| 00:10.55 | *** join/#asterisk zuez (i=steve@66.103.132.86) |
| 00:11.22 | zuez | Is BroadVoice generally a reputable company to deal with if I just need a TiSP with SIP termination? |
| 00:14.48 | zuez | heh, ITSP rather. |
| 00:20.30 | *** join/#asterisk CrazyTux (n=CrazyTux@ppp-70-244-43-191.dsl.hstntx.swbell.net) |
| 00:21.01 | CrazyTux | Hey guys, why would exten => test,1,MeetMe(123|d), give me an error of no application found for 1.4? |
| 00:22.24 | *** join/#asterisk mosty (n=mostyn@60-241-198-194.static.tpgi.com.au) |
| 00:31.31 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1128684591.dsl.bell.ca) |
| 00:36.35 | *** join/#asterisk adker (n=chatzill@74-47-52-122.dr02.glvv.ny.frontiernet.net) |
| 00:37.07 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@c-98-198-162-26.hsd1.tx.comcast.net) |
| 00:37.42 | *** join/#asterisk nirz (n=nir@bzq-79-181-155-4.red.bezeqint.net) |
| 00:38.40 | CrazyTux[m] | Hey guys, anyone know why application not found for 'MeetMe()' would becaused? |
| 00:39.59 | *** join/#asterisk RoyK (n=roy@ip-61-4-149-91.dialup.ice.no) |
| 00:41.12 | caio1982 | CrazyTux[m]: did you compile this 1.4 with a timing source available, like ztdummy? if not, that might be the cause, you can check if it's ok with 'make menuselect' |
| 00:41.53 | CrazyTux[m] | caio1982, let me take a look. |
| 00:42.33 | mosty | or perhaps chan_zap isn't loaded? |
| 00:45.36 | CrazyTux[m] | mosty, looks like for that dependencies have not been met, what are its dependencies? |
| 00:46.11 | mosty | zaptel |
| 00:46.31 | CrazyTux[m] | mosty, where can I pickup those drivers? asterisk.org? |
| 00:46.37 | mosty | yes |
| 00:49.21 | *** join/#asterisk knarfly (n=vtserije@c-75-74-155-198.hsd1.fl.comcast.net) |
| 00:50.20 | CrazyTux[m] | mosty, caio1982 thanks. |
| 00:54.16 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 00:55.58 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkgroup.com.au) |
| 01:00.40 | CrazyTux[m] | mosty, caio1982 anything I should look out for when installing? It configures, makes, fine, however the make install does not appear to place any files/ |
| 01:03.40 | RoyK | does digium still stick to this split license idiocity or are there any chance of asstrix going true gpl? |
| 01:05.55 | Nugget | the GPL is icky. |
| 01:10.08 | knarfly | RoyK: How much money you got....that will be the determining factor |
| 01:11.16 | RoyK | knarfly: and what do you mean with this rather stout question? |
| 01:15.47 | *** join/#asterisk kingsob (n=kingsob@HMTNON14-1242538569.sdsl.bell.ca) |
| 01:16.26 | kingsob | I have my asterisk setup so when someone dials my number, it calls my cell phone... but the caller id is set as my number, not the person who is calling me |
| 01:16.40 | kingsob | is there a way to set the calller is to the person who called me? |
| 01:16.56 | *** part/#asterisk RoyK (n=roy@ip-61-4-149-91.dialup.ice.no) |
| 01:18.44 | mosty | how is asterisk dialing your mobile? via what service? |
| 01:23.11 | kingsob | unlimitel |
| 01:23.21 | kingsob | i think its iax2? |
| 01:23.31 | mosty | you will need to ask them if they support that |
| 01:24.30 | kingsob | is there a way i could just try, and see if it works |
| 01:24.44 | kingsob | i dont even know what i woudl set tho |
| 01:24.59 | kingsob | exten => _1NXXNXXXXXX,1,SetCallerID(?????) |
| 01:26.19 | mosty | look up the SetCallerID application on voip-info.org |
| 01:26.39 | mosty | but it's unlikely to work, it depends on the provider how this is done |
| 01:28.42 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 01:29.36 | *** join/#asterisk egypcio (n=egypcio@200.150.132.61) |
| 01:29.59 | kingsob | ok cool, ill ask them |
| 01:34.47 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@ppp-70-244-43-191.dsl.hstntx.swbell.net) |
| 01:50.16 | *** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 01:51.08 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 01:52.39 | joat | with "exten => _205,1,Dial(SIP/300@192.168.2.38,60,D(3844#7775551212#)" is there a way to insert a pause between "3844#" and "777"? |
| 01:55.27 | *** join/#asterisk mltlnx (n=mltlnx@cpe-72-229-184-242.nyc.res.rr.com) |
| 01:57.05 | *** join/#asterisk becks` (n=flux_@218-173.5-85.cust.bluewin.ch) |
| 01:57.15 | becks` | hi, are multiple Allow: lines allowed in a SIP message? |
| 01:57.26 | knarfly | kingsob: I'm not sure but it sounds like you need to set a variable first with the original callerID and then assign that to the extension when it calls your cell phone |
| 01:57.26 | knarfly | otherwise your cell phone will just show the callerID of the phone which is forwarding the call |
| 01:59.06 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@c-98-198-162-26.hsd1.tx.comcast.net) |
| 01:59.37 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 02:09.11 | *** join/#asterisk mohshami (n=mohshami@86.108.90.197) |
| 02:09.32 | joat | ah! I think a comma does it |
| 02:11.04 | mohshami | hey guys, I configured an asterisk PBX and got it to communicate with an ericsson PBX using H323, the thing is, I can make a call from an extension on the ericsson PBX to an extension on asterisk, that works fine, but if I make a call to a meetme conference the call gets dropped instantly. There is nothing happening in the logs, does it have anything to do with the fact that I'm running * on a VM? |
| 02:24.24 | joat | for info, using "exten => _205,1,Dial(SIP/300@192.168.2.38,60,D(3844#,7775551212#)" with a comma between "3844#" and "777" works for TalkShoe dial-ins |
| 02:32.10 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 02:34.12 | *** join/#asterisk bkruse_home (n=kruz@76.73.154.120) |
| 02:34.12 | *** mode/#asterisk [+o bkruse_home] by ChanServ |
| 02:48.25 | NovceGuru | Has anybody toyed with the * appliance? I was wondering how the "paid" version of the webgui was (end user wants to be invloved or its cli all the way) |
| 02:49.02 | bkruse_home | NovceGuru: I have 'toyed' with it :] |
| 02:49.36 | bkruse_home | the end user gui has a ton more work into it, and its friendliness and its ability to adapt to the appliance as unit, instead of a gui in general |
| 02:49.55 | NovceGuru | Do you know if it lets you manually edit the cfgs and not blow away your changes after you make a change in the gui? |
| 02:50.09 | NovceGuru | sounds promising |
| 02:52.28 | bkruse_home | yes |
| 02:52.34 | bkruse_home | It does, it even has a file editor in the gui |
| 02:54.06 | NovceGuru | awesome |
| 02:54.52 | NovceGuru | (offtopic) I was so frustrated helping a friend out that was using DirectAdmin, you make a change in httpd.conf, he'd make one a few days later and it'd just blow away your custom config and regenerate from its own config |
| 02:55.04 | NovceGuru | was fun to track down :D |
| 02:57.13 | bkruse_home | It does not do that anymore, a lot of work has gone into keeping that functionality, as with a system like asterisk, you cannot limit it to a gui, especially for users who are used to vim and /etc/asterisk/*.conf (and ael, im not hatin :] ) |
| 03:00.51 | NovceGuru | yeah |
| 03:01.48 | NovceGuru | Looks like it'll be nice to integrate with an existing POTS system to ease into an existing setup |
| 03:03.56 | NovceGuru | My biggest "Caveat" will be taking an existing analog line on their panasonic system and trying to make it an extension, so people on the panasonic system can call "extension 401" (analog extension of the Panasonic) and it call a VOIP client |
| 03:04.00 | NovceGuru | seamlessly, :P |
| 03:04.14 | *** part/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 03:12.17 | *** join/#asterisk Kalijawan (n=Kal@static24-89-72-128.regina.accesscomm.ca) |
| 03:13.31 | *** join/#asterisk kand (n=kanderso@139.148.188.72.cfl.res.rr.com) |
| 03:17.38 | *** join/#asterisk LeddyHM (i=hidden-u@polar.artica.net) |
| 03:20.19 | Kalijawan | does anyone know if there is a good way to interrupt in process calls and then transfer them? |
| 03:23.20 | kand | Kalijawan: depending on what you want to do features.conf has a blindxfer option |
| 03:25.15 | Kalijawan | well, i think what i would like to do is from the CLI (or with a py script) is transfer a call in-progress |
| 03:25.39 | Kalijawan | its actually for my electronics final project, what i want to do is make a voice follow you though a house from speaker phone to speaker phone :) |
| 03:26.19 | kand | Kalijawan: There is a usefull application ChannelRedirect in 1.4 |
| 03:27.18 | kand | Kalijawan: how about the system tracking the individual sends sip signaling (masq. as the phone) to transfer from intercom to intercom.... :) |
| 03:28.57 | Kalijawan | well, there is an RF tag on a person and then that gets sent back to the main server and then a py script runs on asterisks |
| 03:29.16 | Kalijawan | and then it calls the next phone in advance and the phone is modified to pickup when it phone rings. |
| 03:29.44 | Kalijawan | hehe, sorry is that what you're kind of suggesting? whats masq mean? |
| 03:29.47 | Kalijawan | hehe |
| 03:30.09 | kand | masqurading (I cant spell)... Cool project tho! Have you looked into the asterisk manager |
| 03:30.16 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@c-98-198-162-26.hsd1.tx.comcast.net) |
| 03:31.37 | Kalijawan | no |
| 03:31.38 | kand | Kalijawan: Sounds like it would probably fit the bill, here is an example on processing a transfer: http://www.voip-info.org/wiki/index.php?page=Asterisk+manager+Example%3A+Transfer |
| 03:31.52 | *** join/#asterisk dickyjoe (n=richardl@124.149.56.181) |
| 03:32.03 | kand | Kalijawan: Think of it like a call management API (sortof) |
| 03:32.33 | dickyjoe | Hello all, Richard from Wodonga, Australia here. |
| 03:33.29 | kand | Kalijawan: This will get you started, examples on use for lots of languages. http://www.voip-info.org/wiki-Asterisk+manager+API |
| 03:33.41 | kand | Hey! Orlando, Fl |
