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| 05:49.51 | rock1979 | hello |
| 05:50.36 | rock1979 | does any one know a documentation how to install spandsp on asterisk 11 |
| 05:50.51 | rock1979 | i mean uptodate document |
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| 06:24.45 | rock1979 | hello shido6 |
| 06:24.50 | shido6 | hello |
| 06:25.03 | rock1979 | i got the link that you send me yesterday |
| 06:25.20 | shido6 | please refresh my memory |
| 06:25.22 | shido6 | :) |
| 06:25.34 | rock1979 | T38 faxing |
| 06:25.46 | rock1979 | from sip to dahdi |
| 06:25.46 | shido6 | oh yes - the training |
| 06:26.11 | shido6 | did the image help? |
| 06:26.15 | rock1979 | yes |
| 06:26.16 | rock1979 | but |
| 06:26.20 | rock1979 | the problem starts |
| 06:26.29 | rock1979 | with the invite to T38 |
| 06:26.42 | rock1979 | asterisk sends a reply with sip 200 ok |
| 06:26.49 | rock1979 | but in sdp there is G711 |
| 06:27.41 | rock1979 | so basicly if asterisk decidec that it can do faxing with T38 then in sip 200 ok message there should be T38 in sdo |
| 06:27.43 | rock1979 | so basicly if asterisk decidec that it can do faxing with T38 then in sip 200 ok message there should be T38 in sdp |
| 06:28.00 | rock1979 | am i right ? |
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| 06:35.15 | bulkorok | rock1979: in theory asterisk should send a T38 re-invite if it recognizes a CNG tone from the other side... |
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| 06:40.45 | rock1979 | yes but in my case t38 re-invite is coming from my customer |
| 06:41.01 | bulkorok | yeah... and?! what is happening at your side then!? |
| 06:41.57 | ChannelZ | debauchery and mayhem |
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| 06:44.13 | rock1979 | on my side asterisk is sending sip 200 ok with g711 in sdp |
| 06:47.41 | bulkorok | rock1979: please paste a full trace |
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| 06:58.54 | rock1979 | https://www.sharefest.me/b43b5244966522dc76acd1732b2cb9c9 |
| 07:00.17 | rock1979 | bulkorok i shared |
| 07:10.15 | rock1979 | bulkorok do you have any idea what was happened in the traces |
| 07:10.35 | bulkorok | i dont get the file |
| 07:11.23 | rock1979 | sorry ? |
| 07:15.14 | rock1979 | what did you mean by "i dont get the file" |
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| 07:18.09 | bulkorok | rock1979: I see nothing... |
| 07:18.32 | rock1979 | i shared the link here |
| 07:18.41 | rock1979 | <PROTECTED> |
| 07:19.03 | bulkorok | yeah... nothing happens when I click it... |
| 07:19.08 | rock1979 | copy and paste it in your browser |
| 07:19.18 | bulkorok | Connected to 0 peers |
| 07:21.12 | rock1979 | save as screen should pop up |
| 07:21.21 | bulkorok | firewall... |
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| 07:22.15 | rock1979 | i sent you the file from irc |
| 07:22.27 | bulkorok | yeah... firewall blocks it... |
| 07:23.00 | rock1979 | what about the link doesnt it open a pop up save as screen ? |
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| 07:26.13 | Stefan27 | if i send BridgeInfo action through AMI with <id> at time T_1, I will eventually get a sequence of BridgeInfoChannel events as response, and then a BridgeInfoComplete at time T_2. What would happen if at time t with T_1<t<T_2 some channel leaves bridge <id>. Will asterisk deliver a BridgeLeave event before it delivers BridgeInfoComplete or after? |
| 07:27.04 | bulkorok | rock1979: nope |
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| 12:41.21 | MasterChen | good morning |
| 12:41.54 | matthew-moretalk | morning! |
| 12:42.53 | MasterChen | oh cool someone is awake! |
| 12:43.07 | MasterChen | the internet has no timezobe. kol |
| 12:43.16 | MasterChen | lol* |
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| 12:48.54 | matthew-moretalk | haha well its 1:49pm here! :) |
| 12:56.11 | MasterChen | ah cool |
| 12:56.32 | MasterChen | dping a pbx install today for my new job. |
| 12:56.51 | MasterChen | figured i'd hang out here to get in the spirit |
