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| 00:29.18 | drmessano | marlinc: Use the one from mariadb |
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| 08:22.02 | ruied | Hi. I'm starting to use pjsip, local extensions are working and trunk registers ok, but when I make an outbound call it says busy/congested. With pjsip debug on it reports 489 bad event. http://pastebin.com/UJVjcsYN |
| 08:22.38 | wyoung | dang |
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| 09:20.03 | AviiNL | Hey all, is it possible to check from inside an AGI script if an extension exist in the current context? I listen for DTMF and go to extension n if someone presses a key, but I don't want it to go to that extension if it does not exist. |
| 09:21.16 | wasanzy | hello |
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| 09:23.12 | wyoung | hi wasanzy |
| 09:23.14 | wyoung | sup? |
| 09:23.18 | wasanzy | usually when you call an IVR, you are presented with a prompt. eg. for english press1.. if you did not press 1, the prompt will repeat again say 3 times before the next sequence is executed. |
| 09:23.25 | wasanzy | how do I do that in Asterisk? |
| 09:23.36 | wasanzy | I want a propmpt to be repeated for at least 3 times |
| 09:23.42 | wasanzy | prompt |
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| 10:06.00 | stefan27 | after doing a Dial(SIP/stefansnomdevpbx3) to a sip device that responds 480, DIALSTATUS is set to BUSY. But can I also get access to the value 480 In the dialplan succeeding the Dial call? |
| 10:11.30 | wasanzy | any help please? |
| 10:17.56 | Rasputin3711 | Background(prompt1&prompt2&prompt3) |
| 10:18.29 | wasanzy | Rasputin3711: me? |
| 10:18.32 | Rasputin3711 | yep |
| 10:18.49 | Rasputin3711 | https://wiki.asterisk.org/wiki/display/AST/Background+and+WaitExten+Applications |
| 10:18.58 | wasanzy | ok |
| 10:19.01 | wasanzy | let me read |
| 10:19.25 | *** join/#asterisk pawiecki (~pawiecki@router.dir.pl) |
| 10:24.26 | pawiecki | In my scenario - caller dials peer, but before peer answers, he hangs up. I want to go through next priorities, instead of Hangup. I tried with Dial option 'F', but then ${DIALSTATUS} is not set. I need dialstatus in next steps. Is it possible? |
| 10:27.25 | KValchev | pawiecki: yes, use dial option 'g' |
| 10:28.57 | pawiecki | KValchev: yes, i'm using 'g' option, and it works, but only if the destination channel hangs up. I want to have the same behaviour if the calling party hangs up. |
| 10:29.52 | KValchev | stefan27: use ${HASH(SIP_CAUSE,<channel-name>)} |
| 10:31.54 | KValchev | pawiecki: which asterisk version you use ? |
| 10:34.06 | pawiecki | 11.18.0 |
| 10:34.11 | Samot | Uhm. |
| 10:34.38 | Samot | Catch the Hang Up Cause |
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| 10:47.08 | pawiecki | Samot: how can i do it? |
| 10:51.29 | KValchev | pawiecki: ok, use h ext in same context |
| 10:52.10 | pawiecki | KValchev: that makes sense, will try it out. |
| 10:53.08 | Samot | https://www.irccloud.com/pastebin/3k6YVmBx/ |
| 10:53.15 | Samot | Something like that. |
| 10:53.18 | KValchev | pawiecki: have, try it |
| 10:58.51 | stefan27 | thanks KValchev, that did not seem to work on asterisk 13.5.0, however ${HANGUPCAUSE(<channel-name>,tech)} did work |
| 10:59.09 | stefan27 | but I don't know how to get channel-name of the dialed channel |
| 10:59.51 | stefan27 | ${HANGUPCAUSE_KEYS()} happened to do it in this case, but only coincidently, since it returns a list of all channel names that have hang up causes |
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| 11:02.42 | Samot | Set(CHANNEL=${CHANNEL}) |
| 11:02.52 | Samot | Boom, channel name. Of course this is SIP channels. |
| 11:03.54 | stefan27 | I have exten => s,n,Dial(SIP/stefansnomdevpbx3) followed by exten => s,n,NoOp(${DIALSTATUS}) how would I access the dialed (hung up) channel name ? e.g. NoOp(${SOME_FUNC()}) would print NoOp(stefansnomdevpbx-0000445). the variable $CHANNEL points to SIP/molly (the executing channel) |
| 11:06.56 | KValchev | to get access to next priority must use dial option g and F |
| 11:08.26 | KValchev | but if caller hangup you cant access it. For that reason use h ext in same contex to hanle hangup |
