| 00:00.27 | michael-i | true |
| 00:00.47 | michael-i | just investing some time to see if I can be lazy |
| 00:01.46 | *** join/#asterisk Netgeeks (~chris@gw1.netgeeks.net) |
| 00:10.42 | *** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com) |
| 00:14.58 | *** join/#asterisk MarcWeber (~marc@li142-245.members.linode.com) |
| 00:15.19 | *** join/#asterisk Netgeeks (~chris@gw1.netgeeks.net) |
| 00:15.21 | beccara | anyone here know much about asterisk RTP bridging? I'm seeing my calls bridge in locally bridge but want to get the RTP out of the asterisk core so need to figure out why I can't get packet2packet bridging running |
| 00:16.03 | MarcWeber | Is someone interested in sending me an offerabout setting up a server which is put in between phonalite VOIP apps and sipgate recording all calls so that quality of a very small call center can be judged by its owner? |
| 00:16.33 | MarcWeber | Which is the place looking for payed asterisk support? |
| 00:16.50 | ruied | I'm trying two different set of dialplan rules using 2 blf keys. I've changesd the state of the keys with: "Set(DEVICE_STATE(Custom:lamp1)=NOT_INUSE)". The problem is when I set one blf key state the other changes also. Is there a way to have this separated ? |
| 00:17.09 | *** join/#asterisk coppice (~chatzilla@183178203098.ctinets.com) |
| 00:22.39 | ruied | nevermind... I've seen the problem... |
| 00:22.48 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip24-56-6-80.ph.ph.cox.net) |
| 00:25.22 | *** join/#asterisk batphone (~boo@unaffiliated/batphone) |
| 00:25.31 | batphone | hey fellas, check out my one liner |
| 00:25.57 | batphone | while true; do DATE=`date`; ping -c 3 oblivion.box | grep loss | cut -f 6 -d " " > results; RESULTS=`cat results`; if [[ "$RESULTS" == "100%" ]] ; then echo "$DATE - $RESULTS packet loss. Calling you now." ; cp /tmp/call-me.call /var/spool/asterisk/outgoing/; chmod 777 /var/spool/asterisk/outgoing/call-me.call ; RESULTS=0 ; break ; else echo "$DATE - $RESULTS packet loss."; fi ; done |
| 00:26.16 | batphone | ;D |
| 00:26.44 | *** join/#asterisk BuenGenio (~Gene@cm61-15-218-59.hkcable.com.hk) |
| 00:26.55 | batphone | this saved my butt last night. we had some carrier maintenance and i didnt want to wait the full six hour window until our network was affected |
| 00:27.08 | batphone | so i set this in motion and went to bed ;P |
| 00:27.24 | batphone | it beat our monitoring systems by a few minutes |
| 00:31.05 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 00:43.13 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 00:48.11 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 00:49.45 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 01:03.53 | *** join/#asterisk LiuYan (~liu.yan@211.154.128.135) |
| 01:14.18 | p3nguin | I have an SPA-3102 which typically communicates with asterisk for calling. I'm trying to dial *86 to check voicemail... I pick up the phone and hear the stutter dial tone, dial *86 (which is the voicemail extension used on other IP phones), and I just get another stutter dial tone. There is no call going to asterisk. sip debug reveals no call at all comes in when I dial *86 via the SPA-3102. Is this normal? How does one ... |
| 01:14.25 | p3nguin | ... usually check voicemail when using an ATA? |
| 01:15.12 | batphone | check that the ATA is not interpreting the * codes as something to interface with the device itself |
| 01:15.31 | batphone | sometimes you can configure the codes that the phone can dial that can be used to interact with the ATA |
| 01:15.35 | batphone | make sense? |
| 01:15.52 | p3nguin | Where would those be found? |
| 01:15.56 | ChannelZ | remember the ATA has its own dialplan |
| 01:16.46 | p3nguin | Found 'em. They are on the "Regional" tab. |
| 01:16.54 | p3nguin | Now to see if *86 is configured to do anything. |
| 01:17.09 | hardwire | bow chicka bow wow |
| 01:17.23 | p3nguin | Call Back Deact Code: *86 |
| 01:17.26 | p3nguin | :/ |
| 01:18.01 | *** join/#asterisk coppice (~chatzilla@m121-202-101-207.smartone-vodafone.com) |
| 01:18.45 | ChannelZ | I'm trying to rememebr if there was a global toggle to turn off all those vertical service codes, or if it was something in the dialplan that allows those to happen |
| 01:19.21 | p3nguin | I was looking for something to turn off all of those. I really prefer asterisk to handle everything. |
| 01:19.49 | p3nguin | Maybe if I explicitly configure *86 in the dialplan it will override the VSC. |
| 01:20.05 | p3nguin | tries |
| 01:20.16 | ChannelZ | I think you just have to erase them all. |
| 01:20.53 | ChannelZ | I use 500 for vm anyway.. but do have a *xxx in my dialplan for my own codes |
| 01:20.53 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 01:21.43 | p3nguin | I prefer *VM (*86) for voice mail just like Verizon. |
| 01:22.50 | p3nguin | I do have *xx in the dial plan, and the VSC was still hijacking my call to *86. |
| 01:23.59 | ChannelZ | yeah I actually just tried some on my SPA922 here and they are being picked up |
| 01:24.27 | ChannelZ | I swore I'd shut all this crap off but apparently not |
| 01:25.00 | p3nguin | Okay, I explicitly set *86 in the dial plan and it was still ignored. The only thing I can see to do is change the matching VSC to something else. |
| 01:26.19 | p3nguin | batphone: Good call on that. I hadn't opened my eyes and looked at other tabs to find that crap. |
| 01:27.44 | ChannelZ | I guess you can set all the 'Supplementary Services' to no on the appropriate Line tab rather than erasing all the codes (if you want to keep them in there for reference) |
| 01:27.47 | *** join/#asterisk binbash_ (~peter@server.digitog.nl) |
| 01:32.54 | autofsckk | night everybody |
| 01:33.21 | autofsckk | if i upgrade the firmware of a spa3102, does that deletes configurations? |
| 01:34.59 | ChannelZ | maybe |
| 01:35.42 | autofsckk | :S |
| 01:36.21 | autofsckk | p3nguin: my asterisk box is working excelent now, thanks a lot again :D |
| 01:37.21 | p3nguin | Great. That's what I like to hear. |
| 01:37.57 | autofsckk | i hope soon i can help people here too :D |
| 01:38.31 | p3nguin | I guess Call Back Serv: would be the setting pertaining to my *86. |
| 01:39.30 | autofsckk | tomorrow im helping a friend configuring a spa3102 that is now working with freepbx in an unbuntu box, but the spa has an old firmware, has some echo problems too, an delay when receiving calls and dialing too, it takes too long to ring |
| 01:39.48 | p3nguin | I do not recommend upgrading it. |
| 01:40.02 | p3nguin | I use 3.3.6(GW) happily. |
| 01:40.33 | autofsckk | really? i have read in some pages that the upgrades helps a lot with the echo problem |
| 01:44.11 | coppice | there is no software which fixes the echo problem in an SPA3102. many people have simply dumped them because of it |
| 01:46.44 | autofsckk | i've read that too |
| 01:56.53 | *** join/#asterisk zerohalo (~zerohalo@74.60.136.128) |
| 01:58.06 | coppice | a lot of the second hand ones on e-bay are from ISPs who gave up and had to rip them out en-mass. they haven't updated the software in years, so I guess they never will. interestingly they still ship from the factory with the 3.3.6 code in them |
| 01:58.52 | p3nguin | I've heard various bad things about the v5 firmwares, so I don't bother updating. |
| 02:00.23 | coppice | I have 5.whatever loaded in the one I use for test work. I haven't had any specific problems that 3.3.6 doesn't have, and the T.38 code is largely functional in 5.x.x |
| 02:03.27 | autofsckk | well i really dont know somebody that works with asterisk, but what i have read about te firmare upgrade to 5.x is that it helps with the echo problem, cant really fix it but it improves a lot |
| 02:12.17 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 02:13.36 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 02:22.04 | *** join/#asterisk eAndi_ (~eAndi_@177.29.66.32) |
| 02:22.19 | autofsckk | is there a way to put asterisk config into freepbx? my friend use freepbx, and iam going to edit the config files, but what about when i want to put the configuration files to his box? they wont work right? |
| 02:25.52 | SeRi | autofsckk, no |
| 02:26.07 | SeRi | most of freepbx is macrobase |
| 02:27.07 | autofsckk | i was trying to see where the sip.conf info was on there, and i couldnt find it |
| 02:28.42 | SeRi | Its all different and unsupported here |
| 02:28.51 | SeRi | ~freepbx |
| 02:28.51 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 02:29.01 | SeRi | :) |
| 02:30.41 | *** join/#asterisk andresmujica (~andres@ubuntu/member/andresmujica) |
| 02:33.59 | *** join/#asterisk F2Knight (~ben@c-24-22-60-186.hsd1.or.comcast.net) |
| 02:35.00 | F2Knight | Q: As I recall , the call-limit does not work the same in 1.8... but I notice that the 1.8 realtime setups do not even have the field in the db for it.. has it been depreciated all together? |
| 02:36.15 | WIMPy | Yes. YOu need to use the group functions. |
| 02:36.54 | F2Knight | Ahh Group thats what it was.. |
| 02:37.07 | F2Knight | thanks WIMPy forgot that. |
| 02:37.47 | *** join/#asterisk JuanCri (~JuanCri@pc-205-210-86-200.cm.vtr.net) |
| 02:39.54 | p3nguin | Depreciated, no. Deprecated, yes. |
| 02:40.07 | F2Knight | ~oink |
| 02:40.07 | infobot | from memory, oink is onomatopoeia for "Obviously you didn't calibrate the lateral phase stabilzers correctly because we're heading towards a giant bananna!", or a torrent site with an annoying habit of disabling accounts that haven't gone to the website in 6 weeks with no warning |
| 02:40.35 | p3nguin | call-limit should still work the same way, even though it is deprecated. |
| 02:40.49 | F2Knight | I am updating a script to import sip.conf to asteriskRT |
| 02:41.19 | p3nguin | autofsckk: If you're going to use FreePBX, use FreePBX. If you're going to use Asterisk, don't even think about using FreePBX. |
| 02:41.51 | F2Knight | and walking through some things that the digium folk did not put in the sipfriends.sql file... Preferd to double check on some of the ones I hardly seen in use.. |
| 02:42.13 | p3nguin | autofsckk: Everything should be configured via FreePBX if that's the way you're going to choose to admin the system. |
| 02:42.15 | F2Knight | like 'requirecalltoken', and t38pt_... |
| 02:44.06 | *** join/#asterisk aot2002_ (~aot2002__@cpe-74-67-42-78.nycap.res.rr.com) |
| 02:59.29 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1176139514.dsl.bell.ca) |
| 02:59.41 | dijib | can anyone tell me why this wouldnt have been completed yet? |
| 02:59.51 | dijib | http://www.crtc.gc.ca/public/cisc/lnp/owen.doc |
| 03:00.12 | dijib | second column is LNP thrid is WNP |
| 03:00.12 | dijib | Owen Sound 2006-02-03 2007-03-14 owen.doc - 63KB |
| 03:00.21 | dijib | completion dates |
| 03:04.36 | p3nguin | February 3, 2006 |
| 03:04.44 | p3nguin | It should have been done a LONG time ago. |
| 03:05.32 | *** join/#asterisk sorressean (~tyler@tds-solutions.net) |
| 03:05.53 | SeRi | waz up dijib |
| 03:05.57 | sorressean | I have a quick question, Is there a way for me to just autoconfirence everyone that dials in? like pickup, then confirence them? |
| 03:06.05 | dijib | pissed that LNP isnt available |
| 03:06.15 | sorressean | The goal is just to let everyone talk back and forth |
| 03:06.34 | p3nguin | sorressean: You can dump all calls into a ConfBridge() or MeetMe(). |
| 03:07.16 | sorressean | Gotcha. Is there an official list of all configuration values/extension functions? not sure what that one supports as args. |
| 03:07.23 | dijib | in ConfBridge and going for a smoke. join me guys 2663@asterisk.serveirc.com |
| 03:07.36 | p3nguin | core show application ConfBridge |
| 03:07.40 | p3nguin | core show application MeetMe |
| 03:08.11 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 03:08.32 | p3nguin | Remember that ConfBridge does not auto-answer the channel, so make sure you either have a Playback(), a BackGround(), or an Answer() before it. |
| 03:08.58 | p3nguin | MeetMe() should answer the channel, though. |
| 03:10.47 | sorressean | extin=>#,n,Answer() would work then? then I can bridge? |
| 03:11.01 | p3nguin | You're close, but that would fail. |
| 03:12.50 | dijib | SeRi, u gunna join? p3nguin ? im stepping out now for a smoke |
| 03:12.52 | p3nguin | If your extension is 123: exten => 123,1,Answer() exten => 123,n,ConfBridge(M1s) |
| 03:13.07 | dijib | get sorressean in here too |
| 03:13.09 | p3nguin | I might join after I go home. |
| 03:13.45 | p3nguin | Actually, I probably will once I get home, just to see if anything is going on. |
| 03:14.00 | sorressean | p3nguin: I'm basically setting up asterisk to listen on a port and just let a few friends chat back and forth. I don't have an extension, it'll just be the incoming call gets answered, then confirenced. |
| 03:14.20 | p3nguin | You certainly do have an extension. |
| 03:14.31 | SeRi | dijib, ill join in a few minutes |
| 03:14.32 | p3nguin | All calls to asterisk are processed by an extension. That's what it does. |
| 03:15.05 | SeRi | dijib, hopefully it does not crash or your phone dies :) |
| 03:15.10 | p3nguin | So figure out what extension you're having people dial. Use that extension instead of 123 like in my example. |
| 03:15.16 | sorressean | p3nguin: but there has to be a "default" extension, right? |
| 03:15.23 | p3nguin | There is no default extension. |
| 03:15.45 | p3nguin | Figure out what extension people call to reach your conference. |
| 03:16.42 | p3nguin | If you don't know what extensions are configured, "dialplan show" will list all of them. |
| 03:16.48 | sorressean | p3nguin: they just point their sip client at myurl.com |
| 03:17.02 | sorressean | ah. gotcha. thanks. |
| 03:17.11 | p3nguin | But there has to be an extension. A SIP URI consists of an extension and a hostname. |
| 03:17.32 | p3nguin | jack@myurl.com ; extension here is 'jack' |
| 03:17.36 | WIMPy | Does it have to? |
| 03:17.47 | p3nguin | 123@myurl.com ; extension here is '123' |
| 03:18.29 | sorressean | p3nguin: o, makes sense, thanks. so I need a dialplan to catch that, then pass it off to extensions. |
| 03:18.46 | p3nguin | dialplan consists of extensions. |
| 03:18.55 | p3nguin | There is no passing anything off to anywhere. |
| 03:19.17 | sorressean | ah. |
| 03:19.20 | sorressean | thanks. |
| 03:20.01 | p3nguin | So what extension are they entering? |
| 03:20.29 | p3nguin | Perhaps their phone is providing an extension if all they are putting in is the host. |
| 03:20.32 | WIMPy | To answer to myself: Yes, a user is mandatory. |
| 03:21.36 | p3nguin | If they are only entering the host name without an extension, either the call will fail, or you'll have to debug it to see what extension their phone added silently. |
| 03:22.35 | sorressean | p3nguin: sweet. I just set it to conf. so conf@myurl.com:3000 |
| 03:23.02 | p3nguin | You configured your asterisk to listen on port 3000 instead of the normal port? |
| 03:23.05 | *** join/#asterisk gajini (~root@61.12.17.170) |
| 03:32.54 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 03:32.54 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0-beta2 (2011/09/27), 1.8.7.1 (2011/10/17), dahdi-linux 2.5.0.2 (2011/10/21), dahdi-tools 2.5.0.2 (2011/10/21), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 03:36.57 | *** part/#asterisk gajini (~root@61.12.17.170) |
| 03:41.01 | sorressean | http://pastebin.com/6BhwAxbb |
| 03:41.14 | sorressean | That's my sip.conf and extensions.conf. the port is still closed though, and it accepts no incoming calls |
| 03:41.50 | p3nguin | Lines 12 and 13 are failure. |
| 03:42.20 | p3nguin | It is exten, not extin... and your spaces are in the wrong places. |
| 03:42.30 | p3nguin | (2112.52) <p3nguin> If your extension is 123: exten => 123,1,Answer() exten => 123,n,ConfBridge(M1s) |
| 03:43.26 | sorressean | p3nguin: well, I feel retarded. Asterisk didn't print error swhen I started that my conf files were broken though. sorry, exten and extin sounds the same with my reader, should've guessed though. :p |
| 03:44.12 | ChannelZ | Reader? Are you blind/hard of sight? |
| 03:44.17 | *** join/#asterisk Rholk (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 03:44.48 | ChannelZ | or "partially blind", I'm making up terms now. |
| 03:45.36 | sorressean | ChannelZ: yeah, I'm blind |
| 03:45.40 | sorressean | ChannelZ: total |
| 03:46.10 | ChannelZ | Ok, just curious if that's what that meant |
| 03:46.23 | p3nguin | sorressean: I was thinking something more like this: http://pastebin.com/vL8JbTQS |
