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| 00:29.42 | klow | I've asked this same question a few different ways but I'm still not exactly clear ... |
| 00:29.51 | klow | res_pjsip_logger.c: <--- Transmitting SIP request (989 bytes) to TCP:73.53.31.109:53843 ---> |
| 00:30.15 | klow | Does that mean that Asterisk needs to be able to start a NEW tcp connection to that endpoint at that IP for that call to go through? |
| 00:31.37 | klow | In other words to Asterisk endpoints need to be reachable for a NEW signaling connection for a call, not just the first connection that was initiated by the endpoint to the server |
| 00:31.41 | file | if rewrite_contact is on then Asterisk will reuse the connection if available |
| 00:32.01 | file | if it's off then it will attempt to contact the IP address and port in the Contact header as provided |
| 00:33.27 | klow | I see |
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| 00:53.52 | klow | if you have an endpoint that is a soft phone app on a phone, so it's IP address may be changing frequently, what would be the best config options? (pjsip) |
| 00:54.11 | klow | I had max_contacts=1 for example, and had trouble because i could not register because there was already a contact |
| 00:54.30 | klow | but if i increment this number, it allows multiple contacts to be shown in pjsip show endpoints, which is confusing |
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| 01:04.44 | file | that's why the remove_existing option exists |
| 01:07.29 | klow | will look at that. so i put rewrite_contact=yes into my endpoint configurations, but still asterisk is trying to make a new tcp connection back to my endpoints |
| 01:07.39 | klow | which are behind another nat, and the tcp connections fail |
| 01:08.06 | klow | I'm not sure how to get this to work, this seemed to work automagically with freepbx/chan_sip but I'm working with pjsip from the ground up now |
| 01:09.26 | klow | i switch to tcp from tis so i could troubleshoot but it seem like its just trying to connect to ports that are not open every time. the ports listed in the contact field for the endpoints ... i didn't know asterisk would actually reach out with a new connection instead of simply using the existing control connection that the client would keep open? |
| 01:11.58 | file | rewrite_contact=yes only works on a new registration, and when Asterisk has not been restarted |
| 01:12.26 | file | and a transport= can't be set for the endpoint |
| 01:12.36 | klow | I was just going to ask that |
| 01:12.39 | klow | my endpoints say transport=transport-tcp |
| 01:14.36 | file | then PJSIP won't reuse connections |
| 01:17.46 | klow | hmm.. so I've set rewrite_contact=yes , remove_existing=yes , and remove the transport=transport-tcp from my endpoints |
| 01:17.50 | klow | can't register currently |
| 01:17.57 | klow | i did restart asterisk a few times |
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| 01:35.18 | klow | file : but don't i need to specify a transport type for the endpoint ? what if i want to restrict the endpoint to only being able to register via TLS for example (which I will need to do eventually) |
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| 01:51.33 | klow | Doh. Cant get any of my phones to register now |
| 01:51.41 | klow | http://pastebin.com/gAwcqn5P anyone able to have a look? Thanks |
| 01:52.00 | klow | "no identify sections to match against" in the log when I try to connect with my soft phone |
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| 02:07.51 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.7.1 (2016/01/15), 11.21.1 (2016/01/15); 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 |
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| 05:22.32 | qloogkm | anybody know how to get pjsip to dial out without any authentication to a sip trunk? provider allows anything from my ip address so no need to send username and password |
| 05:23.44 | qloogkm | it complains unless i put in the "outbound_auth" parameter |
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| 06:01.47 | rexwin_ | from where can I get telecom switch partitioning support? |
| 06:07.08 | rexwin_ | https://www.dropbox.com/s/273qywcmuo5momz/Screenshot%202016-02-03%2020.52.58.png?dl=0 |
| 06:16.45 | [TK]D-Fender | rexwin_, We don't support their stuff. We don't use their stuff. This isn't the place to ask for this. |
| 06:17.16 | [TK]D-Fender | Microsoft does not support Photoshop |
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| 08:44.21 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.7.1 (2016/01/15), 11.21.1 (2016/01/15); 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 |
