| 00:00.09 | jaytee | <etfonhomey> jaytee, 1.39 cents / minute |
| 00:01.05 | jaytee | if you're talking .139 a minute then that's a whole different ball game and cost effective but you f'd up on the decimal point. Hope you don't work as an accountant :-) |
| 00:02.35 | etfonhomey | jaytee, maybe I'm missing something here. 1.39 cents / minute = $0.0139 / minute right? |
| 00:03.02 | etfonhomey | $0.0139 * 10 (for a 10 minute call) = $0.139, right |
| 00:03.19 | etfonhomey | $0.139 is most certainly not 1/3rd of $44, right? |
| 00:03.22 | jaytee | etfonhomey, hey, you typed 1.39/minute to begin with. that's where all this confusion started. |
| 00:03.43 | etfonhomey | Actually, I typed 1.39 cents / minute |
| 00:03.45 | seanbright | jbot, .0139 * 10 |
| 00:03.46 | jbot | 0.139 |
| 00:03.50 | seanbright | ok, good. |
| 00:03.53 | seanbright | math checks out |
| 00:04.32 | etfonhomey | Hey, didn't know jbot was a calculator, too! |
| 00:04.38 | seanbright | he's versatile |
| 00:04.46 | jaytee | etfonhomey, well thanks I'll check out vitelity's website and see |
| 00:05.16 | etfonhomey | jaytee, if I were to give you a "con" for Vitelity it's that if you need immediate response to a help ticket, they charge a fee. |
| 00:05.18 | jaytee | jbot is smarter than most humans I've met |
| 00:05.32 | Carlos_PHX | And certainly more likeable. |
| 00:06.15 | *** join/#asterisk Akiyuki (n=jimi@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 00:06.44 | Carlos_PHX | ROFL, way to give a noob a complex. |
| 00:06.55 | jaytee | was that aimed at me? or the noob? |
| 00:07.01 | etfonhomey | [TK]D-Fender, welcome back, you missed a good one while you were gone. |
| 00:08.15 | etfonhomey | jaytee, I'm talking about aep |
| 00:09.41 | Akiyuki | haha |
| 00:09.59 | Akiyuki | Carlos_PHX: Check your PM |
| 00:10.05 | Akiyuki | Ready to set up an account, possibly |
| 00:12.12 | etfonhomey | jaytee, I forgot to mention the 1.39 CENTS / minute was outbound calling rate. |
| 00:13.03 | etfonhomey | jaytee, inbound is 1.1 CENTS / minute. |
| 00:17.35 | etfonhomey | jaytee, Just looked at the advertised rates and they are a little higher. Guess they raised them since I signed up. |
| 00:18.39 | jaytee | etfonhomey, just to get this straight. that's one dollar and thirty nine center PER MINUTE? |
| 00:19.17 | etfonhomey | jaytee, no that would be 139 CENTS / minute (see, no decimal point) |
| 00:20.26 | etfonhomey | jaytee, from now on, I'll give it to you in dollars. $0.0139 / minute outgoing and $0.011 / minute incoming $1.49 / month / DID |
| 00:21.47 | jaytee | etfonhomey, I think I'll just read vitelity's page, thanks |
| 00:22.18 | jaytee | but I get it now |
| 00:22.54 | jaytee | 1.39 CENTS |
| 00:23.15 | seanbright | jbot, stab them |
| 00:23.15 | jbot | ACTION runs at them with an origami Swiss Army knife, and inflicts a nasty paper cut. |
| 00:26.16 | etfonhomey | jaytee, I just tried some more math. It looks like for $44 at Vitelity / month, you could get 1 DID and make 3,058 minutes of outgoing calls and I believe it's free long distance in the continental US. |
| 00:26.20 | etfonhomey | Dinner time. |
| 00:27.20 | aep | i cant get the sip client connect, it always says timeout ;/ |
| 00:27.40 | aep | any idea how to test a udp connection? telnet obviously doesnt work |
| 00:28.08 | [netman] | nc (netcat) -u |
| 00:28.16 | etfonhomey | telnet = tcp |
| 00:29.09 | aep | etfonhomey: that was my point |
| 00:29.13 | aep | [netman]: oh cool thanks |
| 00:29.33 | [netman] | you are welcome |
| 00:29.34 | *** part/#asterisk E-bola (i=psybnc@ip181.rev112.brygge.net) |
| 00:29.35 | etfonhomey | aep, you know you tried it. admit it. |
| 00:30.33 | aep | it does something |
| 00:30.42 | aep | anything i could type to expect a response? |
| 00:31.54 | aep | erm i have the bad feeling that this "black box" router from my ISP simply doesnt support udp |
| 00:33.33 | seanbright | doesn't support UDP? |
| 00:33.57 | seanbright | i find that hard to believe |
| 00:34.08 | aep | end users dont need udp |
| 00:34.17 | aep | modern routers even proxy your http and smtp |
| 00:34.31 | aep | becouse end users love it... |
| 00:34.35 | jql | hrm... a world without udp |
| 00:34.44 | aep | anyway i just transfered data via udp and netcat |
| 00:34.48 | aep | so that isnt the problem |
| 00:34.48 | jql | no WoW... no traceroute |
| 00:34.53 | jql | my world would be bereft |
| 00:35.13 | aep | wow needs udp? great |
| 00:35.19 | aep | then they wont shut that down :D |
| 00:35.35 | jaytee | ~itsplist-us |
| 00:35.36 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 00:36.15 | aep | i can type whatever i want, asterisk won't respond |
| 00:36.21 | aep | tryes from localhost |
| 00:36.37 | aep | nope |
| 00:36.51 | aep | nc -u localhost 5060 |
| 00:36.56 | aep | nothing i type does anything |
| 00:37.05 | aep | (yes i pressed enter...) |
| 00:38.10 | aep | udp 0 0 sophia:32824 sophia:sip ESTABLISHED |
| 00:38.15 | aep | so not a connection problem |
| 00:38.35 | [netman] | 1) netman@apolo:~$ nc -l -u -p 1111 |
| 00:38.52 | [netman] | 2) netman@apolo:~$ nc -u localhost 1111 |
| 00:39.02 | [netman] | 3) hey |
| 00:39.16 | [netman] | and u should see "hey" on the window 1) |
| 00:39.16 | aep | that prints hey for me on the other one |
| 00:39.18 | aep | yes |
| 00:39.21 | aep | i tryed that |
| 00:39.25 | aep | everything fine with the connection |
| 00:39.36 | aep | works the other way round as well |
| 00:40.08 | aep | however doing the same on port 5060 where asterisk is, does nothing |
| 00:40.15 | aep | no log entry, nothing on the cli |
| 00:40.26 | [netman] | and sip debug? |
| 00:41.04 | *** join/#asterisk loather-work (n=khudson@69.43.168.134) |
| 00:41.20 | aep | i'f i'msending random crap into the port, asterisk cpu usage goes up |
| 00:41.20 | [netman] | maybe u should try something like traceroute o ngrep to find out what is wrong on 5060 port |
| 00:41.35 | loather-work | ok, i'm out of channels on my PRI, but i also have a SIP provider. Is there an easy way to have asterisk try dialling on the PRI first, then dial via the sip provider when all the PRI channels are full? |
| 00:41.53 | aep | <--- SIP read from 127.0.0.1:32825 ---> |
| 00:41.53 | aep | HI! |
| 00:42.10 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 00:42.28 | jaytee | loather-work, yep, test for chanuavail when the Dial fails on the PRI and then redial out the SIP trunk |
| 00:42.52 | aep | [Nov 25 17:48:18] WARNING[16021]: chan_sip.c:6830 determine_firstline_parts: Bad request protocol SIP/2.0/UDP 192.168.3.250:5060; branch=1 |
| 00:43.03 | aep | i guess i dont understand SIP :D |
| 00:43.07 | aep | going to try a real client |
| 00:43.30 | jaytee | aep, order a copy of SIP Demystified from Amazon |
| 00:43.49 | seanbright | that assumes he wants to understand SIP :) |
| 00:43.51 | *** join/#asterisk ElCheapo (n=elcheapo@d137-186-181-17.abhsia.telus.net) |
| 00:43.52 | loather-work | jaytee: what woudl the config for that look like? http://pastebin.ca/1267111 is my dialout macro |
| 00:45.33 | aep | okay, i think that client is just crap |
| 00:45.39 | seanbright | what client? |
| 00:45.42 | aep | xlite |
| 00:45.46 | seanbright | it's not |
| 00:45.57 | seanbright | we use eyeBeam at work with asterisk and it works fine |
| 00:45.57 | aep | debug sip clearly shows it is sending SIP/2.0 401 Unauthorized |
| 00:46.03 | seanbright | eyeBeam is xlite but not free |
| 00:46.10 | aep | however xlite sayd the connection timed out |
| 00:46.19 | aep | i just misstyped the password |
| 00:46.21 | aep | retarded... |
| 00:46.28 | aep | thanks |
| 00:46.30 | seanbright | so it's working now? |
| 00:46.31 | seanbright | good. |
| 00:46.36 | aep | no |
| 00:46.47 | aep | but... i have a debug |
| 00:46.50 | aep | thats awesome |
| 00:47.16 | seanbright | i'm confused. but you don't seem to be asking for assistance so i will wander off. |
| 00:48.25 | aep | yeah well i need to check what i did wrong |
| 00:48.33 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-498931fc3f8ac7fa) |
| 00:48.44 | aep | it says unauthorized, but aprantly i DIDNT type the password wrong |
| 00:49.29 | seanbright | are you still using the sip.conf you pastebin'd? |
| 00:49.50 | seanbright | because there was no secret (password) specified there |
| 00:49.59 | aep | yep |
| 00:50.04 | aep | i did remove that from the paste |
| 00:50.07 | aep | as well as the realm |
| 00:50.11 | seanbright | gotcha |
| 00:50.35 | aep | the client uses aep@servername.domain.tld |
| 00:50.40 | *** join/#asterisk etfonhomey (n=chatzill@74-143-196-254.static.insightbb.com) |
| 00:50.42 | aep | it should use aep@doman.tld :( |
| 00:51.53 | jaytee | loather-work, try something like this in your dial macro between the first dial statement and the second. it needs a labeled priority to go to called unavail so you'd have to add the label to the priority of the Dial statement for your SIP trunk. |
| 00:51.56 | jaytee | exten => s,n,GotoIf($["${DIALSTATUS}" = "CHANUNAVAIL"]?UNAVAIL) |
| 00:52.05 | aep | now it does, yey |
| 00:52.18 | aep | now it's a SIP/2.0 401 Unauthorized |
| 00:53.46 | *** join/#asterisk velts (n=velts@van-fw.blastradius.com) |
| 00:54.18 | *** join/#asterisk ZaVoid (n=zavoid@nj-76-6-39-193.dhcp.embarqhsd.net) |
| 00:54.24 | velts | if i have an asterisk server can i get rid of my landline? |
| 00:54.31 | etfonhomey | aep, I've been away. Where are you at in getting your * up now? |
| 00:54.45 | etfonhomey | velts, if you trust your Internet connection to always be on. |
| 00:54.54 | aep | trying to dial into sip using xlite |
| 00:54.56 | etfonhomey | velts, and if you sign up with an ITSP |
| 00:54.57 | velts | hmm |
| 00:55.02 | aep | i get a 401 unauthorizes |
| 00:55.10 | etfonhomey | aep, what's your sip.conf look like? |
| 00:55.37 | velts | an ITSP like vonage? |
| 00:55.56 | aep | http://rafb.net/p/ftv3yN48.html |
| 00:56.06 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 00:56.06 | etfonhomey | velts, or Vitelity. Ask jaytee about pricing! Just kidding jaytee. :) |
| 00:56.36 | velts | oh so even with asterisk i still need to pay for a land phone |
| 00:56.43 | aep | maybe its a problem that the host isnt called domain.tld but host.domain.tld? |
| 00:56.57 | loather-work | jaytee: and then the second Dial() line becomes exten => s-UNAVAIL,3,Dial(...) right? |
| 00:57.55 | etfonhomey | aep, under [aep], do "canreinvite=no" |
| 00:58.33 | aep | ok |
| 00:58.37 | etfonhomey | aep, also under [aep], remove the secret line. |
| 00:59.23 | aep | ok |
| 00:59.32 | aep | trying with no password then |
| 00:59.43 | velts | etfonhomey, if you didnt want to recieve incoming calls, just make outgoing would you still need to register with an ITSP? |
| 00:59.59 | aep | asterisk says 200 OK and xlite says "connection timed out" |
| 01:00.06 | etfonhomey | aep, Then, in xlite, in the SIP Properties, do this: Display Name: aep User name: aep Auth. user name: aep Domain: (your * server IP) |
| 01:00.14 | aep | yep did that |
| 01:00.44 | aep | there is also: |
| 01:00.45 | aep | Scheduling destruction of SIP dialog '616887F272DC1AF1B206BAE0D8C93253@asgaartech.com' in 32000 ms (Method: REGISTER) |
| 01:00.50 | etfonhomey | aep, Check "register with domain and receive incoming calls" then select "domain" under "send outbound via" |
| 01:00.54 | aep | right after it says 200 OK |
| 01:01.28 | etfonhomey | aep, that's how my xlite is setup currently and it works. Once you have that, if it still doesn't work, let's do some work on your sip.conf |
| 01:02.03 | aep | i dont find the last options you names |
| 01:03.20 | *** part/#asterisk UnixDawg (n=UnixDawg@cpe-98-28-149-231.cinci.res.rr.com) |
| 01:03.34 | etfonhomey | aep, see the sub area of the SIP Properties box that says: "Domain Proxy" ? |
| 01:04.02 | aep | there is Domain/Realm and SIP Proxy |
| 01:04.47 | etfonhomey | aep, go to Help -> About in X-Lite and tell me your version and build #. |
| 01:04.54 | etfonhomey | aep, yours may be newer than mine. |
| 01:05.26 | aep | help redirects to a website |
| 01:05.30 | aep | but its a fresh download |
| 01:05.59 | etfonhomey | aep, are you on a linux box or Mac? |
| 01:06.05 | aep | anyway do you think those options have any effect? |
| 01:06.06 | aep | linux |
| 01:06.57 | *** join/#asterisk stencil_ (i=asr33@unaffiliated/stencil) |
| 01:07.10 | etfonhomey | aep, ah, I have the Windows version. Do you have the option to check "Register with domain" ? |
| 01:07.30 | aep | Register: |
| 01:07.48 | aep | which i guess just means if it tryes registration at startup |
| 01:07.57 | Akiyuki | heh |
| 01:08.08 | *** part/#asterisk stencil_ (i=asr33@unaffiliated/stencil) |
| 01:08.12 | Akiyuki | I just hooked NES nintendo up to my 50 inch tv and put on mario brothers |
| 01:08.17 | Akiyuki | my little boy is having a blast |
| 01:09.12 | *** join/#asterisk TOrrIeri (n=torrieri@nelug/crew/torrieri) |
| 01:09.31 | etfonhomey | aep, OK. Is this a correct screenshot of what you're seeing? http://linux.softpedia.com/progScreenshots/X-Lite-Screenshot-5595.html |
| 01:10.17 | aep | yeah |
| 01:10.20 | *** join/#asterisk Schmee (n=zaphod@ppp100-124.static.internode.on.net) |
| 01:10.46 | etfonhomey | aep, OK. Here we go. Enabled: Yes |
| 01:11.15 | aep | yeah did that |
| 01:11.59 | etfonhomey | aep, Display Name: aep Username: aep Authorization User: aep Password: (blank) Domain/Realm: (* IP address) |
| 01:12.20 | etfonhomey | aep, SIP Proxy: (blank) Out Bound Proxy: (blank) |
| 01:12.38 | aep | ok i can try ther ip adress |
| 01:12.39 | etfonhomey | aep, what are you choices under the "Use Outbound Proxy" |
| 01:13.08 | aep | Default Always Never |
| 01:13.24 | *** join/#asterisk Stressor69 (n=joea@wsip-70-167-2-66.sd.sd.cox.net) |
| 01:13.36 | etfonhomey | same options for "Register" ? |
| 01:13.55 | *** join/#asterisk jeffspeff (i=Administ@c-98-240-113-191.hsd1.ky.comcast.net) |
| 01:14.00 | aep | yep |
| 01:14.54 | etfonhomey | aep, Register: Always Use Outbound Proxy: Never |
| 01:15.12 | etfonhomey | aep Send Internal IP: Always |
| 01:15.26 | etfonhomey | aep, that should do it. |
| 01:16.02 | Stressor69 | anyone familiar w/ 7970 phones? |
| 01:16.38 | aep | ok trying |
| 01:16.50 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 01:17.22 | etfonhomey | aep, any luck? |
| 01:17.42 | aep | the client doesnt save the settings |
| 01:17.51 | aep | i'm trying manual hacking |
| 01:18.37 | aep | now i'm having no debug output at all |
| 01:19.00 | etfonhomey | aep, I don't have a linux box with X handy. Got a windows box handy? |
| 01:19.08 | aep | yep |
| 01:19.21 | aep | should i download the windows version? |
| 01:19.28 | etfonhomey | aep, download X-Lite onto it and I'll be able to give you the exact settings that I know work. |
| 01:19.36 | aep | ok going to |
| 01:21.32 | etfonhomey | velts, you still here? |
| 01:23.34 | aep | the windows client says 408 request timed out |
| 01:23.42 | aep | cannot see debug output on the server |
| 01:23.48 | aep | it didnt try to connect |
| 01:25.10 | etfonhomey | aep. gonna pastebin the settings you should have, given your sip.conf |
| 01:25.54 | aep | ah now it tryed |
| 01:26.04 | aep | 404 Unauthorized again |
| 01:26.07 | aep | err 401 |
| 01:26.16 | aep | propably becouse the realm is wrong |
| 01:26.25 | etfonhomey | aep, take the realm out of your sip.conf |
| 01:26.31 | aep | ok |
| 01:27.46 | Stressor69 | I have a quick question on my asterisk install regarding the "Dial" button on my 7970 phone. When I dial a number and press the "Dial" button it says "No active call to put on hold." If I press "New Call" to get a dial tone first there is no problems. |
| 01:27.46 | etfonhomey | aep, http://www.pastebin.ca/1267133 |
| 01:28.31 | aep | yes thats my setup |
| 01:28.41 | etfonhomey | aep, OK. Then the issue is with your * box. |
| 01:28.58 | etfonhomey | aep, have you done a "sip reload" via the * CLI since we made changes to sip.conf ? |
| 01:29.08 | aep | yep |
| 01:29.19 | aep | btw it says SIP/2.0 200 OK |
| 01:29.19 | aep | <PROTECTED> |
| 01:29.26 | etfonhomey | aep, OK. pastebin your current sip.conf |
| 01:29.28 | aep | but the client wont connect anyway |
| 01:29.45 | aep | it drops the connection right away saying: |
| 01:29.55 | aep | Scheduling destruction of SIP dialog 'M2E4NTI3ZjZkYTk3NjZkMzg0ZTJmYmUwNDkyOWNjNjY.' in 32000 ms (Method: REGISTER) |
| 01:30.12 | etfonhomey | aep, pastebin your sip.conf |
| 01:30.32 | aep | http://rafb.net/p/x8RM2m97.html |
| 01:30.35 | etfonhomey | aep, do you have a firewall on your linux box? |
| 01:30.40 | aep | yes |
| 01:30.44 | aep | but port 5060 is open |
| 01:30.51 | aep | and tested to work perfectly fine |
| 01:31.32 | etfonhomey | aep, turn it off anyway. Less variables to deal with. |
| 01:31.39 | *** join/#asterisk ZaVoid (n=zavoid@nj-76-6-39-193.dhcp.embarqhsd.net) |
| 01:31.40 | aep | ok |
| 01:31.57 | aep | err wait. something is broken here |
| 01:31.59 | aep | --- Transmitting (no NAT) to 192.168.2.21:56160 -- |
| 01:32.11 | aep | thats the local ip of my box |
| 01:32.15 | aep | i AM behind a nat |
| 01:33.13 | aep | so this answer will go straight to hell instead to my local box |
| 01:33.44 | etfonhomey | aep, what's your network topology? |
| 01:34.33 | aep | usual home internet |
| 01:34.39 | aep | windows box 192.168.2.21 -> NAT -> internet |
| 01:34.50 | etfonhomey | aep, I don't see * in that topology. |
| 01:34.53 | aep | the machine running SIP has a public internet adress |
| 01:35.47 | etfonhomey | aep, so 192.168.2.21 -> NAT -> Internet -> * Box ??? |
| 01:36.20 | aep | yeah |
| 01:36.30 | etfonhomey | aep, or is it 192.168.2.21 -> NAT -> Internet -> NAT -> * Box ??? |
| 01:36.44 | aep | no |
| 01:37.04 | etfonhomey | aep, ok add nat=yes to your sip.conf under [aep] |
| 01:37.23 | etfonhomey | aep and qualify=yes |
| 01:37.24 | Akiyuki | Is gtk+-2.0 mandatory for asterisk install from source? |
| 01:37.35 | etfonhomey | Isn't that a graphics library? |
| 01:37.50 | *** join/#asterisk zackz (n=zack@rrcs-24-123-106-250.central.biz.rr.com) |
| 01:37.57 | Akiyuki | Yes, but the config is complaining it can't find it |
| 01:37.58 | mosty | yes it's a gui library, and no it's not mandatory |
| 01:38.06 | aep | etfonhomey: connected!! |
| 01:38.19 | etfonhomey | Now, do you have audio? |
| 01:38.21 | etfonhomey | :) |
| 01:38.45 | aep | dunno if i press the green button it does a sound |
| 01:38.51 | aep | but no clue what else to try |
| 01:38.54 | zackz | so...my 1.4.22 doesn't have chan_zap, why would this be? |
| 01:39.10 | etfonhomey | zackz, dahdi only I think. |
| 01:39.15 | etfonhomey | I might be wrong. |
| 01:39.25 | zackz | well, it didnt even build the DAHDI module either |
| 01:39.49 | Stressor69 | yes, dahdi |
| 01:40.00 | etfonhomey | aep, now create a dialplan so you can call something that will pay music back to you. |
| 01:40.13 | mosty | zackz, you need to build and install dahdi separately before building asterisk |
| 01:40.19 | zackz | from the doc: This version of Asterisk can be built using either Zaptel or DAHDI, |
| 01:40.23 | Stressor69 | zackz: caught me off gaurd as well. setup is basically the same as zap |
| 01:40.28 | zackz | so, basically their documentation is inaccurate |
| 01:40.46 | aep | etfonhomey: yeah now i can go further in that book, awesome |
| 01:40.50 | aep | etfonhomey: thanks a lot! |
| 01:41.00 | mosty | zackz, zaptel or dahdi, either will do. the documentation is accurate |
| 01:41.15 | zackz | i built zaptel |
| 01:41.17 | aep | i get "the person you are calling is unavailable" |
| 01:41.18 | aep | hehe |
| 01:41.24 | zackz | but my asterisk does not even have a chan_zap module |
| 01:41.28 | aep | * has built in sounds for that, awesome |
| 01:41.30 | etfonhomey | aep, that's from X-Lite. |
| 01:41.31 | mosty | zackz, is zaptel installed? |
| 01:41.33 | zackz | there arent even any .h or .c files |
| 01:41.34 | aep | oh :( |
| 01:41.36 | zackz | yes i installed it first |
| 01:41.43 | mosty | zackz, modules loaded? |
| 01:41.44 | zackz | cards are detected fine |