| 03:34.01 | dickyjoe | I have a bit of an issue I am having trouble fixing. Thought I would post it here... |
| 03:34.36 | Kalijawan | okay cool, thanks kand, i think i will look into that :D |
| 03:34.40 | kand | np |
| 03:34.54 | Kalijawan | does anyone do TTS with anything other than festival? |
| 03:35.12 | dickyjoe | I am trunking my own personal asterisk box (debian 4, lastest zap, asterisk) trying to setup incoming IAX2 trunking from free world dialup |
| 03:35.22 | dickyjoe | I get the following error |
| 03:35.34 | dickyjoe | on an incoming call.. |
| 03:36.06 | dickyjoe | [Dec 16 13:53:07] NOTICE[6550]: cryptostub.c:42 stub_ast_key_get: Crypto support not loaded! |
| 03:36.06 | dickyjoe | [Dec 16 13:53:07] WARNING[6550]: chan_iax2.c:5189 authenticate_verify: requested inkey 'freeworlddialup' for RSA authentication does not exist |
| 03:36.06 | dickyjoe | [Dec 16 13:53:07] NOTICE[6550]: chan_iax2.c:7702 socket_process: Host 192.246.69.186 failed to authenticate as iaxfwd |
| 03:36.40 | dickyjoe | as far as i can see rsa is loaded, no compile issues... |
| 03:37.04 | kand | It would seem you are missing the keys, permissions are missing or module isn't...nm check /var/lib/asterisk/keys |
| 03:37.40 | dickyjoe | i did check the keys were there and even redownloaded them from fwd |
| 03:38.32 | kand | what about permissions, can asterisk read them? According to your error asterisk doesnt know the exist. |
| 03:38.35 | dickyjoe | is there a console command to check rsa keys from the cli? |
| 03:39.19 | kand | not that I know off |
| 03:39.27 | dickyjoe | what chmod should i pass on the files> |
| 03:40.06 | kand | dickyjoe: just for testing/now try 777 |
| 03:40.18 | dickyjoe | k |
| 03:40.27 | kand | dickyjoe: narrow it down later |
| 03:40.49 | dickyjoe | they both were -rw-r--r-- |
| 03:40.59 | dickyjoe | now are -rwxrwxrwx |
| 03:41.22 | *** join/#asterisk egypcio (n=egypcio@200.150.132.61) |
| 03:41.26 | kand | dickyjoe: then that problem was not the problem but give it a shot |
| 03:42.15 | dickyjoe | kand: yeah same same |
| 03:42.41 | *** join/#asterisk Wilddev (n=chris@ns.wilddev.net) |
| 03:43.02 | dickyjoe | asterisk wont compile without openssl will it? |
| 03:43.35 | Wilddev | can anyone help me with why I dont get a dialtone from my iaxy? |
| 03:44.03 | Kalijawan | you configured the iax.conf? |
| 03:44.27 | kand | dickyjoe: oh, I believe it would. If you have 1.4 go to your sources and run 'make menuselect' |
| 03:44.28 | Wilddev | yes |
| 03:44.49 | Wilddev | I can see the iaxy receives calls due to the orange light blinking |
| 03:44.57 | Kalijawan | usually you won;t get a dial tone if the phone is not registering |
| 03:45.01 | Wilddev | just the phone never rings or gets dialtone |
| 03:45.13 | Wilddev | the phone works on my tdm400p card fine |
| 03:45.23 | Wilddev | its registered, I checked |
| 03:45.36 | kand | Kalijawan: then 8. Resource Modules and make sure res_crypto is available and selected. |
| 03:46.10 | Wilddev | wierd things is it worked one day then the next time I went to use the phone it didnt work anymore |
| 03:46.19 | kand | g/Kalijawan/dickyjoe/s |
| 03:47.43 | Wilddev | Kalijawan: by register you mean it shows up with iax2 show registry? |
| 03:48.13 | *** join/#asterisk Maliuta (n=nikolai@ppp214-92.static.internode.on.net) |
| 03:48.31 | Kalijawan | yeah |
| 03:48.41 | dickyjoe | kand: i ran make menuselect and it said "install ncurses to use the menu interfaces" |
| 03:49.15 | Wilddev | hmm, it shows with iax2 show peers, but not registry |
| 03:49.55 | kand | dickyjoe: Ok, try running 'module reload res_crypto.so' from the asterisk cli |
| 03:50.01 | Kalijawan | maybe you got the case wrong on one of the id's i had that happen at work before, because some things are case sensitive and some things aren't |
| 03:51.15 | dickyjoe | ok kand getting somewhere... |
| 03:51.26 | kand | dickyjoe: At the moment I believe if you didn't have openssl then this resource wouldn't compile and you would have this issue. Actualy now that I look back at your error I see Crypto support not loaded! |
| 03:51.30 | kand | DOH! |
| 03:51.45 | dickyjoe | it says... |
| 03:51.50 | dickyjoe | *CLI> module reload res |
| 03:51.51 | dickyjoe | res_adsi.so res_features.so res_indications.so res_musiconhold.so res_smdi.so |
| 03:51.51 | dickyjoe | *CLI> module reload res_ |
| 03:52.02 | dickyjoe | its missing as you just said |
| 03:52.46 | kand | Just install openssl with your favorite package manager then: make clean; ./configure; make; make install |
| 03:53.50 | kand | should be good to go. If you install ncurses you can use 'make menuselect' between ./configure and make to controll in a sudo-graphical way what is going to be compiled |
| 03:54.19 | dickyjoe | ok |
| 03:54.28 | kand | just remeber you need to ./configure so it see the new dependencies are satisfied. |
| 03:54.46 | Wilddev | Kalijawan: I am using freepbx to configure it |
| 03:55.24 | dickyjoe | is ./configure an asterisk thing or a linux thing? |
| 03:55.45 | Kalijawan | oh |
| 03:55.50 | dickyjoe | asterisk |
| 03:55.57 | dickyjoe | why is the file put there> |
| 03:56.42 | kand | dickyjoe: linux program compiling thing |
| 03:56.53 | dickyjoe | ok |
| 03:57.34 | dickyjoe | i think that becuase it compiles so fast and i don't know how to read the install log, i miss the output... |
| 03:57.39 | kand | dickyjoe: It is put there becuase it contains info on how to confiure the make process specificly for asterisk (or whatever program you are compiling) |
| 03:57.48 | dickyjoe | i'm learning though... |
| 03:58.00 | dickyjoe | thanks for all the help.. |
| 03:58.21 | kand | dickyjoe: That is why the made menuselect, it gives you a graphical means to see and controll the results of configure. np, good luck! |
| 04:04.25 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@c-98-198-162-26.hsd1.tx.comcast.net) |
| 04:04.55 | dickyjoe | when i try and run 'make menuselect' it tells me i need ncurses but i already have it |
| 04:04.57 | dickyjoe | hmmmm |
| 04:06.40 | *** join/#asterisk SwK (n=SwK@user-24-214-55-149.knology.net) |
| 04:08.05 | CrazyTux[m] | Hey guys |
| 04:08.08 | CrazyTux[m] | Ok, ztdummy is loaded |
| 04:08.11 | CrazyTux[m] | modprobe ztdummy |
| 04:08.15 | CrazyTux[m] | asterisk reloaded.... / restarted |
| 04:08.18 | Wilddev | Kalijawan: do you use IAX2/<extension> in your extensions files for these things? |
| 04:08.31 | CrazyTux[m] | now, a simple exten => 10,1,Playback(auth-thankyou), has no audio? |
| 04:08.33 | CrazyTux[m] | Any idea what is going on? |
| 04:11.40 | ectospasm | CrazyTux[m]: does the console say it's playing the file? |
| 04:11.48 | CrazyTux[m] | ectospasm, yes |
| 04:12.19 | CrazyTux[m] | ectospasm, I don't think it's a NAT issue, normal call audio is fune |
| 04:12.22 | CrazyTux[m] | s/fune/fine/ |
| 04:13.28 | CrazyTux[m] | ectospasm, any thoughts? |
| 04:13.39 | ectospasm | I'm thinking (-; |
| 04:13.48 | CrazyTux[m] | Asterisk 1.4.9 |
| 04:14.04 | CrazyTux[m] | 1.4.7.1 version zaptel |
| 04:14.08 | ectospasm | I dunno, I don't think I've troubleshot an IAX2 problem in a while |
| 04:14.28 | ectospasm | What kind of endpoint are you using to listen to the playback? |
| 04:15.13 | CrazyTux[m] | ectospasm, pap2t |
| 04:15.17 | CrazyTux[m] | ectospasm, PSTN |
| 04:15.23 | CrazyTux[m] | ectospasm, etc |
| 04:15.56 | ectospasm | pap2t? I'm not familiar with that |
| 04:16.26 | CrazyTux[m] | ectospasm, linksys |
| 04:16.55 | ectospasm | I assume it uses SIP... is it behind a NAT? |
| 04:17.17 | dickyjoe | what cut of linux do people here prefer when running asterisk |
| 04:17.32 | ectospasm | I've got it running on Ubuntu Server right now |
| 04:18.28 | CrazyTux[m] | ectospasm, endpoint is behind a nat, yes SIP. |
| 04:18.52 | CrazyTux[m] | ectospasm, however, PSTN (is not behind a nat) |
| 04:18.52 | ectospasm | but a call into the PSTN to that 10 extension exhibits the same behavior? |
| 04:18.58 | CrazyTux[m] | ectospasm, yes |
| 04:19.22 | ectospasm | What happens if you use a call file to create a call to the PSTN, which does a playback? |
| 04:20.12 | CrazyTux[m] | ectospasm, so outbound audio? |
| 04:20.43 | ectospasm | that should test whether it happens for both inbound and outbound call |
| 04:20.57 | CrazyTux[m] | ectospasm, same result |
| 04:21.14 | ectospasm | Actually, if you call into your system, does it play an IVR? Can you get to voicemail, does that play back any audio? |
| 04:22.19 | CrazyTux[m] | ectospasm, from the looks, like all application audio is dead |
| 04:22.23 | CrazyTux[m] | ectospasm, let me try ivr |
| 04:22.45 | *** join/#asterisk osiris (n=osiris@c-71-205-29-230.hsd1.mi.comcast.net) |
| 04:22.57 | ectospasm | any reason why you're using Asterisk 1.4.9 instead of the current version? |
| 04:23.53 | CrazyTux[m] | no |
| 04:24.05 | CrazyTux[m] | figured newer? |
| 04:24.14 | CrazyTux[m] | i was running 1.4.4 |
| 04:24.53 | ectospasm | The latest at this moment is 1.4.15 |
| 04:25.11 | ectospasm | check downloads.digium.com to be sure |
| 04:25.17 | CrazyTux[m] | latest stable? |
| 04:25.19 | ectospasm | that's http://downloads.digium.com |
| 04:25.27 | ectospasm | latest stable |
| 04:25.52 | CrazyTux[m] | oh shit |
| 04:26.19 | ectospasm | you can use subversion to get it, svn co http://svn.digium.com/svn/asterisk/tags/1.4.15 asterisk-1.4.15 ... |
| 04:29.16 | CrazyTux[m] | I wonder where I got the impression |
| 04:29.19 | CrazyTux[m] | 1.4.9 was stable |
| 04:29.19 | CrazyTux[m] | lol |
| 04:29.20 | CrazyTux[m] | wow |
| 04:29.22 | CrazyTux[m] | I feel stupid. |
| 04:29.46 | Wilddev | isnt 1.2 the stable branch and 1.4 the dev branch? |
| 04:30.00 | ectospasm | Wilddev: no |
| 04:30.09 | ectospasm | 1.4 has been considered stable for some time |
| 04:30.16 | Wilddev | oh? how long? |
| 04:30.18 | ectospasm | 1.2 is no longer actively being developed. |
