| 00:32.25 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 00:40.37 | *** join/#asterisk elkng (~elkng@unaffiliated/elkng) |
| 00:41.05 | elkng | File: info.txt Line 1 Col 0 69 bytes 100% |
| 00:41.40 | elkng | http://www.youtube.com/watch?v=hCvu2qgcsVQ this better to have in topic |
| 00:43.48 | *** join/#asterisk minotaur01 (~minotaur@S01060018e7f9c7df.hm.shawcable.net) |
| 00:47.48 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 01:15.13 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 01:15.41 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 01:16.30 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 01:25.04 | *** join/#asterisk gown (~phonetic@gateway/tor-sasl/martinphone) |
| 01:27.10 | *** join/#asterisk jonno11 (~jonno11@cpc2-walt12-2-0-cust286.13-2.cable.virginmedia.com) |
| 01:28.25 | *** join/#asterisk gonewage (~gonewage@c-68-54-124-223.hsd1.il.comcast.net) |
| 01:28.28 | *** join/#asterisk steve222 (~steve22@S0106e0cb4e3ba0b9.ed.shawcable.net) |
| 01:29.20 | steve222 | Hello all, running asterisk + freepbx on debian. Getting the error "Unable to query table description!! Logging disabled." |
| 01:29.31 | steve222 | with regards to the cdr_mysql module |
| 01:29.47 | jonno11 | Hi - hit a massive dead end here. Running Ubuntu 10.04 on a VPS, attempting an install of Asterisk 11. I compiled/installed from a tarball, and it seemed to be successful. However when running /usr/sbin/asterisk I get "Illegal instruction". Where am I going wrong? |
| 01:30.00 | steve222 | The PBX system was running no problems up until a few days ago |
| 01:30.19 | steve222 | I checked the mysql tables today as well as user priv's for the asteriskuser and all seems good. |
| 01:30.26 | steve222 | Any suggestions? |
| 01:37.18 | steve222 | Ahh okay some new info.. |
| 01:37.43 | steve222 | mysql> DESC cdr; |
| 01:37.43 | steve222 | ERROR 1 (HY000): Can't create/write to file '/tmp/#sql_124a_0.MYI' (Errcode: 13) |
| 01:37.51 | steve222 | running chmod 1777 /tmp fixed this |
| 01:38.05 | steve222 | what caused the permissions on my /tmp directory to change? |
| 01:38.20 | steve222 | I'm curious if there's a way to prevent this in the future |
| 01:40.50 | jpsharp | jonno11: That's usually a case of the configure script or something detecting the wrong CPU type. |
| 01:40.50 | jonno11 | Nobody's here to help us… haha |
| 01:40.57 | jonno11 | I spoke too soon! |
| 01:41.09 | apb1963_ | steve222: Noob here but... have you checked to see if there is a table called "description" by some chance? Regarding the second error... what user did you invoke mysql as? |
| 01:41.25 | steve222 | I fixed it actually :) |
| 01:41.25 | jonno11 | Seeing as I'm running on a VPS, how can I check the processor type and correct the issue? |
| 01:41.38 | jonno11 | jpsharp: Seeing as I'm running on a VPS, how can I check the processor type and correct the issue? |
| 01:41.40 | steve222 | was just that the permissions on /tmp were incorrect |
| 01:41.47 | apb1963_ | so now it can query the table? |
| 01:41.49 | steve222 | just wondering why the permissions had changed... |
| 01:41.51 | steve222 | yes. |
| 01:41.57 | apb1963_ | cool |
| 01:42.40 | jpsharp | jonno11: cat /proc/cpuinfo |
| 01:43.08 | apb1963_ | while I was trying to figure out your problem, I tried getting into mysql... and I get access denied for user root... even if I try to run as the asterisk user. |
| 01:43.53 | apb1963_ | how it knows I'm root when I'm asterisk, I don't know. |
| 01:44.15 | apb1963_ | oh well :) |
| 01:44.18 | jonno11 | jpsharp: Ok thanks, how can I reconfigure asterisk? |
| 01:44.54 | jpsharp | mysql root != unix root |
| 01:45.11 | apb1963_ | hmmm.. that's interesting. |
| 01:46.03 | cusco | that is a mysql issue |
| 01:46.05 | apb1963_ | interesting to know... but all I did was "mysql <dbname>" both as root, and as asterisk. |
| 01:46.06 | cusco | try #mysqç |
| 01:46.09 | cusco | #mysql rather |
| 01:46.42 | cusco | jonno11: tried /etc/init.d/asterisk start/stop |
| 01:46.48 | cusco | and asterisk -vvvvr to attach? |
| 01:47.14 | jpsharp | jonno11: Or type "uname -m" and see what it returns. |
| 01:47.39 | jonno11 | jpsharp: uname -m returns x86_64 |
| 01:48.11 | jonno11 | and cusco: Apparently asterisk isn't in that location... |
| 01:48.58 | cusco | asterisk -vvvvvvvvvvvvvvvc |
| 01:49.08 | cusco | what does it state before the illegal instruction? |
| 01:49.15 | jonno11 | Nothing |
| 01:49.24 | jonno11 | That's the only output |
| 01:49.55 | cusco | and you compiled from source? |
| 01:50.01 | jonno11 | Yep |
| 01:50.06 | jpsharp | and can you pastebin cat /proc/cpuinfo? |
| 01:50.08 | cusco | that does not seem right |
| 01:50.18 | cusco | did you uninstall asterisk packages from ubuntu? |
| 01:50.32 | jonno11 | erm |
| 01:50.36 | jonno11 | apt-get packages? |
| 01:50.59 | cusco | yes |
| 01:51.19 | jonno11 | I'm not actually sure |
| 01:51.36 | jonno11 | would doing it now help? apt-get remove |
| 01:51.41 | jonno11 | apt-get purge etc. |
| 01:52.05 | jonno11 | http://pastebin.com/L2MCknvL |
| 01:52.07 | cusco | dpkg -l|grep asterisk|cut -d " " -f 3|xargs apt-get remove --purge -y |
| 01:58.52 | jonno11 | cusco: Ok done |
| 01:59.39 | cusco | go back to the asterisk source dir |
| 01:59.44 | cusco | and run: make install |
| 01:59.44 | cusco | again |
| 01:59.54 | jonno11 | cusco: It's still giving the same error |
| 01:59.58 | jonno11 | aha |
| 02:00.20 | cusco | did you re-install the just compiled binaries? |
| 02:00.27 | cusco | did you compile on the same machine? |
| 02:00.41 | jonno11 | Nope, re-compiling now! |
| 02:00.57 | cusco | on the same machine, right? |
| 02:01.02 | jonno11 | Yep |
| 02:01.36 | jonno11 | Re-compiled, running /usr/sbin/asterisk still returns "Illegal instruction" |
| 02:01.53 | cusco | make clean; make; make install |
| 02:02.16 | cusco | ok |
| 02:02.17 | cusco | first |
| 02:02.20 | cusco | make uninstall-all |
| 02:02.25 | cusco | then make clean |
| 02:02.26 | cusco | make |
| 02:02.28 | cusco | make install |
| 02:06.31 | jonno11 | running... |
| 02:08.34 | jonno11 | Okay done |
| 02:08.52 | cusco | k |
| 02:08.55 | jonno11 | cusco: Seems to have done something |
| 02:08.59 | cusco | asterisk -vvvvvvvvvvvvc |
| 02:09.17 | cusco | yes you had ubuntu's precompiled binaries and libs intrefering |
| 02:09.41 | jonno11 | ah |
| 02:09.41 | jonno11 | no |
| 02:09.50 | jonno11 | "/usr/sbin/asterisk -vvvvvvvvvvvvc |
| 02:09.51 | jonno11 | Illegal instruction |
| 02:09.52 | jonno11 | " |
| 02:10.00 | cusco | are you sure its /usr/sbin? |
| 02:10.05 | jonno11 | No |
| 02:10.21 | jonno11 | But 'asterisk' does the same |
| 02:10.32 | cusco | it is... |
| 02:10.37 | jonno11 | wow actually |
| 02:10.40 | jonno11 | just running asterisk |
| 02:10.41 | cusco | no output before that? |
| 02:10.46 | jonno11 | "Asterisk already running on /var/run/asterisk/asterisk.ctl. Use 'asterisk -r' to connect." |
| 02:10.49 | cusco | ah |
| 02:10.54 | cusco | stop it |
| 02:11.01 | jonno11 | just the -vvvvvvvvvvc option |
| 02:11.03 | jonno11 | threw it out |
| 02:11.04 | cusco | /etc/init.d/asterisk stop |
| 02:11.13 | cusco | or asterisk -vrx "core stop now" |
| 02:11.28 | jonno11 | Illegal instruction |
| 02:11.42 | jonno11 | and "/etc/init.d/asterisk" is not found |
| 02:13.16 | cusco | kill it |
| 02:13.25 | cusco | that is still THE OLD asteirsk running |
| 02:13.27 | cusco | from ubuntu repo |
| 02:14.25 | jonno11 | asterisk never returned anything before |
| 02:14.33 | jonno11 | how can I kill it? |
| 02:15.41 | cusco | ps ax|grep asterisk |
| 02:15.45 | cusco | check the PID number |
| 02:15.54 | cusco | kill -9 <pid-num> |
| 02:15.59 | jonno11 | returns nothing |
| 02:16.04 | cusco | you should learn some linux basics |
| 02:16.07 | cusco | nothing? |
| 02:16.10 | jonno11 | I do know linux |
| 02:16.10 | cusco | :/ |
| 02:16.22 | jonno11 | yes, nothing |
| 02:16.41 | jonno11 | ha ok my bash terminal hung |
| 02:17.02 | jonno11 | kill -9 25997 |
| 02:17.18 | jonno11 | Ok done |
| 02:18.08 | jonno11 | But that process running was the /usr/sbin/asterisk one. |
| 03:16.09 | *** join/#asterisk greenwolf (~root@198.57.44.170) |
| 03:16.33 | greenwolf | sup |
| 03:19.01 | *** join/#asterisk greenwolf (~root@198.57.44.170) |
| 03:32.15 | *** join/#asterisk kresp0 (~kresp0@81.61.24.25.dyn.user.ono.com) |
| 03:35.46 | *** join/#asterisk greenwolf (~root@198.57.44.170) |