| 03:47.20 | p3nguin | I don't remember if you indicated your asterisk version, so I assume you have ConfBridge(). |
| 03:49.52 | sorressean | p3nguin: yeah, it shows up when I use asterisk -v to launch. Granted I don't see configuration errors, but it shows it loading that |
| 03:50.36 | p3nguin | sorressean: Make sure you allow UDP 5060 and the UDP range configured in rtp.conf through the firewall. |
| 03:50.48 | sorressean | and for whatever reason 5060 is still closed. hrm |
| 03:51.26 | p3nguin | How are you determining that it is closed? |
| 03:52.31 | sorressean | p3nguin: I nmap myself and my sip client doesn't connect. |
| 03:52.41 | SeRi | dijib, you in? |
| 03:54.37 | p3nguin | sorressean: How are you using nmap? Show me the exact command. Feel free to omit your IP address if you wish; I am only interested in the options you are passing to nmap. |
| 03:57.31 | sorressean | I'm using nmap -sU dev.tds-solutions.net and nmap -P0 dev.tds-solutions.net |
| 03:58.34 | p3nguin | Is your asterisk on a public IP address, or is it behind a NAT? |
| 04:00.54 | sorressean | p3nguin: public IP |
| 04:01.21 | p3nguin | sorressean: Are you using a firewall, such as iptables? |
| 04:02.27 | sorressean | p3nguin: yeah. I've flushed and set input to accept just to test. |
| 04:03.50 | p3nguin | sorressean: Can you see if "lsof -i udp:5060" shows asterisk is listening? |
| 04:05.01 | sorressean | p3nguin: it's not. |
| 04:05.40 | sorressean | I've got it running too. I've compiled asterisk from source, config files are in /etc/asterisk, I've tried doing asterisk -C /etc/asterisk/sip.conf to make sure. |
| 04:09.51 | sorressean | man, this is going to take a lot of beer, but I'm going to make it work. |
| 04:10.42 | *** join/#asterisk beccara (~beccara@180.222.64.254) |
| 04:17.26 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 04:19.13 | p3nguin | sorressean: I think -C would take the path to asterisk.conf, not sip.conf. |
| 04:25.03 | autofsckk | p3nguin: what firefox version do you have? its now 8? what is going on with ff :S |
| 04:25.14 | p3nguin | 3.6.18 |
| 04:27.20 | sorressean | haha. that's the best version to have. mozilla got into a e-penis versioning match with IE and chrome. |
| 04:27.27 | autofsckk | firefox-8.0-1 |
| 04:30.23 | sorressean | is there a way to see what dialplans I have? dialplan show just shows one registered |
| 04:30.32 | *** join/#asterisk radic (~radic@dslb-178-002-229-101.pools.arcor-ip.net) |
| 04:33.41 | sorressean | meh. scrue this. it's still not binding, so I'm still dead in the water. I'll play with it tomorrow. :p peace |
| 04:36.53 | *** join/#asterisk irroot (~irroot@197.109.94.237) |
| 04:38.29 | ChannelZ | you sure chan_sip is even loaded? |
| 04:41.13 | *** join/#asterisk rdegges (QL8Hx7jfJT@69.164.197.143) |
| 04:41.17 | rdegges | Hey guys, quick question. |
| 04:41.30 | rdegges | I'm running a teleconferencing server, and I've been getting this message a lot lately: "Unable to open DAHDI pseudo channel: Cannot allocate memory" |
| 04:41.42 | rdegges | It happens when a user attempts to join a conference via MeetMe |
| 04:41.49 | p3nguin | sorressean: dialplan show shows everything loaded. |
| 04:41.58 | rdegges | I tried googling it, but didn't find much. |
| 04:42.07 | rdegges | My server has ~10g of free ram according to free -m |
| 04:42.14 | rdegges | When this error occured. |
| 04:42.19 | rdegges | Any idea what could be causing it? |
| 04:42.35 | rdegges | I'm running Asterisk 1.8.7.0 on ubuntu-server 11.04 |
| 04:42.40 | rdegges | with the latest release of dahdi |
| 04:46.23 | rdegges | Furthermore, I'm using dahdi_dummy, if that matters. |
| 04:46.25 | rdegges | Not hardware timing. |
| 04:46.56 | ChannelZ | is it really running? |
| 04:47.14 | rdegges | yeah, i can see it via 'lsmod' output |
| 04:47.25 | p3nguin | There is no dahdi_dummy. |
| 04:47.40 | rdegges | dahdi 216069 1021 xpp,dahdi_transcode,wcb4xxp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,dahdi_voicebus,wct4xxp |
| 04:47.40 | rdegges | crc_ccitt 12667 2 wctdm24xxp,dahdi |
| 04:47.40 | p3nguin | If you see it, you're doing something wrong. |
| 04:48.25 | ChannelZ | You should save yourself the pain and move over to ConfBridge if you can anyway |
| 04:48.29 | rdegges | The behavior isn't consistent. I've called in about 10 times in a row, and maybe 6 of the times I got the error--the other few times it worked alright. |
| 04:48.52 | rdegges | Yah--I'm in the process of porting some of our code to use it, but there's still a lot of work to be done. |
| 04:49.37 | p3nguin | If you're going to go to that trouble, better develop for asterisk 10's confbridge. It has to be better than the minimalistic confbridge we have in 1.8. |
| 04:58.10 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 05:11.42 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 05:16.35 | rdegges | Well, if any of you happen to think of anything that could be causing the dahdi memory issues, let me know via PM. Thanks! |
| 06:00.11 | *** join/#asterisk singler (~singler@81-7-123-162.static.zebra.lt) |
| 06:01.18 | *** join/#asterisk tyman (~tyman@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 06:10.21 | *** join/#asterisk WiretapSeven (~Wiretap@unaffiliated/wiretap) |
| 06:30.38 | *** join/#asterisk AmirBehzad (~behzad@31.184.187.1) |
| 06:31.22 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 06:32.18 | *** join/#asterisk depressed (~depressed@ca16.v6.us.gnics.net) |
| 06:32.48 | *** join/#asterisk depressed (~depressed@ca16.v6.us.gnics.net) |
| 06:33.00 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 06:43.08 | leifmadsen | anyone know why a D-Channel would continuously flap using TDMoE? |
| 06:47.20 | ChannelZ | flap flap |
| 06:47.57 | ChannelZ | Unhappy switch? |
| 06:49.14 | irroot | leifmadsen TDMoE bleg |
| 06:49.33 | leifmadsen | irroot: ya... new client, never used this before. RedFone device. |
| 06:50.28 | irroot | leifmadsen had endless problems with it in past it needs really stable enviroment and good switching before you have a chance |
| 06:50.43 | irroot | there was some kit made locally called farsouth |
| 06:50.48 | *** join/#asterisk depressed (~depressed@ca16.v6.us.gnics.net) |
| 06:51.18 | *** join/#asterisk depressed (~depressed@ca16.v6.us.gnics.net) |
| 06:53.46 | leifmadsen | irroot: ya it seems to be fine on one of the servers, but not the other, so need to figure out what the difference in configuration is. Also, unloading dahdi causes a kernel panic |
| 06:54.47 | irroot | leifmadsen that is not good its not some custom hack ? |
| 06:55.01 | leifmadsen | irroot: no idea, only been lookin at this server for 4 hours |
| 06:55.06 | leifmadsen | didn't configure this server |
| 06:56.31 | irroot | leifmadsen need more red bull ?? |
| 06:56.40 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-afskucrqrdhjrdad) |
| 06:56.41 | leifmadsen | irroot: no I just want to go to bed soon |
| 06:56.46 | leifmadsen | I was already in bed, then they called back |
| 06:57.25 | irroot | leifmadsen bugger well its 9am here so some scarry hour your side if i can help buzz |
| 06:57.52 | leifmadsen | irroot: coolio, just got this flapping D channel on one of the boxes and no audio for some reason |
| 06:57.59 | leifmadsen | other side seems to be fine |
| 07:04.47 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 07:12.07 | *** join/#asterisk hetii (~hetii@194.181.154.25) |
| 07:22.03 | *** join/#asterisk binbash_ (~peter@server.digitog.nl) |
| 07:22.16 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:22.33 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:26.32 | leifmadsen | irroot: ok all working now -- you ever run into a kernel panic when unloading dahdi? |
| 07:27.24 | irroot | leifmadsen no not for long time and then it was the hacked version |
| 07:27.44 | irroot | possibly a problem in tdmoe |
| 07:27.55 | irroot | glad its working grab some sleep |
| 07:28.02 | leifmadsen | hmmm interesting, this is a new checkout of dahdi (2.4.1.2) and another box has it and doesn't do it |
| 07:28.03 | leifmadsen | just this one |
| 07:28.13 | leifmadsen | ya not quite, just the kernel panic |
| 07:31.23 | *** join/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4) |
| 07:52.13 | *** join/#asterisk dom| (~domi@mail.tas.de) |
| 07:56.35 | *** join/#asterisk pietro1 (~pietro@88-149-224-154.dynamic.ngi.it) |
| 07:59.00 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:b1dc:54ed:81d2:84f0) |
| 08:04.05 | *** join/#asterisk bulkorok (d9468199@gateway/web/freenode/ip.217.70.129.153) |
| 08:07.02 | *** join/#asterisk ayrjola (~ayrjola@89.18.236.11) |
| 08:12.43 | *** join/#asterisk waschtl (~waschtl@HSI-KBW-078-043-090-014.hsi4.kabel-badenwuerttemberg.de) |
| 08:13.04 | *** join/#asterisk oej (~olle@static-213-115-251-100.sme.bredbandsbolaget.se) |
| 08:19.10 | *** join/#asterisk phix (~threat@123.243.44.131) |
| 08:19.21 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 08:22.24 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 08:27.50 | *** join/#asterisk elliot98 (~elliot@unaffiliated/elliot98) |
| 08:30.08 | *** join/#asterisk elliot98 (~elliot@unaffiliated/elliot98) |
| 08:30.15 | elliot98 | hello |
| 08:30.36 | elliot98 | when a call is connected through a queue, what accountcode is set up for the agent? |
| 08:34.39 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 08:42.36 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 08:44.57 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 08:48.25 | *** join/#asterisk _N1x (~n1x@mail.orient-logic.com) |
| 08:49.56 | *** join/#asterisk Nasga (~Nasga@186.212.10.93.rev.sfr.net) |
| 08:49.59 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 08:50.01 | *** join/#asterisk lovetide (~lovetide@211.154.128.135) |
| 08:50.08 | *** join/#asterisk LiuYan (~liu.yan@211.154.128.135) |
| 08:52.18 | _N1x | guys i have one strange problem , im testing my asterisk stress test with sipp , when i had 2000 active channel ( active call) everything is ok and after 30 mins later , i have this output http://pastebin.com/5YCS4n3F , at 600 channels . why?... |
| 08:52.26 | _N1x | i have unlimited everything |
| 08:52.34 | _N1x | i think ... |
| 08:54.03 | _N1x | my cpu loading isn't much , max 20 - 25 % and memory also... |
| 08:54.58 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 08:57.49 | *** join/#asterisk Hanumaan (~Hanumaan@dslb-092-074-233-014.pools.arcor-ip.net) |
| 08:58.27 | *** join/#asterisk cusco (~tralala@88.157.128.26) |
| 09:00.50 | irroot | _N1x the packets are leaving but no response are the phones / network been kludged up |
| 09:01.05 | *** join/#asterisk xpot (~xpot@166-70-100-198.ip.xmission.com) |
| 09:04.25 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 09:04.31 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 09:04.33 | schmidts | good morning |
| 09:05.29 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 09:13.25 | *** part/#asterisk AmirBehzad (~behzad@31.184.187.1) |
| 09:16.16 | *** join/#asterisk krotos (~d3v1l@host211-3-dynamic.5-87-r.retail.telecomitalia.it) |
| 09:16.20 | krotos | hi :) |
| 09:16.35 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 09:18.44 | krotos | there is a way to modify query for selecting data in asterisk realtime? |
| 09:19.05 | krotos | i'm only interested in sip_users, not peers |
| 09:23.42 | schmidts | krotos you can change the source code but i think this is not what you want to do ;) why do you only have users and no peers? |
| 09:26.37 | krotos | good question :) for peers, is only 3 or 4 (provider) but i can place all in mysql ( realtime) |
| 09:27.04 | krotos | so, for the users i've already got a table in my CRM with the data , and i want to use this table for doing realtime |
| 09:29.09 | krotos | at this time, i have a script that generate plain text file for sip.conf file |
| 09:30.19 | irroot | krotos is there a need for all this ?? why not have a default context where calls come and handle it in the dialplan its for inbound only ?? |
| 09:33.49 | krotos | there is a global context where all inbound call come in |
| 09:34.36 | krotos | and i handle it in dialplan, |
| 09:35.14 | *** join/#asterisk garymc (~chatzilla@81.138.225.164) |
| 09:35.26 | krotos | my problem is that i have about 700 users ( type=friend) in plain text file (sip.conf) and if i type "reload" in asterisk about 50/60 of this users become unreachable |
| 09:35.33 | krotos | and after 1 minute come up again |
| 09:37.27 | krotos | and i want to try using realtime , for fixing this issue |
| 09:47.17 | schmidts | krotos forget realtime this will become even worst with it |
| 09:47.23 | schmidts | krotos which asterisk version do you use? |
| 09:51.32 | krotos | 1.8.7.1 |
| 09:51.47 | schmidts | hmm strange |
| 09:52.00 | schmidts | do you do a "sip reload" or just "reload" |
| 09:52.11 | krotos | if i do sip reload, nothing happen |
| 09:52.19 | krotos | but with "reload" |
| 09:52.20 | schmidts | ah ok |
| 09:52.34 | krotos | sometimes 50/60 peer become unreacheable,...somtimes only 20.. |
| 09:52.36 | krotos | is randomly |
| 09:52.47 | schmidts | but why do you use reload and not sip reload? if there are any chances in the config file they will be used even with sip reload |
| 09:57.49 | krotos | i always use sip reload, but yesterday i've modify something in sip.conf, extension.conf , voicemail.conf |
| 09:57.54 | krotos | and i type reload |
| 09:58.10 | krotos | (and not sip reload, dialplan reload, voicemail ...) |
| 09:58.21 | krotos | and i've see this behavior |
| 10:00.43 | *** join/#asterisk elliot98 (~elliot@unaffiliated/elliot98) |
| 10:00.59 | elliot98 | gives a wave to all |
| 10:01.01 | elliot98 | hello |
| 10:01.02 | garymc | right im trying to connect my soft phone to the new server the same way i do to my current setup. here is my pastebin. i dont know what is wrong http://pastebin.com/jccDhB43 |
| 10:04.39 | kaldemar | garymc: what you're showing is a trace of asterisk sending a registration message to a polycom phone. it should be the other way around. |
| 10:05.59 | *** join/#asterisk _N1x (~n1x@mail.orient-logic.com) |
| 10:06.30 | _N1x | guys , for high performance i need more CPU or memory resources? |
| 10:06.41 | _N1x | i think CPU is important yes? |
| 10:07.31 | garymc | right i tried to conect my softphone |
| 10:07.55 | garymc | i thought it was showing that too |
| 10:09.21 | schmidts | _N1x it depens on your asterisk version :D mostly you will not need more cpu if you just use the wrong version |
| 10:09.57 | schmidts | krotos never noticed this but i guess this could happens when sip tries to ping the users before its completly loaded and so the response isnt handled right |
| 10:10.02 | kaldemar | garymc: fo to #freepbx and ask how to configure a phone with it. |
| 10:11.18 | garymc | yeah i know how to, its just not connecting |
| 10:12.42 | garymc | or I dont know how to. I do have a current PBX working fine. I m just replicating it and I cant get it to work |
| 10:12.42 | krotos | schmidts: ok :) and about realtime? there is a way to have only sip users ( type=friend) over mysql-realtime and type=peer on flat-file?and i think to modify query in source for adapt to my table already present |
| 10:13.21 | _N1x | schmidts: i use 1.6 version |
| 10:13.29 | _N1x | and when i'm starting sipp stress test |
| 10:13.37 | _N1x | cpu is loaded than memory |
| 10:13.53 | _N1x | for more calls purpose i think i need more cpu yes? |
| 10:13.56 | _N1x | im right?... |
| 10:14.11 | schmidts | type=friend is peer & user at the same time, so if you want to use realtime it will not make a difference but i think this reload stuff is a bug but if sip reload works without any problems i dont see a big need for you to change these things |
| 10:15.38 | schmidts | _N1x do you want to have concurrent calls or more calls per second? you may be interested in this graph ;) https://docs.google.com/spreadsheet/ccc?key=0AuJ3v0yn3iv-dG1CZC1RR184Mk05XzN5UnM3cC1pMmc |
| 10:16.00 | krotos | yea, but i think realtime is better then flat-file. My script for generating flat-file it takes 5 minute |
| 10:16.08 | schmidts | _N1x imho you will need another asterisk version first not another cpu |
| 10:16.48 | schmidts | krotos realtime is much slower than a flat-file cause you will have a database lookup on every sip dialog which would kill your performance |
| 10:16.56 | krotos | and for example, if i modyfi a password for a single friend, the script regenerate all file, including |
