| 00:00.23 | *** join/#asterisk murdock_ut (n=chatzill@c-98-202-154-226.hsd1.ut.comcast.net) |
| 00:02.17 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:08.18 | *** join/#asterisk mosty (n=mosty@ppp191-34.static.internode.on.net) |
| 00:09.43 | *** join/#asterisk korihor (n=korihor@190.78.32.60) |
| 00:17.57 | jeev | if my asterisk server has client configuration for phones to connect and also register lines, it's called a server and client ? |
| 00:19.05 | mosty | if you like |
| 00:20.08 | jeev | mosty, do you know how hosted solutions work? |
| 00:20.50 | mosty | yes. "hosted" just means "on someone else's server" |
| 00:21.05 | jeev | ok |
| 00:21.25 | *** join/#asterisk coppice (n=chatzill@61.157.17.210.dyn.pacific.net.hk) |
| 00:21.47 | jeev | so do you agree that some forms of that will be: multiple phones behind a firewall/nat network on lets say some type of lower quality broadband.. and they will be provisioned to connect to lets say a server at a datacenter, correct? |
| 00:22.27 | *** join/#asterisk irisht (n=irisht@cpe-70-122-11-142.austin.res.rr.com) |
| 00:25.01 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 00:25.13 | mosty | sure, why not? |
| 00:27.39 | jeev | so if you're behind nat and you connect to the asterisk server/client that has the register line and also your login credentials to allow your phone on the broadband line, there shouldn't be an issue.. correct? |
| 00:28.27 | AndyML | mind if I chime in? I'm curious... - that would depend pretty heavily on the NAT, and the config for the phone (in asterisk AND on the phone.) |
| 00:28.40 | jeev | hmm |
| 00:28.50 | AndyML | but it should work, yeah |
| 00:28.50 | jeev | all these providers do it.. |
| 00:28.55 | AndyML | exactly |
| 00:29.10 | AndyML | we do it |
| 00:29.12 | AndyML | it works |
| 00:30.07 | jeev | do you mind if i message you? |
| 00:30.12 | AndyML | not at all |
| 00:31.22 | mosty | jeev, you typically don't register asterisk to your phones, it's the other way around |
| 00:34.09 | jeev | yea, that's what i meant |
| 00:35.09 | mosty | well yes, phones register to asterisk, asterisk can deliver calls to a SIP/IAX/PSTN provider however you like |
| 00:35.20 | drmessano | http://www.slash7.com/pages/vampires |
| 00:35.36 | jeev | yes |
| 00:35.46 | jeev | http://www.psystar.com/hosted.jpg pretty much. |
| 00:35.48 | jeev | right ? |
| 00:35.49 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 00:37.58 | *** part/#asterisk `paul (n=aldee@125.252.68.126) |
| 00:46.24 | jeev | ~book |
| 00:46.24 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
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| 00:59.15 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 01:05.30 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 01:23.13 | drmessano | jeev: Want to buy a PDF copy of the book? |
| 01:23.21 | drmessano | Downloadable and reusable? |
| 01:24.52 | trev | my goal is to have asterisk use a voip provider to receive/place calls. when asterisk picks up a sip connection it should select an operator from a list of remote #'s and place the call through my voip provider |
| 01:25.49 | trev | so if I set up my inbound,outbound sip stuff correctly--am I just basically creating a dialplan? I can't find a good howto of doing call-transfers in the * book |
| 01:26.34 | mosty | trev: call transfers are performed by the endpoints usually. eg by hitting the transfer button on your phone |
| 01:26.34 | jeev | call transfers? |
| 01:26.37 | jeev | like to different extension ? |
| 01:26.51 | trev | jeev: not extensions |
| 01:26.54 | mosty | trev: but it sounds like you just need to use the Dial application in your dialplan |
| 01:27.16 | trev | my voip providers enables me to receive calls from 1-800-trev through SIP, then I want to have a queue of operators sitting in memory somewhere |
| 01:27.39 | trev | when asterisk gets a call it should pick next avail operator and place a call to a remote # through the voip provider (for termination) |
| 01:27.45 | AndyML | sounds like he needs to use the queue application |
| 01:27.46 | trev | I don't think a remote # is technically an extension is it? |
| 01:28.01 | trev | AndyML: ok any keywords are welcome |
| 01:28.03 | jeev | ah, i'm not even doing queues. i dont need queues. |
| 01:28.08 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 01:28.18 | jeev | i just have mine ring like 10 people, whoeer answers it answers it, likely that 9 people are free at once. |
| 01:28.21 | jeev | it's not an inbound office anyway |
| 01:28.29 | AndyML | you can just string |
| 01:28.31 | AndyML | oops |
| 01:28.36 | trev | mosty: so when you create a dialplan you just put normal telephone #'s in, no sip-specific stuff if you have your sip.conf set up right? |
| 01:28.42 | AndyML | you can just string extensions together in a dial - for example... |
| 01:28.55 | mosty | trev, you should read the section on the dialplan in the book... |
| 01:29.04 | trev | trying to... |
| 01:29.14 | AndyML | Dial(sip/username@host/15551212&sip/username@host/15551213&sip/username@host/15551214) |
| 01:29.39 | [TK]D-Fender | trev: using PSTN #'s for "agents" in a call-out queue is serious work. |
| 01:30.03 | AndyML | the queue might be more elegant but the Dial would work. D-Fender is right |
| 01:30.20 | trev | [TK]D-Fender: how serious? |
| 01:30.20 | drmessano | jeev: are you giving out Asterisk help? |
| 01:30.30 | drmessano | jeev: Shouldnt yours be WORKING first? |
| 01:30.43 | [TK]D-Fender | trev: You have to ask the agent for confirmation, because what happens if their VOICEMAIL answers? Or you target a csell thats out of range? |
| 01:30.48 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-4c6592a6e842ee65) |
| 01:31.13 | [TK]D-Fender | trev: Go learn the basics first then branch out. |
| 01:31.22 | trev | yeah I should start with something easier |
| 01:31.34 | [TK]D-Fender | trev: Or better yet, find a consultant |
| 01:31.51 | trev | I am the consultant, this is an experiment in creating a distributed call-center thing |
| 01:32.29 | trev | it may be that it works better for the agents to just use voip clients |
| 01:32.42 | trev | but then there are NAT issues, so I'm considering PSTN |
| 01:32.56 | drmessano | NAT is fine, you just fix it |
| 01:33.10 | trev | well the agents might not be that competent at fixing router settings |
| 01:33.27 | [TK]D-Fender | trev: If you're worried about NAT, and don't know how to deal with PSTN agents you sound thoroughly unqualified for this project |
| 01:34.02 | trev | hah do you feel better? |
| 01:34.09 | *** join/#asterisk moy (n=moy@189.169.61.29) |
| 01:34.42 | [TK]D-Fender | trev: Only in knowing I have no need to feel guilty about losing all inclination to advise further. |
| 01:34.44 | *** join/#asterisk salzh (n=chatzill@58.247.193.104) |
| 01:34.51 | trev | noted |
| 01:35.07 | [TK]D-Fender | trev: keep treading Noah ;) |
| 01:36.02 | drmessano | Are trev and jeev related? |
| 01:36.33 | AndyML | they're on as different IPs but they smell similar |
| 01:37.06 | trev | no we just have the most similar nicks of any two people here |
| 01:37.24 | drmessano | "How much do you know about SIP?" |
| 01:37.26 | *** join/#asterisk Nasra (n=Nasra@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 01:37.28 | drmessano | "NAT a clue" |
| 01:37.37 | [TK]D-Fender | drmessano: Each of them have 4 letters in their nicks and the last to are even IDENTICAL! |
| 01:37.43 | AndyML | whoa |
| 01:37.44 | drmessano | lol |
| 01:37.51 | [TK]D-Fender | two* |
| 01:38.11 | [TK]D-Fender | plays a drum-crash to drmessano's las ZING |
| 01:38.42 | drmessano | lol |
| 01:38.55 | *** join/#asterisk macros73 (n=cs@c-67-163-224-69.hsd1.pa.comcast.net) |
| 01:43.54 | [TK]D-Fender | goes to depopulate the planet again... |
| 01:43.58 | drmessano | Jeev: Regardless of what I say, I admire the fact that someone with a career in pet store management is giving asterisk a shot. |
| 01:45.16 | [TK]D-Fender | drmessano: I remember a waiter once. A waiter with big dreams and budgets.... |
| 01:45.17 | drmessano | So, sorry to "dog" you out, I "cat" tell you enough how sorry I am and "iguana" make it up to you. |
| 01:45.30 | drmessano | lol |
| 01:45.56 | jeev | :) |
| 01:47.00 | drmessano | jjev: If you and I worked in the same datacenter, I would be honored to have you manage a completely different VLAN. |
| 01:48.18 | drmessano | Matter of fact, if you need some one on one asterisk help, call me at home |
| 01:48.33 | drmessano | sip/101@127.0.0.1 |
| 01:48.46 | Qwell | drmessano: don't be giving out my personal number |
| 01:48.47 | drmessano | I'm always here to help |
| 01:48.53 | AndyML | or mine |
| 01:49.31 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 01:50.00 | [TK]D-Fender | drmessano: There's no place like 127.0.0.1 |
| 01:50.07 | [TK]D-Fender | clicks his red shoes |
| 01:53.58 | *** part/#asterisk mosty (n=mosty@ppp191-34.static.internode.on.net) |
| 01:54.06 | *** join/#asterisk mosty (n=mosty@ppp191-34.static.internode.on.net) |
| 01:54.22 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-809095423b9fdb64) |
| 01:55.15 | *** part/#asterisk trevAway (n=laa@c-67-160-226-244.hsd1.ca.comcast.net) |
| 01:56.09 | *** join/#asterisk murdock_ut (n=chatzill@c-98-202-154-226.hsd1.ut.comcast.net) |
| 01:57.40 | *** join/#asterisk DarkRift (n=dark@70.48.126.41) |
| 01:58.39 | drmessano | Pop quiz |
| 01:58.49 | drmessano | Little twirp wrecks my ferrari |
| 01:58.53 | drmessano | Do I: |
| 01:59.09 | drmessano | 1) Beat him.. $125,000 worth |
| 01:59.34 | drmessano | 2) Beat him.. (Sorry, high interest loan) |
| 01:59.50 | drmessano | 3) Make him set up Asterisk for me |
| 02:00.19 | [TK]D-Fender | 4) Realize that "Wait... only in my dreams could I get my hands on a Ferrari, legally or otherwise...." and WAKE UP :p |
| 02:00.50 | drmessano | jeev is another mrdigital |
| 02:01.18 | jeev | heh |
| 02:01.27 | jeev | i had a ferrari, then it broke. |
| 02:01.52 | [TK]D-Fender | jeev: Not to fear, Matchbox has a 5 year warranty! |
| 02:01.55 | jeev | maybe my fault... :/ |
| 02:02.09 | jeev | actually, it was expensive to repair, so the car was just sold. |
| 02:02.12 | jeev | 575M |
| 02:02.21 | jeev | i only broke the carbon fiber lip |
| 02:02.23 | jeev | was my friends. |
| 02:03.14 | *** join/#asterisk gones (n=gones@203.193.37.251) |
| 02:03.30 | drmessano | I am finding this story hard to believe |
| 02:03.40 | drmessano | The part about you having friends.. |
| 02:05.11 | [TK]D-Fender | ouch |
| 02:05.27 | drmessano | I know :( |
| 02:05.49 | jeev | Fender, i'm trying an iax trunk. i have asterisk in house and at the datacenter. |
| 02:05.51 | *** join/#asterisk johnantypas (n=jantypas@mail.antypas.net) |
| 02:06.27 | johnantypas | Good evening folks.... Some success (thank you all), but I fix one thing -- I break another (that's m job....) |
| 02:06.48 | johnantypas | Per the suggestions, I removed the nf_sip modules from my Linux router and Asterisk no longer has dropouts. |
| 02:07.07 | johnantypas | However, now a remote ATA connects, but it has no audio on calls.... (I hate UVerse) |
| 02:07.24 | [TK]D-Fender | ~sipnat |
| 02:07.25 | 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 |
| 02:07.43 | [TK]D-Fender | johnantypas: ^ |
| 02:08.05 | johnantypas | Will check it out thanks..... Any tricks I should know about the wonderful UVerse router? |
| 02:08.12 | jeev | fender, any suggestions as to how my incoming dial plan would be? accepting the phone call on server 1 and having to pass it to server 2 to make it ring the local extensions.. |
| 02:08.25 | jeev | johnantypas, no idea.. haven't used their stuff. |
| 02:08.38 | [TK]D-Fender | johnantypas: Never heard of them. Normally you shouldn't have to care. Does it do VoIP as well? |
| 02:08.40 | drmessano | jeev: Why dont you just install Trixbox? |
| 02:09.02 | AndyML | the question of the day. then you can ask these questions at #trixbox |
| 02:09.04 | [TK]D-Fender | jeev: its your dialplan, do whatever you want. |
| 02:09.15 | johnantypas | Yes. AT&T UVerse. |
| 02:09.25 | johnantypas | Just like FIOS, only slower. |
| 02:09.55 | AndyML | how fast? |
| 02:10.11 | [TK]D-Fender | johnantypas: was that "yes it does VoIP"? |
| 02:10.24 | jeev | any suggestion on how it's done ? |
| 02:10.37 | johnantypas | Yes, but AT&T's own VOIP. Not real SIP. |
| 02:10.45 | jeev | how should i do it, IAX extension it ? |
| 02:10.49 | [TK]D-Fender | johnantypas: SIP or NOT? |
| 02:10.58 | [TK]D-Fender | johnantypas: what's "fake SIP"? |
| 02:11.03 | johnantypas | No SIP. |
| 02:11.34 | [TK]D-Fender | johnantypas: nthen with any luck it won't interfere. Also here's hoping your provider doesn't as well. For everything else follow the guide. |
| 02:11.42 | drmessano | [TK]D-Fender: TELL ME.. TELL ME NOW |
| 02:11.54 | AndyML | jeev: put an entry in iax.conf on both machines. |
| 02:12.10 | johnantypas | The box claims to provide IP, IPTV and VOIP, but the VOIP is non-SIP as the box is also an ATA. I ahve a PAP-2T attached on the inside of the device. The box itself does NAT. |
| 02:12.13 | jeev | [TK]D-Fender, if incoming calls go to server 1, server one accepts it.. how should i have it routed to server 2? IAX also? |
| 02:12.19 | jeev | AndyML, how so? i have iax working just fine. |
| 02:12.35 | drmessano | wget http://www.aclue.com//aclue.tar.gz |
| 02:12.37 | johnantypas | So, I have PAP2T (inside NAT) -- AT&T UVerse NAT Router et al ---- Asterisk server (outside NAT) |
| 02:12.38 | AndyML | there are tutorials for this stuff. |
| 02:13.15 | [TK]D-Fender | jeev: do you WANT to send this call to another server? Make up your own damn mind. You don't even seem to know what you WANT, let alone a clue to do it. |
| 02:13.26 | [TK]D-Fender | johnantypas: Ok, should be fine then. |
| 02:13.56 | johnantypas | I assume I must tell the PAP-2T it is going through a NAT router, and tell my Asterisk server that extension is NAT. |
| 02:14.20 | [TK]D-Fender | ~sipnat |
| 02:14.21 | 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 |
| 02:14.22 | [TK]D-Fender | ^^^^^^^^ |
| 02:14.29 | [TK]D-Fender | johnantypas: Follow. The. Guide. |
| 02:14.32 | jeev | hm |
| 02:17.49 | *** join/#asterisk pkunkra (n=chris@cpe-74-73-8-115.nyc.res.rr.com) |
| 02:18.21 | jeev | Fender, the internal server is where the sip clients are on. they call through internal, iax takes it to the datacenter, datacenter makes the call. for incoming, i'm lost as to how i'm gonna do it, incoming is caught in the datacenter but eventually needs to ring at the office... |
| 02:18.35 | jeev | phones - asterisk - iax through internet to datacenter - asterisk |
| 02:18.55 | [TK]D-Fender | jeev: Get a clue. Its all just calls. |
| 02:19.03 | *** join/#asterisk C4colo (n=DJpyro@66.185.107.193) |
| 02:19.09 | C4colo | gooten morgan |
| 02:19.17 | C4colo | hola a todos |
| 02:19.18 | jeev | guzuntite C4colo. |
| 02:19.27 | C4colo | Hello Everyone! |
| 02:19.40 | C4colo | tis a beautiful day somewhere in the world |
| 02:19.47 | jeev | so should i send the call through iax and set up a dial plan to send it via sip to the phone ? |
| 02:19.57 | C4colo | sure |
| 02:20.16 | mosty | jeev, incoming is the same as outgoing, just in the opposite order |
| 02:20.24 | C4colo | I usually use the protocol my ITSP requires for the trunk and the protocol required by the phone for the phone |
| 02:20.27 | jeev | ok cool |
| 02:20.44 | [TK]D-Fender | mosty: Yesterday I met somebody exactly like you.... only different! |
| 02:20.45 | jeev | C4colo, with so many nat problems.. i turned both the datacenter and local asterisk server on, set up iax in between.. |
| 02:21.05 | johnantypas | Guide followed -- My asterisk server is IP public (No NAT), and the client is behind NAT so my Asterisk sip.conf per the guide has the NAT=yes and qualify statements. However, |
| 02:21.09 | *** part/#asterisk johnantypas (n=jantypas@mail.antypas.net) |
| 02:21.22 | *** join/#asterisk chendy (n=chatzill@58.61.40.100) |
| 02:21.33 | C4colo | iax2 is pretty good for nat traversal |
| 02:21.44 | mosty | [TK]D-Fender, in terms of call routing, a hop is a hop |
| 02:21.49 | C4colo | is a hop |
| 02:22.10 | C4colo | hip to the hop and you don't stop |
| 02:22.23 | C4colo | ok, I'm sleep-deprived and it's raining |
| 02:22.33 | C4colo | not that those two things correlate |
| 02:23.33 | AndyML | anyone know a way to find out if a database key exists in the dialplan? |
| 02:23.48 | jaytee | show dialplan? |
| 02:24.06 | AndyML | from the dialplan... |
| 02:24.06 | [TK]D-Fender | mosty: in pretty much ANYTHING, "a hop is a hop". |
| 02:24.11 | mosty | AndyML, use the DB function, along with GotoIf or similar |
| 02:24.16 | [TK]D-Fender | unload chan_tautology.so |
| 02:24.27 | AndyML | mosty: i'm trying. i can't seem to get it right. let me get you an example |
| 02:24.42 | mosty | [TK]D-Fender, that's the beauty of hops |
| 02:24.44 | [TK]D-Fender | AndyML: "core show functions like DB" |
| 02:24.54 | [TK]D-Fender | mosty: mmmmmmmmmmmmmmm BEER |
| 02:24.56 | AndyML | ok - i'll look there. |
| 02:25.30 | AndyML | DB_EXISTS! thanks mosty |
| 02:25.48 | AndyML | and Fender |
| 02:26.26 | jeev | if i were to have something like exten =>_X.,1,Dial(IAX2/x:password@HOSTNAME.COM/${EXTEN}) in.. how often would the ip be resolved? as often as the ttl/min ? |
| 02:28.44 | mosty | jeev: i would not recommend that for dynamic hosts. set host=dynamic in the account in iax.conf and make the client register |
| 02:29.50 | jeev | ahh |
| 02:30.09 | jeev | how would i send the dial command then to that ? |
| 02:30.10 | jeev | ahh |
| 02:31.15 | jeev | yay, it works. |
| 02:44.27 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 02:49.11 | *** join/#asterisk Sargun (n=xbmodder@atarack/staff/sargun) |
| 02:52.59 | *** join/#asterisk attila_dexter (n=dexter@205.231.130.2) |
| 02:53.02 | attila_dexter | hello |
| 02:53.22 | attila_dexter | i need some help to setup our new te220 with redhat |
| 02:53.43 | attila_dexter | can some one help? |
| 02:54.07 | jeev | ask away dood, if anyone knows, they'll try |
| 02:54.16 | [TK]D-Fender | ~docs |
| 02:54.16 | jbot | somebody said docs was for basic documentation of Asterisk ask see http://voip-info.org/ (~voip-info) and TheBook (~book) |
| 02:54.48 | attila_dexter | i have the red light flashing |
| 02:54.56 | attila_dexter | zaptel is configured and started |
| 02:55.02 | attila_dexter | no errors shows |
| 02:55.17 | attila_dexter | i'm lost now, reading docs for more than 2 hours now |
| 02:58.15 | [TK]D-Fender | attila_dexter: What signalling are you supposed to be getting from the telco? |
| 02:58.21 | *** join/#asterisk ipso (n=ipso@S0106005004c32d38.ok.shawcable.net) |
| 02:58.56 | attila_dexter | they said that from their part everything is all right |
| 02:59.08 | attila_dexter | i changed cable |
| 02:59.12 | *** join/#asterisk hunmonk (n=hunmonk@drupal.org/user/22079/view) |
| 02:59.17 | attila_dexter | check the whole circuit |
| 02:59.23 | attila_dexter | and it is still red |
| 02:59.24 | [TK]D-Fender | attila_dexter: Now try answering the question I asked |
| 03:00.12 | attila_dexter | sorry I'm from Hungary |
| 03:00.27 | jaytee | that's ok, I just ate |
| 03:00.36 | attila_dexter | it is a long distance pri |
| 03:00.46 | attila_dexter | and they said that it must be national |
| 03:00.48 | [TK]D-Fender | jaytee: hukt on fonix werkt 4 u! |
| 03:00.53 | attila_dexter | i mean the signalling |
| 03:00.54 | jaytee | lol |
| 03:00.59 | [TK]D-Fender | attila_dexter: FULL specs please. |
| 03:01.46 | attila_dexter | [TK]D-Fender: what do you want to know? |
| 03:02.10 | [TK]D-Fender | attila_dexter: EXACTLY what signalling is being used. |
| 03:02.22 | [TK]D-Fender | attila_dexter: this is usually about 4 different parameters |
| 03:02.48 | attila_dexter | i do not have this info :/ what should i ask |
| 03:03.48 | attila_dexter | is a way to find out without calling the telco? |
| 03:04.06 | *** join/#asterisk tobias (n=tobias@cpe-076-182-118-165.nc.res.rr.com) |
| 03:04.08 | [TK]D-Fender | attila_dexter: Maybe reading papers they gave you |
| 03:04.44 | attila_dexter | i do not have them |
| 03:04.50 | [TK]D-Fender | attila_dexter: brilliant |
| 03:04.54 | attila_dexter | yes |
| 03:05.10 | attila_dexter | unfortunattly such is life |
| 03:05.11 | [TK]D-Fender | attila_dexter: Your card might have a jumper on it to set it from T1 to E1 mode. Verify its setting |
| 03:05.46 | attila_dexter | as far as i know the line is 24 channel |
| 03:06.13 | [TK]D-Fender | attila_dexter: In Hungary I doubt that very much. |
| 03:06.22 | [TK]D-Fender | ~e1 |
| 03:06.22 | jbot | [~E1] E1 is the basic digital telephony circuit used everywhere except the US,Japan,Taiwan & Hong Kong (who use T1 and similar variants). E1 runs at 2.048Mbps. It can be an unstructured channel for data. It can be channelized, to provide 31 time slots of voice or data, each of 64kbps. Time slot 16 is used for D-Chan when used with PRI signalling. |
| 03:06.25 | [TK]D-Fender | ^^^^^^^^ |
| 03:06.46 | *** join/#asterisk jameswf-home (n=james@ip72-223-0-183.ph.ph.cox.net) |
| 03:07.18 | attila_dexter | no I'm in Fargo, ND |
| 03:07.29 | attila_dexter | but i am Hungarian |
| 03:08.04 | [TK]D-Fender | attila_dexter: Great, what little you tell us is misleading too. |
| 03:08.11 | [TK]D-Fender | attila_dexter: pastebin your configs. |
| 03:08.14 | [TK]D-Fender | ~pb |
| 03:08.15 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines 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 |
| 03:08.16 | [TK]D-Fender | ^^^^^^^^^^^^^^^^ |
| 03:15.59 | attila_dexter | http://pastebin.com/d609b40bb |
| 03:16.22 | attila_dexter | here is /etc/sysconfig/zaptel and the result of make config from zaptel |
| 03:17.25 | [TK]D-Fender | attila_dexter: I want your zaptel.con and zapata.conf. |
| 03:19.13 | attila_dexter | http://pastebin.com/m52ca5bc4 |
| 03:20.11 | attila_dexter | http://pastebin.com/m2ff75479 |
| 03:20.14 | attila_dexter | here you are |
| 03:21.15 | [TK]D-Fender | attila_dexter: Ok, that looks largely sane. You've fully configured your first port. |
| 03:21.45 | [TK]D-Fender | attila_dexter: do "ztcfg -vvvv" and pastebin the results |
| 03:22.02 | *** join/#asterisk PepOSX (n=angeldav@190.72.129.75) |
| 03:22.27 | attila_dexter | http://pastebin.com/d6cc216d1 |
| 03:24.41 | [TK]D-Fender | attila_dexter: ok, now start * manually "asterisk -gvvvvvvvc" |
| 03:25.26 | attila_dexter | ok |
| 03:25.28 | attila_dexter | started |
| 03:25.37 | [TK]D-Fender | attila_dexter: "pri show span 1" |
| 03:26.05 | attila_dexter | http://pastebin.com/d6d5432f4 |
| 03:27.23 | [TK]D-Fender | attila_dexter: Double check the port youplugged it into, and what kind of cable did you use? |
