| 00:01.19 | boom^time | I swear it wasn't doing this early last week when I tested it. |
| 00:03.31 | *** part/#asterisk slicknick5181 (~slick@207-255-114-073-dhcp.unt.pa.atlanticbb.net) |
| 00:07.04 | apb1963 | navaismo: Thank you |
| 00:07.31 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 00:13.50 | dar123 | i am trying to get my fxo working, never done it before and i might be missing a step. I am supposed to play a audio file once i recieve an inbound call, nothing happens. |
| 00:14.53 | dar123 | i do not see any difference on the led when i hear ringback |
| 00:14.59 | dar123 | is it normal? |
| 00:16.51 | *** join/#asterisk joesmoe (~joesmoe@admins.phreefilez.com) |
| 00:24.13 | navaismo | can anyone tell me what is the issue, a connection to asterisk 12 via websocket generate a segmentation fault, this is the backtrace http://pastebin.com/TEXtGcB6 i have compiled asterisk with better backtrace and dont_optimize but in the backtrace there is the <optimize out value> present |
| 00:24.34 | navaismo | boom^time, the "blah" field exist in the cdr.h? |
| 00:25.13 | navaismo | dar123, did you see anything in the cli, what is the outpout of dahdi show channels? |
| 00:25.15 | boom^time | yes, well, in the table, cdr_adaptive_odbc fins it |
| 00:25.17 | boom^time | finds |
| 00:25.37 | boom^time | it populates blah for the original CDR prior to the fork, even though it's called after the fork. |
| 00:25.46 | boom^time | with ForkCDR I'm setting (A,a,v) |
| 00:26.02 | navaismo | oh i see, i dont use odbc so im kind useless |
| 00:26.32 | *** join/#asterisk d45h (~d45h@unaffiliated/dash-/x-7576607) |
| 00:28.22 | boom^time | no worries, thanks for trying |
| 00:33.39 | dar123 | dahdi shows the channel in service |
| 00:34.09 | dar123 | i ran dahdi_monitor so the call is coming in |
| 00:39.58 | dar123 | http://pastebin.ca/2477625 |
| 00:42.17 | navaismo | dar123, Hangup(/var/lib/asterisk/sounds/en/hello-world.gsm)<----?????? |
| 00:42.45 | *** join/#asterisk andrewyager (~andrewyag@syd02s26-fw01.thecore.net.au) |
| 00:42.46 | navaismo | try playback(hello-world) |
| 00:42.53 | navaismo | then hangup() |
| 00:43.08 | navaismo | do not use the .format |
| 00:43.15 | dar123 | okies |
| 00:47.02 | dar123 | same i keep hearing ringback |
| 00:57.51 | navaismo | show us the cli output |
| 01:00.55 | *** join/#asterisk colinwielga (~colinwiel@c-69-254-123-57.hsd1.fl.comcast.net) |
| 01:03.29 | dar123 | 's' => 1. Answer() [pbx_config] |
| 01:03.29 | dar123 | <PROTECTED> |
| 01:03.29 | dar123 | <PROTECTED> |
| 01:04.33 | *** join/#asterisk camerin (hoax@elite.bshellz.net) |
| 01:06.04 | WIMPy | Playback does an implied Answer. |
| 01:06.10 | Penguin | The Answer is superfluous. |
| 01:06.41 | navaismo | dar123, that is the dialplan we need the call attempt |
| 01:09.27 | mjordan | navaismo: please file a bug for that crash. It looks like the security event module is making an assumption it shouldn't about the transport objects. |
| 01:10.12 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-pgaerpocxdtzdqva) |
| 01:10.25 | navaismo | oki |
| 01:10.29 | navaismo | too late |
| 01:13.27 | dar123 | not sure what's the command for it |
| 01:13.56 | navaismo | just dial and copy and paste all the output |
| 01:15.32 | dar123 | i do not get anyting on the CLI |
| 01:16.40 | navaismo | hmmm |
| 01:17.27 | navaismo | is connected on the correct span? with a good cable? |
| 01:17.40 | dar123 | i found one mistake i was using channel = 3 instead of channel => 3 |
| 01:18.29 | dar123 | cable is good, just removed fromt he phone and port number is correct too |
| 01:21.27 | navaismo | on the linux shell run lsdahdi and pastebin the output |
| 01:23.42 | dar123 | http://pastebin.ca/2477634 |
| 01:27.04 | navaismo | dahdi show channels in asterisk cli(again please) |
| 01:28.14 | dar123 | http://pastebin.ca/2477635 |
| 01:31.46 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 01:33.18 | navaismo | and when you dial to your analog line in the cli nothing happens right? |
| 01:33.26 | navaismo | antoher phone is using the line? |
| 01:35.30 | dar123 | nothing appears on the CLI and the line is not in use |
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| 02:17.02 | *** part/#asterisk colinwielga (~colinwiel@c-69-254-123-57.hsd1.fl.comcast.net) |
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| 02:25.24 | din3sh | hello all |
| 02:26.20 | din3sh | I have a bunch of cisco phones calling my ISDN line, I hear a double ring out tone on the cisco, how do i fix this? |
| 02:29.47 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 02:48.12 | m0sphere | what are some of the cheapest termination companies like voip.ms? |
| 02:49.00 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 02:53.17 | Penguin | ~itsp |
| 02:53.17 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 02:57.29 | dar123 | ahhh i can't get my fxo card working. Have been working on it since 6 hrs , it's 6am here :( |
| 03:13.58 | din3sh | dar123:whats the problem? |
| 03:19.44 | dar123 | i do not see any call comming into asterisk |
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| 04:00.56 | Shawn0228 | Hey. Question. Is there an easy way to install PIAF on a headless computer when i have no access to a monitor? |
| 04:01.56 | Penguin | I don't know specifically about PiaF, since it's junk and I wouldn't recommend it to anyone... but in most cases, you can install things if you have a serial connection from your laptop to the target machine. |
| 04:04.57 | Shawn0228 | ive not had any issues with piaf. im not doing anything special with piaf, just a quick dirty way for us to have google voice throughout the hosue |
| 04:04.59 | Shawn0228 | *house |
| 04:09.10 | Penguin | But why would choose specifically that distro? |
| 04:10.07 | *** join/#asterisk ziz212 (~chatzilla@203.115.12.246) |
| 04:11.12 | ziz212 | Pls be kind enough to sort this out. I have installed Asterisk and it doesnt came with TLS security certificate creation script. How can we generate a certificate in that case? |
| 04:15.51 | *** join/#asterisk dfighter (~someone@arcemu/staff/dfighter) |
| 04:19.29 | din3sh | serial connection? really? |
| 04:20.20 | *** join/#asterisk bkruse (~Adium@24.42.229.8) |
| 04:23.03 | Penguin | That's how I installed my last OS. |
| 04:23.43 | Penguin | Serial cable between the laptop and the other computer, boot the media, set the serial terminal options, and go at it. |
| 04:24.52 | din3sh | does usb to serial work? |
| 04:25.08 | din3sh | I mean not all laptops have serial ports nowadays |
| 04:27.13 | Penguin | As long as your adapter works with your laptop, it shouldn't be a problem. I still have a DB-9 on this laptop. |
| 04:28.56 | ziz212 | If the Asterisk is not coming with ast_tls_cert script means , it is not ready for TLS and SRTP ? |
| 04:30.11 | din3sh | I have a bunch of cisco phones calling the pstn via my ISDN line, I hear a double ring out tone on the cisco, how do i fix this? |
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| 06:05.31 | youjelly | anyone used LVS/ultramonkey to loadbalance asterisk |
| 06:08.32 | *** join/#asterisk j4jackj (~jack@99.199.8.60) |
| 06:08.48 | j4jackj | Hello! |
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| 06:53.47 | tomodachi | anyone used asterisk with polycom devices for audio/video conferencing? |
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| 07:23.25 | m0sphere | I'm being flooded with things like this: Executing [7777700972597368637@from-pstn:1] Set("SIP/71.19.241.12-00000705", "__FROM_DID=7777700972597368637") in new stack |
| 07:23.40 | m0sphere | is it just some random tool looking for an exploit? |
| 07:23.44 | j4jackj | Yes it is. |
| 07:23.49 | j4jackj | It has happened to me. |
