| 00:07.46 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 00:08.47 | *** join/#asterisk luckman212_ (~luckman21@2001:470:8abb:0:211:32ff:fe10:cdc1) |
| 00:20.02 | *** join/#asterisk bmg505 (~leon@196-209-163-155.dynamic.isadsl.co.za) |
| 00:26.41 | *** join/#asterisk asr33 (~asr33@unaffiliated/asr33) |
| 00:28.47 | *** join/#asterisk engrxyz (~cvzv@host81-150-217-167.in-addr.btopenworld.com) |
| 00:36.41 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 00:36.41 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 00:43.33 | *** join/#asterisk Russ (~russ@pool-74-100-57-85.lsanca.fios.verizon.net) |
| 00:43.41 | pbxbrian | tompaw: ask the parcipiants would they like mince pies |
| 00:44.01 | pbxbrian | tompaw: if you don't get response they are stalled calls |
| 00:56.16 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 00:56.45 | joako | What command is needed for Asterisk 1.8 to set the callerid name? |
| 00:57.39 | joako | The Master.csv shows the correct name, but the phones only display the number |
| 01:01.21 | joako | Also on some Polycom phones they no longer show any recieved call or missed call history |
| 01:02.00 | *** join/#asterisk ATS63 (dce90b63@gateway/web/freenode/ip.220.233.11.99) |
| 01:11.12 | carrar | show function CALLERID |
| 01:11.39 | carrar | Polycom admin guide FTW |
| 01:15.19 | *** join/#asterisk nightrid3r (~kvirc@62.205.66.220) |
| 01:15.32 | Primer | Hi, I created a digitmap on my polycom 501 that goes: 100x|...the default.... but when I type 1004 it goes back to a dial tone and the display reads "Enter more digits". If I type 1002, 1003, 1005,...1009, it works fine. There is no other digitmap pattern that matches 1004. Anyone know what this might be? |
| 01:15.33 | carrar | See Feature in the admin guide |
| 01:16.21 | carrar | user error in the digit map |
| 01:16.27 | Primer | How so? |
| 01:16.28 | SuperNull | they all need to standardize on a dialplan scheme we have like 3-4 ATAs and they are all different.. |
| 01:16.35 | carrar | How so what |
| 01:16.35 | Primer | 100x is not what I think it is? |
| 01:16.47 | SuperNull | 100x is probably 100 times more than you think it is .. |
| 01:16.56 | carrar | heh |
| 01:17.32 | Primer | Then how is it 100{1,2,3,5,6,7,8,9} all work? Only 1004 does not |
| 01:17.38 | carrar | The adminguide is a beautifull thing |
| 01:19.07 | carrar | I got a gas can and a set of keys, why am I not going anywhere!!@#!@# |
| 01:19.36 | carrar | ~pastebin |
| 01:19.37 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 01:27.16 | carrar | That was a hint to showus your complete config |
| 01:27.54 | carrar | stops hinting |
| 01:29.07 | joako | carrar: My CLI log shows: Set("SIP/6000-00000083", "CALLERID(name)="TECH ASSTNC CTR"") and my Master.CSV shows the correct name, but the phone doesn´t show the name. Everything was working under Asterisk 1.4 |
| 01:29.43 | carrar | Do you see the name being passed to the phone in a SIP Capture? |
| 01:30.00 | carrar | also |
| 01:30.04 | carrar | remove the " |
| 01:30.10 | carrar | no quote marks |
| 01:31.01 | *** join/#asterisk serafie (~erin@76.73.167.231) |
| 01:33.22 | joako | carrar: Yes I see in the INVITE From: ""TECH ASSTNC CTR"" <sip:xxxxxxxxx@192.168.0.165>;tag and also P-Asserted-Identity |
| 01:33.37 | carrar | double quotes |
| 01:33.39 | carrar | remove a set |
| 01:34.26 | carrar | Your phone is probably seeing "" |
| 01:34.35 | Primer | SuperNull: btw, I see what you did there (missed it at first) |
| 01:35.14 | Primer | So anyhow, I've tried this as well: |
| 01:35.16 | Primer | <digitmap dialplan.digitmap="100[0-9]|[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT" dialplan.digitmap.timeOut="3|3|3|3|3|3" /> |
| 01:35.46 | carrar | [0-1][2-9]xxxxxxxxx |
| 01:35.50 | Primer | Note that the part after "100[0-9]|" is the default polycom digitmap |
| 01:36.19 | carrar | what were you dialing? |
| 01:36.22 | Primer | 1004 |
| 01:37.01 | Primer | 100[0-9] and 100x will match 1001, 1002, 1003, 1005, 1006, 1007, 1008, 1009, but NOT 1004 |
| 01:44.51 | carrar | What happens if you do |
| 01:44.54 | carrar | dialplan.digitmap.timeOut="0|3|3|3|3|3|3" /> |
| 01:45.26 | carrar | Though it's a exact match |
| 01:45.41 | carrar | maybe you have something else set |
| 01:51.49 | joako | carrar: I have the name in a variable, It has no quotes. Asterisk is inserting the quotes around the variable |
| 01:51.58 | joako | And the same configuration works fine with Asterisk 1.4 |
| 01:52.33 | joako | In the Master.csv it shows up as """""TECH ASSTNC CTR"""" <+1xxxxxxxxxx>" |
| 01:53.19 | carrar | yeah you have a quote issue |
| 01:53.45 | carrar | fix that |
| 01:53.52 | carrar | and you probably fix the caller id name on the phone |
| 01:54.03 | *** join/#asterisk SupaYoshi (SupaYoshi@ip4da5d319.direct-adsl.nl) |
| 01:54.09 | SupaYoshi | can asterisk actually play mp3 files? |
| 01:54.25 | carrar | it can |
| 01:54.43 | joako | carrar: If I remove the quotes when I set the variable, then the variable becomes just ¨TECH¨ (no quotes) |
| 01:55.29 | carrar | so make it one word without quotes to test |
| 01:58.09 | Primer | carrar: I just stuck with the default of 3 for that |
| 02:00.41 | joako | I guess tomorrow if it doesn´t work I´ll just ¨downgrade¨ it back to Asterisk 1.4. Not woth this hassle. |
| 02:06.48 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 02:16.57 | Penguin | joako: Downgrade from what? |
| 02:17.08 | carrar | 1.8 works great |
| 02:17.12 | *** join/#asterisk jsjc (~Adium@91.Red-83-60-132.dynamicIP.rima-tde.net) |
| 02:17.26 | carrar | even 10 |
| 02:17.31 | carrar | I use 10 in production |
| 02:19.01 | Penguin | I've been using 1.8 for a while now, and it's pretty good. |
| 02:19.11 | Penguin | And it doesn't have issues with CALLERID(). |
| 02:20.14 | Penguin | And never quote variables. Ever. |
| 02:20.47 | carrar | well we haven't seen anything |
| 02:20.52 | Penguin | No version that I know of would drop the rest of the data beyond the first space. That just simply is not how it works. |
| 02:24.22 | Penguin | Set(CALLERID(name)=TECH ASSTNC CTR) <---- joako: this is the correct way to set only the name portion of the callerid. |
| 02:24.44 | *** join/#asterisk appleboy (~appleboy@about/cooking/nakedchef/apple/tarts) |
| 02:26.27 | Penguin | To set the name and number at one time, Set(CALLERID(all)=TECH ASSTNC CTR <3145551212>) |
| 02:29.01 | *** join/#asterisk deo (~deo@58.71.19.178) |
| 02:33.01 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 02:35.49 | joako | Penguin: ¨Downgrade from 1.4 to 1.8. I understand that. The Master.CSV has the correct name. The SIP debug shows the correct name. The Polycom phone doesn´t show the name or even any recieved or missed call logs. Only the number while its ringing |
| 02:36.08 | *** join/#asterisk retentiveboy (~retentive@74-95-28-38-Atlanta.hfc.comcastbusiness.net) |
| 02:36.30 | Penguin | Then the problem is the phone, not asterisk. |
| 02:37.17 | *** join/#asterisk elico (~Thunderbi@bzq-79-182-198-188.red.bezeqint.net) |
| 02:37.32 | joako | Penguin: The same phone with the same configuration was working just fine with Asterisk 1.4. The same Asterisk configuration with the same phone configuration doesn´t work with Asterisk 1.8 |
| 02:38.58 | SupaYoshi | can someone tell me a converter |
| 02:39.00 | SupaYoshi | for mp3 to ulaw |
| 02:39.01 | SupaYoshi | or gsm |
| 02:39.53 | *** join/#asterisk MissionCritical (~MissionCr@unaffiliated/missioncritical) |
| 02:39.55 | Penguin | I would use sox. |
| 02:40.04 | SupaYoshi | I tried that |
| 02:40.11 | SupaYoshi | but i cant find the correct syntax |
| 02:40.17 | SupaYoshi | can you help me? i been busy 3 hours+ |
| 02:40.29 | Penguin | If it can't go straight to ulaw, I'd go from mp3 to wav, then use asterisk's built-in tool to convert wav to ulaw. |
| 02:40.45 | SupaYoshi | em |
| 02:40.49 | SupaYoshi | yeah i tried that |
| 02:40.53 | SupaYoshi | then i got all sorta errors |
| 02:40.59 | SupaYoshi | please help me :( i am desperate |
| 02:41.25 | Penguin | sox infile.mp3 -c1 -r 8000 outfile.wav |
| 02:41.34 | *** join/#asterisk moy (~moy@173.239.155.74) |
| 02:42.58 | SupaYoshi | ok and then :D |
| 02:43.12 | SupaYoshi | that worked yay! :) |
| 02:43.18 | SupaYoshi | now to gsm or ulaw correct? |
