IRC log for #asterisk on 20170108

`00:11.24`	`*** join/#asterisk Y04NN (~y04nn@2a01:e34:ef37:5870:bde2:3f1e:9171:a07c)`
`00:59.13`	`*** join/#asterisk _corrupt (~chalumnin@199.241.146.163)`
`01:23.10`	`*** join/#asterisk marlinc (~marlinc@185.167.204.13)`
`01:53.12`	`*** join/#asterisk marlinc (~marlinc@1.0.0.127.13.204.167.185.in-addr.arpa)`
`02:12.18`	`*** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net)`
`02:20.23`	`*** join/#asterisk friedrich (~friedrich@aextron.de)`
`02:58.43`	`*** join/#asterisk Rini (uid196547@gateway/web/irccloud.com/x-uavcvnvosvkbeyep)`
`03:36.55`	`*** join/#asterisk friedrich (~friedrich@aextron.de)`
`03:50.42`	`*** join/#asterisk robink (~quassel@unaffilated/robink)`
`03:56.27`	`*** join/#asterisk fstd_ (~fstd@unaffiliated/fisted)`
`04:28.56`	`*** join/#asterisk mophed (~mophed@104-0-251-34.lightspeed.jcvlfl.sbcglobal.net)`
`04:29.01`	`mophed`	`greetings`
`04:29.38`	`mophed`	`could someon perhaps lend me a hand with configuring my server?`
`04:30.45`	`mophed`	`im trying to configure it to work with flowroute. using the x-lite softphone`
`04:34.22`	`mophed`	`i can connect to other users with the softphone as long as the "host=dynamic" in sip.conf the user i have to connect to has "host=sip.flowroute.com" and if i change it to dynamic i can connect although not make any calls`
`04:41.28`	`*** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com)`
`04:54.02`	`[TK]D-Fender`	`What "user"?`
`04:54.46`	`[TK]D-Fender`	`You need peers for your phone, you need a peer for your provider.`
`04:55.00`	`[TK]D-Fender`	`Phones are normally dynami, and your provider is fixed`
`04:55.21`	`[TK]D-Fender`	`Show us an actual failed call so we can see what you're actually doing.`
`04:55.57`	`mophed`	`[201]`
`04:56.00`	`mophed`	`type=friend`
`04:56.26`	`mophed`	`so im reffering to 201 as being a "user" although i guess it would be an extension right?`
`04:56.40`	`Samot`	`It would be the device.`
`04:56.51`	`Samot`	`You can make any digit pattern route to that device.`
`04:57.03`	`mophed`	`how can i show you a failed call? i was strying to connect to the server with the xlite phone so that i could place the call`
`04:57.32`	`Samot`	`You're only making one peer.`
`04:57.34`	`Samot`	`For the device.`
`04:57.39`	`Samot`	`Like TK said, you need two peers.`
`04:57.41`	`[TK]D-Fender`	`201 is a phone.. it should be dynamic`
`04:57.48`	`Samot`	`One for the device and one for the provider trunk.`
`04:57.54`	`[TK]D-Fender`	`You should have another peer for your flowroute account`
`04:58.22`	`mophed`	`can i paste in here or do you guys use a seprerate site?`
`04:58.36`	`Samot`	`~pb`
`04:58.37`	`infobot`	`extra, extra, read all about 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.`
`04:58.50`	`mophed`	`thank you`
`05:00.59`	`mophed`	`http://pastebin.ca/3754943`
`05:01.17`	`mophed`	`that is what i have for sip and extensions`
`05:02.14`	`Samot`	`That's just the Flowroute peer.`
`05:02.18`	`Samot`	`You need one for your device.`
`05:02.19`	`[TK]D-Fender`	`Ok, so far that doesn't show 201`
`05:02.28`	`[TK]D-Fender`	`Do you have it?`
`05:02.35`	`[TK]D-Fender`	`Show us the call attempt that fails`
`05:02.50`	`[TK]D-Fender`	`"core set verbose 10", "sip set debutg on"`
`05:05.04`	`mophed`	`the 201 wont do anything at all. but i can connect to it with xlite as long as the host=dynamic`
`05:05.23`	`[TK]D-Fender`	`What do yuo mean won't do anything?`
`05:05.36`	`[TK]D-Fender`	`The point is to use X-Lite to call OUT flowroute, no?`
`05:05.59`	`mophed`	`yessir.`
`05:06.11`	`Samot`	`You don't have any proper context's in your dialplan.`
`05:06.13`	`[TK]D-Fender`	`So it SHOULD do something`
`05:06.18`	`[TK]D-Fender`	`Show us some kind of failure`
`05:06.21`	`Samot`	`The peers need to point to a context to handle the call.`
`05:06.27`	`[TK]D-Fender`	`It's still not clear quite what the problem is`
`05:06.35`	`Samot`	`No, a debug would help.`
`05:06.40`	`mophed`	`http://pastebin.ca/3754945`
