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| 00:46.24 | dasjoe | Hi, I'm having an issue with my dialplan. My phone dials its contacts with an international prefix (+49) at all times, but I can't match that pattern: http://paste.debian.net/379069/ |
| 00:49.26 | ChannelZ | Your NoOp seems to be priority 2 |
| 00:50.13 | ChannelZ | Unless there is a priority 1 of that extension we're not seeing, there is effectively no entry point to that extension by the test call |
| 00:53.10 | dasjoe | ChannelZ: oh wow, that was it. Thanks! |
| 00:53.30 | ChannelZ | sure |
| 00:55.05 | dasjoe | So much for copy-pasting from other people's dialplans |
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| 01:08.18 | ChannelZ | That never ends well |
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| 02:43.31 | ChannelZ | Grrph. I have a sip peer with dtmfmode=inband but when I Monitor() a call it makes, the resulting file has the DTMF muted (I hear tiny little fragments of them where it misses.) How can I get the raw channel audio? |
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| 08:11.27 | andycol_500 | Hi All |
| 08:11.38 | Kunsi | hi |
| 08:11.39 | andycol_500 | is there anyway to display the internal callerid during a blind transfer |
| 08:12.13 | andycol_500 | currently if a call comes from outside and the switchboard answers then does a blind transfer it shows the external callerid not the internal |
| 08:12.26 | andycol_500 | is there anyway to declare the internal with some variable |
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| 08:51.53 | snadge | with res_config_mysql .. how do you force it to reconnect to the database? |
| 08:52.01 | snadge | with cdr_mysql .. you can just reload the module |
| 08:52.34 | snadge | if you kill the database connection from the db itself.. it will reconnect to the newly configured database.. but that seems a bit rude |
| 08:55.20 | andycol_500 | module reload |
| 08:55.54 | snadge | yeah i just tried that |
| 08:56.11 | snadge | but if you module reload res_config_mysql .. it doesn't actually reconnect to the database |
| 08:56.15 | snadge | odd |
| 08:56.39 | snadge | might be dns lookup related |
| 08:57.01 | snadge | eg.. it already goes "hey, i've already got an open connection to x.y.z .. i dont need to reconnect" |
| 08:57.13 | snadge | but if you close the connection from the server.. then it reconnects to the new ip |
| 08:57.54 | snadge | yet it works for the cdr backend |
| 08:58.55 | snadge | unload/load might do it.. but im not game enough to try that |
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| 09:07.41 | andycol_500 | is there anyway to display the internal callerid during a blind transfer |
| 09:07.50 | andycol_500 | currently if a call comes from outside and the switchboard answers then does a blind transfer it shows the external callerid not the internal |
| 09:07.56 | andycol_500 | is there anyway to declare the internal with some variable |
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| 10:13.53 | ChannelZ | Why would you not want the called party to know who is calling them? |
| 10:15.18 | resist0r | he wants to uniquely identify the ext responsible for the blind ransfer |
| 10:15.26 | resist0r | I think? |
| 10:16.16 | ChannelZ | Could just do an attended transfer but complete the transfer immediately without waiting for an answer |
| 10:18.39 | resist0r | I am guessing he wants to declare the internal as a variable or something so it can be used dynamically elsewhere |
| 10:19.02 | snadge | when you execute core stop gracefully .. you cant execute core show calls, or sip show channels etc.. thats a bit sucky |
| 10:19.26 | snadge | im supposed to just trust that asterisk will exit when its finished all of the calls, like the documentation states |
| 10:19.41 | resist0r | while you are waiting for the conditions to be met which makes it gracefully |
| 10:20.22 | snadge | yeah but i wouldn't mind knowing the channel count |
| 10:20.43 | resist0r | can you log into the console twice maybe? |
| 10:20.55 | snadge | i like your thinking |
| 10:20.56 | resist0r | would that allow you to run them while waiting? |
| 10:21.06 | resist0r | the way you want |
| 10:21.24 | resist0r | I believe if I understand what you are saying corectly that might be a solution |
