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| 00:08.32 | newtonr | zerick, you are dialing the peer (chan_sip) or the endpoint (chan_pjsip). The dialplan context entered is determined by the configuration for the peer or endpoint |
| 00:10.12 | newtonr | if your call needs to potentially match extensions in multiple contexts, you need to include one context in another. |
| 00:10.49 | newtonr | zerick, https://wiki.asterisk.org/wiki/display/AST/Include+Statements+Basics |
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| 03:38.24 | Kobaz | i have a really weird problem |
| 03:38.29 | Kobaz | http://pastebin.com/d4uEFcr0 |
| 03:38.47 | Kobaz | i can turn on insecure=port,invite and that invite will work |
| 03:38.57 | Kobaz | but i shouldn't need to do that |
| 03:39.28 | Kobaz | 5506 successfully registered, the secret is correct |
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| 03:55.17 | Penguin | What's weird about that? insecure invite means it doesn't have to authenticate when making calls. |
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| 04:02.43 | Kobaz | oh hmm |
| 04:02.52 | Kobaz | i never knew that's actually what that meant |
| 04:02.52 | Kobaz | heh |
| 04:03.00 | Kobaz | also i'm getting one way audio |
| 04:03.00 | Kobaz | ugh |
| 04:03.19 | Kobaz | you would think with 5 years of using and developing for asterisk i would be able to figure this sort of stuff out more easily |
| 04:03.35 | Kobaz | some part of the sip stack are still confusing sometimes |
| 04:03.59 | Kobaz | Penguin: so why would (using the same authentication) registering work, but invite not? |
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| 04:12.47 | Kobaz | just weird stuff going on |
| 04:16.08 | Kobaz | fixed my one way audio, that was a silly typo |
| 04:16.16 | Kobaz | this authentication thing is driving me crazy |
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| 04:41.20 | Penguin | Did I miss where you pastebinned your peer definition, masking only the secret? |
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| 07:03.43 | c|oneman | I'm having an issue with our phone system IVR, which fails for australian customers |
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| 07:04.02 | c|oneman | How can I test this without having a real Australian phone? |
| 07:05.31 | MaliutaLap | I can't see how that would be a problem |
| 07:05.52 | MaliutaLap | I do have a telstra touchtone phone - but I only run naked DSL and SIP |
| 07:06.19 | c|oneman | who is your voip provider? |
| 07:06.26 | MaliutaLap | c|oneman: exactly how are .au people having problems? |
| 07:06.50 | c|oneman | at one point in the IVR, our system forwards to another DID. At the point, the call drops for our australian customers only |
| 07:06.52 | MaliutaLap | c|oneman: here at home I use * attached to pennytel. elsewhere I use * attached to netsip |
| 07:07.32 | MaliutaLap | c|oneman: "forwards"??? are you using a Dial()? and do you have "re-invite" on? |
| 07:08.01 | c|oneman | I'm not really using asterisk, It's switchvox |
| 07:08.20 | c|oneman | I don't know how it forwards it internally |
| 07:09.39 | MaliutaLap | can't help you with that - but if the connections are coming in via direct SIP then I'd be looking for things like re-invite and firewall issues |
| 07:11.14 | c|oneman | the forward to a DID |
| 07:11.19 | c|oneman | is* |
| 07:11.34 | c|oneman | the call comes in over SIP but gets forwarded to an external DID at the point of failure |
| 07:11.41 | c|oneman | if I point to another DID, there's no problem. |
| 07:13.28 | MaliutaLap | yeah, if it was * I'd ask for a debug of a failing call - switchvox I have NFI about |
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| 07:34.51 | Zogot | morning all |
| 07:35.46 | Zogot | perhaps a stupid question, but for Registration in PJSIP. what is the difference between a server_uri and a client_uri https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip#Configuringres_pjsip-REGISTRATION |
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| 07:59.10 | jepperl | i am having an issue with the asterisk (11.12.1) where sometime during some SIP REGISTER are being received, the asterisk just dies and the CLI becomes unresponsive. The only thing helping is a complete reboot of the machine. Have anyone had these issues? It happens kind of randomly so there is not really a specific SIP trace i can show you :s But |
| 07:59.11 | jepperl | it often happens during REGISTER, but also at some other times, so it is hard to tell what is causing the problem |
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| 08:02.56 | Chainsaw | jepperl: It could be something as mundane as you running out of available file descriptors. |
| 08:03.10 | Chainsaw | jepperl: What did you set the limits to? |