`05:07.05`	`[TK]D-Fender`	`perhaps over basic, but show us what happens`
`05:07.24`	`mophed`	`thats all that 201 has. and if i connect or login with that on xlite. i get the green light but if i dial a number its just a busy signal`
`05:07.33`	`[TK]D-Fender`	`Show the call`
`05:07.35`	`Samot`	`Show us THAT call.`
`05:07.44`	`[TK]D-Fender`	`and the following to extensions.conf:`
`05:07.46`	`[TK]D-Fender`	`^`
`05:07.56`	`[TK]D-Fender`	`you didn't show us the context header`
`05:08.05`	`[TK]D-Fender`	`so we have no proof where those lines are under`
`05:09.10`	`mophed`	`http://pastebin.ca/3754946 is my extensions.conf`
`05:09.29`	`mophed`	`thats the whole thing`
`05:09.46`	`[TK]D-Fender`	`ok, let's see the call`
`05:11.03`	`mophed`	`"originate SIP/flowroute/test number here application Playback hello-world"`
`05:11.08`	`mophed`	`like this?`
`05:11.13`	`[TK]D-Fender`	`No`
`05:11.26`	`[TK]D-Fender`	`I though we were dealing with a X-Lite issue with 201`
`05:11.39`	`[TK]D-Fender`	`You need to be a LOT clearer about the actual failure`
`05:12.05`	`Samot`	`asterisk -r`
`05:12.09`	`Samot`	`core set verbose 10`
`05:12.11`	`Samot`	`sip set debug on`
`05:12.20`	`mophed`	`no the xlite issue is that i cannot get it to log in with the "flowroute" device. i just get a 408 error`
`05:12.20`	`Samot`	`Those commands need to be run.`
`05:12.23`	`Samot`	`Then you make a call.`
`05:12.31`	`Samot`	`So we can see what is happening.`
`05:12.39`	`[TK]D-Fender`	`<mophed> no the xlite issue is that i cannot get it to log in with the "flowroute" device. i just get a 408 error <- you don't`
`05:12.47`	`[TK]D-Fender`	`X-line has nothing to do with flowroute`
`05:12.53`	`[TK]D-Fender`	`It talks to *`
`05:12.57`	`[TK]D-Fender`	`* talks to flowroute.`
`05:13.01`	`[TK]D-Fender`	`they know NOTHNG about each other`
`05:13.18`	`[TK]D-Fender`	`when x-lite dials, * matches that call, sends it into the dialplan and follows those steps`
`05:13.30`	`[TK]D-Fender`	`If it leads to dialing out, that is coincidental`
`05:13.41`	`[TK]D-Fender`	`X-lite inherently knows NOTHING about flowroute`
`05:13.48`	`[TK]D-Fender`	`X-lite = dynamis on its peer`
`05:13.57`	`[TK]D-Fender`	`flowroute = FIXED under its peer`
`05:16.18`	`mophed`	`http://tinypic.com/r/358uybo/9`
`05:16.26`	`mophed`	`http://tinypic.com/r/2zdnm7k/9`
`05:17.11`	`mophed`	`i understand that.`
`05:17.30`	`Samot`	`That's completely wrong.`
`05:17.35`	`Samot`	`X-Lite should have the details for 201`
`05:17.37`	`[TK]D-Fender`	`No`
`05:17.40`	`mophed`	`flowroute is the name of the device that is in sip.conf`
`05:17.45`	`Samot`	`No.`
`05:17.51`	`[TK]D-Fender`	`you are putting flowroute INTO your X-lite`
`05:17.51`	`Samot`	`Flowroute = not device.`
`05:18.00`	`[TK]D-Fender`	`X-lite is NOT to be configured with flowroute!`
`05:18.08`	`[TK]D-Fender`	`X-lite = 201`
`05:18.16`	`[TK]D-Fender`	`it talks to ASTERISK, not flowroute`
`05:18.30`	`[TK]D-Fender`	`clear?`
`05:18.36`	`[TK]D-Fender`	`put 201 in ther`
`05:18.55`	`Samot`	`With the password for 201`
`05:18.56`	`mophed`	`okay. ill do taht and i can see 201 connected in *`
`05:19.11`	`*** join/#asterisk Nugget (~nugget@mezger.macnugget.org)`
`05:19.37`	`[TK]D-Fender`	`X-lite is 201 and is a SEPARATE thing. It knows NOTHING about flowroute.`
`05:19.37`	`Samot`	`Once X-Lite is registered with Asterisk, make the call.`
`05:20.14`	`[TK]D-Fender`	`your extensions.conf may take a call from X-Lite and CHOSE to send it out using dial, but X-Lite has no idea of what * is actually doing`
`05:22.46`	`*** join/#asterisk matrix1233 (~matrix123@197.3.231.61)`
`05:28.37`	`mophed`	`ok i made the call. let me copy the debug`
`05:34.51`	`mophed`	`http://pastebin.ca/3754948`
`05:35.12`	`mophed`	`i dont know if thats enough information`
`05:36.35`	`[TK]D-Fender`	`it isn't`
`05:36.38`	`[TK]D-Fender`	`get the call from the start`
`05:43.44`	`mophed`	`http://pastebin.ca/3754949`
`05:44.19`	`[TK]D-Fender`	`that isn't from the start`
`05:44.25`	`[TK]D-Fender`	`we don't see the call arrive from X-lite`
