| 00:15.18 | bitvilag | i should install fail2ban too then |
| 00:15.28 | bitvilag | I must go now |
| 00:15.33 | bitvilag | thanks for the info |
| 00:15.44 | bitvilag | everytime i come up you always have an answer for me:D |
| 00:15.45 | bitvilag | thanks |
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| 02:55.44 | xaristax | Hi for me is impossible to get CURL working on Asterisk12 have someone know an alternative to this im just trying to send and HTTP GET request to a page i dont need to get info of it |
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| 04:34.18 | [TK]D-Fender | xaristax: So just call curl from CLI |
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| 07:36.56 | Manu18 | J'ai un serveur dedié chez Online.net ou j'ai installé mon serveur asterisk. Depuis chez moi avec ma Freebox tout fonctionne parfaitement. |
| 07:37.11 | Manu18 | Par contre de mon magasin derrière ma Livebox pro rien ne fonctionne. Le telephonne ne s'enregistre même pas. Cela peut venir de la livebox ou d'un soucis de paramétrage de mon sip? |
| 07:52.50 | Manu18 | personne pour m'aider? |
| 07:55.05 | bulkorok | Manu18: english would be better… |
| 07:55.21 | Manu18 | no speak english |
| 07:56.52 | Manu18 | <PROTECTED> |
| 07:57.15 | Manu18 | I aiPar against my shop behind my Livebox pro nothing works. The telephonne does not even register. This can come from a box or concerns setting my sip? a dedicated server at Online.net where I installed my Asterisk server. From home with my Freebox everything works perfectly. |
| 07:57.46 | Manu18 | bulkorok: |
| 08:20.21 | Manu18 | please help me |
| 08:20.36 | *** join/#asterisk KaR]V[aN (~root@95.39.224.52) |
| 08:20.42 | KaR]V[aN | hello |
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| 08:23.55 | Manu18 | bonjour |
| 08:24.25 | KaR]V[aN | I have a question regarding callgroups/pickgroups: I would like to limit pickup only if a call comes from my trunk (ie external call) |
| 08:25.50 | KaR]V[aN | i've been reading about the callgroup/pickupgroup properties but they only seem to apply to extensions |
| 08:26.03 | KaR]V[aN | cant apply them to queues or trunks |
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| 08:43.24 | KaR]V[aN | so, is possible to limit call pickup just to external calls? thanks |
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| 08:57.19 | robscow | on Dial(), to be able to use transfers, do I need to use the r flag? I had it before, had to remove it (my new provider doesn't like it), but now transfers don't work |
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| 09:42.12 | pppingme | anyone have opinion on the grandstream gs-dp715 phones? |
| 09:45.29 | mjt | what's the point in using phone hardware when there are sip softphones which are integrated with everything and can give presence information and other stuff, has built-in bluetooth support, unlimited addressbook and lots'a other goodies? |
| 09:45.36 | mjt | just.. curious, really. |
| 09:46.39 | mjt | modern phones costs about the same as an average office computer |
| 09:52.58 | KaR]V[aN | hardware phones doesn't have crash problems, neither virus. People can answer calls even if their computer is not working properly |
| 09:53.31 | KaR]V[aN | I've faced that when deploying call centers |
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| 10:11.54 | davlefou | hi, some one use goautodial? |
| 10:12.34 | davlefou | I have put it in predictive mode but call is very slow |
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| 11:08.04 | sammoers | hi, any body here ? |
| 11:10.29 | sammoers | I have written a new resource module which listens on a tcp port |
| 11:10.53 | sammoers | from this module I need to get ip address of a sip peer |
| 11:11.07 | sammoers | how to do so ? |
| 11:17.58 | davlefou | is it agi or asterisk module? |
