| 00:02.10 | *** join/#asterisk threesome (~threesome@ip-94-113-13-143.net.upcbroadband.cz) |
| 00:13.17 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 00:42.52 | *** join/#asterisk qakhan (~qakhan@pool-71-120-31-72.washdc.fios.verizon.net) |
| 00:43.16 | qakhan | how we can compare 2 dates in dialplan? |
| 00:47.07 | *** join/#asterisk jasonwert (~w3rt@71-89-135-86.static.aldl.mi.charter.com) |
| 00:49.53 | navaismo | gotoif |
| 00:54.05 | qakhan | navaismo i am getting date from user and comparing with system date |
| 00:54.07 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 00:54.14 | qakhan | but it does not work |
| 00:55.59 | qakhan | http://pastebin.com/s3hph1Xq |
| 00:56.15 | qakhan | it takes caller entered date as number |
| 01:01.08 | navaismo | date <??? |
| 01:01.51 | qakhan | date is veriable where caller entering date |
| 01:02.00 | qakhan | ${date} |
| 01:02.18 | navaismo | yes but are trying to compare with > not sure if the best choice |
| 01:02.34 | navaismo | imo I prefer to do all in an AGI script |
| 01:02.56 | qakhan | i tried to do with < , =, > |
| 01:03.07 | qakhan | i used all but same result |
| 01:04.24 | navaismo | so you are handling the date as a number right? like 01012013 < 09022013 |
| 01:04.49 | *** join/#asterisk Pullphinger (~Pullphing@c-24-13-69-42.hsd1.il.comcast.net) |
| 01:06.01 | qakhan | lets say today is 10022013 but user enter 01022014 (which is greater than) |
| 01:06.50 | navaismo | my advise is to check that individually at realtime i.e: you ask the year and inmediatly check if match this year or the next but if is the past year ask again, and so on for the day & month |
| 01:07.00 | navaismo | you cant compare dates like that |
| 01:07.43 | qakhan | is there any function which convert in date |
| 01:12.04 | navaismo | jesus guys why did you change the pipe | for the comma , |
| 01:12.13 | navaismo | just checked one post in digium forum hehe |
| 01:14.01 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 01:24.44 | *** join/#asterisk jasonwert (~w3rt@71-89-135-86.static.aldl.mi.charter.com) |
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| 01:32.40 | *** part/#asterisk a1chemist (~a1chemist@178.77.16.39) |
| 01:38.32 | phix | navaismo: Because it makes more sense to C programmers? :) |
| 01:43.24 | navaismo | sorry phix i forgot the </sarcasm off> |
| 01:49.07 | phix | ah yeah |
| 01:49.24 | phix | (no spaces in xml closing tags!) |
| 01:52.32 | *** join/#asterisk tristero (~al.f.zero@unaffiliated/transfinite) |
| 01:54.56 | navaismo | got to go |
| 02:06.00 | jeev | when debugging iax2, i have registration between with office A and B, they're connected in iax2 show peers but if i debug it, it says registration refused.. what is this ? |
| 02:22.17 | *** join/#asterisk jpoz (~jpoz@c-24-22-121-42.hsd1.or.comcast.net) |
| 02:23.48 | *** join/#asterisk jpoz_ (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
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| 03:36.02 | *** join/#asterisk gartral (~gartral@unaffiliated/gartral) |
| 03:37.07 | gartral | hey all, I've installed asterisk-gui and got it up but not "running" persay.. it's stuck on "Checking write permission for gui folder" and it's been like this for 15-20 minutes |
| 03:37.33 | ChannelZ | asteriskgui is deaaaaad |
| 03:37.59 | gartral | ChannelZ: well what panel would you recommend instead? |
| 03:38.43 | ChannelZ | None. None panel. |
| 03:39.14 | gartral | ChannelZ: I sense the Troll is strong with this one.. |
| 03:39.19 | ChannelZ | but if you feel you must FreePBX is at least actively worked on |
| 03:39.45 | ChannelZ | You asked, I answered. |
| 03:52.17 | Penguin | Just because you disagree with the response a person gives does not make the person a troll. |
| 04:26.52 | *** join/#asterisk dgeary2 (~debian@d110-33-193-44.mas801.nsw.optusnet.com.au) |
| 04:39.21 | *** join/#asterisk Defraz (~Defraz@209.141.122.71) |
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| 05:05.40 | *** join/#asterisk loggiew (~logan@65.35.91.48) |
| 05:06.39 | loggiew | i have an asterisk build on my router (no NAT) and an external and internal client both connect but when I begin a call I get no audio between them at all. Any thoughts on where to begin? |
| 05:17.37 | ChannelZ | Is the 'external' client behind NAT? |
| 05:18.04 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-tiqcgasmrtovfokc) |
| 05:18.38 | ChannelZ | and while your router might not being doing NAT, does it block traffic in any other way? |
| 05:21.37 | ChannelZ | (like a 'default DENY' unless otherwise told) |
| 05:21.58 | *** join/#asterisk kemmler (tully@r74-192-178-52.htvlcmta01.hnvitx.tl.dh.suddenlink.net) |
| 05:23.29 | kemmler | I have this exten => 66,n,Dial(SIP/${peerip}/${numcalled},10) in my dialplan, do I catch the timeout with a exten => t,1,....? |
| 05:23.36 | loggiew | no... no NAT involved and as far as I can tell I got the rules correct, opening 10,000-20,000 for RTP was added the same time as the other rules for 5060 and things connect in fine |
| 05:24.53 | loggiew | its an ipfire router I upgraded and put the newest asterisk on, then I edited the local rules manually and added rules for 5060 and the 10k range. since I did that external clients connect fine so I assume the rules were added appropriately before any drops |
| 05:25.10 | loggiew | and when I run rtp debug I do see traffic passing, if thats what that means |
| 05:26.18 | loggiew | I dont believe this to be a problem with traffic getting to it as those ports all appear to work and it's not NAT'ed |
| 05:28.22 | ChannelZ | it's sending and receiving rtp, or just sending? |
| 05:28.31 | loggiew | appears to go both ways |
| 05:28.40 | loggiew | and the lack of audio is on both ends, not one way |
| 05:34.20 | ChannelZ | Does the local-side phone work on an echo test to asterisk? |
| 05:34.54 | loggiew | ah.. a place to start. looking up how to do an echo test |
| 05:36.57 | ChannelZ | make an extension that does a Playback() of something and then an Echo() |
| 05:38.20 | loggiew | does it have to be a particular format like wav? where can I get a demo sound |
| 05:38.44 | ChannelZ | you should have some that came with it. Playback(echo-test) |
| 05:39.27 | loggiew | ok, no sound |
| 05:39.45 | loggiew | again it appeared to pass traffic with the rtp debug |
| 05:39.51 | loggiew | but I hear nothing |
| 05:39.54 | ChannelZ | What is the device? |
| 05:40.08 | loggiew | it's my cell phone with a sip client installed |
| 05:40.09 | ChannelZ | A softphone, hardphone |
| 05:40.21 | loggiew | android |
| 05:40.31 | ChannelZ | The built-in one or something else |
| 05:40.40 | loggiew | the program is called zoip |
| 05:40.47 | loggiew | didnt realize there is a built in one, let me look |
| 05:40.55 | ChannelZ | it kind of sucks |
| 05:40.57 | ChannelZ | don't bother |
| 05:41.05 | loggiew | cool |
| 05:41.50 | ChannelZ | In any event seems like you have a problem with Zoiper |
| 05:41.59 | ChannelZ | FWIW I use CSipSimple on android and it works |
| 05:42.04 | loggiew | hm |
| 05:42.51 | kemmler | Does anyone know how to detect a timeout set in a Dial function like this Dial(SIP/${peerip}/${numcalled},10)? |
| 05:43.24 | ChannelZ | well generally the channel status will be NOANSWER |
| 05:43.43 | ChannelZ | ${DIALSTATUS} |
| 05:45.10 | kemmler | ok, so the t,1,.... section of the dialplan is for something else |
| 05:45.26 | kemmler | i know T,1,... is for absolute timeout |
| 05:45.34 | kemmler | but i don't want that |
| 05:45.56 | ChannelZ | t is for extension timeouts |
| 05:46.18 | ChannelZ | Like if you're WaitExten()ing |
| 05:46.27 | kemmler | ahh ok gotcha thanks |
| 05:47.14 | kemmler | so a Goto(s-${DIALSTATUS} and then a s-NOANSWER,1,.. will suffice? |
| 05:49.10 | [TK]D-Fender | just jump to a label withing the current extension rather than invent hybrid patterns like that |
| 05:51.02 | kemmler | like 66,n(NOANSWER),...? |
| 05:51.52 | [TK]D-Fender | if you're in 66 |
| 05:52.26 | kemmler | kk that clears it up. thanks guys |
| 06:00.41 | jeev | fender, iax2 between the two locations is VERY unstable. i tried it off all wans, it goes from 25 ms to 1027ms for example, it drops.. it however is 100% stable over it's openvpn trunk. any ideas? |
| 06:07.41 | [TK]D-Fender | nope |
