| 00:00.09 | [TK]D-Fender | That description is pretty broken |
| 00:00.22 | [TK]D-Fender | Please rephrase |
| 00:01.20 | MorpheuBr | im new to this, i work for a company that use voip, i have to connect to a server that works with g729 now, and zoiper dont work anymore |
| 00:01.55 | [TK]D-Fender | So you are looking for a FREE softphone that supports G.729? |
| 00:02.08 | WIMPy | You have to buy some. Either a aicence or a client that includes it. |
| 00:02.09 | MorpheuBr | YES |
| 00:02.30 | MorpheuBr | theres no free option? |
| 00:02.46 | WIMPy | No legal one. |
| 00:02.53 | MorpheuBr | im using x-pro, but is very bad, ppl dont hear correctly |
| 00:02.54 | [TK]D-Fender | MorpheuBr: G.729 is a licensed codec and costs money.... there shouldn't be any "free" solutions out there unless omeone has just started a charity |
| 00:03.53 | MorpheuBr | i never heard about that, so i have to pay for a softphone client now |
| 00:03.54 | [TK]D-Fender | Softphones suck. All of them. Some slightly harder than others. |
| 00:04.31 | [TK]D-Fender | You have to pay for G.729. It is a patented and licensed codec |
| 00:04.33 | MorpheuBr | what you recomends? i have some linkys phone, but i prefer to use in the computer |
| 00:04.46 | [TK]D-Fender | If G.729 is the only thing your other end accepts... poor you I guess |
| 00:05.12 | Penguin | That's weird. I thought my android soft phones had g729 capability. |
| 00:06.03 | WIMPy | Penguin: Some have. But they state very clearly that you must not use it unless you got the licence yourself. |
| 00:06.18 | WIMPy | It's up to you what you make of that. |
| 00:06.30 | Penguin | It must not have been all that clear, since I never saw that. |
| 00:07.19 | MorpheuBr | where do i find a payd option? |
| 00:07.52 | MorpheuBr | i thougt qotecom was one with support, but i cant connect on it |
| 00:07.57 | [TK]D-Fender | counterpath.com |
| 00:08.02 | [TK]D-Fender | there are several. |
| 00:08.08 | WIMPy | Penguin: Maybe you found the "free" option that must not exist. |
| 00:08.32 | WIMPy | But you can alway move to some country where patents don't matter. |
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| 00:16.30 | Penguin | gets out the VoBAN |
| 00:34.00 | *** join/#asterisk mattwj2002 (~Matt@wikisource/pdpc.active.mattwj2002) |
| 00:34.21 | mattwj2002 | hi all :) |
| 00:35.48 | pabelanger | Hello, yes. Dog here |
| 00:36.38 | mattwj2002 | hi pabelanger |
| 00:46.44 | *** join/#asterisk lordvadr (~lordvadr@jose-tc.ctc.biz) |
| 00:47.13 | lordvadr | Is there a dialplan function to get the remote IP of an anonymous SIP invite? I can't seem to get anything out of SIPPEER, which I didn't really expect. |
| 00:48.33 | Penguin | ${CHANNEL(from)} perhaps |
| 00:48.58 | lordvadr | Penguin, I'll give that a try...one sec. |
| 00:49.03 | *** part/#asterisk mattwj2002 (~Matt@wikisource/pdpc.active.mattwj2002) |
| 00:49.05 | lordvadr | Basically, I'm trying to do this... |
| 00:49.06 | lordvadr | http://pastebin.com/eK3fRK8C |
| 00:49.22 | WIMPy | CHANNEL(peerip) |
| 00:50.04 | Penguin | ${CHANNEL(recvip)} |
| 00:50.36 | Penguin | I use this for the same purpose that you're trying to use it. |
| 00:50.48 | WIMPy | That's for RTP, not SIP. |
| 00:51.01 | lordvadr | WIMPy: Jesus h christ, thank you. |
| 00:51.17 | lordvadr | What's the difference between PEERIP and RECVIP? |
| 00:51.31 | WIMPy | Oh. It's been a long time, someone called me Jesus. |
| 00:51.37 | WIMPy | SIP/RTP |
| 00:51.46 | WIMPy | They need not be the same. |
| 00:51.51 | Penguin | <PROTECTED> |
| 00:51.51 | Penguin | <PROTECTED> |
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| 00:52.26 | Penguin | R/O means it is a read-only value. |
| 00:52.33 | lordvadr | Penguin: Yeah, I read that. That doesn't seem to make a lot of sense. Is one the setup ip, and the other the one to which the RTP stream is being/will be sent to? |
| 00:53.20 | WIMPy | sourece=received |
| 00:53.27 | WIMPy | or at least that's what I remember. |
| 00:54.03 | Penguin | I never could tell the difference, so I chose the one that said source. |
| 00:54.24 | Penguin | It has worked for me, so I keep using it. |
| 00:54.54 | Penguin | Log(NOTICE,${CHANNEL(recvip)} is attempting to make unauthorized calls) |
| 00:55.15 | Penguin | Then I use a regex to match that string to create the ban. |
| 00:55.53 | Penguin | I could probably change to peerip and have the exact same results every time, but I probably won't change it. |
| 00:56.44 | WIMPy | Probably |
| 00:57.59 | Penguin | That was my workaround for the lack of the security logging level in 1.8. |
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| 01:12.49 | mattwj2002 | hi guys |
| 01:13.00 | mattwj2002 | speech to text for asterisk voicemails? |