| 11:22.57 | *** join/#asterisk ipalmer (~IceChat77@host81-133-140-79.in-addr.btopenworld.com) |
| 11:24.49 | ipalmer | Afternoon all, I have * 11.5 and need to do the follwoing queue a call > agent1 rings for 10 seconds no answer > agent2 rings for 10 seconds no answer > call requeued into overflow queue. I can't seem to get the agent2 ringing I have set the timeout = 10 and retry = 10 in queues.conf am I missing something? |
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| 11:30.22 | sammoers | sorry davlefou, I thought my connection to irc is facing a problem |
| 11:30.28 | sammoers | I did not notice your answer |
| 11:30.36 | Ice_Strike | Has anyone use Asterisk VM (Asterisk) and DC VM on a same host ESXi? |
| 11:30.58 | sammoers | thank you replying me, it is asterisk module |
| 11:31.23 | sammoers | I have created as a new resource module, it gets loaded |
| 11:31.32 | sammoers | but if I want to be more specific in my question |
| 11:32.33 | sammoers | I should say that because the sip peer table is static to chan_sip module, I do not know how to access this table to fetch ip address of the sip peers which are currently registered |
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| 11:44.43 | Kobaz | what companies do polycom phone repair? |
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| 12:05.18 | xaristax | Hey im trying to install CURL on asterisk but at some point in most tutorials tell me that i have to run ./configure from the asterisk source folder but i cant manage to find this folder im using centos version 6.4 and asterisk 12 install with yum im very newby on this one can someone point me in the right direction to find this folder or install CURL with other method |
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| 12:19.01 | Gugge | xaristax: if you want to compile asterisk yourself, you should not use packages |
| 12:19.14 | Gugge | and if you use packages, only what is in the package is supported |
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| 12:31.02 | Manu18 | Bonjour |
| 12:31.22 | Manu18 | Est ce que qqun pourrait m'aider pour un leger soucis de config? |
| 12:34.56 | *** join/#asterisk valeech (~valeech@wsip-98-172-29-239.dc.dc.cox.net) |
| 12:36.44 | jkroon | hi all, i was wondering whether there is a subscribe: type hint. For example, I'd like to create a hint like exten 3XX,hint,SUB/SIP:peername/${EXTEN}, now, if we get a subscribe for eg 305, we need to subscribe to SIP/peername to receive events for 305, which we then just send back to whomever subscribed for the hint. |
| 12:37.36 | jkroon | is this possible and I'm just being blind or possibly a useful feature? Keep in mind I can't in my particular use-case cluster-style share hints with the remote peer - but I can subscribe to events there. |
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| 12:52.22 | file | jkroon, that functionality does not currently exist |
| 12:53.10 | jkroon | file, crap. any ideas how easy/difficult it would be to implement? |
| 12:53.30 | file | if chan_sip is involved the answer is an automatic "hard enough" |
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| 13:29.36 | ipengineer | Does anyone know of a way from within a macro to run a given context and come back to the macro if macroExit isnt encountered on that given context? Something like this: https://gist.github.com/zconkle/1b18bdd7a8300e90c72a |
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| 13:35.17 | [TK]D-Fender | yoru sample looks messed up and circular |
| 13:35.56 | [TK]D-Fender | And cryptic over-use of pseudo code. |
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| 13:37.57 | ipengineer | [TK]D-Fender: how so? Let me better explain what I am trying to do.. I am wanting to be able to set monitor-times once. The macro will be called several times from different locations in the dialplan but I dont want to have to define those times in the macro. I want the macro to be able to look at the monitor-times context and if the time is not matched it exits the macro because we dont need to record during hours other |