| 06:16.33 | RudyValencia | Hi, i'm trying to set up Asterisk realtime configuration for use with Kamailio but whenever I try to register Asterisk says the password is incorrect. |
| 06:30.29 | kemmler | RudyValencia, dunno anything about kamailio. do you know how it connects to asterisk? |
| 06:30.49 | RudyValencia | well, this even happens when I directly connect to asterisk |
| 06:31.11 | kemmler | from a sip client? |
| 06:31.22 | RudyValencia | yes |
| 06:31.25 | kemmler | which client |
| 06:31.30 | RudyValencia | any client |
| 06:31.40 | kemmler | and you've set up the client in your sip.conf? |
| 06:32.07 | RudyValencia | when you say client do you mean having an entry in sip.conf for each user in the database? |
| 06:32.33 | RudyValencia | I'm using a mysql database for the users |
| 06:33.21 | kemmler | asterisk looks in sip.conf for sip clients |
| 06:34.42 | RudyValencia | there's also the ability to use mysql |
| 06:35.35 | RudyValencia | since 1.2 |
| 06:35.47 | RudyValencia | I'm using 11.5.1 |
| 06:37.19 | kemmler | what error are you getting when trying to connect |
| 06:38.42 | RudyValencia | NOTICE[967]: chan_sip.c:27919 handle_request_register: Registration from '"Name" <[user]@[domain>' failed for '<ip>:<port>' - Wrong password |
| 06:39.52 | kemmler | have you tried to connect using a config in sip.conf to make sure it's not your mysql setup? |
| 06:40.12 | RudyValencia | I'll try that |
| 06:40.14 | RudyValencia | one sec |
| 06:40.16 | kemmler | kk |
| 06:43.16 | RudyValencia | works as a user in sip.conf |
| 06:43.25 | RudyValencia | but it's not supposed to be that way |
| 06:44.12 | RudyValencia | oh |
| 06:44.13 | RudyValencia | got it |
| 06:44.20 | RudyValencia | sippeers vs. sipusers |
| 06:44.47 | kemmler | sip show users shows your mysql users? |
| 06:45.20 | wdoekes | sip show users won't show (uncached) realtime users |
| 06:45.27 | wdoekes | sip show user XYZ load |
| 06:45.34 | wdoekes | sip show peer XYZ load |
| 06:45.46 | *** join/#asterisk jhlavacek (~jirka@78.208.220.3) |
| 06:46.02 | RudyValencia | i have caching enabled |
| 06:46.33 | *** join/#asterisk CeBe (~CeBe@port-92-206-29-235.dynamic.qsc.de) |
| 06:47.16 | RudyValencia | is it possible to have proper usernames with asterisk?> |
| 06:47.40 | RudyValencia | instead of 3000 can I have rudy.valencia@example.com for instance? |
| 06:48.54 | kemmler | i know you can use letters, don't know if there are any character restrictions though |
| 06:49.45 | kemmler | http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf |
| 06:51.52 | RudyValencia | can I have a numeric extension and a username? |
| 06:53.19 | kemmler | callerid="Tuomas Tammisalo" <1000> |
| 06:53.39 | RudyValencia | ah |
| 06:53.50 | RudyValencia | now I just need to figure out how to tie flowroute into this |
| 06:54.03 | kemmler | what are you trying to do |
| 06:54.49 | RudyValencia | I'd like to be able to have options for caller ID |
| 06:55.16 | RudyValencia | for example, by default we show our toll-free number, but you can choose to hide your number or show your real phone number |
| 06:55.26 | RudyValencia | (using *67/*82 as prefixes) |
| 06:55.34 | kemmler | oh easy, with flowroute you can just set the From: header |
| 06:56.21 | RudyValencia | Right |
| 06:56.30 | kemmler | exten => 66,1,Set(CALLERID(num)=2095551214) |
| 06:56.40 | kemmler | that'll set your outbound cid just fine with flowroute |
| 06:56.55 | RudyValencia | do I just make *67NXXXXXXXXX and *82NXXXXXXXXX outbounds? |
| 06:57.34 | kemmler | *671NXXNXXXXXX |
| 06:57.45 | kemmler | with flowroute since you need your country code |
| 06:57.53 | kemmler | or set it up to add that itself |
| 06:57.59 | RudyValencia | I also could set it up to add the 1 automatically on 9-digit calls |
| 07:03.24 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 07:07.51 | *** join/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net) |
| 07:07.57 | *** part/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net) |
| 07:08.01 | gnu_d | Hi, do I need to configure TLS for SRTP ? |
| 07:09.00 | peektoseen | Hi everyone, I have a trouble with asterisk. Trunks goes ureachable and reachable again. In "top" I see load average '2.15'. Is high load average? |
| 07:09.41 | ChannelZ | depends, how many CPUs/cores? |
| 07:12.37 | peektoseen | "nproc" show me 4 units |
| 07:13.09 | *** join/#asterisk linocisco (~linocisco@193.134.242.12) |
| 07:13.25 | linocisco | hi all |
| 07:13.29 | ChannelZ | So a loadav of 2 would be 50% |
| 07:13.39 | peektoseen | 2 core, 2 threads |
| 07:13.41 | ChannelZ | High maybe, what else is the server doing |
| 07:14.00 | ChannelZ | (did you use top or anything to see what was going on) |
| 07:15.31 | peektoseen | nothing else, only asterisk and system process |
| 07:15.54 | peektoseen | I see two "D" state process in "ps": |
| 07:15.55 | peektoseen | root 280 0.0 0.0 0 0 ? D Oct02 0:04 \_ [md0_raid1] |
| 07:15.56 | peektoseen | root 322 0.0 0.0 0 0 ? D Oct02 0:01 \_ [flush-9:0] |
| 07:17.13 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 07:17.21 | ChannelZ | is your raid verifying its self or rebuilding or something |
| 07:17.25 | ChannelZ | cat /proc/mdstat |
| 07:21.01 | peektoseen | Personalities : [linear] [multipath] [raid0] [raid1] [raid6] [raid5] [raid4] [raid10] |
| 07:21.01 | peektoseen | md0 : active raid1 sda1[0] sdb1[1] |
| 07:21.01 | peektoseen | <PROTECTED> |
| 07:21.01 | peektoseen | unused devices: <none> |
| 07:22.40 | ChannelZ | hmm. Well you must have caught it busy doing something. no matter |
| 07:23.41 | ChannelZ | Is there a crapload of network traffic on the network(s) it's sitting on? Or do you process tons of simultaneous calls? Lots of transcoding going on? |
| 07:23.48 | loggiew | im not so familiar with voip yet but when you say trunks go reachable/unreachable is it a network load issue or server load issue? |
| 07:26.01 | RudyValencia | ok whatever happened to autofallthrough in extensions.conf? I set it to no and asterisk acts as if it's not there |
| 07:26.09 | RudyValencia | it says "Unknown directive: autofallthrough at line 2 of extensions.conf -- IGNORING!!!" |
| 07:27.23 | ChannelZ | in [general] ? |
| 07:27.29 | RudyValencia | yes |
| 07:27.49 | RudyValencia | oh, oops, I had it as default |
| 07:27.54 | RudyValencia | derp :P |
| 07:28.32 | ChannelZ | heh |
| 07:28.56 | ChannelZ | I was squinting at my font making sure you spelled 'through' right |
| 07:29.17 | RudyValencia | now it doesn't end unless I want it to |
| 07:30.09 | ChannelZ | exten => t,1,Playback(bitch-i-dont-have-time-for-this) |
| 07:31.01 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 07:31.53 | ChannelZ | and then Hangup() |
| 07:34.00 | *** join/#asterisk caveat- (hoax@shell.bshellz.net) |
| 07:34.28 | RudyValencia | ah cool |
| 07:34.35 | RudyValencia | indeed |
| 07:35.48 | RudyValencia | I'm getting better at writing extensions.conf :) |
| 07:46.21 | *** join/#asterisk threesome (~threesome@customer-79-127-150-148.net.angelnet.cz) |
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| 08:25.56 | peektoseen | I suppose I found problem: |
| 08:25.57 | peektoseen | http://img854.imageshack.us/img854/3528/1zvn.png |
| 08:27.04 | peektoseen | In iotop every ~5 second shown jbd2 with 99.9% IO |
| 08:27.41 | RudyValencia | I'm trying to set up asterisk with realtime database-backed configuration but I can't get flowroute to register |
| 08:31.28 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 08:33.14 | RudyValencia | nevermind |
| 08:33.21 | RudyValencia | forgot the register => line |
| 08:37.54 | *** join/#asterisk teeteewhy (~teeteewhy@no.ra.pe) |
| 08:41.24 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
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| 08:53.09 | gartral | ok, i'm stuck.. I can't get any sip client to actually connect to my asterisk server.. I've tryed everything I can think of and nada... the weird thing, all sip clients SAY they've registered correctly, yet I can't even call my mailbox! |
| 08:53.27 | gartral | and the log shows that no connections are being made |
| 08:55.06 | Neoti | gartral: whats teh output of asterisk -rx "sip show peers" |
| 08:55.22 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 08:56.12 | gartral | Neoti: http://pastebin.com/3UW68rHx |
| 08:56.45 | Neoti | gartral: what ext are you registering |
