| 00:15.58 | *** join/#asterisk ShapeShifter499 (~ShapeShif@unaffiliated/shapeshifter499) |
| 00:21.05 | cusco | hey folks |
| 00:21.10 | cusco | getting a backtrace |
| 00:21.21 | cusco | gdb gets all the debug it seems |
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| 00:21.25 | cusco | but states: warning: Can't read pathname for load map: Input/output error. |
| 00:21.35 | cusco | is this a problem, or can be ignored ? |
| 00:21.51 | cusco | brb |
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| 00:56.55 | cusco | hmm |
| 00:56.58 | cusco | any answer ? |
| 00:58.10 | slackie | :-/ |
| 00:59.29 | slackie | looks like there is no masterisk here. |
| 01:03.18 | cusco | lol |
| 01:03.33 | cusco | I've got a core :) |
| 01:03.44 | cusco | in /var/spool/asterisk |
| 01:05.39 | *** join/#asterisk freetown2 (~freetown2@ipvpn098241.netvigator.com) |
| 01:06.18 | freetown2 | hi all, could someone give me a pointer on how to update the following to syntax that 1.8 will accept? ExecIf($["${SIPPEER(${PEERCHECK3},status):0:2}" = "OK"]|Set|VOIPCHECK3=1) |
| 01:06.56 | freetown2 | is it ExecIf($["${SIPPEER(${PEERCHECK3},status):0:2}" = "OK"]?Set(VOIPCHECK3=1))? |
| 01:07.13 | [TK]D-Fender | "core show application execif" |
| 01:07.25 | [TK]D-Fender | And | is long dea as a delimiter |
| 01:07.31 | [TK]D-Fender | dead* |
| 01:08.36 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 01:09.14 | freetown2 | [TK]D-Fender, yeah, just found out...kinda got forced to upgrade from 1.4 to 1.8 |
| 01:09.58 | WIMPy | Pipes where pre 1.4. |
| 01:10.13 | [TK]D-Fender | 1.8 itself is old |
| 01:10.20 | [TK]D-Fender | We're on * 12 now.... |
| 01:21.44 | Penguin | 1.8.29.0 (2014/07/10) |
| 01:21.45 | Penguin | I suppose that isn't really THAT old. |
| 01:22.08 | WIMPy | Maybe not by date. |
| 01:22.27 | Penguin | How else is age calculated? |
| 01:22.51 | WIMPy | Features |
| 01:23.28 | Penguin | My dog is 6 features old. |
| 01:27.41 | freetown2 | [TK]D-Fender, using whatever the distro provides...right now, that is Ubuntu Precise. Never managed to build asterisk on illumos. |
| 01:28.28 | freetown2 | dialplan updated. |
| 01:28.38 | [TK]D-Fender | "the distro"? |
| 01:28.44 | [TK]D-Fender | What distro? By whom? |
| 01:28.58 | freetown2 | [TK]D-Fender, sorry, not the asterisk distro |
| 01:29.22 | freetown2 | Linux distro...upgraded from hardy to lucid and then to precise. |
| 01:30.13 | freetown2 | WIMPy, yesterday's problem solved after resetting the accounts...the provider uses Nortel. -_- |
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| 02:47.31 | Sean-Der | Can anyone recommend a SPA with 4 FXO ports that supports ZRTP+SIP-TLS ? |
| 02:49.27 | Sean-Der | blah I meant ATA, I keep getting that mixed up |
| 02:50.14 | Sean-Der | I have used two of Cisco's and played around with some from Patton electronics |
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| 03:19.36 | mutable_malachi | Sean-Der: I do not know, but I would very much like to know. |
| 03:19.48 | mutable_malachi | Sean-Der: You need 4 FXO ports for the telco |
| 03:19.50 | mutable_malachi | ? |
| 03:30.03 | Sean-Der | mutable_malachi: hey! I need 4 FXO ports for hooking into an existing analog PBX |
| 03:30.14 | Sean-Der | I am slowly replacing on old system |
| 03:30.26 | mutable_malachi | Ahh, understood. |
| 03:30.35 | Sean-Der | I have an FXO right now that works fine, but would like to dump the VPN eventually and just use encryption |
| 03:30.41 | Sean-Der | just trying to dump as many moving parts as possible |
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| 06:36.14 | rox | hello, is there a setting regarding the time period in which asterisk is going to process .call files in /var/spool/asterisk/outgoing? |
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| 06:44.00 | bulkorok | rox: no nothing...as soon as a file is in there it is processed... |
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| 06:50.51 | rox | bulkorok: that was so in my experience, but I have never tested with a multitude of files; so generally I can expect the files to be processes pretty much right away |
