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| 00:04.53 | navaismo | ill back tomorrow LOL |
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| 01:34.39 | raspberrypifan | im trying to build chan dongle but i get no makefile.am |
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| 01:59.07 | raspberrypifan | anyone |
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| 02:35.28 | ruben23 | hi guys, i have a hosted asterisk server and my office is accessing via SIP, but my issue is the voice traffic and data traffic are on the same route, any ways i can implement to separate voice traffic and data from accessing my asterisk server somehow..any suggestions..??? |
| 02:41.57 | spicyramen_ | normally you separate Voice and Data traffic via VLANs, that's highly recommendable. |
| 02:42.21 | spicyramen_ | You can place Asterisk in Voice VLAN if you use asterisk for VM, play announcements |
| 02:45.05 | spicyramen_ | Any idea what this error means in * 12.5 "[2014-09-14 06:05:01] ERROR[23029]: pjsip:0 <?>: icess0x7fa8bc0 ..Error sending STUN request: Invalid argument" |
| 02:45.05 | spicyramen_ | stun show status shows right public IP, but ICE candidates are not showing in SDP just local IP address |
| 02:45.06 | spicyramen_ | just updated to * 13 beta release same problem my public stun is google stun server/ Happens with stun.counterpath.net:3478 as well |
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| 07:26.47 | Stefan27 | hi americans |
| 07:31.14 | MaliutaLap | That's right, don't say hi to the Australians :P |
| 07:31.19 | wdoekes | americans? at 0730 UTC? |
| 07:31.30 | MaliutaLap | Stefan27: racist :P |
| 07:32.07 | MaliutaLap | wdoekes: leave the racist alone - let them fit their own problems ;) |
| 07:32.20 | wdoekes | ;P |
| 07:37.31 | coppice | racist? is that someone who prefers field over track? |
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| 07:49.40 | Stefan27 | who's racist? |
| 07:49.49 | Stefan27 | Oh lol |
| 07:50.21 | Stefan27 | I can't greet everyone MaliutaLap... |
| 07:52.20 | wdoekes | so you just greet the ones who sleep.. |
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| 08:10.07 | MaliutaLap | Stefan27: sure you can, it goes something like "Greeting and salutations one and all" :P |
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| 08:16.02 | newmember | hey, I am registering asterisk but asterisk sends a contact with an 's' what option changes that 's'? <sip:s@10.88.1.21> |
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| 08:16.41 | newmember | I troed adding a '/1234' to the registration line but no change |
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| 08:17.17 | newmember | tried* |
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| 08:22.14 | MaliutaLap | 's' is the context that the call is going to. |
| 08:22.45 | MaliutaLap | newmember: pastbin the sip.conf, could help fix the issue |
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| 08:27.46 | newmember | MaliutaLap: http://pastebin.com/W6mbzeZE |
| 08:28.31 | newmember | MaliutaLap: Gafachi works, callcentric has that issue |
| 08:31.15 | MaliutaLap | newmember: you might want to add a context to the callcentric entry. In that context you should have an 's' context, because that is where it would seem to be sending incoming calls |
| 08:32.08 | MaliutaLap | newmember: from the CLI what does 'sip show peers' produce |
| 08:32.26 | MaliutaLap | you have the syntax for registrations fine |
| 08:33.13 | newmember | MaliutaLap: 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline] |
| 08:33.58 | MaliutaLap | newmember: you might want to look at 'sip show peer callcentric' |
| 08:34.28 | MaliutaLap | it is also valid to put 'register => XXXXXX:pppppp@callcentric.com/s' |
| 08:35.33 | newmember | http://pastebin.com/fujSj89a |
| 08:38.46 | MaliutaLap | newmember: adding "qualify=yes" to the entry would also tell you if it was registered. It would show up in 'sip show peers'. But from the output of that it already looks as though it is registering properly |
