| 00:32.36 | *** join/#asterisk theron (~theron@69.63.185.56) |
| 00:37.27 | maruen | anybody awaked now? |
| 00:37.33 | ChannelZ | nope |
| 00:38.13 | maruen | I couldnt install chan dongle to work with asterisk, anyone could help on it? |
| 00:56.16 | *** join/#asterisk vravn (~vravn@syn.rook.sx) |
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| 01:38.01 | WIMPy | maruen: You will have to wait for the magic glass shperer to return from repair to tell someone what your issue is with it. |
| 01:41.39 | WIMPy | Well, or you could tell us what it is. |
| 01:53.17 | *** join/#asterisk vravn (~vravn@syn.rook.sx) |
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| 02:51.28 | Kattyroo | hello my asterisk does not work at all how to fix plz |
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| 03:47.06 | ChannelZ | maybe put the freepbx on it, solve everything |
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| 03:49.43 | Kattyroo | gags. |
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| 04:22.41 | maruen | WIMPy: no |
| 04:22.45 | maruen | I already fixed |
| 04:22.52 | maruen | and installed successfully :) |
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| 05:20.42 | ChannelZ | needs a Frosty |
| 05:30.41 | Penguin | Will you eat it with a fpoon? |
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| 05:34.52 | ChannelZ | Yeff |
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| 05:48.22 | boratynskikamil | Hello. Question to you. Is it possible to hangup a call when everyone canceled it in Queue? |
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| 08:46.55 | mattwj2002 | howdy room |
| 08:48.47 | j4jackj | . |
| 09:09.58 | ChannelZ | That was quick. |
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| 10:45.16 | boratynskikamil | Any suggestion? |
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| 13:51.22 | boratynskikamil | Question. |
| 13:51.43 | SpeedEvil | answer |
| 13:52.22 | boratynskikamil | Where should look for mistake. When I'm SIP1 and I call SIP2. I am not able to hear SIP2, and SIP2 does not hear me. |
| 13:52.34 | *** join/#asterisk gerhard7 (~gerhard7@77-172-35-234.ip.telfort.nl) |
| 13:52.58 | boratynskikamil | After that, when I call to ISDN card, and SIP1 and SIP2 receives this connection. We are not able to hear caller. |
| 13:53.01 | boratynskikamil | Suggestion? |
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| 14:04.20 | boratynskikamil | http://wklej.org/id/1286686/ |
| 14:04.23 | boratynskikamil | extensions.conf |
| 14:04.35 | boratynskikamil | This case is: HD01 calls HD02. |
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| 14:41.52 | WIMPy | boratynskikamil: Firewall |
| 14:42.39 | boratynskikamil | WIMPy: On Asterisk server? |
| 14:42.53 | WIMPy | The side that doesnt hear anything. |
| 14:43.36 | boratynskikamil | WIMPy: Both sides. |
| 14:43.46 | WIMPy | If you don't hear anything the audio doesn;t get to you. Either because your FW doesn't allow it through or because you have a NAT issue. |
| 14:44.25 | boratynskikamil | And after that SIP<->SIP call, when a call comes from ISDN then SIP client doesn't hear a caller. |
| 14:47.43 | boratynskikamil | WIMPy: ^ is more intereseting then SIPs cannot hear each other. |
| 14:50.37 | WIMPy | It doesn't matter where the call comes from. |
| 14:52.08 | *** join/#asterisk iulhk (~iulhk@94.201.234.28) |
| 14:52.13 | WIMPy | Should I go on site and try to fiddle with the fiber optics or do I stay at home and try to build a dialplan that works for bith SIP and DAHDI (and get brain cancer instead)? |