| 10:17.01 | krotos | ouch :( |
| 10:17.04 | schmidts | and btw krotos you should fix your sync script ;) |
| 10:17.33 | schmidts | my sync script generates a flat file for 4222 sip peers in 34 ms ;) |
| 10:18.01 | krotos | you read data from mysql? |
| 10:18.06 | schmidts | yes |
| 10:19.03 | schmidts | what do you use for your sync file? a bash script or something like this? |
| 10:19.09 | krotos | my script is written in php.. |
| 10:19.19 | schmidts | ok :D |
| 10:19.43 | krotos | and run every 10 minutes if there is change ( i use a flag over mysql if there is something to re-sync) |
| 10:20.01 | krotos | (placed in crontab) |
| 10:20.22 | schmidts | ;) |
| 10:21.37 | krotos | in my script i've got only a while and three if |
| 10:21.44 | *** join/#asterisk Hanumaan (~Hanumaan@92.74.233.14) |
| 10:21.44 | schmidts | hehe |
| 10:21.46 | krotos | so the probelm for slow |
| 10:21.48 | krotos | is the if.. |
| 10:22.04 | schmidts | give me a minute to clean my c file of private data and i will paste it for you |
| 10:22.05 | krotos | but 422 sip peers in 34 ms is like a Lamborghini |
| 10:22.29 | krotos | or...is a php problem |
| 10:23.14 | schmidts | 4222 not only 422 ;) |
| 10:23.29 | krotos | ops..i lost a 2 :P |
| 10:23.36 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 10:23.36 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 10:27.13 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 10:30.16 | schmidts | krotos here you are: http://pastebin.com/2Zp3JUBg |
| 10:31.39 | schmidts | you only have to change the mail recipient, database credentials and the sql query itself and i use a file called sip.generated.conf which is just linked into the normal sip.conf |
| 10:32.17 | krotos | yes, i have sip_users.conf in my case linked into sip.conf |
| 10:32.32 | schmidts | ok ;) |
| 10:32.37 | krotos | thankyou a lot :) |
| 10:33.09 | schmidts | your welcome |
| 10:33.14 | schmidts | i hope this works for you |
| 10:35.50 | krotos | wiht some change for my structure , i think it works better then php script :) |
| 10:40.20 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 10:48.39 | _N1x | schmidts: thanks for link , can you provide me asterisk stress examples and guides? thanks |
| 10:51.47 | schmidts | krotos i really hope so, if ever a c binary is slower than a php script which does the same, something has gone completly wrong ;) |
| 10:54.38 | schmidts | _N1x just use sipp ;) |
| 10:57.04 | _N1x | schmidts: im using but need for examples how to test , call per second and concurrent calls |
| 10:57.47 | schmidts | _N1x i use this: http://pastebin.com/umQThUsv |
| 10:58.31 | schmidts | with a sipp command like this: sipp -sf sipload.xml -d 10000 -s 2002 destinationip -mp 5606 -i sourceip -m 12000 -r 300 |
| 10:58.42 | singler | does anyone know is Sangoma's NSG SS7 passes SIP header to asterisk with redirect reason? |
| 10:59.13 | _N1x | schmidts: what is mp , m and r options? |
| 10:59.59 | schmidts | -r is the call per second rate, -m is the max amount of calls and mp is the local rtp echo port |
| 11:00.48 | _N1x | schmidts: amount of calls is maybe concurrent calls yes? |
| 11:01.23 | schmidts | it depends on your scenario if you wait long enough this could be the value of concurrent calls yes |
| 11:02.11 | _N1x | schmidts: thanks :) i'll test using this xml file |
| 11:02.17 | schmidts | ok ;) |
| 11:02.24 | schmidts | have fun :D |
| 11:03.21 | _N1x | schmidts: and which is sipp's directory to copy this xml? |
| 11:03.52 | schmidts | whereever you want just use the -sf param to give the right path |
| 11:04.03 | schmidts | so it could look like this: sipp -sf /tmp/testload.xml .... |
| 11:04.14 | _N1x | yes , 10x |
| 11:13.29 | *** join/#asterisk blizzow (~jburns@67.50.165.59) |
| 11:25.16 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 11:39.52 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-67-253-245-174.rochester.res.rr.com) |
| 12:05.51 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:05.51 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:49.39 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 12:57.06 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:b1dc:54ed:81d2:84f0) |
| 13:02.33 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 13:05.42 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 13:07.59 | ollii | schmidts: i tried your sipload xml scenario on asterisk 1.4 and asterisk 10...created 12000 calls in 5mins, successfull with 1.4 2500 and on asterisk 10 3300...are these "normal" results? |
| 13:10.12 | *** join/#asterisk scubes13 (~scubes13@cpe-024-168-196-000.sc.res.rr.com) |
| 13:12.07 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 13:12.33 | schmidts | ollii yes looks very normal to me |
| 13:12.49 | schmidts | 12000 calls needs some fine tuning on the system side and also for asterisk |
| 13:18.51 | *** join/#asterisk luckman212_phone (~luckman21@pool-108-41-8-176.nycmny.fios.verizon.net) |
| 13:21.03 | saisoma | hey guys, having an issue with voicemail and a branch office: http://pastebin.com/1zZfMmZ9 any assistance is greatly appreciated. |
| 13:25.36 | schmidts | saisoma i dont know the exact state of development for this, but AFAIK is asterisk allready able to subscribe itself to a MWI state of another asterisk. |
| 13:26.19 | *** join/#asterisk saisoma (~saisoma@client72.jdcc.edu) |
| 13:27.28 | saisoma | not sure if this came through before (irc client crashed) so forgive me if you saw this just a few minutes ago |
| 13:27.30 | saisoma | hey guys, having an issue with voicemail and a branch office: http://pastebin.com/1zZfMmZ9 any assistance is greatly appreciated |
| 13:28.26 | ollii | schmidts: increasing open files to 32768 help in asterisk 10 (12000/12000 successful) ... asterisk 1.4.42 complains about sockets ("cannot create socket") |
| 13:29.12 | schmidts | olliii maybe you have to check your rtp.conf file and there is also a limit for numothersock (sorry dont know how its called for your system) |
| 13:31.26 | *** join/#asterisk felimwhiteley (~quassel@46.7.101.58) |
| 13:32.15 | ollii | debian/ubuntu |
| 13:37.08 | ollii | seems also good on asterisk 1.4.42 with the help of increasing open file limit...just forgot that i have to increase it per shell .. ;) |
| 13:40.06 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-aefgagkjfiukpgjt) |
| 13:46.24 | schmidts | ;) |
| 13:47.24 | r0m|u | so I did not know that some voip provider can make there number appear as they where land lines.... Is this true? |
| 13:48.12 | *** join/#asterisk grandpapadot (~grandpapa@99.175.248.81) |
| 13:48.50 | irroot | r0m|u have done it myself for customers and pranking my family |
| 13:50.08 | r0m|u | irroot, how? I thought voip numbers can not be pass cid names or in any natured masked....? |
| 13:51.26 | irroot | r0m|u the interconnect needs to be CLI and you need to pass the number to them legislation may be a barrier |
| 13:51.47 | *** join/#asterisk wesphillips (~wphill04@137.237.195.4) |
| 13:51.49 | *** join/#asterisk eAndi_ (~eAndi_@177.29.35.18) |
| 13:52.08 | irroot | we request a affidavit that the numbers displayed are valid and assigned to them by the telco as a precaution |
| 13:52.43 | r0m|u | o wow! so it did took some work. |
| 13:52.51 | r0m|u | but totally possible |
| 13:53.39 | r0m|u | interesting. Thanks for the info. |
| 13:54.41 | WIMPy | Here almost all VOIP numbers are normal landline numbers. |
| 13:55.00 | r0m|u | WIMPy, where you at? US? |
| 13:55.10 | WIMPy | But OTOH most landlines are VOIP accounts now. |
| 13:55.13 | WIMPy | de |
| 13:55.19 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:55.22 | r0m|u | ah. I see. |
| 13:55.53 | WIMPy | There was a plan to shut down the PSTN by 2012. |
| 13:56.21 | r0m|u | Apparently there is a company called reducefraud that can determine if your line is voip or land line |
| 13:56.25 | WIMPy | But the current version is if there are less tan 7M customers left. |
| 13:56.44 | r0m|u | WIMPy, wow seriously? Is that even a good idea? |
| 13:57.10 | WIMPy | No. it's an extremely bad idea, but it's a cheap idea. |
| 13:58.01 | r0m|u | Well I guess its all geo base :) in PR if you shit PSTN down and everybody uses voip.... we will be fucked once a Hurricane comes :) we get one hurricane a year and pstn most of the time still work and we can at least communicate. :) |
| 13:58.05 | WIMPy | Makes one of the two networks redundant. |
| 13:58.15 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 13:58.17 | r0m|u | ah. I see. |
| 13:58.33 | r0m|u | shut* (rofl shit) |
| 13:59.26 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 14:00.56 | WIMPy | Yes, average down time has already increased dramatically. |
| 14:01.04 | WIMPy | And it will probably continue to do so. |
| 14:01.38 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 14:03.12 | r0m|u | I see. |
| 14:10.46 | *** join/#asterisk master_of_master (~master_of@p57B54E40.dip.t-dialin.net) |
| 14:17.34 | *** join/#asterisk Ulrar (~Ulrar@2a01:e0b:1:136:62eb:69ff:fe8f:18a0) |
| 14:18.55 | Ulrar | Hi, is there a way for an AGI script to get the result of a command like "show dialplan number@context" ? |
| 14:19.13 | Ulrar | Without actualy forking and executing asterisk -rx .. |
| 14:19.23 | *** join/#asterisk irroot (~irroot@197.109.88.133) |
| 14:20.51 | *** join/#asterisk francisvgarcia (~francis.g@190.80.239.124) |
| 14:22.27 | *** join/#asterisk cerberus_za (~coert@8ta-151-47-57.telkomadsl.co.za) |
| 14:25.15 | *** join/#asterisk coppice (~chatzilla@183178203098.ctinets.com) |
| 14:27.40 | *** join/#asterisk fireman_biff (~biff@65.48.133.103) |
| 14:28.33 | fireman_biff | I have dundi peers with dynamic addresses and they don't update the addresses until I do "asterisk -rx reload". How can I fix this? |
| 14:29.53 | fireman_biff | asterisk 1.6 |
| 14:42.45 | p3nguin | irroot: Care to explain the remark about making VoIP phone numbers appear as PSTN numbers? Does not parse for me. |
| 14:43.36 | *** join/#asterisk zerohalo (~zerohalo@74-94-141-204-ma.hfc.comcastbusiness.net) |
| 14:43.39 | *** join/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl) |
| 14:43.42 | irroot | p3nguin CLI presentation VOIP providers on a CLI route can pass any arbitary number to carriers to present to the end party |
| 14:43.44 | jacc0 | hi all |
| 14:45.04 | irroot | so when i call from my voip interconnect my advertised CLI is that of my main telco "hardline" |
| 14:45.47 | irroot | also falls into the category of CLI spoofing |
| 14:45.55 | p3nguin | Still doesn't make sense. I just don't see how a call can say, "Hey, I'm a VoIP phone number!" |
| 14:46.30 | p3nguin | If I pick up a phone and dial my neighbor, he has no idea how my call is delivered to him. |
| 14:46.38 | irroot | correct |
| 14:46.53 | irroot | but one step further is to display my assigned DDI |
| 14:47.18 | p3nguin | He can even look up the switch information, and there's nothing to say how my number arrived at said switch. |
| 14:47.18 | saisoma | question regarding integrating a branch office. is there a way to "share" hint status between * servers? |
| 14:48.11 | irroot | could be on a PRI with a legacy telco so calls coming back will come in on the "normal" line and the call is displayed from the contacts as normal and not a "private number" |
| 14:49.37 | [TK]D-Fender | There is no such thing as a "VoIP phone number". PSTN is just that. If one leg changes the signalling to a VoIP protocol along the way, that is irrelevant. CallerID is not a "DID", nor a "VoIP number". It is caller ID. You can pretty much fake whatever you want into there including numbers that are not valid for dialing anywhere. |
| 14:49.51 | p3nguin | As far as I can tell, the only person/place/thing that will know if a phone call is originating from VoIP or a real line would be the first carrier taking the call. From there on out, it's staying on the PSTN, even if the PSTN is using VoIP. |
| 14:50.45 | irroot | p3nguin indeed [TK]D-Fender that said we have a range of NGN numbers +2787 that are VOIP numbers |
| 14:51.10 | p3nguin | That must be something specific to your region. |
| 14:51.26 | ollii | in germany theres also a numbering block for voip |
| 14:51.26 | [TK]D-Fender | irroot, not a VoIP number. the fact that some telco (ITSP) converts protocols is irrelevant. Nothing about the NUMBER is "VoIP" |
| 14:51.33 | ollii | but that is only a number.. |
| 14:51.34 | francisvgarcia | fireman_biff: are u using iax |
| 14:51.40 | *** join/#asterisk Liability (~gfilmer@196.1.57.28) |
| 14:51.43 | francisvgarcia | fireman_biff: iax for the peering |
| 14:52.01 | fireman_biff | yes iax for the calls |
| 14:52.57 | ollii | i dont know why such a thing exists...mostly our telcos provides us sip with a sip <-> bri/pri gateway ... so we think we use bri/pri |
| 14:53.05 | p3nguin | Here, phone numbers are just phone numbers. I can port a phone number away from a landline carrier who is providing copper to the house and stick it on an ITSP. When I call you using said phone number, you have no way to know if I'm connected with copper or if I'm talking to you via VoIP protocols. |
| 14:53.09 | irroot | [TK]D-Fender indeed a number is a number technically there is no distinction |
| 14:53.12 | fireman_biff | francisvgarcia: ^ |
| 14:53.18 | francisvgarcia | fireman_biff: and what abbout qualifing the peers |
| 14:54.05 | francisvgarcia | qualify=whatevertimeyoulikeinms |
| 14:54.34 | francisvgarcia | I actually have 2 dundi peers runing on dyndns and they are working fine |
| 14:54.36 | *** join/#asterisk edge (~edge@97.64.216.2) |
| 14:54.36 | eppigy | mornin |
| 14:55.18 | irroot | p3nguin there is a possiblility here of porting the number to the ITSP for them to terminate but it can only be ported in geographical area so ITSP's have to install data centres in all areas now |
| 14:55.42 | fireman_biff | francisvgarcia: my dundi.conf has "qualify=yes", its supposed to be a number? |
| 14:55.59 | p3nguin | Okay... so how did you arrive at this distinction of "VoIP calls" vs. "landline calls?" |
| 14:56.03 | schmidts | irroot i hope you mean all countries and not all areas? |
| 14:56.37 | jacc0 | @fireman_biff: yes or a number is both proper |
| 14:56.59 | francisvgarcia | fireman_biff: and under iax.conf |
| 14:57.04 | irroot | schmidts no locations and not provinces either its a mess some areas are several hundered km some are <100km |
| 14:57.55 | francisvgarcia | fireman_biff: I looks to happen when one of your internet services disconnects and renew a new ip address? |
| 14:57.56 | schmidts | irroot ok you are talking about to serve landlines |
| 14:58.22 | p3nguin | Let us assume that there is a data center in your location, and that you are using an ITSP as well as landline services. How is anyone going to know the difference between a call you make via landline and a call you make via VoIP. |
| 14:58.35 | fireman_biff | francisvgarcia: would the iax config come into play? its the dundi peer that doesn't have the ip address update when the ISP gives a new one |
| 14:58.36 | p3nguin | s/IP./IP?/ |
| 14:59.21 | francisvgarcia | fireman_biff: that's what I am talking about |
| 14:59.28 | irroot | p3nguin the original question was can you display your legacy landline number on voip calls so the assumption is the landline is still inplace not ported and that there is a voip carrier ... the idea would be to dial from a unified number accross all carriers the answer is sure send the CLI down the voip link and i the carrier allows this and passes it on the called party will not know the difference between calls coming from legacy line and voip |
| 14:59.31 | r0m|u | wow I missed a lot. p3nguin how is that compnay's such as http://www.reducefraud.com/ can tell you use voip vs pstn? |
| 15:00.10 | r0m|u | this is why ask the question |
| 15:00.16 | *** join/#asterisk Kernel_Core (~IceChat7@h-213-136-53-142.na.cust.bahnhof.se) |
| 15:00.44 | r0m|u | Than it was brought up that they can be differentiated.... |
| 15:00.56 | Kernel_Core | is there any hope for useing SILK codec and IAX2? |
| 15:01.21 | p3nguin | irroot: You're indicating that there is some way to know if a number has been ported to an ITSP or remains on a telco. |
| 15:01.37 | p3nguin | irroot: Explain to me how anyone would know that information. |
| 15:01.44 | jacc0 | all blockers for 1.8.8.0 are closed(https://issues.asterisk.org/jira/browse/ASTERISK-18499) ; will it be released today? |
| 15:02.02 | irroot | no not at all there is no way to determine this however legislation requires ported numbers to emit a tone on connect |
