| 00:09.56 | *** join/#asterisk wonderworld (~ww@ip-62-143-156-254.hsi01.unitymediagroup.de) |
| 00:13.47 | Elio19 | ChannelZ, are you that Bob who left his job and wrote the asterisk primer? |
| 00:14.01 | ChannelZ | Indeed I am |
| 00:15.37 | Elio19 | Nice to meet you |
| 00:16.56 | ChannelZ | You too |
| 00:18.42 | Elio19 | Are you listening to music? |
| 00:19.19 | ChannelZ | Sort of, I'm watching a Twitch stream and there's music on that.. |
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| 00:53.48 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 01:25.57 | Elio19 | I was following your Primer ChannelZ ~more or less~ because i used packages from the repo rather than compile it and picked a soft-phone from the repo. They didnt seemed to start well; soft-phone wont connect to my local SIP endpoint. It did not work immediately. So, I poked around in the Asterisk command line and noticed that it had not load two (of 6) config files: rtp.conf and sip.conf. |
| 01:26.54 | Elio19 | ...assuming that my config files have some different settings from your example. |
| 01:34.21 | ChannelZ | What version of asterisk? |
| 01:34.43 | Elio19 | "Connected to Asterisk 11.14.1 currently running on localhost" |
| 01:35.03 | ChannelZ | what happens if you "module load chan_sip" |
| 01:36.08 | Elio19 | "Loaded chan_sip.so => (Session Initiation Protocol (SIP))" |
| 01:36.56 | ChannelZ | So now try your softphone and see what you get |
| 01:37.36 | Elio19 | Well, i checked port 5060 and its not open |
| 01:38.07 | ChannelZ | define 'not open' - do you really mean nothing is listening on 5060 |
| 01:38.22 | ChannelZ | (IE netstat -alpn |grep 5060) |
| 01:38.43 | Elio19 | bash: IE: command not found |
| 01:38.49 | Elio19 | ^jk jk |
| 01:38.55 | ChannelZ | YOUR SYSTEM IS SCREW |
| 01:39.27 | Elio19 | netstat and netcat come back blank |
| 01:39.41 | ChannelZ | hmm |
| 01:40.15 | ChannelZ | check your sip.conf doesn't have a syntax error or some weird chars or something. pastebin if you like |
| 01:40.28 | Elio19 | I'm replacing the distro-supplied config files with .samples from asterisk.org |
| 01:40.38 | ChannelZ | That's not a good idea |
| 01:40.53 | Elio19 | because there was another file, rtp.conf that did not load, i was not thinking "syntax" |
| 01:41.15 | ChannelZ | The sample configs aren't particularly usable/useful (except for a couple like asterisk.conf and rtp.conf) |
| 01:41.34 | ChannelZ | They are more documentation than workable |
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| 01:50.12 | ChannelZ | wanders off for dinner |
| 01:50.49 | Elio19 | I swapped the distro-supplied config files for ones bundled with source. Now it only loads one config file, but it does listen on 5060 according to netstat |
| 01:51.00 | Elio19 | OK, thanks for help! |
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| 02:51.33 | Penguin | elio19: Any files that come with asterisk are sample files, not intended to be used as they are. |
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| 04:22.29 | Katty | Whrrrrrrrr |
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| 04:23.19 | WIMPy | pets Katty |
| 04:23.39 | c|oneman | does anyone know how to prepend digits on a Polycom diaplan? My SIP provider doesn't put calls through unless they start with a country code |
| 04:24.01 | c|oneman | I want the 1 to be inserted before the call is sent to the provider |
| 04:24.48 | ChannelZ | is this not an Asterisk question? does your polycom talk directly to your ITSP? |
| 04:25.28 | WIMPy | Yes. It's easy with Asterisk. No idea about phones dialplans. |
| 04:27.08 | c|oneman | yeah the polycom talks directly to the provider |
