| 00:01.42 | apb1963 | easy is good... I like easy |
| 00:01.57 | apb1963 | ty |
| 00:04.11 | *** join/#asterisk maxus2 (~k3xenn@ftp.r-group.com.au) |
| 00:07.37 | maxus2 | Hi People! We have an odd issue with IAX. we have two virtuals running on xen. I try to make a call from the first to the second but the second never sees any iax messages. I can see the Recv-Q filling up and I can packet capture the iax packets. but the packets never seem to make it to asterisk, anyone know what would be causing this? |
| 00:12.18 | *** join/#asterisk hackish (b8a2b885@gateway/web/freenode/ip.184.162.184.133) |
| 00:13.12 | hackish | My internet died today and when it came back asterisk is busted... I've tried restarting it but it seems my iax2 trunk isn't registering... not sure why. |
| 00:13.28 | [TK]D-Fender | maxus2: Clearly a networking SNAFU |
| 00:13.28 | hackish | <Unregistered> 60 Request Sent |
| 00:13.44 | hackish | ideas on how to debug it beyond that? |
| 00:14.39 | ChannelZ | Did your IP change? Is it a firewall issue? |
| 00:14.47 | maxus2 | D-Fender - any ideas on where to look? |
| 00:15.12 | maxus2 | I have been looking at it for three days, and just cant seem to make headway |
| 00:15.19 | maxus2 | thier ont he same network |
| 00:15.26 | maxus2 | they are on the |
| 00:15.56 | hackish | hehe. silly ISP... It seems that their modem gave me a dummu IP and the NAT firewall picked that up. |
| 00:16.06 | hackish | ipnat -C -F -f /etc/ipnat.rules |
| 00:16.25 | hackish | and... 184.162.184.133:4569 60 Registered |
| 00:16.34 | [TK]D-Fender | maxus2: Your virtuals are either NAT'd, or otherwise firewalled |
| 00:17.02 | maxus2 | Nope no firewalling all are in a netowkr togther |
| 00:17.10 | maxus2 | no nat either' |
| 00:17.29 | maxus2 | the packets make it to the other machine |
| 00:17.33 | [TK]D-Fender | maxus2: Show us what you've set up and the debugging attempts |
| 00:17.40 | maxus2 | its just that asterisk doesn't grab them fromt he recv-q |
| 00:18.48 | maxus2 | Okay so I have two virtuals one with the ip of 172.16.4.120 talking to another asterisk machine on 172.16.4.100 |
| 00:19.43 | apb1963 | igcewieling: Will it only see output to stderr? Or will it also see stdout from my script? |
| 00:20.56 | maxus2 | just getting a pastie together |
| 00:21.19 | apb1963 | because I'm still not getting output |
| 00:21.53 | *** join/#asterisk suneye (~atcmmi@119.122.152.6) |
| 00:27.40 | maxus2 | [TK]D-Fender: http://pastie.org/private/ek6mt4y5vcfmjpckinanma |
| 00:30.11 | [TK]D-Fender | maxus2: maxus2 Where do i see you proving the firewall is empty on the dest machine, and get to see the packets, and see that * is listening for it let alone running? |
| 00:31.01 | maxus2 | I will add those |
| 00:31.16 | *** join/#asterisk italorossi (~italoross@ec2-54-232-199-58.sa-east-1.compute.amazonaws.com) |
| 00:33.21 | maxus2 | whats the best was to show you asterisk is listening? |
| 00:34.16 | ChannelZ | netstat -alp |
| 00:34.50 | apb1963 | ok I modified the script to output to both stdout & stderr... still nothing. Script here: http://ix.io/5P0 |
| 00:35.25 | maxus2 | thanks! |
| 00:36.47 | *** join/#asterisk troyt (~troyt@2001:1938:240:2000::3) |
| 00:38.09 | maxus2 | [TK]D-Fender: http://pastie.org/private/gigwaqffdhmhtrwt849xq |
| 00:38.28 | maxus2 | Just ask if there is anything else you need? |
| 00:39.37 | [TK]D-Fender | apb1963: that is not an AGI |
| 00:39.54 | [TK]D-Fender | apb1963: you are not flushing the inbound data first which you HAVE to do for an AGI |
| 00:40.12 | [TK]D-Fender | apb1963: Just because it is a shell script doesn't mean it will work as an AGI |
| 00:40.20 | [TK]D-Fender | apb1963: Go re-read the chapter on it |
| 00:40.25 | apb1963 | ok |
| 00:50.51 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 00:53.43 | igcewieling | or use an AGI library |
| 00:59.40 | *** join/#asterisk dtcrshr (~datacrush@unaffiliated/datacrusher) |
| 01:00.12 | apb1963 | [TK]D-Fender: This any better? http://ix.io/5P6 |
| 01:01.20 | apb1963 | That gives me: ERROR[28513][C-00000002]: utils.c:1187 ast_carefulwrite: write() returned error: Broken pipe |
| 01:01.29 | [TK]D-Fender | apb1963: No, you clearly haven't read the book at all. |
| 01:01.42 | apb1963 | the examples are in perl, php and python |
| 01:02.45 | [TK]D-Fender | yes well it clearly describes what interfaces you should read & write from. And if you don't know enough about the language you are writing yours in to do that appropriately then you have a real problem.... |
| 01:03.16 | apb1963 | what interfaces.... something other than stdin and stdout and stderr? I guess I missed that. |
| 01:04.25 | [TK]D-Fender | Those exactly |
| 01:04.37 | apb1963 | so that's what the script does |
| 01:04.45 | apb1963 | it reads from stdin until there is no more to read |
| 01:04.51 | apb1963 | and then writes to stdout & stderr |
| 01:05.39 | apb1963 | well ok it writes to stdout first |
| 01:05.48 | apb1963 | that was more of a test for the command line |
| 01:06.48 | maxus2 | [TK]D-Fender, any ideas on this iax issue? |
| 01:09.19 | [TK]D-Fender | maxus2: not yet. Enable iax debug on the receiving end and take a look. |
| 01:11.07 | maxus2 | i have, nothing appears at all |
| 01:11.18 | maxus2 | its got me baffeled. |
| 01:17.18 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 01:21.56 | jayk | nou |
| 01:26.31 | *** join/#asterisk gnudna (~sklav@unaffiliated/sklav) |
| 01:27.05 | gnudna | Hi guys how does one add his callerid to asterisk so when i go out of a trunk i get my CallerID displayed? |
| 01:29.17 | gnudna | here is the pastebin of my extension.conf |
| 01:29.19 | gnudna | http://pastebin.com/Dp35qeeK |
| 01:30.29 | [TK]D-Fender | gnudna: exten => _1NXXNXXXXXX,1,Set(CALLERID(name)=1234567890) |
| 01:30.36 | [TK]D-Fender | gnudna: By setting the NUMBER, not the NAME |
| 01:30.45 | [TK]D-Fender | gnudna: You don't get to set the name that way. |
| 01:31.01 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 01:32.35 | gnudna | Ho does one set the name? |
| 01:33.07 | gnudna | how^ |
| 01:35.32 | gnudna | i have that already i was hoping to set the name or worst coes to worst override what the client sends. |
| 01:36.18 | gnudna | by the way this is for a new home phone setup im working on |
| 01:37.39 | [TK]D-Fender | you almost never get to. |
| 01:37.46 | [TK]D-Fender | Names are held in a central DB. |
| 01:37.51 | [TK]D-Fender | not YOURS. |
| 01:38.04 | [TK]D-Fender | go read up on CNAM |
| 01:38.47 | gnudna | ok if i can get the number that should be enough i guess |
| 01:39.23 | gnudna | im trying to get all the phone working like a basic pots with asterisk aka to easy the family into the new setup |
| 01:39.26 | gnudna | ;) |
| 01:40.18 | gnudna | just need to deal with the echo now :( |
| 01:40.26 | *** join/#asterisk Bradada (~Bradada@220-135-49-159.HINET-IP.hinet.net) |
| 01:45.53 | [TK]D-Fender | gnudna: Voip.ms should not be a source of echo. what are you using as an enpoint for the other half of the call? |
| 01:48.34 | gnudna | an ata device in this case |
| 01:48.52 | gnudna | assuming im not mistake by your question |
| 01:49.10 | gnudna | i have a phone plugged into an ata device to convert analog to digital |
| 01:51.01 | gnudna | works well in general but some testing today showed echo on a regular line which i was not hearing on my end. |
| 01:51.04 | jeev | any t38modem users around ? |
| 01:51.22 | [TK]D-Fender | What ATA? |
| 01:51.57 | gnudna | dta-310 |
| 01:52.26 | gnudna | working pretty good overall but it is slightly dated |
| 01:53.06 | gnudna | although sip show peers shows this for voip.ms OK (1031 ms) |
| 01:53.12 | gnudna | which seems damn high to me |
| 01:53.46 | gnudna | i jump from a high number to a low 32ms evey so often |
| 01:54.13 | gnudna | this happens on both mtl servers and the toronto one |
| 01:54.33 | gnudna | might need to enable some tos on my firewall |
| 01:56.57 | [TK]D-Fender | That is a SIP QUALIFY time and may have no relation to actual PING & call perfomance |
| 01:58.49 | gnudna | ah ok |
| 01:58.59 | gnudna | i misunderstood what that was for |
| 01:59.09 | [TK]D-Fender | gnudna: Test with another device and see if the echo is the same |
| 01:59.27 | [TK]D-Fender | It's more fo a keep-alive and check that they are even responding. |
| 01:59.38 | [TK]D-Fender | Don't take the number itself as good/bad necessarliy |
| 01:59.46 | gnudna | i do not perceive echo from my end sadly |
| 01:59.55 | gnudna | the person i called noticed echo on the line |
| 01:59.58 | [TK]D-Fender | Like if you yell help and I see you are actually on fire I might react faster... |
| 02:00.05 | gnudna | on my end the call was crystal clear |
| 02:00.20 | gnudna | :) |
| 02:00.20 | [TK]D-Fender | Not that I didn't see you just as fast.. it's a question of how the answer is prioritized on top of basic latency |
| 02:00.40 | [TK]D-Fender | So test with another device on your end |
| 02:00.53 | gnudna | enabling jitterbufffer is that a good thing or a bad thing? |
| 02:00.55 | [TK]D-Fender | If that acts the same then it is your provider.. thoguh it shouldn't be |
| 02:01.00 | gnudna | im looking for the safe options |
| 02:01.16 | [TK]D-Fender | Start with the "verify my ATA isn't crap" test first |
| 02:01.42 | gnudna | sadly i do not have a spare at the moment |
| 02:02.17 | gnudna | i did change my route to the more expensive option on voip.ms to see if there is a difference |
| 02:02.56 | *** join/#asterisk hebber (~hebber@node-14pg.pool-125-25.dynamic.totbb.net) |
| 02:03.18 | [TK]D-Fender | gnudna: Softphoe, etc... |
| 02:03.31 | [TK]D-Fender | just ANYTHING other than your ata |
| 02:03.39 | gnudna | yeah i got a few of those |
| 02:03.57 | gnudna | will test and see |
| 02:04.59 | gnudna | just tweaking extensions.conf at the moment. |
| 02:05.06 | gnudna | got the dial-in working as expected. |
| 02:05.34 | gnudna | care to look at the pastebin to see if it looks good. meaning nothing obviously bad |
| 02:09.02 | gnudna | voila http://pastebin.com/57BVDMgd |
| 02:10.28 | [TK]D-Fender | exten => _91NXXNXXXXXX,n,Dial(Gtalk/gmail/${EXTEN}@voice.google.com,,r) <- very rarely have to fake ringing... |
| 02:11.04 | gnudna | that was from a howto i had followed a while back |
| 02:11.17 | gnudna | i guess i can comment that line out anyways |
| 02:11.30 | [TK]D-Fender | As I don't see any contaner contexts I'm not sure what uses [voipms-outbound] vs [longdistance] |
| 02:11.55 | gnudna | ah sorry i have a context called users where i include them |
| 02:14.57 | gnudna | on voip.ms i have to set the caller id sadly and can not do it from asterisk :( |
| 02:15.25 | gnudna | if i remove the caller id i get unknown when i dial my cell or home number |