| 15:02.07 | p3nguin | I could call you from any of 50 phone numbers I have and you're not going to know if I dialed it over copper or VoIP. |
| 15:02.33 | r0m|u | p3nguin, did you see my question? |
| 15:05.26 | r0m|u | p3nguin, To what I understand thats the company the CL uses to verify users. If the systems detects the use of voip than it will flag it. |
| 15:05.55 | ollii | maybe you just ask them |
| 15:06.35 | p3nguin | r0m|u: I don't see any way they could determine if I have ported my landline number from AT&T over to VoIP.ms |
| 15:06.36 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 15:07.35 | schmidts | p3nguin maybe they have just a list of known voip providers and if the call is routed over one of this know voip providers it wil be flaged |
| 15:08.07 | p3nguin | How can they know the route of the call they are originating? |
| 15:08.30 | p3nguin | If I dial a phone number, I can't see the route it takes. |
| 15:08.56 | schmidts | are we talking about inbound or outbound calls? |
| 15:09.07 | r0m|u | p3nguin, I see. A "anonymous" person approach me asking about acquiring 10,000 VoIP numbers for resale. I struck me as unusual I proceed to ask questions. Than he explain to me about CL, reducefraud and how numbers can be flagged. I knew the guys was up to no good. but indeed raced a question about how it all works. |
| 15:09.37 | r0m|u | raised* |
| 15:10.27 | r0m|u | p3nguin, one way they where getting around it was by selling prepaid sim cards from low level wireless carriers. |
| 15:10.35 | schmidts | if you have a lot of ss7 direct connections then you would be able to see where your call will be routed but normally its just going upwards to a real big provder which has this interconnections to mostly all other ss7 carriers |
| 15:11.43 | schmidts | and btw in ss7 you will get a redirect if a number is ported to the target network, the call will not be routed by the anchor (sorry dont know if this is the real name for it) network, only redirected |
| 15:12.04 | irroot | r0m|u these folks congolese / nigerian by any chance ?? |
| 15:12.38 | schmidts | irroot or maybe palistine :D |
| 15:12.59 | irroot | schmidts yeah they active in europe |
| 15:13.19 | irroot | part of johhannesburg is called little lagos |
| 15:13.32 | r0m|u | rolf. Not sure. It was PM over a forum. the forum allows anon posting but it displays the isp your from. the guy "seems" local as the domain is rr.com |
| 15:13.32 | schmidts | all of my fraud cases the last 5 months were coming from there |
| 15:13.37 | p3nguin | I guess I can buy the bit about SS7 and seeing a redirect. |
| 15:14.30 | p3nguin | But if I make a call TO you, how are you going to see if I am a VoIP caller or landline caller? |
| 15:16.06 | schmidts | p3nguin as i said, the only thing which comes to my mind would be the origin network, but i dont think this will really work, cause mostly voip providers use a bigger pstn carrier as an uplink provider so they dont have to make contracts with all other carriers |
| 15:16.13 | r0m|u | p3nguin, I dont think your regular house hold/business can. Its more of 3r psrty service that has to be involve. |
| 15:16.21 | schmidts | so you will only see the ss7 id of this carrier in front of the voip provider |
| 15:16.54 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:16.55 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:18.01 | r0m|u | irroot, its funny you mention that... I was reading about this guy who apparently got hacked not to long ago... http://www.rowetel.com/blog/ |
| 15:18.35 | *** join/#asterisk timeshell (~chatzilla@gw.lusi.on.ca) |
| 15:18.42 | timeshell | How to change your email address in JIRA? |
| 15:19.18 | p3nguin | seraphie |
| 15:19.43 | r0m|u | p3nguin, can I ask you a quick question regarding voip.ms? |
| 15:19.52 | dym | touches p3nguin |
| 15:20.00 | dym | (: |
| 15:20.10 | p3nguin | Sure, but I can't promise I know the answer. |
| 15:20.11 | dym | throws p3nguin a lustful look |
| 15:20.49 | p3nguin | Just keep your hands off my penis. |
| 15:21.01 | timeshell | This IS a public forum. |
| 15:22.45 | r0m|u | cool. Here is what I want to do. I want to take the disa function away from my gsm gateway and use it sole for cell calls and as back up gateway just in case internet fails. the same goes for incoming calls I just want to use it to be able to recive calls and make calls as backup. Is it possible I can create a sub account in voip.ms and make it route internally to my primary number and gain access to my asterisk server to be able to dial out extensions? |
| 15:23.09 | *** join/#asterisk zamba (marius@flage.org) |
| 15:23.10 | r0m|u | rofl! hahahaha! |
| 15:23.30 | p3nguin | You lost me at sub account. |
| 15:24.08 | r0m|u | p3nguin, I can call my gsm gateway and get a tone and I am able to dial internal extensions inside my pbx. |
| 15:24.18 | p3nguin | right |
| 15:24.27 | zamba | r0m|u: which gsm gateway? |
| 15:24.30 | p3nguin | That's what DISA does. |
| 15:24.37 | p3nguin | secondary dial tone |
| 15:25.02 | r0m|u | I want to dom something similar with voip.ms. I want to call it via a sub account and be able to reach my asterisk server and get a tone via disa. |
| 15:25.18 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 15:25.19 | *** join/#asterisk mmcji (~mmcji@65.172.54.254) |
| 15:25.21 | r0m|u | zamba, PorTech MV-370 |
| 15:26.11 | zamba | r0m|u: aight, ok.. i'm using a dinstar myself |
| 15:26.39 | p3nguin | You can set up internal extensions in VoIP.ms, and then you can put your gateway on a sub account... and then dial Asterisk's voipms extension. Is that what you mean? |
| 15:26.50 | r0m|u | p3nguin, the more I talk about it the more I realize is not possible without been charged. |
| 15:27.03 | p3nguin | Calls between accounts are always free. |
| 15:27.22 | r0m|u | zamba, how is it working out for you? |
| 15:28.27 | r0m|u | p3nguin, thats not a bad idea. I guess I could do it that way. though Is it possible to activate disa base on the cid calling in? |
| 15:28.32 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 15:29.07 | r0m|u | all other cid's go over there. one specific cid gets disa. |
| 15:29.15 | p3nguin | I'm not sure if you can do it on voipms, but you can certainly do it on asterisk. |
| 15:29.46 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:29.49 | zamba | r0m|u: terrible sound quality |
| 15:29.55 | zamba | r0m|u: but i haven't done very much testing |
| 15:30.22 | r0m|u | Yes I want to do it in asterisk. Ill look in to it. I took your advice about all cell call threw the gsm gateway and now I want to do it inwards as well. since is all free |
| 15:30.35 | *** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby) |
| 15:30.37 | wcselby | o/ |
| 15:30.38 | r0m|u | zamba, the portech sound quality is superb! |
| 15:31.15 | zamba | r0m|u: interesting.. but that's the price? |
| 15:32.03 | r0m|u | zamba, well its a bit on the high price. some where around 250.00+ dollars |
| 15:32.19 | zamba | yeah, i see now.. £ 182 |
| 15:32.24 | r0m|u | It can also do mass sms which is a plus. |
| 15:32.41 | zamba | how's the configuration? |
| 15:33.02 | r0m|u | I have it all scripted and all incoming sms route to my email as well as txt. |
| 15:33.12 | r0m|u | zamba, very simple. not complicated at all. |
| 15:33.15 | *** join/#asterisk Katty (~Katty@mail.copi-rite.com) |
| 15:33.18 | Katty | HAI LOVABLES |
| 15:33.45 | r0m|u | zamba, I had more of a hard time making the dial plan that I did configuring the box :) |
| 15:35.39 | r0m|u | I can create acl's base on cid's on the gateway which is a plus. in top of that I added secure disa for incoming calls that way I can dial out long distance at voip prices. |
| 15:36.03 | r0m|u | I also have set as a backup trunk |
| 15:36.22 | r0m|u | no internet. all call goe out threw the gsm gateway. |
| 15:37.48 | wcselby | o/ Katty |
| 15:38.36 | *** join/#asterisk DanFromUK (~DanFromUK@2.27.27.255) |
| 15:38.42 | DanFromUK | hi all. |
| 15:38.57 | Katty | hugs on wcselby |
| 15:39.00 | Katty | hai dan! |
| 15:39.08 | DanFromUK | is there any way to reload cdr_addon_mysql.so without restarting asterisk? |
| 15:39.51 | wcselby | module reload cdr_addon_mysql.so ? |
| 15:40.15 | DanFromUK | ill give it a try. it didnt come up when i hit tab. |
| 15:40.31 | DanFromUK | ah |
| 15:40.33 | DanFromUK | Module 'cdr_addon_mysql.so' does not support reload |
| 15:40.37 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:40.42 | DanFromUK | never mind. i'll have to wait till this evening. |
| 15:40.58 | DanFromUK | thanks |
| 15:46.30 | *** join/#asterisk umay (~chris@67-6-158-37.hlrn.qwest.net) |
| 15:48.53 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 15:54.39 | r0m|u | today comcast phone stop working with a "this number is no longer in service" I call CC support and I was told that there is an issue and that engineers are working on it. I wonder wtf is going on. |
| 15:54.58 | r0m|u | they dont want to release information. |
| 15:57.54 | Qwell | my guess? there's an issue, and they're working on it |
| 15:58.17 | Kobaz | heh |
| 15:58.27 | Kobaz | usually you can ask the tech what was wrong |
| 15:58.31 | Kobaz | and they will tell you something |
| 15:58.36 | Qwell | "something" |
| 15:58.36 | wcselby | r0m|u the issue is that you're using Comcast |
| 15:58.38 | wcselby | :) |
| 15:58.41 | Qwell | it won't be correct, but it'll be something. :p |
| 15:58.47 | Kobaz | hah, yeah sometimes |
| 16:00.00 | r0m|u | wcselby, I dont use it as my primary number. Its my alarm system |
| 16:00.56 | r0m|u | lol Qwell |
| 16:01.16 | r0m|u | I only wonder because they are a big provider and is unusual for this to happen. |
| 16:01.29 | wcselby | r0m|u I was being silly. i have had bad experiences with comcast over the past few weeks |
| 16:01.37 | wcselby | but for cable tv service, not phone |
| 16:02.09 | r0m|u | wcselby, fuck my internet just went out! |
| 16:02.10 | wcselby | in fact, you reminded me I had to follow up with someone over at Comcast, so thanks :) |
| 16:02.23 | r0m|u | your welcome? lol :P |
| 16:02.24 | wcselby | heh |
| 16:02.28 | wcselby | afk a few |
| 16:02.41 | r0m|u | My internet at home just went out and this sucks. |
| 16:02.51 | r0m|u | All my remote sessions died |
| 16:03.58 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip24-56-6-80.ph.ph.cox.net) |
| 16:07.30 | r0m|u | p3nguin, do you have problems with voip.ms re registering when your internet goes out? |
| 16:07.45 | p3nguin | no |
| 16:09.54 | r0m|u | I just lost internet and voip.ms cant register. All of the other provider did except voip.ms |
| 16:10.03 | r0m|u | Is all ways that way |
| 16:10.10 | p3nguin | can't, or just hasn't done it yet? |
| 16:10.12 | r0m|u | Is there a way that I can fprce it? |
| 16:10.32 | r0m|u | is in attempt #67 |
| 16:10.55 | r0m|u | It looks like it cant. |
| 16:11.05 | r0m|u | "time out" |
| 16:12.02 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 16:12.22 | p3nguin | See what sip show registry says about it. |
| 16:12.27 | r0m|u | well I forced it with a port change. |
| 16:12.38 | r0m|u | "Request Sent" |
| 16:12.44 | r0m|u | Now is registered |
| 16:13.30 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 16:14.41 | r0m|u | ROFL!!!!! HAHAHAHAHAAH!!!!!! my forwarding number that I set in comcast to forward to got switch as my primary number and my comcast number was given away to somebody else!!!!!!!!!! How in the hell? that number belongs to voip.ms! |
| 16:16.02 | p3nguin | You ported it from Comcast to VoIP.ms? Then Comcast just gave it to someone else? |
| 16:17.02 | r0m|u | no. nether. all I did was go to the comcast voip portal and set it to forward all calls to my voip.ms. Than comcast used the forward number that I put that belongs to voip.ms as my primary number and gave away my comcast number to somebody else. |
| 16:17.50 | r0m|u | so when you call my comcast number you get an error because the account does not exist. |
| 16:18.10 | p3nguin | Tell them to give it back. |
| 16:18.12 | *** join/#asterisk eAndi_ (eAndi_@201.14.144.128) |
| 16:18.12 | r0m|u | Thats what the eng told me just now. |
| 16:18.41 | r0m|u | I did. I just cant explain how in the world this happen. |
| 16:19.09 | Qwell | They gave away your number? |
| 16:19.15 | Qwell | That is fantastic. |
| 16:19.23 | p3nguin | sounds more like took away rather than gave away. |
| 16:19.32 | r0m|u | ^^ |
| 16:19.44 | p3nguin | I think giving to someone else kind of implies that someone else has it and can use it. |
| 16:20.47 | p3nguin | I don't quite understand why they would do either, though. |
| 16:21.47 | Katty | starving :< |
| 16:21.52 | r0m|u | p3nguin, The eng told me that the number was been populated to another account :/ |
| 16:22.15 | p3nguin | Tell him to put it back. |
| 16:22.22 | wcselby | r0m|u tell them you never authorized that, and you want it back. if they won't, ask to speak with a supervisor |
| 16:22.54 | wcselby | there are actualy legal terms about this sort of thing |
| 16:23.06 | wcselby | i don't recall off the top of my head, but "slamming" may be close |
| 16:23.41 | *** join/#asterisk lhfnet (~lhfnet@net-cdd-fw01.cddlasmercedes.com) |
| 16:24.05 | r0m|u | wcselby, excuse my ignorance but what is "slamming"? |
| 16:24.24 | r0m|u | damn my internet went out again |
| 16:24.24 | wcselby | http://www.fcc.gov/encyclopedia/slamming |
| 16:24.54 | r0m|u | ah! |
| 16:25.09 | wcselby | or http://en.wikipedia.org/wiki/Telephone_slamming |
| 16:25.13 | lhfnet | Hi, I am having problems with the stdexten macro in Asterisk 1.8, I can put it to work, if I do an dial plan show (exten)@(context) i get Dial(${HINT}) instead of the macro stdexten activation |
| 16:25.38 | wcselby | lhfnet please show us your dialplan using pastebin |
| 16:25.40 | wcselby | ~pb |
| 16:25.40 | infobot | [~pb] 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.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 16:25.44 | r0m|u | wow. thanks for the info. I doubt I can mention to comcast "legal" I am just another customer :( |
| 16:25.58 | wcselby | sure you can |
| 16:26.16 | wcselby | tell them what they've done is illegal and sounds a lot like slamming, and if they don't return your number you'll report them to the FCC |
| 16:26.35 | wcselby | the FCC takes those kinds of complaints seriously and will come down on Comcast hard, and they want to avoid that |
| 16:27.06 | wcselby | well let me rephrase, the FCC *could* come down hard on Comcast based on your complaint, if they find it has merit |
| 16:27.44 | r0m|u | Mhhhh interesting.... |
| 16:27.48 | r0m|u | ponders |
| 16:27.52 | lhfnet | http://pastebin.com/0PacavRg |
| 16:28.02 | wcselby | don't bring it up unless they won't return the number |
| 16:29.15 | wcselby | lhfnet which context are your phones in? Or which context are you expecting to call the std-exten macro from? |
| 16:29.50 | wcselby | because you never call it in the dialplan you pasted |
| 16:29.59 | lhfnet | wcselby international, mobile, national, local, internal depending on the privileges of the user |
| 16:30.26 | r0m|u | got it wcselby. Thanks for the info. |
| 16:31.20 | lhfnet | wcselby if I deactivate the default context they can't call each other, but I don't know where I set for a normal call to use stdexten instead of Dial(${HINT}) |
| 16:31.20 | Qwell | wcselby: slamming is more like adding services to an account, like switching long distance providers |
| 16:31.39 | dym | Then again - Qwell has been known to lie. |
| 16:31.44 | dym | hides |
| 16:31.47 | r0m|u | lol |
| 16:31.55 | Qwell | dym: I lie all the time. |
| 16:32.06 | dym | fact! |
| 16:32.17 | wcselby | lhfnet you've never called the Macro() anywhere in your dialplan |
| 16:32.20 | *** join/#asterisk alemos (~Adium@62.28.143.10) |
| 16:32.29 | wcselby | so it's never going to know to execute the stdexten macro |
| 16:32.48 | p3nguin | lhfnet: Contexts are assigned PER PEER. |
| 16:33.00 | p3nguin | If you want a peer to use a different context, change the context you have set for that peer. |
| 16:33.12 | alemos | is there a diagram with the workflow of the manager events when a ZAP call is inbound? |
| 16:33.14 | wcselby | Qwell I realize that, but it's close. it's about switching things on your account, which this loosely falls under. the point is, it should freak out a supervisor enough to look into the issue a little more |
| 16:33.50 | lhfnet | wcselby: I did that also but I got the Dial(${HINT}) in first priority than the macro |
