| 08:40.05 | *** join/#asterisk infobot (ibot@69-58-76-73.ut.vivintwireless.net) |
| 08:40.05 | *** 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), 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 |
| 09:18.31 | *** join/#asterisk denbeiren (~quassel@ptr-178-51-223-24.dyn.mobistar.be) |
| 09:26.17 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
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| 09:42.22 | *** join/#asterisk andycol_500 (~Andrew@41.79.82.246) |
| 09:42.44 | andycol_500 | ChannelZ i do have type=auth |
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| 09:52.26 | *** join/#asterisk marlinc (~marlinc@ip2.weert.li.nl.cvo-technologies.com) |
| 09:53.53 | andycol_500 | has anyone seen that error before |
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| 11:09.13 | Andrew__ | does anyone know how i can fix this pjsip error |
| 11:09.17 | Andrew__ | WARNING[6707]: res_pjsip_outbound_authenticator_digest.c:135 digest_create_request_with_auth_from_old: Unable to create request with auth.No auth credentials for any realms in challenge. |
| 11:09.51 | *** join/#asterisk u0m3 (~u0m3@92.80.98.70) |
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| 11:50.48 | *** join/#asterisk tiuman (~botto@mail.ooonmk.ru) |
| 11:51.41 | tiuman | hi all. my problem with nat? random mute call... not always mute random |
| 11:51.59 | tiuman | Sent RTP P2P packet to 193.201.229.19:11372 (type 08, len 000160) |
| 11:52.11 | tiuman | Sent RTP P2P packet to 193.201.229.19:11372 (type 08, len 000160) |
| 11:52.18 | tiuman | Sent RTP P2P packet to 193.201.229.19:11372 (type 08, len 000160) |
| 11:52.24 | tiuman | its mute... |
| 11:52.30 | tiuman | in cicle |
| 11:52.42 | tiuman | Sent RTP P2P packet to 193.201.229.19:10132 (type 08, len 000160) |
| 11:52.48 | tiuman | Sent RTP P2P packet to 193.201.229.19:19310 (type 08, len 000160) |
| 11:52.53 | tiuman | Sent RTP P2P packet to 193.201.229.19:10132 (type 08, len 000160) |
| 11:53.01 | tiuman | Sent RTP P2P packet to 193.201.229.19:19310 (type 08, len 000160) |
| 11:53.06 | tiuman | its not mute |
| 11:53.17 | *** join/#asterisk defsdoor (~andy@cpc37-sutt4-2-0-cust84.19-1.cable.virginm.net) |
| 11:53.22 | tiuman | what the my prblem?? |
| 11:54.18 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
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| 12:14.33 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:15.48 | *** join/#asterisk slavi (~slavi@217.145.93.100) |
| 12:16.02 | slavi | hey all :) |
| 12:16.40 | slavi | can any1 help me to configure asterisk ? |
| 12:18.19 | tiuman | ?? |
| 12:18.25 | [TK]D-Fender | ~ask |
| 12:18.26 | 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. |
| 12:22.02 | slavi | the problem is that i try to make custom extension_custom.conf file and my idea is to filter phone numbers and remove some characters from it this is example of file |
| 12:22.04 | slavi | [from-custom] |
| 12:22.04 | slavi | include => from-analog |
| 12:22.04 | slavi | exten => _X!,1,Set(CALLERID(num)=123123132131) |
| 12:22.06 | slavi | exten => _X!,n,Goto(from-analog,${EXTEN},1) |
| 12:22.41 | slavi | but when i do this telephone not ringing |
| 12:22.49 | *** join/#asterisk SeRi (~wtf@unaffiliated/seri) |
| 12:23.03 | [TK]D-Fender | That does tell us anything at all |
| 12:23.14 | [TK]D-Fender | We don't see you dialing anything |
| 12:23.31 | [TK]D-Fender | And we don't know the rest of your dialplan or if this particular dialplan even gets used |
| 12:24.51 | slavi | if i use only from-analog part |
| 12:25.10 | slavi | but i wont first to edit the phone |
| 12:25.26 | slavi | and then go to from-analog |
| 12:26.12 | [TK]D-Fender | That context doesn't eman anything yet.... |
| 12:26.22 | [TK]D-Fender | How do we know what you put in there? |
| 12:26.49 | slavi | for context i put form-custom |
| 12:27.09 | [TK]D-Fender | Again you are just mentioning context names for which we have no idea what is IN THEM |
| 12:29.06 | slavi | is this [form-custom] that i send is with right syntax |
| 12:30.29 | [TK]D-Fender | Ok, this doesn't seem to be sinking in. |
| 12:31.37 | [TK]D-Fender | We have no idea what the CONTENTS of those contexts are. Stop giving us context names without proof of what is IN THEM. |
| 12:32.51 | slavi | from-analog just include from-dahdi |
| 12:33.08 | [TK]D-Fender | Why are you still just giving us context names? |
| 12:33.13 | [TK]D-Fender | What don' you understand?\ |
| 12:33.19 | juned | slavi: provide complete dialplan like content of from-analog .. that will help us to understand the problem |
| 12:33.25 | tiuman | Why sometimes I can not hear??? |
| 12:34.00 | slavi | [from-custom] |
| 12:34.00 | slavi | include => from-analog |
| 12:34.00 | slavi | exten => _X!,1,Set(CALLERID(num)=123123132131) |
| 12:34.00 | slavi | exten => _X!,n,Goto(from-analog,${EXTEN},1) |
| 12:34.02 | [TK]D-Fender | tiuman: Can't hear what? Using what? |
| 12:34.11 | [TK]D-Fender | slavi: that code doesn't dial anything. |
| 12:34.11 | tiuman | asterisk -> nat -> provider... |
| 12:34.24 | slavi | [from-dahdi] |
| 12:34.24 | slavi | exten => _X.,1,Set(DID=${EXTEN}) |
| 12:34.25 | slavi | exten => _X.,n,Goto(s,1) |
| 12:34.25 | slavi | exten => s,1,Noop(Entering from-dahdi with DID == ${DID}) |
| 12:34.25 | slavi | exten => s,n,Ringing() |
| 12:34.25 | slavi | exten => s,n,Set(DID=${IF($["${DID}"= ""]?s:${DID})}) |
| 12:34.27 | slavi | exten => s,n,Noop(DID is now ${DID}) |
| 12:34.27 | tiuman | asterisk -> nat -> provider -> GSM telephone |
| 12:34.28 | [TK]D-Fender | tiuman: Where are "you" in that picture? |
| 12:34.29 | slavi | exten => s,n,GotoIf($["${CHANNEL:0:5}"="DAHDI"]?dahdiok:checkzap) |
| 12:34.31 | slavi | exten => s,n(checkzap),GotoIf($["${CHANNEL:0:3}"="Zap"]?zapok:neither) |
