| 00:03.34 | devdvd | TehRabbitt, |
| 00:03.50 | devdvd | that is probably because your extension isnt in the incoming context |
| 00:04.26 | devdvd | is your incoming line sip ? |
| 00:04.29 | TehRabbitt | now I can get it to answer,but i just hear a "click" when it answers |
| 00:04.33 | TehRabbitt | yes incoming line is SIP |
| 00:04.48 | TehRabbitt | [inbound] exten => 12013992818,1,Answer() exten => s,n,Wait(1) exten => s,n,Playback(tt-weasels) |
| 00:05.06 | devdvd | does it play weasels ? |
| 00:05.23 | TehRabbitt | again, nothing, just a "click click" you can try dialing it yourself, it's just silence 0_o |
| 00:06.06 | devdvd | is your asterisk box behind a nat? |
| 00:06.23 | TehRabbitt | Hm maybe I didn't open the ports 0_o |
| 00:07.11 | TehRabbitt | Yep, I feel retarted now, I never opened any ports for it... :-\ |
| 00:07.24 | devdvd | does it work now? |
| 00:08.40 | TehRabbitt | hold on just opened the ports... going to enable port triggering as well |
| 00:09.00 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-138-161.ks.ks.cox.net) |
| 00:09.27 | TehRabbitt | Ok i'm gonna try it now |
| 00:09.54 | TehRabbitt | still nothing :-\ |
| 00:10.01 | devdvd | rtp set debug on |
| 00:10.21 | devdvd | also make sure the rtp ports are open |
| 00:10.26 | TehRabbitt | Got RTP packet from 4.55.17.2:24840 (type 00, seq 000000, ts 000000, len 000160) |
| 00:10.42 | TehRabbitt | that port is WAY outside what I specified in sip.conf.... |
| 00:10.55 | devdvd | sip.conf has nothing to do with it |
| 00:10.58 | TehRabbitt | 24840 is outside 10000-20000.. |
| 00:11.05 | TehRabbitt | should I make it 10000-30000? |
| 00:11.17 | devdvd | what is 4.55.17.2? |
| 00:11.26 | TehRabbitt | flowroute i suppose |
| 00:11.45 | devdvd | that's just a source endpoint, that dont matter, its about what port its received on |
| 00:12.19 | devdvd | i have a suspicion that your problem is nat related |
| 00:12.27 | devdvd | does your router support dmz? |
| 00:12.32 | TehRabbitt | yes |
| 00:12.38 | devdvd | try throwing your asterisk box in there |
| 00:12.45 | devdvd | might help |
| 00:12.55 | TehRabbitt | thing is, it was working perfectly fine before upgrading to 1.6... worked back in october on 1.4 |
| 00:13.09 | TehRabbitt | i'm "starting from scratch" since I felt the original install I had was kinda just thrown togehter |
| 00:13.24 | devdvd | soemthing may have changed between 1.4 and 1.6 in regards to how nat is handled |
| 00:13.42 | TehRabbitt | i've thrown it on dmz |
| 00:13.45 | TehRabbitt | still no luck |
| 00:14.04 | devdvd | whats your entry in sip.conf look like for your provider? |
| 00:14.06 | devdvd | pastebin it |
| 00:14.33 | TehRabbitt | hm, my PuTTY session keeps timing out... maybe it's my server... hmph |
| 00:15.32 | TehRabbitt | http://pastebin.com/iv3ErLMP |
| 00:16.28 | devdvd | ok, now that its in the dmz, try nat=no |
| 00:17.30 | *** join/#asterisk FransWillem (53533603@gateway/web/freenode/ip.83.83.54.3) |
| 00:17.38 | TehRabbitt | nope |
| 00:18.18 | TehRabbitt | it's litterly answering, then hanging up |
| 00:18.23 | FransWillem | Would anyone be able to troubleshoot my simple PBX system to have a Blink win32 client talk to an android gingerbread client ? Ringing works, but whenever a call is answered it just drops with a 58 "Bearer capability not available" code :/ |
| 00:21.09 | FransWillem | From the debug logs it appears that Blink has a problem with an o= line somewhere :/ |
| 00:23.32 | *** join/#asterisk TehRabbitt (~TehRabbit@pool-71-172-89-155.nwrknj.fios.verizon.net) |
| 00:23.34 | TehRabbitt | back |
| 00:23.44 | TehRabbitt | devdvd: == Using SIP RTP CoS mark 5 -- Executing [12013992818@inbound:1] Answer("SIP/flowroute-0000000c", "") in new stack -- Auto fallthrough, channel 'SIP/flowroute-0000000c' status is 'UNKNOWN' |
| 00:23.50 | TehRabbitt | thats the latest message I got at the console |
| 00:24.16 | devdvd | hmm, sip show registry |
| 00:24.19 | devdvd | what does it show |
| 00:24.32 | TehRabbitt | Host dnsmgr Username Refresh State Reg.Time sip.flowroute.com:5060 N 00076354 105 Registered Sun, 27 Feb 2011 19:23:18 1 SIP registrations. |
| 00:25.39 | devdvd | show us your dialplan |
| 00:25.56 | devdvd | whatever the problem is i have a feeling its in your dialplan |
| 00:26.22 | TehRabbitt | http://pastebin.com/pjr7AWUE |
| 00:26.32 | TehRabbitt | again very basic just trying to get it working lol |
| 00:26.42 | FransWillem | Hmmm, Android<->SIP Communicator works fine, but Android<->Blink fails for some unknown reason. Does anyone know how I could troubleshoot this ? |
| 00:28.00 | TehRabbitt | hey devdvd, i'll brb in like 15 min, i gotta take care of a couple things |
| 00:28.14 | devdvd | http://pastebin.com/GgqjXgJj |
| 00:28.17 | devdvd | try that |
| 00:28.24 | TehRabbitt | alright hold on |
| 00:30.01 | TehRabbitt | still nothing but got more info from debug... i'll brb in 15... i'll paste bin it in the meantime |
| 00:30.20 | TehRabbitt | http://pastebin.com/1zNu4veM |
| 00:36.13 | *** join/#asterisk strehlow (~strehlow@97-83-152-238.dhcp.eucl.wi.charter.com) |
| 00:42.33 | FransWillem | Are there any known problems between Asterisk and the sipsimpleclient skd ? |
| 01:03.32 | *** join/#asterisk coppice (~chatzilla@60.157.17.210.dyn.pacific.net.hk) |
| 01:09.55 | *** join/#asterisk capitan (~captain@76.91.206.32) |
| 01:19.13 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 01:21.23 | *** join/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110) |
| 01:22.27 | *** join/#asterisk nestAr (nester@blackbox.ninjaz.net) |
| 01:23.01 | TehRabbitt | devdvd: hey you still around? |
| 01:23.50 | devdvd | yea |
| 01:23.58 | TehRabbitt | did you see the last pastebin i sent? |
| 01:24.01 | devdvd | yea |
| 01:24.07 | devdvd | looking at it |
| 01:24.10 | TehRabbitt | aight |
| 01:25.03 | devdvd | Do this |
| 01:25.06 | devdvd | change Playback |
| 01:25.07 | devdvd | to echo() |
| 01:25.12 | devdvd | and see if you can echo |
| 01:25.42 | TehRabbitt | hm aight |
| 01:26.00 | devdvd | im wondering if the playback function isn't registered or something |
| 01:27.44 | TehRabbitt | 0_o got weasels working |
| 01:27.48 | devdvd | ok |
| 01:27.49 | devdvd | :) |
| 01:27.54 | TehRabbitt | had a space between tt- weasels |
| 01:27.58 | TehRabbitt | i didnt' realize it 0_o |
| 01:27.59 | TehRabbitt | lol |
| 01:28.08 | devdvd | hmm |
| 01:28.14 | devdvd | the paste didnt show that |
| 01:28.19 | TehRabbitt | which is weird :-\ |
| 01:28.29 | devdvd | 13992818,n,Playback(tt-weasels) |
| 01:28.29 | TehRabbitt | i'm thinking maybe it was my terminal buffer or something? |
| 01:28.34 | devdvd | who knows |
| 01:29.15 | TehRabbitt | i litterly just re-typed "tt-weasels" inside the () and realized it was shorter, so i hit undo, and noticed the space |
| 01:29.16 | TehRabbitt | lol |
| 01:29.23 | devdvd | haha |
| 01:29.29 | devdvd | so now is your problem solved |
| 01:30.19 | TehRabbitt | well part 1 is... which is getting inbound calling to work... second part is a much larger headache but that i'll tackle later.. (getting chan_sccp-b working) |
| 01:34.11 | devdvd | haha |
| 01:34.15 | devdvd | i just tackled that myself |
| 01:34.30 | TehRabbitt | you did? 0_o i might need your help in a few haha |
| 01:34.38 | devdvd | yep |
| 01:39.07 | *** join/#asterisk [netman] (~netman@241.Red-80-39-202.dynamicIP.rima-tde.net) |
| 01:39.46 | *** join/#asterisk shapr (~shapr@nat/digium/x-iesecaqjbpcmgnhb) |
| 01:55.51 | *** join/#asterisk joobie (~joobie@unaffiliated/moo0o0ooo00o0o0o) |
| 01:56.49 | joobie | hey guys.. got an issue where a particular phone comes up as "in use" when i do 'sip show inuse' or look at the device in the queue using 'queue show' .. only problem is the phone is not in use. The user can make calls, but because they come up as 'in use' in the queue, they dont get any queu calls |
| 01:57.00 | joobie | any ideas how to resolve this one? the user has rebooted the phone, same deal |
| 01:59.07 | TehRabbitt | devdvd: well I just downloaded the source, gonna compile,cross my fingers and see if I can get it working lol |
| 01:59.19 | devdvd | teh, grab the svn |
| 01:59.58 | devdvd | then just ./configure |
| 01:59.59 | devdvd | make |
| 02:00.01 | devdvd | make install |
| 02:00.11 | devdvd | then restart asterisk |
| 02:00.20 | TehRabbitt | how do I grab the svn? |
| 02:00.26 | devdvd | when it comes back up you can test if it installed right by doing sccp ? |
| 02:00.31 | devdvd | and it shoudl give you a list of options |
| 02:01.05 | TehRabbitt | where can I find the svn |
| 02:01.07 | devdvd | svn co https://chan-sccp-b.svn.sourceforge.net/svnroot/chan-sccp-b/trunk chan-sccp-b |
| 02:01.48 | *** join/#asterisk Kumbang (~kumbang@180.245.137.5) |
| 02:03.58 | *** join/#asterisk daxt (~daxt@112.135.86.91) |
| 02:05.08 | daxt | anybody used mydivert ? |
| 02:05.20 | Freeaqingme | I'm trying to visualize an IVR, and I want to make a distinction between local extensions, and calls that are forwarded to an external number. Any ideas? |
| 02:05.24 | TehRabbitt | devdvd: got the svn, compiling lol |
| 02:05.44 | Freeaqingme | daxt, nope, never heard of. Though maybe you can just ask your q? |
| 02:06.00 | daxt | i want to integrate mydivert with * |
| 02:06.08 | daxt | but confused |
| 02:08.18 | TehRabbitt | how do I kill asterisk if I used that "safeasterisk" start thing |
| 02:08.31 | TehRabbitt | basically I need to restart it / I don't have an /etc/init.d/ script for it |
| 02:15.51 | devdvd | you can either go into the asterisk console and do core stop now |
| 02:16.05 | devdvd | or just do killall -9 asterisk |
| 02:16.07 | TehRabbitt | devdvd, ok... it's installed but how do I know it's working lol |
| 02:16.17 | devdvd | did you restart asterisk |
| 02:16.26 | TehRabbitt | yep |
| 02:16.31 | devdvd | go into the console |
| 02:16.33 | devdvd | type sccp ? |
| 02:16.41 | TehRabbitt | No such command 'sccp' (type 'core show help sccp' for other possible commands) |
| 02:16.46 | devdvd | then its not installed right |
| 02:16.59 | devdvd | what's your modules directory? |
| 02:17.10 | devdvd | look in asterisk.conf |
| 02:17.14 | devdvd | see where your modules are |
| 02:17.28 | devdvd | it should either be /usr/lib/asterisk or /usr/lib64/asterisk |
| 02:18.54 | TehRabbitt | chan_sccp.so |
| 02:18.59 | TehRabbitt | i see it here in /modules |
| 02:19.11 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 02:19.18 | ruben23 | hi guys |
| 02:19.31 | TehRabbitt | noload => chan_skinny.so load => chan_sccp.conf |
| 02:19.37 | TehRabbitt | i even put that in... :-\ |
| 02:24.03 | TehRabbitt | idk why it wont find the module |
| 02:25.31 | TehRabbitt | nvermind i found it |
| 02:25.37 | TehRabbitt | i'm 2 for 2 tonight 0_o lmfao |
| 02:26.25 | devdvd | lol |
| 02:27.43 | TehRabbitt | devdvd: nvm still nothing |
| 02:27.44 | TehRabbitt | sigh |
| 02:27.57 | TehRabbitt | thought it was because I had it as chan_sccp.conf instead of .so but that was only part of it |
| 02:27.58 | TehRabbitt | lol |
| 02:30.47 | ruben23 | guys how do i add a dial string or prefix here -----> 992303201 -----> i got this dial plan ----> http://pastebin.com/9NN5tP3r |
| 02:31.09 | ruben23 | it like 992303201 _ 10 digit US number |
| 02:31.13 | devdvd | TehRabbitt, look at /etc/asterisk/asterisk.conf and show me what astmoddir says |
| 02:31.30 | devdvd | mine looks like astmoddir => /usr/lib/asterisk/modules |
| 02:32.38 | TehRabbitt | astmoddir => /usr/lib/asterisk/modules |
| 02:32.44 | TehRabbitt | but mine should be the lib64 i think |
| 02:32.47 | TehRabbitt | since thats what I compiled to |
| 02:33.29 | devdvd | just copy (or symlink) chan_sccp.so into /usr/lib/asterisk/modules |
| 02:34.32 | ruben23 | <PROTECTED> |
| 02:34.47 | ruben23 | any suggestion guys |
| 02:35.39 | devdvd | so you want to add 992303201 in front of the number |
| 02:35.59 | devdvd | ? |
| 02:36.19 | TehRabbitt | usr/lib/asterisk/modules/chan_sccp.so': File exists |
| 02:36.31 | devdvd | ok |
| 02:36.34 | devdvd | restart asterisk |
| 02:37.04 | devdvd | ruben23, |
| 02:37.28 | TehRabbitt | still nothing |
| 02:37.47 | devdvd | did you compile asterisk from source? |
| 02:37.55 | TehRabbitt | Yes |
| 02:38.09 | TehRabbitt | since I dont want to use the one from aptitude through debian's sources |
| 02:38.09 | devdvd | ok, this is gonna sound really dumb cuz you shouldnt have to do this |
| 02:38.10 | devdvd | but |
| 02:38.14 | devdvd | go into the source directory |
| 02:38.20 | devdvd | make menuselect |
| 02:38.24 | devdvd | go to channel drivers |
| 02:38.25 | TehRabbitt | 0_o |
| 02:38.32 | devdvd | unselect chan_skinny |
| 02:38.33 | devdvd | save |
| 02:38.36 | devdvd | ./configure |
| 02:38.36 | devdvd | make |
| 02:38.40 | devdvd | make install |
| 02:38.49 | TehRabbitt | well i already have noload =>chan_skinny :-\ |
| 02:38.58 | TehRabbitt | No such command 'skinny' (type 'core show help skinny' for other possible commands) |
| 02:39.04 | devdvd | i know |
| 02:39.11 | devdvd | i had to actually recompile to make it work |
| 02:39.33 | TehRabbitt | 0_o |
| 02:39.57 | TehRabbitt | sigh i dont have the source anymore, i've gotta dl it again |
| 02:40.16 | ruben23 | devdvd: yes i want to add it...that |
| 02:41.39 | devdvd | exten => _992303201X.,n,Dial(${TrunkGRNvoip}/${EXTEN:9},,To) |
| 02:41.50 | TehRabbitt | so what am I doing now that I have the source? |
| 02:41.55 | devdvd | recompiling asterisk |
| 02:42.03 | TehRabbitt | :-\ |
| 02:42.04 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 02:42.04 | TehRabbitt | lol |
| 02:42.20 | devdvd | well..that process takes a few seconds if you still had your old source :P |
| 02:42.34 | devdvd | and what did we learn here children? always keep the source |
| 02:42.35 | TehRabbitt | takes all of a minute on this machine ;-D |
| 02:42.55 | TehRabbitt | dual 3.8Ghz xeons with 4MB cache and 4GB ram |
| 02:42.56 | TehRabbitt | lol |
| 02:43.00 | devdvd | thats good |
| 02:43.21 | TehRabbitt | still could have shaved 40 seconds or so off of the compile time :-\ |
| 02:43.21 | TehRabbitt | lol |
| 02:43.33 | TehRabbitt | and all the makemenuconfig options I had selected before |
| 02:43.38 | TehRabbitt | which werent any i beleive |
| 02:44.52 | *** join/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110) |
| 02:44.53 | ruben23 | devdvd: why there is ----> /${EXTEN:9}...? |
| 02:45.10 | shapr | That gives you only the ... last? ... nine digits. |
| 02:45.15 | shapr | or the first nine, I forget which. |
| 02:45.23 | devdvd | strips off the first 9 |
| 02:45.35 | TehRabbitt | so once I recompile, just run make install and hope it works this time? lol |
| 02:45.46 | devdvd | haha |
| 02:46.00 | devdvd | ruben23, do you want to pass those 9 digits onto your sip trunk? |
| 02:46.08 | ruben23 | yes |
| 02:46.13 | devdvd | or just have them be a code that someone has to dial before they can get an outside line |
| 02:47.06 | devdvd | if you honestly want to pass them onto your trunk (which i cant see a reason to unless your sip provider requires it) |
| 02:47.09 | devdvd | then take off the :9 |
| 02:47.34 | ruben23 | devdvd: ---> http://pastebin.com/KMTk06uD |
| 02:49.00 | ruben23 | devdvd: what you think should it be if thats my carrier requires me |