`05:44.33`	`[TK]D-Fender`	`it's too far back`
`05:44.39`	`[TK]D-Fender`	`look for the INVITE <-`
`05:46.27`	`mophed`	`not like the invite on line 2?`
`05:48.09`	`[TK]D-Fender`	`no, that is the OUTBOUND leg of a call`
`05:48.15`	`[TK]D-Fender`	`I don't see the call FROM X-Lite`
`05:48.27`	`[TK]D-Fender`	`Contact: <sip:442036427900@192.168.1.227:5060>`
`05:48.46`	`[TK]D-Fender`	`However I can see you didn't configure the rest of your system to work from behind NAT properly`
`05:49.01`	`[TK]D-Fender`	`localnet=192.168.1.0/255.255.255.0`
`05:49.03`	`[TK]D-Fender`	`nat=yes`
`05:49.16`	`[TK]D-Fender`	`externaddr=YOURWANIPORFQDN`
`05:49.20`	`[TK]D-Fender`	`directmedia=no`
`05:49.28`	`[TK]D-Fender`	`add those under [general] in sip.conf`
`05:49.40`	`mophed`	`http://pastebin.ca/3754950`
`05:49.56`	`[TK]D-Fender`	`You need to have 5060 & your rtp.conf range forwarded to your server as well`
`05:49.56`	`mophed`	`thats from the very start`
`05:50.41`	`[TK]D-Fender`	`make the changes I just told you to do and retest`
`05:50.42`	`[TK]D-Fender`	`"core set verbose 10" <-`
`05:50.46`	`[TK]D-Fender`	`And set verbose`
`05:57.59`	`mophed`	`http://pastebin.ca/3754951`
`05:58.12`	`mophed`	`still just busy signal`
`05:58.50`	`[TK]D-Fender`	`Don't mask anything`
`05:58.54`	`[TK]D-Fender`	`and confirm your forwarding`
`05:59.04`	`[TK]D-Fender`	`and set the verbose like I told you`
`05:59.49`	`mophed`	`i set core set verbose 10`
`06:00.07`	`[TK]D-Fender`	`nope`
`06:00.20`	`[TK]D-Fender`	`otherwise we'd see the dialplan execution`
`06:00.42`	`[TK]D-Fender`	`Also set "nat=no" for flowroute`
`06:02.22`	`mophed`	`k got that to "no" i have not done any port fwd in the router`
`06:02.36`	`mophed`	`i can do that as well if needed`
`06:04.12`	`[TK]D-Fender`	`it is`
`06:06.14`	`mophed`	`just 5060?`
`06:06.36`	`[TK]D-Fender`	`<[TK]D-Fender> You need to have 5060 & your rtp.conf range forwarded to your server as well`
`06:06.37`	`[TK]D-Fender`	`^`
`06:12.43`	`mophed`	`i think its ports 10000-20000 in rtp do i really need that many? im only going to ever be making 1 call at a time`
`06:14.52`	`[TK]D-Fender`	`feel free to reduce that to 4/call`
`06:15.06`	`[TK]D-Fender`	`ar at least a 100 would be nice`
`06:15.09`	`[TK]D-Fender`	`just in case`
`06:15.17`	`mophed`	`100 it is`
`06:15.28`	`mophed`	`10000-10100`
`06:21.13`	`mophed`	`http://pastebin.ca/3754964`
`06:21.16`	`mophed`	`is that right>`
`06:21.40`	`[TK]D-Fender`	`yes`
`06:21.51`	`[TK]D-Fender`	`make sure to restart * itslef completely`
`06:21.58`	`[TK]D-Fender`	`core restart now`
`06:24.54`	`mophed`	`check`
`06:33.36`	`*** join/#asterisk slav3_sergal (~frankthet@unaffiliated/slav3-kitten/x-0866809)`
`06:36.24`	`mophed`	`i must have messed something up. now i cant get xlite to connect again :(`
`06:36.25`	`mophed`	`http://tinypic.com/r/v8euee/9`
`06:36.30`	`mophed`	`does that look right?`
`06:37.13`	`[TK]D-Fender`	`Assuming that is a starting port for the range`
`06:37.45`	`mophed`	`yeah. the global port range is 10000-10100`
`06:37.56`	`[TK]D-Fender`	`What router is that?`
`06:38.40`	`mophed`	`its a motorola NVG510 from ATT`
`06:39.47`	`[TK]D-Fender`	`ok.`
`06:39.53`	`[TK]D-Fender`	`Show the next call attempt`
`06:46.51`	`mophed`	`sec. i think i need to reastart the router. apperently att blocks port 5060 and now i cannot get back into the custom services tab.`
`06:47.30`	`*** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com)`
`06:52.00`	`*** join/#asterisk kambei (~kambei@unaffiliated/kambei)`
`07:08.46`	`*** join/#asterisk mophed (~mophed@104-0-251-34.lightspeed.jcvlfl.sbcglobal.net)`
`07:14.57`	`*** join/#asterisk miralin (~Thunderbi@194.8.128.48)`
`07:34.51`	`*** join/#asterisk Jack17 (~jackkk@212.34.23.49)`
`07:35.53`	`Jack17`	`Morning fellas , how can i play a music for quick answered calls , like if call is answered within 10 seconds play a music if not continue the call normally`
`07:46.58`	`[TK]D-Fender`	`elaborate on that`
`07:47.47`	`*** join/#asterisk MaliutaLap (nikolai@unaffiliated/maliuta)`
