| 00:00.26 | BiG^DoG | I'm building up my first asterisk box from scratch... centos 5 and asterisk 1.4.14... Do I still need to download mpg123 and compile it if I want mp3 music on hold? |
| 00:00.32 | BiG^DoG | or is it natively supported now? |
| 00:01.31 | Qwell | BiG^DoG: use native mog |
| 00:01.32 | Qwell | moh |
| 00:01.51 | BiG^DoG | ok...thx |
| 00:04.36 | angler | Qwell, you have been talking to mog to much |
| 00:04.49 | Qwell | g and h are too close together :( |
| 00:04.53 | Qwell | I do that all the time |
| 00:05.42 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:05.44 | fujin_ | native MOH is best; |
| 00:05.47 | fujin_ | don't use Mp3 though |
| 00:09.42 | *** join/#asterisk hawky (n=geoff@c-71-231-188-226.hsd1.or.comcast.net) |
| 00:13.58 | Siya | what would my syntax be if I needed an IF statement to check for a numbered extension less than say 5 digits? |
| 00:14.03 | `Sauron | native moh meaning .wav? |
| 00:14.35 | Siya | plenty of = and != statements but I need something to distinguish local from global enum |
| 00:19.14 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584588.dsl.bell.ca) |
| 00:27.26 | *** join/#asterisk flewid (n=flewid@mail.flewid.ca) |
| 00:35.51 | *** join/#asterisk macros73 (n=cs@c-24-3-246-27.hsd1.pa.comcast.net) |
| 00:39.46 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 00:40.08 | [hC] | Anyone know how to modify the polycom's in recent firmware versions so that when a second call comes in, you dont hear a call waiting beep in your ear, but instead the base of the phone rings? |
| 00:40.21 | [hC] | I know that this feature was introduced but i havent seen any docs on how to do it yet |
| 00:45.07 | [hC] | well nevermind i found it! |
| 00:45.09 | [hC] | :) |
| 00:47.25 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-5644b921ab733457) |
| 00:47.33 | tzanger | [hC]: how about sharing? |
| 00:49.03 | *** join/#asterisk coppice (n=chatzill@102.204.17.210.dyn.pacific.net.hk) |
| 00:49.08 | [hC] | <PROTECTED> |
| 00:49.08 | [hC] | <PROTECTED> |
| 00:49.08 | [hC] | <PROTECTED> |
| 00:49.12 | *** join/#asterisk jsaunders (n=super@d142-179-4-240.bchsia.telus.net) |
| 00:49.13 | [hC] | that normally says beep |
| 00:49.16 | [hC] | in phone1.cfg |
| 00:49.30 | tzanger | so you change it from beep to ring? |
| 00:49.51 | [hC] | yessir |
| 00:49.55 | tzanger | nice, thanks! |
| 00:49.58 | [hC] | what i pasted to you i have in an override file since i dont modify phone1.cfg |
| 00:50.01 | [hC] | but thats the place yes |
| 00:50.03 | tzanger | right |
| 01:33.26 | florz | is it possible that EAGI stops supplying incoming audio while executing a Playback? |
| 01:45.58 | *** join/#asterisk hawky (n=geoff@c-71-231-188-226.hsd1.or.comcast.net) |
| 01:49.53 | *** join/#asterisk bkw_ (n=brian@adsl-64-149-41-151.dsl.tul2ok.sbcglobal.net) |
| 01:52.42 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 01:55.55 | asdx | what is A()? |
| 01:56.11 | asdx | in the dialplan |
| 01:56.57 | bkw_ | ASS? |
| 01:58.41 | asdx | A(tt-monkeys) |
| 01:59.06 | [TK]D-Fender | bkw_, yup... YOU would think that ;) |
| 01:59.26 | asdx | A() == answer? |
| 01:59.29 | bkw_ | announce |
| 01:59.32 | bkw_ | help DIAL |
| 01:59.37 | bkw_ | or check the freakin wiki |
| 01:59.43 | rob0 | That's a syntax error: tt-monkeys must be preceded by tt-monkeys-intro. |
| 01:59.47 | bkw_ | good lord thats like first grade questions |
| 01:59.54 | [TK]D-Fender | asdx, and you should be speicif when reffering to a PARAMETER of a SPECIFIC application. |
| 02:00.25 | *** join/#asterisk Ryushin (n=Ryushin@windwalker.openinnovations.com) |
| 02:00.59 | rob0 | Is it true that tt-monkeys was recorded in Digium's break room? |
| 02:01.17 | Qwell | we have a break room? |
| 02:01.19 | asdx | [TK]D-Fender: ok |
| 02:01.36 | rob0 | well, the junk room with a cluttered table |
| 02:01.37 | file | Qwell: on the roof. |
| 02:01.44 | Qwell | file: your office? |
| 02:01.51 | file | Qwell: beside it |
| 02:01.54 | Qwell | ahh |
| 02:04.37 | *** join/#asterisk Galleon (i=ryan@c-98-197-41-138.hsd1.tx.comcast.net) |
| 02:07.17 | *** join/#asterisk [zender] (n=ender@72.52.23.4) |
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| 02:19.59 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-4717712428afed0b) |
| 02:21.18 | linagee | if there's a lot of cell calls going on, does it degrade cell quality? |
| 02:21.24 | linagee | (GSM) |
| 02:22.44 | *** join/#asterisk Strom_M (n=strom@ip70-170-60-8.lv.lv.cox.net) |
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| 02:32.00 | *** mode/#asterisk [+o russellb] by ChanServ |
| 02:33.19 | florz | How would you go about streaming audio in an AGI bidirectionally? Or at least play some sound file while still receiving the incoming audio? |
| 02:36.24 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 02:36.50 | *** join/#asterisk bkw_ (n=brian@adsl-64-149-41-151.dsl.tul2ok.sbcglobal.net) |
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| 02:40.46 | BigCanOfTuna | I need to jump to various extensions based on (I think) the TrySystem or System command...does anyone have a good example of how it works? |
| 02:41.02 | BigCanOfTuna | Specifically, I need to know how to check return codes. |
| 02:44.07 | BigCanOfTuna | Ah, I got it...needed to look at SYSTEMSTATUS |
| 02:46.13 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584588.dsl.bell.ca) |
| 02:50.44 | *** join/#asterisk mrtelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 02:50.52 | mrtelephone | i wonder where all the docsis nerds hang out |
| 02:53.10 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-2ba315d950d2db38) |
| 02:54.17 | *** join/#asterisk jetlagmk2 (i=jetlag@pool-70-17-37-23.pskn.east.verizon.net) |
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| 03:01.25 | *** part/#asterisk BigCanOfTuna (n=chatzill@dsl-mac-66-18-226-119-cgy.nucleus.com) |
| 03:07.43 | BiG^DoG | OK... I know I'm begging to be ridiculed but here goes... I'm following along in the * o'reilly book... just installed fresh * install, configured a basic sip device in sip.conf and tried to get my xlite device (which connects to my live * box fine) to talk to this new * server and get a 408 timeout on registration. Am I missing something? |
| 03:09.05 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 03:09.05 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:10.12 | [TK]D-Fender | BiG^DoG, timeout tells you that someones asking and not even getting an answer. This says "networking issue". Around here the first guess is that NAT's involved and you didn't set your system up right. |
| 03:10.49 | BiG^DoG | sip phone is 192.168.1.102 and asterisk is 192.168.1.108 |
| 03:11.04 | BiG^DoG | iptables turned off on * server and local firewall turned off on my xlite machine |
| 03:11.21 | BiG^DoG | I agree it sounds like networking but I'll be damned if I can figure out where |
| 03:11.54 | [TK]D-Fender | BiG^DoG, check your sip.conf. enable SIP debug and restart the phone to see what packets come in/outr |
| 03:12.17 | BiG^DoG | set sip debug ip 192.168.1.102, right? |
| 03:12.33 | BiG^DoG | backwards |
| 03:12.36 | [TK]D-Fender | "sip debug" grab everything until you know whats wrong |
| 03:12.37 | BiG^DoG | sip set debug ip |
| 03:12.57 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 03:13.47 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-7520958c2583f2c9) |
| 03:15.00 | BiG^DoG | interesting... no sip traffic at all |
| 03:16.28 | *** join/#asterisk [hC] (n=hardcore@66.119.172.82) |
| 03:16.54 | [TK]D-Fender | BiG^DoG, start pinging and check everything else. dump your iptables, etc |
| 03:17.03 | mrtelephone | the host is 5 bits for 224 |
| 03:17.28 | BiG^DoG | can't telnet to * on 5060 cause it's udp only, right? |
| 03:17.28 | Nugget | telnet is eeeeeeevil! |
| 03:17.30 | mrtelephone | 0.0.0.31? |
| 03:18.40 | Maliuta | Nugget: telnet is a useful tool |
| 03:19.00 | Maliuta | Nugget: if you can't speak http/smtp/dns then it probably is evil |
| 03:19.15 | Maliuta | BiG^DoG: yes, SIP is all UDP |
| 03:19.59 | mrtelephone | sip can be tcp as well sometimes |
| 03:20.09 | mrtelephone | udp and sip is a crappy combo |
| 03:20.45 | BiG^DoG | gonna completely turn off iptables and reboot |
| 03:21.01 | BiG^DoG | I'm still not seeing any sip traffic on the * box but I can ping the * box from the xlite box |
| 03:21.42 | rob0 | nc(1) can do UDP, but you won't get a TCP-like connection. |
| 03:22.08 | rob0 | nmap -sU can also tell if a UDP port is open. |
| 03:22.31 | BiG^DoG | the port's open -- I checked that... I just want to see if I can connect to it from another machine |
| 03:24.15 | *** join/#asterisk ManxPower (n=manxpowe@34.sub-70-196-71.myvzw.com) |
| 03:30.23 | mrtelephone | <mrtelephone> pay me and I will service you :P |
| 03:30.23 | mrtelephone | <mrtelephone> I am fully unqualified |
| 03:31.47 | mrtelephone | bigdog, netstat -ln | grep 5060 |
| 03:32.00 | mrtelephone | if it listens on 0.0.0.0 it will accept connections from anywhere |
| 03:40.48 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 03:40.49 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:45.55 | *** join/#asterisk ming_zy1 (i=ming_zym@nat/yahoo/x-e92ccdf09c7ec891) |
| 03:47.56 | ManxPower | actually, it will accept connections TO any ip address on the machine |
| 03:50.36 | BiG^DoG | I blew it away and am going to start over |
| 03:50.45 | BiG^DoG | practice makes perfect |
| 03:50.57 | [TK]D-Fender | BiG^DoG, blew what away? |
| 03:51.04 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 03:51.04 | mrtelephone | yeah whatever |
| 03:51.08 | mrtelephone | it will accept stuff |
| 03:51.14 | BiG^DoG | my asterisk install |
| 03:51.15 | mrtelephone | from other palces |
| 03:51.19 | Grizzy | I think "nc" == "netcat" has a UDP mode, doesn't it? |
| 03:51.28 | mrtelephone | sure |
| 03:51.43 | mrtelephone | grizzy, try ngrep |
| 03:51.47 | [TK]D-Fender | BiG^DoG, You know, for all the talk and asking for help... you didn't once show us anything of value... |
| 03:51.59 | mrtelephone | ngrep -W byline port 5060 |
| 03:52.03 | JT | Maliuta: no point talking to the bot :P |
| 03:52.09 | mrtelephone | or are you trying to send stuff? |
| 03:52.41 | Grizzy | ngrep sounds yummy. : o ) |
| 03:52.46 | BiG^DoG | [TK]D-Fender, you're right... I didn't. |
| 03:53.01 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 03:53.01 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 03:53.32 | mrtelephone | ngritz |
| 03:53.40 | [TK]D-Fender | BiG^DoG, hit for next time : pastebin everything. SIP configs, debug CLI, iptables dumps, ifconfig, etc. |
| 03:53.46 | mrtelephone | hmm |
| 03:53.56 | mrtelephone | whats the best way for checking to see if asterisk is accepting connections |
| 03:53.59 | mrtelephone | rmon? |
| 03:54.12 | BiG^DoG | I will do that |
| 03:54.18 | JT | nmap? |
| 03:54.20 | JT | netcat? |
| 03:54.25 | JT | netstat |
| 03:54.27 | *** join/#asterisk mihinomenest (i=G7tx@66.255.220.17) |
| 03:54.31 | JT | lsof |
| 03:54.38 | mrtelephone | netcat sounds interesting |
| 03:54.43 | mrtelephone | you can send a registration attempt |
| 03:54.50 | mrtelephone | or something |
| 03:55.16 | JT | if it's sip specific, then sipp can test |
| 03:55.23 | *** join/#asterisk the007killer (n=the007ki@61.29.2.98) |
| 03:55.46 | the007killer | hi everyone |
| 03:56.16 | mrtelephone | nice |
| 03:56.46 | the007killer | i have a problem with my asterisk server, and can't work it what to do to fix it, can anyone help? |
| 03:56.50 | mrtelephone | i phoned a guy one time.. he was in the RINGING stage or just finished a call and I got a busy signal when I phoned him |
| 03:56.54 | mrtelephone | a fast busy |
| 03:57.06 | mrtelephone | i have to check it out |
| 03:57.26 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 03:57.41 | mrtelephone | asterisk likes to send busy when callee is RINGING another party |
| 03:57.49 | mrtelephone | but once the call is established then callwaiting works |
| 03:57.58 | *** join/#asterisk webman (n=adamg@124.246.8.196.static.nexnet.net.au) |
| 03:58.25 | *** join/#asterisk [hC] (n=hardcore@66.119.172.82) |
| 03:59.23 | the007killer | anyone know how to get mysql support to work |
| 03:59.29 | the007killer | i have a strange error |
| 03:59.30 | mrtelephone | pretty broad question |
| 03:59.35 | the007killer | i don't know how to fix it |
| 04:00.33 | mrtelephone | whats the error for starters |
| 04:02.55 | the007killer | res_config_mysql.c:627 mysql_reconnect: MySQL RealTime: Failed to connect database server on (err 2002). Check debug for more info. |
| 04:03.07 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 04:04.33 | mrtelephone | make sure u can connect to mysql database using the command line |
| 04:05.07 | the007killer | i can |
| 04:05.22 | the007killer | but im now sure which config file it is using to get the settings |
| 04:05.33 | the007killer | not* |
| 04:05.46 | mrtelephone | not sure either |
| 04:05.49 | mrtelephone | did you read voipinfo? |
| 04:06.01 | the007killer | its where i got the address for IRC |
| 04:06.20 | the007killer | i suppose i can go through it again |
| 04:06.20 | [TK]D-Fender | ~book |
| 04:06.21 | 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 |
| 04:07.45 | mrtelephone | sometimes i have to reread shit 02435-0432 times |
| 04:08.02 | [TK]D-Fender | 867-5309~ |
| 04:08.24 | file | Jenny? |
| 04:09.19 | [TK]D-Fender | uNF! |
| 04:17.39 | the007killer | do you know where this debug file is? |
| 04:17.39 | the007killer | MySQL RealTime: Failed to connect database server on (err 2002). Check debug for more info. |
| 04:20.54 | mrtelephone | try /var/log/asterisk/debug |
| 04:23.35 | [TK]D-Fender | try looking at your * configs. |
| 04:24.19 | [TK]D-Fender | and ask yourself why it doesn't list where its looking for a server in the first place |
| 04:25.51 | *** join/#asterisk PepOSX (n=pepOSX@190.72.148.12) |
| 04:27.27 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-68-93-83-7.dsl.hstntx.swbell.net) |
| 04:38.32 | the007killer | thats a good point |
| 04:40.32 | the007killer | there isn't a debug file in /var/log/asterisk or anywhere in the /var/log folder |
| 04:41.58 | \2Legit | does anyone here happen to own an atm? |
| 04:42.57 | *** join/#asterisk Nukemizer (n=Nukemize@15.249.sfcn.org) |
| 04:43.05 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 04:47.37 | flenders | hey, does anyone know if I can change a voicemail password without going through the voicemail menus? |