| 12:34.33 | [TK]D-Fender | slavi: PASTBIN! |
| 12:34.33 | slavi | exten => s,n(neither),Goto(from-pstn,${DID},1) |
| 12:34.36 | [TK]D-Fender | ~pb |
| 12:34.36 | 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. |
| 12:34.37 | slavi | exten => s,n,Macro(Hangupcall,dummy) |
| 12:34.39 | slavi | exten => s,n(dahdiok),Noop(Is a DAHDi Channel) |
| 12:34.41 | slavi | exten => s,n,Set(CHAN=${CHANNEL:6}) |
| 12:34.43 | slavi | exten => s,n,Set(CHAN=${CUT(CHAN,-,1)}) |
| 12:34.44 | [TK]D-Fender | slavi: PASTEBIN! |
| 12:34.45 | slavi | exten => s,n,Macro(from-dahdi-${CHAN},${DID},1) |
| 12:34.46 | [TK]D-Fender | slavi: PASTEBIN! |
| 12:34.47 | slavi | exten => s,n,Noop(Returned from Macro from-dahdi-${CHAN}) |
| 12:34.47 | [TK]D-Fender | slavi: PASTEBIN! |
| 12:34.49 | tiuman | [TK]D-Fender russia |
| 12:34.49 | slavi | exten => s,n,Goto(from-pstn,${DID},1) |
| 12:34.51 | slavi | exten => s,n(zapok),Noop(Is a Zaptel Channel) |
| 12:34.53 | slavi | exten => s,n,Set(CHAN=${CHANNEL:4}) |
| 12:34.55 | slavi | exten => s,n,Set(CHAN=${CUT(CHAN,-,1)}) |
| 12:34.57 | slavi | exten => s,n,Macro(from-dahdi-${CHAN},${DID},1) |
| 12:34.59 | slavi | exten => s,n,Noop(Returned from Macro from-dahdi-${CHAN}) |
| 12:35.01 | slavi | exten => s,n,Goto(from-pstn,${DID},1) |
| 12:35.03 | slavi | ;--== end of [from-dahdi] ==--; |
| 12:35.05 | [TK]D-Fender | tiuman: No, what part of that chain you drew is YOU listening? |
| 12:35.07 | slavi | [from-analog] |
| 12:35.09 | slavi | include => from-dahdi |
| 12:35.15 | [TK]D-Fender | ~ops |
| 12:35.17 | [TK]D-Fender | ~op |
| 12:35.17 | infobot | hmm... op is a wallet |
| 12:35.27 | juned | slavi: http://pastebin.com/ |
| 12:35.28 | [TK]D-Fender | slavi: do NOT flood in here like that |
| 12:35.45 | [TK]D-Fender | ~pb |
| 12:35.45 | 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. |
| 12:35.47 | [TK]D-Fender | ^^^^^ |
| 12:36.07 | slavi | http://pastebin.com/bNX4Q4uh |
| 12:36.10 | [TK]D-Fender | slavi: pastebin THE CALL so we can see what is actually HAPPENING |
| 12:36.42 | tiuman | i call from GSM in the provider-> nat -> asterisk -> call to via provider -> gsm telephone |
| 12:36.54 | tiuman | i call from GSM To the provider-> nat -> asterisk -> call to via provider -> gsm telephone |
| 12:37.22 | [TK]D-Fender | tiuman: So in from provider to Asterisk, then Asterisk dials back out your provider? |
| 12:37.30 | slavi | this is the error from call http://pastebin.com/uE0ujJXL |
| 12:37.44 | [TK]D-Fender | WARNING[8851][C-0000000b] pbx.c: Channel 'DAHDI/5-1' sent to invalid extension but no invalid handler: context,exten,priority=from-custom,s,1 |
| 12:37.51 | [TK]D-Fender | it is TELLING you where it is sending the call. |
| 12:37.55 | [TK]D-Fender | And it does not EXIST |
| 12:37.57 | tiuman | not provider to gsm phone |
| 12:38.01 | tiuman | [TK]D-Fender |
| 12:38.07 | [TK]D-Fender | slavi: context,exten,priority=from-custom,s,1 <------------------------------ |
| 12:38.16 | tiuman | not via provider to gsm phone |
| 12:38.33 | tiuman | Multifon = provider |
| 12:38.41 | [TK]D-Fender | tiuman: fix your description |
| 12:38.45 | tiuman | have the nuber 79241124511 |
| 12:39.08 | [TK]D-Fender | tiuman: redo your call-path description |
| 12:39.17 | tiuman | Multifon = russina SIP provider... i have the account in the him |
| 12:40.24 | tiuman | I call from GSM telephone in the this number 79241124511 it number it account this SIP provider... |
| 12:40.27 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 12:40.51 | tiuman | my asterisk have registered this number |
| 12:41.27 | [TK]D-Fender | tiuman: Fix your description of the entire call path |
| 12:41.28 | tiuman | so after asterisk call to the other GSM Phone |
| 12:41.42 | [TK]D-Fender | tiuman: And don't give us phone number and account specifics |
| 12:41.42 | tiuman | sometimes i not hear |
| 12:42.14 | tiuman | what description* |
| 12:42.15 | tiuman | ??? |
| 12:42.46 | [TK]D-Fender | CALL ARRIVES AT aSTERISK FROM x. aSTERISK CALLS OUT USING x. aND THE NETWORKING BETWEEN THEM |
| 12:43.39 | tiuman | CALL ARRIVES AT aSTERISK FROM x. aSTERISK CALLS OUT USING y. aND THE NETWORKING BETWEEN THEM |
| 12:44.14 | tiuman | x my phone y phone my friend |
| 12:44.24 | [TK]D-Fender | "my phone" means nothing |
| 12:44.26 | [TK]D-Fender | what KIND of phone. |
| 12:44.28 | tiuman | i want call him |
| 12:44.35 | [TK]D-Fender | how does the call arrive at Asterisk? |
| 12:44.37 | tiuman | via asterisk |
| 12:44.40 | [TK]D-Fender | HOW |
| 12:45.07 | [TK]D-Fender | I asked for a simple and clear chain. |
| 12:45.15 | tiuman | GSM PHONE CALLED TO ASTERISK |
| 12:45.33 | [TK]D-Fender | GSM doesn't magically send a call to Asterisk |
| 12:45.38 | [TK]D-Fender | Asterisk is SOFTWARE |
| 12:45.41 | tiuman | ASTERSIK CALLED TO OTHER PHONE |
| 12:45.55 | [TK]D-Fender | Microwaves do not magically fly through the air and land in software |
| 12:46.16 | tiuman | asterisk connected to sip provder |
| 12:46.28 | [TK]D-Fender | So GSM is NOT a valid description all by itself. |
| 12:46.33 | tiuman | have the account him |
| 12:46.42 | [TK]D-Fender | So this start from your PROVIDER first sending a call to Asterisk? |
| 12:47.06 | tiuman | i call in the provider |
| 12:47.19 | [TK]D-Fender | Is that a "yes"? |
| 12:47.38 | tiuman | i dont understanding you answer |
| 12:48.05 | tiuman | Y -> astersik - > X |
| 12:48.27 | [TK]D-Fender | What is the first thing sending a call TO your server? |
| 12:48.32 | [TK]D-Fender | What is the START of this process? |
| 12:48.42 | [TK]D-Fender | What is the FIRST action happening in this chain? |
| 12:48.53 | tiuman | y -> via sip provider -> asterisk -> via sip provider -> x |
| 12:49.07 | [TK]D-Fender | FINALLY. |
