| 00:05.53 | dar123 | <PROTECTED> |
| 00:07.11 | *** join/#asterisk Champi (Champi@rootshell.fr) |
| 00:10.06 | dar123 | i am killing the windows on the box, getting rid of the vmware thingy too |
| 00:10.28 | dar123 | btw do you guyz recommend to go for asterisk 12 now or should i wait for some time? |
| 00:10.56 | WIMPy | I'd recommend 11. |
| 00:10.57 | [TK]D-Fender | Asterisk 12 is not released yet |
| 00:11.18 | [TK]D-Fender | And will not be LTS |
| 00:11.32 | WIMPy | has a feeling that 12 will be the new 1.6. |
| 00:11.44 | WIMPy | Or possibly worse. |
| 00:12.17 | WIMPy | But upgrades are like beer. The last one is often bad. |
| 00:12.28 | dar123 | saying byebye to my very stable winxp aaah |
| 00:13.33 | *** join/#asterisk gravspeed (gravspeed@66.242.174.34) |
| 00:13.47 | *** part/#asterisk tm1000 (tm1000@2600:3c01::f03c:91ff:fe70:fce6) |
| 00:13.52 | gravspeed | anyone familiar with using adtrans with asterisk? |
| 00:14.06 | karl-s | they work |
| 00:14.10 | gravspeed | lol |
| 00:14.14 | karl-s | but you probably want to be more specific |
| 00:14.17 | karl-s | :) |
| 00:14.31 | gravspeed | they do... i am having a problem with my ring group |
| 00:14.59 | gravspeed | when line 1 if off the hook it rings in silence for 32 seconds before going to line 2 |
| 00:15.06 | gravspeed | "rings" |
| 00:15.30 | karl-s | are we talking analog? sip? pri? or what? |
| 00:15.31 | *** join/#asterisk tm1000 (tm1000@2600:3c01::f03c:91ff:fe70:fce6) |
| 00:15.35 | gravspeed | sip to analog |
| 00:15.44 | gravspeed | i guess i should have included that... sorry |
| 00:16.08 | karl-s | probably would need to see a call and some dialplan (how you are doing the 'ring group') |
| 00:16.28 | gravspeed | i have each analog line registering as an extension, extensions as part of a ring group using round |
| 00:16.50 | karl-s | ah ok so its a channelbank type setup? |
| 00:17.05 | gravspeed | to be connected to an old school pbx |
| 00:17.43 | karl-s | so the analog 'extensions' off of the adtran channel bank are extensions on a legacy pbx? |
| 00:19.03 | gravspeed | no the analog ports are tied to extensions and did's route through the ring groups... the existing pbx will simply be connecting to the analog ports |
| 00:19.26 | Penguin | I missed how this related to asterisk. |
| 00:20.54 | karl-s | i'm a little lost too. Are we talking Adtran SIP to Asterisk SIP? who has the DID's ? where does the legacy PBX sit? |
| 00:21.26 | gravspeed | adtran sip to asterisk sip, legacy pbx is on customer site |
| 00:22.27 | gravspeed | i feel like it's ringing to call waiting even though i do not hear a beep... |
| 00:23.22 | gravspeed | we will have the dids when the lmp is complete |
| 00:24.31 | gravspeed | balls... i think i just found the (incredibly simple) command i was looking for... |
| 00:24.40 | gravspeed | no call-waiting... shit |
| 00:27.24 | gravspeed | yep, that worked... |
| 00:27.33 | gravspeed | i feel like a dummy now... thanks guys ;-) |
| 00:28.04 | karl-s | cool |
| 00:28.12 | karl-s | glad its solved |
| 00:30.03 | gravspeed | i kinda expected that to be part of asterisk and not the adtran... but i suppose the adtran is responsble for saying weather or not it is busy so i guess it does make sense |
| 00:56.37 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.51) |
| 01:13.05 | *** join/#asterisk vtone (~vtone@cpc35-bagu10-2-0-cust138.1-3.cable.virginm.net) |
| 01:13.46 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 01:27.46 | dar123 | You do not appear to have the sources for the 2.6.32-358.el6.i686 kernel installed. |
| 01:27.47 | dar123 | make[1]: *** [oct612x-lib] Error 1 |
| 01:28.13 | dar123 | getting this error while running make for dahdi |
| 01:28.33 | dar123 | i have kernel-devel installed |
| 01:29.11 | [TK]D-Fender | you also need headers and a whack of other stuff |
| 01:30.56 | dar123 | hearders are already tehre |
| 01:31.02 | dar123 | and lots of whack too |
| 01:31.08 | dar123 | not sure exactly whats missing |
| 01:31.25 | dar123 | its 4:31 am i want to finish this install and goto bed :( |
| 01:36.52 | dar123 | LOL, just needed a reboot |
| 01:40.02 | [TK]D-Fender | helps when it matches the running kernel and everything ;) |
| 01:40.33 | *** join/#asterisk onizo (~onizo@cpe-98-155-117-134.san.res.rr.com) |
| 01:41.29 | dar123 | yups, figured that out |
| 01:42.06 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 01:42.22 | *** join/#asterisk bipul (~bipul@unaffiliated/bipul) |
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| 02:27.18 | SupaYoshi | heya |
| 02:27.32 | SupaYoshi | i got a new router, and i think it blocks asterisk SIP |
| 02:27.38 | SupaYoshi | I opened up port 5060 UDP and TCP |
| 02:27.41 | SupaYoshi | but no luck so far. |
| 02:28.20 | vtone | SupaYoshi, are you seeing anything on the asterisk host with ngrep -W byline '.*' port 5060 ? |
| 02:29.03 | SupaYoshi | no i dunt have that installed |
| 02:29.09 | vtone | tshark ? |
| 02:29.13 | SupaYoshi | chan_sip.c:4164 retrans_pkt: Retransmission timeout reached on transmission 03b67ff6583d2ef76d0378c87b09cc72@127.0.0.1 for seqno 107 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions |
| 02:29.14 | SupaYoshi | Packet timed out after 6400ms with no response |
| 02:29.21 | SupaYoshi | Is what im getting. :) internal calls work fine.. |
| 02:29.50 | SupaYoshi | Do i need to open port 5004 too maybe. |
| 02:30.03 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 02:30.05 | vtone | what does netstat -anp | grep asterisk say ? |
| 02:30.50 | SupaYoshi | http://paste.ubuntu.com/6440693/ |
| 02:30.55 | outtolunc | notes.. he probably got a cable internet wideband router.. and it has a ALG that doesn't work ;) |
| 02:31.40 | vtone | argh ALGs - the demon of infrastructure teams |
| 02:32.21 | SupaYoshi | nah i got a openwrt router |
| 02:32.35 | vtone | SupaYoshi, install ngrep and verify the asterisk host is sending out traffic using the command from before |
| 02:32.36 | SupaYoshi | everything works, but i have UPNP disabled.. could be it. |
| 02:34.18 | *** join/#asterisk jasonwert (~w3rt@71.89.137.28) |
| 02:34.44 | SupaYoshi | http://paste.ubuntu.com/6440702/ |