| 06:51.49 | bulkorok | yeah... the system is sof fast, that on slower machines a "cp" can mak the call file read by asterisk when it's not finished with copying... so you should "mv" them |
| 06:56.56 | rox | bulkorok: thanx for that tip :-) |
| 06:57.37 | bulkorok | sure... |
| 06:57.38 | rox | bulkorok: i'm using reasonably fast machines, though, 8 processors, raid1 disks and all |
| 06:58.14 | bulkorok | yeah... should do it usually.. but taking care of it, when you know it can help in some special situations... |
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| 07:02.49 | rox | bulkorok: i would always mv them now that I know about it, you never know when such things bite you in the behind |
| 07:03.15 | bulkorok | alright :-D |
| 07:04.51 | rox | do calls go into 'h' extension upon hangup in any context, or only when in macros? |
| 07:05.20 | rox | i am asking in the context of the asterisk dialplan |
| 07:06.50 | bulkorok | rox: https://wiki.asterisk.org/wiki/display/AST/Special+Dialplan+Extensions?src=search |
| 07:07.00 | bulkorok | When a call is hung up, Asterisk executes the h extension in the current context. |
| 07:07.18 | rox | ok, thank you, that will help a lot |
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| 10:15.15 | goddva | is there a way to control which one that should receive the first rtp packet? |
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| 10:40.00 | XATRIX | Hi, i have asterisk 11.11.0 |
| 10:40.21 | XATRIX | How can i >module load chan_dongle.so & dongle show devices ? |
| 10:40.33 | XATRIX | In this version there's no such commands |
| 10:42.45 | XATRIX | [2014-08-01 13:42:28] WARNING[14511]: loader.c:439 load_dynamic_module: Error loading module 'chan_dongle.so': /usr/lib64/asterisk/modules/chan_dongle.so: wrong ELF class: ELFCLASS32 |
| 10:42.45 | XATRIX | [2014-08-01 13:42:28] WARNING[14511]: loader.c:918 load_resource: Module 'chan_dongle.so' could not be loaded. |
| 10:42.52 | ghost75 | this appeared during sendfax, what does it mean: -- Auto fallthrough, channel 'SIP/arcor_out998780-000000e5' status is 'UNKNOWN' |
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| 10:55.51 | goddva | XATRIX, sounds like your so file was not compiled on last compile? |
| 10:56.17 | XATRIX | Yes, look like it was compiled for the different version |
| 10:56.29 | XATRIX | And currently i can't compile it for my own verison |
| 10:56.37 | XATRIX | http://fpaste.org/122498/68901571/ |
| 10:56.40 | goddva | what is that chan_dongle? |
| 10:56.42 | XATRIX | I don't know why :( |
| 10:57.07 | XATRIX | It's for USB 3G modem , actually i want to use it as a voice gateway |
| 10:58.10 | goddva | so where did you get the code from? |
| 10:58.53 | XATRIX | https://code.google.com/p/asterisk-chan-dongle/ |
| 10:59.21 | goddva | Have you checked https://github.com/jstasiak/asterisk-chan-dongle ? |
| 10:59.37 | goddva | **project with my Asterisk 10 and 11 compatibility patches** |
| 11:00.01 | ghost75 | how reliable is fax show stats ? |
| 11:00.36 | XATRIX | goddva: nope... let me do it |
| 11:01.32 | XATRIX | goddva: actually i don't know how to compile it from SVN |
| 11:01.51 | goddva | The link I sent uses git.. |
| 11:02.02 | goddva | # git clone https://github.com/jstasiak/asterisk-chan-dongle.git |
| 11:02.48 | XATRIX | Ok, i did, but no ./configure |
| 11:03.12 | goddva | aclocal && autoconf |
| 11:03.27 | goddva | then you have it |
| 11:03.51 | XATRIX | yes, i trid it |
| 11:03.56 | XATRIX | *tried |
| 11:04.03 | XATRIX | But i only have http://fpaste.org/122501/40689101/ |
| 11:06.04 | goddva | strange, works here.. |
| 11:06.14 | goddva | please read the INSTALL |
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| 12:17.00 | tparcina | How to define the location of sound files? |
| 12:17.12 | tparcina | In asterisk.conf, there is no sound directory. :( |
| 12:17.23 | [TK]D-Fender | varlib <---- |
| 12:19.29 | tparcina | [TK]D-Fender: But there are other things as well. :( |
| 12:19.47 | WIMPy | ln -s |
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| 12:22.06 | tparcina | WIMPy: If I have dir /var/lib/asterisk/sounds/ with files and subdirs, and I have /usr/share/asterisk/sounds with files and subdirs. Can I link the first dir to another dir, so if file (dir) isn't find in first dir, the system searches for it in another? |
