| 00:07.47 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 00:08.46 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.166) |
| 00:11.40 | *** join/#asterisk serafie (~erin@user-24-214-173-250.knology.net) |
| 00:42.16 | *** join/#asterisk pontis (~pontis@80.74.227.132) |
| 00:44.35 | floren | ChannelZ: what codec do you use, wav? |
| 00:55.34 | *** join/#asterisk andrewyager (~andrewyag@2-104-141-114.static-dsl.realworld.net.au) |
| 01:03.04 | *** join/#asterisk andrewyager (~andrewyag@syd02s26-fw01.thecore.net.au) |
| 01:09.25 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.166) |
| 01:21.30 | ChannelZ | Yeah. I don't get 10 minute long voicemails that it matters much |
| 01:23.01 | Katty | a 10 minute voicemail could be very.... |
| 01:23.12 | Katty | erm. nevermind. |
| 01:23.54 | apb1963 | Look! It's the bird lady of Al-CAT-raz! |
| 01:23.55 | ChannelZ | People know better. I'm armed. |
| 01:25.53 | apb1963 | Seen any good pr0n lately Katty? |
| 01:26.35 | Katty | apb1963: does good porn exist? |
| 01:26.52 | Katty | why has no one told me about good porn? |
| 01:27.01 | apb1963 | only that which I make myself |
| 01:27.13 | Katty | well there you go, i guess. |
| 01:27.33 | apb1963 | yes, I come and I go |
| 01:27.43 | Katty | aren't you just clever. |
| 01:28.03 | apb1963 | sharp as a cleaver |
| 01:28.23 | Katty | *hee* |
| 01:28.48 | apb1963 | hums the theme from Jaws |
| 01:29.31 | apb1963 | Now I feel like a shark sandwich |
| 01:29.48 | Katty | maybe you are shark bait. |
| 01:30.09 | apb1963 | Are you calling me a chum? |
| 01:30.48 | Katty | or incredibly adorable, like nemo? |
| 01:31.01 | apb1963 | I'm not a cartoon! |
| 01:31.09 | apb1963 | I happen to be a master baiter. |
| 01:31.24 | Katty | i'm not even going to ask |
| 01:31.42 | apb1963 | Well how else ya gonna catch a shark? |
| 01:31.48 | apb1963 | Gotta have bait |
| 01:31.52 | apb1963 | chum |
| 01:31.57 | Katty | here kitty kitty kitty? |
| 01:32.00 | Katty | oh wait, that's catfish. |
| 01:32.20 | apb1963 | I'm going to name my next pet Commander Snookums |
| 01:32.44 | Katty | that's adorkable |
| 01:33.15 | apb1963 | well there was a stray dog I called Kitty |
| 01:33.26 | apb1963 | he was a scaredy cat |
| 01:33.55 | Katty | aww, poor thing :< |
| 01:33.59 | apb1963 | yeah |
| 01:34.00 | Katty | i'm glad you took him in. |
| 01:34.16 | apb1963 | took him in? Are you kidding? I couldn't take him anywhere! |
| 01:34.53 | Katty | why not? |
| 01:35.03 | apb1963 | terrible manners |
| 01:35.15 | Katty | facepalms |
| 01:35.59 | apb1963 | Seriously... he would eat food right off the floor! |
| 01:36.50 | apb1963 | And I couldn't get him to put on any clothing! He wanted to run around naked! |
| 01:37.03 | *** join/#asterisk serafie (~erin@user-24-214-173-250.knology.net) |
| 01:37.48 | ChannelZ | crapped in the yard, peed everywhere |
| 01:38.27 | apb1963 | jeez ChannelZ... at your age? |
| 01:38.30 | ChannelZ | humped in front of the neighbors |
| 01:38.34 | *** join/#asterisk italorossi (~italoross@187.61.168.117) |
| 01:39.14 | *** join/#asterisk xcom (~wtf@pdpc/supporter/professional/seri) |
| 01:39.35 | apb1963 | well if the neighbors want to watch, I suppose that's their perogative |
| 01:40.03 | apb1963 | personally i'm a bit too shy for that |
| 01:40.43 | apb1963 | what about you Katty? Ever hump in front of the neighbors? |
| 01:40.54 | Katty | no, most certainly not. |
| 01:41.09 | apb1963 | well that's why you don't have any good porn I suppose |
| 01:41.20 | Katty | you're probably right. |
| 01:41.32 | Katty | i think the trade off is fair tho. |
| 01:41.37 | apb1963 | how so? |
| 01:41.39 | Katty | i still have my self respect |
| 01:41.47 | apb1963 | I see |
| 01:41.59 | apb1963 | well, there is that |
| 01:42.05 | Katty | it's kind of nice, at times. |
| 01:42.06 | *** join/#asterisk mnathani (~mnathani@198-84-231-11.cpe.teksavvy.com) |
| 01:42.42 | apb1963 | any time in particular? |
| 01:43.09 | Katty | when i'm awake. |
| 01:45.40 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 02:01.36 | *** join/#asterisk Maliuta (~nobusines@kiev.lusan.id.au) |
| 02:04.56 | *** join/#asterisk janeUbuntu (~jane@2001:3c8:c103:a001:75ff:db68:9248:e98c) |
| 02:06.07 | carrar | what |
| 02:06.21 | janeUbuntu | hello, my topology is ATA-FXS --- PBX CO (FXO), so can i write diaplan for dial to ata(FXS) and wait xx second before send DTMF extension to PBX? |
| 02:10.14 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.166) |
| 02:10.48 | *** join/#asterisk Impossibl (Impossibl@cpe-172-251-29-116.socal.res.rr.com) |
| 02:12.24 | Impossibl | Good evening everyone.. |
| 02:14.00 | Impossibl | Have a question. Have setup a Cisco 7940 and I can call other extensions from this phone but other phones cannot call this phone. CLI shows this: http://pastebin.com/raw.php?i=2ZCWXJb7 |
| 02:14.21 | Impossibl | it is setup as sip |
| 02:23.50 | Impossibl | Anyone using Cisco phones wiht their setup? |
| 02:29.35 | Impossibl | !help |
| 02:29.42 | Impossibl | @help |
| 02:33.17 | Impossibl | doesn't appear to be registering 2005 (Unspecified) D A 0 Unmonitored |
| 02:35.39 | floren | Impossibl: i managed to get my Cisco 8961 registered using this patch: https://issues.asterisk.org/jira/browse/ASTERISK-13145 |
| 02:35.40 | LieutPants | [ASTERISK-13145] [Status: Open] [patch] Presence subscription on Cisco SIP phone needs special Cisco-styled XML - https://issues.asterisk.org/jira/browse/ASTERISK-13145 |