`07:48.51`	`*** join/#asterisk J0hnSteel (~J0hnSteel@92.55.116.125)`
`07:49.13`	`*** join/#asterisk Maliuta (~nobusines@unaffiliated/maliuta)`
`07:55.12`	`Jack17`	`when my client send call using my asterisk , if the call is answered within 5sec i want to play a greeting message else continue the call if the call took more than 5 sec to be answered`
`08:03.35`	`WIMPy`	`I would understand it the other way round.`
`08:03.54`	`Jack17`	`WIMPy :D`
`08:04.26`	`Jack17`	`some calls are answred by voicemail`
`08:04.32`	`Jack17`	`we would like to leave a message`
`08:04.46`	`Jack17`	`instead of wasting our call center to talk`
`08:11.57`	`[TK]D-Fender`	`Speed of answer != voicemail`
`08:12.11`	`[TK]D-Fender`	`That is a shitty guage`
`08:12.21`	`[TK]D-Fender`	`"core show application amd" <----------------`
`08:12.52`	`[TK]D-Fender`	`"core show application dial" <----------------`
`08:13.07`	`[TK]D-Fender`	`use a macro or gosub to run AMD and determine what to do.`
`08:13.10`	`[TK]D-Fender`	`heads to be`
`08:13.13`	`[TK]D-Fender`	`bed`
`08:13.44`	`*** join/#asterisk tzafrir (~tzafrir@local.xorcom.com)`
`08:41.22`	`*** join/#asterisk sekil (~sekil@cable-89-216-195-139.dynamic.sbb.rs)`
`08:42.31`	`*** join/#asterisk sekil (~sekil@cable-89-216-195-139.dynamic.sbb.rs)`
`08:54.37`	`*** join/#asterisk miralin1 (~Thunderbi@194.8.128.48)`
`08:59.35`	`Jack17`	`threshold in amd`
`08:59.41`	`Jack17`	`if i want to only detect words not air :D`
`08:59.42`	`Jack17`	`how`
`08:59.50`	`Jack17`	`should it be 256 or 40`
`09:08.54`	`Jack17`	`when we do originate calls dont hangup when we hangup`
`09:08.55`	`Jack17`	`why`
`09:12.10`	`*** join/#asterisk defsdoor (~andy@cpc35-sutt4-2-0-cust184.19-1.cable.virginm.net)`
`09:58.02`	`*** join/#asterisk fahmad (~fahmad@unaffiliated/fahmad)`
`09:58.04`	`fahmad`	`hello`
`09:59.14`	`fahmad`	`I have a corrupted ASTDB and there are many extensions i have found the way but it will take too much time is there any other way to fix the problem ?`
`09:59.22`	`fahmad`	`to rebuild the astdb`
`10:02.31`	`WIMPy`	`Does it contain anything anything important? If not, just delete it.`
`10:04.32`	`fahmad`	`then what about my extensiions brother ?`
`10:05.53`	`WIMPy`	`They will register again in a few minutes.`
`10:06.09`	`WIMPy`	`Probably bertter than not running at all, isn't it?`
`10:06.23`	`fahmad`	`Ok`
`10:07.29`	`WIMPy`	`And don't call peers extensions.`
`10:25.39`	`Jack17`	`WIMPy`
`10:25.45`	`Jack17`	`when originating`
`10:25.50`	`Jack17`	`user hangup but call still ringing`
`10:35.58`	`WIMPy`	`That description is far too vague to comment.`
`10:41.21`	`*** join/#asterisk miralin (~Thunderbi@194.8.128.48)`
`10:45.36`	`*** join/#asterisk Tiffon (~name@unaffiliated/tiff0n)`
`10:45.59`	`Jack17`	`WIMPy`
`10:46.02`	`Jack17`	`?`
`10:46.56`	`dan_j`	`Jack17: You need to explain with more detail. Include the CLI output and dialplan. Use pastebin for that`
`10:48.38`	`Jack17`	`~pb`
`10:48.38`	`infobot`	`from memory, 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.`
`10:49.17`	`Jack17`	`https://gist.github.com/anonymous/21634425aec3cadd585d9d1415c57dfa`
`11:23.20`	`Jack17`	`dan_j`
`11:23.21`	`Jack17`	`?>`
`11:23.48`	`dan_j`	`That was not the whole output`
`11:24.00`	`dan_j`	`and you didnt post your dialplan`
`11:24.12`	`dan_j`	`So I cannot comment`
`11:39.17`	`Jack17`	`[2017-01-08 12:39:00] WARNING[12004][C-00000012]: features.c:8216 bridge_exec: Bridge failed because channel SIP/proxy1-0000001a does not exist`
`11:39.17`	`Jack17`	`<PROTECTED>`
`12:20.16`	`*** join/#asterisk wyoung (~wyoung@wesleyy.com)`
`12:21.11`	`*** part/#asterisk wyoung (~wyoung@wesleyy.com)`
`12:26.29`	`*** join/#asterisk Y04NN (~y04nn@2a01:e34:ef37:5870:343e:eac4:a37d:ac44)`
`12:31.32`	`Samot`	`Jack17 = follow-me`
`12:45.52`	`*** join/#asterisk miralin (~Thunderbi@194.8.128.48)`
`13:09.37`	`*** join/#asterisk Y04NN (~y04nn@2a01:e34:ef37:5870:1de7:3337:730b:a1ce)`
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`13:24.42`	`*** join/#asterisk matrix1233 (~matrix123@197.3.231.61)`
