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| 01:03.22 | chandoo_ | hi |
| 01:03.38 | chandoo_ | how to find out which ports i have to forward |
| 01:05.55 | newtonr | Depends on what protocols and features for those protocols are in use |
| 01:06.30 | newtonr | for unsecure SIP communication probably UDP 5060 and your RTP port range (default 10000-20000) |
| 01:06.41 | newtonr | insecure even |
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| 01:22.13 | vader- | Any of you guys run Asterisk in AWS? |
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| 03:01.13 | cmendes0101 | vader-: yes |
| 03:01.40 | vader- | how is it working for you? |
| 03:02.02 | vader- | I have a small business which has had their computer fail a few times and im looking for an alternative |
| 03:02.31 | vader- | they have 5 phones |
| 03:02.40 | vader- | so im trying to figure out which AWS instance would work |
| 03:03.05 | cmendes0101 | Pretty good. Haven't had an issue since we moved related to AWS infrastructure |
| 03:03.34 | vader- | how many lines do you support? |
| 03:05.16 | cmendes0101 | We have it autoscaling(not really HA) and hit a few hundred concurrent. It really matters on instance resources |
| 03:05.51 | vader- | ya im thinking i only need a small instance |
| 03:05.56 | cmendes0101 | 5 is pretty light. I wouldn't think latency or bandwidth would be an issue. |
| 03:06.23 | cmendes0101 | Try with tiny |
| 03:06.47 | cmendes0101 | If for some reason 5 concurrent is sucking up all the resources, change instance type |
| 03:07.43 | cmendes0101 | you won't be able to use dahdi or anything. Only difference you would hit is its behind NAT |
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| 03:13.44 | vader- | any other issues i would run into? |
| 03:15.22 | vader- | i would just be using a sip trunk |
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| 06:21.23 | seik0 | Continued investigation of asterisk performance degradation. Got rid of memory leak and problem is still here |
| 06:23.06 | seik0 | See suspicious hanged channels from incoming (external, from Telco) SIP, processing on voicemail, but so long that they are probably dead. Also, getting a lot of "Failed to write frame" in CLI |
| 06:23.41 | seik0 | failed to write frame, most possibly, is when trying to play sound on dead channels |
| 06:24.16 | seik0 | but why are they still in asterisk if they are dead - do not know |
| 06:25.34 | seik0 | Maybe some dialplan issue in hangup extension, but "core show channels" does not show any app(data) from hangup extensions (and for hanged channels only Voicemail ) |
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| 07:45.22 | seik0 | maybe some mutex issue, because cpu is free |
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| 08:39.53 | seik0 | can someone give advice on strace to check for possible mutex problem? |
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| 09:13.25 | seik0 | what is the best way to reject call emulating busy? |
| 09:13.34 | seik0 | is it always necessary to Answer() ? |
| 09:18.43 | markusl | Busy() |
| 09:18.52 | markusl | ? |
| 09:19.21 | markusl | don't answer, after Answer() you have to Hangup() i guess. |
| 09:19.43 | markusl | Busy after answer doesn't make sense to me |
| 09:21.07 | seik0 | most examples use Answer-Playtones-Busy/Congestion-Hangup |
| 09:21.14 | seik0 | and I cannot understand why |
| 09:22.05 | seik0 | logically use Busy or Congestion on unanswered channel to indicate Busy using signalling only |
| 09:22.14 | markusl | that's faking busy if you have analog phones |
| 09:23.37 | seik0 | ook |
| 09:23.45 | markusl | i think you can't signal busy to an analog line so you pick up and play busy tone |
| 09:24.00 | markusl | but with sip and isdn phones you can just busy() |
| 09:24.51 | markusl | imho this workaround should be applied by the ata (or asterisk w/ dahdi fxs) right where the analog phone is connected |