| 16:33.58 | Qwell | Here's the problem. If it's illegal to take a number from a customer and give it to somebody else... What is the appropriate solution for a number that has already been given to somebody else? Take it from them? Illegally? |
| 16:34.03 | wcselby | Dial(${HINT}) is not a call to the macro |
| 16:34.33 | wcselby | Qwell if you buy something that was stolen, it was still stolen, and you have no legal claim ove rit |
| 16:34.36 | wcselby | over* |
| 16:34.56 | Qwell | He didn't own the number. Comcast did. |
| 16:35.23 | Qwell | Now, I'm not saying that they should be raked over coals by the FCC. They absolutely should. |
| 16:35.33 | Qwell | err, that they shouldn't be* |
| 16:35.34 | wcselby | so the other customer that was inappropriately assigned the number doesn't own it either |
| 16:36.38 | Qwell | My point is... make sure you manage your expectations. |
| 16:36.48 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 16:36.52 | lhfnet | wcselby: I understand, what I trying to say is that if I call the stdexten macro in the default context in extensions.conf like exten => _XXXX,1,Macro(stdexten,${MACRO_EXTEN}) and then I use the command show dialplan (EXTENSION)@(CONTEXT) it shows the Dial(${HINT}) in the first priority and the Macro in the second, so the macro does not start |
| 16:37.41 | lhfnet | wcselby: My problem is that I don't know where the Dial(${HINT}) comes from |
| 16:37.44 | wcselby | lhfnet then fix your dialplan. you're doing it wrong |
| 16:38.13 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v021-124.mobile.uci.edu) |
| 16:38.19 | Katty | pokes Qwell |
| 16:38.47 | wcselby | lhfnet so what extension are you doing "dialplan show EXTEN@CONTEXST", i.e show us a copy / paste from the CLI of you doing just that |
| 16:39.00 | wcselby | and the output of th ecommand |
| 16:39.25 | p3nguin | lhfnet: Don't use the default context for anything. |
| 16:39.31 | wcselby | Qwell obviously :) I'm saying he should have an expectation of getting the number back going into the call |
| 16:39.48 | p3nguin | lhfnet: Set up appropriate contexts, then assign contexts to peers. |
| 16:40.10 | wcselby | if comcast fails to do that, he needs to elevate the issue to a super, if the super doesn't want to do that, mention the fcc and slamming. if the super still doesn't care, contact the fcc and file a complaint |
| 16:40.58 | wcselby | p3nguin the thing is, he has appropriate contexts setup in his dialplan, but he keeps talking about the default context, so I'm not sure what he's doing |
| 16:41.24 | p3nguin | <wcselby> lhfnet then fix your dialplan. you're doing it wrong <--- he's doing more than just dialplan wrong. |
| 16:42.52 | lhfnet | http://pastebin.com/SjiFJiaq |
| 16:43.04 | lhfnet | here the dialplan show result |
| 16:43.10 | lhfnet | of the extension 1010 |
| 16:43.36 | lhfnet | as you can see, the Dial(${HINT}) is first, and I didn't set this anywhere |
| 16:44.40 | wcselby | the dialplan you pasted doesn't match what your dialplan show is revealing |
| 16:44.57 | wcselby | have you reloaded your dialplan since you last made changes? |
| 16:45.05 | lhfnet | yo told me to add the call to the stdexten and I did it |
| 16:45.53 | r0m|u | wcselby, They are giving back mynumber in 24 to 72hrs and a tech has to come to my house to provision my modem. They fucked up. |
| 16:46.02 | lhfnet | I add this line to the default context section in extensions.conf: exten => _XXXX,1,Macro(stdexten,${MACRO_EXTEN}) |
| 16:46.03 | *** join/#asterisk Hanumaan (~Hanumaan@132.230.17.77) |
| 16:46.19 | *** join/#asterisk singler (~singler@84.15.187.216) |
| 16:46.20 | wcselby | what you showed me of your dialplan, your default context has two extens and no pattern matching. what you showed me of your dialplan show output, there's hint definitions and pattern matching going on, so I'm not sure...... |
| 16:46.23 | r0m|u | they fucked up I say. God...... what a cluster fuck. |
| 16:46.34 | [TK]D-Fender | lhfnet, Stop showing 1 little line at a time and pastebin your entire dialplan |
| 16:46.45 | wcselby | r0m|u that's good to hear, sorry about the fuckup |
| 16:47.29 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 16:47.51 | r0m|u | wcselby, mentioning a supervisor was the key. things got moving a bit quicker. |
| 16:48.11 | wcselby | r0m|u lol that's dealing with customer service 101 - always mention the super :) |
| 16:48.22 | r0m|u | lol |
| 16:48.25 | lhfnet | http://pastebin.com/ft3SAY8M |
| 16:48.39 | pigpen | Hi all. Anyone know if you can reflash a Allworx 9112 with the Aastra 9112 to work successfully with asterisk? I prefer Polycom, but I believe funding is an issue |
| 16:49.45 | Qwell | reflash to work with Asterisk? SIP is SIP is SIP (except when it's Cisco SIP) |
| 16:50.02 | pigpen | heh, tks. |
| 16:50.03 | wcselby | lol @ Qwell |
| 16:50.15 | Qwell | oh! |
| 16:50.29 | Qwell | The Cisco/Linksys dude at Astricon told me that the 79xx series phones no longer officially support SIP. |
| 16:50.34 | wcselby | and really only cisco 79xx sip isn't sip....the spa 5xx sip line works great! |
| 16:50.45 | Qwell | wcselby: yeah, those aren't legit Cisco though |
| 16:50.56 | Qwell | those guys aren't bloody idiots. They're just regular idiots. |
| 16:51.04 | wcselby | lol |
| 16:51.42 | wcselby | lhfnet give me the output of dialplan show in a pastebin please |
| 16:52.16 | lhfnet | http://pastebin.com/9EaW11hg |
| 16:52.41 | wcselby | just dialplan show |
| 16:53.27 | Qwell | How does ${HINT} get set? O.o |
| 16:53.47 | wcselby | Qwell that's what I'm wondering, hence why I've asked to see the entire dialplan show output |
| 16:54.05 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-yvvuatuvzlpafyjf) |
| 16:54.21 | *** join/#asterisk azv4 (~azv4@50-73-175-237-pennsylvania.hfc.comcastbusiness.net) |
| 16:54.27 | lhfnet | Qwell: that is my question |
| 16:54.32 | azv4 | any phone system sales people out there? I need a quote! |
| 16:54.39 | Qwell | lhfnet: If you don't know, why are you trying to use it? |
| 16:54.42 | *** join/#asterisk azv4 (~azv4@50-73-175-237-pennsylvania.hfc.comcastbusiness.net) |
| 16:54.51 | Qwell | azv4: $0. |
| 16:54.56 | wcselby | azv4 lol do you need some contact info? |
| 16:55.05 | [TK]D-Fender | lhfnet, Now PB "dialplan show" All of it. |
| 16:55.09 | wcselby | lhfnet please just type "dialplan show" on the cli, then pastebin the output |
| 16:55.24 | *** part/#asterisk lhfnet (~lhfnet@net-cdd-fw01.cddlasmercedes.com) |
| 16:55.29 | azv4 | I just need a general idea how much it will cost to buy an IP based phone system |
| 16:55.40 | wcselby | 500 bucks |
| 16:55.50 | wcselby | that's one server and one small phone |
| 16:55.50 | [TK]D-Fender | azv4, The price of whatever computer you install * on. |
| 16:55.53 | Qwell | azv4: $0. |
| 16:55.59 | Qwell | asterisk.org/downloads/ |
| 16:56.01 | wcselby | give us some details :) |
| 16:56.03 | [TK]D-Fender | 3 easy payments of $49.95 |
| 16:56.12 | azv4 | 5 users, Avaya IP Office please |
| 16:56.15 | azv4 | 55 users |
| 16:56.29 | Qwell | Why would we know the pricing of Avaya? Why would you even ask about that here? |
| 16:56.37 | [TK]D-Fender | azv4, ... This isn't #avaya |
| 16:56.38 | azv4 | only phone channel around |
| 16:56.39 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 16:56.45 | [TK]D-Fender | azv4, Go call a local reseller |
| 16:57.05 | shido6 | I have coupons for that $10,000 /mo service contract somewhere |
| 16:57.08 | [TK]D-Fender | azv4, We laugh at Avaya's crap |
| 16:57.14 | Qwell | I love people that enjoy throwing money at proprietary garbage. |
| 16:57.38 | Qwell | We should start charging for app_voicemail. We'd make millions. |
| 16:57.40 | azv4 | Our company needs something that works without bugs and random crap where you wait 3 days to never for support |
| 16:57.56 | *** mode/#asterisk [+b *!*azv4@*.hfc.comcastbusiness.net] by Qwell |
| 16:57.56 | *** kick/#asterisk [azv4!~north@pdpc/sponsor/digium/Qwell] by Qwell (bye) |
| 16:58.03 | wcselby | azv4 contact digium and look for switchvox |
| 16:58.06 | wcselby | oh |
| 16:58.09 | wcselby | well, nevermind then |
| 16:58.24 | wcselby | i suppose he wont' be looking at any digium solutions then |
| 16:59.05 | Qwell | he was only here to troll *shrug* |
| 16:59.41 | *** mode/#asterisk [-b *!*azv4@*.hfc.comcastbusiness.net] by Qwell |
| 17:01.56 | r0m|u | rofl! kicked in tha face! |
| 17:02.07 | r0m|u | iiiinnnn thhhhhaaaa ffffaaaaccccceee! |
| 17:02.10 | irroot | gets a troll grenade |
| 17:02.25 | r0m|u | lol |
| 17:04.39 | wcselby | did lhfnet leave after we asked him for more info or did he give a reason? |
| 17:04.51 | Qwell | * lhfnet (~lhfnet@net-cdd-fw01.cddlasmercedes.com) has left #asterisk |
| 17:05.19 | [TK]D-Fender | wcselby, Yeah he just quit. |
| 17:05.24 | wcselby | lovely |
| 17:05.28 | wcselby | ah well |
| 17:06.07 | [TK]D-Fender | wcselby,Probably "I realize I made a dumb mistake and don't even want to own up to it" |
| 17:06.21 | wcselby | lol |
| 17:06.24 | wcselby | yeah |
| 17:07.37 | wcselby | http://imgur.com/Odluu nice warni9ng |
| 17:08.50 | *** join/#asterisk navaismo (~navaismo@187.170.0.233) |
| 17:12.04 | umay | howdy y'all |
| 17:12.18 | umay | anybody get sonicwall to not kill IAX ? |
| 17:12.45 | Qwell | kill it how? I can't imagine it being any more difficult than opening the single port.. |
| 17:13.19 | umay | seems to time out IAX connections, regardless of qualify settings |
| 17:13.39 | umay | actually regardless of active call or not |
| 17:14.03 | umay | in one instance, getting 10-20 seconds dropped audio, then comes back |
| 17:14.11 | shido6 | boost your timeout to something more than the registration refresh period. |
| 17:14.21 | umay | on active call tho ? |
| 17:14.27 | shido6 | Firewall > Advanced : Default udp conenction timeout |
| 17:19.00 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 17:20.24 | *** join/#asterisk hardwire (~spencersr@cl-36.anc-01.us.sixxs.net) |
| 17:23.09 | r0m|u | wcselby, Another call from comcast eng. They said that my voip.ms was in pool to be ported over. |
| 17:25.55 | Qwell | *that* would have been slamming. |
| 17:26.42 | *** join/#asterisk serafie (~erin@nat/digium/x-xketpkglpazwnjcc) |
| 17:28.04 | r0m|u | Qwell, i GOT A CASE? |
| 17:28.08 | r0m|u | ops |
| 17:28.10 | r0m|u | sorr for the caps |
| 17:28.30 | r0m|u | I am about to mention "slamming" I want to see how far I can go |
| 17:28.35 | Qwell | r0m|u: Only if they had successfully taken your number from your ITSP. |
| 17:28.47 | *** join/#asterisk rdahlin_1 (~rdahlin_1@78-73-19-238-no168.tbcn.telia.com) |
| 17:28.54 | r0m|u | so I sort of caught it on time? |
| 17:29.01 | *** join/#asterisk luckman212 (~irc@pool-108-41-8-176.nycmny.fios.verizon.net) |
| 17:29.05 | Qwell | yes |
| 17:29.07 | *** join/#asterisk dwayne (~dwayne@c-76-29-230-45.hsd1.al.comcast.net) |
| 17:29.11 | *** join/#asterisk Deeewayne (~dwayne@c-76-29-230-45.hsd1.al.comcast.net) |
| 17:29.11 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 17:29.51 | r0m|u | ah... I see. |
| 17:30.35 | wcselby | lol |
| 17:30.40 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 17:30.42 | wcselby | you sound so diappointed |
| 17:33.01 | r0m|u | lmao |
| 17:33.51 | r0m|u | I am. I wanted to shout... You practice slamming! Very illegal in the US! Do you know what that means?!? |
| 17:34.32 | r0m|u | and that I would of waited for the supervisor to shit on hes pants and escalate the call to corporate |
| 17:34.39 | r0m|u | than* |
| 17:35.04 | wcselby | lol |
| 17:35.04 | wcselby | you mean then* |
| 17:35.04 | wcselby | :P |
| 17:35.30 | r0m|u | I been escalated to corporate before. They get things done quick and professionally :) |
| 17:35.34 | r0m|u | ooo yea that |
| 17:35.35 | r0m|u | lol |
| 17:36.05 | *** join/#asterisk xpot (~xpot@166-70-100-198.ip.xmission.com) |
| 17:36.15 | wcselby | heh :) |
| 17:36.17 | *** join/#asterisk xpot-mobile (~james@dhcp69.emcb.utah.edu) |
| 17:46.31 | sorressean | Someone mind taking a look at this? These are the three files in /etc/asterisk, I'm trying to just get it to bind and confirence anyone that calls in, but it doesn't bind at all. I'm not sure if I have an error or what's up: http://pastebin.com/UTPJSHZ8 |
| 17:46.43 | Qwell | r0m|u: You probably aught to call your ITSP and tell them not to release the number. Explain the issue. |
| 17:47.15 | r0m|u | voip.ms? |
| 17:47.16 | Qwell | sorressean: I suspect you need a modules.conf |
| 17:47.18 | Qwell | r0m|u: yes |
| 17:47.21 | wcselby | if the itsp isn't a lec of some sort, they will have no control over if the number gets ported |
| 17:47.28 | sorressean | nods |
| 17:47.30 | wcselby | it will happen above their heads |
| 17:47.35 | Qwell | wcselby: fun |
| 17:47.44 | wcselby | i know this from personal experience, btw |
| 17:47.53 | Qwell | I had a number taken from me once. My ITSP couldn't do a thing about it. :( |
| 17:47.59 | r0m|u | ok you guys are freaking me out.... so who do I call? |
| 17:48.05 | Qwell | 800-4latimes >.> |
| 17:48.25 | wcselby | call your lec and ask them to notify you if they receive a port out request, but they don't always receive those, depending on who the upper level lec is |
| 17:48.31 | Qwell | it was in the list of available numbers, so I had them reg it for me. It worked for about 2 days. |
| 17:49.24 | wcselby | we're a small hosting company and we've got roughly 900 tn's spread out over multiple carriers. i was auditing our numbers and noticed that several of the ones we're paying for monthly, are no longer routed through our carriers. |
| 17:49.48 | r0m|u | ok let me call voip.ms and see where I can go with this.... |
| 17:49.51 | wcselby | we're actually not a hosting company (not our primary business), but a side of the business they started a while back included hosted pbx type stuff |
| 17:49.52 | r0m|u | brb |
| 17:50.10 | paulc | Looking for a recommendation on SIP DECT phones.. Panasonic vs Gigaset S675 IP.. leaning towards the latter, perhaps.. anyone got any thumbs up or down for either? |
| 17:50.25 | wcselby | the panasonic phones looked cool at astricon |
| 17:50.32 | wcselby | that's about my only experience though |
| 17:55.00 | sorressean | sweet. so it's actually listening now. so like, is there something I need to do to load my extensions? |
| 17:55.05 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 17:55.12 | paulc | wcselby: thanks - I half think the Gigaset's have more functionality and look nicer.. but at the same time Panasonic are well known for decent cordless.. their PBXs aren't bad either.. |
| 17:55.15 | sorressean | it's just extensions.conf, there's not an extensions.eel |
| 17:55.48 | Katty | QWELL |
| 17:59.25 | r0m|u | just got of the phone with voip.ms... I was instructed to email there lnp department |
| 18:01.10 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 18:01.27 | p3nguin | ... and we're back. |
| 18:02.06 | r0m|u | guys I have been improving my dial plan as I keep learning more and more.... I wanted to see if you guys can tell me if this a good practice dial plan: http://pastebin.com/raw.php?i=T6H7tDq8 |
| 18:02.12 | Qwell | KATTY |
| 18:02.47 | *** join/#asterisk batphone (~boo@unaffiliated/batphone) |
| 18:03.13 | r0m|u | p3nguin, you missed the part where comcast had my number in for port over.... |
| 18:03.24 | p3nguin | What were they doing? |
| 18:03.26 | *** join/#asterisk fiz- (~fiz@89-69-231-34.dynamic.chello.pl) |
| 18:03.28 | r0m|u | without my request |
| 18:04.05 | r0m|u | They said "that some how" my forwarding number was requested to be port over to them |
| 18:04.31 | wcselby | r0m|u it's good, but there are things you could be doing to make it smaller |
| 18:04.37 | Katty | Qwell: :> |
| 18:04.37 | r0m|u | any who.... now I have to write to voip.ms to make sure they stop just in case they still try |
| 18:04.39 | Katty | glomps Qwell |
| 18:04.44 | Qwell | is glomped |
| 18:04.51 | r0m|u | wcselby, Please advise me :) |
| 18:04.52 | paulc | wonders what glomping is |
| 18:05.01 | Qwell | paulc: it's hot, is what it is |
| 18:05.08 | p3nguin | I thought you said you had NOT ported that number into voipms. |
| 18:05.15 | paulc | ..because it's Katty - of course! ;-) |
| 18:05.20 | Qwell | p3nguin: other way around |
| 18:05.26 | Qwell | p3nguin: Comcast is trying to take his number *from* voip.ms |
| 18:05.33 | r0m|u | ^^ |
| 18:05.35 | Qwell | (a second number) |
| 18:05.38 | p3nguin | That wasn't what he told me originally. |
