| 00:02.19 | *** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au) |
| 00:04.04 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
| 00:07.53 | *** join/#asterisk loompek (n=NoName@noname.rula.net) |
| 00:07.55 | loompek | morning |
| 00:09.03 | loompek | i've got a little ol' question.. how come asterisk doesn't register at some other sip server even though i have all the necesary stuff in sip.conf (register => ... and [server]...) |
| 00:10.13 | loompek | i'm looking the tcpdump output for port 5060... asterisk just sends OPTIONS to all of the serververs in sip.conf with register command and qualify !=no |
| 00:11.25 | JT | ~pb |
| 00:11.26 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 00:11.33 | JT | pastebin sip.conf minus passwords |
| 00:12.51 | *** join/#asterisk __freedom__lover (n=eduardo@clipper.provale.com.br) |
| 00:13.32 | litage | is it possible to determine whether ``fxotune -s'' actually worked, or what changes it made? |
| 00:13.36 | *** join/#asterisk PepOSX (n=pepOSX@190.72.153.233) |
| 00:16.32 | bungalow | hi: im trying to get asterisk to reinvite after dial, to bridge two sip channels... but the media stream is continuing to go through asterisk... |
| 00:16.39 | bungalow | any idea on how to debug this |
| 00:16.40 | bungalow | ? |
| 00:18.03 | fujin_ | are both sides of the conversation set up with canreinvite=yes? |
| 00:18.54 | *** join/#asterisk SwK (n=SwK@user-24-214-55-149.knology.net) |
| 00:20.24 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-11ea7502cc51e34b) |
| 00:20.39 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 00:21.39 | *** join/#asterisk z001 (n=mooserfu@i-195-137-39-237.freedom2surf.net) |
| 00:21.43 | bungalow | fujin_: yes |
| 00:22.57 | fujin_ | is there any reason for the audio to still be routing through asterisk? meetme.. monitor? |
| 00:23.41 | bungalow | I was running mixmonitor before the dial, but I've turned that off... so now nothing... the call comes in, processed by agi, and then dial (w/ g option) |
| 00:24.15 | bungalow | with sip debug I see some INVITES, but not sure what I should be looking for exactly |
| 00:27.01 | loompek | JT hope i didn't miss anything |
| 00:27.01 | loompek | http://rula.net/121 |
| 00:27.37 | ManxPower | bungalow: HOW do you know the media stream is still going thru Asterisk?? |
| 00:28.22 | bungalow | ManxPower: I see rtp traffic going to and from asterisk with rtp debug... is this a valid way of checking? |
| 00:29.22 | bungalow | ManxPower: also, earlier when I had mixmonitor on before the dial I also was recording audio (not sure if this causes asterisk to not re-invite, though) |
| 00:29.52 | ManxPower | yes, but rtp debug is a new feature and not everyone knows about it. |
| 00:29.59 | loompek | JT any ideas? |
| 00:30.11 | ManxPower | bungalow: ANYTHING that causes asterisk to listen to the audio will make it not reinvite. |
| 00:30.27 | ManxPower | bungalow: I assume you are forcing both legs of the call to use the same codec? |
| 00:30.47 | fujin_ | that'll do it |
| 00:30.49 | bungalow | ManxPower: yes, both ulaw... but canreinvite=yes. |
| 00:31.00 | bungalow | ManxPower: rtp debug on is included in that ANYTHING? |
| 00:31.14 | *** join/#asterisk jsidhu2 (n=atomik@66.206.163.184) |
| 00:31.23 | [TK]D-Fender | bungalow, No. |
| 00:31.59 | [TK]D-Fender | bungalow, any of these Dial options : "tTwWr", Use of "Monitor", etc. |
| 00:32.17 | jsidhu2 | aight, i need some help. I have a sip trunk from voipvoip, i can make outgoing calls, but inbound route isnt working.. i create a new any DID/any CID route to goto an IVR, but it doesnt do anything, just hangs up.. anyone help? |
| 00:32.17 | [TK]D-Fender | bungalow, Basically antyhing you tell * to do that requires it to snoop in. |
| 00:32.51 | [TK]D-Fender | jsidhu2, enable SIP debug. Place another call to test. PASTEBIN the *entire* attempt. |
| 00:32.52 | [TK]D-Fender | ~pb |
| 00:32.52 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 00:32.53 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 00:33.02 | bungalow | [TK]D-Fender: I should be ok then.... the only dial option I have set is g |
| 00:33.21 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 00:33.29 | [TK]D-Fender | bungalow, that is non-invasive. Another thing that precludes most reinvites : NAT <- |
| 00:34.29 | bungalow | I have nat=no |
| 00:35.34 | loompek | JT didya check the pastebin? |
| 00:35.48 | bungalow | asterisk is reporting 'native bridging <channelid1> with <channelid2>' |
| 00:35.56 | bungalow | is that something I should be seeing? |
| 00:36.16 | [TK]D-Fender | bungalow, that can be fine |
| 00:37.05 | z001 | hi; I'm looking to make a Queue that will ring phones (A+B+C) with 'ringall', and at the same time ring mobiles (M+N) with 'rrmemory', so the first ring will be (A,B,C,M), the second (A,B,C,N)... |
| 00:37.08 | [TK]D-Fender | bungalow, a native bridge is where TRIES to let the channels connect directly. It does not imply that it actually succeeding though. I presum that'd show up on RTP debug |
| 00:37.09 | z001 | This doesn't seem possible with a single queue... Can I put a call in multiple queues, or put some kind of combined/alias extension in a queue? |
| 00:37.50 | jsidhu2 | [TK]D-Fender: http://pastebin.com/d77ae3d7f |
| 00:37.51 | [TK]D-Fender | z001, You can do that. |
| 00:38.47 | [TK]D-Fender | z001, What you need to do for "M+N" though is dial a LOCAL CHANNEL in which you'll use a "toggle" stored value to select which to ring. Check which, change the flag, dial the guy. |
| 00:38.47 | loompek | maybe anybody else? |
| 00:38.48 | bungalow | [Tk]D-Fender: so is it an indication that it's trying the reinvite? what sort of INVITE sequence should I expect? I assume once the call is connected it performs a series of SIP invites and the RTP stream would vanish from the server... |
| 00:39.17 | [TK]D-Fender | bungalow, I can't tell you any more on the final detail unfortunately. That is the limit of my experience |
| 00:39.42 | [TK]D-Fender | bungalow, I presume you could do a port dump against those IP's to config RTP is going direct |
| 00:39.55 | [TK]D-Fender | confirm* |
| 00:40.23 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 00:40.27 | *** join/#asterisk skymeyer (n=skymeyer@bxlsrvit03.itconnect.be) |
| 00:40.41 | bungalow | [TK]D-Fender: rtp debug should do the same, no? |
| 00:40.47 | bungalow | show the same... |
| 00:41.02 | [TK]D-Fender | bungalow, Possibly, I've never actually used it myself, though I have heard of it |
| 00:41.10 | z001 | [TK]D-Fender: so my queue members will be SIP/A, SIP/B, SIP/C and {something}/123 where 123 is an extension in the dialplan with a global variable and some toggling logic? |
| 00:41.42 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
| 00:41.45 | [TK]D-Fender | z001, "Local/123@context/n" (yes, keep the /n) |
| 00:42.06 | [TK]D-Fender | z001, Now there is a CATCH, and a big one.... |
| 00:42.13 | bungalow | [TK]D-Fender: ok thanks |
| 00:42.17 | bungalow | ManxPower: still there? |
| 00:42.20 | [TK]D-Fender | z001, What kind of interface are you going to call those Cell's on? |
| 00:42.27 | z001 | [TK]D-Fender: thanks - I was looking for an example like that in the queues.conf.sample and the voip-info wiki, but they only cover ZAP/SIP and Agent lines. |
| 00:42.42 | z001 | [TK]D-Fender: via an outgoing SIP trunk |
| 00:43.00 | [TK]D-Fender | z001, remember those samples were showing you how to use a give channel, and LOCAL is a channel type too... |
| 00:43.31 | [TK]D-Fender | z001, Ok, be warned that if your ITSP considers the call "answered" the moment you PLACE it your entire plan goes out the window |
| 00:44.02 | [TK]D-Fender | z001, and the other thing : If a cell has voicemail and is busy, etc, VM could pick up iNSTANTLY thus answering the call. Not a good thing. |
| 00:44.23 | z001 | ah, it might just connect the caller to the first mobile while it's ringing and that's it? That's quite a catch |
| 00:44.25 | [TK]D-Fender | z001, I typically advise against using Cell's as queue members |
| 00:44.34 | [TK]D-Fender | z001, it is indeed. |
| 00:44.49 | z001 | but what alternative is there? |
| 00:44.58 | z001 | handling it entirely in dialplan logic? |
| 00:46.41 | jsidhu2 | aight, i need some help. I have a sip trunk from voipvoip, i can make outgoing calls, but inbound route isnt working.. i create a new any DID/any CID route to goto an IVR, but it doesnt do anything, just hangs up.. anyone help? SIP DEBUG: http://pastebin.com/d4f6d2b78 (The inbound route is set to forward all calls to a queue with an announcement).. |
| 00:47.15 | litage | when calling using an FXO port, i hear echo but the other person doesn't. when i hook up an analog phone to the same POTS line, there's a little bit of echo. this leads me to believe that there's echo being generated somewhere in the building's wiring, or between the building and the exchange. what would you recommend to reduce echo? |
| 00:48.01 | [TK]D-Fender | z001, No, the catch is the "answering" possibilities that come up in trying to call a cell. This could happen automatically because of your ITSP, or becase the cell is out of range and VM's, etc. |
| 00:48.31 | [TK]D-Fender | z001, to this means you had better have control and confidence in how you call that you know you will actually get to cyctle around. |
| 00:48.59 | *** part/#asterisk atomicd (n=AtomicDa@74-206-0-80.static-ip.m.telepacific.net) |
| 00:50.17 | *** join/#asterisk CrazyTux (n=CrazyTux@216-110-94-231.static.twtelecom.net) |
| 00:50.42 | CrazyTux | Hey guys, I'm working on voicemail, how much configuration can I do as far as email templates go? Can I pipe information from external source for example into the emails that are sent out as well? |
| 00:50.50 | z001 | [TK]D-Fender: ah. Well, this is for a night service option - a ring-all queue, but after hours ring-all and some mobiles. So there isn't really an option to not call mobiles. I'll have to experiment, I guess. |
| 00:51.03 | z001 | [TK]D-Fender: Thanks for the help |
| 00:51.44 | [TK]D-Fender | jsidhu2, The call is being answered and the dialplan is processing I see it calling apps and reaching the end of that context. This is FreePBX and you are at the end of the help you should expect in this channel. FreePBX is *not* supported here. |
| 00:52.26 | [TK]D-Fender | z001, What I MIGHT suggest for this : Queue the inter ones and only let them rotae around a few times, then maybe TIMEOUT to *dump* them to the cells.... |
| 00:53.10 | [TK]D-Fender | z001, All jsut ideas. As long as you understand the cirmstances you are dealing with you'll know which was best servers your employees and your callers |
| 00:56.58 | z001 | I'll get them to run VPNs and SIP clients over GPRS on their phones and pretend they're internal then. ;-) |
| 00:57.08 | z001 | ugh |
| 01:00.40 | [TK]D-Fender | z001, Go test how your ITSP reacts, etc, and make sure not to ring them too long so that they fall to VM. I did a bit of this just for basic OOO forwarding here |
| 01:01.05 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 01:02.26 | z001 | it's a bit late now (I can't IRC from work), but I will tomorrow |
| 01:04.39 | JerJer | z001: setup a proxy at home :D |
| 01:05.38 | z001 | oh technically I could, but politically I can't. |
| 01:05.41 | *** join/#asterisk shido6 (n=shido6@74-130-56-203.dhcp.insightbb.com) |
| 01:05.42 | fujin_ | ssh tunnel |
| 01:06.47 | z001 | on another note, if I want to query an SQL database for caller ID, that's going to have to be an AGI script, isn't it? |
| 01:07.01 | Qwell | z001: func_odbc |
| 01:07.21 | Qwell | func_odbc is pretty much one of the most awesome dialplan functions ever |
| 01:07.23 | z001 | ah... I have caller ID going to an odbc database, and was wondering if I could piggy-back on that |
| 01:07.42 | z001 | I thought I'd seen odbc related functions somewhere, but haven't spotted it since wanting to find it |
| 01:08.02 | rob0 | fujin_: an ssh tunnel can't do VoIP, because it's TCP, and voice protocols are UDP. |
| 01:08.25 | fujin_ | that's a pretty stupid assumption |
| 01:08.29 | fujin_ | but yeah, sure, if you say |
| 01:09.14 | rob0 | Stupid assumption? |
| 01:09.28 | rob0 | Maybe you can explain it to me then. |
| 01:09.38 | fujin_ | my comment was referring to him not being able to not IRC from work |
| 01:09.43 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 01:10.06 | rob0 | oh ok :) |
| 01:10.08 | fujin_ | you do realise you can tunnel both udp and tcp over a SSH tunnel |
| 01:10.16 | fujin_ | using netcat to redirect the udp |
| 01:10.20 | rob0 | no I did not / do not |
| 01:10.20 | fujin_ | I've done it, works fine. |
| 01:10.22 | litage | when changing echo cancelers in /usr/src/zaptel/zconfig.h , is it necessary to reboot the box, or just unload then load all zaptel-related modules? |
| 01:10.26 | rob0 | hmmm |
| 01:10.33 | Qwell | litage: just unload the modules |
| 01:10.38 | fujin_ | perhaps even later revisions of openssh allow udp forwarding through ssh |
| 01:10.44 | rob0 | I'd still think openvpn would be better and easier. |
| 01:10.56 | z001 | pingtunnel is one of the neater tunnels I've heard of |
| 01:11.17 | fujin_ | dnstunnel is awesome too :) |
| 01:11.24 | fujin_ | slow, though; |
| 01:11.43 | fujin_ | rob0: openvpn is an abstraction layer which is usually a pain in the ass, I'd generally go with PPTP vs. openvpn. |
| 01:11.55 | z001 | ha, nice |
| 01:11.56 | rob0 | yikes, not me. BTDT. |
| 01:11.56 | litage | Qwell: thanks |
| 01:12.20 | rob0 | <== retired pptpd admin |
| 01:12.21 | Qwell | litage: while you're playing with echo cans, try the jpah "echo can" |
| 01:12.24 | Qwell | it's pretty funny :D |
| 01:12.35 | litage | Qwell: yeah, i've heard it's a bit strange |
| 01:12.45 | Qwell | (I wrote it because I actually needed something like that to test something) |
| 01:13.15 | Qwell | it drops 2 out of every 3 frames of audio, heh |
| 01:13.22 | litage | after reloading the zap-related modules, dmesg says "Zaptel Echo Canceller: MG2". however, MG2 is commented in zconfig.h , and KB1 is uncommented |
| 01:13.26 | Qwell | it sounds *horrible* |
| 01:13.37 | Qwell | litage: is this zaptel 1.4.5? |
| 01:13.39 | Qwell | if so, use 1.4.5.1 |
| 01:13.47 | litage | Qwell: this is zap 1.2.20.1 |
| 01:14.00 | Qwell | lemme check something... |
| 01:16.07 | JT | fujin_: can you get ssh to run over udp? |
| 01:16.21 | Ryushin | Okay, I'm at a bit of a loss here. Do polycom phones by default register themselves with the server after a reboot, or only when they make their first call? |
| 01:16.29 | litage | JT: not that i know of.. |
| 01:16.39 | Qwell | grr |
| 01:17.10 | litage | Qwell: ? |
| 01:17.36 | JT | litage: i didn't think so either, i wonder if fujin_ knows otherwise |
| 01:18.20 | Qwell | litage: get svn branch 1.2 |
| 01:18.34 | Qwell | I thought those changes got released... they never did though |
| 01:18.53 | litage | Qwell: which revision? |
| 01:18.57 | Qwell | latest |
| 01:19.50 | litage | Qwell: what are the differences between MG2 and KB1? |
| 01:19.55 | Qwell | no idea |
| 01:20.14 | Qwell | what hardware do you have? |
| 01:20.49 | litage | Qwell: TDM400P |
| 01:20.52 | Qwell | how old? |
| 01:20.59 | litage | Qwell: couple of weeks |
| 01:21.08 | Qwell | call Digium sales tomorrow, tell them you want some HPEC licenses. :) |
| 01:21.23 | Qwell | say you bought a card a few weeks ago, and that MG2 isn't cutting it |
| 01:21.32 | litage | Qwell: how much are HPEC licences? |
| 01:21.47 | Qwell | For Digium customers with certain cards, $0 :) |
| 01:21.58 | Qwell | otherwise it's $10 per port I think |
| 01:22.23 | Qwell | it's *far* better than any of the open source ones in zaptel |
| 01:22.39 | bungalow | hi...trying to do an external rtp bridge via sip reinvite -- I see that asterisk attempts the re-invite via notice in the X-asterisk-info header, and it appears the INVITE attempt is properly acked, but the RTP stream continues to go through my server. Any idea how to debug this? |
| 01:23.07 | Qwell | but call them up tomorrow, they'll hook you up |
| 01:23.28 | litage | Qwell: should i get one HPEC license for each FXO port? |
| 01:23.28 | Qwell | I'm not sure what you'll need - maybe just the cards serial number |
| 01:23.32 | Qwell | yep |
| 01:23.46 | Qwell | shouldn't cost you anything |
| 01:23.56 | litage | Qwell: so should i bother with the zaptel svn trunk? |
| 01:24.04 | Qwell | you still should, yes |
| 01:24.11 | Qwell | the same bug is going to affect HPEC |
| 01:24.43 | litage | ah gotcha |
| 01:25.27 | Qwell | svn branch 1.2 that is, not trunk |
| 01:25.57 | litage | Qwell: ie?: svn checkout http://svn.digium.com/svn/zaptel/branches/1.2 zaptel-1.2 |
| 01:26.00 | Qwell | yep |
| 01:26.05 | litage | awesome possum |
| 01:26.52 | rob0 | hmmm, this definitely sounds like an Alabammy channel. :) |
| 01:27.03 | Qwell | rob0: howso? O.o |
| 01:27.14 | rob0 | 01:26 < litage> awesome possum |
| 01:27.20 | litage | Qwell: btw, if i reduce the rxgain in zapata.conf to -20.0 , most echo is gone, but i can barely hear the other person. i'm assuming the echo mostly disppears simply because echo's produced by a combo of delay and loudness, and the loudness makes the echo inaudible, but doesn't eliminate the delay? |
| 01:27.32 | litage | rob0: hah, i'm in .au . nowhere near .us |
| 01:27.44 | Qwell | litage: yeah, if the audio isn't received on the far end, it can't echo back ;) |
| 01:27.47 | rob0 | Aha! Even MORE marsupials down under! |
| 01:28.06 | litage | rob0: yup, we've got a few. we even have the world's only monotremes! |
| 01:30.22 | *** join/#asterisk defswork (n=andy@83.105.96.154) |
| 01:30.39 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 01:34.56 | litage | Qwell: this zaptel-1.2 branch that i'm exporting...is it considered 1.2.20.1 r3055, or just r3055, or ...? |
| 01:44.03 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 01:49.50 | litage | Qwell: alive? |
| 01:52.08 | russellb | i killed him |
| 01:52.10 | russellb | sorry |
| 01:53.34 | [TK]D-Fender | russellb, Don't just sit there! Go clean up the mess! |
| 01:54.02 | russellb | i pay people to handle that part for me. |
| 01:54.17 | [TK]D-Fender | russellb, outsourcing is entirely acceptable |
| 01:54.26 | russellb | thanks for the approval :) |
| 01:55.29 | litage | hahah |
| 01:55.37 | litage | cya guys! thanks for your help, Qwell |
| 01:56.11 | Ryushin | I'm trying to work on some polycom phones remotely. 430's and a 601. When I reboot them, they don't seem to register themselves with the sip server automatically after they reboot. Am I missing something? |
| 01:56.38 | Ryushin | If they initiate a call, then they register themselves, but not before then. |
| 02:00.19 | [TK]D-Fender | Ryushin, without seeing your configs we can't know |
| 02:01.18 | Ryushin | Which configs do you need to see? The sip.cfg file? |
| 02:04.17 | [TK]D-Fender | Ryushin, Everything applicable, plus logs, and CLI SIP debug on reboot |
| 02:07.36 | Ryushin | http://www.pastebin.ca/705368 This is the app log from the phone. |
| 02:08.45 | Ryushin | phone boot log: http://www.pastebin.ca/705371 |
| 02:11.04 | [TK]D-Fender | erver 'voipdenver.hq.xpulseusa.com' said 'x7072/0004f214dc21-phone.cfg' is not present |
| 02:11.17 | [TK]D-Fender | uBLFCompressed: File /ffs0/local/0004f214dc21-phone_cfg.zzz does not exist or is empty |
| 02:11.44 | *** join/#asterisk Buhntz (i=Boones@port-212-202-170-97.dynamic.qsc.de) |
| 02:12.19 | Ryushin | Well, these phones were in another state and the vpn was down between the server for a couple of days. |