| 02:34.47 | SupaYoshi | i think that works |
| 02:35.20 | Penguin | Why 5004? |
| 02:35.53 | SupaYoshi | cus internet said so? |
| 02:35.59 | Penguin | What is 5004 for? |
| 02:37.14 | Penguin | I know what a certain document, which I am not going to name, says it is, but I want to know what you think it is. |
| 02:37.37 | [TK]D-Fender | SupaYoshi: Show us actual SIP debug from * CLI |
| 02:37.46 | [TK]D-Fender | SupaYoshi: a complete call..... |
| 02:37.49 | SupaYoshi | RTP |
| 02:37.57 | SupaYoshi | Yeah sure! Hold on |
| 02:39.48 | SupaYoshi | http://paste.ubuntu.com/ |
| 02:39.56 | SupaYoshi | http://paste.ubuntu.com/6440711/ |
| 02:39.58 | SupaYoshi | oops sorry |
| 02:40.53 | SupaYoshi | see line 97 |
| 02:42.07 | *** join/#asterisk SushiB (~Thunderbi@187.184.23.221.cable.dyn.cableonline.com.mx) |
| 02:42.36 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 02:43.20 | [TK]D-Fender | You seem to have missed the words SIP DEBUG |
| 02:43.28 | [TK]D-Fender | "sip set debug on" <---------- |
| 02:44.09 | SupaYoshi | O.o |
| 02:44.15 | SupaYoshi | thats alot of info |
| 02:44.23 | [TK]D-Fender | Yeah... like the IMPORTANT stuff... |
| 02:44.52 | SupaYoshi | <_< yeah ok mister . let me just increase the amount of lines I can scroll back |
| 02:45.24 | outtolunc | thats the first thing i change |
| 02:46.58 | SupaYoshi | http://paste.ubuntu.com/6440740/ |
| 02:47.44 | *** join/#asterisk eharris (~eharris@99-179-7-82.lightspeed.austtx.sbcglobal.net) |
| 02:48.55 | *** join/#asterisk atcmmi (~atcmmi@119.122.155.152) |
| 02:49.05 | [TK]D-Fender | Reliably Transmitting (NAT) to 77.72.169.134:5060: REGISTER sip:sip.voipbuster.com SIP/2.0 Contact: <sip:0031857840891@10.15.1.101:5060> |
| 02:49.27 | [TK]D-Fender | You are registering to voipbuster and we can clearly see that you have NOT set your server up properly to work from behind NAT |
| 02:49.36 | [TK]D-Fender | it is telling them your PRIVATE IP as the return point |
| 02:50.00 | SupaYoshi | same errors gain |
| 02:50.15 | SupaYoshi | ah. |
| 02:50.54 | SupaYoshi | Well im very sorry, its just that it has always been working and I havent changed anything.. and em yeah after I changed my router and address pool its doing this =/ |
| 02:51.37 | SupaYoshi | im going to check my trunks |
| 02:51.49 | [TK]D-Fender | This is your server... you should be looking at where you told it to look for your WAN IP |
| 02:51.53 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 02:51.56 | Penguin | I always like how people say things like, "It worked before, so why do I need to set it up correctly now?" Well, because it is wrong and it doesn't work now. |
| 02:51.57 | [TK]D-Fender | and no, this has nothing to do with your trunks |
| 02:52.03 | [TK]D-Fender | this is SERVER SETTINGS |
| 02:52.09 | *** join/#asterisk bipul (~bipul@unaffiliated/bipul) |
| 02:52.16 | [TK]D-Fender | Your [general] section of sip.conf is out of whack |
| 02:52.31 | [TK]D-Fender | You didn't specify your locanets or externaddr properly |
| 02:52.37 | [TK]D-Fender | localnet* |
| 02:53.00 | Penguin | ~sipnat |
| 02:53.00 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the wiki at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet. |
| 02:53.18 | [TK]D-Fender | [21:27]SupaYoshiI opened up port 5060 UDP and TCP <- this also isn't enough.. that is only SIP... not RTP which carries the actual voice. |
| 02:53.42 | Penguin | He wanted to forward port 5004 for RTP. |
| 02:53.53 | SupaYoshi | i know. |
| 02:54.00 | outtolunc | just 1 |
| 02:54.06 | Penguin | Just 5004. |
| 02:54.11 | outtolunc | ;) |
| 02:54.14 | SupaYoshi | Hold on, I forgot about the setting that was there in asterisk.. I had to change that from the old IP to the new IP |
| 02:54.15 | Penguin | He didn't even bother to look at rtp.conf. |
| 02:54.17 | SupaYoshi | manully... |
| 02:55.32 | Penguin | If your IP address changes, consider externhost and a dynamic DNS host name. |
| 02:55.44 | *** join/#asterisk suneye (~atcmmi@116.7.102.180) |
| 02:56.35 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.53) |
| 02:56.49 | SupaYoshi | thnx |
| 02:57.03 | SupaYoshi | I got a dynamic address that doesnt change (lucky enough) |
| 02:57.29 | Penguin | Doesn't change, or doesn't change very often? |
| 02:58.03 | [TK]D-Fender | Latter, otherwise it isn't dynamic |
| 02:58.26 | SupaYoshi | My ISP sets it stuck for atleast 5 yrs |
| 02:58.30 | SupaYoshi | from my experience. |
| 02:59.18 | [TK]D-Fender | Anyway, go fix your settings |
| 02:59.38 | SupaYoshi | im trying :) |
| 02:59.42 | Penguin | [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet |
| 03:01.41 | SupaYoshi | mweh i will ask freepbx, im actually not supposed to ask here |
| 03:01.47 | SupaYoshi | Im using that =/ cus its easier. |
| 03:01.53 | SupaYoshi | (i thought) |
| 03:02.00 | Penguin | You thought incorrectly. |
| 03:02.19 | Penguin | It seems to be a common misconception. |
| 03:04.05 | SupaYoshi | gui = fail |
| 03:05.06 | [TK]D-Fender | SupaYoshi: "Asterisk SIP Settings" <- install the module. Actually fill in the blanks. |
| 03:06.00 | SupaYoshi | i got it >< filling it in but it doesnt work =/ blah |
| 03:06.44 | [TK]D-Fender | SupaYoshi: Next lesson : SHOW US |
| 03:07.19 | [TK]D-Fender | SupaYoshi: Don't say "it doesn't work" SHOW us what happens when you fill it it... and the actual screen-shots of what you've filled in ... and testing for hostnames at OS CLI |
| 03:07.27 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 03:07.27 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 03:09.30 | SupaYoshi | http://prntscr.com/2563dy |
| 03:12.02 | Penguin | Curious about that settings page... |
| 03:12.11 | SupaYoshi | ok calling work again |
| 03:12.14 | Penguin | How can you have NAT and a public IP address? |
| 03:12.41 | SupaYoshi | em |
| 03:13.12 | SupaYoshi | ah wait.. yeah |
| 03:13.46 | SupaYoshi | i did that beacuse it wasnt working xD nvm fixing it |
| 03:14.11 | Penguin | I don't understand the settings page. I'm asking you. |
| 03:14.13 | SupaYoshi | em no |
| 03:14.20 | *** join/#asterisk DrkShadow (~Jack@host-72-175-240-62.static.bresnan.net) |
