| 00:06.53 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 00:35.55 | eppigy | hello |
| 00:35.56 | eppigy | i am dave |
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| 03:10.17 | Schmee | Hi all. Is there any way to force SIP endpoints to have to register before making a call? Asterisk 1.8 (will be replaced with 11.x at some point soon). Endpoints are X-lite with "register with domain" unchecked. |
| 03:11.01 | WIMPy | You could check in your dialplan. |
| 03:15.05 | Schmee | WIMPy: I'm not quite sure what you mean. I don't see anything in extensions.conf (or anywhere else) that actually checks if an endpoint is registered. Authentication is still required, but it happens on a per-call basis, which doesn't list the endpoint in sip show peers |
| 03:15.34 | WIMPy | core show function SIPPEER |
| 03:17.50 | Schmee | would I be looking at the regexten entry? |
| 03:19.13 | WIMPy | Nope. The IP. |
| 03:19.22 | WIMPy | Or port. |
| 03:19.43 | Schmee | neither shows up, unless the endpoint is actually registered |
| 03:20.20 | WIMPy | At least the IP should be 0, probably the port as well. |
| 03:20.25 | WIMPy | Try it out. |
| 03:20.43 | Schmee | the IP shows as Unspecified and the port is 0 |
| 03:21.39 | Schmee | however, fr the registered endpoints, I get an IP and a port |
| 03:21.39 | WIMPy | Sounds usable. |
| 03:24.26 | Schmee | Quick thought. I just found an obscure little entry. Would setting the endpoint as dynamic force it to register? |
| 03:24.49 | WIMPy | That enables it to register. |
| 03:25.13 | WIMPy | If you specify a host, it can't register. |
| 03:25.55 | Schmee | so I'm misinterpreting the config for host=dynamic ? |
| 03:26.42 | WIMPy | Not sure. You already found out that registration has nothign to do with accepting calls. |
| 03:27.56 | Schmee | actually registration is required to accept calls to the endpoint, but they can make calls vai asterisk without being registered |
| 03:28.25 | WIMPy | It's just there to tell them where they can be reached, yes. |
| 03:29.11 | Schmee | that's what I thought. so there's no setting I can use to force an endpoint to register before it will allow calls out? |
| 03:29.22 | WIMPy | no |
| 03:29.46 | WIMPy | Well, that's in. At least from Asterisk point of view. |
| 03:30.02 | Schmee | bummer. |
| 03:30.29 | Schmee | Oh well, it was worth a shot |
| 03:30.37 | WIMPy | You know ow to do it... It's a one liner. |
| 03:30.49 | WIMPy | how |
| 03:32.38 | Schmee | service asterisk stop ? |
| 03:33.15 | WIMPy | Your dialplan. |
| 03:33.38 | Schmee | can the dialplan determine if an endpoint is registerd? |
| 03:34.20 | WIMPy | Err. We already discussed that. And you already found out how. |
| 03:43.52 | Schmee | excuse me while I bash my head repeatedly against the desk. I must have a distinct lack of caffeine in my bloodstream today |
| 03:44.18 | Schmee | If I understand you correctly, the SIPPEER command is something I can actually use in a dialplan? |
| 03:44.58 | WIMPy | Yes. |
| 03:45.10 | WIMPy | But it's a function. |
| 03:45.16 | Schmee | Sorry, I'm just really slow today |
| 03:45.49 | WIMPy | A simple execif peer ip=unknown Hangup |
| 03:46.08 | WIMPy | or goto somewhere else where you tell people to register, but that might be a bag idea. |
| 03:46.37 | WIMPy | Or maybe noy. They need authentication anyway. |
| 03:46.58 | Schmee | unfortunately it's a call center, in a different state to my current location. They managed to get authentication working, but they fogot to tick the register to server part |
| 03:47.37 | Schmee | I'll make the change in a few hours when they knock off for hte day, and when they try to make calls tomorrow, they won't be able to and have to call me instead. |
| 03:48.13 | Schmee | thanks for the assist, and my apologies again for being really slow at getting the idea throug hmy thick skull |