| 02:15.48 | gnudna | even though i have it defined in the config as you saw from the pastebin |
| 02:16.06 | [TK]D-Fender | config still says the NAME, not the NUMBEr |
| 02:17.03 | *** join/#asterisk Tarso (~Tarso@189.61.52.46) |
| 02:17.10 | gnudna | _1NXXNXXXXXX,1,Set(CALLERID(name)=1234567890 is not correct then? |
| 02:17.23 | [TK]D-Fender | NUMBER, not NAME |
| 02:17.30 | gnudna | ah ok |
| 02:18.43 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 02:26.33 | gnudna | [TK]D-Fender, when dialing in aka calling from outside how do i get it to show who's number is calling in my case it shows my asterisk number in this example 1234567890 |
| 02:27.28 | gnudna | anything wrong with my voipms-inbound? |
| 02:27.47 | gnudna | i even commented out the line same => n,Set(CALLERID(name)=DI-${CALLERID(num)}) and still no change. |
| 02:28.53 | [TK]D-Fender | pastebin your sip.conf for [603] masking only the secret |
| 02:29.05 | gnudna | i had callerid set in there |
| 02:29.11 | gnudna | i just removed it |
| 02:29.20 | gnudna | was wondering where it was getting my nickname from |
| 02:29.22 | gnudna | ;) |
| 02:29.32 | gnudna | btw damn your good at this |
| 02:29.38 | gnudna | new my issue before me |
| 02:29.41 | gnudna | ;) |
| 02:30.26 | gnudna | awesome that was it |
| 02:31.21 | [TK]D-Fender | :) |
| 02:31.33 | [TK]D-Fender | It's usually buried in the big print. |
| 02:31.38 | *** join/#asterisk ivcivc (~kvirc@80.237.55.26) |
| 02:32.51 | gnudna | so the idea before was if i dial 9 and then number i go threw google talk |
| 02:33.04 | gnudna | if i dial regularly i go threw voip.ms |
| 02:33.19 | gnudna | im guessing you figured that part out |
| 02:33.44 | [TK]D-Fender | Yeah, I read patterns and stuff. |
| 02:33.45 | gnudna | the google was a test option i was using before i jumped onto voip.ms |
| 02:34.04 | gnudna | but it worked well enough that i left it in |
| 02:34.08 | [TK]D-Fender | Especially "stuff". I specialise in that... |
| 02:34.32 | gnudna | so you do asterisk full time? |
| 02:34.33 | [TK]D-Fender | It's in the wind now that GV's days are numbered... |
| 02:34.47 | [TK]D-Fender | No, it only looks like I do :) |
| 02:34.54 | gnudna | yeah i heard reports it stopped working since hangouts got released |
| 02:35.10 | gnudna | well its good look for you ;) |
| 02:35.32 | [TK]D-Fender | I've been in here for nearly a decade, I do consult on the side but don't really work in the field besides nominal side contracts. |
| 02:35.46 | [TK]D-Fender | I've got a general IT day-job and budding music career |
| 02:36.00 | gnudna | i figured the music due to the name |
| 02:36.09 | [TK]D-Fender | Curiously non-related |
| 02:36.18 | gnudna | ;) |
| 02:36.30 | gnudna | i figured fender guitar |
| 02:36.36 | gnudna | well it works |
| 02:36.53 | [TK]D-Fender | Nope, I play Jackson & Epiphone... |
| 02:37.03 | gnudna | i got a epiphone myself |
| 02:37.07 | [TK]D-Fender | Technically Jackson is owned by FMC... |
| 02:37.10 | gnudna | one day i will be able to play it |
| 02:37.11 | [TK]D-Fender | but that's just being picky |
| 02:37.25 | *** join/#asterisk italoros_ (~italoross@187.61.168.117) |
| 02:42.44 | gnudna | btw i keep getting this every litte while |
| 02:42.55 | gnudna | May 27 22:39:51] NOTICE[10069]: chan_sip.c:20841 handle_response_peerpoke: Peer 'voipms' is now Lagged. (2041ms / 2000ms) |
| 02:42.56 | gnudna | [May 27 22:40:01] NOTICE[10069]: chan_sip.c:20841 handle_response_peerpoke: Peer 'voipms' is now Reachable. (32ms / 2000ms) |
| 02:43.34 | gnudna | is this my end or voip.ms? |
| 02:43.59 | [TK]D-Fender | yes :) |
| 02:45.41 | gnudna | weird i could not ping them properly yesterday but today it working almost as if there was a delay |
| 02:45.57 | gnudna | no issues to google or anybody else |
| 02:46.04 | [TK]D-Fender | could be issues anywhere along the path. |
| 02:46.25 | gnudna | agreed |
| 02:46.38 | gnudna | my isp has been a little iffy lately |
| 02:48.32 | gnudna | brb smoke break |
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| 03:32.55 | linocisco | hi all |
| 03:33.00 | linocisco | i have very simple question |
| 03:33.03 | linocisco | can I ask? |
| 03:33.11 | [TK]D-Fender | Don't ask to ask |
| 03:33.25 | WIMPy | thinks that WAS the question. |
| 03:33.34 | [TK]D-Fender | that WAS the answer :) |
| 03:34.32 | linocisco | ok. I have configure asterisk on QNAP TS-269 Pro asterisk ver 1.4 and cisco Phones SPA502G and 7942G. they are all working fine. audio quality is really 5 x 5(loud and clear). I dont know why !!. too good to believe to be true |
| 03:35.15 | WIMPy | 1.4??? |
| 03:35.33 | [TK]D-Fender | WIMPy: He's running a crappy port on a junk appliance |
| 03:35.59 | [TK]D-Fender | linocisco: Ok ... so where's the question? |
| 03:36.29 | linocisco | [TK]D-Fender, i dont know how to make it lower quality . :D |
| 03:36.55 | WIMPy | Install a bittorrent client. |
| 03:37.11 | linocisco | WIMPy, at the time of installation, it is with 1.4. now I can upgrade to 1.8 at least with its asteriskNow GUI builtin |
| 03:37.30 | linocisco | WIMPy, i m now thinking to install Dropbx on it |
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| 03:38.21 | linocisco | WIMPy, bittorent can make audio quality poor? |
| 03:38.48 | WIMPy | If you have enough traffic, that works extremely well. |
| 03:39.35 | gnudna | i would have though switching from u/a=law would do the trick |
| 03:39.36 | [TK]D-Fender | linocisco: Do you have even a basic understanding of CPU load & networlk load? Then if you wanted to MAKE it bad... then you should already know. |
| 03:39.38 | gnudna | ;) |
| 03:39.41 | WIMPy | Unless you know tc, which most people don;t seem to do for whatever reason. |
| 03:40.11 | gnudna | = should have been a - |
| 03:40.22 | linocisco | WIMPy, what is tc? |
| 03:40.34 | WIMPy | man tc |
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| 03:40.58 | linocisco | WIMPy, no linux box around. ok. let it be |
| 03:41.39 | WIMPy | You don't want to live without it. Especially not when doing VOIP. |
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| 03:49.30 | linocisco | btw, anybody have tried GSM or WCDMA usbdongle with asterisk instead of buying GSM gateway PCI cards or devices ? |
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| 04:11.38 | gnudna | goodnight guys |
| 04:11.40 | gnudna | im off |
| 04:11.54 | gnudna | [TK]D-Fender, thanks for all the help ;) |
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| 06:13.02 | gdeeble | Is there a way to have a menu read a variable and then immediately push it to the Dial Application? It's not actually going to a phone but an internal extension that's executing a script/going to voicemail menu, maybe a phone extension from time to time but normally not. The path is: Menu -> SubMenu -> Read -> Dial . Is there a way to do this or am I looking at changing things around. |
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| 06:16.42 | kaldemar | gdeeble: what have you tried to do so far? |
| 06:18.36 | ChannelZ | Keep in mind that you can use separate contexts to build IVRs, and the extensions within them can be the options without having to use Read and build the logic yourself |
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| 06:21.48 | bulkorok | hi |
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| 07:28.02 | mirela666 | Good Morning :) |
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| 08:05.24 | *** join/#asterisk bash_noob (~User@cpc25-lewi14-2-0-cust701.2-4.cable.virginmedia.com) |
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| 08:09.05 | bash_noob | Integrating with home automation in future. Need POTS, SIP and h264 video at minimum. before I commit is there a competing asterisk alternative? |
| 08:09.05 | bash_noob | http://www.voipon.co.uk/grandstream-gxe-5024-p-899.html?gclid=CLKr5_epuLcCFbLMtAod5iEAmQ |
| 08:10.37 | *** join/#asterisk Freeaqingme (~Freeaqing@91.214.168.110) |
| 08:15.18 | bulkorok | bash_noob: freeswitch |
| 08:17.01 | bash_noob | bulkorok: looks rack mountable. . .exactly what I was looking for. . .I need a read a little into this. Do you have any models in mind? |
| 08:20.59 | bash_noob | bulkorok: ok I meant the cudatel. I want to buy the hardware. |
| 08:29.37 | *** join/#asterisk jsjc (~Adium@151.Red-83-39-24.dynamicIP.rima-tde.net) |
| 08:31.38 | apb1963 | bash_noob: YATE |
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| 08:33.14 | bash_noob | apb1963: looking into YATE now as cudatel not available in UK:( looks good that thing. |
| 08:34.40 | bash_noob | apb1963: Yet Another Telephone Exchange? |
| 08:39.55 | apb1963 | Yet Another Telephony Engine |
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| 09:07.33 | linocisco | side car= trishaw in my country |
| 09:08.32 | *** join/#asterisk xoveruk (~user@196.200.85.77) |
| 09:08.36 | xoveruk | hi |
| 09:08.51 | xoveruk | Is it possible to increase the jitter buffer size for asterisk? |
| 09:09.30 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 09:12.45 | *** join/#asterisk giany (~giany@shifu.x83.org) |
| 09:13.03 | giany | 000302252 has 0 calls (max unlimited) in 'rrmemory' strategy (134s holdtime, 512s talktime), W:0, C:201, A:36, SL:0.0% within 0s |
| 09:13.25 | giany | that 134s holdtime is for one day or for Queue lifetime(last time asterisk was restarted?) |
| 09:16.10 | kaldemar | xoveruk: which buffer? |
| 09:16.20 | xoveruk | jitter buffer |
| 09:16.58 | kaldemar | xoveruk: ... which jitter buffer and on what version of asterisk? |
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| 09:17.19 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 09:17.28 | xoveruk | 1.6.0.26 |
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| 09:33.07 | Freeaqingme | What's the best way to give a group of phones one common number (freepbx calls it a ring group)? |
| 09:33.12 | Freeaqingme | Should I best use a queue? |
| 09:35.38 | kaldemar | Freeaqingme: exten => 123,1,Dial(Tech/a&Tech/b&...) |
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| 09:39.25 | Freeaqingme | kaldemar, thanks |
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| 09:39.34 | phpboy | hi there, I have a audio problem which is very difficult to pin point because they server is quite busy. I'm hoping for a couple of tips maybe. in around 5% of all calls which is about 30,000 a day the audio becomes one way. what's weird though is if you take a listen to the recording you can hear both parties voices but they can't hear each other. seems mostly like the agent can't here the caller any more |
| 09:39.40 | Freeaqingme | kaldemar, why would you propose this, rather than a queue. Because it's simpler? |
| 09:40.19 | kaldemar | Freeaqingme: if you don't need what a queue offers, then that is the simplest. |
