| 00:00.11 | WindowsUser | ;exten => 123,1,Answer() ; having a priority 1 is more important than Answer() call :) |
| 00:00.37 | ajmcello | ah hmm |
| 00:00.47 | *** join/#asterisk joobie (n=joobie@mx01.anric.com.au) |
| 00:01.44 | dshap | WindowsUser: do you know how to get "asterisk -r" to pull up the CLI on an already-running asterisk server and get it to retain the colors that it has when you start with "asterisk -c" ? |
| 00:02.01 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 00:02.24 | WindowsUser | -cr |
| 00:02.43 | WindowsUser | i dont actually use the colors :) |
| 00:04.10 | jaytee | they're pretty! I'm a big fan of magenta |
| 00:04.18 | *** join/#asterisk buttons840 (n=buttons8@c-24-10-149-58.hsd1.ut.comcast.net) |
| 00:04.54 | dshap | once you're in the Asterisk CLI |
| 00:04.56 | dshap | how do you get out of it |
| 00:04.56 | buttons840 | if i put myfile.blah into the /var/spool/asterisk/outbound will it be parsed and sent out? or does it have to be a *.call file? |
| 00:05.00 | dshap | without stopping the server? |
| 00:05.18 | dshap | buttons840: pretty easy to test, lol |
| 00:06.02 | buttons840 | dshap, true... |
| 00:06.22 | joat | you can also mess with the timestamp and cause the call to be made some time in the future |
| 00:07.32 | buttons840 | joat, i know but thanks, and for the record, it will parse any text file, regardless of it's name or extension, i just tested it |
| 00:08.34 | dshap | ok i have an interesting problem...i have an extension that lets a user record an audio file and then when they press pound it triggers an AGI script which creates a call file that calls some other phone and attempts to play the audio file that the user just recorded |
| 00:08.42 | dshap | the issue is that there is some processing time necessary |
| 00:08.54 | dshap | after the user records and presses pound, the file does not immediately appear in the directory |
| 00:09.03 | dshap | and so the extension that tries to play it can't find it |
| 00:09.24 | dshap | what do you guys recommend i do? |
| 00:09.52 | buttons840 | so the user presses pound and the file doesn't immediately appear? where is it being created at? |
| 00:09.56 | dshap | somehow get the length of the recording and pass that as a variable to the AGI script so that it can tell the extension (via a call file variable) to wait a proportional amount of time? |
| 00:10.03 | dshap | buttons840: sounds/myrecordings |
| 00:10.15 | dshap | buttons840: if it's like a 15 sec recording, it takes some time for it to appear (slow server) |
| 00:10.31 | buttons840 | the agi script creates the call file? |
| 00:11.18 | buttons840 | will Wait(2) solve your problems? ;) |
| 00:12.12 | joat | if the filename is known, pass it to a script which waits for the file to appear in the appropriate directory before creating the call file |
| 00:13.28 | buttons840 | ^^ good idea, just wait in the script, that would ensure the fast processing possible, you have to wait for the file to appear no matter what |
| 00:14.18 | joat | or just set wait high enough (i.e., limit the length of the call and set wait high) |
| 00:14.26 | joat | either way should work |
| 00:14.39 | joat | err... sorry... Wait() |
| 00:14.42 | joat | vice wait |
| 00:15.35 | joat | i think dshap answered his own question though |
| 00:16.50 | buttons840 | indeed, i find asking in irc always helps me to answer my own questions. |
| 00:17.42 | joat | heh |
| 00:17.53 | dshap | hah |
| 00:17.55 | dshap | well |
| 00:18.02 | dshap | waiting until the file exists could be risky |
| 00:18.07 | dshap | what if for some reason the recording fails |
| 00:18.10 | dshap | then it'll be stuck |
| 00:18.12 | *** join/#asterisk Joelito (n=joel@189.220.24.163) |
| 00:18.22 | dshap | well i guess i could juts set some upper limit on the wait time |
| 00:18.26 | joat | yeah, then you'd have to add a timeout to the script |
| 00:18.28 | joat | heh |
| 00:18.34 | dshap | how do u wait until a file exists? |
| 00:18.37 | buttons840 | what if the recording fails? better code some error handling into that script |
| 00:18.47 | dshap | a Goto loop? |
| 00:18.51 | joat | check for file, sleep, loop |
| 00:18.57 | joat | :) |
| 00:19.01 | dshap | is there an asterisk app to check for file? |
| 00:19.34 | *** part/#asterisk Joelito (n=joel@189.220.24.163) |
| 00:19.37 | buttons840 | i don't know |
| 00:19.39 | joat | you're already using an AGI, right? |
| 00:19.48 | joat | err... script? |
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| 00:23.53 | dshap | sounds like a pain |
| 00:23.54 | dshap | but ok |
| 00:24.01 | dshap | well actually |
| 00:24.06 | dshap | this is going to happen once the user has hung up |
| 00:24.14 | dshap | and DeadAGI is deprecated in 1.6 |
| 00:24.17 | dshap | so i dunno what i do then |
| 00:24.44 | Qwell | dshap: it's only deprecated because AGI can act in that mode now |
| 00:25.45 | dshap | oh so i can call AGI() and send commands to Asterisk even on a hung up channel then? |
| 00:25.48 | dshap | it would only be a Set() command anyways |
| 00:28.06 | *** join/#asterisk s14ck (n=s14ck@190-76-102-13.dyn.movilnet.com.ve) |
| 00:28.10 | dshap | anyone here have any experience with audio detection? i'd like to have asterisk call someone's voicemail (it will know that it is a voicemail) and wait for the "beep" before it starts to use Playback() to play a message. |
| 00:28.18 | dshap | how would i make asterisk wait for the beep? |
| 00:29.03 | jaytee | dshap, use AMD() |
| 00:29.55 | dshap | isn't that for detecting if the audio on the line is human or a machine? |
| 00:30.04 | *** join/#asterisk s14ck (n=s14ck@190-76-102-13.dyn.movilnet.com.ve) |
| 00:30.09 | dshap | i'm not interested in that distinction because i *know* it will be the answering machine |
| 00:30.19 | dshap | i'm just interested in waiting until the beep |
| 00:31.57 | *** join/#asterisk pagec (n=cpage@ip67-152-24-104.z24-152-67.customer.algx.net) |
| 00:32.00 | outtolunc | silence and non-silence |
| 00:32.13 | pagec | hey what is the comment to get dahdi to install the base files in /etc? |
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| 00:32.57 | WindowsUser | dshap: making a service to break up with people via voicemail? |
| 00:33.33 | pagec | s/comment/make command |
| 00:33.49 | WindowsUser | make help? |
| 00:33.51 | dshap | WindowsUser: haha nah |
| 00:34.24 | pagec | no such target |
| 00:34.25 | voxter | any of you guys have a working OSX asterisk dialer? |
| 00:34.28 | dshap | outtolunc: not a bad idea but it's a bit more complicated...i'm calling an IVR and sending DTMF to get to the correct mailbox |
| 00:34.52 | dshap | outtolunc: so basically i get some non-silence, then i sendDTMF, then some silence, then some talking, then beep |
| 00:35.13 | dshap | outtolunc: maybe i could figure out how many periods of silence there are and do it that way |
| 00:36.40 | WindowsUser | is it dialing beyond stuff like "please enter mailbox number" |
| 00:37.28 | WindowsUser | I think the hardest part will be waiting for the beep, figuring out the different tones used by different systems |
| 00:37.43 | dshap | WindowsUser: nope, that's it really |
| 00:37.55 | dshap | WindowsUser: and im basically programming each system it will call on a 1-by-1 basis |
| 00:38.13 | dshap | WindowsUser: i guess the takeaway here is that i should learn how to use silence detection in asterisk |
| 00:38.15 | *** join/#asterisk denesh (n=chatzill@216.105.80.149) |
| 00:38.20 | denesh | hi all... |
| 00:39.19 | denesh | i have incoming calls through an SIP trunk working fine... but outgoing calls have no sound... i have get these warnings Asked to transmit frame type 4, while native formats is 1024 (read/write = 4/4) |
| 00:42.04 | denesh | Jul 2 20:35:34 WARNING[10398]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 1024 (read/write = 4/4) |
| 00:42.59 | dshap | denesh: u behind a NAT? |
| 00:43.31 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 00:43.33 | denesh | dshap: yes i am.. but i have the ports forwarded 5060-5061 and 10000-20000 UDP |
| 00:44.08 | WindowsUser | asterisk might need to be told its IP address |
| 00:44.28 | dshap | denesh: you MUST have externip set under [general] in sip.conf |
| 00:44.28 | denesh | its public IP ? |
| 00:44.32 | dshap | yes |
| 00:44.40 | dshap | there's another thing |
| 00:44.42 | dshap | i had your exact issue |
| 00:44.51 | dshap | and once everything else was fixed in my sip.conf |
| 00:45.13 | dshap | adding "fromdomain=yourSIPhost" underneath the particular trunk that was giving me issues |
| 00:45.17 | dshap | fixed it |
| 00:45.27 | dshap | so my SIP host is sip.flowroute.com |
| 00:45.36 | denesh | dshap: good... i'll give it ago... |
| 00:45.57 | dshap | yea try that |
| 00:46.04 | dshap | but obviously u could have other issues w/ur sip.conf |
| 00:46.07 | dshap | but that one seemed to be the least trivial |
| 00:46.12 | dshap | i literally spent hours trying to get my shit to work |
| 00:46.17 | dshap | so annoying |
| 00:46.17 | dshap | hah |
| 00:46.27 | dshap | all because i had a dumbass router |
| 00:52.05 | *** part/#asterisk buttons840 (n=buttons8@c-24-10-149-58.hsd1.ut.comcast.net) |
| 00:52.32 | denesh | hmm i guess i am going to run into the dynamic dns issue... does asterisk take domain names instead of ip addresses |
| 00:55.12 | joat | use externhost instead of externip |
| 00:55.28 | denesh | thanks |
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| 01:47.05 | implicit | dshap, adding fromdomain=sip.flowroute.com caused an issue or fixed one? |
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| 02:18.45 | pagec | why would the asterisk 1.6.0 setup say Notice: Configuration file is /etc/zaptel.conf when i am modprobing wct4xxp? aka what do i need to change in /etc to change that behavior |
| 02:19.03 | kb3ien | is there or is there not support for multiple parking lots at this time in the default trunk? |
| 02:20.21 | kb3ien | there is an esoteric mention of parkinglot=plaza in sip.conf but no clues as to in which contexts one may override the defualt vales. |
| 02:20.24 | kb3ien | values. |
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| 02:24.57 | *** part/#asterisk lanning (n=lanning@173.8.187.197) |
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| 02:58.07 | vegbox | Hello All |
| 03:02.15 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-1e4d9aded66f02d4) |
| 03:08.24 | kb3ien | nhin |
| 03:08.27 | kb3ien | hi |
| 03:10.17 | vegbox | I just got an FXO card but I am having a hard time getting it to work, it works on the default option |
| 03:10.35 | vegbox | like in the default plan, exten => s,1,VoiceMail |
| 03:11.28 | vegbox | but I want my asterisk box to pick up calls, know which ext is calling and do our night time/day time messages |
| 03:32.36 | *** join/#asterisk andresmujica1 (n=andresmu@ubuntu/member/andresmujica) |
| 03:39.53 | vegbox | Who has the best rates for outgoing calls? |
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| 03:54.05 | dshap | hey is there anyone here who knows how to get the current dialplan that asterisk is running? i have no idea how this happened, but i think my FTP client deleted my current extensions.conf |
| 03:54.19 | dshap | but when i do "dialplan show", it shows me everything |
| 03:54.25 | dshap | is there any way to get the dialplan-formatted dialplan? |
| 03:54.28 | dshap | ugh |
| 04:00.28 | *** join/#asterisk sah-work (n=Bawbatos@99-40-7-14.lightspeed.sntcca.sbcglobal.net) |
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| 04:28.11 | Alian | Hello |
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| 05:07.28 | kb3ien | anyone else noticed that udpbindaddr=0.0.0.0:5050 |
| 05:07.39 | kb3ien | doesnt do what you'd expect? |
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| 05:53.59 | ajmcello | is it possible to customize the voicemail system prompts and greetings? |