| 08:39.17 | MaliutaLap | newmember: currently it looks fine to me - are you having problems with calls in a specific direction? (in vs. out) |
| 08:39.25 | newmember | in |
| 08:40.03 | newmember | here is the error: [Sep 14 19:37:37] NOTICE[3370]: chan_sip.c:15432 handle_request_invite: Failed to authenticate user <sip: |
| 08:41.10 | newmember | this happens when I try to call in |
| 08:43.17 | newmember | I added the 's' and reloaded |
| 08:43.18 | newmember | [Sep 15 04:49:46] NOTICE[3370]: chan_sip.c:15432 handle_request_invite: Failed to authenticate user <sip: |
| 08:45.31 | MaliutaLap | newmember: is there a reason you have all those callcentric host entries? |
| 08:46.00 | newmember | they asked me to add it, I think its all their hosts |
| 08:46.19 | MaliutaLap | newmember: it would seem that you should have the host entry in that set to sip.callcentric.com - the other hostnames you have there don't match the dns entry for sip.callcentric.com |
| 08:47.06 | newmember | I think those hosts are found with srv |
| 08:47.44 | newmember | or any of the hosts could be found with srv lookup of callcentric.com |
| 08:50.14 | MaliutaLap | newmember: have you seen http://sysadminman.net/blog/2008/callcentric-trunk-setup-with-asteriskfreepbx-210 or http://apetec.com/voip/Asterisk-Config-Callcentric.htm ? |
| 08:50.20 | newmember | looking at wireshark, I see the INVITE from CC then I see a return packet: SIP/2.0 403 Forbidden |
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| 08:51.34 | newmember | checking |
| 08:52.08 | MaliutaLap | yeah, that normally means there is an issue with the user,fromuser,defaultuser, or secret entries |
| 08:52.36 | MaliutaLap | I use defaultuser and fromuser in my ITSP setups |
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| 09:09.58 | newmember | reading the links you provided I see there is a 's', what is the 's' for? exten => s,1,Dial(SIP/123) |
| 09:11.50 | newmember | I tried adding the 's' and it failed with the same error |
| 09:13.48 | MaliutaLap | s is a generic target |
| 09:14.21 | MaliutaLap | calls coming in can come directly to a 's' extension, or go out to an 's' extension |
| 09:18.30 | MaliutaLap | http://pastebin.com/YGc9bSPa is an excerpt from one of my my ITSP confs |
| 09:18.43 | MaliutaLap | using an s extension |
| 09:22.29 | newmember | ty |
| 09:22.38 | newmember | I am take the night to sleep on it |
| 09:22.43 | newmember | its 3am here |
| 09:22.52 | newmember | MaliutaLap: thanks for the ideas |
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| 09:30.32 | cervajs2 | hi, i want to connect from SIP.js (webrtc client) to asterisk. i enabled http in http.conf and ws transport is ok. but where i can configure wss? |
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| 10:12.37 | Zogot | hey, is there an incorrect column name in the pjsip realtime columns? i svn checkout branch 12, i use alembic and i have in my ps_transports table, a column verifiy_server (http://svn.asterisk.org/svn/asterisk/branches/12/contrib/ast-db-manage/config/versions/2fc7930b41b3_add_pjsip_endpoint_options_for_12_1.py Ctrl+F verifiy) |
| 10:18.17 | wdoekes | Zogot: yes, it's wrong |
| 10:18.46 | wdoekes | I'd fix it, but I have no idea how that alembic stuff works. just replacing the existing version would be wrong. |
| 10:18.59 | wdoekes | please file a bug report |
| 10:19.28 | Zogot | wdoekes: alrighty |
| 10:19.31 | wdoekes | or if you happen know how to write a migration to fix the column name, I can commit it |
| 10:20.04 | Zogot | ah ok, il look to it |
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| 10:33.37 | Zogot | wdoekes: not sure how right it is. i followed the quick guide on alembic website. i tested up and downgrade |
| 10:33.38 | Zogot | https://gist.github.com/zogot/6a584bb4165e78d5e80b |
| 10:33.56 | Zogot | and it renamed correctly, gonna eat back in a little while |
| 10:37.22 | wdoekes | Zogot: looks good to me, tnx |
| 10:37.57 | LiohAu_ | any baresip user here? |
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| 11:04.11 | Zogot | back |
| 11:04.17 | Zogot | wdoekes: cool |