| 12:23.33 | tparcina | I have strange situation, where half files in one, and another half of files is in another (upgrades, change of GNU/Linux distribution, diferent Repo for asterisk and asterisk-sounds...) :( |
| 12:25.32 | WIMPy | No, they need to be in one location. |
| 12:25.52 | WIMPy | Linux doesn't support merging directories. |
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| 12:26.26 | WIMPy | . o O ( one more thing I was used to before "changing" to Linux ) |
| 12:27.11 | tparcina | WIMPy: OK; thank you. |
| 12:29.19 | XATRIX | goddva: Tried multiple ways, still no go. It doesn't want to compile |
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| 12:47.08 | roxlu | Hey guys, does someone maybe knows what RFC contains information about how to reply to a USE-CANDIDATE attribute in STUN? I can't find it in the ICE RFC5245 (or STUN, 5389) or I'm overlooking it |
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| 13:50.33 | dan_j | Hi. Has anyone noticed music on hold starting for seemingly no reason in asterisk 11? |
| 13:50.58 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:50.59 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:51.02 | dan_j | It started happening ever since i moved from 11.10.0 to 11.10.2. |
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| 14:00.53 | tparcina | dan_j: It starts during the call? |
| 14:01.19 | dan_j | Yes. |
| 14:01.30 | dan_j | One side hears silence, the other side hears the music. |
| 14:01.48 | dan_j | Happening to different clients. |
| 14:01.56 | tparcina | What do logs say? |
| 14:02.20 | dan_j | Nothing exciting. Just music started. I've not checked the sip packets yet. |
| 14:02.44 | dan_j | Also, we are seeing dropped calls, and our carrier has investigated and the BYE is coming from our asterisk server. |
| 14:02.52 | dan_j | Nothing obvious on the CLI though. |
| 14:02.54 | [TK]D-Fender | that is what would tell * to start it... |
| 14:03.05 | [TK]D-Fender | You need to be looking at that |
| 14:03.14 | dan_j | I'm going to capture some sip packets and see whats happening, but i wanted to know if anyones seen anything recently like this. |
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| 14:10.07 | captain118_ | I'm new to asterisk and I am having trouble configuring my sip connection. Can someone help me troubleshoot? |
| 14:12.03 | [TK]D-Fender | dan_j: Did you look on the tracker? |
| 14:12.17 | captain118_ | Tracker? |
| 14:12.18 | [TK]D-Fender | captain118_: pastebin what you've got and tell us what's going wrong. |
| 14:12.20 | [TK]D-Fender | ~pb |
| 14:12.20 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 14:12.22 | [TK]D-Fender | ^^^^ |
| 14:12.46 | dan_j | [TK]D-Fender: i was being hopeful! I looked at the changelog :-) |
| 14:13.31 | [TK]D-Fender | dan_j: They don't list new bugs in a changelog |
| 14:13.43 | captain118_ | I get "This number is not in service" when I attempt to dial out. |
| 14:13.47 | dan_j | I know. I was hoping it had already been fixed |
| 14:13.48 | [TK]D-Fender | That's like saying"Yeah there's an error, and we're releasing it in there anyway" |
| 14:13.54 | captain118_ | I used the config from this site. http://switch2voip.us/index.php/customer-support/byod/sip-trunking-configuration |
| 14:14.28 | [TK]D-Fender | captain118_: Their instruction are bad and very dated |
| 14:14.28 | captain118_ | I used the Digest Authentication config type. |
| 14:14.34 | captain118_ | Oh |
| 14:16.03 | [TK]D-Fender | captain118_: So what's the actual error? |
| 14:17.53 | captain118_ | I get this number is not in service when I dial out. My ip trunk is online and registered. So I think all my credentials are right. |
| 14:18.11 | *** join/#asterisk asteriskuser123 (~callmanag@host213-120-124-225.in-addr.btopenworld.com) |
| 14:18.28 | [TK]D-Fender | Show us the registration status & call attempts, etc |
| 14:18.41 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:18.41 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:19.35 | asteriskuser123 | hey, having some trouble with DAHDI and was hoping you could help. It seems that asterisk is ignoring my chan_dadhi.conf and dahdi-channels.conf files as it isn't allowing me to set the context or description of my channels. any ideas? |
