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| 00:57.51 | rexwin_ | I have instances of agent getting calls after logging out and without logging in to asterisk server. why does this happen? |
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| 01:20.11 | pabelanger | rexwin_, check the logs |
| 01:20.20 | pabelanger | there is a reason for asterisk dialing an extension |
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| 02:56.52 | loggiew | Ive setup asterisk before successfully but this time Im having a problem. I must be goofing something and it's been a while. |
| 02:56.59 | loggiew | no audio on either device |
| 02:57.08 | loggiew | otherwise the call seems to connect |
| 02:57.15 | Penguin | NAT must be involved. |
| 02:57.21 | loggiew | I see traffic with tcpdump on ports 10000-20000 |
| 02:57.34 | loggiew | its a virtual server and the ports are wide open |
| 02:57.39 | loggiew | im playing around so no firewall at all |
| 02:58.25 | loggiew | i figured it was nat on the clients end, but when i use netstat -anup and others, I dont see those ports open for service |
| 02:58.55 | loggiew | Im half wondering if I screwed up something about the config thats obvious that should be listening on those ports? |
| 02:59.00 | loggiew | I setup rtp.conf |
| 02:59.10 | loggiew | it's asterisk 11.8 |
| 02:59.17 | Penguin | Asterisk will not listen on the RTP ports. |
| 02:59.37 | loggiew | ok, so its something obvious im missing |
| 03:00.54 | loggiew | is it a poor fundamental understandign of asterisk? or something additional I need to be doing? |
| 03:01.32 | Penguin | Is NAT involved or not? |
| 03:01.39 | loggiew | not on the server end |
| 03:02.07 | loggiew | the phone is probably ipv6 translated at some ipv4 NAT point and the web client is at home behind a router/nat |
| 03:02.32 | Penguin | Did you configure the peer entry to be aware of NAT? |
| 03:03.05 | loggiew | is it not sufficient to turn on those options at the general level? |
| 03:03.17 | loggiew | or do I need to specify per peer that it may encounter nat? |
| 03:03.57 | Penguin | You can enable that at the general level and it will precipitate down to the peers as long as you do not disable it within the peer. |
| 03:04.10 | loggiew | checking to confirm i didnt disable.. |
| 03:04.47 | loggiew | it appears fine, at the peer level i have canreinvite=no |
| 03:04.57 | loggiew | and i commented out directmedia=no |
| 03:05.03 | loggiew | where I set it under general |
| 03:05.09 | Penguin | directmedia replaced canreinvite ages ago. |
| 03:05.31 | loggiew | ya. at a certain point I was throwing shit at the wall |
| 03:05.57 | Penguin | I'm surprised it didn't puke on the presence of canreinvite. |
| 03:06.11 | loggiew | maybe it is :D |
| 03:06.14 | loggiew | let me try without |
| 03:07.26 | loggiew | that didnt fix it |
| 03:07.36 | loggiew | i get audio for the first moment as it pickes up |
| 03:07.50 | loggiew | and i seem to recall that being what happens as it switches to using the higher port numbers |
| 03:07.56 | loggiew | 10000+ |
| 03:08.04 | loggiew | and hence why nat is an issue blah blah |
| 03:08.35 | loggiew | but the server isn't on NAT and nothing listens on those ports |
| 03:08.38 | Penguin | But media doesn't start out in one place and then "switch to using higher port numbers." |
| 03:08.47 | loggiew | but i see traffic on those ports with tcpdump |
| 03:08.55 | loggiew | ok. ? |
| 03:09.11 | Penguin | The port numbers are determined in SDP. Check your sip debug to see what it is doing to set up the media. |
| 03:10.25 | loggiew | looking |
| 03:11.32 | loggiew | ug. 24 pages of packet data or whatever |
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| 04:21.38 | FuriousGeorge | hey all |