| 04:48.19 | flenders | and not through the voicemail.conf file either |
| 04:52.27 | [TK]D-Fender | flenders, nothing normal. |
| 04:52.42 | [TK]D-Fender | flenders, How do you have in mind? |
| 04:53.42 | flenders | [TK]D-Fender: calling a number, that will just ask for your current password, and after you type it in, it asks for the new password, like option 5 on the voicemail prompts |
| 04:54.00 | flenders | but just says "password changed" and return to previous menu |
| 04:54.22 | De_Mon | flenders well, uh, you could dial an extension that calls voicemail main and then sends some dtmf codes to get you to the change password menu? |
| 04:54.33 | [TK]D-Fender | flenders, And whats bad about going into your box and just changing it like normal? |
| 04:54.47 | flenders | De_Mon: hm, yeah, I guess I could do that |
| 04:54.58 | [TK]D-Fender | De_Mon, I was waiting on that :) |
| 04:55.27 | De_Mon | flenders you are still crazy |
| 04:55.50 | flenders | is there an easier way to change passwords on meetme rooms? |
| 04:56.06 | flenders | I was gonna authenticate using voicemail passwords |
| 04:56.15 | De_Mon | [TK]D-Fender nana I beat you |
| 04:56.25 | flenders | on a different voicemail context |
| 04:56.31 | [TK]D-Fender | De_Mon, I didn;t want to come out and say it is all. |
| 04:56.51 | De_Mon | flenders then do an Auth() before sending them to the conference. |
| 04:56.54 | JT | it's probably more crazy how inflexible asterisk's voicemail system |
| 04:58.06 | flenders | De_Mon: you mean Authenticate()? |
| 04:58.16 | De_Mon | probably |
| 04:58.25 | flenders | De_Mon: how do you change that password though? |
| 04:58.47 | *** join/#asterisk ghento (n=ghento@d216-232-170-111.bchsia.telus.net) |
| 04:58.48 | flenders | I just want to give users a way of changing their personal conf rooms passwords whenever they want |
| 04:58.49 | [TK]D-Fender | flenders, use your IMAGINATION as to what you will tel Authenticate to accept <--- |
| 04:58.55 | De_Mon | flenders um |
| 04:59.34 | De_Mon | flenders you should use func_odbc to query a database and get the password and another extension to set the database record. |
| 04:59.54 | De_Mon | make sure to get a processor with atleast 4 cores |
| 04:59.59 | [TK]D-Fender | De_Mon, And you're suggesting running a SQL DB for a meetme room? :) |
| 05:00.08 | De_Mon | so you can dedicate atleast one for this task... |
| 05:00.15 | De_Mon | [TK]D-Fender he asked! |
| 05:00.15 | [TK]D-Fender | ... |
| 05:00.21 | flenders | De_Mon: ha |
| 05:00.23 | [TK]D-Fender | *cough* |
| 05:00.42 | De_Mon | thats one core, incase you didn't catch it |
| 05:00.48 | flenders | De_Mon: that's a bit too much, don't you think? |
| 05:01.00 | De_Mon | maybe, but it'd work |
| 05:01.03 | flenders | what's bad about using VM passwords to authenticate conferences? |
| 05:01.05 | [TK]D-Fender | flenders, Ok, HALF a core, and that's De_Mon final offer! |
| 05:01.15 | flenders | :D |
| 05:01.31 | [TK]D-Fender | flenders, that works. |
| 05:02.16 | flenders | [TK]D-Fender: yeah, I know it does, as I'm doing it, but just wanted to see if there was a way to change the passwords without going to voicemailmain |
| 05:03.21 | [TK]D-Fender | flenders, so first you want to use Vm's PW for Meetme, and then you don't want to use VMM's normal means to change its own PW. Do you realize just how retarded that sounds? :) |
| 05:04.05 | [TK]D-Fender | flenders, Apparently every problem is a nail.... so its time for you to pull out.. a CHAIN-SAW! |
| 05:05.03 | [TK]D-Fender | flenders, And frankly you coudl do Vm without even using VM's pass, and just Authenticate for BOTH. |
| 05:07.03 | flenders | [TK]D-Fender: I just didn't want to have to build all the menus to change passwords stored on asterisk DB |
| 05:07.24 | flenders | as it would be a lot easier to just change it on VM |
| 05:07.33 | [TK]D-Fender | flenders, consumately lazy, yup! :) |
| 05:08.27 | flenders | [TK]D-Fender: simplicity. |
| 05:08.27 | [TK]D-Fender | flenders, that'd be 20 lines tops. |
| 05:09.23 | [TK]D-Fender | flenders, meanwhile you're trying to get every app to do ANOTHER apps job. Yeah, that's "simple" |
| 05:10.09 | flenders | man, I don't know what's wrong with you. it was just a simple question. don't know why all questions here have to become a debate. |
| 05:13.15 | [TK]D-Fender | flenders, Well lets answer it quick. No there isn't a 1-setp app to change a VM pass outside of VMM. |
| 05:13.52 | flenders | [TK]D-Fender: great! see... much better, isn't it? thanks |
| 05:14.06 | [TK]D-Fender | *sigh* |
| 05:20.44 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-68-93-83-7.dsl.hstntx.swbell.net) |
| 05:20.47 | *** part/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 05:24.50 | *** join/#asterisk anujsingh (n=anuj@59.90.65.14) |
| 05:24.59 | anujsingh | hello |
| 05:27.07 | anujsingh | is there someway possible to play dialled call recordings with soft phone(something lik Xlite)to |
| 05:31.35 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 05:32.39 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 05:42.49 | *** join/#asterisk [zender] (n=ender@72.52.23.3) |
| 05:46.21 | *** join/#asterisk osiris (n=osiris@c-71-205-29-230.hsd1.mi.comcast.net) |
| 05:54.22 | *** join/#asterisk osiris (n=osiris@c-71-205-29-230.hsd1.mi.comcast.net) |
| 05:59.54 | *** join/#asterisk osiris (n=osiris@c-71-205-29-230.hsd1.mi.comcast.net) |
| 06:14.00 | De_Mon | what is autoanswer? |
| 06:19.27 | De_Mon | oh, great.. its something debian added |
| 06:25.54 | De_Mon | omg omg omg autoanswer >>> parking a call and picking it up! |
| 06:28.42 | the007killer | does anyone here use the linksys SIP phones with their system? |
| 06:30.32 | De_Mon | suure |
| 06:31.12 | SwK | lotta people use liksys with their asterisk boxen |
| 06:38.14 | *** join/#asterisk jamesrdorn (n=jamesdor@adsl-75-63-124-255.dsl.rcsntx.sbcglobal.net) |
| 06:40.44 | jamesrdorn | hey guys, I need a little assistance trying to get the current sip ext # in VoiceMailMain so it does not ask for a mailbox. I can force one extension by VoiceMailMain(1000), but have not found a way to generate the current caller |
| 06:40.47 | jamesrdorn | any ideas? |
| 06:41.04 | jamesrdorn | in my setup, the sip ext is the mailbox number |
| 06:43.13 | jamesrdorn | so maybe if I understood why there is a "mailbox" veriable for the extensions.conf, I could use it effectivly |
| 06:44.31 | SwK | jamesrdorn, why not just do something like VoicemalMain(${CALLERID(num}) |
| 06:45.51 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:46.58 | jamesrdorn | Executing [100@internal:3] VoiceMailMain("SIP/1000-081b2038", "") in new stack |
| 06:47.03 | jamesrdorn | returns nothing in that string |
| 06:47.15 | jamesrdorn | however, in the sip config, I do specify the callerid |
| 06:47.36 | jamesrdorn | interesting enough |
| 06:47.50 | jamesrdorn | [Nov 22 00:46:37] ERROR[5467]: pbx.c:1523 ast_func_read: Function CALLERID not registered |
| 06:48.06 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:51.04 | kaldemar | jamesrdorn: what version of asterisk are you running? |
| 06:51.49 | jamesrdorn | Asterisk 1.4.9 |
| 06:54.39 | kaldemar | have you deliberately compiled asterisk without all functions? |
| 06:58.10 | jamesrdorn | kaldemar: I did not exclude any functions that I know of. voicemail and meetup work fine |
| 06:58.29 | jamesrdorn | or meetme |
| 06:58.31 | jamesrdorn | rather |
| 06:58.45 | kaldemar | voicemail and meetme are applications, not functions. |
| 06:59.24 | jamesrdorn | kaldemar, I just did a general compile with no flags. |
| 06:59.35 | kaldemar | do you have func_callerid.so in /usr/lib/asterisk/modules/ ? |
| 07:00.00 | Siya | anyone who can point me to a good source for dialplan logic? |
| 07:00.03 | kaldemar | if you do, check that you don't have a noload in /etc/asterisk/modules.conf for it. |
| 07:00.11 | jamesrdorn | it's there |
| 07:00.16 | Siya | I need more than a mere = or != |
| 07:00.18 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 07:01.05 | jamesrdorn | load => func_callerid.so ; Caller ID related dialplan functions |
| 07:01.26 | jamesrdorn | ah |
| 07:01.28 | jamesrdorn | just fixed it |
| 07:01.39 | jamesrdorn | I had set it to load earlier |
| 07:01.56 | jamesrdorn | but just did a reload on the config, aparently I needed to restart asterisk totally |
| 07:01.59 | jamesrdorn | works fine now |
| 07:04.19 | *** join/#asterisk zeeesh (n=zeeesh@202.166.161.45) |
| 07:05.13 | jamesrdorn | so, I still have to beg the question... what is the use of the mailbox string in the sip.conf, just good book keeping? |
| 07:06.43 | *** join/#asterisk red9012 (n=marc3234@76-10-149-62.dsl.teksavvy.com) |
| 07:06.44 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 07:10.32 | *** join/#asterisk slavon_net (n=slavon@slavon.bigtelecom.ru) |
| 07:11.27 | kaldemar | jamesrdorn: it tells which box to check for new messages for a possible MWI to the user. |
| 07:12.14 | jamesrdorn | MWI meaning a data stream letting the client phone know that there is a voicemail waiting? |
| 07:12.36 | kaldemar | yes |
| 07:12.41 | jamesrdorn | or rather, the phone checks asterisk for this function durring re-registration |
| 07:12.44 | jamesrdorn | ok |
| 07:12.45 | kaldemar | message waiting indication |
| 07:12.46 | jamesrdorn | understood |
| 07:12.50 | jamesrdorn | nice |
| 07:12.51 | jamesrdorn | ok |
| 07:12.53 | jamesrdorn | make sence |
| 07:12.57 | jamesrdorn | Thanks for that |
| 07:15.01 | *** join/#asterisk gerphimum (n=trekkie@cpe-70-125-148-108.satx.res.rr.com) |
| 07:16.40 | slavon_net | hi all. simple question... how to get retrun value of app? |
| 07:16.59 | *** join/#asterisk vgster (n=vgster@psc.navonline.net) |
| 07:17.23 | slavon_net | example app_xxx may rerurn -1 and 0... how to check returned value? =( |
| 07:19.29 | slavon_net | or its only say to dialplan that it can continius ? |
| 07:22.53 | Siya | no-one with dailplan if boulean knowledge? |
| 07:23.34 | kaldemar | Siya: http://www.voip-info.org/wiki/view/Asterisk+Expressions |
| 07:30.57 | Siya | kaldemar: brilliant, I knew it was out there was just not finding it on the wiki :) |
| 07:32.10 | Grizzy | could we please have a lisp-like configuration language, instead of inventing yet another syntax? : o ) |
| 07:33.55 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-79-197.lns10.syd6.internode.on.net) |
| 07:36.38 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 07:39.12 | *** join/#asterisk mathesis (n=kukako@unaffiliated/mathesis) |
| 07:39.12 | *** join/#asterisk rati (n=rati@124.125.255.223) |
| 07:43.05 | *** join/#asterisk ghento (n=ghento@d216-232-170-111.bchsia.telus.net) |
| 07:45.22 | *** join/#asterisk mirco (n=mirco@pd95b6029.dip0.t-ipconnect.de) |
| 07:51.41 | Siya | would this be correct if I wanted to test the first two digits of the callerid? |
| 07:51.44 | Siya | exten => s,1,GoToIf($["${CALLERID(num)}" : "00"]?true:false) |
| 07:52.09 | *** join/#asterisk Op3r (n=edwin@222.127.86.171) |
| 07:52.21 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 07:52.28 | Siya | "The regular expression is anchored to the beginning of the string with an implicit '^'." |
| 07:58.01 | *** join/#asterisk tehfox (n=tehfox@netlab-162.netlab.kis.fri.utc.sk) |
| 08:01.24 | Siya | cool if this works then I've set my default callerid on all outbound calls except if the extension has a callerid set that starts with 00 meaning it's a qualified enum (or user error...) |
| 08:01.54 | Siya | so I can keep my internal callerid's equal to the extension numbers |
| 08:01.56 | Siya | :) |
| 08:02.02 | Siya | tx kaldemar |
| 08:02.21 | *** join/#asterisk harpal (n=Harpal@124.125.255.223) |
| 08:02.56 | Siya | tx fujin_ |
| 08:08.28 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 08:11.29 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-1e32e3a594e992ed) |
| 08:12.20 | *** join/#asterisk Bananaskin (n=Banana@aop54.internetdsl.tpnet.pl) |
| 08:12.56 | badcfe | is the evadetranscoding patch incorporated into * 1.4 ? |
| 08:13.28 | harpal | I want to give my voice before it go to put that in to voice mail. What should I do? |
| 08:14.51 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 08:15.16 | kaldemar | harpal: rephrase your question. you're not being clear on what you want to do. |
| 08:17.02 | harpal | kaldemar, I want that some one can call me If I am not available than it play that msg I have puted over there (like, I am not available please leave message) and than calling person leave msg and than it goes to mailbox |
| 08:17.41 | harpal | kaldemar, Can you get what I mean? |
| 08:17.43 | kaldemar | well, you put the message after the dial in your dialplan, and voicemail after the message. |
| 08:18.18 | kaldemar | doesn't voicemail have a feature for that? |
| 08:18.40 | jamesrdorn | kaldemar: the VoiceMailMain app has a built in sound recorder to replace the default "leave a message" tone |
| 08:19.33 | jamesrdorn | nite |
| 08:20.28 | badcfe | is it possible to enable algorithm that evade transcoding when preparing bridge of to sip chan's? |
| 08:21.49 | harpal | jamesrdorn, so to leave that msg I have to configure that my voice mailserver? |
| 08:23.24 | kaldemar | are you using asterisk's internal voicemail? if so, just dial your own voicemail and use the IVR to record your message. |
| 08:23.30 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:24.50 | harpal | kaldemar, I dont know more about that, because I am new be. but in voicemail.conf I have given my yahoo mail id |
| 08:25.16 | harpal | So Is that mean i am using yahoo's voice mail? |
| 08:25.29 | kaldemar | then you should study the whole voicemail concept and start configuring when you know what you're doing. |
| 08:27.28 | *** join/#asterisk bakermd (n=none@72.5.80.5) |
| 08:30.00 | *** join/#asterisk obnauticus (n=obnautic@c-71-236-181-11.hsd1.or.comcast.net) |
| 08:30.08 | obnauticus | is there some type of web-based meetme managment system? |
| 08:31.40 | harpal | kaldemar, ok |
| 08:36.03 | bakermd | I am trying to use realtime for voicemail config, and it is seeing the users I have in the DB, however when it goes to record a message it hangs up with the error app_voicemail.c:3145 leave_voicemail: No format for saving voicemail? |
| 08:36.31 | bakermd | And I put a column in the table for "format" which is set to "wav|wav49" |
| 08:36.35 | bakermd | Any ideas? |
