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| 00:21.40 | snadge | <PROTECTED> |
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| 02:42.40 | captain118 | I've got a weird problem I need help with. |
| 02:43.17 | captain118 | I have an extension that is onsite that is configured both on the phone and the freepbx instance the exact same. |
| 02:43.34 | captain118 | *the same as all the other extensions |
| 02:44.31 | captain118 | This one extension will be ok for 12 min. Then go into an unknown state, then 38 min later it will come back to being ok. |
| 02:44.45 | captain118 | It is very consistent. |
| 02:44.54 | captain118 | Thoughts? |
| 02:47.10 | drmessano | None of the other phones have an issue? |
| 02:47.25 | captain118 | Nope |
| 02:47.47 | drmessano | Check cabling, the phone itself, switch port.. |
| 02:47.57 | drmessano | Reset it |
| 02:47.58 | captain118 | Yes. |
| 02:48.02 | drmessano | Start over with ti |
| 02:48.03 | drmessano | it* |
| 02:48.04 | captain118 | No errors on the switch either. |
| 02:48.14 | drmessano | Is the port staying up? |
| 02:48.18 | captain118 | Yes |
| 02:48.28 | captain118 | Calls can be made going out just not incoming |
| 02:48.46 | captain118 | And the PC connected to the phone doesnt have any issue |
| 02:49.36 | captain118 | Incoming calls while in the "unknown" state go to voicemail |
| 02:49.36 | drmessano | The phone is not on some funky different network segment? |
| 02:49.53 | captain118 | One vlan its a 8 person office. |
| 02:50.09 | drmessano | What make is the phone? |
| 02:50.22 | captain118 | Its a grandstream 1610 |
| 02:50.44 | captain118 | All of them are. |
| 02:51.19 | drmessano | I would reset the phone, upgrade firmware.. Try swapping it out for another device |
| 02:51.42 | drmessano | Provision another phone as this extension, put it in place |
| 02:51.43 | captain118 | The thing that worries me is it was working fine till I added a different extension. |
| 02:52.12 | captain118 | Once I added the other extension (which is working just fine) it started doing this. |
| 02:54.19 | captain118 | I'll go ahead and try resetting the phone and reconfiguring it from scratch. It just worries me that the changes made that day do not involve any settings on that device. |
| 02:55.08 | drmessano | I cant see how they are related, unless you're using some kind of provisioning and used the same MAC address |
| 02:56.21 | captain118 | The phones are manually configured at the moment. So wouldnt be that either. |
| 02:57.26 | drmessano | Yep, sounds even more like coincidence. Beat the phone up a bit.. update, reset.. then move on to replacing it, even temporarily. |
| 02:58.17 | captain118 | Ok I'll do that. While I'm testing do you know the asterisk command to see the full config for an extension in the cli? |
| 02:58.32 | captain118 | its a sip extension |
| 02:58.58 | drmessano | I assume you mean SIP debug.. are you using chan_sip? |
| 02:59.07 | captain118 | yes |
| 02:59.54 | drmessano | sip set debug on or sip set debug peer <peer> |
| 03:01.04 | captain118 | ok. I guess I'm too used to cisco. I was thinking show config... thanks |
| 03:01.24 | drmessano | No probs |
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| 03:20.26 | captain118 | when you run 'sip show peer <extension>' what is the Expire variable? |
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| 04:04.34 | captain118 | drmessano: apparently it was a bug in asterisk. I'm running 11.20.0. |
| 04:04.44 | captain118 | drmessano: https://issues.asterisk.org/jira/browse/ASTERISK-25476 |
| 04:04.52 | captain118 | Time for an upgrade |
| 04:05.15 | drmessano | You said just 1 device is acting up? |
| 04:05.32 | captain118 | Yea |
| 04:05.49 | captain118 | At least that was what was reported to me. |
