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| 00:02.21 | MarkS- | [TK]D-Fender: I want to combine it in a dialplan part, as that is easier for users to remember/use |
| 00:03.04 | [TK]D-Fender | there is no "combine" |
| 00:04.49 | MarkS- | ok, so looking for phones with a key that isn't used is probably the easiest solution (to configure it to do <blindtransfer key>number) to make it easier for them. Thank you for the information so I can stop searching for it! |
| 00:06.18 | Penguin | You can do attended transfers and/or blind transfers. Just not both at the exact same time from the same device. |
| 00:06.43 | Penguin | features.conf has examples of using both. |
| 00:10.19 | MarkS- | Penguin: I know how to use both, however making 1 key on the device to be used for both options would be nice (currently they are using another option for it) |
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| 00:11.10 | WIMPy | Maybe you should tell us how this "both option" could work from your/the users point of view. |
| 00:14.22 | MarkS- | if you transfer to number 01/02/03 -> do it like blind transfer (disconnect the phone direct, no feedback required), for everything else -> do it like attended transfer (first talk to the one talking to the client after you before transferring) |
| 00:15.25 | WIMPy | So it has to depend on the destination number? |
| 00:15.41 | [TK]D-Fender | Completely not happening... |
| 00:16.40 | [TK]D-Fender | Transfering is <how> <where>, not <where> (processing) <how> |
| 00:19.49 | MarkS- | WIMPy: yes, that is the idea |
| 00:24.49 | WIMPy | As [TK]D-Fender said, that's not going to happen from within Asterisk. |
| 00:24.58 | WIMPy | It might be possible with some external trickery. |
| 00:25.38 | [TK]D-Fender | Perhaps via an applicationmap dynamic feature that redirects the call to a specific spot |
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| 00:26.00 | WIMPy | OTOH, if you want it fast and easy, better find some spare keys for the quick destinations. |
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| 00:27.44 | MarkS- | thanks for the information, maybe something to check tomorrow but it is likely that we will go for some seperate keys) |
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| 01:16.14 | bambu | Hi. |
| 01:16.45 | bambu | Is it possible to use my POTS telephone with asterisk directly plugged into my computers rj11 jack? |
| 01:18.15 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 01:18.47 | bambu | When the POTS phone is picked up I want a dial tone to be presented from asterisk |
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| 01:19.10 | bambu | so I assume I want DISA? |
| 01:19.25 | Penguin | No. |
| 01:20.07 | Penguin | DISA is for when you want to dial a phone number, get an answer, and then get another dial tone to make another call out of the first one. |
| 01:21.27 | Penguin | When you have a proper TDM interface to asterisk on your computer, it will provide the dial tone. |
| 01:21.59 | bambu | Is that the only method? |
| 01:22.10 | Penguin | You can also use an ATA. |
| 01:23.47 | bambu | So there is no way with my existing hardware (rj11 jack) and software to do this? |
| 01:24.00 | Penguin | Not likely. |
| 01:24.21 | Penguin | The jack on your computer is most probably a modem jack. |
| 01:24.28 | bambu | Correct. |
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| 02:01.16 | fling | Hello. |
| 02:01.44 | fling | [TK]D-Fender: Is not it possible to mix video streams together in a conference? |
| 02:03.55 | [TK]D-Fender | no |
| 02:04.16 | [TK]D-Fender | "Follow The Speaker" only |
| 02:04.25 | [TK]D-Fender | Same condec, no transcoding offered |
| 02:04.28 | [TK]D-Fender | codec* |
| 02:05.23 | Penguin | 0014.46) <fling> Is it possible to mix videostreams? |
| 02:05.23 | Penguin | (0017.48) <WIMPy> no |
| 02:05.23 | Penguin | (0017.49) <WIMPy> They can only be switched, depending on who's taling. |
| 02:05.44 | Penguin | Let's ask a few more times to see if the answer ever becomes YES. |
| 02:06.15 | WIMPy | Oh, I guess it will become yes sometime. But that might take a few years. |
| 02:06.25 | fling | Penguin: but what if there is some insane solution like sending all the video streeams to a standalone app for mixing? |
| 02:06.31 | fling | What if I will use ffmpeg somehow? |
| 02:06.49 | fling | Is it possible to send all the videostreams to an external app? |
| 02:07.05 | [TK]D-Fender | fling: * doesn't pass the frames anywhere for rewriting |