| 03:28.31 | [TK]D-Fender | attila_dexter: While you're at it : "cat /proc/interrupts" |
| 03:29.38 | attila_dexter | port 1 |
| 03:29.46 | attila_dexter | and seams to be a cat 5 patch cable |
| 03:30.14 | attila_dexter | http://pastebin.com/dc4909db |
| 03:33.23 | [TK]D-Fender | attila_dexter: try "signalling=ni1" in zapata.conf. Stop *. restart it |
| 03:34.02 | *** join/#asterisk astassistant (n=jtknapp@h-72-244-204-146.sttnwaho.dynamic.covad.net) |
| 03:35.06 | attila_dexter | no pri in a* |
| 03:36.01 | [TK]D-Fender | ? |
| 03:36.08 | jaytee | attila_dexter, your T1 goes from the demarc to a CSU, right? |
| 03:36.16 | jaytee | and then to your card? |
| 03:37.32 | [TK]D-Fender | attila_dexter: sorry, "signalling=pri_cpe" and "switchtype=ni1" <- |
| 03:37.41 | [TK]D-Fender | attila_dexter: My bad. |
| 03:38.18 | attila_dexter | jaytee sorry what is demarc? |
| 03:38.38 | jaytee | the smartjack the T1 comes into your premises on |
| 03:38.39 | [TK]D-Fender | attila_dexter: try what I've now corrected for you. |
| 03:39.11 | attila_dexter | Status: Provisioned, In Alarm, Down, Active |
| 03:39.23 | [TK]D-Fender | attila_dexter: With the new settings? |
| 03:39.27 | attila_dexter | yes |
| 03:39.42 | [TK]D-Fender | attila_dexter: Ok, you need to get the telco on the line with you |
| 03:39.42 | jaytee | probably the cable |
| 03:40.06 | attila_dexter | ok |
| 03:40.16 | attila_dexter | i will do it tomorrow 1st hour |
| 03:40.20 | attila_dexter | will you be here? |
| 03:40.28 | jaytee | he's always here |
| 03:40.32 | attila_dexter | ;) |
| 03:40.44 | attila_dexter | thanks! |
| 03:41.08 | attila_dexter | will let you know tomorrow morning what is the situation |
| 03:41.16 | attila_dexter | gn all |
| 03:42.45 | *** join/#asterisk mltlnx (n=mltlnx@cpe-24-193-11-14.nyc.res.rr.com) |
| 03:43.29 | jaytee | if you're not using a CSU or your CSU doesn't do internal crossover then you'd need a T1 crossover cable or your circuit would be in alarm and down. |
| 03:43.59 | [TK]D-Fender | jaytee: VERY unlikely however |
| 03:44.06 | attila_dexter | jaytee will try |
| 03:44.20 | attila_dexter | i have no access to the rack where the cable goes |
| 03:44.32 | jaytee | attila_dexter, unlikely but a possiblity. |
| 03:44.39 | attila_dexter | so will need to wait the telco team tomorrow morning |
| 03:44.49 | jaytee | and a T1 crossover is not the same as a CAT 5 crossover |
| 03:44.54 | attila_dexter | i will check, it cost nothing :) |
| 03:45.01 | jaytee | good luck |
| 03:45.04 | attila_dexter | thnx |
| 03:46.52 | attila_dexter | gn |
| 03:47.26 | jaytee | time to snooze |
| 03:47.35 | jaytee | nite [TK]D-Fender |
| 03:53.49 | *** join/#asterisk ar3dam (n=ar3dam@189.156.217.142) |
| 03:54.25 | ar3dam | hi there, somebody can help with the incoming call?... pls :) |
| 03:54.52 | *** join/#asterisk jasonwoot (n=jasonrot@user-69-73-40-171.knology.net) |
| 03:55.51 | jasonwoot | is anyone doing multiple button maps on polycom 501? |
| 04:00.03 | mosty | ar3dam, be more specific |
| 04:00.06 | *** join/#asterisk shinao1 (n=shinao1@62.173.48.176) |
| 04:01.20 | ar3dam | ok, when i dial, the asterisk answer with demo. |
| 04:02.20 | *** join/#asterisk gmaruzz (n=gmaruzz@213-140-6-122.ip.fastwebnet.it) |
| 04:02.28 | *** part/#asterisk gmaruzz (n=gmaruzz@213-140-6-122.ip.fastwebnet.it) |
| 04:04.59 | mosty | ok, so what's the problem? |
| 04:13.54 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 04:28.38 | ar3dam | mosty, i dont know how to route a incoming call, u can send some tuto? |
| 04:30.43 | [TK]D-Fender | ~book |
| 04:30.43 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 04:34.23 | *** join/#asterisk steliosk (n=Stelios@athedsl-309630.home.otenet.gr) |
| 04:35.36 | *** join/#asterisk steliosk (n=Stelios@athedsl-309630.home.otenet.gr) |
| 04:40.36 | *** join/#asterisk shinao1 (n=shinao1@62.173.48.176) |
| 04:43.59 | ar3dam | join #asterisk-gui |
| 04:50.30 | jeev | [TK]D-Fender, iax for life. |
| 04:51.27 | JT | iax is fail |
| 04:55.51 | *** join/#asterisk CunningPike (n=arodgers@d75-155-184-81.bchsia.telus.net) |
| 04:56.29 | *** join/#asterisk hoegaatit (n=laa@c-67-160-226-244.hsd1.ca.comcast.net) |
| 05:00.13 | drmessano | IAX is for life when you can't follow simple instructions to solve SIP NAT issues |
| 05:00.23 | drmessano | Otherwise, IAX is for week |
| 05:00.27 | drmessano | or month |
| 05:05.05 | *** join/#asterisk rfernandez (n=nobody@189.136.64.210) |
| 05:06.20 | rfernandez | hiya!! i have a proyect to use asterisk as an ivr for a pbx ericcson but the ericcson only accepts E1 channels there is a HW like a channel bank to emulate the e1 signal for the ericcson? E1 -- asterisk --- signal e1 ---- ericcson --- extensions |
| 05:06.54 | [TK]D-Fender | rfernandez: Digium and Sangoma make a large variety of T1/E1 hardware |
| 05:07.06 | [TK]D-Fender | rfernandez: Which were all made to work with * |
| 05:08.13 | rfernandez | [TK]D-Fender, okies! but for my question its possible right? (i dont want maybe to know the "solution" only a yes youre right or no youre wrong) |
| 05:08.27 | [TK]D-Fender | rfernandez: that was a "yes" |
| 05:08.50 | [TK]D-Fender | rfernandez: and no need for a channel bank, you can connect * to your PBX via E1 directly |
| 05:08.51 | rfernandez | [TK]D-Fender, jejejeje sorry i dont got it cause my english its a little roasty |
| 05:08.59 | rfernandez | [TK]D-Fender, really? |
| 05:09.01 | [TK]D-Fender | rfernandez: without any extra analog steps |
| 05:09.20 | rfernandez | E1 ---- asterisk ---- ericcsom -- extensions handled by ericcson? |
| 05:10.26 | rfernandez | checking sangomas page.. |
| 05:10.50 | [TK]D-Fender | rfernandez: E1 -> (digium/sangoma card in * box) -> * (processes calls) -> (out another E1 port on that card) -> your other PBX |
| 05:11.13 | [TK]D-Fender | rfernandez: So you'd want a 2-port or more card |
| 05:11.27 | rfernandez | im talking about 300 lines (inbound) |
| 05:11.38 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:13.15 | Wayhigh | if I install a new tdm400 and when I pickup a line I see the logs say it was ZAP/2-1.. does that mean the channel really is 2-1? |
| 05:13.49 | rfernandez | [TK]D-Fender, thx!! im checking the e1 hardware plus your suggest! |
| 05:13.50 | rfernandez | =D! |
| 05:14.34 | ManxPower | Wayhigh: Channel 2, call instance 1 |
| 05:15.01 | ManxPower | you never specify the call instance, but you'll see it in the log/cli |
| 05:15.52 | Wayhigh | ok.. I'm asking because it'm having some problems with getting the extension in place.. but I'm not gonna bother ya'll with that part of it |
| 05:15.53 | [TK]D-Fender | rfernandez: E1 supports 31 channels tops... |
| 05:16.17 | rfernandez | [TK]D-Fender, yup... im checking the maximum E1 cards... or maybe an astribank.... |
| 05:16.21 | [TK]D-Fender | Wayhigh: And we all know why... |
| 05:16.22 | Wayhigh | hey.. I have a tdm400 with 1 fxo.. what's a decent price to sell it for? |
| 05:16.33 | rfernandez | (im like a virgin! its my first E1 project lol!) |
| 05:16.56 | [TK]D-Fender | rfernandez: Why are you looking at channel bank technology in the first place? |
| 05:16.57 | JT | generally E1s only support 30 voice channels |
| 05:17.07 | Wayhigh | and I can get another tdm400 with 1fxo 1fxs |
| 05:17.28 | [TK]D-Fender | Wayhigh: Why not jsut by an FXS module for the one you have? |
| 05:17.43 | rfernandez | [TK]D-Fender, cause here in mexico its the myth that 3 digium cards cannot live in the same box (cause IRQ troubles) |
| 05:18.08 | [TK]D-Fender | rfernandez: Not a myth. It IS ill advised to run more than 2 cards in a single system |
| 05:18.10 | Wayhigh | fender: I got 2 fxs modules today |
| 05:18.20 | Wayhigh | pulled them out of this good card |
| 05:18.46 | rfernandez | [TK]D-Fender, ok if i want to handle 300 lines in E1 and came in packets or 30 dids i need 10 ports of E1 |
| 05:18.48 | Wayhigh | my old card has a broken port on it |
| 05:19.06 | Wayhigh | so I don't want to try to unload that on people |
| 05:19.08 | JT | a DID does not equal a channel on a PRI |
| 05:19.27 | JT | if you have 300 channels, buy PRI to SIP gateways |
| 05:20.02 | [TK]D-Fender | rfernandez: 1 system with a Sangoma A108d, another with A104D = 12 E1 |
| 05:21.15 | *** join/#asterisk maagic (i=maagic@fsck.fi) |
| 05:21.22 | rfernandez | 8 and 4..... |
| 05:21.24 | rfernandez | sure! |
| 05:21.27 | rfernandez | youre right... |
| 05:21.47 | rfernandez | [TK]D-Fender, man i need to invite you a beer and a dinner cause that sounds a good solution =D |
| 05:24.04 | [TK]D-Fender | rfernandez: It might be better still to split it up into 3 x 4Port cards. |
| 05:24.36 | rfernandez | oh ok |
| 05:25.41 | obnauticus | heh, how would one execute playback on a meetme conference? |
| 05:26.00 | mosty | use chan_local |
| 05:26.08 | [TK]D-Fender | obnauticus: call-file / AMI originate off a local channel |
| 05:30.18 | Wayhigh | gosh dang it.. I figured out my issue.. friggin.. fraggin.. farking bohr bohr bohr |
| 05:34.25 | [TK]D-Fender | ok, bed time. Later all |
| 05:50.20 | *** join/#asterisk blq (n=Bl@dslb-088-067-044-063.pools.arcor-ip.net) |
| 05:56.57 | *** join/#asterisk skyNomad (n=ralfe@196.212.110.130) |
| 06:03.25 | *** join/#asterisk oej (n=olle@ns.webway.se) |
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| 06:24.33 | *** part/#asterisk gones (n=gones@203.193.37.251) |
| 06:26.41 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 06:28.03 | skyNomad | Does anyone know how to determine the terminate cause of an Asterisk Dial command using AGI? |
| 06:30.05 | *** join/#asterisk h-idrisi (n=h-idrisi@212.103.170.132) |
| 06:31.49 | mosty | ${DIALSTATUS} |
| 06:32.29 | skyNomad | mosty: That is if I was using the dialplan from within extensions.conf . |
| 06:32.46 | skyNomad | mosty: But with AGI, once the call hangs up, the script looses connection to the Asterisk server. |
| 06:37.48 | mosty | perhaps you want deadagi? |
| 06:37.54 | mosty | i think that's what it's called |
| 06:39.33 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 06:42.53 | *** join/#asterisk jeffspeff (i=jeff@c-68-53-54-229.hsd1.ky.comcast.net) |
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| 06:46.39 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-f3ad9511cf702037) |
| 06:46.44 | miloux | skyNomad: You need to catch the SIGHUP and ignore it, also as far as i know you cant communicate with the asterisk server without getting a SIGPIPE |
| 06:48.09 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 06:48.18 | skyNomad | miloux: Thanks. |
| 06:48.24 | skyNomad | miloux: I'll look into that. |
| 06:55.31 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 07:04.10 | linuxstb | Is it normal to see lots of active IAX channels with remote IAX hardphones (Atcom AT-530s)? No calls are being made. http://www.pastebin.ca/1169302 |
| 07:06.30 | mosty | if there are no calls, there should be no active channels |
| 07:06.36 | mosty | most likely |
| 07:07.06 | linuxstb | That's what I would have thought... |
| 07:11.20 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 07:25.29 | *** join/#asterisk genin (n=phrame@ANice-252-1-81-109.w83-201.abo.wanadoo.fr) |
| 07:25.33 | genin | mornin folks |
| 07:27.37 | *** join/#asterisk Greek-Boy (n=email@41.221.58.5) |
| 07:31.15 | *** join/#asterisk shinao1 (n=shinao1@62.173.48.176) |
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| 07:32.43 | *** part/#asterisk zydoon (n=zydoon@41.225.154.46) |
| 07:34.26 | *** join/#asterisk GhOnDiE (n=ghondie@dsl78-143-210-38.in-addr.fast.co.uk) |
| 07:36.43 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 07:57.54 | *** join/#asterisk cjk (n=cjk@vodsl-10830.vo.lu) |
| 07:57.55 | cjk | hi, how can i play a sound file to the caller and continue in the dialplan? |
| 07:59.15 | *** join/#asterisk zydoon (n=zydoon@41.225.154.46) |
| 07:59.38 | *** part/#asterisk zydoon (n=zydoon@41.225.154.46) |
| 08:00.28 | *** join/#asterisk blackest (n=john@cpc1-linc10-0-0-cust514.nott.cable.ntl.com) |
| 08:04.58 | skyNomad | cjk: I think the command is Background(filetoplay) |
| 08:05.25 | *** join/#asterisk blackest (n=john@cpc1-linc10-0-0-cust514.nott.cable.ntl.com) |
| 08:10.05 | mvanbaak | If you just want to play a file, use Playback() |
| 08:10.23 | mvanbaak | if you want to play a file, and let the user input a digit (IVR) use Background |
| 08:11.35 | cjk | thanks, unfortunately i do not want to do any of this |
| 08:11.52 | cjk | playback waits until the file is finished before it continues in the dialplan |
| 08:18.17 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
| 08:18.26 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 08:20.53 | *** join/#asterisk xenonex (n=xenonex@92.47.1.175) |
| 08:23.49 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 08:25.22 | *** join/#asterisk qdk (n=qdk@85.235.253.139) |
| 08:37.01 | wizzy_ | I am installing asterisk + astribank on Ubuntu 8.04. Everything from svn HEAD. re: astribank - on boot it comes up, and loads the firmware courtesy /etc/udev/rules.d/xpp.rules (which should be /etc/udev/rules.d/50-xpp.rules ??) |
| 08:37.46 | wizzy_ | However, it does not do the "mount procbususb /proc/bus/usb -t usbfs" thing - needed for dahdi_hardware |
| 08:38.38 | wizzy_ | Do I add that somewhere in /etc/udev/rules.d ? Separate rule ? Addition to 60-persistent-storage.rules ? |
| 08:51.56 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 09:01.20 | *** join/#asterisk oilinki3 (n=oil@ppp-124-121-248-252.revip2.asianet.co.th) |
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| 09:03.22 | *** part/#asterisk blackest (n=john@cpc1-linc10-0-0-cust514.nott.cable.ntl.com) |
| 09:10.17 | skyNomad | I've just upgraded to 1.6 Beta. Can anyone tell me what is going on with the sound files? |
| 09:10.28 | skyNomad | How is the new directory structure working? |
| 09:11.45 | *** join/#asterisk micheluntu (n=michele@dhcp197-193.cu.mi.it) |
| 09:12.17 | micheluntu | hi all i'm having some problems with dundi.. |
| 09:13.41 | micheluntu | this is the log messages http://pastebin.com/d1c594068 |
| 09:14.12 | *** join/#asterisk foexle (n=heiko@router.moltomedia.de) |
| 09:15.50 | *** join/#asterisk onet-shaun (n=chatzill@82-33-19-117.cable.ubr01.stav.blueyonder.co.uk) |
| 09:15.55 | *** join/#asterisk vale-ICS (n=vale@icsnet.demon.co.uk) |
| 09:17.35 | onet-shaun | Morning, im new here but have a zaptel question, i have a t210p digium card, and want port 1 on isdn30 (uk) and port 2 to act as a bridge to a normal phone system, zap show status in asterisk says both lines are ok but rinignin in generates nothing in the CLI even in debug mode and i just get a solid but slightly broken tone ! any ideas ? |
| 09:18.58 | onet-shaun | in zaptel.conf i have: |
| 09:18.59 | onet-shaun | # Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER) |
| 09:19.01 | onet-shaun | span=1,1,0,ccs,hdb3,crc4 |
| 09:19.02 | onet-shaun | # termtype: te |
| 09:19.04 | onet-shaun | bchan=1-15,17-31 |
| 09:19.06 | onet-shaun | dchan=16 |
| 09:19.07 | onet-shaun | # Span 2: TE2/0/2 "T2XXP (PCI) Card 0 Span 2" HDB3/CCS/CRC4 |
| 09:19.09 | onet-shaun | span=2,2,0,ccs,hdb3,crc4 |
| 09:19.10 | onet-shaun | # termtype: te |
| 09:19.12 | onet-shaun | bchan=32-46,48-62 |
| 09:19.13 | onet-shaun | dchan=47 |
| 09:19.15 | onet-shaun | # Global data |
| 09:19.16 | onet-shaun | loadzone = uk |
| 09:19.18 | onet-shaun | defaultzone = uk |
| 09:20.24 | creativx | pastebin plz |
| 09:23.09 | onet-shaun | http://pastebin.com/d5f2fcd5 :) taa |
| 09:28.10 | foexle | hi guys i've a question, astersik write logs in /var/log/asterisk .... but which user use asterisk ??? .... the background are, im starting an agi-script(php) and i would write my own log-files in /var/log/asterisk but i think this php skript dont have permissions at this folder ... i could change the owner but i dont know which user are using by asterisk |
| 09:28.18 | foexle | i hope u understand ;) |
| 09:29.07 | micheluntu | foexle: the user is a command line parameter -U username |
| 09:29.27 | skyNomad | foexle: Check which group the log files are owned by, and then add the user php is running under to that group. |
| 09:29.28 | micheluntu | foexle: you can see doing something like ps -ef | grep asterisk |
| 09:29.46 | foexle | ok thx |
| 09:30.06 | EmleyMoor | A codec "AMR" is listed on my N95 - is this known by any other name? |
| 09:30.08 | skyNomad | foexle: In php, how are you writing to the logs? |
| 09:30.27 | foexle | error_log() |
| 09:30.45 | *** join/#asterisk LND (n=Lee@89.192.144.184) |
| 09:30.53 | skyNomad | foexle: Have you set your error stream? |
| 09:31.22 | foexle | no i havnt ... error_log creates a file when this not exists ... |
| 09:32.27 | foexle | but i need permissions for that :> ... ok i look which user use asterisk an chwon :> ... i test it |
| 09:33.07 | skyNomad | foexle: You might need to do this : define ('STDERR', fopen('php://stderr', 'w')); |
| 09:33.34 | skyNomad | foexle: Then you can just let Asterisk handle the error logging. |
| 09:34.13 | wizzy_ | when I start asterisk, I get an error message "ERROR[5741] asterisk.c: Asterisk has detected a problem with your DAHDI configuration and will shutdown for your protection." |
| 09:34.29 | wizzy_ | looking at the source, this seems to be a timer problem. |
| 09:34.45 | foexle | hmmm ... ok. In which file write asterisk all errors ? only in /var/log/asterisk/error.log? |
| 09:35.15 | wizzy_ | in /proc/dahdi, I have 1 2 3 4 - but they all refer to my AStribank - there is no dummy driver |
| 09:35.55 | wizzy_ | Do I need the dummy driver for a timer ? |
| 09:36.23 | wizzy_ | foexle: I have them in /var/log/asterisk/full - but you have to turn all the logging on |
| 09:36.37 | *** join/#asterisk mltlnx (n=mltlnx@cpe-24-193-11-14.nyc.res.rr.com) |
| 09:37.09 | foexle | ok thanks a lot |
| 09:38.06 | *** join/#asterisk wil_syd (n=root@c211-30-113-159.rivrw2.nsw.optusnet.com.au) |
| 09:38.58 | micheluntu | i'm trying to put sip agent information dinamically into a mysql database using real time engine |
| 09:39.56 | micheluntu | but there's now rows in table and i see no errors in asterisk... |
| 09:40.19 | skyNomad | micheluntu: How are you inserting the data into MySQL? |
| 09:40.46 | micheluntu | with extconfig.cong and res_mysql.conf |
| 09:42.18 | skyNomad | micheluntu: If you use AGI, you would have more control and be able to better debug the problem. |
| 09:43.45 | *** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
| 09:43.54 | micheluntu | skyNomad: i found this message MySQL RealTime: No database socket found, using '/tmp/mysql.sock' as default. |
| 09:44.20 | micheluntu | but the database is not local is running on another server |
| 09:47.46 | skyNomad | micheluntu: Is the database server definately running? |
| 09:47.59 | skyNomad | micheluntu: Can you connect to the database manually? |
| 09:48.06 | micheluntu | yes |
| 09:49.05 | wizzy_ | micheluntu: change /etc/asterisk/res_mysql.conf ? |
| 09:49.42 | skyNomad | micheluntu: Do you have mysql client libraris/headers installed on the asterisk box? |
| 09:50.14 | micheluntu | yes |
| 09:50.36 | wizzy_ | mine has "dbsock = /var/run/mysqld/mysqld.sock" - I would try removing that, and setting dbhost/dbport |
| 09:51.34 | wizzy_ | and you can "telnet dbhost 3306" from the asterisk box ? |
| 09:51.34 | Nugget | telnet is eeeeeeevil! |
| 09:52.58 | micheluntu | wizzy_: yes |
| 09:53.19 | skyNomad | micheluntu: I can't seem to see a simple way of doing it without using ODBC rather. |
| 09:53.27 | micheluntu | FYI i have a lookupmysql extensions that is working |
| 09:53.41 | micheluntu | in fact if I drop the table I get an error |
| 09:54.03 | micheluntu | so the broblem seems to be realtime |
| 09:54.08 | micheluntu | *problem |
| 09:54.13 | skyNomad | micheluntu: Hmmm.. |
| 09:54.37 | skyNomad | micheluntu: Or, it could be a permissions problem, or a database structure issue. |
| 09:55.23 | skyNomad | micheluntu: Is logging enabled on the MySQL server? |
| 09:55.49 | skyNomad | micheluntu: If it is, then you can check out /var/log/mysql/mysqld.err and see if the error message is comming up there. |
| 09:55.56 | micheluntu | skyNomad: no.. but i'll enable now |
| 09:56.39 | *** join/#asterisk SanityIO (n=SanityIO@77.242.105.93) |
| 09:56.40 | *** join/#asterisk SanityIO_ (n=SanityIO@77.242.105.93) |
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| 10:01.18 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 10:05.04 | micheluntu | skyNomad: the message is misleading.. the connection is up |
| 10:05.29 | micheluntu | so the problem is that registration doesn't write into db... |
| 10:07.41 | *** part/#asterisk wil_syd (n=root@c211-30-113-159.rivrw2.nsw.optusnet.com.au) |
| 10:10.51 | *** join/#asterisk SanityIO (n=SanityIO@77.242.105.93) |
| 10:18.28 | *** join/#asterisk tkbeat (n=tk@80.64.182.204) |
| 10:19.12 | skyNomad | micheluntu: Is there no error messages in the mysqld.err file? |
| 10:19.17 | tkbeat | hi |
| 10:19.23 | tkbeat | <PROTECTED> |
| 10:19.58 | skyNomad | tkbeat: core set verbosity 10 |
| 10:20.42 | skyNomad | tkbeat: oops....... core set verbose 10 |
| 10:21.36 | skyNomad | tkbeat: execute that command and try receive a call. See if anything comes up on asterisk CLI then. |
| 10:28.24 | micheluntu | skyNomad: I think i didn't understood how extconfig.conf works.. |
| 10:28.56 | tkbeat | skyNomad, absolute nothing comes in on the cli |
| 10:28.59 | micheluntu | I put which configuration files will be load from database |
| 10:29.05 | micheluntu | right? |
| 10:29.07 | skyNomad | micheluntu: hehe... that could be a problem |
| 10:29.25 | tkbeat | (when i make an incoming call) |
| 10:29.38 | skyNomad | brb |
| 10:30.19 | micheluntu | skyNomad: can you explain me? |
| 10:31.23 | skyNomad | tkbeat: Check that your channels are up. |
| 10:31.52 | tkbeat | channels ? |
| 10:33.14 | tkbeat | what does it mean ? |
| 10:33.52 | skyNomad | micheluntu: Have you configured the mappings in extconfig.conf ? |
| 10:34.03 | tkbeat | skyNomad, where can i find this information ? |
| 10:34.05 | wizzy_ | sip.conf => mysql,voiceone,ast_config |
| 10:34.27 | micheluntu | skyNomad: I hope.... sippeers => mysql,asteriskdb,sip |
| 10:37.09 | *** join/#asterisk ccesario_ (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 10:37.46 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 10:38.22 | micheluntu | skyNomad: one question.. if i must put database name in res_mysql.conf (dbname) what the second parameter is in extconfig.conf? (was voiceone in your example) |
| 10:40.13 | skyNomad | tkbeat: Check out /proc/zaptel/ |
| 10:40.28 | skyNomad | tkbeat: In there are some files. You can check out the status there. |
| 10:40.37 | skyNomad | tkbeat: Otherwise, there are some CLI commands you can use. |
