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| 00:13.59 | file | tick tock |
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| 01:09.22 | phix | file: sup? |
| 01:10.03 | phix | file: Are you opened or closed? |
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| 03:19.42 | ganbold_ | how to enable h235 authentication with cisco gk using ooh323? |
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| 07:56.42 | imihaylov | <PROTECTED> |
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| 08:22.39 | ChannelZ | Well you need to just write some dialplan to do it. Play a prompt, use Read() to get the number, then store it somewhere like the AstDB |
| 08:23.03 | ChannelZ | Then you'll need to check the blacklist on your incoming calls and respond accordingly |
| 08:30.04 | ChannelZ | Very simplistic starter: http://pastebin.com/0pWRDtYN |
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| 08:31.29 | rox | Hello, |
| 08:32.22 | ChannelZ | herro |
| 08:32.35 | rox | I am using 1.8 LTS version, and when monitoring SIP-SIP bridged calls, monitoring sometimes breaks off during call, so I am left with a partial call recording |
| 08:32.57 | rox | has anybody heard of such an issue? |
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| 08:34.21 | rox | oh, and it's not when transferring calls |
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| 08:38.18 | imihaylov | thank you! I'm already using blakclist function but I had to enter them into cli. |
| 08:42.19 | ChannelZ | no prob |
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| 08:58.26 | imihaylov | what is the function to remove number from blacklist |
| 08:58.41 | imihaylov | I want to use it in dialing *31 for example |
| 08:59.46 | imihaylov | exten => *31,n,Set(DB(blacklist/${NumToBlacklist})=1) |
| 08:59.57 | imihaylov | set to be equal to 0 ? |
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| 09:08.53 | ChannelZ | there is the DBdel application |
| 09:09.32 | ChannelZ | exten => *31,n,DBDel(blacklist/${NumToBlacklist}) for instance |
| 09:10.07 | ChannelZ | Actually, do you have the DB_DELETE function in 1.8? |
| 09:10.43 | ChannelZ | core show function DB_DELETE |
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| 09:31.58 | rox | imihaylov: you can DB_DELETE() the number |
| 09:32.07 | imihaylov | yes |
| 09:32.21 | imihaylov | I have DB_DELETE |
| 09:34.20 | imihaylov | thank you |
| 09:34.48 | rox | Regarding my Monitor cutting the calls short - I have found 3 related bugs on issues.asterisk.org but none on Asterisk 1.8. Has anyone ever heard of such an issue on Asterisk 1.8? |
| 09:34.55 | imihaylov | I have strange behavour |
| 09:35.03 | imihaylov | my asterisk extensions are not ringing |
| 09:35.06 | imihaylov | [Feb 17 11:33:57] WARNING[5411]: chan_sip.c:4210 __sip_autodestruct: Autodestruct on dialog '11-36884747-54E30AB50006C2BB-0B440700' with owner SIP/46.19.210.14-00000013 in place (Method: BYE). Rescheduling destruction for 10000 ms |
| 09:35.20 | imihaylov | and I am getting all bunch of that |
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| 09:45.14 | imihaylov | It was vtiger asterisk connector fault |
| 09:49.45 | imihaylov | [Feb 17 11:49:32] WARNING[6025]: app_db.c:129 del_exec: The DBdel application has been deprecated in favor of the DB_DELETE dialplan function! |
| 09:50.09 | imihaylov | [Feb 17 11:48:26] WARNING[6010]: pbx.c:4458 pbx_extension_helper: No application 'DB_DELETE' for extension (cc, *31, 2) |
| 09:50.15 | imihaylov | what now |
| 09:51.47 | wdoekes | imihaylov: you're calling it as an app, not as a function |
| 09:52.11 | wdoekes | SomeApp() vs. Set(SOMEFUNCTION([parameters])=value) |
| 09:52.18 | imihaylov | exten => *31,n,DB_DELETE(blacklist/${NumToBlacklist}) |