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| 04:26.03 | ChannelZ | say it like you mean it! |
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| 06:44.42 | DonX | Is there a way to receive mwi information on a PRI dahdi span? |
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| 07:05.18 | WIMPy | From/to where? |
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| 10:28.09 | tuxx- | hi guys. is it possible to nest asterisk functions like CUT? |
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| 10:38.30 | kaldemar | tuxx-: CUT takes a variable name instead of a string as an argument so nesting CUTs would be a no. |
| 10:38.53 | kaldemar | nesting functions in general is possible. |
| 10:38.54 | tuxx- | tnx. |
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| 10:59.39 | napnap | yesterday I talk about my problem. I have a OpenVox B400P card and all the led blink in red. I cheked my wires and I have this one : http://www.ordinoscope.net/index.php/Informatique/Th%C3%A8mes/R%C3%A9seaux/C%C3%A2blage_crossover under ISDN BUS-S(BRI) title |
| 11:00.14 | napnap | the four wires in the center are used.. |
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| 11:12.35 | napnap | do I change wires order, modify configuration :-\ , |
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| 11:21.57 | napnap | jameswf, what do you think about my wores order for a BRI line ? |
| 11:22.20 | *** join/#asterisk bulkorok (~Adium@85.183.61.47) |
| 11:23.15 | napnap | is not a RJ48 but it is need for my type of lines ? |
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| 11:53.00 | WIMPy | napnap: What are you trying to connect to that card? |
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| 11:56.29 | WIMPy | And why are you talking about crossover cables? |
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| 12:02.14 | michael_work | Penguin, i wanted to ask you if you are the same Penguin from Openrepos (meego/jolla community) or it's just same nick :) |
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| 13:11.27 | napnap | WIMPy, I trying to connect two T0 (EO) (BRI) lines |
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| 14:08.32 | wasanzy | hello |
| 14:09.20 | file | hi |
| 14:09.23 | dan_j | Hi, without fail every night at midnight, asterisk loses all sip registrations and stops responding to SIP packets. Nagios has witnessed this. However, ping tests are fine and the server seems to be running normally. |
| 14:09.36 | wasanzy | is there a command to show a status of an app in asterisk cmd? |
| 14:09.41 | dan_j | The outage lasts for about 30 seconds, and then it runs fine all day. |
| 14:09.56 | dan_j | Here is the debug from last night, but not really sure what i'm looking for. |
| 14:09.57 | dan_j | http://pastebin.com/eRDpV5aV |
| 14:10.18 | dan_j | All sip peers went offline at 00:01:25 |
| 14:11.17 | WIMPy | napnap: You doun;t need a crossover cable under any normal circumstances. |
| 14:12.29 | file | dan_j, that post is private |
| 14:12.36 | file | wasanzy, can you elaborate on what you mean by status? |
| 14:13.14 | dan_j | argh. 1sec |
| 14:13.33 | dan_j | ok. try now. |
| 14:13.51 | jeffspeff | on a fresh install, not all modules are being loaded when * starts, specifically chan_sip and chan_iax2. from the console, I can manually load the module and it works, but doesn't automatically load after i restart *. i have checked modules.conf and autoload is on, sip and iax modules aren't set to noload |
| 14:14.51 | file | "module show" these days will tell you if a module is running (loaded fine with sane enough configuration), otherwise check your console output at startup |
| 14:15.44 | file | dan_j, looks like your database is being slow to respond and due to the way chan_sip is done that will block other stuff |