| 02:36.13 | floren | no idea is 79xx is supported, i only dealt with 89xx and 99xx |
| 02:36.17 | carrar | Cisco 7940's do not need a patch to work |
| 02:36.25 | carrar | they just need to be registered |
| 02:56.02 | Impossibl | yes it can call out but can't call it |
| 02:56.06 | *** part/#asterisk ryan_turner (Ryan@2600:3c02::f03c:91ff:fe70:c6b0) |
| 02:56.11 | Impossibl | its not registering |
| 02:57.04 | carrar | is the phone on the same subnet as the server? |
| 02:57.33 | Impossibl | no the asterisk resides on an ubuntu 12.04 box on amazon ec2 |
| 02:58.08 | carrar | Lets see your sip.conf |
| 02:58.14 | carrar | and phone config |
| 02:58.19 | Impossibl | ok |
| 02:58.24 | Impossibl | I will pastebin |
| 02:58.28 | Impossibl | just second |
| 03:00.16 | Impossibl | here is the SIPMACADDRESS.cnf http://pastebin.com/raw.php?i=CyWB4f7d |
| 03:00.52 | carrar | is the phone behind nat? |
| 03:01.28 | *** join/#asterisk guitarHester (~guitarHes@mobile-166-147-112-237.mycingular.net) |
| 03:03.33 | Impossibl | http://pastebin.com/raw.php?i=DqSt65Y3 |
| 03:03.50 | Impossibl | yes |
| 03:03.52 | carrar | who sip.conf file pls |
| 03:03.54 | carrar | whole |
| 03:04.10 | carrar | You can add this to your cisco config |
| 03:04.10 | carrar | nat_enable: "1" |
| 03:04.10 | carrar | proxy_register: 1 |
| 03:04.10 | carrar | proxy1_port: 5060 |
| 03:04.21 | carrar | and or replace as appropriate |
| 03:04.24 | floren | quick question: what means 4 into ${EXTEN:4} |
| 03:04.47 | floren | cant find the documentation for this |
| 03:05.12 | Impossibl | http://pastebin.com/raw.php?i=K9FDfJbk |
| 03:05.50 | carrar | ugg |
| 03:05.59 | Impossibl | this is freepbx |
| 03:06.00 | carrar | you need to include all the includes |
| 03:06.04 | carrar | you missing everything |
| 03:06.20 | Impossibl | sorry its a pain with freepbx sorry about that |
| 03:06.39 | carrar | make sure you have this in the [general] section |
| 03:06.40 | carrar | pedantic=no |
| 03:07.13 | carrar | might start by using Asterisk from Source |
| 03:07.45 | floren | pedantic is set by default to no, no? |
| 03:08.00 | carrar | floren, pelase read: |
| 03:08.01 | guitarHester | floren: {variable:x:y} where x is the starting position and y is the number of digits to return |
| 03:08.01 | carrar | ~book |
| 03:08.02 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 03:08.22 | floren | thanks guitarHester |
| 03:08.36 | guitarHester | floren: ${EXTEN:4} would return everything after the fourth digit. ... I think |
| 03:08.47 | floren | carrar: i'm reading this as we speak |
| 03:08.50 | guitarHester | floren: you're welcome, hope that helps |
| 03:09.09 | guitarHester | floren: pp 144 in the 4th ed. |
| 03:09.23 | guitarHester | Just happened to have it open at the moment. |
| 03:09.31 | floren | :) |
| 03:10.03 | floren | ChannelZ: i can fix the cisco xml vomit with this: exten => _[x]-cisco-serviceuri-.,1,Goto(${EXTEN:19},1) |
| 03:10.23 | ChannelZ | huh? |
| 03:10.28 | floren | which explains the EXTEN question :) |
| 03:10.35 | carrar | floren: https://wiki.asterisk.org/wiki/display/AST/Selecting+Characters+from+Variables |
| 03:10.50 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.166) |
| 03:10.52 | floren | remember in all the traces i had tons of xml cisco related data? |
| 03:11.08 | floren | apparently this fixes it |
| 03:11.24 | ChannelZ | I can't possibly imagine how. |
| 03:11.35 | *** join/#asterisk smirker (~x@101.162.216.118) |
| 03:11.48 | carrar | it's a MAGIC STATEMENT |
| 03:12.07 | carrar | :4 spells instant death in Asia if it doesn't work |
| 03:12.16 | carrar | works out of fear |
| 03:12.20 | Impossibl | those other includes are empty |
| 03:12.29 | ChannelZ | I didn't remember you ever getting as far as being able to dial anything, the phone would quit responding. |
| 03:12.51 | carrar | Impossibl, then you are missing a lot of settings that need to exist in sip.conf |
| 03:13.13 | carrar | but at least try the changes I said for yoru cisco config |
| 03:13.19 | carrar | you will need that |
| 03:13.34 | smirker | Howdy. Using Asterisk 1.8, I'm using Dial() with the A()nnounce option to play a prompt to the called channel, while the calling channel hearings ringing. My issue is that the calling channel hears silence while the called channel hears the announcement. Is there a way to continue ringing on the calling channel until the announcement finishes playing on the called channel? |
| 03:13.48 | smirker | s/hearings/hears |
| 03:13.58 | carrar | make sure all the *name* are the same in the config too for the cisco |
| 03:14.00 | Impossibl | thanks I did lets see what happens |
| 03:14.04 | carrar | cept for Display name |
| 03:14.09 | carrar | that can be whatever |
| 03:14.36 | carrar | sorry |
| 03:14.42 | carrar | shortname can be wahtever you want |
| 03:14.47 | carrar | all the others need to match |
| 03:15.25 | carrar | AKA line1_displayname: "2005" |
| 03:16.59 | Impossibl | you rock |
| 03:17.02 | Impossibl | that did it |
| 03:17.40 | Impossibl | btw I did compile asterisk from src and then install freepbx |
| 03:17.49 | carrar | I'm sorry to hear that :) |
| 03:17.54 | Impossibl | lol |
| 03:18.05 | Impossibl | I know I need to do it all from command |
| 03:18.08 | Impossibl | its cleaner |
| 03:18.16 | Impossibl | I am with you |
| 03:18.24 | carrar | looks behind him |
| 03:18.36 | Impossibl | Thanks for your help |
| 03:18.42 | Impossibl | :P |
| 03:18.45 | carrar | it happens |
| 03:18.50 | Impossibl | btw you are funny |
