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| 00:52.49 | Docfxit | Can I get any help here for FreePBX? |
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| 01:31.50 | snadge | would it greatly upset anyone if i asked a question relating to asterisk 1.6? |
| 01:32.05 | snadge | i know [TK]D-Fender has been around for a while.. i might need to consult the wise elders on this ;) |
| 01:32.57 | snadge | we actually have plans to move to asterisk 11.. but some changes are required in our billing system (web management interface) relating to the realtimedb and , and | etc |
| 01:33.35 | snadge | why not 13 you ask? .. well.. there's apparently some other issue relating to CDRs.. they're not coming out in an expected format, and some compatibility options have been removed :| |
| 01:35.39 | snadge | so im looking at our 5 core 1.6 servers.. the first four were using phtread timing, the fifth.. dahdi.. and we were having issues with the fourth one that were load related |
| 01:36.11 | snadge | being the knob that i am.. i decided to uncomment the noload for timerfd.. and switched 4 and 5 to timerfd |
| 01:37.18 | snadge | so for about 10 minutes.. we were seeing peers flapping etc on both 4 and 5.. and im just wondering if theres a known issue with 1.6.2.21 and timerfd |
| 01:41.05 | *** join/#asterisk O47m341 (~Suzeanne@75-103-145-152.ccrtc.com) |
| 01:44.41 | snadge | my inclination tells me to just use dahdi.. thats what shitty old versions of asterisk seem happy with ;) |
| 01:44.57 | snadge | especially running in kvm |
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| 01:50.11 | *** join/#asterisk aneesiqbal (~steelbrai@103.255.5.44) |
| 01:54.25 | aneesiqbal | Anybody know if asterisk emits any security events on AMI? Events like failed login |
| 01:56.36 | drmessano | snadge: What kernel is it running? |
| 02:05.08 | [TK]D-Fender | aneesiqbal, You know Google would have answered that requiring fewer words .... https://www.google.ca/#q=asterisk+AMI+security+events |
| 02:07.22 | aneesiqbal | [TK]D-Fender: I did some google searches but my term only found me some SIP security events (not emitted over AMI) and some log file stuff that is not my target, but Thanks :) |
| 02:07.57 | [TK]D-Fender | What specifically is your target? |
| 02:09.04 | aneesiqbal | I have a node js daemon that monitors everything AMI throws at it, I want to get PJSIP authentication failure events and if some IP tries to login too many times, to execute `ufw reject from $IP` on it |
| 02:11.37 | [TK]D-Fender | There is a new security event system as of * 12 |
| 02:12.40 | aneesiqbal | Yep, that's what I read there, but still searching for them here https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+AMI+Events |
| 02:13.00 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ManagerEvent_AuthMethodNotAllowed |
| 02:13.14 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ManagerEvent_ChallengeResponseFailed |
| 02:13.15 | [TK]D-Fender | etc |
| 02:13.36 | aneesiqbal | Wow, how did I look over them? Thanks man! You da MVP |
| 02:16.34 | *** join/#asterisk italorossi (~Adium@179.156.47.253) |
| 02:16.41 | [TK]D-Fender | Google hits the page. Just read the list. |
| 02:16.51 | [TK]D-Fender | You ask specific to AMI.. and the list is right there |
| 02:16.58 | [TK]D-Fender | just have to be throrough about it |
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| 02:53.44 | Lope | i've got playback as my first item for one of my extensions. Sometimes it starts playing before the audio is connected on the call and I miss about 2 seconds of audio. I tried adding Wait(5) as my first instruction and then Playback with n priority as the 2nd instruction, but it still plays the sound at 0 seconds or -2 seconds etc. |
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| 02:57.21 | Lope | WaitForSilence(1000) seems to do the trick |
| 02:58.10 | Lope | But might be a problem if the caller is calling from a noisy environment. I'll try it again, screaming into the phone. "nanananana can't hear anything nanannanana" |
| 02:58.14 | Lope | :p |
| 02:59.25 | Lope | Lol it doesn't seem to wait for silence, even if I start the call making a noise. But it does wait until the call is connected before playing the sound, which I wanted. |
| 03:00.17 | Lope | The one time I tried WaitForSilence(5000) then it did make me wait... Perhaps it detects 1 second of "silence" while the call is still connecting. |
| 03:05.33 | [TK]D-Fender | forcible make it wait 1-2 seconds |