| 01:41.51 | zackz | there is no chan_zap module |
| 01:41.52 | zackz | at all |
| 01:42.02 | mosty | zackz, and did you select chan_zap / chan_dahdi in the asterisk config? |
| 01:42.07 | zackz | there are no files on the system besides my old backups form 1.2 |
| 01:42.08 | etfonhomey | aep, pastebin your extensions.conf file and I'll tell you a number to dial in order to test your audio. |
| 01:42.55 | zackz | mosty: there is no option for chan_zap |
| 01:43.03 | mosty | zackz, it's called chan_dahdi in 1.4.22 |
| 01:43.27 | aep | etfonhomey: http://rafb.net/p/pbEnG466.html |
| 01:43.28 | zackz | but will it work with zaptel 1.4.12.1? |
| 01:43.31 | aep | its huge, i didnt modify it |
| 01:44.40 | zackz | i dont even see dahdi on the asterisk.org downloads page |
| 01:44.58 | zackz | oh i see on the sidebar |
| 01:45.00 | zackz | nm |
| 01:45.21 | Akiyuki | hmm |
| 01:45.27 | Akiyuki | what happened to "sip set debug" in 1.6? |
| 01:45.31 | Akiyuki | I have only used 1.4 |
| 01:46.11 | etfonhomey | aep, in sip.conf change the context for [aep] to "demo" and do a sip reload |
| 01:47.25 | aep | yup |
| 01:47.59 | etfonhomey | aep, now dial 600 in X-Lite and hit Enter. |
| 01:48.11 | *** join/#asterisk Spirits-Sight (n=christop@c-24-218-128-159.hsd1.ma.comcast.net) |
| 01:48.49 | aep | SIP/2.0 407 Proxy Authentication Required |
| 01:48.58 | aep | oO |
| 01:49.24 | Akiyuki | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 01:49.47 | etfonhomey | Akiyuji, that means * is most likely NOT running. |
| 01:49.54 | etfonhomey | Akiyuki* |
| 01:50.37 | etfonhomey | aep, that shows up in the SIP debug on your * box? |
| 01:50.59 | Akiyuki | <PROTECTED> |
| 01:51.00 | Akiyuki | Asterisk died with code 1. |
| 01:51.00 | Akiyuki | cat: /var/run/asterisk.pid: No such file or directory |
| 01:51.00 | Akiyuki | Automatically restarting Asterisk. |
| 01:51.21 | zackz | do asterisk -cvvvvvvvvvvv |
| 01:52.15 | Akiyuki | Unable to bind to 0 .0.0.0 port 4520: Address already in use |
| 01:52.41 | etfonhomey | aep, hello? |
| 01:52.52 | aep | uh sorry. |
| 01:52.56 | aep | yes, it does |
| 01:53.11 | etfonhomey | aep, in sip.conf, change type=peer |
| 01:53.14 | aep | and no connection |
| 01:53.39 | aep | ok |
| 01:53.41 | *** join/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.ut.comcast.net) |
| 01:54.10 | Akiyuki | Where is the 4520 coming from? its 5060 in sip.conf |
| 01:54.12 | aep | lots of output and |
| 01:54.13 | aep | [Nov 25 18:59:43] WARNING[17777]: chan_sip.c:1958 retrans_pkt: Maximum retries exceeded on transmission MDhlNTRjY2EyMTBlYWI3YzJjZGY2ZGE3MTJlZGU1ODE. for seqno 1 (Critical Response) -- See doc/sip-retransmit.txt. |
| 01:54.17 | aep | this might be relevant |
| 01:54.42 | aep | it tryes to send a 200 OK |
| 01:54.44 | [T]ank | anyone here able to recommend a sip provider supporting t.38 here in the US? would prefer a pay as you go, but whatever I can find. was getting set up with one carrier that advertised that they supported it, they bailed out on me last minute and said they were no longer supporting in. |
| 01:54.45 | [T]ank | it |
| 01:55.43 | *** join/#asterisk brut- (n=brut@h66-173-4-254.mntimn.dedicated.static.tds.net) |
| 01:55.51 | Carlos_PHX | ipcomms.net |
| 01:56.02 | etfonhomey | aep, still on the * CLI? Pastebin the output of "sip show peers" |
| 01:56.13 | Akiyuki | stupid asterisk |
| 01:56.17 | Akiyuki | shoulda installed from rpm |
| 01:56.21 | Carlos_PHX | Asterisk sucks |
| 01:56.33 | aep | etfonhomey: it's just "aep/aep 84.56.121.XXX D N 48982 OK (42 ms) " |
| 01:56.41 | Carlos_PHX | RPMs are for little girls and "men" who pee sitting down. |
| 01:56.45 | iCEBrkr | um yeah fax over voip aint happening |
| 01:56.49 | Akiyuki | You must not be married :) |
| 01:56.53 | Akiyuki | pees sitting down |
| 01:57.01 | Carlos_PHX | iCEBrkr: I'll tell my servers doing fax over VoIP to stop it then. |
| 01:57.11 | iCEBrkr | uh huh. |
| 01:57.21 | iCEBrkr | you gotta have a gateway |
| 01:57.32 | *** join/#asterisk brunner (i=4223ac7b@gateway/web/ajax/mibbit.com/x-f90edf7589f2f57d) |
| 01:57.33 | Carlos_PHX | rm -rf / There, no move fax over VoIP. |
| 01:57.49 | *** join/#asterisk egypcio (n=egypcio@unaffiliated/egypcio) |
| 01:58.02 | Carlos_PHX | iCEBrkr: Yes, you do. |
| 01:58.11 | iCEBrkr | haha ok then |
| 01:58.31 | brunner | O |
| 01:58.32 | Carlos_PHX | Otherwise what, you'd fax from nothing to nothing? |
| 01:58.41 | iCEBrkr | theres no way that shit will work say oveer voicepulse or bandwidth, etc etc |
| 01:59.04 | Carlos_PHX | Sure. Right. I'll turn it off then. |
| 01:59.09 | Carlos_PHX | Since it's not actually working. |
| 01:59.16 | Carlos_PHX | Would you like to try one of my T.38 numbers? |
| 01:59.33 | iCEBrkr | which probably termminate into some cisco device |
| 01:59.40 | Carlos_PHX | Asterisk on my side. |
| 01:59.50 | Akiyuki | etfonhomey: Can you take a look at http://pastebin.ca/1267155 |
| 02:00.02 | Carlos_PHX | Dunno what the providers (Bandwidth, etc) use on their side. |
| 02:00.12 | *** part/#asterisk zackz (n=zack@rrcs-24-123-106-250.central.biz.rr.com) |
| 02:00.37 | etfonhomey | aep, do you still have the firewall on? |
| 02:00.50 | etfonhomey | Akiyuki, where is this text from? |
| 02:01.02 | Akiyuki | etfonhomey: Command line when using /etc/init.d/asterisk start |
| 02:01.05 | aep | etfonhomey: no |
| 02:01.07 | pcrane | anyone know anything about this: |
| 02:01.07 | brunner | I'm looking to build a web interface using MySQL and PHP that allows my call screener to transfer people in a queue to a MeetMe room, and then allow my radio hosts to unmute people in the MeetMe room. I see that I can use a couple different applications to have asterisk INSERT rows into my DB, when a call comes in, goes into the queue, gets transferred into the MeetMe room, but what's the best interface to use to allow |
| 02:01.07 | pcrane | Unable to request channel Local |
| 02:01.32 | etfonhomey | Akiyuki, what if you just do "asterisk" from the command line? |
| 02:01.35 | brunner | for example, I could probably do it using the manager interface, but is that the preferred way? |
| 02:01.47 | Akiyuki | etfonhomey: Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory |
| 02:02.06 | *** join/#asterisk pyite (n=pyite@c-76-21-105-195.hsd1.ca.comcast.net) |
| 02:02.09 | iCEBrkr | brunner, good luck. thats quite a bit of work |
| 02:02.25 | iCEBrkr | managing the queus outside of the queue app is pita |
| 02:02.54 | iCEBrkr | im mobile on my g1... |
| 02:03.00 | etfonhomey | aep, It looks like SIP is working, but RTP is having problems. |
| 02:03.02 | iCEBrkr | atrsisk Icvvvvvv |
| 02:03.12 | iCEBrkr | oop |
| 02:03.23 | iCEBrkr | atrisk -cvvvvvv |
| 02:03.33 | aep | etfonhomey: is that a different port? |
| 02:03.41 | [T]ank | Carlos_PHX: do you use these guys? |
| 02:03.50 | etfonhomey | aep, yes, why do you still have your firewall on? |
| 02:04.09 | aep | i dont, but i would try ther port anyway |
| 02:04.14 | aep | my isp blocks various ports |
| 02:04.32 | Carlos_PHX | [T]ank: You mean ipcomms? Just a little, we looked at them for a wholesale subcarrier but they're a little small for us. |
| 02:04.37 | Akiyuki | etfonhomey: Should i `touch` that file? |
| 02:04.41 | Carlos_PHX | I do have a test account and it does work fine. |
| 02:04.57 | brunner | iCEBrkr: couldn't I just have a PHP script use the Asterisk Manager API to redirect whatever user in the queue to an extension that puts them into the MeetMe room? |
| 02:04.59 | etfonhomey | aep, /etc/asterisk/rtp.conf the ports are rtpstart-rtpend |
| 02:05.06 | Carlos_PHX | Our production T.38 is with Vitelity, but they don't support that on the consumer level. |
| 02:05.30 | iCEBrkr | brunner, whats the end goal? |
| 02:05.47 | Carlos_PHX | I was pleased with the support from ipcomms, and the prices are good for end-user accounts. |
| 02:05.48 | etfonhomey | Akiyuki, * creates that file when it is running. |
| 02:05.55 | iCEBrkr | brunner, reminder, im mobile on a tiny keyboard, responses will be abbvr. |
| 02:06.40 | etfonhomey | Akiyuki, what user are you trying to run it as? Sounds like a permissions issue. |
| 02:06.44 | Akiyuki | root |
| 02:06.52 | *** join/#asterisk subdolus (n=subdolus@subby.afraid.org) |
| 02:07.18 | aep | [Nov 25 19:13:00] WARNING[17777]: chan_sip.c:1980 retrans_pkt: Hanging up call YWViMGQzMzRhYTc1NmUwYzYzYzc4NmUwYWUxOTAxMTk. - no reply to our critical packet (see doc/sip-retransmit.txt). |
| 02:07.23 | iCEBrkr | akiyuki, add -cvvvvvv cmd line opts |
| 02:07.27 | aep | etfonhomey: the ports are fine however |
| 02:07.50 | etfonhomey | Akiyuki, remove any trace of * from your system and start over from scratch. |
| 02:07.54 | [TK]D-Fender | aep: Thats a firewall / routing / NAT error |
| 02:08.04 | iCEBrkr | yo tk |
| 02:08.06 | brunner | iCEBrkr: that's fine. This is what I want to do: 1) Incoming calls go into a queue 2) A call screener (an agent) screens each call, and then transfers the caller to an extension that puts them into a MeetMe room muted. 3) A web interface would show what calls are in the MeetMe room, and the web user clicks on someone to unmute them |
| 02:08.13 | aep | [TK]D-Fender: firewall is of |
| 02:08.21 | aep | wait, except on that faildows machine |
| 02:08.52 | pcrane | anyone know why I can't create a local channel? |
| 02:08.53 | [T]ank | Carlos_PHX: test account, meaning one of their free dids? |
| 02:08.59 | [TK]D-Fender | aep: What networking sits between * and your client? |
| 02:09.03 | iCEBrkr | brunner, as long as you use the built in queue and meet me. you can use funcodbc to write things to the db |
| 02:09.31 | brunner | iCEBrkr: yes, but that's the best way for my PHP script to transfer callers from the queue to the MeetMe room? |
| 02:09.39 | aep | [TK]D-Fender: the client is behind a NAT. some sort of black box home router which btw has Voip on its own. then the server is a public machine in the internet |
| 02:09.42 | iCEBrkr | brunner, i.m thinking there are ami cmds to mute/unmute |
| 02:09.59 | [TK]D-Fender | aep: NAT requires several specific things configured to work : |
| 02:10.01 | [TK]D-Fender | ~sipnat |
| 02:10.02 | 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 |
| 02:10.03 | [TK]D-Fender | ^^^^^^^^^^ |
| 02:10.27 | aep | allright, thank you |
| 02:10.50 | Akiyuki | i got it working |
| 02:10.51 | Akiyuki | or atleast up |
| 02:11.06 | aep | [TK]D-Fender: did what is suggests, etfonhomey already told me |
| 02:11.13 | iCEBrkr | brunner, id just make socket connections to ami and issue the proper cmds |
| 02:11.15 | Carlos_PHX | beats head on wall repeatedly. The danger of provisioning customer equipment at home is that you may accidentally provision your own gear instead, then wonder why it doesn't work. |
| 02:11.17 | aep | [TK]D-Fender: SIP apears to connect but voice isnt transported |
| 02:11.26 | etfonhomey | aep, I'm confinced it's a firewall issue. |
| 02:11.30 | brunner | Does anyone know how to unmute callers in a MeetMe room using AMI or any other interface? |
| 02:11.39 | [TK]D-Fender | aep: that is the problem. |
| 02:11.45 | [TK]D-Fender | aep RTP is the hard part |
| 02:11.51 | aep | oh heh |
| 02:11.52 | [TK]D-Fender | aep: Now go read the guide |
| 02:12.02 | iCEBrkr | brunner, the commands are documented on voip-info |
| 02:12.25 | brunner | iCEBrkr: I'm there now, but I can't find a command that unmutes callers |
| 02:12.27 | [TK]D-Fender | brunner: "help meetme" |
| 02:13.03 | aep | ok turned all firewals in the middle of |
| 02:13.15 | aep | and yes i added those two options to sip.conf |
| 02:13.39 | [TK]D-Fender | aep: TWO? more than that.. keep reading it till your eyes bleed |
| 02:13.45 | Akiyuki | huzzah |
| 02:13.52 | brunner | [TK]D-Fender: I have to do this through the Asterisk Manager API |
| 02:13.53 | Akiyuki | I installed and got asterisk started from source! |
| 02:13.54 | iCEBrkr | haha |
| 02:14.24 | iCEBrkr | brunner, look into 'command' |
| 02:14.25 | [TK]D-Fender | brunner: And my answer is EVERY bit as applicable to that. |
| 02:14.29 | [TK]D-Fender | ^^^ |
| 02:14.33 | iCEBrkr | there might be a way from the cli |
| 02:14.42 | [TK]D-Fender | iCEBrkr: There is... |
| 02:14.59 | aep | [TK]D-Fender: maybe a misunderstanding, but i have #9 from that list |
| 02:15.03 | iCEBrkr | i know theres a way hehe |
| 02:15.09 | aep | it mentiones nat=yes and qualify=yes |
| 02:15.09 | [TK]D-Fender | aep: FIRST link... |
| 02:15.14 | brunner | [TK]D-Fender: Oh, I see. Thank you. |
| 02:15.27 | aep | [TK]D-Fender: that doesnt reflect my setup. but i'll read it anyway |
| 02:15.30 | brunner | Are there any other interfaces that could be used for this kind of interaction? |
| 02:15.39 | [TK]D-Fender | aep: Which is? |
| 02:15.55 | aep | [TK]D-Fender: i have only one nat. that one has two |
| 02:15.57 | iCEBrkr | brunner, such as? |
| 02:15.58 | [TK]D-Fender | bruCLI or AMI. Thats it. |
| 02:16.31 | brunner | iCEBrkr: any other API's |
| 02:16.39 | brunner | [TK]D-Fender: okay, thanks! |
| 02:16.58 | iCEBrkr | brunner, just socket_open() |
| 02:17.07 | iCEBrkr | or whatever the php cmd is |
| 02:17.14 | iCEBrkr | attach to the ami port |
| 02:17.19 | aep | [TK]D-Fender: sorry, cant find something new there. any hints for me? |
| 02:17.21 | brunner | iCEBrkr, [TK]D-Fender: thanks for the information. Sorry for my ignorance. I just started playing with asterisk for the first time yesterday. |
| 02:17.22 | iCEBrkr | loging, issue the cmds |
| 02:17.43 | [TK]D-Fender | bruNot bad, barely in and already up to your neck in it... |
| 02:17.52 | iCEBrkr | haha |
| 02:17.58 | [TK]D-Fender | brunner: Keep treading, Noah! |
| 02:18.00 | iCEBrkr | tk, isnt that always the case?? |
| 02:18.15 | brunner | this can't be that hard. |
| 02:18.37 | [TK]D-Fender | brunner: Heard the same thing in Sex Ed. |
| 02:18.43 | iCEBrkr | lol |
| 02:19.53 | iCEBrkr | oh cool i found tab |
| 02:20.06 | iCEBrkr | so i can do nick coompletion |
| 02:20.27 | Carlos_PHX | Yes, but have you found Jesus? |
| 02:20.43 | iCEBrkr | 404 |
| 02:21.23 | brunner | Is there any reason I shouldn't use the MYSQL command instead of an external PHP script to keep track of calls as they come in? |
| 02:21.47 | jaytee | I found Jesus. He was hiding in Housewares in the WalMart in Brownsburg,IN. Looked like he hadn't slept in weeks. |
| 02:21.58 | iCEBrkr | you really wanna keep as much as possiible in the dial plan |
| 02:22.15 | brunner | So I suppose that's a no. |
| 02:22.58 | [TK]D-Fender | Carlos_PHX: Well, I've witnessed many things... but never Jehovah. |
| 02:23.17 | iCEBrkr | brunner, dialplam is native processing, no shelling out and less cpu needed |
| 02:23.21 | *** join/#asterisk Fairman (n=Fairman@c-76-105-10-247.hsd1.ca.comcast.net) |
| 02:24.02 | Fairman | hey everybody |
| 02:25.02 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 02:25.23 | aep | could it be a problem that this dsl home router box does sip on its own? |
| 02:25.42 | [TK]D-Fender | aep: yes |
| 02:25.52 | aep | awesome |
| 02:25.58 | [TK]D-Fender | aep: THAT COULD BE A RATHER SERIOUS ISSUE |
| 02:25.58 | aep | any idea to fix that? |
| 02:26.06 | [TK]D-Fender | aep: R&R <- |
| 02:26.40 | aep | hm? |
| 02:27.24 | Carlos_PHX | ~R&R |
| 02:27.24 | jbot | somebody said 3r was Rotate and Resize (and Reflect) extensions for XFree86, that allow the xserver to inform the x-clients of desktop size (etc) changes, so they can update themselves. http://www.xfree86.org/~keithp/talks/randr/ |
| 02:27.28 | [TK]D-Fender | aep: Rip & Replace |
| 02:27.48 | Carlos_PHX | jbot was way off. |
| 02:28.08 | aep | [TK]D-Fender: there are no replacements. the router boxes are vendor locked |
| 02:28.19 | aep | you may not connect to the internet with ppp anymore |
| 02:28.31 | [TK]D-Fender | aep: get a new vendor |
| 02:28.33 | aep | they roll they propriatary stuff instead |
| 02:28.35 | [TK]D-Fender | NEXT!!@@@!@! (c) BKW |
| 02:28.39 | iCEBrkr | ha |
| 02:28.42 | aep | for that i'd need a new country first ;) |
| 02:28.48 | aep | becouse there are only 3 there |
| 02:28.54 | aep | so anyway, problem not fixable? |
| 02:29.09 | [TK]D-Fender | aep: See, who knew that * could so completely change your life! |
| 02:29.17 | iCEBrkr | haha |
| 02:29.18 | aep | heh |
| 02:29.28 | iCEBrkr | [TK]D-Fender: yeah im bald now |
| 02:29.36 | [TK]D-Fender | aep: But on that note, try to put their router behind one of your own |
| 02:29.52 | aep | yeah thats what i was thinking. ie tunnel |
| 02:30.16 | Fairman | * may soon need a 12 step program... *A |
| 02:30.20 | aep | i need to tunnel from everywhere nowadays anyway |
| 02:30.41 | aep | becouse ISPs are so mart |
| 02:30.43 | aep | *smart |
| 02:31.06 | iCEBrkr | i almost made the switch to freeswitch, but then all my callmanager dtuff would have to be rewritten |
| 02:31.12 | Fairman | is probably not as funny as he thinks he is... |
| 02:31.15 | aep | anyone knows a cheap voip service? i hate wasting on of my server IPs for that all the time |
| 02:32.26 | [TK]D-Fender | aep: That router might FUBAR everyhting jsut the same... but then again, please describe the networking between * and your client like I asked. |
| 02:33.14 | *** join/#asterisk VoipForces (n=courchea@67.55.25.219) |
| 02:33.18 | aep | [client] -> [some weird locked router that does voip on its own] -> internet -> asterisk |
| 02:33.25 | VoipForces | Hi all, anyone has a recommended version of spandsp to use with asterisk 1.4? |
| 02:33.33 | aep | one nat between me and the internez |
| 02:34.06 | aep | was that a sufficant description? |
| 02:34.14 | iCEBrkr | spandex went out in the 90s |
| 02:34.27 | [TK]D-Fender | aep: So * is on a public IP? |
| 02:34.31 | aep | it's 192.168.2.20 -> 192.168.2.1 -> 98.something -> 76.something |
| 02:34.36 | aep | [TK]D-Fender: yes |
| 02:34.37 | VoipForces | :-P iCEBrkr |
| 02:34.40 | iCEBrkr | hehe |
| 02:35.02 | iCEBrkr | spandsp is for faxing? i forget |
| 02:35.10 | [TK]D-Fender | aep: Ok, then pastebin your sip.conf masking only passwords and verify that your server's firewall is not in the way. |
| 02:35.13 | [TK]D-Fender | ~pb |
| 02:35.14 | 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, or , http://bin.cakephp.org/ |
| 02:35.16 | [TK]D-Fender | ^^^^^ |
| 02:35.22 | [TK]D-Fender | iCEBrkr: yes |
| 02:35.36 | iCEBrkr | i gave up on faxing |
| 02:35.51 | aep | [TK]D-Fender: http://rafb.net/p/paNej055.html |
| 02:35.57 | aep | [TK]D-Fender: all firewalls are of |
| 02:36.01 | aep | including the clients |
| 02:36.24 | VoipForces | iCEBrkr: Yes. Works good. But right now it's my first real attemps ar outbound broadcasting of fax. |
| 02:36.56 | VoipForces | I'm writing an outbound fax broadcaster. Right now it works sending faxes on 23 channels (PRI). But asterisk core dumps once in a while. I'm suspecting spandsp |
| 02:37.03 | [TK]D-Fender | aep: So what happens so far? |
| 02:37.28 | aep | [TK]D-Fender: SIP connects fine , then in try to dial 600 and i get several retryes of sendin 200 OK |
| 02:37.40 | aep | then the server decices to stop trying and drops the connection |
| 02:37.48 | [TK]D-Fender | aep: ok, pastebin a call attempt with "sip set debug on" at CLI |
| 02:37.49 | aep | it claims the client never answered |
| 02:37.53 | aep | ok |
| 02:38.05 | [TK]D-Fender | aep: And what client are you using? |
| 02:38.08 | aep | xlite |
| 02:38.11 | iCEBrkr | sure is spandsp? |
| 02:38.20 | [TK]D-Fender | aep: Ok, You're doing pretty good so far.... |
| 02:38.29 | iCEBrkr | asterisk wll expode on reloads under high traffic |
| 02:38.46 | *** part/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.ut.comcast.net) |
| 02:39.35 | VoipForces | iCEBrkr: I have a voice broadcaster dialing on 200-250 sip channels no problem |
| 02:39.50 | aep | [TK]D-Fender: http://rafb.net/p/SqzuzZ84.html |
| 02:39.59 | iCEBrkr | VoipForces: you generating call files, |
| 02:40.00 | iCEBrkr | , |
| 02:40.02 | iCEBrkr | ? |
| 02:40.27 | VoipForces | Yup |
| 02:41.08 | iCEBrkr | i wonder about it grabbing the call files |