| 02:07.06 | fling | audio/video sync problem will appear |
| 02:07.08 | WIMPy | Ot's open source. You can make it whatever you want. |
| 02:07.15 | WIMPy | It's... |
| 02:07.39 | [TK]D-Fender | fling: YES... it IS possible... if YOU go massively rewrite the modules code YOURSELF. |
| 02:07.50 | [TK]D-Fender | fling: Get to work.... |
| 02:08.06 | fling | OK. |
| 02:08.10 | [TK]D-Fender | fling: till then there is no means of passing these multiple streams anywhere via * |
| 02:08.23 | fling | [TK]D-Fender: why? |
| 02:08.30 | [TK]D-Fender | fling: Because the code has no hooks |
| 02:08.39 | [TK]D-Fender | because nobody decided to go through the work |
| 02:08.43 | [TK]D-Fender | because they don't care. |
| 02:08.44 | fling | ohh |
| 02:09.27 | [TK]D-Fender | heads out for the evening |
| 02:09.51 | fling | Thanks for the info guys! |
| 02:10.05 | fling | I'm asterisk lover I don't want to use different software for my calls. |
| 02:10.39 | fling | But now I have a task to implement this thing -> http://mirror.dno.so/incoming/2014.10.21-12%3a25%3a22.991988040.png |
| 02:11.01 | fling | So I will look for an app to use together with asterisk somehow⦠|
| 02:11.11 | fling | hmm hmm probably freeswitch or something |
| 02:14.37 | TazzNZ | since it's in russian mind explaning what it is ? |
| 02:16.39 | fling | TazzNZ: 300 hardware video phones in a sip/xmpp video conference. |
| 02:17.07 | fling | TazzNZ: two of them shown on the main part of the screen on top |
| 02:17.23 | fling | TazzNZ: others are shown at the bottom. |
| 02:17.37 | fling | TazzNZ: pbx is switching video when they start talking (probably). |
| 02:32.09 | fling | https://www.youtube.com/watch?v=T7gz9Ny9WVc |
| 02:32.21 | fling | [TK]D-Fender: ^ is not it based on asterisk actually? |
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| 03:39.13 | jpsharp | It's been a while since I mucked with Asterisk and analogue lines, and I'm having a heck of a time with them. I'm trying to place outgoing calls via an analogue line and the call gets dropped about 20 seconds afterwards. The far end does answer. I'm trying to figure out why the call is being dropped. |
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| 03:54.37 | jpsharp | Okay, the dahdi outgoing call never sees the line as answered. |
| 03:54.49 | jpsharp | So, now I just have to figure out why. |
| 04:25.05 | jpsharp | shouldn't a dahdi analog channel immediate go to answered imediately after dialing if call progress is turned off |
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| 06:11.30 | ChannelZ | jpsharp: Yes. Have you done a dahdi_monitor to listen to what is going on? You might be dialing before the dialtone.. try putting a 'w' or two in front of the number |
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| 07:43.17 | gciotta | Hi, is there a place (ideally a database) where I can find information related to my ACD queues? I'm thinking of something like the CDR tables but focused on ACD data. Thanks! |
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| 07:51.04 | gciotta | Just found something.. http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL |
| 07:53.10 | Pernat | hi how redirect call from gsm to voip and answer another call via voip from the same gsm number? |
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| 11:11.00 | Stefan27 | ¨hi |
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| 12:27.44 | Pernat | hi how redirect voip calls via asterisk - i have 4 voip numbers and i don't know how redirect call when one number is busy to free one |
| 12:31.45 | cunningpike | Pernat: Consecutive Dial() commands. If the first fails, the second one will be executed, and so on. |
| 12:34.10 | Pernat | how it works? |
| 12:35.22 | Pernat | i want make that if someone calls first voip number and it is busy that he call to next number |
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| 12:40.04 | jdzielny | hello everyone. I've got asterisk 12.6.1 running on a Ubuntu x86_64 system, and I'm not getting any CDR entries |
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| 12:58.13 | zpotoloom | hi, anybody knows in which version the queue penalty system is fixed ? |
| 12:58.13 | zpotoloom | with 1.8.30 it's not working although according to jira it was fixed in 1.8.26 ASTERISK-19368 |
| 12:58.13 | zpotoloom | http://pastebin.com/YecubyGi |
| 12:58.32 | zpotoloom | so I guess that the fix was never pushed to 1.8 branch and only to 11 |
| 12:58.48 | zpotoloom | https://issues.asterisk.org/jira/browse/ASTERISK-19368 |