| 13:37.57 | ipengineer | those specified |
| 13:40.18 | ipengineer | almost like if the times are matched we “return” to the macro else we exitMacro |
| 13:40.56 | [TK]D-Fender | looks like your "monitor times context... show BE a macro. |
| 13:41.03 | [TK]D-Fender | should* |
| 13:41.47 | ipengineer | Well I am wanting to store them in realtime is the problem.. We have several different departments that have different times and they change so I am wanting them to be able to update that in realtime.. That is why I have it outside of the macro |
| 13:42.25 | [TK]D-Fender | so have it do a lookup |
| 13:42.47 | [TK]D-Fender | instead of hard-coding |
| 13:43.00 | ipengineer | have the macro do a lookup? |
| 13:43.04 | [TK]D-Fender | sure |
| 13:43.35 | [TK]D-Fender | Query the "times I care about" and cycle through them |
| 13:44.15 | *** part/#asterisk ipalmer (~IceChat77@host81-133-140-79.in-addr.btopenworld.com) |
| 13:44.26 | ipengineer | Should I still use the gotoiftime method and put that in the realtime db or are you saying create a new table and query that table? |
| 13:44.44 | [TK]D-Fender | any way you want |
| 13:45.15 | [TK]D-Fender | Or you could just make duplicate contexts for each dept |
| 13:45.51 | [TK]D-Fender | I don't see multiple accessesw in yrou sample so that make it hard to see the end scenario you are having trouble with |
| 13:46.18 | ipengineer | Ok.. Makes sense… I have no issue creating the records my problem is from the macro I do not know how to query the dialplan for a specific context.. Is there a function for that? |
| 13:46.33 | [TK]D-Fender | Who said query it for a CONTEXT? |
| 13:46.40 | [TK]D-Fender | Query for the TIMES |
| 13:46.58 | [TK]D-Fender | Make a better sample.... |
| 13:47.19 | [TK]D-Fender | Pretty sure you are overcomplicating this |
| 13:48.30 | etc5000 | Hi. Where is better to ask a question related to writing/developing a DSP chip driver? at #asterisk-dev? |
| 13:49.29 | *** join/#asterisk volga629 (~bendersky@76-10-130-18.dsl.teksavvy.com) |
| 13:50.28 | volga629 | Hello Everyone, is there way supply multiply public ip for asterisk in sip settings ? |
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| 13:54.13 | [TK]D-Fender | etc5000: yup |
| 13:54.17 | ipengineer | Prob so.. Let me explain it like this because maybe my example isnt helping.. I am creating one macro that can be called from anywhere in the dialplan and will set monitor on a channel.My problem is I only want monitor to be set during certain hours.. I would ideally create multiple timeframes and if the call falls in those time frames the macro would proceed and set monitor on the channel… So one macro with indefinite |
| 13:54.17 | ipengineer | number of time frames. I would pass the timeframe context or identifier into the macro as an argument so it would know which time frame to check… If that doesnt help I will go to the drawing board again and try to find a better way to illustrate this in an example |
| 13:54.24 | [TK]D-Fender | volga629: No. |
| 13:55.21 | volga629 | That what I though. Will be nice to have to have this feature. |
| 13:55.51 | [TK]D-Fender | ipengineer: Multiple time frames "names" is easy... that is a value you can check ALONGF WITH the time itself |
| 13:56.20 | [TK]D-Fender | ipengineer: Or use as LABEL names for your macro's exten |
| 13:57.26 | [TK]D-Fender | checking times should be a macro itself |
| 13:57.33 | [TK]D-Fender | and easily dynamic in realtime |
| 13:59.53 | Kobaz | do de do |
| 14:00.00 | Kobaz | maybe everyone has had their coffee by now |
| 14:00.10 | Kobaz | anyone know of companies that do polycom repair? |
| 14:01.16 | jameswf | Maybe Polycom |