| 02:41.52 | iCEBrkr | i wrote a system to dial and deliver surveys. i found it blowing up from time to time under the right conditions |
| 02:42.34 | VoipForces | iCEBrkr: Did not have that problem till this fax thing. |
| 02:42.39 | [TK]D-Fender | aep: Either your provider is screwing with the inbound SIP, or that router is. |
| 02:42.45 | iCEBrkr | interesting |
| 02:42.53 | VoipForces | Right now I'm trying the different version of spandsp |
| 02:42.57 | aep | [TK]D-Fender: uh, what does happen? |
| 02:43.01 | iCEBrkr | i was using ast 1.2 |
| 02:43.02 | [TK]D-Fender | aep: You curiously seem to have done everything right the first time... |
| 02:43.13 | iCEBrkr | so i figured its just old |
| 02:43.23 | VoipForces | Have someone dialing on 4 PRI with asterisk 1.2 and call files |
| 02:43.36 | aep | so its that router :( |
| 02:43.40 | VoipForces | I tested it with up to 6 PRI |
| 02:43.49 | aep | [TK]D-Fender: could i try using different ports for rdp? |
| 02:43.50 | VoipForces | With the right hardware it works. |
| 02:43.50 | iCEBrkr | VoipForces: hrrm then maybe it is spandsp |
| 02:44.08 | iCEBrkr | its been over 4yrs since ive looked at spandsp |
| 02:44.22 | [TK]D-Fender | aep: could always try... but thats client-side, and really isn't anything you should ever have to mess with.... |
| 02:44.51 | VoipForces | Does 1.6 still require spandsp? |
| 02:44.58 | Carlos_PHX | Woohoo, got a fax over an ATA. Yeah, I know, it should be easy, but first time trying. |
| 02:45.00 | iCEBrkr | im all about client apps now days |
| 02:45.05 | Carlos_PHX | VoipForces: Yes. |
| 02:45.13 | Carlos_PHX | VoipForces: You trying to do T.38? |
| 02:45.29 | VoipForces | no straight faxing outbound via a PRI |
| 02:45.36 | iCEBrkr | Carlos_PHX: yeah he is, asterisk is seg faulting tho |
| 02:45.49 | Carlos_PHX | Which version of SpanDSP? |
| 02:46.02 | VoipForces | Fax over a PRI or analog if T.30 is I'm not mistaking |
| 02:46.13 | Carlos_PHX | Yes |
| 02:46.17 | iCEBrkr | ok time to collect the bar tab and head home.. the shield is on |
| 02:46.23 | VoipForces | Carlos: 0.0.4pre16 |
| 02:46.29 | Carlos_PHX | You still need SpanDSP. |
| 02:46.39 | aep | ok i'll go sleep now |
| 02:46.41 | Carlos_PHX | Hmm, with Asterisk 1.6 release version? |
| 02:46.42 | aep | thank [TK]D-Fender |
| 02:46.45 | VoipForces | Carlos: Right now I'm under asterisk 1.4 |
| 02:46.51 | aep | going to try a tunnel tomorow |
| 02:46.53 | VoipForces | About to try 0.0.4pre18 |
| 02:47.00 | Carlos_PHX | I might go lower. |
| 02:47.09 | Carlos_PHX | Hold on, let me see what I have in our production 1.4 server. |
| 02:47.29 | [TK]D-Fender | aep: What is your target audeince going to connect from? |
| 02:47.43 | VoipForces | Carlos_PHX: Thanks. |
| 02:48.00 | aep | [TK]D-Fender: everone from some kind of home internet /DSL |
| 02:48.06 | Carlos_PHX | spandsp-0.0.5pre4.tgz |
| 02:48.14 | aep | so everyone behind a nat |
| 02:48.15 | Carlos_PHX | Yes, it's old, but it is working. |
| 02:48.22 | [TK]D-Fender | aep: So basically you're the only one a little screwed while testing? |
| 02:48.23 | Akiyuki | I need to install asterisk sounds now |
| 02:48.27 | VoipForces | Carlos_PHX: Tried it and it bombed |
| 02:48.40 | Carlos_PHX | VoipForces: There's another potential issue...let me look at notes. |
| 02:48.43 | aep | [TK]D-Fender: no. my setup is standard and enforced by our ISPs in the entire country |
| 02:48.52 | [TK]D-Fender | aep: and they aren't so like to be in your unfortunate situation? |
| 02:48.54 | aep | they shutdown ppp |
| 02:49.03 | VoipForces | Carlos_PHX: 0.0.6 series won't even compile app_txfax |
| 02:49.08 | [TK]D-Fender | aep: ***OUCH*** |
| 02:49.09 | aep | [TK]D-Fender: they're in the same situation, why? |
| 02:49.16 | Carlos_PHX | VoipForces: Is it possible to try out 1.6? I have an install script that works, I've used it many times. |
| 02:49.30 | Carlos_PHX | The fax in 1.6 is WAYYYY better |
| 02:49.41 | [TK]D-Fender | aep: your clients shouldn't have to fight so hard. Ok, here's plan B : they are going to connect via softphones, right? |
| 02:49.48 | VoipForces | Carlos_PHX: Right now I'm keeping 1.6 as a last resort. |
| 02:50.00 | Carlos_PHX | Is there a concrete reason, or fear of the new? |
| 02:50.18 | aep | [TK]D-Fender: whatever they have (its for internal co,munications only btw,not a service, so hacks are fine) |
| 02:50.18 | Carlos_PHX | Because I've got 1.6RC6 in production, solid. Didn't bother upgrading to release. |
| 02:50.29 | aep | [TK]D-Fender: i havea nokia smartphone that does voip. i'd like to use it |
| 02:50.49 | [TK]D-Fender | aep: MORE than excellent. Go download Zoiper, and use IAX2 instead of SIP for your protocol. |
| 02:50.51 | [TK]D-Fender | ~zoiper |
| 02:50.52 | jbot | [~zoiper] Zoiper (Formerly known as Idefisk) is a free SIP and IAX soft-phone for Windows, MacOSX, and Linux that can be found at http://www.zoiper.com |
| 02:50.54 | VoipForces | Carlos_PHX: I need something dead stable. Plus I only deploy released version. |
| 02:50.55 | Akiyuki | Ok, I installed asterisk-sounds, but the demo files are missing. Is that in addons? |
| 02:51.07 | VoipForces | Carlos_PHX: Plus I remember the first versions of 1.4.xx |
| 02:51.18 | Carlos_PHX | VoipForces: I think 1.6 is stable. Yes, things have changed a lot since then. |
| 02:51.21 | aep | oh? going to do so right away |
| 02:51.23 | Carlos_PHX | How Digium manages coding. |
| 02:51.31 | [TK]D-Fender | aep: Odds are your provider is just ignorant enough of it not to get in IAX's way and its FAR friendler with NAT |
| 02:51.39 | Carlos_PHX | I can tell you that fax on 1.6 is 100 times improved over 1.4. |
| 02:52.07 | Carlos_PHX | In the next couple weeks we will be moving all our fax servers to 1.6. |
| 02:52.14 | [TK]D-Fender | aep: SIP+RTP takes more ports and is commonly filtered, etc. IAX2 flies under most radars with 1 port needed and media and signalling on the same one |
| 02:52.18 | Carlos_PHX | On top of that, we run them in VMware... |
| 02:52.24 | [TK]D-Fender | aep: You're a prime candidate |
| 02:52.25 | VoipForces | Carlos_PHX: Hmmm. ok, I'll try a few spandsp versions first than see about 1.6. Do you know if freePBX is 1.6 frendly? |
| 02:52.48 | [TK]D-Fender | VoipForces: 2.5 is |
| 02:53.02 | VoipForces | Carlos_PHX: Production in VMWare??? That I would never do. |
| 02:53.20 | VoipForces | [TK]D-Fender: Thanks. |
| 02:53.36 | aep | [TK]D-Fender: sounds like what i need |
| 02:53.37 | Carlos_PHX | VoipForces: I know, VMware "can't" work, yet here we are. |
| 02:53.57 | [TK]D-Fender | aep: give it a try. client is free and setup is not harder than SIP |
| 02:54.07 | VoipForces | Carlos_PHX: I use it for developement/test purpose, that's it. |
| 02:54.32 | Carlos_PHX | VoipForces: See my notes, particularly the configure line for spandsp here: http://televolve.pastebin.com/m5e0874fd |
| 02:54.43 | Carlos_PHX | VMware runs most of our infrastructure, and we want to do more. |
| 02:54.57 | Carlos_PHX | VMware VI3 Enterprise of course, not the freebie. |
| 02:55.13 | Carlos_PHX | Asterisk does NOT scale at all on the freebie. |
| 02:55.46 | Akiyuki | [Nov 25 21:52:54] WARNING[21282]: file.c:891 ast_streamfile: Unable to open hawaii (format 0x4 (ulaw)): No such file or directory but hawaii.gsm exists in /var/lib/asterisk/sounds |
| 02:56.05 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 02:56.19 | aep | [TK]D-Fender: hm my mobile doesnt support IAX aparantly. |
| 02:56.37 | [TK]D-Fender | aep: that was pretty mush assured from the start.. |
| 02:56.41 | Akiyuki | Ah, that worked. I had to do the full path to it. Maybe need tochange a global var |
| 02:56.55 | [TK]D-Fender | aep: IAx2 isn't exactly popular |
| 02:57.00 | aep | oh |
| 02:57.20 | [TK]D-Fender | aep: 3rd option..... so... how's Skype work in your parts? :) |
| 02:57.29 | aep | pretty good. |
| 02:57.41 | VoipForces | Carlos_PHX: LOL banged my head on that /usr also LOL |
| 02:57.55 | aep | skype is considered an end user application, so they dont break it |
| 02:57.57 | [TK]D-Fender | aep: Ok, the only real options for connecting * to skype are per-channep, but for your needs would probably do OK. |
| 02:58.18 | [TK]D-Fender | aep: Digium is preparing to relase an OFFICIAL channel driver (checp licensed per channel). |
| 02:58.21 | Carlos_PHX | VoipForces: That caused most of my problems. |
| 02:58.24 | aep | cool |
| 02:58.27 | aep | thanks so far. |
| 02:58.35 | [TK]D-Fender | aep: not sure exactly how long that may take, but there are 3rd party options |
| 02:58.40 | VoipForces | Carlos_PHX: Damed that fax dialer is working so well if not for those core dups. |
| 02:58.41 | Carlos_PHX | I don't know if anyone other than coppice could help further, but do consider trying 1.6. |
| 02:58.56 | *** join/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.ut.comcast.net) |
| 02:58.56 | aep | i'll try to get some more information on that sip problem though and maybe sue my ISP. together with all the other broken parts. |
| 02:59.10 | aep | even http is broken |
| 02:59.33 | [TK]D-Fender | aep: I'm giving the extra bit because you're the most non-newb newb I've seen in here in ages and are being FUBAR'd by forces beyond "reasonable" control |
| 02:59.50 | *** part/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.ut.comcast.net) |
| 02:59.54 | aep | heh thanks |
| 03:00.08 | [TK]D-Fender | aep: Skype is a BASTARD protocol but if its what you need to make it work, then thats what we're about. |
| 03:00.31 | aep | nice to hear that * can connect to skype :) |
| 03:00.35 | [TK]D-Fender | aep: Working first, working idealy second |
| 03:00.44 | aep | hehe |
| 03:01.07 | [TK]D-Fender | aep: Oh don't get me wrong... CURRENT Skype connectivity is quick hackish... but if its what it takes, so be it |
| 03:01.20 | [TK]D-Fender | aep: Digium's official one will be clean I'm sure |
| 03:01.32 | [TK]D-Fender | aep: Keep an eye out |
| 03:01.34 | [TK]D-Fender | ~skype |
| 03:01.35 | jbot | [~skype] Skype is a free VoIP software and service using a closed client and propritary protocol. Only commercial channel drivers exist, with most solutions being complex, complicated, and hack-ish . Digium's SkypeForAsterisk (see ~SkypeForAsterisk) is a new solution that is a cleaner non-dependent option. |
| 03:01.43 | [TK]D-Fender | ~skypeforasterisk |
| 03:01.44 | jbot | [~skypeforasterisk] is a Digium-made channel driver that allows Asterisk to connect to the Skype network using a Skype-supported connection method. Unlike all other solutions it does not require X11, kernel modules, or a Windows machine. See http://www.astricon.net/skype for beta details. |
| 03:01.58 | aep | will do |
| 03:02.03 | etfonhomey | aep, based on your config, your * box is correctly setup. |
| 03:02.28 | etfonhomey | aep, I just connected to it via my xlite and dialed 600 successfully. |
| 03:02.35 | [TK]D-Fender | etfonhomey: Nice to see you've caught up to 5 steps ago :) |
| 03:02.44 | aep | etfonhomey: to my box? cool |
| 03:02.59 | etfonhomey | [TK]D-Fender, ask aep how he got this far |
| 03:03.10 | aep | points at etfonhomey |
| 03:03.13 | [TK]D-Fender | etfonhomey: Been hand-holding since I left? |
| 03:03.20 | aep | yup |
| 03:03.36 | etfonhomey | [TK]D-Fender, I've been there before. |
| 03:03.38 | [TK]D-Fender | aep: I'm still convinced you're not a twit, don't ruin it! |
| 03:03.50 | aep | heh :) |
| 03:04.05 | etfonhomey | aep, It's your home setup that's jacked up. |
| 03:04.10 | *** join/#asterisk neurosys (n=neurosys@adsl-225-10-162.mia.bellsouth.net) |
| 03:04.14 | aep | yeah i guess. |
| 03:04.15 | aep | sad |
| 03:04.25 | *** join/#asterisk UnixDawg (n=UnixDawg@cpe-98-28-149-231.cinci.res.rr.com) |
| 03:04.50 | etfonhomey | aep, now add the secret back into your sip.conf for [aep] so other people don't look at your sip debug output and connect to your box. |
| 03:05.14 | aep | i'll shut it down before i go. no worries. but thanks for the hint |
| 03:05.40 | etfonhomey | aep, what country are you in? |
| 03:05.44 | aep | germany |
| 03:06.16 | etfonhomey | aep, You get your home phone via SIP from your ISP and by their supplied router? |
| 03:06.18 | [TK]D-Fender | aep: I'd be guessing not a major city region... |
| 03:06.50 | aep | etfonhomey: yes |
| 03:06.56 | aep | [TK]D-Fender: nope. |
| 03:07.29 | aep | well yes, here's a university, but its not like anyone is allowed to connect to them |
| 03:07.35 | etfonhomey | aep, They are probably filtering SIP not bound for there servers. Probably via their QoS config. |
| 03:07.45 | [TK]D-Fender | aep: My condolences for your challenging working environment. Do Give IAX2 a try for the PC clients that can support it, just know that you won't find so myc available for Cell clients, etc |
| 03:07.46 | etfonhomey | their* |
| 03:08.18 | etfonhomey | aep, is your last hope |
| 03:08.23 | etfonhomey | IAX2 that is. |
| 03:08.38 | aep | [TK]D-Fender: if that works i can get away with proxiing me out via another * i guess ;) |
| 03:09.04 | aep | phone->SIP-> IAX2->router->servr |
| 03:09.11 | aep | something like that |
| 03:09.24 | aep | actually thats pretty cool. asterisk is awesome |
| 03:09.32 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 03:10.19 | etfonhomey | [TK]D-Fender, didn't FWD have an IAX enabled client? |
| 03:10.46 | aep | allright. bedtime. thanks alot again for your support [TK]D-Fender and etfonhomey . i'll name you when i'm a famous rockstar. hehe, oh well. have a good day |
| 03:10.55 | riddlebox | hrmm I have to check to see I think I may have a fwd acccount |
| 03:11.06 | [TK]D-Fender | etfonhomey: at one point they had a gateway... they were always more of a service than a client |
| 03:11.08 | etfonhomey | aep, you're welcome. |
| 03:11.15 | [TK]D-Fender | aep: yup |
| 03:11.21 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 03:12.02 | etfonhomey | [TK]D-Fender, I thought they had the FWD Communicator client that would use IAX. It was probably hard coded to only connect to their servers like an AIM or Yahoo! IM client. |
| 03:12.48 | [TK]D-Fender | etfonhomey: protocols are protocols.. who cares about the client? |
| 03:14.06 | etfonhomey | You should see my AMI client, then... |
| 03:14.22 | VoipForces | Carlos_PHX: How well dahdi and wanpipe behave together? |
| 03:14.28 | *** join/#asterisk ZaVoid (n=zavoid@nj-76-6-39-193.dhcp.embarqhsd.net) |
| 03:15.03 | Carlos_PHX | VoipForces: Don't know, never tried it. |
| 03:15.18 | Carlos_PHX | Last used Sangoma in Asterisk 0.9 |
| 03:15.43 | [TK]D-Fender | Carlos_PHX: VoipForces Should be normal with latest releases |
| 03:16.34 | VoipForces | Thanks again TK |
| 03:20.27 | VoipForces | is dahdi using the same zaptel.conf configuratio file and format? |
| 03:21.05 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
| 03:21.41 | *** join/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.ut.comcast.net) |
| 03:22.14 | jasonwoot | can I setup linux as 'multihomed' on two NICs to the same network, or is that asking for trouble? |
| 03:22.37 | [T]ank | this is slightly off topic, but does anyone know how to install xlite or something similar to that (in windows) to a usb thumb drive so that it is executable and all settings saved no matter what machine it plugs into? |
| 03:22.52 | [T]ank | i can install and run it from there... but the settings look like they are stored on the local machine |
| 03:22.57 | VoipForces | jasonwoot: yes, not problem at all |
| 03:23.40 | VoipForces | [T]ank: xlite saves it's setting in the registry |
| 03:23.47 | VoipForces | [T]ank: try zoiper |
| 03:24.03 | jasonwoot | VoipForces: I wouldn't attempt it, but the scenario is: my ISP is offering SIP trunking, but only over an additional port on my fiber service |
| 03:24.15 | [T]ank | VoipForces: thats a new one... heading to google. does it work well? |
| 03:24.22 | jasonwoot | meaning I have to plug it into 2nd NIC, but its to the same ISP |
| 03:24.32 | jasonwoot | the setup is sorta baking my noodle |
| 03:25.17 | etfonhomey | [T]ank, there is a command-line swith for X-Lite that may do something. -argfile= is the switch. |
| 03:25.19 | VoipForces | jasonwoot: well, you just have to do yor routing correctly |
| 03:26.48 | VoipForces | [T]ank: zoiper been around for a while. It's SIP and IAX, Linux, Windoze and Mac. Works like a charm. |
| 03:27.33 | jasonwoot | VoipForces: I've added a static route with what I *think* is the correct syntax, but it's not pingable over that interface, |
| 03:27.34 | jasonwoot | must networking be restarted for it to route? |
| 03:29.41 | VoipForces | should not. |
| 03:30.10 | VoipForces | jasonwoot: I recommend you go to a networking specific channel. |
| 03:30.53 | UnixDawg | ok anyone here using the asterisk-now 1.5 beta |
| 03:31.01 | UnixDawg | and the reports are not working |
| 03:31.08 | UnixDawg | for the cdrs? |
| 03:32.30 | jasonwoot | VoipForces: know a good linux networking channel? |
| 03:36.06 | VoipForces | jasonwoot: Your local LUG is probably the best source. |
| 03:36.28 | VoipForces | jasonwoot: Your will not get a lot of attention here as it's mainly asterisk stuff |
| 03:37.10 | *** part/#asterisk UnixDawg (n=UnixDawg@cpe-98-28-149-231.cinci.res.rr.com) |
| 03:39.32 | *** part/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.ut.comcast.net) |
| 03:42.34 | [TK]D-Fender | jasonwoot: Multi-homed has been lots of trouble for the few I've heard of who've tried it |
| 03:43.14 | [TK]D-Fender | jasonwoot: On safer option is to run * on on IP, and SER on the other proxying the call |
| 03:48.11 | *** join/#asterisk km2 (n=x@32.178.45.73) |
| 03:49.12 | mosty | i tried multi-homed asterisk, it simply doesn't work |
| 03:49.58 | etfonhomey | [TK]D-Fender, when is SER necessary? |
| 03:51.44 | VoipForces | I get No such command 'dahdi show channels' any ideas? |
| 03:52.05 | mosty | is chan_dahdi.so loaded? |
| 03:52.06 | [TK]D-Fender | etfonhomey: just a way to have each app bind to 1 port so * doesn't get F-d up over which IP to originate a response from,etc |
| 03:54.03 | VoipForces | mosty: loking... |
| 03:54.32 | VoipForces | mosty: chan_dahdi.so does not even show... |
| 03:55.04 | [TK]D-Fender | VoipForces: did you INSTALL it? |
| 03:55.29 | [TK]D-Fender | VoipForces: And did you recompile * after? |
| 03:55.35 | VoipForces | Yup it is installed. |
| 03:55.44 | VoipForces | Recompiling * just to be sure. |
| 03:56.34 | VoipForces | Now it compiles: |
| 03:56.42 | VoipForces | <PROTECTED> |
| 03:56.43 | VoipForces | <PROTECTED> |
| 03:58.55 | VoipForces | Now that's better: wanpipe1 card 0 OK 0 0 0 ESF B8ZS YEL 0 db (CSU)/0-133 feet (DSX-1) |
| 04:00.20 | Carlos_PHX | Damn, doing T.38 with an ATA was too easy. |
| 04:03.58 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 04:05.09 | VoipForces | Carlox_PHX: Just completed the install of 1.6, but it refuses to compile app_fax. Menuconfig says that I need spandsp, which I have (same version as you). |
| 04:08.27 | VoipForces | Carlos_PHX: Just completed the install of 1.6, but it refuses to compile app_fax. Menuconfig says that I need spandsp, which I have (same version as you). |
| 04:10.33 | [TK]D-Fender | VoipForces: Trash your whole source folder and start again |
| 04:11.50 | Carlos_PHX | VoipForces: Did you do the configure line as I had noted? |
| 04:14.48 | VoipForces | Yup |
| 04:15.10 | VoipForces | [TK]D-Fender: you mean the * folder? |
| 04:15.23 | VoipForces | hmmm: checking for minimum version of SpanDSP... no |
| 04:15.43 | [TK]D-Fender | VoipForces: your source folder |
| 04:16.44 | VoipForces | [TK]D-Fender: Looks more like a minimu version thing. trying with latest spandsp |
| 04:17.30 | Carlos_PHX | Did you use the same version I did, or another? |
| 04:19.21 | VoipForces | Tried with the same spandsp version, but released 1.6.0.1 of asterisk |
| 04:19.36 | VoipForces | with spandsp-0.0.6pre2 app_fax shows up in menuconfig |
| 04:20.31 | Carlos_PHX | Which one didn't work? |
| 04:21.08 | VoipForces | spandsp-0.0.5pre4 with asterisk 1.6.0.1 |
| 04:21.30 | VoipForces | Great now app_fax does not compile |