| 02:12.35 | Ryushin | I'll kick that phone again and look at the log to see if it does it again. |
| 02:17.49 | Ryushin | To reboot the phone I'm logging into the web server on the phone, going to sip, choosing the last setting and without changing anything, clicking submit to reboot the phone. The the phone truly reboot using this method? |
| 02:20.21 | [TK]D-Fender | should |
| 02:21.31 | jarrod | hmm, has anyone used the asterisk appliance? |
| 02:21.31 | *** join/#asterisk Buhntz (i=Boones@port-212-202-170-97.dynamic.qsc.de) |
| 02:22.19 | Ryushin | Oh well, I'm burnt out on this. I'll pick it up in the morning. This just bugs the crap out of me since this stuff was working fine before. |
| 02:23.08 | Ryushin | Thanks [TK]D-Fender. You awesome to have around here. |
| 02:23.18 | Ryushin | be back in 10 hours. :( |
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| 03:18.47 | yidiyuehan | hi, any one can tell me the website that I can report a IAX channel bug? |
| 03:20.58 | [TK]D-Fender | yidiyuehan, on the Bug Tracker |
| 03:21.43 | yidiyuehan | hi, D-Fender,where is it |
| 03:21.56 | [TK]D-Fender | yidiyuehan, its linked from www.asterisk.org |
| 03:22.08 | [TK]D-Fender | under support I believe |
| 03:22.09 | yidiyuehan | ok. thanks man. |
| 03:24.53 | *** join/#asterisk Math` (n=math@modemcable028.21-56-74.mc.videotron.ca) |
| 03:25.25 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 03:25.34 | Math` | is it possible than OPT_CALLER_HANGUP doesnt work if the call is packet-to-packet bridged? |
| 03:28.30 | *** join/#asterisk theajaysharma (n=ssharma@cpe-76-86-242-163.socal.res.rr.com) |
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| 03:43.59 | perd | are there any good web interfaces for voicemail out there? |
| 03:45.48 | perd | is ari considered 'the best' i guess? |
| 03:49.13 | litage|w | perd: "good", "best" etc are all relative to your needs |
| 03:49.23 | *** join/#asterisk markgreene (n=markgree@c-76-18-121-40.hsd1.al.comcast.net) |
| 03:50.59 | perd | well, i am a simpleton and i like features and flashy looks |
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| 03:57.58 | [TK]D-Fender | perd, Go by an Avaya |
| 03:58.02 | [TK]D-Fender | buy* |
| 03:59.17 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:59.27 | markgreene | Does anyone have any advice to offer for this situation: When I want to record a call I go into the asterisk CLI and type "mixmonitor start SIP/EXTEN filename.wav" and it records the call just fine. Once the call is hung up or I manually stop the recording asterisk restarts, disconnecting any current calls. I am running asterisk 1.2 |
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| 04:07.17 | Math` | I think its worth mentioning in app_dial's documentation that the 'H' option will use the Disconnect feature code once the call is established |
| 04:13.10 | russellb | Math`: yeah, there is a bug report open for exactly that ... to clarify when features become available |
| 04:20.46 | Math` | oh, I'll look it up |
| 04:20.53 | Math` | and add a note to the wiki |
| 04:23.48 | tzanger | weird |
| 04:24.00 | tzanger | zttest reports 100% consistently |
| 04:24.08 | tzanger | but coppice's sliptest program is all over teh map |
| 04:34.38 | russellb | tzanger: have you been writing a zaptel driver lately or something? |
| 04:34.38 | tzanger | russellb: I've got a 288 channel zaptel driver to a proprietary pbx |
| 04:34.38 | russellb | interesting |
| 04:34.38 | tzanger | it works fine so long as I'm taking timing from the PBX |
| 04:34.38 | tzanger | since teh PBX won't let me slave it, I have the DSP implementing elastic buffers and decoupling the TDM and TDMoE sides |
| 04:34.38 | tzanger | think one PC with 4 288-span TDMoE connections to 4 different PBXes |
| 04:34.38 | tzanger | but something is fucking my timing right in the ear |
| 04:34.38 | luke-jr | btw, why doesn't zaptel use the Linux telephony interface? |
| 04:34.38 | tzanger | the elastic buffers seem to be working just fine |
| 04:34.38 | russellb | lol.. |
| 04:34.42 | luke-jr | wtf |
| 04:34.42 | tzanger | ztdummy was teh first suspect, even though the elastic buffer is sized for 5 frames |
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| 04:34.46 | tzanger | but periodically I get a flurry of underrruns/overruns, depending on which eleastic buffer you're looking at |
| 04:34.46 | tzanger | there's a tdm400p in there right now doing nothing but timing |
| 04:35.01 | tzanger | zttest shows perfect consistent results |
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| 04:35.23 | tzanger | but sliptest (sends awgn out an unconnected zaptel port and listens for the echo, trying to correlate it to determine loop length) |
| 04:35.30 | tzanger | but sliptest is ALL OVER the map |
| 04:35.37 | tzanger | sliptest on my home machine's tdm400p is consistent |
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| 04:35.53 | tzanger | so at the moment I don't know who to trust :-) |
| 04:37.30 | rob0 | You can trust your car to the man who wears the Star. |
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| 04:56.01 | *** mode/#asterisk [+o jsmith-away] by ChanServ |
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| 05:09.05 | JT | luke-jr: what are you wtfing at? |
| 05:11.48 | luke-jr | JT: the netsplit ☺ |
| 05:12.11 | JT | yeah, they happen. |
| 05:12.36 | luke-jr | ☺ |
| 05:17.14 | TJNII | To include another file into extensions.com I use #include [filename] correct? |
| 05:17.31 | TJNII | s/extensions.com/extensions.conf/ |
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| 05:19.11 | ManxPower | TJNII: correct |
| 05:19.56 | TJNII | ManxPower: ty |
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| 05:21.28 | ManxPower | To be safe, use the fulll path and don't use quotes unless you understand what you are doing. |
| 05:21.47 | TJNII | Roger |
| 05:24.34 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-47-107.socal.res.rr.com) |
| 05:25.46 | luke-jr | so is Voxee dead? |
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| 05:37.27 | TJNII | With the background command, if I want it to look for file xyz in subdirectory abc of the sounds directory should I do Background(/var.../sounds/abc/xyz) or is this done another way? |
| 05:40.13 | TJNII | survey says Background(abc/xyz) works. Is that the correct way, though? |
| 05:41.00 | Qwell | TJNII: yes |
| 05:41.05 | TJNII | Cool. Thanks |
| 05:41.07 | Qwell | if the files are in /var/lib/asterisk/sounds/ |
| 05:41.15 | Qwell | if not, you'll need to use full path |
| 05:41.27 | Qwell | I wonder if backwards relative paths would work... |
| 05:41.32 | TJNII | Right. Just wanted to make sure what was working for me was actually supposed to work. |
| 05:41.39 | Qwell | mv myprompt.gsm /var/lib/asterisk/blah |
| 05:41.46 | Qwell | Background(../blah/myprompt) |
| 05:41.49 | Qwell | I should try that some time |
| 05:41.58 | luke-jr | mkdir /var/lib/asterisk/blah first |
| 05:42.05 | luke-jr | ☺ |
| 05:42.06 | Qwell | yeah, yeah |
| 05:42.18 | Qwell | somebody test that for me :p |
| 05:42.29 | luke-jr | so any good VoIP providers yet? :P |
| 05:42.34 | Qwell | ~itsp |
| 05:42.34 | jbot | An ITSP (Internet Telephony Service Provider.) is a "VoIP Phone Company". Example : Vonage, Broadvoice, Teliax, VoicePulse, etc. "All ITSPs suck. Some suck less than others. Teliax seems to suck less than most.." (tm) (c) 2007 ManxPower |
| 05:42.41 | Qwell | survey says - nope |
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| 05:42.45 | luke-jr | heh |
| 05:43.05 | luke-jr | Voipjet should do origination |
| 05:43.07 | luke-jr | they seem stable |
| 05:43.24 | kiscokid | Voicepulse seems fine |
| 05:43.29 | luke-jr | Voicepulse? |
| 05:43.47 | luke-jr | Voxee seemed fine when I signed up |
| 05:44.20 | kiscokid | http://connect.voicepulse.com/ |
| 05:44.33 | luke-jr | err |
| 05:44.39 | luke-jr | VoicePulse looks like a ripoff |
| 05:44.41 | luke-jr | ☺ |
| 05:44.52 | kiscokid | why is that? |
| 05:45.21 | *** join/#asterisk nohup- (i=hmmmph@203.81.239.64) |
| 05:46.10 | luke-jr | $15 for the cheapest thing |
| 05:46.58 | *** join/#asterisk Zipper_32 (n=None@d205-250-2-107.bchsia.telus.net) |
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| 05:49.59 | kiscokid | that's the consumer side. Look at the asterisk page |
| 05:50.19 | luke-jr | aha |
| 05:50.37 | luke-jr | pfft, $50 minimum deposit |
| 05:51.12 | kiscokid | what's wrong with that? |
| 05:51.45 | luke-jr | it's not a trivial amount to bet on an unproven service |
| 05:51.55 | kiscokid | its proven to me |
| 05:52.01 | luke-jr | I've seen too many ITSPs die to risk that |
| 05:52.52 | luke-jr | let alone having to sign up just to see their full rate table |
| 05:54.00 | kiscokid | which itsp do you like, if any? |
| 05:55.51 | luke-jr | Voipjet works great for outbound calls |
| 05:56.11 | luke-jr | MyPhoneCompany has pretty much always worked, but their site is Flash so I can't really do anything there if I had to |
| 05:56.45 | luke-jr | iConnectHere *usually* works, but has total crap for support personel |
| 05:57.08 | luke-jr | VoiceStick usually works and actually added a "feature" when asked by enough people |
| 05:57.16 | kiscokid | I tried Voipjet a few months ago. It didn't work |
| 05:57.26 | luke-jr | SellVoip was nice, but unreliable and absolutely no support |
| 05:57.30 | luke-jr | didn't work? O.o |
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| 06:22.28 | loompek | i've got a little ol' question.. how come asterisk doesn't register at some other sip server even though i have all the necesary stuff in sip.conf (register => ... and [server]...) |
| 06:22.51 | loompek | i was checking the tcpdump output for port 5060... asterisk just sends OPTIONS to all of the serververs in sip.conf with register command and qualify !=no |
| 06:22.56 | loompek | here's my sip.conf |
| 06:22.56 | loompek | http://rula.net/121 |
| 06:24.42 | loompek | even though 'sip show peers shows status ok for all |
| 06:29.45 | zeeesh | installing asterisk-addons-1.4.2 getting error, This content is stored as http://sial.org/pbot/27618.? |
| 06:30.51 | loompek | any ideas? |
| 06:31.45 | *** join/#asterisk lbow (n=lbow@dsl-146-6-159.telkomadsl.co.za) |
| 06:36.51 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 06:37.37 | tengulre | hi,all |
| 06:37.43 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
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| 06:37.59 | tengulre | what's different between asterisk-1.4 and asterisk-1.2? |
| 06:39.05 | JT | whatever it says in UPGRADE.txt |
| 06:39.26 | loompek | tengulre 1.4 has more features but it's prolly not so stable because the code is quite fresh |
| 06:39.59 | loompek | JT awsome.. you're here! did you happen to spend some time with my sip.conf? :) |
| 06:40.50 | hmmhesays | in mother russia your sip.conf spends time with you |
| 06:40.52 | JT | sorry i was pretty busy |
| 06:42.35 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
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| 06:44.25 | loompek | JT did you even look at the conf file? i don't think there are any syntax errors and i guess a trained eye (like yours) should probably find the 'gremlin' in a matter of minutes :) here's my sip.conf again... any help would be appreciated - http://rula.net/121 |
| 06:45.54 | *** join/#asterisk r00tlz (n=Cero@190.41.12.173) |
| 06:45.58 | r00tlz | hi |
| 06:47.22 | loompek | morning to you too |
| 06:50.56 | luke-jr | loompek: posting the error might be a good idea |
| 06:51.52 | loompek | luke-jr asterisk doesn't REGISTER but only sends OPTIONS |
| 06:52.08 | loompek | so i can call through a peer |
| 06:52.13 | loompek | but i can't receive a call |
| 06:57.35 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 07:00.16 | kaldemar | loompek: have you checked that with sip set debug? |
| 07:01.10 | loompek | kaldemar don't know how to use sip set debug |
| 07:01.23 | loompek | i've enabled it just right now in console |
| 07:01.27 | loompek | what next? |
| 07:02.30 | kaldemar | it prints all the SIP traffic that asterisk sends or receives. look for the REGISTER messages. |
| 07:02.44 | kaldemar | that's a way to confirm that asterisk tries to register. |
| 07:02.58 | loompek | there aren't any REGISTER |
| 07:03.02 | loompek | that's the whole point! |
| 07:06.57 | luke-jr | loompek: why do you have 2 [general] headers? |
| 07:07.38 | loompek | luke-jr just because.. is it wrong? |
| 07:08.59 | luke-jr | could be |
| 07:10.06 | loompek | no it isn't |
| 07:10.23 | loompek | deleted the second [general] and still no luck |
| 07:10.42 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 07:11.25 | loompek | i can see only OPTIONS in sip set debug... and in tcpdump |
| 07:13.38 | *** join/#asterisk Lawbringer (n=Lawbring@212.183.136.193) |
| 07:13.39 | loompek | i mean.. combinations of OPTIONS and 404 Not found (on the remote asterisk) and OPTIONS and 200 Ok (on all the other) |
| 07:14.47 | loompek | any mode ideas? |
| 07:15.11 | kaldemar | loompek: did you remove the first or the second [general]? what did you do after that? sip reload, restarted asterisk? |
| 07:15.44 | kaldemar | umm. i assume you deleted the second. :P |
| 07:16.43 | *** join/#asterisk lbow (n=lbow@165.146.6.152) |
| 07:17.40 | loompek | restart asterisk |
| 07:17.57 | loompek | of course i deleted the one in between, not the one in the beginning :p |
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| 07:20.32 | *** join/#asterisk linagee (n=linagee@about/linux/staff/linagee) |
| 07:21.06 | kaldemar | loompek: comment out registerattempts=10 and try to debug it again. |
| 07:22.38 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 07:23.07 | loompek | okay.. sip reload... still only options |
| 07:24.49 | loompek | like i said.. asterisk sends ONLY options request... there is no REGISTER anywhere (except from the internal clients) |
| 07:24.55 | loompek | the phones 1,2,3,4 |
| 07:25.41 | *** join/#asterisk Haris (n=Haris@unaffiliated/haris) |
| 07:25.46 | Haris | Hello people |
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| 07:42.44 | henkoegema | <PROTECTED> |
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| 08:04.45 | awk | hrm, anyone advise on these issues? |
| 08:05.24 | awk | ~pbb |
| 08:05.26 | awk | ~pb |
| 08:05.27 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 08:05.57 | *** join/#asterisk adeel (n=adeel@c-24-7-132-155.hsd1.ca.comcast.net) |
| 08:06.08 | awk | http://paste.debian.net/37653 |
| 08:06.15 | awk | I can't see what has changed since 1.4 zaptel |
| 08:06.27 | awk | but it doesn't like these things for some reason, i'm pasting zapata.conf now |
| 08:07.06 | awk | http://paste.debian.net/37654 |
| 08:07.08 | awk | that is my zapata |
| 08:07.31 | awk | any sugestions would be great, i'm getting major issues with dtmf now with this upgrade and it is not due to echo training |
| 08:08.02 | awk | i'm sure if I work this zapata issue out i will have resolved it, I cant find naything on voip-info telling me what I have is wrong, nor does google have replys to these warnings. |
| 08:08.03 | *** join/#asterisk AstNewbie (n=chatzill@058177245004.ctinets.com) |
| 08:10.27 | AstNewbie | Hi everyone, I have a TE120P T1 card connected to my Asterisk server. It works flawlessly since it started to provide service. |
| 08:10.36 | defswork | but ? |
| 08:11.51 | AstNewbie | However, with the latest version, 1.4.11, our users will hear some noise occasionally when dialing out ... |
| 08:12.20 | *** join/#asterisk Uatec (n=uatecuk@adsl.ntsols.com) |
| 08:12.26 | Uatec | hi there |
| 08:12.29 | Uatec | i'm getting Sep 21 09:12:01 WARNING[13087]: file.c:229 ast_writestream: Natural write failed |
| 08:12.30 | Uatec | Sep 21 09:12:01 WARNING[13087]: format_sln.c:166 slinear_write: Bad write (256/320): File too large |
| 08:12.33 | Uatec | every single second |
| 08:12.36 | Uatec | what the hell does it mean? |
| 08:12.42 | Uatec | what file is it failing to write to? |
| 08:12.46 | AstNewbie | More specifically, the problem exists since |
| 08:12.58 | AstNewbie | around 1.4.9 ... |
| 08:14.16 | AstNewbie | It seems the problem relating to the Zap channels .... cos we dont hear any noise when calling internal extension (all are SIP channels) |
| 08:16.34 | Strom_C | what kind of noise? |
| 08:17.08 | AstNewbie | Sili Sala ..... |
| 08:17.27 | Uatec | Asterisk is writing about 5 log files a second |
| 08:17.32 | Uatec | under the name event_log.XXX |
| 08:17.32 | Strom_C | i don't know what a "sili sala" noise is |
| 08:17.36 | Uatec | and it's always empty |
| 08:17.52 | Uatec | there are 6000 files in this directory and counting |
| 08:17.59 | Uatec | and they're all empty |
| 08:18.10 | Uatec | the total size of the directory is like 500k |
| 08:18.41 | luke-jr | lol |
| 08:19.22 | Uatec | it's not funny because after a while asterisk slows right down |
| 08:19.24 | Uatec | this is not right |
| 08:19.36 | Uatec | let me show you some of the stuff from my cli |
| 08:19.50 | luke-jr | out of disk space? |
| 08:20.17 | Uatec | http://rafb.net/p/sUYo6D98.html |
| 08:20.19 | Uatec | no |
| 08:20.19 | Uatec | it's not |
| 08:20.33 | Uatec | why is it creating all these files? |
| 08:20.55 | AstNewbie | Hmmm ... the noise is squelched ... |
| 08:20.57 | luke-jr | read the message |
| 08:20.58 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 08:21.01 | Uatec | i am reading the message |
| 08:21.02 | *** join/#asterisk lokkju_wrk (n=lokkju@unaffiliated/lokkju) |
| 08:21.03 | luke-jr | your files are too large ☺ |
| 08:21.07 | Uatec | but it doesn't tell me which file |
| 08:21.16 | Uatec | i've moved all my log files |
| 08:21.17 | luke-jr | the event_logs I'd presume |
| 08:21.19 | Uatec | effectively deleted them |
| 08:21.28 | luke-jr | it seems to think you have a 0 byte limit |
| 08:21.36 | luke-jr | which could be disk space restrictions |
| 08:21.38 | luke-jr | or quota |
| 08:21.46 | luke-jr | or perhaps a log size limit |
| 08:21.49 | luke-jr | maybe a ulimit |
| 08:22.27 | Uatec | i emptied the logs |
| 08:22.32 | Uatec | but the error continued |
| 08:22.42 | luke-jr | check the other possibilities I mentioned |
| 08:23.18 | Uatec | and i've got 46 gig of free space |
| 08:23.24 | Strom_M | Uatec: it's the slin format driver, which means it's trying to write a sound file |
| 08:24.00 | Uatec | and ulimit is unlimited |
| 08:24.12 | Uatec | ahhhhh |
| 08:24.21 | Uatec | WTF? |
| 08:24.40 | Uatec | there is a 2 gig wav file in /var/spool/asterisk/monitor/ |
| 08:24.45 | luke-jr | LOL |
| 08:24.52 | Uatec | why the hell? |
| 08:25.14 | Strom_M | are you recording any calls? |
| 08:26.15 | Uatec | yes |
| 08:26.17 | Uatec | all of them |
| 08:26.22 | Uatec | but no 2.5 gig long calls |
| 08:26.26 | Uatec | or supposedly not |
| 08:27.16 | Strom_M | perhaps you have a call up that never cleared properly |
| 08:32.39 | luke-jr | it* |
| 08:33.48 | AstNewbie | Hi everyone, I have a TE120P T1 card connected to my Asterisk server. It works flawlessly since it started to provide service. |
| 08:33.49 | AstNewbie | However, since the last few version, 1.4.9 - 1.4.11, our users heard some buzz noise occasionally when dialing out while the called parties didn't ... |
| 08:33.51 | AstNewbie | It seems the problem is relating to the Zap channels .... cos we dont hear any noise when calling internal extension which are SIP channels only. |
| 08:37.43 | AstNewbie | The underlying OS is Debian. |