| 01:13.03 | mattwj2002 | is that possible |
| 01:13.03 | WIMPy | hi mattwj2002 |
| 01:13.14 | mattwj2002 | howdy WIMPy |
| 01:13.33 | [TK]D-Fender | mattwj2002: Sure.... if you don't mind the high error rate |
| 01:13.54 | mattwj2002 | that is fine |
| 01:13.55 | mattwj2002 | :) |
| 01:14.04 | mattwj2002 | I am use to it |
| 01:14.05 | mattwj2002 | :D |
| 01:14.17 | WIMPy | Oh, this voicemail to email stuff seems to work quite well. |
| 01:14.54 | mattwj2002 | WIMPy: here is what I was thinking would be awesome |
| 01:15.32 | mattwj2002 | you get a voicemail it e-mails the mp3 of the voicemail and the speech to text result |
| 01:15.34 | WIMPy | The thing is people feel bad about it. It always reminds them that the NSA does the same. |
| 01:16.00 | Penguin | Google voicemail transcription style? |
| 01:16.16 | mattwj2002 | exactly Penguin |
| 01:16.22 | WIMPy | sipgate only mails the text. |
| 01:16.26 | mattwj2002 | WIMPy: the NSA is a watching |
| 01:16.32 | mattwj2002 | ;) |
| 01:16.37 | WIMPy | I know |
| 01:16.58 | [TK]D-Fender | You should be able to issue an automated FOIA request upon receipt of an e-mail saying "Hey bud, mind sharing a copy with me?" :) |
| 01:17.08 | [TK]D-Fender | voicemail* |
| 01:17.34 | WIMPy | Luckily I use openvpn with keys, not certs. |
| 01:18.26 | mattwj2002 | WIMPy: openvpn access server? |
| 01:18.37 | WIMPy | What? |
| 01:18.40 | mattwj2002 | or the other one |
| 01:18.53 | mattwj2002 | access server is the paid option |
| 01:18.54 | mattwj2002 | :) |
| 01:19.03 | WIMPy | Oh, no. |
| 01:19.36 | mattwj2002 | https://openvpn.net/index.php/access-server/overview.html |
| 01:19.37 | mattwj2002 | :) |
| 01:20.37 | WIMPy | Hmm. Doesn't make it obvious what the point is. |
| 01:20.53 | WIMPy | Oh man, was that nice to compile Asterisk. |
| 01:21.32 | WIMPy | Just a few seconds instead of an hour with freeswitch. |
| 01:23.07 | mattwj2002 | WIMPy: I agree freeswitch compiling is slow |
| 01:23.08 | mattwj2002 | :) |
| 01:23.15 | mattwj2002 | try it on a rpi |
| 01:23.16 | mattwj2002 | ;) |
| 01:23.41 | WIMPy | Plus a quarter of an hour for autoconf and configure each. |
| 01:23.53 | WIMPy | How many days does that take? |
| 01:24.01 | mattwj2002 | less than a day |
| 01:24.06 | mattwj2002 | but it was a long compile |
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| 01:24.20 | WIMPy | is impressed with the Pi. |
| 01:24.56 | WIMPy | Ok, I have to admit that my test box is only a dual PIII-1266. So it's not that fast, either. |
| 01:25.00 | mattwj2002 | buys WIMPy a pi |
| 01:25.17 | WIMPy | I've got two :-) |
| 01:25.51 | WIMPy | I was thinking of abusing one as a telephone adapter. |
| 01:26.09 | mattwj2002 | so how go about vm to text in asterisk? |
| 01:26.15 | mattwj2002 | sphinx? |
| 01:26.42 | WIMPy | Google doesn't like longer files, does it? |
| 01:27.00 | mattwj2002 | send it to google? :O |
| 01:27.06 | mattwj2002 | no idea WIMPy |
| 01:27.42 | WIMPy | I think they have a (rather short) length limt. |
| 01:28.27 | mattwj2002 | google or sphinx? |
| 01:29.03 | WIMPy | google |
| 01:29.40 | mattwj2002 | no idea |
| 01:31.50 | mattwj2002 | check this out! |
| 01:31.51 | mattwj2002 | http://zaf.github.io/asterisk-speech-recog/ |
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| 01:32.13 | mattwj2002 | it uses google but for a hobby pbx it should be fine |
| 01:32.18 | mattwj2002 | whips slav3_kitten |
| 01:32.37 | slav3_kitten | why!? |
| 01:32.54 | mattwj2002 | just kidding :) |
| 01:40.49 | mattwj2002 | is working on this |
| 01:40.50 | mattwj2002 | http://zaf.github.io/asterisk-speech-recog/ |
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| 05:43.37 | justdave | if I have a local route for _011. and a route visible to me via a DUNDi switch statement for _01186. will it pick the more specific route or will it ignore DUNDi because there's a local match? |
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| 08:48.08 | jmls1 | morning all |
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| 08:49.01 | jmls1 | do you know if it's possible to make the # key on a polycom 331 act the same as the "dial" button. My users are used to a cisco 7940 , where pressing # dials the entered number |
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| 09:10.15 | bitwize | Hi! I have a question regarding filters on ami events. I have successfully added filters for some events but "NewStateEvent" and "NewExtenEvent" does not seem to be affected by the blacklist. I'm working with version 11.3.0 and the blacklist-filters not working is "!Event: NewState*" and "!Event: NewExten*". |
| 09:10.16 | bitwize | Is there anyone who can point me in the right direction? |
| 09:11.04 | bitwize | I have tried to blacklist them in multiple ways but still receiving the events... |