| 12:49.15 | *** join/#asterisk sparetire_ (~sparetire@unaffiliated/sparetire) |
| 12:49.31 | tiuman | astreisk call to y |
| 12:49.44 | tiuman | and bridging 2 channels |
| 12:50.01 | tiuman | x and y Speaking |
| 12:50.05 | tiuman | sometimes |
| 12:50.08 | [TK]D-Fender | If you don't have audio then you have a NETWORKING issuee |
| 12:50.19 | [TK]D-Fender | Likely due to imporper NAT configuration |
| 12:50.22 | tiuman | Some times not speaking beacause MUTE |
| 12:50.31 | [TK]D-Fender | What ver of *? |
| 12:50.55 | tiuman | why only sometimes? |
| 12:51.09 | tiuman | why always? |
| 12:51.12 | slavi | i repair the error and now this is call log http://pastebin.com/0ihWFZUx |
| 12:51.24 | slavi | but number still not changed |
| 12:52.10 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 12:52.16 | slavi | i wont to change number when someone is calling |
| 12:52.17 | tiuman | http://pastebin.com/83900rRQ not hear |
| 12:52.32 | tiuman | <PROTECTED> |
| 12:52.32 | tiuman | <PROTECTED> |
| 12:52.40 | tiuman | it hear... |
| 12:52.52 | tiuman | -- SIP/android-out-00000014 is making progress passing it to SIP/android-in-00000013 |
| 12:52.56 | tiuman | it not hear |
| 12:53.02 | tiuman | it different |
| 12:53.19 | tiuman | without 2 string |
| 12:53.22 | [TK]D-Fender | Reliably Transmitting (NAT) to 193.201.229.35:5060: <- BAD |
| 12:53.33 | [TK]D-Fender | Your provider is NOT behind NAT. Fix your peer for them |
| 12:53.59 | [TK]D-Fender | slavi: Clearly your code is not what's being executed and there is another context definition with other stuff in it |
| 12:54.29 | tiuman | [out-priveider]nat=force_rport,comedia |
| 12:54.42 | [TK]D-Fender | slavi: Which means you should make your OWN context name for your code... send the call THERE and then jump back to wherever you want to resume |
| 12:54.54 | [TK]D-Fender | tiuman: "nat=NO" |
| 12:55.00 | tiuman | ok! |
| 12:55.04 | [TK]D-Fender | -> NO <- |
| 12:55.40 | tiuman | directmedia=no |
| 12:55.40 | tiuman | ??? |
| 12:55.44 | slavi | Could it be because in the beginning of from-custom we do a 'include => from-analog' ? Does an include immediately start running the included context? |
| 12:55.48 | [TK]D-Fender | You are telling your server to IGNORE the IP's offered for media. If they use separate media & signaling servers thiis means * will no send to the right place |
| 12:56.01 | [TK]D-Fender | tiumandirectmedia=no <- this is required |
| 12:56.23 | [TK]D-Fender | slavi: You are clearly making a DUPLICATE context name |
| 12:56.29 | tiuman | in the peer? |
| 12:56.36 | [TK]D-Fender | slavi: All that other stuff is clearly already in a context of the same name. |
| 12:56.39 | tiuman | [provider-out]??? |
| 12:56.47 | [TK]D-Fender | slavi: Make you custom stuff in it's OWN context |
| 12:56.59 | [TK]D-Fender | tiuman: In the peers for your provider |
| 12:57.34 | tiuman | ok |
| 12:57.49 | tiuman | [general] nat=yes? |
| 12:58.06 | tiuman | directmedia=no |
| 12:58.08 | tiuman | ? |
| 12:58.12 | tiuman | in the general? |
| 12:58.13 | [TK]D-Fender | that is fine for general |
| 12:58.32 | tiuman | directmedia=no all ??? |
| 12:58.33 | [TK]D-Fender | YOU are behind NAT... your provider(s) are NOT |
| 12:58.40 | [TK]D-Fender | NEVER allow directmedia |
| 12:58.40 | tiuman | ok! |
| 12:59.03 | tiuman | YOU GOD!!! |
| 12:59.50 | slavi | [TK]D-Fender: i rename the context but the log is same and my code not executed |
| 13:00.09 | [TK]D-Fender | And are you sending the CALL to this new context? |
| 13:00.15 | slavi | yes |
| 13:00.30 | [TK]D-Fender | You can't just invent new contexts and expect them to get used without sending calls in there... |
| 13:00.31 | [TK]D-Fender | show us |
| 13:00.46 | [TK]D-Fender | "dialplan show YOURCONTEXT" <- |
| 13:00.49 | [TK]D-Fender | PB along with the new call |
| 13:03.37 | slavi | http://pastebin.com/CLNMDbw3 |
| 13:03.47 | slavi | this is the context and the call |
| 13:05.18 | tiuman | [TK]D-Fender problem not solved |
| 13:05.26 | tiuman | sometimes mute |
| 13:05.35 | [TK]D-Fender | slavi: Executing [s@from-hukuto:1] NoOp("DAHDI/5-1", "Entering from-dahdi with DID == ") in new stack |
| 13:05.42 | [TK]D-Fender | the call is landing on "s" just like it says |
| 13:05.52 | [TK]D-Fender | And you did NOT make a match for "s" |
| 13:05.56 | [TK]D-Fender | \It is using that INCLUDE |
| 13:06.03 | [TK]D-Fender | Why the hell do you have that include there? |
| 13:06.07 | [TK]D-Fender | that is NOT your code. |
| 13:06.14 | *** part/#asterisk juned (~juned@202.131.119.122) |
| 13:06.15 | [TK]D-Fender | Why should you allow it to go somewhere ELSE than your code? |
| 13:06.36 | [TK]D-Fender | tiuman: And I'm not seeing a new & complete call.... |
| 13:08.30 | tiuman | http://pastebin.com/9MQkZaV5 |
| 13:09.54 | slavi | when i remove include the call not proceed to from-analog (context) |
| 13:10.15 | slavi | it give me error that it cant find this context |
| 13:10.49 | [TK]D-Fender | [09:05][TK]D-Fenderthe call is landing on "s" just like it says |
| 13:10.50 | [TK]D-Fender | [09:05][TK]D-FenderAnd you did NOT make a match for "s" |
| 13:11.50 | slavi | could you please help me with a match for "s" |
| 13:12.23 | tiuman | http://pastebin.com/KRNA2mfU |
| 13:12.44 | tiuman | [TK]D-Fender |
| 13:12.53 | tiuman | its mutted |
| 13:12.56 | [TK]D-Fender | slavi: '_X!' => 1. Set(CALLERID(num)=123123132131) [pbx_config] <- does this look like "s" will land on it to you? |
| 13:13.10 | [TK]D-Fender | slavi: that is a match for a DGITI plus any other number of other chars |
| 13:13.11 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 13:13.25 | [TK]D-Fender | slavi: that will NOT match "s". "s" is a letter of the alphabet. Not a digit |
| 13:14.27 | [TK]D-Fender | [09:06][TK]D-Fendertiuman: And I'm not seeing a new & complete call.... |