| 08:37.44 | AstNewbie | Any settings would be related to such situation ?? |
| 08:37.46 | AstNewbie | Thanks in advance. |
| 08:44.48 | AstNewbie | I have also called the telecom ... but telecom seems not having any clues also .... |
| 08:44.49 | AstNewbie | I am not so sure which part causing the problem .... |
| 08:44.51 | AstNewbie | Do you have any idea ?? |
| 08:46.05 | luke-jr | no |
| 08:46.35 | Strom_M | AstNewbie: I'd suggest waiting until the start of business hours in Alabama, and then call Digium for support |
| 08:46.50 | awk | grr, anyone know why the lights are reversed now with parking |
| 08:46.56 | awk | when a call is not parked lights are on the phone |
| 08:47.01 | awk | but when the call goes onto park it goes off? |
| 08:47.06 | awk | what couldbe the reason for this |
| 08:47.13 | AstNewbie | Thanks Strom_M ... |
| 08:47.14 | AstNewbie | ^_^ |
| 08:47.15 | awk | this is some bug with asterisk, any way to resolve this |
| 08:55.35 | *** join/#asterisk nou (i=Chaton@unaffiliated/nou) |
| 09:08.19 | *** join/#asterisk yidiyuehan (n=yidiyueh@58.185.151.162) |
| 09:08.54 | yidiyuehan | Hi, Any one can give me a sample configuration file for ISDN card with bristuff driver? |
| 09:08.57 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 09:13.19 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:17.31 | *** join/#asterisk zeeesh (i=zeeesh@202.166.161.45) |
| 09:19.56 | *** join/#asterisk pimouss (n=bmedici@62.210.194.120) |
| 09:23.18 | pimouss | anyone here could help me on asterisk-gui ? |
| 09:25.04 | awk | @*$%(*$U&@^*$!@*^T*#$^T$*#T$UQ#*$T#(*#$ |
| 09:25.12 | awk | anyone know of any issue with hints on 1.4 |
| 09:25.18 | awk | my lights are reversed |
| 09:25.28 | awk | it doesn't show the lights on the snom when a call is parked |
| 09:26.24 | loompek | well |
| 09:26.29 | loompek | any asterisk gurus here? |
| 09:26.42 | loompek | or is there really noone that could help me |
| 09:26.42 | loompek | :S |
| 09:27.39 | thewiizle | do you have your register string enabled? |
| 09:27.45 | thewiizle | what happens when you type sip show registry |
| 09:30.09 | pimouss | it seems to be dead |
| 09:30.13 | pimouss | nobody answers us on this chan |
| 09:31.22 | *** join/#asterisk Buhntz (i=Boones@port-212-202-170-97.dynamic.qsc.de) |
| 09:35.42 | loompek | sip show registry |
| 09:35.42 | loompek | Host Username Refresh State Reg.Time |
| 09:35.42 | loompek | *CLI> |
| 09:35.43 | loompek | blank |
| 09:35.51 | awk | grrr no hints what so ever |
| 09:36.03 | *** join/#asterisk yannj_fr (n=yannj@chilli.esiee.fr) |
| 09:36.22 | *** part/#asterisk pimouss (n=bmedici@62.210.194.120) |
| 09:38.09 | loompek | so ... how do i enable register strings? |
| 09:38.41 | *** join/#asterisk KpoH (n=AID@host-89-41-66-8.moldtelecom.md) |
| 09:39.27 | KpoH | why "realtime mysql status" causes asterisk crash?! |
| 09:39.48 | *** join/#asterisk Corydon76-dig (i=pink@pdpc/supporter/sustaining/Corydon76-home) |
| 09:39.48 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 09:43.17 | *** join/#asterisk pimouss (n=bmedici@62.210.194.120) |
| 09:44.11 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 09:44.20 | *** join/#asterisk pimouss (n=bmedici@62.210.194.120) |
| 09:44.49 | pimouss | anyone to help please, guys ? |
| 09:46.28 | *** join/#asterisk Corydon76-home (n=tilghman@pdpc/supporter/sustaining/Corydon76-home) |
| 09:46.28 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 09:47.19 | tzafrir_home | help about what? I didn't you you actually asking anything |
| 09:47.55 | awk | tzafrir what is the issue with hints in 1.4 |
| 09:48.02 | awk | my parked is all messed up |
| 09:48.09 | awk | it doesn't show a parked call, the lights dont show on |
| 09:48.17 | awk | is there something i should know about? |
| 09:49.20 | pimouss | yes, i had a problem about using asterisk-hui |
| 09:49.21 | pimouss | gui |
| 09:50.21 | tzafrir_home | pimouss, you don't seem to want others to help you with your problems |
| 09:50.41 | tzafrir_home | Or are otherwise quite shy at mentioning them |
| 09:50.58 | tzafrir_home | Unless you call asterisk-gui your problem ;-) |
| 09:51.14 | tzafrir_home | This is the only piece of information we have lerned about you so far |
| 09:51.54 | tzafrir_home | awk, what have you configured? what do you expect to happen? what actually happens? |
| 09:53.12 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 09:57.31 | *** join/#asterisk Tili (n=tili@cm48.gamma244.maxonline.com.sg) |
| 10:03.00 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 10:07.52 | yidiyuehan | Hi, Any one can give me a sample configuration file for ISDN card with bristuff driver? |
| 10:08.06 | awk | tzafrir: I have setup a parked group 701 right |
| 10:08.15 | awk | now I call into an extension right |
| 10:08.23 | awk | I have added on the snom to monitor 701 |
| 10:08.37 | awk | now I put the call on park but it does not show the light on the phone saying their is a call parked |
| 10:08.39 | awk | yet if |
| 10:08.47 | awk | i hit that function key it goes to that parked call |
| 10:08.52 | awk | so its not showing the parked calls |
| 10:08.57 | *** join/#asterisk zcionn_ (n=a@58.69.243.203) |
| 10:09.00 | awk | if I do s a show hints its just unavialble |
| 10:09.24 | awk | and we using metermaid to assign blf to parking bays |
| 10:09.51 | *** join/#asterisk bintut (n=bintut@cm47.gamma178.maxonline.com.sg) |
| 10:10.05 | awk | 701@uditelco-local : Local/701@parkedcall State:Unavailable Watchers 0 |
| 10:11.11 | bintut | tzafrir: faidon forwarded already the http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=438702 to http://bugs.digium.com/bug_view_page.php?bug_id=10780 |
| 10:11.32 | awk | in extensions.conf I have exten => 701,hint,Local/701@parkedcalls |
| 10:11.42 | awk | what am I missing, it has a hint |
| 10:12.53 | awk | and I have set in a include extensions-app exten => 701,1,ParkedCall(701) |
| 10:14.02 | *** join/#asterisk pepesz76 (n=pepesz76@wolin.et.tudelft.nl) |
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| 10:22.19 | *** part/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-7d526ee367fb0bd3) |
| 10:36.32 | *** join/#asterisk thieums (n=Mathieu@rny93-4-82-231-54-139.fbx.proxad.net) |
| 10:36.53 | *** join/#asterisk disposable (n=michal@82-69-60-92.dsl.in-addr.zen.co.uk) |
| 10:36.59 | *** join/#asterisk jsmith-away (n=jsmith@000-181-428.area3.spcsdns.net) |
| 10:37.00 | *** mode/#asterisk [+o jsmith-away] by ChanServ |
| 10:37.44 | disposable | is it possible to have a ringgroup among extensions defined in a ringgroup? |
| 10:41.29 | thieums | Hello, i would need some help regarding mysql realtime. I would like to link a context from the sip table to the extension table (ie without using the switch statment in extensions.conf). Do you know if it's possible ? |
| 10:45.42 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 10:49.44 | ai-a[out] | Question about Faxing on Asterisk... how come we can fax from uk -> auz. guessing its going over internet and other stuff but we cant fax from one fax machine to asterisk on the same switch with no other data ? |
| 10:53.48 | thewiizle | anyone pretty up to speed on SPA dialplans? |
| 10:54.52 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:56.15 | nDuff | ai-a: you generally can't fax over the internet, at all. |
| 10:56.26 | nDuff | ai-a: even faxing over LANs is iffy, unless you're using T.37 or T.38. |
| 10:57.08 | nDuff | ...in either of those cases, faxing over the Internet works fine, but Asterisk doesn't natively support those protocols, and (in the US, at least) using either of them gives up legal protections which faxes otherwise enjoy. |
| 10:58.00 | *** join/#asterisk masus (n=tet@88.248.73.2) |
| 10:58.13 | masus | hi all, which ports use asterisk anybody know ? |
| 10:58.15 | nDuff | ai-a: faxing is best done as a 100%-zaptel affair, or using something like iaxmodem where the only VoIP span is internal to the server. (Something like TDMoE which guarantees timing is fine too, of course) |
| 10:58.23 | nDuff | masus: depends, what are you doing with it? |
| 10:58.47 | nDuff | masus: Asterisk has support a wide variety of protocols; what ports it needs depends on what protocols you use. |
| 10:59.02 | nDuff | masus: if you want to know which ports your specific installation is using, use a tool like netstat to tell you. |
| 11:03.36 | *** join/#asterisk TUplink_ (n=mythtv@c-24-126-34-203.hsd1.wv.comcast.net) |
| 11:03.57 | TUplink_ | i keep getting a funny error....... [Sep 21 06:47:01] WARNING[78048]: chan_sip.c:8128 check_auth: username mismatch, have <20001>, digest has <> |
| 11:03.57 | TUplink_ | [Sep 21 06:47:01] NOTICE[78048]: chan_sip.c:13388 handle_request_invite: Failed to authenticate user "Tommy Huff"<sip:20001@24.126.34.203>;tag=cb227e18-2710-3d7f27c1-8605-597a0dac |
| 11:03.57 | TUplink_ | <PROTECTED> |
| 11:04.12 | TUplink_ | it use to work... then my ATA lost its config... and now its all FOBAR |
| 11:06.29 | nDuff | TUplink_: check the "auth user" setting in the ATA's settings. |
| 11:06.47 | TUplink_ | ok... i will but im sure its right |
| 11:06.48 | nDuff | TUplink_: should match the regular username. |
| 11:06.58 | nDuff | TUplink_: are you sure there's an entry in that field at all? |
| 11:07.02 | TUplink_ | have to get on the winblows comp to check the xml then upload it |
| 11:07.07 | nDuff | TUplink_: the error you gave reads like "auth user" is blank. |
| 11:07.08 | thieums | any realtime specialist here ? |
| 11:07.36 | TUplink_ | when i paste that error in a webbrowser it comes up as a funnt box |
| 11:08.03 | TUplink_ | its noty even registering |
| 11:09.13 | TUplink_ | let me go check |
| 11:11.19 | masus | nDuff: what i have done is a outgoing with 2 internet connections thrue ADSL + GSHDSL , it's working but atADSL i hav NAT problems so i'll do port forwarding |
| 11:11.49 | masus | so i need , which ports are used or which range of ports |
| 11:12.42 | nDuff | masus: you still haven't told me what protocols you're using. |
| 11:13.08 | nDuff | IAX? SIP? |
| 11:13.08 | masus | i haven't told u have ask ? |
| 11:13.15 | masus | SIP |
| 11:13.32 | masus | sorry my english is not good |
| 11:13.42 | thieums | 5060 udp |
| 11:15.26 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 11:23.00 | Haris | Guys, can I configure myself, Vonage inbound on a linksys RTP300 ? |
| 11:27.22 | *** join/#asterisk Polis_ttt (n=Polis_tt@194-237-172-225-no48.business.telia.com) |
| 11:28.23 | ai-a | nDuff: well we have a softfax on this server that works quite well. However we've got only the pbx and ata box on a switch with nothing else. and its 100% failure so far. |
| 11:31.32 | *** join/#asterisk TUplink_ (n=mythtv@c-24-126-34-203.hsd1.wv.comcast.net) |
| 11:31.34 | TUplink_ | nDuff..... not sure what i did i removed the proxy and changed the caller id name to match the extension and now it works |
| 11:31.43 | *** join/#asterisk s0lid (n=jlq@210.213.198.122) |
| 11:32.43 | *** part/#asterisk TUplink_ (n=mythtv@c-24-126-34-203.hsd1.wv.comcast.net) |
| 11:33.47 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 11:34.19 | *** join/#asterisk hieunm_vips (n=hieunm_v@210.245.57.162) |
| 11:35.21 | hieunm_vips | hi all, I want to change volume of sound when playback to user, how could I do this with "Playback" application? |
| 11:35.38 | *** part/#asterisk masus (n=tet@88.248.73.2) |
| 11:35.58 | hieunm_vips | Or is there any other applications support my case ? |
| 11:47.48 | Uatec | hey if i monitor() a call to the same file every time, does it concatenate the file? |
| 11:53.30 | hwt | hey. i am looking for a way to tap into the INCOMING voice to a meetme conference, to kick users that are making noise. |
| 11:53.33 | hwt | any suggestions? |
| 11:57.41 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 11:57.51 | lirakis | morning |
| 11:59.48 | hwt | is it at all possible? |
| 12:03.05 | *** join/#asterisk guillote_GNU (n=bancaria@host35.201-253-17.telecom.net.ar) |
| 12:07.49 | *** join/#asterisk lolscorruptionof (i=aaron@nat/digium/x-d1184c8549e6a01d) |
| 12:08.45 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
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| 12:10.14 | *** join/#asterisk PepOSX (n=pepOSX@190.72.153.233) |
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| 12:45.48 | _x86_ | anyone use OctWare SoftEcho with Sangoma cards that already have HWEC? |
| 12:46.23 | Uatec | hey there |
| 12:46.31 | Uatec | well i'm about to buy an a500 |
| 12:46.31 | _x86_ | morning ;) |
| 12:46.38 | _x86_ | a500? |
| 12:46.42 | _x86_ | whats that? |
| 12:46.43 | Uatec | the sangoma BRI |
| 12:46.46 | _x86_ | ah |
| 12:46.55 | Uatec | and i want to decide if i want to get the echo cancellation module |
| 12:46.57 | Uatec | it's £150 |
| 12:47.19 | _x86_ | dunno about with BRI/PRI, because those are digital interfaces anyway |
| 12:47.43 | _x86_ | POTS in .us are analog, and you almost always need HWEC |
| 12:48.17 | Uatec | HWEC? |
| 12:48.22 | Uatec | Hard white enveolope cuttings? |
| 12:48.27 | _x86_ | hardware echo cancellation |
| 12:48.32 | Uatec | ahh |
| 12:48.50 | Uatec | £150, hmmm |
| 12:48.53 | Uatec | i think we do want it |
| 12:49.33 | tzafrir_home | Uatec, is it actually supported? do current Sangoma drivers support it? |
| 12:50.59 | Uatec | well, the sangoma cards come with an optional echo cancellation module |
| 12:51.12 | Uatec | it would seem pointless if the drivers didn't support it |
| 12:59.10 | _x86_ | tzafrir_home: yeah, sangoma drivers support it |
| 12:59.41 | _x86_ | Uatec: it's not enabled by default, you have to use the wancfg utility (included with the drivers) to manually enable it on each span |
| 13:00.24 | *** part/#asterisk kippi (n=Cowen@untrust-gct.equinoxit.net) |
| 13:01.14 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 13:01.18 | tzafrir_home | right, I see now |
| 13:01.26 | tzafrir_home | http://wiki.sangoma.com/sangoma-wanpipe-smg-asterisk-bri-installation |
| 13:01.38 | tzafrir_home | nice of them to use bristuff |
| 13:03.23 | *** join/#asterisk blinky42 (n=me@c-75-69-77-42.hsd1.vt.comcast.net) |
| 13:03.46 | *** join/#asterisk af_ (n=getsmart@81-174-46-123.dynamic.ngi.it) |
| 13:05.01 | pimouss | hello |
| 13:05.04 | pimouss | I have a problem trying to get GUI to work |
| 13:05.04 | pimouss | asterisk's http is responding on 8080 but answers File not found on / |
| 13:05.04 | pimouss | any idea please ? |
| 13:05.49 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.37.205) |
| 13:07.56 | *** join/#asterisk ivanfm (n=ivanfm@c906b486.virtua.com.br) |
| 13:09.36 | *** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 13:09.45 | tzafrir_home | pimouss, I don't think that / should have a file |
| 13:09.58 | tzafrir_home | A redirection from / as an optional setup, IIRC |
| 13:11.53 | Kurin- | Is there any reason a polycom phone would break a router or a network connection? |
| 13:12.10 | pimouss | yes, indeed, but none of the expected URLs work |
| 13:12.11 | *** join/#asterisk Corydon76-dig (i=peuce@pdpc/supporter/sustaining/Corydon76-home) |
| 13:12.11 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 13:12.16 | Kurin- | We're installing these phones and three times now when we've plugged the phone in the router's have died |
| 13:12.39 | *** join/#asterisk Corydon76-home (i=eleven@pdpc/supporter/sustaining/Corydon76-home) |
| 13:12.39 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 13:12.39 | deeperror | Kurin: are phone lines plugged into network jacks? |
| 13:12.44 | Kurin- | and one was a fbsd box, and when the nic died ifconfig said "no carrier" but the nic showed a link light |
| 13:12.47 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:13.06 | Kurin- | The patch cord goes from the phone's "LAN" jack to the switch |
| 13:14.24 | *** join/#asterisk nacer (n=nacer@l.alcolo.a.mpl.pastIX.net) |
| 13:14.29 | nacer | hey hey |
| 13:14.49 | nacer | someone know a good documentation for make update of asterisknow ? |
| 13:15.38 | lolscorruptionof | conary update conary |
| 13:15.52 | lolscorruptionof | conary update asterisk zaptel libpri |
| 13:15.54 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:16.03 | nacer | oki |
| 13:16.08 | nacer | how to have a shell ? |
| 13:16.30 | lolscorruptionof | Change to another virtual term |
| 13:17.15 | keulin | why my genzaptelconf script is displaying for about on hour "Generating '/etc/zaptel.conf'" |
| 13:17.18 | keulin | ? |
| 13:18.06 | nacer | ok |
| 13:18.51 | modu | just a question about IAX/RSA : there is no way to sign IAX paquet to prevent man in the middle attacks ? |
| 13:19.07 | [TK]D-Fender | nacer: You're in the wrong channel, please read the topic. |
| 13:19.17 | nacer | ok |
| 13:19.44 | nacer | [TK]D-Fender, tks i am in the good one now :) |
| 13:20.05 | nacer | tks lolscorruptionof for your help |
| 13:20.53 | keulin | no idea about this issue ? |
| 13:22.22 | [TK]D-Fender | keulin: just hit enter. if you're not at aprompt, just ctrl-c out |
| 13:23.07 | *** join/#asterisk shtoom (n=shtoom@59.93.127.243) |
| 13:24.07 | keulin | [TK]D-Fender, yes, i've allready done that but in this case /etc/zaptel.conf is not generated |
| 13:24.17 | keulin | I don't understand why |
| 13:24.29 | [TK]D-Fender | just build them yourself. its a handful of lines... |
| 13:26.10 | rob0 | ./gen[TK]D-Fender.conf && echo it worked |
| 13:27.33 | keulin | yes i know how to do that, but i would like to make this work |
| 13:27.37 | modu | is IAX encryption is stable now ? |
| 13:31.20 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 13:32.23 | *** join/#asterisk VJFROMGT (n=vjfromgt@static-68-161-227-229.ny325.east.verizon.net) |
| 13:32.51 | VJFROMGT | i have an outbound route with matching pattern but evertime i try to make a call i get "no rotue to destination" |
| 13:36.51 | [TK]D-Fender | VJFROMGT: pastebin the call CLI with channel debug enabled. |
| 13:42.03 | Kurin- | So no one's ever had a polycom phone break the switch it's plugged into? |
| 13:42.39 | [TK]D-Fender | Kurin-: nope....I have seen 2 Polycoms simply "die" however |
| 13:42.54 | Kurin- | No, this is definitely the switch |
| 13:43.10 | Kurin- | He plugs his phone in and his PC gets knocked offline and can't ping the gateway |
| 13:43.19 | Kurin- | but when he brings his phone in it still works fine |
| 13:43.26 | JT | username=jesse |
| 13:43.26 | JT | secret=jghb572 |
| 13:43.30 | JT | oops |
| 13:43.35 | *** join/#asterisk rexile (i=elixer@65.207.74.18) |
| 13:43.39 | keulin | lol |
| 13:43.42 | *** join/#asterisk zerohalo (n=zeroHalo@pool-72-70-79-233.bstnma.east.verizon.net) |
| 13:43.58 | Kurin- | We all have the same password |
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| 13:44.07 | Kurin- | I wish you could put in your SIP credentials via the phone |
| 13:44.18 | [TK]D-Fender | Kurin-: ? |
| 13:44.28 | Kurin- | Just musing at JT's misspaste |
| 13:44.31 | Kurin- | mispaste |
| 13:44.55 | [TK]D-Fender | Kurin-: No, wondering about the "sip credentials on phone" bit... |
| 13:44.56 | Kurin- | Yeah so I'm pretty sure the phone is somehow breaking the switch, since when I plugged it into my fbsd machine it took the NIC offline |
| 13:45.15 | [TK]D-Fender | Kurin-: What model? |
| 13:45.16 | deeperror | it probably has a hard coded ip? |
| 13:45.17 | Kurin- | Well the polycom downloads its config file, which has the sip username/password in it |
| 13:45.28 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:45.28 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:45.28 | Kurin- | Soundpoint 550 and 330 |
