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| 04:01.21 | bsdice | is anyone using jitter buffer on Asterisk (adaptive?) and if so, to counter what effects? WAN jitter? |
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| 05:21.55 | ChannelZ | jitter buffers only help your incoming audio and usually to correct inconsistent internet latency |
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| 05:39.57 | Penguin | and it adds delay. |
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| 06:39.33 | jeev | anyone here config an adtran 924 before for hunt groups ?. |
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| 06:59.27 | BeeBuu | hi,all. |
| 07:00.02 | BeeBuu | why my cdr billsec is 0 when channel is local? |
| 07:00.20 | BeeBuu | how to fix it? any one help please? |
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| 08:06.18 | aamu11 | buaaaaaaaaa |
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| 08:09.14 | aamu11 | buaaaaaaaaaa |
| 08:12.06 | ChannelZ | take your meds kids |
| 08:18.07 | MaliutaLap | ChannelZ: now you sound like my oncologist :P |
| 08:19.33 | ChannelZ | Well you don't want to end up like aamu11 do you? |
| 08:21.57 | MaliutaLap | ChannelZ: it may be too late for that - he may be approaching my level of "whackiness" ;) |
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| 13:53.51 | beebuu | "put /n into the dial command when transferring the local channel" any one tell me what's that mean please? |
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| 14:03.05 | beebuu | i got the way that can fix billsec was set 0 when using local channle by google,it told me "put /n into the dial command when transferring the local channel",anyone help me please?i don't what is the /n mean? |
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| 14:37.50 | woopstar | hi all.. i've setup successfully a clone of my asterisk server. Everything works fine. Sip registration etc. But when i do a call, the call just ends. Nothing is written in the logs or asterisk cli. How would i debug what happens? |
| 14:39.05 | bulkorok | 'core set verbose 5' |
| 14:43.15 | woopstar | oh.. would i be able to have the same sip peers on two asterisk servers? |
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| 15:42.13 | beebuu | i got the way that can fix billsec was set 0 when using local channle by google,it told me "put /n into the dial command when transferring the local channel",anyone help me please?i don't what is the /n mean? |
| 15:42.32 | beebuu | no one here now? |
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| 15:42.58 | mjordan | beebuu: https://wiki.asterisk.org/wiki/display/AST/Local+Channel+Optimization |
| 15:43.22 | mjordan | beebuu: and https://wiki.asterisk.org/wiki/display/AST/Local+Channel+Modifiers |
| 15:44.05 | beebuu | thanks mjordan! |
| 15:44.12 | beebuu | tons of thanks! |
| 15:44.18 | mjordan | np |
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| 16:01.39 | tom_w | I have a strange issue. Have a server with a 2x te410p it has always worked fine. After a reboot last night one of the cards is in a blue/yellow state. the other card is fine. I have restarted dahdi several times and still no luck. when I try I server reboot it hangs on shutdown on the dahdi card. If I power cycle it comes back up but stays in blue/yellow alarm. There are now errors in /var/log/asterisk/messages. Any ideas on where |
| 16:02.41 | tom_w | */no errors |
