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| 01:02.36 | phix | mornin' |
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| 07:24.25 | Manu18 | Bonjour |
| 07:25.27 | Manu18 | Je viens d'installer freepbx sur une debian 6 et j'ai une page blanche dans la page admin |
| 07:25.38 | Manu18 | est ceque qqun aurait une solution a se probleme? |
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| 07:36.19 | Manu18 | ? |
| 07:42.25 | Manu18 | personne pour m'aider? |
| 07:44.16 | wdoekes | Manu18: tu veux chercher à #freepbx |
| 07:44.40 | wdoekes | *dans |
| 07:55.11 | Manu18 | Personne ne repond wdoekes |
| 07:58.32 | wdoekes | dommage. toutefois, nous ne soutenons pas freepbx ici |
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| 09:26.02 | hindi | hi, Asterisk 1.8 sip.conf... is it deny/permit or contactdeny/contactpermit ? |
| 09:27.56 | wdoekes | hindi: those are two different settings |
| 09:28.14 | wdoekes | deny is for the register/invite itself |
| 09:28.38 | wdoekes | contactdeny is for the registration-destination (possibly also invite-contact) |
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| 09:29.56 | hindi | wdoekes, thanks... but doesn't get the difference |
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| 09:32.24 | hindi | now i've got it :D |
| 09:33.35 | hindi | permit/deny is acl. contactpermit/contactdeny register filter :) |
| 09:34.32 | tictacbum | hi, I am trying to forward incoming calls to a mobile number, I configured an extension for the mobile and added it to Dial, it's working but when I answer a call there's no sound on either side, any clue what can be wrong? |
| 09:35.50 | hindi | firewall, nat? |
| 09:37.06 | hindi | wdoekes, should i use both? Using contactpermit/deny for years ... working like a charm.. so instead there are no unauth calls allowed there is no need for acl? |
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| 09:41.23 | wdoekes | hindi: I'm not entirely sure when which is checked. I'd guess that for nat=yes account types, the contact is always equal to the source-ip, so deny matches the same as contactdeny |
| 09:42.51 | wdoekes | if everything is logical, you should limit source-ip by deny line, and destination-ip by contactdeny. in most cases they are the same |
| 09:43.34 | hindi | ok thanks |
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| 11:19.35 | W0rmDr1nk | hi |
| 11:19.59 | W0rmDr1nk | Maybe not best place to ask - but how does crediting work with DCCA ? |
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| 11:25.25 | W0rmDr1nk | is refund used ? |
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| 12:59.19 | Katty | morning |
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| 13:09.27 | mbowie | Moin |
| 13:09.37 | mbowie | goes for teh coffee... |
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| 13:20.05 | roxlu | hi guys, I'm reading up on SRTP and skimmed over the RFC. I was wondering, if my understanding is correct ... |
| 13:20.38 | roxlu | When using srtp, the agents keep their own cryptographic context that they use to encrypt the payload |
| 13:21.30 | roxlu | and can I assume that the exchange/handshake of the crypto. context is done 'outside' of SRTP? e.g. no info is transferred over srtp regarding this context ? |
| 13:21.43 | file | correct |
| 13:22.03 | file | in the case of SDES the keys are exchanged in the signaling which should be done over a secure transport to prevent eavesdropping |
| 13:22.19 | file | in the case of DTLS the keys are derived from the cryptographic details of the DTLS session itself |
| 13:23.31 | roxlu | file: thanks! |
| 13:24.11 | roxlu | So in webrtc DTLS is only used to exchange these cryptographic details ... (things falling into place now) |