| 08:57.07 | gartral | Neoti: right now, 6000 and 6007 at once, on two seperate clients |
| 08:57.22 | gartral | I don |
| 08:57.25 | gartral | oops |
| 08:58.02 | Neoti | well there not registered... do you have a firewall or something on the machine.... also whats the output of *sip.conf and *extensions.conf |
| 08:58.33 | Neoti | whats teh out put in asterisk when you try and dial something .... |
| 08:59.19 | gartral | Neoti: Nothing, there's absolutly *NO* indication that the clients are interacting with asterisk, period. |
| 09:00.19 | gartral | asterisk says clients arn't connected, clients say they ARE connected, but there is no way it seems for me to dertermine to *WHAT* they're connecting too.. |
| 09:01.02 | gartral | Neoti: also No such command '*sip.conf' (type 'core show help *sip.conf' for other possible commands) |
| 09:01.08 | Chainsaw | gartral: What you need is a packet capture. I would recommend wireshark. |
| 09:01.09 | Neoti | have you configured the client to connect to the IP and Port that you have specified in asterisk.conf etc |
| 09:01.38 | kaldemar | bind addresses and ports are not configured in asterisk.conf. |
| 09:01.43 | Neoti | Chainsaw: gartral: yes do do a wireshark trace too and see if there is network traffic etc |
| 09:02.01 | Chainsaw | Neoti: Well it's hard to deny that there is traffic. The question is just where the mismatch is. |
| 09:02.55 | Neoti | kaldemar: yeah sorrty i meant sip.conf .... |
| 09:02.57 | gartral | Neoti: there IS traffic, to the server at port 5060, and traffic comming back, but asterisk is just.. saying i'm a liar |
| 09:04.13 | gartral | and I'm testing after stopping my firewall daemons, just to make damn well sure |
| 09:04.40 | gartral | (obviously I've closed my public-facing firewalls off for security here) |
| 09:04.43 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 09:05.49 | kemmler | i'm finally done. asterisk can tongue punch my fart box for a few weeks |
| 09:07.18 | gartral | kemmler: I'm about right there with you |
| 09:07.41 | *** join/#asterisk kresp0 (~kresp0@1.Red-83-50-237.dynamicIP.rima-tde.net) |
| 09:08.11 | kemmler | crawls into bed and pretends to be a bear |
| 09:08.18 | gartral | there's just no connection reflected from within asterisk |
| 09:08.55 | kaldemar | gartral: if you don't see anything in sip debug, then asterisk is not getting any messages. |
| 09:09.52 | gartral | kaldemar: I've started asterisk -vvvvvr and been watching it.. |
| 09:09.54 | Neoti | gartral: cat /etc/asterisk/extensions.conf && cat /etc/asterisk/sip.conf and past bin it ? also be good to wireshark the traffic with something like tshark -p -ta -R "sip" OR tethereal -p -ta -R 'sip' |
| 09:09.54 | kaldemar | check the bind address and port in sip.conf and see that it actually listens on it with netstat -nlp for instance. |
| 09:10.56 | kaldemar | gartral: if it listens on the port, use wireshark, tshark, tcpdump or some other tool of your preference to verify that the server is getting traffic from the clients. |
| 09:10.58 | gartral | kaldemar: bindaddr = 0.0.0.0 |
| 09:12.43 | gartral | kaldemar: I have, and it is. but if asterisk isn't listening and responding, wtf *IS* and why is it responding the Asterisk's configs, if I change off too another extension, the register fails. but asterisk says nothing about a connection |
| 09:13.01 | *** join/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net) |
| 09:13.19 | Blashyrkh | what are the basic configuration files for asterisk? |
| 09:13.22 | kaldemar | gartral: netstat... |
| 09:13.26 | gartral | bindaddr = 0.0.0.0 |
| 09:13.32 | gartral | bindport 5060 |
| 09:13.37 | kaldemar | Blashyrkh: that depends on what you configure asterisk to do. |
| 09:13.38 | Blashyrkh | i just installed asterisk and created a sip.conf and a extension.conf |
| 09:13.59 | kemmler | so did asterisk skip from 1.8 to 11? |
| 09:14.01 | Blashyrkh | but obviously sip module is not loaded automatically, i can do it manually though |
| 09:14.10 | kaldemar | kemmler: no. to 10. |
| 09:14.18 | kaldemar | ~asterisk10 |
| 09:14.18 | infobot | Asterisk 10 -- http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/, or a Standard Release. It was released on 2011-12-15, with maintenance until 2012-12-15. Asterisk 10 will be end of life on 2013-12-15. |
| 09:14.27 | kemmler | that's cool |
| 09:14.30 | kemmler | thanks |
| 09:14.57 | kemmler | my boss is still on 1.4 |
| 09:14.58 | kaldemar | Blashyrkh: modules.conf would help with that. |
| 09:15.52 | Blashyrkh | yes, i copied the sample module configuration into my asterisk folder but there is nothing in there |
| 09:16.59 | Blashyrkh | mhm i have installed asterisk from the debian repositories, i guess thats the difference, up until now i always compiled it myself |
| 09:19.57 | kemmler | is there any cool new features between 1.6 and 11 |
| 09:20.52 | kaldemar | kemmler: loads. see CHANGES and UPGRADE*.txt in a v11 source package. |
| 09:21.25 | *** join/#asterisk janelleb (~janelle@unaffiliated/janelleb) |
| 09:21.33 | kemmler | cool, im going to test it out tomorrow. I hope there's some major improvements to the AMI. |
| 09:21.39 | gartral | kaldemar: ok, so if I were too dive deeper into this with netstat, what would I grep for? |
| 09:21.57 | jploh | asterisk |
| 09:22.01 | kaldemar | gartral: netstat -nlp | grep 5060 |
| 09:22.17 | jploh | oops, spoke too soon. follow kaldemar |
| 09:22.28 | janelleb | hey all: netstat anyone had success sending multiline SMSs with chan_dongle? |
| 09:23.28 | gartral | oh mother.. f^$%& kamailio didn't stop when I told it too! AARRRGH |
| 09:24.34 | RudyValencia | I'm trying to set up voicemail using res_config_mysql |
| 09:24.54 | RudyValencia | and whenever I dial the voicemail it hangs up and gives me three errors |
| 09:25.20 | gartral | kaldemar: now... I feel really stupid |
| 09:25.43 | RudyValencia | http://dpaste.com/1404462/ |
| 09:25.49 | RudyValencia | what is happening? |
| 09:26.10 | kaldemar | gartral: that happens. |
| 09:31.02 | gartral | kaldemar: YAY! -- Registered SIP '6000' at 192.168.1.8:5060 |
| 09:34.47 | gartral | note to self, never call your own extension... |
| 09:36.36 | kemmler | if you do that through at&t you get thrown right into your voicemail if you don't have a passcode forced. which means you can just waltz into someone else's voicemail |
| 09:38.50 | *** join/#asterisk PLMg (PLMg@78.96.151.225) |
| 09:40.37 | PLMg | hey, can anyone help me understand about 10 lines of code? |
| 09:43.09 | Chainsaw | PLMg: Let's have it linked on pastebin. |
| 09:44.31 | RudyValencia | Is it possible to have voicemail use MySQL for the user list and/or voicemail storage? |
| 09:44.39 | PLMg | http://pastebin.com/x3eSRR8d |
| 09:45.20 | PLMg | so, sip extension 10 is fax, the call in the pastebin is a fax call. What happens to it? |
| 09:45.35 | Chainsaw | PLMg: It is detected as a fax, that's for sure. |
| 09:45.48 | PLMg | yep Channel 3 detected a CED tone towards the network. |
| 09:45.59 | PLMg | but after that I do not really understand |
| 09:46.11 | RudyValencia | I'm trying to get voicemail to work with res_config_mysql and I am getting the following whenever I call voicemail and enter my information: http://dpaste.com/1404462/ |
| 09:46.21 | Chainsaw | PLMg: SIP/010 hung up the call, for what I can discern. |
| 09:46.31 | Chainsaw | PLMg: You may need debug 10 & verbose 10 to see just that bit more. |
| 09:46.46 | PLMg | good point! ty Chainsaw |
| 09:47.34 | RudyValencia | Any suggestions as to why I'm getting those errors in the asterisk console? |
| 09:48.18 | Blashyrkh | if i see ok as a status for a peers, dows taht mean it is succesfully registerd? |
| 09:55.22 | gartral | I think I just broke my router's poor little brain >.< |
| 09:56.52 | RudyValencia | wow |
| 09:57.51 | gartral | tried to open about 100 test calls by batch adding a lot of extensions to a page group and paging them |
| 09:58.22 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 09:59.51 | RudyValencia | any idea why I can't get mysql-based voicemail to work? |
| 10:01.01 | Chainsaw | gartral: Overflowed the NAT table, forcing it to disconnect sessions. Fun :) |
| 10:09.30 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 10:14.45 | gartral | Chainsaw: netgear WNR2000v3 "home/small buissiness" router, running dd-wrt.. little thing handles a lot for what it is, but it does have it's limits |