| 16:07.57 | IMS77 | Hi, is there a limitation with the loops in dialplan ? I'm trying something like this : http://pastebin.com/YLLQ2Nx1 => this simple example works but when I include a second loop in the first (to check the status for x peers) => the loop exit (after 12 or 14 decrement) and I go back to the caller of the macro (I never see the counter = 0). Maybe the treatment is too long ? Can i change a parameter to have more time or not check the infinite loop... Than |
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| 16:14.51 | IMS77 | In debug i can see exited non-zero on 'SIP/ms5-00000118' in macro 'mymacro' and then Internal Gosub(handler,s,1) start. Any ideas ? |
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| 16:28.48 | IMS77 | Nobody can help ? |
| 16:33.39 | pabelanger | IMS77, stop using maros |
| 16:33.41 | pabelanger | macro* |
| 16:33.43 | pabelanger | and use gosub |
| 16:34.28 | pabelanger | There are limits to Macro() so, you might be having issue with that |
| 16:35.01 | IMS77 | ok but i search a doc about these limitations but i can't find them |
| 16:35.32 | IMS77 | i will try with gosub |
| 16:35.35 | pabelanger | Not sure they are documented, but you don't want to be using Macro() |
| 16:55.18 | IMS77 | I'm trying to use Gosub => with the macro when calling first a macro1 with 3 args and then calling a macro2 => macro2 has in parameter the same args than macro1. How to do with gosub ? |
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| 17:22.27 | ghost75 | using nat, do i need to port forward udptl ports? |
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| 17:35.24 | tom_w | I have a strange issue. Have a server with a 2x te410p it has always worked fine. After a reboot last night one of the cards is in a blue/yellow state. the other card is fine. I have restarted dahdi several times and still no luck. when I try I server reboot it hangs on shutdown on the dahdi card. If I power cycle it comes back up but stays in blue/yellow alarm. There are now errors in /var/log/asterisk/messages. Any ideas on where |
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| 17:53.33 | wonderworld | tom_w: enable dahdi debugging |
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| 18:10.18 | tom_w | I have and the only error is waiting for the t200 timer |
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| 18:18.16 | paule32 | hello |
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| 18:18.34 | paule32 | [Mar 7 18:49:10] WARNING[10220]: chan_sip.c:20720 handle_response_register: Forbidden - wrong password on authentication for REGISTER for '4936916580458' to 'sip.1und1.de' |
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| 18:18.45 | paule32 | i give right password |
| 18:19.13 | paule32 | 49 is the country code, 4691 is the area code the rest is my no. |
| 18:19.30 | WIMPy | tom_w: Are those spans PRIs to a telco or what? |
| 18:22.04 | WIMPy | paule32: How did you get from KDG to UI in such short time? |
| 18:22.46 | paule32 | haha, strong dicipline |
| 18:23.49 | WIMPy | Enable sip debug and read what they tell you. They semm to be quite explicit about what they don't like. |
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| 18:29.55 | paule32 | WIMPy: http://codepad.org/LfT3SMe8 |
| 18:31.11 | WIMPy | YOU enable sip debug and YOU read it :-) |
| 18:32.06 | WIMPy | points at line 23 |
| 18:33.09 | WIMPy | They couldn't be much more helpfull, I guess. |
| 18:33.23 | paule32 | oh |
| 18:33.35 | paule32 | 403 RFC1918-IPs not permited |
| 18:43.51 | *** join/#asterisk powerunits (272a9976@gateway/web/freenode/ip.39.42.153.118) |
| 18:44.35 | powerunits | dear team. i need some info on bluetooth with asterisk .. can i configure more than 2 blue tooth with 1 asterisk server? |
| 18:46.19 | powerunits | ? |