| 04:27.27 | c|oneman | I guess the provider is more accustomed to talking with a pbx |
| 04:28.00 | c|oneman | got it. |
| 04:28.49 | c|oneman | RR1R[2-9]xxxxxxxxx |
| 04:28.51 | ChannelZ | I don't know Polycom's dialplan syntax specifically |
| 04:29.28 | c|oneman | http://support.polycom.com/global/documents/support/technical/products/voice/Digit_Map_Changes_TB11572.pdf . This was useful, I'd read it before but I must have missed something |
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| 06:43.05 | Psil0Cybin | hey guys i have a problem i am trying to get asterisks to connect to the internet i got it working perfect locally but when my friend tries to connect he gets a SIP 408 - Timeout issue |
| 06:43.13 | Psil0Cybin | and i can see on the asterisks CLI he is timing out |
| 06:43.39 | Psil0Cybin | i have been googling for a while, I port forwarded almsot every port mentioned online |
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| 15:18.08 | *** join/#asterisk infobot (ibot@rikers.org) |
| 15:18.08 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.2.0 (2015/02/06), 11.16.0 (2015/02/06), 1.8.32.2 (2015/01/28); Standard: 12.8.1 (2015/01/28); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 15:18.20 | tuks92 | mjordan |
| 15:18.24 | tuks92 | can you help me |
| 15:18.36 | tuks92 | are you there? |
| 15:19.33 | tuks92 | i need help if someone is willing to spare some time to help me, in return I will pay you! This is nothing hard for people who know what they are doing, here is what I need: http://pastebin.com/ijVfa6Cw |
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| 15:21.21 | [TK]D-Fender | you say "But we still donât get anything." abut you haven't actually shown the call.... |
| 15:21.46 | mjordan | tuks92: I don't do contract work. There are others who may however. If you need contract work, you should probably ask on the asterisk-biz mailing list. |
| 15:23.08 | *** join/#asterisk infobot (ibot@rikers.org) |
| 15:23.08 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.2.0 (2015/02/06), 11.16.0 (2015/02/06), 1.8.32.2 (2015/01/28); Standard: 12.8.1 (2015/01/28); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 15:29.33 | mjordan | the asterisk-biz list is generally the appropriate place |
| 15:29.44 | tuks92 | ok thank you very much |
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| 15:33.46 | mutilator | sooooo, any SNOM peeps around yet? https://forum.snom.com/index.php?showtopic=15009 i'm having an issue someone may have an idea how to work around |
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| 16:01.45 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.2.0 (2015/02/06), 11.16.0 (2015/02/06), 1.8.32.2 (2015/01/28); Standard: 12.8.1 (2015/01/28); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 16:08.23 | Katty | is anyone using veeam? |
| 16:10.36 | vomit | Hello! We are experiencing outgoing calls to be cut of if not picked up and Providers support are unable to point the problem. Looks like some timeout is being reached and the call disconnects at 30 sec mark - asterisk logs show a regular hangup, the caller gets fast busy. One provider states issue on the other end. Another provider responded stating we send two requests to attempt to mask an IP (?) and refers to B2BUA (Details here: http://pastebin.co |
| 16:13.02 | *** join/#asterisk infobot (ibot@rikers.org) |
| 16:13.02 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.2.0 (2015/02/06), 11.16.0 (2015/02/06), 1.8.32.2 (2015/01/28); Standard: 12.8.1 (2015/01/28); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 16:24.30 | [TK]D-Fender | Katty: My head office does. I'm looking to do with ProxMox here.... |