| 09:40.27 | Freeaqingme | kk, tnx |
| 09:41.01 | As001 | Hello I just installed Asterisk 11.4 hoping that web rtc might work with Firefox 22 (using sipml demo) but I reseive this warning "WARNING[24136][C-00000003]: chan_sip.c:10420 process_sdp: Rejecting secure audio stream without encryption details: audio 43420 UDP/TLS/RTP/SAVPF 109 0 8 101" Does anyone succeed to do test call to asterisk using sipml demo at http://sipml5.org/call.htm?svn=179 ? |
| 09:41.16 | *** join/#asterisk BorjaGVO (d51beb92@gateway/web/freenode/ip.213.27.235.146) |
| 09:41.47 | As001 | I can do it from Google Chrome without problems but Firefox seems to use other encryption. |
| 09:43.09 | BorjaGVO | Hi all. I'm using eventfilter in manager.conf but it seems like you are only able to filter events by Event (i.e. QueueMemberStatus). I would like to filter all QueueMemberStatus events but the ones that have a certain string in Location: field. Is this possible? |
| 09:43.20 | As001 | I can register to server but when I call extension (echotest) I get above warning. |
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| 10:05.36 | vedic | I am getting this error as I try to load meetme in modules.conf: *** Failed to load module app_meetme.so - Required |
| 10:07.47 | Chainsaw | vedic: Did you just upgrade from 1.6 to 1.8, 10 or 11? |
| 10:08.43 | tparcina | Incoming call rings extension that is entered in the Asterisk database. |
| 10:09.55 | tparcina | Since we have more extensions, and we don't use agents, I have to rotate those extensions (that are entered in database) every day. |
| 10:10.28 | tparcina | Does anyone have suggestion what is the best way to do it? |
| 10:10.47 | tparcina | I figured out two options: |
| 10:11.33 | tparcina | 1. Cron sends single command to asterisk (dials extension) and in dialplan I do the rotating job. |
| 10:12.02 | tparcina | 2. Cron sends all commands to Asterisk and changes DB values directly. |
| 10:12.39 | tparcina | If anyone has any other suggestion I'll be happy to hear it. :) |
| 10:13.46 | vedic | Chainsaw: I have done fresh install of * 11.4.0 |
| 10:14.39 | Chainsaw | vedic: Asterisk 11 does not have a "meetme" application. |
| 10:14.46 | Chainsaw | vedic: You should convert to "ConfBridge" instead. |
| 10:15.46 | Chainsaw | vedic: To have "meetme" even mentioned in the config suggests it was for 1.6 or older. |
| 10:16.43 | vedic | Chainsaw: It does have meetme. I can see in menu select and also the .so file |
| 10:16.56 | vedic | How to check if a module is loaded or not? |
| 10:17.24 | Chainsaw | vedic: module show |
| 10:17.51 | Chainsaw | vedic: The meetme file will have been left behind by an older Asterisk installation. |
| 10:17.52 | *** join/#asterisk din3sh (2988f997@gateway/web/freenode/ip.41.136.249.151) |
| 10:17.57 | linocisco | hi all |
| 10:18.04 | Chainsaw | linocisco: Good morning. |
| 10:18.32 | linocisco | a friend of mine has bought E1 card to be used on asterisk server. he said 100 concurrent calls can be made per second |
| 10:18.48 | vedic | Chainsaw: I think I have loaded it. |
| 10:19.10 | linocisco | i m very new to E1 or T1 lines because my country none of them. With E1, how many phone no. can we get? |
| 10:19.22 | Chainsaw | vedic: You haven't, because 11 will only offer you app_confbridge.so |
| 10:20.25 | aberrios | linocisco, on one E1/ISDN30/PRI interface you can make 30 calls. as for DDIs on the circuit thats down to the provider, but I see no reason to have a limit on DDIs |
| 10:21.03 | vedic | Chainsaw: i can see it showing details when I write: core show application MeetMe. also it appears in "module show". And you can also enable it in "make menuselect" . I haven't read anywhere that meetme has been removed |
| 10:21.25 | linocisco | aberrios, 30 calls means 30 phone No.s ? |
| 10:22.07 | Chainsaw | linocisco: You can have more than 30 phone numbers. I have 80 numbers pointing at 3 ISDN BRI interfaces, which means 6 concurrent calls. |
| 10:22.24 | aberrios | linocisco, 30 calls means 30 conversations. As for phone numbers on the circuit that is down to the telco. I know BT usually give you a range of 10 DDIs with a PRI contract. |
| 10:22.48 | Chainsaw | linocisco: The number of phone numbers you can be assigned and the number of phone *calls* you can make at the same time are unrelated. |
| 10:23.48 | vedic | linocisco: with E1, you get 30 channels and 2 control channels. Effectively you can use 30 phone lines concurrently. Usually Telecom operator provides 1 pilot number and 29+ DID numbers. |
| 10:24.42 | *** join/#asterisk darkskiez (~mhb@fsf/member/darkskiez) |
| 10:25.25 | linocisco | <PROTECTED> |
| 10:26.11 | Chainsaw | linocisco: Yes, life is a bit different in the digital world. |
| 10:28.44 | linocisco | Chainsaw, is there any link i can refer about E1 and T1 lines and how many no. i can get according to them |
| 10:28.46 | linocisco | ? |
| 10:28.53 | *** join/#asterisk malmut (~malmut@176.104.148.155) |
| 10:29.03 | Chainsaw | linocisco: You may find this article a little helpful: http://en.wikipedia.org/wiki/Primary_Rate_Interface |
| 10:29.27 | Chainsaw | linocisco: But generally, the sky is the limit. If you want 200 numbers pointed at an E1... your telco can do that. |
| 10:31.06 | linocisco | Chainsaw, thanks let me read |
| 10:34.23 | msaraiva | Chainsaw: Asterisk 11 does have MeetMe. ConfBridge is not a direct replacement for meetme, but it works quite good for 90% of the cases. |
| 10:35.06 | msaraiva | There are still features missing, like the start and endtime functionalities. |
| 10:38.30 | *** part/#asterisk As001 (~uros@82.117.198.142) |
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| 10:54.57 | din3sh | linocisco: you need to understand the difference between an analogue line and digital one |
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| 11:15.29 | BorjaGVO | In manager.conf.sample appears ";eventfilter=!Channel: DAHDI*". I interpret asterisk (*) as wildcard, while in comments it says that regex are used. Can anyone clarify this? |
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| 11:22.48 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 11:23.48 | kaldemar | BorjaGVO: DAHDI* will do as a wildcard but .* for example works too... |
| 11:25.02 | BorjaGVO | kaldemar: alright. And what about this: eventfilter=!Location: Local\/[237]8.* ? |
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| 11:31.04 | BorjaGVO | kaldemar: flooking for example for this: Location: Local/382@from-queue/n |
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| 11:37.00 | kaldemar | BorjaGVO: you try it. |
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| 12:40.27 | Freeaqingme | I'm going to have a very long dial() statement. Is it possible to put some line endings in there? |
| 12:41.16 | WIMPy | Buid a variable? |
| 12:41.32 | Freeaqingme | hadn't thought of that. tnx |
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| 13:23.07 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.6.195) |
| 13:23.29 | FireAndIce | Hi all!! |
| 13:23.53 | FireAndIce | Can anyone help understand this, http://pastebin.centos.org/2524/ ? |
| 13:24.28 | *** join/#asterisk robl^ (~robl@pdpc/supporter/active/robl) |
| 13:25.24 | FireAndIce | I'm unable to register to asterisk through a client api, doubango ngn stack. I just want to understand what the sip debug message means. |
| 13:26.08 | [TK]D-Fender | looks like a mangled packet. Your API is probably messed up |
| 13:27.10 | FireAndIce | Ok. However, it works with the test code provided with api. |
| 13:27.21 | FireAndIce | I dont understand where am I going wrong. |
| 13:27.33 | *** join/#asterisk skrusty (~support@168.63.14.171) |
| 13:28.04 | FireAndIce | [TK]D-Fender, thanks anyway. |
| 13:29.01 | *** join/#asterisk afink (4172ce22@gateway/web/freenode/ip.65.114.206.34) |
| 13:29.09 | [TK]D-Fender | If you're passing it a collected variable make sure it's initialized and proper.... |
| 13:29.18 | [TK]D-Fender | Could be a junk reference in your code to it |
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| 13:30.20 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 13:42.01 | FireAndIce | [TK]D-Fender, ok, I'll check it out. thanks. |
| 13:45.23 | *** join/#asterisk polysics (~Adium@host96-50-dynamic.21-79-r.retail.telecomitalia.it) |
| 13:45.29 | polysics | hi there |
| 13:45.47 | *** join/#asterisk zendel (zendel.fer@124.43.125.184) |
| 13:46.34 | polysics | someone mentioned having CONNECTEDLINE working with a peer in a ConfBridge |
| 13:46.43 | polysics | could I please see their working code? :-D |
| 13:48.28 | WIMPy | I posted that line. |
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| 13:49.01 | polysics | oh, I know, I jsut can't find the log |
| 13:49.20 | polysics | what do you set and with which payload? I would like to try with something I know works |
| 13:49.32 | WIMPy | mom |
| 13:50.11 | WIMPy | dprintf(amisock, "action: setvar\r\nchannel: %s\r\nvariable: CONNECTEDLINE(name)\r\nValue: test count=%d\r\n\r\n", a_chan, ++aoccurr); |
| 13:50.21 | WIMPy | That's what I used for testing. |
| 13:50.35 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 13:50.57 | WIMPy | And that's what I used for (un)muted events: |
| 13:51.01 | WIMPy | dprintf(amisock, "action: setvar\r\nchannel: %s\r\nvariable: CONNECTEDLINE(name)\r\nValue: %s\r\n\r\n", a_chan, muted?"muted":"talking"); |
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| 13:51.38 | polysics | that would translate to setting CONNECTEDLINE(name) to "muted", right? |
| 13:51.42 | polysics | a simple string |
| 13:51.49 | WIMPy | yes |
| 13:52.45 | WIMPy | CONNECTEDLINE ist not really a propper place for that information, but probably the one most likely to work. |
| 13:53.31 | polysics | then I think it's the phone's fault |
| 13:53.42 | polysics | I tried setting it to "testing" to no avail |
| 13:53.44 | WIMPy | sip debug will tell you. |
| 13:53.45 | polysics | thanks anyway |
| 13:54.23 | polysics | I think the phone doesn't send an "unsupported" packet but it might be worth checking |
| 13:55.04 | Greenlight | You can see what's being *sent* to the phone though |
| 13:55.17 | Greenlight | I did say last week that it appeared my softphone didn't support it |
| 13:56.04 | polysics | and it really didn't? |
| 13:56.54 | WIMPy | Even though I haven't seen a good SIP phone, yet, they don't have to be completely dysfunctional :-) |
| 13:57.06 | Greenlight | That's what it looked like |
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| 14:14.11 | FireAndIce | [TK]D-Fender, I'm able to register through the api. Just restarted my server. |
| 14:14.19 | *** join/#asterisk serafie (~erin@user-24-214-173-250.knology.net) |