| 09:53.31 | imihaylov | ok |
| 09:53.35 | imihaylov | [Feb 17 11:53:17] ERROR[6133]: pbx.c:3974 ast_func_write: Function DB_DELETE cannot be written to |
| 09:53.38 | imihaylov | when i use |
| 09:54.01 | imihaylov | Set(DB_DELETE(blacklist/{NumToBlacklist})=1) |
| 09:54.36 | imihaylov | syntaxis is DB_DELETE(family/key) |
| 09:55.53 | wdoekes | according to the [Description], it *returns* a value |
| 09:56.29 | wdoekes | i.e. you'll call it like this: NoOp(${DB_DELETE(blacklist/${NumToBlacklist})}) |
| 09:56.39 | wdoekes | or this: Set(return_value=${DB_DELETE(blacklist/${NumToBlacklist})}) |
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| 09:58.33 | imihaylov | thank you very much! |
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| 10:39.16 | wizbit | i have these errors in my asterisk messages log: http://dpaste.com/25126ZB.txt |
| 10:39.22 | wizbit | are they anything to worry about? |
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| 10:51.18 | kleszcz | check /etc/asterisk |
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| 10:51.32 | kleszcz | and then check /etc/asterisk/asterisk.conf |
| 10:52.00 | Tuju | i'm having trouble with this http://lists.digium.com/pipermail/asterisk-users/2014-March/282561.html fromdomain doesn't work with outbound INVITE |
| 10:52.04 | kleszcz | astetcdir => /etc/asterisk |
| 10:52.25 | Tuju | all my outbound invites have ip-address instead of FQDN. |
| 10:52.58 | Tuju | and hence my trunk lines and contexts get mixed at receiving end, again :-/ |
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| 11:58.14 | yun1989 | hello all |
| 11:58.34 | yun1989 | i have one problem some times asterisk put the CPU to 95% usage |
| 11:58.43 | yun1989 | any ideias for this problem ? |
| 12:01.16 | Tuju | i manged to get partial fix to my problem by removing 'invite' from insecure setting, hence putting all invites under digest auth --> realm setting is used. |
| 12:01.39 | Tuju | but invite messages still appear to appear with ip-address, something that i find filthy. |
| 12:02.19 | Tuju | calls work (if my bookkeeper would answer) - but it's not perfect yet. |
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| 12:35.54 | stefan27 | i disabled coredumps some time ago and now i dont remember how to get them back... If i do 'asterisk -c -vvvg' from /tmp and then "core stop gracefully" isn't asterisk supposed to produce a coredump file in tmp? |
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| 12:44.48 | imihaylov | is there any way to make a script that calls some number pointing to asterisk |
| 12:44.48 | imihaylov | and asterisk automatically pickups and hang after 10 seconds, |
| 12:44.48 | imihaylov | and returns success as result or failure if anything goes wrong. |
| 12:44.48 | imihaylov | The script would be at other asterisk server. |
| 12:46.16 | file | everything required to do it exists |
| 12:46.20 | file | in pieces. |
| 12:56.13 | wizbit | does my iax.conf look secure: http://dpaste.com/1B9831K.txt ? |
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| 13:03.56 | stefan27 | im completely lost... why won't asterisk produce core dump files, I tried starting it as root with asterisk -c -vvvg |
| 13:04.12 | stefan27 | ive produced tons of core dumps in the past |
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| 13:17.27 | rox | is there a real difference between monitor with m flag and mixmonitor? |
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| 13:17.39 | rox | stefan27: check your ulimit |
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| 13:25.25 | Rad | Hi , i've created a sip trunk for one of our customer, with a call limit feature , i'm not using the sip.Conf "call-limit" parameter , instead i am using the Group() commands . When my customer reaches the limit , i want to send "Call Limit Reached", how can i do that ? |