| 08:06.45 | jepperl | chainsaw: i did not set up the server, i was put on the task of fixing the issue, but i will definitely check out is it is a fd issue (which, when you said id, did not seem very unlikely). I will dig a little more on this and return with results :) thanks! |
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| 09:02.51 | hurdman | hello all, it seems my asterisk doesn't forward my SIP/2.0 183 Session progress during the call, is there an option to check or something like that ? |
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| 09:44.04 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.12.1 (2014/09/18), 1.8.30.0 (2014/08/19); Standard: Asterisk 12.5.1 (2014/09/18); 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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| 10:01.47 | dan_j | Hi. I'm currently using a macro to make the callee press 1 to accept an incoming call. However, when two callees answer and try to press 1, one of the callee's are connected to the caller and the other disconnected. Is there a way I can inform the disconnected callee that the call has been answered elsewhere? |
| 10:04.52 | ovoshlook | Hello all! I try to record call by MixMonitor. It recordes my calls, but it creates file Even call not answered (fie is empty and small size but it is). I use option b, but it does not help |
| 10:06.00 | dan_j | I had that problem. I discovered that it was only making recordings for ANSWERED calls, but i wasnt doing StopMixMonitor. So i had small incomplete files for answered calls. |
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| 12:31.00 | dkorras | <PROTECTED> |
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| 13:04.25 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 13:05.00 | AAmit | Hi All |
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| 13:09.53 | Tuju | why tcpdump shows like this: Via: SIP/2.0/UDP 212.1\000\000\021\000\000\000port 5060\000\000\000\021\000\000\000\340\207+\010 \2 |
| 13:10.08 | Tuju | that ip-address gets escaped from half-way. |
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| 13:20.59 | ovoshlook | hello all when doing call have this error |
| 13:21.01 | ovoshlook | ast_rtp_new: Oh dear... we couldn't allocate a port for RTP instance '0x7f00b81c7028' |
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| 13:21.56 | ovoshlook | only one call |
| 13:22.07 | *** part/#asterisk Marquis42 (~mrbmw@d149-67-95-208.try.wideopenwest.com) |
| 13:22.19 | ovoshlook | tested system, rtp ports - 6000 |
| 13:22.49 | ovoshlook | asterisk 12.5.0 |
| 13:23.34 | mjordan | when you say "rtp ports 6000", what do you mean? |
| 13:23.57 | mjordan | what is the value of rtpstart/rtpend in rtp.conf? |
| 13:24.11 | ovoshlook | start 30 000 end 36 000 |
| 13:24.27 | ovoshlook | 6000 RTP ports at asterisk |
| 13:24.33 | ovoshlook | opened |
| 13:25.27 | mjordan | that error happens for one of two reasons |
| 13:25.29 | ovoshlook | sometime call is Ok |
| 13:25.46 | mjordan | 1. We wrapped around the allowed RTP port range and can't find a free port |
| 13:26.01 | mjordan | 2. An error occurred while opening the port |
| 13:26.29 | mjordan | If you don't have a massive number of ports opened - which can be confirmed using netstat or similar tools - then my guess is you have a permissions problem somewhere. |
| 13:27.11 | WIMPy | Do you test with a softphone running on the same machine? |
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| 13:35.15 | ovoshlook | all ports opened |
| 13:35.20 | ovoshlook | No firewall |
| 13:36.04 | ovoshlook | this is testing system. only one call goes through |
| 13:36.26 | ovoshlook | so number of calls not more that opened RTP ports |
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| 13:43.43 | ovoshlook | Any Ideas with this issue? |
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| 14:12.18 | Tuju | what's the today's ipsec-tunnel solution on linux? |
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| 14:23.56 | litn | hey guys, I'm getting your call cannot be completed as dialed, trying to figure out what's going on |
| 14:24.02 | litn | inbound calls are working ok but not outbound |
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| 14:26.05 | litn | when I look at the log, it just shows this: http://pastebin.com/Pk7nH3mE |
| 14:26.48 | litn | for making a new call |
| 14:26.53 | litn | that 10.10 is the phone |
| 14:27.04 | litn | but it doesn't show trying to make the call out via our provider... |
| 14:28.07 | newtonr | It looks like what is happening is exactly what you have configured to happen. Perhaps you have another extension that matches a prefix that you are not dialing? |
| 14:28.49 | newtonr | You are hitting 7276373317@from-internal , so if that dialplan exists there, then that is what should be executed |