| 13:45.45 | Kurin- | No it gets its IP from DHCP |
| 13:45.52 | [TK]D-Fender | Kurin-: You can enter auth infor directly in the phones LCD menus, in the web admin, through provisioning files, and even through DHCP. |
| 13:45.58 | JT | heh you have no sip hostname anyway, mwuahaha |
| 13:46.07 | Kurin- | And in fact it's on the network for a few seconds |
| 13:46.12 | Kurin- | Since I can see packets going through |
| 13:46.16 | Kurin- | But then everything just stops |
| 13:46.17 | JT | i meant to paste http://youtube.com/watch?v=AmCc6MEhNGM rofl! |
| 13:46.25 | Kurin- | I'm not really concerned about auth info |
| 13:46.26 | [TK]D-Fender | Kurin-: Any chance he doesn't have PoE at home and doesn't have an injector? |
| 13:47.03 | Kurin- | Well he doesn't have PoE, no, but these phones all use regular power cords |
| 13:47.55 | bintut | gtg.. |
| 13:49.02 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 13:49.18 | Kurin- | Could it be that the phone is doing some PoE stuff out the LAN interface and needs the patch cable that came with it? |
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| 13:52.54 | tzanger | morning |
| 13:54.32 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 13:56.10 | *** join/#asterisk coppice (n=chatzill@50.206.17.210.dyn.pacific.net.hk) |
| 13:56.18 | tzanger | he's here!! |
| 13:56.21 | tzanger | good morning, coppice |
| 13:56.28 | tzanger | or evening for you I suspect |
| 13:57.14 | coppice | well, I feel sleepy. either its bedtime, or I'm in an important meeting |
| 13:57.20 | tzanger | haha |
| 13:57.21 | hwt | hey. i am looking for a way to tap into the INCOMING voice to a meetme conference, to kick users that are making noise. any suggestions? |
| 13:57.33 | tzanger | I have a datapoint and question for you, if you have time |
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| 13:58.35 | tzanger | I have access to a system with a tdm400 in which the zaptel-provided zttest program consistently reports 100% for timing accuracy (8192 samples in 8192 sample periods), yet sliptest returns results all over the map (inconsistent correlation between emitted sound and detected audio) |
| 13:58.57 | tzanger | replaced the card and sliptest is now reporting what I'd expect (about 508/516) |
| 13:59.18 | Kurin- | Yeah that must be it |
| 13:59.30 | Kurin- | I plugged the phone on my desk in with a different cable and made a call |
| 13:59.38 | Kurin- | Within about a second it died |
| 13:59.53 | tzanger | my question is how would the two (zttest and sliptest) be at such huge odds with each other? I understand they're measuring timing differently (one strictly time, the other measuring audio path loop) but it seems strange |
| 13:59.55 | Kurin- | though it recovered |
| 14:00.13 | tzanger | sliptest appears to work equally well on FXS and FXO ports, at least on my TDM400P at home here |
| 14:07.21 | [TK]D-Fender | hwt : why not jsut MUTE them out instead? |
| 14:09.06 | [TK]D-Fender | hwt: And you could always parse out "show meetme X" to see who's talking... |
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| 14:25.09 | MindTheGap | hello all, i need to answer a zap channen call and if I get an specific sound i want * to drop the call. so far i'm having 60% success at this using backgrouddetect because sometimes it detects this sound as a "DTMF 2". but its not consistent. is a pre recorded call from the telco warning me that this is a collect call and I should hangup if i do not wish to pay for the call. As you may have notice, the telco is not willing to block collect calls at the |
| 14:25.10 | MindTheGap | ir end, so im stuck. Its a E1 ISDN. |
| 14:26.03 | *** part/#asterisk hieunm_vips (n=hieunm_v@210.245.57.162) |
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| 14:27.54 | tzafrir_home | MindTheGap, "drop the call if I get a specific sound": that's busydetect? |
| 14:28.04 | tzafrir_home | But what if htat sound occours randomly? |
| 14:31.12 | MindTheGap | tzafrir_home, it is a pre recorded message, its the same all the time, but theres music in it and sometimes its recognized as DTMF |
| 14:31.58 | tzafrir_home | MindTheGap, if it is ISDN, I suspect that there are smarter ways to know you should hang up |
| 14:33.24 | MindTheGap | tzafrir_home, please enlighten me :) |
| 14:33.58 | Uatec | hey, is there any hardware i can use to simulate an ISDN line? |
| 14:34.15 | JT | an isdn simulator |
| 14:34.25 | russellb | asterisk with a T1/E1 card in it? :) |
| 14:34.54 | tzafrir_home | Uatec, hardware? sure. A dual ISDN adapter. Did you ask about software? |
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| 14:35.33 | Uatec | hardware |
| 14:35.34 | MindTheGap | tzafrir_home, i know it is possible on a E1/R2 using unicall, but im not aware of any methosd on ISDN. |
| 14:35.39 | Uatec | i have just bought a sangoma a500 |
| 14:35.51 | tzafrir_home | that's ISDN BRI, not PRI |
| 14:35.54 | Uatec | but i don't have an ISDN BRI line to test it on |
| 14:36.04 | Uatec | i know, i didn't say PRI |
| 14:36.37 | tzafrir_home | Uatec, Asterisk with a cheapo HFC-s card? |
| 14:37.30 | Uatec | HFC-s card? |
| 14:37.33 | Uatec | hmm |
| 14:38.16 | *** part/#asterisk MtJB (n=warthawg@cpe-66-68-180-85.austin.res.rr.com) |
| 14:38.37 | Uatec | and hfc-s doesn't appear too useful for finding cards on google |
| 14:38.50 | tzafrir_home | cologne HFC-s is the name of the chipset |
| 14:39.45 | Uatec | know any cards which use it? |
| 14:40.29 | tzafrir_home | Billion , hmm |
| 14:40.47 | [TK]D-Fender | tzafrir_home: No, I'm sure he jsut needs one ;) |
| 14:42.27 | Uatec | WAIT |
| 14:42.37 | nacer | ?fxo fxs |
| 14:42.54 | tzanger | NT = network termination, TE = terminal endpoint? I can't remember |
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| 14:44.00 | [TK]D-Fender | ouch |
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| 14:44.36 | santibiotico | TE = Terminal Equipment |
| 14:45.40 | Uatec | so i could actually set one of my ports of my b410p to TE and one of them to NT and have them connect to each other ? |
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| 14:45.58 | Uatec | that sounds too simple |
| 14:46.06 | JT | Uatec: correct |
| 14:46.08 | tzafrir_home | Uatec, right |
| 14:46.10 | JT | i've done it before |
| 14:46.43 | tzafrir_home | Uatec, just make sure you use a cable with all 8 wires |
| 14:46.47 | tzafrir_home | non-crossed |
| 14:47.13 | tzafrir_home | (all 8 wires: ISDN BRI uses just 4, but not exactly the same 4 as ethernet) |
| 14:47.28 | Uatec | ok, well all our cables here have all 8 |
| 14:47.30 | Uatec | awesome |
| 14:47.38 | Uatec | i just became very very very happy and excited |
| 14:47.41 | tzafrir_home | ~fxsfxo |
| 14:47.42 | jbot | well, fxsfxo is an FXO port expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 14:49.18 | tzanger | coppice: any ideas about my question regarding HUGE difference between zttest and sliptest? |
| 14:50.19 | coppice | tzanger maybe it matches so fantastically well, there is no echo |
| 14:50.57 | tzanger | coppice: an unterminated line on a TDM400 acting as a perfectly terminated line? surely you are joking with me :-) |
| 14:51.40 | coppice | how does it sound? is the card creating a horrible distorted mess that won't correlate? |
| 14:52.18 | tzanger | I must admit I have not listened to a thing, I'm only using the card as a source of timing for tdmoe |
| 14:52.34 | *** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 14:53.26 | tzanger | sliptest should work equally well with FXO and FXS ports, right? I think it would (but I'm not the brightest, etiher) |
| 14:53.41 | *** join/#asterisk KpoH (n=AID@host-89-41-66-8.moldtelecom.md) |
| 14:54.04 | tzanger | it seems to work on my home * box anyway on both types of ports, and with digitla lines that are looped back it also seems to respond as I'd expect it to |
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| 14:57.02 | tzafrir_home | tzanger, I commited some fixes to zttest a week ago or so. With them as well? |
| 14:57.42 | tzafrir_home | Make sure you use zttest -v, and also look at the final result after you press ctrl-C |
| 14:57.55 | tzanger | no this is a 1.4.3 pull of zaptel (i'm locked to thsi version for the client) |
| 14:58.13 | tzanger | tzafrir_home: yeah I am getting what I expect to be normal results with the new tdm400 |
| 14:58.26 | tzafrir_home | locked version? just pull it from somewhere as source / binary |
| 14:58.33 | tzanger | I know :-) |
| 14:58.43 | tzanger | just haven't got to it yet,a nd they're all off for Yom Kipur now anyway |
| 14:58.46 | tzanger | so I have to wait for Monday |
| 14:59.19 | *** join/#asterisk mtaht4 (n=m@172-110-62-200.enitel.net.ni) |
| 14:59.45 | tzafrir_home | What is sliptest? |
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| 15:04.29 | tzanger | tzafrir_home: it's a little utility the dsp guru coppice came up with which opens a zap channel and spews AWGN on the line. it then listens to the line and auto-correlates the transmitted audio to the received audio |
| 15:04.50 | tzanger | essentially it determines the loop length, which for any TDM circuit should be pretty much consistent at *some* value |
| 15:04.58 | tzafrir_home | ~AWGN |
| 15:05.19 | tzanger | additive gaussian white nose |
| 15:05.27 | coppice | A White Guy's Noise |
| 15:05.36 | tzanger | you mean like Eminem? |
| 15:05.41 | tzafrir_home | looks like my toolkit is missing some pretty useful tools from spandsp |
| 15:05.47 | [TK]D-Fender | "He didn't have Ice Cube, so he brought Vanilla Ice" |
| 15:05.56 | tzanger | I didn't realize you could write his music mathematically :-) |
| 15:07.17 | tzanger | tzafrir_home: yeah I love this app |
| 15:07.45 | tzanger | it tells you in one quick test whether a) timing is consistent b) path is consistent and c) audio should work |
| 15:09.10 | nDuff | ai-a: what codec is the ata using? |
| 15:09.38 | tzafrir_home | coppice, is it included in spandsp? |
| 15:09.48 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 15:10.06 | tzanger | tzafrir_home: no |
| 15:10.35 | tzanger | http://www.soft-switch.org/downloads/sliptest.c |
| 15:10.38 | nDuff | ai-a: you should be using alaw or ulaw. |
| 15:11.15 | tzanger | er sli8ptest |
| 15:11.19 | tzanger | dammit |
| 15:11.19 | tzanger | sliptest |
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| 15:14.04 | *** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136) |
| 15:14.27 | tzafrir_home | coppice, mind you that you use <linux/zaptel.h> which has changed to <zaptel/zaptel.h> in zaptel 1.4 |
| 15:14.44 | Aeudian | Question: Is there a way to stop extensions.ael from being parsed on reload? my guess would be add a noload line in modules, but i am not sure what. |
| 15:17.24 | nDuff | Aeudian: pbx_ael |
| 15:18.19 | coppice | tzafrir_home: 1.4 didn't exists when I wrote that code |
| 15:18.27 | Aeudian | nDuff: noload => pbx_ael in modules.conf right? |
| 15:18.52 | nDuff | Aeudian: sounds about right, yes. |
| 15:19.22 | Aeudian | nDuff: still getting notifications from pbx_load_module |
| 15:19.26 | outtolunc | rename it <G> |
| 15:19.29 | [TK]D-Fender | noloca => pbx_ael.so |
| 15:19.41 | [TK]D-Fender | noload => pbx_ael.so |
| 15:20.46 | tzafrir_home | Anyway, now zttest (in svn) should count only the time it actually spends reading. I sitll suspect it is a bit inaccurte |
| 15:22.26 | coppice | what does zttest actully do? it used to be something entirely meaningless |
| 15:22.40 | Aeudian | TK: i did that but pbx_load_module still attempts to find the file, my guess cause i see an auto load all? |
| 15:22.46 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 15:23.22 | nDuff | coppice: counts percentage of interrupts serviced, if I understand correctly (!= guaranteed) |
| 15:23.47 | coppice | nDuff: that is awfully vague |
| 15:23.59 | nDuff | coppice: that *is* useful -- if for no other reason for telling people for whom the percentage is too low that they need to go fix their systems before faxing will wore correctly. |
| 15:24.16 | tzanger | coppice: my understanding is that it waits 8192 sample periods, and then counts the actual number of samples receieved in that period |
| 15:24.17 | coppice | it tells them nothing of the sort |
| 15:24.18 | [TK]D-Fender | <PROTECTED> |
| 15:24.45 | tzafrir_home | compare time as messured by the clock generated by Zaptel to the system clock. One thing it does very well is detecting when the Zaptel clock is very bad: non-existant, badly lagging, etc. |
| 15:25.02 | Aeudian | TK: ya it works, i forgot i had to restart asterisk service a simple reload wouldn't do it, thanks |
| 15:25.27 | coppice | tzafrir_home: if that is what it does, it is entirely useless |
| 15:25.30 | tzanger | tzafrir_home: I've found sliptest to be far far more useful in that regard |
| 15:25.55 | nDuff | coppice: there's a well-established relationship between zttest numbers being below 99.98% and faxing being iffy. |
| 15:26.06 | tzanger | nDuff: I don't believe that |
| 15:26.10 | *** join/#asterisk ManxPower (n=manxpowe@202.sub-70-220-216.myvzw.com) |
| 15:26.19 | tzanger | I have several systems where zttest tells me that faxing will never work, where it in fact works very well |
| 15:26.30 | tzafrir_home | tzanger, well, does sliptest need an actual Zaptel channel? |
| 15:26.31 | tzanger | zttest ~95% but consistent results |
| 15:26.34 | tzanger | tzafrir_home: yes |
| 15:26.50 | tzafrir_home | or can it use a single pseudo channel? |
| 15:27.09 | etfonhomey_ | [TK]D-Fender Which Polycom phone do you recommend for the secretary/power user? |
| 15:27.11 | tzanger | tzafrir_home: it requires an actual channel, as it actually spits audio out and listens for its reflection |
| 15:27.40 | coppice | nDuff: its telling you something completely bogus |
| 15:27.44 | tzafrir_home | well, you don't always have a handy channel. In fact, when Asterisk runs, you don't |
| 15:27.50 | *** join/#asterisk |NexT| (n=next@netcache1.altecom.net) |
| 15:27.52 | tzanger | tzafrir_home: agreed |
| 15:28.02 | tzanger | but without an actual channel, how do you measure actual path timing? |
| 15:28.02 | coppice | faxing is *not* dependent on the system clock at all |
| 15:28.26 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 15:29.09 | nDuff | coppice: *shrug*. I'm relaying what's quite typically painted as a scapegoat on the iaxmodem list |
| 15:29.33 | coppice | if there is an echo anywheer down the line, sliptest will tell you is the loop length is stable. that is a meaningful test of what will work for faxing |
| 15:29.33 | nDuff | coppice: ...including by folks who should know better if it's not true. |
| 15:29.53 | FXOL | anyone got some time to help w/ a basic php problem? |
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| 15:30.33 | tzafrir_home | coppice, right. But if the zaptel clock is not very close to the system clock, chances are that the Zaptel clock is bogus. This is why zttest is very good at detecting simple problems with it |
| 15:32.47 | |NexT| | 8192 zaptel samples in 8192.217 system clock sample intervals (100.003%) |
| 15:32.53 | |NexT| | this is normal? |
| 15:33.13 | tzafrir_home | nDuff, I am actually yyet to messure a clear 100% on systems where Zaptel is clocked by our devices. But they pass faxes pretty well |
| 15:33.28 | tzafrir_home | We do have other indications of where things slip |
| 15:33.32 | jsmith | FXOL: Well, I could probably help you out, but this isn't really the right channel for it... what's your question, and then we'll take this off the channel |
| 15:34.04 | nDuff | tzafrir: I've actually gotten 100%, but only on multicore systems using Sangoma's PCI-E boards. |
| 15:34.08 | tzafrir_home | zttest is not entierely accurate. And its display of multiple digits of precision just makes it appear "accurate" |
| 15:34.36 | FXOL | (jsmith): I got help... fixed, thanks ;P |
| 15:35.06 | FXOL | (jsmith): Had to modify my PHP script to help get Cepstral working right, and apparently screwed up one little thing :P |
| 15:35.40 | [TK]D-Fender | etfonhomey_: Power users don't normally factor in, receptionists = IP 650 |
| 15:35.57 | etfonhomey_ | Thanks! |
| 15:36.15 | ManxPower | Charter Cablemodem tech support - when you need a good laugh. |
| 15:36.44 | etfonhomey_ | What artifacts would you hear in the audio of a phone call if there were issues with jitter? |
| 15:36.49 | rob0 | "What version of Windows are you running?" |
| 15:36.55 | [TK]D-Fender | etfonhomey_: Basically its not a big premium over the 601 any more and its a backlit screen, HD (like that matters....), USb expansion, etc... |
| 15:37.06 | ManxPower | etfonhomey_: think bad cellphone connection |
| 15:37.08 | [TK]D-Fender | etfonhomey_: Mayan <- |
| 15:37.12 | jsmith | etfonhomey_: Pops, crackles, something that sounds like a bad cell phone |
| 15:37.32 | jsmith | [TK]D-Fender: I think they're Aztec, not Mayan |
| 15:37.34 | jsmith | :-) |
| 15:37.44 | [TK]D-Fender | jsmith: The jury is out.. |
| 15:37.57 | ManxPower | you don't get pops and crackles bad cell connections unless you are using ANALOG |
| 15:39.45 | Corydon76-dig | or eating rice krispies on the phone |
| 15:40.17 | FXOL | (jsmith): wanna test? :P |
| 15:40.58 | jsmith | FXOL: What exactly would I be testing? |
| 15:41.07 | FXOL | naw.. it's ok ;P |
| 15:41.35 | FXOL | just part of my IVR |
| 15:42.12 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 15:42.20 | flujan | hi all. |
| 15:42.42 | *** join/#asterisk swombat (n=KDan@87-194-122-30.bethere.co.uk) |
| 15:42.43 | flujan | is it possible to asterisk record a extension state on a postgresql database? |
| 15:43.02 | flujan | swombat: here I am ... :D |
| 15:43.12 | flujan | [TK]D-Fender: hi man... how are you doing? |
| 15:43.19 | swombat | flujan: I think your question should be: "Is it possible to get an AGI script to be called every time the extension state changes? |
| 15:43.22 | swombat | " |
| 15:43.35 | swombat | if you can get an AGI script called, you can do whatever you want to the db from there |
| 15:43.38 | flujan | swombat: yeap.. It will also solve the probem... :D |
| 15:43.46 | flujan | swombat: for sure... |
| 15:44.02 | swombat | so. now we await the answers from all these lovely people here :-) |
| 15:44.07 | FXOL | wow |
| 15:44.11 | [TK]D-Fender | swombat: It would not be AGI you'd be wanting to do, and yes. You could have a script monitoring AMI for the state change notification messages and act accordingly. |
| 15:44.12 | FXOL | multiport Cepstral is $ |
| 15:44.40 | swombat | [TK]D-Fender: flujan is trying to resolve a performance issue because monitoring 500 extensions gets a bit slow |
| 15:45.02 | swombat | [TK]D-Fender: so i suggested he should update the status in the db when it changes, rather than polling it every 5 seconds across hundreds or even thousands of users |
| 15:45.30 | [TK]D-Fender | swombat / flujan :elaborate on how you expect to monitor them. |
| 15:46.01 | swombat | I was suggesting that he has some sort of AGI call whenever the state changes, and that that call then updates the db |
| 15:47.19 | flujan | [TK]D-Fender: yeap... Actually I am using AJAM to do this... But I run a loop on each extension, parsing the results from each on them and updating each user on the database. |
| 15:47.41 | [TK]D-Fender | AJAM? |
| 15:48.15 | outtolunc | its the backend asterisk-gui uses to talk to manager |
| 15:48.53 | [TK]D-Fender | :/ |
| 15:48.54 | flujan | [TK]D-Fender: yeap... the manager api that I can acess like a web-service. |