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| 14:48.06 | zpotoloom | looking at the code i see no diff in the queue code related to penaltys between 1.8 and 11 branch |
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| 15:28.11 | stevenm | Lo, does anyone know of a box (analogue preferably) that'd auto answer any inbound call and pass the audio it receives from that call out of a 3.5mm or other audio type jack? |
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| 16:43.51 | Stefan27 | I have a weird asterisk-issue which might only be producable under my special environment and load. I tested having 20 simultaneous calls, which worked fine for ~2 hours but then the sip-udp-socket seemed to become unresponsive and asterisk entered a weird state. I described it in http://pastebin.com/e8psZePG including a threaddump and other checks I did. |
| 16:43.59 | Stefan27 | If anyone has the time to look at it id be grateful :) |
| 16:48.01 | [TK]D-Fender | #2 0x08225d93 in ast_sem_wait (sem=0x952a860) at /usr/src/debug/asterisk-12.6.0/include/asterisk/sem.h:59 <- 12.6.1 is out. As usual you'll need to replicate this on the latest |
| 16:49.29 | pabelanger | anybody know a SIP client that supports SIP REFER transfers? |
| 16:51.12 | [TK]D-Fender | Bria, etc all do I'm sure |
| 16:51.40 | [TK]D-Fender | Basically probably just about everything except crippled versions of commercial products |
| 16:51.48 | [TK]D-Fender | ie: X-Lite |
| 16:52.47 | pabelanger | more specifically, if I send a SIP REFER to the softphone, it will react properly |
| 16:52.52 | Stefan27 | ok ill download new source tonight |
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| 16:55.36 | Stefan27 | what I found weird was the following: strace -c -o /tmp/strace3.out -s 2000 -fp <asterisk-pid> ; gives (partial) output after 10 seconds: |
| 16:55.36 | Stefan27 | % time seconds usecs/call calls errors syscall |
| 16:55.36 | Stefan27 | <PROTECTED> |
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| 16:59.29 | Stefan27 | i'm using an installation of fedora 20 on a virtual machine - has anyone had experience with such a setup where nanosleep bugs out? |
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| 17:04.44 | pabelanger | okay, Jitsi does it |
| 17:04.49 | pabelanger | appartently blink doesn't |
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| 17:10.49 | jpsharp | I'm getting back to my analogue dial issue and I'm still rather confused. I have two boxes, appear to be configured identically, but one behaves and the other doesn't. |
| 17:13.53 | jpsharp | One box gives me an answer on outbound calls, the other box just gives me "making progress" and never sees an answer. |
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| 17:34.13 | qakhan | Hi all, is it possible we can run 10 sec 1st file and 10 sec 2nd file of MOH in queue |
| 17:36.19 | [TK]D-Fender | qakhan: No. Go edit your files |
| 17:37.41 | qakhan | musiconhold.conf? |
| 17:37.50 | [TK]D-Fender | AUDIO FILES |
| 17:37.54 | jpsharp | No, your actual audio files |
| 17:39.26 | qakhan | ok so i have to murge them in single file |
| 17:39.33 | qakhan | right? |
| 17:40.06 | [TK]D-Fender | qakhan: You see what order they play in. Change the files themselves however you want |
| 17:40.32 | qakhan | ok |
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| 18:23.37 | burnbrighter | Hi All - Iâm now getting a busy tone on my inbound calls from viatalk. I havenât changed anything on my set up. If anyone cares to have a look, here are all of my relevant SIP settings. Thanks in advance for any help. Outgoing calls are working ok - I still need to figure out why dialplan forces me to use 1XXX (area code) locally. http://pastebin.com/yQJ1X9Xg |
| 18:24.36 | Penguin | It's your dialplan, so it's not forcing anything. You make it do what you want it to do. |
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| 18:24.49 | burnbrighter | right :) |
| 18:25.01 | [TK]D-Fender | burnbrighter: And you aren't showing us the call. |
| 18:25.23 | burnbrighter | <â lack of knowledge forcing that problem |
| 18:25.25 | [TK]D-Fender | burnbrighter: Which is like showing us a sales flyer for a new car and then immediately asking us why yours crashed |
| 18:26.07 | [TK]D-Fender | "sip set debug on", "core set verbose 10" <----------- |
| 18:26.40 | burnbrighter | [TK]D-Fender: youâve been extremely helpful in the past, thanks for that. I did set verbosity up, will try another call again. Give me a minute please. |