| 14:02.16 | Chainsaw | Kobaz: We do it ourselves, turning 3 or so "bad" ones in a single good one. |
| 14:02.57 | Kobaz | hah |
| 14:03.06 | Kobaz | resolder the headset jack and etc? |
| 14:03.09 | Kobaz | that's generally what goes bad |
| 14:03.29 | ipengineer | [TK]D-Fender: Sorry I am having a hard time grasping this.. Should be simple IDK what my deal is over here.. I think my biggest issue is HOW to query times from the db from within a macro |
| 14:04.29 | [TK]D-Fender | NO NEED FOR QUERY AT ALL |
| 14:04.41 | *** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 14:04.42 | [TK]D-Fender | oops caps |
| 14:04.54 | ipengineer | They may be in order! |
| 14:05.03 | [TK]D-Fender | ipengineer: You have not specified how you expected to change those hours in the first place |
| 14:05.19 | anonymouz666 | hi ipengineer |
| 14:05.24 | [TK]D-Fender | How is this not just dumbe striaghtforward dialplan with a single exten with labels, etc? |
| 14:05.45 | [TK]D-Fender | If you're going to edit it in realtime.... you should know what you're doing for this. |
| 14:06.07 | [TK]D-Fender | pass it the dept to check and use that as the label to Goto the portion of the macro that pertains to them |
| 14:10.59 | ipengineer | So the times will be in the macro itself. Each department will have a label and the times willk be under there. IF this going to be dynamic I am assuing the macro can be placed in the realtime db? |
| 14:12.08 | WIMPy | Macros are deprecated. |
| 14:13.14 | Kobaz | yeah dont use macro |
| 14:13.20 | ipengineer | I am using macros all over the place.. If I use a gosub on this.. Which I could see working. Will that not create an issue as long as it is not called from within another macro? I have read that mixing macros and gosub is not advised |
| 14:13.36 | Kobaz | just don't use macro at all |
| 14:13.38 | Kobaz | and you'll be fine |
| 14:15.14 | anonymouz666 | the old get_ilbc_source.sh does not work anymore. I need to download the ilbc codec :/ |
| 14:16.07 | ipengineer | I am not at a place I can change everything over.. That is why I have avoided gosub. At some point I am going to have to redo all of this but for the time being I am stuck with them. Can gosub and macros co-exist as long as they are not being called one from the other? |
| 14:16.12 | *** join/#asterisk vlad_sta_ (~vlad_star@212.34.52.66) |
| 14:29.11 | [TK]D-Fender | Macros work fine, Gosub works fine |
| 14:29.17 | [TK]D-Fender | All the same really for your needs |
| 14:29.30 | [TK]D-Fender | And "deprecated" doesn't mean anything and hasn't for years |
| 14:29.42 | [TK]D-Fender | Digium gave up REMOVING functionality pretty much. |
| 14:33.15 | ipengineer | I would prefer to keep using the macro since everything else is.. If they get rid of them at some point I will have to cross that bridge but we are not doing anything overly complicated that has caused us a problem with them. I am going to go back to the drawing board and try to think about this again. [TK]D-Fender I appreciate your help and you have given me some things to think about. |
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| 14:35.21 | jkroon | file, not sure if chan_sip is directly involved. but in order to be able to subscribe via sip would probably require support from chan_sip - in which case yes - it becomes insane. ok, for now the answer is going to be no inter-server BLF. |
| 14:35.39 | [TK]D-Fender | One easy hack they should create is a "macro function" in the dialplan so that there is an inline return value that is stacked |
| 14:35.46 | [TK]D-Fender | It'd be a pretty easy hack. |
| 14:37.20 | ipengineer | Yes.. That would make this a heck of a lot easier.. That is my issue now I cant GoTo anyting becasue I cant return to where I was in the macro if the conditions arent met. |
| 14:40.08 | [TK]D-Fender | Why not? |
| 14:40.17 | [TK]D-Fender | YOU choose when to return. |