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| 18:49.59 | powerunits | any help ?? |
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| 18:50.25 | paule32 | you can register |
| 18:50.32 | paule32 | endless |
| 18:50.41 | paule32 | if the phones voip |
| 18:51.21 | WIMPy | powerunits: I don;t think there are many chan_mobile users. You will probably need some patience. |
| 18:51.42 | powerunits | rite thanks |
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| 18:53.42 | paule32 | WIMPy: thanks for the hint, i think the problem is the private address of the server? |
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| 19:04.05 | *** join/#asterisk ibercom (58196563@gateway/web/freenode/ip.88.25.101.99) |
| 19:08.45 | ibercom | I have this bugs https://issues.asterisk.org/jira/browse/ASTERISK-22945 with asterisk 11. And i need know where or when chan_sip executes sip_destroy_peer ... |
| 19:13.10 | ibercom | expire_register runs ok. But the other function never happens. |
| 19:13.54 | *** join/#asterisk cyford (~cyford33@2601:0:9300:2fc:6911:d42d:bd83:833c) |
| 19:14.33 | *** part/#asterisk ibercom (58196563@gateway/web/freenode/ip.88.25.101.99) |
| 19:15.25 | cyford | hi, i have a problem where my voicemails are all garbled up and i can not understand any of them |
| 19:16.09 | paule32 | memory problem? |
| 19:16.18 | paule32 | running in vm? |
| 19:18.26 | *** join/#asterisk SGjunior (~sgjunior@96.127.222.20) |
| 19:20.55 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 19:23.16 | *** join/#asterisk plm (~neo@65.111.165.20) |
| 19:24.13 | *** join/#asterisk SGjunior (~sgjunior@96.127.222.20) |
| 19:25.40 | cyford | yes running in vm |
| 19:25.55 | cyford | <PROTECTED> |
| 19:26.46 | cyford | <PROTECTED> |
| 19:27.16 | paule32 | wat? |
| 19:27.21 | paule32 | confused |
| 19:28.13 | cyford | when people call me and leave a voicemail i cant undwerstanfd it |
| 19:28.24 | cyford | *understand |
| 19:29.48 | Penguin | If you listen to the voice mail on the phone by calling into asterisk, does it sound like garbage? Or is it only garbage when you listen to the file from the email? |
| 19:29.59 | paule32 | played with codec? |
| 19:30.40 | cyford | .WAV and ubuntu defualt player |
| 19:30.51 | cyford | testing from phon enow |
| 19:33.18 | paule32 | what says: voicemail.conf format ? |
| 19:39.32 | *** join/#asterisk Milarepa (~Milarepa@209.33.195.241) |
| 19:41.38 | cyford | <PROTECTED> |
| 19:44.48 | WIMPy | paule32: Yes, Read about how to set up Asterisk behind NAT. |
| 19:48.56 | Penguin | If it plays good from the phone but plays bad in the media player, my guess is the codec used in the file is not what the media player expected. |
| 19:50.39 | Penguin | paule32: Maybe '4936916580458' is not your username, so when they said wrong password or authentication, they meant it. |
| 19:52.19 | *** join/#asterisk SGjunior (~sgjunior@96.127.222.20) |
| 19:54.27 | paule32 | hmmm |
| 19:55.18 | Penguin | Especially if your area code is 4691 and you told asterisk to use 3691. |
| 19:55.54 | paule32 | must check .. |
| 20:01.45 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 20:20.56 | paule32 | WIMPy: http://codepad.org/i8rnzBAd |
| 20:21.08 | paule32 | my sip.conf (partly) |
| 20:21.48 | WIMPy | 192.168:0.0 is not a valid IP. |
| 20:22.02 | eirirs | \o/ |
| 20:22.17 | WIMPy | And omdomain is not a valid configuration keyword. |
| 20:22.29 | paule32 | oh shit. im blind dumb ehing |
| 20:22.43 | Qwell | and Asterisk would have given warnings for all three of those issues. |
| 20:22.49 | Qwell | WIMPy: missed one :p |
| 20:23.22 | WIMPy | Since when does chan_sip complain about garbage in it's config file? |