`13:40.43`	`dan_j`	`Where did chan_local move to in asterisk 13?`
`13:41.18`	`dan_j`	`In v11, i have to pre-load it, but i cant find the module in v13. I can see that it was removed but cant see where to`
`13:47.20`	`*** join/#asterisk Y04NN (~y04nn@nayon.fr)`
`14:05.34`	`*** join/#asterisk Rasputin3711 (5d511371@gateway/web/freenode/ip.93.81.19.113)`
`14:05.41`	`Rasputin3711`	`Hello`
`14:06.17`	`Rasputin3711`	`<PROTECTED>`
`14:07.02`	`Rasputin3711`	`I have got many warnings in logs in a local network, i don't have NAT. How can i troubleshoot that?`
`14:07.41`	`Rasputin3711`	`Between clients only switches`
`14:20.07`	`*** join/#asterisk miralin1 (~Thunderbi@194.8.128.48)`
`14:53.18`	`tzafrir`	`dan_j, main/core_local.c (source), /usr/sbin/asterisk (runtime)`
`15:01.02`	`*** join/#asterisk fstd (~fstd@unaffiliated/fisted)`
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`15:08.34`	`dan_j`	`tzafrir: thanks`
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`20:23.59`	`surye`	`So moving to a new server (from sip to pjsip), and I have a weird issue where the phones are getting registered as Contacts with the wrong port (ie: 514/sip:514@10.193.10.169:50769;user=phone), is there any way to make sure it's using 5060 (looks like maybe it's using the source port of the register packet?`
`20:24.04`	`*** join/#asterisk [TK]D-Fender (~joe@64.235.216.2)`
`20:24.58`	`surye`	`The phones are crappy cisco 7940s with SIP firmware, but I'd imagine they should still work..`
`20:25.09`	`surye`	`or at least did under chan_sip`
`20:30.33`	`*** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com)`
`20:30.54`	`surye`	`Ah, Rewrite Contact seems to be the key here.`
`20:31.08`	`surye`	`Or rather, the problem`
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`22:03.25`	`*** join/#asterisk infobot (ibot@rikers.org)`
`22:03.25`	`*** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.13.1 (2016/12/08), 11.25.1 (2016/12/08), Standard: 14.2.1 (2016/12/08); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu`
`22:05.43`	`*** join/#asterisk simplydrew_ (~simplydre@unaffiliated/simplydrew)`
`22:07.27`	`*** join/#asterisk tzafrir (~tzafrir@bzq-82-81-175-197.red.bezeqint.net)`
`22:10.55`	`*** join/#asterisk moreder (~moreder@88-222-225-208.meganet.lt)`
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`22:17.59`	`moreder`	`i'm stuck with external alarm passtrough. i'm using inband dtmf mode on both call sides. tones are ok on incoming side but distorted on outgoing part. using ver 1.8.26. seems like asterisk is doing something with inband dtmf's but i cant figure out what exactly and how to disable it`
`22:19.18`	`igcewieling`	`moreder: what codec?`
`22:19.29`	`*** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com)`
`22:19.37`	`moreder`	`alaw`
`22:20.38`	`*** join/#asterisk mophed (~mophed@104-0-251-34.lightspeed.jcvlfl.sbcglobal.net)`
`22:23.51`	`mophed`	`greetings`
`22:25.14`	`mophed`	`today is the 4th solid day that i am spending trying to get asterisk to work with flowroute. with 0 succsess. i will admit that i know nothing about the phone system or SIP, Trunks, DID and the sort.`
`22:25.29`	`[TK]D-Fender`	`And I left asking you for an updated call & configs`
`22:25.33`	`[TK]D-Fender`	`and you disappeared`
`22:25.53`	`[TK]D-Fender`	`After finally (supposedly) getting you to NOT put flowroute info into your soft-phone`
`22:25.55`	`*** join/#asterisk simplydrew_ (~simplydre@unaffiliated/simplydrew)`
`22:26.36`	`mophed`	`im sorry, i caught you when you said you were off to bed. i was able to finally get my ports forwarded, which wasnt working becuase the router was giving me an ipv6 address. so i couldnt see my ip in the forward list`
`22:26.59`	`[TK]D-Fender`	`pretty big fail there...`
`22:27.05`	`[TK]D-Fender`	`ok, let's see where we are now`
`22:28.57`	`mophed`	`im back to square one. after i put all the changes you suggested into the conf files xlite was no longer able to connect to asterisk and asterisk wasnt showing up on the flowroute site anymore.`