| 03:05.58 | Lope | I see some people use a fancier syntax, same => n,Blah() |
| 03:06.03 | [TK]D-Fender | and do so by playing silence (as it forces audio to be established) |
| 03:06.05 | Lope | is that recommended? |
| 03:06.09 | [TK]D-Fender | Playback(silence/2) |
| 03:06.14 | [TK]D-Fender | in those cases, yes |
| 03:06.30 | Lope | Forcible? |
| 03:07.01 | Lope | Okay I'll check if I have a silence sound file? or is it built into asterisk? |
| 03:07.56 | Lope | cool, found the page, thanks for the tip :) https://wiki.asterisk.org/wiki/display/AST/Answer%2C+Playback%2C+and+Hangup+Applications |
| 03:12.35 | Lope | Playback(silence/2&/path/to/sound) works |
| 03:22.32 | Lope | Is there any point in including the Answer() command? |
| 03:25.05 | [TK]D-Fender | Gives an answer state aside from audio |
| 03:25.13 | [TK]D-Fender | When in doubt always do both |
| 03:27.49 | Lope | Ok. can you explain what the logic of this is? http://codepad.org/6FMVzevD |
| 03:28.15 | [TK]D-Fender | Just read it |
| 03:28.37 | Lope | basically I'm transferring calls from a DID with unlimited channels to my softphone using Dial. If there's already a call in progress I want to divert them to voicemail |
| 03:28.44 | [TK]D-Fender | it's just 1 step after another... and an occasional jump |
| 03:29.15 | [TK]D-Fender | "core show function DEVICE_STATE" <- |
| 03:29.24 | Lope | It says if the dialstatus is busy, goto busy, otherwise go to unavailable. What's the difference? |
| 03:29.41 | [TK]D-Fender | that means exactly what it says |
| 03:29.53 | Lope | what does unavailable mean? |
| 03:29.58 | [TK]D-Fender | if the phone SAYS it's busy... it'll go to one place... if not ... it'll go to another |
| 03:30.07 | [TK]D-Fender | that's a LABEL |
| 03:30.10 | [TK]D-Fender | GOTO FRED |
| 03:30.12 | [TK]D-Fender | GOTO HOME |
| 03:30.17 | Lope | oh right |
| 03:30.24 | [TK]D-Fender | Read up on your dialplan basics |
| 03:30.34 | [TK]D-Fender | And read the app's instructions |
| 03:30.40 | Lope | What confuses me is what happens if it dials successfully? |
| 03:30.50 | Lope | then it goes to unavailable? |
| 03:31.02 | [TK]D-Fender | read Dial()'s instructions |
| 03:31.08 | Lope | Ok |
| 03:31.45 | [TK]D-Fender | Don't shortcut the actual understanding of the individual dialplan apps. This is what is your job to read in the first place to know what you'll need to do what you want. |
| 03:31.52 | [TK]D-Fender | And this isn't it |
| 03:32.14 | [TK]D-Fender | because that sample looks at the result of TRYING to call it. |
| 03:32.32 | [TK]D-Fender | You are trying to determine whether you SHOULD call it it in the first place |
| 03:32.38 | [TK]D-Fender | And I already gave you the tool for that |
| 03:33.09 | Lope | yes. So that dialplan would not be suitable for me. |
| 03:33.25 | Lope | i thought it was a bit strange that it was dialling without checking if it's busy first. |
| 03:33.41 | Lope | kind of like it's doing things backwards. |
| 03:34.06 | Lope | Reading up on the 2 things you recommended. |
| 03:36.25 | Lope | Is a softphone only cabable of answering one call simultaneously? |
| 03:36.37 | Lope | (having 1 conversation at a time) |
| 03:44.02 | Lope | SIP/testsip0 state NOT_INUSE |
| 03:44.07 | Lope | nice |
| 03:44.56 | Lope | can I query variables right from the terminal? |
| 03:45.45 | Lope | I tried running this in the terminal, but it didn't like it. (works in my dialplan) NoOp(SIP/testsip0 state ${DEVICE_STATE(SIP/testsip0)}) |
| 03:50.00 | Lope | I've tried a few things like `echo NoOp(hello)` `show NoOp(hello)` `core show NoOp(hello)` etc |
| 03:51.22 | Lope | I've tried many googles like "asterisk variables console" 'asterisk variable state terminal" etc. |
| 03:52.35 | Lope | core show ${DEVICE_STATE(SIP/testsip0)} etc |
| 03:54.04 | Lope | `dialplan show chanvar ${DEVICE_STATE(SIP/testsip0)}` Channel '${DEVICE_STATE(SIP/testsip0)}' not found |
| 03:55.10 | Lope | `dialplan global show chanvar ${DEVICE_STATE(SIP/testsip0)}` |
| 03:58.03 | Lope | Okay I'll just debug by calling for now. So it looks like basically my softphone device state must be NOT_INUSE for me to attempt dialing it. |
| 03:58.11 | Lope | Anything else is a job for voicemail. |
| 04:29.14 | *** join/#asterisk aneesiqbal (~steelbrai@103.255.5.81) |
| 04:31.35 | Lope | I've finished my dialplan, now to test it. http://codepad.org/RA9ebjeh |
| 04:33.02 | Lope | It seems I need to make an entry for my extension in the voicemail config. |
| 04:38.24 | [TK]D-Fender | Looks like progress... |
