| 00:00.14 | gusto | i do not understnad why ppl here are bitching so often about loosing the registration states of an astdb as if ppl would register for long time, the client is re-registering anyway some time |
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| 00:03.26 | [TK]D-Fender | gusto: And the loss of AstDB as a persistent storage for standard dialplan use, etc |
| 00:04.09 | Penguin | They're taking away astdb? |
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| 00:05.22 | WIMPy | All your AstDB are belong to us! |
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| 00:10.31 | gusto | [TK]D-Fender: yes, i ve read about that, but i am not using that yet |
| 00:10.47 | gusto | Penguin: no, its not your mirrors |
| 00:11.08 | gusto | CeBe: Congstar? |
| 00:11.19 | gusto | WIMPy: what do you mean by that? |
| 00:11.20 | [TK]D-Fender | AstDB is normally a pretty important things. Starting with that off of standard storage is normally a bad idea |
| 00:11.35 | CeBe | gusto: ? |
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| 05:59.20 | snadge | 12.0.0 .. do i want it? :p |
| 06:00.41 | MaliutaLap | Probably better off with 11, if you want LTS |
| 06:00.54 | MaliutaLap | a 12 install will only last you so long |
| 06:21.30 | ChannelZ | It's fun to play with, running it at home. Not sure I'd run a big system on it |
| 06:21.48 | ChannelZ | just because so much has changed |
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| 08:42.12 | vader-- | hello |
| 08:43.08 | vader-- | Have any of you guys seen an issue where show sip peers would show a sip trunk with an unspecified ip and unknown status? I am using Flowroute for SIP and flowroute shows a connection but asterisk isn't. When you make a call you get a channel unavailable. |
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| 08:51.38 | kaldemar | vader--: define an ip address for the peer in the host parameter, instead of "dynamic". |
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| 08:55.48 | vader-- | teh host is set to host=sip.flowroute.com |
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| 08:57.14 | kaldemar | show the peer from sip.conf and sip show peers. mask the secret. |
| 08:58.20 | vader-- | ok one second |
| 08:58.24 | vader-- | here is what is in the cli |
| 08:58.26 | vader-- | [Jan 7 15:58:05] NOTICE[3439]: chan_sip.c:8178 sip_reregister: -- Re-registration for 55151064@sip.flowroute.com |
| 08:58.27 | vader-- | [Jan 7 15:58:06] NOTICE[3439]: chan_sip.c:13779 handle_response_register: Outbound Registration: Expiry for sip.flowroute.com is 120 sec (Scheduling reregistration in 105 s) |
| 08:58.51 | kaldemar | hmm.. you might have srvlookup=no which prevents the name resolution. |
| 08:59.13 | kaldemar | that is unrelated, it is a registration from your asterisk to the provider. |
| 09:00.01 | vader-- | http://pastebin.com/Ki5UDC6u |
| 09:01.41 | DigiDaz | vader--: Do a sip trace on the flowroute IP, you may find you have a NAT issue there |
| 09:03.40 | vader-- | http://pastebin.com/UK8GEbU5 |
| 09:03.49 | vader-- | srvlookup is = yes |
| 09:05.02 | vader-- | http://pastebin.com/UK8GEbU5 is the sip trace |
| 09:06.55 | vader-- | i don't really see anything in it |
| 09:07.06 | vader-- | that sends up a red flag other than the SIP 2.0 / 404 Not Found |
| 09:13.18 | kaldemar | you have a config issue or a dns issue if you don't see an ip address for the peer. |
| 09:14.08 | vader-- | if i drop to the command line |
| 09:14.13 | vader-- | i can ping the host |
| 09:14.15 | vader-- | and get an ip |
| 09:14.21 | vader-- | for sip.flowroute.com |
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| 09:17.11 | kaldemar | vader--: change type to peer. |
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| 10:07.13 | IamTrying | http://www.asteriskguru.com/tools/bandwidth_calculator.php - Where is the OPus codec calculator? is it missing there? |
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| 11:06.06 | vader-- | is there a good dialplan that covers US and internal calling? |
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| 11:19.20 | MaliutaLap | vader--: there used to be a decent patter match for north america in the examples |
| 11:19.33 | MaliutaLap | vader--: it's not a hard one to write though |
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| 12:30.38 | andi | Hi |
| 12:32.27 | andi | I configured sipgates trunking to my asterisk and I just tested if it's possible to add an exten which answers to all incoming calls. which should work with exten => _X,1,Answer(). But when I call the sipgate number I get: Call from 'SIP-ID' to extension 'sipgate phone number' rejected because extension not found in context 'fromInternet'. |
| 12:33.09 | andi | When adding a exten => 'sipgate phone number',1,Answer() the call gets answered. |
| 12:36.58 | andi | Ah, ok, my number matching was wrong. For german numbers _ZZZX. should work. Which now answers my call. |
| 12:38.02 | Gugge | andi: what did you expect _X to match? :) |
| 12:39.22 | Gugge | It only matches a single 0-9 |
| 12:40.21 | andi | Now I know. I thought _X would match every call. It was my mistake. But s should match every call, right? |
| 12:40.28 | kaldemar | andi: no. |
| 12:40.39 | kaldemar | andi: s matches s. |
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| 12:41.17 | kaldemar | andi: https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching https://wiki.asterisk.org/wiki/display/AST/Handling+Special+Extensions |
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| 12:56.30 | Ibrahim22 | Hi, is it safe to use Asterisk 12.0.0 in production, or would you recommend sticking to 11 for the time being? |
| 12:56.58 | Chainsaw | Using a .0.0 of anything in production seems a risky endeavour. |