| 14:29.36 | [TK]D-Fender | litn: 7276373317 <- you have no outbound route to match this number you are dialing |
| 14:30.11 | [TK]D-Fender | litn: And you should be asking in #freepbx since that's what you[re running. |
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| 15:08.24 | seanbright | in asterisk 11, i have asterisk configured such that a local (LAN) SIP device can dial out to an ITSP |
| 15:08.46 | seanbright | in dialplan i set the caller id name and number before i Dial |
| 15:09.09 | seanbright | if i look at the channel from asterisk to the ITSP, it has garbage set on it's caller id |
| 15:09.17 | seanbright | how do i set caller id on that channel? |
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| 15:25.43 | Tuju | seanbright: i do Set(CALLERID(num)=+46${CALLERID(num):1}) |
| 15:26.17 | Tuju | it removes first digit and puts prefix in place. |
| 15:27.14 | *** join/#asterisk zoid_ (~zz@181.29.135.98) |
| 15:27.39 | MaliutaLap | Generally the way to do that - I have a few setups where I need to use Set(CALLERID(num)="") in extensions |
| 15:29.43 | *** join/#asterisk ketas- (~ketas@92.38.190.90.dyn.estpak.ee) |
| 15:32.00 | seanbright | that appears to set the caller id on the local channel |
| 15:32.04 | seanbright | the caller id on the peer channel is not set |
| 15:32.04 | *** join/#asterisk fling (~fling@fsf/member/fling) |
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| 15:33.32 | MaliutaLap | seanbright: it sets the outgoing CID on any Dial() |
| 15:34.05 | MaliutaLap | seanbright: it can also be used to set in CID on an incoming |
| 15:34.40 | *** join/#asterisk bipolar (~bipolar@offsite.guru) |
| 15:34.41 | MaliutaLap | seanbright: if the ITSP resets the outgoing CID there isn't much you can do |
| 15:34.48 | seanbright | it's not the ITSP |
| 15:34.54 | seanbright | the CID is presented correctly to the callee |
| 15:35.13 | seanbright | but if i do a "core show channel SIP/channel-to-itsp" the caller ID is 's' |
| 15:35.21 | MaliutaLap | seanbright: using Dial()???? |
| 15:35.23 | seanbright | or '~~s~~' because i am using AEL |
| 15:35.26 | seanbright | MaliutaLap: yes |
| 15:35.54 | MaliutaLap | seanbright: so you do the Set() before the Dial()??? |
| 15:35.55 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-stfpamouorxeqkco) |
| 15:35.55 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:36.00 | seanbright | yes |
| 15:36.08 | MaliutaLap | seanbright: because that works for me |
| 15:36.26 | seanbright | exten => _NXXNXXXXXX,1,Set(CALLERID(num)=4445556666) |
| 15:36.37 | seanbright | exten => _NXXNXXXXXX,n,Dial(SIP/foobar...) |
| 15:37.22 | seanbright | if i do a 'core show channel SIP/foobar-00000000ad' (or whatever the remote channel is), the Caller ID is not set to 4445556666 |
| 15:37.33 | seanbright | but the callee is presented with the correct CID |
| 15:37.39 | seanbright | the channel is just wrong |
| 15:38.15 | MaliutaLap | seanbright: http://pastebin.com/cirnt1yt <- exert from a working extensions.ael |
| 15:39.32 | seanbright | MaliutaLap: do a 'core show channel FOO' on an established call where FOO is the channel name out to the ITSP |
| 15:39.44 | seanbright | not the local channel name |
| 15:40.03 | seanbright | "local channel" meaning the local sip phone's channel to asterisk |
| 15:40.52 | MaliutaLap | http://pastebin.com/7SV1UZRV more fully |
| 15:41.00 | seanbright | heh |
| 15:41.06 | seanbright | ok, well thanks for your help |
| 15:41.17 | MaliutaLap | it is the correct way |
| 15:41.25 | MaliutaLap | and is proven to work |
| 15:41.25 | seanbright | i am aware of that |
| 15:41.29 | seanbright | i am also aware of that |
| 15:41.33 | *** join/#asterisk stevenm (~stevenm@stevenm.keele.netcentral.co.uk) |
| 15:41.39 | seanbright | but you are either not reading what i am saying, or ignoring it completely |
| 15:41.46 | seanbright | in either case, i appreciate you taking the time to try and help |
| 15:41.53 | *** join/#asterisk tuxx- (~tuxx@pantoff0l.nl) |
| 15:41.54 | seanbright | thank you |
| 15:42.02 | MaliutaLap | what is the exact ael you are using? |
| 15:42.06 | stevenm | Lo, anyone know any USB phones similar to a Cyberphone V651 (i.e. no keypad - just basically a USB sound card in the shape of a handset) ? |
| 15:42.32 | *** join/#asterisk mbowie (~mbowie@162.212.36.9) |
| 15:42.49 | MaliutaLap | seanbright: pastebin the actual ael and maybe I can help further, that and a 'sip debug' would help |
| 15:42.56 | seanbright | heh |
| 15:43.14 | MaliutaLap | "it doesn't work" isn't exactly useful |
| 15:43.40 | MaliutaLap | "Carefully explaining your problem is half the solution." |
| 15:44.25 | seanbright | reading back through the history of this conversation, not once did i say "it doesn't work" |
| 15:44.40 | seanbright | let me try one more time. |
| 15:44.58 | Tuju | seanbright: i hate that too, don't understand why things cannot be discussed in general level without going into details. |