| 14:16.25 | dan_j | is it normally to have zero log data for around 1 minute? Between 00:00:10 and 00:01:20 |
| 14:17.03 | file | if the database blocks, chan_sip will block |
| 14:17.49 | dan_j | Ok. I'll take a look at the database, however nagious hasnt reported any issues with that. |
| 14:17.50 | dan_j | Thanks |
| 14:17.52 | wasanzy | file: I have this app ControlPlayback and I want to check if it is working, but didn't want to use dialplan |
| 14:18.25 | file | wasanzy, the only way to know is to try it. |
| 14:19.09 | wasanzy | ok |
| 14:20.30 | DruidZ | I have a client with no sound. Usually a NAT issue, right? |
| 14:20.38 | file | DruidZ, usually yes |
| 14:20.49 | dan_j | DruidZ: was it working before? |
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| 14:21.02 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:21.22 | DruidZ | Various pages talk about sip_nat.conf but the server is not behind a firewall so that is irrelevant I think. |
| 14:21.32 | dan_j | DruidZ: SIP ALG on some routers can also mess things up with one-was sound. |
| 14:21.36 | DruidZ | It works and then stops working. |
| 14:21.50 | dan_j | Reboot their router, and then see if the sound comes back. |
| 14:22.48 | jeffspeff | file, if i start asterisk using "asterisk -cv" then all modules work fine. If i start asterisk using "service asterisk start" then they don't all load |
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| 14:24.28 | DruidZ | They have an SPA-122 vers. 1.2.???. Should I upgrade to the latest 1.3.3 (015)? |
| 14:25.28 | DruidZ | They have no sound in either direction. |
| 14:26.26 | DruidZ | Am I correct in assuming that sip_nat.conf is irrelevant since I am not behind a NAT firewall? |
| 14:26.44 | dan_j | Is the SPA behind a nat? |
| 14:26.52 | DruidZ | Yes. |
| 14:27.21 | DruidZ | And my phone on the same switch works so it must be at his end. |
| 14:27.26 | napnap | WIMPy, anyway , I just want to know why led are in red alarm. A wires order problem ? (What this card expect ?) (I use this cable on another asterisk box) Or it is possible that the problem comes from my configuration ? ...:-\ |
| 14:27.35 | dan_j | Its probably a nat issue, but sip_nat.conf is irrelevnat |
| 14:28.02 | dan_j | When you say 'the same switch' do you mean, the same switch as the asterisk box, or as the spa? |
| 14:28.08 | DruidZ | Right. Do you know of any pages that discuss this issue from the SPA side? |
| 14:28.33 | DruidZ | We are both registered to the same * server. |
| 14:28.55 | dan_j | Thats not what I asked. |
| 14:29.11 | dan_j | The asterisk server has a public IP, correct? |
| 14:29.27 | DruidZ | Oh, we are in different cities. |
| 14:29.33 | DruidZ | Yes. |
| 14:30.01 | file | jeffspeff, this sounds vaguely familiar but nothing springs to mind |
| 14:30.01 | dan_j | Ok. I would firstly check his router and look for any firewall settings that could be blocking the audio |
| 14:30.31 | DruidZ | The weird thing is that it works sometimes. |
| 14:31.08 | dan_j | You should be able to set the RTP ports on the SPA, and then map the ports on the router to force it to work. |
| 14:31.21 | dan_j | However, it sounds like a dodgy router if it works sometimes. |
| 14:31.40 | dan_j | Did you try rebooting the router as I said? |
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| 14:44.40 | DruidZ | Doing that now. |
| 14:44.46 | DruidZ | Brand new Linksys. |
| 14:45.36 | DruidZ | Should I disable SPI on the router? |
| 14:46.31 | *** join/#asterisk rmudgett (~rmudgett@nat/digium/x-jfcfvemnoxpntuhn) |
| 14:47.58 | [TK]D-Fender | SPA != ALG |
| 14:48.03 | [TK]D-Fender | Shouldn't need to touch that |