| 14:14.43 | [TK]D-Fender | Soulds like Windows Admin 101 |
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| 14:18.21 | FireAndIce | Iol.. |
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| 14:18.24 | FireAndIce | lol.. |
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| 14:27.56 | Kobaz | anyone have any problems with the polycom 3.3.5 firmware with the idle browser crapping out and people getting call dropouts? |
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| 14:38.21 | Kobaz | is everyone still sleeping from memorial day weekend |
| 14:40.04 | robl^ | shhh! some of them are hungover -- too much "celebrating" |
| 14:40.17 | igcewieling | looks at robl^ |
| 14:40.32 | Kobaz | hahaha.... reading this bit on detecting packet loss |
| 14:40.43 | Kobaz | However, the Grandstream GXP-2000 (version 1.1.0.14) we have in our test bed turned out to be an invaluable tool since it has the uncanny ability to turn the 1/50 second gap caused by a single lost packet into a multi-second garbled mess. So, if like me, you enjoy testing for dropped packets, I highly recommend adding the Grandstream to your toolbox - it is a great tool and worth far more than the retail price. |
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| 14:42.51 | *** join/#asterisk miRobolant (~miRe@9-voip-telecom.10-cust.tasfrance.com) |
| 14:42.56 | miRobolant | Hi there ! |
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| 14:47.27 | Chainsaw | Kobaz: The GXP-2000 packet loss detector, now with phone function! |
| 14:47.44 | Chainsaw | Kobaz: Seriously though. You can pry my Polycoms from my cold, dead hands. |
| 14:47.51 | Kobaz | hah |
| 14:47.51 | Kobaz | yeah |
| 14:47.57 | Kobaz | except for when they have problems |
| 14:48.12 | Kobaz | early 4.0 firmware was b.a.d. |
| 14:48.18 | Chainsaw | It was, agreed. |
| 14:48.29 | Chainsaw | I'm not sure when it became good. At some point I tried it and all was well. |
| 14:48.51 | Kobaz | i'm on 3.3.5 for most sites |
| 14:48.57 | Chainsaw | (It got confused between UDP & TCP when it came to SIP, at least that's what it looked like on the Asterisk side) |
| 14:49.04 | aberrios | went from using 3.2.3 and 3.2.4 straight to UC 4.1.4 |
| 14:49.09 | Chainsaw | checks |
| 14:49.11 | Chainsaw | 4.0.3.7562 apparently. |
| 14:49.16 | Kobaz | 3.2 was nice and borken |
| 14:49.29 | Kobaz | phone returns 200 ringing |
| 14:49.34 | Kobaz | but the phone isnt actually ringing |
| 14:49.57 | Kobaz | happened on one out of every 10ish calls |
| 14:50.05 | Chainsaw | Oh hey. 4.0.4 is out. |
| 14:50.07 | Chainsaw | Have you tried that? |
| 14:50.11 | Kobaz | nope |
| 14:51.20 | miRobolant | Hi, I would like to know if you can help me about a problem I try to resolve for weeks. |
| 14:51.44 | miRobolant | Asterisk 1.6.2.20 |
| 14:51.50 | miRobolant | <PROTECTED> |
| 14:51.50 | miRobolant | <PROTECTED> |
| 14:51.50 | miRobolant | [2013-05-28 16:49:46] WARNING[27016]: chan_sip.c:18480 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog '7ab616882bee0b874ce9d2543e60331d@X.X.X.X'. Giving up. |
| 14:51.50 | miRobolant | <PROTECTED> |
| 14:52.52 | Kobaz | you'll need some sip debugs |
| 14:53.05 | miRobolant | As ? |
| 14:53.18 | Kobaz | as in sip set debug on |
| 14:54.53 | Kobaz | and don't remove anything except for things you want private like passwords. don't completely xxxx out ip addresses... if you want to mask, remove the first part ie: x.80.3.20 |
| 14:55.52 | *** join/#asterisk Rumbles (~Rumbles@host81-149-239-223.in-addr.btopenworld.com) |
| 14:56.45 | miRobolant | Okay, I'm on it |
| 14:59.09 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 15:00.18 | *** join/#asterisk mihamina (~mihamina@41.188.46.121) |
| 15:01.05 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 15:01.05 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 15:08.03 | *** join/#asterisk Linkforsoad (~Linkforso@D9799130.cm-3-2c.dynamic.ziggo.nl) |
| 15:08.09 | Kobaz | what cable tester is it that can do like, signal testing.. verify that the run is not only conencted but can actually pass data properly |
| 15:09.06 | WIMPy | Sounds like network tester vs cable tester. |
| 15:09.26 | robl^ | higher end Fluke Cable/Network Certification Testers |
| 15:09.41 | miRobolant | Actually, the peer is connecting via Internet |
| 15:12.25 | miRobolant | <-------------> |
| 15:12.25 | miRobolant | [KPARISAST2*CLI> |
| 15:12.25 | miRobolant | [0K--- (8 headers 0 lines) --- |
| 15:12.25 | miRobolant | [KPARISAST2*CLI> |
| 15:12.25 | miRobolant | [0K[2013-05-28 17:01:33] WARNING[0m[27016]: mchan_sip.c[0m:m18480[0m mhandle_response_invite[0m: Re-invite to non-existing call leg on other UA. SIP dialog '2597885620067f9a27e2c87b57653533@X.1.1.113'. Giving up. |
| 15:13.45 | ChannelZ | snickers |
| 15:19.46 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 15:19.46 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:22.23 | [TK]D-Fender | Mars |
| 15:24.41 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.170) |
| 15:25.28 | robl^ | M&Ms |
| 15:28.15 | *** join/#asterisk navaismo (~navaismo@189.241.9.57) |
| 15:32.32 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
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| 15:43.07 | BeeBuu | anyone help me please? I'm using chan_ss7 now. Is there anything like pridialplan in ISDN? |
| 15:46.00 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 15:46.00 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 15:47.54 | coppice | marathon (I'm old school) |
| 15:51.31 | slav3_kitten | hmm, i really need another cisco phone. |
| 15:51.48 | slav3_kitten | so it matches the rest of the phones in the house |
| 15:52.03 | saint_ | slav3_kitten: you put cisco in your house ? |
| 15:52.32 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 15:53.52 | BeeBuu | I'm using chan_ss7 now. Is there anything come with ss7.conf like pridialplan in ISDN config? |
| 15:54.08 | slav3_kitten | saint_, i also put a 42 unit rack, 48 port patch panel, 12 port catv patch panel, 2x 2u UPSs, 2x cisco 24 port switches, though thou i'm going to put a 24 port hp gige managed switch in there, 2 2U servers, 1 1U pf sense router, PoE access points in each end of the house... |
| 15:54.46 | slav3_kitten | infrastructure, i got that. |
| 15:54.47 | *** join/#asterisk Rumbles (~Rumbles@host81-149-239-223.in-addr.btopenworld.com) |
| 15:55.46 | igcewieling | BeeBuu: almost nobody here uses chan_ss7. have you searched the asterisk ss7 mailing lst? see lists.digium.com |
| 15:55.54 | saint_ | slav3_kitten: sweet. I have alcatel-lucent omniswitch routers, 1Gb network over Cat 6, and digium / yealink phones |
| 15:56.03 | slav3_kitten | nice :) |
| 15:56.09 | saint_ | all poe too |
| 15:56.16 | saint_ | I just dumped my vonage accounts |
| 15:56.20 | saint_ | switched to VoIP.MS |
| 15:56.23 | slav3_kitten | i had got some cisco 7900 phones really cheap... i'll never do them again |
| 15:56.41 | slav3_kitten | i'm voip.ms for local stuff and i use flowroute for my international calling |
| 15:56.47 | saint_ | I have 1 here actually which is pretty cool.. Alcatel-Lucent My IC Phone.. Totally touch screen. |
| 15:57.05 | saint_ | check this out: http://www.wirelessgoodness.com/wp-content/uploads/2011/02/image102.png |
| 15:57.10 | BeeBuu | igcewieling: thanks |
| 15:57.11 | gorkish | i just got one of the polycom vvx600 -- it is really great |
| 15:57.11 | saint_ | awesome |
| 15:57.24 | slav3_kitten | yea if i was going again, i'd go polycom |
| 15:57.35 | slav3_kitten | way more documentation than cisco bullshit |
| 15:57.38 | saint_ | for what part ? |
| 15:57.50 | slav3_kitten | that phone is stupid large |
| 15:57.51 | saint_ | regular phones, or conference ones ? |
| 15:57.57 | slav3_kitten | regular phones |
| 15:58.03 | gorkish | touchscreen, video calling option with a camera you can move around as needed, plus it has a webkit browser so you can do some really great applications easily |
| 15:58.04 | saint_ | slav3_kitten: it's awesome on my desk. |
| 15:58.15 | slav3_kitten | saint_, my desk isn't that big |
| 15:58.24 | saint_ | slav3_kitten: you must not be that important, lol |
| 15:58.26 | slav3_kitten | the thing looks larger than my 12" laptop |
| 15:58.37 | slav3_kitten | i'm not important lol |
| 15:58.38 | gorkish | the webkit version on it supports websockets so i have a realtime display of all extensions in the office on the screen at all times |
| 15:59.09 | igcewieling | gorkish: in our experience even with nothing special VVX phones randomly reboot |
| 15:59.44 | saint_ | slav3_kitten: i got the new Yaealink for sip over wlan (it's not really sip over wlan) they are awesome too |
| 15:59.49 | saint_ | with a color display.. |
| 15:59.55 | gorkish | igcewieling, this is the first one i have gotten to test. rest are spip 650 but considering slowly rolling to vvx when they get worked out. i demoed a vvx1500 but hated it |
| 16:00.17 | saint_ | slav3_kitten: http://www.yealink.com/product_info.aspx?ProductsCateID=308&CateId=307&BaseInfoCateId=308&Cate_Id=308&parentcateid=307 |
| 16:00.18 | gorkish | the 500/600 seem much more solid |
| 16:00.28 | gorkish | 600 espeically |
| 16:00.33 | Chainsaw | It's all 670 here. |
| 16:00.39 | slav3_kitten | i'm kinda like ... wired all the things |
| 16:00.41 | gorkish | plus bluetooth is built in for headsets which is huge |
| 16:00.44 | slav3_kitten | i really dislike wireless |
| 16:00.59 | [TK]D-Fender | gorkish: the spec look really nice and the price-point is surprisingly good |
| 16:00.59 | saint_ | slav3_kitten: agree, but it's cool to have a couple in the house .. |
| 16:01.13 | gorkish | spip670 is deprecated. they are advancing the 650 but 670 is EOL; they are really screwing 670 adopters |
| 16:02.01 | Chainsaw | I use them for calls. They seem quite good at that. |
| 16:02.02 | slav3_kitten | yea i can understand that thinking |
| 16:02.05 | gorkish | [TK]D-Fender, yes i recommend you get one to test out, especially if you are interested in developing local applications or having video as an easy option |
| 16:02.31 | *** join/#asterisk FLeiXiuS (~FLeiXiuS@unaffiliated/fleixius) |
| 16:04.17 | [TK]D-Fender | Seems you need to buy the camera separate |
| 16:04.58 | igcewieling | I hate the VVX's UI. |
| 16:05.01 | gorkish | You buy the camera separate but it can be hot plugged so you can share them around if you dont need it all the time |
| 16:06.12 | gorkish | ui occasionally slugs and could stand improvement especially when juggling multiple calls |
| 16:07.10 | [TK]D-Fender | http://www.ipphone-warehouse.com/Polycom-VVX-s/774.htm |
| 16:07.31 | [TK]D-Fender | This is surprisingly affordable all things considered |