`22:29.38`	`[TK]D-Fender`	`PASTEBIN <-`
`22:29.42`	`mophed`	`so i backed up all the changes we made. and restored the original files just to see if xlite and asterisk would again talk to eachother and show up on the flowroute site`
`22:30.14`	`[TK]D-Fender`	`If you undid everything I told you to do then my time has been wasted.`
`22:30.39`	`[TK]D-Fender`	`Show the configs & debug`
`22:31.00`	`mophed`	`no no. i backed it all up`
`22:32.02`	`*** join/#asterisk Y04NN (~y04nn@2a01:e34:ef37:5870:5d50:e36a:32c7:87ac)`
`22:32.33`	`mophed`	`http://tinypic.com/r/wgzqjs/9`
`22:32.49`	`mophed`	`this is what i mean when i say showing up on the flowroute site. i dont know if its important of not`
`22:33.17`	`[TK]D-Fender`	`Seems to say it's registered`
`22:33.22`	`[TK]D-Fender`	`but I asked for proof from your side`
`22:33.47`	`*** join/#asterisk friedrich (~friedrich@aextron.de)`
`22:34.10`	`mophed`	`i understand. when i load the conf files we did yesterday its not registered anymore and xlite wont connect to the * server`
`22:36.16`	`mophed`	`im just wondering if i am missing something somewhere. i am running this on a debian distro and installed * with the package manager by simply doing apt-get install asterisk i do have all the dependencies`
`22:36.47`	`mophed`	`should uninstall and and reinstall from source`
`22:37.55`	`mophed`	`i also just reloaded last nights configs and i can get the core to set verbose to 10 but sip set debug on is not working anymore`
`22:38.05`	`mophed`	`CLI> sip set debug on`
`22:38.06`	`mophed`	`No such command 'sip set debug on' (type 'core show help sip set' for other possible commands)`
`22:38.40`	`*** join/#asterisk ketas (~ketas@200-62-46-176.dyn.estpak.ee)`
`22:43.13`	`*** join/#asterisk friedrich (~friedrich@aextron.de)`
`22:43.46`	`mophed`	`with the verbose set to 10 i cannot see anything connecting to * and the xlite phone gives me a 408 error and just keeps retrying`
`22:44.43`	`mophed`	`and the active SIP registration is no longer seen on the flowroute site.`
`22:47.26`	`[TK]D-Fender`	`You clearly don't have chan_sip loaded`
`22:47.47`	`mophed`	`i really just want to pull my hair out. this should not be this difficult. i just want to make a simple phone call. its not like im setting up an entire call center. could you maybe recommend a good begginers guide i could read or follow on this very basic setup?`
`22:48.18`	`[TK]D-Fender`	`So either the package is messed up, or your configs (starting wiht modules.conf not autoloading or explicitly loading), or the content of the sip config being invlaid, or in conflict and unable to bind`
`22:48.46`	`[TK]D-Fender`	`You made the mistake of trashing all the evidence instead of continuing to work with what was already so close`
`22:48.57`	`[TK]D-Fender`	`And introduced more potential for failure`
`22:49.10`	`[TK]D-Fender`	`go manually load it from CLI FIRST`
`22:49.18`	`[TK]D-Fender`	`"module load chan_sip.so"`
`22:49.28`	`[TK]D-Fender`	`if it DOES load then immediately verify your modules.conf`
`22:51.35`	`mophed`	`== Parsing '/etc/asterisk/users.conf': Not found (Permission denied)`
`22:51.38`	`mophed`	`line 2`
`22:53.35`	`moreder`	`hi D-Fender. maybe you have any clue how to disable inband dtmf processing in asterisk?`
`22:56.52`	`[TK]D-Fender`	`Tell * not to listen for inband`
`22:57.07`	`*** join/#asterisk Y04NN (~y04nn@2a01:e34:ef37:5870:14f6:a7c3:3c34:c01d)`
`22:57.13`	`[TK]D-Fender`	`they are YOUR peers. Set the value for this accordingly`
`22:57.52`	`[TK]D-Fender`	`mophed, and I don't; see the REST of the lines. This is not confirming that is NOT loaded.`
`22:58.31`	`[TK]D-Fender`	`mophed, But while you're at PERMISSIONS is something you should check on your configs in general`
`23:00.39`	`moreder`	`D-Fender: could you suggest which peer parameter can turn it off?`
`23:00.48`	`mophed`	`the users.conf file does exist http://pastebin.ca/3755271 thats the only things uncommented in it. i havent touched the users.conf at all`
