| 00:00.01 | snadge | thanks wimpy.. then i just found this: https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Codecs+and+Audio+Formats |
| 00:00.14 | snadge | so apparently older versions of asterisk support both 8 and 16khz.. and thats it.. i can experiment with this |
| 00:00.39 | snadge | if it will automatically halve the sampling rate of 16khz to 8khz.. then we can use 16khz files, and that will sound better on g722 or g711.1 codecs |
| 00:00.49 | snadge | i dont think anyone actually uses g711.1 though.. i would have to check |
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| 01:21.58 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.14.0 (2017/02/13), 11.25.1 (2016/12/08), Standard: 14.3.0 (2017/02/13); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 02:13.01 | igcewieling | Anyone see what is wrong? Mar 8 21:09:24] ERROR[8796][C-0000001b]: chan_sip.c:17341 get_pai: Bad PAI header: <sip:9739462289@152.188.40.167;user=phone>. |
| 02:16.40 | overyander | ;user=phone shouldn't be within that tag should it? |
| 02:26.33 | igcewieling | Good point. I'll have to double check. |
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| 02:40.30 | igcewieling | the source IP appears to be owned by a company which has some old UUNET address blocks and is a carrier / tower management company. Just what we need, invalid headers from carriers. |
| 02:41.41 | igcewieling | ATT Wireless. joy. |
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| 04:09.59 | Samot | user=phone <- Remote Party Header attribute. |
| 04:10.20 | Samot | chan_sip.c:17341 get_pai: Bad PAI header: <-- != PAI header. |
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| 08:49.21 | Haris | hello all |
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| 09:14.06 | Haris | Can we configure multiple SIP trunks or exchanges or remote SIP peers for call routing in asterisk ? |
| 09:14.14 | Haris | its a dumb Q. but still asking |
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| 09:27.08 | Haris | how to check queues via cli on 11 ? |
| 09:27.14 | Haris | show queues doesn't seem to work |
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| 09:30.50 | *** mode/#asterisk [+o danjenkins] by ChanServ |
| 09:57.25 | *** join/#asterisk wasanzy (~wasanzy@41.66.254.58) |
| 09:57.28 | wasanzy | hello |
| 09:58.01 | wasanzy | does asterisk 11.12.0 supports web sip client? |
| 09:59.02 | wasanzy | I mean using WebRTC |
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| 10:57.13 | Haris | Is there a nagios plugin or tool to monitor asterisk sip trunks or sip peers ? |
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| 11:38.50 | wasanzy | Haris: https://exchange.nagios.org/directory/Plugins/Network-Protocols/*-VoIP/SIP/check_sip/details and https://exchange.nagios.org/directory/Plugins/Telephony/Asterisk/check_asterisk_peers/details |
| 11:38.57 | wasanzy | I use those |
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| 12:09.29 | sotoz | Hello |
| 12:09.31 | Haris | hmm |
| 12:10.17 | sotoz | I'm trying to use the rfc2833 dtmf_mode with pjsip but I get the "ERROR[17236]: config_options.c:738 aco_process_var: Error parsing dtmf_mode=rfc2833 at line 46" message. |
| 12:10.49 | sotoz | If I switch to rfc4733 it works but I'm not sure if the operators that I'm using support rfc4733. |
| 12:11.32 | sotoz | They say they only support rfc2833 and I'm not sure if those two are fully compatible. I know that 4733 obsoleted 2833 but what that means in terms of compatibility? |
| 12:28.25 | sotoz | anyone ? ^^ :) |
| 12:34.21 | *** join/#asterisk MacroMan (~MacroMan@host213-123-31-77.in-addr.btopenworld.com) |
| 12:35.27 | Haris | how to check queue on 11 ? |
| 12:35.45 | Haris | on cli |
| 12:35.52 | MacroMan | queue show |
| 12:36.08 | MacroMan | or `queue show queuename` |
| 12:36.11 | Haris | rasterisk -x "show queue" or rasterisk -x "sip show queue" doesn't work |
| 12:36.44 | MacroMan | `queue show` not `show queue` |
| 12:37.29 | Haris | great |
| 12:37.33 | Haris | works |
| 12:38.44 | Haris | is there a way to check queue config ? |
| 12:38.47 | Haris | on cli |