| 04:06.06 | drmessano | Im familiar with the issue.. It should be affecting more than 1 phone |
| 04:06.14 | drmessano | Worth a try |
| 04:06.29 | drmessano | But in my experience, it wasnt just 1 phone.. |
| 04:06.30 | captain118 | After we talked last I noticed the conference rooms were having the same issue. But no one was reporting those because no one calls into those. |
| 04:06.37 | drmessano | Ah |
| 04:06.47 | drmessano | Well good.. 2 is better than 1 |
| 04:06.51 | captain118 | 2 of the 3 conference rooms were having the issue |
| 04:07.01 | drmessano | Even better.. |
| 04:07.09 | drmessano | I would say theres a good chance thats it.. upgrade away |
| 04:07.22 | captain118 | thanks for your help |
| 04:07.53 | drmessano | Had we known it was more than 1 the symptoms you described on the 1 phone sound exactly like that issue |
| 04:08.02 | drmessano | We had a lot of it reported at the time.. same deal |
| 04:08.12 | drmessano | So good luck |
| 04:08.17 | captain118 | thanks |
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| 08:04.23 | Haris | https://issues.asterisk.org/jira/browse/ASTERISK-24597 |
| 08:13.41 | Haris | pkg names need to be updated in contrib/scripts/install_prereq install |
| 08:13.44 | Haris | script |
| 08:13.51 | Haris | for EL7 |
| 08:13.56 | Haris | centos 7 |
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| 09:54.57 | ashysaini | Does asterisk 13 consumes more resources (CPU, Memory) than asterisk 11? any idea? |
| 09:56.03 | JamesO | in my personal experience i found 13 to have less overhead than 11, but it wasn't on a live production system |
| 09:58.33 | ashysaini | @James, i am running same application on asterisk 11 and 13(production), getting over 15 cpu load on just 50 to 60 calls |
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| 10:00.12 | JamesO | I never ran more than 25 calls on mine, it was only testing/playing but i never hit anything like that or even looked to creep up - you may wanna wait for someone more knowledgable on the subject |
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| 10:39.16 | Haris | I'm installing it as per freepbx wiki for centos 7 |
| 10:39.22 | Haris | I can't get it to work |
| 10:40.13 | Haris | at first when I installed on ubuntu, freepbx was storing extension config in db. now, on centos 7, its storing it in files ? and then its complaining about there being no vm provider ? wt! h |
| 10:40.43 | Haris | installing 13 from source |
| 10:40.50 | Haris | as per http://wiki.freepbx.org/display/FOP/Installing+FreePBX+13+on+CentOS+7 |
| 10:41.35 | Haris | do I have to select a freepbx or asterisk mod in freepbx gui for it to store extension config in db ? |
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| 11:21.48 | pa | hi |
| 11:21.54 | pa | maybe this has been asked already |
| 11:22.15 | wyoung | hi pa |
| 11:22.50 | pa | but starting from today asterisk began to vomit an infinite amount of errors "TLS connection for client 'gtalkusername' cannot be established. OpenSSL initialization failed. |
| 11:22.55 | pa | is it some known bug? |
| 11:23.17 | pa | or did google plug off the xmpp support? |
| 11:23.40 | wyoung | pa: *shrugs* |
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| 11:28.32 | pa | maybe certificates changed? |
| 11:28.34 | pa | donno.. |
| 11:28.38 | pa | it started right now.. |
| 11:29.34 | Haris | I have asterisk + freepbx both version 13. I'v configured two ext. when I connect through fpbx ucp for one user. it says; |
| 11:29.42 | Haris | [2016-06-14 07:27:30] VERBOSE[27503] res_http_websocket.c: WebSocket connection from '192.168.2.61:64984' for protocol 'sip' accepted using version '13' |
| 11:29.42 | Haris | [2016-06-14 07:27:30] NOTICE[27503] chan_sip.c: Registration from '<sip:991060@192.168.0.98>' failed for '192.168.2.61:64984' - Wrong password |
| 11:29.49 | Haris | how can I fix the wrong password part |