| 10:20.47 | gartral | how do I check rather or not the asterisk I'm using has a specific module/driver installed? |
| 10:21.43 | gartral | or was compiled with said driver/module |
| 10:23.18 | Chainsaw | gartral: module show |
| 10:24.30 | gartral | doesn't look like I have motif or xmpp >.< |
| 10:25.48 | Chainsaw | gartral: module load res_xmpp.so |
| 10:26.04 | Chainsaw | gartral: This will force it to admit whether the module exists, and if so, why it cannot be loaded. |
| 10:26.15 | *** join/#asterisk lnb (~lnb@CPE0002b3c8018e-CM602ad06bec2f.cpe.net.cable.rogers.com) |
| 10:27.06 | gartral | Chainsaw: "no such file or directory" |
| 10:27.13 | gartral | not compiles |
| 10:27.18 | gartral | compiled* |
| 10:27.38 | Chainsaw | gartral: Indeed. You have your work cut out for you :) |
| 10:28.38 | gartral | so, next question.. is there any remote chance I can just compile these missing modules and inject them, or am I going to have to rip out my asterisk install and replace it with a self-compiled build? |
| 10:31.45 | Chainsaw | gartral: Depends on where the main Asterisk came from. |
| 10:32.03 | Chainsaw | gartral: If it is a distro package, then you need the distro edition of those modules too. Separate packages for you to install presumably? |
| 10:32.23 | gartral | ducks behind a bunker, waving the Ubuntu Server flag |
| 10:32.38 | Chainsaw | Then presumably you can apt-get them. |
| 10:33.01 | Chainsaw | Don't worry, it's just a semi-adulterated Debian to me. |
| 10:33.44 | gartral | whew, some people go completly nuclear whenever they hear *buntu* |
| 10:35.22 | gartral | sadly, however, it seems that I'm doomed, as there isn't a readily compiled version of the modules I require in repos |
| 10:35.47 | gartral | considers using a PBX bridge for this.. |
| 10:35.51 | Chainsaw | Not some "extra" package either? |
| 10:36.00 | Chainsaw | Can you get me "apt-cache search asterisk" output on pastebin? |
| 10:36.34 | Chainsaw | gartral: Anyone who doesn't use Gentoo uses something foreign to me. Whether it's RedHat Expensive Linux or some Debian, I'll just have to cope. |
| 10:37.01 | gartral | applauds Chainsaw on the backronym |
| 10:37.09 | Chainsaw | bows |
| 10:38.07 | gartral | i can't believe I've met a reasonable person on IRC.. |
| 10:38.16 | PLMg | Chainsaw , u are saying Gentoo is a better alternative for asterisk then CentOS? |
| 10:38.43 | gartral | is waiting on the the search to dump to file, before pastebinning it |
| 10:39.05 | Chainsaw | PLMg: Probably. You've met the Gentoo packager for Asterisk now. |
| 10:39.26 | PLMg | :) I have to try it out sometime |
| 10:40.01 | gartral | I can't wrap my grey-matter around gentoo.. debian/ubuntu is what I learned first, and it's what I'm comfortable on |
| 10:40.09 | Chainsaw | PLMg: Just don't ask Digium for support. We've had a falling out over the amount of patches I deem required to get Asterisk stable. They feel it is perfect when they release it. |
| 10:40.51 | gartral | my poor server |
| 10:41.00 | PLMg | I don;t think I will ever meet digium support, since the want money for it :) |
| 10:41.50 | Chainsaw | gartral: The transition can be made. I started out out Slackware, then went to Debian, and ended up on Gentoo as a best-of-both. |
| 10:41.58 | Chainsaw | gartral: out out? out *on*. Typing is hard. |
| 10:42.25 | Chainsaw | likes the flexibility that compilation offers, but wants the convenience of a package system |
| 10:43.56 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 10:46.55 | gartral | Chainsaw: what? |
| 10:50.35 | Chainsaw | gartral: I was correcting my atrocious grammar mistakes. Not to worry. |
| 10:51.50 | gartral | Chainsaw: Ahh, I see |
| 10:52.10 | gartral | has a sneaking suspicion his hdd is dying of old age |
| 10:53.16 | gartral | Chainsaw: http://paste.ubuntu.com/6187659/ |
| 10:57.27 | gartral | yea.. dying |
| 10:57.29 | gartral | http://paste.ubuntu.com/6187667/ |
| 10:59.29 | Chainsaw | gartral: I don't see sector reallocations, calibration failures or spin-up retries. |
| 10:59.30 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 11:00.34 | gartral | Chainsaw: yea, but it's getting slower and slower to respond too commands every day |
| 11:02.13 | gartral | Chainsaw: it shouldn't take 20-30 seconds to open a man page, ever |
| 11:02.31 | Chainsaw | gartral: Agreed, that isn't acceptable. |
| 11:04.05 | gartral | has just bought a replacement drive, ironically, it cost more than th server |
| 11:05.04 | gartral | I picked up my server for $35 US, I just payed $37.59 for the replacement disk >.< |
| 11:08.52 | *** join/#asterisk jploh (~textual@122.2.37.42) |
| 11:09.00 | gartral | Chainsaw: http://us.msi.com/product/server/K1-1000-Series.html#?div=Overview |
| 11:09.35 | Chainsaw | gartral: http://www8.hp.com/uk/en/products/proliant-servers/product-detail.html?oid=5249584#!tab=features |
| 11:09.37 | *** join/#asterisk teeteewhy (~teeteewhy@no.ra.pe) |
| 11:10.40 | gartral | Chainsaw: yea, yea, I'm not going to find those in a datacenter closing sale any time soon |
| 11:11.11 | Chainsaw | gartral: That's what I run Asterisk on, and the only platform I can vouch for. |
| 11:11.33 | *** join/#asterisk Dpunkt (~Dpunkt@p5DE5EDED.dip0.t-ipconnect.de) |
| 11:12.03 | gartral | for $35/ea I got 2 single core systems with 4gb ecc ram on board.. only thing missing was hdds.. |
| 11:12.14 | Chainsaw | AMD or Intel? |
| 11:12.16 | gartral | AMD |
| 11:12.23 | Chainsaw | That is a good deal, yes :) |
| 11:12.52 | gartral | Chainsaw: correction: single-core/dual socket, populated |
| 11:13.26 | Chainsaw | gartral: Yes, along the lines of what I used to run. Well, that was dual socket dual core Opteron 2218. |
| 11:13.34 | Chainsaw | gartral: But it's a good vintage for stability. |
| 11:14.10 | gartral | Chainsaw: I can upgrade to dual-core chips, but frankley, these things run hot enough as-is |
| 11:14.10 | Chainsaw | gartral: Don't try and virtualise on them though, those CPUs do not have the acceleration for it. |
| 11:14.22 | gartral | oh I know |
| 11:14.34 | gartral | these are dedicated servers only |
| 11:15.24 | Chainsaw | gartral: If you see any Proliant DL365 or DL385 come up, jump on it. |
| 11:15.46 | gartral | but I can certainly vouch for their stability.. A) they were made by MSI for Atipa and installed in a NASA datacenter that recently closed by me. |
| 11:15.49 | Chainsaw | gartral: Working out of band is something every server should have. And few do. |
| 11:16.24 | gartral | Chainsaw: mine have nice BMCs built in |
| 11:16.45 | Chainsaw | gartral: Bare IPMI is no iLO4... |
| 11:17.19 | gartral | Chainsaw: B) they have transferrable warrenties that are still good for another 3 years |
| 11:18.16 | Chainsaw | gartral: I buy nearly everything refurb or renew these days. It's cheaper to run. |
| 11:18.23 | gartral | Chainsaw: meh, for what I paid.. having the ability to send a command to a BMC and instantly know why server just went castors-up is good enough |
| 11:18.27 | Chainsaw | gartral: Not to mention, you basically get free burn-in testing and no DOAs. |
| 11:18.40 | gartral | Chainsaw: NOT TRUE |
| 11:19.12 | Chainsaw | gartral: I've had more DOAs on new equipment then I've ever had on the refurb I've purchased. |
| 11:19.27 | eirirs | agrees on that |
| 11:19.33 | eirirs | refurb tends to be in better quality lol |
| 11:20.00 | gartral | I had to exchange one of these servers the day I bought them because the thing tried to kill me when it went suicidal.. 8x40 mm fans make an excellent flame-thrower! |
| 11:21.28 | gartral | still would like to know why it decided to catch fire on my testing bench, but.. yea, death-by-server is NOT something I want on my gravestone |
| 11:22.39 | gartral | but the ones I have now are stable, have been for months |
| 11:23.12 | gartral | I need a cooling solution for them though |
| 11:23.59 | gartral | is rambling, it's 7:20 AM and been up all night getting asterisk working |
| 11:28.11 | *** join/#asterisk andrewyager (~andrewyag@syd02s26-fw01.thecore.net.au) |
| 11:28.52 | *** part/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net) |
| 11:28.53 | Chainsaw | Catching fire? Generally accumulated dust. |
| 11:29.25 | Chainsaw | As long as it is on and the air is moving, it is *just* within thermal parameters and will not shut down. |
