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| 09:55.17 | cbeyerlein | Hi. Using Asterisk13, is there a way to dump ALL callerid information of a channel? There are dozens of possible datatypes in CALLERID(datatype,CID), and I would like to have an overview.. |
| 09:55.52 | cbeyerlein | currently trying to migrate from chan_ss7 to libss7-2.0.0 and need to verify same behaviour on ss7 level.. |
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| 10:10.44 | janicez | Meow |
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| 12:28.05 | tobiP | Hi folks |
| 12:28.36 | janicez | h |
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| 12:32.05 | tobiP | I've encountered an issue with confbridge and leve/join sound. It is only played to the joining participant (which is kind of useless) |
| 12:32.08 | pmantis | Hi guys. Is there a way to adjust the length of DTMF sent over FXO when dialing out? |
| 12:32.13 | tobiP | *leave |
| 12:38.05 | tobiP | anyone who can at least validate this behaviour? |
| 12:39.35 | robmal | announce_join_leave |
| 12:39.59 | tobiP | announce_join_leave=yes |
| 12:40.58 | tobiP | Only the caller who is joining gets the join sound |
| 12:41.23 | cbeyerlein | hmm.. what is the PJSIP equivalent of CHANNEL(from) ? |
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| 12:45.29 | cbeyerlein | CHANNEL(from) already extract the URI from the From-Header |
| 12:45.45 | cbeyerlein | and I have massive troubles doing the same in dialplan with PJSIP_HEADER(read,From) |
| 12:46.00 | cbeyerlein | as asterisk expressions are having troubles with the embedded whitespace and " quotes |
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| 12:54.45 | cbeyerlein | exten => _X.,n,NoOp($["\"1234\" <sip:5678@192.168.1.1>;tag=1618837779" =~ "sip:\+?([0-9]+)@"]) |
| 12:54.57 | cbeyerlein | what do I miss here? shouldnt this work? |
| 12:55.02 | cbeyerlein | and give 5678? |
| 12:55.38 | cbeyerlein | (still trying to parse From header with pjsip...) |
| 13:02.25 | cbeyerlein | exten => _X.,n,NoOp(From user: $["${REPLACE(PJSIP_HEADER(read,from),\"\\"\")}" =~ "sip:\+?([0-9]+)@"]) |
| 13:02.30 | cbeyerlein | now this is crazy... |
| 13:03.52 | cbeyerlein | vs. chan_sip: $["${CHANNEL(from)}" : "sip:\+([0-9]+)@"] |
| 13:04.01 | cbeyerlein | more straight forward ^^ |
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| 13:48.34 | leftleg | hello all |
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| 13:49.05 | [TK]D-Fender | goodbye |
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| 14:07.56 | last1 | anyone has a solution for circumventing ISP's that block port 5060 ? |
| 14:08.01 | last1 | I'm trying to register with a remote provider |
| 14:10.23 | [TK]D-Fender | use another port |
| 14:10.42 | [TK]D-Fender | because networking is netowrking. If they filter a port then there is no way for you to use that one on that interface |
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| 14:13.24 | last1 | I can't use another port, other people connect to port 5060 |
| 14:13.33 | last1 | I saw some people use something called siproxd |
| 14:13.41 | last1 | as an outbound proxy |
| 14:13.46 | last1 | wanted to know if anyone else played with that |
| 14:15.48 | [TK]D-Fender | Yes, others have |
| 14:16.07 | [TK]D-Fender | For you I'd recommend just using PJSIP as well as chan_sip |
| 14:16.17 | [TK]D-Fender | So you don't have to run some sort of external proxy on top |
| 14:17.06 | last1 | is that like a proxy ? |
| 14:17.15 | [TK]D-Fender | no |
| 14:17.18 | last1 | I guess I need something that listens on port 5061 and proxies to our asterisk on port 5060 |
| 14:17.19 | [TK]D-Fender | it's another channel driver |
| 14:17.29 | [TK]D-Fender | just use it |
| 14:17.52 | last1 | I install this on the same server as asterisk ? |
| 14:18.13 | file | if they block outgoing 5060... then no matter what you do it won't work, unless you proxy the traffic elsewhere on a different port |
| 14:18.51 | last1 | that's what I'm trying to do... run a proxy on the asterisk server on port 5061 which will forward to asterisk on port 5060 |
| 14:19.00 | last1 | I thought siproxd does that |
| 14:19.40 | file | it acts as a SIP proxy, although Asterisk can also listen and work on a different port |
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| 14:20.11 | last1 | yes, but only one at a time |
| 14:20.27 | last1 | I need both 5060 and 5061 |
| 14:20.28 | file | PJSIP can listen on multiple - but ultimately thay may or may not solve your problem |
| 14:20.47 | file | if your ISP is blocking OUTGOING port 5060 then it won't |