| 11:29.56 | wyoung | Haris: use the correct password |
| 11:30.13 | Haris | how or where do I set the password, so they are the ones I use ? |
| 11:30.28 | wyoung | /etc/asterisk/sip.conf |
| 11:31.16 | Haris | I have two sets of credentials. one in db, one in sip.conf. which is applicable ? |
| 11:31.57 | Haris | sip.conf says ext no = password. but when I login to fpbx ucp, it says the above |
| 11:34.00 | wyoung | Haris: I only use asterisk |
| 11:34.07 | wyoung | so I dont know about freepbx |
| 11:34.38 | Haris | I'v removed everything from sip_additional.conf, included in sip.conf. there's no ext/secret configured in the sip.conf files now. the set of credentials are in db |
| 11:35.03 | pa | well i guess i have to wait for someone else having the same problem.. |
| 11:35.14 | Haris | hmm |
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| 11:45.41 | pa | wyoung, found this: http://pbxinaflash.com/community/threads/gv-stopped-connecting-today.19394/ |
| 11:46.39 | wyoung | pa: I am lazyt |
| 11:50.14 | Haris | I deleted and re-created user/pass. its still saying wrong pass |
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| 11:54.22 | Haris | why does asterisk say status=unreachable for an ext on which I just registered ? |
| 11:56.18 | Haris | as soon as I register, I get the msg --> |
| 11:56.20 | Haris | [2016-06-14 07:54:59] VERBOSE[4382] chan_sip.c: Unregistered SIP '991060' |
| 11:56.20 | Haris | [2016-06-14 07:55:45] VERBOSE[4382] chan_sip.c: Registered SIP '991060' at 192.168.2.61:52443 |
| 11:56.20 | Haris | [2016-06-14 07:55:49] NOTICE[4366] chan_sip.c: Peer '991060' is now UNREACHABLE! Last qualify: 0 |
| 11:56.36 | Haris | wth happened ? |
| 11:56.49 | Haris | 99xxxx is the web socket clone of the original ext |
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| 12:08.58 | wyoung | ? |
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| 12:10.45 | Haris | unreachable status problem |
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| 12:18.57 | Samot | Haris: You're asking a lot of FreePBX related questions in here. They aren't going to be able to answer most in here because most don't use any GUIs in their Asterisk installs. |
| 12:29.42 | Haris | hmm |
| 12:30.19 | Alblasco1702 | i have a problem with asterisk and grandstream ht-503 fxo and the IVR as asterisk picks up and when the ivr is playing the caller hansup asterisk leaves the line open. |
| 12:31.01 | Alblasco1702 | where do i need to look? |
| 12:31.20 | file | it would be the responsibility of the HT-503 to inform Asterisk that the call has ended |
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| 12:31.38 | scv | and considering its a grandstream |
| 12:31.41 | scv | heh |
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| 12:33.49 | wyoung | hey wil_syd2 |
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| 12:56.29 | [TK]D-Fender | The HT clearly doesn't know it ended, because there isn't some option to have it not actually kill the call with * |
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| 12:58.04 | [TK]D-Fender | Which means you have either no CDS on the line, or an inproper standard selected based on your line. |
| 12:58.06 | [TK]D-Fender | ~cds |
| 12:58.06 | infobot | [~cds] Call Discconect Supervision is a service placed on analog lines to be able to signal you that that the calling party has hung up. This is typically done either by a momentary battery cut, or by a polarity reversal on the line. |
| 13:00.03 | Alblasco1702 | [TK]D-Fender, ok it is on the netherlands maybe you know what is used here? |
| 13:00.38 | [TK]D-Fender | Since I don't live there... no. |
| 13:00.42 | [TK]D-Fender | Check the setting in your HT |
| 13:01.19 | Alblasco1702 | [TK]D-Fender, ok i look at it thnx for the direction. |