| 13:14.41 | *** join/#asterisk tparcina (~tomo@212.92.200.41) |
| 13:15.07 | *** join/#asterisk bulkorok (~Adium@89.245.151.228) |
| 13:15.21 | tiuman | the difference is deaf and hearing? |
| 13:16.11 | [TK]D-Fender | sits back and awaits what he asked for... |
| 13:17.21 | tiuman | http://pastebin.com/uWRNfdnX |
| 13:18.58 | tiuman | [TK]D-Fender its not hear calling |
| 13:21.42 | *** join/#asterisk jameswf (uid27319@gateway/web/irccloud.com/x-zvemgkmpjskrohwp) |
| 13:22.53 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 13:23.04 | slavi | IT WORKS THANKS !!!!!!!!!!!!!!!! |
| 13:24.23 | tiuman | [TK]D-Fender??? |
| 13:26.43 | [TK]D-Fender | [09:14][TK]D-Fender[09:06][TK]D-Fendertiuman: And I'm not seeing a new & complete call.... |
| 13:27.04 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 13:27.19 | tiuman | [TK]D-Fender sip debug on?? |
| 13:27.32 | tiuman | sip set debug on |
| 13:27.35 | tiuman | ??? |
| 13:27.44 | [TK]D-Fender | COMPLETE CALL |
| 13:27.47 | [TK]D-Fender | ALL of it |
| 13:27.49 | [TK]D-Fender | You are showing HALF |
| 13:28.02 | tiuman | aaaa |
| 13:28.03 | tiuman | ok |
| 13:31.43 | tiuman | http://pastebin.com/9pzRP5Ub |
| 13:31.51 | tiuman | it hear |
| 13:31.54 | tiuman | call |
| 13:35.47 | [TK]D-Fender | Contact: <sip:79244668977@188.244.188.181:9980 |
| 13:35.54 | [TK]D-Fender | this is a very unusual port to be sending from |
| 13:36.05 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 13:36.05 | [TK]D-Fender | Also, what do you have forwarded to your server, and what router are you using? |
| 13:38.58 | tiuman | [TK]D-Fender it my router |
| 13:39.09 | [TK]D-Fender | I asked you what MODEL. |
| 13:39.24 | [TK]D-Fender | and what you FORWARDED to your server from it |
| 13:40.00 | tiuman | port forwardin 188.244.188.181 9980 to 192.168.0.80 5060 |
| 13:41.56 | [TK]D-Fender | This is not good. * passing the ports int he CONTENT of the packets, rewriting them is BAD |
| 13:42.50 | *** join/#asterisk SPERman (~botto@mail.ooonmk.ru) |
| 13:42.57 | SPERman | port forwardin 188.244.188.181 9980 to 192.168.0.80 5060 |
| 13:43.01 | SPERman | [TK]D-Fender |
| 13:43.05 | SPERman | it i tiuman |
| 13:43.24 | [TK]D-Fender | [09:41][TK]D-FenderThis is not good. * passing the ports int he CONTENT of the packets, rewriting them is BAD |
| 13:43.25 | SPERman | mikrotik |
| 13:43.31 | [TK]D-Fender | What else do you have forwarded? |
| 13:43.58 | SPERman | must be 5060?? |
| 13:44.11 | SPERman | RTP |
| 13:44.40 | SPERman | 6001-20000 |
| 13:44.42 | SPERman | UDP |
| 13:44.52 | SPERman | 9980 udp |
| 13:44.52 | [TK]D-Fender | What aee you messing with the port in the first place? |
| 13:45.10 | SPERman | what??? |
| 13:45.31 | SPERman | 9980 to 192.168.0.80:5060 |
| 13:45.56 | SPERman | second 6001-20000 to 192.168.0.80:6001-20000 |
| 13:46.02 | SPERman | udp |
| 13:47.09 | SPERman | When was the default port had the same problem |
| 13:47.11 | SPERman | 5060 |
| 13:47.37 | SPERman | externaddr=188.244.188.181:5060 |
| 13:50.57 | SPERman | i disable 5060 in microtik. |
| 13:51.03 | SPERman | service ports |
| 13:51.05 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 13:51.21 | SPERman | DIRECT SIP media |
| 13:55.02 | SPERman | asterisk(192.168.0.80) -> mikrotik(192.168.0.33 lan,192.168.81.1 pptp local) -PPTP_tunnel_without_nat-> mikrotik(PORT FORWARDING)(192.168.81.1 pptpremote,192.168.88.1 lan ,188.244.188.181 wan) -> INTERNET |
| 13:55.54 | SPERman | pptp it's bad?? |
| 13:56.28 | SPERman | mikrotik 192.168.81.1 can see 192.168.0.80 |
| 13:56.44 | SPERman | ping socekt |
| 13:59.13 | *** join/#asterisk malcolmd (malcolmd@pdpc/sponsor/digium/malcolmd) |
| 13:59.14 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 14:00.04 | [TK]D-Fender | Ok, too much extra manipulation going on and the story is changing... |
| 14:01.06 | [TK]D-Fender | drops this case |
| 14:02.14 | SPERman | SIP ALG |
| 14:02.22 | SPERman | ip firewall service-port disable sip |
| 14:03.08 | SPERman | [TK]D-Fender it is problem routing??? |
| 14:03.22 | SPERman | not asterisk settings? |
| 14:04.44 | SPERman | how tunneling protocol like asterisk ??? |
| 14:06.21 | *** join/#asterisk glNito (6ca1d582@gateway/web/cgi-irc/kiwiirc.com/ip.108.161.213.130) |
| 14:06.49 | *** join/#asterisk kharwell (kharwell@nat/digium/x-lhmtudrkvzjflyxt) |
| 14:08.29 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-nbkccyifaphvavbs) |
| 14:08.30 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 14:11.07 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 14:24.02 | *** join/#asterisk Synthase_ (uid63346@gateway/web/irccloud.com/x-tcjzylyppwaladmw) |
| 14:34.03 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 14:53.12 | SPERman | [TK]D-Fender it is problem routing??? |
| 14:57.17 | *** part/#asterisk etienne1 (~eallovon@avencall-gw-vbo.alionis.net) |
| 14:59.57 | *** join/#asterisk pjensen00 (~per@ip-69-178-218-71.far.ideaone.net) |
| 15:01.28 | pjensen00 | I'm trying to update the name of a bridge via the ARI. I'm getting a 200 back but the name remains unchanged. I don't see any difference in the doc between creating a new bridge and updating an existing bridge. The only difference is if the ID supplied in the query exists as a bridge or not. |
| 15:02.12 | pjensen00 | http://pastebin.com/iqMMJUpq |
| 15:03.17 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 15:03.46 | *** join/#asterisk tiuman (~botto@mail.ooonmk.ru) |
| 15:05.38 | pjensen00 | Anyone have any ideas? |
| 15:11.29 | *** join/#asterisk italorossi (~Adium@187.60.66.11) |
| 15:16.55 | *** join/#asterisk superscrat (~asanders@173-21-89-217.client.mchsi.com) |
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| 16:06.53 | *** part/#asterisk c0rnoTa (~c0rnoTa@91.221.232.65) |
