| 00:03.40 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 00:03.45 | *** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net) |
| 00:04.28 | syzygyBSD | any asterisk devs in here? |
| 00:05.00 | *** join/#asterisk bjohnson (n=bjohnson@i209-195-72-9.cia.com) |
| 00:05.08 | *** join/#asterisk etfonhomey (n=etfonhom@74-140-206-4.dhcp.insightbb.com) |
| 00:07.56 | Nugget | Yes, lots, but that question usually sends them hiding. :) |
| 00:08.04 | inv_arp[work] | lol |
| 00:08.12 | perd | just /who digium |
| 00:08.18 | perd | and start msg'ing like crazy |
| 00:08.19 | perd | they love that |
| 00:08.25 | hads | hah |
| 00:09.27 | syzygyBSD | lol, I was just going to ask them a question on difficulty of development before I dive into something I shouldn't |
| 00:10.13 | tzafrir_laptop | syzygyBSD, so ask |
| 00:10.43 | tzafrir_laptop | one of the great astersk devs will eventually drop by and notice it |
| 00:10.46 | tzafrir_laptop | ;-) |
| 00:10.51 | syzygyBSD | well, I was thinking about rewriting the TCP stack SIP uses so it would correctly handle multiple IPs |
| 00:11.39 | tzafrir_laptop | or, consider #asterisk-dev |
| 00:11.57 | syzygyBSD | ya, there is never any discussion in there though |
| 00:12.04 | tzafrir_laptop | I figure some folks are still there... |
| 00:12.14 | *** join/#asterisk Rhiliam (n=gary@CPE001310426d31-CM0012256ea75c.cpe.net.cable.rogers.com) |
| 00:12.22 | syzygyBSD | ya, that is just for show though, they are all in the secret dev channel |
| 00:12.37 | Rhiliam | has anyone hooked up a vonage ata to your * box? |
| 00:12.51 | Waverly360 | Is there anyone here who can walk me through a dual pbx config via the sip.conf file? |
| 00:13.11 | syzygyBSD | Rhiliam: lots of people have |
| 00:13.24 | *** join/#asterisk etfonhomey1 (n=etfonhom@74-140-206-4.dhcp.insightbb.com) |
| 00:13.45 | Rhiliam | I am thinking all it would take is routing the FXO port on the back into ab FXS port on the Digium card? |
| 00:13.47 | syzygyBSD | the problem is with unlocking it so you can... |
| 00:14.03 | Rhiliam | maybe my simple idea won't work then :) |
| 00:14.07 | syzygyBSD | Rhiliam: wait.. how are you wanting to hook it up? |
| 00:14.52 | syzygyBSD | I was thinking you wanted to use the sip feature to asterisk, not the analog |
| 00:14.52 | syzygyBSD | why not just go with another sip provider? |
| 00:14.54 | Rhiliam | long story. |
| 00:15.03 | syzygyBSD | k, you are correct in what you think you have to do |
| 00:15.49 | Rhiliam | cool. While I am here another quetsions. Is it possible to add a sip channel to a ZAP group? Essentially, have all available channels, whether zap/sip or iax part of a group? |
| 00:15.50 | syzygyBSD | should be straight forward, ask if you have any problems :) |
| 00:16.49 | dj-fu | where can I find the voice files provided by asterisk-sound? default directory? |
| 00:17.05 | syzygyBSD | <PROTECTED> |
| 00:17.58 | dj-fu | anyone suggest a nice sound for "the person you have dialed is not available, transferring you through to another extension" |
| 00:18.00 | dj-fu | or similar? |
| 00:18.17 | [TK]D-Fender | Rhiliam : no. |
| 00:18.54 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 00:19.26 | Rhiliam | thats what I though. Any thoughts on g729 vs speex vs gsm? |
| 00:21.08 | JT_ | why would you need to add a sip channel to a zap group? |
| 00:21.08 | dj-fu | if the zaps are full, call out over the sip |
| 00:21.08 | *** part/#asterisk Cresl1n (i=matt@nat/digium/x-fc964a4a48423abd) |
| 00:21.09 | dj-fu | I was thinking of doing the same but realised that it's probably impossible to do with type/group/exten style |
| 00:21.31 | JT_ | dj-fu: it's unnecessary in reality |
| 00:21.42 | Rhiliam | I am not really upto snuff with my asterisk coding - how would I route the call to the sip channel if the zaps are full? |
| 00:21.45 | riddlebox | what is needed to get zapscan.conf? |
| 00:21.45 | JT_ | you just add another priority to the extension that calls the second priority |
| 00:21.55 | dj-fu | right, true |
| 00:22.17 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 00:22.49 | dj-fu | is there a way for me to return the call back to my incoming as 's'? |
| 00:23.06 | dj-fu | like, if an extension is unavailable and they leave a message, instead of hanging up, I'd like to put them back into the incoming context |
| 00:23.22 | dj-fu | kinda like a fallback I guess |
| 00:24.35 | elriah | Anyone have the Cisco file term41.default.loads? |
| 00:29.25 | *** join/#asterisk Givemelove (n=bozoo@cre94-1-81-57-163-2.fbx.proxad.net) |
| 00:29.27 | Givemelove | Hi there |
| 00:29.38 | Givemelove | anybody have already seen this onto the console? |
| 00:29.51 | Givemelove | handle_response_invite: Received response: "Forbidden" from '"XXXXXXXXX" <sip:818XXXXXXX@10.1.69.243>;tag=as47c0f24d' |
| 00:30.09 | elriah | Anyone have the Cisco file term41.default.loads? |
| 00:30.43 | JT_ | dj-fu: GoTo? |
| 00:31.14 | dj-fu | I suppose. I thought there may have been another way, thanks |
| 00:31.44 | JT_ | there might be |
| 00:31.52 | JT_ | depends on exactly what you're try to do |
| 00:34.27 | dj-fu | nah I think goto will suffice, thanks |
| 00:34.39 | dj-fu | is the Voicemail() application case sensitive? I've seen it in the book as Voicemail and VoiceMail |
| 00:34.45 | dj-fu | and also do I have to specify @default? |
| 00:35.40 | *** join/#asterisk matiasds19 (n=matiasds@200.55.35.159) |
| 00:35.50 | elriah | Anyone have the Cisco file term41.default.loads? If not, does anyone have a hammer so I can smash 25 Cisco IP phones? |
| 00:36.21 | elriah | ... since we've fried them all apparently, lol |
| 00:37.58 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 00:39.05 | *** join/#asterisk coppice (n=chatzill@118.202.17.210.dyn.pacific.net.hk) |
| 00:39.42 | elriah | Wow, Cisco is really friendly. I called their 800 number, got a body after hitting one number, and got access to what I needed. I dont' have any contracts or anything. Whoever said they were a bumbling buracracy (sp?) |
| 00:40.14 | JT_ | you were just lucky |
| 00:40.24 | elriah | lol, maybe, but I'm greatful ... |
| 00:40.35 | dj-fu | was abou to say |
| 00:40.40 | dj-fu | you usually need a contract to get those images |
| 00:41.12 | elriah | He just asked me for my serial number ... |
| 00:41.28 | dj-fu | nice |
| 00:41.32 | elriah | Downloading the files now, if they work I'm a happy camper... |
| 00:41.39 | matiasds19 | hey guys, im having a problem reciving inbound calls from my sip provider, i dont know why the providers hang up the call 1 second after i answer. If i register the provider with eyebeam, they dont hang up, i analized the traffic and the RTP session is stablish for the second the call was up. Someone knows wahts going on? |
| 00:42.36 | matiasds19 | i know they hang up, because analizing the traffic they send me the bye signal |
| 00:42.40 | elriah | matiasds19: Describe the setup ... |
| 00:43.04 | matiasds19 | shall i pastebin the sip.conf? |
| 00:43.13 | CunningPike | elriah: You found the secret way in. They are going to have to kill you now |
| 00:44.35 | *** part/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 00:44.54 | elriah | CunningPike: Oh great, well, eye for an eye, we just killed 25 of their nice new IP phones (hence the call)... So i guess we're due.. |
| 00:45.02 | CunningPike | lol |
| 00:51.28 | matiasds19 | here is the sip.conf |
| 00:51.29 | matiasds19 | http://www.pastebin.ca/352804 |
| 00:51.50 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 00:53.31 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 00:56.38 | sweeper | AMI can check what clients are registered to the server, neh? |
| 00:57.05 | *** join/#asterisk _Thor (i=CS@user-vc8fl7l.biz.mindspring.com) |
| 00:57.15 | _Thor | Hello everybody |
| 00:58.17 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 00:59.45 | *** join/#asterisk Stp1800 (n=Stp1800@67-22-111-205.atlsfl.adelphia.net) |
| 00:59.57 | Stp1800 | Hi guys. Have a firewall question. |
| 01:00.25 | Stp1800 | I have a Linksys router in front of my Asterisk server. What ports do I need to forward in to my Asterisk box to allow iax2 traffic to pass? |
| 01:02.36 | *** join/#asterisk sharp (n=sharp@c-68-46-30-7.hsd1.pa.comcast.net) |
| 01:05.14 | JT_ | UDP 4569 |
| 01:05.39 | *** join/#asterisk Dr-Linux|home (n=Dreamer@DSL-202-59-73-131.nexlinx.net.pk) |
| 01:11.11 | *** join/#asterisk anthonyl (n=fbfff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 01:11.15 | *** join/#asterisk DigitalSoul (i=Sim0n@renegade.ws) |
| 01:15.14 | kervel | anyone knows a way to make a reasonably cheap voip doorbell ? |
| 01:15.35 | kervel | i find solutions that need an analog convertor + a 500 EUR doorbell |
| 01:16.22 | tzafrir_laptop | 500 euro? |
| 01:16.30 | Givemelove | embed a Cisco phone to your doorwall |
| 01:16.31 | kervel | http://forums.whirlpool.net.au/forum-replies.cfm?t=416104&p=1 |
| 01:16.40 | Givemelove | with a quick dial button :) |
| 01:16.58 | kervel | Givemelove in this neighbourhood it wouldn't survive 2 hours :) |
| 01:17.08 | Givemelove | depends |
| 01:17.11 | Givemelove | where do U live? |
| 01:17.21 | kervel | belgium |
| 01:17.28 | Givemelove | be sure |
| 01:17.42 | Givemelove | il sera vole chez moi avant chez toi |
| 01:17.43 | Givemelove | :D |
| 01:17.46 | CunningPike | kervel: Check out Viking |
| 01:17.47 | Givemelove | Creteil |
| 01:18.02 | kervel | :) |
| 01:18.04 | Givemelove | ;) |
| 01:18.05 | kervel | viking ? |
| 01:18.43 | CunningPike | http://www.vikingelectronics.com/ |
| 01:18.59 | kervel | going to look .. |
| 01:19.16 | *** join/#asterisk shodan- (n=shodan@ip065.96-113-216.pppoe1.joliette.intermonde.net) |
| 01:19.20 | perd | viiiiiiikinggggg |
| 01:20.16 | Givemelove | E50 --> vandal resistant, |
| 01:20.21 | Givemelove | sweet |
| 01:20.39 | kervel | pity they are analog |
| 01:20.39 | perd | f that, you want vandal resistant just hook up the case to a high voltage power line |
| 01:20.57 | perd | makes killer BBQ, too |
| 01:21.13 | robl^ | mmm. BBQ vandals and chips!! |
| 01:21.14 | kervel | they make ip phones for 50 euro , why not a simple button with a network plug |
| 01:23.03 | Givemelove | check this one |
| 01:23.03 | Givemelove | http://www.vikingelectronics.com/products/view_product.php?pid=383 |
| 01:23.11 | Givemelove | good luck |
| 01:23.33 | kervel | hehe |
| 01:24.35 | robl^ | ohhh! do they sell those embedded in a clown's head? "I want a double cheese burger, milk shake, and large fries." |
| 01:25.06 | *** join/#asterisk wubba (n=kmurrey@cable-76-215.sssnet.com) |
| 01:25.34 | The_DoC^ | Dude Where's my car |
| 01:26.23 | wubba | Is there a log file that shows if an extension is registered or not? |
| 01:26.43 | dlynes_laptop | wubba: sip show registry |
| 01:29.10 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqu6.cable.mindspring.com) |
| 01:29.16 | perd | what is the bare minimum required for an asterisk configuration |
| 01:29.22 | perd | which files need to exist |
| 01:29.35 | perd | asterisk.conf, features.conf, modules.conf ? |
| 01:29.42 | wubba | Where are the log files that shows the mysql info? |
| 01:29.48 | perd | oh and codecs.conf i guess? |
| 01:30.05 | dlynes_laptop | perd: delete them all and find out |
| 01:30.14 | perd | yea im about to do that |
| 01:30.18 | dlynes_laptop | perd: just start adding back the files that it complains about |
| 01:30.19 | perd | i wanted to expedite the process :/ |
| 01:30.23 | dlynes_laptop | until it stops complaining |
| 01:30.24 | perd | hehe ok |
| 01:30.29 | perd | no lazy way out ofr me |
| 01:30.40 | dlynes_laptop | Well, I could tell you |
| 01:30.44 | dlynes_laptop | But I've never done it myself |
| 01:31.05 | dlynes_laptop | Well...I did it once, but I didn't document it |
| 01:31.07 | perd | and then i wouldnt learn anything :P |
| 01:31.16 | dlynes_laptop | and the files that it needed then have probably changed |
| 01:31.31 | dlynes_laptop | because that was back in the early days of 1.2 |
| 01:31.37 | perd | ah |
| 01:32.13 | dlynes_laptop | back when most people were still refusing to upgrade to 1.2 from 1.0 |
| 01:33.10 | dlynes_laptop | I'm encountering an error trying to compile zaptel 1.2.12 with Linux 2.6.20 |
| 01:33.19 | dlynes_laptop | I'm just curious if someone might have a fix for it |
| 01:33.24 | dlynes_laptop | My pastebin is at http://ipv4.pastebin.ca/352860 |
| 01:34.48 | dlynes_laptop | Fwiw, 1.2.13 compiles and links just fine with Linux 2.6.20 |
| 01:35.18 | dlynes_laptop | But Sangoma doesn't patch against 1.2.13 and it does patch against 1.2.12 |
| 01:35.19 | Qwell | well...uhh...1.2.12 came out before 2.6.20, so... |
| 01:35.26 | i3inary | anyone here know much about .call files and methods of bridging the calls together? |
| 01:36.15 | i3inary | im currently using originate: and exten: to create the 2 call legs |
| 01:36.17 | dlynes_laptop | Qwell: well, do you know what the newest kernel version is that works with 1.2.12, then? |
| 01:36.24 | Qwell | 2.6.19 |
| 01:36.30 | *** join/#asterisk droemel (n=droemel@p548EB7A3.dip0.t-ipconnect.de) |
| 01:36.30 | dlynes_laptop | Ok, thanks |
| 01:36.46 | dlynes_laptop | I just needed a newer kernel version so that my network card driver works |
| 01:37.48 | i3inary | im kinda stumped on how to get my originate call leg and my exten call leg to show in my cdr records |
| 01:38.07 | i3inary | both my originate and my exten call legs are outbound calls |
| 01:39.28 | i3inary | right now they show up as 1 cdr record...but in reality i am paying for 2 calls to the pstn...am i wrong in assuming that i should be making 2 outbound calls via exten instead of originate and exten? |
| 01:41.10 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 01:41.28 | *** join/#asterisk Kumbang (n=kumbang@167.205.24.67) |
| 01:42.29 | *** join/#asterisk Exstatica (i=exstatic@redline.mednor.net) |
| 01:43.02 | _Thor | Hello Everyone |
| 01:43.09 | i3inary | hi |
| 01:43.22 | _Thor | Anybody here good with ASTCC? |
| 01:46.10 | _Thor | I guess no |
| 01:46.16 | *** part/#asterisk _Thor (i=CS@user-vc8fl7l.biz.mindspring.com) |
| 01:49.55 | *** join/#asterisk ez` (n=supermar@c207.134.229-230.clta.globetrotter.net) |
| 01:52.53 | *** part/#asterisk wubba (n=kmurrey@cable-76-215.sssnet.com) |
| 01:55.26 | *** join/#asterisk dmz (n=dmz@64.20.147.237) |
| 01:56.04 | dmz | hey y'all |
| 01:56.05 | *** join/#asterisk fholmes (n=fholmes@cpe-72-177-234-192.houston.res.rr.com) |
| 01:56.35 | dmz | what is the recommended way to have an agent who receives a call via queue onto his/her phone (hard phone, not soft) to get a url or something to direct the agent to a client specific page? |
| 01:59.05 | *** join/#asterisk iq (n=iq@unaffiliated/iq) |
| 01:59.07 | iq | Hi |
| 01:59.32 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 02:05.41 | Exstatica | dmz: i've used wildfire with the asterisk plugin |
| 02:06.02 | dmz | thanks let me google that |
| 02:06.54 | rue_mohr | heh kb1 needs the T1 card back :) |
| 02:07.55 | Exstatica | its from jivesoftware.org |
| 02:14.15 | *** join/#asterisk hoowa (n=chatzill@210.83.203.100) |
| 02:14.32 | hoowa | morning |
| 02:18.15 | *** join/#asterisk J4k3 (i=jsuter@12.197.130.202) |
| 02:21.44 | *** join/#asterisk yxa (n=lonari@58.185.90.101) |
| 02:21.55 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 02:22.01 | yxa | anyone knows what does dtmfthreshold is for in misdn? |
| 02:24.26 | JT_ | rue_mohr: kb1? |
| 02:31.21 | yxa | anyone familiar with misdn? |
| 02:31.49 | JT_ | familiar enough to say i don't like it :) |
| 02:32.50 | yxa | JT_ the volume is very soft on and i have adjusted the txgain but its still the same. any ideas? |
| 02:32.55 | *** join/#asterisk tim0123 (n=cash247@adsl-75-39-213-70.dsl.rcsntx.sbcglobal.net) |
| 02:33.11 | *** join/#asterisk sahafeez (n=sahafeez@ip68-6-215-70.sd.sd.cox.net) |
| 02:33.51 | JT_ | not sure |
| 02:34.00 | tim0123 | How do you fix this [cdr_addon_mysql.so]Aug 20 15:18:38 ... cannot open shared object file: No such file or directory |
| 02:34.28 | JT_ | is cdr_addon_mysql.so there? |
| 02:34.35 | tim0123 | no |
| 02:35.04 | tim0123 | Im using red hat 4 |
| 02:35.14 | tim0123 | Im using red hat enterprize 4 |
| 02:37.46 | *** join/#asterisk J4k3- (i=jsuter@dhcp-12-197-128-58.intrastar.net) |
| 02:38.27 | JT_ | i'm not sure how your distro is relevant |
| 02:40.29 | *** join/#asterisk DukeOfURL (n=chatzill@vc7-1-39.dsl.netrack.net) |
| 02:40.48 | *** part/#asterisk alindeman (i=adml@freenode/staff/alindeman) |
| 02:40.57 | DukeOfURL | Hello |
| 02:40.57 | *** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140) |
| 02:41.04 | tim0123 | Haha that was alot of help |
| 02:41.37 | JT_ | what was? |
| 02:42.44 | *** part/#asterisk connecta (n=Administ@175.6.188.72.cfl.res.rr.com) |
| 02:43.26 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 02:44.04 | JT_ | tim0123: you have failed to mention how you installed asterisk/etc |
| 02:44.31 | JT_ | tim0123: logic would dictacte if a required file is missing, you would have to replace that file |
| 02:44.44 | JT_ | only if you are using mysql cdrs do you require that object |
| 02:45.34 | tim0123 | I installed the addon |
| 02:46.12 | JT_ | how did you install all this? |
| 02:46.29 | *** join/#asterisk mhayk (n=mhayk@200.199.53.56) |
| 02:47.37 | *** join/#asterisk jpe-nyc (n=jpe@p77-37.acedsl.com) |
| 02:48.11 | DukeOfURL | anyone here know about writing an application to run on a motorola cell phone? where to start? |
| 02:49.17 | perd | uhh |
| 02:49.21 | perd | yea, learn asm |
| 02:49.58 | perd | or maybe some kind of CE crap |
| 02:50.06 | perd | depending on what type of phone it is |
| 02:50.34 | etfonhomey1 | You can also right java apps for some phones. |
| 02:50.43 | perd | that would be sweet |
| 02:50.44 | mhayk | anyone here know about what I need to integrate my speech recognize with asterisk..? |
| 02:50.52 | perd | mhayk: sphinx or lumenvox |
| 02:50.55 | DukeOfURL | need access to GPS on the least expensive phone |
| 02:51.01 | perd | sphinx is free, it works well, it's a bit more involved to set up than lumenvox |
| 02:51.03 | DukeOfURL | need Internet access |
| 02:51.04 | perd | lumenvox costs a lot. |
| 02:51.17 | perd | <PROTECTED> |
| 02:51.29 | JT_ | DukeOfURL: all new phones sold in the US have gps |
| 02:51.36 | JT_ | dunno how accessible they are though |
| 02:51.48 | DukeOfURL | ok, but which phones allow my Java app to access GPS? |
| 02:56.28 | CJLinst | Is it possible to use subscriptions to display things like whether or not an agent is logged into a queue? On a phone like an snom 360 or poly 601? |
| 02:58.16 | *** join/#asterisk dennisharrison (n=dennisha@68-114-106-133.dhcp.slid.la.charter.com) |
| 03:01.50 | *** join/#asterisk Milk_ (n=None@74-134-97-185.dhcp.insightbb.com) |
| 03:01.53 | Milk_ | good evening |
| 03:02.08 | Milk_ | am Im I allowed to ask a question, despite it being a freepbx/trixbox install? |
| 03:02.41 | *** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir) |
| 03:04.02 | Qwell | is it in any way related to freepbx or the config files it creates? |
| 03:04.11 | dlynes_laptop | DukeOfURL: what do GPS, Java, or cellphones have to do with Asterisk, anyways? |
| 03:04.59 | Milk_ | Qwell, oh... possibly, but only indirectly. As well, no one is answering in #freepbx or #trixbox, and my current forums searches arn't helping yet |
| 03:05.18 | dlynes_laptop | DukeOfURL: It seems like your question might be better answered on a variety of other avenues such as forums for WinCE, Java mobile edition, or SymbianOS |
| 03:05.18 | elriah | Anyone have a Cisco term41.default.loads file? |
| 03:05.37 | Qwell | elriah: should be the same firmware as a 7961 |
| 03:05.40 | DukeOfURL | dlynes_laptop: This cell phone application will communicate with Asterisk |
| 03:05.47 | Qwell | elriah: and no, you'll have to call Cisco |
| 03:05.55 | dlynes_laptop | DukeOfURL: using sip? |
| 03:06.23 | elriah | Qwell: Thanks. I called Cisco, they gave me the SIP firmware but before I knew I needed term41.default.loads, my TAC guy went home for the evening... Oh well... |
| 03:06.30 | elriah | Anyone have a term61.default.loads file? |
| 03:06.30 | DukeOfURL | no, GSM/DTMF and GPRS. |
| 03:06.58 | Milk_ | I'm taking that as a big no then |
| 03:07.15 | Qwell | Milk_: you can ask, but don't expect an answer if it's related to freepbx |
| 03:07.23 | *** join/#asterisk etfonhomey (n=etfonhom@74-140-206-4.dhcp.insightbb.com) |
| 03:07.35 | dlynes_laptop | DukeOfURL: Ummm...GSM is a codec..asterisk uses version 2 if I remember correctly; most cell phones use version 4 which is patented |
| 03:07.46 | dlynes_laptop | DukeOfURL: DTMF is just touch tones |
| 03:07.59 | Milk_ | Qwell, well.. I think I've decided to reinstall and reseat the card, but here goes |
| 03:08.13 | dlynes_laptop | DukeOfURL: and GPRS is protocol for overlaying TCP/IP on top of PCS |
| 03:08.19 | DukeOfURL | dlynes_laptop: GSM = Global System Mobile (or something like that). It's not the codec. |
| 03:08.20 | Qwell | dlynes_laptop: gsm is more than a codec |
| 03:08.32 | etfonhomey | DukeofURL: did anyone send you this link: http://developer.motorola.com/products/handsets/ |
| 03:08.39 | *** join/#asterisk ModocNet (n=d82e036a@69-12-147-8.dsl.static.sonic.net) |
| 03:08.41 | dlynes_laptop | Qwell: Yeah, but still...none of what he's talking about has anything to do with asterisk |
| 03:08.45 | DukeOfURL | yup--been there |
| 03:08.52 | Qwell | dlynes_laptop: I agree. :p |
| 03:08.56 | Milk_ | I've got an FXO card installed, its connected to my lingo line. I'm trying to get it to send incomming calls to my polycom sip phone, and allow the phone to call out, but its not functioning properly, despite no red alarms on the fxo card, its an X100P |
| 03:08.59 | DukeOfURL | I've looking for someone that's already fought the battles. |
| 03:09.28 | dlynes_laptop | DukeOfURL: but you still haven't told us how asterisk is going to interface with the cellphone |
| 03:09.49 | dlynes_laptop | DukeOfURL: gprs, cellphone gsm are not valid ways of interfacing with asterisk |
| 03:10.33 | dlynes_laptop | DukeOfURL: I think what you probably want to use is SIP |
| 03:10.39 | DukeOfURL | oh my god! I can't call an Asterisk system with my cell phone? What have I been doing for four years??? |
| 03:10.51 | dlynes_laptop | DukeOfURL: SIP would go over UDP, which goes over TCP, which goes over GPRS |
| 03:11.00 | dlynes_laptop | DukeOfURL: Well...GPRS, EDGE, or 3G |
| 03:11.17 | Qwell | udp over tcp? |
| 03:11.17 | DukeOfURL | Have you investigated the cost of GPRS? |
| 03:11.33 | dlynes_laptop | Qwell: ooops...meant UDP over IP |
| 03:11.37 | *** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net) |
| 03:12.24 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 03:12.28 | dlynes_laptop | DukeOfURL: no |
| 03:12.49 | DukeOfURL | It's real expensive. |
| 03:12.49 | dlynes_laptop | DukeOfURL: another method you could use is a GSM gateway which still talks SIP to asterisk |
| 03:13.02 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 03:13.02 | *** mode/#asterisk [+o mog] by ChanServ |
| 03:13.08 | dlynes_laptop | DukeOfURL: and that'll broadcast a GSM, CDMA or TDMA signal |
| 03:13.08 | DukeOfURL | That's in the plan. But the cell phone is still GSM. |
| 03:13.32 | dlynes_laptop | DukeOfURL: GSM is the easiest way to go, but that's in the gateway...still nothing to do with asterisk |
| 03:13.49 | dlynes_laptop | DukeOfURL: the only thing that's common for all of those solutions that asterisk can interface with is SIP |
| 03:14.11 | elriah | Qwell: Do you know the filename that term41.default.loads comes in? |
| 03:14.37 | elriah | Qwell: And is it, by chance, just one of the firmware bins renamed? |
| 03:14.45 | DukeOfURL | Well, the IVR running in Asterisk is most definitely part of the plan. But I've got that covered. What I'm looking for is someone with J2ME/Motorola development experience |
| 03:15.10 | dlynes_laptop | DukeOfURL: Like I said...for that part you're probably better off asking in a J2ME forum |
| 03:15.27 | puzzled | elriah: not a single filename but many since it's part of Cisco's firmware tarballs |
| 03:15.34 | dlynes_laptop | DukeOfURL: But what you're looking for to interface asterisk with your cellphone is called a 'GSM gateway' |
| 03:15.56 | DukeOfURL | Yup. Company in Genmany makes them. |
| 03:16.01 | DukeOfURL | Germany |
| 03:16.09 | dlynes_laptop | DukeOfURL: so effectually no special programming is needed on the asterisk side...you just need to know how to set up a SIP channel |
| 03:16.11 | DukeOfURL | Deutchland |
| 03:16.16 | elriah | puzzled: I have a bunch of the tarballs but can't find term41.defaults.load, is it in one of the bins or sbns? |
| 03:16.38 | puzzled | elriah: sorry don't remember |
| 03:16.48 | DukeOfURL | No, there's a bunch of IVR scripting that coordinates with the app in the phone. |
| 03:17.23 | dlynes_laptop | DukeOfURL: so you want a DTMF application in the phone then |
| 03:17.33 | dlynes_laptop | DukeOfURL: it's still nothing to do with asterisk |
| 03:17.46 | dlynes_laptop | DukeOfURL: that's to do with J2ME, or SymbianOS |
| 03:18.00 | dlynes_laptop | DukeOfURL: I would think that's something you're going to have to write using a native application for the cellphone |
| 03:18.10 | DukeOfURL | Yes, I need to develop an application for the phone. And I'm not the first person to post OT on this list, nor the last. |
| 03:18.12 | dlynes_laptop | DukeOfURL: I would imagine it's not possible using J2ME |
| 03:18.39 | dlynes_laptop | DukeOfURL: do you know what the native OS is for the cellphone? |
| 03:18.44 | JT_ | this isn't a list |
| 03:19.00 | dlynes_laptop | DukeOfURL: i.e. for all the Nokia Series 60 cellphones, they use SymbianOS |
| 03:19.21 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-248-142.dc.res.rr.com) |
| 03:19.25 | DukeOfURL | There are several. I would like to start with J2ME because it's available on many phones. |
| 03:19.37 | DukeOfURL | Symbian is too fragmented. |
| 03:19.40 | dlynes_laptop | DukeOfURL: Well, do whatever you like |
| 03:19.43 | DukeOfURL | Too many forks. |
| 03:19.49 | DukeOfURL | I will. Thanks. |
| 03:19.50 | dlynes_laptop | DukeOfURL: but as I said...I don't think it's possible with J2ME |
| 03:20.05 | DukeOfURL | I hope it is. |
| 03:20.08 | dlynes_laptop | DukeOfURL: Too many forks of Symbian? It's a commercial OS...it's not open source |
| 03:20.29 | DukeOfURL | Each manufacturer has gone its own way with Symbian. |
| 03:20.46 | dlynes_laptop | Yeah...each manufacturer has created their own extensions to Symbian |
| 03:21.02 | dlynes_laptop | They haven't forked it, though |
| 03:21.04 | DukeOfURL | And it's not open. |
| 03:21.12 | dlynes_laptop | Correct..it's not open |
| 03:21.46 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 03:22.18 | DukeOfURL | And I'm thinking of an FSK modem for both the phone and Asterisk to pass data over GSM. |
| 03:22.29 | DukeOfURL | like what caller id uses |
| 03:22.48 | tengulre | hi,all |
| 03:22.58 | perd | oh man |
| 03:23.03 | perd | exercise is the worst thing in the world |
| 03:23.29 | JT_ | it can be |
| 03:23.34 | JT_ | but it can be like a drug |
| 03:23.37 | JT_ | done regularly enough |
| 03:23.53 | DukeOfURL | The girls at the gym are real cute, however. |
| 03:24.14 | JT_ | gyms seem like such an artificial environment to exercise in |
| 03:24.32 | dlynes_laptop | yeah...badminton and tennis courts are much better |
| 03:24.39 | dlynes_laptop | meet a lot of asian girls there :) |
| 03:24.46 | tengulre | :) |
| 03:25.02 | JT_ | oh they sound much more boring, smashing a ball around a hard court |
| 03:25.02 | dlynes_laptop | Most around college age, too :) |
| 03:25.33 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 03:26.48 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 03:26.48 | *** mode/#asterisk [+o anthm] by ChanServ |
| 03:27.30 | *** join/#asterisk eject_ck (n=eject@ll-110.213.223.85.sovam.net.ua) |
| 03:28.42 | tengulre | Happy spring day of chinese here!!! |
| 03:29.59 | eject_ck | Hi all, I'm using spa3k with asterisk and have one big trouble - when I call through my digital office PBX (samsung) to Spa3k and then hold call I see in spa3k interface that it still connected. How fix it ? |
| 03:30.24 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 03:30.45 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 03:30.56 | JT_ | i've heard that the birth rates will increase by 10% in asia because of some sort of superstitious belief |
| 03:30.59 | JT_ | bloody ridiculous |
| 03:31.13 | perd | i need to get to the point where exercising is like a drug |
| 03:31.15 | perd | that would be so awesome |
| 03:31.21 | perd | right now it's like a poison |
| 03:31.24 | perd | a bad kind of drug. |
| 03:31.42 | JT_ | heh |
| 03:31.48 | JT_ | adrenaline |
| 03:31.54 | JT_ | yeah it's mostly mental really |
| 03:31.56 | sevard | it takes a long time for exercise to be like a drug |
| 03:32.03 | sevard | i was at that point once |
| 03:32.06 | sevard | it was great. |
| 03:32.11 | DukeOfURL | perd: maybe you need a cell phone that reminds you. |