| 11:09.07 | KValchev | some like h,1,NoOp(${DIALSTATUS}) |
| 11:09.27 | stefan27 | execution does proceed to the next priority after Dial even without g option |
| 11:09.57 | KValchev | h,n,NoOp(${HANGUPCAUSE(${CHANNEL},tech)}) |
| 11:10.12 | stefan27 | the use case is that the caller never hangs up, but the CALLEE responds with 480 |
| 11:12.54 | KValchev | aha |
| 11:13.40 | Samot | I don't know. I don't send a g or f option with the Dial() command |
| 11:14.04 | *** join/#asterisk sekil (~sekil@jabber.net011.net) |
| 11:14.37 | Samot | I don't send any options except for timeout and I'm able to catch the DIALSTATUS, HANGUPCAUSE and process the results after Dial() has completed. |
| 11:15.20 | stefan27 | even in the event the call was never established? (rejected by 480)? |
| 11:15.32 | KValchev | immediately after dial use NoOp(${HANGUPCAUSE(${CDR(dstchannel)},ast)}) |
| 11:15.46 | Samot | Generally a 480 mean Temporarily Unavailable. |
| 11:16.09 | Samot | stefan27: In most cases that code is the result of an ISDN code. |
| 11:16.21 | Samot | Multiple ISDN codes translate to the same SIP code. |
| 11:16.54 | WIMPy | And off course the location is missing. |
| 11:16.58 | KValchev | stefan27: try it |
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| 11:17.09 | stefan27 | yes, that's a different issue, $CDR does not work (the cdr module is off i think, and i want it off) |
| 11:17.37 | stefan27 | ${HANGUPCAUSE(<hung-up-channel-name>,tech)} is the only thing that worked so far |
| 11:18.31 | stefan27 | but I have no yet no reliable way of getting <hung-up-channel-name> |
| 11:18.53 | stefan27 | thanks for the tips though, i'll continue experimenting |
| 11:19.09 | *** join/#asterisk sekil (~sekil@jabber.net011.net) |
| 11:19.09 | file | https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_HANGUPCAUSE_KEYS |
| 11:19.18 | file | and https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause |
| 11:19.53 | stefan27 | yes I used a combination of those to begin with, but HANGUPCAUSE_KEYS says: Returns a comma-separated list of channel names to be used with the HANGUPCAUSE |
| 11:19.53 | stefan27 | function. |
| 11:20.21 | file | right... |
| 11:20.24 | stefan27 | would that list return all channels in the pbx that currently have hangup causes, or only those related to my channel? |
| 11:20.30 | file | only those related to your channel. |
| 11:21.01 | KValchev | for failover dial based on remote sip cause you must know remote channel name, this is imposible only with CDR module |
| 11:21.31 | *** join/#asterisk rcp (~z@217.155.200.159) |
| 11:21.31 | stefan27 | Ah, then it will work, and if i do multiple dials and I'm only interested in the hangup cause in one of them I guess i can do processing on the output of HANGUPCAUSE_KEYS |
| 11:21.57 | rcp | guys i've got a voip server, but the call isnt reaching the handset, how can I find out what ports I should be forwarding on the router where the handset is? |
| 11:21.59 | stefan27 | thanks, i'll glue together something that works will work |
| 11:22.11 | rcp | it was working now stopped. |
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| 12:28.31 | tzafrir | Any idea when do the patents required for implementing G.729 and/or G.723 expire? |
| 12:32.11 | Samot | No idea. |
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| 12:34.20 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 12:36.14 | [TK]D-Fender | rcp, Show us the failed call. |
| 12:36.16 | [TK]D-Fender | ~pb |
| 12:36.17 | infobot | rumour has it, 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://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 12:36.18 | [TK]D-Fender | ^^^ |
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| 12:54.48 | *** mode/#asterisk [+o cresl1n] by ChanServ |
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| 13:19.31 | ruben23 | hi there guys any idea how to dial from asterisk CLI> (what command on it please) |
| 13:20.50 | [TK]D-Fender | consle <tab> |
| 13:21.04 | [TK]D-Fender | console <tab> |
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| 13:25.15 | *** join/#asterisk s8y (~s_8y@host217-36-11-53.in-addr.btopenworld.com) |
