| 00:00.31 | marsmars | that was on my ISDN to IAX side (Fritz!Box with Freetz) |
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| 00:30.29 | TazzNZ | hey tm1000 you around ? |
| 00:30.35 | tm1000 | yes |
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| 00:42.51 | jameswf | That was easy |
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| 01:20.17 | tm1000 | jameswf: pms for the winz |
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| 01:24.32 | phix | hey, i am having issues where the caller can hear voice but the receiving can't |
| 01:25.09 | phix | s/ving/ver/ |
| 01:25.24 | phix | lol |
| 01:25.27 | phix | regex fail |
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| 01:32.12 | phix | when do i need to have nat=force_rpot,comedia / (old nat=yes)? For my sip peers? internal handsets? handsets that may change from being internal to external? |
| 01:32.31 | phix | I also set externip (as my sip box is behind NAT but with port forwarding) where do I put that? in global? |
| 01:33.40 | phix | also I have qualify to yes, and one of my peers is unreachable however dial still rings it, the phone picks up but cannot hear anything |
| 01:34.17 | phix | isn't the point of qualify so the asterisk box knows which extensions are unreachable and hence not try to call them/> |
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| 01:42.49 | phix | ? |
| 01:46.52 | phix | ? |
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| 02:11.06 | phix | for seqno 102 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions |
| 02:11.09 | phix | :/ |
| 02:11.11 | phix | That is not helpful |
| 02:11.48 | phix | ok I need a better example on how to set this ip |
| 02:11.50 | phix | up* |
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| 02:18.34 | phix | ? |
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| 02:48.13 | phix | ? |
| 02:48.28 | phix | Any one awake? |
| 02:48.58 | WIMPy | Yes |
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| 02:49.31 | phix | Hey WIMPy! |
| 02:57.39 | carrar | we're hunting wabbits |
| 02:58.58 | WIMPy | Are they edible? |
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| 03:53.03 | *** join/#asterisk Lepon (Lepon@14-202-40-1.tpgi.com.au) |
| 03:53.31 | Lepon | Greetings fantastic Asterisk people |
| 03:54.10 | Lepon | Hoping someone can point me in the right direction, I would like to play a message to both channels when the caller presses a key combination |
| 03:54.42 | Lepon | At the moment I have been using a feature to call a macro that retrieves the right sounds file and uses playback() to play it |
| 03:55.00 | Lepon | unfortunately only one side of the call hears the sound, I need both parties to hear it |
| 03:55.14 | Lepon | any ideas of a direction I should investigate? |
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| 05:46.34 | Lepon | Hey all, anyone know a way to playback a sound file to both ends of a call at once (ie: caller A dials caller B -> they talk -> at somepoint I then want to play both of them a sound file without interrupting their call -> then they continue talking) |
| 05:49.59 | [TK]D-Fender | Lepon: When& how do you expect to have this action decided? |
| 05:52.42 | Lepon | hopefully when one of the users presses a key combination |
| 05:53.02 | Lepon | I was trying to use features.conf to execute a macro but I couldnt only get it to output to one channel and blank the other |
| 05:57.57 | [TK]D-Fender | call a macro in there and Originate a new call to a Local Channel. This will connect to the issuer's channel with Chanspy + Barge and the dialplan end will hit the playback. |
| 05:59.10 | Lepon | fantasic thank you. Will give that ago. I had been trying chanspy but not with a new call |
| 05:59.12 | Lepon | cheers |
| 06:00.33 | [TK]D-Fender | Needs to be a new call so that it can hook in. |
| 06:00.41 | [TK]D-Fender | while both ends a bridged |
| 06:00.44 | [TK]D-Fender | are* |
| 06:01.52 | Lepon | thanks |
| 06:02.51 | [TK]D-Fender | You're welcome |
| 06:10.13 | [TK]D-Fender | bed time... |
| 06:10.15 | [TK]D-Fender | heads off |