| 04:21.40 | VoipForces | but I saw reference for that on google |
| 04:22.13 | VoipForces | 0013688: [patch] Update app_fax to work with spandsp-0.0.6 |
| 04:23.42 | VoipForces | Even with that patch it does not compile... |
| 04:25.50 | VoipForces | This is also related: http://bugs.digium.com/view.php?id=13756 |
| 04:28.50 | VoipForces | Ok, for those iterested, looks like that for asterisk 1.6.0.1, you need spandsp-0.0.6pre1 + the patch in bug 13688 |
| 04:29.07 | VoipForces | + if you want to to t.38 you need the patch in bug 13756 |
| 04:30.11 | VoipForces | And I need to familiarize with those new 1.6 commands... |
| 04:31.41 | *** join/#asterisk CunningPike_ (n=arodgers@204.239.8.149) |
| 04:32.31 | *** join/#asterisk etfonhomey (n=chatzill@74-143-196-254.static.insightbb.com) |
| 04:32.58 | VoipForces | Well, found a bug with 1.6 and freepbx 2.5.1.0 |
| 04:36.31 | VoipForces | Ok, I'm brain dead. Good night everyone |
| 04:43.04 | *** join/#asterisk RypPn (n=Sally@rosscom.demon.co.uk) |
| 04:43.52 | mosty | i have a problem with MixMonitor in asterisk 1.4 becoming out of sync, so i'm trialling Monitor as a replacement. the logs show the Monitor command, but i can't find the file that it creates. /var/spool/asterisk/monitor/ is empty (and writable), and it fails even when i put the full path in the Monitor call. what could be wrong? |
| 04:47.04 | *** join/#asterisk ricko73 (n=dhartman@wilug/newlug/ricko73) |
| 04:47.31 | *** join/#asterisk jameswf (n=james@ip68-3-60-66.ph.ph.cox.net) |
| 05:03.10 | mosty | seems to be a bug with the 'b' option |
| 05:07.14 | *** part/#asterisk subdolus (n=subdolus@subby.afraid.org) |
| 05:14.20 | *** join/#asterisk zakiazigazi (n=zakiazig@fw1.ngigroup.com) |
| 05:19.27 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 05:24.12 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 05:24.16 | phix | hey |
| 05:25.28 | jameswf | ho |
| 05:26.24 | ricko73 | hip hop hip hop |
| 05:26.52 | ricko73 | jameswf: you kickin it with Naughty by Nature? |
| 05:27.27 | jameswf | NWA -Networks with Admins... |
| 05:28.59 | jameswf | Rap had 2pac, I am 2pair |
| 05:29.39 | jameswf | oh I should start a band called twisted pair |
| 05:29.46 | styelz | iam LimeLight Cool J |
| 05:30.02 | ricko73 | oh you're twisted alright... |
| 05:30.19 | styelz | hook me up some of that Captain Rock |
| 05:30.26 | jameswf | J- Java |
| 05:30.35 | ricko73 | cranking out new image files for AstLinux (0.6.2 will be uploaded before the weekend) |
| 05:31.13 | jameswf | ricko73: you should use our beta drivers |
| 05:31.28 | ricko73 | jameswf: perhaps in trunk |
| 05:31.45 | ricko73 | we're using 2.2.6 currently |
| 05:32.12 | jameswf | bryce may have em built, we push em quite a bit just dont call em release.. |
| 05:32.56 | ricko73 | gotcha |
| 05:33.26 | ricko73 | still need to get someone from you co on the voip-user-conference call |
| 05:33.31 | ricko73 | your co |
| 05:34.24 | jameswf | I am always there just lurking :) we should have our BRI card within 90 or so maybe I can do an intro... |
| 05:34.45 | *** join/#asterisk VeKTeReX (n=kevin@58-70-61-222.eonet.ne.jp) |
| 05:35.18 | *** part/#asterisk VeKTeReX (n=kevin@58-70-61-222.eonet.ne.jp) |
| 05:38.27 | jameswf | off to bed |
| 05:38.43 | ricko73 | yeah...heading there shortly |
| 05:38.53 | *** join/#asterisk freakazoid0223 (n=mattc@pool-71-242-217-94.phlapa.east.verizon.net) |
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| 06:27.19 | C4away | Playtones only works on early media if audio has been played back previously on that channel? |
| 06:27.42 | C4away | Asterisk 1.6.0.1 |
| 06:29.59 | C4away | tested on a sip phone registered to an asterisk server calling another asterisk server over a SIP trunk, and from the PSTN through an SS7 gateway to the asterisk server ... both ways the Playtones(!950/330,!1400/330,!1800/330,0) was silent the first time it played, then the "not in service" message plays and can be heard, then the second Playtones(!950/330,!1400/330,!1800/330,0) is heard on the channel |
| 06:31.04 | C4away | I fixed it by playing silence/1 first then the playtones |
| 06:31.15 | C4away | is this a known issue? |
| 06:31.57 | *** join/#asterisk djin (n=djin@84-105-70-127.cable.quicknet.nl) |
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| 06:45.03 | Paige_ | hi365, has anyone managed to get a cisco 7911 phone to work with asterisk? |
| 06:46.35 | hi365 | Paige_: did I become the mod for cisco phone issues? |
| 06:47.38 | Paige_ | hi365 sorry autocomplete fail |
| 06:47.47 | hi365 | cool then |
| 06:50.15 | *** join/#asterisk sinelaw (n=sinelaw_@217.132.93.220) |
| 06:52.25 | sinelaw | can asterisk do failover for phone calls? (if a server goes down during a call, the call will be continued on another server)? |
| 06:53.40 | afink | um yes |
| 06:56.31 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
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| 07:09.10 | sinelaw | how? |
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| 08:07.08 | farah | hi all |
| 08:07.51 | farah | anyone knows how to monitor the command "iax2 show netstats" using SNMP |
| 08:08.34 | farah | plz i need some help |
| 08:38.16 | Paige_ | hi365, has anyone managed to get a cisco 7911 phone to work with asterisk? |
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| 08:55.51 | aiksa[LV] | Hi everyone |
| 08:56.27 | *** join/#asterisk seldev (i=test@firewall.selmoni.ch) |
| 08:56.32 | aiksa[LV] | is there an option how to pass early audio from zaptel E1 channel to end device without Answer()ing the channel before? |
| 08:57.38 | seldev | hi everyone |
| 08:57.56 | seldev | i'm looking for help with a digium tdm410p with an fxs module |
| 08:58.43 | seldev | is someone arround who has experience with this type of card? |
| 09:14.55 | tzafrir_laptop | ~ask |
| 09:14.56 | jbot | somebody said ask was 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. |
| 09:15.03 | tzafrir_laptop | seldev, ==^ |
| 09:17.14 | seldev | ok, I've got the mentioned card and working kernel modules but no dial tone at all |
| 09:18.32 | seldev | when starting zaptel i get this |
| 09:18.32 | seldev | Loading zaptel framework: [ OK ] |
| 09:18.32 | seldev | Waiting for zap to come online...OK |
| 09:18.32 | seldev | Loading zaptel hardware modules: wctdm24xxp. |
| 09:18.32 | seldev | Running ztcfg: ioctl(ZT_LOADZONE) failed: Invalid argument |
| 09:18.33 | seldev | Notice: Configuration file is /etc/zaptel.conf |
| 09:18.35 | seldev | line 268: Unable to register tone zone 'ch' |
| 09:18.37 | seldev | <PROTECTED> |
| 09:18.57 | seldev | zaptel version is 1.4.7.1 |
| 09:21.03 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-b5f00438443dca1d) |
| 09:21.54 | seldev | my /etc/zaptel.conf: |
| 09:21.54 | seldev | fxoks=1 |
| 09:21.54 | seldev | loadzone=ch |
| 09:21.54 | seldev | defaultzone=ch |
| 09:22.39 | *** join/#asterisk subdolus (n=subdolus@subby.afraid.org) |
| 09:23.55 | *** part/#asterisk Paige_ (n=Paige@c-67-165-114-31.hsd1.wa.comcast.net) |
| 09:23.56 | seldev | at which point i should have at least a dialtone when i pickup my phone? after loading kernel modules or after starting asterisk? |
| 09:28.01 | *** join/#asterisk Chris-NB (n=chris@85-126-61-10.work.xdsl-line.inode.at) |
| 09:34.53 | *** join/#asterisk Rico29 (n=Rico@lns-bzn-48f-81-56-220-28.adsl.proxad.net) |
| 09:49.31 | farah | anyone knows how to monitor the command "iax2 show netstats" using SNMP |
| 09:49.38 | farah | plz i need some help |
| 09:50.07 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 09:50.51 | mort_gib | farah; TK had a point yesterday |
| 09:51.10 | tzafrir_laptop | seldev, you need a newer version of zaptel, IIRC |
| 09:52.14 | tzafrir_laptop | http://docs.tzafrir.org.il/#_past_incompatibilities |
| 09:53.09 | tzafrir_laptop | Alternatively, build asterisk vs. 1.4.7.1's zaptel.h |
| 09:53.19 | tzafrir_laptop | or do you use asterisk 1.4.22 ? |
| 09:53.58 | hi365 | is it posible to send the vm email to more than one address? |
| 09:54.13 | tzafrir_laptop | err.. it's pure zaptel |
| 09:54.45 | tzafrir_laptop | hi365, you can always use an alias in your mta configuration... |
| 09:55.14 | *** join/#asterisk Rico29 (n=Rico@lns-bzn-48f-81-56-220-28.adsl.proxad.net) |
| 09:55.23 | hi365 | tzafrir_laptop: without that? |
| 10:03.29 | *** join/#asterisk suvir (n=suvir@ppp-124-120-130-33.revip2.asianet.co.th) |
| 10:05.32 | djin | hi |
| 10:05.43 | suvir | hi |
| 10:05.57 | djin | Does anyone use SNOM (300) with Freeswitch? |
| 10:06.04 | seldev | tzafrir_laptop thanks. that resolved the error message on zaptel start but still no dialtone |
| 10:06.38 | djin | calling from SNOM to other SIP (soft-)phones works by default, but calling to SNOM fails. |
| 10:06.44 | tzafrir_laptop | djin, ask on #freeswitch (if there is indeed such a channel) ? |
| 10:06.50 | djin | haha |
| 10:07.00 | djin | sorry, wrong window :) |
| 10:08.24 | tzafrir_laptop | seldev, what version of asterisk do you have? |
| 10:08.34 | tzafrir_laptop | and what did you actually do? |
| 10:08.42 | seldev | 1.4.21.2 |
| 10:08.46 | farah | anyone knows how to monitor the command "iax2 show netstats" using SNMP plz? |
| 10:13.01 | seldev | I've got a TDM410 card with one FXS module on socket 1. On port 1 I have an analog phone. It's a modified Trixbox system. |
| 10:13.43 | seldev | my zapata.conf looks like this |
| 10:14.07 | seldev | language=de |
| 10:14.07 | seldev | signalling=fxo_ks |
| 10:14.07 | seldev | context=Internal |
| 10:14.07 | seldev | channel => 1 |
| 10:14.37 | seldev | kernel modules zaptel and wctdm24xxp are loaded |
| 10:15.27 | seldev | in modprobe.conf I've got the "line install wctdm24xxp /sbin/modprobe --ignore-install wctdm24xxp opermode=SWITZERLAND fxshonormode=1 boostringer=1 fastringer=1 && /sbin/ztcfg" as this was mentioned in a forum |
| 10:16.31 | seldev | module loading is successfull, asterisk loads zapata.conf, but no dialtone when I pickup the phone nor ringing when calling the extension |
| 10:17.16 | seldev | do you need anything else? |
| 10:18.37 | *** join/#asterisk musse- (n=musse@static-212.214.40.123.addr.tdcsong.se) |
| 10:18.52 | *** join/#asterisk BrokenNoze (n=magic@host81-137-172-233.in-addr.btopenworld.com) |
| 10:20.14 | *** part/#asterisk musse- (n=musse@static-212.214.40.123.addr.tdcsong.se) |
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| 10:20.35 | BrokenNoze | Hi all, has anyone ever had a problem with 1.4.2 and music on hold cutting in half way through a call? |
| 10:22.10 | tzafrir_laptop | seldev, do you see anterisk complaining about a failed TONEZONE ioctl in its logs? |
| 10:23.40 | seldev | no complaining |
| 10:25.27 | *** join/#asterisk grEvenX (n=even@apb9hb.ip.ssc.net) |
| 10:26.46 | seldev | tzafir_laptop should there be a channel entry on "zap show channels" ? |
| 10:30.35 | seldev | tzafrir_laptop there is a supply voltage on the line as I can hear dtmf tones when i press the phone keys |
| 10:31.24 | tzafrir_laptop | seldev, what's the output of 'zap show channels' ? |
| 10:37.22 | seldev | tzafrir_laptop it shows only the title row => Chan Extension Context Language MOH Interpret |
| 10:38.10 | tzafrir_laptop | what is the output of: cat /proc/zaptel/1 |
| 10:38.45 | seldev | Span 1: WCTDM/0 "Wildcard TDM410P Board 1" (MASTER) |
| 10:38.45 | seldev | <PROTECTED> |
| 10:38.45 | seldev | <PROTECTED> |
| 10:38.46 | seldev | <PROTECTED> |
| 10:38.46 | seldev | <PROTECTED> |
| 10:38.46 | seldev | <PROTECTED> |
| 10:48.13 | tzafrir_laptop | any chance you got the wrong channel? |
| 10:48.36 | tzafrir_laptop | what is the output of: genzaptelconf -l |
| 10:49.53 | seldev | ### Span 1: WCTDM/0 "Wildcard TDM410P Board 1" (MASTER) |
| 10:49.53 | seldev | 1 FXS |
| 10:49.53 | seldev | # channel 2, WCTDM, no module. |
| 10:49.53 | seldev | # channel 3, WCTDM, no module. |
| 10:49.53 | seldev | # channel 4, WCTDM, no module. |
| 10:51.28 | seldev | tzafrir_laptop where do you mean I could have gotten the wrong channel? zapata.conf? |
| 10:52.14 | tzafrir_laptop | no. it's correct |
| 10:52.43 | tzafrir_laptop | can you try 'zap restart' in asterisk? If that doesn't help: restart asterisk |
| 10:52.57 | seldev | <PROTECTED> |
| 10:52.57 | seldev | <PROTECTED> |
| 10:53.26 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 10:53.31 | seldev | restart when convenient |
| 10:53.31 | seldev | Waiting for inactivity to perform restart |
| 10:53.31 | seldev | selphone-01*CLI> |
| 10:53.31 | seldev | Disconnected from Asterisk server |
| 10:53.33 | joelsolanki | jql: Hi |
| 10:54.11 | joelsolanki | [Nov 26 05:48:45] WARNING[24426]: chan_sip.c:3036 sip_call: No audio format found to offer |
| 10:54.27 | seldev | still no channels tzafrir_laptop |
| 10:54.32 | joelsolanki | this seems to be codec problem. i am trying to transcode. let me present the scenario |
| 10:54.51 | joelsolanki | ipphone --> asterisk1 --> asterisk2 --> asterisk3 |
| 10:55.05 | tzafrir_laptop | seldev, look for errors in the asterisk logs |
| 10:55.13 | tzafrir_laptop | look for chan_zap.c there |
| 10:55.25 | joelsolanki | ipphone send ulaw and asterisk1 accepts ulaw and forwards to asterisk2 and asterisk must do transcoding and send the g729 call to asterisk3 |
| 10:55.39 | joelsolanki | but i am getting error [Nov 26 05:48:45] WARNING[24426]: chan_sip.c:3036 sip_call: No audio format found to offer on asterisk2 box |
| 10:55.52 | joelsolanki | let me pastebin the configs |
| 10:56.18 | angryuser | joelsolanki: are you able to transcode on asterisk2 g729 ? check locally |
| 10:57.33 | joelsolanki | angryuser: how do i check locally ? |
| 10:57.54 | joelsolanki | yes i think i can connect eyebeam to asterisk2 and see if it does transcoding |
| 10:57.59 | angryuser | joelsolanki: setup a peer with ulaw&alaw and call a peer with g729 forced |
| 10:58.12 | joelsolanki | let me do that and also you the configs of all boxes |
| 10:58.12 | *** join/#asterisk chris`_ast (i=chris@penguin.curious3.co.uk) |
| 10:58.24 | seldev | tzafrir_laptop yes there's an error |
| 10:58.25 | seldev | [Nov 26 11:55:56] ERROR[24453] chan_zap.c: Unable to load config zapata.conf |
| 10:58.43 | tzafrir_laptop | and before that? |
| 10:58.55 | tzafrir_laptop | ls -l /etc/asterisk/zapata.conf |
| 10:59.04 | chris`_ast | Asterisk 1.4.2 and g729 codecs, are there none bugs with that or am I just doing it wrong :P |
| 10:59.21 | chris`_ast | Registered 2 channels but when ever I make a sip > sip or sip > zap call, it complains about being out ofl icences. |
| 10:59.33 | chris`_ast | asterisk cli show g729 shows both licences thoguh |
| 11:00.33 | seldev | [Nov 26 11:58:31] WARNING[24468] config.c: parse error: No category context for line 1 of /etc/asterisk/zapata.conf |
| 11:00.33 | seldev | [Nov 26 11:58:31] ERROR[24468] chan_zap.c: Unable to load config zapata.conf |
| 11:00.49 | seldev | that would be the language entry |
| 11:01.41 | joelsolanki | angryuser: http://www.pastebin.ca/1267513 |
| 11:01.55 | joelsolanki | this is the current config. let me check the transcoding locally too |
| 11:02.01 | joelsolanki | plz take a look at config |
| 11:02.22 | seldev | i commented out the first line now is the error on line 2 |
| 11:04.08 | tzafrir_laptop | seldev, I guess you're missing '[channels]' at the top of zapata.conf |
| 11:05.10 | seldev | correct. i don't have such a line... I took the example on voip-info.org.. |
| 11:05.33 | joelsolanki | angryuser: you are right asterisk2 itself is not able to transcode |
| 11:05.39 | seldev | selphone-01*CLI> zap restart |
| 11:05.39 | seldev | <PROTECTED> |
| 11:05.39 | seldev | <PROTECTED> |
| 11:05.39 | seldev | <PROTECTED> |
| 11:05.39 | seldev | <PROTECTED> |
| 11:06.03 | seldev | looks much better tzafrir_laptop. thanks a lot so far. |
| 11:06.22 | joelsolanki | angryuser: you there ? |
| 11:07.13 | angryuser | joelsolanki: yes sec |
| 11:08.10 | joelsolanki | ok |
| 11:08.27 | angryuser | joelsolanki: so check the licenses on asterisk2 box |
| 11:09.33 | joelsolanki | hmm. |
| 11:09.46 | joelsolanki | let me checkout what happend. maybe something went during restore |
| 11:12.31 | chris`_ast | [Nov 26 11:13:43] WARNING[16232] codec_g729a.c: out of G.729 decoder licenses |
| 11:12.31 | chris`_ast | [Nov 26 11:13:43] WARNING[16232] translate.c: g729tolin did not update samples 0 |
| 11:12.35 | chris`_ast | Theres 2 licences :( |
| 11:13.22 | seldev | tzafrir_laptop i have a dialtone but when i can only dial 1 digit then it hangs up |
| 11:13.34 | seldev | calling the extension works like a charm |
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| 11:46.11 | itguru | I'm trying to configure outbound calls on a UK ISDN30e setup. incoming and outgoing calls work fine for local numbers, 0800, and international calls do not - any tips? |
| 11:54.11 | *** join/#asterisk itguru (n=p@5ad906d0.bb.sky.com) |
| 12:03.30 | *** join/#asterisk xrmx__ (n=user@host116-173-dynamic.10-79-r.retail.telecomitalia.it) |
| 12:04.49 | xrmx__ | hi, anybody knows if thre's a way to trace call flow from asterisk cli? |
| 12:05.42 | tzafrir_laptop | what type of call? |
| 12:05.48 | xrmx__ | incoming call |
| 12:06.05 | tzafrir_laptop | what channel? sip? iax? zap? |
| 12:06.23 | tzafrir_laptop | And what type of "flow"? |
| 12:06.37 | xrmx__ | iax, i mean the call arrive but i hear a "Goodbye" prompt |
| 12:06.49 | xrmx__ | and phones don't ring |
| 12:06.58 | tzafrir_laptop | (geenrally I believe that the answer is "yes" if I guess your meaning correctly) |
| 12:07.34 | tzafrir_laptop | generally with verbosity level 3 you also see dialplan flow |
| 12:08.03 | tzafrir_laptop | in logger.conf you should set the 'console' to show 'verbose' messages |
| 12:08.12 | tzafrir_laptop | (see 'logger show channels') |
| 12:09.31 | xrmx__ | yeah, that's it , thanks :) |
| 12:09.58 | xrmx__ | call are going to voicemail |
| 12:10.05 | xrmx__ | s/call/calls |
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| 12:35.34 | jyfletcher | I have a sip client that works, but the Status line in "sip show peer xxxx" is always UNKNOWN. I have set debug for the number and see the registration and it looks ok to me... |
| 12:36.17 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:36.17 | *** mode/#asterisk [+o lmadsen] by ChanServ |
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| 12:39.44 | farah | anyone knows how to monitor the command "iax2 show netstats" using SNMP plz |
| 12:45.08 | *** join/#asterisk Meaw (n=dino@213.244.81.144) |
| 12:46.49 | Meaw | hello, im testing out asterisk in a vps but i got some errors in compiling it , im following this link http://www.vsppanel.com/gettingstarted_installing.html |
| 12:50.14 | *** join/#asterisk ZaVoid (n=zavoid@75.147.121.177) |
| 12:51.01 | Daviey | Meaw: what errors? |
| 12:52.25 | Meaw | in the link they are compiling asterisk addons first, maybe i should compile asterisk first then asterisk addons? |
| 12:58.14 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 12:58.56 | *** join/#asterisk sasargen (n=chatzill@72.58.82.74) |
| 12:59.56 | Akiyuki | Is there a way to set a asterisksoundsdir variable? |
| 13:00.12 | Akiyuki | So I dont have to Playback(/var/lib/asterisk/sounds/foo) each time |
| 13:00.57 | Meaw | ah i compiled asterisk first and it works |
| 13:01.12 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 13:01.40 | *** join/#asterisk lou_gr (n=lou@212-70-216-131.ath.static.tee.gr) |
| 13:06.18 | *** join/#asterisk h-idrisi (n=h-idrisi@212.100.196.195) |
| 13:09.16 | *** join/#asterisk nikko (n=nikko@173-17-214-107.client.mchsi.com) |
| 13:11.09 | Meaw | when i start asterisk : /usr/sbin/safe_asterisk |
| 13:11.17 | Meaw | i got an error says /usr/sbin/safe_asterisk: line 130: /dev/tty9: Permission denied |
| 13:11.38 | Meaw | any other way to start it? |
| 13:11.58 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 13:12.40 | Akiyuki | Try /etc/init.d/asterisk start |
| 13:13.20 | Meaw | still getting permission denied /usr/sbin/safe_asterisk |
| 13:13.43 | Akiyuki | What user are you executing this ass? |
| 13:14.01 | Meaw | root |