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| 14:49.28 | DruidZ | OK. So now it doesn't even ring and I get "Retransmission timeout reached on transmission" in my log. |
| 14:49.47 | dan_j | Restart the SPA now. |
| 14:50.21 | DruidZ | Just did. |
| 14:55.17 | dan_j | [TK]D-Fender: SPA != ALG ?? ALG can mess with anything SIP. |
| 14:56.00 | DruidZ | ALG is off. |
| 14:56.25 | dan_j | Ok. Then check the firewall settings on the router. |
| 14:56.43 | dan_j | In sip.conf what do you have NAT= ? |
| 14:58.11 | DruidZ | nat=force_rport,comedia |
| 14:58.54 | [TK]D-Fender | SPA rather |
| 14:58.58 | [TK]D-Fender | SPI* |
| 14:59.01 | [TK]D-Fender | did it again |
| 14:59.03 | [TK]D-Fender | gah |
| 14:59.07 | [TK]D-Fender | [10:45]DruidZShould I disable SPI on the router? <- no |
| 14:59.57 | [TK]D-Fender | [10:49]DruidZOK. So now it doesn't even ring and I get "Retransmission timeout reached on transmission" in my log. <- logs = trash. live CLI only. "sip show settings" <- PB first, then a call attempt "sip set debug on" <- |
| 15:01.37 | DruidZ | Sorry, I meant in CLI. |
| 15:03.33 | *** part/#asterisk bulkorok (~Adium@85.183.61.47) |
| 15:04.59 | DruidZ | So what setting[s] are you interested in? |
| 15:05.43 | [TK]D-Fender | all of them |
| 15:08.33 | DruidZ | http://www.vex.net/~darcy/asterisk_settings.txt |
| 15:09.52 | [TK]D-Fender | <PROTECTED> |
| 15:10.03 | [TK]D-Fender | you have clearly not set your system up to work from behind NAT |
| 15:10.56 | DruidZ | Right. That's because I am not behind a NAT. |
| 15:11.17 | DruidZ | My user is but my server is not. |
| 15:12.19 | DruidZ | SIP works entirely on UDP, right? |
| 15:12.26 | [TK]D-Fender | So there is a public IP directly on that box? |
| 15:13.41 | DruidZ | Yes. |
| 15:14.02 | [TK]D-Fender | ok, pastbein the peer in question masking only the scret and a failed call |
| 15:14.03 | DruidZ | It sits right on the Internet backbone. |
| 15:17.54 | napnap | I understood that the B400P card with E0 lines can work with DAHDI but perhaps I should see on the side of mISDN or bristuff ? |
| 15:18.03 | napnap | work with * |
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| 15:20.47 | [TK]D-Fender | napnap: Whose card? |
| 15:26.36 | napnap | [TK]D-Fender, OpenVox 400P |
| 15:27.27 | napnap | (I use asterisk 1.8.11 under Elastix) |
| 15:29.22 | [TK]D-Fender | They are supported by DAHDI.... |
| 15:29.33 | [TK]D-Fender | and I don't think E0 is the term you're looking for... |
| 15:31.44 | napnap | as I understood T0=E0(european notation) |
| 15:32.02 | Katty | wants a napnap :< |
| 15:32.59 | napnap | when I connect to my asterisk console, I have dahdi command, pri command (I think I sould have bri * command and not pri commands no ?) |
| 15:33.48 | napnap | which explain why the card are in red alarm because wires order of PRI != BRI |
| 15:33.53 | napnap | :-\ |
| 15:34.03 | napnap | I swim |
| 15:35.06 | Katty | i run. |
| 15:35.11 | Katty | but swimming is fun too! |
| 15:35.15 | mirela666 | i drink |
| 15:35.21 | Katty | Nugget: did you see my toasty run yesterday afternoon? |
| 15:36.50 | Nugget | nope, missed it |
| 15:37.24 | *** join/#asterisk cbwest (~cbwest@rrcs-96-10-75-242.se.biz.rr.com) |
| 15:37.32 | Nugget | looks |
| 15:38.15 | Nugget | TOAST! |
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| 16:07.37 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 16:09.26 | *** join/#asterisk HelHound (~hellhound@mgmt.ngcommunications.com) |
| 16:10.43 | HelHound | hi all, is there a way to disable device stats in 1.4 ? We're using SIP/<number>@termination-server as members, and states don't seem to get registered correctly |
| 16:13.14 | Qwell | umm |
| 16:13.17 | Qwell | ~asterisk versions |
| 16:13.17 | infobot | Asterisk versions and their support levels are documented at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 16:14.31 | HelHound | ok, i'm using 1.8, can i disable device states ? *g* |