| 16:09.30 | igcewieling | Sales was all excited when the VVX500 came out. We stopped deploying them after a single customer. |
| 16:14.03 | *** join/#asterisk YoMomma (~YoMomma@71-95-164-90.static.mtpk.ca.charter.com) |
| 16:32.27 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
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| 16:43.17 | *** part/#asterisk mihamina (~mihamina@41.188.46.121) |
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| 16:46.36 | leifmadsen | igcewieling: +100000 |
| 16:46.49 | leifmadsen | igcewieling: we've had tons of issues with VVX{500,600} |
| 16:46.52 | igcewieling | leifmadsen: the VVXs? |
| 16:46.55 | igcewieling | ah |
| 16:46.55 | leifmadsen | yep |
| 16:47.01 | leifmadsen | mostly, phantom calls |
| 16:47.10 | leifmadsen | just places calls randomly without touching the device |
| 16:47.15 | leifmadsen | had to RMA a bunch of devices |
| 16:47.26 | igcewieling | heh, ours usually reboot in the middle of the call |
| 16:47.27 | leifmadsen | Polycom sent a rep on site... I think the VVX deploy has been a nightmare fo rthem |
| 16:47.43 | igcewieling | leifmadsen: maybe they should stop releasing more models |
| 16:47.46 | leifmadsen | igcewieling: nice :) also the phone would spam the system with directory updates like woah |
| 16:48.11 | igcewieling | leifmadsen: ah, we also had lockups when monitoring large numbers of buddies |
| 16:48.31 | leifmadsen | good times... |
| 16:51.03 | gorkish | yeah im not trying to shill for them other than that the 600 i have works ok; i would not immediately suggest it for a 100+ phone deployment without a lot more testing but it has the promise of being a solid product if they can get the kinks worked out |
| 16:52.05 | gorkish | i have only had this one a couple weeks |
| 16:53.39 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.186) |
| 17:01.35 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:221:6aff:feb8:e0b2) |
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| 17:17.54 | Zipper_32 | I've just setup a new asterisk box after migrating from the Zap analog system. I've setup entries in /etc/dahdi/system.conf and /etc/asterisk/chan_dahdi.conf; additionally, the status of Dahdi shows all my FXO ports, but I cannot see incoming calls in my asterisk console. I was hoping somebody might be able to point me in the right direction to figure out where my incoming calls are currently going? |
| 17:19.03 | igcewieling | look at the output of /proc/dahdi/1 if you have RED in ports you have lines plugged into then there is a wiring issue |
| 17:22.09 | Zipper_32 | Thanks; so far there is one red, which is the unplugged port. The rest are just showing inactive. |
| 17:23.29 | igcewieling | does "dahdi show channels" show the channels? |
| 17:26.14 | Zipper_32 | I'm assuming that is through the asterisk console? Then no, I don't even have the option to auto-complete the "dahdi" command |
| 17:30.14 | Zipper_32 | i can see the channels when I do dahdi_scan from the cli, but nothing through the asterisk console. |
| 17:30.47 | igcewieling | then chan_dahdi.so was not build or is not loaded |
| 17:31.00 | Zipper_32 | Perhaps I need to do another compile of the source? I did make some hpec changes after compiling, but I didn't think that would do it. |
| 17:31.06 | Zipper_32 | I'll investigate with that. Thank you very much! |
| 17:35.14 | igcewieling | you MUST rebuild Asterisk when installing DAHDI |
| 17:35.26 | igcewieling | go into make menuconfig make sure chan_dahdi is enabled |
| 17:38.38 | Zipper_32 | igcewieling, Nailed it. Thanks for the help. |
| 17:43.01 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
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| 17:52.49 | *** join/#asterisk midori-rus (~symbaba@85.26.164.114) |
| 17:53.09 | midori-rus | hello, any speack russian? |
| 17:53.44 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.171) |
| 17:54.15 | Qwell | midori-rus: Have you tried #asterisk-ru ? |
| 17:55.27 | *** join/#asterisk tonyclewis (uid6025@gateway/web/irccloud.com/x-inkpdbwyonbujezi) |
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| 18:02.05 | raidghost | Do i need DahDi when installing asterisk, if i dont use isdncards or digium cards? |
| 18:04.03 | Qwell | raidghost: no |
| 18:08.15 | [TK]D-Fender | Qwell: newer * doesn't need it for even Meetme/IAX trunking anymore does it? |
| 18:08.53 | Qwell | it's needed for meetme |
| 18:12.55 | drmessano | I hope one day someone will look back on the days meetme and the awful concept of using Zaptel/DAHDI for mixing, and wonders what the hell someone was thinking |
| 18:13.08 | Qwell | one day? |
| 18:13.16 | drmessano | One day when it's not so painful |
| 18:14.01 | igcewieling | drmessano: somehow I suspect using an existing audio mixing source for MeetMe instead of writing your own audio mixing stuff seemed like a good idea at the time |
| 18:18.07 | drmessano | igcewieling, no doubt.. |
| 18:23.49 | *** join/#asterisk Arman (~howlader@117.18.231.50) |
| 18:24.45 | leifmadsen | drmessano: linux timing didn't exist at the time |
| 18:25.01 | leifmadsen | kernel timing* |
| 18:30.52 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.124.171) |
| 18:33.08 | Arman | tzafrir, hi |
| 18:33.24 | tzafrir | hi |
| 18:34.28 | Arman | tzafrir, its been after long time who inspire me to knowledge share at irc |
| 18:34.44 | Arman | tzafrir, thanks to you |
| 18:36.00 | Arman | tzafrir, my another name is Adnan from Bangladesh, who deploy you xorcom first time at bangladesh xr1000 |
| 18:36.18 | tzafrir | What are you doing nowadays? |
| 18:37.29 | Arman | tzafrir, i am working at www.banglatracker.com as Network & System administrator |
| 18:38.54 | Arman | now development is become my hobby, for self and man kind. |
| 18:44.13 | Arman | tzafrir, by the way how are you sir? |
| 18:44.32 | tzafrir | great |
| 18:46.40 | Arman | tzafrir, i need some time your guide & tutorial, i hope you may help us. |
| 18:47.00 | tzafrir | What about? |
| 18:48.21 | Arman | i would like to develop a module for asterisk and contribute to community |
| 18:48.56 | Arman | tzafrir, not much special rather than websocket |
| 18:49.43 | Arman | tzafrir, any way this is my hobby |
| 18:52.14 | Arman | tzafrir, i will leave now as dark night here. as well also need to catch office today morning. thanks once again for inspire & kind cooperation. |
| 18:56.20 | *** join/#asterisk jameswf (~james@unaffiliated/jameswf-home) |
| 18:56.35 | *** join/#asterisk robert_ (~hellspawn@objectx/robert) |
| 18:56.52 | jameswf | should gain adjustments in chan_dahdi reflect in dahdi show channel X |
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| 18:58.48 | robert_ | so I'm in the process of converting asterisk to use mysql for a backend; however, can we use setvar somehow in conjunction with a mysql database somehow? |
| 18:58.57 | robert_ | (RudyValencia and I, that is) |
| 18:59.19 | igcewieling | robert_: 1) it won't be as easy as yo think it is and 2) yes. |
| 18:59.45 | robert_ | igcewieling: I didn't say I thought it would be easy :p |
| 18:59.54 | robert_ | I said "somehow" |
| 19:00.18 | igcewieling | <PROTECTED> |
| 19:00.43 | igcewieling | robert_: "1" was referring to use Realtime |
| 19:00.44 | *** join/#asterisk Nickinator (~Nickinato@123-243-142-239.static.tpgi.com.au) |
| 19:00.58 | robert_ | oh |
| 19:01.34 | igcewieling | 2) is easy |
| 19:04.05 | robert_ | oh, hm. |
| 19:05.30 | robert_ | useclientcode? accountcode? forgive me, I'm sort of stumbling into his. RudyValencia is more the ast person than I am, but he isn't available right now, and I'd like to get this finished. |
| 19:05.50 | robert_ | where do those go? |
| 19:05.53 | robert_ | lol |
| 19:06.25 | igcewieling | robert_: those are part of our sippeers database table |
| 19:06.31 | igcewieling | for realtime |
| 19:06.37 | robert_ | oh, okay |
| 19:06.45 | igcewieling | we don't actually use it, we have a script to read the table and generate sip.conf for us. |
| 19:06.52 | *** join/#asterisk MLNoah (~chatzilla@noc.metalink.net) |
| 19:07.04 | igcewieling | I pasted it so you could see how to do a setvar |
| 19:07.09 | robert_ | oh, okay. |
| 19:07.54 | MLNoah | i'm trying to set up hints for a Realtime-based system, set up exten => _X.,hint,${ODBC_GET_HINT(${EXTEN},${ODBC_GET_CUST_FROM_PEER(${CHANNEL(peername)})})} in my hint context. but i get a warning "unknown or unavailable item requested 'peername'" when the peer tries to subscribe |
| 19:08.11 | MLNoah | is CHANNEL(peername) not available on subscriptions? is there an alternative I can use? |
| 19:08.12 | robert_ | igcewieling: http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb < we're following this, only I'm doing half of the work. :p |
| 19:08.28 | jameswf | bueller? |
| 19:08.51 | igcewieling | robert_: ah. can't help |
| 19:09.23 | igcewieling | MLNoah: I doubt it works that way |
| 19:09.33 | igcewieling | peers and channels are "call" things |
| 19:09.48 | igcewieling | since you don't have a call up, I doubt you'll have a CHANNEL() |
| 19:09.56 | robert_ | igcewieling: oh, okay. I'm not ready for it "this very moment", I was just making sure somehow that we can set our PSTN_CID variable accordingly. |
| 19:10.02 | MLNoah | hm. so then I'd be down to separate hint contexts per customer |
| 19:10.24 | igcewieling | MLNoah: welcome to the world of Asterisk Hacks For Multitennant Systems. |
| 19:11.07 | MLNoah | which is about 3000% more fun when you try to add in asterisk clustering |
| 19:11.23 | igcewieling | I'd think you'll want separate contexts anyway to prevent one customer from subscribing to another customer's extensions |
| 19:11.41 | MLNoah | well, my thought was to have a database of <customer>,<hint extension>,<hint detail> |
| 19:11.51 | MLNoah | and then let ODBC go to town and figure out what the hint should be |
| 19:12.00 | MLNoah | but i suppose that wouldn't work anyway, would it. |
| 19:12.01 | MLNoah | dur. |
| 19:12.12 | robert_ | igcewieling: That amounts to, I'm pretty sure, Egon Spengler's definition of "bad." :p |
| 19:18.05 | igcewieling | Since Asterisk 1.6 and later, Asterisk has had significant performance improvements, making Kamailio less useful in many situations |
| 19:18.09 | MLNoah | and i realize a lot of my problem with this will go away once the system isn't under development any more... but is there any way to keep the ability to see "sip show peers" on realtime peers without the drawback of rtcachefriends making your settings "sticky" until you manually prune? |
| 19:19.05 | igcewieling | MLNoah: We moved all our peers into realtime, realized all those annoyances and wrote a script to generate the peers for sip.conf from the database data |
