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09:04.47 | mccar | Hi! I have a problem. Using asterisk 1.8.15. I'd like to transfer incoming SIP call from my provider to mobile phone using another SIP account. Everything goes well, I got incoming call and than redirect call with Dial command (SIP/line_89128223344/89123332322). And after that I see incoming call on my mobile phone. But when I answer on call I can't hear on both side in about 20 seconds and... |
09:04.49 | mccar | ...then everything ok. Why this delays appear? Asterisk says that its a "Locally bridged" call. Is this a reason? |
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10:08.28 | WimDV | Hello. I have an issue with ccss on an asterisk 11.10.0 (can't upgrade this). The ccss works well. But when device A does a cc request to device B, when device B becomes available, device A receives a callback, so far no problem. However, when device A rejects the call, device B still receives the call. |
10:09.16 | WimDV | also maybe usefull, it's a freepbx 2.11 |
10:17.10 | mccar | Nobody there, friend. Except me) |
10:17.35 | mccar | And I can't help you.. |
10:20.27 | WimDV | basically the channel is not cleared when the phone sends back a 486 Busy... |
10:21.06 | WimDV | while my phone rings and i do core show channels it lists sip/102-something, is normal |
10:21.31 | WimDV | then when i press reject i see on the cli "got sip response 486...." also normal |
10:21.47 | WimDV | but then core show channels still lists sip/102-something, not so normal :-) |
10:23.31 | WimDV | maybe another sip response on a reject would help |
10:24.44 | WimDV | sip 603 "decline" has the same result :( |
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10:31.36 | WimDV | actually returning sip 404 works better |
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11:56.05 | snadge | is there such a thing as re-invite with IAX? |
11:56.30 | snadge | basically.. scratching my head trying to figure out how calls are going out a trunk, which for certain types of calls.. is supposed to be inbound only |
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12:03.55 | snadge | so the call comes in .. is sent to customers pbx via IAX.. then call makes its way back out the same trunk |
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12:23.26 | [TK]D-Fender | snadge: It's your dialplan... itt goes where you tell it to. |
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12:30.26 | snadge | might actually be a glitch in the monitoring tool we use.. apparently if a call is longer than 4 hours, it gets confused.. hehe |
12:30.59 | snadge | so according to asterisk logs, on both the incoming trunk for that number, and the core call termination server.. there was only an inbound call |
12:31.08 | snadge | and that matches with the cdr |
12:31.46 | snadge | 6 hours, 11 minutes and 28 seconds.. longest call i've seen in a while.. maybe it was a lost BYE |
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14:31.10 | qakhan | i have setup a queue with 15 agnets 7801 - 7015 and they are receiving calls, i want to setup a new number to send calls to first 8 agents |
14:31.55 | qakhan | is it possible if someone dialing number 123-456-7890 then call only ring on first 8 agents |
14:34.14 | [TK]D-Fender | Member penalties <- |
14:34.17 | [TK]D-Fender | Tthis is queue basics |
14:34.24 | [TK]D-Fender | Read the sample configs and the apps instructions |
14:34.33 | [TK]D-Fender | Like we keep telling you to |
14:36.52 | keithf | higher priority extensions would ring first then the call would move down to the extensions with lower. Thats how i am doing it here. |
14:37.11 | keithf | have priority setup so that the top guys get calls first since they are rock solid then from there the calls trickle down |
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14:50.10 | monsterco | What is a good free IAX2 client out there for windows? |
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14:50.48 | monsterco | What is a good free IAX2 client out there for windows? |
14:56.18 | [TK]D-Fender | There are vey few at all |
14:56.27 | [TK]D-Fender | Zoiper is one of the few tthat is maintained |
15:02.51 | qakhan | [TK]D-Fender call coming on 123-456-7890 only ring on first 8 agents. i dont want to send call to remaing last agents |
15:04.49 | [TK]D-Fender | So either set a different penalty on entry, or another queue |
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15:28.34 | ModFather | Hi There, i am using: same => n(menu-ivr),Background(/var/lib/asterisk/sounds/open-hours) |
15:29.00 | ModFather | to playback a .wav file with a voice message, and after that i have WaitExten |
