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| 01:49.24 | WIMPy | yawns |
| 02:13.44 | volga629 | https://issues.asterisk.org/jira/browse/ASTERISK-17410 is any body know if patches was included in 10.7 can't find release note give 404 |
| 02:15.08 | ChannelZ | well the bug says "unresolved" so I'm going to say 'no' |
| 02:18.28 | volga629 | that I see [2012-09-20 22:00:58] VERBOSE[30688] chan_sip.c: Found RTP video format 117, [2012-09-20 22:05:19] VERBOSE[30688] chan_sip.c: Found video description format H264 for ID 99 |
| 02:18.46 | volga629 | I am trying troubleshoot this video issue |
| 02:19.01 | volga629 | 117 set on phone |
| 02:20.03 | volga629 | and all starts from this message WARNING[30688] chan_sip.c: ignoring 'video' media offer because port number is zero |
| 02:20.22 | ChannelZ | because the port number is probably 0 |
| 02:21.18 | ChannelZ | so either your device is giving a wacky port number in its offer or if you are still messing with conntrack_sip perhaps that is where the SIP packet is getting munged |
| 02:21.25 | volga629 | and can't really 100% determine if it is affected be reported bug, I capture with tcpdump rtp sessions never so port 0 all going above 10000 |
| 02:22.16 | volga629 | yes that possible I can start test tomorrow in Europe night hours |
| 02:22.17 | WIMPy | nf_conntrack_sip is a good thing. nf_nat_sip is the bad one. |
| 02:22.58 | WIMPy | It IS night here now. |
| 02:23.15 | volga629 | nf_conntrack_sip ports=5060 sip_direct_signalling=0 sip_direct_media=0 I modified shorewall today want to see if it make any different |
| 02:24.12 | volga629 | and nf_nat_sip is loaded too, but if understand correctly they should loaded together because one is helper ? |
| 02:24.29 | WIMPy | Not exactely. |
| 02:24.50 | ChannelZ | Shouln't nf_conntrack_sip watch the SIP traffic such that it can map incoming RTP without having to port-forward? Cuz last time I tried I could never get it to do anything. |
| 02:24.50 | WIMPy | The conntrack module does what the name says. |
| 02:25.07 | WIMPy | the nat module mangles the user data. DO NO USE IT. |
| 02:25.26 | WIMPy | Exactely. |
| 02:25.40 | ChannelZ | I'll have to screw with it again. Maybe I had something wrong. |
| 02:26.02 | WIMPy | I have no ports open for RTP. |
| 02:26.11 | volga629 | or I see this one I will unload, from vpn tunnel box to asterisk one hop no any nat needed |
| 02:27.25 | volga629 | even if on tunnel him self NAT traversal in place ? |
| 02:28.06 | WIMPy | I my experience nf_nat_sip is very good at making things worse. |
| 02:28.29 | ChannelZ | That one would be the one that would manipulate outgoing SIP to fudge IP addresses and port numbers then |
| 02:28.42 | WIMPy | At least with Asterisk. Let Asterisks nat support handle the NAT. It's better at it. |
| 02:28.52 | WIMPy | yes |
| 02:28.54 | ChannelZ | like for FTP |
| 02:29.38 | WIMPy | Well, it works quite well for FTP. |
| 02:29.59 | *** join/#asterisk SeRi (~wtf@c-98-200-53-71.hsd1.tx.comcast.net) |
| 02:30.09 | WIMPy | The other issue with nf_nat_sip is that it may magle the payload in the wrong direction. |
| 02:30.20 | volga629 | that from lsmod nf_conntrack_sip 29181 1 nf_nat_sip |
| 02:30.47 | volga629 | it loading as dependency |
| 02:33.16 | ChannelZ | maybe your shorewall script is, it doesn't happen here if I just modprobe nf_conntrack_sip |
| 02:33.35 | WIMPy | Off course not. It's the other way round. |
| 02:34.41 | volga629 | yes I am looking on modinfo and no dependency like nf_nat_sip |
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| 02:40.27 | volga629 | nf_conntrack 92343 34 nf_conntrack_sip, this look better, but where module line 0 is this normal nf_conntrack_sip 29181 0 ? |
| 02:45.38 | volga629 | I will test in morning and I will post results here if I will make some progress on it |