| 02:43.22 | Penguin | In asterisk, file convert outfile.wav outfile.ulaw |
| 02:43.33 | SupaYoshi | sox WARN rate: rate clipped 1 samples; decrease volume? |
| 02:43.57 | *** join/#asterisk retentiveboy (~retentive@74-95-28-38-Atlanta.hfc.comcastbusiness.net) |
| 02:44.20 | SupaYoshi | emm |
| 02:44.22 | SupaYoshi | Ok! :D |
| 02:44.25 | SupaYoshi | lets just do that |
| 02:52.21 | SupaYoshi | HEYYYY |
| 02:52.25 | SupaYoshi | IT WORKS! thnx :D |
| 02:52.32 | SupaYoshi | however the quality of the song is horrible... |
| 02:58.58 | SupaYoshi | how do i make it play mp3 files on hold? |
| 02:59.07 | SupaYoshi | or wav in high quality |
| 02:59.33 | Penguin | It's pretty much the same way for both mp3, wav, or ulaw. |
| 03:00.02 | Penguin | If you built asterisk with mp3 support, configure it as per the moh sample config file. |
| 03:00.56 | Penguin | If you didn't build it with mp3 support, you'll need to take care of that part first. |
| 03:01.12 | SupaYoshi | well i got a raspberry pi |
| 03:01.20 | SupaYoshi | with freepbx and the latest asterisk :) |
| 03:01.30 | SupaYoshi | however i dont know how to check if i have mp3 support |
| 03:02.43 | Penguin | module show like mp3 |
| 03:03.34 | Penguin | If you see format_mp3.so, I think you're good. |
| 03:03.46 | SupaYoshi | app_mp3.so |
| 03:03.58 | SupaYoshi | Desc: Silly MP3 Application |
| 03:05.16 | Penguin | Did you build that asterisk yourself? |
| 03:05.23 | SupaYoshi | no :P |
| 03:05.43 | SupaYoshi | http://www.raspberry-asterisk.org/ |
| 03:07.59 | SupaYoshi | hello? |
| 03:23.14 | SupaYoshi | hello? |
| 03:24.00 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 03:45.29 | ChannelZ | Can you hear me now? |
| 03:50.44 | SupaYoshi | ugh |
| 03:50.52 | SupaYoshi | lol :P i cant set it to play mp3s |
| 03:51.00 | SupaYoshi | it just wont accept mp3 files lol |
| 03:51.32 | Penguin | (2100.56) <Penguin> If you didn't build it with mp3 support, you'll need to take care of that part first. |
| 03:51.42 | Penguin | You clearly didn't take care of that part. |
| 03:51.53 | SupaYoshi | :P i just tried |
| 03:51.56 | SupaYoshi | I intalled asterisk-mp3 |
| 03:52.05 | SupaYoshi | But it doesnt seem to work just yet. |
| 03:52.16 | SupaYoshi | It's on a raspberry pi, and running make doesnt work |
| 03:52.20 | SupaYoshi | help please? |
| 03:52.42 | SupaYoshi | perhaps anyone that has teamviewr can help? ive been up from 5 pm till 5 am now |
| 03:52.47 | SupaYoshi | trying to set this thingup lol |
| 03:53.05 | Penguin | Did you load format_mp3.so? |
| 03:53.05 | SupaYoshi | make menuselect etc wont do |
| 03:53.10 | SupaYoshi | No i didnt, |
| 03:53.13 | SupaYoshi | I dont know how to |
| 03:53.21 | SupaYoshi | internet tells me bs. cus its not working for this thing |
| 03:53.35 | SupaYoshi | make menuselect etc. all bs |
| 03:53.36 | Penguin | If the asterisk-mp3 package provides it, module load format_mp3.so |
| 03:54.12 | SupaYoshi | [2012-12-21 03:53:59] WARNING[18183]: loader.c:783 inspect_module: Module 'format_mp3.so' was not compiled with the sa |
| 03:54.19 | SupaYoshi | me compile-time options as this version of Asterisk. |
| 03:54.28 | Penguin | So you'll need to upgrade the asterisk pacakge, too. |
| 03:54.39 | SupaYoshi | how? :D |
| 03:54.46 | SupaYoshi | sudo apt-get upgrade? :) |
| 03:55.14 | Penguin | Whatever you did to install asterisk-mp3, consider using the same method to install the compatible asterisk package. |
| 03:55.33 | SupaYoshi | k doin the sudo apt-get upgrade then |
| 03:55.44 | SupaYoshi | :) i got the raspberry pi today |
| 03:57.20 | SupaYoshi | or should i type sudo apt-get install asterisk |
| 03:57.24 | SupaYoshi | but i already have asterisk :P |
| 03:57.29 | SupaYoshi | im updating atm. takes awhile |
| 04:07.45 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 04:08.24 | *** join/#asterisk radic (~radic@dslb-178-010-255-098.pools.arcor-ip.net) |
| 04:13.20 | *** join/#asterisk rue_mohr (~rue@24-207-103-226.eastlink.ca) |
| 04:13.37 | rue_mohr | I have an mgcp problem |
| 04:13.50 | ChannelZ | better than a meth problem |
| 04:13.54 | rue_mohr | :) |
| 04:14.00 | rue_mohr | I have two gateways |
| 04:14.13 | rue_mohr | and when I try to specify the second one, the first one stops working |
| 04:14.30 | rue_mohr | does mgcp.conf have issues? |
| 04:15.14 | rue_mohr | both configs work fine on their own |
| 04:15.34 | rue_mohr | if there are both in there, the first one dosn't work |
| 04:19.34 | ChannelZ | to which parameter are you referring? |
| 04:26.05 | *** join/#asterisk freeedrich| (friedrich@perplexa.be) |
| 04:28.13 | *** join/#asterisk b2 (~ion@pdpc/supporter/active/beckb) |
| 04:29.15 | *** join/#asterisk camerin (hoax@newelite.bshellz.net) |
| 04:29.22 | *** join/#asterisk caveat- (hoax@gateway/shell/bshellz.net/x-tstanifpalukbbrw) |
| 04:31.27 | rue_mohr | the [...] section for the gateway |
| 04:31.39 | rue_mohr | if I have one, its ok, if I have two specified, the first one tanks |
| 04:32.17 | rue_mohr | I get errors of the same nature as if the gateway is not accessable via the network |
| 04:32.26 | ChannelZ | sound like possibly a syntax error preventing the whole thing from being parsed |
| 04:32.42 | rue_mohr | then why doe sthe last one work ok? |
| 04:34.53 | rue_mohr | http://paste.debian.net/217884/ |
| 04:35.12 | ChannelZ | do the 'mgcp show endpoitns' commands show both? |
| 04:35.13 | rue_mohr | I dont have access to the system now, but in that example, vog-2 would work, and vog-1 wont |
| 04:35.38 | rue_mohr | yes it does |
| 04:35.59 | rue_mohr | but there are packet timeouts to (in example vog-1) |
| 04:36.27 | rue_mohr | if I swap the sections, again, the first one defined wont work |
| 04:36.33 | ChannelZ | define 'work' - do you get an error on the console or.. |
| 04:36.53 | rue_mohr | network errors to do iwth packets failing |
| 04:38.02 | rue_mohr | as I dont have the system infront of me, guesses would be nice |
| 04:38.22 | rue_mohr | I didn a lot of things to try to diagose |
| 04:38.35 | rue_mohr | oh actually |
| 04:39.00 | rue_mohr | an important point is that the errors didn't happen till the rtp session tried to be created |
| 04:39.05 | ChannelZ | I don't know crap of MGCP so I'm just hitting basics |
| 04:39.15 | rue_mohr | ok |
| 04:39.46 | rue_mohr | both boxes use an rtp start port of 10000, its all local network, is that cosher? |
| 04:40.14 | rue_mohr | I should think so |
| 04:40.30 | ChannelZ | you specify a range but yes |
| 04:40.57 | rue_mohr | hmm, I wonder if the gateways cant go over 4 rtp ports, |
| 04:41.12 | rue_mohr | but I didn't even populate 4 channels, so it still should have been fine |
| 04:41.23 | ChannelZ | is that all the larger the range you specified? |
| 04:41.49 | *** join/#asterisk Mission-Critical (~MissionCr@unaffiliated/missioncritical) |
| 04:42.32 | rue_mohr | the gateways are trickey, http://www.voip-info.org/wiki/view/VoipPack+VOG+4000+and+asterisk thats my horrid article on what I know about them |
| 04:43.59 | ChannelZ | oh I thought you were referring to rtp.conf or something |
| 04:44.49 | rue_mohr | no, rtp.conf is standard |
| 04:45.06 | rue_mohr | I modified the gateways to asterisk for rtp |
| 04:50.54 | *** join/#asterisk Carp1 (~Carp1@cpe-74-70-40-35.nycap.res.rr.com) |
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| 05:49.55 | *** join/#asterisk carrar (tim@osburn.com) |
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| 06:26.23 | *** join/#asterisk Hesit (~asteriskp@ip70-171-207-117.tc.ph.cox.net) |
| 06:26.29 | Hesit | Question for someone |
| 06:26.38 | Hesit | cp: cannot create regular file `/etc/asterisk/indications.conf': Permission denied |
| 06:26.50 | Hesit | This is on my asteriskpbx user |
| 06:27.06 | carrar | probably owned by root and you are not root |
| 06:27.10 | Hesit | I feel like it's going to do this with every config sample I need to make |
| 06:27.42 | Hesit | sudo chown name:name /etc/asterisk/ ? |
| 06:27.43 | carrar | Why don't you install the samples as part of the initial isntall? |
| 06:27.58 | Hesit | I'm following the docs |
| 06:28.33 | Hesit | would it just be chown name:name /etc/asterisk/? |
| 06:28.37 | Hesit | no sudo? |
| 06:28.43 | carrar | Are you installing from source? |
| 06:28.44 | Hesit | Centos btw |