| 04:30.32 | ectospasm | I'd like to say since 1.4.0, but I'm not that certain |
| 04:30.38 | Wilddev | aha ok |
| 04:30.39 | ectospasm | Probably for at least 8 months |
| 04:30.54 | Wilddev | well that makes me feel better about using it |
| 04:30.57 | ectospasm | 1.2 will still receive security fixes |
| 04:31.12 | Wilddev | at least if I can figure out why my Iaxy isnt working |
| 04:32.33 | ectospasm | my IAXy works, though I don't use it often |
| 04:33.12 | Wilddev | well mine appears to accept calls from the blinking orange light, just the phone never rings or gets a dialtone |
| 04:33.25 | Wilddev | the phone itself works fine on my tdm400 |
| 04:35.08 | ectospasm | Wilddev: does the blue light ever come on? |
| 04:35.18 | Wilddev | yes, its on now |
| 04:36.13 | Wilddev | only things I'm finding thru google suggest the hardware might be broken :-( |
| 04:38.25 | ectospasm | does anything show up in the console when you take it off hook, or try dialing it? |
| 04:39.41 | Wilddev | nothing when I take the phone off hook, but when I dial the extension, iax2 debug shows the incoming call and extension |
| 04:44.06 | CrazyTux[m] | ectospasm, ok I reverted back to * version 1.4.4 where everything was working previously, all good and dandy, without zaptel driver |
| 04:44.13 | CrazyTux[m] | ectospasm, however same problem, with zaptel? |
| 04:44.41 | ectospasm | what version of zaptel are you using? |
| 04:44.48 | ectospasm | 1.4.7.1? |
| 04:45.02 | CrazyTux[m] | 1.4.7.1 |
| 04:45.03 | CrazyTux[m] | yea |
| 04:45.06 | CrazyTux[m] | let me guess, EDGE? |
| 04:45.46 | ectospasm | so, use Asterisk 1.4.15... I cannot vouch for zaptel 1.4.7 or later working with any Asterisk version prior to 1.4.14... |
| 04:46.17 | ectospasm | It may work, but you may be seeing that it doesn't work... |
| 04:46.43 | CrazyTux[m] | ectospasm, so 1.4.7.1 works with 1.4.14? |
| 04:46.48 | CrazyTux[m] | ectospasm, and MAY work with 1.4.15 |
| 04:46.59 | ectospasm | no... it works with 1.4.14 or later |
| 04:47.31 | ectospasm | But you're trying to use it with 1.4.9 and/or 1.4.4, which isn't working for you. |
| 04:47.32 | CrazyTux[m] | so 1.4.(4|9) should be fine with the driveR? |
| 04:47.37 | ectospasm | NO |
| 04:47.39 | CrazyTux[m] | ectospasm, 1.4.4 worked without it |
| 04:47.43 | CrazyTux[m] | let me try 1.4.15 |
| 04:47.45 | ectospasm | without Zaptel? |
| 04:47.52 | CrazyTux[m] | yes |
| 04:48.02 | ectospasm | well, your TDM400 will be worthless without the zaptel drivers... |
| 04:48.43 | CrazyTux[m] | ectospasm, oh, no hardware |
| 04:48.52 | CrazyTux[m] | ectospasm, I'm just trying to setup zaptel, for meetme |
| 04:48.55 | CrazyTux[m] | ectospasm, using ztdummy |
| 04:49.14 | ectospasm | Ah |
| 04:49.25 | ectospasm | How are you connecting to the PSTN? |
| 04:49.34 | ectospasm | Through an VoIP provider? |
| 04:49.54 | CrazyTux[m] | ectospasm, yea |
| 04:51.07 | ectospasm | So neither an internal SIP phone nor your VoIP can do audio? ztdummy should have zero effect on that... |
| 04:51.17 | ectospasm | I mean playback audio |
| 04:51.35 | ectospasm | You call in from the PSTN, and it doesn't play your IVR, or any other audio, right? |
| 04:51.41 | dickyjoe | does anyone know of any ata's that support decadic dialling? |
| 04:52.22 | CrazyTux[m] | ectospasm, well, it's definitely not a NAT/Audio issue, as far as that -- incoming/outgoing calls are fine |
| 04:52.34 | CrazyTux[m] | ectospasm, just no audio on calls dealing with applications, playback, ivr, etc. |
| 04:52.41 | RypPn | so its just hold music? |
| 04:52.53 | CrazyTux[m] | ectospasm, and I had this working previous to installing zaptel stuff. |
| 04:53.06 | CrazyTux[m] | RypPn, was that to me? |
| 04:53.30 | RypPn | CrazyTux[m]: try adding internal_timing = yes under [options] in asterisk.conf |
| 04:55.22 | CrazyTux[m] | RypPn, hmm whats that do? that seemed to do the trick |
| 04:55.28 | RypPn | ;) |
| 04:55.54 | RypPn | tells asterisk to use ztdummy for timing |
| 04:56.18 | CrazyTux[m] | RypPn, oh :) |
| 04:56.27 | CrazyTux[m] | RypPn, thanks! |
| 04:56.32 | RypPn | np |
| 04:57.09 | RypPn | just remember to remove it if you get zaptel hardware at some point :) |
| 04:57.21 | *** join/#asterisk shyam_k (i=shyam@59.91.255.237) |
| 04:58.48 | CrazyTux[m] | RypPn, will do |
| 05:02.09 | CrazyTux[m] | RypPn, hmm, weird, I reverted back to 1.4.9 to see if it'd work with that version, and it didn't seem to, reverted back to 1.4.4, and now its broken again hmmm |
| 05:05.00 | RypPn | I'm using 1.4.15 with zaptel 1.4.7.1 on 2 setups, one pure ztdummy and the other with zaptel hardware, no issues on either |
| 05:05.14 | CrazyTux[m] | RypPn, I'm just going to revert to 1.4.15 |
| 05:05.29 | CrazyTux[m] | zaptel 189928 4 zttranscode,ztdummy, lsmod shows that they are both loaded. |
| 05:05.56 | RypPn | how about crc_ccitt ? |
| 05:06.29 | CrazyTux[m] | in asterisk.conf ? |
| 05:06.40 | RypPn | no, a kernel module |
| 05:06.45 | *** join/#asterisk [gnubie] (n=[gnubie]@cm174.gamma179.maxonline.com.sg) |
| 05:06.53 | RypPn | I always see it attahced to zaptel |
| 05:06.56 | RypPn | attached* |
| 05:07.00 | CrazyTux[m] | RypPn, yea its loaded. |
| 05:07.05 | CrazyTux[m] | crc_ccitt 2400 1 zaptel |
| 05:07.46 | CrazyTux[m] | compiling 1.4.15 hopefully no problems, with this one :) |
| 05:08.10 | RypPn | what version of zaptel are you using? |
| 05:08.45 | CrazyTux[m] | 1.4.7.1 |
| 05:12.02 | RypPn | you could run zttest to check the timing, best done when the system is quiet tho |
| 05:12.18 | CrazyTux[m] | RypPn, yea this is 100% dev |
| 05:12.30 | CrazyTux[m] | RypPn, hmm 1.4.15 does the same thing |
| 05:14.10 | CrazyTux[m] | RypPn, how long does the test take normally? |
| 05:14.20 | ectospasm | zttest will run indefinitely |
| 05:14.21 | RypPn | you cancel out manually |
| 05:14.24 | CrazyTux[m] | oh |
| 05:14.26 | CrazyTux[m] | alrighty |
| 05:14.39 | CrazyTux[m] | Best: 0.000 -- Worst: 100.000 -- Average: 100.000000, Difference: 100.000000 |
| 05:14.39 | ectospasm | but don't kill it after only a couple of scores... |
| 05:14.45 | CrazyTux[m] | that dosent look very good? |
| 05:15.00 | ectospasm | run it for at least 20 cycles |
| 05:15.34 | CrazyTux[m] | that was for like a good few minutes |
| 05:15.39 | CrazyTux[m] | dosent look like its doing anything? |
| 05:16.08 | ectospasm | Actually, come to think of it I don't know if zttest works with just ztdummy |
| 05:16.31 | ectospasm | at work we usually use it to see how timing is going with our digital cards... |
| 05:16.31 | RypPn | it does |
| 05:17.24 | CrazyTux[m] | yea everything |
| 05:17.27 | CrazyTux[m] | results in 0 passes? |
| 05:18.04 | RypPn | http://rafb.net/p/Tf8cCh81.html |
| 05:19.03 | CrazyTux[m] | Yea mine, makes 0 passes |
| 05:19.08 | CrazyTux[m] | Any idea on what would block it? |
| 05:19.16 | CrazyTux[m] | lsmod shows the modules loaded? |
| 05:20.00 | RypPn | for reasons I have still to fathom, timings are better when usb is enabled on the board and I usually built wcusb in zaptel just in case |
| 05:20.34 | RypPn | that could be rubbish tho |
| 05:21.17 | ectospasm | yeah, it uses the kernel as a timing source in 2.6 kernels |
| 05:21.27 | ectospasm | if you're still running a 2.4 kernel, then the USB applies |
| 05:21.53 | ectospasm | (and for 2.6 kernels prior to 2.6.15 or something like that) |
| 05:22.07 | RypPn | why would the timings noticeably improve when usb is enabled on the motherboard tho under 2.6? |
| 05:22.38 | ectospasm | That's a good question, I would think that wouldn't be an issue... |
| 05:22.58 | RypPn | I went through a phase of disabling motherboard devices not in-use thinking it was helping |
| 05:23.10 | *** join/#asterisk Idle (n=brian@S010600a024969312.ed.shawcable.net) |
| 05:23.21 | Idle | http://voipsa.org/pipermail/voipsec_voipsa.org/2007-December/002522.html |
| 05:23.27 | Idle | anyone seen that trixbox vuln yet? |
| 05:25.07 | ectospasm | Interesting |
| 05:25.12 | Qwell | very |
| 05:25.18 | ectospasm | Another reason one shouldn't use trixbox |
| 05:25.23 | Qwell | well, asterisknow doesn't have a rootkit |
| 05:25.31 | ectospasm | heheheh |
| 05:25.55 | ectospasm | I can't remember, does Switchvox Free Edition phone home at all? |
| 05:26.35 | Qwell | phoning home and running arbitrary commands are two entirely different things |
| 05:26.39 | Qwell | having said that - I don't know |
| 05:26.45 | ectospasm | true enough |
| 05:26.56 | CrazyTux[m] | RypPn, should there be a need to recompile zapatel, from switching * versions? |
| 05:27.03 | Qwell | well, that, and not telling people that you're doing it |
| 05:27.37 | ectospasm | And to think, I almost got a job with Fonality! |
| 05:27.49 | Qwell | I'm sorry to hear that |
| 05:27.53 | RypPn | CrazyTux[m]: I wouldn't have said so, but you could try removing and reloading those modules in the kernel to see if it makes any difference first |
| 05:28.23 | ectospasm | well, I didn't get it, because they needed someone who could be in LA like tomorrow |
| 05:28.35 | CrazyTux[m] | RypPn, already tried that a bunch |
| 05:28.36 | ectospasm | it was way too fast for me |
| 05:28.44 | Qwell | ectospasm: was this about 18 months ago? heh |
| 05:28.52 | Qwell | more than that now, maybe |
| 05:28.57 | ectospasm | actually, yeah |
| 05:28.59 | Qwell | heh |
| 05:29.03 | Qwell | I know the guy who got that job :p |
| 05:29.07 | ectospasm | like May of 2006 |
| 05:29.14 | Qwell | sounds about right, yeah |
| 05:29.15 | ectospasm | heheheh |
| 05:29.55 | Qwell | that's kinda funny |
| 05:30.25 | ectospasm | I got a really bad feeling from that, like they were rushing too fast to get me in there |
| 05:30.37 | Qwell | no comment |
| 05:30.46 | ectospasm | heheheh |
| 05:30.59 | ectospasm | I like where I'm at now, though (-; |
| 05:31.15 | Qwell | erm, who are you again? |