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| 09:24.16 | bitwize | FYI I just solved the "NewExtenEvent" by restricting read access for "dialplan" in manager configuration |
| 09:25.17 | ayn | hello everyone , need your help to achieve something i setted up asterisk 1.8.xx and everything is working well i have 4 sip trunk,a communication time limit is on each trunk. I want a config that keeps me from overlapsing the time limit on each trunk. If one trunk(trunk1) is being used for a communicatin till the time limit, the call is hanged up and the next call should use trunk2;and after trunk3... How can i achieve that? |
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| 11:32.02 | zamba | i'm not sure if this is asterisk specific or not.. but we have some polycom phones and when we have an active dial tone and then try dialing an international line, the phone goes into busy signal.. i suspect this is due to the ring pattern on the polycom phones, or? |
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| 12:32.29 | zamba | dead channel? |
| 12:33.07 | deeshu | Is there any way of sending the DID of the agent using originate (via AMI ) .. currently the DID is sending first agent DID instead of the calling agent DID |
| 12:33.59 | SpeedEvil | It's not dead. It's just resting. |
| 12:57.29 | WIMPy | zamba: Yes, it's the phones config (usually "dislplan" or such). |
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| 13:33.41 | lampe2 | hey just a quick question. we wont to setup a asterisk server but we have multiple offices which we connect through VPN. will this work with asterisk? or should we better get a asterisk server on every location and connect them ? |
| 13:35.08 | warenbe | hi, quick answer: it will work as long as you network bandwitch is good :) |
| 13:35.42 | WIMPy | You can do it either way. Depends on how reliable your connection is and if you have alternatives in case of failure. |
| 13:35.42 | Chainsaw | lampe2: Running multiple Asterisk servers will certainly help your failover. |
| 13:35.55 | file | options, options, options |
| 13:36.04 | Chainsaw | lampe2: You could at least play an announcement to tell your users that the link is down, or send the call out over backup phone lines, etc. |
| 13:36.28 | warenbe | Chainsaw: or you can use things like Xorcom TS for reall failover without having one pbx on each office |
| 13:37.43 | lampe2 | thx for the info ! the problem is one incoming number is a service number and people should answer the phone in the 3 offices that we have |
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| 13:38.44 | warenbe | well event with 1 pbx on each office, you can send call to phones. you just have to link every pbx with a sip trunk. but yes it's easier to use a vpn and only 1 pbx |
| 13:38.50 | warenbe | *even |
| 13:39.24 | lampe2 | okay thx! |
| 13:39.59 | WIMPy | Why would you use SIP between two Asterisks? |
| 13:40.45 | Chainsaw | IAX is made for it. |
| 13:41.07 | warenbe | he tells that one phone number should ring all phones in differents office. let's say your number is incoming in the PBX 1, and you have phones on pbx 2 and 3, then i think one simple solution is to link every pbx with the 1st one using sip trunk |
| 13:41.11 | warenbe | well |
| 13:41.16 | warenbe | you're right |
| 13:41.20 | warenbe | IAX is fine |
| 13:41.38 | warenbe | i usually use SIP but just because i'm better in SIP than IAX :) |
| 13:42.08 | WIMPy | The goos thing about IAX is that you don't have to be good at itto make it work. |
| 13:42.09 | warenbe | but it does not change the fact that it's possible with 1 or multiple PBX :) |
| 13:42.32 | warenbe | yes, it's just an habit |
| 13:42.33 | Chainsaw | NAT/VPN awkwardness is also less likely to cause one-way audio with IAX. |
| 13:42.37 | WIMPy | Which is waht I said in the beginning as well. |
| 13:43.25 | warenbe | personnaly i would use only 1 pbx and vpn (less support for users, less configuration on the pbx -> less cost) |
| 13:43.40 | WIMPy | And IAX allows you to send variables across. |
| 13:43.47 | warenbe | but it depends on the internet connection... |
| 13:45.19 | lampe2 | in the mainofficce we have like 15 phones in the second 15 to and in the 3th office 3 |
| 13:45.43 | warenbe | so i would use 2 pbx |
| 13:46.02 | warenbe | because if your internet connection is down, then it's not good having 15 phones down... |
| 13:46.41 | lampe2 | we have 2 internet connections between the main and second office so thats not a problem |
| 13:46.49 | warenbe | ho ok :) |
| 13:47.27 | lampe2 | but we are working on local setups |
| 13:47.38 | lampe2 | so that everyoffice will have its own setup |
| 13:48.15 | lampe2 | thats why we are looking for a pbx system ;) |