| 08:42.31 | *** join/#asterisk cypherdelic (n=cypher@p5B27CB7B.dip.t-dialin.net) |
| 08:50.44 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584588.dsl.bell.ca) |
| 08:59.27 | *** join/#asterisk dmz (n=dmz@64.253.5.180.dyn-cm-pool75.pool.hargray.net) |
| 08:59.45 | dmz | howdy y'all, is anyone here use multiple FWD accounts w/1 asterisk box? and if so, what's the trick? |
| 08:59.58 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 09:01.27 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
| 09:10.36 | *** join/#asterisk duckz (n=duckz@85.204.47.228) |
| 09:10.44 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 09:10.56 | obnauticus | Has anyone here used webmeetme/CBmySQL? |
| 09:11.10 | obnauticus | (conference bridging between Asterisk in MySQL (no idear what it is)). |
| 09:14.07 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:19.06 | *** join/#asterisk mkl1525 (n=qwertz@89.246.184.245) |
| 09:22.18 | mkl1525 | Hi, is there a cli way to direct callers in a queue directly to an agent? we've sometimes the problem that agents don't ring although there are callers in queue |
| 09:32.50 | *** join/#asterisk anujsingh (n=Administ@59.94.134.54) |
| 09:32.57 | anujsingh | hello all |
| 09:33.24 | anujsingh | i want to play recorded files of /var/spool/monitor/DONE threw soft phone. |
| 09:33.57 | *** join/#asterisk rati (n=rati@124.125.255.223) |
| 09:33.58 | ronr | how can I dial multiple extensions in a dialplan (at once and stop dailing if one of them answers)? |
| 09:34.32 | anujsingh | can someone guide me , target is to develop something which can play recorded files in /var/spool/asterisk/DONE folder? |
| 09:34.59 | anujsingh | i used apache web interface so that one can access recorded files. |
| 09:34.59 | mvanbaak | ronr: Dial(SIP/1&SIP/2&SIP/3) |
| 09:35.38 | ronr | mvanbaak: thanks |
| 09:35.48 | anujsingh | can i play recorded files with soft phone? |
| 09:36.38 | anujsingh | i am trying to create a voicemailbox linked to /var/spool/asterisk/monitor/DONE folder |
| 09:36.52 | obnauticus | mkl1525 you can originate the call. |
| 09:37.46 | anujsingh | i wonder if this thing someone has already done and to save myself from reinventing the wheel. |
| 09:37.47 | *** join/#asterisk _ys (n=yuri@236-069.nat.mns.ru) |
| 09:37.55 | *** join/#asterisk MrMister2 (n=mrmister@195-23-221-215.net.novis.pt) |
| 09:38.06 | anujsingh | to play the monitor/DONE sound recording via softphone |
| 09:38.27 | mvanbaak | anujsingh: you'll have to use Playback and some AGI to present a list of files |
| 09:38.45 | MrMister2 | Hi. Can anyone recomend a ISDN BRI card to connect to Asterisk? I'm looking for something cheap, less than 100USD if possible. |
| 09:39.18 | MrMister2 | I've seen Billion and AVM cards, any advice on those? |
| 09:45.56 | anujsingh | mvanbaak can i have link to the relative page? |
| 09:46.11 | *** join/#asterisk harpal (n=Harpal@124.125.255.223) |
| 09:47.23 | mkl1525 | obnauticus thanks will have a deeper look at it, but as far as I sees this is for AMI not for the cli? |
| 09:47.30 | *** join/#asterisk ai-a (n=jake2@megan.healthnet.co.uk) |
| 09:49.25 | MrMister2 | None has any advice or tips on cheap BRI ISDN cards? |
| 09:50.12 | obnauticus | mkl525 yes. |
| 09:50.38 | obnauticus | I don't know how to originate via AMI but you can, it's easier via asterisk's console. |
| 09:51.19 | ai-a | MrMister2: www.ebay.com |
| 09:51.22 | obnauticus | originate technology/extension application <application> [arg1] [arg2] |
| 09:57.28 | MrMister2 | ai-a: Thanks, but it wasn't so much _where_ to buy as _what_ to buy :) |
| 09:57.53 | *** join/#asterisk ReD-MaN (i=root-rox@172-220.static.golden.net) |
| 09:57.54 | ai-a | http://www.asteriskguru.com/tutorials/bri.html |
| 09:58.00 | ai-a | MrMister2: GIYF |
| 09:58.37 | *** join/#asterisk sergey (n=sergey@91.189.233.71) |
| 09:58.46 | obnauticus | Man i need an ISDN BRI :\ |
| 10:00.19 | anujsingh | thanks mvanbaak |
| 10:00.29 | *** join/#asterisk defswork (n=andy@77.44.54.34) |
| 10:02.47 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-9b1432262e25679a) |
| 10:06.27 | *** join/#asterisk qdk (n=qdk@85.235.253.139) |
| 10:10.46 | MrMister2 | ai-a: Yep. I tried Google but most of the stuff I found is for PRI, not BRI and the rest is motly ppl moaning on how ISDN support on Asterisk is crap :( |
| 10:11.17 | ai-a | well then theres your answer. |
| 10:11.32 | JT | MrMister2: does the BRI in your area use ETSI signalling? |
| 10:14.42 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 10:17.04 | anujsingh | can i have tutorial link for writing agi scripts? |
| 10:18.14 | tzafrir_home | ai-a, that link is outdated |
| 10:18.28 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 10:18.36 | tzafrir_home | lots of misleading comments |
| 10:20.29 | ai-a | we're all outdated and obsolete |
| 10:20.53 | JT | eh |
| 10:21.07 | JT | ai-a: so in reality you don't know much about setting up a bri then? |
| 10:21.26 | ai-a | have no idea ;) |
| 10:21.44 | ai-a | virtually i might have done it in a game once. |
| 10:26.06 | *** join/#asterisk harpal (n=Harpal@124.125.255.223) |
| 10:28.48 | mvanbaak | anujsingh: depends on the language in what you want to create the agi in |
| 10:29.04 | *** join/#asterisk vgster (n=vgster@psc.navonline.net) |
| 10:29.41 | *** join/#asterisk rati (n=rati@124.125.255.223) |
| 10:31.19 | MrMister2 | JT: I'm in Portugal and I have no idea if it uses ETSI or not :). Let me see if Google returns anything. |
| 10:31.55 | *** join/#asterisk achu (n=achu@125.17.244.2) |
| 10:32.14 | JT | MrMister2: chances are yes |
| 10:32.37 | MrMister2 | JT: http://portal.etsi.org/at/TRAC/ATAAB/Advisory%20Notes/an01r000.pdf |
| 10:33.04 | anujsingh | mvanbaak i want to play recorded files in monitor/DONE folder via soft phone |
| 10:33.29 | MrMister2 | JT: I think so since that document specifies it. Again, any advice on a cheap BRI card? I was looking for something on the order of 100USD. I've seen mention of Billion and AMV Fritz cards. |
| 10:33.47 | achu | I have to asterisk boxes, which is inter connected using IAX |
| 10:33.53 | achu | if a caller caller on box1 he is not able to get the Name Directory of the second box |
| 10:33.57 | JT | MrMister2: yeah anything that uses the HFC-s is usually pretty cheap |
| 10:34.00 | achu | how we can configure the box to get both server's Name Directory ? |
| 10:34.18 | achu | any configuration need to be changed ? |
| 10:34.30 | anujsingh | i dont know how to do this , i used apache web page linked to the folder where a user can download files, but have to develop someway ,so that a user can listen to recorded files via soft phone. |
| 10:34.39 | MrMister2 | My problem is that I have 0 experience with ISDN and Asterisk, only analog and Asterisk, so was looking for more experienced ppl to give me some pointers on cheap cards and if I should avoid any. |
| 10:34.40 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 10:34.46 | bintut | hello all.. |
| 10:34.47 | mkl1525 | just read that "retry = 5" in queue.conf should ring all available agents on that queue after 5 seconds - is this correct? seems not to work for my configuration |
| 10:34.54 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 10:35.04 | JT | MrMister2: using bristuff would be ideal |
| 10:35.18 | achu | MrMister2, can you help me ? |
| 10:35.37 | MrMister2 | JT: Are there any known issues with HFC chipsets? voice quality, caller id not working, etc...? |
| 10:35.53 | bintut | i am following the book "Asterisk: The Future of Telephony 2nd Edition" on connecting 2 asterisk boxes together via iax but i can't seem to call the phones on the other side. |
| 10:35.54 | ai-a | mkl1525: hmm, the file say "retry = 5; //How long do we wait before trying all the members again?" so what do you think ? |
| 10:36.03 | JT | MrMister2: no, callerid works fine. it is digital. |
| 10:36.30 | JT | MrMister2: the main issues are the cheap ones are single port only, and no hardware echo cancellation |
| 10:37.43 | MrMister2 | achu: I have 2 Asterisk boxes connected with AIX, yes. I had that very same issue and the only way I found to make it work (didn't try very hard :)) was to replicate the NameDir from one box to another since I would get a empty string from one box to another on the caller id. |
| 10:37.47 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 10:39.18 | achu | MrMister2, you mean copy the extensions from extensions.conf to the second server ? |
| 10:39.20 | MrMister2 | JT: I only need one port (2 channels). mmm... the hardware echo cancellation is more concerning. Could I use one of the echo cancellation software on the market? If it's only 10 or 15USD I don't mind paying for it. |
| 10:40.07 | *** join/#asterisk FreezeS (n=bruno@82.208.157.125) |
| 10:40.33 | mkl1525 | ai-a problem is that it doesn't work for us in this way so I thought I missed something |
| 10:40.41 | FreezeS | hey guys |
| 10:41.15 | ai-a | mkl1525: can you show us the cli output while using the queue... |
| 10:41.15 | FreezeS | there is a problem with the redirector on the asterisk.org site |
| 10:43.03 | bintut | anyone cares to check my current configurations of my 2 asterisk boxes at http://www.privatepaste.com/113AdU24gK |
| 10:43.52 | bintut | i must admit that i see both boxes are registered to each other by checking the command of "iax2 show registry" on the asterisk shell.. |
| 10:44.48 | ai-a | privatepaste - Yet Another Pasting Site Wanting Donations ;) |
| 10:45.23 | achu | MrMister2, any idea ? |
| 10:45.51 | bintut | but whenever i tried to call from my sip phone A registered to pbx A to sip phone B registered to pbx B or the other way around, i always get a warning message like "unable to create channel type iax2" or "no such host: b" |
| 10:46.13 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-f6113de94b2be266) |
| 10:46.49 | bintut | what's wrong with my setup that i can't make calls to other's end? |
| 10:48.25 | Mw3 | bintut: your iax.conf is wrong |
| 10:49.01 | bintut | Mw3: which part? |
| 10:49.11 | MrMister2 | achu: sorry, was AFK. No, I meant the astdb where the caller id names are stored. see http://www.voip-info.org/wiki-Asterisk+Database and http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+LookupCIDName |
| 10:49.13 | Mw3 | bintut: at least the peers in dialplan and in the iax.conf does not match. in dialplan you wrote Dial(IAX/a/...) but in iax.conf you hava b as peer |
| 10:49.41 | bintut | ok, i'll try that.. |
| 10:49.44 | MrMister2 | achu: I don't have access to both boxes here but those URL's should be a good starting point. |
| 10:50.10 | achu | k |
| 10:50.36 | MrMister2 | JT: any advice on the echo cancellation with the cheap cards? |
| 10:51.04 | tzafrir_home | MrMister2, software echo cancellation, for starters |
| 10:51.05 | MrMister2 | That might be something to be concerned about :( |
| 10:51.41 | tzafrir_home | BRI (digital telephony in general) does not generate echo. It is only there to try to cancel echo from other sources |
| 10:51.46 | MrMister2 | tzafrir_home: Any advice on which one should work better in this instance? cheap ISDN BRI card, one port only. I don't mind paying 10 or 15 USD for it :) |
| 10:52.16 | JT | MrMister2: it's not cheap is my advice |
| 10:52.36 | achu | MrMister2, can you please explain if you have time, how you do that ? |
| 10:52.38 | tzafrir_home | none of them have hardware echo cancellation. But then again, the performance hit of a software echo canceller on one or two ports , not an issue |
| 10:52.43 | MrMister2 | JT: "it's not cheap"? sorry? |
| 10:54.14 | JT | hardware echo cancellation |
| 10:54.43 | MrMister2 | tzafrir_home: sorry, the last question was asking about the software echo cancellation, not the card itself since it seems that (what I understood from JT's response) the cheap ones mostly use the same chipset and work fine. |
| 10:55.08 | MrMister2 | JT: I've seen hardware echo cancellation and it bloody expensive, yes :( |
| 10:55.20 | tzafrir_home | oslec is nice |
| 10:55.22 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 10:55.29 | tzafrir_home | seems to work fine here |
| 10:55.37 | achu | MrMister2, did you tried to change the values in zapata.conf ? |
| 10:55.46 | achu | for echo cancellation |
| 10:55.50 | MrMister2 | achu: ? |
| 10:56.05 | bintut | Mw3: thanks. i got it working already. :) |
| 10:56.06 | MrMister2 | achu: It's not analog, it's digital. |
| 10:56.14 | achu | oh , sorry |
| 10:56.34 | achu | which card it is ? |
| 10:57.02 | harpal | ai-a, yesterday we have setup calling between two asterisk server. Its working fine. but when user from 1st asterisk server calls to user to 2nd asterisk server than it doesnt shows number of caller. its showing IAX user |
| 10:57.40 | ai-a | harpal: hmmm. |
| 10:57.41 | harpal | So can it shows me who has called from another server. so i can call it back? |
| 10:57.43 | MrMister2 | I'll have a Asterisk box hooked up to a ISDN BRI and a SIP provider for termination to the national PSTN on the trunk side and 7 Siemens C450 IP mobile phones plus one Linksys PAP2T on the extensions side. Not sure if echo will be a big issue but don't think so. |
| 10:58.05 | achu | harpal use switch=> |
| 10:58.18 | MrMister2 | achu: That's what I was asking advice on since I have 0 experience with ISDN and Asterisk. |
| 10:58.36 | *** join/#asterisk myiagy (n=myiagy@201-67-138-60.bnut3703.dsl.brasiltelecom.net.br) |
| 10:59.33 | harpal | how to use switch? |
| 10:59.36 | *** join/#asterisk _ys (i=ys@91.151.196.254) |
| 10:59.45 | anujsingh | can i have some tutorial link of AGI howto ? |
| 11:00.01 | MrMister2 | ~AGI |
| 11:00.04 | jbot | methinks agi is the Asterisk Gateway Interface... similar to CGI for web applications AGI lets you script call control and access databases using your favorite language. AGI wrappers are available for Python (pyst), Perl (astperl?) and other languages |
| 11:00.31 | achu | I am not sure about it, but you can try with switch=> statement |
| 11:00.32 | MrMister2 | anujsingh: http://www.voip-info.org/wiki-Asterisk+AGI |
| 11:00.40 | harpal | ok |
| 11:01.07 | MrMister2 | anujsingh: I've done a full AGI app in PHP and it's really easy to control the call from start to finish. |
| 11:01.58 | achu | harpal, hold on a second |
| 11:02.05 | anujsingh | MrMister2 I am trying to find a way to play recorded files of ../monitor/DONE folder using soft phone. |