| 03:32.31 | JT_ | sevard: then what happened? |
| 03:32.31 | DukeOfURL | Bluetooth interface with the chip in your sneaker |
| 03:32.39 | perd | you mean like the ring tone says 'you enjoy exercise, it feels good' ? |
| 03:32.53 | perd | subliminal messages |
| 03:32.53 | DukeOfURL | something like that |
| 03:32.58 | perd | oh shock therapy |
| 03:33.03 | perd | bluetooth shock chip in my shoe |
| 03:33.05 | perd | good idea |
| 03:33.33 | DukeOfURL | no. more like risk/reward. 20K volt arc to your leg. |
| 03:34.14 | DukeOfURL | the shoes are already available--they keep track of your miles, etc. |
| 03:34.30 | perd | oh |
| 03:34.37 | sevard | ))<>(( |
| 03:34.47 | perd | is that an ascii vagina or something? |
| 03:34.52 | sevard | no. |
| 03:34.58 | perd | i was gonna say, it looks pointy. |
| 03:35.04 | sevard | http://www.backandforthforever.com |
| 03:35.31 | perd | anyone here use vista and happen to know how to set sloppy focus for the mouse |
| 03:35.33 | tengulre | anybody using asterisk for telecome corp? |
| 03:35.53 | DukeOfURL | telecome? |
| 03:36.02 | DukeOfURL | is that like phone sex? |
| 03:36.05 | tengulre | telcom |
| 03:36.25 | perd | wth is this sevard |
| 03:36.27 | sevard | tengulre: lots of people are. hint, hint. |
| 03:36.43 | perd | fecal fetish site? |
| 03:36.52 | sevard | no. |
| 03:37.06 | perd | liars |
| 03:37.10 | perd | both of you |
| 03:37.12 | sevard | no lies. |
| 03:37.22 | perd | Say, "You poop into my butt hole and I poop into your butt hole ... back and forth ... forever." - Robby (age 6)????? |
| 03:37.29 | sevard | right. |
| 03:37.36 | sevard | click on Robby |
| 03:38.23 | tengulre | DukeOfURL: don't discuss SEX here! pls! |
| 03:38.32 | DukeOfURL | ME???? |
| 03:38.43 | tengulre | yes , It |
| 03:38.46 | tengulre | 's u! |
| 03:38.52 | perd | i guess that's taken out of context? |
| 03:38.52 | sevard | DukeOfURL: you're a naughty, dirty boy. |
| 03:39.00 | DukeOfURL | ME??? |
| 03:39.11 | tengulre | ;) |
| 03:39.25 | tengulre | make a joke!! |
| 03:39.49 | tengulre | I m playing snooker here! anybody want join us? |
| 03:39.53 | *** join/#asterisk InHisName (n=Administ@c-68-38-105-1.hsd1.pa.comcast.net) |
| 03:39.58 | tengulre | anybody know DingJunHui? |
| 03:40.06 | JT_ | people can discuss what they want |
| 03:40.14 | *** join/#asterisk infernix (n=nix@spirit.infernix.net) |
| 03:41.52 | *** part/#asterisk DukeOfURL (n=chatzill@vc7-1-39.dsl.netrack.net) |
| 03:41.58 | *** join/#asterisk wubba (n=kmurrey@cable-76-215.sssnet.com) |
| 03:42.23 | wubba | If you do an upgrade - is there a log that records what is going on during the upgrade? |
| 03:43.21 | elriah | Anyone have this Cisco firmware file: cmterm-7941_7961-sip.8.0.2SR1.cop |
| 03:48.28 | *** part/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 03:52.14 | tengulre | I will back home tomorrow!!! HAPPY SPRING DAY OF CHINESE!!! say hello to everyone!! |
| 03:55.24 | *** part/#asterisk wubba (n=kmurrey@cable-76-215.sssnet.com) |
| 03:56.56 | eject_ck | as I understand my offcice PBX (samsung officeserv100) have disconnect tone that not eq with SPA3k - 480@-30,620@-30;4(.25/.25/1+2) |
| 03:57.08 | JT_ | tengulre: we call it chinese new year |
| 04:04.07 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 04:07.27 | InHisName | How do I create voicemail accounts in * ? |
| 04:10.24 | rue_mohr | JT_, yea, he needed his the T1 card back so he call tell the nortel foks how their equiptment works |
| 04:10.28 | rue_mohr | :) |
| 04:10.40 | rue_mohr | InHisName, voicemail.conf |
| 04:12.03 | InHisName | rue_mohr, I tried making the one liner in the voicemail.conf. I still don't have a directory nor will that new user have a voicemail to respond. |
| 04:12.26 | elriah | InHisName: How did you install asterisk? From source? |
| 04:12.38 | InHisName | Came with astlinux |
| 04:12.55 | elriah | Did you issue a reload command after modifying voicemail.conf? |
| 04:12.57 | elriah | Or restart? |
| 04:13.04 | InHisName | yes |
| 04:13.37 | elriah | Post your voicemail.conf and extensions.conf and sip.conf into a pastebin and post that url back here and I'll look at it. |
| 04:14.40 | InHisName | hold on gonna take a while to get there. |
| 04:14.46 | JT_ | rue_mohr: i have no idea who he is? |
| 04:15.47 | elriah | JT_ ?? |
| 04:16.39 | JT_ | sorry elriah ? |
| 04:17.20 | *** join/#asterisk JunK-Y (n=junky@modemcable140.185-70-69.mc.videotron.ca) |
| 04:19.44 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
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| 04:26.21 | wubba | Asterisk Call Manager - if you telnet into this - will it ask for a login? |
| 04:27.47 | JunK-Y | the Asterisk Manager Interface? yes. |
| 04:27.47 | *** join/#asterisk hematitec (n=cratz@adsl-71-159-206-4.dsl.pltn13.sbcglobal.net) |
| 04:28.21 | wubba | junk-y - OK - if it doesn't does that mean it's not connecting properly? |
| 04:28.37 | JunK-Y | see manager.conf, make sure its enabled. |
| 04:29.55 | wubba | It is enable - I have ssh into my box and telnet into it - but it doesn't give a login |
| 04:30.27 | wubba | it just says: Asterisk Call Manager/1.0 |
| 04:30.36 | wubba | and then sits at a blank line |
| 04:30.53 | JunK-Y | wubba: so its fine, now read how to login correctly. |
| 04:31.18 | JunK-Y | look for Asterisk Manager Login |
| 04:32.12 | *** join/#asterisk topping (n=topping@204.152.96.238) |
| 04:33.18 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 04:34.38 | bkruse_home | action: login |
| 04:34.39 | bkruse_home | username: yourusername |
| 04:34.39 | bkruse_home | secret: yoursecret |
| 04:34.54 | bkruse_home | theres so much more, thats just the start |
| 04:34.57 | JunK-Y | (and 2 enters ) :) |
| 04:34.59 | bkruse_home | what are you using this for? |
| 04:35.14 | bkruse_home | JunK-Y: i was hoping hed be able to figure it out. But your right ;] |
| 04:35.20 | wubba | testing the paswword |
| 04:36.23 | wubba | I am having trouble - when I add an extension it is not showing up. |
| 04:36.55 | bkruse_home | how is the manager interface going to help that? |
| 04:37.45 | wubba | I'm looking all over the place |
| 04:38.00 | *** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net) |
| 04:39.30 | dlynes_laptop | perd: sloppy focus? is that something like sloppy seconds? |
| 04:40.36 | *** part/#asterisk wubba (n=kmurrey@cable-76-215.sssnet.com) |
| 04:47.58 | *** join/#asterisk markit (n=konversa@host119-245-static.72-81-b.business.telecomitalia.it) |
| 04:48.41 | markit | hi, loading the dialplan I've noticed the message: WARNING[13476]: chan_iax2.c:9119 set_config: Section 'guest' lacks type, but I've [guest] with type=user (* 1.4svn), any clue? |
| 04:53.40 | JT_ | dlynes_laptop: pretty much |
| 04:53.48 | JT_ | dlynes_laptop: sloppy focus is HORRIBLE |
| 04:54.05 | JT_ | you move the moust around the screen, and whatever window it's hovering over, gains focus |
| 04:54.10 | JT_ | s/moust/mouse/ |
| 04:55.26 | bkruse_home | </3 sloppy focus |
| 04:55.29 | bkruse_home | i just alt+tab, easy |
| 04:56.33 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 04:57.20 | dlynes_laptop | JT_: oh yeah...like X11...twm |
| 04:57.23 | dlynes_laptop | JT_: it sucks |
| 04:58.11 | dlynes_laptop | CunningPike: you describe it like you don't have to pick any particular orifice to stick it in...just pick the first one that fits |
| 04:58.22 | dlynes_laptop | CunningPike: that's sloppy focus :) |
| 04:58.38 | CunningPike | :) |
| 04:59.08 | *** join/#asterisk eald (n=eald@189.157.105.134) |
| 05:04.03 | eald | hi guys, which version of asterisk you actually use for production? last 1.2? |
| 05:04.46 | *** join/#asterisk ManxPower (n=manxpowe@67.sub-70-216-41.myvzw.com) |
| 05:04.48 | danp | that's what i use, yeah |
| 05:05.06 | danp | i actually use an svn checkout from the 1.2 branch |
| 05:08.40 | eald | good to know, I guess is very stable |
| 05:11.57 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 05:11.57 | *** mode/#asterisk [+o russellb] by ChanServ |
| 05:24.07 | *** join/#asterisk HockeyInJune (i=HockeyIn@pool-151-204-139-13.ny325.east.verizon.net) |
| 05:26.18 | *** part/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
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| 05:33.43 | zeeesh | hi |
| 05:38.42 | rue_mohr | hi |
| 05:38.46 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
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| 05:46.54 | mosty | i'm trying to figure out how to use an ericsson f220m gsm gateway, i don't have a manual (looking on the web now), does anyone know how to set one of these up with asterisk? |
| 05:47.57 | JT_ | what interface does it deliver? |
| 05:49.24 | mosty | i'm trying to figure that out now. i've found a manual fortunately |
| 05:49.45 | JT_ | ethernet, pots... |
| 05:50.08 | mosty | the diagrams show it plugged into a telephone |
| 05:50.46 | mosty | so i guess it is fxo, and would connect to an fxs card in my asterisk box |
| 05:51.12 | JT_ | right |
| 05:51.24 | JT_ | how annoying ;) |
| 05:51.45 | mosty | i have a telephone plugged in now, trying to figure out how to enter my sim cards pin |
| 05:52.24 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 05:52.24 | *** mode/#asterisk [+o russellb] by ChanServ |
| 05:53.40 | mosty | ahh, that worked :) |
| 05:54.18 | *** join/#asterisk tonyheb (n=dub@modemcable060.80-200-24.mc.videotron.ca) |
| 06:05.41 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 06:06.44 | *** join/#asterisk PhilKC (n=PhilKC@freenode/staff/about.linux.philkc) |
| 06:09.32 | k-man | how would i set up a voicemail box for unanswered calls to our main line? so any of the office staff can check those messages? |
| 06:11.21 | mosty | k-man: create a voicemail box, give out the PIN number for when people dial in to retrieve voicemail |
| 06:11.33 | k-man | ah, ok |
| 06:19.45 | *** join/#asterisk hoowa (n=chatzill@210.83.203.100) |
| 06:26.52 | *** join/#asterisk foobar778 (n=bryan@ip68-100-41-120.dc.dc.cox.net) |
| 06:30.24 | k-man | so... say i set my extension to 500, what should happen if i dail 500? |
| 06:30.29 | *** join/#asterisk eject_ck (n=eject@ll-110.213.223.85.sovam.net.ua) |
| 06:30.31 | k-man | from my phone? |
| 06:30.40 | k-man | will it go to voicemail? |
| 06:30.54 | k-man | or could i set it to go to voicemail i should say |
| 06:33.15 | mosty | you could, but limiting the number of channels that this particular sip account is allowed |
| 06:35.51 | k-man | mosty, well... currently i get an engaged signal if i ring my own extension number |
| 06:36.12 | k-man | as i don't have more than one phone, i can't really tell if my extension number is working |
| 06:36.26 | mosty | depends on the phone too |
| 06:36.38 | mosty | get another phone to test with |
| 06:36.56 | k-man | can i test with a softphone? |
| 06:37.15 | k-man | which windows softphone is recommended for use with asterisk? |
| 06:37.29 | k-man | btw, its an SPA942 |
| 06:37.36 | mosty | i don't know of any softphones that are good, they all suck |
| 06:37.42 | mosty | but try xlite or whatever it's called |
| 06:37.47 | k-man | ok |
| 06:37.50 | k-man | thanks |
| 06:38.46 | *** join/#asterisk lyGP (n=l@60.51.217.146) |
| 06:38.53 | lyGP | Hi everyone |
| 06:39.13 | lyGP | i have a problem here, when i make a call.. |
| 06:39.37 | lyGP | the duration counter automatically start counting even when i havent pick up the phone |
| 06:40.23 | *** join/#asterisk KnowWhat (n=KnowWhat@host210-2-166-243.isb.dancom.net.pk) |
| 06:40.27 | lyGP | is there i any configuration i need to do in order for me to start the duration counting only when i pick up the calls? |
| 06:40.28 | k-man | oh, you have to restart after installing x-lite *sigh*( |
| 06:40.30 | k-man | bbiab |
| 06:40.52 | KnowWhat | k-man well |
| 06:40.57 | KnowWhat | if you dont , it will still work |
| 06:41.03 | k-man | oh.. will it? |
| 06:41.03 | k-man | cool |
| 06:41.04 | KnowWhat | i did that and it is working fine |
| 06:41.16 | k-man | ok |
| 06:41.17 | k-man | thanks |
| 06:41.21 | KnowWhat | i didnt restart my laptop for a few days now lol |
| 06:41.33 | mosty | lygp: what kind of call? |
| 06:41.56 | lyGP | i am calling from SIP SOftphone to PSTN |
| 06:42.24 | lyGP | when my PSTN extension rings, the duration timer started to count, but i still havent pick up the phone |
| 06:42.50 | mosty | does your asterisk box Answer the call before dialing out to the PSTN? |
| 06:45.11 | lyGP | i'm using sip:extension@ip:5060 to call out |
| 06:45.25 | lyGP | i'm using a voip gateway machine |
| 06:47.14 | dlynes_laptop | ok...this is pretty stoopid |
| 06:47.17 | k-man | xlite says i need to give it the domain, what is the domain for my asterisk server? |
| 06:47.29 | dlynes_laptop | Add in a remora daughterboard, and all your modules get renumbered |
| 06:51.18 | *** join/#asterisk Waverly360 (n=mirc@209.149.58.214) |
| 06:51.38 | Waverly360 | Hey guys, I'm having a hell of a time trying to get one pbx to dial an extension on another via sip. |
| 06:52.19 | Waverly360 | At it's simplest form, is it possible to put a Dial string in extensions.conf to directly dial an extension on another pbx? |
| 06:52.50 | lyGP | no solution for my query? |
| 06:53.24 | dlynes_laptop | Waverly360: yes...look up the switch statement for extensions.conf |
| 06:54.51 | dlynes_laptop | Waverly360: it allows sharing of dial plans between two iax2 connected asterisk boxes |
| 06:54.57 | dlynes_laptop | Waverly360: but not using sip |
| 06:55.34 | Waverly360 | dlynes_laptop: So how can I do it with sip? |
| 06:55.40 | dlynes_laptop | Waverly360: you don't |
| 06:55.59 | dlynes_laptop | Waverly360: except through complex dialplan involving checking caller ids and that kind of thing |
| 06:56.06 | Waverly360 | dlynes_laptop: You have to use IAX? What about using register? |
| 06:56.23 | dlynes_laptop | Waverly360: register just allows a sip server to know what ip address to expect calls from |
| 06:56.35 | dlynes_laptop | Waverly360: it's useful for dynamic ips |
| 06:57.05 | Waverly360 | dlynes_laptop: Well, assume for a second that I'm not concerned with the dialplan checking. I just want to simply make a call to another pbx's extension. |
| 06:57.12 | dlynes_laptop | lyGP: mosty asked you a question |
| 06:57.13 | Waverly360 | I mean the callerid checking... |
| 06:57.47 | dlynes_laptop | Waverly360: You can do that with IDD's |
| 06:58.00 | Waverly360 | IDD's? |
| 06:58.06 | dlynes_laptop | inward direct dialing |
| 06:58.20 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 06:58.23 | dlynes_laptop | basically one external phone number maps to a local directly |
| 06:58.43 | k-man | does this look right for an internal extension? exten => 500,1,Dial(SIP/jason,,r) |
| 06:58.49 | lyGP | <mosty> does your asterisk box Answer the call before dialing out to the PSTN? |
| 06:58.49 | lyGP | <lyGP> i'm using sip:extension@ip:5060 to call out |
| 06:58.49 | lyGP | <lyGP> i'm using a voip gateway machine |
| 06:59.05 | dlynes_laptop | lyGP: you're not answering mosty's question |
| 06:59.26 | lyGP | i'm not sure whether the asterisk box answer the call, how do i determine that? |
| 06:59.41 | dlynes_laptop | lyGP: Do you have an 'Answer()' statement in the dialplan for the call? |
| 06:59.49 | lyGP | oh, then now |
| 06:59.59 | lyGP | my extension keep on ringing until i pick up |
| 07:00.10 | dlynes_laptop | lyGP: there's also certain dialplan apps that do an implicit answer as well |
| 07:00.11 | lyGP | but the duration timer start counting when my extension is ringing |
| 07:00.30 | dlynes_laptop | k-man: yeah..looks fine |
| 07:00.38 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 07:00.44 | dlynes_laptop | k-man: but i would drop the ',,r', or you'll get a secondary ringing |
| 07:00.52 | k-man | dlynes_laptop, can't call myself or a softphone |
| 07:00.54 | k-man | ok |
| 07:01.03 | k-man | what is secondary ringing? |
| 07:01.07 | lyGP | exten => _ZXX,1,Dial(SIP/${EXTEN}@192.168.7.92:5060) |
| 07:01.07 | lyGP | exten => _ZXX,2,Hangup() |
| 07:01.22 | lyGP | this is my dial plan |
| 07:01.24 | dlynes_laptop | k-man: ,r generates a ring |
| 07:01.28 | Waverly360 | dlynes_laptop: Is that the only way? I found this line on the "dual servers" voip wiki, but I can't seem to make it work, even if I add the proper sip.conf entries. I get a congestion/busy error |
| 07:01.33 | dlynes_laptop | k-man: sip should automatically supply a ring if one is expected |
| 07:01.39 | Waverly360 | dlynes_laptop: exten => _7XXX,2,Dial(SIP/myserver:passwordA@SIPserverA/${EXTEN:1},30,r) |
| 07:01.39 | k-man | oh, i see |
| 07:02.19 | dlynes_laptop | Waverly360: that '${EXTEN:1}' will only dial an extension, if your sip server routes it as such |
| 07:02.27 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 07:02.48 | dlynes_laptop | Waverly360: but even then, it'll drop the first digit |
| 07:03.37 | dlynes_laptop | Waverly360: i.e. in the incoming context for that sip account, you'll need to handle extensions |
| 07:04.14 | Waverly360 | dlynes_laptop: Well, here's what I did to modify it...keep in mind this was only for proof of concept purposes. exten => 7115,2,Dial(SIP/myserver:passwordA@SIPserverA/115,30,r) |
| 07:04.30 | Waverly360 | When I dial 7115 on my pbx, I want it to dial 115 on another |
| 07:04.51 | dlynes_laptop | Waverly360: yeah...then you would strip the first digit |
| 07:04.57 | dlynes_laptop | Waverly360: so ${EXTEN:1} |
| 07:09.40 | Waverly360 | well...if I use that exact line above..that should work, right? My problem is, I keep getting a " No such host: SIPserverA/115" when I try dialing 7115. I've created the entry in sip.conf, but I must be doing something wrong. |
| 07:09.46 | k-man | dlynes_laptop, ah, i didn't have the internal calls in the general context |
| 07:09.54 | k-man | so i couldn't make internal calls |
| 07:09.59 | k-man | its working now |
| 07:10.24 | k-man | hehe |
| 07:10.29 | k-man | worked that one out for myself! |
| 07:11.10 | lyGP | any solution on the duration timer? |
| 07:13.29 | Waverly360 | Ok..now I'm getting a "SIP/SIPserverA-0a21a4f8 is circuit-busy" message when I try dialing. |
| 07:13.35 | Waverly360 | What would cause that in this situation? |
| 07:17.13 | Waverly360 | Ok..maybe I can take a step back a second |
| 07:18.29 | Waverly360 | When I call this "Dial(SIP/SIPserverA/115,30,r)", asterisk looks for the definition of SIPserverA in my sip.conf file...correct? |
| 07:19.54 | Waverly360 | Anyone? |
| 07:24.18 | reivilo | Waverly360: the easier way is to create a register and a peer entry and at the other side an authorized context who switch in the extensions and context you want |
| 07:26.08 | reivilo | Waverly360: exten => _0.,1,Dial(SIP/${EXTEN:1}@SIPServerB,30,r) |
| 07:29.11 | Waverly360 | reivilo: So I need to create a peer and a register on each server I want to do this to? |
| 07:31.09 | Waverly360 | I had thought about doing it that way...but in all of the reading I've done today, I've found multiple places where it was stated that if both (or all) pbxes had static ips and weren't separated by a NAT, that it would be easy to simply dial from one to another. |
| 07:31.31 | Waverly360 | It just doesn't seem to be as simple, or even possible. |
| 07:35.56 | *** join/#asterisk topping (n=topping@204.152.96.238) |
| 07:36.26 | mafkees | Waverly360: why not use iax ? |
| 07:39.16 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 07:39.33 | Waverly360 | mafkees: Honestly? My manager doesn't want me to for some reason. |
| 07:40.10 | Waverly360 | mafkees: I suggested IAX...he said he's rather do it with SIP if at all possible *shrugs* |
| 07:41.45 | mosty | Waverly360: you don't need to register if they're static ip's |
| 07:41.54 | mafkees | fair enough |
| 07:42.00 | dlynes_laptop | Waverly360: You can do what you want without registering, no problem |
| 07:42.22 | dlynes_laptop | ARgggggh |
| 07:42.26 | dlynes_laptop | Don't tell me |
| 07:42.55 | dlynes_laptop | The latest asterisk is not compatible with zaptel 1.2.12? |
| 07:43.46 | *** join/#asterisk Renacor (n=vircuser@p54B9E3C9.dip.t-dialin.net) |
| 07:43.59 | Renacor | I keep getting [Feb 12 23:39:44] WARNING[2620]: app_dial.c:1081 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) |
| 07:44.04 | Renacor | can somebody help me please? |
| 07:44.16 | dlynes_laptop | Renacor: your zaptel channel is down |
| 07:44.26 | dlynes_laptop | Renacor: do you have your zaptel channel driver loaded? |
| 07:44.36 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 07:46.10 | Waverly360 | mosty, dlynes_laptop: Ok...that's great...how do I do it? I can't seem to create a dial string that will successfully dial an extension on another pbx. |
| 07:46.30 | dlynes_laptop | Waverly360: in the simplest instance: |
| 07:46.33 | Waverly360 | I don't want to deal with registering and all of that, so barebones minimum, what do I need to have. |
| 07:46.45 | dlynes_laptop | Waverly360: Dial(SIP/sippeername/${EXTEN}) |
| 07:46.53 | mosty | Waverly360: Dial(SIP/exten@thehostname,30,r) should work i think |
| 07:46.58 | Renacor | dlynes_laptop: yeah its loaded and ztcfg shows its fine |
| 07:47.15 | dlynes_laptop | Renacor: i didn't ask about the zaptel driver |
| 07:47.17 | Renacor | dlynes_laptop: shows 4 channels configured |
| 07:47.25 | dlynes_laptop | Renacor: i asked about the zaptel channel driver |
| 07:47.32 | dlynes_laptop | Renacor: i.e. chan_zap.so |
| 07:47.43 | Renacor | how can I see if its loaded or not |
| 07:47.51 | dlynes_laptop | Renacor: show modules |
| 07:47.52 | Waverly360 | dlynes_laptop, mosty: So I assume I'll need to define sippeername in sip.conf on the calling asterisk box AND the called asterisk box? |
| 07:47.56 | dlynes_laptop | Renacor: also zap show status |
| 07:48.01 | dlynes_laptop | Renacor: zap show channels |
| 07:48.17 | dlynes_laptop | Waverly360: correct |
| 07:48.33 | dlynes_laptop | Waverly360: will it be two way communication? |
| 07:48.43 | dlynes_laptop | Waverly360: i.e. will the pbxes be able to call each other? |
| 07:48.47 | Waverly360 | Yes |
| 07:48.51 | Renacor | well zap show channels show Chan Extension Context Language MOH Interpret |
| 07:48.55 | mosty | Waverly360: you don't need to- you can specify the hostname or ip in the dial command |
| 07:48.56 | Renacor | with nothing in it |
| 07:48.57 | Waverly360 | I need both pbxes to be able to call extensions on each other. |
| 07:49.02 | dlynes_laptop | Waverly360: just define them as friends then |
| 07:49.13 | dlynes_laptop | Waverly360: forget peers and users |
| 07:49.19 | *** join/#asterisk dj-fu (n=deejay@202-89-157-131.dyn-dsl.ubs.qsi.net.nz) |
| 07:49.29 | dlynes_laptop | Renacor: what about zap show status? |
| 07:49.52 | Renacor | chan_zap.so Zapata Telephony 0 |
| 07:49.53 | Waverly360 | mosty: but I need to be able to setup authentication. I need to define a user and password for each pbx. |
| 07:49.57 | Renacor | in show modules |
| 07:50.04 | mosty | Waverly360: then do that instead |
| 07:50.33 | Waverly360 | mosty: Well, I swear I've done that..and it's not working...I'll try it again from scratch, and make sure I'm not missing something. |
| 07:50.58 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 07:51.25 | dlynes_laptop | Waverly360: you can set up usernames and passwords even when defining the sip peers as friends |
| 07:51.48 | dlynes_laptop | Waverly360: you define those in your sip.conf, not in your dialplan |
| 07:52.22 | foobar778 | Ok this is my problem I have asterisks setup on a Linux box Kanotix with asterisk 1.214 I setup 3 sip users in my sip config and three coreesponding extensions I have a dvg-1120s ata router with one analog phone it registers on the lan ands I put a softphone on the lan and cvall the extension for the anaolog phone and it rings The PC is in the dmz and the iptables -F ALL PORTS ARE OPEN yet no softphonew from the wan can register Yet h |
| 07:52.23 | foobar778 | ermatetic earlier today was able to connect using his pbx across the wan any help please on ghetting softphones or ata to connect to my asterisk all ports are open |
| 07:52.29 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 07:53.03 | Renacor | zap show status shows: |
| 07:53.05 | Renacor | Wildcard TDM400P REV I Board 1 OK 0 0 0 |
| 07:53.32 | Renacor | i don't understand what is wrong here |
| 07:55.57 | Waverly360 | dlynes_laptop: Yeah, it took me a bit to understand that. It was a little confusing thinking of them that way. I'm assuming that means, that I can use the exact same definition on both pbxes to define the credentials that I'll use to connect from one pbx to the next... |
| 07:55.58 | foobar778 | dlynes_laptop did u see my question |
| 07:55.58 | dlynes_laptop | Renacor: maybe you don't have any ports defined in your zapata.conf file? |
| 07:55.58 | dlynes_laptop | foobar778: yes, I saw it....why do you ask? |
| 07:56.12 | foobar778 | I want an answer if u can help please |
| 07:56.42 | dlynes_laptop | foobar778: sorry, but I don't even know what that hardware is that you're using |
| 07:57.13 | foobar778 | does it matter I just want a softphone tp connect to asterisk |
| 07:57.24 | foobar778 | from the wan |
| 07:57.42 | dlynes_laptop | you mentioned soemthing about an ata and kanotix and a whole bunch of other stuff I know nothing about |
| 07:57.52 | dlynes_laptop | so I just went back to what i was doing |
| 07:57.59 | foobar778 | kanotix is just a linix distro |
| 07:57.59 | *** join/#asterisk frogzoo (n=frogzoo@202.155.165.25) |
| 07:58.12 | christel | Qwell: ping |
| 07:58.17 | dlynes_laptop | foobar778: restate your issue without all the other stuff that doens't matter, then |
| 07:58.18 | foobar778 | ata doesnt a play a part in the wan |
| 07:58.22 | dlynes_laptop | chrisq: he's probably sleeping |
| 07:58.45 | foobar778 | ok basically can register behind lan not wan |
| 07:59.02 | foobar778 | <PROTECTED> |
| 07:59.24 | foobar778 | yet not yest |
| 07:59.54 | foobar778 | so another pbx could connect from the wan but softphones wont |
| 08:00.10 | foobar778 | also a pap2 from the wan could not |
| 08:00.39 | foobar778 | But I put my Moms pc on a vpn and yes it worked because thats a big lan |
| 08:01.22 | foobar778 | Xten lite connects lan not wan PC in the DMZ |
| 08:01.23 | dlynes_laptop | sounds like a configuration issue on your softphones and on the pap2 |
| 08:01.33 | *** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it) |
| 08:01.43 | foobar778 | again same config just wan versus lan |
| 08:02.12 | foobar778 | Moms pc before vpn couldnt connect same setting wan |
| 08:02.21 | mosty | firewall issue? |
| 08:02.22 | foobar778 | <PROTECTED> |
| 08:02.28 | foobar778 | DMZ] |
| 08:02.44 | foobar778 | and today an asterisk pbx connecteregisterd |
| 08:02.44 | Waverly360 | Ok, sorry to keep beating this issue. It's still not working. I'm getting a circuit busy/congestion error in the console on the box I'm trying to call from. This is the pbx I'm calling from. extensions.conf: exten => 7115,1,Dial(SIP/PBXA/115,30,r); sip.conf: [PBXA] |
| 08:02.44 | Waverly360 | type=friend |
| 08:02.44 | Waverly360 | username=PBXA |
| 08:02.45 | Waverly360 | secret=mypasswordA |
| 08:02.45 | Waverly360 | host=172.30.1.199 |
| 08:02.47 | Waverly360 | insecure=very |
| 08:02.52 | Waverly360 | oops |
| 08:03.05 | Waverly360 | sorry about the scrolling..didn't mean to click that |
| 08:03.26 | foobar778 | waverly restart asterisks if congetsion works for me |
| 08:03.38 | dlynes_laptop | mosty: not a firewall issue...he's got a softphone on the lan that's not working, and several clients on the wan connecting, but one that isn't |
| 08:04.11 | dlynes_laptop | Waverly360: what's the name of the sip context? is it [PBXA]? |
| 08:04.17 | Waverly360 | yes |
| 08:04.54 | Waverly360 | and I have that same context defined in the sip.conf on the other server. |
| 08:05.00 | foobar778 | does anyone do vnc remotes??? |
| 08:05.11 | foobar778 | to sort this out?? |
| 08:05.52 | mosty | is the format for SIP channels different than that for IAX? i mean, you use IAX/peer/exten for IAX, but the asterisk sip channel page on the wiki says for SIP that it's SIP/exten@peer |
| 08:06.19 | Waverly360 | mosty: I think I've seen it used interchangably |
| 08:07.56 | *** join/#asterisk andrew` (i=andrew@69-12-140-101.dsl.dynamic.sonic.net) |
| 08:10.53 | Waverly360 | dlynes_laptop: Shouldn't that just work? |
| 08:11.58 | Waverly360 | I'm wondering if I've left something out of the 2nd server's config |
| 08:13.23 | dlynes_laptop | I'm sorry...just a bit busy atm |
| 08:13.51 | dlynes_laptop | All the incompatibilities between the latest zaptel, asterisk, sangoma wanpipe drivers, and the linux kernel are killing my free time |
| 08:13.56 | mosty | up till now, i've only been using SIP/peer, but now i want to send an extension also (like I have been using IAX/peer/someextension) so that the sip client can see what DID they were being called on, what format should i use in my dial commands? |
| 08:14.09 | mosty | neither the one in the wiki, nor the one mentioned here seem to work |
| 08:14.44 | dlynes_laptop | mosty: SIP/peer/${EXTEN} |