| 03:38.04 | *** join/#asterisk kresp0 (~kresp0@81.61.24.25.dyn.user.ono.com) |
| 04:01.26 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 04:01.26 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 04:21.39 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.82.82) |
| 04:21.51 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 04:51.48 | *** join/#asterisk volga629 (~volga629@76-10-130-18.dsl.teksavvy.com) |
| 05:10.38 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
| 05:14.32 | *** join/#asterisk greenwolf (~root@198.57.44.170) |
| 05:14.43 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 05:15.10 | greenwolf | im having a problem with installing and compiling the dahdi-linux tools package installed on this machine for some reason |
| 05:15.18 | greenwolf | i keep getting the error msg You do not appear to have the sources for the 2.6.32-042stab062.2 kernel installed. |
| 05:15.21 | greenwolf | make[1]: *** [modules] Error 1 |
| 05:15.56 | greenwolf | but when i run -uname -a my linux kernal is that 2.6.32-042stable062.2 kernal |
| 05:16.00 | ChannelZ | You need the kernel headers |
| 05:16.13 | greenwolf | umm ok so how do i retrieve those? |
| 05:16.34 | ChannelZ | Your favorite package manager? |
| 05:16.40 | greenwolf | apt-get |
| 05:16.56 | ChannelZ | Well there you go then. |
| 05:17.10 | greenwolf | but i already ran apt-get upgrade and apt-get update |
| 05:17.23 | ChannelZ | You need the kernel headers package |
| 05:17.24 | greenwolf | i thought that would have retrieved those files already |
| 05:17.32 | greenwolf | ok thanks channelz for that info |
| 05:17.37 | greenwolf | shit had me stuck for hours lol |
| 05:17.58 | ChannelZ | The kernel sources/headers aren't typically installed by default |
| 05:18.18 | greenwolf | ah i see |
| 05:18.30 | greenwolf | usually for centOS they are but im running stupid ubuntu |
| 05:18.38 | greenwolf | i totally miss slackware or gentoo OS |
| 05:20.02 | greenwolf | ok thanks channelz it worked :) |
| 05:20.04 | greenwolf | brb |
| 05:22.04 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 05:32.51 | *** join/#asterisk greenwolf (~root@198.57.44.170) |
| 05:33.00 | greenwolf | hey channelz sorry man its not working again |
| 05:33.17 | greenwolf | i keep getting that same error msg so i prob didnt get the right headers?" |
| 05:33.43 | greenwolf | keeps saying everything is up to date on the system and thatthe package can't be found |
| 05:33.58 | greenwolf | y is it so hard to get the linux kernal headers installed |
| 05:34.18 | Sicelo | u have to get headers that fit the exact version of kernel you have |
| 05:35.08 | Sicelo | and no, it's not so hard. dahdi modules were the first modules i have ever built in my life, and it worked ok the first time |
| 05:35.09 | greenwolf | yes i understand but i google that and there doesnt seem to much info on getting thm properly installed and workign |
| 05:35.49 | Sicelo | apt-get install linux-headers-`uname-r` |
| 05:35.50 | greenwolf | well when i try to install and make the dahdi drivers for linux i get that error msg |
| 05:35.54 | Sicelo | something like that |
| 05:36.02 | greenwolf | yup i did that didnt work |
| 05:36.30 | Sicelo | weird. try a reboot :-/ |
| 05:37.26 | greenwolf | if i type just apt-get install linux headers i get a list of headers to install |
| 05:37.37 | greenwolf | im tyring to figure out which one this system needs |
| 05:37.59 | Sicelo | it is the one that fits your kernel |
| 05:38.16 | greenwolf | vserver |
| 05:38.23 | greenwolf | openvz |
| 05:38.28 | greenwolf | i dont kno which one to chose |
| 05:38.36 | Sicelo | uname -r |
| 05:38.39 | greenwolf | its just a regular 2.6.32 kernal |
| 05:38.43 | greenwolf | unam -r |
| 05:38.45 | greenwolf | oops lol |
| 05:38.58 | Sicelo | what does uname -r say? |
| 05:39.15 | greenwolf | 2.6.32-042stab062.2 |
| 05:39.22 | greenwolf | nothing in the list matches that tho |
| 05:40.57 | greenwolf | hold on i think i got it here |
| 05:41.01 | greenwolf | its installing now |
| 05:41.37 | greenwolf | what directory should i install those into? |
| 05:41.53 | Sicelo | default? |
| 05:42.39 | greenwolf | make[1]: Entering directory `/usr/src/dahdi-linux-complete-2.6.1+2.6.1/linux' |
| 05:42.40 | greenwolf | make -C drivers/dahdi/firmware firmware-loaders |
| 05:42.40 | greenwolf | make[2]: Entering directory `/usr/src/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/firmware' |
| 05:42.43 | greenwolf | make[2]: Leaving directory `/usr/src/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/firmware' |
| 05:42.46 | greenwolf | You do not appear to have the sources for the 2.6.32-042stab062.2 kernel installed. |
| 05:42.49 | greenwolf | make[1]: *** [modules] Error 1 |
| 05:42.51 | greenwolf | make[1]: Leaving directory `/usr/src/dahdi-linux-complete-2.6.1+2.6.1/linux' |
| 05:42.54 | greenwolf | make: *** [all] Error 2 |
| 05:42.56 | greenwolf | still not working |
| 05:43.00 | Sicelo | ~pb |
| 05:43.00 | 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. |
| 05:43.06 | greenwolf | sorry |
| 05:45.30 | Sicelo | maybe install linux-headers-2.6.32-x-all |
| 05:45.50 | Sicelo | that should likely find the right version for you |
| 05:46.18 | greenwolf | unable to locate package |
| 05:46.32 | Sicelo | lol. |
| 05:46.38 | greenwolf | shouldnt the headers be installed when i upgrade or update the system using apt-get |
| 05:46.50 | Sicelo | where i had 'x' you should put in the right value |
| 05:47.02 | greenwolf | oh shit lol |
| 05:47.26 | Sicelo | you can also try linux-headers-2.6-openvz-amd64 |
| 05:47.40 | Sicelo | beyond that, i can't help anymore |
| 05:47.56 | greenwolf | ok thank sicelo i will work with that info |
| 05:58.27 | *** join/#asterisk nopain (~kt@203.81.80.94) |
| 05:58.47 | nopain | can someone help me? |
| 05:58.53 | nopain | i got an problem with my asterisk |
| 05:59.07 | greenwolf | i can try |
| 05:59.25 | greenwolf | nopain: go ahead someone here will help if i cant |
| 06:00.11 | nopain | it's that call is stucked in asterisk console after forwarding the call to other number |
| 06:01.51 | greenwolf | are you forwarding a particular call from the console or within the dial plan? |
| 06:02.42 | nopain | greenwolf: I put a dedicated number for forwarding on the cisco SIP phone |
| 06:03.02 | nopain | greenwolf: when someone calls, it forward to a dedicated number |
| 06:03.32 | nopain | greenwolf: but after finishing conversation, the call is still remained in the Asterisk console |
| 06:03.58 | greenwolf | have u tried to setup call forwarding within asterisk dial plan in extensions.conf or followme feature rather than on the SIP device? |
| 06:04.35 | jpsharp | Sounds like asterisk isn't detecting hangups. How do the calls get into and out of asterisk? analogue lines? |
| 06:04.49 | greenwolf | oh i see |
| 06:04.55 | nopain | greenwolf: yes analogue line |
| 06:04.59 | greenwolf | i think yourddvevice is failing to send the BYE packet to asterisk to end the call |
| 06:05.09 | nopain | jpsharp: yes analogue line |
| 06:05.37 | nopain | so most of the times, i have to soft hangup the calls manually |
| 06:05.56 | nopain | i want the better way to solve this issue |
| 06:06.34 | greenwolf | you should setup wireshark and watch which device is failing to send the BYE packet then u can go from there |
| 06:06.44 | greenwolf | it could be your ata OR PHONE OR ASTERISK |
| 06:07.02 | greenwolf | gotta see where its failing no to send hangup( |
| 06:07.04 | jpsharp | theres no bye with analog lines |
| 06:07.25 | greenwolf | 0h yes thats right he did say anologue |
| 06:07.36 | jpsharp | Are they analog lines into a Digium card in the server? |
| 06:07.40 | greenwolf | sorry i dont do much anaolgue anymore just VOIP |
| 06:09.10 | nopain | jpsharp: yes.. analog lines plug in to Sangoma A-200 card in the server |
| 06:09.14 | nopain | not the Digium |
| 06:11.22 | jpsharp | Hmm. Obviously the cards are not detecting the hangup. Do you have the signalling on the lines set for fxs_ks in your zapata/chan_dahdi.conf? |
| 06:13.02 | nopain | jpsharp: yes i have it |
| 06:13.20 | nopain | it looks like signalling = fxs_ks |
| 06:14.55 | jpsharp | Then your phone lines aren't sending any kind of disconnect supervision when the calls tear down. You'll have to do it the ghetto method using busy detect. |
| 06:16.59 | nopain | jpsharp: could you please explain me a bit more how to do it? |
