| 00:03.02 | iulhk | <@newtonr>: but what to do with issue, if i send dtmf by using linphone asterisk works fine, if i send dtmf from imsdroid, its crashing, is it just bcoz imsdroid dtmf mode is rfc4733 and my asterisk has rfc2833 or its bcoz of something else ? |
| 00:03.23 | file | until the backtrace is examined that can't be answered |
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| 00:38.05 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 00:44.55 | iulhk | i just compiled asterisk on machine and gave command core show applications , and unable to voicemail voicemailmain , what i missed during compilations ? |
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| 01:10.56 | chandoo__ | phix, i configured 4 extenstions |
| 01:11.39 | chandoo__ | when i dail extenstion i am getting you have dailed an invalid feature access code |
| 01:11.43 | chandoo__ | what is that mean |
| 01:21.04 | [TK]D-Fender | We have no idea |
| 01:21.16 | [TK]D-Fender | You should be SHOWING us the call |
| 01:21.29 | [TK]D-Fender | Not making us guess why something may be happening. |
| 01:21.45 | [TK]D-Fender | If you want an autopsy then give us a body. |
| 01:24.15 | superscrat | ^^^ I've never heard that expression before :) funny |
| 01:40.13 | chandoo__ | [TK]D-Fender, http://pastebin.com/Dq6Mh1fk this is extenstions.conf |
| 01:40.38 | chandoo__ | i am able to sucessfully call outside numbers , but i am not able to call extenstions i configured |
| 01:40.44 | chandoo__ | what is that i am doing wrong |
| 01:41.22 | [TK]D-Fender | chandoo__, I told you to show us THE CALL |
| 01:41.27 | [TK]D-Fender | chandoo__, Not your CONFIGS |
| 01:41.35 | [TK]D-Fender | Asterisk CLI |
| 01:41.40 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 01:42.11 | chandoo__ | okay |
| 01:44.37 | chandoo__ | http://pastebin.com/nRQ7ce5X |
| 01:44.45 | chandoo__ | [TK]D-Fender, that is the error |
| 01:45.39 | [TK]D-Fender | Since when does the PSTN care about 103? |
| 01:45.47 | [TK]D-Fender | What is phonepower supposed to do with that? |
| 01:45.53 | [TK]D-Fender | That isn't a real phone number |
| 01:46.04 | [TK]D-Fender | <PROTECTED> |
| 01:46.24 | chandoo__ | phonepower is the provider i configured in sip.conf |
| 01:46.28 | [TK]D-Fender | I know |
| 01:46.37 | [TK]D-Fender | you sent them a number to dial |
| 01:46.46 | [TK]D-Fender | 103 is not a real phone number |
| 01:46.51 | [TK]D-Fender | Why are you sending them 103? |
| 01:46.56 | chandoo__ | i configured 4 extenstions |
| 01:47.05 | [TK]D-Fender | No |
| 01:47.14 | [TK]D-Fender | you are sending that number TO YOUR PROVIDER |
| 01:47.19 | [TK]D-Fender | WHY? |
| 01:47.38 | chandoo__ | oh |
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| 01:48.36 | chandoo__ | so how it should be |
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| 01:49.56 | chandoo__ | [TK]D-Fender, can you pls show me the sample exten line how it should be |
| 01:50.06 | [TK]D-Fender | chandoo__, Depends what you are trying to do. |
| 01:50.11 | [TK]D-Fender | And you are using a lot of wrong words. |
| 01:50.16 | chandoo__ | calling extenstions 103 |
| 01:50.59 | [TK]D-Fender | SIP DEVICE 103 |
| 01:51.13 | [TK]D-Fender | Do not call a "device" an "extension" |
| 01:51.30 | [TK]D-Fender | your [103] in sip.conf is a "device". Not an "extension" |
| 01:51.37 | [TK]D-Fender | Memorize this now. |
| 01:51.41 | [TK]D-Fender | For real. |
| 01:51.45 | chandoo__ | okay |
| 01:51.59 | chandoo__ | pardon me i am new to this |
| 01:52.05 | [TK]D-Fender | Once that's done you dial a normal device that you don't need to pass a number to like SIP/103 |
| 01:52.49 | [TK]D-Fender | To use a peer device you defined for a provider, etc you do SIP/mphonepower/1234567890 |
| 01:55.55 | *** join/#asterisk infobot (ibot@rikers.org) |
| 01:55.55 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.4.0 (2015/06/04), 11.18.0 (2015/06/04), 1.8.32.3 (2015/04/08); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 01:57.27 | chandoo__ | http://pastebin.com/bCSM2bH8 |
| 01:58.15 | [TK]D-Fender | <[TK]D-Fender> Once that's done you dial a normal device that you don't need to pass a number to like SIP/103 |
| 01:58.17 | [TK]D-Fender | ^^^^^ |
| 01:58.26 | [TK]D-Fender | 103 is not a number you pass to your PROVIDER |
| 01:58.32 | [TK]D-Fender | TEHY do no have anything to do with it |
| 01:58.55 | [TK]D-Fender | 103 is a peer on YOUR system, not a number you pass via ANOTHER peer |