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| 12:03.42 | file | spicyramen_, that can occur if the client provides a TCP based candidate which is presently unsupported |
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| 13:46.19 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.12.0 (2014/08/19), 1.8.30.0 (2014/08/19); Standard: Asterisk 12.5.0 (2014/08/19); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
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| 13:54.16 | cervajs2 | which is the best configuration for webrtc endpoints if i dont know if the endpoint will be behind nat? icesupport=yes, nat=yes ? |
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| 14:18.04 | file | WebRTC requires ICE. |
| 14:19.12 | cervajs2 | file: what about nat settings? |
| 14:19.20 | file | doesn't matter, it'll use ICE |
| 14:20.43 | cervajs2 | file: thanks. i'll note it |
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| 14:26.04 | cervajs2 | any personal recomendation for sip stack for JS? i'm trying SIP.js. REGISTER is succesfull, but peer is UNREACHABLE. sipml5 works better and peer is ok |
| 14:26.35 | Stefan27 | i use sipml5 |
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| 14:32.37 | Stefan27 | but we dont use asterisk12's pjsip stack, we use a setup of asterisk+webrtc2sip+sipml5, the latter two are from doubango telecom http://doubango.org/ |
| 14:33.22 | Stefan27 | im gonna try to get rid of the middle guy (webrtc2sip) and use asterisk's webrtc support instead |
| 14:33.57 | cervajs2 | is webrtc2sip still necessary? i have working wss asterisk 11.12.0 |
| 14:34.04 | Stefan27 | it's not |
| 14:34.18 | Stefan27 | Trying to get rid of it |
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| 14:34.33 | dan_j | Hi. I'm using Asterisk 11.10.2 and trying to pause mix monitor. However, i've just discovered that pausemonitor isn't listed when i do 'core show applications' |
| 14:34.40 | dan_j | Any idea what could be wrong? |
| 14:35.15 | Stefan27 | we only recently installed the ast 12, i guess webrtc support came with 11? |
| 14:35.41 | Stefan27 | wss that is |
| 14:38.39 | marceloamorim | putnopvut: hey man, how are you, I asked few days ago about ccss, if the ccss put the requests in some file or db that I could look, do you know if there is? |
| 14:38.44 | mjordan | 11 had SIP over WS/WSS, DTLS-SRTP, ICE/STUN/TURN |
| 14:39.53 | putnopvut | marceloamorim: hm, no I don't believe it does anything like that. |
| 14:40.52 | mjordan | the only thing added in 12 that was WebRTC related was pass through support for VP8/Opus and format attribute negotiation for Opus |
| 14:41.06 | dan_j | Assuming that pausemonitor is not compatible with mixmonitor, how does one pause mixmonitor? |
| 14:41.20 | marceloamorim | putnopvut: ok, I`ll keep try to find it, thx man |
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| 14:51.28 | mjordan | dan_j: you should be able to use StopMixMonitor, then, if you want to restart it, use MixMonitor with the 'a' option. |
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| 16:03.12 | qakhan | hi all. i am trying to do call out through php script but is it not working. here is script http://pastebin.com/33W7Kczt |
| 16:03.27 | qakhan | here is asterisk cli http://pastebin.com/EcrjjBYe |
| 16:08.46 | drale | does $number need to be a string and not an int |
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| 16:10.24 | wdoekes | qakhan: I don't think asterisk knows you want php(1) to interpret that |
| 16:10.37 | wdoekes | try adding '#!/usr/bin/env php' at the top of the .php file |
| 16:10.56 | wdoekes | and make it executable |
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| 16:28.03 | dan_j | mjordan: thanks. works perfectly. |
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| 17:41.04 | qakhan | drale number will be int |
| 17:42.08 | qakhan | wdoekes same script was working on centos 5.6 asterisk 1.4 now i upgreaded to centos 6.5 and asterisk 11.18 |
| 17:50.35 | qakhan | wdoekes #!/usr/bin/env php removed the error but dialpaln is not getting $number value . |
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| 19:03.29 | Pegasus_RPG | Hello. I'm sitting on hold on a single-line phone connected to * via an ATA. What dialplan do I need to add to allow it to transfer the call to another extension? |