| 20:23.23 | Qwell | crap, I missed 2 |
| 20:23.54 | Qwell | shrugs |
| 20:24.19 | WIMPy | Yes, quality and cnreinvite don't exist, either. |
| 20:24.27 | Qwell | crap, missed 3 |
| 20:24.42 | Qwell | WIMPy: and the *expirey ones |
| 20:25.17 | WIMPy | Oh well. Maybe we should see if the file contains valid lines. |
| 20:25.43 | Qwell | 15/21 ain't bad |
| 20:28.24 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 20:29.02 | *** join/#asterisk g_r_eek (~g_r_eek@46-236-35.adsl.cyta.gr) |
| 20:31.12 | WIMPy | Oh, and not to mention that abfuscating the username was senseless. |
| 20:31.27 | Qwell | WIMPy: You mean because of the non-obfuscated context? |
| 20:32.05 | WIMPy | yepp |
| 20:32.29 | *** join/#asterisk bford (~bford@nat/digium/x-vdilxuowjjtkqotj) |
| 20:34.07 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 20:34.53 | Penguin | It's all jacked up. Did a kindergartener write the config? |
| 20:37.04 | ghost75 | was somebody successfull with sendfax (res_fax) on 11.x with t.38 ? |
| 20:38.56 | ghost75 | receivfax works, but sendfax not |
| 20:39.28 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 20:42.21 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-uvctezvvwnztcksm) |
| 20:42.21 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 20:42.58 | *** join/#asterisk bkruse (~Adium@24.42.229.8) |
| 20:43.53 | paule32 | now, i bevome : SIP/2.0 403 Contact User und Anrufernummer verschieden |
| 20:44.46 | WIMPy | Then your register is wrong. |
| 20:44.55 | WIMPy | I wouldn't use register anyway. |
| 20:45.11 | ghost75 | omdomain=1und1.de <- omnomnom? |
| 20:45.43 | paule32 | i fix that |
| 20:45.46 | WIMPy | omnomnomdomain=pizza.de |
| 20:45.48 | paule32 | fromdomain |
| 20:46.14 | Qwell | WIMPy: German pizza? |
| 20:47.03 | WIMPy | Everyone eats pizza, don't they? All around the world. |
| 20:47.34 | WIMPy | had pizza today. But that come from the freezer. |
| 20:47.44 | ghost75 | fromdomain=sip.1und1.de |
| 20:47.49 | ghost75 | insecure=port,invite |
| 20:48.12 | ghost75 | classical it food |
| 20:50.20 | ghost75 | and in peer its not secret, its remotesecret |
| 20:52.42 | paule32 | thank you, had some options misspellings |
| 20:53.01 | paule32 | now, the registerrar is registered on server |
| 20:53.35 | ghost75 | somebody from vodafone said, if there are "too many" requests on their peer, then they treat the registered host as untrusted <- wtf |
| 20:54.27 | ghost75 | paule32: sip show registry, what refresh time do you have? |
| 20:54.33 | paule32 | o_O |
| 20:55.55 | *** join/#asterisk jasonwert (~w3rt@97-86-79-162.static.klmz.mi.charter.com) |
| 20:58.02 | paule32 | next |
| 20:58.04 | paule32 | i call the number but nothing happends |
| 20:58.11 | paule32 | dialplan? |
| 20:58.33 | ghost75 | u have one? |
| 20:58.48 | WIMPy | So we can call you now? |
| 21:00.45 | Penguin | ghost75: remotesecret vs secret has nothing to do with peer or a different type. It has to do with if you require authentication from them or if they require auth from you. |
| 21:01.04 | ghost75 | i know |
| 21:01.05 | paule32 | 936916580458 |
| 21:01.13 | paule32 | 4936916580458 |
| 21:01.39 | WIMPy | paule32: We already knew |
| 21:01.42 | Penguin | If they require auth from you and they never send auth to you, use remotesecrete and forget insecure. |
| 21:01.52 | Penguin | eww... remote secrete |
| 21:02.15 | WIMPy | filth |
| 21:02.26 | ghost75 | almost everybody uses secret wrong in isp peers |
| 21:02.49 | Qwell | ghost75: If remotesecret is not set, it will use secret. |
| 21:02.58 | WIMPy | You need additional peers for incomming calls based on host anyway. |