| 11:29.48 | Chainsaw | Then when the air stops, the long "beard" that it has grown settles down on hot components and gets stuck down. |
| 11:30.18 | Chainsaw | Next round... barely any air cooling and flammable, highly compacted dust stuck on metal heatsinks. |
| 11:30.18 | *** join/#asterisk vlad_starkov (~vlad_star@194.67.31.159) |
| 11:30.19 | gartral | Chainsaw: this wasn't just "dust" as it was off, I pressed the power, and 4 feet of flame shot out it's ass-end |
| 11:30.40 | Chainsaw | Oh right. Metal shavings shorting out a power supply. |
| 11:30.53 | Chainsaw | That's just sloppy though. Suggests bad air management in the data centre. |
| 11:31.07 | gartral | it was the oposite side of the machine, I had just cleaned it |
| 11:31.12 | eirirs | I am freezing when I'm inside the server room |
| 11:31.49 | Chainsaw | gartral: Shorting out a switch-mode power supply the right way will send very high voltages to components that do not appreciate it. |
| 11:31.57 | Chainsaw | gartral: The PSU was still the twisted fire-starter. |
| 11:32.18 | gartral | it was open, and upon inspection, the dvd rom had fused to itself to it's retaining bracket |
| 11:32.32 | Chainsaw | gartral: So it was your 5V rail. |
| 11:32.37 | gartral | likely |
| 11:32.52 | gartral | just glad I hadn't droppen my hdd in yet |
| 11:32.59 | gartral | dropped* |
| 11:33.25 | Chainsaw | gartral: Indeed, that would have been fatal. |
| 11:34.11 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 11:34.23 | gartral | moot point now.. that was 3 months ago.. and like I said, death-by-server doesn't lend itself to a honor-worthy tombstone |
| 11:34.59 | gartral | I don't want people to laugh as they read my grave message |
| 11:35.51 | Chainsaw | Agreed :D |
| 11:37.47 | *** join/#asterisk Nickinator (~Nickinato@123-243-142-239.static.tpgi.com.au) |
| 11:37.50 | gartral | I don't fear electricity, I don't fear inertia, Not to scared of water, suffocation would suck, but fire.. fire and me don't get along all to well |
| 11:44.50 | gnu_d | Why do I get: "tcptls.c:95 ssl_close: SSL_shutdown() failed: 5" ? |
| 11:48.43 | Chainsaw | Generally because the connection is forcefully closed. |
| 11:48.49 | *** join/#asterisk bombev (~User@73.109.Globcom.Net) |
| 11:48.51 | Chainsaw | It's the SSL equivalent of "you didn't even say goodbye!" |
| 11:48.51 | bombev | is it possible to send sms via Asterisk in the dialplan |
| 11:51.43 | Chainsaw | bombev: core show application SMS |
| 11:52.04 | bombev | now I am looking at Sendsms() |
| 11:52.09 | bombev | isn't it better? |
| 11:53.00 | gnu_d | Chainsaw: but I configured it correctly. |
| 11:53.21 | Chainsaw | bombev: I do not have that application on Asterisk 11.5.1 |
| 11:53.41 | Chainsaw | gnu_d: Sure. The other end is hanging up without saying goodbye. It's not you. |
| 11:53.44 | bombev | well I have it on Asterisk 1.8 |
| 11:54.00 | Chainsaw | bombev: It's worth stating that you are using an old version. |
| 11:54.10 | gnu_d | so that's why the WebRTC needs SSL, so it be a pain to be configured. |
| 11:54.38 | gnu_d | Chainsaw: but even a stable client is not working. |
| 11:54.51 | gartral | ubuntu seems to come with asterisk 1.8.13.1 |
| 11:55.18 | gartral | needs too recompile |
| 11:58.00 | gnu_d | Chainsaw: and what client to use then ? |
| 11:58.09 | gnu_d | for WebRTC |
| 11:58.45 | Chainsaw | gnu_d: You would have to ask others about WebRTC. |
| 11:59.55 | *** join/#asterisk kresp0 (~kresp0@1.Red-83-50-237.dynamicIP.rima-tde.net) |
| 11:59.58 | gnu_d | what others, I thought it would be easy using Asterisk, but as far as I see it can't even establish a TLS connection properly. |
| 12:00.29 | *** join/#asterisk kresp0 (~kresp0@1.Red-83-50-237.dynamicIP.rima-tde.net) |
| 12:01.05 | Chainsaw | gnu_d: I apply patches to Asterisk to fix the SSL handling. |
| 12:02.03 | Chainsaw | gnu_d: You should find that gentoo-asterisk-patchset-1.17.tar.bz2 applies cleanly to 1.8.23.1 if you compile it yourself. |
| 12:02.43 | Chainsaw | gnu_d: Relevant bug reports are https://issues.asterisk.org/jira/browse/ASTERISK-18345 & https://issues.asterisk.org/jira/browse/ASTERISK-17727 |
| 12:04.58 | Chainsaw | gnu_d: (The patch set is basically a collection of patches that are in the bug tracker, but not applied upstream by Digium. Every diff in the set has a link to a bug report, if applicable.) |
| 12:07.04 | gnu_d | Chainsaw: but I don't know which bug is it. |
| 12:07.31 | Chainsaw | gnu_d: 18345 |
| 12:09.10 | gnu_d | Chainsaw: do I need to download the diff files along with the patch file ? |
| 12:09.37 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 12:10.02 | gnu_d | Chainsaw: and I got Asterisk 12.0, same error with 11.5 too. |
| 12:11.45 | *** join/#asterisk Pullphinger (~Pullphing@12.40.23.68) |
| 12:12.23 | Chainsaw | gnu_d: I'll get you a patchset for 11.5.1 |
| 12:12.49 | Chainsaw | gnu_d: gentoo-asterisk-patchset-3.7.tar.bz2 please, for 11.5.1 |
| 12:13.08 | Chainsaw | gnu_d: I do not package 12, because it's not released yet. I have enough trouble making a release go, let alone an alpha, beta or RC. |
| 12:13.16 | *** join/#asterisk Prosouth__ (~sabayonus@62-2-198-100.static.cablecom.ch) |
| 12:16.34 | gnu_d | Chainsaw: how do I apply the patch ? |
| 12:19.29 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:21b:63ff:fe31:8426) |
| 12:20.32 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 12:21.39 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:25.13 | *** join/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net) |
| 12:26.18 | *** join/#asterisk Nickinator (~Nickinato@123-243-142-239.static.tpgi.com.au) |
| 12:26.44 | *** join/#asterisk Draecos (~Draecos@124-169-96-140.dyn.iinet.net.au) |
| 12:27.34 | Chainsaw | gnu_d: With path -p1 -i /wherever/the/diffs/are/stored/*.diff in the Asterisk source directory please. |
| 12:27.55 | gnu_d | Chainsaw: so I need the diff files as well ? |
| 12:28.23 | SuperNull | derp. |
| 12:29.17 | Chainsaw | gnu_d: Those are inside the .tar.bz2 that I supplied you. |
| 12:30.57 | gnu_d | Chainsaw: so try1, try2, try3 apply for each SSL version, if so I only need try1 ? |
| 12:31.26 | Chainsaw | gnu_d: Where is this "try" coming from? |
| 12:31.42 | gnu_d | Chainsaw: the diff files |
| 12:32.01 | gnu_d | Chainsaw: in the middle of the file somewhere there is a "try" word |
| 12:32.28 | Chainsaw | gnu_d: The patches are numbered, and will be applied in order. |
| 12:32.29 | *** join/#asterisk heffer (felix@fedora/heffer) |
| 12:32.49 | Chainsaw | gnu_d: By using *, they'll go in the right order. Starting with 01, and ending with whatever the highest number is. About 15. |
| 12:33.25 | gnu_d | Chainsaw: I got Archlinux, will that affect the patch ? |
| 12:36.50 | *** join/#asterisk Dovid (~Dovid@ool-457f6ac5.dyn.optonline.net) |
| 12:37.24 | Chainsaw | gnu_d: It will apply as long as it is the Asterisk release from the Digium website. |
| 12:39.29 | gnu_d | Chainsaw: I patched 11, do I need the others, cause they don't sound TLS related ? |
| 12:39.47 | Chainsaw | gnu_d: You need all of them, applied in sequence, as I have by now repeatedly explained. |
| 12:41.40 | *** join/#asterisk nRonn13 (~nRonn13@65.207.151.254) |
| 12:42.52 | nRonn13 | Astricon 2013 woOoO! |
| 12:45.45 | *** join/#asterisk hypknight (~hypknight@unaffiliated/hypknight) |
| 12:46.31 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 12:46.31 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 12:51.26 | *** join/#asterisk davlefouAMD (~david@197.15.69.187) |
| 13:04.06 | gnu_d | Chainsaw: I can't use Asterisk 1.8, the WebRTC support is in Asterisk 1.11, these patches are for 1.8 apparently :(. |
| 13:04.29 | *** join/#asterisk tuxx- (tuxx@pantoff0l.nl) |
| 13:06.08 | *** join/#asterisk zigg (~matt@unaffiliated/zigg) |
| 13:07.52 | Chainsaw | gnu_d: I gave you gentoo-asterisk-patchset-3.7.tar.bz for 11.5.1 |
| 13:08.28 | Chainsaw | gnu_d: Please clear out your directory and unpack the tarball again to make sure all patches apply cleanly. |
| 13:14.09 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 13:15.45 | gnu_d | Chainsaw: is this ok: for p in *.patch; do patch -p1 -d ../asterisk < "$p"; done ? |
| 13:16.49 | Chainsaw | gnu_d: Should do, yeah. |
| 13:16.57 | gnu_d | Chainsaw: it doesn't |
| 13:17.35 | gnu_d | Chainsaw: says: can't find file to patch at input line 12 |