| 16:15.33 | HelHound | or has this issue been resolved with other (higher) versions ? I understand that normally you use SIP endpoints, but not in our case |
| 16:15.59 | *** join/#asterisk cmendes0101 (~cmendes01@pool-96-251-59-96.lsanca.fios.verizon.net) |
| 16:16.27 | *** join/#asterisk navaismo (~navaismo@200-52-45-221.dynamic.axtel.net) |
| 16:16.28 | *** join/#asterisk illuder (illuder@105-236-34-179.access.mtnbusiness.co.za) |
| 16:16.56 | HelHound | And can't find any information on device states with using SIP/<number>@server members |
| 16:18.09 | *** join/#asterisk cmendes0101| (~cmendes01@pool-96-251-59-96.lsanca.fios.verizon.net) |
| 16:18.44 | *** join/#asterisk bkruse (~Adium@24.42.229.8) |
| 16:21.23 | [TK]D-Fender | [12:16]HelHoundAnd can't find any information on device states with using SIP/<number>@server members What states? What members? Show us actual debug, configs, and status dumps/... |
| 16:22.20 | *** join/#asterisk Ferdroid (~Fershun@unaffiliated/ferdroid) |
| 16:27.18 | HelHound | [TK]D-Fender: example of a member with a Ring+Inuse state in a queue, where there are no connected or waiting calls in the queue, and member/number is available: Klantenservice (SIP/1*08052XXXXXXXXXX@trunk-out) (realtime) (Ring+In use) has taken no calls yet |
| 16:27.52 | [TK]D-Fender | You cannot do states on devices like that |
| 16:28.05 | [TK]D-Fender | You COULD if you used a Local channel instead |
| 16:28.23 | HelHound | ok, figured as much, is there a way to 'disable' it, so that the queue will always call members even if they are busy ? |
| 16:28.25 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 16:29.00 | HelHound | [TK]D-Fender: and device state will not be used when using Local channels ? |
| 16:29.10 | HelHound | or will just return 'Not in use' state |
| 16:29.11 | [TK]D-Fender | If you allow ring-in-use on the queue itself it shouldn't track them |
| 16:29.20 | [TK]D-Fender | (or care about the result) |
| 16:30.34 | HelHound | mmm.. we have ringinuse=yes in the queue.conf in the general context.. but queue does not send calls to members |
| 16:30.42 | michael_work | hey |
| 16:30.54 | [TK]D-Fender | always make your settings explicit in the queue you're working with |
| 16:31.13 | [TK]D-Fender | Then provide actual completee configs, debug, etc so we can see what's really happening. |
| 16:31.17 | michael_work | is there any varibales used or other ways to detect if calls had attended transfer? |
| 16:31.20 | HelHound | ok, will check that out. And the ringinuse=yes will work even if the members are in 'Ring+In use' state ? |
| 16:32.00 | HelHound | because the calls get correctly sent to members who are in 'In use' state, but not when 'ring+In use' |
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| 16:40.02 | [TK]D-Fender | HelHound: Show all of it |
| 16:42.42 | [TK]D-Fender | michael_work: Not really. |
| 16:45.20 | michael_work | [TK]D-Fender :( |
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| 17:19.58 | jeffspeff | if i start asterisk using "asterisk -cv" then all modules work fine. If i start asterisk using "service asterisk start" then they don't all load |
| 17:22.29 | newtonr | jeffspeff, do all of your modules have the same permissions? |
| 17:22.41 | jeffspeff | yes |
| 17:23.04 | jeffspeff | this is a fresh compile |
| 17:23.40 | newtonr | You might verify anyway, if you had an old installation there before. What do the verbose and debug messages say about why the modules won't load when you start it with service? |
| 17:24.43 | jeffspeff | it's a fresh OS as well. no previous versions have been on this machine. when i start * via service asterisk start i don't get any output in /var/log/asterisk/messages |