| 16:07.43 | *** join/#asterisk busymind (~textual@c-24-21-79-235.hsd1.wa.comcast.net) |
| 16:08.07 | busymind | how many simultaneous connections can AMI handle in Asterisk 11? |
| 16:09.20 | WIMPy | I guess "as many as your system can handle" would be the answer. |
| 16:10.49 | busymind | WIMPy - is there a best practice? I've seen a few sources online that say that asterisk doesnt do well with many different connections |
| 16:11.00 | busymind | but i figured I'd ask the source :) |
| 16:11.27 | WIMPy | There was an issue in older versions, I think. But why do you need many connections? |
| 16:12.32 | busymind | I'm developing an operator panel |
| 16:12.49 | WIMPy | That sounds like one connection so far :-) |
| 16:12.56 | busymind | hahaha |
| 16:12.59 | busymind | i know |
| 16:14.02 | busymind | we're just doing research right now - and we're just trying to decide if we want to build a connector for asterisk as its own application, or have each agent connect to AMI directly |
| 16:14.49 | WIMPy | That should be more of a question of how you organize things within your application. |
| 16:16.13 | WIMPy | Or if tyhere's an application or if the frontend contains the application. |
| 16:16.55 | busymind | well a part of that planning was understanding Asterisk limitations |
| 16:17.03 | busymind | thats why I figured I'd ask the group :) |
| 16:27.41 | reveal | can someone make sense of this http://pastebin.com/sNgMSEGj |
| 16:30.58 | WIMPy | What is agi-OSDagent_conf.agi? Does it use an interpreter that isn't executable? |
| 16:31.57 | reveal | WIMPy: its a perl script for a dialer |
| 16:32.27 | WIMPy | So does it use a working version of perl? |
| 16:32.55 | reveal | perl is installed on the system if thats what you are asking |
| 16:33.05 | WIMPy | Nope |
| 16:33.41 | WIMPy | I want to know if *exactely* what that script refers to in the #!/... line works. |
| 16:33.43 | reveal | can you be a little bit more specific with me (sorry) |
| 16:33.52 | reveal | #!/usr/bin/perl |
| 16:34.15 | WIMPy | And can you start that agi manually? |
| 16:34.47 | reveal | ive never started an agi manually before |
| 16:34.55 | reveal | im still green to asterisk and etc |
| 16:35.22 | WIMPy | It's just an executable. |
| 16:35.33 | reveal | oh so in theory i can perl file.agi |
| 16:35.35 | reveal | ? |
| 16:35.48 | reveal | or do i need to execute this in the CLI for asterisk |
| 16:36.15 | WIMPy | Yes. But you should try without the "perl" part, as Asterisk would. |
| 16:36.37 | WIMPy | bbiab |
| 16:39.41 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 16:39.47 | *** join/#asterisk lbazan (~LoKoMurdo@fedora/LoKoMurdoK) |
| 16:41.03 | *** join/#asterisk mac_ified (~mac_ified@67-9-150-210.res.bhn.net) |
| 16:50.13 | *** join/#asterisk infobot (ibot@69-58-76-73.ut.vivintwireless.net) |
| 16:50.13 | *** 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), 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 |
| 16:51.07 | reveal | [2015-10-14 15:38:18] <DAHDI/i2/9116-8ad9>AGI Rx << verbose "Failed to execute '/var/lib/asterisk/agi-bin/agi-OSDagent_conf.agi': Permission denied" 1 |
| 16:51.10 | reveal | [2015-10-14 15:38:18] agi-OSDagent_conf.agi,agentmon: Failed to execute '/var/lib/asterisk/agi-bin/agi-OSDagent_conf.agi': Permission denied |
| 16:53.00 | craigify | chown asterisk:asterisk |
| 16:53.04 | craigify | on the AGi script |
| 16:53.09 | craigify | see if that works |
| 16:53.34 | reveal | it is |
| 16:53.37 | craigify | it's pretty normal for asterisk to run as the asterisk user, or even chroot as asterisk |
| 16:53.37 | reveal | getfacl /var/lib/asterisk/agi-bin/agi-OSDagent_conf.agi |
| 16:53.37 | reveal | getfacl: Removing leading '/' from absolute path names |
| 16:53.38 | reveal | # file: var/lib/asterisk/agi-bin/agi-OSDagent_conf.agi |
| 16:53.38 | reveal | # owner: asterisk |
| 16:53.38 | reveal | # group: asterisk |
| 16:53.44 | craigify | ok |
| 16:53.49 | reveal | and it still errors |
| 16:54.04 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 16:54.47 | craigify | and asterisk is running as what user? |
| 16:55.00 | reveal | root when i grep the pid |
| 16:56.21 | craigify | ok |
| 16:56.39 | craigify | the executable bit is set? |
| 16:56.47 | craigify | what else could be going on... |
| 16:57.39 | craigify | I'd also check for the existence of /usr/bin/perl |
| 16:57.44 | craigify | just to make sure |
| 16:58.05 | craigify | and make sure it's not something silly like perl was installed in /usr/local/bin |
| 16:58.20 | reveal | <PROTECTED> |
| 16:58.20 | reveal | /usr/bin/perl |
| 16:58.21 | reveal | [root@rrsi-o1 agi-bin]# |
| 16:58.52 | craigify | you verified that the executable bit is set on the agi script? |
| 16:59.21 | reveal | user::rwx |
| 16:59.21 | reveal | group::rwx |
| 16:59.22 | reveal | other::rwx |
| 16:59.34 | craigify | of course it is |
| 16:59.35 | craigify | lol |
| 16:59.49 | reveal | 777 |
| 17:00.00 | craigify | did you do your rain dance this morning? |
| 17:00.10 | reveal | no lol |
| 17:00.21 | craigify | pray to the god of unix or something |
| 17:00.26 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
| 17:00.36 | reveal | lol |
| 17:00.43 | reveal | i shouldnt have to pray :( |
| 17:00.49 | craigify | those are all of the things I would have done |
| 17:01.16 | craigify | how do you run a syntax check on a perl script? |
| 17:01.47 | craigify | I've had AGI scripts with syntax errors cause weird AGI errors in asterisk with no useful error message |
| 17:02.03 | craigify | specifically PHP scripts |
| 17:02.19 | reveal | maybe perl -c %F |
| 17:02.23 | craigify | php would exit non zero, but stderr doesn't make it way back to asterisk |