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| 13:45.55 | lmkone | hello all - newbie question - I have got an external sip account configured in my sip.conf but I am unable to make outside calls via that trunk, it seems that asterisk is trying to use my "internal" username and password to auth on invite to external sip provider (Failed to authenticate on INVITE to '"{internal_name}" <sip:{internal_number}@{external_sip_fqdn}>;tag=as4653dd7a') |
| 13:46.40 | lmkone | have been trying different things but still no luck - any ideas? |
| 13:49.00 | [TK]D-Fender | lmkone: fromuser= <------------- |
| 13:49.23 | [TK]D-Fender | Set to your account username |
| 13:50.15 | Katty | hi fenderbender. are you feeling better today? |
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| 13:51.35 | lmkone | fromuser <- that is it |
| 13:51.47 | lmkone | thanks very much [TK]D-Fender |
| 13:53.10 | [TK]D-Fender | You're welcome |
| 13:53.26 | [TK]D-Fender | Katty: Another day, another dollar.... |
| 13:54.33 | lmkone | one more thing - sometimes i am also getting No matching peer for {caller_number} from {external_sip_fqdn} on incoming calls - it does not happen all the time |
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| 13:55.32 | Katty | [TK]D-Fender: *nod* |
| 13:55.50 | [TK]D-Fender | lmkone: We'd have to look at the specifics |
| 13:59.52 | lmkone | from log just before 'No matching...' got Using INVITE request as basis request 6172efa3601cc09b57338ecb0b00a88c@{external_sip_fqdn} where 6172efa3601cc09b57338ecb0b00a88c is not a username |
| 14:00.14 | lmkone | and SIP/2.0 401 Unauthorized right after No matching peer for {caller_number} from {external_sip_fqdn} |
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| 14:02.09 | Katty | waves to sruffell |
| 14:02.21 | sruffell | Hi Katty |
| 14:02.27 | Katty | how're you dear? |
| 14:02.53 | sruffell | Hehâ¦good. You? |
| 14:03.34 | [TK]D-Fender | lmkone: We'd need to see the full SIP debug for the call... |
| 14:03.36 | *** join/#asterisk DruidZ (~darcy@dilbert.druid.net) |
| 14:03.37 | sruffell | I just shudder to imagine the day that I answer that question any differently. So bad *and* I had to come to work. /me shudders |
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| 14:11.51 | *** join/#asterisk whizzi (~whizzi@D4B2620B.static.ziggozakelijk.nl) |
| 14:12.00 | whizzi | hi all :) |
| 14:12.24 | whizzi | I think I may have found a pretty serious bug in Asterisk 11.10.0 |
| 14:12.32 | pabelanger | hello? yes dog here |
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| 14:12.57 | whizzi | I was wondering if somebody could test it with 11.11 before I file it as a bug |
| 14:13.23 | DruidZ | Would it by any chance have anything to do with calls not hanging up? |
| 14:13.32 | whizzi | nope |
| 14:13.47 | whizzi | itâs the ACL-part which is not acting like it is supposed to |
| 14:13.58 | whizzi | itâs a very easy test |
| 14:14.08 | DruidZ | Can I test it without disrupting my server? |
| 14:14.14 | whizzi | yes, you can |
| 14:14.39 | DruidZ | OK, email me at darcy@Vex.Net. |
| 14:14.53 | DruidZ | Or you can SIP call me at the same address. |
| 14:15.02 | whizzi | I can also write it down here ;) |
| 14:15.15 | DruidZ | OK. |
| 14:15.21 | whizzi | find a sip peer without a deny= or allow=, so no ACL |
| 14:15.36 | whizzi | or rather, type=friend |
| 14:15.56 | [TK]D-Fender | [10:15]whizzifind a sip peer without a deny= or allow=, so no ACL <- those 2 things do not correlate |
| 14:16.17 | [TK]D-Fender | whizzi: "allow" has nothing to do with ACL |
| 14:16.18 | whizzi | true, thatâs why I already wrote type=friend |
| 14:16.57 | DruidZ | OK, I have one with type=friend. |
| 14:17.08 | whizzi | no allow or deny? |
| 14:17.18 | DruidZ | It also has "disallow all" followed by "allow ulaw" if that matters. |
| 14:17.25 | whizzi | sip show user $username shows: ACL: No |
| 14:17.44 | whizzi | aar.. |
| 14:17.50 | whizzi | Iâm talking about permit= and deny= |