| 13:48.46 | warenbe | Question: what is the best solution when you have multiple PBX and you want all phones to see state of other pbx's phone? |
| 13:53.39 | *** join/#asterisk serafie (~erin@nat/digium/x-edxgseiinlzafcpm) |
| 13:53.41 | Katty | morning |
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| 14:17.36 | zamba | i'm trying to play an mp3 file using MP3Player().. [Mar 5 15:17:14] NOTICE[26386]: app_mp3.c:133 timed_read: Poll timed out/errored out with 0 |
| 14:17.39 | zamba | what am i doing wrong? |
| 14:17.41 | *** join/#asterisk ajay (aj@124-148-99-135.dyn.iinet.net.au) |
| 14:18.20 | zamba | it's just a simple same => n,MP3Player(/path/to/file/) |
| 14:18.25 | ergodicsum | hello, I'm having phone issues and one warning that keeps coming up is Mar 5 09:17:13 NOTICE[445]: cdr.c:445 ast_cdr_free: CDR on channel 'Local/#40157833@neoagent-0719,2' lacks end |
| 14:18.38 | ergodicsum | what does lacks end mean? |
| 14:19.46 | zamba | got it :D |
| 14:21.18 | *** join/#asterisk jetlag (~jetlag@pool-71-168-193-221.cmdnnj.east.verizon.net) |
| 14:24.55 | ergodicsum | does anyone know what "lacks end" mean? |
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| 14:53.12 | pabelanger | zamba, Gah, convert your mp3 into something native, so you don't have to transcode audio everytime |
| 14:53.54 | zamba | pabelanger: nah, that's ok.. it's never going to be used :) |
| 14:54.01 | zamba | pabelanger: or do you have a command for that? |
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| 14:57.51 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 15:01.55 | [TK]D-Fender | zamba: "core show help file convert" |
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| 15:06.36 | ghost75 | with comedia the remote peer has to send rtp to the port i specify in rtp.conf? |
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| 15:25.53 | newtonr | ghost75, "; nat = comedia ; Send media to the port Asterisk received it from regardless |
| 15:25.54 | newtonr | ; ; of where the SDP says to send it. |
| 15:26.25 | file | the remote side has to send media to Asterisk and Asterisk has to receive it in order to know the source address information |
| 15:26.31 | newtonr | there is also a more detailed description of the behavior a little further down in the sample sip.conf |
| 15:26.54 | ghost75 | what is this all for? |
| 15:27.22 | file | do you mean what is that option for? |
| 15:27.29 | ghost75 | yes |
| 15:27.54 | file | if the other side is behind NAT and the NAT device allows the UDP mapping to be bidirectional it will allow RTP to flow without any configuration on the other side |
| 15:28.31 | ghost75 | and when nat is used local? |
| 15:29.03 | file | if Asterisk is behind NAT then you need to forward ports and configure isp.conf with localnet/externip or externhost |
| 15:29.10 | file | unless the other side is doing NAT traversal stuff |
| 15:29.25 | ghost75 | so comedia not used then |
| 15:29.40 | file | it could be |
| 15:29.58 | file | if you configured Asterisk with localnet/externip you could turn on comedia and if the remote side is behind NAT then RTP would flow |
| 15:30.03 | ghost75 | when i do rtp debug i see that rtp ports are different than in rtp.conf |
| 15:30.22 | file | rtp.conf controls local ports, it doesn't control the ports the other side will send from |
| 15:32.55 | ghost75 | k, then i debug it doesnt show local port |
| 15:33.54 | *** join/#asterisk navaismo (~navaismo@200-52-45-221.dynamic.axtel.net) |
| 15:34.14 | ghost75 | so then comedia ignores header and uses port where packet comes in? |
| 15:34.17 | *** join/#asterisk lovea (~smuxi@212.250.220.116) |
| 15:35.04 | file | it will use the SDP provided address+port until it receives a packet and then it will switch |
| 15:36.43 | ghost75 | will sip debug show the provided port? couldnt find it |
| 15:37.11 | file | for RTP? it's in the SDP and it also spits out a media is at <address> line |
| 15:37.42 | ghost75 | SIP/2.0 200 OK <- is this sdp ? |
| 15:38.14 | file | it is not |
| 15:38.20 | file | said message may contain an SDP |
| 15:39.26 | lovea | CallerID question. Using Asterisk 11.8, Digium TDM411PE Asterisk PCI Card + H/W EC; 1-Port FXO + 1-Port FXS. London based with VirginMedia POTS. Cannot for the life of me get CallerID to display on inbound analogue calls even though enabled with VirginMedia. I've tried all suggestions found in various forums. Run out of ideas. Has anyone any experience of VirginMedia in the UK in this respect? |
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| 16:04.48 | ghost75 | file, this is also confusing me: https://www.ietf.org/mail-archive/web/mmusic/current/msg01326.html |