| 10:42.35 | skyNomad | micheluntu: I'm busy looking for a link that will help you,. |
| 10:43.04 | micheluntu | skyNomad: thanks, i'm looking |
| 10:43.11 | *** join/#asterisk mltlnx (n=mltlnx@96.232.20.139) |
| 10:43.21 | skyNomad | micheluntu: Check out http://www.voip-info.org/wiki-Asterisk+RealTime and http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip |
| 10:43.47 | *** join/#asterisk Dovid (n=Dovid@tony09-118-62.inter.net.il) |
| 10:45.11 | *** join/#asterisk quentusrex (n=quentusr@c-71-197-244-228.hsd1.or.comcast.net) |
| 10:45.55 | skyNomad | tkbeat: Some URL's which will help you : http://www.voip-info.org/wiki/view/Asterisk+Zaptel+Installation | http://www.voip-info.org/wiki/view/Asterisk+config+zaptel.conf | http://www.voip-info.org/wiki/view/Asterisk+config+zapata.conf |
| 10:46.01 | quentusrex | Does anyone know of a good piece of hardware that would help move a home phone from pots to voip? I'm looking for a device that has 1 ethernet for sip, 1 FXS and 1 FXO. |
| 10:46.12 | skyNomad | voip-info.org is the ultimate reference for all things VoIP and Asterisk related. |
| 10:46.22 | quentusrex | I already have a voip account with a providor. |
| 10:46.22 | mvanbaak | ~ata |
| 10:46.23 | jbot | from memory, ata is Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 10:47.09 | *** join/#asterisk ElSonico (n=tav@nat/ibm/x-8265f7000b96343b) |
| 10:47.24 | quentusrex | I have found a few devices, but are there hardware recommendations? |
| 10:48.03 | *** part/#asterisk ElSonico (n=tav@nat/ibm/x-8265f7000b96343b) |
| 10:51.42 | *** join/#asterisk MaliutaLap (n=nikolai@kiev.lusan.id.au) |
| 10:52.27 | quentusrex | I'm looking for something that will take all incoming calls from the pots system, and send all out going calls to sip |
| 10:52.54 | quentusrex | And while I wait for the DID to transfer to the voip providor. When the DID moves, I need to only use the sip... |
| 10:58.24 | tkbeat | yes i have one trunk to my sip provider and it is registered |
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| 11:06.46 | skyNomad | Is anyone here using version 1.6? |
| 11:08.09 | wizzy_ | yes |
| 11:09.13 | skyNomad | For some reason, now that I have upgraded from 1.4 to 1.6, my READ() commands don't work. |
| 11:09.16 | skyNomad | It is so odd. |
| 11:09.23 | skyNomad | any thoughts? |
| 11:09.32 | skyNomad | PS: I've just upgraded about 2 hours ago. |
| 11:12.24 | wizzy_ | skyNomad: I am installing, not upgrading. So - no comments |
| 11:13.07 | skyNomad | ah... ok then. |
| 11:13.43 | *** join/#asterisk samad (n=samad@116.71.184.168) |
| 11:14.32 | samad | heloo every one |
| 11:16.07 | *** join/#asterisk exodos (n=notch@mach.xs4all.nl) |
| 11:16.23 | samad | can anybody helpme regarding telephony application |
| 11:16.39 | samad | i am trying recieve call and dail from our dotnet application |
| 11:16.40 | samad | ? |
| 11:18.45 | exodos | i'm using trixbox 2.6.1 with asterisk 1.4.20 and i'm trying to make it work with mISDN. It look like there is no misdn command in asterisk anymore. Anyone knows where to look for it? |
| 11:18.50 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 11:19.33 | wizzy_ | exodos: you need to load the module / compile the module |
| 11:19.42 | samad | wizzy? |
| 11:20.02 | wizzy_ | samad: where is your question ? |
| 11:20.08 | wizzy_ | help me ? |
| 11:21.24 | wizzy_ | exodos: module load chan_misdn ? |
| 11:23.39 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 11:24.04 | kaldemar | exodos: yes, go to #trixbox |
| 11:24.07 | skyNomad | samad: How are you interacting with Asterisk from .NET? |
| 11:24.35 | samad | that what i am looking for:) |
| 11:24.48 | creativx | samad: use asterisk.net libs |
| 11:24.53 | creativx | a port from asteriskjava |
| 11:25.15 | samad | creativx |
| 11:25.16 | skyNomad | samad: What exactly are you trying to do? |
| 11:25.30 | creativx | samad: google "asterisk.net sourceforge" |
| 11:25.35 | samad | actually i am developing application in dotnet |
| 11:25.46 | creativx | are you making a CRM-like app, or a softphone |
| 11:25.50 | samad | where agent will recieve call and dail back to cutomer |
| 11:26.14 | samad | i need help where i can see asterisk with dotnet |
| 11:26.23 | skyNomad | samad: Why don't you just customize an opensource sip client? |
| 11:26.38 | samad | sip? |
| 11:26.53 | skyNomad | hmmm... I think you have some research to do. |
| 11:27.21 | skyNomad | samad: SIP is the protocol you will probably want to use for communicating with Asterisk. |
| 11:27.27 | samad | ok |
| 11:27.58 | skyNomad | samad: You should read through the articles at http://www.voip-info.org |
| 11:28.08 | samad | ok |
| 11:28.08 | qp | we use sip for trunks and iax2 for softphones |
| 11:28.08 | skyNomad | samad: Also, look specifically at http://www.voip-info.org/wiki-SIP |
| 11:28.17 | samad | ok |
| 11:28.27 | samad | so what type of hardware need for that |
| 11:28.41 | skyNomad | qp: And we use sip for softphones and iax2 for trunks. It all depends on the situation. |
| 11:28.48 | qp | samad, we have developed a c# .net app for talking to asterisk, and use iaxclient and this wrapper: http://www.asteriasgi.com/?q=IAXClient-Wrapper |
| 11:28.51 | qp | true |
| 11:29.04 | qp | samad, if you want to chat more about .net stuff, pop into #iaxclient |
| 11:29.18 | samad | thanks all of u |
| 11:29.18 | skyNomad | samad: You could write something which doesn't even need hardware, apart from speakers and a microphone. |
| 11:29.23 | skyNomad | sure |
| 11:32.14 | *** part/#asterisk Tobsn (n=luxus@freshmeat.de) |
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| 11:36.27 | *** join/#asterisk samad (n=samad@116.71.184.168) |
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| 11:38.19 | samad | skyNomad |
| 11:40.41 | Wayhigh | finally got my home line setup to answer as an IVR between 8:30pm and 7am. No more annoying calls from drunk people thinking my number's a cab company |
| 11:41.13 | skyNomad | samad: yes? |
| 11:41.46 | samad | have u exp developing telephony application |
| 11:42.27 | skyNomad | samad: Yes. |
| 11:42.51 | skyNomad | samad: My job involves developing large Asterisk-based applications. |
| 11:42.57 | skyNomad | samad: Why do you ask? |
| 11:43.08 | samad | sir i am begginer |
| 11:43.21 | samad | i have some developing exp in c# |
| 11:43.23 | qp | sir? |
| 11:43.28 | qp | looks around for a sir |
| 11:43.58 | skyNomad | samad: That link that qp posted looks like a great way for you to get started. |
| 11:45.19 | skyNomad | samad: I suggest you try out that wrapper by qp. |
| 11:45.29 | the_5th_wheel | samad: skyNomad is special, but he is definitly not good enough to be called sir :-p |
| 11:45.54 | qp | our c#.net app is quite feature rich now, attended transfers, multi lines, etc |
| 11:46.08 | skyNomad | the_5th_wheel: hehe.... One can always count on you to be scathing. |
| 11:46.42 | the_5th_wheel | skyNomad: and violent :-) |
| 11:48.42 | Wayhigh | makes a note not to stand behind 5thwheel |
| 11:50.08 | skyNomad | dodges yet another flying object from the_5th_wheel |
| 11:50.38 | the_5th_wheel | cant get something that wont make too much of a mess flinging it across the office |
| 11:50.55 | skyNomad | is very relieved |
| 11:50.58 | *** join/#asterisk ElSonico (n=tav@nat/ibm/x-a00616c80837aef4) |
| 11:51.53 | *** join/#asterisk Sniper_VOIP (n=michofr@mail.splendor.net) |
| 11:52.25 | Wayhigh | 5thwheel: for office warfare, I recommend chocolate, eggs, cabbages, beans, and mountain dew |
| 11:52.27 | Sniper_VOIP | [TK]D-Fender...Fuck you and your family |
| 11:52.35 | Wayhigh | eat that.. wait a few hours.. and begin barnstorming |
| 11:53.24 | skyNomad | ewww |
| 11:53.35 | skyNomad | If he did that, I'd have to relocate. |
| 11:54.30 | quentusrex | What are your thoughts on the grandstream 503? |
| 11:54.56 | Nugget | ~gs |
| 11:54.56 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 11:54.56 | Wayhigh | quent: it's not nearly as good as some good crop dusting of annoying officemates |
| 11:55.33 | skyNomad | I really like the grandstreams. I use the Grandstream 100. |
| 11:55.40 | quentusrex | Well, what piece of hardware would be a better replacement? |
| 11:55.50 | skyNomad | hahaha.... I think the jbot is a little bias. |
| 11:55.59 | skyNomad | I've never had a problem with a Grandstream. |
| 11:56.06 | skyNomad | And they are quick and simple to configure. |
| 11:56.08 | the_5th_wheel | Wayhigh: he would enjoy the chocolate... eggs and cabbage wold smell |
| 11:56.38 | the_5th_wheel | skyNomad: Are you smoking dope again? These grandstreams a kak! |
| 11:56.46 | skyNomad | the_5th_wheel: I wouldn't enjoy anything that came out any of your orifices. |
| 11:57.00 | *** join/#asterisk scampbell (n=scampbel@199.105.195.156) |
| 11:57.50 | quentusrex | I've used a grandstream phone, and it worked well for 3 years and then crapped out. But 3 years is fine. |
| 11:57.52 | the_5th_wheel | whoops, missed the second message... |
| 11:58.39 | quentusrex | What is a better version of the grandstream handytone 503? |
| 11:58.47 | quentusrex | a higher quality piece of hardware? |
| 11:58.57 | the_5th_wheel | The grandstreams wwe've had here gave us lots of issues, not a single one lasted longer than 6 months |
| 11:59.48 | Nugget | polycoms seem to be the consensus favorite in this channel, but I don't speak from experience. I only have non-voip polycom stuff. |
| 12:00.08 | Nugget | I have cisco phones which I absolutely do not recommend. |
| 12:00.51 | quentusrex | ok, so there is one vote for polycoms |
| 12:00.57 | quentusrex | what else? |
| 12:01.00 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:01.00 | *** mode/#asterisk [+o lmadsen] by ChanServ |
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| 12:05.30 | the_5th_wheel | quentusrex: in the lowbudget range, the atcoms work decentish |
| 12:06.18 | quentusrex | what about an american voip supplier? |
| 12:07.49 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 12:09.13 | *** join/#asterisk ElSonico (n=tav@nat/ibm/x-bbdc3af3c0720551) |
| 12:11.18 | wizzy_ | Are we anywhere close to having asterisk authenticate users against LDAP ? |
| 12:11.34 | *** join/#asterisk hunmonk (n=hunmonk@drupal.org/user/22079/view) |
| 12:11.46 | micheluntu | where can i find the most simple extensions.conf ? For testing purpose I need only sip phone to sip phone call... |
| 12:12.08 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:14.27 | qp | might be some here: http://www.voip-info.org/wiki/view/Asterisk+Configuration+Examples |
| 12:18.32 | micheluntu | qp thanks I take a look |
| 12:21.34 | *** join/#asterisk Gat0rvean (n=gredish@64.191.128.145) |
| 12:33.46 | KermitTheFragger | quentusrex: i would also like to cast my vote for the 'stay as far away from grandstream as possible'-option |
| 12:34.14 | quentusrex | KermitTheFragger, I can only stay away from grandstream if I have a replacement. |
| 12:34.26 | quentusrex | What could do what the grandstream handytone 503? |
| 12:34.38 | KermitTheFragger | quentusrex: we use cisco 7931's here, with chan-sccp-b |
| 12:34.56 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 12:35.02 | qp | can anyone explain to me what "timing issues" actually refers too when talking about vmware etc and if its still applicable to a VOIP only solution? |
| 12:35.17 | KermitTheFragger | biggest advantage imho is that you can easily downgrade the firmware, so you can find a stable one which suits your needs |
| 12:35.33 | quentusrex | I don't like cisco... |
| 12:35.37 | quentusrex | anything else? |
| 12:35.45 | [TK]D-Fender | quentusrex: What do you require of that device? |
| 12:36.09 | KermitTheFragger | yeah requirements would make it easier :-) |
| 12:36.10 | quentusrex | I have a pots phone system. It'll be three weeks until the DID is transfered to the voip providor. |
| 12:36.22 | KermitTheFragger | we choose the cisco 7931's because of the BLF's |
| 12:36.33 | quentusrex | I want to plug in a device, register with the voip account. |
| 12:36.53 | quentusrex | and while the pots is handling my DID, my home phones ring through the pots. |
| 12:37.10 | quentusrex | but when the DID moves to voip, my phones still ring.... |
| 12:37.26 | [TK]D-Fender | quentusrex: Linksys SPA-3102 <- |
| 12:38.39 | *** join/#asterisk wonderworld (n=ww@ip-62-143-216-14.hsi.ish.de) |
| 12:38.48 | [TK]D-Fender | KermitTheFragger: Do those buttons really light up? Supported under SIP? |
| 12:39.03 | KermitTheFragger | [TK]D-Fender: yeah i made some change to the source |
| 12:39.30 | [TK]D-Fender | KermitTheFragger: For SIP or SCCP? |
| 12:40.03 | KermitTheFragger | [TK]D-Fender: ow right, srry i think i misunderstood you. SCCP image that is |
| 12:40.10 | [TK]D-Fender | KermitTheFragger: And those are pricey buggers.... and look strangely like the LinksysOne phones (PHM) I saw previewed a year & a half ago |
| 12:40.17 | KermitTheFragger | [TK]D-Fender: afaik there isnt a sip image for the 7931 |
| 12:40.42 | KermitTheFragger | [TK]D-Fender: I buy them for 120 euro's a piece |
| 12:40.57 | [TK]D-Fender | KermitTheFragger: http://www.telephonyware.com/telephonyware/tw00522.html?id=BjoA6Yis <- this SAYS SIP.... but who knows... |
| 12:41.32 | [TK]D-Fender | KermitTheFragger: And the buttons on the side actually light up? All of the stock photos make it hard to tell |
| 12:41.58 | KermitTheFragger | [TK]D-Fender: Yes the buttons lit up when some one makes, receives a call |
| 12:42.22 | *** join/#asterisk pukkita (n=pukkita@137.Red-80-59-10.staticIP.rima-tde.net) |
| 12:42.27 | KermitTheFragger | [TK]D-Fender: and in the display it also lists all the BLF fields and even shows the number of the other party |
| 12:42.53 | pukkita | anyone familiar with primaries? |
| 12:43.31 | [TK]D-Fender | pukkita: Yes, we're all following the Presidental race like hawks... |
| 12:43.46 | KermitTheFragger | [TK]D-Fender: but thats all with the SCCP image |
| 12:44.03 | [TK]D-Fender | KermitTheFragger: Yeah, hte 7914 only worked under it as well |
| 12:44.37 | pukkita | I'm having an odd problem, have my asterisk box connected via a primary to an old "digital" PBX. If I route a call from the asterisk box to the PBX, the call enters just fine, but if I try to route a call from the PBX to the asterisk box, it looks like the PBX detects somehow the line as unavailable and hangs without even sending the number to call |
| 12:44.44 | pukkita | [TK]D-Fender: :) |
| 12:45.12 | samad | qp? |
| 12:45.34 | [TK]D-Fender | pukkita: pastebin CLI debug (PRI/etc) to match |
| 12:45.36 | [TK]D-Fender | ~pb |
| 12:45.36 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines 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 |
| 12:45.38 | [TK]D-Fender | ^^^^^^^^^^^ |
| 12:45.45 | pukkita | I can't see any errors on the asterisk side, when the PBX tries to make a call through that connection asterisk just just shows as if the PBX would have dialed the s extension |
| 12:45.59 | KermitTheFragger | [TK]D-Fender: from the research i did a while back, SIP with cisco phones isn't much of a success |
| 12:46.17 | pukkita | [TK]D-Fender: there are no errores, already debugged using wanpipemon |
| 12:46.43 | KermitTheFragger | [TK]D-Fender: you loose half the phones functionality, and its buggy |
| 12:46.44 | [TK]D-Fender | KermitTheFragger: Well it works for 79/4/6x pretty normally last I heard. You do lose some functionality SCCP enjoys of course |
| 12:46.58 | [TK]D-Fender | KermitTheFragger: SIP is an afterthought to Cisco.... |
| 12:47.37 | pukkita | I suspect it could be a parameter in zapata.conf??? |
| 12:47.52 | [TK]D-Fender | pukkita: Could be. pastebin it along with the CLI debug I requested |
| 12:47.54 | KermitTheFragger | [TK]D-Fender: hopefully that will change in the future, but for now im quite content with chan-sccp-b ;-) |
| 12:49.17 | *** join/#asterisk ManxPower (n=manxpowe@25.sub-75-249-159.myvzw.com) |
| 12:53.41 | KermitTheFragger | [TK]D-Fender: http://www.cisco.com/en/US/products/hw/phones/ps379/prod_release_notes_list.html#anchor18 SIP support was added last march |
| 12:56.41 | *** join/#asterisk albertoandrade (n=alberto@200.195.161.164) |
| 12:58.30 | pukkita | [TK]D-Fender: doing it... I have just thought, the Asterisk primary is set up as master, I'm using a E1 crossover cable, and the timing source for that span is set to be sourced from a "real" Telco primary. Should signalling in zapata.conf be pri_cpe???? |
| 12:59.02 | *** join/#asterisk samad (n=samad@116.71.186.147) |
| 12:59.29 | *** join/#asterisk nate187 (n=nate@gw.bigrivertel.net) |
| 12:59.33 | samad | skyNomad |
| 13:03.50 | *** join/#asterisk n3hxs (n=HAMming@151.196.87.132) |
| 13:07.52 | pukkita | [TK]D-Fender: I cannot debug an outgoing call from the PBX to the asterisk box right now, I can't change the PBX programming and the tech guy isn't reachable right now. I pasted an incoming pri call thorugh that same primary debug, and the zapata and zaptel .conf files: http://pastebin.com/m7f5072a7 |
| 13:10.11 | pukkita | [TK]D-Fender I wonder if it would be the pri signalling in zapata.conf that should be changed to pri_cpe, why are incoming calls working fine then? |
| 13:10.45 | pukkita | [TK]D-Fender or is it maybe the pridialplan parameters? |
| 13:11.09 | pukkita | [TK]D-Fender: I'd like to have some clues to try when the PBX guy comes... |
| 13:11.36 | *** join/#asterisk ar3dam (n=ar3dam@189.156.217.142) |
| 13:12.03 | ar3dam | hello there... |
| 13:12.46 | ar3dam | somebody can showme a tipical incoming call? |
| 13:13.14 | ar3dam | some example, pls? |
| 13:13.18 | ManxPower | ar3dam: it all depends on your dialplan. |
| 13:13.20 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:13.45 | *** join/#asterisk intralanman (n=lanman@216.40.253.210) |
| 13:14.47 | *** join/#asterisk tkbeat (n=tk@80.64.182.204) |
| 13:14.54 | tkbeat | hi |
| 13:15.12 | tkbeat | what can i do when a trink can not registered ? |
| 13:15.30 | ManxPower | ar3dam: here is a typical call on MY system: http://pastebin.ca/1169688 |
| 13:16.05 | ManxPower | tkbeat: fix the user or password |
| 13:17.03 | [TK]D-Fender | pukkita: Looks fine. We'll need to see thaqt call |
| 13:18.13 | [TK]D-Fender | ManxPower: Never show that to anyone you're not deliberately trying to scare away from #asterisk forever. |
| 13:18.40 | [TK]D-Fender | ar3dam: There is no such this as a "typical" incoming call. |
| 13:18.59 | ManxPower | [TK]D-Fender: now you know my evil plan! |
| 13:19.05 | pukkita | [TK]D-Fender I have apointed a test for tomorrow with the PBX guy |
| 13:19.06 | [TK]D-Fender | ar3dam: When a call arrives at * it hits an EXTENSION. What you do is entirely up to you |
| 13:19.30 | [TK]D-Fender | ManxPower: And you're macro is a psycho mess I'd shoot any other person for coding! |
| 13:19.47 | ManxPower | [TK]D-Fender: yes, dear. |
| 13:19.49 | pukkita | [TK]D-Fender: it isn't signalling already tried and the primary doesn't work. so it should be something else, maybe priindication or related? |
| 13:20.37 | [TK]D-Fender | pukkita: Could jsut be the way the PBX works. Think I've seen another like it. Dials a "null" number and then DTMF's the intended # |
| 13:20.38 | creativx | hehhe |
| 13:20.42 | creativx | ManxPower: greatest pastebin ever |
| 13:20.48 | [TK]D-Fender | your* |
| 13:21.03 | pukkita | [TK]D-Fender the PBX guy said it looks like the PBX somehow deoesn't detect the primary ready/available for making an outgoing call |
| 13:21.39 | pukkita | [TK]D-Fender: that sounds very feasible, is there any workaround for that null dialling? |
| 13:21.42 | [TK]D-Fender | pukkita: We'll see when you are actually in a position to DO something about this. |
| 13:22.18 | [TK]D-Fender | pukkita: if thats the case then you'll need to change it in your PBX. It's the one doing the dialing after all. |
| 13:23.14 | pukkita | [TK]D-Fender: I appointed the test for tomorrow morning, I'm in Spain and just want to know which things could I try as most of you will be sleeping :) |
| 13:23.36 | [TK]D-Fender | pukkita: Don't know me very well yet it seems ;) |
| 13:24.32 | pukkita | [TK]D-Fender: hahahaha :) the PBX is way old and looks like it has very few changeable parameters. |
| 13:26.03 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 13:26.48 | pukkita | [TK]D-Fender: so if that's the case and the PBX dialling cannot be changed, I'll be out of look? is that not addressable on the asterisk side? |
| 13:27.07 | pukkita | [TK]D-Fender s/look/luck/ |
| 13:27.10 | [TK]D-Fender | pukkita: if its post-dial DTMF then we should be able to deal with that in an IVR |
| 13:27.29 | ar3dam | ok, some can help to create a dialplan? |
| 13:27.51 | ManxPower | ar3dam: no. |
| 13:28.00 | ManxPower | ar3dam: Read the books, the docs, etc. |
| 13:28.09 | pukkita | [TK]D-Fender: mmmm you mean making the default extension answer first, then wait for the dtmf, then dial out? |
| 13:28.46 | [TK]D-Fender | pukkita: basically, yes |
| 13:28.59 | [TK]D-Fender | ar3dam: What do you want it to do? |
| 13:29.25 | pukkita | [TK]D-Fender what puzzles me if that same PBX primary port was connected to a telco primary what is working just fine plugged directly to the asterisk box, telco parameters unchanged (the span number 1 on that same configs) :-? |
| 13:29.45 | ar3dam | i try to make some basic, when incoming call, ring in a ext 100. |
| 13:29.51 | [TK]D-Fender | pukkita: I can't say any more without seeing first-hand what's happening |
| 13:30.11 | [TK]D-Fender | ar3dam: Go read the book and learn about *'s dialplan applications. |
| 13:30.14 | pukkita | [TK]D-Fender: ok, thanks a lot, now I know least what to llok for |
| 13:30.24 | pukkita | see ya guys |
| 13:30.26 | ar3dam | only to have a north. |
| 13:30.47 | ManxPower | ar3dam: Asterisk is not a PBX. Asterisk is a toolkit that lets you build PBX from scratch. |
| 13:32.17 | ar3dam | i know man, i learning asterisk, i make my first install asterisk, before i try with trixbox. |
| 13:32.43 | ManxPower | You learned nothing about Asterisk when you used Trixbox. Now go read the book. |
| 13:32.52 | ar3dam | but a this time, i have learning many thing of asterisk |
| 13:33.47 | *** join/#asterisk hsv-al (n=hsval@66.0.46.210) |
| 13:33.53 | ar3dam | u is right, with trixbox no learn nothing, because i installing asterisk with debian. |
| 13:34.40 | [TK]D-Fender | ar3dam: The dialplan is 95% of Asterisk. Setting up a SIP peer, analog or digital cards is nothing. Asterisk is about processing CALLS and that means the dialplan. |
| 13:35.12 | hsv-al | hello d-fender |
| 13:36.06 | ar3dam | ji ji ji ji .. ok. u have some web to read?, i find some books on amazon. |
| 13:36.17 | ManxPower | ~book |
| 13:36.18 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 13:36.43 | ar3dam | thks! jbot! .. |
