:56.13 | RoyK | it coredumped earlier due to lack of memory |
| 14:56.17 | RoyK | 2.2GB core file |
| 14:56.21 | CtRiX | ROTFL! |
| 14:56.31 | CtRiX | RoyK, update your HD ! |
| 14:56.34 | benjk | you need to install more RAM |
| 14:56.39 | RoyK | yes..... |
| 14:56.48 | RoyK | 1TB to make it work until next upgrade |
| 14:56.52 | CtRiX | that's a marketing decision |
| 14:56.57 | RoyK | rotfl |
| 14:57.05 | CtRiX | M$ vista will need more HD and more Ram |
| 14:57.17 | merlinn | is the asterisk sip filter supposed to work this way? |
| 14:57.18 | CtRiX | that way, 1.4 will be called Asterisk Vista |
| 14:57.27 | CtRiX | to show that it needs more ram && HD |
| 15:00.07 | bbz_ | are there any parameters that you can pass to /dev/dsp, to tell the audio signal which channel to be played over? R/L mono? |
| 15:02.49 | *** join/#asterisk SplasPood (n=jwb@nat-loopback.lga4.us.corp.voxel.net) |
| 15:06.05 | *** join/#asterisk mogorman (i=mogorman@nat/digium/x-0868c293c6bc6db9) |
| 15:06.05 | *** mode/#asterisk [+o mogorman] by ChanServ |
| 15:08.05 | Sir_Diddymus | RoyK: may i ask how many users you've got on your box? Alot of traffic on it? always interested in experiences with big installations... |
| 15:08.10 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:08.10 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:08.29 | Sir_Diddymus | (or better said: "real" installations. Only using it in private, with two user) |
| 15:10.42 | ManxPower | merlinn, I didn't know Asterisk had a SIP filter |
| 15:11.35 | *** join/#asterisk vooduhal (n=vooduhal@tc-proxy2.catt.com) |
| 15:12.15 | blaylock | has anyone ever seen this message come up in the console? !! Got reject for frame 70, retransmitting frame 70 now, updating n_r!! or something like it |
| 15:12.19 | vooduhal | Does anyone know of a simple php/perl/cgi script to show information from the CLI. Like a simple page that just dumps 'show queues' etc to a page? |
| 15:12.44 | blaylock | vooduhal, use astman or gastman |
| 15:13.38 | vooduhal | blaylock, needs to be web based. |
| 15:14.14 | RoyK | Sir_Diddymus: a couple of thousand on this one |
| 15:15.11 | blaylock | vooduhal, ahh ok |
| 15:15.40 | blaylock | vooduhal, then no sorry i dont |
| 15:17.22 | Sir_Diddymus | RoyK: oy!! ok... :) |
| 15:20.02 | Crescendo | Sep 26 11:14:56 NOTICE[5323]: chan_iax2.c:5138 register_verify: No registration for peer 'diax' (from 192.168.0.87) ??? I'm assuming this is my botched version of a setup. What did I do wrong, how do I fix it? |
| 15:20.31 | bbz_ | would the 'nobody' user have permissions to use /dev/dsp? |
| 15:20.58 | ManxPower | bbz_, One would hope not. |
| 15:21.17 | bbz_ | manxpower: is there a good way i could give it permission to? |
| 15:21.52 | ManxPower | bbz_, that would depend on your distro. |
| 15:23.14 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:23.19 | merlinn | ManxPower: the sip parser |
| 15:23.24 | merlinn | it seems to suck slightly |
| 15:23.46 | merlinn | but I might be wrong, my clue isn't magnificent |
| 15:23.57 | ManxPower | merlinn, Asterisk is not a SIP Proxy |
| 15:24.11 | bbz_ | ManxPower: not sure -- its whatever fonality uses to load machines with =/ |
| 15:24.49 | merlinn | no, it shouldn't be |
| 15:27.04 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:27.04 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:27.38 | *** join/#asterisk Jamez^7 (n=martini@modemcable131.214-131-66.mc.videotron.ca) |
| 15:28.36 | merlinn | but it should have an RFC compliant syntax, right? |
| 15:28.50 | De_Mon | if I want to send an email via dialplan, I should use the SYSTEM command, right? |
| 15:29.37 | benjk | its not SIP RFC compliant anyway, no UTF-8 |
| 15:30.45 | merlinn | :( |
| 15:33.47 | *** join/#asterisk svenna_ (n=svenna@p548D12BD.dip0.t-ipconnect.de) |
| 15:33.50 | ManxPower | De_Mon, correct |
| 15:34.37 | ManxPower | merlinn, Yes, the call setup/teardown/control is RFC compliant enough to work ith most SIP devices |
| 15:34.52 | merlinn | but when trunkiing |
| 15:35.00 | merlinn | to non upstream carriers |
| 15:35.05 | ManxPower | merlinn, As far as I know there is no standard for SIP trunking. |
| 15:35.12 | merlinn | well no |
| 15:35.16 | merlinn | but SIP messages have a standard |
| 15:35.26 | ManxPower | merlinn, a standard for what? |
| 15:35.39 | merlinn | syntax |
| 15:35.42 | *** join/#asterisk ichilton (n=ian@gatekeeper.ichilton.net) |
| 15:35.47 | ManxPower | And BTW, it would be RTP trunking that would save you bandwidth |
| 15:35.56 | merlinn | I'm not looking to save bandwidth |
| 15:35.57 | ichilton | hi |
| 15:36.02 | merlinn | just communicate |
| 15:36.05 | ichilton | anyone using a cisco 79xx with asterisk? |
| 15:36.09 | ManxPower | merlinn, What ARE you looking for? |
| 15:36.17 | ManxPower | merlinn, Asterisk communicates with most SIP devices. |
| 15:36.25 | merlinn | I just acquired a VOIP company |
| 15:36.36 | merlinn | who hvae a fairly extensive asterisk roll out |
| 15:36.43 | ManxPower | merlinn, Asterisk communicates with most VOIP companies using SIP as well. |
| 15:36.50 | merlinn | I can see that |
| 15:36.52 | Qwell | ManxPower: SIP trunking is what freepbx folks call "sip" |
| 15:36.57 | Qwell | ie; to a provider |
| 15:37.11 | ManxPower | In fact, I cannot think of any SIP complient product that Asterisk will NOT work with. |
| 15:37.14 | *** join/#asterisk jero (n=jerou@savoirfairelinux.net) |
| 15:37.20 | ManxPower | Qwell, which is why "trunk" should be a banned term. |
| 15:37.23 | merlinn | urm |
| 15:37.26 | jero | hehe |
| 15:37.27 | Qwell | ManxPower: agreed |
| 15:37.27 | merlinn | nortel devices |
| 15:37.31 | merlinn | it seems |
| 15:37.40 | merlinn | I'm being held back considerably by the deployment in place |
| 15:37.43 | ManxPower | merlinn, You'll need to be more specific. |
| 15:37.51 | merlinn | and it seems that the fault lies with the SIP communication |
| 15:37.51 | Nugget | http://slacker.com/photos/misc/pophell <-- extensive asterisk roll out :) |
| 15:37.55 | ManxPower | Nortel has many SIP devices. |
| 15:37.58 | merlinn | there are a number of suppliers that just don't know what I'm talking about |
| 15:38.04 | merlinn | when we try to interconnect |
| 15:38.17 | merlinn | so I'm stuck using IAX trunsk to the small number of upstreams that support htem |
| 15:38.27 | ManxPower | merlinn, we can't even get Nortel BCM to talk to EACH OTHER with SIP. Nortel told us that those issues will be fixed in the next software upgrade to the BCM |
| 15:38.29 | merlinn | or SIP to the suppliers that understand the implementation |
| 15:38.59 | ManxPower | merlinn, What SPECIFIC PRODUCTS do you have issues with? |
| 15:39.09 | merlinn | urm |
| 15:39.17 | ManxPower | Nortel makes a zillion "SIP complient" devices. |
| 15:39.21 | merlinn | integr8 soft switches |
| 15:39.28 | Crescendo | Sep 26 11:14:56 NOTICE[5323]: chan_iax2.c:5138 register_verify: No registration for peer 'diax' (from 192.168.0.87) ??? I'm assuming this is my botched version of a setup. What did I do wrong, how do I fix it? |
| 15:39.30 | Qwell | Nugget: I was having trouble connecting to your site the other day (and again today)... It just kinda sits there, acting stupid, loading |
| 15:39.44 | Qwell | or, rather, not loading |
| 15:39.45 | merlinn | nortel multipath soft switches |
| 15:39.53 | *** join/#asterisk ichilton (n=ian@gatekeeper.ichilton.net) |
| 15:40.05 | merlinn | that's the only ones |
| 15:40.10 | Nugget | Qwell: that's a common problem for some linux kernel versions that have a crippling bug relating to mtu discovery. |
| 15:40.12 | merlinn | but it seems to be like 75% of the carriers in the UK |
| 15:40.17 | Qwell | eh? |
| 15:40.27 | De_Mon | should I be using the system cmd to send email via asterisk or something else? |
| 15:40.31 | Nugget | those versions of linux can't pass traffic through a firewall that does scrubbing |
| 15:40.36 | ManxPower | De_Mon, USE THE SYSTEM COMMAND |
| 15:40.37 | Qwell | nice |
| 15:40.42 | Nugget | using linux 2.6.8 anywhere in between you and me? |
| 15:40.45 | Qwell | Nugget: something as new as 2.6.17 has that? |
| 15:40.47 | carrar | hahahh wouldn't it be cool is that was a valid error message "I'm assuming this is my botched version |
| 15:40.48 | carrar | <PROTECTED> |
| 15:40.59 | De_Mon | ManxPower sorry, I looked away right after I asked the first time ;) |
| 15:41.01 | Qwell | nope |
| 15:41.07 | Nugget | Qwell: I know that 2.6.8.x had it, and 2.6.9 fixed it, but I think maybe it came back again. |
| 15:41.09 | Qwell | .17 and .13 |
| 15:41.10 | Nugget | I'm not sure. |
| 15:41.17 | ManxPower | Nugget, I assumed that MTU discovery problem was caused by people that don't know what they are doing blocking all ICMP |
| 15:41.39 | Nugget | ManxPower: this is different. Linux emits fragmented packets with the "no fragment" flag set. |
| 15:42.01 | Nugget | and many firewalls toss those packets because they're suspicious |
| 15:42.05 | ManxPower | merlinn, Nortel has a TERRIBLE history of implementing standards. Asterisk is better, but not perfect. |
| 15:42.07 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 15:42.18 | ManxPower | Nugget, Well THAT sure is a smart thing to do. |
| 15:42.19 | Qwell | Nugget: fix your firewall :P |
| 15:42.21 | *** join/#asterisk toxap (n=toxap@213.227.193.75) |
| 15:42.22 | Juggie | we have a full nortel rack in our lab of voip stuff. |
| 15:42.24 | Nugget | my firewall is fine. |
| 15:42.28 | Juggie | we turn them on so ppl think they are doing something |
| 15:42.33 | Juggie | but they do absolutely nothing. |
| 15:42.39 | Juggie | ppl like to see the blue nortel light |
| 15:42.39 | ManxPower | merlinn, I'll bet that if you can get together a good problem report, someone will be able to fix it. |
| 15:43.15 | *** part/#asterisk toxap (n=toxap@213.227.193.75) |
| 15:43.30 | *** join/#asterisk toxap (n=toxap@213.227.193.75) |
| 15:43.58 | *** part/#asterisk jmls (n=asterisk@host81-159-167-80.range81-159.btcentralplus.com) |
| 15:44.36 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:44.37 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:46.38 | *** join/#asterisk AsteriskMonkey (n=admin@209.250.138.244) |
| 15:46.41 | *** join/#asterisk Mportnoy (n=test@201.199.68.150) |
| 15:47.02 | AsteriskMonkey | hey has anyone experinced issue with ivrs not working right, in the sense that extensions occasionally say invalid etc? |
| 15:47.37 | ManxPower | AsteriskMonkey, that is usually caused by relaxdtmf=yes or too high or low gain on the port. |
| 15:47.46 | ManxPower | assuming the call is Zap |
| 15:47.50 | AsteriskMonkey | yes |
| 15:48.18 | AsteriskMonkey | rx tx gains could be a likley culprit |
| 15:48.24 | }btorch{ | AsteriskMonkey: yes I had those issues .. relaxdtmf was my problem |
| 15:48.24 | ManxPower | AsteriskMonkey, I once had a similar problem that was fixed by OUTGOING gain being lowered. |
| 15:48.29 | ManxPower | I suspect the DTMF was echoing. |
| 15:49.11 | }btorch{ | AsteriskMonkey: also some tests with ztmonitor helped me out setting up some proper tx/rx gains |
| 15:51.01 | AsteriskMonkey | yes this ones new to me never had this wierdness before, I actually see full digits though in the console :P |
| 15:51.15 | stoffell | exit |
| 15:51.24 | stoffell | oops;) l8errz |
| 15:51.28 | *** join/#asterisk bkw__ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 15:52.34 | AsteriskMonkey | mmm i see levels both at 2500 during calls |
| 15:56.47 | luke-jr_work | Is there a simple way to count minutes on an outbound channel, and autocongest when its monthly usage gets to x minutes? |
| 15:57.01 | mog | show application dial |
| 15:57.18 | mog | dial command has option to terminate after X time |
| 15:57.22 | mog | option L i believe |
| 15:57.24 | luke-jr_work | total, not a single call |
| 15:57.26 | mog | so you could have a variable |
| 15:57.31 | Qwell | astdb? |
| 15:57.32 | mog | for the month time |
| 15:57.34 | mog | and use that |
| 15:57.36 | mog | or astdb |
| 15:57.37 | luke-jr_work | something that would persist across a reboot |
| 15:57.38 | mog | would be easy |
| 15:57.41 | mog | yes |
| 15:57.45 | mog | astdb persists all |
| 15:57.51 | Qwell | mog: What do you think...4-5 lines? |
| 15:57.54 | mog | yeah |
| 15:58.08 | luke-jr_work | hmm |
| 15:58.13 | mog | maybe three, get variable, dial, set variable |
| 15:58.29 | luke-jr_work | doesn't Dial abort the dialplan? |
| 15:58.33 | luke-jr_work | if the call is successful |
| 15:59.12 | *** join/#asterisk c4t3l (n=c4t3l@69.15.174.114) |
| 16:00.40 | mog | you can catch it at h exten |
| 16:00.49 | mog | astcc |
| 16:00.54 | mog | already does all this for ya |
| 16:01.07 | luke-jr_work | hm |
| 16:01.09 | benjk | not if the remote party hangs up first |
| 16:01.22 | luke-jr_work | 'h' is used even if the ... yeah ;p |
| 16:01.32 | mog | you can catch both |
| 16:06.03 | jbeez | anyone know of a cheap IP phone that works with asterisk, and has a passthrough for the ethernet so you can have a pc plugged into the back of it, and it can recieve a field from the dhcp server telling it to switch vlans to the phone vlan? |
| 16:06.32 | AsteriskMonkey | Aastra 480i |
| 16:07.04 | AsteriskMonkey | Anyone know of a good way to page? im having issue with paging.. the meetme room kinda leaves people hanging after the page |
| 16:07.43 | jbeez | that looks pretty nice |
| 16:09.06 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 16:09.17 | De_Mon | mog why not use the cdr data? func_odbc! |
| 16:09.31 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 16:09.34 | mog | or that |
| 16:09.38 | mog | lots of ways to do it |
| 16:09.40 | mog | all easy |
| 16:09.41 | mog | ^_^ |
| 16:12.15 | *** join/#asterisk jake1932 (n=Administ@pool-68-236-5-134.phil.east.verizon.net) |
| 16:13.09 | jake1932 | can someone help me with a iax trace? http://pastebin.ca/183086 |
| 16:13.29 | jake1932 | asterisk answers then hangs up within a couple seconds |
| 16:13.35 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 16:15.11 | De_Mon | jake1932 you didn't include the dialplan |
| 16:15.20 | De_Mon | what's it supposed to do after playing the main greeting? |
| 16:15.38 | jake1932 | it doesn't get all the way through the main greeting before hanging up |
| 16:16.05 | jake1932 | it used to work and just stopped this week - no change in the dialplan (or anything for that matter) |
| 16:16.52 | jake1932 | to answer your question, it waits for an extension |
| 16:17.18 | *** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net) |
| 16:17.32 | hmmhesays | ok, whats the best way to kill a zombie call from the console |
| 16:17.40 | jake1932 | soft hangup? |
| 16:17.40 | De_Mon | hrm, I'd call my IAX provider ;) |
| 16:17.54 | ManxPower | jake1932, it only waits for an extension if autofallthru=no |
| 16:18.04 | ManxPower | jake1932, what version of Asterisk? |
| 16:18.06 | hmmhesays | ahh this build does not have it |
| 16:18.08 | jake1932 | .11 |
| 16:18.28 | ManxPower | jake1932, and you have autofallthru=no |
| 16:18.35 | ManxPower | or whatever the actual option is |
| 16:18.37 | jake1932 | letmesee |
| 16:19.01 | ManxPower | or just throw a WaitExten as the next priority |
| 16:19.01 | Crescendo | The quickstart guide gives me a running asterisk, no how do I set up some client machines? |
| 16:19.07 | ManxPower | Crescendo, client machines? |
| 16:19.21 | Crescendo | Yeah, let's say I want to just phone PC to PC. |
| 16:19.34 | ManxPower | Crescendo, check the docs for the softphone you are using |
| 16:19.44 | *** join/#asterisk richcorbs (i=keefejoh@gateway/web/cgi-irc/ircatwork.com/x-1bac2d06d181f933) |
| 16:19.48 | CunningPike | Crescendo: You need a SIP or IAX softphone - there are plenty about |
| 16:19.48 | Crescendo | I think I've FUBARred a config file somewhere, so that may be an issue. |
| 16:19.59 | Crescendo | Right, I know that. |
| 16:20.08 | Crescendo | I'm using this DIAX or whatever. |
| 16:20.16 | De_Mon | hah, I thought he actually had 'WaitExten' |
| 16:20.17 | Crescendo | But I don't want to call out, yet. |
| 16:20.55 | angryuser | i am using visdn with 2 Isdn lines(cologne chip) when i dial(VISDN/visdn0${exten}) normally i dont need to precise which channel to dial? |
| 16:20.58 | Crescendo | Should I make install again, will that restore everything for astersisk? |
| 16:20.59 | jake1932 | <PROTECTED> |
| 16:21.13 | De_Mon | ManxPower the whole 'it's not playing the whole greeting' might be a problem still |
| 16:21.20 | jake1932 | the call is dropping very fast |
| 16:21.27 | ManxPower | jake1932, It can't hurt to add it. |
| 16:21.43 | ManxPower | jake1932, I can't see any issue with the trace except for the last packet. |
| 16:21.48 | jake1932 | <PROTECTED> |
| 16:21.57 | richcorbs | anyone ever put dialogic T1 boards back-to-back like this: [server [dialogic]]====[[dialogic] Asterisk]? |
| 16:22.32 | jake1932 | ManxPower: tnx for that (should've put it been in there before) |