| 12:39.19 | Haris | ah, queue show rules is empty |
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| 12:40.53 | MacroMan | I'm getting calls dropping several minutes after being bridged. Here is my dialplan: https://paste.ngx.cc/bd3c8cbf9997c7e9 and my call log: https://paste.ngx.cc/a8372db62805fe20 |
| 12:41.43 | MacroMan | The last line in the call log is when the call drops |
| 12:43.01 | MacroMan | Not really sure what it means by abnormal exit from the gosu |
| 12:43.06 | MacroMan | gosub* |
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| 12:47.55 | Haris | has 0 calls (max unlimited) in 'rrmemory' strategy (3s holdtime, 136s talktime), W:0, C:104, A:16, SL:100.0% within 60s <--- what does this output mean ? |
| 12:48.15 | Haris | I understand the 0 calls, rrmemory part. but what does the holdtime and subsequent things mean |
| 12:48.26 | Haris | W, C, A, SL |
| 12:49.13 | MacroMan | I think W is calls waiting, A is calls answered, SL is service level. Don't know what C is |
| 12:49.40 | MacroMan | I think. I don't fully know as I don't use those myself |
| 12:50.05 | Haris | SL seems good |
| 12:50.25 | Haris | A means, the rest were outgoing calls ? |
| 12:51.34 | MacroMan | Possibly. Do you think you've made 104 outgoing calls? |
| 12:52.15 | MacroMan | Haris, Here you go: https://www.voip-info.org/wiki/view/asterisk+cli+command+show+queue |
| 12:52.26 | Haris | hmm .. I'm not seated close to the guys making calls |
| 12:52.55 | MacroMan | The C is cancelled. ie, abandoned calls |
| 12:53.22 | Haris | *ouch* cancelled |
| 12:53.53 | Haris | answered or cancelled ? |
| 12:54.06 | MacroMan | See the link |
| 12:54.27 | Haris | https://www.voip-info.org/wiki/view/asterisk+cli+command+queue+show |
| 12:54.28 | MacroMan | Actually this is if for the right command: https://www.voip-info.org/wiki/view/asterisk+cli+command+queue+show |
| 12:54.47 | MacroMan | Yes. It explains what they mean |
| 12:54.59 | Haris | in the table, 1 is against calls answered |
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| 12:58.18 | MacroMan | You are right. C is answered and A is unanswered. Can't think what they would stand for. |
| 12:58.35 | MacroMan | Ah, connected and abandonded? |
| 12:58.41 | Haris | hmm |
| 12:59.05 | Haris | what does the holdtime mean |
| 12:59.48 | MacroMan | I suppose the average time in the queue before being answered? |
| 13:01.57 | sotoz | I'm trying to use the rfc2833 dtmf_mode with pjsip but I get the "ERROR[17236]: config_options.c:738 aco_process_var: Error parsing dtmf_mode=rfc2833 at line 46" message. If I switch to rfc4733 it works but I'm not sure if the operators that I'm using support rfc4733. They say they only support rfc2833 and I'm not sure if those two are fully compatible. I know that 4733 obsoleted 2833 but what that means in terms of compatibility? |
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| 13:35.46 | jkroon | is there some smart way to try and track where in asterisk CPU usage is being spent? I'm sitting with a system that today suddenly shot up to around 350% (4-core system) CPU utilization. |
| 13:35.57 | jkroon | I can't find anything significant that changed ... |
| 13:37.00 | MacroMan | jkroon, Have you checked 'top' to make sure it's astersik using the cycles? |
| 13:37.02 | tomcruise | Having this strange issue with 14.3.0 and ConfBridge. When recording a participants name (x=0, open writing: /var/spool/asterisk/confbridge/confbridge-name-8888-1489065962.94 format: sln, 0x7f67c0011058), the file does get saved (as .sln). Playback to other participants however is just silence. I noticed the playback tries to open the same filename but then with extension .slin instead of .sln. Is |
| 13:37.04 | jkroon | the CPU spend is marginally higher user than system (55%, 40%) |
| 13:37.08 | tomcruise | this a known issue or config setting maybe? |
| 13:37.32 | jkroon | MacroMan, yes, asterisk: 333.2 <-- straight from top ... but it varies, goes up to 380% ... |
| 13:38.27 | MacroMan | Has your number of calls gone up? |
| 13:39.04 | MacroMan | Asterisk doesn't scale well and once you've hit the ceiling of calls, CPU usage can go sky high. |