| 14:18.01 | whizzi | *gets more coffee* |
| 14:18.21 | whizzi | anyway, no permit or deny rules makes ACL set to no |
| 14:18.56 | DruidZ | <PROTECTED> |
| 14:19.13 | DruidZ | Coffe sounds good. brb. |
| 14:19.19 | whizzi | now, set something on that particular friend, deny=0.0.0.0/0.0.0.0 and allow=whatever.you.are.allowing/sub.net |
| 14:19.32 | [TK]D-Fender | 'PERMIT |
| 14:19.41 | whizzi | thank you [TK]D-Fender |
| 14:19.44 | whizzi | youâre right |
| 14:19.57 | whizzi | *coffee not done yet* |
| 14:19.59 | [TK]D-Fender | whizzi: Caffeinate <- |
| 14:21.04 | whizzi | youâd expect that if you do a âsip reloadâ, the ACL would be set and giving Yes in a sip show user $username |
| 14:21.14 | DruidZ | self-caffeinates - "Mmmm" |
| 14:21.28 | whizzi | and thatâs whatâs not working on my Asterisk 11.10.0 box |
| 14:21.49 | whizzi | the only way I can set Acl to yes, is actually unloading chan_sip.so and loading it again, than ACL: Yes |
| 14:21.53 | whizzi | and working as expected |
| 14:21.56 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 14:23.10 | whizzi | a sip reload doesnât actually add the deny and permit to that user, while I think it should since it changes when you unload and reload the module. @DruidZ: I wouldnât unload chan_sip.so on your system though, unless you like to interrupt your SIP channels ;) |
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| 14:25.54 | DruidZ | I get Yes for ACL after a sip reload. |
| 14:26.01 | DruidZ | 11.11.0 |
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| 14:26.46 | whizzi | hmm.. thatâs odd |
| 14:27.00 | DruidZ | And removing it gives me a no after a sip reload. |
| 14:28.11 | whizzi | ok, than I guess something in my configuration is somehow blocking the sip reload but is loading chan_sip perfectly fine |
| 14:28.26 | whizzi | thanks for testing DruidZ |
| 14:29.13 | DruidZ | Or, it was fixed between 11.10 and 11.11. |
| 14:29.50 | whizzi | itâs not in the changelog |
| 14:30.16 | DruidZ | Maybe it was an accidental fix from some other fix. |
| 14:31.29 | whizzi | maybe |
| 14:31.45 | whizzi | Iâll figure that out later :) |
| 14:31.47 | DruidZ | Code changes can sometimes have unexpected and suprising side effects, mostly bad but could randomly be good too. |
| 14:31.55 | whizzi | thanks again for testing |
| 14:32.03 | whizzi | DruidZ: I know, Iâm a programmer myself ;) |
| 14:32.08 | whizzi | not a good one.. |
| 14:32.30 | DruidZ | That comes with time. |
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| 14:37.38 | DruidZ | Anyone seen my message to asterisk-users about calls not hanging up? |
| 14:38.29 | DruidZ | I hesitate to point the bug finger but when something works in 11.10 and not 11.11 one has to wonder. |
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| 14:46.00 | Nugget | are you calling comcast to cancel your cable? :) |
| 14:46.13 | Nugget | those calls apparently never terminate |
| 14:49.36 | DruidZ | Nugget: Funny guy. |
| 14:49.56 | DruidZ | No, these are all calls, even internal to the switch. |
| 14:50.21 | [TK]D-Fender | DruidZ: Or you could ask Verizon for their .02 cents worth.... |
| 14:52.08 | DruidZ | Biggest issue is that I have to restart * from time to time before I ruun out of channels but I can't tell what calls are really active. |
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| 14:53.27 | mbowie | I've not run into that. Our production pair is still on 1.8... our big migration to 11.x hinges on rolling out real-time LDAP at the same time, which has a lot of work tied to it. |
| 14:53.42 | mbowie | (It seems to be the road less traveled / documented) |
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| 14:55.44 | DruidZ | The fact that no one else is mentioning this suggests that I have a configuration issue but my configuration is pretty simple. |