| 15:01.06 | boratynskikamil | WIMPy: I did a test. Firewall is disable at all, still the same. I didn't NAT anything. |
| 15:01.15 | boratynskikamil | So... let's debug SIP. |
| 15:05.48 | iulhk | i hv spa8000 installed in my office, right now i attached 4 telephone lines with it, and its working, i can receive calls by using normal telephone set, my question, can i take calls in my asterisk box by using spa8000? |
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| 15:26.07 | boratynskikamil | What may be a problem when 2 SIPs cannot call to each-other? Any suggestions? |
| 15:36.29 | boratynskikamil | http://wklej.org/hash/c90a270a047/ - both SIPs do not hear each other. |
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| 18:45.00 | daveyfx | hi all. is this a digium support channel, or just for asterisk? |
| 18:45.13 | daveyfx | i have a question specifically about digium gateways |
| 18:46.16 | WIMPy | They have their own support. |
| 18:58.19 | [TK]D-Fender | then again some people here might also use them and have your answer |
| 18:58.26 | [TK]D-Fender | So just ask. |
| 19:00.54 | *** join/#asterisk matrix1233 (~matrix123@41.228.40.158) |
| 19:03.19 | daveyfx | thanks. i'm using a G100 with the latest firmware to handle sip conversion between my softswitch and a lync mediation server |
| 19:03.37 | daveyfx | my software supports sip over udp, lync uses sip over tcp |
| 19:04.38 | [TK]D-Fender | G100 doesn't sound to be between your softswitch and lync .... |
| 19:04.59 | [TK]D-Fender | considering it's a digital PSTN gateway, not a SIP proxy |
| 19:05.00 | daveyfx | the issue is that for incoming calls from PSTN, the G100 sends a line URI that lync doesn't understand |
| 19:05.15 | [TK]D-Fender | Perhaps more like G100 -> softswitch -> lync |
| 19:05.49 | daveyfx | the G100 needs to be there so that lync can talk to my softswitch |
| 19:06.03 | daveyfx | lync talks SIP over TCP |
| 19:06.10 | daveyfx | my softswitch talks SIP over UDP |
| 19:06.37 | pabelanger | then you need asterisk between lync and softswitch to convert UDP to TCP |
| 19:06.40 | pabelanger | or some proxy |
| 19:06.50 | daveyfx | that's what i'm using the G100 for |
| 19:07.12 | daveyfx | it is converting SIP to TCP just fine |
| 19:07.29 | pabelanger | okay, what is the issue again? |
| 19:08.05 | daveyfx | lync can see the incoming INVITE request, but the request is not routable because the destination line URI is incorrect |
| 19:08.06 | [TK]D-Fender | "the G100 sends a line URI that lync doesn't understand" |
| 19:08.10 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 19:08.38 | pabelanger | likely need to hackup the Dial command some how |
| 19:09.00 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 19:09.00 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 19:10.13 | pabelanger | daveyfx, basically, contact Digium, see if they have integration support with Lync. If this was native asterisk, we could do more since you had access to the dialplan |
| 19:10.15 | daveyfx | here's what my lync server sees from the G100 |
| 19:10.21 | daveyfx | Message-Type: response |
| 19:10.21 | daveyfx | Start-Line: SIP/2.0 504 Server time-out |
| 19:10.21 | daveyfx | From: "XXXXXX,DAVID"<sip:1XXXXXX7949;phone-context=DefaultProfile@xxxxxxxx.com;user=phone>;epid=519F43B2D4;tag=1c96d39e33 |
| 19:10.21 | daveyfx | To: <sip:1SIPT-MNODES-LYNC@172.16.3.24>;tag=1F8D1C802E737BF2C0BE498254B2D7FB |
| 19:10.21 | daveyfx | Call-ID: 8190167b-e183-40f5-980c-cea6317f60e7 |
| 19:10.21 | daveyfx | CSeq: 189 INVITE |