| 15:44.59 | seanbright | 1. my soft phone calls asterisk (a channel is created inside asterisk) |
| 15:45.23 | seanbright | 2. it hits some AEL dialplan in which CALLERID(num) is set to a value, say 4445556666 |
| 15:45.31 | MaliutaLap | Tuju: the devil is in the details |
| 15:45.43 | seanbright | 3. it then uses Dial to dial out an ITSP |
| 15:45.55 | Tuju | MaliutaLap: let it be, it doesn't exclude discussion on general level. |
| 15:46.08 | seanbright | 4. the callee receives the call and - now this is the important part - the caller ID presented to the callee is CORRECT |
| 15:46.21 | seanbright | are we all on the same page so far? |
| 15:46.26 | *** join/#asterisk hexanol (~bibi@modemcable094.94-70-69.static.videotron.ca) |
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| 15:47.13 | Tuju | seanbright: doesn't sound like inventing the black powder yet. :) |
| 15:47.52 | *** join/#asterisk ruied (~ruied@25.172.37.188.rev.vodafone.pt) |
| 15:47.54 | seanbright | NOW... after step 4 above, there are 2 channels inside asterisk |
| 15:48.16 | seanbright | 1. SIP/my-phone-00000001 - the channel from my local phone to asterisk |
| 15:48.27 | seanbright | 2. SIP/my-itsp-00000002 - the channel from asterisk to my ITSP |
| 15:48.41 | *** join/#asterisk gusto (~gusto@2a02:810d:8640:1dd0:221:6aff:feb8:e0b2) |
| 15:48.51 | seanbright | if i do "core show channel SIP/my-phone-00000001" the caller ID is shown as 4445556666 |
| 15:49.04 | MaliutaLap | just as an example I have dealt with situations where, until I did a 'sip debug', the solution of what the CID was set to didn't present itself. This is not something I am unfamiliar with. |
| 15:49.27 | seanbright | if i do "core show channel SIP/my-itsp-00000002" the caller ID is shown as 's' or '~~s~~' or any number of other things that aren't correct |
| 15:49.31 | seanbright | the SIP is FINE |
| 15:49.39 | seanbright | THE OTHER SIDE GETS THE CORRECT CALLER ID |
| 15:49.44 | seanbright | how many fucking times do i have to say that? |
| 15:49.53 | seanbright | this is like troll level 1000 |
| 15:49.56 | seanbright | thank you for your help |
| 15:50.05 | Tuju | seanbright: B-subscriber sees correct caller-id? |
| 15:50.16 | Tuju | so where is the problem then? |
| 15:50.22 | seanbright | haha |
| 15:50.35 | *** join/#asterisk gtjoseph (~gtj0@unaffiliated/gtj) |
| 15:50.42 | seanbright | it took me a while to realize i was being trolled. well played. |
| 15:50.57 | Tuju | if you think so, whatever. |
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| 15:51.07 | Tuju | sorry that i wasted my minutes to you. |
| 15:51.17 | Tuju | won't happen next time. |
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| 15:53.25 | MaliutaLap | sometimes I wonder why I do community level help |
| 15:53.47 | *** join/#asterisk ruied (~ruied@25.172.37.188.rev.vodafone.pt) |
| 15:55.15 | Tuju | with such "bright" thinking capabilities, yes - a caller-id might become big problem. |
| 15:58.02 | MaliutaLap | sometimes I feel like going all [TK]D-Fender - if you don't have 'sip debug' then go away :) |
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| 16:01.06 | Tuju | i wouldn't. |
| 16:01.32 | Tuju | it helps if people can talk and understand generally how sip is supposed to work. |
| 16:02.10 | Tuju | voip is still, pretty new technology for majority and it doesn't hurt to get basics right. |
| 16:02.23 | *** join/#asterisk bkruse (~Adium@24.42.207.11) |
| 16:04.29 | MaliutaLap | "new"??? I've been doing this for 10 years now :) |
| 16:05.20 | Tuju | indeed. exactly. |
| 16:05.53 | Tuju | i got my first voip phones....don't remember the manufacturer, those became cisco's phone after they bought that company. |
| 16:06.08 | Tuju | maybe it started with s letter....or...hmmm |
| 16:07.19 | Qwell | Selsius |
| 16:07.23 | Tuju | bingo |
| 16:07.56 | Tuju | h.323 if i recall correctly. |
| 16:08.02 | Qwell | SCCP |
| 16:08.20 | Tuju | hmmm...maybe it was the siemes phone that had h.323 |
| 16:08.39 | Tuju | Qwell: it might have been that selsius was already s |
| 16:08.43 | Tuju | Qwell: it might have been that selsius was already skinny protocols |
| 16:09.30 | Qwell | Selsius created SCCP. They were definitely SCCP. |
| 16:09.33 | Tuju | http://www.cisco.com/web/about/doing_business/corporate_development/acquisitions/ac_name/about_cisco_acquisition_names_list.html |
| 16:09.44 | Tuju | Selsius is a leading supplier of network PBX systems for high-quality telephony over IP networks. Selsius' technology will enable Cisco to accelerate the transition from conventional, proprietary circuit-switched PBXs to multi-service, open LAN systems capable of enabling the next step in data/voice integration. |