| 04:39.00 | Lope | I'd be interested to know how to check variable's state in the terminal? |
| 04:40.08 | Lope | there are a mountain of settings in voicemail.conf |
| 04:52.50 | *** join/#asterisk O47m341 (~Suzeanne@75-103-145-152.ccrtc.com) |
| 04:54.17 | Lope | Cool I was able to leave voicemail successfully and play it by digging in the /var/spool/... dir and playing the resulting wav file. |
| 04:55.56 | Lope | So the dialplan works perfectly :) thanks for your help. |
| 04:56.34 | Lope | What convenient open source options are available for checking voicemail? I think checking voicemail in a web-browser would be most preferable. |
| 05:02.08 | snadge | drmessano, to answer your question earlier (sigh).. 3.4.4-3.fc17.x86_64 |
| 05:02.12 | snadge | its been one of those days.. |
| 05:02.47 | snadge | yes.. its a fedora 17 system.. yes.. its running a random kernel update from within that release |
| 05:02.57 | snadge | no.. that wasn't my idea |
| 05:03.19 | drmessano | Seems like I recall timerfd not being so great in 1.6.2.. but did you verify it was loaded? |
| 05:03.26 | snadge | yeah its definitely loaded |
| 05:03.39 | snadge | the systems that are "stable" *cough* are using pthread |
| 05:03.54 | snadge | and the fifth one was randomly using dahdi.. gotta love that consistency |
| 05:04.32 | snadge | so at a whim i changed the 4th and 5th one to timerfd.. and they both exhibited a load condition with peers flapping for about 10 minutes today.. none of the other systems did |
| 05:05.05 | snadge | so my thoughts are, mixed with a tiny amount of experience, is that dahdi is probably a better option |
| 05:05.14 | drmessano | Probably |
| 05:05.30 | snadge | since its higher precedence than pthread.. and timerfd apparently may have an issue, despite being the highest priority (normally) |
| 05:06.23 | snadge | its quite possible that they were all once on dahdi.. and they have been rebooted since a kernel has been updated or something like that.. and fallen back on pthread.. and we've just not noticed |
| 05:06.40 | drmessano | I efforted it get it working back then, and #1 issue was I believe I was on CentOS 5 at the time which didn't support it.. and when I moved to Ubuntu (and a newer kernel) it just didn't work well with 1.6.2.. but 1.8 onwards |
| 05:06.41 | snadge | especially since i can see dahdi in /usr/src |
| 05:06.52 | drmessano | Ah |
| 05:06.58 | drmessano | That would make a lot of sense |
| 05:07.33 | drmessano | Kernel update, nothing to notice because everything is "operating" except timing |
| 05:08.24 | drmessano | That might be the route to pursue.. Try and build it for the new kernel |
| 05:09.05 | snadge | yeah we dont actually need dahdi as such.. its all just sip traffic with no hardware as such |
| 05:09.23 | snadge | but.. my understanding is that dahdi provides a timing source, even with the absense of hardware it supports |
| 05:09.32 | drmessano | Right |
| 05:09.45 | snadge | especially on older versions of asterisk and older kernels |
| 05:09.45 | drmessano | and pthread just sucks.. |
| 05:09.48 | snadge | exactly |
| 05:09.53 | [TK]D-Fender | Lope, "core show application voicemailmain" <- |
| 05:09.57 | snadge | amazing that we haven't noticed it |
| 05:10.10 | drmessano | You have now |
| 05:10.13 | drmessano | lol |
| 05:11.02 | drmessano | Im so glad the need for DAHDI is gone in later branches |
| 05:19.58 | Lope | TK-D Fender. Yeah I don't want to call-in to check for messages. I want to be able to browse existing voice messages in a browser, is that possible? |
| 05:20.25 | [TK]D-Fender | Lope, It's your server. You should see where * put those files... go work something up to host them |
| 05:20.30 | [TK]D-Fender | Or whatever |
| 05:23.26 | Lope | TK-DFender. Okay. If I delete msg0000.* on the filesystem, will asterisk realize it's been deleted or do I need to do something special? |
| 05:25.53 | Lope | Anyone tried this? http://sourceforge.net/projects/ast-php-vm/ |
| 05:28.03 | [TK]D-Fender | Virtually no-one probably |
| 05:32.38 | Lope | okay I see I can just delete 0000 and it starts from 0000 again. |
| 05:37.12 | Lope | Is there an easy to amplify voicemail messages to the maximum level before clipping? |
| 05:37.27 | Lope | (or do I need to script that myself) ? |
| 05:41.25 | Lope | Looks like I'll use sox http://www.voip-info.org/wiki/view/Asterisk+sound+files |
| 05:48.53 | Lope | Thanks for all the help guys, much appreciated. I'm on the home stretch of getting my asterisk setup done. |