| 13:27.17 | s8y | hello, I am trying to use if in my dialplan and getting no application 'IF'. can you point me to example how to use if? |
| 13:28.21 | ruben23 | [TK]D-Fender: im having hard time to assemble the correct command |
| 13:28.24 | *** join/#asterisk KValchev (~KValchev@ns.atsoftconsult-bg.com) |
| 13:29.27 | pawiecki | how can i put called party number into voicemail mail template? |
| 13:29.39 | [TK]D-Fender | ruben23, read the CLI help |
| 13:30.01 | [TK]D-Fender | "core show help console dial" |
| 13:30.07 | [TK]D-Fender | or keep hitting <tab> |
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| 13:30.32 | [TK]D-Fender | s8y, There is no "IF" application |
| 13:30.53 | [TK]D-Fender | s8y, "IF" is a FUNCTION. |
| 13:31.11 | s8y | how do I use IF function in dialplan then? |
| 13:31.19 | [TK]D-Fender | Like any other function |
| 13:32.06 | [TK]D-Fender | exten => extension,priority(label),Application(Data - Including FUNCTIONS) |
| 13:32.13 | s8y | that's my offending line same => n,IF(${keeprecording} = 0 ?[NoOp(delete ${MIXMONITOR_FILENAME})]:[NoOp(keep, update mysql)]) |
| 13:33.06 | [TK]D-Fender | no |
| 13:33.19 | [TK]D-Fender | IF is not an APPLICATION |
| 13:33.27 | [TK]D-Fender | Wher you have it on thatt line is as an application |
| 13:33.35 | [TK]D-Fender | n = priority |
| 13:33.46 | [TK]D-Fender | <[TK]D-Fender> exten => extension,priority(label),Application(Data - Including FUNCTIONS) |
| 13:33.47 | [TK]D-Fender | ^^ |
| 13:33.50 | [TK]D-Fender | Whattt comes next? |
| 13:33.53 | [TK]D-Fender | the Applicastion |
| 13:33.57 | [TK]D-Fender | And "IF" isn't one |
| 13:34.48 | s8y | ok. how would you put if else condition in dialplan then? |
| 13:35.01 | [TK]D-Fender | It looks like you're looking to use "ExecIf" |
| 13:37.29 | s8y | ok thanks, will have a look |
| 13:49.10 | s8y | is thre else statement in execif? |
| 13:51.21 | [TK]D-Fender | "core show application execif" <- |
| 14:00.22 | *** join/#asterisk putnopvut (putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:00.23 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 14:04.23 | s8y | i altered my line as per documentation same => n,ExecIf($["${keeprecording}" = 0]?system(rm -f ${MIXMONITOR_FILENAME})[:NoOp(keep, update mysql)]) |
| 14:05.40 | s8y | but its is still not working as expected (file is not getting deleted). Is there a way to validate the systax other than executing it? |
| 14:05.57 | *** join/#asterisk sh_smith (~smuxi@cpe-76-174-26-91.socal.res.rr.com) |
| 14:07.06 | KValchev | remove quota from keeprecording |
| 14:08.40 | KValchev | or put quotes on zero for string comparison |
| 14:10.04 | *** join/#asterisk SunGod- (~SunGod@92.55.116.125) |
| 14:11.35 | s8y | Kvalchev: thanks for suggestion, it doesn't work either way |
| 14:12.26 | s8y | Kvalchev: neither for same => n,ExecIf($["${keeprecording}" = "0"] nor same => n,ExecIf($[${keeprecording} = 0] |
| 14:14.46 | *** join/#asterisk kharwell (kharwell@nat/digium/x-zvdbsgbwhpkwqsvu) |
| 14:19.10 | [TK]D-Fender | We have no proof of what that variable is holding <- |
| 14:19.15 | *** join/#asterisk Chotizei (chotaire@oahu.chotaire.net) |
| 14:19.22 | [TK]D-Fender | Time for some quality control sanity checks |
| 14:19.41 | lee | is libpri/dahdi supported here, or only asterisk? |
| 14:19.49 | [TK]D-Fender | lee, all |
| 14:20.07 | lee | ok great. question time... |
| 14:22.43 | s8y | [TK]D-Fender: i output same => n,NoOp(recorded File ${MIXMONITOR_FILENAME} do we keep it ${keeprecording} ) |
| 14:22.56 | s8y | and it returns NoOp("PJSIP/16405-00000068", "recorded File /var/spool/asterisk/monitor/2016/09/07/16403-16405-1473257459.138.wav do we keep it No ") |
| 14:23.18 | [TK]D-Fender | "No" sure doesn't look like "0" to me.... |
| 14:23.23 | [TK]D-Fender | ^ |
| 14:23.26 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-zqshuavzlqyebnqj) |
| 14:27.00 | s8y | [TK]D-Fender: I changed it to same => n,ExecIf($[${keeprecording} = No ]?system(rm -f ${MIXMONITOR_FILENAME}) |
| 14:27.04 | s8y | but still no luck |
| 14:27.05 | pawiecki | Calling party is anonymous, and calls at VoieMail(123). How can I label caller as 'Anonymous' in VM mail template, that is sent? |
| 14:27.23 | pawiecki | *VoiceMail(123) |