| 13:39.31 | jkroon | MacroMan, no. |
| 13:39.42 | seanbright | does scale well? |
| 13:39.45 | seanbright | err |
| 13:39.47 | seanbright | doesn't* |
| 13:39.50 | jkroon | in fact, it's seemingly lower today than it was yesterday. |
| 13:40.00 | jkroon | MacroMan, 50 concurrent calls isn't a challenge. |
| 13:40.14 | jkroon | we've done benchmarks at a few hundred before. |
| 13:40.22 | jkroon | (not call setups, just switching rtp packets) |
| 13:40.31 | jkroon | and we were not seeing this kind of CPU usage. |
| 13:40.34 | MacroMan | I agree. I'm not really sure what else to check. I'll leave this to see if someone else can hel |
| 13:40.38 | MacroMan | help* |
| 13:42.23 | jkroon | if this is that stupid astdb again i'm going to lose it. |
| 13:43.08 | jkroon | effects are different ... but maybe. |
| 13:43.26 | Samot | What else does the machine do for the users? |
| 13:43.35 | *** join/#asterisk sotoz (~sotoz@095-097-255-066.static.chello.nl) |
| 13:43.40 | jkroon | Samot, dedicated for asterisk. |
| 13:43.56 | Samot | That doesn't tell me anything. |
| 13:44.05 | Samot | What else does the machine do for the users? |
| 13:44.06 | jkroon | then please clarify your question |
| 13:44.17 | jkroon | there is nothing on that machine other than asterisk. |
| 13:44.17 | Samot | Are you recording calls? Running queues? Conf bridges? |
| 13:44.20 | jkroon | oh yes. |
| 13:44.23 | jkroon | recordings. |
| 13:44.33 | Samot | What other things is the machine doing to that could raise the resources? |
| 13:44.39 | jkroon | one queue (which is used for reception which doesn't receive a lot of calls) |
| 13:44.42 | jkroon | no conf bridges |
| 13:44.55 | Samot | Those are important things to know when you're asking "Why is my CPU/RAM in use so much?" |
| 13:45.20 | jkroon | of those call recordings would be the most suspect. |
| 13:45.29 | Samot | Of course. |
| 13:45.46 | jkroon | uno momento, those I can hand off to ramfs (already have code in place for that). |
| 13:46.12 | WIMPy | Directories getting too big? Fragmentation reachig critical levels? What FS are you using? |
| 13:46.18 | jkroon | ext4 |
| 13:46.32 | jkroon | 400GB available on a 2.5TB filesystem. |
| 13:46.47 | Samot | What's this server running for CPU? |
| 13:47.09 | WIMPy | I have certainly had ext4 hit the brakes quite hard a few times. |
| 13:47.10 | jkroon | Intel(R) Core(TM) i3-3220 CPU @ 3.30GHz |
| 13:47.29 | jkroon | ok, let's eliminate filesystem first by having asterisk interact with ramdisk. |
| 13:47.46 | jkroon | i've got 2GB RAM available for that ... |
| 13:48.17 | Samot | So youre using a single dual core? |
| 13:48.48 | Samot | That's handling X amount of users, about 50+ sim calls, call recording and who knows what else... |
| 13:48.49 | jkroon | processor : 3 <-- /proc/cpuinfo claims 4 cores. |
| 13:49.04 | Samot | Dual core 4 threads. |
| 13:49.16 | jkroon | Samot, no config changes since yesterday where it was using <100% CPU now suddenly today it's 350%? |
| 13:49.53 | Samot | So you went from 25% usage to over 75% usage on the CPU over night? |
| 13:50.12 | jkroon | jip ... exactly why i'm so surprized. |
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| 13:50.38 | jkroon | something must have changed but since i have /etc/ in git with an auto-commit on everything every night I can promise you it didn't. |
| 13:50.42 | jkroon | same version of asterisk since Jan. |
| 13:50.48 | jkroon | even rebooted the machine. |
| 13:51.43 | WIMPy | Have you tried copying files around yourself? Do you get a decent speed on that? |
| 13:51.55 | jkroon | WIMPy, will check that now. |
| 13:52.08 | Samot | Show the output of the top command |
| 13:52.48 | jkroon | 3581 asterisk 20 0 2404808 93944 12104 S 320.3 2.4 214:13.30 asterisk |
| 13:54.10 | Samot | No, all the output. |
| 13:54.17 | Samot | So we can see what is being used on the system. |
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| 13:55.09 | WIMPy | IF we can see. |
| 13:56.44 | jkroon | http://pastebin.com/aKUuafpr |
| 13:56.46 | Samot | We'll see more than we do now |