| 19:10.22 | daveyfx | Via: SIP/2.0/TLS 172.16.x.xx:51762;branch=z9hG4bKdd93dd10;ms-received-port=51762;ms-received-cid=13700 |
| 19:10.23 | daveyfx | Content-Length: 0 |
| 19:10.23 | daveyfx | ms-diagnostics: 1022;reason="Cannot process routing destination";destination="sip:1SIPT-MNODES-LYNC@172.16.x.xx";source="lync1.mnodes.local" |
| 19:10.33 | pabelanger | daveyfx, don't flood, use pastebin |
| 19:10.47 | daveyfx | oops, sorry. i'm new to this. |
| 19:11.05 | daveyfx | but, essentially, the destination is wrong |
| 19:11.24 | daveyfx | the destination should be a valid normalized telephone number |
| 19:13.40 | daveyfx | but that destination shown there is from the endpoint i have configured in the g100 |
| 19:14.30 | daveyfx | so for some reason the g100 is not passing the actual destination telephone number |
| 19:14.59 | pabelanger | daveyfx, Contact digium support is your best bet right now |
| 19:15.08 | pabelanger | we don't have access to the source to see what it is doing |
| 19:16.56 | daveyfx | ok, thanks for the advice. thought i'd give it a shot :) |
| 19:23.14 | gusto | hey WIMPy |
| 19:23.20 | gusto | WIMPy: how is it going? |
| 19:23.40 | WIMPy | Bad |
| 19:23.44 | gusto | why? |
| 19:23.52 | gusto | your ISDN adapter still not working? |
| 19:24.04 | WIMPy | Battling with Asterisk again. Or rather DAHDI. |
| 19:24.33 | WIMPy | Well, it is working, but it's not what I'd like to see. |
| 19:25.38 | WIMPy | Same goes for the FXS card. |
| 19:26.04 | WIMPy | But that might be less of an issue as there can't be any en-block dialling. |
| 19:46.41 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
| 20:07.13 | *** join/#asterisk ChannelZ (channelz@burner.com) |
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| 20:31.22 | *** join/#asterisk paule32 (paule32@dslb-188-103-039-150.pools.arcor-ip.net) |
| 20:31.40 | paule32 | hello |
| 20:32.04 | ChannelZ | ohell |
| 20:32.24 | WIMPy | Greetings to Saxonia. |
| 20:32.27 | WIMPy | scnr |
| 20:32.43 | paule32 | how can i use asterisk pbx with software modem (terminal) as dial in server |
| 20:32.59 | paule32 | WIMPy: greetings from thuringa |
| 20:33.12 | WIMPy | Not that far away :-) |
| 20:33.24 | WIMPy | Asterisk is not for dialup. |
| 20:33.26 | paule32 | haha |
| 20:34.22 | paule32 | i would like construct a server, so, that i can dial in with ip number (phone no) |
| 20:34.43 | WIMPy | What? IP or phone? |
| 20:35.00 | paule32 | you know the mailslots in the 1980er? |
| 20:35.24 | WIMPy | Mine is still running. |
| 20:35.38 | paule32 | i have a ip telefon number in my avm fritz.box |
| 20:35.41 | WIMPy | But it's only used as mail server. |
| 20:36.07 | WIMPy | You can't do modem over voip. |
| 20:36.22 | paule32 | so it should be possible to route the number (call) to ,asterisk application? |
| 20:36.46 | WIMPy | There is a modem over IP thing like T.38, but I don;t know of any implementation. |
| 20:36.58 | WIMPy | Sure. |
| 20:37.07 | WIMPy | Or just move it over. |
| 20:37.54 | paule32 | how? |
| 20:38.29 | WIMPy | Take your SIP account data, disable it in the FB and configure it in Asterisk instead. |
| 20:39.36 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
| 20:39.54 | paule32 | ok, when i call and route from fritz,box, i get message from wommen: you have successfully install ... |
| 20:44.12 | paule32 | what's next? |
| 20:44.34 | WIMPy | What do you want next? |
| 20:45.57 | paule32 | simulate a call from an other pc to asterisk, so i can transfer text between pc1 and pc2(asterisk pbx) |
| 20:46.26 | WIMPy | Text? |