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| 04:38.40 | Schmee | got to go. thanks again for your help WIMPy |
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| 10:02.36 | madax | hi guys, i'd like to record everything on a channel using Mixmonitor, especially when someone else is spying/whispering on the same channels, is that a way to do that plz ? |
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| 12:17.37 | ilia_ | Hello, is there a way to change IP in o= (owner) SDP field with pjsip? My Asterisk 13 uses internal IP, but out provider expect to see external IP there. external_media_address is set up correctly in transport section, but this doen't work. |
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| 12:36.51 | pawiecki | have you encountered teleconference calls dropping after about 15 minutes? |
| 12:39.39 | skrusty | 15 minutes sounds like a sip session timer to me |
| 12:41.10 | pawiecki | but it's only for teleconference, regular calls can last 50 mins without this issue |
| 12:41.39 | skrusty | when you say lasts, what happens? |
| 12:41.43 | skrusty | hangup? |
| 12:42.37 | pawiecki | yes, normal hangup after finished conversation |
| 12:43.02 | skrusty | no, after 15 mins |
| 12:44.22 | pawiecki | i have no logs from it right now, just got the information about the issue |
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| 12:48.50 | pawiecki | where can i look for session timer setting in asterisk? |
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| 13:12.46 | ilia_ | I've found media_address parameter for endpoints, this is what I looked for. Thanks ) |
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| 13:36.50 | johnny_|_ | Hi. How can I retain caller id when "register => ..." forwards call to extension? |
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| 13:38.18 | [TK]D-Fender | it doesn't |
| 13:38.20 | roswell | between register => and an extension, there's some dialplan |
| 13:38.26 | [TK]D-Fender | that line tells the other server where you are. |
| 13:38.37 | [TK]D-Fender | It has no impact whatsoever on callerid |
| 13:38.55 | [TK]D-Fender | They send what they send |
| 13:39.07 | [TK]D-Fender | I recommend looking at your call and seeing what you're doing with it |
| 13:39.20 | [TK]D-Fender | ~pm |
| 13:39.20 | infobot | rumour has it, pm is project manager, or private message, or perl mongers, or pathetic moron: when you see someone say pm, they're asking if you think that they're a pathetic moron, or something you don't do without asking permission |
| 13:39.23 | [TK]D-Fender | ^^ Your friend |
| 13:39.29 | [TK]D-Fender | ~pb |
| 13:39.29 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 13:39.30 | [TK]D-Fender | rather |
| 13:39.42 | roswell | my telepathy.so tells me he's asking about incoming calls |
| 13:42.16 | [TK]D-Fender | me reading what he wrote told me that.... |
| 13:42.24 | johnny_|_ | roswell, [TK]D-Fender: yes I am asking about incoming calls. I have "register => .../101" set to linphone. And when I call myself using ekiga sip, that call is forwarded to SIP/101 but callerid says: 101 instead of me@ekiga.net... |
| 13:42.51 | [TK]D-Fender | johnny_|_: Show us the call..... |
| 13:43.02 | johnny_|_ | [TK]D-Fender, roswell: sorry for my lingo, I am new to voip, I am trying to explain my best. |
| 13:43.45 | johnny_|_ | [TK]D-Fender: by show do you mean asterisk log and sip.conf? I will do that. |
| 13:44.11 | [TK]D-Fender | johnny_|_: Yes |
| 13:44.27 | [TK]D-Fender | Though not actually "log" |
| 13:44.29 | [TK]D-Fender | LIV CLI ONLY |
| 13:44.34 | [TK]D-Fender | LIVE* |
| 13:44.37 | roswell | johnny_|_, callerid shows its best, the id, 101. "me@wherever" would be ${CALLERID(name)} |