| 07:24.01 | m0sphere | is reporting it to whoever owns 71.19.241.12 any use? |
| 07:24.47 | j4jackj | could be |
| 07:24.50 | j4jackj | but unlikely |
| 07:27.00 | m0sphere | hmm |
| 07:27.25 | m0sphere | i wanted to go to sleep but now I want to see if they find any type of exploits |
| 07:30.07 | j4jackj | The exploit they want is to get a loop line to their call centre. |
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| 07:39.13 | *** join/#asterisk jpoz (~jpoz@c-24-22-121-42.hsd1.or.comcast.net) |
| 07:45.08 | m0sphere | I was doing some testing with a custom dialplan and it was working. Then it seemed to stop working. I now get a busy tone when dialing the extension I was using and the cli says that no one is available to answer at this time. |
| 07:47.34 | m0sphere | how do I troubleshoot that? |
| 07:50.24 | m0sphere | for an entry in the log file: [2013-11-21 00:01:33] VERBOSE[24866][C-00000498] pbx.c: == Spawn extension (app-announcement-11, s, 6) exited non-zero on 'Local/2@from-internal-000004c3;2' |
| 07:50.42 | m0sphere | what does C-00000498 refer to and what does 000004c3 refer to? |
| 07:51.04 | j4jackj | Umm |
| 07:54.27 | m0sphere | blah i think i'm an idiot |
| 07:56.36 | m0sphere | yep. missing a ; in my php |
| 07:58.58 | ChannelZ | It could be worse, you could have misplaced your colon |
| 07:59.04 | m0sphere | lol |
| 08:03.47 | m0sphere | is there a cli command to show if text2wave is set up/installed/configured properly? |
| 08:07.28 | m0sphere | i need to sleep, i'll deal with it later, goodnight |
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| 08:47.52 | quenode | hi anyone know how to parse emails from voicemail .wav files so I can execute do debug mp3vm.pl it has some bug in it. Tnx :) |
| 08:48.04 | [j4jackj] | uhhh |
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| 09:32.59 | sjs205 | Morning all... what is the best way to initiate two calls from a single call to a specific extension? |
| 09:35.43 | kaldemar | what do you mean by initiating two calls? do you mean to dial two destinations or originating new calls? |
| 09:37.13 | sjs205 | kaldemar, originating two calls and joining them together, lisk Dial with multiplu numbers, but the additional numbers are not hung up when one answers... |
| 09:38.03 | kaldemar | core show application Page |
| 09:38.27 | kaldemar | option d enables audio both ways. |
| 09:39.07 | sjs205 | Nice... that could be what I'm looking for... cheers kaldemar... |
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| 10:16.25 | *** join/#asterisk michael_work (~michael@bzq-112-168-31-118.red.bezeqint.net) |
| 10:16.43 | michael_work | hello |
| 10:19.04 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 10:20.39 | michael_work | just wonder how is it possible that i set specific moh class in Dial app and it works but at some point i see it stoped and started other class. without any change |
| 10:20.54 | michael_work | i don't have debug on to provide more info :( |
| 10:21.08 | michael_work | anyone familiar with that? |
| 10:22.07 | *** join/#asterisk rox (~rox@mx.abraxas.si) |
| 10:22.41 | rox | hello, I am running asterisk 11.5.1, and I am trying to capture SIP debug, however I can't get it to go to any of the files |
| 10:23.13 | tomodachi | rox: you could always tcpdump the traffic itself |
| 10:23.56 | rox | i could, however I am sort of used to the convenience of getting SIP debug into a file and I have some scripts that depend on it |
| 10:24.29 | sjs205 | okay, still working on originating 2 calls from a single call. The issue is that I want one of the new calls to be played a IVR message, and when they press 1 the two calls are joined... any ideas? |
| 10:24.35 | rox | my logger is configured so: |
| 10:24.35 | rox | debug => debug |
| 10:24.35 | rox | console => notice,warning,error |
| 10:24.35 | rox | messages => notice,warning,error,debug |
| 10:24.51 | rox | however, i can see SIP debug packages in console, but not in messages or debug file |
| 10:25.07 | rox | can anybody please give me an idea, how i can get SIP debug to go to any of the files? |
| 10:25.09 | kaldemar | rox: you don't get sip debug to a file with logger. |
| 10:25.10 | michael_work | rox, verbose ? |
| 10:25.46 | michael_work | anyone any idea on my problem? |
| 10:25.52 | rox | michael_work: verbosity would control what goes into file and what not=? |
| 10:26.13 | kaldemar | rox: if you're trying to get it temporarily for debugging purposes, use tee for example: asterisk -vvvr | tee /tmp/ast_out.log |
| 10:26.22 | kaldemar | rox: then enable sip debug in CLI |
| 10:26.29 | michael_work | as you have debug error warn you have verbose as well |
| 10:26.35 | michael_work | try to add it to messages |
| 10:26.45 | kaldemar | verbosity does not include sip debug. |
| 10:26.58 | rox | kaldemar: what you are saying is, I can't get SIP debug to go to a file the old way any more? |
| 10:26.59 | michael_work | ok |
| 10:27.09 | rox | kaldemar: and thank you for the quick solution |
| 10:27.12 | kaldemar | rox: i'm saying you never could. |
| 10:28.02 | rox | kaldemar: i am using Asterisk 1.2 and 1.8 in production, and 'sip set debug peer foo' command started dumping SIP packets into my messages file |
| 10:33.43 | mic_ | has anyone tried running asterisk now on HP Micro Proliant? |
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| 11:13.56 | kaldemar | rox: i was full of crap. chan_sip verbosity prints the sip debugs, so you'll get the debug with the verbose level. |
| 11:16.56 | michael_work | kaldemar, haha i was right |
| 11:17.06 | michael_work | :P |
| 11:17.36 | tomodachi | mic_: i havent but why would it be any different than any other plattform? |
| 11:17.53 | tomodachi | if you can run asterisk on your raspberry pi im sure the ph micro proliant will do just fine |
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| 12:10.28 | sjs205 | Is there an alternative to WaitExten within a macro? |
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| 12:12.53 | mic_ | tomodachi: it was just to do some research |
| 12:13.01 | mic_ | tomodachi: I will be getting one soon to run some tests then :) |
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| 13:15.47 | ziz212 | Hi, |
| 13:16.33 | ziz212 | How to add some settings to context which defined in another file in asterisk ? |
| 13:16.40 | ziz212 | pls help |
| 13:17.45 | kaldemar | ziz212: #include |
| 13:18.56 | *** join/#asterisk jasonwert (~w3rt@71.89.137.28) |
| 13:19.07 | rox | ziz212: edit the other file? |
| 13:20.26 | ziz212 | I mean if I have extension 4000 with some parameters in sip.conf and I am using sip.1.conf with #include sip.1.conf .. and if I need to add more parameters to context 4000 ... how can i add that to the second file (not sip.conf but to sip.1.conf) ? |
| 13:21.41 | *** join/#asterisk longst (~longst@83.223.25.245) |
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| 13:23.17 | longst | I am wondering in somewhere DAHDI configuration I could set Channel identification Preferred/exclusive ? |
| 13:24.47 | *** join/#asterisk Dovid (~Dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 13:24.48 | WIMPy | This time I don't need to fail at understanding the question as I already fail to parse it. |
| 13:25.11 | longst | more over, operator signal PBX over PRI with content of "High layer compatibility" part, but DAHDI board does not send this part.. |
| 13:25.38 | ziz212 | lets say if I create extension as 4000 in sip.conf ... so i am writing [4000] then puting setting bellow lines.. and if i need to add some additional settings to that [4000] ..in some other file such like sip.1.conf whcih it linked to sip.conf .. how can I write that in sip.conf |