| 20:47.09 | paule32 | yes, like text list's - you know the service from tv watching broadcasters? |
| 20:47.29 | WIMPy | No. |
| 20:47.41 | WIMPy | are you sure, you know what Asterisk is? |
| 20:48.12 | paule32 | an open source pbx, with feature to build modules |
| 20:48.46 | WIMPy | Yes, that's what the lid says, I guess. |
| 20:49.50 | paule32 | i have read, that i can do text messangeing, and video streaming, beside call, conferences ... |
| 20:51.22 | paule32 | so, i need text messageing between pc's, when they call the pbx |
| 20:54.07 | *** join/#asterisk serafie (~erin@24.96.64.240) |
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| 20:56.34 | paule32 | do you know what i want? |
| 20:56.57 | WIMPy | No |
| 20:57.07 | paule32 | also: |
| 20:57.29 | paule32 | 1. public people call me |
| 20:58.18 | paule32 | 2. the call is routed into my fritz.box - (in normal way a pluged telephone rings) |
| 20:58.59 | paule32 | 3. the route is to a linux pc (debian) - the asterisk pbx |
| 20:59.41 | paule32 | 4. the caller can now loged in into the pbx application |
| 21:00.14 | paule32 | 5. a list of text is transfered between pbx and call er pc |
| 21:00.48 | WIMPy | So you want an IVr that then sends SMs? |
| 21:00.52 | paule32 | 6. now, the caller can hang up, or initiate a request for more text |
| 21:02.07 | paule32 | i'm not really sure if i can say SMS, the text should send to the caller pc into the software application |
| 21:02.17 | paule32 | not hardware |
| 21:02.29 | WIMPy | What caller PC? |
| 21:03.22 | paule32 | caller pc = 800 123 <- telephone number - to call the server |
| 21:04.09 | WIMPy | Your description is pretty vague. |
| 21:04.49 | paule32 | i don't know how to describe, my slengish is bad i know |
| 21:07.44 | *** join/#asterisk Tim_Toady (~moi@89.249.79.253) |
| 21:12.05 | *** join/#asterisk mnathani (~mnathani@umbozero.winvive.com) |
| 21:22.17 | paule32 | WIMPy: http://img5.fotos-hochladen.net/uploads/phoneex08gy7c2j.png |
| 21:23.20 | *** join/#asterisk gerhard7 (~gerhard7@77-172-35-234.ip.telfort.nl) |
| 21:23.33 | WIMPy | Still nicht very clear. |
| 21:23.43 | WIMPy | What is that PC1 and what's running there. |
| 21:23.52 | WIMPy | And what does Asterisk do? |
| 21:24.07 | WIMPy | And what's that "Text" |
| 21:24.12 | WIMPy | ? |
| 21:24.58 | paule32 | pc1 is a standard ibm personal computer 586 32 amd bit cpu |
| 21:25.48 | paule32 | on pc1, i have to code a application that do transfer requests between pc and asterisk pc |
| 21:26.43 | paule32 | asterisk should do pick up the calling number and response to pc with text |
| 21:26.43 | WIMPy | So why Asterisk? What's the point if you use a custom application? |
| 21:26.51 | paule32 | text is a "string" |
| 21:27.31 | WIMPy | Senseless details. What's the goal of whatever you're trying to do? |
| 21:27.59 | paule32 | i have no custom application, i would like to code with the community help |
| 21:28.12 | WIMPy | So far I can't make your details fit together. |
| 21:28.25 | paule32 | you can say a client/server application |
| 21:29.03 | *** join/#asterisk cmendes0101 (~cmendes01@pool-96-251-59-96.lsanca.fios.verizon.net) |
| 21:29.16 | paule32 | the client applications calls the server (asterisk pbx) |
| 21:29.39 | paule32 | and transfers data - string's |
| 21:29.53 | paule32 | or: "array of chars" |
| 21:32.21 | WIMPy | That does not sound like it could make any sense to use Asterisk there. |
| 21:32.55 | paule32 | why? |
| 21:33.19 | paule32 | i have no problem to code a application that runs over ther internet |