| 12:57.07 | matthew-moretalk | haha well welcome! |
| 12:57.21 | MasterChen | ty |
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| 12:57.53 | MasterChen | most of my work has been software based asterisk. learning the hardware side will be interesting |
| 12:58.34 | matthew-moretalk | not that much to it if your not doing PSTN work |
| 12:58.39 | matthew-moretalk | only sip |
| 12:59.06 | MasterChen | ah |
| 12:59.51 | matthew-moretalk | what tech are you needing to use? |
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| 13:12.10 | MasterChen | othing specific |
| 13:12.21 | MasterChen | i am shadoeing today on a pbx install |
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| 13:12.44 | MasterChen | really jt's just a linux box in a rack that just got built. |
| 13:12.51 | matthew-moretalk | fair enough |
| 13:13.12 | MasterChen | still, I've rrally only dealt with config files and cli |
| 13:13.22 | *** join/#asterisk aurs (~aurs@84.48.58.187) |
| 13:13.29 | MasterChen | nothing like T1 and pstn integration |
| 13:13.50 | MasterChen | all software and dialplan |
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| 13:33.57 | *** join/#asterisk ruied (~ruied@25.172.37.188.rev.vodafone.pt) |
| 13:37.28 | ruied | Hello. I need to get StartRetry or EndRetry or pid call id asterisk variables. Is there any way I can get those values at the dialplan? |
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| 13:50.54 | ruied | I'm using asterisk auto-dial .call to dial "person1" files and I want to track after the 3th missed call the dialplan check that and dial to "person2" Supervisor saying that person1 didn't answered the phone |
| 13:51.30 | ruied | is there a way to get the current call pid variable in the dialplan? |
| 13:53.43 | [TK]D-Fender | You should do your own retry by using a Local channel as your "Channel:" to process it. |
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| 14:14.37 | *** join/#asterisk tinklebear (~tinklebea@cpe-066-057-253-171.nc.res.rr.com) |
| 14:14.54 | tinklebear | Anyone know of a way to auto-dial and send text to speech messages? |
| 14:15.30 | [TK]D-Fender | tinklebear: Originate / Call-file + a TTS app like festival, etc |
| 14:15.33 | matthew-moretalk | im personally just playing with a service for that messagebird.com |
| 14:15.33 | tinklebear | I was looking for an AMI example using google text-to-speech |
| 14:15.58 | [TK]D-Fender | AMI doesn't do TTS. It's only role would be to Originate the outbound call. |
| 14:16.02 | [TK]D-Fender | its* |
| 14:16.37 | ruied | [TK]D-Fender, going to take a look at that... |
| 14:16.38 | tinklebear | D-Fender do you know where I could find some good examples of this? |
| 14:17.19 | tinklebear | So the call file would be dropped in the outgoing spool directory? |
| 14:17.33 | [TK]D-Fender | yes. This is all well documented |
| 14:17.47 | tinklebear | I've done that before but never with any tts |
| 14:18.29 | tinklebear | hmm...so no way to programatically do it through ami without moving files around |
| 14:18.34 | [TK]D-Fender | those 2 parts don't have anything specific to do with one another |
| 14:18.43 | [TK]D-Fender | AMI Originate <- |
| 14:19.19 | file | moo |
| 14:19.31 | tinklebear | I was hoping to not have to build any server on the asterisk to listen for when I wanted to make a call and what to do. |
| 14:20.13 | [TK]D-Fender | tinklebear: Something has to tell it to start launching those calls. |
| 14:20.32 | tinklebear | that's why I was hoping ami could do it |
| 14:20.37 | [TK]D-Fender | tinklebear: AMI/CLI Originate, or Call Files. Take your pick. |
| 14:20.38 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 14:20.55 | [TK]D-Fender | And you CAN use AMI to launch the call. I've told you this 3 times now. |
| 14:20.55 | tinklebear | so it can be done with just ami? |
| 14:21.32 | [TK]D-Fender | timahvo1: AMI does not PROCESS calls. your TTS is still * dialplan |
| 14:21.56 | tinklebear | I've looked for some examples but this is the closest I've found: http://www.voip-info.org/wiki/view/Asterisk+manager+dialout |