| 10:21.25 | snadge | yeah.. 5 calls remaning |
| 10:23.43 | andycol_500 | yes the reason i need the variable is so i can create a macro so that if the call is not answered it will return to the person who dialed |
| 10:24.21 | andycol_500 | so switchboard transfers to ext 201 and if he doesnt answer it comes back to her |
| 10:26.05 | *** join/#asterisk bounceman (~bounceman@185.32.9.250) |
| 10:26.14 | bounceman | Howdy, I have the following in my file.call Context: dialer |
| 10:26.37 | bounceman | however, it does not seem to hit that context at all. It does not output anything in the CLI it just dials |
| 10:28.00 | bounceman | hmm, wierd. When I run dialplan show I see a bunch of demo contextes |
| 10:28.22 | resist0r | have you core reload |
| 10:30.57 | bounceman | yeah, it looks like it loads extensions.lua |
| 10:31.01 | bounceman | and not extensions.conf |
| 10:31.28 | bounceman | asterisk.ael |
| 10:31.28 | bounceman | * |
| 10:32.20 | resist0r | andycol_500: why do you need the call to be returned to the same switchboard operator whom initiated the transfer? I am asking because it seems that the call should get put back into the que of inbound calls tobe sent to an avilable operator because what if the original transfering operator recieves a new inbound call while the call they just released is on the way back? |
| 10:35.10 | bounceman | Why would it load extensions.ael instead of extensions.conf, I have not told it to do so |
| 10:44.08 | resist0r | bounceman: http://lists.digium.com/pipermail/asterisk-users/2012-January/269184.html |
| 10:46.04 | bounceman | Ok, I've never have do to do that before though :p |
| 10:46.50 | ChannelZ | If module autoloading is on, a lot of times they will load if a module's config file exists. |
| 10:47.03 | ChannelZ | If you're not using the lua or ael extensions, don't have config files sitting there for them. |
| 10:48.25 | bounceman | mmkay, make sense. I now remember I removed some of them on previous setups |
| 10:48.42 | bounceman | Anyhow, please see this. http://pastebin.com/3ZprSZjQ when using .call files it does not go through the dialer context |
| 10:51.48 | andycol_500 | unfortunately a lot of customers ask for the call to go back to the person who did the original transfer |
| 10:51.54 | ChannelZ | Nothing is going through the dialplan |
| 10:53.12 | bounceman | ChannelZ: I cannot push the call through the dialplan? |
| 10:53.20 | ChannelZ | It's making a SIP channel and if it answers, running Playback(hello-world) |
| 10:53.30 | ChannelZ | You can |
| 10:53.38 | ChannelZ | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Call+Files |
| 10:54.06 | ChannelZ | You need to use Extension: and Priority: instead of Application: |
| 10:54.17 | bounceman | okay, I'll try |
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| 10:59.25 | ChannelZ | goes to bed |
| 11:00.02 | bounceman | looks at clock |
| 11:00.25 | ChannelZ | 4am here |
| 11:02.24 | andycol_500 | anyone have any ideas i have managed to get it to sort of work with Set(DN=${SIP_HEADER(TO):5}) Set(DN=${CUT(DN,\,,3)}) |
| 11:02.26 | andycol_500 | but then still shows |
| 11:02.35 | andycol_500 | DN=<sip:300 |
| 11:02.46 | andycol_500 | need just the 300 |
| 11:05.06 | andycol_500 | got it working :) |
| 11:06.27 | andycol_500 | actuallu nope doesnt work for external calls :( |
| 11:11.44 | bounceman | ChannelZ: I tried this http://pastebin.com/E8j1G5nk in my .call file but still it does not hit exten => 5000.... |
| 11:12.44 | resist0r | bounceman: are you watching the console with -vvvvvvvvvvv to see what is happening? |
| 11:14.06 | bounceman | no, tripple v |
| 11:14.38 | resist0r | I forget how many it can accept but I know it wont complain if too many are there so to get the most verbose output I over do it |
| 11:15.04 | Kunsi | maximum core verbose level is 21 |
| 11:15.20 | Kunsi | atleast it doesn't go any higher than that |
| 11:17.26 | resist0r | bounceman: is your asterisk.conf pointing to the sane spool path you are saving the .call file? |
| 11:18.02 | bounceman | http://pastebin.com/G2fW71Cu |
| 11:18.06 | bounceman | This is all I receieve |
| 11:18.10 | bounceman | with five thousands v |
| 11:18.51 | ChannelZ | And you don't spot the problem? |