| 14:21.49 | last1 | I can configure my local client to connect to remote port 5061 |
| 14:21.58 | last1 | wouldn't that solve my issue if I have a proxy listening on port 5061 ? |
| 14:23.05 | file | I thought you said you were registering with a remote provider |
| 14:23.54 | last1 | yes, at our office |
| 14:24.23 | last1 | @ home I have a linksys voip device and I am trying to get it to connect to our office asterisk |
| 14:24.28 | last1 | but the isp blocks port 6050 |
| 14:24.31 | last1 | err, 5060 |
| 14:24.35 | [TK]D-Fender | last1: If it's your server, then sure |
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| 14:24.53 | [TK]D-Fender | or you could just setup a VPN, etc (best approach) |
| 14:25.07 | last1 | we have that for other offices using openvpn |
| 14:25.16 | last1 | but that requires a dedicated server so too complicated |
| 14:26.04 | last1 | so running siproxd on our asterisk server, on port 5061 should work ? |
| 14:26.10 | last1 | aka, is it the right software to do this ? |
| 14:29.37 | *** join/#asterisk leftleg (~jburke@nsseofirewall.nsseo.org) |
| 14:29.43 | leftleg | hello all |
| 14:29.57 | *** join/#asterisk superscrat (asanders@nat/digium/x-djzbmenbrttvqxmh) |
| 14:30.45 | leftleg | i have one asterisk box hosting 2 extentions and 1 conference. I am able to call the 2 extension but I am unable to call the conference from my other 6 pbxs. did an update break conferences? |
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| 14:31.16 | [TK]D-Fender | leftleg: What does "break conferences" mean? |
| 14:31.45 | [TK]D-Fender | leftleg: and "from my other 6 pbxs" could mean anything as to how things are connected. |
| 14:31.48 | leftleg | ok, i have 2 extensions, 100 and 101 |
| 14:31.56 | leftleg | sip trunk |
| 14:32.02 | [TK]D-Fender | leftleg: The description of your setup is VERY vague. We have no details to go on. |
| 14:32.11 | leftleg | ok well i'm explaining |
| 14:32.22 | *** part/#asterisk bulkorok (~Benjamin_@217.110.197.225) |
| 14:32.22 | [TK]D-Fender | Show us the actual failed call. |
| 14:32.37 | [TK]D-Fender | ~pb |
| 14:32.37 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 14:32.38 | [TK]D-Fender | ^^^ |
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| 14:32.59 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:34.05 | leftleg | hopefully i got everything in the logs |
| 14:34.06 | leftleg | http://pastebin.com/ARYWRbct |
| 14:37.39 | [TK]D-Fender | that isn't a complete call |
| 14:37.48 | [TK]D-Fender | And that is a FREEPBX setup (which is not supported here) |
| 14:38.07 | leftleg | cool, thanks for the help anyways. |
| 14:38.28 | [TK]D-Fender | For which your incoming call is landing in from-trunk and you don't have an inbound route to handle whatever it arrived as |
| 14:38.42 | [TK]D-Fender | #freepbx <---- |
| 14:38.53 | [TK]D-Fender | Join there and we can resume debugging this |
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| 14:51.34 | Dodewapwap | I'm trying to route SIP calls to XMPP but I get this error every time : http://pastie.org/10248897 |
| 14:51.56 | Dodewapwap | I read other people had this issue and fixed it by adding "icesupport=yes" in rtp.conf and sip.conf but it didn't work fr me |
| 14:55.58 | kfife | Can anyone help me understand the automixmon feature as it relates to generating an tone when activated? |
| 14:56.15 | kfife | Some Asterisk 12.x doc seeems to suggest you can set a courtesytone parameter in features |
| 14:56.24 | kfife | but the parser chokes on this parameter |
| 14:57.09 | kfife | Then I tried enabling the parameter in in res_parking, due to the fact it exists there (and was formerly the same as features) |
| 14:57.12 | kfife | no love |
| 14:57.29 | kfife | Tried setting TOUCH_MONITOR_MESSAGE_START=beep |
| 14:57.49 | kfife | TOUCH_MONITOR_MESSAGE_START=beep (in the active call) |
| 14:57.51 | kfife | no love |
| 14:58.02 | cbeyerlein | how to get rid of chan_dahdi / libss7 console messages that are not having a logging priority? "[1] Unhandled optional parameter 0x37 'Echo Control Information'"? |
| 14:58.26 | kfife | Any suggestions?? It appears to be a documented feature, but my wall is full of dents from head-banging. |
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| 14:59.33 | kfife | Beuller?... Beuller?... :-) |
| 15:06.06 | pankid | who should I talk to about editing this page? https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial |
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| 15:06.42 | pankid | got stuck with tls not working, then figured out it was a permission thing. Probably obvious, but might help someone in the future. |