| 13:44.43 | [TK]D-Fender | it's too early for typing again... |
| 13:44.57 | johnny_|_ | hold on, just a sec :) |
| 13:45.23 | [TK]D-Fender | "sip set debug on" <- And you need to actually be looking at the SIP traffic to qualify as "looking". |
| 13:45.56 | roswell | fender, why not to advise him with tcpdump? ) |
| 13:46.23 | [TK]D-Fender | roswell: Because that won't tell him what * is matching it against |
| 13:46.36 | [TK]D-Fender | roswell: You miss all of the parts where * tells you what it actually does with it |
| 13:46.42 | [TK]D-Fender | roswell: External = waste of time |
| 13:48.03 | roswell | yup perharps but he came up with a quite straightforward quejds |
| 13:48.11 | roswell | gosh i need a nap too |
| 13:49.04 | johnny_|_ | [TK]D-Fender, roswell: http://pastebin.ca/2884154 |
| 13:50.06 | johnny_|_ | that is for call from ekiga to linphone |
| 13:50.09 | [TK]D-Fender | roswell: Yes, and he needs to actuall look at what's happening to confirm at what point things go wrong. Everything he needs is at CLI. |
| 13:50.30 | [TK]D-Fender | [08:45][TK]D-Fender"sip set debug on" <- And you need to actually be looking at the SIP traffic to qualify as "looking". |
| 13:50.32 | [TK]D-Fender | johnny_|_: ^^^^^^^^^^^ |
| 13:50.51 | johnny_|_ | oh sorry |
| 13:51.30 | [TK]D-Fender | johnny_|_: fromuser=101 <--- this is automatically your problem though |
| 13:51.54 | johnny_|_ | thanks, will try to comment that |
| 13:51.58 | [TK]D-Fender | johnny_|_: that forces the "From:" your device will be sent when * calls it |
| 13:52.22 | [TK]D-Fender | johnny_|_: and without RPID setting, that's all you'll get |
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| 13:56.31 | johnny_|_ | [TK]D-Fender: Well there is some progress, now instead of 101, I my id is user@lan.ip... |
| 13:56.42 | johnny_|_ | I will work on those sip logs, just a sec |
| 14:02.12 | johnny_|_ | [TK]D-Fender: http://pastebin.ca/2884159 |
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| 14:04.18 | [TK]D-Fender | johnny_|_: When you're asking for us to help debug things.... fucking with the evidence is not particularly helpful... |
| 14:04.46 | [TK]D-Fender | johnny_|_: You're changing the very things we're supposed to be looking at to confirm formatting, etc |
| 14:04.59 | johnny_|_ | [TK]D-Fender: umm I don't follow. I redacted my publicip and my username. |
| 14:05.23 | johnny_|_ | that is all |
| 14:05.31 | [TK]D-Fender | From: <sip:someuser@192.168.1.122>;tag=as7482f4c4 |
| 14:05.42 | [TK]D-Fender | From: <sip:someuser@sip.linphone.org>;tag=K8hg7F~SC |
| 14:05.52 | [TK]D-Fender | This IS where the callerID typically comes from |
| 14:05.56 | [TK]D-Fender | and that is what you are masking |
| 14:06.13 | [TK]D-Fender | Line 7 |
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| 14:06.27 | [TK]D-Fender | and then sent out as line 95 |
| 14:08.26 | johnny_|_ | [TK]D-Fender: sorry, I see the call did not went the way I expected. I now see that I actually called myself from the same sip (linphone) instead from ekiga one. I blame my client first :/. |
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| 14:19.32 | johnny_|_ | [TK]D-Fender: http://pastebin.ca/2884162, now callid is administrator@192.168.1.122 which is username of my machine (debian). However there is no (expected) From address which I expect it to be foo.bar@ekiga.net. |
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| 14:29.43 | [TK]D-Fender | You're still filtering.... |
| 14:29.44 | johnny_|_ | well I guess this is correct to some degree since @ekiga.net is not shown when calling linphone sip (without redirections) either. What is shown is username@public-ip, but with my settings (as you pointed out), caller id retains username but changes public-ip to local-ip. |
| 14:29.57 | johnny_|_ | [TK]D-Fender: what do you mean by filtering? |