| 14:22.48 | tinklebear | So I would need to feed the ami the text |
| 14:23.09 | [TK]D-Fender | tinklebear: that's the idea. |
| 14:23.43 | tinklebear | ...the world for an example |
| 14:23.56 | *** join/#asterisk gugaua (5e66624d@gateway/web/freenode/ip.94.102.98.77) |
| 14:24.41 | gugaua | Hello, I want every number that starts with a '0' to be changed |
| 14:24.46 | *** join/#asterisk _0x5eb_ (~seb@seb-hpws2.w1.tele.crt1.net) |
| 14:24.47 | gugaua | exten => _0!,n,Set(CALLERID(num)=${CALLERID(num):1}) |
| 14:24.59 | gugaua | but it is not working.... |
| 14:25.35 | *** join/#asterisk fstd_ (~fstd@unaffiliated/fisted) |
| 14:25.35 | [TK]D-Fender | gugaua: Show us. |
| 14:25.55 | [TK]D-Fender | GugAnd realize you didn't WHAT number you are looking to change. |
| 14:25.55 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-dpnqzsrmejfwgtzn) |
| 14:25.55 | [TK]D-Fender | gugaua: And realize you didn't WHAT number you are looking to change. |
| 14:26.11 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 14:26.47 | gugaua | the original number would be 1234 1234567 which is shown as 01234 1234567 |
| 14:27.06 | ruied | [TK]D-Fender, So, the idea of the Local Channel is to make the .call file just running once (MaxRetries=0) and do the retry calls at the dialplan, correct? |
| 14:27.11 | [TK]D-Fender | gugaua: You keep saying n\"number" and being clear onw WHAT KIND of number. |
| 14:27.30 | [TK]D-Fender | ruied: that is what I said. |
| 14:28.10 | gugaua | [TK]D-Fender: what do you mean by that? |
| 14:28.37 | [TK]D-Fender | gugaua: I can have a SIP DEVICE that is numeric : [123] |
| 14:29.01 | gugaua | Oh its from ISDN |
| 14:29.04 | gugaua | dahdi |
| 14:29.14 | [TK]D-Fender | gugaua: I could have a DIALED number that is numeric. I could have a CALLERID that is numeric. EVERYTHING could be a number in Asterisk |
| 14:29.30 | [TK]D-Fender | gugaua: MULTIPLE kinds of numbers come from DAHDI. |
| 14:29.49 | [TK]D-Fender | gugaua: you get a CALLING number and a CALLED number for an inbound call. |
| 14:29.55 | [TK]D-Fender | gugaua: that is still 2 differnt kinds. |
| 14:30.08 | [TK]D-Fender | gugaua: And you aren't being clear about which one you want to "change" and why |
| 14:30.33 | *** join/#asterisk Mango45 (~Mango45@d209-89-208-218.abhsia.telus.net) |
| 14:31.04 | gugaua | [2014-08-27 16:20:49] VERBOSE[14180][C-00000001] pbx.c: -- Executing [h@macro-dial:1] Macro("DAHDI/i1/0123412345678-2", "hangupcall") in new stack |
| 14:31.42 | asterfix | hi there |
| 14:31.54 | gugaua | I want to change the number which is incomming through dahdi(ISDN), this number has got a 0 too much and needs to be changed on every call |
| 14:32.08 | gugaua | outgoing calls are not a problem |
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| 14:32.20 | gugaua | incoming are shown in a wrong way |
| 14:33.25 | gugaua | [TK]D-Fender: CALLERID(num) needs to be changed |
| 14:33.36 | tinklebear | D-Fender...thanks |
| 14:33.40 | tinklebear | I think I found the example |
| 14:33.43 | [TK]D-Fender | gugaua: so match the pattern starting with 0 and Goto a context and exten trimming off the first digit |
| 14:33.52 | tinklebear | http://lists.digium.com/pipermail/asterisk-users/2009-December/242511.html |
| 14:34.29 | [TK]D-Fender | [10:24]gugauaexten => _0!,n,Set(CALLERID(num)=${CALLERID(num):1}) <- this changeds the CALLERID. This not the number they DIALED. You need to GOTO another exten. So do you Goto trimming the EXTEN |
| 14:36.49 | gugaua | [TK]D-Fender: just that I get sure you understand me because of my english.... The incomming call via Dahdi(ISDN) cannot be changed at this line? |
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| 14:37.42 | [TK]D-Fender | gugaua: You see to fail to understand the concept that when a call comes in they are both DIALING a number.. nad have a CALLERID |
| 14:38.07 | [TK]D-Fender | MY phone number is 12345 and I am DIALING 06789 to get to you |
| 14:38.33 | [TK]D-Fender | gugaua: that code you tried showing us looks at the number that was DIALED.... and then changes the CALLERIR. |