| 12:58.04 | Ibrahim22 | Yeah, I know, but the new API's for Asterisk 12 are so tempting :) but if there is a serious argument against using 12.0.0 i won't |
| 13:01.40 | Gugge | Ibrahim22: i would use 11 in production, its gonna be supported longer :) |
| 13:02.10 | Gugge | if you need features from 12, that is another thing :) |
| 13:04.52 | Ibrahim22 | I don't really care about the short support time of 12, as I enjoy working in Asterisk and would more than happy upgrade my system when needed |
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| 13:16.06 | michael_work | hello |
| 13:16.44 | michael_work | anyone here using srtp ? |
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| 14:00.55 | Katty | morning |
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| 14:01.43 | skrusty | is there any reason why some of the functionality being baked into ARI, like the new bridge functions, couldn't also be exposed via AMI? I see in AMI 12 doc there are more bridge functions, and the ability to list and kick channels from a bridge, but i don't see any way of creating an initial channel bridge. I assume this simply isn't being opened up via AMI? |
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| 14:06.12 | [TK]D-Fender | skrusty: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerAction_Bridge |
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| 14:10.56 | Katty | file: Ms Zoe is looking dashing this morning in her pink sweater. |
| 14:11.05 | file | she quite likes it |
| 14:12.00 | ipalmer | hi all, I have 2 vlans with asterisk on one vlan along with handset 1 and handest s on the other vlan, I can register the handset fine but when I can't call handset 2 from handset 1 and when I call handset 1 from handset 2 there is no sound. I have set sip.conf up so that both vlans are in as localnet. Am i missing anything |
| 14:12.31 | Katty | leifmadsen: happy birthday!!! |
| 14:12.45 | [TK]D-Fender | ipalmer: Yes. Debug. |
| 14:12.51 | leifmadsen | Katty: shoooosh! |
| 14:13.18 | Katty | pats leifmadsen |
| 14:13.34 | Katty | leifmadsen: it's ok. you don't look a day over 30 dear |
| 14:14.09 | skrusty | [TK]D-Fender: but under AMI, how do you create an initial bridge? |
| 14:14.19 | skrusty | AMI bridge takes two channels |
| 14:14.44 | skrusty | and allows you to bridge them, not create a bridge and add channels to it - or am i reading the spec wrong? :) |
| 14:14.50 | WIMPy | Bridge or ConfBridge? |
| 14:15.37 | skrusty | bridge |
| 14:15.38 | *** part/#asterisk ipalmer (~IceChat77@host81-133-140-79.in-addr.btopenworld.com) |
| 14:15.38 | skrusty | with ref to asterisk 12 |
| 14:15.39 | leifmadsen | Katty: you haven't seen my epic beard |
| 14:15.40 | [TK]D-Fender | [09:05][TK]D-Fenderskrusty: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerAction_Bridge |
| 14:15.55 | skrusty | Channel1 - Channel to Bridge to Channel2. |
| 14:15.55 | leifmadsen | skrusty: correct, in Asterisk 12, you create a bridge, and attach channels to it |
| 14:16.15 | Greenlight | That's the currernt Asterisk 11 version, which just does 2 channels |
| 14:16.15 | skrusty | Channel2 - Channel to Bridge to Channel1. |
| 14:16.16 | leifmadsen | Asterisk 12 and < Asterisk 12 are totally different beasts |
| 14:16.18 | leifmadsen | Asterisk 12 AMI is version 2.0.0 |
| 14:16.18 | skrusty | [TK]D-Fender: that doesn't create a bridge, it bridges two channels |
| 14:16.22 | leifmadsen | correct |
| 14:16.35 | Katty | leifmadsen: i have not. would you say you're at FreeBSD level or Red Hat level yet? |
| 14:16.36 | leifmadsen | think of bridge in Asterisk 12 as like an empty conference room |
| 14:16.38 | skrusty | so my question was, does the bridge functionality of ARI carry into AMI? |
| 14:16.41 | skrusty | yes |
| 14:16.51 | leifmadsen | Katty: just Red Hat, not FreeBSD, but I'm working on it! |
| 14:16.55 | skrusty | i get the general concept |
| 14:16.57 | Katty | nods |
| 14:16.58 | leifmadsen | skrusty: yes |
| 14:17.00 | leifmadsen | I believe so |
| 14:17.03 | leifmadsen | file: confirmation? |
| 14:17.16 | file | ummm what bridge functionality? |
| 14:17.24 | skrusty | leifmadsen: jsut wondering then how you create a bridge in 12 via AMI, as the AMI Bridge_Action takes two channels |
| 14:17.26 | file | you don't have the control you do with ARI |
| 14:17.41 | skrusty | file: yes you do, https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Bridges+REST+API |
| 14:17.44 | file | AMI you can bridge two channels, ARI you have total control |
| 14:17.53 | skrusty | that's the answer i wanted |
| 14:18.07 | skrusty | so the functionality of ARI in 12 doesn't flow over to AMI also |
| 14:18.14 | file | nope, two different things |
| 14:18.27 | skrusty | irritating though really |
| 14:18.34 | skrusty | why should they be different? |
| 14:18.41 | file | because they do fundamentally different things |
| 14:18.59 | skrusty | but ARI is just a another interface |
| 14:19.09 | file | ARI is for writing applications in Asterisk, AMI is for third party call control and general events |
| 14:19.35 | skrusty | ok |
| 14:19.39 | file | ARI = Write a new queueing application, AMI = receive information from app_queue inside of Asterisk and control it |
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| 14:20.00 | skrusty | just trying to see what direction to take AsterNET in for Asterisk 12 onwards |
| 14:20.33 | skrusty | yeah, i understand, i just didn't know if the functionality has bleed |
| 14:20.36 | Greenlight | I would be nice if all the functionality was avilable via the AMI though |
| 14:20.46 | skrusty | i agree |
| 14:20.57 | skrusty | i don't see why you COULDN'T have bridge functions like ARIs in AMI |
| 14:21.13 | file | you could, but I'd rather spend my time working on ARI then trying to shoehorn all of that into AMI |
| 14:21.15 | Greenlight | I guess we'd have to stick with the ConfBrdige method of doing it, which under the covers will be doing the same |