| 02:49.14 | devdvd | <devdvd> if you honestly want to pass them onto your trunk (which i cant see a reason to unless your sip provider requires it) |
| 02:49.14 | devdvd | <devdvd> then take off the :9 |
| 02:49.33 | *** join/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110) |
| 02:49.43 | devdvd | exten => _992303201X.,n,Dial(${TrunkGRNvoip}/${EXTEN},,To) |
| 02:50.13 | TehRabbitt | make install |
| 02:50.53 | ruben23 | devdvd: thank you so much for your help, ill test this |
| 02:50.57 | devdvd | ok |
| 02:52.46 | TehRabbitt | devdvd: should I be concerned: http://pastebin.com/CJxT3KLm |
| 02:53.32 | devdvd | no |
| 02:53.36 | devdvd | thats what you should see actually |
| 02:53.50 | devdvd | make sure you have noload=>chan_skinny.so in your asterisk.conf |
| 02:53.51 | devdvd | also |
| 02:53.57 | devdvd | do you have autoload=yes |
| 02:53.58 | devdvd | ? |
| 02:56.14 | TehRabbitt | where would autoload=yes be |
| 02:56.25 | devdvd | in modules.conf |
| 02:56.28 | TehRabbitt | yes uit's there |
| 02:56.32 | devdvd | ok |
| 02:56.41 | devdvd | do noload=>chan_skinny.so in modules.conf |
| 02:56.43 | devdvd | and restart asterisk |
| 02:56.50 | leifmadsen | you're still working on this? :) |
| 02:56.55 | TehRabbitt | Asterisk ended with exit status 1 Asterisk died with code 1. Automatically restarting Asterisk. |
| 02:57.01 | devdvd | leifmadsen, different issue now :) |
| 02:57.05 | leifmadsen | sounds like a crash |
| 02:57.07 | TehRabbitt | and now it's stuck in a loop |
| 02:57.11 | devdvd | hmmm |
| 02:57.12 | Freeaqingme | TehRabbitt, look at a strace or logfile |
| 02:57.20 | leifmadsen | ls /tmp/core* |
| 02:57.55 | TehRabbitt | ls: cannot access /tmp/core*: No such file or directory |
| 02:58.01 | leifmadsen | if it's crashing though, there isn't much you can do other than 1) file an issue, 2) fix it with code 3) figure out how to work around the issue |
| 02:58.15 | leifmadsen | did asterisk crash, or did you just get kicked out of the console |
| 02:58.25 | leifmadsen | also are you using safe_asterisk or and init.d script to start asterisk? |
| 02:58.47 | leifmadsen | you could force a core dump by running asterisk in the foreground with asterisk -cg and then reproducing the crash |
| 02:59.11 | TehRabbitt | safe_asterisk |
| 02:59.34 | TehRabbitt | ah looks like it's already running 0_o |
| 02:59.37 | TehRabbitt | hence the crash? |
| 03:00.00 | leifmadsen | well after a crash the safe_asterisk script should restart asterisk |
| 03:00.04 | leifmadsen | which is why it's runnign again |
| 03:00.09 | leifmadsen | *CLI> core show uptime |
| 03:00.19 | leifmadsen | that'll tell you how long asterisk has been running and help to determine if it crashed/restarted |
| 03:00.42 | TehRabbitt | No such command 'core' (type 'core show help core' for other possible commands) |
| 03:00.48 | leifmadsen | safe_asterisk should be storing a core dump file in /tmp/ by default, but you may have to uncomment a line in it (although it should work by default) |
| 03:00.53 | leifmadsen | what version are you using? |
| 03:00.55 | TehRabbitt | 1.6 |
| 03:01.00 | leifmadsen | 1.6.what? |
| 03:01.04 | leifmadsen | 1.6 is not a version |
| 03:01.21 | leifmadsen | 1.6.0 vs 1.6.1 vs 1.6.2 are all separate branches |
| 03:01.23 | TehRabbitt | Asterisk 1.6.2.16.2 |
| 03:01.30 | leifmadsen | tags are made within that branch |
| 03:01.38 | TehRabbitt | Connected to Asterisk 1.6.2.16.2 currently running on thoth (pid = 2043) |
| 03:01.40 | leifmadsen | 'core show version' should definitely work in that version |
| 03:01.51 | leifmadsen | don't just type 'core' |
| 03:01.52 | TehRabbitt | that does... |
| 03:01.54 | TehRabbitt | ah |
| 03:02.07 | leifmadsen | core show uptime |
| 03:02.13 | TehRabbitt | Last reload: 4 minutes, 31 seconds |
| 03:02.23 | leifmadsen | sounds like it crashed and restarted then |
| 03:02.39 | TehRabbitt | well I had just recompiled it |
| 03:02.41 | leifmadsen | not sure why the core dump didn't get put in /tmp/ but it's not really useful to you |
| 03:02.49 | TehRabbitt | and i went to start it but it was already running i think |
| 03:02.54 | leifmadsen | you're using pure stock asterisk right? (no patches?) |
| 03:02.58 | TehRabbitt | Correct |
| 03:03.06 | TehRabbitt | only thing i'm trying to get working is chan_sccp-b right now |
| 03:03.09 | TehRabbitt | and it's still not loading |
| 03:03.21 | leifmadsen | is chan_sccp shipped with stock asterisk? |
| 03:03.39 | leifmadsen | I don't use skinny phones which is why I don't know |
| 03:03.41 | TehRabbitt | no it's a module you need to compile / copy in |
| 03:03.49 | TehRabbitt | it's a replacement for skinny.conf |
| 03:03.50 | devdvd | TehRabbitt, try removing chan_sccp.so from modules |
| 03:03.51 | leifmadsen | hence my question about vanilla asterisk |
| 03:03.51 | devdvd | and restart |
| 03:04.00 | TehRabbitt | alright |
| 03:04.02 | devdvd | mabye the newest svn broke |
| 03:04.04 | leifmadsen | you said yes, but that is not the case |
| 03:04.26 | TehRabbitt | well asterisk itself isn't really modified for the exception of loading one module from outside which wont even load lol |
| 03:04.29 | leifmadsen | I'd make sure you'd running the version of asterisk compatible with the patch/third party code you're using |
| 03:04.42 | TehRabbitt | well it says 1.6 |
| 03:04.44 | leifmadsen | that sentence didn't make any sense to me |
| 03:04.48 | leifmadsen | 1.6 is not a version or a branch |
| 03:04.56 | leifmadsen | we just went over that |
| 03:05.08 | leifmadsen | 1.6.0 vs 1.6.1 is just as different as 1.2 vs 1.4 |
| 03:05.13 | leifmadsen | they are major version changes |
| 03:05.17 | TehRabbitt | http://www.voip-info.org/wiki/view/chan_sccp-b |
| 03:06.01 | leifmadsen | well I've exhausted my ability to help you as you're using a technology I'm not familiar with and are using a third party module |
| 03:06.28 | TehRabbitt | err http://chan-sccp-b.sourceforge.net/ |
| 03:06.45 | devdvd | TehRabbitt, go ahead and remove that module |
| 03:06.49 | devdvd | from your modules dir |
| 03:06.49 | TehRabbitt | aight |
| 03:06.50 | devdvd | and restart |
| 03:06.54 | leifmadsen | I suggest getting support from the developers of that module |
| 03:06.56 | devdvd | see if maybe the svn broke it |
| 03:07.07 | devdvd | i know for a fact it works |
| 03:07.24 | TehRabbitt | so recompile using the latest "stable" on their site you think devdvd? |
| 03:07.26 | devdvd | but svn...hard to say what changed from last night to now :) |
| 03:07.39 | TehRabbitt | lol |
| 03:07.42 | devdvd | if removing the module fixes it |
| 03:07.42 | devdvd | yes |
| 03:08.29 | TehRabbitt | yea seems to be working after removing :-\ |
| 03:08.32 | devdvd | ok |
| 03:08.34 | TehRabbitt | i'm gonna go with stable for now lol |
| 03:08.36 | TehRabbitt | weird |
| 03:08.39 | devdvd | recomile from stable /sadface |
| 03:08.45 | devdvd | not really |
| 03:08.59 | TehRabbitt | well like you said, what might have changed since yesterdray lol |
| 03:11.30 | TehRabbitt | devdvd: recompiling... hopefully this version works |
| 03:13.53 | TehRabbitt | ok so the module is the same exact file in both lib64 and regular lib |
| 03:14.12 | devdvd | well |
| 03:14.22 | devdvd | lib is 32 and lib64 is 64 bit |
| 03:14.22 | TehRabbitt | now how can I tell if it's loaded |
| 03:14.33 | TehRabbitt | 9d17288630756b882ab980b8e7e456a0 /usr/lib/asterisk/modules/chan_sccp.so |
| 03:14.38 | TehRabbitt | 9d17288630756b882ab980b8e7e456a0 /usr/lib64/asterisk/modules/chan_sccp.so |
| 03:14.45 | devdvd | go into the cli |
| 03:14.47 | devdvd | sccp ? |
| 03:14.48 | TehRabbitt | thats what the make install did |
| 03:14.52 | TehRabbitt | sccp "command not found" |
| 03:14.59 | devdvd | in the cli? |
| 03:15.06 | TehRabbitt | yep |
| 03:15.09 | TehRabbitt | "no such command" |
| 03:15.13 | devdvd | i dont know what to tell ya then :\ |
| 03:15.23 | TehRabbitt | OH |
| 03:15.26 | TehRabbitt | chan_sccp |
| 03:15.27 | TehRabbitt | not sccp |
| 03:15.36 | TehRabbitt | thoth*CLI> reload chan_sccp Module 'chan_sccp' does not support reload |
| 03:15.40 | Jouva | Oh yeah speaking of modules, I followed the official instructions to install asterisk on CentOS via yum, and dahdi modules didn't load. |
| 03:15.46 | TehRabbitt | maybe not |
| 03:16.00 | *** join/#asterisk bmg505 (~leon@196-209-71-232.dynamic.isadsl.co.za) |
| 03:16.19 | TehRabbitt | devdvd: and just when I said that debug just spit out: -- SCCP: Accepted connection from 192.168.1.200 -- SCCP: Using ip 192.168.1.70 |
| 03:16.22 | Jouva | http://www.asterisk.org/downloads/yum it seems to indicate that it it should be installed. And I see the files. But they won't load automatically or with insmod |
| 03:16.23 | TehRabbitt | so it looks like it worked somehow |
| 03:19.21 | raden | how can i setup a dialplan that i can call into asterisk from my cell and dial out to mask myself ? |
| 03:25.58 | TehRabbitt | devdvd: so heres the last part of chan_sccp that I cna't seem to figure out... I know how to get the XML set up (already did it)... know how to set up the sccp.conf file (done)... but I don't know how to add it to extensions.conf so that my sccp phone can dial outbound or even call another internal extension |
| 03:27.12 | *** join/#asterisk bjhaid (~abejide@41.220.69.10) |
| 03:27.38 | devdvd | that part is easy |
| 03:28.08 | devdvd | sec |
| 03:28.41 | devdvd | if your context in your sccp.conf is the same as the context for the rest of your phones |
| 03:28.53 | devdvd | then it will have the same rights as the other phones |
| 03:28.57 | devdvd | inc outbound line |
| 03:29.05 | devdvd | so say you have a context called phones |
| 03:29.15 | *** join/#asterisk sorressean (~ty@host-98-127-117-55.gdj-co.client.bresnan.net) |
| 03:29.18 | devdvd | in sccp you will wnat to put that phone into that context |
| 03:29.42 | TehRabbitt | aight |
| 03:29.48 | devdvd | as far as other phones dialing it |
| 03:29.55 | sorressean | I'm curious if someone can help me. I'm looking for a provider that will allow me to receive incoming calls and also call out. I don't know what it's called that I need, so help is appreciated. |
| 03:29.55 | TehRabbitt | then i just do "include=>outbound" etc right? |
| 03:31.39 | devdvd | just do exten=>whatever,1,Dial(SCCP/whatever) |
| 03:34.48 | *** join/#asterisk bjhaid (~abejide@41.206.12.26.vgccl.net) |
| 03:35.33 | sorressean | So I guess what is the terminology I need? I just need inbound and outbound calling. I can configure the asterisk server myself. It'll take a bit, but I'll get it. |
| 03:36.27 | devdvd | ~itsp |
| 03:36.27 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 03:36.33 | *** part/#asterisk bjhaid (~abejide@41.206.12.26.vgccl.net) |
| 03:36.39 | *** join/#asterisk bjhaid (~abejide@41.206.12.26.vgccl.net) |
| 03:36.43 | devdvd | thts what you are lookin for Sorrr |
| 03:37.23 | sorressean | itpslist-us |
| 03:37.29 | sorressean | devdvd: thanks |
| 03:37.47 | sorressean | ~itpslist-us |
| 03:39.32 | sorressean | is a channel what will give me outbound? |
| 03:40.26 | p3nguin | A channel is what gets created every time you dial something from your phone. |
| 03:40.53 | devdvd | yea what the non-mammal aquatic creature said |
| 03:41.25 | devdvd | hiya p3nguin |
| 03:41.58 | TehRabbitt | hey p3nguin |
| 03:42.13 | TehRabbitt | finally got things sorta working lol |
| 03:42.35 | sorressean | sweet. Last thing for a bit. I was looking at Teliax, there's a cheap $24/m plan that is residential. Could I hook asterisk up to that? |
| 03:42.48 | devdvd | are they sip? |
| 03:42.51 | devdvd | or iax2 |
| 03:42.58 | Jouva | p3nguin: Hey, dunno if you saw my messages, but tl;dr: Thanks for the voip.ms tip the other day |
| 03:43.14 | TehRabbitt | sorressean: I use flowroute for my inbound and voip.ms for my outbound calling... they have both been really good for me in the past |
| 03:43.25 | *** join/#asterisk bjhaid (~abejide@41.220.69.17) |
| 03:43.33 | bjhaid | what value does dialstatus returns to ami if a code is written from ami to request the status of a last call? |
| 03:44.12 | Jouva | p3nguin: We didn't put money in the account yet due to weird billing stuff, but I think once that's cleared it'll be great. Their dallas server is hosted by the same people as our server w/ asterisk |
| 03:44.34 | sorressean | TehRabbitt: thanks. and I'm not really sure. I'm new to all this. I just need to figure out a setu pthat will let me get inbound and outbound so I can start playing with this. :p |
| 03:45.52 | TehRabbitt | sorressean: well I chose voip.ms for my outbound because their rates are pretty good for outbound calling... and flowroute have some really good deals on inbound DID #'s |
| 03:46.14 | sorressean | nods. |
| 03:46.29 | sorressean | I assume inbound is called "sip termination?" |
| 03:46.48 | TehRabbitt | so what you do is you set it up so your inbound calls come in via one, and when you dial out, it goes via the other but passes your DID # to your outbound provider so if people have caller ID they can see who is calling lol |
| 03:46.56 | TehRabbitt | yes that is correct |
| 03:47.12 | TehRabbitt | SIP termination is most popular though there are a few IAX2 ones as well |
| 03:47.38 | sorressean | nods. |
| 03:47.50 | sorressean | And asterisk will let me set the caller Id as I call out? |
| 03:48.21 | devdvd | yes |
| 03:48.28 | sorressean | sweet. |
| 03:48.44 | sorressean | smirks. "Sign up today! Free account and 25 cents credit!" horay. I can buy a gumball! |
| 03:48.48 | TehRabbitt | yup haha |
| 03:48.59 | TehRabbitt | 25 cents goes a long way :-p |
| 03:49.07 | cobra2 | TehRabbitt: so what does voip.ms look like on inbound charges? |
| 03:49.08 | TehRabbitt | thats like, 15 minutes or so |
| 03:49.23 | TehRabbitt | cobra2: haven't looked in a while but I remember it being more than flowroute |
| 03:50.15 | TehRabbitt | Monthly Fee: $0.99 to $1.49 |
| 03:50.25 | TehRabbitt | Per minute inbound $0.01 to $0.0149 |
| 03:50.34 | TehRabbitt | which suprisngly isn't bad now that i'm looking at it 0_o |
| 03:50.45 | sorressean | hmm. not bad. I'm kind of looking at a setup that will allow me to skip long distance on my cell, as well as allow for call forwarding so I don't have to give my cell number out, so I'm paying for inbound+outbound. |
| 03:50.54 | cobra2 | that doesn't sound too bad, still would cost me a ton |
| 03:51.13 | TehRabbitt | cobra2: well thats why flowroute is nice, you can buy bundles of DIDs |
| 03:51.16 | p3nguin | Well, that was a good read. It's literally the blind leading the blind for like the past several hours. |
| 03:51.18 | TehRabbitt | like 5 for a discounted price |
| 03:51.23 | TehRabbitt | or 100 0_o |
| 03:51.59 | TehRabbitt | p3nguin: i'm guessin you especially liked my .conf / .so mixup? lmfao |
| 03:51.59 | p3nguin | One of the more interesting tidbits is where we need to recompile Asterisk because we forgot to load chan_sccp. |
| 03:52.13 | TehRabbitt | LOL |
| 03:52.15 | devdvd | LOL |
| 03:52.25 | devdvd | well, i wasn't sure wtf |
| 03:52.40 | devdvd | and when i recompiled it worked |
| 03:53.00 | devdvd | in the end all is good though :) |
| 03:53.20 | p3nguin | Another is where we put Asterisk in a DMZ off a residential grade router, and that magically takes NAT away from the equation. |