| 14:30.15 | [TK]D-Fender | From: "someuser" <sip:administrator@public-ip>;tag=f44a9308-1b7e-e411-913b-00155d014713 |
| 14:31.38 | johnny_|_ | I ran 2 sed commands over logs that's it, I am not deleting lines. That someuser is just "foo" or "johnny" for the sake of this matter. |
| 14:33.17 | [TK]D-Fender | ... |
| 14:33.34 | [TK]D-Fender | You are changing the lines we're supposed to be looking at. |
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| 14:34.46 | johnny_|_ | but it does not make any difference. changing is consistent. you can ran sed yourself and replace someuser with johhny(?) |
| 14:35.04 | johnny_|_ | but I appreciate your help, really. |
| 14:36.58 | [TK]D-Fender | johnny_|_: Those lines are where the callerid IS |
| 14:37.08 | [TK]D-Fender | johnny_|_: What you see coming in is what it is. |
| 14:37.25 | [TK]D-Fender | johnny_|_: Then ther is whatever your setup does to it before calling your device |
| 14:39.21 | johnny_|_ | [TK]D-Fender: well thanks, I guess it is working now. |
| 14:39.56 | [TK]D-Fender | johnny_|_: That's something... |
| 14:40.45 | johnny_|_ | [TK]D-Fender: I have another issue that I want to know more about... |
| 14:41.11 | johnny_|_ | linphone client has chat/conversations. |
| 14:41.36 | johnny_|_ | What kind of messages are these? I think my asterisk setup does not know what to do with them. |
| 14:42.07 | johnny_|_ | I am talking about incoming messages for starters. |
| 14:44.35 | [TK]D-Fender | johnny_|_: that is SIP messaging. |
| 14:45.32 | [TK]D-Fender | johnny_|_: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_MESSAGE |
| 14:45.33 | johnny_|_ | [TK]D-Fender: it seems not many clients support that. (If we are thinking the same thing...) |
| 14:45.45 | [TK]D-Fender | johnny_|_: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_MessageSend |
| 14:46.18 | [TK]D-Fender | johnny_|_: Lots do. It takes some work for * to handle those and it is version dependent |
| 14:50.32 | johnny_|_ | [TK]D-Fender: thanks, I will look into it. For now my CLI says: Executing [101@othersipserver-incoming:1] Dial("Message/ast_msg_queue", "SIP/101") in new stack. Then: Scheduling destruction of SIP dialog 'ZN2L3XMtwT' in 32000 ms (Method: MESSAGE) [Dec 9 15:45:30] NOTICE[25170][C-00000009]: chan_sip.c:29781 sip_request_call: Asked to get a channel of unsupported format (nothing) while capability is |
| 14:50.34 | johnny_|_ | (gsm|law|alaw|ilbc). Unable to create channel of type 'SIP' cause 58: Bearer capability not available. |
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| 14:58.16 | [TK]D-Fender | johnny_|_: You've got some reading to do for this... |
| 14:58.51 | johnny_|_ | [TK]D-Fender: you mean messaging or in general :) |
| 14:59.09 | [TK]D-Fender | YES |
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| 14:59.33 | johnny_|_ | [TK]D-Fender: I was happy to set up gsm/usb dongle and webrtc video telephony so this should not be a big deal. |
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| 15:22.39 | vassilux | I try to understand echocancel option but when I change value from yes to 128, 256 dahdi show channel 1 shows 1 taps. I do unload chan_dahdi, restart dahdi, load chan_dahdi. Do I need to restart my PC ? |
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| 15:44.53 | vassilux | I looked into the function process_echocancel (chan_dahdi.c) and It seems the taps value must be reloaded |
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| 16:12.27 | Dulcin | Hi is it possible to see incoming calls (in some form of log) to test if my external sip trunk is routing the calls correctly? |
| 16:12.54 | Dulcin | As I currently 'believe' I set the SIP trunk up correctly, but my phone is not ringing :) |
| 16:13.13 | johnny_|_ | Dulcin: sip set debug on |