| 14:21.37 | file | and turning AMI into something it was never originally designed to be |
| 14:21.53 | skrusty | :) |
| 14:21.55 | Greenlight | I guess what you're saying then is that if we *need* to do such things, we should be using ARI anyway? |
| 14:21.56 | file | that being said, the fundamentals provide the mechanism to add other external interfaces |
| 14:22.07 | file | Greenlight, if you want to write an Asterisk application then that is what ARI is for |
| 14:22.28 | Greenlight | What if I want to have an application control calls and get events from Asterisk |
| 14:22.39 | file | AMI |
| 14:22.39 | Greenlight | And one of the way it controls calls is bridging them together |
| 14:22.51 | file | WELL, that depends |
| 14:22.55 | Greenlight | Hehe |
| 14:22.56 | skrusty | :) |
| 14:22.59 | file | is that fundamentally an application? |
| 14:23.33 | Greenlight | I wouldn't say so, no |
| 14:23.36 | file | are you trying to construct an application out of a combination of dialplan logic + AMI + AGI? |
| 14:23.40 | Greenlight | It's a feature of the controlling application |
| 14:23.54 | Greenlight | Yes, I guess |
| 14:23.57 | Greenlight | And external resources |
| 14:24.32 | file | then that's what ARI is for |
| 14:24.43 | Greenlight | Like at present, if we want to bridge 3 calls, we redirect them all to ConfBridge in the dialplan. |
| 14:24.54 | Greenlight | Certainly works, but a directbridge seems neater |
| 14:24.59 | file | nods |
| 14:25.08 | skrusty | agreed |
| 14:25.33 | file | there have been pieces here and there to accomplish things, but nothing... pleasant |
| 14:25.39 | Greenlight | Yea :) |
| 14:25.40 | skrusty | i wasn't trying to start a discussion on the topic, more wondering if it had been considered :) |
| 14:25.49 | *** join/#asterisk Draecos (~Draecos@58-7-60-116.dyn.iinet.net.au) |
| 14:25.54 | Greenlight | And would you invisage some applications using *both* ARI and AMI? |
| 14:26.10 | file | nothing to say you can't |
| 14:26.32 | Greenlight | Cool; that works |
| 14:27.26 | file | you wouldn't want to use that to drive your ARI actions though... because since they aren't going over the same connection you aren't guaranteed any sort of synchronization |
| 14:27.38 | WIMPy | would love a dialplan applicatipn that waits for an AMI signal to continue. |
| 14:27.39 | *** join/#asterisk theron (~theron@66.220.145.145) |
| 14:28.37 | Greenlight | Right, so it would make sense to use ARI for everything then |
| 14:28.59 | skrusty | it does seem like ARI is 'almost' a replacement of AGI and AMI |
| 14:29.10 | skrusty | you can subscribe to events you require |
| 14:29.16 | file | AGI and AMI could be used to accomplish some of what ARI can do |
| 14:29.23 | skrusty | and is |
| 14:29.25 | [TK]D-Fender | WIMPy: The is a fairly easy hack to acheive this |
| 14:29.48 | skrusty | i wrote a simple multiserver queue app using AMI/AGI |
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| 14:30.03 | WIMPy | [TK]D-Fender: Probably. I've been doing it with redirects, but it's not ideal. |
| 14:30.12 | [TK]D-Fender | WIMPy: Use a UserEvent to push the channel ID on the "waiting release list along with a dialplan target to redirect to to resume, and then send it in a loop |
| 14:31.01 | [TK]D-Fender | WIMPy: yup, there you go :) |
| 14:31.02 | WIMPy | Would be nice to be able to replace all AGIs in a safe and easy way. |
| 14:31.05 | [TK]D-Fender | No need for AGI for that though.. |
| 14:31.07 | [TK]D-Fender | you can look in dialplan direct while you wait |
| 14:31.38 | WIMPy | No, that's the idea. I have AGIs that set channel variables. It's easy to do that via AMI, but I need to ensure the variables are set before thay are used. |
| 14:31.43 | Greenlight | Hmm the thought of converting everything from AMI to ARI is daunting... |
| 14:32.16 | skrusty | depends what framework your using :) |
| 14:32.16 | Greenlight | My own |
| 14:32.16 | file | is actually working on ARI right now |
| 14:32.37 | skrusty | im just creating the basic classes for AsterNETs ARI implementation at the moment using t4 templates and swagger defninitions |
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| 14:48.58 | *** join/#asterisk AlphaErr0r^ (~error@186.44.152.78) |
| 14:49.15 | AlphaErr0r^ | Good Mornign everyone |
| 14:49.51 | AlphaErr0r^ | i have a question can you have a extension function as a conference room ... (im new to this ) |
| 14:50.08 | AlphaErr0r^ | lets say extension 300 is a conf room ... and all the users can call extension 300 and hear one another |
| 14:50.29 | WIMPy | Extenstions do whatever you tell them to do in your extensions.conf. |
| 14:50.35 | [TK]D-Fender | AlphaErr0r^: "core show application ConfBridge" |
| 14:50.41 | [TK]D-Fender | AlphaErr0r^: "core show application MeetMe" |
| 14:50.51 | [TK]D-Fender | AlphaErr0r^: Indeed, it's your dialplan.. do whatever you want. |
| 14:51.07 | [TK]D-Fender | AlphaErr0r^: Except calling DEVICES "extensions" |
| 14:51.07 | AlphaErr0r^ | thanks |
| 14:51.08 | *** join/#asterisk mmikeym (~mikeym@184.70.65.118) |
| 14:53.27 | AlphaErr0r^ | ok so i just read up the info .. |
| 14:53.57 | AlphaErr0r^ | conf bridge joins the users to a spesific conference |
| 14:54.07 | AlphaErr0r^ | meetme is liek meeting rooms ? |
| 14:54.37 | [TK]D-Fender | AlphaErr0r^: they are the same concept |
| 14:54.54 | [TK]D-Fender | AlphaErr0r^: ConfBridge is newer and not dependent on DAHDI |
| 14:55.05 | AlphaErr0r^ | ah ok ok ... |
| 14:55.09 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:55.09 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:55.29 | [TK]D-Fender | AlphaErr0r^: And is getting all the newer functionality, so in * 10+ You're better off sticking with it |
| 14:58.00 | AlphaErr0r^ | well i'm going to try to get confbridge workign .. not sure if the module is loaded |