| 03:18.51 | carrar | usually by accident |
| 03:19.09 | ChannelZ | commence reach-around |
| 03:19.16 | carrar | please tip your waitresses |
| 03:19.22 | Impossibl | lol |
| 03:19.31 | Impossibl | cya brothas! |
| 03:33.50 | *** join/#asterisk andrewyager (~andrewyag@syd02s26-fw01.thecore.net.au) |
| 03:36.47 | *** join/#asterisk mae_tae (7960eecf@gateway/web/freenode/ip.121.96.238.207) |
| 03:48.50 | *** join/#asterisk AliRezaTaleghani (~Thunderbi@unaffiliated/AliRezaTaleghani) |
| 03:53.10 | *** join/#asterisk guitarHester (~guitarHes@mobile-166-147-112-237.mycingular.net) |
| 03:58.47 | smirker | Wondering if anyone knows any info regarding my question posted ~1 hour ago? (regarding Dial/announcements/ringing) |
| 04:00.43 | ChannelZ | so you're saying the calling channel hears ringing until the called picks up, at which point it goes silent until the accounce to the called finishes? |
| 04:01.51 | ChannelZ | s/accounce/announce/ |
| 04:02.43 | smirker | ChannelZ, that's right |
| 04:03.55 | mae_tae | hi people, im new to asterisk, regarding dahdi trunks, what's the difference between this group <no> & analog channel <no>? and how will i able to identify it? |
| 04:04.01 | smirker | i've played around with the r/R options, and adding Ringing() before the Dial() (undesirable), however i've had no luck |
| 04:06.22 | ChannelZ | hmm yeah I just tried it. When the called channel answers the progress completes.. I guess that'd have to be something re-engineered in the source |
| 04:07.33 | smirker | hmm, in that case, i think i might have to live with it and stick with very short announcements (most are, anyway) |
| 04:08.08 | ChannelZ | mae_tae: not sure what you're talking about. group=1 signifies all the channel preceeding it are in group 1, which you can use to say Dial() out of and have it automatically pick the next free channel |
| 04:08.36 | smirker | ChannelZ, thanks for looking into it |
| 04:09.24 | *** join/#asterisk serafie (~erin@user-24-214-173-250.knology.net) |
| 04:10.22 | mae_tae | ChannelZ: does this mean that port = 1 analog channel? |
| 04:11.20 | ChannelZ | well channel=1 allocates a channel.. |
| 04:11.27 | ChannelZ | What are you actually trying to get the answer to? |
| 04:11.36 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.166) |
| 04:12.07 | ChannelZ | smirker: You could do it ghetto-style, and make a separate MOH context with a looping ringing sound, and use the m() flag along with A() |
| 04:13.22 | mae_tae | ChannelZ: we have several telephone lines but i really wonder why some of the lines on asterisk doesnt trigger that IVR when unanaswered? it just continues to ring |
| 04:13.47 | smirker | ChannelZ, interesting, but that would answer the call through, right? |
| 04:14.00 | ChannelZ | mae_tae: That all depends on how you have your dialplan programmed. |
| 04:14.21 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 04:14.45 | mae_tae | ChannelZ: does this mean that this possible problem on telephone provider's side? |
| 04:14.45 | ChannelZ | smirker: the calling channel would be considered answered at that point if it wasn't already, yes |
| 04:15.00 | ChannelZ | mae_tae: no |
| 04:15.11 | ChannelZ | mae_tae: What is your interface? How many channels do you have? |
| 04:16.29 | mae_tae | ChannelZ: fx0 1-7, i'm really dont understand what's that channel means? sorry im really new |
| 04:16.41 | *** join/#asterisk aruntomar (~Thunderbi@49.248.155.155) |
| 04:17.12 | ChannelZ | so you have like a TDM800 card with 8 FXO ports, but only have 7 phone lines? |
| 04:18.26 | mae_tae | ChannelZ: yeah, you're right |
| 04:19.00 | ChannelZ | and you're saying that at some point when someone calls, asterisk never answers? |
| 04:20.14 | mae_tae | no, if we call some of the phone lines externally, seems IVR on it doesnt work, just some of it |
| 04:21.25 | ChannelZ | well.. barring hardware failure on some of your FXO ports, look at your /etc/asterisk/chan_dahdi.conf for channel => lines. Perhaps you haven't allocated all your channels. pastebin it if you want and we can look |
| 04:22.07 | ChannelZ | or they might not even be allocated by the driver, which is in /etc/dahdi/system.conf |
| 04:22.34 | mae_tae | and other phone lines were busy, if we call from internal, but if we test it line by line, no problem |
| 04:22.37 | mae_tae | ok ill look at it |
| 04:22.40 | ChannelZ | doing 'dahdi show channels' on the asterisk console will tell you all the channels asterisk knows about |
| 04:24.38 | mae_tae | that's why im confuse of what this is GROUP all about and Analog Channel, because i try to change one of the phone lines & specify dahdi trunks to group 1, instead of analog channel, seems that phone is working fine |
| 04:24.53 | mae_tae | ok ill paste the configuration |
| 04:27.15 | ChannelZ | ~pb |
| 04:27.15 | 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. |
| 04:27.25 | ChannelZ | ^^^ |
| 04:30.15 | mae_tae | http://pastebin.com/mKkgzYuz |
| 04:31.06 | smirker | smirks |
| 04:31.10 | ChannelZ | ugh none of those have your channels. see the three #include lines at the bottom, look in those files. |
| 04:31.58 | ChannelZ | and you're running freepbx so thats a wildcard |
| 04:32.08 | *** join/#asterisk killown (~killown@pdpc/supporter/student/killown) |
| 04:32.53 | mae_tae | k, btw i use freepbx (web gui) in configuration |
| 04:33.06 | mae_tae | ok ill look at it |
| 04:33.51 | ChannelZ | yeah chances are your "trunks" or whatever they refer to them as aren't hooked up right through it, but I have no idea cuz I don't use it |