| 06:28.47 | carrar | and read the README? |
| 06:28.51 | Hesit | no |
| 06:29.02 | carrar | well |
| 06:29.03 | Hesit | http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/installing_how_to_install_it.html |
| 06:29.03 | carrar | there you go |
| 06:29.22 | Hesit | I've followed it up to their |
| 06:29.32 | carrar | installing from svn? |
| 06:29.34 | Hesit | I can't seem to set the permissions for my asteriskpbx account though |
| 06:29.38 | Hesit | yes |
| 06:29.45 | Hesit | I can do all this through the root |
| 06:29.46 | carrar | so then you still have the README file |
| 06:29.54 | carrar | 1.4 is old |
| 06:29.59 | carrar | OLD OLD |
| 06:30.06 | Hesit | 1.8 |
| 06:30.11 | Hesit | I'm using 1.8 |
| 06:30.17 | carrar | I'd use 10 |
| 06:30.24 | Hesit | well I'm using 1.8 |
| 06:30.26 | carrar | but 1.8 is fine also |
| 06:30.34 | Hesit | It's just a test run |
| 06:30.39 | carrar | so in your src directory you have a README file |
| 06:30.40 | Hesit | I want to get a feel for the pbx |
| 06:31.53 | carrar | ~book |
| 06:31.53 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 06:32.02 | carrar | hrmm |
| 06:32.03 | carrar | ok |
| 06:32.11 | Hesit | I don't have a readme in my src |
| 06:32.16 | Hesit | just "asterisk-complete" |
| 06:32.18 | carrar | capital |
| 06:32.29 | Hesit | captial? |
| 06:32.39 | carrar | You should have several README files |
| 06:32.53 | Hesit | not on this account I guess |
| 06:33.34 | Hesit | ahhh |
| 06:33.40 | Hesit | It's in my 1.8 file |
| 06:34.03 | Hesit | in /src/asterisk-complete/asterisk/1.8/ |
| 06:34.07 | kaldemar | Hesit: "sudo make samples" |
| 06:34.17 | Hesit | kaldemar |
| 06:34.21 | carrar | ok |
| 06:34.23 | carrar | I just did "svn co http://svn.asterisk.org/svn/asterisk/branches/1.8" |
| 06:34.27 | carrar | you did that also right? |
| 06:34.38 | Hesit | What happens when I want to edit them and I don't have the permissions to do so |
| 06:34.39 | kaldemar | Hesit: that will put sample config files for you in /etc/asterisk. the page does not have the command. |
| 06:34.42 | Hesit | I hate going back to the root everytime |
| 06:34.58 | carrar | in the 1.8 directory you have 3 README files |
| 06:35.03 | carrar | read them |
| 06:35.24 | carrar | for your first run through install everything as root |
| 06:35.25 | kaldemar | Hesit: you either use sudo with your editor command or change the permissions, or better - ownership, of the files. |
| 06:35.31 | carrar | as the user root |
| 06:35.51 | carrar | read the part |
| 06:35.51 | carrar | --- NEW INSTALLATIONS --------------------------------------------------------- |
| 06:36.15 | carrar | key part you are missing is make samples |
| 06:39.01 | Hesit | kaldemar: ty |
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| 07:23.12 | schmidts | good morning |
| 07:26.31 | CommaCrazy | morning |
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| 07:56.23 | horsefish | hello all, im relatively new to asterisk and am hoping someone could offer a helping hand for a task at hand |
| 07:56.59 | *** join/#asterisk horsefish (~storm@109-163-233-42.static.srsvps.com) |
| 07:58.11 | horsefish | hello all, im having some trouble with asterisk for a specific case involving automatic dial |
| 07:58.34 | horsefish | if someone could help out please pm |
| 08:00.24 | kaldemar | ~ask |
| 08:00.25 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 08:01.38 | horsefish | im trying to setup an automatic dial between 2 parties, in this instance they'll be 101 and 102 |
| 08:02.09 | horsefish | both parties have their independent SIP phone (physical) and will be set up to automatically receive a call when requested |
| 08:03.08 | kaldemar | requested how? |
| 08:03.10 | horsefish | now im having trouble configuring a .call script to place in /var/spool/asterisk/outgoing with the correct parameters to dial continuously from 101 to 102 until 102 answers, indefinitely and infinitely recurring |
| 08:03.37 | horsefish | and vice versa |
| 08:04.08 | kaldemar | connect the other end to an extension that keeps on dialing. |
| 08:04.17 | horsefish | the point is that when one picks up the phone, it immediately calls the other party (102 => 101 &c) |
| 08:05.24 | horsefish | ok so is the Channel the originating party or just the SIP line that is performing the call? |
| 08:07.09 | kaldemar | asterisk is the originating party. Channel defines what is dialed first. if that destination answers, then it is connected to an extension or an application. you need to connect it to an extension that does what you want, which is keeps on dialing. |
| 08:07.19 | horsefish | and im assuming the Extension is the destination; enacting this scenario with SIP clients and manually answering the calls, the Extension does not originate from 101 and when 102 answers the calls are not <-> connected, instead they disconnect immediately and says goodbye. |
| 08:08.01 | horsefish | how do i go about doing this in the .call file? |
| 08:08.13 | kaldemar | what do you have now? |
| 08:08.17 | kaldemar | ~pb |
| 08:08.17 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 08:08.35 | horsefish | one sec |
| 08:09.15 | horsefish | http://pastebin.com/GZaYAedF |
| 08:11.22 | kaldemar | the retry parameters are for the channel, which is dialed first. when SIP/203 answers, it is connected to salientes,200,1. that is where you need to dial your other party and do the retries in case of a failure. |
| 08:14.24 | horsefish | what do i append to dial to 200 and maintain the connection/retry after 203 answers? |
| 08:17.29 | kaldemar | horsefish: what does your extension look like now? |
| 08:17.50 | horsefish | still 200 |
| 08:18.21 | horsefish | im using 203 and the other party 200 |
| 08:18.36 | kaldemar | by extension i mean an extension in your dialplan. the "exten => 200,1,..." you have under [salientes] in your extensions.conf. |
| 08:18.48 | horsefish | so this executes and i answer the call here at 203, then it auto says 'goodbye' and doesn't connect to 200 |
| 08:19.03 | horsefish | one sec |
| 08:20.42 | horsefish | actually i thought salientes was an 'extension' of asterisk already since i followed a tut, there is no salientes in the conf file |
| 08:20.49 | horsefish | lol |
| 08:21.05 | kaldemar | [salientes] is a context. 200 is the extension, hence then "exten =>". |
| 08:23.38 | horsefish | so if i create a context 'outgoing' how should the exten => 200,1,... be configured properly to connect to 200? |
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| 08:26.46 | kaldemar | Dial(SIP/200) would be the simplest. |
| 08:26.55 | kaldemar | ~book |
| 08:26.55 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 08:26.58 | *** join/#asterisk k610 (~K610@cred.epid.ucl.ac.be) |
| 08:27.14 | kaldemar | you'll find some nice documentation and examples on dialplan in that book. |
| 08:27.32 | horsefish | thanks |
| 08:28.01 | horsefish | as far as Dial(SIP/200) goes, does that append to Extension: in the .call file? |
| 08:28.16 | kaldemar | it has nothing to do with the call file. |
| 08:28.56 | kaldemar | the call file dials the Channel, and upon answer connects the channel to your dialplan. after that, the call file is out of the equation. |
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| 08:29.36 | horsefish | i meant the context, my bad |
| 08:29.40 | horsefish | Context:** |
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| 08:30.00 | horsefish | so in this condition what would be ideal for the context: param? |
| 08:30.30 | horsefish | for the .call file |
| 08:31.14 | ChannelZ | Whatever context contains the extension you're wanting the call to execute |
| 08:31.54 | kaldemar | what is ideal is completely up to you. you're the one who constructs your dialplan. |
| 08:32.20 | ChannelZ | Unless you're just using the Application: field in which case you don't need a context |
| 08:33.14 | horsefish | so for a context that constantly redials the receiving party when one originates a call, this needs to be constructed as part of a dialplan |
| 08:34.02 | horsefish | when the ONLY OTHER*** party originates THE call |
| 08:34.09 | kaldemar | yes. but the part of dialplan that does the dialing is an extension, not a context. |