| 13:58.11 | Samot | This is showing 50% CPU usage. |
| 13:58.53 | jkroon | the %Cpu line shows as percentage of overall available. |
| 13:59.08 | jkroon | the individual lines shows as percentage a single "thread" |
| 13:59.15 | Samot | Read that wrong, 50% of users processes. |
| 13:59.25 | Samot | 22.% system processes... |
| 13:59.31 | jkroon | that system worries me. |
| 14:00.59 | Samot | Why is the load so high? |
| 14:01.27 | jkroon | if we can figure that one out we've probably found the problem. |
| 14:01.45 | Samot | How many users are on it? |
| 14:02.02 | WIMPy | High load always smells of I/O. |
| 14:02.18 | jkroon | 302 sip peers [Monitored: 104 online, 198 offline Unmonitored: 0 online, 0 offline] |
| 14:02.26 | jkroon | going to astdb as well... |
| 14:02.28 | jkroon | just maybe. |
| 14:02.50 | Samot | AstDB is SQLlite3 |
| 14:02.53 | WIMPy | Have you tried the FS? |
| 14:03.15 | jkroon | Samot, using fsync() every now and again. |
| 14:03.48 | jkroon | WIMPy, FS? |
| 14:04.10 | WIMPy | Fiesystem |
| 14:04.58 | jkroon | dd gives me around 15MB/s outputting to the filesystem. |
| 14:05.11 | Samot | Wait.. |
| 14:05.14 | jkroon | iotop is reporting <500KB total write. |
| 14:05.18 | Samot | Single Dual Core CPU |
| 14:05.20 | WIMPy | That looks pretty little. |
| 14:05.22 | Samot | And how much RAM? |
| 14:05.26 | jkroon | 4GB total. |
| 14:05.35 | Samot | Upgrade your resources. |
| 14:05.50 | jkroon | free is reporting 2.3GB avialable (600MB completely free) |
| 14:05.55 | jkroon | so RAM is adequate. |
| 14:06.01 | Samot | OK. |
| 14:06.03 | jkroon | it's not trying to swap either. |
| 14:06.10 | Samot | SWAP is good. |
| 14:06.14 | jkroon | 2GB buff/cache. |
| 14:06.23 | jkroon | 2GB swap available, not using any of it. |
| 14:08.33 | *** join/#asterisk Y04NN (~y04nn@nayon.fr) |
| 14:08.34 | WIMPy | is also still in search of a performant and reliable FS, BTW. |
| 14:09.25 | *** join/#asterisk Dovid (~dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 14:09.40 | jkroon | WIMPy, you might find this interesting: http://jkroon.blogs.uls.co.za/it/voip/asterisk-massively-speeding-up-those-register-requests |
| 14:09.54 | Samot | I just wouldn't put 300+ users doing call recordings and can get up to 100+ sim calls on a server with those resources. Personally. |
| 14:10.00 | jkroon | it basically describes how I got stick with REGISTER processing in chan_sip and where the issue was exactly this disk IO thing. |
| 14:10.18 | grodriguezdata4 | Samot:Hey, how are you i hope you, or the comunnity can help me. I have the same problem i can´t use my asterisk server with my current internet provider, i have been reading at i seems like if if make multiple SIp petitions They block something on the internet. i have 5 SIP trunks with different SIP trunking providers i noticed that if i register |
| 14:10.18 | grodriguezdata4 | <PROTECTED> |
| 14:10.29 | WIMPy | jkroon: Yes, I was there when you had that topic. |
| 14:10.38 | jkroon | Samot, asterisk is using around 100MB of RAM on there. Majority of RAM is for MySQL. |
| 14:11.10 | jkroon | oh yes WIMPy - recall we discussed that a while back. |
| 14:11.44 | jkroon | grodriguezdata4, what are you using for NAT? |
| 14:11.49 | jkroon | if there is NAT. |
| 14:12.24 | WIMPy | grodriguezdata4: Smells like a router having run out of RAM. |
| 14:12.30 | Samot | jkroon: It's an ISP issue. |
| 14:13.13 | Samot | We can offer all the advice in the world but it won't matter unless we can figure out a way to "hack" his ISPs restrictions. |
| 14:13.43 | Samot | Which, in some parts of the world is to not use standard SIP ports. |
| 14:13.59 | Samot | Like 5060, 5061, etc etc.. |
| 14:14.14 | Samot | They generally end use using other ports that are allowed, like 80 or something. |
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| 14:15.17 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 14:17.32 | [TK]D-Fender | "i have 5 SIP trunks with different SIP trunking providers i noticed that if i register more than three trunks i lost the registry and i can not reach it again the registry until i reboot my server." <- this shouldn't be a limit |