| 13:19.57 | Chainsaw | gnu_d: Then you've done it wrong. You could always do what I actually said instead, which is to run patch -p1 -i /wherever/patchset/is/*.patch |
| 13:20.02 | Chainsaw | gnu_d: Whilst inside the Asterisk source directory. |
| 13:20.33 | gnu_d | Chainsaw: I tried, that's why I needed to modify the command and try something else. |
| 13:22.04 | gnu_d | Chainsaw: is it like this: patch -p1 -i ../asterisk-patchset/*.patch ? |
| 13:24.13 | Chainsaw | gnu_d: Yes. Or try your for loop, but use -i instead of < |
| 13:25.39 | gnu_d | Chainsaw: still doesn't work, for most of the files says can't find file to patch |
| 13:26.15 | Chainsaw | gnu_d: That would suggest that you are not in the correct directory. |
| 13:26.45 | gnu_d | Chainsaw: I'm in the source root directory, 3 files passed if not |
| 13:27.35 | Chainsaw | gnu_d: Fine, in that case only run one patch and leave the other bugs in there. |
| 13:27.49 | Chainsaw | gnu_d: It is the ssl-reader-should-block that's doing you in. |
| 13:32.59 | *** join/#asterisk danjenkins (~danjenkin@62.254.236.250) |
| 13:35.51 | Penguin | Moo. |
| 13:36.21 | Penguin | Yes. That's right, I said it. |
| 13:36.43 | WIMPy | Dangerous. |
| 13:37.04 | [TK]D-Fender | sticks a fork into Penguin to see if he's "done" |
| 13:37.17 | WIMPy | We usually put things that say moo on the grill. |
| 13:37.48 | WIMPy | hifives [TK]D-Fender |
| 13:38.14 | jmetro | i grilled up some moo the other night, it was delishus |
| 13:38.34 | Penguin | Well, this government shutdown is affecting all of us. |
| 13:38.50 | WIMPy | Has anyone ever tried grilled penguin? |
| 13:39.09 | Penguin | I tried to have some Cocoa Puffs and didn't have any clean bowls! |
| 13:40.18 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-111-87.user.veloxzone.com.br) |
| 13:41.05 | gnu_d | Chainsaw: it's not fixed |
| 13:44.22 | Chainsaw | gnu_d: Then I suggest that you also try the certificate chaining support. |
| 13:44.51 | Chainsaw | gnu_d: With those two patches in place Asterisk is more than capable of handling SSL properly and suspicion would then return to whatever the other end is. |
| 13:46.10 | *** join/#asterisk Maliuta (nikolai@donetsk.lusan.id.au) |
| 13:47.06 | gnu_d | Chainsaw: I really hope so, do you have any tips how to debug the whole thing via gdb ? |
| 13:47.45 | Chainsaw | gnu_d: I can't say I've done GDB debugging of Asterisk to that degree. Last time it segfaulted on me it was obvious from reading the source what they'd done. |
| 13:47.53 | Chainsaw | gnu_d: And I sent a patch. |
| 13:57.02 | gnu_d | Chainsaw: other than logs, what else can I monitor ? |
| 13:59.01 | Chainsaw | gnu_d: The experience of others perhaps? |
| 14:02.24 | gnu_d | Chainsaw: as far as I see, no one says anything. |
| 14:07.27 | Chainsaw | I like WebRTC as an idea, but haven't used it enough to be able to advise you. |
| 14:12.48 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:12.48 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:15.41 | *** join/#asterisk Vann (~manny@71-14-7-106.static.stbr.ga.charter.com) |
| 14:18.24 | gnu_d | Chainsaw: if this works with Asteirsk, it would be a blast :) |
| 14:19.55 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:20.21 | file | WebRTC is still being defined and specced... venturing into it can bring about sadness and frustration |
| 14:20.25 | *** join/#asterisk Dovid (~Dovid@ool-457f6ac5.dyn.optonline.net) |
| 14:21.32 | bacobart | well, that about describes the life of the average developer anyway ;p |
| 14:22.18 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 14:23.38 | gnu_d | file: well, better that then veturing with flash :(, I don't support flash. |
| 14:23.58 | gnu_d | file: my webapplication will be slow, unresponsive and ugly. |
| 14:24.03 | file | at this point I'd argue otherwise |
| 14:24.25 | gnu_d | file: besides there is no flash compiler for Linux |
| 14:24.31 | file | in the future not so much |
| 14:25.33 | gnu_d | my goal is to get it working on one computer :) |
| 14:26.00 | gnu_d | after max 2 years it will eventually work everywhere |
| 14:28.09 | file | there's more to it then that... depending on what you are doing and how it's you keeping up to date with the movement of WebRTC as it continues to evolve/be defined/change |
| 14:35.00 | *** join/#asterisk madsara (~madsara@bolgia7.circleeight.net) |
| 14:35.56 | madsara | Hey, I keep seeing this in REGISTERs, and I'm thinking I need to adjust it. How do I set realm in sip.conf? "...,realm="BroadWorks",algorithm=MD5..." |
| 14:36.34 | Penguin | Did you look at the sip.conf sample? |
| 14:37.09 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-vfkvqyfciuplrswa) |
| 14:37.09 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:37.55 | madsara | Yeah, I can set it in "general"... and I've tried it in locally connected stanza, but still no dice. Let me test general. |
| 14:38.08 | madsara | I've got numerous other peers working, just this one. |
| 14:39.51 | madsara | Yeah, even setting it in [general] doesn't alter that. |
| 14:40.52 | Penguin | Did you run sip reload after changing sip.conf? |
| 14:41.42 | madsara | Yes. |
| 14:45.34 | madsara | Actually, that BroadWorks realm isn't coming from me, it's the other end. Disregard. |
| 14:45.39 | madsara | I'll dig into this more on my own |
| 14:58.08 | Blashyrkh | i just try to configure a freepbx |
| 14:58.20 | *** join/#asterisk lvlolvlo (~lvlolvlo@unaffiliated/lvlolvlo) |
| 14:58.31 | Blashyrkh | i have a working asterisk configuratino for the same trunk but i cant make the freepbx to register |
| 14:58.49 | WIMPy | ~freepbx |
| 14:58.49 | 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 |
| 14:58.50 | Blashyrkh | is this the write place to ask about tis |
| 14:58.58 | Blashyrkh | thanks |
| 14:59.04 | Blashyrkh | haha talking to a bot |
| 14:59.25 | *** part/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net) |
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| 15:08.09 | Katty | hello my asterisk does not work at all how to fix plz |
| 15:11.16 | anonymouz666 | Katty: # halt |
| 15:12.01 | Katty | hugs anonymouz666 |
| 15:12.05 | Penguin | [freenode] -ChanServ(ChanServ@services.)- Channel #halt is not registered. |
| 15:12.10 | Penguin | No one there to help. |
| 15:13.49 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 15:19.21 | *** join/#asterisk adeeln (~adeeln@199.48.103.26) |
| 15:20.19 | adeeln | has anyone gotten buddy watching/presence on a polycom 650 running 3.3.2 to work with asterisk 1.8? |
| 15:24.45 | Penguin | What's the problem? |
| 15:25.12 | [TK]D-Fender | adeeln: I have similar if not that specific combo in use |
| 15:26.12 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 15:26.30 | adeeln | i can't seem to get the 650 to subscribe/watch the peers i'm looking for....read through all the docs i can find, but no joy....strange thing is that the exact same config file works for 3.2 |
| 15:27.34 | adeeln | do i need to define a calllimit for each peer? |
| 15:27.52 | *** join/#asterisk danjenkins (~danjenkin@62.254.236.250) |
| 15:30.54 | [TK]D-Fender | callcounter=yes <---- |
| 15:30.56 | [TK]D-Fender | required |
| 15:31.44 | adeeln | ok, so i have that set.... |
| 15:32.19 | *** join/#asterisk navaismo (~navai_000@189.241.90.55) |
| 15:33.12 | [TK]D-Fender | So show us your polycom configs and the debug of a phone booting, registering, and subscribing. |
| 15:33.43 | adeeln | i'll paste the polycom configs; but i don't see any subscribe messages |
| 15:34.02 | adeeln | which implies that my polycom config is probably incorrect |
| 15:34.44 | adeeln | what should the subscribe context be set to? the same context of the peer? |
| 15:36.36 | [TK]D-Fender | adeeln: If you don't set a subscribecontext it uses the standard context |
| 15:37.08 | adeeln | [TK]D-Fender: i'm assuming the default should work in most cases? |
| 15:37.09 | Penguin | Wouldn't you set it to the context where your hints are? |
| 15:37.23 | [TK]D-Fender | adeeln: Its YOUR config... you tell us :) |
| 15:37.35 | [TK]D-Fender | adeeln: "default" is a dangerous term |
| 15:37.43 | Penguin | For example, I keep my hints in my devices context. |
| 15:37.55 | Penguin | Nothing of any importance is in my default context. |
| 15:38.29 | adeeln | ok, just needed a little clarification on that; so it's definitely set to the wrong value |