| 14:58.15 | AlphaErr0r^ | u guys happen to ahve any tutorials for setting this up ? |
| 14:58.24 | AlphaErr0r^ | i know im being lazy but jsut asking |
| 14:58.27 | [TK]D-Fender | wiki.asterisk.org |
| 14:58.28 | Greenlight | confbrdige.conf explains things quite well |
| 14:58.33 | [TK]D-Fender | The sample configs |
| 14:58.35 | [TK]D-Fender | the BOOK |
| 14:58.37 | [TK]D-Fender | ~book |
| 14:58.37 | 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 |
| 15:00.29 | AlphaErr0r^ | cool bbook is downloading ! |
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| 15:09.59 | file | moo |
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| 15:11.58 | Katty | wonders if Ms Zoe Doggie would like to help her knit a blanket this morning |
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| 15:22.58 | fireman_biff | I'm seeing "Span 1: Channel 0/1 got hangup, cause -1" on the CLI around the time a user says a call dropped. Any idea what the -1 is about? It doesn't seem to be a valid cause code. |
| 15:24.03 | AlphaErr0r^ | thanks everyoen i'm working ! |
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| 16:54.59 | Katty | is there a way to hangup an extension from the CLI? |
| 16:55.51 | Katty | like softhangup(SIP/foo) |
| 16:56.22 | [TK]D-Fender | channel request hangup .... |
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| 17:03.15 | Katty | [TK]D-Fender: ty dear |
| 17:03.56 | *** join/#asterisk timholum (~chatzilla@68-117-120-140.static.eucl.wi.charter.com) |
| 17:05.01 | timholum | Hello my asterisk server just crashed, The last line in the log prior to it dieing was "format_wav.c: Bad write (192): No space left on device" , But when I do df -h The fullest drive is less then 40% full |
| 17:05.29 | timholum | Any idea's as to what could cause this? |
| 17:09.37 | michael_work | anyone using srtp/tls on asterisk would like to have input on bug that for me makes 100% crash on several version of astewrisk(including last one) |
| 17:09.52 | michael_work | https://issues.asterisk.org/jira/browse/ASTERISK-22920#comment-213702 |
| 17:12.26 | [TK]D-Fender | timholum: mount <- |
| 17:13.17 | timholum | http://pastebin.com/L9fsrfPD is the output of mount |
| 17:13.31 | timholum | http://pastebin.com/M5DAhKWU is the 10 lines of the log arround the error |
| 17:13.42 | timholum | http://pastebin.com/7CRKrSLk is df -h |
| 17:14.06 | timholum | http://pastebin.com/4ncKRX90 is df -i |
| 17:16.14 | Greenlight | No space left on device seems quite conclusive |
| 17:16.48 | timholum | Greenlight: I agree, but which device? everything is less then 40% full |
| 17:16.48 | Greenlight | I would assume this is getting passed to Asterisk from the OS |
| 17:16.56 | Greenlight | *when* you did the df -h, yes |
| 17:17.15 | Greenlight | How do you know it wasn't writing 1GB file to ramdisk, and then cleared it after it errored out? |
| 17:17.17 | timholum | All that was after the fact |
| 17:17.30 | timholum | I know the only app that is writting to the ramdrive |
| 17:17.38 | *** join/#asterisk Kledio (~Kledio@38.112.241.194) |
| 17:17.50 | Kledio | Hey |
| 17:17.54 | Greenlight | What were you doing in asterisk at the time? |
| 17:18.18 | timholum | The server was taking calls like normal |
| 17:18.48 | Kledio | Can anyone help on an issue with asterisk server giving busy tone for extensions? |
| 17:18.53 | timholum | I will check the cdr log to see if there was an unusually long call that ended at that time, but I doubt that a call recording would be 30 or 40 gb |
| 17:20.45 | Greenlight | Pastebin the log leading up to the issue, and also, what version are you on ? |
| 17:21.05 | timholum | 1.8.11 cert 10 |
| 17:21.28 | timholum | The cdr log does not show any extreamly long calls |
| 17:21.56 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 17:22.04 | [TK]D-Fender | Kledio: pastebin the call. |
| 17:22.04 | [TK]D-Fender | ~pb |
| 17:22.05 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 17:22.07 | [TK]D-Fender | ^^^ |
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| 17:23.45 | Kledio | one sec im generating the log right now |
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| 17:28.30 | Kledio | http://pastebin.com/0bDXpymj |
| 17:28.51 | Kledio | Sorry it took so long the server was hanging |
| 17:29.08 | timholum | http://pastebin.com/fvg32hDz is the log |
| 17:29.14 | timholum | Greenlight |
| 17:31.38 | [TK]D-Fender | Kledio: I do not see a CALL in there |
| 17:31.49 | [TK]D-Fender | Kledio: And I do see dialplan error warnings... |
| 17:32.23 | Greenlight | looks like a disk issue to me... even that log looks corrupt: Check like 262: [2014-01-07 09:58:54] WARNING[15763] file.c: Natural write f[Jan 7 10:27:19] VERBOSE[14434] config.c: == |
| 17:32.35 | Greenlight | It gets cut off half way through |
| 17:33.05 | Greenlight | Maybe you've a faulty disk, or faulty controller, and the error is getting translated to out of space. |
| 17:33.17 | Greenlight | Not really my area of expertise though, perhaps someone else has an idea.. |
| 17:34.21 | [TK]D-Fender | Kledio: "sip set debug on" <-------- |
| 17:34.44 | Kledio | do i run that in the cli |
| 17:35.42 | [TK]D-Fender | yes |
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| 17:36.17 | Kledio | alright then what |
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| 17:40.51 | Kledio | the log doesnt have any new information |
| 17:41.17 | [TK]D-Fender | Asterisk CLI. |
| 17:41.21 | [TK]D-Fender | Not "logs" |
| 17:41.38 | [TK]D-Fender | Log files are worthless. |
| 17:41.47 | [TK]D-Fender | If you still see nothing.... then the packets aren't getting through at all |
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| 17:42.48 | Kledio | where would the packets be dumped? |
| 17:43.28 | [TK]D-Fender | ASTERISK CLI |
| 17:44.25 | Kledio | well nothing shows up when i dial an extension again |