| 08:15.09 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 08:15.18 | dlynes_laptop | mosty: unmless you're wanting it for a SIP phone, and not a SIP server |
| 08:15.22 | *** join/#asterisk af_ (n=getsmart@ip-179-53.sn1.eutelia.it) |
| 08:15.43 | mosty | dlynes_laptop: neither of my SIP phones like that, asterisk says they're circuit busy when i try that |
| 08:15.56 | Waverly360 | dlynes_laptop: It's alright..I appreciate the help you've given me so far. |
| 08:16.04 | mosty | i want something that works with both sip phones and sip servers |
| 08:16.09 | Waverly360 | dlynes_laptop: I'm still working on the issue myself. |
| 08:16.30 | dlynes_laptop | mosty: like i said...it works for sip servers only |
| 08:16.48 | dlynes_laptop | mosty: try modifying the caller id before passing it to the phone |
| 08:17.37 | mosty | dlynes_laptop: do you know if there is an easy way to distinguish the two, so i can send DID extensions to the servers but not to the phones? |
| 08:19.48 | booray | does anyone suppose it's possible to never let certain agents leave a queue? |
| 08:19.48 | Waverly360 | booray: you can set up permanent queue members if you want |
| 08:19.53 | Waverly360 | booray: Just specify a SIP/XXX or ZAP/XXX instead of the agent name |
| 08:20.35 | booray | Waverly360: I found that specifying SIP/XXX initially places them in the queue when the daemon starts, however it will still let them log out |
| 08:20.49 | booray | Waverly360: unless I was on crack. it was a long night |
| 08:22.52 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 08:23.30 | booray | thx wav |
| 08:23.53 | Waverly360 | booray: I'm pretty sure they can't log out. Doing that pretty much sets the phone in the queue..not the actual agent. |
| 08:24.34 | Waverly360 | booray: Just keep in mind that having someone in the queue like that permanently will probably make the queue priorities worthless |
| 08:26.36 | booray | Waverly360: that's fine for this application.. basically so that the receptionist's phone always rings, but that other users can log in and out of the incoming call queue, strategy=ringall |
| 08:27.29 | *** join/#asterisk X-Rob (n=rob-x@124.150.99.11) |
| 08:28.04 | *** join/#asterisk af_ (n=getsmart@ip-179-53.sn1.eutelia.it) |
| 08:28.04 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 08:29.06 | Waverly360 | booray: Yeah, that sounds good. |
| 08:30.20 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 08:30.55 | Waverly360 | dlynes_laptop: I know you're busy...just wanted to let you know that I tried setting them up as registered peers. I'm getting the same circuit-busy congestion error. I've got something else going on weird here. I suppose I need to work that out. |
| 08:31.53 | *** join/#asterisk slima (i=slima@unaffiliated/slima) |
| 08:33.50 | *** join/#asterisk lorinc (n=ang@caracas-4955.adsl.interware.hu) |
| 08:34.34 | *** join/#asterisk mkl1525 (n=qwertz@pD95307A5.dip0.t-ipconnect.de) |
| 08:43.22 | *** join/#asterisk angryuser (n=Miranda@i03v-213-44-169-43.d4.club-internet.fr) |
| 08:43.28 | angryuser | i am searching someone familiar with indications.conf;) |
| 08:44.23 | angryuser | i have a problem with threewaycalling set to yes, and transfer set to yes, http://forums.digium.com/viewtopic.php?p=43796#43796 last messge of the topic describes my problem. |
| 08:47.34 | mkl1525 | Hi, I'm playing with call queues and ring strategy. Is it possible in rrmemory strategy to ring more than one phone at the same time if more than one caller is in the queue? |
| 08:49.07 | *** join/#asterisk FlatFoot (n=simon@80.88.192.113) |
| 08:49.21 | FlatFoot | morning all |
| 08:49.47 | FlatFoot | have just installed on freebsd version 1.2.13 can start using asterisk -cvvvvvvvvvvvvvvv |
| 08:49.59 | FlatFoot | but asterisk will not auto start |
| 08:50.13 | FlatFoot | it says Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) |
| 08:50.28 | FlatFoot | have googled read and tried all the info i can find |
| 08:50.37 | FlatFoot | anyone got any ideas ? |
| 08:52.10 | putzz | os? |
| 08:52.21 | tzafrir | <FlatFoot> have just installed on freebsd version 1.2.13 |
| 08:52.22 | FlatFoot | freebsd 6 |
| 08:52.31 | tzafrir | ps aux | grep asterisk |
| 08:52.33 | FlatFoot | no sorry * 1.2.13 |
| 08:52.41 | putzz | freebsd is the best ;-) |
| 08:52.45 | putzz | I got my boxes on bsd |
| 08:52.55 | tzafrir | A sanity check: is there actually an asterisk process running? |
| 08:53.28 | FlatFoot | no that seems to be the prob no process running |
| 08:53.36 | FlatFoot | sorry having a blonde day |
| 08:53.46 | tzafrir | FlatFoot, do you get any more specific error message? |
| 08:54.15 | FlatFoot | if i try to start * manually i just get the does the .ctl exist |
| 08:54.46 | FlatFoot | sorted it , sorry to bother |
| 08:54.59 | FlatFoot | the asterisk.sh file was missing from /var/run |
| 08:55.06 | FlatFoot | what a plank i am |
| 08:55.41 | putzz | do make install again |
| 08:55.44 | putzz | see if it will help |
| 08:55.50 | dlynes_laptop | FlatFoot: generally unless you supply specific parameters to asterisk startup |
| 08:56.00 | dlynes_laptop | FlatFoot: it expects you to run it as root user |
| 08:56.24 | FlatFoot | first time install on freebsd so i just followed the install instructions on the wiki |
| 08:59.56 | FlatFoot | ah well of to install a radio , thanks for your help |
| 09:02.21 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 09:04.03 | reivilo | Is someone try to set up musiconhold with FM card ? |
| 09:05.27 | mosty | mkl1525: no- asterisk queues suck :( |
| 09:05.59 | *** join/#asterisk Ethon (i=arne@Oldman.steinkamm.com) |
| 09:06.49 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 09:17.06 | canapa | is there a documentation how to install a AVM Fritz Card in Asterisk ? |
| 09:17.12 | *** join/#asterisk inspired (n=mikael@cl-330.sto-01.se.sixxs.net) |
| 09:23.49 | *** join/#asterisk Ahrimanes (n=ma@81.7.159.2) |
| 09:28.50 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 09:28.50 | *** mode/#asterisk [+o russellb] by ChanServ |
| 09:29.09 | angryuser | can anybody help me out ? http://forums.digium.com/viewtopic.php?p=43796#43796 something related with indications.conf |
| 09:29.53 | Ahrimanes | hey russellb |
| 09:30.14 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 09:30.33 | russellb | hey |
| 09:30.57 | key2 | russellb: what time is it for u ? |
| 09:31.02 | kippi | hey |
| 09:31.03 | russellb | 4:30 AM |
| 09:31.21 | key2 | what u doin up at 4:31 AM |
| 09:31.30 | russellb | woke up not feeling well :( |
| 09:31.34 | kippi | I am having problems getting music on hold to work, when I press hold I don't get any music on hold |
| 09:32.15 | kippi | Its just saying Stopped music on hold |
| 09:32.18 | Ahrimanes | russellb: mind if i query you in private with a few q's ? |
| 09:32.40 | russellb | Ahrimanes: it depends what it is about :) |
| 09:32.45 | key2 | russellb: sex |
| 09:32.50 | russellb | lol ... |
| 09:32.57 | Ahrimanes | key2: no that I only do with you... |
| 09:33.00 | key2 | russellb: thats why he doenst do it publicly |
| 09:33.03 | Ahrimanes | russellb: your sla work |
| 09:33.13 | russellb | ah, ok, sure |
| 09:33.30 | key2 | russellb: too bad, no sex |
| 09:35.54 | *** join/#asterisk xcla (n=info@static-b5-251-83.telepac.pt) |
| 09:35.58 | key2 | http://bithumen.ath.cx/bitbucket/slonsoset.gif |
| 09:36.00 | key2 | ;) |
| 09:36.49 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 09:36.49 | *** mode/#asterisk [+o denon] by ChanServ |
| 09:38.05 | kippi | anyone got ideas? |
| 09:44.50 | tzafrir | angryuser, have you copied the sample indications.conf to /etc/asterisk ? |
| 09:46.31 | angryuser | tzafrir:yes i am using the default one |
| 09:46.54 | tzafrir | what language did you set up? |
| 09:47.02 | angryuser | tzafrir:fr |
| 09:47.17 | tzafrir | And what do you see for fr in indications.conf? |
| 09:47.46 | tzafrir | for starters: sanity check that the section for fr appears only once: |
| 09:48.02 | tzafrir | grep '\[fr\]' /etc/asterisk/indications.conf |
| 09:48.16 | angryuser | tzafrir: i dont really understand the way indications are described |
| 09:48.36 | angryuser | tzafrir: il copy/paste fr section |
| 09:48.36 | tzafrir | You don't really need to edit it |
| 09:48.57 | tzafrir | but is there indeed only one? |
| 09:51.00 | angryuser | yes there is one fre section |
| 09:51.29 | *** join/#asterisk Aboulafia (n=adlp@shm67-4-82-242-214-214.fbx.proxad.net) |
| 09:51.33 | Aboulafia | Hi ! |
| 09:51.34 | angryuser | http://pastebin.ca/353378 here it is tzafrir: |
| 09:52.37 | tzafrir | angryuser, the language in zaptel.conf is irrelevant. Have you set the language in zapata.conf or in the dialplan? |
| 09:53.31 | angryuser | tzafrir: not sure which one describes the hook tone;), i have set the language only in zaptel.conf, not in zapata, or dialplan |
| 09:54.23 | *** part/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 09:54.26 | angryuser | tzafrir: il will try to set it now |
| 09:55.27 | Aboulafia | So, Ive problem, with IPPH 202 phone from the tiger compagny. Some time they register, but not always. So I can't dial to them. But they can dial with correct user/password... in sip debug i can see that they try to register, but they are unautorized... |
| 09:55.38 | angryuser | tzafrir: i set language=fr under [channels] in zapata.conf, got the same error |
| 09:55.40 | Aboulafia | (and one our before it was wporking... |
| 09:55.43 | Aboulafia | any idea ? |
| 09:56.30 | tzafrir | The hook tone is probably "dial" |
| 09:56.58 | yansolo90 | hey, anybody knows what login/password are for ssh Cisco 79XX (other than "debug/debug" or "log/log") ? |
| 09:57.11 | tzafrir | Aboulafia, hmmm, not sure I can help, but pastebin some configuration |
| 09:57.25 | angryuser | tzafrir: setting language in zapata doe not help much, the feature is not working under fxs, but for sip client it is working |
| 09:57.25 | Aboulafia | angryuser: french ? :) |
| 09:57.52 | angryuser | Aboulafia: i am living in france bur i am not french |
| 09:57.54 | Aboulafia | tzafrir: yep, but I do not know wich cfg I can give :] |
| 09:58.01 | tzafrir | yansolo90, they listen on port 22?? |
| 09:58.27 | Aboulafia | angryuser: ok :) (I'm a french who live in france, if you are at strasbourg ask for me, to have a drink ! :) |
| 09:58.50 | angryuser | Aboulafia: nice;) |
| 09:59.25 | *** join/#asterisk Dibbler_XP (n=Dibbler_@host217-45-198-229.in-addr.btopenworld.com) |
| 09:59.42 | Aboulafia | tzafrir: http://pastebin.ca/353382 |
| 09:59.59 | Aboulafia | angryuser: what a lovely town... |
| 10:00.03 | angryuser | tzafrir: maybe i should set language in dialplan? it is really pity if i am unable to get it working for fxs's |
| 10:00.24 | Aboulafia | angryuser: which kind of fxo card have you ? |
| 10:00.29 | angryuser | Aboulafia: i line in north, picardie;) |
| 10:00.42 | angryuser | *live |
| 10:01.31 | angryuser | Aboulafia: i have a problem it's a tone related, one button does no work;) |
| 10:01.34 | Aboulafia | angryuser: I've a TDM2400 with 1 FXO modules and 2 FXS, and @ home a TDM400P 3FXO, 1 FXS... connected on french line.. (Nice is a town in the south of france, sorry ;D ) |
| 10:02.09 | Aboulafia | angryuser: one button ? which one ? |
| 10:02.15 | angryuser | Aboulafia: R;) |
| 10:02.20 | tzafrir | angryuser, any chance it is is overriden? |
| 10:02.35 | tzafrir | can you print the language in the dialplan? |
| 10:02.58 | Aboulafia | angryuser: not a dtmf pb nor a language pb... |
| 10:03.22 | Aboulafia | the R is (in my littel memory) the flaash key, it's do on some phone a very fast hang up |
| 10:03.26 | tzafrir | Something of the sort of: exten => PATTERN,n,NoOp(language is ${LANGUAGE}) |
| 10:03.37 | angryuser | tzafrir: i have the feature working for sip, not on fxs's |
| 10:03.55 | Aboulafia | angryuser: have u try with other phone ? (Ive some phone with the R key who s woking and some other not...) |
| 10:03.56 | angryuser | tzafrir:ok il try that |
| 10:04.02 | *** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net) |
| 10:05.03 | Ahrimanes | mm beer |
| 10:05.46 | Aboulafia | angryuser: and if it s to do some line transfer, and other thing like it, have a look about the file feature.conf... muck luck that it work for all phone... |
| 10:05.55 | Aboulafia | Ahrimanes: at 11'o clock ? ;) |
| 10:06.10 | Ahrimanes | Aboulafia: anything after 9 is fine by me ;) |
| 10:06.20 | angryuser | Aboulafia: il will try with other phones later too |
| 10:06.48 | Aboulafia | angryuser: ok, and try to simulate the R, just for me please, a _VERY_ fast hang up :) |
| 10:07.00 | Aboulafia | Ahrimanes: euh.. well... nothing :) |
| 10:07.48 | tzafrir | yeah. A flash is essentially a short hangup |
| 10:08.01 | angryuser | Aboulafia:trafrir: cant reload dialplace, peoples talking;) |
| 10:08.11 | angryuser | *dialplan |
| 10:08.19 | angryuser | nedd to wait a bit |
| 10:08.28 | Aboulafia | angryuser: u can relaod dialplan while people talking : extension reload |
| 10:08.51 | Aboulafia | angryuser: I love command like stop when convinient :) |
| 10:08.52 | tzafrir | dialplan can safely be reloaded (unless you're afraid that your change is bad in itself) |
| 10:09.03 | tzafrir | extensions reload |
| 10:09.18 | tzafrir | (e<tab><tab><tab> or something similar) |
| 10:09.33 | Aboulafia | tzafrir: (ooopss I forget always the s :/ but I use so much of the completition |
| 10:10.14 | angryuser | tzafrir:nope, not working maybe deprecated;) |
| 10:10.58 | tzafrir | If you get a "deprecated" warning, it means that it is still working, but will get removed on next revision of Asterisk. |
| 10:11.26 | tzafrir | So get used to the newer form (what is it?) because on 1.6 this will be the only one you have. |
| 10:11.47 | angryuser | tzafrir:i med deprecated, like i have no messages at all;) 'no such command' extensions reload |
| 10:12.27 | angryuser | tzafrir:but whatever it is done |
| 10:13.20 | *** join/#asterisk DrCron (n=rszasz@c-67-174-231-152.hsd1.ca.comcast.net) |
| 10:13.28 | angryuser | tzafrir: the result is i have nothing set to ${LANGUAGE} |
| 10:15.07 | Aboulafia | Eyh it's my tuen now !! :D |
| 10:15.16 | Aboulafia | s/tuen/turn/g :) |
| 10:15.57 | Aboulafia | It's about my registratiojn pb with ipph202 !!!, they can phone (with their username), but they can't be phone ! |
| 10:17.20 | angryuser | tzafrir: i set this var LANGUAGE=fr in globals, get same error on test, Aboulafia: il will try with another phone, who knows |
| 10:21.34 | tzafrir_laptop | angryuser, this means that the channel's language wasn't set for some reason |
| 10:21.46 | tzafrir_laptop | Maybe incorrect order in zapata.conf or whatever |
| 10:24.15 | *** join/#asterisk susinths (n=susinths@uio-8021x-193-157-188-0-23-197.uio.no) |
| 10:24.26 | *** join/#asterisk nas_lslsa (n=chatzill@87.203.223.210) |
| 10:25.47 | angryuser | tzafrir_laptop: hm, in zapata.conf options must be repeated fo every channel or it is sufficient to write them down once under [channels]? |
| 10:25.56 | tzafrir_laptop | one. But putit near the beginning |
| 10:26.53 | angryuser | tzafrir_laptop: i did so, well i will try to test the feature on Sip phones, brb 15 min |
| 10:27.13 | *** join/#asterisk segal_wor (n=segal_wo@cuscon10658.tstt.net.tt) |
| 10:31.05 | nas_lslsa | angryuser ?? I am trying to make asterisk to work with sjphone - VoIP software , without any luck :( any ideas ? I don't even know if asterisk works okey with SIP |
| 10:31.43 | dj-fu | asterisk works perfectly with sip |
| 10:31.52 | dj-fu | it's just a matter of correctly configuring your sip.conf and the client |
| 10:35.10 | mkl1525 | Does anybody know when the autofill option was added to the queue app? |
| 10:37.15 | angryuser | nas_lslsa: descripe you problem |
| 10:39.35 | angryuser | nas_lslsa:in username under sjphone put not a username configured under sip.con but [whatyouhavetypedhere] |
| 10:39.46 | nas_lslsa | sorry , just a sec :/ |
| 10:39.50 | angryuser | nas_lslsa:on the top of sip user |
| 10:41.05 | nas_lslsa | okey now , |
| 10:41.29 | *** join/#asterisk frogzoo (n=frogzoo@202.155.165.25) |
| 10:41.45 | nas_lslsa | I made several tries ... is there any way to find for example if on asterisk linux the port is open ? |
| 10:42.02 | nas_lslsa | to do telnet 192.168.1.80 5060 |
| 10:42.28 | hads | UDP... |
| 10:42.35 | nas_lslsa | I don't get any error message , like " Wrong username /passwd " or something .. |
| 10:42.42 | angryuser | tzafrir_laptop:i got some news on sip phones test, well actually R puts a person onhold, and opens another line, nothing more, when the conversatio woth other person is finished, asterisk calls you back and put you back on relation with that phone, maybe i need to deactivate the feature |
| 10:42.49 | nas_lslsa | or _ can't connect .. |
| 10:42.56 | nas_lslsa | hads yep |
| 10:43.17 | dj-fu | nas_lslsa, nc -uvv 192.168.1.80 5060 |
| 10:43.26 | angryuser | tzafrir_laptop:you got wrong/username/passwer under CLI? |
| 10:43.41 | tzafrir_laptop | me? |
| 10:43.41 | angryuser | *password |
| 10:43.51 | angryuser | tzafrir_laptop: oops sorry;) |
| 10:44.09 | angryuser | nas_lslsa: him:) |
| 10:48.11 | mitcheloc | digium.com is down? =( |
| 10:50.41 | *** join/#asterisk ctooley (n=ctooley@tx-71-51-44-36.dhcp.embarqhsd.net) |
| 10:53.14 | *** join/#asterisk Charles[NS] (n=neouf@LSt-Amand-152-31-36-33.w82-127.abo.wanadoo.fr) |
| 10:56.14 | *** join/#asterisk DrCron (n=rszasz@c-67-174-231-152.hsd1.ca.comcast.net) |
| 11:01.21 | Renacor | can somebody help me with this: [Feb 13 02:42:30] WARNING[2855]: app_dial.c:1081 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) |
| 11:07.11 | angryuser | modprobe zaptel give what? and ztcfg -vvv no errors? Renacor: |
| 11:09.36 | tzafrir | Renacor, what type of Zap is it? Waht hardware? |
| 11:10.17 | dlynes_laptop | mitcheloc: they continually have issues with their dns server |
| 11:10.18 | tzafrir | Where did you dial to? What channel (Zap/WHAT_EXATLY)? |
| 11:10.32 | dlynes_laptop | mitcheloc: try ftp.digium.com, ftp1.digium.com, ftp2.digium.com |
| 11:14.45 | Ahrimanes | we need to wake up russellb at 4.30 every mrning |
| 11:14.47 | Ahrimanes | morning |
| 11:15.31 | russellb | Ahrimanes: svn update, i made one more small change in rev 54174 |
| 11:15.42 | *** join/#asterisk EyeCue (n=eyecue@miranda/admin/eyecue) |
| 11:15.48 | Renacor | angryuser: zaptel is already there znd ztcfg -vvvv gives no errors but shows 4 channels configured |
| 11:15.59 | Renacor | tzafrir: Its a TDM400P |
| 11:16.12 | Ahrimanes | russellb: oh ok.. need to redo compile anyways |
| 11:17.47 | tzafrir | BTW: does devstate work with zaptel? If not: any idea what it would take to make it work? |
| 11:18.09 | russellb | well, it works by default to some degree with any channel |
| 11:18.13 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:18.15 | dlynes_laptop | russellb: Just thought you might like to know that zaptel 1.2.13 and asterisk 1.2.15 have broken compatibility with a number of things |
| 11:18.24 | tzafrir | ok, I'll have to look into it |
| 11:18.35 | russellb | if the channel driver doesn't provide its own way to determine device state, then asterisk will look for channels with that name that exist |
| 11:18.43 | russellb | for basic INUSE / NOT_INUSE ... |
| 11:18.50 | russellb | i think maybe RINGING too ... |
| 11:18.59 | russellb | dlynes_laptop: um, that's not good. |
| 11:19.04 | Ahrimanes | what have i started.. hehe |
| 11:19.19 | dlynes_laptop | russellb: zaptel 1.2.12 will not work with asterisk 1.2.15 |
| 11:19.20 | russellb | dlynes_laptop: if that's the case, we need reports on bugs.digium.com ... |
| 11:19.32 | russellb | dlynes_laptop: oh, well yeah ... |
| 11:19.37 | russellb | i guess we forgot to mention that :-/ |
| 11:19.42 | dlynes_laptop | russellb: sangoma wanpipe 2.3.4-7 will not work with zaptel 1.2.13 |
| 11:19.53 | dlynes_laptop | russellb: zaptel 1.2.12 will not work with linux 2.6.20 |
| 11:20.01 | russellb | um. |
| 11:20.07 | russellb | zaptel 1.2.12 came out before linux 2.6.20 |
| 11:20.15 | russellb | what do you want me to do about that? |
| 11:20.18 | dlynes_laptop | shurg |
| 11:20.28 | dlynes_laptop | They're both minor version upgrades |
| 11:20.41 | russellb | linux breaks the API a whole lot more than we do :) |
| 11:20.47 | dlynes_laptop | ah..i c |
| 11:21.00 | russellb | they change stuff allllllllll the time |
| 11:21.03 | russellb | it's quite a pain, actually |
| 11:21.10 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 11:21.16 | russellb | but, such is life for out of tree kernel modules |
| 11:21.22 | dlynes_laptop | Yeah..the whole zaptel/asterisk/linux version upgrade was a huge pain in the ass for me today |
| 11:21.35 | dlynes_laptop | Wasted about 4 or 5 hours sorting out the incompatibility issues |
| 11:21.40 | russellb | sorry to hear that |
| 11:21.44 | russellb | the latest of everything should work |
| 11:21.48 | russellb | of asterisk/zaptel/linux |
| 11:21.57 | russellb | i can't do anything abuot wanpipe, obviously. |
| 11:22.04 | dlynes_laptop | Yeah...it works as long as you don't use wanpipe |
| 11:22.11 | dlynes_laptop | Nod, I can understand that |
| 11:22.20 | russellb | guess you shouldn't be using it :-p |
| 11:22.22 | dlynes_laptop | And you're probably happy that it doesn't work, too :) |
| 11:22.32 | russellb | it certainly was not on purpose |
| 11:22.39 | dlynes_laptop | Yeah, I realize taht |
| 11:22.43 | dlynes_laptop | but otoh... |
| 11:22.43 | Ahrimanes | core show function DEVSTATE |
| 11:22.44 | Ahrimanes | :) |
| 11:22.53 | russellb | but, it's just like when linux API breaks our stuff |
| 11:23.06 | russellb | out of tree comes with a price |
| 11:23.07 | dlynes_laptop | zaptel 1.2.12 to zaptel 1.2.13 seems like a huge upgrade though, too |
| 11:23.12 | russellb | it is |
| 11:23.17 | russellb | for a lot of reasons |
| 11:23.26 | dlynes_laptop | zaptel 1.2.13 was more than double the size of 1.2.12 |
| 11:23.35 | russellb | because of firmware files, i guess |
| 11:23.41 | dlynes_laptop | ah... |
| 11:23.49 | russellb | it includes support for some new produces |
| 11:23.52 | russellb | er, products |
| 11:23.53 | dlynes_laptop | nobody seemed to be able to give me a straight answer on that :) |
| 11:24.07 | dlynes_laptop | I guess it must be for all the new octasic support |
| 11:24.13 | russellb | yeah, that |
| 11:24.17 | russellb | as well as the new echo canceller |
| 11:24.21 | russellb | the transcoder board ... |
| 11:24.35 | dlynes_laptop | and probably improved routines in the HPEC |
| 11:24.50 | russellb | right |
| 11:24.58 | russellb | that thing is supposed to really rock ... |
| 11:25.14 | Ahrimanes | HPEC ? |
| 11:25.14 | *** join/#asterisk kanelbullar (n=kanelbul@83.240.200.92) |
| 11:25.15 | dlynes_laptop | HPEC is new as of 1.2.11 I think, right? |
| 11:25.18 | russellb | as far as software echocan goes |
| 11:25.24 | russellb | HPEC is new as of 1.2.13 |
| 11:25.24 | dlynes_laptop | Ahrimanes: the new high performance software echo canceller |
| 11:25.28 | Ahrimanes | codec_zap is used for transcoder board? |
| 11:25.31 | Ahrimanes | dlynes_laptop: ah |
| 11:25.35 | russellb | Ahrimanes: yes |
| 11:25.43 | Ahrimanes | ok |
| 11:25.51 | dlynes_laptop | Ahrimanes: it's supposed to be considerably better than the free one built into zaptel |
| 11:26.03 | dlynes_laptop | Ahrimanes: and I'm guessing for that reason it's not free |
| 11:26.05 | russellb | it's free to all digium customers |
| 11:26.13 | Ahrimanes | hehe |
| 11:26.17 | dlynes_laptop | russellb: to all current digium customers :) |
| 11:26.27 | russellb | right. |
| 11:26.28 | dlynes_laptop | russellb: out of warranty digium customers aren't covered |
| 11:26.35 | russellb | that is true. |
| 11:26.50 | dlynes_laptop | But still |
| 11:26.54 | dlynes_laptop | It sounds quite promising |
| 11:27.01 | russellb | yep, worth giving a try |
| 11:27.07 | dlynes_laptop | Free or not so free |
| 11:27.08 | russellb | i'd be interested in hearing what people thing of it |
| 11:27.16 | russellb | think* |
| 11:27.21 | Renacor | which port is supposed to connect the phone line coming into the building? fxo or fxs? |
| 11:27.28 | russellb | Renacor: fxo |
| 11:27.52 | Renacor | so in zapata.conf its fxo and same for zaptel.conf? |
| 11:28.02 | dlynes_laptop | Renacor: fxs |
| 11:28.03 | kanelbullar | Hi guys, anybody familiar with blocking collect calls in Brazil? |
| 11:28.04 | russellb | actually... |
| 11:28.10 | dlynes_laptop | Renacor: fxo ports use fxs signalling and vice-versa |
| 11:28.14 | russellb | it's an fxo port, but it t an fxo port uses fxs signalling :) |
| 11:28.21 | russellb | so, you configure it as fxs in those files ... |
| 11:28.29 | Renacor | yeah thats what I thought |
| 11:28.41 | Renacor | so zaptel.conf is fxs and so is zapata.conf |
| 11:28.44 | russellb | it's nice and confusing |
| 11:28.48 | Renacor | yeah |
| 11:28.51 | kanelbullar | Should the dialplan "Flash" instruction work well for blocking collect calls in Brazil, using Digital cards? |
| 11:29.15 | Renacor | Im still getting [Feb 13 03:20:40] WARNING[2902]: app_dial.c:1081 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) |
| 11:29.40 | dlynes_laptop | Renacor: does asterisk start up? |
| 11:29.50 | Renacor | dlynes_laptop: yep |
| 11:29.54 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 11:30.09 | dlynes_laptop | Renacor: and do you have chan_zap.so loaded in your modules.conf, or have autoload=yes in your modules.conf file? |
| 11:30.28 | russellb | alright, it has been a fun few hours of insomnia. back to sleep for me ... |
| 11:30.30 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 11:30.38 | dlynes_laptop | russellb: have a good sleep |
| 11:30.44 | dlynes_laptop | russellb: I wish I could do the same :) |
| 11:30.57 | Renacor | dlynes_laptop: no chan_zap.so in modules.conf but I have autoload=yes |
| 11:31.21 | dlynes_laptop | Renacor: ok, what happens when you do unload chan_zap.so |
| 11:31.26 | dlynes_laptop | Renacor: and then load chan_zap.so? |
| 11:31.56 | Renacor | here we go |
| 11:32.03 | Renacor | [Feb 13 03:31:37] WARNING[2919]: config.c:737 process_text_line: parse error: No category context for line 10 of zapata.conf |
| 11:32.09 | Renacor | [Feb 13 03:31:37] ERROR[2919]: chan_zap.c:11107 setup_zap: Unable to load config zapata.conf |
| 11:32.20 | dlynes_laptop | Renacor: there ya go...so you've got a buggy zapata.conf file |
| 11:32.20 | Renacor | much better =D |
| 11:32.26 | dlynes_laptop | Renacor: you haven't specified a context |
| 11:32.43 | dlynes_laptop | Renacor: you need a line like 'context=incoming', or something simliar |
| 11:32.50 | dlynes_laptop | s/simliar/similar/ |
| 11:33.07 | Renacor | yeah I have that |
| 11:33.12 | Renacor | context=ext_germany |
| 11:33.20 | dlynes_laptop | take a look at line 10 then |
| 11:33.21 | Renacor | lemme pastebin it |
| 11:33.29 | dlynes_laptop | pastebin your zapata.conf file in its entirety |
| 11:34.06 | Renacor | http://pastebin.ca/353457 |
| 11:34.19 | angryuser | Is it possible for user to dial some sequence and by it enter ins state "do no disturb"? |
| 11:34.46 | Renacor | one sec thats not the entire file |
| 11:35.05 | Renacor | dlynes_laptop: http://pastebin.ca/353459 |
| 11:35.30 | Renacor | so line 10 is signalling=fxs_ks |
| 11:37.26 | Ahrimanes | on ami, with action: command.. how do i get a noop of a dialplan functions output? |
| 11:38.46 | *** join/#asterisk foobar778 (n=johhny@ip72-196-234-182.dc.dc.cox.net) |
| 11:39.11 | dlynes_laptop | Renacor: add the following lines before # |
| 11:39.11 | dlynes_laptop | ; Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1" |
| 11:39.11 | dlynes_laptop | # |
| 11:39.17 | dlynes_laptop | doh |
| 11:39.26 | dlynes_laptop | Anyways...add it before the Span 1 comment |
| 11:39.30 | dlynes_laptop | [trunkgroups] |
| 11:39.33 | dlynes_laptop | [channels] |
| 11:39.58 | dlynes_laptop | and then move your context= line up so that it's right after '[channels]' |
| 11:40.26 | foobar778 | hi does anyone have strong networking knowledge I need some help maybe ten minutes can call u to ur pbx or to ur pstn whatever |
| 11:42.02 | angryuser | how can i activa DND for user state |
| 11:42.05 | angryuser | ? |
| 11:42.35 | foobar778 | My issue is that my asterisk is working I have three users they can all ring each other across the lan but I have to put my adapter ata in bridge mode in non bridghe mode and in dmz doesnt work has me stumped |
| 11:42.53 | *** join/#asterisk morex (n=m@81.168.125.119) |
| 11:43.04 | morex | Hey, I've got a problem with ISDN I wonder if anyone's seen before |
| 11:43.19 | morex | I've got Asterisk sitting between the ISDN cable in the wall and a call centre's PBX |
| 11:43.26 | morex | With a TE205P |
| 11:43.28 | dlynes_laptop | angryuser: using the dnd button on your phone |
| 11:43.34 | morex | When people try to dial out through Asterisk |
| 11:43.46 | dlynes_laptop | angryuser: dnd is usually a function of your phone |
| 11:43.50 | morex | Asterisk is picking up just the first four digits and interpreting this as the extension |