| 13:26.43 | WIMPy | ziz212: You want to spread the configuration of a single peer over multiple files? |
| 13:27.17 | ziz212 | yes your right |
| 13:27.41 | ziz212 | how can i start that in second file ? |
| 13:27.45 | WIMPy | longst: Yes, Asterisk doesn't know anything beyound the BC. |
| 13:28.09 | WIMPy | ziz212: I doubt that is possible with peers. |
| 13:28.16 | WIMPy | Maybe you should look at templates. |
| 13:28.22 | longst | WIMPy: what is BC please? |
| 13:28.36 | WIMPy | Bearer Capability |
| 13:28.47 | ziz212 | can I use [4000]+ in second file |
| 13:28.56 | ziz212 | I found this in a blog |
| 13:29.03 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 13:29.03 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 13:29.19 | WIMPy | That works for the dialplan, but I doubt it works for others. |
| 13:29.55 | WIMPy | And IIRC that should be (+). |
| 13:30.22 | [TK]D-Fender | ziz212: #include /path/to/file.whatever |
| 13:30.28 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
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| 13:34.25 | kaldemar | [context](+) works in sip.conf aswell. |
| 13:35.11 | longst | WIMPy: what BC used in this case? |
| 13:35.20 | ziz212 | yes it is right |
| 13:35.31 | ziz212 | [4000](+) |
| 13:35.33 | WIMPy | Wow |
| 13:35.39 | ziz212 | thanks |
| 13:35.42 | ziz212 | for the help |
| 13:35.48 | WIMPy | longst: I don't understand your question. |
| 13:36.04 | [TK]D-Fender | kaldemar: that works when you need to "add" to an already accumulating device definition, but he was asking about how to include from an actual file... |
| 13:36.18 | *** join/#asterisk Greenlight (wluke@cpc1-dund9-0-0-cust142.16-4.cable.virginm.net) |
| 13:37.08 | *** join/#asterisk jhlavacek (~jirka@78.208.220.3) |
| 13:37.29 | kaldemar | [TK]D-Fender: he did say "additional settings". |
| 13:38.01 | longst | WIMPy: actually I didn't really understand your answer"Asterisk doesn't know anything beyound the BC." |
| 13:38.08 | dar123 | hi guyz, need help with my fxo. Still not working, my third day trying to get it to work :( |
| 13:38.44 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-59-196.user.veloxzone.com.br) |
| 13:39.02 | WIMPy | longst: It only knows about BC, but not the HLC/LLC you mentioned. |
| 13:40.20 | dar123 | i think calls are not reaching asterisk |
| 13:40.42 | longst | WIMPy, if you understood your correctly DAHDI channel would not set anything in "High Layer Compatibility" correct? |
| 13:41.05 | WIMPy | correct |
| 13:41.43 | longst | WIMPy: may what is meaning for LLC? |
| 13:42.26 | [TK]D-Fender | kalYup, tht could go either way... depends whether he wants the device-name header there or not... |
| 13:42.33 | [TK]D-Fender | kaldemar: |
| 13:42.33 | WIMPy | Low Layer Compatibility |
| 13:45.47 | *** join/#asterisk jasonwert (~w3rt@71.89.137.28) |
| 13:49.38 | *** join/#asterisk Rhomber (~david@60-240-245-17.static.tpgi.com.au) |
| 13:49.41 | Rhomber | hey |
| 13:50.03 | Rhomber | i was wondering if there any guides for forwarding an external ip to a local one and hosting asterisk on that? |
| 13:50.34 | Rhomber | i am having problems with firewalls on a bridged interface, can't seem to work around it.. so figured it might be easier to do that. |
| 13:51.31 | tvmkid | Callers to one of my DAHDI channels is not able to detect hangup in DAHDI channel. I am in India and the provider is BSNL |
| 13:52.39 | tvmkid | I have a recording of dahdi_monitor of the call and RX side shows a change in waveform when call disconnects |
| 13:52.57 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 13:53.08 | tvmkid | but I am unable to figure out what is happening based on the waveform |
| 13:57.13 | *** join/#asterisk LooserOuting (~LooserOut@ip-176-198-134-194.unitymediagroup.de) |
| 13:58.22 | *** join/#asterisk giany (~giany@shifu.x83.org) |
| 13:58.22 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 13:58.34 | dar123 | is there a step by step troubleshooting guide for fxo? |
| 13:58.44 | giany | hi, is there a way I can send 3 beeps in case a ported number is dialed? |
| 14:00.49 | [TK]D-Fender | darJust show us. |
| 14:00.55 | [TK]D-Fender | dar123: Just show us |
| 14:01.05 | [TK]D-Fender | dar123: The request is too vague |
| 14:01.12 | dar123 | wait .. |
| 14:01.43 | [TK]D-Fender | giany: How is * expected to know a number is "ported"? |
| 14:02.13 | *** part/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 14:02.40 | giany | based on the DID called, I can interrogate a DB or something and if the DID is in the ported list then insert 3 beeps, else use normal flow. |
| 14:02.45 | giany | my problem is the beeps :) |
| 14:03.11 | *** join/#asterisk d45h (~d45h@unaffiliated/dash-/x-7576607) |
| 14:06.52 | Kobaz | mm |
| 14:06.52 | Kobaz | so |
| 14:06.55 | dar123 | http://tny.cz/d25b1c74 |
| 14:06.58 | Kobaz | this would be my first time doing this |
| 14:07.07 | Kobaz | who/where do i send polycom phones to get fixed |
| 14:07.11 | dar123 | testing inbound calls on fxo, keep hearing ringback |
| 14:07.22 | Kobaz | google only tells me where to buy refurb phones |
| 14:07.27 | Kobaz | can't find repair info at all |
| 14:09.31 | [TK]D-Fender | giany: Playback(beep)< <- DONE |
| 14:09.45 | [TK]D-Fender | giany: and the "lookup or somthing" ... that is again for you to do in the dialplan |
| 14:10.06 | [TK]D-Fender | Kobaz: An official reseller who supports them |
| 14:10.56 | Kobaz | mm |
| 14:10.56 | Kobaz | k |
| 14:10.59 | [TK]D-Fender | dar123: What is the problem with it? |
| 14:12.11 | [TK]D-Fender | dar123: Show us the call attepmt, and status dumps prior/during. |
| 14:12.25 | [TK]D-Fender | dar123: "dahdhi show status", "dahdi show channels" |
| 14:13.47 | boom^time | Does anybody know why when I use ForkCDR and set a CDR variable afterwards it sets the variable to the original (preforked) CDR? |
| 14:13.55 | boom^time | I'm using ForkCDR(A,a,v) |
| 14:14.43 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 14:17.54 | dar123 | i do not see call attempts on verbose mode on cli of asterisk. how can i check the status dumps |
| 14:18.45 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.37) |
| 14:19.44 | dar123 | http://tny.cz/be14afdf |
| 14:21.40 | [TK]D-Fender | darWhat port is your line plugged into? |
| 14:22.14 | *** join/#asterisk wolfmitchell (~wolfmitch@botters/wolfmitchell) |
| 14:22.33 | dar123 | port 3 |
| 14:24.39 | boom^time | Does anyone have any experience with ForkCDR? I'm completely stumped. I'd really appreciate it. |
| 14:26.11 | [TK]D-Fender | dar123: call out on it.... |
| 14:27.12 | *** join/#asterisk wolfmitchell (~wolfmitch@botters/wolfmitchell) |
| 14:28.33 | dar123 | it will take me a while to do it, still have to learn the dialplan |
| 14:29.17 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 14:29.45 | dar123 | i am located in Saudi Arabia and i am using finland loadzone, read in a forum that it works |
| 14:29.51 | WIMPy | No need for dialplan. channel originate dahdi/3 application sayunixtime |
| 14:29.53 | dar123 | could that be the problem? |
| 14:31.14 | dar123 | http://tny.cz/d014462b |
| 14:36.35 | [TK]D-Fender | dar123: There is no error in there.... listen with a phone in parallel |
| 14:40.00 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:40.06 | dar123 | when the call comes in to the fxo, should the LED color change? its always green |
| 14:42.53 | boom^time | Okay, I came up with a pretty cappy hack for anyone experiencing the same problem. I have to set the variable before the fork, using v in forkcdr to copy the variable to the new forked cdr, and then wipe the variable from the original cdr. Which is not atomic so it feels dirty but I can't find a better solution. |
| 14:47.02 | *** join/#asterisk sjs205 (~sjs205@host81-151-252-147.range81-151.btcentralplus.com) |