| 14:38.39 | [TK]D-Fender | ]It does not change what is being DIALED |
| 14:40.04 | asterfix | I use a Wildcard x101p with dahdi 2.10.0 and asterisk 11.12.0, and the config seems ok BUT when someone dials in, asterisk shows no reaction. no ring, nothing. any hints? |
| 14:40.56 | [TK]D-Fender | asterfix: Show us that dahdi is configured and loaded and your channels acknowledged |
| 14:42.12 | gugaua | is that because of the n? |
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| 14:42.55 | gugaua | If I change that line to exten => _.,1,Set(CALLERID(num)=${CALLERID(num):1}) every call missed the first digit |
| 14:43.21 | [TK]D-Fender | gugaua: No, You want to change the EXTENSION and you are changing the CALLERID in there. If you can't understand the difference between the calling number and the called number then we cannot help you. |
| 14:43.39 | asterfix | TK D-Fender: http://pastebin.com/raw.php?i=aMWtdPrP |
| 14:45.27 | [TK]D-Fender | gugaua: that last line matches anything dialed with one or more characters, then strips the first charater off of the caller ID number. |
| 14:47.54 | [TK]D-Fender | asterfix: Ok, that looks pretty good right now... ramp up core debug and see if anything shows |
| 14:49.39 | gugaua | [TK]D-Fender: okay, I edited the extension so it will take only external calls, now you told me I have to edit EXTENSION to change the callerID there right? |
| 14:50.21 | asterfix | TK D-Fender: nothing, even with the highest debug level. |
| 14:51.27 | [TK]D-Fender | asterfix: Could be a crapped out card. have you tested the line itself? |
| 14:51.56 | asterfix | yes. the phone behind it rings and works properly |
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| 14:52.07 | [TK]D-Fender | gugaua: You are still having issue confirming what you are trying to do . Be SPECIFIC about exactly which number you are trying to modify and where you "see" it. |
| 14:52.09 | asterfix | the card worked fine for years, I just changed pc |
| 14:52.22 | System123 | Is there a way to filter AMI events only to the events related to a specific context? |
| 14:52.45 | [TK]D-Fender | System123: nope |
| 14:53.02 | [TK]D-Fender | System123: unless you are tagging sutom evens |
| 14:53.05 | [TK]D-Fender | custom |
| 14:53.18 | System123 | I am using adhearsion and only want the events related to adhearsion, not all the other AMI events. |
| 14:53.53 | [TK]D-Fender | System123: Highly doubtful... |
| 14:54.09 | System123 | Bummer, would have been nice. |
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| 14:57.19 | gugaua | [TK]D-Fender: Okay I will try to explain it again as good as possible, my Asterisk is connected as a phone to my main telephone system via ISDN, asterisk has got it´s own phone extension 66, when an external call comes in from ISDN like 12341234567 (not from sip) an additional digit (0) is shown at my sip phone (sip phone number 001 for example) which is unnecessary but If I call with my Sip phone to an external number everything is |
| 14:58.19 | [TK]D-Fender | gugaua: the number you see when you dial another phone is the CALLERID of the originating channel. this is not the number that channel DIALED IN. |
| 15:00.24 | gugaua | [TK]D-Fender: and what is the number called that a channel dialed in? |
| 15:01.45 | gugaua | away for 20 min |
| 15:02.29 | asterfix | TK D-Fender: do you have any suggestions on that? |
| 15:05.35 | Mango45 | I'm using WaitForNoise and WaitForSilence to determine (1) whether or not a caller has started to speak, and (2) if they did, when they stop speaking. |
| 15:05.59 | Mango45 | Any suggestions on playing a comfort noise file during this? I tried to use BackgroundDetect, but there seems to be no way to detect if the caller never speaks - it just waits forever. |
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| 15:25.42 | asterfix | TK D-Fender: changing the slots didn't solve the problem |
| 15:25.42 | pjensen00 | Hrm. But can't you hang up every single channel currently up in asterisk with the ARI? It looks like you can arbitrarily manipulate channels via the example at the bottom of the page here https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=29395573 |