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| 10:59.06 | michael_work | if i have extension(sip device) configurated at realtime with dtls and if the key info are provided in [general] (config file) and are not configurated in realtime (they do not exist or they do exist but have NULL) it would not use the one from general and the empty fingerprint would be send which would lead to error |
| 10:59.18 | michael_work | is it feature or should i report it? |
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| 11:03.11 | file | the DTLS options are not supported in the general section |
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| 11:13.52 | michael_work | file, why? |
| 11:14.21 | michael_work | at least dtls key and ca file settings should be there |
| 11:14.34 | file | because I didn't put them in there? |
| 11:14.56 | michael_work | so if i set dtlsenables on peer and do not specify keys it would take default settings |
| 11:15.15 | file | normally in a config file you would just use templates, but in a database you'd have to duplicate |
| 11:15.34 | michael_work | file, okay, i'll rephrase: what is the reason you did not put them there? lack of time or on purpose |
| 11:16.03 | michael_work | yes, template support on realtime or dtls in general should be implemented in my opionion |
| 11:16.04 | file | I tried to minimize the impact to chan_sip as much as possible, and making stuff work in the general section and have it inherited and then overwritten optionally is not great in chan_sip |
| 11:16.37 | michael_work | we had "template" colomn in DB and i thouhgt it works. it took me a while to understand it doesn't |
| 11:16.58 | michael_work | as well as it took me to understand that i need to specify key for each peer in DB |
| 11:17.37 | michael_work | as it did not send any error (maybe the check if the dtls enabled and no keys provided it should through some warning and do not turn dtls on? ) |
| 11:18.39 | michael_work | file, it might be good or not, but it's how it works with most settings and i think it should be done to all or non |
| 11:19.06 | michael_work | there is even functio set_peer_defaults already anyway. |
| 11:20.01 | file | a check could be done, or it could be supported in general if someone wanted to do it |
| 11:20.33 | michael_work | file so if patch submitted it mostly would be accepted? |
| 11:20.42 | file | it would have to go through code review and be tested |
| 11:21.06 | file | but sure |
| 11:21.55 | michael_work | i have one patch in review which got few rejects i fixed and nothing happens to it (i'm already goinf to apply it to productive to new servers, but nothing on review) |
| 11:22.02 | michael_work | should i push "ship it" button? |
| 11:22.21 | file | no - shipping your own stuff is not useful, someone else has to look at it |
| 11:22.27 | file | what review? |
| 11:22.37 | michael_work | i maybe will try to do it (depending on time and depending if i'll be sure i can do it in proper way) |
| 11:22.41 | file | and a ton of people have been focused over the past while on making sure features get up for review, as feature freeze was last week |
| 11:22.59 | michael_work | https://reviewboard.asterisk.org/r/3607/ |
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| 11:23.06 | michael_work | it's been there month |
| 11:23.17 | michael_work | last update from 3 weeks ago |
| 11:24.04 | file | I'll see if I can nudge someone |
| 11:24.56 | michael_work | it's small patch and would be nice to have it for others. as well as it would give me for motivation and believe in myself to try and make more patches :P |
| 11:25.08 | michael_work | thanks |
| 11:26.08 | file | although does that not work with static realtime? |
| 11:26.18 | file | or you wrote it because you want both file and realtime? |
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| 11:28.13 | michael_work | file, what does static realtime means? |
| 11:28.26 | file | you can put config files into realtime and have them loaded from there |
| 11:28.37 | michael_work | as text? blob? |
| 11:28.40 | file | it's read in when you load or reload the module |
| 11:28.59 | file | as text in columns |