| 03:53.23 | *** join/#asterisk Frostix (~Frostix@184-15-205-206.dsl2.chtn.wv.frontiernet.net) |
| 03:53.40 | TehRabbitt | lol |
| 03:53.55 | devdvd | hey, it worked |
| 03:54.08 | p3nguin | No, it didn't. |
| 03:55.04 | *** join/#asterisk TehRabbitt (~TehRabbit@pool-71-172-89-155.nwrknj.fios.verizon.net) |
| 03:55.11 | TehRabbitt | back lol |
| 04:01.13 | p3nguin | sorressean: Inbound to you from the PSTN is called "origination." You'll need a DID for this to work. |
| 04:01.41 | p3nguin | sorressean: Termination is where you make outbound calls to the PSTN through the ITSP. You don't need a DID for this. |
| 04:01.56 | TehRabbitt | 0_o always mix those 2 up |
| 04:02.41 | p3nguin | Like I said, several hours of the blind leading the blind. |
| 04:09.34 | TehRabbitt | still can't seem to get it working... this is what I get when trying to dial out using my SCCP phone |
| 04:09.34 | TehRabbitt | http://pastebin.com/ciSHRkdS |
| 04:11.56 | *** join/#asterisk coppice (~chatzilla@m180-219-213-153.smartone-vodafone.com) |
| 04:12.21 | TehRabbitt | heres the info frome xtensions.conf: http://pastebin.com/bVThFNct |
| 04:14.35 | sorressean | p3nguin: thanks |
| 04:16.23 | TehRabbitt | it works, but i dont hear "ringing" |
| 04:21.58 | TehRabbitt | any idea why my phone doesn't "Ring" when trying to call outbound #'s? |
| 04:22.11 | TehRabbitt | like, i hear a dialtone, but it just stays silent till the person picks up |
| 04:22.34 | shapr | Perhaps the server isn't sending back "RINGING" responses? |
| 04:23.10 | TehRabbitt | how can I check that? |
| 04:23.21 | shapr | packet capture with wireshark? |
| 04:23.25 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-wrkxduipxkblciuh) |
| 04:23.48 | sorressean | hmm. how hard is it to do a press 1 to leave a message, to to call out, if user enters 2, ask for password then number and then allow for them to dial a number to call out? |
| 04:23.56 | TehRabbitt | is there an easier way than using wireshark? |
| 04:26.11 | p3nguin | When I started using sccp recently with VoIP.ms, I lost my ringing sound while calling out. The fix, of course, is to add the r option in the Dial command. |
| 04:26.11 | sorressean | I'm working on trying to figure out how to make it connect at the moment, but that's essentially what I want. just a sort of 1 to leave a message, 2 to call out thing. |
| 04:26.45 | p3nguin | sorressean: That's very basic dial plan logic. It's very simple to do. |
| 04:27.09 | sorressean | p3nguin: sweet. Are there some good resources? right now I'm like beyond lost and confused. |
| 04:27.16 | p3nguin | ~newbook |
| 04:27.16 | infobot | Asterisk: The Definitive Guide. A preview of the upcoming release is available at http://ofps.oreilly.com/titles/9780596517342/. The book is being released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/. Please consider supporting the authors generosity by purchasing a copy at http://oreilly.com/catalog/9780596517342. |
| 04:27.21 | p3nguin | ~book |
| 04:27.21 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 04:27.43 | p3nguin | Those should keep you busy for a while. |
| 04:28.00 | sorressean | p3nguin: thanks |
| 04:29.59 | p3nguin | I just realized that I slightly misworded what I meant: After I started using chan_sccp-b, one day not that long ago I realized I got no ringing sounds when calling out through VoIP.ms. I don't know if the two things are related or not. |
| 04:30.14 | TehRabbitt | p3nguin: like this you mean: exten => _1NXXNXXXXXX,2,Dial(IAX2/voipms/${EXTEN},r) |
| 04:30.21 | p3nguin | yes |
| 04:30.47 | TehRabbitt | still nothing, and I did reload dialplan |
| 04:31.03 | p3nguin | except that won't work because of the wrong syntax. |
| 04:31.17 | TehRabbitt | oh lol |
| 04:31.22 | TehRabbitt | where should the ,r be? |
| 04:31.35 | p3nguin | channeltech/peer/extension,timeout,options |
| 04:31.47 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-zypajquempsvrokm) |
| 04:31.53 | cobra2 | TehRabbitt: thanks for the info a while back man. |
| 04:32.10 | TehRabbitt | np |
| 04:32.17 | TehRabbitt | p3nguin: So like this then: exten => _NXXNXXXXXX,2,Dial(IAX2/voipms/1${EXTEN},120,r) |
| 04:32.22 | p3nguin | If you omit the timeout, you still have to leave an empty spot for it. |
| 04:32.32 | p3nguin | or provide a timeout... either way. |
| 04:32.32 | TehRabbitt | ohhhh |
| 04:33.01 | TehRabbitt | IT RINGS! |
| 04:33.25 | p3nguin | Asterisk is providing the ringing sound. It's not ideal, but it is better than not having any ringing sound at all. |
| 04:34.10 | p3nguin | I guess I need to test VoIP.ms without the r option from a SIP device and see what happens. It very well could be a problem with chan_sccp-b. |
| 04:34.30 | TehRabbitt | hm. could be or maybe voip.ms doesn't provide ringing anymore for IAX? |
| 04:35.12 | *** join/#asterisk bmg505 (~leon@196-209-71-232.dynamic.isadsl.co.za) |
| 04:35.51 | p3nguin | I never had the forethought to test any other scenarios. I use IAX2, and I use chan_sccp-b, and it didn't ring... so I added ringing sound. End. |
| 04:36.31 | p3nguin | I know I had ringing when I started using sccp, but I was using v2 at that time. |
| 04:37.09 | p3nguin | I later upgraded to v3 and I lost the ringing sound at some point. It's possible there's some connection between those two events. |
| 04:41.00 | *** join/#asterisk bmg505 (~leon@196-209-71-232.dynamic.isadsl.co.za) |
| 04:41.40 | *** join/#asterisk bmg505 (~leon@196-209-71-232.dynamic.isadsl.co.za) |
| 04:41.51 | shapr | f -a |
| 04:41.54 | shapr | fail! |
| 04:42.20 | *** join/#asterisk bmg505 (~leon@196-209-71-232.dynamic.isadsl.co.za) |
| 04:42.59 | *** join/#asterisk bmg505 (~leon@196-209-71-232.dynamic.isadsl.co.za) |
| 04:43.38 | *** join/#asterisk bmg505 (~leon@196-209-71-232.dynamic.isadsl.co.za) |
| 04:44.22 | *** join/#asterisk bmg505 (~leon@196-209-71-232.dynamic.isadsl.co.za) |
| 04:45.00 | *** join/#asterisk bmg505 (~leon@196-209-71-232.dynamic.isadsl.co.za) |
| 04:45.54 | *** join/#asterisk bmg505 (~leon@196-209-71-232.dynamic.isadsl.co.za) |
| 04:45.59 | TehRabbitt | whats the code to enable hold for a phone that doesn't have a hold button? |
| 04:46.18 | TehRabbitt | wasnt' tehre some sort of # code or somethign you dial to put the person on hold? |
| 04:46.53 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 04:47.37 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 04:51.07 | *** join/#asterisk bmg505 (~leon@196-209-71-232.dynamic.isadsl.co.za) |
| 04:51.47 | *** join/#asterisk bmg505 (~leon@196-209-71-232.dynamic.isadsl.co.za) |
| 04:52.33 | *** join/#asterisk bmg505 (~leon@196-209-71-232.dynamic.isadsl.co.za) |
| 04:53.13 | *** join/#asterisk bmg505 (~leon@196-209-71-232.dynamic.isadsl.co.za) |
| 04:53.53 | *** join/#asterisk bmg505 (~leon@196-209-71-232.dynamic.isadsl.co.za) |
| 05:09.36 | *** join/#asterisk bmg505 (~leon@196-209-71-232.dynamic.isadsl.co.za) |
| 05:10.16 | *** join/#asterisk bmg505 (~leon@196-209-71-232.dynamic.isadsl.co.za) |
| 05:10.57 | *** join/#asterisk bmg505 (~leon@196-209-71-232.dynamic.isadsl.co.za) |
| 05:13.24 | p3nguin | res_features can probably handle that. |
| 05:16.38 | daxt | p3ngine do you know mydivert ? |
| 05:17.10 | daxt | p3nguin do you know mydivert ? |
| 05:18.07 | *** join/#asterisk bmg505 (~leon@196-209-71-232.dynamic.isadsl.co.za) |
| 05:18.58 | *** join/#asterisk timahvo1 (~rogue@41.215.1.35) |
| 05:19.48 | *** part/#asterisk benngard (~mabe@h30n2-g-ml-a11.ias.bredband.telia.com) |
| 05:23.42 | *** join/#asterisk bmg505 (~leon@196-209-71-232.dynamic.isadsl.co.za) |
| 05:24.23 | *** join/#asterisk timahvo1 (~rogue@41.215.1.35) |
| 05:27.36 | *** join/#asterisk |TEX| (~TEX@119.224.56.245) |
| 05:29.01 | *** join/#asterisk bmg505 (~leon@196-209-71-232.dynamic.isadsl.co.za) |
| 05:29.41 | *** join/#asterisk bmg505 (~leon@196-209-71-232.dynamic.isadsl.co.za) |
| 05:33.04 | *** join/#asterisk timahvo1 (~rogue@41.215.1.35) |
| 05:34.35 | *** join/#asterisk bmg505 (~leon@196-209-71-232.dynamic.isadsl.co.za) |
| 05:35.10 | *** join/#asterisk strehlow (~strehlow@24-213-19-74.static.mrqt.mi.charter.com) |
| 05:35.15 | *** join/#asterisk bmg505 (~leon@196-209-71-232.dynamic.isadsl.co.za) |
| 05:37.57 | *** part/#asterisk TehRabbitt (~TehRabbit@pool-71-172-89-155.nwrknj.fios.verizon.net) |
| 05:40.06 | *** join/#asterisk bmg505 (~leon@196-209-71-232.dynamic.isadsl.co.za) |
| 05:44.31 | *** join/#asterisk bmg505 (~leon@196-209-71-232.dynamic.isadsl.co.za) |
| 05:50.10 | *** join/#asterisk bmg505 (~leon@196-209-71-232.dynamic.isadsl.co.za) |
| 05:50.50 | *** join/#asterisk bmg505 (~leon@196-209-71-232.dynamic.isadsl.co.za) |
| 05:51.28 | *** join/#asterisk bmg505 (~leon@196-209-71-232.dynamic.isadsl.co.za) |
| 05:52.09 | *** join/#asterisk bmg505 (~leon@196-209-71-232.dynamic.isadsl.co.za) |
| 05:52.48 | *** join/#asterisk bmg505 (~leon@196-209-71-232.dynamic.isadsl.co.za) |
| 05:53.28 | *** join/#asterisk bmg505 (~leon@196-209-71-232.dynamic.isadsl.co.za) |
| 05:53.53 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 05:54.08 | *** join/#asterisk bmg505 (~leon@196-209-71-232.dynamic.isadsl.co.za) |
| 05:58.09 | *** join/#asterisk jkroon (~jkroon@dsl-241-247-150.telkomadsl.co.za) |
| 05:59.03 | *** join/#asterisk Subdolus (dexterity@creep.bur.st) |
| 05:59.57 | *** join/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110) |
| 06:01.46 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 06:01.46 | *** mode/#asterisk [+o file] by ChanServ |
| 06:07.35 | p3nguin | daxt: Did I answer you any of the other ~100 times you've asked that same damn question? |
| 06:07.36 | *** join/#asterisk benngard (~mabe@213.88.138.230) |
| 06:11.09 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 06:15.39 | *** join/#asterisk voxter (~voxter@macpro.daytonhome.voxter.net) |
| 06:19.16 | *** join/#asterisk Kumbang (~kumbang@180.245.137.5) |
| 06:22.29 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 06:30.01 | daxt | p3nguin ya i have asked this more than 10 times here and yet to get a convincing answer |
| 06:30.20 | *** join/#asterisk tyman (~tyler@173-12-219-178-Fresno.hfc.comcastbusiness.net) |
| 06:31.04 | drmessano | Apparently nobody has ever heard of them |
| 06:32.59 | p3nguin | Did you ever consider that people don't respond because A) they don't know anything about it, and/or B) they've grown tired of seeing the same question over and over and over and over? |
| 06:33.06 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-bjrmnbbyejglbaty) |
| 06:33.40 | daxt | somebody told me that there was a member of this room who has used it |
| 06:33.50 | daxt | and i am told that he is a regular visitor |
| 06:34.01 | daxt | so i am looking for him :) |
| 06:36.35 | p3nguin | Perhaps he'll answer when he's ready. Or perhaps you can send him a single message about the problem instead of asking the same thing over and over. |
| 06:39.32 | drmessano | Do you know who this alleged member is? |
| 06:43.20 | *** join/#asterisk sourcode (~code@ppp-58-8-118-49.revip2.asianet.co.th) |
| 06:44.23 | daxt | no that is the problem |
| 06:44.41 | daxt | anyways mydivert is a DID service |
| 06:55.26 | p3nguin | Did you ever describe the problem you're experiencing? If you don't ever ask about something, there's nothing anyone can help you solve. |
| 06:58.53 | daxt | i am trying to integrate asterisk with mydivert and followed their guide , but could not integrate |
| 06:58.56 | daxt | that is the problem |
| 06:59.43 | daxt | i added extension in asterisk which dials their test number , but call does not go through |
| 07:00.31 | p3nguin | If this "mydivert" thing is just a service which provides DIDs, how does the provider give the DIDs to you? Are they a SIP service provider? |
| 07:05.10 | daxt | yes |
| 07:07.25 | *** join/#asterisk Tim_Toady (~moi@77.49.3.6.dsl.dyn.forthnet.gr) |
| 07:07.49 | daxt | http://paste.org/pastebin/view/29517 this is what i followed |
| 07:13.59 | p3nguin | If they provider a regular service like every other ITSP out there, it can't be too hard to configure a peer for them in your sip.conf and create an extension to do something useful. I'll take a look at your paste, but I can't promise it will make sense to me. |
| 07:14.51 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-138-161.ks.ks.cox.net) |
| 07:16.37 | p3nguin | I assume this paste is what they gave you as a configuration guide. It actually looks pretty good... a little better than what most ITSPs would give you as a sample configuration. |
| 07:19.25 | p3nguin | When you use terms like "integrate," it makes it seem like software that you must add to your asterisk to increase functionality. That's not the case, so try to use better descriptions for problems. If you would have said, "I can't get calls to work from my provider; I followed the setup guide they gave me, but it's still not working," I would have been a lot more likely to WANT to help you weeks ago when you ... |
| 07:19.31 | p3nguin | ... started spewing the questions about who uses mydivert. |
| 07:20.26 | *** join/#asterisk awclin (~alinford@g0962184.demon.co.uk) |
| 07:23.02 | p3nguin | I'd bet other people would have offered to help weeks ago, too. |
| 07:24.35 | *** join/#asterisk strehlow (~strehlow@24-213-19-74.static.mrqt.mi.charter.com) |
| 07:25.05 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 07:26.27 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 07:26.29 | schmidts | good morning |
| 07:26.32 | wdoekes2 | morning |
| 07:28.57 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 07:30.45 | kaldemar | daxt: you can keep on looking or you can show someone why your call fails. have you done the latter? |
| 07:37.23 | *** join/#asterisk voxter (~voxter@74.114.208.126) |
| 07:41.44 | *** join/#asterisk Sipster_ (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 07:45.12 | *** join/#asterisk awclin (~alinford@g0962184.demon.co.uk) |
| 07:46.50 | *** join/#asterisk sourcode (~code@ppp-58-8-159-11.revip2.asianet.co.th) |
| 07:48.31 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 08:02.53 | *** join/#asterisk megaman5 (~megaman5@c-71-197-149-17.hsd1.wa.comcast.net) |
| 08:03.19 | megaman5 | Does anyone know how I can send a fax using Free Fax for Asterisk and FreePBX? |
| 08:09.30 | wdoekes2 | ~ask |
| 08:09.30 | 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. |
| 08:10.39 | wdoekes2 | I know how to send fax using regular asterisk and the SendFAX app, but I know nothing about FreePBX, nor "Free Fax for Asterisk" |
| 08:14.50 | *** join/#asterisk voxter (~voxter@macpro.daytonhome.voxter.net) |
| 08:15.06 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 08:15.27 | megaman5 | For example, using the sendfax app, does that require command line access and a tiff file or can i setup a email to fax gateway or printer in windows? |
| 08:16.05 | megaman5 | "Free Fax for Asterisk" is the Digium offering for faxes... |
| 08:18.01 | p3nguin | What does Free Fax for Asterisk have to do with FreePBX? |
| 08:19.06 | megaman5 | FreePBX is just the gui that i picked when installing AsteriskNOW, its the default one |
| 08:19.27 | megaman5 | so nothing i guess... |
| 08:22.38 | p3nguin | If you're trying to get help using FreePBX, this isn't the right place. If you're just wanting to know how to use fax for asterisk, I can probably give you a sample dialplan to do it. |