| 14:47.29 | boom^time | Just seems odd that you'd fork a cdr and then never touch the fork but just the original. Especially considering there is a special option of s withing forkcdr for setting the variable on the original cdr, seems pointless. |
| 14:47.46 | boom^time | Okay I guess I'm done talking to myself. :) |
| 14:47.53 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-gcwnxeorqvsulcws) |
| 14:47.54 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 14:50.19 | sjs205 | I know that waitextern doesn't work in a macro, is there another option? |
| 14:50.47 | WIMPy | The manual says use Read(). |
| 14:51.00 | sjs205 | Cheers WIMPy... again :) |
| 14:55.06 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 14:56.27 | tvmkid | Callers to one of my DAHDI channels is not able to detect hangup in DAHDI channel. I am in India and the provider is BSNL |
| 14:57.02 | tvmkid | <PROTECTED> |
| 14:57.33 | tvmkid | Some one here who can help me with understanding the waveform and figuring out configuration |
| 15:02.01 | *** join/#asterisk serafie (~erin@nat/digium/x-jbhgyacqvphffzwq) |
| 15:03.08 | *** join/#asterisk vlad_starkov (~vlad_star@217.26.6.70) |
| 15:04.21 | longst | I am wondering if I could find some guide that could set two DAHDI PRI ports one PRI port received calls and transparently sends to the other PRI port? thanks |
| 15:08.42 | WIMPy | Put each port in to a context and Dial() out the other port. |
| 15:12.17 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-hkbvzymyycghiphd) |
| 15:12.17 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 15:46.16 | *** join/#asterisk slicknick5181 (~slick@207-255-114-073-dhcp.unt.pa.atlanticbb.net) |
| 16:03.32 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 16:05.33 | *** join/#asterisk longst (~longst@83.223.25.245) |
| 16:15.17 | *** join/#asterisk Vendigroth (41254d61@gateway/web/freenode/ip.65.37.77.97) |
| 16:15.24 | sjs205 | I'm getting an "Unable to create/find SIP channel for this INVITE" when I dial into my sever from one peer, but when I dial from another it all works fine... any ideas? |
| 16:16.34 | sjs205 | Sorry, that should be client, not peer |
| 16:16.43 | sjs205 | Although they are different peers... |
| 16:18.27 | tomodachi | how do i enable dtm logging to console? i have set verbose to 10 and debugging to 10 but still dont see anything |
| 16:18.32 | tomodachi | dtmf |
| 16:20.22 | tomodachi | ah found it |
| 16:21.13 | tomodachi | hmm strange i see dtmf on calls to my meetme applcation |
| 16:21.28 | tomodachi | but when i call out on PSTN there are no dtmf entries in the log |
| 16:24.37 | *** join/#asterisk din3sh (298855dd@gateway/web/freenode/ip.41.136.85.221) |
| 16:26.21 | din3sh | can a firewall on a pc be the cause of one-way audio on a softphone installed on that same pc? |
| 16:26.49 | WIMPy | Easily |
| 16:27.28 | tomodachi | din3sh: quite often |
| 16:27.49 | din3sh | uhuh thanks will check that out |
| 16:31.26 | din3sh | next question: I have some cisco phones with sip firmware running on asterisk 1.8.20, call to pstn numbers via my ISDN line generate a double tone when dialing. Dialing SIP extensions the ring out is normal |
| 16:32.07 | Vendigroth | Is there a way to reduce the "Dropping incompatible voice frame" notices? the logs are getting flooded with them |
| 16:32.29 | WIMPy | Did you use option r in your Dial()? |
| 16:32.33 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 16:32.36 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 16:34.22 | din3sh | using option r or not using it doesn't have any effect |
| 16:35.00 | WIMPy | That sounds b0rked. |
| 16:35.27 | Vendigroth | happens mostly on incoming dahdi calls. those are then passed to an AGI script through extensions.conf. no dialing is going on. |
| 16:35.28 | din3sh | b0rked? |
| 16:35.34 | din3sh | :o |
| 16:35.53 | WIMPy | broken |
| 16:36.47 | din3sh | progressinband=never in my sip.conf |
| 16:37.31 | WIMPy | That doesn't seem to work then. |
| 16:37.36 | tomodachi | anyone have any clue why my dtmf over sip trunk isnt working? in the console i see dtm when sending it to an app like meetme, but when i call a number over my sip trunk to pstn there is nothing in the console |
| 16:37.55 | WIMPy | din3sh: WHy do you try to disable it? |
| 16:38.10 | Greenlight | tomodachi: Are you keeping asterisk in the media path ? |
| 16:38.36 | din3sh | the users are pretty annoyed |
| 16:38.53 | din3sh | a dialout tone overlaps another one |
| 16:39.00 | WIMPy | tomodachi: Because you're actually dialling and not sending dtmf? |
| 16:39.12 | WIMPy | din3sh: I haven't had any issues with =yes. |
| 16:39.24 | jansiva | [TK]D-Fender: slicknick5181: still we are getting calls now using 100 number |
| 16:39.28 | jansiva | sip_general_custom.conf <---- allowguest=no |
| 16:39.31 | tomodachi | Greenlight: thats a goodquestion ,you think perphas my phone talkts to the sip trunk server directly? |
| 16:39.38 | tomodachi | reinvite |
| 16:40.14 | Greenlight | directmedia. It's a possibility. I'm also not sure if DTMF tones would always appear in the console for a normal call, even if you are in the media path. |
| 16:40.21 | tomodachi | WIMPy: well its a pure ip phone im already in the call, when i attempt to send dtmf , the other end is not getting them since its not reacting to my menu choices |
| 16:40.23 | din3sh | can progressinband=option be set on a dahdi channel/trunk or works only on sip? |
| 16:40.28 | Greenlight | You've a few methods of DTMF as well, you need to choose on that your provider supports |
| 16:40.31 | WIMPy | tomodachi: WHEN don't you see them? |
| 16:41.06 | tomodachi | WIMPy: when i call to pstn number that requets me to enter some menu option |
| 16:41.45 | Greenlight | Are you using inband DTMF or out of band ? |
| 16:42.18 | tomodachi | i guess out of band is the default? |
| 16:42.26 | WIMPy | din3sh: I'd have to check what I used with dahdi. But I surely wouldn't want to miss more than neccessary. |
| 16:42.40 | Greenlight | Not too sure what the default is, you should be specific on the trunk settings |
| 16:42.41 | tomodachi | dtmfmode is set to auto |
| 16:42.53 | Greenlight | What does your ITSP support ? |
| 16:43.52 | Greenlight | Some ITSPs, use kit where the rfc2833 doesn't "agree" with Asterisk, in which case use inband or "info", or alter the source code as we did. |
| 16:44.20 | Greenlight | I beleive auto will try and use rfc2833 |
| 16:44.51 | tomodachi | ok , ill try the inband setting then , i just tried googline their home page and in their snom config example they actually specify dtmf signaling as inband |
| 16:45.45 | tomodachi | nope didnt work |
| 16:45.45 | din3sh | WIMPy:internal calls to sip extensions does not have a double ring out tone...only numbers through the dahdi trunk |
| 16:45.52 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 16:45.59 | tomodachi | but i should see dtmf in my console no matter what signaling i use shoudnt i? |
| 16:46.25 | tomodachi | or will I only see dtmf events that terminate in my own pbx? |
| 16:46.34 | Greenlight | Not sure on that one to be honest |
| 16:47.10 | tomodachi | i guess I could dump some traffic and see if i see the dtmf stuff there... |
| 16:47.19 | WIMPy | din3sh: Sure. They usually don't have early media. |
| 16:47.55 | WIMPy | din3sh: Looks like the only dahdi box I have left is using mustly defalut parameters. |
| 16:48.21 | din3sh | :| |
| 16:48.39 | din3sh | is there a setting for earlymedia = no in dahdi? |
| 16:49.29 | tomodachi | din3sh: there is a ;overlapdial=yes setting in chan_dahdi.conf , perhaps you have it enabled by accident? |
| 16:49.49 | tomodachi | WIMPy: Greenlight well guys im off, will try some more tomorrow, thnx for the helpfull input! |