`23:02.12`	`mophed`	`http://pastebin.ca/3755273 was the full output. i will try to change the permissions now`
`23:02.23`	`*** part/#asterisk igcewieling (~ewieling@ip98-170-211-145.pn.at.cox.net)`
`23:05.27`	`*** join/#asterisk friedrich (~friedrich@aextron.de)`
`23:10.37`	`[TK]D-Fender`	`moreder, the very clear "dtmfmode".`
`23:11.15`	`[TK]D-Fender`	`So apparently users.conf did NOT stop it from loading`
`23:11.28`	`[TK]D-Fender`	`And it WASN'T loaded at * startup`
`23:11.48`	`[TK]D-Fender`	`So that should tell you to immediately verify modules.conf`
`23:12.53`	`mophed`	`im running all of this as root which should be able to access any file. so when you say permissions is it something i need to setup in a * config file?`
`23:13.15`	`[TK]D-Fender`	`running HOW?`
`23:13.27`	`[TK]D-Fender`	`ps -aux\|grep asterisk`
`23:14.56`	`moreder`	`D-Fender: i have DTMFmode: inband on both peers. tcpdump on incoming peer records clear tones, but on ougoing it's creepy, interrupted and distorted`
`23:15.49`	`mophed`	`asterisk 3604 0.6 1.7 1537704 35272 ? Ssl 05:44 0:35 /usr/sbin/asterisk -g -f -U asterisk`
`23:15.52`	`mophed`	`root 3899 0.0 0.3 112564 7736 pts/5 S+ 06:40 0:00 rasterisk r`
`23:15.55`	`mophed`	`root 4155 0.0 0.0 12720 952 pts/6 S+ 07:18 0:00 grep asterisk`
`23:16.08`	`[TK]D-Fender`	`so it's clearly running as asterisk`
`23:17.42`	`[TK]D-Fender`	`moreder, So you asked how to NOT use inband .. and then confirm you are TELLING it to. I don't what you're trying to do.`
`23:20.19`	`mophed`	`http://pastebin.ca/3755276 is the modules config`
`23:21.01`	`moreder`	`D-Fender: i want inband dtmf on both sides. but for some reason dtmf on outgoing peer is not even close to incoming. i suppose it's because of *`
`23:21.05`	`WIMPy`	`Was it able to read modules.conf?`
`23:21.15`	`mophed`	`should i just add load => chan_sip.so ?`
`23:22.11`	`[TK]D-Fender`	`moreder, You're dealing with audio, so codec, jitter, etc can all play against you.`
`23:22.37`	`[TK]D-Fender`	`mophed, I'm not seeing proof of PERMISSION like I suggest already`
`23:22.57`	`[TK]D-Fender`	`<moreder> hi D-Fender. maybe you have any clue how to disable inband dtmf processing in asterisk?`
`23:23.05`	`[TK]D-Fender`	`moreder> D-Fender: i want inband dtmf on both sides.`
`23:23.32`	`[TK]D-Fender`	`What. The. Fuck.`
`23:24.29`	`mophed`	`im trying to find how i set those permissions chown -R asterisk /var/log/asterisk ?`
`23:24.41`	`[TK]D-Fender`	`Let's try to fix the mentaily`
`23:24.43`	`[TK]D-Fender`	`PROVE IT FIRST`
`23:24.50`	`[TK]D-Fender`	`do NOT fuck with the EVIDENCE`
`23:24.56`	`[TK]D-Fender`	`EYS <- use them`
`23:24.58`	`[TK]D-Fender`	`EYES*`
`23:25.17`	`[TK]D-Fender`	`Thiws is why things take so long. Refusing to look`
`23:25.40`	`[TK]D-Fender`	`Then shooting at random and missing`
`23:26.19`	`mophed`	`i guess im just not understaning`
`23:27.04`	`[TK]D-Fender`	`-----> SHOW THE FILES IN THE FOLDER ** NOW ** <------`
`23:27.17`	`[TK]D-Fender`	`Getting warmer?`
`23:27.24`	`moreder`	`D-Fender: i'm not sure but i think * is processing my inband dtmf. i just want * to pass it as it is.`
`23:27.45`	`[TK]D-Fender`	`is IS processing it`
`23:27.55`	`[TK]D-Fender`	`and you can't stop it if it is sitting in between`
`23:28.09`	`[TK]D-Fender`	`So that measn allowing re-invites and keeping the stream away from *`
`23:28.18`	`[TK]D-Fender`	`Which may or may not be possible depending on your scenario`
`23:29.17`	`WIMPy`	`Inband DTMF might work better if you tell Asterisk that it's somethign else. Off course you can't make Asterisk Read DTMF any more then.`
`23:29.46`	`mophed`	`http://tinypic.com/r/2ah73n6/9 the contents of the asterisk dir`
`23:30.06`	`[TK]D-Fender`	`facepalms`
`23:30.37`	`WIMPy`	`LOL`
`23:30.38`	`mophed`	`im sorry`
`23:31.00`	`WIMPy`	`Seriousely`
`23:31.22`	`[TK]D-Fender`	`Yes, the situation is serious and the partient is critical`
`23:31.29`	`[TK]D-Fender`	`patient*`