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| 09:18.20 | BarthezZ | Hmm, I'm having some trouble getting my peers recognised in asterisk 1.8... What would be the best settings on the peer for the following case; I have calls coming in and going out from 2 ip's, which should be mapped to individual peers (for presence for example) |
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| 09:30.20 | pawiecki_ | BarthezZ: what do you mean by "trouble getting recognised"? |
| 09:31.19 | BarthezZ | hmm well what I want to achieve is that the call for the peer is matched on the from username, but only when coming from one of the two known ip's |
| 09:31.55 | BarthezZ | I think ideally it would be cool if i can get those matched based on the srv record, but I'm afraid 1.8 can't handle that yet |
| 09:35.58 | pawiecki_ | But you should have single peer registered on single device/IP. Then you can control (in/out)going calls with dialplan rules. If i understand what you need. |
| 09:37.21 | BarthezZ | Well, the peer can be coming from two seperate paths |
| 09:38.28 | BarthezZ | but it's identically the same peer |
| 09:39.24 | BarthezZ | so settings host=dynamic and insecure=invite works... just doesn't feel right |
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| 10:45.30 | bouncema_ | if I wish to change the sound files being played in the queue, announcements and such. Do I need to replace the original sound files under /var/lib/asterisk/sounds ? |
| 10:46.02 | bouncema_ | if dialplan changes are excluded |
| 10:57.27 | *** join/#asterisk DelphiWorld (~VOIPER@openvpn/user/DelphiWorld) |
| 10:58.00 | DelphiWorld | hi everyone |
| 10:58.08 | DelphiWorld | i have a gateway registered to my asterisk |
| 10:58.32 | DelphiWorld | gateway is dynamic ip over internet not lan |
| 10:58.37 | DelphiWorld | is it pocible to route call to it? |
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| 11:36.50 | DelphiWorld | guys |
| 11:36.54 | DelphiWorld | how to route call to dynamic peer |
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| 12:02.54 | Demon_VoIP | DelphiWorld, using AGI |
| 12:03.13 | DelphiWorld | Demon_VoIP: no simple way using config? |
| 12:03.23 | Demon_VoIP | DelphiWorld, "dynamic peer" == "host=dynamic" ? |
| 12:03.58 | DelphiWorld | Demon_VoIP: yes, and if i call for example 12125445454, i dont want it to route to the peer username, but change the invite to that number? |
| 12:04.27 | WIMPy | What?!? |
| 12:04.49 | DelphiWorld | WIMPy: my gateway register with username 100 |
| 12:04.52 | Demon_VoIP | Sorry, I don't understand :) |
| 12:04.57 | DelphiWorld | but want to call 222 through it |
| 12:05.32 | Demon_VoIP | Dial(SIP/222@100 ? |
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| 13:15.41 | [TK]D-Fender | Demon_VoIP: Dial(SIP/100/222) |
| 13:15.53 | [TK]D-Fender | "core show application dial" <- |
| 13:22.35 | Demon_VoIP | there are several syntax :) asterisk 13.7 "core show application dial" hasn't such syntax. |
| 13:22.48 | Demon_VoIP | for example: https://wiki.asterisk.org/wiki/display/AST/Dialing+PJSIP+Channels |
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| 13:23.49 | Demon_VoIP | Dial is very overloaded application with syntax changed from version to version |
| 13:25.27 | [TK]D-Fender | Barely changed |
| 13:25.44 | [TK]D-Fender | the basics of tech/peer/number has been around forever |
| 13:26.40 | file | Dial strings are channel driver specific, some just follow the same sort of thing |
| 13:26.50 | [TK]D-Fender | other format for un-authed style calls is more like s tandard URI : Dial(tech/numbertodial@hostname |
| 13:26.53 | file | the only requirement is <name>/ at the front |
| 13:27.14 | [TK]D-Fender | PJSIP is different... and it's documented so you shouldn't take more than 2 minutes to find the example page that explains it all |
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| 13:49.23 | Demon_VoIP | Dial application can use option "m". How can i force to start music from begining every call? :) Or: Any other way I can set a custom melody instead of ringing? :) |
| 13:52.44 | [TK]D-Fender | use the option that uses MoH insttead of ringing |
| 13:52.47 | [TK]D-Fender | That is it |
| 13:53.40 | Demon_VoIP | How can i force to start music on hold from begining every call? Now i dynamic add the same audio with realtime. And delete it after hangup. Is there only way? |
| 13:54.31 | [TK]D-Fender | You aren't being very clear |
| 13:54.48 | [TK]D-Fender | Do you mean DURING the actual bridge or BEFORE the call is answered? |
| 13:55.27 | Demon_VoIP | mesiconhold playing background after first call hangup. Second call started and caller listen NOT start of audio |
| 13:57.38 | [TK]D-Fender | Still not clear |