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| 16:57.56 | SpeedEvil | Random sort-of-related question. In general - if I have multiple phones logged onto one SIP account - will they all ring on calls? |
| 17:00.24 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
| 17:02.04 | [TK]D-Fender | SpeedEvil: "logged" is a vague and bad term. Multiple registrations to the same device will not all ring. Only the last to register wins and will get called. |
| 17:02.21 | SpeedEvil | Thanks. |
| 17:03.56 | ChannelZ | (if you are daring, 12 supports multiple registration.) |
| 17:04.07 | file | chan_pjsip does, not chan_sip |
| 17:04.16 | ChannelZ | yes |
| 17:04.40 | [TK]D-Fender | Good to know.... |
| 17:04.41 | file | when talking about 12 it's best to be specific, or else someone will think chan_sip does |
| 17:04.50 | [TK]D-Fender | needs to get back in the game and try out 12 |
| 17:05.08 | file | FreePBX has a distro with their 12 version and Asterisk 12, allows you to use PJSIP |
| 17:05.28 | ChannelZ | That sounds terrifying |
| 17:05.54 | tm1000 | ChannelZ: stop being a hater. completely unnecessary |
| 17:06.31 | ghost75 | why hater |
| 17:06.46 | ChannelZ | The ones with no sense of humor are fun. |
| 17:07.29 | file | for those who are interested I've got a blog post at http://www.joshua-colp.com/gulp-i-mean-sip-hi-chan_pjsip/ which talks about why we did chan_pjsip and a quick list of what it's capable of |
| 17:08.01 | navaismo | file are you Joshua? |
| 17:08.05 | file | navaismo, yes. |
| 17:08.23 | ChannelZ | I'm glad for it. It's a little scattered to configure (change is difficult!) but overall it's all so much smoother. |
| 17:08.40 | navaismo | w00t! as fle you aosund less hater tahn in twitter, and i love your 'hater-ness' btw |
| 17:08.57 | file | I only hate ... uh... Google and realtime? |
| 17:09.08 | navaismo | hehe |
| 17:09.17 | tm1000 | file: arent you going to kill realtime soon |
| 17:09.27 | file | tm1000, it'll never die |
| 17:09.42 | file | tm1000, but I'm comfortable with where we are |
| 17:09.45 | *** join/#asterisk bkruse (~Adium@74.51.115.113) |
| 17:10.10 | file | ChannelZ, http://www.joshua-colp.com/pjsip-configuration-dont-hate-me/ :D |
| 17:11.19 | ChannelZ | I don't hate it, it's just something new to get used to, not having a single config entity that represents a device as a whole. |
| 17:11.41 | file | I've pondered doing such a thing but I fear a users.conf scenario |
| 17:11.55 | ChannelZ | I'm happy to get rid of the peer type which was always confusing (mostly because it was documented incorrectly all over the place as to what it was really doing) |
| 17:15.04 | file | if you can think of tweaks/documentation changes/etc to make it easier I'm certainly open to helping with making them |
| 17:16.04 | ChannelZ | Is there a roadmap as to how long chan_sip will be maintained going forward? |
| 17:16.21 | [TK]D-Fender | [12:11]fileI've pondered doing such a thing but I fear a users.conf scenario <- DEPRECATE! DEPRECATE! DEPRECATE! </dalek> |
| 17:16.57 | file | I don't think we have an official thing but it'll still be around for a long time to come |
| 17:17.21 | file | [TK]D-Fender, quite |
| 17:17.50 | file | chan_pjsip doesn't have feature parity so there's still some stuff you have to use chan_sip for |
| 17:18.08 | [TK]D-Fender | file: like? |
| 17:19.44 | file | [TK]D-Fender, if you want AOC support then chan_sip, and CCSS (call completion supplementary services) |
| 17:19.48 | file | and other stuff I'm no doubt forgetting |
| 17:19.54 | file | HA! users.conf support :P |
| 17:20.15 | [TK]D-Fender | I think that's referred to as a bug ;) |
| 17:21.26 | file | for a lot of stuff chan_pjsip is viable and the gap is closing |
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| 18:08.16 | file | looks around |
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| 18:18.19 | ChannelZ-Wk | looks asquare |
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| 19:27.19 | qakhan | hi all, what is the problem with my originate code |
| 19:27.20 | qakhan | http://pastebin.com/MYezpWUJ |
| 19:27.53 | navaismo | whats the issue |
| 19:28.47 | *** join/#asterisk ibercom (58196f7d@gateway/web/freenode/ip.88.25.111.125) |
| 19:30.49 | qakhan | when i call this scritp by dialplan there is message |
| 19:30.59 | qakhan | "/data/scripts/bridgecall.php: Failed to execute '/data/scripts/bridgecall.php': Exec format error" |
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| 19:39.49 | navaismo | chmod +x and add the path to the php |
| 19:43.27 | [TK]D-Fender | [14:30]qakhanwhen i call this scritp by dialplan there is message <- you haven't shown us how you're doing it or the actual call attempt.... |
| 19:44.15 | qakhan | ok i send you what i am doing |