| 05:54.11 | *** join/#asterisk Defraz (n=tim@24-117-236-174.cpe.cableone.net) |
| 05:54.31 | joako | ajmcello: Sure but what exactly do you mean? |
| 05:55.01 | ajmcello | i want to change the number options, 1 for busy, 2 for unavailable, 4 to manage folders, etc.. |
| 05:55.18 | joako | ajmcello: Yes but you have to edit the code and recompile |
| 05:55.45 | ajmcello | app_voicemail? |
| 06:04.39 | joako | ajmcello: Yes |
| 06:04.46 | joako | Really I don't see the purpose |
| 06:05.14 | joako | I thought about it a while back but determined the asterisk voicemail wasn't as bad as I made it out to be... it's actually rather easy to use |
| 06:05.24 | ajmcello | minivm might work |
| 06:05.30 | ajmcello | but it has been stripped of odbc/sql |
| 06:05.41 | ajmcello | :( |
| 06:16.04 | WindowsUser | whats the difference between voicemail and minivm? |
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| 07:07.38 | trnzmeta | guys: for asterisk behind firewall, I need to portforward the rtp ports |
| 07:08.01 | trnzmeta | rtp is tcp or udp? |
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| 07:11.15 | ISO9001 | udp |
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| 07:13.50 | tokozedg | hi, does asterisk have built in conference call function, i mean call someone and automatically add in a conference? |
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| 08:04.07 | JackTheNipple | tokozedg: for this question you registered with freenode.net? |
| 08:04.16 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 08:05.14 | JackTheNipple | tokozedg: did you ask your favorite search engine? May I suggest some keywords to search for? |
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| 08:11.41 | *** join/#asterisk Cyorxamp (n=Cyorxamp@212.57.229.111) |
| 08:11.53 | Cyorxamp | Hey is there a way of clearing the Asterisk CLI history? |
| 08:12.30 | jblack | Clear the screen? ctrl-l |
| 08:12.41 | Cyorxamp | no the history of entered commands |
| 08:13.02 | JackTheNipple | .asterisk_history |
| 08:13.08 | JackTheNipple | analog to .bash_history |
| 08:13.21 | Cyorxamp | but it would require an asterisk restart to take effect? |
| 08:13.29 | JackTheNipple | test it |
| 08:13.32 | JackTheNipple | not sure.... |
| 08:13.37 | JackTheNipple | ehm |
| 08:13.54 | JackTheNipple | yes, but quite reasonable ;-) as bash does the same |
| 08:14.07 | JackTheNipple | reading history for the session and re-writing it afterwards |
| 08:14.10 | ickmund | 'sip show peers' tells me that all my peers are unmonitored. How do I turn on monitoring? |
| 08:14.31 | JackTheNipple | what shall this monitoring do? |
| 08:14.35 | JackTheNipple | ...for you? |
| 08:15.08 | Cyorxamp | JackTheNipple, it works, just get rid of the commands you don't want in there and they don't show when pressing up in the console any more - no asterisk restart needed |
| 08:15.09 | Cyorxamp | thanks :D |
| 08:15.29 | JackTheNipple | welcome ;-) |
| 08:16.00 | ickmund | JackTheNipple: I'm guessing asterisk sends out a ping of some sort, maybe an option, to see what peers are still there? |
| 08:17.00 | ickmund | JackTheNipple: In the end I want to be able to get this data via AGI to be able to parse which peers are there (for a simple directory service) |
| 08:19.40 | JackTheNipple | ickmund: try to use "qualify=200" in sip.con |
| 08:20.39 | ickmund | JackTheNipple: exactly what I was looking for, many thanks |
| 08:21.00 | JackTheNipple | ;-) |
| 08:21.49 | JackTheNipple | you might need to adopt value of qualify to something thats fits your needs, but normaly 200 is quite fine. |
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| 08:30.52 | yidiyuehan | hi, guys, i want to send the extension status to windows OS like DOS or VB.net, anybody has any cue or any reference page on this? |
| 08:32.40 | JackTheNipple | yidiyuehan: you are looking for AMI, asterisk manager interface. |
| 08:33.19 | JackTheNipple | yidiyuehan: can be used with VB, perl, java and even assembler, if you're hard enough |
| 08:34.47 | JackTheNipple | yidiyuehan: good point to start: http://www.voip-info.org/wiki/view/Asterisk+manager+API |
| 08:35.35 | JackTheNipple | and follow the white rabbit! |
| 08:37.38 | ISO9001 | lol assembler AMI |
| 08:37.44 | JackTheNipple | ^^ |
| 08:38.13 | ISO9001 | yidiyuehan: you should totally go the assembler route. You'll never get the raw AMI performance with anything but hand-tuned asm. |
| 08:38.20 | JackTheNipple | yeah, give a shit on all this overhead - who need tcp/ip & http-libs! |
| 08:42.50 | yidiyuehan | hi Jackthenipple, |
| 08:43.22 | ISO9001 | incidentally, if you need any help with the asm, just ask jack. He's like a god among men. |
| 08:44.07 | yidiyuehan | I have viewd the AMI and it seems that it's only one-way communication to asterisk, now I am wondering how to send back something from linux to windows to update asterisk status. |
| 08:45.29 | JackTheNipple | no, its two-way |
| 08:45.57 | JackTheNipple | yidiyuehan: you will receive "events" on AMI as well |
| 08:46.12 | *** join/#asterisk guaxinim (n=guaxinim@unaffiliated/guaxinim) |
| 08:46.19 | JackTheNipple | otherwise you could do status-polling to stay informed. |
| 08:47.01 | JackTheNipple | AGI & AMI are the only way (beside retrieving infos from a DBMS) how to get information from asterisk to external systems |
| 08:55.36 | ISO9001 | only info in the db is cdr, no? |
| 08:55.39 | yidiyuehan | Hi, Jack, that's it. AGI |
| 08:56.01 | yidiyuehan | I guess I need to focus on these two parts: AGI and AMI. thanks a lot bro |
| 09:01.59 | *** join/#asterisk merlin8282 (n=merlin82@AStrasbourg-753-1-21-124.w90-56.abo.wanadoo.fr) |
| 09:03.51 | merlin8282 | Hi. I want to compile the AGX addons (enabling rxfax) for my asterisk 1.4.22 installation, but the AGX README says I need * 1.4.17 (for the "OLD" version of AGX) or * 1.4.24 (for the "NEW" version). |
| 09:04.13 | merlin8282 | Can I keep my 1.4.22 installation anyway ? |
| 09:06.15 | JackTheNipple | yidiyuehan: welcome ;-) Maybe sometimes someone will be able to help me with my problem as well ;-) |
| 09:13.27 | *** join/#asterisk daelf (n=elf@adsl-99-130-196-193.dsl.ipltin.sbcglobal.net) |
| 09:13.39 | *** part/#asterisk daelf (n=elf@adsl-99-130-196-193.dsl.ipltin.sbcglobal.net) |
| 09:19.28 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 09:21.22 | yidiyuehan | Hi, Jack, Certainly I am willing if I know the solution:-) |
| 09:27.51 | *** join/#asterisk Borai (i=DYN@S0106001c109e98db.no.shawcable.net) |
| 09:28.20 | Borai | Hello, if i have a server that I want to run asterisk on but have no hardware cards on it do i still need the dahdi? |
| 09:34.03 | *** join/#asterisk drazhar007 (n=drazhar0@LPuteaux-156-16-53-67.w82-127.abo.wanadoo.fr) |
| 09:34.16 | drazhar007 | Hello |
| 09:34.56 | drazhar007 | I try to find the solution for this message : [Jul 3 09:30:38] WARNING[3759]: chan_iax2.c:7312 socket_process: Received trunked frame before first full voice frame |
| 09:35.43 | drazhar007 | I have make a post on the forum with my configuration and some network statistique. Somebody can help me ? Thanks |
| 09:40.14 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-207ed7853c5329b6) |
| 09:44.47 | *** join/#asterisk guaxinim (n=guaxinim@unaffiliated/guaxinim) |
| 09:49.04 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:ed7e:4424:4cd0:d836) |
| 09:52.43 | *** join/#asterisk thomas (i=tm@tm.muc.de) |
| 09:52.44 | thomas | hello. |
| 09:52.59 | thomas | have a extention: _59XXX <<but no match when i dial "59001728210852" |
| 09:53.05 | thomas | what is wrong here? |
| 09:54.30 | yidiyuehan | thomas, it should be _59XXX. |
| 09:54.35 | merlin8282 | you miss a dot: _59XXX. |
| 09:54.36 | yidiyuehan | there is a dot behind it |
| 09:54.50 | thomas | oh ok |
| 09:54.58 | thomas | emm what is "." ? |
| 09:55.24 | yidiyuehan | hi borai, you still need zaptel or dahdi even without any interface card, however you don't need to install the driver. |
| 09:56.01 | yidiyuehan | thomas, _59XXX. means anything with at least 5 digits and match first two as 59 |
| 10:01.15 | thomas | yidiyuehan: ah ok. thank you! :-) |
| 10:02.06 | *** join/#asterisk guaxinim (n=guaxinim@unaffiliated/guaxinim) |
| 10:04.46 | drazhar007 | I try to find the solution for this message : [Jul 3 09:30:38] WARNING[3759]: chan_iax2.c:7312 socket_process: Received trunked frame before first full voice frame |
| 10:12.24 | drazhar007 | In the same time the user complain of the telephony quality. |
| 10:18.03 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 10:24.27 | *** join/#asterisk guaxinim (n=guaxinim@unaffiliated/guaxinim) |
| 10:26.50 | thomas | emm, |
| 10:27.22 | thomas | a good web-gui for administrate asterisk, voipphones (snom with config (xmll).. easy create dialplans and macros.. ? |
| 10:27.25 | thomas | ? |
| 10:27.26 | thomas | only freepbx? |
| 10:27.30 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 10:32.30 | *** join/#asterisk SAT1 (i=515ab00d@gateway/web/freenode/x-148bed44c3144fde) |
| 10:32.44 | *** join/#asterisk Stdht (n=salfrede@212.98.187.129) |
| 10:32.45 | *** join/#asterisk hi365 (n=hi365@94.159.178.170) |
| 10:32.48 | Stdht | hI all |
| 10:33.17 | SAT1 | Hi all |
| 10:33.28 | SAT1 | i am getting this error while installing DADHI WARNING: could not find /usr/src/dahdi-trunk/drivers/dahdi/vpmadt032_loader/.vpmadt032_x86_32.o.cmd for /usr/src/dahdi-trunk/drivers/dahdi/vpmadt032_loader/vp madt032_x86_32.o |
| 10:33.34 | SAT1 | on CentOS |
| 10:33.35 | Stdht | Iam trying SIP -> Asterisk -> GSMgatway -> ASSA |
| 10:33.52 | Stdht | how to cacel echo |
| 10:35.39 | Zhad | Is anyone here using chan_mobile? |
| 10:36.47 | Stdht | I use chan_celliax |
| 10:37.36 | Zhad | coo. not seen that before, looks pretty good. |
| 10:38.00 | Zhad | shame the website is down. |
| 10:39.08 | iksik | damn, disconnected |
| 10:39.10 | iksik | any one have seen my question? |
| 10:39.42 | Zhad | was trying to get chan_mobile working again, but is getting no audio and calls not hanging up. |
| 10:39.55 | *** join/#asterisk lanman__ (n=lanman@173-102-30-248.pools.spcsdns.net) |
| 10:40.07 | iksik | <iksik> What could be a reason of dropping incoming connection with: "Number is not in service" ? |
| 10:40.11 | iksik | <iksik> i'm sure that extension exists... outgoing calls from this extension works fine, but i'm not sure if something is not missconfigured to handle incoming calls |
| 10:42.39 | Stdht | iksik try asterisk -rvvvvvddddddddddddddddd |
| 10:42.56 | Stdht | you''ll see debug |
| 10:43.21 | iksik | ok |
| 10:44.41 | SAT1 | russellb |
| 10:44.53 | Zhad | or core set debug 12 |
| 10:45.19 | iksik | Stdht http://pastebin.com/m646f0070 - can You tell me something about that ? |
| 10:48.50 | *** join/#asterisk Subdolus (i=dexterit@creep.bur.st) |
| 10:48.55 | thomas | a good web-gui for administrate asterisk, voipphones (snom with config (xmll).. easy create dialplans and macros.. ? |
| 10:48.58 | thomas | only freepbx? |
| 10:49.49 | Stdht | thomas asterisk-gui |
| 10:50.08 | *** join/#asterisk yang (i=yang@freenode/sponsor/cacert.assurer.yang) |
| 10:50.10 | iksik | trixbox ? |
| 10:50.26 | thomas | is trixbox better as freepbx? |
| 10:50.30 | Stdht | iksik could you pastebin your dialplan |
| 10:50.33 | thomas | iksik: but trixbox is a complete distri or? |
| 10:50.43 | iksik | yeap, centos based distro |
| 10:50.47 | thomas | mh ok |
| 10:50.50 | iksik | it includes freepbx afaik |
| 10:50.54 | thomas | then i think frepbx... |
| 10:51.04 | thomas | have a exist installed debian :/ |
| 10:51.22 | iksik | Stdht, can I display it from asterisk console? |
| 10:51.54 | iksik | For now I have no idea where that dialplan (in plain text) is... oO |
| 10:53.07 | Stdht | dialplan show |
| 10:53.42 | iksik | oh damn, it's a looooot of informations ;/ |