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| 08:36.11 | horsefish | ok ill be checking out that book then, its getting pretty complicated |
| 08:36.16 | ChannelZ | Extensions live in contexts. Contexts are like separate boxes. You can have extension 200 in 5 different contexts that all do completely different things, so you need to tell Asterisk which context the extension you're wanting to actually use is in. |
| 08:37.22 | horsefish | ok so a context for this case would ideally be - dial and redial until answered ? |
| 08:37.36 | ChannelZ | I'm not sure what you're talking about to be honest |
| 08:37.43 | horsefish | lol |
| 08:38.02 | ChannelZ | Or rather how what you're asking relates to call files or not |
| 08:38.27 | ChannelZ | I didn't scroll back and read this *entire* conversation so I'm unclear what you're trying to do. |
| 08:39.34 | horsefish | 100 & 101 are SIP extensions connected to SIP physical phones. 101 picks up the phone to undoubtedly speak with 100 (and vice versa). using a .call file that would be placed on the server every startup, these calls need to ring and autoconnect when both phones are answered |
| 08:40.51 | kaldemar | horsefish: a context does not do anything. extensions do. a context is just a container for extensions. |
| 08:41.39 | ChannelZ | So every time your server starts you want to create a call between 100 and 101? Not sure I get the application.. |
| 08:41.40 | kaldemar | horsefish: in the future, don't call those SIP extensions when dealing with asterisk. use the term peer for example. |
| 08:42.13 | horsefish | thanks for the tip. |
| 08:42.55 | kaldemar | just to avoid confusion because an extension means the specific thing in the dialplan. |
| 08:43.20 | horsefish | ChannelZ: sorry for the misunderstanding, im working with a guy who knows very little about asterisk and im helping him since i have more experience with systems. pbx is definitely a new topic for me |
| 08:44.30 | ChannelZ | In any event, lets say you have a couple of extensions defined: exten => 100,1,Dial(SIP/who100is) and exten => 101,1,Dial(who101is) that live in the context [internal] |
| 08:44.47 | horsefish | so im about to query what the dialplan command does.. |
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| 08:44.54 | ChannelZ | The peers 'who100is' and 'who101is' are defined in sip.conf and have the same context, internal |
| 08:45.23 | ChannelZ | Which means whenever either of those devices tries to place a call, Asterisk will try to execute whatever extension they dialed in the 'internal' context |
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| 08:46.15 | horsefish | ok |
| 08:46.16 | ChannelZ | Contexts are forms of routing if you will, or a means to restrict access to your dialplan.. so for instance I have an "out_local" context which has extensions that allow people to dial local numbers. I have a separate one called "out_ld" that have extensions allowing long distance. |
| 08:47.04 | ChannelZ | A public phone in the lobby might be set to the out_local context, but a phone on the boss's desk is set to out_ld (which can also include out_local) |
| 08:47.10 | horsefish | how do you route calls out to landlines? that's pretty cool |
| 08:47.31 | ChannelZ | Different question. |
| 08:47.33 | horsefish | ok |
| 08:48.00 | horsefish | yes following |
| 08:48.08 | ChannelZ | Well that was it :) |
| 08:48.52 | ChannelZ | In the case of a call file, what you're saying to Asterisk is "Call this channel, and if they answer, connext them to extension XXX in the dialplan" |
| 08:49.11 | horsefish | could i possibly configure everything through the web interface without reading through the conf files? |
| 08:49.23 | horsefish | yes! |
| 08:49.38 | horsefish | and keep dialing until they answer |
| 08:49.58 | ChannelZ | So if you wanted to put a call file in the spool that made 101's phone ring, and if they answered have it call 100, then you'd specify Channel: SIP/101 Context: internal Extension: 100 Priority: 1 |
| 08:50.04 | ChannelZ | given my example above |
| 08:50.24 | ChannelZ | for it to keep calling you'll have to also set retries and such |
| 08:51.03 | horsefish | dont the retrys apply to the channel and not the extension (which is the 'destination/target' as i understand)? |
| 08:51.15 | ChannelZ | yes |
| 08:52.27 | horsefish | Ok, so how could I make the retrys apply to the extension? |
| 08:52.55 | ChannelZ | make the Dial() never time out if you just want it to ring indefinately |
| 08:53.51 | kaldemar | there's also RetryDial. "core show application RetryDial" |
| 08:54.41 | ChannelZ | personally I would throw my phone out the window onto the street if anyone ever did that.. |
| 08:54.56 | horsefish | lol |
| 08:55.28 | ChannelZ | If you wanted to get swanky with it you could get into call queues |
| 08:55.35 | horsefish | where is the Dial() edited? |
| 08:56.18 | ChannelZ | extensions.conf. But it sounds like you really need to read The Book and get familiar with Asterisk as a whole, this is a bit micro for your experience |
| 08:56.58 | horsefish | you're very right |
| 08:57.21 | horsefish | gonna give youre tips a try |
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| 08:58.04 | ChannelZ | I've been working on an Asterisk Primer that gives you an accellerated crash-course from start to finish to get people started, but unfortunately I haven't had the time to finish it and put it up. Maybe in Jan. |
| 08:59.18 | horsefish | woo nice! |
| 08:59.58 | horsefish | would be a great gift for my friend |
| 09:02.09 | ChannelZ | The book is a great resource but covers a lot of stuff a lot of people don't care about, I wanted to make something that would make the whole thing more approachable and get someone starting from scratch to understand more clearly how the basic pieces fit together. |
| 09:03.53 | horsefish | nice if you get around to that then pm me ! |
| 09:04.11 | ChannelZ | I think newbies get totally overwhelmed when they download and see the dozens of config files with 100s of lines in them |
| 09:04.21 | horsefish | are you using the web interface as the main means of configuration or more through the files? |
| 09:04.56 | ChannelZ | So they either go searching for tutorials on specific things to try to steal config fragments from, but if you don't understand what you're even looking at you just wind up in a hole. Then they throw up their hands and go searching for a GUI, which to me is even worse. |
| 09:05.08 | ChannelZ | No, no GUIs here |
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| 09:06.35 | horsefish | when worse comes to worse |
| 09:06.42 | horsefish | GUI takes over |
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| 09:07.27 | horsefish | so come to think of my situation, SIP phones could probably be preprogrammed to dial a specific number when theyre unhooked |
| 09:07.50 | horsefish | and the receiving party could be programmed to autoreceive calls (my friend confirmed this) |
| 09:07.54 | ChannelZ | I guess it would help to know exactly what the application is |
| 09:08.37 | ChannelZ | I mean beyond what you've said.. what is the actual purpose |
| 09:09.21 | *** part/#asterisk deo (~deo@203.177.9.66) |
| 09:09.38 | horsefish | a radio station has a point-to-point tunnel across 2 towers about 5 miles apart from each other. they asked us to install an always-available comms system whereby one picks up the phone and it is 'auto-connecting' to the other party no hard work on their part (punching in 101 or 102) |
| 09:09.46 | ChannelZ | (and yes some/most devices you can have auto-answer - and some you can send additional SIP headers to do this "on demand" from the dialplan..) |
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| 09:10.18 | horsefish | so using an elastix server and 2 sip phones we thought it'd do the trick, since on both ends of the tunnel its local ips of the same subnet |
| 09:10.31 | ChannelZ | ok that makes more sense. |
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| 09:10.35 | horsefish | yeah, thats a major advantage of using physical phones! |
| 09:10.36 | ChannelZ | Forget call files completely |
| 09:10.53 | horsefish | haha awesome :p |
| 09:11.44 | ChannelZ | First I'm going to call your 2 devices [tower-a] and [tower-b] as configured in sip.conf |
| 09:12.15 | horsefish | yeah |