| 02:46.00 | volga629 | thank you Everyone |
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| 02:51.50 | ChannelZ | wonders does nf_conntrack_sip still work if I have to portforward 5060? IE Asterisk runs on a separate machine from the actual firewall running iptables |
| 02:57.29 | SeRi | test |
| 02:59.12 | ChannelZ | yeah I'm trying |
| 02:59.25 | ChannelZ | but I just fail2ban'd myself |
| 02:59.52 | volga629 | If asterisk in have directly connected network to firewall you need only route on asterisk back without port forward |
| 03:00.07 | volga629 | :-) fail2ban rulez |
| 03:00.50 | ChannelZ | I know, but the point was to try and avoid having to port-forward a bunch of RTP ports and see if nf_conntrack_sip could take care of it |
| 03:01.08 | volga629 | should yes |
| 03:02.58 | volga629 | http://fpaste.org/0NF2/ modinfo |
| 03:04.04 | volga629 | I think only 5060 and so on, but not rtp. Might I am wrong, need check 100% |
| 03:07.04 | ChannelZ | well something that can only connection track the SIP traffic is kind of useless |
| 03:08.14 | ChannelZ | but actually in my testing I have a different problem.. whereby my remote softphone tells Asterisk its LAN IP.. which matches localnet and Asterisk sends RTP to the LAN address instead of the external one, even though nat should be on. |
| 03:09.29 | *** join/#asterisk hiyo (6c09aeb6@gateway/web/freenode/ip.108.9.174.182) |
| 03:09.33 | hiyo | hi |
| 03:09.48 | ChannelZ | hiyo |
| 03:10.26 | *** join/#asterisk d00gster (~doughant@bba773204.alshamil.net.ae) |
| 03:10.27 | hiyo | channelz: so what exactly can I do with Asterisk? (in layman's terms) |
| 03:10.50 | ChannelZ | Do things with phone calls |
| 03:11.14 | hiyo | interesting |
| 03:11.37 | hiyo | can I manage skype calls? |
| 03:12.15 | hiyo | channelz: and what is the difference from it and using a regular SIP client? |
| 03:13.31 | ChannelZ | Yes and no on Skype; You'd have to get Skype For SIP or whatever they call it, which is an extra cost through them. |
| 03:13.45 | ChannelZ | And the difference is that Skype is closed/propriatary, SIP is not |
| 03:14.35 | hiyo | channelz: I meant what is the difference from using Asterisk and a regular SIP client |
| 03:15.40 | ChannelZ | well Asterisk is a PBX not just a client |
| 03:16.51 | hiyo | oh |
| 03:16.52 | ChannelZ | IE it's a whole 'server' not a client-end piece of software to let you make SIP calls. |
| 03:17.57 | hiyo | oh okay then |
| 03:19.37 | hiyo | could it be possible though to set it up so that I could have SIP calls from a landline attached to the computer? |
| 03:20.00 | ChannelZ | yes |
| 03:20.26 | hiyo | could taht be possible with just a SIP client? |
| 03:20.33 | ChannelZ | That's one thing Asterisk does over a simple SIP proxy, is it can act as a bridge between technologies |
| 03:21.13 | ChannelZ | Not to drive an analog phone or line, no.. you need hardware (an ATA) for that. |
| 03:22.01 | hiyo | oh so I couldn't just have it hooked up to the dialup modem in the back to work then could I? |
| 03:23.28 | ChannelZ | No |
| 03:25.08 | hiyo | oh okay then |
| 03:26.21 | hiyo | channelz: well thanks for clearing that up for me and I hope you have a good day/afternoon/night |
| 03:28.46 | ChannelZ | sure you too |
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| 04:35.34 | volga629 | is nobody know when asterisk 11 LTS should be released ? |
| 04:35.56 | volga629 | tired any body |
| 04:37.52 | volga629 | never mind found https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
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| 05:17.14 | sysnod1359 | is there a way to forcefully shut down asterisk? i have tried to kill it but has not been effective |