| 11:43.56 | morex | rather than the whole number dialled |
| 11:44.00 | foobar778 | dlynes_laptop remeber me from earlier |
| 11:44.10 | morex | We think Asterisk is timing out the extension read too quickly |
| 11:44.33 | dlynes_laptop | foobar778: yeah, and I'm still too busy to spend about 1/2 hour or so to debug your system |
| 11:44.34 | morex | Anybody know how to increase the timeout in zapata.conf? |
| 11:44.57 | angryuser | dlynes_laptop: in wiki it's *78 , and if my phone have no this function? |
| 11:45.02 | foobar778 | need 10 minutes can I call u pbx or landline?? |
| 11:45.07 | dlynes_laptop | foobar778: nothing ever takes ten minutes |
| 11:45.16 | morex | Oh, and why is the Digium web site down? |
| 11:45.18 | *** join/#asterisk tg (i=tg@x-net.hu) |
| 11:45.19 | foobar778 | just a few basic questions |
| 11:45.42 | foobar778 | 10 minutes max promise |
| 11:46.07 | dlynes_laptop | angryuser: *78 is a vertical service code |
| 11:46.16 | dlynes_laptop | angryuser: and it might be implemented on zaptel channels |
| 11:46.20 | dlynes_laptop | angryuser: not sip channels |
| 11:47.06 | angryuser | dlynes_laptop: ok, which conf, file list all of them?(functions) |
| 11:48.23 | ez` | is it possible to add time stamp to CLI asterisk ? |
| 11:48.25 | Renacor | dlynes_laptop: I thinkthat fixed the problem :) |
| 11:49.15 | Renacor | now I just gotta figure out which port im supposed to be using heh |
| 11:50.14 | angryuser | is there any list of service codes for asterisk 1.4? |
| 11:50.53 | *** join/#asterisk Ebola (n=Ebola@host86-142-178-37.range86-142.btcentralplus.com) |
| 11:59.53 | dlynes_laptop | angryuser: asterisk doesn't have service codes |
| 12:00.06 | dlynes_laptop | angryuser: some of the connecting technologies may or may not have service codes though |
| 12:00.22 | dlynes_laptop | angryuser: zaptel for instance has some service codes |
| 12:05.24 | susinths | hey guys |
| 12:05.47 | susinths | im planning to set up ast + ser, should i use SER or openSER? |
| 12:05.53 | *** join/#asterisk ComaVN (n=blaargh@unaffiliated/comavn) |
| 12:06.10 | Renacor | whats ser? |
| 12:06.16 | Renacor | sorry just curious |
| 12:06.38 | florz | ser is nothing, everything is hir! |
| 12:06.46 | Renacor | Hehe |
| 12:06.53 | Renacor | sip server eh? |
| 12:07.02 | florz | But apart from that, yep, Sip Express Router |
| 12:07.30 | florz | SIP proxy and registration server, to be exact |
| 12:08.18 | *** join/#asterisk apardo (n=apardo@87.217.144.14) |
| 12:10.46 | Makenshi | just got my new black box catalogue.. they have a 1-port analogue and 2-port digital ata |
| 12:10.57 | tg | hey |
| 12:11.49 | endre | tg? rly? |
| 12:12.01 | tg | endre: o/ |
| 12:12.04 | endre | o/ |
| 12:12.56 | tg | anybody tried atcom ip phones? |
| 12:14.58 | *** join/#asterisk lukketto (n=lukketto@host47-159-dynamic.6-87-r.retail.telecomitalia.it) |
| 12:15.58 | *** join/#asterisk penguinFunk (n=penguin@87.224.86.46) |
| 12:22.48 | markit | hi, loading the dialplan I've noticed the message: WARNING[13476]: chan_iax2.c:9119 set_config: Section 'guest' lacks type, but I've [guest] with type=user (* 1.4svn) in iax.conf, any clue? |
| 12:24.02 | dlynes_laptop | Makenshi: i bet they charge an arm and a leg for them too, right? |
| 12:26.13 | Makenshi | yeah, 66 and 125 gbp respectively |
| 12:26.35 | dlynes_laptop | Double that to get USD? |
| 12:27.13 | Makenshi | well i think the prices in usa are different |
| 12:27.19 | Makenshi | they're selling a lot of avaya stuff too |
| 12:27.27 | dlynes_laptop | Yeah...avaya is lucent |
| 12:27.39 | dlynes_laptop | i.e. Bell Labs |
| 12:27.44 | Makenshi | and hipath |
| 12:27.51 | dlynes_laptop | Siemens |
| 12:27.53 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 12:28.00 | Makenshi | yep :o) |
| 12:28.10 | dlynes_laptop | Hipath isn't really used much in North America |
| 12:28.23 | dlynes_laptop | We had a Siemens rep trying to get us to sell it a few years back |
| 12:28.36 | dlynes_laptop | we basically told him to hit the road :) |
| 12:29.28 | Makenshi | they're doing everything.. nortel and alcatel too o.o |
| 12:29.36 | Makenshi | .. and mitel |
| 12:37.03 | *** join/#asterisk giasai68 (n=giasai@ip-240-130.sn2.eutelia.it) |
| 12:40.45 | *** join/#asterisk squall (n=squall@ns2.squallnetwork.net) |
| 12:45.13 | *** join/#asterisk Jared_Leto (n=Lostprop@80-89-104-241.DSL.ycn.com) |
| 12:48.30 | *** join/#asterisk coppice (n=chatzill@118.202.17.210.dyn.pacific.net.hk) |
| 12:51.13 | *** join/#asterisk droemel (n=droemel@p548EB7A3.dip0.t-ipconnect.de) |
| 12:53.16 | dlynes_laptop | Makenshi: Nortel can't be ignored...same with Mitel and Panasonic |
| 12:53.43 | dlynes_laptop | Makenshi: Nortel's presence in Canada is quite strong; Mitel's presence in the motel industry is quite strong, and Panasonic is quite strong all around |
| 12:54.47 | coppice | nortel is the world's biggest switch maker. hard to ignore them if you are in telecoms :-) |
| 12:55.48 | dlynes_laptop | yep |
| 12:55.54 | dlynes_laptop | besides their former parent company |
| 12:56.15 | dlynes_laptop | Has significant ownership in most of the important world telcos |
| 12:58.34 | dlynes_laptop | well...actually lemme reword that |
| 12:58.55 | dlynes_laptop | in about 8 or 9 large international telcos |
| 12:59.22 | dlynes_laptop | including bt and hk t&t |
| 13:01.50 | Renacor | how do you handle incoming calls in asterisk |
| 13:01.59 | Renacor | do you have to create a special context or something? |
| 13:02.25 | dlynes_laptop | Renacor: all channel drivers require their configuration files to have some kind of directive to specify a context |
| 13:02.47 | dlynes_laptop | Renacor: it is this context that specifies the dialplan context that all calls on that channel are dropped into |
| 13:03.28 | Renacor | i see |
| 13:03.36 | yansolo90 | hey, anybody knows what login/password are for ssh Cisco 79XX (other than "debug/debug" or "log/log") ? |
| 13:08.54 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 13:16.53 | *** join/#asterisk potsboy (n=chrisg@196.211.16.202) |
| 13:17.13 | *** join/#asterisk alephant (n=cmd@c-24-3-52-93.hsd1.mn.comcast.net) |
| 13:17.17 | alephant | Morning all! |
| 13:17.36 | Renacor | dlynes_laptop: thanks for all your help I think i got it going now =) |
| 13:17.37 | alephant | Anybody have any experience with integrating * with Centrex systems? |
| 13:18.15 | alephant | I have a customer who wants to continue using the flash-hook transfer feature of his Centrex lines... |
| 13:18.52 | alephant | ...specifically, he claims that his Centrex system was able to transfer incoming calls such that the main line (with his main phone number) would be freed up after the transfer |
| 13:19.19 | dlynes_laptop | alephant: yeah...that's not a centrex feature, really...it's just a feature you can have on your analog lines...it's called 'link' |
| 13:19.32 | alephant | !!! |
| 13:19.36 | dlynes_laptop | alephant: there's a dialplan app for it called 'flash' |
| 13:19.36 | alephant | okokokokok |
| 13:19.48 | alephant | tell me about this "link" first... |
| 13:20.01 | dlynes_laptop | alephant: exactly what you already stated |
| 13:20.03 | *** join/#asterisk xF|DarkSC (n=dark@server.vinylkind.de) |
| 13:20.06 | xF|DarkSC | hi |
| 13:20.06 | alephant | is there any doc you can point me at, or should I just start the deluge of questions? |
| 13:20.22 | dlynes_laptop | alephant: a call comes in, you hit link, you dial another number, you hit link again, your line is freed up |
| 13:20.26 | xF|DarkSC | anyone else having problems with sipgate.de? |
| 13:20.44 | dlynes_laptop | alephant: and you don't have an incoming line or outgoing line used up |
| 13:21.16 | alephant | and the dialplan app for "link" is called "flash"? |
| 13:21.30 | dlynes_laptop | alephant: head on over to voip-info.org, click on asterisk on the left hand side, then scroll down until you see 'application commands', click on that, then scroll down until you see the application 'flash' |
| 13:21.38 | dlynes_laptop | alephant: correct |
| 13:21.39 | alephant | will do |
| 13:21.50 | alephant | thanks so much! Talk about the right place at the right time! |
| 13:22.22 | dlynes_laptop | alephant: yeah...the link feature is quite useful |
| 13:22.46 | alephant | so what is "flash" then? |
| 13:22.48 | dlynes_laptop | alephant: one gotcha though...the flash application is not in active development I don't think, and so it apparently has issues on some systems |
| 13:23.13 | dlynes_laptop | alephant: on some phone systems it's called 'flash', and on others it's called 'link' |
| 13:23.21 | alephant | Ah. So it's the same thing. |
| 13:23.29 | dlynes_laptop | alephant: i think the phone company officially calls it 'flash' |
| 13:23.31 | alephant | Just that it's not a feature of Centrex per se. |
| 13:23.34 | *** join/#asterisk nitram (i=foo@superblob.com) |
| 13:23.58 | dlynes_laptop | alephant: No...it's a centrex feature, but not a centrex line feature |
| 13:24.18 | dlynes_laptop | alephant: i.e. the phone company has that feature available, but it's on analog lines |
| 13:24.25 | alephant | so it's a Centrex implementation of a feature available on analog lines |
| 13:24.27 | alephant | gotcha |
| 13:24.34 | dlynes_laptop | and it's done at the central switch |
| 13:24.49 | alephant | whereas the Centrex version is done at the Centrex hardware? |
| 13:25.24 | dlynes_laptop | alephant: well, the centrex line version, you need centrex lines for |
| 13:25.31 | dlynes_laptop | alephant: and centrex phones |
| 13:26.03 | dlynes_laptop | alephant: in Canada...I don't know about elsewhere, those phones are always Nortel Meridian phones |
| 13:26.51 | dlynes_laptop | and that's because the central switches at the telcos here are all nortel equipment |
| 13:29.12 | *** join/#asterisk chiang_sg (i=chiang_s@bb219-74-175-228.singnet.com.sg) |
| 13:30.10 | *** join/#asterisk zamba (i=marius@flage.org) |
| 13:30.10 | alephant | Ok... so this Flash() is what I've already implemented... |
| 13:30.28 | zamba | i'm trying to figure out what the different alarms in the zapata driver means |
| 13:30.28 | alephant | ...the voip-info doc has an example using a macro for an extension |
| 13:30.40 | zamba | red/yellow/blue/recovering/no error and so on |
| 13:30.48 | zamba | what do they refer to? |
| 13:31.06 | alephant | I implemented it as the result of the "flash" command (since their softphones and old Centrex phones have "flash" buttons) |
| 13:31.25 | alephant | so their dialing *7 now gives them dialtone and puts the other party on hold |
| 13:31.46 | dlynes_laptop | red means something's desperately wrong, yellow means you've got a configuration error, recovering usually means it's coming up, and i have no idea what blue is...'no error' should be pretty self explanatory |
| 13:32.23 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:32.43 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:33.08 | zamba | dlynes_laptop: configuration error? zapata.conf, in other words? |
| 13:33.17 | dlynes_laptop | zamba: or zaptel.conf or both |
| 13:33.23 | zamba | at the moment i have yel/rec |
| 13:33.26 | dlynes_laptop | zamba: generally zaptel.conf though |
| 13:33.30 | zamba | Digium Wildcard TE110P T1/E1 Card 0 |
| 13:33.33 | dlynes_laptop | zamba: on a pri channel? |
| 13:33.38 | zamba | .. i think so? :) |
| 13:33.48 | dlynes_laptop | zamba: yeah...issue is almost guaranteed to be in your zaptel.conf configuation |
| 13:33.56 | dlynes_laptop | s/configuation/configuration/ |
| 13:34.38 | zamba | what's the difference between zapata and zaptel? |
| 13:34.53 | zamba | i only have zapata.conf in /etc/asterisk |
| 13:35.02 | dlynes_laptop | zamba: zaptel.conf is in /etc |
| 13:35.04 | zamba | zaptel.conf is in /etc? hm.. |
| 13:35.05 | zamba | oh, ok |
| 13:35.10 | dlynes_laptop | zamba: zaptel is the driver configuration |
| 13:35.17 | dlynes_laptop | zamba: zapata is the channel driver configuration |
| 13:35.18 | zamba | not much in there |
| 13:35.27 | zamba | but most probably zaptel.conf? |
| 13:35.34 | dlynes_laptop | zamba: correct |
| 13:35.36 | zamba | now i've only got RED |
| 13:35.44 | zamba | we did some rewiring of the pairs on the cable :) |
| 13:35.49 | zamba | it's probably that? |
| 13:35.54 | dlynes_laptop | zamba: you probably haven't configured your t1/pri parameters |
| 13:36.04 | dlynes_laptop | zamba: or that, too |
| 13:36.38 | dlynes_laptop | zamba: anyways...blitzrage might be able to help you more... |
| 13:36.43 | dlynes_laptop | zamba: I've gotta run shortly |
| 13:37.01 | alephant | dlynes_laptop: I think I know what we were doing wrong... |
| 13:37.22 | dlynes_laptop | alephant: working on too little sleep? |
| 13:37.24 | alephant | dlynes_laptop: I think the transferring party needs to HANG UP in order to complete the transfer :-| |
| 13:37.39 | dlynes_laptop | on the link? |
| 13:37.41 | alephant | alephant: Not at all. Just woke up, feel great! |
| 13:37.47 | alephant | dlynes_laptop: yep. |
| 13:37.56 | dlynes_laptop | alephant: yeah...got you mixed up with zamba :) |
| 13:38.04 | chiang_sg | hi anyone has a script that will detect of asterisk is down(crash) then autostart again ? |
| 13:38.22 | dlynes_laptop | alephant: perhaps...with normal phone systems |
| 13:38.30 | dlynes_laptop | alephant: you just do the link, then do a dial |
| 13:38.34 | dlynes_laptop | alephant: and then do link again |
| 13:38.47 | *** join/#asterisk lyGP (n=l@60.51.217.146) |
| 13:38.53 | dlynes_laptop | alephant: erm wait...or maybe it was hangup |
| 13:38.57 | alephant | I've gotta wait until these guys wake up to test it |
| 13:39.08 | alephant | flash --> dial --> flash just toggles between the two calls |
| 13:39.17 | dlynes_laptop | alephant: nah...flash dial flash |
| 13:39.17 | zamba | dlynes_laptop: i set the t1/pri parameters in /etc/zaptel.conf? |
| 13:39.24 | zamba | or zapata.conf? :) |
| 13:39.26 | alephant | so in a coupla hours I can actually test it |
| 13:39.29 | lyGP | i plug and directline into the fxo port |
| 13:39.34 | dlynes_laptop | zamba: zaptel.conf |
| 13:39.43 | lyGP | when i call to the line number, the extension just cut off |
| 13:39.44 | zamba | one sec, pastebin coming up |
| 13:39.48 | alephant | dlynes_laptop: you've "gotta run shortly"... will you be around later? |
| 13:39.49 | lyGP | i wish to knwo whwat is the problem? |
| 13:40.03 | dlynes_laptop | zamba: you should have a sample zaptel.conf file with zaptel |
| 13:40.10 | zamba | http://www.pastebin.ca/353603 |
| 13:40.14 | zamba | that's all i've got in zaptel.conf |
| 13:40.15 | dlynes_laptop | zamba: please don't expect me to read the pastebin |
| 13:40.20 | potsboy | Hi all, is anybody aware of a app_devstate type of equivilent for asterisk 1.4.. What I am trying to achieve is to be able to illuminate leds on a snom on demand? |
| 13:40.22 | zamba | dlynes_laptop: ok, i'm sorry :) |
| 13:40.23 | dlynes_laptop | zamba: as I said...I'm on my way out |
| 13:40.28 | zamba | ok |
| 13:40.33 | zamba | thanks for your help anyway :) |
| 13:40.47 | dlynes_laptop | anyways |
| 13:40.52 | dlynes_laptop | have a good morning peeps |
| 13:40.54 | dlynes_laptop | i'm out of here |
| 13:41.15 | dlynes_laptop | I'll be glad after I get this system installed |
| 13:41.23 | dlynes_laptop | Then I can go home again and go to sleep :) |
| 13:41.33 | dlynes_laptop | Pulled an all-nighter tonight :( |
| 13:45.54 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 13:46.31 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 13:46.58 | lyGP | i plug and directline into the fxo port |
| 13:46.58 | lyGP | when i call to the line number, the extension just cut off |
| 13:46.59 | lyGP | it wont go into asterisk IVR |
| 13:47.00 | lyGP | but terminate the call |
| 13:47.00 | lyGP | anybody can help me |
| 13:47.00 | lyGP | i'm using a ICSTEL VoiP Gateway |
| 13:54.24 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 14:08.14 | lyGP | exten => s,1,Answer ; Answer the line |
| 14:08.15 | lyGP | exten => s,2,DigitTimeout,5 ; Set Digit Timeout to 5 seconds |
| 14:08.15 | lyGP | exten => s,ResponseTimeout,10 ; Set Response Timeout to 10 seconds |
| 14:08.16 | lyGP | exten => s,4,BackGround(demo-congrats) |
| 14:08.31 | *** join/#asterisk dsphunxion (n=sil@the.internet.is.broken.com) |
| 14:08.39 | lyGP | this command seems like not working to anser my fxo port |
| 14:08.48 | lyGP | any help? |
| 14:09.52 | mosty | well the third line is broken, it doesn't have a priority, so the call dies there and doesn't get to priority 4 |
| 14:11.11 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net) |
| 14:11.23 | *** join/#asterisk foobar778 (n=bryan@ip68-100-41-120.dc.dc.cox.net) |
| 14:11.41 | foobar778 | Yea I got it working!!!!!!!!! |
| 14:11.45 | lyGP | mosty: there is a s,3, responsetimeout,10 |
| 14:12.09 | mosty | lygp: what does the console show, with debug and verbose set to 10, when a call comes in? |
| 14:13.14 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 14:13.27 | foobar778 | if anyone followed my plighthe reso;ution on the router evn though dmz was set the ports still needed to be fowarded apparently just dmz doesnt pass all through atleast with wrt54g with ddwrt firmware |
| 14:14.33 | *** join/#asterisk docelmo (n=vircuser@c-68-32-61-251.hsd1.de.comcast.net) |
| 14:17.34 | foobar778 | can anyone test a call |
| 14:17.57 | foobar778 | I mean register to my pbx and call an extension?? |
| 14:18.31 | canapa | well |
| 14:18.35 | canapa | host |
| 14:18.49 | foobar778 | want my host?? |
| 14:19.04 | canapa | if i wana register to it? |
| 14:19.07 | foobar778 | canap talking to me? |
| 14:19.12 | foobar778 | yes |
| 14:19.18 | foobar778 | <PROTECTED> |
| 14:19.27 | foobar778 | 712 |
| 14:19.31 | foobar778 | boy125 |
| 14:19.32 | endre | www.whatismyip.org |
| 14:19.49 | foobar778 | dial 7777 |
| 14:20.03 | foobar778 | endre??? |
| 14:20.16 | tzafrir_laptop | lyGP, working? |
| 14:20.20 | endre | nevermind |
| 14:20.33 | endre | i tought u wanted to know your ip, sorry |
| 14:20.34 | tzafrir_laptop | lyGP, is the space before the comma also in the original? |
| 14:20.54 | lyGP | nope |
| 14:21.30 | mosty | lygp: can you answer my previous question? |
| 14:21.40 | lyGP | hold on, i'm checking |
| 14:21.47 | *** join/#asterisk _DAW (n=chatzill@adsl-35-241-116.msy.bellsouth.net) |
| 14:21.53 | lyGP | coz it seems like the console not displaying anything |
| 14:21.54 | foobar778 | canapy cant refister??? |
| 14:21.59 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 14:21.59 | foobar778 | register |
| 14:22.18 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 14:22.22 | mosty | lygp: is your zapata.conf sending incoming calls to the correct context? |
| 14:23.00 | _DAW | Hi. Has anyone here had problems with Te110p pri immediatley hanging up on incoming calls once they are answered. No errors, just hangs up. |
| 14:23.38 | foobar778 | rang no sound |
| 14:23.45 | *** join/#asterisk heh_v_water (n=heh_v_wa@71-210-49-107.hlna.qwest.net) |
| 14:23.56 | lyGP | context=from-pstn |
| 14:24.01 | foobar778 | canapy it rang |
| 14:24.31 | lyGP | actually i;m newbie, still figuring around how to get it to work |
| 14:24.48 | lyGP | i'm using a ICSTEL voip gateway |
| 14:24.49 | *** join/#asterisk lowlevel (n=Stuart@CPE00145176d140-CM000f9f1e356e.cpe.net.cable.rogers.com) |
| 14:25.01 | lyGP | i believe ZAPATA is for the digium card right? |
| 14:25.36 | mosty | lygp: when you said FXO port, i assumed you meant a digium wctdm card |
| 14:25.46 | foobar778 | <PROTECTED> |
| 14:26.11 | foobar778 | itn rang no souncodecs? |
| 14:26.34 | foobar778 | canapy? |
| 14:26.47 | foobar778 | did u hear me? |
| 14:26.53 | lyGP | oh, sorry. it is a voip gateway with 2FXO + 2FXS |
| 14:27.22 | *** join/#asterisk stoffell (n=stoffell@d51A582B2.access.telenet.be) |
| 14:27.36 | *** join/#asterisk sambal (n=ivo@sd5116ceb.adsl.wanadoo.nl) |
| 14:27.51 | mosty | lygp: and how is it connected to asterisk? |
| 14:27.52 | sambal | hi, i have a problem with meetme, when a user press a 0 to go to the main menu, all users are going to the main menu |
| 14:28.07 | lyGP | my asterisk is 192.168.7.200 |
| 14:28.09 | sambal | does anyone know why? must be something with dtmf |
| 14:28.22 | lyGP | while the gateway is 192.168.7.92 |
| 14:28.36 | *** join/#asterisk canapa (i=canapa@83-64-148-102.wolfsberg.xdsl-line.inode.at) |
| 14:28.38 | canapa | argh |
| 14:28.45 | lyGP | exten => _ZXX,1,Dial(SIP/${EXTEN}@192.168.7.92:5060) |
| 14:28.45 | lyGP | exten => _ZXX,2,Hangup() |
| 14:28.54 | lyGP | i configure this in my dial plan as well.. |
| 14:29.14 | canapa | it works, just IRC did kick me for osme reason :d |
| 14:29.24 | *** join/#asterisk angryuser (n=Miranda@i03v-213-44-169-43.d4.club-internet.fr) |
| 14:29.25 | mosty | lygp: so you are dialling out via that gateway, but how is it configured to dial to the asterisk box? |
| 14:29.44 | lyGP | ya, dial out not a problem now |
| 14:29.46 | Defend | quick question pressing 0 from vm needs a extention defined is it 0 or o that needs to be defined? |
| 14:30.03 | lyGP | mosty, only when dial in, the connection being cut off |
| 14:30.26 | mosty | lygp: how is this gateway configured to dial to the asterisk box? |
| 14:31.02 | lyGP | i did not configure anything on the gateway |
| 14:31.11 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:31.11 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:31.25 | lyGP | i just assign it with an IP |
| 14:31.31 | lyGP | then configure asterisk |
| 14:31.45 | mosty | lygp: and you're trying to get calls to go from the gateway to asterisk right |
| 14:31.46 | mosty | ? |
| 14:31.57 | *** join/#asterisk blitz[training] (n=blitzrag@66.135.99.122) |
| 14:32.04 | lyGP | ya |
| 14:32.20 | mosty | well you need to configure the gateway to do that |
| 14:32.28 | Makenshi | anyone using trip care to place a test call for me? |
| 14:32.54 | *** part/#asterisk markit (n=konversa@host119-245-static.72-81-b.business.telecomitalia.it) |
| 14:32.56 | *** join/#asterisk PMantis (n=pmantis@66.251.89.34) |
| 14:34.16 | lyGP | a seconds, let me have a look |
| 14:36.54 | angryuser | ho to use Followme in asterisk dialplan? i have exten => xxx,1,Followme(Sipname) but it tells me application is not found, i have checked out app_followme.so is loaded in modules, and also followme.conf is present |
| 14:37.25 | angryuser | asterisk 1.4 changed the way followme is used? |
| 14:37.30 | lyGP | mosty: any idea what to look for in the voip gatwey? |
| 14:37.46 | *** join/#asterisk Ahrimanes (n=ma@81.7.159.2) |
| 14:38.41 | mosty | lygp: sip forwarding, or create a sip account, i would imagine |
| 14:38.55 | [TK]D-Fender | Defend: "o" has to be defined in the context calling VoiceMail, and must be enabled to the box in question. |
| 14:40.46 | lyGP | i need to look into the manual |
| 14:41.47 | lyGP | mosty: regarding the question previous, when i dial out the counter already start counting the duration even though the device is not pick up |
| 14:42.10 | lyGP | mosty:is there any configuration that i need to change, either in asterisk or voip gateway |
| 14:42.19 | mosty | lygp: the gateway is probably answering the call before forwarding it on |
| 14:43.14 | lyGP | mosty: any hints that what i can look into for this? |
| 14:43.53 | mosty | lygp: if you're not answering the call before dial'ing the gateway, it must be done on the gateway somewhere |
| 14:45.57 | lyGP | seems like the gateway giving alot of problem :( |
| 14:46.41 | *** join/#asterisk lowlevel (n=Stuart@CPE00145176d140-CM000f9f1e356e.cpe.net.cable.rogers.com) |
| 14:48.08 | *** join/#asterisk JackEStorm (n=no@ip68-225-72-125.no.no.cox.net) |
| 14:51.39 | lyGP | mosty: any question, how do i dial for example, phone number wait for pause then key in pin number wait for pause then phone number |
| 14:51.49 | lyGP | it need to dialled from sjphone |
| 14:51.57 | lyGP | not in the dial plan |
| 14:52.01 | *** join/#asterisk heh_v_water (n=heh_v_wa@71-210-49-107.hlna.qwest.net) |
| 14:52.07 | lyGP | can i configure the dial plan to behave like this? |
| 14:52.43 | mosty | lygp: sounds like you want to script a softphone, but i know jack about that softphone |
| 14:54.02 | lyGP | ya, each user will have their own pin, so i cannot script in dial plan |
| 14:54.29 | lyGP | for example, from handphone i usuall 12355,,,,,,1234,,,,123134502 |
| 14:54.46 | qdk | lyGP: isnt the account authentication enough? |
| 14:55.14 | lyGP | 12355(to dial into the system) then 1234(pin).. then only destination calls |
| 14:55.44 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 14:55.44 | lyGP | nope, each user are given a prepaid voice card, so they will have to managed their own pin. |
| 14:56.36 | lyGP | so i can script the dial plan in asterisk to preconfigured with pin. else all the user will use the same prepaid pin to makes call |
| 14:57.19 | *** join/#asterisk bvm (n=bvm@c-67-180-84-16.hsd1.ca.comcast.net) |
| 14:57.19 | qdk | lyGP: well it sounds like an easy job for a dialplan, but you seem to go about the wrong way. |
| 14:58.56 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:59.31 | lyGP | qdk, current all those prepaid card is to keep track all the user usage. |
| 14:59.51 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 15:00.32 | angryuser | i want my users enter pin from any phone (fxs asterisk) someting like <targer sip user> then <user call to be routed to> after that all the target call's must be routed to <user> is it possible? |
| 15:00.35 | qdk | lyGP: why not just activate the card once, add the value to an account and let people dial "normally"? |
| 15:01.24 | qdk | lyGP: but either way its just some DP-magic, give it a go. |
| 15:01.54 | qdk | angryuser: how would you know that X is located on phone Y? |
| 15:02.23 | tzafrir_laptop | angryuser, sounds like basic dialplan/IVR stuff |
| 15:02.39 | tzafrir_laptop | You just need several contexts |
| 15:02.51 | angryuser | tzafrir_laptop: not at all it's like follow me |
| 15:03.09 | *** join/#asterisk saktek1000 (n=saktek10@alphan17.lnk.telstra.net) |
| 15:03.32 | saktek1000 | did any one know what is wrong with G729 license server ? |
| 15:03.34 | lyGP | basically, the user are dialling like this on a normal extension (service number).wait.(pin).wait.(destination number) |
| 15:03.47 | angryuser | tzafrir_laptop: here is the situation, day 1 user 1 user Phone 1, day 2 user1 uses phone 2 and wants his call to be routed thre, how do i realize that? |
| 15:04.01 | lyGP | so basically i wish to have dial plan in asterisk that can do this dialling |
| 15:04.05 | angryuser | *his calls |
| 15:04.26 | lyGP | if want to call service number , is not a problem.. |
| 15:04.28 | qdk | lyGP: Asterisk can do that. |
| 15:04.31 | angryuser | qdk: the user know where he is located;) |
| 15:05.23 | [TK]D-Fender | angryuser: You need to use somthing like a stanardized macro for dialing extension in which you can see if you set a forwarding value. Go read about AstDB. That is the likely tool for your job. |
| 15:05.23 | qdk | angryuser: sure, but people rarely call themselves. |
| 15:05.23 | lyGP | qdk:- what is the dial plan will be like? |
| 15:05.35 | saktek1000 | did any one know what is wrong with G729 license server ? |
| 15:05.36 | *** join/#asterisk Bruno_Volvo (i=PJirc@81-174-39-150.f5.ngi.it) |
| 15:05.37 | qdk | lyGP: that is up to you to figure out. |
| 15:05.43 | JackEStorm | is there a howto/faq on Asterisk SIP+NAT+multiple nics and one way audio problems? (I can't seem to find one) |
| 15:05.45 | angryuser | [TK]D-Fender: ok, asdb comes implemented and preinstalled with asterisk? |
| 15:06.00 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 15:06.02 | [TK]D-Fender | angryuser: Yes. |
| 15:06.05 | lyGP | qdk:- that is why i need help, dialling normal ways, i have no problem now |
| 15:06.18 | lyGP | is just this unique dialling method, have no idea to input |
| 15:06.19 | [TK]D-Fender | angryuser: "show function DB" |
| 15:06.31 | angryuser | gone to read page and pages.....;) |
| 15:06.59 | Bruno_Volvo | Someone can help me with agents? |
| 15:07.11 | Bruno_Volvo | I can register my agents with queuenumber* |
| 15:07.24 | Bruno_Volvo | On FOP panel i see yellow light |
| 15:07.36 | mercestes | lyGP: Are you talking about DISA or are you talking about IVR's? |
| 15:07.42 | qdk | lyGP: Ok, though you had a problem or wanted to know if it was possible, but sine you are looking for a spoon feeding i cant help you. |
| 15:07.42 | mercestes | lyGP: What are you trying to do exactly? |
| 15:07.51 | Bruno_Volvo | but phone not ring on incoming call, and show agents show me all agents logged off |
| 15:07.53 | angryuser | forgot to tell you people, thank you for help;) |
| 15:08.21 | saktek1000 | Is any one from Digium here ?? |
| 15:08.31 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:08.31 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:08.42 | mercestes | qdk: You scared him off..:) |
| 15:08.52 | [TK]D-Fender | JackEStorm: You need to define all of your clocal subnets, and the external IP to use if communicating with resources outside your local subnets. |