| 17:02.30 | craigify | so you're left wondering why it didn't work |
| 17:02.37 | craigify | but, it wasn't a permission denied error |
| 17:03.56 | reveal | <PROTECTED> |
| 17:03.57 | reveal | agi-OSDagent_conf.agi syntax OK |
| 17:04.26 | craigify | I'd start changing up the situation so I can test parts. essentially do unit testing |
| 17:04.29 | craigify | make a simple script |
| 17:04.33 | craigify | see if that works |
| 17:04.42 | craigify | perhaps not even use perl |
| 17:04.45 | craigify | then try using perl |
| 17:04.55 | craigify | add in complexity on each step until you reproduce the error |
| 17:05.31 | reveal | can you execute an agi script inside the CLI in asterisk |
| 17:05.52 | craigify | assuming that you'll get it to work using a very simple unit test, then hopefully discover the problem as you add complexity, such as using perl, or python, or php |
| 17:05.59 | craigify | never tried |
| 17:06.02 | craigify | not sure |
| 17:06.06 | reveal | bc i have a agi-test script |
| 17:08.22 | pjensen00 | Ha, nevermind...... I finally got an error message back from Stasis "Changing bridge name is not implemented". Weird that it will give 500's if the name is blank to start with but still fail the same way. |
| 17:08.37 | pjensen00 | *give 200's |
| 17:09.39 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 17:15.37 | *** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright) |
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| 17:53.11 | *** join/#asterisk talntid (~talntid@173-160-189-58-Washington.hfc.comcastbusiness.net) |
| 17:57.20 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 17:57.39 | talntid | has anyone noticed that the db schema, naming schema for realtime stuff is horrible? |
| 17:58.14 | talntid | no primary keys for the most part, no foreign keys, inconsistent naming |
| 17:59.56 | WIMPy | You are new to Asterisk? |
| 18:01.29 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-icwlkrvyrletabpt) |
| 18:04.32 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 18:05.10 | talntid | i suppose so :P |
| 18:06.31 | reveal | WIMPy: hey |
| 18:10.37 | *** join/#asterisk estranger (~russ@204.148.2.242) |
| 18:19.44 | *** join/#asterisk karelk (~karel@31.10.145.202) |
| 18:27.27 | *** join/#asterisk ThomasKeller (~Thomas@vmx.ethz.ch) |
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| 18:43.41 | *** join/#asterisk retentiveboy (~retentive@2001:558:6011:71:ad79:e98a:280a:2221) |
| 18:57.09 | newtonr | talntid, Good thing it is open source so you can make patches to improve things! ;) |
| 18:57.44 | *** join/#asterisk vandyk (~vandyk@177.19.184.162) |
| 18:58.05 | WIMPy | Bad thing that it's a little complicated to get them upstream. |
| 19:02.18 | reveal | WIMPy: is it possible to be walked through creating a dialplan to use an agi-test script |
| 19:02.47 | reveal | or is there a way to call it from the CLI |
| 19:02.56 | WIMPy | You just call it via AGI(). That's all. |
| 19:03.15 | reveal | so like AGI() /path/too/agi? |
| 19:03.35 | WIMPy | Yes, just start it. But it probably won't do much sensible as the data from Asterisk is missing. |
| 19:03.48 | WIMPy | If you put it within the brackets, yes. |
| 19:04.13 | reveal | well when i do perl agi-test.agi it runs but produces no output outside the CLI |
| 19:04.36 | WIMPy | Sounds normal. |
| 19:04.48 | WIMPy | Did you try without "perl"? |
| 19:05.07 | reveal | from root prompt it just sits there no output |
| 19:05.41 | WIMPy | So it's running and waiting for input. |
| 19:05.53 | reveal | input from asterisk? |
| 19:06.22 | WIMPy | Yes. If the script is calld from Asterisk, Asterisk will send some data. |
| 19:06.46 | reveal | http://pastebin.com/XQMEM3ds |
| 19:06.47 | newtonr | WIMPy, if you have a way to make it easier, it is always worth bringing it up on the dev list. Obviously that is not a brief or easy conversation, but I think it is always worthwhile as we want things to be as easy as possible within reason. |
| 19:06.50 | reveal | that is agi-test.agi |
| 19:07.44 | WIMPy | newtonr: Having had a look at the relevant part on the wiki was rather discouraging. |
| 19:08.05 | reveal | so i executed the script via ./agi-test.agi and am sitting in asterisk CLI |
| 19:09.48 | WIMPy | I don't speak perl, but pressing return might make it continue. |
| 19:10.27 | reveal | you are correct |
| 19:10.51 | reveal | ================== Complete ====================== |
| 19:10.52 | reveal | 7 tests completed, 0 passed, 7 failed |
| 19:10.52 | reveal | ================================================== |
| 19:10.53 | reveal | but i think in order to test it, it needs to call me right |
| 19:11.37 | WIMPy | It needs to be called from dialplan. |
| 19:12.01 | vandyk | Hi, I have a question, I have this infrastructure |
| 19:12.02 | vandyk | http://ibin.co/2J9yvMfIZhpX |
| 19:12.25 | vandyk | But most of my customers register only via DNS using voip.mydomain.com |
| 19:13.16 | vandyk | when I have issues with this link, I try to ask the customers to register on voip2.mydomain.com or voip3.mydomain.com but they can't make it work correctly |
| 19:13.31 | vandyk | I think that it's related with some firewall rules |
| 19:13.50 | robmal | I think you should use PI addresses and bgp. |
| 19:14.38 | vandyk | just to make sure, if someone connects to Asterisk using the link1, the firewall should maintain the session with the link1, I mean, Asterisk need to send traffic back to the customer by link1. Right? |
| 19:15.29 | newtonr | WIMPy, yeah I think the wiki tries to cover many potential scenarios and provide a lot of information whereas the typical patch submission is be pretty simple. The wiki errors on the side of providing too much information for patch submission I think. |
