| 00:00.38 | *** join/#asterisk ipengineer (~zconkle@108-206-120-218.lightspeed.mckntx.sbcglobal.net) |
| 00:02.41 | *** join/#asterisk j0 (~dan@S0106c0eae432889d.va.shawcable.net) |
| 00:03.58 | j0 | i'm having voice problems on my pri with a sangoma a101. just get dead air and now a screetching kind of sound.. everything else appears normal so i'm not sure how to troubleshoot it.. any ideas on where to start? |
| 00:05.02 | *** join/#asterisk ipengineer (~zconkle@108-206-120-218.lightspeed.mckntx.sbcglobal.net) |
| 00:06.21 | ipengineer | file: You here? |
| 00:10.23 | navaismo | j0, sangoma's troubleshooting wiki page |
| 00:15.09 | ipengineer | If I set CALLERID(name-pres)=Sales will that change the name sent to an extension but preserve the name for CDR? |
| 00:16.02 | Penguin | CallerID isn't something sent to extensions. CallerID is data carried on a channel. |
| 00:17.28 | ipengineer | Penguin: Ok.. So what I am wanting to accomplish is prepend custom text in front of caller name sent to an extension without affecting CDR.. What would be the best way to accomplish that? |
| 00:18.05 | Penguin | I just told you that Caller ID isn't something that is sent to extensions, and you come back with a question about sending it to an extension. |
| 00:19.09 | WIMPy | ipengineer: That's not a valid value for *pres. |
| 00:19.32 | ipengineer | Penguin: Sorry.. We always call a device and extension.. Dial(PJSIP/123@abcdefg) |
| 00:20.17 | Penguin | If you're going to use asterisk and ask questions pertaining to asterisk, you'll have to learn that extensions are not devices and devices are not extensions. |
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| 00:23.11 | ipengineer | Penguin: Point taken |
| 00:23.18 | Penguin | I don't see why you can't set the CALLERID() and the CDR() separately. |
| 00:24.38 | ipengineer | I wasn't sure if setting the CALLERID would effect the CALLERID being written to the CDR |
| 00:30.03 | Penguin | Yeah, it does. |
| 00:30.29 | Penguin | If you change the CALLERID(name), the CDR will show the modified name rather than the original name. |
| 00:32.32 | ipengineer | What if I set name-pres is that propagated to the CDR? I don't really understand the difference between the two and am not finding any documentation differentiating them. |
| 00:33.20 | *** join/#asterisk navaismo (~navaismo@189.191.246.129) |
| 00:36.42 | WIMPy | core show functio CALLERID |
| 00:36.48 | WIMPy | core show function CALLERPRES |
| 00:37.43 | WIMPy | *pres sets presentation indicators. |
| 00:39.07 | ipengineer | WIMPy: Thanks |
| 00:40.04 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 00:58.04 | j0 | well I learned lots on Sangoma's troubleshooting page, but still no dice.. Our PRI seems to be working perfectly. No errors that I can see anywhere. It's just that we don't get any audio either way. It is either silent or a constant tone. |
| 00:59.48 | WIMPy | Wrong codec? |
| 01:01.01 | WIMPy | Doy you see other transmission errors? |
| 01:08.32 | j0 | no transmission errors on the interface. pri settings are the same as they have been for years |
| 01:08.43 | j0 | how does a codec come into play for a pri? |
| 01:09.24 | WIMPy | Depending on your area you will use alaw or ulaw. |
| 01:10.00 | j0 | is that configured in zaptel or wanrouter? |
| 01:10.21 | WIMPy | Possibly both. |
| 01:10.29 | WIMPy | hasn't used wanpipe |
| 01:10.40 | Penguin | If it was working before and now it isn't, you've got to look at what got changed between then and now. |
| 01:11.28 | j0 | it stopped working this morning.. no-one has logged in for a couples months and no configuration files other than voicemail and extensions have been modified |
| 01:11.56 | WIMPy | Have you tried turning it off and on again? |
| 01:12.33 | j0 | yeah.. restarted it in every way possible :) |
| 01:14.25 | WIMPy | I don't see how you could get bad audio without other issues as well. |
| 01:16.33 | j0 | yeah.. so i think it's on the service provider side |
| 01:17.52 | WIMPy | I don't see how that would work out unless their switch is on fire. |
| 01:18.59 | j0 | yeah.. and that's my problem |
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| 02:29.11 | DssTech | whats up yall check this out http://youtu.be/NAtDsM2-nNc |
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| 03:52.24 | volga629 | Hello Everyone, issue with cert have being resolved new certs did the trick |
| 03:53.05 | volga629 | Is anybody was be able setup presence with jitsi ? |
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| 04:04.27 | volga629 | Hello Everyone, issue with cert have being resolved new certs did the trick |