| 15:48.57 | *** join/#asterisk yannj_fr (n=yannj@vpn.intelunix.fr) |
| 15:49.20 | flujan | http://www.voip-info.org/wiki/view/Aynchronous+Javascript+Asterisk+Manager+%28AJAM%29 |
| 15:49.40 | ManxPower | using an AGI everytime an extension state changes will |
| 15:49.50 | ManxPower | NOT "improve performance" |
| 15:50.07 | swombat | ManxPower: not on the asterisk side |
| 15:50.27 | swombat | but the bit where performance is dying is where he loops through a thousand states every 5 seconds to keep the db up to date |
| 15:50.29 | flujan | ManxPower: I run a script that collects the extensions state every 5 seconds and stores it on a db. |
| 15:50.48 | flujan | the script takes about 5 seconds to run... |
| 15:51.24 | flujan | ManxPower: how much performance I will loose using this behavior of asteirsk calling a agi script ? |
| 15:51.49 | ManxPower | flujan: I don't know, but there are several forks and execs involved. |
| 15:52.15 | swombat | that's irrelevant |
| 15:52.26 | swombat | i presume there are not 500 state changes per second |
| 15:52.32 | flujan | swombat: the forks and execs? |
| 15:52.33 | swombat | or per 5 seconds even |
| 15:52.48 | swombat | flujan: assumption: state changes much less than 500 times per 5 seconds |
| 15:52.50 | swombat | correct? |
| 15:52.59 | flujan | swombat: for sure... |
| 15:53.03 | swombat | (either that, or you have 500 users online all the time and switching like crazy) |
| 15:53.21 | swombat | therefore, even if the AGI takes 10x as long as the AJAM, it doesn't matter because it's called much more rarely |
| 15:53.40 | flujan | swombat: nops they not swich states like crazys... |
| 15:53.50 | flujan | yeap. |
| 15:53.57 | twisted | i've got the world on a string |
| 15:53.58 | swombat | agi calls are near-instantaneous on my server. Compared to your 5 second loop, you might as well consider them free |
| 15:54.14 | flujan | ManxPower [TK]D-Fender how can I put in on the dialplan? |
| 15:54.34 | ManxPower | flujan: Did any say you CAN run an AGI everytime an extension state changes? |
| 15:54.42 | ManxPower | I'm not aware of that feature |
| 15:55.48 | *** join/#asterisk bmd (n=bmd@72.54.252.34) |
| 15:55.49 | flujan | ManxPower: ... [TK]D-Fender suggest a AMI that will notify asterisk on each state change... |
| 15:56.07 | CCFL_Man2 | is Mark here? |
| 15:56.15 | [TK]D-Fender | flujan: What place does this have in the DIALPLAN? |
| 15:56.59 | [TK]D-Fender | flujan: What do you want to do on state change exactly? |
| 15:57.26 | swombat | he wants to update a single field in a pgsql db |
| 15:57.42 | flujan | [TK]D-Fender: just have the control of which users are on the phone, which is not logged and so on... I will display it on a report. |
| 15:58.05 | flujan | [TK]D-Fender: to know if the users are on the phone... Dialing and so on. |
| 15:58.15 | [TK]D-Fender | flujan: the just make a COMPLETELY seperate script (no AGI or dialplan involved), which will poll AMI for state change messages and do your DB work. |
| 15:58.27 | coppice | tzafrir_home: sure, its some kind of sanity check, but people try reading significance into 99% vs 99.9%, and are told that above some figure is perfect. its complete fiction |
| 15:58.33 | swombat | [TK]D-Fender: that is SLOW. |
| 15:58.45 | flujan | [TK]D-Fender: I already did it... |
| 15:59.17 | flujan | [TK]D-Fender: this is working right now... but it is slow... for each extension I need to update a row on the database... |
| 15:59.29 | flujan | this update have a serious cost... |
| 15:59.33 | [TK]D-Fender | how is it slow? Its REALTIME |
| 15:59.52 | [TK]D-Fender | you don't poll EVERYBODY, you just bloody well wait for INDIVIDUAL events! |
| 16:00.03 | [TK]D-Fender | psychos.... |
| 16:00.04 | swombat | ah well. that would work. |
| 16:00.11 | swombat | if it's possible |
| 16:00.13 | [TK]D-Fender | :p |
| 16:00.14 | flujan | [TK]D-Fender: hum... |
| 16:00.17 | tzafrir_home | coppice, I agree with you there |
| 16:00.32 | flujan | [TK]D-Fender: yeap it will solve the issue, but how to wait for individual events from the AMI? |
| 16:00.33 | swombat | flujan: that's another possible solution then - if you can make the poll only return rows for users which have state changes |
| 16:00.48 | swombat | and with this, i'm out. have fun :-) |
| 16:00.50 | flujan | swombat: for sure... |
| 16:00.58 | [TK]D-Fender | flujan: Open a tcp socket and sit around waiting for events. Its called PROGRAMMING <- |
| 16:01.18 | flujan | [TK]D-Fender: I am currently using the extensionstate command... |
| 16:01.27 | coppice | tzafrir_home: but people do report cases where that test gives lousy results, but there is not evidence of actual data loss |
| 16:01.43 | [TK]D-Fender | flujan: STOP |
| 16:01.46 | CCFL_Man2 | crap, this guy doesn't take paypal |
| 16:02.00 | [TK]D-Fender | flujan: that YOU asking for it. this is the opposite of what I am advising. |
| 16:02.23 | flujan | [TK]D-Fender: do you recommend creating a socket on the ami port and just process the output? |
| 16:02.40 | tzafrir_home | coppice, a short test will not catch an occasional slip |
| 16:02.53 | [TK]D-Fender | flujan: Yes, watch the events go by in real-time. Your script will stay open the WHOLE TIME. |
| 16:02.57 | tzafrir_home | may or may not catch |
| 16:03.00 | CCFL_Man2 | [TK]D-Fender: the Mark who collects western electric phones, whats his nick? |
| 16:03.19 | flujan | [TK]D-Fender: thanks for the tip... I will meditate on that... :D |
| 16:03.40 | [TK]D-Fender | CCFL_Man2: Don't recall for certain. Go check a channel archive |
| 16:03.41 | JT | programming > prayers |
| 16:03.43 | coppice | tzafrir_home: but why would people get a lousy result when they see no errors at all when faxinging without ECM |
| 16:03.51 | flujan | swombat: thanks for the help too... I will see what can I do with the socket solution |
| 16:04.01 | JT | coppice: either strom or his friend |
| 16:04.02 | swombat | good stuff. good luck! |
| 16:04.11 | [TK]D-Fender | JT : I keep a private reserve of buddist monks aside in case of urgencies regardless ;) |
| 16:04.12 | JT | s/coppice/CCFL_Man2/ |
| 16:04.15 | coppice | it certainly won't catch the occassional slip. its bogus when you use it for that fine detail |
| 16:04.25 | CCFL_Man2 | ahh |
| 16:04.27 | JT | [TK]D-Fender: always useful |
| 16:04.48 | [TK]D-Fender | JT : My karma ran over your dogma :p |
| 16:05.10 | CCFL_Man2 | JT: it's Strom_M's friend |
| 16:05.18 | JT | heh |
| 16:05.26 | davevg-btwtech | flujan: If you are somewhat ok in perl, try POE::Component::Client::Asterisk::Manager |
| 16:05.36 | Strom_M | which friend? |
| 16:05.42 | Strom_M | oh, Mark |
| 16:05.44 | CCFL_Man2 | Strom_M: Mark |
| 16:05.44 | tzafrir_home | or Asterisk::Manager |
| 16:05.50 | Strom_M | CCFL_Man2: his handle is rudholm |
| 16:05.55 | tzafrir_home | which is probably not that good |
| 16:06.10 | flujan | thanks davevg-btwtech :) |
| 16:06.13 | CCFL_Man2 | Strom_M: ahh, ok, i think i talked to him before |
| 16:06.59 | *** join/#asterisk fskrotzki (n=fskrot@host198.textwise.com) |
| 16:07.56 | *** join/#asterisk doughecka (n=doug@unaffiliated/doughecka) |
| 16:10.56 | CCFL_Man2 | Strom_M: i'm glad you're here, i need advice |
| 16:11.26 | Strom_M | ok? |
| 16:11.56 | CCFL_Man2 | i need a ringer for my green imperial WE202 |
| 16:12.00 | pots_line | D-Fender: What do you do to monitor state and presence for 100s of phones? |
| 16:12.48 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 16:13.04 | CCFL_Man2 | Strom_M: there is a 684A on ebay, i'm guessing it's better for authenticity because it's newer than the 543 ringers? |
| 16:13.42 | Strom_M | i dont know a damn thing about 202s |
| 16:13.48 | [TK]D-Fender | pots_line: Depends on poll-frequency, etc |
| 16:13.56 | pots_line | for BLF |
| 16:14.00 | CCFL_Man2 | Strom_M: neither do i :P |
| 16:14.10 | pots_line | would need to be pretty often |
| 16:14.31 | CCFL_Man2 | Strom_M: i'm guessing since mine is dated 50s i should get a newer ringer box with network |
| 16:15.31 | CCFL_Man2 | apparently the network provides sidetone |
| 16:15.43 | [TK]D-Fender | pots_line: Never had an install for someone to monitor that many. I might use an Aastra 5i series phone + 2 LCD consoles. Or a web script on 5s refresh. |
| 16:17.16 | CCFL_Man2 | Strom_M: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&rd=1&item=170149395255&ssPageName=STRK:MEWA:IT&ih=007 |
| 16:17.17 | pots_line | Just curious . . . We have several 100+ phone installations that require it. Presence stuff kind of whigs out the phones when you are monitoring that many. |
| 16:17.22 | pots_line | Going to have to go to an application instead of using the BLF on the phones |
| 16:18.18 | [TK]D-Fender | pots_line: I've only jsut recently heard of some Polycom setup whigging out on a mass-page, but no details on model/firware combo. |
| 16:18.21 | defswork | [TK]D-Fender: is there existing web scripts to show that ? I could do with one |
| 16:18.31 | JT | mmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmm |
| 16:18.33 | [TK]D-Fender | pots_line: But franly a web setup is jsut so much more readable. |
| 16:18.44 | pots_line | I agree |
| 16:18.57 | [TK]D-Fender | defswork: Dunno, never looked. I jsut code my own |
| 16:19.07 | defswork | [TK]D-Fender: open source it ! ;) |
| 16:19.08 | JT | oops |
| 16:19.19 | CCFL_Man2 | Strom_M: i'm not sure what i should do :P |
| 16:19.22 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 16:19.36 | pots_line | Polycom actually called and wanted to have us generate packet traces and debug to fix the problem. But, they didn't call it a bug because they don't officially support Asterisk. |
| 16:19.51 | pots_line | At least they are being helpful. |
| 16:20.01 | Zeeek | Hey yall, join #voip-users-conference for the um voip users conference at http://voipusersconference.org |
| 16:20.05 | [TK]D-Fender | My polycoms monitor 2 queues, 4 agents login/pause/call status, and I have a general MB script for full company presence. |
| 16:20.49 | pots_line | None of the consoles 601s monitor less than 40 phones. . . . And, they make a ton of state change noise. |
| 16:20.50 | defswork | [TK]D-Fender: would you care to give me a copy so I can see how you go about it? |
| 16:20.59 | CCFL_Man2 | pots_line: ahh, my favorite kind of phone line |
| 16:21.12 | pots_line | :-) |
| 16:21.15 | GoRK | does anyone have polycom's technical bulletin 25751 that explains the SRTP options in 2.2.0 firmware? |
| 16:21.25 | [TK]D-Fender | defswork: I just dump "show hints" via AMI and parse away nice & dirty like :) |
| 16:21.34 | defswork | oh :) |
| 16:21.38 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 16:21.41 | flujan | [TK]D-Fender: I am creating that script that connects to ami via socket an grab the information... |
| 16:21.43 | pots_line | That'll do it. |
| 16:21.49 | defswork | I did something similar last night to create phonebooks for aastra phones |
| 16:21.58 | flujan | Does AMI echos all state changes to the port or I need to ask for it? |
| 16:22.00 | [TK]D-Fender | <- the Red Green of coding..... |
| 16:22.14 | flujan | [TK]D-Fender: my program is connecting to it put just outputs nothing... :( |
| 16:22.37 | [TK]D-Fender | flujan: Your code-fu is weak :| |
| 16:22.56 | defswork | ideally you'd have a monitor script that gets the ifno only once - save multiple clients getting it multiple times |
| 16:23.01 | pots_line | AMI requires you to make a request right . . . |
| 16:23.19 | [TK]D-Fender | pots_line: My normal way, yes. |
| 16:23.21 | pots_line | So, if you loop through every 5 secs or so |
| 16:23.26 | flujan | [TK]D-Fender: but I thought I was a Jedi... :( |
| 16:23.32 | pots_line | you can pretty well keep track of state |
| 16:23.35 | Zeeek | russellb is anyone available today? |
| 16:23.51 | flujan | [TK]D-Fender: any way.. I need to send something to ami to get the new changes right? |
| 16:24.10 | [TK]D-Fender | defswork: For my polycom CSR polling I have *1* process poll, and it then generates a STATIC page for the others to load. |
| 16:24.17 | file | Zeeek: hrm! |
| 16:24.32 | [TK]D-Fender | flujan: nope. Go WIKI up the AMI to learn how it sends those events |
| 16:24.58 | defswork | [TK]D-Fender: yeah thats the right way to do it |
| 16:25.17 | russellb | Zeeek: not sure, i'm not ... |
| 16:25.23 | [TK]D-Fender | defswork: My code is ugly but functional :) |
| 16:25.29 | defswork | [TK]D-Fender: and only poll when someone is watching too :) |
| 16:25.29 | russellb | it's going to be crazy around here for the next few weeks |
| 16:25.49 | [TK]D-Fender | defswork: Mine are always watching, its on the Polycom IDLE scrren <- |
| 16:26.09 | russellb | we have astricon and then moving into our new building |
| 16:26.21 | defswork | [TK]D-Fender: do you cross reference with extension names ? |
| 16:26.23 | GoRK | if you have to make a lot of AMI connections you can also use 'astmanproxy' that reduces the load on asterisk |
| 16:26.46 | CCFL_Man2 | i like b8zs instead of ami |
| 16:26.50 | [TK]D-Fender | defswork: only on the on-demand one. the idle one has no room on screen. |
| 16:27.00 | [TK]D-Fender | defswork: I suppose I could do initials, but NAH.... |
| 16:27.07 | defswork | :) |
| 16:27.07 | [TK]D-Fender | defswork: There's only 4 of them. |
| 16:27.17 | defswork | I'll do something |
| 16:27.20 | Strom_M | CCFL_Man2: http://www.stromcarlson.com/misc/alternate_mark_inversion.png |
| 16:27.43 | russellb | o.O |
| 16:28.09 | russellb | Strom_M: http://www.russellbryant.net/DTMF_Task_Force.jpg |
| 16:28.21 | Strom_M | bahaha |
| 16:28.33 | CCFL_Man2 | Strom_M: stud |
| 16:28.58 | *** join/#asterisk yannj_fr (n=yannj@vpn.intelunix.fr) |
| 16:29.35 | CCFL_Man2 | hah |
| 16:29.50 | CCFL_Man2 | sweet |
| 16:31.19 | CCFL_Man2 | i've let that bell box go |
| 16:31.30 | CCFL_Man2 | i want a 685A |
| 16:31.37 | *** join/#asterisk CVirus (n=GoD@82.201.174.251) |
| 16:31.42 | [TK]D-Fender | Stud.... thats something that gets nailed to the wall and buried behind gyproc and only comes out when the whole house comes tumbling down, right? :) |
| 16:31.54 | *** join/#asterisk ManxPower (n=manxpowe@71-8-61-95.dhcp.leds.al.charter.com) |
| 16:32.09 | CCFL_Man2 | i'll paint the cover to match the green imperial WE202 |
| 16:32.14 | CCFL_Man2 | [TK]D-Fender: heh |
| 16:35.51 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 16:37.07 | Ryushin | I'm trouble shooting why the polycom phones aren't working on a remote network that connects to the asterisk server over a VPN. I set up a SIP softphone on the server there, and it registered with asterisk just fine. It's just that the polycom phones aren't. |
| 16:38.16 | Ryushin | The polycom log is here: http://www.pastebin.ca/706081 |
| 16:38.32 | *** join/#asterisk ikk (n=ikk@195.50.105.113) |
| 16:38.43 | Ryushin | The entries that I'm seeing are this: Registration failed User: 7070, Error Code:480 Temporarily not available |
| 16:39.10 | Ryushin | But I'm not seeing any registration attempts when I'm watching asterisk using "asterisk -vvvvvvvvvr" |
| 16:39.19 | ikk | people where would i look to see why i cant make external connections - connections within the network work fine - but connections from internet do not :( |
| 16:40.18 | Ryushin | ikk: What protocol? |
| 16:40.31 | ikk | iax2 |
| 16:40.49 | Ryushin | Try using tcpdump. |
| 16:40.53 | ikk | when trying to use zopier i just get timeout when trying to connect |
| 16:41.18 | Ryushin | Do you have the port available to connect to the internet? It is UDP. |
| 16:41.24 | *** join/#asterisk bmg505 (n=leon@196.209.180.191) |
| 16:42.07 | ikk | yes i can see the attempts via tcpdump but i just get timeouts in logs :( |
| 16:42.25 | ikk | 17:42:09.105937 IP 192.168.1.66.4569 > XXX.XXX.XXX.XXX.4570: UDP, length 12 |
| 16:43.14 | ikk | anyone on the internet 192.168.1.xx network can connect fine - but im external too ti and need to connect if possible |
| 16:43.49 | davevg-btwtech | Ryushin, can the phones route correctly to the * box via IP? |
| 16:43.52 | ManxPower | did you forward the port 4569. You need to do that when your asterisk is behind nat and you are using IAX2 |
| 16:44.52 | ikk | yes i believe it is forwarded |
| 16:46.01 | ikk | (it must be otherwise i would be able to get as far as it showing in tcpdump on that server) |
| 16:46.07 | ikk | would / would not |
| 16:46.11 | Ryushin | davevg-btwtech: Well, the thing is, that that phones are pulling their configs fine via ftp. I'm watching the vsftpd.log and that shows all the files are going there fine. |
| 16:46.38 | Ryushin | So they would have to route at least tcp traffic correctly. |
| 16:47.01 | davevg-btwtech | and the ftp server is on the other side of the vpn? |
| 16:47.18 | Ryushin | The only difference is that this is a new Cisco VPN when they got rid of the Fortigate Firewall VPN. |
| 16:47.27 | ManxPower | and 1.66 is the IP of your server? |
| 16:47.30 | Ryushin | Yea, FTP is on the same server as asterisk. |
| 16:47.57 | ikk | ManxPower, yes thats the ip of the asterisk server |
| 16:48.01 | Ryushin | The IP of the server is 172.17.127.15 |
| 16:48.40 | Ryushin | They have a windows server on the same remote network, and I installed nmap to run a scan to make sure it could see UDP 5060. |
| 16:48.42 | deeperror | are you fwd only tcp packets or udp as well? |
| 16:49.08 | Ryushin | I wonder if zoiper will fall back to tcp on sip? |
| 16:49.27 | Ryushin | If zoiper did that, then I can see what the problem is. |
| 16:49.27 | ManxPower | Asterisk does not support TCP for VoIP |
| 16:49.54 | Ryushin | Then if zoiper worked via sip, I have to think that udp is working fine. |
| 16:50.06 | ManxPower | Ryushin: what about 10000-20000/UDP? |
| 16:53.03 | *** join/#asterisk mtaht4 (n=m@239-106-62-200.enitel.net.ni) |
| 16:53.16 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 16:56.29 | |NexT| | Hi, I have 1 TE420B, all the 4 spans are configured in 0 in timing sync (zttest show 100%), but in the messages log, show this: kernel: Zaptel: Master changed to TE4/0/1 |
| 16:57.42 | Ryushin | Will the Asterisk server have 10000-20000 available to scan? |
| 16:57.56 | ManxPower | |NexT|: what field is that timing option? 1st field, 2nd field, etc |
| 16:58.00 | ManxPower | Ryushin: no idea. |
| 16:58.13 | ManxPower | but you need to make sure those ports are not blocked. |
| 16:58.17 | |NexT| | this message is showed only if any of the spans reset |
| 16:58.30 | Ryushin | Okay. I'll log into the cisco router and take a look. |
| 16:58.44 | |NexT| | span=1,0,0,ccs,hdb3 |
| 16:58.46 | davevg-btwtech | 10000-20000 will only affect rtp, not the initial SIP registration I think |
| 16:59.18 | |NexT| | the second is the timing |
| 16:59.52 | ManxPower | none of your spans come from the telco? |
| 17:01.15 | |NexT| | all spans come from PSTN |
| 17:01.44 | ManxPower | |NexT|: that is not going to work very well. |
| 17:02.08 | ManxPower | you want to get your sync source (timing) from one of the spans. |
| 17:02.49 | ManxPower | BTW, zttest does NOT show sync timing |
| 17:03.20 | ManxPower | it also does not test for sync timing |
| 17:05.21 | |NexT| | ok, if i put the span 4 for primary sync and the span 3 for secondary sync, when our span 1 (sync 0) is reset, the message "kernel: Zaptel: Master changed to TE4/0/1" appears again. |
| 17:05.58 | tru_`z24 | Is there some zaptel test I can run to test my zaptel hardwarE? |
| 17:06.33 | ManxPower | |NexT|: is it causing problems? |