| 15:08.54 | tzafrir | saktek1000, probably |
| 15:09.50 | saktek1000 | there is problem with G729 licsense server, I think it is down , can not connect to it when trying to activate my G729 lisecne |
| 15:11.04 | qdk | mercestes: Guess he wasnt here for help, but for leeching. |
| 15:11.15 | *** join/#asterisk hyphen (n=hyphen@c-69-136-84-149.hsd1.pa.comcast.net) |
| 15:11.37 | mercestes | saktek1000: You have been blocked specifically, your card has been declined. Please contact your card agency and have the issue resolved and we will remove the block. |
| 15:13.03 | mercestes | saktek1000: I'm kidding btw..you can quit checking your online statements. |
| 15:13.53 | *** join/#asterisk lukketto (n=lukketto@host47-159-dynamic.6-87-r.retail.telecomitalia.it) |
| 15:14.12 | Bruno_Volvo | How I can route incoming call with DB? (ex. I have 100 CID that I need to route to particular queue, how I can make this?= |
| 15:15.47 | *** join/#asterisk jarg (n=jarg@200.56.225.61) |
| 15:15.58 | docelmo | saktek1000 check with digium |
| 15:16.02 | [TK]D-Fender | Bruno_Volvo: "show function ODBC" , "show function DB" |
| 15:16.04 | docelmo | only they can help you at this point |
| 15:16.08 | mosty | bruno_volvo: what do you mean by CID? |
| 15:16.15 | [TK]D-Fender | mosty: CallerID |
| 15:17.00 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-72-65-23-44.bflony.east.verizon.net) |
| 15:17.06 | JackEStorm | [TK]D-Fender: I thought that asterisk when a sip phone is set 'nat=yes' would proxy all the streams? (because it's ignoring the header and knows where that traffic really should go) |
| 15:17.43 | [TK]D-Fender | JackEStorm: * needs to know when to forge its OWN return IP. Thats what localnet & externip are for. |
| 15:18.18 | [TK]D-Fender | JackEStorm: Syaing the remote end is behit a NAT of its own is a very seperate thing altogether. * needs to know ITS behind NAT. |
| 15:18.27 | Bruno_Volvo | [TK]D-Fender: very thanks! |
| 15:18.52 | *** join/#asterisk bertrand^ (n=bertrand@ATuileries-151-1-75-176.w86-212.abo.wanadoo.fr) |
| 15:19.09 | [TK]D-Fender | Bruno_Volvo: You're welcome. You could do this with fixed entries in the dialplan, but I don't think you want to hand-key them in. up to you naturally... |
| 15:21.39 | Bruno_Volvo | [TK]D-Fender: Now I use Trixbox for my particular Custumer, 60 lines, 40 phones... On March I have Asterisk BootCamp, and after this I discard trixbox. |
| 15:22.22 | [TK]D-Fender | Bruno_Volvo: Welcome to the brave and rewarding world of "Do It Yourself", where you actually GET what you want... |
| 15:22.45 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 15:23.26 | Bruno_Volvo | [TK]D-Fender: Cool :D I need other little help... I need request of password for agent login and logout |
| 15:24.45 | Bruno_Volvo | [TK]D-Fender: but simply Asterisk ask me for ID and # without password |
| 15:24.59 | Bruno_Volvo | I have this log http://brunovolvo.pastebin.co.uk/10444 |
| 15:25.17 | [TK]D-Fender | Bruno_Volvo: * asks for a password, Trixbox on the otherhand might just automatically auth people. |
| 15:26.09 | mosty | is it correct that SIP phones won't accept calls to a particular extension, and that only SIP servers can handle that? (I tested a snom phone and a linksys, they refused the call) |
| 15:26.09 | [TK]D-Fender | Bruno_Volvo: Which is exactly what its doing. |
| 15:26.20 | [TK]D-Fender | mosty: What kind of "extensions" do you have ona phone? |
| 15:26.34 | *** join/#asterisk jlewis (n=jlewis@solo.atlantic.net) |
| 15:26.42 | Bruno_Volvo | [TK]D-Fender: mmmmm, i understand! Now I install pure Asterisk on other PC, and copy source for agent-login/logout |
| 15:27.20 | [TK]D-Fender | Bruno_Volvo: You can do whatever you want with it.... once you're coding it for yourself. Until then FreePBX owns your soul.... |
| 15:27.30 | mosty | [TK]D-Fender: i have multiple DID numbers mapped to a single SIP account. i don't know if that account is a SIP phone or a SIP server. if it's a SIP server, they want to route incoming calls based on the specific DID that was dialled |
| 15:27.36 | jlewis | with a relatively inflexible PBX exchanging calls with * over sip, what's the best way to either change or munge the callerid that PBX sends to *? |
| 15:28.16 | sweeper | jlewis: well, * can set the callerid on an incoming call |
| 15:28.25 | jlewis | in dialplan? |
| 15:28.29 | sweeper | yep |
| 15:28.33 | jlewis | that was the first way I thought of |
| 15:28.34 | *** join/#asterisk codazoda (n=chatzill@mail.hurdmanivr.com) |
| 15:28.42 | jlewis | but I think it complicates things a bit |
| 15:28.59 | Bruno_Volvo | [TK]D-Fender: on 19 March I start BootCamp in Bologna, Italy! another month! :( |
| 15:29.00 | sweeper | if the pbx can't change the callerid, then * has to |
| 15:29.01 | [TK]D-Fender | mosty: Well you don't register TO a phone, so they are clearly a SERVER. They should probably be dialing a target exten when sending calls to you for which you'd create entries in their user context to choose whre to route them to. |
| 15:29.18 | jlewis | we can't just have their incoming context "include => " the relevant contexts if we need to accept the call and setcallerid |
| 15:29.29 | [TK]D-Fender | Bruno_Volvo: Why wait for boot camp? Don't go in blind. Get your hands dirty... |
| 15:29.31 | codazoda | I'm trying to do a sing line transfer on a tdm2400e card from an AGI script. Does anyone know if this is possible? |
| 15:29.32 | [TK]D-Fender | ~book |
| 15:29.35 | jbot | [book] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 15:30.13 | mosty | [TK]D-Fender: i am not registering to any servers, they are registering to me (both phones and servers) |
| 15:30.14 | mercestes | codazoda: define sing line transfer. |
| 15:30.34 | Bruno_Volvo | [TK]D-Fender: I try It from 4 month, but with my normal job, I have very little time! On March I take one week from my work only for asterisk! |
| 15:30.40 | JackEStorm | [TK]D-Fender: in this case [short example] phone A boots and configs from *, gets 192.168.1.x, and is on NIC A, NIC B is a real IP, and phone B connected via NIC B, and gets it's ip (192.168.1.x in this case) from the NAT device on that end. ...so it's more like 'every case' in this case. |
| 15:30.44 | [TK]D-Fender | mosty: That doesn't soud right... since when do DIDI *providers* register to YOU? |
| 15:30.57 | *** join/#asterisk Shazaum (n=shazaum@200.175.61.250.static.gvt.net.br) |
| 15:31.02 | *** join/#asterisk ToyMan (n=Stuart@12.23.30.130) |
| 15:31.04 | mosty | [TK]D-Fender: they're not providers. we are the provider. they are a client's pbx |
| 15:31.06 | Shazaum | chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 3 |
| 15:31.09 | Shazaum | hi alll |
| 15:31.10 | codazoda | What I'm trying is "EXEC Flash" then "EXEC SendDTMF 1234567". On my analog lines, when I do a hook flash I get a dialtone. Then, if I dial 7 digits, that phone number rings. Disconnecting the line then connects the two callers (but I'm off). |
| 15:31.19 | Shazaum | somebody ? |
| 15:31.23 | jlewis | if I goto(contextA,${EXTEN},1) and contextA doesn't have a matching ${EXTEN} but includes a context that does, will that work?...I know...I should just set it up and test it to see |
| 15:31.37 | codazoda | So, I'm doing a transfer without bridging two lines. |
| 15:31.47 | mercestes | mosty: Phones do not accept calls to extensions. phones accept a Dial via Sip technology to a SIP Peer name, for example, Dial(Sip/YourPhone). |
| 15:31.54 | [TK]D-Fender | JackEStorm: Well you need to define your 2 local subnets, disable reinvites, and THEN tell * that some phones/services on the outside are behind their own NAT's (since * is publlic you don't need NAT=yes in [general]) |
| 15:32.14 | [TK]D-Fender | jlewis: Indeed you should. |
| 15:32.18 | mercestes | mosty: You can trigger this event by using an extension, 1234,1,Dial(SIP/Phone1234.) but there is no reason for the extension to match the peer name, but it often happens that way just for sanity |
| 15:32.28 | jlewis | :) guess I'll find out |
| 15:33.09 | JackEStorm | [TK]D-Fender: doh! reinvites |
| 15:33.18 | mercestes | codazoda: Oh, I did code for that yesterday |
| 15:33.23 | angryuser | hm, why do i have application DBdel, but i dont have DBput? aster 1.4 |
| 15:33.35 | mercestes | codazoda: yes, it's very possible...and no, you don't even need AGI...but if you want AGI...then yes. |
| 15:33.50 | mosty | mercestes: i give out sip accounts, and sell DID numbers, which are mapped to those sip accounts. a single sip account may have multiple DID's mapped to it. is there a way (with a single Dial command) to send those calls to both sip phones and sip server clients along with the specific DID that was dialled? |
| 15:34.04 | [TK]D-Fender | JackEStorm: Reinvites=EVIL. |
| 15:34.38 | *** join/#asterisk awannabe (n=brad@207-114-155-213.static.twtelecom.net) |
| 15:34.39 | [TK]D-Fender | mosty: You are PROVIDING this service and don't know the answer already> OMG.... |
| 15:34.44 | *** join/#asterisk sav_mcfly (n=R00T@pergamo.zonaz.net) |
| 15:34.56 | mercestes | mosty: exten => 1,1,Dial(Sip/1&Sip/2&Sip/3&Sip/4&Sip/5&Sip/OMGSpam&Sip/SomebodyStopMe,9000,wt) |
| 15:34.59 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:35.10 | mercestes | [TK]D-Fender: i second that emotion. |
| 15:35.11 | [TK]D-Fender | service mercestes stop |
| 15:35.15 | mosty | [TK]D-Fender: i don't think i'm explaining myself well |
| 15:35.23 | mosty | mercestes: no that is not what i want to do |
| 15:35.42 | mosty | i map multiple DID's to one sip account |
| 15:35.49 | [TK]D-Fender | angryuser: Because "DB" is now a FUNCTION. |
| 15:35.53 | awannabe | hey guys, im using AgentCallbackLogin() and having a problem once I call the agent, it then does all the steps in dialplan for the extension the agent is logged into, not the steps i have setup when the agent's number is called, is there a way to fix this I hope?? |
| 15:36.26 | mosty | mercestes: i map multiple DID's to one sip account, not one did to multiple sip accounts (as your Dial command did) |
| 15:36.39 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 15:36.45 | [TK]D-Fender | awannabe: You told it where to go, and its doing it. rethink where you send calls for that agent.... |
| 15:36.51 | mercestes | mosty: Ok. |
| 15:37.03 | angryuser | [TK]D-Fender: bhaha, you are right;) |
| 15:37.06 | JackEStorm | [TK]D-Fender: I forgot about all about reinvites, been about 4~6 months since I've had to touch this box |
| 15:37.09 | angryuser | [TK]D-Fender: found it |
| 15:37.11 | mosty | mercestes: now if that sip client is a phone, i Dial(SIP/peername) and they get all the calls. this works ok. |
| 15:37.24 | sav_mcfly | ftp://ftp.digium.com/pub/telephony/hpec/README says about zaptel 1.4.1 but there is no such version |
| 15:37.31 | awannabe | [TK]D-Fender, yeah i see that, but these people move around all the time, so I want it to roll to their personal voicemail box, which isnt at that phone they are calling, is there another way around that? |
| 15:37.36 | mercestes | mosty: exten => 123,1,Dial(Sip/FreeHelp) exten => 234,1,Goto(123,1) exten => 456,1,Goto(123,1) |
| 15:37.40 | mosty | mercestes: but in some cases, the sip client is a server (like asterisk, or epygi or whatever), and they want to know which DID the call came in on |
| 15:37.59 | [TK]D-Fender | sav_mcfly: Its about to be released... don't let the cat out of the bag! |
| 15:38.13 | mercestes | mosty: So you use lines? Or mangle CID? |
| 15:38.21 | Shazaum | <PROTECTED> |
| 15:38.29 | Shazaum | chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 3 ???? |
| 15:38.31 | mercestes | mosty: Set(CALLERID(name)="123") |
| 15:38.40 | [TK]D-Fender | mosty: exten_X,1,Goto(${EXTEN},1) |
| 15:39.00 | mosty | mercestes: so the only way is with callerid? |
| 15:39.30 | [TK]D-Fender | mercestes: You don't mangle CID to indicate where you route calls to. I'm quite sure they'd WANT to have the originating CID.... |
| 15:39.55 | mercestes | mosty: no. |
| 15:39.56 | [TK]D-Fender | mosty: Dial(SIP/mypeer/stupidtagetnumberduh) |
| 15:40.19 | codazoda | mercestes: thanks for the info. Knowing that it works got me thinking of a couple other things to look at. I just did it successfully. Thanks again! |
| 15:40.29 | mosty | [TK]D-Fender: i tried that, but it does not work when the sip peer is a phone |
| 15:40.31 | mercestes | codazoda: your welcome. |
| 15:41.24 | [TK]D-Fender | mosty: In thoses cases you can't do it, and should have settings to accomodate that. |
| 15:41.25 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 15:41.56 | mosty | [TK]D-Fender: ok, that is all i wanted to confirm- that i need to seperate the two groups and use two different kinds of dial commands |
| 15:43.24 | mosty | sorry if i wasn't very coherent, it's late here |
| 15:44.08 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 15:44.23 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 15:48.31 | [TK]D-Fender | mosty: NP. Just took a few extra questions to iron it out. |
| 15:49.07 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 15:49.52 | *** part/#asterisk Shazaum (n=shazaum@200.175.61.250.static.gvt.net.br) |
| 15:49.53 | PMantis | !seen file |
| 15:49.56 | puzzled | hi |
| 15:51.09 | PMantis | file, There is a bug report for chan_spy crashing asterisk on disconnect - I have that issue on 1.4.0. Should I file a new report, or update the existing one? |
| 15:51.35 | PMantis | The existing one is for a different asterisk version |
| 15:53.58 | *** join/#asterisk funxion (n=nunya@63.214.236.169) |
| 15:55.33 | zamba | i need help setting up a digium wildcard e100p card.. i've compiled the modules and they seem to be working just fine.. i can also see the card when issuing 'zap show status', but alarms return "yel/rec".. |
| 15:55.48 | zamba | my pastebin for /etc/zaptel.conf is at http://www.pastebin.ca/353603 |
| 15:58.33 | mercestes | zamba: This is BRI? |
| 15:58.39 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 15:58.40 | *** join/#asterisk bertrand^ (n=bertrand@ATuileries-151-1-82-232.w86-212.abo.wanadoo.fr) |
| 15:58.51 | zamba | mercestes: no, pri |
| 15:59.13 | mercestes | zamba: If I recall correctly, PRI only has 23 channels. |
| 15:59.25 | mafkees | mercestes: not if it's E1 |
| 15:59.28 | mafkees | that one has 30 |
| 15:59.33 | mercestes | oh. |
| 15:59.44 | mercestes | that's what I meant. |
| 15:59.46 | mercestes | really it was. |
| 16:00.02 | mafkees | :) |
| 16:00.11 | mercestes | zamba: What does your provider say? |
| 16:01.11 | zamba | it should be PRI/E1 |
| 16:01.36 | zamba | the "provider" is a ericsson pbx (called mx?) |
| 16:02.22 | zamba | mx-one? |
| 16:02.32 | zamba | but it is definitely PRI/E1 |
| 16:04.40 | sweeper | zamba: if you're in yellow alarm, you probably don't have the right framing set |
| 16:04.54 | sweeper | check with your provider to find the correct t1 settings |
| 16:04.58 | bertrand^ | hello |
| 16:05.22 | zamba | well, it should either be cas or ccs |
| 16:05.22 | bertrand^ | while trying dial 12345678 from the asterisk console, asterisk should call me, right? |
| 16:05.29 | zamba | according to http://www.voip-info.org/wiki/index.php?page=Zaptel.conf+span+syntax |
| 16:07.24 | zamba | how can i test the configuration in zaptel.conf? do i need to reload the module? |
| 16:08.38 | *** join/#asterisk D3V|L (n=d3v@200.118.122.44) |
| 16:08.59 | *** part/#asterisk D3V|L (n=d3v@200.118.122.44) |
| 16:09.38 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 16:09.58 | bertrand^ | zamba, i used ztcfg -v |
| 16:10.09 | bertrand^ | zamba, but maybe don't trust a newbie |
| 16:10.26 | Bobthehunter | ben explique l |
| 16:10.53 | bertrand^ | chouette ça cause français ici |
| 16:11.27 | Bobthehunter | lol sorry |
| 16:11.32 | Bobthehunter | trying to apste something |
| 16:11.51 | Bobthehunter | socket_read: Rejected connect attempt from 69.159.245.243, requested/capability 0x2/0xf802 incompatible with our capability 0x |
| 16:12.04 | Bobthehunter | ok hjow we suppose to HEX 2 DEC this |
| 16:12.55 | [TK]D-Fender | Bobthehunter: 63490 |
| 16:28.42 | *** join/#asterisk jbot_ (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 16:28.43 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- Asterisk 1.2.15 and Zaptel 1.2.13 released! (Feb. 9, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits -=- Join #freepbx for freepbx/trixbox support. -=- |
| 16:28.57 | angryuser | [TK]D-Fender hm, good idea, make more peers instead, but how can i verify if "peer" is Unreachable? |
| 16:30.05 | angryuser | [TK]D-Fender in dialplan i mean |
| 16:30.38 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 16:30.38 | mercestes | angryuser: ${DIALSTATUS} is much more useful than simply checking ot see if a peer is unreachable. |
| 16:31.17 | angryuser | ok i will search that var |
| 16:32.31 | *** join/#asterisk samueltc (n=samuel@206-248-163-197.dsl.teksavvy.com) |
| 16:32.33 | *** join/#asterisk edison\ (n=edison@c-67-176-75-179.hsd1.co.comcast.net) |
| 16:32.35 | samueltc | hi |
| 16:33.14 | angryuser | DIALSTATUS wont do coz i need to know before calling if peer is reachable, not analyze last call information |
| 16:33.14 | *** join/#asterisk JoNate (n=noone@mail.wmelec.com) |
| 16:33.53 | *** join/#asterisk docelmo (n=vircuser@c-68-32-61-251.hsd1.de.comcast.net) |
| 16:33.54 | *** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 16:34.22 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 16:35.24 | sav_mcfly | angryuser: also use qualify=yes |
| 16:35.45 | [TK]D-Fender | angryuser: If the call fails for whatever reason, jsut dial the next option. |
| 16:36.01 | [TK]D-Fender | angryuser: You don't even need dialstatus |
| 16:36.50 | angryuser | [TK]D-Fender: i will call it BruteDialing;) |
| 16:37.11 | edison\ | hey all. |
| 16:37.20 | *** join/#asterisk w0ls0n (n=Me@43-141-135-64.dsl.sacoriver.net) |
| 16:37.31 | edison\ | my upstream sip provider and * aren't talking since I upgraded to the latest SVN. |
| 16:37.36 | w0ls0n | I would just like some opinions on trixbox |
| 16:37.44 | docelmo | w0ls0n it sucks |
| 16:37.57 | edison\ | anyone know of conf changes that would affect this, or should I check in -dev? |
| 16:37.59 | docelmo | and is the wrong channel to ask such questions |
| 16:38.10 | *** join/#asterisk intralanman (n=lanman@pool-71-253-242-197.nrflva.east.verizon.net) |
| 16:38.17 | w0ls0n | I was asking for opinions thats all |
| 16:39.03 | *** part/#asterisk Stp1800 (n=Stp1800@67-22-111-205.atlsfl.adelphia.net) |
| 16:39.10 | file | w0ls0n: there are 3 branches of Asterisk in SVN that are active so without saying which one you are talking about... it's hard to say, and you should also try to do some investigation to elaborate on what "aren't talking" means |
| 16:40.11 | brodiem | What options do I have to be able to have for example a "standard" queue, and then a "priority" queue where callers entering the priority queue will of course be prioritized over those entering the standard queue? (using the same agents for both queues) |
| 16:42.50 | [TK]D-Fender | brodiem: Just st the priority when you call App_queue |
| 16:42.54 | ModocNet | DAMN!!!! Forgot to put mailbox= in sip.conf!!!!! just wasted 1/2 day! |
| 16:43.35 | [TK]D-Fender | w0ls0n: I concur. It sucks. |
| 16:43.41 | w0ls0n | hmmm ok |
| 16:43.42 | ModocNet | 112 extensions install and forgot 1 line....customer kept say, "isn't there an indicator for voicemail?" |
| 16:43.53 | w0ls0n | is there a GUI backend for asterisk I can install seperately? |
| 16:43.54 | [TK]D-Fender | ModocNet: Now go pat yourself on the back for having figured it out.... |
| 16:43.57 | ModocNet | it's always the little thing the get me |
| 16:44.02 | ModocNet | lol |
| 16:44.03 | *** part/#asterisk edison\ (n=edison@c-67-176-75-179.hsd1.co.comcast.net) |
| 16:44.20 | [TK]D-Fender | w0ls0n: FreePBX is the backend that Trixbox uses, and can be installed on its own. |
| 16:44.22 | ModocNet | the hard stuff no problem...it's always something little |
| 16:44.28 | w0ls0n | ah ok |
| 16:44.38 | mercestes | w0ls0n: Asterisknow has some GUI stuff as well. |
| 16:44.56 | brodiem | [TK]D-Fender I was looking at Queue() on voip-info and doesn't look to be an option to set a priority on entering the queue.I know I can set penalties on agents but I'm wondering if say agents logged into "standard" have penalty of 5 and those logged into "priority" have no penalty if it'll actually apply its priotities or not, i.e. whether or not penalties will take other queues into consideration |
| 16:45.49 | w0ls0n | so freepbx is the gui extension of asterisk? |
| 16:46.04 | *** join/#asterisk supjigatr (n=syslod@152.53.16.10) |
| 16:46.15 | *** part/#asterisk supjigatr (n=syslod@152.53.16.10) |
| 16:46.21 | brodiem | [TK]D-Fender n/m sorry, I found the QUEUE_PRIO variable :) |
| 16:46.26 | mercestes | w0ls0n: no. freepbx is just a GUI thing. |
| 16:46.31 | w0ls0n | ok |
| 16:46.50 | mercestes | w0ls0n: There is another gui extension of asterisk somewhere on asterisknow if you care to search for it. |
| 16:47.04 | w0ls0n | I will :-) |
| 16:47.36 | [TK]D-Fender | w0ls0n: FreePBX is a system that completely configure all the major bits of * allowing you to build a cookie-cutter PBX with limited intelligence and flexibility via a web browser |
| 16:49.02 | w0ls0n | man I need RR at work. 500MB files are just taking too long |
| 16:49.38 | w0ls0n | ok thanks all |
| 16:50.09 | w0ls0n | strange. Asterisknow is an ISO? |
| 16:50.45 | mafkees | yeah |
| 16:50.49 | mafkees | it's a complete os |
| 16:51.05 | file | it's a linux distro with Asterisk and the GUI |
| 16:51.22 | file | but the GUI itself is a separate project and can be installed on a regular Asterisk install, provided you use 1.4 |
| 16:51.23 | [TK]D-Fender | Just like Trixbox... yay! |
| 16:51.38 | mafkees | lol |
| 16:51.47 | mercestes | guis are for script kiddies. |
| 16:51.50 | mafkees | yeah, but created by the digium folks |
| 16:52.10 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 16:53.36 | Dr-Linux | anybody knows the Digigum tdm400p 4 fxo port card price? |
| 16:53.37 | w0ls0n | yea well the only problem I have is just looking at the code. Using the GUI will be a bit more freindlier to me. Anyways this is just a test machine until I get myself more fimiliar (SP) with it |
| 16:53.51 | sweeper | hmmmm |
| 16:54.01 | xermesx | [TK]D-Fender, have you ever tried misdn with asterisk ? |
| 16:54.11 | sweeper | I wonder if I can do cool voodoo with tftp or ftp and rails |
| 16:54.16 | [TK]D-Fender | xermesx: nope |
| 16:54.21 | sweeper | like, generate polycom configs on the fly from a DB :D |
| 16:54.23 | mercestes | w0ls0n: if you require a GUI to do whatever it is you are trying to do, then I suggest you steer clear of whatever it is you are trying to do. |
| 16:54.30 | w0ls0n | hehe |
| 16:54.56 | *** join/#asterisk badcfe (n=cso@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr) |
| 16:55.01 | [TK]D-Fender | w0ls0n: These QUI tools witll teach you NOTHING, and are a poor stepping stone to learning * |
| 16:55.02 | sweeper | [TK]D-Fender: can polycoms pull config over something like HTTP:XML? |
| 16:55.15 | [TK]D-Fender | sweeper: Over HTTP/HTTPS yes |
| 16:55.28 | sweeper | what does the config file look like? |
| 16:55.30 | sweeper | text? |
| 16:55.38 | [TK]D-Fender | sweeper: Just like ti does over FTP. |
| 16:55.47 | sweeper | never done it over ftp :P |
| 16:56.13 | [TK]D-Fender | sweeper: The go download a firmware pack and theAdmin guide and crack open the books. Stop trying to run before you can walk... |
| 16:56.22 | sweeper | :P |
| 16:57.24 | *** join/#asterisk apardo (n=apardo@87.217.144.23) |
| 16:58.08 | mercestes | gentoo is mean |
| 16:58.44 | bertrand^ | "Auto fallthrough, channel 'OSS/dsp' status is 'CHANUNAVAIL'" |
| 16:58.54 | bertrand^ | does that ring a bell to someone?? |
| 17:00.07 | angryuser | [TK]D-Fender:ok i have put on the solution to duplicate peers on each ip, and they have registered.....all(!!) |
| 17:00.39 | angryuser | [TK]D-Fender:my provider is a strange phenomen |
| 17:02.05 | CunningPike | ~seen gork |
| 17:02.26 | jbot | gork <n=4cb3dd7b@bz.bzflag.bz> was last seen on IRC in channel #bzflag, 10d 23h 7m 16s ago, saying: 'shit!'. |
| 17:02.27 | mitcheloc | ~seen my ass |
| 17:02.33 | jbot | mitcheloc: i haven't seen 'my ass' |
| 17:02.45 | mitcheloc | ahh... sorry, couldn't resist |
| 17:02.52 | CunningPike | ~lart mitcheloc |
| 17:03.00 | mitcheloc | meh! |
| 17:03.06 | file | mitcheloc: uh oh, it's you |
| 17:03.09 | file | mitcheloc: and you're awake! |
| 17:03.21 | Qwell[] | at 9am even |
| 17:05.51 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 17:05.55 | sweeper | oh, does anyone know of documentation that may exist for multi-site, fault-tolerant * installs? |
| 17:09.03 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 17:09.39 | bkw_ | +list |
| 17:09.43 | bkw_ | doh |
| 17:10.34 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 17:12.41 | mitcheloc | bkw_: wrong room! |
| 17:13.01 | *** join/#asterisk ToyMan (n=Stuart@12.23.30.130) |
| 17:13.13 | mitcheloc | fileQwell: lol, i put my head back down to go back to sleep....was woke up by a phone call, meh! |
| 17:13.25 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
| 17:22.34 | *** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it) |
| 17:22.46 | jlewis | Just got back and tested it. If I goto(contextA,${EXTEN},1) and contextA doesn't have a matching ${EXTEN} but includes a context that does, it does work as desired. Very cool! |
| 17:23.30 | jlewis | which means I can accept calls, do all the CID munging desired, and then hand the calls back to the dialplan as if nothing has happened |
| 17:23.58 | jlewis | which should be an easy fix for my friend's CID issues |
| 17:24.45 | *** join/#asterisk qdk (n=qdk@90.184.3.249) |
| 17:25.39 | [TK]D-Fender | jlewis: Andimagine all you had to do was get off your butt and TRY it! |
| 17:37.07 | *** join/#asterisk nosbig (n=nosbig@rrcs-70-62-223-6.central.biz.rr.com) |
| 17:37.13 | *** join/#asterisk nortex (n=nortex@64.136.65.144) |
| 17:38.10 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 17:40.49 | *** join/#asterisk apardo (n=apardo@87.217.144.23) |
| 17:48.59 | *** join/#asterisk Skaag (n=skaag@212.199.180.157.static.012.net.il) |
| 17:49.54 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 17:50.51 | Skaag | I just purchased two IAX2 trunks, I tried adding them to iax.conf but now I get this message: chan_iax2.c:8183 build_peer: unable to support trunking on peer '4020' without zaptel timing |
| 17:50.56 | angryuser | to read with DB i need to set only <family> and in db_result i will have a key? |
| 17:55.18 | *** join/#asterisk wulfy814 (n=wulfy814@dsl093-061-214.pit1.dsl.speakeasy.net) |
| 17:55.22 | wulfy814 | afternoon folks |
| 17:55.53 | wulfy814 | I'm getting an AM radio on my TA750 with 8 fxo , I've made sure it's properly grounded and the asterisk PC is grounded - any other tips to eliminate this? |
| 17:58.16 | blitz[training] | encase the box within' lead |
| 17:59.05 | wulfy814 | seriously? what about aluminum foil? |
| 17:59.18 | blitz[training] | that might work too, lol |
| 17:59.23 | blitz[training] | I was just being an ass |
| 17:59.42 | *** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir) |
| 18:00.17 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 18:00.32 | diclophis-work | i htought asterisk 1.4 was supposed to come with this kickass web-config component? |
| 18:00.46 | Qwell[] | no, but if you /j #asterisk-gui, you can read the topic and see how to get it |
| 18:01.04 | Waverly360 | In sip.conf, what are the available options for 'host'? I know you can use an ip address or hostname, as well as 'dynamic', but can you do ranges of ips? |
| 18:01.38 | diclophis-work | Qwell: how stable do you reckon it is? |
| 18:01.53 | Qwell[] | it seems to work - people are using it in production |
| 18:01.57 | *** join/#asterisk codazoda (n=chatzill@mail.hurdmanivr.com) |
| 18:02.19 | diclophis-work | ok, so i have the 1.4 tarball unpacked, i just run the svn co command in that dir, and it will add the proper components? |
| 18:03.01 | codazoda | Hi. I have a TDM2400E card. The volume was very low (both in and out), so I turned up TX and RX to 20 each. That solved my volume problems, but it seems to cause cut-outs in the callers I'm talking to. Any ideas? |
| 18:03.23 | Waverly360 | I want to allow any pbx with the proper username and password, to dial an extension on this pbx via sip. Surely I don't have to create a separate context in sip.conf for each server..right? |
| 18:07.21 | blitz[training] | Waverly360: I think you can define a type=user with no host= definition and then use deny/permit to control the ACL |
| 18:07.25 | blitz[training] | although I have no tried that |
| 18:08.04 | [TK]D-Fender | Skaag: You don't have a Zaptel timing source on your server so you can't do trunking. |
| 18:08.22 | Skaag | I don't even know what a timing source is |
| 18:08.34 | diclophis-work | Skaag: a source of timing |
| 18:08.40 | Skaag | like NTP? |