`23:32.55`	`*** join/#asterisk matrix1233 (~matrix123@197.3.231.61)`
`23:34.42`	`mophed`	`i just dont understand what you are saying i need to show proof of. i see that the chan_sip.so was loaded. and that permission was denied for both the users.conf and sip_notify.conf files`
`23:35.17`	`WIMPy`	`I guess some computer/Unix/Linux basics would help.`
`23:36.03`	`mophed`	`i mean, i feel like i have "basic" skills`
`23:36.05`	`WIMPy`	`I didn't really follow up so far, but that screenshot really made me laugh.`
`23:36.31`	`[TK]D-Fender`	`ls -la /etc/asterisk`
`23:36.32`	`[TK]D-Fender`	`^^^^^^^^^^^^^^^^^^^^^`
`23:36.45`	`[TK]D-Fender`	`US RTHE OPTION FOR LS THAT EVEN ***LISTS*** PERMISSIONS`
`23:36.51`	`moreder`	`WIMPy: i will try your suggestion, thank you`
`23:37.25`	`[TK]D-Fender`	`Let's try this AGAIN.............`
`23:37.46`	`moreder`	`D-Fender: directmedia not possible in my scenario :(`
`23:37.56`	`[TK]D-Fender`	`moreder, then game over`
`23:39.51`	`mophed`	`every file is -rw-r----- except sip.conf manager.d and extensions.conf which are -rw-r--r-- drwxr-xr-x`
`23:40.27`	`[TK]D-Fender`	`gives up`
`23:40.54`	`mophed`	`http://pastebin.ca/3755282`
`23:41.07`	`mophed`	`well i really apreciate the help you have tried to provide`
`23:41.29`	`WIMPy`	`So obviousely Asterisk isn't allowed to read those files.`
`23:41.46`	`WIMPy`	`That's a pretty good reason for failing.`
`23:42.01`	`moreder`	`mophed, change owner: chown asterisk:asterisk -R /etc/asterisk`
`23:42.18`	`mophed`	`and trust me when i say that i am the most frustrated here. i thought this was going to be an easy setup process. and now 4 days later i just want to throw this computer out the window.`
`23:43.00`	`mophed`	`and if you could even come over here and just punch my face. i wouldnt mind.`
`23:43.04`	`mophed`	`moreder: thank you`
`23:48.28`	`[TK]D-Fender`	`Look at the pain it tool for you to prove who * is running as and what the permissions on your configs was`
`23:48.36`	`[TK]D-Fender`	`Asterisk wasn't the problem.`
`23:48.44`	`[TK]D-Fender`	`And even told you "permissions"`
`23:48.57`	`[TK]D-Fender`	`This one is on you`
`23:49.12`	`[TK]D-Fender`	`So NOW........`
`23:50.30`	`mophed`	`[Jan 8 07:53:01] NOTICE[4442]: chan_sip.c:15752 sip_reg_timeout: -- Registration for âXXXXXXXX@sip.flowroute.com' timed out, trying again (Attempt #18)`
`23:50.41`	`mophed`	`now at least i see its trying`
`23:51.09`	`moreder`	`D-Fender: i found something that maybe could help me: Asterisk will consistently intercept, buffer and regenerate all DTMF tones it detects. This may be a problem, as some of those appliances use distinct DTMF-silence cadences, and by regenerating those tones they are output with Asterisk's signal-silence cadences.In order to avoid this behaviour in Asterisk, what you can do is to modify the`
`23:51.09`	`moreder`	`frequency of the DTMF tones Asterisk expects, so that the true DTMF tones are not recognized and therefore let alone by the DSP module.`
`23:52.17`	`WIMPy`	`Unless you have dtmfmode=inband or "features" activated, they shouldn't reach any DSP routines.`
`23:52.19`	`[TK]D-Fender`	`<mophed> [Jan 8 07:53:01] NOTICE[4442]: chan_sip.c:15752 sip_reg_timeout: -- Registration for âXXXXXXXX@sip.flowroute.com' timed out, trying again (Attempt #18) <- this is NOT what I should have seen after all this time`
`23:52.53`	`[TK]D-Fender`	`WIMPy, * still interprets it from each side and retransmits. It does not actuall pass THROUGH as is ever`
`23:53.02`	`[TK]D-Fender`	`it's supposed to be "always translated"`
`23:53.10`	`[TK]D-Fender`	`in full B2BUA style`
`23:53.15`	`[TK]D-Fender`	`Which is his gripe`
`23:53.39`	`WIMPy`	`That's broken.`
`23:54.05`	`WIMPy`	`Or would be.`
`23:55.52`	`*** join/#asterisk sh1rtybird (~quassel@beepbeep.serverpit.com)`
`23:58.49`	`moreder`	`many thank D-Fender and WIMPy. will try few thing tomorrow`
`23:59.45`	`mophed`	`http://pastebin.ca/3755289`
`23:59.59`	`[TK]D-Fender`	`SIP DEBUG`

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