| 13:14.05 | Akiyuki | ouch |
| 13:14.07 | Akiyuki | I'm not sure then. |
| 13:14.43 | *** join/#asterisk zeeesh (n=zeeesh@203.215.176.22) |
| 13:15.00 | Meaw | :/ |
| 13:15.02 | *** join/#asterisk n3hxs (n=HAMming@63.68.135.4) |
| 13:16.33 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:17.10 | *** mode/#asterisk [+o [TK]D-Fender] by ChanServ |
| 13:17.41 | *** topic/#asterisk by [TK]D-Fender -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0.1 (2008/10/09), 1.4.22 (2008/10/02), *-Addons 1.6.0 (2008/10/02), 1.4.7 (2008/06/04), dahdi-linux-complete 2.0.0+2.0.0 (2008/10/02), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-commits #asterisk-bugs #asterisk-dev |
| 13:19.14 | *** join/#asterisk feeds (n=feeds@85-135-245-223.adsl.slovanet.sk) |
| 13:19.42 | *** join/#asterisk protocols (n=protocol@p5791FD1E.dip.t-dialin.net) |
| 13:21.29 | feeds | hi, what does this mean: Unable to install capabilities. <-- It prints when I try to run asterisk as user asteriskserver, when he is in the group asterisk and he's owner of all the * files, except /etc/asterisk, which he can access and /usr/share/asterisk . Why is this happening? |
| 13:21.52 | feeds | and he's runuser in asterisk.conf ... |
| 13:22.00 | *** join/#asterisk UnixDawg (n=UnixDawg@cpe-98-28-149-231.cinci.res.rr.com) |
| 13:22.39 | *** join/#asterisk xrmx__ (n=user@host116-173-dynamic.10-79-r.retail.telecomitalia.it) |
| 13:24.25 | *** join/#asterisk neurosys (n=neurosys@c-66-229-81-161.hsd1.fl.comcast.net) |
| 13:26.28 | feeds | someone? |
| 13:26.57 | *** join/#asterisk donnib (n=aaaa@0x555281d0.adsl.cybercity.dk) |
| 13:27.54 | donnib | hi all, i have a problem with a phone which when i call the other party picks up but the call fails. see http://pastebin.com/d4b7d45d4 around line 381 to 400. |
| 13:29.19 | donnib | here is the settings for that particular extension. http://pastebin.com/d8966dc4 |
| 13:29.35 | donnib | the call i make is between two extensions |
| 13:30.17 | donnib | i can make calls fine between a hard phone and a xlite client but problems between hardware and a specific hardware client (Linksys SPA942) |
| 13:30.47 | donnib | anyone have an idea what is wrong ? |
| 13:32.05 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 13:33.13 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 13:35.22 | *** part/#asterisk Spirits-Sight (n=christop@c-24-218-128-159.hsd1.ma.comcast.net) |
| 13:35.27 | *** join/#asterisk WimpMan (n=wimpy@213.240.181.251) |
| 13:36.37 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.195.82) |
| 13:38.39 | [TK]D-Fender | donnib: Provide * CLI output with SIP debug, and show the config of BOTH sides. |
| 13:40.39 | *** join/#asterisk UnixDawg_ (n=UnixDawg@cpe-98-28-149-231.cinci.res.rr.com) |
| 13:42.50 | rwaite | god i have a headache today |
| 13:43.40 | donnib | @[TK]D-Fender: here is the CLI output of a call i made now. i am calling from 100 to 110 http://pastebin.com/d2edf9266 |
| 13:43.44 | *** join/#asterisk UnixDawg_ (n=UnixDawg@cpe-98-28-149-231.cinci.res.rr.com) |
| 13:46.32 | [TK]D-Fender | donnib: Where are these 2 phones located relative to *? |
| 13:46.45 | donnib | @[TK]D-Fender: the config on the asterisk side for 100 is http://pastebin.com/d8966dc4. the config for 110 is : http://i36.tinypic.com/210hz5c.jpg. the debug on 110 is : http://pastebin.com/d4b7d45d4 |
| 13:47.05 | donnib | they are in two different parts of the world. one in india and one in Denmark |
| 13:48.14 | [TK]D-Fender | donnib: Well they both seem to list a PRIVATE IP, so unless you're VPN'd then you did not configure them for NAT properly. |
| 13:48.21 | donnib | i know there is almost 400ms latency between these two extensions |
| 13:48.30 | donnib | they are on a vpn |
| 13:48.39 | donnib | they run on same network |
| 13:48.48 | [TK]D-Fender | Retransmitting #5 (no NAT) to 10.115.91.36:5060: |
| 13:49.00 | donnib | yes that is correct |
| 13:49.02 | [TK]D-Fender | donnib: then its another routing / firewall issue |
| 13:49.16 | donnib | but why on that particular phone |
| 13:49.34 | donnib | as mentioned it works fine calling other clients which uses x-lite client |
| 13:49.38 | [TK]D-Fender | donnib: Go examine his environment closer. test with a remote softphone |
| 13:49.49 | [TK]D-Fender | donnib: at that specific location? |
| 13:51.35 | donnib | what to you mean a specific location ? in the same offices i have many users using x-lite. they all work but this hardphone does not work. all users are same place physically |
| 13:51.45 | donnib | and running on same connection |
| 13:52.15 | [TK]D-Fender | donnib: thst precise location. |
| 13:52.22 | donnib | i have run out of ideas |
| 13:52.34 | donnib | you mean the plug on the wall ? |
| 13:52.48 | [TK]D-Fender | donnib: you have other users in that same office that work ok? |
| 13:52.56 | donnib | i can access the web server for the phone |
| 13:53.02 | donnib | Yes but they all use x-lite |
| 13:53.12 | donnib | besides this one i have problems with |
| 13:53.34 | [TK]D-Fender | donnib: Web doesn't prove that other things will works, but if you have X-Lite running on their network, then you should be OK. |
| 13:53.44 | donnib | the phone rings. it's registered in asterisk but when they pick up then it doesn't work |
| 13:54.04 | donnib | yes x-lite works fine on that network. |
| 13:54.31 | donnib | all users are set to not use NAT since we all are running on the same network |
| 13:55.58 | donnib | does the debug from the phone tell you anything ? i can see there are some Internal server error and some 487 errors |
| 13:56.13 | [TK]D-Fender | donnib: Ok, I'm not sure at this point... if its VPN'd and X-Lite works from that location, There must be something wrong with your config of that phone |
| 13:56.18 | donnib | don't know if u get something from that |
| 13:56.28 | [TK]D-Fender | donnib: what the debug tells me is that packets go out, but never come back. |
| 13:57.08 | coppice | what do you think they are? boomerangs? |
| 13:57.21 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 13:58.28 | donnib | i am looking into the manual for SPA-942 to see if i did something wrong |
| 13:59.13 | [TK]D-Fender | coppice: g'day |
| 13:59.38 | coppice | i guess I fostered such a response |
| 13:59.57 | [TK]D-Fender | coppice: Fosters.... Australian for beer! |
| 14:00.55 | coppice | Fosters - like making love in a punt |
| 14:01.15 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 14:01.40 | [TK]D-Fender | coppice: http://www.youtube.com/watch?v=f3RYHKWXIwI |
| 14:02.13 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:02.16 | *** join/#asterisk Dovid (n=annon@tony09-118-62.inter.net.il) |
| 14:02.51 | Dovid | TK: Just noticed that you got mod status. Congrats |
| 14:04.21 | coppice | [TK]D-Fender: pointing to the "foreplay" commercial would have been a better response to my quip |
| 14:04.21 | [TK]D-Fender | Dovid: Been about a year now. |
| 14:04.29 | *** mode/#asterisk [-o [TK]D-Fender] by [TK]D-Fender |
| 14:04.59 | [TK]D-Fender | coppice: Didn't want to waste too much time browsing, and its been 5 years since I've had standard television and seen those commercials |
| 14:05.20 | coppice | Dovid: he finally passed the Mr Grumpy test, and qualified |
| 14:05.25 | Dovid | ;) |
| 14:05.46 | [TK]D-Fender | coppice: Not entirely inaccurate ;) |
| 14:09.47 | Dovid | When i script stuff I put in on top the date created and time for future refrence. here is the last one that I did: Created on 11-25-2008 @ 0939 By (a very tired and grumpy.....) Dovid |
| 14:10.38 | coppice | one offs don't count. you need to establish a track record |
| 14:10.50 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 14:11.08 | hi365 | Dovid: hello |
| 14:11.21 | Dovid | hi hi |
| 14:11.33 | hi365 | Dovid: do you work for Moshe M? |
| 14:11.38 | Dovid | hi365: hi |
| 14:11.54 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:12.11 | Dovid | define work ;) |
| 14:12.24 | hi365 | oh, stop it! |
| 14:13.21 | coppice | work is what you're doing when you're not having fun |
| 14:13.32 | *** join/#asterisk freezey (i=hidden-u@gw.mypublisher.com) |
| 14:14.14 | freezey | question.. i am about to try and load the RingList.xml file to my cisco 7940's this file only has 1 custom ringtone that i wanted to try.. is this going to overwrite all the current default ringers the phone came with? |
| 14:16.47 | Dovid | TK: I no longer think for myself and actually *try* to follow manuals. installing Vicidialr and they have exten => _*NXXNXXXXXX*3429 trying to figure out if the * has any sagnifigance as far as asterisk is concerned or just the way they have it ? They also have exten => _**3429 which i do not see any X so why the leading _. any guess ? I want to write it **the correct way** but want to make sure they do not signify anything to Asterisk |
| 14:17.11 | *** join/#asterisk telnettech (i=telnette@gw.percipia.com) |
| 14:22.12 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-b114e28f8f516bec) |
| 14:22.12 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:22.12 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 14:23.11 | [TK]D-Fender | Dovid: You're right..... you no longer think. |
| 14:23.18 | *** join/#asterisk feeds (n=feeds@85-135-245-223.adsl.slovanet.sk) |
| 14:23.33 | [TK]D-Fender | Dovid: Not too much further before bottom now! |
| 14:23.45 | Dovid | TK: Thinking gets me in to too much trouble |
| 14:24.10 | [TK]D-Fender | Dovid: Lets see how this "not thinking" thing pans out... |
| 14:24.20 | Dovid | so far real bad |
| 14:25.04 | [TK]D-Fender | Dovid: Rock. Hard place. Hard place. Rock. Good, now we've passed the formailties and you can continue on to greater suffering |
| 14:26.48 | donnib | hmm....tried to look in the manual and everything is setup correct |
| 14:27.19 | mikealeonetti | [TK]D-Fender: what do you do for a living? |
| 14:27.21 | donnib | i even compared all the settings between the two phones. they are both Linksys and everyting is exactly the same |
| 14:27.45 | [TK]D-Fender | mikealeonetti: I am the IT dept for a non-tech company |
| 14:28.17 | mikealeonetti | [TK]D-Fender: anyone I know? |
| 14:28.35 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 14:29.09 | [TK]D-Fender | mikealeonetti: Small place, very unlikely |
| 14:29.17 | mikealeonetti | [TK]D-Fender: why asterisk? |
| 14:29.25 | [TK]D-Fender | mikealeonetti: meaning? |
| 14:30.10 | mikealeonetti | [TK]D-Fender: why chose to become proficient in Asterisk, or act like it if you're not |
| 14:30.25 | [TK]D-Fender | mikealeonetti: Because I LIKE it maybe? |
| 14:30.45 | [TK]D-Fender | mikealeonetti: And enough to get my own company to switch to it when we moved, perhaps. |
| 14:30.59 | [TK]D-Fender | mikealeonetti: and the fact I am an * consultant on the side as well.... |
| 14:31.37 | mikealeonetti | [TK]D-Fender: interesting |
| 14:32.20 | mikealeonetti | antikkkx |
| 14:32.24 | mikealeonetti | err |
| 14:33.19 | jaytee | good god! I could have written a Gui based IDE, compiler and library in the amount of time Visual Studio SP1 takes to install. |
| 14:34.21 | telnettech | jayte: is that the Visual dialplan? |
| 14:36.19 | [TK]D-Fender | ... extrapolation FAIL |
| 14:36.41 | jaytee | telnettech, no the Visual Dialplan is something else. |
| 14:36.54 | freezey | i am using the P0S3-08-2-00.sb2 is it possible to even force a ringtone onto the phones if this file is being used? |
| 14:37.02 | feeds | what does the astdb file in the lib directory hold? |
| 14:37.18 | jaytee | feeds, whatever data you wish to put in it |
| 14:37.25 | feeds | and how? |
| 14:37.31 | jaytee | ~book |
| 14:37.31 | jbot | well, book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 14:37.33 | jaytee | that's how |
| 14:37.58 | feeds | ... thanks ... |
| 14:39.58 | jaytee | feeds, pages 160-162 show you some examples of it's uses. it's more of a "flat" file database, not a true relational database but useful nonetheless. |
| 14:41.37 | [TK]D-Fender | feeds: "core show function DB" |
| 14:41.48 | [TK]D-Fender | feeds: CLI "help db" |
| 14:42.28 | [TK]D-Fender | feeds: and for the quickest start to what it holds "database show" |
| 14:42.39 | [TK]D-Fender | feeds: CLI "help database" <- correction |
| 14:45.56 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:50.19 | *** part/#asterisk jyfletcher (n=justin@82.150.65.31) |
| 14:57.33 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 14:57.54 | *** part/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 14:59.29 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 15:03.53 | *** join/#asterisk VoipForces (n=courchea@firewall.privalodc.com) |
| 15:04.01 | VoipForces | Hi, anyone is awaya of a NVFaxDetect port for asterisk 1.6 ? |
| 15:04.09 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 15:06.12 | *** join/#asterisk jcape (n=jcape@209.120.251.66) |
| 15:07.36 | jcape | My apologies if this is a FAQ: I'm trying to accept an inbound FAX on a PRI, and then forward it back out via an FXS card to a real fax machine, using Digium single-port T1 and 8-port AEX800 FXS. Are there any writeups on this, and/or what gotchas may exist? |
| 15:07.50 | SuPrSluG | can you use asterisk to route traffic to another asterisk box? kind of like a proxy. need to decommission a box and need something in front * until the other is ready. |
| 15:08.20 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 15:09.09 | SuPrSluG | it would do nothing other than pass request to pstn a lucent box |
| 15:09.52 | [TK]D-Fender | SuPrSluG: * is not a proxy |
| 15:10.24 | *** join/#asterisk flohack (n=fhackenb@mk090152235252.a1.net) |
| 15:10.37 | [TK]D-Fender | jcape: Any fax risks instability. EC should disable itself on detection of the tones. |
| 15:10.48 | VoipForces | SuPrSluG: Have done it with a Meridian system. Put the asterisk in front and tie the asterisk to the legacy PBX wia a T1. |
| 15:10.54 | SuPrSluG | opensips only way right. i thought that. the boss wants to decom and this didin't sound right |
| 15:11.34 | jcape | [TK]D-Fender: Well, right now our faxes are coming in over the PRI and getting redirected backout and over some inbound POTS lines (not my setup) |
| 15:11.51 | SuPrSluG | VoipForces:it would look like Lucent TNT -> asterisk -> asterisk1 and asterisk2 |
| 15:11.53 | jcape | I'd like to have it just run through Asterisk if possible |
| 15:12.10 | [TK]D-Fender | jcape: it is, though tiny timing issues etc risk losing faxes. |
| 15:12.14 | [TK]D-Fender | pcYMMV |
| 15:12.20 | *** join/#asterisk feeds (n=feeds@85-135-245-223.adsl.slovanet.sk) |
| 15:12.43 | VoipForces | SuPrSluG: What is your current setup and what would you like to achieve? |
| 15:12.53 | SuPrSluG | yes |
| 15:12.59 | jameswf | [TK]D-Fender: you know the reliability of zaptel and tone detection |
| 15:13.04 | SuPrSluG | sorry |
| 15:13.33 | SuPrSluG | VoipForces:it would look like Lucent TNT -> asterisk1 and asterisk2 (current) |
| 15:13.37 | jameswf | often the that detection only works on a fluke |
| 15:13.49 | [TK]D-Fender | jcape: Make sure to disable EC for the TM ports you're going to sue and on bridge |
| 15:14.08 | SuPrSluG | VoipForces:it would look like Lucent TNT -> asterisk -> asterisk1 and asterisk2 ( needed until old boxes decommissioned) |
| 15:14.10 | [TK]D-Fender | jameswf: Fluke makes great diagnostic tools :) |
| 15:14.26 | jcape | [TK]D-Fender: TM ports? |
| 15:14.33 | [TK]D-Fender | TDM |
| 15:14.35 | VoipForces | SuPrSluG: which is your old box? |
| 15:14.42 | [TK]D-Fender | Your analog ones with the fax machines on it |
| 15:15.17 | jcape | So disable EC on the analog ports *and* the PRI, or just the analog? |
| 15:15.25 | jcape | Is there a writeup somewhere, or is that something I'm contributing later? |
| 15:15.28 | SuPrSluG | VoipForces: right now it goes from a lucent tnt to asterisk |
| 15:16.36 | SuPrSluG | they want something in the middle until they decide to roll over to a new architecture |
| 15:17.13 | [TK]D-Fender | jcape: There is no writeup. This is 2 values in zapata.conf / chan_dahdi.conf |
| 15:17.31 | [TK]D-Fender | jcape: It ain't Raw-Cat Science |
| 15:17.47 | jcape | OK, voip-info suggests this is some dark art. |
| 15:18.02 | jcape | And seeing as I'm going to be begging for money to do it, I want to know it works... |
| 15:18.05 | jcape | Thanks |
| 15:18.28 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com) |
| 15:20.33 | [TK]D-Fender | jcape: Depends on the stability of your server, tmiing, * / zaptel versions, CPU load, BUS load, etc |
| 15:20.57 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
| 15:21.11 | VoipForces | SuPrSluG: Well, you can configure your asterisk server for forward all calls from one box to the other though an iax2 trunk. |
| 15:22.01 | freezey | for some reason when i create the RINGLIST.DAT the phone doesnt take it |
| 15:22.51 | SuPrSluG | VoipForces: seems kinda odd going sip->iax->sip |
| 15:23.14 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
| 15:23.33 | SuPrSluG | [TK]D-Fender: is opensips the way to go for this/ |
| 15:24.17 | [TK]D-Fender | SuPrSluG: Is all the auth setup on the dest *? |
| 15:24.37 | SuPrSluG | yes it |
| 15:24.45 | SuPrSluG | is in production |
| 15:24.46 | [TK]D-Fender | SuPrSluG: Is it local? |
| 15:24.52 | SuPrSluG | yes |
| 15:24.56 | SuPrSluG | in house |
| 15:25.27 | [TK]D-Fender | SuPrSluG: Considered just assigning it another IP interface on the LAN? |
| 15:25.38 | [TK]D-Fender | SuPhave it take over the old IP as well. |
| 15:26.03 | VoipForces | SuPrSluG: I do it all the time. Here is a sample config: PRI <--> asterisk1 <--- IAX2 over internet ---> asterisk2 <--SIP phones--> |
| 15:26.17 | VoipForces | SuPrSluG: SIP phones are also connected to the asterisk1 |
| 15:26.42 | *** join/#asterisk SiberAIR (n=SibRphre@ip67-93-6-162.z6-93-67.customer.algx.net) |
| 15:26.47 | VoipForces | SuPrSluG: Plus on the asterisk2 I have an ATA on which a fax is connected and I can fax though asterisk2, asterisk1 via the PRI |
| 15:27.05 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.34.14) |
| 15:27.34 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 15:37.24 | freezey | Since i already built the phones using the P0S3-08-2-00.loads for the sip do i have to reset the phones in order to push the ringtones? i couldnt imagine having to do that but does anybody have any ideas? |
| 15:40.21 | *** join/#asterisk flohack (n=fhackenb@mk084020179087.a1.net) |
| 15:40.31 | flohack | Hi! I'm trying to login a queue member at the touch of a button (no phone involved) using the Originate AMI action. I'm establishing a new channel from Local/dummy@dummy/n (which does an Answer()) to the extension the agent would call when logging in. Unforunately the call from Originate always terminates at the first GotoIf. Can someone please have a look at the debug log at http://pastebin.com/m61d0bd1b (debug log and the relevant dialplan |
| 15:40.31 | flohack | logic). |
| 15:46.58 | jameswf | http://www.edgepbx.cn/shop/index.php?controller=review_info&review_id=18 <- $450 and you dont get a case |
| 15:47.40 | flohack | Any ideas concerning my Originate problem? |
| 15:50.05 | farah | anyone confortable with the command "exec" in the snmpd.conf? |
| 15:51.15 | *** join/#asterisk Assimilate (n=Assimila@72.22.242.66) |
| 15:52.35 | [TK]D-Fender | flohack: Forget debug and somehow you felt you didn't have to show us this exten, or your Originate contents... Local/dummy@dummy |
| 15:54.45 | flohack | [TK]D-Fender: Sorry, give me a second please |
| 15:54.59 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 15:56.21 | SiberAIR | anyone have a particular basis towards and SIP trunk/DID providers in the US? |
| 16:00.45 | freezey | [TK]D-Fender: you usually always have the answer to things... howcome when i reboot the phone its not picking up my RINGLIST.DAT... i checked the binary file of the .loads and i do see in there that it specifies ringtones.. i am just wondering why it wont grab |
| 16:00.49 | jameswf | ~itsplist-us |