| 08:28.23 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 08:32.19 | *** join/#asterisk joachim_- (~joachim@post.comvie.no) |
| 08:33.36 | joachim_- | My Queues wont show until I call in the queues once.. Anybody know how to fix this? |
| 08:34.20 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:f068:8f8:578e:ea90) |
| 08:44.04 | *** join/#asterisk fenlander (~fenlander@82.152.81.57) |
| 08:45.35 | *** join/#asterisk E-bola (~bola@188.120.76.228) |
| 08:46.27 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 08:47.22 | E-bola | Morning |
| 08:49.13 | *** join/#asterisk Dovid (~Dovid@office.mypbxmanager.net) |
| 08:52.30 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 08:54.53 | p3nguin | joachim_-: What do you mean won't show? |
| 09:00.14 | *** join/#asterisk RamaskeZ (~RamaskeZ@ip-87-86-154-74.easynet.co.uk) |
| 09:01.43 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 09:02.39 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 09:11.38 | E-bola | [Dec 21 16:10:13] WARNING[8235] chan_iax2.c: Max retries exceeded to host 193.105.190.200 on IAX2/318422_peer-12881 (type = 6, subclass = 2, ts=252005, seqno=53) |
| 09:11.38 | E-bola | [Dec 21 16:10:21] |
| 09:11.46 | E-bola | it keeps coming in 1.8 :( |
| 09:19.09 | wdoekes2 | megaman5: the SendFAX app requires a nice tiff.. but you don't need command line access. you do need filesystem access so you can put the tiff there. (command line access is nice to put a system there that does conversion to tiff for you) |
| 09:25.31 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 09:29.03 | joachim_- | p3nguin: I cant log in any agents to my queues.. I have to call the queue first. Then it gets active and i can log in agents |
| 09:32.47 | p3nguin | joachim_-: How often does it happen? Does it happen only after a restart? What causes the behavior to happen again? |
| 09:34.11 | joachim_- | p3nguin: it happends after restarting asterisk.. queue reload doesnt fix it either |
| 09:34.20 | creativx | <3 magic reboots |
| 09:34.44 | joachim_- | I have to pick up my phone and call in to my ivr and queue, then the queue becomes active, and i can log in agents |
| 09:36.11 | p3nguin | joachim_-: All queues or only certain ones? |
| 09:36.24 | joachim_- | All queues |
| 09:37.13 | strehlow | dumb question: where do you configure the smtp server for voicemail emails? |
| 09:37.25 | p3nguin | Pick a queue name for me. Run this on your acli, replacing qname with the name you picked: queue show qname |
| 09:37.36 | p3nguin | Then show me the members. |
| 09:38.03 | p3nguin | strehlow: If you just want the setting for it, it's in voicemail.conf. |
| 09:38.51 | joachim_- | queue1 has 0 calls (max unlimited) in 'leastrecent' strategy (34s holdtime, 108s talktime), W:0, C:1, A:1, SL:0.0% within 60s |
| 09:38.51 | joachim_- | <PROTECTED> |
| 09:38.51 | joachim_- | <PROTECTED> |
| 09:38.51 | joachim_- | <PROTECTED> |
| 09:40.29 | p3nguin | joachim_-: In your modules.conf, add these two settings: preload => pbx_config.so and preload => chan_local.so |
| 09:40.59 | p3nguin | joachim_-: Then restart again and see if it works properly or is still acting the same way. |
| 09:44.01 | joachim_- | p3nguin: Thanks! Will try that after work... It's a production system so cant restart now. :) |
| 09:44.53 | p3nguin | I had a feeling you might be using a local channel as the q member. |
| 09:47.05 | *** join/#asterisk no1peanut (~rudolf@h59ec0d32.stgertrud.dyn.perspektivbredband.net) |
| 09:47.20 | *** join/#asterisk af_ (~getsmart@78.134.22.218) |
| 09:50.01 | no1peanut | I have a weird issue in a ivr menu - I have 2 choices - both leading to a playback in another context - first one plays immediately but the other one has a 4-5 seconds pause before it plays |
| 09:56.05 | *** join/#asterisk killdashnine (~matthias@82.161.138.210) |
| 09:57.01 | fenrus | is there silence in the beginning of the audio file? :) |
| 09:58.26 | *** join/#asterisk iprouteth0 (~iprouteth@unaffiliated/iprouteth0) |
| 09:59.06 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 09:59.43 | p3nguin | core set verbose 4 |
| 10:00.02 | p3nguin | Make a call to the one with the problem. Pastebin everything that happens. |
| 10:05.39 | no1peanut | fenrus, yes |
| 10:06.26 | fenrus | no1peanut, so if you just playback the file you have 4 seconds of silence before the content begins ? |
| 10:06.34 | fenrus | then you might need to edit the file with some audio software |
| 10:07.03 | no1peanut | fenrus, I have no silence when listening to the file from outside asterisk |
| 10:13.40 | no1peanut | p3nguin, http://pastebin.com/0h4abRhL |
| 10:13.43 | *** join/#asterisk Bryanstein (bryan@shellium/admin/bryanstein) |
| 10:14.16 | *** join/#asterisk TimeRider (~steve@188-220-38-155.zone11.bethere.co.uk) |
| 10:17.16 | p3nguin | no1peanut: enter_recording_code.slin is the one giving you the problem? |
| 10:17.27 | no1peanut | yes |
| 10:22.03 | AdvoWork | hi there, after a bit of advice regarding dial rules in trixbox, i need to dial an international number 00352... but this fails, how would I add a specific dial rule to allow this one number? |
| 10:28.55 | fenrus | ~trixbox |
| 10:28.55 | infobot | well, trixbox is SH1TB0X. Basically a CRAPPY, closed-source distro. STAY AWAY! |
| 10:29.06 | fenrus | is there some information about that? :D |
| 10:34.02 | kaldemar | AdvoWork: ask in #trixbox or #freepbx |
| 10:35.22 | E-bola | Can any1 give me a foolproof way of getting a crash/debug file when asterisk hangs? I've failed to follow the guideslines on the website twice now, and the bug is still there :( |
| 10:35.58 | iprouteth0 | E-bola: are you watching asterisk CLI output with higher verbosity? |
| 10:38.27 | iprouteth0 | or maybe... |
| 10:38.36 | iprouteth0 | cat /var/log/asterisk/messages |
| 10:41.44 | E-bola | im using verbose typically between 16-25 and i got full debug logging turned on to the full log |
| 10:41.55 | E-bola | thats not whats needed to debug a bug, the devs want a crashfile |
| 10:44.11 | iprouteth0 | ah |
| 10:44.25 | iprouteth0 | traceback sorta thing? |
| 10:45.53 | E-bola | Yep |
| 10:48.15 | iprouteth0 | if you start asterisk by command in a terminal rather than with initscript, and if it should crash, wouldn't that just output to that term;s stdout? |
| 10:48.58 | E-bola | it doesnt crash, it hangs, thats the issue |
| 10:49.06 | E-bola | if it wuld crash it would produce a coredump i could use |
| 10:50.26 | iprouteth0 | i see. Clearly I've still much to learn... Really just started digging into asterisk more heavily lately in gentoo |
| 10:53.30 | strehlow | Would someone mind helping me with my voicemail.conf to get outgoing email working? I've been at it for a couple hours and have been failing. It doesn't seem like it should be too complicated. |
| 10:54.13 | E-bola | strehlow: its not, just make sure u can send email from the console of ur server |
| 10:54.14 | p3nguin | Mostly you just need to leave the mailcmd setting alone (which is probably commented out). |
| 10:54.18 | E-bola | it has very little to do with asterisk |
| 10:54.38 | Dovid | afternoon all |
| 10:55.22 | strehlow | ok. i have my outgoing smtp server and port set. Humor me here, i'm new to this stuff, but is there something I'm missing? |
| 10:55.26 | iprouteth0 | ah, sometimes email is such a pain..... I work for an ISP and we have many business customers on dynamic IPs who get upset when we tell them we block port 25 |
| 10:56.53 | p3nguin | strehlow: Care to show me how you set the outgoing smtp server and port? |
| 10:57.07 | strehlow | imapserver=smtp.charter.net |
| 10:57.07 | strehlow | imapport=25 |
| 10:57.26 | p3nguin | Neither of those are smtp. |
| 10:57.31 | p3nguin | They clearly say IMAP. |
| 10:57.38 | strehlow | holyshit |
| 10:57.42 | Chainsaw | I was about to say, I've never configured this option before but that looks very wrong. |
| 10:57.54 | p3nguin | That's for IMAP storage of voicemail. |
| 10:57.54 | strehlow | i've been overlooking that |
| 10:58.19 | p3nguin | Like I mentioned a while ago... there's really nothing you need to change in voicemail.conf. |
| 10:58.45 | p3nguin | You just need to configure some type of basic smtp on the host. Asterisk will use it. |
| 10:58.46 | iprouteth0 | that was nice and quick |
| 10:59.10 | strehlow | well, i'm back to square one. Where is the correct location for configuring the outgoing mailserver? And I'll get rid of that in voicemail.conf; i now realize it's not what i intended it to be. |
| 10:59.35 | iprouteth0 | make sure your sendmail works first |
| 11:00.16 | p3nguin | Why do I need to keep repeating this? |
| 11:00.20 | p3nguin | (0458.44) <p3nguin> You just need to configure some type of basic smtp on the host. Asterisk will use it. |
| 11:00.24 | p3nguin | (0454.13) <p3nguin> Mostly you just need to leave the mailcmd setting alone (which is probably commented out). |
| 11:00.42 | strehlow | sorry, i'm tired. thanks for your help, i'll dive back in. |
| 11:00.53 | iprouteth0 | p3nguin: I feel your pain. Tis the nature of support |
| 11:01.59 | p3nguin | Asterisk does not have any smtp configuration. It only has one basic setting for the mail command, and that is mailcmd. You typically don't have to touch it, but if you do, you'll configure the mail command, not a server address, not a host name... just a mail COMMAND. |
| 11:02.02 | iprouteth0 | My work partially consists of supporting POTS and DSL, FTTP/IPTV customers over the telephone |
| 11:03.06 | iprouteth0 | among other groups |
| 11:04.37 | tzafrir_laptop | FTTP? |
| 11:04.48 | AdvoWork | hi there. anyone got any ideas about this? set up a sip account using ekiga, if i ring from ekiga > a normal extension, it works and i can hear both sides. if i ring from a normal account > ekiga sip account I cant hear anything. any ideas please? |
| 11:04.49 | iprouteth0 | Fiber to the Permise |
| 11:04.50 | tzafrir_laptop | Noise file transfer protocol? |
| 11:04.58 | tzafrir_laptop | noisy |
| 11:05.03 | p3nguin | You probably don't want or need much smtp feature, so something basic like ssmtp or msmtp would be great for your PBX mail relay. |
| 11:05.23 | iprouteth0 | premise* |
| 11:05.24 | tzafrir_laptop | p3nguin, that depends. They don't queue |
| 11:05.47 | tzafrir_laptop | It depends what's the impact of losing a mail message |
| 11:06.10 | p3nguin | I would expect the real mail server would be able to handle the queuing. |
| 11:06.45 | iprouteth0 | We're also doing SIP based managed voip and we just started doing SIP trunking i understand... |
| 11:06.49 | strehlow | I just want it to forward voicemails to an email address. It's running on CentOS, so i'm trying to figure it out |
| 11:08.23 | iprouteth0 | maybe something like yum install sendmail, than configure your smtp server appropriately I imagine |
| 11:10.26 | p3nguin | strehlow: That is a duty of the mail server. So far you haven't even indicated you have one of those. |
| 11:10.27 | strehlow | i'm researching. i was working on this before and had to work on some firewall rules for a few hours, then got back to this. my head is pretty far up my rear-end. I don't know what i was thinking messing around in voicemail.conf |
| 11:11.08 | strehlow | i was trying to use gmail's mail server. i thought relaying it through that would work, similar to how i have my notifications sent out via my pfSense firewall |
| 11:11.10 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 11:12.08 | p3nguin | If you configure your MTA to use gmail as the relay, then it will work that way. |
| 11:12.39 | iprouteth0 | We utilize gmail for our customers, and for us it will only use secure smtp(443) with authentication |
| 11:12.50 | iprouteth0 | oops |
| 11:12.53 | iprouteth0 | wrong port |
| 11:13.34 | iprouteth0 | 465 or 587 |
| 11:14.59 | *** join/#asterisk eject_ck (~eject_ck@62.205.134.210) |
| 11:15.35 | eject_ck | Hi all |
| 11:16.02 | eject_ck | how can I create dialplan pattern for numbers with + at the beginning ? |
| 11:16.13 | eject_ck | this not work _[+49]0XXXXXXXXX |
| 11:16.30 | p3nguin | You can't dial a + from a phone's keypad, anyway. |
| 11:17.12 | eject_ck | p3nguin: I'm using android phone with sip client |
| 11:17.24 | p3nguin | 1 2 3 4 5 6 7 8 9 * 0 # |
| 11:17.26 | eject_ck | I'm using my phonebook where I have all numbers in international format |
| 11:17.39 | p3nguin | Reformat them appropriately. |
| 11:18.14 | *** join/#asterisk fIorz (nobody@2001:1a50:503c::1) |
| 11:18.31 | kaldemar | eject_ck: are you aware what [+49] stands for? |
| 11:18.58 | eject_ck | it's recommended international format for calls in roamind |
| 11:19.01 | eject_ck | roaming |
| 11:19.09 | kaldemar | eject_ck: + OR 4 OR 9. not +49. |
| 11:19.09 | eject_ck | + not work there |
| 11:19.12 | no1peanut | eject_ck, maybe this can help ? http://www.freepbx.org/support/documentation/howtos/how-to-strip-or-replace-the-character-at-the-beginning-of-a-called-numb |
| 11:19.20 | eject_ck | ah, sorry |
| 11:19.36 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
| 11:19.45 | eject_ck | [(+49)] |
| 11:19.57 | iprouteth0 | the omnipotent regular expression..... |
| 11:20.29 | kaldemar | eject_ck: that's ( or + or 4 or 9 or ). |
| 11:21.28 | *** join/#asterisk TimeRider (~steve@188-220-38-155.zone11.bethere.co.uk) |
| 11:21.29 | kaldemar | eject_ck: if you put something inside square brackets, it represents a single character listed inside the brackets. |
| 11:21.59 | kaldemar | eject_ck: just use exten => _+490XXXXXXXXX |
| 11:22.15 | iprouteth0 | what does the _ do? |
| 11:22.34 | eject_ck | iprouteth0: Dialplan patterns always begin with the underscore (_) character |
| 11:22.47 | eject_ck | A common error is to forget the underscore ("_") character at the beginning of the pattern |
| 11:22.51 | eject_ck | :) |
| 11:23.04 | p3nguin | The underscore indicates it's a pattern instead of a literal match. |
| 11:23.13 | p3nguin | s/literal/explicit/ |
| 11:23.23 | iprouteth0 | I see. I have only been experimenting with explicit matches so far |
| 11:23.25 | kaldemar | iprouteth0: a pattern means you can use wildcards like X, N, X, Z, . or !. |
| 11:24.03 | kaldemar | X is nice and useful, hence i typed it twice. i think. |
| 11:24.39 | iprouteth0 | I've some famility with dial-plans, but more in the way cisco does them, specifically dial peers like in call manager express |
| 11:24.58 | eject_ck | kaldemar: thanks! :) |
| 11:25.13 | eject_ck | created 3 extens instead of 1 |
| 11:25.36 | no1peanut | eject_ck, Did you look at the link ? looks like it is what you where asking |
| 11:25.38 | eject_ck | 3 working instead 1 not usable I mean :))) |
| 11:25.57 | eject_ck | no1peanut: it works with + as well |
| 11:26.14 | eject_ck | problem was with wrong pattern |
| 11:27.45 | eject_ck | I have another question to you guys :) I'm using GSM VoIP gateway and then I'm forwarding incoming calls to SIP accounts |
| 11:28.17 | eject_ck | I need to replace caller id for GSM->GSM VoIP GATEWAY->SIP CLIENT |
| 11:28.29 | eject_ck | is it possible ? |
| 11:28.32 | kaldemar | yes |
| 11:28.42 | eject_ck | where I can read about that ? |
| 11:29.09 | eject_ck | Now when someone calls to SIP client via gateway I see gateway's CALLER ID |
| 11:29.34 | eject_ck | I know about Set(CALLERID(name) |
| 11:29.56 | eject_ck | Set(CALLERID(name)=$GSM_CALLER) |