| 19:03.58 | Pegasus_RPG | I can already do a hook flash and dial another extension, but hook flashing again just switches between the calls. |
| 19:05.13 | sgriepentrog | Pegasus_RPG: After dialing the "another" extension, just hang up to complete the transfer. There's a bit of blind faith that it will go through... |
| 19:06.17 | Pegasus_RPG | Yeah, and I can't lose this call so I'll wait to try that when I experiment by calling my cell phone :) |
| 19:06.35 | [TK]D-Fender | Read its manual. |
| 19:06.54 | WIMPy | Well, there are two answers: Either 1. Reat the manual of your ATA or 2. look at features.conf for an alternative. |
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| 19:08.40 | Pegasus_RPG | OH, didn't realize the ATA would handle it |
| 19:08.54 | WIMPy | And next time, read the manual befor using whatever for something important. |
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| 19:10.38 | cervajs2 | is it possible use webrtc in asterisk 11 only for signaling? media (opus+vp8) directly to endpoints |
| 19:10.39 | Pegasus_RPG | WIMPy: yes, fair point. My original problem is actually that my regular VoIP desk phone can't get back SRTP calls on hold for longer than about 2 minutes. (I don't hear anything and neither does the other party) |
| 19:10.50 | Pegasus_RPG | so I have to transfer it to another extension to get it going |
| 19:11.12 | Pegasus_RPG | And I'm assuming that's a firmware problem in the phone |
| 19:13.04 | file | cervajs2, no. |
| 19:14.54 | cervajs2 | file: this means that "directmedia" in endpoint configuration doesnt have any effect? |
| 19:15.15 | file | not when ICE, TURN, or SRTP is in play |
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| 19:16.57 | cervajs2 | file: thank you. this information is very interesting |
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| 19:22.28 | cervajs2 | file: last question. i promise ;) i dont understand what's the role of TURN server in rtp.conf. can you explain it? or do you have doc for it (no read the source luke pls ;) |
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| 19:23.05 | file | it's not something Asterisk specific |
| 19:23.18 | file | https://en.wikipedia.org/wiki/Traversal_Using_Relays_around_NAT |
| 19:24.46 | cervajs2 | file: i know theory about TURN. i dont understand how asterisk using it. you can configure that into SDP that for relay is used external TURN server and not internal asterisk RTP relay? |
| 19:25.22 | file | no, Asterisk acts as a TURN client if it is behind NAT and direct communication is not possible |
| 19:25.35 | file | traffic still flows through Asterisk, it just goes through the TURN server to get there |
| 19:26.00 | cervajs2 | file: got it. thanks ! |
| 19:27.01 | LiohAu | hey guys, any recommendation for the sound codec ? |
| 19:27.16 | LiohAu | g711 is the most used right? |
| 19:28.08 | Qwell | LiohAu: It depends on what your endpoints support. |
| 19:28.30 | cervajs2 | opus is the new star. g711 is legacy which works always |
| 19:28.43 | LiohAu | are there default common codec installed on OSX and windows ? |
| 19:29.01 | LiohAu | (I only target softphones on osx and windows) |
| 19:29.24 | cervajs2 | codecs are typically in softphones |
| 19:29.38 | cervajs2 | check jitsi.org |
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| 19:31.34 | [TK]D-Fender | [15:29]cervajs2codecs are typically in softphones <- ALL VoIP Phones... not just softphones... |
| 19:31.53 | [TK]D-Fender | LiohAu: G.711 is what's used on the PSNT almost universally. |
| 19:32.10 | [TK]D-Fender | LiohAu: What you choose will depend on what you're communicating with. |
| 19:32.36 | [TK]D-Fender | LiohAu: You could use 1 codec for 1 kind of call, but another for a different call. |
| 19:32.45 | LiohAu | cervajs2: yes but are there codec already setup in windows and osx ? (If you make your own SIP client and you don't want to handle the distribution of codec for examples) |
| 19:33.21 | [TK]D-Fender | LiohAu: They probably don't come with codecs. Go find some and include them |