| 16:24.36 | [TK]D-Fender | go* |
| 16:24.45 | Katty | [TK]D-Fender: have you personally worked with it? |
| 16:24.52 | [TK]D-Fender | Katty: Nope. |
| 16:25.15 | Katty | [TK]D-Fender: do you know if they're dump snapshots onto local, san, or nas? |
| 16:25.28 | [TK]D-Fender | Katty: QNAP NAS |
| 16:26.07 | *** join/#asterisk nizda (~nizda@50.242.169.133) |
| 16:26.22 | Katty | [TK]D-Fender: cloud qnap or regular qnap? |
| 16:26.42 | [TK]D-Fender | Katty: Regular TS420 rack or some such |
| 16:26.47 | Katty | ok |
| 16:27.01 | Katty | [TK]D-Fender: do you have any idea what the raid configuration is? |
| 16:27.24 | [TK]D-Fender | Katty: After which they use USB attached HD's for removable off-site storage |
| 16:27.34 | Katty | [TK]D-Fender: trying to get feel for what kind of I/O veeam generates |
| 16:27.59 | [TK]D-Fender | Either 5 or 6 I'm betting |
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| 16:45.57 | dimitry7 | what is a trunk? |
| 16:46.02 | dimitry7 | hi everybody! :) |
| 16:46.34 | [TK]D-Fender | ~trunk |
| 16:46.34 | infobot | methinks trunk is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. the nose of an elephant |
| 16:50.36 | *** part/#asterisk nizda (~nizda@50.242.169.133) |
| 16:53.29 | dimitry7 | [TK]D-Fender, okay good, now |
| 16:53.44 | dimitry7 | how do I see how many trunks and DID's do I have in a asterisk pbx server? |
| 16:53.53 | [TK]D-Fender | Look at your configs |
| 16:54.00 | dimitry7 | yeah I have them open |
| 16:54.02 | dimitry7 | where do I look? |
| 16:54.14 | [TK]D-Fender | Where you configured them |
| 16:54.24 | dimitry7 | I didn't configure them |
| 16:54.28 | dimitry7 | there where already configured |
| 16:54.31 | [TK]D-Fender | Which obviously depends on what kind of things you're loking for. |
| 16:54.36 | dimitry7 | which parameters do I look for? |
| 16:54.41 | dimitry7 | just 2 thinks |
| 16:54.43 | dimitry7 | thinGs |
| 16:54.45 | dimitry7 | trunks and did's |
| 16:54.46 | [TK]D-Fender | Depends what KIND <- |
| 16:54.55 | *** join/#asterisk Chotaire (chotaire@vegetarian.cannibal.club) |
| 16:54.55 | dimitry7 | trunks from service provider |
| 16:54.58 | [TK]D-Fender | dimitry7trunks and did's <- these are CONCEPTS |
| 16:55.02 | [TK]D-Fender | WHAT KIND OF TRUNK? |
| 16:55.12 | dimitry7 | from the service provider |
| 16:55.19 | [TK]D-Fender | that isn't a "kind |
| 16:55.20 | [TK]D-Fender | " |
| 16:56.22 | dimitry7 | <PROTECTED> |
| 16:56.23 | dimitry7 | that one |
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| 16:58.55 | [TK]D-Fender | ... |
| 16:58.56 | [TK]D-Fender | no. |
| 16:59.02 | [TK]D-Fender | what TECHNOLOGY? |
| 17:04.51 | *** join/#asterisk bayar (488f38d2@gateway/web/freenode/ip.72.143.56.210) |
| 17:05.24 | bayar | hi |
| 17:07.47 | dimitry7 | oh |
| 17:07.57 | dimitry7 | [TK]D-Fender, PRI |
| 17:08.15 | bayar | i am trying to connect 2 sipml5 instances using Asterisk 13.2.0. Voice calls are fine. Can not make video calls. asterisk has public ip address, sipml5 client are behind nat. |
| 17:08.22 | bayar | i saw https://issues.asterisk.org/jira/browse/ASTERISK-24735 |
| 17:08.24 | [TK]D-Fender | ~pri |
| 17:08.25 | infobot | well, pri is [~pri] Primary Rate Interface, often called T1 or E1 (European Standard). E1 offers 30 ISDN B-Channels a 64kBit/s + 1 D-Channel with 64kBit/s. The T1 has 23 B-Channels + 1 D-Channel. Cards to use with *: T100P, E100P, TE410P, R1T1,R2T1,R4T1, etc. |
| 17:08.26 | [TK]D-Fender | ^^^ |
| 17:08.58 | dimitry7 | yeah |