| 16:10.16 | Tuju | http://www.cisco.com/web/about/doing_business/corporate_development/acquisitions/about_cisco_acquisitions.html pretty exhaustive list there. |
| 16:10.32 | Tuju | like a death-star or black hole or smthing |
| 16:11.01 | MaliutaLap | that's what happened recently to linksys :) |
| 16:11.24 | MaliutaLap | cisco didn't half rape their voip section |
| 16:11.25 | Tuju | i've tried to have a word or two about phones on #cisco channel but i think i've never seen anyone there who knows about phones. |
| 16:11.37 | Qwell | Recently? That was like 10 years ago. |
| 16:12.06 | MaliutaLap | Qwell: well it was after my Bone Marrow Transplant |
| 16:12.06 | Tuju | they also ate sipura, i think it was a company that made good ATA-boxes. and instant neckshot for that product. |
| 16:12.24 | MaliutaLap | Tuju: true |
| 16:12.46 | MaliutaLap | although I never really did like the sipura call routing stuff |
| 16:13.05 | Tuju | i was too late to get the first box alltogether. |
| 16:13.08 | MaliutaLap | #cisco is good for ios stuff |
| 16:13.20 | Tuju | and career-chit-chat. |
| 16:13.32 | MaliutaLap | I still use cisco phones and switches |
| 16:13.34 | Tuju | but that's not very technical for most people. |
| 16:14.01 | MaliutaLap | true, I do mostly use other sources for my ios troubles |
| 16:14.06 | Tuju | MaliutaLap: i gave up with switches, have HP all over nowdays. don't really know what I'm missing thou. |
| 16:14.06 | MaliutaLap | when they happen |
| 16:14.28 | MaliutaLap | Tuju: I like to keep my hands dirty |
| 16:14.55 | Tuju | yep, that happens very easily with ios-based configuring :) |
| 16:15.17 | Tuju | i also get fustrated and pissed when working with deadline. |
| 16:15.17 | MaliutaLap | lol |
| 16:15.44 | Tuju | 99% of moving parts on ios-based systems are not there for you, don't need 'em. |
| 16:16.02 | MaliutaLap | Oh, I develop terrets and multiple personality symptoms when on a deadline |
| 16:16.07 | Tuju | and once in a while when you do need something complex, you find that there is a bug on that feature. |
| 16:16.24 | MaliutaLap | that's the same with all software |
| 16:16.36 | Tuju | i once burned myself with cisco ISDN lines badly. |
| 16:16.56 | MaliutaLap | the best thing about OSS is you can fix 99.99% of those bugs |
| 16:17.00 | Tuju | a lot of people went to see that box and all said that it was correctly done. just didn't work so. |
| 16:17.07 | Tuju | indeed. |
| 16:17.27 | MaliutaLap | I once broke the apache community with mod_rewrite issue |
| 16:17.31 | Tuju | or you can ask someone, ecosystem is likely more open and willing to listen. |
| 16:17.51 | Tuju | did you get to keep all the pieces? :) |
| 16:17.55 | MaliutaLap | turns out the solution "fell out" of the docs back in 1.1 days, this was in 2.0 |
| 16:18.11 | MaliutaLap | I tried :) |
| 16:19.07 | Tuju | is generating prime background and it's taking ages. :-/ |
| 16:19.46 | Tuju | why cisco doesn't buy dlink and give it a neckshot? everyone would appreciate that. |
| 16:19.50 | MaliutaLap | Tuju: not dong large number theory in your head? |
| 16:20.01 | Tuju | nope, vpn's |
| 16:20.18 | MaliutaLap | kill tp-link first! |
| 16:20.32 | Tuju | not the Kenny? |
| 16:20.47 | MaliutaLap | ok, tp-link second then |
| 16:21.13 | MaliutaLap | tp-link in the network? I don't want to know |
| 16:21.37 | Tuju | what is tp? toilet paper? this one (on deck) http://ayay.co.uk/backgrounds/historical/world_war_two/1944-Normandy-landing.jpg |
| 16:21.53 | coppice | what's wrong with tp-link? |
| 16:22.19 | Tuju | i always wondered what they used when got inland if the tp was left on landing craft. |
| 16:22.22 | MaliutaLap | coppice: it's like herpes |
| 16:22.38 | MaliutaLap | even their windows drivers don't work |
| 16:22.40 | *** part/#asterisk gtjoseph (~gtj0@unaffiliated/gtj) |
| 16:23.31 | coppice | dunno about that. their linux drivers have never given me problems, and their routers are very reliable |
| 16:23.35 | MaliutaLap | coppice: the correct question is "what's right with tp-link" :) |
| 16:23.55 | MaliutaLap | coppice: okaaaaaay |
| 16:24.08 | coppice | I've had far less trouble with tp-link products than with most other makes |
| 16:24.14 | MaliutaLap | coppice: I've yet to have a good experience with tp-link |
| 16:24.52 | coppice | they are the world's biggest customer for networking chips, so if they aren't getting the attention of the silicon vendors I don't know who would |
| 16:25.02 | MaliutaLap | netgear works, intel works, tp-link normally has huge issues |
| 16:25.40 | coppice | netgear stuff I've used needs resetting a couple of time a week. intel has be rather variable |
| 16:25.55 | coppice | s/be/been |
| 16:26.12 | MaliutaLap | I find intel the most reliable |
| 16:26.46 | coppice | for motherboard stuff, probably |
| 16:27.11 | MaliutaLap | well that is how you get most intel wifi stuff |
| 16:27.38 | *** join/#asterisk lorsungcu (~anonymous@204.17.231.116) |