| 19:44.39 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 19:46.08 | qakhan | first correct me if i am wrong, i am trying to do bridge call between 2 channels. |
| 19:47.24 | [TK]D-Fender | that is an orignate... not a bridge. They are nothing alike |
| 19:49.17 | qakhan | i asked you about bridge and you said AMI script |
| 19:49.54 | [TK]D-Fender | qakhan: Correct. AMI has a LOT of different command. That is not BRIDGE |
| 19:50.25 | qakhan | ok so i cannot use originate to make bridge call |
| 19:50.29 | qakhan | right? |
| 19:51.33 | [TK]D-Fender | qakhan: I'm not sure I trust your wording VS your intent |
| 19:52.04 | [TK]D-Fender | You don't MAKE a Bridge call. Bridge TAKES 2 calls you already have and connects them |
| 19:52.53 | *** join/#asterisk startledmarmot (~startledm@cpe-76-172-72-246.socal.res.rr.com) |
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| 19:58.42 | qakhan | ok let me describe again what i want to achive |
| 20:01.30 | qakhan | customer will call to asterisk and enter the confirmation number of his order, astersik will check the phone number against the confirmation number and dial the number and connect to customer |
| 20:02.42 | [TK]D-Fender | qakhan: What happens to that first call? When? |
| 20:03.28 | qakhan | then customer can talk to sales person |
| 20:03.40 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 20:03.43 | qakhan | first call will be on hold |
| 20:03.55 | [TK]D-Fender | Why is this not just a DIAL? |
| 20:04.07 | [TK]D-Fender | So far that first person is STAYING on that call in. |
| 20:04.34 | qakhan | as asterisk dial the sales person number it connect both calls |
| 20:06.06 | *** join/#asterisk bkruse (~Adium@74.51.115.113) |
| 20:06.34 | [TK]D-Fender | So far there is 1 customer who calls in. They are then just going through the dialplan and entering IVR's, etc. Then you call OUT to a sales person. that is 2 people talking. That is a DIAL. |
| 20:08.33 | qakhan | man you are Amazing |
| 20:08.43 | qakhan | i got it |
| 20:09.20 | [TK]D-Fender | Not seeing this as a simple call like any other is avery bad sign.... |
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| 20:26.11 | *** join/#asterisk giesen__ (~ggiesen@2001:4900:2:650:20c:29ff:fef5:4836) |
| 20:27.41 | giesen__ | having an issue with asterisk. I have a peer that is using TCP SIP (only peer I have doing so) and it seems RTP is not being sourced from the address in bindaddr |
| 20:27.59 | giesen__ | all my udp-based peers are sourced properly |
| 20:28.18 | giesen__ | running 1.8.20.0 on CentOS 6 |
| 20:28.38 | Chainsaw | You're 6 versions out of date on the 1.8 branch, please update. |
| 20:29.03 | giesen__ | that is the latest in EPEL |
| 20:29.22 | gusto | that at first, and second, why is he doing sip through tcp? |
| 20:29.32 | giesen__ | MS Lync |
| 20:29.34 | giesen__ | requires it |
| 20:29.35 | giesen__ | don't ask |
| 20:29.44 | giesen__ | if I could I'd fire the customer |
| 20:29.47 | gusto | because if he has some problems with udp ... for example a firewall that would filter it, than nothing helps there |
| 20:30.03 | Chainsaw | gusto: I do SIP over TCP because it is more resilient to subtle packet loss. |
| 20:30.13 | gusto | lol |
| 20:30.15 | Chainsaw | gusto: Particularly when doing attended transfers over a congested link. |
| 20:30.18 | giesen__ | the underlying issue is if the lync server call forwards the call |
| 20:30.23 | giesen__ | there is no audio |
| 20:30.37 | giesen__ | because the IP that the UDP is being sourced from is firewalled |
| 20:30.48 | gusto | nated? |
| 20:31.00 | Chainsaw | gusto: Losing the odd audio packet is unfortunate. Having your caller fall through the cracks and be connected to neither the caller nor the callee... that's a dealbreaker. |
| 20:31.04 | giesen__ | no |
| 20:31.49 | gusto | Chainsaw: i understand, however, we are not in 1949 ... packets should not just drop ... no matter if UDP or TCP |
| 20:32.16 | Chainsaw | gusto: I control the telephony infrastructure, not the ethernet. |
| 20:32.30 | Chainsaw | gusto: You fix the bits you control. |
| 20:32.55 | gusto | yes |
| 20:33.18 | gusto | now, i do not see how to resolve that problem with his lync there |
| 20:33.39 | *** join/#asterisk pzn (~pzn@pdpc/supporter/active/pzn) |
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| 20:34.05 | gusto | lync server is forwarding the call, right? |
| 20:34.13 | giesen__ | yes |
| 20:34.15 | giesen__ | no reinvites |
| 20:34.43 | gusto | and asterisk thinks that the call came from the lync server so he tries to negotiate the RTP ports with the lync server instead of the source |
| 20:35.39 | Chainsaw | gusto: What I'm saying is that SIP over TCP is used in the field. It's not as baroque as you're trying to make out. |