| 11:29.31 | file | https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration?src=search#RealtimeDatabaseConfiguration-Twomodes:StaticandRealtime |
| 11:29.32 | michael_work | i wanted realtime as it's easier to sync and validate data |
| 11:29.38 | file | someone put the table spec at the bottom |
| 11:29.44 | michael_work | to have single config (one mysql) for several servers |
| 11:29.57 | file | pros and cons, pros and cons |
| 11:30.07 | michael_work | sure, that's one it's optional |
| 11:30.32 | michael_work | it's does not change, but add current capabilities :P) |
| 11:32.51 | michael_work | it's easier to handle and validate if you have a lot of enries |
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| 11:52.15 | phpboy | cdr_mysql has been depricated, what do you use instead for CDR mysql? |
| 11:53.13 | WIMPy | odbc |
| 11:53.32 | phpboy | :( |
| 11:53.44 | phpboy | I'm not a fan of obdc |
| 11:53.56 | phpboy | it is what is is I guess |
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| 12:13.55 | michael_work | phpboy, you can use php |
| 12:14.00 | michael_work | agi + php |
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| 12:39.19 | dipanjan | I have a prompt in gsm format. It plays fine when I test with a soft phone like Twinkle. But when I test it with a real phone the prompt stops playing at a particular place and goes to the next prompt. Can anybody say why this is happening? |
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| 13:00.43 | bmurt | hey ya'll we are looking at separating the sip and RTP traffic to be more direct between endpoints, some behind NATs and some not, but i came across mediaproxy and rtpproxy. i also came across asterisk's own canreinvite, etc. would asterisk's side need to be used inconjunction with the proxies or would the config be done at the proxy level? |
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| 13:27.05 | phpboy | michael_work: that's a little silly |
| 13:27.11 | phpboy | will stick to odbc |
| 13:27.23 | phpboy | no point in introducing an additional point of failure |
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| 13:41.06 | lbarth | Hi, i try to replace a commercial anouncement server with asterisk. I recieve a lot information sin the initial Request URI eg INVITE sip:annc@192.168.16.5;play=file%3Anull.wav;group_id=1;mix=0;tone_id=10001;ntimes=1;next_tone=-1 SIP/2.0 I know i can read with the sip_header method sip headers, but how can i get the request URI? |
| 13:42.04 | anonymouz666 | If I understand correctly request URI will be the ${EXTEN} |
| 13:42.17 | anonymouz666 | annc,1,... |
| 13:42.26 | lbarth | but it does not inclue parameters |
| 13:42.39 | anonymouz666 | correct |
| 13:43.02 | lbarth | and i need to know in dialplan that i need to play anouncement 10001 from tonegroup=1 |
| 13:43.41 | [TK]D-Fender | lbarth: Show us the call |
| 13:44.01 | [TK]D-Fender | with SIP debug |
| 13:45.38 | anonymouz666 | lbarth: it will be easier to you if you put this as a custom header not params. |
| 13:45.46 | lbarth | http://pastebin.com/7Z1KMgJP thats the invite |
| 13:46.11 | lbarth | the rpoblem ist that rfc4240 defines params for anouncements :-( |
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| 14:05.49 | Katty | SUPERLATIVES. |
| 14:07.12 | [TK]D-Fender | lbathe FULL call so we can see what * does with it... |
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| 15:05.30 | hforjehikiasq6n2 | Can asterisk terminate an encrypted IAX2 audio stream, decrypt it, and send it to my provider in the clear? I only really care about encryption from my end devices to the pbx, since it goes over the internet. |
| 15:06.38 | WIMPy | It will always be decrypted. It's only encrypted for the pne link. |
| 15:06.45 | Qwell | isn't your provider on the Internet too? O.o |
| 15:07.22 | hforjehikiasq6n2 | Yeah, but if I'm in a coffee shop I'd rather not have Jimmy McScriptKiddy sniff my phone calls |
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| 15:08.50 | hforjehikiasq6n2 | I tried forceencrypt=yes but then the call wouldn't go through at all |
| 15:09.31 | WIMPy | Are you using an IAX softphone? Or what's the setup? |