| 13:04.03 | [TK]D-Fender | Best of luck with it |
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| 13:09.36 | Alblasco1702 | thnx |
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| 15:42.32 | monsterco | What is the recommended technology to use for a trunk on Asterisk 13? pjsip or sip? *ALSO*, what what are the cons/benefits of using either one when it comes to management headache? |
| 15:43.14 | scv | https://i.gyazo.com/6fee544ca12079d3a6f7917c8aa14910.png |
| 15:43.55 | drmessano | ROFL |
| 15:44.22 | scv | can't make this stuff up |
| 15:44.25 | WIMPy | "trunk" implies IAX, doesn't it :-) |
| 15:44.47 | drmessano | A SIP Trunk is a box you store your unused phones in |
| 15:45.08 | WIMPy | Sounds usefull. |
| 15:45.40 | drmessano | Whereas an IAX Trunk is a rare, endangered create not often spotted in the wild |
| 15:45.50 | drmessano | creature |
| 15:46.17 | drmessano | "Look Ma, an IAX trunk.. " "Shut up, Billy. no one uses IAX" |
| 15:51.19 | Samot | No, they use IAX2 |
| 15:51.30 | Samot | Because it's better. |
| 15:51.59 | Greenlight | IAX2 is great, especially trunked over low bandwidth links, or though funky firewalls |
| 15:55.03 | [TK]D-Fender | If you're running multiple calls. |
| 15:55.12 | [TK]D-Fender | For single there is no real savings |
| 15:55.17 | Greenlight | Ofc |
| 15:55.46 | [TK]D-Fender | monsterco, pjsip is recommended over chan_sip so long as it works. |
| 15:55.58 | drmessano | Is bandwidth still an issue anymore? |
| 15:56.17 | Greenlight | From time to time |
| 15:56.20 | Samot | Geographically, yes. |
| 15:56.21 | [TK]D-Fender | It is in those shittry remote areas where DSL is all you can get your hands on. |
| 15:56.29 | Samot | In North America, probably a no. |
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| 15:56.38 | Greenlight | Some areas I bet it still is |
| 15:56.45 | Samot | Small. |
| 15:56.46 | [TK]D-Fender | Actually.... North America has it worst almost... |
| 15:56.47 | Greenlight | DSL can be slow still, especially upload |
| 15:57.10 | Samot | It's still better than like India. |
| 15:57.32 | drmessano | Sure, but that many places where the slight savings of using IAX2 trunked over SIP saves the day? |
| 15:57.42 | Samot | Oh I doubt it. |
| 15:57.44 | Greenlight | I've had 200 seat call centres get let down by internet provider, and so going over 4G metered dongle for weeks |
| 15:57.49 | drmessano | That was my point |
| 15:57.50 | Samot | First you have to find a provider that does IAX. |
| 15:58.00 | Greenlight | G729 IAX2 trunked gets that down to like 2meg |
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| 15:58.27 | drmessano | You said the G-dash-dash-dash word |
| 15:58.41 | Greenlight | It's a shame more providers don't but it's peanuts to get a server in a datacentre and change from IAX2 to SIP if needed |
| 15:59.27 | freebs | trolls for drmessano |
| 15:59.40 | drmessano | bans freebs from the salad bar |
| 15:59.43 | freebs | lol |
| 16:00.50 | drmessano | Greenlight: I'm mostly kidding about the G729 thing.. I actually run G729 on my home system because I can |
| 16:01.09 | Greenlight | It's great for what it is |
| 16:01.13 | drmessano | I dont care about the call quality, or the bandwidth, it just makes me feel elite |
| 16:01.23 | Greenlight | Squeezing it down to that sort of bandwidth is impressive |
| 16:02.00 | Greenlight | And it also does that thing, can't think of the name off hand, where it fills in missed packets |
| 16:05.16 | drmessano | G729+TLS+SRTP = Secret Squirrel Bell |
| 16:06.39 | drmessano | Someone needs to create a digital artifact module that intentionally adds weird digital artifacts every so often to a call to make it sound more secret and stuff |
| 16:07.12 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 16:07.29 | [TK]D-Fender | And the visual call analyzer should have scrolling text stripes like in the Matrix. |