| 16:51.00 | WIMPy | I think overlap dialling will actually give you a premature connect, not just early media. |
| 16:51.14 | WIMPy | But that's all extremely iffy with Asterisk anyway. |
| 16:52.36 | din3sh | my dahdi settings: |
| 16:52.37 | din3sh | http://pastebin.com/Rt67wdBn |
| 16:53.24 | WIMPy | Maybe that's the trick, now that I think of it. I have definitely always had overlap enabled. |
| 16:57.16 | din3sh | i should enable it? |
| 16:57.38 | WIMPy | You could try it. |
| 16:57.57 | din3sh | btw all the other phones, polycom/grandstream etc work with single dialout tone, only Cisco phones acting wild |
| 16:58.09 | WIMPy | I'm not much in to dahdi any more. |
| 16:58.12 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 16:59.17 | WIMPy | So if you have progrssinband=never and get double rings, that must be a double bug. 1. Asterisk sending out inband and the phone using it without disabling its own tones. |
| 17:03.47 | din3sh | i think the progessinband=never might only work when calling SIP peers |
| 17:04.34 | WIMPy | Doesn't make sense to me, but that doesn't mean much. |
| 17:10.58 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 17:18.59 | din3sh | asterisk sure gives some sleepless nights |
| 17:21.15 | file | certainly sounds (ha!) like that is what is happening... Asterisk won't mix that stuff itself |
| 17:29.35 | ipengineer | Hello everyone! Any thoughts on why two phones would be setting up native_rtp when I have direct_media set to no? http://paste.laravel.com/19iF |
| 17:30.51 | file | the native_rtp bridge technology supports local and remote bridging |
| 17:30.53 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 17:30.53 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 17:31.11 | file | if remote bridging is not possible (you have it disabled) it will try to locally bridge within the RTP stack |
| 17:31.25 | ipengineer | hmm ok not sure why all of a sudden media has stopped working sporadically and placing a call on hold ends up in disconnects every time |
| 17:32.24 | file | to answer that would require complete console output with debug and looking at what is going on at the time |
| 17:32.54 | ipengineer | file: Ok. What I am wanting to do is have asterisk handle the bridging every time. I don't want the devices to handle rtp between themselves under any circumstances. The only thing I did in ps_endpoints was set direct_media=no and disable_direct_media_on_nat=yes |
| 17:33.05 | file | they aren't |
| 17:33.10 | file | media is flowing through Asterisk |
| 17:34.21 | ipengineer | file: Ok. What is the best way to get that output? Do I need to set pjsip debug enable? The output I pasted was just the console running with verbose set high |
| 17:34.42 | file | it's not PJSIP level, bridging is a core operation thus "core set debug 9" and debug output going to a file or the console |
| 17:38.09 | ipengineer | file: is there an option to show debug output in console? I am not seeing anything different in the console output. It states I changed debug from 0 to 9 but the output is the same |
| 17:38.30 | file | it's configured in logger.conf where it goes |
| 17:40.38 | ipengineer | file: Got it, Thanks.. http://paste.laravel.com/19iI |
| 17:40.53 | mjordan | file: we could, I suppose, emit a verbose message from native_rtp that says whether or not the bridge is local or remote |
| 17:41.01 | file | mjordan, aye |
| 17:41.27 | mjordan | it just happens to be DEBUG right now |
| 17:41.30 | mjordan | [Nov 21 17:39:40] DEBUG[11973][C-00000148]: bridge_native_rtp.c:154 native_rtp_bridge_start: Locally RTP bridged 'PJSIP/sip100_abcdefg-00000082' and 'PJSIP/sip101_abcdefg-00000083' in stack |
| 17:42.04 | file | ipengineer, I don't have an immediate answer for you |
| 17:42.14 | mjordan | ipengineer: you do have strictrtp enabled, which does mean that for a short period of time, RTP is going to get dropped until the source of the RTP locks. |
| 17:42.20 | *** join/#asterisk generalhan_ (~generalha@about/windows/staff/generalhan) |
| 17:42.21 | mjordan | what are you actually seeing happen? |
| 17:42.55 | ipengineer | mjordan: I am not sure if it is or not. It is not something I have changed from how it was set when installed.. |
| 17:43.09 | mjordan | k - the media is passing through Asterisk based on that log file |
| 17:43.21 | mjordan | it's just doing a packet to packet bridge as opposed to a full translation |
| 17:43.32 | file | quite |
| 17:43.39 | ipengineer | mjordan: I can place a call between two extensions, once I answer media may or may not work. If media does work and I place a call on hold the devices I pressed hold on disconnects while the other side sits in limbo with no media |
| 17:44.37 | mjordan | I don't think we have enough information to help. The only way to see what is going on would be to look at a full SIP trace, along with RTP debug enabled |
| 17:45.13 | mjordan | It does look like you have some retransmissions going on: [Nov 21 17:39:43] DEBUG[1684]: pjsip:0 <?>: tsx0x7f77e800a .Retransmiting Response msg 200/INVITE/cseq=1121 (tdta0x7f77e44a64c0), count=2, restart?=1 |
| 17:45.23 | mjordan | which may mean you have some NAT issues |
| 17:45.29 | file | need it allllllll |
| 17:45.56 | ipengineer | so I need to pjsip set logger on and rtp debug? |
| 17:46.04 | ipengineer | rtcp set debug on |
| 17:46.14 | file | rtp set debug on |
| 17:46.37 | *** join/#asterisk cmendes0101 (~cmendes01@office.phone.com) |
| 17:47.38 | ipengineer | http://paste.laravel.com/19iL |
| 17:50.47 | file | it looks like something is not ACKing the 200 OK |
| 17:51.46 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 17:52.11 | ipengineer | Can you see which IP is sending the 200 OK? |
| 17:53.09 | file | Asterisk is sending the 200 OK to 192.168.3.121 |
| 17:53.28 | file | to answer the call. |
| 17:54.31 | file | wow, a Contact of 127.0.0.1... |
| 17:54.57 | ChannelZ-Wk | 'I talk to myself, cos there is no one to talk to..' |
| 17:55.07 | file | how do you have this configured... |
| 17:55.22 | [TK]D-Fender | Think of it as Èa conversation with a very interesting individualÈ |
| 17:55.37 | ipengineer | Which part? pjsip module? |
| 17:55.46 | file | cause something is wacky... 192.168.3.169 is putting 127.0.0.1 in its messaging |
| 17:56.54 | ipengineer | file: Ok so I just restarted asterisk and everything is working fine. Back to like it was media passing, hold working. No configuration changes.. Thats a little concerning though. Nothing has been touched in days. |
| 17:57.42 | file | interesting |
| 17:58.04 | file | are you using any NAT settings? |
| 17:58.32 | file | heck that wouldn't even modify the From header... |
| 17:59.06 | file | I'd definitely look at the GXP2020 at 192.168.3.169... it's SIP signaling should not have 127.0.0.1 in it |
| 17:59.46 | *** part/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net) |
| 18:01.46 | WIMPy | file: I'm not sure how things are now, but Asterisk certainly could mix it's own ringing with early media. |
| 18:02.12 | ipengineer | Ok.. Well it was going it between two 2100's as well and a 2100 calling out to the PSTN |
| 18:02.13 | file | WIMPy, it won't mix the audio if dialing out unless you are creative and do it while in a conference bridge |
| 18:02.23 | ipengineer | I captured it again working if you are curious at looking |
| 18:02.24 | ipengineer | http://paste.laravel.com/19iU |
| 18:02.44 | file | there's no longer 127.0.0.1 in there |
| 18:02.48 | WIMPy | file: It definitely used to do so. I've had it from dahdi to dahdi. |
| 18:03.09 | file | WIMPy, that would be a DAHDi-ism |
| 18:03.51 | WIMPy | Does the channel itself create inband ringing? |
| 18:04.28 | file | ipengineer, yeah... the signaling for that is now fine |