| 04:36.09 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 04:36.44 | mae_tae | http://pastebin.com/LF67YR6U |
| 04:37.09 | mae_tae | thats for chan_dahdi_group.conf |
| 04:39.51 | ChannelZ | ok so all 7 of your channels are being consumed |
| 04:39.57 | ChannelZ | the rest is probably a freepbx issue |
| 04:41.03 | ChannelZ | you can test from the console by doing 'core set verbose 3' and then dialing each number in turn from an external number, see if you get 'Starting simple switch on DAHDI/X-X' for each one, just to make sure they're all physically working |
| 04:41.48 | mae_tae | yeah right, that's really now that im trying to solve, which i dont have idea at all |
| 04:48.39 | *** join/#asterisk andrewyager (~andrewyag@syd02s26-fw01.thecore.net.au) |
| 04:48.49 | mae_tae | ChannelZ: core: command not found |
| 04:50.32 | ChannelZ | on the asterisk console, not the shell |
| 04:50.35 | ChannelZ | asterisk -rvvv |
| 04:50.40 | ChannelZ | that will do it from the shell |
| 04:51.29 | mae_tae | ok |
| 04:52.56 | ChannelZ | oh dear, there are little 3D book thumbnails on amazon.com now |
| 04:54.25 | mae_tae | ChannelZ: im now in the console, so i will try to call those trunk line nos of telco's & then see if there is that 'starting....' something that would come out, is that what you mean? |
| 04:55.05 | mae_tae | xtension Changed 108[ext-local] new state Idle for Notify User 1203 -- Connected line update to SIP/213-000004ed prevented. -- SIP/131-000004ee answered SIP/213-000004ed |
| 04:56.59 | ChannelZ | You should see something like -- Starting simple switch on 'DAHDI/1-1' and then probably a bunch of other crap will flow after, being freepbx |
| 05:08.46 | mae_tae | ChannelZ: phone no's having that message 'starting simple....', has that working IVR, but other nos are just ringing or busy, no 'starting.....' message |
| 05:12.43 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.181) |
| 05:13.35 | mae_tae | ChannelZ: so what would I trace/do if that 'starting simple...' would not come out? |
| 05:15.07 | mae_tae | [2013-07-10 13:14:53] NOTICE[4849]: chan_sip.c:26387 sip_poke_noanswer: Peer '203' is now UNREACHABLE! Last qualify: 14 |
| 05:15.51 | ChannelZ | Do you not see it EVER? |
| 05:17.08 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 05:20.15 | mae_tae | ChannelZ: yes i find some but those numbers having that are numbers with fine IVR |
| 05:20.46 | mae_tae | but those numbers with just continue ringing, i dont find that display |
| 05:22.42 | ChannelZ | What do you get if you do 'dahdi show channels' on the asterisk console? |
| 05:23.01 | mae_tae | for a while ill try it |
| 05:23.47 | mae_tae | Service 1 from-zaptel en default <--- somethng like that up to service 8 |
| 05:24.48 | mae_tae | up to channel 8 i mean |
| 05:25.15 | ChannelZ | it doesn't say 'In Service' at the end of each? |
| 05:26.46 | mae_tae | ChannelZ: yap you're right... |
| 05:35.49 | ChannelZ | well it's hard to say for sure, but two possibilites are that those phone lines aren't functioning correctly from your telco, or you've got dead ports on your card. Did they ever work before? |
| 05:35.59 | ChannelZ | the numbers that aren't working now, that is |
| 05:39.40 | mae_tae | ChannelZ: if i'll use the phone line directly, i can call any number externally |
| 05:42.06 | ChannelZ | as in you hook a telephone up to the physical line? |
| 05:42.46 | mae_tae | yes.... |
| 05:43.17 | ChannelZ | and if you call that number from a different phone does that work? (since that's what you're having issues with) |
| 05:44.21 | mae_tae | yes, if it is not hook up in the freepbx server |
| 05:45.19 | ChannelZ | Of your 8 lines, how many don't work, and are they next to each other (IE the ports on the TDM card the plug into) |
| 05:45.48 | mae_tae | example if i call 3419948 here, no IVR working, it just continues ringing, but other phones IVR works very fine |
| 05:46.31 | mae_tae | out of 7, there 3 which is not working well |
| 05:46.31 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-141.moldtelecom.md) |
| 05:47.46 | ChannelZ | and are all 3 on consecutive ports? |
| 05:49.14 | mae_tae | no... 2 are placed in ports 6 and 7 (no IVR working) , the other is place somewhere between 1 & 5 |
| 05:49.56 | ChannelZ | hmm.. well the last thing to try is to take one of the non-working lines, and plug it into the port of one of the working ones, and see if that number starts working |
| 05:50.06 | ChannelZ | if so, you might have a hardware problem |
| 05:50.37 | mae_tae | port 1 (IVR works fine) |
| 05:52.27 | ChannelZ | so a line belonging to a number that didn't previous worked, now works when you put it in port 1? |
| 05:52.43 | mae_tae | ChannelZ: last thing, if assuming all ports are working well, what other configure that i will look into |
| 05:53.14 | mae_tae | no, i haven't transferred it yet... |
| 05:53.26 | mae_tae | ill try to transfer it now |
| 05:54.47 | mae_tae | ChannelZ: here's another thing lines in ports 6 & 7 is different telco's with those ports 1-5, is that alright if i just move in the port 6 line to port 1 example? |
| 05:55.09 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.197) |
| 05:58.34 | ChannelZ | shouldn't matter |
| 06:00.41 | mae_tae | k |
| 06:04.50 | *** join/#asterisk v0lZy (~Thunderbi@mail.silk-group.net) |
| 06:11.44 | floren | ChannelZ: i made some progress, bob. i can now dialout but i cannot dialin anymore. spent some time on reading the asterisk book and i'm getting a hang of everything... |