| 11:30.04 | eject_ck | how can I get $GSM_CALLER ? |
| 11:30.21 | joachim_- | I think I have a NAT issue.. What is rport? |
| 11:30.22 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 11:30.49 | kaldemar | eject_ck: if it doesn't come through the GSM gateway, you can't. see in SIP debug if you get the real callerid somehow. |
| 11:30.58 | joachim_- | My nat setting on my asterisk server is nat=route |
| 11:32.03 | joachim_- | And it's not behind nat, the asterisk is listening on a static ip. |
| 11:36.16 | eject_ck | kaldemar: it comes I belive |
| 11:36.46 | eject_ck | how that SIP header is called ? CallerID ? |
| 11:38.20 | eject_ck | On GSM VoIP gateway (portech 370, btw) I have setting Mobild -> Settings -> CLID Presentation = On |
| 11:38.25 | eject_ck | I think that's it |
| 11:39.45 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 11:40.25 | kaldemar | eject_ck: there is no separate header for caller id information. see what your asterisk gets. |
| 11:41.07 | eject_ck | ok |
| 11:41.19 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-cnldrfrzxoemvveq) |
| 11:41.23 | eject_ck | so it's From: |
| 11:41.27 | eject_ck | ? |
| 11:42.20 | kaldemar | maybe. see the trace. |
| 11:43.23 | joachim_- | Should i use nat=no i sip.conf [general] when my asterisk is not behind nat, its listening on a public static ip.. Or should i use nat=yes.. All of my sip clients is behind nat |
| 11:43.38 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 11:44.11 | joachim_- | But when some of our cisco phones call each other, (local call) The sound is gone, and i get "Prodding channel SIP/10111191" Failed |
| 11:44.42 | Chainsaw | joachim_-: directmedia=no; do not trust a Cisco to get SIP right. |
| 11:44.47 | kaldemar | joachim_-: nat=no in general and nat=yes for clients. |
| 11:45.28 | joachim_- | kaldemar: Thats what i thought too.. [general] in sip.conf currently says nat=route |
| 11:45.55 | joachim_- | kaldemar: and now some of out phones are having trouble calling each other, but outband calls in or out works just fine |
| 11:55.46 | *** join/#asterisk saxa (~brastrak@189.105.43.78) |
| 11:56.25 | saxa | hi ppl, can anybody help me understand how to transfer a call from one phone to another ? |
| 11:56.44 | saxa | I have 3 sip phones, so I call from 1001 to 1002 |
| 11:56.55 | saxa | then I try to transfer the call to a 1003 |
| 11:57.09 | benngard | on 1002 when u have answered press # 1 0 0 3 |
| 11:57.15 | saxa | I try just to dial 1003 on 1002 but nothing happens |
| 11:57.28 | saxa | oh I need to press # ? |
| 11:57.32 | benngard | u need tT in deil command for 1002 |
| 11:57.39 | benngard | sec |
| 11:57.55 | saxa | I have added tT to all sip accounts |
| 11:58.04 | saxa | thats in extensions.conf |
| 11:58.14 | benngard | core show application Dial |
| 11:59.28 | benngard | exten => 0317998975-NOT_INUSE,1,Dial(SIP/0317998975&OOH323/00733025975@Avaya,20,rtT) |
| 11:59.39 | benngard | thats my extension |
| 11:59.50 | *** join/#asterisk jkroon (~jkroon@dsl-241-238-153.telkomadsl.co.za) |
| 12:00.10 | jkroon | segfault at 4 ip 00007fbdfe551d60 sp 00007fbde107c268 error 4 in pbx_dundi.so[7fbdfe54d000+1e000] <-- any ideas on figuring out WHAT exactly goes pop ? (* 1.6.2.16.1) |
| 12:00.15 | benngard | if u dial 0317998975 i answer in my fixed phoen, then i just press # and the extension i want to trasfer to |
| 12:02.10 | *** join/#asterisk chasing`Sol (~chasingSo@82.201.135.150) |
| 12:03.04 | saxa | benngard: ok, I understand |
| 12:03.22 | saxa | benngard: will just try to figure one thing out, brb |
| 12:03.52 | saxa | it seems that 1003 doesnt ring at all, and make asterisk freeze |
| 12:03.59 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
| 12:04.22 | saxa | i need to figure what is wrong, since the phone and the sip.conf of 1002 and 1003 are the same |
| 12:04.48 | benngard | from 1002, dial 1003, does it work? |
| 12:05.33 | benngard | when u answer a call in 1002 and press # do u hear a voice saying "transfer"? |
| 12:07.26 | benngard | check features.conf, ;blindxfer => #1 ; Blind transfer (default is #) -- Make sure to set the T and/or t option in the Dial() or Queue() app call! |
| 12:09.19 | saxa | Send DTMF is better to be via RTP rfc2833 or in audio or via SIP INFO ? |
| 12:10.03 | saxa | benngard: I was trying to call from 1002 to 1003 and nothing was happening, now I rebooted the 1003 and I can call from 1003 to 1002, let me try the other stuff. |
| 12:10.20 | coppice | DTMF via SIP INFO is non-standard, so RFC2833 is preferred for that if nothing else |
| 12:11.15 | saxa | coppice: i have also "in audio" option |
| 12:12.04 | coppice | sending it as audio only really works with G.711 |
| 12:12.42 | saxa | ok so I will check the RTP option which is rfc2833 |
| 12:13.11 | saxa | I suppose that all phones must have the same setting ? |
| 12:15.30 | saxa | thats what I get when transferring to 1003, http://pastebin.com/iewFX1d1 |
| 12:15.42 | saxa | it puts the calling party on moh |
| 12:15.51 | saxa | but then it doesnt ring at 1003 |
| 12:16.30 | saxa | seems it freezes asterisk, since if I cann from outside to dahdi, it even doesnt answer or ring on any phone. |
| 12:18.22 | saxa | 09:46 < benngard> when u answer a call in 1002 and press # do u hear a voice saying "transfer"? |
| 12:18.25 | saxa | no |
| 12:18.42 | saxa | i do not hear that voice |
| 12:19.10 | saxa | but i have a transfer button on that phones. |
| 12:27.42 | *** join/#asterisk Maliuta (~nobusines@kiev.lusan.id.au) |
| 12:32.36 | saxa | hmm, I do not have features.conf in my /etc/asterisk dir |
| 12:36.23 | *** part/#asterisk benngard (~mabe@213.88.138.230) |
| 12:42.23 | saxa | ok, I added features.conf |
| 12:42.30 | *** join/#asterisk Dovid (Dovid@office.mypbxmanager.net) |
| 12:42.54 | saxa | which are the minimum .conf files asterisk should have to function ? |
| 12:43.34 | Dovid | depends on what functions you need |
| 12:43.41 | Dovid | when installing asterisk clean to make samples |
| 12:43.49 | Dovid | and you will get sample files |
| 12:44.36 | saxa | Dovid: yes, since I'm having a script doing that I would like to doublechek which ones need to be in there |
| 12:45.23 | saxa | but features.conf and logger.conf should be in there |
| 12:45.35 | saxa | which I had not copied to that dir |
| 12:45.48 | saxa | so I have already added them now to my script |
| 12:48.09 | Dovid | i would put them all in and clean them up |
| 12:48.14 | Dovid | so you have the basics. |
| 12:48.25 | Dovid | does asterisk 1.8.X still need a time for MeetMe ? |
| 12:48.37 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:48.37 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:50.51 | saxa | thx Dovid |
| 12:51.27 | saxa | Dovid: i use 1.8.x but have not even tried meetme |
| 13:06.42 | *** join/#asterisk coppice (~chatzilla@60.157.17.210.dyn.pacific.net.hk) |
| 13:13.44 | *** join/#asterisk saxa (~sasa@189.105.43.78) |
| 13:15.44 | *** join/#asterisk hugorebelo (~hugo@c951b485.virtua.com.br) |
| 13:16.53 | leifmadsen | ~telcomjoshleifvoxmartenterprisesinc |
| 13:17.06 | *** join/#asterisk benngard (~mabe@h30n2-g-ml-a11.ias.bredband.telia.com) |
| 13:17.35 | leifmadsen | infobot: telcomjoshleifvoxmartenterprisesinc is a subsidiary of Qwell Communications Inc |
| 13:17.35 | infobot | leifmadsen: that's too long |
| 13:17.41 | leifmadsen | no it's not! |
| 13:17.52 | leifmadsen | infobot: telcomjoshleifvoxmartenterprises is a subsidiary of Qwell Communications Inc |
| 13:17.53 | infobot | leifmadsen: okay |
| 13:20.00 | saxa | lol |
| 13:20.20 | leifmadsen | infobot: qwellcommunicationsinc is the evil empire portrayed in the television show Better Off Ted (aka Veridian Dynamics) |
| 13:20.20 | infobot | leifmadsen: okay |
| 13:20.46 | leifmadsen | ~qwell |
| 13:20.47 | infobot | extra, extra, read all about it, qwell is a patented liquid formula that contains three plant-based bio-active agents that work together in a perfectly balanced combination. These agents act synergistically to boost your good cholesterol and slash the bad. |
| 13:21.25 | leifmadsen | infobot: qwell is also qwellcommunicationsinc, the holding company of telcomjoshleifvoxmartinc |
| 13:21.25 | infobot | okay, leifmadsen |
| 13:21.36 | coppice | does slash the bad play with the evil guns and roses |
| 13:25.48 | *** join/#asterisk sourcode (~code@ppp-58-8-159-11.revip2.asianet.co.th) |
| 13:32.23 | *** join/#asterisk felimwhiteley (~quassel@109.255.104.145) |
| 13:33.58 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 13:37.05 | joachim_- | "Real time queues will not show ANYTHING until a caller calls the queue." <--- Is this true? |
| 13:37.09 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 13:37.21 | joachim_- | Having same problem with my realtime queue setup |
| 13:41.25 | leifmadsen | joachim_-: yes because the database isn't accessed until data is requested |
| 13:42.46 | joachim_- | leifmadsen: hmmmmmmmm.. Then my agents cant login to their queues until someone calls in first. Thats kinda stupid |
| 13:45.11 | *** part/#asterisk dhorner_mb (~dhorner_m@pool-96-238-239-251.aubnin.fios.verizon.net) |
| 13:45.26 | leifmadsen | that's not what I said at all |
| 13:46.31 | leifmadsen | queue_members is a separate entry in extconfig.conf |
| 13:46.56 | leifmadsen | you were talking about callers into queues, not queue members |
| 13:48.01 | *** join/#asterisk mechbangirc (~mechbangi@mbl-65-157-159.dsl.net.pk) |
| 13:48.26 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 13:49.36 | mechbangirc | can i take user input during record() just like i can take control input in controlplayback(). thing is i want to give user an option to pause/resume recording. |
| 13:49.53 | E-bola | I'm testing collecting a backtrace from my locking asterisk, but when i run the command: gdb -ex "thread apply all bt" --batch /usr/sbin/asterisk 26611 > /tmp/backtrace-threads.txt |
| 13:49.59 | leifmadsen | mechbangirc: no, Record() just listens for audio |
| 13:50.03 | E-bola | I get the following: Cannot access memory at address 0x1 |
| 13:50.09 | leifmadsen | (and the # key to end recording) |
| 13:51.10 | mechbangirc | leifmadsen: so i can make use of # to give that functionality like create multiple recordings and at the end combine them right? or is there a better work around? |
| 13:51.47 | leifmadsen | E-bola: try it this way: gdb -ex "thread apply all bt" --batch /usr/sbin/asterisk `pidof asterisk` > /tmp/backtrace-threads.txt |
| 13:52.10 | leifmadsen | mechbangirc: I don't know what you mean -- when you press the # key, Record() stops and saves the file |
| 13:52.22 | leifmadsen | mechbangirc: beyond that you can program your dialplan to do whatever you want |
| 13:52.46 | E-bola | leifmadsen: Same error. Also core show locks doesnt exist, is it because i didnt enable some flag when compiling? |
| 13:52.50 | *** join/#asterisk cnu (cnu@the.ultimate.lamer.la) |
| 13:52.54 | leifmadsen | possibly |
| 13:53.02 | leifmadsen | did you enable DONT_OPTIMIZE? |
| 13:53.09 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 13:53.10 | leifmadsen | I've never seen that error before |
| 13:53.42 | E-bola | Its possible I dont, i have up on trying to get a backtrace |
| 13:54.03 | E-bola | brrr. "Its possible i didnt, i gave up on trying to get a backtrace |
| 13:54.39 | leifmadsen | mechbangirc: I suggest doing a 'core show application Record' and look at the available options and see how they can be applied to the problem you are trying to solve |
| 13:54.42 | E-bola | You wouldnt happen to be in the process of tagging 1.8.3 final? Then i could throw that on tonight with the proper flags |
| 13:54.51 | leifmadsen | it's already tagged |
| 13:55.08 | leifmadsen | the tarball just isn't signed yet. Should be out later today. |
| 13:55.23 | E-bola | Is it identical to rc3? |
| 13:55.38 | leifmadsen | as far as I can remember |
| 13:55.55 | leifmadsen | possible I merged in the fax security release |
| 13:55.59 | leifmadsen | but that'd be the only change |
| 13:56.09 | leifmadsen | we don't make changes for the full tag from the last RC |
| 13:56.16 | E-bola | Ok thanks. Ile just recompile my rc3 then |
| 13:56.23 | leifmadsen | (except in cases we need to merge a security change or whatever) |
| 13:56.33 | *** join/#asterisk asterisk-Tester (~chatzilla@77.42.241.114) |
| 13:56.41 | leifmadsen | releases are always made from the last RC |
| 13:56.53 | E-bola | Ahh thats clever, good to know |
| 13:57.13 | *** join/#asterisk mechbangirc (~mechbangi@mbl-65-157-26.dsl.net.pk) |
| 13:58.00 | leifmadsen | well it's pretty obvious why that would be the case |
| 13:58.13 | mechbangirc | leifmadsen: where would # put me to next priority or explicitly handup? |
| 13:58.25 | leifmadsen | why would we do RCs and then make changes that weren't tested for the release? |
| 13:58.35 | leifmadsen | mechbangirc: I suggest you try it |
| 13:59.53 | mechbangirc | sure. is there any other app that you know of for such functionality besides Record()? |
| 14:00.12 | *** join/#asterisk espiceland (~erin@nat/digium/x-cvtvtzlarhstbuok) |
| 14:00.28 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 14:05.14 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 14:07.38 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 14:09.06 | *** join/#asterisk hrhrhr (~c1@unaffiliated/hrhrhr) |
| 14:14.28 | *** join/#asterisk waKKu (~blah@unaffiliated/wakku) |
| 14:14.32 | waKKu | hi folks.. morning all |
| 14:17.31 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 14:19.26 | benngard | any knows of an iphon app that sends the status of the phone to asterisk, for example when the iphone receives or plcae a call that information is sent to asterisk |
| 14:20.18 | *** join/#asterisk dhorner_mb (~dhorner_m@pool-96-238-239-251.aubnin.fios.verizon.net) |
| 14:22.52 | *** join/#asterisk Mhaddog__ (~Mhaddog@adsl-072-149-063-056.sip.bct.bellsouth.net) |
| 14:23.42 | *** part/#asterisk waKKu (~blah@unaffiliated/wakku) |
| 14:24.27 | *** join/#asterisk killdashnine (~matthias@82.161.138.210) |
| 14:25.24 | *** join/#asterisk Katty (~Katty@mail.copi-rite.com) |
| 14:25.41 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:26.54 | *** join/#asterisk wonderworld (~ww@port-92-201-55-88.dynamic.qsc.de) |
| 14:28.06 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 14:43.32 | *** part/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 14:45.14 | *** join/#asterisk Tim_Toady (~moi@77.49.3.6.dsl.dyn.forthnet.gr) |
| 14:47.40 | *** join/#asterisk kriegerod (~krieger@chaf.tenet.odessa.ua) |
| 14:49.13 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 14:50.02 | *** join/#asterisk RamaskeZ (~RamaskeZ@ip-87-86-154-74.easynet.co.uk) |
| 14:51.21 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:51.21 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:52.01 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 14:54.34 | *** join/#asterisk BlackBishop (dexter@ipv6.d3xt3r01.tk) |
| 14:56.52 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 14:59.08 | *** join/#asterisk chasing`Sol (~chasingSo@82.201.135.150) |
| 15:02.39 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 15:05.34 | *** join/#asterisk drmessano-lt (~nonya@pdpc/supporter/active/drmessano) |