| 14:17.50 | [TK]D-Fender | Show us actual registration attempts |
| 14:17.52 | [TK]D-Fender | ~pb |
| 14:17.52 | infobot | it has been said that 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://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 14:17.53 | [TK]D-Fender | ^^^ |
| 14:19.30 | grodriguezdata4 | jkroon: I do not use nothing for NAT, i just setting up the sip trunks, dialplan rules, dialplan prefixes and just start calling, i had been working like three yearas ago with my old internet provider and i worked ok, unfortunately i had to change my internet provider |
| 14:19.36 | jkroon | @ WIMPy + Samot - no single asterisk thread is using more than ~12% of a core. |
| 14:20.34 | jkroon | or a thread then ... |
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| 14:30.25 | ddickenson | Can anyone think of a reason that a system when dialing between two sip endpoints would take an additional 5-6 seconds to show up anything on the CLI (having already taken into account the digitmap timeout by pressing "dial") then another 5-6 seconds before it begins to ring the other phone? It only does this for proper matching extensions, if it's |
| 14:30.25 | ddickenson | something not defined in the dialplan you get reorder immediately. Have tried two flavors of Asterisk 13 Cert. and the issue has happened with both D45s and D62s as the phone sets. Also issue happens when using both pjsip as well as chan_sip. Running out of ideas to try... |
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| 14:31.36 | ddickenson | No level of verbosity or debug seems to give any idea of what's happening during that timeout between when dial is pressed and the call shows up on the CLI |
| 14:31.39 | WIMPy | ddickenson: Listening on AMI is great to find out where exactely the time is spent. |
| 14:31.49 | ddickenson | ahh |
| 14:32.01 | ddickenson | Thx, will research |
| 14:32.27 | WIMPy | Debug should help, but that usually produces so much other output that it's impossible to parse. |
| 14:32.45 | jkroon | ddickenson, ICE i've seen to that, as well as plain old STUN to a non-responsive server. |
| 14:32.47 | ddickenson | luckily this is not a production system yet so nothing going on except me trying to call these two phoens |
| 14:33.05 | WIMPy | I had a similar problem and traced it down to using DUNDi. |
| 14:33.12 | [TK]D-Fender | ddickenson, Do you see the delay before dialplan start executing? |
| 14:33.32 | ddickenson | [TK]D-Fender, yes so that made me think it's in the phone itself |
| 14:33.36 | [TK]D-Fender | It is |
| 14:33.42 | ddickenson | and same delay before it hits the receiving phone |
| 14:34.42 | [TK]D-Fender | * doesn't delay call processing once the call hits. If no dialplan is executing in those seconds it's because the call hasn't arrived yet |
| 14:34.50 | [TK]D-Fender | And that's the phone's issue |
| 14:35.02 | ddickenson | Is the digit map the only timeout on the digium phones? Because I'd changed it during testing to match precise numbers and such with no "T" at the end and still no joy. Tough in all other installs pressing the DIAL soft key has negated that timeout |
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| 14:35.24 | WIMPy | Oh yes, there can be massive delays in Asterisk. |
| 14:35.44 | ddickenson | So why does it immediately reject if I dial an undefined number and only delay if I dial one that is defined? |
| 14:35.48 | WIMPy | Well, I haven't seen that much yet, but 1-2 seconnds for no aparent reason does happen. |
| 14:35.50 | ddickenson | if it hasn't hit asterisk yet |
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| 14:36.23 | ddickenson | This is a first for me as well... I've seen the short delays you mention a bunch, not real worried about those though |
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| 14:37.05 | [TK]D-Fender | ddickenson> So why does it immediately reject if I dial an undefined number and only delay if I dial one that is defined? <- because you didnt' actually define it in a way that it accepts immediately |