| 16:27.55 | coppice | I guess their router business has completely gone now, but that used to be uiffy |
| 16:28.46 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-tuykxaisgzxvgvvm) |
| 16:28.47 | Tuju | last one i bought was used cisco 3620, still have it. |
| 16:29.01 | MaliutaLap | most people have seen their router businesses disappear |
| 16:29.19 | coppice | not tp-link. their's is huge :-) |
| 16:29.27 | Tuju | nowdays i would but advanced MCA cards |
| 16:29.39 | MaliutaLap | trying to remember the units I have sitting at the end of my dining table |
| 16:29.57 | MaliutaLap | would use J if I could get someone to spend the $$$'s |
| 16:30.24 | Tuju | http://en.wikipedia.org/wiki/Advanced_Mezzanine_Card |
| 16:31.05 | MaliutaLap | like I prefer HP servers over Dell |
| 16:31.42 | MaliutaLap | there are 2 pieces of Dell kit I like ... monitors and one specific keyboard model :) |
| 16:32.41 | MaliutaLap | gah! people are stupid |
| 16:32.59 | *** join/#asterisk ruied (~ruied@25.172.37.188.rev.vodafone.pt) |
| 16:33.06 | Tuju | http://www.advantech.com/products/1-2JKLPJ/MIC-5401/mod_36C21D71-4A86-40D1-829B-EFFB49F61E5F.aspx does that open to you? |
| 16:33.08 | MaliutaLap | just looked at the console of a specific * install - people not dialing complete numbers |
| 16:33.51 | Tuju | http://www.picmg.org/product-showcase/?search_query=voip&term=&company=&submit=Submit |
| 16:35.47 | Tuju | http://www.nateurope.com/ that looks very interesting. |
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| 16:37.46 | Tuju | heads to sauna for prime-number-generation |
| 16:39.27 | MaliutaLap | I should head to bed - it's 02:40 and I have about 10 hours 'til demanding female show up. How I put up with all the "grope my tits" I'll never know :) |
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| 17:19.46 | adrellias | hey got a quick question, we want to jump a call from a que to a outbound sip trunk, if there is no answer |
| 17:20.22 | adrellias | is the correct way to do a Dial(SIP/outbout/number,20) ? |
| 17:20.27 | adrellias | should that work / |
| 17:21.08 | adrellias | exten => quename,2,Dial(SIP/sip_outbound/0012345667778,20) |
| 17:21.11 | adrellias | ? |
| 17:25.38 | *** join/#asterisk darkdrgn2k (~darkdrgn2@209.90.253.66) |
| 17:25.40 | darkdrgn2k | <PROTECTED> |
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| 17:57.35 | *** join/#asterisk cj (~cjac@2607:ff08:f5:3a::3) |
| 17:58.32 | cj | hey guys |
| 17:58.59 | cj | for an iax2 peering session to be established, do both peers need to see one another as the IP to which they register? |
| 17:59.15 | *** join/#asterisk AAmit (~amit@1.39.13.197) |
| 17:59.53 | cj | I have a multi-path route from sip0 to sip1@172.16.78.1 over a set of GRE tunnels and a reverse over the same GRE tunnels from sip1 to sip0@100.65.12.78 |
| 18:00.37 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 18:01.14 | cj | the GRE tunnels have IP addresses 100.65.43.[1,2], 100.65.43.[5,6] and 100.65.43.[9,10], and these are the IPs on the return packet |
| 18:02.18 | cj | so sip1 has register => sip1:y3rm0m@100.65.12.78 |
| 18:02.34 | cj | and permit=100.65.43.0/255.255.255.240 |
| 18:03.24 | cj | while sip0 has register => sip0:y3rm0m@172.16.78.1 |
| 18:03.35 | cj | and permit=100.65.43.0/255.255.255.240 as well |
| 18:05.32 | cj | is it even possible to balance the traffic over these GRE tunnels this way? |
| 18:08.01 | cj | I guess I could register on each IP: |
| 18:08.02 | cj | <PROTECTED> |
| 18:08.05 | cj | <PROTECTED> |
| 18:08.08 | cj | <PROTECTED> |
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| 18:46.35 | adrellias | anyone around ? |
| 18:46.37 | adrellias | ping |
| 18:46.40 | adrellias | :) |
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| 18:47.22 | WIMPy | feels triangular today |
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| 19:04.51 | AAmit | adrellias, Hi |
| 19:07.59 | AAmit | adrellias, Youu there??????? |
| 19:25.45 | AAmit | Hi All |
| 19:25.56 | AAmit | Anybody......... |
| 19:26.49 | [TK]D-Fender | Many people don't feel obligated to respond to people just walking in and saying "hi". |
| 19:27.18 | [TK]D-Fender | If you have something you wanted to discuss people are much more likely to respond once you've at least started a topic |
| 19:27.45 | MaliutaLap | also "Anybody" isn't here right now - check the listing of peoples with a "presence" in the channel :) |
| 19:28.09 | AAmit | [TK]D-Fender, LOL :) |
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| 20:29.09 | *** join/#asterisk kraptv (~kraptv@unaffiliated/kraptv) |
| 20:29.45 | kraptv | Guys - I'm mostly configured with dahdi, but I need a bit of help. I've got a TDM400p with two FXO cards connected to the PSTN. |
| 20:30.33 | kraptv | I can query the card with dahdi show status |