| 17:44.25 | [TK]D-Fender | If you see nothing there with ISP debug enabled... then packets either aren't reaching your server at all or are lost at the firewall on it |
| 17:44.27 | [TK]D-Fender | SIP* |
| 17:44.27 | Kledio | its just a busy tone |
| 17:44.29 | [TK]D-Fender | you have a networking issue |
| 17:44.47 | *** join/#asterisk SunTsu_ (miyamoto@unaffiliated/suntsu) |
| 17:44.58 | Kledio | the calls work if i call the main line and then get transfered to an extension |
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| 17:49.23 | [TK]D-Fender | Kledio: You have not given us any details whatsoever for this scenario, or any other |
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| 17:53.23 | timholum | Looks like somehow the disk was actually full at that time, apache and mysql also reported disk full errors |
| 17:53.29 | Kledio | Alright so the company phones in our NYC office work fine by dialing extensions, but the company phones at denever are experiencing issue with dialing an extension |
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| 17:58.03 | [TK]D-Fender | [12:52]KledioAlright so the company phones in our NYC office work fine by dialing extensions, but the company phones at denever are experiencing issue with dialing an extension <- this is not "details. |
| 17:58.08 | Greenlight | timholum: Indeed |
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| 18:02.23 | *** join/#asterisk dumby (~dumby@204.246.140.162) |
| 18:03.53 | Kledio | what details do you need? |
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| 18:08.53 | Kledio | [Jan 7 13:08:33] WARNING[15573] app_dial.c: Unable to create channel of type 'IAX2' (cause 20 - Unknown) |
| 18:09.06 | Kledio | thats what i get in the log when i get a busy tone |
| 18:09.22 | Greenlight | heh no wonder you're not seeing anything in the SIP logs... |
| 18:10.59 | *** join/#asterisk Hyper_Eye (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 18:10.59 | [TK]D-Fender | Kledio: You've shown us no calls. I've told you you have a NETWORKING issue... and you described NONE OF IT |
| 18:11.19 | [TK]D-Fender | Kledio: "new york" and "denver" mean nothing to us. |
| 18:11.26 | Greenlight | Looks like he's using IAX not SIP though |
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| 18:12.03 | [TK]D-Fender | Greenlight: http://pastebin.com/0bDXpymj <- and if you read you'd see almost all of that is SIP |
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| 18:12.11 | Greenlight | Ahh sorry hadn't been reading the pb's just noticed talk of no packets over SIP and then mention of IAX... |
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| 18:15.16 | Kledio | this is the log for the other asterisk http://pastebin.com/Wc7gxAYM |
| 18:15.39 | Kledio | the one that works correctly with extensions |
| 18:16.57 | *** join/#asterisk mcf3782 (~Adium@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 18:19.17 | mcf3782 | Chainsaw: You around? |
| 18:24.05 | [TK]D-Fender | Kledio: Showing us another asterisk that is working is also of no help at all. |
| 18:26.26 | [TK]D-Fender | Kledio: and I don't see anything "working" in there at all. I see a TONG of peer dial failures, DAHDI & dialplan configuration errors. These systems are a mess |
| 18:31.01 | Chainsaw | mcf3782: Yes. |
| 18:31.32 | *** join/#asterisk caveat- (hoax@2a01:4f8:191:9111:30::10) |
| 18:32.52 | Chainsaw | mcf3782: Or at least, I was. Now I'm off to a LUG meeting. |
| 18:33.04 | mcf3782 | drats |
| 18:33.56 | mcf3782 | I'll catch up with him later. |
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| 18:37.02 | AlphaErr0r^ | hey guys in the process of installign the conf bridge after readon g the steps in the book |
| 18:37.05 | AlphaErr0r^ | i'm having a problem |
| 18:37.27 | AlphaErr0r^ | == Using SIP RTP CoS mark 5 |
| 18:37.27 | AlphaErr0r^ | <PROTECTED> |
| 18:37.27 | AlphaErr0r^ | <PROTECTED> |
| 18:37.37 | AlphaErr0r^ | 300 is my conf ext |
| 18:37.50 | AlphaErr0r^ | 213 is my extn but its not answering .. |
| 18:38.15 | AlphaErr0r^ | - <SIP/213-00000000> Playing 'conf-onlyperson.ulaw' (language 'en') |
| 18:38.20 | AlphaErr0r^ | dont get a sound at all |
| 18:38.42 | [TK]D-Fender | perhaps your DEVICE is misconfigured |
| 18:38.54 | [TK]D-Fender | next... PASTEBIN... do NOT flood in here. |
| 18:38.55 | [TK]D-Fender | ~pb |
| 18:38.56 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 18:38.58 | [TK]D-Fender | ^^^ |
| 18:39.00 | AlphaErr0r^ | sorry |
| 18:39.12 | AlphaErr0r^ | misconfigured FOR SURE haha |
| 18:39.59 | AlphaErr0r^ | any suggestions ? |
| 18:40.11 | *** join/#asterisk yahh (~mitesh@59.164.82.26) |
| 18:40.34 | [TK]D-Fender | provide actual useful information. |
| 18:40.46 | [TK]D-Fender | like what your device is. How it is networked relative to your server, etc... |
| 18:41.41 | AlphaErr0r^ | ok i have 2 clients via SIP accessing the server. |
| 18:41.51 | AlphaErr0r^ | the server is on my network here 192.168.2.x |
| 18:42.03 | AlphaErr0r^ | both clients are register and can call eachother |
| 18:42.20 | AlphaErr0r^ | im running cent os 6.5 |
| 18:42.25 | [TK]D-Fender | Still no client netowrking details... |
| 18:43.12 | AlphaErr0r^ | Asterisk 1.8.25.0 |
| 18:43.45 | AlphaErr0r^ | the clients are connected via wirless and are on the same subnet |
| 18:43.55 | AlphaErr0r^ | the clients are runnign linphone on a ipad and a blackberry |
| 18:44.45 | AlphaErr0r^ | they are using udp |
| 18:44.56 | AlphaErr0r^ | i dont know what other information u may need? |
| 18:45.43 | [TK]D-Fender | verify your firewall on your server |
| 18:45.53 | AlphaErr0r^ | iptables are turned off |
| 18:46.02 | AlphaErr0r^ | sip clients can call one another |
| 18:46.12 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 18:46.13 | AlphaErr0r^ | i can access my trunks incoming and outgoign as well |