| 03:14.25 | SupaYoshi | well the weird thing is atm... |
| 03:14.32 | Penguin | I understand NAT = yes, no, never, and route |
| 03:14.35 | SupaYoshi | If i fill in it the way i am supposed to. it doesnt work somehow.. |
| 03:14.51 | DrkShadow | In Asterisk 1.4 I had it set up to log the whole execution of the dialplan to the full log.. in Elastix, 1.6 or 1.8, I think, I can't figure this out. How can I do it? |
| 03:14.55 | Penguin | If you mark nat yes, why is there the option for a public IP address? |
| 03:15.34 | SupaYoshi | https://secure.join.me/199-668-313 |
| 03:15.37 | SupaYoshi | click and i can show ya :) |
| 03:15.54 | Penguin | I understand that if you use the static IP setting, it will write externaddr to the settings file. I understand that if you use the dynamic IP setting that it will write exterhost to the file. |
| 03:16.37 | SupaYoshi | yes but look it is theyre setup. |
| 03:17.48 | Penguin | What if you mark the public IP button? |
| 03:19.35 | SupaYoshi | then the boxes dissapear :) |
| 03:19.39 | Penguin | I see that. |
| 03:19.48 | Penguin | But the NAT setting is still there. |
| 03:20.01 | SupaYoshi | yeah |
| 03:20.03 | SupaYoshi | it stays there |
| 03:20.06 | Penguin | So my question is: how do you have NAT *and* a public IP address? |
| 03:20.33 | SupaYoshi | :P I got no clue atm.. I also think I have it setup incorrectly atm |
| 03:20.36 | SupaYoshi | quite sure. |
| 03:21.13 | Penguin | Set it to nat yes, static IP, and put in your externaddr in the box, and put in your localnet into the other boxes. |
| 03:21.50 | SupaYoshi | yeah it works again :p |
| 03:21.54 | SupaYoshi | lol |
| 03:22.00 | SupaYoshi | This is my problem. |
| 03:22.07 | SupaYoshi | I have a modem from my ISP that cannot be bridged. |
| 03:22.13 | Penguin | ick! |
| 03:22.16 | Penguin | That's a problem. |
| 03:22.32 | SupaYoshi | So I have: Experiabox (MODEM) -> Router -> PBX |
| 03:22.48 | Penguin | You have two NAT layers, then. |
| 03:23.07 | SupaYoshi | So... since DHCP cannot be put off on the Experiabox... I have it DMZ'ed to the router... and the router gets a address from the Experiabox. |
| 03:23.33 | Penguin | DMZ is always the wrong answer. |
| 03:23.35 | SupaYoshi | Yeah.. exactly so that kind of sucks... so that made me confused, so now ive set the IP to 192.168.2.1 which is the WAN address of my router... |
| 03:23.57 | SupaYoshi | DMZ is the right answer when every other option is killed |
| 03:24.06 | Penguin | I disagree. |
| 03:24.14 | SupaYoshi | :P What would you do then? |
| 03:24.20 | SupaYoshi | Add a route to my ISP modem? :D |
| 03:24.41 | Penguin | I'd get the modem to be bridged. |
| 03:24.59 | SupaYoshi | :P not possible thanks to my lovely ISP |
| 03:25.19 | SupaYoshi | where ma dad works for, (lucky enough not for that part of the company) |
| 03:25.37 | SupaYoshi | basically i have an extra HOP |
| 03:25.53 | SupaYoshi | but it works again, so im glad :P lol |
| 03:26.18 | SupaYoshi | one question, why would i portforward ports? if it works as is? |
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| 05:14.35 | runfromnowhere | Does anyone have experience with Asterisk 11 MixMonitor recorded calls getting out of sync? |
| 05:17.04 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.44) |
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| 05:32.44 | DrkShadow | In Asterisk 1.4 I had it set up to log the whole execution of the dialplan to the full log.. in Elastix, 1.6 or 1.8, I think, I can't figure this out. How can I do it? |
| 05:33.09 | [TK]D-Fender | logger.conf <- |
| 05:33.16 | [TK]D-Fender | same place as always |
| 05:33.20 | Penguin | We don't support Elastix, but asterisk handles it in logger.conf. |
| 05:40.15 | DrkShadow | Would you care to elaborate? I have full => notice,warning,error,verbose -- and the log file full does not contain the dial plan traversal informamtion. |
| 05:41.18 | DrkShadow | Sorry, looking at the wrong server. The one I want the dialplan traversal information has full => notice,warning,error,debug,verbose |
| 05:41.38 | [TK]D-Fender | And what verbose level is it running at? |
| 05:42.05 | DrkShadow | how can I determine that? |
| 05:42.35 | DrkShadow | It's not set in any configuration file. |
| 05:42.52 | [TK]D-Fender | go set it at CLI |
| 05:43.05 | Penguin | When you connect to the CLI, it will tell you the verbose level. |
| 05:43.09 | DrkShadow | If it's set in the CLI then it's set to 9 |
| 05:43.15 | Penguin | E.g., Verbosity is at least 3 |
| 05:43.16 | [TK]D-Fender | dgo do it |
| 05:43.21 | DrkShadow | as that's the last I set. But it doesn't echo when I start the CLI. |
| 05:43.28 | Penguin | Then it must be 0. |
| 05:43.49 | DrkShadow | set 9, exited, started cLI, didn't echo verbosity. |
| 05:44.01 | Penguin | What version is this? |
| 05:44.06 | DrkShadow | Asterisk 11.5.0 |
| 05:44.07 | [TK]D-Fender | And what did it say when you set it? |
| 05:44.23 | DrkShadow | sip*CLI> core set verbose 9; Set remote console verbosity to 9 |
| 05:45.40 | DrkShadow | so... |
| 05:46.13 | [TK]D-Fender | go test it now |
| 05:47.04 | DrkShadow | sip*CLI> core set verbose 9; Set remote console verbosity to 9; sip*CLI> exit; Asterisk cleanly ending (0).; Executing last minute cleanups; [root@sip asterisk]# asterisk -r; Asterisk 11.5.0, Copyright (C) 1999 - 2012 Digium, Inc. and others.; Created by Mark Spencer ; markster@digium.com>; Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.; This is free |
| 05:47.04 | DrkShadow | software, with components licensed under the GNU General Public; License version 2 and other licenses; you are welcome to redistribute it under; certain conditions. Type 'core show license' for details.; =========================================================================; Connected to Asterisk 11.5.0 currently running on sip (pid = 2785) |
| 05:47.57 | Penguin | Huh. I just realized that I'm not sure I have ever typed exit to leave the CLI. |
| 05:48.17 | DrkShadow | quit works too. |
| 05:48.25 | Penguin | Never done that, either. |
| 05:48.44 | DrkShadow | I'm afraid ctrl-d would kill my shell |