| 06:17.17 | jpsharp | Can you pastebin your zapata/dahdi conf? |
| 06:18.37 | jpsharp | ~pb |
| 06:18.37 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 06:20.16 | nopain | ok |
| 06:21.48 | nopain | jpsharp: http://pastebin.com/PUXUk4xH |
| 06:21.50 | nopain | pls check it |
| 06:23.52 | jpsharp | You need to set busydetect=yes |
| 06:24.31 | nopain | ok |
| 06:24.34 | nopain | then ? |
| 06:24.45 | jpsharp | And, if you're not in the US, you need to set your tonezone to the appropriate country. |
| 06:26.55 | *** join/#asterisk dijib (~dijib@208-96-84-35.eastlink.ca) |
| 06:28.28 | nopain | ic |
| 06:28.57 | jpsharp | I can't guarantee that will solve your problem, but it should help it quite a bit. |
| 06:33.46 | nopain | ok |
| 06:33.53 | nopain | thanks for your answer |
| 06:36.27 | jpsharp | analogue lines: The bane of telephony engineers everywhere. |
| 06:41.13 | *** join/#asterisk Nickinator (~Nickinato@220-244-173-9.static.tpgi.com.au) |
| 06:44.32 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 07:07.30 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.82.82) |
| 07:22.07 | *** join/#asterisk kontinuity (~kontinuit@122.178.195.61) |
| 07:25.18 | *** join/#asterisk miltux (~miltux@62.1.230.190.dsl.dyn.forthnet.gr) |
| 07:30.54 | *** join/#asterisk Romeyo (Romeyo@115.111.8.254) |
| 07:31.13 | Romeyo | hiya people! |
| 07:33.56 | Romeyo | can anyone point me to a guide where i can setup my audio code mp124 fxs with asterisk for voice mail? |
| 07:38.48 | jpsharp | I don't think you're going to find something that specific. |
| 07:39.19 | jpsharp | The audiocodes box will look like a regular SIP device to Asterisk, so go from there. |
| 07:41.05 | Romeyo | alright :) |
| 07:41.11 | Romeyo | thanks |
| 07:42.51 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 08:08.11 | *** join/#asterisk evilman_home (kvirc@2.93.212.24) |
| 08:30.20 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 08:31.16 | *** join/#asterisk ziro_axis (~h.sabrey@62.240.32.90) |
| 09:05.47 | ziro_axis | hello what i the deference between A* 1.8 & A* 11, or A* 10 |
| 09:11.32 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 09:12.14 | gavimobile | folks, I don't want to use a dialplan for my linksys ata spa2102. I would like to disable this option so that the ata device uses the dialplan of my pbx. is this possible? |
| 09:13.17 | WIMPy | Probably, but with the usual side-effects. |
| 09:13.58 | gavimobile | WIMPy: for example? |
| 09:14.48 | WIMPy | Not being able to dial anything conatining # because you need it as an enter key or you have to live with (possibly long) timeouts. |
| 09:15.46 | WIMPy | If you really want the PBX to do it, you need overlap dialing. But I don't know if any ATA supports that. |
| 09:16.19 | WIMPy | The big advantage of using interface cards. |
| 09:16.50 | gavimobile | WIMPy: side effects are no good. let me ask my question like this. my problem is that the dialing time is really short. when I pickup my phone I must enter the number in within a few seconds or the call gets droped. then I need to hang up and redial. what would be the ideal way of solving this issue? |
| 09:17.22 | WIMPy | Increase the timeout. |
| 09:17.22 | ziro_axis | after installing A* framework i need to install a unified communication (which is better FreePBX or Elastix) |
| 09:17.34 | WIMPy | Or what I said before. |
| 09:18.22 | WIMPy | ziro_axis: I have no idea of any of them fall in to that category, but |
| 09:18.27 | WIMPy | ~freepbx |
| 09:18.27 | 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 |
| 09:18.31 | WIMPy | ~elastix |
| 09:18.31 | infobot | rumour has it, elastix is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes Elastix VERY difficult to support, and is not supported in #asterisk. Try asking in #Elastix or on their forums at http://www.elastix.org |
| 09:22.17 | *** join/#asterisk fritz09 (~Adium@46.115.64.29) |
| 09:38.45 | ziro_axis | so when you install A* what kind of agent you use to configure the users & the dialing plans |
| 09:46.49 | WIMPy | vi |
| 09:59.23 | ziro_axis | WIMPy> what you mean? |
| 10:00.21 | gavimobile | WIMPy: I changed the timeout, but its only effective once I press the first key |
| 10:07.34 | *** join/#asterisk areski (~areski@80.174.255.71.dyn.user.ono.com) |
| 10:16.54 | *** join/#asterisk elico (~Thunderbi@109.64.229.90) |
| 10:27.38 | *** join/#asterisk vlad_starkov (~vlad_star@79.104.4.186) |
| 10:28.11 | Romeyo | ziro_axis WIMPy means to edit at the CLI using the vi editor [if am not wrong] |
| 10:34.50 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 10:36.09 | WIMPy | yes |
| 10:36.23 | WIMPy | gavimobile: What else would you expect? |
| 10:41.51 | gavimobile | WIMPy: well my polycom doesn't drop ever, doesn't matter how long I wait |
| 10:42.01 | gavimobile | it waits for me to send the call out |
| 10:48.07 | WIMPy | I don't know what you can configure on Polycoms. Only their Soundstations seem to make it across the atlantic. |
| 10:48.28 | *** join/#asterisk ghost75 (~trechber@dslb-178-002-147-242.pools.arcor-ip.net) |
| 10:56.09 | ziro_axis | gents i have configured my new installed *Now server and added some extensions but when i use x-light it is not connecting |
| 10:56.59 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:c11e:a73:25a:5dce) |
| 10:57.17 | ziro_axis | <PROTECTED> |
| 10:57.51 | ziro_axis | hello any body cam support? |
| 11:01.12 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 11:02.51 | WIMPy | never undestood the x-lite configuration. |
| 11:09.50 | *** join/#asterisk kresp0 (~kresp0@81.61.24.25.dyn.user.ono.com) |
| 11:13.01 | *** join/#asterisk joshuahh (~joahahua@harradence.net) |
| 11:13.23 | joshuahh | Hello, can someone help me with an issue iam having with freepbx ? |
| 11:13.33 | joshuahh | FATAL ERROR |
| 11:13.34 | joshuahh | DB Error: insufficient permissions |
| 11:13.34 | joshuahh | Trace Back |
| 11:13.34 | joshuahh | /var/www/html/admin/libraries/db_connect.php:63 die_freepbx() |
| 11:13.34 | joshuahh | <PROTECTED> |
| 11:13.46 | joshuahh | i get that error message when trying to access /admin/config.php |
| 11:13.55 | WIMPy | Don't flood the channel |
| 11:13.57 | WIMPy | and |
| 11:14.02 | WIMPy | ~freepbx |
| 11:14.02 | 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 |
| 11:14.12 | joshuahh | sorry for flooding channel |
| 11:14.23 | joshuahh | i have been in freepbx for quite a long time and have not got a response... |
| 11:16.39 | ziro_axis | WIMPy> so what kind of softphone you are using |
| 11:17.09 | WIMPy | Usually none. If I do, I use zoiper. |
| 11:28.38 | *** join/#asterisk slav3_kitten (~kitten@unaffiliated/slav3-kitten/x-0866809) |
| 11:48.57 | ghost75 | to use voicemail login somewhere else, is the only way to check voicemail.conf if password was correctly given? |
| 11:50.49 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.83.36) |
| 11:56.34 | kaldemar | ghost75: yes. use functions AST_CONFIG and CUT for parsing. |
| 11:57.32 | ghost75 | i want to login outside of asterisk |
| 12:05.17 | kaldemar | then use something else. parsing voicemail.conf is the only way unless you use realtime for the config. |
| 12:05.32 | ghost75 | this looks good http://www.voip-info.org/wiki/view/Asterisk::config |
| 12:05.58 | ghost75 | but old |
| 12:09.49 | *** join/#asterisk Dovid (~Dovid@128.sub-70-215-65.myvzw.com) |
| 12:25.30 | gavimobile | my ata device is now rining later than my other phones in my queue. how can I have my ata device ring the same time as my polycom. I don't think I made changes to my pbx, could this be an ata setting? |
| 12:26.26 | *** part/#asterisk joshuahh (~joahahua@harradence.net) |
| 12:29.24 | *** join/#asterisk vlad_starkov (~vlad_star@79.104.4.186) |
| 12:31.57 | weinerk | call-confirmation-macro from AGI - DIAL(M(confirmcall^^^)) |
| 12:31.58 | weinerk | How can I tell after Dial() returns if the call was ACCEPTED or REJECTED. |
| 12:36.00 | *** join/#asterisk vlad_sta_ (~vlad_star@83.149.8.106) |
| 12:48.33 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 13:06.26 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 13:16.01 | *** join/#asterisk Neptu (~Neptu@c-af90e255.113-1-64736c14.cust.bredbandsbolaget.se) |
| 13:20.47 | ziro_axis | what is the domain settings, is it the same IP of my server? |
| 13:24.43 | ziro_axis | hello |
| 13:41.56 | *** join/#asterisk oej (~olle@2001:16d8:cc57:1000::42:1003) |