15:29.09 | ModFather | how i can loop music while waitingExten ? |
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15:33.50 | [TK]D-Fender | background it several times |
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15:44.05 | ModFather | [TK]D-Fender i thought about it.. but i am wondering if i do that, how i can make "WaitExten" to be active while i am background it |
15:44.28 | [TK]D-Fender | that's what background IS |
15:44.46 | [TK]D-Fender | you fire it of.. then CONTINUE to do other stuff... |
15:44.50 | [TK]D-Fender | including WaitExten |
15:46.41 | ModFather | [TK]D-Fender |
15:46.42 | ModFather | http://pastebin.ca/3401875 |
15:47.08 | [TK]D-Fender | No. |
15:47.18 | [TK]D-Fender | I did NOT say to call WaitExten multiple times |
15:47.29 | [TK]D-Fender | BACKGROUND Multipl, THEN Waitexten |
15:47.30 | ModFather | yes sorry |
15:47.51 | [TK]D-Fender | Make sure to have an adequate pause in what you are playing back. |
15:48.25 | ModFather | i want WaitExten to be available from 1st second of Playback |
15:48.42 | [TK]D-Fender | It is |
15:49.13 | [TK]D-Fender | You also put duplicate labels there... bad |
15:49.37 | [TK]D-Fender | And that looks like the usual stock sounds folder. You should be able to use the REALTIVE path instead of specifying it like thatt |
15:49.50 | [TK]D-Fender | same => n,WaitExten(20),Background(/var/lib/asterisk/sounds/open-hours),Background(/var/lib/asterisk/sounds/open-hours),Background(/var/lib/asterisk/sounds/open-hours) |
15:49.54 | [TK]D-Fender | And this... no |
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15:50.04 | [TK]D-Fender | You can't jsut glue 10 things together on a line like that |
15:50.13 | ModFather | http://pastebin.ca/3401879 |
15:50.24 | ModFather | i understand, so in a new line |
15:53.05 | [TK]D-Fender | .... |
15:53.14 | [TK]D-Fender | [11:47][TK]D-FenderBACKGROUND Multipl, THEN Waitexten <------------------------ |
15:53.32 | ModFather | yes already fixed ;) |
15:53.35 | [TK]D-Fender | same => n,WaitExten(20),Background(/var/lib/asterisk/sounds/open-hours) <- NO |
15:54.27 | ModFather | open-hours is a .wav file |
15:55.26 | ModFather | is it bad advance to store my "custom" voice files on the standard asterisk sound folder? |
15:57.22 | [TK]D-Fender | no, but it's best to put the in a convenient sub-folder so you can back them up easier and not mix them with the others |
15:57.31 | [TK]D-Fender | Background(custom/myfile) |
15:58.55 | ModFather | yep thanks for sharing , very nice! |
16:05.05 | ModFather | [TK]D-Fender http://pastebin.ca/3401894 |
16:05.18 | [TK]D-Fender | ... |
16:05.22 | [TK]D-Fender | facpalms |
16:05.28 | ModFather | lol |
16:05.29 | [TK]D-Fender | NO |
16:05.39 | [TK]D-Fender | YOU CAN'T CALL MULTIPLE APPS ON THE SAME DAMN LINE |
16:06.05 | [TK]D-Fender | exten => extension,priority(label),application(data) |
16:06.18 | [TK]D-Fender | not : exten => extension,priority(label),application(data),application(data),application(data),application(data),application(data),application(data),application(data) |
16:06.43 | ModFather | oh got it |
16:06.56 | [TK]D-Fender | Dialplan 101 |
16:07.06 | ModFather | haha |
16:09.19 | ModFather | http://pastebin.ca/3401896 |
16:09.21 | ModFather | finally |
16:09.43 | [TK]D-Fender | .... |
16:09.45 | ModFather | i had a typo next to the WaitExten interval (20 it should be (20) |
16:09.50 | [TK]D-Fender | 2 FAILS |
16:09.59 | [TK]D-Fender | background |
16:10.01 | [TK]D-Fender | BACK TO BACK |
16:10.12 | [TK]D-Fender | NOT WITH A WAITEXTTEN between them |
16:10.32 | [TK]D-Fender | [11:53][TK]D-Fender[11:47][TK]D-FenderBACKGROUND Multipl, THEN Waitexten <------------------------ |
16:10.37 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
16:11.16 | [TK]D-Fender | aND THAT IS NOT A relative PATH, YOU PUT A LEADING SLASH IN FRONT |
16:11.23 | ModFather | yes |
16:11.25 | ModFather | i know fixed |
16:12.12 | ModFather | the waitextens should be 1st |
16:12.24 | ModFather | then Mulitpl background |
16:12.35 | ModFather | this was my logic before |
16:15.27 | ModFather | [TK]D-Fender ^^ http://pastebin.ca/3401900 |
16:15.41 | [TK]D-Fender | NO |
16:15.49 | ModFather | [11:47] [TK]D-Fender BACKGROUND Multipl, THEN Waitexten <------------------------ |
16:15.57 | [TK]D-Fender | NOT ON THE SAME DAMN LINE |
16:16.50 | ModFather | [TK]D-Fender how i can multipl Background and get Active WaitExten? |
16:17.18 | ModFather | http://pastebin.ca/3401904 |
16:18.32 | [TK]D-Fender | http://pastebin.ca/3401905 |
16:18.59 | [TK]D-Fender | Clear now? |
16:19.13 | [TK]D-Fender | [12:06][TK]D-Fenderexten => extension,priority(label),application(data) |