| 09:25.13 | seik0 | do not have analog lines, so not a problem |
| 09:25.14 | markusl | not generically for peers that call you over sip |
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| 09:53.56 | seik0 | I have hanged channels (shown by "core show channels") |
| 09:54.01 | seik0 | they hang for 4 hours already |
| 09:54.14 | seik0 | I want to make strace for theirs threads |
| 09:54.40 | seik0 | but can't find threadID of those threads |
| 09:54.46 | seik0 | I have channels names |
| 09:54.53 | seik0 | I can get threads list |
| 09:55.20 | seik0 | but can't link channels to threads |
| 09:55.23 | seik0 | any help? |
| 10:07.09 | seik0 | should "core show channels verbose" have been not so trunked it may have helped |
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| 11:04.41 | WIMPy | 'core show locks' |
| 11:04.53 | WIMPy | But you might have to recomplile to enable it. |
| 11:06.15 | WIMPy | is AFK again |
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| 11:11.05 | seik0 | WIMPy, yes, but now built without lock debug |
| 11:11.15 | seik0 | is it the only way? |
| 11:16.43 | seik0 | WIMPy, recompiling already |
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| 11:52.35 | seik0 | asterisk hangs when debug_threads enabled |
| 11:53.17 | seik0 | not a for a while |
| 11:53.21 | seik0 | though |
| 11:54.13 | seik0 | I saw such behavior when running asterisk under hyperv on 3.0 kernel |
| 11:54.31 | seik0 | connecting with gdb fixes problem |
| 11:54.42 | seik0 | it is a mutex problem |
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| 12:31.41 | seik0 | i'm tired =( |
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| 12:47.47 | densh | Hi, any hints on ] ERROR[50660]: netsock2.c:263 ast_sockaddr_resolve: getaddrinfo("SAME_PBX_IP:5060:60628", "(null)", ...): Name or service not known ? |
| 12:48.00 | densh | where to start digging? |
| 12:48.29 | mjordan | SAME_PBX_IP:5060:60628 is not a valid DNS name |
| 12:48.43 | mjordan | for one thing, I'm not sure why you've got two ports appended to whatever the hostname is |
| 12:49.02 | mjordan | beyond that, whether or not SAME_PBX_IP is a valid host/IP address is something you'd have to check |
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| 12:54.20 | densh | Thx, I see that two ports are glued together, that's weird! SAME_PBX_IP is valid IP address of the PBX server. |
| 12:54.41 | densh | Any ideas how to find out why these two port numbers are glued together? |
| 12:54.56 | mjordan | no without knowing where it came from |
| 12:55.12 | mjordan | if you've specified SAME_PBX_IP:5060 some place, well, that's your problem |
| 12:55.21 | densh | it's from /var/log/asterisk/messages |
| 12:55.27 | mjordan | densh: that's just a log file |
| 12:55.32 | mjordan | if it came from "outside", you'd want to dig into what sent it to you |
| 12:56.18 | densh | no, it's the main IP address of the PBX machine |
| 12:56.24 | densh | same PBX |
| 12:56.32 | mjordan | then I'd check your config files |
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| 12:56.56 | densh | already looked there, nothing has changes, looks ok. |
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| 13:16.30 | ChB82 | Hello! I use "PauseQueueMember(,SIP/123)" to pause the member 123 in any queue. Now i want to remove this member from every queue he is logged in (and paused). But "RemoveQueueMember(,SIP/123) is not supported??? Any ideas? |
| 13:18.06 | mirela666 | ChB82: you can use queue_members_table as dynamic memeber table for each queue |
| 13:18.28 | mirela666 | ChB82: and then just delete or insert memeber to that table |
| 13:19.01 | mirela666 | ChB82: http://www.voip-info.org/wiki/view/Asterisk+RealTime+Queue |
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| 13:20.00 | mirela666 | ChB82: queues can be static if you like and have only dinamic members if that suits you |
| 13:26.57 | densh | Hi, any hints on ] ERROR[50660]: netsock2.c:263 ast_sockaddr_resolve: getaddrinfo("SAME_PBX_IP:5060:60628", "(null)", ...): Name or service not known ? |