| 17:09.03 | [TK]D-Fender | dimitry7: As for how many DID's... there is no proof in the way a system is configured. You might have a CLUE based on the extensions you defined to process those calls. |
| 17:09.03 | bayar | Has anybody working setup with simpl5 endpoints and asterisk 13 ? |
| 17:09.49 | dimitry7 | [TK]D-Fender, aren't there commands to see that? |
| 17:09.56 | [TK]D-Fender | dimitry7: No. |
| 17:09.57 | dimitry7 | pri show channels? |
| 17:10.29 | [TK]D-Fender | dimitry7: The telco could send you 5 thousand different DID's. What your dialplan looks like to process them may not be indicative of how many they might send you |
| 17:10.29 | dimitry7 | or this: channel =>1-15,17-31 in chan_dadhi.conf ? |
| 17:10.52 | [TK]D-Fender | dimitry7: That will show you ow many simultaneous channels you can support, but that has nothing to do with how many DID's |
| 17:11.13 | [TK]D-Fender | dimitry7or this: channel =>1-15,17-31 in chan_dadhi.conf ? <- this suggests a 30-channel E1 PRI |
| 17:11.14 | dimitry7 | ok there is no way possible I can see how many Dids im getting |
| 17:11.22 | dimitry7 | its all up to my Service provider, ok |
| 17:11.24 | [TK]D-Fender | dimitry7: no PROOF |
| 17:11.41 | [TK]D-Fender | dimitry7: Your dialplan MIGHT account for all them in a way that it is evident... or it might not |
| 17:11.46 | dimitry7 | I mean proof |
| 17:11.55 | dimitry7 | yeah |
| 17:12.04 | dimitry7 | ok 30-channel E1 PRI |
| 17:12.22 | dimitry7 | those aren't trunks? |
| 17:12.47 | [TK]D-Fender | your PRI is the trunk |
| 17:13.05 | [TK]D-Fender | you have ONE PRI, with 30 usable channels defined |
| 17:13.18 | dimitry7 | its because the service provider said we had 30 trunks |
| 17:14.15 | [TK]D-Fender | You could view each channel on your PRI as a "trunk" if you wish |
| 17:14.29 | [TK]D-Fender | You're just mixing terms at that point |
| 17:14.30 | dimitry7 | [TK]D-Fender, oooh, that's where I got confused |
| 17:14.47 | dimitry7 | yeah ... I see :S |
| 17:15.13 | dimitry7 | and for example... with 30 TRUNKS (real ones)... we would be talking about 30 PRIs no? |
| 17:15.15 | [TK]D-Fender | A trunk is usually reserved to refer to the physical or encapsulating connection |
| 17:15.55 | dimitry7 | ok ok... is there a way to differentiate in the config or commands... a trunk (meaning physical or encapsulated connection) |
| 17:16.06 | dimitry7 | versus a channel of them..? |
| 17:16.14 | [TK]D-Fender | huh? |
| 17:16.26 | dimitry7 | Yeah for example I have 1 E1 PRI |
| 17:16.42 | dimitry7 | with 30 channels, but how about having 4 PRIs? |
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| 17:16.48 | dimitry7 | where do I see how many PRIs do I have? |
| 17:17.06 | [TK]D-Fender | system.conf |
| 17:17.14 | [TK]D-Fender | <PROTECTED> |
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| 17:18.06 | dimitry7 | I have this: http://pastebin.com/QVW7b2RH |
| 17:18.52 | bayar | anybody have experience with sipml5 + asterisk ? |
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| 17:19.29 | [TK]D-Fender | dimitry7: Only 1 defined there |
| 17:19.49 | [TK]D-Fender | span=1,1,0,ccs,hdb3 <--- one span definition = 1 PRI |
| 17:19.57 | dimitry7 | oooh... great :D |
| 17:20.07 | [TK]D-Fender | bchan=1-15,17-31 <- these channels on it defined as voice. |
| 17:20.07 | dimitry7 | now I got it all man! |
| 17:20.54 | dimitry7 | yeah 1 trunk (above) and that trunk supports 30 channels... with somultaneous calls right?? |
| 17:21.21 | dimitry7 | * Simultaneous |
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| 17:24.05 | [TK]D-Fender | yes |
| 17:25.11 | dimitry7 | [TK]D-Fender, excellent! thank you so much for all the explainations man! :-) |