| 14:40.37 | [TK]D-Fender | You wouldn't "goto" anyway.. you'd GOSUB |
| 14:40.56 | ipengineer | But I cant gosub from within a macro.. Or so I have read |
| 14:41.07 | [TK]D-Fender | News to me. Have you tried? |
| 14:41.34 | ipengineer | No. I know it works.. I have read in the wiki and somehwere else that they shouldnt be mixed because it can cause asterisk to crash |
| 14:41.50 | ipengineer | If that is not the case then this is an open and closed issue I can have it fixed in 2 minutes |
| 14:44.10 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 14:44.11 | [TK]D-Fender | if it caused a crash.. that'd be a bug and would have been fixed |
| 14:44.13 | [TK]D-Fender | Which wiki? |
| 14:44.23 | ipengineer | wiki.asterisk |
| 14:44.36 | ipengineer | I am looking for it to send you the link. I know I have seen it atleast two times |
| 14:50.35 | ipengineer | Well either I cant find it or I dreamt that up and am confusing the macro depth problem with mixing gosub/macros and need to have my head examined.. The way this day is starting out I would bet money on the latter |
| 14:51.34 | [TK]D-Fender | There was an old quoted macro depth of around 7 or so I recall... |
| 14:51.39 | [TK]D-Fender | Not sure if it still applies. |
| 14:51.42 | [TK]D-Fender | I would doubt it |
| 14:52.40 | ipengineer | Yea and I have never been concerned with it because I never exceed a depth of two.. I have caused myself a world of hurt avoiding gosub if that is in fact not an issue |
| 14:55.43 | file | it depends on stack depth, Macros execute within... themselves |
| 14:56.03 | file | GoSub uses clever logic to not execute within itself, but to jump around accordingly |
| 14:57.38 | ipengineer | I will use gosub inside the macro and see if any issues arise.. That will def make this issue simple to fix |
| 14:58.27 | ipengineer | file: While you are here did you ever check with Mark on the dialog-info/xml implementation? |
| 15:00.26 | file | ipengineer, I did - there's some work to be done with the new pubsub stuff to add support for what is required to implement it again |
| 15:00.55 | file | right now there is no way to store stuff persistently from the modules which generate the payload (ie: dialog-info+xml) and that is required |
| 15:01.25 | file | it's all doable - just needs to be done |
| 15:01.46 | ipengineer | Do you have any thoughts on when that might happen? |
| 15:01.46 | file | my *goal* is to get it done before 13 |
| 15:02.44 | file | but I have no timeline as it's just a side project |
| 15:03.05 | ipengineer | Ok sounds good. Like I said let me know if there is anything I can do.. I have two developers here with lots of C experience that may be able to take a look if I could point them in the right direction |
| 15:05.14 | *** join/#asterisk bluOxigen (~a@unaffiliated/bluOxigen) |
| 15:07.25 | *** join/#asterisk marceloamorim (bd5ac048@gateway/web/freenode/ip.189.90.192.72) |
| 15:09.24 | *** join/#asterisk navaismo (~navaismo@189.191.228.79) |
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| 15:31.34 | *** join/#asterisk Manu18 (56ddaf8b@gateway/web/freenode/ip.86.221.175.139) |
| 15:31.41 | Manu18 | Bonjour |
| 15:32.05 | Manu18 | Il y a des franacais pour un soucis d'enregistrement de tel? |
| 15:32.09 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-ejwyoqvtlillrvtr) |
| 15:32.10 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:33.57 | navaismo | w00t? |
| 15:34.14 | Manu18 | Je n'arrive pas a enregistrer mon tel |
| 15:34.19 | Manu18 | mon sip est configuré |
| 15:34.25 | Manu18 | est ce que je te le paste ? |
| 15:34.55 | navaismo | [TK]D-Fender, is your call ^ |
| 15:35.20 | navaismo | only see some letters |
| 15:35.50 | [TK]D-Fender | Manu18: eVIDEMENT... SI TU DEMONTRE RIEN ON A RIEN A DIRE... |
| 15:36.31 | ageis | [TK]D-Fender: can enabling callcounter and counteronpeer result in peers being deemed busy? |