| 05:48.53 | Penguin | Ctrl+c is less keystrokes. I prefer pushing less buttons. |
| 05:49.04 | DrkShadow | so anyway... |
| 05:49.19 | Penguin | I don't think ctrl+d will do anything. |
| 05:49.30 | Penguin | Nope, it doesn't. |
| 05:49.38 | Penguin | So that's interesting. |
| 05:50.06 | Penguin | I'm going to tell you now that I have not used 11, so I don't know its quirks. |
| 05:50.28 | Penguin | 1.8 shows the verbose level when connecting, but that doesn't seem relevant at this point. |
| 05:53.31 | DrkShadow | I can set verbosity via init script. I'll do that. |
| 05:53.47 | Penguin | You can also set it in a config file, which seems to make more sense. |
| 05:53.55 | DrkShadow | Thanks for that. |
| 05:53.57 | DrkShadow | where? |
| 05:54.01 | DrkShadow | I haven't seen any config file setting. |
| 05:54.09 | Penguin | asterisk.conf |
| 06:00.15 | DrkShadow | Thanks, will wait and see what happens. |
| 06:13.49 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 06:16.24 | drmessano | Hmmm |
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| 10:05.30 | linocisco | hi all |
| 10:05.41 | linocisco | skype for asterisk is not free |
| 10:05.44 | linocisco | which one is free? |
| 10:06.08 | Chainsaw | Skype for Asterisk doesn't exist anymore. |
| 10:06.13 | Chainsaw | So even if you were willing to pay for it, it wouldn't help. |
| 10:08.07 | linocisco | Chainsaw, so what should I use? |
| 10:08.20 | Chainsaw | linocisco: Not Skype? |
| 10:08.39 | linocisco | Chainsaw, ok. if skype doesn't work, what else should I use? |
| 10:08.50 | Chainsaw | linocisco: VoIP. That's what Asterisk is designed for. |
| 10:09.14 | Chainsaw | linocisco: There's all sorts of cool stuff you can do with WebRTC if you don't want people to have to install clients. |
| 10:09.40 | linocisco | Chainsaw, I would like to enable visitors on my website click to call button on my website that should reach my pbx and ring at reception |
| 10:10.09 | Chainsaw | linocisco: Yes, WebRTC. |
| 10:10.54 | linocisco | Chainsaw, is there any documentation on how to implement or integrate WebRTC with asterisk to accomplish my goal.? any useful URLs? |
| 10:11.33 | Chainsaw | linocisco: https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support |
| 10:12.00 | linocisco | Chainsaw, Tango Yankee! |
| 10:12.19 | Chainsaw | linocisco: Yankee Whiskey. |
| 10:12.41 | linocisco | Chainsaw, out |
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| 10:35.38 | linocisco | Chainsaw, webRTC doesn't work on IE |
| 10:36.12 | Chainsaw | linocisco: People still use that? |
| 10:36.42 | linocisco | Chainsaw, yes. should support on all browsers as all websites can be run on all browsers |
| 10:37.06 | Chainsaw | linocisco: Could have an ActiveX WebRTC control for that. |
| 10:37.56 | linocisco | Chainsaw, dont know how to |
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| 11:23.10 | ocholetras | hi guys! |
| 11:24.04 | ocholetras | im starting with asterisk and have to replicate an asterisk install, and im refactoring all because the existing installation uses a lot of macros |
| 11:24.10 | ocholetras | i have seen that macros are deprecated |
| 11:24.18 | ocholetras | should i go by goSub? |
| 11:24.27 | ocholetras | because i have found this https://wiki.asterisk.org/wiki/display/AST/app_macro+Deprecation |
| 11:24.35 | ocholetras | and looks a little tricky |
| 11:24.38 | ocholetras | right? |
| 11:29.39 | ocholetras | wow! not all at once ^^ |
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| 13:01.17 | danfromuk | Hi, when transfering calls to a queue, userfield doesnt seem to be copied over. Is there any way to force this? |
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| 13:02.02 | ocholetras | what you mean with user field? |
| 13:02.06 | ocholetras | src ext? |
| 13:02.18 | danfromuk | userfield is a cdr field |
| 13:02.40 | danfromuk | CDR(userfield) |
| 13:03.42 | Geek-Linux | Hi, all do any body have mtp3d experience ? |
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| 13:42.36 | zafu | hi, when a sip client forwards a call with a "302 moved" is there a way to know which extension orginated the forwarding? |
| 13:43.05 | zafu | as the "new" call is a Local/* channel with no reference to the sip client |
| 13:43.56 | WIMPy | Extensions don't originate calls. They are destinations. |
| 13:44.23 | zafu | I meant "which sip client" |
| 13:44.37 | WIMPy | At least while ringing you will only have the CALLERID and FORWARDING functions. |
| 13:45.28 | zafu | functions or variables? |
| 13:45.47 | WIMPy | functions |
| 13:46.39 | *** join/#asterisk Merlin (merlin@omni.gcinfotech.com) |
| 13:47.04 | Merlin | hi all, can someone point me to a perl script or documentation on how to read the time/date stamps in polycom log files please? |
| 13:47.31 | zafu | WIMPy: can't find a reference to that function |
| 13:47.51 | WIMPy | 'core show function CALLERID' |
| 13:48.18 | WIMPy | Oh, sorry. The other one is REDIRECTING. |
| 13:48.25 | zafu | ah :) |
| 13:51.55 | zafu | how does one use that function? |
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| 13:56.30 | WIMPy | Like any other function :-) 'core show apoplication REDIRECTING' will give you the list of options. |
| 13:56.46 | WIMPy | s/apoplication/function/ |
| 13:58.04 | [TK]D-Fender | Merlin: 1 - absolute time with minute resolution 0210281716 -- 2002 October 28, 17:16 |
| 13:58.05 | [TK]D-Fender | Merlin: 2 - absolute time with seconds resolution 1028171642 -- October 28, 17:16:42 |
| 13:58.10 | [TK]D-Fender | Merlin: One of those |
| 13:58.46 | Merlin | oh interesting |
| 13:58.48 | Merlin | thank you |
| 13:59.04 | [TK]D-Fender | Merlin: BTW it was in the Admin Guide .... you should really go download it... |
| 13:59.26 | Merlin | i swear i searched that |
| 13:59.28 | Merlin | over and over again |
| 13:59.34 | Merlin | i guess i didn't know what to search for |
| 13:59.58 | [TK]D-Fender | "stamp" would have gotten it |
| 14:00.11 | [TK]D-Fender | Certainly did for me. |
| 14:00.41 | Merlin | interesting |
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| 14:09.59 | Merlin | ok, i found it now |
| 14:10.00 | Merlin | thank you |