| 18:04.39 | ipengineer | file: Right.. i am seeing this though where it wasn't there before "No remote address on RTP instance '0x7fb230005868' so dropping frame" |
| 18:05.21 | file | ipengineer, cause it is on hold... which involves changing the stream state and to support the older standard setting the target for RTP to 0.0.0.0 |
| 18:05.32 | file | ipengineer, since that address goes nowhere... Asterisk can't send media |
| 18:05.36 | *** join/#asterisk kresp0_ (~kresp0@gateway/tor-sasl/kresp0) |
| 18:05.36 | file | WIMPy, which channel? |
| 18:05.46 | WIMPy | dahdi |
| 18:06.03 | WIMPy | (in that case, tho the question was of a general nature) |
| 18:06.05 | ipengineer | file: What are the chances Asterisk could have been manipulating the headers or translating them wrong? Like you said I don't know the grand stream would set to 127.0.0.1 and after a reboot why would that change? I guess I could have ran wireshark on the grand stream itself |
| 18:06.21 | ipengineer | file: Ok that makes sense |
| 18:06.40 | file | WIMPy, DAHDi does it internally and due to the way DAHDi channels work stuff can be mixed at that layer |
| 18:07.00 | file | WIMPy, the core has the capability to do tone generation and just replaces the outgoing stream to a device with the generated tones, it can't mix it with the existing stream |
| 18:07.08 | WIMPy | Far too much channeltype specific stuff in Asterisk. |
| 18:07.45 | file | ipengineer, it's POSSIBLE but I can't name anywhere we would modify all of that... if rewrite_contact is on we'll rewrite the Contact with the source address, but not the rest |
| 18:08.22 | file | a tcpdump when it occurs would see if it is being rewritten |
| 18:09.26 | ipengineer | file: Ok.. Well I will let it sit and see if it creeps up again.. No sense in beating a dead horse. Thanks for your help! |
| 18:09.58 | ipengineer | file: Yea I will grab that if it happens again. |
| 18:14.18 | Qwell | WIMPy: patches welcome |
| 18:15.22 | *** join/#asterisk TSM (~the_softw@fw-lon1.wenn.com) |
| 18:15.22 | WIMPy | Patches? |
| 18:15.28 | *** part/#asterisk TSM (~the_softw@fw-lon1.wenn.com) |
| 18:15.31 | WIMPy | That sounds like a complete rewrite to me. |
| 18:16.45 | Qwell | k, well let us know when you've got something for us to test then |
| 18:17.22 | WIMPy | Maybe I would have had something if I hadn't battled with Asterisk for years. |
| 18:24.28 | *** join/#asterisk kenlik (~alberto@187.23.86.91) |
| 18:27.54 | kenlik | hi! i'm using asterisk 1.8, how to added a new value in "userfield" column of Master.csv file? |
| 18:30.52 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 18:32.03 | *** join/#asterisk chris_n (~Chris@koha/developer/chris-n) |
| 18:35.10 | anonymouz666 | WIMPy: does your battle ended? |
| 18:35.36 | WIMPy | No :-( |
| 18:36.55 | *** join/#asterisk Micc (~Micc@static-50-125-113-34.frr01.both.wa.frontiernet.net) |
| 18:37.26 | *** part/#asterisk slicknick5181 (~slick@207-255-114-073-dhcp.unt.pa.atlanticbb.net) |
| 18:38.32 | anonymouz666 | talking about deadlocks asterisk just did a big step forward. seg faults it is still a battle, at least for me. what makes things even harder it's the fact I can't provide useful backtraces (I think) |
| 18:38.56 | anonymouz666 | debug_threads dont_optimize kills the performance |
| 18:39.26 | WIMPy | The impact of dont_optimize shouldn't be big. |
| 18:45.43 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 18:51.32 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 19:04.57 | lvlinux | does anyone know why my global variables in the dialplan don't get loaded before the dialplan? I have to manually run "dialplan reload" to get them to be loaded. |
| 19:05.33 | Penguin | The ones you have defined in the globals section? |
| 19:05.38 | lvlinux | yes |
| 19:06.36 | lvlinux | i have several phones on a variable called $LOCALPHONES (for a ring group) |
| 19:07.24 | Penguin | Show me how you set it. |
| 19:07.27 | lvlinux | when an incoming call comes it, it dials ${LOCALPHONES} |
| 19:07.33 | lvlinux | ok i'll pastebin |
| 19:08.01 | *** join/#asterisk longst (~longst@83.223.25.245) |
| 19:09.03 | *** part/#asterisk longst (~longst@83.223.25.245) |
| 19:10.33 | lvlinux | http://pastebin.com/LYDkMAn4 |
| 19:11.49 | WIMPy | What do you expect in LOCALPHONES while BOB isn't set, yet? |
| 19:12.09 | Penguin | LOCALPHONES is being set from a variable named BOB. Where does BOB get set? |
| 19:12.14 | drmessano | Beat me to it |
| 19:12.20 | lvlinux | lol yes I just figured that out... |
| 19:12.52 | Penguin | I don't know for sure that it will help to set it first, but I certainly would give it a try. |
| 19:13.13 | lvlinux | should've pastebinned it for myself to look at lol when i was putting it in there I thought that must be it so just moved the LOCALPHONES variable below BOB and it works fine. |
| 19:13.24 | lvlinux | doh! |
| 19:13.33 | Penguin | Good. |
| 19:15.43 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.65) |
| 19:15.59 | lvlinux | now if i can just figure out why chan_skinny says bad input every couple seconds... |
| 19:16.01 | *** join/#asterisk Dpunkt (~Dpunkt@p5DE5F715.dip0.t-ipconnect.de) |
| 19:16.17 | Penguin | Which asterisk branch are you using? |
| 19:16.54 | lvlinux | 11.4 |
| 19:17.09 | lvlinux | no sorry 11.5 |
| 19:17.11 | Penguin | And you are using Cisco phones with SCCP on them? |
| 19:17.22 | lvlinux | just one---a 7920 wifi cordless |
| 19:17.32 | lvlinux | no SIP firmware for it :-( |
| 19:17.42 | Penguin | I remember. |
| 19:17.58 | lvlinux | but it did it on 11.4 too, and I think maybe 1.8 too. |
| 19:18.42 | Penguin | I haven't had the opportunity to use chan_skinny on >1.8 yet. |
| 19:18.43 | lvlinux | seems to work ok, but fills my console window with the messages "Starting Skinny session from 172.16.1.202" and then 2-3 secs later "Ending Skinny session from 172.16.1.202 (bad input)" |
| 19:19.04 | lvlinux | calls RX/TX fine though |
| 19:19.05 | Penguin | But on <=1.8, it sucks. For that, I use chan_sccp-b. |
| 19:21.02 | lvlinux | I've heard about chan_sccp but I'm using Astlinux on this machine, so it would be a total pain to compile it... |
| 19:22.10 | lvlinux | Astlinux rocks otherwise btw |
| 19:30.43 | *** join/#asterisk bkruse (~Adium@24.42.229.8) |
| 19:37.24 | Vendigroth | Anyone knows how to reduce the "Dropping incompatible voice frame" notices? every time a call comes in one gets logged. |
| 19:37.54 | Martin` | Hello ppl. I've a question. I have now asterisk 11 running. Now I want to forward a call to my mobile phone. but When I do not pickup my mobile phone it goes to voicemail. I Don't want that ppl who call my asterisk server get my mobile voicemail. What is the best way to protect it? |
| 19:38.16 | Martin` | I had the followme method. so I've to press 1 before I have caller |
| 19:38.31 | Martin` | is this still the best way to do it? |
| 19:38.51 | [TK]D-Fender | Martin: yes |
| 19:39.51 | Martin` | ok |
| 19:40.10 | *** join/#asterisk navaismo (~navaismo@189.241.101.4) |
| 19:44.48 | lvlinux | [5~[5~ |
| 19:45.19 | lvlinux | sry |
| 19:45.27 | lvlinux | wrong window... |
| 19:45.53 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 19:46.53 | mjordan | anonymouz666: if you're seeing a crash, DEBUG_THREADS doesn't help. DEBUG_THREADS only helps with finding deadlocks. If you have crashes, DONT_OPTIMIZE and BETTER_BACKTRACES are very helpful, and have only a minimal impact on performance |
| 19:52.13 | anonymouz666 | alright |
| 19:52.21 | anonymouz666 | will rebuild the software |
| 19:53.43 | *** join/#asterisk Joel_re (~jr@103.20.64.231) |
| 19:54.01 | Joel_re | hey |
| 19:54.15 | Joel_re | does anyone think this is possible with asterisk - http://vimeo.com/3747070 |
| 19:55.06 | Joel_re | I feel it might be asterisk driving that "MegaPhone" system |