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| 06:12.04 | floren | configs: http://codepad.org/oeW4GIME |
| 06:12.18 | floren | sip trace for incomming call: http://codepad.org/JPjD0ksq |
| 06:12.31 | floren | i'm missing something small, i guess |
| 06:13.14 | mae_tae | ChannelZ: i tried to transfer the line to line having good port, but still doesnt work, failed |
| 06:16.20 | ChannelZ | hmm so then it's something about the line. You said some of them are from different telcos, are the non-working ones from the same telco, and a different one than the lines that work? |
| 06:16.27 | mae_tae | i mean to port having a good line, hehe |
| 06:16.43 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.197) |
| 06:19.28 | mae_tae | something like this port 6 & 7 (DTPL, can call if physically externally if hook up, no ivr working), port 1 - 5 (Bajan, 4 of it with working IVR), |
| 06:20.59 | mae_tae | i tried totransfer one bajan not working port to good port, but not working |
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| 06:22.27 | mae_tae | something like this port 6 & 7 (DTPL, can call external if physically hook up, no ivr working), port 1 - 5 (Bajan, 4 of it with working IVR), |
| 06:23.23 | ChannelZ | well without being there I don't know what else to suggest. You've basically ruled out the ports being bad on your TDM card, by moving a non-working line to a port you know works with a different line |
| 06:24.40 | mae_tae | but i suspect its a DAHDI trunks issue, that GROUP option & Analog Channel option |
| 06:24.50 | mae_tae | but anyway thank you for time |
| 06:24.54 | mae_tae | and help |
| 06:25.18 | ChannelZ | groups have nothing to do with it |
| 06:25.57 | ChannelZ | you never answered one of my earlier questions, did all of these lines ever work, is it something that just stopped working recently? |
| 06:29.10 | ChannelZ | floren: well it looks like your incoming call works fine, but all your dialplan does is answer and then hangup. |
| 06:29.26 | ChannelZ | and your dialplan is back to being a general mess |
| 06:30.04 | floren | :) |
| 06:30.22 | ChannelZ | all of your contexts are mixed up, you have potential security problems with the way it's setup |
| 06:30.22 | floren | don't you love it messy? :) |
| 06:30.33 | ChannelZ | mess is one thing but none of this is going to work |
| 06:30.37 | floren | i understand |
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| 06:31.10 | ChannelZ | you're sending incoming calls into [101-local] which also has extensions that can dial back OUT |
| 06:31.44 | floren | the good thing from all of this is that now I understand perfectly what you mean |
| 06:31.49 | ChannelZ | At minimum it's a loop, like you started with 2 days ago or whenever |
| 06:32.03 | floren | let me change that. i'll start fresh |
| 06:32.10 | floren | and remote the config in a bit |
| 06:32.21 | floren | repost* |
| 06:32.37 | ChannelZ | Contexts are like rooms in your house, and you put the inlaws from your side of the family in a separate room from the inlaws on your wife's side, so they can't interact with each other and cause problems. |
| 06:32.53 | floren | ya :) |
| 06:33.19 | ChannelZ | Incoming calls from your ITSP should go to one context. Your local devices should go into another. |
| 06:33.37 | floren | doing this right now, i see what you mean |
| 06:33.39 | ChannelZ | And when you include => one from the other, you've effectively broken down that wall. |
| 06:35.23 | ChannelZ | I use 3 contexts. [incoming], [internal], and [out]. incoming has my exension(s) for handling incoming calls. internal has my internal extensions for dialing between my local phones, accessing voicemail, etc. out has extensions for dialing outside numbers. |
| 06:35.40 | mae_tae | ChannelZ: i dont know if all lines are really working, we havent trace 1 by 1 b4, because our concern b4 that we can call internally & externally until these problem now arises |
| 06:35.57 | floren | i see, i'll draw the same logic |
| 06:36.05 | ChannelZ | My ITSP goes into 'incoming'. My devices go into 'out' which then includes 'internal', so my local devices can call out and each other. |
| 06:36.44 | ChannelZ | (and in reality I have 2 'out' contexts, 'out_local' and 'out_ld" for local and long distance, so I can restrict certain phones from calling numbers I don't want them to.) |
| 06:37.12 | ChannelZ | but you get the idea. Contexts are logical separation, but more importantly, ACCESS separation. |
| 06:37.55 | floren | ya |
| 06:38.27 | ChannelZ | mae_tae: well it's hard to say if you don't know if they all ever functioned correctly at one time. The systematic way to do this is to unplug ALL your lines. |
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| 06:39.11 | ChannelZ | mae_tae: then, one by one, plug each line into port 1, and call its phone number from some other place (mobile, whatever) and see if it works. |
| 06:39.42 | mae_tae | ChannelZ: does analog channel also matters? |
| 06:39.53 | ChannelZ | mae_tae: identify which lines don't work. (Asterisk doesn't know the difference. If one works in port 1, they should ALL work in port 1.) |
| 06:40.08 | ChannelZ | mae_tae: then call your telco for those lines and ask why they are different. |
| 06:40.17 | ChannelZ | ALL of your lines are analog, are they not? |
| 06:41.16 | mae_tae | yes... they are all analog |