| 16:14.48 | Dulcin | Is that in a conf file somewhere? |
| 16:15.00 | johnny_|_ | Dulcin: asterisk CLI |
| 16:15.12 | Dulcin | ah ok |
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| 16:33.42 | vassilux | put echocancel=256 into users.conf works for me after reload module |
| 16:38.49 | [TK]D-Fender | users.conf should never be used. |
| 16:39.23 | [TK]D-Fender | It was a dead idea to support a long-dead GUI. |
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| 16:42.07 | vassilux | Your are right , but ... :-) |
| 16:42.43 | [TK]D-Fender | Let me guess... you're using an equally dead install? |
| 16:43.45 | vassilux | it is a long story |
| 16:44.25 | vassilux | I hope to migrate to freepbx |
| 16:44.34 | [TK]D-Fender | I'm sure the story isn't that long.... |
| 16:45.12 | WIMPy | Short story, long history? |
| 16:46.26 | vassilux | yep , an old quick decision make a long story |
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| 17:28.51 | johnny_|_ | [TK]D-Fender: callerid is again broken. AFAIK call goes: "register => .../101" picks incoming call [linphone] block looks up context(=from-sip) => extensions.conf [from-sip] exten => 101,n,Dial(SIP/101). Verbose(${CALLERID(all)}) displays (just): "" <foo>. |
| 17:29.21 | [TK]D-Fender | johnny_|_: You know the drill... |
| 17:29.39 | [TK]D-Fender | And "register" has nothing to do with CallerID |
| 17:30.09 | [TK]D-Fender | We've been through that... |
| 17:30.50 | johnny_|_ | [TK]D-Fender: the thing is, somewhere along call routing From address gets reshaped from foo@sip.linphone.org to foo@192.168.1.122... Why? |
| 17:30.59 | [TK]D-Fender | .... |
| 17:31.15 | johnny_|_ | I will post logs but they will be the same as before. |
| 17:31.25 | johnny_|_ | I will include extensions.conf part. |
| 17:31.31 | [TK]D-Fender | Show actual logs. Do NOT mess with the edivence |
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| 17:40.07 | johnny_|_ | [TK]D-Fender: http://pastebin.ca/2884264 . bar@sip.linphone.org is calling foo@sip.linphone.org. |
| 17:43.29 | [TK]D-Fender | From: <sip:bar@192.168.1.122>;tag=as20a01f6f |
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| 17:43.35 | [TK]D-Fender | <PROTECTED> |
| 17:43.54 | [TK]D-Fender | Says"bar" there twice. |
| 17:43.57 | [TK]D-Fender | So....? |
| 17:44.29 | johnny_|_ | yes, this is where "sip.linphone.org" is replaced (not by me!!) with 192.168.1.122 |
| 17:45.32 | johnny_|_ | I am not sure what are you point at other than what I just said. |
| 17:45.54 | [TK]D-Fender | * is NEVER going to tell your phone "linphone" |
| 17:45.59 | [TK]D-Fender | Asterisk is NOT a SIP rpoxy |
| 17:46.08 | [TK]D-Fender | Your phone talks to ASTERISK |
| 17:46.17 | [TK]D-Fender | Linphone talks to ASTERISK. |
| 17:46.24 | [TK]D-Fender | They do not talk directly together |
| 17:46.27 | [TK]D-Fender | * is a |
| 17:46.31 | [TK]D-Fender | ~b2bua |
| 17:46.31 | infobot | extra, extra, read all about it, b2bua is a Back 2 Back User Agent. Additional information is available on wikipedia: http://en.wikipedia.org/wiki/Back-to-back_user_agent |
| 17:47.10 | [TK]D-Fender | If you want to fake up the callerID to look pretty then by all means. Just keep in mind what'll happen when you try to dial it back. |
| 17:47.16 | [TK]D-Fender | (based on that callerid) |
| 17:48.15 | johnny_|_ | So this is the correct behaviour I am experiencing? |
| 17:48.51 | [TK]D-Fender | perfectly normal |
| 17:49.04 | johnny_|_ | Just to make sure we are on the same line. * is acting as SIP client ("register =>../101") line, right? |
| 17:49.17 | [TK]D-Fender | that's what a B2BUA is |
| 17:49.22 | [TK]D-Fender | it is a client to EVERYTHING |
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| 17:52.49 | hrnt | [TK]D-Fender: hmm, even to the SIP phones / extensions? |