| 15:09.48 | hforjehikiasq6n2 | iax2 show peer <line number> shows aes128 encryption and keyrotate |
| 15:09.52 | hforjehikiasq6n2 | Yeah, I'm using Zoiper |
| 15:10.03 | hforjehikiasq6n2 | on my android phone |
| 15:10.25 | WIMPy | Does it support encryption? |
| 15:11.13 | hforjehikiasq6n2 | If iax2 show peer says it does, doesn't it? Or is that just a line capability that hasn't been verified yet? |
| 15:11.29 | WIMPy | Asterisk does, but does Zoiper? |
| 15:11.37 | WIMPy | Or maybe only in the paid version? |
| 15:14.41 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 15:15.45 | hforjehikiasq6n2 | I got the premium one that supports SIP TLS, but I honestly can't tell if it supports IAX2 encryption |
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| 15:17.18 | hexanol | I have a question about context inclusion in dialplan |
| 15:17.34 | hexanol | is it safe for context A to include B and context B to include A ? |
| 15:17.57 | hforjehikiasq6n2 | authenticate_verify: Call Terminated, Incoming call is unencrypted while force encrypt is enabled. |
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| 15:24.19 | *** join/#asterisk [Outcast] (~Outcast]@50-195-8-201-static.hfc.comcastbusiness.net) |
| 15:27.01 | [TK]D-Fender | hexanol: That's pretty pointless as a design. NOt sure how far * will protect against circular searches. |
| 15:27.28 | hexanol | [TK]D-Fender: yeah I know this is pointless, but I have seen it on some system, and was just wondering |
| 15:27.39 | hexanol | how safe it is actually |
| 15:27.46 | [TK]D-Fender | no idea. go test.... |
| 15:27.49 | hexanol | i.e. is asterisk coded in a way that this is safe |
| 15:27.59 | hexanol | well it seems to work |
| 15:28.03 | hexanol | but I wonder if it's mostly luck |
| 15:28.28 | [TK]D-Fender | No, the outcome is predictable I'm sure. Maybe not what you expect, but predictable |
| 15:32.00 | *** join/#asterisk anonymouz666 (~anonymouz@179.236.136.49) |
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| 15:39.16 | rmudgett | I'd expect that if you were searching for something that was in the dialplan it would work fine. However, if it wasn't in the dialplan then the circular include is likely to cause an infinite loop with bad consequences. |
| 15:39.32 | Katty | fender bender. |
| 15:42.31 | [TK]D-Fender | crashes |
| 15:42.43 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 15:47.06 | drale | I setup a test asterisk VM to handle an pattern of extensions, say 87XX. but send anything else back to the main asterisk. accidently created a loop :) |
| 15:47.36 | drale | Got to account for all scenerio in dialplan |
| 15:49.17 | hforjehikiasq6n2 | also, be careful where you put your recursive include? If it's at the beginning of both dial plans, you'll probably have a bad day |
| 15:50.25 | WIMPy | It doesn't matter where you put an include, only the order of includes. |
| 16:10.01 | [TK]D-Fender | There is no "careful" with a recursive include. It is a novice mistake and should never be made. |
| 16:20.05 | *** join/#asterisk igcewieling1 (~ewieling@ip98-183-26-100.pn.at.cox.net) |
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| 16:35.19 | *** part/#asterisk igcewieling1 (~ewieling@ip98-183-26-100.pn.at.cox.net) |
| 16:47.03 | *** join/#asterisk dralezero (~drale@unaffiliated/drale) |
| 16:50.08 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 16:53.07 | gavimobile | does 'sip show channels' show anything other than concurrent calls? |
| 16:53.45 | gavimobile | or is it possible to see activity there with a connection to and from a trunk |
| 16:55.28 | dralezero | what about 'core show channels' it shows the application the channel is on |
| 16:56.08 | gavimobile | dralezero: ill monitor this too |
| 16:56.41 | file | define "activity" and "connection" |
| 16:57.41 | gavimobile | file, well doing a sip debug I get this randomly. it looks like attackers trying to make a call but it might be something small like the connection between my pbx and the trunk http://pastebin.com/drMRrq43 |
| 16:58.12 | file | that's Asterisk sending an OPTIONS message to determine if a remote server is alive or not |