| 13:39.09 | ar3dam | this pdf is to my begin in asterisk, very good :D ;) |
| 13:40.29 | *** part/#asterisk MindTheGap (n=MindTheG@201.17.149.252) |
| 13:40.53 | *** join/#asterisk MindTheGap (n=MindTheG@201.17.149.252) |
| 13:41.22 | creativx | hehe |
| 13:41.27 | creativx | thats a first.. thanking the jbot |
| 13:41.51 | [TK]D-Fender | creativx: Far from |
| 13:42.03 | *** join/#asterisk razu_ (n=razu@195.222.7.33) |
| 13:42.09 | [TK]D-Fender | ~areyouadog |
| 13:42.09 | jbot | Bark! Bark! |
| 13:42.13 | [TK]D-Fender | ~botsnack |
| 13:42.13 | jbot | :), [TK]D-Fender |
| 13:42.23 | creativx | heheh |
| 13:42.25 | [TK]D-Fender | pets jbot |
| 13:42.53 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 13:43.22 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.95.130) |
| 13:43.38 | *** join/#asterisk mgroman (n=miles@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 13:43.50 | mgroman | Hello, what is this res_cepstral? How does it differ from app_swift? |
| 13:48.25 | ManxPower | mgroman: what do the readmes say? |
| 13:49.50 | mgroman | There is no mention off app_swift in res_cepstral and vice versa... I think res_cepstral is the "official digium supported" module for the swift engine? |
| 13:51.09 | tkbeat | hi again |
| 13:54.05 | tkbeat | what do i have to check when i am unable to make a outgoing call ? trunk is registered but when i make an outgoing call after a while i get the all 'circuits are busy now' message ?! |
| 13:55.51 | ManxPower | tkbeat: HANGUPCAUSE, prlocaldialplan, pridialplan, priindication, switch type, incorrectly formatted Dial line, wring settings in zapata.conf, zaptel.conf, and there are about 40 more reasons I can think of. If you have a SPECIFIC question then ask it. |
| 13:56.17 | ManxPower | use pastebin to show us the output of a failed call. <-- DO THIS. |
| 13:57.08 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001a6682d7b6.cpe.net.cable.rogers.com) |
| 13:57.53 | *** join/#asterisk Drognan (n=Drognan@rrcs-24-129-157-34.se.biz.rr.com) |
| 13:59.11 | ManxPower | Well, I can't wait around here all day for a response. |
| 13:59.22 | tkbeat | hier is the paste |
| 13:59.22 | tkbeat | v |
| 13:59.24 | tkbeat | http://pastebin.com/d54c17540 |
| 13:59.28 | Drognan | I'm told I can get a device with 6 fxs ports that will then have an IP trunk that is more bandwidth efficient than 6 separate g729 connections, is there such a thing and who makes it? |
| 13:59.45 | ManxPower | tkbeat: I cannot and will not help you with a GUI Asterisk. |
| 14:00.14 | ManxPower | Drognan: no such things exist. |
| 14:00.32 | Drognan | it didn't sound right to me either |
| 14:01.12 | ManxPower | If you are using IAX2 there are trunking options to reduce bandwidth usage, but SIP (actually RTP) does not support that |
| 14:01.18 | [TK]D-Fender | ~freepbx |
| 14:01.19 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 14:01.21 | [TK]D-Fender | tkbeat: ^^^^^^^^^^ |
| 14:01.35 | ManxPower | There's 10 mins of my life I'll never get back. |
| 14:01.53 | anonymouz666 | heh. |
| 14:02.12 | Nugget | heh |
| 14:02.13 | [TK]D-Fender | Drognan: the one you make yourself. Setup a small server with a TDM card and IAX it up. |
| 14:03.17 | [TK]D-Fender | tkbeat: do "sip debug" from CLI and try the call again and pastebin it. |
| 14:03.49 | ManxPower | For people with NON-GUI questions, I'll be in #asterisk-cli |
| 14:05.47 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 14:08.04 | *** join/#asterisk mort_gib (n=mjensen@dsl-p4-177.gibconnect.com) |
| 14:09.03 | *** join/#asterisk MrNaz (n=naz@ppp59-167-169-142.lns1.mel4.internode.on.net) |
| 14:09.04 | *** join/#asterisk samad (n=samad@116.71.187.66) |
| 14:12.03 | mgroman | python |
| 14:12.46 | tkbeat | [TK]D-Fender, http://pastebin.com/m89bed80 |
| 14:15.47 | [TK]D-Fender | tkbeat: You have NOT set your system up properly to work from behind NAT -> From: "chris" <sip:231@192.168.30.200>;tag=as716d6664 |
| 14:16.12 | [TK]D-Fender | tkbeat: Go find a guide on how to do this in FreePBX. |
| 14:17.57 | *** join/#asterisk tobias (n=tobias@user-0c2hj2e.cable.mindspring.com) |
| 14:18.59 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 14:19.08 | samad | hello every one |
| 14:19.13 | samad | i need some help |
| 14:19.21 | samad | need to dail number using windowform |
| 14:19.32 | samad | iam developing small application in c# |
| 14:19.46 | ManxPower | [TK]D-Fender: Seems like a monday around here today |
| 14:20.08 | Wayhigh | oh dang.. was someone asking fender about freepbx again? |
| 14:20.40 | Wayhigh | repeat after me.. "Only Wayhigh is allowed to ask Fender about FreePBX." |
| 14:20.42 | samad | yeah |
| 14:20.52 | samad | Wayhigh |
| 14:21.11 | Wayhigh | yeah? |
| 14:21.19 | samad | sir any help |
| 14:21.29 | samad | i mean i need some help:) |
| 14:21.42 | samad | i was using tapi |
| 14:21.48 | Wayhigh | I think nerdvittles has something about click2call on it |
| 14:22.02 | samad | nervittles? |
| 14:22.10 | Wayhigh | certainly you can look at how he did his asteridex application |
| 14:22.14 | *** join/#asterisk cerbianguard (n=mark@c-71-232-59-18.hsd1.ma.comcast.net) |
| 14:22.34 | Wayhigh | nerdvittles.. it's a site run by a guy that does all kinds of cool applications for freepbx |
| 14:22.55 | samad | do i realy need asterisk plateform for just callback? |
| 14:23.46 | samad | ? |
| 14:24.04 | ManxPower | samad: We are not here to convince you to use Asterisk. |
| 14:24.10 | Wayhigh | do you need a cordless drill to put screws in? |
| 14:24.24 | samad | yeah |
| 14:25.00 | Wayhigh | samad: you could look at a voicexml/callxml solution I suppose |
| 14:25.10 | Wayhigh | but that would be hosted by someone else normally |
| 14:25.37 | samad | any other simple solution you think? |
| 14:25.54 | Wayhigh | simplest solution for callback? |
| 14:25.59 | samad | yes sir |
| 14:26.07 | Wayhigh | voice? |
| 14:26.08 | wizzy_ | I am using asterisk 1.6.0 from svn HEAD, with a Digium B410P quad ISDN card. "misdn show stacks" says "Port 4 Type TE Prot. PTP L2Link UP L1Link:UP Blocked:0 Debug:4". A call does this :- [9329@internal:3] Dial("SIP/1000-08346750", "mISDN/g:telkom/7879329") in new stack // P[ 4] There is no free channel on port (4) // WARNING[6166]: chan_misdn.c:3320 misdn_request: Could not Dial out on group 'telkom'. |
| 14:26.35 | samad | yes |
| 14:26.50 | wizzy_ | What is "no free channel" ? |
| 14:27.21 | Wayhigh | samad: I suppose you could write an app to monitor the serial port.. pull the CID and then call that back but bridging to another serialport is hard |
| 14:28.11 | samad | hmm |
| 14:28.13 | *** join/#asterisk MaliutaLap (n=nikolai@kiev.lusan.id.au) |
| 14:28.23 | samad | y serialport? |
| 14:29.51 | [TK]D-Fender | samad: Please define your use of the word "callback". |
| 14:30.30 | EmleyMoor | wizzy: It means the port you asked it to use is busy |
| 14:30.50 | *** join/#asterisk coppice (n=chatzill@61.157.17.210.dyn.pacific.net.hk) |
| 14:31.07 | samad | simple callback |
| 14:31.13 | samad | dail number |
| 14:31.16 | samad | make call |
| 14:31.41 | ManxPower | that is a call, not a call BACK. |
| 14:31.42 | wizzy_ | EmleyMoor: Its not really busy. What else might it be ? Do I have to register or something ? |
| 14:31.49 | EmleyMoor | samad: Under what circumstances? What initiates it? |
| 14:32.14 | samad | what is the simple solution to make call to customer using simple pc(window xp) with IP phone |
| 14:32.15 | [TK]D-Fender | samad: WHAT is dialing this "number"? What should your system do with it exactly? |
| 14:32.26 | wizzy_ | is there another way I can report the 'busyness' of the port ? |
| 14:32.46 | samad | just make a call and talk to customer |
| 14:32.53 | samad | using headphone mic |
| 14:32.54 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 14:33.02 | [TK]D-Fender | samad: So the customer is using a softphone on a windows PC? |
| 14:33.04 | ManxPower | samad: then just dial the call and talk to the destination. This is STANDARD Asterisk stuff. |
| 14:33.19 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.static.dsl.tele.dk) |
| 14:34.03 | samad | we dont know what customer is using |
| 14:34.21 | samad | it could be anything |
| 14:34.22 | [TK]D-Fender | samad: what is on the other side of the call? |
| 14:34.33 | samad | you mean customer? |
| 14:34.48 | [TK]D-Fender | samad: please describe what EACH END of the call is using. |
| 14:34.55 | samad | it could be cell phone, hardphone, softphone, |
| 14:34.57 | qp | sorry to sound bullish, but I have told samad everything already, about c# wrapping or simply using a trixbox + hud, he needs a simple call center setup on voip, 24x7 support desk |
| 14:35.10 | samad | qp |
| 14:35.15 | ManxPower | qp: but he is unable to express that. |
| 14:35.22 | samad | i just have meeting with mgmt |
| 14:35.24 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:35.24 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 14:35.24 | qp | we have had a long chat already :) |
| 14:35.31 | [TK]D-Fender | qp: Our condolences |
| 14:35.38 | samad | they not going for trixbox rightnow |
| 14:35.42 | ManxPower | qp: you poor thing. |
| 14:35.48 | qp | most activity #iaxclient has seen for months |
| 14:36.07 | samad | they just want callback from pc simply |
| 14:36.08 | [TK]D-Fender | samad: please describe what you want each end of this call to do and be using. |
| 14:36.20 | ManxPower | samad: Expect to spend a couple of months working with Asterisk before you manage to build a CRM system, which is what you are looking for. |
| 14:36.42 | ManxPower | samad: you are not looking for callback. |
| 14:36.44 | EmleyMoor | CallBACK? From where? |
| 14:36.45 | samad | i know writting crm takes time |
| 14:36.48 | qp | samad, you just want a phone then |
| 14:36.58 | ManxPower | samad: callback = user dials PBX, PBX hangs up and calls them back. This is a call back. |
| 14:37.18 | samad | sorry about wrong term callback |
| 14:37.35 | samad | i am trying to explain make a call |
| 14:37.36 | ManxPower | samad: if you continue to use the wrong term then nobody will help you. |
| 14:37.41 | [TK]D-Fender | samad: please stop and answer my question. |
| 14:37.46 | samad | ok |
| 14:38.02 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 14:38.11 | qp | I think samad just needs an app that pops info from a crm when a call comes in and also a headset for making calls, but all via existing lines, and nothing at all to do with Asterisk |
| 14:38.14 | [TK]D-Fender | samad: and I expect to hear what TECHNOLOGY is used on EACH SIDE of this. |
| 14:38.24 | ManxPower | qp: but he has not SAID that. |
| 14:38.25 | [TK]D-Fender | qp: let him answer. |
| 14:38.31 | qp | ok |
| 14:38.45 | samad | qp is right |
| 14:38.50 | samad | because i was using tapi |
| 14:38.57 | samad | the problem i faced |
| 14:38.58 | [TK]D-Fender | samad: I want to see YOU say exactly what it is you want. |
| 14:39.26 | samad | i want to make a call using my application |
| 14:39.32 | samad | like dailer |
| 14:39.39 | [TK]D-Fender | samad: What application? What protocols? What hardware? |
| 14:39.40 | wizzy_ | "misdn show stacks" reports fine (port 4 up). "misdn show channels" says "Channel List: (nil)". Where should I start looking ? |
| 14:39.47 | ManxPower | samad: Good. Now are you going to change your story again or are you going to stick with it this time? |
| 14:39.50 | *** join/#asterisk vale-ICS (n=vale@icsnet.demon.co.uk) |
| 14:40.10 | samad | this is very simple application i am developing in dot net |
| 14:40.16 | samad | i used tapi |
| 14:40.31 | [TK]D-Fender | samad: Fine, so you've got this hooked to a winmodem, right? |
| 14:40.34 | *** join/#asterisk seanmh (i=seanmh@216.31.101.25) |
| 14:40.34 | samad | but reciver wasnt able to hear me |
| 14:40.40 | ManxPower | I give up. [TK]D-Fender, you can handle this guy. |
| 14:40.42 | samad | yes |
| 14:41.01 | [TK]D-Fender | samad: then go ask for help in a programming channel. |
| 14:41.22 | [TK]D-Fender | samad: I fail to see what this has to do with * in any way. |
| 14:41.36 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:41.47 | samad | you know simple dailer application |
| 14:41.51 | samad | or you dont?> |
| 14:41.57 | samad | using modem? |
| 14:42.04 | ManxPower | samad: a simple dialer application has nothing to do with asterisk or a modem. |
| 14:42.51 | samad | any one who wrote simple dailer application to make call |
| 14:43.21 | samad | leave asterisk... or callxml solution |
| 14:44.01 | samad | qp told me very good solution trixbox one ... but my company is not going for that solution for a while |
| 14:44.13 | [TK]D-Fender | samad: There are dialers for *. Go visit the WIKI and look them up. |
| 14:44.21 | ManxPower | samad: Most of us have done dialer applications. |
| 14:44.29 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:44.44 | ManxPower | Mine makes a call to the user to notify them they have voicemail. |
| 14:44.44 | Wayhigh | samad: there are other ways but they're probably more difficult than an asterisk solution. |
| 14:44.46 | samad | WIKI? |
| 14:44.56 | *** join/#asterisk ChicagoBud (n=Bud@38.104.180.122) |
| 14:45.04 | ManxPower | ~wiki |
| 14:45.06 | ManxPower | ~mailinglist |
| 14:45.07 | jbot | [~mailinglist] The Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search. |
| 14:45.08 | ManxPower | ~docs |
| 14:45.08 | jbot | methinks docs is for basic documentation of Asterisk ask see http://voip-info.org/ (~voip-info) and TheBook (~book) |
| 14:45.09 | ManxPower | ~book |
| 14:45.10 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 14:45.11 | Wayhigh | sure they're simpler.. but that comes at the expense of your manpower to set it up |
| 14:45.21 | samad | ok |
| 14:45.34 | samad | thanks all for your help |
| 14:46.07 | Wayhigh | samad: can you at least do us a favor and google "asterisk click2call"? |
| 14:46.16 | micheluntu | I'm trying to use extconfig.conf to load sip peers from database |
| 14:46.44 | samad | ok |
| 14:47.01 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 14:47.15 | [T]ank | what is a good rate to pay for residential sip? |
| 14:47.45 | micheluntu | someone get it work? |
| 14:48.07 | wizzy_ | micheluntu: I have used voiceone to do that |
| 14:48.41 | micheluntu | wizzy_: I'm searching for voiceone.. what is it? |
| 14:49.01 | micheluntu | like trixbox? |
| 14:49.58 | ManxPower | [T]ank: 2/cents/min |
| 14:50.13 | wizzy_ | micheluntu: I believe most asterisk 'distributions' are ISOs - because of all the tricky drivers. Thus they come with their own management interfaces, like trixbox and elastik |
| 14:50.38 | wizzy_ | voiceone appears to be a frontend for 'your' asterisk setup |
| 14:50.49 | wizzy_ | i.e. is not wedded to an iso |
| 14:51.01 | [T]ank | ManxPower: ok.. thats what I am paying... just making sure. |
| 14:51.18 | *** join/#asterisk jadams_ (n=jadams@rrcs-24-73-180-234.se.biz.rr.com) |
| 14:51.19 | ManxPower | ~zeeek |
| 14:51.20 | jbot | rumour has it, zeeek is someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 14:51.38 | jadams_ | can anyone tell me a piece of software I can use to play GSM files other than Apple Quicktime? |
| 14:52.12 | ManxPower | jadams_: windows media player, but it needs to have a WAV49 wrapper |
| 14:52.43 | jadams_ | ManxPower, what is the wav49 wrapper? A codec, or something that asterisk puts into the gsm file itself? |
| 14:53.01 | wizzy_ | can anyone enlighten me on my disconnect between "misdn show stacks" (works) and "misdn show channels" ("Channel List: (nil)") |
| 14:53.16 | ManxPower | jadams_: It is a GSM file in a WAV wrapper. Asterisk calls it WAV49 in voicemail.conf |
| 14:53.33 | jadams_ | ManxPower, thanks a ton |
| 14:53.58 | ManxPower | jadams_: I assume you REALLY want to just listen to voicemail that's been sent via e-mail, but of course you made us work for it by not saying. |
| 14:54.35 | jadams_ | ManxPower, not exactly |
| 14:54.39 | jadams_ | ManxPower, sorry, I'll explain |
| 14:54.47 | jadams_ | it's just being used for call recording and playback |
| 14:55.03 | jadams_ | and it was using wav, but that took up too much space comparatively |
| 14:55.09 | *** join/#asterisk Assimilate (n=Assimila@216.83.78.108) |
| 14:55.28 | jadams_ | so we switched to GSM, only to discover that nothing can play straight gsm without extra software installation |
| 14:55.31 | jadams_ | this is our first asterisk install ever |
| 14:56.15 | wizzy_ | micheluntu: sippeers => mysql,voiceone,sip_buddies <-- mysql,database,table |
| 14:56.16 | micheluntu | wizzy_: so can you show me your extconfig.conf ? |
| 14:57.27 | Wayhigh | jadams: http://www.voip-info.org/wiki-Asterisk+sound+files |
| 14:58.12 | micheluntu | wizzy_: thanx! this is my "sip_buddies" table http://pastebin.com/d20a3d7b3 looks like your? |
| 14:59.32 | jadams_ | Wayhigh, thanks |
| 14:59.58 | Wayhigh | jadams: no problem.. it is a confusing issue.. got me too for a bit |
| 15:00.36 | wizzy_ | I added mine after - http://pastebin.com/m1a3523b0 |
| 15:01.36 | *** join/#asterisk cjk (n=cjk@vodsl-10830.vo.lu) |
| 15:01.51 | micheluntu | wizzy_: thanks, differs |
| 15:02.37 | wizzy_ | micheluntu: it might - but I think that all asterisk is looking for are the fields similarly named in the equiv. text file |
| 15:02.40 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:03.15 | micheluntu | I think so, but I can't see my peers with sip show peers |
| 15:03.33 | wizzy_ | the other stuff might be voiceone stuff |
| 15:03.57 | wizzy_ | did you fix your remote db issues ? |
| 15:04.35 | wizzy_ | add some logging to mysql - maybe you will see the query |
| 15:05.30 | wizzy_ | *sigh* - no misdn users ? sux to be in 3rd world |
| 15:05.53 | dominic1 | is it possible to generate a event if why pickup was successfully (in ami) I only get a zombie hangup of the original destination? Hope somebody can help me... |
| 15:06.17 | *** part/#asterisk ManxPower (n=manxpowe@25.sub-75-249-159.myvzw.com) |
| 15:07.10 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 15:07.54 | *** join/#asterisk cerbianguard (n=mark@c-71-232-59-18.hsd1.ma.comcast.net) |
| 15:09.25 | *** join/#asterisk russellb (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:09.25 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:11.25 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 15:12.17 | *** join/#asterisk emiller (n=ed@st001.nfdc.net) |
| 15:13.50 | emiller | Hello all, when i am making a call, and try to enter a number, asterisk doesn't see it, any place i can look at? |
| 15:14.18 | emiller | sorry, when i am currently on a call, and try to input |
| 15:14.36 | *** part/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 15:15.19 | [TK]D-Fender | emiller: fix your dtmfmode setting |
| 15:15.29 | emiller | thanks D-Fender |
| 15:17.03 | *** join/#asterisk MrNaz (n=naz@ppp59-167-169-142.lns1.mel4.internode.on.net) |
| 15:17.45 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:17.56 | emiller | hmm, my dtmfmode = rfc2833, that should be correct, no? |
| 15:18.12 | hsv-al | hmmm the great debate........a pint of blue berries, and pom pomegranite juice for lunch........or a $10.00 splurge of taco bell, 5 big taste tacos, 2 orders of nachos w/ cheese, ranchero chicken soft taco, and a large diet pepsi.......... |
| 15:18.31 | emiller | taco bell. |
| 15:18.41 | qp | every time, and I'm in England |
| 15:21.17 | [TK]D-Fender | emiller: Says who? You offer no details. |
| 15:21.52 | [TK]D-Fender | emiller: If that choice is always correct, why is it a choice? Why not just make an option like "works=yes" so people don't make the wrong choice. |
| 15:22.52 | [TK]D-Fender | hsv-al: "and a large diet pepsi.........." because you've gotta watch your weight |
| 15:23.03 | hsv-al | thats not the point as everyone assumes |
| 15:23.09 | hsv-al | regular soda is disgusting, to sugary, corn syrup hell |
| 15:23.16 | hsv-al | diet tastes better |
| 15:23.18 | [TK]D-Fender | hsv-al: Sarcasm is ALWAYS the point |
| 15:24.02 | [TK]D-Fender | hsv-al: And yes I prefer "diet" drinks for the taste myself. |
| 15:24.44 | *** join/#asterisk hubguruJR (n=hubguruJ@mail.ntegratedsolutions.com) |
| 15:24.45 | hsv-al | i have psychosis, if i eat fast food |
| 15:24.53 | hsv-al | its mandatory i run 5-6 miles at night |
| 15:25.00 | hsv-al | if i eat bb's+pom juice, its only 3-4 heh |
| 15:25.04 | [TK]D-Fender | hsv-al: But could we identify the difference? |
| 15:28.33 | hubguruJR | Hi all |
| 15:28.50 | hubguruJR | any asterisk udptl to cisco gateway experts out there? |
| 15:31.17 | *** join/#asterisk deeperror (n=deeperro@76.226.177.255) |
| 15:31.20 | deeperror | ~grandstream |
| 15:31.21 | jbot | it has been said that grandstream is the Yugo of VoIP hardware. Run. Run away now. |
| 15:32.04 | bminish | got a weird issue with an aastra 480i it hangs up incoming calls after about 5 seconds on outgoing calls it's fine |
| 15:32.07 | Nugget | heh |
| 15:32.31 | [TK]D-Fender | bminish: PASTEBIN is your friend... |
| 15:33.23 | bminish | [TK]D-Fender, I know about pastebin, just not got anything to pate in there yet.. |
| 15:33.55 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:34.18 | [TK]D-Fender | bminish: then go get CLI + SIP DEBUG. |
| 15:35.45 | wizzy_ | On svn HEAD, in /etc/asterisk/misdn.conf I have "misdn_init=/etc/misdn-init.conf". Should I have "misdn_init=/etc/mISDN.conf" (the new, shiny, XML config file ?) |
| 15:36.10 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 15:36.38 | *** join/#asterisk ipso (n=ipso@S0106005004c32d38.ok.shawcable.net) |
| 15:38.26 | x86 | how do i determine if both APIC as well as ACPI are enabled in my running kernel? |
| 15:39.17 | x86 | sangoma keeps going in circles as to why i'm getting massive amounts of overruns (and thus, horrible quality), on a single span (on a dual-port T1 card but only one port in use) |
| 15:39.31 | x86 | it's very frustrating that their support can't help me properly |
| 15:39.32 | bminish | [TK]D-Fender, here http://pastebin.com/d2fb8dabe |
| 15:43.14 | *** part/#asterisk samad (n=samad@116.71.187.66) |
| 15:43.23 | [TK]D-Fender | bminish: pastebin both sip.conf entries and describe where they are located relative to the server |
| 15:44.45 | bminish | [TK]D-Fender, here http://pastebin.com/d47663a2f they are both on the same local lan, no nat |
| 15:45.20 | bminish | the phone radio<202> has the same issues no matter how the call arrives at the asterisk box <201> has no issues |
| 15:46.01 | [TK]D-Fender | bminish: What ver of *? |
| 15:46.20 | bminish | [TK]D-Fender, 1.4.21.2 |
| 15:46.42 | [TK]D-Fender | bminish: "notifyringing=yes" <- remove. Set "nat=no", "canreinvite=no" and "type=peer" for both of them, and set ONE codec only for each |