| 14:14.52 | [TK]D-Fender | Merlin: You're welcome |
| 14:15.11 | *** join/#asterisk dar123 (~DAR@77.30.146.69) |
| 14:15.56 | dar123 | which echo canceller is recommended i am setting up my home pc for asterisk |
| 14:16.43 | [TK]D-Fender | darthat depends on your hardware |
| 14:16.47 | [TK]D-Fender | dar123: that depends on your hardware |
| 14:16.58 | [TK]D-Fender | dar123: What do you have? |
| 14:16.58 | WIMPy | There seem to be different experiences. Do you need one at all? |
| 14:17.45 | dar123 | i have TDM410 from digium |
| 14:18.10 | dar123 | 1fxs + 1fxo port |
| 14:19.02 | [TK]D-Fender | dar123: Is the default not cutting it? |
| 14:19.21 | [TK]D-Fender | dar123: by that I mean the one in the sample config which is mg2 |
| 14:19.48 | [TK]D-Fender | darotherwise there are 2-3 others included and then time to move up to OSLEC if none of those do |
| 14:26.55 | dar123 | i havent tried that, just following the instructions step by step |
| 14:27.15 | dar123 | i will go ahead with mg2, will see if i have any problems |
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| 14:27.45 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 14:34.04 | dar123 | as per the doc i am following it says that driver for my card is wcaxx |
| 14:34.34 | dar123 | when i "modprobe wcaxx", i get FATAL: Module wcaxx not found. |
| 14:35.18 | [TK]D-Fender | Never heard of that module name before... |
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| 14:36.23 | [TK]D-Fender | Should be "wctdm" |
| 14:39.14 | kaldemar | the digium's user manual really says wcaxx. that's odd. |
| 14:40.00 | dar123 | yeah it does |
| 14:40.20 | kaldemar | dahdi does not have such a module. |
| 14:40.30 | dar123 | is there a way from command line to know on which channel i have fxo and on which fxs |
| 14:40.37 | dar123 | Channel map: |
| 14:40.37 | dar123 | Channel 03: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 03) |
| 14:40.37 | dar123 | Channel 04: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 04) |
| 14:40.37 | dar123 | 2 channels to configure. |
| 14:40.37 | dar123 | Changing signalling on channel 3 from FXS Kewlstart to FXO Kewlstart |
| 14:40.38 | dar123 | DAHDI_CHANCONFIG failed on channel 3: Invalid argument (22) |
| 14:40.39 | dar123 | Selected signaling not supported |
| 14:40.42 | dar123 | Possible causes: |
| 14:40.44 | dar123 | <PROTECTED> |
| 14:40.46 | dar123 | <PROTECTED> |
| 14:40.47 | dar123 | <PROTECTED> |
| 14:40.58 | dar123 | oops i am glad i am not kicked or banned ... sorry for this |
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| 14:44.30 | [TK]D-Fender | Apology came before the kick, all is good :) |
| 14:44.38 | [TK]D-Fender | hopefully you'll be more careful |
| 14:45.03 | [TK]D-Fender | Sounds like you got your modules backwards |
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| 14:48.08 | dar123 | i checked on the card carefully and from the docs definately i have the wrong information |
| 14:49.10 | dar123 | is there a way to validate the information from the machine instead of physically looking at the card |
| 14:49.47 | Chainsaw | dar123: With a bit of luck it is in dmesg output. Do you have a /var/log/dmesg at all? |
| 14:52.12 | protocoldoug | Does anyone know why a B() or b() dial option, when calling a subroutine WILL set a variable within the sub, but... With a G() or a U() dial option, I can read an infinitely inheritable variable, but, I cannot set it? |
| 14:52.15 | protocoldoug | example: http://pasteall.org/47412 |
| 14:53.01 | protocoldoug | if you run that example as is, __TESTVARIABLE will not get set again, using a U() dial option, however, with a B() dial option, it will get set -- both using the same subroutine |
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| 14:55.06 | protocoldoug | (I'm running Asterisk 11.2-cert2) |
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| 15:06.10 | dar123 | no luck from the dmesg log |
| 15:06.47 | dar123 | can't find anything that can be helpful |
| 15:06.56 | *** part/#asterisk ocholetras (~alorenzo@taburiente.ccti.ull.es) |
| 15:07.17 | Chainsaw | dar123: I would prefer it if you could make that available to me using pastebin. |
| 15:07.28 | Chainsaw | dar123: dmesg &> kernel.txt |
| 15:07.32 | Chainsaw | dar123: And then get me that kernel.txt |
| 15:08.21 | dar123 | okies |
| 15:08.46 | kaldemar | protocoldoug: U() executes the gosub for the called channel, B() for the calling channel. you defined the variable for the calling channel. |
| 15:09.42 | kaldemar | protocoldoug: if you use U() to set a variable, set it with func MASTER_CHANNEL |
| 15:10.17 | [TK]D-Fender | protocoldoug: IIRC morst variables die as you reach "h" so that shouldn't be a valid test. next you're answering and killing a channel almost instantly there.. you should ad a delay. |
| 15:11.11 | [TK]D-Fender | protocoldoug: Next if you're throwing in another layer like that you might want to set the __ version in your local channel before the gosub, etc |
| 15:11.19 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:11.19 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:11.33 | [TK]D-Fender | protocoldoug: And if using a gosub you can pass it as a PARAMETER anyway |
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| 15:16.52 | dar123 | http://pastebin.ca/2477263 |
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| 15:20.21 | Chainsaw | dar123: Right, so your FXO is in slot 3 (FXS signalling please). And your FXS is in slot 4 (FXO signalling please). |
| 15:20.58 | Chainsaw | dar123: You are set for North American signalling, if that is not where you are then you will need to adjust your "tone zone". |
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| 15:30.26 | dar123 | Chainsaw: Thanks, this is exactly how i have configured |
| 15:30.54 | dar123 | Channel 03: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 03) |
| 15:30.54 | dar123 | Channel 04: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 04) |
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| 15:32.10 | dar123 | http://pastebin.ca/2477266 |