| 15:05.43 | *** join/#asterisk cyphorious (~cyphoriou@212-183-78-196.adsl.highway.telekom.at) |
| 15:07.12 | BlackBishop | any way I can make the odbc functions run multiple queries ? |
| 15:08.32 | wdoekes2 | stored procedure? function? union? |
| 15:09.12 | wdoekes2 | multiple calls to odbc from your dialplan? |
| 15:09.52 | BlackBishop | well.. I was thinking more of something like "query1;query2;" |
| 15:10.04 | wdoekes2 | why would you want to do that? |
| 15:10.25 | BlackBishop | it seems simpler than creating a function/stored procedure .. or calling multiple functions.. |
| 15:10.54 | *** join/#asterisk SaiSoma (~chatzilla@client105.jdcc.edu) |
| 15:11.42 | wdoekes2 | it seems to me like you're trying to do things you shouldn't |
| 15:12.09 | wdoekes2 | if you cannot union a select but needs multiple queries, how would the returned data be any sensible |
| 15:12.42 | wdoekes2 | and if you're doing multiple inserts/updates, you cannot get a proper result, probably |
| 15:15.17 | BlackBishop | well, I don't care about the returned data |
| 15:15.20 | BlackBishop | I just want 2 updates |
| 15:15.59 | wdoekes2 | you might be able to join those updates ;) |
| 15:16.19 | BlackBishop | googles about sql join-ing updates ... |
| 15:17.10 | E-bola | why do u want to do them in 1 line? |
| 15:17.14 | E-bola | cant u just have 2 lines? |
| 15:17.39 | BlackBishop | well, I was wondering if I can have to lines ! |
| 15:18.02 | BlackBishop | like .. 2 lines of writesql=... |
| 15:18.16 | BlackBishop | in the same odbc function |
| 15:18.21 | E-bola | just call the function twice? |
| 15:18.33 | wdoekes2 | mysql> update customer_customer c inner join reseller_reseller r on r.id = c.reseller_id set c.modified = now(), r.modified = now() where c.id = 1; |
| 15:18.37 | wdoekes2 | Query OK, 2 rows affected (0.03 sec) |
| 15:18.50 | BlackBishop | wouldn't it execute the that sounds like a pain in the ass |
| 15:18.51 | BlackBishop | :| |
| 15:18.55 | BlackBishop | the join thing |
| 15:19.02 | BlackBishop | I think I'll just create 2 functions |
| 15:19.09 | BlackBishop | and then call each one |
| 15:19.20 | wdoekes2 | smart move |
| 15:19.22 | BlackBishop | E-bola: calling it twice would only execute them twice ... |
| 15:19.34 | BlackBishop | or the same one twice .. |
| 15:19.52 | E-bola | BlackBishop: umm huh? Call function with the first insert, then call the function again with the second insert |
| 15:20.24 | BlackBishop | how can I call it like that ? |
| 15:22.13 | E-bola | I think we're missunderstanding each other |
| 15:22.44 | E-bola | <BlackBishop> I think I'll just create 2 functions |
| 15:22.44 | E-bola | <BlackBishop> and then call each one |
| 15:22.47 | E-bola | thats what im saying |
| 15:23.17 | BlackBishop | ow |
| 15:23.27 | BlackBishop | :) |
| 15:23.28 | BlackBishop | ok |
| 15:23.51 | E-bola | its either that or fix ur sql to do 2 things like wdoekes2 showed |
| 15:25.20 | BlackBishop | yeah .. would of been nice to be able to separate them by ";" ... |
| 15:26.21 | E-bola | why? |
| 15:26.27 | E-bola | its not a huge difference :) |
| 15:26.30 | leifmadsen | sounds like you want to use a stored procedure :) |
| 15:27.03 | E-bola | I agree a stored procedure would be overkill for this, then i'm more inclined two join the inserts or whatever they are |
| 15:27.05 | leifmadsen | or as wdoekes2 said, you could create a join'd statement |
| 15:27.15 | tzanger | I'll store your procedure! |
| 15:27.18 | BlackBishop | not really .. it was just .. alot simpler to just query1;query2; |
| 15:27.37 | BlackBishop | E-bola: they're 2 updates |
| 15:31.33 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 15:37.24 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 15:39.51 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 15:39.56 | krion | hi |
| 15:40.21 | krion | how can i check from asterisk cli that it correctly read asterisk.extensions mysql table |
| 15:41.07 | BlackBishop | dialplan show ? |
| 15:41.12 | BlackBishop | I think |
| 15:41.31 | krion | because my connection seems ok but asterisk doesn't load context from extensions table, core show config mappings is ok and one peer in mysql are correctly loaded |
| 15:41.51 | krion | BlackBishop: indeed i've done this, and it doesnt show my contexte |
| 15:41.53 | *** join/#asterisk moy (~moy@CPE003048b11058-CM00222d6b4d65.cpe.net.cable.rogers.com) |
| 15:42.15 | krion | that's why i'm asking that in order to be sure this is not a mysql connection trouble |
| 15:42.34 | leifmadsen | check your mysql logs |
| 15:42.52 | leifmadsen | enable query logging on mysql and monitor it while you're loading asterisk to see what is happeneing |
| 15:52.18 | krion | hum, i tcpdump the port 3306 since i just have phpmyadmin access to the db |
| 15:52.36 | krion | and looks like it query only asterisk.sip_peers, no asterisk.extensions |
| 15:52.42 | krion | thanks for the hint |
| 15:54.42 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 15:58.12 | krion | *CLI> realtime mysql status |
| 15:58.25 | krion | shows ok, so my trouble is somewhere else i guess |
| 16:00.19 | krion | even *CLI> realtime load extensions context <mycontext> shows correct infos... |
| 16:00.44 | krion | so why is dialplan show <mycontext> not working... |
| 16:01.21 | *** join/#asterisk cyphorious (~cyphoriou@212-183-71-225.adsl.highway.telekom.at) |
| 16:01.21 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 16:04.27 | *** join/#asterisk odin917 (~Gavin_Sil@24.136.120.162) |
| 16:04.39 | odin917 | hey all |
| 16:05.16 | odin917 | im having an issue where my asterisk server is remote to our location, all our sip devices can call out fine, but asterisk reports them as unreachable after a period of time, preventing extension to extension dialing |
| 16:07.13 | *** join/#asterisk geek_cl (~lletelier@200.73.2.58) |
| 16:15.30 | *** join/#asterisk atan2 (~atan@unaffiliated/atan) |
| 16:21.58 | *** join/#asterisk zerohalo (~zerohalo@173-13-92-27-NewEngland.hfc.comcastbusiness.net) |
| 16:22.18 | *** join/#asterisk GreatSUN (~greatsun@178-190-211-140.adsl.highway.telekom.at) |
| 16:22.20 | GreatSUN | hi all |
| 16:22.54 | ChannelZ | odin917: are they set to qualify? firewall issues? |
| 16:23.08 | GreatSUN | can someone possibly help me with ISDN Phone connected to asterisk via dahdi ? |
| 16:25.20 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 16:26.39 | GreatSUN | I got my configs pasted to nopaste.info |
| 16:26.45 | GreatSUN | http://nopaste.info/04b007dbed.html |
| 16:27.21 | GreatSUN | hopefully someone of you can help me to get the phone working |
| 16:27.39 | GreatSUN | I was working on it for more than a day now :o( |
| 16:27.59 | GreatSUN | guess I am just missing some configuration (or I need to upgrade asterisk) |
| 16:28.35 | WIMPy | GreatSUN: And what's the issue with it? |
| 16:29.13 | GreatSUN | I can not call it and it neither gets a dial tone |
| 16:29.44 | GreatSUN | so neither incoming call nor outgoing is working |
| 16:30.06 | WIMPy | Do you have the correct crossover cable or jumper setting? What about Power? |
| 16:30.10 | krion | FYI my problem was that an agi is called in order to create context and hints, this agi failed to execute properly |
| 16:30.40 | GreatSUN | WIMPy: yeah, all correct |
| 16:30.51 | GreatSUN | The Phone has its own power |
| 16:30.56 | E-bola | leifmadsen: Ok got the backtrace command working, now ile wait for the next crash |
| 16:31.09 | GreatSUN | WIMPy: Port is running under NT not TE |
| 16:31.21 | E-bola | I wonder if core show locks will work, since normally the console isnt responsive |
| 16:31.52 | WIMPy | GreatSUN: That makes it easier. Do you have reasonably current versions of dahdi, libpri and asterisk? |
| 16:32.23 | GreatSUN | dahdi: 2.4.0 |
| 16:32.58 | odin917 | ChannelZ: they are set to qualify |
| 16:33.12 | GreatSUN | asterisk 1.6.2.13 |
| 16:33.31 | GreatSUN | libpri: I dunno. Can you tell me how to find out version there ? |
| 16:33.42 | odin917 | i think i "resolved" the issue by setting the phones to reregister every 2 minutes x.x |
| 16:33.42 | WIMPy | Err, wait. Your ports 3 and 4 are commented out. |
| 16:34.34 | GreatSUN | WIMPy: yeah, I found out that elsewise if I configure the extension dahdi isn't working anymore in asterisk |
| 16:35.08 | WIMPy | It the ports aren't configured, they surely won't work. |
| 16:35.59 | WIMPy | But IIRC Asterisk 1.6 can't handle NT-ptmp mode. |
| 16:36.09 | GreatSUN | WIMPy: they get configured through chan_dahdi.conf and chan_dahdi_additional.conf |
| 16:36.26 | GreatSUN | WIMPy: hmmm... that might be the main problem |
| 16:36.41 | GreatSUN | WIMPy: I got elastix running here in 64bit mode |
| 16:37.01 | GreatSUN | do you know if I can upgrade asterisk without big problems? |
| 16:37.18 | GreatSUN | or is there a howto or such? |
| 16:37.20 | WIMPy | No, from wha I have noted, you need Asterisk 1.8 and libpri 1.4.11, better 1.4.12 to use NT-ptmp with dahdi. |
| 16:37.35 | WIMPy | No idea. |
| 16:37.41 | GreatSUN | bad luck though |
| 16:37.42 | *** join/#asterisk garymc (~chatzilla@host81-148-15-59.in-addr.btopenworld.com) |
| 16:38.08 | WIMPy | But NT-ptmp works with older Asterisks if you use misdn or LCR. |
| 16:38.09 | GreatSUN | in this case I could have been trying a whole life :-D |
| 16:38.43 | GreatSUN | WIMPy: hmmm... I think I will try to switch to newer versions if I got the time |
| 16:38.56 | GreatSUN | elsewise, it's just one phone that's missing... |
| 16:39.08 | WIMPy | Ja, I'd recommend 1.8, no matter what channel driver. |
| 16:39.41 | GreatSUN | WIMPy: yesterday I installed pbx from a flash or likewise name |
| 16:40.07 | GreatSUN | then I wanted google voice and fax and was installing incredible pbx after everything worked |
| 16:40.15 | GreatSUN | that destroyed all my configurations |
| 16:40.25 | GreatSUN | and then I switched back to elastix |
| 16:40.30 | WIMPy | Welcome to the world of GUIs. |
| 16:40.46 | GreatSUN | WIMPy: it even was a shellscript :-D |
| 16:41.06 | GreatSUN | but true, I like freepbx |
| 16:41.21 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 16:41.24 | GreatSUN | but you have to find workarounds to get everything working like it should to use it |
| 16:43.09 | drmessano-lt | GreatSUN, I would suggest AsteriskNOW over Elastix. At least you can run 1.8 and wont have all the bloat |
| 16:47.12 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 16:47.19 | GreatSUN | drmessano-lt: maybe I will try that laters |
| 16:47.34 | GreatSUN | but for now I have to work with what I got |
| 16:47.39 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 16:48.01 | GreatSUN | and before creating a backup of the configurations I will anyways never reinstall |
| 16:48.55 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 16:50.06 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:59.56 | *** join/#asterisk GabrielPiassetta (~gabriel@200.175.61.250) |
| 17:00.21 | GreatSUN | just created backup of asterisk files |
| 17:00.31 | GreatSUN | now I will try to install 1.8 |
| 17:00.39 | GreatSUN | http://www.freepbx.org/forum/freepbx/installation/asterisk-1-8-installation |
| 17:00.50 | WIMPy | What card do you have, BTW? |
| 17:01.04 | GreatSUN | junghanns |
| 17:01.13 | GreatSUN | quadbri |
| 17:01.13 | *** join/#asterisk Blackthorn (Blackthorn@76.77.160.10) |
| 17:01.38 | Blackthorn | just did a clean install of asterisk 1.8 and everything is up and running but I get a cli error of WARNING[2323]: db.c:112 dbinit: Unable to open Asterisk database '/var/lib/asterisk/astdb': Permission denied |
| 17:01.42 | WIMPy | You need the te_nt_override patch for dahdi then. |
| 17:01.52 | *** join/#asterisk LemensTS (~matthew@adsl-70-238-156-107.dsl.stlsmo.sbcglobal.net) |
| 17:02.17 | *** join/#asterisk ClintGoudie-Nice (~clint@smtp.callware.com) |
| 17:02.50 | WIMPy | Blackthorn: That't pretty obvious, I'd say. Asterisk needs to be able to write there. |
| 17:03.01 | GreatSUN | or access |
| 17:03.11 | ClintGoudie-Nice | Heya all. How do I enable srtp for a sip trunk? I've got libSRTP installed on the box, and I set srtpcapable=yes on the sip trunk. |
| 17:03.20 | GreatSUN | ls -al /var/lib/asterisk/astdb will show you root.root I bet |
| 17:03.45 | WIMPy | ClintGoudie-Nice: Are you sure you want it? |
| 17:03.46 | GreatSUN | WIMPy: nope, I think I will try 1.8 |
| 17:04.03 | ClintGoudie-Nice | W |
| 17:04.10 | Blackthorn | it does shwo root |
| 17:04.16 | Blackthorn | it does show root |
| 17:04.16 | ClintGoudie-Nice | WIMPy: sadly yes |
| 17:04.18 | GreatSUN | WIMPy: what version would you use? 1.8.3 or .4-RC2? |
| 17:04.23 | WIMPy | GreatSUN: You need the patch to be able to switch to NT mode at all. |
| 17:04.42 | WIMPy | GreatSUN: Should be ok. |
| 17:04.54 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 17:05.05 | GreatSUN | WIMPy: yesterday I had 1.8 running and there was no patch needed |
| 17:05.22 | Blackthorn | i do not know how to set the permission for asterisk to gain access to the file. |
| 17:05.25 | GreatSUN | I bet I just need to recompile the dahdi drivers after upgrade |
| 17:05.26 | WIMPy | You could use the phone? |
| 17:05.27 | *** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 17:05.33 | GreatSUN | WIMPy: yeah |
| 17:06.04 | GreatSUN | WIMPy: I just got crazy today cause I didn't see the problem anymore |
| 17:06.09 | WIMPy | Ok, that never worked for me. |
| 17:06.16 | GreatSUN | but I think it was just asterisk 1.6 ;o) |
| 17:06.32 | GreatSUN | WIMPy: you got the same card(s)? |
| 17:06.36 | LemensTS | Anyone ever have problems with Acoustic echo on Polycom 321's? |
| 17:06.47 | GreatSUN | so 1.8.3 or .4 ? |
| 17:06.57 | WIMPy | I've got two octoBRIs. |
| 17:07.05 | GreatSUN | WIMPy: nice :o) |
| 17:07.23 | GreatSUN | which driver (+version) are you using? |
| 17:07.47 | WIMPy | Different manufacturers, but the only obvious difference is the colour of the pcb :-) |
| 17:07.59 | WIMPy | I'm using misdn2 with LCR. |
| 17:10.42 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 17:12.57 | Blackthorn | is this the correct command "chown asterisk:asterisk /var/lib/asterisk/astdb" to fix the permission denied error? |
| 17:14.26 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.3 (2011/02/28), 1.6.2.17 (2011/02/28), 1.4.40 (2011/02/28), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 17:15.08 | leifmadsen | Asterisk 1.4.40, 1.6.2.17, and 1.8.3 are now available. Additionally, Asterisk 1.4.41-rc1, 1.6.2.18-rc1, and 1.8.4-rc2 are available. For more information please see the release announcements at http://www.asterisk.org. |
| 17:21.56 | *** join/#asterisk drivefast (~radu@cpe-67-49-43-2.socal.res.rr.com) |
| 17:23.16 | Chainsaw | leifmadsen: Excellent, thanks. Packaging for Gentoo now. |
| 17:23.21 | leifmadsen | nice |
| 17:27.25 | Dovid | hi. If in my invite I have G729, Ulaw and the carrier has Ulaw, G729 asterisk does ulaw. any way to have asterisk go by my prefrence and not my carrier's ? |
| 17:28.16 | Chainsaw | Dovid: That would involve transcoding, having your CPU convert between the two formats. |
| 17:28.32 | Chainsaw | Dovid: Which is wasteful, that is why Asterisk will not do it by default. |
| 17:28.34 | leifmadsen | Dovid: change the order you enable the codecs |
| 17:29.21 | leifmadsen | Chainsaw: he says the provider offers both so transcoding isn't necessary |