| 19:15.58 | robmal | Asterisk will send rtp data to the address it recieved in the sip invite, which is sent to the address from which the sip register came. |
| 19:16.07 | robmal | So, yes. |
| 19:16.39 | WIMPy | vandyk: Not neccessarily. Depends on their security settings. |
| 19:17.31 | vandyk | WIMPy: what option of security settings handles this? |
| 19:18.01 | WIMPy | Depends on the used application/device. |
| 19:18.18 | vandyk | this option is not set at my side? |
| 19:18.29 | vandyk | I mean, only at my side |
| 19:18.39 | WIMPy | No. |
| 19:19.19 | vandyk | Let's assume that I have at least 2 different devices: ATA and other Asterisk server |
| 19:19.39 | WIMPy | Generelly it's fine for SIP to send messages from different IPs, but some implementations may choose to accept messages ony from where they expect them. |
| 19:19.42 | vandyk | for the Asterisk server, which setting should I look? |
| 19:20.08 | WIMPy | insecure |
| 19:21.39 | WIMPy | Oh, and you should consider upgrading, I guess. |
| 19:22.01 | *** join/#asterisk jonmasters (~jcm@edison.jonmasters.org) |
| 19:22.55 | robmal | Also, PI addresses and bgp, maybe some round robin dns. |
| 19:23.14 | vandyk | robmal: what is PI addresses? |
| 19:23.24 | vandyk | I work with Amazon Route 53 DNS |
| 19:23.26 | robmal | PI stands for Provider Independent. |
| 19:24.11 | robmal | They're not connected to any ISP, only the organisation who owns them and by using bgp protocol you can announce them through any provider. |
| 19:24.46 | vandyk | humm, this I think that would be more difficult at this time. |
| 19:26.01 | robmal | To be honest afaik IANA doesn't give away any new PI addresses now because they've run out :-) So you'd have to buy or lease it. Times are hard ;-) |
| 19:27.47 | *** join/#asterisk Cyphor (~Cyphor@69.9.239.74) |
| 19:28.56 | vandyk | :-) |
| 19:29.15 | WIMPy | The trouble is that most providers ignore routes to small networks anyway. |
| 19:30.14 | vandyk | but considering that I have the DNS on Amazon, I think that I can do some of DNS round robin there, right? So I could let the customers register only at voip.mydomain.com but at Amazon deal with the IP that points to that |
| 19:30.45 | vandyk | and I need to change all my extensions and trunks to use insecure=port,invite and set them as peer, right? |
| 19:31.15 | WIMPy | No, the others. |
| 19:31.36 | WIMPy | Unless your firewall makes sure you reply from the address they sent to. |
| 19:31.49 | WIMPy | Asteriks doesn't. |
| 19:32.10 | vandyk | what do you mean the others? :-) |
| 19:32.19 | vandyk | I can't have insecure=invite,ports |
| 19:32.22 | vandyk | ? |
| 19:32.24 | WIMPy | The others need the insecure setting. |
| 19:32.38 | vandyk | Ahh OK |
| 19:32.51 | WIMPy | chan_sip doesn't really work with multi-homed systems. |
| 19:33.03 | WIMPy | I don't know how well that works with chan_pjsip. |
| 19:33.53 | vandyk | and to upgrade my server, what is the more stable version to use now? |
| 19:34.17 | WIMPy | So unless you can make sure all your customers have the insecure setting or something equivaletn for what they use, you need som advanced firewall trickery. |
| 19:35.10 | WIMPy | Asterisk 13 has chan_pjsip that might handle the situation in a sensible way. But I don't know. |
| 19:36.48 | vandyk | I see. Well thank you very much WIMPy and robmal |
| 19:37.01 | vandyk | I'll do some planning to make this changes |
| 19:37.53 | robmal | I don't see any reason why your clients couldn't register to voip. and voip2. at the same time. |
| 19:38.07 | robmal | Almost all devices/platforms support that. |
| 19:39.08 | WIMPy | That still doesn't make Asterisk reply from the expected IP. |
| 19:39.37 | robmal | Sure, that's the firewalls job. |
| 19:40.03 | WIMPy | Not really, but you might make it do so. |
| 19:41.15 | vandyk | I tried to configure on Fortigate a rule that stick the client session on same link |
| 19:41.30 | vandyk | but it seems that this not works well with UDP |
| 19:41.53 | vandyk | this is the only explanation to me |
| 19:42.16 | vandyk | and the issue that customers has when this occurs is one way sound |
| 19:42.34 | vandyk | what lead me to some NAT issues |
| 19:42.46 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
| 19:43.03 | WIMPy | Not too bad. At least you got the session set up :-) |
| 19:44.09 | *** join/#asterisk bkruse (~Adium@67.63.143.34) |
| 19:44.32 | WIMPy | The FW needs to replace the IP in the SDP. The famous SIP ALG. for possible side effects consult a glass sphere. |
| 19:47.01 | WIMPy | Or in other words: Your direwall rules for the SIP sessions seem to be working. |
| 19:47.08 | WIMPy | The next stage ist RTP. |
| 19:47.57 | robmal | I think he disabled something important ;-) |
| 19:47.58 | WIMPy | That was the wrong firewall rule. |
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| 20:31.15 | *** join/#asterisk napsi (82b42afa@gateway/web/freenode/ip.130.180.42.250) |
| 20:31.27 | napsi | need some help with |
| 20:31.31 | napsi | asterisk |
| 20:32.38 | napsi | <PROTECTED> |
| 20:32.43 | napsi | <PROTECTED> |
| 20:43.24 | pjensen00 | *return to provider |
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| 21:09.10 | *** join/#asterisk millsu2 (~bgardner@69-195-66-124.unifiedlayer.com) |
| 21:14.30 | millsu2 | I have about 600 sip phones connected to asterisk (freepbx) and had an issue today where about 30 pf them suddenly could not take calls. Asterisk was able to dial them through Dial(SIP/XXXX) but freepbx wouldn't even attempt to dial beacuse it thought the phones were busy. |
| 21:14.51 | millsu2 | It seems freepbx uses devce hints and the hints weren't updating: |
| 21:14.52 | millsu2 | 42802@ext-local : SIP/42802&Custom:DND State:InUse |
| 21:15.31 | WIMPy | #freepbx |