| 04:04.30 | volga629 | Is anybody was be able setup presence with jitsi ? |
| 04:07.08 | *** join/#asterisk str8uplinux (~dschuett@ip70-171-169-110.om.om.cox.net) |
| 04:10.21 | str8uplinux | I am trying to set up google voice on asterisk 11.7, but am getting the following error when making an outbound call: chan_motif.c:821 jingle_add_google_candidates_to_transport: Unable to add Google ICE candidates as ICE support not available or no candidates available |
| 04:10.43 | str8uplinux | i have set icesupport=true in rtp.conf |
| 04:10.47 | str8uplinux | any ideas? |
| 04:11.20 | Penguin | Did you consider what the message says and disable ICE support? |
| 04:12.17 | str8uplinux | i get the same error if i comment out icesupport=true |
| 04:13.45 | str8uplinux | Penguin: besides chan_motif requires icesupport. |
| 04:13.54 | str8uplinux | according to the asterisk docs |
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| 05:25.17 | ChannelZ | hmmm |
| 05:25.47 | ChannelZ | lots of "sipvicious" trolling lately |
| 05:28.25 | ChannelZ | str8uplinux: is your asterisk behind a NAT? odd that it can't figure out ANY IP to add.. |
| 05:40.44 | *** part/#asterisk str8uplinux (~dschuett@ip70-171-169-110.om.om.cox.net) |
| 05:41.03 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.118) |
| 05:41.04 | Penguin | vlad_starkov: Please fix your connection. |
| 05:41.36 | ChannelZ | GRRRR ban 50.30.32.0/20 |
| 05:41.41 | ChannelZ | motherfucker |
| 05:42.07 | ChannelZ | excuse me |
| 06:13.52 | DssTech | I would like to be part of this channel. I own a VoIP Company and a i an engineer and I design Modules and PBX software. also I own a computer company and a datacenter. So anyone needs Virtual hosting let me know. I will hook you up. if you want to host your PBX servers at my datacenter. I am willing to help anyone. |
| 06:17.46 | Penguin | Free VPS? |
| 06:18.04 | ChannelZ | maybe you could fix this piece of crap YouTube. argh. |
| 06:18.07 | DssTech | not free lol well maybe for the channel owners |
| 06:18.32 | DssTech | ChannelZ: is it not tubing right? |
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| 06:21.03 | ChannelZ | it's just being random stupid like always. An video, in spite of showing minutes buffered, plays for 3 and stops for 15 seconds for no reason |
| 06:22.25 | ChannelZ | An video. Jesus. A video. |
| 06:28.27 | drmessano | Mmmm Advertising. Dinner |
| 06:33.56 | drmessano | I'll take a free virtual host |
| 06:36.31 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.144) |
| 06:36.32 | Penguin | vlad_starkov: Please fix your connection. |
| 06:39.15 | ChannelZ | I imagine if he was actually at his workstation, he would. |
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| 11:03.40 | Ice_Strike | Hey |
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| 12:54.24 | Ice_Strike | Has anyone written Asterisk AMI process in go before? |
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| 14:38.49 | boratynskikamil | Hello. |
| 14:39.55 | boratynskikamil | I have got a OpenVox G400E GSM card. Question: is it nescessary to install drivers from extra-chan and the whole asterisk from sources one more time? Some suggesstions? |
| 14:41.47 | boratynskikamil | As I see in lspci, there is no driver and dahdi cannot identify card, what is obvious. :-) |
| 14:44.14 | boratynskikamil | http://wiki.openvox.cn/index.php/OpenVox_G400E_Installation_Manual#Chapter_3_Software_Installation_and_Configuration - wiki. |
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| 15:05.48 | WIMPy | wonders if there's a language called "go" |
| 15:14.18 | LiuYan | google go? |
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| 16:28.14 | volga629 | Hello Everyone, I am looking for sip client which supports distributed device state |
| 16:45.35 | *** join/#asterisk Penguin (~xwQ5kwYl6@20264.odci.gov.united-states.rltk.us) |
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| 18:33.16 | gusto | so |
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| 18:59.29 | ChannelZ | us |
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| 19:03.48 | csd199 | hi. I have 2 asterisk-linux boxes and a VPN (openvpn). I want to put all the IAX trafic inside the VPN. When I configure the LAN IP's from servers to access the remote asterisk box, it doesn't work. If I use the TUN IP's, it works, I can connect the two server via IAX channels. My question, is there a way I can use instead the TUN IP's, the server LAN IP's to access the IAX remote asterisk |
| 19:03.48 | csd199 | server? |
| 19:04.30 | [TK]D-Fender | the IP used is up to your networking stack, not * |