| 05:51.35 | Lope | how can I specify a custom greeting for my different voicemail-boxes? I've searched for the word greeting in voicemail.conf and it only has greetingsfolder=INBOX in hte section that says "IMAP configuration settings only" I'm not using imap, the stuff is just dumping in /var/spool/asterisk/voicemail/extNUMBER/INBOX |
| 05:54.01 | Lope | okay apparently it's /var/spool/asterisk/voicemail/default/extNUMBER/INBOX/unavail.wav ? |
| 05:58.19 | Lope | currently I've got softphones defined in my sip.conf. How can I get Asterisk to connect to a SIP account that my VoIP provider has given me, so I can route the incoming calls to my softphones? |
| 05:59.18 | Lope | Seems like the type would be peer. |
| 05:59.52 | Lope | host would be the hostname of my SIP provider... |
| 06:00.01 | *** join/#asterisk giesen (~ggiesen@2001:4900:2:650::180) |
| 06:01.02 | Lope | How do you tell sip.conf how to distinguish for a listening connection (where a softphone can initiate a connection and REGISTER) vs a SIP account where my asterisk config must go and REGISTER at the VoIP provider? |
| 06:01.54 | *** join/#asterisk jjrh (~jjrh@2607:f0b0:1:6e12:94da:efb1:c314:a84b) |
| 06:03.44 | [TK]D-Fender | You create a peer to match calls your provider will send you, and to define how to auth to your provider to send them calls. |
| 06:03.52 | [TK]D-Fender | As for them knowing where to call you ... |
| 06:03.54 | [TK]D-Fender | ~sipregister |
| 06:03.54 | infobot | [~sipregister] SIP registration is to tell your provider what IP address & EXTEN to send INCOMING calls to. Some ITSPs let you use a fixed address or host rather than registering. Registration is NOT normally needed to PLACE calls, as those are typically auth'ed independently. Others accept unauth'ed calls once you are registered (saves on negotiation BW). |
| 06:03.56 | [TK]D-Fender | ^^^ |
| 06:22.34 | *** join/#asterisk robmal (r@wporzo.pl) |
| 06:25.14 | Lope | Okay. Thus far for accepting incoming numbers I've gotten the DID providers to send incoming calls to my server's IP. |
| 06:26.02 | Lope | But I thought it might be cool to maintain the flexibility of keeping my SIP account at my VoIP provider where I can either connect a softphone to it, or connect my asterisk server to it... |
| 06:26.14 | Lope | But I suppose I'm just complicating things unnecessarily. |
| 06:27.19 | Lope | Okay lastly...so far I've not learned how to make outgoing calls with my asterisk server... |
| 06:28.23 | Lope | How can I setup my asterisk server such that if I dial 0 it will use provider 0 to place the call, if I dial 1 it will use provider 1 to place the call etc. So then it will take the digits after the first digit, and dial those at the provider? |
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| 06:56.50 | [TK]D-Fender | Lope, There is no need for a prefix |
| 06:57.00 | [TK]D-Fender | Your dialplan processes whatever you tell it to. |
| 06:57.30 | [TK]D-Fender | as for the provider.. you make your peer ... maybe you register. You make dialplan to Dial() out that peer. The end |
| 06:57.34 | [TK]D-Fender | #bedtime |
| 06:57.38 | Lope | Okay, can you please show me an example of a dialplan which will make use of different providers? |
| 06:58.42 | Lope | So my new VoIP provider who have given me a SIP account... if they forward incoming numbers to my IP, how can I make outgoing calls with their service? |
| 07:03.16 | *** join/#asterisk bof22 (~Thunderbi@185.13.183.107) |
| 07:05.51 | Lope | To register my asterisk at my provider, This page says you use a line in sip.conf like http://www.voip-info.org/wiki/view/Asterisk+SIP+register |
| 07:05.55 | Lope | register => user[:secret[:authuser]]@host[:port][/extension] |
| 07:06.26 | Lope | but this page mentions pjsip.conf, which doesn't exist on my asterisk installation: https://wiki.asterisk.org/wiki/display/AST/Configuring+Outbound+Registrations |
| 07:18.20 | Lope | I'm getting forbidden, wrong password. |
| 07:19.44 | Lope | I've added register => user[:secret[:authuser]]@host[:port][/extension] to sip.conf under [general] as follows register => john:SUPERSECRET:john@sipprovider.com:5060/johnext |
| 07:19.57 | Lope | as described here: http://www.voip-info.org/wiki/view/Asterisk+SIP+register |
| 07:43.03 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 07:44.06 | Lope | This doesn't share any light on it either http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-UnderstandingVoIP-SECT-6.html#asterisk-UnderstandingVoIP-SECT-6.2 |