| 17:53.09 | Mango45 | Hi. I'm using a service provider that demands a 60 second registration interval. Asterisk is not respecting this and insists on registering every 5 minutes. This doesn't work well for obvious reasons. |
| 17:53.18 | johnny_|_ | Ok, incoming call from bar@linphone.org to foo@linphone. My "internal" sip (101), the username I am connected with to my *, is called from [from-sip] extent - basically rerouting call to 101. I would expect to retain From address (bar@linphone.org) but instead it is changed to bar@192.168.1.122. In fact it would even be ok if callerid would be set to bar_linphone.org@192.168.122. |
| 17:53.22 | Mango45 | Is there any way I can configure this specific registration to register every 60 seconds, but not the others? |
| 17:53.32 | [TK]D-Fender | hrnt: You see that word "EVERYTHING" that I shoved in all-caps? I did mean it..... |
| 17:53.52 | [TK]D-Fender | johnny_|_: It is NOT "re-routing". |
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| 17:54.06 | [TK]D-Fender | johnny_|_: nAsterisk sits in the MIDDLE. Linphone never talks directly to your phone |
| 17:54.22 | [TK]D-Fender | johnny_|_: Asterisk bridges AUDIO when your phone accepts the call that ASTERISK initiates to it. |
| 17:54.47 | hrnt | [TK]D-Fender: do you also mean that it acts as a client every time to everything? |
| 17:55.10 | [TK]D-Fender | hrnt: Read the link on what a B2BUA is.... |
| 17:55.14 | Mango45 | Huh, never mind,this seems to be an open bug. |
| 17:55.18 | johnny_|_ | [TK]D-Fender: I know that, but why is callerid changed or can I somehow reshape it back so that I can see where is the user calling me from (which domain). Right now, I can only see user |
| 17:55.19 | Mango45 | https://issues.asterisk.org/jira/browse/ASTERISK-23267 for anyone curious |
| 17:55.28 | johnny_|_ | ... user's username. |
| 17:55.50 | hrnt | [TK]D-Fender: yes, it says that B2BUA only acts as a client the the destination (when it acts as a UAC) |
| 17:55.56 | hrnt | to the destination |
| 17:56.23 | [TK]D-Fender | hrnt: Everything is a destination. Everything is a client. A talks to B. B talks to C. A does NOT talk to C |
| 17:57.00 | [TK]D-Fender | hrnt: Asterisk sits in the middle. |
| 17:57.06 | hrnt | [TK]D-Fender: that is not what the linked article says |
| 17:57.16 | hrnt | [TK]D-Fender: here is from the RFC: "A back-to-back user agent (B2BUA) is a logical entity that receives a request and processes it as a user agent server (UAS). |
| 17:57.53 | [TK]D-Fender | [12:55]johnny_|_[TK]D-Fender: I know that, but why is callerid changed or can I somehow reshape it back so that I can see where is the user calling me from (which domain). Right now, I can only see user <- change the callerid if you want. |
| 17:58.06 | [TK]D-Fender | johnny_|_: [12:47][TK]D-FenderIf you want to fake up the callerID to look pretty then by all means. Just keep in mind what'll happen when you try to dial it back. |
| 17:58.12 | [TK]D-Fender | jihnWhich is what I've already told you |
| 17:58.21 | hrnt | so when extension 1234 calls 5678, * acts as a UAS for the 1234->* part and UAC as *->5678 part |
| 17:58.53 | [TK]D-Fender | hrnt: 1234 is 100% separate from 5678. |
| 17:58.58 | johnny_|_ | [TK]D-Fender: but where oh where can I pick that the user is calling from sip.linphone.org? I can see it in SIP debug log, but how to materialize that in sip.conf or extenstions.conf |
| 17:59.28 | [TK]D-Fender | johnny_|_: You dump that peer into a CONTEXT right? THAT is how you know where the call came from. |
| 17:59.55 | hrnt | [TK]D-Fender: they are not in the B2BUA definition |
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| 18:00.44 | johnny_|_ | [TK]D-Fender: but what if I get call to foo@sip.linphone.org from ekiga.net. Surely that user won't have user@linphone but user@ekiga(?). What am I missing here? |