| 05:31.24 | ectospasm | Trey, in Support |
| 05:31.28 | Qwell | (sorry, I keep forgetting) |
| 05:31.30 | Qwell | ectospasm: see msg :D |
| 05:40.30 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-3.dllstx.fios.verizon.net) |
| 05:42.54 | *** part/#asterisk shyam_k (i=shyam@59.91.255.237) |
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| 07:11.15 | *** join/#asterisk ZX81 (n=ZX81@202.20.97.211) |
| 07:12.31 | ZX81 | ~ping |
| 07:12.32 | jbot | pong |
| 07:14.36 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
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| 07:45.33 | *** part/#asterisk [gnubie] (n=[gnubie]@cm174.gamma179.maxonline.com.sg) |
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| 07:47.37 | troy- | i currently have asterisk running and dundi seems to be listening on port 4520; since i wont be using it can i disable it? |
| 08:01.39 | *** join/#asterisk smooth_penguin (n=smooth_p@59.95.55.34) |
| 08:03.23 | smooth_penguin | is it possible to send or recieve a phone signal / dialtone through a cordless phone from its base, I should be able to extract the phone signal from my cordless reciever and plug it into my modem etc ? |
| 08:04.05 | smooth_penguin | the cordless phone would be the substitute for wiring |
| 08:04.38 | smooth_penguin | but can I take the dialtone out and plug it into my asterisk server or another phone? |
| 08:20.57 | *** join/#asterisk mosty (n=mostyn@60-241-198-194.static.tpgi.com.au) |
| 08:53.29 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 08:59.33 | *** join/#asterisk ZX81 (n=ZX81@202.20.97.211) |
| 09:01.00 | ZX81 | ~ping |
| 09:01.00 | jbot | pong |
| 09:02.56 | *** join/#asterisk insane_1 (n=jimbean@cpe-76-187-239-242.tx.res.rr.com) |
| 09:03.07 | insane_1 | hey guys, anyone around? |
| 09:03.36 | ZX81 | maybe |
| 09:03.43 | insane_1 | :> |
| 09:03.53 | insane_1 | are you well versed in the ways of asterisk / zaptel |
| 09:03.56 | insane_1 | i'm praying yes |
| 09:03.57 | ZX81 | :) |
| 09:03.59 | ZX81 | why |
| 09:04.07 | insane_1 | i'm having hell with these zaptel drivers |
| 09:04.09 | ZX81 | bit of a leading question |
| 09:04.12 | ZX81 | what's up? |
| 09:04.39 | insane_1 | short answer: the most famous error message by far i've seen |
| 09:04.41 | insane_1 | ZT_CHANCONFIG failed on channel 1: Invalid argument (22) |
| 09:04.41 | insane_1 | Did you forget that FXS interfaces are configured with FXO signalling |
| 09:04.41 | insane_1 | and that FXO interfaces use FXS signalling? |
| 09:04.53 | ZX81 | and are they right? |
| 09:04.56 | insane_1 | EVEN THOUGH it's done right in zaptel.conf |
| 09:05.05 | ZX81 | in zapata.conf and zaptel.conf |
| 09:05.10 | ZX81 | tried swapping them? |
| 09:05.15 | insane_1 | fxsks=1-8 |
| 09:05.24 | insane_1 | how doyou mean swap |
| 09:05.27 | ZX81 | tried fxoks=1-8? |
| 09:05.27 | insane_1 | do you* |
| 09:05.51 | insane_1 | it's for a FXO module |
| 09:05.55 | insane_1 | it uses FXS singaling |
| 09:06.02 | insane_1 | right? |
| 09:06.04 | ZX81 | should do yes |
| 09:06.09 | ZX81 | try it first though |
| 09:06.12 | ZX81 | just make the change |
| 09:06.16 | ZX81 | then type ztcfg -v |
| 09:06.24 | insane_1 | whoa |
| 09:06.25 | ZX81 | also is this a tdm2400p? |
| 09:06.38 | insane_1 | i have 2 separate cards in there |
| 09:06.42 | ZX81 | ah |
| 09:06.44 | insane_1 | 1 is an 8 channel fxs |
| 09:06.50 | insane_1 | other is a 8 channel fxo |
| 09:06.52 | ZX81 | likely 1-8 are not the ones you think |
| 09:07.00 | insane_1 | also it didn't error out that time |
| 09:07.02 | insane_1 | interesting |
| 09:07.04 | ZX81 | so maybe the fxs one is coming up first |
| 09:07.20 | ZX81 | so fxs=card 1 and fxo=card2 |
| 09:07.31 | insane_1 | that's true i hadn't though of that |
| 09:07.40 | ZX81 | :) |
| 09:07.41 | insane_1 | that would change the port numbers then? |
| 09:07.44 | ZX81 | yep |
| 09:07.53 | insane_1 | so instead of 1-8 it would be 9-18 |
| 09:07.59 | ZX81 | possibly |
| 09:08.02 | insane_1 | interesting that zttool doesn't show them as ready though |
| 09:08.09 | ZX81 | or maybe 25-(25+8) |
| 09:08.11 | ZX81 | :) |
| 09:08.11 | insane_1 | whoa |
| 09:08.19 | insane_1 | actually now it shows one of them ready |
| 09:08.23 | ZX81 | :) |
| 09:08.27 | insane_1 | i love you |
| 09:08.30 | ZX81 | :) |
| 09:08.33 | ZX81 | ~adn |
| 09:08.33 | jbot | hmm... adn is hmm... adn is is the Asterisk Daily News - http://www.venturevoip.com/news.php for HTML and http://feeds.feedburner.com/asterisknews for RSS |
| 09:08.43 | ZX81 | :) |
| 09:09.15 | insane_1 | you know of all the dialplan samples and all the reading i did of the $50 book asterisk: the future of ip telephony |
| 09:09.20 | insane_1 | i haven't seen 1 that does what i want it too |
| 09:09.32 | ZX81 | :) that's the beauty of Asterisk |
| 09:09.38 | ZX81 | we create default dialplans |
| 09:09.43 | insane_1 | not really, it makes it hell for me, lol |
| 09:09.51 | ZX81 | but it takes a week or so before our customers are happy |
| 09:09.58 | ZX81 | everyone wants something different |
| 09:10.16 | ZX81 | the language is a bit clunky - but its like trying to make a game with BASIC |
| 09:10.24 | ZX81 | you just use what you're given :) |
| 09:10.41 | insane_1 | both cards show ready now |
| 09:10.45 | insane_1 | though |
| 09:10.45 | ZX81 | sweet as |
| 09:10.48 | insane_1 | i'm concerned |
| 09:10.52 | ZX81 | how come? |
| 09:11.07 | insane_1 | ZT_CHANCONFIG failed on channel 17: No such device or address (6) |
| 09:11.19 | insane_1 | that's my fxsks=9-18 |
| 09:11.19 | ZX81 | um |
| 09:11.26 | insane_1 | and it only errored on channel 17 |
| 09:11.43 | ZX81 | 1-8 9-18 should be right I guess |
| 09:11.50 | ZX81 | unless they're six port modules |
| 09:11.53 | insane_1 | 8 |
| 09:11.56 | ZX81 | k |
| 09:12.02 | insane_1 | 8 fxs, 8 fxo |
| 09:12.07 | ZX81 | anything else in the machine? |
| 09:12.26 | ZX81 | isn't 8+8 16? |
| 09:12.31 | insane_1 | oh yeah |
| 09:12.32 | insane_1 | LOL |
| 09:12.33 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 09:12.40 | ZX81 | :) |
| 09:13.15 | insane_1 | thanks for pointing that out |
| 09:13.20 | ZX81 | :) np |
| 09:13.34 | insane_1 | are you connected with digium/asterisk in any way? |
| 09:13.44 | ZX81 | nah just write the daily asterisk news |
| 09:13.53 | ZX81 | run a company or two in Dunedin, New Zealand |
| 09:14.03 | insane_1 | so you're good with programming it then? :P |
| 09:14.18 | ZX81 | not as good as tzafrir :) |
| 09:14.32 | ZX81 | brb cigarette |
| 09:15.02 | insane_1 | k |
| 09:16.40 | insane_1 | tzafrir, might you be around? |
| 09:16.47 | tzafrir | yes |
| 09:16.53 | insane_1 | sweet! |
| 09:17.22 | insane_1 | while I'm not asking you to write my entire dialplan for me, and i'm sure there's probably no real incentive for you to help me, i was wondering if you might be able to help |
| 09:17.34 | insane_1 | i can't seem to wrap my head around this asterisk logic completely |
| 09:17.59 | insane_1 | and i'm going out of my head trying too |
| 09:18.07 | insane_1 | but basically this is what i want to happen: |
| 09:18.13 | tzafrir | I figure someone has already pointed you to a good book on the subject |
| 09:18.28 | insane_1 | i've read the o'reilly's book cover to cover |
| 09:18.32 | insane_1 | it's very basic |
| 09:18.42 | insane_1 | and while i've gotten the basic stuff to work |
| 09:18.51 | insane_1 | and have basically lived on voip-info.org for the last week |
| 09:18.54 | insane_1 | i can't get what i want to done |
| 09:18.56 | tzafrir | ok, so ask ahead |
| 09:19.13 | insane_1 | and i'm close to having to fork out thousands for a consulting firm and i really would not like to do that |
| 09:19.20 | insane_1 | ok so here it goes: |
| 09:20.06 | insane_1 | inbound call to pots line, rings all extensions on the network, as well as external numbers via sip |
| 09:20.47 | insane_1 | that's just inbound calls to a particular pots line |
| 09:21.23 | insane_1 | outbound calls from actual linksys SPA942's: lines to be dialed out on need to be selected by pressing the little line buttons on the side of the phone |
| 09:21.34 | insane_1 | via pots or sip, whichever may be the case |
| 09:22.12 | insane_1 | if incoming pots call is not answered, forward to a single voicemail box |
| 09:22.18 | insane_1 | accessible from all extensions |
| 09:23.15 | insane_1 | any advice you may have is of course greatly appreciated, i realize of course you're not obligated |
| 09:23.24 | insane_1 | but i'd be very grateful |
| 09:24.15 | insane_1 | i have 6 SIP trunks and 5 pots lines to work with, and depending on which number the inbound POTS call comes from |
| 09:24.21 | insane_1 | depends on what i want it to do |
| 09:24.27 | tzafrir | insane_1, sounds like quite a basic dialplan setup |
| 09:24.44 | insane_1 | are you serious? |
| 09:24.56 | tzafrir | Unless I miss something |
| 09:25.30 | tzafrir | "incoming call does XYZ" -> an incoming call gets to somewhere in your dialplan. Do XYZ there |
| 09:26.06 | insane_1 | the planning of it sounds simple yes, it's not hella complicated like IVR's and things like that |
| 09:26.11 | insane_1 | but the CODING of it |
| 09:26.13 | insane_1 | is difficult |
| 09:26.23 | tzafrir | Doing something in case of no answer: check the DIALSTATUS of dial |
| 09:26.25 | tzafrir | e.g: |
| 09:27.15 | tzafrir | exten => foo,n,Dial(SIP/bar&SIP/baz&IAX/foo) |
| 09:27.39 | insane_1 | i only have SIP here |
| 09:27.40 | insane_1 | no iax |
| 09:27.43 | tzafrir | exten => foo,n,Goto(s-${DIALSTATUS},1) |
| 09:28.11 | tzafrir | hmm... surely you don't have IAX. Maybe IAX2. |
| 09:28.48 | insane_1 | i don't have iax anything |
| 09:28.55 | insane_1 | simply SIP and 5 pots lines |
| 09:28.56 | tzafrir | Dial sets the variable DIALSTATUS according to the result. See 'core show application dial' |
| 09:30.33 | insane_1 | hmm |