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| 16:58.20 | aquaguy | Hello! Is there a way to make an user access his voicemail when he/she dials his/her own extension number? RIght now we access the voicemail at ext. 0000 but the Grandstream GXV3275s that we use call the user extension by default to check the voicemail is there a way to change this behaviout either on asterisk itself or the Ip phone |
| 16:58.24 | aquaguy | Thank you |
| 16:58.44 | [TK]D-Fender | It's your dialplan.. do whatever you want |
| 17:00.00 | aquaguy | I don't know how to do it that's why Im asking. |
| 17:00.55 | [TK]D-Fender | compare the callerid/devicename, or whatever else you want to what was dialed... then go to voicemailmain instead |
| 17:01.26 | WIMPy | You can configure the voicemail extension announced in the MWI for checking. Most phones will use it. |
| 17:02.43 | WIMPy | And you could for example exit to VoiceMailMain from VoiceMail on a keypres which is hand for remore operation as well. |
| 17:07.16 | aquaguy | WIMPy: thanks |
| 17:07.48 | [TK]D-Fender | Or do it the way you asked |
| 17:08.00 | [TK]D-Fender | Or another different way |
| 17:08.25 | [TK]D-Fender | #100%venn |
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| 18:05.53 | malcolmd | pankid: make a comment on the page with the suggested changes; i'll take a look at it |
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| 20:21.37 | [TK]D-Fender | checkout time, BBL |
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| 21:00.59 | [red] | after making my own rpm, i've finally realized how large or a project asterisk is |
| 21:01.18 | [red] | *of a |
| 21:09.38 | aquaguy | Hello, I'm trying to play some music on hold but does not work / I'm not doing it right. I've created a 8Khz mono .wav file using Adobe Audition. I've uploaded the file to /var/lib/asterisk/moh/icarus. In musiconhold.conf under [default] context i've added mode=files; directory=/var/lib/asterisk/moh/icarus |
| 21:09.44 | aquaguy | the ; are really new lines |
| 21:10.23 | [red] | what error message does the CLI or logs give you? |
| 21:10.49 | aquaguy | On extensions.conf default context I've added: exten=1000,1,Answer; exten=1000,2,MusicOnHold(); The call is answered but I can not hear the music. |
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| 21:12.02 | [red] | 17:10 < [red]> what error message does the CLI or logs give you? |
| 21:13.22 | aquaguy | sorry, no errors |
| 21:14.47 | [red] | i would double check the file format with the file tool |
| 21:15.00 | [red] | file file.wav |
| 21:15.17 | aquaguy | [red]: which tool is that? |
| 21:15.18 | aquaguy | ok |
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| 21:17.32 | aquaguy | [red]: file command not found I've listed the commands with core show help and I've got a command called file convert but not file. |
| 21:18.09 | [red] | it's not an asterisk tool |
| 21:18.34 | [red] | exit the cli |
| 21:18.36 | [red] | man file |
| 21:19.21 | aquaguy | ok sorry again |
| 21:19.38 | aquaguy | moh_high.wav: RIFF (little-endian) data, WAVE audio, mono 8000 Hz |
| 21:20.21 | [red] | that seems fine |
| 21:20.30 | [red] | strange that asterisk isn't throwing an error |
| 21:21.20 | [red] | are you sure asterisk has permissions to read that file? |
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| 21:21.49 | aquaguy | I've set it to 777 so yes. |
| 21:27.47 | aquaguy | this is the only error I can see in the /var/log/asterisk/messages |
| 21:27.48 | aquaguy | [Jun 19 23:26:35] ERROR[14404] ari/config.c: No configured users for ARI |
| 21:28.49 | robmal | manager.conf |
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| 21:46.00 | aquaguy | [red]: http://pastebin.com/FadmuQUF |
| 21:46.30 | aquaguy | That's the log. All I've done is call 1000 and hung up. |
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| 21:59.36 | aquaguy | it's something related to the file. the wav files that came bundled with asterisknow are played |
| 22:00.28 | aquaguy | I don't understand what else I have to do/change in the file it's already a 8Khz mono wav file :S |
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| 22:10.02 | WIMPy | The file must not contain any meta data. |
| 22:14.52 | aquaguy | WIMPy: thank you SO SO much, it was the metadata that Audition was adding |
| 22:16.16 | aquaguy | The music on hold loops? I only have one audio file, when it finishes it replays the same audio or goes silent? |
| 22:16.33 | WIMPy | It repeats |
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