| 05:17.58 | ChannelZ | well if you can't kill -9 it then something is horribly awry |
| 05:18.08 | ChannelZ | Make sure you're not running safe_asterisk and it's not just starting a new one |
| 05:18.20 | sysnod1359 | oh ok |
| 05:18.34 | sysnod1359 | let me try again. did try to kill -9 asterisk |
| 05:21.13 | sysnod1359 | well yeah. asterisk is off. but i still cannot access into the etc/asterisk/ folder |
| 05:21.58 | sysnod1359 | i want to back-up the sip.conf file and extensions.conf ...and then will uninstall/reinstall asterisk |
| 05:25.13 | ChannelZ | I don't understand the question |
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| 05:26.30 | sysnod1359 | i cannot open the directory : etc/asterisk/ -----> it just wont open on console or GUI |
| 05:26.48 | ChannelZ | Do you not have permissions? |
| 05:26.56 | sysnod1359 | i want to back up the sip.conf and extensions.conf file . i am root |
| 05:27.18 | ChannelZ | define "won't open" |
| 05:28.25 | sysnod1359 | when i type dir in the etc/asterisk/ folder |
| 05:28.29 | sysnod1359 | nothing comes up. |
| 05:28.31 | sysnod1359 | and it just hangs |
| 05:28.54 | ChannelZ | well it's /etc/asterisk unless you're running some prebuilt package that moves it elsewhere |
| 05:29.25 | ChannelZ | but I don't understand what you're typing where. Are you using gnome or KDE or something? |
| 05:29.37 | sysnod1359 | yes. i am also running gnome |
| 05:32.06 | ChannelZ | well with asterisk running if you can access the console you can do 'core show settings' to see where the config directory is, if it's someplace else. |
| 05:32.32 | ChannelZ | otherwise if you're root I have no idea why you can't see inside it, so that's my only guess is that it's not there. |
| 05:32.37 | sysnod1359 | cannot access asterisk console |
| 05:33.10 | sysnod1359 | asterisk console wont load. |
| 05:33.25 | ChannelZ | well from a shell do 'locate sip.conf' |
| 05:33.35 | ChannelZ | or otherwise search for it. |
| 05:34.46 | sysnod1359 | i am trying to search for the sip.conf file |
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| 05:38.24 | sysnod1359 | i think the filesystem is gone |
| 05:38.42 | ChannelZ | it certainly sounds like you have "issues" |
| 05:39.41 | sysnod1359 | yes. i cannot get into the asterisk directory.. |
| 05:41.30 | sysnod1359 | i'll try to copy the etc/asterisk folder |
| 05:41.36 | carrar | Did you click go first? |
| 05:41.58 | sysnod1359 | click 'go first" ? |
| 05:42.26 | carrar | That was my attempt at being sarcastic |
| 05:42.51 | carrar | You change directory |
| 05:43.00 | carrar | not open |
| 05:43.04 | ChannelZ | this isn't FreePBX is it? |
| 05:43.08 | carrar | opens sounds like a GUI term |
| 05:43.11 | sysnod1359 | no. this is just asterisk |
| 05:43.26 | carrar | You are in a shell right? |
| 05:43.32 | sysnod1359 | fine i'll close the vncserver. and will run it via termina |
| 05:43.39 | sysnod1359 | shell |
| 05:43.51 | ChannelZ | yeah... do it like a man, man! |
| 05:44.06 | sysnod1359 | yeah. lets be man and do shell |
| 05:44.08 | sysnod1359 | shelling |
| 05:44.35 | sysnod1359 | so i logged in. if i type cd /etc/asterisk/ ...my system will hang. |
| 05:44.47 | ChannelZ | sounds like time to fsck it |
| 05:44.49 | sysnod1359 | if i type in cd /etc/ its fine. |
| 05:45.25 | sysnod1359 | i can get into the etc folder. but not into the asterisk folder ....if i go into the asterisk folder it just hangs...it just become dead |
| 05:45.41 | carrar | show me |
| 05:45.43 | carrar | ls -ald /etc/asterisk/ |
| 05:46.04 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 05:46.26 | sysnod1359 | drwxr-xr-x 3 asterisk asterisk 4096 Sep 20 18:05 /etc/asterisk/ |
| 05:46.50 | jpsharp | sounds like you've got a jacked up filesystem. |
| 05:46.51 | carrar | and the output of 'whoami' |