| 16:08.39 | Samot | Just remember TLS+SRTP is a resource hog. |
| 16:09.10 | drmessano | Samot: It's ok.. I purchased one Sangoma DSP card for each call |
| 16:09.18 | drmessano | I should be Lit |
| 16:09.38 | Samot | Yes you should. |
| 16:10.16 | Greenlight | I would have thought G729 is more of a resource hog than TLS+SRTP ? |
| 16:10.26 | Greenlight | Assuming no hardware cards |
| 16:10.44 | drmessano | I also bought a 2 port PRI card.. all the calls are looped out one channel and back in another, and I have a bunch of ferrite chokes on the cable to choke out stray resources |
| 16:10.47 | Samot | I was being facetious. |
| 16:10.53 | drmessano | Greenlight: Theres a story there |
| 16:11.32 | drmessano | Like someone selling a software SBC that requires a DSP add-in card to do SRTP because of it's high resource usage |
| 16:11.34 | drmessano | Allegedly |
| 16:11.38 | Samot | The moral of it was VMs suck and can't do TLS+SRTP without sucking up 50% of the resources and not having a DSP card. |
| 16:11.48 | Greenlight | You joke about the PRI loopback... but before chan_local that's what you had to do... |
| 16:12.39 | drmessano | Samot: You know what I find hilarious? |
| 16:12.59 | Samot | Teacup Pigs inside a large tea cup? |
| 16:13.08 | Samot | Wearing top hats. |
| 16:13.24 | Greenlight | I wish they'd bring back top hats |
| 16:13.25 | drmessano | When you have a VM utilizing extensive resources, do you know the best way to alleviate that? |
| 16:13.39 | Samot | What's that? |
| 16:13.44 | drmessano | Allocate more resources |
| 16:13.51 | drmessano | So like.. couldnt you do that |
| 16:13.52 | Samot | No. |
| 16:14.00 | Samot | Stop. |
| 16:14.07 | drmessano | Like add a core |
| 16:14.08 | Samot | mind blown. |
| 16:14.10 | drmessano | or RAM |
| 16:14.22 | drmessano | or another PS/2 port |
| 16:14.26 | drmessano | or something |
| 16:14.45 | Samot | I'm all like that news reporter in Scanners. |
| 16:14.49 | drmessano | But the DSP card sounds legit too |
| 16:15.07 | drmessano | See if you can get a couple of their USB timing dongles too |
| 16:15.34 | Greenlight | I've had great results running Asterisk on ESXi with high load. Granted it's not as fast as it would be on the raw hardware but it's close enough that the benifits of it being a VM outweigh that |
| 16:16.14 | monsterco | as for as PJSIP and SIP go, does ARI care? |
| 16:16.20 | Greenlight | Nope |
| 16:16.23 | monsterco | or can the two be interchanged easily? |
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| 16:19.34 | drmessano | monsterco: Well you obviously need to use the correct channel type in your request |
| 16:19.37 | drmessano | So yes it does |
| 16:20.12 | drmessano | PJSIP is PJSIP, SIP is SIP... they are the same protocol, but they are not the same channel driver.. they are each other own island |
| 16:20.32 | drmessano | Just the same as IAX2 != SIP |
| 16:20.48 | drmessano | Just the same as IAX2 != CHAN_SIP != CHAN_PJSIP |
| 16:21.05 | monsterco | :) got it |
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| 16:37.26 | drmessano | One of these days I need to spend more time with PJSIP |
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| 16:54.49 | Samot | Yeah. |
| 16:55.30 | Samot | I might have to start messing with it a lot more in the next few weeks. It's looking like the best option for a client and how they want to deploy. |
| 16:57.13 | Samot | You know, phone in the office with a softphone on the mobile and a phone at the home office. |
| 16:59.00 | Samot | But the IP phones will have 4 extensions and about 16 BLFs for other extensions and parking lots. |
| 17:00.57 | Samot | And I think they're going to want call pickup groups |
| 17:01.09 | Samot | So I know some of that has been problematic with PJSIP. |