| 16:22.52 | Crescendo | Asterisk* |
| 16:23.06 | ManxPower | Crescendo, Ypi read The Book |
| 16:23.07 | ManxPower | ~book |
| 16:23.09 | jbot | [book] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 16:23.41 | ManxPower | richcorbs, I'm not aware that anyone has actually used a Dialogic board in an Asterisk server. |
| 16:23.41 | jake1932 | ManxPower: any clue on the lat packet? |
| 16:23.46 | Crescendo | ...I did. |
| 16:23.47 | ManxPower | jake1932, nope. |
| 16:24.48 | richcorbs | ManxPower, thanks...the boards are on the approved hardware list but I guess that doesn't mean anyone has actually used them :) |
| 16:25.08 | Crescendo | Will a fresh make install after editing configs restore settings to defaults? |
| 16:25.14 | ManxPower | richcorbs, the drivers are closed source and require purchase from Digium |
| 16:25.28 | *** join/#asterisk angom (n=angom@red-corp-200.79.133.11.telnor.net) |
| 16:25.46 | richcorbs | ManxPower, thanks, that is very good to know |
| 16:26.17 | ManxPower | richcorbs, I suspect Digium has to pay Intel a license fee for every driver sold |
| 16:26.51 | richcorbs | ManxPower, thanks again...do you work for Digium? |
| 16:27.13 | ManxPower | richcorbs, no. If I worked for Digium I would not be able to insult people that really deserve it. |
| 16:27.34 | GerbilWrk | Can anyone point me in the direction of a linux sftp server? |
| 16:27.41 | ManxPower | richcorbs, but I've been around for quite a long time |
| 16:28.38 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 16:28.52 | TripleFFFF | can one point me to valid app_Rxfax for 1.2..12 ?please |
| 16:29.01 | richcorbs | ManxPower, what I really need is an affordable way to load test a voice application across multiple T1s connecting in to Dialogic boards |
| 16:29.16 | ManxPower | richcorbs, Asterisk may not be your answer. |
| 16:29.36 | richcorbs | ManxPower, any other ideas off the top of your head? |
| 16:30.03 | *** join/#asterisk Bigfoot_home (n=simon@host86-139-134-16.range86-139.btcentralplus.com) |
| 16:30.42 | richcorbs | ManxPower, (don't want to take too much of your generous time) |
| 16:30.42 | ManxPower | richcorbs, you could always do Asterisk[digium/sangoma card] -> [dialogic[yourdialogicivr |
| 16:30.43 | ManxPower | Use asterisk to generate the calls |
| 16:30.44 | richcorbs | exactly |
| 16:31.05 | ManxPower | if you put dialogic boards in Asterisk you would not be testing the boards, you would be testing the drivers. |
| 16:31.05 | richcorbs | ManxPower, would I need DSUs between the sangoma-dialogics? |
| 16:31.14 | ManxPower | richcorbs, no, just a T-1 crossover |
| 16:32.18 | ManxPower | richcorbs, just be prepared to learn ALOT about Asterisk just to get to the point where you can generate calls. |
| 16:33.15 | richcorbs | ManxPower, could I not use SIPP or another SIP-based tool to generate calls through Asterisk? |
| 16:37.39 | ManxPower | richcorbs, you could, but there are much easier ways, called ".call files" |
| 16:37.53 | *** join/#asterisk Cyt (n=danielcy@athedsl-162092.otenet.gr) |
| 16:37.56 | Juggie | even easier then call files, Action Originate |
| 16:38.19 | richcorbs | ManxPower, can .call files be scripted and play audio? |
| 16:38.37 | Crescendo | http://www.voip-info.org/wiki/view/Asterisk+Step-by-step+Installation - this document provides VoicePulse. I don't want it. |
| 16:38.50 | Crescendo | How do I configure Asterisk without it? |
| 16:38.55 | ManxPower | richcorbs, the call files can be used to generate calls then connect the local leg of the call to an asterisk application or extension or dialplan |
| 16:39.34 | ManxPower | richcorbs, create the text file, drop it into /var/spool/asterisk/outgoing and asterisk will process it. |
| 16:39.54 | Hymie | hmm... just where does the 'o' extension go, I tried in the context the call came in on, and that doesn't work |
| 16:40.13 | ManxPower | Hymie, same context as the Voicemail app is |
| 16:40.25 | ManxPower | for this macro == context |
| 16:40.36 | Hymie | sec |
| 16:40.41 | *** join/#asterisk edguy3 (n=edguy@host-208-115-200-88.patmedia.net) |
| 16:40.54 | Hymie | hmm, ok, thanks |
| 16:40.56 | Hymie | trying |
| 16:40.59 | richcorbs | ManxPower, thanks again...you got me pointed in the right direction...much appreciated |
| 16:41.30 | ManxPower | richcorbs, just send a paypal to eric@fnords.org for %25 of the money my advice saved you 8-) |
| 16:42.25 | mog | lol ManxPower |
| 16:43.31 | ManxPower | mog, I think the integer size is too small. Obviously the amount wrapped around. |
| 16:43.42 | mog | lol |
| 16:43.52 | ManxPower | 8-) |
| 16:45.40 | Hymie | hmm |
| 16:45.47 | Hymie | still don't get the option to press 0 to reach the operator |
| 16:45.50 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 16:45.50 | *** mode/#asterisk [+o mog] by ChanServ |
| 16:46.10 | richcorbs | MaxPower, lol...thanks, my appreciation is all I can offer at this point |
| 16:46.16 | *** part/#asterisk richcorbs (i=keefejoh@gateway/web/cgi-irc/ircatwork.com/x-1bac2d06d181f933) |
| 16:46.57 | jake1932 | ManxPower: you gotta negotiate it beforehand |
| 16:49.16 | *** join/#asterisk x86_ (n=x86@p3m/member/x86) |
| 16:50.22 | *** join/#asterisk aptura (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 16:51.05 | *** part/#asterisk jake1932 (n=Administ@pool-68-236-5-134.phil.east.verizon.net) |
| 16:51.43 | *** join/#asterisk jtodd (n=jtodd@adsl-75-24-91-217.dsl.pltn13.sbcglobal.net) |
| 16:52.08 | *** join/#asterisk DanTMG (n=danielga@203-206-234-123.dyn.iinet.net.au) |
| 16:57.13 | [TK]D-Fender | ManxPower: Got that statement scripted don't you? :) Saw it letter for letter yesterday :) |
| 16:57.28 | Crescendo | http://www.voip-info.org/wiki/view/Asterisk+Step-by-step+Installation - this document provides VoicePulse. I don't want it. |
| 16:57.31 | Crescendo | How do I configure Asterisk without it? |
| 16:59.16 | *** join/#asterisk Givemelove (n=foo@208.57.229.162) |
| 17:01.34 | ManxPower | [TK]D-Fender, nope. |
| 17:01.49 | ManxPower | I figure that if I'm going to beg for money, the least I can do it type it out each time. |
| 17:02.00 | ManxPower | (well other than the .sig for email) |
| 17:02.39 | *** join/#asterisk deb_user (n=none@70-59-108-105.albq.qwest.net) |
| 17:03.03 | *** join/#asterisk champster (n=asterisk@AH.tescogroup.com) |
| 17:03.09 | deb_user | anybody have any insight on getting echo on a TDM interface? |
| 17:03.20 | deb_user | can't seem to get rid of an echo |
| 17:03.35 | [TK]D-Fender | lol |
| 17:04.21 | *** join/#asterisk smackus (n=ckwall@63.149.122.93) |
| 17:04.22 | Hymie | ManxPower: you negected to tell me that the important aspect of getting the operation option to work.. the sort of crux of it all, is actually uncommenting the operator= line in voicemail.conf, and not just THING YOU HAVE #@$@($@)# hehe |
| 17:04.26 | deb_user | i've got echo cancellation and training enabled in zapata.conf |
| 17:05.00 | ManxPower | Hymie, I find that it builds character to make the user work a little. |
| 17:05.49 | Hymie | ManxPower: well, shit.. I looked at it, I thought I uncommented it... I even had another copy of the voicemail.conf, on another user's server in front of me, thinking that was my copy ;P |
| 17:06.00 | Hymie | #$@#$ TOO MANY SHELLS |
| 17:06.13 | CtRiX | <deb_user> anybody have any insight on getting echo on a TDM interface? |
| 17:06.13 | CtRiX | <deb_user> can't seem to get rid of an echo |
| 17:06.15 | CtRiX | ROTFL! |
| 17:06.19 | CtRiX | another one ! |
| 17:06.26 | CtRiX | buy sangoma |
| 17:06.27 | Crescendo | http://www.voip-info.org/wiki/view/Asterisk+Step-by-step+Installation - this document provides VoicePulse. I don't want it. |
| 17:06.28 | Crescendo | How do I configure Asterisk without it? |
| 17:06.28 | deb_user | why is that so funny? |
| 17:06.50 | CtRiX | because echo is a free addon with that cards. it's bundled. |
| 17:07.25 | sx-wks | CtRiX: echo ? with what card ? |
| 17:07.32 | *** join/#asterisk razu (n=rasmus@tln-kontor.norby.ee) |
| 17:07.40 | sx-wks | I don't get no echo on my TDM400P |
| 17:07.51 | Hymie | CtRiX: explain |
| 17:07.57 | Hymie | CtRiX: where is this card? |
| 17:08.00 | Hymie | CtRiX: can I have three? |
| 17:08.02 | Corydon-w | Echo is an engineering problem with lots of attempts to solve it, with nothing providing a 100% solution |
| 17:08.35 | Corydon-w | Most solutions are simply "good enough" |
| 17:08.35 | Hymie | Corydon-w: you are wrong, CtRiX has a solution that will fix it |
| 17:08.35 | Hymie | Corydon-w: it is perfect |
| 17:09.16 | Corydon-w | Yeah, I forget. Nothing can ever compete with Sangoma. All praise the magical beast that is Sangoma |
| 17:09.22 | aptura | BY any chance anyone personally know ian murdock? I think I was driving past him on highway 520 in redmond and his licence plate said Debian |
| 17:10.33 | deb_user | so i pretty much am stuck with echo on my tdm? |
| 17:10.47 | deb_user | funny thing is, it doesn't happen all the time |
| 17:11.24 | Corydon-w | Oh, your're hearing the echo, then? |
| 17:11.36 | sx-wks | well... I don't get no crappy echo on my card. dunno what you guys are talkin' about |
| 17:11.41 | Corydon-w | The problem is on the other side of the connection |
| 17:12.20 | *** join/#asterisk ToTo (n=ToTo@host149-109-dynamic.58-82-r.retail.telecomitalia.it) |
| 17:13.02 | sx-wks | Corydon-w: as in, get a better phone ? |
| 17:13.05 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 17:13.13 | sx-wks | mine are $5 walmart implements |
| 17:13.22 | Corydon-w | sx-wks: no, as in, call someone with better equipment |
| 17:13.27 | Dr-Linux | anybody is using Shpinx2 voice recognition with asterisk? |
| 17:13.51 | deb_user | corydon-w: well, i'm connected to central office via vpn |
| 17:14.05 | deb_user | echo comes in from fxs port to my iax softphone |
| 17:14.38 | Corydon-w | deb_user: right, the central office relays echo without doing anything to it |
| 17:14.44 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 17:14.53 | Corydon-w | deb_user: it's coming from the person you called |
| 17:14.56 | *** join/#asterisk linagee (n=linagee@unaffiliated/linagee) |
| 17:15.12 | sx-wks | Corydon-w: ah well :D |
| 17:15.13 | deb_user | the person I called is in the central office where the asterisk server is located |
| 17:15.23 | deb_user | i can hear the echo, she can't |
| 17:15.26 | *** join/#asterisk Cyt (n=danielcy@athedsl-162092.otenet.gr) |
| 17:15.43 | linagee | is there a such thing as a nextel voip gateway? (a free one? :-D ) |
| 17:15.48 | Corydon-w | deb_user: Ask her to press the handset tightly against her ear |
| 17:16.06 | Corydon-w | deb_user: that will confirm that the echo is coming from her crappy phone |
| 17:16.37 | deb_user | corydon: no, because even when I call out from the zap interface in hear the echo |
| 17:16.48 | deb_user | to a cell phone or someting |
| 17:16.51 | deb_user | on the fxo port |
| 17:17.42 | linagee | Corydon-w: so you want him to say, "mam, calm down. put the crappy phone to your ear" LOL |
| 17:18.28 | Corydon-w | deb_user: okay, so you hear it only when you use the PSTN? |
| 17:18.31 | linagee | deb_user: don't they have those echo reducing fxo/fxs ports? |
| 17:18.46 | linagee | (or a software way to do it?) |
| 17:18.53 | deb_user | corydon-w: not exactly...i hear it only when I use the tdm400 |
| 17:19.00 | CtRiX | yeah |
| 17:19.07 | deb_user | corydon: I have an iax termination service that doesn't give me any echo at all |
| 17:19.39 | deb_user | but when I receive a call from the central office, it goes from a legacy pbx via an fxs port and then via iax2 to my softphone here at home office |
| 17:19.41 | Corydon-w | You can compensate by lowering the txgain to negative |
| 17:19.41 | Cresl1n | deb_user what kind of zap card are you calling on? |
| 17:19.57 | deb_user | digium tdm400 |
| 17:20.43 | Corydon-w | Ah, yeah, you can also use the fxotune command |
| 17:21.11 | Corydon-w | That usually does an awesome job at wiping out echo |
| 17:21.16 | deb_user | corydon: how do I use that? |
| 17:21.58 | linuxbangalore | Sep 27 01:20:01 NOTICE[18214]: res_musiconhold.c:515 monmp3thread: Request to schedule in the past?!?! |
| 17:22.03 | Corydon-w | shutdown Asterisk, then run '/usr/src/zaptel/fxotune -i 4' |
| 17:22.21 | Cresl1n | deb_user: use fxotune |
| 17:22.24 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 17:22.25 | linagee | deb_user: http://www.voip-info.org/wiki/view/Asterisk+fxotune |
| 17:22.26 | linuxbangalore | is that line because I have wrong time setup on my machine |
| 17:22.28 | Dr-Linux | anybody tried sphinx2 with asterisk? |
| 17:22.30 | Corydon-w | then run '/usr/src/zaptel/fxotune -s' on boot |
| 17:22.31 | ManxPower | linagee, no. |
| 17:22.34 | ManxPower | linuxbangalore, no. |
| 17:22.35 | deb_user | ok... |
| 17:22.37 | linagee | ManxPower: no? |
| 17:22.38 | deb_user | i'll try it |
| 17:22.46 | linagee | ManxPower: ah, about the nextel gateway? |
| 17:22.55 | ManxPower | linuxbangalore, that is just a message that says "I didn't get to the MOH in time, maybe you'll her an MoH blip. |
| 17:23.05 | ManxPower | linagee, the no was for linuxbangalore |
| 17:23.36 | Cresl1n | deb_user: I'll bet that that will fix your problem for you |
| 17:23.48 | linuxbangalore | ManxPower, here get this other error... |
| 17:23.48 | linuxbangalore | Sep 27 01:20:19 NOTICE[18219]: pbx.c:1741 pbx_extension_helper: Cannot find extension context 'sip' |
| 17:23.49 | Cresl1n | deb_user: if not, contact me on IRC and I'll help you fix it |
| 17:23.50 | linagee | deb_user: from the wiki: "(`fxotune -i 4`) It should discover which zap channels are FXO modules and tune them accordingly. Be warned however, it takes a significant amount of time for EACH module to test, I would say somewhere around 2-3 minutes." |
| 17:24.05 | Cresl1n | deb_user: also, before you do, make sure you update zaptel to the beta release of it |
| 17:24.35 | deb_user | linagee: thanks...I was just looking at that too after googling it :) |
| 17:24.55 | deb_user | tuning module three right now |
| 17:25.15 | deb_user | anybody care to explain in laymens terms what the tuning exactly does? |
| 17:25.24 | deb_user | just curious... |
| 17:26.04 | linagee | deb_user: you've never seen dark city? :-D |
| 17:26.10 | linagee | deb_user: TUNING.... :-D |
| 17:26.16 | deb_user | linagee: nope |
| 17:26.26 | linagee | aw. then it was wasted. :(O |
| 17:26.41 | ManxPower | linuxbangalore, you do not have a context called [sip] in extensions.conf |
| 17:26.50 | linagee | deb_user: search for tuning. http://www.mediacircus.net/darkcity.html |
| 17:26.51 | Corydon-w | deb_user: it sets the echo coefficients on the card to be better aligned with your particular telco installation |
| 17:27.01 | *** join/#asterisk tRSS (n=tRSS@193.220.221.2) |
| 17:27.05 | deb_user | corydon: ok |
| 17:27.08 | linagee | deb_user: "Armed with a telepathic ability that can shape reality, called 'tuning', they roam the streets of the unnamed city" |
| 17:27.13 | *** part/#asterisk a1fa (n=a1fa@unaffiliated/a1fa) |
| 17:27.24 | linagee | deb_user: very strange/interesting movie |
| 17:27.25 | deb_user | sheesh, its still tuning |
| 17:27.34 | deb_user | linagee: i'll put it on my netflix |
| 17:27.34 | linuxbangalore | ok |
| 17:27.37 | DanTMG | just about to setup a confrencing box, what hardware specs do you guys think woul be suitable for 60 users i.e. any indicitive CPU usage per user? |
| 17:27.51 | CtRiX | i like echo coefficients |
| 17:27.55 | Corydon-w | deb_user: because it's aligned to a particular installation, that's why it couldn't be preset to levels which helped you at the outset |
| 17:28.06 | tRSS | what are the values that I can set for context in sip.conf? Can I tell not to go to any context under [general] |
| 17:28.07 | tRSS | ? |
| 17:28.12 | Cresl1n | deb_user: what version of zaptel are you using? |
| 17:28.17 | linagee | deb_user: then i won't go into the movie in too much detail. :) (but it's awesome) |
| 17:28.36 | deb_user | cresl1n: 1.2.7 |
| 17:28.44 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 17:28.51 | Cresl1n | deb_user: tune it with fxotune from zaptel-beta |
| 17:28.58 | linagee | Corydon-w: i don't get it either. tuning. aligned. i see a sine wave graph on the wiki. maybe we just don't know enough about how the PSTN works. |
| 17:29.07 | deb_user | cresl1n: why? it won't work otherwise? |
| 17:29.18 | Cresl1n | deb_user: it performs a LOT better with that version |
| 17:29.25 | deb_user | really? |
| 17:29.35 | Corydon-w | deb_user: Cresl1n should know. He wrote it. |
| 17:29.39 | Cresl1n | deb_user: and also there's an updated version of the zaptel echo canceler in the beta release |
| 17:29.45 | linagee | 'balancing the hybrid' |
| 17:29.47 | deb_user | cresl1n: do I have to install the beta version of zaptel for it to work? |
| 17:29.48 | Cresl1n | which will help you with your echo problems also |