| 15:38.38 | adeeln | but i still don't see any subscribe messages from the polycom |
| 15:40.29 | adeeln | the polycom configs: http://bpaste.net/show/NJ9uyfsbYngbJQBM3PzI/ |
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| 15:45.46 | [TK]D-Fender | adeeln: that looks mostly fine. Now the directory.... |
| 15:46.26 | adeeln | pretty standard stuff....http://bpaste.net/show/uX7teT5tAEgQFzWNzrQE/ |
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| 15:52.06 | [TK]D-Fender | adeeln: Looks like it should. Do you see "buddies" on-screen? |
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| 15:59.55 | adeeln | [TK]D-Fender: last i checked, no i didn't see it on the interface...unfortunately i'm not in front of that phone at the moment |
| 16:00.20 | [TK]D-Fender | adeeln: Go check, and reboot the phone after you do and watch it. |
| 16:00.29 | adeeln | yeah, i will |
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| 16:12.29 | Chainsaw | Oh neat, that's a new attack: |
| 16:12.31 | Chainsaw | <PROTECTED> |
| 16:12.31 | Chainsaw | [2013-10-03 17:11:53] WARNING[31764]: tcptls.c:269 handle_tcptls_connection: FILE * open failed! |
| 16:13.36 | *** join/#asterisk danjenkins (~danjenkin@62.254.236.250) |
| 16:15.27 | jeev | fender, i couldn't stop iax2 from having a bad day, i'll put them both on public ips and see if that fairs better.. i tried off all t he wans, each side has multiple wans, so i'm not sure, must be iptables cause but it wasn't always. |
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| 17:47.25 | k1ll3rwabb1t | I need to disable voicemail for a single extension in FreePBX 2.11.0.2, ideas? |
| 17:47.54 | Penguin | ~freepbx |
| 17:47.54 | 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 |
| 17:48.11 | jmetro | ^ |
| 17:48.20 | k1ll3rwabb1t | aye, thanks anyway |
| 17:48.45 | *** part/#asterisk k1ll3rwabb1t (~Joey@static-96-254-155-194.tampfl.fios.verizon.net) |
| 17:55.38 | boom^time | aw I was going to tell him to change vmb password to something unguessable and leave messages until it was full. |
| 17:56.01 | Penguin | This isn't Verizon Wireless. |
| 17:56.07 | boom^time | lol |
| 17:56.27 | boom^time | not with that attitude. |
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| 18:05.49 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 18:15.19 | velus | hello is anoyone about? |
| 18:15.30 | Chainsaw | No, we have collectively gone for dinner. |
| 18:15.41 | Chainsaw | But if you leave us a message, we will reply as soon as we are able. |
| 18:15.58 | velus | lo9l |
| 18:16.15 | velus | im wondering if there is any asterisk developers in here |
| 18:16.32 | WIMPy | ~ask |
| 18:16.32 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 18:19.43 | velus | ok well im looking for either a astrisk developer that can create a system that can take credit card numbers and then sed the data onto a certain website for collection and then listen on a certain php script to then give either a successfull or denied responce to the caller |
| 18:19.58 | [TK]D-Fender | velus: that is not an "Asterisk developer" |
| 18:20.13 | [TK]D-Fender | velus: that is just some person to CONFIGURE Asterisk. This is not core code. |
| 18:20.21 | jmetro | ^ its dialplan |
| 18:20.37 | velus | ok then can someone configure it then? |
| 18:20.40 | [TK]D-Fender | And yes... pretty much just call processing. |
| 18:20.53 | velus | yes call processing for credit cards |
| 18:21.20 | Penguin | Call processing, not credit card processing. |
| 18:21.36 | velus | but this needs to process creidtcards |
| 18:21.46 | jmetro | no, it needs to process dtmf |
| 18:21.49 | velus | ok |
| 18:22.02 | [TK]D-Fender | YOUR code do all this extra communication, not Asterisk |
| 18:22.08 | velus | ok |
| 18:22.13 | jmetro | your code + php scripting at least. |
| 18:22.34 | velus | ok can someone then be able to create the code and php script? |
| 18:22.41 | velus | well get a price on this |
| 18:26.03 | paulc | I'll do it for half a million dollars |
| 18:26.24 | velus | ok paulc can i borrow half a mill of you pleasse |
| 18:26.26 | boom^time | $3.6 mil. Bitcoin only. |
| 18:26.44 | paulc | (but in all seriousness - I've done a bunch with Asterisk using CURL to call PHP scripts in the back end, passing/returning data..) |
| 18:27.08 | jmetro | boom^time: what are you, the fbi? |
| 18:27.18 | velus | lol |
| 18:27.28 | boom^time | jmetro, my pockets are a little light after yesterday is all. |
| 18:27.48 | velus | so how much would this cost about paulc its im wanting to offer a pay by phone service to suplement other services |
| 18:28.20 | paulc | velus: I can't quote a price without understanding a bit more.. specs.. user experience etc.. but it's less than half a million dollars, if that helps ;-) |
| 18:28.30 | paulc | in fact.. it's probably less than $100k |
| 18:28.34 | boom^time | My price is firm. |
| 18:28.48 | paulc | thinks his price is more realistic than boom^time's |
| 18:28.51 | velus | one sec need to get the burnt chicken out a sec |
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| 18:35.28 | jmetro | i could totally write it. |
| 18:41.52 | boom^time | jmetro, competition doesn't help my cause. |
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| 18:48.30 | Katty | telnet |
| 18:48.30 | Nugget | telnet is eeeeeeevil! |
| 18:48.34 | Katty | i win. |
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| 18:50.52 | leifmadsen | I almost did that too :) |
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| 19:04.31 | *** join/#asterisk qakhan (~qakhan@50-200-52-14-static.hfc.comcastbusiness.net) |
| 19:05.03 | qakhan | how can we convert months number in name |
| 19:05.23 | qakhan | like 01=January, 02=February, 03=March |
| 19:06.19 | boom^time | SSH using non elliptical diffie-hellman key exchange while utilizing a Dual_EC_DRBG random number generating algorithm. Now that is evil. |
| 19:06.57 | boom^time | qakhan, convert them in what way? Do you want them verbally spoken? |
| 19:07.44 | boom^time | if so record them yourself, 01.gsm says "January", then Playback(/months/${MONTH}) or something similar |
| 19:08.46 | qakhan | boom^time i am getting months from caller and i want TTS say month name instead of number |
| 19:09.37 | qakhan | caller entered 03 so TTS says your entered 03. i want to say you entered march |
| 19:09.58 | boom^time | qakhan, did you miss what I typed? |
| 19:16.17 | qakhan | boom^time see this http://pastebin.com/E64ukfWi |
| 19:17.58 | velus | bck sorry had to also work on soemthing for my mum |
| 19:21.13 | boom^time | qakhan, learn how gotoif's work |
| 19:22.03 | Chainsaw | qakhan: You could call a bash shell to use the date command I suppose? date -d X/1/1 "+%B" |
| 19:22.34 | velus | so basically i need a system that when a user gets to a certain point in te pbx (something like option 2 then option 3 then option 3) it gets to the automated credit card processing (though i know its call processing) to then prompt them to put in a user account number (i think it will be 8 didgits) then put in a pi number (for there account) then ask them how much they want to put on the account in whole £'s then prompt them for there credit card nu |
| 19:22.35 | velus | mber xpire date snf there ccv code (three didgits ono the back) then send that data to a card prosessing center then when the center does a call back on the php script the script would then need to uupdate the amount in there account |
| 19:22.35 | Chainsaw | qakhan: Where X is the numeric month entered by the user. |
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| 19:25.50 | qakhan | ok |
| 19:27.37 | anonymouz666 | OK, guys. Next Astricon I will be present. |
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| 19:28.30 | anonymouz666 | Maybe the next will be in Vegas, NY, LA, SF or Honolulu... |
| 19:28.43 | anonymouz666 | :-) |
| 19:28.57 | velus | maybe UK |
| 19:29.37 | velus | paulc: u there |
| 19:35.23 | qakhan | anonymouz666 what ppl do in Astricon? |
| 19:36.38 | jmetro | its like a comiccon but with phones and phonebooks |
| 19:37.34 | WIMPy | You dress up in phone suits? |
| 19:38.49 | WIMPy | ... and push each others buttons? |
| 19:40.22 | jmetro | Oh yeah, we have phone piles and dance to on-hold music. |
| 19:41.19 | navaismo | hahahaha |