| 10:54.13 | *** join/#asterisk guaxinim (n=guaxinim@unaffiliated/guaxinim) |
| 10:55.43 | Stdht | etc/asterisk/extension.conf |
| 10:55.49 | Stdht | * etc/asterisk/extensions.conf |
| 10:56.20 | iksik | Stdht http://pastebin.com/m719d65ae |
| 10:57.02 | iksik | nope. there is no dialplans for this extension in /etc/asterisk/extensions.conf :) |
| 11:02.09 | Stdht | hm I do not understand.. but there no commands in dialplan like gotoif |
| 11:02.27 | *** join/#asterisk callenb (n=callenb@host86-168-62-181.range86-168.btcentralplus.com) |
| 11:03.00 | iksik | Stdht it's generated with some script I think... I'm setting it up via web interface |
| 11:03.29 | Stdht | that is why I do not use .. asterisk-gui for instance.. |
| 11:04.03 | Stdht | because I am newb too... |
| 11:04.06 | Stdht | ) |
| 11:05.10 | Stdht | GotoIf("SIP/arek-0a176138", "0?from-trunk|717180310|1") in new stack |
| 11:05.37 | Stdht | I think here call is rerouted to ss-noservice |
| 11:07.23 | iksik | uhm |
| 11:07.40 | iksik | any ideas how to fix this? :/ |
| 11:09.20 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 11:09.45 | *** join/#asterisk zeeesh (n=zeeesh@203.215.176.22) |
| 11:10.12 | Stdht | at his line some var is compared to 0 as I understand and ... if yes goes from-trunk|717180310|1 |
| 11:10.33 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 11:10.38 | Stdht | otherwise goes the next line |
| 11:11.24 | Stdht | unfortunatelly asterisk do not writes what variable .... |
| 11:11.36 | Stdht | is compared |
| 11:13.25 | *** join/#asterisk qdk (n=qdk@81.7.168.130) |
| 11:13.27 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 11:13.57 | iksik | mhm |
| 11:26.45 | *** join/#asterisk maxxer (n=maxxer@host-78-14-229-86.cust-adsl.tiscali.it) |
| 11:27.33 | maxxer | hi. i tried asterisk-gui branches/2.0 on debian (asterisk 1.4), but when I try to go to web config page says there's nothing |
| 11:29.51 | Zhad | It's a shame ther eisn't a softATA from symbian |
| 11:32.06 | Stdht | maxxer try inec again |
| 11:32.14 | Stdht | yesterday there was commit |
| 11:32.20 | Stdht | make update |
| 11:32.39 | maxxer | Stdht, inec? i checked out 5 minutes ago |
| 11:33.02 | maxxer | r4963 |
| 11:34.15 | Stdht | yes .. |
| 11:34.17 | maxxer | make checkconfig says everything ok |
| 11:34.32 | Stdht | asterisk/static/config/index.html |
| 11:34.39 | Stdht | ? |
| 11:34.51 | maxxer | Stdht, says not found |
| 11:35.01 | Stdht | hm |
| 11:35.14 | Stdht | http://yourhost:8088/asterisk/static/config/index.html |
| 11:35.38 | maxxer | Stdht, exactly |
| 11:35.54 | Stdht | hm. I am not at 4962 may be there new errors... |
| 11:36.05 | maxxer | what r are you? |
| 11:36.05 | Stdht | goto channel asterik-gui |
| 11:36.25 | Stdht | sorry mistake..I am at 4962 |
| 11:36.47 | maxxer | asterisk version? |
| 11:37.25 | Stdht | at asterisk-gui channel ..there is one an who can help you ... But I do not remember nick .. awkR.. or something like that |
| 11:37.32 | Stdht | >asterisk version? 1.4 |
| 11:38.31 | maxxer | so the same... |
| 11:39.05 | Stdht | I tried this morning make update .. but thre R is still 4962 |
| 11:39.18 | Stdht | on my asterisk.. I don;t wkonw why |
| 11:39.36 | Stdht | and it still working |
| 11:39.52 | Stdht | but I made decision not to use it |
| 11:39.57 | maxxer | why? |
| 11:40.01 | Stdht | I prefer direct extension.conf |
| 11:40.12 | maxxer | I was trying to use voiceone but had troubles with that as well |
| 11:40.18 | maxxer | also they only support asterisk 1.2 |
| 11:40.25 | Stdht | asterisk-gui bugzzz) |
| 11:40.36 | Stdht | but gui is lovely |
| 11:41.02 | Stdht | maxxer: where are you from? |
| 11:43.39 | maxxer | also, make samples doesn't work. is it right? |
| 11:43.43 | maxxer | Stdht, italy |
| 11:45.00 | *** join/#asterisk ramindia (n=balajibh@202.63.96.10) |
| 11:53.24 | *** part/#asterisk Stdht (n=salfrede@212.98.187.129) |
| 11:53.35 | iksik | huh |
| 11:53.46 | *** join/#asterisk AsteriskDom (n=amunoz@75.112.88.200.m.sta.codetel.net.do) |
| 11:54.10 | iksik | ok, I've got it... ALLOW_SIP_ANON = yes - fix the problem, then there is another one... why I can't get the incoming call ID from SIP :| |
| 11:56.02 | Pan3D | iksik: please provide DETAILS |
| 11:59.06 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 12:02.48 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:10.43 | *** join/#asterisk voipheroes (n=voiphero@AMontsouris-757-1-4-36.w92-128.abo.wanadoo.fr) |
| 12:16.36 | iksik | Pan3D I don't know how to set it up in freepbx, to pass this caller ID. It should be configured in trunk? |
| 12:22.39 | ramindia | Hi can some one tell me how can record voice ( when the calling card says you have 50$ balance) |
| 12:26.13 | *** join/#asterisk KingOfDos (n=irssi@ht1-ix.kingofdos.net) |
| 12:26.53 | *** part/#asterisk merlin8282 (n=merlin82@AStrasbourg-753-1-21-124.w90-56.abo.wanadoo.fr) |
| 12:34.22 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:34.22 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:40.33 | carrar | Y*A*W*N |
| 12:46.38 | leifmadsen | OMG VACATION! |
| 12:48.16 | *** join/#asterisk davevg-btwtech (n=davevg__@75.97.64.33.res-cmts.senj.ptd.net) |
| 12:48.28 | carrar | MG |
| 12:48.30 | carrar | O |
| 12:53.32 | carrar | Z |
| 13:01.21 | *** join/#asterisk Deeewayne (n=dwayne@76.29.245.9) |
| 13:01.21 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 13:03.39 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
| 13:03.48 | Zeeek | Friday, it's VUC day |
| 13:04.09 | Zeeek | TGIVUCD! |
| 13:05.31 | *** join/#asterisk gscmans (n=guna@94-170-141-94.cable.ubr16.haye.blueyonder.co.uk) |
| 13:06.55 | [TK]D-Fender | ramindia: huh? |
| 13:07.46 | Zeeek | Where's my girl? Eh? Where is she? |
| 13:07.59 | Zeeek | reflects 10 seconds on remaining PC |
| 13:08.41 | Zeeek | Two protocols walk into a bar, SIP and IAX2... |
| 13:09.06 | Zeeek | The bartender asks, "what'll it be, fellows?" |
| 13:09.32 | *** join/#asterisk devyll (n=email@89.36.24.2) |
| 13:09.42 | Zeeek | [TK]D-Fender : |
| 13:09.52 | Zeeek | The bartender asks, "what'll it be, fellows?" |
| 13:10.37 | devyll | can anybody tell me why do I get this warning message ? : format_ogg_vorbis.c:535 ogg_vorbis_seek: Seeking is not supported on OGG/Vorbis streams! . i use mixmonitor app to record every queue conversation, and I don't need to seek the recorded call. I'm a little confused. I would appreciate any help. |
| 13:10.40 | [TK]D-Fender | Zeeek: .... Don't quit your "day job" :p |
| 13:11.00 | Zeeek | I'm hooking up the drum sounds as we speak |
| 13:11.48 | [TK]D-Fender | devyll: Last I checked you can't write to a format you don't have a "codec" module for. the "format" modules are for playback only. |
| 13:11.53 | Zeeek | I guess you could continue: SIP says "tell me all the drinks you serve" and IAX2 says ... |
| 13:13.11 | *** join/#asterisk Heretic (n=BuRn@216.32.93.241) |
| 13:14.29 | *** join/#asterisk ruben23 (n=RPL@203.177.91.194) |
| 13:14.55 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 13:15.15 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 13:15.19 | Zeeek | {{{{Katty}}}} |
| 13:17.00 | devyll | [TK]D-Fender , but everything works. the call is recorded in .ogg format and I can listen to it. The only problem is the warning message which appears right when I hungup. |
| 13:18.05 | [TK]D-Fender | devyll: Hrm... no idea... |
| 13:19.09 | ruben23 | hi |
| 13:35.27 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 13:38.09 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 13:39.44 | Zhad | okay, a few small changes in audio.conf, still can't answer phone, there is a longer fdelay before deskphone rings and I get 1-way audio (yay), spooky thing is, the mobile handset has 2-way audio. |
| 13:47.58 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com) |
| 13:52.39 | *** join/#asterisk _Steve_ (n=Steve@unaffiliated/steve/x-520345) |
| 13:57.59 | sysreq | hi! i have a sangoma a200 (4 fxo ports) card in an asterisk 1.6.1.1 server, installed with the latest beta wanpipe drivers.. my problem seems to be with the ulaw codec on dahdi channels; playing wav files works as asterisk plays them as slin, but as soon as i try to play a ulaw file, or call a SIP user (they all use ulaw as their codec), i have no audio whatsoever |
| 13:58.59 | sysreq | calls between SIP users using the ulaw codec works fine |
| 13:59.43 | Zhad | do you get transcode.c errors in the console? |
| 13:59.45 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 14:00.29 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 14:02.24 | *** join/#asterisk amessina (n=amessina@2001:470:1f11:a4:20e:cff:fe01:d5ec) |
| 14:02.27 | sysreq | Zhad: no, the only thing i get from the console is channel.c setting time ticks (ie. "channel.c:2308 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second") |
| 14:02.57 | Zhad | thought it was worth asking, because I get transcode.c errors with 1.6.1.1 quite a bit. |
| 14:03.06 | Zhad | coupled with no audio. |
| 14:03.50 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
| 14:04.22 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 14:07.48 | sysreq | here's the cli output, if anyone wants to take a look at it: http://pastebin.ca/1483019 |
| 14:08.40 | *** join/#asterisk AndyML (n=AndyML@pool-173-49-143-205.phlapa.fios.verizon.net) |
| 14:15.51 | sysreq | in fact, i just tested and if i play another wav file.. it doesn't work either, only the first wav file works, then as soon as it switches to using ulaw.. it ends; no matter if you play wav or ulaw files beyond that point, you don't hear anything. |
| 14:20.57 | defswork | I have a sangoma a200 at home |
| 14:21.07 | defswork | not on dahdi though |
| 14:30.26 | *** join/#asterisk maxagaz (n=maxagaz@222.128.36.151) |
| 14:30.29 | *** join/#asterisk jicksta (n=jicksta@c-67-169-183-106.hsd1.ca.comcast.net) |
| 14:30.59 | sysreq | i'll try with 1.6.0.10 |
| 14:31.00 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 14:32.01 | *** part/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 14:32.09 | *** join/#asterisk maxagaz (n=maxagaz@222.128.36.151) |
| 14:35.45 | *** join/#asterisk wchance_work (n=wchance_@69.25.185.115) |
| 14:35.59 | Zeeek | conference today in 90: come by #voip-users-conference anytime and join us at 12 Noon EDT for the live conf |
| 14:36.09 | Zeeek | http://VUC.me |
| 14:36.33 | wchance_work | trying to figure out why my digium board is not working. Seems that after reboot the drivers did not load. When I do a lspci I see the board |
| 14:36.49 | wchance_work | what can I do next to check further using ZAPTEL |
| 14:37.30 | [TK]D-Fender | wchance_work: mAYBE YOU COULD SHOW US WHAT "NOT WORKING" MEANS & LOOKS LIKE... |
| 14:37.36 | [TK]D-Fender | Whhe caps-day! |
| 14:37.37 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 14:37.40 | [TK]D-Fender | Whee* |
| 14:37.41 | [TK]D-Fender | dangit |
| 14:37.49 | brah | "It's just you. http://downloads.digium.com is up." |
| 14:38.09 | brah | I'm raging here, trying to install * but some route seems to be down |
| 14:38.24 | wchance_work | I am using a TC400B card with Freeswitch and when I run FS it does not load because it does not find the dev |
| 14:38.30 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 14:38.41 | *** join/#asterisk darkskiez (n=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 14:38.52 | wchance_work | according to FS it is a driver issue |
| 14:39.17 | [TK]D-Fender | wchance_work: "ztcfg -vvvv" <- what does it give you? |
| 14:39.55 | wchance_work | line 0: Unable to open master device '/dev/zap/ctl' |
| 14:40.12 | defswork | moudles loaded ? |
| 14:40.16 | defswork | moudles ? |
| 14:40.22 | defswork | patents new word |
| 14:40.55 | wchance_work | how do I check if modules loaded ? |
| 14:41.28 | *** join/#asterisk moy (n=moy@74.12.123.90) |
| 14:41.59 | defswork | wchance_work: have you built/installed them ? |