| 14:29.25 | [TK]D-Fender | s8y, Show complete code with the complete debug |
| 14:29.41 | [TK]D-Fender | s8y, You are doing this in pieces and new erros keep coming to light |
| 14:31.51 | s8y | [TK]D-Fender: good point, just a sec |
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| 14:34.21 | *** mode/#asterisk [+o mjordan_] by ChanServ |
| 14:36.00 | *** join/#asterisk sekil (~sekil@jabber.net011.net) |
| 14:39.24 | s8y | [TK]D-Fender: http://fpaste.scsys.co.uk/533263?tx=on&ln=on&submit=Format+it%21 |
| 14:39.58 | [TK]D-Fender | ($[${keeprecording} = No ] |
| 14:40.14 | [TK]D-Fender | immediate lesson, never shove extra spaces where not called for |
| 14:40.39 | *** join/#asterisk ambrogiodev (~ambrogiod@31.44.120.18) |
| 14:40.40 | ambrogiodev | 444 |
| 14:40.53 | ambrogiodev | Hi |
| 14:41.04 | lee | I have a Digium TE133 card, I'm trying to run the loopback test but it's saying "no such device", and I have no idea why. http://pastebin.com/qTuHw2Ma |
| 14:41.07 | ambrogiodev | I'll need some help about Asterik |
| 14:41.10 | [TK]D-Fender | s8y, And you have extra junk on the end in [] |
| 14:41.21 | [TK]D-Fender | [: <- not appropriate |
| 14:42.25 | [TK]D-Fender | ~ask |
| 14:42.26 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 14:42.28 | [TK]D-Fender | ambrogiodev, ^^^ |
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| 14:43.07 | ambrogiodev | What is the best library to use with Asterisk server between PJSIP, PortSIP, Linphone ? |
| 14:44.18 | [TK]D-Fender | ambrogiodev, * doesn't care whatt your client uses |
| 14:44.32 | ambrogiodev | why ? |
| 14:44.38 | [TK]D-Fender | ambrogiodev, Because it doesn't. |
| 14:44.42 | [TK]D-Fender | SIP is SIP |
| 14:44.52 | [TK]D-Fender | That call to * is either placed properly, or it isn't. |
| 14:47.48 | ambrogiodev | according with your experience, did you find some particular problems with PortSIP ? |
| 14:48.25 | ambrogiodev | I know that the SIP protocol is the same but my boss ask for more details |
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| 14:51.04 | pawiecki | s8y: you might want to put quotes around values: $[ "${keeprecording}" = "No" ] |
| 14:55.38 | s8y | pawiecki: thanks for suggestion, it is still no luck. |
| 14:55.56 | s8y | i changed it to same => n,ExecIf($[ "${keeprecording}" = "No"]?system(touch /tmp/test_execifYes)system(tout /tmp/test_execIfNo)) same => n,system(touch /tmp/test_system) |
| 14:56.49 | s8y | and then list filesystem context with !ls /tmp/test* and only /tmp/test_system file is there |
| 14:57.20 | [TK]D-Fender | Stil a mess there |
| 14:57.34 | [TK]D-Fender | execifYes)system(tout <--- no delimiters, etc |
| 14:58.12 | [TK]D-Fender | Also do ANOTHER app to prove from CLI thatt it's executing at all |
| 14:58.18 | [TK]D-Fender | ie NoOp |
| 14:58.19 | pawiecki | in first line you have an error "toutc" instead of "touch". Maybe check what is the value of ${keeprecording} => NoOp(${keeprecording}) |
| 14:59.00 | pawiecki | *'tout' instead of 'touch' |
| 14:59.35 | [TK]D-Fender | As well |
| 15:00.11 | [TK]D-Fender | Then while you're at itt.... I'd look at the file and prove it's there and you even have rights to it |
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| 15:04.29 | s8y | thanks for looking into it. I have done sanity check with same => n,ExecIf(1=1?system(touch /tmp/test_execifYes)system(touch /tmp/test_execIfNo)) |
| 15:04.52 | [TK]D-Fender | no |
| 15:04.57 | [TK]D-Fender | STILL have shit glued together |
| 15:05.07 | lee | lol, awesome. can't register for a digium account to get support because of an automatic IP address check failure |
| 15:05.15 | lee | fucking stupid |
| 15:05.20 | [TK]D-Fender | 1=1 <- NOT an expression |
| 15:05.33 | [TK]D-Fender | fYes)system <- again, NO DELIMITER where there should be one |
| 15:05.42 | [TK]D-Fender | And not prrof that the file is in the right place & condition |
| 15:06.53 | [TK]D-Fender | And I told you to test with ANOTHER Application |
| 15:06.57 | [TK]D-Fender | Noop <- |
| 15:13.16 | ruied | pjsip receiving some NOTICE from opensips softswitch provider and can't receive calls... |
| 15:13.32 | ruied | http://pastebin.com/ELMdCpzG |
| 15:14.21 | ruied | "No matching peer endpoint found" |