| 20:30.50 | kraptv | I do NOT see my two idle channels when I do a "dahdi show channels" however. |
| 20:32.53 | marceloamorim | dahdi show channels you should see the state |
| 20:33.07 | marceloamorim | and if you need more informations you can do dahdi show channel 1 |
| 20:33.14 | marceloamorim | 1 is the number of your channel |
| 20:33.43 | marceloamorim | kraptv: sorry if this was not what you wanted |
| 20:34.25 | kraptv | Yeah, sadly, I don't see any channels available. |
| 20:34.35 | kraptv | I'll dig in a little more. |
| 20:35.02 | marceloamorim | try to find the device on dmesg |
| 20:35.24 | kraptv | Yeah, I find it. I think I need to configure dahdi-channels.conf. (hmm...) |
| 20:35.48 | marceloamorim | if you are using the linux, the dahdi_genconf works pretty well |
| 20:35.56 | kraptv | So, looks like dahdi-channels.conf is configured automatically - seems like it's fine there. |
| 20:36.01 | kraptv | yeah, I did dahdi_genconf. |
| 20:36.14 | kraptv | and definitely the right signalling, etc. |
| 20:36.41 | *** join/#asterisk tuxx- (~tuxx@pantoff0l.nl) |
| 20:37.19 | kraptv | when I do a "dahdi show channels" it doesn't show any channels, but should it show the channels and just report them as idle? |
| 20:38.35 | *** join/#asterisk tuxx- (~tuxx@2a01:7c8:aab5:336:0:1c1:c0c4:c014) |
| 20:41.40 | kraptv | Can I share my /etc/dahdi/system.conf files on pastebin? |
| 20:41.55 | kraptv | and chan_dahdi and dahdi-channels config files in /etc/asterisk? |
| 20:42.56 | kraptv | system.conf is http://pastebin.com/YUyHFWnX |
| 20:44.58 | kraptv | chan_dahdi.conf is http://pastebin.com/ys7qF6J2 |
| 20:45.43 | kraptv | and finally my dahdi-channels.conf is here |
| 20:45.44 | kraptv | http://pastebin.com/vBvBSwXT |
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| 20:46.10 | kraptv | System sees the card, asterisk sees dahdi. |
| 20:46.49 | kraptv | channels are listed in the right configs (did I miss something?) |
| 20:48.10 | kraptv | Man. It's doing my head in. I love asterisk, too. |
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| 20:54.07 | kraptv | Alright. doing dahdi_monitor, I see the rings visually. |
| 20:54.35 | kraptv | dahdi_monitor 1 -vv definitely showed the visual 'spikes' when I rang the line. |
| 21:06.33 | kraptv | Alright. Figured it out. |
| 21:06.57 | kraptv | Under asterisk, I did a "dahdi restart" which complained about a lack of an #include file. |
| 21:08.18 | *** join/#asterisk techshell (~techshell@ec2-176-34-130-192.eu-west-1.compute.amazonaws.com) |
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| 21:09.42 | techshell | experts ... i am running an asterisk |
| 21:09.43 | techshell | asterisk18-1.8.30 |
| 21:09.50 | techshell | in EC2 |
| 21:10.00 | techshell | and having terrible issues with meetme |
| 21:10.43 | techshell | the issues are all intermittent and related to call quality. |
| 21:11.29 | techshell | taking a pcap trace confirms that rtp stream for one caller goes into asterisk clean ... mixed audio out to another caller comes out stuttering |
| 21:11.55 | techshell | running tcpdump directly on the host .. not a span port or mirror. |
| 21:12.18 | techshell | confirmed for sure this is not a network related problem .. 100% |
| 21:12.49 | techshell | running dahdi 2.9.1 |
| 21:13.07 | techshell | any help will much much appreciated |
| 21:13.08 | techshell | thanks |
| 21:15.47 | techshell | I have been using asterisk since 1.2 and I would consider myself and advanced user, I have tried everything but cannot for the life of me figure out why I am seeing this behaviour. |
| 21:16.31 | techshell | there is a chance the issue is caused by Amazon .. but how do I confirm. |
| 21:16.48 | techshell | look forward to hear from ya |
| 21:18.24 | WIMPy | Sounds like bad timing, which seems pretty normal for virtual systems. |
| 21:18.44 | WIMPy | I'd upgrade away from meetme. |
| 21:19.45 | lvlinux | hey WIMPy should he try the ConfBridge in 1.8 just to see if it makes any diff? It doesn't rely on DAHDI like meetme does it? |
| 21:19.55 | Qwell | ConfBridge requires at least Asterisk 11 (which you should be very highly considering at this point) |
| 21:20.17 | lvlinux | i thot a very featureless Confbridge was in 1.8? |
| 21:20.18 | WIMPy | No, it doesn't, but it should be 11. |
| 21:20.39 | Qwell | lvlinux: It's not useful until 11. Or maybe it was 10 (which is no longer supported, so my statement stands) |
| 21:21.34 | lvlinux | yes i know it's not as far as features, but would it not work enough for him to try it in 1.8 without an upgrade---that way he could determine if it was indeed a timing problem, without going to 11 immediately---which he should do of course asap... |
| 21:22.01 | Qwell | lvlinux: No need to test. ConfBridge will certainly fix such issues. |
| 21:22.15 | lvlinux | ah ok. |
| 21:23.02 | techshell | righto .. thanks folk |