| 14:42.16 | wchance_work | defswork oh yes it was working previously |
| 14:42.25 | defswork | previous to ? |
| 14:42.35 | wchance_work | previous to rebooting the server |
| 14:43.20 | defswork | you done a kernel upgrade by any chance ? |
| 14:43.34 | wchance_work | yum update (YES) |
| 14:45.26 | *** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk) |
| 14:45.37 | *** join/#asterisk moa_ (n=moa_@lab.vision.net) |
| 14:45.40 | Zhad | lo coppice |
| 14:45.51 | [TK]D-Fender | wchance_work: Then you need to recompile the kernel modules <- |
| 14:46.05 | wchance_work | ah ok |
| 14:46.23 | wchance_work | lets see if I can figure that out |
| 14:47.08 | wchance_work | I am in "/usr/src/zaptel-1.4.12.1" |
| 14:47.31 | wchance_work | that is where I recompilre right? |
| 14:47.42 | [TK]D-Fender | wchance_work: Same place you did it the first time |
| 14:49.07 | *** part/#asterisk callenb (n=callenb@host86-168-62-181.range86-168.btcentralplus.com) |
| 14:49.14 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 14:50.10 | Zeeek | how are we doing in here? |
| 14:50.16 | Zeeek | everything ok? |
| 14:51.12 | *** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 14:51.25 | *** join/#asterisk MrNaz (n=mrnaz@203.214.68.222) |
| 14:51.34 | *** join/#asterisk fred-tmft (n=fred-tea@c-69-244-180-112.hsd1.mi.comcast.net) |
| 14:52.12 | *** join/#asterisk outtolunc (n=me@c-98-248-96-110.hsd1.ca.comcast.net) |
| 14:53.25 | *** join/#asterisk fred-tmft (n=fred-tea@c-69-244-180-112.hsd1.mi.comcast.net) |
| 14:53.41 | wchance_work | thanks |
| 14:54.20 | *** join/#asterisk devyll (n=email@89.36.24.2) |
| 14:54.59 | *** part/#asterisk lipatden (n=ldennehy@sbpofw1.schubergphilis.com) |
| 14:55.28 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
| 14:55.56 | devyll | pri_dchannel: Ring requested on unconfigured channel 0/31 span 1 ?! who is responsable for the channel request ? I configured 1 to 30 channels . (16 is the D channel) |
| 14:58.02 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:ed7e:4424:4cd0:d836) |
| 14:59.44 | [TK]D-Fender | devyll: I'm not seeing your configs, and CLI status dumps for your channels |
| 14:59.46 | [TK]D-Fender | ~pb |
| 14:59.47 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/, or apt-get install pastebinit |
| 14:59.48 | [TK]D-Fender | ^^^^ |
| 14:59.48 | *** join/#asterisk ccesario_ (n=ccesario@189.88.2.216) |
| 15:06.33 | *** join/#asterisk ccesario_ (n=ccesario@189.88.2.216) |
| 15:08.04 | iksik | hummm |
| 15:10.23 | iksik | [Jun 28 11:27:32] dialparties.agi: Caller ID name is 'unknown' number is 'my_sip_peer_username' |
| 15:10.37 | iksik | any ideas why? |
| 15:10.43 | [TK]D-Fender | ~freepbx |
| 15:10.44 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 15:10.48 | *** join/#asterisk ManxPower (n=manxpowe@141.sub-70-223-111.myvzw.com) |
| 15:10.57 | iksik | o.O |
| 15:11.18 | iksik | ok |
| 15:12.35 | devyll | [TK]D-Fender , |
| 15:12.35 | devyll | "dahdi show channels" shows http://pastebin.com/m6785f045 "dahdi show status" shows http://pastebin.com/m2d00f7d "system.conf" shows http://pastebin.com/m166b032f |
| 15:12.35 | devyll | and from chan_dahdi.conf http://pastebin.com/m4e318cc5 |
| 15:14.15 | *** join/#asterisk ccesario_ (n=ccesario@189.88.2.216) |
| 15:14.36 | [TK]D-Fender | devyll: I'm worndering if span 1 = port1 with the other being port 0. |
| 15:14.54 | *** join/#asterisk |Rain| (i=rain@ev.il.net) |
| 15:15.20 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 15:15.57 | devyll | [TK]D-Fender : dahdi_scan shows: http://asterisk.pastebin.com/m2e4c1268 |
| 15:16.03 | |Rain| | is there a way to jump back to the start of a macro (gosub now, I guess) in AEL in 1.6? I want to restart a macro from within a catch{} block, and it was easy enough in 1.4... |
| 15:16.13 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 15:16.13 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:16.28 | [TK]D-Fender | |Rain|: Goto <--- |
| 15:16.30 | devyll | so, my understanding is that Span 1 is Port 0 |
| 15:16.33 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:17.04 | *** join/#asterisk dmz (n=dmz@64.203.233.195.dyn-cm-pool35.hargray.net) |
| 15:17.52 | devyll | [TK]D-Fender and i only have e1 cable in Port 0 , and only Span 1 has green led light, Port 1 has no cable and therefore Span 2 has RED alarm. |
| 15:19.16 | [TK]D-Fender | devyll: Hrm... does LOOK OK. |
| 15:19.29 | [TK]D-Fender | devyll: Restart * and see if it repeats |
| 15:20.13 | *** join/#asterisk ccesario (n=ccesario@189.88.2.216) |
| 15:22.18 | Joel | Qwell ping - questions for you about the rpms up on digium |
| 15:22.34 | Joel | Qwell what's the difference between asterisk16-1.6.0.10-1_centos5.i386.rpm and asterisk16-core-1.6.0.10-1_centos5.i386.rpm ? |
| 15:23.20 | *** join/#asterisk callenb (n=callenb@host86-168-62-181.range86-168.btcentralplus.com) |
| 15:23.24 | *** join/#asterisk ccesario (n=ccesario@189.88.2.216) |
| 15:24.03 | devyll | [TK]D-Fender , the error said: Ring requested on unconfigured channel 0/31 span 1. That happend when I tried calling from outside. My question is: who decides which free channel to choose when recieving a call ? And if it picks randomly somehow ? Because I could restart everything (already did that actually) and try calling 15 times with dahdi choosing 15 diffrent open channels. |
| 15:24.05 | devyll | That won't prove the problem is fixed |
| 15:25.05 | ramindia | [TK]D-Fender: Hi |
| 15:25.22 | [TK]D-Fender | devyll: I think I head of a different issue between who is using a 0-based vs 1-based port # sequence on the dchan... |
| 15:25.25 | *** join/#asterisk ccesario (n=ccesario@189.88.2.216) |
| 15:26.18 | Deeewayne | Joel, my bet is the non-core rpm is a metapackage |
| 15:26.52 | ramindia | [TK]D-Fender: how can i record Voice when i call calling card and send DTMF PIN and ( it says card balance is 50$)...i want to record this 50$ .. any suggestions |
| 15:27.13 | [TK]D-Fender | ramindia: How are you calling? |
| 15:27.55 | ramindia | [TK]D-Fender: iam calling from Asterisk Dial plan picking up from Database and sending the call to Calling card provider |
| 15:28.02 | *** join/#asterisk jerryeguru (n=Administ@41.222.2.65) |
| 15:28.10 | [TK]D-Fender | ramindia: "core show application monitor" <- |
| 15:28.49 | jerryeguru | i am trying to identify my interface card, when i lspci i see >>Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
| 15:29.24 | jerryeguru | what kind of card is this, digital or analog, E1...? |
| 15:30.08 | [TK]D-Fender | jerryeguru: How is it you don't know what card you have? |
| 15:30.43 | ramindia | [TK]D-Fender: how can i integrate same to my dial plan and store in a path and file |
| 15:30.49 | *** join/#asterisk ccesario (n=ccesario@189.88.2.216) |
| 15:30.55 | [TK]D-Fender | ramindia: "core show application monitor" <- |
| 15:31.55 | *** join/#asterisk ccesario (n=ccesario@189.88.2.216) |
| 15:32.08 | ramindia | [TK]D-Fender: yes i excuted command.. in the * cli.... how can i automate in the Dial plan record and hangup/// |
| 15:32.22 | jerryeguru | [TK]D-Fender: someone handed me this card to test and give back incase it doesnt suit my needs, there was absolutely no words to indentify the name or type of card |
| 15:32.38 | [TK]D-Fender | ramindia: I asked you how you CALL THEM. |
| 15:33.29 | [TK]D-Fender | jerryeguru: You sure he didn't ahnd you a loaded pistol? If you see something that looks like a loop for your finger Put it through and try pulling! |
| 15:33.49 | [TK]D-Fender | jerryeguru: And that should be a TDM400P, and it would depend on what modules are on it as to if it suits your needs |
| 15:36.04 | *** join/#asterisk decimalz (n=pbxk1064@203.171.196.209) |
| 15:36.13 | ramindia | [TK]D-Fender: its auto process.. using my own code of dialplan.. .. is i answered correct.. ? |
| 15:36.26 | jerryeguru | [TK]D-Fender: it didnt scare me so i thght it wldnt kill me either, u are saying this is TDM400P? and how come it is showing me Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
| 15:36.42 | [TK]D-Fender | ramindia: doesn't describe WHAT is initiating the call out. |
| 15:37.05 | [TK]D-Fender | jerryeguru: Because thats the chipset its based on. Do you have the card in hand? |
| 15:37.42 | ramindia | [TK]D-Fender: hmm then ? |
| 15:37.54 | *** join/#asterisk moa_ (n=moa_@lab.vision.net) |
| 15:37.58 | jerryeguru | [TK]D-Fender: i have put the card into the server but not setup/installed asterisk yet |
| 15:38.33 | [TK]D-Fender | jerryeguru: this card has 4 ports on the side of it for little daughter-card modules, correct? |
| 15:38.55 | jerryeguru | [TK]D-Fender: exactly |
| 15:39.19 | [TK]D-Fender | jerryeguru: Indeed a TDM400P type card. what COLOURS were the modules, and how many of the 4? |
| 15:39.21 | jerryeguru | [TK]D-Fender: i wonder if it is an E1 card though, i am not so well versed with asterisk |
| 15:39.39 | [TK]D-Fender | jerryeguru: It is not an E1 card, it is an ANALOG card |
| 15:40.34 | jerryeguru | <PROTECTED> |
| 15:40.48 | jerryeguru | so this is not what i am looking for :( |
| 15:40.50 | [TK]D-Fender | jerryeguru: Well that isn't an E1 card. |
| 15:41.22 | *** join/#asterisk |Rain| (i=rain@warped.bluecherry.net) |
| 15:42.07 | |Rain| | [TK]D-Fender: that's the logical answer, but goto says it can't find s|1 (even though it's visible in dialplah show if I remove the goto so that the AEL compiles) |
| 15:42.22 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 15:42.39 | *** join/#asterisk SlipperyChicken (n=andrew@LONDON14-1168107385.sdsl.bell.ca) |
| 15:43.30 | [TK]D-Fender | |Rain|: pastebin your code and the failed attempt, and a CLI dump of that dialplan context |
| 15:43.39 | [TK]D-Fender | |and quick, I'm off in 4 mins |
| 15:44.03 | jerryeguru | <PROTECTED> |
| 15:44.11 | *** join/#asterisk martyn-job (i=be18869a@gateway/web/freenode/x-41fbf818381de492) |
| 15:44.19 | martyn-job | Hi Asterisk Users... |
| 15:44.39 | [TK]D-Fender | jerryeguru: E1 is a digital trunk to the telco, analog is analog. you don't need one for the other |
| 15:44.44 | [TK]D-Fender | jerryeguru: Totally separate techs |
| 15:45.13 | [TK]D-Fender | jerryeguru: And you haven't confirmed the COLOUR of the modules on that card. RED is for pluggin in LINES, GREEN is for plugging in PHONES. |
| 15:45.23 | jerryeguru | <PROTECTED> |
| 15:45.27 | [TK]D-Fender | jerryeguru: If you have the wrong kind well... do the math |
| 15:45.35 | martyn-job | I have in my musiconhold random=yes and in my directory i have 10 gsm files ..but always play the same gsm file :S |
| 15:45.45 | [TK]D-Fender | jerryeguru: The telco doesn't care what you have on your line so long as it picks up, dials, etc |
| 15:45.50 | martyn-job | What can i do to play random...? |
| 15:46.21 | jerryeguru | <PROTECTED> |
| 15:46.31 | jerryeguru | <PROTECTED> |
| 15:46.34 | [TK]D-Fender | jerryeguru: DON'T |
| 15:47.00 | [TK]D-Fender | jerryeguru: Go confirmt he colour, because the wrong module will FRY if a call comes in and its not the right one. |
| 15:47.28 | [TK]D-Fender | Lunch time, BBIAB |
| 15:48.42 | *** part/#asterisk ManxPower (n=manxpowe@141.sub-70-223-111.myvzw.com) |
| 15:50.23 | *** join/#asterisk ccesario (n=ccesario@189.88.2.216) |
| 15:51.25 | jerryeguru | [TK]D-Fender: ok let me first identify the colors and revert back |
| 15:52.34 | *** join/#asterisk ccesario (n=ccesario@189.88.2.216) |
| 15:52.51 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com) |
| 15:54.15 | devyll | what does it mean when after a 6 label the next one is 102 .. (like here: http://asterisk.pastebin.com/m52af7464 ) . I'm trying to understand the Goto(7) after seting those variables and I can't seem to understand the logic behind that dialplan. |
| 15:54.42 | Joel | devyll huh? where do you see 102? |
| 15:54.54 | devyll | 103 |
| 15:54.57 | devyll | sorry Joel |
| 15:55.03 | Joel | 2 fails, so it adds 101 |