| 13:27.10 | densh | why it glues two port numbers together? |
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| 13:30.22 | ChB82 | mirela666: Thank you very much ... seems to be much easier with working realtime config... |
| 13:30.56 | mirela666 | ChB82: indeed, you then depend on DB accessibility but much easyer |
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| 13:48.54 | *** join/#asterisk alex999 (~alex999@host172-102-static.15-188-b.business.telecomitalia.it) |
| 13:49.50 | alex999 | Is someone using the last release 13.4? |
| 13:50.50 | alex999 | Upgrading from 13.3 to 13.4 presence state doesn't work anymore. |
| 13:51.01 | WIMPy | The download page should have a counter. |
| 13:51.11 | alex999 | have you experinced the same? |
| 13:51.31 | WIMPy | No, I'm back on 11. |
| 13:51.46 | WIMPy | And I have no use for presence, either. |
| 13:52.02 | alex999 | Thanks WIMPy |
| 13:53.08 | alex999 | Someone else .... using BLF on phone |
| 13:53.46 | WIMPy | BLF is somethig else. |
| 13:54.50 | alex999 | BLF is directly connected to presence, isnt'it |
| 13:54.51 | phix | I wish DAHDI worked |
| 13:55.29 | WIMPy | phix: define "work" |
| 13:56.10 | phix | I pick up analogue phone and I have no dial tone, someone rings my POTS number and nothing rings |
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| 14:04.48 | alex999 | is someone experiencing issue with BLF after upgrading to Asterisk 13.4? |
| 14:07.07 | mjordan | alex999: there's nowhere near enough information to answer your question. The presence/device state tests are all passing with both chan_sip and chan_pjsip, so without specific details, the answer is "no, we aren't seeing that problem" |
| 14:10.03 | alex999 | thanks mjordan... I will check through out logs to see if I can find something to report |
| 14:10.53 | WIMPy | Might be a good start to describe the issue first. |
| 14:11.05 | seik0 | Loaded asterisk with 50 simultaneous calls each ending in voicemail and asterisk crashed |
| 14:11.20 | seik0 | twice already |
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| 14:27.04 | dan_j | Hi. Can anyone shed some light on this? There are two different brand SIP phones behind a NAT device which has SIP ALG enabled. I curious to know why one brand works fine without disabling SIP ALG and the other doesn't. |
| 14:27.12 | dan_j | Any ideas what the difference could be? |
| 14:27.51 | WIMPy | Luck? |
| 14:28.03 | WIMPy | ICE? TURN? |
| 14:29.00 | mjordan | dan_j: Usually that depends on where the SIP phones (or Asterisk - let's just say a UA) decides where to send the requests/responses |
| 14:29.18 | mjordan | that is, do you ignore the headers in the packet, or do you use that information? |
| 14:29.40 | mjordan | If you ignore the headers, where do you send the packets? To the IP address you received it from? Do you also send it to the port you received it from? |
| 14:29.48 | mjordan | NAT is fun. |
| 14:30.38 | rmudgett | mjordan: A lot of people wouldn't say that. :) |
| 14:30.49 | mirela666 | lol fun indeed |
| 14:31.29 | alex999 | mjordan: I have checked logs.... here what I found.... WARNING[9550] presencestate.c: No provider found for label CustomPresence... after any extension registration |
| 14:31.47 | alex999 | Any idea |
| 14:32.04 | dan_j | It looks like the headers on the broken phone say the following: |
| 14:32.09 | dan_j | <--- SIP read from UDP:SOURCEIP:5060 ---> |
| 14:32.09 | dan_j | Via: SIP/2.0/UDP SOURCEIP:5060;branch=z9hG4bKsf0175accf |
| 14:32.17 | dan_j | and the reply |
| 14:32.18 | dan_j | <--- Transmitting (NAT) to SOURCEIP:5060 ---> |
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| 14:32.33 | dan_j | So the ports match both the header and the actual port that was used to send |
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| 14:36.06 | mjordan | alex999: do you have the func_presencestate module loaded? |