| 18:05.44 | Qwell | which is what caused the release of the first number |
| 18:05.58 | Qwell | I'm surprised one was done before the other... |
| 18:06.07 | r0m|u | p3nguin, I guess I didnt expressed it correctly but yes Qwell is right |
| 18:06.31 | Katty | paulc: a glomp is a running jump/pounce |
| 18:06.32 | *** join/#asterisk lowtek (~grandpapa@99.175.248.81) |
| 18:06.41 | Katty | paulc: i am PRO at the glomp |
| 18:06.52 | Katty | paulc: i think helps that i'm short. maximum glompness |
| 18:07.08 | wcselby | r0m|u this line in your outbound context - "exten => _NXXNXXXXXX,1,Set(TRUNKCHECK=0)" could just be rewritten to be "exten => _NXXNXXXXXX,1,Goto(1${EXTEN})" and you don't need to repeat all the same stuff. pattern matching in your internal context. also, you need to rework your inbound fax detection, you've got NUMBER,1,... twice in the same context, and it's because you're trying to do fax detection. |
| 18:07.32 | paulc | Katty: haha fair play :-) |
| 18:07.56 | wcselby | that was just from a quick glance :) |
| 18:08.24 | r0m|u | thanks wcselby! Ill oook in to it. I do intend to shorten it out... its massive :/ |
| 18:08.34 | *** join/#asterisk michael-i (~anonymous@204.11.230.58.static.etheric.net) |
| 18:08.55 | wcselby | np |
| 18:09.33 | lowtek | Hey guys, what's the best way to keep track of a loop in a 1.8.x dialplan? Just increment ${X}? |
| 18:09.53 | Qwell | lowtek: pretty much. see While() |
| 18:10.24 | lowtek | Qwell - oooh, while, missed that one ... tnx! ;) |
| 18:11.19 | michael-i | Hi all. I'm having trouble with vm2email suddenly. I'm using msmtp and Asterisk v10. I can send mail on the command line using the same config but sending via Asterisk fails. debug==10000 shows the sendmail function in app_voicemail reporting mail sent correctly |
| 18:11.47 | michael-i | Any other debug tips? I'm in low-sleep frustration mode, apologies if I've missed something. |
| 18:12.22 | wcselby | anything in /var/log/maillog or whatever the log file is for msmtp? |
| 18:12.59 | michael-i | Gotta check the source for a length limit to the mailcmd string. Just realized this is a monster command I'm constructing. |
| 18:13.09 | michael-i | wcselby: nothing…very frustrating |
| 18:13.18 | wcselby | as in, nothing at all? |
| 18:13.23 | wcselby | or just no errors? |
| 18:14.03 | michael-i | wcselby: an empty, mocking file |
| 18:14.28 | michael-i | just saw mailcmd[160] :) this HAS to be it |
| 18:14.42 | michael-i | sorry for the typed inner-monologue |
| 18:15.01 | wcselby | r0m|u also, you've got a lot of instances of pattern matching without any regex patterns (e.g _311 or _411). You don't need to use the _ option unless you're actually going to match against a regex pattern (i.e you could do _[34]11) |
| 18:15.33 | wcselby | michael-i is there anything in the mail log from when you manually send the mail from the command line?\ |
| 18:15.52 | p3nguin | If you have only explicit extensions, do not use an _ on the front. |
| 18:16.05 | wcselby | r0m|u otherwise just use 311 or 411, since that's the entireity of the extensions |
| 18:16.09 | wcselby | extension* |
| 18:16.11 | michael-i | wcselby: yes, that works perfectly. I just looked at my msmtp mailcmd in voicemail.conf and it's 280 chars long |
| 18:16.17 | wcselby | yeah, p3nguin said it much better than I just did :) |
| 18:16.21 | michael-i | time to write a quick patch |
| 18:16.32 | wcselby | michael-i cool |
| 18:16.49 | Katty | gets her dance on |
| 18:17.04 | Katty | need moar caffeines! |
| 18:17.09 | Katty | grooves over to soda machine |
| 18:17.10 | lowtek | lol, will $[${X}++] work? |
| 18:17.45 | Katty | crap i need a dime :< |
| 18:17.52 | lowtek | bag? |
| 18:17.58 | Katty | sooo not cool soda machine. |
| 18:18.05 | Katty | why can't they take debit cards? |
| 18:18.14 | rdegges | Hrm, I'm getting a weird error: "asterisk.c:1337 listener: Unable to create pipe: Too many open files". |
| 18:18.20 | wcselby | Katty i've seen machines that do that |
| 18:18.23 | rdegges | And I've only got ~100 calls on this box :o |
| 18:18.29 | wcselby | Katty they charge like 1.25 for a soda though |
| 18:18.31 | Qwell | Katty: I stayed in a hotel once that had those. |
| 18:18.32 | lowtek | rdegges: ulimit |
| 18:18.50 | Qwell | they put a hold of like $50 on my account to buy a damn soda, every time I used it. |
| 18:18.51 | Katty | okay well why can't they be here in this itty bitty city in missouri?! pbbbffft |
| 18:19.03 | rdegges | lowtek: cat /proc/sys/fs/file-max shows 1604167 |
| 18:19.07 | wcselby | Qwell - OUCH! I never thought to check that |
| 18:19.10 | rdegges | That's enormous--I'm not using that many files. |
| 18:19.22 | lowtek | rdegges: it's prolly set less for the user running asterisk |
| 18:19.30 | rdegges | lsof | grep asterisk | wc -l outputs 1211 |
| 18:19.38 | rdegges | How do I set it on a per-user basis? :o |
| 18:19.59 | r0m|u | wcselby, good catch. Thanks |
| 18:20.14 | umay | rdegges: are you using the FILE() function ? |
| 18:20.19 | lowtek | rdegges: what distro? |
| 18:20.21 | r0m|u | _[349]11 |
| 18:20.23 | rdegges | umay: let me look really quick |
| 18:20.41 | rdegges | umay: yah I am, actualy :o |
| 18:20.44 | Katty | eppigy: buy me a soda! |
| 18:21.01 | rdegges | lowtek: ubuntu-server, 11.04 |
| 18:21.05 | umay | i think theres a bug in that function |
| 18:21.10 | rdegges | umay: :ooooooo |
| 18:21.12 | Katty | i bet i'd ask eppigy for a soda and he'd buy me a whole cube |
| 18:21.13 | umay | prolly should go up on the issue tracker |
| 18:21.21 | Katty | he seems like that kinda guy |
| 18:22.26 | Qwell | lowtek: ${INC(X)} |
| 18:22.45 | rdegges | umay: dude, thanks for that. I never would have thought of that as a possible issue. |
| 18:22.45 | wcselby | so, there's been a "management" meeting going on in the next room over all morning (I'm the only non-mgmt employee). it's suddenly very quiet....I think they went ot lunch without me. :/ |
| 18:22.52 | rdegges | but I just realized that there is no FCLOSE() or anything |
| 18:22.57 | lowtek | Qwell ... just as good and thanks again ;) |
| 18:23.01 | rdegges | So that makes sense that asterisk would keep the file descriptor open |
| 18:23.04 | rdegges | maybe that's fucking this up :( |
| 18:23.16 | Qwell | lowtek: next time there will be a small* fee |
| 18:23.19 | Qwell | (*not small) |
| 18:23.50 | wcselby | Qwell small is a relative term. compared to 1,000,000,000 dollars, 100,000 is actually quite small |
| 18:24.13 | lowtek | lol |
| 18:24.17 | wcselby | so you should say : next time there will be a relatively small fee |
| 18:24.48 | umay | rdegges: i am going to try to add a test for this to the asterisk test suite in #asterisk-testing |
| 18:24.53 | wcselby | (actual example I've seen used) |
| 18:25.01 | rdegges | umay: awesome :o |
| 18:42.12 | michael-i | Well, the patch to app_voicemail.c to expand the mailcmd field to 640 chars works. Debug output reports the entire msmtp monster-command. But, no dice… Still no e-mails sent or logged. |
| 18:48.57 | sorressean | I'm watching in the Asterisk console when I call in and I see MeetMe exited with non error code 1, how do I see why it failed? My exten looks something like: exten => conf,3,MeetMe(1|cdp) |
| 18:49.09 | sorressean | I use d so it'll create the confirence if noone is in it |
| 18:51.02 | batphone | mp3player app does not render sound when i dial into that extension |
| 18:51.07 | batphone | i have logging on verbose/debug |
| 18:51.12 | batphone | nothing indicating a reason why |
| 18:51.13 | p3nguin | sorressean: | should be , |
| 18:51.48 | batphone | should resample mp3 into 8khz? |
| 18:52.11 | sorressean | weird. I seen | in an example, thanks |
| 18:54.53 | sorressean | it still fails though. is there a way to show why? |
| 18:55.00 | p3nguin | core set verbose 3 |
| 18:55.03 | p3nguin | make another call |
| 18:55.06 | sorressean | <PROTECTED> |
| 18:55.09 | p3nguin | Pastebin the output. |
| 18:56.10 | sorressean | http://pastebin.com/m7YFa3R4 |
| 18:58.13 | sorressean | it doesn't give any indication as to why it failed, just exits |
| 18:58.50 | lowtek | hmm.. here's a fun one.. I need to evaluate a string to determine if it's 0123456789*#, is there a fast way to do this or should I just gotoif(x|x|x)? |
| 18:59.03 | lowtek | .. rather .. 0 or 1 or 2 ..... or * or # |
| 19:01.10 | *** join/#asterisk Dibbler (~Dibbler@87-194-103-72.bethere.co.uk) |
| 19:02.03 | p3nguin | sorressean: Do you have dahdi installed and loaded? MeetMe uses the dahdi pseudo channel. |
| 19:02.15 | p3nguin | Also, what was wrong with ConfBridge? |
| 19:02.30 | ChannelZ | infastructure |
| 19:02.53 | eppigy | hands Katty a soda |
| 19:03.28 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 19:04.53 | Katty | :> |
| 19:05.11 | eppigy | :] |
| 19:05.21 | Katty | hugs on eppigy |
| 19:05.34 | eppigy | squinty eye smiles |
| 19:05.37 | Katty | eppigy: i'm not feeling so perky tday |
| 19:05.42 | ChannelZ | sorressean: does the channel exit immediately after the MeetMe executes? |
| 19:05.45 | eppigy | aww thats no good |
| 19:05.49 | Katty | i know :< |
| 19:06.14 | eppigy | i am a little sleepy myself for some reason |
| 19:06.37 | Katty | same. |
| 19:06.49 | Katty | but i didn't keep you awake all night. |
| 19:06.52 | Katty | so not my fault. this time. |
| 19:07.58 | eppigy | lol sadly |
| 19:12.35 | lowtek | lol, is there an easier way to do this: exten => s,n,GotoIf($[${YESNO}=yes | ${YESNO}=0 | ${YESNO}=1 | ${YESNO}=2 | ${YESNO}=3 | ${YESNO}=4 | ${YESNO}=5 | ${YESNO}=6 | ${YESNO}=7 | ${YESNO}=8 | ${YESNO}=9 | ${YESNO}=* | ${YESNO}=#]?s,yes) |
| 19:12.41 | *** join/#asterisk sbrath (~sbrath@unaffiliated/sbrath) |
| 19:13.14 | Qwell | uhh |
| 19:13.15 | sbrath | does anyone know how to setup a BLF hint to monitor a queue, IE: flash a blf light when a queue has members? ( Other than write a script ) |
| 19:13.25 | Qwell | lowtek: Goto(${YESNO}) |
| 19:13.43 | Qwell | exten => _[0-9*#],1,DoStuff() |
| 19:13.49 | lowtek | Qwell: Thought of that, but this is in a macro ... |
| 19:14.12 | lowtek | It's dtmf coming back from SpeechBackground() |
| 19:14.19 | lowtek | trying to catch "any key" |
| 19:14.29 | ChannelZ | What else could someone possibly hit on a keypad that wouldn't match that whole thing? |
| 19:14.32 | Qwell | so then why not just do something if it's empty? |
| 19:14.39 | Qwell | ChannelZ: timeout |
| 19:14.54 | ChannelZ | ah |
| 19:15.10 | ChannelZ | yeah turn it around and do something if it's empty |
| 19:15.34 | lowtek | Qwell: "so then why not just do something if it's empty?" to me? Well, I'm doing something else if there is no response ... |
| 19:15.36 | Qwell | aren't there vars it sets based on success/fail? |
| 19:15.57 | lowtek | Basically "accept this call" -> yes/no or any dtmf key |
| 19:16.25 | Qwell | You just described an else clause. I don't see how this is difficult. O.o |
| 19:16.28 | ChannelZ | your 'if' has an 'else' regardless of which way you look at it |
| 19:17.05 | ChannelZ | if they didn't enter nothing, the else must be that they hit something. |
| 19:17.22 | lowtek | hmm... processing that ... |
| 19:18.29 | irroot | lowtek i put FUNC DIALPLAN_EXISTS together for this purpose |
| 19:19.01 | irroot | create a context with all th bits in it and check it with this func |
| 19:19.59 | irroot | GotoIf(${DIALPLAN_EXISTS(mycont,${INPUT})}?mycont,${INPUT},1) |
| 19:21.04 | lowtek | guess I could just check the length of the returned bit |
| 19:21.15 | lowtek | irroot: ahh! |
| 19:21.18 | lowtek | interesting ... |
| 19:21.22 | lowtek | thanks guys, let me rework this |
| 19:21.23 | lowtek | :) |
| 19:21.51 | michael-i | Finally found my problem. Both the mailcmd buffer and the tmp2 buffer in sendmail() in app_voicemail.c need widening. Now everything's fine. It failed silently before this :) Glad to check this one off!!! |
| 19:22.05 | [TK]D-Fender | sbrath, Use a DEVICE_STATE and add it in your login/out extensions |
| 19:22.35 | [TK]D-Fender | sbrath, And run a counter. Or perhaps just a monitoring scrip that will execute a flag update at a polling frequency |
| 19:24.40 | sbrath | I already have a BLF for queue members, I want to make a light flash when there are people waiting... I think I can just add a macro to the inbound and to the "mambermacro" that will check the QUEUECALLS and set a DEVICE_STATE |
| 19:27.10 | sorressean | p3nguin: I was just trying to see if something else works. ConfBridge wouldn't work for some reason. well, it may ahve just been a client issue, I'll have to have them check. |
| 19:27.55 | lowtek | Ok, easy fix, LEN(${YESNO})=1 |
| 19:28.10 | lowtek | .. and clean |
| 19:28.41 | sorressean | It also never played any sounds. the console said it was playing the onlyoneuser.gsm or whatever that is, but I never heard anything when I tested, either. |
| 19:29.58 | p3nguin | sorressean: Sounds like you didn't configure it for NAT. |
| 19:30.10 | p3nguin | I see you called it from a phone behind a NAT. |
| 19:31.29 | sorressean | weird. ok, I'll look into that. thanks |
| 19:37.24 | *** join/#asterisk Russ (~russ@206.29.182.216) |
| 19:38.08 | *** join/#asterisk beccara (~beccara@mail.ubergroup.co.nz) |
| 19:44.23 | *** join/#asterisk simNIX (~simNIX@156-60.bbned.dsl.internl.net) |
| 19:45.54 | [TK]D-Fender | sbrath, if there are people waiting? Then you'll have to poll the queues... |
| 19:46.30 | Qwell | sbrath: You can use custom hints on the meetme itself. |
| 19:46.51 | [TK]D-Fender | Qwell, Queue, not meetme.. |
| 19:47.05 | Qwell | err, yeah, I totally read queue too. |
| 19:47.17 | Qwell | I wonder if queue provides a devstate |
| 19:47.46 | Qwell | guess not. weird. |
| 19:48.02 | Qwell | it wouldn't be that hard to add |
| 19:52.04 | *** join/#asterisk nix8n82-phone (~AndChat@71-32-137-67.chyn.qwest.net) |
| 19:57.50 | r0m|u | wcselby, you till around? |
| 20:01.35 | *** join/#asterisk b2 (~ion@pdpc/supporter/active/beckb) |
| 20:05.16 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 20:10.18 | p3nguin | Well that's annoying... |
| 20:11.04 | p3nguin | In my vyatta system, I thought RTP would be considered "related" to SIP, so the NAT rule was disabled for the RTP ports. It was blocking audio from remote phones. |
| 20:11.23 | p3nguin | At least that appears to be the cause. |
| 20:11.34 | p3nguin | I enabled the rule and audio worked. |
| 20:11.39 | p3nguin | I guess I should disable it again and make sure it doesn't work. |
| 20:12.14 | *** join/#asterisk lystra (~lystra@hammer.thewrittenword.com) |
| 20:13.28 | p3nguin | That seems to be the problem. Disabled it, no audio again. |
| 20:14.57 | wcselby | r0m|u i'm arond now |
| 20:15.20 | wcselby | i'm around* |
| 20:15.21 | r0m|u | wcselby, I implemented some of your advices.... I am trying to understand one part.... |
| 20:15.46 | r0m|u | _NXXNXXXXXX,1,Set(TRUNKCHECK=0)" could just be rewritten to be "exten => _NXXNXXXXXX,1,Goto(1${EXTEN})" and you don't need to repeat all the same stuff. |
| 20:16.22 | r0m|u | How is that going to help me if I am trying to set a virable as 0 before the context start to be able to determin if I fail over or not |
| 20:17.08 | wcselby | think of it this way |
| 20:17.17 | wcselby | are you doing the same thing if it starts with a 1 or without a 1? |
| 20:18.04 | wcselby | if so, then you don't need to duplicate the entire stanza of code |
| 20:18.24 | wcselby | just use a goto on the first line to point it to the section of code you want it to use |
| 20:18.36 | p3nguin | That's what I told him. |
| 20:19.11 | p3nguin | http://pastebin.com/Piqv4Egj see lines 118-124 |
| 20:19.33 | r0m|u | wcselby, I think thats the problem the putput is nether a 0 or a 1 so I have to cut set it to do a 0 as default an in "OK" a 1 |
| 20:19.34 | p3nguin | But who the fuck ever listens to me?! |
| 20:19.39 | *** join/#asterisk Cubber (~ronny@150.156.193.100) |
| 20:19.53 | wcselby | p3nguin lol |
| 20:19.55 | r0m|u | p3nguin, when did you tell me? |
| 20:19.56 | wcselby | r0m|u like this - http://pastebin.com/z7UA3qd9 |
| 20:20.32 | Cubber | I am trying to add custom files for the default moh in asterik now, however when I remove the default moh files and add my own they do not play, I just get silence. If I add back in the default .ulaw files it works again, but my custom moh files do not play. |
| 20:20.36 | p3nguin | I even include one feor 7-digit dialing. |
| 20:20.38 | r0m|u | wcselby, ah!!!! |
| 20:20.47 | r0m|u | I understand what you mean now |
| 20:20.56 | wcselby | r0m|u it's the same thing that p3nguin does :) |