| 15:14.25 | [TK]D-Fender | opttions != call |
| 15:15.32 | ruied | the endpoint g9 seems to exist |
| 15:16.20 | [TK]D-Fender | and that endpoint does not have an AOR to match |
| 15:16.29 | [TK]D-Fender | So .. that endpointt is not going to happen... |
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| 15:55.57 | s8y | hello, with your help I managed to resolve my execIf statement. in case you will ever need to use it correct syntax is same => n,ExecIf($[ "${keeprecording}" = "No"]?system(touch /tmp/test_execifYes):system(touch /tmp/test_execIfNo)) |
| 15:56.05 | s8y | thank you all for your help |
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| 15:58.35 | [TK]D-Fender | Only told him 5 times that he was missing a delimiter... |
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| 17:17.49 | zhold | Hey guys, having trouble here using Amfeltec Piranha USB-FXO Adapter on Asterisk 11.13 on my BeagleBone running Debian 8.4 |
| 17:18.01 | zhold | anyone ever used one of these adapters before? |
| 17:21.11 | zhold | $100 BTC reward for anyone who can help me config this adapter with dahdi/asterisk |
| 17:21.31 | zhold | F#@!# :( |
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| 17:28.07 | robmal | zhold: Never seen it, what's the problem? |
| 17:28.10 | lorsungcu | zhold: does the usb port output enough power? |
| 17:28.36 | lorsungcu | looks like it needs very close to the maximum available |
| 17:34.56 | zhold | lorsungcu: ohh gee the power :/ let me check |
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| 17:53.30 | somepoortech | zhold: you can check lsusb and make sure the device is usable |
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| 18:00.11 | robmal | Money, making people help you since 350 B.C. |
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| 18:01.28 | somepoortech | I think currency has been around a bit longer than that, but yeah it is a good motivator |
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| 18:25.43 | jrun | i am under the impression that there can be a section in pjsip.conf of type 'contact' |
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| 18:27.05 | jrun | but when i try to pass the name of that section as the value of the var 'contact' under a given aor i get `exceeding uri lenght limit of pjsip' error msg and in fact aor object is not created |
| 18:27.08 | jrun | any idea? |
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| 18:32.48 | pawiecki | Hi. Is #including external * config files viewed as a bad practice, or is it irrelevant? I find it hard to trace and manage dialplan flow, with many included files, without explicit notice about jumping between them. |
| 18:33.48 | cervajs2 | hi, what has better performance FastAGI or CURL in dialplan? |
| 18:34.45 | cresl1n | Not sure |
| 18:34.55 | cresl1n | They both have pros and cons |
| 18:35.34 | cervajs2 | cresl1n: what are cons with CURL? |
| 18:42.00 | cresl1n | how were you planning on using CURL? |
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| 18:48.28 | igcewieling1 | pawiecki: I keep #include to a minimum, but it is very helpful. |
| 18:49.33 | igcewieling1 | cervajs2: I'd expect fastAGI to be faster, if for no other reason than you have one fewer fork/exec, but I doubt it makes any differenct other than a highly loaded system. |
| 18:50.50 | igcewieling1 | I do heavy AGI processing on one of our servers and at 102 channels (51 calls) load average is usually under 1 |
| 18:51.01 | igcewieling1 | that's not even using FAST agi, just regular AGI. |
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| 19:50.30 | cervajs2 | <PROTECTED> |
| 19:51.57 | cervajs2 | developing and testing web app server is much easier than agi server development and automated testing |
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| 20:58.54 | KNERD | i got 11.20 running..some are complaining of delayed vociemail in box..meaning they check the box and says "no new messages" but then some time later, it will say new messages from some days, weeks ago. Anyone know of any bug offhhand? |
| 20:59.30 | KNERD | upgrading now to 11.23 (current) to see if that makes any difference |
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