| 11:03.58 | anujsingh | the thing so far done is manually , using apache , a web page linked to the target directory, but trying to use softphone only. |
| 11:04.34 | MrMister2 | anujsingh: you mean like the voicemail does? you dial a number, it gives you a menu and plays files on the pbx hd? |
| 11:04.43 | anujsingh | exactly :) |
| 11:04.48 | MrMister2 | My advice is to hack the voicemail code then ;) |
| 11:05.19 | MrMister2 | why re-invent the whell? |
| 11:05.22 | MrMister2 | *wheel |
| 11:05.34 | anujsingh | yes you are right . but i am not good with C. |
| 11:05.59 | anujsingh | so trying to find some middle way. |
| 11:06.13 | MrMister2 | anujsingh: what languages do you know? php? |
| 11:06.22 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:06.42 | anujsingh | shell scripting , but i have few friends who know php . |
| 11:06.43 | ai-a | why not write a script on an ext dialled that places audio files ? |
| 11:08.14 | anujsingh | ai-a then how to show which files are there. ? |
| 11:08.19 | achu | harpal, http://forums.whirlpool.net.au/forum-replies-archive.cfm/549784.html |
| 11:08.51 | ai-a | anujsingh: where is your call design ? |
| 11:09.00 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 11:09.02 | ai-a | if you have 10,000 files,, how do you expect to 'show it' ? |
| 11:09.26 | achu | harpal, see whether its help you |
| 11:09.32 | anujsingh | day by day file number will increase. |
| 11:10.03 | anujsingh | yes ai-a |
| 11:10.09 | harpal | achu, ok I am checking that. but I have one another question |
| 11:10.18 | ai-a | anujsingh: i am not writing it for you. i dont want to know. |
| 11:10.38 | harpal | I have caller from different context than how can they call each other? |
| 11:10.47 | harpal | on a single server |
| 11:11.55 | anujsingh | yes ai-a . |
| 11:12.35 | anujsingh | but the thing you said is correct , what after 6 months of such a setup, there will be thousands of files. |
| 11:13.12 | anujsingh | best way seems a web interface linked to the recorded files. easy to manage . |
| 11:13.19 | achu | harpal, I don't understand |
| 11:13.49 | *** join/#asterisk MRH2 (n=Mr_happy@62.49.242.3) |
| 11:13.50 | ai-a | anujsingh: why not have a meeting (not in here) with your staff and design it. |
| 11:14.31 | harpal | I have one user from context harpal and second user from context default. now here it doesnt make call between them |
| 11:14.56 | anujsingh | yep i raised the same thing , how to handle thousnads of files after few months, |
| 11:15.11 | anujsingh | web interface i have done already , |
| 11:15.29 | MRH2 | hi - anyway to lookup a text list in the dial plan as in: if callerid=listed on the textfile then goto blah |
| 11:15.29 | anujsingh | but the client wants to see such a way, |
| 11:15.52 | anujsingh | to play recorded files via softphone. |
| 11:17.13 | harpal | achu, Can you understand that what I am trying to say? |
| 11:17.31 | *** join/#asterisk stolpskott (n=andrej@191.66.227.87.static.ghkn.siw.siwnet.net) |
| 11:18.47 | achu | harpal, read this , its not the solution but you can get an idea |
| 11:18.48 | achu | http://www.automated.it/asterisk/lah-3-6-05_2.html |
| 11:19.14 | stolpskott | Hi does anyone have a clue about this: call parking works fine internally, extension to extension, byt if a call comes in trough a trunk, then to a queue ant someone picks that up, he cannot park the call (transferes work fine) |
| 11:19.37 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-974792a82fc2f23a) |
| 11:21.44 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 11:22.40 | achu | anybody have idea about name directory sharing between two asterisk servers ? |
| 11:24.28 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 11:29.58 | dandre | Hello, |
| 11:30.02 | stolpskott | so parking a call that comes trouch a queue, anyone have any hints what to look at if it doesn't work? I se app_queue soes not have a "k" option. What to to? |
| 11:30.31 | dandre | what is the value of maxlen in queues.conf that means 'no limit'? |
| 11:39.51 | *** part/#asterisk stolpskott (n=andrej@191.66.227.87.static.ghkn.siw.siwnet.net) |
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| 11:43.37 | jm|laptop | Skype annoys me |
| 11:45.12 | mkl1525 | Does anybody know if persistenmembers/queue works with an expected * restart only or also if * crashes (afaik it saves the values in ast db don't know if this is stored on hd)? |
| 11:48.15 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 11:54.34 | dandre | I have queue whose members are set to Local/extnum@context to have queues in wich members are automatically joined. The problem is that if one extension in the members list, the call is routed to that member voicemail. How show I have the correct behaviour? |
| 11:59.08 | *** join/#asterisk McDouglas (n=mcd@mmcomp.adsl.datanet.hu) |
| 11:59.18 | McDouglas | hi |
| 11:59.43 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 12:00.07 | McDouglas | is it possible to let my users change whether they want voicemail on their extension or not? (basicly allowing them to edit the dialplan, obviously not through an ssh connection) |
| 12:02.19 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
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| 12:35.35 | gazzerh | i everyone. If I add a analog card to my asterisk box will I be able to use one of the lines for faxing over the ISDN30 line I also have a card for? |
| 12:35.47 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 12:39.54 | defswork | I added a rule to stop certain extensions from being able to dial out - I am sure it worked but apparently it doesnt :O |
| 12:40.07 | defswork | it's - exten => _0X./110,n,Hangup |
| 12:40.09 | *** join/#asterisk qdk (n=qdk@85.235.253.139) |
| 12:40.11 | defswork | does that look ok? |
| 12:45.08 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 12:47.30 | *** join/#asterisk ltd (n=z@pat.transact.net.au) |
| 12:47.57 | McDouglas | i know there is a ${BLINDTRANSFER} variable, but is there such a thing for attended transfer? |
| 12:48.13 | McDouglas | (because is think it isnt set if i do an attended transfer) |
| 12:48.37 | *** join/#asterisk anujsing (n=Administ@59.94.133.230) |
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| 13:07.45 | rantsh | Good morning people |
| 13:15.20 | *** join/#asterisk vykarian (n=risadinh@server.pennacchi.com.br) |
| 13:23.10 | vykarian | Hi guys.. I'm having a issue when calling from a SIP account a ZAP channel.. I receive the message Bad Gateway |
| 13:23.21 | vykarian | my confs and verbosity are here: http://pastebin.ca/793629 |
| 13:23.24 | vykarian | any tip? |
| 13:24.21 | vykarian | the zap channels are all available |
| 13:26.54 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
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| 13:33.15 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
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| 13:52.10 | *** join/#asterisk masus (n=burdan@88.248.14.186) |
| 13:52.59 | masus | have anybody experience how to use Application mysql query with "mysql stored procedures"... Thanks |
| 13:53.41 | masus | "exten => s,n,MYSQL(Query resultid ${connid} call sp_ca("${ARG1}",@VARMI);SELECT @VARMI;);" |
| 13:53.47 | masus | is this wrong ? |
| 13:55.24 | dandre | <PROTECTED> |
| 13:58.38 | mkl1525 | HI, trying the safe_asterisk script. it starts without problems. but when i do a kill on the * process it isn't restarted - is this intended behavior? |
| 13:59.27 | [TK]D-Fender | dandre: Stop using extens that call voicemail <----- |
| 14:00.39 | McDouglas | i know there is a ${BLINDTRANSFER} variable, but is there such a thing for attended transfer? |
| 14:01.20 | [TK]D-Fender | McDouglas: No, because the actual transfer is a hand-off transaction and the call you place is seperate up until that point. |
| 14:01.54 | [TK]D-Fender | McDouglas: For instancee on a Polycom SPIP you can start a transfer and at any point hit "Split" and the calls will be maintained completely independant. |
| 14:02.26 | *** part/#asterisk masus (n=burdan@88.248.14.186) |
| 14:04.06 | dandre | [TK]D-Fender: ok so I must use two extens for my users: one regular with voicemail and the other as "agent"? |
| 14:05.18 | *** join/#asterisk pacor (n=paco@86.111.66.1) |
| 14:05.31 | pacor | hi all |
| 14:06.15 | tzafrir | mkl1525, err.... yes |
| 14:06.27 | tzafrir | did I mention I don't really like safe_astrerisk? |
| 14:06.52 | *** join/#asterisk masus (n=burdan@88.248.14.186) |
| 14:07.10 | pacor | somebody knows a voip provider that permit to call to land phones all over the world? |
| 14:07.30 | pacor | i want to for a solidary project |
| 14:07.34 | pacor | thats posible? |
| 14:07.40 | pacor | sorry for my english |
| 14:09.35 | pacor | in fact i prefer for africa or south america |
| 14:09.37 | [TK]D-Fender | dandre: Well that one calls voicemail, and you don't want that. Do the math. |
| 14:10.16 | [TK]D-Fender | pacor: most let you call internationally... |
| 14:11.14 | pacor | [TK]D-Fender total free? |
| 14:11.55 | pacor | i want to build a free locutory with foneras, and asterisk to permit foreings call home in the street |
| 14:12.41 | [TK]D-Fender | pacor: FREE?! lol |
| 14:12.53 | pacor | yeah i want |
| 14:13.06 | [TK]D-Fender | pacor: Of course not. Who give free world-wide calling?! Why do you think we pay LD? |
| 14:13.49 | pacor | write more clearly my english is so bad |
| 14:13.51 | pacor | sorry |
| 14:14.15 | rob0 | What is not clear? You pay for calls. |
| 14:14.31 | pacor | is for a protest versus phone providers |
| 14:14.41 | pacor | LD is not clear |
| 14:14.42 | pacor | xD |
| 14:14.43 | coppice | but why do you pay for calls? its not written in law |
| 14:15.22 | pacor | i look for providers and is free to call but normally in first world |
| 14:15.37 | dandre | [TK]D-Fender: ok when app_queue dials Local/extennum, it runs stdexten macro. Is that right? Is this behaviour hard coded or is it cutomisable? |
| 14:15.39 | [TK]D-Fender | coppice: No, only in the invoice you're going to receive :) |
| 14:16.16 | [TK]D-Fender | dandre: This is YOUR dialplan. YOU sent that call where its going. What is there to "assume"? It runs what you TOLD it to. |
| 14:16.33 | pacor | africa and south america, i see allways have a cost |
| 14:20.28 | dandre | oups, sorry ;-) |
| 14:21.11 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
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| 14:31.25 | McDouglas | [TK]D-Fender: so, you say there is no way i can detect an attended transfer? how can i implement a callback feature then? (i could do it with blind transfer) |
| 14:32.10 | [TK]D-Fender | McDouglas: if you're doing an attended transfer then you never lost the call you are planning on transferring. So what need is there to callback? |
| 14:32.38 | [TK]D-Fender | McDouglas: Thats like asking for car insurace without even having a car :) |
| 14:33.06 | McDouglas | well, you see i have some dense users who wont press the blind transfer key but isntead they press the attended transfer and they hang up imediatelly |
| 14:33.38 | McDouglas | so basicaly its a blind transfer (not technically of course) |
| 14:33.53 | [TK]D-Fender | McDouglas: Sorry, we don't sell "Dense Employee Insurance" here, please try again later! |
| 14:35.00 | McDouglas | [TK]D-Fender: okay, dense might be the wrong word. The old analog phones only had one transfer key and it was the attended transfer. the new SPA phones have a blind transfer but you have to press that "jog" key to movi in the menu and get the bxfer button |
| 14:35.18 | McDouglas | *move |
| 14:35.59 | [TK]D-Fender | McDouglas: Sorry, but the answer isn't changing. They can be as "used to" the old way as they want, but the rules have changed and they will need to adapt. |
| 14:36.15 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 14:36.15 | *** mode/#asterisk [+o russellb] by ChanServ |
| 14:36.27 | McDouglas | so it is not possible to mimic the hw-pbx in regards of handling the callback? |
| 14:37.50 | [TK]D-Fender | McDouglas:I just described how it works. There really isn't anything more to say. |
| 14:37.56 | *** join/#asterisk macros73 (n=cs@c-24-3-246-27.hsd1.pa.comcast.net) |
| 14:39.23 | McDouglas | something else then: is it possible to let my users change whether they want voicemail on their extension or not? (basicly allowing them to edit the dialplan, obviously not through an ssh connection) |
| 14:40.29 | [TK]D-Fender | McDouglas: No, generally they cannot change your dialplan (You don't let them mess with your code, do you?). YOU however can add some extra checks to make a decision based on a choice they made about how they want to process calls to their extens. |
| 14:41.11 | McDouglas | any chance there is some simple web frontend or something to allow them to make a choice? |
| 14:41.14 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:41.14 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 14:41.53 | [TK]D-Fender | McDouglas: Since I can have extens vary their result based on time of day, who's calling, if its raining in Portugal, and if there is a reference to Paris Hilton on the front page of Digg, I'm sure its not big deal for you to put a "does ext X want VM?" |
| 14:42.22 | [TK]D-Fender | McDouglas: This isn't the channel to be asking about GUI's. And there is no one specifically for that. Feel free to write your own. |
| 14:42.47 | McDouglas | well, writing a simple gui is all right |
| 14:42.55 | McDouglas | but how can i integrate it into my dialplan? |
| 14:43.14 | McDouglas | use the management interface to modify the dialplan? |
| 14:43.20 | McDouglas | or there is something simpler? |
| 14:43.43 | [TK]D-Fender | McDouglas: make your dialplan check a value that will detemine what it is to do. |
| 14:44.19 | [TK]D-Fender | McDouglas: "core show application gotoif" <- this is not Raw Car Science. |
| 14:44.24 | [TK]D-Fender | Raw Cat* |
| 14:44.37 | [TK]D-Fender | darn, I hate mangling that punch-line |
| 14:44.53 | McDouglas | i understand what you say and i know how to make a decision |
| 14:45.13 | McDouglas | but how do i interract with the "value" that gotoif uses? |
| 14:45.36 | [TK]D-Fender | McDouglas: How about you actually make an exten so users can set this "value"..... |
| 14:46.42 | McDouglas | i was thinking something "more" visual :P |
| 14:46.55 | *** join/#asterisk harpal (n=Harpal@124.125.255.223) |