| 01:59.12 | [TK]D-Fender | SIP/103 <-------------------------------- |
| 01:59.44 | chandoo__ | so what changes i have to make to my sip.conf |
| 02:00.14 | [TK]D-Fender | NONE |
| 02:00.20 | [TK]D-Fender | PAY ATTENTION. |
| 02:00.26 | [TK]D-Fender | Fix what you are DIALING |
| 02:00.48 | chandoo__ | in dailing i am entering 103 and press enter |
| 02:00.53 | [TK]D-Fender | .... |
| 02:00.58 | [TK]D-Fender | IN EXTENSIONS.CONF |
| 02:01.14 | chandoo__ | paying attention |
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| 02:01.36 | [TK]D-Fender | You have no dialing rules in your extemnsions.conf to take that number and dial the specific device |
| 02:02.07 | [TK]D-Fender | You need MORE dialplan. |
| 02:02.30 | [TK]D-Fender | Right now You take EVERY number and just throw it out the peer you defined for your provider |
| 02:02.36 | [TK]D-Fender | This is not good enough. |
| 02:02.45 | [TK]D-Fender | You need MORE |
| 02:03.00 | [TK]D-Fender | So make new extensions for these local devices you want to dial |
| 02:04.58 | chandoo__ | okay |
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| 02:09.33 | chandoo__ | [TK]D-Fender, i made like this |
| 02:09.35 | chandoo__ | exten => 101,1,Dial(SIP/101,15) |
| 02:09.35 | chandoo__ | exten => 101,n,Playback(number-not-answering) |
| 02:11.53 | [TK]D-Fender | ok... and? |
| 02:13.59 | chandoo__ | i got it now |
| 02:14.01 | chandoo__ | it is working |
| 02:14.36 | [TK]D-Fender | You should not be throwing _X to your provider. You should have patterns that at least vaguely look like what a real phone # should look like |
| 02:15.12 | [TK]D-Fender | exten => _X.,1,Dial |
| 02:15.23 | [TK]D-Fender | that is anything 2 or more long |
| 02:15.44 | [TK]D-Fender | think about how long the SHORTEST real number should be at least and accomodate that |
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| 02:19.10 | chandoo__ | okay |
| 02:19.31 | chandoo__ | i am getting call ended between devices after 32secs |
| 02:19.34 | chandoo__ | Packet timed out after 32000ms with no response |
| 02:19.45 | chandoo__ | even thought we are talking |
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| 08:10.29 | esupra | ping |
| 08:13.07 | esupra | WARNING[15023] res_config_mysql.c: MySQL RealTime: Failed to update database: Lost connection to MySQL server during query |
| 08:13.07 | esupra | [Jun 26 09:25:36] ERROR[15023] res_config_mysql.c: MySQL RealTime: Ping failed (2006). Trying an explicit reconnect. |
| 08:13.43 | esupra | anybody any idea how to get rid of this? asterisk 11, apt-get installed. this only happens on registrations of sip devices |
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| 10:12.42 | mastahh | Does anyone know much about L5 Asterisk support from Digium? |
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| 10:52.08 | doome_ | hi all! I am just getting into PJSIP, and there is something I cannot solve: I would like to send a fixed PAI header to an endpoint. Anyone knows a way how to do this? |
| 10:52.34 | doome_ | I tried to set the callerid, from_user of the endpoint with no success |
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| 11:04.31 | chandoo__ | hi |
| 11:04.40 | chandoo__ | phix, hi |
| 11:06.18 | chandoo__ | my call is getting disconnected after 32secs |
| 11:06.21 | chandoo__ | [Jun 25 22:50:46] WARNING[101635]: chan_sip.c:3995 retrans_pkt: Retransmission timeout reached on transmission f93a47946ebe4ae2bae93c9785c79062 for seqno 18231 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions |
| 11:06.21 | chandoo__ | Packet timed out after 32000ms with no response |
| 11:06.31 | chandoo__ | phix, this is the error |
| 11:22.30 | doome_ | I managed to do it: have to set callerid from dialplan |
| 11:23.00 | doome_ | is there a nicer way to set outbound caller id? |
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| 11:59.29 | mirela666 | doome_, you can set it in sip.conf for the specific peer/user |
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| 11:59.50 | doome_ | you mean pjsip conf? |
| 12:00.29 | doome_ | i think it just sets the callerid of the calls coming in from the endpoint |
| 12:00.46 | WIMPy | correct |
| 12:01.29 | mirela666 | oh |
| 12:01.59 | mirela666 | i think it might be set on outgoing also |
| 12:02.44 | mirela666 | callerid="Joe Smith" <15555551212> |
| 12:02.49 | mirela666 | for example |
| 12:03.25 | WIMPy | That wouldn't make sense. |