| 18:48.13 | AlphaErr0r^ | im thinkign it could be my dial plan ... |
| 18:48.21 | [TK]D-Fender | AlphaErr0r^: First... do the sane thing and add an ANSWER before you dump them in conf |
| 18:48.42 | *** join/#asterisk jetlag (~jetlag@pool-71-168-196-225.cmdnnj.east.verizon.net) |
| 18:48.44 | [TK]D-Fender | AlphaErr0r^: If that doesn't do it, then enable SIP DEBUG and pastebin the complete attempt from beginning to end |
| 18:48.48 | *** join/#asterisk vinhdizzo (~vinh@cpe-98-154-220-20.socal.res.rr.com) |
| 18:48.50 | [TK]D-Fender | "sip set debug on" <- |
| 18:53.31 | *** join/#asterisk AlphaErr0r^ (AlphaErr0r@0-1f-fb-c-50-dd.dynamic.wimax.tstt.net.tt) |
| 18:53.39 | AlphaErr0r^ | sorry gor bumped |
| 18:53.47 | AlphaErr0r^ | i may have missed soem of your answers |
| 18:54.00 | [TK]D-Fender | [13:48][TK]D-FenderAlphaErr0r^: First... do the sane thing and add an ANSWER before you dump them in conf |
| 18:54.02 | [TK]D-Fender | [13:48][TK]D-FenderAlphaErr0r^: If that doesn't do it, then enable SIP DEBUG and pastebin the complete attempt from beginning to end |
| 18:54.04 | [TK]D-Fender | [13:48][TK]D-Fender"sip set debug on" <- |
| 18:54.09 | AlphaErr0r^ | ok on it now ! |
| 18:55.03 | *** join/#asterisk Kyosh (~whoa@pool-74-108-131-147.nycmny.fios.verizon.net) |
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| 19:10.23 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 19:12.06 | *** join/#asterisk ThoMe (~tm@tm.muc.de) |
| 19:12.06 | ThoMe | hiho |
| 19:12.39 | ThoMe | the pattern for ONE or TWO digits is _XX or ? |
| 19:13.27 | Katty | looks in |
| 19:13.32 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 19:14.02 | ThoMe | kalz: hm. only "9" doesnt works |
| 19:14.30 | [TK]D-Fender | ThoMe: There is no "or" |
| 19:14.46 | ThoMe | [TK]D-Fender: and what is OR? _X. ? |
| 19:16.00 | [TK]D-Fender | ThoMe: THERE IS NONE |
| 19:18.23 | ThoMe | [TK]D-Fender: pls, can yu help me? :-( |
| 19:18.39 | ThoMe | _X. is one or more or? |
| 19:18.53 | [TK]D-Fender | ThoMe: What part of "THERE IS NO PATTERN TO SAY EITHER JUST 1 OR 2 DIGITS" ARE YOU HAVING TROUBLE WITH? |
| 19:18.57 | [TK]D-Fender | ThoMe: IT DOE NOT exist. |
| 19:19.00 | [TK]D-Fender | ThoMe: AT ALL |
| 19:19.27 | [TK]D-Fender | ThoMe: This requires to you to make TWO different extens. |
| 19:19.33 | yahh | Hello. I am facing lot of voice cut after installing wctw235b |
| 19:19.37 | [TK]D-Fender | ThoMe: one to handle SINGLE digits. |
| 19:19.41 | [TK]D-Fender | ThoMe: the other for TWO digits. |
| 19:20.07 | [TK]D-Fender | ThoMe: There is no pattern to handle BOTH of these exclusively. |
| 19:20.08 | yahh | I can see error (http://pastebin.com/yJEgnnq7) on console |
| 19:20.37 | Katty | pokes fenderbender. |
| 19:21.04 | yahh | i m using libpri-1.4.14, dahdi-linux-complete-2.8.0.1+2.8.0, asterisk-11.7.0 |
| 19:21.30 | yahh | can someone help in this |
| 19:21.37 | _Corey_ | [TK]D-Fender: Actually, I believe _X! will do the trick |
| 19:21.52 | _Corey_ | !=0 or more digits |
| 19:21.58 | [TK]D-Fender | _Corey_: No, that is not appropriate. |
| 19:22.27 | _Corey_ | is looking to limit it to two digits? |
| 19:22.33 | [TK]D-Fender | _Corey_: depending what it is mixed with it is likely to break. |
| 19:22.51 | [TK]D-Fender | wonders why everyone has reading problems today.... |
| 19:23.20 | _Corey_ | lol |
| 19:23.25 | [TK]D-Fender | _Corey_: if I put that in a context with an IVR runing off "s", I'll NEVER get to enter a 2 digit exten |
| 19:23.40 | Katty | [TK]D-Fender: for the same reason you have typing problems some days. |
| 19:23.42 | [TK]D-Fender | _Corey_: and that doesn't prevent LONGER than 2 in other cases |
| 19:23.55 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 19:23.55 | _Corey_ | X shouldn't match s |
| 19:23.56 | Katty | [TK]D-Fender: but i still love you. |
| 19:24.10 | _Corey_ | Agreed on the "won't prevent longer", definitely |
| 19:24.19 | [TK]D-Fender | _Corey_: I know... I specified running it off "s" so that your pattern is the only NUMERIC one in there so you could see why it'd fail to allow you to dial a TWO digit number |
| 19:24.37 | [TK]D-Fender | ! is not just "0 or more" |
| 19:25.01 | [TK]D-Fender | it is "as soon as this match is unique:" |
| 19:25.38 | [TK]D-Fender | This handles pattern overlap |
| 19:25.59 | *** join/#asterisk GameGamer43 (sid5533@gateway/web/irccloud.com/x-erndanscwykfypqo) |
| 19:26.09 | _Corey_ | It's generally not good practice to use !, though occasionally it can work... I didn't read the whole thread here, so feel free to ignore the suggestion... |
| 19:26.59 | [TK]D-Fender | _Corey_: doesn't work for you what you figured it would though.... |
| 19:27.40 | [TK]D-Fender | _Corey_: As sole numeric pattern you'll never hit 2 digits, and the only reason to survive to reachit is having other extens in there.... and then again.. it's those ones that will cath it |
| 19:28.03 | [TK]D-Fender | "!" has it's place... just not here. |
| 19:28.32 | *** join/#asterisk davlefouAMD (~david@197.15.224.185) |
| 19:31.40 | *** join/#asterisk dwayne (~dwayne@c-71-207-208-112.hsd1.al.comcast.net) |
| 19:33.22 | ThoMe | hm. |
| 19:33.31 | ThoMe | I recieve an call and asterisk say atched device setup to use SRTP, but request was not! |
| 19:33.49 | ThoMe | what can I do? i would like use srtp, when posible |
| 19:33.55 | ThoMe | outging calls works fine |
| 19:48.56 | Katty | pesters dwayne |
| 19:58.00 | dwayne | tickles Katty |
| 19:58.17 | yahh | I have added some more information - service dahdi status and system.conf including error at http://pastebin.com/a0Zg881P |
| 19:58.28 | yahh | kindly have a look |
| 20:15.12 | *** join/#asterisk jsjc (~Adium@153.Red-88-14-68.dynamicIP.rima-tde.net) |
| 20:21.57 | *** join/#asterisk mic__ (~mic@90.184.7.102) |
| 20:25.51 | yahh | can someone help in - http://lists.digium.com/pipermail/asterisk-users/2014-January/281696.html |