| 21:15.42 | millsu2 | Anyone have any idea why these hints wouldn't update to idle, or how to force the state to update? |
| 21:16.41 | millsu2 | does freepbx decide the state for these hints, or is that something asterisk handles? |
| 21:16.46 | WIMPy | Maybe someone/something set DND? |
| 21:16.53 | WIMPy | Both. |
| 21:17.15 | WIMPy | One each. |
| 21:19.23 | millsu2 | The freepbx featurecode for DND is disabled, but this does seem like it might be related to that. |
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| 22:26.44 | *** join/#asterisk CurtN (51f3d734@gateway/web/freenode/ip.81.243.215.52) |
| 22:26.49 | CurtN | Hi people |
| 22:27.07 | CurtN | anyone here can give me a little help for Asterisk ? :) |
| 22:27.17 | WIMPy | ~ask |
| 22:27.17 | 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. |
| 22:28.03 | CurtN | Well, i am trying to install Asterisk with WebRTC & sipML5. i am following this tutorial http://highsecurity.blogspot.be/2012/12/webrtc-and-asterisk-11-using-sipml5.html |
| 22:28.06 | CurtN | to problem is |
| 22:28.58 | CurtN | at step 30 i get a "Unable to connect" when i try to acces to http://<myIP>/myphone/call.htm |
| 22:29.11 | CurtN | (sorry if my english is bad :) ) |
| 22:30.45 | robmal | It looks like you've failed one of the 29 steps before! |
| 22:30.56 | CurtN | haha :) |
| 22:31.26 | robmal | Also, once one of TheGuys wakes up - go to #freepbx |
| 22:31.29 | CurtN | i've done all steps excepted the one with " [FREEPBX USERS] " |
| 22:31.51 | robmal | It's not an asterisk issue. |
| 22:32.30 | CurtN | Maybe it's a missconfiguration of centos ? |
| 22:32.45 | CurtN | thank you for helping me :) |
| 22:33.08 | robmal | Maybe. I'd love to help but i have no idea about freepbx inner workings so i'm sure you'll get better help on their channel. |
| 22:33.26 | CurtN | ok thank you robmal :) |
| 22:33.32 | CurtN | i'll ask them |
| 22:34.48 | CurtN | but last question, is it a way to do a visioconference (asterisk, webrtc, sipml5) on FreeBSD without FreePBX ? I am a bit new to this ^^ |
| 22:36.47 | robmal | Asterisk supports video in the latest versions so afaik you can make a 2 way call with video, maybe a conference but i haven't had a chance to test it yet. |
| 22:37.37 | CurtN | what does afaik mean ? |
| 22:38.03 | robmal | Abbrevation for: As Far As I Know |
| 22:38.19 | CurtN | oh, thank you robmal ! |
| 22:38.52 | CurtN | i guess i have to change my VM settings or Centos centings |
| 22:39.52 | CurtN | btw, do you know how to make a VM machine ip public, so i can connect to it anywhere ? |
| 22:41.51 | robmal | Well, this is a complicated question... Firstly - vms are threated by asterisk as any other number so yes, any call can be directed to a VM and the VM can be on any asterisk server on public ip. |
| 22:43.20 | robmal | s/threated/treated/; # not my first language |
| 22:44.19 | CurtN | what is s/threated/treated/ for ? |
| 22:44.34 | robmal | Oh. |
| 22:44.40 | robmal | Nevermind that. |
| 22:44.48 | CurtN | ^^ |
| 22:44.48 | robmal | Just... IT stuff. |
| 22:45.19 | CurtN | ok :) |
| 22:45.37 | robmal | K, so what's going on now and what do you want to get? |
| 22:48.55 | CurtN | well i have to make a website which allows users to communicate from all around the world. I have a FreeBSD server and i must use Asterisk integration. I have decide to use webrtc & sipml5 to make it easier. I am only at the installation step, asterisk is up, phones calls : ok but when i try to acces to the default web page wich allows u to register and call my web browser only displays "unable to connect" i dont know if i have faile |
| 22:50.09 | CurtN | moreover, my vm is set on bridged & replication |
| 22:50.20 | *** join/#asterisk ModFather (~johnd@athedsl-145228.home.otenet.gr) |
| 22:50.23 | ModFather | hi all |
| 22:50.41 | CurtN | hi @ModFather |
| 22:52.09 | CurtN | maybe i should set up apache ? |
| 22:54.28 | *** join/#asterisk nbjoerg (~joerg@netbsd/developer/joerg) |
| 22:55.01 | nbjoerg | I need to get a log of the sip exchange from my asterisk with a given peer into a file |
| 22:55.05 | nbjoerg | what's the easiest way? |
| 22:56.27 | WIMPy | wireshark/tshark/tcpdump |
| 22:56.28 | DivideBy0 | usually tcpdump |
| 22:57.24 | DivideBy0 | try tcpdump -n host <ip.ad.re.ss> |
| 23:00.51 | nbjoerg | not an option, since it istls traffic |
| 23:01.19 | nbjoerg | and getting wireshark to decode tls is even more tricky... |
| 23:04.47 | WIMPy | Then there's only one option. |
| 23:08.42 | *** join/#asterisk elguero (~miguel323@2001:470:1f06:12c4::2) |
| 23:10.04 | robmal | Jews? |
| 23:10.42 | WIMPy | They surely sell some appliance to do it. |
| 23:12.00 | robmal | So, once again, it's all because of the jews. |
| 23:12.28 | *** join/#asterisk [NC] (~nc@rv1.sabius.net) |
| 23:13.34 | DivideBy0 | nbjoerg: can you just turn debug on in the CLI? or do you need more info |
| 23:14.52 | nbjoerg | DivideBy0: that's lots of data scrolling by |
| 23:15.11 | robmal | nbjoerg: /etc/asterisk/logger.conf has some clues, set messages=>(whatever),all |
| 23:15.45 | robmal | Or if the live log is enough, there're clues there too (too lay to check) |
| 23:19.52 | nbjoerg | robmal: even if I add an entry for level debug, sip debug doesn't seem to end up there? |
| 23:20.50 | nbjoerg | hm. some other level |
| 23:20.53 | nbjoerg | verbose maybe? |
| 23:20.53 | robmal | Try sip set debug peer {whatever} |
| 23:21.01 | robmal | core set verbose 10 |
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| 23:22.28 | robmal | All hail SuperMario. |
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| 23:41.19 | ModFather | hi [TK]D-Fender |
| 23:41.24 | ModFather | hi robmal |
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