| 06:41.44 | ChannelZ | It's possible (though unlikely) that your non-working lines have a different signalling method from the telco(s). You are using kewlstart on all of them, but maybe your telco(s) have some on loop-start or ground-start or something. |
| 06:42.27 | *** join/#asterisk russum (~russum@78-3-120-238.adsl.net.t-com.hr) |
| 06:43.14 | ChannelZ | Because for whatever reason Asterisk is not realizing a call is coming in on those lines, which all things being equal (IE the port it's plugged into on your TDM card is otherwise functional) means it's not recognizing the signal. |
| 06:43.18 | *** part/#asterisk russum (~russum@78-3-120-238.adsl.net.t-com.hr) |
| 06:44.36 | ChannelZ | Could be a negligible line voltage or something. Or the wrong signalling method as I said. Given the one-by-one test I mentioned above, I think you'll need to move on to debugging it with your telcos. |
| 06:46.13 | mae_tae | ok, thanks again ill contact the telco |
| 06:46.54 | ChannelZ | good luck |
| 06:47.53 | ChannelZ | wanders off to shower |
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| 06:49.59 | floren | good night ChannelZ |
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| 08:24.23 | ChannelZ | yawns |
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| 09:31.15 | xoveruk | hi |
| 09:31.42 | xoveruk | I have two server connected to each other using IAX. One side shows IAX peer reachable and the other not. |
| 09:32.30 | xoveruk | I can see the reason why is because the dst port number is incorrect. however I am confused why the side sending the wrong port can place calls, yet the side sending the correct port number cannot. |
| 09:34.53 | xoveruk | How is it possible for a one way communication |
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| 11:36.34 | mic_ | it sounds like a rhetorical question |
| 11:36.43 | mic_ | but are openvox cards mega picky |
| 11:36.53 | mic_ | or they just require some special conditions? |
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| 12:25.12 | WIMPy | mic_: Why? |
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| 12:43.12 | rolandow | hi .. i'm installing asterisk on a public server, to run our pbx in the cloud.. |
| 12:43.18 | rolandow | now, what should i use for localnet? |
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| 12:43.53 | itgrl | Morning all |
| 12:44.32 | rolandow | because the sample file says ; IF LOCALNET IS NOT SET, THE EXTERNAL ADDRESS WILL NOT BE SET CORRECTLY. |
| 12:45.49 | WIMPy | Did you also read the headline? |
| 12:46.05 | WIMPy | That's all for NAT situations. |
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| 12:46.57 | rolandow | ok |
| 12:47.05 | rolandow | so i don't have to set externaddr as well? |
| 12:47.32 | WIMPy | nly if your Asterisk is behind NAT. |
| 12:47.49 | WIMPy | O |
| 12:47.57 | rolandow | ok |
| 12:49.44 | rolandow | thanks! |
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| 13:06.57 | Kobaz | wow this is the third time this has happened |
| 13:07.03 | Kobaz | asterisk starts up and manager is not running |
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| 14:25.03 | Katty | pokes people, looking for signs of life |
| 14:25.54 | WIMPy | Asterisk made us all to zombies. |
| 14:26.01 | trdillon1 | nice |
| 14:27.00 | jmetro | consumes Katty with T-virus |
| 14:34.15 | Katty | hugs jmetro |
| 14:35.04 | jmetro | notes that zombies approve of hugs. Rawr |
| 14:35.32 | Katty | ^_^ |
| 14:37.45 | jmetro | :D whats new on the ast front |
| 14:39.04 | Katty | i heard someone put asterisk on a pi, and then in a phone |
| 14:39.08 | Katty | i think it was ken. |
| 14:39.12 | Katty | but maybe he was using freeswitch |
| 14:40.29 | jmetro | i think i heard of freeswitch doing that |
| 14:40.38 | jmetro | you could definitely root a phone and stick it on there though |
| 14:41.18 | Katty | the photos were pretty fancy. |
| 14:41.27 | Katty | sadly we don't do much with asterisk here these days. |
| 14:41.38 | Katty | no one wanted to buy it, so pretty much all they sell is Toshiba |
| 14:41.54 | Katty | i think the owner may have some Hosted Solutions in the works using asterisk tho |
| 14:42.06 | Katty | who knows, maybe i'll get to tinker with it again soon |
| 14:45.28 | jmetro | where do you work at? |
| 14:45.46 | jmetro | its easy enough to setup a VM of it and tinker around with softphones |
| 14:52.15 | Katty | oh it's just a little company out in the middle of nowhere. about 10 people. their primary focus is copiers and printers |
| 14:52.28 | Katty | and yes, vmware was the plan. |
| 14:52.36 | Katty | things just move slowly here. |
| 14:57.32 | jmetro | MFP's and big stuff or small stuff too? |
| 14:58.47 | Katty | both. |
| 14:58.52 | Katty | even wide format printers. |
| 14:59.12 | Katty | we're a kyocera, toshiba, hp, and konica minolta dealer.. i think |
| 14:59.14 | jmetro | wide formats are annoying, they blow up with the HP Universal driver =p |
| 14:59.20 | Katty | i don't work on the printer stuffs here, so not really sure |
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| 15:07.18 | FlashDeluxe | hi@all! I got a problem. I am using asterisk 1.8.10 and dahdi 2.4.0, everything is working fine, except sending faxes. If i try to send a Fax, it builds up the connection, but then it fails :-( I see a debug message "res_rtp_asterisk.c:1555 process_dtmf_rfc2833: Ignoring RTP 2833 Event: 00000036" but i don know what it means... Here is the complete log: http://pastebin.com/QXcERWLq |