| 05:47.26 | ChannelZ | fsck her, fsck her hard |
| 05:47.52 | sysnod1359 | thats the best thing to do? |
| 05:48.00 | sysnod1359 | so i'll have to unmount her and then fsck her |
| 05:48.02 | carrar | what does dmesg say |
| 05:48.06 | ChannelZ | as opposed to what, nothing? |
| 05:48.17 | carrar | check your messages log file for system errors |
| 05:48.19 | carrar | get on it |
| 05:48.26 | ChannelZ | is your sip.conf worth saving? |
| 05:48.35 | carrar | Call your admin |
| 05:49.10 | sysnod1359 | dmesg has a very long message |
| 05:49.13 | ChannelZ | ring ring ring ring ring ring ring ring bananaphone |
| 05:49.24 | *** part/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 05:49.33 | carrar | Look for "bad" things |
| 05:49.35 | sysnod1359 | sip.conf doesnt have much...i am just running it as experimental...but i guess fsck |
| 05:50.57 | sysnod1359 | Thunar[3915]: segfault at b1772820 ip b1772820 sp bf80b7fc error 14 in gconv-modules.cache[b1779000+7000] |
| 05:51.43 | jpsharp | That's a bad thing. |
| 05:52.05 | carrar | thats it? |
| 05:52.12 | sysnod1359 | there is more ... |
| 05:52.35 | sysnod1359 | xfce4-settings-[3531]: segfault at 1 ip b80661f5 sp bfd02d10 error 4 in xfce4-settings-helper[b805d000+10000] |
| 05:53.21 | jpsharp | YOu've got hardware issues. |
| 05:53.46 | carrar | do your testing on a vm without X |
| 05:54.25 | carrar | but yeah, run fsck |
| 05:54.46 | carrar | You are root right? |
| 05:54.54 | sysnod1359 | yeah. i guess i'll run a fsck. the fie system seems bad. yeah .root |
| 05:55.06 | sysnod1359 | i'll have to unmount and run fsck |
| 05:55.22 | carrar | tail your messages file after you have verified rsyslog is logging systme errors to the messages file |
| 05:55.25 | carrar | then cd into it |
| 05:55.40 | carrar | see what happens lucy |
| 05:55.54 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.38) |
| 05:56.35 | carrar | anything on console when you cd into /etc/asterisk? |
| 05:57.26 | sysnod1359 | <PROTECTED> |
| 05:58.47 | sysnod1359 | not sure how to execute it |
| 06:00.41 | carrar | might start a fresh clean install |
| 06:00.50 | carrar | since this is a test box |
| 06:01.16 | carrar | don't install X/gui stuff |
| 06:01.23 | sysnod1359 | yes... i just checked the etc/default/rcS |
| 06:01.37 | sysnod1359 | fsckfix was off. just turned it on |
| 06:01.45 | carrar | centos seems to work well for others and is simple |
| 06:02.30 | sysnod1359 | yes. i did GUI/vncserver just as an experimental thing |
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| 10:34.58 | sysnod1359 | i want to make asterisk playback a mp3 file. how do i specify which folder asterisk should look in? |
| 10:41.47 | *** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com) |
| 10:43.38 | specing | *directory! |
| 10:46.00 | sysnod1359 | directory? |
| 10:46.47 | sysnod1359 | @specing: yes. how to specify the directory. |
| 10:46.58 | WIMPy | *ex |
| 10:47.15 | sysnod1359 | i am not getting it.... |
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| 11:34.26 | Shishir | hello |
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| 13:52.17 | shawny_ | Hello Everyone! Hoping to get a quick answer here........ I just installed asterisk-10.9.0-rc1 and when I go into the CLI I can't use any show commands, I can't do any SIP commands, I cannot load any modules (there doesn't seem to be a /modules in /etc/asterisk) is my install FUBAR, or is it just my lack of knowledge? |
| 13:52.37 | *** join/#asterisk evilman_home (kvirc@2.92.147.77) |
| 13:58.23 | WIMPy | They are usually in /usr/lib/asterisk/modules. |
| 14:00.52 | gavimobile | if I want my pbx to connect to calls, would I need something other than Dial() to both parties? |