16:19.15 | [TK]D-Fender | [12:06][TK]D-Fendernot : exten => extension,priority(label),application(data),application(data),application(data),application(data),application(data),application(data),application(data) |
16:19.56 | ModFather | Clear clear |
16:20.05 | ModFather | http://pastebin.ca/3401904 |
16:20.09 | [TK]D-Fender | [12:17]ModFatherhttp://pastebin.ca/3401904 <- 32nd times the charm |
16:20.32 | ModFather | is that right? |
16:20.35 | [TK]D-Fender | yes |
16:20.38 | ModFather | 1 application per line |
16:21.08 | [TK]D-Fender | Yes, finally... |
16:21.12 | ModFather | [TK]D-Fender http://pastebin.ca/3401907 is that legal? |
16:21.23 | [TK]D-Fender | yes |
16:21.37 | ModFather | hmm |
16:22.35 | ModFather | so if i dial exten 1 while on the 1st playback WaitExten should get it? |
16:23.07 | [TK]D-Fender | watch |
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16:26.51 | ModFather | [TK]D-Fender worked but regarding the: adequate pause .. can i have a "pause" between playbacks? |
16:27.00 | ModFather | 2 seconds pause ? |
16:27.09 | [TK]D-Fender | Background(silence/5) |
16:27.09 | ModFather | or i need to make re record the audio file? |
16:27.13 | ModFather | oh |
16:27.16 | [TK]D-Fender | Look at the increments in that folder |
16:27.20 | ModFather | yep |
16:30.11 | ModFather | [TK]D-Fender thanks worked |
16:32.07 | ModFather | [TK]D-Fender i had this: same => n,Dial(SIP/twilio0/+1${EXTEN}) |
16:32.21 | ModFather | my phones when i am calling over US, i dont need to put +1 front of the number |
16:32.24 | ModFather | and its working |
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16:32.46 | ModFather | its any regex or "tip" that i can have both ways to work? |
16:32.59 | ModFather | i mean even if i dial with +1 or without +1 |
16:33.02 | litn | hello, after changing manager information in manager.conf, how do I reload the agi or whatever so that the changes go into affect without restarting asterisk? |
16:33.58 | [TK]D-Fender | multiple patterns |
16:34.17 | [TK]D-Fender | litn: "manager reload" |
16:34.26 | [TK]D-Fender | litn: And this isn't AGI..... |
16:36.20 | litn | ah |
16:37.05 | ModFather | [TK]D-Fender yep did it, and worked thank you |
16:38.29 | jameswf | should unc_periodic_hook just run without being called? |
16:38.37 | jameswf | *func_periodic_hook |
16:38.48 | jameswf | or set |
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17:38.31 | jwpierce3 | How do I create a "dummy extension" for voicemail purposes? |
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17:44.08 | jameswf | voicemail.cong |
17:44.11 | jameswf | *conf |
17:44.15 | mub | jwpierce3: Through whatever web interface you've got for it... |
17:46.51 | jwpierce3 | mub, there is no interface. The extension doesn't exist. I want to be able to dial the "dummy extension" to check voicemail, as well as send calls to it after dialing a ring group in extensions.conf |
17:48.15 | mub | jwpierce3: Open the file jameswf said (/etc/asterisk/voicemail.conf) and add an entry there |
17:48.45 | jameswf | jwpierce3: http://www.asteriskguru.com/tutorials/asterisk_voicemail.html |
17:49.46 | jwpierce3 | thanks, mub jameswf |
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18:46.08 | Kobaz | soooo |
18:46.24 | Kobaz | i have a customer on a sonicwall and they have some phones connected to a hosted box |
18:46.30 | Kobaz | aaaaaand all the phones just dropped off completely... can't register |
18:46.50 | Kobaz | anything to keep in mind when debugging this? I have all the SIP translations turned off, and there's consistent nat enabled |
18:47.07 | Kobaz | not seeing any sip traffic at all from their network |
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19:52.31 | _boot | Hi, was hoping someone could point me in the right direction... have an IP phone connected to Asterisk which then talks to sipgate, when making/receiving calls the audio from the ip phone seems to hit the other end, but i can't hear anything on the ip phone. if i use the ip phone to transfer the call to a confbridge and then join the bridge as well, the audio seems to work from both sides - any ideas as to |
19:52.37 | _boot | <PROTECTED> |
19:55.07 | [TK]D-Fender | You're allowing reinvites between your phone & provider |
19:55.21 | [TK]D-Fender | "directmedia=no" <- this should be set for ALL peers |
19:55.41 | _boot | oh, nice - will try that. thanks! |
19:57.13 | _boot | worked! that's been driving me crazy for the last few days - thank you very much indeed |
20:03.26 | [TK]D-Fender | You're welcome |
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