| 13:29.06 | stefan27 | ulimit -c unlimited didn't help, was that what you meant? |
| 13:30.25 | Rad | stefan27 sorry , are you trying to answer my question ? |
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| 13:32.00 | stefan27 | Sorry, i was asking rox, he was helping me :D |
| 13:32.19 | stefan27 | I can't answer your question :/ |
| 13:32.40 | Rad | its ok. |
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| 13:40.39 | Rad | hi ? |
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| 13:47.02 | Apteryx | Hello! I was wondering if there was a way to quickly switch between chan_sip and pjsip using the modules.conf file? |
| 13:47.50 | WIMPy | Apteryx: Exactely that. |
| 13:47.58 | Apteryx | Up to now I was maintaining a custom modules.conf for trying both |
| 13:48.40 | Apteryx | WIMPy: but in order to load pjsip, there does not appear to be a single line, I need to load tens of res_pj_*. I hope I'm wrong. |
| 13:49.38 | WIMPy | No. That's correct. But you can write two versions of modules.conf that you replace. |
| 13:50.13 | Apteryx | WIMPy: ok! So if I'm using autoload, what happens? Won't pjsip clashes with chan_sip? |
| 13:50.29 | WIMPy | It will. |
| 13:50.46 | Apteryx | ok. So the default config is to load pjsip or chan_sip? |
| 13:51.00 | WIMPy | But you can use both if you configure them correctly to not interfere. |
| 13:51.29 | WIMPy | The default is to load wht exists. That would probably be both. |
| 13:52.01 | Apteryx | WIMPy: ok. Thanks for the explanation. |
| 13:53.05 | Apteryx | Is there a bare minimum example somewhere as to what is required to be minimally in modules.conf and why? |
| 13:54.19 | Apteryx | I got something working by trial and error, but a nice example would still be nice :) |
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| 13:55.22 | stefan27 | nevermind my problem im an idiot i mixed up (misread) the kill signals |
| 13:55.36 | stefan27 | kill -11 was for forcing core |
| 13:56.50 | Rad | any idea how to inform a specific trunk that it has reached the limit of outbounded calls ? |
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| 14:10.34 | WIMPy | Apteryx: No it depends on what YOU need. You couls also still use autoload an some noload's. |
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| 14:11.23 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:11.49 | rox | I am having problems, Monitor does not record entire calls, it stops recording randomly in the middle of call. If I replace Monitor with MixMonitor, can I reasonably expect the problem to go away? |
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| 15:46.56 | jamicque | Hi, while using asterisk with webscoket I get an error in console "No DTLS-SRTP support present on engine for RTP instance '0xa7c0eb4', was it compiled with support for it?" It happnes on my asterisk 11 compiled on lenny, on ubuntu 14.04 everything is ok, do you have any ideas what is wrong or which lib am I missing? |
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| 15:48.10 | file | OpenSSL has to have DTLS-SRTP support, and libsrtp has to be present |
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| 15:49.21 | jamicque | I tried to compile the openssl from source to be sure it has DTLS-SRTP support, libsrtp is present |
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| 15:49.45 | jamicque | in make menuselect I see that module res_srtp is present |
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| 16:01.01 | yun1989 | hello all |
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| 16:01.03 | mutilator | anyone use the snom870 w/ openvpn? |