| 14:01.34 | gavimobile | I am trying to call an extention => pbx answers => pbx asks what telephone number to dial => pbx dials the number => pbx also calls my cell phone |
| 14:03.45 | shawny_ | so, if I am looking in /usr/lib/asterisk/modules and I still cannot see chan_sip.so or anything regarding sip....did my install pooch? |
| 14:04.00 | shawny_ | or is it something I forgot to do? |
| 14:04.39 | WIMPy | How did you install? |
| 14:06.05 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 14:06.14 | shawny_ | i did the ./configure - make - make install - make configure - make samples when I first did the ./configure it told me I needed to install SQLite3, so I did that and started over. |
| 14:06.57 | WIMPy | Between ./configue and make, you should insert a make menuselect. |
| 14:07.14 | shawny_ | am I right in assuming that I should be able to do a 'sip show peers' right from the asterisk cli? |
| 14:07.22 | WIMPy | yes |
| 14:07.33 | WIMPy | If you have SIP support. |
| 14:09.10 | shawny_ | does the sip support come from the make menuselect part? I thought sip support would have just been included with an asterisk install ( I know I shouldn't assume) |
| 14:09.52 | WIMPy | I'm not sure what gets built if you don't configure anything. |
| 14:13.48 | shawny_ | hmmm, looks like someone will be doing a re-install. |
| 14:16.47 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
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| 14:27.17 | gavimobile | I cannot get asterisk to continue with the dialplan after dialing a cellphone. it only continues once the callphone calls hangs up. exten => start,1,Dial(Local/201@motekPC) |
| 14:28.21 | gavimobile | I would like asterisk to go to a Goto context after that first call has been initiated |
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| 14:45.56 | gavimobile | folks? |
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| 15:02.09 | volga629 | hello Everyone |
| 15:02.59 | volga629 | WIMPy that what I was be able fish out http://fpaste.org/u7eI/ I am testing right now rtp to short |
| 15:05.17 | *** join/#asterisk serafie (~erin@75.76.38.159) |
| 15:05.51 | WIMPy | Looks ok from the SIP side, I think. |
| 15:06.46 | volga629 | I wonder 2 think if I need disable on asterisk config for peer nat settings and id some trouble with rtp payload id |
| 15:07.47 | WIMPy | Usually the best way to handle NAT is to enable it in Asterisk and nowhere else. |
| 15:08.39 | *** join/#asterisk evilman_home (kvirc@78-106-157-83.broadband.corbina.ru) |
| 15:08.43 | volga629 | so might rtp id issue ? |
| 15:09.09 | volga629 | I can try change on video phone, currently set to 117 |
| 15:09.18 | WIMPy | I'm not that much in to SPI and even less in to SDP. |
| 15:09.25 | WIMPy | SIP |
| 15:16.59 | volga629 | [2012-09-23 11:14:41] VERBOSE[30688] chan_sip.c: --- (11 headers 0 lines) --- |
| 15:16.59 | volga629 | [2012-09-23 11:14:41] WARNING[13570] res_rtp_asterisk.c: RTP Read too short |
| 15:17.06 | volga629 | o lines |
| 15:17.10 | volga629 | 0 line |
| 15:17.11 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 15:18.05 | volga629 | WIMPy I am going [TK]D-Fender might he is can give more info |
| 15:18.30 | WIMPy | Probably |
| 15:19.31 | volga629 | <PROTECTED> |
| 15:20.28 | Shishir | hello |
| 15:20.29 | volga629 | I corrected codec on remote side and at least don't see trouble with codec g729 in place |
| 15:21.16 | [TK]D-Fender | I don't see actual dialplan execution r any outboud call attempt to follow through with that.... |
| 15:21.57 | Shishir | how can I use asterisk (with public address) to forward calls to remote gateways behind nat |
| 15:22.15 | WIMPy | Just do it. |
| 15:22.27 | volga629 | let do test call |
| 15:22.28 | WIMPy | Or ask a more precise question. |
| 15:22.45 | volga629 | in sip debug ? |
| 15:23.13 | [TK]D-Fender | There is no call processing in there. |