| 14:38.52 | alex999 | module show like func_presencestate.so says.... func_presencestate.so Gets or sets a presence state in the dia 0 Running core |
| 14:39.53 | mjordan | hm. |
| 14:39.57 | mjordan | What is one of the hints that fails? |
| 14:40.02 | mjordan | And what does it show in 'core show hint'? |
| 14:41.19 | alex999 | all hints fail like "exten => 621,hint,PJSIP/621,CustomPresence:621" |
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| 14:43.16 | alex999 | to make more sense I will follow with my extension |
| 14:43.39 | mjordan | alex999: First, remove the CustomPresence label (for now) and see if the subscriptions work at that point. |
| 14:43.58 | mjordan | alex999: Then, I'd look into why the CustomPresence isn't being picked up. My guess is that you probably have some warnings at your startup |
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| 14:44.28 | mjordan | func_presencestate should be providing the CustomPresence provider, so your culprit is also possibly a module load order issue there. |
| 14:47.19 | alex999 | strange .I haven't altered any conf file ... just upgraded from 13.3 to 13.4 ... How can i check which is the correct order to load modules |
| 14:47.58 | mjordan | alex999: it may or may not be a problem. It depends on what your module.conf is. But I'm just guessing. I'd first check to make sure that things work without the CustomPresence provider. Make sure that's your actual issue. |
| 14:48.09 | mjordan | alex999: after that, make sure things are starting up without errors or warnings. |
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| 14:48.39 | mjordan | alex999: since func_presencestate provides the CustomPresence provider, make sure that it also is loaded and running. Try using some of its provided functions to make sure there aren't any issues. |
| 14:50.02 | alex999 | mjordan: note ... core show hints reports my extension in this way: 610@ext-local : PJSIP/610,CustomPres State:Idle Presence: Watchers 0 |
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| 14:53.43 | mjordan | alex999: k. I'd follow the steps I gave to diagnose what is actually happening. |
| 14:53.54 | mjordan | the custom presence provider error may be a red herring. |
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| 15:12.33 | linocisco | hi all, anybody has provisioned Grandstream GXP1405 phones ? |
| 15:17.12 | alex999 | mjordan: I would like to show you the log but i do not remember to what url I have to paste the text... |
| 15:17.48 | alex999 | I am not a frequent IRC user |
| 15:19.41 | linocisco | so many parameters I don't know what to do. its config files are not like polycom or others |
| 15:24.51 | alex999 | can someone tell me what website I should use to create a link with my text? |
| 15:26.54 | linocisco | alex999, pastebin.net |
| 15:33.09 | alex999 | linocisco: thanks indeed |
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| 15:34.24 | linocisco | there is always silent whisper in this room. not so many active users like #ubuntu #python. But asterisk said it has strong community support |
| 15:35.15 | alex999 | mjordan.... looking carefully at log I found this twice: [2015-06-24 16:56:55] WARNING[6019][C-0000003a] func_presencestate.c: PRESENCE_STATE unknown |
| 15:37.26 | cervajs2 | hello. i switched to asterisk13 from ast1.8 and i have problem with app_queue. i'm using static agents with this syntax member => Local/201@route_phones_1,1,mrbond,hint:201@subscribe_1 . the state of the agent is not changed when call appears. any ideas? i checked changelogs about queue/hints but no clues |
| 15:43.11 | alex999 | linocisco: unfortunately I cannot give you any help... I'm usign SNOM |
| 15:45.22 | cervajs2 | hmm. downgrade to 13.3.0 from 13.4.0. it looks like bug |
| 15:45.32 | mjordan | alex999: I'm going to go out on a limb and say that something went wrong with your upgrade. func_presencestate provides the PRESENCE_STATE function, which means ... something has gone strange. |