| 17:29.56 | Cresl1n | deb_user: yes |
| 17:30.03 | Cresl1n | but you shouldn't have to update asterisk or anything else |
| 17:30.21 | deb_user | cresl1n: is an upgrade just like a fresh install? |
| 17:30.30 | deb_user | pretty much follow the instructions as is? |
| 17:30.35 | Cresl1n | yep |
| 17:30.38 | deb_user | ok |
| 17:30.45 | deb_user | i'll have to do that outside of working hours |
| 17:30.48 | deb_user | or during the weekend |
| 17:31.01 | deb_user | already the tuning is taking a little longer than my users will tolerate |
| 17:31.16 | deb_user | still tuning the first module...its almot been ten minutes |
| 17:31.16 | Juggie | Deb, stop *, unload /etc/init.d/zaptel stop, make install on zaptel-beta dir, /etc/init.d/zaptel start, safe_asterisk |
| 17:31.17 | Juggie | huzzah |
| 17:31.29 | linagee | Cresl1n: phone wires are like a "shared bandwidth", right? computer talks out, it gets what it said back to itself (and there is latency with the phone network.) this is the echo? |
| 17:31.39 | Juggie | linagee, no. |
| 17:32.15 | Cresl1n | linagee: well, it's a phenomenon that occurs when you have a hybrid (a 2 wire interface to 4 wire interface converter) |
| 17:32.27 | Cresl1n | primarily electrical in nature |
| 17:32.55 | linagee | Cresl1n: does the tdm400 have echo cancelers on the card? is that what it adjusts, or is it just all in software? |
| 17:33.23 | linagee | "fxotune does not do anything with the echo canceler algorithms themselves - instead, it optimizes the signal before it gets to the echo canceler, making it easier for the echo canceler to do it's work" |
| 17:33.30 | linagee | hm |
| 17:33.44 | Cresl1n | linagee: it has a hybrid that can be tuned |
| 17:33.46 | Cresl1n | that's part of it |
| 17:33.58 | Cresl1n | and it has a little itty bitty echo canceler on the card |
| 17:34.02 | *** join/#asterisk zeppelin_ (n=zeppelin@201.41.0.159) |
| 17:34.11 | Cresl1n | but it's not a true echo canceler, it's just a filter that you have to pre-tune |
| 17:34.18 | CtRiX | it tunes echo coefficients |
| 17:34.23 | Cresl1n | exactly |
| 17:34.33 | Juggie | and sometimes makes Cresl1n breakfast |
| 17:34.35 | Juggie | though not allways |
| 17:34.35 | linagee | Cresl1n: it really does sound like a lot of effort just to save two additional wires of copper to every household. lol. |
| 17:34.40 | Cresl1n | CtRiX: and if you buy sangoma cards, you get echo free with them as well |
| 17:34.41 | Cresl1n | :-P |
| 17:34.58 | Juggie | or, you could just go digital :) |
| 17:35.03 | Juggie | which saves us all alot of trouble. |
| 17:35.08 | CtRiX | probably... but far less echo |
| 17:35.15 | Cresl1n | linagee: if you're using a regular phone, you don't notice the echo |
| 17:35.18 | Cresl1n | it comes out as side tone |
| 17:35.25 | linagee | hmm |
| 17:35.31 | Cresl1n | you don't have echo cancelers on regular telephones |
| 17:35.50 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 17:35.50 | *** mode/#asterisk [+o mog] by ChanServ |
| 17:35.52 | *** join/#asterisk xming (n=xming@gentoo/user/xming) |
| 17:35.57 | trelane_ | Cresl1n, yes it does, it's hte little LCX245 |
| 17:35.57 | Cresl1n | it's just when you start introducing significant delays, the sidetone becomes echo |
| 17:35.59 | Juggie | Cresl1n, an got an order of 10 TE207's :P |
| 17:36.05 | Cresl1n | so that's why you need echo cancelers |
| 17:36.06 | CtRiX | mog has fixed his client ! |
| 17:36.35 | Juggie | er, i just got my order of 10 TE207's... i need more coffee. |
| 17:36.59 | mog | was it acting crazy |
| 17:37.02 | *** join/#asterisk Ox0F0-0FF (n=pierre@200.216.238.226) |
| 17:37.03 | mog | my appologies |
| 17:37.12 | Cresl1n | mog had a runaway IRC client |
| 17:37.15 | Juggie | hah. |
| 17:37.16 | mog | its what i get for running svn trunk |
| 17:37.40 | Juggie | are you guys going to have any of the appliances @ astricon? |
| 17:37.49 | Cresl1n | probably |
| 17:38.34 | mog | or more aptly what you get |
| 17:38.36 | mog | yes |
| 17:38.51 | Cresl1n | deb_user: have you got zaptel-beta yet? |
| 17:39.14 | deb_user | Cresl1n: I'm going to wait until after business hours to do any more changes, just in case |
| 17:39.51 | Cresl1n | deb_user: you won't have any problems |
| 17:39.58 | Cresl1n | it won't break anything |
| 17:39.58 | deb_user | cresl1n: i believe you |
| 17:39.59 | linagee | Cresl1n: is echo cancellation so different on every unique install that it can't just have preset values? |
| 17:40.01 | Cresl1n | it'll just work |
| 17:40.16 | Cresl1n | linagee: yeah, that's why fxotune was created |
| 17:40.19 | Juggie | linagee, echo isnt a static thing. |
| 17:40.20 | deb_user | cresl1n: but, it just makes it easier |
| 17:40.29 | jbeez | anyone use linksys ip phones? Are they any good since cisco owns linksys, or are they junk like I expect? |
| 17:40.30 | deb_user | because while its tuning I have to shut down asterisk |
| 17:40.35 | linagee | Cresl1n: i see. things like line length, impedance... hrm |
| 17:40.40 | Cresl1n | so that it would tune the card to the characteristics inherent in the line |
| 17:40.50 | Cresl1n | all figments of analogue lines, of course |
| 17:40.56 | deb_user | cresl1n: but the 1.2.7 did the tuning...so I'll wait this evening to install beta |
| 17:41.00 | Cresl1n | for digital lines, this is all moot |
| 17:41.21 | deb_user | and...I've got some other stuff pressing that has priority, thanks for all your help |
| 17:41.22 | Cresl1n | deb_user: well, you could try it with the tuning on 1.2.7, but I'd tune it with the beta version |
| 17:41.40 | Cresl1n | and not only that, you get the newer echo canceler |
| 17:41.48 | deb_user | cresl1n: I already tuned with 1.2.7, now I'll wait until this evening to install the beta version |
| 17:41.57 | Cresl1n | ok |
| 17:42.06 | deb_user | again, thanks for your help |
| 17:42.08 | Cresl1n | see if it sounds better with the tuning done by the 1.2.7 version |
| 17:42.09 | Cresl1n | ok |
| 17:42.11 | Cresl1n | cya! |
| 17:42.16 | linagee | Cresl1n: i think one nice use of an fxo would be to have an emergency landline set up just for 911. then you don't have to worry about 911 over voip and if the intarweb goes down. |
| 17:42.16 | deb_user | I'll check it right now |
| 17:42.23 | jbalcomb | Asterisk does not know that my multi-queue member is on wrap up for Queue#1 when a call comes in for Queue#1. Bug? How to fix? |
| 17:42.34 | *** join/#asterisk pingwin (n=pingwin@216.249.143.62) |
| 17:42.57 | Cresl1n | deb_user: make sure you run fxotune -s so that it sets the coefficients |
| 17:43.15 | deb_user | oh...whoops, ok |
| 17:43.15 | Cresl1n | you need to run that at boot time, after the driver for the card has loaded |
| 17:43.24 | deb_user | do I need to run it right now? |
| 17:43.26 | *** join/#asterisk flujan (i=flujan@201-42-70-91.dsl.telesp.net.br) |
| 17:43.33 | pingwin[work] | i'm having stupid fever with my dialplan. I'm sorry. but how do I make logic for if someone dials 1 for sales, 2 for support, 3 for blah type of example? |
| 17:43.37 | deb_user | i almost never reboot my * machine |
| 17:43.48 | pingwin[work] | WaitExten wants a real extensions within my default context and not within my macro |
| 17:43.49 | flujan | hi guys, i want to understand the output from the iax2 jb debug command. |
| 17:44.09 | flujan | I am having problems in the communication |
| 17:44.46 | linagee | dial 1 for sales, dial 2 for shell access, etc. :-D |
| 17:45.17 | pingwin[work] | well like I said linagee it's looking for the extension in the wrong context, not within my macro |
| 17:45.29 | flujan | http://pastebin.ca/183178 here you can take a look. :) |
| 17:45.42 | flujan | I dunno how to interpret this... |
| 17:45.50 | linagee | pingwin[work]: *shrug*. i'm a freepbx user. i know how to poke around extensions_custom.conf and that's all i need to know. :-D |
| 17:46.03 | justinu|laptop | flujan: it's completely intuitive, what's the problem? ;) |
| 17:46.15 | pingwin[work] | linagee: well thanks for your input |
| 17:46.20 | Cresl1n | deb_user: sure you should run it |
| 17:46.30 | CunningPike | linagee: That's exactly what we do for 911 - all our calls go out through the PRI at our main office, except for 911 calls from our remote offices, which are sent to SPA-3000s and then out over 1Bs |
| 17:46.36 | flujan | justinu|laptop, what means v, L, l, 1, s, G |
| 17:46.37 | Cresl1n | you can run it any time after the driver has been loaded |
| 17:46.44 | Druken | does asterisk still have the problem with vmwi usinf realtime? |
| 17:46.48 | flujan | justinu|laptop, to me it is not intuitive... :( |
| 17:47.13 | linagee | CunningPike: that's smart. then your only fault points are UPS power for phones, UPS power for the asterisk servers, and then it's all in the hands of your CO. :-D |
| 17:47.24 | benjk | its into-itive |
| 17:47.25 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 17:47.40 | benjk | you have to be "into" it, for it to be intuitive |
| 17:47.41 | deb_user | cresl1n: ok, I'm still getting echo, but only when there's someone speaking or noise on the other end of the line |
| 17:47.53 | linagee | CunningPike: what if there was a fire in the server room? lol |
| 17:48.13 | justinu|laptop | flujan: i know, i was just being sarcastic. honestly you will probably have to find the person who wrote the jitter buffer, or figure out the code yourself. |
| 17:48.27 | CunningPike | linagee: Phones are on PoE which is on UPS/genset, as are the SPA-3000. If all that fails, each SPA-3000 has a red phone in the FXS port that can be used to dial direct to the PSTN |
| 17:48.37 | Cresl1n | deb_user: so it's better? |
| 17:48.38 | linagee | CunningPike: maybe you could have the voip phone hunt for asterisk servers |
| 17:48.53 | deb_user | cresl1n: not really...that was the problem all along |
| 17:48.57 | *** join/#asterisk oej (n=oej@231.Red-88-14-197.dynamicIP.rima-tde.net) |
| 17:49.04 | Cresl1n | what does your zapata.conf look like? |
| 17:49.04 | CunningPike | w/b, oej |
| 17:49.17 | deb_user | i'll put it on pastebin |
| 17:49.18 | deb_user | one sec |
| 17:49.21 | Cresl1n | ok |
| 17:49.30 | flujan | justinu|laptop, ok ... :D |
| 17:49.36 | flujan | justinu|laptop, thanks anyway. |
| 17:50.42 | linagee | CunningPike: so maybe if there's a huge cube farm with lots of voip, you'd have red phones strategically placed with signs that say, "911 only" |
| 17:51.18 | *** join/#asterisk Cinen (n=Cinen@208.70.20.3) |
| 17:51.26 | CunningPike | linagee: Only at our remote offices - our main office where the PRI is, we just use the PRI |
| 17:51.49 | CunningPike | linagee: The issue is with the ANI/ALI, rather than fault tolerance |
| 17:52.04 | linagee | can i emulate a modem over voip? i'm using PCM to my voip provider... |
| 17:52.23 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 17:52.23 | *** mode/#asterisk [+o mog] by ChanServ |
| 17:52.43 | deb_user | sheesh, pastebin is slow |
| 17:52.55 | *** join/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net) |
| 17:53.12 | ichilton | anyone use cisco 79xx phones here? |
| 17:55.08 | deb_user | cresl1n: http://pastebin.com/794916 |
| 17:55.54 | *** join/#asterisk liran_ (n=liran@212.199.177.208.static.012.net.il) |
| 17:56.44 | jbalcomb | ichilton: yes |
| 17:57.39 | CunningPike | deb_user: Use pastebin.ca |
| 17:57.41 | CunningPike | ~pb |
| 17:57.47 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 17:57.47 | tRSS | I have a question: How can i make people from one context (B) to be able to call people in another context(A), without being able to use anything else from context (A) (e.g. include => statements to be able to dial international numbers that is allowed to context A people, but not to context B people)? |
| 17:58.09 | deb_user | cresl1n: its pretty much the default zapata.conf file...with a few config changes for my interfaces |
| 17:58.13 | *** part/#asterisk liran_ (n=liran@212.199.177.208.static.012.net.il) |
| 17:58.36 | jtexter3 | tRSS: it's all in how you segment your dialplan, and what includes you use |
| 17:58.56 | jtexter3 | For your setup, I would have a context named internal-extensions, which contains how to call other people internally |
| 17:59.05 | tRSS | jtexter3: should I pastebin, to give you a better understanding of my dialplan? |
| 17:59.11 | ichilton | jbalcomb: I just got a 7940 today - i've got it connecting to asterisk and it's making outgoing calls ok but it wont work when you try and ring it |
| 17:59.12 | jtexter3 | Sure |
| 17:59.23 | ichilton | jbalcomb: asterisk gives a SIP error |
| 17:59.38 | jbalcomb | ichilton: whats your dtmfmode? |
| 17:59.43 | ichilton | jbalcomb: there seems to be a small cross next to the line on the screen though - is that normal? |
| 17:59.48 | ichilton | jbalcomb: the rfc one |
| 18:00.02 | Cresl1n | deb_user: try turning echotraining off |
| 18:00.09 | ichilton | jbalcomb: dtmfmode=rfc2833 |
| 18:00.12 | deb_user | really? |
| 18:00.12 | deb_user | ok |
| 18:00.57 | benjk | its normal only if you are catholic |
| 18:01.24 | tRSS | jtexter3: give me just a min, putting in on pb |
| 18:01.57 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 18:02.19 | twisted[work] | omg its the virgin mary on my desktop |
| 18:02.20 | jbalcomb | ichilton: ok, thats right. |
| 18:02.25 | twisted[work] | its a miracle! |
| 18:02.47 | jbalcomb | ichilton: whats the SIP error? |
| 18:03.07 | *** join/#asterisk re-pete (n=repete@24.96.201.72) |
| 18:03.24 | jbalcomb | twisted[work]: thats not the virgin mary, its Virtual Stripper. |
| 18:03.29 | re-pete | is mpg123 still required for 1.2? |
| 18:03.47 | jbalcomb | re-pete: no, mpg123 is bad. use madplay. |
| 18:03.54 | jtexter3 | re-pete: I'm using format_mp3, and it's working great for me |
| 18:03.54 | deb_user | cresl1n: still pretty much the same |
| 18:03.57 | ManxPower | tRSS, I gave you the answer yesterdat |
| 18:04.21 | re-pete | so mpg123 is bad now... ok, thanks guys |
| 18:04.26 | jbalcomb | ichilton: i beleive the cross means its not registered |
| 18:04.39 | benjk | excommunicated |
| 18:04.43 | smackus | I see the variable dnis, but I do not know how, or if it is possible to populate the cdr data with dnis. Is there something out there for this? Right now I do not see any data in my cdr with dnis |
| 18:04.44 | ManxPower | re-pete, mpg123 was ALWAYS BAD. However was still required |
| 18:04.53 | *** join/#asterisk kristalino (i=kristali@gateway/tor/x-e424111546898b1d) |
| 18:04.54 | jbalcomb | re-pete: yeah, its related to issues with alsa and compiling and libs or something. its unsupported. |
| 18:05.23 | jbalcomb | Why is Asterisk stupid about members in multiple queues? |
| 18:05.50 | tRSS | ManxPower: you did, but your solution is not fitting into my dialplan, which is much more complex |
| 18:07.03 | ManxPower | tRSS, eventually you'll realize your dialplan is badly designed and go to using my design |
| 18:07.22 | ichilton | jbalcomb: just a normal couldn't create SIP something... (sorry, at home now so dont have the real error to hand) |
| 18:07.24 | jbalcomb | haha.. manxpower, you're so bogus. |
| 18:07.31 | tRSS | ManxPower: either I do that, or I try to fix mine. I won't give up this easily ;) |
| 18:07.37 | ichilton | jbalcomb: is there any way I can find out why it's not registering? |
| 18:07.40 | deb_user | cresl1n: i guess i'll try the beta version tonight, and hope that helps |
| 18:08.08 | jbalcomb | ichilton: hrmm.. yeah, sounds like its not registering. turning of guest access will probably break being able to make outgoing calls. |
| 18:08.30 | ManxPower | tRSS, Your design is flawed. |
| 18:08.33 | jbalcomb | ichilton: check your logs, find the error, report back. |
| 18:08.36 | justinu|laptop | ~seen r_evolution |
| 18:09.05 | jbot | r_evolution <i=r_evolut@208.251.203.246> was last seen on IRC in channel #asterisk, 13d 17h 33m 1s ago, saying: 'realtime or flat file'. |
| 18:09.05 | ManxPower | jbalcomb, in all honestly, it's not "my design", it is "the correct design", which many many many people use. |
| 18:09.08 | *** join/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 18:10.20 | ichilton | jbalcomb: ok |
| 18:10.21 | *** join/#asterisk file2 (n=IrcNet@out.clearnet.com) |
| 18:10.21 | *** mode/#asterisk [+o file2] by ChanServ |
| 18:10.24 | ichilton | jbalcomb: thanks |
| 18:10.37 | ManxPower | You MUST seperate extensions that ring phones and extensions that dial out to the PSTN into different contexts if you want any access control at all |
| 18:12.07 | *** join/#asterisk mafkees (n=michiel@vanbaak.xs4all.nl) |
| 18:12.09 | jbalcomb | ichilton: np |
| 18:12.18 | mafkees | hello all :) |
| 18:12.50 | mafkees | how can I uninstall zaptel if I compiled from source ? |
| 18:13.08 | jbalcomb | ManxPower: How do you get Asterisk to consider wrapuptime per agent rather than per queue so multiqueue members don't get backs back to back to back? |