| 14:48.05 | [TK]D-Fender | McDouglas: If you think this singular decision is woth the work, then by all means. |
| 14:48.38 | [TK]D-Fender | McDouglas: However for perspective keep in mind that in the span of this conversation I could have coded it all in the dialplan :) |
| 14:49.01 | McDouglas | [TK]D-Fender: obviously you had much more experience with asterisk so, its not a suprise ;) |
| 14:49.29 | [TK]D-Fender | McDouglas: next hint "core show function DB" |
| 14:49.31 | McDouglas | lets say i do it your way, how can i store the value? using astdb ? |
| 14:49.40 | [TK]D-Fender | ^^^^ |
| 14:49.58 | [TK]D-Fender | McDouglas: thats the "freebie" way. Is it worth doing something bigger? |
| 14:50.52 | McDouglas | but you have to provide them with some voice instructions like "if you want VM press 1, else 0", right? |
| 14:51.08 | McDouglas | or mybe i can send an email about it, but they will forget that :P |
| 14:51.26 | McDouglas | so i was thinking making it graphical they wont forget, and i dont have to record a message either |
| 14:52.06 | [TK]D-Fender | McDouglas: HP LaserJets make great graphics, and 3M a way to make the idea "stick" ;) |
| 14:52.37 | [TK]D-Fender | McDouglas: and the recordings could have been done int he time since I first told you the code was completed till now ;) |
| 14:53.25 | McDouglas | okay, then lets just talk about this hypothetically |
| 14:53.35 | McDouglas | if i wanted to make, lets say a php frontend |
| 14:53.41 | McDouglas | what is the best way to set the db values |
| 14:54.48 | [TK]D-Fender | McDouglas: get a BDB PHP module, or use AMI, or direct call to "asterisk -rx ....", or whatever. |
| 14:55.31 | blitzrage | happy thanksgiving everyone! |
| 14:56.17 | De_Mon | [TK]D-Fender Parking/UnParking has some problems -- like either the caller or called party hanging up while being parked (as I can't get a hangup extension in parkedcalls) and such. |
| 14:57.12 | [TK]D-Fender | De_Mon: guess you'll have to come up with something a bit more "creative" |
| 14:58.26 | *** join/#asterisk harpal (n=Harpal@124.125.255.223) |
| 14:58.31 | De_Mon | I discovered an app in bristuff called Autoanswer[Login] that might work well |
| 14:59.36 | *** join/#asterisk bmg505 (n=leon@196.209.179.149) |
| 15:01.41 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
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| 15:02.39 | *** join/#asterisk MRH2 (n=Mr_happy@62.49.242.3) |
| 15:04.25 | MRH2 | hi i seem to recall that you used to not be able to change the cdr channel in the dialplan - is this still the case? |
| 15:05.18 | [TK]D-Fender | MRH2: "show function CDR" <- go try |
| 15:06.02 | De_Mon | what is a cdr 'channel' |
| 15:06.06 | De_Mon | oooh nevermind I know |
| 15:06.27 | slavon_net | why i listen distortion if channel is ULAW and file is GSM? |
| 15:06.46 | De_Mon | slavon_net voip? |
| 15:06.50 | slavon_net | yep |
| 15:07.04 | De_Mon | I blame that |
| 15:07.56 | MRH2 | i'm gonna guess channel is still read only |
| 15:07.57 | [TK]D-Fender | slavon_net: Certain releases of trunk have caused that, as have certain GCC compiler versions. |
| 15:08.12 | slavon_net | have last brunch1.4 |
| 15:10.30 | MRH2 | anyway easy way to make cdr channel writable? |
| 15:11.08 | [TK]D-Fender | MRH2: You've got the source. |
| 15:12.15 | MRH2 | i so h8 that answer on irc - no offence |
| 15:12.17 | MRH2 | lol |
| 15:12.21 | De_Mon | MRH2 why? |
| 15:12.28 | De_Mon | why do you want that writable? |
| 15:12.32 | MRH2 | cause it answers every single question |
| 15:12.36 | blitzrage | slavon_net: fyi -- branch, not brunch (although brunch is a delicious idea) |
| 15:12.58 | MRH2 | you might as well have an autoresponder |
| 15:13.07 | slavon_net | blitzrage =) |
| 15:13.16 | De_Mon | MRH2 changing the 'writable' flag on a field isn't very hard. but that doesn't tell me why you want to do it. |
| 15:13.51 | [TK]D-Fender | MRH2: Sorry, what you want requires a source mod. This is not a cop-out, this is a REALITY. |
| 15:14.10 | [TK]D-Fender | blitzrage: I find branch rather hard to swallow ;) |
| 15:14.43 | De_Mon | MRH2 you could always create a new cdr field ChannelCustom() and set it to whatever you want, and in your cdr queries choose that field over the real one if its not null |
| 15:14.58 | De_Mon | CDR(channelCustom) rather |
| 15:15.08 | [TK]D-Fender | De_Mon: Viable alternative... |
| 15:15.15 | MRH2 | using in an agent channel changes it, need to keep other calls consistent |
| 15:15.47 | blitzrage | MRH2: in trunk there is cdr_adaptive_odbc which lets you setup fields and change them |
| 15:15.52 | De_Mon | so you want to know the channel that answered for the agent? |
| 15:15.56 | blitzrage | there is a 1.4 backport, but I can't find it |
| 15:16.02 | jamesrdorn | ugh, I have asterisk installed on a CF card, I tried to have the log directory setup for /dev/null, but asterisk will fail to start. I dont really need logs at all. I am hoping I dont have to create a ramdisk to solve this problem |
| 15:16.38 | tzafrir | jamesrdorn, the log directory is a directory, not a file |
| 15:16.54 | tzafrir | you should edit logger.conf more carefully |
| 15:17.22 | tzafrir | make sure nothing is sent to a file. Only to the console (and maybe to syslog?) |
| 15:17.33 | De_Mon | can you ln -s /dev/null to asterisk/logfile ^_^ |
| 15:17.36 | jamesrdorn | tzafrir, can I just turn logging off? |
| 15:17.47 | De_Mon | ^^^ thats what he said |
| 15:18.56 | MRH2 | i reckon i'll have to mess about with making cdr channel writable and set cdr - was hoping if there was maybe a config option somewhere though. |
| 15:22.26 | jamesrdorn | there we go... loooks like there was already a ramdisk mounted for tmpfs |
| 15:25.07 | jamesrdorn | just did a symlink from /var/logs to the tmpfs also |
| 15:25.09 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 15:25.25 | jamesrdorn | that should cause almost no writing to the CF card for logging |
| 15:26.53 | tzafrir | MRH2, you can send CDR to a remote storage if it helps |
| 15:27.07 | MRH2 | not me ;) |
| 15:27.42 | mort_gib | Hi |
| 15:28.31 | tzafrir | MRH2, what's the point in the CDR if you don't have where to store it? |
| 15:29.09 | mort_gib | questions :-) |
| 15:29.37 | MRH2 | sorry misread cdr as that logger topic (although just log to console is the way to go) |
| 15:30.54 | mort_gib | Anyone have a minute?? |
| 15:31.52 | tzafrir | jbot, tell mort_gib about ask |
| 15:32.22 | mort_gib | :-) New to Asterisk, but a few questions comes up! |
| 15:32.30 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 15:32.46 | mort_gib | Can you do a transfer to an external line from a phone, IP or analog handset?? |
| 15:33.27 | [TK]D-Fender | mort_gib: If you have a line free to call out on, sure |
| 15:33.38 | mort_gib | Any hints?? |
| 15:34.08 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.134.26) |
| 15:34.10 | tzafrir | mort_gib, in short: yes |
| 15:34.24 | tzafrir | read a bit about transfer features |
| 15:34.26 | [TK]D-Fender | mort_gib: Just transfer the call to an exten that dials out a line. End of story. |
| 15:34.29 | tzafrir | ~docs |
| 15:34.30 | jbot | well, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
| 15:34.54 | tzafrir | This one needs updating, right? |
| 15:35.12 | mort_gib | Okay, sorry for bothering you guys! |
| 15:35.21 | *** join/#asterisk ghento (n=ghento@d216-232-170-111.bchsia.telus.net) |
| 15:35.21 | [TK]D-Fender | tzafrir : just a little ;) |
| 15:37.29 | *** join/#asterisk michael-i (n=michael-@141.41.40.55) |
| 15:38.04 | *** join/#asterisk exvito (n=exvito@195.245.132.93) |
| 15:38.09 | mort_gib | The question that I haven't found an answer to is the following, when a user leaves the office, can they direct their calls to say a mobile of choice directly from the IP handset? |
| 15:38.43 | michael-i | Hey everybody |
| 15:38.52 | mort_gib | That would require changes to the extensions.conf |
| 15:39.29 | defswork | mort_gib: I just divert on no answer |
| 15:39.43 | defswork | mort_gib: and they can set themselves on dnd |
| 15:39.44 | mort_gib | Yes, but the point is "to any mobile" |
| 15:40.12 | defswork | mort_gib: I have feature code that prompts for the number |
| 15:40.31 | mort_gib | -Yeah?? :-) |
| 15:40.36 | *** join/#asterisk bakermd (i=bakermd@dhcp12-11-95-189.lebp.atl.wayport.net) |
| 15:40.50 | bakermd | I am trying to use realtime for voicemail config, and it is seeing the users I have in the DB, however when it goes to record a message it hangs up with the error app_voicemail.c:3145 leave_voicemail: No format for saving voicemail? |
| 15:40.53 | bakermd | And I put a column in the table for "format" which is set to "wav|wav49" |
| 15:40.54 | bakermd | Any ideas? |
| 15:41.51 | defswork | mort_gib: *72 on my boxes |
| 15:41.51 | *** join/#asterisk masus (n=burdan@88.248.73.2) |
| 15:41.57 | michael-i | I had a problem earlier where, upon answering an analog phone connected to my Asterisk install, several DTMF tones were played before the call was connected. I've finally have some logs here (http://pastebin.ca/793734). Anyone heard of this happening before? |
| 15:42.33 | defswork | mort_gib: steal the dial plans for it from freepbx |
| 15:42.48 | masus | i cant use mysql stored procedure with application MYSQL within asterisk, have anybody any usage examples |
| 15:43.37 | [TK]D-Fender | mort_gib: When you forward on an phone, you cansend the call to any exten THEY can dial. |
| 15:44.09 | *** join/#asterisk stimpie (n=stimpie@84-104-5-115.cable.quicknet.nl) |
| 15:44.13 | mort_gib | I haven't had a look at freepbx yet, but I suppose I should! |
| 15:44.20 | [TK]D-Fender | defswork: thats if you want * doing the thinking for you. Or you can leave it up to the phone. |
| 15:44.36 | defswork | [TK]D-Fender: sure |
| 15:45.05 | defswork | and I got a glowing commendation from a user today |
| 15:45.33 | mort_gib | I have two this month, a 12 user and a 22 user, this feature is one of the top 5 features |
| 15:45.47 | defswork | mort_gib: I use trixbox |
| 15:45.59 | mort_gib | -Why Tribox?? |
| 15:46.07 | defswork | I tried a couple - it worked |
| 15:46.40 | defswork | my first install was totally blind so I needed a big step up to start with |
| 15:46.58 | florz | Any ideas as to how to have a program make a call and then receive all the incoming audio, with the possibility to send some audio, at least via file playback? |
| 15:47.45 | [TK]D-Fender | florz: elaborate on the "receive all the incoming audio, with the possibility to send some audio" part. |
| 15:47.59 | [TK]D-Fender | florz: What do you plan on doing with received audio? |
| 15:48.01 | *** join/#asterisk orn (n=orn@85.197.193.24) |
| 15:48.01 | florz | [TK]D-Fender: erm, what's missing, you think? |
| 15:48.02 | mort_gib | Well, it took me two days (still doing my job) to get a systems to do lots of stuff, including using VOIPSTUNT/POTS conditional |
| 15:48.19 | florz | [TK]D-Fender: process it? =:-) - well, I'd prefer to get SLIN |
| 15:48.36 | orn | I can't unload res_odbc for some reason: |
| 15:48.37 | orn | [Nov 22 15:46:36] WARNING[25417]: loader.c:492 ast_unload_resource: Firm unload failed for res_odbc.so |
| 15:48.40 | [TK]D-Fender | florz: that isn't an action. Try again. |
| 15:49.10 | orn | Any ides? |
| 15:49.24 | florz | [TK]D-Fender: Well, I want to extract data that was encoded into it by the remote side, but I don't quite see how the details would affect this part of the problem?! |
| 15:49.26 | orn | module show like odbc: |
| 15:49.28 | orn | res_odbc.so ODBC Resource 0 |
| 15:49.30 | [TK]D-Fender | florz: Your description doesn't tell us enough to suggest anything yet. You're going to need to give real details... |
| 15:50.07 | [TK]D-Fender | florz: "extract data" and "encoded", are you trying to encrypt your request for help too? |
| 15:50.22 | orn | furthermore, when I load the voicemail app it complains about not being able to connect to the odbc database, but odbc isn't configured as enabled in voicemail.conf |
| 15:50.39 | [TK]D-Fender | florz: You are being painfully evasive and we will not be able to assist you if you don't get down to the specifics |
| 15:50.49 | florz | [TK]D-Fender: Gee, no, do you really wanna know all the details of how I am encoding and decoding? |
| 15:51.14 | florz | [TK]D-Fender: I'm full willing to explain it all if you really think that will help you in any way - I just doubt it will ... |
| 15:51.18 | blitzrage | orn: you don't have 'odbcstorage' uncommented anywhere in your voicemail.conf file? |
| 15:51.35 | [TK]D-Fender | florz: You say "listen to audio" and "send audio". What kind of answer do you think we can give for something so abstract? |
| 15:52.00 | orn | only appears in two lines, as odbcstorage=asterisk and odbctable=voicemessages, both of which are commented |
| 15:52.46 | orn | but the weirdest thing is that i can't unload the odbc module |
| 15:52.56 | florz | [TK]D-Fender: Well, if I knew the answer, I wouldn't be asking, right? But I simply don't see how the further processing of the data affects the interface via which I get and send SLIN audio?! |
| 15:53.37 | florz | [TK]D-Fender: Maybe I should make clearer that both operations should be working in parallel. |
| 15:53.48 | [TK]D-Fender | florz: If you want absolute raw audio stream from a channel you'll have to write an app for it. |
| 15:54.24 | [TK]D-Fender | florz: As in a dialplan application. Go look at app_echo for some inspiration. |
| 15:55.51 | *** join/#asterisk ghento (n=ghento@d216-232-170-111.bchsia.telus.net) |
| 15:55.58 | florz | [TK]D-Fender: In particular, there is no way to make bidirectional transmission work with EAGI? Like, playback without stopping the incoming stream? |
| 15:56.51 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 15:56.57 | [TK]D-Fender | florz: I seriously doubt it. Dialplan and AGI are not the places for the kind of dangerously vague global audio control you're asking for. |
| 15:58.07 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 15:58.58 | florz | [TK]D-Fender: Well, hackish ideas are welcome, too, if I can avoid touching the asterisk source. Like, receiving via Monitor() writing to a pipe (which doesn't work because Monitor does an unlink() first ...). |
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| 15:59.48 | [TK]D-Fender | florz: Sorry, you asked the kind of open-ended question that only be summed up with "yes you're writing a whole dialplan app" |