| 07:56.36 | *** join/#asterisk bulkorok (~btietz@89.245.151.228) |
| 07:58.24 | Lope | Maybe that outbound registration method is obsolete. I've setup pjsip.conf according to the example but it's not connecting, no errors etc. https://wiki.asterisk.org/wiki/display/AST/Configuring+Outbound+Registrations |
| 08:08.37 | Lope | how can i install res_pjsip on ubuntu? |
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| 09:30.17 | *** join/#asterisk infobot (ibot@rikers.org) |
| 09:30.17 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.6.0 (2015/10/09), 11.20.0 (2015/10/09); 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 |
| 09:39.22 | tparcina | Right now, my outgoing calls have SIP HEADER TO format: number@IP_ADD_OF_PROVIDER example: 0981234567@123.123.123.123 |
| 09:40.03 | tparcina | How to change it to this format: number@mydomain.com ? |
| 09:47.42 | wdoekes | tparcina: chan_sip? fromdomain=mydomain.com |
| 09:48.00 | wdoekes | oh wait, the TO domain |
| 09:48.14 | wdoekes | that's.. probably also possible.. with the right Dial syntax |
| 09:49.00 | wdoekes | <PROTECTED> |
| 09:49.00 | wdoekes | <PROTECTED> |
| 09:49.01 | wdoekes | ... |
| 09:49.02 | tparcina | wdoekes: Yes, I'm looking for that syntax. :D |
| 09:49.08 | wdoekes | <PROTECTED> |
| 09:49.08 | wdoekes | <PROTECTED> |
| 09:49.38 | wdoekes | clearly documented on line 29985 in chan_sip.c |
| 09:49.51 | wdoekes | I'm amazed you didn't spot that |
| 09:49.53 | wdoekes | :troll: |
| 09:50.01 | tparcina | :D |
| 09:50.12 | tparcina | wdoekes: Thank you, I'll try that. |
| 09:50.13 | *** join/#asterisk defsdoor (~andy@cpc73037-sutt4-2-0-cust62.19-1.cable.virginm.net) |
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| 10:54.29 | Lope | I'm trying to compile pjproject `--with-external-speex`. I have installed libspeex-dev with apt-get. Any tips on how to specify the path? checking if external Speex devkit is installed... aconfigure: error: Unable to use external Speex library. If Speex development files are not available in the default locations, use CFLAGS and LDFLAGS env var to set the include/lib paths |
| 10:58.16 | wdoekes | Lope: did you reconfigure after installing speex-dev? |
| 11:00.52 | Lope | hmm, I didn't reconfigure pjproject. I only reconfigured asterisk. And the func_ speexsomething wasn't available. So I figured I need to recompile pjproject. This page https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject seems to suggest that I should use --with-external-speex ... or is that unnecessary, can I just let it detect it? |
| 11:04.45 | Lope | is it possible to use a mobile phone as a FXO with bluetooth with asterisk 13? |
| 11:05.27 | Lope | func_speex is still unavailable. |
| 11:05.54 | Lope | I recompiled pjproject using the standard configure "./configure --prefix=/usr --enable-shared --disable-sound --disable-resample --disable-video --disable-opencore-amr CFLAGS='-O2 -DNDEBUG'" |
| 11:10.08 | Lope | Depends on: speex(E), speex_preprocess(E) |
| 11:10.09 | Lope | Can use: speexdsp(E) |
| 11:10.13 | Lope | What is the (E) ? |
| 11:12.37 | *** join/#asterisk Panther_Modern (~Panther_M@unaffiliated/panther-modern/x-6168176) |
| 11:15.05 | Lope | awesome! I needed to install libspeexdsp-dev and then run ./configure again. |
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| 11:50.34 | tparcina | I'm trying to modify SIP TO HEADER on outgoing phone calls. |
| 11:51.35 | tparcina | I have put the line: exten => 8380,n,Set(SIP_HEADER(To)=mydomain.hr) just before the dial command. |
| 11:52.24 | tparcina | But outgoing phone call still has the same old TO header. :( |
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| 12:02.56 | tparcina | I have tried with 8380,n,Set(SIPAddHeader(To: <sip:0989970434@mydomain.hr>) but this doesn't work neither. :( |
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| 12:09.42 | tparcina | OK, so the right syntax is: exten => 8380,n,SIPAddHeader(To: <sip:0989970434@mydomain.hr>) |
| 12:10.09 | tparcina | But, when I do that, outgoing call has 2 "To" headers. :( |
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| 12:10.53 | tparcina | First "To", that was there before, and second "To" that I have added. |
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| 12:21.12 | snadge | had some pretty serious outages lately.. isn't it interesting how a little bit of latency on the network, causes everything to fail |
| 12:21.22 | snadge | instead of just perform more slowly |