| 15:25.43 | file | some stuff doesn't stop you, but the intention is not arbitrary control |
| 15:25.43 | pjensen00 | I see, thanks. |
| 15:25.43 | [TK]D-Fender | asterfix: test the card with another machine to see if it's simply died somewhere in the transition |
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| 15:45.55 | Mango45 | [TK]D-Fender: I have a comfort noise file that I created. Basically I want to use Playback to play it, while the other dial plan logic is going on. Possible? |
| 15:46.17 | [TK]D-Fender | Mango45: Like? |
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| 15:49.33 | Katty | hello my asterisk does not work at all how to fix plz is urgent thx |
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| 16:00.33 | posixninja | what's the difference between asterisk 11 and asterisk 1.8? |
| 16:00.46 | Mango45 | [TK]D-Fender: I'm using WaitForNoise and WaitForSilence to determine (1) whether or not a caller has started to speak. |
| 16:01.01 | Mango45 | However, since the caller hears nothing, they think the call has been disconnected. |
| 16:01.28 | Mango45 | Therefore, I want to play a file in the background. |
| 16:01.44 | [TK]D-Fender | posixninja: 9.2 |
| 16:01.49 | Chainsaw | posixninja: Asterisk 11 has much improved T38 (fax bypass) support compared to 1.8 |
| 16:01.51 | Mango45 | :D |
| 16:02.00 | posixninja | ;P |
| 16:02.20 | Chainsaw | posixninja: And since both are long-term support branches, I would recommend 11 for new deployments. |
| 16:02.31 | posixninja | ok, but configuration is essentially the same |
| 16:02.50 | Chainsaw | posixninja: Correct. |
| 16:03.13 | Chainsaw | posixninja: The gigantic jump in version numbers is more of a marketing thing if you ask me. |
| 16:03.29 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-qquymfhrqnqubabp) |
| 16:03.38 | posixninja | just curious all the documentation I read referred to asterisk11 and when I checked the version ubuntu installed and saw 1.8 I was really confused |
| 16:03.57 | Chainsaw | posixninja: Seems a bit lazy if they still haven't packaged 11 by now. |
| 16:04.46 | newtonr | posixninja, https://wiki.asterisk.org/wiki/display/AST/New+in+11 |
| 16:05.53 | newtonr | https://wiki.asterisk.org/wiki/display/AST/Call+Identifier+Logging is reason enough to use 11 over 1.8 :) |
| 16:06.33 | Chainsaw | And aside from that one "security fix" that didn't work for SIP-over-TCP or TLS, the 11 branch has been very good to me. |
| 16:06.52 | [TK]D-Fender | 11 has T.38 TERMINATION, as well as hangup & predial handlers, CEL, and tons more. you skipped the fact there another entire branch between them. |
| 16:07.14 | [TK]D-Fender | And * 12 is released with 13 on the way |
| 16:08.26 | Chainsaw | "Much improved" T38, yes. |
| 16:08.33 | Chainsaw | And using 12 in production seems ill advised. |
| 16:08.41 | newtonr | posixninja, it is good to consult the support lifetimes of the various versions: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 16:09.09 | posixninja | I was just being lazy by not compiling from source myself ;-P |
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| 16:53.33 | file | there are people with 12 in production, just as an fyi |
| 16:53.43 | file | but they understand what it is. |
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| 17:34.49 | Katty | blames file |
| 17:35.52 | cusco | blames society |
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| 17:37.48 | carrar | moo |
| 17:40.09 | Nugget | ]:8) Moo |
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| 17:46.15 | slylock99 | hi guys |
| 17:46.30 | p00h | hi there) |
| 17:48.03 | slylock99 | I am trying to modify a dialplan in asterisk |
| 17:48.14 | cusco | moo |
| 17:54.49 | Katty | nugget! |
| 17:54.51 | Katty | hugs Nugget |
| 17:55.22 | Nugget | huggles Katty |
| 17:55.57 | Katty | congrats on your 25+ mile run |
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| 17:56.34 | Nugget | it was nice to have some hills for a change |