| 14:56.56 | DruidZ | And it didn't happen under 11.10 |
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| 15:56.17 | DruidZ | Hmm. Someone on the mailing list suggests that my dialplan continues after the dial() command if the call is answered but I am pretty sure that is not the case. |
| 15:56.45 | DruidZ | Certainly not in 11.10. |
| 15:58.10 | Qwell | Not while the other end of the call is up, anyways. |
| 15:58.30 | Qwell | Given certain options to Dial(), it *can* continue on after the other end hangs up. |
| 16:00.19 | Qwell | Also, ignore that guy. Given what you posted, there's nothing to suggest that the call is in Voicemail. |
| 16:17.34 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 16:26.46 | DruidZ | I was wondering about that. |
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| 17:15.55 | DruidZ | Just to confirm I removed the voicemail command. No change. |
| 17:16.52 | DruidZ | I also tried removing the hangup() command. I think that that's superflouos isn't it? |
| 17:17.08 | DruidZ | Belt and suspenders. |
| 17:19.42 | [TK]D-Fender | [11:56]DruidZHmm. Someone on the mailing list suggests that my dialplan continues after the dial() command if the call is answered but I am pretty sure that is not the case. <- it does not continue DURING the call. It can continue AFTER the call if you tell it to. |
| 17:24.34 | DruidZ | Yes, option g. It only applies if the destination hangs up. |
| 17:24.55 | DruidZ | Makes sense. Not much point in continuing if the caller hangs up. |
| 17:26.13 | Katty | sruffell: oops, sorry. didn't see your reply (= i'm glad you are doing well! i'm doing great!!! |
| 17:26.16 | DruidZ | Although there is an 'F' option. Doesn't make much sense to me. |
| 17:27.02 | DruidZ | "If caller hangs up send caller elsewhere." |
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| 17:52.49 | DruidZ | Is anyone here running 11.11.0? |
| 17:59.12 | WIMPy | Still on 11.10. What breaks if I upgrade? |
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| 19:29.27 | lirakis | hey guys |
| 19:29.53 | lirakis | im experiencing an issue with dynamic queues. If an agent logs in from a domain that does not have an A record, it is marked as "invalid" |
| 19:30.06 | lirakis | if the domain does have an A record, it works fine |
| 19:30.25 | lirakis | the domain DOES have an SRV record |
| 19:30.51 | lirakis | but my real question is: does asterisk do some "pre-qualification" of agents when they log in, and if so, can i disable it |
| 19:32.09 | lirakis | we route through an outbound proxy any way, and our domains have SRV records that the proxy will resolve via rfc3263 dns mechanisms |
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| 20:00.09 | lirakis | chirp chirp |
| 20:06.07 | mbowie | Crickets? |
| 20:06.16 | *** join/#asterisk niekniek (~niek@77-172-219-210.ip.telfort.nl) |
| 20:07.33 | niekniek | hello ppl, what is TTL called in the world of ISDN or GSM? |
| 20:07.40 | niekniek | i.e. how many hops will be used before a reject |
| 20:08.26 | niekniek | i want to make a call go to a mobile phone. If the phone is off the call should not be redirected to the voicemail system or the system that plays the unavailable message, in stead it should just stop. This system is the next hop (or whatever itâs called), so if the TTL is expired that could happen maybe? |
| 20:15.54 | [TK]D-Fender | niekniek: No such thing |
| 20:16.40 | niekniek | hmmm. not even if you have a connection to the GSM provider with SIP? |
| 20:16.47 | niekniek | a thing like that exists in SIP right? |
| 20:17.47 | [TK]D-Fender | You have a connection to the GSM provider.... over GSM |
| 20:19.23 | niekniek | yeah⦠but you have to get TO the GSM provider with your call⦠some mobile providers offer that your GSM is coupled with a SIP registration... |
| 20:20.09 | niekniek | max-forwards is what i mean |