| 15:45.37 | *** join/#asterisk rsdvd (~rsdvd@90.206.88.142) |
| 15:45.39 | mjordan | And no, I don't think it's a bug |
| 15:45.57 | mjordan | at least for presence state.' |
| 15:46.46 | alex999 | well ... I'll try recompiling the source |
| 15:47.31 | alex999 | and see what happen |
| 15:47.49 | alex999 | mjordan: thank you a lot for now! |
| 15:48.20 | mjordan | np |
| 15:48.51 | mjordan | I'm doing a clean install from the tarball as well, just to double check. But we have a _lot_ of presence state tests that run against 13 nightly, and we haven't had an issue with them at any point during the development of 13.3/13.4 |
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| 15:59.25 | qakhan | hi all, h exten runs when call hangup, but why it runs when user answer the call. |
| 16:03.23 | qakhan | here is pastbin http://pastebin.com/tA4vwf3G |
| 16:19.02 | alex999 | mjordan: I did recompile the source and the issue is still present. Here is the link to the log http://pastebin.com/Xa4tkNXX . I appreciate what you are doing. I'm going home now. If you do not mind I will contact you tomorrow to know whether you are experiencing the same issue with the fresh installation from the tarball. |
| 16:19.49 | alex999 | Have nice day to you all! |
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| 17:07.54 | alex999 | mjordan.. hello .... I was able to get an irc app working on my phone. ... |
| 17:09.41 | alex999 | On my mobile phone ... of course |
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| 17:40.35 | dan_j | Hi, what does this mean? Not seen it before. |
| 17:40.56 | dan_j | chan_sip.c:3902 __sip_xmit: sip_xmit of 0x7fc2d47c2270 (len 595) to IPMASKED:0 returned -1: Invalid argument |
| 17:41.00 | dan_j | Over and over. |
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| 17:51.41 | caseyd | hey so I have asterisk set up and it seems to be impossible to get through my checkpoint firewall. I gave up and have two nic cards in my asterisk server one local and one external. Is this guide still valid - https://www.didww.com/Knowledgebase/sip_with_firewall_nat_using_asterisk/ Starting from the line: The easiest solution to this is to avoid NAT entirely. |
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| 17:53.52 | caseyd | I thought if there was a way I could have the traffic pass through into the local nic card and then out with the external nic, that it would work.. and it does,but now I can't make extension to extension calls |
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| 18:08.01 | *** join/#asterisk c0ldg0ld (~c0ldg0ld@unaffiliated/c0ldg0ld) |
| 18:08.51 | c0ldg0ld | hello all, is there a doc anywhere that I'm missing that shows proper procedure for doing kernel updates with a production asterisk system? I'm assuming you just need to recompile but want to make sure before I get into a maint window |
| 18:09.31 | c0ldg0ld | CentOS 6.x and vanilla Asterisk |
| 18:11.30 | carrar | perhaps the document that explains not to do that on a production box? |
| 18:12.10 | c0ldg0ld | alrighty... assuming you need to do a kernel update for security reasons. It has to get done at some point |
| 18:12.11 | *** join/#asterisk theron (~theron@199.201.64.131) |
| 18:12.21 | c0ldg0ld | these are businesses and things can be done in off hours |
| 18:13.56 | carrar | as long as you have a good backup in case you need to re-install the OS and re-install your asterisk setup |
| 18:14.02 | c0ldg0ld | Are you saying that the only way to EVER update the underlying linux system is to forklift the box and start over? |
| 18:14.16 | c0ldg0ld | Yes, all these systems are backed up remotely every night |
| 18:14.43 | c0ldg0ld | I'm honestly not worried about having to rebuild I just thought there must be a way to update the OS without blowing everything away |
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| 18:15.09 | carrar | do it in a vm so you can just copy the image, upgrade it, then make it the primary |
