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| 02:32.39 | wh1t3c0d3r | Hi, I need help with my AsteriskNow installation. I have no audio when calling using two different SIP client. |
| 02:34.51 | wh1t3c0d3r | If I'm not in the correct channel, please tell where to go. |
| 02:35.00 | wh1t3c0d3r | *tell me |
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| 02:41.23 | WIMPy | Are all on the same network? |
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| 02:43.49 | wh1t3c0d3r | no, seperated. I think NAT is having trouble. |
| 02:44.02 | WIMPy | That's pretty normal. |
| 02:44.06 | WIMPy | ~sipnat |
| 02:44.06 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the wiki at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet. |
| 02:44.54 | wh1t3c0d3r | When directly connecing to it via the local network, it work. What I don't understand is it was working yesterday.. |
| 02:45.05 | wh1t3c0d3r | Sorry for my english |
| 02:46.02 | wh1t3c0d3r | I've tried enabling the NAT via the SIP setting page and in the extension's configuration |
| 02:46.14 | wh1t3c0d3r | but no luck |
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| 02:54.57 | dan_j | Hi. Is it possible to specify two IP addressed in sip.conf bindaddr ? |
| 02:55.18 | WIMPy | no |
| 02:56.06 | dan_j | oh dear. Ok, next question, if ive got a floating IP which i want to listen on, and also want to listen on 127.0.0.1. Is it possible? |
| 02:56.17 | dan_j | ie. i dont want to listen any one other public ips. |
| 02:56.23 | WIMPy | You can listen on one or all. |
| 02:56.49 | WIMPy | iptables is your friend then. |
| 02:57.08 | WIMPy | Or nftables if your're state of the art. |
| 02:57.30 | dan_j | I know i could just block it, but asterisk keeps putting the wrong ip in the sip dialogs |
| 02:58.08 | WIMPy | Yes. chan_sip doesn't like to be used on a multi-homed system. |
| 03:00.57 | wh1t3c0d3r | no much help in the links you gave me. First one sent a 404 and the second does not contains new info |
| 03:01.36 | wh1t3c0d3r | Nat is already set to yes, localnet and directmedia are set too but still no audio (I restarted the service) |
| 03:01.51 | WIMPy | google should have lots of info on the topic. |
| 03:02.04 | WIMPy | But sometimes it just doesn't work. |
| 03:02.10 | wh1t3c0d3r | port forwarding is not a problem since the server is using an ip behing a DMZ.. |
| 03:02.12 | WIMPy | It really depends on the NATting device(s). |
| 03:02.46 | wh1t3c0d3r | yeah but I still does not see why it stopped working without modification.. |
| 03:03.03 | wh1t3c0d3r | both on the router and the server. |
| 03:03.18 | WIMPy | Any dynamic IPs involved? |
| 03:03.27 | wh1t3c0d3r | no, static ips |
| 03:03.46 | WIMPy | Or anyone else (re)using some port? |
| 03:04.33 | wh1t3c0d3r | no, I don't think, the external ip is unique and all data sent to it is forwarded to the server |
| 03:05.12 | wh1t3c0d3r | two ip on the router, one is for general purpose and the other is dedicated |
| 03:05.51 | wh1t3c0d3r | I've tried restarting the server, the router and even the device and computer connecting to the server |
| 03:06.18 | wh1t3c0d3r | I'm thinking about reinstalling... |
| 03:06.40 | wh1t3c0d3r | but want to avoid that.. |
| 03:07.27 | WIMPy | It's unlikely to help. |
| 03:08.34 | wh1t3c0d3r | all the settings is done via the freepbx admin panel. |
| 03:09.12 | dan_j | i think i'm going to have to rewrite the pacemaker RA that i'm using so that it doesnt need to use 127.0.0.1, then i can force asterisk to bind to one ip. |
| 03:09.36 | dan_j | If asterisk is bound to one IP, does that guarantee that it wont put a different IP in the sip packets? |
| 03:11.17 | WIMPy | NFI. But you might have to turn off ICE. |
| 03:11.48 | dan_j | Is that for me? |
| 03:11.55 | WIMPy | yes |
| 03:12.07 | dan_j | NFI? |
| 03:12.21 | WIMPy | No F***ing Idea |
| 03:12.33 | dan_j | ah |
| 03:12.50 | dan_j | Ok. looks like ive got more testing to do. |
| 03:12.51 | dan_j | thanks |
| 03:17.40 | wh1t3c0d3r | hum.. I will reinstall, nothing is blocking any packets. |
| 03:17.56 | wh1t3c0d3r | thanks for time trying to help me |
| 03:18.05 | wh1t3c0d3r | *for your time |
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| 09:23.58 | StaRetji | howdy folks |
| 09:24.12 | StaRetji | can someome confirm if * is working as DTMF tone in Asterisk? |
| 09:24.37 | StaRetji | we tried activating promotion on sim card, over sip, using DTMF and operator requires pressing * |
| 09:24.53 | StaRetji | however, it seems it is not working, while digits work |
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| 09:46.40 | buhman | if asterisk servers have device state distribution configured correctly, how is information about which asterisk server is currently available distributed to devices? |
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| 14:33.51 | dan_j | Hi. I'm trying to get named pickup groups to work, but it appears that the pickupexten feature doesnt look at the named pickup groups. Can that be right? |