| 14:06.39 | *** join/#asterisk elkng (~elkng@unaffiliated/elkng) |
| 14:07.00 | *** join/#asterisk blee (~blee@67.8.206.215) |
| 14:11.28 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
| 14:13.03 | *** join/#asterisk deo (~deo@112.198.79.71) |
| 15:15.55 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.122) |
| 15:19.12 | *** join/#asterisk ruben231 (~OpenDial@112.198.91.121) |
| 15:40.00 | *** join/#asterisk gonewage (~gonewage@c-68-54-124-223.hsd1.il.comcast.net) |
| 15:42.56 | *** join/#asterisk jonno11 (~jonno11@cpc2-walt12-2-0-cust286.13-2.cable.virginmedia.com) |
| 15:50.08 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 15:51.38 | *** join/#asterisk tonyclewis (uid6025@gateway/web/irccloud.com/x-suozyjxudccnzhbf) |
| 15:55.55 | *** join/#asterisk Dovid (~Dovid@ool-1826d7a8.dyn.optonline.net) |
| 15:56.06 | jonno11 | Hi - I've got a very basic, beginner setup. My sip.conf contains this: http://pastebin.com/VD2RpvTu but I'm getting a 403 Forbidden error when attempting to connect with my softphone |
| 15:57.07 | jonno11 | Sorry, massive N00b here |
| 15:58.20 | *** join/#asterisk wonderworld (~w@dsdf-4db5ef4a.pool.mediaWays.net) |
| 15:59.02 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 16:00.14 | Sicelo | first, you seem to have changed default port. did you set your softphone config to reflect that too? |
| 16:02.03 | Sicelo | jonno11: ^^ |
| 16:02.12 | jonno11 | I'm copying from a tutorial here unfortunately! Shall I remove the 'port' var? |
| 16:02.56 | Sicelo | maybe change it to 5060 insted. |
| 16:03.54 | jonno11 | Ok have done… that doesn't seem to have done it |
| 16:04.08 | Sicelo | sip reload in asterisk |
| 16:04.48 | Sicelo | every time you change the sip.conf, you have to 'sip reload' in the asterisk console |
| 16:06.28 | jonno11 | Okay, so core restart won't do that? |
| 16:07.14 | Sicelo | ah, that should as well. why restart the whole thing for a sip-only change though? |
| 16:07.23 | jonno11 | Aha! |
| 16:13.08 | *** join/#asterisk jonno11 (~jonno11@cpc2-walt12-2-0-cust286.13-2.cable.virginmedia.com) |
| 16:16.17 | Dovid | i all |
| 16:16.35 | Dovid | jonno11: We were all n00b's at some point |
| 16:16.55 | Dovid | jonno11: Glad you are learning and asking here and not off using some gui... |
| 16:19.19 | pabelanger | heh, figured we'd have realtime cdr.sql example in contrib by now |
| 16:19.51 | jonno11 | Haha thanks Dovid |
| 16:20.21 | jonno11 | Problem solved Sicelo, thanks |
| 16:21.01 | Dovid | pabelanger: hi there. how is the cold Canada? |
| 16:22.14 | Sicelo | cool jonno11. i'm noob too. honestly |
| 16:22.34 | jonno11 | Sicelo: we will get there! |
| 16:22.57 | *** part/#asterisk Romeyo (Romeyo@115.111.8.254) |
| 16:23.08 | jonno11 | Dovid: I have to say, those GUI's confuse me more than the raw config files |
| 16:23.14 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 16:23.50 | jonno11 | exten => 2000,1,Dial(SIP/2000,20) makes so much more sense than millions of menus |
| 16:24.10 | Dovid | jonno11; Most people are lazy and just use GUI's. |
| 16:24.42 | jonno11 | Dovid: all it takes is a bit of learning to use CLI |
| 16:24.54 | jonno11 | Then everything can be done faster! |
| 16:24.59 | *** join/#asterisk ziro_axis (~h.sabrey@41.254.2.189) |
| 16:25.08 | ziro_axis | hello all |
| 16:27.06 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 16:27.37 | Dovid | anyone here ever get mplayer of VLC to work with Asterisk MOH? |
| 16:29.35 | ziro_axis | my problem still exists and i do not know what is the reason |
| 16:30.04 | *** join/#asterisk jkroon (~jkroon@105.243.231.242) |
| 16:30.06 | ziro_axis | after installing the AsteriskNow and configuring the eth0 then adding some extenssions |
| 16:30.52 | ziro_axis | extension cannot register to the server?? any one can help |
| 16:31.28 | ziro_axis | pebelanger>> can you help in that |
| 16:31.29 | carrar | ~book |
| 16:31.29 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 16:32.30 | slash213 | can someone help me with queues, guys? |
| 16:32.53 | carrar | ~ask |
| 16:32.53 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 16:33.04 | slash213 | yeah, i'm typing, one moment |
| 16:33.17 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 16:34.03 | slash213 | i have wrapuptime set to 0, but if the call enters queue when one of the operators is busy (strategy is ringall), when the operator becomes available, the call in question doesn't get transferred to this operator |
| 16:34.22 | slash213 | everyone else is ringing alright, but not the guy who was initially busy |
| 16:34.42 | slash213 | asterisk is 1.8.15.1 |
| 16:34.45 | Dovid | morning TK |
| 16:36.22 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 16:36.22 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 16:40.43 | *** join/#asterisk plantseeker (~Plantseek@77.240.63.97) |
| 16:41.15 | ziro_axis | after i used the wireshark to trace the SIP messages i got 401 unauthorized? what this mean? |
| 16:41.47 | [TK]D-Fender | ziro_axis: Depends where it occurs in the conversation |
| 16:41.54 | [TK]D-Fender | ziro_axis: PASTEBIN is your friend... |
| 16:41.56 | WIMPy | slash213: It's probably just not as smart as you'd like it to be. |
| 16:42.00 | [TK]D-Fender | ~pb |
| 16:42.01 | 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. |
| 16:42.01 | [TK]D-Fender | ^^^ |
| 16:42.54 | [TK]D-Fender | slash213: If the other users were already ringing the newly free person does not start ringing immediately. The others' timeout has to have finished. |
| 16:44.36 | slash213 | [TK]D-Fender, oooh. my timeout is set to 60. will setting it to 10 solve my problem? |
| 16:44.54 | slash213 | won't people fall out of queue for the reason of "no one took the call"? |
| 16:45.56 | slash213 | i don't set any timeout for the 'queue' application in the dialplan, if that's important |
| 16:47.46 | [TK]D-Fender | slash213: AGEENT timeout |
| 16:47.58 | [TK]D-Fender | slash213: not Queue() |
| 16:48.52 | slash213 | [TK]D-Fender, yeah! it completely solved my puny problem |
| 16:48.58 | slash213 | [TK]D-Fender, thanks a lot |
| 16:49.23 | slash213 | asterisk documentation can be quite confusing sometimes |
| 16:49.28 | [TK]D-Fender | slash213: You're welcome. |
| 16:51.52 | *** join/#asterisk rockyjo3 (~rockyjoe@212.29.154.149) |
| 16:52.11 | rockyjo3 | hello All |
| 16:52.44 | rockyjo3 | this is my first time in the channel so please apologize if am posting this help request to the wrong place |
| 16:54.07 | rockyjo3 | I am struggling to make my asterisk box (set up using trixbox) work with the analog pbx in my office. |
| 16:56.19 | rockyjo3 | I have a TDM410 with a FXO and a FXS module and what I've done is plugging the FXO module to an extension of the analog pbx. but it doesn't work |
| 16:56.25 | *** join/#asterisk kleszcz (tick@linuxmafia.pl) |
| 16:57.45 | Dovid | rockyjo3: If you are using trixbox you may want to ask in #trixbox |
| 16:58.17 | [TK]D-Fender | rockyjo3: You'll need to tell us a lot more than "it doesn't work" if we're to help you |
| 16:59.06 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 16:59.16 | rockyjo3 | ok. thanks for the suggestions. I will try to give some more details here first if you don't mind |
| 17:00.04 | rockyjo3 | I used freepbx to set up the incoming route, the trunk and the three SIP extensions |
| 17:02.04 | rockyjo3 | the SIP extensions are able to talk to each other so I would say they are properly set up. what I am not able to do is getting any calls from the analog pbx extension I connected to the FXO port |
| 17:03.00 | *** join/#asterisk Natureshadow (nik@shore.naturalnet.de) |
| 17:04.03 | Natureshadow | Hi. Is there a way to have asterisk ignore non-zero returns from dialplan applications? I use ReceiveFax and it terminates non-zero because of transmission errors, but the fax does go through in fact ... |
| 17:04.35 | rockyjo3 | in the Inbound section of Freepbx I created an empty route leaving everything as default and only setting as destination the IVR I created |
| 17:04.55 | Natureshadow | The TIFF file appears and the dial plan should e-mail it after receiving. The file is intact and I really do want it mailed, just Asterisk won't do it because any WARNING from ReceiveFax makes it return from the dialplan extension non-zero |
| 17:05.03 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 17:05.33 | rockyjo3 | but by trying to call the analog extension it keeps ringing without getting to the IVR until it hangs up |