| 14:37.27 | WIMPy | I used to be annoyed by the fact that I already had the caller name pop up on my desktop and then had to wait 2 seconds for the phone to ring. |
| 14:37.33 | [TK]D-Fender | Look at SIP debug. No packets during that delay = phone sent nothing |
| 14:37.54 | [TK]D-Fender | This does NOT require guessing |
| 14:39.39 | ddickenson | Will look, what I'm saying though, is that if the phone is not sending anything, and it must at least be sending some initial packets, then how would the phone know that if I dial say 1001 that it IS in the dialplan but if I dial 1002 it is not and immediately reject. Has to be talking to asterisk at some point during the delay. |
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| 14:50.50 | ddickenson | https://hastebin.com/uvowubutad.xml |
| 14:51.12 | ddickenson | It sends quite a bit right when the call is initiated |
| 14:51.17 | ddickenson | then waits... |
| 14:53.42 | ddickenson | I looked down at the time stamps and between line 255 and 256 is a 5 second delay but not obvious (to me) why |
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| 14:59.27 | sriharsha | grodriguezdata4, you can verify if your ISP is blocking or not by using a VPN to a server on the Internet ; but there you may have to deal with NAT |
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| 15:17.50 | tomcruise | Having this strange issue with 14.3.0 and ConfBridge. When recording a participants name (x=0, open writing: /var/spool/asterisk/confbridge/confbridge-name-8888-1489065962.94 format: sln, 0x7f67c0011058), the file does get saved (as .sln). Playback to other participants however is just silence. I noticed the playback tries to open the same filename but then with extension .slin instead of .sln. Is |
| 15:17.56 | tomcruise | this a known issue or config setting maybe? |
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| 15:20.13 | sotoz | Hi |
| 15:20.22 | sotoz | I'm trying to use the rfc2833 dtmf_mode with pjsip but I get the "ERROR[17236]: config_options.c:738 aco_process_var: Error parsing dtmf_mode=rfc2833 at line 46" message. If I switch to rfc4733 it works but I'm not sure if the operators that I'm using support rfc4733. They say they only support rfc2833 and I'm not sure if those two are fully compatible. I know that 4733 obsoleted 2833 but what that means in terms of compatibility? |
| 15:22.09 | roswell | sotoz, is it mandatory to use 2833? |
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| 15:37.59 | lvlinux | sotoz: pjsip evidently doesn't support pure 2833, so just set it on 4733 and it will probably work. Your operators may even be using 4733 and just still referring to it as 2833. 4733 has been out for a LONG time. |
| 15:39.19 | lvlinux | They are mostly compatible. |
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| 15:52.40 | grodriguezdata4 | Samot: i've already call to my ISP provider so they changed my cable modem, but the trouble its the same, they said that the make a pass through to my mac but guess what it doesn not work... I tried with an IAX2 trunk and it works but my SIp providers said that their servers will increase the bandwith consumption, i really apreciate your help and a |
| 15:52.40 | grodriguezdata4 | dvices but im going to be crazy, thanks in advance |
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| 16:24.19 | grodriguezdata4 | dear: [TK]D-Fender, jkroon,, Samot: this is my server log: https://thepasteb.in/p/0ghJV4j5BJls5 |
| 16:25.22 | grodriguezdata4 | sriharsha, community: whats is the best and easy way to handle with NAT? |
| 16:27.00 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
| 16:30.22 | [TK]D-Fender | Properly |
| 16:30.32 | [TK]D-Fender | Retransmitting #2 (NAT) to 174.142.94.143:5060: |
| 16:30.32 | [TK]D-Fender | REGISTER sip:multiphone.com.co SIP/2.0 |
| 16:30.36 | [TK]D-Fender | Contact: <sip:s@192.168.20.200:5060> |
| 16:30.47 | [TK]D-Fender | You are sending your PRIVATE IP as the return contact |
| 16:31.07 | [TK]D-Fender | You have not done the job of setting your system up to work behind NAT properly |