| 14:19.58 | [TK]D-Fender | asteriskuser123: show us |
| 14:20.00 | [TK]D-Fender | ~pb |
| 14:20.00 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 14:20.02 | [TK]D-Fender | ^^^ |
| 14:20.21 | asteriskuser123 | If i delete the files, it still shows the two lines when i 'dahdi show channels' |
| 14:20.24 | asteriskuser123 | ok gimmie a sec |
| 14:20.45 | captain118_ | http://pastebin.com/J56Ze1bd |
| 14:21.43 | davlefou | hi, how a made dialplan debug with sql dialplan? |
| 14:22.27 | [TK]D-Fender | captain118_: Requires a login... |
| 14:23.49 | *** join/#asterisk ShapeShifter499 (~ShapeShif@unaffiliated/shapeshifter499) |
| 14:23.57 | [TK]D-Fender | captain118_: enable SIP debug and try a call |
| 14:26.09 | *** join/#asterisk dym (~patrick@unaffiliated/dym) |
| 14:26.41 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-atvgaprnbgnyqyod) |
| 14:27.35 | asteriskuser123 | dadhi-channels.conf > pastebin.com/5Bin16R1 |
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| 14:31.54 | asteriskuser123 | chan_dahdi.conf > pastebin.com/aSmqjMHE |
| 14:32.47 | asteriskuser123 | regardless of what's in these files, the two channels (2 and 4) always show up as having a default context and no descriptions or callerID set when i run 'dahdi show channels' even if i kill and restart both the dahdi service and the asterisk server |
| 14:32.51 | asteriskuser123 | any ideas? |
| 14:35.35 | asteriskuser123 | sighs |
| 14:35.43 | asteriskuser123 | I was super hoping you might be able to help |
| 14:35.51 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-egiiqjiafjlrkziq) |
| 14:36.53 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-kolybehoopozabqs) |
| 14:36.53 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 14:38.17 | asteriskuser123 | [TK]D-Fender: any ideas? |
| 14:38.29 | [TK]D-Fender | #import dahdi-channels.conf <- not valid |
| 14:38.41 | asteriskuser123 | oh, what's my mistake? |
| 14:38.47 | [TK]D-Fender | descripton=Admin Line <- misspelled... assuming it even iexists |
| 14:39.05 | [TK]D-Fender | You've invented a bunch of syntax there. |
| 14:39.10 | [TK]D-Fender | Refer to the sample config |
| 14:39.14 | captain118_ | lol |
| 14:39.57 | asteriskuser123 | is the dahdi-channels.conf file ok? |
| 14:40.27 | *** join/#asterisk gsimpson1712 (~gsimpson1@77.74.196.154) |
| 14:41.09 | [TK]D-Fender | asteriskuser123: You also have 2 "channel => 4" lines because it's in dahdi-channels.conf AND in the main after the broken "include" |
| 14:43.37 | asteriskuser123 | hmm ok |
| 14:45.49 | captain118_ | I did a "sip set debug on" what should I do to add additional verbose logging? |
| 14:46.09 | [TK]D-Fender | "core set verbose 10" |
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| 14:48.35 | captain118_ | http://pastebin.com/p7pw99ZS |
| 14:49.44 | asteriskuser123 | [TK]D-Fender: I fixed the include and commented out everything but the first block of settings so the only channel related things are in dahdi-channels.conf, restarted the dahdi service and asterisk and it still shows the context as 'default' and not 'from-pstn' :( |
| 14:50.35 | asteriskuser123 | it's reading the files because I see it parsing them in the asterisk startup output, but it's almost as if it's just ignoring the content all together. |
| 14:51.21 | asteriskuser123 | [TK]D-Fender: thoughts? |
| 14:52.08 | [TK]D-Fender | Show your 2 new files |
| 14:52.50 | [TK]D-Fender | captain118_: You don't ahve an inbound route set up to handle that call |
| 14:57.05 | asteriskuser123 | pastebin.com/AWKia0Dd pastebin.com/0LyWpR4Z |
| 14:57.14 | asteriskuser123 | [TK]D-Fender: those are the new files |
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| 15:01.19 | captain118_ | Progress I set the DID for the internal route to the actual phone number of the SIP account. This is my first install. |
| 15:01.24 | WIMPy | asteriskuser123: You might get more people to look if you post complete URLs. |
| 15:01.30 | captain118_ | http://pastebin.com/xgMUf3A2 |
| 15:01.31 | [TK]D-Fender | asteriskuser123: You should have 2& 4 as from-pstn now |