| 19:56.55 | [TK]D-Fender | Joel_re: Voice analysys would require you to write a dialplan app, etc to process, but the rest... sure.. it'll take the CALL. All the work is your own custom code though |
| 19:58.02 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 20:00.53 | Joel_re | [TK]D-Fender: yeah |
| 20:01.05 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 20:02.23 | Martin` | [TK]D-Fender: do you maybe know if it is possible to just continue the ring when I use followme? I don't want that woman who say "Please wait while I'm trying to locate the caller" :) |
| 20:02.30 | Martin` | I don't like woman |
| 20:03.09 | [TK]D-Fender | Martin: use the Macro or Gosub option in your Dial() yourself.... and forget that "followme" function |
| 20:03.27 | Penguin | I'm the caller. I don't need her to locate me when I call you. |
| 20:03.40 | Martin` | lol |
| 20:04.08 | Martin` | [TK]D-Fender: how can I do that? |
| 20:04.23 | Penguin | Did you ues FollowMe's d option? |
| 20:04.49 | Martin` | no |
| 20:04.55 | Penguin | You should try it once. |
| 20:05.56 | volga629 | what is asterisk default tos values |
| 20:06.00 | *** join/#asterisk dxd828 (~dxd828@host-92-24-127-29.ppp.as43234.net) |
| 20:06.06 | volga629 | <PROTECTED> |
| 20:06.08 | volga629 | <PROTECTED> |
| 20:06.10 | volga629 | <PROTECTED> |
| 20:06.17 | [TK]D-Fender | Martin: "core show application dial" <------------ |
| 20:06.41 | Martin` | Penguin: but now the muziek starts directly |
| 20:06.51 | Martin` | I want the ringing sound :P |
| 20:07.02 | Penguin | If it's not one thing you're complaining about, now it's another. |
| 20:07.25 | Penguin | volga629: That seems like defaults to me. |
| 20:08.03 | Martin` | Penguin: sorry :P |
| 20:08.15 | Penguin | martin`: Do you also need it to hold your hand while it calls you? |
| 20:08.48 | Martin` | if it was possible |
| 20:09.05 | Penguin | martin`: Do you want normal followme behavior minus announcement, minus music, plus ringing? |
| 20:09.26 | Penguin | And the call recipient needs to press 1 to accept? |
| 20:10.02 | Martin` | yes/ checking if how to do it with macro instead followme. I don't need the followme features |
| 20:10.08 | Penguin | How do you determine if followme needs to be used? |
| 20:10.09 | Martin` | it is only the pressing one before conenct |
| 20:10.10 | Martin` | connect |
| 20:10.27 | Penguin | It's easy. I'll try to help you in a few minutes. |
| 20:10.33 | outtolunc | thinks it would be nice to be able to set the order of http handlers |
| 20:11.04 | Penguin | martin`: Under what condition do you want to followme? |
| 20:11.45 | Martin` | It forwards to my mobilephone, I don't want that it will come in my voicemail so I have to press 1 first to protect that |
| 20:12.25 | Penguin | You want the call to go to your mobile? But you do not want to press 1 to accept? |
| 20:12.51 | Penguin | or you do want to press 1 to accept? |
| 20:13.01 | Martin` | I want to press one |
| 20:13.19 | Penguin | You want the call to go to your mobile, but you do not want the caller to reach your mobile's voicemail? |
| 20:13.33 | Martin` | yes |
| 20:13.38 | Penguin | Okay. |
| 20:13.43 | volga629 | AF41 DSCP 34 is this correct ? |
| 20:13.45 | volga629 | = |
| 20:13.54 | Penguin | martin`: Under what condition does the call go to your mobile? |
| 20:14.09 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 20:14.32 | anonymouz666 | the SIP devices must support PAI/RPID header to be able to use connectedline updates? |
| 20:14.37 | Penguin | Are you enabling call forwarding on the desk phone? Are you only sending to mobile if the call is NOANSWER after ringing the desk phone? |
| 20:14.48 | Martin` | when somebody calls my phonenr it forwards to my deskphone and mobile phone |
| 20:15.02 | Penguin | That isn't forwarding. |
| 20:15.37 | Penguin | Do you want the call to ring both your desk phone and mobile phone at the same time? |
| 20:15.53 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.59) |
| 20:16.32 | Martin` | yes |
| 20:16.43 | Martin` | http://moo.net/code/callscreen.html maybe this is what I need? |
| 20:18.49 | Penguin | Kind of. |
| 20:20.18 | Penguin | If you need to ring the desk phone without having to press 1 to accept the call, and you want to require pressing 1 to accept it on the mobile, you can't call both phones from the same Dial, or the macro will run when you answer both phones. |
| 20:21.01 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 20:21.10 | Penguin | You'll need to use a local channel and a separate piece of dial plan to make it so one does and one doesn't use screening. |
| 20:21.54 | Martin` | ok |
| 20:22.00 | Martin` | already have that |
| 20:22.01 | Martin` | :) |
| 20:22.45 | Martin` | only need to add macro then :) |
| 20:22.54 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 20:29.08 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.59) |
| 20:32.01 | Martin` | <PROTECTED> |
| 20:32.50 | Penguin | Why would you need to set it? If someone calls you, there should already be callerid on the channel. |
| 20:33.52 | Martin` | I'm testing now, my callerid is now 2000 |
| 20:34.10 | Martin` | my sip provider does not accept to use that als callerid |
| 20:34.20 | Penguin | Oh, you had an internal callerid number on the channel. |
| 20:34.51 | *** join/#asterisk dar123 (~DAR@77.30.146.69) |
| 20:35.03 | Penguin | I have externalCID and internalCID for phones, plus checks in place to prevent that. |
| 20:41.35 | *** join/#asterisk cmendes0101 (~cmendes01@office.phone.com) |
| 20:41.53 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.59) |
| 20:43.04 | Martin` | Penguin: ok, I need to check it before it goes to oudside, if it is a small nummber set it :) |
| 20:43.21 | Martin` | but I want to be able to use other callerids too. it is fun :P |
| 20:43.38 | Martin` | call to somebody and uses his own callerid :P |
| 20:43.43 | Penguin | Are you in the US? |
| 20:44.38 | Martin` | no, the netherlands |
| 20:44.56 | Martin` | it is an old macro, need to change a lot :P |
| 20:45.17 | Penguin | Yeah, there are some problems in it. |
| 20:47.46 | Martin` | I have it working now. but when I wait to long it connects :P |
| 20:48.57 | Penguin | I think that means it's wrong. |
| 20:51.58 | karl-s | I think i'm missing something, isn't G.729 passthrough as simple as ensuring allow=g729 is enabled on the phone and trunk? (knowing that both support it) |
| 20:52.33 | karl-s | yet on the SDP message from Asterisk to the Trunk, g.729 is stripped from the SDP message leaving only alaw and ulaw... |
| 20:54.53 | Martin` | [Nov 21 21:54:05] WARNING[2259][C-0000000f]: ast_expr2.fl:470 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '=', expecting $end; Input: |
| 20:54.56 | Martin` | <PROTECTED> |
| 20:54.56 | Martin` | ah thats wrong |
| 20:55.52 | Penguin | martin`: http://pastebin.com/qXeWTwzN |
| 20:56.12 | dar123 | where do we set the value of debug and verbosity to 10, is it in the logger.conf? |
| 20:56.46 | *** join/#asterisk LeandroDardini (~leandro@ip-224-67.sn2.eutelia.it) |
| 20:56.53 | *** join/#asterisk cmendes0101 (~cmendes01@office.phone.com) |
| 20:57.01 | LeandroDardini | hello all |
| 20:58.20 | Penguin | dar123: On the CLI or in asterisk.conf |
| 20:58.49 | Penguin | You almost never need to set debug level above 0 in the conf. |
| 20:59.07 | Martin` | Penguin: tnx, that works |
| 20:59.35 | lordvadr | If I say "pri set debug file whatever", and turn it on, I still see the output on the cli verbose output. Is there anyway to turn that off? |
| 20:59.41 | Martin` | I like the confbridge-join sound, then you are sure that you are connected :) |