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| 15:39.17 | igcewieling | FlashDeluxe: fax over voice over IP is not expected to work. Try using T.38 if your provider supports it. What ATA are you using to convert fax to analog? |
| 15:39.45 | FlashDeluxe | i am using the grandstream ht-502 |
| 15:40.16 | FlashDeluxe | i configured it to use t.38 and now, the error doesn`t occur, but the faxes are not sended anyway :-( |
| 15:40.49 | igcewieling | you know Asteirsk does not support SIP -> DAHDI fax calls until Asterisk 11? This is called gateway functionality |
| 15:41.14 | igcewieling | FlashDeluxe: fax and t.38 are some of the hardest things to make work with asterisk |
| 15:42.15 | drmessano | Don't expect any of this to work with 1.8 |
| 15:42.26 | Katty | hi danny |
| 15:42.27 | FlashDeluxe | igcewieling, but i can remember that it worked once... :-( So i got to upgrade my asterisk or what is your suggestion? |
| 15:42.34 | Katty | and mister ice weasel |
| 15:42.36 | WIMPy | It's probably best not to expect anything to work. |
| 15:43.12 | igcewieling | FlashDeluxe: If you keep it as fax over voice over IP over ulaw it will work some of the time and will never be reliable. |
| 15:43.22 | WIMPy | tried to set up a box using dahdi on the weekend with horribles results. |
| 15:43.31 | igcewieling | because asterisk doesn't even consider it a fax call |
| 15:43.51 | igcewieling | WIMPy: The failure is entirely yours. 8-) |
| 15:44.21 | WIMPy | Yes, I used Asterisk and dahdi. Big mistage obviousely. |
| 15:44.24 | igcewieling | make && make install && make config && dahdi_genconf && service dahdi restart |
| 15:45.12 | FlashDeluxe | igcewieling: Soo its not reliable with asterisk 11, too? |
| 15:45.20 | WIMPy | Once again something that should have been simple but will require extensive testing to find out why it failed. |
| 15:45.31 | igcewieling | FlashDeluxe: it should be mostly reliable if you use T.38 |
| 15:45.36 | jmetro | I think my lower back has turned into a giant knot of pain |
| 15:45.50 | igcewieling | WIMPy: *nod* |
| 15:46.18 | igcewieling | jmetro: any sign of alien abduction? |
| 15:46.47 | WIMPy | Additional signs? |
| 15:46.52 | FlashDeluxe | igcewieling: mhh ok and is there a possible solution to override asterisk and use it stand-alone? |
| 15:47.35 | igcewieling | FlashDeluxe: I guess you could connect your ATA directly to an ITSP or you can do what people who want reliable fax do: install a POTS analog line. |
| 15:48.33 | igcewieling | That's not a knot, that is the future conqueror of Earth! |
| 15:48.39 | jmetro | or do what people who want reliable fax AND know what the internet is: email. |
| 15:50.28 | FlashDeluxe | tell it to my customer.. |
| 15:53.18 | igcewieling | We use Adtran TA9xx and NetVanta boxes when we need analog or T.38 |
| 15:53.33 | igcewieling | direct to carrier, running them through Asterisk makes faxes fail for us. |
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| 19:23.43 | WIMPy | has a feeling that Asterisk is trying to tell him that he shouldn;t use it. |
| 19:23.55 | WIMPy | Now I gut a stuck IAX trunk. |
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| 19:38.55 | jmetro | well thurs your problem |
| 19:39.38 | WIMPy | Might be ASTERISK-21996 |
| 19:39.39 | LieutPants | [ASTERISK-21996] [Status: Waiting for Feedback] [Assigned: modelnine] chan_iax2 fails to process network packets after a while - https://issues.asterisk.org/jira/browse/ASTERISK-21996 |
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| 20:06.38 | Hive | I know you can select "MixMonitor" or "Monitor" for "monitor-type" in queues.conf, but is there a way to completely disable recording for queue calls all together? |
| 20:07.23 | [TK]D-Fender | not-specifying to record = not rcord |
| 20:08.05 | Hive | Not specifying "monitor-type" defaults to "monitor-type=Monitor" |
| 20:08.53 | Hive | I'm looking through the options for queues.conf and i just dont see anything that says 'set this to yes to enable monitoring' |
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| 20:09.31 | [sr] | dahdi-linux is 2.7.0 now |
| 20:09.32 | [sr] | :p |
| 20:10.26 | [sr] | guys, i track changes on asterisk and dahdi-linux, but i don't see changes for a longgggggggg time, but new releases always come out, like 2.6.2 and 2.7.0, but no change on the repo, |
| 20:10.35 | [sr] | did it changed? the one that is on the webpage |
| 20:12.13 | Hive | [TK]D-Fender, maybe I'm misreading this; By setting monitor-type = MixMonitor, when specifying monitor-format to enable recording of queue member conversations, app_queue will now use the new MixMonitor application instead of Monitor so the concept of "joining/mixing" the in/out files now goes away when this is enabled. You can set the default type for all queues here, and then also change monitor-type for individual queue |
| 20:12.13 | Hive | s within queue by using the same configuration parameter within a queue configuration block. If you do not specify or comment out this option, it will default to the old 'Monitor' behavior to keep backward compatibility. |
| 20:13.06 | WIMPy | [sr]: It works with current Linuxes. |
| 20:13.21 | Hive | So if you don't specify "monitor-type" then you would need to use the Monitor() command to cause a recording to be started? |
| 20:13.38 | [TK]D-Fender | http://svnview.digium.com/svn/asterisk/branches/11/configs/queues.conf.sample?view=markup |
| 20:13.59 | [sr] | WIMPy: didn't get it, you mean 2.7x ? |