| 18:13.27 | jbalcomb | s/backs/calls/ |
| 18:13.49 | *** join/#asterisk rbd (n=rbd@adsl-074-229-183-112.sip.rmo.bellsouth.net) |
| 18:14.27 | *** join/#asterisk _alex_mx_ (n=alex@200.78.229.18) |
| 18:14.27 | ichilton | jbalcomb: just trying it on my home asterisk system |
| 18:14.34 | ichilton | jbalcomb: can you not dial # numbers at all? |
| 18:14.51 | rbd | in the case of accessing an agi script via asterisk-java (or some other agi server... agi:// URL)...is the script simply sent to asterisk and cached all at once, or 'streamed' to asterisk (asterisk requesting parts of the script as it needs it)? |
| 18:15.15 | ichilton | jbalcomb: I use like #9<number> for outgoing calls |
| 18:15.38 | GerbilWrk | can asterisk delete a file from the dialplan? |
| 18:15.44 | jbalcomb | ichilton: if its not registered that would make sense |
| 18:15.58 | jbalcomb | ichilton: you did up date to the SIP firmware? |
| 18:16.03 | tRSS | jtexter3: i have pastebin'ed my problem: http://pastebin.ca/183205 |
| 18:16.17 | *** join/#asterisk syzygyBSD_ (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 18:16.26 | *** part/#asterisk syzygyBSD_ (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 18:16.28 | CtRiX | GerbilWrk, System() |
| 18:16.43 | ichilton | jbalcomb: no, I mean generally - # seems to be the send key |
| 18:16.43 | CtRiX | GerbilWrk, show application system |
| 18:16.53 | ichilton | jbalcomb: at work I could make calls but nothing starting with # |
| 18:16.53 | mafkees | rbd: asterisk does not get the script at all |
| 18:17.01 | ichilton | jbalcomb: it already had the sip firmware on when I bought it |
| 18:17.01 | GerbilWrk | CtRiX, cool, thanks |
| 18:17.09 | jbalcomb | ichilton: ok |
| 18:17.23 | Crescendo | Sep 26 14:16:45 NOTICE[4753]: chan_iax2.c:6902 socket_read: Rejected connect attempt from 192.168.0.87, who was trying to reach '1@Phone1' --- what does this mean, and how do I fix it? |
| 18:17.43 | jbalcomb | ichilton: oh, i though you meant # as in number not the key. i'm not sure about that. |
| 18:18.01 | jbalcomb | ichilton: off the rip though I'd say that little x is your uppermost issue |
| 18:18.36 | jtexter3 | tRSS: Okay, so far, so good. What's the issue? |
| 18:18.40 | ichilton | jbalcomb: ok |
| 18:18.56 | ichilton | jbalcomb: just trying to get it to connect at home and see if I have the problem with an already working asterisk box instead of a fresh one |
| 18:19.04 | jtexter3 | ManxPower: Could you fill me in on the solution you have tRSS yesterday? |
| 18:20.03 | tRSS | jtexter3: in the current situation, butters and chef are unable to dial any us-it people (i.e. cartmen and tweak) |
| 18:20.04 | ichilton | jbalcomb: can these phones do stun? |
| 18:20.12 | *** join/#asterisk clive- (n=pirch@dsl-145-28-211.telkomadsl.co.za) |
| 18:20.33 | clive- | hi, hows the 1.4 beta testing going? |
| 18:20.46 | jbalcomb | ichilton: have not looked into it |
| 18:20.51 | ichilton | jbalcomb: ok, ta |
| 18:21.05 | jbalcomb | ichilton: a quick search in the PDF manual will tell ya of course |
| 18:21.16 | ichilton | jbalcomb: do you have that or a url handy? |
| 18:21.36 | jbalcomb | ichilton: nah, just google |
| 18:21.40 | ichilton | k |
| 18:21.45 | ichilton | jbalcomb: you use yours in asterisk ok? |
| 18:22.00 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 18:22.07 | jbalcomb | ichilton: i only have two. i set them up four months ago and haven't touched 'em since. |
| 18:22.16 | ichilton | in asterisk? |
| 18:22.20 | jbalcomb | ichilton: yeah, they work great. |
| 18:22.20 | IOscanner | Does anyone have a openser.cfg example that has both NAThelper adn PSTN to Asterisk box? |
| 18:22.24 | ichilton | ok, cool |
| 18:22.25 | smackus | is there a way to call a CLI command from within extensions.conf? for example, I want to have an extension run the command Agent Loggof Agent/1002 |
| 18:22.29 | jbalcomb | ichilton: /with/ asterisk. |
| 18:22.45 | ichilton | jbalcomb: do you know where you can download the latest sip firmware? |
| 18:23.01 | ichilton | jbalcomb: I gather you need an account with cisco to get it from them? |
| 18:23.05 | Crescendo | Sep 26 14:16:45 NOTICE[4753]: chan_iax2.c:6902 socket_read: Rejected connect attempt from 192.168.0.87, who was trying to reach '1@Phone1' --- what does this mean, and how do I fix it? |
| 18:23.08 | jbalcomb | ichilton: you can get it from Cisco if you have a contract, otherwise it's off limits. |
| 18:23.24 | ichilton | jbalcomb: ah, ok :( |
| 18:23.48 | tRSS | jtexter3: any luck, so far? |
| 18:24.42 | jbalcomb | Crescendo: I think it means that 192.168.0.87 rejected the servers attempt to make an IAX connection to extension 1 in the phone1 context. |
| 18:24.43 | jtexter3 | tRSS: Looking at it, you are trying to prevent finance from calling outside the office? |
| 18:25.08 | tRSS | jtexter3: exactly, while still allowing them to be able to speak to people in us-it |
| 18:25.31 | jtexter3 | tRSS: In which case, I would just create another context for people who can dial internal and local |
| 18:25.35 | tRSS | and vice versa |
| 18:25.47 | ichilton | jbalcomb: argh, at home every time I go into network configuration it just reboots and does all it's configuring stuff again.... |
| 18:26.14 | tRSS | jtexter3: have you made any changes to the pb file? |
| 18:26.14 | Crescendo | jbalcomb, it appears as if the server rejected 87, from where I'm standing. I also get a "Sep 26 14:26:09 NOTICE[4753]: chan_iax2.c:5138 register_verify: No registration for peer 'phone1' (from 192.168.0.87)" |
| 18:28.14 | jbalcomb | Crescendo: ah, yes, it is reverse as you say. so 'sip show peer phone1' maybe? |
| 18:28.27 | jtexter3 | tRSS: just submitted it |
| 18:28.44 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.242) |
| 18:29.00 | tRSS | jtexter3: checking it now |
| 18:30.18 | Crescendo | jbalcomb, it does exist, but I think the context is set differently? |
| 18:30.29 | Crescendo | Is "default" an actual value, or just a placeholder? |
| 18:31.03 | jbalcomb | Crescendo: as in the context 'default'? it only real if its there [default] |
| 18:31.15 | jbalcomb | Crescendo: 'sip show peer 1' |
| 18:31.37 | Crescendo | Peer 1 not found. |
| 18:34.58 | pjz | anyone seen: app_meetme.c:667 conf_run: Error getting conference |
| 18:35.01 | pjz | ? |
| 18:35.12 | pjz | I can't seem to transfer two different people into a meetme conference. |
| 18:35.15 | syzygyBSD | maybe... |
| 18:35.27 | pjz | if I try, I get that, and the 2nd person gets dropped instead of transferred |
| 18:35.52 | syzygyBSD | do you have dynamically create conference? and keep conference open? |
| 18:35.58 | tRSS | jtexter3; the problem is still there. How will chef be able to dial cartman? internal context says nothing about the context [us-it] |
| 18:36.49 | wunderkin | he has to make sweet love first |
| 18:36.54 | syzygyBSD | well chef is off the show now so ti doesnt' matter |
| 18:37.01 | *** join/#asterisk Cinen (n=Cinen@208.70.20.3) |
| 18:37.03 | pjz | syzygyBSD: hrm? I have conf => 410 in my meetme.conf. The first person goes in fine. Any number of people can dial into it by dialing. But the 2nd person to transfer in gets hung up on instead. |
| 18:37.18 | wunderkin | :( |
| 18:37.30 | syzygyBSD | pjz: hmm, never done it that way... |
| 18:38.28 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 18:38.29 | smackus | here is my dilemma, I have agents logged in using AgentLogin who for whatever reason are getting locked into the queue, even though they have hung up the phone, which is supposed to log them out of the queue... who are not actually logged out of the queue. So I have to go to the CLI and issue a Agent Logoff Agent/blah to get them out. can anyone recommend a better way? if i am not around I cannot issue that command. |
| 18:39.29 | smackus | I was hoping there was some way to assign an extension to a command that could log them out, but I am not finding the appropriate command. Remove Queue Member works, but it completely messes them up. So now they log in using AgentLogin, but they are not added to the queue |
| 18:39.55 | smackus | Is there no way to issue a cli command from within the dial plan? |
| 18:39.59 | psk | hi all! i've 2 TE210P boards on my * pbx, but i'm not able to configure zapata.conf for the second one. Is there anyone who can help me? |
| 18:40.38 | [TK]D-Fender | smackus: System(/usr/sbin/asterisk -rx "restart now") |
| 18:41.09 | smackus | I cannot restart asterisk to do this... other people still using the system. |
| 18:41.50 | smackus | agent logoff from the cli does exactly what i want it to, i just have to be available to do it. which means 24/7 |
| 18:42.04 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
| 18:42.06 | hmmhesays | ok, on a reinvite... does asterisk issue the new invite or does the originating endpoint |
| 18:42.44 | tRSS | jtexter3: the problem is still there! |
| 18:45.44 | jbalcomb | MySQL replication server went down. I'm off. |
| 18:46.04 | sx-wks | wierd... the FWD 612 clock doesn't do anything |
| 18:46.10 | sx-wks | I get silence |
| 18:48.46 | kristalino | i'd like to replace my phone by an asterisk server. What is the hardware i need for that (for my home) ? |
| 18:49.56 | re-pete | what the he|| is exit status 1 ??? |
| 18:50.35 | *** join/#asterisk KranZ (n=user@sme.bestline.net) |
| 18:50.36 | *** part/#asterisk KranZ (n=user@sme.bestline.net) |
| 18:50.38 | *** join/#asterisk KranZ (n=user@sme.bestline.net) |
| 18:51.11 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 18:55.22 | Crescendo | Any suggestions? |
| 18:55.25 | jtexter3 | tRSS: Is there a reason Chef and Butters can't be in the internal context |
| 18:55.55 | wunderkin | heh |
| 18:56.04 | wunderkin | <3 butters |
| 18:56.06 | *** join/#asterisk kannan (i=1000@61.8.147.164) |
| 18:56.20 | justinu|laptop | butters scotch? |
| 18:56.38 | wunderkin | that's me! heh |
| 18:57.03 | rob0 | kristalino: something with one FXO and one FXS interface. Cheapest solution would be something like a Linksys/Sipura SPA-3000. |
| 18:57.34 | rob0 | kristalino: You plug your phone line into the FXO port, and telephone into the FXS port. |
| 18:59.39 | TripleFFFF | hey |
| 18:59.51 | TripleFFFF | any reason whhy qviewer.pl only returns first queue info ? |
| 19:00.55 | tRSS | jtexter3: internal contexts have dial patterns for other sites |
| 19:01.22 | Juggie | does anyone know of a active branch which has SRTP support? |
| 19:02.00 | tRSS | jtexter3: reason for having different contexts (i.e. us-it, finance, etc.) is that they all have certain restrictions on them to dial out and they have their custom voicemail IVR. but they should still be able to dial one and another |
| 19:03.02 | Juggie | tRSS, are the clients on sip phones? |
| 19:03.10 | clive- | juggie, I think 1.4 does |
| 19:03.24 | Juggie | clive-, 1.4 doesnt have it included by default that i know of. |
| 19:03.44 | tRSS | Juggie: yes, almost all of them, have sip phone (soft- and hard-phones) |
| 19:03.58 | Juggie | so whats the problem :) |
| 19:04.08 | Juggie | group them into certain context's based on their area |
| 19:04.30 | tRSS | Juggie, have you looked at the pastebin? |
| 19:04.36 | Juggie | no, whats the link |
| 19:04.40 | Juggie | i'm joining the conversation late. |
| 19:04.44 | twisted[work] | late |
| 19:05.28 | Juggie | tRSS, link? |
| 19:05.34 | tRSS | Juggie: no problem, here you go: this is what I have: http://pastebin.ca/183205, and this is what jtexter3 is suggesting: http://pastebin.ca/183223 |
| 19:06.19 | *** join/#asterisk IAmPostman (n=postman@adsl-209-30-166-99.dsl.rcsntx.swbell.net) |
| 19:06.35 | Juggie | tRSS, with your pastebin, whats not working. |
| 19:07.08 | IAmPostman | im a newbie, can someone point me to the right direction? |
| 19:07.10 | tRSS | Juggie: with my pastebin, finance people are unable to dial us-it people |
| 19:07.37 | Juggie | tRSS, what extensions would us-it be. |
| 19:07.59 | Juggie | 8234 & 8235? |
| 19:08.18 | IAmPostman | anybd pls?? |
| 19:08.18 | tRSS | IAmPostman: go left, then turn right, take 5 steps to the left again, turn around, make a u turn, go back straight, and then stop at the stop light for 5 mins. go straight, then make a sharp left turn. this is where you need to be! :) |
| 19:08.30 | pingwin[work] | i can't get "press 1 for blah" style function from a macro to work. any hints? |
| 19:08.36 | tRSS | Juggie: 8234 & 8235 |
| 19:08.59 | Juggie | well, thats because no where does finance have the us-it context included. |
| 19:09.03 | *** join/#asterisk kink0 (n=k@161.pool62-37-205.static.uni2.es) |
| 19:09.19 | Juggie | and of course you woudnt want to do that |
| 19:09.24 | Juggie | lets see what jtexter suggested |
| 19:09.36 | tRSS | Juggie: exactly, but if I do an include in finance, then finance people are able to dial international and local trunks (thru us-it) |
| 19:09.41 | IAmPostman | tRSS: cmon' man ppl come here for help not made fun off.. |
| 19:09.46 | kink0 | hello, anyone knows why my asterisk dead ( still running, but not calls ) when I got: |
| 19:10.03 | kink0 | chan_zap.c:8274 pri_dchannel: Restart requested on odd/unavailable channel |
| 19:10.13 | tRSS | IAmPostman: that was a very sincere joke. What is the problem you are facing? |
| 19:10.16 | Juggie | tRSS, this is obviiously a fake dialplan right |
| 19:10.25 | Juggie | eg @somesip and @someothersip |
| 19:10.48 | tRSS | Juggie: it is. but the essence is there. the actual dialplan is much more complicated and huge |
| 19:11.06 | Juggie | ok |
| 19:11.10 | Juggie | i will rewrite, standby. |
| 19:11.19 | tRSS | standing by ..... ;) |
| 19:11.28 | rob0 | The Right Direction: Wiki |
| 19:12.00 | Juggie | tRSS, what uses default? |
| 19:12.02 | *** join/#asterisk Aces1UP (n=Aces1UP@209.101.89.82) |
| 19:12.36 | IAmPostman | i hv 2 locations and want to be able to make longdistance call by dialing in. If I can do away using DSL and not getting any IP provider.. |
| 19:12.38 | Aces1UP | i have a few questions about asterisk and gsm gateways, does someone here have experience using gsm gateways with asterisk that can help? |
| 19:12.45 | tRSS | default is basically default. I believe I would be phasing this out. i will remove it soon, I guess |
| 19:13.02 | kink0 | Aces1UP, I am ussing Asterisk with Stargate's |
| 19:13.11 | IAmPostman | tRSS: if this is possible, can u tell what i need and should do.. thans |
| 19:13.36 | Juggie | so, what should finance get access to? |
| 19:14.31 | tRSS | Juggie: Just to internals (i.e. 6XXX & 5XXX) and the 8XXX extensions. they should not have access to PSTN, Long distance, international, etc. |
| 19:15.32 | Juggie | what context handles incomming calls |
| 19:15.39 | Juggie | does this pbx have pstn? incomming iax, sip? |
| 19:15.59 | tRSS | IAmPostman: I am not exactly aware of your situation, may be a bit more details would be helpful |
| 19:16.13 | blaylock | Juggie, default can handle incomming calls |
| 19:16.22 | IAmPostman | can any1 comment pls!!! --- i hv 2 locations and want to be able to make longdistance call by dialing in. If I can do away using DSL and not getting any IP provider.. what sort of cards i need and setups i have to do? |
| 19:16.54 | tRSS | Juggie: this pbx works with other pbxs, it doesn't have direct termination in it of PSTN. Reason for using this pbx, is that we can put restrictions on per-user basis. |
| 19:16.56 | Juggie | blaylock, i know. |
| 19:17.09 | Juggie | tRSS, does it receive any incomming sip calls? |
| 19:17.14 | blaylock | Juggie, then why you ask? |
| 19:17.14 | Juggie | or is it all outbound. |
| 19:17.24 | Juggie | because thats not necessairy what hes doing |
| 19:17.31 | Juggie | default doesnt HAVE to take incomming calls. |
| 19:17.39 | Juggie | nor is that necessairly how he has it configured |
| 19:17.42 | tRSS | Juggie: it does recieve incoming sip calls as well. e.g. someone from 6500 (6XXX internals) might call 8324. |
| 19:17.44 | pingwin[work] | is it possible to dial a different context without forwarding to a specific extension? as in just transfering to a different context? |
| 19:17.52 | Juggie | tRSS, ok. |
| 19:18.32 | blaylock | pingwin[work], i think you can use Goto() for that |
| 19:19.02 | Crescendo | Any suggestions? |
| 19:19.05 | *** join/#asterisk Cherebrum (n=jgarland@207.210.228.172) |
| 19:19.11 | tRSS | IAmPostman: have you checked the wiki |
| 19:19.22 | Cherebrum | I think I may have some timing slips on my digium T1 card. How can I check this? |
| 19:19.29 | IAmPostman | wiki:: where can i see that? |
| 19:19.29 | tRSS | wiki is always a good place to start. it has some really nice & short how tos |
| 19:19.41 | blaylock | pingwin[work], so for example exten => 100,1,GoTo(diffcontext,s,1) |
| 19:19.49 | tRSS | voip-info.org (i hope putting urls is not an offense here) |
| 19:19.50 | Juggie | trss, http://pastebin.ca/183269 |
| 19:19.58 | tRSS | Juggie: checking, thanks. |