| 20:21.14 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 20:21.19 | niekniek | [TK]D-Fender: What about this? https://www.dialogic.com/webhelp/BorderNet2020/1.1.0/WebHelp/hopcounter_maxforwards_iw.htm |
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| 20:23.26 | [TK]D-Fender | niekniek: that is all SIP./ |
| 20:23.48 | [TK]D-Fender | The fact they mention some endpoint that is not SIP is besides teh point |
| 20:24.03 | [TK]D-Fender | the actually count only exists within the SIP routing infrastructure |
| 20:24.28 | niekniek | Look at example 1. Hop Counter is coming from SS7 |
| 20:24.54 | niekniek | itâs being converted to max-forward |
| 20:25.18 | niekniek | thatâs not sip... |
| 20:26.14 | [TK]D-Fender | Well * has no mechanism for dealing with this.... |
| 20:32.30 | niekniek | [TK]D-Fender: why do you think that? then the remote systems wonât have loop detection⦠thatâs not very healthy⦠|
| 20:33.29 | lirakis | i dont see what is difficult about your problem niekniek |
| 20:33.42 | [TK]D-Fender | * is a B2BUA anyway.. it is not a "router" |
| 20:33.55 | lirakis | if the GSM gateway returns that the subscirber is off line - 404 the call and hangup |
| 20:34.06 | lirakis | why would it "wait" at all? |
| 20:34.15 | lirakis | its either there or its not |
| 20:34.17 | *** join/#asterisk am1n0 (~devnull@unaffiliated/am1n0) |
| 20:34.19 | am1n0 | hi |
| 20:34.43 | niekniek | lirakis: thatâs my issueâ¦. overhere providers wonât. Theyâll play some message and not early media⦠it just means answer and then you canât behave like that |
| 20:35.15 | lirakis | you mean the cell provider |
| 20:35.19 | niekniek | yeah |
| 20:35.38 | lirakis | so your real question is how do you determine if a cell user is actually off line |
| 20:36.06 | am1n0 | i am trying to use app_flite, so i added it to my modules.conf, but it complains about an undefined symbol snd_pcm_hw_params_sizeof, which i found out to be in chan_alsa.so. i added preload => chan_alsa.so much before app_flite, and added chan_alsa.so=yes in [global], and asterisk does now load chan_alsa as one of the first modules, but app_flite still complains about the missing symbol. |
| 20:36.08 | am1n0 | any ideas? |
| 20:36.11 | lirakis | if you have your own GSM gateway - there is likely some indication GSM side that the subscriber is unavailable |
| 20:36.13 | *** part/#asterisk roxlu (~roxlu@5469B62A.cm-12-2c.dynamic.ziggo.nl) |
| 20:36.18 | niekniek | yeah, if thereâs any other option then what iâm suggesting, please :) |
| 20:36.36 | niekniek | yeah, of course, then iâm in control |
| 20:36.48 | niekniek | but i want to use a regular sim card |
| 20:37.02 | niekniek | the answering telling the user is offline is a next hop |
| 20:37.38 | lirakis | ... i dont understand what you just said |
| 20:37.51 | niekniek | so if i make sure that hop counter is at 0 trying to reach the user I might fool the router that wants to connect my call to the message in believing thereâs a loop |
| 20:38.17 | niekniek | rephrase |
| 20:38.19 | lirakis | checking a max-fwd is not the right answer here |
| 20:38.31 | niekniek | so if i make sure that hop counter is at 0 when the GSM provider trying to reach the user I might fool the router that wants to connect my call to the message in believing thereâs a loop |
| 20:38.49 | niekniek | iâm not checking it⦠iâm changing the value |
| 20:38.59 | niekniek | so that number will travel to itâs destination |
| 20:38.59 | lirakis | i understand |
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| 20:39.30 | lirakis | you are underthe assumption that there are no other b2bua between your SIP provider and the cell network |
| 20:39.43 | lirakis | unless you are buying direct from some one who is doing gsm origination |
| 20:39.47 | niekniek | no; probably a few |
| 20:39.53 | lirakis | so your solution is broken |