| 20:21.10 | r0m|u | p3nguin, sorry I never caught your reply? |
| 20:21.16 | wcselby | Cubber how are you adding them? |
| 20:21.17 | p3nguin | It was WEEKS ago. |
| 20:21.23 | wcselby | are you using freepbx or asterisk-gui? |
| 20:21.24 | Cubber | via freepbx |
| 20:21.32 | r0m|u | ~freepbx |
| 20:21.32 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 20:21.37 | wcselby | lol |
| 20:21.40 | r0m|u | lol |
| 20:21.43 | r0m|u | lmao |
| 20:21.46 | wcselby | I've done this once or twice before in freepbx, trying to remember |
| 20:21.48 | r0m|u | p3nguin, sorry!!!!!! :( |
| 20:21.59 | r0m|u | didnt catch it. I am sometimes lost in limbo |
| 20:22.06 | r0m|u | wel most of the time. |
| 20:22.08 | r0m|u | lol |
| 20:22.10 | r0m|u | :P |
| 20:22.50 | r0m|u | p3nguin, your help is much appreciated and I am sure people do listen to masta p3nguin |
| 20:22.53 | wcselby | Cubber are you reloading asterisk / freepbx between when you load the files and when you don't? |
| 20:23.06 | r0m|u | :P |
| 20:23.13 | r0m|u | I do for once! |
| 20:23.15 | wcselby | and when you say you delete them and re-add them, do you mean you're doing this all through the gui, or are you messing around on the box itself, as in command line? |
| 20:23.25 | r0m|u | wcselby, Thanks. |
| 20:23.32 | wcselby | r0m|u np |
| 20:23.43 | Cubber | wcselby yes sir |
| 20:23.53 | p3nguin | I give lots of examples and they get ignored. |
| 20:23.54 | wcselby | i didn't ask a yes or no question.... |
| 20:24.07 | Cubber | wcselby the first one was the second one wasnt |
| 20:24.07 | wcselby | or well I guess my first question was |
| 20:24.10 | r0m|u | I was not following. I though you where talking about the initial context |
| 20:24.12 | wcselby | lol sorry |
| 20:24.28 | wcselby | r0m|u nope - and you've got that repeated somewhere down the file too |
| 20:24.48 | wcselby | in to [to-callcentric] context |
| 20:24.54 | Cubber | wcselby as for the second I am adding and removing through the gui. However I did move the default files in the CLI and then did /etc/init.d/asterisk reload |
| 20:25.05 | r0m|u | wcselby, yes sr. fixing it now. |
| 20:25.15 | wcselby | Cubber do a complete amportal restart |
| 20:25.22 | wcselby | from the command line |
| 20:25.28 | Cubber | wcselby ok |
| 20:25.33 | wcselby | you should never just restart asterisk itself if you're on a freepbx box |
| 20:25.35 | r0m|u | p3nguin, never have I ignore you! :D |
| 20:25.35 | wcselby | weird things happen |
| 20:25.51 | r0m|u | lol you are masta |
| 20:26.07 | r0m|u | your kung fu is strong! |
| 20:26.11 | p3nguin | slams down his iron fist and upsets the channel |
| 20:26.15 | Cubber | amportal restart did not work |
| 20:26.26 | wcselby | Cubber you've at least got the default music now, right? |
| 20:26.28 | r0m|u | bows to the masta! |
| 20:26.35 | p3nguin | Now clean yourselves up! |
| 20:26.41 | r0m|u | rofl! |
| 20:26.43 | r0m|u | hahahah |
| 20:26.46 | Cubber | wcselby no that is in a backup folder right now |
| 20:26.57 | Cubber | wcselby it works if I just restore it then reload asterisk, I get it back that way |
| 20:26.57 | p3nguin | So... |
| 20:27.08 | Cubber | wcselby just custom mp3 or wav files do not play |
| 20:27.12 | p3nguin | Why isn't RTP considered to be related to SIP in mv Vyatta router? |
| 20:27.19 | p3nguin | s/mv/my/ |
| 20:27.27 | wcselby | what are the names of the custom mp3 files? do they have any brackets or parens? |
| 20:27.38 | Cubber | orig_Back In Black.mp3 |
| 20:27.47 | Cubber | is how it was named when uploaded from freepbx |
| 20:28.02 | Cubber | was originally Back In Black.mp3 before I uploaded |
| 20:28.14 | wcselby | Cubber version of asterisk / freepbx ? |
| 20:28.18 | r0m|u | p3nguin, I dont think they are related? are they? |
| 20:28.26 | p3nguin | I was told they are. |
| 20:28.27 | r0m|u | RTP is a general media transport |
| 20:28.32 | Cubber | wcselby asterisknow 1.6 with freepbx 2.9.0.7 |
| 20:28.35 | p3nguin | But I don't think they are. |
| 20:28.41 | r0m|u | so lots of applicatins use it |
| 20:28.46 | p3nguin | That's why I created the rule... but when I was told they ARE related, I set the rule to disabled. |
| 20:28.59 | Cubber | Asterisk 1.6.2.20 |
| 20:29.01 | Cubber | to be specific |
| 20:29.02 | wcselby | what happens in the cli / /var/log/asterisk/full when you try to play the musiconhold (put someone on hold, etc) |
| 20:29.21 | Cubber | wcselby cli reports that it is playing the default moh |
| 20:29.28 | r0m|u | I think who ever told you si wrong. RTP and SIP are not married |
| 20:29.48 | p3nguin | I think they are wrong, too, considering I had no audio when I had the rule disabled. |
| 20:29.50 | r0m|u | nether divorced.... just two different transport one which uses the other |
| 20:29.55 | Cubber | <PROTECTED> |
| 20:29.59 | p3nguin | I enabled it again, and audio is restored. |
| 20:30.08 | Cubber | that is /var/log/asterisk/full |
| 20:30.26 | Cubber | res_musiconhold.c: Unable to open file '/var/lib/asterisk/moh//orig_Back In Black': No such file or directory |
| 20:30.35 | Cubber | spaces? |
| 20:30.42 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-wudfjsspxziykpil) |
| 20:30.51 | r0m|u | p3nguin, I think you found your answer :) |
| 20:31.05 | p3nguin | Sometimes that happens. |
| 20:31.24 | r0m|u | lol I think it happens more than just "sometimes" |
| 20:31.46 | r0m|u | I tend to find things that way too |
| 20:31.49 | p3nguin | You're right. People are wrong much more often than just "sometimes." |
| 20:31.50 | r0m|u | just by trying it. |
| 20:32.41 | wcselby | Cubber that shoudln't be an issue |
| 20:32.42 | Cubber | wcselby: i just renamed the file with no spaces and get the same error in logs |
| 20:32.46 | r0m|u | d00d is cold here in Texas..... ugh cold... ahte it |
| 20:32.46 | Cubber | <PROTECTED> |
| 20:32.51 | *** join/#asterisk afink (~afink@wsip-184-187-15-226.om.om.cox.net) |
| 20:32.52 | wcselby | cold? |
| 20:32.54 | r0m|u | hate* |
| 20:32.57 | wcselby | what part of texas are you in? |
| 20:33.19 | wcselby | aren't you in spring? |
| 20:33.32 | wcselby | it's like 67-70 degrees in houston right now, that's perfect weather |
| 20:33.33 | r0m|u | Yes |
| 20:33.42 | r0m|u | I am freezing. LOL |
| 20:33.56 | wcselby | okay so my iphone says 63 degrees |
| 20:34.13 | r0m|u | lol |
| 20:34.13 | wcselby | lol, not freezing, but it's still nice |
| 20:34.13 | r0m|u | I am from PR. 70 is cold! |
| 20:34.17 | p3nguin | 45 F over here today. |
| 20:34.17 | r0m|u | lol :P |
| 20:34.21 | wcselby | and it's better than 100+ |
| 20:34.25 | r0m|u | true |
| 20:34.30 | wcselby | sorry Cubber , r0m|u distracted me |
| 20:34.34 | *** join/#asterisk shido6_ (~shido6@209.131.62.113) |
| 20:34.40 | wcselby | ;) |
| 20:34.46 | Cubber | wcselby no problem so for some reason the file is not being found? |
| 20:34.49 | r0m|u | damn p3nguin! |
| 20:35.11 | p3nguin | It was in the 60s for the past several days. |
| 20:35.23 | r0m|u | wcselby, you in Houston? |
| 20:35.24 | wcselby | it snowed at astricon |
| 20:35.32 | wcselby | r0m|u yeah, live in friendswood, work out by katy |
| 20:35.51 | r0m|u | ah! My old Friend is from Friendswood. |
| 20:36.01 | r0m|u | My old Supervisor too :) |
| 20:36.01 | Cubber | wcselby this looks like my issue: https://issues.asterisk.org/view.php?id=12115 |
| 20:36.08 | r0m|u | nice are |
| 20:36.23 | r0m|u | nice area* |
| 20:36.39 | r0m|u | p3nguin, man thats just to cold for me. |
| 20:36.48 | wcselby | you said you're on 1.6.2.20? |
| 20:36.51 | p3nguin | I'm not real fond of it right now, either. |
| 20:37.09 | p3nguin | 60s was nice, 45 is rather chilly. |
| 20:37.18 | r0m|u | I can concur. |
| 20:37.29 | *** part/#asterisk fireman_biff (~biff@65.48.133.103) |
| 20:37.30 | wcselby | Cubber that issue was resolved back in 1.6.0 |
| 20:37.52 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-sblijqgitsgmkwba) |
| 20:39.09 | Cubber | wcselby the error is saying the filname without the .mp3 extension in the error: '/var/lib/asterisk/moh//orig_bbb |
| 20:39.18 | Cubber | wcselby but the file is orig_bbb.mp3 |
| 20:39.22 | Cubber | may be the issue |
| 20:39.28 | wcselby | Cubber yeah I realize that, that's not the issue I don't think |
| 20:39.30 | wcselby | can you get to the cli? |
| 20:39.36 | Cubber | been there |
| 20:39.42 | wcselby | type moh show files |
| 20:40.17 | Cubber | Class: default File: /var/lib/asterisk/moh//orig_bbb |
| 20:40.28 | wcselby | it's the extra slash I think |
| 20:40.42 | Cubber | hmm so how to get rid of it |
| 20:40.43 | wcselby | in freepbx, is there somewhere to define the path for moh? |
| 20:40.50 | wcselby | or for that class? |
| 20:41.00 | Cubber | nope |
| 20:41.27 | Cubber | wcselby http://www.freepbx.org/forum/freepbx/users/moh-problem-with-mp3 |
| 20:41.28 | r0m|u | I know who would know Cubber |
| 20:41.49 | p3nguin | someone in a channel that isn't #asterisk? |
| 20:42.31 | wcselby | lol Cubber ask around in the #asterisk-now and #freepbx channels, I'll continue to try and help where I can |
| 20:43.59 | wcselby | i guess also the question should be asked - do you have a directory and file named /var/lib/asterisk/moh/orig_bbb if you check on the command line? |
| 20:44.00 | Cubber | wcselby freepbx channel is telling me to re encode the file to 8000hz |
| 20:44.14 | p3nguin | I wasn't paying attention... but are you using mode mp3 or mode files? |
| 20:44.20 | wcselby | ls -lh /var/lib/asterisk/moh/ |
| 20:46.01 | wcselby | hmmm |
| 20:46.05 | r0m|u | wcselby, I dont have to same => n,Hangup() because with the GoTo I am telling it to jum to the same dial plan as 1NXXN right? |
| 20:46.06 | wcselby | okay, disappear then |
| 20:46.16 | wcselby | correct |
| 20:46.19 | wcselby | just like I showed it |
| 20:46.40 | wcselby | assuming we're talking about the same thing |
| 20:46.42 | wcselby | lol |
| 20:46.50 | p3nguin | Nothing runs after a Goto() unless the Goto() fails and the call doesn't actually go anyhwere else. |
| 20:46.54 | r0m|u | its all clear now. like boobs slaping you in the face! |
| 20:47.18 | p3nguin | So what you're saying is that sometimes you have to motorboat your dialplan? |
| 20:47.21 | r0m|u | clearness has struck me |
| 20:48.02 | wcselby | p3nguin LOL! |
| 20:48.43 | r0m|u | I am lost. but ill laugh! LOL |
| 20:48.45 | *** join/#asterisk Azrael808 (~peter@cpc17-walt12-2-0-cust657.13-2.cable.virginmedia.com) |
| 20:49.02 | p3nguin | ~motorboat |
| 20:49.12 | p3nguin | :/ |
| 20:49.43 | r0m|u | lol |
| 20:49.59 | r0m|u | the bot is drunk. probably at the strip joint... is not even friday |
| 20:50.00 | wcselby | lol, I think that should just be a link to r0m|u description of "<r0m|u> its all clear now. like boobs slaping you in the face!" |
| 20:50.06 | *** join/#asterisk grandpapadot (~grandpapa@99.175.248.81) |
| 20:50.25 | r0m|u | rofl!!!!! |
| 20:50.41 | p3nguin | infobot: motorboat is <reply> <r0m|u> its all clear now. like boobs slaping you in the face! |
| 20:50.41 | infobot | p3nguin: okay |
| 20:50.45 | p3nguin | done. |
| 20:50.55 | grandpapadot | Hey guys, trying to evaluate with precedence the following without luck, any help would be mucho appreciated as I'm in my second hour and brain fried -> GotoIf($[ $[$["${A}"="yes"] | $["${A}"="1"]] & $["${V}">"800"] ]?s,connect:s,abort) |
| 20:51.14 | r0m|u | ~motorboat |
| 20:51.15 | infobot | <r0m|u> its all clear now. like boobs slaping you in the face! |
| 20:51.18 | grandpapadot | Tried $[(...)..] assuming () would work like math, lol |
| 20:51.19 | wcselby | nice' |
| 20:51.23 | r0m|u | lmao |
| 20:52.01 | r0m|u | my nick is embedded in to a bot in irc. wtf! LOL |
| 20:52.02 | wcselby | grandpapadot I don't think you can use the & in precedence evaluation in asterisk |
| 20:52.12 | wcselby | but I could be incorrect on that, which versino of asterisk are you using? |
| 20:52.15 | grandpapadot | 1.8 |
| 20:52.36 | p3nguin | For future reference, that's a branch not a version. |
| 20:52.41 | grandpapadot | Basically I just want GotoIf((a=yes|a=1) & b>800) |
| 20:53.13 | *** join/#asterisk joshaidan (~brianj@S0106001c1023e838.tb.shawcable.net) |
| 20:53.38 | wcselby | grandpapadot - check this link https://wiki.asterisk.org/wiki/display/AST/Operators |
| 20:54.06 | wcselby | also, be careful with quotes |
| 20:54.09 | wcselby | just saying |
| 20:54.14 | grandpapadot | Yea, I read that, what gets me is the statement: Parentheses are used for grouping in the usual manner |
| 20:54.22 | grandpapadot | .. but it's not working .. |
| 20:54.33 | grandpapadot | I have the quotes in there in case the var comes back with nothing |
| 20:54.53 | autofsckk | hello |
| 20:55.12 | wcselby | grandpapadot what does the CLI show? |
| 20:55.23 | wcselby | it should show you the evaluation as a 1 or 0 |
| 20:55.27 | grandpapadot | evals to "0" |
| 20:55.42 | grandpapadot | Right... but it's the OR that's jacking it up, lol ... |
| 20:56.00 | grandpapadot | I need the A=yes | A=1 to be evaulated before and against B>800 |
| 20:56.18 | grandpapadot | (a=yes | a=1) & b>800 |
| 20:56.30 | autofsckk | when dialing through spa3102 it dial right but hangs up at the first ring |
| 20:56.54 | wcselby | grandpapadot I take it you've tried to manually set the evaluations ? |
| 20:56.58 | grandpapadot | so if a is "yes" or "1" that should eval to "1" and then if b > 800 that should eval to "1" so the result should be true or "1" |
| 20:57.08 | p3nguin | autofsckk: A phone on the ATA or a call coming in from the PSTN? |
| 20:57.25 | grandpapadot | I could break it up but I was trying to be "clean" and learn something new ... |
| 20:57.30 | grandpapadot | oh well, lol |
| 20:58.07 | autofsckk | p3nguin: a call from the ATA, havent test yet a call coming |
| 20:58.33 | wcselby | grandpapadot I meant, have you manually set the values of a and b prior to the execution of the gotoif in the dialplan, for testing purposes? |
| 20:59.09 | grandpapadot | wcselby: oh, yea, I'm showing them the line before with NoOp( ** ${thevar} ** ) and they are setting right |
| 20:59.21 | p3nguin | autofsckk: I assume you mean a phone attached to Line 1. Can you show me the dial plan on Line1 of the ATA? |
| 20:59.30 | grandpapadot | .. so my eval is getting the stuff |
| 20:59.51 | autofsckk | p3nguin: it only rings once and it imediatly hangs up |
| 21:00.05 | grandpapadot | Basically, I'm just trying to find a clean answer on how to do precedence with asterisk and a pattern like: (a=yes | a=1) & b>800 |
| 21:00.07 | autofsckk | p3nguin: of course, gimme a minute |
| 21:01.11 | pigpen | Hi all, having an annoying issue here with dtmf. Polycom 650, Asterisk 1.8.7.1, Audiocodes FXO, PSTN lines: calling into, lets say a bank. DTMF works fine for the "press 1 for?" part, but when you go to punch in the account number, it screws up. |
| 21:01.37 | grandpapadot | pigpen: if you type the dtmf slower does it work? |
| 21:01.38 | pigpen | asterisk sip.conf is set with dtmfmode=inband |
| 21:01.47 | wcselby | grandpapadot check with leifmadsen or Qwell , they may have some ideas about nifty ways to do what you're talking about |
| 21:01.51 | pigpen | yeah, but not too slow. but not too fast. |
| 21:01.53 | p3nguin | inband sucks. |
| 21:02.01 | pigpen | ;-) you know a woman is involved on the description. |
| 21:02.30 | pigpen | p3nguin, everything sucks at the moment. I swear. |
| 21:02.53 | r0m|u | I agree with p3nguin inband is problematic |
| 21:03.06 | pigpen | example: Day 1: "Yeah, everything is great!!!" Day 2: Everything is HELL and has been for weeks!!!" |
| 21:03.10 | r0m|u | I had to go away from it on my setup |
| 21:03.24 | p3nguin | rfc2833 is preferred most of the time. |
| 21:03.49 | pigpen | yeah, i had the dtmfmode=rfc2833 set for every other deployment I have done, but it wasn't working with it either. |
| 21:03.57 | pigpen | any chance the audiocodes is mucking it up? |
| 21:04.26 | p3nguin | Yes. |
| 21:04.38 | p3nguin | There are DTMF settings for your card. |
| 21:04.38 | pigpen | yeah, now just to figure out where. |
| 21:05.02 | p3nguin | I'd imagine in the dahdi config file. |
| 21:05.10 | pigpen | Audiocodes is a sip device. |
| 21:05.18 | pigpen | I wish it was DAHDI. |
| 21:05.29 | p3nguin | Oh, it's a SIP/FXO appliance? |
| 21:05.33 | pigpen | yeah |
| 21:05.39 | pigpen | complication x 20 |
| 21:05.58 | r0m|u | As much as I like them sometimes Audiocodecs can be chaotic |
| 21:06.16 | p3nguin | Surely there is some DTMF tuning options in that device. |