| 15:46.51 | bminish | built from source and running on centos x86_64 |
| 15:51.36 | *** join/#asterisk hi365_m (n=hi365@213.151.56.78) |
| 15:52.07 | bminish | [TK]D-Fender, done that reloaded sip in * restarted phones, same issue |
| 15:52.40 | [TK]D-Fender | bminish: Ok, not sure from this point... |
| 15:53.11 | bminish | ok, really odd thing is that this came out of the blue |
| 15:53.42 | bminish | BTW what does notifyringing=yes do |
| 15:58.55 | zamba | what's that feature that uses no bandwidth when there's no talking? |
| 15:59.23 | zamba | it doesn't send any packets |
| 15:59.25 | zamba | what's it called? |
| 15:59.33 | NoxIn- | voice activity detection ? |
| 15:59.35 | NoxIn- | VAD |
| 15:59.39 | *** join/#asterisk azeey (n=addisu@c-98-223-159-126.hsd1.in.comcast.net) |
| 15:59.47 | zamba | aight, ok |
| 15:59.52 | zamba | how do i turn it on and off? |
| 16:00.30 | zamba | it's an endpoint configuration, maybe? |
| 16:00.36 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 16:00.41 | *** join/#asterisk HKhan (i=hkhan@hamzah.is.an.evil.g3nius.net) |
| 16:04.08 | NoxIn- | zamba: I think it's not supported on asterisk so you can't enable it |
| 16:04.16 | NoxIn- | but my information on this may be outdated |
| 16:07.10 | *** join/#asterisk Entr4nced (n=IMG001@67-129-213-39.dia.static.qwest.net) |
| 16:12.08 | [TK]D-Fender | zamba: "what's that feature that uses no bandwidth when there's no talking?" - Not compatible with * |
| 16:12.40 | zamba | ok |
| 16:12.47 | zamba | so it's basically up to the endpoints? |
| 16:13.27 | *** join/#asterisk Sargun (n=xbmodder@atarack/staff/sargun) |
| 16:13.28 | micheluntu | I have : exten => 2109,1,Dial(SIP/2109) exten => 2106,1,Dial(SIP/2106) in extensions.conf |
| 16:14.02 | micheluntu | could I set exten => XXXX,1,Dial(SIP/${SOMETHING}) ?? |
| 16:15.45 | [TK]D-Fender | micheluntu: Close. Go read up on dialplan PATTERNS. |
| 16:16.13 | [TK]D-Fender | micheluntu: And note that that idea will allow you do attempt to dial things you don't want them to. |
| 16:17.37 | micheluntu | [TK]D-Fender: ok, thanks |
| 16:19.13 | micheluntu | [TK]D-Fender: i'm only testing some scenarios |
| 16:19.50 | [TK]D-Fender | micheluntu: The idea isn't too bad, and you're really close. |
| 16:23.35 | *** part/#asterisk hubguruJR (n=hubguruJ@mail.ntegratedsolutions.com) |
| 16:32.37 | *** join/#asterisk mvanbaak_ (n=michiel@vanbaak.xs4all.nl) |
| 16:41.18 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 16:45.09 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.95.130) |
| 16:45.49 | *** join/#asterisk qp| (n=mg@82-71-8-45.dsl.in-addr.zen.co.uk) |
| 16:45.54 | *** join/#asterisk Hertzy3 (n=ahertz@ahertz.atlantic.net) |
| 16:46.05 | Hertzy3 | ~centos52bug |
| 16:46.05 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889 |
| 16:47.26 | exothermc | There really isn't anything that can be done SIP to SIP about echo is there? |
| 16:47.44 | [TK]D-Fender | exothermc: Purely SIP, no. |
| 16:48.21 | [TK]D-Fender | Katty: http://ecoworldly.com/2008/08/12/big-cats-banned-from-australia/ |
| 16:48.23 | jeev | hi Fender |
| 16:49.20 | wizzy_ | on an idle system, should "misdn show channels" say "Channel List: (nil)" ? |
| 16:53.31 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 16:55.51 | *** join/#asterisk bradleyprice86 (n=chatzill@fw.datafax.net) |
| 16:56.21 | *** join/#asterisk cerbianguard (n=mark@c-71-232-59-18.hsd1.ma.comcast.net) |
| 16:56.59 | cerbianguard | Ok guys, heres another problem |
| 16:57.02 | cerbianguard | in the logs, I get |
| 16:57.03 | cerbianguard | loader.c: chan_zap.so: load_module failed, returning -1 |
| 16:57.16 | Qwell | cerbianguard: there should be errors/warnings before that |
| 16:57.28 | [TK]D-Fender | cerbianguard: do "ztcfg -vvvv" before starting *. then start * manually |
| 16:57.41 | cerbianguard | I'll try that |
| 16:58.39 | jeev | tonight is the night is the night of love |
| 16:58.47 | cerbianguard | "ztcfg -vvvv" give me "Channel map: |
| 16:58.47 | cerbianguard | Channel 01: FXS Kewlstart (Default) (Slaves: 01) |
| 16:58.47 | cerbianguard | Channel 02: FXS Kewlstart (Default) (Slaves: 02) |
| 16:58.47 | cerbianguard | Channel 03: FXS Kewlstart (Default) (Slaves: 03) |
| 16:58.47 | cerbianguard | Channel 04: FXS Kewlstart (Default) (Slaves: 04) |
| 16:58.47 | cerbianguard | 4 channels configured. |
| 16:58.49 | cerbianguard | ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 16:58.51 | cerbianguard | " |
| 16:59.08 | [TK]D-Fender | cerbianguard: PASTEBIN! |
| 16:59.12 | [TK]D-Fender | ~pb |
| 16:59.13 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines 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 |
| 16:59.26 | [TK]D-Fender | cerbianguard: and next please pastebin your zapata.conf |
| 16:59.54 | [TK]D-Fender | cerbianguard: and did ztcfg give you the "ZT_CHANCONFIG failed on channel 1: No such device or address (6)" error? |
| 17:00.05 | cerbianguard | yes |
| 17:01.04 | [TK]D-Fender | cerbianguard: pastebin your zapata.conf and zaptel.conf. |
| 17:01.32 | jeev | Fender, why do i consistently get stuff like this *CLI> [Aug 13 09:57:53] NOTICE[16468]: chan_sip.c:15851 sip_poke_noanswer: Peer 'chrisviatalk' is now UNREACHABLE! Last qualify: 1392, i have a 3000ms qualify.. are they that bad? |
| 17:01.56 | [TK]D-Fender | jeev: "DUH" |
| 17:02.02 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:02.18 | jeev | scratches head and waits to get yelled at |
| 17:03.43 | cerbianguard | sorry for the noob mistakes...I learn fast though |
| 17:03.44 | cerbianguard | ; |
| 17:03.44 | cerbianguard | ; Zapata telephony interface |
| 17:03.44 | cerbianguard | ; |
| 17:03.44 | cerbianguard | ; Configuration file |
| 17:03.44 | cerbianguard | [trunkgroups] |
| 17:03.46 | cerbianguard | ; |
| 17:03.48 | cerbianguard | ; Trunk groups are used for NFAS or GR-303 connections. |
| 17:03.50 | cerbianguard | ; |
| 17:03.52 | cerbianguard | ; Group: Defines a trunk group. |
| 17:03.54 | cerbianguard | ; group => <trunkgroup>,<dchannel>[,<backup1>...] |
| 17:03.56 | cerbianguard | ; |
| 17:03.58 | cerbianguard | ; trunkgroup is the numerical trunk group to create |
| 17:04.00 | cerbianguard | ; dchannel is the zap channel which will have the |
| 17:04.02 | cerbianguard | ; d-channel for the trunk. |
| 17:04.04 | cerbianguard | ; backup1 is an optional list of backup d-channels. |
| 17:04.04 | _ShrikE | omg |
| 17:04.06 | cerbianguard | ; |
| 17:04.08 | cerbianguard | ;trunkgroup => 1,24,48 |
| 17:04.10 | cerbianguard | ; |
| 17:04.12 | cerbianguard | ; Spanmap: Associates a span with a trunk group |
| 17:04.14 | cerbianguard | ; spanmap => <zapspan>,<trunkgroup>[,<logicalspan>] |
| 17:04.16 | cerbianguard | ; |
| 17:04.18 | cerbianguard | ; zapspan is the zap span number to associate |
| 17:04.20 | cerbianguard | ; trunkgroup is the trunkgroup (specified above) for the mapping |
| 17:04.22 | cerbianguard | ; logicalspan is the logical span number within the trunk group to use. |
| 17:04.23 | DagMoller | lol |
| 17:04.24 | cerbianguard | ; if unspecified, no logical span number is used. |
| 17:04.25 | *** kick/#asterisk [cerbianguard!i=north@pdpc/sponsor/digium/Qwell] by Qwell (pastebin.com) |
| 17:05.14 | russellb | heh, cat configs/zapata.conf.sample > #asterisk |
| 17:05.43 | *** part/#asterisk micheluntu (n=michele@dhcp197-193.cu.mi.it) |
| 17:05.52 | Sargun | hehe |
| 17:06.59 | Qwell | /exec -o cat configs/zapata.conf,sample |
| 17:07.00 | Qwell | :D |
| 17:07.26 | Qwell | I love his comment before the paste too |
| 17:08.12 | [TK]D-Fender | Qwell: aFTER i ALREADY YELLED AT HIM AND LINKED HIM TO IT.. |
| 17:08.20 | Qwell | naturally |
| 17:08.21 | [TK]D-Fender | (and fixed my darn caps-lock) |
| 17:10.33 | *** join/#asterisk btfx (n=chatzill@c-76-19-45-11.hsd1.ma.comcast.net) |
| 17:11.20 | *** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
| 17:11.41 | jeev | hm |
| 17:11.59 | *** join/#asterisk NOT_guru (n=NOT_guru@24-241-103-142.static.stls.mo.charter.com) |
| 17:13.07 | *** join/#asterisk famicom (i=famicom@c51447b09.cable.wanadoo.nl) |
| 17:13.19 | famicom | I got a stupid question |
| 17:14.00 | [TK]D-Fender | makes another toast to "truth in advertising" |
| 17:14.09 | russellb | ~ask |
| 17:14.10 | jbot | [ask] Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 17:14.19 | famicom | are phone numbers to astrisk as ip adresses are to apache? |
| 17:14.28 | [TK]D-Fender | famicom: No. |
| 17:14.52 | russellb | IP addresses are IP addresses to both Asterisk and Apache |
| 17:15.04 | [TK]D-Fender | famicom: "phone number" by the most basic generalization implies access to the PSTN. This is not implicit |
| 17:15.20 | [TK]D-Fender | russellb: parse error ;) |
| 17:15.35 | famicom | thanks for the acronymns |
| 17:15.37 | russellb | falls over |
| 17:15.48 | famicom | googles to figure out what it's about |
| 17:16.21 | tzafrir_laptop | wikipedia tends to be handy for telephony words you don't understand |
| 17:16.25 | tzafrir_laptop | ~wiki PSTN |
| 17:17.25 | famicom | well, im trying to fill the gap between internets < - > phone lines |
| 17:17.46 | [TK]D-Fender | famicom: * can process calls to and from MANY different technolgies |
| 17:17.52 | famicom | yup |
| 17:18.03 | famicom | PTSN SIP, the works |
| 17:18.07 | famicom | BUT |
| 17:18.16 | [TK]D-Fender | famicom: Analog lines, BRI, T1, E1, VoIP protocols (SIP, H.323, IAX, etc) |
| 17:18.33 | famicom | softphone > sip > ????? > landlines |
| 17:19.03 | [TK]D-Fender | famicom: softphone > SIP > * > (somehting line a PCI card with FXO ports on it) > analog line |
| 17:19.11 | [TK]D-Fender | like* |
| 17:19.17 | famicom | ah |
| 17:19.18 | famicom | thanks |
| 17:19.56 | famicom | another stupid question |
| 17:20.03 | [TK]D-Fender | famicom: Can you narrow down the scop of your needs? How many ports? Only really thinkking about soft-phones for "users"? |
| 17:20.08 | [TK]D-Fender | scope* |
| 17:20.14 | famicom | why can services like voipbuster etc offer such cheap landlines |
| 17:20.56 | famicom | [TK]D-Fender: for now i'm just trying to find a way to hook up 3 phonelines + 2 fax lines in one central management place |
| 17:21.03 | [TK]D-Fender | famicom: because they have huge inexpensive pipes, and they have X channels and Y customers where Y is a nice multiple of X. |
| 17:21.04 | famicom | which isn't really that hard to do |
| 17:21.53 | famicom | hmmm |
| 17:22.14 | famicom | replaces his current tubes with pipes |
| 17:24.04 | famicom | anyhow |
| 17:24.42 | famicom | you just mentioned that these companies have phat pipes |
| 17:25.02 | famicom | i take it that you don't mean TCP/IP connections but PTSN |
| 17:25.09 | famicom | PSTN* |
| 17:28.04 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 17:29.01 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 17:29.58 | *** join/#asterisk watchy (n=watchy@adsl-69-152-41-251.dsl.ltrkar.swbell.net) |
| 17:30.11 | *** join/#asterisk VaNNi (n=VaNNi___@38.98.61.143) |
| 17:30.12 | watchy | any reason why i would get this 203/202 192.168.0.122 D 5060 Unmonitored |
| 17:30.20 | *** part/#asterisk HKhan (i=hkhan@hamzah.is.an.evil.g3nius.net) |
| 17:31.08 | [TK]D-Fender | famicom: big internet BW + survivable % of free channels |
| 17:31.27 | [TK]D-Fender | watchy: multiple phones behind a remote NAT improperly configured |
| 17:31.42 | watchy | its not behind nat |
| 17:31.47 | watchy | its a local phone |
| 17:31.48 | hsv-al | hi japanese nintendo |
| 17:31.51 | hsv-al | famicom whats up |
| 17:31.59 | famicom | not too much |
| 17:32.14 | watchy | i guess i will check the conf of the phone next time im out on site |
| 17:32.19 | watchy | its working fine though |
| 17:32.20 | famicom | trying to figure out the magic that is voip |
| 17:32.32 | watchy | they have wildblue so remotely looking at the phone is super slow |
| 17:32.32 | famicom | plus, i want to get my own EFAX type setup |
| 17:32.39 | famicom | cause they're a ripoff |
| 17:33.40 | *** join/#asterisk pta200 (n=paolo@dsl211-220-225.nyc1.dsl.speakeasy.net) |
| 17:33.45 | mgroman | Writing to STDERR will display in the *CLI> ? |
| 17:34.49 | *** join/#asterisk vale-ICS (n=vale@boyne.demon.co.uk) |
| 17:34.58 | Qwell | mgroman: from AGI, I believe so |
| 17:35.11 | pta200 | How do you do date comparisons in a diaplan since iftime and gotoiftime don't take the year as a parameter? |
| 17:35.32 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 17:38.07 | watchy | hey tk: i got a customer that says their greetings arent coming up when someone dials their voicemail |
| 17:38.13 | watchy | is there a special setting for that |
| 17:38.24 | [TK]D-Fender | watchy: |
| 17:38.32 | [TK]D-Fender | watchy: "core show application voicemail" |
| 17:39.43 | watchy | i love you |
| 17:39.50 | seanbright | i love you too |
| 17:39.53 | seanbright | oh |
| 17:39.56 | seanbright | you weren't talking to me |
| 17:39.59 | seanbright | we'll talk later. |
| 17:40.08 | watchy | haha |
| 17:40.13 | famicom | oooh |
| 17:40.20 | famicom | I got another question |
| 17:40.43 | [TK]D-Fender | 42 |
| 17:40.47 | famicom | no |
| 17:40.50 | watchy | thats the greatest answer ever tk |
| 17:40.59 | famicom | yes |
| 17:41.03 | famicom | but what was the question! |
| 17:41.14 | [TK]D-Fender | ascii stupid question, get a stupid ansi |
| 17:41.43 | Nugget | 42 is the number I use on my car. |
| 17:41.43 | Paige_ | anyone have any php agi scripts that process cdr records that i could see? |
| 17:42.02 | Nugget | http://macnugget.org/photos/tws200806/IMG_14905 |
| 17:42.07 | famicom | | < joke |
| 17:42.08 | famicom | | |
| 17:42.08 | famicom | | < -- you |
| 17:42.08 | famicom | |_____ |
| 17:44.45 | pta200 | Any throughts on how to do a date comparisons in a diaplan since iftime and gotoiftime don't take the year as a parameter? |
| 17:44.56 | *** part/#asterisk wizzy_ (n=andyr@musselcracker.aims.ac.za) |
| 17:45.48 | [TK]D-Fender | pta200: obvious answer, do it OUTSIDE of *. |
| 17:46.19 | *** part/#asterisk azeey (n=addisu@c-98-223-159-126.hsd1.in.comcast.net) |
| 17:47.23 | *** join/#asterisk jasonwoot (n=jasonrot@user-69-73-40-171.knology.net) |
| 17:48.22 | Paige_ | can anyone give a good reason to stick with asterisk instead of changing over to freeswitch? |
| 17:49.18 | Paige_ | and no, a few asterisk books in print is not a good reason |
| 17:50.07 | [TK]D-Fender | Paige_: On what basis are we supposed to convince you? |
| 17:50.31 | [TK]D-Fender | Paige_: What you like / not like about *? What do you need out of either? |
| 17:50.36 | Paige_ | features, stability and functionality |
| 17:50.38 | jasonwoot | freeswitch does have fewer syllables... |
| 17:50.46 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 17:50.51 | [TK]D-Fender | jasonwoot: But more letters |
| 17:50.56 | Paige_ | * still has deadlock issues |
| 17:51.00 | DarienWork | and it can't be shortened to a single ASCII character |
| 17:51.02 | Paige_ | and core dumps a lot |
| 17:51.09 | [TK]D-Fender | DarienWork: Good point. |
| 17:51.17 | [TK]D-Fender | Paige_: Not in my world it doesn't |
| 17:51.21 | jasonwoot | touche |
| 17:51.51 | [TK]D-Fender | Paige_: Perhaps you could could describe your needs a little better. Maybe even a lot. |
| 17:51.51 | jasonwoot | you'll all want to switch to my new open source pbx soon anyway |
| 17:52.10 | jasonwoot | ampersand... it's a whole key higher than asterisk |
| 17:52.11 | Paige_ | i need a realtime conferencing server for one |
| 17:52.27 | Paige_ | asterisk realtime support is very limited |
| 17:52.38 | Paige_ | especially for extensions |
| 17:52.44 | [TK]D-Fender | Paige_: Where I come from you get locked up in a soft room for having a "conference for one" |
| 17:52.59 | *** join/#asterisk nny_1 (n=nny_1@64.203.237.47) |
| 17:53.15 | DarienWork | I find that if I have a conference involving anyone other than myself, I get frustrated with how much more awesome I am than everyone else |
| 17:53.16 | Paige_ | for one, i need a real time conferencing server |
| 17:53.20 | Paige_ | is that better? |
| 17:53.37 | DarienWork | much |
| 17:53.39 | [TK]D-Fender | Paige_: How many callers? what kind of channels? |
| 17:53.40 | Paige_ | ok |
| 17:54.03 | Paige_ | as many callers as needed, sip channels |
| 17:54.05 | [TK]D-Fender | PaigeHow many over users (outside of in a single conference)? |
| 17:54.12 | Paige_ | able to be created on the fly |
| 17:54.19 | jasonwoot | <[TK]D-Fender> , thanks for the additional metaphor, always looking for more of those |
| 17:54.24 | [TK]D-Fender | Paige_: "as many callers as needed" so what... 5 -6 million? |
| 17:54.32 | Paige_ | 20, 30 |
| 17:55.23 | [TK]D-Fender | Paige_: total user count? |
| 17:55.36 | Paige_ | could be as many as 500 on a server |
| 17:55.46 | Paige_ | in various rooms |
| 17:56.20 | Paige_ | with ability to moderate the conferences via a web portal |
| 17:56.24 | [TK]D-Fender | Paige_: thats a lot of MUX going on... |
| 17:56.35 | Paige_ | yes it is |
| 17:56.50 | [TK]D-Fender | Paige_: 500 people worth of conferencing would best be plit up |
| 17:56.53 | [TK]D-Fender | split* |
| 17:57.55 | Paige_ | right now a client uses 3 servers to achieve this. 2 asterisk and one opensip |
| 17:58.48 | [TK]D-Fender | Paige_: Feel free to go FreeSWITCH a go. If it scales to your needs, whatever, more power to you. |
| 17:59.16 | Paige_ | i want to go freeswitch but my boss wants to stay with asterisk |
| 17:59.20 | jasonwoot | I can't imagine a 500 person conference... |
| 17:59.48 | Paige_ | and frankly, support in this channel is hard to come by sometimes |
| 17:59.51 | jasonwoot | my favorite thing in the whole world is when I have 30 people on a conf and the d channel on a t1 resets, and they all go bye bye |
| 17:59.55 | deeperror | jasonwoot, that would be a circus |
| 18:00.18 | Paige_ | jasonwoot, that would only be 23 clients |
| 18:00.30 | jasonwoot | "... has left the conference" "... has left the confrerence" |
| 18:00.46 | Paige_ | t1 = 23 bri and 1 d channel |
| 18:01.01 | tzafrir_laptop | 23 B channels |
| 18:01.02 | jasonwoot | your T1's only have 24 channels? that's so 2007 |
| 18:01.18 | tzafrir_laptop | (BRI is 2 B channels and an extra D channel) |
| 18:01.38 | jasonwoot | makes a note not to exagerate anymore |
| 18:04.04 | jeev | Fender, i saw that thing you wrote about the Unmonitor in the peers status.. you said improperly configured phones.. uh, is |
| 18:04.12 | jeev | what is proper config, ~sipnat one ? |
| 18:04.16 | *** join/#asterisk JenniferAkemi (n=akemi@76-10-152-16.dsl.teksavvy.com) |
| 18:04.18 | tzafrir_laptop | jasonwoot, still using those T1? why not move to E1s |
| 18:05.30 | watchy | whats the point of recording your name in * voicemail? where is it used |
| 18:05.44 | [TK]D-Fender | watchy: in the automated message, in Directory, etc |
| 18:05.54 | watchy | ah |
| 18:06.09 | jasonwoot | sip trunks for everything except the conf calls... keeping those lousy T1s for that |
| 18:06.09 | watchy | everytime i get sent to my voicemail it says the temp message. |
| 18:06.40 | jasonwoot | <watchy> dial by name directory |
| 18:07.02 | [TK]D-Fender | watchy: As well it should. |
| 18:07.02 | jasonwoot | has anyone created multiple button maps for polycom 501s? |
| 18:07.13 | [TK]D-Fender | jasonwoot: Meaning? |
| 18:07.20 | watchy | tk: so wouldnt the temp message actually be a perminant message? |
| 18:07.33 | jasonwoot | need to lose the forward button for some, but not for all, by class |
| 18:07.55 | [TK]D-Fender | watchy: the temp gest played in PLACE of the regular ones as long as its there. When you delete it to old one plays. |
| 18:07.57 | *** part/#asterisk btfx (n=chatzill@c-76-19-45-11.hsd1.ma.comcast.net) |
| 18:08.18 | watchy | hrm oh i see |
| 18:08.23 | watchy | so if i delete the temp message |
| 18:08.27 | watchy | the regular one will play |
| 18:08.30 | [TK]D-Fender | jasonwoot: make multiple sip.cfg's and specify them per <mac>.cfg |
| 18:08.58 | thomas | hello |
| 18:09.06 | thomas | how i can install a stun server? |
| 18:09.49 | MindTheGap | hello all... im not sure this will be an asterisk question but here it goes... i have a 1.6beta9 asterisk using realtime ldap for sip peers and it works great, now it comes the time to add voicemail as well and although my users have their sip info under their "cn=user, ou=people, dc=etc, dc=com", i've seen voicemails being populated somewhere else like cn=600,ou=voicemail,dc=etc,dc=com. |
| 18:09.52 | MindTheGap | Does it have to be like this or can I just stuff in voicemail settings under their cn like with sip info? what are the advantages? i think its cleaner than having multiple OUs like people, sippeers, voicemail. Of course, i might just be missing something as there's priority settings under voicemail i still cannot fully understand since i inderstand the diaplan should handle prioritys and logic... can someone enlighten me? |
| 18:10.11 | [TK]D-Fender | thomas: Depends on the stun server |
| 18:10.26 | [TK]D-Fender | thomas: And this has nothing to do with * as * doe not support stun |
| 18:10.47 | thomas | oh, ok |
| 18:10.53 | thomas | [TK]D-Fender: sorry and thank you |
| 18:12.07 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:14.45 | *** part/#asterisk pta200 (n=paolo@dsl211-220-225.nyc1.dsl.speakeasy.net) |
| 18:16.32 | *** join/#asterisk oilinki (n=oil@ppp-124-120-5-63.revip2.asianet.co.th) |
| 18:19.38 | watchy | tk you have saved my sagging nuts yet again |
| 18:19.43 | watchy | i wish i could paypal you love |
| 18:19.56 | NOT_guru | has anyone in here used the grandstream 8 port FXO network thing? |
| 18:20.17 | NOT_guru | if so did it work well? or did you start RMA'ing it within 10 minutes? |
| 18:20.20 | NOT_guru | =D |
| 18:20.32 | NOT_guru | TIA |
| 18:21.17 | watchy | grandstrema sucks |
| 18:21.21 | NOT_guru | I am just curious about network based FXO and how they work |
| 18:21.35 | NOT_guru | I have heard that in general before watchy |
| 18:21.37 | watchy | no one buys grandstream |
| 18:21.51 | watchy | i bet the people at grandstream use polycom phones for tech support |
| 18:22.04 | NOT_guru | ok well lets discuss the linksys spa400 4FXO then |
| 18:22.06 | NOT_guru | LOL |
| 18:22.15 | watchy | is that an ata? |
| 18:22.21 | NOT_guru | yes 4 port |
| 18:22.32 | watchy | i use many linksys atas cuz they are sipura |
| 18:22.36 | watchy | and they work wonderful |
| 18:22.53 | NOT_guru | you you have several single ports in se? |
| 18:22.57 | NOT_guru | use |
| 18:23.00 | watchy | i have many 2ports in use |
| 18:23.00 | NOT_guru | on same system? |
| 18:23.03 | watchy | like 40 |
| 18:23.08 | *** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 18:23.17 | watchy | i use the 2 ports in bunkers out at a .gov missle contractor |