| 15:55.11 | devyll | ok, so in case of fail |
| 15:55.25 | devyll | why whouldn't you use 50 ? |
| 15:55.39 | devyll | instead of 103. or closer. |
| 15:55.40 | devyll | ? |
| 15:56.26 | jerryeguru | [TK]D-Fender: still here |
| 15:56.39 | *** join/#asterisk ccesario (n=ccesario@189.88.2.216) |
| 15:56.39 | devyll | you can see there . s,7,blabla1 and the next one s,103,SetVar(blabla). and .. I was trying to understand why "103" and not "8" for example. ?! |
| 15:56.58 | jerryeguru | [TK]D-Fender: there seem to be no colors displayed on the card modules, its hard to identify which is which! |
| 15:58.31 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 15:58.53 | ajmcello | i have nat=yes in my sip.conf and am able to register my phone and call extensions, but dtmf isn't working. anybody have any ideas? |
| 15:59.21 | ajmcello | the phone is on an inside nat network and the asterisk server is at a remote location on an internal ip |
| 15:59.33 | *** join/#asterisk hfb (n=hfb@pool-98-112-239-34.lsanca.dsl-w.verizon.net) |
| 15:59.51 | ajmcello | i opened up UDP 10000 to 20000 on the router side of the asterisk server but still have no luck |
| 16:01.53 | CryWolf | ajmcello: is "internal ip" a public or private ip? |
| 16:02.02 | drmessano | jerryeguru: The PC boards are all translucent? |
| 16:02.02 | ajmcello | private |
| 16:02.24 | CryWolf | ajmcello: So you have: phone - nat - internet - nat - pbx ? |
| 16:02.41 | drmessano | jerryeguru: What color are the BOARDS |
| 16:02.48 | ajmcello | crywolf: yup |
| 16:02.59 | CryWolf | ajmcello: Ouch. |
| 16:03.02 | ajmcello | haha. |
| 16:03.23 | drmessano | <PROTECTED> |
| 16:03.39 | carrar | what, no double natting? |
| 16:03.49 | ajmcello | everything works fine from the pbx inside nat network |
| 16:04.13 | ajmcello | it seemed to all work fine using asterisk 1.4.x |
| 16:04.34 | drmessano | I regularly run nat'ed phones to my boxes, which are all behind NAT |
| 16:05.40 | jerryeguru | drmessano: i wanted to indentify the module on the card, from some research off googling i am told module 1 is a green FXS module, and module 2 is an orange-red FXO module on the TDM400P card |
| 16:05.59 | jerryeguru | drmessano: got it here >> http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-CHP-4-SECT-4.html |
| 16:08.23 | drmessano | ok |
| 16:08.29 | ajmcello | oh maybe that is the problem |
| 16:08.34 | ajmcello | sip show peers shows port 1060 instead of 5060 |
| 16:08.35 | ajmcello | hrmmm |
| 16:09.44 | Zeeek | #voip-users-conference and http://VoipUsersConference.org with Twilio live |
| 16:09.48 | Zeeek | bye for now |
| 16:09.54 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
| 16:12.30 | *** part/#asterisk jerryeguru (n=Administ@41.222.2.65) |
| 16:16.26 | dwschool | i have two modem devices, i want to be able to choose on which device faxmail sends out a fax - is this possible? |
| 16:16.30 | |Rain| | [TK]D-Fender (or anyone else): http://www.themuffin.net/j/tmp/gotofail |
| 16:20.12 | *** join/#asterisk ccesario_ (n=ccesario@189.88.2.216) |
| 16:22.09 | *** join/#asterisk phurl (n=mdupont@82.114.94.9) |
| 16:23.10 | *** part/#asterisk phurl (n=mdupont@82.114.94.9) |
| 16:27.08 | *** join/#asterisk martyndev (i=be18869a@gateway/web/freenode/x-9eea3f691fbff186) |
| 16:27.15 | martyndev | Hi . :D I have in my musiconhold random=yes and in my directory i have 10 gsm files ..but always play the same gsm file :S |
| 16:28.24 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com) |
| 16:29.04 | martyndev | hey, sorry, now the problem is solved :D |
| 16:29.12 | martyndev | thanks. just moh reload and it's works.. |
| 16:29.50 | martyndev | i dont know why with reload res_musiconhold.so dont work.-. |
| 16:29.57 | martyndev | but, anyway, thank you ;) |
| 16:32.10 | *** join/#asterisk JonCup (n=JonCup@72.34.90.74) |
| 16:32.50 | *** join/#asterisk asaleem (n=asaleem@modemcable210.78-70-69.static.videotron.ca) |
| 16:33.06 | asaleem | I recording calls from queue. It works fine but at the end I have all A's part of conversation and when it is finished, then B's part. Is that normal |
| 16:33.19 | asaleem | I am using AsteriskNOW |
| 16:33.45 | Joel | sounds like a bad mix |
| 16:34.23 | *** part/#asterisk martyndev (i=be18869a@gateway/web/freenode/x-9eea3f691fbff186) |
| 16:34.29 | asaleem | Joel, so it is not the normal behavior |
| 16:34.50 | asaleem | The system used sox with m |
| 16:34.53 | asaleem | to mix |
| 16:35.53 | *** join/#asterisk gscmans (n=guna@94-170-141-94.cable.ubr16.haye.blueyonder.co.uk) |
| 16:36.09 | *** join/#asterisk ramindia (n=balajibh@202.63.96.10) |
| 16:36.24 | ramindia | [TK]D-Fender: i lost connection |
| 16:39.00 | ramindia | regarding recording you are suggesting something |
| 16:40.28 | asaleem | Is there any way I can fix that? |
| 16:43.25 | |Rain| | is sox installed and working? asterisk isn't very noisy if the mix fails for some reason |
| 16:44.00 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 16:47.21 | *** join/#asterisk Shazaum (n=shazaum@unaffiliated/shazaum) |
| 16:48.20 | Shazaum | hi |
| 16:48.33 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 16:49.44 | Shazaum | is possible set acl "permit" in configuration default of sip? |
| 16:53.33 | *** join/#asterisk rob0 (n=rob0@cardinal.lizella.net) |
| 16:56.01 | *** join/#asterisk bongy (n=sooka@93-42-53-123.ip85.fastwebnet.it) |
| 17:00.43 | |Rain| | allow=0.0.0.0/0? |
| 17:01.31 | |Rain| | make that permit= |
| 17:01.31 | asaleem | |Rain|, yes, sox is installed, as it joins the in-out |
| 17:01.32 | Shazaum | |Rain|: u sure? |
| 17:02.06 | Shazaum | this is a peers? or type a call from guest? |
| 17:04.27 | |Rain| | Shazaum: nope! I'm pretty sure there's no ACL by default, though |
| 17:04.36 | |Rain| | asaleem: oh, so it concatenated the inside and outside halves instead of mixing them? |
| 17:05.16 | asaleem | |Rain|, exactly |
| 17:05.23 | Shazaum | |Rain|: ok |
| 17:05.33 | asaleem | |Rain|, how do I resolve that? |
| 17:07.57 | devyll | where is the apps directory defined ? |
| 17:07.58 | bongy | Hi *, someone know if for default the SIP connections from the same host where asterisk is installed are forbidden ? |
| 17:08.01 | |Rain| | good question, I've never heard of that happening before. what version of sox? |
| 17:09.31 | ramindia | [TK]D-Fender: are u around |
| 17:11.16 | JonCup | any body have any expierience with grandstream ata 502 or simular, ive got one setup, incoming calls ring in, but I get one way audio, they can hear me, but I cant hear them, AND i cant make any outgoing calls |
| 17:12.58 | *** join/#asterisk vegbox (n=kevinle@24-205-54-7.dhcp.gldl.ca.charter.com) |
| 17:16.04 | *** join/#asterisk tobias (n=tobias@201.sub-97-143-183.myvzw.com) |
| 17:16.04 | *** part/#asterisk callenb (n=callenb@host86-168-62-181.range86-168.btcentralplus.com) |
| 17:16.17 | [TK]D-Fender | back |
| 17:19.20 | ramindia | [TK]D-Fender: hi |
| 17:19.23 | devyll | I'm trying to install spandsp with Asterisk but I can't seem to find the Asterisk apps folder. More then that there is no "asterisk" folder in /usr/src/ . also no apps folder in any asterisk directory. Can anybody help me understand what I'm doing wrong ? |
| 17:19.57 | [TK]D-Fender | devyll: Well where DID you extract your source? |
| 17:20.22 | ramindia | [TK]D-Fender: i lost connection.. u were suggesting me for that recordings |
| 17:20.47 | [TK]D-Fender | ramindia: You call MONITOR before you dial and it will record the call. |
| 17:22.35 | ramindia | [TK]D-Fender: let me try that |
| 17:23.37 | devyll | [TK]D-Fender Asterisk is installed from a pre-built package (i'm tied to that package, can't use anything else.. company policy) .. |
| 17:24.12 | [TK]D-Fender | devyll: Go look for a prepackaged set of apps then. |
| 17:24.23 | [TK]D-Fender | devyll: And good luck with that |
| 17:24.47 | devyll | [TK]D-Fender , ok . got the ideea. thanks. |
| 17:33.12 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com) |
| 17:37.44 | JonCup | Hey guys, need help with this grandstream handytone 502, i cannot make any outgoing calls, I just get a fast busy, and I dont see anything on the CLI, and on incoming calls, i I cant hear anything, but the caller can hear me, any ideas here anything anybody? |
| 17:38.07 | *** join/#asterisk ccesario_ (n=ccesario@189-19-6-236.dsl.telesp.net.br) |
| 17:38.49 | JonCup | server and ATA are on the same LAN |
| 17:41.45 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:46.44 | ramindia | [TK]D-Fender: i have used "Monitor(wav,${CALLFILENAME},m)" ......... is this ends after the call hangups ? |
| 17:47.29 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 17:50.26 | *** part/#asterisk SAT1 (i=515ab00d@gateway/web/freenode/x-148bed44c3144fde) |
| 18:00.25 | *** join/#asterisk qdk (n=qdk@81.7.168.130) |
| 18:02.02 | ramindia | how can convert Speech conversation of wav file to Text ? |
| 18:03.35 | joako | There is lumenvox... |
| 18:04.16 | KavanS | ramindia, use a hangup at the end |
| 18:04.25 | KavanS | and monitor will end |
| 18:04.54 | ramindia | KavanS: if i dont mention at the end it keep recording the same file ? |
| 18:05.25 | ramindia | joako: any open source to test.. not for commercial use |
| 18:08.10 | joako | Best I know is they offer a single channel license for $50 or $99 |
| 18:08.50 | ramindia | k |
| 18:10.14 | *** join/#asterisk cusco_ (n=pcmedic@213.63.137.210) |
| 18:10.15 | cusco_ | hi |
| 18:10.48 | cusco_ | is there a compilling option that makes somehow asterisk go extra verbose |
| 18:11.12 | cusco_ | so we know for example the source files that are doing what and the lines in the source files etc |
| 18:11.19 | *** join/#asterisk Corydon76-dig (i=black@pdpc/supporter/bronze/Corydon76-home) |
| 18:11.19 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 18:11.33 | *** join/#asterisk jeff (i=jeff@unaffiliated/jeff) |
| 18:11.45 | Corydon76-dig | cusco_: no, but there is "core set debug 15" |
| 18:11.56 | cusco_ | I know that |
| 18:12.07 | cusco_ | tho that does not tell me when asterisk process is just "killed" |
| 18:12.13 | Corydon76-dig | cusco_: and "core set verbose 15" |
| 18:12.40 | Corydon76-dig | Heh, if the Asterisk process is killed, then it received a signal that it could not ignore |
| 18:12.41 | cusco_ | any other extra verbose options you may indicate? |
| 18:12.56 | Corydon76-dig | cusco_: So the Asterisk process is showing "Killed"? |
| 18:13.02 | cusco_ | not now |
| 18:13.03 | cusco_ | it happened |
| 18:13.15 | Corydon76-dig | cusco_: How much memory do you have in the machine? |
| 18:13.18 | cusco_ | we are having one queue only that stops reaching extentions |
| 18:13.21 | cusco_ | let me check |
| 18:13.28 | WindowsUser | check /var/log/messages it'll tell you if it gets killed due to sucking up all the rams |
| 18:13.39 | cusco_ | 2 gigs |
| 18:13.40 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 18:13.43 | cusco_ | only running asterisk |
| 18:13.48 | cusco_ | and mysql |
| 18:13.55 | Corydon76-dig | cusco_: Likely a memory leak, then. Are you running the latest version? |
| 18:14.06 | cusco_ | 1.6.1.1 |
| 18:14.33 | Corydon76-dig | cusco_: in make menuselect, Compiler Options, turn on MALLOC_DEBUG |
| 18:14.38 | cusco_ | what could cause one queue only to stop reaching the extentions, when all other go thru |
| 18:14.43 | cusco_ | ok |
| 18:15.04 | cusco_ | is that in ./configure (if I do not use menuselect) |
| 18:15.14 | Corydon76-dig | cusco_: turns on memory tracking with 'core show memory allocations' and 'core show memory summary' |
| 18:15.18 | Corydon76-dig | It is only in menuselect |
| 18:16.19 | cusco_ | ok let me check |
| 18:16.49 | Corydon76-dig | cusco_: how are you connecting to mysql? |
| 18:17.08 | Corydon76-dig | cusco_: res_config_mysql or res_config_odbc? |
| 18:17.27 | cusco_ | mysql |
| 18:17.45 | cusco_ | im in menuselect for the first time |