| 17:06.50 | ManxPower | tru_`z24: several tests depending on what you are trying to test. |
| 17:07.02 | tru_`z24 | To make sure zaptel is operating correctly with the hardware |
| 17:07.36 | tzafrir_home | <ManxPower> BTW, zttest does NOT show sync timing |
| 17:07.38 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 17:07.55 | tzafrir_home | What do you mean? It does show the timing from the Zaptel sync master |
| 17:08.06 | |NexT| | I don't know, but my first span resets (red alarm) every 5-10 minutes |
| 17:08.49 | ManxPower | tzafrir_home: as I understand it it tests the jitter in receiving test frames to/from the card. |
| 17:09.06 | ManxPower | You should be able to run zttest without a line even plugged into the card. |
| 17:09.23 | tzafrir_home | ManxPower, it simply cmpares ticks it gets from Zaptel to system clock |
| 17:09.25 | ManxPower | and without a line to get sync from...... |
| 17:09.43 | ManxPower | tzafrir_home: so it has nothing to do with the line sync source at all |
| 17:09.47 | tzafrir_home | Because the card can also provide ticks. Just like analog cards do |
| 17:10.13 | tzafrir_home | ok |
| 17:10.31 | ManxPower | zaptel timing and T-1/E-1 timing are two different things. |
| 17:12.37 | tzafrir_home | ManxPower, hmmm... can be. Or can be the same. The zaptel clock is just the ticks of the syncing span |
| 17:13.00 | ManxPower | If you say so. |
| 17:13.16 | flujan | [TK]D-Fender: I can achive the events showing using telnet? |
| 17:13.16 | Nugget | telnet is eeeeeeevil! |
| 17:13.37 | [TK]D-Fender | flujan: Thats all the TCP connection is effectively. |
| 17:13.50 | ManxPower | according to Juggie on Asterisk-dev, zttest has NOTHING to to with T-1/E-1 timing. |
| 17:13.54 | ManxPower | It only tests IRQ stuff. |
| 17:15.42 | twisted | MANBOY |
| 17:16.07 | *** join/#asterisk saftsack (n=saftsack@pD9E04F92.dip.t-dialin.net) |
| 17:17.02 | *** join/#asterisk Sci_05 (i=Sci_05@ts.bestserversllc.net) |
| 17:17.08 | Sci_05 | afternoon all |
| 17:17.48 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 17:17.53 | tru_`z24 | ManxPower: I'm just looking for a test to make sure the card can operate, without having to have asterisk configured to test it, that way i can install zaptel, make sure it passes all tests, then move on to installing asterisk without worrying if i have zaptel configured. |
| 17:19.30 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 17:19.39 | Ryushin | Is there a way to get asterisk to log failed registrations for sip? |
| 17:20.52 | chemikk | yes |
| 17:21.56 | Ryushin | I couldn't remember what it was. It was sip set debug. Sorry for that. |
| 17:23.23 | *** part/#asterisk ming_zym (n=ming_zym@124.254.56.170) |
| 17:23.27 | tzafrir_home | tru_`z24, there are a bunch of tests in the zaptel directory. I have really no clue what they are for |
| 17:23.48 | tru_`z24 | roger. |
| 17:23.58 | tzafrir_home | If one of them is useful, please document it... |
| 17:24.07 | tru_`z24 | k |
| 17:24.17 | ManxPower | tru_`z24: there isn't a lot you can do. If zttest works, then the system is talking to the card. |
| 17:24.26 | ManxPower | not much else you can test without a real connection |
| 17:24.41 | |NexT| | ManxPower, now I change span 1 sync 1, span 2 sync 2, etc..., the span 2-3 show in the intense debug: "Got RR response to our frame" and "T203 counter expired, sending RR and scheduling T203 again" with two diferent frames, but the first span, show "Unsolicited RR with P/F bit, responding" |
| 17:24.58 | tru_`z24 | Real connection ? Do you mean real application connecting with zaptel, or real connection as in it plugged into a phoneline? |
| 17:25.25 | tru_`z24 | This is an analog card, so I can have a phone line plugged in very easily |
| 17:25.25 | ManxPower | tru_`z24: I means sending calls over the line. |
| 17:25.39 | tru_`z24 | k |
| 17:25.59 | tru_`z24 | I was having problems with zapata.conf, so i wanted to test with zaptel only |
| 17:26.10 | ManxPower | |NexT|: what actual problem are you having? |
| 17:26.12 | |NexT| | the span 1 is one carrier and the 2 to 4 is other carrier, what's the problem? |
| 17:26.16 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 17:26.27 | tru_`z24 | If i put the wrong thing in zapata, the zaptel module doesn't load in asterisk |
| 17:26.39 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 17:26.39 | |NexT| | the 1rt span reset every 5-10 minutes with a red alarm |
| 17:26.41 | ManxPower | |NexT|: failed calls, poor audio quality, lockups? |
| 17:26.51 | ManxPower | |NexT|: then call your telco and yell. |
| 17:27.08 | ManxPower | red alarm = physical cable or line issue |
| 17:27.26 | |NexT| | I called to my telca and put new cabling, reconfiguring with pri_net anf pri_cpe but the problem persist |
| 17:27.43 | ManxPower | |NexT|: then they have to come out and test the line. |
| 17:27.46 | tzafrir_home | tru_`z24, so use genzaptelconf :-) |
| 17:27.57 | |NexT| | I change the t203 timers from -1 to 600 |
| 17:28.00 | ManxPower | for more than the 1 min they usually test it for. |
| 17:28.14 | ManxPower | |NexT|: you are screwing with T-1 timers? |
| 17:28.34 | *** join/#asterisk saftsack (n=saftsack@pD9E04F92.dip.t-dialin.net) |
| 17:28.38 | |NexT| | the red alarm is only for 3 seconds |
| 17:28.45 | |NexT| | I'm using E1 |
| 17:28.48 | ManxPower | Anyway, it does not matter. A red alarm is a physical issue. |
| 17:29.06 | |NexT| | sorry |
| 17:29.24 | |NexT| | Sep 21 19:28:33 veuip2nou kernel: wct4xxp: Setting yellow alarm on span 1 |
| 17:29.35 | |NexT| | Sep 21 19:28:38 veuip2nou kernel: wct4xxp: Clearing yellow alarm on span 1 |
| 17:29.44 | ManxPower | YELLOW alarms are different. |
| 17:29.51 | |NexT| | is a yelow alarm ;) |
| 17:29.53 | ManxPower | There is 20 mins of my life I will never get back. |
| 17:30.01 | *** join/#asterisk ming_zym (n=ming_zym@124.254.56.170) |
| 17:30.29 | ManxPower | do you now have span 1 set as 1 as the sync/timing source? |
| 17:30.40 | Ryushin | This makes no sense! Zopier using sip connects just fine to the asterisk server across the VPN. The Polycom phones don't. |
| 17:31.06 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 17:31.10 | Ryushin | There is not a peep from the polycom phones over sip. FTP works fine. SIP works fine using Zoiper. |
| 17:32.02 | |NexT| | yes, now the config is this: span=1,1,0,ccs,hdb3,crc4 | span=2,2,0,ccs,hdb3 | span=3,3,0,ccs,hdb3 |
| 17:32.34 | |NexT| | I test the first span with and without crc4 acording our carrier |
| 17:33.32 | |NexT| | I test with resetinterval to never, 600 and 3600 |
| 17:36.19 | *** join/#asterisk katsuodo (n=katsuodo@pool-72-68-117-42.nwrknj.east.verizon.net) |
| 17:36.31 | |NexT| | with my old TE411P, the problem is the same |
| 17:36.59 | *** join/#asterisk saftsack (n=saftsack@pD9E04F92.dip.t-dialin.net) |
| 17:52.30 | *** join/#asterisk jbot_ (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 17:52.30 | *** topic/#asterisk is Asterisk: The Open Source Telephony Application Platform -=- Asterisk 1.4.11 (Aug. 21, 2007) -=- Join #asterisknow or #asterisk-gui for AsteriskNOW and Asterisk-GUI info -=- Join #asterisk-commits to monitor svn changes -=- Join #freepbx for freepbx/#trixbox for trixbox support. -=- 1.2 is in security maintenance mode. No non-security related bug fixes will be applied. -=- Going to AstriCon? Join us in #astricon! |
| 17:52.53 | [TK]D-Fender | MACscr: Sort fingers the guilty party directly, now doesn't it? |
| 17:53.35 | MACscr | But why would the jitter only happen on recordings and not regular talk talkie? =P |
| 17:54.02 | MACscr | Whoops, meant talkie talkie, but eh, whatever, you get my point. =P |
| 17:54.47 | tru_`z24 | What is the pseduo zap interface? |
| 17:54.48 | Kwakwa | U got a lot of hd activity going on on the machine? |
| 17:55.03 | tru_`z24 | Ran zttest, and I have a bunch fo 100%'s and a couple 99%'s |
| 17:55.30 | [TK]D-Fender | MACscr: Whats on the other end? |
| 17:55.44 | [TK]D-Fender | tru_`z24: pseudo = ztdummy |
| 17:55.59 | MACscr | [TK]D-Fender : no idea, it happens with most calls i make from my grandstream |
| 17:56.07 | tru_`z24 | k, so does that mean it's not running a test on the real hardware? |
| 17:56.36 | [TK]D-Fender | tru_`z24: what "real hardware"? |
| 17:56.38 | Kwakwa | MACscr: Could also be if you're using a different codec, voice might be fine using u/alaw when talking to someone but if you playback a .gsm file it's a bit poor. |
| 17:56.45 | tru_`z24 | a x100p clone |
| 17:56.51 | Zeeek | ACE hardware sells it |
| 17:56.53 | *** join/#asterisk lbow (n=lbow@41-195-77-250.access.uunet.co.za) |
| 17:56.55 | [TK]D-Fender | tru_`z24: not sure. |
| 17:57.19 | [TK]D-Fender | MACscr: and that doesn't answer my question at all. |
| 17:57.33 | Kwakwa | :) |
| 17:57.53 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 17:58.14 | MACscr | [TK]D-Fender : could you clarify your question a bit? |
| 17:58.31 | [TK]D-Fender | MACscr: I asked you whats on the other end of those calls.... |
| 17:59.36 | Kwakwa | MACscr: Make sure the audio you're playing back is in the same codec as codec used for the call. |
| 18:01.36 | tru_`z24 | So is it possible to run zttest on /dev/zap/ctl instead of pseudo? |
| 18:02.17 | MACscr | [TK]D-Fender : what do you mean, whats on the other end? Its a recording, could be music, voices, whatever. I have no idea what equipment/software they use |
| 18:04.28 | [TK]D-Fender | MACscr: you don't know whats on the other end of "phone-to-phone" calls? Can't come up with an answer like "digium digital/analog card", "ITSP using codec XXX", etc? |
| 18:05.11 | *** join/#asterisk Shido6 (n=shido6@74-130-56-203.dhcp.insightbb.com) |
| 18:07.35 | *** join/#asterisk scastano (n=scastano@72.165.82.2) |
| 18:08.17 | scastano | crazy question on IAX2 trunks |
| 18:08.20 | scastano | who's got some time?! |
| 18:08.21 | scastano | haha |
| 18:08.25 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 18:08.28 | Katty | :> |
| 18:08.53 | Kwakwa | Ask ur question scastano and if someone has time they'll answer :) |
| 18:09.01 | Katty | i have a question!!! |
| 18:09.09 | Katty | how many ferrets does it take to screw in a lightbulb? |
| 18:09.34 | Kwakwa | none, they bite their owners fingers until they do it |
| 18:09.42 | Katty | lies. |
| 18:09.47 | rob0 | None, because weasels have eaten our phone system? |
| 18:09.49 | Katty | ferrets do not bite unless overly wound up and excited. |
| 18:09.56 | scastano | hahahaha |
| 18:09.57 | Katty | rob0: EXACTLY! |
| 18:09.58 | Katty | <PROTECTED> |
| 18:10.08 | Katty | they've been carried away by monkies... |
| 18:10.12 | scastano | ok... so... i've got IAX trunks setup between 2 boxes.... |
| 18:10.21 | Katty | we've been transfering this at&t rep to that wav file |
| 18:10.24 | Katty | so fun. |
| 18:10.27 | scastano | calls from my house to my office always go through...... |
| 18:10.35 | rob0 | Only a * user would get the humor. |
| 18:10.37 | scastano | call from the office back out to my house seem to die after a while |
| 18:10.41 | Katty | rob0: clearly. |
| 18:10.44 | scastano | the extensions never ring |
| 18:10.45 | Katty | rob0: now that the rain is gone |
| 18:10.50 | scastano | it just goes straight to voicemail |
| 18:10.55 | Kwakwa | U must only know know nice ferrets Katty, the ones I know will do all kinds of terrible things. |
| 18:11.06 | ManxPower | scastano: you have trunk=yes? |
| 18:11.11 | scastano | but the RTP stream is obviously going back and for cause I get my home voicemail recording |
| 18:11.19 | scastano | where? |
| 18:11.22 | ManxPower | scastano: IAX2 does not use RTP> |
| 18:11.24 | Kwakwa | iax.conf |
| 18:11.32 | scastano | well... I mean.. I see the packets go |
| 18:11.47 | Kwakwa | iax show debug ? |
| 18:11.52 | scastano | the call comes through, the voice is transmitted, but it doesn't ever try to ring the extensions |
| 18:11.55 | Kwakwa | iax2 show debug rather |
| 18:11.56 | ManxPower | scastano: well you can't have an iax2 trunk unless you have trunk=yes, unless you were using the term to mean "IAX2 connection", in which case STOP USING THE WRONG TERMS |
| 18:12.01 | scastano | did that.... didn't see anything funny |
| 18:12.08 | scastano | no no |
| 18:12.11 | scastano | trunks |
| 18:12.17 | scastano | one side is asterisk 1.2 |
| 18:12.24 | ManxPower | an IAX2 trunk has a very specific meaning. |
| 18:12.24 | scastano | the other is trixbox running the same |
| 18:12.26 | Kwakwa | Have you set up the right context? |
| 18:12.29 | scastano | yup |
| 18:12.34 | ManxPower | scastano: You don't call connections "trunks" in Asterisk |
| 18:12.34 | Kwakwa | scastano: are you sure? |
| 18:12.35 | scastano | calls work... for a while |
| 18:12.38 | ManxPower | never. ever. |
| 18:12.46 | scastano | yes.... |
| 18:13.04 | Kwakwa | scastano: Do they die or does audio only go 1 way? |
| 18:13.04 | scastano | if I call 0201 right now at work.. extension 201 rings at home |
| 18:13.11 | scastano | in about half hour when I do that.... |
| 18:13.13 | scastano | no ring |
| 18:13.21 | scastano | but I do get extension 201's voicemail |
| 18:13.56 | scastano | and not sure about the audio... when it dies again |
| 18:14.01 | scastano | I'll try to record a voicemail |
| 18:14.02 | scastano | haha |
| 18:14.24 | Kwakwa | scastano: You're kind of confusing me, "the call comes through, the voice is transmitted, but it doesn't ever try to ring the extensions" |
| 18:14.33 | scastano | exactly.... |
| 18:14.35 | scastano | kinda! |
| 18:14.53 | ManxPower | scastano: that statement makes no sense. |
| 18:14.56 | Kwakwa | If it isn't ringing the extension, how do you know its not working? |
| 18:15.02 | scastano | my trixbox says the call is "established" |
| 18:15.07 | scastano | but..... |
| 18:15.11 | scastano | the extension never rings |
| 18:15.13 | ManxPower | scastano: trixbox is not supported here. |
| 18:15.13 | scastano | I just get voicemail |
| 18:15.16 | scastano | I know |
| 18:15.20 | scastano | the trixbox works fine |
| 18:15.21 | ManxPower | scastano: use the Asterisk CLI |
| 18:15.28 | scastano | I'm having a problem with the asterisk box |
| 18:15.30 | *** join/#asterisk katsuodo (n=katsuodo@pool-72-76-11-31.nwrknj.east.verizon.net) |
| 18:15.52 | ManxPower | so put the CLI output of a failed call on pastebin.ca |
| 18:16.06 | katsuodo | Hello Everyone |
| 18:16.07 | Kwakwa | ManxPower: Is trixbox just a GUI shoved over *? |
| 18:16.17 | scastano | basically... yes |
| 18:16.38 | scastano | its asterisk with freepbx preinstalled |
| 18:16.39 | Kwakwa | ahh, get in the CLI then.. that'll help u solve the problem |
| 18:16.41 | scastano | but its the same core |
| 18:16.44 | scastano | I'm in the CLI |
| 18:16.50 | ManxPower | Kwakwa: no. Trixbox is a large complex THING that turns Asterisk config files into a maze of spaghetti dialplan, macros, and AGIs. |
| 18:16.50 | scastano | thats where I'm seeing this |
| 18:17.00 | ManxPower | you can't even debug the thing on the CLI because it spews out so much crap. |
| 18:17.14 | Kwakwa | :/ |
| 18:17.32 | scastano | well the thing is.... even in a packet capture outside the box |
| 18:17.33 | ManxPower | scastano: I'm still waiting for that pastebin |
| 18:17.43 | scastano | what do you want a pastbin of |
| 18:17.43 | scastano | ? |
| 18:17.50 | scastano | from which side? |
| 18:17.52 | ManxPower | THE CLI OUTPUT OF A FAILED CALL. |
| 18:17.58 | scastano | thats the thing |
| 18:18.03 | scastano | the call doesn't "fail" |
| 18:18.05 | pjz | anyone have recommendations for a good linux SIP softphone? |
| 18:18.06 | tzafrir_home | Kwakwa, no. Trixbox is a PBX built on top of Asterisk and other stuff. |
| 18:18.11 | ManxPower | scastano: if you paste the tricbox CLI I'll feed you to the aligators. |
| 18:18.13 | scastano | the cli says its going fine |
| 18:18.16 | Kwakwa | Capturing packets etc... seems a bit over kill mate :) |
| 18:18.22 | scastano | haha |
| 18:18.24 | katsuodo | I inherited a asterisk 1.2 box with tdm400p card and trying to dial from the console and I am receiving no such extension '4004' in context local first time introduce to asterisk |
| 18:18.28 | ManxPower | scastano: *sigh* Did it occur to you that I might see something you might miss? |
| 18:18.32 | tzafrir_home | Asterisk is more of a toolkit for building PBX systems than a PBX |
| 18:18.34 | scastano | very possible |
| 18:18.44 | scastano | hang on... I'll see what I can grab |
| 18:19.01 | scastano | go figure... its working right now |
| 18:19.13 | ManxPower | scastano: sounds like a NAT problem to me. |
| 18:19.22 | scastano | thats exatly what I thought |
| 18:19.22 | katsuodo | Any suggestions? I am also reading the asterisk blue and white asterisk bible |
| 18:19.37 | GoRK | Does anyone have Polycom's Technical Bulletin 25751 that details the SRTP configuration options for polycom phones? |
| 18:19.41 | Kwakwa | scastano: I think your ambiguity helped heal it :) |
| 18:19.46 | ManxPower | and you know what the standard fix for random failed calls in a nat enviroment, right? |
| 18:19.47 | scastano | but I enabled DMZ setting on my linksys at home |
| 18:19.57 | scastano | ManxPower: no idea? |
| 18:20.18 | ManxPower | scastano: qualify=yes will keep enough traffic on the link to keep the router from closing the NAT translation |
| 18:20.24 | scastano | got that |
| 18:20.37 | ManxPower | then I have no more suggetions. |
| 18:20.41 | scastano | even dropped qualifyfreqok=5000 n there |
| 18:20.59 | scastano | just to ensure the nat stays open |
| 18:21.08 | *** join/#asterisk drwelby (n=mpfister@mail.enplan.com) |
| 18:21.13 | scastano | and like I said... the stream gets from one place to another... I hear the voicemail |
| 18:21.20 | scastano | but the extension doesn't ring |
| 18:21.33 | scastano | its almost like its not "ACK"ing |
| 18:22.55 | Kwakwa | and I'm assuming u also have nat=yes in the config? |
| 18:23.23 | scastano | no |
| 18:23.25 | scastano | :( |
| 18:24.41 | Kwakwa | :/ |
| 18:25.26 | scastano | but without it... it works |
| 18:25.31 | scastano | I'm gonna through it in there anyway |
| 18:25.53 | Kwakwa | I'd like to think there was method to adding that option :p |
| 18:26.13 | MACscr | [TK]D-Fender : i am using a SIP to PTSN provider (callcentric) for making my calls. When you say other end, I am thinking about who receives the call. Which is why I said I have no idea what their equipment is. |
| 18:26.16 | scastano | I didn't know it needed it with IAX2 |
| 18:26.19 | scastano | I thought that was the point |
| 18:26.51 | [TK]D-Fender | MACscr: ok, how is GXP -> PAP2? PAP2 = fine all the time? |
| 18:26.54 | ManxPower | you can add nat=yes to the iax.conf, but it is not a valid option and will be ignored. |
| 18:27.51 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
| 18:27.53 | scastano | thats what I thought |
| 18:27.59 | scastano | IAX2 doesn't care about nats |
| 18:28.01 | Kwakwa | haha, good point :b |
| 18:28.35 | Kwakwa | I gave up with IAX2 in favour of SIP because of multiple call issues, think I'm mixing the two in my head *hangs head in shame* |
| 18:29.18 | MACscr | [TK]D-Fender : my pap2 is just another extension of mine. I meant that its fine when i make calls with it. Calling from GXP to PAP2 is not going to result in any recordings, thus i could not test anything like that |
| 18:30.02 | [TK]D-Fender | MACscr: ok, your description is getting spotty. is it ONLY GXP -> * prompts that is the problem? |