| 18:23.18 | NOT_guru | both fxo or the 1 fxo 1fxs kind? |
| 18:23.23 | watchy | we put them in explosive proof boxes |
| 18:23.34 | watchy | oops |
| 18:23.37 | watchy | fxs's |
| 18:23.40 | NOT_guru | oops indeed |
| 18:23.40 | watchy | not fxos |
| 18:23.45 | NOT_guru | ah |
| 18:23.47 | NOT_guru | indeed |
| 18:23.52 | watchy | we plug big red phones into them |
| 18:23.54 | NOT_guru | I am specifically looking at FXO's |
| 18:23.59 | watchy | incase of explosion pickup the phone |
| 18:24.04 | NOT_guru | LOL |
| 18:24.07 | watchy | serious |
| 18:24.15 | watchy | the phones are like $800 |
| 18:24.17 | NOT_guru | thats harsh |
| 18:24.26 | watchy | and they are $5 walmart phones certified by the dod |
| 18:24.30 | watchy | but bright red |
| 18:24.41 | NOT_guru | screw the dod |
| 18:24.43 | NOT_guru | LOL |
| 18:24.46 | NOT_guru | j/k |
| 18:24.48 | NOT_guru | sorta |
| 18:24.49 | watchy | im sure they have to have some other kinda stuff to keep from making sparts |
| 18:24.50 | watchy | sparks |
| 18:24.58 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 18:24.59 | watchy | so they dont blow up the munitions in the bunker |
| 18:25.05 | *** part/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 18:25.12 | NOT_guru | again.... harsh |
| 18:25.17 | watchy | yea |
| 18:25.20 | [TK]D-Fender | watchy: Paypal can send me money, which is currently interchangeable with love :) |
| 18:25.25 | watchy | hahah |
| 18:25.28 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.static.dsl.tele.dk) |
| 18:25.33 | NOT_guru | so the linksys SPA's are basically siporas? |
| 18:25.38 | watchy | whats the going exchange rate to US currency and love? |
| 18:25.42 | watchy | yea |
| 18:25.43 | Qwell | 1=1 |
| 18:25.57 | NOT_guru | US dollar is weak |
| 18:26.01 | NOT_guru | so .50 = 1 |
| 18:26.07 | watchy | haha |
| 18:26.13 | watchy | ill go get some euros then |
| 18:26.19 | NOT_guru | LOL |
| 18:26.45 | *** join/#asterisk Linker3000L (n=chatzill@78.32.25.201) |
| 18:26.56 | watchy | brb i gotta fill out some po req forms i need some phones |
| 18:27.02 | *** join/#asterisk l2cache (n=l2cache@155.132.204.68.cfl.res.rr.com) |
| 18:27.23 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 18:29.59 | *** part/#asterisk Paige_ (n=Paige@208.89.241.31) |
| 18:31.02 | *** join/#asterisk metastable (n=superfly@212.71.24.170.res.static.edpnet.net) |
| 18:31.10 | oilinki | I'm trying to understand realtime. |
| 18:31.36 | NOT_guru | a show on showtimne hosted by Bill Mahr? |
| 18:31.45 | NOT_guru | oh sorry wrong room |
| 18:31.53 | oilinki | if I'm using realtime (odbc,mysql), should the peers be seen with command sip show peers ? |
| 18:31.55 | metastable | i'm having problems compiling the b410p module |
| 18:32.37 | metastable | the error says something about using EXTRA_CFLAGS |
| 18:32.41 | MindTheGap | oilinki, with rtcachefriends=yes youll see them |
| 18:32.51 | oilinki | currently I get nothing, but when testing with > realtime load sippeers name voip.foo.com -> I get results |
| 18:32.54 | metastable | I have a lot of output, for anyone interested in helping |
| 18:32.58 | metastable | including digiuminfo |
| 18:33.03 | oilinki | MindTheGap: ok. thanks. I'll try that one. |
| 18:33.42 | metastable | zaptel 1.4.11 compiles fine |
| 18:33.50 | metastable | but 'make b410p' keeps failing |
| 18:39.23 | metastable | anyone with experience about the b410p module ? |
| 18:41.22 | Qwell | metastable: failing how? |
| 18:41.24 | Qwell | ~pb |
| 18:41.24 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines 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 |
| 18:41.50 | metastable | qwell: i can send you a .tgz with digiuminfo and the output of all make commands |
| 18:42.04 | Qwell | just pastebin the output of make |
| 18:42.17 | metastable | ok |
| 18:43.19 | *** join/#asterisk banzaika (n=banzaika@rrcs-24-136-116-90.nyc.biz.rr.com) |
| 18:44.25 | metastable | Qwell: http://pastebin.com/m21467f39 |
| 18:44.56 | Qwell | metastable: 2.6.26? |
| 18:45.03 | Qwell | err, nm, it says right there |
| 18:45.25 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 18:45.49 | Qwell | tzafrir_laptop: what's that variable you can set to get rid of the kernel CFLAGS errors? |
| 18:45.52 | metastable | Qwell: 2.6.24-19-generic |
| 18:45.56 | metastable | ok :) |
| 18:49.20 | *** join/#asterisk lunaphyte (n=lunaphyt@unaffiliated/lunaphyte) |
| 18:49.23 | lunaphyte | hi |
| 18:49.34 | metastable | it's an ubuntu hardy heron 8.04.1 server |
| 18:50.24 | lunaphyte | have any of you guys that use voipjet had trouble lately (the last day or so maybe) sending calls to test.voipjet.com? |
| 18:51.34 | jasonwoot | <[TK]D-Fender> divert.fwd.1.enabled = "0" , correct? |
| 18:53.32 | [TK]D-Fender | jasonwoot: Never tried |
| 18:54.54 | jasonwoot | polycom problem resoultion = try once, replace with software phone |
| 18:55.19 | jasonwoot | why did they have to put the friggin forward soft key right where everyone would hit it accidentally? |
| 18:55.45 | *** join/#asterisk DarienWork (n=dan@209.17.173.13) |
| 18:56.17 | *** join/#asterisk ghenry (n=ghenry@ghenry.plus.com) |
| 18:56.19 | ghenry | Hi all, I'm trying to debug why an outbound analogue trunk is not being reached by my dialpla |
| 18:56.34 | ghenry | my users in in users.conf as expected via the gui and are SIP and they in the default context |
| 18:56.43 | ghenry | that's the asterisk gui. |
| 18:56.49 | ghenry | http://rafb.net/p/0FEjvD55.html |
| 18:56.56 | ghenry | but _X. isn't triggering say a mobile 11 digit number etc. |
| 18:57.07 | ghenry | Should I have _XXXXX. instead in the trunk? |
| 18:57.13 | ghenry | with sip set debug ip blah on |
| 18:57.16 | nny_1 | so anyone use outcall/ have a copy of the last 1.4.X release or know where to get it? |
| 18:57.20 | ghenry | I can see that any non 4 digit number in not found in the default context |
| 18:57.44 | nny_1 | working on some basic outlook (cough) integration and outcall is borked with exchange it seems |
| 18:57.45 | mgroman | How does one disable safe_asterisk ? |
| 18:58.28 | *** join/#asterisk irieKen (n=chatzill@rrcs-74-87-28-55.west.biz.rr.com) |
| 18:59.21 | irieKen | Hi guys... For some reason, when people call into our Asterisk system, everything seems OK, then after we transfer them to another extension, we can hear them, but they can't hear us... Anyone have an idea of what's going on? (Happens on both polycom and grandstream phones). |
| 19:02.12 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:03.20 | mgroman | The answer to my question is to change the /etc/init.d/asterisk script (on RH) and just have it execute asterisk, not safe_asterisk |
| 19:03.49 | mgroman | and then reformat because RH is weak |
| 19:05.09 | jaytee | RH is only as weak as the skillset of the system administrator |
| 19:05.12 | *** join/#asterisk oilinki3 (n=oil@ppp-124-120-3-31.revip2.asianet.co.th) |
| 19:05.23 | deeperror | :) |
| 19:05.41 | mgroman | jaytee: lies! |
| 19:06.05 | jaytee | mgroman: pffft! |
| 19:06.23 | *** join/#asterisk lowtek (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 19:06.43 | *** join/#asterisk washburnello (n=johnsy@h23.238.30.71.dynamic.ip.windstream.net) |
| 19:06.57 | jaytee | ghenry, pastebin the full extensions.conf |
| 19:07.06 | ghenry | ok |
| 19:07.15 | metastable | i don't like rh either |
| 19:07.22 | metastable | then again, i don't like ubuntu as a server either |
| 19:07.26 | metastable | and i'm using both |
| 19:07.34 | irieKen | Anyone have issues with the calling party not being able to hear you after a call is transfered to you? |
| 19:07.35 | deeperror | how about centos? |
| 19:07.45 | lowtek | Hi all. Is there a way to find the voicemailbox for a sip peer in the dialplan? For example, if someone dials extension 855 but that extension's mailbox is 866@default, is there a way to call VoiceMail() in a way that uses the correct mailbox? ${EXTEN}@default will obviously not work in this condition. |
| 19:07.57 | metastable | i liked gentoo |
| 19:08.02 | metastable | on a decent machine |
| 19:08.08 | metastable | and debian too |
| 19:08.10 | jaytee | I run Ubuntu on my desktop here and at home. I run * on RHEL 5 64 bit and works superbly. I also run my test system on CentOS 5.2 and have no problems with it. |
| 19:08.12 | metastable | package managers ! |
| 19:08.20 | ghenry | jaytee: not much to it: http://nopaste.com/p/aDqHSLpE |
| 19:09.19 | Alan_Hicks | Howdy folks. |
| 19:09.25 | deeperror | lowtek, why would the boxes be split up? |
| 19:09.38 | [TK]D-Fender | lowtek: For your peer do "SetVar=RealBox=123" and you can do ${Realbox} to retrieve it |
| 19:09.51 | lowtek | deeperror: We have a bunch of sales people that share a single voicemailbox ... |
| 19:10.39 | lowtek | [TK]D-Fender: Thanks! That's a step in the right direction for sure, is there a way to retreive a sip.conf entrie's mailbox on the fly in the dialplan? |
| 19:11.32 | [TK]D-Fender | lowtek: As you can have multiple, no. This way is pretty much the same... just 1 extra entry to make... |
| 19:12.24 | lowtek | [TK]D-Fender: Thanks, TK |
| 19:14.34 | *** join/#asterisk coppice (n=chatzill@175.202.17.210.dyn.pacific.net.hk) |
| 19:15.01 | lowtek | TK: Ahh! I see, you can do a setvar in the peer's sip.conf entry and that will end up being a channel variable whenever the peer is in use? Did I read that right? |
| 19:15.50 | jaytee | ghenry, set core verbose 10 and pastebin the results of the failed call, please. |
| 19:15.59 | jblack | <PROTECTED> |
| 19:16.05 | ghenry | there is nothing listed there jaytee |
| 19:16.22 | ghenry | you can only see for example 07245908280 not found in default |
| 19:16.31 | ghenry | when you sip set debug ip |
| 19:16.32 | *** join/#asterisk korihor (n=korihor@201.211.168.130) |
| 19:17.24 | jaytee | did you write the [numberplan-custom-1] context section yourself or was it generated by the GUI? |
| 19:17.51 | ghenry | by the crap gui, but I picked _X. |
| 19:18.40 | [TK]D-Fender | ghenry: Stop picking stupid psycho macth-alls like that and set up PROPER patterns |
| 19:18.54 | [TK]D-Fender | lowtek: Yes |
| 19:18.58 | ghenry | I'm just testing for now [TK]D-Fender |
| 19:19.12 | jaytee | ghenry, your default context is a mess |
| 19:19.16 | [TK]D-Fender | ghenry: Then make a better test. You are asking for trouble and GETTING IT |
| 19:19.23 | ghenry | I want proper patterns once I've confirmed the trunks |
| 19:19.29 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 19:19.33 | jaytee | when you match on _X. it only does a NoOp and then does nothing |
| 19:19.39 | lowtek | [TK]D-Fender: Thanks again! :) |
| 19:19.46 | [TK]D-Fender | lowtek: np |
| 19:20.08 | ghenry | jaytee: exten = _X.,1,NoOp('This is default') was me for testing |
| 19:20.14 | ghenry | forgot to take that out |
| 19:20.26 | ghenry | I didn't even see that on the console with verbose and debug 10 |
| 19:20.40 | ghenry | but there's two |
| 19:20.54 | ghenry | one in default and another in customblah |
| 19:21.38 | *** part/#asterisk washburnello (n=johnsy@h23.238.30.71.dynamic.ip.windstream.net) |
| 19:24.09 | *** join/#asterisk neurosys (n=neurosys@adsl-067-032-132-060.sip.bct.bellsouth.net) |
| 19:24.14 | ghenry | jaytee: I've adjusted now to (missing out other stuff for paste) http://nopaste.com/p/aYAKMFwzN |
| 19:26.24 | irieKen | NM, seems that my end is muted when transferring from the polycoms... |
| 19:26.28 | irieKen | Anyone have any ideas? |
| 19:32.56 | *** join/#asterisk sorenruck_ (n=sorenruc@p5DC0ECF5.dip.t-dialin.net) |
| 19:34.42 | *** join/#asterisk nitam (n=eskali@173-205-231-201.fibertel.com.ar) |
| 19:34.48 | nitam | hi |
| 19:36.21 | irieKen | Hi Nitam. |
| 19:36.29 | nitam | does anybody know how to start recording a phone conversation from a SIP phone nor softphone ? |
| 19:37.02 | jaytee | ghenry, sorry but I had to step away. your default context doesn't send ANY calls to your [numberplan-custom-1] so the trunkdial macro will never be run. |
| 19:37.44 | ghenry | yeah, but if it's included in that context, surely the SIP exten can reach the trunk? |
| 19:37.54 | ghenry | or should I include that in the default context? |
| 19:37.58 | jaytee | ghenry, as [TK]D-Fender said, you need to correct your matches and either include the numberplan-custom-1 context in default or put a pattern match that jumps to it in default. |
| 19:38.09 | ghenry | ok, thanks |
| 19:38.17 | ghenry | why doesn't the gui do that. that's what it's for |
| 19:38.25 | jaytee | no, it's not for that |
| 19:38.53 | sorenruck_ | Hi, |
| 19:38.53 | sorenruck_ | doe?s any body know what this log means? |
| 19:38.53 | sorenruck_ | <PROTECTED> |
| 19:38.53 | sorenruck_ | <PROTECTED> |
| 19:38.54 | sorenruck_ | <PROTECTED> |
| 19:38.54 | sorenruck_ | <PROTECTED> |
| 19:38.56 | sorenruck_ | <PROTECTED> |
| 19:38.58 | sorenruck_ | <PROTECTED> |
| 19:39.00 | sorenruck_ | <PROTECTED> |
| 19:39.02 | sorenruck_ | <PROTECTED> |
| 19:39.05 | *** mode/#asterisk [+o [TK]D-Fender] by ChanServ |
| 19:39.06 | sorenruck_ | <PROTECTED> |
| 19:39.08 | sorenruck_ | <PROTECTED> |
| 19:39.10 | sorenruck_ | My System is: |
| 19:39.11 | *** kick/#asterisk [sorenruck_!n=joe@216.191.106.163] by [TK]D-Fender ([TK]D-Fender) |
| 19:39.20 | ghenry | jaytee: I may have well just wrote it myself instead of debugging the gui writes |
| 19:39.41 | [TK]D-Fender | ghenry: You learn slowly grasshopper |
| 19:39.42 | ghenry | this was out of the box, just adding users and a trunk |
| 19:39.56 | ghenry | some f***ing appliance ;-) |
| 19:40.21 | ghenry | ok, back to basics and more control. Thanks [TK]D-Fender and jaytee |
| 19:40.31 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 19:40.39 | *** join/#asterisk sorenruck_ (n=sorenruc@p5DC0ECF5.dip.t-dialin.net) |
| 19:40.53 | [TK]D-Fender | sorenruck_: do NOT span in here like that again |
| 19:41.04 | [TK]D-Fender | sorenruck_: use a PASTEBIN |
| 19:41.05 | sorenruck_ | ok, sorry |
| 19:41.06 | [TK]D-Fender | ~pb |
| 19:41.06 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines 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 |
| 19:41.08 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^ |
| 19:41.09 | [TK]D-Fender | spam* |
| 19:41.45 | [TK]D-Fender | sorenruck_: And for your next attempt, enable pri debug on that span |
| 19:42.23 | nitam | [TK]D-Fender, do you know how to start recording a phone conversation from a SIP phone ? .. i know about the Monitor cmd, etc, but i just wanted to know if there is a default way to do that ? |
| 19:42.39 | [TK]D-Fender | nitam: There is no such thing as "default" |
| 19:42.53 | [TK]D-Fender | nitam: * does what YOU configure it to do. |
| 19:43.05 | Alan_Hicks | Howdy. My struggles with BLF continue. I'm using Polycom 320 phones and have gotten presence to work, but am unsure what else needs to be done to enable BLF. I've included what I think are the relevant portions of all config files here: http://pastebin.com/d12dff9ed |
| 19:43.40 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 19:43.42 | nitam | [TK]D-Fender, I agree, but im wondering if trixbox has something predefined for that |
| 19:43.46 | Alan_Hicks | The 320 phone displays an icon for "Alan Hicks" and this icon changes with "Alan Hicks"'s phone (exten => 210) so presence is working. |
| 19:44.06 | [TK]D-Fender | nitam: Don't know, don't care |
| 19:44.15 | Alan_Hicks | If I hit "Line 2" on the CTS phone, it immediately rings exten 210 as expected from speed dial. |
| 19:44.43 | NOT_guru | natim try #trixbox for trixbox specific questions |
| 19:44.51 | Alan_Hicks | What I'd like to do is get the light beside "Line 2" to turn on or blink whenever the phone at exten 210 is in use. How can I accomplish this? |
| 19:44.53 | NOT_guru | just trying to help |
| 19:45.03 | nitam | [TK]D-Fender, like the Asterisk Dial Commands: tTrwW.. it suppouse that "w" allow callers to start recording calls |
| 19:45.14 | nitam | oh .. my bad then .. ty NOT_guru |
| 19:45.20 | NOT_guru | no worries |
| 19:45.27 | [TK]D-Fender | Alan_Hicks: please pastebin ALL of the relevent bits. |
| 19:45.28 | NOT_guru | again |
| 19:45.31 | NOT_guru | just trying to help |
| 19:45.45 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 19:45.51 | Alan_Hicks | [TK]D-Fender: You want all of sip.conf and extensions.conf? |
| 19:46.22 | [TK]D-Fender | Alan_Hicks: First your phones hsould all have "type=peer" ,"call-limit=99". Do that no and restart. |
| 19:46.24 | [TK]D-Fender | now* |
| 19:46.37 | [TK]D-Fender | Alan_Hicks: and kill the incoming-limit |
| 19:46.48 | Alan_Hicks | Ok. |
| 19:47.06 | [TK]D-Fender | Alan_Hicks: and exten => 210,1,Dial(SIP/alan,,T) <- remove the "T". This is for loser interfaces |
| 19:47.37 | *** join/#asterisk Gringo_ (n=raf@213.219.166.60.adsl.dyn.edpnet.net) |
| 19:48.14 | Gringo_ | i'm looking for a way to do attended transfers with the manager API |
| 19:48.55 | Gringo_ | is this possible? |
| 19:49.53 | Alan_Hicks | [TK]D-Fender: sip.conf changes completes per your instruction. This seems to have broken presence though. |
| 19:50.10 | [TK]D-Fender | Alan_Hicks: Complete pastebin this time. |
| 19:50.19 | Alan_Hicks | The icons on the phones' LCD screens no longer change. |
| 19:50.27 | irieKen | WTF? My Asterisk server has suddently started refusing SSH connections:S |
| 19:50.48 | Alan_Hicks | [TK]D-Fender: http://pastebin.com/d2c13bcdb |
| 19:51.46 | [TK]D-Fender | Alan_Hicks: What ver of *? |
| 19:52.13 | Alan_Hicks | Asterisk 1.4.20 |
| 19:52.21 | [TK]D-Fender | Alan_Hicks: You've restarted *? |
| 19:52.33 | Alan_Hicks | [TK]D-Fender: Yes. |
| 19:52.58 | [TK]D-Fender | Alan_Hicks: PB : "core show hints" and place a call. (verbose 10) |
| 19:53.01 | Alan_Hicks | I just restarted the phones, and that seems to have brought presence back to working, but still no BLF. |
| 19:55.24 | irieKen | Anyone have any idea why my Asterisk server would suddunly stop accepting SSH? |
| 19:55.42 | [TK]D-Fender | irieKen: * doesn't do SSH |
| 19:55.54 | Qwell | irieKen: let me guess - trixbox? |
| 19:56.11 | irieKen | Qwell: Close; AA50 Asterisk Appliance. |
| 19:56.19 | Qwell | is it enabled? |
| 19:56.25 | Alan_Hicks | [TK]D-Fender: Just a moment. Restarting the phones again. Have to do that everytime I restart asterisk to bring presence back online apparently. |
| 19:56.26 | irieKen | Qwell: yes, it is enabled. |
| 19:56.32 | Qwell | verify |
| 19:56.45 | irieKen | Qwell: Verified by looking at network.conf . |
| 19:57.06 | *** part/#asterisk Gringo_ (n=raf@213.219.166.60.adsl.dyn.edpnet.net) |
| 19:57.24 | Qwell | and did you let it rerun the networking script? |
| 19:57.30 | Alan_Hicks | [TK]D-Fender: http://pastebin.com/d636fa022 |
| 19:58.01 | Qwell | (by using the gui like you should...) |
| 19:58.21 | [TK]D-Fender | Alan_Hicks: 210@internal : SIP/alan State:InUse Watchers 1 <- so * works. the rest is your phone config outside of * |
| 19:58.38 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:58.38 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 19:58.54 | lmadsen | Yourname`: !!! |
| 19:59.04 | Alan_Hicks | [TK]D-Fender: Thank you. Are you able to assist there? |
| 19:59.06 | irieKen | Qwell: How do I make it rerun the networking script? I simply restarted the box. |
| 19:59.20 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 19:59.20 | Yourname` | lmadsen!! So is it this weekend?!? |
| 19:59.20 | Qwell | by using the gui to fix it |
| 19:59.24 | lmadsen | Yourname`: wow, you're here! :) |
| 19:59.34 | Yourname` | Nick highlight :D |
| 19:59.35 | [TK]D-Fender | Alan_Hicks: Trash your configs, work off the base provisioning files and not that puny attempt at "override-only. |
| 19:59.38 | lmadsen | Yourname`: tonight is my condo warming! I am in the building! You should stop by 3405 anytime after 5:30 |
| 19:59.47 | lmadsen | bring the wife! |
| 19:59.52 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 19:59.57 | *** join/#asterisk taintman (n=root@74.95.210.124) |
| 19:59.58 | Yourname` | Congratulations man! |
| 19:59.59 | lmadsen | I like exclamation marks!!! |
| 20:00.07 | lmadsen | Yourname`: thanks! |
| 20:00.08 | irieKen | Qwell: I unchecked it, then checked the "SSH" box again, saved it, then clicked apply settings... nochange. |
| 20:00.14 | Yourname` | I'll try to make it, I was going to go to the datacentre real quick, but let's see. |
| 20:00.20 | Yourname` | Yes, she comes. |
| 20:00.51 | lmadsen | Yourname`: of course -- there will be free wine |
| 20:00.58 | lmadsen | 5:30 -> probably... 10pm tonight? |
| 20:01.03 | lmadsen | something like that |
| 20:01.06 | lmadsen | it IS a school night |
| 20:01.15 | Qwell | pfft, excuses |
| 20:01.18 | Alan_Hicks | [TK]D-Fender: Well, I started over from scratch with the original phone1.cfg except for changing the digitmap line. |
| 20:01.28 | Yourname` | I'm wondering why you picked a wednesday for this good deed!? lol |
| 20:01.41 | lmadsen | Yourname`: because most people have weekend plans since it is summer |
| 20:01.53 | lmadsen | and weekdays don't stop me from having a party :) |
| 20:02.04 | Yourname` | Oh come on, it hasn't been summer for as long as it started. :( |
| 20:02.08 | Yourname` | Thunderstorms and what not.. |
| 20:02.18 | *** join/#asterisk washburnello (n=johnsy@h23.238.30.71.dynamic.ip.windstream.net) |
| 20:02.21 | Alan_Hicks | [TK]D-Fender: I can pastebin my entire phone1.cfg if you think that will help, but it's very generic. |
| 20:02.43 | lmadsen | Yourname`: that is true, but most people seem to have their weekends booked well in advance of anything I'd plan, plus there is less chance of people bailing on a weeknight |
| 20:02.47 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:02.50 | lmadsen | at least that's how I see it, heh |
| 20:02.52 | Yourname` | lol true |
| 20:03.21 | Yourname` | I'll try to bring a neighbor buddy too if that's ok? ;) |
| 20:03.24 | Alan_Hicks | MAC.cfg pulles in phone1.cfg and then over-rides with values in MAC-phone.cfg so I don't have to carry around a lot of extra bits. |
| 20:03.33 | lmadsen | Yourname`: absolutely! the more the merrier! |
| 20:03.35 | [TK]D-Fender | Alan_Hicks: start from scratch. take phone1 and template it with everything except the user & pass. Then COPY IT to phone-210.cfg and have your MAC file use sip.cfg (complete base) + the specific phone-XXX for it |
| 20:03.46 | lmadsen | I'm gonna put a sign up I think to invite the local neighbors as they return home from work |
| 20:03.56 | Yourname` | Cool! |
| 20:03.58 | Yourname` | lol you should |
| 20:04.04 | lmadsen | I think I just might... |
| 20:04.05 | Alan_Hicks | [TK]D-Fender: Alright. I'll do that and report back in a few minutes. |
| 20:04.08 | Yourname` | Usually it happens at the bar though. |
| 20:04.20 | Yourname` | A day or two before the party, the "host" ends up at the bar and invites errrrbody. |