| 21:24.00 | techshell | changing to ConfBridge and Asterisk11 is quite a bit of work for me. |
| 21:24.34 | file | it works better, but if your timing is still wacked it can only help so much |
| 21:24.38 | file | like - really wacked |
| 21:25.08 | techshell | other than dahdi is there any other timing source I could use |
| 21:25.19 | techshell | I have no hw .. all sip |
| 21:25.33 | techshell | so shouldn't need dahdi |
| 21:25.35 | WIMPy | You need dahdi for meetme. |
| 21:26.03 | Qwell | dahdi doesn't provide timing for meetme, it provides mixing. |
| 21:26.23 | techshell | i had a similar system running in EC2 for the last 3 yrs without any problems |
| 21:26.27 | WIMPy | And timing. |
| 21:27.22 | techshell | only diff is that the new instance uses hvm while the old uses pv |
| 21:27.45 | techshell | would tweaking kernel hz timers help in this case |
| 21:28.00 | techshell | old system that worked was running on ubuntu |
| 21:28.05 | techshell | this one runs on Centos |
| 21:28.57 | WIMPy | Possible. No way to know without trying. |
| 21:29.13 | WIMPy | But I doubt dahdi uses that timer. |
| 21:30.06 | techshell | me too .. dahdi i suspect either takes timing from line card (isdn) or some dummy psudo driver |
| 21:30.24 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-jcciqypjecqlfsvd) |
| 21:34.22 | WIMPy | It's best to have hardware, obviousely. |
| 21:37.04 | techshell | yeah ... but not very 2014 is it. |
| 21:37.23 | cunningpike | techshell: Have you gotten timing numbers from your pseudo inerface? |
| 21:37.49 | WIMPy | Neither is meetme. |
| 21:38.06 | techshell | agreed |
| 21:38.26 | techshell | I am using Thirdlane to provide web interface for my users |
| 21:38.44 | techshell | usability is also a factor |
| 21:39.08 | techshell | I wish I can tell them you can't setup followme or see whose on the conference call |
| 21:39.17 | techshell | would make my life sooo much easier |
| 21:39.56 | techshell | @cunningpike whats the best way to do that? |
| 21:40.12 | techshell | @cunningpike dahdi_test? |
| 21:40.28 | cunningpike | techshell: Yup. |
| 21:40.55 | techshell | --- Results after 36 passes --- |
| 21:40.55 | techshell | Best: 99.999% -- Worst: 99.605% -- Average: 99.947849% |
| 21:40.55 | techshell | Cummulative Accuracy (not per pass): 99.994 |
| 21:41.15 | techshell | I guess i'll need to run for longer |
| 21:41.23 | techshell | but here is sample |
| 21:42.26 | cunningpike | techshell: Those numbers don't indicate a timing issue to me... |
| 21:42.27 | techshell | @cunningpike note issue is intermittent. |
| 21:42.39 | cunningpike | Oh, I love those :D |
| 21:43.02 | techshell | @cunningpike yeh ... the best ones. |
| 21:43.05 | cunningpike | Is this all SIP? |
| 21:43.13 | techshell | @cunningpike yep 100% |
| 21:43.53 | cunningpike | So it's not transcoding. Hmm. |
| 21:44.31 | techshell | @cunningpike no funky codes either ... everything is g711 |
| 21:44.48 | techshell | @cunningpike *codecs |
| 21:47.22 | techshell | The old system that worked for the last 2-3 yrs was 1.8.10 |
| 21:47.38 | techshell | this one is 1.8.30 |
| 21:48.27 | cunningpike | Well, I'd go back to 1.8.10 to see if the problem stays. |
| 21:48.43 | cunningpike | Failing that, I'd try the ConfBridge, as other people suggested earlier. |
| 21:49.10 | techshell | whats really strange is all other calls are perfect |
| 21:49.47 | cunningpike | So it must be something in the MeetMe mixing then. |
| 21:49.59 | techshell | looks like it to me |
| 21:50.10 | cunningpike | But I'd try those steps to try and isolate the issue. |
| 21:51.15 | techshell | what i'll do for now is resurrect the installation that worked - 1.8.10 - and send conf calls there, just to alleviate the issues. |
| 21:51.24 | techshell | then I'll re-evaluate the options. |
| 21:52.03 | techshell | nice thing in AWS is just restore from a snapshot and away it goes .. lazy but aren't we all |
| 21:53.31 | cunningpike | Any sysop worth their salt is lazy. It's what drives us to automate stuff ;-) |
| 21:54.31 | techshell | only prob I have is time .. I am doing this in the middle of umpteen other - completely unrelated - client work. |
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| 21:54.50 | techshell | I am sure you can relate .. |
| 21:55.12 | techshell | the queue is always full .. can never get it to zero so you can do what you really *want* to do. |
| 21:55.17 | techshell | righto ... |
| 21:55.52 | techshell | many thanks @cunningpike and @WIMPy |
| 21:55.55 | techshell | have smashing weekend |
| 21:56.01 | cunningpike | You too! |
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| 22:32.57 | darkdrgn2k | so anyone else know if there is any other way to stop asterisk from killing som inband tones from my alarm panel? |
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