| 16:58.40 | gavimobile | ok well ipcomfs has a ~3600 at the end of the register string |
| 16:59.03 | gavimobile | shouldn't it be doing this less frequently? also it clearly says FROM & TO as if its initiating a call |
| 16:59.19 | file | From and To are part of SIP signaling, and this is NOT a registration |
| 16:59.34 | file | it's done as a result of the "qualify" functionality |
| 16:59.35 | gavimobile | file: fyew |
| 16:59.46 | gavimobile | qualify is set to yes |
| 16:59.51 | gavimobile | I always set it to yes |
| 17:00.00 | file | then that is why those are going out |
| 17:00.11 | gavimobile | also, why don't my other trunks do this as frequently |
| 17:00.20 | gavimobile | nevemind |
| 17:00.31 | gavimobile | sip set debug peer onlythisone |
| 17:00.35 | gavimobile | silly me |
| 17:01.07 | gavimobile | file: thanks for the clarification!! I appreciate it |
| 17:01.34 | gavimobile | oh, while on this topic, can I change the frequency of this? |
| 17:01.39 | gavimobile | once an hour to check? |
| 17:02.29 | *** part/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 17:02.29 | file | it can be configured in sip.conf, and once an hour isn't exactly useful |
| 17:03.04 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 17:03.13 | gavimobile | sorry, I closed the window accidently |
| 17:03.28 | gavimobile | could you repeat that if you said something |
| 17:03.45 | file | it can be configured in sip.conf, and once an hour isn't exactly useful |
| 17:04.06 | gavimobile | true, but easier to debug other issues |
| 17:04.21 | gavimobile | what's the downside of doing this less frequently? |
| 17:04.48 | file | you won't know if the remote server is unreachable, in which case any calls placed to it will probably hang for oh 30 seconds before failing |
| 17:09.42 | *** join/#asterisk _grim_ (~grim@23.30.198.161) |
| 17:12.37 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 17:13.03 | _grim_ | I need some help clearing my understanding of variables.... a variable is set on a channel, call changes channel, variable is gone. prepended by a single _ and it's carried from parent to child channel, but not past. prepended by 2 __ and it's carried forever.... Am I correct in assuming it's carried forever during that call. so if I set varUUID on call A to value A, and that call goes from macro to queue to macro to gosub....etc... it will always carry the |
| 17:13.04 | _grim_ | <PROTECTED> |
| 17:14.18 | _grim_ | assuming I use set(__varUUID... with 2 _ |
| 17:19.32 | gavimobile | file: I see, what about performance. does all this frequent activity put extra stress on the pbx? |
| 17:19.40 | file | not a ton |
| 17:21.30 | gavimobile | to me, every hour to check if there is still a connection is sufficent for me |
| 17:21.43 | gavimobile | what's asterisk 11's default |
| 17:21.51 | gavimobile | do you know off hand |
| 17:22.10 | file | 60 seconds. |
| 17:36.07 | *** join/#asterisk bkruse (~Adium@24.42.207.11) |
| 17:38.44 | gavimobile | file: hrm.. what's the variable to change it please |
| 17:39.05 | file | search sip.conf.sample for qualify |
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| 18:03.54 | Katty | Wash, tell me i'm pretty. |
| 18:05.11 | [TK]D-Fender | Were I unwed, I would take you in a manly fashion... |
| 18:05.22 | Katty | ty, fender. i feel better now. |
| 18:05.32 | Katty | i have faults on all 4 power supplies on my blade server. |
| 18:05.39 | Katty | and intel said my SLA is 2 hours. |
| 18:05.58 | Katty | i think i will take up a new thumb twiddling hobby while all my servers are down! |
| 18:11.40 | mjordan | that ... sucks. |
| 18:19.27 | *** join/#asterisk isgastis (~isgastis@86.100.160.165) |
| 18:22.19 | voip | sounds like lightning / bad gruonds or something like that |
| 18:22.24 | voip | surge supression ftw |
| 18:33.38 | bmurt | anyone offloading RTP to mediaproxy or rtpproxy? |
| 18:34.27 | *** join/#asterisk JasonL (~jason@216.223.114.3) |
| 18:41.12 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 18:44.02 | JasonL | Hello, does anyone know if there is a way to know if asterisk is running with core dumps enabled? |