| 16:00.50 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 16:01.20 | tzafrir_laptop | likes the name chan_mob (from asterisk-users mailing list post) |
| 16:01.23 | *** join/#asterisk ccesario (n=ccesario@linux.unialco.com.br) |
| 16:01.23 | *** join/#asterisk demiv (n=demiv@dsl-emcali-190.1.227.208.emcali.net.co) |
| 16:01.25 | SiberAIR | thanks jbot |
| 16:01.27 | flohack | the dummy context is simply: |
| 16:01.27 | flohack | [ Context 'dummy' created by 'pbx_config' ] |
| 16:01.27 | flohack | <PROTECTED> |
| 16:01.32 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-87-202.w86-215.abo.wanadoo.fr) |
| 16:01.41 | *** part/#asterisk Ericounet (n=Ericoune@ACaen-151-1-87-202.w86-215.abo.wanadoo.fr) |
| 16:01.44 | flohack | Here is the action: |
| 16:01.46 | flohack | action: Originate |
| 16:01.48 | flohack | actionid: 4329094_49# |
| 16:01.50 | flohack | callerid: "6001" <6001> |
| 16:01.50 | *** join/#asterisk chuck (n=charlie@tangocms/developer/chuck) |
| 16:01.52 | flohack | async: true |
| 16:01.54 | flohack | priority: 1 |
| 16:01.56 | flohack | context: agents |
| 16:01.58 | flohack | exten: 2001ag1 |
| 16:02.00 | chuck | How do I get that httpd enabled for configuring asterisk? |
| 16:02.00 | flohack | channel: Local/dummy@dummy/n |
| 16:02.01 | tzafrir_laptop | ~pb |
| 16:02.02 | 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, or , http://bin.cakephp.org/ |
| 16:02.02 | flohack | timeout: 30 |
| 16:02.06 | flohack | [TK]D-Fender: anything else I can provide? asterisk is v. 1.4.22 BTW. |
| 16:04.18 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:04.18 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 16:04.20 | farah | anyone confortable with the command "exec" in the snmpd.conf? |
| 16:09.18 | *** join/#asterisk neurosys (n=neurosys@c-66-229-81-161.hsd1.fl.comcast.net) |
| 16:10.23 | *** part/#asterisk farah (n=fauf@11-71-72-193.adsl.switzerland.net) |
| 16:13.49 | *** join/#asterisk hi365_m (n=hi365@213.151.43.64) |
| 16:15.27 | *** join/#asterisk flohack (n=fhackenb@mk084020161158.a1.net) |
| 16:15.53 | flohack | [TK]D-Fender: Is there a channel log? My connection (UMTS, AARRGGHHH) dropped |
| 16:18.21 | flohack | [TK]D-Fender: The last thing I saw was my message specifying the Originate action |
| 16:18.35 | [TK]D-Fender | flohack: Please pastebin the bits I requested and be complete |
| 16:20.55 | flohack | [TK]D-Fender: Sure |
| 16:21.58 | flohack | [TK]D-Fender: Here is the second pastebin: http://pastebin.com/m449aa7da |
| 16:22.15 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 16:23.08 | [TK]D-Fender | flohack: That will start your originate and then HANGUP almost instantly on it |
| 16:23.25 | [TK]D-Fender | flohack: because the channel you are calling ends immediately after the answer |
| 16:24.07 | flohack | [TK]D-Fender: Ok, what would I have to place in the dummy context to keep the channel up until the other end (the extension I'm calling) hangs up? |
| 16:24.35 | [TK]D-Fender | flohack: Something that obviously keeps the local channel GOING. |
| 16:24.47 | [TK]D-Fender | flohack: Think on this... |
| 16:29.26 | flohack | [TK]D-Fender: Sorry, but I'm stuck here. The only thing I could think of is a goto loop with a wait(xx) in between... |
| 16:30.11 | [TK]D-Fender | flohack: And did you go and do that? |
| 16:30.39 | flohack | [TK]D-Fender: Not yet, I usually think and confirm before going aheas :-) |
| 16:30.43 | flohack | ahead |
| 16:31.13 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
| 16:31.37 | [TK]D-Fender | flohack: Well that would keep that channel going, now wouldn't it? Since you are triggering this externally, what you would seem to want is the effectively jsut run some dialplan apps and having a local channel on the other end from an Originate is the only real way to do it. |
| 16:31.45 | [TK]D-Fender | flohack: This would be the way for this. |
| 16:32.05 | flohack | [TK]D-Fender: Ok, thanks a lot! I'll have a try! |
| 16:32.08 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 16:32.33 | [TK]D-Fender | flohack: SO, yes, just having the Local channel "wait" a bit is what you want, though I doubt you want or need it to stay up very long. pick a certain limit to prevent something from hanging and you should be fine. |
| 16:32.57 | flohack | [TK]D-Fender: Alright, I'll keep that in mind |
| 16:32.58 | [TK]D-Fender | flohack: If you're looking to use it for an auto logon/logonff, etc, then I'd say 10s ought to do it |
| 16:33.29 | [TK]D-Fender | flothe context you're dumping them into for that exten will probably run its course far faster and end the call by itself |
| 16:34.25 | flohack | [TK]D-Fender: Thanks! |
| 16:34.59 | *** join/#asterisk |Torg| (n=mdm@adsl-70-136-110-111.dsl.rcsntx.sbcglobal.net) |
| 16:35.46 | |Torg| | can someone give me some help with a x100p fxo that is always offhook? |
| 16:35.52 | flohack | [TK]D-Fender: That did the trick! |
| 16:36.15 | flohack | [TK]D-Fender: I'll put it on http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate |
| 16:36.42 | [TK]D-Fender | flohack: Sure, why not... |
| 16:43.52 | *** join/#asterisk jcape (n=jcape@209.120.251.66) |
| 16:46.38 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 16:49.35 | Katty | dumb question |
| 16:49.42 | Katty | why are sip trunks cheaper than getting lines from AT&T? |
| 16:51.28 | rwaite | trying to start an economics debate, are we? |
| 16:52.00 | drmessano | Katty: copper |
| 16:52.26 | Katty | i'm just curious. |
| 16:52.31 | drmessano | Actually |
| 16:52.52 | drmessano | Think of it like Gas prices |
| 16:52.56 | rwaite | Katty: probably to offset the cost of having physical "lines" |
| 16:53.01 | drmessano | We are used to gas costing X |
| 16:53.19 | carrar | the equipment that runs a SIP trunk less expensive, but the service itself is cheaper and simpler to operate |
| 16:53.32 | carrar | from: |
| 16:53.33 | carrar | http://www.lightreading.com/document.asp?doc_id=167567 |
| 16:53.39 | drmessano | So if I came out with newgas tomorrow, and I could see it for $0.75 and it worked in the same cars |
| 16:53.41 | carrar | google is your friend! |
| 16:53.46 | rwaite | it's all bush's fault, kerry would've lowered the prices! |
| 16:53.47 | drmessano | sell |
| 16:54.10 | drmessano | AT&T would still be stuck with their legacy pricing structure |
| 16:54.17 | rwaite | the price system is hard, let's go shopping |
| 16:54.45 | drmessano | You don't just wack off the 75% of overprice you have been adding for years and not collapse horribly |
| 16:55.17 | *** join/#asterisk erth64net (n=erth64ne@mail.opensourcery.com) |
| 16:55.35 | rwaite | your spending will always expand to accommodate for your income, in other words |
| 16:55.39 | drmessano | So AT&T will keep the legacy customers, and as they start to migrate away at a faster rate, they will slash in response.. eventually their pricing will even out |
| 16:56.08 | Katty | thanks (= |
| 16:56.40 | rwaite | i have a question ... why is it still called at&t |
| 16:56.54 | rwaite | do they still do telegraphs? |
| 16:57.00 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 16:57.35 | drmessano | Because if you're gonna sell the same service you have been for the last 100 years at the same inflated price, keeping the same name helps with the price recognition |
| 16:58.12 | rwaite | well, truly, prices are based upon what people are willing to pay |
| 16:58.36 | rwaite | their prices are no more inflated than my local milk company's, or wonder bread's |
| 16:58.50 | drmessano | Not true |
| 16:59.01 | [TK]D-Fender | rwaite: Wonderbread is greatly inflated. "host" is the future! |
| 16:59.10 | [TK]D-Fender | prays |
| 16:59.19 | SiberAIR | hrm |
| 16:59.31 | SiberAIR | anyone know where to get a grandsteam 286 ATA for less than voipsupply wants? |
| 16:59.47 | rwaite | Very true. If you're in a room with 99 others, and I walk in with a card full of milk bottles, and ask for 10 dollars a piece, if in the end enough people pay that price to maximise my profits - the price has been set |
| 17:00.02 | drmessano | rolls eyes |
| 17:00.23 | rwaite | If not, it will go up, or down. Can't know until you do it. |
| 17:00.55 | drmessano | I think Katty's questions was reference to what makes AT&T's pricing so much higher and not "Because" |
| 17:01.02 | rwaite | I know :) |
| 17:01.13 | rwaite | I agree with you, for the most part |
| 17:01.32 | Katty | grins |
| 17:01.33 | |Torg| | AT&T or more exactly the Bells sell many lines at a profit and quite a few at a loss. You actually PAY for that loss in the forms orf rual tax fees. And it has EVERYTHING to do with copper, or more exactly what the proders refer to as the "last mile" problem |
| 17:01.36 | Katty | is staying out of this debate. |
| 17:01.44 | |Torg| | the prices have more to do with the PUC then anything else |
| 17:02.13 | rwaite | like I said, price is a complicated concept. and then you get into cost... |
| 17:02.44 | drmessano | "The Bells"? |
| 17:02.47 | drmessano | You mean both of them? |
| 17:02.54 | |Torg| | Bell south, pacbel, swbell, etc |
| 17:03.14 | |Torg| | they still operate as seperate companies, but realistcly yes I mean both of them |
| 17:03.28 | jameswf | ~simon |
| 17:03.29 | jbot | That was utterly and completely mind numbingly painful I would rather debug windows |
| 17:03.29 | drmessano | No, actually, they are all AT&T now |
| 17:03.38 | drmessano | AT&T + Verizon |
| 17:03.40 | rwaite | I wonder, if the Iraq war had never happened, how much of the country could have been connected by fiber with the money since spent |
| 17:03.49 | rwaite | Do you think everywhere? |
| 17:04.02 | rwaite | And I'm speaking of the US, of course |
| 17:04.12 | drmessano | No |
| 17:04.33 | rwaite | A great deal? |
| 17:04.53 | jameswf | rwaite: ZERO due to the fact if the money wasn't invested in Iraq the government would invest it in $200,000 pencils. we do not have a good history of spending on communications infrastructure |
| 17:05.04 | drmessano | The lack of connectivity in the US has nothing to do with a War.. It has more to due with an economy that was ripe with inbalance due to the greed of corporations run amok due to the republican government we had in place |
| 17:05.04 | rwaite | Heh heh. |
| 17:05.28 | rwaite | Oh yes, I'm being purely technical here. AS if God came down and said "DO THIS" |
| 17:05.52 | jameswf | I imagine the money would be spent on a 4 lain bridge to hawaii |
| 17:05.56 | jameswf | *lane |
| 17:06.00 | rwaite | I guess a clearer question would be, "how much do you suppose it would cost to connect every home, business, or 'location' in the US by fiber" |
| 17:06.02 | drmessano | Mandated competition may be communistist, but I don't mind saluting Czar Obama with my FiOS |
| 17:06.13 | drmessano | communistic |
| 17:06.33 | drmessano | Think about it this way |
| 17:06.37 | angryuser | anyone from voiceroute here ? |
| 17:06.37 | |Torg| | like drmessano said, it has to do with a PUIC who controlls (well they think they do) pricing and lies that is told to hem about connectivity. Fiber Optic cables to only densly populated affluent areas is one example. Removing tarrifs on ISDN in favor of DSL is anoher |
| 17:06.43 | jameswf | to much fiber causes constipation |
| 17:07.49 | drmessano | AT&T is pushing speeds that are 1/2 to 1/4 what cable is pushing, and somehow, they are providing us a "service". So after putting up with their shitty, noisy dialup lines for years, they gave us slow, shitty internet access over the same copper |
| 17:08.05 | *** join/#asterisk RobertLaptop (n=rmiddle@m9f0736d0.tmodns.net) |
| 17:08.10 | jameswf | the senate and congress is controlled by people who have assistants to do that computer voodoo they just dont get it so the government will not advance technology |
| 17:08.13 | drmessano | Thats how AT&T rolls |
| 17:08.36 | rwaite | Ok. Well, what if my goal today was to gather up as much money as possible and lay fiber myself. How much money do I have to gather up? Enough to pay off municipalities, etc, too. |
| 17:08.45 | drmessano | Heres one for ya.. rwaite.. Since you're into the purely economic part of it |
| 17:08.54 | *** join/#asterisk CunningPike (n=arodgers@204.239.10.119) |
| 17:09.07 | rwaite | they see me rollin, providing inferior telcom services |
| 17:09.17 | jameswf | haiter |
| 17:10.05 | jameswf | AT&T is the future.... AT&T Bringing communications to the 20th century |
| 17:10.17 | drmessano | DSL capable of providing me with bandwidth to carry ONE VoIP call over a vonage ATA would run $24.95, closer to $35 in the end. |
| 17:10.23 | angryuser | i was banned from voiceroute forums 'druid' for no reason ;( |
| 17:10.31 | drmessano | Vonage would run $32 a month total |
| 17:10.38 | angryuser | i claim justice! :) |
| 17:10.51 | SiberAIR | anyone here using voicepulse? |
| 17:11.17 | drmessano | For $67 a month I can have an unlimited telephone line brought into my house, with an ass of features that Bell is still making you buy packages to accomplish |
| 17:11.49 | angryuser | drmessano: analog line ? |
| 17:11.51 | drmessano | My parents had unlimited local, unlimited long distance on ONE phone line from Bell for $90 a month |
| 17:11.53 | rwaite | true. but why do people pay that? |
| 17:11.54 | drmessano | No |
| 17:12.02 | drmessano | Read up |
| 17:12.12 | *** join/#asterisk fogo (n=Paul@69.169.132.35.provo.static.broadweave.net) |
| 17:12.31 | rwaite | looks at clock. time for lunch :D |
| 17:12.36 | angryuser | here in france we have 1⬠unlimited sip account, which is nice |
| 17:12.39 | drmessano | So thats $23 a month more for AT&T to charge me for what they would supply with an ATA to my door over DSL |
| 17:14.00 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 17:14.02 | drmessano | hang on, heres one better |
| 17:14.14 | drmessano | <drmessano> DSL capable of providing me with bandwidth to carry ONE VoIP call over a vonage ATA would run $24.95, closer to $35 in the end. |
| 17:14.20 | |Torg| | peopel pay that for ease, its why they buy bundlesd service. If it is seen as easier they do it, even it is more exansive. Take a product, raise the price 50% and five then a 25% "discount" it will be seen as somehow beter |
| 17:14.27 | drmessano | $24.95 was ALSO the price of the former AT&T callvantage service |
| 17:14.43 | drmessano | So AT&T would charge me $32 a month from one department for DSL |
| 17:14.50 | drmessano | $24.95 from another for Callvantage |
| 17:15.09 | drmessano | Then to get similar service on an analog line, $90 from another department |
| 17:15.23 | *** join/#asterisk `Sean (i=Un1x@CPE001cc031aa0d-CM0011ae925de0.cpe.net.cable.rogers.com) |
| 17:15.47 | drmessano | $35 and $32 with taxes.. so about that $67 |
| 17:16.02 | jameswf | Qwest is offering like 24M DSL but not to my house... |
| 17:16.02 | drmessano | So yeah.. I could underprice AT&T by going with... AT&T |
| 17:16.15 | drmessano | How assinine is that.. |
| 17:20.40 | angryuser | that's why i am happy to be my own provider of internet ;) |
| 17:21.06 | drmessano | Nobody is their own provider of internet |
| 17:21.13 | drmessano | It all comes from somewhere |
| 17:21.25 | angryuser | yes but the next hop is me |
| 17:22.11 | *** join/#asterisk Alan_Hicks (i=alan@cardinal.lizella.net) |
| 17:22.36 | angryuser | need to go bye all |
| 17:22.59 | Alan_Hicks | Howdy. Do I really need to compile wanpipe in order to use a Sangoma analogue card? I was in here the other day and I could have sworn some one told me the drivers were included with zaptel. |
| 17:24.43 | [TK]D-Fender | Alan_Hicks: Yes you need Wanpipe |
| 17:24.59 | [TK]D-Fender | Alan_Hicks: And what else do your Rice Crispies say to you? :p |
| 17:25.33 | Alan_Hicks | Thanks. I could have sworn it was actually you that told me otherwise, but I guess it could have been Snap, Crackle, or Pop. :^) |
| 17:25.57 | Alan_Hicks | starts writing a SlackBuild script for wanpipe. |
| 17:26.02 | SiberAIR | hey [TK]D-Fender who do you use as a SIP trunk? |
| 17:26.41 | [TK]D-Fender | SiberAIR: I don't. My clients use a mix of les.net , unlimitel.ca , etc |
| 17:30.30 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:31.16 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 17:31.20 | SiberAIR | ah canada |
| 17:31.21 | SiberAIR | hrm |
| 17:31.26 | SiberAIR | i looked up voicepulse prices |
| 17:31.32 | SiberAIR | not bad but i think i can find better |
| 17:32.34 | VoipForces | Anyone knows if there is supposed to be a change in the system command between 1.4 and 1.6? |
| 17:32.51 | VoipForces | My system calls were working in 1.4 and don't work in 1.6... |
| 17:33.10 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 17:37.09 | *** join/#asterisk Great_Anta_Baka (i=c419ffd2@gateway/web/ajax/mibbit.com/x-34f10633bbfc7275) |
| 17:38.10 | [TK]D-Fender | VoipForces: changes.txt upgrade.txt "core show application system" |
| 17:38.31 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 17:42.28 | *** join/#asterisk wiscados (n=mint@81.25.184.155) |
| 17:48.41 | jameswf | classic: http://trixbox.org/forums/trixbox-forums/open-discussion/please-vote-make-trixbox-support-chat-room |
| 17:49.26 | jameswf | VoipForces: is probably one of those ( , to |) people |
| 17:50.02 | jameswf | dislexic |
| 17:50.21 | [TK]D-Fender | jameswf: sucksess :p |
| 17:51.03 | jameswf | I am proud to announce I am in full remission and have best anorexia |
| 17:51.23 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 17:51.45 | jameswf | *beat |
| 17:52.08 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 18:01.11 | *** join/#asterisk Simon-- (n=sim@staff-nat.netnation.com) |
| 18:01.53 | Simon-- | anybody seen VNAK<->DPREP IAX2 packet storming with 1.4.21.2-1.4.22? |
| 18:03.45 | Katty | Qwell: 75 on healadin! 74 on hunter! |
| 18:03.50 | Katty | Qwell: :> |
| 18:06.44 | *** join/#asterisk feeds (n=feeds@85-135-245-223.adsl.slovanet.sk) |
| 18:11.30 | *** join/#asterisk LND (n=LND@89.193.250.227) |
| 18:15.30 | Alan_Hicks | Boy I don't like compiling these sangoma drivers.... |
| 18:15.39 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 18:16.18 | jameswf | then don't :) |
| 18:16.28 | *** join/#asterisk mohawk (n=mohawk@host217-40-110-154.in-addr.btopenworld.com) |
| 18:16.30 | Alan_Hicks | hehehe. no choice in the matter. |
| 18:16.38 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 18:16.48 | Alan_Hicks | For some reason ./Setup isn't honoring --builddir for /etc/wanpipe. |
| 18:17.17 | jameswf | if you weren;t using sangoma you wouldn;t have to worry about it :) |
| 18:19.08 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 18:22.22 | *** join/#asterisk StephenF (n=none@198.144.201.106) |
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| 18:26.58 | Alan_Hicks | What the hell? |
| 18:27.06 | [TK]D-Fender | Alan_Hicks: it prompts your during the process anyways... |
| 18:28.28 | Alan_Hicks | [TK]D-Fender: Yeah, but I'm using --silent in my build-script to create a Slackware package. |
| 18:29.01 | Alan_Hicks | Should work fine as root, but I'm building as a mortal user right now to prevent borking my box. |
| 18:29.01 | [TK]D-Fender | Alan_Hicks: then be a good little masochist and stop whining :p |
| 18:29.13 | Alan_Hicks | [TK]D-Fender: hehehe |
| 18:29.35 | Alan_Hicks | Basically, it does honor builddir for /etc/wanpipe... *if* it finds a currently existing /etc/wanpipe directory. |
| 18:29.55 | Alan_Hicks | I think as root it will create this directory if it doesn't exist, but as a mortal user it couldn't. Go figure. |
| 18:30.22 | drmessano | Anyone here support 3COM NBX? |
| 18:33.33 | *** join/#asterisk etfonhomey (n=chatzill@www2.askpri.org) |
| 18:38.37 | rwaite | meow |
| 18:39.48 | [TK]D-Fender | spays rwaite |
| 18:39.49 | *** join/#asterisk donnib (n=donnib@0x555281d0.adsl.cybercity.dk) |
| 18:41.14 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 18:41.39 | *** join/#asterisk jcape (n=jcape@209.120.251.66) |
| 18:44.52 | donnib | hey |
| 18:45.13 | donnib | anyone using a Linksys VoIP phone ? SPA-962 or SPA-942 ? |
| 18:46.09 | scooby2 | 941 and 942 here |