| 20:40.07 | lirakis | it goes through a b2bua and max-fwd is totally reset by the b2bua |
| 20:40.29 | niekniek | ah |
| 20:40.40 | niekniek | they reset max-fwd? sobâs |
| 20:40.43 | lirakis | duh |
| 20:40.43 | niekniek | ;) |
| 20:40.45 | lirakis | its a b2bua |
| 20:40.47 | lirakis | they all do |
| 20:40.56 | lirakis | because .... its a b2bua |
| 20:41.27 | niekniek | hmmm⦠never thought about thisâ¦. |
| 20:41.47 | niekniek | so you suggest itâs not just âroutersâ in between but also b2buaâs? |
| 20:41.54 | lirakis | yes |
| 20:42.06 | lirakis | unless you are buying direct from someone who operates the GSM gateway |
| 20:42.07 | niekniek | how are things in ss7 then? |
| 20:42.33 | niekniek | because my call will be very shortly, probably mostly ss7 right? |
| 20:42.39 | lirakis | no |
| 20:42.43 | niekniek | because my call will be sip very shortly, probably mostly ss7 right? |
| 20:42.48 | lirakis | it will probably go voip to who ever has a gsm gateway |
| 20:42.58 | lirakis | why would ss7 be involved at all |
| 20:43.38 | niekniek | i donât know how these providers connect there calls⦠|
| 20:43.58 | niekniek | so if i have a e1, i have no idea what happens on the remote site |
| 20:44.00 | niekniek | side |
| 20:44.23 | lirakis | im assuming you are originating this call SIP |
| 20:44.28 | lirakis | to a sip provider |
| 20:44.36 | niekniek | well, thatâs one option |
| 20:44.45 | niekniek | but also e1 a lo |
| 20:44.46 | niekniek | t |
| 20:44.54 | lirakis | if you are going tdm |
| 20:45.02 | lirakis | ... why are you talking about max fwd |
| 20:45.26 | niekniek | because the max-fwd parameter is converted to itâs tdm equivalent |
| 20:45.41 | niekniek | at least thatâs what i theorize... |
| 20:45.44 | niekniek | not sure :) |
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| 20:47.03 | lirakis | all the same ... that sounds completely broken |
| 20:47.23 | lirakis | you have no idea how many switches are between you and where the conversion happens |
| 20:47.31 | lirakis | and b/c you are operating in a switched environment |
| 20:47.36 | lirakis | that path can change |
| 20:47.43 | lirakis | so i will reiterate |
| 20:47.51 | lirakis | 1. this is a terrible idea |
| 20:48.07 | niekniek | so you think thereâs a big change the number will be reset somewhere right? |
| 20:48.21 | lirakis | 2. it will only work if you are buying direct, and can gaurantee the number of hops for either SIP and or ss7 |
| 20:48.42 | lirakis | niekniek, certainly if you are sending to a sip provider |
| 20:48.50 | lirakis | it will almost certainly be reset immediately |
| 20:49.15 | niekniek | well, i have direct lines e1 to the mobile provider, which only can be used to call the cells in a closed user group |
| 20:49.33 | niekniek | maybe i have a shot there |
| 20:53.22 | niekniek | lirakis: thanks for the discussion |
| 20:53.42 | niekniek | let me just tell you about the possibilities |
| 20:54.08 | niekniek | when i can influence if voicemail/na message kicks in i can use the mobile phone as a callcenter agent |
| 20:54.24 | niekniek | if the user gets calls normally the voicemail can kick in |
| 20:55.08 | niekniek | if the user is called by the callcenter it shouldnât and then the user can be set to offline |
| 21:00.44 | phix | Mornin' |
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| 21:11.48 | kaziklu-bey | afternoon |
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| 21:12.01 | markit | hi, I'm trying to understand how sip works, expecially behind nat. Is it correct that, with asterisk acting as a sip client, the "register" activity is just to inform the server about the ip where can be reached, and does not "open a permanent connenction" with it? |
| 21:12.10 | kaziklu-bey | tm1000: we will miss Landon Donovan |
| 21:12.21 | markit | I mean, you have to port forward port 5060-5061 if you want to be called, right? |