| 19:19.25 | MLNoah | yeah, i'm starting to lean that way myself |
| 19:19.48 | MLNoah | "can't wait" until i get to custom parking lots |
| 19:19.54 | igcewieling | Realtime is one of Asterisk's "bluebells" features. |
| 19:24.08 | MLNoah | ...sigh... I thought I saw articles suggesting you could, in fact, do hints on realtime peers... |
| 19:25.39 | robl^ | I store peers in a DB, but generate a sip.conf from the DB as needed. not quite a perfect solution, but works when you need to use hints |
| 19:25.59 | igcewieling | robl^: it is as close to a perfect solution as you are likely to get |
| 19:26.07 | MLNoah | which it does, as long as the retard setting up sip.conf remembers callcounter=yes |
| 19:26.07 | *** join/#asterisk imox (~imox@91-66-32-57-dynip.superkabel.de) |
| 19:26.15 | MLNoah | every single asterisk system i do... *headdesk* |
| 19:26.42 | MLNoah | thanks for the help, igcewieling |
| 19:27.33 | igcewieling | MLNoah: have you considered purchasing a multi-tennant Asterisk solution like Bicom's PBXware? |
| 19:27.57 | igcewieling | We use them for our "hosted" solution |
| 19:29.01 | *** join/#asterisk Kraln (~kraln@69.169.90.240) |
| 19:30.52 | jameswf | seems you can't define gains in an include... thats kinda stupid |
| 19:35.19 | *** join/#asterisk medve (~medve@92-249-193-115.pool.digikabel.hu) |
| 19:35.35 | MLNoah | my budget is $0 |
| 19:35.38 | *** part/#asterisk medve (~medve@92-249-193-115.pool.digikabel.hu) |
| 19:35.48 | MLNoah | and apparently that's what the boss thinks my time costs him too *shrug* |
| 19:36.05 | talntid | your choice to stay in an environment like that :) |
| 19:42.03 | *** join/#asterisk Tarso (~Tarso@189.61.52.46) |
| 19:45.31 | *** join/#asterisk thehar_ (~thehar@diddlebox.thehar.com) |
| 19:46.30 | *** join/#asterisk krapper (krapper@territory.krapper.net) |
| 19:47.42 | krapper | sip peers becoming unreachable when behind a DD-WRT router... anyone have experience with this issue? |
| 19:51.14 | igcewieling | krapper: qualify=yes |
| 19:51.45 | krapper | igcewieling, already there... never becomes unreachable on another router |
| 19:53.42 | *** join/#asterisk itgrl (~Adium@70.94.92.176) |
| 19:53.59 | *** part/#asterisk itgrl (~Adium@70.94.92.176) |
| 19:54.51 | drmessano | krapper, which firmware version? |
| 19:55.40 | *** join/#asterisk thebmw (~thebmw@i.am.thebmw.net) |
| 19:56.00 | krapper | drmessano, DD-WRT v24-sp2 (03/25/13) mega |
| 19:56.27 | drmessano | What model router? |
| 19:56.35 | krapper | linksys e900 |
| 19:57.33 | igcewieling | Customer bought a 4-port poe switch for his 5 phones |
| 19:57.45 | drmessano | I guess that's not supported by 14929. Will 15778 run on it? |
| 19:58.49 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:ccba:e252:2dc9:b90) |
| 19:59.10 | robl^ | igcewieling: a 4 port poe switch? I bet its one of those cheap trendnets. It will power 3 phones, you add a 4th and things start failing. |
| 19:59.40 | drmessano | Yeah, all 4 ports HAVE PoE, you just can't use them all at the same time |
| 20:00.18 | drmessano | Sounds like some of the others.. 24 ports PoE *for convenience*, but you can only really connect 16 phones |
| 20:00.35 | krapper | drmessano, i'm running 21061, only version supported by this router. :-/ |
| 20:01.42 | drmessano | 21676 is out.. I was going to suggest 21286 |
| 20:03.08 | drmessano | I would not run 21061, 21153, or 21223... They're too close together.. Gives me the impression he was trying to fix something seriously borked |
| 20:04.32 | krapper | my router on the dd-wrt site only shows 21061... can that be disregard and go with other versions? |
| 20:07.04 | drmessano | Where on the dd-wrt site did you find that info? Other than the recommended build of 14929 for older hardware, there is no router database and no official post for any newer hardware |
| 20:07.26 | krapper | http://www.dd-wrt.com/site/support/router-database ... type e900 |
| 20:07.28 | drmessano | I would suggest 21286. I have been using it for a couple weeks and its very stable |
| 20:07.38 | drmessano | The router database doesn't work |
| 20:09.25 | krapper | you just follow through forums? |
| 20:12.22 | drmessano | Somewhat.. I check the ftp site for new builds, check the forums for success/failures.. |
| 20:12.36 | drmessano | When a build sucks, I make note of it.. and I don't try it again |
| 20:12.38 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 20:12.39 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 20:14.00 | drmessano | dd-wrt is in a sort of limbo right now as far as recommended releases. 14929 is stable as hell.. as long as your hardware is supported. Beyond that you're kinda on your own. 14929 is the only *officially recommended* build |
| 20:14.04 | drmessano | It's a little ugh |
| 20:16.11 | krapper | i'm digging around ftp.dd-wrt.com, don't see those new builds? |
| 20:16.38 | drmessano | ftp://ftp.dd-wrt.com/others/eko/BrainSlayer-V24-preSP2/2013/ |
| 20:17.16 | krapper | deep in the directories :-D |
| 20:20.45 | krapper | drmessano, thx! |
| 20:43.41 | *** part/#asterisk YoMomma (~YoMomma@71-95-164-90.static.mtpk.ca.charter.com) |
| 20:46.16 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 20:51.35 | DBordello | Does anybody have any experience running Asterisk on a Raspbeery Pi? |
| 20:52.23 | navaismo | o/ |
| 20:52.55 | navaismo | that means me |
| 20:55.15 | DBordello | navaismo, care to share your experiences? :) |
| 20:55.32 | igcewieling | navaismo: sometimes it is prudent to not admit some of the things you know. |
| 20:57.14 | navaismo | DBordello, Well in the past i have worked with Alix Boards so using Raspberry pi was easier and little fast to do a native compile, about asterisk it works fine an so far 3 calls works well, my setup use apache+mysql for static realtime |
| 20:57.22 | navaismo | igcewieling, uh, why? |
| 20:57.39 | *** join/#asterisk philm (~phil@97.101.207.243) |
| 20:57.42 | DBordello | navaismo, great, I was hopig it was up to the task |
| 20:57.58 | navaismo | DBordello, there are som stuff yu cant run on it like g729, dpma |
| 20:58.19 | navaismo | and there are also imges with freepbx-asterisk or PIAF for raspberry pi |
| 20:58.44 | DBordello | aaah, very nice. |
| 20:58.58 | DBordello | I am looking forward to playing with it |
| 20:59.49 | igcewieling | navaismo: because then you become the expert everyone pesters for information since nobody else can help |
| 21:00.30 | igcewieling | same reason you don't imply you "work with computers" at a party. |
| 21:00.43 | navaismo | DBordello, I would recommend to install from sources, if you want faster compilation you can use sdcc, the I love my pi-pbx more than my pi-media center |
| 21:01.04 | navaismo | igcewieling, ah true, or with your family |
| 21:01.06 | DBordello | navaismo, install from source on what distribution? |
| 21:01.23 | navaismo | igcewieling, but im just telling my experience here |
| 21:01.54 | navaismo | DBordello, Well thats up to you, I use Fedora Remix but alot lot people prefer Raspbian |
| 21:02.09 | DBordello | navaismo, fair enough. |
| 21:02.21 | DBordello | To be honest, I have a raspberry pi sitting my drawer, looking for a use ;) |
| 21:02.25 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 21:03.21 | navaismo | I was suggesting to the freepbx people to make a module for the freepbx-pi when a call failed the pi give us a beer |
| 21:04.25 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.147) |
| 21:05.07 | navaismo | the rpi is like a swiss-knife, you can do a lot on it |
| 21:05.28 | drmessano | or a little.. depends on your need |
| 21:05.31 | robl^ | I'm using Asterisk on an Raspberry Pi -- but its dedicated, limited purpose. It's basically fail-over and delivers intercept messages. I don't use it as a PBX or conferencing server |
| 21:06.12 | drmessano | I have that's my "bench NAS" at home for transferring files from dead machines, and is also the wifi bridge to my bench |
| 21:06.19 | drmessano | have one* |
| 21:07.55 | drmessano | Rather limited use for such a complex little piece of equipment, but then again, what else could fill both those roles in a small package? |
| 21:08.15 | igcewieling | robl^ sort of has a "thing for phones". send them that link of all your phones, robl^ |
| 21:08.23 | drmessano | lol |
| 21:08.59 | robl^ | igcewieling: hush! I don't have that many in my lab |
| 21:09.34 | drmessano | I want to see |
| 21:09.40 | drmessano | !!!11!!!1!!!! |
| 21:09.58 | robl^ | its a mix of asterisk, nortel, and cisco |
| 21:10.03 | robl^ | ohh.. and sipxecs |
| 21:11.17 | talntid | nortel BCM400! |
| 21:11.21 | talntid | yayyyyy |
| 21:11.28 | talntid | (kill me now.) |
| 21:11.47 | robl^ | I have a BCM50 & CS1000 |
| 21:11.59 | talntid | want a BCM400? |
| 21:12.03 | talntid | just pay shippin'! |
| 21:12.04 | talntid | lol |
| 21:12.05 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 21:12.24 | robl^ | maybe if it were an R6 BCM450 ;-) |
| 21:12.36 | talntid | no idea what that is |
| 21:13.40 | robl^ | BCM450 was the latest and largerst model of the BCMs. R6.0 is the software release |
| 21:23.50 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:23.53 | *** join/#asterisk Dovid (~Dovid@pool-100-1-174-118.nwrknj.fios.verizon.net) |
| 21:24.04 | raidghost | When i have added the information that is needed to connect a trunk, and it says: Not registered? Then its very easy to loose its temper. |
| 21:29.11 | talntid | I see |
| 21:29.26 | talntid | I hate that I always had to use java 6 revision 13 or whatever to manage it |
| 21:29.35 | talntid | and it just.. always sucked. the bcm400 |
| 21:30.32 | *** join/#asterisk threesome (~threesome@81.183.251.5) |
| 21:30.48 | DBordello | does anybody have any recommendations for a good cordless IP phone for home use? |
| 21:30.56 | DBordello | I am not afraid of a good ebay value :) |
| 21:31.22 | drmessano | I prefer ATA's and cheap DECT phones |
| 21:31.45 | DBordello | drmessano, why is that? |
| 21:32.15 | drmessano | I personally don't see or have the need for an expensive cordless SIP device. |
| 21:32.23 | drmessano | ATA works fine for me |
| 21:32.32 | drmessano | I can use any phone I want |
| 21:32.40 | WIMPy | ATAs? |
| 21:32.41 | DBordello | Multiple lines perhaps? Transfering? |
| 21:33.00 | drmessano | When DECT 12.0 comes out, I can just swap out my $14 phone with another $14 phone |
| 21:33.25 | drmessano | DBordello, perhaps.. Two lines isn't difficult, beyond that, sure. Transfers can be done with the ATA |
| 21:33.42 | WIMPy | Is that the US version of DECT 2.0 AKA CAT-iq? |
| 21:34.25 | drmessano | WIMPy, I wasn't being specific. Point was that when this $14 DECT phone becomes obsolete, I can swap it out for another $14 phone |
| 21:34.34 | *** join/#asterisk navaismo (~navaismo@189.241.9.57) |
| 21:34.51 | drmessano | I guess I should have picked something more obscure like DECT 3.141 |
| 21:34.53 | drmessano | :) |
| 21:34.56 | DBordello | My grand plan is to use Asterisk for a home PBX. I want to tie in my phone, my ladies phone and a "home" line using google voice (or something else if that stops working). I'd like each line to have a distinct ring, and be able to be answered from any phone. I assume this would require 3 line phones? |
| 21:35.12 | [TK]D-Fender | DBordello: Google Voice is about to DIE |
| 21:35.27 | DBordello | [TK]D-Fender, i'll probably just get a SIP line then and forward to that |
| 21:35.33 | drmessano | DBordello, distinctive ring doesn't require a multi-line phone.. |
| 21:35.59 | drmessano | "Your phone, my special lady-friends phone, and another phone" is little vague. |
| 21:36.00 | DBordello | [TK]D-Fender, or, is google voice compeltly going to die? Or just the asterisk integration? |
| 21:36.14 | drmessano | Google Voice + XMPP is going to die, so no Asterisk |
| 21:36.18 | drmessano | voice is moving to hangouts |
| 21:36.18 | [TK]D-Fender | DBordello: they are shifting protocols again..... |
| 21:36.35 | DBordello | But I can always just forward to a SIP line, right? |
| 21:36.40 | DBordello | drmessano, agreed. |
| 21:36.57 | DBordello | I want to have 3 incoming lines to the house, and any phone to be able to answer, and make calls to any of the lines. |
| 21:37.32 | drmessano | DBordello, my case use at home. I have probably 6 DID's that ring into my Asterisk box. 3 of them can eventually hit my single extension cordless phone either by ringing it directly or being a failover from another extension |
| 21:38.13 | drmessano | I don't _need_ for my cordless to have another extension present on the device. |
| 21:38.13 | DBordello | drmessano, how do you determine outgoing line if you are at a given extension? |
| 21:39.03 | drmessano | Are you asking how I choose which line to use for outgoing? I don't choose beyond the default one for each extension |
| 21:39.12 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 21:39.18 | drmessano | I don't interactively select it |
| 21:39.40 | DBordello | okay, that is something I'd like to be easily done from each handset |
| 21:39.48 | DBordello | If myself, or my girlfriend is using it |
| 21:41.07 | drmessano | If *I* had to do that, I would expand to a 2 line cordless phone. Same concept. But that's me. |
| 21:41.19 | drmessano | As in, ATA+Cordless |
| 21:41.49 | DBordello | ah |
| 21:42.06 | drmessano | What bothers me about putting down money on *this weeks* cordless phone tech is that I have to invest several hundred bucks again to upgrade |
| 21:42.22 | DBordello | well, new ones don't make old ones less functional |
| 21:42.41 | DBordello | I was thinking of cruising ebay for some deals |
| 21:42.51 | drmessano | From the time I started using Asterisk, my ATA I have dedicated to cordless has been through 3 different technology upgrades and i've replaced 2 phones in addition to that due to failures |
| 21:43.02 | WIMPy | Why don;t you use a SIP base? |
| 21:43.07 | drmessano | Tell that to my 2.4GHZ phones :) |
| 21:43.31 | DBordello | WIMPy, any suggestions? |
| 21:43.47 | WIMPy | Gigaset |
| 21:44.14 | DBordello | WIMPy, those look great, thanks |
| 21:44.27 | drmessano | Although things have settled in a bit with DECT. I don't there being a significant reason to upgrade, as DECT seems to do everything pretty well |
| 21:44.38 | drmessano | Eh.. |
| 21:45.04 | WIMPy | Sure, but an analog base? |
| 21:45.17 | drmessano | Whats the problem with that? |
| 21:45.35 | WIMPy | less functionality. |
| 21:45.39 | igcewieling | heh, we hit 150 calls per server today |
| 21:46.20 | igcewieling | Aren't there SIP/DECT phones |
| 21:46.45 | WIMPy | That's waht I said. |
| 21:46.48 | drmessano | If I had to buy a SIP based device for the 5 I have replaced in the last 7 years, I would be out more than i've spent on desk phones at home lol |
| 21:47.36 | igcewieling | drmessano: as long as you don't want SIP/WIFI, the SIP/DECT are not all that expensive |
| 21:48.09 | drmessano | What's the price point now for a single extension SIP/DECT device? |
| 21:48.10 | igcewieling | WIMPy: USA DECT runs at 1.8Ghz, IIRC. |
| 21:48.40 | WIMPy | NFI. I only know that it's a different band. |
| 21:49.24 | igcewieling | drmessano: http://store.vitelity.com/panasonic-phones/ |
| 21:49.43 | DBordello | Looks like you can get a gigaset, 2 handsets for ~$60- |
| 21:49.50 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 21:50.27 | drmessano | $60 would be well worth it. |
| 21:50.39 | drmessano | I wouldn't pay $200 for a DESK phone for home, nevermind a cordless |
| 21:50.57 | igcewieling | drmessano: silly! get work to pay for it! |
| 21:51.14 | drmessano | haha |
| 21:51.20 | DBordello | http://www.ebay.com/itm/GIGASET-A510A-2B-Gigaset-Cordless-With-2-Handsets-/120955151098?pt=US_Cordless_Telephones_Handsets&hash=item1c297d22fa |
| 21:51.23 | *** join/#asterisk kresp0 (~kresp0@35.Red-83-39-25.dynamicIP.rima-tde.net) |
| 21:51.36 | DBordello | Are Wifi/SIP phones more expensive than DECT/SIP? |
| 21:51.46 | igcewieling | I have a Polycom 550 w/plantronics headset, a Polycom VVX500, and a Polycom 335 on my desk. |
| 21:51.54 | carrar | usually cheaper cause they suck more |
| 21:52.04 | WIMPy | They are definitely more shitty. |
| 21:52.08 | igcewieling | DBordello: wifi uses much more power and so battery life sucks |
| 21:52.09 | DBordello | carrar, good point |
| 21:52.15 | DBordello | aah, okay, scratch that idea then |
| 21:52.26 | DBordello | I just oredered a Polycom 501 off ebay for ~$60 |
| 21:52.33 | drmessano | That Gigaset looks interesting |
| 21:52.42 | carrar | Polycom makes great phones |
| 21:52.48 | DBordello | I can't tell if it is single line, or dual line |
| 21:52.54 | igcewieling | DBordello: I have on my desk what we sell to customers 8-) |
| 21:53.00 | DBordello | igcewieling, :) |
| 21:53.01 | drmessano | At $65, you're at the cost of an ATA + Walmart DECT phone. Can't argue with that |
| 21:53.21 | igcewieling | drmessano: worth looking into at least. |
| 21:53.31 | carrar | But while at Walmart you could get chips |
| 21:53.32 | DBordello | drmessano, I am just discussing options. I can certainly see the advantage of moving the SIP side to an ATA |
| 21:54.08 | drmessano | No, I think at $65 it doesn't make sense to use the ATA+cordless |
| 21:54.27 | drmessano | You're talking almost the same price for much more functional device |
| 21:54.28 | DBordello | Does polycom make any cordless phones? |
| 21:54.41 | drmessano | Ye$ they certainly do |
| 21:54.47 | WIMPy | I would go for a BRI base, but that's not an option for you, I guess. |
| 21:54.49 | drmessano | They are quite nice $$$ |
| 21:54.50 | carrar | probably pick up a aastra NBU-400 fairely cheap |
| 21:54.56 | carrar | for SIP DECT |
| 21:55.04 | WIMPy | No, they buy from Kirk. |
| 21:55.11 | WIMPy | Or did they buy Kirk? |
| 21:55.25 | gorkish | the spectralink line has both wifi and dect however i believe polycom doesnt actually make them |
| 21:55.35 | DBordello | Any signifcant advantage of going purely polycom, versus a mix/match setup? |
| 21:56.08 | igcewieling | DBordello: Polycom has two cordless lines. Neither are in any way similar to their desk phones. |
| 21:56.10 | carrar | Your comphortlevel |
| 21:56.18 | igcewieling | one line is dect, and one is wifi, IIRC |
| 21:56.29 | carrar | http://www.ebay.com/itm/Aastra-MBU-400-Wireless-DECT-Solution-Handset-Base-Bundle-/151031771150?pt=US_VoIP_Home_Phones&hash=item232a31f00e |
| 21:56.30 | gorkish | dbordello: well they are all provisioned with the same tree and basically the same config setup so its easier in theory |
| 21:56.52 | DBordello | gorkish, I thought that might help |
| 21:56.55 | igcewieling | DBordello: learning one phone well is hard enough, why make three times the work |
| 21:57.14 | carrar | Learning is good |
| 21:57.29 | gorkish | the spectralink 8400 (wifi) provisioning is the same as the soundpoint ip and vvx but they are expensive as all get-out |
| 21:57.34 | DBordello | carrar, that looks nice, but a bit steep $$$ |
| 21:57.35 | igcewieling | carrar: you can always go back and add more phones just to complicate things later. |
| 21:57.45 | igcewieling | Polycoms are expensive. |
| 21:57.53 | DBordello | got it :) |
| 21:57.56 | carrar | DBordello, if this is for a office you need to consider what you are doing it for |
| 21:58.11 | DBordello | carrar, it is for home, with no real good use :) |
| 21:58.14 | carrar | oh |
| 21:58.21 | *** join/#asterisk War_Bear (~War_Bear@warbear.co.uk) |
| 21:58.25 | carrar | you don't need cordless sip phone |
| 21:58.33 | carrar | Get a bluetooth headset |
| 21:58.48 | gorkish | one other roaming wireless option is to get a regular desk phone and put a DECT headset on it |
| 21:58.57 | drmessano | Bluetooth? Isn't that what you get when you eat a Blackberry? |
| 21:58.58 | WIMPy | How do you dial with your BT headset? |
| 21:59.16 | carrar | with your fingers on the phone |
| 21:59.34 | carrar | didn't say he needed to dial from the yard |
| 21:59.38 | carrar | DFTY |
| 21:59.43 | gorkish | actually vvx600 has built in bluetooth and it works very well. |
| 21:59.50 | WIMPy | Why do you need a wiereless headset on a wired phone? |
| 21:59.51 | DBordello | wireless dialing as well :) |
| 21:59.52 | gorkish | dialing with DECT handset+lifter you can do with DISA |
| 22:00.08 | drmessano | I still say if we're talking about a comparison involving $200 phones, an ATA + Walmart DECT phone wins all. Though I am impressed with the $65 price point of that Gigaset, as long as its not junk |
| 22:00.23 | carrar | I use a bluetooth plantronics binoral headset on my Cisco 7941, it's just awesome |
| 22:00.25 | drmessano | Though I can't see it being much worse than a low end POTS DECT phone |
| 22:00.30 | carrar | (at home) |
| 22:00.37 | gorkish | linksys used to sell a cheap sip wifi handset didnt they? like 80 bucks? |
| 22:00.42 | drmessano | Yeah |
| 22:00.54 | drmessano | WIP something? |
| 22:00.58 | carrar | I got a Spectralink 8020 too, but it's kinda lame |
| 22:01.03 | drmessano | WIP300 or some crap |
| 22:01.06 | carrar | Looks pretty though |
| 22:01.11 | drmessano | Work In Progress |
| 22:01.29 | drmessano | IIRC D-Link had a flip phone |
| 22:01.33 | DBordello | I see no real good reason for a Wifi phone. You need the charger anyways |
| 22:01.33 | carrar | If I have to call someone, I have to be at my desk anyways |
| 22:01.51 | DBordello | Might as well use DECT to backhaul |
| 22:02.18 | carrar | NSA will record your calls if you do cordless |