| 18:46.51 | donnib | do you connect the ethernet cable to WAN ? |
| 18:47.19 | donnib | 942 has both a PC and a WAN |
| 18:47.19 | *** join/#asterisk hfb (n=hfb@pool-96-229-38-185.lsanca.dsl-w.verizon.net) |
| 18:47.44 | StephenF | When specifiying a new client * setup what kind of hardware do you guys recommend as far as servers go? |
| 18:47.58 | StephenF | any specific brands that work especially well with *? |
| 18:47.58 | scooby2 | yes |
| 18:48.08 | [TK]D-Fender | donnib: Yes, you use the WAN |
| 18:48.39 | scooby2 | most anything works |
| 18:48.46 | donnib | [TK]D-Fender: thx, trying to solve my problem from earlier |
| 18:48.56 | donnib | :( |
| 18:49.00 | scooby2 | StephenF: we were using Dell but now use SuperMicro |
| 18:49.10 | StephenF | any specifc reason for the change? |
| 18:49.38 | scooby2 | We could get two SuperMicros for the price of one Dell with 4 hour support |
| 18:50.10 | StephenF | hmm |
| 18:50.21 | StephenF | Been happy with SuperMicro support? |
| 18:50.36 | scooby2 | on site spare versus having to wait for Dell if something breaks |
| 18:51.54 | scooby2 | there are many resellers out there. Most are very good support wise. |
| 18:52.01 | *** join/#asterisk dmz (n=dmz@64.203.203.232.dyn-cm-pool-64.hargray.net) |
| 18:52.38 | *** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 18:52.38 | *** mode/#asterisk [+o russellb] by ChanServ |
| 18:53.33 | *** join/#asterisk jcape (n=jcape@209.120.251.66) |
| 18:54.41 | *** join/#asterisk Akiyuki (n=jimi@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 18:54.57 | etfonhomey | scooby2, did you have any issues with the Dell's other than price? |
| 18:55.03 | Akiyuki | Carlos_PHX: You around? |
| 18:55.17 | scooby2 | etfonhomey: not at all |
| 18:56.00 | etfonhomey | scooby2, what model Dell were you using? (or did you standardize on one?) |
| 18:59.08 | scooby2 | PowerEdge 1850's we then bought two TE412p's (i think they are) and it caused the dells to crash. Tried them in a 1950 and they worked fine. |
| 18:59.27 | Akiyuki | Poweredge ftw |
| 18:59.36 | sinelaw | how can i do this? if a server goes down during a call, the call will be continued on another server |
| 18:59.50 | Alan_Hicks | just builds his own servers and doesn't blame anyone else when they fail. |
| 19:00.25 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:00.25 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 19:00.28 | sinelaw | I mean, i want to have say two servers, with one handling the calls of the other if it fails |
| 19:00.46 | *** part/#asterisk mikealeonetti (n=mikel@static-72-68-153-122.nycmny.fios.verizon.net) |
| 19:00.46 | sinelaw | btw, i don't really want, the client wants. |
| 19:01.02 | scooby2 | yeah the TE412p. We have been moving to SuperMicro for all other servers so I showed the boss that one Dell w/ 24x7x365 4 hour support was almost exactly the same price as two supermicros. We lose money if our call center goes down. |
| 19:01.42 | Alan_Hicks | sinelaw: I would think most of that could be accomplished using the linux-ha project, but I'm not sure on having the failover server seamlessly pick up calls in progress. |
| 19:02.29 | Alan_Hicks | Getting it to take-over for all future calls should be a walk in the park though. |
| 19:03.28 | scooby2 | switching the t1's is the problem I have found |
| 19:04.06 | Alan_Hicks | I've only worked with analogue lines, so I can't help you there. |
| 19:05.31 | *** join/#asterisk qdk (n=qdk@94.191.225.182.bredband.3.dk) |
| 19:05.43 | *** join/#asterisk KJ5T (n=steve@65-120-138-35.dia.static.qwest.net) |
| 19:05.47 | sinelaw | theoretically there is no reason it shouldn't be possible to continue SIP calls |
| 19:06.03 | sinelaw | but theory and practice aren't really related |
| 19:06.33 | scooby2 | in theory you should be able to keep recording calls after an atxfer |
| 19:06.58 | lmadsen | there are open bugs about this very issue on the bug tracker |
| 19:07.15 | *** join/#asterisk fskrotzki (n=fskrotzk@host.textwise.com) |
| 19:07.20 | Alan_Hicks | Looks like a good place to start looking. |
| 19:07.34 | lmadsen | if you use Monitor() and start recording a call, then perform a SIP attended transfer, the recording continues, but my understanding is the reverse is true with MixMonitor() |
| 19:07.44 | *** join/#asterisk macros73_ (n=cs@dsl093-063-232.pit1.dsl.speakeasy.net) |
| 19:08.00 | KJ5T | I am in the process of setting up special extensions so I can forward calls to my cellphone |
| 19:08.32 | KJ5T | However, I can't figure out how to send caller ID information. So that when a call is transferred it says something like "Asterisk" |
| 19:08.46 | KJ5T | So I will actually answer if it is an important call and someone else transferred it to me. |
| 19:09.10 | scooby2 | lmadsen: I know MixMonitor() does not work. I will have to try Monitor() |
| 19:10.03 | lmadsen | scooby2: I know Monitor() works (at least in my latest ABE release, although I'm pretty confident it'll be the exact same code in open source asterisk as well), and it doesn't stop recorded on a SIP attended transfer (even though I want it to in my scenario :)) |
| 19:13.06 | scooby2 | lmadsen: thanks |
| 19:14.34 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 19:16.26 | Katty | [TK]D-Fender: great. next week we have a toshiba phone system meeting )=< |
| 19:16.29 | VoipForces | FYI, the system command in 1.6 does not require escaping of comma and single quote like it used to be in 1.4 |
| 19:16.42 | Katty | [TK]D-Fender: is this company TRYING to kill itself?! )=< |
| 19:16.45 | [TK]D-Fender | Katty: Dear God... you're working for WhoreCo |
| 19:16.52 | Katty | yep |
| 19:17.07 | Katty | i don't want to setup shitty toshiba systems |
| 19:17.10 | Katty | or shitty samsung systems |
| 19:17.15 | Katty | why can't we just do asterisk?! |
| 19:17.16 | [TK]D-Fender | Katty: Why they'd practically bend over backwards for a customer! |
| 19:17.22 | [TK]D-Fender | looks around nervously |
| 19:17.38 | Katty | [TK]D-Fender: they'd sell a cheeseburger to the garden center next door if they could make a buck. |
| 19:18.31 | Deeewayne | O.O I like cheeseburgers |
| 19:19.15 | Katty | cries over her fate. |
| 19:19.25 | Katty | they're going to make me install analog systems |
| 19:19.28 | Katty | i know it. i can feel it. |
| 19:19.38 | Deeewayne | Katty: you need a job in a zoo |
| 19:19.39 | jaytee | mmmm, fried meat patty from hooved ungulate |
| 19:19.47 | Katty | my spidersense is tingling )= |
| 19:19.56 | Katty | jaytee: you hear that. i need a zoo job. |
| 19:20.08 | Katty | Deeewayne: but i already work with a bunch of monkies and asses. |
| 19:20.25 | Deeewayne | :-( |
| 19:20.33 | jaytee | Katty, we're a 5 person shop and I have to figure out how to get rid of our resident "village idiot" to make a vacancy first. |
| 19:21.32 | Deeewayne | lion's cage ? |
| 19:21.58 | jaytee | but as soon as there's an opening I'll give ya a shout out to get your resume in here. |
| 19:22.36 | jaytee | Deeewayne, I wouldn't foist bad meat like this guy on Mwangi or Shawmfa, they're nice lions. :-) |
| 19:23.51 | Katty | thanks jaytee, but i can't leave this area |
| 19:23.56 | jaytee | and Cila, Kisa and Roser, our Amur tigers are almost as picky eaters as my cats at home. |
| 19:23.59 | Katty | not while my mom is suffering from Alzheimers. |
| 19:24.13 | jaytee | Katty, that's understandable |
| 19:24.14 | Katty | i pray i'll be here another 40 years with her (= |
| 19:24.17 | Alan_Hicks | jaytee: Just go to his (her?) house and take the warning labels off everything. Let evolution handle it for you. |
| 19:24.34 | jaytee | heheehee |
| 19:25.41 | jaytee | It's very tough having a 40 something year old coworker with ADHD. And that's the Deluxe Version of ADHD. |
| 19:27.10 | VoipForces | Carlos_PHX: currently faxing on 21 channels with asterisk 1.6 |
| 19:27.39 | jaytee | recent studies of some of the cannabinoid family show that they slow the growth of tumors in the brain and the formation of the type of plaque on neurons that is thought to be the cause of Alzhiemer's. |
| 19:28.04 | jaytee | so maybe we shouldn't just "say no". |
| 19:28.17 | jaytee | maybe we should say, "Hell yeah!!!" |
| 19:29.30 | Alan_Hicks | You know, people that want to be able to legally use marijuana should just say "I want to be able to smoke marijuana in the privacy of my own home, just like I can drink a beer there." |
| 19:29.30 | Alan_Hicks | They would get a lot further than coming up with all these reasons other than getting high. |
| 19:29.56 | jaytee | Alan_Hicks, but that would mean we'd have to change to a form of government that fosters freedom and the right to pursue happiness. |
| 19:30.22 | [TK]D-Fender | jaytee: Nah, that'd never work! |
| 19:30.24 | Alan_Hicks | jaytee: Don't worry. The USA only has another 20 or 30 years before it collapses like a house of cards under the weight of its government. |
| 19:30.29 | jaytee | Capitalist oligarchies like this one are based on creating a climate of control, fear and depression in order to fuel consumer spending. |
| 19:30.35 | Alan_Hicks | When that happens, vote for me. :^) |
| 19:31.25 | jaytee | I believe that America's only hope is in abandoning capitalism and democracy in favor of a benevolent dictatorship, with me as it's leader. |
| 19:31.43 | Alan_Hicks | Yeah, that'll never work. |
| 19:31.53 | Alan_Hicks | I'd be a much better dictator. :^) |
| 19:33.08 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 19:33.50 | Carlos_PHX | VoipForces: Cool! Success! |
| 19:34.02 | jaytee | first off, all telemarketers would be rounded up and sent to gulags, all liquor stores would be forced to remain open 24/7/365. McDonalds would be forced to offer the double quarter pound Big Mac and Taco Bell would be forced to bring back the Cheesy Gordita Crunch as a permanent menu choice. |
| 19:34.39 | Alan_Hicks | See, I told you I'd make a better dictator. |
| 19:34.46 | VoipForces | so far it's steady. Had to struggle to get nvfacdetect to compile in 1.6 plus lost a good 2 hours because of the system escape change between 1.4 and 1.6 |
| 19:35.00 | jaytee | and the SciFi channel would be forced to split into two channels..... Real SciFi and LameAssWrestling&GhostHuntingCrap |
| 19:36.01 | Alan_Hicks | I'd write a new Constitution with term limits for Congressmen and Senators. Senators would be appointed by the State legislatures. SCOTUS would have term limits too, and anyone found guilty of voting for a bill (even one that doesn't pass) that is found to be unconstitutional will be guilty of treason and hung from the neck until dead. |
| 19:36.19 | Alan_Hicks | But I fully support forcing the SciFi channel to split like that. I'm behind you there 100%. |
| 19:36.43 | jaytee | oooooh, yeah. I'm starting to like the idea. Can I be one of your Capo De Tuti Capos? |
| 19:36.52 | Alan_Hicks | Yes. |
| 19:37.03 | jaytee | All Hail Emperor Alan!!!!! |
| 19:37.17 | Alan_Hicks | Congressmen get 4 two-year terms (don't have to be consequitive). |
| 19:37.29 | Alan_Hicks | I'm flexible on having or not-having term limits for Senators. |
| 19:37.54 | Alan_Hicks | I'd rather that Senators simply serve entirely at the will of the State legislatures and can be replaced for any reason or no reason at any time. |
| 19:38.10 | jaytee | term limits and really cracking down on lobbyists would do a lot towards fixing this broken machine. |
| 19:38.22 | Alan_Hicks | Also, the dollar would be based entirely on the value of 1/100th of a troy ounce of gold. |
| 19:38.55 | jaytee | Alan, good idea. after all, most employees in America today are employees at will meaning their employer can can them for any reason whatsoever. |
| 19:39.10 | Alan_Hicks | The federal reserve could not print money it could not backup with gold, and the government would be banned from borrowing money except in time of war. Also, the budget must be balanced or anyone holding an elected seat will never be elegible for re-election. |
| 19:40.23 | Alan_Hicks | Wanna build that bridge to no-where so you'll get re-elected by the people getting all that money? Sure, as long as the budget is balanced that year. If it ain't, your pork-barrel spending just cost you your entire career. |
| 19:40.56 | Alan_Hicks | One of these days I'm gonna write that Constitution and put it up on the web somewhere. |
| 19:41.21 | telnettech | Alan: How many senators would you allow to serve? 1 per state? and if so, what if the senator voted for the balance budget? wouldnt you want to say that they are fiscally responsible and should be allowed to retrun? |
| 19:41.37 | Alan_Hicks | And anyone accepting government aid would be ineligable to vote. |
| 19:41.51 | jaytee | I'm with you an all but the gold standard thing, while it sounds good to have money backed by something of real value instead of a "promise" or based on debt the idea of using gold is flawed. Russia could easily destabilize any other countries gold based economy by dumping vast reserves onto the global market. |
| 19:41.52 | *** join/#asterisk ccesario (n=ccesario@189.20.219.10) |
| 19:42.06 | Alan_Hicks | telnettech: 2 per State. I like having the Vice President cast the deciding vote if there's a tie. |
| 19:42.06 | Nugget | telnet is eeeeeeevil! |
| 19:42.54 | VoipForces | Carlos_PHX: 21 channels faxing and load average: 0.08, 0.10, 0.08 |
| 19:42.58 | Alan_Hicks | jaytee: They could do the same thing by printing vast reserves of conterfeit US currency. |
| 19:43.05 | telnettech | nugget: how do you figure that telnet is evil |
| 19:43.36 | jaytee | Nugget, telnet is a thing and therefore not evil in and of itself, it's only when wielded by someone unscrupulous and malicious that it becomes a "tool" of evil. |
| 19:43.47 | Nugget | I'm one of those openbsd, neckbeard "encrypt your swap" freaks. |
| 19:43.53 | Alan_Hicks | telnettech: As for returning, that depends. If Senators are appointed by the States, they would be exempt from dismissal on the budget issue. If they are elected, they're gone. No questions asked. |
| 19:43.58 | jaytee | Alan_Hicks, Iran has already done that in the 90's |
| 19:44.16 | Alan_Hicks | Even if they voted for a lower budget that would have been balanced, they should have fought harder for it. :^) |
| 19:44.38 | jaytee | which is why our money has color in it, a small embedded strip in it, etc. |
| 19:44.46 | Alan_Hicks | jaytee: Right. At least by using gold, you limit the amount of inflation that can occur, and who can do it to you. |
| 19:45.02 | jaytee | Platinum or germanium would be a better base |
| 19:45.15 | Alan_Hicks | Platinum I can go for. No problem with that. |
| 19:45.29 | telnettech | etfon: you on here |
| 19:45.31 | Alan_Hicks | And, oh yes... one other thing.... |
| 19:45.42 | drmessano | germanium? |
| 19:45.46 | Alan_Hicks | The US government may not give any money to the State government with strings attached. |
| 19:46.08 | jaytee | drmessano, never heard of it? rare earth element used in semiconductors |
| 19:46.13 | Alan_Hicks | In other words, no more green-mail that makes the States entirely reliant and subserviant on the federal government for cash. |
| 19:46.20 | drmessano | Like 1N34A germanium signal diodes? I have shitloads of them |
| 19:46.23 | drmessano | Am I rich? |
| 19:46.38 | drmessano | OMG, _ I AM_ |
| 19:46.43 | drmessano | YES.. RICH |
| 19:46.51 | jaytee | drmessano, rich? nope, but who knows what they'll be worth in 20 years |
| 19:47.15 | jaytee | I'd still hang on to your X-Men comics though. |
| 19:47.38 | drmessano | I do have GI-JOE #1 |
| 19:48.50 | drmessano | I actually have like 2 boxes of comic books |
| 19:48.54 | drmessano | Hmmm |
| 19:49.22 | jaytee | my mom threw out all my comics and baseball cards I was 11 and we moved. They'd be worth a fortune today. |
| 19:50.11 | hardwire | beats druid |
| 19:50.14 | drmessano | I have all my baseball and hockey bards |
| 19:50.16 | drmessano | cards |
| 19:50.17 | telnettech | jaytee: so is the training helping you in your day to day operation of your * box |
| 19:50.46 | jaytee | telnettech, somewhat |
| 19:51.38 | jaytee | I've thought of a few improvements to my dialplan but alot of things I'd based on the book and the way I manage my contexts is almost exactly how Jared recommended doing in class. |
| 19:52.24 | jaytee | I'm focusing on doing more with database and making things dynamic and adding features based on that. |
| 19:52.53 | jaytee | as well as a Windows based remote management GUI. |
| 19:53.00 | jameswf | in my application list on my blackberry I have an application called "phone" I wonder what happens if i remove it |
| 19:53.12 | jaytee | Because there is such a high demand for GUIs nowadays. |
| 19:54.37 | telnettech | jaytee: yeah i have been going back thru recent installs and changing things....I also have sent the development team a few ideas on how to make certain hospitality features work |
| 19:54.38 | *** join/#asterisk BadPacket (n=Bad@unaffiliated/badpacket) |
| 19:55.05 | jaytee | I've been working on that white paper I promised Ron for integrating dialplans with Nortel Meridian Option 11c-81c switches with Asterisk. |
| 19:55.30 | drmessano | Anyone here know if YATE or Freeswitch works with SIP TCP? |
| 19:55.37 | drmessano | ZOMG offtopic |
| 19:55.48 | drmessano | I need it for integration |
| 19:56.10 | *** join/#asterisk neurosys (n=neurosys@c-66-229-81-161.hsd1.fl.comcast.net) |
| 19:57.54 | etfonhomey | telnettech, did you figure out the weird ringing thing from befoe? |
| 19:58.03 | telnettech | no i havent |
| 20:00.08 | telnettech | et: i havent tried the canreinvite=no option cause the customer is going into a busy holiday weekend and has forbidden us to make any changes until after |
| 20:00.25 | *** join/#asterisk jtodd (n=jtodd@blob.fox-den.com) |
| 20:01.31 | telnettech | et: did you have any other suggestions? |
| 20:01.51 | *** join/#asterisk RobH (n=RobH@130-227.127-70.tampabay.res.rr.com) |
| 20:02.24 | etfonhomey | telnettech, No, just curious. |
| 20:03.46 | RobH | Hrmm, someone asked me this and I am not sure of the answer. When I call the Record() application, is there a way to tell it to record in more than one file format during its runtime? For example, I want to record 'nightmenu' in .gsm, .wav and .au all at once? |
| 20:05.24 | donnib | what does line 0: Unable to open master device '/dev/zap/ctl' mean ? i try to run genzaptelconf |
| 20:07.52 | jameswf | donnib: no zaptel loaded |
| 20:08.08 | VoipForces | RObH: after the record spawn a or multiple System call to do the conversion |
| 20:08.16 | donnib | so how do i get it loaded ? |
| 20:08.23 | *** join/#asterisk RypPn (n=Sally@rosscom.demon.co.uk) |
| 20:08.30 | RobH | VoipForces: use dialplan to make shell command line calls? |
| 20:08.46 | VoipForces | RObH: Yup works just fine |
| 20:08.49 | RobH | was hoping there was a way to do that within asterisk, but i guess i could do that to run sox and convert |
| 20:08.57 | jameswf | donnib: do you have hardware? |
| 20:09.01 | donnib | nope |
| 20:09.02 | VoipForces | RobH: asterisk -rx "show application system" |
| 20:09.10 | RobH | VoipForces: i didnt really think of that either, thanks =] |
| 20:09.11 | jameswf | then dont worry about it |
| 20:09.18 | donnib | i need the conferencing |
| 20:09.26 | donnib | and was told that i need the zaptle |
| 20:09.33 | donnib | zaptel* |
| 20:09.38 | RobH | donnib: yea, you need the ztdummy timer running |
| 20:09.41 | jameswf | modprobe zaptel; modprobe ztdummy |
| 20:09.44 | VoipForces | donnib: you need ztdummy for timing that's it |
| 20:10.05 | *** join/#asterisk Micc (n=dotirc@c-76-121-255-52.hsd1.wa.comcast.net) |
| 20:10.06 | donnib | so i need to run mobprobe zaptel ? |
| 20:10.39 | jameswf | donnib: if you use the zaptel init scrit it will work it out for you |
| 20:10.46 | jameswf | *script |
| 20:10.55 | Micc | Ok, so I figured out the problem I was having yesterday was that the aastra phones don't like to have anything in the callerid field. |