| 18:49.48 | malcolmd | JasonL: cat /proc/`pid of asterisk`/cmdline |
| 18:52.23 | *** join/#asterisk Defraz (~Defraz@24-117-69-71.cpe.cableone.net) |
| 18:53.07 | JasonL | malcolmd: thanks, it doesn't show as running core dump there... I had previously set dumpcore = yes in asterisk.conf, but then I realized that after running for a while it would crash and output a core dump because the system would run out of memory. I disabled dumpcore in asterisk.conf, but it still seems to be using high memory, still crashing and still outputting the core file... Asterisk |
| 18:53.08 | JasonL | is not being started with -G. |
| 18:54.48 | malcolmd | -G is for the group to start with, -g is for dumping a core in case of a crash |
| 18:55.15 | JasonL | oops, i meant it is not running with -g argument |
| 19:02.30 | JasonL | is there a way to kill asterisk in a way that it would output a core dump if it was enabled? |
| 19:13.00 | *** join/#asterisk n3hxs (~Ed@pool-96-245-157-123.phlapa.fios.verizon.net) |
| 19:21.00 | malcolmd | kill -3 `pid` ? |
| 19:23.43 | JasonL | that didn't work, but kill -4 did :) |
| 19:41.09 | TazzNZ | morning all |
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| 21:01.07 | Katty | paces. |
| 21:02.09 | invader | So does anyone know of a magical place I can download firmware 4.X for Polycom phones? |
| 21:02.26 | invader | 3.1.3 does not support LLDP and I require it. |
| 21:03.23 | Katty | http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html |
| 21:03.37 | *** join/#asterisk bkruse (~Adium@64.89.97.113) |
| 21:03.45 | Katty | resumes pacing. |
| 21:03.54 | *** join/#asterisk drale (~drale@unaffiliated/drale) |
| 21:06.00 | invader | ..... |
| 21:06.08 | invader | hugs Katty |
| 21:06.13 | invader | goes back to work |
| 21:12.43 | [TK]D-Fender | invader: on polycom's site.... |
| 21:16.49 | invader | You know, I never checked, Why? I thought I would have to pay for it like Cisco. |
| 21:17.43 | [TK]D-Fender | no |
| 21:26.27 | *** join/#asterisk MarcoZink (~marcozink@189.146.108.107) |
| 21:27.03 | Chainsaw | invader: I know of a few vendors who paywall their stuff, but Cisco is definitely the worst. |
| 21:27.37 | Chainsaw | invader: Polycom on the other hand has all the firmware and the docs out there. This is why I buy their handsets, even though they're a bit dearer. |
| 21:27.51 | Chainsaw | invader: (They comply to standards better too) |
| 21:38.26 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 21:38.26 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 22:31.56 | Lepon | Good morning all (or good *insert time where you are* all) |
| 22:32.19 | Lepon | Just wondering if there is a way with Chanspy to whisper or talk to both channels of a call at once? |
| 22:33.10 | newtonr | one of the chanspy options is for barging, that should do the trick IIRC |
| 22:33.48 | newtonr | option 'B' |
| 22:33.52 | Lepon | I tried that like "exten => s,n,Chanspy(${ARG1},bw)" but it only played the sound on one channel |
| 22:34.02 | newtonr | check the help text again.. |
| 22:34.18 | newtonr | lowercase b is "Only spy on channels involved in a bridged call." |
| 22:34.24 | Lepon | r crap |
| 22:34.29 | Lepon | damn typos |
| 22:34.30 | newtonr | uppercase B is "Instead of whispering on a single channel barge in on both channels |
| 22:34.31 | newtonr | <PROTECTED> |
| 22:34.32 | Lepon | thank you |
| 22:34.36 | newtonr | no problemo |
| 22:37.36 | Lepon | awesome sauce I love you guys on here! |
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| 23:34.59 | Delphiworld-Away | hi guys |
| 23:35.13 | Delphiworld-Away | is asterisk 11 or 12 the current release? |
| 23:37.04 | [TK]D-Fender | Which number is bigger? |
| 23:37.35 | Delphiworld-Away | [TK]D-Fender: surely 12 but at i see 11 is LTS. |
| 23:37.58 | [TK]D-Fender | Yes, and Windows Vista is still "supported" |
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| 23:39.11 | TazzNZ | that is almost a quote for bash.org..... |
| 23:46.44 | Delphiworld-Away | anyway to enable modules without menuselect? |
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| 23:59.52 | Blue2000k | Evening |