| 19:20.10 | justinu|laptop | Cherebrum: afaik, zaptel cards don't report slips |
| 19:20.37 | KranZ | Cherebrum: do you get HDLC Abort 6 errors? |
| 19:20.37 | IAmPostman | tRSS: yes i did.. but somehow.. can't find the setup im looking for |
| 19:20.37 | Juggie | is that clear? |
| 19:21.31 | *** join/#asterisk joako (n=joako@64.238.175.93) |
| 19:21.50 | IAmPostman | what i wanted it to be able to make longdistance call between 2 asterisk boxes at different location without using any voip provider. and alsno not using any SIP phones.. but preferably dialing in |
| 19:22.04 | IAmPostman | any taker?/ |
| 19:22.13 | joako | How can I get more than 1 SIP device behind the same NAT to register with Asterisk?? |
| 19:22.25 | Cherebrum | no errors |
| 19:22.25 | Cherebrum | just users complaining about dropped calls |
| 19:22.50 | Cherebrum | I am seeing this: http://pastebin.ca/182184 |
| 19:22.58 | tRSS | Juggie: just give me a sec, i am still going thru |
| 19:23.02 | Cherebrum | But it doesn't happen all the time |
| 19:23.06 | CunningPike | IAmPostman: Look at IAX trunking between your two servers |
| 19:23.06 | Juggie | k |
| 19:24.15 | IAmPostman | CunningPike: do i need TDM400P in order to make my calls (dialing in).. |
| 19:24.38 | tRSS | Juggie: what you have left as an exercise for me, is, I guess the real problem I am facing. in the current setup, you have put all the extensions, in one big context,which is what I am trying to avoid. reasons being, finance has a different voicemail ivr then the us-it. |
| 19:24.48 | CunningPike | IAmPostman: Your route is PSTN -> Asterisk -> IAX -> Asterisk -> PSTN, correct? |
| 19:24.50 | IAmPostman | is IAX responsible for routing my calls to each boxes? |
| 19:25.17 | IAmPostman | CunningPike: correct |
| 19:25.18 | Juggie | tRSS, then you can just create local-extension-finance etc |
| 19:25.59 | CunningPike | IAmPostman: Then each Asterisk server needs a way to connect to the PSTN - how depends on your concurrent call volume - how many concurrent calls are you looking at? |
| 19:26.17 | Juggie | tRSS, a couple of macros would also be very useful. |
| 19:26.28 | IAmPostman | CunningPike: less than 5 |
| 19:26.38 | Juggie | actually, you could do it with one. |
| 19:26.44 | tRSS | Juggie: can I IM you pvt. |
| 19:26.51 | IAmPostman | CunningPike: what crads do i need?? |
| 19:26.57 | Juggie | ok i guess :P |
| 19:26.59 | CunningPike | IAmPostman: So you have 5 POTS lines? |
| 19:28.00 | IAmPostman | CunningPike: i dont know how im gonna do it yet.. but we hv traditional pbx here, and 2 extra POTS.. im thinking of just using 1 POTS |
| 19:37.29 | IAmPostman | CunningPike: if u hve a 4 ports card, do u need to hv 5 landlines? |
| 19:37.29 | CunningPike | IAmPostman: 1 POTS => 1 concurrent call, so for that you can simply use an FXO gateway - an SPA-3000 or similar |
| 19:37.29 | zparta | anyone got HUDlite server running on freebsd? |
| 19:37.30 | CunningPike | IAmPostman: For more lines, get an Audiocodes FXO gateway or similar |
| 19:37.30 | IAmPostman | CunningPike: FXO gateway.. is this like the TDM400P digium cards? |
| 19:37.30 | CunningPike | IAmPostman: I would steer away from cards if I were you, and get an external gateway |
| 19:37.30 | IAmPostman | CunningPike: ok, what can u tell me about those third party X100P cards?? |
| 19:37.30 | CunningPike | IAmPostman: They're shite |
| 19:37.30 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 19:37.30 | IAmPostman | CunningPike: are they low on quality? |
| 19:37.30 | clive- | iampostman they dont work too great on xeons |
| 19:37.30 | CunningPike | IAmPostman: They are barely adequate as a timing source - I would never use one for making calls |
| 19:37.30 | IAmPostman | CunningPike:ok, so the spa 3000,, how does it connect to pc? |
| 19:37.30 | clive- | actually not at all on xeons |
| 19:37.31 | *** join/#asterisk sudhir492 (n=sudhir@leesburg-bsr3-68-65-168-202.chvlva.adelphia.net) |
| 19:37.31 | CunningPike | IAmPostman: It connects to the LAN via Ethernet and registers with the Asterisk server as a SIP UA. It has an FXO port to connect to the PSTN, and an FXS port if you want to connect an analog handset |
| 19:37.31 | IAmPostman | CunningPike: ohh ok... if i hv that on both sides... can u give me a background how to setup IAX |
| 19:37.31 | *** join/#asterisk Seba_soy (n=s@64.76.126.29) |
| 19:37.31 | Seba_soy | hello |
| 19:37.31 | clive- | cunningpike you still need a timming source... |
| 19:37.31 | Seba_soy | i have a question |
| 19:37.31 | CunningPike | IAmPostman: There is a good article in the wiki |
| 19:37.32 | clive- | unless you want to rely on ztdummy |
| 19:37.32 | CunningPike | ~wiki |
| 19:37.33 | CunningPike | clive-: Yup - use ztdummy - should be fine for a handful of calls, especially on 2.6 kernel |
| 19:37.33 | *** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 19:37.33 | Seba_soy | I have 2 asterisk box, first one register against 2nd... but when I try to place calls from 2nd to 1st it give me FORBIDDEN |
| 19:37.33 | CunningPike | ~thewiki |
| 19:37.34 | jbot | methinks thewiki is at http://www.voip-info.org/wiki-Asterisk |
| 19:37.34 | IAmPostman | CunningPike: ok ill check that.. but is IAX kinda like a replacement for VOIP providers? |
| 19:37.35 | Seba_soy | how have I configure it? |
| 19:37.35 | CunningPike | ~iax |
| 19:37.36 | jbot | extra, extra, read all about it, iax is port 5036 for the original (deprecated) IAX protocol. Port 4569 is for the the current IAX2 protocol. IAX is pronounced "Eeks". stands for Inter-Asterisk Exchange |
| 19:37.36 | clive- | cunningpike in my expereince ztdummy dies under heavier load |
| 19:37.36 | CunningPike | Goddam it, jbot, wakey wakey |
| 19:37.36 | CunningPike | clive-: Agreed, but for a handful of concurrent calls, it should be fine |
| 19:37.36 | clive- | 100% |
| 19:37.37 | IAmPostman | CunningPike: when u say ~thewiki, would this be under voip-info.org?? |
| 19:37.37 | CunningPike | IAmPostman: Yes- I apologize for my friend jbot, but he has been drinking again |
| 19:37.37 | IAmPostman | CunningPike: haha.. |
| 19:37.37 | justinu|laptop | he'll get around to it |
| 19:37.37 | CunningPike | IAmPostman: Search for 'iax trunk' |
| 19:37.37 | Seba_soy | any clue? |
| 19:37.37 | sudhir492 | Hi All |
| 19:37.37 | Seba_soy | hi |
| 19:37.37 | CunningPike | Seba_soy: You could do worse than look at iax trunking, too |
| 19:37.37 | sudhir492 | Is there a good IP phone for under US$100 ? |
| 19:37.38 | *** join/#asterisk techie (n=techie@c-67-182-22-174.hsd1.ca.comcast.net) |
| 19:37.38 | IAmPostman | CunningPike: ok.. what about if i hv more lines would spa 3000 still works.. or do i need diffrent cards? |
| 19:37.38 | Seba_soy | I know, I like to try on SIP |
| 19:37.38 | Cherebrum | Could this be a bug in Asterisk 1.2.1: http://pastebin.ca/182184 |
| 19:37.38 | Cherebrum | ? |
| 19:37.38 | CunningPike | Seba_soy: Why? :) |
| 19:37.38 | Crescendo | I'm having the worst trouble setting up SIP. |
| 19:37.38 | Seba_soy | I know IAX works, I want to make SIP work... |
| 19:37.39 | joako | How can I get more than 1 SIP device behind the same NAT to register with Asterisk?? |
| 19:37.58 | CunningPike | jbot lives! |
| 19:38.00 | jbot | did you doubt that, cunningpike? |
| 19:38.17 | CunningPike | For a time there, jbot, I feared for your wellbeing |
| 19:38.17 | Seba_soy | CunningPike: did you read my problem? |
| 19:38.27 | IAmPostman | CunningPike: ok.. what about if i hv more lines would spa 3000 still works.. or do i need diffrent cards? |
| 19:38.54 | [TK]D-Fender | sudhir492: The closest is : http://www.telephonydepot.com/product_p/105-057-9112.htm |
| 19:38.58 | CunningPike | IAmPostman: For more than one line, look at an Audiocodes FXO gateway - same principle, more ports |
| 19:39.07 | *** join/#asterisk euphobot (n=don@adsl-64-170-69-114.dsl.lsan03.pacbell.net) |
| 19:39.47 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 19:39.54 | IAmPostman | CunningPike: with the specs and setup, u've given me.. would the quality be the same than skype or worse? |
| 19:40.14 | CunningPike | Cherebrum: I don't think so - it is simply an indication that an Information Element was sent that libpri doesn't know what to do with - does the message coincide with the call dropping |
| 19:40.26 | CunningPike | IAmPostman: Yes |
| 19:40.26 | IAmPostman | CunningPike: currently im using cuphone Personal phone gateway, basically just forwarding the call from skype to PSTN, but lately quality is unstable.. |
| 19:40.53 | Cherebrum | I will have to have the users document dropped calls and match it up with a timestamp of some sort because I'm not at that site... |
| 19:40.55 | CunningPike | IAmPostman: The quality will depend on the latency between your 2 asterisk servers, but it should be reasonable |
| 19:41.05 | IAmPostman | CunningPike: ic.. |
| 19:41.07 | CunningPike | Cherebrum: Yup, you will |
| 19:41.14 | CunningPike | ~wglwat |
| 19:41.16 | jbot | methinks wglwat is well, good luck with all that |
| 19:41.20 | CunningPike | :) |
| 19:41.39 | IAmPostman | CunningPike: any background how will I be able to connect/upgrade/ integrate our old pbx system.? |
| 19:41.48 | CunningPike | IAmPostman: What is it? |
| 19:41.54 | *** join/#asterisk oQPa (i=Spain@178.Red-83-40-182.dynamicIP.rima-tde.net) |
| 19:42.30 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 19:42.30 | *** mode/#asterisk [+o mog] by ChanServ |
| 19:42.56 | Seba_soy | i am having another problem, I have an asterisk with user 777 defined, and a call comes from a PBX-IP with 777 as from user, then asterisk reject the call because wrong password... |
| 19:42.59 | *** join/#asterisk mr_canny (n=mr_canny@200.138.113.82) |
| 19:43.06 | IAmPostman | CunningPike: we basically hv one main number and it rolls for open trunk line.. |
| 19:43.18 | Seba_soy | i am accepting the call swith insecure=invite,port to match against ip address |
| 19:43.31 | CunningPike | IAmPostman: What type of PBX is it? |
| 19:43.35 | IAmPostman | CunningPike: any suggestion i can integrate or upgrade this using asterisk? |
| 19:43.45 | IAmPostman | merlin |
| 19:43.49 | IAmPostman | CunningPike: merlin |
| 19:44.04 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 19:44.04 | Cherebrum | I'm going to upgrade to the latest 1.2 release for now and see if the problem goes away. |
| 19:44.10 | CunningPike | IAmPostman: You want to keep it? Or replace it |
| 19:44.21 | CunningPike | Cherebrum: What are you running now? |
| 19:44.22 | mr_canny | Hi folks I search libnewt for slackware but not found |
| 19:44.27 | Cherebrum | CunningPike: 1.2.1 |
| 19:44.36 | *** part/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 19:44.36 | IAmPostman | CunningPike: improve it I guess, if it nots too much money to do so |
| 19:45.00 | CunningPike | IAmPostman: How does it connect to the PSTN? PRI? |
| 19:45.18 | mr_canny | where I found this libnewt for slackware ? |
| 19:45.23 | *** join/#asterisk Mathesis (n=chatzill@unaffiliated/mathesis) |
| 19:45.30 | IAmPostman | CunningPike: yes |
| 19:46.15 | CunningPike | IAmPostman: Well, ymmv, but you could replace it with asterisk with either a PRI card or a redphone PRI bridge |
| 19:46.18 | Seba_soy | does asterisk authenticate call when it comes matching from IP ADDRESS ? |
| 19:46.29 | Seba_soy | I think it is authenticating with from-user |
| 19:46.32 | CunningPike | Seba_soy: Pastebin your sip.conf |
| 19:46.35 | CunningPike | ~pb |
| 19:46.37 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 19:46.46 | Seba_soy | let me explain config.... |
| 19:46.52 | CunningPike | Nice to see you've sobered up, jbot |
| 19:46.56 | Seba_soy | I have a pbx-ip NOT asterisk |
| 19:47.01 | Seba_soy | and I have an asterisk box |
| 19:47.03 | mr_canny | please help me !! |
| 19:47.27 | CunningPike | mr_canny: Ask a sensible question |
| 19:47.33 | CunningPike | ~suggestions |
| 19:47.34 | jbot | suggestions is probably 1) Don't ask to ask. Just say your problem, 2) Don't repeat until 5 mins after, 3) Read and re-read the docs first, then admit it if you REALLY don't understand. You're wasting your time and ours if you haven't at least tried. 4) If your problem ain't solved, come back in 12 hrs or 24 hrs later. We're very international. 5) Be polite and ... |
| 19:48.03 | IAmPostman | CunningPike: You've been a great help!! Thank you sir!! |
| 19:48.09 | CunningPike | IAmPostman: ytw |
| 19:48.09 | *** join/#asterisk Qb3rt (n=jhgjkgui@kyle.colba.net) |
| 19:48.22 | mr_canny | CunningPike, I need libnewt for slackware ...I search in google but not found |
| 19:48.45 | CunningPike | mr_canny: Serves you right for using slackware ;) |
| 19:48.45 | mr_canny | CunningPike, zttool requires libnewt |
| 19:48.57 | Qb3rt | my company is currently looking for voip asterisk consultant in montreal... Thanks! |
| 19:49.15 | sudhir492 | D-Fender: Have you used aastra 9112i phones? Is the quality good? |
| 19:49.20 | CunningPike | Qb3rt: Try the asterisk-biz mailing list |
| 19:49.23 | Seba_soy | on asterisk I have defined [7777] type=friend, i have definde [pbx] type=friend, insecure=very to accept calls from pbx |
| 19:49.40 | Seba_soy | call comes from pbx like this fromuser: 7777@ip-pbx |
| 19:50.00 | Seba_soy | then asterisk tries to mach 7777 with his internal user and return forbidden |
| 19:50.06 | mr_canny | CunningPike, what linux you use with asterisk ? |
| 19:50.12 | Seba_soy | it is like the pbx cant have same extensions than asterisk |
| 19:50.33 | CunningPike | mr_canny: RHEL |
| 19:51.17 | mr_canny | sniff sniff sniff |
| 19:52.28 | wunderkin | allergy problems? |
| 19:53.26 | Crescendo | Alright, I've had a lot of problems getting SIP off the ground. Where is definitive guide, or can someone actually help me? |
| 19:53.57 | *** join/#asterisk Didour (n=didour@caracas-4668.adsl.interware.hu) |
| 19:54.00 | Didour | Hi! |
| 19:54.05 | [TK]D-Fender | sudhir492: I haven't used the 9112i personally, but I hear its decent and I've worked with the 480i which is actually a pretty nice phone. |
| 19:54.30 | *** join/#asterisk lukketto (n=lukketto@host43-106-dynamic.59-82-r.retail.telecomitalia.it) |
| 19:54.36 | [TK]D-Fender | mr_canny: If you did a complete install of Slackware you would have libnewt...... |
| 19:55.13 | mr_canny | [TK]D-Fender, I have instaled full |
| 19:55.38 | *** part/#asterisk clive- (n=pirch@dsl-145-28-211.telkomadsl.co.za) |
| 19:56.11 | IAmPostman | CunningPike: with PSTN -> Asterisk -> IAX -> Asterisk -> PSTN, i dont need a voip provider right?? |
| 19:56.33 | [TK]D-Fender | mr_canny: I use Slackware all the time and if you do "everything" you won't be missing anything... |
| 19:57.09 | mr_canny | [TK]D-Fender, I use slackware 10.2 and you ? |
| 19:58.02 | *** join/#asterisk ajedwards (n=justacha@unaffiliated/ajedwards) |
| 19:58.29 | Didour | Sorry, I don't know the rules, may I put a question to You here? |
| 19:58.54 | kristalino | rob0, ok. Is Linksys/Sipura SPA-3000 easy to find. And does this interface have a phone rj11 input ? |
| 20:00.22 | Crescendo | Alright, I've had a lot of problems getting SIP off the ground. Where is definitive guide, or can someone actually help me? |
| 20:00.59 | rob0 | kristalino: FXO and FXS ports are RJ-11. Different function, same form. And yes, lots of vendors sell them. I mentioned that as only one possibility, minimum needs. You can also get zaptel hardware from Digium or competitors. |
| 20:01.36 | [TK]D-Fender | mr_canny: Same |
| 20:02.50 | mr_canny | [TK]D-Fender, I search in cd of slackware |
| 20:03.21 | rob0 | Libnewt is fairly new in Slackware, what version is this? |
| 20:03.41 | kristalino | rob0, yeap, that's exactly what i need : minimum need and minimum price. Any usb hardware (i'd like to use asterisk on a laptop) ? |
| 20:04.02 | *** mode/#asterisk [+b *!*=bkw_@*] by Corydon-w |
| 20:04.09 | *** kick/#asterisk [bkw_!n=tilghman@pdpc/supporter/sustaining/Corydon76-home] by Corydon-w (Enough) |
| 20:04.21 | Seba_soy | I read on voip-info that sip match first a "user" using field "fromuser: xxxx@ip-pbx" |
| 20:04.35 | Seba_soy | what if ip-pbx have same extensions that my Asterisk? |
| 20:04.51 | rob0 | Um, wait, I don't think libnewt *is* in Slackware. |
| 20:05.02 | Seba_soy | for example, I have extension 2000 on my asterisk and a call comes in from extension 2000 from ip-pbx |
| 20:05.42 | Seba_soy | how asterisk match it? |
| 20:05.45 | mr_canny | someone have libnewt in slackware... please send me.... |
| 20:06.31 | joako | How can I get more than 1 SIP device behind the same NAT to register with Asterisk?? |
| 20:07.14 | mog | Corydon-w, what was taht about? |
| 20:07.15 | *** join/#asterisk MikeJ (n=mikej@d14-69-8-30.try.wideopenwest.com) |
| 20:07.20 | rob0 | mr_canny: "./configure && make && sudo make install" is your friend |
| 20:07.38 | Corydon-w | mog: There's some history |
| 20:07.42 | rob0 | mr_canny: checkinstall in extra/ is nice too. |