| 09:30.44 | insane_1 | i don't think i've quite explained it the way i intended too |
| 09:30.48 | insane_1 | i apologize if i was unclear |
| 09:32.08 | insane_1 | really i guess the outbound dial is really what i need to concentrate on this moment |
| 09:32.18 | insane_1 | i have 6, 4 line phones |
| 09:32.27 | insane_1 | Linksys SPA942's |
| 09:32.42 | insane_1 | each one of them needs to do the same thing |
| 09:32.51 | insane_1 | and all be linked to the same pots line |
| 09:33.32 | *** join/#asterisk dty (n=dertybiz@195.225.54.221) |
| 09:33.34 | insane_1 | 4 separate phone numbers |
| 09:33.48 | insane_1 | so when line 1 is picked up, it dials out as that phone number |
| 09:34.37 | insane_1 | my situation isn't the typical call center where you have many users doing lots of things, i have 1 user doing the same thing from multiple phones |
| 09:35.34 | insane_1 | and when 1 of those lines is called, it rings all of the phones as well a few outside numbers, such as my cell phone |
| 09:36.18 | *** join/#asterisk rvhi (n=chatzill@66.175.65.82) |
| 09:37.03 | tzafrir | If you have 6 lines on the phone, each of them can be an independent SIP user/peer |
| 09:37.37 | insane_1 | i'm needing to concentrate on pots right this moment |
| 09:37.43 | insane_1 | and i'll get to the multiring |
| 09:37.54 | insane_1 | 4 line phone |
| 09:37.55 | rvhi | after * restart, all hint extensions became "unavailable", any idea? |
| 09:38.23 | insane_1 | inbound call on my FXO Card channel 1 |
| 09:38.30 | insane_1 | is assigned to line 1 of my SIP phone |
| 09:38.35 | insane_1 | for incoming and outgoing calls |
| 09:38.54 | insane_1 | fxo card channel 2 is assigned to line 2 on my sip phone |
| 09:39.12 | tzafrir | rvhi, how can a fresh new Asterisk know that a phone is available? |
| 09:39.19 | insane_1 | and ongoing until i'm out of lines on my phone |
| 09:40.03 | insane_1 | think of it this way, ports 1-4 on FXO card need to link directly to lines 1 through 4 of my SIP phone |
| 09:40.08 | insane_1 | for incoming and outgoing calls |
| 09:40.40 | rvhi | tzafrir: how does * know? i thought * knows about the channel availability, no? |
| 09:41.01 | tzafrir | Is it a SIP phone? |
| 09:41.07 | rvhi | polycom |
| 09:41.42 | tzafrir | And has no specific host settings, right? It has to be registered |
| 09:41.51 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 09:42.11 | tzafrir | So until it registers, Asterisk simply cannot direct calls to it |
| 09:42.46 | tzafrir | ('host=dynamic' , that is) |
| 09:44.16 | rvhi | all extensions register |
| 09:44.32 | rvhi | 'show hints' show unavailable |
| 09:45.45 | tzafrir | They show up with the proper IP on 'sip show peers'? |
| 09:45.53 | rvhi | ya |
| 09:46.06 | rvhi | can make calls and receive calls |
| 09:46.55 | tzafrir | Next: are there actually hint priorities in your dialplan? |
| 09:47.23 | rvhi | yes, |
| 09:47.28 | rvhi | it works before restart |
| 09:47.42 | adelas | is there a way to create an extension, and have it create a config file for the cisco phone, and host it in a tftp server? |
| 09:47.56 | adelas | on the asterisk itself? |
| 09:48.28 | adelas | long time ago i saw ampportal or something like that have it, but i don't see it in like asteriskNow |
| 09:49.04 | insane_1 | hrm |
| 09:51.58 | insane_1 | :/ |
| 09:53.06 | insane_1 | have i pissed you off tzafrir |
| 09:56.58 | *** join/#asterisk ltd (n=z@pat.transact.net.au) |
| 09:57.42 | *** join/#asterisk _ys (n=yuri@80.70.236.69) |
| 09:59.22 | *** join/#asterisk coppice (n=chatzill@235.202.17.210.dyn.pacific.net.hk) |
| 10:04.28 | *** join/#asterisk lsodi (n=lsodi@84-50-34-145-dsl.kjj.estpak.ee) |
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| 10:07.04 | lsodi | to include extensions from extensions.ael file in extensions.conf I must include them in context as "include => customerservice"? |
| 10:09.20 | tzafrir | insane_1, no. Just me bing elsewhere |
| 10:10.28 | insane_1 | ah |
| 10:10.43 | insane_1 | so does what i said make sense? |
| 10:11.06 | insane_1 | zap channel 1 rings line 1 of my sip phone, line 1 of my sip phone rings out of zap channel 1 |
| 10:12.28 | tzafrir | analog zap? |
| 10:12.33 | insane_1 | yes |
| 10:12.53 | tzafrir | So it goes to extension s in the context you defined to it in zapata.conf . |
| 10:13.19 | tzafrir | So there, you just use Dial(SIP/my-sip-phone) |
| 10:13.46 | tzafrir | Or: |
| 10:13.55 | tzafrir | So there, you just use Dial(SIP/my-sip-phone&SIP/another-sip-phone) |
| 10:14.12 | tzafrir | that part works? |
| 10:14.23 | insane_1 | i'm sorry i don't think it does |
| 10:15.29 | insane_1 | Incoming analog Zap channel 1, rings to Line 1 of Linksys SPA942 |
| 10:15.36 | insane_1 | on the same token |
| 10:15.51 | insane_1 | when dialing out on line 1 of linksys spa942, it uses Zap channel 1 |
| 10:19.02 | tzafrir | insane_1, dialing out is a different flow |
| 10:19.16 | tzafrir | I asked you about a call that comes from the Zap channel |
| 10:19.28 | tzafrir | We'll deal with calls that come from the SIP channel later |
| 10:20.17 | insane_1 | i'm telling you about the calls that come from the zap channel |
| 10:20.20 | insane_1 | that's what i want to happen |
| 10:20.30 | insane_1 | i don't care about sip incoming yet |
| 10:20.49 | insane_1 | i only want to talk inbound and outbound pots analog zap right now |
| 10:21.12 | insane_1 | but the phones i have for that are SIP Phones |
| 10:21.24 | insane_1 | the pots numbers need to register as lines to the SIP phone |
| 10:27.34 | tzafrir | insane_1, all SIP users should be able to make outbound Zap calls? |
| 10:30.24 | insane_1 | yes |
| 10:30.54 | insane_1 | SIP Phones need to make outbound Zap calls |
| 10:31.00 | insane_1 | 4 separate pots lines |
| 10:46.23 | _x86_ | morning |
| 10:48.03 | *** join/#asterisk jiboneus (n=jibone@140.82.50.60.klj05-home.tm.net.my) |
| 10:57.14 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 10:57.38 | hi365 | hello. can someone help with a mysql querie? |
| 10:59.07 | hi365 | nevermind |
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| 11:58.51 | knarfly | anyone from Amsterdam online this morning? |
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| 12:41.16 | *** join/#asterisk [gnubie] (n=[gnubie]@cm174.gamma179.maxonline.com.sg) |
| 12:42.14 | [gnubie] | can anyone point me to a website that discusses all about call (attended/blind) transfer on asterisk 1.4.x? |
| 12:46.42 | [gnubie] | i want that my analog phone connected to my fxs port of the digium tdm dev kit can initiate a call (attended/blind) transfer.. |
| 12:49.33 | d-k-t | [gnubie], you seen this page? http://www.voip-info.org/wiki-Asterisk+config+features.conf |
| 12:49.56 | [gnubie] | lemme check it.. thanks.. |
| 12:51.39 | *** join/#asterisk Op3r (n=Op3r@203.177.210.151) |
| 12:58.15 | tzafrir | [gnubie], also chan_zap has a built-in "flash", which is attended transfer as well |
| 12:59.03 | Op3r | does ./genzaptelconf can really configure the zap cards ? |
| 12:59.08 | [gnubie] | tzafrir: how do i make use of that one? |
| 12:59.51 | tzafrir | Op3r, analog ones: sure. digital ones: basically yes, but there is more of a gueswork there |
| 13:02.29 | Op3r | so basically it will detect whats the bchan and dchan and if it is euroisdn or what not right? |
| 13:03.06 | tzafrir | [gnubie], threewaycalling=yes transfer=yes |
| 13:03.10 | tzafrir | in zapata.conf |
| 13:03.55 | [gnubie] | tzafrir: that's it? i mean, no other configs to add on my dialplan? |
| 13:03.56 | tzafrir | Op3r, no, it won't detect that. It assumes that if you have E1 or BRI you also use euroisdn :-) |
| 13:04.10 | Op3r | errr |
| 13:04.12 | Op3r | that sucks |
| 13:04.14 | Op3r | :( |
| 13:04.31 | tzafrir | and if you have T1 you use national. And if you have 24 channels: you have T1 |
| 13:04.51 | tzafrir | (those assumptions break in some cases. e.g: J1. But are mostly correct) |
| 13:05.04 | tzafrir | Op3r, you use BRI or E1? |
| 13:05.59 | Op3r | nope |
| 13:06.04 | tzafrir | That's much better than trying to detect from a connected line. Because then you cannot configure anything before connecting to the telco |
| 13:06.06 | Op3r | im just asking |
| 13:06.28 | tzafrir | Or consider the case of two ports connected in a loopback |
| 13:07.14 | tzafrir | So if you have a better suggestion, I'd love to hear it |
| 13:07.23 | tzafrir | (or even read it) |
| 13:09.01 | *** join/#asterisk Greek-Boy (n=email@41.221.58.5) |
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| 15:10.54 | dty | hi |
| 15:11.04 | dty | any courses/training in europe? |
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| 15:32.14 | *** join/#asterisk moprilo (n=nada@190.10.0.84) |
| 15:36.15 | moprilo | hi, I have this asterisk mesh, but when i terminate a call to pstn, I the pstn call is answered but it takes about 10 sec to the incomming *voip sound to connect. (independent of the delay). |
| 15:36.34 | moprilo | Anyone know how to treat this.. or how this is called, so I can search more easily..? |
| 15:37.00 | Qwell | probably firewall issues |
| 15:37.04 | Qwell | ~sipnat |
| 15:37.04 | 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 |
| 15:37.24 | moprilo | really?.. i'll check on that, thanks |
| 15:38.10 | Qwell | normally one-way audio would occur, but I could see mappings being eventually setup, allowing the audio through |
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| 16:30.48 | hardwire | so |
| 16:30.58 | hardwire | LNP can only operate within the same "ratecenter" right? |
| 16:48.06 | *** join/#asterisk errr (n=errr@fedora/errr) |
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| 17:06.08 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 17:10.43 | *** mode/#asterisk [+o mog] by ChanServ |
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| 17:15.09 | mocker | How difficult is it to split a PRI into both voice and data? |