| 17:26.31 | overyander | is there a way to set a sip registration within a realtime db or do you have to do it in the sip.conf file? |
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| 17:29.47 | bipolar | I'm trying to use the periodic beep option in MixMonitor that was added in v13, but something is wrong. I've created a pastebin with both my test extension configuration and the full output from the log with debug set to 5. The beep when it starts recording works, and I hear that. http://pastebin.com/aGjckK84 Does anyone have an idea what's going on? |
| 17:32.23 | [TK]D-Fender | [2015-02-27 12:25:18] ERROR[4065][C-0000000c] app_chanspy.c: Could Not Set Write Format. |
| 17:32.29 | [TK]D-Fender | [2015-02-27 12:25:18] WARNING[4065][C-0000000c] channel.c: Unable to find a codec translation path from (slin) to (slin) |
| 17:32.55 | [TK]D-Fender | Sounds like a whole lot of "didn't set your codecs right" |
| 17:33.15 | bipolar | I figured that had something to do with it, but I'm not sure how it could possibly not be able to 'translate' a codec when it's the same one. |
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| 17:37.00 | bipolar | I created a new pastebin that shows the translations available. http://pastebin.com/4bbmHgiy |
| 17:38.20 | bipolar | I wish I knew more about it, but all I can tell is that slin is there. twice actually. hmm. |
| 17:41.52 | bipolar | Ok. This build must be borked. I'll take it over to the people who compiled it. I thought it might be something in my configuration. |
| 17:43.56 | [TK]D-Fender | Doesn't matter if it's avaiable. |
| 17:44.07 | [TK]D-Fender | What matters is what you TOLD it to use |
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| 18:23.51 | Skywalker | guys i'm in trouble. i work at this sip carrier company. we connect to both, our cliens and suppliers via SIP Proxies. alright. (opensips + media proxy ag project). some of our customers use asterisk (as dialer) which works as passive rtp. how could i set it to work as active ? |
| 18:24.35 | Skywalker | i ask that because my suppler work as passive rtp too. so the calls are mute (neither customer or supplier begin the rtp flow) |
| 18:27.47 | pabelanger | what is passive rtp? |
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| 18:35.07 | babadoo | i try to use tls, but get "due to reason 'tlsv1 alert unknown ca', terminating" |
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| 18:40.13 | babadoo | how can i check if the ca file is in the correct format? |
| 18:46.58 | pabelanger | https://www.openssl.org/docs/apps/verify.html |
| 18:47.22 | babadoo | it tells me its OK |
| 18:47.40 | babadoo | but i get the error unknown ca ni asterisk |
| 18:47.45 | babadoo | *in |
| 18:49.59 | babadoo | its a free cert from comodo, maybe its not accepted by chrome? i use it in webrtp (sipml5) |
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| 19:00.24 | Katty | he's dead jim )= *sob* |
| 19:02.55 | Kobaz | man, 13.2 already |
| 19:03.01 | Kobaz | i'm all behind and everything |
| 19:03.10 | Kobaz | all our production systems are 1.8, heh |
| 19:03.16 | Kobaz | 1.8.12 with lots of patches |
| 19:08.28 | pabelanger | babadoo, best you could do is try following: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial |
| 19:08.32 | pabelanger | and see if you get the same results |
| 19:08.44 | pabelanger | Kobaz, 1.8.7.1 here with backports |
| 19:09.09 | pabelanger | bonus thing, none of instances are public facing |
| 19:09.17 | Kobaz | yay |
| 19:09.36 | Kobaz | public facing *'s are interesting to get right |
| 19:09.50 | Kobaz | i had to write my own security firewall/rate limiter |