| 18:01.57 | [TK]D-Fender | johnny_|_: Then you're going to have to look at your headers int he dialplan. |
| 18:02.08 | [TK]D-Fender | johnny_|_: "core show function SIP_HEADER" |
| 18:02.50 | [TK]D-Fender | johnny_|_: Or you could set them up so they don't daisy chain the way you have them now |
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| 18:03.45 | johnny_|_ | [TK]D-Fender: SIP_HEADER seems promising, thanks. |
| 18:04.05 | [TK]D-Fender | johnny_|_: Far less ideal than setting those 2 sources up separately |
| 18:04.39 | johnny_|_ | [TK]D-Fender: I don't understand the last part. Do you mean to set up separate clients for ekiga an linphone? |
| 18:04.57 | [TK]D-Fender | Yes |
| 18:05.14 | [TK]D-Fender | I don't see a rason to have to chain them through one another. |
| 18:05.16 | johnny_|_ | [TK]D-Fender: and what about 1000+ other providers? |
| 18:05.33 | [TK]D-Fender | johnny_|_: Are YOU going to be using 1000? |
| 18:05.55 | [TK]D-Fender | johnny_|_: Or are they just using one to contact your ekiga? |
| 18:06.01 | [TK]D-Fender | (or linphone actually" |
| 18:06.23 | [TK]D-Fender | are these 2 account you have, or is the linphone being proxied into ekiga from the client's end? |
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| 18:06.57 | johnny_|_ | [TK]D-Fender: I actually only have linphone, since ekiga stopped working as client (from *) |
| 18:07.30 | [TK]D-Fender | I'm pretty sure it didn;t stop. Maybe you had issue trying to use it... |
| 18:07.50 | [TK]D-Fender | But I'm fairly use the service itself works fine... |
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| 18:08.20 | johnny_|_ | [TK]D-Fender: Maybe I am missing something but what about other users. They don't have to have account set up at linphone to actually call me at linphone. |
| 18:08.44 | [TK]D-Fender | johnny_|_: * is NOT a proxy. Get ready for pain... |
| 18:09.18 | [TK]D-Fender | johnny_|_: You're going to have a lot of work trying hack all the andom stuff you seem to be expecting will be calling you. |
| 18:09.18 | hardwire | heh |
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| 18:11.39 | johnny_|_ | [TK]D-Fender: I get it to some degree. I don't need 100% correct sip address to callback as as you pointed out am not going to set up with 1000 voip services. I just want to be able to see who (username) is calling me from where (domain). I don't mind if it is spoofed on my end. |
| 18:12.25 | [TK]D-Fender | johnny_|_: Well the headers on your inbound INVITE is what you've got to hack up. Go for it. |
| 18:12.49 | johnny_|_ | [TK]D-Fender: yes, thanks again, I was looking for that the whole time :). |
| 18:14.39 | [TK]D-Fender | johnny_|_: Hope what you hack up works to your expectations. |
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| 18:20.22 | johnny_|_ | [TK]D-Fender: meh, it's my new hobby. This is the first time anybody actually answered my questions here. |
| 18:22.00 | [TK]D-Fender | johnny_|_: Better questions tends to get better answers. Same with having more complete information. Sprinkle with luck. |
| 18:23.28 | johnny_|_ | [TK]D-Fender: voip terminology is weak on my end. Sorry, pretty new on this field. |
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| 18:57.49 | johnny_|_ | Since I am here. What do you guys think about 3g/umts video (h.324m)? I only recently noticed that none of major smart phones supports it. When did this happen? Why? So they can promote their proprietary software? |
| 19:00.45 | [TK]D-Fender | that is a protocol, not just a codec |
| 19:01.08 | [TK]D-Fender | http://en.wikipedia.org/wiki/3G-324M |
| 19:01.31 | [TK]D-Fender | Several of the audio & video codecs that spec lsits are supported, but the control channel itself is not. |