| 17:26.08 | jeffspeff | when started as a service like that, i can manually start the module from the * cli and the module loads fine |
| 17:26.58 | newtonr | what about permissions on your .conf files? maybe something goofy with modules.conf ? |
| 17:31.23 | WIMPy | napnap: No. Asterisk doesn't make a difference between BRI and PRI. It's all called PRI. |
| 17:31.31 | navaismo | Anyone have hints on this: Asterisk's extension receive a call from avaya sip trunk, the asterisk's extension will transfer the call to an IVR when the asterisk's extension transfer the call the Avaya number hear the first audio then the call is Dropped. |
| 17:31.55 | WIMPy | napnap: And I prefer to use mISDN for several reasons, but it won't make your alarm go away. |
| 17:32.30 | navaismo | IN the sip debug i can see the REFER from the asterisk's extension to the IVR then the accept then the playback of the sound and suddenly the read application said "User disconnect" and the call terminate |
| 17:33.10 | jeffspeff | newtonr, all privs are the same for all .conf files |
| 17:33.29 | napnap | WIMPy, ok...for now I go to home... I continue tomorow.. I hope I will fix that...Thanks for your answer... |
| 17:34.16 | WIMPy | napnap: And make sure to use a normal cable. |
| 17:41.52 | *** join/#asterisk brittel (~jack@542713e4.ftth.concepts.nl) |
| 17:42.52 | brittel | Hi, I have an issue with my asterisk box not being able to recongnize busy signals/ tones (and others) on certain networks |
| 17:43.17 | brittel | is there a way to manually change this for certain networks so it can recognize them? |
| 17:45.57 | WIMPy | Go digital and forget about fecognizing tones. |
| 17:47.42 | brittel | This is SIP -> Mobile |
| 17:48.19 | WIMPy | Then there is something wrong on the other side of your ITSP. |
| 17:48.33 | brittel | For example, when calling MTN network, the remote party rings forever, but thats not being triggered |
| 17:48.59 | brittel | It simply times out with any indication |
| 17:49.24 | WIMPy | What is not being triggered? |
| 17:49.36 | brittel | The ringing indication |
| 17:49.57 | brittel | When calling a handset on a different network, i'm getting a ringing indication from the remote party |
| 17:50.17 | WIMPy | Side effects of everyone going VOIP. Things just don't work any more. |
| 17:50.35 | brittel | True |
| 17:51.59 | brittel | However, the remote party is sending the tones; no way to tweak the tone handling? |
| 17:52.15 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 17:52.24 | WIMPy | Tones aren't handled. |
| 17:52.38 | WIMPy | All signalling is out of band. |
| 17:53.00 | brittel | in the pure digital world that is |
| 17:53.45 | WIMPy | There isn't anything else except for remaining subscriber lines. |
| 17:57.56 | *** join/#asterisk apb1963 (~apb@174.134.232.101) |
| 17:58.59 | apb1963 | Good morning all! Anyone know how to setup a Samsung Galaxy S4 to do 3 way calling by way of Asterisk? |
| 17:59.22 | apb1963 | I'm trying to help someone else... |
| 17:59.25 | *** part/#asterisk cbwest (~cbwest@rrcs-96-10-75-242.se.biz.rr.com) |
| 18:00.38 | WIMPy | Where (or what) is the link between the S4 and Asterisk? |
| 18:00.48 | apb1963 | I found a youtube video that describes how to set it up for voip in general... but no mention that I could find, of how to do 3 way. |
| 18:01.43 | WIMPy | There's no such feature. |
| 18:02.12 | WIMPy | Either you use Asterisks "Features" or you transfer the calls to a conference. |
| 18:03.00 | WIMPy | Terminal controlled 3way calling is not possible. |
| 18:03.34 | WIMPy | The terminal might offer you such a function, try to check its manual. |
| 18:04.10 | apb1963 | ok, all of these sound like possibilities |