| 18:08.50 | diclophis-work | more along the lines of CPU clock timing |
| 18:08.56 | [TK]D-Fender | Skaag: A zaptel compatible card or ZTDUMMY which has to be compiled in with Zaptel BEFORE * is compiled. |
| 18:09.00 | blitz[training] | Waverly360: yep, just tried it -- there is no host= for users |
| 18:09.28 | [TK]D-Fender | Skaag: In order to time-synch packet data, etc. Required for IAX2 trunking, Meetme, and another thing or two. |
| 18:09.48 | [TK]D-Fender | Skaag: No, a high-resolution system clock, not NTP. |
| 18:10.14 | blitz[training] | Waverly360: so do something like: [my_pbxs]\n type=user\n context=incoming\n deny=0.0.0.0/0.0.0.0\n permit=192.168.1.0/255.255.255.0 |
| 18:10.36 | Waverly360 | blitz: Thank you so much. I'll try that out. |
| 18:15.51 | *** join/#asterisk joeTSUNAMI (n=joeTSUNA@72.243.118.59) |
| 18:16.12 | *** join/#asterisk docelmo (n=vircuser@c-68-32-61-251.hsd1.de.comcast.net) |
| 18:19.51 | Waverly360 | blitz[training]: No luck man. If I leave out the host=[server I'm connecting from], then all I get is a busy signal |
| 18:20.13 | Waverly360 | blitz[training]: even with type set as user, or peer, and with permit=0.0.0.0/0.0.0.0 |
| 18:20.56 | Skaag | so how do i get this timing source setup? |
| 18:21.03 | blitz[training] | hrmmm... then it must be a limitation of SIP. I've just always created multiple definitions and never thought of what you were looking for :) |
| 18:23.18 | [TK]D-Fender | Skaag: Go lookup ZTDUMMY on the WIKI |
| 18:23.21 | [TK]D-Fender | ~wikis |
| 18:23.30 | jbot | from memory, wikis is http://www.voip-info.org |
| 18:23.40 | Skaag | thanks |
| 18:24.03 | Skaag | that kind of thing should be installed and setup by default, if you ask me |
| 18:25.04 | *** join/#asterisk genz (n=erdo@im.jobdig.com) |
| 18:25.37 | [TK]D-Fender | Skaag: Do you even HAVE zaptel installed? |
| 18:25.46 | Skaag | I don't need any digium hardware |
| 18:26.13 | Skaag | I just need IAX2 to function, to connect a trunk to an IVR i'm making |
| 18:26.19 | putzz | lol |
| 18:26.22 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 18:26.24 | Skaag | maybe i'm going the wrong way about this |
| 18:26.34 | putzz | u still need ztdummy |
| 18:26.40 | putzz | for timing |
| 18:26.41 | Skaag | yes, unfortunately |
| 18:26.51 | [TK]D-Fender | Skaag: Its also neatly explained on Page 36 of THE BOOK. |
| 18:26.53 | [TK]D-Fender | ~book |
| 18:27.02 | jbot | it has been said that book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 18:27.18 | Skaag | oh god, a kernel module... |
| 18:27.24 | Skaag | kill me now |
| 18:27.24 | Waverly360 | lol |
| 18:27.31 | Skaag | I want to die, please |
| 18:27.39 | Waverly360 | It's not so bad Skaag |
| 18:27.44 | CunningPike | Skaag: You're not Douglas Garstang, are you? |
| 18:27.50 | Qwell[] | haha |
| 18:27.53 | wunderkin | heh heh |
| 18:27.56 | Skaag | no, but if I was, I would probably say the same thing |
| 18:28.06 | *** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 18:28.48 | Qwell[] | Skaag: unlikely :) |
| 18:28.58 | Skaag | oh well |
| 18:29.00 | Skaag | futz. |
| 18:30.34 | [TK]D-Fender | Skaag: If this is enough to leave you so dismayed, try not to do it during roush hour.... |
| 18:31.16 | *** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it) |
| 18:31.54 | *** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir) |
| 18:32.45 | joeTSUNAMI | Hi guys.. I have a Polycom 430 that is stuck in a constant reboot cycle and can't seem to do anything... does anyone one know how to do a hard reset on this? |
| 18:33.33 | HarryR | joeTSUNAMI, take the power cable out? |
| 18:34.19 | joeTSUNAMI | HarryR: tried.. after waiting even after 30 min, it goes back to the reboot cycle |
| 18:34.42 | [TK]D-Fender | joeTSUNAMI: Hold 1,3,5,7 simultaneously while in the bootrom countdown. |
| 18:34.56 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
| 18:35.10 | [TK]D-Fender | joeTSUNAMI: And fix your configs. |
| 18:35.25 | joeTSUNAMI | [TK]D-Fender: did that too, put in the admin pass, it said it reset, but still boot cycles |
| 18:35.29 | CunningPike | joeTSUNAMI: If it is setup to look for a config file that isn't there, and it has no existing config to fall back on, it will keep rebooting until a config file is found |
| 18:35.40 | funkmaster | hi ppl :D are there any skypho users here? |
| 18:35.44 | [TK]D-Fender | joeTSUNAMI: Ditch your provisioning server config |
| 18:35.52 | joeTSUNAMI | CunningPike: ah, that might be the case.. maybe it got corrupted |
| 18:36.07 | [TK]D-Fender | joeTSUNAMI: Then create a new folder and up/downgrade it. that should cause a local filesystem reformat. |
| 18:36.14 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:36.31 | [TK]D-Fender | joeTSUNAMI: From there you can begin doing this properly from scratch. |
| 18:36.53 | joeTSUNAMI | [TK]D-Fender: so just needs the config file? not trying to do a firmware upgrade, right? |
| 18:37.41 | [TK]D-Fender | joeTSUNAMI: You want to do the upgrade/downgrade as a means of ensuring that the filesystem is flushed. You can revert to your standard versions afterwards. |
| 18:38.17 | joeTSUNAMI | ah, ok.. i'll try that out. thanks all! |
| 18:42.30 | Waverly360 | Does anyone know if there's a limit to the length of a context name? |
| 18:43.21 | Corydon-w | 80 characters |
| 18:44.15 | Waverly360 | Wow..that many? |
| 18:44.22 | Waverly360 | I must be doing something else wrong then. |
| 18:44.26 | Corydon-w | Okay, it might only be 79 |
| 18:44.37 | wunderkin | .5 |
| 18:45.48 | Corydon-w | Yep, it's only 79. |
| 18:45.56 | wunderkin | that's an odd number |
| 18:46.06 | Qwell[] | no it's not |
| 18:46.12 | Corydon-w | 80 - 1 for the terminating null |
| 18:46.19 | wunderkin | oh |
| 18:46.19 | Waverly360 | well.. "technically" 79 is an odd number |
| 18:46.27 | Waverly360 | :P |
| 18:46.37 | Corydon-w | ditto for extensions. 79 max characters |
| 18:46.56 | Corydon-w | and channel names have the same limitation |
| 18:47.01 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 18:47.01 | *** mode/#asterisk [+o russellb] by ChanServ |
| 18:47.05 | *** join/#asterisk knobenheimer (n=knoben@c-71-205-184-144.hsd1.mi.comcast.net) |
| 18:47.33 | Corydon-w | Language names and music classes are both max 19 characters |
| 18:48.05 | russellb | in 1.2 anyway |
| 18:48.16 | Corydon-w | Well, okay |
| 18:48.24 | russellb | in 1.4, they may be unlimited ... |
| 18:48.27 | russellb | :-p |
| 18:48.38 | Corydon-w | Actually, no |
| 18:48.51 | russellb | you're right |
| 18:48.51 | *** join/#asterisk poppo (n=adas@S0106004063d8e527.ed.shawcable.net) |
| 18:48.52 | Corydon-w | Music classes were expanded to 80 in 1.4 |
| 18:48.58 | russellb | it's still limited in some places |
| 18:49.15 | russellb | i was thinking since it was just a stringfield on the channel struct ... oh well |
| 18:49.20 | poppo | Does anybody have experience with Asterisk and ruby on rails? |
| 18:49.20 | *** join/#asterisk JT (n=jon@unaffiliated/jt) |
| 18:49.36 | Corydon-w | Even those static sizes are incredibly large for just about every usage imaginable |
| 18:49.47 | russellb | yep |
| 18:50.01 | Corydon-w | I don't know that the PSTN could handle a 79-digit phone number |
| 18:50.02 | Waverly360 | Ok..well what about a max size on username in sip.conf? |
| 18:50.55 | Corydon-w | 2255 |
| 18:50.58 | Corydon-w | oops, 255 |
| 18:51.10 | Waverly360 | damnit..I have no idea what's going on here |
| 18:52.48 | perd | is there a working chan_sccp for 1.4 yet? |
| 18:53.01 | Corydon-w | Why not use chan_skinny for 1.4? |
| 18:53.10 | perd | doesnt work on 7902 |
| 18:53.18 | Corydon-w | File a bug |
| 18:53.58 | Corydon-w | chan_sccp is one of the worst-coded modules I've seen recently. I was looking through it yesterday and found several huge memory leaks |
| 18:56.13 | perd | blah |
| 18:56.49 | perd | maybe i should try svn |
| 18:56.54 | perd | seems like a bad idea for production though |
| 18:57.44 | [TK]D-Fender | Skinny phones outside of CM/CME is a bad idea period. When in Roma.... |
| 18:58.54 | perd | <PROTECTED> |
| 18:58.57 | perd | hehe |
| 18:59.33 | *** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net) |
| 18:59.53 | Waverly360 | Ok, I need someone to look at this part of my sip.conf and tell me if I'm missing something. [pbx]\n type=friend\n context=phones\n username=pbx\n password=pbx\n |
| 19:00.12 | blitz[training] | host= |
| 19:00.14 | perd | ho |
| 19:00.15 | perd | bah |
| 19:00.25 | Waverly360 | but why do I have to have host? |
| 19:00.34 | perd | it wont work otherwise |
| 19:00.35 | blitz[training] | something to parse on |
| 19:00.35 | Waverly360 | I don't want it to match on the hostname |
| 19:00.44 | perd | dynamic = no matching |
| 19:00.51 | Waverly360 | yeah, but then you have to register |
| 19:01.00 | cpatry | im getting: [Feb 13 13:57:08] WARNING[1889]: chan_sip.c:2836 sip_call: No audio format found but if im doing sip show peer <peer> ive: Codecs : 0x6 (gsm|ulaw) |
| 19:01.01 | cpatry | <PROTECTED> |
| 19:01.10 | cpatry | how come? |
| 19:03.02 | *** join/#asterisk Janos (n=Janos@nat1.inalambrica.net) |
| 19:03.51 | oej | cpatry: Does the call really match the peer? |
| 19:03.52 | poppo | Does anybody have experience with Asterisk and ruby on rails? |
| 19:04.15 | *** join/#asterisk PhilKC (i=greece@freenode/staff/about.linux.philkc) |
| 19:04.40 | cpatry | yep |
| 19:07.34 | putzz | ~pb |
| 19:07.37 | jbot | methinks pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 19:07.58 | *** join/#asterisk dansmith (n=dan@gw0.danplanet.com) |
| 19:08.05 | Waverly360 | Ok...well maybe one of you guys can help me solve my problem. I have multiple asterisk pbxes, and I want them to be able to dial extensions on each other freely. |
| 19:08.20 | Waverly360 | I do not, however, want one server to be the master. |
| 19:08.38 | Waverly360 | I want each of them to be equals if at all possible. |
| 19:08.46 | dansmith | anyone have any comments about the PolyCom 430 vs. the 301? |
| 19:09.12 | dansmith | not sure how much their "echo cancellation" would make a difference, or if the speakerphone is much better |
| 19:09.14 | robl^ | 430 nice, 301 not so nice |
| 19:09.24 | robl^ | the 30 has no speaker phone |
| 19:09.28 | *** join/#asterisk topping (n=topping@h-67-100-91-18.snfccasy.covad.net) |
| 19:09.32 | robl^ | er.. 301 |
| 19:09.32 | dansmith | oh, really? I just assumed |
| 19:09.42 | oej | cpatry: Then you have a misconfiguration or a bug |
| 19:09.43 | dansmith | well, that settles it.. thanks :) |
| 19:09.57 | [TK]D-Fender | dansmith: Do you have PoE already? |
| 19:10.08 | dansmith | [TK]D-Fender: no |
| 19:10.16 | [TK]D-Fender | dansmith: If not just bypass the IP 430 and get an IP 501 for the extra $20 |
| 19:10.24 | CunningPike | Waverly360: DUNDi? |
| 19:10.48 | blitz[training] | CunningPike: that just passes information and nothing to do with SIP authentication |
| 19:11.05 | Waverly360 | DUNDi? |
| 19:11.08 | dansmith | [TK]D-Fender: oh, do I have to have PoE for the 430? |
| 19:11.10 | blitz[training] | ~dundi |
| 19:11.11 | jbot | methinks dundi is http://www.dundi.com |
| 19:11.11 | brodiem | I have a question about setting TIMEOUT(absolute). If I specify it in a dial plan, which its end result is to launch Queue(), will that timeout carry over into the Queue application channel? |
| 19:11.41 | dansmith | [TK]D-Fender: nm, I see that it comes with an AC adapter |
| 19:11.59 | [TK]D-Fender | dansmith: No, but thats the economic reason to choose it over the IP 501. Because IP 501 + the added PoE cable would be around 40+$ more expensive. |
| 19:12.09 | dansmith | ah |
| 19:12.12 | Waverly360 | CunningPike: I don't think that's what I'm lookin for, but thanks :) |
| 19:12.18 | [TK]D-Fender | dansmith: its a question of bang/buck |
| 19:12.19 | blitz[training] | yah, it's not |
| 19:12.21 | CunningPike | Waverly360: OK |
| 19:12.24 | dansmith | [TK]D-Fender: thanks |
| 19:12.47 | [TK]D-Fender | dansmith: IP 501 adds another line-key, larger display with greater resolution, and a slightly sturdier base. |
| 19:12.58 | dansmith | yea, sounds worth it |
| 19:13.18 | [TK]D-Fender | dansmith: The advantage also carries through when you consider the MicroBrowser |
| 19:13.28 | dansmith | what's the microbrowser? |
| 19:13.48 | dansmith | and what about the headset jack? Can I use any RJ-11 headset there? |
| 19:13.53 | [TK]D-Fender | dansmith: And XHTML capable browser for showing live content on your phone. |
| 19:13.59 | dansmith | whoa |
| 19:14.01 | dansmith | crazy |
| 19:14.29 | [TK]D-Fender | dansmith: It uses the standard RJ12 (I believe its numbered), connector. The same size as a handset jack, but the wiring is different IIRC. |
| 19:14.49 | dansmith | ah |
| 19:14.54 | dansmith | so I have to use a polycom headset? |
| 19:15.15 | dansmith | ...or get creative :P |
| 19:15.22 | [TK]D-Fender | dansmith: No itsa common enough stardard used by many companies like GN Netcom, Plantronics, etc. |
| 19:15.26 | mcab | no, standard headsets like plantronics will work |
| 19:15.27 | Janos | i have an asterisk server also working as the firewall/gateway of a couple of xlite soft-phones, i also have another soft-phone that registers from the internet, the problem is that when i try to talk between the external and one of the internal phones i get a one way audio, can anyone give me some hints on this ? |
| 19:15.29 | dansmith | ah, ok |
| 19:15.35 | dansmith | [TK]D-Fender: you're just full of information :) |
| 19:15.54 | brodiem | dansmith make sure you use a headset with an amplifier |
| 19:16.12 | dansmith | brodiem: yea, that's what I have at work, I figured I'd get the same thing if it would work |
| 19:16.17 | [TK]D-Fender | Indeed. Unamped headsets = suck. Plantronics M12 is quite decent. |
| 19:16.22 | dansmith | it's a plantronics |
| 19:16.38 | [TK]D-Fender | dansmith: Then you should be able to use them with it jsut fine |
| 19:16.52 | brodiem | [TK]D-Fender I opted to use snom 360's since they have a built-in amp = less clutter and vol/mic controls |
| 19:17.13 | dansmith | well, at work, I have an Avaya phone and was just given all the pieces to plug together |
| 19:17.15 | [TK]D-Fender | dansmith: And avoid the Plantronics Voice-Tube headsets. Too damn sensitive and jittery. You'd swear thats what they used for MCI commercials... |
| 19:17.25 | dansmith | although it doesn't have a headset jack, the amp just goes between the handset and the phone |
| 19:17.29 | Qwell[] | jittery mci commercials? |
| 19:17.34 | dansmith | heh |
| 19:17.40 | dansmith | but the tube looks so cool :) |
| 19:17.41 | Qwell[] | so much jitter you can hear a pin drop? |
| 19:17.44 | [TK]D-Fender | brodiem: I opt for Polycom, because I'd like to ensure that my sound source doesn't suck BEFORE amping :) |
| 19:17.51 | brodiem | lol |
| 19:17.57 | [TK]D-Fender | Qwell[]: 2 points to you.... |
| 19:18.27 | [TK]D-Fender | Snom is pretty far down my suggest list.... |
| 19:18.39 | *** join/#asterisk scarfy (n=scarfy@84-73-85-22.dclient.hispeed.ch) |
| 19:18.59 | brodiem | [TK]D-Fender why the hate for snom? |
| 19:19.29 | scarfy | Hi, i just compiled/installed/configured asterisk for the first time and after that i get this error when i open the asterisk console.. what does it mean? is a conf file wrong? |
| 19:19.30 | scarfy | [Feb 13 20:07:46] WARNING[32486]: chan_sip.c:1735 __sip_xmit: sip_xmit of 0xb7923fa0 (len 374) to 86.64.162.35:5060 returned -1: Bad file descriptor |
| 19:19.34 | *** join/#asterisk ucfMethod (n=ucfmetho@c-69-143-112-70.hsd1.va.comcast.net) |
| 19:19.35 | [TK]D-Fender | brodiem: Crappy display, flakey firmware, marginal sound quality, and too close in price to Polycom to have a prayer for consideration. |
| 19:20.06 | mitcheloc | they are nice and programmable though |
| 19:21.38 | brodiem | I'd have to agree that the snom's SQ doesn't compare to the polycom conference phone we have (which is actually one of the analog ones plugged into an ATA) but I don't think I've used a phone with more configuration and customization options |
| 19:22.08 | *** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it) |
| 19:22.27 | brodiem | the newer firmware reduces the text size on the menus though |
| 19:22.46 | [TK]D-Fender | mitcheloc / brodiem : I don't disagree that it can be more configurable, but I wouldn't sacrifice stability & quality for it. Its that's the case, the 480I is my runner-up and offers a boatload of features. |
| 19:23.06 | *** join/#asterisk juice (n=juice@mo-76-0-14-195.dhcp.embarqhsd.net) |
| 19:23.23 | russellb | scarfy: you probably have another application running that is using port 5060, such as Ekiga or something ... |
| 19:23.36 | brodiem | i've got some 480i's also.. I like the look&feel of the 480i better but still find overall the snom is better, IMO of course.. |
| 19:23.38 | [TK]D-Fender | Which for sure beats out Polycom on features, but sacrifices higher-end call handling easy of use and flexibility. |
| 19:23.39 | mitcheloc | then we could fix the quality/reliability issues ourselves :) |
| 19:23.45 | scarfy | russellb, aaaaaaaaaaah ok THANKS so much |
| 19:23.55 | russellb | scarfy: you're welcome |
| 19:24.41 | [TK]D-Fender | mitcheloc: Probably not opensourcing the firmware so as not to incur a large number of phone returns due to "bricking". |
| 19:25.37 | mitcheloc | they could make more money chargine a service fee for bricked devices! or a work around to keep people from bricking them, i don't think thats a valid excuse... |
| 19:25.49 | Qwell[] | jtag FTW |
| 19:25.59 | mitcheloc | ahh just thinking of running asterisk directly on the phone heh |
| 19:26.13 | mitcheloc | (jk) |
| 19:26.20 | brodiem | the snom's run a linux-based firmware |
| 19:26.33 | Qwell[] | brodiem: very little |
| 19:26.41 | *** join/#asterisk KnowWhat (n=KnowWhat@125.209.64.75) |
| 19:26.50 | Corydon-w | mitcheloc: were you aware that the snom 100 had an IAX firmware image? |
| 19:27.03 | brodiem | figured that, since the linux kernel has no real telephony features built in:) |
| 19:27.30 | mitcheloc | Corydon-w: no i wasn't, is the snom 100 a current product? firmware complete? |
| 19:27.44 | Corydon-w | No, the snom 100 was snom's first phone |
| 19:27.50 | Corydon-w | now discontinued |
| 19:27.59 | brodiem | they have a new phone coming out in a couple of months |
| 19:28.13 | mitcheloc | i see |
| 19:28.16 | brodiem | so I was told by one of them at the IT expo |
| 19:28.17 | Corydon-w | brodiem: when is that ever not true? |
| 19:28.48 | mitcheloc | it's too bad nokia's n800 wasn't a tad bit better, it serves as such as sweet wifi voip phone (though over gizmo/skype) that it would make a nice desk phone too with a proper headset |
| 19:29.20 | Qwell[] | Corydon-w: I just had a silly thought |
| 19:29.29 | Qwell[] | Corydon-w: bluetooth HFP softphone |
| 19:29.29 | Corydon-w | Qwell[]: just one? |
| 19:30.02 | Corydon-w | That's not all that silly |
| 19:30.14 | Qwell[] | not bluetooth for a headset |
| 19:30.18 | Qwell[] | I mean bluetooth for the protocol |
| 19:30.21 | Qwell[] | :p |
| 19:30.31 | Corydon-w | I know |
| 19:31.10 | Corydon-w | It's basically the same as doing SIP over wireless, only with a different protocol |
| 19:31.16 | kippi | whats the best way so that a user can divert there extenstion to a mobile phone? |
| 19:31.51 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 19:32.02 | Corydon-w | Qwell[]: however, I can think of a very useful application: recording your cell phone calls... |
| 19:32.11 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 19:32.13 | genz | kippi: Call forwarding? like *72? |
| 19:32.18 | Qwell[] | eh? |
| 19:32.24 | Corydon-w | (on your laptop) |
| 19:32.30 | kippi | is it easy to setup? |
| 19:32.36 | Qwell[] | but what's it gonna bluetooth...to? |
| 19:32.50 | genz | *72 should work with Asterisk out of the box. |
| 19:32.57 | brodiem | I have a question about setting TIMEOUT(absolute). If I specify it in a dial plan, which its end result is to launch Queue(), will that timeout carry over into the Queue application channel? |
| 19:32.57 | Corydon-w | You just put Asterisk on the laptop and put the laptop via bluetooth between the cell phone base and the cell phone headset |
| 19:32.58 | kippi | ah ok |
| 19:33.09 | Qwell[] | then you wouldn't need a softphone |
| 19:33.16 | Qwell[] | I mean, unless you did a reinvite :P |
| 19:33.25 | Qwell[] | which would be hilarious |
| 19:33.29 | Corydon-w | Qwell[]: well, conferencing, too |
| 19:33.58 | Qwell[] | and yeah, I could see that scenario |
| 19:33.59 | Corydon-w | Qwell[]: you wouldn't really need a softphone as you could use the console driver |
| 19:34.14 | Qwell[] | well, if you had a (bluetooth) headset... |
| 19:34.25 | Corydon-w | Yep |
| 19:34.53 | CJLinst | Anyone to have time to discuss the relationship between the sip.conf call-limit, extension subscriptions, and sending more than one call to the call-limited extension? |
| 19:35.47 | mercestes | CJLinst: For fun or profit? |
| 19:35.50 | *** join/#asterisk Kritter (i=kritter@cain.strange.com) |
| 19:36.10 | CJLinst | for fun, of course. |
| 19:36.29 | mercestes | sorry, I wasted away all my fun time playing video games at work..:( |
| 19:36.46 | CJLinst | how about I explain how I figure it works and you can tell me if I'm either warm or cold. |
| 19:37.12 | *** join/#asterisk jpe-nyc (n=jpe@p77-37.acedsl.com) |
| 19:38.05 | mercestes | Ooo, I like this game. |
| 19:38.11 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 19:39.11 | CJLinst | Using 1.4-SVN... If I want the subscriptions to work on my snom, I have to set call-limit=1 on the SIP channel. or the BLF lamp will not illuminate until the call-limit is reached on that device. |
| 19:39.45 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 19:40.15 | *** part/#asterisk w0ls0n (n=Me@43-141-135-64.dsl.sacoriver.net) |
| 19:40.30 | CJLinst | What I'm noticing, is that if I do set call-limit=1, calls to that device are returned CHANUNAVAIL, not BUSY as the SIP driver never tries the extension because of the call-limit=1. |
| 19:41.04 | mercestes | ok |
| 19:41.06 | mercestes | with you so far |
| 19:41.10 | poppo | Anybody Has experience with RAGI???????? Need freelancer!! |
| 19:41.24 | mercestes | poppo: ragi? Ruby AGI? |
| 19:41.26 | *** join/#asterisk hegars (n=hegars@203.161.78.66.static.amnet.net.au) |
| 19:41.28 | CJLinst | unfortunately, this means I have to choose between BLF functionality and being able to detect BUSY (and forward to VoiceMail(b${EXTEN}), etc. |
| 19:41.30 | hegars | hello all |
| 19:41.45 | mercestes | CJLinst: Lost you there. Cold. Abort. |
| 19:41.55 | mercestes | CJLinst: Why not have both? |
| 19:42.00 | poppo | mercestes: Yes |
| 19:42.14 | genz | Does anyone know how to set it up so when a user logs off a phone (*12) it can be forwarded to another number (not just extension)? |
| 19:42.23 | mercestes | poppo: Isn't ruby loosely c based? |
| 19:42.33 | CJLinst | Is this where peer vs user vs friend comes in? |
| 19:42.33 | Qwell[] | mercestes: not at all |
| 19:42.45 | poppo | merceses: Not sure |
| 19:42.57 | [TK]D-Fender | CJLinst: This is a conceptual conflict. Technically the SIP device MAY be able to accept another call, so * is reporting that its just "not happening". If you can't forsee any other reason why the channel should REALLY be unreachable, you should be able to treat this accordingly and jsut do "b" for it |
| 19:43.01 | poppo | merceses: Just looking for somebody to do some work with my rails app and asterisk |
| 19:43.54 | mercestes | poppo: ...why don't you explain on a more fundamental level what you want to happen v/s what is happening now? or must it absolutely be ragi? |
| 19:44.02 | CJLinst | Right. and switch to bEXT on BUSY or UNAVAIL... |
| 19:44.16 | sweeper | poppo: I can give you some TIPS |
| 19:44.23 | sweeper | but RAGI is pretty straightforward |
| 19:44.35 | sweeper | if you know rails, might as well diy |
| 19:45.06 | sweeper | mercestes: ragi > you! |
| 19:45.39 | CJLinst | It'd be nice if subscriptions would return as BUSY if calls > 0 instead of calls > call-limit, but I'm sure that'd break something someone else is trying to do. |
| 19:46.23 | poppo | sweeper: What i am trying to do is i have this application rails and user buys events from my site |
| 19:46.33 | poppo | What i want to do is 3 days before the event happens |
| 19:46.42 | *** join/#asterisk AJaymn (n=mypocket@24-159-236-181.dhcp.mdsn.wi.charter.com) |
| 19:46.47 | mercestes | CJLinst: unavailable is different than unreachable. unavailable is "called with no answer." unreachable is "cannot be rung." You can still divert to VM on an unreachable. |
| 19:46.51 | poppo | asterisk will call the register users and will do a txt to speak of there event |
| 19:46.57 | poppo | Thats all |
| 19:49.11 | nortex | Has anyone here used the systemname option in asterisk.conf with realtime to set the system a sip device is registered to? |
| 19:49.33 | sweeper | poppo: don't even need RAGI, really |
| 19:49.52 | *** join/#asterisk christel (i=christel@freenode/staff/gentoo.christel) |
| 19:50.20 | poppo | ok |
| 19:50.36 | poppo | Sweeper: what would you say i use |
| 19:50.39 | CJLinst | mercestes: thanks |
| 19:50.49 | poppo | Sweeper: I know i can create a call file no problem with Rails |
| 19:50.59 | poppo | sweeper: but what about the txt to speach part |
| 19:51.03 | CJLinst | mercestes: Do you know if the queue app treats CHANUNAVAIL the same as BUSY? |
| 19:51.15 | poppo | sweeper: can i have that info store on the call file |
| 19:51.29 | sweeper | poppo: you can do text to speech in your dialplan, and thus, you can do it in a call file |
| 19:52.03 | sweeper | timtowdi and such |
| 19:52.07 | poppo | but i need the info grab from the ruby on rails app |
| 19:52.10 | poppo | or mysql |
| 19:52.24 | sweeper | right |
| 19:52.31 | sweeper | to create the callfile |
| 19:52.45 | sweeper | I'd suggest you create the call file when the user buys the event |
| 19:53.00 | sweeper | and future-date it |
| 19:53.00 | poppo | yes i get that |
| 19:53.07 | *** join/#asterisk Kritter (i=kritter@cain.strange.com) |
| 19:53.23 | poppo | What about the txt to speak |
| 19:53.29 | sweeper | also generate whatever text file you want read when the user buys the event |
| 19:53.37 | poppo | ohh ok |
| 19:53.49 | sweeper | then the future-dated call file tells asterisk to read the text file to the user |
| 19:54.12 | poppo | sweeper: can i have 1 txt file per event |
| 19:54.17 | sweeper | sure |
| 19:54.35 | sweeper | in which case, that text file would be created when you create the event |
| 19:54.37 | poppo | then in the call file have a txt to speak like Hello %Name% then the file to play |
| 19:54.45 | sweeper | yep! |
| 19:54.49 | sweeper | tre simple |
| 19:54.55 | poppo | sweeper: cause that might be better if i have like 40 users |
| 19:55.10 | sweeper | eh, text files are small :P |
| 19:55.16 | *** join/#asterisk kervel (n=kervel@cable-213-132-140-61.upc.chello.be) |
| 19:55.32 | sweeper | but it would definately be easier, at least you wouldn't have to keep track of individual call files |
| 19:55.45 | kervel | hello, i'm using asterisk with snom 300 phones, and DTMF won't work |
| 19:55.50 | Kritter | I have a question that I think has to do with SIP reinvtes and nat... two sip phones have zero sound passing. :( |
| 19:56.05 | kervel | when i hit a key on the snom phone, i can hear the dtmf tone, but the other side doesn't react |
| 19:56.16 | *** join/#asterisk heh_v_water (n=heh_v_wa@71-210-49-107.hlna.qwest.net) |
| 19:56.27 | kervel | in the sip.conf i use rfcXXXX out of band dtmf |
| 19:56.27 | Kritter | check how dtmf is being sent. |
| 19:57.06 | kervel | the setup is like this : snom 300 <-> ethernet <-> asterisk <-> digium BRI ISDN |
| 19:57.37 | kervel | Kritter is there a way to see if asterisk receives the DTMF tones ? |
| 19:57.45 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:58.04 | kervel | and is sip the only place where things can go wrong ? or is the BRI interface a potential problem too ? |
| 19:58.26 | Kritter | might be able to debug it in the console, though I came for help on my no sound issue from sip phone to sip phone, so am hardly an expert. |
| 19:58.37 | kervel | hehe :) |
| 19:58.43 | kervel | ,r |
| 19:58.47 | kervel | Kritter NAT issues ? |
| 19:58.48 | Kritter | I know for my dtmf problem I had to change a setting on the phones. |
| 19:59.12 | kervel | the SNOM phones auto-negotiate dtmf settings, i cannot change manually |
| 19:59.27 | Kritter | all the research I've done has pointed me at NAT as suspect, but they're on a local segment - and I think I've tried all permuation of the nat values. |
| 19:59.47 | kervel | and also, i have to wait untill my ISP calls me to tell me a SMS was received before i can debug :) |
| 20:00.11 | kervel | Kritter they are on the same subnet ? |
| 20:00.30 | Kritter | yes, and that subnet is declared. |
| 20:00.36 | kervel | Kritter softphones on machines with multiple IP addresses ? |
| 20:00.47 | kervel | or ain't it the trivial stuff ? :) |
| 20:00.54 | Kritter | it's a proof of concept test, nope, grandstream 4 lines. |