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| 19:02.14 | johnny_|_ | But why no support in android? ios I think also doesn't have it. Is this dead technology? |
| 19:03.35 | [TK]D-Fender | Well it dates back to 2001. Could very well be dead |
| 19:05.04 | johnny_|_ | But currently there is no alternative then. |
| 19:06.44 | hrnt | for video calling? really? |
| 19:08.15 | johnny_|_ | I think so. Is it possible to make (software independent, heh) ios <--> android video call using cellular network. |
| 19:09.08 | hrnt | sure, if you install a third party app |
| 19:09.49 | johnny_|_ | software independent somehow hints without 3rd party apps |
| 19:10.03 | johnny_|_ | it used to be possible with 3g video. |
| 19:10.34 | hrnt | if i had to guess they dropped the support because IIRC that had to go through the carrier |
| 19:10.37 | hrnt | (and they billed separately for it) |
| 19:11.24 | johnny_|_ | who is "they"? phone manufacturers? |
| 19:11.34 | hrnt | carriers |
| 19:11.53 | hrnt | there used to be a separate pricing for video calls |
| 19:11.54 | johnny_|_ | But i don't think carriers dropped it? |
| 19:12.14 | johnny_|_ | That depends on operator and/or country. |
| 19:12.30 | hrnt | sure, but the phone call was still routed over a circuit switched connection |
| 19:14.29 | johnny_|_ | Do you have an opinon on why android does not support 3g video calls? It doesn't make sense. |
| 19:15.46 | [TK]D-Fender | johnny_|_: Fron where? |
| 19:16.34 | hrnt | johnny_|_: well, the standard is retarded. they would have to interoperate will all kinds of crappy feature phones which would mean that they couldn't easily extend the functionality themselves. |
| 19:17.43 | hrnt | johnny_|_: the calls would go through the carrier. as far as i know for example video calls have never worked properly in the US because of carrier resistance. |
| 19:18.18 | hrnt | on the other hand they could implement their own proprietary app they would have full control of (facetime or whatever) and have much higher quality voice calls with the added benefit of vendor lock-in ;P |
| 19:19.29 | johnny_|_ | [TK]D-Fender, hrnt, all: this is what I am talking about: https://code.google.com/p/android/issues/detail?id=2664 |
| 19:21.24 | johnny_|_ | hrnt: well if they are already at it, why not resist audio calls as well and let's surround ourselves with closed proprietary solutions :P. |
| 19:21.42 | johnny_|_ | with better audio :P |
| 19:23.02 | hrnt | johnny_|_: (by the way, my galaxy S3 does support video calling) |
| 19:23.16 | [TK]D-Fender | johnny_|_: Interesting read.... |
| 19:23.30 | hrnt | johnny_|_: it could just as well be that the manufacturers decided that 3G video calling was a dying technology |
| 19:23.36 | hrnt | johnny_|_: the future is packet switched after all ;P |
| 19:24.14 | johnny_|_ | [TK]D-Fender: the more you know :) |
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| 21:07.05 | Micc | Anyone know how to move the Buddies button on a polycom back on to the first screen? I have 3 custom efks but only one is showing on the idle screen so I would like the Buddies key to be on the first screen so I don't have to hit more. |
| 21:07.40 | Micc | Why isn't there a #polycom channel? They could sure use one. |
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| 22:54.27 | *** join/#asterisk s7r_r (~s7r@openvpn/user/s7r) |
| 23:02.34 | *** join/#asterisk talntid (~talntid@216.229.186.226) |
| 23:02.51 | talntid | is there an easy way to dial all sip devices that start with a certian prefix? |
| 23:03.08 | talntid | like, dial(SIP/htdsk*) |
| 23:03.10 | WIMPy | No |
| 23:03.21 | talntid | didn't think so. |
| 23:21.36 | *** join/#asterisk CeBe (~CeBe@port-92-200-94-14.dynamic.qsc.de) |
| 23:37.53 | *** part/#asterisk jhlavacek (~jirka@84.19.95.180) |