| 15:32.36 | Chainsaw | dar123: No, your signalling is the wrong way round. Read what I said again. |
| 15:33.03 | Chainsaw | <Chainsaw> dar123: Right, so your FXO is in slot 3 (FXS signalling please). And your FXS is in slot 4 (FXO signalling please). |
| 15:33.43 | Chainsaw | dar123: I understand that this inverse relationship between signalling & port types may be confusing at first. But please, do it the way I suggested. It will work a lot better. |
| 15:35.39 | Weezey | When I'm authenticated to an asterisk box with my phone, and show up as a SIP peer, then make a call out to a different asterisk box and then my call comes back to the first one where I'm authenticated, the first box is handing back 401 Unauthorized even though I should be coming back to the first box as the peer for the second one, not my local phone. |
| 15:35.47 | Weezey | is that a directmedia=no problem? |
| 15:35.52 | protocoldoug | [TK]D-Fender: ah ha! Thank you, now if I output the channel inside the sub, I see that it's for the other channel! Awesome. What I wound up doing was using a SHARED(varable)=value, and then passing the original channel into the sub as an argument |
| 15:36.14 | protocoldoug | really appreciate the eyeballs, and thank you too kaldemar (I couldn't get MASTER_CHANNEL to work for, somehow) |
| 15:37.24 | dar123 | i wasn't able to find saudi arabia in zonedata.c, thats why left it to default |
| 15:37.39 | Chainsaw | dar123: I really doubt you have North American signalling. |
| 15:42.26 | dar123 | not sure which one to select, did a lot of search before on google too |
| 21:33.28 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 21:33.28 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.6.0 (2013/10/21), 10.12.3 (2013/08/27), 1.8.24.0 (2013/10/21), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 21:34.11 | navaismo | ~late |
| 21:34.23 | navaismo | ¬¬ |
| 21:34.26 | jansiva | ok |
| 21:35.06 | jansiva | i just discussed with my team mate he says when we has issue earlier he made change http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_and_Asterisk |
| 21:35.47 | jansiva | it stopped anonymous call ringing |
| 21:37.15 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 21:41.23 | navaismo | basically what we told you all the morning allowguest=no!!! |
| 21:42.06 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.99) |
| 21:45.36 | boom^time | Hey guys, what's the best way to set a cdr variable on a cdr record that has been forked? |
| 21:45.53 | boom^time | When I try it sets the variable for the record prior to the fork. Not sure why. |
| 21:46.49 | *** join/#asterisk grummund (~user@unaffiliated/grummund) |
| 21:47.08 | jansiva | navaismo: how do i test it is working or not |
| 21:48.33 | jansiva | i mean test from 1001 exten etc |
| 21:48.37 | jansiva | how to do |
| 21:55.57 | *** join/#asterisk jknox (~jknox@act.douglasfast.net) |
| 21:56.58 | jknox | Hey guys getting a compile error with Asterisk-12-beta-1 |
| 21:57.25 | jknox | http://pastebin.com/wnE31EPX |
| 21:57.34 | jknox | Has anyone encountered this? |
| 21:59.05 | *** join/#asterisk jpoz_ (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 21:59.46 | slicknick5181 | Jansiva, Download a sip softphone on a pc and try to dail your server ip addres |
| 22:00.18 | slicknick5181 | Jansiva, SJ Phone is perfect for such occasion |
| 22:00.58 | amessina | Hello. I am having trouble with Asterisk-12-beta1 placing calls through an outbound trunk that requires the "fromuser" parameter. I have fromuser defined in pjsip.conf, but looking at the logs, the fromuser is not prepended to the "From:" header. |
| 22:02.03 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-afdbwxiazjjisvnp) |
| 22:02.03 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 22:02.35 | dar123 | need help guyz http://pastebin.ca/2477349 |
| 22:03.03 | lordvadr | Using AMI, if I'm trying to cross reference messages between events, is Channel sufficient to use, or should I be using UniqueID? |
| 22:03.58 | *** part/#asterisk slicknick5181 (~slick@207-255-114-073-dhcp.unt.pa.atlanticbb.net) |
| 22:04.29 | mjordan | lordvadr: depends. In versions of Asterisk prior to 12, channel names can change as a result of Masquerades (although you'll be informed via a sequence of Rename events). |
| 22:04.51 | mjordan | lordvadr: what messages are you trying to tie together? |
| 22:05.17 | lordvadr | mjordan: I just need to map a NewChannel and a DAHDIChannel together. I get direction from NewChannel, and span and timeslot from DAHDIChannel. |
| 22:06.00 | jknox | When installing asterisk 12 do I need to disable chan_sip when enabling chan_pjsip |
| 22:06.14 | mjordan | jknox: no, but you don't want them binding on the same addr/port |
| 22:06.15 | lordvadr | Seems like I could use either, but if any sort of renaming were to occur, Uniqueid would be the best one to use? Those events come in back to back so I don't know if its even possible to rename the channel between them. |
| 22:06.24 | mjordan | lordvadr: uniqueid would be better in that case |
| 22:06.34 | mjordan | and no, in general, it would not be possible to rename between them |
| 22:06.42 | lordvadr | mjordan: I figured as much. Thank you. |
| 22:07.23 | mjordan | jknox: so long as you have chan_sip bind to one port (say, 5061) and res_pjsip bind to another (say, 5060), the two can run concurrently |
| 22:07.47 | jknox | mjordan: Im getting compile errors and was hoping that was the issue. When you get a chance check this out and let me know your thoughts: res_pjsip/pjsip_distributor.c:137: error: ‘pjsip_transaction’ has no member named ‘mutex’ |
| 22:08.29 | mjordan | jknox: hm... what version of 12? |
| 22:08.33 | jknox | woops http://pastebin.com/wnE31EPX |
| 22:08.43 | jknox | <PROTECTED> |
| 22:08.53 | mjordan | I think this might have just gotten fixed in head of 12 |
| 22:09.00 | jknox | ok |
| 22:09.00 | mjordan | do you mind trying from the source branch? |
| 22:09.07 | jknox | Ill do it now |
| 22:11.43 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