| 20:04.37 | lmadsen | Yourname`: for sure... I need to come down to meet these people you speak of one of these days :) |
| 20:05.01 | Yourname` | It's not many, just a small group of regulars, lol |
| 20:05.09 | lmadsen | Yourname`: even better :) |
| 20:05.16 | lmadsen | I shall soon become a regular |
| 20:05.33 | Yourname` | hehe nice |
| 20:05.37 | lmadsen | just that I'm a shy small town farmer boy... I need to be introduced, heh |
| 20:05.42 | lmadsen | after that though... watch out |
| 20:05.43 | Yourname` | I'll txt you before I come down.. |
| 20:05.48 | Yourname` | LOL |
| 20:05.54 | Yourname` | All good, I'll handle the intros. |
| 20:05.54 | lmadsen | sweet deals! feel free to just come down if I don't reply right away |
| 20:06.16 | lmadsen | some friends should be coming by around 5:30 since they work at Q9 (Queen and York I believe) |
| 20:06.22 | lmadsen | then the rest will... show up whenever |
| 20:06.33 | Yourname` | Q9? Isn't that at Front/Uni? |
| 20:06.50 | Yourname` | Oh, wait. I'm talking about the datacentre, I guess. |
| 20:08.04 | [TK]D-Fender | <lmadsen> I shall soon become a regular <- try prune juice |
| 20:09.04 | *** mode/#asterisk [-o [TK]D-Fender] by [TK]D-Fender |
| 20:12.15 | lmadsen | [TK]D-Fender: uh.... huh |
| 20:12.50 | jeev | lmadsen |
| 20:12.57 | jeev | Fiber One cerial |
| 20:12.59 | jeev | cereal |
| 20:13.01 | jeev | get that. |
| 20:13.03 | lmadsen | serial |
| 20:13.12 | jeev | no you dork, it's not rs232 or whatever |
| 20:13.27 | jeev | fiber one, best cereal, i eat it every day and it makes mje crap well |
| 20:13.40 | lmadsen | Yourname`: ya perhaps... most of my friends work in the main building (not the data centre) |
| 20:13.56 | [TK]D-Fender | jeev: Not well enough..... you're still quite "full" ;) |
| 20:14.06 | jeev | no fender.. i always think of you when i'm pushing hard |
| 20:14.43 | *** join/#asterisk l2trace99 (n=jr@70-9-51-67.area4.spcsdns.net) |
| 20:15.48 | jeev | loves iax |
| 20:15.52 | jeev | how about that fender? |
| 20:16.03 | jeev | fender, though. seriously, please. i have a serious question!!! and dont say ~sipnat. |
| 20:16.23 | lmadsen | ~sipnat |
| 20:16.23 | 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 |
| 20:16.25 | jeev | and dont say ~ask |
| 20:16.27 | jeev | damnit,. |
| 20:16.38 | mgroman | lmadsen: your site is down |
| 20:16.53 | jeev | the web version was never up for me, get the pdf. |
| 20:16.59 | lmadsen | mgroman: that is true... I haven't setup my primary switch again and gotten the external IP set back up for the site... :( |
| 20:17.10 | jeev | lmadsen, i will gladly host your site. |
| 20:17.11 | lmadsen | I just recently moved |
| 20:17.32 | jblack | I didn't know you had a site. I bet it would be worth a daily visit. |
| 20:17.48 | jeev | it's the html version of the book |
| 20:18.01 | lmadsen | jblack: I wouldn't say "daily" :) |
| 20:18.04 | jblack | That's it/ |
| 20:18.07 | lmadsen | I need to do a better job of updating it |
| 20:18.10 | jeev | the book i helped him write.. and NEVER got credit for it. |
| 20:18.34 | jblack | That's because your nic doesn't have a strong relationship to your name? |
| 20:19.01 | jeev | no, he didn't give me credit because we had a disagreement. |
| 20:20.13 | [TK]D-Fender | jeev: IAX is of no use to me. |
| 20:20.30 | mgroman | doing "sys.stderr.write(msg)" is giving me absolutely nothing, has anyone here successfully used python with AGI? |
| 20:20.38 | jeev | Fenderino, i used iax to set up the internal asterisk with the external ones, then did dial plans.. to vaoid all the nat crap i was having issues with.. |
| 20:20.43 | [TK]D-Fender | jeev: Only if you're desparate for bandwidth or are dealing with a truly unavoidable firewall. |
| 20:20.53 | jblack | mgroman: Use the agi function, rather than stderr. |
| 20:20.54 | [TK]D-Fender | jeev: Otherwise is a cop-out. |
| 20:21.07 | jeev | but in all seriousness, how do these hosted pbx's work if there are 10 phones behind a firewall... how does it work? it just traverses it fine ? |
| 20:21.20 | jeev | yea, i couldn't avoid the firewall so i did iax to the external asterisk. |
| 20:21.27 | jeev | but i'm at a loss for how these hosted pbx companies work.. |
| 20:21.36 | *** join/#asterisk l2cache (n=l2cache@155.132.204.68.cfl.res.rr.com) |
| 20:21.38 | [TK]D-Fender | jeev: No, none of them work. They don't really have customers, Santa Clause is real, and you aren't incompetant. |
| 20:21.44 | mgroman | jblack: myexten,n,AGI(my_agi_script.py) ... and then in my_agi_script.py i write to stderr. You are saying that is wrong? |
| 20:21.51 | [TK]D-Fender | </paralleluniverse> |
| 20:21.52 | jeev | now i know the last statement was false.. but everything else was sarcasm |
| 20:21.54 | jblack | mgroman: I am saying that is wrong. |
| 20:21.54 | jeev | enlightenme |
| 20:21.57 | jeev | enlighten me |
| 20:22.16 | [TK]D-Fender | blasts jeev with a 10 megawatt laser |
| 20:22.17 | mgroman | ~agi |
| 20:22.17 | jbot | well, agi is the Asterisk Gateway Interface... similar to CGI for web applications AGI lets you script call control and access databases using your favorite language. AGI wrappers are available for Python (pyst), Perl (astperl?) and other languages, or <reply> See also http://www.voip-info.org/wiki-Asterisk+AGI |
| 20:22.21 | jblack | mgroman: Try the noop for a start. That'll get into the asterisk logs. There's probably some sort of debug function as well. |
| 20:22.49 | jblack | [TK]D-Fender: Hmmm. Can you think of any argument against IAX2 when both endpoints support it? |
| 20:23.19 | [TK]D-Fender | jblack: seems more susceptable to jitter & other issues. |
| 20:23.39 | jeev | just recovered.. that laser was weak |
| 20:23.46 | [TK]D-Fender | jblack: Version compatibility, etc |
| 20:24.07 | jeev | fender, in all seriousness, could you please tell me what you think about the method of using asterisk as a client and server outside a firewalled network with phones behind the firewall ? |
| 20:24.33 | [TK]D-Fender | jeev: 10 MILLION watts of light in a focused beam. We should be smoking you now. |
| 20:24.40 | jeev | i've destroyed all my credibility with you.. now you dont even take me seriously! |
| 20:24.48 | jblack | [TK]D-Fender: I suppose jitter can become a problem since users have the ability to tune it, thus screwing it up. |
| 20:24.50 | jeev | and it all started with the damn WIP330! |
| 20:25.03 | [TK]D-Fender | jeev: what do you mean "NOW"? |
| 20:25.12 | jeev | bastard |
| 20:25.20 | jblack | Perhaps the defaults allow for too much jitter for high latency links as well... |
| 20:25.51 | [TK]D-Fender | jeev: * works fine for the rest of us. |
| 20:26.12 | jeev | how about the method of using asterisk as a client and server outside a firewalled network with phones behind the firewall ? |
| 20:26.46 | [TK]D-Fender | jblack: I also don't like closed standards. While * may be a documented protocol, unless its NEEDED why would I use a less supportable solution? |
| 20:27.16 | jblack | [TK]D-Fender: Because gopher sucks? |
| 20:27.28 | [TK]D-Fender | jeev: and FFS stop using " as a client and server". That whole rip you were on the other day is drivel. a call is a call is a call. |
| 20:27.37 | [TK]D-Fender | IAX* |
| 20:27.53 | jblack | But anyways, those seem like reasonable points to me. |
| 20:27.58 | [TK]D-Fender | jblack: gopher? Sorry, I'm referring to SIP vs IAX2 |
| 20:28.33 | jblack | [TK]D-Fender: i was being obscure. Every protocol starts off as less supported. Even http. |
| 20:28.36 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 20:28.54 | jeev | shoots fender with a 100 megawatt crap creator machine |
| 20:29.25 | jeev | so i figure the only reason why i failed at what i wanted to do was cause of the multi wan situation. |
| 20:29.28 | jblack | Anyways, like I said, I can understand and respect your viewpoint. |
| 20:29.50 | [TK]D-Fender | jblack: I'm all about having the freedom to choose every pieve of my solution and not be owned by any of them. Thats my beef against the Astribank. Not because of any inherent failing in its function. Rather that it physically ties via USB (ball & chain to server) with the limits associated along-with. Then add the fact that it only talks to *. Non-reusable tech. I refuse to buy dead-ends |
| 20:30.24 | [TK]D-Fender | jblack: Also another reason I like Sangoma cards. I could get Windows drivers (if for some psychotic reason I should want such a thing) |
| 20:30.38 | jblack | I highly value flexibility too. |
| 20:31.05 | [TK]D-Fender | jblack: I place quality & flexibility over loyalty as far as tech goes. |
| 20:31.27 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 20:31.40 | jblack | And regarding Sangoma, I'm sold as well too. The one I'm testing is flawless. Free software drivers, excellent documentation and support, well built, and solid as a rock. |
| 20:32.04 | [TK]D-Fender | jblack: For instance if Polycom were to do do something that seriously pisses me off I'd dump them like a rock. It'd be sad of course |
| 20:32.14 | jblack | I wish I could return that rhino I "bought" a couple months ago via some dark,damp location on a rhino engineer. |
| 20:32.37 | [TK]D-Fender | jblack: Go visit jameswf-home ;) |
| 20:32.57 | jblack | jameswf-home recommended the card. When I talk about a dark, damp location, I'm referring to his. |
| 20:33.05 | [TK]D-Fender | jblack: lol |
| 20:33.13 | [TK]D-Fender | ok, well its time to head home. BBIAB |
| 20:34.08 | jeev | no matter how much he pretends he hates me.. he loves me |
| 20:38.08 | *** join/#asterisk dlynes (n=daniel@S01060016b68219f1.vs.shawcable.net) |
| 20:39.52 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 20:39.52 | *** mode/#asterisk [+o denon] by ChanServ |
| 20:51.56 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 20:53.43 | *** join/#asterisk l2cache (n=l2cache@155.132.204.68.cfl.res.rr.com) |
| 20:54.21 | *** join/#asterisk quentusrex (n=quentusr@c-71-227-241-183.hsd1.or.comcast.net) |
| 20:54.57 | _fury | is there a good tutorial anywhere on extensions.conf ? |
| 20:55.19 | _fury | need to program a simple menu that will answer the call, ring once, play a greeting, accept a phone number (10 digits), execute a mysql query |
| 20:55.59 | jeev | lots of tutorials |
| 20:56.02 | jeev | hvae you read the book ? |
| 20:56.06 | _fury | can't seem to find a good one |
| 20:56.08 | jeev | or at least glanced at it ? |
| 20:56.10 | jeev | ~book |
| 20:56.11 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 20:56.15 | _fury | I glanced yeah |
| 20:56.19 | jeev | buy the book for archive and download the pdf |
| 20:58.10 | [TK]D-Fender | _fury: "core show applications" , "core show functions" <- go read the list of the apps & functions you can use tin thedialplan and read up |
| 20:58.30 | jeev | crap,, when did you come back |
| 20:58.58 | *** join/#asterisk GhOnDiE (n=ghondie@dsl78-143-210-38.in-addr.fast.co.uk) |
| 20:59.19 | GhOnDiE | hi peeps |
| 20:59.27 | jeev | hi |
| 20:59.40 | GhOnDiE | dont suppose anybody know of a prferably free web provisioning tool? |
| 20:59.45 | GhOnDiE | dont need to do loads |
| 20:59.50 | GhOnDiE | i just want to have a play |
| 21:00.31 | [TK]D-Fender | GhOnDiE: www.webmin.com |
| 21:00.58 | GhOnDiE | is there a module for that or something |
| 21:01.04 | GhOnDiE | i mean to provision my phones |
| 21:01.05 | GhOnDiE | ? |
| 21:01.41 | [TK]D-Fender | GhOnDiE: See you never went and said WHAT you wanted to provision so naturally I'm thinking "Yeah, a mail-server! Thats it!" |
| 21:02.02 | jeev | lol |
| 21:02.06 | jeev | what phones GhOnDiE? |
| 21:02.11 | [TK]D-Fender | GhOnDiE: Yes, and all "phone provisioning" tool work for all models by all manufacturers |
| 21:02.15 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 21:02.27 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:02.33 | GhOnDiE | lol |
| 21:02.51 | GhOnDiE | well i have a couple of crappy grandstreams |
| 21:02.55 | GhOnDiE | gxp2000 |
| 21:03.16 | [TK]D-Fender | GhOnDiE: Never heard of one for those. Get Google-ing |
| 21:03.35 | *** join/#asterisk Entr4nced (i=IMG001@65.sub-75-218-30.myvzw.com) |
| 21:03.56 | GhOnDiE | i have the way to provision it, just did not know if there was a reliable web based tool. |
| 21:03.59 | GhOnDiE | thanks anyway |
| 21:04.13 | [TK]D-Fender | GhOnDiE: Even stranger to hear someone ask for a web provisioning tool for a phone that meant to be configured via a web interface on the phone itself |
| 21:04.24 | GhOnDiE | well it is |
| 21:04.25 | jaytee | GhOnDiE, just use TFTP with the Grandstreams. |
| 21:04.48 | GhOnDiE | i am, |
| 21:04.49 | [TK]D-Fender | jaytee: He wants a config generator. |
| 21:04.54 | Alan_Hicks | [TK]D-Fender: Ok, I've done what you asked me to do and have no change to my situation. |
| 21:05.00 | GhOnDiE | i was using the voicepulse tool thing |
| 21:05.08 | jasonwoot | wants a pat on the back... after a year, the "forward" softkey is gone on his polycoms |
| 21:05.09 | *** part/#asterisk washburnello (n=johnsy@h23.238.30.71.dynamic.ip.windstream.net) |
| 21:05.15 | GhOnDiE | will stick with how im doing it now |
| 21:05.30 | Alan_Hicks | I've used the stock sip.cfg, modified only to enable presence and to specify my server address (172.16.200.1). |
| 21:05.40 | irieKen | How can I tell which codecs are installed on my asterisk box? |
| 21:05.42 | jaytee | ah, then just use a browser. If you want real centralized provisioning buy a better brand of phone. |
| 21:05.49 | [TK]D-Fender | Alan_Hicks: enable sip debug pastebin a full call attempt which should show the presence notification. |
| 21:05.53 | Alan_Hicks | The stock phone1.cfg was copied over to phone-21{0,1}.cfg and used in each one. |
| 21:06.02 | [TK]D-Fender | irieKen: "show modules like codec" |
| 21:06.19 | jeev | wow |
| 21:06.26 | jeev | i think fender told me never to get a grandstream |
| 21:06.36 | jeev | he also said dont get a wip330... what did i do? got polycoms but got the wip330 too |
| 21:06.50 | GhOnDiE | its not a bad phone |
| 21:06.54 | jaytee | how's that wip330 working out for ya? |
| 21:06.55 | irieKen | D-Fender: thank you. |
| 21:06.55 | GhOnDiE | but i would not recomend it |
| 21:07.12 | Alan_Hicks | [TK]D-Fender: Whoa! That's a lot of lines to pastebin. Give me a few minutes. :-) |
| 21:07.13 | [TK]D-Fender | jeev: Which just goes to show you can lead a horse to the water, but PETA won't let you hold its head under. |
| 21:07.31 | [TK]D-Fender | Alan_Hicks: Yeah, avaluable 10 seconds required... |
| 21:07.43 | jaytee | Iiiiii jjjjjjusst cccouldnttttt handdddle the jjjjittter on the ggggrandstreamszzzzz. |
| 21:07.59 | jaytee | hahahaha, PETA |
| 21:08.03 | jeev | lol |
| 21:08.07 | dlynes | Wwwwwwhhhhhhaaattttt jjjjittttteeerrrr???? |
| 21:08.07 | jaytee | People Eating Tasty Animals |
| 21:08.34 | GhOnDiE | yeah cant say i ever had any jitter |
| 21:08.42 | jaytee | PETA protests all the time outside our offices. |
| 21:08.47 | jaytee | flaky nutcases |
| 21:08.55 | irieKen | Jeev: i made the mistake of buying a bunch of Grandstreams:s |
| 21:09.09 | jaytee | irieKen, sucks doesn't it? |
| 21:09.27 | irieKen | jaytee: yeah.... Works OK, except that incoming calls have this weird choppiness... |
| 21:09.34 | jaytee | there's always Ebay and a world full of fools to dump them on |
| 21:09.49 | jeev | fender is mad mean to me |
| 21:09.50 | irieKen | Jaytee: I can't tell if its the phone or asterisk causing it, but the polycoms don't have that problem. |
| 21:10.21 | irieKen | jayteee: However, the polycoms have their own troubles; when I transfer a call, the person calling can't hear me, but I can hear them... |
| 21:10.27 | [TK]D-Fender | jaytee: http://s38.photobucket.com/albums/e144/freekinacaij/Demotivators/?action=view¤t=9a51a34a453d2e924a0084eea858b52a.jpg |
| 21:10.28 | jaytee | irieKen, it's the bootrom version most likely. I had choppy MOH till I upgraded and had tons of jitter on calls. |
| 21:10.44 | jeev | lol |
| 21:10.46 | irieKen | jaytee: what version are you using now? |
| 21:10.52 | [TK]D-Fender | jaytee>flaky nutcases <- yes, but its getting us a lot of mostly naked celebs ;) |
| 21:11.18 | jaytee | [TK]D-Fender, you know that I work at a zoo, right? |
| 21:11.38 | [TK]D-Fender | jaytee: So do I, but don't tell my co-workers that ;) |
| 21:12.13 | jaytee | [TK]D-Fender, no I'm serious. A real zoo. Elephants, lions, tigers and bears Oh My! and even giraffes. |
| 21:12.36 | jaytee | I've fed a couple of our elephants Krispy Kremes. They love em. |
| 21:13.00 | jaytee | but everytime we have an animal death even due to normal natural causes PETA shows up to protest. |
| 21:13.43 | Alan_Hicks | [TK]D-Fender: I hope this is what you wanted. ;^) http://pastebin.com/d1c8f7ae7 |
| 21:13.53 | jeev | jaytee, fender is the head of peta, he just wont admit it |
| 21:14.19 | TJNII | Don't worry too much about PETA. They don't live in the same world we do and almost everyone knows it. |
| 21:14.32 | jaytee | I just laugh at them |
| 21:14.44 | lowtek | Hey all, anyone that uses MYSQL calls within their dialplan, if asterisk can't make a connection to a database, does it fallthrough or terminate the channel? |
| 21:14.48 | [TK]D-Fender | Alan_Hicks: #823. Its there. Still just did your phones wrong somewhere |
| 21:15.15 | TJNII | http://www.chron.com/disp/story.mpl/front/5937293.html |
| 21:15.29 | Alan_Hicks | [TK]D-Fender: Wish I knew where that was. |
| 21:16.13 | [TK]D-Fender | Alan_Hicks: Verify your buddies screen. |
| 21:16.14 | lunaphyte | have any of you guys that use voipjet had trouble lately (the last day or so maybe) sending calls to test.voipjet.com? |
| 21:17.33 | *** join/#asterisk chevap (n=chevap@89.201.196.182) |
| 21:21.19 | jaytee | irieKen, sorry but I had a call. The GXP2000 program and bootrom versions I upgraded to are Program-- 1.1.5.15 Bootloader-- 1.1.5.6 |
| 21:22.15 | Alan_Hicks | [TK]D-Fender: Yeah the buddies screen is fine. Like I said before, it continued to update the buddy's status, but just plain flat out doesn't do BLF. |
| 21:22.43 | lowtek | Alan_Hicks / [TK]D-Fender: I have this same issue after upgrading to 1.4.22.1 and bootrom 3.0/latest sip_ld |
| 21:22.56 | [TK]D-Fender | Alan_Hicks: You may need to upgrade your firmware. What ver? |
| 21:23.22 | Alan_Hicks | BootRom_4_0_0_release_sig.zip That what you're looking for? |
| 21:23.27 | [TK]D-Fender | Alan_Hicks: no, SIP |
| 21:23.32 | Alan_Hicks | 2201-06642-001.sip.ld |
| 21:23.37 | [TK]D-Fender | .... |
| 21:23.50 | Alan_Hicks | I'll read it off the phone. :-) |
| 21:23.52 | [TK]D-Fender | Alan_Hicks: NORMAL # please |
| 21:25.00 | Alan_Hicks | 2.2.0.0047 |
| 21:25.02 | jaytee | be back later |
| 21:25.04 | jeev | uh, is there a level of debug if you enable iax2 debug? |
| 21:25.22 | jeev | i'm makin g a call and it's saying service unavailable, half the time it's working and half it's not.. it's not putting anything in the log when it doesn't work |
| 21:26.20 | jeev | i wonder if it's an X-Lite issue cause i'm connected remotely (just messing around with voicemail) |
| 21:26.30 | Kobaz | is there a way to cause a channel to go on hold through the AMI (or something similar) |
| 21:26.43 | [TK]D-Fender | jeev: Because yeah.... X-Lite's IAX stack has "issues".... |
| 21:27.36 | jeev | Fender, it's connected to SIP, allowed it on the firewall of the dsl conncetion and i'm at another office.. and when i dial my number, i either get service available (nothing in the logs) or i get the call going through and the log works fine |
| 21:27.49 | jeev | i will just assume it's an x-lite issue as this reallhy isn't that importnat. |
| 21:27.51 | jeev | important, orry |
| 21:27.53 | jeev | sorry. |
| 21:29.12 | lowtek | ~c |
| 21:29.13 | jbot | hmm... c is for maniacs C code. C code run. Run, code, run. Please? |
| 21:29.21 | lowtek | ~centos |
| 21:29.22 | jbot | from memory, centos is an Enterprise-class Linux Distribution derived from sources freely provided to the public by a prominent North American Enterprise Linux vendor. Check it out at http://www.centos.org/projects/centos, or http://www.voip-info.org/wiki/view/Asterisk+Linux+Centos |
| 21:30.06 | *** join/#asterisk aliver (n=aliver@ip-216-17-160-99.rev.frii.com) |
| 21:30.36 | jeev | fender, i have dial plan issues i need to solve. i make retards look good. |
| 21:30.55 | aliver | is there a way I can verify that a specific SIP client is "online" and working with asterisk? I know I can do a "sip show peers" but it shows SIP phones that I know are offline. |
| 21:31.12 | jeev | try making a call to it |
| 21:31.42 | aliver | Well, right now I only have the one SIP phone. |
| 21:31.52 | aliver | I guess I could fire up a softphone and try that. |
| 21:32.03 | jeev | i wanna beat fender with a hard phone |
| 21:32.04 | [TK]D-Fender | aliver: And what does it show exactly for such a phone? |
| 21:32.23 | [TK]D-Fender | goes to oil up his katana again.... |
| 21:32.23 | aliver | it shows this: 2251 10.10.12.85 A 5060 Unmonitored |
| 21:32.40 | [TK]D-Fender | aliver: And thats because you did not tell * to LOOK for its status. |
| 21:32.43 | aliver | it's the "unmonitored" thing I'm guessing? |
| 21:32.48 | [TK]D-Fender | aliver: "qualify=yes" <---- |
| 21:32.53 | aliver | Ahhhh... |
| 21:33.05 | jeev | fender, your katana against my g26 or xd40. pick one! |
| 21:34.23 | aliver | okay I take it that needs to go into the phones section in sip.conf |
| 21:34.45 | aliver | I'm not worth using the katana on. You don't want retard blood on it. |
| 21:34.46 | [TK]D-Fender | Glocks and Glock wanna-bes. |
| 21:34.47 | [TK]D-Fender | bleh |
| 21:35.24 | jeev | lol |
| 21:35.43 | [TK]D-Fender | Sig Sauer FTW |
| 21:36.08 | tzanger | AWWWWWWW FUCK YEAH |
| 21:36.10 | tzanger | 01:0b.0 Bridge: Unknown device 1895:0001 (rev 01) |
| 21:36.18 | tzanger | w00ty w00t woot! |
| 21:36.55 | jasonwoot | yes? |
| 21:36.57 | jasonwoot | oh |
| 21:37.04 | tzanger | HAHAHAH |
| 21:37.13 | tzanger | now _that_ is funny |
| 21:37.38 | Qwell | ? |
| 21:38.03 | Qwell | tzanger: you trying to get calls out of your toaster again? |
| 21:38.17 | TJNII | A noble cause |
| 21:38.27 | TJNII | has plans to network the Washer |
| 21:38.43 | tzanger | Qwell: that is my first ever programmed PCI FPGA |
| 21:38.48 | Qwell | I see |
| 21:39.10 | lowtek | There needs to be a dialplan function to lookup the mailbox as defined in a sip peer ... I wish I knew C well enough to write my own ... |
| 21:39.14 | TJNII | tzanger: Nice. Very nice |
| 21:39.18 | tzanger | it does absolutley nothing but show up on the PCI bus, but once I hook something up to the other side of that PCI interface (DMA bus master, TDM bus, anything really) it's all mine |