| 21:03.14 | ghost75 | yeah, but remotesecret is still enough |
| 21:03.27 | WIMPy | More work |
| 21:03.31 | WIMPy | =bad |
| 21:03.57 | paule32 | how shows the dialplan, when i try call 3242 (out lan no.)? |
| 21:04.15 | WIMPy | What? |
| 21:04.33 | paule32 | i have register me on a sip gateway |
| 21:04.41 | paule32 | here 1und1.de |
| 21:04.41 | WIMPy | Set verbose to 3 to see what you configured as dialplan. |
| 21:05.51 | paule32 | eehm nothing till yet |
| 21:06.55 | WIMPy | If you don't see anything, then there's no call. |
| 21:07.54 | ghost75 | prolly no phone in config |
| 21:09.43 | paule32 | ok, thanks all for helping |
| 21:10.02 | paule32 | have to go to bed - tomorrow is yet another day |
| 21:10.03 | paule32 | bye |
| 21:10.26 | ghost75 | bed verboten until works |
| 21:11.10 | WIMPy | Versager |
| 21:11.17 | WIMPy | ;-) |
| 21:18.24 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 21:18.24 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 21:19.06 | ghost75 | maybe i should try res_fax_digium |
| 21:20.12 | WIMPy | If I ever get boared, I will try fax. |
| 21:20.30 | WIMPy | -a |
| 21:20.32 | ghost75 | just for the kicks |
| 21:23.55 | WIMPy | We should call paule32 and tell hoim to finish work before going to sleep. |
| 21:24.06 | ghost75 | lol |
| 21:24.08 | *** join/#asterisk Bryanstein (~Bryanstei@shellium/admin/bryanstein) |
| 21:25.11 | ghost75 | create a call file |
| 21:29.31 | *** join/#asterisk Milarepa (~Milarepa@host-74-211-92-125.beyondbb.com) |
| 21:30.43 | *** join/#asterisk igcewieling (~igcewieli@ip98-183-26-100.pn.at.cox.net) |
| 21:31.50 | igcewieling | Has anyone tried linking conference bridges on different Asterisks using multicast? The traditional way is using a standard SIP call to do this. |
| 21:33.47 | igcewieling | An example of the traditional way is shown here: http://leifmadsen.wordpress.com/2011/08/11/connecting-two-conferences-on-initial-join-with-cleanup/ I'd like to try something similar with multicast. |
| 21:35.49 | hardwire | don't forget to add yoda voice clips while this is all happening. |
| 21:43.00 | *** join/#asterisk _abc_ (~user@unaffiliated/ccbbaa) |
| 21:43.25 | _abc_ | What is the oldest maintained release please? Was reading about the patches now? |
| 21:43.43 | _abc_ | 1.8.x? |
| 21:46.42 | Qwell | ~asterisk versions |
| 21:46.43 | infobot | Asterisk versions and their support levels are documented at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 21:48.10 | _abc_ | Thank you . |
| 21:50.37 | _abc_ | So 1.4.x is oldest LTS |
| 21:50.46 | hardwire | and sneakiest. |
| 21:50.53 | hardwire | runs away |
| 21:51.00 | _abc_ | ? |
| 21:58.17 | *** join/#asterisk issackelly (sid18035@gateway/web/irccloud.com/x-mvjercuaknqcctft) |
| 21:58.45 | issackelly | I'm making a game, and I'd like to record and playback at the same time, so there is background noise while I'm recording the player input. Is that possible? |
| 22:01.27 | *** join/#asterisk theron (~theron@69.63.185.56) |
| 22:06.30 | igcewieling | _abc_: read the page again. 1.4.x is the oldest LTS asterisk, but not a currently supported one. |
| 22:07.25 | igcewieling | issackelly: what version of Asterisk? |
| 22:07.59 | issackelly | igcewieling currently running what is default in wheezy, which I believe is 1.8. I'm not afraid to upgrade or compile even if that will solve this |
| 22:08.44 | ghost75 | same with digium fax: channel 'SIP/personalvoip_out-00000000' refused to negotiate T.38 |
| 22:09.54 | _abc_ | issackelly: Yes, I saw that. Thanks. |
| 22:10.20 | _abc_ | * igcewieling ^ |
| 22:14.53 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 22:25.25 | *** join/#asterisk theron (~theron@69.63.185.56) |