| 20:35.52 | giesen__ | a number of astra phones use it as well |
| 20:36.18 | gusto | what now? TCP? |
| 20:36.29 | giesen__ | yes |
| 20:36.39 | gusto | well, i have no problems with that |
| 20:36.50 | gusto | i can even do SSL on my infrastructure ;-) |
| 20:37.04 | gusto | or "TLS" or what is its name now |
| 20:37.31 | gusto | thats the only case i am forced to use TCP |
| 20:37.39 | pzn | I'm with trouble with my ISP provider today. the DNS gets down/up several times. when DNS is down, I lose all internal network extensions, and they are configured with IP addresses, not names. is this expected in asterisk? is this a bug? |
| 20:37.43 | Chainsaw | Yes, TLS certificate support worked for the first time in 12.1.0 |
| 20:37.50 | Chainsaw | Before then you had to patch it. |
| 20:38.14 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 20:38.14 | Chainsaw | pzn: Asterisk 11 and lower will not tolerate DNS delays. Host a local BIND/named instance on the box if you have to. |
| 20:38.45 | gusto | well, i am surprised by this, because my asterisk used to work fine even configured with ip address w/o dns at all |
| 20:39.10 | pzn | Chainsaw, ok. tks. I'll use a local bind server |
| 20:39.12 | Chainsaw | gusto: Any DNS lookup can and will kill it. Could be something as mundane as a STUN server setting. |
| 20:39.18 | gusto | pzn: what kind of network extensions you mean? |
| 20:39.24 | Chainsaw | gusto: Even for a peer that isn't supposed to use it. |
| 20:39.31 | gusto | lol |
| 20:39.56 | pzn | gusto, I have some external network extensions and many internal network extensions. all extensions die on DNS failure |
| 20:40.05 | gusto | i had once an infinite loop on asterisk 1.8 putting my cpu load at 100% while resolving SRV records in a loop |
| 20:40.53 | leifmadsen | yo dawg, I heard you like loops, so we put loops in your loops |
| 20:41.00 | leifmadsen | fruit loops |
| 20:41.00 | Chainsaw | pzn: And point the BIND instance at 8.8.8.8 as a forwarder if you can, should your ISP DNS be substandard. |
| 20:41.03 | leifmadsen | drops the mic |
| 20:41.26 | ghost75 | puts speaker to mic |
| 20:41.53 | [TK]D-Fender | kills the gain |
| 20:42.24 | gusto | it is true what i wrote .. it was really a dns loop because it was an srv record pointing at the same name again |
| 20:43.00 | gusto | something like srv of sip.example.com being sip.example.com A |
| 20:43.03 | *** join/#asterisk Pici (~Pici@ubuntu/member/pdpc.professional.pici) |
| 20:43.05 | *** part/#asterisk Pici (~Pici@ubuntu/member/pdpc.professional.pici) |
| 20:43.27 | pzn | Chainsaw, can I configure asterisk to use a specific dns server (instead of getting from /etc/resolv.conf) ? |
| 20:43.59 | Chainsaw | pzn: I would fix DNS for everyone & everything on that server and adjust resolv.conf |
| 20:44.56 | pzn | Chainsaw, I meant, just for testing... before changing the dns for the whole machine |
| 20:45.01 | hrnt | what is so wrong with SIP over TCP :( |
| 20:45.37 | Chainsaw | hrnt: It upsets gusto's delicate sensibilities. We better not talk of it again. |
| 20:46.10 | Chainsaw | pzn: Not aware of a way to override DNS on a per application basis like that, no. |
| 20:46.43 | pzn | Chainsaw, ok. tks. I'll change it system wide |
| 20:46.57 | giesen__ | gutso the asterisk server and the lync server are the ones that should be negotiating RTP ports |
| 20:47.12 | giesen__ | the problem is the asterisk server is negotiating RTP ports using a different IP |
| 20:47.19 | giesen__ | that it's not supposed to |
| 20:47.25 | giesen__ | and it only happens with this peer |
| 20:47.39 | Chainsaw | pzn: I predict the end of a number of "weird" and/or "intermittent" outages. |
| 20:47.41 | giesen__ | and the only thing really different about this peer is it uses TCP SIP |
| 20:50.00 | *** join/#asterisk brynjar (~brynjar@fubar.no) |
| 20:50.56 | *** join/#asterisk jeffspeff (~jeffspeff@12.49.160.131) |
| 20:51.16 | jeffspeff | is there something like web-meetme but an active project? |
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| 20:55.10 | *** join/#asterisk paule32 (paule32@dslb-188-106-237-043.pools.arcor-ip.net) |
| 20:55.15 | paule32 | hello |
| 20:55.23 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 20:55.48 | paule32 | what password is requierd: http://codepad.org/c08PVhn7 ?? |
| 20:57.18 | jeffspeff | paule32, the password you set for that sip user |
| 20:58.09 | paule32 | oh shit, i dump |
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| 21:06.49 | *** join/#asterisk bkruse (~Adium@74.51.115.113) |
| 21:20.02 | *** join/#asterisk Asterisk_User (d0450a06@gateway/web/freenode/ip.208.69.10.6) |
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| 21:27.19 | paule32 | <PROTECTED> |