| 17:02.59 | drmessano | I like the whole multiple endpoints using the same extension thing, and PJSIP looks like the way to go there. |
| 17:07.34 | Samot | Yeah. |
| 17:07.52 | Samot | Back in the day we used OpenSER and IPCentrex for that. |
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| 17:08.06 | Samot | OpenSER for regular users and IPCentrex for PBX. |
| 17:08.33 | Samot | So there could be multiple AORs. |
| 17:24.55 | pa | ok, i think i found out what was the nasty problem with motif |
| 17:25.14 | pa | apparently google stopped authenticating if there's more than one account (per IP i guess) |
| 17:30.12 | pa | sorry, not |
| 17:31.26 | pa | failing also with only one account.. |
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| 17:46.59 | drmessano | pa: Im interested to know what you find |
| 17:47.16 | drmessano | This looming Google pulling the plug thing has been a long time coming |
| 17:47.36 | Samot | Well even their ToS says, "Expect it to not work from time to time" |
| 17:47.39 | Samot | Or close to it. |
| 17:48.22 | Samot | It would be interesting once they do. |
| 17:48.38 | Samot | Because iirc the ToS says they don't need to tell you. |
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| 17:53.23 | drmessano | Google has become an app company that also have web interfaces to those apps |
| 17:53.31 | pa | drmessano, for now it stopped working no matter what. I'm trying to upgrade to asterisk 13 to see if that helps |
| 17:53.55 | drmessano | I dont think 13 will make a difference |
| 17:53.59 | pa | ah.. |
| 17:54.16 | drmessano | You're on Motif, right? |
| 17:55.49 | pa | yes |
| 17:56.15 | drmessano | So moving from 1.8 to 11, which was the older res_jabber/chan_gtalk to res_xmpp/chan_motif.. Completely different apps. Thats why "upgrading" was a magic bullet |
| 17:56.21 | drmessano | Because the old modules sucked |
| 17:56.34 | drmessano | The new ones are fine, and moving to 13 won't make anything work better |
| 17:56.46 | pa | ah i see |
| 17:57.04 | pa | well i found around others with my problem, saying that things restarted working by themselves after a while |
| 17:57.16 | drmessano | It may just be "down" |
| 17:57.24 | pa | do you know how to enable motif debugging? |
| 17:57.28 | drmessano | Maybe they have an XMPP entry point that crapped the bed |
| 17:57.29 | pa | maybe it tells a bit more |
| 17:58.06 | pa | drmessano, i thought about that.. but i find strange that nobody else complains.. Is it only me using asterisk with google voice? |
| 17:58.50 | drmessano | "an XMPP entry point", not "the".. Im guessing, and suggesting a very limited outage |
| 17:59.19 | drmessano | If they multiple servers hosting XMPP sessions and one is down, that could be you |
| 18:00.19 | drmessano | set debug=yes in the general section of xmpp.conf |
| 18:00.41 | pa | ah thanks i try |
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| 18:07.54 | MrKeuner | hi, basic voip user here... I have a pap2 device, can I block callers? Do i need Asterisk for this? |
| 18:09.15 | drmessano | Is your PAP2 locked to your provider? |
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| 18:10.58 | drmessano | *60 may or may not block the last caller depending on how its set up |
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| 18:51.23 | pa | drmessano, so this is what happens apparently |
| 18:51.24 | pa | http://codepad.org/N7opOsaX |
| 18:51.36 | pa | doesn't look like an error |
| 18:51.49 | pa | but i don't understand how xmpp works, maybe it is |
| 18:52.14 | pa | it might be that the error should be below the debug not above |
| 18:52.17 | pa | it's just an infinite stream |
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| 19:02.35 | drmessano | How do you run any updates recently |
| 19:02.40 | drmessano | Have you |