| 17:07.46 | [TK]D-Fender | rockyjo3: Show us your configs for the channels |
| 17:08.06 | [TK]D-Fender | rockyjo3: And the CLI of when you place a call to ir |
| 17:08.14 | *** join/#asterisk elico (~Thunderbi@109.64.229.90) |
| 17:08.52 | [TK]D-Fender | Natureshadow: There is no "after" with faxing. It ends the call. Do your processing in "h" <- |
| 17:09.47 | Natureshadow | [TK]D-Fender: I tried that. I defined the fax stuff in a macro that is called from the fax extension in the inbound context, and I also tried to put the post-processing in h there. It is never executed |
| 17:10.41 | [TK]D-Fender | Natureshadow: Then you've put things in the wrong place |
| 17:10.50 | rockyjo3 | *********************** |
| 17:10.52 | rockyjo3 | [channels] |
| 17:10.52 | rockyjo3 | language=it |
| 17:10.52 | rockyjo3 | context=from-zaptel |
| 17:10.52 | rockyjo3 | signalling=fxs_ks |
| 17:10.52 | rockyjo3 | rxwink=300 ; Atlas seems to use long (250ms) winks |
| 17:10.52 | rockyjo3 | ; |
| 17:10.53 | rockyjo3 | ; Whether or not to do distinctive ring detection on FXO lines |
| 17:10.53 | rockyjo3 | ; |
| 17:10.54 | rockyjo3 | ;usedistinctiveringdetection=yes |
| 17:10.54 | rockyjo3 | usecallerid=yes |
| 17:10.55 | [TK]D-Fender | Natureshadow: And have forgotten how macros interact with the calling context |
| 17:10.58 | [TK]D-Fender | rockyjo3: PASTEBIN! |
| 17:11.05 | [TK]D-Fender | rockyjo3: do NOT fllod in here |
| 17:11.07 | [TK]D-Fender | ~pb |
| 17:11.08 | 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. |
| 17:11.09 | [TK]D-Fender | ^^^ |
| 17:11.20 | [TK]D-Fender | flood* |
| 17:11.39 | rockyjo3 | yup.. :( |
| 17:12.46 | rockyjo3 | http://pastebin.com/xKKbBttS |
| 17:15.13 | [TK]D-Fender | rockyjo3: AND the files that includes |
| 17:17.04 | rockyjo3 | http://pastebin.com/MvzKU8jJ |
| 17:20.49 | *** join/#asterisk phix (~threat@123-243-44-131.static.tpgi.com.au) |
| 17:21.23 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 17:21.39 | rockyjo3 | the FXO module is on port #1 and FXS in port #4 |
| 17:22.12 | [TK]D-Fender | rockyjo3: You have no "channel =>" line therefor you have NO channels defined at all on that card |
| 17:23.03 | rockyjo3 | oh.. wow! sorry but as you could see I am a very beginner.. how can I configure them? |
| 17:24.18 | *** join/#asterisk cyborg-one (~cyborg-on@188-115-135-131.broadband.tenet.odessa.ua) |
| 17:26.39 | [TK]D-Fender | rockyjo3: There should be a place in the GUI to specify the FXO channels |
| 17:29.58 | rockyjo3 | is it anything to do with /etc/zaptel.conf ? here is its content http://pastebin.com/Dpbp7NST |
| 17:31.59 | [TK]D-Fender | rockyjo3: I just told you the problem |
| 17:33.42 | rockyjo3 | yes [TK]D-Fender, I know. but in that file I see the entries fxsks=1 and fxsko=4 and would like to know if they are the channels you were talking about or not |
| 17:34.04 | [TK]D-Fender | [12:22][TK]D-Fenderrockyjo3: You have no "channel =>" line therefor you have NO channels defined at all on that card |
| 17:34.18 | [TK]D-Fender | I was not talking about zaptel.conf. |
| 17:34.21 | Sicelo | those lines are for the signalling |
| 17:34.32 | [TK]D-Fender | I was clearly talking about zapata.conf and the ones linked to it |
| 17:35.57 | rockyjo3 | ok fair enough, thank you. as I am not finding any place in the gui to set them up, is there a way to set them manually by CLI? |
| 17:37.24 | [TK]D-Fender | rockyjo3: AnorockI see no normal place that is protected from being overwritten. try this : |
| 17:37.39 | [TK]D-Fender | rockyjo3: callerid=asreceived |
| 17:37.48 | [TK]D-Fender | rockyjo3: channel => 1 |
| 17:37.55 | [TK]D-Fender | rockyjo3: At the the bottom of zapata.conf |
| 17:38.06 | [TK]D-Fender | rockyjo3: Restart * and do "zap show channels" at * CLI |
| 17:38.47 | rockyjo3 | ok, thx. will do it now and let you know |
| 17:41.02 | rockyjo3 | trixbox1*CLI> zap show channels |
| 17:41.02 | rockyjo3 | <PROTECTED> |
| 17:41.02 | rockyjo3 | <PROTECTED> |
| 17:41.02 | rockyjo3 | <PROTECTED> |
| 17:41.07 | phix | Lets aterisks |
| 17:41.15 | phix | rockyjo3: dont spam |
| 17:41.25 | phix | [TK]D-Fender: <3 |
| 17:41.26 | p3nguin | It isn't spam. he didn't send you unwanted email. |
| 17:41.28 | *** join/#asterisk gonewage (~gonewage@c-68-54-124-223.hsd1.il.comcast.net) |
| 17:41.53 | phix | p3nguin: He sent an unwanted number of lines of text |
| 17:42.15 | p3nguin | He was kind of asked to do it. |
| 17:42.21 | phix | p3nguin: Spam != email |
| 17:42.34 | p3nguin | And a bunch of lines != spam |
| 17:42.42 | WIMPy | Don't spam about spam. |
| 17:43.00 | phix | bunch of lines == spam if p3nguin == true |
| 17:43.27 | phix | for i in p3nguin do print "shut up p3nguin" |
| 17:43.57 | rockyjo3 | didn't mean to let people argue... I only though a few lines wouldn't bother anyone but as I see I was wrong so will keep using pastebin.com even for a couple of lines |
| 17:44.09 | p3nguin | Did you read the rules about the pastebin? |
| 17:44.24 | phix | rockyjo3: few == 3. you printed 4 |
| 17:44.26 | p3nguin | You were given it just 33 minutes ago. |
| 17:44.36 | p3nguin | I'll give it to you again. |
| 17:44.41 | phix | that is unacceptable :) |
| 17:44.42 | p3nguin | ~pb |
| 17:44.42 | 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. |
| 17:44.58 | phix | p3nguin: thnx |
| 17:45.08 | phix | rockyjo3: use pastebin :) |
| 17:45.31 | phix | I actually dont care, I am only saying this as I was spanked hard last time I pasted that many lines |
| 17:45.45 | rockyjo3 | will do. thx |
| 17:46.07 | p3nguin | Four lines isn't nearly as bad as a screen full. |
| 17:46.13 | WIMPy | Now we're back on topic. |
| 17:46.16 | phix | I mean some people like p3nguin like getting spanked but I dont |
| 17:46.21 | phix | I am weird that way |
| 17:46.23 | WIMPy | It's all about S&M here. |
| 17:46.33 | phix | WIMPy: <3 |
| 17:46.43 | phix | you got your leather helmet on? |
| 17:47.11 | WIMPy | No. A Pickelhaube :-) |
| 17:47.13 | *** join/#asterisk ruben231 (~OpenDial@112.198.91.121) |
| 17:47.25 | phix | WIMPy: that's nasty |
| 17:47.36 | [TK]D-Fender | rockyjo3: the channel is listed at least. It has a chance to work now. |
| 17:47.48 | phix | [TK]D-Fender: <3 |
| 17:48.04 | p3nguin | You can keep your Pickelhelmet to yourself. |
| 17:48.05 | phix | Lets reminise` |
| 17:48.19 | *** join/#asterisk Gugge (gugge@kriminel.dk) |
| 17:48.37 | phix | p3nguin: true words have never been spokem so well as that |
| 17:49.17 | rockyjo3 | was writing you [TK]D-Fender. Thanks a lot! I am now able to get inbound calls. I am now working to let the outbound ones work as well |
| 17:50.15 | [TK]D-Fender | rockyjo3: Keep an eye that config updates don't blow away your lines. You may have to re-enter them again. |
| 17:51.59 | rockyjo3 | I will do that! and thanks again for the great and quick help! |
| 17:53.15 | Natureshadow | [TK]D-Fender: I converted the macro into an context, put the argument givent to the macro into a variable and replaced Macro() with Goto() |
| 17:53.21 | Natureshadow | [TK]D-Fender: Everything works fine now :) |
| 17:53.37 | Natureshadow | [TK]D-Fender: Thanks a lot, fixing this would have cost me hours without your help :)! |
| 17:55.02 | [TK]D-Fender | Natureshadow: You're welcome. Do not forget that Macro's merge with the calling context |
| 17:55.13 | phix | The very blood that flows through you! It's time to die when skin turns blue!!!!11 |
| 17:55.13 | *** join/#asterisk jacekowski (jacekowski@jacekowski.org) |
| 17:56.18 | phix | http://mwomercs.com/ |
| 17:56.29 | phix | Mech warriors are the best |
| 18:15.14 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:15.14 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 18:17.51 | *** join/#asterisk ruben231 (~OpenDial@112.198.90.121) |
| 18:18.28 | *** join/#asterisk mtbf (~ewilded@static.124.15.9.5.clients.your-server.de) |
| 18:20.13 | rockyjo3 | one more issue.. when I call the asterisk extension from another analog extension and leave a message and hangup, by listening to the voicemail of the asterisk extension the message is three minutes long. it seems the hangup is not recognized so it keeps recording till its 3 minutes limit... |
| 18:22.07 | mtbf | Hi guys, I am trying to estimate idle time for some of extensions on my PBX. First of all, I am grabbing all active peers with rasterisk -x 'sip show channels', for the rest I am looking for traces of activity in CDR. |