| 22:33.25 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 22:40.38 | issackelly | igcewieling: Is there maybe a way in a newer version? I'm not seeing it in the docs |
| 22:41.43 | WIMPy | Get a little creative and use Monitor. |
| 22:42.16 | *** join/#asterisk camerin (hoax@elite.bshellz.net) |
| 22:43.46 | mjordan | Or ChanSpy, Local channel to a Playback, and drop them into Record |
| 22:55.03 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 22:55.44 | powerunits | dear team. i need some info on bluetooth with asterisk .. can i configure more than 2 blue tooth with 1 asterisk server? |
| 23:00.29 | *** join/#asterisk JeffC_NN (~JeffC_NN@mail.nownerd.com) |
| 23:01.37 | JeffC_NN | Any thoughts on doing script-controlled warm-transfer via AMI? I've been using conferences so far, but I want a real transfer, and not all the baggage of a conference. |
| 23:03.33 | JeffC_NN | I don't know anything about AMI's Bridge command. Can you bridge more than 2 calls together? |
| 23:04.27 | JeffC_NN | I'm assuming you can't, and that you need an application like ConfBridge to mix the audio of 2+ channels, but how then does a warm-transfer work? |
| 23:06.44 | drmessano | powerunits, yes, but you'll need two GOOD dongles that have unique MAC addresses |
| 23:06.56 | drmessano | Most of the eBay stuff has the same MAC |
| 23:08.50 | powerunits | drmessano: can i use any dongle or there are some special bluetooth dongles? |
| 23:09.13 | issackelly | WIMPy: mjordan Thanks for the pointers, looking into it |
| 23:09.20 | drmessano | If its supported by the BT stack in Linux, it will work |
| 23:09.58 | drmessano | Unless you crap up some old BT 1.0 dongle |
| 23:10.51 | powerunits | drmessano: thanks... if possible can you give som model number that i can buy? |
| 23:11.35 | drmessano | Not really.. I buy $5 eBay dongles. But they're almost guaranteed not work with an N+1 scenario due to the aforementioned MAC address issue |
| 23:11.56 | ghost75 | bluez can be challanging |
| 23:12.13 | drmessano | That part is actually pretty easy |
| 23:12.29 | drmessano | and configuring a single dongle is pretty easy |
| 23:12.31 | ghost75 | if its not too old yes |
| 23:12.39 | drmessano | Anything 2.0 or newer |
| 23:12.41 | mjordan | JeffC_NN: warm transfers work natively in 12 |
| 23:12.53 | drmessano | Thats what, 8 years? |
| 23:12.55 | mjordan | before that, it's pretty hard. |
| 23:13.17 | ghost75 | in debian squeeze it was not working |
| 23:13.56 | drmessano | I had it working on CentOS in 2006, and Ubuntu thereafter |
| 23:14.06 | drmessano | Using EDR 2.1 dongles |
| 23:14.17 | drmessano | I think I got like 10 of them |
| 23:14.20 | ghost75 | in wheezy had it working too |
| 23:15.10 | drmessano | But anyway.. powerunits you'll probably need to buy some overpriced brand name dongles if you want unique MAC addresses |
| 23:15.16 | drmessano | I pity you |
| 23:15.25 | drmessano | No better quality other than not being stamped out clones |
| 23:15.57 | powerunits | drmessano: i got it.. but any good brand name and how much it can cost? |
| 23:16.04 | ghost75 | why need unique mac |
| 23:16.07 | drmessano | Again, no.. |
| 23:16.11 | powerunits | rit |
| 23:16.28 | mjordan | JeffC_NN: if you're interested, take a look at the atxferswap, atxferthreeway, and atxfercomplete options here - https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Configuration_features |
| 23:16.41 | drmessano | ghost75, N+1 you need different MAC addresses on the 2+ dongles. The cheap chinese dongles all have the same MAC |
| 23:16.50 | ghost75 | omg |