| 20:32.23 | *** join/#asterisk wrxed (~rrittgarn@75-146-129-225-SouthBend.hfc.comcastbusiness.net) |
| 20:33.18 | wrxed | Looking to do a Dial with two Local channels, but if one of the resulting channels declines, I don't want to stop ringing the other channel. I didn't see any of the options that seemed to apply to my situation, unless I'm missing something (which I probably am). Anyone able to point me in the right direction? |
| 20:34.00 | wrxed | (options as far as Dial() options) |
| 20:35.32 | [TK]D-Fender | wrxed: instead of cascading back... just kill time... |
| 20:36.58 | wrxed | TK: i don't exactly follow. just have it sit at a wait on a dialresult of busy? |
| 20:37.10 | *** join/#asterisk afidegnum (~chatzilla@41.207.162.254) |
| 20:37.53 | afidegnum | hi, anyone online ? |
| 20:37.57 | Qwell | no |
| 20:38.24 | [TK]D-Fender | wrkcorrect |
| 20:38.39 | [TK]D-Fender | wrxed: correct. Run the clock. |
| 20:38.45 | wrxed | makes sense. Thanks |
| 20:39.16 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 20:42.34 | afidegnum | I am building an online application which is going to be like a PBX system but needs to be installed into Iphone and android phones. |
| 20:42.34 | afidegnum | is it possible? |
| 20:43.34 | *** join/#asterisk felipealmeida (~user@179.210.237.70) |
| 20:43.39 | [TK]D-Fender | afidegnum: You're the programmer.. how do you not know the answer already? |
| 20:44.08 | [TK]D-Fender | an "online application" is vage.... |
| 20:44.57 | [TK]D-Fender | At what point it becomes "like a PBX" system is unclear, including what part is "client side" and what part is "server side" (as alluded to by your use of the term "online" |
| 20:45.27 | afidegnum | ok, going to do more reading, I will be back |
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| 21:02.19 | *** join/#asterisk mlg9000 (~mlg9000@174.84.212.34) |
| 21:03.36 | mlg9000 | Hi all. With Google voice shutting down xmpp in May, are there any free US calling alternatives available to switch to? |
| 21:03.48 | mlg9000 | Incoming/outgoing |
| 21:04.09 | mlg9000 | or maybe even just one or the other |
| 21:05.04 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 21:06.12 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 21:16.44 | *** join/#asterisk alsuren (~dlaban@host86-147-224-61.range86-147.btcentralplus.com) |
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| 21:27.43 | [TK]D-Fender | checkout time, BBIAB |
| 21:38.01 | Katty | shivers |
| 21:38.23 | Katty | file: take your weather back :< |
| 21:41.17 | *** part/#asterisk mjordan (~matt@nat/digium/x-jtwdpusbvqwrdpbq) |
| 21:49.33 | drmessano | mlg9000, nothing reliable |
| 21:50.53 | *** part/#asterisk mlg9000 (~mlg9000@174.84.212.34) |
| 21:51.54 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-renwigbrmzklaayb) |
| 21:51.54 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 21:54.01 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 22:13.04 | *** join/#asterisk Worsedah (~worsedah@69-92-77-41.cpe.cableone.net) |
| 22:13.14 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 22:13.38 | Worsedah | Hi, I'm having a strange, probably NAT related, problem trying to help out someone with that I'm stuck on |
| 22:13.54 | WIMPy | ~ask |
| 22:13.54 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 22:14.56 | *** join/#asterisk davlefouAMD (~david@197.15.216.102) |
| 22:15.47 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:16.24 | Worsedah | They are renting a pbx and trying to connect 3 phones from their office. The office has a single public IP. Each phone is a Cisco 7940 and registers fine but SIP show peers show them all having the same IP and port 5061. Incoming calls go straight to the extensions voicemail, I assume because the router at their office gets confused as to which phone it should actually go to. Is there a configuration change I can make asterisk side or |
| 22:16.24 | Worsedah | <PROTECTED> |
| 22:16.41 | dwayne | Katty, file is way too generous with his weather :-( |
| 22:17.36 | Katty | nods sadly |
| 22:17.45 | Katty | and the australians are being stingy with theirs! |
| 22:17.57 | Katty | stingey? |
| 22:17.58 | dwayne | yes! with their kangaroos too |
| 22:17.59 | Katty | something like that. |
| 22:18.05 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 22:18.25 | drmessano | hmmm |
| 22:18.49 | Katty | hi danny. |
| 22:19.06 | drmessano | Worsedah, sounds like the firewall needs to be swapped out |
| 22:19.16 | drmessano | Worsedah, shatty NAT |
| 22:19.21 | drmessano | Hi Katty |
| 22:19.29 | Katty | how's your day going dear? |
| 22:19.36 | drmessano | It's COLD |
| 22:19.41 | drmessano | Been out in it too much |
| 22:19.44 | Katty | yeah :< |
| 22:19.55 | Katty | you haven't gotten frostbite have you? |
| 22:20.24 | *** join/#asterisk reconwireless (uid10170@gateway/web/irccloud.com/x-cfzpcrxumroxxabk) |
| 22:20.31 | drmessano | I have a UPS going tits up, made a "quick trip" to the country to check on it. Froze fingers to the bone just locking and unlocking the gates |
| 22:20.42 | Katty | poor thing :< |
| 22:21.06 | Worsedah | very shitty nat, linksys EA4500 |
| 22:21.35 | drmessano | Well, I am kinda an idiot. I was trying to do it "really fast" and after fumbling with it for a good 60 seconds or so, I put on my gloves |
| 22:21.51 | drmessano | Then did the same on the way out |
| 22:21.52 | drmessano | "I dont need my glov... yeah" |
| 22:22.01 | Katty | grins |
| 22:22.03 | Katty | you goober! |
| 22:23.18 | drmessano | No kidding |
| 22:23.34 | drmessano | Book smart, no CUMIN CENTS |
| 22:24.01 | drmessano | It's the Story of My Life |
| 22:24.09 | drmessano | and I hope that One Direction song is stuck in your head now |
| 22:24.33 | Katty | nope. the game of love by daft punk ;) |
| 22:24.36 | Katty | nice try tho. |