| 12:21.52 | snadge | we've got layer 2 spanning multiple locations now.. and on an ordinary day.. you'll see sub 1ms pings between the 3 sites |
| 12:22.14 | snadge | but today.. we were seeing spikes and jitter.. anywhere from 8ms.. to 40ms |
| 12:22.19 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 12:23.03 | snadge | when you've got around 500 to 800 registered peers on an asterisk instance.. this makes it spew pages of unreachable, lagged.. reachable messages.. like a mexican wave |
| 12:23.15 | snadge | flapping in and out like the door on a brothel |
| 12:24.21 | snadge | im a sysadmin.. so im looking at system load, cpu utilisation, everything looks good there.. wtf is going on.. yep everything pings etc.. no loss of connectivity |
| 12:25.03 | snadge | of course we're using a realtime db and cdrs stored in an ndb cluster etc.. so im looking at that.. custom agi scripts etc |
| 12:26.04 | snadge | then im like.. hey network guys.. those pings dont look right to me.. "are you sure? 8ms is bad?" .. "um yes.. apparently its very bad" |
| 12:27.06 | snadge | turns out some random link in the chain, which had some other customers and apparently no qos.. was topping out and causing the latency spikes.. which blew the whole network up.. good times :D |
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| 12:53.25 | tparcina | So, the question still stands, how to modify SIP header for outgoing call? (not how to add new field in header, but how to modify existing "to" field) |
| 12:54.10 | wdoekes | so... |
| 12:54.26 | wdoekes | what about the [!dnid] in the dial string then? |
| 12:55.12 | wdoekes | you decided to go ahead and ignore that, and instead make up something about SIP_HEADER(To) ? |
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| 13:20.31 | [TK]D-Fender | You don't get to modify headers like that |
| 13:20.48 | [TK]D-Fender | To: is formed from the HOST & EXTENSION dialed |
| 13:22.04 | wdoekes | yea, I pointed him to the Dial syntax this morning, for which he said thanks, and then ignored |
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| 13:41.10 | tparcina | wdoekes: I haven't ignored, just I haven't found the way how to use it. |
| 13:42.11 | tparcina | wdoekes: Then, searching for way how to use it, I found SIP_HEADER and SIPAddHeader, which seamed the right way to do things. |
| 13:43.01 | tparcina | [TK]D-Fender: wdoekes: Can you share a example code? |
| 13:43.18 | [TK]D-Fender | There is no code |
| 13:43.25 | [TK]D-Fender | I just told you what it is generatted from |
| 13:43.38 | [TK]D-Fender | you can't ADD this header, you're getting it whether you like it or not |
| 13:49.07 | tparcina | [TK]D-Fender: OK, I can't ADD the header, but can I modify it? |
| 13:50.51 | [TK]D-Fender | No,. |
| 13:50.55 | [TK]D-Fender | I told you how it is MADE |
| 13:50.59 | [TK]D-Fender | change those |
| 13:58.32 | tparcina | is searching how to change HOST from dialplan. |
| 14:00.16 | [TK]D-Fender | You don'tt. |
| 14:00.21 | [TK]D-Fender | It's what you dial when you dial it |
| 14:00.26 | [TK]D-Fender | facepalms |
| 14:01.42 | tparcina | [TK]D-Fender: Yes, but Sonetel (SIP provider) is asking that I in TO and FROM field have mydomain (instead of their IP address). |
| 14:02.00 | tparcina | [TK]D-Fender: It didn't make much sense to me. |
| 14:02.15 | [TK]D-Fender | [08:20][TK]D-FenderTo: is formed from the HOST & EXTENSION dialed |
| 14:02.25 | [TK]D-Fender | wants his last 40 minutes back .... |
| 14:02.49 | tparcina | [TK]D-Fender: And thank you for saving me time. :) |
| 14:06.55 | Gugge | tparcina: how about Dial(SIP/1234@mydomain/remoteip) then? |
| 14:07.25 | tparcina | Gugge: I haven't tried that one yet. |
| 14:07.37 | Gugge | or ask the provider how your asterisk dial command should be :P |
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| 14:34.46 | wdoekes | tparcina: Dial(SIP/mydevice/number!number@somedomain.com) ? |
| 14:36.30 | tparcina | wdoekes: What would be number!number in this case? |
| 14:38.43 | *** join/#asterisk pathcl (~pathcl@159.203.136.187) |
| 14:38.46 | pathcl | hi there! |
| 14:38.50 | Gugge | mydevice/number would be the part you want to dial number@somedomain.com would be what you want in To: |
| 14:38.54 | pathcl | Im playing around phpagi |
| 14:39.28 | pathcl | though for some reason after dtfm the call is hanged up |
| 14:39.30 | pathcl | :( |
| 14:39.59 | pathcl | http://pastebin.com/vVn7niU0 |
| 14:40.20 | pathcl | if I remove sendDTMF call is ok |