| 21:39.43 | TJNII | tzanger: DIY card or are you using a breakout board? |
| 21:39.52 | jeev | i want a sig. |
| 21:39.54 | jeev | i want a ppk |
| 21:40.07 | tzanger | that was just proof of concept, it's the opencores PCI master core with nothing on the other side |
| 21:40.09 | jeev | but the most important thing is fender's head stuffed on my den wall |
| 21:40.13 | TJNII | ~trout jeev |
| 21:40.13 | jbot | ACTION slaps jeev around a bit with a large trout! |
| 21:40.39 | tzanger | TJNII: it's an Enterpoint minican2 dev board |
| 21:40.55 | jeev | i hate seafood. |
| 21:40.56 | tzanger | basically a 1.5mil gate spartan 3 on a PCI edge and nothing else |
| 21:41.14 | TJNII | I see |
| 21:41.37 | [TK]D-Fender | PPK? a "ladies" gun, and don't give me that "Bond" crap. |
| 21:41.42 | tzanger | next I have to get pci fake hotplugging to work so I don't have to keep rebooting the machine |
| 21:41.50 | jeev | no, it looks cool |
| 21:42.01 | TJNII | Interesting. I've only played with gates and MCUs on serial ports. I was never brave enough to plug somehing I mage into a card slot. |
| 21:42.19 | tzanger | I made an ISA I/O board about 16 years ago |
| 21:42.21 | tzanger | I still hae it somewhere |
| 21:42.27 | tzanger | got sick of trying to work wiht parallel ports |
| 21:42.41 | tzanger | had 3 8255 I think the chip was |
| 21:43.25 | lowtek | Or better yet a function that would let you assign the different sections of the sip definition to a channel variable.. i.e., Set(MAILBOX=SipLookup(801)) |
| 21:43.42 | lowtek | er... Set(MAILBOX=SipLookup(801,Mailbox)) |
| 21:45.37 | TJNII | USB will probably be my next adventure. I have some PICs with built in USB trancievers in them but I messed up and didn't get flash based chips. |
| 21:45.50 | TJNII | One time programmable is not goot for prototypes. |
| 21:45.54 | TJNII | s/goot/good/ |
| 21:46.23 | jasonwoot | can someone please pastebin me a sample of their dial plan for agent login? |
| 21:46.42 | [TK]D-Fender | jeev: the XD reminds me a lot of the SIG P series actually... minus the hammer |
| 21:46.55 | [TK]D-Fender | X D |
| 21:46.57 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 21:49.11 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 21:49.35 | jeev | i love it, |
| 21:49.41 | aliver | I started asterisk with "-v -v -v" but when I pick up my sip phone and dial a wrong number or my own number it just says 404 not found |
| 21:49.56 | aliver | Can I make * show me some errors or what not? |
| 21:50.02 | aliver | or is it my crappy grandstream SIP phone? |
| 21:50.22 | aliver | when I do a "sip show peers" it does show up now. |
| 21:50.31 | aliver | or at least it shows the latency to ping it. |
| 21:50.44 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 21:51.10 | TJNII | Sounds like a dialplan problem...... |
| 21:51.12 | flujan | hello all. |
| 21:51.25 | jeev | fender, io love my glock 26 though |
| 21:51.46 | aliver | Well, that's probably true, but is there a way to get * to complain about something I can google? |
| 21:51.50 | lowtek | I got a Glock 20 (10mm) |
| 21:52.24 | flujan | guys I am having a trouble with asterisk using two sip trunks. The first one is on the network 10.213.129.x and the second on the 172.16.0.1 network. I just have one network card on this machine |
| 21:52.54 | jeev | 10mm heh, such a weird projectile |
| 21:53.12 | flujan | both sip trunks connects and works. I can place calls and when I heard the other users... but they can't hear me in the second sip trunk . |
| 21:53.21 | *** join/#asterisk hron84 (n=hron@s8u9dyor7k.adsl.datanet.hu) |
| 21:53.27 | flujan | it is a nat issue |
| 21:53.44 | flujan | so, how can I solve this? can I have multiple sip trunks on the same machine with different networks? |
| 21:54.04 | lowtek | Are sip.conf variables accessible via the DB functions? |
| 21:54.24 | lowtek | flujan: Are you binding on the required addresses in sip.conf? |
| 21:54.55 | flujan | lowtek: do you mean the host option? |
| 21:55.08 | hron84 | Hi, somebody can help me? I installed * 1.6 to vmware. I would like test echo but it doesn't play any file. error: app_playback.c: no such file or directory. languageprefix has been enabled, but nor enabled nor disabled state not helps |
| 21:55.22 | lowtek | flujan: I didn't say anything about a "host option" |
| 21:55.23 | *** join/#asterisk l2cache (n=l2cache@155.132.204.68.cfl.res.rr.com) |
| 21:55.32 | hron84 | ast can reach these files ofc |
| 21:55.34 | lowtek | bindaddr=0.0.0.0 |
| 21:55.43 | lowtek | flujan: Which ver of asterisk? |
| 21:56.50 | flujan | 1.4.21.2 |
| 21:57.28 | hron84 | how can i view where finds asterisk its sound files? |
| 21:57.29 | lowtek | In sip.conf, bindaddr=0.0.0.0 |
| 21:58.01 | aliver | Are Polycom SIP phones as good as the corporate asterisk-review sites say or is that hype? |
| 21:58.18 | aliver | Or do hardphones really differ that much in terms of quality? |
| 21:58.24 | lowtek | When asterisk 1.4.x loads up it's configuration from the various files (sip.conf, extensions.conf, static-realtime) does it store it in the Asterisk DB or just in memory? |
| 21:58.51 | lowtek | aliver: They are the best for most SIP applications, imho |
| 21:59.04 | flujan | lowtek: http://pastie.org/252564 |
| 21:59.36 | lowtek | Yea, like I said, bindaddr=0.0.0.0 |
| 21:59.43 | lowtek | under general |
| 21:59.46 | jasonwoot | aliver polycom=run away |
| 21:59.53 | aliver | lowtek Well, it figures, they are pretty expensive. |
| 22:00.09 | lowtek | flujan: i think the default is to bind to all but try it anyway |
| 22:00.13 | aliver | jasonwoot Really? So the hype is undeserved? What do you use instead? |
| 22:00.32 | *** join/#asterisk Gat0rvean (n=gredish@64.191.128.145) |
| 22:00.49 | jasonwoot | we've been converting away from desktop sets to software phones as often as possible, |
| 22:00.49 | lowtek | flujan: pastebin your ifconfig ... |
| 22:00.51 | TJNII | lowtek: You have nat=yes set, correct? |
| 22:00.58 | jasonwoot | the 501s aren't that expensive in bulk, |
| 22:01.09 | lowtek | TJNII: For what? |
| 22:01.23 | TJNII | Sorry lowtek that was for flujan |
| 22:01.31 | lowtek | aliver: The 330's are great for the price ... |
| 22:01.35 | lowtek | $100 |
| 22:02.37 | aliver | jasonwoot at my gig, software phones wouldn't fly. People here are too low tek. |
| 22:03.18 | jasonwoot | aliver: yeah, they're still necessay for people who are married to the handset, |
| 22:03.34 | lowtek | soft phones suck, imho |
| 22:03.58 | lowtek | not to mention you inherit PC support when there's a problem |
| 22:03.58 | jasonwoot | z o i p e r |
| 22:04.17 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 22:04.19 | lowtek | zoiper is one of the better ones, I agree, this is the one I use for testing |
| 22:04.27 | flujan | lowtek: binaddr=0.0.0.0 do not work in the sip.conf |
| 22:04.31 | lowtek | ExpressTalk is fair, I hate X-Lite |
| 22:04.48 | flujan | TJNII: yeap... i set both trunks with the nat=yes |
| 22:04.54 | TJNII | flujan: Pastebin the output of your ifconfig |
| 22:04.54 | [TK]D-Fender | flujan: works better when you spell it right |
| 22:05.00 | lowtek | flujan: That's all I got. I bind to multiple addresses that way. |
| 22:05.11 | lowtek | flujan: But I have two physical nics |
| 22:05.15 | flujan | [TK]D-Fender: hello man... how are you doing? lol |
| 22:05.45 | lowtek | flujan: Oops, yea, try bindaddr=0.0.0.0 |
| 22:05.45 | jasonwoot | if someone will pastebin me a sample of their agent login dial plan I'll give them a million dollars |
| 22:05.53 | lowtek | jasonwoot: PayPal? |
| 22:06.10 | jasonwoot | two party, out of state, bad check |
| 22:06.30 | flujan | lowtek: you are right... I made the mistake typing it here again. |
| 22:06.46 | lowtek | I'm out, thanks some for the help, others for nothing |
| 22:07.01 | flujan | lowtek: thanks for the help you too |
| 22:08.13 | _fury | In my dial plan, I'm trying to use WaitExten, I have extensions like _NXXNXXXXXX to accept a phone number. If a user dials just three numbers for instance, I want it to time out after about 2 seconds. I've used Set(TIMEOUT(digit)=2) before WaitExten(10). is that correct? it doesn't seem to have any effect. |
| 22:09.04 | TJNII | So you want a two second timeout? |
| 22:09.44 | flujan | TJNII: http://pastie.org/252564 |
| 22:10.51 | flujan | TJNII: I can hear people talking, but they cannot hear me... if I change the default gw... the 172.16 trunk stop working and the 10.0.x works fine |
| 22:10.55 | _fury | TJNII: yeah |
| 22:11.12 | TJNII | _fury: Then why are you telling waitexten you want a 10 second timeout? |
| 22:11.15 | [TK]D-Fender | _fury: Set digit & response timeouts separately and do not pass a timeout to waitexten. let your other setups do that |
| 22:11.38 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 22:12.02 | _fury | well if there's no input at all after 10 seconds, I want it to play "goodbye" and hang up, but if there is only say 2 digits, I don't want it to wait for the 5 seconds it waits now before it reports the error |
| 22:12.18 | jeev | i need to learn contexts and stuff, bad. |
| 22:13.51 | TJNII | flujan: I'm confised. Can you post your routing table with the 10. peer working and the routing table with the 172.16 routing table so I can compare them? |
| 22:14.23 | *** join/#asterisk rotzak (n=rotzak@64-42-108-202.atgi.net) |
| 22:15.53 | [TK]D-Fender | _fury: pastebin what you've done. |
| 22:16.59 | flujan | TJNII: http://pastie.org/252564 |
| 22:19.54 | flujan | it is like the packages from my network is not hitting the trunk ... but the trunk is sending the packages to right to me ... |
| 22:19.59 | [TK]D-Fender | flujan: "canreinvite=no" <---------- |
| 22:20.07 | *** join/#asterisk mags3 (n=stryfe@pool-151-203-62-170.bos.east.verizon.net) |
| 22:20.12 | [TK]D-Fender | nvm |
| 22:20.12 | *** part/#asterisk mags3 (n=stryfe@pool-151-203-62-170.bos.east.verizon.net) |
| 22:21.03 | TJNII | flujan: But in all cases it is one way audio? The SIP initiation packets get through ok? |
| 22:21.04 | flujan | ? |
| 22:21.43 | flujan | TJNII: yes for both ... |
| 22:21.46 | jeev | Fender, have i told you that i love iax? |
| 22:22.07 | flujan | [TK]D-Fender: what is wrong with that option? |
| 22:22.13 | jeev | have i told you that i dont know anything iax? i just use it to traverse my firewall? |
| 22:22.27 | flujan | [TK]D-Fender: i set to canreinvite=yes but nothing changed... |
| 22:22.38 | [TK]D-Fender | flujan: pastebin a call. |
| 22:22.48 | [TK]D-Fender | flujan: And check your firewall |
| 22:23.07 | TJNII | pastebin it with sip debug on |
| 22:23.17 | TJNII | So we can see what IP addresses are in the headers |
| 22:23.24 | jeev | pastebin it with core debug too |
| 22:24.29 | flujan | [TK]D-Fender: http://pastie.org/252591 |
| 22:24.43 | *** join/#asterisk justmehere (n=justmehe@24-176-158-178.dhcp.kgpt.tn.charter.com) |
| 22:25.42 | justmehere | Howdy all, I know Asterisk doesn't like vm's, but was wondering if there is any difference in using on ESXi? |
| 22:25.59 | justmehere | I mean, is it practical to run * on a virtual machine? |
| 22:26.18 | drmessano | Isn't ESXi a virtual machine? |
| 22:26.27 | justmehere | no, it's the host ;-) |
| 22:26.50 | drmessano | Oh, an expert |
| 22:27.32 | justmehere | I'm not interfacing with any fxo or fxs cards, what kind of possible side effects am I facing moving it to vm? |
| 22:27.37 | drmessano | Doesn't ESXi exist to host virtual machines? |
| 22:27.57 | drmessano | So therefore, if you're using ESXi, one can assume you're using Virtual machines |
| 22:27.58 | *** join/#asterisk l2cache (n=l2cache@155.132.204.68.cfl.res.rr.com) |
| 22:28.15 | justmehere | haha, nicely worded, yeah, I use it for other servers, and understand *'s need for consistent clock cycles, |
| 22:28.17 | drmessano | Which referring back to comment --> "I know Asterisk doesn't like vm's |
| 22:28.27 | drmessano | You have answered your own question |
| 22:28.37 | drmessano | So paypal me the $25 and have a nice day |
| 22:28.48 | justmehere | umm, if you just want to run in circles, then stfu, I'm asking serious question |
| 22:28.55 | justmehere | I don't know much about the subject, and wanting to |
| 22:29.03 | TJNII | flujan: Which IP was that supposed to go to? |
| 22:29.05 | justmehere | know the real world effects others might have experienced |
| 22:29.12 | drmessano | You stated that Asterisk doesnt like VMs, then asked if it can run on ESXi, which hosts VMs |
| 22:29.16 | drmessano | Are you stupid? |
| 22:29.21 | _fury | [TK]D-Fender: http://rafb.net/p/0JdTvY12.html |
| 22:29.30 | _fury | [TK]D-Fender: Also I can't get the Ringing() to work in the beginning of the call |
| 22:29.38 | justmehere | sure, but what was my question you fucking retard, did I ask, "Duh, does * like VM's? |
| 22:29.49 | justmehere | no, I asked about repercussions |
| 22:29.57 | justmehere | and experience other may have had |
| 22:29.58 | drmessano | No, you stated it does not.. As in, based in fact |
| 22:30.12 | drmessano | Then asked if you could run it on a particular VM product |
| 22:30.14 | justmehere | still what was my question? |
| 22:30.16 | drmessano | Which, is, stupud |
| 22:30.19 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 22:30.20 | drmessano | s t u p u d |
| 22:30.27 | TJNII | flujan: NM. I see it. |
| 22:30.35 | flujan | ow ok |
| 22:30.42 | flujan | lol I was just typing it |
| 22:30.46 | [TK]D-Fender | _fury: What happens on timeout? |
| 22:30.47 | flujan | the sip negotiation is goog |
| 22:30.49 | flujan | ops |
| 22:30.50 | flujan | good |
| 22:30.59 | flujan | the issue lies with the rtp IMHO. |
| 22:31.03 | _fury | [TK]D-Fender: it plays vm-goodbye and hangs up |
| 22:31.09 | drmessano | "but was wondering if there is any difference in using on ESXi?" <--- no, RTFM |
| 22:31.17 | TJNII | flujan: I believe that is correct |
| 22:31.28 | [TK]D-Fender | _fury: So what DOESN'T it do properly? |
| 22:31.42 | flujan | [TK]D-Fender: TJNII: any ideas i can try? |
| 22:32.04 | _fury | [TK]D-Fender: I want to use 10 digits for input. If I do 3 digits, it takes about 5 seconds to go to the _. rule which brings it back up |
| 22:32.05 | TJNII | flujan: I want to say it has to do with the IP addresses * is putting in the RTP headers, but with 3 different subnets on that box I don't know what to tell you to do. |
| 22:32.09 | _fury | I want it to take more like 2 seconds |
| 22:32.56 | flujan | the audio from my extension 40005 do not hit the phone but the phone audio hit my peer |
| 22:33.03 | flujan | so the problem is with me sending audio |
| 22:33.46 | justmehere | Anyone know the possible side effects of running a * in a VM? |
| 22:33.48 | [TK]D-Fender | flujan: As in type 3 digits, wait for 5 seconds? |
| 22:34.29 | _fury | [TK]D-Fender: if you're in the US and want to call my number, you can see what happens - if I type just 3 digits (it expects 10), I want it to wait 2 seconds before going back to ask for input again. Instead, it waits it seems like somewhere around 5 or 7 |
| 22:34.39 | flujan | [TK]D-Fender: ? |
| 22:34.43 | _fury | more importantly, if somebody presses 0, I want it to go right away |
| 22:35.09 | TJNII | flujan: This call in the debug was from some device with a 192.168 address (40005) to the 10. device at 99514177? |
| 22:36.04 | flujan | TJNII: yeap my internal subnet... i have tree one with the extensions and hardphones, and two with sip trunks... I am activating the second one with the sip trunk right now |
| 22:37.31 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:37.55 | flujan | TJNII: this is the sip trunk debug http://pastie.org/252599 with a call from my extension 40005 to the trunk |
| 22:38.54 | justmehere | Anyone know what the possible side effects are when running Asterisk in a VM? it's a light load server (~40 calls per hour, only a few concurrently, no FXO/FXS boards, all SIP phones, no transcoding) |
| 22:39.45 | jaytee | nausea, diarrhea, vomitting, itching, boils and possible blindness |
| 22:39.55 | *** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
| 22:39.59 | jaytee | oh! and erectile dysfunction |
| 22:40.12 | drmessano | Running asterisk in a VM can also lead to early termination |
| 22:40.13 | justmehere | oh, drmessano said that was from his wanking |
| 22:40.29 | justmehere | I've ignored him, let me know if he says anything worthwhile |
| 22:40.35 | drmessano | or premature retardation, ala justmehere |
| 22:40.40 | TJNII | flujan: Can you control the servers on the other end? |
| 22:40.54 | TJNII | flujan: NM, that won't work. |
| 22:40.58 | drmessano | If he ignore me, I must have gotten to him.. Job DONE |
| 22:41.03 | flujan | TJNII: no that is from the pstn |
| 22:41.04 | drmessano | ignored* |
| 22:41.11 | flujan | TJNII: I GOT IT |
| 22:41.30 | flujan | TJNII: the problem is with the route... i need to add a route to the entire 10.0.0 network |
| 22:41.30 | TJNII | What did you do? |
| 22:41.38 | flujan | not only to the gateway |
| 22:41.45 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 22:41.45 | flujan | the other end ( telco ) |
| 22:42.06 | flujan | is using a lot of 10.x.x.x peers in the dialog and in the rtp... thanks to wireshark... :D |
| 22:42.15 | *** part/#asterisk justmehere (n=justmehe@24-176-158-178.dhcp.kgpt.tn.charter.com) |
| 22:42.22 | TJNII | flujan: Pastebin your routing table now. I'd like to see it. |
| 22:43.23 | TJNII | Oh. You had no 10.0.0.0 |
| 22:43.26 | TJNII | I see it now |
| 22:43.51 | flujan | yeap |
| 22:44.04 | flujan | thanks for the help TJNII |
| 22:44.16 | TJNII | And when you changed 0.0.0.0 to test 10. you took out the gateway for 172.16 so of course that didn't work. |
| 22:44.26 | flujan | YES |
| 22:44.29 | flujan | that is it |
| 22:44.36 | flujan | thanks for the help [TK]D-Fender :) |
| 22:45.03 | TJNII | wishes he saw it sooner, he was looking for it. ;) |
| 22:45.33 | [TK]D-Fender | flujan: I missed it too... |
| 22:46.34 | flujan | thanks for the help guys |
| 22:46.39 | flujan | you are cool :D |
| 22:46.46 | flujan | ok I am dumb lol |
| 22:47.24 | drmessano | This kinda nuts |
| 22:47.50 | drmessano | I just checked one Asterisk box, and it has 127.0.0.1 assigned |
| 22:47.58 | drmessano | I checked another, SAME IP |
| 22:48.14 | drmessano | So for the hell of it, I checked my desktop... 127.0.0.1!!! |
| 22:48.21 | drmessano | I think I have been ROOTED |
| 22:51.26 | DarienWork | rewt! |
| 22:51.35 | TJNII | NOES! You have a virus! Boil your hard drive! |
| 22:51.58 | DarienWork | lol ur dum u should be using windows it is moar secur then stupod linux crap lol! |
| 22:53.27 | jameswf-home | yay |
| 22:53.29 | metastable | qwell: any word on the compile error on 'make b410p' yet ? |
| 22:53.31 | drmessano | Linux is lame.. If I wanted MS-DOS, I would use stupid old MS-DOS |
| 22:54.15 | jameswf-home | is still searching for the alias file that gives funny errors on all dos commands |
| 22:54.19 | metastable | if i wanted ms-dos, i'd fire up my toaster |
| 22:56.10 | *** join/#asterisk l2cache (n=l2cache@155.132.204.68.cfl.res.rr.com) |
| 22:57.12 | *** part/#asterisk hron84 (n=hron@s8u9dyor7k.adsl.datanet.hu) |
| 22:59.49 | metastable | the 127.0.1.1 assignments are stupid |
| 23:00.06 | metastable | they don't interfere, but i'm uncomfortable with it |
| 23:01.08 | *** join/#asterisk irieKen (n=chatzill@rrcs-74-87-28-55.west.biz.rr.com) |
| 23:03.51 | scooby2 | i wish i could figure out the circuit settings this old * 1.2 SigMAN system |
| 23:05.24 | *** join/#asterisk mvanbaak_ (n=michiel@vanbaak.xs4all.nl) |
| 23:12.26 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
| 23:23.10 | *** join/#asterisk moos3 (n=richard@pool-70-105-227-38.port.east.verizon.net) |
| 23:23.19 | moos3 | anyone know anything about qsig? |
| 23:25.51 | moos3 | anyone there? |
| 23:27.21 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 23:31.09 | moos3 | anyone know what port q.sig uses |
| 23:34.33 | jaytee | qsig doesn't use TCP ports, it's an ISDN signalling protocol that uses the D channel on PRIs |
| 23:35.24 | moos3 | I was told that you can do qsig over tcp |
| 23:36.05 | jaytee | really? don't know why that would be. can you ask the person that told you that what port it uses? |
| 23:36.41 | moos3 | i want to interface my samsung officeserv 100 IP phone system with my home asterisk |
| 23:37.04 | moos3 | qsig over SIP is that possible? |
| 23:37.57 | jaytee | I've read about q.sig tunneling over h.323 and SCTP with Cisco Call Manager but never used it. |
| 23:40.10 | jameswf-home | heh http://dontcallmyboss.blogspot.com/2008/08/cox-communications-wow.html |
| 23:40.53 | jameswf-home | isp FAIL |
| 23:46.36 | *** join/#asterisk ZX81 (n=matt@202.20.97.211) |
| 23:47.04 | ZX81 | grrrr! Anyone know how to get call-limit into a realtime table in mysql bearing in mind you can't use - in a field name? |
| 23:47.19 | irieKen | Does anyone know why Asterisk would handle calls transfered by a polycom differently than another phone? |
| 23:48.37 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 23:48.38 | ZX81 | hmmm |
| 23:48.38 | ZX81 | `call-limit` smallint(5) unsigned default NULL, |
| 23:48.46 | ZX81 | but mine (5) won't allow it |
| 23:49.14 | *** join/#asterisk pcrane (n=pcrane@202.49.106.158) |
| 23:49.18 | ZX81 | hey |
| 23:49.23 | pcrane | hey hey |
| 23:49.23 | ZX81 | ALTER TABLE sip_buddies ADD COLUMN call-limit int fails |
| 23:49.25 | ZX81 | :) |
| 23:49.31 | ZX81 | will join room in a sec |
| 23:49.34 | pcrane | kk |
| 23:55.03 | [TK]D-Fender | irieKen: It doesn't in any way I can think of. Could you be more specific? |
| 23:55.18 | *** join/#asterisk Assimilate (n=Assimila@24-116-182-58.cpe.cableone.net) |
| 23:55.43 | *** join/#asterisk l2cache (n=l2cache@155.132.204.68.cfl.res.rr.com) |
| 23:55.59 | irieKen | [TK]D-Fender: For some reason, when I transfer a call from my polycom to any other extension, the caller that I transfered can't hear me, but I can hear them. |
| 23:56.40 | [TK]D-Fender | irieKen: Makes little sense unless you're dealing with multiple subnets & reinvites |
| 23:57.11 | irieKen | [TK]D-Fender: Subsequently, if I transfer the call again (from a different phone, after having it transfered from the polycom), the problem persists. However, if I transfer from any other kind of phone when the call first comes in, it is fine, until it touches a Polycom:S |
| 23:57.30 | [TK]D-Fender | irieKen: strange..... |
| 23:57.38 | irieKen | [TK]D-Fender: These are all on the same subnet... I'm not quite sure what reinvites would entail. |
| 23:57.54 | irieKen | [TK]D-Fender: yeah, very strange... Been trying to figure it out all day. |
| 23:58.24 | ZX81 | [solved] use backticks around field name with hypen |
| 23:58.28 | ZX81 | *hyphen |