| 16:00.42 | florz | [TK]D-Fender: Well, where do you think could I be more specific? |
| 16:01.40 | [TK]D-Fender | florz: there is no sense of sequence in the order of events (which you never gave". Everything in AGI / dialplan is purely linear. one complete step at a time. NOTHING in parallel. So what you wan't can't be done there. |
| 16:03.30 | florz | [TK]D-Fender: Well, but Monitor()ing does work in parallel with, say, AGI execution, too!? And no, there is no order to sending and receiving, I want to do both at the same time, just that sending via file playback would be good enough, I don't need continuous streamed transmission. |
| 16:04.23 | [TK]D-Fender | florz: Doesn't have to be continuous to be a problem, only simultaneous. Then there's the thought of trying to hack into the Monitor and synchronize with some other "actions". |
| 16:04.51 | [TK]D-Fender | florz: Perhaps app_queue is a better sample. |
| 16:05.32 | florz | But, I mean, asterisk generally does have full duplex voice, doesn't it? =:-) |
| 16:05.44 | florz | well |
| 16:06.50 | orn | ok, so when i uncomment odbcstorage in voicemail.conf and set it to something other than asterisk the error message changes accordingly, but when I comment out that line it complains about not being able to connect to database object asterisk... |
| 16:07.04 | orn | why the hell would the voicemail app be trying to use odbc when the line is commented out? |
| 16:07.19 | orn | does voicemail require odbc? |
| 16:10.47 | [TK]D-Fender | florz: I'm not even going to answer that one... |
| 16:11.05 | jamesrdorn | orn, no it does not require odbc |
| 16:11.12 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:11.16 | florz | [TK]D-Fender: That one wasn't even a question ... |
| 16:11.48 | jamesrdorn | ord, make sure all odbc entries are set for "noload" in the modules.conf |
| 16:12.01 | orn | they are now, but it was res_odbc was loaded |
| 16:12.04 | orn | and i can't unload it |
| 16:12.12 | orn | is that normal perhaps? isn't it possible to unload res_odbc.so ? |
| 16:12.28 | [TK]D-Fender | florz: You seem to be desperate for an easy answer that does not exist. Unless you're able to rework what you want to do, I have already told you the only real way it can be done. |
| 16:12.40 | orn | but even if it were loaded, shouldn't app_voicemail only use what is specified in voicemail.conf? |
| 16:12.41 | jamesrdorn | yes, set it for noload, then reboot your asterisk with "restart gracefully" |
| 16:13.05 | jamesrdorn | orn, unfortunatly, I dont have that answer |
| 16:13.06 | [TK]D-Fender | orn, not, thats what exconfig is for. |
| 16:13.47 | orn | from extconfig: ;voicemail => odbc,asterisk |
| 16:14.18 | *** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 16:15.15 | [TK]D-Fender | orn: Do you have a reason to be using odbc? |
| 16:15.26 | mort_gib | Orn, I have errors on that one too. Dosn't affect my setup though (as far as can tell) |
| 16:15.26 | orn | no, and i'm trying not to |
| 16:15.45 | orn | [TK]D-Fender: since it's commented out in extconf, voicemail shouldn't try to use odbc, right? |
| 16:15.50 | [TK]D-Fender | orn: If you don't need odbc, then just noload it. |
| 16:16.03 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 16:16.12 | [TK]D-Fender | orn: I missed the part where you showed that voicemail was indeed trying to use it at all. can you pastebin that? |
| 16:16.18 | orn | sure |
| 16:16.34 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 16:17.01 | jamesrdorn | ugh, about to leave for work |
| 16:17.49 | orn | http://pastebin.com/d19590f81 |
| 16:18.45 | [TK]D-Fender | orn: Ok, pastebin up all related configs. |
| 16:19.15 | florz | [TK]D-Fender: Well, I suspected that. Even though I guess dropping the unlink()s from Monitor would be easier, but we'll see ... |
| 16:19.54 | orn | doing so now, sorry i didn't at first.. realized right after |
| 16:20.10 | [TK]D-Fender | orn: voicemail.conf , asterisk.conf , res_odbc.conf , func_odbc.conf |
| 16:20.10 | orn | you want the whole configs or excerpts? |
| 16:20.14 | *** join/#asterisk saint_ (n=saint@c-69-242-118-124.hsd1.nj.comcast.net) |
| 16:20.20 | [TK]D-Fender | orn: EVERYTHING. |
| 16:20.35 | [TK]D-Fender | orn: When you've got a problem, don't start trying to debug showing little bits a t a time. |
| 16:22.00 | anonymouz666 | Great deadlock with chan_local using Asterisk 1.2. |
| 16:22.13 | anonymouz666 | ahhh fuck I guess I will need to update to 1.4 |
| 16:26.08 | orn | [TK]D-Fender: http://pastebin.com/d19cd216c |
| 16:27.32 | nestAr | heh |
| 16:27.48 | [TK]D-Fender | orn: enabled => yes <- From res_odbc.conf: |
| 16:28.36 | [TK]D-Fender | orn: try "no", and restart |
| 16:28.46 | orn | restart * or reload module? |
| 16:28.57 | [TK]D-Fender | orn: each if needed |
| 16:29.11 | [TK]D-Fender | orn: and do you have a noload re_odbc.so for modules.conf? |
| 16:30.38 | orn | I do now, but I removed it a while back because * wouln't start when I had it as noload... Will try to restart when I can. module reload doesn't work |
| 16:31.03 | orn | nor does module unload, but that had already been established |
| 16:31.39 | orn | But even so, I still don't understand why voicemail is trying to use odbc, given that it's commented out in voicemail.conf |
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| 16:36.04 | [TK]D-Fender | orn: I'm not sure, try cseeting the pre-load to "no" in there as well though... |
| 16:36.10 | [TK]D-Fender | setting* |
| 16:37.08 | orn | ok, thanks |
| 16:37.56 | orn | hmm, do you mean pre-connect in res_odbc or pre-load in modules.conf? |
| 16:39.50 | orn | hmmm wow |
| 16:39.55 | orn | when i load voicemail app now the server crashes |
| 16:40.04 | [TK]D-Fender | orn: Progress! |
| 16:40.11 | orn | :) |
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| 18:02.13 | bakermd | In realtime voicemail, how do you configure the formats for voicemail to be saved in? |
| 18:03.28 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
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| 18:09.04 | moemoe | could it be that medianix sucks? i boot it from cd and asterisk only dumps core and quit |
| 18:10.01 | mvanbaak | hhmm |
| 18:10.14 | mvanbaak | I'm rewriting my 1.0.9 extensions.conf to 1.4-svn |
| 18:10.41 | mvanbaak | DBGet and DBPut |
| 18:10.45 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 18:10.47 | mvanbaak | I bet those are no longer allowed |
| 18:13.19 | [TK]D-Fender | moemoe: Entirely possible. Of course it is not supported here... |
| 18:13.32 | [TK]D-Fender | mvanbaak: Correct |
| 18:15.18 | moemoe | [TK]D-Fender: yes, i just wanted to have a debian-based "base" with zaptel already working, and hoped i would get it with this cd |
| 18:15.55 | mvanbaak | DBGet(mvb2vm=mvb/2vm) |
| 18:16.19 | mvanbaak | will that be: Set(vmb2vm=${DB(mvb/2vm)}) |
| 18:16.31 | mkl1525 | trying to configure persistent agents, enabled it in agents.conf and persistentmembers in queue.conf but when using CallBackLogin nothing is written to astdb at least "database show" doesnt show an agent entry - any hints? |
| 18:16.48 | moemoe | or is it more recommendet to use visdn/misdn? |
| 18:16.59 | moemoe | at the moment i'm completely free in my choice ;) |
| 18:16.59 | mvanbaak | CallBackLogin is deprecated :) |
| 18:18.11 | mkl1525 | mvanbaak I know but looking at the source persistance should by working with it too |
| 18:18.31 | mvanbaak | and DBPut(mvb/2vm=1) will that be Set(DB(mvb/2vm)=1) ? |
| 18:19.36 | [TK]D-Fender | mvanbaak: Yes |
| 18:21.02 | mvanbaak | [TK]D-Fender: thanks :) |
| 18:21.11 | mvanbaak | I finally decided to ditch asterisk 1.0.9 |
| 18:25.19 | Putzz | wow |
| 18:25.37 | mvanbaak | gheh |
| 18:25.47 | mvanbaak | SetMusicOnHold(mvb) |
| 18:25.49 | mvanbaak | hahahahaha |
| 18:25.58 | mvanbaak | that's not going to work in 1.4 |
| 18:26.15 | nestAr | yeah, they kinda changed a lot of stuff... |
| 18:26.19 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 18:27.04 | nestAr | i still have one machine running a 1.0 release |
| 18:27.07 | nestAr | Asterisk CVS-v1-0-11/19/04 |
| 18:27.08 | nestAr | nice. |
| 18:27.20 | nestAr | i don't know why it's running though, no calls going through it. |
| 18:28.08 | mvanbaak | gheh |
| 18:28.17 | mvanbaak | I have 2 1.0.9 boxen in production |
| 18:28.27 | mvanbaak | doing roughly 1500 calls a day |
| 18:29.18 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 18:30.44 | *** join/#asterisk mathesis (n=kukako@unaffiliated/mathesis) |
| 18:32.51 | *** join/#asterisk macros73 (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
| 18:34.46 | mvanbaak | there's no: 'dialplan syntaxcheck' right ? |
| 18:35.05 | [TK]D-Fender | mvanbaak: Correct |
| 18:35.17 | mvanbaak | hhmm |
| 18:35.19 | mvanbaak | how to test ? |
| 18:35.36 | [TK]D-Fender | mvanbaak: Place calls. |
| 18:36.19 | mvanbaak | this is going to be a loooooooooooooooooooooooooooooooong night |
| 18:39.52 | *** join/#asterisk angom (n=angom@200.38.31.239.dsl.dyn.telnor.net) |
| 18:49.46 | [hC] | fender, have you played much with iperf? |
| 18:52.40 | *** join/#asterisk xxoxx (n=xxoxx@tor/regular/xxoxx) |
| 18:54.30 | [TK]D-Fender | [hC]: Never heard of |
| 18:54.44 | [TK]D-Fender | [hC]: But have just Googled |
| 18:55.53 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 18:56.01 | TJNII | Has anyone installed flite on asterisk on debian? |
| 18:56.01 | [hC] | ive been looking into it for sending udp traffic between two hosts at a particular packet size, to see what sort of kbit/s lines can handle before they tank |
| 18:56.08 | hi365 | can asterisk stream moh without a theird part program? |
| 18:56.21 | [hC] | since im using g729 on ADSL circuits, its very low pps, which can screw up some adsl modems. |
| 18:56.30 | [hC] | er, very high pps, very low packet size. |
| 18:58.34 | TJNII | hi365: stream it to where |
| 18:58.45 | hi365 | TJNII: moh |
| 18:59.02 | TJNII | Okay, then from where |
| 18:59.12 | hi365 | mpg123 has been giving me problems, im wondering if i even need it |
| 18:59.29 | TJNII | Oh. You're trying to use mp3s? |
| 18:59.33 | hi365 | TJNII: shoutcast mainly (http://scfire-chi-aa04.stream.aol.com:80/stream/1074) |
| 18:59.42 | hi365 | TJNII: an mp3 stream |
| 19:00.11 | TJNII | I believe you will need an external program to handle mp3 streams. |
| 19:00.56 | TJNII | Asterisk will handle wav, gsm, and some other formats I don't remember off the top of my head. I know 1.2 won't natively handle mp3s, not sure about 1.4 |
| 19:01.47 | hi365 | TJNII: interesting, cause 1.2 can do native mp3 -> moh (localy) |
| 19:01.57 | TJNII | With mpg123 |
| 19:02.34 | TJNII | By natively I mean all in *, no externam programs |
| 19:02.48 | hi365 | TJNII: problem is with mpg123 it sometimes stops after 2-4 seconds |
| 19:03.03 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 19:03.14 | hi365 | TJNII: (yes, asterisk can NATIVELY play mp3's in 1.2) |
| 19:03.24 | hi365 | and it doesnt rebuffer |
| 19:05.51 | hi365 | also - mpg123 doesnt stop streaming after a call is disconnected |
| 19:06.24 | TJNII | I couldn have sworn it didn't. Maybe it just didn't work for me |
| 19:06.44 | TJNII | It's been a long time since I played with moh |
| 19:07.56 | [TK]D-Fender | hi365: Be very careful of the word STREAM you were using. MPG123 & Native can both play FIXED files. This has nothing to do with STREAMING. |
| 19:08.28 | hi365 | [TK]D-Fender: true, but can asterisk do streaming as well? |
| 19:08.34 | [TK]D-Fender | hi365: MPG123 may also be capable of accessing an actual stream itself, but there is no * native tool to do so. |
| 19:08.38 | [TK]D-Fender | ^^^ |
| 19:08.51 | hi365 | ok. have you used mpg123 befor? |
| 19:09.02 | TJNII | [TK]D-Fender: He was asking about shoutcast streams earlier. I think I steered him OT. |
| 19:09.22 | [TK]D-Fender | hi365: Yes, but never for streaming. |
| 19:09.33 | hi365 | oh well |
| 19:10.21 | [TK]D-Fender | hi365: And I don't believe that it WOULD stop streaming jsut because its not being used. This might cause networking / startup headaches, so in all likelyhood ones the stream is started with * it stays on the whole time. |
| 19:10.43 | [TK]D-Fender | s/ones/once/ |
| 19:11.13 | hi365 | seems so. and it totaly sucks! why in the world would i want the stream all the time??!! |
| 19:11.43 | tzafrir | hi365, IIRC it does stop streaming when nobody listens |
| 19:11.50 | tzafrir | But that's easy to test |
| 19:11.53 | hi365 | tzafrir: not my mpg123 |
| 19:12.38 | [TK]D-Fender | hi365: Since only 1 specific version is supported by * your point seems moot at best |
| 19:12.41 | hi365 | sterisk 1721 0.0 0.1 4372 1140 ? S 21:02 0:00 /usr/bin/mpg123 -q -r 8000 -f 8192 -s --mono http:/.... <------- i used it last like ten mimutes ago |
| 19:12.55 | hi365 | which version is that? |
| 19:12.56 | tzafrir | You does really have to use mp3. You just need a program that shows signs of life (e.g: CPU usage, log messages) when running, and use it as a custom moh program |
| 19:13.23 | tzafrir | You can use sox, if you actually want an output that makes sense |
| 19:13.46 | *** join/#asterisk funxion (n=x@adsl-065-013-053-031.sip.mia.bellsouth.net) |
| 19:13.53 | tzafrir | But I figure that basically 'cat </dev/zero' may be good enough. Not sure |
| 19:14.15 | [TK]D-Fender | hi365: This is very well documented. 0.59r |
| 19:14.21 | hi365 | tzafrir: can sox stream? |
| 19:14.51 | tzafrir | [TK]D-Fender, 0.59r is because for very long time mpg123 was not developed. |
| 19:14.54 | funxion | if I have a call coming in to asterisk on an e1 pri with the destination field blank how can I get the call to being without using immediate=yes |
| 19:15.06 | tzafrir | In fact, it has quite a few known security holes |
| 19:15.25 | tzafrir | And thus not recommended for streaming from a remote server |
| 19:15.45 | [TK]D-Fender | funxion: huh? |
| 19:15.57 | hi365 | do you guys have a better sugestion for shoutcats streaming? |
| 19:16.01 | funxion | exactly |
| 19:16.12 | funxion | inbound call on e1 pri |
| 19:16.22 | funxion | no destination number |
| 19:16.39 | funxion | as if its connected to a channel bank with a handset going off hook |
| 19:17.01 | funxion | I get extension '' in context 'blah' not found |
| 19:17.16 | funxion | I have an s extension in that context |
| 19:17.43 | funxion | got me? |
| 19:17.53 | [TK]D-Fender | funxion: "s" will not catch [blank] |
| 19:17.57 | funxion | I know |