| 18:22.34 | mtbf | However sometimes extensions appear in channels but nothing gets logged into cdr table and this confuses me. |
| 18:25.15 | *** join/#asterisk gonewage (~gonewage@c-68-54-124-223.hsd1.il.comcast.net) |
| 18:27.02 | mtbf | I know CDR entry is created after the call is hung up, that's why I am trying to eliminate active extensions by analysing show channels output before I ask the database. It looks like I have to consider 'Last message' value too. |
| 18:27.02 | [TK]D-Fender | rockyjo3: Your PBX is not sending CDC to your card |
| 18:27.04 | [TK]D-Fender | ~cdc |
| 18:27.04 | infobot | rumour has it, cdc is the cult of the dead cow - see http://www.cdc.org |
| 18:27.09 | p3nguin | haha |
| 18:27.09 | [TK]D-Fender | ~CDS |
| 18:27.09 | infobot | [~cds] Call Discconect Supervision is a service placed on analog lines to be able to signal you that that the calling party has hung up. This is typically done either by a momentary battery cut, or by a polarity reversal on the line. |
| 18:27.15 | [TK]D-Fender | ^^ |
| 18:27.18 | [TK]D-Fender | s/c/s |
| 18:27.28 | p3nguin | sed: -e expression #1, Unterminated `s' command |
| 18:28.00 | *** join/#asterisk TriJetScud (~TriJetScu@d216-232-208-44.bchsia.telus.net) |
| 18:28.13 | p3nguin | mtbf: Does "core show channels" indicate that any channels are active? |
| 18:31.29 | mtbf | p3nguin: not at the moment, since it's afterhours, however I am going to make some tests with my extension, thanks for the hint with distinguishing active channels. |
| 18:31.55 | p3nguin | I don't know what it means to test with your extension. What is this extension you speak of? |
| 18:32.35 | mtbf | There are no active channels currently, I am going to test how core show channels output looks like when I am making a call. |
| 18:32.50 | p3nguin | That's your phone, not an extension. |
| 18:33.45 | p3nguin | core show channels will show you what extension your phone called, but I'm not sure how that is going to help you. |
| 18:38.00 | p3nguin | The only way I know to find out if extensions are not being called is to check the CDR for the extension called, the start time, and the duration of the call. |
| 18:38.17 | p3nguin | But I don't understand the purpose of this exercise. |
| 18:39.15 | mtbf | p3nguin: I want to know for how long particular extension is idle. As far as I know the CDR record is created after the call, so I want to first check if such call is in the place by checking active channels. |
| 18:39.21 | kaldemar | mtbf: sip show channels shows all sip dialogs, not just those associated with calls. |
| 18:39.42 | mtbf | If the extension is not involved in any active channel, I am asking cdr and calculating now()-(calldate+duration) |
| 18:39.49 | p3nguin | I don't understand the concept of extensions being idle. It doens't make sense to me. |
| 18:40.24 | mtbf | Idle - not calling/being called. |
| 18:40.50 | p3nguin | Extensions are only called. Phones call, extensions get called. |
| 18:41.36 | mtbf | But phones are registered as extensions. |
| 18:41.45 | p3nguin | No, they aren't. |
| 18:41.51 | p3nguin | Phones are registered as peers. |
| 18:41.56 | *** join/#asterisk jonno11 (~jonno11@cpc2-walt12-2-0-cust286.13-2.cable.virginmedia.com) |
| 18:42.14 | rockyjo3 | uhm... so it isn't anything I can do from a configuration perspective as it is something missing from the analog PBX, am I right? |
| 18:42.35 | mtbf | Ok, so I want to estimate for how long particular peers are idle. |
| 18:43.45 | p3nguin | Are you using call queues? |
| 18:44.32 | p3nguin | Queues have the ability to measure stats similar to that for devices in queues, but that requires your calls to be through queues rather than all calls to and from phones. |
| 18:51.43 | [TK]D-Fender | mtbf: You'll clearly have to look at active channels and CDR for this |
| 18:55.05 | *** join/#asterisk yoyolala (~yoyolala@unaffiliated/yoyolala) |
| 19:03.33 | *** join/#asterisk minotaur01 (~minotaur@S01060018e7f9c7df.hm.shawcable.net) |
| 19:04.28 | ziro_axis | challenging question, a project to cover 2000 online user with 200 concurrent calls using asterisk, what is the proper hardware for it? |
| 19:05.26 | ziro_axis | taking in mind that the users are softclients over iPhone or android |
| 19:06.24 | [TK]D-Fender | ziro_axis: What exactly will these calls be doing? Any transcoding involved? Recording? Conferencing? |
| 19:06.30 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 19:06.39 | [TK]D-Fender | ziro_axis: Any kind of more intense processing? |
| 19:07.14 | ziro_axis | the codec of the iphone may uses g723 |
| 19:09.18 | [TK]D-Fender | ziro_axis: * doesn't speak g.723 except with an expensive transcoder card. So forget conferencing without it, or recording in any other format directly, and you'll need all your recordings pre-made in it. |
| 19:10.27 | pabelanger | Well that is fucked up, cdr_adaptive_odbc does seem to work unless you also have cdr_cvs loaded |
| 19:10.32 | pabelanger | csv* |
| 19:10.37 | pabelanger | will have to see why |
| 19:11.40 | *** join/#asterisk hehol (~Adium@ip-176-199-50-218.unitymediagroup.de) |
| 19:20.01 | ziro_axis | in this case no need for it |
| 19:20.18 | ziro_axis | i can use the g711 for iphone |
| 19:21.01 | ziro_axis | it will take more bandwidth in wireless, but it worth sacrify |
| 19:21.33 | ziro_axis | i need to disconnect i'll joinback form home soon |
| 19:25.35 | *** join/#asterisk kleszcz (tick@linuxmafia.pl) |
| 19:34.24 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
| 19:38.41 | weinerk | call-confirmation-macro from AGI - DIAL(M(confirmcall^^^)) |
| 19:38.42 | weinerk | How can I tell after Dial() returns if the call was ACCEPTED or REJECTED. |
| 19:50.18 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 19:51.37 | *** join/#asterisk mjordan (~mjordan@user-24-214-136-35.knology.net) |
| 19:51.37 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 19:55.53 | *** join/#asterisk k1ng (~k1ng@unaffiliated/k1ng) |
| 19:56.58 | k1ng | Hi, i am having issue with user 0722 and 0723 can make call but no other |
| 20:01.59 | phix | EU? |
| 20:02.21 | [TK]D-Fender | k1ng: And what are these 2 numbers? |
| 20:02.46 | [TK]D-Fender | k1ng: How do they relate to an actualy technology that Asterisk can use? |
| 20:03.17 | [TK]D-Fender | k1ng: What does "make call" mean? What is this "no other" you mention afterwards? |
| 20:03.33 | k1ng | um those 2 numbers are username.. |
| 20:03.47 | k1ng | i dont know what you mean by second question |
| 20:04.13 | *** join/#asterisk shadar (~eugene@37.113.133.194) |
| 20:04.43 | [TK]D-Fender | k1ng: What is a "username" What KIND of device are you talking about and what is the actual problem? |
| 20:04.58 | k1ng | my asterisk server is just for internal call. no trunk. only user 0722 can make call to 0723 and user 0723 can make call to 0722 |
| 20:05.53 | [TK]D-Fender | k1ng: What kind of devices are those? |
| 20:06.05 | [TK]D-Fender | k1ng: You have still not told us what the problem is. |
| 20:07.53 | k1ng | [TK]D-Fender, no device used. just internal. |
| 20:08.35 | k1ng | [TK]D-Fender, i am not sure what is the problem. there is more than 20 users but they cannot make call to another asterisk user at all |
| 20:09.01 | [TK]D-Fender | k1ng: a "number" isn't a magic thing that talks to Asterisk. |
| 20:09.06 | [TK]D-Fender | k1ng: DEVICES talk to Asterisk |
| 20:09.36 | k1ng | [TK]D-Fender, can i show you any file?? |
| 20:09.43 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 20:09.44 | [TK]D-Fender | k1ng: k1ng You can answer my question |
| 20:10.00 | k1ng | by devices you mean? |
| 20:10.10 | k1ng | SIP trunk? |
| 20:10.26 | k1ng | or the ip phone? |
| 20:10.32 | [TK]D-Fender | k1ng: If I beat you over the head with a stick, then the STICK is the device. |
| 20:10.44 | [TK]D-Fender | k1ng: Calls do not come out of thin air. a DEVICE places them. |
| 20:11.03 | [TK]D-Fender | k1ng: If I phone my friend I might be using a CELL PHONE to place a call. That si a device. |
| 20:11.03 | k1ng | a vps with ubuntu |
| 20:11.08 | [TK]D-Fender | k1ng: WHAT is placing these calls? |
| 20:11.32 | [TK]D-Fender | [15:11]k1nga vps with ubuntu That is your SERVER, not the DEVICES that are esending it calls |
| 20:12.09 | k1ng | lol. i am really confused with term devices |
| 20:12.25 | [TK]D-Fender | .... |
