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| 00:35.10 | lordvadr | Is there a good way to play a soundfile to the caller while waiting for some sort of progress indication from a remote peer? The problem I have is that a remote peer, when busy, waits 20 seconds before returning the 486 BUSY. I want to play some kind of "please wait while we connect your call", but if I get a 180 RINGING back, I want to give the caller ringback at that point. |
| 00:44.30 | WIMPy | I havent tried that, but from what I found from trying to make that FXS card usable, Background() might work. |
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| 00:54.27 | alami | any one can help with elastix 3 gui default password |
| 00:55.59 | WIMPy | Yes. #elastix |
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| 05:04.32 | monsterco | Hi everyone - my Android phone has a hard time reading the voicemail .WAV files - is there an app for .WAV files I can download or should I just change to MP3 format? |
| 05:05.42 | [TK]D-Fender | VLC reads everything |
| 05:06.57 | ChannelZ | the wav's are basically gsm in a wav wrapper which is odd and no surprise not very well supported. |
| 05:07.18 | ChannelZ | I've never actually tried putting just a .gsm file on the phone and see if it knows what to do with it.. |
| 05:09.27 | monsterco | [TK]D-Fender - "VLC encountered an error with this media. Please try refreshing the media library" <= this comes from VLC Beta for Android |
| 05:10.31 | [TK]D-Fender | try wav instead |
| 05:11.10 | ChannelZ | (thats what I wound up doing, bigger but people aren't leaving 10 minute voicemails so.. eh..) |
| 05:11.40 | monsterco | I cant dictate the file type right now - have plans for future MP3 - but for now do you know if any apps play .WAV on Android? |
| 05:13.22 | [TK]D-Fender | I've never had problems between the stock player + VLC |
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| 05:16.01 | monsterco | [TK]D-Fender - have you tried a .WAV file? |
| 05:16.06 | monsterco | I think it's different from .wav |
| 05:16.45 | [TK]D-Fender | it is |
| 05:17.05 | monsterco | stock playe doesn't play it - i think it's a licensed ... |
| 05:17.28 | [TK]D-Fender | it isn't |
| 05:19.05 | monsterco | v-player used to play it but not anymore |
| 05:19.21 | monsterco | if not licensed wonder why there is such scarce support... |
| 05:20.39 | monsterco | I just re-installed v-player and now it plays it |
| 05:20.52 | monsterco | [TK]D-Fender - how are you online and active here 24/7? |
| 05:21.21 | [TK]D-Fender | I'm not... |
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| 05:25.27 | Penguin | I started using WAV because it could be played on Windows computers, Linux computers, and even Apple mobile devices. It doesn't play on Android, though, so I was actually just recently thinking about putting both wav and WAV in the email. |
| 05:28.02 | monsterco | Penguin - yeah, it's not supported by stock |
| 05:28.06 | monsterco | on Android |
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| 05:57.35 | volga629 | Hello Everyone, is possible pull missed calls through ami ? |
| 06:00.38 | [TK]D-Fender | Not unless you put it in something AMI can grab |
| 06:00.48 | [TK]D-Fender | have you actually READ the function list? |
| 06:05.45 | volga629 | yes I do, I am think on idea of xmpp bot for missing calls and vm |
| 06:06.51 | volga629 | I though on cdr, but not yet |
| 06:07.03 | volga629 | sure |
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| 06:22.56 | volga629 | need some how associate extension with jid |
| 06:43.21 | volga629 | is this right way to check dial status |
| 06:43.23 | volga629 | exten => s,n,GotoIf($[ ${DIALSTATUS} = "NOANSWER"]?notify) |
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| 08:53.26 | gavimobile | im having issues with what I think is an issue with my vpn router. I have a sip phone which has 2 accounts, 1 connecting to my pbx internally and one externally. the problem is that when I finish a call using the internal one (through vpn to server) the call doesn't hangup when the other party hangs up. can someone confirm that this is a problem with my vpn router? |
| 08:54.16 | gavimobile | I removed both accounts on the phone than recreated a new account using the wan ip and it works as expected. than same account only changed the host to the local address and the problem occurs |
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| 09:26.28 | ChannelZ | Sounds like a BYE is getting lost somewhere. |
| 09:28.02 | ChannelZ | But alas, this BYE isn't.. off to bed.. |
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| 10:21.56 | Rico | hi there |
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| 10:23.27 | Rico | I have a really basic problem that I can't understand : using asterisk 1.8.26.0, when I call "Queue(testqueue,,,,5)" in my dialplan, if nobody answers the call in the queue in the 5 seconds, my dialplan does not continue and call is dropped |
| 10:24.46 | Rico | Executing [...@incoming_calls:1] Queue("SIP/1002-000002b7", "ocetest-tous,nC,,,5") in new stack |
| 10:25.26 | Rico | does not work better without "n" option |
| 10:26.00 | Rico | any idea ? |
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| 10:33.13 | ghost75 | password = *( unreserved / escaped / |
| 10:33.14 | ghost75 | <PROTECTED> |
| 10:33.22 | ghost75 | those are chars NOT to be used in password? |
| 10:33.35 | ghost75 | http://www.ietf.org/rfc/rfc3261.txt |
| 10:34.45 | wdoekes | ghost75: they're allowed in a password, but they need to be (percent) escaped if they're used in a sip-uri |
| 10:35.02 | wdoekes | ehm |
| 10:35.18 | wdoekes | invert that, those are the allowed characters |
| 10:36.01 | ghost75 | allowed for uri or password |
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| 10:38.01 | ghost75 | was just curious about the register string where i use such chars |
| 10:38.06 | ghost75 | in password ^^ |
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| 10:41.22 | Rico | wdoekes: any idea about my really basic question ? |
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| 10:55.51 | MaliutaLap | curses ghost75 really didn't need to know that rfc number - now I'm going to have to read it |
| 10:56.20 | ghost75 | read all :> |
| 10:59.40 | MaliutaLap | ghost75: yeah, I have a tendency to do that with RFC's ... most of which are long winded. It takes up alot of time, and ends up being like legislation - most lawyers don't read the amount of legislation I do, and misquote it all the time. RFC's are to my tech life as legislation is to my political life |
| 11:00.24 | MaliutaLap | and that's the second RFC I've opened today |
| 11:01.23 | Rico | anybody about my queue timeout problem ? |
| 11:01.29 | Rico | please ? |
| 11:14.41 | ghost75 | Rico: u not even showed dialplan |
| 11:18.00 | Rico | ghost75: http://pastebin.com/MumnkP3y |
| 11:18.08 | Rico | she simplest diaplan you can imagine |
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| 11:34.26 | ghost75 | Rico: i never did queues but what should it do other than your playback and hangup from dialplan |
| 11:34.43 | Rico | ?? |
| 11:34.56 | Rico | did not understood |
| 11:37.19 | bsdice | Good morning |
| 11:37.59 | bsdice | quite a few bugfix commits in 11-SVN lately, great :) |
| 11:38.30 | ghost75 | Rico: i would look at timeout and retry in queues.conf |
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| 12:20.21 | woopstar | Hi ppl... Running make menuconfig, where does asterisk save it's compile options? I want to setup an asterisk on another server, and compile it the exact same way as the former. |
| 12:21.17 | WIMPy | menuselect.makeopts |
| 12:21.55 | woopstar | cool, thanks |
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| 12:43.10 | Rico | ghost75: find out why : timeout configured for the queue in mysql (realtime) preempts on the one configured when calling queue with timeout option |
| 12:43.20 | Rico | is that the normal behaviour ? |
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| 12:46.04 | *** join/#asterisk hypfer (~hypfer@unaffiliated/hypfer) |
| 12:46.57 | hypfer | hi, I have this analog telephone line and I want asterisk to "share" this and act like a VOIP server |
| 12:47.04 | hypfer | is that possible? |
| 12:47.16 | hypfer | what hardware can I use for that? (USB would be awesome) |
| 12:48.11 | hypfer | basically I want a reverse ATA (i think) |
| 12:48.51 | WIMPy | An FXO card. I don't know if Sangoma has that direction as USB. Possibly Xorcom. |
| 12:49.17 | WIMPy | Bu you should be aware that using POTS lines can be somehow frustrating. |
| 12:49.29 | WIMPy | Better get something digital. |
| 12:50.01 | hypfer | well I have no other choice |
| 12:50.24 | hypfer | It's voice over cable and the router only provides ports for analog phones |
| 12:50.33 | hypfer | Kabel Deutschland Hitron whatever |
| 12:50.49 | hypfer | or is there another way? |
| 12:50.58 | WIMPy | Port your numbers to sipgate an be happy. |
| 12:51.35 | hypfer | thats more of a workaround |
| 12:52.29 | WIMPy | Trying to use what you have, doesn't make sense. |
| 12:52.49 | hypfer | why? is it really impossible? |
| 12:53.05 | WIMPy | No, but it won't work very well. |
| 12:53.30 | hypfer | -> ? |
| 12:53.34 | WIMPy | POTS is good enough to connect a single phone. And only for that. |
| 12:54.56 | hypfer | USB FXO |
| 12:54.58 | hypfer | EUR 114,06 |
| 12:54.59 | hypfer | woah. |
| 12:55.12 | WIMPy | Or hope that the publication by BNetzA on router restraints will make a difference. |
| 12:55.27 | WIMPy | Well, analogue is expensive. |
| 12:55.50 | hypfer | protip: it won't |
| 12:55.54 | hypfer | this is germany. |
| 12:56.17 | hypfer | oh well. thanks anyways |
| 12:56.53 | WIMPy | Upgrade to the pseudo-ISDN service or use an independant ITSP. |
| 12:57.35 | *** join/#asterisk gerhard7 (~gerhard7@77-172-35-234.ip.telfort.nl) |
| 12:58.57 | hypfer | uhm.. don't I have to rent a FritzBox for that ISDN stuff? |
| 13:00.12 | WIMPy | It used to be that way. Don't know about the current terms. |
| 13:00.19 | WIMPy | But that would still be cheaper than the USB FXO. |
| 13:00.43 | hypfer | thats more of a ideologial question |
| 13:01.02 | bsdice | hypfer drop analog, go digital (fritzbox, et al) |
| 13:01.02 | hypfer | I'd rather pay 120€ than renting a fritzbox |
| 13:01.30 | hypfer | that hitron something supports bridge mode at least |
| 13:01.51 | bsdice | voice over cable will not give you many options |
| 13:02.04 | bsdice | architecturally imho even a dead end |
| 13:02.48 | hypfer | well I could just put a cheap phone there and accept that the situation is pretty much fucked |
| 13:03.08 | WIMPy | Wehn I was ther you bought th FB. No rent. |
| 13:03.19 | hypfer | 5€/month |
| 13:03.28 | hypfer | also I want to own my hardware |
| 13:03.34 | hypfer | just because.. reasons |
| 13:03.36 | WIMPy | Seems ok to me. |
| 13:04.14 | hypfer | and you're 100% sure that the USB FXO way won't work? |
| 13:04.15 | WIMPy | You don't have to use th FB. Save it's config and directly configure the accounts in Asterisk. |
| 13:04.29 | WIMPy | It will work. |
| 13:04.36 | WIMPy | For some definition of "work". |
| 13:04.43 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:04.50 | hypfer | 5€ per month to read out a config.. thats not a good deal |
| 13:04.55 | hypfer | WIMPy: what won't work? |
| 13:05.03 | WIMPy | You made that deal. |
| 13:05.05 | *** join/#asterisk madman2021 (d58f097a@gateway/web/freenode/ip.213.143.9.122) |
| 13:05.09 | madman2021 | Hello everyone, I've just deployed my own asterisk server, Current have 60 active channels and the CLI being spammed with "Rejecting unknown SIP connection from |
| 13:05.22 | bsdice | what 60! |
| 13:05.25 | madman2021 | Yes |
| 13:05.31 | WIMPy | You won't find out when the remote end answers or hangs up. |
| 13:05.33 | madman2021 | Total active calls sorry |
| 13:05.55 | hypfer | *sigh* |
| 13:06.15 | WIMPy | has to go |
| 13:06.16 | hypfer | why isn't there just a simple "bridge analog phone to VOIP" device which just works |
| 13:06.18 | bsdice | hypfer I been where you are, using ISDN card even. too much pain... |
| 13:06.19 | hypfer | :( |
| 13:06.53 | WIMPy | ISDN card is easy. |
| 13:06.58 | WIMPy | Any way... CUL |
| 13:07.09 | hypfer | renting the fritzbox for 24 month would be equal to buying that FXO |
| 13:07.27 | hypfer | can't I just use a cheap modem or something? :( |
| 13:09.48 | bsdice | if something won't work you'll spend a boat load of time to debug it |
| 13:11.03 | hypfer | a simple dumb cheap phone is the way to go i'll guess |
| 13:13.13 | madman2021 | Should I be worried about console spam regarding this.--: "Rejecting unknown SIP connection from {ip} |
| 13:13.25 | bsdice | is it a legit IP? |
| 13:13.29 | bsdice | not normal |
| 13:14.22 | madman2021 | Looks like some VPS company in Netherlands |
| 13:14.36 | madman2021 | So I guess this is telling me someones trying to brute force access? |
| 13:14.37 | bsdice | ahhh |
| 13:14.52 | bsdice | if you don't have a customer in NL, yes |
| 13:15.08 | *** join/#asterisk jhlavacek (~jirka@78.208.220.3) |
| 13:15.55 | madman2021 | There is only 1 sip trunk connected to this box, Would blocking all access but the one from my SIP trunk be a good idea ? |
| 13:17.10 | bsdice | is SIP upstream on your outside and all clients on inside (i.e. LAN)? |
| 13:17.55 | bsdice | allowguest=no alwaysauthreject=yes and then you could firewall off outside port 5060 udp |
| 13:18.06 | madman2021 | This sip box is for my company only, We have 2 phyical phones on the LAN and then ofcourse the SIP trunk is the only thing on the WAN |
| 13:18.12 | bsdice | ie put qualify=yes qualifyfreq=50 into peer |
| 13:19.11 | bsdice | then do something like $IPT -A INPUT -i $EXT_IF -m conntrack --ctstate ESTABLISHED,RELATED -j ACCEPT ; $IPT -A INPUT -i $EXT_IF -j L_DROP with iptables |
| 13:19.39 | bsdice | this will protect the box from incoming port 5060 udp probes |
| 13:20.17 | bsdice | while still allowing your upstream to function correctly |
| 13:20.29 | madman2021 | Dumb question, Is $IPT an enviroment variable or something snazzy with iptables |
| 13:20.38 | bsdice | iptables |
| 13:20.51 | bsdice | just /sbin/iptables |
| 13:20.51 | madman2021 | Ok I shall put some more reasearch into iptables |
| 13:21.08 | bsdice | the trick is to make it semi permeable |
| 13:21.39 | madman2021 | What is $EXT_IF. Feeling really dumb right now. |
| 13:21.40 | bsdice | your asterisk punches a hole into it that stays open for 180 seconds (UDP) |
| 13:21.48 | bsdice | hence the qualify=50 |
| 13:21.59 | bsdice | your external interface, its just a variable we use |
| 13:22.17 | madman2021 | Ahh that makes sense to me now haha |
| 13:22.47 | bsdice | L_DROP is a custom target, you can use DROP |
| 13:23.51 | bsdice | you have to keep the UDP port open though, i.e. qualify=50 or something |
| 13:23.59 | madman2021 | Okay, I'll see what I can do when the phones are off the hook |
| 13:24.35 | madman2021 | Thank you so much for the help. I'm ripping my hair out atm. Just migrated a old SBS 2003 to server 2012 R2. Not good. |
| 13:25.16 | bsdice | 2012 architecture requires some good thinking |
| 13:25.48 | bsdice | especially if you want to employ Exchange on same machine... AND run disaster backup and finegraned backup |
| 13:25.53 | madman2021 | In what aspect ? |
| 13:26.28 | madman2021 | I just backed up the PST's and removed entries from the active directory scheme then deployed exchange 2013 and injected the PST back in |
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| 14:11.12 | boratynskikamil | Hello. Question. If I have some person in the queue, and if an agent canceled phone, is it possible to do not recall him? |
| 14:13.20 | leifmadsen | boratynskikamil: the way that comes to mind is that you could use a Local channel to call the agent, then have a check that looks at the value returned by ${DIALSTATUS}, and if ${DIALSTATUS} = CANCEL or something else like that, then execute QueueMemberPause() |
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| 14:16.04 | boratynskikamil | leifmadsen: Ok. And one more question. If someone calls, Asterisk periodically redials Agent. Then I see lots of connections. Possible to disable it? |
| 14:16.16 | boratynskikamil | And change this redial time? |
| 14:18.35 | leifmadsen | sure, there are settings in queues.conf for this |
| 14:19.39 | leifmadsen | timeout and retry options |
| 14:21.16 | boratynskikamil | See. |
| 14:21.21 | boratynskikamil | TIMEOUT options. (: |
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| 14:25.18 | woopstar | leifmadsen: You still do paid consulting? |
| 14:27.56 | leifmadsen | woopstar: sorry I do not |
| 14:28.23 | woopstar | :) ok |
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| 15:23.33 | ghost75 | <madman2021> In what aspect ? <- exchange is a disaster to restore without good software |
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| 15:48.15 | alami | hello, i want to install asterisk, but i have never use CentOS, but i know good debian/ubuntu, what do you thing, should i install asterisk in debian or just use elastix |
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| 16:02.22 | leifmadsen | alami: use the distro you are most comfortable with |
| 16:02.37 | leifmadsen | asterisk will run well on both RHEL and Debian based systems |
| 16:06.17 | alami | leifmadsen: what do you recomended, just install asterisk, or something like elastix |
| 16:07.13 | leifmadsen | I'm not sure how to answer that question |
| 16:07.20 | leifmadsen | I have no idea what you're trying to accomplish |
| 16:09.03 | ghost75 | what speaks against debian/ubuntu? |
| 16:09.05 | alami | leifmadsen: http://elastix.org/ include a nice Gui to manage Asterisk |
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| 16:09.18 | leifmadsen | it sure does |
| 16:09.36 | alami | and i don't know what profesionell people use when running Asterisk |
| 16:10.08 | leifmadsen | depends entirely what they are trying to accomplish. I have never used any of the GUIs in production, but I also build fairly complex systems. |
| 16:10.11 | alami | but i thing everyone use GUI :) |
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| 16:10.20 | leifmadsen | that would be inaccurate |
| 16:10.30 | alami | lol |
| 16:10.34 | ghost75 | anyone here uses gui? |
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| 16:11.03 | alami | leifmadsen: also for big and complex systems you don't use GUI? |
| 16:11.18 | alami | FreePBX :) |
| 16:11.29 | alami | respect |
| 16:11.54 | leifmadsen | alami: we do, but we build our own |
| 16:12.09 | leifmadsen | because freepbx doesn't manage what I build |
| 16:12.34 | leifmadsen | again, it depends entirely what you're building, your resources, your capabilities |
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| 16:12.48 | leifmadsen | freepbx is fine for some people, but it would never work in the environments I work in |
| 16:13.01 | alami | leifmadsen: you build your own GUI? |
| 16:13.03 | _Corey_ | alami: FreePBX wasn't designed for "big and complex"... More "small and medium" with generally simple needs. |
| 16:13.11 | leifmadsen | alami: well our portal team does |
| 16:13.19 | leifmadsen | I just build the features and controls |
| 16:13.30 | bulkorok | I suppose thousand things changed in voicemail config and behaviour from 1.4.23 to 1.8 ?! |
| 16:13.42 | alami | leifmadsen: but if i learn CLI, i can also manage big and complex system? |
| 16:13.42 | leifmadsen | 1.4 to 1.8 is a huge jump |
| 16:13.53 | leifmadsen | alami: I can't speak to your ability to do that :) |
| 16:13.59 | bulkorok | leifmadsen: that scares me -.- |
| 16:14.11 | leifmadsen | do not upgrade a 1.4 to 1.8 system |
| 16:14.13 | leifmadsen | you will have a bad time |
| 16:14.22 | leifmadsen | you must migrate to a 1.8 system |
| 16:14.23 | bulkorok | njo... just the voicemail thing |
| 16:14.24 | alami | lol :) if you can do it, i will also do it leifmadsen :) |
| 16:14.37 | leifmadsen | alami: I have a 12 year head start on you :) |
| 16:15.11 | alami | mhh Okay :D pretty good |
| 16:16.01 | alami | i have CCNA voice, but that's nothing face of your 12 year experience, but i want also only learn that |
| 16:16.13 | leifmadsen | http://asteriskdocs.org |
| 16:16.34 | alami | thanks a lot leifmadsen |
| 16:18.30 | bulkorok | multiple options in application without colon seperation, right!? |
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| 16:23.10 | giesen__ | I have an issue on a trunk to a lync server. When a call is call forwarded on the lync server (ie hairpinned), on the second leg of the call asterisk sends RTP to the internal IP (the lync server is NAT'd) |
| 16:23.12 | bulkorok | I have no idea how the voicemail thing on the 1.4 server works... |
| 16:23.18 | giesen__ | anyone seen this/know how to correct it |
| 16:23.29 | giesen__ | calls to extensions on the lync server work just fine |
| 16:23.46 | giesen__ | it's when the lync server makes the second call out that asterisk is sending media to the internal IP |
| 16:25.51 | *** join/#asterisk sliske_ (~sliske_@2001:4800:780e:510:6c33:9987:ff05:245f) |
| 16:26.16 | Penguin | There isn't much functional or configurational change in voice mail between 1.4 and 1.8. I performed the upgrade from the old branch to the new branch quite easily. |
| 16:26.55 | bulkorok | Penguin: yeah... but in the 1.4 voicemail.conf file there is only a general and zone section... |
| 16:26.55 | Penguin | But of course there is the matter of what things you do with your asterisk that I don't do with my asterisk. |
| 16:27.16 | bulkorok | just searchcontexts=yes |
| 16:27.52 | Penguin | 1.8 also has just a general section and the zone(s) of your choosing. |
| 16:28.03 | bulkorok | no voicemails defined -.- |
| 16:28.07 | Penguin | See the sample voicemail.conf from 1.8.latest |
| 16:28.15 | Penguin | 1.8.26.0 |
| 16:28.59 | bulkorok | 1.4 |
| 16:28.59 | bulkorok | exten => _X.,n,Voicemail(${EXTEN}|u|s) |
| 16:29.06 | Penguin | Oh, they call it voice mail contexts, not zones. |
| 16:29.11 | bulkorok | where exten is a 5 digit number |
| 16:29.17 | Penguin | That's dialplan, not voicemail. |
| 16:29.22 | bulkorok | sure |
| 16:29.32 | Penguin | Dialplan syntax changed on certain apps. |
| 16:29.46 | bulkorok | and how gets Voicemail() the personal voicemail?! |
| 16:30.06 | Penguin | core show application Voicemail |
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| 16:31.10 | bulkorok | anyway... I'm done... tomorrow more :-) |
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| 16:35.53 | Katty | SCOTCH TAPE DESPENSER |
| 16:36.12 | Katty | telnet |
| 16:36.12 | Nugget | telnet is eeeeeeevil! |
| 16:36.14 | Katty | hugs Nugget |
| 16:36.18 | Katty | Nugget: texas |
| 16:36.18 | Nugget | Don't mess with Texas. |
| 16:36.53 | ghost75 | how do you manage large dialplans? |
| 16:37.02 | Katty | with a whiteboard. |
| 16:37.06 | Penguin | vim and a few files |
| 16:37.06 | Katty | or your noggin. |
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| 16:37.24 | Katty | mister Penguin's noggin can manage anything, i hear. |
| 16:37.40 | Katty | eyes Iamnacho with a fork |
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| 16:38.02 | ghost75 | extensions.conf can has subfiles? |
| 16:38.13 | Katty | it can include things. |
| 16:38.14 | Penguin | Sure. See #include |
| 16:38.49 | Penguin | #Include extensions.conf.d/*/extensions.conf; |
| 16:38.56 | Penguin | That's what I do. |
| 16:39.01 | Katty | OR DO YOU. |
| 16:39.06 | Penguin | Well... |
| 16:39.09 | Penguin | Yes, I do! |
| 16:39.17 | Iamnacho | no forks! |
| 16:39.23 | Penguin | DANGER!!! |
| 16:39.35 | Katty | how about a spork? |
| 16:39.52 | Katty | I'm not sure spooning you would be appropriate. |
| 16:39.59 | Iamnacho | a spork maybe :) |
| 16:40.20 | Katty | titanium spork? |
| 16:40.26 | Iamnacho | even better |
| 16:40.31 | Penguin | If you're going to spoon me, you've got to let me fork you. |
| 16:40.32 | Katty | *hee* |
| 16:41.00 | ghost75 | brings knife |
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| 16:41.26 | Katty | woahwoahwoah. one does not simply knife nachos. |
| 16:42.03 | ghost75 | dont remember me on yesterdays cheese nachos |
| 16:42.33 | Katty | i don't think i was here yesterday. well i was here, but not. |
| 16:42.55 | ghost75 | i had a full bag of them, now empry |
| 16:43.10 | Katty | gosh. |
| 16:43.16 | Katty | i hope you drank lots of water! they are high in sodium |
| 16:43.18 | Penguin | Where I come from, nachos do not come in a bag. |
| 16:43.22 | Iamnacho | saddly i haven't had nachos for a while |
| 16:43.30 | Katty | Penguin: i think he means doritos or something. |
| 16:44.00 | ghost75 | wow, my gpu just crashed |
| 16:44.14 | Katty | that's what you get when you eat a whole bag of chips! |
| 16:44.15 | Penguin | The tortilla chips come in a bag, but then I have to load them up with cheese, meat, sour cream, green onions, maybe refried beans. |
| 16:44.28 | ghost75 | yes tortilla chips |
| 16:44.41 | Katty | mmm that sounds good. needs tomato :> and lettuces! |
| 16:44.46 | Katty | ALL THE LETTUCES. |
| 16:45.14 | Penguin | I don't eat tomato and I typically don't put lettuce on the nachos. |
| 16:45.24 | Katty | more for me :> |
| 16:45.34 | Penguin | You may have all of it. |
| 16:46.07 | Penguin | Maybe some guacamole if it isn't nasty. |
| 16:46.10 | Katty | \o/ |
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| 17:39.54 | toot | hi folks, been a long time since playing with asterisk but going to buy a Digium 1A4B04F, just checking if there is anything i need to know, are they fine/good with pci-e 3 or? any advice |
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| 17:52.01 | giesen__ | Anyone have any thoughts on my issue? I have a trunk to a lync server. When a call is call forwarded on the lync server (ie hairpinned), on the second leg of the call asterisk sends RTP to the internal IP (the lync server is NAT'd). calls to/from extensions on the lync server work just fine, it's when the lync server makes the second call out that asterisk is sending media to the internal IP |
| 17:52.59 | giesen__ | asterisk seems to ignore the nat=yes setting and is not translating the internal IP to the external IP |
| 17:54.29 | Penguin | When I run into that, it is almost always a problem with my NAT, not with asterisk configuration. |
| 17:56.46 | giesen__ | the weird thing is it only happens with forwarded calls |
| 17:56.57 | giesen__ | all other calls asterisk correctly translates the IP |
| 17:57.11 | Penguin | And what I mean by that is that I can leave all configurations the same but change the router which does the NAT and it magically works. |
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| 17:58.31 | Penguin | I had a Cisco 800 series router that provided that same problem to me, which I wasn't amused by. I switched to a Linux-based solution and it was fixed. |
| 17:59.23 | volga629 | because on cisco by default enabled sip media and rtp filtering |
| 17:59.33 | giesen__ | unfortunately I have no control over the router |
| 17:59.43 | Penguin | I've also run into similar problems using the Linux router and fixed them by clearing the conntrack info. It was like it was retaining useless data and causing trouble. |
| 18:00.14 | volga629 | and you can't disable it completely, because it intended work with call manager |
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| 22:15.39 | jeffspeff | what (if any) wireless voip phones are recommended? we use polycom soundpoints for desk phones and somebody is wanting a wireless phone. |
| 22:16.35 | WIMPy | Use a CAT-iq phone. Wifi is oncool for voice. |
| 22:20.27 | jeffspeff | thanks WIMPy |
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| 22:42.09 | *** join/#asterisk Xylit00l (~XyliBox@147.213.67.121) |
| 22:45.25 | Xylit00l | hello all |
| 22:45.47 | *** join/#asterisk Milarepa (~Milarepa@host-74-211-92-125.beyondbb.com) |
| 22:48.14 | Xylit00l | I have a strange problem with audio using voicenetwork.ca trunk; i'm behind nat with ~500 ports fw to asterisk machine, same port range set up in rtp.conf; problem comes randomly with sometimes when concurent calls, sometimes when only one call.. audio just dissapears for ~5 seconds.. or more.. or at all, randomly |
| 22:48.32 | Xylit00l | is there any setting in asterisk config to help this ? |
| 22:48.50 | dijib | are you doing any bit shaping on your router? or QOS? |
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| 22:52.07 | Xylit00l | bandwidth is not a problem, same time these audio dissapear ping is stable, cpu is normal |
| 22:53.57 | MasterSenpai | hey. i just installed AsteriskNOW 3.0.1 and I am trying to figure out how to make sure that the unistim module is enabled? |
| 23:08.06 | Penguin | mastersenpai: module show like unistim |
| 23:13.00 | Xylit00l | dijib: after audio silence, sometimes it gives reinvite |
| 23:14.46 | MasterSenpai | Penguin: I do see that it is loaded. The problem I am running into is using the Web interface to set up Nortel ip phones and I don't see anything about unistim. There isn't even an unistim.conf file in /etc. Is there something I need to do? |
| 23:16.23 | MasterSenpai | They are i2004 models |
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| 23:36.02 | Xylit00l | any suggestion why audio is intrerrupting duiring message play then CDR updated on SIP and menu starts from begining ? |
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