| 17:17.25 | hardwire | if its point to point, easy.. otherwise have fun coordinating that with your ISP. |
| 17:17.55 | hardwire | some ISP's even offer dynamic channel allocation to proprietary hardware.. then split it up to a Network interface and T1 |
| 17:26.18 | SplasPood | hrm... anyone having voicepulse issues? |
| 17:29.17 | hardwire | I'm kind acurious why broadvoice only allows one incoming call per account |
| 17:32.20 | mocker | hardwire: So a company that has like 8 analog lines and DSL, probably best to just get a PRI and keep the DSL line? |
| 17:32.44 | hardwire | mocker: yeh, offers you a bit more scalability anyways |
| 17:32.55 | mocker | (assuming the 8 lines are appx equal to cost of PRI) |
| 17:33.05 | mocker | Not an easy way to get rid of the DSL too. |
| 17:35.11 | hardwire | if you're ISP supports it, you can mix em up |
| 17:35.25 | hardwire | except it won't be as fast as your dsl (maybe) |
| 17:35.31 | mocker | *nod* |
| 17:35.36 | mocker | hardwire: Thanks. |
| 17:35.50 | mocker | I'm used to just straight PRI anyway, so that works. :) |
| 17:35.53 | hardwire | you would be making your asterisk box into the internet router |
| 17:36.01 | hardwire | ah |
| 17:42.15 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 17:45.42 | hardwire | so ChanIsAvail won't lock the channel.. hmm.. |
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| 18:57.48 | *** join/#asterisk ManxPower (n=manxpowe@240.sub-70-222-86.myvzw.com) |
| 18:58.40 | jer | i need to rename an extension, and i keep getting a username mismatch, have <new_exten>, digest has <old_exten> ... i've replaced all references on the device itself, and that i can see in *; what am i missing? |
| 19:01.00 | tzafrir | jer, what we are missing is the trace of such a call and a copy of your dialplan |
| 19:01.59 | [TK]D-Fender | tzanger, No, not dial... sip.conf and some backupt of what was done on the "phone" |
| 19:02.07 | [TK]D-Fender | tzanger, Dialplan doesn't figure in at all |
| 19:02.14 | [TK]D-Fender | darn auto-complete |
| 19:03.04 | *** join/#asterisk ZX81 (n=ZX81@202.20.97.211) |
| 19:04.55 | jer | [TK]D-Fender, http://pastebin.ca/819035 .. what was done on the phone was going into its web gui, going to the identity page, and changing all occurances of the old extension number to the new extension number (in this case, the new extension number is 600) |
| 19:08.47 | [TK]D-Fender | jer, well *'s side looks pretty simple. Guess its your phoen |
| 19:08.49 | [TK]D-Fender | phoen* |
| 19:08.51 | [TK]D-Fender | asjkdjkalsdha |
| 19:09.18 | jer | hrmm, i'll give it a hard reboot just in case |
| 19:09.45 | [TK]D-Fender | jer, And of course.... we can't see what you did in your phone |
| 19:13.19 | jer | [TK]D-Fender, right now i know... doing a hard reboot of the phone somehow fixed it.. a little disturbing but anyway |
| 19:13.22 | jer | sorry for the noise |
| 19:15.30 | *** join/#asterisk Cyorxamp (n=Cyorxamp@212.57.229.111) |
| 19:16.29 | ZX81 | morning all |
| 19:18.12 | tzafrir | wow, it's getting late |
| 19:19.04 | ZX81 | tzafrir, you been on this whole time? |
| 19:19.46 | tzafrir | ZX81, for some definition of "here" |
| 19:19.48 | ZX81 | [23:37] tzafrir insane_1, all SIP users should be able to make outbound Zap calls? |
| 19:20.15 | ZX81 | [08:19] tzafrir wow, it's getting late |
| 19:20.37 | tzafrir | ZX81, some people have the strangest time zones |
| 19:20.45 | ZX81 | :) |
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| 19:48.14 | ZX81_ | aha the reason the UPS didn't come back up is that a cleaner unplugged it to plug in a vacuum cleaner! |
| 19:48.22 | ZX81_ | lol |
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| 20:03.18 | ManxPower | don't you mean the former cleaner? |
| 20:03.30 | ZX81 | :) heh yeah |
| 20:06.16 | ManxPower | I had a couple of power outages at home last night. So I plugged in my TiVo to a UPS. Of course there were no more power outages. |
| 20:06.27 | d-k-t | another good reason to use somewhat different sockets for stuff like that :) |
| 20:07.25 | d-k-t | or fit a different standard plug to the vacuum and make sure there are enough sockets to match around the place |
| 20:07.32 | d-k-t | or do both :) |
| 20:17.37 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 20:19.33 | *** join/#asterisk hmmhesays (n=Neg@24-119-176-74.cpe.cableone.net) |
| 20:19.41 | hmmhesays | what up folks |
| 20:22.51 | *** join/#asterisk mog (n=mog@nat/digium/x-6e7a7bb1cdcdcf76) |
| 20:22.51 | *** mode/#asterisk [+o mog] by ChanServ |
| 20:37.16 | *** join/#asterisk troy- (n=tabmeist@CPE00907f17e478-CM00186845db94.cpe.net.cable.rogers.com) |
| 20:38.52 | troy- | what component of asterisk is listening on port 5353? |
| 20:39.28 | ManxPower | troy-: TCP or UDP? |
| 20:39.35 | troy- | udp |
| 20:40.23 | *** join/#asterisk techie (n=techie@adsl-76-247-10-106.dsl.lsan03.sbcglobal.net) |
| 20:40.26 | ManxPower | Asterisk does not use that port. |
| 20:40.45 | ManxPower | However, a 5-second google search shows that it is used by Zero Conf (a linux thing) as well as Apple TV |
| 20:41.10 | ManxPower | The #Linux channel is 4 doors down the hall to the left. They should be able to help you more. |
| 20:41.22 | troy- | haha, thanks :-) |
| 20:42.03 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 20:43.07 | *** part/#asterisk ManxPower (n=manxpowe@240.sub-70-222-86.myvzw.com) |
| 20:44.16 | tzafrir | netstat -lnup should show which process listens on that port |
| 20:45.03 | tzafrir | ZeroConf is also a OSX thing, though by a different buzzword |
| 20:46.36 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 20:49.13 | troy- | tzafrir, its showing as avahi-daemon listening on 5353/16384/16385 |
| 20:49.46 | tzafrir | there you go. zeroconf |
| 20:50.05 | troy- | is there any reason i need to have avahi running if i'm using only static external IPs? |
| 20:50.54 | *** join/#asterisk Haq (n=liquid@6a.ab.5446.static.theplanet.com) |
| 20:51.43 | d-k-t | troy-, it's nothing asterisk related, and if you don't know what it does, you probably don't need it |
| 20:52.14 | Haq | This is annoying. I can make an outbound call and hear the call recipient but they cannot hear me. My softphone is setup & working w/the microphone correctly. |
| 20:52.26 | troy- | d-k-t, the system is a VPS so i'm not really sure whether its something i need to keep or not |
| 20:54.15 | d-k-t | troy-, typically could be useful if you want to access a printer or other zeroconf compatible device on the local network, if not, you probably don't need it |
| 20:54.16 | *** join/#asterisk Hadi- (n=Hadi@CPE001310492769-CM001225e00576.cpe.net.cable.rogers.com) |
| 20:54.20 | Hadi- | hello everyone |
| 20:54.40 | troy- | thanks d-k-t |
| 20:54.54 | d-k-t | troy-, np |
| 20:55.11 | Hadi- | i'm having a little issue.. we are using a cisco 7960 connected to asterisk with the g729 codec... when we make calls.. we see the following on the asterisk CLI: |
| 20:55.12 | Hadi- | 2007-12-16 15:53:23 NOTICE[8554]: rtp.c:415 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 209.47.1.15 |
| 20:55.28 | Hadi- | and right when we hear this.. there is silence on the phone |
| 20:55.34 | Hadi- | for 1-2 seconds.. |
| 20:55.40 | Hadi- | (on the 7960) |
| 20:56.22 | Hadi- | any suggestions |
| 20:56.57 | kaldemar | have you turned comfort noise off on the phone? |
| 20:57.08 | Hadi- | diable VAD? yes |
| 20:57.10 | Hadi- | same.. |
| 20:57.40 | Haq | Any and it magically starts working. |
| 20:58.31 | Hadi- | well i disable it in the options |
| 20:58.35 | Hadi- | but the issue is still there |
| 21:00.06 | d-k-t | Hadi-, isn't it Enable VAD NO? |
| 21:00.19 | d-k-t | ok, nm :) |
| 21:00.19 | Hadi- | yes |
| 21:00.25 | Hadi- | I did Enable VAD = no |
| 21:01.03 | Hadi- | but still see that message on CLI |
| 21:01.08 | Hadi- | and the silence |
| 21:04.13 | *** join/#asterisk Daviey (n=dave@ubuntu/member/daviey) |
| 21:04.31 | *** join/#asterisk Odie_flocon (n=odie@S0106000f3d5eae81.cg.shawcable.net) |
| 21:05.44 | Daviey | What sort of processor should i get for <300 extensions, 15 concurrent calls with the possibility of some of them needing G.729? |
| 21:06.39 | Daviey | oh, thats 15 PRI calls + 15-20 internal |
| 21:14.13 | d-k-t | Daviey, it's unlikely anyone could give you a solid answer for that, the asterisk book recommends multiple 'modern' CPUs for that sort of load or multiple systems |
| 21:15.49 | mocker | I would avoid celeron though |
| 21:15.52 | mocker | :) |
| 21:16.23 | hmmhesays | they are crippled |
| 21:17.22 | *** join/#asterisk Aces1up (n=dude@ip70-173-52-152.lv.lv.cox.net) |
| 21:17.45 | Aces1up | anyone know of a good voip provider? seems all the reviews i read, no company is worth buying from. |
| 21:17.59 | d-k-t | I personally would avoid celerons for anything except for where your only requirement is low cost and performance is a non-issue |
| 21:18.38 | d-k-t | Aces1up, totally depends on your requirements |
| 21:19.32 | d-k-t | Aces1up, I've not seen any telco, be they a voip provider or traditional telco that's got 100% good reviews |
| 21:20.22 | Aces1up | hrmm need a voip line for business purposes and a 1-800 number. |
| 21:20.52 | d-k-t | Aces1up, in the US? |
| 21:21.27 | russellb | i've been working with junction networks a bit lately, it has been working exellent |
| 21:21.33 | *** part/#asterisk techie (n=techie@adsl-76-247-10-106.dsl.lsan03.sbcglobal.net) |
| 21:21.35 | russellb | and it was very quick and easy to set up... |
| 21:22.29 | Aces1up | d-k-t yes |
| 21:22.39 | russellb | but there are plenty of good ones |
| 21:23.07 | russellb | i also like nufone and voicepulse |
| 21:23.25 | Aces1up | i was looking at voicepulse they seems on average to have better reviews. |
| 21:23.56 | hmmhesays | yeah they all will give you problems |