| 19:18.13 | tzafrir | hi365, sox can't stream. I suggested to use it to test moh's behaviour |
| 19:18.22 | funxion | is there anyway to populate the destination data at the zapata level? |
| 19:18.23 | tzafrir | why not mpg123? |
| 19:18.27 | [TK]D-Fender | funxion: so make a pattern that does and whlie you do that ask your telco why they aren't sending a DID on your PRI |
| 19:18.46 | funxion | how would I make a pattern taht matches blank? |
| 19:19.03 | hi365 | tzafrir: I will try .59r (no resone - just looking for something stable/reliable) |
| 19:19.22 | tzafrir | hi365, get mpg123 from your distro, I guess |
| 19:19.34 | tzafrir | (at least if $DISTRO=Debian) |
| 19:19.34 | hi365 | k |
| 19:19.38 | [TK]D-Fender | funxion: Give a good read to your Asterisk Dialplan Patterns list. It should become apparent. |
| 19:19.45 | funxion | ok |
| 19:20.17 | tzafrir | funxion, or try to use _X. as a pattern, and see what extension you actually get... |
| 19:21.00 | [TK]D-Fender | tzafrir : won't work. |
| 19:21.07 | [TK]D-Fender | funxion: Keep reading for a little bit :) |
| 19:21.11 | funxion | I am |
| 19:21.13 | tzafrir | [TK]D-Fender, why? |
| 19:21.29 | [TK]D-Fender | tzafrir : [14:17]<funxion>I get extension '' in context 'blah' not found |
| 19:21.39 | [TK]D-Fender | tzafrir : [14:17]<[TK]D-Fender>funxion: "s" will not catch [blank] |
| 19:21.43 | funxion | _ maybe? |
| 19:21.56 | funxion | or _. |
| 19:22.00 | [TK]D-Fender | funxion: "_" implies its a pattern that follows, but yuo need more... |
| 19:22.07 | [TK]D-Fender | funxion: getting warmer... |
| 19:22.55 | funxion | i |
| 19:23.04 | tzafrir | If the number is empty, then s is needed, indeed |
| 19:23.23 | funxion | but s wont respond to blank |
| 19:23.31 | *** join/#asterisk Voicemeup (n=VoiceMeU@modemcable132.108-83-70.mc.videotron.ca) |
| 19:23.34 | Voicemeup | ERROR: Unable to determine hostid, You must have at least one NIC! |
| 19:23.35 | funxion | why not i for invalid |
| 19:23.39 | [TK]D-Fender | tzanger: "s" doesn't work for PRI, the extension is known, its just that its BLANK. |
| 19:23.42 | Voicemeup | anyway to to get digium g729 crap to work ? |
| 19:23.47 | Voicemeup | the register thing |
| 19:23.52 | tzafrir | [TK]D-Fender, s sure does. |
| 19:23.56 | [TK]D-Fender | funxion: "i" only works for ivr's |
| 19:24.09 | [TK]D-Fender | tzafrir : He's just confirmed negative, and I've seen this before. |
| 19:24.14 | tzafrir | If you send an empty number, you get to s |
| 19:24.21 | mvanbaak | nope |
| 19:24.27 | funxion | tzafrir its doesnt work |
| 19:24.41 | mvanbaak | use _. |
| 19:24.49 | [TK]D-Fender | mvanbaak: Close but no cigar :) |
| 19:25.04 | Voicemeup | in the registration utility readme file (for Linux) is said that |
| 19:25.04 | Voicemeup | >> first nic always must be eth0 for registration to work |
| 19:25.08 | Voicemeup | omfg you serious ? |
| 19:25.09 | Voicemeup | POS |
| 19:25.10 | *** part/#asterisk Voicemeup (n=VoiceMeU@modemcable132.108-83-70.mc.videotron.ca) |
| 19:26.00 | mvanbaak | I have to admit I didn't ;) |
| 19:26.11 | funxion | [TK]D-Fender []? |
| 19:26.24 | tzafrir | [TK]D-Fender, no! |
| 19:26.40 | [TK]D-Fender | tzafrir : No what? :) |
| 19:26.55 | funxion | lol |
| 19:26.55 | mvanbaak | exten => [\d|\w]{0,},1,.... |
| 19:26.56 | funxion | ! |
| 19:26.57 | mvanbaak | ;) |
| 19:27.31 | [TK]D-Fender | *sigh* |
| 19:27.34 | mvanbaak | lol |
| 19:28.01 | funxion | is ! right? |
| 19:28.59 | funxion | [TK]D-Fender? |
| 19:29.41 | tzanger | [TK]D-Fender: I had a patch for that |
| 19:29.53 | tzanger | [TK]D-Fender: it made 'i' work for PRI just like it did for every other channel type out there |
| 19:29.57 | tzanger | but it was rejected |
| 19:30.12 | [TK]D-Fender | funxion: Yes. |
| 19:30.17 | funxion | thnx |
| 19:30.33 | mvanbaak | -su: less: command not found |
| 19:30.36 | mvanbaak | hahahahaha |
| 19:30.38 | [TK]D-Fender | tzanger: Doesn't work for SIP. invalid exten 404's |
| 19:31.04 | tzanger | [TK]D-Fender: yeah, it fixed that too... 404 if 'i' wasn't defined, i if defined |
| 19:31.39 | *** join/#asterisk bkw_ (n=brian@adsl-64-149-41-151.dsl.tul2ok.sbcglobal.net) |
| 19:32.13 | [TK]D-Fender | tzanger: and... REJECTED... lol. |
| 19:32.36 | tzanger | heh |
| 19:32.39 | [TK]D-Fender | tzanger: like... WTF. |
| 19:35.37 | hi365 | tzafrir: im trying to compile 0.59 of mpg123 and im getting an errors (undefined reference) any ideas? |
| 19:37.31 | *** join/#asterisk CCFL_Man2 (i=CCFL_Man@argon.pureshells.com) |
| 19:38.20 | [TK]D-Fender | hi365: Pastebin is an idea..... |
| 19:38.28 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 19:38.48 | tzafrir | hi365, looking at mpg123.org I see 2007-11-04 Thomas: Bugfix release 0.68 |
| 19:39.26 | hi365 | tzafrir: [TK]D-Fender mentioned that .59r is the best for * |
| 19:39.36 | hi365 | http://pastebin.ca/793949 |
| 19:39.51 | tzafrir | If that old code fails to build with some new compilers and stuff, feel free to solve that |
| 19:40.08 | hi365 | crap! |
| 19:41.14 | *** part/#asterisk bkw_ (n=brian@adsl-64-149-41-151.dsl.tul2ok.sbcglobal.net) |
| 19:42.00 | steve | is it possible to set a dtmfmode across all trunks/extensions? |
| 19:42.01 | hi365 | well how CAN i reliable stream moh? |
| 19:42.31 | steve | for some reason I can't hear DTMF tones on a sip trunk at all |
| 19:43.13 | hi365 | [TK]D-Fender: (in case you didnt see it: http://pastebin.ca/793949) |
| 19:44.11 | [TK]D-Fender | hi365: Oh, I saw it, I just have nothing to suggest to you for it unfortunately. |
| 19:44.20 | hi365 | no prob. |
| 19:44.30 | [TK]D-Fender | steve: Go set your modes in every entry. |
| 19:47.41 | steve | [TK]D-Fender: to set it on the sip trunk inbound and outbound, do I just need one dtmfmode=rfc2833 entry in the trunk config? |
| 19:50.29 | [TK]D-Fender | steve: Yes |
| 19:50.50 | [TK]D-Fender | steve: And I'd suggest the same for [general] as well |
| 19:57.14 | *** part/#asterisk myiagy (n=myiagy@201-67-138-60.bnut3703.dsl.brasiltelecom.net.br) |
| 19:57.58 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 20:01.22 | *** join/#asterisk Strom_M (n=strom@ip70-170-60-8.lv.lv.cox.net) |
| 20:01.47 | *** join/#asterisk [_HaGGarD_] (n=info@pD9E7F20C.dip.t-dialin.net) |
| 20:01.54 | [_HaGGarD_] | Hi @all |
| 20:03.47 | [_HaGGarD_] | anybody here, using * as B2BUA / Media-Proxy who could give some hints bout media flow-through ? |
| 20:04.08 | [TK]D-Fender | [_HaGGarD_]: * is ONLY a B2BUA. |
| 20:04.28 | [TK]D-Fender | [_HaGGarD_]: So what are you looking to do? |
| 20:04.36 | [_HaGGarD_] | [TK]D-Fender: yes...sure :-) |
| 20:05.03 | *** join/#asterisk mkl1525 (n=qwertz@p5098c328.dip0.t-ipconnect.de) |
| 20:05.12 | [_HaGGarD_] | [TK]D-Fender: is there any way to allow reinvites while forcing media through * ? |
| 20:05.26 | fujin_ | canreinvite=yes |
| 20:05.29 | fujin_ | in the sip definition |
| 20:05.31 | [TK]D-Fender | [_HaGGarD_]: that is a contradiction in terms. |
| 20:05.36 | fujin_ | actually yeah |
| 20:05.40 | [_HaGGarD_] | fujin_: shure, i know :-) |
| 20:05.47 | fujin_ | canreinvite=no would force media through * |
| 20:05.54 | fujin_ | the opposite would bridge the end-to-end devices |
| 20:05.58 | fujin_ | (if available) |
| 20:06.01 | [_HaGGarD_] | [TK]D-Fender: But there are some situations where it would be very helpful.. |
| 20:06.03 | [TK]D-Fender | [_HaGGarD_]: how can it be "direct" while being "indirect"? |
| 20:06.31 | mkl1525 | Hi, trying to debug why persistentmembers doesn't work for me, but my c knowledge is quite rusty so is there a function to print a structure (ast_config) to cli similar to php var_dump? |
| 20:06.39 | [TK]D-Fender | [_HaGGarD_]: Helpful? More like impossible. * can't be OUTSIDE the stream, and IN it at the same time. |
| 20:07.26 | fujin_ | mkl1525: persistentmembers will be in the database, show database |
| 20:07.34 | fujin_ | err |
| 20:07.36 | fujin_ | database show |
| 20:07.47 | [_HaGGarD_] | [TK]D-Fender: I need to have full features of SIP compatible reinvites and having media flow through *. In the "native" way, media would flow around if canreinvite=yes... |
| 20:07.53 | fujin_ | /Queue/PersistentMembers/Helpdesk : Local/710@agents;0;0;Local/710@agents|Local/729@agents;0;0;Local/729@agents|Local/735@agents;0;0;Local/735@agents|Local/734@agents;0;0;Local/734@agents |
| 20:08.04 | fujin_ | /Queue/PersistentMembers/Helpdesk_ko : Local/996@agents;0;0;Local/996@agents|Local/710@agents;0;0;Local/710@agents |
| 20:08.54 | [TK]D-Fender | [_HaGGarD_]: I think you are failing to understand the rules of physics. If the phones have reinvited then the traffic goes DIRECT. Not through *. Is something about this not clear? |
| 20:09.30 | [_HaGGarD_] | [TK]D-Fender: why it should be impossible when cisco can do that ? Or do you mean it would be a violation of * architecture ? |
| 20:10.03 | fujin_ | cisco is a law unto themselves |
| 20:10.05 | [_HaGGarD_] | [TK]D-Fender: No, not in every situation. Image the phones would only do sdp/sendonly ? |
| 20:10.07 | fujin_ | RFC abiding? never! |
| 20:10.12 | [TK]D-Fender | [_HaGGarD_]: How can media go to * and NOT go to it at the same time? |
| 20:10.32 | fujin_ | yes, I wasn't aware that was possible |
| 20:10.46 | fujin_ | and more importantly, why would you do it? |
| 20:10.50 | [TK]D-Fender | [_HaGGarD_]: you aren't making any sense. |
| 20:10.53 | [_HaGGarD_] | [TK]D-Fender: Because some cpes are using reinvites just for simple things like moh... |
| 20:10.54 | fujin_ | let's look at the problem that you need to solve |
| 20:11.08 | fujin_ | what's wrong with that? |
| 20:11.26 | [_HaGGarD_] | fujin_: I need to see rtp for lawful interception ? :-) |
| 20:11.34 | fujin_ | then canreinvite=no |
| 20:11.38 | fujin_ | force all of your traffic |
| 20:11.42 | [_HaGGarD_] | ...in every situation.. :-) |
| 20:11.50 | fujin_ | then canreinvite=no |
| 20:11.53 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 20:11.54 | [TK]D-Fender | [_HaGGarD_]: Your choices are "all" or "nothing" |
| 20:12.17 | [_HaGGarD_] | fujin_: yes, canreinvite=no would work around in some ways. But if cpe trys to reinvite for moh, it fails :-) |
| 20:12.32 | fujin_ | replace cpe with $non_broken_cpe |
| 20:12.37 | [_HaGGarD_] | hehehe |
| 20:12.40 | fujin_ | why would it try to reinvite if SIP is telling it that it cannot? |
| 20:12.55 | fujin_ | It's no joking matter. your CPE is non-compliant. |
| 20:12.59 | [_HaGGarD_] | fujin_: just for fu**** music on hold... :-( |
| 20:13.27 | fujin_ | Are you sure you've got canreinvite=no in the sip.conf definitions? |
| 20:13.43 | [_HaGGarD_] | fujin_: sure... |
| 20:13.44 | [TK]D-Fender | fujin_: http://www.youtube.com/watch?v=Vav6b5F-_64 |
| 20:14.25 | fujin_ | o_0 |
| 20:14.26 | [_HaGGarD_] | fujin_: but then, reinvites for music on hold fails...logical. :-) |
| 20:14.38 | fujin_ | [_HaGGarD_]: no, illogical |
| 20:14.41 | fujin_ | you're a stupid idiot |
| 20:14.42 | fujin_ | eof |
| 20:14.51 | fujin_ | take it up with $cpe_supplier |
| 20:14.53 | [_HaGGarD_] | fujin_: *grmpf* |
| 20:15.10 | *** part/#asterisk Strom_M (n=strom@ip70-170-60-8.lv.lv.cox.net) |
| 20:15.23 | fujin_ | As I commonly say: you're doing it wrong. |
| 20:15.39 | *** join/#asterisk Cresl1n (n=matt@65.4.30.167) |
| 20:15.39 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 20:16.13 | [_HaGGarD_] | fujin_: thats very long way to tell a cpe supplier the their products are trash :-) |
| 20:16.41 | fujin_ | I'd go so far as to say you've got broken configuration somewhere. |
| 20:17.17 | [TK]D-Fender | In Soviet Russia phone dials YOU! |
| 20:17.39 | [_HaGGarD_] | fujin_: config is very simple. Just accept all and dial the destination |
| 20:19.01 | [_HaGGarD_] | fujin_: But anyway, it seems that the backround of that question would need too much time |
| 20:19.20 | [_HaGGarD_] | :-) |
| 20:20.01 | fujin_ | lol |
| 20:20.03 | fujin_ | in soviet stupid |
| 20:20.07 | fujin_ | fujin_ beats you |
| 20:20.31 | [_HaGGarD_] | hm... |
| 20:20.48 | mamep | can someone help with pwlib for oh323? |
| 20:27.52 | *** join/#asterisk phix (i=threat@123-243-44-131.tpgi.com.au) |
| 20:33.31 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 20:34.54 | *** join/#asterisk torrr (n=opera@bzq-79-181-124-111.red.bezeqint.net) |
| 20:35.13 | torrr | can anyone explain to me about gateway? |
| 20:35.40 | [TK]D-Fender | torrr: Gateway was a US based PC manufacturer bought out by Acre. |
| 20:35.43 | [TK]D-Fender | Acer* |
| 20:36.43 | torrr | I mean a voip gateway like : Grandstream GXW-4008 8 Port FXS IP Analog Gateway |
| 20:36.51 | [TK]D-Fender | torrr: What about it? |
| 20:37.07 | torrr | I see it and I don't understand if this is what I want or not |
| 20:37.29 | [TK]D-Fender | torrr: Tell us what you want and we'll tell you if that does the job. |
| 20:37.41 | *** join/#asterisk berniv6 (n=berni@fliwatuet.birkenwald.de) |
| 20:38.06 | torrr | I want to add to a business old phone switch, some voip ability |
| 20:38.23 | torrr | I hope also getting incoming calls from skype |
| 20:38.41 | torrr | or other pc software |
| 20:38.47 | torrr | that's it |
| 20:39.36 | [TK]D-Fender | torrr: if your PBX has CO (telco FXO ports), then yuo can plug them into this and use them for SIP calls. |
| 20:39.43 | [TK]D-Fender | torrr: as for skype... |
| 20:39.45 | [TK]D-Fender | ~skype |
| 20:39.46 | jbot | [~skype] Skype is a free VoIP software and service using a closed client and propritary protocol. Only commercial channel drivers exist, all of which are poor implementations of convoluted hacks. In general, forget about using Skype with Asterisk... |
| 20:39.54 | berniv6 | hi ... I just tried to use the asterisk ipv6 branch (asteriskv6.org) and noticed extremely bad audio quality when the codec selected is alaw/ulaw, even with just playing local files. Choosing another codec (e.g. GSM) gives way better results, using the official (not ipv6-enabled) 1.4 SVN branch does not change anything at all |
| 20:39.57 | [_HaGGarD_] | nice :-) |
| 20:40.20 | berniv6 | phones involved are snom hardphones and twinkle softphone |
| 20:41.04 | [TK]D-Fender | berniv6: this is a question for #asterisk-dev |
| 20:41.05 | torrr | ~SIP |
| 20:41.06 | jbot | i guess sip is http://www.cs.columbia.edu/sip/ X11 PPP dialer interface written in gtk+. URL: http://www.geocities.com/SiliconValley/Campus/3104/sip/ Session Initiation Protocol (see RFC 3261) |
| 20:41.31 | berniv6 | [TK]D-Fender: sure? It happens with non-ipv6 as well |
| 20:42.24 | torrr | I don't know if my PBX has it |