| 21:33.26 | WIMPy | So far I don't see anything where it could be usefull. |
| 21:33.43 | WIMPy | Sounds more likely that a webserver could help. |
| 21:34.16 | paule32 | but i want a application that use a 56k modem to call to asterisk |
| 21:34.56 | paule32 | the speed is not the thing |
| 21:36.38 | WIMPy | You can't. |
| 21:37.37 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 21:37.38 | WIMPy | I told you an hour ago. |
| 21:38.04 | WIMPy | Neither can you do modem over VOIP nor does Asterisk do modem. |
| 21:38.31 | WIMPy | And AFAIK there is only 3rd party modem emulation for fax modems. |
| 21:39.04 | WIMPy | Get a real modem or a NAS if you plan something bigger. |
| 21:39.24 | WIMPy | But you still need a real phone line. |
| 21:44.37 | *** join/#asterisk matrix1233 (~matrix123@41.231.9.186) |
| 21:45.45 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 21:47.52 | matrix1233 | hello |
| 21:48.55 | matrix1233 | am searching a solution to remove echo i found PBXMate but is a commercial component ..there is a module similar opensource ? |
| 21:53.11 | paule32 | WIMPy: lol, i have 4 of 7 active phone lines |
| 21:53.50 | WIMPy | What does that mean? |
| 21:54.27 | paule32 | you told, i need a real phone line? |
| 21:54.38 | WIMPy | yes |
| 21:55.14 | j4jackj | Bello |
| 21:55.38 | WIMPy | But there's also that thing calld the Internet, which might be more suitable. |
| 21:56.34 | *** join/#asterisk reconwireless (uid10170@gateway/web/irccloud.com/x-jyupmtyjsigbhegj) |
| 21:57.16 | paule32 | my isp is a dsl provider which offers voip |
| 21:58.00 | WIMPy | Which is not a phone line. |
| 21:59.14 | paule32 | i have applicance of "dsl flat + phone flat" - 2 packets in one |
| 21:59.38 | WIMPy | And what does that tell us? |
| 22:00.54 | paule32 | i can call the world with 3 phones on custom end side (me) the rest of 4 numbers are individual line numbers |
| 22:01.36 | WIMPy | I don't know what that means. |
| 22:02.02 | WIMPy | Nor do I have an idea what you're tryiong to tell us now. |
| 22:02.22 | paule32 | for me it is clear: i have phone lines for the world |
| 22:02.33 | WIMPy | No, you don't. |
| 22:03.04 | WIMPy | I explaied that about an hour and a half ago and later again. |
| 22:04.19 | paule32 | i am confused - i can do surf into the web, and call my girl friend, too |
| 22:06.53 | [TK]D-Fender | [16:48]matrix1233am searching a solution to remove echo i found PBXMate but is a commercial component ..there is a module similar opensource ? <- there is no software fix for your server. the problem is at the endpoints HAS to be fixed there |
| 22:23.51 | matrix1233 | [TK]D-Fender: i have no access to the endpoints .. for now i used DENOISE and VOLUME to reduse the echo |
| 22:24.28 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 22:25.57 | matrix1233 | [TK]D-Fender: jitterbuffer can resolve also ? |
| 22:26.07 | matrix1233 | [TK]D-Fender: to reduse ? |
| 22:29.26 | *** join/#asterisk CeBe (~CeBe@port-92-206-47-149.dynamic.qsc.de) |
| 22:45.54 | [TK]D-Fender | matrix1233: Ok it seems you are not getting the message. I've said this several times now and I'll try one last time. Hopefully this will make the situation clear for you. THIS CANNOT BE FIXED BY ANYTHING ON YOUR SERVER. This is not an "Asterisk" limitation. This is basic science. |
| 22:46.20 | [TK]D-Fender | matrix1233: Echo cancellation HAS to be done at the analog>digital CONVERSION point. |
| 22:47.14 | matrix1233 | [TK]D-Fender: ok thanks :) i understand |