| 19:09.56 | Kobaz | since fail2ban was failing miserably |
| 19:11.02 | overyander | i'm having an issue recieving calls over a sip trunk between a cisco call manager express and my asterisk box. from asterisk i can dial extensions connected to the cme fine. when a phone on the cme tries to call something on the asterisk server, i see a sip registration request on asterisk from the remote extension. any suggestions? |
| 19:11.24 | pabelanger | Kobaz, Ya, if I need public facing stuff, I dropped a proxy in front and bridge stuff into asterisk |
| 19:12.06 | Kobaz | what proxy are you using |
| 19:13.29 | pabelanger | kamailio |
| 19:13.36 | pabelanger | but, might give OpenSIPs a go |
| 19:13.43 | pabelanger | mostly for the event system they have |
| 19:13.51 | pabelanger | but, I think kamailio has an event system now |
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| 19:38.01 | overyander | i'm having an issue recieving calls over a sip trunk between a cisco call manager express and my asterisk box. from asterisk i can dial extensions connected to the cme fine. when a phone on the cme tries to call something on the asterisk server, i see a sip registration request on asterisk from the remote extension. any suggestions? |
| 19:38.43 | pabelanger | ~collectdebug |
| 19:38.44 | infobot | somebody said collectdebug was a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 19:38.47 | pabelanger | overyander, ^ |
| 19:38.56 | pabelanger | get a SIP debug log and see what is happening |
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| 19:41.37 | overyander | pabelanger, http://pastebin.com/cvyVJP03 |
| 19:42.40 | [TK]D-Fender | We don't see the actual attempt |
| 19:42.50 | [TK]D-Fender | you restricted it to a single peer and it's not flagged as coming from there |
| 19:43.05 | pabelanger | right, need more data |
| 19:43.46 | overyander | that's what i get when a call tries to come into asterisk from cme. i don't want to create sip peers for all of the extensions connected to cme. |
| 19:44.48 | [TK]D-Fender | No. |
| 19:44.55 | [TK]D-Fender | We are NOT seeing that attempt you're talking about |
| 19:45.13 | [TK]D-Fender | your attempt to restict what you are vieing in debug... has resticted us from actually SEEING this attempt |
| 19:45.56 | pabelanger | what is the IP address of your cisco device? |
| 19:47.05 | overyander | 172.16.100.254 |
| 19:47.25 | pabelanger | then we are not seeing the INVITE from it |
| 19:47.31 | overyander | there's a lot of traffic on this asterisk box which is why i didn't turn on global sip debug |
| 19:47.44 | pabelanger | so, either the debug is outside the time the INVITE was sent |
| 19:47.51 | pabelanger | or cisco is not actually sending it |
| 19:48.07 | overyander | would a pcap help? |
| 19:48.21 | pabelanger | redo your debug |
| 19:48.26 | pabelanger | look for an INVITE |
| 19:48.30 | pabelanger | if you don't see it |
| 19:48.35 | pabelanger | then you are not getting call |
| 19:53.02 | overyander | here's a more detailed debug |
| 19:53.04 | overyander | http://pastebin.com/qNRteTdD |
| 19:53.47 | [TK]D-Fender | [Feb 27 13:50:24] No matching peer for '1159' from '172.16.100.254:65381' |
| 19:53.50 | [TK]D-Fender | ^^ |
| 19:54.12 | overyander | <PROTECTED> |
| 19:54.44 | [TK]D-Fender | Is that just the IP of a single phone and others will have different IP as well? |
| 19:55.08 | overyander | 172.16.100.254 is the IP of the Cisco CME, the phones have different IP's |
| 19:55.22 | [TK]D-Fender | well the call is SOURCED as the CME |
| 19:55.32 | [TK]D-Fender | So make a match for it |