| 21:13.48 | kaziklu-bey | ^Only if your IP isn´t public or behind Firewall |
| 21:13.56 | kaziklu-bey | for more info Try the RFC 3621 |
| 21:14.33 | markit | kaziklu-bey: my asterisk, as I stated, IS behind a nat |
| 21:14.56 | kaziklu-bey | so yes forward the ports |
| 21:14.59 | markit | but I've a friend of mine that says that is not needed, he does not even forward rtp ports |
| 21:15.19 | markit | probably he uses Qualify and his router "keeps the nat association" alive |
| 21:15.24 | kaziklu-bey | uh |
| 21:15.35 | kaziklu-bey | is he talking about a peer or the PBX? |
| 21:15.35 | markit | but is a sort of "trick", and I can't imagin how could work for rtp |
| 21:15.55 | kaziklu-bey | peers usually dont need the forwarding but the pbx/server does |
| 21:16.09 | markit | he is talking about the PBX that registers to a sip provider (trunk), and the asterisk pbx is behind a nat |
| 21:18.25 | kaziklu-bey | well, in my nat case i do the fwd ports to my PBX because if i dont redirect the ports then no audio |
| 21:18.34 | kaziklu-bey | nor registration |
| 21:18.37 | kaziklu-bey | nada |
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| 21:22.30 | markit | mmm registration should work, since is a connection originated inside your local lan to outside |
| 21:22.44 | markit | the problem should be with the "invite" (receiving calls) |
| 21:24.46 | kaziklu-bey | but there are different types of NAT so maybe you send the REGISTER but never receive the Unauthorized, |
| 21:24.48 | kaziklu-bey | http://en.wikipedia.org/wiki/Network_address_translation |
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| 21:44.41 | markit | sleep time bye kaziklu-bey |
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| 22:33.25 | bluefirecorp | I'm looking at trying to get google voice setup with my actual phone and figured asterisk would be the best solution. For the life of me, I really can't get anything to work correctly. |
| 22:33.38 | bluefirecorp | Does anyone have a decent tutorial for what I want to do? |
| 22:39.15 | newtonr | bluefirecorp, I believe Google threw a wrench in that plan a few or several months back. That is, I don't believe google voice has direct xmpp voice support any longer. |
| 22:39.32 | newtonr | bluefirecorp, https://productforums.google.com/forum/#!topic/voice/CdojHZ5_WBU[1-25-false] |
| 22:40.22 | bluefirecorp | So, absolutely no chance? |
| 22:40.29 | newtonr | just google "google voice third party xmpp clients" |
| 22:40.39 | newtonr | bluefirecorp, maybe? but unlikely |
| 22:42.09 | bluefirecorp | Well, now that idea is out of the window. Anyone have any suggestions for a voip provider for a home phone system? |
| 22:47.56 | c|oneman | voip.ms |
| 22:48.05 | c|oneman | or maybe voipo is cheaper if you're US based |
| 22:48.45 | bluefirecorp | Hmm, seems like voipo is worth it if I pay 2 years in advance, wow. |
| 22:49.25 | c|oneman | voip.ms is great to have because it lets you tinker without having to use asterisk |
| 22:49.35 | bluefirecorp | Right. |
| 22:49.40 | c|oneman | you can setup call hunts, queues, etc. without having to write any code |
| 22:53.00 | c|oneman | theres also something like Anveo Direct |
| 22:53.06 | c|oneman | I think they would be very competitive on price |
| 22:53.14 | c|oneman | but you're on your own for setting things up |
| 22:53.27 | c|oneman | free incoming , half cent outgoing probbaly |
| 23:03.14 | bluefirecorp | Meh, it's setup now. |
| 23:03.36 | bluefirecorp | let's test. |
| 23:07.10 | *** join/#asterisk velusuniverse (~velus@unaffiliated/velus) |
| 23:07.31 | velusuniverse | hello im having problemns installing asterisk and freepbx on centos |
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| 23:31.26 | kaziklu-bey | why? try #freepbx |
| 23:50.55 | kleszcz | hmm |
| 23:51.22 | kaziklu-bey | ? |
| 23:57.38 | kleszcz | ;) |