| 22:47.48 | paule32 | WIMPy: i have found a simple solution: the caller app have to recieve sounds like the option "Playback" |
| 22:48.01 | [TK]D-Fender | matrix1233: If your provider has issues, you are screwed if you stay with them. You cannot fix a problem that starts on their side. Change providers if that is the case. Lowering volume may impact echo, but not entirely fix it and you will be pumping out weak calls. then when the other side jacks up their volumes to hear you you'll see more things go wrong and more quality lost. |
| 22:48.49 | [TK]D-Fender | matrix1233: If your PHONES are causing this... replace them. |
| 22:48.54 | paule32 | the next resulting question is, how can interact the caller with the server? |
| 22:48.59 | [TK]D-Fender | matrix1233: Softphones are the WORST. |
| 22:49.20 | paule32 | the caller must send sounds , too |
| 22:49.29 | [TK]D-Fender | paule32: like what? |
| 22:49.34 | *** join/#asterisk vlad_sta_ (~vlad_star@77.41.82.191) |
| 22:49.38 | *** join/#asterisk Milarepa (~Milarepa@host-74-211-92-125.beyondbb.com) |
| 22:50.00 | paule32 | exten => 600,1,Playback(demo-echotest) |
| 22:50.21 | matrix1233 | [TK]D-Fender: thanks i will change the provider |
| 22:50.39 | [TK]D-Fender | paule32: that is * playing a sound to the device in that call being processed.... |
| 22:51.11 | paule32 | yes, my idea was: the sound modem |
| 22:52.07 | [TK]D-Fender | paule32: Asterisk cannot act as a modem. That is not happening. Game over. Time for a new plac |
| 22:52.10 | paule32 | also the caller application calls server, get sound as response, the caller check the sound (sound processing) ... |
| 22:52.12 | [TK]D-Fender | plan* |
| 22:52.59 | [TK]D-Fender | paule32: "get sound as a response" <- this is very vague.... |
| 22:53.21 | paule32 | e.g: |
| 22:54.14 | paule32 | caller initiate a session, file "1000" is play - the frequency: 1000 is send to caller |
| 22:54.49 | [TK]D-Fender | paule32: "core show application playtones" <- |
| 22:54.50 | paule32 | the caller check the frequence, and build a char |
| 22:55.19 | [TK]D-Fender | paulFORGET about asterisk analyzing the frequency and processing it. |
| 22:55.34 | [TK]D-Fender | paule32: it has no function to do so |
| 22:56.15 | paule32 | yes, this with the dialtones was my second thinking |
| 22:56.38 | [TK]D-Fender | paule32: You'd have to develop your own dialplan application directly in C to gain full no-(additional) latency to the stream and code this 100% yourself. |
| 22:57.27 | [TK]D-Fender | paule32: You seem to talka lot about how you'd like things to work.... what actual GOAL are you trying to accomplish? |
| 22:59.20 | paule32 | create a software modem application - back to root's ... so analog modems can call my pbx and get informations - a lack of sysop mail box in the 80er |
| 23:00.13 | paule32 | before internet was born, amature funkers has pocket radio |
| 23:00.42 | paule32 | they used a software modem as endpoint |
| 23:01.54 | [TK]D-Fender | paule32: What start of communication are you looking at? |
| 23:02.05 | [TK]D-Fender | s/start/standard/ |
| 23:03.06 | [TK]D-Fender | paule32: VoIP using G.711 is lucky to survive short 2400 baud comms in passthrough. |
| 23:03.25 | [TK]D-Fender | paule32: Asteriskhas not PROCESSING for this and you'd have to code all the DSP work yourself in C |
| 23:03.48 | [TK]D-Fender | paule32: This sounds like a tragic waste of time & effort if not an outright pipe-dream. |
| 23:04.02 | paule32 | nothing, i want my own internet with my friends, so i have 1 friend that home is in a funj hole, and only 56k modems have changes to the internet |