| 20:12.36 | [TK]D-Fender | k1ng: a CELL PHONE is a device |
| 20:12.41 | [TK]D-Fender | an ANALOG PHONE is a device |
| 20:12.52 | [TK]D-Fender | a SOFT-PHONE on a PC using a VoIP protocol is a DEVICE |
| 20:12.57 | [TK]D-Fender | a ***THING*** places a call. |
| 20:13.04 | k1ng | soft phone |
| 20:13.26 | [TK]D-Fender | k1ng: When I ask you what is placing the call... the answer in this case is a SOFT PHONE. |
| 20:13.27 | k1ng | [TK]D-Fender, a custom soft phone. |
| 20:13.54 | [TK]D-Fender | k1ng: the problems a soft phone can have are nothing like the problems an analog phone connected to a PCI TDM card in my server will have |
| 20:14.03 | [TK]D-Fender | k1ng: what does "custom" mean? |
| 20:14.10 | [TK]D-Fender | k1ng: what is "custom:" about it? |
| 20:14.19 | [TK]D-Fender | k1ng: What is the actual problem with your call? |
| 20:16.23 | k1ng | custom mean a softphone was created by developer who works in my client's office and it was intergrate to a software |
| 20:17.41 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 20:21.17 | phix | yawn |
| 20:23.10 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.171) |
| 20:23.53 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 20:27.37 | *** join/#asterisk tonikasch (~tonikasch@unaffiliated/tonikasch) |
| 21:06.07 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 21:29.10 | *** join/#asterisk mjordan (~mjordan@user-24-214-136-35.knology.net) |
| 21:29.10 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 21:29.28 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 21:41.02 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 21:52.44 | *** join/#asterisk xn|adept[ks] (~pi@rrcs-97-77-208-162.sw.biz.rr.com) |
| 21:59.10 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 22:13.45 | *** join/#asterisk F|ReSTaRT (~dlyh@unaffiliated/firestart) |
| 22:30.15 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 22:42.06 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 22:47.01 | *** join/#asterisk gonewage (~gonewage@c-68-54-124-223.hsd1.il.comcast.net) |
| 22:52.33 | *** join/#asterisk jonno11 (~jonno11@cpc2-walt12-2-0-cust286.13-2.cable.virginmedia.com) |
| 22:52.57 | jonno11 | Hi - does anybody have any experience with configuring Sipgate with Asterisk? |
| 22:52.59 | *** join/#asterisk Robotman321 (~brad@50-194-126-9-static.hfc.comcastbusiness.net) |
| 22:53.27 | jonno11 | I just can't get it to work |
| 22:53.48 | *** join/#asterisk elico (~Thunderbi@bzq-79-179-201-209.red.bezeqint.net) |
| 22:55.10 | WIMPy | That is bot the kind/amount of information that enables anybody to help. |
| 22:55.26 | WIMPy | s/bot/not/ |
| 22:56.25 | jonno11 | http://pastebin.com/a0Z1hHyw |
| 22:56.45 | jonno11 | That is my sip.conf file |
| 22:57.13 | jonno11 | I am such a n00b, I can't even tell how to check if it's connecting! |
| 22:58.05 | WIMPy | SIP is stateless so there is no "connection". |
| 22:58.08 | jonno11 | WIMPy: sip show registry says: 0 SIP registrations. |
| 22:58.15 | WIMPy | For the register see 'sip show registry'. |
| 22:58.24 | WIMPy | Otherwise place a call. |
| 22:58.36 | WIMPy | Is that the complete sip.conf? |
| 22:58.44 | jonno11 | No |
| 22:58.56 | jonno11 | I have 2 functioning extensions setup |
| 22:59.04 | jonno11 | 2000 + 2001 |
| 22:59.26 | p3nguin | Extensions are not found in sip.conf. |
| 22:59.40 | *** join/#asterisk iceyp (~icepick@watcher.vibecommunications.co.nz) |
| 23:00.12 | WIMPy | If you have 0 registrations, your sip.conf must be broken in a way that it doesn't see that register line you PBd. |
| 23:00.21 | p3nguin | Care to guess where extensions are configured? |
| 23:00.33 | iceyp | hey guys, I'm having issues disabling T.38 fax support, I want to pass voice calls from our SIP device direct to the SIP gateway without asterisk trying to do T.38 on it... How can I disable this as I continue to see errors like Dec 10 11:41:37] NOTICE[3694]: res_rtp_asterisk.c:2190 ast_rtp_read: Unknown RTP codec 100 received from |
| 23:01.13 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:01.14 | jonno11 | Ok well whatever this is http://pastebin.com/dUZeX65t |
| 23:01.22 | jonno11 | p3nguin ^ |
| 23:01.28 | p3nguin | I'll give you a hint: extensions.conf <----- |
| 23:01.46 | jonno11 | p3nguin: Wonderful sarcasm, thanks for your help |
| 23:01.47 | p3nguin | This is part of a sip.conf |
| 23:01.53 | jonno11 | Yes |
| 23:02.14 | p3nguin | You need a register statement and a peer for sipgate. |
| 23:02.24 | WIMPy | So where has that register gone? |
| 23:02.48 | WIMPy | Don;t show us out of context sniplets. If we can't see what your doing we can;t help. |
| 23:03.29 | jonno11 | The full sip.conf is this http://pastebin.com/qs25y1d1 |
| 23:03.35 | jonno11 | WIMPy ^ |
| 23:04.02 | p3nguin | Register statments must go in the general section. |
| 23:04.08 | p3nguin | sipgate isn't behind nat. |
| 23:04.14 | WIMPy | Ok, so there we have it. Register lines go under [general]. |
| 23:04.20 | p3nguin | You should make them a peer, not a friend. |
| 23:04.45 | p3nguin | username and defaultuser are the old and new parameters for the same thing. |
| 23:05.02 | p3nguin | Bunches of problems there. |
| 23:05.12 | WIMPy | And fromuser, defaultuser and outboundprovy are most probably not needed. |
| 23:05.14 | jonno11 | WIMPy: Aha okay thanks |
| 23:05.28 | iceyp | Can anyone tell me how to get asterisk not to do T.38 faxing? I just want asterisk to transparently deliver the call |
| 23:05.35 | p3nguin | Do you call outbound via sipgate? |
| 23:05.51 | jonno11 | WIMPy: is there a good, n00b friendly resource I can read? |
| 23:06.02 | p3nguin | I'm giving you the answers. |
| 23:06.04 | jonno11 | p3nguin: No, inbound |
| 23:06.09 | p3nguin | Just inbound? |
| 23:06.24 | WIMPy | Doesn't sipgate have a sample config on their help pages? |
| 23:06.33 | p3nguin | It's likely to be wrong. |
| 23:06.39 | p3nguin | All ITSPs have wrong samples. |
| 23:06.53 | jonno11 | no they don't unfortunately |
| 23:07.00 | WIMPy | Might be right. |
| 23:07.12 | WIMPy | Althoug I wouldn't blame the ITSPs for that. |
| 23:07.45 | jonno11 | Ok so now that issue is corrected, how can I test? |
| 23:07.55 | WIMPy | call |
| 23:08.49 | jonno11 | Ok so calling doesn't work |
| 23:08.56 | jonno11 | but sip show registry does |
| 23:09.18 | p3nguin | Here's an example of a good sipgate peer entry: http://pastebin.com/HiXLjUC2 |
| 23:09.26 | WIMPy | Did you see the call in the CLI? |
| 23:09.36 | p3nguin | Put your register statement in the general section. Save, sip reload. |
| 23:10.41 | p3nguin | The register statement syntax is register => username:password@host/phone-number |
| 23:10.58 | WIMPy | Plus a lot more. |
| 23:11.05 | p3nguin | Nope. |
| 23:11.10 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.216) |
| 23:11.13 | p3nguin | That is the syntax he needed for sipgate. |
| 23:11.21 | jonno11 | no secret? |
| 23:11.34 | WIMPy | That's not what you said. |
| 23:11.38 | p3nguin | For yours, for example: register => 1234567890:aBcDeFgH@sipgate.com/3145551212 |
| 23:11.50 | jonno11 | I meant in your example |
| 23:12.16 | p3nguin | Copy it, paste it in your sip.conf, change your defaultuser and host. |
| 23:12.32 | jonno11 | Ok have done |
| 23:12.39 | p3nguin | Fix your register statement. |
| 23:12.40 | jonno11 | but I don't need to add secret=*? |
| 23:12.49 | p3nguin | Correct. They will NEVER authenticate calls to you. |
| 23:13.24 | p3nguin | secret is the "incoming password" for lack of a better description. |
| 23:13.29 | WIMPy | You only need that when placing calls to/via them. |
| 23:13.34 | jonno11 | aha okay |
| 23:13.35 | p3nguin | Nope, wrong again. |
| 23:13.48 | p3nguin | If the peer is able to authenticate on invites TO you, then you can use a secret. |
| 23:14.08 | jonno11 | I'm going to need to clue up on this terminology |
| 23:14.10 | WIMPy | Yes, but they don't. |
| 23:14.28 | p3nguin | If they auth calls to you, and if you send calls to them and they need authentication, you use secret. |
| 23:14.43 | p3nguin | If they do not auth calls to you, but you have to auth to them, use remotesecret. |
| 23:15.00 | jonno11 | YES okay brilliant |
| 23:15.05 | jonno11 | Console showed the call |
| 23:15.09 | p3nguin | But you said you do not make calls to them. |
| 23:15.15 | p3nguin | And they don't auth calls to you. |
| 23:15.21 | p3nguin | So no secret and no remotesecret. |