| 15:23.28 | [TK]D-Fender | I see no second call so that 2 enpoints have a reason to use video. |
| 15:30.11 | volga629 | this dial plan http://fpaste.org/WBo5/ |
| 15:31.06 | [TK]D-Fender | http://www.politicususa.com/elizabeth-warren-dominates-scott-brown-debate.html |
| 15:31.16 | [TK]D-Fender | oops |
| 15:31.36 | [TK]D-Fender | Capabilities: us - (g729|h264), peer - audio=(ulaw|g729)/video=(nothing)/text=(nothing), combined - (g729) |
| 15:31.43 | [TK]D-Fender | First call in, NO video |
| 15:31.58 | [TK]D-Fender | Peer doesn't provide video |
| 15:32.00 | [TK]D-Fender | ^ |
| 15:33.31 | [TK]D-Fender | Liine 530 : other side already OK's, now the answer back to the calling phone keeps saying "just G.729" |
| 15:35.05 | volga629 | Or on peer setting need allow ulaw ? |
| 15:36.59 | [TK]D-Fender | 1st call came in not offering video. First odd thing |
| 15:37.37 | [TK]D-Fender | NExt at least the codec's seem OK for negotiation and we see no debug that that warning is critical and I don't see a proper description of the PROBLEM it is causing. |
| 15:37.40 | *** join/#asterisk Z_God (~julius@83.161.151.96) |
| 15:38.09 | Z_God | it seems asterisk runs about 7 threads here by default, is there a simple way to reduce this? |
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| 15:40.01 | volga629 | I see how possible catch this moment |
| 15:40.32 | volga629 | I went trough peers settings and make sure set to g729&h264 |
| 15:40.55 | [TK]D-Fender | What peer settings? The damn PHONE didn't start with it. |
| 15:41.06 | [TK]D-Fender | BRIA should be adjusted. |
| 15:41.13 | [TK]D-Fender | And you have not stated the PROBLEM |
| 15:41.14 | volga629 | on phones and on asterisk |
| 15:42.56 | volga629 | problem video phone on remote end through vpn tunnel. Voice working no issue video not working I see errors about video port 0 and rtp too short |
| 15:43.54 | *** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com) |
| 15:48.11 | volga629 | I adjusted BRIA for asterisk to g729&h264 |
| 15:49.38 | [TK]D-Fender | Do 2 local Bria's work? |
| 15:50.55 | volga629 | http://fpaste.org/WzLD/ this call after change |
| 15:51.21 | volga629 | second phone is hard phone and another end is Bria |
| 15:51.35 | [TK]D-Fender | What "change"? |
| 15:51.42 | [TK]D-Fender | You are not being clear or thorough\ |
| 15:52.03 | volga629 | codec on bria. I set to g729&h264 |
| 15:56.20 | volga629 | I see those message only between peers when dial out no problems |
| 16:02.34 | volga629 | <PROTECTED> |
| 16:18.07 | volga629 | what else can be done to try resolve it ? |
| 16:20.16 | *** join/#asterisk d00gster (~doughant@92.98.59.187) |
| 16:22.38 | volga629 | this look ok right now Capabilities: us - (g729|h264), peer - audio=(alaw|g729)/video=(h264)/text=(nothing), combined - (g729|h264) ? |
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| 17:06.59 | *** join/#asterisk pcangel (~pcangel@S0106b8a38659e04a.vc.shawcable.net) |
| 17:07.37 | pcangel | Hi guys.. I have one server on 10.1.2 and another on 10.7.0, with seemingly identical sip.conf and features.conf |
| 17:07.57 | pcangel | The 10.1.2 is able to Atxfer to an exten that starts with a # key, the 10.7.0 is not |
| 17:08.13 | pcangel | The 10.7.0 plays back invalid extension, even though the extension is valid and can be dialed manually from the same context |
| 17:08.29 | pcangel | I'm using Atxfer in the manager interface |
| 17:08.48 | pcangel | Most items in features.conf are commented out to use defaults |
| 17:20.40 | [TK]D-Fender | pastebin the call output, and your configs |
| 17:23.52 | *** join/#asterisk iulhk (iulhk@119.154.97.22) |
| 17:32.58 | volga629 | [TK]D-Fender is any other suggestion about rtp video issue ? |