| 21:00.45 | LeandroDardini | I am still trying to dial directly using a format like Dial(SIP/username:password@some.host.com/number) but it seems no more possible with asterisk 11.6 ... do you know the correct syntax? |
| 21:05.44 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 21:09.30 | dar123 | <PROTECTED> |
| 21:10.15 | dar123 | i ran core set debug 9 and core set verbose 9 |
| 21:10.22 | navaismo | hey dar123 sorry yesterday suffer a power outage, still nothing on the cli? |
| 21:10.33 | dar123 | nope |
| 21:10.49 | dar123 | still the same issue |
| 21:12.00 | *** join/#asterisk JonathanD (~JonathanD@freenode/staff/jonathand) |
| 21:15.40 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 21:17.30 | navaismo | if you disconnect the cable and the run lsdahdi the channel is marked as red? |
| 21:19.52 | dar123 | yes , when i remove the cable i see it as red |
| 21:21.17 | dar123 | http://pastebin.ca/2477792 |
| 21:24.20 | navaismo | hmmm |
| 21:24.49 | navaismo | when you call in the full log with core set debug 3 show anything? |
| 21:24.59 | dar123 | nope |
| 21:26.42 | navaismo | jesus |
| 21:27.23 | dar123 | somehow i think driver not handing the call to asteirsk |
| 21:27.33 | navaismo | dahdi version? |
| 21:27.36 | dar123 | not sure how to put in better words |
| 21:27.57 | dar123 | 2.7 |
| 21:28.51 | [TK]D-Fender | checkout time, BBIAB |
| 21:30.18 | *** join/#asterisk g_r_eek (~g_r_eek@176.92.243.222) |
| 21:30.23 | *** join/#asterisk ghost75 (~quassel@dslb-088-066-182-024.pools.arcor-ip.net) |
| 21:31.36 | ghost75 | somebody using cidname in mysql ? |
| 21:31.38 | outtolunc | dar123: that lsdahdi showed port 3 in red alarm, what does dahdi_tool show |
| 21:34.38 | dar123 | yes it's on port 3. i just was running "dahdi_monitor 3 -vv" and i could see the audio activity |
| 21:35.16 | outtolunc | you already know 'as a basic channel' it works from when you did the originate test wimpy had you do.. |
| 21:35.17 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 21:35.43 | outtolunc | you are now working on why the 'line' is not being seen by it |
| 21:35.56 | *** join/#asterisk saxa (~saxa@187-127-186-95.user.veloxzone.com.br) |
| 21:37.59 | outtolunc | pretty sure wimpy was also trying to get you to understand how fxo and fxs must be set.. and got frustrated that he thought you had them backward |
| 21:38.50 | outtolunc | the 'line' from the wall is FXO which means that port on the digium card needs to be set to FXS to talk to it.. so if the 'line' from the wall is plugged into port 3, you have them backward |
| 21:38.57 | dar123 | i ran the command but didn't know what the output meant http://pastebin.ca/2477797 |
| 21:39.31 | dar123 | yeah i got the figured out latter , so signaling is ok no |
| 21:39.33 | dar123 | now |
| 21:39.49 | outtolunc | is your line plugged into port 4 now? |
| 21:40.56 | outtolunc | what you need to do is plug a phone into port 3, go offhood and punch some digits and watch the console |
| 21:40.59 | dar123 | my line is plugged into 3 |
| 21:41.13 | outtolunc | which you have set to FXO |
| 21:41.23 | dar123 | port 3 is fxo, yes right |
| 21:41.34 | outtolunc | an FXO (line) will not talk to a FXO (port) |
| 21:42.06 | outtolunc | <PROTECTED> |
| 21:42.35 | outtolunc | that is a port on my card (quad t1) for talking to a FXS channel bank |
| 21:42.50 | outtolunc | see how one side is FXO, the other side is FXS |
| 21:42.56 | *** join/#asterisk beardy (~beardy@unaffiliated/beardy) |
| 21:43.15 | outtolunc | (channel bank isn't plugged in, hense the red) |
| 21:44.42 | dar123 | 3 FXO FXSKS (In use) (EC: MG2 - INACTIVE) |
| 21:46.39 | dar123 | 4 FXS FXOLS (In use) (EC: MG2 - INACTIVE) |
| 21:47.17 | dar123 | my PSTN line goes in port 3, my analog phone goes in port 4. correct |
| 21:47.45 | outtolunc | right, it's like your system.conf and chan_dahdi.conf are not matching |
| 21:48.07 | dar123 | let me just go through them once more |
| 21:49.48 | outtolunc | as an example.. my system.conf has fxols=73-96 |
| 21:50.07 | outtolunc | chan_dahdi.conf has signalling=fxo_ls |
| 21:50.24 | outtolunc | the actual device (channel bank) is an 24 port FXS |
| 21:50.58 | dar123 | http://pastebin.ca/2477798 |
| 21:52.52 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 21:52.55 | outtolunc | i bet if you pulled the card, swapped the modules your issue would go away |
| 21:53.45 | dar123 | can u plz tell me what config i can modify to fix it i pasted both system.conf and chan_dahdi.conf |
| 21:53.50 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:53.50 | outtolunc | hmm 12-beta isn't showing the status cleared after firing up my CB |
| 21:54.05 | saxa | Something went wrong. Try running dahdi_cfg -vvv. |
| 21:54.21 | saxa | DAHDI_SPANCONFIG failed on span 1: Invalid argument (22) |
| 21:54.32 | saxa | any idea why do I get this ? |
| 21:55.48 | saxa | http://pastebin.com/kudRSxmZ <- here is my system.conf |
| 21:56.02 | saxa | which iirc was working |
| 21:57.13 | saxa | ok I got it, forget about it |
| 21:57.25 | saxa | I need to pass the correct mode type to the card |
| 21:57.31 | saxa | I reinstalled my system |
| 22:00.31 | outtolunc | never seen a cas= before (its not even in my zaptel/system.conf's |
| 22:02.17 | navaismo | R2 signalling |
| 22:02.20 | navaismo | ?¿ |
| 22:03.26 | navaismo | saxa did you installed libopenr2? |
| 22:03.56 | saxa | navaismo: yes, I was missing the module loading options in modprobe.d |
| 22:04.26 | saxa | since i reinstalled everything i forgot to add this option manually |
| 22:04.44 | saxa | option wcte13xp default_linemode=e1 |
| 22:04.57 | saxa | its ok now, already up and running |
| 22:05.19 | saxa | I think I lost my Grandstream GXP2120 |
| 22:05.22 | saxa | :( |
| 22:06.00 | saxa | I had it working today, but with an old firmware, and then I added the correct link for the firmware update and rebooted the phone |
| 22:06.23 | saxa | the phone downloaded everything, and now is stuck at the boot. |
| 22:06.33 | saxa | sorry for the OT |
| 22:14.06 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.66) |
| 22:21.07 | *** join/#asterisk nyc-h0st (~nyc-h0st@84.40.101.126) |
| 22:25.28 | nyc-h0st | all, for some reason incomming calls from a SIP trunk do not dial the 'start' extension in my context for it but rather try to dial the phone number for the trunk...very odd |
| 22:27.31 | WIMPy | That's not odd at all, but what should hopefully happen. |
| 22:28.49 | nyc-h0st | for some reason i recall it dialing 'start' in previous versions, could be just me |
| 22:29.24 | WIMPy | Or your register statement was missing your extension. |
| 22:30.12 | nyc-h0st | okay, so say my trunk # is 212-555-1212, call somes in i have exten => _nxxnxxxxxx catching it |
| 22:30.46 | nyc-h0st | and then dialing an internal extension, it then swaps the caller phone # with the trunk # on the receiving end |
| 22:31.53 | nyc-h0st | ie the sip clients see 212-555-1212 as calling them not the original caller phone #, and i do not replace caller id anywhere, its as if Asterisk swaps out the caller number when i handle the _NXXNXXXXXX exten for the incomming call from the trunk |
| 22:32.48 | pabelanger | asterisk can basically do anything, you just need to tell it how to do it |
| 22:33.11 | WIMPy | But you have to tell it to change something. |
| 22:33.18 | [TK]D-Fender | * swaps nothing. |
| 22:33.27 | [TK]D-Fender | that you don't tell it to |
| 22:34.02 | [TK]D-Fender | You should be looking at the actual call. |
| 22:34.08 | [TK]D-Fender | This will answer all of this for you |
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| 23:01.40 | nyc-h0st | i think i figured it out, this is turning into a two leg call, ie same as A dials B then B transfers to C but C sees B instead of A |
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