| 22:24.42 | drmessano | lol |
| 22:24.45 | *** join/#asterisk sectech (~sectech@stjhnbsu0ww-142134150058.dhcp-dynamic.FibreOP.nb.bellaliant.net) |
| 22:25.04 | drmessano | http://fixitwizkid.com/attachments/xaeam-jpg.3217/ |
| 22:25.21 | Katty | awww. |
| 22:25.35 | Katty | mick jagger foxy! |
| 22:26.00 | drmessano | Theres a demotivational version too |
| 22:26.06 | drmessano | methinks its better |
| 22:27.40 | *** join/#asterisk APT1 (~APT1@unaffiliated/apt1) |
| 22:27.49 | Worsedah | So no option to force the Ciscos to all be on a different port then 5061 that may make them work with the NAT being how it is? |
| 22:40.44 | drmessano | Worsedah, not going to work. |
| 22:41.18 | drmessano | y |
| 22:41.20 | sectech | Anyone hear of an issue with iax2 where it only negotiates low bandwidth codecs in Asterisk 11.6? |
| 22:41.33 | sectech | Upgraded to 11.7, and it works fine... |
| 22:41.40 | drmessano | 11.6 was crap |
| 22:41.50 | drmessano | I had several issues with it, and skipped it |
| 22:42.17 | sectech | agh ok... |
| 22:42.18 | Worsedah | drmessano, why not, what component needs changed/replaced? |
| 22:42.27 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 22:42.35 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 22:42.42 | drmessano | [17:19:06] <drmessano> Worsedah, sounds like the firewall needs to be swapped out |
| 22:42.42 | drmessano | [17:19:15] <drmessano> Worsedah, shatty NAT |
| 22:43.53 | drmessano | Some routers just don't work. In this case, we're not talking about a $500, $1000, or $10000 router, so there's no "There has to be some way" |
| 22:44.23 | drmessano | Upgrade the firmware, use DD-WRT, swap it completely |
| 22:44.25 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:a42:224:1dff:fecd:234c) |
| 22:47.24 | *** join/#asterisk kuruption (kuruption@vato.is.a.big.black.cock.addikt.org) |
| 22:50.34 | Worsedah | drmessano, is there a router/nat that is recommended and known to work? this person doesn't have the technical ability really to do much other than buy one known to work |
| 22:50.55 | Worsedah | hence them using a hosted solution |
| 22:52.16 | drmessano | Worsedah, there's thousands that work and very few that do not. Bad luck. Another consumer router off the shelf in all likelihood will work |
| 22:53.26 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 22:54.08 | drmessano | Worsedah, does that router have SIP ALG? |
| 22:59.07 | drmessano | Katty, I tagged you in a post |
| 22:59.12 | drmessano | It's quite cute |
| 22:59.59 | Worsedah | drmessano, I was just checking that, it appears to have it yes |
| 23:03.17 | drmessano | Could try disabling it. It doesn't sound like a typical ALG issue, but it could just be very broken |
| 23:03.55 | drmessano | WHAT DOES THE FAX SAY? |
| 23:05.51 | drmessano | ~whatdoesthefaxsay |
| 23:05.51 | infobot | SCREEEE-AWWWW WAAAHH-PHFLIBAP |
| 23:27.39 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 23:29.50 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 23:33.42 | DigiDaz | Worsedah: I'm not a big asterisk user, can you force rport on asterisk? I know the 7940 doesn't have that support in the phone |
| 23:35.31 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.178) |
| 23:36.18 | Worsedah | drmessano, it seems to be working after changing the ALG SIP setting, the peers all got different ports in the sip show peers list and seem to be all ringing through now. Time will tell if it is consistent though |
| 23:37.19 | drmessano | Worsedah, it could be that simple. Kinda scary though |
| 23:39.49 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 23:40.57 | *** join/#asterisk saint_ (~saint@c-50-166-85-78.hsd1.nj.comcast.net) |
| 23:40.57 | saint_ | whois *!*@78.46.76.141 |
| 23:41.15 | saint_ | hi all - |
| 23:41.30 | saint_ | is there a document that explains how to upgrade asterisk while keeping the same options in menuselect ? |
| 23:41.59 | saint_ | i remember seeing a document where it was said to do the make config (I think) then copy the results into the new source folder, and launch the make ... or something like that ..? |
| 23:42.11 | WIMPy | copy menselect.makeopts |
| 23:42.17 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.178) |
| 23:42.24 | saint_ | that's all it is to copy ? |
| 23:42.35 | WIMPy | yes |
| 23:43.26 | saint_ | WIMPy menuselect-tree ? |
| 23:43.31 | WIMPy | The result might not be the same however if some changed dependencies aren't met. |
| 23:43.40 | WIMPy | what? |
| 23:44.03 | saint_ | WIMPy i just don t want to go through all the options |
| 23:44.10 | Worsedah | drmessano, year, I have no experience with multiple phones connecting to a server from behind the same nat, so this one is out of my confort zone |
| 23:45.36 | drmessano | Worsedah, this is as simple as multiple browsers connecting from behind a NAT. Some people have tried to simplify the SIP configuration and have only broken it horribly |
| 23:47.12 | Worsedah | It makes sense to my how it is supposed to work, each one registers on 5060 and specifies/get an rport to listen on after correct? It seems like these Ciscos have a hard time adhering to anything but what they want though |
| 23:49.11 | *** join/#asterisk darkbasic_ (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 23:50.36 | DigiDaz | Worsedah: If you have any grief, there is a voip control port setting in the 7940s, this will allow you to set individual ports for each. It will really depend upon the nat you have. My 790 registers with 5061 to an external asterisk box but that is the port it is set at. |
| 23:51.33 | DigiDaz | 7960 even |
| 23:52.54 | *** join/#asterisk vinhdizzo (~vinh@cpe-98-154-220-20.socal.res.rr.com) |
| 23:57.03 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 23:58.57 | WIMPy | Y |
| 23:59.59 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |