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| 00:35.35 | *** join/#asterisk asteriskmonkey1 (~philip@74-51-38-204.telnetcommunications.com) |
| 00:38.26 | asteriskmonkey1 | anyone know why asterisk wouldnt process the next line after a faxrecieve command it seems in older versions this was possible but in 1.10 it stops porcessing after executing the faxrecieve |
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| 01:01.56 | [TK]D-Fender | asteriskmonkey1: Those apps always disconnect after completion. You always had to use "h" for the cleanup |
| 01:03.49 | asteriskmonkey1 | yeah just realized i couldnt use h, inside the macro |
| 01:04.04 | asteriskmonkey1 | thanks :) |
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| 06:42.18 | ChannelZ | WTF |
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| 07:51.23 | ChannelZ | OooOOo... make menuconfig is all pretty now |
| 07:57.45 | ChannelZ | oh.. hmm.. actually this must depend on some other package |
| 08:00.37 | ChannelZ | Ahhh... newt |
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| 09:04.24 | doolittlework | any one know if asterisk can be used as a hosted pabx solution high end 1000 plus users |
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| 09:17.01 | doolittlework | hi |
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| 09:30.43 | jkroon | are there any people from digium around? I'm stuck with technical support and really need one of the devs to take a look at an issue. |
| 09:32.10 | WIMPy | 1. It's weekend and 2. |
| 09:32.14 | WIMPy | ~ask |
| 09:32.15 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 09:35.21 | jkroon | WIMPy, it's not an issue digium wishes me to discuss in public, nor does the technical support guy seem to understand the gravity of the situasion. |
| 09:35.58 | jkroon | thus, i need to point someone from digium with an understanding of how things actually work to a specific support issue, and handle it from there. |
| 09:36.22 | jkroon | but i'll wait till monday and then try again. |
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| 09:47.24 | WIMPy | Ok, so some mysterious thing someone already needs to know about. |
| 09:48.10 | cusco_ | why the mystery? |
| 09:48.29 | WIMPy | Is it still safe to have Asterisk connected to the internet? |
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| 11:50.49 | Roelt | I would switch to, http://www.youtube.com/watch?v=K_q_k_QDraQ better security.. :) |
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| 12:30.41 | jkroon | WIMPy, yes, it is, but i don't think digium would appreciate me just releasing this info. it's not asterisk itself but it pertains to what I suspect is a reasonable portion of their income stream. |
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| 13:32.14 | drmessano | I hope this isn't about G729 |
| 13:36.38 | *** join/#asterisk vassilux (~vassilux@31.35.29.247) |
| 13:40.07 | vassilux | hi alls, I use Asterisk-GUI and want to add MixMonitor for outgoing and incoming calls by a macro for example. I look into pbx.js and pbx2.js and found the place for outgoing but can't see the place where I can change for incoming calls. Any idea ? |
| 14:04.28 | cusco_ | hmm... hi |
| 14:05.36 | cusco_ | vassilux: I never used GUI, but how about at the s extension in the context the call come sin? |
| 14:05.45 | cusco_ | or.. just before Queue() cmd ? |
| 14:05.51 | cusco_ | ... |
| 14:05.52 | cusco_ | well |
| 14:06.06 | cusco_ | <PROTECTED> |
| 14:06.09 | cusco_ | NoOp(zzzzzzzzz ${Contacto} ${LEN(Contacto)}); |
| 14:06.12 | cusco_ | what is wrong here ??? |
| 14:06.58 | cusco_ | nvm |
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| 14:33.50 | jkroon | drmessano, ... ? |
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| 17:16.36 | xoveruk | what does the (S) mean in the CLI command 'iax2 show peers'? |
| 17:21.29 | xoveruk | is there anyway of test iax2 authentication from the CLI rather than trying to place a call? |
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| 17:44.54 | vassilux | cusco, I can do it on the dialplan and it works I asked just for one who worked with GUI , sorry I my PC was down |
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| 18:59.09 | TyGuy | does anybody know if 64 bit windows drivers exist for the TDM410? |
| 19:01.03 | jkroon | racks his head wondering why anybody would even want to try and use a TDM card on Windows ?!? |
| 19:01.32 | jkroon | but to answer your question - no, i don't even think windows drivers exist for the digium cards ... at least, not as far as I know. |
| 19:03.47 | TyGuy | ok cool. wanting to try to avoid reinstalling windows |
| 19:11.54 | jkroon | ok, but what exactly are you trying to use the TDM card for? |
| 19:14.21 | jkroon | ok, i can't find any references to windows drivers for the TDM cards anywhere, why not just leave the device unconfigured or do you need access to the functionality from windows? |
| 19:15.04 | TyGuy | I'm just going to install linux lol |
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| 19:36.51 | [TK]D-Fender | Digium doesn't make windows drivers' |
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| 20:14.39 | gusto | hey does the stun server also check for the RTP port number and reports it back to tell the peer where the RTP stream will be at? |
| 20:15.27 | jkroon | gusto, no, a stun server cannot do that. |
| 20:15.52 | jkroon | not sure about res_stun_monitor though, but I don't think so at this stage. |
| 20:16.27 | gusto | so a stun server reports only the public ip address, right? |
| 20:16.53 | jkroon | no, from what I recall it reports both the IP and port it received the request from. |
| 20:17.24 | jkroon | however, depending on the type of NAT this is not sufficient for determining the external rtp port (fully symmetric NAT this won't work) |
| 20:17.25 | gusto | so it theoretically could be also used for finding out which RTP port it will be |
| 20:17.52 | jkroon | in specific types of NAT (which are not particularly common from my experience) |
| 20:18.53 | gusto | so how they do it then? |
| 20:19.07 | jkroon | who's they? |
| 20:19.21 | gusto | the SIP client applications for example |
| 20:19.53 | jkroon | they don't. at least, not reliably. they typcially use rport, and have the registrar/proxy deal with re-routing the rtp. |
| 20:19.55 | gusto | you have a port set to say 7078 but it comes out as external port 8078, how does he find out what number it was? |
| 20:20.31 | jkroon | http://en.wikipedia.org/wiki/Network_address_translation |
| 20:20.55 | [ctrl][alt][del] | good day all |
| 20:20.58 | gusto | so it was only luck that my asterisk got the package |
| 20:21.01 | jkroon | i highly recommend you read that, specifically try and understand the different types of NAT |
| 20:21.12 | jkroon | no, it has nothing to do with luck. |
| 20:21.34 | *** join/#asterisk ilj (~ilj@sourcemage/grimoire/apprentice/ilj) |
| 20:22.54 | jkroon | SIP tries to be clever at dealing with NAT, and there are various RFCs and strategies to deal with it. rport seems to do the trick quite nicely. |
| 20:23.15 | gusto | well, but rport is the easiest one |
| 20:23.27 | gusto | it just says that you take the port the connection came from |
| 20:23.52 | jkroon | no, not quite, the IP+port the first rtp frame came from |
| 20:24.06 | gusto | that would be that RTP would connect to a port that was given by asterisk and through rtport would find out where it came from |
| 20:24.14 | jkroon | but if I'm not mistaken, the IP should be restricted to the same IP the SIP packet came from. |
| 20:24.42 | jkroon | yes, and what if that matches the advertised sdp description? |
| 20:24.48 | gusto | sip is not a problem, because on sip you have always one server with static port |
| 20:25.17 | jkroon | then it becomes possible to redirect the traffic directly from another end-point if I'm not mistaken, but again, I may be wrong. [TK]D-Fender might know better. |
| 20:25.48 | jkroon | gusto, no, SIP is a peer-to-peer protocol. in many cases you require a registration, but yea, you're right, the SIP portion of the whole mess isn't the major problem. |
| 20:26.06 | jkroon | but the SIP part carries descriptions regarding the RTP, which is where the issues stem from. |
| 20:26.06 | gusto | it could go like that ... asterisk says on SIP that the peer should hit the port 12000 with his port "whatever port" 1:1, so say 7078 <--> 12000 |
| 20:26.43 | *** join/#asterisk vlad_starkov (~vlad_star@ppp91-79-78-104.pppoe.mtu-net.ru) |
| 20:26.51 | gusto | so it could find out, that the peer hit the port 12000 with 8078 and could then choose 8078 as the port, on the other side the peer stays at his 7078 1:1 conf |
| 20:26.53 | jkroon | yea, but if it doesn't match the sdp then you can't redirect because you have no way of knowing whether you're dealing with a full-cone, restricted or symmetric NAT |
| 20:27.18 | gusto | do not care so much about the NAT type. that is nothing i can change |
| 20:27.46 | jkroon | is both your endpoints behind NAT? |
| 20:27.55 | gusto | no |
| 20:28.02 | jkroon | or do you only care about RTP from behind a single NAT'ed host? |
| 20:28.15 | gusto | i do not care at all |
| 20:28.48 | gusto | i only just had a call with a friend and he/she was behind a carrier NAT (mobile broadband) and it worked -> to my surprise |
| 20:28.49 | jkroon | then why ask the question about nat? |
| 20:29.02 | jkroon | ah ok |
| 20:29.09 | gusto | you was the one who brought up NAT |
| 20:29.21 | jkroon | you asked about stun which implies nat :) |
| 20:29.26 | gusto | no |
| 20:29.47 | gusto | with stun he only knows which host it was, so that he does not mix that up |
| 20:29.49 | jkroon | explain why not please |
| 20:30.11 | gusto | what could happen when there would be more peers making a call at the same time |
| 20:30.33 | jkroon | they would be allocated different source port numbers if it's to the same outbound host |
| 20:30.34 | gusto | but that could be also hold apart through some RTP header messages |
| 20:30.55 | gusto | jkroon: yes, you are right, but that would happen for SIP |
| 20:31.05 | jkroon | and rtp too |
| 20:31.22 | gusto | but what if two are going to make a call and the nat mixes that up, so that the first one ends up on that o... |
| 20:31.25 | gusto | no |
| 20:31.47 | gusto | he would not mix that up, becaouse asterisk would give them different port numbers to hit at |
| 20:31.49 | jkroon | initial return rtp would probably get dropped unless rport was specified, as soon as the first outbound rtp goes from behind nat a mapping would be created and rtp would be able to flow in both directions. |
| 20:31.50 | gusto | OK |
| 20:32.17 | jkroon | that is indeed true yes :) |
| 20:32.22 | gusto | you are thinking too complicated |
| 20:32.32 | gusto | they register from different ports at SIP |
| 20:32.33 | jkroon | but asterisk would not know where to send the return rtp traffic :) |
| 20:32.58 | gusto | and asterisk chooses two ports, and gives each one in the SIP messages through, so they can not mix up the calls |
| 20:33.10 | gusto | sure it knows |
| 20:33.16 | jkroon | how? |
| 20:33.19 | gusto | it sends it to the same port |
| 20:33.30 | gusto | that's what i found out today |
| 20:34.02 | jkroon | yes, with rport enabled it does. otherwise it tries to use what's in the sdp description, which in the case you described would be wrong. |
| 20:34.15 | gusto | when some idiot comes in with rport - it takes the port which was then found out through hitting - as the RTP port and sends both traffic to the same port |
| 20:34.34 | jkroon | both traffic? |
| 20:34.43 | gusto | well, but we are talking about rport now, right? |
| 20:34.48 | jkroon | yes |
| 20:35.08 | gusto | and he knows already that someone is connection from behind nat, so he does not create two port streams then |
| 20:35.21 | gusto | i saw it on the debug |
| 20:35.32 | gusto | 'rtp set debug on' |
| 20:36.48 | jkroon | * would still open two rtp ports on it's side (one for rtp and one for rtcp, with the even port for rtp - as advertised in it's sdp, +1 for rtcp). |
| 20:36.50 | gusto | http://pastebin.com/7fzQCWC9 |
| 20:37.20 | jkroon | with rport it will wait for something to hit the rtp port to establish the other endpoint |
| 20:37.37 | jkroon | without rport it will revert to using the sdp description from the peer for the remote endpoint |
| 20:37.38 | gusto | jkroon: yes, but i do not care about how it is handeled by asterisk internally and also not about how it is handeled by NAT, because i can not change that thing |
| 20:38.05 | jkroon | then what is it that you want to know? |
| 20:38.22 | gusto | 1.) take a look at the paste |
| 20:38.35 | jkroon | already did |
| 20:38.37 | gusto | 2.) we pretty much found out about the basic functionality |
| 20:39.03 | jkroon | ok ... |
| 20:39.22 | gusto | 3.) that paste was my computer - not the peer behind nat we were taling about - that would have different port numberg |
| 20:39.26 | gusto | number |
| 20:39.54 | gusto | however, it appears to the SIP debug what is the peer's port number |
| 20:41.26 | jkroon | ok, you're now confusing the daylight out of me. |
| 20:42.15 | gusto | Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) |
| 20:42.18 | gusto | Peer audio RTP is at port 62.216.201.80:7078 |
| 20:42.31 | gusto | that's inside the SIP debug |
| 20:43.05 | gusto | the question is, why |
| 20:43.54 | jkroon | is your friend with M-net Telekommunikations ? and is he connecting directly to your asterisk server? |
| 20:44.02 | gusto | no no, that's me |
| 20:44.11 | gusto | the friend's debug is gone already |
| 20:45.02 | jkroon | ok, but your debug is sending TO that address, and receiving FROM, so if 62.216.201.80 is YOU then you've got three SIP agents involved? or just two? |
| 20:45.06 | gusto | but i can put some credit on my SIM and experimend as well, with mobile broadband |
| 20:45.27 | jkroon | i wouldn't bother. |
| 20:45.45 | gusto | well, but then i could get a realistic scenario |
| 20:46.09 | gusto | however, only for that purpouse it's not worth |
| 20:46.37 | jkroon | please describe the previous scenario in full, as in all agents involved please, and at what point the SIP and RTP debug stuff was captured. |
| 20:47.09 | gusto | my asterisk server is not on my ip, at least that's not the one i am talking about |
| 20:47.19 | *** join/#asterisk justdave_ (~dave@unaffiliated/justdave) |
| 20:47.29 | gusto | and to test it i connect sometimes from my ip to see if it works or not |
| 20:47.43 | gusto | however, that peer was on t-mobile, so of course had a different ip |
| 20:49.52 | gusto | but this debug is cool, i am going to throw on something |
| 20:50.00 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 20:54.13 | gusto | hm |
| 20:54.21 | jkroon | sip debug ... saved my ass on MANY occasions. |
| 20:54.25 | gusto | it seems that he always opens only one RTP port per call |
| 20:54.42 | gusto | well, me too |
| 20:54.51 | gusto | but i am now looking at rtp debug |
| 20:55.05 | gusto | but it's OK |
| 20:55.13 | jkroon | yea, asterisk doesn't give much feedback re the rtcp port ... |
| 20:55.21 | jkroon | but it's always there +1 on the rtp port. |
| 20:55.39 | gusto | you mean SIP+RTP |
| 20:55.54 | gusto | maybe to asterisk-asterisk he opens SIP+RTP+RTP |
| 20:56.18 | gusto | but i see asterisk-ata/soft-phone always SIP+RTP |
| 20:56.56 | jkroon | no, SIP + RTP + RTCP (SIP describes the RTP and manages/controls the RTP setup/flow/breakdown) |
| 20:57.21 | jkroon | RTCP on the other hand gives feedback re the QUALITY of the RTP stream (jitter, packet loss) |
| 20:57.23 | gusto | however, but that's not an extra port |
| 20:57.46 | gusto | and we are talking about the network ports here now |
| 20:58.01 | jkroon | http://en.wikipedia.org/wiki/RTCP |
| 20:58.10 | jkroon | yes, there is an extra port :) |
| 20:58.16 | gusto | really? |
| 20:58.23 | gusto | so ... how do i monitor that? |
| 20:58.33 | jkroon | run a netstat -nulp next time you have a call ongoing. |
| 20:58.55 | jkroon | tcpdump would be one way, or sip show channels iirc. |
| 21:02.02 | jkroon | i have yet to implement logging the rtcp values (made available as channel variables after the call has been hung up) to my CDRs |
| 21:02.25 | jkroon | should be do-able in the h extension if I'm not mistaken |
| 21:02.43 | gusto | ok |
| 21:02.53 | gusto | but somehow that rtcp port seems to be always rtp+1 |
| 21:03.24 | jkroon | it is |
| 21:03.28 | gusto | so |
| 21:03.30 | gusto | you see |
| 21:03.37 | gusto | there is nothing interesting about that one |
| 21:03.50 | jkroon | except the info it contains is something i'm after :) |
| 21:04.20 | gusto | yes, and good to know for firewall configurations |
| 21:06.15 | jkroon | that really depends on how you configure your firewalls to begin with. |
| 21:09.43 | *** join/#asterisk tamiel (~tamiel@c-69-181-254-90.hsd1.ca.comcast.net) |
| 21:10.35 | jkroon | grr, ok, so let's say I have SIP/x<=>Local/something<=>SIP/y ... now I'd really like to log the rtcp values on both SIP legs ... suspect a better way to do that will be to actually do it from some kind of AMI connection point. |
| 21:14.13 | gusto | well, it works even easier, because it does not 1:1 but asteriskport:foreignpeerrport |
| 21:15.21 | gusto | ok, now we figured out that down to a packer ;-) |
| 21:15.24 | gusto | packet |
| 21:15.58 | gusto | jkroon: what is that AMI? |
| 21:16.08 | jkroon | asterisk manager interface |
| 21:16.21 | gusto | ah |
| 21:17.49 | gusto | btw. did you get that windows 7 has netstat in it's powershell |
| 21:18.16 | gusto | i am running that shit in a virtualbox and i was just trying, and wow, it has netstat :-D |
| 21:21.58 | gusto | and there are a lot of ports open on that ;-) |
| 21:23.35 | jkroon | windows always had netstat, it was just always a second class one. |
| 21:31.13 | gusto | hm |
| 21:31.20 | gusto | since when they had netstat? |
| 21:34.04 | jkroon | at least 98 |
| 21:34.42 | jkroon | at least until 2000 ... which is the last time I seriously used windows |
| 21:57.28 | *** join/#asterisk xoveruk (~rich@196.200.85.125) |
| 21:57.38 | xoveruk | hi |
| 21:57.59 | xoveruk | please, how can I test iax2 channel authenticates to remote server using CLI without placing physical call. |
| 21:59.51 | *** join/#asterisk jsjc (~Adium@27.Red-83-59-180.dynamicIP.rima-tde.net) |
| 22:00.22 | ChannelZ | Shorter: "how can I do something without doing something"? |
| 22:11.54 | [TK]D-Fender | xoveruk: You can't |
| 22:13.23 | xoveruk | ok |
| 22:13.54 | xoveruk | is there anyway that I can show the callerID of the person I am calling when I place my call. |
| 22:14.01 | xoveruk | sendID if you like |
| 22:24.39 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 22:25.46 | xoveruk | please. what does (S) mean when I issue iax2 show peers? |
| 22:56.35 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 23:17.14 | *** join/#asterisk fulcan (~brads@li503-84.members.linode.com) |
| 23:17.40 | fulcan | does asterisk support h.323 video? |
| 23:23.28 | fulcan | http://www.voip-info.org/wiki/view/Asterisk+video this makes no sense and is it current with asterisk 10? -> The H.323 channels (chan_h323, chan_oh323, chan_ooh323) do not allow video calls at this moment; however, the ChangeLog of ooh323 notes for release 0.6: "Added H.263 video codec negotiation handling"? |
| 23:23.33 | *** join/#asterisk DarthExpeditor (~IceChat9@rrcs-71-43-7-26.se.biz.rr.com) |
| 23:25.01 | *** join/#asterisk fergus_ (~fergus@178.124.149.113) |
| 23:26.52 | *** join/#asterisk gusto (~gusto@ppp-62-216-201-80.dynamic.mnet-online.de) |
| 23:41.21 | *** join/#asterisk Z_God (~julius@83.161.151.96) |
| 23:48.21 | *** join/#asterisk xoveruk (~rich@196.200.85.125) |
| 23:48.59 | xoveruk | please, what does the (S) mean when issuing 'iax2 show peers'? |
| 23:51.17 | *** join/#asterisk amessina (~amessina@2001:470:1f11:a4:d6be:d9ff:fe8d:7c1e) |
| 23:52.10 | xoveruk | is there a way that i can make a test call from the CLI? |
| 23:59.18 | WIMPy | xoveruk: 'channel originate' |