| 18:15.10 | c0ldg0ld | I've left it in the past, some of these systems have been running for years |
| 18:15.27 | carrar | who does bare metal anymore |
| 18:15.32 | c0ldg0ld | can't do VM, have hardware telephony cards that don't play well with vm |
| 18:15.50 | c0ldg0ld | or at least didn't when I installed back when |
| 18:16.21 | carrar | as long as you know what you're doing, it really has nothing to do with Asterisk |
| 18:16.26 | c0ldg0ld | ha, people who have physical T1s coming in |
| 18:16.37 | carrar | use a adtran |
| 18:16.44 | carrar | off load that to a external box |
| 18:16.50 | c0ldg0ld | I know that, I just broke a few boxes back in the early days when the kernel up dated |
| 18:17.09 | c0ldg0ld | T1 ->> adtran -> adtran -> sip trunk?> |
| 18:18.15 | c0ldg0ld | Or rather t1 -> adtran -> siptrunk ->asterisk |
| 18:18.16 | carrar | only 1 adtran |
| 18:18.24 | c0ldg0ld | yeah realized that was confusing |
| 18:18.29 | carrar | but yeah it turns your tdma PRI into sip |
| 18:18.37 | c0ldg0ld | play nice with fax? |
| 18:18.43 | carrar | seems to |
| 18:18.48 | carrar | they supprot t.38 |
| 18:18.49 | c0ldg0ld | Hmmm |
| 18:19.01 | c0ldg0ld | will check that out. Thanks |
| 18:19.29 | carrar | they are pretty much a telco standard |
| 18:20.00 | carrar | (Adtran Total Access 924e) |
| 18:20.01 | c0ldg0ld | Using some old tsu600's for analog channel banks currently |
| 18:21.36 | carrar | If you are heavy on faxs then Adtran NetVanta 644 |
| 18:21.50 | c0ldg0ld | Would make failover much easier too |
| 18:22.02 | carrar | yup |
| 18:22.16 | c0ldg0ld | time to hit the boss up for a test unit |
| 18:22.29 | c0ldg0ld | Not heavy faxing, just a few per site |
| 18:22.31 | carrar | Try for a NetVanta 644 |
| 18:22.46 | carrar | then settle for for a 924e 3rd gen |
| 18:22.48 | carrar | heh |
| 18:23.00 | c0ldg0ld | just had issues with some of the audiocodes mediant deals from the telco at one point so I always ask about faxing |
| 18:23.23 | carrar | Well T.38 is already a bit tricky |
| 18:23.26 | carrar | err always |
| 18:23.53 | carrar | keeping it analog will probably always work better |
| 18:24.11 | c0ldg0ld | Yeah with the old school PRIs we haven't had a bit of trouble |
| 18:25.36 | c0ldg0ld | oh yeah, looking at the prices I'll never get a netvanta, much less 17 of them! |
| 18:27.57 | carrar | firmware is free, so if you can buy a used one you can update it |
| 18:29.34 | c0ldg0ld | yeah even used he balked but I do now have a 924e inbound |
| 18:30.02 | carrar | well if you can't afford one, your buisness model is broken |
| 18:30.32 | carrar | heh |
| 18:30.56 | c0ldg0ld | yeah well... budgets. |
| 18:31.06 | c0ldg0ld | this aint a small operation, I'm a little fish |
| 18:31.33 | c0ldg0ld | so not my model |
| 18:31.35 | carrar | 17 asterisk boxes is not a little fish |
| 18:32.07 | c0ldg0ld | each with around 30 or less endpoint for the most part. My nortel is another story |
| 18:32.39 | c0ldg0ld | But suffice it to say they make you do more with less, hence asterisk in the first place |
| 18:33.05 | c0ldg0ld | these clinics would have ended up with 1/4 of the features using used norstar key systems probably otherwise |
| 18:33.34 | carrar | and cost 100 adtran's worth |
| 18:33.53 | c0ldg0ld | It's not that it's impossible it's just super painful once things get to a certain dollar amt and become capital projects |
| 18:34.24 | c0ldg0ld | so when I tell them. I'd like to buy this $4000 used piece of equip because I wanna test something. They laugh |
| 18:34.28 | carrar | pain to me, is when I have to wake up at night |
| 18:34.33 | carrar | painfull that is |
| 18:34.37 | c0ldg0ld | Yeah me too, but they don't care about htat |
| 18:34.42 | c0ldg0ld | and these places are closed at night |
| 18:35.03 | c0ldg0ld | The nortel though is full time and I've had plenty of 4am calls |
| 18:35.06 | carrar | So ask for a eval unit |
| 18:36.05 | carrar | try it out |