| 19:56.18 | overyander | 1159 is the extension connected to CME trying to make the call. |
| 19:56.44 | [TK]D-Fender | Don't care about 1159 |
| 19:56.50 | [TK]D-Fender | make a peer to match the CME IP |
| 19:56.57 | overyander | i did |
| 19:57.02 | [TK]D-Fender | it doesn't amtch |
| 19:57.09 | *** join/#asterisk babadoo (~babadoo@planeshift/art/associate/kinea) |
| 19:57.13 | [TK]D-Fender | probably for challenging where it shouldn't <- |
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| 19:59.23 | overyander | the peer name in asterisk for the CME is CME-PBX. i'm not understanding why the calls from the CME are trying to match the CME extensions with peers on asterisk. 1159 is an extension on the CME. if 1159 calls asterisk it tries to auth 1159 on asterisk. if 1050 on the cme tries to call asterisk, it tries to auth 1050 with a peer on asterisk. |
| 20:02.46 | [TK]D-Fender | no, it doesn't |
| 20:03.01 | [TK]D-Fender | Forget those numbers NOW. |
| 20:03.46 | overyander | ok |
| 20:04.29 | [TK]D-Fender | Show us your peer for that IP. |
| 20:04.43 | [TK]D-Fender | * config... |
| 20:04.48 | [TK]D-Fender | not some status dump |
| 20:06.52 | overyander | it's in a db, do you want the raw data of the row or the data with the collumn names? |
| 20:09.42 | [TK]D-Fender | with names |
| 20:11.22 | overyander | [TK]D-Fender, http://pastebin.com/YjeKPEzS |
| 20:12.25 | [TK]D-Fender | very wrong for what you need |
| 20:12.30 | [TK]D-Fender | type = PEER |
| 20:12.35 | [TK]D-Fender | insecure=port,invite |
| 20:12.43 | [TK]D-Fender | set those equivalents |
| 20:14.54 | overyander | ok, changed and about to test |
| 20:18.06 | overyander | [TK]D-Fender, i think that solved the problem. not getting the same auth error |
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| 20:35.17 | SalmaHaya | Hi everyone - I have two Asterisk servers. One is in Data center and one is on site. I would like to create a failover to cell phone solution. DIDs will come to Server-A and then forward to Server-B (on site server). I would like to have Server-A failover to cell phones if Server-B is offline. What part of dialplan can help me with this? to detect the Server-B being down? |
| 20:38.02 | [TK]D-Fender | SalmaHaya: You'd dial the other server YOURSELF... and check the result |
| 20:38.13 | [TK]D-Fender | ? book |
| 20:38.22 | [TK]D-Fender | ~book |
| 20:38.22 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 20:38.24 | [TK]D-Fender | ^^^^^ |
| 20:39.17 | [TK]D-Fender | And answered in #freepbx |
| 20:43.00 | overyander | [TK]D-Fender, thanks for your help with that issue. just to clarify my understanding. peer for trunks, friend for actual phones? |
| 20:47.19 | [TK]D-Fender | overyander: friend = user + peer. |
| 20:47.23 | [TK]D-Fender | user = match NAME |
| 20:47.26 | [TK]D-Fender | peer = match IP |
| 20:47.34 | [TK]D-Fender | We don't WANT to amtch by name |
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| 20:48.33 | overyander | That makes sense. |
| 20:49.38 | overyander | I have my outside sip trunk provider configured as a friend. just like i had the cme settings. why doesn't it try auth like then when i get calls from them? |
| 20:55.34 | [TK]D-Fender | overyander: Depends how the SENDER sends you a call as well |
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| 23:07.50 | litn | hey guys, none of my channels are closing after hangup, my log shows a lot of " Rescheduling destruction for 10000 ms" |
| 23:07.59 | litn | resources on the server are fine- cpu/ hd space, memory |
| 23:08.04 | litn | no locks in mysql |
| 23:08.14 | litn | anything I can do here? |
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