| 15:08.19 | asteriskuser123 | i should, but they're still as default |
| 15:08.38 | asteriskuser123 | like I said, it's as if its ignoring the conent of the files |
| 15:08.39 | asteriskuser123 | it's weird |
| 15:09.26 | [TK]D-Fender | do a module reload from CLI and prove what files it's hitting |
| 15:09.49 | [TK]D-Fender | And remove the extra crap for chan_dahdi.conf |
| 15:13.02 | asteriskuser123 | ok, new chan_dahdi.conf --> pastebin.com/7PFruivQ |
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| 15:16.33 | asteriskuser123 | ok, I unloaded and reloaded the module and this is the output from the console --> pastebin.com/3cPgRgwA |
| 15:17.12 | asteriskuser123 | the show channels output is the same as at the top of the paste |
| 15:17.50 | smkelly | CHOOSE FILE |
| 15:17.59 | file | CHOOSE FILE |
| 15:18.26 | WIMPy | I chose not to choose file. I chose something else. |
| 15:18.38 | file | https://fbcdn-sphotos-g-a.akamaihd.net/hphotos-ak-xfp1/t1.0-9/10298986_10100105696031426_1840395437362156511_n.jpg |
| 15:20.47 | asteriskuser123 | [TK]D-Fender: this is the users file that it reads: pastebin.com/q82Dr4r4 |
| 15:20.54 | asteriskuser123 | help :'( |
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| 15:24.15 | *** part/#asterisk gsimpson1712 (~gsimpson1@77.74.196.154) |
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| 15:27.02 | asteriskuser123 | hi [TK]D-Fender |
| 15:27.10 | asteriskuser123 | ok, new chan_dahdi.conf --> pastebin.com/7PFruivQ |
| 15:27.12 | asteriskuser123 | ok, I unloaded and reloaded the module and this is the output from the console --> pastebin.com/3cPgRgwA |
| 15:27.14 | [TK]D-Fender | [11:26][TK]D-Fenderpastebin.com/q82Dr4r4 <- users.conf can DIAF |
| 15:27.15 | [TK]D-Fender | [11:26][TK]D-FenderWhat you showed in your base configs can be in CONFLICT with this. |
| 15:27.17 | [TK]D-Fender | [11:27][TK]D-FenderStop using it |
| 15:27.40 | asteriskuser123 | ok, i'll delete the users file, it was autogenerated anyway and I dont use it |
| 15:28.28 | asteriskuser123 | holy shit, that worked |
| 15:28.33 | asteriskuser123 | fuck me |
| 15:28.35 | asteriskuser123 | thanks man |
| 15:28.58 | asteriskuser123 | thanks [TK]D-Fender |
| 15:29.23 | [TK]D-Fender | You're welcome |
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| 15:33.12 | captain118_ | I'm getting an invalid number now: http://pastebin.com/QdbNLaDQ |
| 15:35.13 | [TK]D-Fender | <PROTECTED> |
| 15:35.19 | [TK]D-Fender | Go ask them why |
| 15:35.25 | [TK]D-Fender | <PROTECTED> |
| 15:35.29 | [TK]D-Fender | Is that number proper? |
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| 15:36.00 | captain118_ | no looks like my dial rules need to be modified. |
| 15:37.01 | sip | due to my name and this channel, im constantly getting queued with a purple indicator that someone is talking to me |
| 15:37.07 | sip | however it's just typical SIP chatter :> |
| 15:37.11 | sip | long live asterisk |
| 15:41.32 | *** join/#asterisk sip (~0x2e@wan.siptrunk.us) |
| 15:42.33 | captain118_ | I got it to dial out by removing the requirement to hit 9 to dial out. |
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| 15:56.01 | xaristax | Hi can someone give a hand im trying to make a call file and everything goes fine when i create the file and move it to the outgoing folder but when i create a bash script to do this asterisk gives me permission denied |
| 15:56.11 | xaristax | even when i change the owner to asterisk and the permissions |
| 15:56.38 | [TK]D-Fender | PASTEBIN is your friend... |
| 15:57.26 | xaristax | http://pastebin.com/74ddcPMM |
| 15:58.42 | [TK]D-Fender | And the output when running that as *? |
| 15:59.16 | xaristax | i get this: WARNING[2221]: pbx_spool.c:520 queue_file: Unable to stat /var/spool/asterisk/outgoing/call61328.call: Permission denied |
| 15:59.52 | xaristax | the funny thing is that if i move back the file and move it again into outgoing all works well |
| 16:01.54 | [TK]D-Fender | chown asterisk:asterisk /var/www/html/panda/scripts/*.call |
| 16:02.07 | [TK]D-Fender | What makes you think * has the rights to change the ownership of these files? |
| 16:02.33 | [TK]D-Fender | And if it has the right.. why would it NEED to? |
| 16:02.43 | [TK]D-Fender | touch /var/www/html/panda/scropts/*.call <- I also think you have spelling issues |
| 16:03.17 | xaristax | maybe it`s a bad practice but im using * to access all files with .call at the end |
| 16:03.49 | xaristax | and its working because change the owner and update the date on the files but what drives me crazy is why asterisk send a permission denied |
| 16:04.40 | [TK]D-Fender | You're also not confirming WHICH item caused that message. |
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| 16:06.54 | xaristax | iam generating a call file with a php script and i set the name to call and some random numbers after that a cron job executes the bash script set the permissions and move this files into outgoing but doest work |
| 16:07.39 | xaristax | asterisk outputs that ugly message |
| 16:09.28 | [TK]D-Fender | I'm waiting to see what condition things are in as you run that script in the first palce... |
| 16:09.34 | [TK]D-Fender | place* |
| 16:10.32 | xaristax | sorry if is a dumb question but what you mean by condition things? |
| 16:10.58 | [TK]D-Fender | "condition things are in" <- |
| 16:11.17 | [TK]D-Fender | I don't see dumps proving what perms are actually getting set, if any of it is working at all. |
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| 16:11.31 | [TK]D-Fender | You aren't proving which part(s) are specifically failing |
| 16:13.08 | xaristax | the part that is failing from my perspective is the bash file someway is moving the files but asterisk says no.... one question will be can i see some configuration files of the call files? |
| 16:13.50 | [TK]D-Fender | huh? |
| 16:14.01 | [TK]D-Fender | call files don't have "configuration files" |
| 16:14.05 | [TK]D-Fender | What are you talking about? |
| 16:15.26 | xaristax | haha sorry for the confusing chatting im experimenting with this part of asterisk so im pretty new on this section what i still dont get is what the files get move into outgoing but the call cannot be place but when i move it from shell works fine |
| 16:17.55 | [TK]D-Fender | Then you aren't thinking about what user is issuing the commands. |
| 16:18.04 | [TK]D-Fender | Once clearly does not have the permission to do so |
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| 16:40.48 | *** join/#asterisk ZenMaster (~zenmaster@unaffiliated/auckla) |
| 16:41.33 | ZenMaster | Hi. I have a freePBX system that for the most part works great. I am having this issue where when a user makes a outbound call on his polycom 501. When the customer does not answer, he gets about half way through his msg on their voicemail and the call drops. |
| 16:41.43 | ZenMaster | I am using asterisk 11. |
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| 17:09.01 | kafal | Team, I am building an app for Asterisk 12.3 where users can navigate IVR by either pressing keys in the app which sends the DTMF via PlayDTMF action. The other way to send DTMF is by pressing key on the voip phone. In both the cases AMI sends BeginDTMF and EndDTMF event. Is there any way to know which method PlayDTMF/Phone press raised the DTMF event? |
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| 17:24.58 | kafal | Anyone has any idea? |
| 17:30.35 | newtonr | kafal, I don't! |
| 17:31.05 | [TK]D-Fender | kafal: No. |
| 17:31.07 | kafal | Is there a way I can add a custom varibale in an event |
| 17:31.09 | kafal | ? |
| 17:31.34 | [TK]D-Fender | kafal: You can create a custom EVENT from your dialplan... |
| 17:31.50 | kafal | ok so i fire my own custom DTMF event |
| 17:31.55 | kafal | with variables |
| 17:31.57 | kafal | got it |
| 17:31.59 | kafal | i will try |
| 17:32.19 | kafal | I will not use AMI in built event. I will extend it and fire a cusomt event |
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| 20:01.47 | icarusglider | good day. quick question regarding configuration of DAHDI PRI (Digium TE133f) |
| 20:02.35 | icarusglider | Need to adjust the CALLING data being sent, provider is telling me for my local numbers im sending as A1 which is national, to C1 which should be local or subscriber. Any ideas? |
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