| 17:30.50 | Dovid | leifmadsen: I did. But asterisk seems to be going by the carrier and not me. i am the one making the call |
| 17:31.20 | Dovid | in my invite I send G729, G711u. In carriers 183 he sends Ulaw, G729 and asterisk says ok to G711u. |
| 17:32.01 | leifmadsen | then the other end must get the preference |
| 17:32.10 | leifmadsen | don't offer g711u then |
| 17:32.28 | Dovid | leifmadsen: since we both agree to both why does asterisk go by the carriers decision ? |
| 17:32.54 | Dovid | leifmadsen: I can't since some times the carrier can only do ulaw - then i have no choice |
| 17:33.01 | leifmadsen | shrugs |
| 17:33.19 | Dovid | maybe we need a bounty for: use Asterisk codec prefrence over remote system = YES !!! |
| 17:33.57 | leifmadsen | no I imagine it's an architectural issue -- The_Boy_Wonder I believe is going to be looking at codec negotiation for asterisk 1.10 |
| 17:34.03 | leifmadsen | goes to eat lunch |
| 17:34.38 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 17:34.56 | *** part/#asterisk Montys (~dmartinez@nat/digium/x-uuvvlsnwqbamajxg) |
| 17:37.15 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-156-107.dsl.stlsmo.sbcglobal.net) |
| 17:47.06 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 17:57.55 | Katty | HELLO MY ASTERISK DOES NOT WORK HOW TO FIX PLZ???? |
| 17:58.09 | WIMPy | Use a larger hammer. |
| 18:01.54 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-yftwfxuttgaricxo) |
| 18:03.13 | odin917 | does anyone have any information regarding how to identify myself separate from the callerid when making an outgoing call, my voip provider (SIP) is dropping my call if i dont use the callerid assigned to that trunk because they dont know who to charge the call to |
| 18:03.25 | fauxalliance | nice |
| 18:03.56 | odin917 | my provider is broadsoft |
| 18:04.16 | WIMPy | Ask them. |
| 18:04.24 | WIMPy | Or use another. |
| 18:04.41 | odin917 | thanks that was very helpful |
| 18:07.07 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 18:07.34 | gr0mit | hi guys |
| 18:07.52 | gr0mit | struggling to allow incoming calls from a a.b.c.d/27 range |
| 18:08.12 | Faustov | kill it! |
| 18:08.13 | gr0mit | the host= seems to only allow a single IP address |
| 18:09.16 | gr0mit | hey Faustov |
| 18:09.36 | Faustov | deny=0.0.0.0/0.0.0.0 |
| 18:09.36 | Faustov | permit=192.168.200.0/255.255.255.0 |
| 18:09.36 | Faustov | permit=193.93.205.34/255.255.255.255 |
| 18:09.41 | Faustov | what you're possibly looking for |
| 18:09.53 | gr0mit | yes, but i want to match on a peer |
| 18:10.22 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 18:10.32 | gr0mit | the iuncoming call does nto get as far as finding the correct peer |
| 18:10.41 | gr0mit | so it goes into [general] |
| 18:11.08 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 18:11.49 | *** join/#asterisk Jouva (Jouva@fluffy.moufette.com) |
| 18:12.11 | Faustov | hmm, not sure if I understand what you're trying to do |
| 18:12.30 | gr0mit | ok, I have a voip provider X |
| 18:12.46 | gr0mit | who says 'SIP calls will come from a.b.c.d/27 |
| 18:13.34 | Faustov | iptables -I INPUT -s a.b.c.d/27 -j ACCEPT && iptables -P INPUT DROP <--- perhaps? |
| 18:13.55 | gr0mit | and I want to send calls from any of these IP addresses into Provider_X_incoming context |
| 18:14.02 | gr0mit | firewall is fine. |
| 18:14.06 | Faustov | ah ok |
| 18:14.07 | Faustov | I see |
| 18:14.09 | gr0mit | not worried re firewall |
| 18:14.32 | gr0mit | want to have * find the correct peer based on host=a.b.c.d/27 |
| 18:14.45 | gr0mit | but host= only takes a single IP, not a subnet |
| 18:14.52 | Faustov | tricky |
| 18:15.54 | Faustov | I guess you wouldn't be asking if you only had just one such case? |
| 18:16.17 | gr0mit | well, they hve 2 /27 from which I have to accept calls |
| 18:16.19 | WIMPy | AFAIK the only option is to generate 32 peers. |
| 18:17.03 | Jouva | Hey guys, I've got issues trying to get dahdi kernel modules to work. Running almost latest 1.6 (didn't get today's update) on CentOS 5.2 and used the asterisk.org instructions to install via yum. Rebooted the server, and the kernel modules still won't load automatically or with insmod. |
| 18:17.23 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 18:17.46 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-172.dslextreme.com) |
| 18:18.04 | WIMPy | Load them manually or ask someone how is into CentOS. |
| 18:18.11 | WIMPy | s/how/who/ |
| 18:18.44 | Jouva | WIMPy: Again, they won't load with insmod |
| 18:19.06 | WIMPy | What happens? |
| 18:19.26 | Faustov | gr0mit: how about you SNAT it to one address and then use host? ;) ugly I know... |
| 18:20.02 | *** join/#asterisk phidelta (~philipp@85-126-26-62.heiligenstadt.xdsl-line.inode.at) |
| 18:20.07 | WIMPy | Ok, a little less to write, but indeed ugly. |
| 18:20.08 | gr0mit | eeeew! |
| 18:20.20 | Faustov | well |
| 18:20.21 | odin917 | i have been instructed to set "p-charge-identity: 1112223333" for my outgoing call via the trunk, where does that go in asterisk? |
| 18:20.21 | Jouva | Well normally I should be able to just do "insmod dahdi" yes? Or would I need to explicitly use a full path for the file? |
| 18:20.34 | Faustov | gr0mit: I just checked sip.conf.example from 1.6.2.11 |
| 18:20.47 | *** part/#asterisk phidelta (~philipp@85-126-26-62.heiligenstadt.xdsl-line.inode.at) |
| 18:20.48 | gr0mit | oh yes? |
| 18:20.50 | Faustov | it does mention the ACL can be configured per [account] in sip conf |
| 18:20.57 | Faustov | did it not work for you? |
| 18:21.05 | gr0mit | runs 1.4.x |
| 18:21.11 | Faustov | mhm |
| 18:21.23 | WIMPy | Jouva: You should only need to modprobe the driver for your hardware. the main dahdi module will be loaded automatically if everything works. |
| 18:21.43 | Jouva | aha |
| 18:21.58 | Jouva | THAT'S the command I needed. I kept using insmod |
| 18:22.14 | Jouva | Cause it's been so long since I loaded a module and didn't see anything about modprobe :P |
| 18:22.21 | Jouva | Totally forgot about it |
| 18:22.28 | Faustov | gr0mit: final idea, if you don't use host=dynamic elsewhere, you could define the dynamic range as your range and put it there... |
| 18:22.38 | WIMPy | But insmod dahdi should work non the less. |
| 18:22.47 | Jouva | Yeah it didn't though |
| 18:23.06 | Jouva | insmod dahdi first said: insmod: can't read 'dahdi': No such file or directory |
| 18:23.30 | Jouva | So I went into the dahdi directory in the kernel's version under modules and tried insmod dahdi.ko |
| 18:23.34 | Jouva | insmod: error inserting 'dahdi.ko': -1 Unknown symbol in module |
| 18:23.38 | Jouva | But modprobe loaded it fine |
| 18:23.42 | gr0mit | tried host=dynamic |
| 18:23.50 | gr0mit | no worky |
| 18:23.51 | Jouva | Now to see how to make it load automatically on CentOS |
| 18:24.13 | Jouva | But that did what I needed for dahdi to use the MeetMe call |
| 18:24.39 | Faustov | gr0mit: no other hints from the config file ;( |
| 18:25.11 | Faustov | gtg, have a nice evening o/ |
| 18:25.25 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 18:29.54 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 18:33.27 | retentiveboy | I replaced a circa-2005 asterisk box with a new 1.8 machine over the weekend. Things tested great but this morning I found outbound calls resulting in "604 Does not exist anywhere" response from the SIP provider. They're stumped so far. Inbound calls from them are good. Any thoughts on what to look for? |
| 18:34.33 | *** join/#asterisk cusco_ (~tralala@pcmedic.pt) |
| 18:34.35 | cusco_ | hi |
| 18:34.48 | cusco_ | is there some reason why I can't use a context named 'Geral' ?= |
| 18:35.40 | cusco_ | in extensions.ael I have loads of includes like: #include "extensions_partner/extensions_Testing.ael" |
| 18:35.52 | cusco_ | ael reload shows me loading each of those ael's |
| 18:36.00 | retentiveboy | Do you mean [general]? |
| 18:36.12 | cusco_ | but if one of them starts as 'context Geral {' |
| 18:36.15 | cusco_ | it doesn't show |
| 18:36.17 | cusco_ | no |
| 18:36.20 | cusco_ | 'Geral' |
| 18:36.41 | cusco_ | (has the same meaning in portuguese, so when someone calls reception say, I call it geral) |
| 18:36.49 | retentiveboy | no ideas. sorry |
| 18:37.46 | cusco_ | if I change the include line from "extensions_partner/extensions_Geral.ael" to "extensions_partner/extensions_Geral1.ael" ael reload complains that cannot open that file |
| 18:37.56 | cusco_ | else, shows loading everything else but that ael |
| 18:38.09 | cusco_ | dialplan show, shows nothing for context Geral also |
| 18:39.35 | *** join/#asterisk drivefast (~radu@adsl-99-92-126-154.dsl.lsan03.sbcglobal.net) |
| 18:39.56 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 18:43.53 | cusco_ | ok ok |
| 18:43.54 | cusco_ | file permissions |
| 18:43.59 | cusco_ | bloody hell! |
| 18:48.57 | *** part/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 18:52.30 | *** join/#asterisk lanning (~lanning@208.87.233.137) |
| 18:55.01 | Katty | ohai |
| 18:55.11 | Katty | how's everyone's monday |
| 18:55.33 | *** join/#asterisk voxter (~voxter@macpro.daytonhome.voxter.net) |
| 18:55.57 | Qwell | it's monday |
| 18:56.44 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 18:57.03 | *** join/#asterisk timahvo1 (~rogue@41.223.57.77) |
| 18:57.17 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 18:57.25 | nathan7 | Qwell: D: |
| 18:58.11 | *** join/#asterisk johncblandii (~johncblan@c-98-195-123-246.hsd1.tx.comcast.net) |
| 18:58.38 | johncblandii | anyone around to chat real quick? |
| 18:59.48 | Katty | hugs Qwell |
| 19:00.05 | *** join/#asterisk bullium (~wbradshaw@216.68.250.30) |
| 19:00.11 | Qwell | Katty: \o/ |
| 19:00.51 | leifmadsen | johncblandii: if you have a question just ask it |
| 19:01.06 | johncblandii | coolio. was quiet so was unsure if folks were around. : |
| 19:01.07 | johncblandii | :) |
| 19:01.09 | leifmadsen | most of us probably don't have time to just chat it up |
| 19:01.17 | johncblandii | nah, not to just chat. |
| 19:01.24 | leifmadsen | this is IRC -- it can take some time for people to respond |
| 19:01.34 | johncblandii | so, we were gearing up to use Cloudvox but they announced today they are shutting down. |
| 19:01.43 | leifmadsen | never heard of it |
| 19:01.44 | Katty | anyone know the syntax to wget an entire directory? |
| 19:01.47 | johncblandii | are there any similar providers w/ a clean API? |
| 19:01.59 | johncblandii | basically hosted asterisk w/ a simple REST api to access it |
| 19:02.25 | johncblandii | try this katty: http://stackoverflow.com/questions/273743/using-wget-to-recursively-fetch-a-directory-with-arbitrary-files-in-it |
| 19:02.50 | BlackBishop | something -r or -R .. recursive thing .. |
| 19:03.43 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v001-135.mobile.uci.edu) |
| 19:03.45 | johncblandii | so no one knows of an asterisk host w/ a REST api [or similar]? |
| 19:03.58 | Katty | johncblandii: tytyty |
| 19:04.02 | johncblandii | np |
| 19:04.41 | johncblandii | We're looking into the AMI approach but that extends our dev time considerably, it seems |
| 19:05.46 | *** join/#asterisk nicetry (~root@unaffiliated/nicetry) |
| 19:06.34 | nicetry | Hello there, I was wondering if your open source software can be used on a hardware PBX, eg. an Allworx voip pbx |
| 19:10.58 | *** join/#asterisk tuxx- (tuxx@vps460.directvps.nl) |
| 19:11.03 | leifmadsen | nicetry: if your hardware PBX runs Linux, sure |
| 19:11.20 | leifmadsen | I doubt the hardware interfaces (PRI or analog) would be compatible though |
| 19:11.42 | leifmadsen | the likely answer to your question though is "no" |
| 19:12.02 | _Corey_ | nicetry: As far as I know, the Allworx phones are also proprietary |
| 19:16.01 | johncblandii | using the AMI, can you dynamically create dial plans? |
| 19:17.16 | sbrath | has faxdetect changed somehow in 1.8 from 1.6? |
| 19:17.33 | *** join/#asterisk benngard (~mabe@h30n2-g-ml-a11.ias.bredband.telia.com) |
| 19:18.48 | *** join/#asterisk clintc (~clintc@n128-227-2-246.xlate.ufl.edu) |
| 19:20.13 | tzafrir_laptop | johncblandii, and how about writing your own thin wrapper around Asterisk? |
| 19:20.46 | johncblandii | it looks like that's what we're going to have to do |
| 19:20.47 | tzafrir_laptop | (useful more for issueing commands, I guess) |
| 19:21.06 | johncblandii | we'd prefer a prebuilt REST api than having to build our own dial plans, etc |
| 19:22.15 | johncblandii | we see the http server as an option as well |
| 19:27.35 | *** join/#asterisk blackgecko (~blackgeck@189.216.103.155) |
| 19:28.35 | blackgecko | hi: can anyone helpmeto find out how can i a know if a call was ended by the dialplan or the caller ? |
| 19:34.39 | cusco_ | hi... |
| 19:35.37 | cusco_ | in my context I have exten XXX => {bla bla} |
| 19:36.20 | *** join/#asterisk blackgecko (~blackgeck@189.135.80.51) |
| 19:36.34 | *** join/#asterisk NightMonkey (debian-tor@pdpc/supporter/professional/nightmonkey) |
| 19:37.24 | cusco_ | before that I use goto mycontext,260,1 |
| 19:37.34 | cusco_ | it says invalid extension... |
| 19:37.47 | cusco_ | and goes to i => {bla} instead |
| 19:37.49 | cusco_ | :| |
| 19:37.53 | blackgecko | anyone know how can i get to know if a call was hanged up by asterisk or the caller ? |
| 19:37.59 | Qwell | cusco_: because you don't have an extension that matches 260 |
| 19:38.02 | Qwell | XXX != _XXX |
| 19:38.08 | *** join/#asterisk Micc (~Micc@74.93.108.114) |
| 19:38.23 | cusco_ | ow, what is the differente ce Qwell ? |
| 19:38.36 | cusco_ | difference |
| 19:38.36 | Qwell | One way does what you want, the other doesn't. |
| 19:38.44 | cusco_ | got that.. |
| 19:38.55 | cusco_ | ok thanks, but still what is the difference? |
| 19:39.41 | Micc | why is there about a 1.5 second delay in receiving audio from a parked caller after they have been picked up? I've noticed this for years. I thought it would be fixed by now, but maybe its not a bug but a problem with my configuration. |
| 19:40.03 | Micc | and the delay gets shorter over time as the call goes on. |
| 19:40.59 | jaytee | cusco_, the leading _ indicates to Asterisk that it is a pattern match, not an actual extension. |
| 19:41.18 | nicetry | thanks _Corey_ and leifmadsen - yes I think its all "proprietary" but you see a lot of 'nix running on a lot of proprietary stuff these days so I though I'd take a shot in the dark. I'm just sick of this useless & closed UI |
| 19:41.20 | cusco_ | ah ok thanks jaytee |
| 19:41.38 | cusco_ | blackgecko: perhaps you can check the cli logs? |
| 19:41.49 | _Corey_ | nicetry: Send Paetec a "thank you" note... :) |
| 19:42.01 | johncblandii | anyone curious, here is a service I found w/ a simple API to Cloudvox: https://www.tropo.com/. |
| 19:42.55 | blackgecko | cusco: but is there a variable that i can evaluate to know that? ive tried with HANGUPCAUSE but i always get Normal_clearing |
| 19:43.07 | johncblandii | l8rz |
| 19:43.08 | *** part/#asterisk johncblandii (~johncblan@c-98-195-123-246.hsd1.tx.comcast.net) |
| 19:43.46 | blackgecko | cusco: this is inside an AGI doing some kind of call limit |
| 19:44.46 | cusco_ | so you don't know if it is either the call limitting or the callee hunging up ? |