| 23:15.43 | jonno11 | So how do I handle this call? |
| 23:16.07 | WIMPy | The message you got should tell you wat's missing. |
| 23:16.09 | jonno11 | Also, just a quick question, how can I disconnect from the CLI without ending the asterisk session? |
| 23:16.18 | p3nguin | In the context that you have assigned to the sipgate peer, you have to create an extension that matches your phone number. |
| 23:16.48 | WIMPy | Probably the customer number. |
| 23:16.53 | p3nguin | When asterisk is running in the background, you use asterisk -R or asterisk -r to connect to the running console. |
| 23:17.03 | jonno11 | Yep, but how do I disconnect |
| 23:17.04 | p3nguin | Send a sigint to exit. |
| 23:17.12 | WIMPy | But you can register whatever you want with them IIRC. |
| 23:17.12 | p3nguin | aka ctrl+c |
| 23:17.26 | jonno11 | ^C exits |
| 23:17.31 | jonno11 | it stops the server |
| 23:17.37 | *** join/#asterisk elico (~Thunderbi@bzq-79-182-206-190.red.bezeqint.net) |
| 23:17.40 | p3nguin | Then you aren't running it in the background like I said. |
| 23:17.48 | p3nguin | Start asterisk in the background first. |
| 23:18.01 | jonno11 | how can I do that with the verbosity set? |
| 23:18.11 | p3nguin | You should have an init script to do it for you. |
| 23:18.20 | jonno11 | aha gotcha |
| 23:18.22 | jonno11 | cheers |
| 23:19.02 | p3nguin | Run asterisk as a server. Connect to it with asterisk -R or asterisk -r. |
| 23:19.09 | p3nguin | To change verbose levels, core set verbose <level> |
| 23:19.24 | p3nguin | 3 is suitable for most things. |
| 23:19.52 | jonno11 | I'm getting "Using SIP RTP CoS mark 5" on incoming |
| 23:19.56 | jonno11 | but that's it |
| 23:20.10 | WIMPy | increase verbose. |
| 23:20.19 | p3nguin | If you need to know what is happening on the call, you'll want sip debug, not core verbosity. |
| 23:20.23 | ChannelZ | MORE COWBELL! |
| 23:20.29 | WIMPy | That message is not helpful. |
| 23:20.36 | p3nguin | sip set debug on |
| 23:20.40 | p3nguin | Then make your call. |
| 23:21.00 | WIMPy | Isn't that a bit overkilll? |
| 23:21.31 | p3nguin | To see what the call is doing? I wouldn't think so. |
| 23:21.55 | p3nguin | I'm sure he doesn't have an appropriate extension in the context where the call is going. The sip debug will reveal it all to him. |
| 23:22.00 | WIMPy | You usually get a very detaild message giving context and extension. |
| 23:22.23 | *** join/#asterisk lukejt (~luke@149.241.199.189) |
| 23:22.30 | p3nguin | If the context and extension do not exist, will you expect to see it still? |
| 23:22.41 | WIMPy | yes |
| 23:23.32 | jonno11 | Ok so I have this configured under sip gate-inbound |
| 23:23.37 | jonno11 | sipgate-inbound* |
| 23:23.39 | jonno11 | exten => 1643437,1,Dial(SIP/2000) |
| 23:23.56 | p3nguin | Is 1643437 your phone number? |
| 23:23.57 | jonno11 | exten => USERNAME,1,Dial(SIP/2000) |
| 23:24.05 | jonno11 | No it's the sipgate username |
| 23:24.12 | p3nguin | Is it also the phone number that you used in the register statment after the slash? |
| 23:24.34 | jonno11 | Yep |
| 23:24.40 | p3nguin | You should consider using the phone number for standardization. |
| 23:24.51 | *** join/#asterisk areski (~areski@80.174.255.71.dyn.user.ono.com) |
| 23:25.00 | WIMPy | The customer number is a phone number as well. |
| 23:25.03 | p3nguin | But if you used the same thing in the register and in the extension, it should be working. |
| 23:25.09 | p3nguin | The user name is not a phone number. |
| 23:25.14 | jonno11 | It's just hanging up unfortunately |
| 23:25.22 | WIMPy | With sipgate it is. |
| 23:25.29 | jonno11 | Yeah it is, you can call others |
| 23:25.30 | p3nguin | For example, my sipgate user name is something like 7654e6fw6 |
| 23:25.34 | WIMPy | core set verbose 3. |
| 23:25.44 | p3nguin | But my phone number is something like 4154499909. |
| 23:25.56 | p3nguin | So no, it isn't that way with sipgate. |
| 23:26.03 | WIMPy | Other sipgate customers can call both of them. |
| 23:26.28 | jonno11 | Either way, as it's setup, it's not working |
| 23:26.33 | jonno11 | Phone is just hanging up |
| 23:26.38 | p3nguin | Still waiting on that sip debug. |
| 23:26.43 | lukejt | from the AGI, how can I get the (dial) answered time? |
| 23:26.48 | WIMPy | So if you have team or plus or something, they will be able to directly call all your accounts. |
| 23:30.12 | lukejt | or more to the point, once a call completes, why are CDR() variables empty |
| 23:30.53 | jonno11 | p3nguin: Have sent that debug over |
| 23:31.03 | p3nguin | I don't see it. |
| 23:31.33 | jonno11 | PM |
| 23:31.35 | jonno11 | http://pastebin.com/fqGMLRym |
| 23:31.39 | p3nguin | I don't do PM. |
| 23:31.39 | jonno11 | nevermind |
| 23:31.40 | jonno11 | haha |
| 23:32.14 | p3nguin | Where's the rest of it? |
| 23:32.14 | WIMPy | lukejt: What about ANSWEREDTIME? |
| 23:32.26 | p3nguin | That isn't a call. |
| 23:32.40 | p3nguin | sip set debug on, then make a call to your sipgate number. |
| 23:33.28 | jonno11 | It's extremely difficult to find the beginning/end |
| 23:34.05 | p3nguin | I would rather see too much as opposed to not enough. |
| 23:34.15 | jonno11 | Ok here it is |
| 23:34.16 | jonno11 | http://pastebin.com/cMZ3hLsV |
| 23:35.40 | p3nguin | Run this for me: dialplan show 1643437@sipgate-inbound |
| 23:36.03 | jonno11 | There is no existence of 'sipgate-inbound' context |
| 23:36.08 | p3nguin | That's a problem. |
| 23:36.14 | WIMPy | Wonders if the call actually succeeded. |
| 23:36.18 | p3nguin | Do you have any context related to sipgate already? |
| 23:36.53 | jonno11 | http://pastebin.com/S7dPHr6X |
| 23:37.07 | jonno11 | That's my extensions.conf |
| 23:37.15 | p3nguin | Did you run dialplan reload? |
| 23:37.56 | p3nguin | You have to reload the dialplan after you make changes to it, just like you have to sip reload after you change sip.conf. |
| 23:38.00 | jonno11 | Aha okay |
| 23:38.06 | jonno11 | I thought I had rebooted asterisk |
| 23:38.26 | p3nguin | unnecessary in any event. |
| 23:38.52 | lukejt | WIMPy: DIALEDTIME, ANSWEREDTIME are populated, but it seems hacky to me. Ideally I'd like to access CDR(answer) rather than working it out myself (which could be error-prone) |
| 23:39.27 | jonno11 | Brilliant it works |
| 23:39.29 | jonno11 | however |
| 23:39.33 | WIMPy | lukejt: CDR is written after the call has ended. |
| 23:39.50 | jonno11 | Is there any Asterisk reason why I can't hear audio on my computer's end |
| 23:40.07 | WIMPy | lukejt: If it is available in the h extension depends on cdr configuration. |
| 23:40.16 | *** join/#asterisk elico (~Thunderbi@bzq-79-182-198-188.red.bezeqint.net) |
| 23:40.43 | WIMPy | jonno11: Your firewall. |
| 23:40.45 | p3nguin | NAT problems often cause lack of audio, but I don't know if that's your actual issue or not since I don't know how your computer comes into play. |
| 23:41.11 | jonno11 | Hmm |
| 23:41.17 | jonno11 | Audio is fine on my phone |
| 23:42.04 | lukejt | WIMPy: I'm accessing CDR after the call is finished (in my destructor) - once the channel is dead. Using MySQL to store CDRs. Perhaps they are being written to DB and flushed or something? |
| 23:42.15 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 23:42.48 | WIMPy | lukejt: What "destructor"? |
| 23:46.04 | lukejt | WIMPy: Part of my AGI code that gets called once the call is finished |
| 23:46.36 | lukejt | where i'm calculating the call costs and trying to access CDR/channel variables |
| 23:46.41 | WIMPy | When / from where exactely? |
| 23:47.29 | p3nguin | Don't you have to use DeadAGI() for a call that has already ended? |
| 23:47.48 | jonno11 | this call quality is really bad |
| 23:48.00 | WIMPy | That's probably true, yes. |
| 23:48.05 | p3nguin | Do you have low bandwidth? |
| 23:48.55 | lukejt | p3nguin: not anymore, DeadAGI is deprecated. the same functionaity is now included in the normal agi call |
| 23:49.04 | p3nguin | Okay, that's good to know. |
| 23:49.26 | WIMPy | didn't know that, either. |
| 23:54.45 | lukejt | http://pastebin.com/1C7cF9pn |
| 23:54.55 | lukejt | that's what I'm doing - slightly sanitised |
| 23:55.25 | lukejt | and I'm getting.. (coming) |
| 23:56.53 | lukejt | AGI debug log |
| 23:56.57 | lukejt | http://pastebin.com/tZqq7x2m |
| 23:58.05 | lukejt | as you see, most of the CDR() vars are empty, and CDR(start) is showing an incorrect value |
| 23:58.37 | lukejt | it's actually showing the value for CDR(end) |