| 20:00.59 | genz | Does anyone know how to set it up so when a user logs off a phone (*12) it can be forwarded to another number (not just extension)? |
| 20:01.07 | kervel | Kritter and call setup works, just no sound ? |
| 20:01.40 | Kritter | ringing happens, and solo they can call the VM system, issue dtmf tones, leave messages for each other etc. |
| 20:01.54 | kervel | weird |
| 20:01.56 | Kritter | but if you actually wanted to talk... good luck |
| 20:02.07 | kervel | Kritter no codec issue ? |
| 20:02.14 | *** join/#asterisk Dr-Linux|home (n=Dreamer@DSL-202-59-73-131.nexlinx.net.pk) |
| 20:02.27 | Kritter | the phones do display different codecs on the screen. |
| 20:02.47 | Kritter | but they're the same model phones with the same codec's availible. |
| 20:03.09 | Kritter | that's my other debugging line after the NAT investigation. |
| 20:03.12 | kervel | Kritter and you cannot set the codec manually |
| 20:03.54 | Kritter | each phone has a webstack, with codec choices. but they match, not sure why they are picking different ones. |
| 20:05.50 | Kritter | is there a best choice codec folks like? |
| 20:05.54 | Kritter | I'll disable all but it. |
| 20:06.06 | *** join/#asterisk X-Rob (n=rob-x@124.150.99.11) |
| 20:07.47 | kervel | Kritter maybe just pick one to find out if that's the problem |
| 20:08.16 | JT | g.711a and g.711u are the best quality codecs |
| 20:08.28 | endre | since they are not codecs |
| 20:08.31 | endre | only pcm stream |
| 20:08.40 | JT | threy are the native telephone network codecs |
| 20:08.45 | JT | incorrect |
| 20:08.51 | JT | they are companded audio |
| 20:08.53 | *** join/#asterisk loufoque (n=loufoque@smj33-1-82-233-64-48.fbx.proxad.net) |
| 20:08.56 | JT | not quit pcm streams |
| 20:08.59 | JT | quite |
| 20:09.00 | endre | umm |
| 20:09.03 | endre | 64kbit, rite? |
| 20:09.12 | JT | right |
| 20:09.24 | endre | that's 8000Hz 8 bit audio? |
| 20:09.26 | loufoque | I would like to forward calls to mobile phones to a mobile phone, what kind of hardware do I need to do that? |
| 20:09.30 | Kritter | well my lovely options appear to be PCMU, PCMA, G.723.1 G.729A/B |
| 20:09.46 | JT | no, they are NOT pcm straight, that's what the Mu law and A law are all about |
| 20:09.59 | cpatry | loufoque: gsm gateways? |
| 20:10.13 | JT | companding is like a form of compression, they have closer to 16bits of dynamic range than 8bits as a result |
| 20:10.18 | endre | afaik, u is big-endian, a is little-endian or wrat |
| 20:10.24 | JT | NO |
| 20:10.29 | endre | ok, you win |
| 20:10.36 | endre | you can have all my internets |
| 20:12.02 | kervel | loufoque http://www.voipsolutions.be/pbx/gsm-gateway/mv-370-voip-gsm-gateway.html ? but it's expensive apparently |
| 20:14.19 | endre | i rtfm'd |
| 20:14.23 | endre | JT: sorry for bothering |
| 20:15.49 | loufoque | what about that Celliax thing? |
| 20:16.17 | loufoque | couldn't it act as a gsm gateway? The adapter should be less costly. |
| 20:16.18 | poppo | Anybody can point me to an call example were there is txt to speach in the call file without modifying the extension or call files? |
| 20:17.56 | mercestes | poppo: festival the txt file maybe? |
| 20:18.00 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-154-6-99.red.bezeqint.net) |
| 20:18.04 | poppo | yes |
| 20:21.47 | *** part/#asterisk joeTSUNAMI (n=joeTSUNA@72.243.118.59) |
| 20:22.27 | Kritter | well I locked one down the GSM, and now it works. |
| 20:24.05 | *** join/#asterisk thekidrio (n=thekidri@66.107.42.13) |
| 20:30.26 | *** join/#asterisk elriah (i=elriah@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 20:30.40 | elriah | Anyone have a sample dialplan.xml for asterisk and the Cisco phones? |
| 20:30.45 | thekidrio | how important is hard drive space on an asterisk box? |
| 20:31.03 | elriah | thekidrio: Not very, even if you record calls its minimal. |
| 20:31.06 | j0 | thekidrio: well, funny things happen when it fills up.. |
| 20:31.10 | thekidrio | heh |
| 20:31.19 | thekidrio | i have 5 10krpm drives in there |
| 20:31.23 | thekidrio | but they are only 36 gig each |
| 20:31.28 | j0 | lol |
| 20:31.31 | thekidrio | i am used to proxy/image serving |
| 20:31.37 | thekidrio | so thats not much space for that haha |
| 20:31.40 | Qwell[] | asterisk requires at least 500GB |
| 20:31.51 | Qwell[] | (I'm kidding, obviously) |
| 20:31.53 | elriah | Qwell[]: lol |
| 20:31.57 | [TK]D-Fender | elriah: Perhaps you should look on the WIKI for a more complete guide to setting up Cisco phones.... |
| 20:32.26 | elriah | [TK]D-Fender: Got the phone working great with the lastest "generation 2" firmware direct from TAC. I just can't outsmart the dialplan.xml and I've been all over the wiki... |
| 20:33.24 | thekidrio | so 36x5 should be enough for a while right? |
| 20:33.41 | thekidrio | the server is going to be fairly far away from me so i dont want to drive to it often |
| 20:33.42 | elriah | thekidrio: 2gb would be enought for quite a while... |
| 20:34.17 | thekidrio | hahhaha |
| 20:34.17 | thekidrio | hrmm |
| 20:34.17 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 20:34.17 | thekidrio | maybe i should take back some of these drives |
| 20:34.17 | *** join/#asterisk Hmmhesays (n=Neg@24-117-135-28.cpe.cableone.net) |
| 20:34.18 | thekidrio | slap them in some other server |
| 20:34.18 | KnowWhat | hello all |
| 20:34.18 | KnowWhat | exten => _991NXXXXXXXXX,2,Dial(${TRUNKASTERLINK}/${EXTEN:2},55,o) |
| 20:34.21 | KnowWhat | what doesthat line means? |
| 20:35.47 | [TK]D-Fender | KnowWhat: "show application dial" |
| 20:36.13 | *** join/#asterisk ruied (n=ruied@bl9-252-99.dsl.telepac.pt) |
| 20:36.28 | [TK]D-Fender | KnowWhat: http://www.voip-info.org/wiki-Asterisk+variables |
| 20:36.38 | [TK]D-Fender | ~book |
| 20:36.45 | jbot | book is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 20:39.08 | KnowWhat | thanks |
| 20:42.05 | *** join/#asterisk dhill (i=dhill@fog.mindcry.org) |
| 20:42.18 | dhill | anyone know what http://pastebin.ca/336644 is caused by? |
| 20:42.25 | dhill | i am getting that with a few sip clients |
| 20:45.40 | poppo | I am moving call file to /var/spool/asterisk/outgoing and it not doing anything |
| 20:46.58 | poppo | never mind |
| 20:49.14 | mercestes | Done! |
| 20:49.34 | sevard | Done! |
| 20:49.40 | KnowWhat | _1NXXNXXXXXX == 1 530 354 5639 |
| 20:49.45 | KnowWhat | am i right? |
| 20:49.56 | sevard | NERRGG |
| 20:49.58 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 20:50.41 | KnowWhat | sevard what? |
| 20:50.50 | mercestes | KnowWhat: damnit! I got 1 208 704 7658 I guess we're both right. |
| 20:51.27 | sevard | Bahwahgha. |
| 20:51.56 | KnowWhat | mercestes i want some one who have faith in it :P |
| 20:52.51 | sevard | hoooooooooe lordy somebody's got the faith, can I get an AMEN choir? |
| 20:53.04 | zamba | anyone got a TE110P card here and can tell me what the lights on the back of the card means? |
| 20:53.10 | zamba | .. if anything |
| 20:53.28 | KnowWhat | back of... the card kool |
| 20:53.31 | sevard | zamba: if/ifnot zombies are atacking |
| 20:53.43 | KnowWhat | sevard well i am unable to dial out through the vitelity |
| 20:53.55 | zamba | oh, come on.. :) |
| 20:53.59 | sevard | serious! |
| 20:54.02 | sevard | do you have red lights? |
| 20:54.04 | KnowWhat | Everyone is busy/congested at this time (1:0/0/1) |
| 20:54.08 | zamba | i'm trying to find the manual for the card on the net, but i can't find it |
| 20:54.13 | Bobthehunter | having a hard time figuring out MSD stripping or not for pri |
| 20:54.19 | Bobthehunter | depedning if its local or not |
| 20:54.27 | zamba | sevard: i have one red light, yeah |
| 20:54.27 | sevard | if [redlights]; then hideWithGuns.g ; fi |
| 20:54.36 | KnowWhat | i am getting this error .. Everyone is busy/congested at this time (1:0/0/1) |
| 20:54.37 | Bobthehunter | EX: 303640 could be local bu 303641 not.. |
| 20:54.48 | CJLinst | do polycoms really need dtmfmode=inband? |
| 20:55.57 | *** join/#asterisk GreenK (i=GreenK@207-119-216-204.dyn.centurytel.net) |
| 20:56.56 | *** join/#asterisk scarfy (n=scarfy@84-73-85-22.dclient.hispeed.ch) |
| 20:57.07 | zamba | ok, i finally found something in the knowledge base at digium.. thanks for your help, sevard :) |
| 20:57.12 | sevard | no problem! |
| 20:58.03 | KnowWhat | sevard what about me sevard |
| 20:58.17 | sevard | KnowWhat: what about me KnowWhat |
| 20:58.28 | KnowWhat | sevard tell you KnowWhat |
| 20:58.30 | scarfy | Hi, i'm trieng to configure my asterisk to connect to my ekiga.net account.. but it always give this error.. what does it mean? |
| 20:58.31 | scarfy | [Feb 13 21:56:34] NOTICE[3556]: chan_sip.c:7085 sip_reg_timeout: -- Registration for 'X@ekiga.net' timed out, trying again (Attempt #1) |
| 20:58.31 | scarfy | <PROTECTED> |
| 21:01.00 | Hmmhesays | does anyone know if squid reads the config file every time a request comes in? |
| 21:01.02 | Hmmhesays | for the acls |
| 21:01.12 | kervel | hmm, still having DTMF problems ... all the internal asterisk functions work, but when i call outside (ISDN) dtmf doesnt work anymore |
| 21:01.40 | kervel | so the phone<->asterisk does DTMF well, the asterisk<->digium BRI doesn't work well |
| 21:01.41 | *** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir) |
| 21:01.49 | [TK]D-Fender | Talks does Yoda funny hmmmmMMMM??!?!? |
| 21:01.50 | kervel | maybe some regional settings ? is that possible ? |
| 21:02.12 | sevard | Hmmhesays: that'd be horribly inefficient, i thought it loaded a binary version into ram that it checks against and you reloaded the squid file into ram with an application |
| 21:06.54 | *** join/#asterisk gr0mit_home (n=Tim@extrt.txrx.org.uk) |
| 21:09.59 | scarfy | anyone an idea what it could mean? |
| 21:11.05 | sevard | zombies? |
| 21:11.11 | *** join/#asterisk Skaag (n=skaag@212.199.180.157.static.012.net.il) |
| 21:11.32 | scarfy | zombies? :-/ |
| 21:11.50 | sevard | jump to the first conclusion |
| 21:11.54 | sevard | and solve the problem from there |
| 21:12.05 | sevard | therefore, I think you've got zombies in your airducts |
| 21:12.26 | sevard | you should go up there with a flamethrower and a 9mm, make sure you scortch them to slow them down then line up your sights and pump a single round into the forehead |
| 21:12.33 | scarfy | but my first conclusion was that it's something wrong my the sip.conf |
| 21:12.54 | *** join/#asterisk matiasds19 (n=matiasdo@201.253.208.136) |
| 21:13.02 | sevard | look buddy |
| 21:13.05 | sevard | who's the expert here |
| 21:13.06 | genz | CJLinst: still there? |
| 21:13.15 | *** join/#asterisk gmcinnes (n=gmcinnes@bas2-toronto63-1177698148.dsl.bell.ca) |
| 21:13.21 | scarfy | on zombies or asterisk? |
| 21:13.28 | gmcinnes | hi all. |
| 21:13.29 | sevard | zombies ON asterisk |
| 21:13.33 | sevard | not or. |
| 21:13.45 | scarfy | and flamethrowers |
| 21:13.49 | scarfy | uhu |
| 21:13.53 | sevard | flamethrowers and 9mm pistols are the remedy |
| 21:13.57 | sevard | not the issue at hand |
| 21:14.05 | sevard | dear boy, just go into the air duct |
| 21:14.37 | scarfy | so in short you have no clue what it means |
| 21:14.42 | sevard | i never said that |
| 21:15.09 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 21:15.10 | [TK]D-Fender | Graaaahhhgh! |
| 21:15.12 | scarfy | but you think its more fun to talk about airducts instead of giving a hint how to solve the problem |
| 21:15.27 | sevard | see?! |
| 21:15.32 | gmcinnes | what would y'all recommend as an analog FXS card? |
| 21:15.49 | genz | What could be the cause of this "ast_cdr_free: CDR on channel 'SIP/539-b6c19610' not posted"? |
| 21:15.49 | Qwell[] | gmcinnes: digium tdm400p |
| 21:15.53 | sevard | gmcinnes: they won't sell you one if you use "y'all" |
| 21:16.01 | sweeper | Qwell[]: I suspect bias! |
| 21:16.06 | gmcinnes | sevard: their loss :) |
| 21:16.07 | Qwell[] | of course |
| 21:16.19 | Qwell[] | sevard: please, digium is based in the south :) |
| 21:16.26 | CJLinst | genz: yes |
| 21:16.27 | sevard | niwgga plzz |
| 21:16.30 | Qwell[] | we give discounts (not really) when people say y'all! |
| 21:16.36 | sevard | "we"? |
| 21:16.44 | CJLinst | extra point for all y'all? |
| 21:16.51 | genz | CJLinst: Why do you ask about the Polycom dtmfmode setting? |
| 21:16.54 | sevard | since when do you work for digium? |
| 21:17.11 | gmcinnes | Its going in a dell poweredge, and I know there are problems with those servers and the tdm400 |
| 21:17.12 | scarfy | so would any of y'all know what my problem is? |
| 21:17.38 | gmcinnes | Is that resolved? |
| 21:18.08 | poppo | I have a call file with the application doing the following Application: Festival |
| 21:18.08 | poppo | Data: 'Hello asterisk user, how are you today?' |
| 21:18.11 | cpatry | Qwell[]: say CANTALOUPE to matt b. |
| 21:18.14 | CJLinst | Well, I dialed an extension with the 't' modifier and it seemed as though I heard just the first millisecond of DTMF when he tried to transfer using # but nothing happened. There's an example on voip-info that states dtmfmode=inband... |
| 21:18.18 | poppo | now when the user pickup it dosent say the msg ? |
| 21:18.23 | poppo | what is it that i am missing |
| 21:18.25 | xF|Nakaori | does anyone know, what it could be, that the asterisk mailbox doesnt say a thing if called from outside? |
| 21:18.44 | drako | where is located the asterisk database file? |
| 21:19.00 | CJLinst | genz: it's set up as rfc2833 at the moment. |
| 21:19.01 | sweeper | today must be esl |
| 21:19.04 | sweeper | *day |
| 21:19.23 | genz | CJLinst: What would be the benefit of the change? |
| 21:20.36 | [TK]D-Fender | cpatry: CANTALOUPE : When you get turned down for a Vegas... wedding |
| 21:20.49 | cpatry | drako: by default: /var/lib/asterisk/astdb |
| 21:21.09 | matiasds19 | hey, i need an asterisk expert, im going crazy with a SDP problem, someone here who can help? |
| 21:21.10 | drako | ty |
| 21:21.18 | *** join/#asterisk kuku5 (n=kuku5@c-71-201-219-72.hsd1.il.comcast.net) |
| 21:21.28 | kuku5 | Is 169$ for a cisco 7940 decent price ? |
| 21:22.04 | CJLinst | genz: well, the transfer didn't work. isn't it better to have out-of-band DTMF from the SIP phones and let Zap add the tones if they're in the loop. so to speak? |
| 21:22.23 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 21:22.29 | BSDTech | http://rapidshare.com/files/16327272/spip_ssip_2_1_0_release_sig.zip.html |
| 21:22.32 | BSDTech | enjoy |
| 21:23.26 | gmcinnes | no thoughts about tdm400p and dell poweredge servers? |
| 21:25.00 | sweeper | gmcinnes: you'd get more info if you knew what chipset and harddrive specs were |
| 21:25.16 | gmcinnes | yeah. I"m trying to check that. Is the harddrive an issue? |
| 21:25.25 | sweeper | mmm |
| 21:25.30 | sweeper | come to think of it |
| 21:25.37 | sweeper | I was thinking of T1 cards |
| 21:25.51 | *** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir) |
| 21:25.57 | JT_ | poweredge, that's not very specific |
| 21:26.57 | [TK]D-Fender | JT_ : Oh... its a Dell... does that help? ;) |
| 21:28.06 | gmcinnes | looks like its a xeon with an intel 5000x chipset. that help more/ |
| 21:28.07 | gmcinnes | ? |
| 21:29.44 | gmcinnes | I only say poweredge because this page from digium notes a whole lot of poweredge models that are incompatible. |
| 21:31.01 | JT_ | mm dual or quad core? |
| 21:31.35 | *** join/#asterisk etfonhome (n=etfonhom@74-140-206-4.dhcp.insightbb.com) |
| 21:31.39 | gmcinnes | dual core |
| 21:31.45 | *** join/#asterisk mpruett (n=mpruett@24-240-203-82.static.stls.mo.charter.com) |
| 21:31.50 | *** join/#asterisk JT (n=jon@unaffiliated/jt) |
| 21:32.27 | JT_ | gmcinnes: doing any virtualisation? |
| 21:32.50 | gmcinnes | no. It's probably overspeced for the problem at hand. |
| 21:32.57 | JT_ | ah ok |
| 21:33.32 | gmcinnes | I tried to talk them out of getting that much, but they want to 'use it in the future'. dumb, but its their money :) |
| 21:33.48 | JT_ | fair enough |
| 21:33.53 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 21:33.55 | gmcinnes | anyhow, I don't see any reports of problems with that intel 5000x chipset. |
| 21:34.02 | JT_ | as long as it's got dual power supplies and RAID1 |
| 21:34.06 | gmcinnes | yes. |
| 21:34.15 | [TK]D-Fender | bbiab |
| 21:34.23 | JT_ | yeah, VIA is usually the biggest problem chipset |
| 21:34.24 | JT_ | as always |
| 21:34.31 | Cyon | Hmmmm, so, I want to implement that fandy-dancy ability to prompt for a name, and then play it back to the person being called before the call goes through -- is there a nice doc someone on this on? I couldn't find it on voip-info, but it's gotta be there. |
| 21:35.00 | *** join/#asterisk lirakis (n=lirakis@ool-45775b9b.dyn.optonline.net) |
| 21:35.00 | gmcinnes | um. actually I'm not sure about the power supplies, but it's not *that* mission critical, and there will be a staging server we can promote into production quickly and easily. |
| 21:35.04 | lirakis | hello |
| 21:35.24 | *** join/#asterisk saftsack (n=oliver@pD9E0774D.dip.t-dialin.net) |
| 21:35.26 | JT_ | unless you're running a cluster, you should *always* go for redundant power supplies :) |
| 21:35.27 | lirakis | has anyone heard of "Sanford Duitz" ..? |
| 21:36.18 | JT_ | especially at initial system ordering time |
| 21:36.36 | lirakis | ive read that he is an "American VoIP Pioneer" |
| 21:36.38 | gmcinnes | I guess. I wish they'd spend less on the machine and more on the power supply :) |
| 21:36.47 | lirakis | .. anyone know anything about him? |
| 21:36.47 | JT_ | had a poweredge 2600 power supply die last year |
| 21:36.53 | JT_ | wasn't replaced until the next day |
| 21:36.54 | lirakis | join #voip |
| 21:36.57 | lirakis | oops |
| 21:36.58 | lirakis | lol |
| 21:37.04 | gmcinnes | yeah, I think that has happened to me too. |
| 21:37.14 | gmcinnes | I can lose a day of production without too much hassle though. |
| 21:37.19 | gmcinnes | only a day tho. |
| 21:37.29 | gmcinnes | brb |
| 21:37.33 | JT_ | i ordered a second power supply from the parts department |
| 21:37.41 | JT_ | took at least 1.5months to arrive! |
| 21:37.49 | JT_ | worldwide shortage my arse |
| 21:39.30 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 21:39.54 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 21:40.08 | *** join/#asterisk Charles[NS] (i=neouf@81.91.66.232) |
| 21:42.04 | Cyon | So, it's called call screening...any good web site references for it? |
| 21:42.14 | *** join/#asterisk atlantia (n=user@64.253.25.242) |
| 21:42.33 | *** part/#asterisk atlantia (n=user@64.253.25.242) |
| 21:51.21 | Bobthehunter | eb 13 11:31:01 NOTICE[10188] chan_iax2.c: Rejected connect attempt from 69.159.245.243, requested/capability 0x2/0xf802 incompatible with our capability 0xf904. |
| 21:51.27 | Bobthehunter | so i guess one is trying to hack us lol |
| 21:51.34 | Bobthehunter | time for retribution |
| 21:52.05 | JT_ | maybe it is, or maybe something is just configured wrong |
| 21:52.18 | Bobthehunter | nope |
| 21:52.24 | JT_ | retribution, sounds illegal |
| 21:52.30 | JT_ | it's just a connection attempt |
| 21:52.31 | JT_ | who cares |
| 21:52.33 | Bobthehunter | reporting should of been the word |
| 21:52.39 | JT_ | heh |
| 21:52.44 | JT_ | is it repeated? |
| 21:52.50 | Bobthehunter | gettign someone scanning from winsip device trying clls to 00441727844387, |
| 21:52.57 | Bobthehunter | SIP/5060-08476498" |
| 21:53.04 | JT_ | and iax? |
| 21:53.35 | Bobthehunter | "WinSIP 28" <10027> |
| 21:53.42 | Bobthehunter | yeah wel we got a few here ;) |
| 21:54.07 | Bobthehunter | SIP/5060-08436408 |
| 21:55.17 | *** join/#asterisk tg_ (n=tg@cube0.sch.bme.hu) |
| 21:55.26 | Bobthehunter | somehow hes spoofing src addy |
| 21:58.00 | scarfy | ok i think i know now WHAT it means.. took me long enough on my own.. error 406 not accebtable comes when the other server somehow gets my local ip instead of an internet ip. But how do i configure asterisks to do it correctly? |
| 21:59.02 | JT | it's asterisk, not asterisks |
| 21:59.06 | JT | is nat=yes |
| 22:00.15 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 22:00.22 | scarfy | JT i tried that option didn't change much |
| 22:01.22 | scarfy | ah and the asterisks was a typo sry |
| 22:01.42 | *** join/#asterisk canapa (i=canapa@83-64-148-102.wolfsberg.xdsl-line.inode.at) |
| 22:01.59 | scarfy | JT, and it is nat=yes right now yes |
| 22:02.07 | Cyon | Ok, dial with option p; now I have to get the deletion of the caller name working |
| 22:02.08 | JT | are you sure that's the problem? |
| 22:02.11 | JT | tried an iax2 debug? |
| 22:03.16 | scarfy | i don't know what an iax2 debug is.. its the first time i ever set up asterisk today |
| 22:03.34 | JT | it switches on debugging for iax |
| 22:04.50 | scarfy | iax2 debug is an option in sip.conf? or what? sry i really have no clue what it is.. a programm? |
| 22:05.15 | Strom_C | ~book |
| 22:05.25 | jbot | i heard book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 22:05.25 | JT | it's a command |
| 22:05.47 | blitz[training] | iax2 != sip |
| 22:05.48 | scarfy | ah in the asterisk console? |
| 22:05.53 | blitz[training] | iax2 is a protocol, sip is a different protocol |
| 22:06.10 | *** join/#asterisk atlantia (n=scott@64.253.25.242) |
| 22:06.41 | scarfy | ah but i'm using sip... for ekiga.net |
| 22:06.52 | JT | indeed, for some reason i thought he was using iax |
| 22:06.53 | atlantia | i read in the book that asterisk picks automagically the best format when you specify a recording in the extension list. I have a main_menu.mp3 in there, but it ignores it, is mp3 not an option? |
| 22:07.02 | JT | the error was ridiculously far back in scrollback |
| 22:07.09 | atlantia | tested with main_menu.gsm and it works |
| 22:07.25 | JT | atlantia: since when was mp3 ever an option for prompts? |
| 22:07.29 | JT | it's an option for MoH |
| 22:07.35 | atlantia | JT, heh that answers my q |
| 22:07.41 | scarfy | JT, yes i had a long time finding anything usefull out on my own since nobody would give me a hint back then |
| 22:07.48 | atlantia | is there a program to convert mp3 to gsm |
| 22:07.51 | *** join/#asterisk xx8xx (n=pride_32@wsip-66-210-250-2.ph.ph.cox.net) |
| 22:07.53 | JT | scarfy: sip debug then |
| 22:08.00 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 22:08.04 | scarfy | JT ok thx |
| 22:08.05 | xx8xx | hey guys.. In asteriskNOW how do I simulate an incoming call ? |
| 22:08.15 | JT | atlantia: are your channels running the gsm codec? |
| 22:08.31 | atlantia | JT yes |
| 22:09.00 | Dr-Linux|home | atlantia: us wavpad |
| 22:09.01 | JT | i guess if all your channels are running gsm, then that'd be the best format to store them in |
| 22:09.05 | JT | i think sox can do it |
| 22:09.15 | perd | xx8xx make another context, add your sip client to that context, include your external context within that context |
| 22:09.19 | JT | sox can definately do gsm to wav |
| 22:09.31 | JT | and sox is the favourite for tinkering with asterisk prompts |
| 22:13.34 | *** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net) |
| 22:14.01 | perd | anyone know if there's a way to change what the 4 LCD panel buttons do on 7912 phones? |
| 22:14.07 | *** part/#asterisk hyphen (n=hyphen@c-69-136-84-149.hsd1.pa.comcast.net) |
| 22:14.11 | perd | i would like to add xfer/dnd/park options |
| 22:14.19 | Qwell[] | perd: hack the code |
| 22:14.24 | Qwell[] | or do you mean sip? |
| 22:14.33 | perd | sip |
| 22:14.46 | perd | i was hoping there was a template file |
| 22:14.48 | scarfy | JT, ah yeah now i see in the sip debug it says Contact: <sip:101@192.168.0.4> wich IS my local ip.. i don't know what contact stands for but i don't think he needs to know my local ip |
| 22:14.49 | Qwell[] | then no |
| 22:14.51 | perd | i didnt see any in the docs i read though |
| 22:14.55 | perd | man that's lame |
| 22:15.55 | ChicagoBud | can anyone help with a sangoma A200D install? |
| 22:16.59 | scarfy | JT? |
| 22:17.28 | JT | scarfy: is this to a sip provider? |
| 22:17.46 | ChicagoBud | I have two fxo modules but the generated zaptel.conf seems to have detected fxs modules |
| 22:17.53 | scarfy | JT, yes. its to ekiga.net an sip provider |
| 22:18.20 | JT | are you using the right username and password? |
| 22:18.33 | JT | maybe pastebin.ca your sip.conf, after blanking the secret |
| 22:18.47 | scarfy | JT, well.. yes but wouldn't it then give another error if that would be the case? |
| 22:18.48 | wunderkin | ChicagoBud, fxs uses fxo signalling and fxo uses fxs so yes |
| 22:19.00 | JT | scarfy: maybe |
| 22:19.16 | ChicagoBud | wunderkin, OK |
| 22:19.55 | ChicagoBud | wunderkin, now in asterisk, should I be able to do a 'zap show channels' or something like that? |
| 22:21.03 | wunderkin | as long as you recompiled asterisk and loaded the modules it should work |
| 22:22.48 | ChicagoBud | wunderkin, maybe I didn't enable it when building * |
| 22:23.47 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 22:23.54 | scarfy | JT, http://pastebin.ca/354321 shoudl i also pastebin the stufff sip set debug gives out? |
| 22:24.00 | ChicagoBud | wunderkin, Humm. make menuselect shows chan_zap as checked |
| 22:24.54 | wunderkin | i'm not familiar with sangoma, fxo, or fxs installs so i can't really help any further |
| 22:27.30 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 22:28.37 | ChicagoBud | wunderkin, thanks. I stoped asterisk and did an asterisk -cvvvv and I don't see chan_zap getting loaded. I'll have to try to figure out why not. |
| 22:29.21 | mafkees | gheh |
| 22:29.33 | mafkees | today my boss asked me to review all the moh classes |
| 22:29.48 | mafkees | some customers dont like it to listen to Cradle of Filth while waiting |
| 22:30.00 | JT | whingers |
| 22:30.13 | scarfy | jt? |
| 22:31.02 | *** join/#asterisk frogzoo (n=frogzoo@202.155.165.25) |
| 22:31.44 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-c6bc09938db2324f) |
| 22:31.49 | anonymouz666 | how much it's a fair salary/month in NY... a high technician with asterisk/ser/networking/unix skillz ? |
| 22:32.14 | Qwell[] | anonymouz666: fair for who? |
| 22:32.14 | *** join/#asterisk Opperior (n=chatzill@75.69.247.108) |
| 22:32.24 | JT | scarfy: i guess you should? |
| 22:32.57 | JT | Qwell[]: fair to someone who says "high technician" and "skillz" |
| 22:33.05 | anonymouz666 | lol |
| 22:33.06 | anonymouz666 | haha |
| 22:33.26 | anonymouz666 | take easy |
| 22:33.41 | anonymouz666 | 4000k? |
| 22:33.43 | anonymouz666 | 5000k? |
| 22:33.51 | bkruse | anonymouz666: thats what i make, in a day. |
| 22:34.00 | bkruse | i am teh awesome! |
| 22:34.00 | mafkees | wow |
| 22:34.01 | Qwell[] | bkruse: you're uberleet though |
| 22:34.04 | mafkees | ur a pimp ? |
| 22:34.09 | anonymouz666 | bkruse: you are my hero |
| 22:34.11 | *** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 22:34.13 | bkruse | Qwell: i guess so |
| 22:34.20 | bkruse | thats what the digium ladies say, at least |
| 22:34.32 | anonymouz666 | I need an idea |
| 22:34.43 | anonymouz666 | you live in america, not me. |
| 22:34.44 | JT | no seriously, anonymouz666, english skills often impact upon your pay rate |
| 22:34.52 | bkruse | it really does. |
| 22:34.57 | scarfy | JT, http://pastebin.ca/354329 |
| 22:35.00 | bkruse | Qwell: rrdtool! |
| 22:35.01 | bkruse | :P |
| 22:35.07 | *** join/#asterisk voipanywhere (n=pirch@a81-84-60-32.cpe.netcabo.pt) |
| 22:35.14 | anonymouz666 | JT: ok, how much it's fair for you to accept? |
| 22:35.16 | [[blah]asfd | is there any way to make a call jump to the front of the line in a queue, based on how long they have been in another queue? for example if someone is in queue number 1 for 15 minutes, can it go to another queue which is a back up for queue one, but put the caller in front of anyone else that is in that queue? |
| 22:36.42 | anonymouz666 | my english sucks because I miss lots of classes to drink beer while at school. :) |
| 22:36.48 | ChicagoBud | fyi, I re-ran ./configure; make -- in *. Iit looks like chan_zap was not built |
| 22:37.23 | *** join/#asterisk Jared_Leto (n=Lostprop@80-89-104-241.DSL.ycn.com) |
| 22:37.41 | JT | anonymouz666: and being a "high" technician? :) |
| 22:38.02 | diclophis-work | why would asterisk grab the called extension (phone number for a zap channel) as the username for a SIP connection? |
| 22:38.23 | [[blah]asfd | diclophis-work: show your dial command. |
| 22:38.27 | [[blah]asfd | chances are you told it to |
| 22:38.45 | diclophis-work | Dial("SIP/192.168.44.68-09a88290", "SIP/212@jawa.internal|20") |
| 22:38.52 | scarfy | JT, isn't it the problem that somehow my internal ip gets used instead of my external? |
| 22:38.54 | ChicagoBud | and that seems to have worked. Thanks wunderkin for the tip on the fxo/fxs |
| 22:39.12 | diclophis-work | where jawa.internal is a hostname |
| 22:39.18 | JT | scarfy: try setting externip= |
| 22:39.23 | JT | it might help |
| 22:39.24 | diclophis-work | and is not specified at all in sip.conf |
| 22:39.33 | [[blah]asfd | diclophis-work: is 192.168.44.68-09a88290 your sip user? |