| 20:08.21 | Corydon-w | mog: he's been told too many times not to troll, so he get kicked at the slightest attempt |
| 20:08.36 | *** join/#asterisk krylen (n=krylen@70.91.221.169) |
| 20:08.43 | rob0 | (I thought bkw_ was an op here.) |
| 20:08.53 | mr_canny | rob0, thanks |
| 20:08.59 | Corydon-w | Was, being the key word |
| 20:09.08 | rob0 | ah, when did this happen? |
| 20:09.18 | Didour | Is anybody experienced, that the g729 license get stucked if use mixmonitor, or I don't know something? |
| 20:09.40 | krylen | hey guys, i've got an extension that always says "user at blah not availible" no matter what i do, i even removed the extension, and all it's references and re-added them |
| 20:10.07 | krylen | i should say, it goes right to voicemail |
| 20:10.11 | Corydon-w | rob0: back when bkw_ decided that a fork was a good idea |
| 20:10.13 | Seba_soy | anyclue? |
| 20:10.17 | *** join/#asterisk dhahn (n=dhahn@ip-216-17-139-63.rev.frii.com) |
| 20:10.54 | Corydon-w | rob0: it's been close to a year now |
| 20:11.12 | dhahn | Has anyone had any success using the Manager and the AGI to originate calls? |
| 20:12.06 | dhahn | I seem to only be able to get orignate failed with no route to destination |
| 20:12.17 | dhahn | bkruse: hello |
| 20:12.21 | Corydon-w | dhahn: what channel are you using? |
| 20:12.43 | dhahn | Corydon-w: An IAX2 channel. |
| 20:12.54 | Corydon-w | dhahn: specifically, what? |
| 20:13.20 | dhahn | Corydon-w: line reads - "Channel: IAX2/u23193293" |
| 20:13.59 | Corydon-w | So "u23193293" exists as an entry in /etc/asterisk/iax.conf ? |
| 20:15.25 | *** part/#asterisk krylen (n=krylen@70.91.221.169) |
| 20:15.31 | dhahn | Corydon-w: Yes. And I can make outbound calls from the CLI using that IAX2 channel |
| 20:15.31 | Seba_soy | I cant make calls to an asterisk if it calls comes from another pbx and uses the same extension |
| 20:16.05 | Corydon-w | dhahn: are you dialling a number after that? i.e. IAX2/u23193293/2345678 ? |
| 20:16.56 | MikeJ | Corydon-w, I think you are mistaken.. to my knowledge, bkw_ did no forking of asterisk |
| 20:17.07 | kristalino | rob0, yeap, that's exactly what i need : minimum price. Do you know any usb hardware that has similar functions as the spa 3000 (i'd like to use asterisk on a laptop) ? |
| 20:17.07 | jtexter3 | Is anyone here using, or providing, a hosted pbx that is Asterisk based? |
| 20:17.14 | dhahn | Corydon-w: No I'm not. |
| 20:17.24 | De_Mon | if I download royalty free music for free, are there any limitations on using it for music on hold? |
| 20:17.29 | Corydon-w | dhahn: is this an iaxy or... ? |
| 20:17.44 | ManxPower | De_Mon, royalty free music is not free. |
| 20:17.46 | mog | possibly De_Mon |
| 20:17.53 | mog | you are redistribbing it by playing it to people |
| 20:17.56 | ManxPower | you pay for it ONCE, rather than once per year |
| 20:17.57 | *** join/#asterisk robin_sz (n=robin@adsl.redpoint.org.uk) |
| 20:18.02 | mog | you would have to have a license to do so |
| 20:18.10 | dhahn | Corydon-w: trying to develop an application that uses the manager. |
| 20:18.11 | mog | the songs with asterisk you can use as much as you want |
| 20:18.19 | dhahn | Corydon-w:phpagi based |
| 20:18.34 | robin_sz | hmm .. outgoing audio on this X100P seems very quiet ... |
| 20:18.34 | Corydon-w | dhahn: but what's the IAX device that you're connecting to? |
| 20:18.44 | dhahn | Corydon-w:had problems getting it to work, so, I thought I would try just sending some commands to the Server |
| 20:18.57 | dhahn | Corydon-w: An outbound channel to an IAX provider |
| 20:19.00 | Cherebrum | jtexter3: I was using asterisk for hosted PBX but switched to OpenPBX because it could run in Xen. I was able to get 30 instances of OpenPBX running on the same box using Xen. So 30 customers per box. :) Only reason I couldn't do more is because each box only had 8 gigs of ram. |
| 20:19.06 | De_Mon | mog why would I be able to use music included with asterisk? |
| 20:19.13 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 20:19.15 | De_Mon | can I get more music included with asterisk ;) |
| 20:19.17 | Corydon-w | dhahn: ah. Yeah, I think you need to provide it a number to dial... |
| 20:19.21 | mog | yes |
| 20:19.25 | _alex_mx_ | anyone know how reliable zttool's report on irq misses is and if it's realtime or it's a running total since...(last module load, restart, what) |
| 20:19.30 | ManxPower | De_Mon, because Digium has special permission |
| 20:19.31 | robin_sz | presumably there is some way of winding 60 or 10db of gain into a zap outgoing? |
| 20:19.33 | Corydon-w | dhahn: so the format is IAX2/peername/number-to-dial |
| 20:20.06 | dhahn | Corydon-w:in the call manager speak, the extension is on a different line, so, I assumed it would go there. I'll test with the number on the line. |
| 20:20.20 | ManxPower | robin_sz, txgain |
| 20:20.26 | Corydon-w | dhahn: when you just dial IAX2/peername, you're not giving it a destination, just a node name, and the peer is responding with that error message |
| 20:20.30 | ManxPower | I can't imagine why you would want 60db |
| 20:20.31 | mog | but you have to have right to do so by the person who owns the recording |
| 20:20.33 | mog | we got permission from freeplay music to give everyone one who uses asterisk access to those three files we distrib |
| 20:20.34 | Cherebrum | jtexter3: You could do multiple customers on the same asterisk instance but that would mean a NASTY mess of a dialplan and it's impossible to support |
| 20:20.36 | mog | most music you would have to pay for to use the way we do with asterisk |
| 20:20.38 | mog | but im sure you could find some free music somewhere |
| 20:20.43 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 20:36.21 | *** part/#asterisk lukketto (n=lukketto@host43-106-dynamic.59-82-r.retail.telecomitalia.it) |
| 20:36.21 | Corydon-w | dhahn: you can do that with some IAX2 destinations, but I'm pretty sure a provider won't let you do that |
| 20:36.22 | carrar | does 2B release work with Asterisk? |
| 20:36.22 | De_Mon | mog royalty free(beer) or free as in speech free |
| 20:36.22 | Seba_soy | why I cant make a call from an ip-pbx to an asterisk pbx if ip-pbx have same extensions that asterisk? |
| 20:36.22 | *** part/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 20:36.22 | Cherebrum | jtexter3: or maybe you could use fastagi and run all the dialplan stuff thru an AGI script |
| 20:36.22 | anthm | bkw forked asterisk? what's it called? bkwsterisk? |
| 20:36.22 | ManxPower | I thought bkw was INVOLVED with FreePBX, which I thought was a fork of Asterisk |
| 20:36.22 | De_Mon | ManxPower good to know, thanks |
| 20:36.22 | Cherebrum | ManxPower: FreePBX is a gui for asterisk I think |
| 20:36.22 | _alex_mx_ | openpbx, freepbx is a gui |
| 20:36.22 | Cherebrum | ManxPower: OpenPBX is the fork. |
| 20:36.22 | ManxPower | Cherebrum, I sit corrected |
| 20:36.22 | De_Mon | anthm speaks? cool |
| 20:36.22 | Seba_soy | any clue? |
| 20:36.23 | Cresl1n | carrar: with DMS100 switch type |
| 20:36.23 | *** join/#asterisk champster (n=asterisk@AH.tescogroup.com) |
| 20:36.23 | anthm | isnt freepbx a gui frontend to asterisk? |
| 20:36.23 | CtRiX | i think so |
| 20:36.23 | trelane_ | yes |
| 20:36.23 | trelane_ | due to dialplan fuglyness it is not supported here |
| 20:36.23 | robin_sz | => Set default values of rxgain/txgain to 1.0 |
| 20:36.24 | robin_sz | default should be zero, obviously |
| 20:36.24 | trelane_ | -.03 |
| 20:36.24 | trelane_ | it's ANALOG |
| 20:36.24 | trelane_ | not Digital |
| 20:36.24 | CtRiX | ANAL |
| 20:36.24 | trelane_ | 0 in a perfect world is clip |
| 20:36.24 | trelane_ | -3db |
| 20:36.24 | trelane_ | but rxgain/txgain aren't really in db |
| 20:36.25 | *** part/#asterisk smackus (n=ckwall@63.149.122.93) |
| 20:36.25 | pingwin[work] | is there a way to check original context of a call? ie i have a incoming call over zap, then this is shifted to my default context to be able to access extensions. but my internal extensions can call out back over zap. Obviously this needs to be prevented |
| 20:36.25 | robin_sz | well, I was assuming the docs were correct in saying they were in db |
| 20:36.25 | trelane_ | robin_sz, you sir are out of your mind, analog sound is always mixed 3db below clip, while digital sound is mixed at 0 |
| 20:36.25 | trelane_ | you mix right on the pin and your audio will clip |
| 20:36.25 | robin_sz | trelane, you are not understanding the difference between levels (measured in dbm) and gain (measured in db) |
| 20:36.25 | trelane_ | no because I've pushed +20 on bad lines and I know what a 20dB gain does |
| 20:36.25 | trelane_ | I understand full well the difference, here we set the gain to 0 |
| 20:36.25 | trelane_ | but you record analog at -3 |
| 20:36.26 | robin_sz | you still got it worng, sorry and all that. |
| 20:36.26 | robin_sz | what level will a gain of 0db give? |
| 20:36.26 | robin_sz | answer: depends on input level of course |
| 20:36.26 | jtexter3 | Cherebrum: How do the users connect? For example, if I'm a small business, do I just get a set of SIP phones that connect to your hosted service? Is that just using the WWW, or VPN? |
| 20:36.26 | De_Mon | pingwin[work] put the extensions they need to call in a separate context and INCLUDE that context in your 'incoming-zap' context |
| 20:36.26 | robin_sz | so, yes, the desired LEVEL may well be -3db, the GAIN setting to achieve that could be ANYTHING ... +20 even, if the input signal is at -23 |
| 20:36.26 | De_Mon | pingwin[work] and keep the outgoing calling ability in a separate context so you can say [local] includes [local-extensions] and [outoing-zap] |
| 20:36.26 | jtexter3 | Cherebrum: I've seen several people talk about it, just wondering how it works. Seems like an interesting idea |
| 20:36.26 | De_Mon | and incoming includes local-extensions and not outgoing-zap |
| 20:36.27 | pingwin[work] | ahhhh I think I get it, thanks alot De_Mon |
| 20:36.27 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 20:39.19 | Cherebrum | jtexter3: sorry I had a phone call... We didn't run a web interface of any sort. Just sip phones for the customer |
| 20:40.07 | Cherebrum | jtexter3: We also required them to have a SIP NAT traversal device. Edgewater Edgemark works well. Or use our session border controller for far end NAT traversal |
| 20:40.14 | TripleFFFF | wahts on port 2000 ? |
| 20:40.24 | Cherebrum | jtexter3: You can use openser with mediaproxy to do far end nat traversal |
| 20:40.55 | Cherebrum | Near end NAT traversal always works best tho |
| 20:41.02 | TripleFFFF | dundio ? |
| 20:41.03 | Cherebrum | so check out the Edgewater Edgemark products |
| 20:41.21 | TripleFFFF | hmm skinny |
| 20:42.18 | Seba_soy | somebody can explain me how can I make calls if I have same extensions on both pbx? |
| 20:42.24 | Cherebrum | SIP and NAT won't work well together... you might be able to get it to work with some of the hacks that asterisk has built in but it won't be reliable enough for some random business to connect over the interweb to |
| 20:42.33 | Seba_soy | I got "failed to authenticate user..." |
| 20:43.10 | *** join/#asterisk quid246 (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 20:43.17 | Cherebrum | jtexter3: And don't use Cisco phones.. They are certified garbage. ;) |
| 20:43.33 | TripleFFFF | ? cher .. myu 7960 is perfect whats up ? |
| 20:44.05 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 20:44.19 | *** join/#asterisk Heimidal (n=Heimidal@phpbb/styles/heimidal) |
| 20:44.30 | Cherebrum | TripleFFFF: The SIP stack is a bit lacking and it doesn't do well with outbound proxy servers.. IE: you tell it to you an outbound proxy server and it bypasses the outbound proxy and sends ACK messages directly |
| 20:44.34 | Heimidal | has anyone configured a dial plan for Cbeyond's SIPconnect |
| 20:44.51 | Cherebrum | Polcom phones are a much better choice IMHO |
| 20:44.55 | Cherebrum | er Polycom |
| 20:45.28 | Cherebrum | TripleFFFF: It also seems to break pretty easily when you send it OPTIONS for the SIP keepalive stuff |
| 20:45.38 | Cherebrum | TripleFFFF: IE: qualify=yes |
| 20:47.39 | robin_sz | thats better, echolearning=yes and 6db of gain has helped a lot |
| 20:48.24 | jbeez | and how do you add gain? |
| 20:48.40 | robin_sz | zapata.cong, rxgain=6.0 |
| 20:48.44 | robin_sz | cong? |
| 20:48.46 | robin_sz | conf. |
| 20:48.56 | *** part/#asterisk Cherebrum (n=jgarland@207.210.228.172) |
| 20:49.23 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-88-154-106-254.red.bezeqint.net) |
| 20:49.44 | robin_sz | and a quick inspection with ztmonitor |
| 20:50.48 | *** join/#asterisk Prelius (n=pzotov@70.88.213.25) |
| 20:51.46 | Seba_soy | any help :(:(:( |
| 20:52.18 | Prelius | folks, I wonder if I could ask for some help compiling zaptel 1.2.9 under Ubuntu 6.06.1 AMD 64... |
| 20:52.28 | *** join/#asterisk GaryH (n=GaryH@wallace.garysoft.co.uk) |
| 20:53.09 | Prelius | getting the following error: make[1]: Entering directory `/usr/src/linux-headers-2.6.15-23-amd64-generic' |
| 20:53.09 | Prelius | <PROTECTED> |
| 20:53.09 | Prelius | make[2]: *** No rule to make target `arch/x86_64/kernel/../../i386/kernel/cpuid.o', needed by `arch/x86_64/kernel/msr.o'. Stop. |
| 20:53.09 | Prelius | make[1]: *** [arch/x86_64/kernel] Error 2 |
| 20:53.09 | Prelius | make[1]: Leaving directory `/usr/src/linux-headers-2.6.15-23-amd64-generic' |
| 20:53.10 | Prelius | make: *** [linux26] Error 2 |
| 20:53.16 | jtexter3 | Cherebrum: I've been using Polycom, and so far have been very happy. |
| 20:53.30 | jtexter3 | Cherebrum: Did you give them minimum requirements for the amount of bandwidth they need? |
| 20:53.56 | ManxPower | robin_sz, uh, it's echotraining |
| 20:54.01 | ManxPower | echolearning does NOTHING |
| 20:54.07 | hmmhesays | Prelius: pastebin is your friend |
| 20:54.10 | stoffell | Prelius: use pastebin.com (or pastebin.ca) |
| 20:54.25 | robin_sz | ManxPower, my typo in irc I expect |
| 20:54.38 | ManxPower | robin_sz, that's why you should always PASTE |
| 20:54.48 | robin_sz | paste bad :) |
| 20:54.52 | jbalcomb | ~seen [TK]D-Fender |
| 20:56.01 | jbot | [tk]d-fender <n=joe@MTRLPQ02-1177745839.sdsl.bell.ca> was last seen on IRC in channel #asterisk, 54m 25s ago, saying: 'mr_canny: Same'. |
| 20:56.03 | robin_sz | pastebin! |
| 20:56.14 | ManxPower | pasting one or 2 lines is usually OK |
| 20:56.17 | hmmhesays | riker and his father are going at it |
| 20:56.20 | robin_sz | echotraining=yes |
| 20:56.21 | Prelius | ok, I don't know what pastebin is, but I guess I am about to find out.... |
| 20:56.22 | robin_sz | ~paste |
| 20:56.23 | jbot | from memory, paste is see http://paste.husk.org, or http://paste-it.net |
| 20:56.24 | stoffell | ~pastebin |
| 20:56.25 | jbot | somebody said pastebin was a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 20:56.27 | *** join/#asterisk Cherebrum (n=jgarland@207.210.228.172) |
| 20:56.27 | jbalcomb | ~wastebin |
| 20:56.30 | stoffell | lol robin_sz , guess someone has to type it all .. (or paste it ? :p) |
| 20:56.30 | Cherebrum | Hey guys.. digg my bluetooth Western Electric 500 article. ;) http://digg.com/mods/Bluetooth_Western_Electric_500_handset_mod |
| 20:56.30 | robin_sz | ~nopaste |
| 20:56.31 | jbot | from memory, nopaste is http://nopaste.snit.ch/ or http://pastebin.com/, or preferably http://pastebin.ca/ |
| 20:56.31 | robin_sz | bugger. :) |
| 20:56.31 | jbalcomb | i think jbot is a little busy |
| 20:56.32 | jtexter3 | Cherebrum: I've been using Polycom, and so far have been very happy. |
| 20:56.37 | jtexter3 | Cherebrum: Did you give them minimum requirements for the amount of bandwidth they need? |
| 20:56.46 | robin_sz | ahh, happy bot. |
| 20:57.08 | quid246 | Err.. is there a way to "test" callerid that is sent, to ensure it's #'s onyl? |
| 20:57.13 | Cherebrum | jtexter3: we would evaluate their network |
| 20:57.28 | Cherebrum | jtexter3: and then charge for the upgrades. ;) |
| 20:58.07 | robin_sz | sigh ... but even with the patches, ukcallerid doesnt seem to happen on this X100P |
| 20:58.34 | *** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 20:58.46 | Cherebrum | ... I turned a Western Electric 500 handset from a 1960's style phone into a bluetooth headset.. :) |
| 20:58.58 | Cherebrum | er handset |
| 20:59.42 | hmmhesays | oh hell yeah, they re-maid from russia with love, for ps2 |
| 21:00.46 | Cherebrum | From Russia with love, from Japan with love. |
| 21:00.55 | Cherebrum | ;) |
| 21:01.21 | hmmhesays | You sir are obviously not a james bond fan |
| 21:01.38 | jtexter3 | Cherebrum: I assume you use a bandwidth calculator to determine if it's up to snuff? If so, which one do you use? I see so many, it's hard to know what's good and what's not |
| 21:02.46 | Cherebrum | jtexter3: the Edgewater Edgemark does "call admission controll" You tell it how much bandwidth you have and it won't allow you to exceed your bandwidth limit with phone calls. |