| 16:01.14 | yun1989 | <PROTECTED> |
| 16:01.20 | yun1989 | any ideias for this problem ? |
| 16:01.34 | yun1989 | @[TK]D-Fender |
| 16:01.42 | yun1989 | hello how are you ? |
| 16:02.16 | [TK]D-Fender | Sorry, can't help you on this one... |
| 16:02.33 | yun1989 | ok |
| 16:03.06 | JerJer | meep meep |
| 16:03.07 | yun1989 | thanks |
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| 16:15.21 | stefan27 | When NEWCIDNAME="test1,test2", I execute Macro(set-callerid-name,"${NEWCIDNAME}") but in context set-callerid-name ARG1 is not "test1,test2" as expected but just test1 |
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| 16:15.58 | stefan27 | how can i escape "," in calls to Macro app |
| 16:16.25 | [TK]D-Fender | <PROTECTED> |
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| 16:17.36 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 16:20.51 | stefan27 | It seems like even with exten => _Y.,n,Set(NEWCIDNAME="test1\,test2") and exten => _Y.,n,Macro(set-callerid-name,"${NEWCIDNAME}") . ARG1 in set-callerid-name is still test1\ |
| 16:21.49 | stefan27 | I want ARG1 to refer to "test1,test2" if NEWCIDNAME has the value "test1,test2" |
| 16:24.07 | stefan27 | is there any other general way to escape , in app-calls |
| 16:24.46 | [TK]D-Fender | pass it the variable name instead of the value |
| 16:30.16 | *** join/#asterisk Defraz (~Defraz@209.141.122.71) |
| 16:31.00 | stefan27 | Yeah I guess that works, if ${ARG1} is a variable-name - ${${ARG1}} is the value? |
| 16:33.33 | *** join/#asterisk linocisco (~linocisco@111.84.193.64) |
| 16:33.49 | linocisco | hi all |
| 16:34.29 | [TK]D-Fender | <PROTECTED> |
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| 16:46.13 | linocisco | how can we setup aseterisk for sending Fax from computers and receiving fax from shared network drives or NAS drive? |
| 16:47.14 | linocisco | instead of receiving Fax via email attachement |
| 16:58.42 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 16:59.08 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 16:59.08 | [TK]D-Fender | Asterisk doesn't do "e-mail" |
| 16:59.47 | [TK]D-Fender | It's your dialplan, do whatever you want with it. |
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| 17:57.46 | yun1989 | hello all |
| 17:58.05 | yun1989 | it's possible to connect asterisk to this https://www.grandstream.com/index.php/products/ip-voice-telephony/consumer-analog-telephone-adaptors? |
| 17:58.40 | [TK]D-Fender | yes |
| 18:01.28 | yun1989 | @[TK]D-Fender i have one problem some times asterisk put the CPU to 95% usage |
| 18:01.48 | yun1989 | thanks for your help one more time |
| 18:01.53 | *** join/#asterisk Tom-M (~Tom-M@541CB5AA.cm-5-5c.dynamic.ziggo.nl) |
| 18:02.19 | yun1989 | do you have any ideia why ? |
| 18:03.00 | [TK]D-Fender | [11:02][TK]D-FenderSorry, can't help you on this one... |
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| 19:08.15 | h8 | hi everyone |
| 19:09.09 | cyford | helo |
| 19:09.12 | h8 | running asterisk 11 on a gentoo machine, I'm getting the following error: chan_sip.c:23028 handle_response_invite: Received response: "Forbidden", where's the catch, what is the usual issue here? I've tried going through NAT problems, opened the server up totally, same problem, the call goes out through a SIP Gateway, if I connect to the gateway through a softphone it works, through asterisk it doesn't |
| 19:11.04 | [TK]D-Fender | Because it's authing wrong |
| 19:11.37 | cyford | yep just what i was about to say lol |
| 19:13.08 | h8 | so it's auth issues, okay, I'll look in to that |
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| 19:56.14 | *** join/#asterisk talntid (~talntid@173-160-189-58-Washington.hfc.comcastbusiness.net) |
| 19:56.33 | talntid | i have timeout=10 in a queue, and the queue is setup as ringall |