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| 22:17.52 | newtonr | amessina, try with latest SVN of 12, if that still doesn't work, post a JIRA issue including an Asterisk log with "pjsip set logger on" and a pcap |
| 22:19.04 | amessina | newtonr: actually, I am using branches/12@402864 |
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| 22:21.41 | mjordan | amessina: sorry, I'm late to the party: what issue were you seeing? |
| 22:23.26 | file | fromuser not working |
| 22:24.35 | amessina | mjordan: I have a sip trunk that I'm trying to get to work with PJSIP. Their configuration (for Asterisk 11 and below) required the use of the "fromuser" parameter. Using branches/12@402864 with the fromuser parameter defined for the endpoint, calls are rejected |
| 22:24.48 | amessina | and the From line looks like: From: <sip:50.XX.XX.XX>;tag=2e8f576f-45b3-4035-bf5e-357a524ea773 |
| 22:25.02 | newtonr | amessina, can you pastebin your sanitized pjsip.conf ? |
| 22:27.25 | amessina | newtonr: mjordan: http://ur1.ca/g2chk |
| 22:28.11 | amessina | For testing, I added the fromdomain=mydomain.com parameter, and that gets put in properly: From: <sip:mydomain.com>;tag=2e8f576f-45b3-4035-bf5e-357a524ea773 |
| 22:30.54 | file | erm, what happens if you originate a call out the trunk from the CLI? |
| 22:31.08 | amessina | file, are you asking me? |
| 22:31.14 | file | yes |
| 22:31.19 | amessina | ok lemme try |
| 22:31.22 | jknox | mjordan: that did it! thanks! |
| 22:33.26 | mjordan | jknox: good to hear :-) |
| 22:35.18 | dar123 | FXO signaling is being used on a FXO interface (use a FXS signaling variant) |
| 22:36.14 | dar123 | <PROTECTED> |
| 22:36.35 | dar123 | i have loop start configured for fxo |
| 22:37.53 | amessina | file: from the cli, it seems to work: From: <sip:42925@50.196.241.75>;tag=54f1267e-3013-42af-a3b3-e2d12591e1b8 |
| 22:40.30 | [TK]D-Fender | Daran FXO interface uses FXS signaling. IT's the reverse |
| 22:41.13 | [TK]D-Fender | dar123: When I have an FXO interface (to interface with a Foreign eXchange Office), I would have to act as an FXS (Foreign eXchange Station) |
| 22:43.11 | WIMPy | Probably because you don't have a FXO interface, but an interface to an FXO? |
| 22:43.18 | WIMPy | Analog is complicated. |
| 22:43.49 | lordvadr | Using AMI, and having a call come in on one DAHDI span, and then go out another DAHDI span, if I set CDR(userfield) on the "inbound" channel, and then call Getvar on the outbound channel, it doesn't show up. This isn't entirely unexpected, but that call will have only one CDR entry. If I know the outbound channel, do I have to run MASTER_CHANNEL on the outbound to get at the actual CDR data? |
| 22:43.52 | dar123 | sorry i am not getting this concept. My understanding is FXO expects dialtone and FXS generates |
| 22:44.03 | amessina | file: The problem is that I used to be able to set CALLERID(num)=NXXNXXXXXX even with fromuser and the CALLERID was passed through to the trunk. |
| 22:44.05 | mjordan | lordvadr: what version? |
| 22:44.10 | lordvadr | 1.8.7 |
| 22:44.10 | dar123 | can you plz refer me a link to understand this concept |
| 22:44.26 | file | amessina, you'd have to pass that as rpid or pai... |
| 22:44.59 | mjordan | lordvadr: yes |
| 22:45.17 | mjordan | lordvadr: your other option would be, if using Asterisk 11, to use a pre-dial handler to set the userfield on the outbound channel |
| 22:45.27 | mjordan | but Asterisk 1.8 doesn't have that feature |
| 22:45.38 | amessina | file: right. In asterisk 11: sendrpid=yes, fromuser=12345 and in Dialplan CALLERID(num)=NXXNXXXXXX works |
| 22:46.55 | file | it should also work in PJSIP land, if not then that's a bug |
| 22:46.58 | lordvadr | mjordan: Essentially I want to log span and timeslot of my callls in userfield, and I want two tags if it came in on one and went out on another. My only option to get span and timeslot on the outbound is to use AMI. |
| 22:47.31 | lordvadr | mjorday: What happens if I set the CDR(userfield) on both to something different? Will the inbound channel always win or will it get overwritten by what I set on the outbound channel? |
| 22:48.13 | lordvadr | And is there a good way to know if MASTER_CHANNEL will return something without checking it? |
| 22:48.56 | amessina | in asterisk 12: send_rpid=yes, fromuser=12345 and in Dialplan CALLERID(num)=NXXNXXXXXX doesn't work (the <from>@<domain> is set to NXXNXXXXXX@<domain> |
| 22:49.46 | file | aye the callerid module shouldn't set it if fromuser is configured, that's the bug |
| 22:51.32 | file | I'm eating dinner so I can't fix it immediately |
| 22:51.41 | mjordan | lordvadr: with userfield, and userfield only, the two fields are appended with a ';' |
| 22:51.46 | mjordan | if you set it on both channels |
| 22:52.28 | mjordan | in general, unless you see it noted otherwise, values on the inbound channel are logged to the CDR. There are exceptions, for example, peeraccount (or other 'peer' fields) |
| 22:52.57 | mjordan | and of course, sometimes outbound channels become inbound channels (which is why CDRs are goofy) |
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| 22:53.02 | mjordan | so things can get murky |
| 22:53.14 | amessina | file: ok |
| 22:53.26 | lordvadr | mjordan: That's fantastic. I couldn't ask for anything better. |
| 22:53.44 | mjordan | amessina: in case file gets amnesia due to a food coma, could you file an issue? I'd hate for that to get dropped |
| 22:53.52 | lordvadr | mjordan: Is that the same in all versions of asterisk going forward? |
| 22:54.04 | mjordan | lordvadr: yup, that behavior was even preserved in the great CDR migration of 2013 |
| 22:54.13 | mjordan | (aka: Asterisk 12) |
| 22:55.07 | lordvadr | What happened to CDR in Asterisk 12? |
| 22:55.21 | amessina | mjordan: ok lemme put it all together well enough for a ticket |
| 22:55.23 | file | mjordan rewrote it |
| 22:56.02 | outtolunc | FUNC_CDR |
| 22:57.19 | newtonr | amessina, thanks! |
| 22:57.45 | newtonr | amessina, feel free to copy paste the IRC chat and attach as a .txt on the issue as well |
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| 22:59.37 | amessina | newtonr: ok thanks |
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| 23:07.33 | navaismo | is realtime supported by chan_pjsip? |