| 11:19.38 | bounceman | Why wouldnt it hit the 5000 extension before dialing? It should hit it before forbidden is recieved |
| 11:19.49 | ChannelZ | Why would it? |
| 11:20.15 | ChannelZ | It tries to create the channel you specified, SIP/siptrunk.foo.bar/4610101010 - once that channel answers, it sends it into the dialplan. |
| 11:20.26 | ChannelZ | That channel is never successfully made. |
| 11:20.35 | bounceman | It sends out an invite, what if I want to do changes in the context regarding callerid? |
| 11:20.37 | bounceman | for an example |
| 11:21.05 | Kunsi | bounceman: i use Local/ channel for that |
| 11:21.14 | ChannelZ | Well there's a CallerID: for thatspecific case.. But that's not how call files work in general |
| 11:21.17 | bounceman | and then you dial from the context? |
| 11:22.10 | Kunsi | bounceman: http://paste.debian.net/379123/ - callfile then uses channel Local/phoes@callfiles |
| 11:22.15 | Kunsi | phones* |
| 11:33.08 | bounceman | Ok I got it running now, however it seems that the dialer sends an BYE instantaneously when the call is anwered |
| 11:37.10 | bounceman | It seems it starts the playback not when the call is answered. |
| 11:37.16 | bounceman | but when it first dials |
| 11:37.43 | bounceman | ah nvm, my issue |
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| 13:14.47 | mub | Y'all da real MVP |
| 13:14.55 | mub | Just wanted to let y'all know |
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| 14:56.46 | danielyk | My problem occured after upgrade to 13.7.2. After an re-Register, the peer ip address changed (thats normal). But OPTIONS and INVITEs are now still sent to the old address, so I become for INVITEs an Forbidden from my provider. |
| 14:59.12 | danielyk | The OPTIONS should be sent to the new peer address. But that does not. |
| 15:03.33 | Samot | Does the Asterisk box still think the External IP is the old IP? |
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| 15:07.32 | danielyk | The external IP is updated without problems (all SDP has the correct public ip). |
| 15:09.20 | danielyk | I have dnsmgr turned on, so the peer ip address is updated. But Asterisk still sends all to the old ip address |
| 15:11.29 | danielyk | Only REGISTER are sent to the new ip address... |
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| 15:16.59 | danielyk | After a manually "core reload" all is fine. |
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| 15:17.41 | seik0 | Hi! Can't find whether asterisk 1.4 supports U(x) option for Dial. Does anybody know? |
| 15:18.50 | seik0 | Ok, 've found |
| 15:18.58 | seik0 | looks like it not |
| 15:19.01 | seik0 | thanks |
| 15:20.46 | [TK]D-Fender | Hoply crap ancient garbage... |
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| 15:22.39 | Sonic5093 | Hey guys, I have a quick question. Did PJSIP on asterisk 13 (with realtime enabled) change how the contact header is set. On chan_sip it sets the header to sip:Extension@[ip-adddress] while PJSIP seems to be setting it to just sip:[ip-address]. |
| 15:22.43 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-laqrbyskumsbulor) |
| 15:22.43 | [TK]D-Fender | 6 entire branches behind something that is even supported anymore, and 8 before what is now current LTS |
| 15:23.09 | Sonic5093 | This seems like it is having some effect on a cisco SPA504G that I am using to test directed pick up. |
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| 15:26.37 | seik0 | [TK]D-Fender: just can't find time to migrate. And it work,s so you know |
| 15:27.13 | Sonic5093 | http://pastebin.com/HhUvpjUR is an example of the sip messages I captured on Asterisk 13 and 1.8 to make sure it was not some issue with PJSIP or asterisk 13. If necessary I can get more debug info. |
| 15:28.03 | [TK]D-Fender | Sonic5093: Always provide actual debug, and use the same version for both. Forget 1.8 |
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| 15:29.28 | Sonic5093 | Okay, will go grab that then will ask again. The only reason I brought up the 1.8 server as that is what some of the production servers are still running. I had been tasked with testing everything my company uses in Asterisk 13 before migrating /updating to it. |
| 15:31.09 | seik0 | [TK]D-Fender: it's not just old asterisk, it's old os, old hardware, old handwritten drivers patches. Lots of fun! |