| 14:40.49 | pathcl | I need to have stream_file and sendDTMF at the same time |
| 14:40.59 | pathcl | is that possible? |
| 14:44.05 | [TK]D-Fender | You should actually look at the call to see why it hung up |
| 14:44.26 | [TK]D-Fender | And SendDTMF is not done "in the background" |
| 14:45.05 | pathcl | how can I do it in the background ? |
| 14:45.15 | pathcl | I'll try sip debug to catch why |
| 14:46.49 | *** join/#asterisk hexanol (~bibi@modemcable094.94-70-69.static.videotron.ca) |
| 14:47.19 | [TK]D-Fender | You can't in any normal way |
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| 14:48.21 | tparcina | wdoekes: Gugge: I have tried that. But, just as [TK]D-Fender has said, TO field is composed from HOST and EXTENSION dialed. |
| 14:48.23 | pathcl | for some unknown reason the stream_file is not played completely |
| 14:48.35 | pathcl | thats my theory |
| 14:49.12 | pathcl | uhm |
| 14:49.18 | pathcl | I could try exec playback |
| 14:50.19 | wdoekes | tparcina: what *did* it do when you added "!<to-header>" ? |
| 14:50.28 | wdoekes | did it not dial? did it ignore the value? |
| 14:50.37 | wdoekes | did it put in only the number? only the domain? |
| 14:50.40 | wdoekes | details please |
| 14:51.46 | tparcina | wdoekes: It has dialed and the SIP message head, in header, 2 TO fields. |
| 14:52.01 | wdoekes | works fine over here |
| 14:52.01 | wdoekes | *CLI> channel originate SIP/12345@1.2.3.4!number@mydomain.com application Wait |
| 14:52.09 | wdoekes | INVITE sip:12345@1.2.3.4 SIP/2.0 |
| 14:52.10 | tparcina | wdoekes: First one, the old (problematic one), and second. |
| 14:52.13 | wdoekes | To: <sip:number@mydomain.com> |
| 14:54.55 | tparcina | wdoekes: Thank you! :D |
| 14:56.25 | tparcina | wdoekes: In your favorite cafe bar, can you pay for drink over the Internet? :) |
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| 15:21.33 | saint_ | morning |
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| 16:04.16 | WIMPy | http://www.mail-archive.com/misc@openbsd.org/msg144351.html |
| 16:08.54 | wdoekes | tnx WIMPy |
| 16:09.48 | *** join/#asterisk keithf (~keithf@47.21.29.194) |
| 16:11.02 | WIMPy | Someone who seems to know more said that on a thread scale from 0-10 that's an 11. |
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| 16:27.12 | dan_j | Is there a way in the CLI to check that voicemail IMAP is functioning? |
| 16:27.24 | dan_j | or even included in the current build? |
| 16:31.07 | [TK]D-Fender | Use it? |
| 16:36.27 | dan_j | Doesnt seem to work. And i dont see any reference to it when starting asterisk. Just double checked by menuselect and it is on imap storage. |
| 16:37.42 | dan_j | Had it working previously but been working on some source code and then had problems compiling with the imap library. |
| 16:37.47 | keithf | guys im new to asterisk and pbx's i have been dealing with freepbx and i have learned the gui fairly well but i want to get deeper into asterisk.. is there anything you guys recommend for learning understand and getting an overall better understanding ? |
| 16:38.03 | dan_j | Managed to get it to work with the help of wdoekes but now having this problem |
| 16:38.26 | WIMPy | ~book |
| 16:38.26 | 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 |
| 16:38.33 | WIMPy | keithf: That ^ |
| 16:39.58 | keithf | other than the asterisk fast track training class this you say is good? |
| 16:40.17 | keithf | im trying to get to the training in dubai in april but im trying to do everything i can to get btter versed |
| 16:40.21 | dan_j | It's the official book. You can view it online. |
| 16:40.33 | dan_j | http://www.asteriskdocs.org/ |
| 16:40.35 | keithf | sweet |
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| 16:42.33 | dan_j | keithf: There is also information at https://wiki.asterisk.org/wiki/display/AST/Home |
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| 16:48.44 | keithf | thanks |
| 16:49.04 | keithf | im really trying to improve my skills as a pbx admin and asterisk is big part of it so i have to get better |
| 16:49.13 | keithf | will be doing a lot of sitting and watching and reading |
| 16:49.23 | dan_j | If you are using freepbx, i would not mess with any asterisk config files. |
| 16:50.21 | keithf | yeah i know that |
| 16:50.26 | keithf | but i still want a better understanding |
| 16:50.35 | keithf | of what all the modules are doing and not doing |