| 18:29.04 | burnbrighter | here you go: http://pastebin.com/0Kj4pZzk |
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| 18:31.19 | [TK]D-Fender | burnbrighter: There is no inbound call there at all |
| 18:31.58 | burnbrighter | [TK]D-Fender: I dial from my mobile and it getsa a busy signal |
| 18:32.08 | [TK]D-Fender | busNot from your server you don't |
| 18:32.22 | [TK]D-Fender | burnbrighter: There are no packets for an inbound call in what you showed us |
| 18:33.30 | burnbrighter | I donât see anyting from â198.8.60.11â hitting the firewall |
| 18:34.31 | burnbrighter | thatâs âsanfrancisco-1.vtnoc.net" |
| 18:34.48 | [TK]D-Fender | Just because that is the IP you are registering to doesn't mean that's where they send you calls from |
| 18:36.16 | burnbrighter | right, looking also for incoming to port 5060,5061 and 5080-5081 (alternate ports which Iâm not sure are enabled) |
| 18:37.35 | burnbrighter | [TK]D-Fender: how does the SIP provider know what your IP is and where to send traffic to? |
| 18:37.45 | burnbrighter | in my case - VIATALK |
| 18:38.00 | burnbrighter | there is some kind of registration process, right? |
| 18:38.20 | Penguin | That's what registration does. |
| 18:38.25 | Penguin | that's the purpose of it. |
| 18:38.43 | burnbrighter | I think I show the device is registered, right? |
| 18:38.48 | burnbrighter | in the first post |
| 18:38.53 | burnbrighter | (pastebin) |
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| 18:40.09 | burnbrighter | VIATALKs logs show the call was answered |
| 18:42.07 | [TK]D-Fender | [14:40]burnbrighterVIATALKs logs show the call was answered <- your sip debug shows no call. |
| 18:42.20 | [TK]D-Fender | burnbrighter: Did you LIMIT what * was looking at for debug? |
| 18:42.24 | burnbrighter | [TK]D-Fender: I know, I agree. |
| 18:42.40 | [TK]D-Fender | burnbrighter: As is by peer/ip? |
| 18:42.51 | burnbrighter | I set it up exactly as you sent |
| 18:43.04 | [TK]D-Fender | burnbrighter: Well there is no call there... |
| 18:43.25 | [TK]D-Fender | burnbrighter:Again, the server that sends you calls may not be at the same IP as the one you register to. |
| 18:43.35 | burnbrighter | right |
| 18:43.50 | burnbrighter | I was looking on the firewall log for anything related |
| 18:44.34 | burnbrighter | thinking maybe firewall was blocking it or something, but I donât think thatâs the case |
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| 18:48.08 | [TK]D-Fender | Look > Think |
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| 18:51.13 | burnbrighter | [TK]D-Fender: not seing anyting inbound or outbound to any of those ports |
| 18:54.02 | [TK]D-Fender | asky to provider |
| 18:54.10 | [TK]D-Fender | ask you* |
| 18:54.12 | [TK]D-Fender | +r |
| 18:55.13 | burnbrighter | yes, I put in a ticket |
| 18:55.25 | burnbrighter | their CS is horrible |
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| 19:44.52 | Penguin | I have exten h with 1 priority in a parent context and another exten h with 2 priorities in an including context. When the call hangs up while in the parent context, it runs h,1 like it's supposed to, but then jumps over to h,2 from the including context. I realize this is behaving properly... |
| 19:45.35 | Penguin | Correction, the included context's h has 3 priorities, not 2. |
| 19:46.10 | Penguin | So can I run Hangup() at h,2 in the parent context to prevent it from hitting h,3 in the included context? |
| 19:46.32 | Penguin | or would two NoOp()s be more appropriate? |
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| 20:40.35 | burnbrighter | [TK]D-Fender: This turned out to be a DNS issue. I am back in order |
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| 20:47.56 | edong23 | using the L dial option to limit the call to x ms... im having a bit of trouble finding an example of this... any references? |
| 20:48.37 | edong23 | or should i use Set(TIMOUT)? |
| 20:50.56 | Penguin | Dial(SIP/itsp/${EXTEN},,L(1199990:120000:60000)X); |
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| 20:52.03 | Penguin | core show application Dial |
| 20:55.30 | edong23 | Penguin: but the first option after L is the only requirement, right? |
| 20:55.58 | edong23 | core show gives me what i find on google, but didnt have a real example |
| 20:56.06 | edong23 | thank you Penguin |
| 20:56.11 | edong23 | i saw a few online... that were... ok |
| 20:56.23 | edong23 | but they were usually blabbering lines in freepbx |
| 20:56.30 | edong23 | i assume generated from the web gui |