| 13:57.51 | [TK]D-Fender | What do you mean hearing music after hangup? |
| 13:58.04 | [TK]D-Fender | What is this business about FIRST call and then more things happening? |
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| 14:03.44 | Demon_VoIP | first call: caller listen audio from begining. second call: caller listen audio NOT from begining. Is it clear? |
| 14:04.00 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 14:04.23 | Demon_VoIP | audio = musiconhold instead of ringing |
| 14:05.51 | Demon_VoIP | it's specific of musiconhold :( |
| 14:23.27 | [TK]D-Fender | http://svnview.digium.com/svn/asterisk/branches/13/configs/samples/musiconhold.conf.sample?revision=420494&view=markup |
| 14:23.29 | [TK]D-Fender | ^^^^^^^ |
| 14:27.08 | Demon_VoIP | Not very clear. Did I miss some option? I see the standard config module. |
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| 14:27.42 | [TK]D-Fender | You did |
| 14:27.45 | [TK]D-Fender | read it again |
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| 14:28.54 | Demon_VoIP | cachertclasses=no ? |
| 14:29.35 | Demon_VoIP | announcement= ? |
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| 14:45.42 | [TK]D-Fender | Yes |
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| 14:49.30 | *** join/#asterisk mub (~jub@107.1.220.2) |
| 14:49.42 | mub | Hey asterisk buddies :3 |
| 14:49.56 | onebree | hello mub |
| 14:52.03 | cresl1n | hey mub |
| 14:53.03 | *** join/#asterisk tparcina (~tomo@212.92.200.41) |
| 14:54.17 | mub | Sooo, last night I built a demo vicidial system from scratch. I compiled asterisk first, then dahdi-linux-complete, then amfeltec's voicesync module. I was testing everything this morning and everything works except when I tried to call a conference. Asterisk complains that app_meetme.so is missing. |
| 14:55.28 | mub | I must have missed a step, because this never happened to me before |
| 14:57.22 | mub | I recompiled the dahdi-linux-complete, dahdi_test shows all 99.999%, everything is running as it should.. yet app_meetme is still missing |
| 14:58.45 | *** join/#asterisk hfb (~hfb@96.247.52.240) |
| 14:58.53 | [TK]D-Fender | So go look at menuconfig to see what it thinks is missing. |
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| 15:05.16 | acidfu | it is possible to specify the "caller id" when using "channel originate" from the command line ? |
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| 15:05.46 | [TK]D-Fender | no |
| 15:05.55 | [TK]D-Fender | acidfu: Use AMI Originate or a call-file |
| 15:06.06 | acidfu | a call file ? |
| 15:06.09 | acidfu | what is that :) |
| 15:06.41 | [TK]D-Fender | GOOGLE-ABLE |
| 15:07.07 | acidfu | and you can trigger it via the cli too ? |
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| 15:07.56 | [TK]D-Fender | * CLI? Make a back-door method. |
| 15:10.55 | mub | BAH I lied, I didn't compile the voicesync module |
| 15:11.12 | mub | This is what I get for trying to drink and work |
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| 15:13.37 | mub | I don't understand how dahdi_test worked and reported such good results |
| 15:14.16 | mub | I guess the init script loaded the dahdi_dummy... Which is just as good as a timing card apparently |
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| 17:05.37 | socomm | Does asterisk have some form of Click to Dial integration? |
| 17:08.05 | [TK]D-Fender | Click-to-dial is YOU |
| 17:08.10 | [TK]D-Fender | Asterisk is just the back end |
| 17:08.31 | [TK]D-Fender | You crewate your interface (the thing you click on) and have it tell * to place the call |
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| 17:28.06 | mub | socomm: I might be able to give you an example of how ours works with our freepbx box |
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| 17:43.55 | mub | socomm: I'm looking around aimlessly on the server. Our programmers aren't here |
| 17:44.11 | mub | I doubt they'd explain it to me anyways; they hate me |
| 17:44.20 | mub | sorry, can't help you |
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| 18:35.53 | *** join/#asterisk grim_ (~bikerGrim@rrcs-98-102-12-82.central.biz.rr.com) |
| 18:36.13 | grim_ | module load res_fax_digium.so crashes the ast process... |
| 18:36.56 | grim_ | I've tried the recommended barcelona, generic, opteron |
| 18:37.53 | grim_ | full log doesn't have anything in it. where would I start in terms of troubleshooting this? |
| 18:40.47 | grim_ | running ast in console. then connecting a second console to load the mod. the process seg faults. |
| 18:42.55 | *** join/#asterisk italorossi (~Adium@177.65.203.21) |
| 18:45.32 | KValchev | grim_: if you buid is from source try to recompile all . |