| 18:36.27 | carrar | return it when your done |
| 18:36.28 | c0ldg0ld | What so much better about the netvanta vs the total access? |
| 18:36.35 | carrar | more DSP's |
| 18:36.44 | carrar | if you are heavy on faxes |
| 18:36.52 | carrar | which requires a DSP |
| 18:36.54 | c0ldg0ld | like I said only 1-2 faxes per box |
| 18:37.18 | carrar | 924 can do 4 concurrent per box |
| 18:37.25 | c0ldg0ld | ah |
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| 18:37.30 | carrar | 908e might also be fine |
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| 18:38.57 | c0ldg0ld | will evaluate the 924 and maybe graduate from there on the boxes that might need more |
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| 18:40.10 | russlevy | hi, i recorded a call using Monitor, and I have the -in and -out files saved separately. they are approx 600s, but there is a 9s difference between the two. they start at the right time, but it looks like one is recorded either slower, or faster, than the other as they finish at different times, and as the call goes on, the lag between the two gets longer |
| 18:40.34 | russlevy | both are through chan_sip, one through an opus websocket, the other standard sip |
| 18:40.39 | russlevy | is there anything that may cause this? |
| 18:40.49 | c0ldg0ld | This whole deal is started out as a "lets see if we can make it work" in 2009 and they added clinic after clinic since then till I got to a point where I had to step back and say "I need to kinda standardize on what I've learned in these years, get all the boxes the same etc" which is where I am now. Retroactively trying to standardize because the early boxes I didn't really do things as well as I could |
| 18:42.17 | cmendes0101 | russlevy: Do you need separate recordings? Mixmonitor might solve the issue since its mixing them together for the recording |
| 18:42.44 | carrar | c0ldg0ld, what about using a hosted model? |
| 18:43.10 | c0ldg0ld | carrar: All hosted at one site |
| 18:43.22 | carrar | in a DC |
| 18:43.49 | carrar | then when they loose power, things still work and will go to their cell |
| 18:44.11 | c0ldg0ld | carrar: I've considered it, the only reason I haven't is that relies on a lot of factors and any one of them going down causes all locations to go down |
| 18:44.25 | c0ldg0ld | right but there's also a centralized point of failure for all of them a |
| 18:44.48 | carrar | so the cost you will save on 17 boxes you can build a little redundency |
| 18:45.13 | c0ldg0ld | we've had issues were our "MAN" connections have gone down but the clinics were able to operate independently with their phones |
| 18:46.07 | c0ldg0ld | We've got some guys here that run the network that I'm not sure I want to trust my stuff to... unfortunately |
| 18:46.13 | russlevy | cmendes0101: i do need separate |
| 18:46.49 | russlevy | is this a normal issue? |
| 18:47.05 | c0ldg0ld | They have had issues that spanned months where their cores were maxed out for no reason apparent to them. That's the reason I haven't had all PRI come to this location and then distribute out. That and e911 stuff |
| 18:47.24 | c0ldg0ld | which I'm sure can be overcome in software |
| 18:49.48 | c0ldg0ld | carrar: Have IAX2 trunks between a few locations already for 4 digit dialing. Honestly, like I said I got into this with just enough know-how to make a box work and have grown on it ever since but am probably not following best practices very well by consequence |
| 18:51.14 | carrar | yeah, time to think about scalability |
| 18:51.25 | *** part/#asterisk cervajs2 (~cervenka@gatekeeper.bm.ipex.cz) |
| 18:51.35 | c0ldg0ld | Ha, that time was probably 3 years ago but was too busy! |
| 18:51.36 | carrar | or move to a cookie cutter setup |
| 18:53.22 | c0ldg0ld | That's kinda what I was thinking. I've started building files to #include in my dialplan that have all the subroutines I generally will add to a box to control time-of-day routing, toll free, LD, and local calls, queue logins and outs, etc and was thinking about setting up a kickstart install for whichever version of Centos I decide to use and making all the boxes the same basic starting point then adding |
| 18:53.24 | c0ldg0ld | the custom-to-that-location stuff |
| 18:54.59 | c0ldg0ld | Also thankfully, ditching the 7960 sets and moving to D40's |
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| 18:59.45 | cmendes0101 | russlevy: Haven't really seen it before. Check the sample rate and everything of both to compare if there are any differences in that. I think soxi should be good |
| 19:06.23 | russlevy | cmendes0101: everything is the same -- just different sample #s |
| 19:06.39 | russlevy | cmendes0101: during the conversation everything was fine |
| 19:07.28 | cmendes0101 | and the duration total also showed the 9 second difference? |
| 19:07.52 | russlevy | that's where i see it |
| 19:08.14 | russlevy | and when i manually mix the -in and -out file together you hear that they start off nicely, and then one is just sampling badly |
| 19:08.32 | russlevy | one has a really nice length of 626.0000, the other 617.2800 |
| 19:08.47 | cmendes0101 | default format wav? |
| 19:09.01 | cmendes0101 | from the monitor() command |
| 19:09.03 | russlevy | yes -- didn't do anything fancy |
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| 19:10.34 | cmendes0101 | Hmm, ok I don't really know. Seems like everything should be good. Haven't played with monitor in a while but don't remember recording at that length |
| 19:10.45 | russlevy | Monitor("SIP/1060-00000000", "wav,/filename,bm") |
| 19:11.38 | cmendes0101 | Just curious, does it matter if they are different if you don't need to mix them together or you need separate and mixed? |
| 19:14.36 | cmendes0101 | Actually isn't the m parameter telling it to mix together and delete -in and -out? |
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| 19:24.25 | russlevy | cmendes0101: m means run an external program |
| 19:25.16 | russlevy | i need separate and mixed -- i need to process the files, and then mix them together afterwards |
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| 19:53.26 | coreyfarrell | has anyone done multiple instances of asterisk on the same system using alternate config paths like the testsuite does? I'm looking for any information about issues with doing this in production. |
| 19:57.21 | carrar | coreyfarrell, how is it working out in testing? |
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| 20:01.16 | coreyfarrell | the testsuite runs successfully every night, so it works fine in testing. but the testsuite rarely handles more than a handful of calls before destroying the instances and recreating new ones. |
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| 20:11.53 | carrar | No better place to test then production :) |
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| 20:17.30 | pcAngel | If there's any jira admins or bug marshals, I'm hoping issue #25157 can be updated. It's set to "Waiting for feedback" and assigned to me (username Dade) but our network admin (username atna99) has posted the debug logs and core traces. I'm unsure how to proceed with any development and am hoping this issue can get back to a non-waiting-on-me-state. Separate debug messages have been posted |
| 20:17.31 | pcAngel | on three occasions including today. |
| 20:18.53 | pcAngel | I just found the "Enter feedback" link, so you guys can ignore that last message, thanks! |
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| 23:20.19 | roler | We have an Elastix NLX4000 with a Sangoma B601 PRI card. Ocasionally when dialing into conference calls, the DTMF tones aren't recognized. Dialing the test line 253-243-3381 does NOT report back what I dial in, and it even adds random digits. It works perfect from my cell phone. Is there a setting some where to change how DTMF tones are recognized? |
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