| 15:31.36 | [TK]D-Fender | PAIN. |
| 15:32.17 | seik0 | reality is pain, it is |
| 15:33.46 | seik0 | btw, res_odbc in asterisk 1.4 works the best with oracle odbc in comparison to all later releases |
| 15:34.16 | seik0 | it somehow does not eat memory |
| 15:35.09 | seik0 | cause in oracle odbc drivers, but they are completely proprietary and closed and so on, so we cannot do anything |
| 15:35.16 | seik0 | memory just leaks |
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| 17:24.33 | Sonic5093 | Okay, so I gathered the debug for the question earlier: Sip messages: http://pastebin.com/HhUvpjUR; Relevant Config files: http://pastebin.com/09VtT219 http://pastebin.com/gdE7Hmha PJSIP Debug: http://pastebin.com/3Ni93yW3 http://pastebin.com/X455PB75; Chan_sip debug: http://pastebin.com/pDW5Sd2y |
| 17:24.54 | Sonic5093 | Question from earlier: Hey guys, I have a quick question. Did PJSIP on asterisk 13 (with realtime enabled) change how the contact header is set. On chan_sip it sets the header to sip:Extension@[ip-adddress] while PJSIP seems to be setting it to just sip:[ip-address]. |
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| 17:46.30 | cmendes0101 | If I'm using originate and settings variables, do these get passed into the failed extension? Doesn't appear to be coming through |
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| 18:35.28 | eschmidbauer | do you have to restart asterisk to reload rtp.conf ? |
| 18:37.23 | Sonic5093 | Did PJSIP on asterisk 13 (with realtime enabled) change how the contact header is set. On chan_sip it sets the header to sip:Extension@[ip-adddress] while PJSIP seems to be setting it to just sip:[ip-address]. Sip messages: http://pastebin.com/HhUvpjUR; Relevant Config files: http://pastebin.com/09VtT219 http://pastebin.com/gdE7Hmha PJSIP Debug: http://pastebin.com/3Ni93yW3 http://pastebin.com/X455PB75; Chan_sip debug: http://pa |
| 18:38.02 | Sonic5093 | I can get more debug logs if necessary. In those pastebin links I included debug 5 and verbose 5 as listed on the gathering debug page of the asterisk docs. |
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| 21:47.09 | xochilpili | hello all |
| 21:47.47 | xochilpili | i have no audio in one channel, i can hear them, but they cant hear me |
| 21:48.20 | xochilpili | i have check reinvite=no, and directmedia=yes and "no" |
| 21:48.31 | xochilpili | also i disable iptables for the moment |
| 21:49.04 | xochilpili | this happend only over incomming calls, if i make an outgoing call, everything works fine |
| 21:50.40 | xochilpili | any hand? |
| 21:51.01 | xochilpili | also rtp set debu on << doesnt display anything |
| 22:09.17 | xochilpili | hello? |
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| 22:42.24 | xochilpili | hello? |
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| 23:05.09 | resist0r | xochilpili: are you using NAT and if so what traversal are you attempting to implimet? |
| 23:05.23 | xochilpili | resist0r, yes, im using nat |
| 23:05.39 | xochilpili | i can hear them, but they can not |
| 23:05.58 | xochilpili | and this is only with incoming calls, outgoing calls works fine |
| 23:06.06 | xochilpili | also, extension to extension |
| 23:06.06 | resist0r | right, I think you should check your port forwarding entries |
| 23:06.19 | xochilpili | i have disable iptables |
| 23:06.23 | xochilpili | no firewall |
| 23:06.55 | resist0r | at the switch/router how are you managing SIP and RTP ? |
| 23:07.31 | xochilpili | resist0r, i have nothing in the middle between my isp and this asterisk |
| 23:07.46 | resist0r | I thought you were behind NAT? |
| 23:08.04 | xochilpili | i only have this server which is connected to my isp trunk |
| 23:08.58 | resist0r | so where does the asterisk machine get its IP address? |
| 23:09.37 | xochilpili | i have 2 eth's eth0 direct to my isp provider via an ont, then eth1 to my local lan |
| 23:09.57 | drab | I'm testing a super simple AR menu going from the asterisk wiki and it works except that when I punch in an extension I don't hear the phone ringing even tho I can see in the logs that it is ringing (it eventually goes to voicemail). any thoughts what might be ahppening? |
| 23:10.08 | drab | if I dial that extension directly it works just fine (ie it rings as expected) |