| 21:24.50 | hmmhesays | what is the best h.323 implementation for asterisk now? |
| 21:25.05 | russellb | whichever one you can get to work |
| 21:25.05 | Aces1up | hmmhesays really? |
| 21:25.10 | russellb | none of them are really actively supported |
| 21:26.02 | hmmhesays | Aces1up: yes, you are sending calls across the public internet... just think about that |
| 21:27.59 | hmmhesays | junction is a bit spendy |
| 21:30.00 | d-k-t | It never stops dismaying me when so many people ask me 'so what's wrong with using the internet to provide 'critical service a' to this customer, it's much cheaper' when I suggest they get a couple of dedicated circuits |
| 21:33.06 | hmmhesays | pretty much yeah, |
| 21:33.34 | hmmhesays | I do some lcr out an itsp for international calls, but that is about it |
| 21:36.02 | hmmhesays | I have a 7ms ping time to voipjet |
| 21:36.13 | d-k-t | the safety issues with IP phones connected to a remote PBX, especially one in another country without provision of locally connected phones, training and signage is another one... |
| 21:40.50 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 21:48.46 | *** join/#asterisk mog (n=mog@nat/digium/x-69c3438e788da769) |
| 21:48.46 | *** mode/#asterisk [+o mog] by ChanServ |
| 21:59.13 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 22:00.55 | *** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 22:11.39 | *** join/#asterisk RoyK (n=roy@ip-172-3-149-91.dialup.ice.no) |
| 22:11.54 | *** join/#asterisk ZX81 (n=ZX81@202.49.106.158) |
| 22:16.06 | *** join/#asterisk troy- (n=tabmeist@CPE00907f17e478-CM00186845db94.cpe.net.cable.rogers.com) |
| 22:26.50 | *** join/#asterisk jetlagmk2 (i=jetlag@70.106.121.28) |
| 22:27.15 | X-Filez | Ppls, As it is possible to adjust that calling heard music or words during that moment when there is a call (instead of hooters) ? |
| 22:27.51 | *** join/#asterisk CVirus (n=GoD@62.135.96.207) |
| 22:29.44 | russellb | that question didn't make sense ... |
| 22:30.44 | X-Filez | brr. rewrite my question ? |
| 22:33.32 | *** join/#asterisk mltlnx (n=mltlnx@cpe-72-229-184-242.nyc.res.rr.com) |
| 22:34.15 | X-Filez | rewrite my question: PPls, I try call in phone and I hear BEEEP BEEEP, i want hear sound play from file, this possible ? |
| 22:35.47 | hmmhesays | anything is possible |
| 22:36.21 | X-Filez | you can say name cmd in extention ? |
| 22:36.28 | X-Filez | or where this configure |
| 22:43.54 | De_Mon | english isn't even your 2nd language yet |
| 22:45.33 | ZX81 | X-Filez, maybe use the exten => h,1 extension |
| 22:45.42 | ZX81 | and see if you can play a file |
| 22:46.13 | ZX81 | or for a call thats up |
| 22:46.18 | ZX81 | use a musiconhold class |
| 22:46.26 | ZX81 | (defined in musiconhold.conf) |
| 22:46.36 | ZX81 | and use a recording of whatever sound you want |
| 22:46.43 | ZX81 | then when you dial use: |
| 22:47.01 | ZX81 | exten => _X.,1,Dial(Zap/${EXTEN}|30|m) |
| 22:47.09 | ZX81 | the |m will play the music on hold |
| 22:47.26 | ZX81 | I've used a New Zealand ring tone recorded as a file for this in the past |
| 22:47.56 | ZX81 | the extensions are defined in extensions.conf (assuming you have a clean Asterisk install and not Trixbox etc) |
| 22:48.45 | X-Filez | ZX81: hm :) thanks.. |
| 22:49.01 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 22:49.15 | ZX81 | also, you might want to check out the wiki for info |
| 22:49.25 | ZX81 | ~voip-info |
| 22:49.25 | jbot | hmm... voip-info is the Voice Over IP wiki. It is a community resource which will answer all of your questions, from Asterisk to ZTDummy. You can find it over at http://www.voip-info.org - well worth bookmarking |
| 23:01.05 | *** join/#asterisk mrtelnet (n=mrtelnet@c-67-173-191-235.hsd1.in.comcast.net) |
| 23:01.34 | mrtelnet | can anyone help me with a dialplan issue i have? |
| 23:02.16 | mrtelnet | hello? |
| 23:03.19 | ZX81 | mrtelnet, I have a hint for you |
| 23:03.19 | Nugget | telnet is eeeeeeevil! |
| 23:03.20 | ZX81 | :) |
| 23:03.31 | mrtelnet | true, i now use ssh |
| 23:03.44 | ZX81 | if you just ask your problem rather than asking if you can ask a problem you're a lot more likely to get an answer :) |
| 23:04.00 | mrtelnet | Im sorry, ive never really used irc before |
| 23:04.09 | ZX81 | :) sweet |
| 23:04.14 | ZX81 | so what's your problem? |
| 23:04.15 | mrtelnet | the rare times i've been on, its been derelict |
| 23:04.20 | ZX81 | :) |
| 23:04.54 | mrtelnet | I am using a call file to call a sip based hardphone to connect to a pots number on a sip gateway |
| 23:05.05 | ZX81 | yep |
| 23:05.50 | mrtelnet | It fails with a message to the full log that is "pbx_spool.c: Call failed to go through, reason 3" |
| 23:05.59 | ZX81 | lol nice |
| 23:06.02 | ZX81 | that's helpful |
| 23:06.03 | ZX81 | :) |
| 23:06.04 | mrtelnet | yup |
| 23:06.05 | ZX81 | so |
| 23:06.12 | ZX81 | when you make a call manually does it work? |
| 23:06.13 | mrtelnet | they dont seem do have the errors docs |
| 23:06.17 | mrtelnet | yes, |
| 23:06.26 | ZX81 | ok can you pastebin your callfile? |
| 23:06.28 | ZX81 | i.e. |
| 23:06.29 | mrtelnet | the extension, if i dial it, will connect fine |
| 23:06.30 | ZX81 | ~pastebin |
| 23:06.31 | jbot | [~pastebin] 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 |
| 23:06.44 | mrtelnet | thanks |
| 23:06.55 | ZX81 | np haven't fixed it yet :) |
| 23:07.32 | mrtelnet | http://pastebin.ca/819410 |
| 23:07.37 | mrtelnet | is the call file |
| 23:07.45 | ZX81 | k sec |
| 23:07.58 | mrtelnet | Local/352 is the hardphone and 9991 is the dp ext |
| 23:08.07 | ZX81 | ok |
| 23:08.16 | ZX81 | can you show me the from-internal context? |
| 23:08.20 | ZX81 | again in pastebin |
| 23:08.25 | ZX81 | this is trixbox isn't it :) |
| 23:08.32 | mrtelnet | yup |
| 23:08.36 | ZX81 | :) |
| 23:08.37 | mrtelnet | http://pastebin.ca/819411 |
| 23:08.41 | mrtelnet | is the dialplan |
| 23:08.48 | ZX81 | k |
| 23:09.33 | ZX81 | does the first leg of the call come up? |
| 23:09.39 | ZX81 | i.e. does your sip phone ring? |
| 23:09.42 | mrtelnet | no |
| 23:09.58 | ZX81 | is there a 352 or something similar in from-internal? |
| 23:10.04 | mrtelnet | should be |
| 23:10.21 | ZX81 | does your phone support two lines? |
| 23:10.38 | ZX81 | or do you have another phone you can use to call 352? |
| 23:10.44 | mrtelnet | if i change the dp to just play a sound instead of dialing, the sip phone rings and i hear the sound |
| 23:10.48 | mrtelnet | yes |
| 23:11.04 | mrtelnet | its an X-Lite softphone |
| 23:11.10 | ZX81 | if you change the destination |
| 23:11.17 | ZX81 | to play a sound file? |
| 23:11.24 | ZX81 | it calls the sip phone |
| 23:11.33 | ZX81 | but if you change it to dial the number it doesn't? |
| 23:11.36 | ZX81 | that seems strange |
| 23:11.37 | mrtelnet | yup |
| 23:11.43 | ZX81 | as it would normally call the softphone |
| 23:11.47 | ZX81 | then dial the number |
| 23:11.53 | ZX81 | or play the file or whatever |
| 23:12.09 | mrtelnet | if i change line 5 to exten => _9991.,n,Playsound(tt-weasels) |
| 23:12.14 | mrtelnet | or Playback |
| 23:12.19 | mrtelnet | rathrt |
| 23:12.26 | ZX81 | so it is not supposed to start trying the other end till you pickup the phone |
| 23:12.33 | mrtelnet | yup |
| 23:12.58 | ZX81 | so theoretically if you made it context: jkhkjhkjh extension: jkhgjhgjhg priority 999 it should still call you |
| 23:13.09 | mrtelnet | a crm system should be handling a hyperlink to call a client that rings your phone connecting to the clien |
| 23:13.10 | ZX81 | but hangup once you answer |
| 23:13.34 | mrtelnet | Ive not found that to be true |
| 23:13.40 | ZX81 | sec |
| 23:13.43 | mrtelnet | k |
| 23:15.10 | ZX81 | OMFG |
| 23:15.14 | ZX81 | my server just crashed |
| 23:15.19 | ZX81 | lol |
| 23:15.24 | Iamnacho | :( |
| 23:15.28 | mrtelnet | um... |
| 23:15.31 | mrtelnet | thats not good |
| 23:15.34 | ZX81 | no shit |
| 23:15.40 | ZX81 | has 500 users |
| 23:15.49 | ZX81 | lucky there's multiple redundancy |
| 23:15.49 | ZX81 | :) |
| 23:15.52 | ZX81 | ok |
| 23:15.53 | mrtelnet | lol |
| 23:15.57 | ZX81 | so I'm not trying that again |
| 23:15.58 | ZX81 | :) |
| 23:16.02 | ZX81 | but it did make the call |
| 23:16.08 | d-k-t | ZX81, I was going to say, more servers, but you're already there :) |
| 23:16.24 | ZX81 | :) yep |
| 23:16.31 | ZX81 | Starting Local/691@freevoip_nz-ef38,1 at freevoip_nzasdasd,6434742112asdasd,1 failed so falling back to |
| 23:16.45 | ZX81 | ok, so if you change the call file to: |
| 23:17.04 | ZX81 | http://pastebin.ca/819431 |
| 23:17.13 | ZX81 | does it make a call between you and you? |
| 23:18.20 | mrtelnet | no, |
| 23:18.27 | ZX81 | what happens? |
| 23:18.35 | mrtelnet | it said somthing about delaying somthing currently running call file |
| 23:18.42 | ZX81 | hmmm |
| 23:18.50 | mrtelnet | is there a cli command to view the queue |
| 23:19.18 | ZX81 | nah |
| 23:19.24 | ZX81 | it won't retry |
| 23:19.31 | ZX81 | cos you have maxretries=0 |
| 23:19.37 | mrtelnet | right |
| 23:20.04 | ZX81 | can you change it back |
| 23:20.12 | mrtelnet | to? |
| 23:20.17 | ZX81 | and then try again exten => _9991.,n,Playback(tt-weasels) |
| 23:20.18 | mrtelnet | the original 9991 |
| 23:20.21 | mrtelnet | sure |
| 23:20.21 | ZX81 | yeah |
| 23:20.38 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@c-98-198-162-26.hsd1.tx.comcast.net) |
| 23:20.58 | ZX81 | maybe it needs an accountcode to make an outgoing pstn call? |
| 23:21.15 | mrtelnet | no accounting enabled, but it is trixbox |
| 23:21.30 | ZX81 | is there an accountcode in the sip_additional.conf section for 352? |
| 23:21.49 | mrtelnet | hold on |
| 23:21.51 | ZX81 | k |
| 23:22.03 | mrtelnet | my playback(ttweasles |
| 23:22.08 | mrtelnet | failed with Delaying retry since we're currently running '/var/spool/asterisk/outgoing/temp.call' |
| 23:22.57 | ZX81 | lol weird |