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| 20:58.06 | Penguin | L(x) or L(x:y) or L(x:y:z) or L(x::z) |
| 20:58.11 | Penguin | Yes, x is required. |
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| 21:45.18 | burnbrighter | Penguin: we were talking about dialing plans earlier - can you help me with this part? |
| 21:47.03 | burnbrighter | assuming that âdial patterns that will use this routeâ in my route is the same thing - my problem is, as I mentioned above, I always have to dial 1+area code to dial a local number |
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| 22:41.06 | pederindi | Hi, I'm getting no sound in the two places of a call in a lan |
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| 22:42.10 | pederindi | but the sip connection works, so the data stream is not transmited, any help? thanks |
| 22:42.34 | [TK]D-Fender | "sip set debug on", "core set verbose 10" |
| 22:42.36 | [TK]D-Fender | ~pb |
| 22:42.36 | 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. |
| 22:42.37 | [TK]D-Fender | ^^^^ |
| 22:42.40 | [TK]D-Fender | Show us the call |
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| 22:43.37 | pederindi | [TK]D-Fender: how can I put the debug in a file? if not, I will use scroll in tmux... |
| 22:45.55 | pederindi | [TK]D-Fender: this is a "sip set debug on" that I did yesterday http://pastebin.com/j0tTrYCw |
| 22:46.17 | pederindi | meanwhile I will do another with "core set verbose 10", thanks for the help |
| 22:49.33 | [TK]D-Fender | pederindi: Peer audio RTP is at port 37.133.33.141:7078 |
| 22:49.46 | [TK]D-Fender | pederindi: Your load device is giving the PUBLIC IP as the source for audio. |
| 22:50.07 | [TK]D-Fender | Either fix the NAT detection or it, or se your peer to "nat=yes" |
| 22:53.57 | edong23 | does "nat=yes" still work? |
| 22:54.06 | edong23 | i thought it was comedia.. something |
| 22:54.22 | edong23 | nat=force_rport,comedia |
| 22:54.42 | edong23 | guess it depends on the version, but im thinking it will be removed soon |
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| 22:55.49 | pederindi | puting the devices with nat=yes or nat=force_rport,comedia, it still not working |
| 22:56.00 | pederindi | i do not understand the concept of "load device" |
| 22:56.14 | pederindi | but is strange that if is a LAN call, why it takes the public ip |
| 23:05.28 | [TK]D-Fender | pederindi: You should also put "directmedia=no" in each |
| 23:07.24 | pederindi | [TK]D-Fender: that's true, now it works, but I do not understand |
| 23:07.40 | pederindi | another test was to use (free) linphone voip service |
| 23:08.03 | pederindi | by default directmedia=yes, and in that case I put "nat=yes" |
| 23:08.06 | pederindi | and it worked... |
| 23:08.26 | pederindi | I mean, I did not put directmedia=yes, but by default it is enabled, it is said in sip.conf commentaries |
| 23:08.36 | pederindi | (really thanks for your help!!) |
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| 23:10.12 | pederindi | Aha, I know why I took the public ip, the linphone sip devices have implemented stun server... that's how they got it |
| 23:10.29 | [TK]D-Fender | id you allow directmedia the endpoints will renegotiate RTp which will lead to to bogus public IP offer and harpin-NAT scenario resulting in no audio |
| 23:12.17 | pederindi | no directmedia means that asterisk pass the stream, so it is centralizing the communication... no? |
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| 23:13.04 | pederindi | that asterisk process the stream * (relays it) |
| 23:21.25 | [TK]D-Fender | correct |
| 23:21.40 | [TK]D-Fender | enforcing the NATrestriction on the offered media IP\ |
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| 23:26.12 | pederindi | [TK]D-Fender: I could not get it without your help, thanks again |
| 23:29.41 | pederindi | however, this is a bad solution |
| 23:30.29 | pederindi | the sip clients need stun server to arrive to outside calls, but also need to do internal calls without the needs on asterisk local bridging... |
| 23:36.41 | [TK]D-Fender | Why should the clients be talking directly with the outside? |
| 23:38.35 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 23:39.23 | pederindi | imagine a system with a federated voip system |
| 23:39.47 | pederindi | only it makes sense to centralize calls to outside in cases of sip providers |
| 23:53.07 | *** join/#asterisk newtonr (~newtonr@wsip-24-120-166-97.lv.lv.cox.net) |
| 23:53.07 | *** mode/#asterisk [+o newtonr] by ChanServ |