| 23:17.15 | mjordan | JeffC_NN: That lets you swap back and forth between the transfer destination and the transfer target; convert an attended transfer into a 3 party bridge; and 'complete' the transfer between the target and destination in the 'normal' way (respectively) |
| 23:17.43 | ghost75 | i have this http://www.dlbilder.de/produkte/orig/068e0ca21d889ed62d0a7ff4c85033bb.jpg |
| 23:17.54 | drmessano | Matter of fact, I think all my $5 dongles from way back when were 00:00:00:00:00:00 |
| 23:18.47 | drmessano | powerunits, there's a brand name |
| 23:18.55 | drmessano | Maybe you can buy 2 of those |
| 23:19.03 | drmessano | Otherwise, Google is your friend |
| 23:19.55 | powerunits | drmessano: thanks alot for your help and guide |
| 23:20.57 | drmessano | I really should go back to chan_mobile |
| 23:21.14 | ghost75 | i stopped using because my phone crashed |
| 23:21.17 | drmessano | With an Asterisk PBX at home and at work, I can put a dongle on each one, pair my phone, and route my calls that way |
| 23:21.25 | drmessano | Google voice be damned |
| 23:21.48 | file | Google Voice? What's that? |
| 23:22.04 | drmessano | file: Exactly |
| 23:22.18 | ghost75 | i remember something that cambridge chipset works good |
| 23:22.32 | drmessano | Google Voice was the OS before Vista that is being retired in April |
| 23:22.37 | drmessano | Maybe I have my wires crossed |
| 23:23.23 | drmessano | XP is the Google telephony thing they connected to XMPP that they're beheading in May |
| 23:23.28 | drmessano | THATS IT |
| 23:25.15 | drmessano | I really dont need Google Voice so much as the one-to-many allowed me to answer calls on Asterisk at home and work. chan_mobile would do the same, and I could port my GV number to my cell |
| 23:25.46 | ghost75 | i think google voice came never to my country |
| 23:26.05 | drmessano | I'll have to test the iPhone and the chan_mobile+dongle thing to see how well it works with that device |
| 23:26.41 | drmessano | I had an OLD ass Samsung in 2006 and even that far back I could tether from 40 feet away and the quality was amazing |
| 23:26.57 | powerunits | drmessano: as you said "powerunits, there's a brand name" please can tell me those brandnames? |
| 23:27.29 | drmessano | I could forget my phone in the car, go upstairs to my apartment, and never miss a call |
| 23:27.52 | *** part/#asterisk mjordan (~matt@nat/digium/x-yixnzucsdxchtxol) |
| 23:27.55 | file | gets a lot of missed calls since he doesn't answer his phone |
| 23:28.12 | drmessano | powerunits, seriouslly.. we went around in circles last time you were in here. ghost75 posted a pic of the dongle he uses. OPEN IT and LOOK. Beyond that, GOOGLE |
| 23:28.25 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 23:28.32 | ghost75 | i also hate it getting mobile calls at home |
| 23:28.39 | drmessano | *I do not use BRAND NAME DONGLES* and thats what YOU NEED |
| 23:29.01 | powerunits | rite thanks |
| 23:29.35 | drmessano | http://www.dlbilder.de/produkte/orig/068e0ca21d889ed62d0a7ff4c85033bb.jpg <---- |
| 23:29.55 | issackelly | mjordan: can you correct me if I'm wrong? -- If I want to use ChanSpy to play "background" audio, -- caller calls "agent" .. starts normal extension flow. When the call starts, I make a new outgoing call to the same extension which enables ChanSpy in barge mode and then does a Playback, I'm not sure how I can do this in all one dialplan |
| 23:30.22 | drmessano | powerunits, http://www.newegg.com/Network-Bluetooth/SubCategory/ID-295 |
| 23:31.13 | powerunits | drmessano: thanks alot |
| 23:41.15 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 23:51.15 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