| 23:23.06 | ZX81 | just rm -f /var/spool/asterisk/outgoing/temp.call |
| 23:23.09 | ZX81 | then do it again |
| 23:23.19 | ZX81 | maybe maxretries of 0 means retry forever lol |
| 23:23.20 | mrtelnet | file doesn't exist |
| 23:23.24 | mrtelnet | lol |
| 23:23.27 | ZX81 | lol nice |
| 23:23.40 | mrtelnet | i restarted asterisk and am trying again |
| 23:23.40 | ZX81 | anything in /var/spool/asterisk/outgoing/ |
| 23:23.43 | ZX81 | lol ok |
| 23:23.43 | mrtelnet | nope |
| 23:23.52 | ZX81 | bit heavy handed but all good |
| 23:23.53 | ZX81 | :) |
| 23:24.29 | mrtelnet | devel system, not worried |
| 23:24.46 | ZX81 | sweet |
| 23:24.46 | ZX81 | :) |
| 23:24.59 | mrtelnet | happened again |
| 23:25.31 | mrtelnet | but if i call 9991, it plays fine |
| 23:25.45 | ZX81 | was that with the weasel sounds? |
| 23:25.50 | mrtelnet | yeah |
| 23:26.19 | ZX81 | so can you pastebin what you have for 9991 stuff? |
| 23:26.54 | *** join/#asterisk taxilian (n=richard@rbateman.dsl.xmission.com) |
| 23:27.15 | mrtelnet | http://pastebin.ca/819447 |
| 23:27.22 | mrtelnet | 9990 is weasels |
| 23:27.23 | taxilian | anyone here able to help me with a quick SIP trunk question? |
| 23:28.36 | ZX81 | taxilian, :) ask the question not if you can ask a question |
| 23:28.48 | ZX81 | mrtelnet, so you changed the call file to 9990? |
| 23:29.11 | mrtelnet | yeah, but ive also tried it with changing 9991 to playback |
| 23:29.29 | ZX81 | but neither worked? |
| 23:29.35 | mrtelnet | no, |
| 23:29.45 | mrtelnet | im trying a different ext right nowe |
| 23:29.53 | ZX81 | the 9991 one won't work |
| 23:29.53 | [TK]D-Fender | mrtelnet, Would be a nice idea if you would pastebin your CALL FILE since thats likely where the error is |
| 23:29.58 | ZX81 | unless you're dialing a number |
| 23:30.16 | ZX81 | i.e. 9991 then something |
| 23:30.20 | [TK]D-Fender | ZX81, "sialing a number"? What a ridiculous redundant statement.... |
| 23:30.26 | ZX81 | :) |
| 23:30.30 | ZX81 | stupid keyboard |
| 23:30.37 | ZX81 | call file looks kinda fine |
| 23:30.53 | [TK]D-Fender | ZX81, Of course it does.. that why it isn't working. Now chow us what you've done |
| 23:30.53 | mrtelnet | heres even weirder |
| 23:30.59 | [TK]D-Fender | show* |
| 23:30.59 | ZX81 | http://pastebin.ca/819454 |
| 23:31.00 | [TK]D-Fender | kl;asdj;ashdkl |
| 23:31.07 | ZX81 | :D |
| 23:31.19 | [TK]D-Fender | # |
| 23:31.19 | [TK]D-Fender | [from-internal-custom] |
| 23:31.40 | [TK]D-Fender | Context: from-internal |
| 23:31.47 | [TK]D-Fender | do these look the same to YOU? |
| 23:32.05 | ZX81 | yeah I don't do custom :) |
| 23:32.14 | ZX81 | lol |
| 23:32.15 | mrtelnet | I made my callfile http://pastebin.ca/819455 |
| 23:32.17 | [TK]D-Fender | ZX81, look what context your extens are in! |
| 23:32.21 | taxilian | lol. alright. I'm trying to set up a trunk so that all calls that match a certain pattern get called as "number@sip.byu.edu" |
| 23:32.32 | [TK]D-Fender | http://pastebin.ca/819447 <---- [from-internal-custom] |
| 23:32.35 | taxilian | I'm using asteriskgui, so that would be best, but if there is a way to do it manually that's fine |
| 23:32.37 | ZX81 | mrtelnet, [TK]D-Fender found it |
| 23:32.50 | taxilian | it's an outgoing only sip trunk |
| 23:32.55 | [TK]D-Fender | taxilian, GUI's are not supported here |
| 23:33.04 | [TK]D-Fender | ~siptrunk |
| 23:33.05 | jbot | siptrunk is probably Asterisk does not support SIP Trunks. Set trunk=no in sip.conf and then set up the device normally in sip.conf. |
| 23:33.17 | ZX81 | maybe #asteriskgui |
| 23:33.19 | [TK]D-Fender | those 2 words don't really belong in the same sentence |
| 23:33.20 | russellb | what? that comment makes no sense at all |
| 23:33.21 | taxilian | I probably have my terms confused |
| 23:33.22 | taxilian | sorry =] |
| 23:33.26 | russellb | "trunk" is not a valid option for sip.conf ... |
| 23:33.28 | taxilian | just a sip call, really. |
| 23:33.28 | mrtelnet | include => from-internal-trixbox? |
| 23:33.46 | [TK]D-Fender | russellb, I know... I'll tell you later who even MADE that jbot bit... it wasn't me ;) |
| 23:33.54 | ZX81 | mrtelnet, change the context in your call file to from-internal-custom |
| 23:33.54 | russellb | jbot: forget siptrunk |
| 23:33.54 | jbot | i forgot siptrunk, russellb |
| 23:34.05 | taxilian | I'll ask in #asterisk-gui, then. thanks |
| 23:34.08 | russellb | [TK]D-Fender: heh, who was it? |
| 23:34.12 | *** part/#asterisk VJFROMGT (n=vjfromgt@user-387g9sf.cable.mindspring.com) |
| 23:34.14 | taxilian | just nobody seems real active there =] |
| 23:34.15 | mrtelnet | k |
| 23:34.57 | mrtelnet | Context: from-internal-cuistom or Channel: Local/352from-internal-custom |
| 23:35.35 | ZX81 | both |
| 23:35.37 | russellb | jbot: siptrunk is <reply> test |
| 23:35.38 | jbot | russellb: okay |
| 23:35.41 | russellb | jbot: siptrunk |
| 23:35.41 | jbot | test |
| 23:35.44 | russellb | yay |
| 23:35.47 | russellb | jbot: forget siptrunk |
| 23:35.48 | jbot | i forgot siptrunk, russellb |
| 23:36.22 | russellb | jbot: siptrunk is <reply> There is nothing special about a SIP trunk in the protocol like there is in the case of IAX2, for example. You set up a SIP trunk like a regular peer in sip.conf. |
| 23:36.22 | jbot | russellb: okay |
| 23:36.30 | russellb | better, i think.. |
| 23:36.46 | mrtelnet | lol |
| 23:37.41 | mrtelnet | No such extension/context 352@from-internal-custom creating local channel |
| 23:41.24 | mrtelnet | any ideas |
| 23:41.57 | *** join/#asterisk sts3c (n=bryan@66-43-34-10.misn.com) |
| 23:43.22 | mvanbaak | hhmm, that sounds like freepbx |
| 23:44.33 | mvanbaak | mrtelnet: open extensions_custom.conf in vim, find the [from-internal-custom] context and add extension 352 there |
| 23:45.48 | *** part/#asterisk RoyK (n=roy@ip-172-3-149-91.dialup.ice.no) |
| 23:46.57 | mrtelnet | speaking as a non-freebpx newb thats trying to learn, how should i put 352 in? |
| 23:47.55 | mvanbaak | depends on what you want to do with extension 352 |
| 23:48.02 | mrtelnet | the Dial(SIP\352) is in from-internal |
| 23:48.04 | *** join/#asterisk saftsack (n=saftsack@83.218.162.174) |
| 23:48.12 | mvanbaak | that cant be right |
| 23:48.18 | mrtelnet | why |
| 23:48.30 | mvanbaak | because it's SIP/<something> |
| 23:48.35 | *** join/#asterisk s0lid (n=_freq@210.213.199.24) |
| 23:48.40 | mvanbaak | the \ is not used for that |
| 23:48.41 | [TK]D-Fender | mrtelnet, "No such extension/context 352@from-internal-custom" <--- its in NOT [from-internal] ! Wake up time! |
| 23:48.43 | mrtelnet | sorry, doing that from memory |
| 23:48.58 | mrtelnet | no, I changed it to that as suggested |
| 23:49.11 | mvanbaak | mrtelnet: add this to [from-internal-custom] |
| 23:49.14 | [TK]D-Fender | mrtelnet, changed WHAT exactly? |
| 23:49.27 | mvanbaak | exten => 352,1,Dial(SIP/352) |
| 23:49.33 | mvanbaak | and be done with it |
| 23:49.45 | mrtelnet | the call file is http://pastebin.ca/819477 |
| 23:50.02 | mvanbaak | btw |
| 23:50.07 | mvanbaak | ~freepbx |
| 23:50.08 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 23:50.20 | mrtelnet | I know... |
| 23:50.37 | [TK]D-Fender | mrtelnet, Fine... go pastebin your entire [from-internal] context for us to look at. |
| 23:50.41 | mrtelnet | i dont really like it, but am too inexpirenced to do it myself |
| 23:50.55 | mvanbaak | mrtelnet: go read the book |
| 23:50.58 | mvanbaak | ~book |
| 23:50.58 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 23:51.07 | mrtelnet | thanks |
| 23:51.22 | *** join/#asterisk litage|w (n=nick@70.55.220.203.static.comindico.com.au) |
| 23:51.36 | mvanbaak | that way you'll be able to do it yourself |
| 23:51.57 | mvanbaak | and the config will be way cleaner then the endless mess that freepbx generates |
| 23:52.31 | mrtelnet | true, but I have no time to do it myself at the moment, Ill get to as soon as i can |
| 23:52.42 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-219-11.hsd1.al.comcast.net) [NETSPLIT VICTIM] |
| 23:53.01 | mvanbaak | mrtelnet: I can do your config. 65 euro/hour :) |
| 23:53.01 | [TK]D-Fender | mrtelnet, Please pastebin that entire context from your dialplan as I requested. |
| 23:53.27 | mrtelnet | this is going to sound horrible, but i cannot find from-internal |
| 23:53.57 | mvanbaak | it's in extensions.conf or extensions_additional.conf |
| 23:54.21 | [TK]D-Fender | mrtelnet, Call us when you find a CLUE. |
| 23:54.26 | mrtelnet | k |
| 23:54.28 | mrtelnet | sry |
| 23:54.47 | [TK]D-Fender | mrtelnet, You're picking a context for your call-file and you can't even find WHY YOU CHOSE IT! |
| 23:55.10 | mrtelnet | http://www.trixbox.org/forums/trixbox-forums/help/timed-announcements-paging |
| 23:56.18 | taxilian | does anyone have a suggestion for which the most powerful gui is? (yeah, I know it's better to do it by hand, command line... I just have to leave something my parents can figure out) |
| 23:57.00 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 23:57.08 | [TK]D-Fender | taxilian, thedecent ones cost. |
| 23:57.30 | taxilian | :-/ |
| 23:57.45 | taxilian | I'm trying to use asterisknow, but there are some basic things that I can't seem to do with it |
| 23:57.49 | taxilian | and nobody is in the forums right now |
| 23:58.34 | *** join/#asterisk techie (n=techie@adsl-76-247-10-106.dsl.lsan03.sbcglobal.net) |
| 23:58.36 | [TK]D-Fender | taxilian, Well thats not to say it isn't any good. So far that says that not enough people are around to answer your questeion, and you don't know any better what to do with it. |
| 23:59.10 | taxilian | yeah I know |
| 23:59.31 | taxilian | just frustrating =] I hadn't intended any criticism to them |
| 23:59.39 | taxilian | I'm just trying to find a solution to the problem |
| 23:59.54 | taxilian | I'm a programmer myself, just dont' have any voip experience. I know you can't always be everywhere |