| 23:10.17 | resist0r | okay you are probably piping data to the wrong adapter |
| 23:10.21 | drab | this is my AR menu: https://dpaste.de/Tpr3 |
| 23:10.36 | xochilpili | resist0r, how it's that? |
| 23:11.31 | drab | could it be something with Background? by the time I started keying in numbers it was still playing |
| 23:12.43 | xochilpili | resist0r, the rare thing is that i have not rtp debug |
| 23:12.53 | xochilpili | rtp set debu on << tells nothing |
| 23:12.58 | resist0r | okay... so the issue here is you need to be sure the rtp path and SIP ports are available with iptables down that removes one place but what about the adapter that is connected to your ISP, where does it get an IP address? from your cable modem? |
| 23:15.04 | resist0r | when you connect to the asterisk console use -vvvvvvvvvvvvvvvvvvvvvv to make is give lots out output |
| 23:15.07 | xochilpili | resist0r, sip ports are: 10000 and 20000 declared into my sip.conf file, then, my eth0 is directly connected to an ont provided by my isp, they give me an static ip address, and there's no internet, just voice |
| 23:15.28 | xochilpili | asterisk -rvvvvvvvvvvvvvvvvvvvvvvvvvvvvv is what i use |
| 23:16.00 | [TK]D-Fender | And after all this time you've spent here ... we STILL aren't looking at the actual call |
| 23:16.11 | [TK]D-Fender | And you are talking about issues with auiod for a SIP call |
| 23:16.15 | [TK]D-Fender | and NOT SHOWING SIP DEBUG |
| 23:16.23 | xochilpili | [TK]D-Fender, wait please |
| 23:16.34 | [TK]D-Fender | Wait? |
| 23:16.53 | [TK]D-Fender | You came in asking about his issue an hour & half ago.. and DON'T have the actuall call to show us still? |
| 23:17.33 | xochilpili | http://pastebin.com/RyHW9CbF << sip debug on |
| 23:18.01 | resist0r | I wonder if a module in not loaded that should be? |
| 23:19.01 | xochilpili | resist0r, http://pastebin.com/sQHB5cMd << rtp modules |
| 23:20.53 | snadge | argh.. i just did some maintenance last night to an asterisk realtimedb.. mysql.. and now im getting a bunch of calls being rejected from inbound |
| 23:20.57 | snadge | U 2016/02/09 09:19:56.180901 192.168.0.14:5060 -> 192.168.0.22:5060 |
| 23:20.57 | snadge | SIP/2.0 480 Temporarily unavailable |
| 23:21.09 | snadge | temp unavailable? whut? |
| 23:21.41 | resist0r | is the database up running and logged into? |
| 23:23.17 | snadge | yes |
| 23:23.18 | resist0r | snadge: it seems that 192.168.0.22 is not available |
| 23:23.27 | snadge | except it is.. and passing calls |
| 23:23.32 | snadge | only some inbound calls are doing this |
| 23:23.48 | snadge | i cant find a pattern to it yet |
| 23:23.57 | snadge | if the customer re-regs.. or changes server.. then its fine |
| 23:24.46 | [TK]D-Fender | xochilpili, Why is everyting on local subnets? |
| 23:24.46 | resist0r | are some extensions configured with differing codec preference orders than the expected order on the asterisk machine? |
| 23:25.07 | [TK]D-Fender | show us your actual network interface |
| 23:25.09 | [TK]D-Fender | ifconfig |
| 23:26.06 | resist0r | snadge: have you tried resetting your switch/router? |
| 23:28.11 | resist0r | or maybe fail2ban or similiar is installed on one or both machines and is sensitive and added the other machine to black list? |
| 23:28.13 | resist0r | hmm |
| 23:29.57 | snadge | will check fail2ban |
| 23:29.58 | xochilpili | [TK]D-Fender, http://pastebin.com/EWyuMcMu << ifconfig |
| 23:31.32 | xochilpili | [TK]D-Fender, i created an extension in Asterisk_A, then when i make an call to my inbound route; i have audio in both directions |
| 23:31.58 | xochilpili | the issue is when i redirect the call to Asterisk_B |
| 23:35.02 | [TK]D-Fender | You are talking about 2 * systems |
| 23:35.09 | [TK]D-Fender | and I don't see 2 sets of debug there |
| 23:35.25 | [TK]D-Fender | And you're not telling us what is on the far end of the call |
| 23:35.41 | [TK]D-Fender | You are loking at HALF the call |
| 23:40.04 | xochilpili | [TK]D-Fender, im trying to connect to server 2 |
| 23:40.24 | xochilpili | but i need to move myself to the office |
| 23:40.41 | [TK]D-Fender | And we're not seeing half of the picture |
| 23:41.31 | [TK]D-Fender | punches his card on this issue and retires for the evening... |
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