| 09:12.34 | ChannelZ | I'd put them in two different contexts, context-a and context-b respectively |
| 09:14.09 | ChannelZ | ..sec... |
| 09:14.23 | horsefish | ok |
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| 09:15.35 | ChannelZ | If you wanted it to be so that if you picked up either phone's handset (or hit the speakerphone button for instance) and it auto-dialed the other side without them having to dial *anything*, you'll need to do some programming on the phones to get them to dial *something* automatically |
| 09:16.25 | horsefish | correct |
| 09:16.46 | ChannelZ | Beyond the scope of Asterisk though. But let's say you get both of them to just dial a 1 when you pick them up. |
| 09:17.20 | ChannelZ | In extensions.conf, make a context [context-a] and put in exten => 1,1,Dial(SIP/tower-b) |
| 09:17.44 | ChannelZ | Make another context [context-b] and put in exten => 1,1,Dial(SIP/tower-a) |
| 09:17.56 | ChannelZ | That's literally it. |
| 09:18.26 | horsefish | and configure the SIP phones to identify as the respective [tower-a] and [tower-b] |
| 09:18.39 | ChannelZ | When tower-a picks up their phone and either the phone dials a 1 or they hit it, it executes extension 1 in context-a (because that's the context set for the tower-a device in sip.conf) |
| 09:18.56 | ChannelZ | And vice-versa for tower-b. it dials 1 in context-b |
| 09:19.04 | horsefish | nice. |
| 09:19.20 | ChannelZ | Yeah. I don't like to call devices extensions because they aren't the same thing. |
| 09:19.59 | ChannelZ | A device is a device. An extension is a virtual thing that exists in asterisk's mind, and has NO relation to any device. |
| 09:20.15 | horsefish | well extensions are arbitrary user-names if you will TMU and devices can be programmed to assume any extension not so? |
| 09:20.25 | horsefish | yeah |
| 09:20.25 | ChannelZ | no |
| 09:20.48 | ChannelZ | An extension is just some digits (or not just digits) that can be dialed, and that when dialed, make Asterisk do something. |
| 09:21.03 | horsefish | ahh |
| 09:21.07 | ChannelZ | If that something happens to be to dial another device, so be it. But they can obviously do lots of other things. |
| 09:22.28 | ChannelZ | But a device is still a device, it only has an associated "extension" if you write that down on a piece of paper and program your dialplan to call that device when someone dials that number. |
| 09:22.30 | horsefish | sort of like a 9 for an outside tone? e.g. hotels? |
| 09:23.18 | ChannelZ | We call those 'steering digits' but it along with the rest of the digits they dial is an extension. |
| 09:23.56 | ChannelZ | Which can be configured to place an outside call... or it could play screaming monkey sounds and then hang up. |
| 09:24.34 | horsefish | one sec.. |
| 09:25.38 | horsefish | the extensions file says not to edit it and instead edit through the web gui? |
| 09:26.43 | horsefish | maybe we'll program 2 to play something of the sort, just for kicks |
| 09:27.37 | ChannelZ | well if you're using Elastix you are a slave to the GUI and I can't help you there |
| 09:28.43 | horsefish | lol ill reboot see what happens |
| 09:29.16 | ChannelZ | If all your system is going to do is not much more than what you've described, Elastix seems like massive overkill |
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| 09:30.30 | horsefish | we were thinking of getting a cubox (http://solid-run.com) for the task, simple debian and slap freepbx on it |
| 09:31.11 | ChannelZ | In reality, you could probably do this without Asterisk at ALL if you could configure the phones to automatically dial by IP address |
| 09:31.12 | horsefish | what say you |
| 09:31.35 | horsefish | damn |
| 09:31.52 | ChannelZ | Do you already have the devices? |
| 09:32.17 | ChannelZ | (and I wouldn't run FreePBX on that little thing you pated) |
| 09:33.11 | ChannelZ | by devices I mean phones |
| 09:33.31 | horsefish | no not yet |
| 09:34.16 | horsefish | but that definitely takes the weight off of configuring a stable system not meant to be touched by the client |
| 09:34.36 | horsefish | pated? |
| 09:34.42 | ChannelZ | pasted, sorry |
| 09:34.48 | ChannelZ | that cubox thing |
| 09:34.54 | horsefish | lol |
| 09:35.22 | horsefish | still want that thing for me :) |
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| 09:36.15 | ChannelZ | I don't have an actual SIP phone here to try but I'd think you'd probably be able to trick most devices into just dialing another one without having to use a SIP proxy at all. |
| 09:36.33 | horsefish | call me a treehugger but there is a warm feeling in my heart when im running servers 100% and consuming 3, tops 4W. |
| 09:37.33 | horsefish | yeah, assuming a nice modern one has RJ45 and can be assigned a static ip |
| 09:37.42 | ChannelZ | yeah they all can |
| 09:37.59 | horsefish | an agenda or contacts list can assign a speed dial 1 to each phone for each other |
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| 09:38.21 | horsefish | problem solved |
| 09:38.25 | horsefish | cash the check |
| 09:38.52 | ChannelZ | Well in that case the actual number dialed (the extension) is irrelavent, it just has to send the call to the right IP. |
| 09:40.23 | ChannelZ | I'll have to try this at work tomorrow for fun |
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| 09:40.41 | horsefish | righthttp://www.umpc.net/store/?fun=hone |
| 09:40.43 | ChannelZ | In the meantime I should have been in bed an hour ago.. |
| 09:40.45 | horsefish | http://www.umpc.net/store/?fun=hone *** |
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| 09:42.28 | horsefish | good, you've been a real help |
| 09:42.50 | horsefish | thanks a mil |
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| 09:46.47 | ChannelZ | sure, nighty |
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| 12:41.59 | *** join/#asterisk wilfredor (c85a2731@wikipedia/wilfredor) |
| 12:42.02 | wilfredor | Hi |
| 12:43.20 | wilfredor | I need a comunication between fisical pbx´s (no ip) and mobiles phones, around 30,000 |
| 12:43.38 | wilfredor | And a Call accounting for asterisk |
| 12:43.48 | wilfredor | I'm not a expert in this topic |
| 12:43.59 | wilfredor | someone could help me?, thanks |
| 12:44.57 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.6.103) |
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| 13:41.39 | jeffspeff | i'm messing around with meetme, using it in realtime with mysql. when i set the user pin and admin pin, and specify option w, it doesn't mark the admin user as the actual marked user. how can i change that? if someone enters the admin pin instead of user pin, i want them to be the marked user. |
| 13:48.44 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 13:50.11 | saint_ | hi all |
| 13:51.32 | saint_ | I'm working out of a VM in iCloud where I do not have any logs on the fire wall.. even though I can setup rules.. When I receive calls to my Asterisk, I can hear the audio path if I use Playback() function, but if I use echo() function I can't hear myself. What ports do I need to open on this box for that to work ? I tried 5060 and 10000:20000 with no success. |
| 13:51.52 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 13:52.13 | WIMPy | UDP? |
| 13:52.23 | saint_ | 5060 udp / tcp |
| 13:52.27 | saint_ | and 10000:20000 udp |
| 13:52.33 | WIMPy | ok |
| 13:52.54 | WIMPy | Is it NATting? |
| 13:52.58 | saint_ | if I set the default behaviour on the box to accept all incoming connection, then it s working |
| 13:53.01 | saint_ | no, public IP |
| 13:53.11 | saint_ | but it's a pain because I can't see what's being used |
| 13:53.23 | *** join/#asterisk CunningPike (~CunningPi@d28-23-24-84.dim.wideopenwest.com) |
| 13:54.29 | [TK]D-Fender | saint_, Show us the call |
| 13:55.10 | *** join/#asterisk serafie (~erin@nat/digium/x-clxkdfxwzriwxpan) |
| 13:55.27 | saint_ | ok,s tand by |
| 13:56.35 | WIMPy | Are the Digium guys aredy off for the holidays? |
| 13:59.07 | saint_ | [TK]D-Fender: / WIMPy : never mind it s working now... |
| 14:00.18 | *** join/#asterisk SupaYoshi (SupaYoshi@ip4da5d319.direct-adsl.nl) |
| 14:00.27 | SupaYoshi | Anyone experience setting up mp3s to be played in asterisk? |
| 14:00.39 | SupaYoshi | I am reading guides everywhere. Everyone is doing something different! |