| 19:33.27 | LiohAu | ok |
| 19:33.35 | LiohAu | thought they had default :s |
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| 19:38.55 | cervajs2 | if you want develop softphone, you will probably use VoIP SDK like www.pjsip.org. free codecs are included (like g711) |
| 19:39.45 | WHiZZi | ahoy all, perhaps a stupid question here. Is there any reason I should enable allowguest=yes on an external reachable system (port 5060) where only incoming phonecalls are received from registered SIP peers? |
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| 19:40.03 | WHiZZi | other than, yeh.. I like anonymous INVITEs on my SIP port ? |
| 19:40.41 | [TK]D-Fender | WHiZZi: No. |
| 19:41.28 | WHiZZi | ok, thnx :). |
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| 19:43.47 | DrewBeer | is there a way to transfer a call from iax to iax using a specific ext and context? or do i need to just dump into the context and have some route in there? |
| 19:44.57 | [TK]D-Fender | DrewBeer: All call processing = dialplan |
| 19:45.09 | DrewBeer | figured |
| 19:45.29 | DrewBeer | does transfer context still work? |
| 19:45.39 | DrewBeer | i know in a peer you can set the peercontext |
| 19:46.47 | [TK]D-Fender | DrewBeer: As in? |
| 19:48.38 | ipengineer | When sending a pjsip_notify message should Asterisk handle the 401 Unauthorized? Right now it is just dropping it |
| 19:50.09 | mjordan | ipengineer: do you have outbound authentication configured? |
| 19:50.22 | ipengineer | On Asterisk? No |
| 19:50.31 | mjordan | k, is someone sending the 401 to Asterisk? |
| 19:50.57 | ipengineer | mjordan: Yes, sending a notify to a Grandstream device and it sends back a 401 |
| 19:51.13 | ipengineer | mjordan: https://gist.github.com/zconkle/3822cbb8741869e150ae |
| 19:51.26 | mjordan | you need to have an outbound_auth set up for your endpoint in question. |
| 19:51.32 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_res_pjsip |
| 19:51.47 | mjordan | Look at outbound_auth. Asterisk doesn't know how to authenticate against the 401, so it doesn't do anything |
| 19:52.26 | ipengineer | mjordan: Ok. Thanks |
| 19:53.37 | ipengineer | mjordan: Real quick before I get too far in on this side. What is there to Auth against on a device? Other than the user/pass for configuration changes |
| 20:01.42 | mjordan | set outbound_auth to the section that defines the credentials the device expects |
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| 20:01.55 | mjordan | other than that, it should match an auth section, such as those in the examples |
| 20:02.04 | mjordan | pretty sure the wiki page has examples of setting up outbound auth on a trunk |
| 20:02.46 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip#Configuringres_pjsip-ASIPtrunktoyourserviceprovider,includingoutboundregistration |
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| 20:12.42 | ipengineer | Alright Ill give that a go |
| 20:17.01 | marceloamorim | guys, I`m trying to forward my skills on asterisk. I put the logging from the call using the mariadb and connect cel with that db, but when I call for some phones that state is busy, the cel don`t tell me that, just show me the hangup at the same time from chan_start. Should I try to set more informations in another table using the same linkedid? |
| 20:22.45 | spicyramen_ | which version of * includes a fix for this https://issues.asterisk.org/jira/browse/ASTERISK-23026 ? |
| 20:33.42 | rmudgett | If you follow the link that ASTERISK-23026 was closed as a duplicate of: https://issues.asterisk.org/jira/browse/ASTERISK-22961 You will see the target release was v11.11.0 and v12.4.0 |
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| 23:29.55 | dan_j | Hi. Is it possible to specify two separate macro's in the dial command? |
| 23:33.05 | [TK]D-Fender | What's you goal? |
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| 23:44.51 | dan_j | Its ok. I've combined the two macros into one and used a channel variable to determine what code should be run within the macro |
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