| 18:32.11 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 18:33.45 | MACscr | [TK]D-Fender : ok, obviously my GXP is using my * to make any calls, thus why I am in this channel. When I call a business with my GXP, which would obviously be GXP -> * -> CallCentric -> Business (no idea what they are using, everyone would be different). Most businesses have some type of recording whe you call them or are on hold. I get a lot of "crackle" (poort quality) during these recordings. If i would call with a regular (ptsn) phone or even my pap2 |
| 18:33.45 | MACscr | my * box), i do not get this type of poor quality. |
| 18:35.20 | ManxPower | The SIPura line of devices frequently default to 0.30 (30ms) voice packet size. Asterisk uses 0.20 (20ms). Change the SIPura box to use 20ms packet size. |
| 18:35.51 | ManxPower | perhaps the GXP (may it rot in hell) has the same issue. |
| 18:37.17 | [TK]D-Fender | MACscr: so basically the GXP is shitty on all calls then? |
| 18:37.46 | MACscr | [TK]D-Fender : only on recordings, thats why its odd |
| 18:38.18 | MACscr | But your right, the GXP is a shitty phoen in general =P |
| 18:38.22 | [TK]D-Fender | Audio is audio at that point... the GXP blows. Period |
| 18:38.41 | *** join/#asterisk adorah (n=Michael@87.69.57.246.cable.012.net.il) |
| 18:40.27 | *** join/#asterisk atomicd (n=AtomicDa@74-206-0-80.static-ip.m.telepacific.net) |
| 18:43.32 | *** join/#asterisk GaryH (n=chatzill@wallace.garysoft.co.uk) |
| 18:51.22 | *** join/#asterisk GreggB (n=GreggB@66.206.86.107) |
| 18:53.19 | atomicd | My boss says I get to go to Astricon! :-) |
| 18:55.15 | the_lalelu | atomicd: your boss should talk to my boss. ;) |
| 18:56.18 | atomicd | I'm going on the cheap though... My Dad live in Scottsdale so I don't need a hotel. Plus I'm driving from Anaheim, CA. |
| 18:56.23 | *** join/#asterisk funxion (n=nunya@63.214.236.169) |
| 18:59.23 | the_lalelu | atomicd: well, i'm from hamburg, germany. *g* |
| 18:59.58 | atomicd | the_lalelu: That would be a long drive... :-) |
| 19:00.19 | *** join/#asterisk Yourname` (n=IM@unaffiliated/yourname/x-837320) |
| 19:00.20 | hmmhesays | ejabberd is driving me insane |
| 19:00.31 | the_lalelu | i guess your right. :D |
| 19:00.46 | mog | hmmhesays, whats wrong with ejabberd |
| 19:00.52 | *** join/#asterisk lbow (n=lbow@41-195-77-250.access.uunet.co.za) |
| 19:02.16 | hmmhesays | well I can't get my registration to work |
| 19:02.48 | mog | do you have it allowed in your acl? |
| 19:03.14 | hmmhesays | I have everything open, but it seems to be trying to use ssl to auth my client, and I can't find in my config file where it says to do that |
| 19:04.12 | hmmhesays | {s2s_use_starttls, false}. |
| 19:05.39 | hmmhesays | also where you configure the listen ports, I if I comment out the tls lines and uncomment the "use these if no tls support" I get no log files at all |
| 19:11.22 | *** join/#asterisk PorkSale (n=barney@c-98-202-51-107.hsd1.ut.comcast.net) |
| 19:13.07 | PorkSale | I just learned about asterisk and I'm interested in setting up a system for up to 8 lines. I'm trying to figure out what an FXS and FXO module is for. |
| 19:13.59 | hmmhesays | you plug a phone in to fxs modules and you plug a phone line into fxo modules |
| 19:14.50 | jsmith | (And then configure the signalling just opposite of that) |
| 19:15.48 | PorkSale | http://www.digium.com/en/wheretobuy/digiumdirect/productview.php?product_code=1TDM844BF-01 |
| 19:16.07 | PorkSale | so that would allow me to have 4 lines connected for incoming or outgoing calls? |
| 19:17.14 | PorkSale | maybe I don't understand asterisk as I thought. Do you need lines going into the fxo modules if it is all voip? |
| 19:17.27 | jsmith | Nope... the FXO ports are just for connecting to analog lines |
| 19:17.36 | jsmith | If you're doing all VoIP, you don't need any cards |
| 19:17.46 | PorkSale | ok, so I would just get the 8 port with 2 quad fxs modules |
| 19:18.04 | jsmith | If you do have a card though, then you can mix/match/bridge calls across VoIP and analog |
| 19:19.44 | PorkSale | ok, but if I just have the fxs module I can still mix/match/bridge calls all on the VoIP right? Like I can forward a call to another location though the VoIP |
| 19:20.24 | PorkSale | just not from VoIP to analog lines I've got coming in |
| 19:20.30 | [TK]D-Fender | At 8 channels I would suggest looking into a partial PRI |
| 19:21.58 | pots_line | Anyone ever use isymphony? |
| 19:22.09 | pots_line | op panel |
| 19:23.12 | jsmith | Poincare: Exactly |
| 19:23.23 | jsmith | [TK]D-Fender: His eight ports would be FXS ports, not FXO ports |
| 19:23.36 | jsmith | [TK]D-Fender: Oh, and a temporary asteriskdocs.org is back up |
| 19:23.48 | jsmith | (while I re-design the site, again) |
| 19:23.52 | [TK]D-Fender | jsmith: Enough that all the links work? |
| 19:23.56 | jsmith | Yes |
| 19:24.04 | [TK]D-Fender | ok, I'll de-list mine then |
| 19:24.45 | jsmith | No, that's fine |
| 19:24.55 | [TK]D-Fender | mirror* |
| 19:25.03 | *** join/#asterisk tomcats (n=fgonzale@189.157.152.170) |
| 19:25.44 | *** join/#asterisk zeromobile (n=zero@64.78.21.135) |
| 19:26.11 | tomcats | What could cause asterisk to only enable sound coming from the agent to the caller, meaning that the caller CAN hear the agent but not the other way? |
| 19:26.43 | Wonka | NAT? |
| 19:27.01 | [TK]D-Fender | entirely |
| 19:27.08 | tomcats | ohh and this only happens sometimes.. like 1 out of 10 calls the agent recieves... |
| 19:27.15 | zerohalo | DID provider issue? |
| 19:27.45 | tomcats | we checke dthe local loop T1 acces and they say that everything is fine... |
| 19:28.03 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 19:28.25 | tomcats | a full log doesn't show any errors too... |
| 19:28.34 | *** join/#asterisk jtoy (n=jtoy@mail.backchannelmedia.com) |
| 19:28.46 | jsmith | tomcats: Did you check the output of "rtp debug"? |
| 19:28.51 | jtoy | can sending emails of voicemails be turned on/off through the manager? |
| 19:29.01 | *** part/#asterisk dasuberdavid (i=david@nat/digium/x-e6a5d9c3adf33b2d) |
| 19:29.11 | tomcats | what's rtp debug? |
| 19:29.13 | jsmith | tomcats: That'll tell you one way or the other if RTP is getting through (assuming this is SIP, of course) |
| 19:29.39 | jsmith | jtoy: Well, yes, but it's convoluted. You'd have to use the UpdateConfig action to rewrite voicemail.conf |
| 19:29.52 | jsmith | jtoy: So it's *possible*, just not so easy |
| 19:30.09 | jtoy | is that way safe? is there a better solution? |
| 19:31.07 | *** part/#asterisk zeromobile (n=zero@64.78.21.135) |
| 19:31.24 | tomcats | jsmith: ohh interesting.. is there a way to filter rtp debug output? |
| 19:31.53 | jsmith | tomcats: Not really... |
| 19:32.02 | jsmith | jtoy: It's safe |
| 19:32.16 | jtoy | is realtime voicemail a better option? |
| 19:33.11 | *** join/#asterisk defsmac (n=andy@defsdoor.gotadsl.co.uk) |
| 19:33.22 | PorkSale | are there any modules or options that'll send out a transcript of your voicemail to your email? |
| 19:33.39 | defsmac | PorkSale, trixbox does that out of the box |
| 19:33.57 | defsmac | so you should be able to find out how |
| 19:34.20 | defsmac | part of freepbx I guess |
| 19:34.33 | *** join/#asterisk r00tlz (n=Cero@190.41.12.173) |
| 19:34.56 | PorkSale | is trixbox built on top of asterisk or they're totally seperate? |
| 19:35.08 | defsmac | its a distribution |
| 19:35.09 | scastano | defsamc: I don't think it does it out of the box, there's a few modules to figure |
| 19:35.31 | scastano | and trixbox is CentOS with asterisk and freepbx already running on it |
| 19:35.34 | scastano | with a few other goodies |
| 19:35.44 | scastano | like hylafax and asterisk recording interface |
| 19:35.56 | defsmac | ok - by transcript do you mean voice recognition to text ? if so - then no :) |
| 19:36.14 | pepse | has asterisk gui had any improvements? |
| 19:36.21 | pepse | like, is it fully functional yet |
| 19:36.38 | *** join/#asterisk ivrc (n=chatzill@74.228.54.150) |
| 19:36.40 | tomcats | jsmith: rtp debug show packets in and out... any other idea? |
| 19:37.25 | pepse | it made some -craaazy- configs when i tried it.. but it's problem was some bugs that made a lot of stuff not work |
| 19:38.56 | jsmith | tomcats: If it shows packets coming and going, then everything is fine on Asterisk's side... is there a firewall between the Asterisk box and the phone that might be blocking the audio? |
| 19:40.43 | *** join/#asterisk blinkbot2k (n=me@c-75-69-77-42.hsd1.vt.comcast.net) |
| 19:42.26 | tomcats | jsmith: I am suspecting is not a connection problem since the problem happens randomly.. I am having trouble to replicate the issue, althought the problem keeps happening... |
| 19:45.13 | PorkSale | Any idea what a partial PRI costs? I'm looking at hosting our service through vitelity. Our phones aren't really that busy most of the time so I'm thinking it'll be less than $200 per month total to do it this way. |
| 19:45.55 | jsmith | PorkSale: It all depends on your location and your telco |
| 19:47.15 | scastano | Vitelitys prices are hard to beat |
| 19:47.26 | scastano | I'm in DC in a really well lit area |
| 19:47.40 | scastano | a partial PRI for me, 12 channels is around 280 |
| 19:49.27 | scastano | and also... I used Vitelity here for rollovers when my 2 PRI's fill up |
| 19:49.43 | *** join/#asterisk thunter (n=Tee@rawb.fttp.xmission.com) |
| 19:49.57 | *** join/#asterisk Defraz (n=t0tal@208.98.184.140) |
| 19:50.36 | thunter | how do I send a # key to a caller instead of # acting like the transfer key? |
| 19:52.09 | *** join/#asterisk r00tlz (n=Cero@190.41.12.173) |
| 19:54.49 | ManxPower | thunter: don't put t/T/w/W on the Dial lone |
| 19:54.49 | defsmac | scastano, a month ? |
| 19:54.49 | ManxPower | lne |
| 19:55.23 | ManxPower | My last quote for a 11 channel PRI was $800/month |
| 19:55.32 | defsmac | wtf ? |
| 19:55.41 | defsmac | in uk it a tenner a channel |
| 19:55.50 | [TK]D-Fender | ManxPower: Kinda sucky |
| 19:55.51 | defsmac | so that would be 80 a month |
| 19:55.59 | scastano | and yes... per month |
| 19:56.05 | defsmac | minimum 8 channels |
| 19:56.18 | ManxPower | defsmac: I've never heard of PRI that cheap. |
| 19:56.45 | ManxPower | Generally PRIs are priced based on the loop + channels. |
| 19:57.14 | JerJer | ManxPower: unless the loop is a few feet long :D |
| 19:57.32 | JerJer | and your workin with a friendly xLEC |
| 19:57.43 | defsmac | maybe I'm misunderstanding what you are meaning by PRI ? |
| 19:57.50 | Wonka | wtf? we paid about 400EUR/month for a PRI with 30 B-channels |
| 19:57.58 | ManxPower | defsmac: T-1 or E-1 PRI |
| 19:58.22 | scastano | Wonka: where the hell are you?! |
| 19:58.22 | ManxPower | Wonka: Prices vary by quite a bit. |
| 19:58.24 | JerJer | back a year or so ago we could get real T-1 PRI loops for $125 a month - not any more though :( |
| 19:58.26 | scastano | I wanna live there! :-P |
| 19:58.28 | Wonka | scastano: northern germany |
| 19:58.28 | defsmac | ManxPower, well an E1 in the uk is about a 10 a channel - minimum of 8 |
| 19:58.43 | ManxPower | defsmac: no loop charge? |
| 19:58.44 | scastano | E1 is actually 30 channels |
| 19:58.49 | scastano | T1 is 24 |
| 19:58.58 | scastano | each way you loose one channel for the D channel |
| 19:59.09 | defsmac | ManxPower, not that I am aware of |
| 19:59.10 | ManxPower | JerJer: there no facilities CLECs in my area |
| 19:59.20 | [TK]D-Fender | E1=31, T1=24, E1 PRI=30, T1 PRI=23 <--- |
| 19:59.30 | Wonka | scastano: E1 is 32 timeslots. one ist used for sync, one for D channel, leaves 30 for B channels |
| 19:59.34 | scastano | doh |
| 19:59.35 | scastano | you're right |
| 19:59.39 | scastano | I was off by 1 |
| 19:59.40 | scastano | hahaha |
| 19:59.56 | defsmac | D channel is channel 16 on BT |
| 20:00.08 | defsmac | I learned that the hard way on my first install |
| 20:00.15 | JerJer | ManxPower: find a couple mil and start one |
| 20:00.18 | JerJer | ....or not |
| 20:00.31 | ManxPower | JerJer: Only a couple of mil? |
| 20:00.47 | scastano | ManxPower: I've get $8 man... .I'm in |
| 20:00.52 | scastano | but I wanna be a partner! |
| 20:00.53 | scastano | :-P |
| 20:00.58 | JerJer | if you had a market a couple mil would be a start |
| 20:01.16 | [TK]D-Fender | scastano: You'll be a very small "part" of PARTner ;) |
| 20:01.58 | scastano | hahahahaha |
| 20:02.05 | ManxPower | JerJer: I shutdown my micro isp, telco, cableco about a month ago. |
| 20:02.34 | ManxPower | "irreconcilable differences" |
| 20:03.06 | J4k3 | wtf, cable? :) |
| 20:03.33 | ManxPower | J4k3: *nod* |
| 20:03.46 | JerJer | ManxPower: that sucks |
| 20:03.55 | J4k3 | but then I thought better of it |
| 20:04.04 | J4k3 | ManxPower: buy an old cable modem head end and just back-feed your node? :) |
| 20:04.11 | J4k3 | err back-fed? |
| 20:04.19 | ManxPower | J4k3: I would need a NOC first. |
| 20:04.23 | JerJer | i've got about 9 customer radios online here in my neck of the woods |
| 20:04.35 | ManxPower | those "irreconcilable differences" means I don't have a NOC anymore |
| 20:04.41 | J4k3 | I've got around ~70 subs |
| 20:05.13 | JerJer | 3 of which are family (but they still pay :) |
| 20:05.15 | ManxPower | I'll revisit the idea in 18 - 24 months |
| 20:05.39 | J4k3 | ManxPower: my "noc" is a wide hallway in my home, with a water heater closet (2Mx2M) holding most of the gear |
| 20:05.55 | J4k3 | and a 45m (150') tower in my side yard. |
| 20:06.51 | J4k3 | well, I call it a 'wide hallway'... its got open doorways on both ends feeding other rooms... its maybe 5Mx6M |
| 20:06.57 | scastano | hahaha |
| 20:10.31 | JerJer | my 'NOC' is an equipment box mounted to a 100' tower out back, on the big hill |
| 20:10.38 | JerJer | powered by solar :) |
| 20:13.44 | hmmhesays | I got the msn transport working but this stupid client doesn't support nicknames |
| 20:14.37 | *** join/#asterisk lbow (n=lbow@41-195-77-250.access.uunet.co.za) |
| 20:15.55 | *** part/#asterisk thunter (n=Tee@rawb.fttp.xmission.com) |
| 20:19.06 | FXOL | anyone alive that might have some ideas on a problem with a service provider? |
| 20:19.34 | FXOL | We were using an IAX trunk to connect to them... and calls always connected, but with alot of breakup in the call |
| 20:19.50 | FXOL | They had us switch to SIP, and call quality was pefect |
| 20:20.05 | FXOL | however... system keeps acting like connection drops betweeen us |
| 20:20.09 | FXOL | outbound calls wont work |
| 20:20.16 | FXOL | and inbound calls just ring and don't hit * |
| 20:20.26 | FXOL | it's Random tho when it works |
| 20:22.10 | [TK]D-Fender | FXOL: sounsd like a NAT problem. |
| 20:22.25 | FXOL | why would it work 50/50 tho? |
| 20:23.35 | tru_`z24 | when i do zap show channels at the asterisk cli, it only shows the pseudo interface.... |
| 20:23.59 | ManxPower | tru_`z24: then you don't have any channels configured |
| 20:24.15 | ManxPower | FXOL: because nat routers close |
| 20:24.21 | ManxPower | "inactive" connections |
| 20:24.25 | FXOL | intersting |
| 20:24.32 | FXOL | so how might this get resolved? And why only w/ SIP? |
| 20:24.33 | tru_`z24 | I have the channel configured in zaptel, so let me see what is going on in the zapata.conf then |
| 20:25.05 | ManxPower | tru_`z24: /etc/zaptel.conf is the CARD config file /etc/asterisk/zapata.conf is the ASTERISK config file. |
| 20:25.13 | tru_`z24 | right, the card config file is right |
| 20:25.23 | ManxPower | tru_`z24: the asterisk config is not |
| 20:25.30 | [TK]D-Fender | FXOL: ... |
| 20:25.31 | [TK]D-Fender | ~sipnat |
| 20:25.32 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 20:25.33 | [TK]D-Fender | ^^^^^^^^^^^^^^^^ |
| 20:25.38 | ManxPower | FXOL: IAX is more chatty than SIP and could easily keep the NAT reanslation active. |
| 20:25.42 | tru_`z24 | right, is there a similar configuration program like genzaptelconf ? |
| 20:25.54 | ManxPower | tru_`z24: I've never used that program. |
| 20:25.59 | ManxPower | config them manually |
| 20:29.22 | *** join/#asterisk Schumie (i=SteveWri@cpc2-rdng2-0-0-cust382.winn.cable.ntl.com) |
| 20:29.32 | FXOL | (ManxPower): I assume this would still be a problem if I put on a DMZ? |
| 20:29.51 | ManxPower | FXOL: that would totally depend on your router. |
| 20:30.04 | FXOL | Netgear WR614 :P |
| 20:30.12 | FXOL | WRG614 I mean |
| 20:30.13 | FXOL | :P |
| 20:30.17 | ManxPower | I'm not a magical router fairy. |
| 20:30.21 | FXOL | I know ;P |
| 20:31.02 | tru_`z24 | Got it now :-) |
| 20:32.19 | tzafrir_home | tru_`z24, zapconf :-) |
| 20:34.56 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 20:35.15 | tru_`z24 | well, genzaptelconf generated a zapata.conf.channels file |
| 20:35.23 | tru_`z24 | i just appended it to my current zapata.conf file and it works |
| 20:35.42 | *** join/#asterisk ivanfm (n=ivanfm@c906b486.virtua.com.br) |
| 20:36.41 | tzafrir_home | zapata-channels.conf , but yes |
| 20:38.21 | *** join/#asterisk guillote_GNU (n=bancaria@host35.201-253-17.telecom.net.ar) |
| 20:42.45 | lirakis | night all |
| 20:42.47 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 20:43.59 | *** join/#asterisk copantl (n=copantl@63.161.232.126) |
| 20:44.24 | copantl | any body know howto install phpagi on asterisk? |
| 20:45.07 | ManxPower | it doesn't come with any docs? |
| 20:46.59 | *** join/#asterisk inv_arp[work] (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 20:47.24 | copantl | i saw it but i dont see a place to install it |
| 20:47.32 | copantl | is not a module? |
| 20:47.47 | ManxPower | IT is not an asterisk module. |
| 20:47.54 | ManxPower | I dunno if it is a PHP "module" or not. |
| 20:48.30 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:48.39 | copantl | but i dont see any make install or something like |
| 20:50.14 | *** join/#asterisk jimmysolis (n=jimmy@190.41.82.1) |
| 20:51.15 | jimmysolis | Hello i cant compile the zaptel 1.4.5 =( |
| 20:51.24 | jimmysolis | who know that: make: *** No rule to make target `install-inlcude', needed by `install-programs'. Stop. |
| 20:52.00 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 20:52.55 | [TK]D-Fender | jimmysolis, there is a typo in the makefile. |
| 20:53.17 | [TK]D-Fender | jimmysolis, install-inlcude should be install-include' |
| 20:53.33 | [TK]D-Fender | jimmysolis, load it up in a text editor and fix it and "make install" should work |
| 20:53.53 | jimmysolis | i use debian and i dont found this package "install-include" |
| 20:54.18 | [TK]D-Fender | jimmysolis, there is nothing to find. |
| 20:54.24 | jimmysolis | ok |
| 20:54.35 | [TK]D-Fender | jimmysolis, there is an ERROR in the make file. a simple tex-edit corrects this |
| 20:54.42 | [TK]D-Fender | text-edit |
| 20:54.46 | jimmysolis | sorry but i dont understand english very well |
| 20:54.49 | jimmysolis | Gracias =) |
| 20:55.00 | [TK]D-Fender | jimmysolis, hopefully you just "got it" |
| 20:55.26 | *** join/#asterisk dlynes_laptop (n=dlynes@s142-179-114-141.bc.hsia.telus.net) |
| 20:56.21 | dlynes_laptop | Is there a way to make it so that when asterisk picks a line from a zaptel channelgroup to call out on, that it starts picking from line 8, instead of line 1, on an 8 line card? |
| 20:56.46 | tzafrir_home | jimmysolis, right. Get 1.4.5.1 |
| 20:56.46 | dlynes_laptop | That way the user isn't picking up a ringing line when they try to dial out? |
| 20:56.47 | jsmith | dlynes_laptop: Use a capital G |
| 20:56.55 | dlynes_laptop | jsmith: ah, ok |
| 20:56.56 | jsmith | dlynes_laptop: Zap/G1 instead of Zap/g1 |
| 20:56.58 | dlynes_laptop | jsmith: thanks |
| 20:57.03 | jsmith | dlynes_laptop: No problem |