| 18:17.45 | Corydon76-dig | cusco_: There's a deadlock in the current release of res_config_mysql for 1.6.1 |
| 18:18.00 | cusco_ | really? :-| |
| 18:18.05 | cusco_ | is it documented? |
| 18:18.13 | cusco_ | anywhere on the web? |
| 18:18.19 | *** part/#asterisk Shazaum (n=shazaum@unaffiliated/shazaum) |
| 18:18.20 | Corydon76-dig | cusco_: yeah, there's a release candidate for addons-1.6.1.1 about to go out on Monday |
| 18:18.38 | Corydon76-dig | cusco_: The fix is already in SVN... You'd just need to do a checkout |
| 18:18.41 | cusco_ | so I could take note and pass it on the the propper local technitian |
| 18:18.55 | cusco_ | we are in a production place |
| 18:19.02 | cusco_ | I don't think svn is recommended |
| 18:19.14 | Corydon76-dig | cusco_: if you like, you can checkout the patch directly from SVN and apply it |
| 18:19.17 | cusco_ | my boss says that 1.4 was stable comparing to 1.6 |
| 18:19.22 | Corydon76-dig | cusco_: SVN branch, not SVN trunk |
| 18:19.28 | *** join/#asterisk mykhyggz (n=mykhyggz@evolone.org) |
| 18:19.39 | |Rain| | is it possible to jump to the start of an AEL macro in 1.6.1.1? I want to restart a macro in a catch{} block, but I haven't been able to make the parser cooperate: http://www.themuffin.net/j/tmp/gotofail |
| 18:19.43 | cusco_ | (sorry but what is the difference bwtween branch and trunk) |
| 18:20.05 | Corydon76-dig | cusco_: trunk is the bleeding edge. A branch is the location from which releases are tagged |
| 18:20.11 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 18:20.21 | cusco_ | so in branch there are the releases candidates etc |
| 18:20.40 | Corydon76-dig | From the branch, release candidates are made, yes |
| 18:20.53 | Corydon76-dig | and the final release candidate is the release |
| 18:20.58 | cusco_ | ok |
| 18:21.01 | cusco_ | are they stable? |
| 18:21.16 | Corydon76-dig | Yes, especially when you're talking about -addons |
| 18:21.24 | Corydon76-dig | Addons hardly ever changes |
| 18:22.26 | cusco_ | can you point me to a place on teh web that talks about the mysql deadlock? |
| 18:23.21 | cusco_ | and where is that mallock option in menuselect? |
| 18:23.25 | Corydon76-dig | https://issues.asterisk.org/view.php?id=15023 |
| 18:23.36 | Corydon76-dig | cusco_: MALLOC_DEBUG is under Compiler Options |
| 18:23.47 | [TK]D-Fender | |Rain|: In your code it looks like everything in Catch 1 is CoMMENTEd OUT |
| 18:24.05 | |Rain| | that's because it won't compile to get the dialplan show... unless I comment it |
| 18:24.13 | Corydon76-dig | cusco_: "Compiler Flags - Development" |
| 18:24.16 | *** join/#asterisk ccesario_ (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 18:25.08 | cusco_ | thanks a lot for your tips Corydon76-dig |
| 18:25.16 | [TK]D-Fender | |Rain|: Well if its commented out then you haven't DONE anything |
| 18:25.23 | Corydon76-dig | cusco_: You can get the patch with this command: svn diff -c913 http://svn.asterisk.org/svn/asterisk-addons |
| 18:25.29 | |Rain| | the compile errors are after each line |
| 18:25.31 | cusco_ | My boss wants me to reproduce them problems before I update anything |
| 18:25.51 | cusco_ | ok let me check that too |
| 18:26.18 | dwschool | anyone here running hylafax? |
| 18:26.21 | cusco_ | Modified: trunk-merged |
| 18:26.24 | cusco_ | thanks Corydon76-dig |
| 18:27.17 | cusco_ | there is a error now building asterisk-addons lol |
| 18:28.15 | cusco_ | http://pastebin.com/m12464089 |
| 18:28.33 | cusco_ | with res_config_mysql |
| 18:29.16 | *** join/#asterisk cesar_CR (i=cesar@celord.ice.co.cr) |
| 18:29.26 | Corydon76-dig | cusco_: Dunno why that would be, if you're running 1.6.1 |
| 18:29.40 | Corydon76-dig | cusco_: perhaps you were mistaken about the version you're running? |
| 18:29.57 | Corydon76-dig | cusco_: or at least about the version that is installed? |
| 18:30.29 | cusco_ | I will download again asterisk-addons |
| 18:30.31 | cusco_ | hold |
| 18:30.33 | Corydon76-dig | asterisk -rx 'core show version' |
| 18:31.04 | *** join/#asterisk cesar_CR (n=cesar@201.199.168.170) |
| 18:31.23 | cusco_ | Asterisk 1.6.1.1 built by root @ perfpbxr on a x86_64 running Linux on 2009-06-15 15:48:54 UTC |
| 18:31.30 | *** join/#asterisk cesar_CR (i=cesar@celord.ice.co.cr) |
| 18:31.38 | Corydon76-dig | cusco_: Ah, appears you downloaded the wrong version of addons |
| 18:31.54 | cusco_ | could be, hold |
| 18:32.01 | Corydon76-dig | Probably trunk or 1.6.2, not 1.6.1 |
| 18:32.09 | cusco_ | (it was not installed by me, rather by my boss) |
| 18:32.25 | cusco_ | that means that that mysql thingie should be fix |
| 18:32.27 | *** join/#asterisk cesar_CR (i=cesar@celord.ice.co.cr) |
| 18:32.33 | cusco_ | fixed? |
| 18:33.15 | Corydon76-dig | The issue is in the released version of 1.6.1 |
| 18:33.31 | cusco_ | ok I qill have to ask my boss about that |
| 18:33.40 | cusco_ | thank you once again Corydon76-dig |
| 18:33.43 | Corydon76-dig | There are two possible ways around: install the patch or use res_config_odbc |
| 18:33.57 | cusco_ | I will patch it up later |
| 18:34.43 | cusco_ | do you reckon that one queue only stops getting to our softphones, could be related with that mysql issue? |
| 18:35.18 | Corydon76-dig | It's possible |
| 18:35.43 | cusco_ | ok mate |
| 18:35.48 | cusco_ | I have to go now |
| 18:36.08 | cusco_ | I will be bugging you guys later. Thank you for your tips |
| 18:38.50 | *** part/#asterisk fred-tmft (n=fred-tea@c-69-244-180-112.hsd1.mi.comcast.net) |
| 18:40.47 | *** join/#asterisk JimVanM (n=jimvanm@bas1-toronto63-1088800059.dsl.bell.ca) |
| 18:41.34 | *** join/#asterisk ingenius (n=alektro@host50.190-229-165.telecom.net.ar) |
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| 18:43.13 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 18:43.57 | JimVanM | question about EVAL() and variable names within variables |
| 18:44.08 | JimVanM | I have a base variable name VALUE_ |
| 18:44.17 | JimVanM | I want to iterate through several of them |
| 18:44.27 | JimVanM | so VALUE_1, VALUE_2, VALUE_3, etc |
| 18:44.40 | JimVanM | the actual integer will be contained in another variable |
| 18:44.41 | JimVanM | so |
| 18:44.48 | JimVanM | I'm trying to refer to something like |
| 18:45.07 | JimVanM | ${VALUE_${digit}} |
| 18:45.12 | JimVanM | naturally this doesn't work |
| 18:45.27 | JimVanM | I've been messing around with EVAL() |
| 18:45.32 | JimVanM | and quotes of all kinds |
| 18:45.38 | JimVanM | and can't get the syntaxt right |
| 18:45.48 | JimVanM | any advice would be appreciated |
| 18:47.11 | Corydon76-dig | JimVanM: AEL or dialplan? |
| 18:47.24 | JimVanM | dialplan |
| 18:47.35 | Corydon76-dig | Then what you have should work fine |
| 18:47.44 | Corydon76-dig | ${foo_${n}} |
| 18:48.50 | Corydon76-dig | EVAL doesn't come into it unless you have the literal characters '${' WITHIN a variable |
| 18:49.34 | Corydon76-dig | EVAL allows you to evaluate the CONTENTS of a variable |
| 18:50.10 | Corydon76-dig | normally the dialplan only evaluates literal variables |
| 18:50.48 | [TK]D-Fender | |Rain|: "A catch block can be specified to catch special extensions. " <- you are trying "catch" against a boring number. I suspect this is illegal like that instruction implied |
| 18:51.01 | [TK]D-Fender | |Rain|: Yuo do NOR make IVR's in macros |
| 18:51.03 | [TK]D-Fender | NOT* |
| 18:51.37 | JimVanM | Corydon76-dig: OK, thanks for that. I think I may have a logic problem somewhere else. |
| 18:53.15 | |Rain| | [TK]D-Fender: I wasn't, originally... I originally had this stuff in a gosub, but 1.6 whined about it so I tried converting it to a macro |
| 18:53.19 | JimVanM | Corydon76-dig: found it. I wasn't assigning the value properly, so the variable wasn't even getting created |
| 18:53.33 | Corydon76-dig | JimVanM: EVAL is all explained in TFOT... you should read it sometime. ;-) |
| 18:53.35 | JimVanM | Corydon76-dig: keyboard-chair interface problem |
| 18:53.39 | [TK]D-Fender | |Rain|: And figured you could be the rules? |
| 18:53.49 | JimVanM | Corydon76-dig: LOL! touche |
| 18:54.00 | [TK]D-Fender | ~osmosis |
| 18:54.01 | infobot | [~osmosis] Osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ... |
| 18:54.03 | [TK]D-Fender | JimVanM: ;) |
| 18:54.21 | |Rain| | [TK]D-Fender: it wouldn't be the first time (and I suspect it would work if I did it in extensions.conf) |
| 18:54.25 | [TK]D-Fender | .. Just sayin'! |
| 18:54.54 | JimVanM | [TK]D-Fender: oh believe me, that book has caused me my fair share of head trauma |
| 18:54.57 | |Rain| | I guess I'll just go back to a regular context with gosub and ignore the AEL parser's whining |
| 18:55.40 | [TK]D-Fender | |Rain|: AEL = means wel, but can only do less that you can do yourself in dialplan and creates extra greif while saving little |
| 18:55.54 | JimVanM | Corydon76-dig: the first (and only) programming language I ever learned really well was REXX |
| 18:56.16 | JimVanM | Corydon76-dig: if dialplan is a language (and I tend to think it is), then it is the second |
| 18:56.25 | JimVanM | Corydon76-dig: so I'm sorta damaged goods |
| 18:57.23 | [TK]D-Fender | puts JimVanM up for sale on eBay with an "as-is / caveat emptor" label |
| 18:57.30 | |Rain| | oh well, thanks |
| 18:57.57 | JimVanM | [TK]D-Fender: make sure it includes a picture that clearly shows the damage |
| 18:58.35 | JimVanM | [TK]D-Fender: http://smartmortgageadvice.files.wordpress.com/2007/08/head-up-ass.jpg |
| 18:58.38 | [TK]D-Fender | "Cat scan not included" |
| 18:59.31 | coppice | is a PET scan a more generic form of cat scan? :-\ |
| 19:02.19 | [TK]D-Fender | POSSIBLY |
| 19:18.16 | *** join/#asterisk ingenius (n=alektro@host50.190-229-165.telecom.net.ar) |
| 19:23.41 | *** join/#asterisk s0lid (n=s0lid@122.53.99.2) |
| 19:42.44 | jaytee | actually a PET scan uses radioactive tracers to map active regions of the brain using a similar tomographic technique of mapping 2D slices into a 3D image. CAT is just 2D xray slices composed into a 3D image but doesn't register active vs inactive regions of the brain. |
| 19:43.03 | *** join/#asterisk The_TiK (n=Seymour@76.30.103.42) |
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| 19:44.29 | The_TiK | is there a way to set the volume for cepstral using the swift.conf file or another way? I am trying using the ssml tags but it is not working |
| 20:04.17 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 20:10.10 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
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| 20:14.35 | *** join/#asterisk Stdht (n=salfrede@212.98.187.129) |
| 20:14.41 | Stdht | Hi all |
| 20:14.47 | Stdht | Where could I find asterisk svn |
| 20:15.05 | *** join/#asterisk pulpster (n=pulpster@p16.eregie.pub.ro) |
| 20:15.16 | *** join/#asterisk callenb (n=callenb@host86-168-62-181.range86-168.btcentralplus.com) |
| 20:15.29 | Stdht | May be anyone tried celliax under asterisk 1.6 also?? |
| 20:16.23 | [TK]D-Fender | Stdht: www.asterisk.org |
| 20:17.46 | *** join/#asterisk mchou (n=quassel@unaffiliated/mchou) |
| 20:18.38 | Stdht | http://www.asterisk.org/developers/get-source |
| 20:18.48 | mchou | anyone here notice that google voice now seems to accept DTMF "1" from * now on inbound calls? |
| 20:18.51 | Stdht | thanks |
| 20:19.25 | mchou | I dunno what happened but it now all seems to work |
| 20:19.46 | *** join/#asterisk degrade (n=degrade@unaffiliated/degrade) |
| 20:20.28 | degrade | I'm trying Asterisk with a 128 thread server. |
| 20:22.59 | Stdht | celliax??? is there anybody out there?) |
| 20:25.08 | dwschool | WARNING[14411]: chan_iax2.c:2309 __attempt_transmit: Max retries exceeded to host 127.0.0.1 on IAX2/school_iaxmodem-6837 (type = 6, subclass = 11, ts=50379, seqno=15) |
| 20:25.13 | dwschool | what does this mean? |
| 20:26.35 | [TK]D-Fender | checkout time, BBIAB |
| 20:28.13 | *** join/#asterisk hamush (n=hamush@24-155-177-34.dyn.grandenetworks.net) |