| 20:11.03 | VoipForces | RobH: I use it intensivly to run mysql queries, beats the hell of writing agi to do simple table update/insert |
| 20:11.04 | donnib | so what do i need to run exactly ? |
| 20:11.56 | Micc | but now I've got a problem where asterisk is getting into a bad state and when I show channels theres a ton of outoing lines and a bunch of people in queues. |
| 20:12.37 | Micc | and once I do a show queues it doesn't show me anything and then I can't give it any commands after that. |
| 20:13.07 | Micc | If I get a message about avoiding initial deadlock, is that bad or is that a normal thing? |
| 20:14.00 | lmadsen | Micc: you can ignore it |
| 20:14.08 | lmadsen | the deadlock was avoided |
| 20:15.22 | Micc | yes it was |
| 20:17.02 | *** join/#asterisk pramz (n=pramod@c-24-18-141-135.hsd1.wa.comcast.net) |
| 20:23.32 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 20:24.36 | Micc | Any ideas why it would get stuck in the strange state? |
| 20:25.50 | drmessano | Micc: I think that old ass version of Asterisk, pre-fixing a lot of deadlock issues, is going to keep giving you problems |
| 20:26.22 | Micc | hmmm.. |
| 20:26.55 | Micc | well I guess its time to upgrade then. |
| 20:27.06 | Micc | We're running asterisk 1.2 with some modifications. |
| 20:27.20 | donnib | if i have two subnets 10.116.x.x and 10.115.91.x is localnet=10.0.0.0/255.0.0.0 good enough ? |
| 20:27.34 | jaytee | yay, getting out of work early! be back later |
| 20:27.40 | Micc | I'm getting another virtual server to install asterisk on, I can make that one version 1.4 or 1.6 |
| 20:27.40 | Katty | jealous :< |
| 20:28.35 | Micc | Does anyone know much about this bicom stuff thats advertised on voip-info.org? |
| 20:28.45 | donnib | anyone ? |
| 20:28.48 | Micc | They look like they have some cool stuff but they don't give any prices. |
| 20:28.57 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 20:28.59 | Micc | donnib, yes you should be fine. |
| 20:29.08 | donnib | ok thx |
| 20:29.30 | Micc | donnib, you can turn on full logging and you'll see it doing the compares. |
| 20:29.44 | donnib | sip set debug ? |
| 20:29.57 | Micc | donnib, no, edit logger or logging.conf I forget the name. |
| 20:30.05 | Micc | uncomment the full line. |
| 20:30.17 | Micc | then tail -f /var/log/asterisk/full |
| 20:30.26 | Micc | and watch what it does. |
| 20:31.06 | donnib | at the moment i have this line in logger full => notice,warning,error,debug,verbose |
| 20:31.15 | donnib | it's not uncommented |
| 20:31.32 | drmessano | <Micc> We're running asterisk 1.2 with some modifications. <-- and a very old 1.2 to boot |
| 20:31.39 | drmessano | Thats the problem |
| 20:32.02 | Micc | drmessano, you have the same problems, huh? |
| 20:32.06 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 20:32.07 | drmessano | You're expecting a lot of help debugging this thing, and you're using something ancient |
| 20:32.20 | drmessano | No, but things have improved over time |
| 20:32.57 | drmessano | I probably did use that version, 2 years ago |
| 20:33.03 | Micc | drmessano, I know, I just haven't taken the time the get all the modifications written down so we can upgrade and make the same mods. |
| 20:33.24 | Micc | oh, I see what your saying now. |
| 20:33.57 | Micc | I know its old, but you guys have been a big help still. |
| 20:34.06 | StephenF | Do you guys typically run RAID 5 on your * boxes? Or maybe just RAID 1? |
| 20:34.50 | StephenF | How much do you price a normal Asterisk server at for about 25 users |
| 20:34.57 | StephenF | just the server hardware I mean |
| 20:35.14 | drmessano | heh.. trade secret.. Want a quote? |
| 20:35.20 | StephenF | lol |
| 20:35.33 | StephenF | No im just looking for industry averages |
| 20:35.39 | Micc | drmessano, Its been running fine without a hitch for the last 3 or 4 years. Its just recently we've taken on some external customers. |
| 20:35.52 | *** join/#asterisk jcape (n=jcape@209.120.251.66) |
| 20:36.02 | StephenF | Im building a server for a client and want to make sure im in the ballpark here |
| 20:36.22 | drmessano | $500 |
| 20:36.39 | StephenF | plus any interface cards they need right? |
| 20:36.47 | drmessano | ... |
| 20:37.07 | drmessano | dude, price the hardware, price the labor and time, mark it up a bit.. thats how you price something out |
| 20:37.18 | drmessano | It isnt about "What do you guys do" |
| 20:37.18 | *** join/#asterisk highzeth (n=highzeth@hoiseth.no) |
| 20:38.07 | StephenF | yup, just wanted to make sure my prices were sane |
| 20:38.31 | drmessano | Your prices are your prices |
| 20:38.37 | drmessano | There is no comparison |
| 20:38.46 | drmessano | Its based on what you decide needs to be the in the box |
| 20:42.11 | *** join/#asterisk jdnWEST (n=jdn@mx1.westparkcom.net) |
| 20:43.12 | rwaite | w00t, four day weekend |
| 20:43.13 | rwaite | ! |
| 20:46.23 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) |
| 20:51.33 | *** join/#asterisk sh0tt (n=sh0t@83.19.145.67) |
| 20:52.39 | Akiyuki | Anyone know a company that makes phone cards? |
| 20:53.36 | lmadsen | I think digium does |
| 20:54.34 | Akiyuki | That would be cool. Those kind that you scratch the little bar off ? |
| 20:54.39 | *** join/#asterisk fogo (n=Paul@69.169.132.35.provo.static.broadweave.net) |
| 20:54.40 | psykx-out | and diginum are worth supporting IMHO beacause they are pro opensource |
| 20:54.44 | lmadsen | you mean calling card? |
| 20:54.54 | psykx-out | lol |
| 20:55.02 | Akiyuki | I guess |
| 20:55.16 | Akiyuki | English isn't my first langauge. |
| 20:55.28 | Akiyuki | But yeah, you purchase the card at a store, then you scratch the back and dial the # |
| 20:55.55 | *** join/#asterisk meuserj (n=meuserj@indianalifesciences.com) |
| 20:56.26 | *** join/#asterisk telecos (n=sergio@67.166.219.87.dynamic.jazztel.es) |
| 20:58.18 | Akiyuki | Does digium make that product? |
| 21:00.01 | Qwell | Akiyuki: No, Digium makes telephony interface hardware |
| 21:01.40 | *** join/#asterisk edibrac (n=edibrac4@206.173.193.34.ptr.us.xo.net) |
| 21:02.05 | meuserj | Ok.. I'm working to migrate my asterisk configuration from a server running 1.2.13 to a server running 1.4.21.2... it's not starting and exiting with exit status 1, and no matter how much I crank up the verbosity, I don't see anything indicating what is causing the fatal error... can someone educate me? http://pastebin.com/m129bd70c |
| 21:03.00 | jameswf | meuserj: /var/log/asterisk |
| 21:03.38 | edibrac | i'm getting one irqmiss when i do zttool -- and reading up on the irqmisses, one culprit is the dma settings for my hard drives. my question is about hdparm output -- how do I know what the current udma it is using? http://pastebin.com/m74111706 |
| 21:03.58 | edibrac | it says the current one is indicated with a * but for my output it doens't say. |
| 21:04.13 | tzafrir_laptop | meuserj, is it the latest Lenny package? |
| 21:04.25 | meuserj | tzafrir_laptop: yeah |
| 21:05.43 | tzafrir_laptop | grep voicemail /etc/asterisk/modules.conf |
| 21:06.20 | tzafrir_laptop | I think you need to make voicemail_imap.so and voicemail_odbc.so noload => |
| 21:06.40 | Akiyuki | Qwell: Oh, I thought that is waht lmadsen was saying |
| 21:06.45 | tzafrir_laptop | I have no idea why I don't see the error |
| 21:06.45 | tzafrir_laptop | did you miss pasting the last line? |
| 21:08.22 | *** join/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 21:08.34 | meuserj | tzafrir_laptop: no, the last line is the last line |
| 21:08.40 | *** part/#asterisk donnib (n=donnib@0x555281d0.adsl.cybercity.dk) |
| 21:08.49 | ruben23 | hi to all |
| 21:08.57 | *** join/#asterisk lucasb (n=lbussey@office.telifon.com) |
| 21:09.10 | ruben23 | hi i have to refer something if what error log could this be... |
| 21:09.33 | ruben23 | Nov 6 14:38:48 NOTICE[7376]: rtp.c:587 ast_rtp_read: Unknown RTP codec 90 received |
| 21:09.42 | *** join/#asterisk ElCheapo (n=elcheapo@d137-186-181-17.abhsia.telus.net) |
| 21:09.43 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 21:10.07 | tzafrir_laptop | meuserj, how do you run it? |
| 21:10.11 | ruben23 | this si the output of my asterisk CLI. |
| 21:10.19 | tzafrir_laptop | can you use strace to get more clues? |
| 21:10.34 | *** join/#asterisk [netman] (n=netman@181.Red-88-17-242.staticIP.rima-tde.net) |
| 21:10.52 | meuserj | tzafrir_laptop: I'm running it with this command: /usr/sbin/asterisk -dfvvvv -p -U asterisk |
| 21:11.01 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:11.01 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 21:12.01 | tzafrir_laptop | meuserj, so next thing to do would be strace /usr/sbin/asterisk -p -U asterisk |
| 21:12.02 | ruben23 | hi anyone can help me with this.. |
| 21:12.06 | tzafrir_laptop | errr... |
| 21:12.28 | tzafrir_laptop | meuserj, so next thing to do would be strace /usr/sbin/asterisk -dvvvv -p -U asterisk |
| 21:12.42 | meuserj | tzafrir_laptop: I tried to noload voicemail_imap and voicemail_odbc, and there is no change |
| 21:14.07 | ruben23 | hi can anyone help me with this error log on asterisk CLI Nov 6 14:38:48 NOTICE[7376]: rtp.c:587 ast_rtp_read: Unknown RTP codec 90 received |
| 21:15.16 | tzafrir_laptop | Are RTP codecs numbers or bitmasks? |
| 21:17.20 | ruben23 | actually i have no idea... |
| 21:18.15 | ruben23 | you have idea with my error log on CLI. |
| 21:18.33 | tzafrir_laptop | meuserj, so how about running that strace |
| 21:18.35 | tzafrir_laptop | ? |
| 21:18.40 | meuserj | tzafrir_laptop: http://pastebin.com/m4130b4bb |
| 21:18.52 | meuserj | tzafrir_laptop: JUST pasted it |
| 21:19.40 | VoipForces | Anyone has bright ideas to mix "publicity announcements" within music on hold without doing a copy/paste job over the actual moh files? |
| 21:20.40 | C4away | is it a known issue that Playtones isn't audible during early media unless a Playback command has been issued previously? |
| 21:21.36 | tzafrir_laptop | meuserj, grep app_voicemail /etc/asterisk/modules.conf |
| 21:23.28 | meuserj | tzafrir_laptop: no output |
| 21:23.36 | *** join/#asterisk etfonhomey (n=chatzill@74-143-196-254.static.insightbb.com) |
| 21:23.44 | tzafrir_laptop | So you didn't add those two lines |
| 21:25.09 | meuserj | I added noload lines for voicemail_imap.so and voicemail_odbc.so.... didn't realize that I needed to prepend app_ to it |
| 21:25.59 | jtodd | voipforces: I'm not certain it works out of the box, but I've heard of people use streaming audio as MOH sources, and then using common radio station automation tools for that purpose. |
| 21:26.03 | meuserj | when I prepend app_, it still exits with an error, but the output is new... |
| 21:26.23 | meuserj | I'll come back if I get stuck again. Thanks. |
| 21:26.38 | VoipForces | jtodd: in theory that would work, but it's a pain in the ass to setup. |
| 21:27.23 | jtodd | voipforces: well, not too many ways to do sophisticated audio mixing without PITA. |
| 21:27.50 | *** join/#asterisk `Sean (i=Un1x@CPE001cc031aa0d-CM0014045acc3c.cpe.net.cable.rogers.com) |
| 21:28.22 | jtodd | voipforces: easiest way is to just create messages as MOH files and stick them in the directory for random play. |
| 21:28.48 | ruben23 | hi can you look into this its an output to my asterisk CLI Nov 6 14:38:48 NOTICE[7376]: rtp.c:587 ast_rtp_read: Unknown RTP codec 90 received |
| 21:29.59 | *** join/#asterisk iCEBrkr (i=icebrkr@cyberdyne.org) |
| 21:31.09 | VoipForces | jtodd: proble with that is that you might get 2-3 ties back to back same message. |
| 21:32.18 | jtodd | VoipForces: no, not if you use the ordered method. Just alphabetize your filenames, and put enough music/messgaes in the directory that it doesn't repeat within the typical span of listening. |
| 21:32.46 | jtodd | VoipForces: From musiconhold.conf: ; This plays files directly from the specified directory, no external |
| 21:32.46 | jtodd | ; processes are required. Files are played in normal sorting order |
| 21:32.46 | jtodd | ; (same as a sorted directory listing)... |
| 21:33.26 | jtodd | There is a "random" and a "sorted" method - use what makes sense. |
| 21:35.25 | ruben23 | hi anyone here have ideas what would be the changes on my asterisk configuration when i do inbound calls currently im doing outbound... |
| 21:35.50 | ruben23 | i have asterisk ver 1.4.22 |
| 21:36.39 | ruben23 | SIP + sofphones. |
| 21:38.21 | VoipForces | Boss just had e put xmas moh... yuk |
| 21:38.55 | *** join/#asterisk mchou (n=mchou@unaffiliated/mchou) |
| 21:38.59 | trogs | heh, should play 5 seconds of xmas then switch to some metal. |
| 21:39.04 | *** join/#asterisk DarkRift (n=dark@65.92.171.125) |
| 21:40.00 | mchou | not a asteris problem per se, but who do people recommend for sending and receiving faxes over ip? |
| 21:40.10 | *** join/#asterisk erth64net (n=erth64ne@mail.opensourcery.com) |
| 21:42.42 | Qwell | mchou: email. |
| 21:44.23 | mchou | Qwell: not realistic in this case |
| 21:44.45 | mchou | Qwell: business still has fax needs |
| 21:44.57 | Qwell | why over ip? |
| 21:45.09 | Qwell | get an analog line, and a fax machine... |
| 21:45.42 | puppet | mchou: fax over IP will never work 100% |
| 21:45.51 | puppet | not safe enough for companies anyway |
| 21:45.54 | mchou | this is a small business (2 people) trying to get rid of analog all toghether |
| 21:46.00 | puppet | sure for home use sure |
| 21:46.12 | *** part/#asterisk chuck (n=charlie@tangocms/developer/chuck) |
| 21:46.13 | puppet | mchou: i wouldn't do it if you req. fax in the buisness |
| 21:46.20 | puppet | mchou: if it is something you use form time to time |
| 21:46.23 | puppet | mchou: then it is no problem |
| 21:46.29 | C4away | t.38 works much better than ulaw passthrough |
| 21:46.38 | C4away | but finding a t.38 provider is the tough part |
| 21:46.45 | mchou | C4away: yup |
| 21:47.02 | drmessano | Indeed |
| 21:48.08 | ruben23 | hi anyone here have ideas what would be the changes on my asterisk configuration when i do inbound calls currently im doing outbound... |
| 21:50.52 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 21:51.09 | Akiyuki | <PROTECTED> |
| 21:51.09 | Akiyuki | <PROTECTED> |
| 21:51.19 | Akiyuki | This is the messages I see in my CLI, what could be causing this? |
| 21:51.24 | drmessano | hackers |
| 21:51.54 | voxter | Soo, I've got our main asterisk server sometimes going from 5-10% cpu all the way up to 80-90% and its not clearly evident what is causing it. Is there a way to use gdb or something to determine which component is causing the spike? I think it is related to exceeding a certain number of simultaneous calls, or transcoded calls, or something.. But I am having a hell of a time isolating it |
| 21:52.00 | drmessano | 00405A1468c5 is the IP address of the gibson |
| 21:52.17 | drmessano | voxter: FreePBX? |
| 21:52.23 | drmessano | 1.6? |
| 21:52.24 | voxter | drmessano: no freepbx there. |
| 21:52.26 | drmessano | ok |
| 21:52.27 | voxter | drmessano: 1.2 |
| 21:52.36 | drmessano | FOP? |
| 21:52.45 | voxter | drmessano: no. its a switch for all intents and purposes. |
| 21:52.49 | drmessano | Ok |
| 21:53.00 | drmessano | Flash operator panel as of late has been causing huge fits, just like that |
| 21:53.01 | voxter | drmessano: its the asterisk process gobbling up the cpu. |
| 21:53.13 | drmessano | I got that |
| 21:53.23 | voxter | drmessano: no AMI users connected |
| 21:53.27 | drmessano | ok |
| 21:53.34 | Akiyuki | Is there a way to see why this MGCP phone is resetting every few minutes? |
| 21:53.51 | Akiyuki | s/minutes/seconds |
| 21:54.07 | voxter | drmessano: im pretty sure it is related to call setup/transcoding, but its just not clearly evident, it can happen when im using 17 g729 licenses, but other times be at 5% cpu when im using 25. |
| 21:54.35 | *** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-e6ceb371ed2025be) |
| 21:55.17 | voxter | Hm, maybe its coming from translating between IAX and SIP |
| 21:57.44 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 22:01.41 | Akiyuki | What does MGCP Auditing endpoint d001@00405A1468C5 for hookstate |
| 22:01.43 | Akiyuki | mean? |
| 22:05.33 | *** join/#asterisk iCEBrkr (i=icebrkr@cyberdyne.org) |
| 22:09.56 | *** join/#asterisk RobertLaptop (n=rmiddle@96.244.48.200) |
| 22:15.26 | voxter | note to self: dont attach strace to a busy asterisk. it freaks the hell out and crashes. |
| 22:15.33 | voxter | (asterisk 1.2 at least) |
| 22:16.34 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 22:18.00 | *** join/#asterisk __Markus___ (n=markus@a89-182-141-114.net-htp.de) |
| 22:18.27 | __Markus___ | help |
| 22:18.29 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 22:18.41 | __Markus___ | Hi |
| 22:19.38 | __Markus___ | I have questions about asterisk, chan_capi and AsteriskGui |
| 22:19.47 | *** join/#asterisk UnixDawg_ (n=UnixDawg@cpe-98-28-149-231.cinci.res.rr.com) |
| 22:23.35 | *** part/#asterisk Titanous (n=titanous@unaffiliated/titanous) |
| 22:25.16 | ruben23 | hi anyone familiar with DID in inbound calls |
| 22:27.21 | *** join/#asterisk UnixDawg (n=UnixDawg@cpe-98-28-149-231.cinci.res.rr.com) |
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| 22:39.45 | WHYS | Looking for a version of * to run production in a University. Am I limiting myself by installing *NOW over a CLI version? |
| 22:40.53 | russellb | Maybe. It depends what you want to do |
| 22:40.59 | russellb | and also, you don't _have_ to use the GUI with *NOW |
| 22:41.31 | russellb | if you're running in that big of an environment, it's likely that the GUIs are not going to suit your needs |
| 22:41.50 | WHYS | Just wondering if I would break teh GUI by changing things - ODBC, adhearsion, etc |
| 22:43.24 | WHYS | <PROTECTED> |
| 22:43.56 | WHYS | OK, two (clustering) |
| 22:44.05 | *** join/#asterisk SiberAIR (n=SibRphre@pool-96-224-252-126.nycmny.fios.verizon.net) |
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| 23:09.00 | Akiyuki | yeah, thats me |
| 23:11.31 | *** join/#asterisk ElDios (n=ElDios@ip-216-105.sn2.eutelia.it) |
| 23:12.55 | ElDios | guys a question which could start flames in here ...but anyway I really think is the most precious place to ask: what is the best *automagic* PBX software between Elastix, Trixbox e AsteriskNOW and why? |
| 23:13.04 | ElDios | no flames and vegetables pls :) |
| 23:13.33 | *** join/#asterisk TOrrIeri (n=torrieri@nelug/crew/torrieri) |
| 23:17.00 | thedonvaughn | ElDios: in short, there is no best. whichever one fits your needs |
| 23:17.58 | ElDios | thedonvaughn the need is simple.. have a working, fast and simple PBX with the least crap around the conf files as possible and a good user interface |
| 23:19.06 | ElDios | the problem is, which fits *best* this need... (taking out the *fast* require pure Asterisk from scratch will probably be the answer, I know... shame on me -_-' ) |
| 23:22.15 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 23:24.32 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
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| 23:30.59 | ElDios | the fact that asterisknow is that sponsorized by Digium takes me off that product but infact from the forums notes *that* product is the one which keeps your conf file more clean and neat |
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| 23:40.25 | mm2knet | hi there, anyone can help me with my asterisk installation? xD |
| 23:40.41 | mm2knet | . /config |
| 23:43.23 | mm2knet | i have connected an anologue an one isdn phone to a fritz!box and i want them to register to my asterisk 1.4. I can call the phones from a softphone (zoiper) and also can access the voicemailbox with it, but the phones theirself can't do any of that things -.- |
| 23:48.37 | Assimilate | Happy turkey day to all who practices the sacrifice of a turkey for consumption on the last thursday of the November month. |
| 23:48.39 | *** part/#asterisk Assimilate (n=Assimila@72.22.242.66) |
| 23:49.49 | *** join/#asterisk newmember (n=chatzill@static-66-11-81-77.ptr.terago.net) |
| 23:52.34 | DrAk0 | a good softphone for mac? |