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| 20:14.18 | WIMPy | yes |
| 20:14.25 | [sr] | WIMPy: ah, i'm saying, that new release has been out, but i don't see the change in the svn repo, got it? |
| 20:14.28 | [TK]D-Fender | 389 |
| 20:14.52 | [sr] | maybe the link on the webpage is outdated no? |
| 20:15.39 | [sr] | as for [TK]D-Fender link, maybe it changed from svn.asterisk.com to svn.digium.com ? |
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| 20:30.54 | jmetro | same => n,Goto(1) will just jump to the beginning right <.< |
| 20:31.07 | jmetro | of the current context |
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| 20:40.11 | Hive | [TK]D-Fender, apparently it's based on "monitor-format", not specifying this will disable all recording |
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| 20:56.40 | Katty | fender bender. |
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| 21:29.29 | artyx | Is there a command line test to see if dial tone is available on a dahdi device |
| 21:30.03 | artyx | FOr example. istheredial dahdi-1 or something ideal return would be 1 or 0 |
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| 21:32.22 | WIMPy | You are aware that you only get a dialtone after seizing a channel and befoe dialling? |
| 21:32.58 | artyx | My problem is there is a moh module that is being used, and when the moh source dies (stream) my dahdi trunk also dies |
| 21:33.52 | artyx | Now a simple solution would be to go to normal boring static moh sources... But my desired solution (aside from fixing the code to NOT do that) would be just to check to see if the trunk has hung yet, and if so then reboot the asterisk srv |
| 21:33.55 | [TK]D-Fender | And this is related how? |
| 21:34.00 | WIMPy | I don't understand the relation. |
| 21:34.39 | artyx | START streaming via mpg123 ... start asterisk, initialize dahdi trunks... NOw sometime later the mpg123 stream gets snipped/bumped/kicked off |
| 21:34.48 | artyx | AS soon as that happens, my dahdi trunk locks up as well.... |
| 21:35.13 | artyx | So when i attempt to dial in, i get a busy signal, when i attempt to dial out, i get a busy signal |
| 21:35.56 | artyx | I am 70% certain my sip trunks quit working as well, but i dont remmeber atm if thats the case |
| 21:36.39 | artyx | Problem: WHEN the mpg123 source dies, my phone line quits "working" ;) |
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| 22:17.35 | jmetro | huh |
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| 22:31.55 | Freeaqingme | hi folks; I'm trying to deploy my pbx on our production server, but am running into nothing but 401 and 403 unauthorized/forbidden responses. Is there any obvious cause if there's no NAT involved? |
| 22:32.32 | WIMPy | How is that related to NAT? |
| 22:33.43 | Freeaqingme | well, most of the times, errors are caused by a nat setup in my case. But that's not the case this time ;) |
| 22:33.54 | jmetro | sounds like you are unauthorized or forbidden |
| 22:34.04 | jmetro | i would check the things you use to become authorized |
| 22:34.05 | artyx | Yeh, you need to get your permit first |
| 22:34.16 | Freeaqingme | jml |
| 22:34.19 | Freeaqingme | oops |
| 22:35.42 | Freeaqingme | what I see happen is client does a REGISTER, server responds with 401 unauthorized, client does a Register (with auth header), asterisk responds with a 403 forbidden |
| 22:37.16 | jmetro | sounds like you have the wrong username or password |
| 22:37.27 | jmetro | i would try the forgot password link in sip.conf and the phones web ui |
| 22:38.15 | artyx | Also while your at it, can you post your sip.conf contents (censored as needed) up to pb ? |
| 22:38.19 | WIMPy | likes that one. |
| 22:38.50 | artyx | at the very least, the extension in question |
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| 22:40.00 | Freeaqingme | sec |
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| 22:40.19 | artyx | Man the guy who had my job before i won it should be drawn and quartered |
| 22:40.38 | artyx | err. castrated and fed back to him, then drawn and quartered |
| 22:42.10 | Freeaqingme | jmetro, artyx WIMPy https://gist.github.com/Freeaqingme/c23ccbda46199645f0c1 |
| 22:42.28 | Freeaqingme | client is on 172.17.3.59, server on 172.18.3.3 |
| 22:42.47 | Freeaqingme | I've btw copy/pasted the password from 'sip show users' ( a dozen times) |
| 22:42.55 | Freeaqingme | and config seems to be loaded properly |
| 22:45.11 | jmetro | i seem to remember something about type=friend vs type=peer |
| 22:45.13 | jmetro | and one being awful. |
| 22:45.44 | jmetro | does your password contain ""s |
| 22:45.47 | jmetro | because your sip.conf does. |
| 22:46.24 | Freeaqingme | ha! |
| 22:46.40 | Freeaqingme | after stripping the password of all special chars I rembmer thinking, " i should also try without quotes" |
| 22:46.43 | Freeaqingme | that's it... |
| 22:46.49 | jmetro | =) |
| 22:50.06 | Freeaqingme | < afk to bed. tnx guys |
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| 23:55.03 | nick123456 | asterisk 11 how do I cange incomming caller ID to include a few digits before it |
| 23:55.45 | nick123456 | example incoming id is 800-555-1212 I want to add 79 so it would be 79-800-555-1212 |
| 23:56.06 | WIMPy | core show function CALLERID |
| 23:56.38 | paulc | exten => blah,n,Set(CALLERID(num)=79${CALLERID(num)}) something like that maybe |
| 23:57.10 | nick123456 | Ok |
| 23:57.15 | nick123456 | I will try |
| 23:57.22 | nick123456 | Thank you! |