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| 15:24.16 | StaRetji | folks, can asterisk send * dtmf ? |
| 15:27.27 | [TK]D-Fender | yes |
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| 15:30.02 | StaRetji | ok, thanks |
| 15:30.27 | StaRetji | then I must be doing something wrong, I send 0-9 normaly, but * not recognized for some reason |
| 15:30.41 | StaRetji | I will debug, just needed to know if it is supported or no |
| 15:31.24 | [TK]D-Fender | Shouldn't have to ask |
| 15:31.48 | [TK]D-Fender | There's dialplan patterns for that sort of thing, and you can't go and not support those 2. |
| 15:31.55 | [TK]D-Fender | A-D is another funny story... |
| 15:32.54 | dan_j | StaRetji: could it be triggering a feature, and therefore not forwarding the * onwards? |
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| 15:33.32 | StaRetji | hm, could be, thanks for the tip dan_j |
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| 16:29.00 | navaismo | ~Valar Morghulis |
| 16:29.43 | marsmars | Hello, I have got an modified router with Asterisk 11.11.0 and the IAX2 packages does not receive to the asterisk instance, but they are registrated by dtrace on the lan interface at the Router. netstat -nlp says, that the asterisk server has got the udp socket at port 4569 Waht could be wrong? |
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| 16:33.20 | [TK]D-Fender | marsmars: Show us |
| 16:33.21 | [TK]D-Fender | ~pb |
| 16:33.22 | 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. |
| 16:33.23 | [TK]D-Fender | ^^^ |
| 16:39.59 | *** join/#asterisk Mango45 (~Mango45@d209-89-208-218.abhsia.telus.net) |
| 16:42.18 | Mango45 | Is there a way to disable transcoding in Asterisk, if there is a codec that's common to both peers? |
| 16:42.42 | Mango45 | Actually, let me ask that a different way. |
| 16:44.02 | Mango45 | If my phone supports G.722 and G.711, and a carrier only sometimes supports G.722, what will happen for calls in which it doesn't? |
| 16:44.51 | ChkDigit | They will pick the most preferred codec that they share. |
| 16:45.50 | ChkDigit | call by call. |
| 16:47.13 | [TK]D-Fender | Also depends if the call has already been answered by * or not, who has what for preferences, and whose order takes precedence |
| 16:48.35 | Mango45 | Assume * has not answered the call, and directmedia and directrtpsetup are turned on. |
| 16:49.02 | Mango45 | Some users in http://www.dslreports.com/forum/r29246600-Asterisk-How-to-avoid-unwanted-transcoding-to-from-G.722- this thread claim that Asterisk insists on transcoding from G.722 to G.711, even though the phone supports G.711. |
| 16:50.28 | ChkDigit | Do you mean the phone supports G.722 and G.711? |
| 16:50.40 | [TK]D-Fender | [12:48]Mango45Assume * has not answered the call, and directmedia and directrtpsetup are turned on. <- last 2 don't matter |
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| 16:50.49 | ChkDigit | G.722 -> G.711 would need a transcode. |
| 16:52.02 | Mango45 | ChkDigit, yes the phone supports both. The goal is to pick the most preferred codec that both peers share, though apparently that doesn't happen. |
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| 16:54.39 | mjordan | Mango45: generally, Asterisk attempts to avoid transcoding. In some situations, however, there may be a reason why that is difficult. The SIP_CODEC function lets you pick which codec you want to offer. Using a pre-dial handler, you could control that however you want. |
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| 16:55.47 | mjordan | And, just to make things more fun: generally, attempting to always avoid transcoding is not what everyone wants. If you really want to avoid transcoding, then you should set the preference order of codecs on your peers/endpoints to the order that you want |
| 16:56.00 | mjordan | And then use the dialplan to set/restrict things further if desired |
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| 16:58.01 | Mango45 | Out of curiosity, why would you prefer to transcode, if both peers have a common codec? |
| 16:58.21 | Mango45 | Why not just pick the first preference that is common to both? |
| 16:58.35 | mjordan | bandwidth. |
| 16:58.48 | Mango45 | Ah. |
| 16:58.51 | mjordan | There are situations where you want to use g729, for example, because you know the network is getting congested. |
| 17:03.01 | marsmars | I have post some conf and logs at http://pastebin.com/7egN8GjG |
| 17:04.52 | [TK]D-Fender | marsmars: So far * is seeing a call there.... |
| 17:05.13 | [TK]D-Fender | marsmars: kill the code debug and include verbose on your next attempt |
| 17:06.34 | *** join/#asterisk Penguin (~xwQ5kwYl6@20264.odci.gov.united-states.rltk.us) |
| 17:08.45 | marsmars | TK D-Fender: new version with verbose: http://pastebin.com/7egN8GjG |