| 19:56.49 | talntid | after 10 seconds i'd like it to continue down the dialplan... am i doinitwrong? |
| 19:59.21 | rrittgarn1 | timeout in a queue is how long to ring a queue member. unless you do it in the Queue() call |
| 20:10.14 | talntid | so since they are ringall, they all ring at once |
| 20:10.36 | *** join/#asterisk JerJer (~Adium@asterisk/original-h323-guy/JerJer) |
| 20:10.41 | talntid | so 10 seconds later, it should fall through, or does it do 10 * [number of queue members] ? |
| 20:15.00 | [TK]D-Fender | that is the inter-agent dial timeout, not the TTL of the call staying in the queue |
| 20:15.11 | [TK]D-Fender | "core show application queue:" <-- |
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| 20:21.52 | talntid | roger that |
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| 21:45.33 | jab416171 | ugh, I keep getting this error from google: |
| 21:45.35 | jab416171 | Too many created bindings per hour. |
| 21:48.01 | Penguin | Maybe you should pastebin your configuration. |
| 21:50.57 | ChannelZ-Wk | Making lots of calls? |
| 22:07.14 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 22:12.21 | gusto | jab416171, how do you connect asterisk to google? XMPP? |
| 22:12.48 | jab416171 | ChannelZ, not making any calls, it's "disconnected" in the UI |
| 22:12.53 | jab416171 | gusto, I guess so... whatever asterisk does by default |
| 22:12.55 | jab416171 | using Motif |
| 22:12.58 | jab416171 | Penguin, which file is that? |
| 22:13.22 | gusto | chan_motif.conf? |
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| 22:22.52 | jab416171 | it's working now |
| 22:22.53 | jab416171 | lol |
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| 23:34.00 | *** join/#asterisk jamicque (~jamicque@89-71-40-164.dynamic.chello.pl) |
| 23:34.35 | jamicque | which rtp engine has the DTLS config and what dependency is required to enable it? First time getting this message: ERROR[31285][C-00000001]: chan_sip.c:5701 dialog_initialize_dtls_srtp: No DTLS-SRTP support present on engine for RTP instance '0xa4181a4', was it compiled with support for it? |
| 23:35.02 | jamicque | OpenSSL 1.0.1i 6 Aug 2014 and SRTP installed |
| 23:35.13 | jamicque | but no luck :( |
| 23:49.45 | ChannelZ-Wk | libsrtp probably |
| 23:50.35 | *** join/#asterisk ndb (~ndb@191.191.120.246) |
| 23:50.40 | jamicque | libsrtp is in version 1.4.5 |
| 23:50.49 | ndb | I wanna build a new module with libcurl deps, how do I add a -lcurl? |
| 23:50.55 | ndb | on compilation flags* |
| 23:51.53 | jamicque | during the configuration of sources i get in config.log - conftest.c:291: undefined reference to `SSL_CTX_set_tlsext_use_srtp' |
| 23:52.19 | jamicque | I guess that's the point, any solutions? |
| 23:53.18 | ChannelZ-Wk | SSL_CTX_set_tlsext_use_srtp seems like an openssl thing |
| 23:53.31 | ChannelZ-Wk | You have the -dev of both installed? Not just the libs? |
| 23:54.42 | jamicque | jep, however it might be something wrong with the package |
| 23:54.47 | jamicque | becouse it's on lenny |
| 23:55.07 | jamicque | do you think thah compiling openssl from source will solve the issue? |
| 23:56.07 | ChannelZ-Wk | ? FWIW I have libssl-dev 1.0.1-4 and libsrtp0-dev 1.4.4 and res_srtp builds OK |
| 23:57.45 | jamicque | on lenny normally was package 0.9.8 |
| 23:57.56 | jamicque | the one 1.0.1 is backported it might be corrupted |
| 23:58.30 | ChannelZ-Wk | <- Ubuntu 12.04 (precise) |
| 23:58.36 | jamicque | i might tre installing openssl 1.0.1 from source, maybe this will help, or do you have any other ideas? |
| 23:58.47 | jamicque | 12.04 is newer than lenny :) |
| 23:59.32 | [TK]D-Fender | GLACIERS are newer than Lenny... |
| 23:59.37 | [TK]D-Fender | ... and faster |