| 19:41.28 | velus | lol |
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| 20:28.00 | paulc | velus: I am now :) |
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| 21:31.24 | *** join/#asterisk jeffspeff (~Jeff@173.218.60.193) |
| 21:32.41 | jeffspeff | i'm looking for an easy way to allow users an outbound CID(num) that matches their DID while allowing their extension to remain their CID(num) when calling internally |
| 21:34.58 | jeffspeff | is there a channel variable set in sip.conf already designed for this purpose? i'm brainfarting right now, you'll have to excuse me. |
| 21:35.40 | [TK]D-Fender | There is... if you MAKE it... |
| 21:35.47 | [TK]D-Fender | SetVar= <----- |
| 21:35.51 | [TK]D-Fender | Read the sample configs.... |
| 21:37.13 | jeffspeff | in my sip db there's (which i'm pretty sure i only generated from the included sip mysql tables in the source resources) a cid_number what is that for? |
| 21:40.01 | roswell | jeffspeff, it's actually callerid in a peer's conf |
| 21:40.26 | [TK]D-Fender | jeffspeff: that is the callerID assigned over calls from that device definition |
| 21:41.11 | roswell | jeffspeff, it remains intact unless you will to redefine it further in a dialplan |
| 21:41.15 | jeffspeff | does cid_num override callerid ? |
| 21:41.26 | roswell | callerid includes cid_num |
| 21:41.57 | jeffspeff | my callerid is defined as "User" <1234> |
| 21:41.58 | *** join/#asterisk navaismo (~navai_000@189.241.90.55) |
| 21:42.06 | jeffspeff | and the cid_number is blank |
| 21:42.37 | jeffspeff | callerid and cid_number are two seperate columns in my sip db |
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| 21:43.06 | roswell | i'm not sure which table of that db we're talking about |
| 21:43.31 | jeffspeff | the sippeers table for asterisk |
| 21:45.26 | roswell | ehm... possibly the asterisk doesn't know nothing about that cid_number column it the db and ignores it |
| 21:45.42 | roswell | where did you get the schema for your db? |
| 21:47.16 | jeffspeff | when you extract the asterisk source tar go to contrib/realtime/mysql/ |
| 21:47.27 | jeffspeff | it's in the sippeers.sql file |
| 21:47.31 | jeffspeff | i just verified it |
| 21:52.25 | roswell | well try to put some desired values into that field |
| 21:52.57 | [TK]D-Fender | [17:41]jeffspeffdoes cid_num override callerid ? <- it overrides whatever the DEVICE itself may suggest |
| 21:57.46 | jeffspeff | [TK]D-Fender, typically, in my setup, the actual callerid displayed (when recieving a call from another extension/phone) is what is defined in the "callerid" field for that extension in the sippeers table. Currently, there is nothing set in any of the "cid_number" fields, they're all NULL. If I were to put numbers in the "cid_number" field for an extension, would that override the number defined in the "callerid" field? |
| 21:58.17 | jeffspeff | the "callerid" field contains data in the following format -> "Username" <1234> |
| 21:58.46 | jeffspeff | will "cid_number" value replace the 1234 of the "callerid"? |
| 21:59.17 | [TK]D-Fender | jeffspeff: there is only ONE callerid field for * |
| 22:01.12 | jeffspeff | the sippeers.sql file that is included with the source of asterisk includes a "callerid" column and a "cid_number" column |
| 22:01.54 | [TK]D-Fender | jeffspeff: Possibly just storing what the device itself offered |
| 22:02.04 | [TK]D-Fender | this has nothing to do with "alternate outbound. |
| 22:02.13 | jeffspeff | what? |
| 22:02.37 | [TK]D-Fender | What what? |
| 22:02.46 | jeffspeff | the "cid_number" field is provided in the source, the field doesn't actually contain any information, they're all NULL, i already said that |
| 22:03.04 | [TK]D-Fender | Is a call in progress by the device right now? |
| 22:03.10 | jeffspeff | i don't know why you keep going back to what the device is offering. the devices aren't offering anything |
| 22:03.13 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-fwjhcsvlmvxvtyyy) |
| 22:03.22 | [TK]D-Fender | And maybe that's why it's still blank |
| 22:03.38 | jeffspeff | ok, that's been my question |
| 22:03.51 | jeffspeff | i didn't know if i could set that value and use it as a channel variable |
| 22:04.05 | jeffspeff | or if it was deprecated and just not removed from the .sql file |
| 22:04.26 | jeffspeff | i didn't know if setting that would override another setting in the sippeers table |
| 22:05.44 | [TK]D-Fender | [17:35][TK]D-FenderThere is... if you MAKE it... |
| 22:05.46 | [TK]D-Fender | [17:35][TK]D-FenderSetVar= <----- |
| 22:05.47 | [TK]D-Fender | [17:35][TK]D-FenderRead the sample configs.... |
| 22:05.49 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 22:06.31 | jeffspeff | what does setvar have to do with me trying to figure out what "cid_number" field is for and how it behaves in relation to "callerid" field |
| 22:06.45 | jeffspeff | i'm aware of setvar |
| 22:07.19 | [TK]D-Fender | [18:03]jeffspeffi didn't know if i could set that value and use it as a channel variable <- wanna set a value and use it as a VARIABLE? |
| 22:07.26 | [TK]D-Fender | SetVar <_ |
| 22:07.28 | [TK]D-Fender | That IS the answer |
| 22:07.54 | Penguin | What does the cid_number field in the sql do? |
| 22:08.37 | jeffspeff | I'm wondering the same |
| 22:09.02 | Penguin | What does it do, what does it mean, what is it for? |
| 22:09.07 | Penguin | Any other way to ask the same damn thing? |
| 22:10.14 | [TK]D-Fender | I would not, could not, on a train! |
| 22:10.27 | Penguin | What is the purpose of it? |
| 22:10.40 | jeffspeff | Penguin, is that directed at me? |
| 22:10.43 | Penguin | No. |
| 22:10.48 | [TK]D-Fender | How is it meant to be used? |
| 22:10.55 | jeffspeff | ok, i had already asked the same thing |
| 22:10.58 | Penguin | It was clear to me that you didn't know. |
| 22:11.02 | jeffspeff | <jeffspeff> in my sip db there's (which i'm pretty sure i only generated from the included sip mysql tables in the source resources) a cid_number what is that for? |
| 22:11.06 | [TK]D-Fender | What did the designer have in mind for it? |
| 22:11.23 | Penguin | I'm just trying to get the answer to THAT question instead of some other tangent. |
| 22:13.46 | *** join/#asterisk _Nep`_ (~neo@host-94-251-36-150.bbcustomer.zsttk.net) |
| 22:13.55 | _Nep`_ | hi all |
| 22:14.14 | _Nep`_ | guys, i need your help :( |
| 22:14.25 | _Nep`_ | (sorry for my english) |
| 22:14.40 | _Nep`_ | i have problem with iax2 peer |
| 22:15.30 | *** join/#asterisk Dovid (~Dovid@ool-457f6ac5.dyn.optonline.net) |
| 22:15.32 | _Nep`_ | all afk? |
| 22:16.28 | jeffspeff | if you state you problem or question and somebody knows what the problem is or how to help, they will usually answer. |
| 22:16.59 | navaismo | ~ask |
| 22:17.00 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 22:17.10 | [TK]D-Fender | _Nep`_: No, we're all waiting for you tel describe the problem. |
| 22:17.16 | [TK]D-Fender | to* |
| 22:17.42 | _Nep`_ | oh, sorry ) |
| 22:19.11 | _Nep`_ | i have peers sip and iax2 (sip peers is cisco spa122, iax2 is softphone from android (wifi/3g)), problem with iax2 peers, when i call - |
| 22:19.11 | _Nep`_ | remote party can hear me well |
| 22:19.22 | _Nep`_ | ... no |
| 22:19.28 | _Nep`_ | remote party hears me badly |
| 22:19.38 | _Nep`_ | and I hear - good |
| 22:19.48 | _Nep`_ | (sorry, its google translate) :) |
| 22:19.57 | _Nep`_ | all peers use g729 codec |
| 22:20.48 | _Nep`_ | in remote side my voice breaks and hiccups |
| 22:21.07 | _Nep`_ | and voice of remote side i lessen nice! |
| 22:21.11 | _Nep`_ | ;( |
| 22:22.01 | _Nep`_ | and i dont know how i can fix this problem ( |
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| 22:23.53 | [TK]D-Fender | Is that android phone even in direct WIFI range of the same local subnet as the server? |
| 22:25.36 | _Nep`_ | yes |
| 22:25.54 | _Nep`_ | ping with android <--> server < 10ms |
| 22:26.11 | _Nep`_ | if you ask about it ) |
| 22:35.00 | _Nep`_ | anybody can help me? |
| 22:35.30 | [TK]D-Fender | so it sounds bad on loacl lan? |
| 22:35.34 | [TK]D-Fender | local lan |
| 22:37.28 | jeffspeff | Penguin, [TK]D-Fender, I have found that if "cid_number" is anything other than NULL (including an empty string) then it will override the number represented in the "callerid" field |
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