| 21:27.19 | paule32 | <PROTECTED> |
| 21:27.19 | paule32 | <PROTECTED> |
| 21:27.19 | paule32 | <PROTECTED> |
| 21:27.19 | paule32 | [Mar 5 21:58:06] WARNING[10220]: chan_sip.c:28964 reload_config: Section 'genernal' lacks type |
| 21:27.19 | paule32 | <PROTECTED> |
| 21:27.32 | newtonr | ~pastebin |
| 21:27.32 | 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. |
| 21:28.09 | [TK]D-Fender | [16:27]paule32[Mar 5 21:58:06] WARNING[10220]: chan_sip.c:28964 reload_config: Section 'genernal' lacks type <--- you probably want to check your SPELLING |
| 21:28.21 | [TK]D-Fender | heads home for the day..... |
| 21:28.29 | hardwire | NOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOO |
| 21:28.36 | leifmadsen | \o/ |
| 21:28.49 | hardwire | I haven't talked to TK in years! AND NOW THEY ARE GONE! |
| 21:29.04 | paule32 | that is the log output of asterisk |
| 21:29.06 | newtonr | they? |
| 21:29.21 | hardwire | newtonr: they as in he or she or it. |
| 21:29.39 | hardwire | newtonr: thems? is that better? |
| 21:29.55 | newtonr | paule32, TK was saying that you have a section "genernal", which should be "general" |
| 21:30.14 | Asterisk_User | Hello. In FreePBX I hit "Go Live" twice in a row. I heard that if you do that - some config files will be overwritten incompletely and cause issues with PBX running Asterisk. My question is why does it take so much time for Go Live changes to apply when my PBX is using less than 1% CPU, has plenty of disk space and enough memory? |
| 21:30.32 | hardwire | that sounds lovely. |
| 21:30.43 | newtonr | ~freepbx |
| 21:30.44 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 21:30.45 | paule32 | newtonr: oh thanks |
| 21:31.05 | Asterisk_User | okay |
| 21:34.53 | paule32 | [Mar 5 22:05:48] WARNING[10220]: chan_sip.c:20720 handle_response_register: Forbidden - wrong password on authentication for REGISTER for '36916580458' to 'sip.1und1.de' |
| 21:35.42 | paule32 | is that the password für sip.conf 2691... or ? |
| 21:35.43 | Asterisk_User | My bad. I'm using Asterisk web ui, not FreePBX. |
| 21:39.01 | tm1000 | Asterisk_User: the Asterisk Web UI is extremely outdated and just sits in source to look pretty. it doesnt work well if it even does work |
| 21:39.30 | Asterisk_User | tmp1000, are you saying that the Web UI is to blame for? |
| 21:40.17 | tm1000 | Asterisk_User: Im saying no one can answer what the problem is since no one works on Asterisk UI |
| 21:40.32 | tm1000 | and if anyone attempts to give you advice it would probably be guessing... |
| 21:45.06 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 21:52.18 | paule32 | is that message correct: http://codepad.org/Ve9lU7Yu |
| 21:52.19 | paule32 | ? |
| 21:52.49 | navaismo | yeah wyhy |
| 21:52.58 | navaismo | s/wyhy/why |
| 21:53.31 | paule32 | must there not stand "registered" ? |
| 21:58.16 | *** join/#asterisk Milarepa (~Milarepa@wsip-24-234-72-245.lv.lv.cox.net) |
| 21:58.58 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:08.17 | navaismo | only if it is registered |
| 22:12.09 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
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| 22:23.00 | paule32 | [Mar 5 22:54:28] WARNING[10220]: chan_sip.c:28528 reload_config: Invalid address for externaddr keyword: test.domain |
| 22:23.14 | paule32 | why get this warning? |
| 22:23.19 | navaismo | wrong addr |
| 22:23.28 | navaismo | use exterhost for domains |
| 22:25.07 | [TK]D-Fender | externaddr should be valid |
| 22:33.50 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 23:20.00 | Penguin | invalid address? It doesn't like a host name with only one dot? |
| 23:22.34 | *** join/#asterisk rrittgarn (~rrittgarn@75-150-221-205-Illinois.hfc.comcastbusiness.net) |
| 23:25.41 | rrittgarn | Is there an easy way to take an established call and turn it into a conference call with AMI. When I redirect one side of the call the other would get dropped without a 'g' on the dial (i think). Goal is to have two bridged channels, dropped into a conf bridge so we can add more channels/callers. |
| 23:27.49 | [TK]D-Fender | AMI redirect them to the conference. |
| 23:31.24 | *** join/#asterisk theron (~theron@69.63.185.56) |
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| 23:40.37 | *** part/#asterisk mjordan (~matt@nat/digium/x-bywizkfklqkbdnsr) |
| 23:47.26 | rrittgarn | Fender, can you redirect both sides of the call at the same time? |
| 23:47.39 | rrittgarn | even with two redirects back to back won't the second get hung up on? |
| 23:48.08 | [TK]D-Fender | Have you rad the instructions? |
| 23:48.12 | [TK]D-Fender | read* |
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| 23:49.43 | rrittgarn | guess that helps... sorry |
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