| 16:33.38 | grodriguezdata4 | [TK]D-Fender: i'm so sorry, im not too expert , can you point me on the right direction? |
| 16:34.57 | [TK]D-Fender | externaddr, nat,directmedia,localnet all need to be configured properl |
| 16:35.02 | [TK]D-Fender | read the sample config |
| 16:40.40 | jkroon | is there a way to request the module loader to recheck modules.conf, and to unload modules that are no longer listed with load => and load ones that got added? (we're using autoload => no) |
| 16:42.28 | [TK]D-Fender | no |
| 16:42.34 | jkroon | thanks |
| 16:42.39 | ddickenson | [TK]D-Fender... was a DNS problem. Apparently there was an issue with the primary DNS server and 5 seconds is the default timeout. Swapped the primary and secondary and works fine now |
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| 16:47.19 | overyander | can multiple servers share the same cdr database table when using cdr odbc? |
| 16:49.17 | jkroon | yes |
| 16:49.27 | overyander | thanks |
| 16:52.09 | rmudgett | jkroon: Try "core restart now" |
| 16:52.51 | jkroon | rmudgett, i was hoping there is a less intrusive way :) |
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| 18:19.58 | grodriguezdata4 | i have freePBX in order to mange my Asterisk server, so i followed this guide: http://wiki.freepbx.org/display/FPG/NAT+Configuration+FreePBX+12, am i missing something? |
| 18:21.08 | [TK]D-Fender | You clearly didn't do your subnet setup properly or the part about specifying your WAN IP |
| 18:21.08 | WIMPy | #freepbx |
| 18:21.18 | [TK]D-Fender | because it was advertising the PUBLIC IP, and not your WAN IP |
| 18:21.25 | [TK]D-Fender | <PROTECTED> |
| 18:21.27 | [TK]D-Fender | ~freepbx |
| 18:21.28 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
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| 18:27.42 | grodriguezdata4 | [TK]D-Fender: yeah i got it, but the thing is that i just follow this guide in order to setting my system up to work behind NAT properly, despite the fact that the same server works ok with another internet provider without this issue |
| 18:28.06 | [TK]D-Fender | Your settings are wrong if you are giving the wrong address |
| 18:28.18 | WIMPy | Most are clever enough to ignore what you send to them. |
| 18:28.19 | [TK]D-Fender | Saying you followed the guide doesn't eman you actually did. |
| 18:28.28 | [TK]D-Fender | We'd have to actually see what you did |
| 18:34.33 | grodriguezdata4 | dear: [TK]D-Fender, WIMPy, please excuseme i dont speak english so sometimes i could write as i want. its my english |
| 18:39.53 | dan_j | grodriguezdata4: Go to the #freepbx channel and ask for help there if you are using freepbx. |
| 18:40.33 | kraylo | oh, snap!! :) |
| 18:41.35 | [TK]D-Fender | snaps |
| 18:41.52 | [TK]D-Fender | starts planning where to bury the latest batch of bodies... |
| 18:43.03 | dan_j | Sorry, was trying to use simple english to see if that would help move things along. |
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| 19:51.13 | igcewieling | [TK]D-Fender: in the Fields of Disillusionment |
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| 20:28.27 | grodriguezdata4 | [TK]D-Fender: im here again: i tried to compile all my trouble just there can you give me another try? https://thepasteb.in/p/58hgERqEAJ6Tv |
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| 20:40.01 | BLAH_BLAH_BLAH | I was wondering if someone could help me go in the right direction. I am currently dialing an extension which in the dialplan dial's a phone number. I would like to know the easiest way to put the caller have moh while the dialed phone number is going through it's own dial plan and if all variables line up correctly, the 2 parties can communicate. Thanks in advance |
| 20:41.56 | [TK]D-Fender | "core show application dial" <--- |
| 20:50.39 | BLAH_BLAH_BLAH | G options looks to be it |
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| 21:35.14 | igcewieling | BLAH_BLAH_BLAH: are the PSTN interfaces analog? |
| 21:36.02 | BLAH_BLAH_BLAH | all sip |
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| 21:41.21 | igcewieling | good. |
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