| 20:57.05 | tzafrir_home | It fixed Makefile typos and such |
| 20:57.51 | dlynes_laptop | jsmith: didn't even know the feature existed...thought 'G' and 'g' were the same thing :) |
| 20:58.14 | tzafrir_home | dlynes_laptop, or you can use r or R for round-robin |
| 20:58.22 | jimmysolis | thanks guys now work :) |
| 20:59.03 | jsmith | Work? |
| 20:59.19 | dlynes_laptop | tzafrir_home: what would round robin be good for? |
| 20:59.35 | jsmith | jimmysolis: Trabajo? No sabemos trabejar! |
| 21:00.12 | tzafrir_home | Make things less predictable? Make sure all channels are used? |
| 21:01.21 | jimmysolis | aka si sabemos trabajar nos negrean jajaja |
| 21:01.54 | *** part/#asterisk zerohalo (n=zeroHalo@pool-72-70-79-233.bstnma.east.verizon.net) |
| 21:03.28 | Katty | oh |
| 21:03.30 | Katty | i'm still in here |
| 21:03.35 | Katty | crazy!! |
| 21:03.39 | Katty | also, hi |
| 21:04.39 | hmmhesays | heh |
| 21:04.40 | hmmhesays | that is crazy |
| 21:05.09 | [TK]D-Fender | Katty, Mew. |
| 21:06.12 | dlynes_laptop | tzafrir_home: but if you have calls coming in from line 1 up, and calls going out from line 8 down, does that not use the lines up? |
| 21:06.30 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 21:06.40 | hmmhesays | I really wish the jabber transports had voice |
| 21:06.57 | dlynes_laptop | tzafrir: I mean I can see round robin for call agents, but that's for extensions, not lines |
| 21:08.18 | jsmith | dlynes_laptop: It's so that if you have a bad channel, your calls don't keep getting stuck on that one bad channel |
| 21:08.18 | Katty | [TK]D-Fender: mew |
| 21:08.24 | Katty | [TK]D-Fender: started a shiny new website. |
| 21:08.56 | Katty | [TK]D-Fender: with ampache, and gallery2, and phpbb |
| 21:09.07 | [TK]D-Fender | dlynes_laptop, That almost never matters. any telco hunt group will work around whichever channels are in use and pick a free one. |
| 21:09.14 | *** join/#asterisk amarzouk2 (n=chatzill@217.54.201.152) |
| 21:09.22 | Katty | [TK]D-Fender: it r purrty. |
| 21:09.24 | [TK]D-Fender | Katty, oooohhh |
| 21:09.30 | [TK]D-Fender | Katty, Link meh! |
| 21:09.37 | Katty | k |
| 21:10.30 | Wonka | purrty? can has pusseh? |
| 21:10.41 | Katty | ^_- |
| 21:11.04 | Wonka | r cute! |
| 21:11.08 | [TK]D-Fender | Wonka, Sure thing.... "Willy" :p |
| 21:11.13 | Katty | oh. |
| 21:11.25 | Katty | i pulled that from a little picture of a kitty that said 'i r not squeeze toy' |
| 21:11.51 | Katty | http://icanhascheezburger.files.wordpress.com/2007/04/i-r-not-squeezy-toy.jpg |
| 21:11.56 | Wonka | icanhascheezburger.com has many of these |
| 21:11.56 | Katty | and so i've been using that ever since :P |
| 21:11.58 | Wonka | :) |
| 21:12.32 | Katty | (= |
| 21:12.39 | amarzouk2 | Hi, I am having a problem with get_data , it does not accurately get the input digits! what can I do to insure more accuracy? |
| 21:12.51 | *** join/#asterisk metfan2007 (n=metfan20@189.135.156.38) |
| 21:13.23 | Katty | amarzouk2: threaten it with a stick. |
| 21:13.32 | Katty | amarzouk2: and tell it no pudding! |
| 21:13.58 | amarzouk2 | :) tried that already no use :( |
| 21:14.22 | Katty | :< |
| 21:15.36 | Wonka | and no jelly donut either. |
| 21:16.33 | atomicd | As I'm registering for Astricon, there's a field labeled: "Enter Discount Code" Anyone know of a discount code? I'm always up for saving my company a couple of bucks... |
| 21:17.17 | *** join/#asterisk h0 (n=Fakhir@unaffiliated/fakhir) |
| 21:17.26 | Qwell | atomicd: check digium.com, I think there's one on the front page...or was? |
| 21:17.56 | BadPacket | Digium-Astricon-2007 or VoicePulse-Astricon-2007 |
| 21:18.33 | Qwell | http://www.digium.com/en/mediacenter/events/viewevent/55 |
| 21:18.35 | Qwell | yeah |
| 21:18.55 | atomicd | You guys rock...thanks! |
| 21:19.24 | *** join/#asterisk yannj_fr (n=yannj@vpn.intelunix.fr) |
| 21:22.16 | tomcats | what could cause rtp packets to be able to get from the phone to the pbx but not the other way arround on a LAN setup? |
| 21:22.43 | atomicd | Just saved me $110 (actually, my company) (20% of $550) |
| 21:22.59 | dlynes_laptop | [TK]D-Fender: ah....well, the whole reasonn i'm doing this is so that when someone picks up a phone, and then dials a number, they're not connected to a call that happens to come in at the same time they finish typing in their phone number |
| 21:23.13 | dlynes_laptop | [TK]D-Fender: and ultimately dialing hte phone number in that person's ear |
| 21:23.32 | [TK]D-Fender | dlynes_laptop, then yeah your best odds are ascending incoming, decending outgoing |
| 21:23.48 | dlynes_laptop | [TK]D-Fender: exactly...that's the way the nortel phones usually do it, too |
| 21:25.32 | metfan2007 | hi All, is there any way to add more G729 licenses to a Asterisk that already has a few licenses registered??? |
| 21:26.02 | Qwell | metfan2007: yes, just drop in another license file, and it'll pick it up |
| 21:26.16 | *** part/#asterisk jtoy (n=jtoy@mail.backchannelmedia.com) |
| 21:26.22 | Qwell | atomicd: so, that means you're buying us beer? on the company, of course :p |
| 21:26.43 | dlynes_laptop | metfan2007: same way |
| 21:26.53 | atomicd | Qwell: Drinks are on Reliant Manufacturing! |
| 21:27.00 | metfan2007 | Qwell: drop?? there? :S do you mean to run againt the registration utility? |
| 21:27.17 | Qwell | metfan2007: well, you get the new license (by running register), and it puts it where it needs to be |
| 21:27.26 | Qwell | atomicd: I am, of course, only joking :) |
| 21:27.40 | perd | where does the voicemailmain app store the passwords if a user changes it? |
| 21:27.44 | metfan2007 | thanks! |
| 21:28.18 | putnopvut | perd: in the voicemail.conf file |
| 21:28.27 | perd | really.. hrm |
| 21:28.34 | putnopvut | or users.conf if that's where you have your mailboxes defined. |
| 21:28.41 | perd | the passwords in there dont work, and i have no users.conf |
| 21:31.09 | perd | haha oh, i'm retarded is why.. |
| 21:31.43 | [TK]D-Fender | perd, the first step is admitting you have a problem.... |
| 21:31.52 | perd | i sure do have a lot of them. |
| 21:32.13 | theHub | /bye Have a nice weekend, everyone. |
| 21:33.11 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 21:33.11 | *** mode/#asterisk [+o anthm] by ChanServ |
| 21:33.51 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 21:33.51 | *** mode/#asterisk [+o anthm] by ChanServ |
| 21:34.52 | _Sam-- | data23: i had a very pleasant experience with ebay pandora purchase -- worked mint, shipped quick. no problems. |
| 21:34.54 | _Sam-- | er |
| 21:35.00 | _Sam-- | wrong win |
| 21:35.48 | [TK]D-Fender | _Sam--, All stable? |
| 21:36.11 | _Sam-- | so far so good, thank you for checking. System uptime: 1 day, 1 hour, 56 minutes, 18 seconds |
| 21:38.26 | [TK]D-Fender | _Sam--, great... |
| 21:39.03 | _Sam-- | thank you again for your help. |
| 21:39.47 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 21:39.51 | [TK]D-Fender | _Sam--, np |
| 21:44.34 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 21:44.44 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 21:44.44 | *** mode/#asterisk [+o anthm] by ChanServ |
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| 21:54.56 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 21:56.53 | riddlebox | hrmm is there a way to get Asterisk to detect a hangup from a TDM fxo? right now if someone hangs up, it rings the full 20 sec I set then goes to voicemail? |
| 21:57.24 | Ryushin | Are there alternative voices for Asterisk beside the girl that can be bought/downloaded? |
| 21:58.34 | *** join/#asterisk zeromobile (n=zero@64.78.21.135) |
| 21:59.20 | fakhir | Ryushin, there are two on the digium site -> http://www.digium.com/en/products/voice/ |
| 22:00.39 | jsmith | riddlebox: Yes, if your line has remote disconnect supervision on it |
| 22:01.03 | jsmith | riddlebox: If it does, the telco will temporarily open the loop within about 6 or 7 seconds of the remote party hanging up |
| 22:01.03 | riddlebox | jsmith, right now I doubt it, as I have a line provided by the cable company |
| 22:01.24 | jsmith | Ryushin: Yes... check the wiki |
| 22:01.46 | Katty | i guess he's getting fast food |
| 22:04.24 | hmmhesays | fast food, fast heart attack? |
| 22:05.19 | Katty | clearly |
| 22:07.08 | perd | mmm vmail.cgi is nice |
| 22:07.18 | perd | just thought i'd throw that out there. |
| 22:13.39 | *** join/#asterisk barrys (n=barrys@ool-4577407d.dyn.optonline.net) |
| 22:14.49 | *** part/#asterisk barrys (n=barrys@ool-4577407d.dyn.optonline.net) |
| 22:15.27 | hmmhesays | i'll throw you out there |
| 22:15.35 | *** join/#asterisk __freedom__lover (n=eduardo@clipper.provale.com.br) |
| 22:15.43 | perd | you're too kind! |
| 22:17.03 | *** join/#asterisk Zipper_32 (n=None@d205-250-2-107.bchsia.telus.net) |
| 22:20.33 | tomcats | for some reason app_queue is bridging the call halfways with the agent.. sound only comes from the agent but not from the caller... any ideas? |
| 22:26.51 | *** join/#asterisk disgrntld81 (n=asdf@CPE-75-81-155-105.wi.res.rr.com) |
| 22:27.59 | disgrntld81 | total newb needs direction... i want asterisk automatically add 1 + area code if i only dial 7 digits, what should i google to find out? |
| 22:28.56 | gremzoid | extensions.conf variables... |
| 22:29.03 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 22:29.04 | disgrntld81 | sweet, thanks |
| 22:29.04 | gremzoid | 1${EXTEN} ? |
| 22:29.14 | perd | GotoIf |
| 22:29.29 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 22:29.32 | gremzoid | yea that to |
| 22:29.52 | perd | if it's 7 chars, do this, otherwise, do that |
| 22:30.02 | perd | there's probably a better way to do it but i'm no guru |
| 22:30.12 | tzafrir_home | Ryushin, http://www.voip-info.org/wiki/view/Asterisk+sound+files+international |
| 22:30.20 | disgrntld81 | great, thanks! |
| 22:31.07 | perd | http://www.voip-info.org/wiki-Asterisk+cmd+GotoIf disgrntl |
| 22:32.24 | disgrntld81 | oh cool, good link |
| 22:32.41 | Ryushin | Thanks for the links. Thats going to get me started. I appreciate it. |
| 22:36.38 | riddlebox | jsmith, my telco isnt providing remote disconnect |
| 22:36.53 | GreggB | Ryushin: there's a company giving out sound files. |
| 22:38.10 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 22:38.25 | GreggB | Ryushin: Ahh, found them. http://www.voicevector.com/ These folks are great to work with, and you can get an entire voice pack for free from them (checkout the link on their homepage) |
| 22:38.55 | GreggB | Ryushin: We've been happily using the voice pack since early this year... |
| 22:39.31 | perd | has anyone here integrated asterisk with a ccm 3.3 server, i have a quick dumb question if you dont mind |
| 22:39.56 | sevard | Everytime somebody mentions a "new service" they've been using for "quite a while" and been "pretty happy with it" i always assume they're the CEO pushing their drabble on IRC. |
| 22:40.12 | perd | haha |
| 22:40.41 | gremzoid | eww pommy voices |
| 22:44.53 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 22:46.31 | GreggB | sevard: you talking about me? heh, no, I'm a lowly SysAdmin running an Asterisk box for my employer (who is a medical records management company).... now that I think of it, the voice pak did have one issue; The American Female Asterisk voice pack's "vm-options" file reads an incorrect list of options (apparently for *1.2). If anyone wants to hear more about this, just lemme know - otherwise, just email Robert Christian (robert@voicevector.com), and he sen |
| 22:49.01 | sevard | he sen |
| 22:49.09 | sevard | t me a packet of cookie chrisps for christmas |
| 22:50.00 | *** join/#asterisk thieums (n=Mathieu@rny93-4-82-231-54-139.fbx.proxad.net) |
| 22:50.06 | GreggB | I dont think you can add those to the wiki |
| 22:55.49 | hmmhesays | new avenged sevenfold song is pretty good |
| 22:57.51 | *** join/#asterisk tsurko (n=tsurko@78.90.100.214) |
| 22:57.56 | *** join/#asterisk alephcom (n=darren@h66-112-187-16.mcsnet.ca) |
| 23:01.18 | *** join/#asterisk alephcom (n=darren@h66-112-187-16.mcsnet.ca) |
| 23:01.27 | alephcom | greetings |
| 23:01.54 | alephcom | Anybody aware of an ftp server that will accept mixed case usernames? |
| 23:02.04 | alephcom | I'm trying to use the default usernames in the polycoms. |
| 23:03.18 | [TK]D-Fender | alephcom, I'd bet vsftpd does as well as proftpd |
| 23:03.26 | hmmhesays | I bet most do |
| 23:03.27 | Qwell | any linux ftpd, no doubt |
| 23:03.39 | alephcom | That's what I thought to be it doesn't appear that way. |
| 23:03.41 | hmmhesays | and probably most windows ftpd's |
| 23:03.44 | alephcom | Maybe I have something screwed up. |
| 23:03.58 | Qwell | and if it doesn't accept them with mixed case, it'll just lowercase everything anyways |
| 23:06.23 | dlynes_laptop | If I have say ten extensions, and each extension has eight accounts, is there an efficient way to pick a free account on every phone, when a call comes in? |
| 23:06.52 | [TK]D-Fender | dlynes_laptop, how can an "extension" have an "account"? |
| 23:06.55 | dlynes_laptop | I'm finding IsChanAvail() and other means are either not reliable, or they're too slow |
| 23:07.10 | dlynes_laptop | [TK]D-Fender: each 'line' appearance on the extension is a separate account |
| 23:07.39 | [TK]D-Fender | dlynes_laptop, are they intended to be completely different identities? |
| 23:08.04 | [TK]D-Fender | dlynes_laptop, And stop calling a PHONE an EXTENSION. |
| 23:09.13 | alephcom | Thanks, I'll snoop around and try a few more things. |
| 23:09.22 | dlynes_laptop | [TK]D-Fender: no...just different 'lines' on the same phone |
| 23:10.05 | [TK]D-Fender | dlynes_laptop, please clarify your dangerously vague terminology..... |
| 23:10.41 | dlynes_laptop | [TK]D-Fender: I've got a call coming in, I need to send it to all ten phones, whether those phones are on a call, or not |
| 23:11.09 | perd | terror alert! |
| 23:11.18 | dlynes_laptop | [TK]D-Fender: each phone is capable of handling up to nine simultaneous calls, but I only have 8 peers and users defined for each |
| 23:11.34 | [TK]D-Fender | dlynes_laptop, what models? |
| 23:11.37 | *** join/#asterisk zeromobile (n=zero@64.78.21.135) |
| 23:11.39 | dlynes_laptop | [TK]D-Fender: 9133i's |
| 23:11.42 | *** join/#asterisk metfan2007 (n=metfan20@189.135.156.38) |
| 23:11.44 | dlynes_laptop | [TK]D-Fender: Aastra's |
| 23:12.05 | [TK]D-Fender | dlynes_laptop, And why all running multiple regs on the same phone if they are not infact "unique"? |
| 23:14.00 | dlynes_laptop | [TK]D-Fender: how else can I handle eight simultaneous calls on one phone? |
| 23:14.24 | [TK]D-Fender | dlynes_laptop, 1 reg spanning all keys. they should not be run as sperate reg's |
| 23:14.33 | *** join/#asterisk grndslm (n=grndslm@24-116-87-97.cpe.cableone.net) |
| 23:14.50 | dlynes_laptop | [TK]D-Fender: I remember trying that a year ago or so on Aastra's, and it never worked |
| 23:14.55 | dlynes_laptop | [TK]D-Fender: I've never tried it since |
| 23:15.14 | dlynes_laptop | [TK]D-Fender: is that working on aastra's now? |
| 23:15.25 | [TK]D-Fender | dlynes_laptop, shove the same auth in each "line" (what a bastardized use of the word they do...) and it should count them as *1* reg and span calls across those keys |
| 23:15.40 | dlynes_laptop | [TK]D-Fender: just the 'auth'? |
| 23:15.41 | [TK]D-Fender | dlynes_laptop, its how I had it on my God-aweful 57i CT |
| 23:15.56 | [TK]D-Fender | dlynes_laptop, set 1-3 up identically and see it span... |
| 23:16.00 | dlynes_laptop | [TK]D-Fender: and the 'auth' matches the '[auth]', or the username=auth? |
| 23:16.10 | [TK]D-Fender | dlynes_laptop, No, nothing to do with *. |
| 23:16.15 | [TK]D-Fender | dlynes_laptop, on the PHONE'S config |
| 23:16.18 | dlynes_laptop | [TK]D-Fender: ok...lemme try |
| 23:16.19 | metfan2007 | hi all, how can I tell Asterisk to always try to start a trunk if the link is down? After some time Asterisk stops to try to look for the other Asterisk box... |
| 23:16.20 | dlynes_laptop | [TK]D-Fender: thanks |
| 23:16.36 | [TK]D-Fender | dlynes_laptop, go into each "lines" definition and set them IDENTICALLY and test to see that it spans. |
| 23:16.47 | [TK]D-Fender | dlynes_laptop, place a call out and have another call come in. |
| 23:17.00 | [TK]D-Fender | dlynes_laptop, manual "channel hunting" is raging BS |
| 23:17.31 | [TK]D-Fender | dlynes_laptop, unless its your INTENTION for each "line" to have seperate identities |
| 23:18.12 | [TK]D-Fender | metfan2007, is the remote side on a fixed IP / host? |
| 23:18.24 | metfan2007 | I see a few messages in the CLI saying that chan_iax2.c:7238 socket_read: Peer 'pedregal' is now TOO LAGGED (2028 ms)! |
| 23:19.44 | metfan2007 | Fender: both Asterisk boxes has dynamic IPs with Dyndns domains |
| 23:20.27 | [TK]D-Fender | metfan2007, then set "host=[thehostnameforthebox]", and "qualify=no" and forget about registering between the two. There's no need. |
| 23:20.33 | metfan2007 | Fender: after some "LAGGED", "UNREACHABLE" and "is now REACHABLE" messages Asterisk stops to reach again the other box |
| 23:21.16 | [TK]D-Fender | metfan2007, this way when you dial out it'll just dial and not care if it think they're up or not. It'll try and if it succeeds, then it succeeds, if not, just try again |
| 23:21.16 | metfan2007 | Fender: currently I have to do a manual iax reload to Asterisk start to reach the other host |
| 23:21.34 | *** join/#asterisk agx (n=badpengu@81-174-46-120.dynamic.ngi.it) |
| 23:21.36 | metfan2007 | Fender: ok ok |
| 23:22.01 | [TK]D-Fender | those lag warnings only come if you tell * to MONITOR (and hence CARE) the remote host. You should not BOTHER caring in this case |
| 23:22.21 | *** part/#asterisk agx (n=badpengu@81-174-46-120.dynamic.ngi.it) |
| 23:24.00 | metfan2007 | Fender: so do I need to change "qualify=no" on both sides? they are acting as a trunk |
| 23:24.10 | metfan2007 | Fender: with "trunk=yes" |
| 23:25.42 | [TK]D-Fender | metfan2007, shouldn't matter |
| 23:26.02 | [TK]D-Fender | metfan2007, once a channel is up it should stil aggregate the calls |
| 23:26.21 | metfan2007 | Fender: ok, I'll try, so what's "qualify=yes" used for? |
| 23:26.48 | [TK]D-Fender | metfan2007, as a way for * to chose to GIVE UP trying to call a dead host |
| 23:27.10 | [TK]D-Fender | metfan2007, and helps as a NAT keep-alive, etc |
| 23:27.28 | metfan2007 | Fender: and "maxregexpire" and "minregexpire"?? |
| 23:27.28 | [TK]D-Fender | metfan2007, none of which applies to your situation. |
| 23:27.56 | [TK]D-Fender | metfan2007, you don't even NEED to register. you have a host name. as long as thats valid there is no point in registering at all |
| 23:28.10 | metfan2007 | Fender: ok ok |
| 23:30.32 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 23:31.13 | *** join/#asterisk ming_zym (n=ming_zym@124.254.52.195) |
| 23:31.30 | [TK]D-Fender | metfan2007, the only thing "register"ing does is inform the server to which IP it should send calls. This doesn't change how the calls themselves are authed, and as you always know where to go it serves no purpose |
| 23:36.22 | *** join/#asterisk r00tlz (n=Cero@190.41.12.173) |
| 23:37.19 | metfan2007 | exit |
| 23:42.45 | *** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
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| 23:47.30 | *** mode/#asterisk [+o anthm] by ChanServ |
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| 23:56.50 | *** part/#asterisk atomicd (n=AtomicDa@74-206-0-80.static-ip.m.telepacific.net) |
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