| 16:51.53 | [TK]D-Fender | FreePBX has a fair number of custom files included that you can do a lot with. First understand how * works, then understand how FreePBX works. Live in the middle-ground |
| 16:58.20 | keithf | ive gone thru the training from sangoma at least on the freepbx software |
| 16:58.34 | keithf | its still a lot and they have a ton of custom modules |
| 17:08.52 | *** join/#asterisk psiforce (7a6bd2ec@gateway/web/freenode/ip.122.107.210.236) |
| 17:10.01 | psiforce | what is the best way of adding a new context to the dialplan WITHOUT reloading the dialplan? Can it be done using asterisk realtime? |
| 17:11.34 | [TK]D-Fender | No, realtime still uses "switch" in the static config |
| 17:12.07 | psiforce | :( |
| 17:13.53 | [TK]D-Fender | Doesn't mean you can't automate it |
| 17:14.01 | [TK]D-Fender | but it's still gotta be there |
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| 17:15.20 | psiforce | ok so the only way to create a context is to run "asterisk -rx 'dialplan reload'" |
| 17:16.57 | psiforce | I was really surprised that these wasn't an action in the ami for this |
| 17:17.24 | [TK]D-Fender | there is... |
| 17:17.40 | [TK]D-Fender | read the AMI command list |
| 17:23.43 | psiforce | [TK]D-Fender: Do you mean there is a way of reloading via the ami or a way of adding a context via the ami? |
| 17:24.17 | psiforce | I don't see a way of adding a new context from the ami (using asterisk 1.8) |
| 17:25.46 | [TK]D-Fender | you can do botth actually |
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| 17:29.14 | psiforce | [TK]D-Fender: I can see how to reload easy enough but from the list of commands (http://pastebin.com/3c2iAH56) I don't see how to add a context via the ami |
| 17:29.47 | [TK]D-Fender | Starte at it a bit longer.... |
| 17:29.51 | [TK]D-Fender | stare* |
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| 17:30.42 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 17:35.51 | psiforce | [TK]D-Fender: I'm losted.... I can see how to add an exten but not a new context |
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| 17:39.58 | [TK]D-Fender | <PROTECTED> |
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| 17:42.47 | psiforce | [TK]D-Fender: but that just includes a context within another context, doesn't it. its not going to create a new context |
| 17:43.23 | [TK]D-Fender | may have stretched that to allowing switch |
| 17:44.14 | [TK]D-Fender | indeed this does require the config to me modded. |
| 17:44.36 | [TK]D-Fender | there was a way to run chell comamnd from a lter ver of CLI, just not sure which, or the syntax offhand |
| 17:46.36 | psiforce | but how does running a shell command help? even if you run "echo [new-context] >> /etc/asterisk/extensions.conf", you still end up having to run "asterisk -rx 'dialplan reload'" |
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| 17:48.04 | psiforce | and "dialplan add include existing-context into new-context", doesn't work as "new-context" has to exist |
| 17:49.59 | [TK]D-Fender | no point to dialplan add" if you're adding to the text file |
| 17:50.10 | [TK]D-Fender | and you can use the CLI dialplan relaon, withoutt using the shell |
| 17:50.20 | [TK]D-Fender | reload* |
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| 17:54.11 | psiforce | [TK]D-Fender: ok so you can't create new contexts without reloading the dialplan |
| 17:54.43 | [TK]D-Fender | psiforce: Not that I can see. Was mistaken on that. Reloading it though, yes you can |
| 17:55.29 | psiforce | ok, well guess we will just have to bite the bullet then :/ |
| 17:56.31 | psiforce | its just that dialplan reload hangs the console for about 7 seconds |
| 17:56.44 | psiforce | hence wanting to avoid doing that |
| 17:58.44 | psiforce | thanks for your input anyway [TK]D-Fender |
| 17:58.58 | [TK]D-Fender | You're welcome |
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| 19:17.14 | Docfxit_ | Penguin: May I PM you? |
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| 19:54.29 | dan_j | What does Asterisk do with Imap if the mailbox has exceeded it's quota? |
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| 20:36.49 | lordvadr | If I set FAXOPT(gateway)=yes,10 in my dialplan or in sip.conf, it causes "Codec mismatch" warnings and transcoding between slin and g.711 on my ISDN circuits. Removing the FAXOPT fixes this. Is this normal behavior due to the gateway having to listen to the stream? Is there a way to suppress the warnings? Version is 10.12.4 (the last version I can get out of the repo for an old el5 box). |