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| 17:35.05 | volga629 | I see pay load id set a=rtpmap:99 H264/90000 is phones need to be adjusted ? |
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| 18:38.07 | luizmachado | Hi all |
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| 19:30.47 | pcangel | [TK]D-Fender: The reason for the problem turned out to simply be that ast_app_dbget breaks from its loop upon receiving '#'... I don't know why I thought it worked on the 10.1.2 system, I must have been testing transfering it to the wrong exten. I thought my problem was going to be another feature trying to take over the channel (I didn't understand the call stack until just now when I |
| 19:30.47 | pcangel | was looking) |
| 19:31.08 | pcangel | So I just have to rewrite all of my auto created extensions to use a different prefix =\ |
| 19:31.41 | pcangel | I was hoping there'd be some option I could flick somewhere to stop it from stopping the extension read when it gets the # |
| 19:34.02 | WIMPy | You're saying you can;t feature transfer to an exteinsion containing a #? |
| 19:34.32 | pcangel | I can't atxfer to a valid extension containing a #, through the asterisk manager interface |
| 19:35.06 | WIMPy | That sounds wrong. |
| 19:35.08 | pcangel | My CRM auto-creates extensions for queues and has them start with ##98, so it's just a minor hassle. I thought it was a config problem and was searching for how to fix it |
| 19:35.10 | pcangel | I know |
| 19:35.13 | pcangel | but I read the code |
| 19:35.34 | pcangel | I'm going to submit it as a bug, but it sounds strange that nobody's considered that... |
| 19:35.54 | WIMPy | Indeed. |
| 19:36.55 | WIMPy | I just wonder what atxfer is good for. |
| 19:37.16 | WIMPy | I always used redirect. |
| 19:37.32 | pcangel | features.c function builtin_channel callis ast_app_dtget, which in app.c shows a loop that will always break on '#', so as soon as I hit #, it stops reading |
| 19:37.57 | pcangel | the person transfering is used to hearing the person on the other line before dropping the call |
| 19:38.35 | WIMPy | Ah, yes. |
| 19:39.28 | WIMPy | But 'as soon as I hit #, it stops reading' doesn't make ant sense to me. |
| 19:39.54 | pcangel | Manager function Atxfer simulates DTMF tones to execute an attended transfer |
| 19:40.04 | WIMPy | Ugh |
| 19:40.04 | pcangel | so by "I" I meant the manager function Atxfer |
| 19:40.09 | WIMPy | Ugly. |
| 19:40.11 | pcangel | it reads in to builtin_atxfer |
| 19:40.42 | WIMPy | But bad enough that the feature transfer does it. |
| 19:42.32 | pcangel | I guess for such an insignificant feature compared to the size of the rest of the code, it's a 'alright' design decision to make to implement things that way, rather than duplicating the builtin_atxfer code in to the manager functions. I'm sure there's a better way to segment it. Unfortunately I don't feel like patching and submitting a patch purely because this is my first time looking at |
| 19:42.33 | pcangel | the asterisk code, and I have no idea what implication any change would make |
| 19:42.36 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 19:43.28 | WIMPy | There probably aren't many who do. |
| 19:43.52 | WIMPy | And I have a bad feeling that fixing that issue may brak some badly designed dialplans. |
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| 19:44.53 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 19:50.50 | WIMPy | Might be better to do the transfer yourself. |
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| 20:21.10 | pcangel | Submitted. Roar I need to go eat some babies, thanks for the brainstorming |
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| 20:40.32 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 21:05.31 | kresp0 | why asterisk don't load my extensions.conf? I'm using default asterisk.conf and the asterisk is the owner of all /etc/asterisk/* files |