| 12:03.41 | WIMPy | (usually) |
| 12:04.37 | mirela666 | what do you mean? if a user is [132] and he dials 123 will be show as CLID |
| 12:04.59 | mirela666 | and if you set specific callerid= than it will be shown |
| 12:05.01 | WIMPy | What? |
| 12:05.22 | WIMPy | That didn't make sense. You don't dial devices. |
| 12:06.36 | mirela666 | ok i might get it wrong, the goal... |
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| 12:19.05 | Pekoh | SayNumber will play the clip (and it can be heard) when passed directly, ie SayNumber(2) but not when it is passed the results of a variable SayNumber(${COUNT}). CLI shows it is playing the clip but its not heard, any help or pointer appreciated. |
| 12:20.27 | WIMPy | Show us |
| 12:21.17 | Pekoh | exten => 4018,1,Set(COUNT=10) |
| 12:21.17 | Pekoh | exten => 4018,n(start),GoToIf($[${COUNT} > 0]?:goodbye) |
| 12:21.18 | Pekoh | exten => 4018,n,SayNumber(${COUNT}) |
| 12:21.18 | Pekoh | exten => 4018,n,Set(COUNT=$[${COUNT} -1]) |
| 12:21.18 | Pekoh | exten => 4018,n,GoTo(start) |
| 12:21.18 | Pekoh | exten => 4018,n(goodbye),Playback(vm-goodbye) |
| 12:21.18 | Pekoh | exten => 4018,n,Hangup() |
| 12:21.46 | WIMPy | ~pb |
| 12:21.46 | 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. |
| 12:22.10 | Pekoh | <PROTECTED> |
| 12:22.11 | Pekoh | <PROTECTED> |
| 12:22.51 | WIMPy | Did you answer the channel? Not all applications imply an auto-answer. |
| 12:22.59 | Pekoh | serious.... |
| 12:23.01 | Pekoh | sec |
| 12:23.56 | Pekoh | how embarassing |
| 12:24.47 | Pekoh | yeh tutorials I am going through are very explantory but for an older verions, thanks, oh an I know I can use same =>, |
| 12:24.53 | Pekoh | Thanks again |
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| 16:02.13 | mastahh | I am trying to build a fresh install of Asterisk 13 and I am getting this error (http://pastebin.com/5L4tCfiS). I am compiling from svn co http://svn.asterisk.org/svn/asterisk/branches/13. Any ideas? |
| 16:04.22 | file | Asterisk is no longer stored in SVN, what you see there is an old version |
| 16:04.50 | file | and that particularly problem was fixed |
| 16:05.19 | mastahh | file: should I use wget instead? |
| 16:05.27 | file | https://wiki.asterisk.org/wiki/display/AST/Gerrit+Usage#GerritUsage-CloningfromGerrit has the command which will check out a branch using git |
| 16:05.32 | file | or you can grab the tarball of the latest release |
| 16:05.45 | file | it's also mirrored on github at https://github.com/asterisk/asterisk |
| 16:05.52 | Qwell | file: Maybe a build warning from svn might be in order? Big ugly "We have migrated to git. If you are building this, you are out of date." |
| 16:06.21 | Qwell | of course, that would mean committing something to svn... |
| 16:06.45 | file | it'd be nice |
| 16:07.21 | mastahh | thanks for the confirmation file :) |
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| 16:39.33 | tenspeed705 | anyone have a copy of the defenitive guide they are willing to "lend" me? |
| 16:41.02 | rmudgett | ~book |
| 16:41.02 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 16:42.15 | tenspeed705 | thanks mudgett. I seen that before, and google just kept showing me paid copys |
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| 16:48.29 | [TK]D-Fender | https://www.google.ca/#q=asterisk+definitive+guide |
| 16:48.46 | [TK]D-Fender | For me 1st seacrh = 1st 2 links DIRECT to reading free |
| 16:49.45 | *** join/#asterisk Panther_Modern (~Panther_M@unaffiliated/panther-modern/x-6168176) |
| 16:50.59 | [TK]D-Fender | Actually almost ALL of the first page of results lead direct to "free onlinoe right here, right now" |
| 16:51.08 | [TK]D-Fender | Also from *'s main site : http://www.asterisk.org/community/documentation |
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| 19:01.37 | BakaKuna | did anyone here ever post json data using the Asterisk Curl application? |
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| 19:34.12 | Katty | russellb: YOU SIR |
| 19:34.24 | Katty | russellb: are to have a lovely birthday. |
| 19:34.33 | Katty | russellb: <3 (also, happy birthday) |
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| 23:02.54 | chandoo | hi |
| 23:02.57 | chandoo | good evening |
| 23:03.22 | chandoo | hi phix |
| 23:03.30 | chandoo | hi [TK]D-Fender |
| 23:08.30 | chandoo | when i dail device i am not able to end the call |
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