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| 17:57.01 | Katty | shakes head |
| 17:57.08 | Katty | hills and nice should not be in the same sentance |
| 17:57.37 | WIMPy | If you just look at them? |
| 17:58.11 | Katty | maybe. |
| 17:58.52 | Nugget | My 50 mile race is in five weeks |
| 17:59.13 | carrar | running should only be required when the spanish inquisition is after you |
| 17:59.58 | WIMPy | thinks it would be appropriate if someone wants to use SIP as well. |
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| 18:03.50 | slylock99 | guys i am trying to write a dialplan |
| 18:04.03 | Katty | Nugget: i see your training hard for the zombies |
| 18:04.07 | slylock99 | problem is that it keeps repearing itself |
| 18:04.17 | slylock99 | *repeating |
| 18:04.24 | Katty | carrar: but you have to be fit enough to run from the spanish inquisition |
| 18:04.28 | slylock99 | i want it to repeat 3 times |
| 18:04.33 | slylock99 | and then quit |
| 18:04.33 | Katty | carrar: you never know when the French Banana War might be on your doorstep |
| 18:04.40 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 18:05.50 | slylock99 | can anyone help me with that |
| 18:08.06 | marceloamorim | guys, there is any explanation about receive audio and transmit audio when I receive a call, but when I do a call I don`t received the audio |
| 18:09.51 | [TK]D-Fender | slylock99: Set a counter. Do your task. increase the counter. Has counter reached your max? Yes? Stop doing it. No? go back and do again. |
| 18:13.23 | marceloamorim | I make a tunnel and I have route for my network, this is my sip.conf http://pastebin.com/PdVc7geT |
| 18:14.40 | [TK]D-Fender | marceloamorim: configs don't tell us what's actually happening. We'd need to se a call with SIP debug enabled.... |
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| 18:21.57 | slylock99 | [TK]D-Fender: could you show me an example |
| 18:23.21 | [TK]D-Fender | slylock99: Set a variable to 0 or 1.. do your task. Increment that varialble. |
| 18:23.32 | [TK]D-Fender | skyShow us your attempt at these first 2 parts |
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| 18:27.58 | navaismo | http://www.voip-info.org/wiki/view/Asterisk+cmd+While |
| 18:28.06 | navaismo | ups sorry TK i saw too late your last msg |
| 18:34.44 | slylock99 | [TK]D-Fender: let me try this |
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| 18:42.59 | marceloamorim | [TK]D-Fender: I tried to get a call, but I`m not used to check this situation, so I don`t know if what I got is the right thing |
| 18:43.12 | carrar | heh |
| 18:44.06 | marceloamorim | http://pastebin.com/QTCuz0d2 |
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| 18:45.57 | [TK]D-Fender | marceloamorim: We clearly don't see the beginning or the end of the call. |
| 18:47.01 | *** join/#asterisk niekniek (~niek@77-172-219-210.ip.telfort.nl) |
| 18:50.33 | marceloamorim | I`ll ask for another call then |
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| 19:00.10 | marceloamorim | [TK]D-Fender: now I`m pretty sure that what I`ll post to you is the call, I opened 2 windows to see |
| 19:00.26 | marceloamorim | I set verbose to 10 and I set debug on the peer |
| 19:01.46 | marceloamorim | http://pastebin.com/1PXrmF51 |
| 19:06.52 | [TK]D-Fender | 2014-08-27 15:56:17] VERBOSE[4854][C-00000001] pbx.c: [2014-08-27 15:56:17] -- Executing [032818357@lvlDDD_star:1] NoOp("SIP/hoc2-00000002", ""Ligando para FIXO, 032818357"") in new stack <- we don't see the INVITE that started this channel |
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| 19:07.34 | marceloamorim | how can I debug this? |
| 19:08.20 | marceloamorim | because this is all I got on that call with set verbose 10 and sip set debug peer |
| 19:08.32 | [TK]D-Fender | Where do YOU see the invite coming in? |
| 19:08.42 | [TK]D-Fender | Evidently you're looking at the wrong thing. |
| 19:08.51 | [TK]D-Fender | Stop being so specific and enable GLOBAL SIp debug |
| 19:10.23 | marceloamorim | ok |
| 19:10.59 | marceloamorim | as I said, I`m not used to see this situation, I even don`t know what to see |