| 19:04.50 | csd199 | yes |
| 19:07.49 | csd199 | or why? |
| 19:09.29 | csd199 | or where can I read about it? |
| 19:10.07 | csd199 | sip channels work fine inside the vpn, but not IAX... |
| 19:17.16 | [TK]D-Fender | * sources the IP based on your routing stack. So does SIP. |
| 19:17.37 | [TK]D-Fender | both use the same mechanism. |
| 19:18.01 | [TK]D-Fender | SIP holds internal routing information as well though since it is a routed protocol. IAX is not. |
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| 19:31.53 | csd199 | ok.... this means I can use sip instead iax to connect the asterisk servers via vpn? |
| 19:36.26 | coreyf1513 | csd199: openvpn is doing net-to-net vpn from the same box as asterisk? |
| 19:37.58 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 19:40.16 | csd199 | I have two separated asterisk servers. One is vpn server and the other vpn client. they are in internet |
| 19:42.39 | csd199 | and if I use the sintax: username:pass@server.something.com works fine, but if I use username:pass@localIP doesn't work, if I use username:pass@TUN-IP also works fine.... |
| 19:43.19 | csd199 | he he |
| 19:44.11 | drmessano | [14:04:26] <[TK]D-Fender> the IP used is up to your networking stack, not * |
| 19:45.30 | drmessano | Your routing is not set up correctly |
| 19:46.21 | csd199 | hmm.... well, the two nets can see each other. |
| 19:46.47 | csd199 | but if you give me a clue, I'll appreciate it. |
| 19:47.22 | csd199 | is there a way to let asterisk know about routing networks? |
| 19:47.28 | coreyf1513 | csd199: ensure you have proper FORWARD rules in iptables, and modify your ip route to specify source addresses. these are things you'll need to ask about in a channel for your distro. |
| 19:47.47 | volga629 | which soft client is good for asterisk |
| 19:48.40 | csd199 | coreyf1513, I think the routing network is working fine. I have other applications working ok. |
| 19:48.55 | drmessano | volga629, http://en.wikipedia.org/wiki/List_of_SIP_software#Free_and_open_source_license_2 |
| 19:48.58 | csd199 | but, Is there a configuration leting know asterisk about routing? |
| 19:49.07 | drmessano | .... |
| 19:49.09 | drmessano | No |
| 19:49.30 | csd199 | do you recommend to sustitute iax with sip? |
| 19:49.33 | drmessano | You've proven your tunnel is working, but your routing is not |
| 19:49.55 | drmessano | If you want the LAN IPs to work, fix your routing.. Thats not an Asterisk issue |
| 19:49.57 | csd199 | ok... |
| 19:50.00 | drmessano | Thats your network stack |
| 19:50.40 | csd199 | something like adding a new route? |
| 19:50.50 | csd199 | route add xxxx etc... |
| 19:50.53 | csd199 | ? |
| 19:51.02 | volga629 | I have distributed device state, tried today might be 10 soft clients, and by far only bria handling device state correctly |
| 19:51.43 | drmessano | Possibly.. You need to check your routing tables, your tunnel config, etc.. Figure out where the error lies. Your distros help channel is a more appropriate place for that sort of investigation |
| 19:51.45 | volga629 | most local client list is not working correctly or subscribe not working |
| 19:52.25 | volga629 | need try blink |
| 19:52.32 | drmessano | volga629, have you tried Jitsi? |
| 19:53.26 | volga629 | yes, contact list is adding by default @domain name and when asking for subscribe getting 404, because extension on another server |
| 19:53.40 | volga629 | but video and sound working good |
| 19:55.34 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 19:55.44 | [TK]D-Fender | We could be looking at -40 for New Years ..... yay |
| 19:56.29 | volga629 | in Ottawa +2 right now :-) |
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| 21:40.35 | Free99 | hey guys. I forgot what the command was to display a peer/user's details, specifically their password |
| 21:40.40 | Free99 | any pointers? |
| 21:42.28 | [TK]D-Fender | Free99: "cat sip.conf" |
| 21:43.21 | Free99 | [TK]D-Fender, the accounts are in an ODBC MySQL database.. If I have to wade through the mysql, I suppose I can try that |
| 21:43.39 | [TK]D-Fender | You'll have to. |
| 21:44.32 | Free99 | (sigh) well, nothing to it but to do it I guess. For some reason I thought there was a command that'd list everything out for a particular user account |
| 21:44.59 | [TK]D-Fender | Free99: Secret is the one thing you DON'T get |
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| 21:58.28 | rampage73 | anyone well versed in paging? all I want to do is page through the speakers of the phone server itself, the speakers work and the server can play other sounds but when I attempt to ask asterisk to do it i get nothing |