| 21:03.48 | Cherebrum | hmmhesays: I'm saying that the Japan dudes that make the PS2 give you "From Russia with Love" |
| 21:03.53 | _alex_mx_ | i have a box with a TE410P 2E1 with one carrier and 2E1 with a second (all isdn pri) I see HDLC errors with one carrier but not the other any tips on how to further debug this |
| 21:03.59 | Prelius | OK, pastebin is a nifty thing... the url to it is as follows: http://pastebin.ca/183363... Wonder if Digium folks (or anyone) else can help? |
| 21:04.56 | Cherebrum | Prelius: type "make modules_prepare" in your kernel source folder... |
| 21:05.26 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 21:05.27 | *** mode/#asterisk [+o mog] by ChanServ |
| 21:05.49 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.242) |
| 21:07.45 | *** part/#asterisk GaryH (n=GaryH@wallace.garysoft.co.uk) |
| 21:08.02 | *** join/#asterisk hotroot (n=michael@pD9E95265.dip.t-dialin.net) |
| 21:08.04 | jtexter3 | Cherebrum: Do you support multiple codecs? Or just low bandwidth? |
| 21:08.59 | *** join/#asterisk Dude345 (n=Aces1UP@209.101.89.82) |
| 21:09.42 | *** join/#asterisk lorinc (n=ang@caracas-2844.adsl.interware.hu) |
| 21:09.49 | Cherebrum | jtexter3: g711u |
| 21:10.21 | *** part/#asterisk Cherebrum (n=jgarland@207.210.228.172) |
| 21:11.31 | jtexter3 | Cherebrum: Does the Edgewater determine how much bandwidth each call consumes? Or do you use a ration? For TCIP/IP networks, I've been using 80Kpbs (64 + overhead). Has that been your experience as well? |
| 21:11.54 | brookshire | Prelius: which verion of zaptel are you trying to compile? |
| 21:12.19 | Prelius | Cherebrum: ran that, no change, |
| 21:12.38 | Prelius | brookshine: 1.2.9 |
| 21:12.59 | anthm | Cherebrum, how did you test end up last night? |
| 21:13.03 | anthm | i had to leave |
| 21:13.34 | hmmhesays | hey anthm: remember that mipsel project I was bugging you about? |
| 21:13.39 | anthm | ya |
| 21:13.59 | hmmhesays | after going through 2 bad pieces of hardward, I'm about 90% there |
| 21:14.29 | brookshire | prelius: i suggest using the newest verison first. 1.2.9.1 |
| 21:14.29 | anthm | you must have lots of patience that was since like july no? |
| 21:14.37 | hmmhesays | anthm: it was off and on |
| 21:14.44 | brookshire | but more importantly, check and make sure you have a /usr/src/linux linked the correct place |
| 21:15.05 | hmmhesays | my last little hangup is gsm support i'm getting some errors compiling, no clue what they mean |
| 21:15.14 | hmmhesays | otherwise asterisk runs |
| 21:15.21 | Prelius | brookshire: sorry, it is the latest version 1.2.9.1, forgot to add .1 |
| 21:15.59 | brookshire | ls -ld /usr/src/linux* |
| 21:16.26 | robin_sz | ~book |
| 21:16.27 | jbot | somebody said book was a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 21:16.43 | Qwell | brookshire: 1.2.12.1 :p |
| 21:16.49 | anthm | isnt there already a dist of it for mipsel now ? |
| 21:17.15 | *** join/#asterisk Mavvie (n=edwin@ppp43-109.lns2.syd6.internode.on.net) |
| 21:17.28 | hmmhesays | for openwrt there is dd-wrt I believe, but they aren't binary compatible with this board |
| 21:17.29 | Givemelove | Anybody has an issue with zap? when I have a call, the 1st second is cut. it's sufficient to cut the initial "Hello?" |
| 21:17.47 | brookshire | qwell: Zaptel Version 1.2.9.1 |
| 21:17.49 | brookshire | nub |
| 21:17.50 | Qwell | oh |
| 21:18.24 | brookshire | *snaps fingers in a "z" formation* |
| 21:18.26 | quid246 | Is there a way to "test" callerid that is sent, to ensure it's #'s only? |
| 21:18.28 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 21:18.58 | Givemelove | Anybody has an issue with zap? when I have a call, the 1st second is cut. it's sufficient to cut the initial "Hello?" |
| 21:19.24 | brookshire | givemelove: is this on the ivr or when someone picks up the line? |
| 21:19.28 | stoffell | Givemelove: , what phone are you using? |
| 21:20.28 | Givemelove | cisco 7912 |
| 21:20.41 | Givemelove | brookshire: when somebody picks up the line |
| 21:21.26 | [Outcast] | have the addons been fixed for 1.4 |
| 21:22.21 | Qwell | [Outcast]: yes |
| 21:22.25 | robin_sz | someone zipped the Asterisk book PDF ... |
| 21:22.29 | robin_sz | weird |
| 21:22.43 | Givemelove | I'm running 1.2.12.1 core shall I upgrade everything to 1.4 beta or only the addons? |
| 21:22.54 | [Outcast] | Qwell, great |
| 21:23.05 | *** join/#asterisk Rez (i=lorez@freenode/staff/lorez) |
| 21:23.43 | hwt | hey, i want to generate a bunch of calls (with rtp) from one asterisk server, through a nortel SIP gateway, and back to another server. |
| 21:23.59 | hwt | how can i achieve this? |
| 21:24.00 | brookshire | givemelove: i do not believe this is a zap issues because you are using an ip phone |
| 21:24.08 | brookshire | zap has already answered, i believe |
| 21:24.11 | hwt | astertest is no good, as far as i can understand. |
| 21:24.19 | [Outcast] | /var/spool/asterisk/outgoing |
| 21:24.23 | Givemelove | brookshire: what would you guess? |
| 21:24.43 | hwt | [Outcast]: i was kinda hoping for something more elegant. |
| 21:24.50 | brookshire | is this phone local to the network of the asterisk server? |
| 21:25.26 | Givemelove | yes, it's onto the same network |
| 21:25.35 | brookshire | hwt: the hammer! |
| 21:25.43 | brookshire | it's a commercial product though |
| 21:25.45 | brookshire | :/ |
| 21:25.59 | [Outcast] | brookshire, it is expense |
| 21:26.03 | [Outcast] | google sipp |
| 21:26.15 | brookshire | i don't know what it costs.. |
| 21:26.30 | brookshire | but.. someone could make a fortune by just renting those things, lol |
| 21:26.36 | Givemelove | :D |
| 21:27.41 | brookshire | Givemelove: i have no idea really, does it do it with other phones too? |
| 21:28.01 | *** join/#asterisk dovid (n=dovi5988@barak.cellcom.co.il) |
| 21:28.04 | *** join/#asterisk dpetersen (n=dpeterse@158.91.216.16) |
| 21:28.20 | *** join/#asterisk GaryH (n=GaryH@wallace.garysoft.co.uk) |
| 21:28.26 | dovid | is it ok for me to remove the directory /var/log/asterisk and then just recreate the folder ? |
| 21:28.27 | hwt | brookshire: URL? |
| 21:28.45 | Givemelove | yeah |
| 21:28.50 | Givemelove | that sucks |
| 21:28.59 | [Outcast] | hwt: it cost thousands |
| 21:29.07 | dovid | Givemelove: talking to me ? |
| 21:30.29 | dovid | is it ok for me to remove the directory /var/log/asterisk and then just recreate the folder ? |
| 21:30.44 | Corydon-w | dovid: yes, as long as you restart Asterisk |
| 21:30.51 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 21:31.20 | Prelius | brookshire: |
| 21:31.21 | Prelius | <PROTECTED> |
| 21:31.21 | Prelius | <PROTECTED> |
| 21:31.24 | *** join/#asterisk QbY_ (n=Kelvin@cm-64-221-172-182.dhcp.southerncoastalcable.net) |
| 21:32.28 | *** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.mn.comcast.net) |
| 21:32.36 | QbY_ | in sip.conf -- i have register => username:pass@host.com/1234567890 ... which context is * going to look for 1234567890 in? |
| 21:33.08 | Corydon-w | dovid: you also need to recreate the cdr-csv directory |
| 21:34.39 | *** join/#asterisk Psykick (n=anon@222.153.207.54) |
| 21:34.44 | Psykick | hi guys |
| 21:35.06 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 21:35.09 | Psykick | I keep getting this error in the logs when I try to place an outbound call |
| 21:35.10 | Psykick | <PROTECTED> |
| 21:35.21 | Psykick | has anyone seen this before? |
| 21:35.50 | Psykick | I've tried googling ... and found someone else with the same problem ... no answers to his problem either |
| 21:36.14 | dpetersen | I've just done a new installation with a TDM2412E and am having a lot of problems with static and occasional dropped calls on the ZAP channels. I've had the phone company test the lines, and they all checked out okay. Anyone have ideas of what else I could look at to fix the problem? |
| 21:38.35 | *** part/#asterisk Jamez^7 (n=martini@modemcable131.214-131-66.mc.videotron.ca) |
| 21:39.13 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 21:39.14 | *** mode/#asterisk [+o mog] by ChanServ |
| 21:39.14 | trelane_ | dpetersen, sure, after hours take asterisk down and fxotune the channels |
| 21:39.32 | trelane_ | dpetersen, it'll take AWHILE... then make sure you load the fxotune echo-coefficients |
| 21:40.22 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 21:40.28 | *** part/#asterisk hotroot (n=michael@pD9E95265.dip.t-dialin.net) |
| 21:40.28 | trelane_ | dpetersen, also, run zttool on the machine and select your tdm2412p and tell me how many interrupts you've dropped, also do the same running cat /proc/interrupt (could be a timing problme) |
| 21:41.05 | dpetersen | trelane_, does that still have an effect with the hardware echo cancellation module enabled? |
| 21:41.16 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 21:41.16 | trelane_ | dpetersen, absolutely |
| 21:41.22 | trelane_ | dpetersen, they do two different jobs |
| 21:41.43 | trelane_ | one deals with echo on the line, the other echo on the bridge (and on the line to some extent) |
| 21:41.51 | dpetersen | ah |
| 21:41.59 | trelane_ | I always fxotune |
| 21:42.00 | Cresl1n | dpetersen: make sure you're using the zaptel-beta version of fxotune |
| 21:42.07 | trelane_ | indeed |
| 21:42.13 | trelane_ | very good advice |
| 21:42.21 | trelane_ | file broke it! |
| 21:42.23 | trelane_ | :) |
| 21:43.03 | droops | hey, im using a conference to bridge 2 calls, is there a way without and agi, to end the conference in 20 seconds, if the second person never joins? |
| 21:43.25 | trelane_ | how is the confrence being started? |
| 21:43.32 | trelane_ | conference |
| 21:43.55 | droops | meetme(444,dMq) |
| 21:44.43 | trelane_ | how are both callers getting there? (transferred from a dialplan, hey, I dialed 444!, or what? |
| 21:44.56 | droops | oh, transfered froma dialplan |
| 21:45.21 | trelane_ | to my knowledge (and there are better asterisk dialplan authors out there than I) you're going to need an AGI in that case |
| 21:45.41 | droops | ok, thanks trelane |
| 21:45.53 | trelane_ | good luck :) |
| 21:50.32 | *** join/#asterisk xAD (n=xAD@host144-199.pool8290.interbusiness.it) |
| 21:51.02 | anthm | add it as a magic var in meetme MEETME_TIMER=20 and MEETME_TIMER_REQ=2 or something to say kick everyone in 20 sec unless there are 2 members |
| 21:51.44 | Corydon-w | droops: just use the w() flag in 1.2.12.1 |
| 21:51.56 | *** join/#asterisk MIXX941 (n=mark@unaffiliated/mixx941) |
| 21:52.01 | trelane_ | anthm, is that in meetme.conf or as a set() prior to the call of meetme |
| 21:52.02 | Corydon-w | i.e. meetme(444,dMqw(20)) |
| 21:52.29 | quid246 | haha, anybody remember the old RCA TV's that had a built in speakerphone |
| 21:52.30 | anthm | i am just proposing it, it would have to be coded |
| 21:52.31 | Corydon-w | droops: as long as the second user is a Marked user, that will work |
| 21:52.49 | anthm | well there ya go |
| 21:53.00 | [hC] | So, im pretty sure ive found a bug in asterisk's voicemail forwarding. If you prepend the voicemail before you send it, it leaves the message in your inbox still. |
| 21:53.17 | Qwell | [hC]: There was something just fixed, relating to that |
| 21:53.21 | [hC] | Ahh |
| 21:53.26 | [hC] | This box is running 1.2.7.1 |
| 21:53.26 | Corydon-w | We had to add it to MeetMe in 1.2 because Page would lock up channels if someone paged and hungup before everybody was added to the conference |
| 21:53.30 | [hC] | maybe i should upgrade and try again. |
| 21:53.31 | Qwell | in fact.. |
| 21:53.41 | Qwell | Corydon-w: Would your fix fix his thing also? |
| 21:53.49 | Corydon-w | Yes |
| 21:54.21 | droops | Corydon-w, the w() flag? |
| 21:54.31 | droops | ohh |
| 21:54.33 | Corydon-w | droops: it's fairly new |
| 21:54.35 | droops | ok, see it now |
| 21:54.46 | droops | jsut set the second caller as a marked user |
| 21:54.48 | *** join/#asterisk smackus (n=ckwall@63.149.122.93) |
| 21:54.59 | droops | thats awesome, you going to phreaknic? |
| 21:55.01 | Corydon-w | droops: right |
| 21:55.15 | [hC] | Qwell: are you referring to corydon about my problem, or did i just not see the previous conversation |
| 21:55.19 | Corydon-w | droops: I better be; I'm the treasurer for the nonprofit |
| 21:55.20 | Qwell | [hC]: yeah |
| 21:55.23 | droops | then i got a cold high life pony just for you |
| 21:56.01 | *** join/#asterisk Aces1UP (n=Aces1UP@209.101.89.82) |
| 21:56.18 | Aces1UP | has anyone used asterisk as a GSM origination Service here? |
| 21:56.59 | smackus | I would like to add a field to my cdr database and populate it with the dnis of the number dialed. Can anyone refer me to a command or some sort of documentation on how to do this? I am not finding anything. |
| 21:57.04 | droops | this wwont work on anything before 1.2.12.1? |
| 21:57.17 | *** part/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net) |
| 21:57.25 | Corydon-w | droops: I think not. Like I said, it's a fairly new fix |
| 21:57.42 | Corydon-w | but you could probably backport the fix, if needed |
| 21:58.02 | droops | i appreciate it man, i was dredding writing an agi to do this |
| 21:58.25 | hmmhesays | duffman says "alright" |
| 21:58.31 | shodan- | is there some phone number to name & address script available for canadian numbers ? 411.ca (I think it's powered by whitepages,com) does reverse lookup , is there a script that uses their site maybe ? |
| 21:58.52 | quid246 | shodan: Don't think so... but it's a good idea, script something up. |
| 21:59.31 | shodan- | it'd be great , once you have the address then use google maps and embed that in the voice-emails :))) |
| 21:59.35 | Corydon-w | droops: went into 1.2 at revision 43003 |
| 21:59.58 | quid246 | shodan: Reverse engineer the US 411 script... then work from there |
| 22:00.27 | Corydon-w | droops: issue 7275 in the bugtracker has part of the fix |
| 22:01.22 | *** join/#asterisk daysmen3 (n=primus@host86-139-53-205.range86-139.btcentralplus.com) |
| 22:01.59 | shodan- | which script ? got an address or do they come with asterisk ? |
| 22:02.00 | smackus | can i do a setvar and push that to the cdr database? |
| 22:02.17 | *** join/#asterisk freebsd_fan (n=ebola@catagiuri305.giuri.unige.it) |
| 22:02.32 | quid246 | Anyone know of a way of testing whether a clients CallerID NUM is valid (numbers instead of anything else)? |
| 22:03.18 | *** part/#asterisk Heimidal (n=Heimidal@phpbb/styles/heimidal) |
| 22:04.17 | sx-wks | any news from the ipv6 bounty ? |
| 22:04.36 | anthm | there's a regex func |
| 22:04.59 | anthm | i think, at least there was when i wrote it unless it was nuked at some point |
| 22:05.02 | Corydon-w | quid246: you can do a match via the dialplan |
| 22:05.09 | Corydon-w | anthm: no, it's still there |
| 22:05.44 | Corydon-w | quid246: exten => 123/_NXXNXXXXXX,n,NoOp(cid is good) |
| 22:06.00 | Corydon-w | quid246: exten => 123,s,Hangup |
| 22:06.36 | *** part/#asterisk GaryH (n=GaryH@wallace.garysoft.co.uk) |
| 22:08.01 | quid246 | Corydon: Good idea |
| 22:08.17 | *** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) |
| 22:08.44 | smackus | wait... |
| 22:08.56 | smackus | what kind of info is in called number? in the cdr database? |
| 22:08.57 | *** part/#asterisk QbY_ (n=Kelvin@cm-64-221-172-182.dhcp.southerncoastalcable.net) |
| 22:09.09 | smackus | is that the dialed number? |
| 22:09.24 | Corydon-w | smackus: ${EXTEN} |
| 22:10.09 | smackus | so that is essentially DNIS then, right? |
| 22:10.17 | Corydon-w | Yes |
| 22:10.40 | Corydon-w | Usually, anyway |
| 22:11.02 | hmmhesays | phew, what a crazy day |
| 22:11.05 | hmmhesays | CRAZY DAY |
| 22:11.10 | smackus | ok, so if my callednumber is not being populated, how do i populate it? |
| 22:11.24 | hmmhesays | in the cdr's? |
| 22:11.27 | smackus | do i just do a setvar(callednumber? |
| 22:11.54 | hmmhesays | doesn't the dst field usually come from the last called ${EXTEN}? |
| 22:12.02 | *** join/#asterisk joako (n=joako@64.238.175.93) |
| 22:12.08 | Corydon-w | smackus: Goto is what you'd need to use |
| 22:12.23 | smackus | why goto? |
| 22:12.28 | smackus | I thought that routed the call... |
| 22:12.33 | Corydon-w | smackus: EXTEN is your instruction address |
| 22:12.34 | smackus | i just want to report the dialed number |
| 22:12.57 | [hC] | ok so i upgraded to the latest asterisk |
| 22:13.00 | smackus | i need to populate somewhere in the cdr database the number that caller dialed to get into the system. |
| 22:13.23 | [hC] | when i try to forward a message to another user, and i prepend it |
| 22:13.26 | Corydon-w | Uh, is that the actual name of the database field? |
| 22:13.28 | [hC] | it prepends, but sticks it back in my inbox prepended. |
| 22:13.36 | [hC] | (voicemail) |
| 22:13.48 | smackus | callednumber yes |
| 22:14.05 | quid246 | Corydon: I'm just trying to figure out how to use your example in the context of a GotoIF statment... as in "if CID is not numbers write 123-456-7890 as CIDNUM"? |