| 18:06.00 | navaismo | Now i remmeber why igceweiling fly away from the irc :'( |
| 18:06.45 | apb1963 | So assuming the asterisk feature code... she'd just hit *whatever and then dial a second number essentially? Probably documented somewhere... I'll google. |
| 18:06.46 | apb1963 | thank you WIMPy |
| 18:07.24 | WIMPy | Err, no, that;s just another option to transfer calls. |
| 18:20.21 | *** join/#asterisk gusto (~gusto@2a02:810d:8600:8d4:21b:63ff:fe31:8426) |
| 18:33.09 | navaismo | [TK]D-Fender, do you have time to take a look on this debug-->http://pastebin.com/Gww4mQVF My brain is dry after many days call hangup after attended transfer |
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| 19:29.55 | jmls | I'm trying to use the Newstate ami event to track calls between extensions. |
| 19:31.30 | jmls | so, 502 (julian_smith) calls 503 (snom), 503 answers. if someone could look at http://pastebin.com/davL9V1Y and tell me why connectedlinenum on the snom event is <unknown> |
| 19:32.31 | jmls | (sorry, the 2nd event is what I meant to say) |
| 19:33.08 | jmls | the first event is the "up" event for 503, and the second is the "up" event for 502 |
| 19:43.00 | jeffspeff | when using centos 64bit, should asterisk compile the modules into /usr/lib/asterisk or /usr/lib64/asterisk ? i'm finding that if i don't specify the libdir during ./configure then it installs to /usr/lib but looks for libasteriskssl.so in /usr/lib64 |
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| 20:01.56 | Penguin | jeffspeff: I would see if there is a 64-bit RPM spec file. I think you'll find the configure options listed there. |
| 20:03.30 | jeffspeff | Penguin, i'm compiling from source, not using an rpm |
| 20:03.36 | Penguin | I understand that. |
| 20:04.26 | Penguin | I knew that before I gave my suggestion. |
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| 20:10.35 | *** join/#asterisk navaismo (~navaismo@189.146.39.29) |
| 20:18.52 | jeffspeff | i think the issue might be related to this http://pastebin.com/DR3wZ7jR any idea how to fix that or manually set it? |
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| 20:22.28 | jwr__ | Is anyone running asterisk/freepbx, with a pci analog card, in a virtual machine? I can't think of any reason why I couldn't do that, is there one? |
| 20:22.54 | DruidZ | Me again. Still having this sound problem. |
| 20:29.32 | *** join/#asterisk neonerz (~g_bove@cpe-076-182-115-090.nc.res.rr.com) |
| 20:29.34 | DruidZ | It is only with one client. |
| 20:29.43 | *** part/#asterisk neonerz (~g_bove@cpe-076-182-115-090.nc.res.rr.com) |
| 20:36.12 | gusto | what sound problem? |
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| 20:46.59 | DruidZ | He can call out just fine but no one can call in. |
| 20:47.42 | DruidZ | Except he can't call me. It rings but when I answer, no sound either way. |
| 20:47.54 | DruidZ | We are both on the same switch. |
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| 20:51.20 | DruidZ | Now he called my cell and there is no sound. It seems to be random. |
| 20:52.06 | DruidZ | I can call other people on the switch or not and other users are not having a problem. |
| 20:52.47 | DruidZ | When he calls me and the switch redirects to my cell it works in one direction. He can hear me but I can't hear him. |
| 20:53.42 | DruidZ | The switch is on the Internet proper, not behind a NAT. |
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| 21:26.35 | qakhan | hi all, i am trying to setup sip trunking, i can receive calls but i cannot make outbound calls |
| 21:27.18 | qakhan | here is my sip debug, http://pastebin.com/0rARVBb1 |
| 21:35.33 | Penguin | qakhan: Pastebin the STC-SIP peer from sip.conf. |
| 21:35.43 | Penguin | mask ONLY the password. |