| 14:00.42 | SupaYoshi | http://www.data4ict.com/tutorials/asterisklinux/music.asp |
| 14:00.43 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:00.43 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:03.02 | [TK]D-Fender | SupaYoshi, Well that guide shows you 2 ways.... |
| 14:03.19 | [TK]D-Fender | 3 depending on your point of view |
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| 14:03.23 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:04.04 | [TK]D-Fender | SupaYoshi, One of which does not actually imply "MP3" is being used. |
| 14:04.26 | [TK]D-Fender | SupaYoshi, And also only shows setting up MoH, not general usage. |
| 14:04.41 | [TK]D-Fender | SupaYoshi, Perhaps you should clarify your goal |
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| 14:08.52 | saint_ | is there a limitation in the password I can assign a client in the sip.conf file ..? Like if I want :68%<76]&9h:9r are all the characters authorized ? Or am I limited to [0-9][a-z][A-Z] ..? |
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| 14:29.56 | jeffspeff | i'm messing around with meetme, using it in realtime with mysql. when i set the user pin and admin pin, and specify option w, it doesn't mark the admin user as the actual marked user. how can i change that? if someone enters the admin pin instead of user pin, i want them to be the marked user. |
| 14:30.53 | *** join/#asterisk _Corey_ (~chatzilla@pool-72-78-178-187.phlapa.fios.verizon.net) |
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| 14:35.26 | RZero | Hi all, I am using an FXO card and trying to get AGI to detect this "DAHDI/1-1 answered SIP/1111-00000013" as this tells me the call has connected but because its an FXO Im not sure what command to use, Ive tired DIALSTATUS and Channel Status |
| 14:35.44 | RZero | neither work |
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| 14:36.52 | WIMPy | When/where exactely do you want to know the status? |
| 14:38.01 | SupaYoshi | well... |
| 14:38.05 | SupaYoshi | I been trying the guide... |
| 14:38.09 | SupaYoshi | I been trying everything |
| 14:38.09 | RZero | After I call the number, I dont care if the other side picks up or not |
| 14:38.17 | SupaYoshi | I cannot set my asterisk to my mp3 |
| 14:38.22 | SupaYoshi | It converts it to a wav. |
| 14:38.24 | SupaYoshi | I've tried everything |
| 14:38.33 | SupaYoshi | Is there anyone that has teamviwer and likes to set it up for 15 dollar? |
| 14:39.04 | RZero | I just testing numbers to make sure they connect |
| 14:40.14 | WIMPy | SupaYoshi: What does "it converts it to wav" mean? And why do you want to play mp3 directly? wav is a good base for whatever COED might be in use. And it's safe on the legal side. |
| 14:40.22 | [TK]D-Fender | RZero, AGI doesn't "detect" anything. |
| 14:40.28 | WIMPy | s/COED/CODEC/ |
| 14:41.05 | [TK]D-Fender | RZero, that line you pasted shows a call you Dial()-ed having been answered. |
| 14:41.47 | RZero | not like ISDN/SIP connection, it just means the call has connected |
| 14:42.06 | RZero | Ie you here the ringing |
| 14:42.12 | RZero | hear * |
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| 14:54.37 | WIMPy | That depends on your line and your configuration. |
| 14:55.02 | WIMPy | But doing it on a POTS line is not the best idea. |
| 14:55.04 | Katty | shivers. |
| 14:55.34 | Katty | lots of birds out this morning :> |
| 14:55.49 | RZero | I know but I have to test our number ranges from different networks |
| 14:55.59 | [TK]D-Fender | RZero, AGI is not a background process. There is no "detection". It is linear jsut like the rest of the dialplan. |
| 14:56.06 | *** join/#asterisk blee (~blee@68.204.217.123) |
| 14:56.20 | [TK]D-Fender | RZero, Your direction is skewed |
| 14:57.10 | _Corey_ | Katty: apocalyptic birds? |
| 14:57.13 | RZero | how so ? |
| 14:57.24 | Katty | _Corey_: mmmm, no they look house finches. |
| 14:57.36 | Katty | infobot: crittercam |
| 14:57.36 | infobot | [crittercam] The Birdie Breakfast Buffet! http://www.ustream.tv/channel/birdie-breakfast-buffet |
| 14:57.39 | Katty | _Corey_: ^- |
| 14:57.45 | _Corey_ | best have a baseball bat handy.... just in case |
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| 14:58.08 | RZero | all I need to know is that the number rings across 3 different PSTN providers |
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| 15:55.10 | Hive | Is it possible for one sip peer to have access to another peer's variables? like PeerList[<peerName>][<variable>] or something? Google is having a hard time understanding my question :[ |
| 15:56.25 | SeRi | Penguin: vuc is @11AM Central?or 10AM? |
| 16:05.32 | *** join/#asterisk elguero_ (04353e8a@gateway/web/freenode/ip.4.53.62.138) |
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| 16:22.51 | [TK]D-Fender | Hive, Peers don't have variables, channels do. |
| 16:23.12 | Penguin | seri: 1100 CST |
| 16:24.11 | WIMPy | [TK]D-Fender: What do you call the stuff you get from the SIPPEER function? |
| 16:24.27 | [TK]D-Fender | WIMPy, those are peer parameters. |
| 16:25.03 | [TK]D-Fender | WIMPy, variable = channel variable. |
| 16:25.40 | WIMPy | They can be set from a peer as well. |
| 16:25.57 | [TK]D-Fender | WIMPy, Now if he wants to grab an actual peer parameter (like username, etc), then yes the function for this should do its job. |
| 16:26.33 | [TK]D-Fender | WIMPy, There is a difference between a default value set when a call is placed by a peer and a "parameter" |
| 16:27.13 | [TK]D-Fender | WIMPy, And that of a "channel variable" set by a call actually in progress. Right no (as usual), the description is not slanted that way and no clarification seems to be on its way. |
| 16:27.14 | WIMPy | A parameter that becomes a variable? |
| 16:28.35 | [TK]D-Fender | WIMPy, "SetVar=var=" may set "var" as a channel variable when a call is placed by it. however that var could change in processing the call. Which is he asking for? Is he really even asking about channel variables? Did he perhaps mean he wants the ""username" "host", etc from the peer? We don't know and right now it's vague |
| 16:29.04 | WIMPy | Correct. |
| 16:29.44 | [TK]D-Fender | WIMPy, I'm not going to waste more time being asked to be psychic. |
| 16:30.15 | *** join/#asterisk dxd828 (~dxd828@195.191.107.205) |
| 16:30.22 | [TK]D-Fender | Hive, please rephrase precisely what it it you're looking to do so we can make sure you get the right answer. |
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| 16:56.47 | Hive | [TK]D-Fender; I have listings in sip.conf that look similar to this: http://pastebin.com/wMTe21M8 . If 101 opens a channel, is there any way for it to check what value [102] has for the variable "recordAudio" |
| 16:57.30 | WIMPy | Hive: core show function SIPPEER |
| 16:58.12 | Hive | thank you thank you, exactly what i need |
| 16:58.46 | [TK]D-Fender | Hive, Might work, not sure if it allows you to inquire on SetVar's |
| 16:59.27 | WIMPy | That's what it says if you enter that. |
| 16:59.47 | Hive | chanvar[name] - A channel variable configured with setvar for this |
| 16:59.49 | Hive | <PROTECTED> |
| 17:00.00 | Hive | ill test it and let you know D-Fender |
| 17:00.12 | *** join/#asterisk shadar (~eugene@37.113.133.194) |
| 17:05.48 | Katty | i feel like i'm missing out on all the Hot Lude Action |
| 17:07.46 | WIMPy | I'm missing out on my presents. Can't afford the taxes. |
| 17:08.01 | [TK]D-Fender | Katty, boogaloo dudes! |
| 17:10.50 | Hive | That works, thanks again WIMPy |
| 17:11.18 | *** join/#asterisk dxd828 (~dxd828@195.191.107.205) |
| 17:11.21 | *** join/#asterisk lorsungcu (~anonymous@24-196-56-142.static.stcd.mn.charter.com) |
| 17:11.56 | [TK]D-Fender | Hive, Good to know.... generally the variable bits might be the thing to break if anything.... |
| 17:12.17 | [TK]D-Fender | Hive, Perhaps you could test setting 2 vars and checking each for us while you're there? |
| 17:12.32 | [TK]D-Fender | Hive, Just to see if you can grab both, or only the first/last |
| 17:16.11 | *** join/#asterisk CunningPike (~CunningPi@198.228.232.119) |
| 17:16.27 | Hive | [TK]D-Fender: sure ill mess around with it gimme a min |