| 19:45.06 | cusco_ | can't you tell your call limitting application to NoOp() if it hangs up? |
| 19:46.39 | blackgecko | what im doing is using the Dial((L):(x)) |
| 19:46.51 | blackgecko | and im really stuck with this |
| 19:47.41 | _Corey_ | blackgecko: Can't you grab the hangup and process it in your AGI? Should be fairly simple to see then. |
| 19:48.36 | _Corey_ | blackgecko: Whatever runs after your Dial with the L() param will get run should the Dial kill the call for that reason and if one or the other party hangs up it'll go to your hangup handler |
| 19:49.40 | blackgecko | sorry but im a newb in agi development |
| 19:49.48 | blackgecko | let me see if i get that |
| 19:49.55 | blackgecko | i should do |
| 19:51.47 | *** join/#asterisk DelphiWorld (~VoIpGuy@41.200.29.140) |
| 19:51.51 | DelphiWorld | hey. |
| 19:52.42 | cusco_ | hi |
| 19:53.56 | *** part/#asterisk nicetry (~root@unaffiliated/nicetry) |
| 19:54.09 | cusco_ | blackgecko: if what _Corey_ says is true, you just for example add NoOp(This exceeded the limits imposed.); after the dial |
| 19:54.38 | blackgecko | exten => _99XX,n,Dial(Local/${DID_EXTENSION},25,trL(240000:120000:30000)M(cti,Cid,startCall,${LOG_ID},${CALLERID(num)},${EXTEN},${CALL_ID})) |
| 19:54.38 | blackgecko | exten => _99xx,n,NoOp(system_nahgup) |
| 19:54.43 | cusco_ | if it noops it was the limit, else the callee |
| 19:56.20 | _Corey_ | blackgecko: I'm assuming you're calling the Dial within your AGI with what I said above... |
| 19:57.50 | *** part/#asterisk DelphiWorld (~VoIpGuy@41.200.29.140) |
| 20:29.59 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 20:29.59 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.3 (2011/02/28), 1.6.2.17 (2011/02/28), 1.4.40 (2011/02/28), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 20:31.03 | p3nguin | You should have an extension for every phone number that you want to use from a provider. |
| 20:33.34 | *** join/#asterisk fofware (~Fabian@host63.190-31-62.telecom.net.ar) |
| 20:33.55 | odin917 | p3nguin: i have many internal users with 2 ring groups, when i get the call it always goes to one ring group, the rule for the inbound route is not triggering because i only see one number in my asterisk log for incoming regardless of which DID i dial |
| 20:39.30 | p3nguin | odin917: Which version of FreePBX are you using? |
| 20:39.58 | odin917 | i dial 2125551111 i see incoming 2125551111, asterisk forward to ringGroup1, i dial 2125552222 i still see "incoming catch all" and it hits ring group1 |
| 20:40.08 | odin917 | p3nguin: 2.6 |
| 20:40.12 | p3nguin | odin917: This channel doesn't support FreePBX, so please ask for help in the appropriate channel. |
| 20:41.02 | chazzam | I see wut you did there |
| 20:41.09 | odin917 | p3nguin: it doesnt have to do with freepbx |
| 20:41.20 | p3nguin | odin917: Yes it does. |
| 20:41.22 | odin917 | im debugging this whole issue from the asterisk log |
| 20:41.39 | chazzam | odin917: did you configure it from freepbx? |
| 20:41.42 | p3nguin | odin917: FreePBX is responsible for cofiguring all of the aspects you're mentioned. |
| 20:42.01 | p3nguin | s/you're/you've/ |
| 20:42.05 | odin917 | ignore everything but the part where i receive the call and asterisk states the DID |
| 20:43.02 | odin917 | OT: infobot expanding sed command is sweet |
| 20:44.06 | *** join/#asterisk espiceland (~erin@207.98.195.107) |
| 20:56.38 | *** join/#asterisk jamesgecko (~james@64.207.3.161) |
| 20:59.39 | *** part/#asterisk clintc (~clintc@n128-227-2-246.xlate.ufl.edu) |
| 21:12.06 | *** join/#asterisk juniorsa (~Miranda@206.248.146.160) |
| 21:12.49 | juniorsa | Hi, I am looking for a rock solid ata (2 ethernet ports, 2 fxs ports) that works with asterisk (sip also), any suggestions? |
| 21:14.06 | leifmadsen | try the Linksys stuff... SPA-3102 I think |
| 21:14.54 | juniorsa | is grandstream stuff any good? |
| 21:15.49 | _Corey_ | juniorsa: Depends on your definition of "good"... ;) |
| 21:15.49 | juniorsa | something that does QOS right? |
| 21:17.11 | *** join/#asterisk SiNGLer (~singler@78-60-139-121.static.zebra.lt) |
| 21:17.39 | *** join/#asterisk dwayne (~dwayne@c-71-207-214-190.hsd1.al.comcast.net) |
| 21:21.13 | *** join/#asterisk IsUp (IsUp@unaffiliated/isup) |
| 21:37.55 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 21:42.31 | Tozz_ | juniorsa: Grandstream ATA's work great here once their configured |
| 21:42.41 | Tozz_ | but the configuration interface has bugs, sucks and more of that kind |
| 21:43.07 | Tozz_ | but I have a few of them running here for years, they have never failed |
| 21:53.31 | IsUp | Tozz_: same here |
| 21:54.23 | juniorsa | Tozz_: would you try and have a 2 port ethernet to put the ATA directly to the cable modem and then a wireless router, do these units enforce qos? or should I just get the 1 ethernet version and plug it into a port behind the wireless router? |
| 21:57.33 | IsUp | juniorsa: just get 1 eth version and plug it to router |
| 22:00.30 | Tozz_ | I would not use the Grandstream as router |
| 22:00.36 | Tozz_ | its a stupid feature anyway |
| 22:00.48 | Tozz_ | as IsUp says, just plug it into the router |
| 22:02.51 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 22:03.28 | *** join/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110) |
| 22:07.09 | *** part/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 22:07.18 | *** join/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110) |
| 22:09.52 | juniorsa | what about QOS? if it's behind the router won't it be affected by bittorrent and stuff? |
| 22:10.38 | Tozz_ | your router should do QoS |
| 22:10.55 | Tozz_ | I doubt the grandstream is any good at it with its router function |
| 22:11.00 | fenrus | and your provider should understand the QoS/COS flags youset. |
| 22:11.44 | juniorsa | ok, sounds good, thanks for all your good advice |
| 22:21.15 | *** part/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 22:24.25 | *** join/#asterisk mountainm2k (~msturtz@gw.booyahnetworks.com) |
| 22:26.15 | mountainm2k | Hi all-- I'm once again in a position of needing some additional Polycom phones, hopefully just like our IP-4000 speakerphone. Can anybody suggest a current-generation replacement? |
| 22:26.57 | mountainm2k | I guess the IP-4000 is still available |
| 22:27.01 | Qwell | http://www.polycom.com/products/voice/conferencing_solutions/conference_phones/soundstation/soundstation_ip7000.html |
| 22:27.31 | Qwell | But yes, 4000s should not be difficult to find. |
| 22:27.33 | mountainm2k | hahah, HD Voice makes me giggle every time I hear it |
| 22:27.51 | mountainm2k | it's only an 8khz audio path, it ain't HD |
| 22:27.54 | mountainm2k | <shrug> |
| 22:28.08 | Qwell | Except it isn't. |
| 22:28.49 | mountainm2k | huh, really? Reading about it... |
| 22:29.28 | mountainm2k | This isn't that system where it sends a tone burst to the other end, and if it finds another Polycom then it goes into a data mode? |
| 22:29.34 | Qwell | no |
| 22:29.42 | Qwell | These are widely available codecs. |
| 22:29.56 | mountainm2k | ...but still SIP... |
| 22:30.00 | Qwell | and? |
| 22:30.24 | mountainm2k | and still limited to the bandwidth of a single circuit of the carrier circuit |
| 22:30.36 | Qwell | Who says the call has to go through a carrier? |
| 22:31.11 | mountainm2k | Well, I guess *I* say so :-) since that's how my * is configured -- it doesn't do VoIP outside our WAN |
| 22:31.22 | mountainm2k | although I guess there's no reason it would have to stay that way |
| 22:31.28 | Qwell | and HD codecs do no good inside a WAN? |
| 22:31.49 | mountainm2k | If we were more than one building, and people regularly made station-to-station calls |
| 22:32.00 | *** join/#asterisk philfine (~philfine@a213-22-45-215.cpe.netcabo.pt) |
| 22:32.02 | mountainm2k | IE, calls within our WAN |
| 22:32.06 | mountainm2k | which they almost never do |
| 22:32.29 | philfine | Hello everyone, how can I make my dial plan Dial through a different channel if one is busy or failed for some reason ? |
| 22:32.49 | *** join/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110) |
| 22:33.24 | philfine | Something like, first try to make a call though cheaper provider. In case it is already being used of failed for some reason, use the other one |
| 22:33.33 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 22:40.55 | *** part/#asterisk mountainm2k (~msturtz@gw.booyahnetworks.com) |
| 22:44.10 | p3nguin | This is really pissing me off. I've got this Cisco 7960G with MGCP on it, and I can't switch it over to SCCP. It just won't "upgrade" to the SCCP image. It reads the files, it acts like it understands the firmware files, but then goes back to looking for MGCP files. |
| 22:45.07 | *** join/#asterisk jro (~jro@66.252.74.2) |
| 22:45.30 | jro | Is there a way to completely reject an inbound call from a certain DID? not answer it then terminate it |
| 22:45.59 | p3nguin | sure |
| 22:46.20 | jro | right now I get recieved a call from an unknown number, which answers it, then says no service avail |
| 22:46.26 | WIMPy | Hangup() |
| 22:46.26 | jro | i dont want to accept it from the trunk at all |
| 22:46.43 | p3nguin | Whichever extension you don't want to answer, use Hangup() instead of an app that answers. |
| 22:47.07 | WIMPy | Hangup(21) would be more appropriate. |
| 22:47.24 | p3nguin | I don't know the hangup codes, so I'll take your word on thAt. |
| 22:47.47 | jro | hmm I'll try it |
| 22:48.41 | p3nguin | You should already have an extension for every DID you own, so this should be pretty trivial to configure on your box. |
| 22:49.25 | *** join/#asterisk Faithful (~Faithful@carame.lnk.telstra.net) |
| 22:52.08 | *** join/#asterisk NightMonkey (debian-tor@pdpc/supporter/professional/nightmonkey) |
| 22:57.31 | p3nguin | GAH! This is so annoying. |
| 22:57.59 | p3nguin | Maybe I need to use an older SCCP version. |
| 22:58.15 | p3nguin | or change it to SIP |
| 22:58.30 | *** join/#asterisk geek_cl (~lletelier@190.151.34.170) |
| 22:58.32 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 22:59.09 | p3nguin | I've never had any trouble going between SIP and SCCP (both ways), but this MGCP is kicking my ass. |
| 22:59.41 | *** join/#asterisk exothermc_ (~miles@74.85.89.146) |
| 23:00.01 | *** join/#asterisk atan2 (~atan@unaffiliated/atan) |
| 23:01.44 | Qwell | p3nguin: Try using one of the other 16 different ways to upgrade. |
| 23:01.51 | Qwell | It's changed about that many times. |
| 23:02.56 | *** join/#asterisk fenlander (~fenlander@82.152.81.57) |
| 23:02.59 | p3nguin | According to http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/7960g_7940g/mgcp/firmware/matrix/frmwrup.html , MGCP images 7.x or later to SCCP images 5.x or later I need to follow Procedure E. Procedure E gives me to ways. I've tried both, and I get the same results. |
| 23:05.25 | *** join/#asterisk arielb27 (~chatzilla@99-1-236-49.lightspeed.miamfl.sbcglobal.net) |
| 23:05.41 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 23:09.12 | *** join/#asterisk dwayne (~dwayne@c-71-207-214-190.hsd1.al.comcast.net) |
| 23:09.54 | ChannelZ | netwerk trubbles |
| 23:09.55 | *** join/#asterisk ghenry (~ghenry@pdpc/supporter/monthlybyte/ghenry) |
| 23:10.54 | *** join/#asterisk Dibbler (~Dibbler@87-194-103-72.bethere.co.uk) |
| 23:11.35 | p3nguin | The phone reads my SEP cnf files, reads the SCCP fimware .loads file and then goes on to ask the tfptd for the MGC cnf files like it didn't know what the SCCP firmware files meant. |
| 23:12.21 | Tozz_ | factory reset? |
| 23:12.27 | p3nguin | Done it twice. |
| 23:12.38 | p3nguin | I think it had even been done before I got here to mess with it. |
| 23:13.03 | p3nguin | I really didn't want to configure MGCP on Asterisk for a single bastard phone. |
| 23:14.01 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 23:14.12 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 23:14.21 | *** join/#asterisk chasing`Sol (~chasingSo@82.201.135.150) |
| 23:14.40 | *** join/#asterisk pabelanger (~pabelange@50.22.5.41-static.reverse.softlayer.com) |
| 23:14.41 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 23:16.39 | *** join/#asterisk alecdavis (~alec@202.36.76.21) |
| 23:19.41 | *** join/#asterisk wonderworld (~ww@port-92-201-55-88.dynamic.qsc.de) |
| 23:23.30 | *** join/#asterisk cyphorious (~cyphoriou@chello062178189196.2.15.tuwien.teleweb.at) |
| 23:44.11 | *** join/#asterisk aut (~aut@c-174-48-60-12.hsd1.fl.comcast.net) |
| 23:44.29 | aut | does anybody know anything about comcast PRIs? how they compare to using a telco like at&t? they say it's delivered over their docsis network. |
| 23:44.52 | aut | comcast is significantly cheaper than at&t, so it makes me wonder what the differences might be |
| 23:45.20 | *** join/#asterisk bjhaid (~abejide@41.206.11.1.vgccl.net) |
| 23:47.42 | Qwell | eww |
| 23:47.46 | Qwell | packetcable>SIP>PRI |
| 23:48.34 | _Corey_ | The TDM PRI is an endangered species... |
| 23:49.06 | _Corey_ | With the exception of a couple ILECs, we see this "integrated" "enhanced" stuff everywhere now |
| 23:49.13 | aut | qwell: what? :) |
| 23:49.45 | Qwell | aut: PacketCable over the DOCSIS network, converted to SIP at the modem, then to PRI at the IAD. |
| 23:50.46 | Qwell | It's only PRI for the sake of being PRI. They could easily deliver it as SIP instead, and skip a layer. |
| 23:51.17 | aut | that's what i kind of figured... hrm |
| 23:51.21 | aut | so it sucks? :) |
| 23:51.32 | Qwell | It's Comcast. It would suck no matter the method used. |
| 23:51.39 | _Corey_ | lol |
| 23:51.46 | Qwell | Do you REALLY want to trust your phone connectivity to Comcast? |
| 23:51.50 | aut | not particularly... |
| 23:51.57 | Qwell | I rest my case. |
| 23:52.04 | aut | is there some nationwide company that is generally recognized for business voice? i've heard paetec a lot |
| 23:52.13 | aut | at&t is a fortune for pri |
| 23:52.15 | Qwell | AT&T? :p |
| 23:52.34 | _Corey_ | What part of the US are you in? |
| 23:52.38 | aut | miami |
| 23:52.55 | _Corey_ | I've got a customer in Ft. Lauderdale on Paetec |
| 23:53.16 | aut | dealing with at&t is retarded too... don't know how to get in touch with someone who understands business needs. i was talking to someone there who didnt understand that i need what paetec calls "direct trunk overflow" |
| 23:53.29 | _Corey_ | They're using their SIP product... it works well enough, though the customer service was pretty rough for them |
| 23:53.32 | aut | _corey_: interesting... how is it? are they cheaper than at&t? |
| 23:53.36 | aut | ugh |
| 23:53.45 | _Corey_ | Once it's up and going... |
| 23:53.52 | aut | do they have a t1? |
| 23:53.56 | aut | with the sip product? |
| 23:54.10 | _Corey_ | Yeah, they have two data T1s |
| 23:55.11 | _Corey_ | I didn't realize Comcast was in Miami |
| 23:55.27 | _Corey_ | I've got a customer in the Philly market with 100MB fiber drop from them that's pretty happy |
| 23:55.41 | _Corey_ | I wouldn't really touch them for voice service though |
| 23:56.23 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-129-161.lsanca.fios.verizon.net) |
| 23:58.57 | AirGuitarist | i'm using PBIAF for my asterisk server and noticed my call logs in mysql don't have the unique id. was wondering if anyone had tried recompiling asterisk with a PBIAF install to change that |