| 20:29.05 | asaleem | I recording calls from queue. It works fine but at the end I have all A's part of conversation and when it is finished, then B's part. Is that normal |
| 20:29.50 | WindowsUser | theres an M flag for the monitor app that triggers a merge when the call is complete |
| 20:29.58 | WindowsUser | lower case m i think |
| 20:31.55 | *** join/#asterisk IPPBX-ARG (n=pirruar@200-112-139-7.bbt.net.ar) |
| 20:32.05 | IPPBX-ARG | helloo |
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| 20:33.09 | *** part/#asterisk IPPBX-ARG (n=pirruar@200-112-139-7.bbt.net.ar) |
| 20:34.36 | hamush | does anyone know of a nice and verbose sip phone for linux that can be used for testing asterisk sip setup? |
| 20:37.27 | WindowsUser | ekiga + wireshark :) |
| 20:38.46 | hamush | that's a good idea... i've been frowning in ekiga's general direction because of its lack of verbosity, but maybe if i throw wireshark up, too, i can get an idea of what's going on |
| 20:42.58 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 20:44.58 | dwschool | for some reason, my iax modems stop sending fax signal, like when i call them sometimes they pick up right away with a fax tone - but other times i get nothing and cannot use the fax |
| 20:45.40 | dwschool | has anyone else experienced this? |
| 20:46.57 | dwschool | after a restart everything is fine again |
| 20:52.18 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 20:55.08 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 20:58.07 | hamush | i am following the o'reilly book |
| 20:58.56 | WindowsUser | I'm going to ~ask and then run away :) |
| 20:58.58 | WindowsUser | ~ask |
| 20:58.58 | 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. |
| 21:00.06 | hamush | 's echo test example, but i can't get the echo test to answer... does anyone know a good way that i tell if i'm experiencing a nat issue or asterisk config issue? (short of getting rid of nat, which isn't an option) |
| 21:01.11 | hamush | those were supposed to be one message, but i have butter fingers |
| 21:11.38 | Joel | does anyone have a nifty tool to take dialplan and turn it into something pretty? like a graph? |
| 21:13.32 | joako | Joel: asterisk -rx "show dialplan" |
| 21:16.51 | Joel | joako that's a graph to you? O.o |
| 21:16.54 | *** join/#asterisk terr_ (n=terr_@dsl-vlan435-66-18-218-43.nucleus.com) |
| 21:17.06 | Stdht | How Could I get * version from CLI? |
| 21:17.27 | [TK]D-Fender | Stdht: "show version" |
| 21:17.40 | Stdht | do not work |
| 21:17.45 | Stdht | I am on >1.4 |
| 21:17.47 | terr_ | I'm considering installing asterisk. Can someone breif me on what it can do relative to SKYPE or commercial VoIP products like something offered by www.magicjack.com |
| 21:17.48 | jaytee | Joel, check out http://projects.abourget.net/astograph |
| 21:18.09 | Joel | jaytee <3 |
| 21:18.26 | ISO9001 | terr_: what do you want it to do? heh. |
| 21:18.41 | [TK]D-Fender | terr_: Not comperable. Skype is a SOFT-PHONE. MajicJack is the same, and specifically tied to their service. * is a PBX & telephony TOOLKIT |
| 21:18.51 | JonCup | Hey guys, need help with this grandstream handytone 502, i cannot make any outgoing calls, I just get a fast busy, and I dont see anything on the CLI, and on incoming calls, i I cant hear anything, but the caller can hear me, any ideas here anything anybody? |
| 21:19.12 | jaytee | Joel, it's not that fancy. just shows relationships between contexts. you might also google Visual Dialplan but I'm not sure if that's open source or not |
| 21:20.02 | joako | Joel: It's "pretty" imo |
| 21:20.54 | terr_ | [TK]D-Fender: yes I know what asterix is. |
| 21:21.17 | *** join/#asterisk tamiel (n=tamiel@ip-25.net-89-2-117.rev.numericable.fr) |
| 21:21.27 | [TK]D-Fender | terr_: Then why are you asking for a comparison between apples & oranges? |
| 21:21.46 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 21:21.51 | terr_ | [TK]D-Fender: I want to set up VoIP for my home office & mobile office. I'll want to interface somehow to POTS in order to allow people to dial in. I'll also probably want an 800 number if possible |
| 21:21.55 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 21:21.57 | [TK]D-Fender | And I know what "asterix" is too... a CARTOON |
| 21:22.24 | terr_ | [TK]D-Fender: because they are both fruit. For $20 per year I have VoIP with magicJack. I also get a 1-408 number if I like |
| 21:22.34 | [TK]D-Fender | terr_: So... can Skype or MajigJack take in your analog to do something rpductive? |
| 21:22.43 | terr_ | [TK]D-Fender: nope. |
| 21:23.27 | terr_ | Its a interface via USB and this means we need to install a deamon in order to handle it. Asterisk will allow me to actually plug in an analog or digital phone. Asterik is a PBX |
| 21:23.41 | [TK]D-Fender | if they can't do what you want, what is the point of comparing with them? |
| 21:23.48 | terr_ | [TK]D-Fender: so now way out of asterisk into POTS? |
| 21:23.50 | Joel | joako might want to look up the definition of a graph! |
| 21:24.00 | [TK]D-Fender | terr_: a TDM card |
| 21:24.30 | terr_ | [TK]D-Fender: TDM - then what? I call my telco and ask for a PRI interface to their system? |
| 21:25.13 | [TK]D-Fender | terr_: I said TDM expecting your needs to be small. maybe you should define those first |
| 21:25.48 | terr_ | [TK]D-Fender: I _might_ look at becomming a full ISP. Currently I do the web hosting part |
| 21:26.52 | [TK]D-Fender | terr_: before dreaming of that go look at who allows you to RESELL their service |
| 21:26.58 | terr_ | [TK]D-Fender: so I'm looking into VoIP as well. At this point - I've know about asterisk for quite a while but I've not fully researched how it can be interfaced to the POTS system. I know if I set up two asterisk boxes I can have free calls between these boxes regardless where on the net they might be located. |
| 21:27.13 | [TK]D-Fender | terr_: As forupstream interface, yes, PRI is the entry point |
| 21:27.25 | terr_ | [TK]D-Fender: my Telco is required to do this. But its expensive. |
| 21:28.40 | terr_ | [TK]D-Fender: what I think I'm reading from you is that if I have two businesses which might have offices at a distance from each other - then installing a pair of asterisk servers is a perfect fit for them. |
| 21:29.05 | [TK]D-Fender | terr_: Sure... |
| 21:29.45 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com) |
| 21:29.58 | terr_ | [TK]D-Fender: what I think I'm also reading is that for a simple home office wanting a local number that something like MagicJack might be simpler. |
| 21:30.36 | [TK]D-Fender | terr_: Except that attempting to use it with * is a violation of their TOS |
| 21:30.52 | terr_ | *? |
| 21:30.55 | terr_ | who? |
| 21:31.36 | [TK]D-Fender | ASTERISK |
| 21:31.38 | [TK]D-Fender | ... |
| 21:32.04 | terr_ | Oh - no - I wasn't thinking of trying to use MagicJack and Asterisk together. |
| 21:32.04 | ISO9001 | lol caps. |
| 21:32.33 | terr_ | [TK]D-Fender: I was wondering if we have asterisk tied into the POTS phone numbers at this time. |
| 21:32.53 | [TK]D-Fender | terr_: huh? |
| 21:32.58 | terr_ | [TK]D-Fender: I might be interested in setting up asterisk and actually interfacing with Pots in this city. |
| 21:34.29 | terr_ | [TK]D-Fender: skype for instance has skypeout. If we want we _can_ set up an asterisk server in each city in the world and provide our own services - but the issue is how to get people who currently have POTS service interfaced with a new VoIP service. |
| 21:34.49 | Stdht | how to decde SVN-trunk-r204919 to version |
| 21:34.54 | terr_ | [TK]D-Fender: I suspect MagicJack is simply doing this. For all I know they run asterisk! |
| 21:35.05 | [TK]D-Fender | terr_extremely unlikely |
| 21:35.10 | Stdht | under 1.6 must type in cli // core show version |
| 21:35.44 | terr_ | [TK]D-Fender: well - whatever they run its pretty simnple. They need 128kb/sec. I beleive asterisk will compress - am I right? |
| 21:36.23 | [TK]D-Fender | terr_: How do you get someone with POTS over to VoIP? Repleace their PBX or put an analog gateway in front of their existing hardware |
| 21:37.10 | [TK]D-Fender | terr_: go read up on CODECS, bandwidth requirements, etc... |
| 21:37.12 | terr_ | [TK]D-Fender: simple way is with a dilogic card |
| 21:37.56 | terr_ | [TK]D-Fender: only good way is to interface to the existing POTS system... a T1 interface will do it for instance. |
| 21:38.13 | [TK]D-Fender | terrHUH? |
| 21:38.30 | terr_ | [TK]D-Fender: T1 will support 24 lines in/out. |
| 21:38.31 | [TK]D-Fender | terr_ Interfce to POTS... T1 Interface.... Dialogic... WTF? |
| 21:39.00 | terr_ | Dilogic makes a card that interfaces T1 to a PC. Been there for more than 10 years. |
| 21:39.35 | terr_ | Pri is even better. But I need to check details on PRI - since I don't need it I never checked. |
| 21:39.39 | [TK]D-Fender | terr_ I know who they are, you are throw little bits of ideas out in an incoherent manner that does not properly express the implementation you have in mind. |
| 21:40.06 | [TK]D-Fender | terr_: Akin to Star Trek "Techno-babble" being spewed out |
| 21:40.38 | terr_ | [TK]D-Fender: I wasn't looking at _any_ specific implementation. I mearly asked if there are people who provide connectivity between POTS and a system running over asterisk servers. |
| 21:40.54 | [TK]D-Fender | terr_: and how does a "T1 interface" interface with "the existing POTS system"? |
| 21:41.26 | terr_ | [TK]D-Fender: I gather perhaps not - this means asterisk is only suitable for a private set of numbers amoung a group of people which might be larger or small. |
| 21:41.39 | [TK]D-Fender | terr_: POTS = analog copper. Using someone to reach regular boring copper means that they are an ITSP for you like MajigJack, etc |
| 21:42.01 | terr_ | [TK]D-Fender: T1 is interfaced by the telco - this is how they provide business class services |
| 21:42.12 | [TK]D-Fender | terr_: PSTN, not POTS. |
| 21:42.19 | [TK]D-Fender | terr_: watch your terminology |
| 21:42.40 | [TK]D-Fender | POTS = dumb analog. |
| 21:43.05 | terr_ | POTS - Plain old telephone service... its analoge typically to a CO and then is digitized adn runs over the ATM system. DSL is flipped into the ATM system at the CO as well. |
| 21:43.20 | ISO9001 | lol |
| 21:43.44 | ISO9001 | good thing you explained that, I'm not sure [TK]D-Fender knew what it was... |
| 21:43.56 | terr_ | At the Co it runs over ATM to another CO where it might get flipped back to analogue if its going to a home to might stay digital if its going to a buisness |
| 21:43.57 | _ShrikE | patience grasshopper |
| 21:44.04 | [TK]D-Fender | terr_: I shudder to think of a service that terminates you to POTS <- |
| 21:44.28 | *** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 21:44.32 | terr_ | [TK]D-Fender: yes... but I actually run this way. |
| 21:44.48 | terr_ | [TK]D-Fender: adn I am tired of paying phone companies |
| 21:45.05 | [TK]D-Fender | terr_: You pay somone to dump a call onto a BORING ANALOG line that your provider has? |
| 21:45.32 | terr_ | [TK]D-Fender: yes - It will cost me much much more to get digital service here. |
| 21:45.35 | [TK]D-Fender | terr_: I'd love to know what termination provider uses POTS. |
| 21:45.43 | terr_ | Telus |
| 21:45.48 | terr_ | in Alberta |
| 21:46.07 | [TK]D-Fender | terr_: Unless you are referring to providing YOURSELF service between multiple system you run |
| 21:46.14 | terr_ | my phone is analogue and it runs about 15000 feet to their CO. |
| 21:46.43 | terr_ | [TK]D-Fender: no - I have been considering cancelling my analogue service and using stricktly VoIP. |
| 21:47.46 | terr_ | [TK]D-Fender: in fact I had to build a TDM - shoot the phone comapnies lines - find their problems and they wanted to charge me $1400 for 1HOUR of work to snip off their stray wires... after I told them were they were |