| 17:10.06 | Penguin | I thought the order of peer definition entries in sip.conf was important. I created a type user before a type peer and incoming calls still use the second entry (the peer instead of the user). The user and peer both use the same host. |
| 17:10.30 | Penguin | Does type peer have precedence over type user regardless of order in sip.conf? |
| 17:11.21 | [TK]D-Fender | marsmars: tail -f /var/log/asterisk/debug <-- LIVE CLI only.. forget "logs" |
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| 17:16.19 | Penguin | Actually, the problem isn't related to type user vs peer. It's still doing the same thing when changed to type peer. |
| 17:16.48 | Penguin | So the new question... doesn't order of peer entries in sip.conf matter? |
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| 17:22.42 | marsmars | TK -Fender: http://pastebin.com/WQcvtZLd |
| 17:24.25 | marsmars | TK D-Fender: I have to change the location. I'm drving home. The akku should be ok, please ask or give hints, but i cound not direct answer... |
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| 17:57.00 | igcewieling | Has anyone experieced weird issues like some commands in the Asterisk CLI causing the CLI to stop working correctly. See http://pastebin.com/QSVtkYc0 for an example, see how the second sip show peers doesn't work. Asterisk 11.11.0 / DAHDI 2.9 / Wanpipe 7.0.10 |
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| 18:05.45 | newtonr | do calls stop getting processed after that happens? |
| 18:05.54 | igcewieling | newtonr: yes. |
| 18:07.04 | newtonr | sounds like a deadlock maybe. probably want to grab the info described here and create a bug report on JIRA if you can reproduce the issue https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace#GettingaBacktrace-GettingInformationForADeadlock |
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| 18:18.41 | igcewieling | I'll see what I can do, but it might be a moot issue, I suspect they will leave us. |
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| 18:28.42 | marsmars | TK D-Fender: my car is broken, seem it would be late comming home |
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| 19:23.50 | dan_j | Hi, how is the g729 license tied to a server? if i've got two servers, with a floating ip, and only one server with asterisk running at a time, do i require licenses for both? |
| 19:24.15 | Qwell | dan_j: It is tied directly to the hardware. You require licenses for both, yes. |
| 19:24.47 | Qwell | That's just the nature of HA. 2 of everything - even licenses. |
| 19:25.04 | dan_j | Ok. I was hoping it was tied to an ip address :-) |
| 19:25.24 | dan_j | Is it possible to transfer licenses from an old machine? |
| 19:26.16 | Qwell | You'd have to ask Digium. |
| 19:26.21 | dan_j | ok. |
| 19:26.22 | dan_j | Thanks |
| 19:26.35 | Qwell | Short version though: Not really. |
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| 19:48.42 | shido6 | how many do you have? :) |
| 19:48.48 | shido6 | licenses... |
| 19:49.01 | dan_j | Only a handful |
| 19:49.05 | dan_j | about 10 |
| 19:50.16 | shido6 | I would talk to âem before buying another license |
| 19:50.41 | shido6 | do u get eyeballs on what it is you do? |
| 19:50.57 | WIMPy | Buy a transcoder card next time. That should be easy to transfer to a new server. |
| 19:51.18 | Qwell | WIMPy: $100 vs $800 (or whatever it costs)... |
| 19:51.52 | dan_j | I think it'll be ok. I'm actually moving from one VM to another, so i can simply decommission the old server, change the MAC on the new server to match the old server, then move the licenses over. |
| 19:51.56 | WIMPy | Sometimes you get the cards for less than the same number of software licenses. |
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| 19:54.26 | dan_j | I assume those cards don't work too well when using guest VMs to run asterisk. |
| 19:54.40 | igcewieling | dan_j: You can "reinstal" the license a few times, so you can move to a new machine it once or twice |
| 19:56.01 | shido6 | dan_j: not a fan of iommu? |
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| 20:05.33 | Katty | Qwell: this is your fault. |
| 20:05.36 | Katty | Qwell: YOUR FAULT. |
| 20:06.16 | igcewieling | It is Seg's Fault! |
| 20:07.54 | Qwell | Katty: most things are |
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| 20:11.08 | Katty | hugs Qwell |
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| 20:23.06 | twanny796 | I need to compile dahdi and then export it to another system. I managed to do this with DESTDIR=/root/destDir, then make a tar file and extract in / of the other system. However the module is not loaded and /dev/dahdi is not created. Anyone done this? |
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| 20:45.06 | navaismo | Create an RPM or deb or whatever package the system use |
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