| 17:20.27 | Hive | [TK]D-Fender, seems to work without issue. I set 3 different variables in the peer listing in sip.conf with servar=VARNAME=something. I NoOp'd the values out of order of their listing and everything came up correct |
| 17:21.02 | [TK]D-Fender | Hive, Thanks. We'll have that info to pass to the next person asking about such things... |
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| 18:18.48 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 18:19.06 | saint_ | re - |
| 18:19.38 | saint_ | if I have 2 sip client that call each other through my asterisk.. does the video feature depend on the clients themselves, or does asterisk has to do anything with it ? |
| 18:21.15 | [TK]D-Fender | saint_, Both |
| 18:21.33 | [TK]D-Fender | saint_, If the client doesn't support video, Asterisk doesn't magically make it support video. |
| 18:21.40 | Katty | hello my asterisk does not work at all how to fix plz is urgent thx. |
| 18:21.59 | [TK]D-Fender | saint_, * has to accept the offer of video from the caller and has to offer it to the callee. All parties must accept |
| 18:22.21 | saint_ | [TK]D-Fender: i bought the video option for Bria (sip client for ipad/iphone from counterpath) for both my ipad and my iphone, but i can t get it to work. so my clients have the possibility. |
| 18:23.16 | _Corey_ | Katty: If I understood Mel Gibson's movie correctly, I think human sacrifices are required today |
| 18:23.23 | Katty | oh? |
| 18:23.48 | [TK]D-Fender | saitthen look at your configs and SIP debug of the calls |
| 18:23.55 | _Corey_ | yeah, that |
| 18:23.58 | _Corey_ | er |
| 18:24.03 | _Corey_ | that'll fix anything |
| 18:24.03 | Katty | i smell a plot most foul. |
| 18:24.18 | Katty | i guess that could also be the trash. |
| 18:24.24 | [TK]D-Fender | Katty, I did my contribution on stage last night airing off the full opening for "Final Countdown" on the keys.... |
| 18:24.37 | _Corey_ | We've watched too many stupid apocalypse-themed movies recently |
| 18:24.41 | Katty | was it appreciated? |
| 18:24.55 | [TK]D-Fender | Katty, my contributions always are... |
| 18:24.57 | _Corey_ | [TK]D-Fender: nice |
| 18:25.15 | Katty | egggggggcelent. |
| 18:25.32 | [TK]D-Fender | egggzachary |
| 18:28.24 | Katty | stitches up the first of the kitty toys. |
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| 18:35.30 | rue_mohr | [TK]D-Fender, hello |
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| 19:38.02 | Katty | i miss jaytee |
| 19:39.51 | talntidd | where'd he go? |
| 19:47.01 | *** part/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 19:47.53 | Katty | [TK]D-Fender: how about a gsm file instead |
| 19:48.27 | *** join/#asterisk parasitodelsur (~wtf@23.30.88.89) |
| 19:54.53 | Katty | [TK]D-Fender: disregard >.< wrong andrew |
| 20:03.59 | lorsungcu | any reason DTMF wouldn't be received by the voicemail() application when using SIP INFO instead of rfc2833 |
| 20:05.11 | [TK]D-Fender | lorsungcu, Application has no impact on DTMF. The mode is either right, or it isn't |
| 20:05.22 | [TK]D-Fender | lorsungcu, Check the peer the call is actually coming in under |
| 20:05.25 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 20:05.59 | lorsungcu | its a phone |
| 20:06.12 | lorsungcu | issue is that with these phones, when set to use rfc2833, they also send audio inband |
| 20:06.17 | lorsungcu | and it fucks with the trunks they have |
| 20:07.22 | [TK]D-Fender | I said check the peer. as in peer entry in sip.conf. |
| 20:07.29 | lorsungcu | yeah, its set right |
| 20:07.38 | [TK]D-Fender | And while you're at it, why NOT the phone. |
| 20:07.42 | [TK]D-Fender | Prove that * is matching the right entry and that it's configured to match |
| 20:07.52 | lorsungcu | alright |
| 20:08.02 | [TK]D-Fender | If it doesn't work then SOMETHING is lying. |
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| 20:44.39 | Katty | weeeeeeeeeeeeeeeeeeeeeeeeeee |
| 20:53.56 | [TK]D-Fender | ~whee |
| 20:53.56 | infobot | [~whee] Weeeeeeeee! http://www.albinoblacksheep.com/flash/weeee |
| 20:54.10 | *** join/#asterisk nicknam12322 (021d14df@gateway/web/freenode/ip.2.29.20.223) |
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| 21:00.39 | *** join/#asterisk polysics (~di@host226-75-dynamic.244-95-r.retail.telecomitalia.it) |
| 21:06.16 | *** join/#asterisk polysics_ (~di@host54-66-dynamic.245-95-r.retail.telecomitalia.it) |
| 21:08.40 | *** join/#asterisk josefig (~josef@unaffiliated/josefig) |
| 21:17.49 | *** join/#asterisk DelphiWorld (~AlfajrFM@openvpn/user/DelphiWorld) |
| 21:17.52 | DelphiWorld | Hi everyone |
| 21:18.02 | DelphiWorld | can someone give me SPA941 firmware please ? |
| 21:19.43 | *** join/#asterisk sonstwo (~garland@unaffiliated/ffs) |
| 21:21.36 | Katty | hmm. |
| 21:21.41 | Katty | 5 cats toys enough? |
| 21:21.43 | Katty | or maybe 6 |
| 21:22.35 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-poqdljmmxuslexyc) |
| 21:23.16 | DelphiWorld | slaps Katty around a bit with a large trout |
| 21:24.35 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 21:29.41 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.9.138) |
| 21:30.29 | DelphiWorld | i got fw... |
| 21:31.56 | *** join/#asterisk zopsi (1818afbd@gateway/web/freenode/ip.24.24.175.189) |
| 21:33.02 | zopsi | How would I go about debugging DPMA connection from digium d70? My phones just time out contacting the configuration server, but they are able to see it. |
| 21:38.58 | Katty | with chocolate. |
| 21:39.30 | SupaYoshi | ive some trouble with a ringgroup here |
| 21:39.44 | zopsi | chocolate heh |
| 21:39.59 | Katty | you can use espresso to fix ring groups. |
| 21:40.20 | zopsi | but actually I need a legitimate answer as none of our phones work right now |
| 21:44.28 | *** join/#asterisk dxd828 (~dxd828@host86-165-22-173.range86-165.btcentralplus.com) |
| 21:45.49 | *** join/#asterisk tonyclewis (uid6025@gateway/web/irccloud.com/x-ubgarvymtmnsypvl) |
| 21:47.40 | SupaYoshi | espresso? |
| 21:48.03 | SupaYoshi | ive trouble that it wont ring my phones, and just hangs up :) because it says they are unavaiible in asterisk -r verbose 10 |
| 21:54.30 | zopsi | what is this referring to WARNING[9888]: xmldoc.c:2037 xmldoc_build_field: Couldn't find manager DPMAMessageResponse in XML documentation? |
| 21:57.00 | blitzrage | zopsi: looks like some missing documentation |
| 21:57.16 | zopsi | yeah so not a big deal |
| 21:57.40 | zopsi | @blitzrage: could that be causing issues of DPMA registrations timing out? |
| 21:57.48 | blitzrage | unknown, I don't use DPMA |
| 21:58.09 | zopsi | @blitzrage: gotcha |
| 21:58.26 | *** join/#asterisk CunningPike (~CunningPi@d28-23-24-84.dim.wideopenwest.com) |
| 22:03.15 | parasitodelsur | Penguin: have you been having issues with voip.ms? |
| 22:03.34 | blitzrage | buddy of mine has |
| 22:03.43 | blitzrage | keeps having to change servers over the last 2-3 weeks |
| 22:03.51 | blitzrage | not getting calls, unable to place calls reliably, etc |
| 22:04.06 | parasitodelsur | blitzrage: in what area? |
| 22:04.11 | blitzrage | Sarnia |
| 22:04.14 | blitzrage | Ontario |
| 22:04.22 | parasitodelsur | ok. That explains it. |
| 22:04.32 | parasitodelsur | There is a shit load of issues coming out of that area |
| 22:04.32 | blitzrage | define: it |
| 22:04.37 | blitzrage | ahhh |
| 22:04.42 | blitzrage | guess who isn't surprised |
| 22:04.43 | blitzrage | <--- this guy |
| 22:04.46 | parasitodelsur | The DC puked |
| 22:04.52 | parasitodelsur | lol |
| 22:04.54 | blitzrage | that whole area is a black hole of technical failure |
| 22:05.03 | parasitodelsur | o wow |
| 22:05.13 | blitzrage | Sarnia still uses Lotus 1-2-3 |
| 22:05.22 | parasitodelsur | There is a bad rash at dslr of people complaining about issues.... |
| 22:05.23 | blitzrage | I wouldn't be shocked if several places use Wordperfect too |
| 22:05.31 | blitzrage | dslr? |
| 22:05.38 | parasitodelsur | dslreports.com |
| 22:05.41 | blitzrage | ahh right |
| 22:05.54 | parasitodelsur | the majority are from Canada |
| 22:06.03 | blitzrage | I don't use that sooooo I always think, "digital single lens reflex" |
| 22:06.03 | parasitodelsur | who ever invented Lotus Notes needs to die |
| 22:06.13 | blitzrage | probably already did |
| 22:06.19 | parasitodelsur | good. |
| 22:06.21 | parasitodelsur | lol |