| 19:02.51 | drmessano | Sorry I'm on my phone using text to speech |
| 19:06.53 | pa | no, not really |
| 19:07.09 | pa | it stopped logging in abruptly this morning |
| 19:07.21 | pa | then i updated, but it didnt help |
| 19:13.15 | Samot | I had a client say their GV was down but now it's back up |
| 19:19.25 | pa | ah.. |
| 19:19.31 | pa | let me retry |
| 19:20.24 | pa | here apparently not yet.. |
| 19:20.48 | pa | Samot, what IP does talk.google.com resolves to, for you? |
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| 20:00.02 | doolittlework | evening gents |
| 20:00.36 | doolittlework | been playing around with dundi linking two asterisk boxes via iax |
| 20:01.02 | doolittlework | i have this working where i can route calls from one to the other with no problems |
| 20:01.57 | doolittlework | i am looking to create a queue on the one box and route calls to a agent on the other box, the problem is that i can no seem to get the call routed to the remote box |
| 20:03.45 | doolittlework | i think it is how one defines the agent in the queues.conf file |
| 20:04.14 | doolittlework | thinking that the call is local where it is actualy remote |
| 20:04.30 | [TK]D-Fender | You call it no different that you would as dialed |
| 20:09.00 | doolittlework | i think the problem is that i still not fully understand the routing when it comes to using dundi, i followed some howtoo's to get to the current state, i think i need to figure out how the howtoo works first |
| 20:10.14 | doolittlework | TK are you one of the lead developers of asterisk? |
| 20:10.38 | [TK]D-Fender | no |
| 20:10.50 | [TK]D-Fender | DUNDI has nothing to do with it |
| 20:10.53 | doolittlework | the reason for asking is that over the years i have notised you are very active on this channel |
| 20:11.03 | [TK]D-Fender | it simply defines what PEER & TECh to use tto call the other server |
| 20:11.27 | [TK]D-Fender | so the member is no different that as if you just used your peer normally |
| 20:12.13 | doolittlework | how does the queue link to them member, does it rely on the context for that user? |
| 20:12.23 | doolittlework | if the member is on a local machine that is |
| 20:12.43 | [TK]D-Fender | ther is no "link" |
| 20:12.45 | [TK]D-Fender | this is not magic |
| 20:12.48 | [TK]D-Fender | Forget dundi. |
| 20:12.50 | doolittlework | lol |
| 20:12.55 | [TK]D-Fender | you are dialing a an IAX2 peer |
| 20:12.56 | doolittlework | ok |
| 20:12.57 | [TK]D-Fender | just like a trunk |
| 20:13.00 | [TK]D-Fender | NO DIFFERNT |
| 20:13.11 | [TK]D-Fender | IAX2/peer/number |
| 20:13.30 | [TK]D-Fender | queue's don't do dundu lookups. |
| 20:13.34 | [TK]D-Fender | dundi* |
| 20:13.45 | [TK]D-Fender | dundi = dialplan concept |
| 20:14.05 | [TK]D-Fender | And mostly worthless if you know what numebers to expect on the other side |
| 20:14.17 | [TK]D-Fender | like Site A = 1XX, Site B = 2XX |
| 20:14.57 | doolittlework | i see thank you for the nudge in the right direction |
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| 23:04.52 | MrKeuner | drmessano, I don't think it is locked. I bought the device myself. I use some voipvoip reseller. What does it do when blocked number calls? I am not home and cannot try right now. |
| 23:05.45 | drmessano | MrKeuner: Dont know.. never used block in the ATA itself.. always used it with a PBX or the provider implemented it |
| 23:06.18 | MrKeuner | drmessano, all right, I'll check it out. Do you think I can store lots of numbers there? |
| 23:06.20 | drmessano | Chances are, youre provider implements call block of some sort.. you just need to remove the options from your ATA |
| 23:06.37 | drmessano | So I would suggest asking your provider |
| 23:06.57 | MrKeuner | Okay, will do. Thank you for replying |
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