| 23:08.39 | file | yes. |
| 23:09.02 | karl-s | is pjsip the new sip stack? how does it differ? |
| 23:09.37 | file | it's the new one, and it differs... a lot |
| 23:09.40 | [TK]D-Fender | Completely new stack |
| 23:09.44 | [TK]D-Fender | no config files. |
| 23:09.46 | [TK]D-Fender | new rules |
| 23:09.48 | [TK]D-Fender | new ups |
| 23:09.50 | [TK]D-Fender | new downs. |
| 23:09.56 | karl-s | no config files? |
| 23:10.01 | [TK]D-Fender | Don't even ASK about "slightly to the left" :p |
| 23:10.04 | [TK]D-Fender | new* |
| 23:10.08 | karl-s | ah ok :{ |
| 23:10.13 | karl-s | world of difference :) |
| 23:10.53 | karl-s | I guess what i want to know is whether there are any new compelling features... |
| 23:12.21 | file | we focused on getting as close to parity as we could first |
| 23:13.04 | karl-s | is this what "codename pinapple" was all about a few years ago? |
| 23:14.25 | file | nope |
| 23:14.35 | file | chan_pjsip is not based on chan_sip at all |
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| 23:17.02 | lordvadr | mjordan: Just FYI, that userfield concatination doesn't work. If the call is answered, the destination channel's userfield is the only thing that makes it into CDR, if it's not answered, the source channel is the only thing that makes it in there. |
| 23:17.58 | file | amessina, I fixed it |
| 23:18.20 | amessina | file: geez, that was fast! I'm still putting together the ticket! |
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| 23:18.55 | newtonr | amessina, don't throw away your ticket yet, maybe SVN up, test again and verify it is fixed for you :D |
| 23:19.17 | file | I'd actually submit the ticket, I'll put it in the commit messages so we have a record why |
| 23:20.28 | navaismo | file is there a structure(sql file) of the pjsip channel, I ddint found anyone on the contrib directory |
| 23:20.49 | amessina | newtonr: I'll file the ticket and it can get marked as closed so at least its there |
| 23:20.58 | amessina | file: ASTERISK-22866 |
| 23:20.58 | LieutPants | [ASTERISK-22866] [Status: Triage] Asterisk 12 (branches/12@402864) doesn't correctly set the fromuser parameter - https://issues.asterisk.org/jira/browse/ASTERISK-22866 |
| 23:21.00 | mjordan | navaismo: alembic will generate the realtime schema for the res_pjsip stack |
| 23:21.14 | mjordan | if you're using a tagged tarball, that should be in the tag already for you |
| 23:21.18 | mjordan | beta1 should have it |
| 23:21.25 | file | amessina, merci |
| 23:21.31 | amessina | file: in the mean time, I'll rebuild with your fix and see what happens after dinner ;) |
| 23:21.40 | navaismo | oh I downloaded from trunk, let me check the tarball |
| 23:22.16 | mjordan | navaismo: you can actually generate the SQL files using alembic yourself, but when we make tarballs we generate the SQL scripts to create the tables for MySQL, Oracle, MS SqlServer, and PostgreSQL |
| 23:22.29 | mjordan | (so those folders that have been there since 1.2? are now actually all populated) |
| 23:23.10 | mjordan | nope, 1.4 |
| 23:23.25 | file | and just like magic, all done |
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| 23:24.38 | navaismo | oh |
| 23:27.25 | navaismo | and most important what is that thing called "alembic" |
| 23:28.07 | amessina | thanks file. BTW, what happened to "accountcode" for pjsip? |
| 23:28.37 | newtonr | navaismo, https://wiki.asterisk.org/wiki/display/AST/Managing+Realtime+Databases+with+Alembic |
| 23:28.57 | file | forgotten and lack of time I would answer that question with |
| 23:29.20 | file | easy to add, though |
| 23:29.48 | file | missing setvar as well |
| 23:29.59 | amessina | was there supposed to be some other way to set endpoints to accountcodes? |
| 23:30.21 | navaismo | uh that link is pure gold thanks newtonr |
| 23:30.28 | file | nope |
| 23:30.38 | newtonr | navaismo, thank mjordan , he wrote it :) |
| 23:30.50 | file | chan_pjsip isn't complete yet, so there is stuff missing |
| 23:32.12 | navaismo | i havent used asterisk12 since chan_gulp so im trying now |
| 23:38.54 | mjordan | amessina: you should be able to set the accountcode on the channel in the dialplan still, but of course, that's dialplan only. |
| 23:39.04 | amessina | mjordan: |
| 23:39.09 | mjordan | we really should just have a setvar, since you should be able to use setvar=CHANNEL(accountcode)=foo |
| 23:39.14 | file | mjordan, yeah. |
| 23:39.37 | amessina | mjordan: yes, i used to do the following for convenience: |
| 23:39.40 | amessina | same => n,ExecIf($["${EXTEN:0:1}" = "1"]?Set(CALLERID(num)=${CHANNEL(accountcode):1}):Set(CALLERID(num)=${CHANNEL(accountcode)})) |
| 23:40.53 | amessina | I suppose I could use the CALLERID(tag) purpose for this, I think. |
| 23:41.07 | amessina | I'm not sure how to retrieve that though. |
| 23:41.35 | amessina | But, I'll wait till file implements 'setvar' ;) |
| 23:42.05 | file | I'm actually creating an issue for that :P |
| 23:42.14 | amessina | file: https://messinet.com/koji/buildinfo?buildID=1551 Ok, i'm gonna give it a go! |
| 23:44.13 | file | braced myself for the goodbye, cause it's all I've ever known |
| 23:45.35 | file | ASTERISK-22868 |
| 23:45.36 | LieutPants | [ASTERISK-22868] [Status: Triage] chan_pjsip: 'setvar' should be supported on endpoints - https://issues.asterisk.org/jira/browse/ASTERISK-22868 |
| 23:47.20 | file | that sorta thing is more of a weekend issue |
| 23:47.30 | amessina | file: your fix for ASTERISK-22866 works like a champ |
| 23:47.31 | LieutPants | [ASTERISK-22866] [Status: Closed] Asterisk 12 (branches/12@402864) doesn't correctly set the fromuser parameter - https://issues.asterisk.org/jira/browse/ASTERISK-22866 |
| 23:47.39 | file | amessina, coolio |
| 23:47.47 | amessina | thanks a ton! |
| 23:47.52 | file | yw |
| 23:50.04 | newtonr | yay |
| 23:50.58 | navaismo | omg omg omg that alembic stuff are great |
| 23:51.09 | navaismo | is all wet now |
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| 23:55.06 | mjordan | navaismo: thank russellb if he's ever in here :-) |
| 23:55.56 | navaismo | got it |