| 21:48.17 | terr_ | [TK]D-Fender: THEN - after I got my DSL service working - then they wanted me to provide FREE consulting so they could get everyone else running. |
| 21:48.47 | terr_ | [TK]D-Fender: THEN they over billed me more than $3000. So I have ZERO love for them. I want to replace them. |
| 21:49.12 | terr_ | MagicJack will do this for Long Distance out. I'm wondering about if I can do local service in as well. |
| 21:49.19 | *** join/#asterisk taxilian (n=richard@216.83.134.36) |
| 21:49.21 | terr_ | MagicJack is $20 per year. |
| 21:49.46 | terr_ | My ISp uplink wants $20/m for the same thing. |
| 21:50.04 | [TK]D-Fender | terr_: For as long as they exist and you follow their TOS |
| 21:50.19 | [TK]D-Fender | terr_: And don't change their terms on you |
| 21:50.22 | terr_ | However - a for instance is that I can set up asterisk in my kids places and then I don't even need the telco in order to call them. |
| 21:50.43 | terr_ | [TK]D-Fender: they MIGHT change the terms. I suspect they will. |
| 21:51.26 | *** join/#asterisk mltlnx (n=mltlnx@cpe-24-193-11-14.nyc.res.rr.com) |
| 21:51.31 | terr_ | [TK]D-Fender: I see MagicJack wants 2.7 cents per minuet into Australia. I can just mail one to a friend in Australia and then its free. |
| 21:51.41 | [TK]D-Fender | terr_: You don't need * there to call them.... |
| 21:52.00 | terr_ | [TK]D-Fender: hmm - how else? |
| 21:52.18 | [TK]D-Fender | terr_: Soft-phone or any other VoIP hardware |
| 21:52.46 | terr_ | [TK]D-Fender: what would you suggest as the cheapest / best VoIP hardware? |
| 21:53.23 | [TK]D-Fender | terr_: Small stuff = Linksys ATA's |
| 21:53.51 | terr_ | [TK]D-Fender: you see - one solution is to set up an asterisk server on a tiny box PC and sell it to people who want to eliminate the telcos of the world. |
| 21:53.53 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:54.09 | terr_ | [TK]D-Fender: I'll look into that solution. |
| 21:54.19 | *** part/#asterisk taxilian (n=richard@216.83.134.36) |
| 21:54.39 | [TK]D-Fender | terr_: * itself doesn't do any of this... |
| 21:54.53 | [TK]D-Fender | terr_: * just sits BETWEEN services & hardware you already have <- |
| 21:55.25 | terr_ | [TK]D-Fender: I'm not surprised. I just barely began to check into *. But I've looked into asterisk a couple years ago... just never set it up |
| 21:55.43 | *** join/#asterisk taxilian (n=richard@216.83.134.36) |
| 21:58.18 | *** join/#asterisk taxilian (n=richard@216.83.134.36) |
| 21:59.02 | taxilian | WindowsUser: do you remember talking to me the other day about doing weird incoming call routing with conferences? |
| 22:00.25 | hamush | i am very new to asterisk and am trying to set up a sip phone. i've followed the echo test example in the o'reilly book, but can't get it to pick up. should i be able to dial the echo test extension in the dialplan with my sip phone and have it pick up? or do dialplan extensions not correspond to dialed sip numbers? |
| 22:01.51 | [TK]D-Fender | hamush: what does "not pick up" mean? |
| 22:02.21 | hamush | i get a "100 Trying" from asterisk, but nothing else (besides keepalives) until my sip phone cancels the call |
| 22:03.11 | [TK]D-Fender | hamush: pastebin the complete failed call attempt from beginning to end with SIP debug enabeld |
| 22:03.13 | [TK]D-Fender | ~pb |
| 22:03.14 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/, or apt-get install pastebinit |
| 22:03.25 | *** join/#asterisk SaiSoma|AFK (n=SaiSoma@74.167.136.30) |
| 22:05.21 | terr_ | [TK]D-Fender: If I get something like a Linksys SPA2100 (not through reading what it can do) then can I interface this into an asterisk server? |
| 22:06.44 | [TK]D-Fender | terr_: 2100 = old discontinued... |
| 22:06.46 | *** join/#asterisk dwayne (n=dwayne@76.29.245.9) |
| 22:06.59 | terr_ | [TK]D-Fender: google came up with it first - that's all. |
| 22:07.06 | [TK]D-Fender | terr_: its a SIP ATA, you can set it up with anything that talks SIP. |
| 22:07.21 | terr_ | SIP =? |
| 22:07.34 | terr_ | ATA=? |
| 22:07.50 | [TK]D-Fender | .... |
| 22:07.53 | [TK]D-Fender | ~ata |
| 22:07.54 | infobot | well, ata is Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 22:08.01 | terr_ | [TK]D-Fender: thanx |
| 22:09.05 | hamush | [TK]D-Fender: http://pastebin.com/m55cc7170 |
| 22:09.39 | terr_ | [TK]D-Fender: ok. So this runs analog into a chip which digitizes it and then I presume SIP is an protocol which probably runs over UDP |
| 22:10.19 | terr_ | ~fx0 |
| 22:10.27 | [TK]D-Fender | hamush: Verbose 10, we can't see what the DIALPLAN is doing <- |
| 22:10.40 | terr_ | ~fx0 terr_ |
| 22:10.47 | [TK]D-Fender | terr_: Go read the book.... |
| 22:10.49 | [TK]D-Fender | ~book |
| 22:10.50 | 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 |
| 22:10.58 | terr_ | _terr ~fx0 |
| 22:11.16 | terr_ | [TK]D-Fender: how to I access the bot? |
| 22:11.40 | [TK]D-Fender | terr_: You don't seem to understand the basic hardware that Joe Blow using Vonage uses to interact with his ITSP yet you want to become one yourself... |
| 22:12.05 | [TK]D-Fender | terr_: Ask the bot about something it KNOWS. |
| 22:12.14 | hamush | [TK]D-Fender: is setting the debug verbosity a command in the command line program? i'm not familiar with how to do it |
| 22:12.29 | [TK]D-Fender | hamush: "core set verbose 10" |
| 22:12.31 | terr_ | [TK]D-Fender: ya gotta start somewhere |
| 22:12.49 | [TK]D-Fender | terr_: Yes, but you could do better than "nowhere" :p |
| 22:13.06 | terr_ | [TK]D-Fender: when I set up the webservers I had never run Linux or OpenBSD so I hired a consultant. That was more than 10 years ago and my servers have been lit since then |
| 22:13.28 | [TK]D-Fender | terr_: So basically even the job you have now you oursourced.... |
| 22:14.00 | [TK]D-Fender | terrGuess you should go hire tha consultant now and save yourself being the poor repeater of information when you go ask them to set things up ;) |
| 22:14.28 | terr_ | [TK]D-Fender: no - I outsourced it 10 years ago. I took it over aboout 9.9 years ago. |
| 22:14.32 | hamush | [TK]D-Fender: i'm getting "No such command 'core' (type 'help' for help)". is the verbosity command something that goes in the config file or in the cli? |
| 22:14.52 | [TK]D-Fender | hamush: "set verbose 10" |
| 22:14.56 | terr_ | [TK]D-Fender: He's in Chilie in his VW westy... |
| 22:15.12 | [TK]D-Fender | hamush: What version are you on anyway? Looks like 1.2- |
| 22:15.34 | hamush | [TK]D-Fender: yeah, 1.2.7.1 |
| 22:15.43 | [TK]D-Fender | EWWWW |
| 22:15.44 | terr_ | [TK]D-Fender: ok - with an FXO I can set up my computer as an answering service! Nice. |
| 22:16.07 | terr_ | ~FXO |
| 22:16.07 | infobot | foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo |
| 22:18.16 | terr_ | [TK]D-Fender: as I understand it now - the ATA is an ethernet to analog interface. I called a supplier and they usually handle analog to USB. So we have both options. |
| 22:18.46 | [TK]D-Fender | terr_: Analog to USB huh.... what device? |
| 22:18.57 | terr_ | hang on. |
| 22:19.41 | hamush | [TK]D-Fender: http://pastebin.com/m72bcc095 |
| 22:21.13 | [TK]D-Fender | hamush: Ok, do an "Answer()" first, then "Playback(silence/1)", then start the echo test |
| 22:23.54 | *** join/#asterisk bijit (i=1000@190.241.15.48) |
| 22:26.15 | hamush | [TK]D-Fender: it worked! thank you! i have been pulling my hair out! |
| 22:27.38 | *** join/#asterisk Olobola (i=Olobola@132.sub-75-210-205.myvzw.com) |
| 22:28.52 | hamush | [TK]D-Fender: i guess it makes sense that i'd need to tell asterisk to answer :) |
| 22:29.12 | Olobola | howdy.trying to get speex up and running.. yum install worked fined, rebuilt asterisk but codec_speex.so doesn't seem to be anywhere. ? |
| 22:32.22 | *** part/#asterisk callenb (n=callenb@host86-168-62-181.range86-168.btcentralplus.com) |
| 22:33.28 | terr_ | [TK]D-Fender: Headset-USB: Logitech Clearchat Pro USB Phone-USB: Sony VN-CX1A |
| 22:33.40 | [TK]D-Fender | Olobola: You must have missed the clear warnings in menuconfig that you were missing prerequisites |
| 22:33.54 | [TK]D-Fender | terr_: that is not an FXO |
| 22:33.57 | *** join/#asterisk implicit- (n=bayan@unaffiliated/implicit) |
| 22:34.08 | terr_ | [TK]D-Fender: no its not |
| 22:34.34 | [TK]D-Fender | terr_: And that only lets you use a boring phone ALONG with a softphone. By itself its worthless |
| 22:35.24 | Olobola | [TK]D-Fender: thanks, I'll reinstall |
| 22:35.40 | terr_ | [TK]D-Fender: actually I think what I'm looking for is a boring phone (for now) |
| 22:37.26 | Olobola | can anyone recommend a good ip phone? utstarcom F3000 maybe? |
| 22:38.09 | joako | Anyone know for Polycom how I can use a plantronics headset WITHOUT lifter? |
| 22:38.39 | terr_ | Keyspan VP-24A Cordless VoIP Phone ??? |
| 22:38.43 | [TK]D-Fender | Olobola: UTS = garbage |
| 22:38.48 | terr_ | Never used it - and I know little about it. |
| 22:39.14 | [TK]D-Fender | [18:38]<terr_>Keyspan VP-24A Cordless VoIP Phone ??? <_ where ARE you finding all these garbage "VoIP Products"? |
| 22:39.51 | terr_ | [TK]D-Fender: I said I know little about it. |
| 22:39.52 | [TK]D-Fender | Olobola: Polycom > All |
| 22:40.09 | [TK]D-Fender | terr_: Yeah, but you offered it as a possible suggestion. |
| 22:40.48 | [TK]D-Fender | joako: Who says a headset needs a lifter? |
| 22:40.49 | terr_ | [TK]D-Fender: nobody else offered anything |
| 22:41.16 | [TK]D-Fender | terr_ : BRILLIANT reaswon to offer up random crap to confuse him! I should have thought of that! |
| 22:41.55 | terr_ | [TK]D-Fender: you're right. Sorry |
| 22:42.07 | joako | [TK]D-Fender: I know it doesn't.... but how to answer/hangup without being at the phone? All the ones I see have both headset and lifter, e.g.: http://www.voiplink.com/Plantronics_CS55_HL10_Handset_Lifter_p/plantronics-cs55-hl10-pc.htm |
| 22:43.05 | [TK]D-Fender | joako: That is another matter. Get an new series (555/650/maybe 450) which supports the Jabra wireless lifter protocol. |
| 22:43.31 | [netman] | is there any way of replacing patterns on strings into the dialplan? |
| 22:44.02 | [TK]D-Fender | joako: that signals the phone to answer electronically over the connector wire rather than physical hook-switch manipulation. These tend to be BT headsets |
| 22:44.21 | [TK]D-Fender | [netman]: "core show function REGEX" |
| 22:44.36 | Olobola | [TK]D-Fender: thank you |
| 22:45.45 | [netman] | thx [TK]D-Fender |
| 22:46.30 | [netman] | [TK]D-Fender: it's only for matching, not for replacing |
| 22:46.37 | joako | [TK]D-Fender: Do you know anything that will work with a Polycom 501? |
| 22:46.46 | [TK]D-Fender | joako: There is non |
| 22:46.49 | [TK]D-Fender | none* |
| 22:47.06 | [TK]D-Fender | joako: Mechanical lifter or replace the phone |
| 22:47.57 | *** join/#asterisk Corydon76-dig (i=beige@pdpc/supporter/bronze/Corydon76-home) |
| 22:47.57 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 22:49.00 | *** part/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com) |
| 22:51.33 | joako | And actually if I use a lifter that wouldn't work either... |
| 23:04.30 | [TK]D-Fender | joaWhy not? |
| 23:05.01 | *** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu1.dynamic.dsl.tele.dk) |
| 23:05.46 | b14ck | hi all |
| 23:05.48 | b14ck | hows it goin |
| 23:08.58 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 23:10.12 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 23:16.04 | *** join/#asterisk ingenius (n=alektro@186.136.6.218) |
| 23:24.07 | *** join/#asterisk dwschool (n=Dwayne@64.42.227.97) |
| 23:24.28 | *** part/#asterisk taxilian (n=richard@216.83.134.36) |
| 23:25.05 | dwschool | i have two iax modems answering for an incoming sip trunk, one or them answers about 85% of the time and the other one answers about 50% of the time - asterisk says its ringing the modem, the line rings but the modem never picks up |
| 23:49.55 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