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| 19:27.30 | marceloamorim | this is a lot of informations |
| 19:27.31 | marceloamorim | http://pastebin.com/weywxZ8k |
| 19:30.02 | pjensen00 | One more question about ARI. Is it required to have Stasis hand control of a channel over to an ARI client for the client to be able to send messages over the websocket? |
| 19:30.26 | pjensen00 | Or can I just connect to the socket and start firing commands |
| 19:34.07 | marceloamorim | now I got it why I didn`t paste the right debug on sip, its because it wasn`t a call on that peer, they called with another peer. sorry about that |
| 19:37.54 | *** join/#asterisk dms1 (~dms1@65.207.151.254) |
| 19:38.02 | marceloamorim | [2014-08-27 16:23:08] VERBOSE[4927][C-00000009] res_rtp_asterisk.c: [2014-08-27 16:23:08] > 0x2d28130 -- Probation passed - setting RTP source address to 192.168.5.114:10108 this is the line when the audio start to receive, I think |
| 19:42.11 | file | pjensen00, the websocket does not accept commands - it's strictly for events from Asterisk |
| 19:42.35 | pjensen00 | ah, thanks for clearing that up |
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| 19:51.24 | [TK]D-Fender | marceloamorim: check your firewalls on all systems, and your sound device on the other end |
| 19:53.28 | marceloamorim | iptables is open |
| 19:54.02 | marceloamorim | in theory, there isn`t problem on the system right? |
| 19:54.22 | [TK]D-Fender | Asterisk itself looks fine |
| 19:55.19 | marceloamorim | I'll check one more time the routers |
| 19:55.39 | Katty | for the love of all that is chocolate, can the day possible drag any more slowly |
| 19:56.52 | marceloamorim | [TK]D-Fender: if you know any book that would help me to check those debugs, I really don`t know what to look =( |
| 19:57.01 | [TK]D-Fender | re-initializes his time-dilation matrix... |
| 19:57.14 | [TK]D-Fender | MarcNothing in th ebook.. this is outside of * at this point |
| 19:57.17 | [TK]D-Fender | check your CLIENT side |
| 19:57.52 | [TK]D-Fender | Sound card, etc |
| 19:57.53 | [TK]D-Fender | and any networking in between |
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| 20:01.13 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 20:22.56 | litn | hey guys, what's the best way to send rtp/sip traffic to another server just to capture for logs? |
| 20:24.38 | sekil | homer |
| 20:26.22 | pabelanger | I don't think homer does RTP |
| 20:26.39 | sekil | oh yeah...it doesn't |
| 20:26.58 | sekil | sip will do great |
| 20:30.11 | litn | hmm |
| 20:30.40 | litn | no other solutions you guys know about? |
| 20:37.40 | *** part/#asterisk sekil (~Ognjen@78.24.104.82) |
| 20:37.41 | pabelanger | You'd have to use rtpproxy to capture rtp into pcap files |
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| 20:49.10 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 20:50.26 | mjordan | litn: do you really want the RTP, or do you want the call quality statistics (RTCP)? |
| 20:53.25 | litn | sometimes we have weird situations, for example, calls that come in with no audio |
| 20:53.34 | litn | and I feel like I would need to have the RTP traffic for debugging |
| 20:53.35 | litn | no? |
| 20:54.05 | mjordan | if you have no audio, your RTCP statistics would tell you that |
| 21:02.48 | mjordan | <PROTECTED> |
| 21:02.50 | mjordan | http://svn.asterisk.org/svn/asterisk/tags/12.5.0/CHANGES |
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| 21:25.34 | pa | WIMPy, you around? |
| 21:26.08 | pa | WIMPy, i contacted cologne, and got the patch for 3.13 to add the loopback device to mISDN2. going 2 try it. do you want to have it too? |
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| 21:32.35 | sip | doot |
| 21:33.18 | [TK]D-Fender | And the coloured girls say... |
| 21:33.40 | sip | ouch! |
| 21:33.56 | sip | touche |
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| 23:21.38 | tinklebear | Does "core reload" reload the extensions.conf? |
| 23:26.00 | [TK]D-Fender | you would see it if it did |
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