| 21:36.38 | [TK]D-Fender | Shouldn't need |
| 21:36.51 | [TK]D-Fender | We see the failure right there |
| 21:37.02 | [TK]D-Fender | SIP/2.0 484 Address Incomplete <---- They don't like what you dialed |
| 21:37.29 | qakhan | http://pastebin.com/b5Kr7m0h |
| 21:38.07 | qakhan | Penguin what do you mean mask Password? |
| 21:38.22 | [TK]D-Fender | Your "user" entry is also pointless |
| 21:38.26 | Penguin | Some people paste their password in the pastebin. |
| 21:38.35 | [TK]D-Fender | [17:36][TK]D-FenderSIP/2.0 484 Address Incomplete <---- They don't like what you dialed |
| 21:38.36 | navaismo | Ok for the record, was the yealink phone using the asterisk feature it works |
| 21:38.38 | [TK]D-Fender | qakhan: ^^^^^^^^^^^ |
| 21:39.31 | qakhan | [TK]D-Fender if i remove username and secret i get same massage |
| 21:39.48 | [TK]D-Fender | qakhan: THEY DON'T LIKE THE NUMBER |
| 21:40.09 | Penguin | You need defaultuser (or username on antique asterisk) and secret... to be able to authenticate TO THE PEER when making a call. |
| 21:40.24 | [TK]D-Fender | it isn't an auth issue |
| 21:40.47 | Penguin | But if he takes away the defaultuser and secret there will be. |
| 21:40.53 | qakhan | but it works fine on elastix |
| 21:41.53 | [TK]D-Fender | Says nothing... and we don't see this working elsewhere |
| 21:42.04 | [TK]D-Fender | all we see is a very clear refusal to process that # |
| 21:49.11 | qakhan | [TK]D-Fender here is my elastix sip debug |
| 21:49.25 | qakhan | http://pastebin.com/ARCKUPGM |
| 21:49.57 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 21:50.20 | qakhan | this i am using sip trunk on other server that is way service is unavaiable on this. otherwise it is working |
| 21:50.31 | [TK]D-Fender | I elasitx... getting NO ANSWER at all |
| 21:50.43 | [TK]D-Fender | that does not look "fine" |
| 21:50.59 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 21:51.10 | qakhan | please ignor last part |
| 21:51.12 | [TK]D-Fender | Retransmitting #5 (NAT) to 10.200.7.157:5060: <--- repeat lack of response at all |
| 21:51.37 | [TK]D-Fender | I see nothing good to compare to there |
| 21:51.55 | qakhan | i told you service is not available |
| 21:51.57 | qakhan | on it |
| 21:52.30 | [TK]D-Fender | [17:40]qakhanbut it works fine on elastix <- you also said it was FINE |
| 21:52.53 | [TK]D-Fender | What was this 2nd call supposed to tell us..... it didn't work.... it didn't get refused differently.... it got IGNORED |
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| 21:53.27 | *** join/#asterisk yago (~kresp0@gateway/tor-sasl/kresp0) |
| 21:53.52 | qakhan | ok help me here, is there any way i can get sip trunk config from elastix |
| 21:54.29 | [TK]D-Fender | "get"? What is there to "get"? |
| 21:54.55 | qakhan | sip trunk works fine on elastrix |
| 21:54.57 | [TK]D-Fender | your first call got rejected, and your 2nd call didn't get any answer at all. |
| 21:55.17 | [TK]D-Fender | [17:54]qakhansip trunk works fine on elastrix <- We have not seen a good call yet |
| 21:56.00 | qakhan | 2nd call didnt get answer because i removed sip trunk connection from that |
| 21:56.09 | qakhan | to use on 1st one |
| 21:56.18 | qakhan | cable unplugged |
| 21:56.50 | [TK]D-Fender | You are not showing us anything useful here. |
| 21:57.08 | qakhan | ok i can show tomorrow |
| 21:57.35 | qakhan | i need someone to be there on remote side |
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| 23:27.43 | file | bananas! |
| 23:28.36 | navaismo | fried ^ |
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| 23:56.16 | Penguin | chocolate covered |
| 23:57.59 | navaismo | or strawberry jam |
| 23:59.10 | navaismo | http://3.bp.blogspot.com/-XBleVqz_csI/UaFNFrWjBTI/AAAAAAAABfg/JWJuO7wcMNY/s1600/P1040646.JPG |