| 23:04.39 | paule32 | but that dosn't matter |
| 23:06.19 | [TK]D-Fender | paule32: If you want to try for some horrible low-end attempt the best I can think of is to get an analog TDM card and a standard modem. Plug them into each other and let your OS handle the data end whil you pray there is enough audio integrity to maintain CARRIER at all. |
| 23:07.19 | paule32 | yes, this is possible |
| 23:07.28 | paule32 | but |
| 23:07.40 | paule32 | 1 line per modem |
| 23:08.09 | paule32 | but i have only 1 line for phone + dsl internet |
| 23:08.12 | WIMPy | Rather it might be possible. At very low speeds with a LOT of luck. |
| 23:09.05 | paule32 | so i can recieve 4 call's over the dsl router into pbx |
| 23:09.17 | paule32 | and the problem is away |
| 23:09.34 | WIMPy | You can have 4 calls? |
| 23:09.50 | [TK]D-Fender | "over the DSL router" |
| 23:09.54 | paule32 | yes, |
| 23:09.57 | [TK]D-Fender | this spells "voip" to me.... |
| 23:10.25 | j4jackj | [TK]D-Fender, that is what it is |
| 23:10.25 | paule32 | my words from beginn |
| 23:10.26 | WIMPy | Yes, I told him that he needs a real phone line, hours ago. |
| 23:11.04 | [TK]D-Fender | paule32: Asterisk can "maybe" take your VoIP connection directly in. Then do what I suggested |
| 23:11.22 | paule32 | i have to register a free phone number at my provider |
| 23:11.22 | [TK]D-Fender | paule32: that's the only thing you've got. |
| 23:11.55 | paule32 | and setup it as internetphone |
| 23:12.30 | paule32 | so i can get calls from world into pbx |
| 23:12.59 | paule32 | its very easy |
| 23:13.58 | paule32 | so i have up to 7 phone lines (theroy) |
| 23:14.19 | WIMPy | Numbers. |
| 23:14.35 | paule32 | yes numbers |
| 23:14.38 | WIMPy | They probably restrict you to two calls. |
| 23:14.43 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 23:15.00 | paule32 | nono |
| 23:15.29 | WIMPy | Sure? |
| 23:15.31 | paule32 | i can call 7 numbers, if i would |
| 23:15.43 | WIMPy | But not at the same time. |
| 23:16.19 | WIMPy | Arcor/VF use a 2nd dedicated PVC with 128kbit. |
| 23:16.31 | WIMPy | Or at least they did the last time I had an encounter. |
| 23:17.05 | paule32 | 1 and 1 offers dsl routers for family |
| 23:17.36 | paule32 | so the wife can call, daughter chat, and me surf |
| 23:17.43 | WIMPy | So it's not about the line you're using now? |
| 23:17.51 | paule32 | in oversee |
| 23:19.28 | paule32 | i rest my case |
| 23:19.51 | WIMPy | o.O |
| 23:20.12 | paule32 | i have a router - 7 numbers (wlan) - 1 internet in one |
| 23:20.55 | WIMPy | tries to reboot |
| 23:23.42 | paule32 | in fact of, that the telecom had open everyone phone holders 8 msn's |
| 23:24.00 | WIMPy | They offered 10 |
| 23:24.06 | paule32 | ok |
| 23:26.18 | paule32 | so, give it example code for a simple "echo" application - also: call - send dial tone e.g. 4 - wait 2 seconds - pong with dial tone e.g.: 2 ?? |
| 23:26.53 | WIMPy | What's the point again? |
| 23:27.15 | paule32 | ping - pong app :) |
| 23:33.59 | WIMPy | That was a pretty cold boot. Lovely how a exploded GPU driver prevents a normal shutdown. |
| 23:36.52 | paule32 | friends... |
| 23:37.06 | paule32 | i found these: http://www.voip-info.org/wiki/view/Asterisk+manager+API |
| 23:37.25 | WIMPy | Great stuff. |
| 23:37.35 | WIMPy | Now you just need to find a use for it. |
| 23:42.23 | [TK]D-Fender | HOW EVERYTHING CAN I?!??!? PLAEZE@!# |
| 23:43.03 | WIMPy | One of the many people who didn't realize he didn;t have a phone line any more. |
| 23:43.38 | [TK]D-Fender | heads out for showtime.... |