| 21:06.24 | ChannelZ | maybe you have a type-o preventing it from being parsed correctly. |
| 21:08.12 | kresp0 | thank you ChannelZ |
| 21:08.37 | kresp0 | now I see the problem when I do module load pbx_config |
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| 21:34.51 | unicron | what's the best open source sip client with zrtp? |
| 21:38.52 | leifmadsen | I'm not sure many really support ZRTP |
| 21:39.00 | leifmadsen | I was under the impression ZRTP kind of died |
| 21:39.32 | unicron | well that sucks |
| 21:39.39 | WIMPy | Just because Asterisk doesn't do it? |
| 21:39.59 | unicron | i want end to end encryption |
| 21:40.30 | leifmadsen | WIMPy: no |
| 21:40.45 | leifmadsen | unicron: SIP TLS + SRTP |
| 21:41.13 | leifmadsen | WIMPy: because nothing ended up implementing it |
| 21:41.46 | kresp0 | unicron: SFLphone works ok and support SIP TLS + SRTP / ZRTP (but I didnt tried the encryption) |
| 21:45.23 | *** join/#asterisk Luke-Jr (~luke-jr@2001:470:5:265:222:4dff:fe50:4c49) |
| 21:45.48 | Luke-Jr | Any way to set Asterisk up to ban IPs when there's more than 2 calls in a second, and log the IP? <.< |
| 21:46.30 | leifmadsen | Luke-Jr: you use fail2ban for that |
| 21:46.39 | leifmadsen | that is outside the scope of Asterisk |
| 21:46.57 | Luke-Jr | well, I don't see any way to find the IPs |
| 21:47.29 | Luke-Jr | short of tcpdump and seeing which one is flooding |
| 21:47.36 | leifmadsen | fail2ban |
| 21:48.17 | specing | Luke-Jr: iptables connlimit |
| 21:48.47 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-2-227-192.ph.ph.cox.net) |
| 21:49.05 | Luke-Jr | any way to get Asterisk to just log the IP of each call somewhere? |
| 21:50.31 | leifmadsen | could look at CHANNEL() function to see if ti has it, then use cdr_adaptive_odbc and add an ipaddress field to the CDR |
| 21:50.36 | leifmadsen | or Log() etc |
| 21:55.59 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 21:57.43 | unicron | could write a simple wrapper for it |
| 21:59.46 | unicron | jitsi seems cool |
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| 22:05.39 | riwarren | Is there anyone here that has worked with NACT BAT-1000 Software before? |
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| 23:00.23 | [TK]D-Fender | Luke-Jrwell, I don't see any way to find the IPs <- If * is processing the call then you're in dialplan and can know the IP and can do this yourself. |
| 23:17.48 | *** join/#asterisk dandate2 (~dan@120.28.230.182) |
| 23:18.13 | dandate2 | when trying to register an iax softphone cli says iax2_chan ... failed MD5 authentication... |
| 23:18.14 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 23:18.15 | dandate2 | what the hell is that |
| 23:26.18 | *** join/#asterisk darkskiez (~mhb@fsf/member/darkskiez) |
| 23:30.31 | [TK]D-Fender | What does it sound like? |
| 23:33.18 | *** join/#asterisk AlfE_ (~quassel@83-215-36-12.bruck.dyn.salzburg-online.at) |
| 23:34.31 | dandate2 | failed MD5 authentication, but no clue why that would happen on a fresh zoiper download and extension creation |
| 23:34.50 | dandate2 | couldnt find anything about it on google |
| 23:38.47 | volga629 | [TK]D-Fender After all looking into issue, I almost positive that trouble in vpn tunnel which NAT Traversal. I corrected all codec and I don't see Capabilities: us - (g729|h264), peer - audio=(ulaw|g729)/video=(nothing)/text=(nothing), combined - (g729), right it negotiate correctly, but issue still persist. I will try rebuild vpn tunnel without NAT T and see if it make any different. Thank you for you Everyone for they help and time |
| 23:39.18 | volga629 | now |
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