| 22:39.41 | anonymouz666 | JT: hehe. I want just a base value. For example: someone offers to you 2000k/month to work in NY. And then you say: Native people won't accept for at least 4000k the same job. |
| 22:39.46 | diclophis-work | [[blah]asfd: i reckon so? |
| 22:39.47 | JT | scarfy: i can see the private ip there, ut it's still getting the error back to you too, hmm |
| 22:40.04 | diclophis-work | i was thinking that was the calling calling |
| 22:40.07 | diclophis-work | er calling channel |
| 22:40.14 | [[blah]asfd | no... Should be your user |
| 22:40.22 | scarfy | JT, yeah.. this a point.. wierd |
| 22:40.26 | [[blah]asfd | Dial(SIP/<user> |
| 22:40.39 | JT | scarfy: that may just be the sip server being smart |
| 22:40.54 | elriah | Woohoo! I just made my first change to asterisk! God help my users now... lol |
| 22:40.57 | [[blah]asfd | your user should exist in the sip.conf |
| 22:40.58 | elriah | chan_sip.c |
| 22:41.15 | diclophis-work | [[blah]asfd: ok here is what i am trying to setup... we have 2 boxes... one has a digium card, and is setup to redirect all calls for a certain number to the second box |
| 22:41.33 | diclophis-work | where that box picks up, and runs a special menu... then if need be, will dial an extension on the first box |
| 22:42.49 | [[blah]asfd | what you have pasted here would dial sip user 192.168.44.68-09a88290 and sip user 212@jawa.internal for 20 second before going to the next line in your dial plan |
| 22:43.18 | diclophis-work | hmm |
| 22:44.03 | scarfy | JT, wierd i set externip now it worked for a second and then timedout :-/ http://pastebin.ca/354348 |
| 22:44.13 | *** join/#asterisk ping2921 (n=marc3234@206-248-163-237.dsl.teksavvy.com) |
| 22:44.38 | JT | scarfy: 406, doesn't look like it worked.. |
| 22:44.58 | diclophis-work | [[blah]asfd: how would i setup what i described |
| 22:45.19 | scarfy | JT, yes but look the first 2 lines "Peer 'ekiga' is now Reachable" |
| 22:45.31 | JT | that has nothing to do with register |
| 22:45.35 | JT | that's just qualify |
| 22:45.38 | diclophis-work | the thing i dont understand is why its trying to used the caller-id of the caller as anything related to do with SIP authentication |
| 22:45.40 | scarfy | ah ok :( |
| 22:45.41 | scarfy | darn |
| 22:45.42 | scarfy | rofl |
| 22:45.44 | JT | qualify does no autg |
| 22:45.47 | JT | auth |
| 22:45.57 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqu6.cable.mindspring.com) |
| 22:45.58 | JT | it just is the sip equivalent of a ping |
| 22:46.02 | scarfy | JT, then its still the same problem |
| 22:46.32 | JT | indeed |
| 22:46.50 | voipanywhere | Anyone has seen dseeb_? (VERY IMPORTANT) |
| 22:47.13 | *** join/#asterisk gork (n=gork@gw.amarillo.energynet.com) |
| 22:47.22 | scarfy | JT, but if its ot the ext ip... i'm clueless |
| 22:47.37 | *** join/#asterisk angom (n=angom@red-corp-201.143.88.126.telnor.net) |
| 22:47.55 | [[blah]asfd | diclophis-work: you would want to do |
| 22:47.56 | [[blah]asfd | exten => whatever,1,Dial(first server|20) |
| 22:47.56 | [[blah]asfd | exten => whatever,2,Dial(second server|20) |
| 22:48.10 | [[blah]asfd | if i understood what you were looking for |
| 22:48.13 | JT | scarfy: sorry, i couldn't understand that |
| 22:48.21 | [[blah]asfd | check out the wiki and look up the dial command. |
| 22:48.28 | [[blah]asfd | there are a lot of helpful examples. |
| 22:48.54 | scarfy | JT, if its ot the external ip... i have no clue amymore at all on what the problem is |
| 22:49.03 | diclophis-work | [[blah]asfd: not quite... all calls for a certain number goto server 2, however all the phones are connected to server 1, so when someone wants to get to a phone connected to server 1, and they are on server 2 they have to dial back into server 1 |
| 22:49.06 | diclophis-work | damn |
| 22:49.13 | diclophis-work | maybe this is over complicating |
| 22:49.30 | gork | is there an audio converter that will allow me to encode some high quality source into the .g722 files compatible with asterisk? |
| 22:49.51 | JT | scarfy: if its ot |
| 22:50.09 | scarfy | JT, ot? |
| 22:50.21 | JT | scarfy: that's what you wrote to me, twice |
| 22:50.40 | scarfy | no i wrote 2 completly different sentences |
| 22:50.41 | diclophis-work | whats the difference between entries in sip.conf that are "register" entries, and [user] entries? |
| 22:50.58 | JT | < scarfy> JT, but if its ot the ext ip... i'm clueless |
| 22:51.02 | JT | < scarfy> JT, if its ot the external ip... i have no clue amymore at all |
| 22:51.43 | scarfy | aah wierd i would miss the same letter twice, it should be not, was a typo |
| 22:51.45 | scarfy | sry |
| 22:51.59 | JT | gork: sox is usually used to do audio conversion |
| 22:52.09 | JT | gork: since when does asterisk support wideband audio? |
| 22:52.16 | *** join/#asterisk hal2385614 (n=chatzill@86.149.55.180) |
| 22:53.01 | gork | sox does not seem to support g722.. which is a split band ADPCM... either that or it's a hella complicated commandline option :(.. 1.4 supports g722 in passthrough and trunk has a full codec |
| 22:53.35 | JT | well if it's a prompt, it needs to be 100% supported |
| 22:53.47 | Qwell[] | gork: no it doesn't |
| 22:53.56 | Qwell[] | it's still only passthrough in trunk, I'm pretty sure |
| 22:54.31 | file | nope. |
| 22:54.39 | perd | anyone have a call forwarding example |
| 22:54.40 | Qwell[] | we can transcode g722 now? |
| 22:54.45 | diclophis-work | doh, why are my SIP invitations using the caller-id as the From ?? |
| 22:54.51 | scarfy | What could cause a 406? |
| 22:54.53 | file | yessir |
| 22:55.04 | Qwell[] | hmm |
| 22:55.04 | diclophis-work | wouldnt the From: be the fromuser or username in sip.conf? |
| 22:55.05 | JT | diclophis-work: that's normal |
| 22:55.05 | Qwell[] | since when? :p |
| 22:55.10 | file | awhile. |
| 22:55.12 | JT | hmm |
| 22:55.26 | diclophis-work | JT: hmm, how do i make it not do that? |
| 22:55.26 | hal2385614 | hi all! I have cisco 79xx phones that I am using with an asterisk 1.2 box with TDM400P card. When a caller phones via a standard telephone line, there is an echo for about 5 seconds and then the line quality is great. Is there any way to get rid of this confusing behaviour? I have tried litterally 100s of different configurations to remove the issue, without any success. Can someone... |
| 22:55.27 | hal2385614 | ...please help? |
| 22:55.31 | gork | yes though everything to transcode g722 to is 8khz so really its just up/downsampling whenever there is a transcode |
| 22:55.48 | gork | the codec works fine on top of 1.4 with my polycom 650.. it sounds really really fantastic |
| 22:55.50 | perd | hal: hardware DSP :/ |
| 22:55.56 | Qwell[] | [TK]D-Fender: see? |
| 22:55.56 | diclophis-work | hal2385614: have you tweaked the echo training times? |
| 22:56.08 | Qwell[] | "it sounds really really fantastic" |
| 22:56.55 | hal2385614 | diclophis - thanks for your response. Is there any way to find out what the settings are after it has trained, and then hard code them in? |
| 22:57.20 | gork | Yes he was discouraging me a couple of weeks ago by claiming it didnt sound that much better but ... well it really does when you can get it going all the way through |
| 22:57.45 | hal2385614 | I am not sure what you mean by tweaking the training times, though - I may have, but maybe I just don't use the same terminology |
| 22:57.48 | diclophis-work | hal2385614: well from what i remember about setting up a zapata devices, there is an option for how long to do echo traning |
| 22:57.56 | [[blah]asfd | diclophis-work: not sure I follow |
| 22:57.59 | hal2385614 | could you explain |
| 22:58.12 | diclophis-work | you also might have two or more echo cancelation mechanisms happening at the same time |
| 22:58.25 | hal2385614 | really? |
| 22:58.31 | gork | the problem is that i cant find anything anywhere to actually take, say a wav file and produce an asterisk compatible .g722 .. maybe i just need to write a little app that uses the asterisk g722 codec itself to produce the file ?? |
| 22:58.52 | hal2385614 | shall I paste my zap config to pastebin? Woudl you care to look? |
| 22:58.59 | diclophis-work | yea i will scope it out |
| 22:59.15 | diclophis-work | tdm400p is a digium card right? |
| 22:59.26 | diclophis-work | i think i am using the 411s which have the hardware echo cancelation |
| 22:59.56 | hal2385614 | yes, it is digium |
| 23:00.07 | JT | 411?? |
| 23:00.18 | Qwell[] | JT: you know, the number before 412? |
| 23:00.20 | hal2385614 | I am able to confiugre echo cancellation in the zapata.conf - is this what you mean? |
| 23:00.29 | diclophis-work | yea |
| 23:00.48 | hal2385614 | the cisco phone is, of course, over sip |
| 23:00.52 | diclophis-work | try increasing the echotraining=100 or so |
| 23:00.59 | JT | Qwell[]: i think he's imagining there's some sort of hardware echo cancellation boards for the tdm400p |
| 23:01.09 | *** join/#asterisk olsen (n=diego@200.61.236.33) |
| 23:01.13 | Qwell[] | no, there is a TE411 |
| 23:01.22 | diclophis-work | yea the TE411 |
| 23:02.08 | wunderkin | P |
| 23:02.10 | JT | what is a te411? |
| 23:02.27 | Qwell[] | quad T1/E1 with the older echo can |
| 23:02.35 | JT | te411p? |
| 23:02.38 | Qwell[] | yes |
| 23:02.51 | JT | right, that's not an analogue tdm400p at allt hough :P |
| 23:02.54 | diclophis-work | yea thats what i am using |
| 23:03.10 | JT | you're either using a tdm400p or you're not, diclophis-work |
| 23:03.10 | diclophis-work | oh.. i only have experience with the TE411p |
| 23:03.29 | diclophis-work | damnit i dont know what i am using anymore |
| 23:03.35 | scarfy | does anybody have an idea what a 406 error could mean? |
| 23:03.39 | diclophis-work | i know i havnt had any echo problems |
| 23:03.39 | scarfy | sip |
| 23:03.53 | Qwell[] | scarfy: What does the rest of the error say? |
| 23:04.11 | scarfy | Got SIP response 406 "Not Acceptable" back from 86.64.162.3 |
| 23:04.20 | Qwell[] | so something you sent wasn't acceptable |
| 23:04.29 | scarfy | in what way? |
| 23:04.42 | Qwell[] | I don't know - you'll have to ask them |
| 23:04.51 | Qwell[] | What did you send? |
| 23:04.59 | JT | diclophis-work: i don't think any digium cards for analogue have hardware EC |
| 23:05.10 | Qwell[] | JT: TDM2400p does |
| 23:05.25 | JT | ah ok :) |
| 23:05.25 | scarfy | Qwell, just a registration of my account at this sip provider |
| 23:05.26 | hal2385614 | everyone - does diclophis' advice still stand for me? Does echotraining=100 wwork for the TDM400P? |
| 23:05.31 | JT | that would mean the tdm800p does? |
| 23:05.38 | Qwell[] | JT: I don't know.. |
| 23:05.51 | JT | is it a function of the modules or the main board? |
| 23:06.02 | kervel | i'm trying music on hold, and for intern lines is okay, but when i call from my cell phone, i hear a bit of the song and then noise |
| 23:06.04 | scarfy | a register => i sent wich he has a problem with |
| 23:06.04 | Qwell[] | on most (all?) of the boards, it's another module |
| 23:06.17 | file | B410P is the only board with onboard |
| 23:06.52 | kervel | i'm using format_wav as i don't have format_mp3 (political correct build) |
| 23:06.55 | elriah | On cisco phones, is the MWI light supposed to blink or just stay solid? |
| 23:07.04 | Qwell[] | elriah: What's it doing? |
| 23:07.09 | elriah | Solid. |
| 23:07.16 | Qwell[] | Then it's supposed to stay solid. |
| 23:07.40 | elriah | Qwell[]: Ok. let me rephrase, lol, is there a way to make the MWI on a Cisco phone blink instead of stay solid? |
| 23:07.51 | Qwell[] | why would you want to do that? |
| 23:09.31 | scarfy | what could there be not acceptable? i did it after a tutorial from theyre wiki in the first place ... is something with my sip.conf? |
| 23:09.48 | [TK]D-Fender | scarfy : Thats a given... |
| 23:09.57 | [TK]D-Fender | scarfy : maybe you should SHOW it to us... |
| 23:10.07 | scarfy | the tutorial? |
| 23:10.08 | [TK]D-Fender | ~pb |
| 23:10.21 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 23:10.21 | scarfy | :-/ |
| 23:10.21 | diclophis-work | ok, i have a laser focused question... i have a valid SIP user defined in sip.conf ([trunk_1] ... calls get delivered to it fine), how do I know Dial() to that "user" (in reality my SIP provider) from my dialplan ? |
| 23:10.21 | [TK]D-Fender | scarfy : no, your CONFIGS |
| 23:10.26 | diclophis-work | Dial(SIP/trunk_1) ?? |
| 23:10.32 | scarfy | i showed them already a bit back |
| 23:10.41 | scarfy | but i can post the pastbins again |
| 23:11.06 | [TK]D-Fender | diclophis-work : Poorly worded, but here's how : Dial(SIP/trunk_1/1234567890) |
| 23:12.06 | diclophis-work | [TK]D-Fender: "No such host: trunk_1" |
| 23:12.19 | BSDTech | ok what dould be the exten setting to check a exten status and direct a call |
| 23:12.26 | BSDTech | is it chanstatus |
| 23:12.26 | diclophis-work | though the definition of "trunk_1" in users.conf, has a host= |
| 23:12.29 | [TK]D-Fender | diclophis-work : then you do not have things done right in your sip.conf. Pastebin it |
| 23:12.33 | scarfy | [TK]D-Fender, thats my sip.conf http://pastebin.ca/354321 |
| 23:12.45 | [TK]D-Fender | diclophis-work : I would avoid users.conf if I were you. |
| 23:13.11 | scarfy | [TK]D-Fender, and thats the error i get when i enabled sip debug http://pastebin.ca/354348 |
| 23:13.45 | [TK]D-Fender | scarfy : pow pastebin the CLI output where you see these errors with SIP debug enabled |
| 23:14.06 | [TK]D-Fender | scarfy : And does your * server have a PUBLIC IP? |
| 23:14.28 | diclophis-work | [TK]D-Fender: here is server #2s sip.conf and users.conf http://pastebin.ca/354400 |
| 23:14.44 | scarfy | [TK]D-Fender, no, he doesn't he goes over a nat/router |
| 23:15.06 | scarfy | [TK]D-Fender, and i already posted the cli output to you |
| 23:15.17 | diclophis-work | this is server #1s sip.conf http://pastebin.ca/354402 |
| 23:15.21 | *** join/#asterisk ez` (n=supermar@c207.134.229-230.clta.globetrotter.net) |
| 23:16.00 | diclophis-work | users.conf is where asterisk 1.4's gui puts service providers.... |
| 23:16.23 | gmcinnes | guys how would I check if asterisk is listening on sip? should I expect: telnet X.X.X.X 5060 to work? |
| 23:16.38 | diclophis-work | ... it could be UDP? |
| 23:16.45 | diclophis-work | try a netstat -a | grep 5060 |
| 23:16.46 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 23:17.03 | [TK]D-Fender | scarfy : Looks like you ARE NAT'd |
| 23:17.24 | gmcinnes | diclophis-work: good idea. |
| 23:17.26 | [TK]D-Fender | scarfy : And are missing ALL of thes ettings required to make it work properly |
| 23:17.37 | gmcinnes | diclophis-work: nothing! ok. I'll go hack again. |
| 23:17.53 | [TK]D-Fender | diclophis-work : Ok, I can't support users.con.... |
| 23:17.59 | diclophis-work | gmcinnes: you could also see if the chan_sip is loaded |
| 23:18.13 | diclophis-work | [TK]D-Fender: you know.. i didnt like users.conf either |
| 23:18.18 | diclophis-work | i am over it |
| 23:18.28 | diclophis-work | so i am gonna add an register => entry to sip.conf right? |
| 23:18.44 | diclophis-work | perhaps server #1 is the problem |
| 23:18.51 | diclophis-work | whats the difference between a "friend" and a "peer" |
| 23:19.34 | [TK]D-Fender | diclophis-work : Whats this in your {trunk_1} entry? # |
| 23:19.34 | [TK]D-Fender | hassip = no |
| 23:19.38 | hal2385614 | could someone tell me what phone.conf is used for? It is not clear in the documentation |
| 23:19.49 | diclophis-work | hmm |
| 23:19.59 | [TK]D-Fender | diclophis-work : Not knowing users.conf verey well I'd still suspect that line... |
| 23:20.06 | [TK]D-Fender | hal2385614 : It is to be forgotten |
| 23:20.31 | [TK]D-Fender | hal2385614 : Was for suppotring some very specific aopen winmodems under Zaptel. |
| 23:20.36 | elriah | Is there a way, via the xml config, to modify the soft keys on a Cisco 79x1? |
| 23:20.38 | hal2385614 | what do you mean by that, fender? |
| 23:20.44 | scarfy | [TK]D-Fender, i am behind a nat with the asterisk server yes.. i said that. and all i could find thourgh google and docs is to set externip and localip... hmm i just seen that in my latest conf file i misspelled local... meaby that helps .. but are there any more settings that are requiered to make it work proberly? |
| 23:20.49 | [TK]D-Fender | hal2385614 :Support for it has since been dropped |
| 23:20.53 | JT | hal2385614: it means you don't need to worry about it |
| 23:21.11 | hal2385614 | oh, good, I hadn't configured it, so I am glad that I shouldn't have, if you know what I mean!! ;-) Thanks |
| 23:21.27 | diclophis-work | wtfreak is this: username mismatch, have <trunk_1>, digest has <s> |
| 23:21.36 | [TK]D-Fender | scarfy : I don't see ANY of those lines in your [general], and you should canreinvite=no in there as well |
| 23:21.39 | diclophis-work | nowhere have i set the username to trunk_1 |
| 23:21.41 | JT | hal2385614: surely you only configure .conf files you need, not all of the ones there? :o |
| 23:22.37 | hal2385614 | yes, that is true, JT - and this is what I had done, but I suddenly noticed echo settings in phone.conf, and thought it may have been an oversight of mine |
| 23:23.11 | JT | ah ok |
| 23:23.31 | scarfy | [TK]D-Fender, ah yeah i posted those pastebins before i added those lines.. sry i just posted again the old pastebin instead of making a new one... but i set externip and localnet since then everything else is the same and yeah i don't have canreinvite=no in global |
| 23:24.15 | scarfy | [TK]D-Fender, Uh WOW =D i think its working now :D |
| 23:24.20 | [TK]D-Fender | scarfy : if you're going to show me something, please make sure its REAL... |
| 23:24.45 | [TK]D-Fender | Ekiga didn't want to hear your private IP as the retun IP.... |
| 23:25.09 | hal2385614 | regarding what diclophis said about "two or more echo cancelation mechanisms happening at the same time", does this mean that I shouldn't have fxo and fxs interfaces both configured to cancel echo, if a call comes in on one and goes straight to the other? |
| 23:25.18 | JT | scarfy: err so setting externip DID work? |
| 23:25.24 | scarfy | [TK]D-Fender, yes sry, i thought it was still real. i fergot about the change cause it didn't change anything. won't happen again |
| 23:25.35 | JT | scarfy: are you telling me you didn't do a sip reload or asterisk reload? |
| 23:25.42 | scarfy | JT, no adding localnet worked only externip didn't do anything |
| 23:25.54 | scarfy | JT, of course i did a reload |
| 23:25.54 | JT | localnet probably did something |
| 23:26.03 | JT | externip probably did too |
| 23:26.20 | scarfy | JT, probably the combination |
| 23:26.26 | JT | i hate trying to solve other's sip problems from afar |
| 23:26.29 | [TK]D-Fender | scarfy : Not realy mad, just that you should apy attention if you execpt "help" to WORK :) |
| 23:26.38 | JT | sip just works when i set it up :) |
| 23:26.53 | [TK]D-Fender | Yes, you NEED the combination. |
| 23:27.07 | [TK]D-Fender | otherwise * won't know when to forge the header |
| 23:27.19 | scarfy | [TK]D-Fender, ah ok.. i ussualy pay attention to that.. must have slipped by |
| 23:27.41 | scarfy | [Feb 14 00:23:34] NOTICE[3556]: chan_sip.c:11961 handle_response_register: Outbound Registration: Expiry for ekiga.net is 120 sec (Scheduling reregistration in 105 s) |
| 23:27.42 | JT | i even get inbound sip calls through nat with no portforwarding, not sure if that's a fluke |
| 23:27.44 | scarfy | means its working right? |
| 23:27.48 | JT | yes |
| 23:28.08 | hal2385614 | if I have a call coming in on my standard telephone line connected to my FXO interface, and that call needs to be connected to an analogue telephone via the FXS interface, is it OK to have echo cancelling enabled on both of the interfaces? |
| 23:28.10 | scarfy | ok :D cool :D great thanks all of your for you're help :D thanks VERY much |
| 23:28.57 | JT | no probs |
| 23:29.10 | [TK]D-Fender | scarfy : Yup (registereing at least) |
| 23:29.23 | [TK]D-Fender | scarfy : I'd put decent odds on other success |
| 23:29.48 | [TK]D-Fender | scarfy : np |
| 23:30.17 | [TK]D-Fender | hal2385614 : You shouldn't need both ends. thats what "echocancelwhenbridged" is for. But you shouldn't typically |
| 23:30.39 | hal2385614 | ah, really? |
| 23:30.44 | [TK]D-Fender | hal2385614 : Esp not if you have an analog fax attached |
| 23:30.54 | [TK]D-Fender | hal2385614 : try it if you're not sure |
| 23:31.01 | scarfy | [TK]D-Fender, its all wroking now :D its great :D thanks again :D |
| 23:31.09 | [TK]D-Fender | At least nabling it on the PHONE's zaptel channel |
| 23:31.16 | [TK]D-Fender | scarfy : Good to hear |
| 23:31.22 | hal2385614 | OMG - I can't tell you the problems that I have had with my analogue fax!! ;-) |
| 23:31.43 | [TK]D-Fender | hal2385614 : You'll keep it the hell away from * if you know whats good for you.... |
| 23:32.00 | [TK]D-Fender | hal2385614 : And you needn't. Been there, done that... |
| 23:32.43 | hal2385614 | It is because I am a cheapskate - in that I only have one line for both fax and phonecalls! |
| 23:33.09 | [TK]D-Fender | hal2385614 : Get a fax switch and put it in front of * |
| 23:33.26 | hal2385614 | ooh, really? |
| 23:33.31 | hal2385614 | that sounds good |
| 23:33.45 | [TK]D-Fender | hal2385614 : No, I have a habit of lying at least 10% of the time to maintain quota... |
| 23:33.50 | hal2385614 | I haven't heard of one - where do I get one from (I am from the UK) and what kind of prices are they? |
| 23:33.54 | [TK]D-Fender | hal2385614 : NOT |
| 23:34.12 | xx8xx | in linux whats the command to get a list of all the IRQs used to check for conflict ? |
| 23:34.12 | hal2385614 | haha! |
| 23:34.17 | [TK]D-Fender | hal2385614 : Around 50-100$USD depending |
| 23:34.31 | [TK]D-Fender | xx8xx : "cat /proc/interrupts" |
| 23:35.26 | kervel | anyone knows why musiconhold over isdn just goes away when the music is silent, but speech keeps working perfectly |
| 23:35.28 | hal2385614 | can you recommend a good brand, fender? |
| 23:35.51 | [TK]D-Fender | hal2385614 : Not offhand.... haven't bought one personally, have just worked with 2 before |
| 23:36.34 | *** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 23:36.36 | hal2385614 | ok - thanks for the advice! |
| 23:36.48 | xx8xx | ok i see thse irqs: 1.3.5.8.11.12.LOC.ERR.MIS |
| 23:36.53 | xx8xx | what are the last 3 mean ? |
| 23:37.50 | [TK]D-Fender | xx8xx : Dunno. How about Pastebinning it... |
| 23:37.52 | [TK]D-Fender | ~pb |
| 23:38.06 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 23:38.06 | [TK]D-Fender | the WHOLE thing |
| 23:38.09 | xx8xx | ok then brb |
| 23:38.52 | JT | kervel: what sort of isdn setup? |
| 23:39.14 | JT | but in general it's a sign of either silence supression or a highly compressed codec |
| 23:39.21 | JT | music often sounds like silence |
| 23:39.25 | JT | or noise |
| 23:39.28 | kervel | JT: B410P: |
| 23:39.32 | JT | to a computer |
| 23:39.47 | kervel | JT when i call from a directly connected sip phone, the music is perfect |
| 23:39.54 | ruied | I have a voip phone conecting to * (* is behind a nat linux) when we established a call we can make a conversation with some audio problems, than the connection goes down due to packet loss |
| 23:40.00 | JT | maybe it's your telco |
| 23:40.01 | kervel | when i call from PSTN, music drops away (changes into silent noise) when it is silent |
| 23:40.04 | JT | is this to a landline, kervel ? |
| 23:40.10 | ruied | what could it be? QOS ? |
| 23:40.27 | kervel | JT nope, cellphone... but the same cellphone works with other pbx's musiconhold |
| 23:40.46 | JT | err cellphones have highly compressed audio codecs with silence detection |
| 23:40.51 | kervel | ahh, |
| 23:41.00 | JT | the only solution is to live with it or to increase the volume a little |
| 23:41.10 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 23:41.12 | kervel | i tried madplay -A3 already ... i'll try -A5 |
| 23:41.23 | JT | but it's quite standard when on a cellphones to hear music on hold fade in and out |
| 23:41.40 | xx8xx | ok here it is |
| 23:41.43 | xx8xx | <PROTECTED> |
| 23:41.43 | xx8xx | <PROTECTED> |
| 23:41.43 | xx8xx | <PROTECTED> |
| 23:41.43 | xx8xx | <PROTECTED> |
| 23:41.43 | xx8xx | <PROTECTED> |
| 23:41.44 | xx8xx | <PROTECTED> |
| 23:41.46 | xx8xx | <PROTECTED> |
| 23:41.48 | xx8xx | <PROTECTED> |
| 23:41.50 | xx8xx | <PROTECTED> |
| 23:41.50 | kervel | okay, then i'll live with it :) |
| 23:41.52 | xx8xx | NMI: 0 0 |
| 23:41.54 | xx8xx | LOC: 649585 649701 |
| 23:41.56 | xx8xx | ERR: 717876 |
| 23:41.58 | xx8xx | MIS: 0 |
| 23:42.00 | JT | yeah just check that it's not ridiculous and overtly loud on landline |
| 23:42.02 | JT | xx8xx: STOP |
| 23:42.02 | *** part/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 23:42.05 | JT | xx8xx: NOW |
| 23:42.17 | xx8xx | u guys said to paste the whole thing |
| 23:42.21 | JT | xx8xx: [TK]D-Fender said pastebin, not "flood 300 people on irc" |
| 23:42.22 | JT | no |
| 23:42.25 | JT | he said pastebin |
| 23:42.27 | JT | pay attention |
| 23:42.38 | JT | he even provided url links |
| 23:42.38 | xx8xx | what is pastebin ? |
| 23:42.42 | JT | ~pb |
| 23:42.43 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 23:43.10 | xx8xx | aha sorry my bad |
| 23:43.11 | blitzrage | JT: you realize that once a paste has happened, there is nothing the person can do right? :) |
| 23:43.33 | JT | blitzrage: closing the client |
| 23:44.04 | kervel | luckily he didn't have too many irq's :) |
| 23:44.15 | JT | most irc clients throttle pastes so they don't get killed for Excess Flood |
| 23:44.17 | *** join/#asterisk qdk (n=qdk@90.184.3.249) |
| 23:44.36 | xx8xx | so what are the last 4 irqs ? |
| 23:44.41 | xx8xx | how come they are not numbers ? |
| 23:44.49 | ruied | <PROTECTED> |
| 23:44.49 | ruied | [Feb 13 23:38:18] WARNING[16998]: chan_sip.c:1881 retrans_pkt: Maximum retries exceeded on transmission 6516208935279c23@192.168.1.1 for seqno 36878 (Critical Response) |
| 23:44.54 | JT | they aren't irqs i believe |
| 23:45.02 | JT | either way, not really relevant to the problem at hand |
| 23:45.15 | ruied | what could be my problem? Quality of service.... |
| 23:45.40 | JT | ruied: it could be, is your Internet connection normally dodgy? |
| 23:46.08 | blitzrage | ruied: is that phone actually on your local network? Because it's sending the call to a LAN IP |
| 23:46.53 | blitzrage | if it is outside, you need nat=yes to tell it to send the replies to the IP address the call comes from, and not that in the Contact header |
| 23:47.26 | ruied | I have several phones in my intranet working well, but a sip phone connected from the Internet to * doesn't work well.... |
| 23:47.32 | blitzrage | see above |
| 23:47.45 | [TK]D-Fender | xx8xx : Well I don't see any Digium cards in there and they are the only ones that are picky about IRPS really... |
| 23:48.00 | [TK]D-Fender | Oh, and probably the elcheapo's like Rhino. |
| 23:48.57 | ruied | blitzrage, where shall I define that? in [mario's] sip.conf |
| 23:49.00 | ruied | ? |
| 23:49.19 | blitzrage | yes -- and I hope it is [mario] and not [mario's] |
| 23:49.27 | ruied | yes :) |
| 23:49.32 | kervel | ruied you could use IAX also instead of SIP ... it works much better with NAT |
| 23:49.33 | hal2385614 | ok, I am back to trying to prevent the echo between the FXO port (incoming telephone line) and the cisco 79xx sip phones. Does anyone have any suggestions about how to remove the first 10-15 seconds of echo (before things settle down) |
| 23:49.45 | blitzrage | kervel: SIP works fine behind NAT |
| 23:49.58 | kervel | blitzrage yeah, if you have a fixed public ip |
| 23:50.09 | *** part/#asterisk xx8xx (n=pride_32@wsip-66-210-250-2.ph.ph.cox.net) |
| 23:50.10 | blitzrage | kervel: totally untrue -- I have 6 phones behind NAT here |
| 23:50.12 | *** join/#asterisk coppice (n=chatzill@118.202.17.210.dyn.pacific.net.hk) |
| 23:50.23 | kervel | ahh ... then i just had bad luck :) |
| 23:50.46 | kervel | blitzrage and NAT = cheap adsl router ? or linux box with sip connection tracking |
| 23:50.49 | blitzrage | just need to know how to configure is all |
| 23:50.59 | blitzrage | Linksys router |
| 23:51.03 | kervel | ahh |
| 23:51.05 | blitzrage | no port forwarding required |
| 23:51.06 | kervel | i stand corrected |
| 23:52.09 | osas | hi all :) I have a SIP related question: Is it possible to relax the SIP checking on the asterisk side? I want asterisk to accept an INVITE with credetials with the same Cseq number as the previous one that was challanged. |
| 23:52.52 | blitzrage | osas: not sure you can do specifically that, but you can relax with insecure=[yes|very] |
| 23:53.06 | osas | by default, the second INVITE is ignored |
| 23:53.18 | osas | tryed that ... :( |
| 23:53.27 | blitzrage | see chan_sip.c :) |
| 23:53.32 | osas | k |
| 23:53.34 | osas | ;-) |
| 23:55.43 | elriah | Anyone else in here using the Cisco 7941G? I'm trying to figure out how to get the horizontal lines off of the LCD (SIP 8.0.4SR2) |
| 23:55.57 | diclophis-work | doh, anyone know why the asterisk gui would have options disabled? |
| 23:56.27 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