| 22:14.29 | Corydon-w | smackus: what cdr driver are you using? |
| 22:14.35 | smackus | cdr_mysql |
| 22:14.50 | smackus | i think i can just do a setvar... |
| 22:14.53 | Corydon-w | quid246: it's a Goto, not a GotoIf |
| 22:15.39 | Corydon-w | smackus: unless you've done some custom coding, cdr_mysql does not know about such a field |
| 22:15.40 | [hC] | Corydon-w: this patch you were referencing earlier, would it have any effect n voicemail forwarding behavior, when prepending the message? |
| 22:15.54 | quid246 | Sorry... I mean like (1) If Caller ID is valid, goto step 3. (2) Write CallerID as 123-456-7890. (3) Dial |
| 22:15.59 | Corydon-w | [hC]: yes, it should fix it |
| 22:16.13 | [hC] | Corydon-w: is it present in 1.2.12.1, sorry i missed it. |
| 22:16.21 | quid246 | Hmm.. I'm almessed up today... maybe not enough food |
| 22:16.27 | Corydon-w | quid246: exten => 123/_NXXNXXXXXX,n,NoOp |
| 22:16.46 | Corydon-w | quid246: exten => 123,Set(CALLERID(num)=123456789) |
| 22:16.56 | Corydon-w | err, that should be... |
| 22:17.00 | Corydon-w | quid246: exten => 123,s,Set(CALLERID(num)=123456789) |
| 22:17.12 | Corydon-w | quid246: exten => 123,n,Dial(...) |
| 22:17.33 | Corydon-w | In other words, if the callerid is okay, do nothing |
| 22:17.39 | Corydon-w | Otherwise, set the callerid |
| 22:18.06 | Corydon-w | The NoOp and the Set are at the same priority |
| 22:18.26 | Corydon-w | The Set happens only if the callerid does not match the pattern |
| 22:18.37 | quid246 | okay, thanks... will give that a try now |
| 22:19.08 | [hC] | Corydon-w: sorry to bug you, is the fix present in 1.2.12.1 or only in trunk? |
| 22:19.42 | Corydon-w | [hC]: current 1.2 |
| 22:20.11 | [hC] | Corydon-w: heh, current 1.2 as in trunk, or as in current release? Cause ive installed the latest 1.2 (1.2.12.1) and the problem persists. |
| 22:20.36 | Corydon-w | current release |
| 22:20.44 | Qwell | [hC]: You'll need to checkout branches/1.2/ |
| 22:21.11 | Qwell | svn trunk, svn branch 1.2, and 1.2 release, are all different thigns |
| 22:21.12 | Corydon-w | [hC]: the fix went into the 1.2 tree yesterday |
| 22:21.22 | [hC] | Qwell: Ah. I see. So newer than the 1.2.12.1 tarball on asterisk.org, but still the current stable branch. |
| 22:21.24 | Qwell | and now 1.4 branch and 1.4 release |
| 22:21.25 | [hC] | gotcha. |
| 22:21.29 | Qwell | exactly |
| 22:21.31 | Corydon-w | So, no, it's not in 1.2.12.1 |
| 22:22.00 | smackus | in the cdr database, there is a field named callednumber. i am just trying to figure out how to populate it |
| 22:22.29 | *** join/#asterisk alexis101 (n=as@MTRLPQ02-1177996380.sdsl.bell.ca) |
| 22:22.51 | Corydon-w | smackus: unless you've altered cdr_mysql.c, it will not and can not be altered from Asterisk |
| 22:23.35 | alexis101 | Hi guys ... i have a little problem im triying to use to function AGENT but i always get the error message telling me that The function AGENT is not registred |
| 22:24.08 | alexis101 | anyone know what i have to do to make the fuction work ? |
| 22:24.48 | Corydon-w | Agent is a channel type, not a function |
| 22:24.55 | alexis101 | there is a fuction to |
| 22:25.07 | alexis101 | http://www.voip-info.org/wiki/index.php?page=Asterisk+func+agent |
| 22:25.41 | *** join/#asterisk joako (n=joako@64.238.175.93) |
| 22:26.05 | joako | How can I get multiple SIP phones behind the same nat to register to an Asterisk server outside the NAT? |
| 22:26.45 | *** join/#asterisk pagec (n=pagec@141.155.63.98) |
| 22:27.24 | Corydon-w | alexis101: what version are you running? |
| 22:27.47 | pagec | i am trying to setup echo canceling (but I have no zaptel cards installed), I have been hunting around on google, but cannot find how i turn on echo canceling without any zaptel cards (i.e. i am only using ztdummy) |
| 22:28.34 | pagec | how do i setup echo canceling on a pure iax connection with ztdummy? |
| 22:29.03 | joako | pagec: Read this: http://www.voip-info.org/wiki/view/Causes+of+Echo |
| 22:29.11 | Corydon-w | alexis101: the AGENT function was introduced in the 1.4-beta2 release. If you're running 1.2, you don't have access to the AGENT function |
| 22:29.32 | alexis101 | ok thank you :( |
| 22:29.41 | *** join/#asterisk adamowitz (n=adamowit@ip68-9-201-27.ri.ri.cox.net) |
| 22:30.10 | Aces1UP | has anyone used asterisk as a GSM origination Service here? |
| 22:30.39 | adamowitz | does anyone here know anything about PsipTN or TeLTeL? |
| 22:31.34 | pagec | joako: yes i reach that, problem is definitely with local telco calls and VOIP. and I read on how to implement echo cancellation, but it all talks about compiling zaptel for hardware, does that also work using ztdummy? |
| 22:31.55 | pagec | joako: s/reach/read |
| 22:32.17 | Corydon-w | Aces1UP: doubtful. What hardware are you using for termination? |
| 22:32.41 | Corydon-w | Aces1UP: and whose driver? |
| 22:33.36 | Corydon-w | I suppose it could be done if your GSM gateway had an FXO/FXS or T1 interface |
| 22:33.36 | Aces1UP | 2n gsm gateway |
| 22:33.46 | joako | pagec: So you are doing a setup with only VoIP? That means the echo is caused by your provider and they should fix the echo issue.... |
| 22:35.15 | Corydon-w | Aces1UP: ah, a SIP gateway |
| 22:35.51 | Corydon-w | Aces1UP: or the PRI gateway? |
| 22:35.51 | Aces1UP | yep |
| 22:35.56 | Aces1UP | sip |
| 22:36.24 | Corydon-w | You should be able to use it same as any SIP provider |
| 22:36.49 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 22:36.59 | Aces1UP | well here is what i would like to do. |
| 22:37.51 | Aces1UP | mobile to gsm gateway to usa via voip |
| 22:38.16 | pingwin[work] | hey I've set the callerid for my pri, why is it still coming up on external lines as the internal extension? |
| 22:38.34 | Corydon-w | Aces1UP: sounds fine |
| 22:38.59 | Cresl1n | pagec: you can't do echo cancelation on pure voip calls |
| 22:39.18 | Cresl1n | pagec: echo cancelation is _only_ done when the call interacts with the PSTN |
| 22:39.23 | Cresl1n | *period* |
| 22:39.24 | *** join/#asterisk RoyK (n=roy@gprs-ggsn6-nat.mobil.telenor.no) |
| 22:39.34 | *** join/#asterisk Kraig (n=sebas@nat1.inalambrica.net) |
| 22:39.49 | Aces1UP | now, can i have sims rollever to another channel to keep to the main access number open for new calls |
| 22:39.50 | pagec | Cresl1n: i have an iax2 connection to the provider, does that mean that echo isn't my fault peroid? |
| 22:40.15 | Cresl1n | pagec: who hears it, the person you're calling through your provider, or do you hear it? |
| 22:40.25 | pagec | Cresl1n: i hear it |
| 22:40.35 | Cresl1n | pagec: it's your iax2 provider then |
| 22:40.50 | Cresl1n | they're not doing a proper job doing echo cancelation |
| 22:41.00 | Aces1UP | corydon think i could use skype for voip in that solution? |
| 22:41.28 | pagec | Cresl1n: so all i can do i call them? |
| 22:41.33 | Cresl1n | yep |
| 22:41.35 | Cresl1n | that's it |
| 22:41.37 | Corydon-w | Aces1UP: no, skype is proprietary |
| 22:41.43 | *** part/#asterisk dhahn (n=dhahn@ip-216-17-139-63.rev.frii.com) |
| 22:41.56 | *** join/#asterisk droops (n=root@adsl-065-005-212-128.sip.jan.bellsouth.net) |
| 22:42.01 | joako | Or find a better provider.... |
| 22:42.12 | Aces1UP | what would you suggest for that solution as a voip provider? |
| 22:43.19 | joako | There are many... the only one I can recommend is www.isphone.net but they have minimums and such.. only good for voip providers |
| 22:45.33 | *** join/#asterisk NEZPERCECOUNTY (n=Nez@co.nezperce.id.us) |
| 22:45.53 | NEZPERCECOUNTY | I'm looking to find the best Video Client available for * |
| 22:46.44 | NEZPERCECOUNTY | Does anyone have any experiance with this? |
| 22:48.16 | NEZPERCECOUNTY | I'm running a standard install of Trixbox 1.2 |
| 22:48.45 | NEZPERCECOUNTY | Perhaps someone has patched h.264 before? |
| 22:49.29 | Corydon-w | Trixbox is not supported here. See the topic |
| 22:50.16 | Corydon-w | ~freepbx |
| 22:50.22 | jbot | well, freepbx is the Microsoft BOB of PBXes and NOT supported here! People using it should join #freepbx (FreePBX is the new name of AMP) |
| 22:50.22 | NEZPERCECOUNTY | Thanks |
| 22:51.40 | [hC] | Anyone know of an issue with a linksys or sipura ATA, where for some reason when you call it, it responds with 'busy here' and passes the caller on to voice mail ? |
| 22:51.54 | [hC] | when either one person is on the phone, or sometimes nobody is on the phone |
| 22:53.52 | joako | well if someone is on the phone, the call waiting could be disabled |
| 22:54.51 | ManxPower | [hC], I'm having a psychic vision! |
| 22:54.58 | ManxPower | [hC], you are using G726, aren't you? |
| 22:55.05 | [hC] | no sir. ulaw. |
| 22:55.19 | ManxPower | Those things never were very accurate. |
| 22:57.03 | *** join/#asterisk DrukenHME (n=jdumais@CPE000854de4ec0-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 22:58.39 | joako | How can I get multiple SIP phones behind the same nat to register to an Asterisk server outside the NAT? |
| 22:59.13 | ManxPower | joako, I don't see why not. |
| 22:59.27 | ManxPower | the NATbox will change the source port, just like it does for every other protocol. |
| 22:59.38 | joako | It does not work right. One phone registers, it works fine. Turn it off and register the other phone, the other one works fine |
| 22:59.55 | ManxPower | joako, What stops working when both are registered? |
| 23:00.04 | [hC] | huh |
| 23:00.04 | [hC] | so |
| 23:00.06 | ManxPower | and what does the port show in "sip show peers" |
| 23:00.07 | [hC] | nobody is on the phone |
| 23:00.13 | joako | turn both on at the same time, one will stop to accept inbound calls and after a while one will be registered and the other will just say ints trying to register |
| 23:00.14 | [hC] | yet calling this linksys pap2 ata returns 'busy here' |
| 23:01.45 | joako | right now my sip show peers shows |
| 23:01.48 | joako | 001099022a78L1/001099022a 64.238.175.226 D N 5060 OK (39 ms) |
| 23:01.51 | ManxPower | joako, set qualify=yes in the section in sip.conf for each device. |
| 23:01.56 | joako | it is.... |
| 23:01.58 | joako | and nat=yes |
| 23:02.21 | joako | when both phones are registered and one stops to accept inbound calls, it says it is using port 1024 in sip show peers |
| 23:02.27 | ManxPower | joako, what is sip show peers when BOTH phones are running and have been powered on within 30 seconds on the sip show peers |
| 23:13.33 | *** part/#asterisk amdtech (n=amd011@ss-5-100.shsu.edu) |
| 23:13.54 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 23:15.15 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 23:15.16 | *** mode/#asterisk [+o denon] by ChanServ |
| 23:17.59 | *** join/#asterisk kamyl (n=kamyl@lion.visual.pl) |
| 23:19.34 | *** join/#asterisk Kerry_G (n=Kerry_G@ip70-187-129-227.oc.oc.cox.net) |
| 23:20.16 | robin_sz | I suspect joako was registerign both phones to the same sip account |
| 23:20.27 | *** part/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 23:21.30 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 23:21.42 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 23:22.31 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 23:23.18 | *** join/#asterisk [shodan] (n=shodan@ip118.96-113-216.pppoe1.joliette.intermonde.net) |
| 23:25.39 | *** join/#asterisk joako (n=joako@64.238.175.93) |
| 23:26.26 | *** part/#asterisk techie (n=techie@c-67-182-22-174.hsd1.ca.comcast.net) |
| 23:26.44 | joako | ManxPower: sorry, I had to step out.... |
| 23:26.47 | *** part/#asterisk NEZPERCECOUNTY (n=Nez@co.nezperce.id.us) |
| 23:27.23 | joako | Right now one phone says Registering.... and the other is registered |
| 23:28.55 | joako | Does anyone know how I can get two SIP phones behind a NAT to register to the same asterisk, outside the NAT? |
| 23:29.35 | Kerry_G | I do it all the time, havent had an issue with it |
| 23:30.36 | *** join/#asterisk Schulich (n=Jazba@165.154.37.45) |
| 23:30.39 | Kerry_G | I have 3 phones at home behind nat connected to a remote * box which is also behind another nat at a remote location |
| 23:31.17 | *** part/#asterisk kamyl (n=kamyl@lion.visual.pl) |
| 23:31.28 | joako | What sort of phones? |
| 23:31.47 | Kerry_G | Linksys SPA941, Linksys SPA942, and GS GXP-2000 |
| 23:32.07 | joako | Hmmm.... and these are Linksys phones I am having issues with |
| 23:32.08 | Kerry_G | and I can fire up X-Lite or SJPhone for testing |
| 23:32.18 | joako | And everything uses port 5060? |
| 23:32.21 | joako | what sort of NAT? |
| 23:32.45 | Kerry_G | At home its a Linksys WRT54GS v4 and the * box is behind a pfsense firewall |
| 23:32.59 | joako | What firmware on the WRT? |
| 23:33.25 | Kerry_G | 4.1.17.1 or something like that |
| 23:33.26 | *** part/#asterisk dasenjo (n=dasenjo@208.195.215.242) |
| 23:33.30 | Kerry_G | Linksys Firmware |
| 23:33.33 | joako | So the stock one... |
| 23:33.39 | Kerry_G | yes, latest version |
| 23:33.49 | joako | Hmmm... I am also using a WRT54GS v 4 but with the DD-wrt firmware |
| 23:33.59 | joako | The problem is, I dont think the stock one supports WDS |
| 23:34.08 | Kerry_G | I also used to use OpenWRT but it chocked if I had the VOIP version |
| 23:34.43 | *** part/#asterisk Agrajag- (n=filip@c211-30-4-5.artrmn1.nsw.optusnet.com.au) |
| 23:34.46 | *** join/#asterisk QbY_ (n=Kelvin@cm-64-221-172-88.dhcp.southerncoastalcable.net) |
| 23:34.48 | joako | I'm not using the VoiP version,... which just has SER installed... I was going to try it but thats not the solution I wanted |
| 23:35.11 | QbY_ | is it possible to customize the voicemail greetings based upon the context of the mailbox? |
| 23:35.17 | orlock | bahh |
| 23:35.29 | orlock | i swapped home phones from a 7940 to a Grandstream |
| 23:35.34 | orlock | my GF no like the grandstream |
| 23:35.45 | Kerry_G | smart GF |
| 23:35.50 | orlock | heh, yeah :) |
| 23:36.01 | orlock | but i didnt pay for either, work did.. so they get the decent one |
| 23:36.05 | *** join/#asterisk gooagle (n=goldenol@65.171.196.4) |
| 23:36.05 | joako | qby: well each mailbox has its own greeting, do you mean the stuff such as "The person at extension ... is on the phone" |
| 23:36.08 | gooagle | hi |
| 23:36.15 | gooagle | has anyone gotten blf working with the thomson st2030? |
| 23:36.38 | QbY_ | joako.. yes.. |
| 23:37.12 | QbY_ | joako.. so that context a would say, "the person at extension...." but context b would say, "the employee at extension" |
| 23:37.31 | joako | QbY_: hmmm... not exactly based on the context of the extension, but you COULD use the multi-language feature to allow for different recordings |
| 23:37.58 | QbY_ | joa.. i was thinking of doing that.. could i define my own language? |
| 23:38.10 | joako | so exten => 123,3,SetLanguage(SOMENAME) exten 123,4,VoiceMail(u123) |
| 23:38.18 | *** join/#asterisk Blackthorn (n=blacktho@w-l4.smyth.net) |
| 23:38.28 | joako | all you need to do is make a folder in /var/lib/asterisk/sounds |
| 23:38.39 | joako | and SetLangauge(FOLDER) |
| 23:38.47 | QbY_ | cools.. |
| 23:39.27 | Blackthorn | Hi All, I just recieved a pair of grandstream 386 units for evaluation (I have been deploying spa-2000). The Grandstream setup was quick and easy and setup on g726/32 just like i was hoping for. But each of the units has an ac humm? Suggestions? |
| 23:39.31 | joako | See http://www.voip-info.org/wiki/view/Asterisk+multi-language for more details... |
| 23:39.38 | ManxPower | joako, you MUST use different SIP userids |
| 23:39.53 | ManxPower | Blackthorn, welcome to grandstream |
| 23:40.03 | joako | ManxPower: yes, they are two different userids...... |
| 23:41.06 | Blackthorn | if thats normal operation with the ac hum thats just not acceptable for deployment... bummer net little looking units though... |
| 23:46.07 | *** join/#asterisk RoyKa (n=roy@ti211210a080-1773.bb.online.no) |
| 23:49.12 | DrukenHME | how do i get an ata to know what it's ip is? |
| 23:51.01 | *** join/#asterisk angom_h (n=Angel@red-corp-200.38.30.241.telnor.net) |
| 23:52.24 | *** join/#asterisk Ciber311 (n=Ciber311@user-1087e94.cable.mindspring.com) |
| 23:54.58 | *** part/#asterisk nettie (n=nettie@85-18-54-38.ip.fastwebnet.it) |
| 23:55.40 | *** join/#asterisk Nivex (n=kjotte@user-0c8hq6g.cable.mindspring.com) |
| 23:56.40 | robin_sz | how can i find out what version the actual kernel module of zaptel is actually running? |
| 23:56.41 | [Outcast] | anyone here from sipwest? |
| 23:57.08 | robin_sz | trying to figure out why my UK callerid is not worky. |
| 23:58.37 | robin_sz | mon aeroglisser est plein des anguilles |
| 23:58.38 | *** join/#asterisk dhahn (n=dhahn@ip-216-17-139-63.rev.frii.com) |
| 23:58.43 | joako | DrukenHME: You can get an ATA to know its IP via DHCP - Dynamic Host Configuration Protocol |
| 23:59.13 | dhahn | Does anyone have an example of using the Manager to originate a call and start an AGI application? |