| 22:02.39 | DBordello | carrar, you mean they will record them twice |
| 22:02.42 | WIMPy | In all other cases as well, I'm sure. |
| 22:02.43 | drmessano | My iPhone with BRIA ends up taking up the role of a "cordless phone" for the most part. Almost no need for it when I can roam the office or home with my PBX extension in hand |
| 22:03.02 | carrar | not if your calls are encrypted |
| 22:03.18 | drmessano | lol |
| 22:03.32 | carrar | at least untill the UTAH datacenter is built |
| 22:04.42 | drmessano | DBordello, that's another option too.. a Softphone app running on your smartphone. Now you have a cellphone/multi-line extension to your PBX |
| 22:04.47 | drmessano | Just a thought |
| 22:04.58 | DBordello | drmessano, I was thinking about that too |
| 22:05.32 | drmessano | My wife and I both use BRIA. It wasn't the cheapest option, but for the iPhone I felt it was the best/only path |
| 22:05.48 | drmessano | Nice answering your home phone on your cell via SIP |
| 22:06.41 | DBordello | Do you do that over the ceullar data while away? |
| 22:06.52 | DBordello | Or is that asking for a bad time |
| 22:06.59 | drmessano | Yes, with G729. It works well |
| 22:07.16 | gorkish | i use groundwire on the iphone; it rocks |
| 22:07.29 | drmessano | As a matter of fact, I have used it quite a bit over a 256k data link out to some of my sites. |
| 22:07.40 | gorkish | the registration can go through their proxy so that it roams very well between cell+wifi |
| 22:07.59 | drmessano | I have Wifi access points out there... so the phone jumps on the wifi and the backhaul to the office is over our RF path |
| 22:08.40 | drmessano | So people call me on my extension at work.. "Can you come to my office".. "Uh, I am out in the sticks" "Uh, what?.. But I..." |
| 22:10.05 | DBordello | WIMPy, since you recommended gigaset, any experience with them? Thinking about pulling the trigger. |
| 22:10.39 | DBordello | I am thinking an A510 |
| 22:10.40 | WIMPy | Only a really old one. Nothing great, but it works. |
| 22:10.57 | gorkish | hey is there a faster/cheaper way to do a simple voice dial by name type application than going the full route with res_speech and unimrcp and lumenvox ? |
| 22:11.35 | WIMPy | As I said, I personally prefer an ISDN base if two simultaneous calls are enough. |
| 22:13.00 | DBordello | WIMPy, fair enough |
| 22:13.17 | DBordello | People still use ISDN? I recall using that for 128k dialup :) |
| 22:13.54 | WIMPy | Is there anythign to replace it? |
| 22:15.05 | DBordello | I am pulling the trigger on this guy: http://www.ebay.com/itm/SLEEK-STYLISH-SIEMENS-GIGASET-CORDLESS-PHONE-HD-AUDIO-COLOFUL-DISPLAY-A510A-2B-/111067143333?pt=US_Cordless_Telephones_Handsets&hash=item19dc1e1ca5 |
| 22:15.07 | DBordello | I'll report back |
| 22:15.25 | DBordello | WIMPy, obviosuly I don't know all of its uses, but for mine (internet), it has been replaced. |
| 22:15.42 | WIMPy | Off ourse. |
| 22:15.50 | WIMPy | But not for telephony. |
| 22:22.37 | DBordello | I wonder how the Obihai Obi devices will deal with the Google Voice protocol change |
| 22:23.17 | drmessano | They wont |
| 22:23.25 | DBordello | I know those are popular as google voice end points |
| 22:23.27 | drmessano | There's no interface to hangouts |
| 22:23.54 | DBordello | Lame. |
| 22:24.07 | drmessano | That's really the big issue.. and anything google is going to offer will be in the form of embedding hangouts into your app, vs having a way of "connecting" |
| 22:24.22 | drmessano | They want Hangouts to be an island |
| 22:24.50 | drmessano | Kinda goes against the whole "Don't be evil" |
| 22:25.19 | DBordello | Will voice be leaving gmail? |
| 22:25.47 | drmessano | As soon as you opt to "upgrade" from Chats to Hangouts it will.. for the time being. The plan is to then move Voice to hangouts |
| 22:25.52 | drmessano | So it will come back, in theory |
| 22:26.16 | DBordello | ah |
| 22:26.31 | drmessano | They're basically creating their own version of Skype |
| 22:26.52 | drmessano | One small painful piece at a time |
| 22:26.57 | DBordello | Will all the external clients break as well? XMPP I believe |
| 22:27.13 | drmessano | Yes, Hangouts do not do XMPP |
| 22:27.19 | drmessano | Thats why Google Voice is going away |
| 22:27.27 | drmessano | from asterisk, anyway |
| 22:27.56 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:28.36 | drmessano | Google decided to drop XMPP from hangouts.. because it was problematic. Google Voice is moving to Hangouts. Therefore, Google Voice will not be accessible with XMPP. |
| 22:28.45 | drmessano | Those are the 3 points |
| 22:30.11 | drmessano | At the end of the day, Google is dropping XMPP.. From what I understand, XMPP never existed inside their network, same as Facebook.. it was merely an interface to their messaging backend |
| 22:30.28 | DBordello | ah, it had a good, partially implemented life |
| 22:30.33 | drmessano | So they're cutting basically cutting away the ailing interface |
| 22:30.54 | drmessano | -cutting |
| 22:31.50 | DBordello | At the end of the day, it isn't a big deal. A DID provider is like $2/month that can be forwarded to from Google Voice |
| 22:32.33 | DBordello | And I assume you can spoof your caller ID anyways to look like you are calling from google voice? |
| 22:33.06 | drmessano | If you have a company that supports it, yes. Flowroute allows it |
| 22:34.37 | DBordello | Flowroute looks intriguing |
| 22:34.45 | drmessano | Flowroute is awesome |
| 22:35.02 | DBordello | great, I will probably go with them |
| 22:35.29 | drmessano | Ok, time to go home. A Reba, durchey. |
| 22:36.15 | DBordello | take care |
| 22:46.07 | *** join/#asterisk esaym (~esaym153@216-45-91-132.gvec.net) |
| 22:47.45 | esaym | ugh, I upgraded my asterisk install from the .deb package provided by digium to the one in debian wheezy. Now asterisk can't find the voicemail files: http://pastie.org/7977196 |
| 22:49.29 | igcewieling | esaym: good luck. |
| 22:49.38 | esaym | more data in this one: http://pastie.org/7977201 |
| 22:49.51 | esaym | igcewieling: yea it stinks, I am missing calls now :( |
| 22:50.45 | igcewieling | try uninstalling the new package, then reinstall the old package, then uninstall the old paclage, then install the new package. Back up /etc/asterisk and /var/spool/asterisk and /usr/lib/asterisk |
| 22:51.33 | esaym | igcewieling: yea I am thinking that.. |
| 22:51.51 | igcewieling | this is why you should install Asterisk from source. 8-| |
| 22:53.39 | *** join/#asterisk dtcrshr (~datacrush@unaffiliated/datacrusher) |
| 22:53.47 | talntid | or properly learn how to use your package manager :) |
| 22:54.32 | igcewieling | talntid: with packages you are stuck with whatever version you get from your repo, even if you want or need to upgrade |
| 22:55.04 | talntid | igcewieling, i fully understand. I use asterisk from my package manager. |
| 22:55.22 | talntid | you can change repos, too |
| 22:57.25 | talntid | it's likely that during the upgrade, it asked to overwrite a config file, and it was told yes |
| 22:57.59 | talntid | backups should solve this, right, esaym? |
| 23:00.26 | esaym | talntid: it is backed up. |
| 23:00.36 | esaym | talntid: but I don't get why just voicemail stopped working |
| 23:00.41 | esaym | everything seems the same |
| 23:00.50 | talntid | it's a config issue. or a permissions issue - one or the other |
| 23:00.59 | talntid | where is it looking for the vm at? where is the vm actually at? |
| 23:01.19 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 23:01.19 | *** mode/#asterisk [+o file] by ChanServ |
| 23:01.32 | talntid | gives file a blueberry muffin |
| 23:02.49 | esaym | talntid: i don't know where it is looking, debug doesn't say.. :( |
| 23:03.25 | talntid | ok, so lets take this the easy way |
| 23:03.48 | talntid | vm-intro |
| 23:03.54 | talntid | where is it currently located? |
| 23:05.46 | talntid | hint: find command |
| 23:05.52 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 23:07.03 | esaym | I posted in the pastie link above |
| 23:08.50 | talntid | the odds of me scrolling up to find all the pastie links you posted above is very low |
| 23:09.15 | talntid | but the one at: [15:50] <esaym> more data in this one: http://pastie.org/7977201 ... doesn't answer my question. |
| 23:09.20 | *** join/#asterisk blee (~blee@67.8.206.215) |
| 23:13.00 | esaym | talntid: /usr/share/asterisk/sounds/en_US_f_Allison |
| 23:13.38 | esaym | I uninstalled and reinstalled and still same thing. Of course I am using the same conf files.. |
| 23:13.46 | esaym | did a full purge |
| 23:13.50 | talntid | pasty /usr/share/asterisk/sounds/en_US_f_Allison |
| 23:13.53 | talntid | ls /usr/share/asterisk/sounds/en_US_f_Allison |
| 23:13.58 | esaym | ls /usr/share/asterisk/sounds/ |
| 23:14.00 | esaym | ls /usr/share/asterisk/sounds/ |
| 23:14.24 | esaym | thought he was at a command prompt |
| 23:14.31 | talntid | hehe |
| 23:14.44 | jmetro | dont ask me how many times i've linux'd in a windows box |
| 23:14.55 | jmetro | "ls" damnit "ls" damnit "cp" damnit |
| 23:15.01 | talntid | lol yup |
| 23:15.03 | esaym | ls /usr/share/asterisk/sounds/en_US_f_Allison/ |wc -l |
| 23:15.05 | esaym | 294 |
| 23:15.22 | esaym | /usr/share/asterisk/sounds/en_US_f_Allison/vm-intro.gsm |
| 23:15.44 | esaym | it is complaining about ulaw. Wonder if that is the issue? |
| 23:17.30 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 23:17.39 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-ciecuxhugifbldgf) |
| 23:17.43 | esaym | installing debian package asterisk-core-sounds-en-wav, |
| 23:17.52 | esaym | not sure why they would matter though |
| 23:19.27 | esaym | talntid: ok that worked.. |
| 23:19.51 | esaym | don't know why. Perhaps I have a gsm module disabled... but on the other deb package from digium there was only gsm files.. |
| 23:20.06 | talntid | in voicemail.conf |
| 23:20.19 | talntid | whats it say for "format = " |
| 23:20.20 | talntid | ? |
| 23:20.47 | talntid | should only matter for writing though |
| 23:21.24 | esaym | talntid: wav |
| 23:21.32 | *** join/#asterisk Dovid (~Dovid@ool-43523afd.dyn.optonline.net) |
| 23:21.41 | esaym | talntid: but I changed it to wav|wav49|gsm |
| 23:21.50 | esaym | talntid: I think that is just for writing vm's |
| 23:21.54 | esaym | yea |
| 23:22.44 | esaym | alright ty |
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| 23:35.05 | *** join/#asterisk boscage (boscage@unaffiliated/boscage) |
| 23:47.46 | WIMPy | These scanners can actually be quite entertaining. |
| 23:56.29 | *** join/#asterisk robl^ (~robl@pdpc/supporter/active/robl) |
| 23:59.26 | *** join/#asterisk xcom (~wtf@pdpc/supporter/professional/seri) |