| 15:36.55 | [TK]D-Fender | ageis: Dpends on your definition of "busy" |
| 15:37.35 | ageis | [TK]D-Fender: definition of busy... "SIP/peer is is busy.. Everyone is busy/congested at this time" |
| 15:37.43 | [TK]D-Fender | that means nothing |
| 15:37.49 | [TK]D-Fender | ageis: and no |
| 15:38.10 | Manu18 | http://pastebin.com/6ymX2Hkp |
| 15:38.21 | ageis | [TK]D-Fender: our client is receiving busy signals which does not mean nothing at the momoent |
| 15:38.37 | [TK]D-Fender | ageis: Wheres the sip debug for it? |
| 15:38.50 | ageis | [TK]D-Fender: ill return when I have it :) |
| 15:39.15 | [TK]D-Fender | ageis: Yeah "I didn't reeally actually look at the call" bit doesn't help much :) |
| 15:39.23 | ageis | it is hard to replicate |
| 15:39.30 | ageis | seems to be working fine right now |
| 15:39.42 | [TK]D-Fender | It's a miracle! |
| 15:39.43 | *** join/#asterisk jpoz (~jpoz@107-1-105-37-ip-static.hfc.comcastbusiness.net) |
| 15:42.34 | Manu18 | [TK]D-Fender: tu as regardé mon sip.conf? |
| 15:42.56 | ageis | [TK]D-Fender: [2014-04-14 15:06:49] VERBOSE[3799] chan_sip.c: -- Got SIP response 486 "Busy Here" back from <local network ip> |
| 15:43.23 | navaismo | your phone return that |
| 15:44.46 | ageis | I wonder if it's just a connectivity problem between the SIP provider and PBX |
| 15:44.52 | ageis | cause this msg then goes to voicemail as it should |
| 15:45.00 | *** join/#asterisk darkdrgn2k (~darkdrgn2@209.90.253.66) |
| 15:45.01 | ageis | but they are reporting busy signals sometimes when calling in |
| 15:45.12 | darkdrgn2k | what field in an invite is used to provide the OUTBOUND call disply on an outgoing call? |
| 15:45.57 | *** join/#asterisk digiv (~digiv@as1.si.umich.edu) |
| 15:47.05 | [TK]D-Fender | ageis: What you showed is a SIP client SAYING that it is busy. What device... you haven't told us |
| 15:47.54 | [TK]D-Fender | darkdrgn2k: Depends. "from:", "Remote Party ID", "p-asserted identity", etc |
| 15:48.01 | ageis | [TK]D-Fender: I realize that, and I am saying this log message is unrelated to the separate report of busy signals |
| 15:48.20 | ageis | [TK]D-Fender: it's probably a network issue between the phone server and SIP provider |
| 15:48.24 | [TK]D-Fender | Manu18: cPis la reste? |
| 15:48.47 | ageis | [TK]D-Fender: because all of these extens are programmed to go to voicemail when they are busy |
| 15:49.07 | [TK]D-Fender | ageis: The "probably" game is my favourite. You can volley it back an forth for hours! |
| 15:49.13 | *** join/#asterisk DougsTech (~DougsTech@c-98-230-105-210.hsd1.al.comcast.net) |
| 15:49.54 | ageis | [TK]D-Fender: I say that cause there is nothing in the full verbose logs. so it likely means Voips cannot contact the PBX. |
| 15:50.00 | ageis | *Voipms |
| 15:51.11 | [TK]D-Fender | Could be a SIP issue... could be networking, could be a lot of things. A better guess is still a guess, but hey... we work with what we've got... |
| 15:51.42 | [TK]D-Fender | could be a "max channels allowed reached, call = busy" |
| 15:52.33 | Manu18 | pour l'enregistrement des telephonne, il ne faut pas que ca ? |
| 15:53.25 | ageis | [TK]D-Fender: how would I know if we reached max channels? |
| 15:53.28 | Manu18 | [TK]D-Fender: |
| 15:54.19 | ageis | [TK]D-Fender: welp. I just pulled the CDR and there are no calls of disposition 'BUSY' or failed. all either ANSWER or NO ANSWER. |
| 15:55.02 | Manu18 | [TK]D-Fender: pour l'enregistrement des telephonne, il ne faut pas que ca ? |
| 15:55.33 | *** join/#asterisk jpoz (~jpoz@ec2-54-193-11-223.us-west-1.compute.amazonaws.com) |
| 15:56.02 | [TK]D-Fender | [11:52]Manu18pour l'enregistrement des telephonne, il ne faut pas que ca ? <- montre pas just un photo de votre auto quand il etait neuf et nous demande pourquoi que t'as eu un accident avec. |
| 15:56.27 | Manu18 | une photo de quoi je n'ai pas compris desolé |