| 18:47.29 | grim_ | nvrmnd, just saw that fax for asterisk is no longer supported in 13. :-( |
| 19:14.33 | *** join/#asterisk poolson (~dddd@74-94-68-241-bellevue.wa.seattle.hfc.comcastbusiness.net) |
| 19:15.29 | poolson | hey im wondering what a good approach to doing something is. currently i have an app that connects the AMI and looks for a certain event (a call coming in) and then taking the CID and doing something with it. |
| 19:15.44 | poolson | from there i'm then looking to see which extension answered the call (if at all) |
| 19:15.57 | drmessano | ?ok |
| 19:16.04 | poolson | im thinking i can just keep wading thru the globs on data streaming from AMI |
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| 19:16.27 | poolson | but can i just take the id from the event of the call coming in and then querying somewhere with that? |
| 19:16.46 | poolson | drmessano ! LTNS ! |
| 19:17.18 | WIMPy | I'm not sure what the actual question is. |
| 19:17.28 | poolson | well dude let me dumb it down |
| 19:17.55 | poolson | whats the best way to identify which extension picked up a call |
| 19:18.08 | poolson | preferably i want to do this from some external app |
| 19:18.43 | [TK]D-Fender | Channel names are all there |
| 19:18.47 | [TK]D-Fender | So go do it |
| 19:18.49 | WIMPy | Extensions don't pick up calls. |
| 19:18.55 | poolson | here we go |
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| 19:19.10 | poolson | yeah ok gurl sorry for the lack of technicality |
| 19:19.16 | poolson | guru* |
| 19:19.22 | poolson | well i know its all there |
| 19:19.27 | [TK]D-Fender | So go do it |
| 19:19.31 | poolson | the question im asking is there a prefered approach |
| 19:19.45 | [TK]D-Fender | Channel name is sent with "new call", when bridging, etc |
| 19:19.45 | WIMPy | CHECK DIALEDPEERNAME AND DIALEDPEERNUMBER. |
| 19:19.56 | [TK]D-Fender | Ther is no "preferred method. AMI is AMI |
| 19:20.04 | [TK]D-Fender | WIMPy: that's dialplan |
| 19:20.07 | poolson | was that so difficult dude ? |
| 19:20.07 | [TK]D-Fender | not AMI |
| 19:20.14 | poolson | well there you go |
| 19:20.14 | [TK]D-Fender | Ther is nothing difficult |
| 19:20.20 | poolson | maybe better to do in the dial plan then ? |
| 19:20.25 | [TK]D-Fender | The events for the newcall & bridge are right there in the list |
| 19:20.36 | poolson | dude seriously are you a full time clown ? |
| 19:20.37 | WIMPy | What is that suppoest to mean? You will get it via AMI as well. |
| 19:20.48 | poolson | i already have that part functioning .. getting when a call comes in |
| 19:20.56 | poolson | im just asking about other approaches (if any) |
| 19:21.07 | [TK]D-Fender | Other than AGI? |
| 19:21.14 | [TK]D-Fender | AMI rather |
| 19:21.55 | poolson | fender how often do you bathe? im just trying to get a grip on who im dealing with here |
| 19:22.11 | WIMPy | DIALEDPEERNAME is the only generic thing. |
| 19:22.52 | WIMPy | If you watch for channel names, you might have to track more information on them, depending on the channeltype in use. |
| 19:23.33 | poolson | so whats the terminology for what happens internally when someone answers the call at an extension |
| 19:23.56 | WIMPy | And you can use dialplan, off course. But that seems like a step backwards. |
| 19:25.04 | [TK]D-Fender | poolsonso whats the terminology for what happens internally when someone answers the call at an extension <- bridge |
| 19:26.06 | WIMPy | Well, first the call is answered which then leads to the two channels being bridged. |
| 19:26.11 | [TK]D-Fender | poolsonim just asking about other approaches (if any) <- other than AMI would be you calling a dialplan app that logs it in a way you want. |
| 19:27.08 | poolson | bravo dude. see when you take the cock outta your mouth you can focus on being sensible |
| 19:27.13 | poolson | thanks for the info tho ! |
| 19:27.14 | poolson | :) |
| 19:27.24 | poolson | and thanks wimpy too |
| 19:28.03 | [TK]D-Fender | Watch your own mouth |
| 19:28.44 | poolson | today is a good day dudes ! |
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| 19:44.43 | drmessano | [TK]D-Fender: isn't he just a joy to talk to? |
| 19:45.07 | [TK]D-Fender | drmessano: While there are words I might use, that isn't currently on the list. |
| 19:45.49 | drmessano | https://usercontent.irccloud-cdn.com/file/jBoY0Gsy/IMG_2001.JPG |
| 19:45.58 | drmessano | ^ |
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| 19:55.56 | mub | Chicks dig it. |
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| 20:28.27 | dan_j | Hi. I'm running a set of asterisk servers in multi-tenant environment. Some clients have call recording and it's working well. However, if client A happens to call client B, the call is only recorded once. As far as I can see, my only option is to install a border asterisk server which the asterisk servers will dial, and then the border asterisk server will |