| 21:59.08 | [TK]D-Fender | Show us |
| 21:59.10 | [TK]D-Fender | ~pb |
| 21:59.11 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 21:59.12 | [TK]D-Fender | ^^^ |
| 21:59.36 | *** part/#asterisk rampage73 (~rampage73@184.2.147.197) |
| 22:01.33 | [TK]D-Fender | And as fast as they ask and are requested to provide backup... off they run... |
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| 22:08.39 | Free99 | [TK]D-Fender, it's because you are so direct that it is frightening haha |
| 22:09.19 | [TK]D-Fender | Patient: Doctor, my arm hurts! |
| 22:09.26 | [TK]D-Fender | Doctor: Ok, lets take a look. |
| 22:09.29 | Free99 | I still remember you most prominently out of this room when I was first learning asterisk lol |
| 22:09.30 | [TK]D-Fender | *Patient flees |
| 22:12.14 | Free99 | generally an expected reaction when the doctor has a bloody lab coat from wading through many configs past, and a gruff demeanor to boot :P |
| 22:12.48 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 22:12.51 | [TK]D-Fender | I said two words. |
| 22:12.56 | [TK]D-Fender | That's it. |
| 22:13.06 | [TK]D-Fender | and pointed to a place to dump the evidence |
| 22:14.06 | [TK]D-Fender | You wanna run after that? What does that say about them? |
| 22:14.27 | [TK]D-Fender | Clearly the problem isn't important otherwise they'd stay, right? |
| 22:19.02 | [TK]D-Fender | Hopefully just an accidentl quit or emergency departure... |
| 22:19.29 | [TK]D-Fender | considering he's been asked the same kind of things before and provided them just fine. |
| 22:20.00 | Free99 | dunno if I'd wish for an emergency departure, but yeah just ribbing you a little |
| 22:20.29 | Katty | hi kiddos |
| 22:20.36 | *** join/#asterisk marlinc (~marlinc@ip565fa73c.direct-adsl.nl) |
| 22:28.59 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 22:30.06 | Free99 | see you later guys, and thanks [TK]D-Fender |
| 22:39.38 | *** join/#asterisk Dpunkt (~Dpunkt@p5DE5F1F0.dip0.t-ipconnect.de) |
| 22:40.26 | *** join/#asterisk jasonwert (~w3rt@71.89.137.28) |
| 22:48.47 | *** join/#asterisk troyt (~troyt@2601:7:6d00:432:44dd:acff:fe85:9c8e) |
| 23:17.42 | volga629 | Is res_corosync still available for asterisk 12 ? |
| 23:19.55 | *** join/#asterisk hayden_ru (~hayden_ru@0896410724.static.corbina.ru) |
| 23:20.12 | gusto | btw. i discovered an interesting problem with my confbridge |
| 23:20.29 | *** join/#asterisk bsdice (~bsdice@embinet.eu) |
| 23:20.38 | gusto | its that when some of the peers goes unexpectedly down, they keep all hanging in |
| 23:20.47 | gusto | and of course traffic is still transfered |
| 23:20.49 | gusto | into nothing |
| 23:21.04 | bsdice | good evening |
| 23:21.45 | gusto | so, one solution would be to have a hard timeout, so that every conference would have a hard limit of say 60 minutes and be killed kicking away all peers from it |
| 23:27.22 | gusto | i read something about a silence timeout that would be interesting to |
| 23:27.23 | gusto | too |
| 23:31.44 | *** join/#asterisk bkruse (~Adium@24.42.229.8) |
| 23:33.34 | ChannelZ | Well you're saying that if a person disconnects their call, they get stuck in the conference? |
| 23:34.12 | gusto | not everytime |
| 23:34.20 | gusto | that's what is so strange about it |
| 23:34.33 | gusto | it does not happen on phones over ATA |
| 23:34.45 | gusto | but it keeps happening when a softphone crashes |
| 23:34.59 | gusto | for example ekiga is crashing all the time |
| 23:36.42 | ChannelZ | you might try rtptimeout on it |
| 23:42.12 | Ice_Strike | Has anyone written Asterisk AMI process in Go language? |
| 23:42.33 | Ice_Strike | I might consider trying that instead of Node.js |
| 23:42.52 | Ice_Strike | Go language is much better when it come to Concurrent processing.. |
| 23:51.52 | ChannelZ | don't think I've ever seen it mentioned, ever |
| 23:57.26 | [TK]D-Fender | Normally making such an oddball choice in language would scream "dead-end project in the making" |
| 23:57.47 | [TK]D-Fender | And that's for anything at all... |
| 23:57.49 | Ice_Strike | Why you say that? |
| 23:58.02 | [TK]D-Fender | Then to say specifically for an AMI framework... |
| 23:58.38 | [TK]D-Fender | Future maintainability. Long term survival of the language itself. Platfor supportm portability, etc |
| 23:59.49 | [TK]D-Fender | "When we launched Go for App Engine in May 2011 the SDK was just a modified version of the Python SDK. At the time, there was no canonical way to build or organize Go programs, so it made sense to take the Python approach." |