| 20:43.06 | [TK]D-Fender | torrr: You do not seem to understand VoIP at all. |
| 20:43.20 | [TK]D-Fender | torrr: You should go download THE BOOK right now and get reading. |
| 20:43.22 | [TK]D-Fender | ~book |
| 20:43.22 | 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 |
| 20:43.27 | torrr | That is corect |
| 20:43.30 | [_HaGGarD_] | so...thought about it again: why it shouldn't be a good idea proxying media and using reinvites for changing session params, if media needs to be proxied ? :-) |
| 20:44.09 | [_HaGGarD_] | reinvites are mainly used for changing session params, I guess... :-) |
| 20:45.59 | torrr | [TK]D-Fender: what software is used to talk with astrisk? |
| 20:46.49 | [TK]D-Fender | torrr: Asterisk *IS* software, and all sorts of other softwares can talk to it. Depends what KIND of software, and HOW you want it to talk, and what you want it to DO. |
| 20:48.13 | torrr | I thought Asterisk is a server that run only on *nix |
| 20:48.51 | torrr | I am asking of client software |
| 20:48.55 | tzafrir | torrr, it also runs on Linux |
| 20:48.56 | [TK]D-Fender | torrr: Yes, Asterisk is made to run in Unix-like environments. But your questions are dangerously vague. |
| 20:49.29 | [TK]D-Fender | torrr: Sounds like you are talking about SOFTPHONES |
| 20:49.33 | [TK]D-Fender | ~softphone |
| 20:49.33 | jbot | [~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria |
| 20:49.42 | tzanger | ~bria |
| 20:49.42 | jbot | [~bria] Bria is a NON-free SIP soft-phone for Windows, MacOSX, and Linux that can be downloaded from http://www.counterpath.com/ This includes licensed audio & video codecs and is full-featured. |
| 20:49.42 | tzanger | ? |
| 20:49.45 | tzanger | ahh |
| 20:49.52 | tzanger | non-free, no wonder I don't look at it |
| 20:50.05 | [TK]D-Fender | tzanger: I was previously called eyeBeam |
| 20:50.08 | [TK]D-Fender | It* |
| 20:50.10 | tzanger | ah |
| 20:50.25 | fujin_ | No, eyeBeam and Bria are seperate products. |
| 20:50.43 | fujin_ | Bria is *newer*, with a different inetrface, eyeBeam is just the non-free version of X-lite (i.e.; all features enabled) |
| 20:50.44 | [TK]D-Fender | tzanger: Should have marketed it as "Chia : the soft-phone that grows on you!" |
| 20:51.25 | torrr | [TK]D-Fender: how do I interface the office real phones? |
| 20:51.32 | tzanger | define real phones |
| 20:51.36 | tzanger | an existing KSU or PBX? |
| 20:51.46 | [TK]D-Fender | fujin_: Ah... |
| 20:52.27 | [TK]D-Fender | torrr: Stop now and read the book. You need to understand your own PBX FIRST and THEN what other technologies are out there so that you can see what kind of interfacing is possible |
| 20:55.07 | [TK]D-Fender | tzanger / fujin_ : eyebeam added |
| 20:56.31 | torrr | tzanger: I don't know :( it is a Panasonic switch, it has 8 incoming lines, it is probably 7 year old |
| 20:56.40 | tzanger | yeah it's a KSU then |
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| 20:57.17 | harlequin516 | How do I implement an Answering Machine Detect using agi? |
| 20:57.24 | torrr | tzanger: what does it mean for my posiblities? |
| 20:57.35 | harlequin516 | I mean from an AGI? |
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| 21:00.01 | harlequin516 | There's no BackgroundDetect application available as an AGI cmd. What to do? |
| 21:00.33 | *** join/#asterisk torr (n=opera@bzq-79-181-124-111.red.bezeqint.net) |
| 21:00.41 | torr | was down |
| 21:01.10 | [TK]D-Fender | [15:52]<[TK]D-Fender>torrr: Stop now and read the book. You need to understand your own PBX FIRST and THEN what other technologies are out there so that you can see what kind of interfacing is possible |
| 21:01.48 | torr | [TK]D-Fender: I have downloaded it , and I intend to read , though it doesn't look like a light reading |
| 21:02.43 | torr | 604 pages |
| 21:02.43 | [TK]D-Fender | torr :read the early chapters that explain about telephony, voip, pstn, etc. |
| 21:02.43 | [TK]D-Fender | torr : You need to understand what kind of hardware relates to what, and how * can fit in. |
| 21:03.28 | torr | [TK]D-Fender: ok |
| 21:05.37 | [TK]D-Fender | harlequin516: http://www.google.ca/search?hl=en&q=asterisk+answering+machine+detection&btnG=Google+Search&meta= |
| 21:06.47 | fujin_ | [TK]D-Fender: roger that |
| 21:08.16 | [TK]D-Fender | Another victory for jfgi! :) |
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| 21:19.33 | icewaterman | hi, i've found a bug in misdn. how can i report it? |
| 21:20.12 | icewaterman | there is actually little info on where or to whom report a bug with misdn |
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| 21:21.35 | [TK]D-Fender | icewaterman: Go look on their site |
| 21:22.36 | icewaterman | [TK]D-Fender: well, the site is not very informative about bugs. |
| 21:23.52 | icewaterman | and sending a mail to isdn4linux devs will probably just cause the message to be written to /dev/null |
| 21:24.19 | [TK]D-Fender | icewaterman: "Life sucks but rarely swallows". |
| 21:28.22 | [TK]D-Fender | ok, checkout time, back in a while... |
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| 21:44.47 | ramonpeek | Question: |
| 21:44.48 | ramonpeek | Many user have issue with the g729 registration utility. |
| 21:44.48 | ramonpeek | I understands how it works and can get succesfull registrations. |
| 21:44.48 | ramonpeek | But the registration utility demands using eth0 for registration. |
| 21:44.48 | ramonpeek | And that it a really nasty demand. |
| 21:44.48 | ramonpeek | Configuring an Asterisk box to use eth0 to have internet access is often a lot of work. |
| 21:44.50 | ramonpeek | In many of my Asterisk boxes eth0 is a glasfiber port!! |
| 21:44.52 | ramonpeek | And it almost impossible to change that port to eth0 when it's running a live configuration. |
| 21:44.54 | ramonpeek | To bad the registration utilty is not open-source itself... ;-) |
| 21:44.56 | ramonpeek | Or else I would have commited code to add an extra argument to set the interface parameter.. |
| 21:44.58 | ramonpeek | Something like "register -i eth3" |
| 21:45.00 | ramonpeek | How difficult can this be? |
| 21:45.02 | ramonpeek | Does anyone at Digium feel the need for this feature request too? |
| 21:46.00 | chrizz- | it should be possible to decide the interface-name of each network card by using udev rules |
| 21:46.32 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 21:47.05 | Qwell | ramonpeek: if eth0 isn't a nic, then...well...rename it |
| 21:47.26 | Qwell | it uses every eth* interface |
| 21:47.28 | JT | ramonpeek: how is it even at all relevant that eth0 is fibre? |
| 21:47.33 | Qwell | JT: it's not |
| 21:47.36 | ramonpeek | I know... but you cant do that on a running system where the current eth0 is in use... you need to disconnect many users.. that is my problem |
| 21:47.46 | JT | ramonpeek: that makes no sense. |
| 21:47.51 | Qwell | what does it being in use have to do with anything? |
| 21:48.05 | JT | and wtf does it matter if it's copper or fibre? |
| 21:48.08 | ramonpeek | well.. |
| 21:48.45 | ramonpeek | the systems are often live systems.. shutting down asterisk for a couple of seconds for a restart is not a big issue |
| 21:48.58 | Qwell | why do you need to restart anything to register your g729 license? |
| 21:49.08 | ramonpeek | but restarting takes alot longer if you first need to rename your interfaces and then back again |
| 21:50.03 | ramonpeek | you need to restart asterisk to activate a newly registred g729 from digium |
| 21:50.08 | Qwell | no you don't |
| 21:50.12 | ramonpeek | oh... |
| 21:50.20 | ramonpeek | the manual says it does |
| 21:50.27 | JT | ramonpeek: so how is the fibre relevant again? |
| 21:50.37 | Qwell | but that isn't the point |
| 21:50.47 | Qwell | what does eth0 have to do with anything? |
| 21:50.50 | ramonpeek | well in my systems eth0 by default is teh fibre port |
| 21:50.53 | Qwell | and? |
| 21:50.56 | Qwell | why is that a problem? |
| 21:50.58 | JT | WHY DOES IT MATTER IF IT'S FIRE? |
| 21:50.59 | ramonpeek | an often its not connected |
| 21:51.01 | JT | FIBE |
| 21:51.03 | Qwell | and? |
| 21:51.03 | JT | bah |
| 21:51.14 | Qwell | again - not a problem |
| 21:51.17 | JT | ramonpeek: you're crazy |
| 21:51.25 | ramonpeek | yeah yeah |
| 21:51.29 | ramonpeek | lol |
| 21:51.32 | Qwell | so what IS the problem? |
| 21:51.33 | JT | ramonpeek: HW MAC addresses are there regardless of if anything is plugged in or not |
| 21:51.39 | JT | ramonpeek: read an Ethernet 101 |
| 21:53.10 | JT | assumptions can lead to much embarrasment |
| 21:53.22 | ramonpeek | come on.. think with me.. if my system is using eth0 for its internal VoIP network with many active users connected to it. I cannot just reprogram the interface to a different setting to get internet access.. that would upset too many users |
| 21:53.37 | Qwell | WHY do you need to "reprogram" anything? |
| 21:53.43 | JT | ramonpeek: you don't need to reprogram the interface. you're crazy. |
| 21:53.56 | JT | ramonpeek: your understanding of the situation is completely incorrect |
| 21:54.09 | JT | delete what you think you know and start again |
| 21:54.30 | ramonpeek | how else am I supposed to get internet access on eth0 when it normally is on an internal LAN that is NOT connected to the internet |
| 21:54.41 | Qwell | umm |
| 21:54.43 | Qwell | proper routes? |
| 21:54.47 | JT | default route |
| 21:54.53 | JT | TCP/IP 101 |
| 21:54.57 | Qwell | -1 |
| 21:54.57 | ramonpeek | Ahh... that's easy.. |
| 21:55.04 | ramonpeek | but that doesn't work! |
| 21:55.08 | Qwell | yes, it does |
| 21:55.23 | ramonpeek | I get connected to the server.. no problem.. but registration fails! |
| 21:55.25 | Qwell | register has NO say *at all* which interface is used for traffic |
| 21:55.44 | ramonpeek | According to the Digium guidelines it does.. |
| 21:56.03 | ramonpeek | and when I try it through eth0 it indeed works. |
| 21:56.12 | Qwell | what "guidelines"? |
| 21:56.22 | ramonpeek | same settings on eth1... no working |
| 21:56.38 | ramonpeek | Hold on let me get you the URL |
| 21:57.15 | ramonpeek | see this: http://ftp.digium.com/pub/telephony/codec_g729/README |
| 21:57.36 | ramonpeek | and look under item 4 |
| 21:58.03 | file | it requires eth0 to create a unique ID for your computer |
| 21:58.03 | Qwell | and? |
| 21:58.06 | ramonpeek | it says: The name of the first |
| 21:58.06 | ramonpeek | <PROTECTED> |
| 21:58.06 | ramonpeek | <PROTECTED> |
| 21:58.15 | Qwell | it does *NOT* require that eth0 be *connected* |
| 21:58.31 | ramonpeek | that was my thought too... in the beginning |
| 21:59.02 | Qwell | if you run ifconfig, do you see eth0? |
| 21:59.09 | ramonpeek | but when today again... a had a system running multiple eth ports.. and the first port was eth0 |
| 21:59.25 | ramonpeek | and it was shown in ifconfig.. yes |
| 21:59.31 | Qwell | then it's working fine |
| 21:59.45 | Qwell | Does it give you a hostid? |
| 22:00.19 | ramonpeek | Registration didnt work... but a did have internet access.. I changed the routes and got internet access through eth0 and wham .... it worked!! |
| 22:00.29 | ramonpeek | that was the second time on a separate machine |
| 22:00.32 | Qwell | then your routes are broken |
| 22:00.39 | Qwell | not a problem with register |
| 22:01.27 | ramonpeek | Uhm.. strange how i can access everything on the internet then whilst the registration utility fails.. |
| 22:01.46 | Qwell | including ssl sites? |
| 22:01.55 | ramonpeek | Could it be that firewall rules may cause problems |
| 22:01.57 | Qwell | do you even get the menu? |
| 22:02.38 | ramonpeek | Does traffic come inbound over a new connection?? |
| 22:02.41 | Qwell | no |
| 22:03.00 | ramonpeek | ssl sites worked too.... weird huh |
| 22:03.16 | Qwell | it's clearly a problem with your network |
| 22:03.51 | ramonpeek | OK.. well I guess you have quite some experience with the registration utility too |
| 22:04.01 | ramonpeek | I'll do some more testing |
| 22:04.21 | ramonpeek | Is there a dummy key that could be used.? |
| 22:04.24 | Qwell | no |
| 22:04.43 | Qwell | about all I can tell you, is to run it with -v |
| 22:04.45 | ramonpeek | Uhmm.... to bad... costly testing :) |
| 22:05.11 | ramonpeek | Ah .. Ok that's usefull verbose mode.. didn't know it exists. |
| 22:06.35 | ramonpeek | thanks so far.. |
| 22:11.23 | JT | you know you can call digium support, right? |
| 22:13.25 | Siya | is the mailbox extension setting callable in extensions.conf? |
| 22:13.40 | Siya | can't find a suitable variable... |
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| 22:45.24 | alephcom | good day everyone |
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| 23:07.12 | Siya | omg there must be an easy way to read the extension from a channel... |
| 23:07.41 | JT | ${EXTEN} ? |
| 23:07.55 | Siya | I want to remove the channel type and random trailer so I just have the channel middle bit which tells me where the call came from |
| 23:08.01 | Siya | JT no |
| 23:08.14 | Siya | EXTEn is the dialled extension |
| 23:08.39 | JT | please be more desriptive then |
| 23:08.40 | Siya | I want the caller and I can't (don't want to) use the callerid |
| 23:08.55 | JT | well that's just silly |
| 23:09.00 | Siya | hehe |
| 23:09.14 | Siya | there must be a way though |
| 23:09.26 | JT | why can't you user callerid? |
| 23:09.32 | Siya | VoiceMailMain("SIP/102-083dce00", "s0031bladibla") |
| 23:09.49 | Siya | mailbox number is 102 not 0031bla... |
| 23:11.05 | Siya | being able to read the 102 out of ${channel} would allow me to set a 'bespoke' callerid per extension if I want to without having to hack extensions.conf too much |
| 23:12.07 | JT | bespoke is a text string |
| 23:12.16 | JT | more suitable for callerid name than number |
| 23:13.45 | Siya | JT I meant customised not "bespoke" as CallerID(name) |
| 23:14.19 | JT | what do you mean? |
| 23:14.28 | Siya | So I've fixed my calledid (external/internal) issue |
| 23:14.37 | fujin_ | I think you can use $CUT |
| 23:14.40 | fujin_ | to achieve that |
| 23:15.14 | Siya | but am left with the extension with a non-default callerid which has to enter it's mailbox number when it dials voicemail |
| 23:15.31 | Siya | fujin_: I saw that on the wiki but don't quite understand how it works |
| 23:15.54 | Siya | Ooh wife is calling bed time, will read up 2morrow morning |
| 23:15.58 | Siya | gnight all |
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