| 21:06.25 | pigpen | yeah, they are much easier to deal with on a ini file level. but man, you get into that dam web interface?. |
| 21:07.06 | pigpen | I am seeing someone referring to "Declare RFC 2833 in DSP = Yes" 1st TX DTMF Option = RFC 2833 and RFC 2833 Payload Type - 101 |
| 21:07.16 | pigpen | but I think the polycom has a payload of 127 |
| 21:07.20 | pigpen | I wonder what the diff is. |
| 21:07.43 | p3nguin | 26 |
| 21:07.51 | p3nguin | ;0 |
| 21:07.52 | r0m|u | lol |
| 21:07.52 | pigpen | I knew it! |
| 21:08.15 | pigpen | I needed that. |
| 21:08.24 | p3nguin | I couldn't decide if I was going to do it or not. |
| 21:08.47 | wcselby | lol |
| 21:08.49 | pigpen | You ultimately had no choice. |
| 21:09.49 | r0m|u | pigpen, tone it down to 120 |
| 21:10.00 | r0m|u | at least that what Ihave mine setup at |
| 21:10.04 | pigpen | sniff?was already set to 96 |
| 21:10.07 | r0m|u | I use a polycom 501 though |
| 21:10.20 | r0m|u | over a different appliance. |
| 21:11.04 | r0m|u | lol p3nguin |
| 21:11.15 | grandpapadot | lol, man I just simplified that down to half of what I had with ISNULL, awesome |
| 21:13.45 | sorressean | p3nguin told me to use dnat, I had to afk so I didn't get to ask about this. I was connecting to a friends confirence and I didn't have to mess with dnat, can I make asterisk do that? when I connect the IP is like 192.168.blah, but I'm connecting across the internet to a linode. |
| 21:14.14 | p3nguin | I never mentioned dnat. |
| 21:14.36 | lystra | Newbie question. If I move off our local phone company and set up an Asterisk server with some Digium analog cards, what type of company do I look for to transfer our phone numbers to and route calls to our Asterisk server? |
| 21:14.42 | p3nguin | What I said was: Configure your asterisk to correctly work with your NAT. I saw you were making a call from a phone behind a NAT, so NAT in clearly involved. |
| 21:14.45 | sorressean | You did I think when you seen that the connecting ip was a 192.x |
| 21:15.08 | p3nguin | I did not say "dnat." I said configure asterisk to work with your NAT. |
| 21:15.33 | wcselby | lystra you can keep your analog phone company if you get an analog digium card |
| 21:15.47 | p3nguin | dnat is a term in routing -- I'm focusing on your asterisk configuration. |
| 21:15.47 | r0m|u | FXO* |
| 21:15.55 | wcselby | if you want to transfer your number to another provider, you should look at SIP providers, which would save you the expense of the digium analog card |
| 21:16.32 | lystra | wcselby: Ok. But then how do I use the existing analog phones? |
| 21:16.42 | p3nguin | lystra: If you're going pure VoIP and no more telco lines, you'll want an ITSP. |
| 21:16.45 | p3nguin | ~itsp |
| 21:16.45 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 21:17.19 | lystra | p3nguin: Ok, thanks. |
| 21:17.30 | lystra | ~itsplist-us |
| 21:17.30 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 21:17.35 | p3nguin | lystra: To use existing analog phones, you'll either use ATAs or an FXS card. |
| 21:17.37 | p3nguin | ~ata |
| 21:17.38 | infobot | extra, extra, read all about it, ata is Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 21:17.40 | p3nguin | ~fxs |
| 21:17.41 | infobot | [fxs] foreign exchange station - type of port you need to connect an analog device (phone, fax machine) to a pbx. This is the type of port found in your wall jack. |
| 21:19.48 | p3nguin | sorressean: On your asterisk system, I'm interested in "ifconfig -a" and "iptables -L -nv" |
| 21:21.47 | pigpen | Ok, trying a few things |
| 21:21.58 | pigpen | everywhere dtmf was listed, I set to rfc2833 |
| 21:22.16 | pigpen | revered asterisk back to 2833 |
| 21:29.27 | pigpen | Well, I think they have line problems. |
| 21:29.51 | pigpen | they are getting a mess of cross-talk on the lines too. |
| 21:34.40 | r0m|u | pigpen, I think your pbx got hacked and is now serving "adult chat lines" |
| 21:35.18 | pigpen | sweet, maybe now I can make some money |
| 21:35.26 | r0m|u | LOL |
| 21:35.35 | *** join/#asterisk willzzz (~Will@gateway.meteor-web.com) |
| 21:35.41 | pigpen | I should have enough lines. only lave like 92 |
| 21:36.34 | r0m|u | nice. |
| 21:36.52 | r0m|u | p3nguin, can help you get thise lines going pretty quick. :P |
| 21:36.56 | r0m|u | those* |
| 21:38.00 | pigpen | I am now trying to setup a number off of my pbx (the one severed with the 4 pri's) to test dtmf function |
| 21:38.03 | pigpen | ie: a test number. |
| 21:38.16 | pigpen | ie: punch in a number, and it plays it back to you. |
| 21:38.30 | wcselby | read(variable) |
| 21:38.36 | wcselby | saydigits(variable) |
| 21:38.50 | pigpen | thanks, I knew I have seen those before. |
| 21:41.18 | wcselby | np |
| 21:42.55 | ruied | I have asterisk 1.8.7.1, I want to receive a fax and convert it to tiff or pdf with ReceiveFAX() and getting an error that asterisk could not locate a FA tecnologie. does spandsp comes with asterisk or do I have to install it? |
| 21:44.07 | willzzz | i have a unique problem |
| 21:44.24 | willzzz | i have a associate with a gsm mobile phone + active SIM + carrier and when they call us |
| 21:44.33 | willzzz | we get DTMF digits from their GSM Mobile carrier |
| 21:44.39 | willzzz | repeated exactly twice |
| 21:44.48 | r0m|u | exten => 69,1,Answer same => n,BackGround(extra/booty-talk) same => n,NoOp(${~motorboat}) |
| 21:45.15 | p3nguin | You have an extraneous Answer() there. |
| 21:45.38 | p3nguin | i.e. BackGround() performs an answer by itself. |
| 21:45.38 | r0m|u | LOL nothing escapes tha masta! |
| 21:46.20 | wcselby | ruied you have to manually install spandsp, then compile the appropriate modules into asterisk |
| 21:46.31 | wcselby | appropriate modules being app_fax and res_fax, I think |
| 21:49.41 | pigpen | packet loss is driving me nuts today |
| 21:50.01 | eppigy | i would address your network issues |
| 21:50.36 | *** join/#asterisk WiretapSeven (~Wiretap@unaffiliated/wiretap) |
| 21:51.26 | pigpen | I wish I could. because, I could fix it. but my dam sip has different ideas. |
| 21:51.46 | eppigy | PUNCH THEM IN THE FACE |
| 21:51.52 | Katty | or not. |
| 21:51.57 | eppigy | :[ |
| 21:51.59 | Katty | hug their face, instead. |
| 21:52.01 | r0m|u | iiiiiinnnnnnn tttthhhhhaaaa ffffaaaaccccceeee!!!! |
| 21:52.02 | Katty | what are we talking about? |
| 21:52.16 | eppigy | ur face is nice i would like to hug it |
| 21:52.16 | ruied | wcselby, going to try |
| 21:52.22 | Katty | aww ty eppigy |
| 21:52.43 | pigpen | yeah, it is nuts. I got 2 GB fiber across two providers 20 miles away?and I am too dam lazy to go there. |
| 21:52.44 | pigpen | ;-) |
| 21:56.05 | *** join/#asterisk navaismo (~navaismo@189.146.48.254) |
| 21:56.10 | r0m|u | yay! its going home time. :) |
| 21:56.27 | r0m|u | off from work.... cya! |
| 21:56.34 | eppigy | :] |
| 21:57.27 | *** join/#asterisk WiretapWork (~wiretap@unaffiliated/wiretap) |
| 21:57.32 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 22:03.32 | *** part/#asterisk wesphillips (~wphill04@137.237.195.4) |
| 22:03.36 | wcselby | later r0m|u |
| 22:03.56 | willzzz | so has anyone else had a legacy dtmf=auto |
| 22:03.58 | willzzz | removed that |
| 22:04.04 | willzzz | now there's only dtmfmode=auto |
| 22:04.09 | willzzz | and in the trunks dtmfmode=auto's |
| 22:04.25 | willzzz | and there's ulaw and alaw in both since the dtmf repeat is coming from overseas from alaw to ulaw |
| 22:12.43 | p3nguin | I don't understand the question. |
| 22:13.34 | WIMPy | Did you see a question mark, yet? |
| 22:14.52 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 22:20.40 | willzzz | i have a user |
| 22:20.44 | willzzz | on a gsm mobile phone |
| 22:20.48 | willzzz | that calls our SIP trunks |
| 22:20.57 | willzzz | in which in our SIP trunk upstream we have dtmfmode=auto |
| 22:21.35 | willzzz | we get DTMF data (entered digits into our internal system) |
| 22:21.41 | willzzz | the digits our asterisk is receieving is CORRECT |
| 22:21.44 | willzzz | but the digits are repeated twice |
| 22:21.52 | willzzz | I want 123, my system is receieving 112233 |
| 22:22.00 | willzzz | and domestically it works fine |
| 22:22.11 | willzzz | someone is abroad and they call in and it's 112233 |
| 22:28.07 | willzzz | http://pastebin.com/NcZ0zbLQ |
| 22:29.08 | WIMPy | why do you use auto? |
| 22:29.25 | WIMPy | Does your ITSp detect DTMF for you or not? |
| 22:35.38 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 22:41.26 | willzzz | ITSp? I don't know. |
| 22:41.27 | *** join/#asterisk nny (~Scott@174.107.223.14) |
| 22:41.31 | willzzz | I use auto because rfc1833 works fine. |
| 22:42.33 | WIMPy | If RFC2833, configure that. |
| 22:42.38 | nny | diagnosing and issue with freepbx/asterisk. Entry in additional.conf is exten => s,n(record),MixMonitor(${EVAL(${MIXMON_DIR})}${CALLFILENAME}.${MIXMON_FORMAT},,${MIXMON_POST}), results are http://pastebin.com/0FDq6m9u . This is *pretty* much gonna come down to a freepbx issue, but trying to figure out how EVAL is being used improperly. Any eyes who see something lemme know |
| 22:42.40 | WIMPy | If RFC2833 works, configure that. |
| 22:42.47 | willzzz | what does that mean |
| 22:42.49 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
| 22:42.50 | willzzz | i use just rf2833 |
| 22:43.13 | nny | sorry exten => s,n(record),MixMonitor(${EVAL(${MIXMON_DIR})}${CALLFILENAME}.${MIXMON_FORMAT},,${MIXMON_POST}) is the entry, the comma is from sentence formatting |
| 23:05.27 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-sblijqgitsgmkwba) |
| 23:07.04 | *** join/#asterisk kaushal (~kaushal@14.97.130.22) |
| 23:07.07 | kaushal | Hi |
| 23:09.22 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 23:11.23 | nny | btw ignore me, freepbx missing some variables, fixing now. Mmmm |
| 23:22.49 | *** join/#asterisk talntid (~t@c-67-168-114-26.hsd1.wa.comcast.net) |
| 23:23.06 | talntid | Who do you guys recommend for SIP voip providers? Flowroute, Vitelity... ? |
| 23:23.29 | p3nguin | VoIP.ms or Flowroute |
| 23:25.16 | talntid | hmm, voip.ms. Hadn't heard of them. Thanks. I'll look into them :) |
| 23:25.26 | SeRi | p3nguin, after I made the change to my dial plan I am now gettin: Timeout, but no rule 't' or 'e' in context 'phones' |
| 23:25.36 | SeRi | Thats the error |
| 23:25.53 | [TK]D-Fender | SeRi: Have you considered making one of those extrensions? |
| 23:26.30 | SeRi | [TK]D-Fender, why is it doing that though? Why is it looking for t or e? |
| 23:26.45 | p3nguin | t is the timeout extension, e is the everything else extension. |
| 23:27.11 | p3nguin | So if you want me to tell you why you are getting a timeout, I need to see the entire dialplan. |
| 23:27.32 | sorressean | I'm trying to set up my confirence onhold music to stream. some places I've seen say to use: /usr/bin/mpg123 -q -r 8000 -f 8192 -s --mono http://69.4.232.112:8168/ that just distorts the stream all to hell though. Is htere a better way to do that? preferably so I can get the mp3? |
| 23:27.53 | SeRi | p3nguin, http://pastebin.com/a9Nkx9Ee |
| 23:28.14 | *** join/#asterisk happylife (~happylife@212.92.145.7) |
| 23:28.14 | [TK]D-Fender | SeRi: becuase you timed out <- |
| 23:28.37 | p3nguin | There's no phones context in that pastebin. |
| 23:28.53 | p3nguin | When I say "entire," I don't mean "partial." |
| 23:29.27 | SeRi | ok one sec |
| 23:29.39 | p3nguin | sorressean: I use mode=custom and application=/usr/bin/mpg123 -q -b 4096 --preload 0.2 -r 8000 -f 4096 -m -s http://some-stream |
| 23:30.08 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 23:30.43 | sorressean | p3nguin: I'll give it a shot. thanks. |
| 23:32.34 | SeRi | p3nguin, http://pastebin.com/nSXs6Eid |
| 23:33.33 | talntid | p3nguin, I just tried live chat twice with voip.ms, and waited 5 minutes for a response each time.. nothing. do'h :( |
| 23:33.37 | p3nguin | How's line 58 working out for you? |
| 23:33.46 | p3nguin | talntid: What are you trying to find out? |
| 23:34.21 | p3nguin | seri: Now, what extension are you calling that ends up giving you the warning? |
| 23:34.27 | SeRi | p3nguin, not at all. I was just testing :) I know I need a ,fax,1 there :) |
| 23:34.50 | SeRi | penguni I am trying to dial out 10 digit number _NXXN |
| 23:34.51 | talntid | how easy they are to get ahold of, in the case of needing support, for one. secondly, I would be terminating 190k minutes each month through them... bulk discount? multiple servers in different locations, that I can backup to? |
| 23:35.26 | sorressean | p3nguin: weird. it still juat makes really weird noises. like it can't stream right. |
| 23:36.37 | p3nguin | seri: line 143 will match a 10-digit NANP number, but when it gets there, it will fail. |
| 23:37.05 | p3nguin | seri: You're trying to go to a priority of 1+whatever 10 digit number you've entered. Never gonna work. |
| 23:37.46 | SeRi | looking. |
| 23:38.06 | p3nguin | You can Goto(priority) or Goto(extension,priority) or Goto(context,extension,priority). Goto(extension) will not work. |
| 23:39.03 | p3nguin | This is why I give people EXACT, WORKING examples. |
| 23:39.18 | SeRi | p3nguin, I wanted to go back to exten => _1NXXNXXXXXX,1,Set(TRUNKCHECK=0) |
| 23:39.37 | p3nguin | Then you should have used Goto(extension,priority) |
| 23:39.47 | SeRi | so I have to do exten => _NXXNXXXXXX,1,Goto(1${EXTEN},1)? |
| 23:39.58 | p3nguin | mumbles something about people not following his fucking examples. |
| 23:40.08 | p3nguin | Correct. |
| 23:40.35 | SeRi | p3nguin, sorry I just dont remember I think I got some fucked up amnesia or something.... :( |
| 23:40.55 | SeRi | staying up too late and getting little to no slee is fucking me up I guess |
| 23:42.00 | p3nguin | Take a day off and sleep. |
| 23:42.19 | p3nguin | It may not help your memory any, but it will make you feel better for a while. |
| 23:42.35 | SeRi | by the way your stuff went out today. I hope the sim works out for you and the ata's as well.. Is not much but is a way of saying thank you for your help. |
| 23:43.37 | p3nguin | I'll check for it in a couple days. |
| 23:43.59 | SeRi | p3nguin, you are right. I am off to bed. again thanks for the help... Ill probable wake up in the middle of the night... ok let me know when it gets there. |
| 23:44.07 | SeRi | ooo I have tracking. its in the car. |
| 23:44.24 | SeRi | ill be back later. time for some rest |
| 23:44.46 | sorressean | wtf. wonder if just recompiling asterisk would work. every example I find just kills the sound. it sounds horrible. |
| 23:45.16 | p3nguin | Did you remember to enable mp3 support? |
| 23:45.30 | sorressean | p3nguin: I did. I have the mp3 modules loaded. |
| 23:46.13 | sorressean | p3nguin: load => app_mp3.so and load => format_mp3.so |
| 23:46.30 | p3nguin | I've never had a problem with mp3s not playing correctly... I've had problems with them not playing at all. |
| 23:46.41 | sorressean | p3nguin: do I need to do something else to get it to detect that it's mp3? |
| 23:46.56 | p3nguin | Nothing that I know of. |
| 23:47.04 | p3nguin | You've got mpg123 installed? |
| 23:47.51 | sorressean | p3nguin: yeah. it plays sound. it's just a lot of distorted hissing. |
| 23:47.54 | p3nguin | What version is it? |
| 23:49.20 | sorressean | p3nguin: mpg123? 0.2.11 |
| 23:49.24 | autofsckk | p3nguin: sorry im back, i have upload the configs, could you check your notice please? |
| 23:49.42 | p3nguin | autofsckk: I just want to see the dialplan for line1 on the ATA. |
| 23:50.56 | p3nguin | (but I'll save this pastebin for the next step) |
| 23:52.17 | autofsckk | you'll make changes there and save it? |
| 23:53.05 | p3nguin | If necessary, yes, but I just want to see the dial plan from Line 1 of the ATA. |
| 23:53.39 | autofsckk | oh i see, sorries, let me c&p |
| 23:54.12 | autofsckk | (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) |
| 23:54.51 | p3nguin | Can you give me an example number that you would be dialing? |
| 23:55.28 | autofsckk | 044551234567 |
| 23:56.45 | p3nguin | As best I can tell, that doesn't match any of your ATA's dial plan. |
| 23:57.26 | p3nguin | But I may not completely understand what the dot is doing there. If it's the same as it is in asterisk, your number does not match. |
| 23:57.40 | p3nguin | If it means 0 or more digits, then it matches. |
| 23:58.58 | autofsckk | p3nguin: i didnt configure the spa, it was already configured |
| 23:59.03 | p3nguin | I know. |
| 23:59.27 | p3nguin | That appears to be a default from-the-factory dial plan. |
| 23:59.37 | p3nguin | for USA |
| 23:59.41 | p3nguin | s/USA/North America/ |