| 15:56.52 | Manu18 | ok j'ai compris |
| 15:57.22 | Manu18 | dans asterisk il ne me dit rien du tout pas de messages d'erreur ou autre [TK]D-Fender |
| 15:58.38 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 16:05.23 | [TK]D-Fender | Manu18: Tu nous donne AUCUN description de la reseautage de votre serveur NI votre telephone. Aucun confirmation que vous etes au CLI ave "sip set debug" en attente d'arrive des donnes, etc... |
| 16:05.40 | Manu18 | ok |
| 16:05.48 | Manu18 | j'ai un telephone thomson st2030 |
| 16:05.53 | Manu18 | et je lance un debug |
| 16:08.14 | Manu18 | http://pastebin.com/9qNA9qFT |
| 16:08.19 | Manu18 | voila [TK]D-Fender |
| 16:10.45 | [TK]D-Fender | ... |
| 16:11.04 | Manu18 | ??? |
| 16:11.28 | [TK]D-Fender | [12:05][TK]D-FenderManu18: Tu nous donne AUCUN description de la reseautage de votre serveur NI votre telephone. <-------------------------- |
| 16:11.33 | [TK]D-Fender | Description NUL |
| 16:11.43 | [TK]D-Fender | DETAILS DE RESEAU |
| 16:11.57 | Manu18 | Que veux tu savoir exactement ? je suis sur un dedié dedibox |
| 16:12.05 | Manu18 | ou est hebergé asterisk |
| 16:12.10 | [TK]D-Fender | Je voit aucun communicate d'origine de votre telehpne dans le debug, aucun essaye d'appel, etc |
| 16:12.17 | Manu18 | je suis de mon magasin derriere une livebox pro |
| 16:12.34 | [TK]D-Fender | Ces noms de marque me dit riens |
| 16:12.35 | Manu18 | [TK]D-Fender: le telephonne ne s'enregistre pas.... |
| 16:12.53 | *** join/#asterisk zamba (marius@flage.org) |
| 16:12.59 | [TK]D-Fender | aucun traffique = problem de reseau |
| 16:13.13 | [TK]D-Fender | Les paquets arrive meme-pas |
| 16:13.27 | Manu18 | La est le sousic |
| 16:13.31 | Manu18 | soucis |
| 16:13.34 | [TK]D-Fender | Ou un telephone mal-configure |
| 16:13.44 | Manu18 | de chez moi ca fonctionne parfaitement |
| 16:13.59 | [TK]D-Fender | Ca nous dit rien |
| 16:14.39 | Manu18 | soucis de routeur alors? |
| 16:15.08 | [TK]D-Fender | On sait pas. |
| 16:15.16 | [TK]D-Fender | Y-a DEUX cotes |
| 16:15.27 | [TK]D-Fender | pis plusiers pieces entre-eux |
| 16:16.55 | Manu18 | de mon mobile ca fonctionne en creant un compte peer |
| 16:19.13 | [TK]D-Fender | Alors ce soit la configuration du telephone, our le reseau entre la telephone & votre serveur |
| 16:19.20 | [TK]D-Fender | ou* |
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| 16:37.33 | anonymouz666 | someone sets [TK]D-Fender channel to language=FR |
| 16:37.36 | anonymouz666 | heh |
| 16:37.39 | *** join/#asterisk jrose_atDigium (~jrose@nat/digium/x-dwlfmcgcporhksax) |
| 16:39.13 | [TK]D-Fender | When my french is is suitably better than their english.... I switch :) |
| 16:39.34 | [TK]D-Fender | git 'er done! |
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| 16:46.17 | mjordan | I've just been impressed over the past few days that you know french. |
| 16:46.52 | [TK]D-Fender | I'd be more impressive if it wasn't the norm here.... |
| 16:47.20 | *** join/#asterisk jpoz (~jpoz@ec2-54-193-11-223.us-west-1.compute.amazonaws.com) |
| 16:47.29 | mjordan | I'm rather mono-linguistic unfortunately. Always impressed by people who are fluent (or conversational) in more than one. |
| 16:50.07 | anonymouz666 | I am trying to improve my english to join the next astricon |
| 16:50.12 | ageis | [TK]D-Fender: what can be done to stop random IPs from sending SIP invites and stuff to my server? |
| 16:50.31 | [TK]D-Fender | ageis: Obviously... FIREWALLS <- |
| 16:50.42 | ageis | ah yes |
| 16:50.56 | anonymouz666 | and share some experience about what we do with asterisk here in brazil |
| 16:52.10 | *** join/#asterisk cmendes0101| (~cmendes01@office.phone.com) |
| 16:52.12 | [TK]D-Fender | anonymouz666: Your english is already pretty good.... |
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| 17:28.11 | file | ipengineer, my train of thought came back re: dialog-info+xml |