| 20:28.27 | dan_j | dial back to the relevant asterisk server, so that the internal asterisk servers see them as two separate calls and record the call twice. |
| 20:28.48 | dan_j | Is there any other option within asterisk that can cause it to treat the call as two totally separate calls? |
| 20:29.09 | dan_j | Specifically, the AMI linkedid should be different for each side of the call. |
| 20:29.42 | WIMPy | A local channel in between should be enough. |
| 20:32.50 | dan_j | Local channels still seem to share the same linkedid |
| 20:33.28 | WIMPy | Never looked at them. But make sure you have the non optimizing version. |
| 20:33.31 | dan_j | I use the linkedid to start the call on the correct channel so that the recording continues even if the call is transferred. |
| 20:34.12 | dan_j | But the linkedid for one client should be different than the linkedid of the other client. |
| 20:34.45 | dan_j | I also thought local channel, but there doesnt seem to be a way to get asterisk to treat it as a totally separate channel. From asterisk's point of view, they are still linked. |
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| 21:16.51 | wdoekes | dan_j: I use an opensips server which mangles the callid, so I get a second call into the original asterisk |
| 21:17.31 | dan_j | I've not looked at opensips yet. How easy was it to set up? |
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| 21:18.27 | wdoekes | specifically this: https://github.com/wdoekes/opensips/commits/wjd-spiral_module , but there are other modules (b2bua) or kamailio with topoh (which is 90% of the aforementioned module) |
| 21:19.16 | wdoekes | the sers (opensips, kamailio) have a bit of a learning curve |
| 21:19.46 | wdoekes | but if you use it for limited purposes like bouncing back the call, you won't need a large config |
| 21:20.32 | WIMPy | A local channel (without optimizing!) really doesn't work? |
| 21:20.42 | dan_j | Is it able to read from a file or database in order to know whether the call should be bounced back or sent out? |
| 21:21.06 | dan_j | WIMPy: Nope. The linkedid all remain the same. |
| 21:21.27 | wdoekes | dan_j: oh yes, it's capable of everything |
| 21:21.31 | dan_j | linkedid = the uniqueid of the first channel that started the call. |
| 21:21.43 | dan_j | wdoekes: but it doesnt handle any media itself? |
| 21:22.05 | wdoekes | it doesn't need to, but it can plug in a proxy (rtpproxy for instance) |
| 21:22.42 | WIMPy | There must be other loopback options. |
| 21:23.35 | wdoekes | I've tried all sorts of things, there is no way to do everything (like call recording on two sides, or get sane CDR/CEL records) with chan_sip |
| 21:25.38 | dan_j | wdoekes: do you run only one opensips server or multiple for redundancy? |
| 21:26.07 | wdoekes | everything is redundant across multiple DCs, obviously |
| 21:26.44 | dan_j | Cool. I'm working on that solution. Not worked out an easy way of taking a DC down and moving to another, and then switching back easily. |
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| 21:55.08 | klow | Assuming that my endpoints are all over hell on different kinds of NATs, as they are mobile apps (Bria) - what is the best strategy for NAT traversal ? is it ICE ? TURN/STUN? ICE uses both of those already? I'm not totally sure what I need to set up to help my client app endpoints make successful calls as often as possible .. any info greatly appreciated! |
| 21:55.44 | klow | currently testing on 13.7 pjsip . |
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| 21:57.22 | klow | I started trying to configure "resiprocate-turn-server" but got a bit confused. It says 2 public IPs are required in the configuration - my Asterisk server are behind a firewall and accessed by endpoints via port forwarding. So can an TURN server be behind a NAT itself with port forwarding? Or must I put this on public IP .. and do I need 2 IPs for it to work? |
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| 22:30.52 | drab | can anybody recommend a simple CDR CSV parser script? googling for stuff like that I get a ton of fancy web apps, but I really just want a script to then gen an email |
| 22:31.24 | drab | not interested in billing or anything like that, just want to review calls from previous day to make sure nothing weird happened |
| 22:31.53 | drab | almost tempted to write something myself in bash, but there ought to be something done already |
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| 23:31.55 | newtonr | drab, I don't know of anything off-hand |
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