| 22:06.32 | blitzrage | well I'll let my buddy know he's not the only one :) |
| 22:06.40 | parasitodelsur | yeap. Good Luck |
| 22:06.44 | blitzrage | thanks? :) |
| 22:06.49 | parasitodelsur | no issues at the Houston/Dallas pop |
| 22:06.57 | parasitodelsur | blitzrage: lol |
| 22:07.05 | blitzrage | he has been trying to use Atlanta, Toronto, New York, etc |
| 22:07.08 | blitzrage | keeps moving around |
| 22:07.51 | parasitodelsur | Thats the reason I come over here to ask Penguin because Houston and Dallas have been solid |
| 22:08.05 | parasitodelsur | but other report that it has issues as well |
| 22:08.12 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:08.17 | parasitodelsur | I cant belive it because I dont have the issue |
| 22:08.35 | parasitodelsur | Than again I am about 30 Miles from the POP |
| 22:08.43 | parasitodelsur | :) |
| 22:08.55 | blitzrage | neat |
| 22:09.56 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.38) |
| 22:11.29 | SupaYoshi | does anyone know what to do about a softphone in a ringgroup? it causes my phones not to ring |
| 22:11.30 | SupaYoshi | <PROTECTED> |
| 22:11.37 | SupaYoshi | i get this error, when i ring my ringgroup :) |
| 22:11.49 | blitzrage | don't use ring groups I guess |
| 22:12.03 | blitzrage | just use Dial() and multiple end points separated with & |
| 22:12.31 | SupaYoshi | http://pastebin.com/raw.php?i=JQjY6Amu |
| 22:12.45 | SupaYoshi | Im using freepbx, how would i do that? |
| 22:13.02 | blitzrage | sorry, never used freepbx, just using vanilla asterisk |
| 22:14.12 | zopsi | so anyone used DPMA before or should I just try and get in touch with Digium support? |
| 22:16.19 | *** join/#asterisk SeRi (~wtf@pdpc/supporter/professional/seri) |
| 22:16.35 | SeRi | there we go. much better |
| 22:29.36 | Katty | i'm outta here kiddos. |
| 22:29.49 | Katty | merry christmas and happy hanukkah and blessed yule... |
| 22:29.52 | Katty | and all that jazz. |
| 22:41.24 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 22:46.11 | *** join/#asterisk navaismo (~Administr@189.191.20.105) |
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| 23:05.22 | SupaYoshi | anyone an idea what this means? |
| 23:05.23 | SupaYoshi | app_dial.c:2274 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 23:06.34 | [TK]D-Fender | Device isn't registered |
| 23:06.40 | SupaYoshi | ah weird. |
| 23:06.47 | SupaYoshi | Because one time it works, and the other time it doesn't |
| 23:07.02 | SupaYoshi | I've 2 devices on that ringroup, one is offline, and the other one is registred. |
| 23:07.35 | SupaYoshi | though the ring strategy is ring-prim |
| 23:07.42 | SupaYoshi | that worked just 10 minutes ago fine :) |
| 23:08.44 | SupaYoshi | think something is wrong in my sip device |
| 23:11.07 | SupaYoshi | think i got it, sipura was on Ans Call Without Reg: No |
| 23:11.12 | SupaYoshi | *yes i mean |
| 23:12.58 | SupaYoshi | mhm ok fixed ;) |
| 23:13.00 | SupaYoshi | pbx.c:4218 pbx_extension_helper: No application 'VoiceMail' for extension (macro-vm, s-BUSY, 3) |
| 23:13.06 | SupaYoshi | that means my voicemail isnt activated right? |
| 23:16.47 | [TK]D-Fender | no it means that you have messed up your core * configuration and the application module has failed to load |
| 23:16.48 | Penguin | "Activated" is a sales/marketing term. |
| 23:16.56 | Penguin | It means you didn't load app_voicemail.so. |
| 23:17.45 | [TK]D-Fender | SupaYoshi: It may be completely non-existant. It may fail due to serious misconfiguration. It may not load because of a modules.conf override or bad target. |
| 23:18.18 | Penguin | seri: I haven't personally had any trouble, but I haven't been making calls today while I was out. |
| 23:18.25 | [TK]D-Fender | perhaps * can't read the config files it's supposed to at all |
| 23:18.43 | SeRi | Penguin: ok. Thanks for the info. |
| 23:18.57 | SupaYoshi | i see. |
| 23:19.05 | SupaYoshi | well i just ugraded all modules |
| 23:19.09 | SupaYoshi | all i done |
| 23:19.26 | *** join/#asterisk dxd828 (~dxd828@host86-165-22-173.range86-165.btcentralplus.com) |
| 23:19.50 | SupaYoshi | + the voicemail webgui seems to work |
| 23:20.08 | zopsi | ugh I'm dropping digium and going for something else |
| 23:20.51 | [TK]D-Fender | SupaYoshi: FreePBX modules mean precisely nothing. ASTERISK is broken |
| 23:21.03 | Penguin | And... we don't support GUIs here, anyway. |
| 23:22.56 | SupaYoshi | how many seconds should a device register? |
| 23:23.44 | Penguin | What does that even mean? |
| 23:24.14 | SupaYoshi | Register Expires |
| 23:24.20 | SupaYoshi | i mean |
| 23:25.25 | Penguin | I usually set the devices' register expiration time to a lower value when there is more potential for short outages. So across the public internet, I might have 60 seconds, where I might have 300 seconds for a device on my LAN. |
| 23:26.12 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 23:27.00 | SupaYoshi | ah ok! |
| 23:27.12 | SupaYoshi | because, em i have a sipura |
| 23:27.22 | SupaYoshi | and every now and then it tells the asterisks that its busy |
| 23:27.25 | Penguin | That's a brand name. You have the entire brand? |
| 23:27.25 | SupaYoshi | but it isnt.. |
| 23:27.32 | SupaYoshi | supura 3000 :P |
| 23:27.38 | Penguin | Okay, so you have an ATA. |
| 23:27.42 | SupaYoshi | Yeah! |
| 23:27.54 | SupaYoshi | and i set it up works nicely, but the registration was on 120 seconds? |
| 23:27.58 | SupaYoshi | i put it on 30 now. |
| 23:28.02 | Penguin | Registered or not registered, busy means busy. |
| 23:28.07 | SupaYoshi | registrated |
| 23:28.12 | SupaYoshi | oh... |
| 23:28.19 | SupaYoshi | what can it be then? cus if i reboot my supura |
| 23:28.22 | SupaYoshi | prboblem is gone |
| 23:28.48 | Penguin | Perhaps your devices was losing registration and you have set it to not allow calls to be received if it is not registered. |
| 23:28.53 | Penguin | There's a setting for that. |
| 23:29.24 | Penguin | I would want to find out why it isn't staying registered, if that is the case. |
| 23:29.52 | Penguin | If it is due to network outages, reduce your expire time in the device so that it reregisters more often. |
| 23:30.13 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 23:30.15 | SupaYoshi | ok! |
| 23:30.25 | Penguin | But remember that the lower you set the expiry, the more traffic it will generate. |
| 23:30.25 | SupaYoshi | i havent had it so far after i changed it to 30 |
| 23:30.33 | SupaYoshi | i understand that yeah.. |
| 23:30.35 | SupaYoshi | http://pastebin.com/raw.php?i=bSmaQVTv |
| 23:30.41 | SupaYoshi | whats going on here? can you see? |
| 23:30.44 | SupaYoshi | this is my voicemail problem |
| 23:31.29 | Penguin | The line numbers are not shown in raw mode, so I can't ask you what line it's on. |
| 23:31.38 | SupaYoshi | http://pastebin.com/bSmaQVTv |
| 23:32.34 | Penguin | I see that line 60 shows the problem that we've already given you the answers to. |
| 23:32.36 | SupaYoshi | 60 |
| 23:32.50 | SupaYoshi | Oh i thought maybe in verbose 10 the answer would be different |
| 23:32.55 | SupaYoshi | sorry.. :P |
| 23:33.07 | SupaYoshi | mhm.. |
| 23:33.20 | Penguin | It won't get more verbose above 3. |
| 23:33.39 | SupaYoshi | ah i see what swrong maybe... |
| 23:33.44 | SupaYoshi | i made a guess. :P and it was that |
| 23:33.48 | Penguin | Verbose 4 adds the dnsmngr stuff, I think it's verbose 7 where CDR is added... |
| 23:34.37 | Penguin | So, until you load app_voicemail.so, you will not have the voicemail app. |
| 23:34.54 | SupaYoshi | :) loaded |
| 23:35.01 | SupaYoshi | sudo apt-get install asterisk-voicemail |
| 23:35.10 | SupaYoshi | -facepalm- it wasnt installed. gui said it was. |
| 23:35.16 | Penguin | That just installs it. You still have to load it in asterisk. |
| 23:35.27 | Penguin | module load app_voicemail.so |
| 23:35.31 | SupaYoshi | ok |
| 23:35.55 | SupaYoshi | LOADED! great |
| 23:36.07 | SupaYoshi | mhm.. but my supura reports to be busy once again. strange |
| 23:36.39 | *** join/#asterisk apb1963_ (~apb1963@174.134.117.244) |
| 23:37.09 | *** join/#asterisk qjc (43a8a76a@gateway/web/freenode/ip.67.168.167.106) |
| 23:38.12 | SupaYoshi | but Call 1 state is idle |
| 23:59.13 | SupaYoshi | anyone that knows the location of vm-theperosn |
| 23:59.17 | SupaYoshi | vm-theperson normally? |