| 17:29.28 | file | ipengineer, http://svn.digium.com/svn/asterisk/branches/12/include/asterisk/res_pjsip_pubsub.h has some documentation about core pubsub support, and http://svn.digium.com/svn/asterisk/branches/12/res/res_pjsip_exten_state.c is an implementation for extension state |
| 17:30.20 | file | ipengineer, those are what needs to be changed some to make it so a body generator (what produces the payload) has a way to store information persistently about the subscription |
| 17:30.59 | file | ipengineer, if you have a developer who can look those over and get an idea and is willing to go down that road that would be awesome and we could discuss on the -dev list to hash out the details and make sure everyone is on an agreed upon right approach |
| 17:32.45 | ipengineer | file OK I will sit down with them and see if they can make some sense out of whats going on and let you know what I find out. |
| 17:33.59 | file | ipengineer, the general premise is: res_pjsip_pubsub takes care of the generic pubsub stuff, res_pjsip_exten_state interfaces with the Asterisk core to receive extension state information, it then uses body generators which produce the body (dialog-info+xml) that is sent to devices |
| 17:34.32 | file | ipengineer, every other body doesn't require state so that functionality does not exist ... but dialog-info+xml does need to store state |
| 17:34.57 | file | and that's where we are. |
| 17:37.25 | ipengineer | file: Ok.. For reference, because I have your previous implementation for 12.0.0 what changed? The move over to res_pjsip_pubsub? |
| 17:37.48 | file | ipengineer, stuff was pretty much rewritten and body generators were added |
| 17:38.49 | file | previously the modules which produced the body were more tightly coupled with the extension state module, allowing the needed access to store state information |
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| 17:41.01 | ipengineer | file: Ok Ill engage them this afternoon and touch base with you on twitter or here if that works for you |
| 17:41.26 | file | IRC, asterisk-dev mailing list, hallucinations, all good |
| 17:41.48 | ipengineer | kk |
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| 18:21.29 | ryan_turner|MTW | I have two asterisk servers, with sip friends able to connect to either one. Is there a way I can setup some sort of dynamic routing to make it such that the extensions route properly, either internally or through the other sip server? |
| 18:22.17 | ryan_turner|MTW | So, both asterisk servers are connected, but they're able to determine if they should route the call to the other server or keep it internal? |
| 18:23.37 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 18:25.56 | WIMPy | What is dynamic about your setup? |
| 18:27.36 | ryan_turner|MTW | which asterisk server the clients are connecting to |
| 18:28.23 | ryan_turner|MTW | The more I think about this, the less value there is in it... |
| 18:28.47 | WIMPy | So you have the same announts on both and do some DNS rr or something? |
| 18:28.54 | ryan_turner|MTW | anycast |
| 18:29.01 | WIMPy | Sorry, but my magic sphere sems to be broken again. |
| 18:29.30 | WIMPy | You can use DUNDi for that. |
| 18:30.16 | ryan_turner|MTW | WIMPy, thanks much! |
| 18:30.25 | WIMPy | Together with regcontext. |
| 18:31.25 | WIMPy | Or just try yourself and fail over to the other server. |
| 18:31.40 | ryan_turner|MTW | Yeah, fantastic |
| 18:32.15 | ryan_turner|MTW | I wonder if many ip phones support certificate authentication |
| 18:32.25 | WIMPy | But if you have more than 2 servers in total, the first option would make sense. |
| 18:33.08 | ryan_turner|MTW | Yeah, idea here is to get a "mesh" going |
| 18:33.40 | WIMPy | Then DUNDi is your friend. |
| 18:34.17 | ryan_turner|MTW | How does DUNDi handle authentication? |
| 18:34.50 | WIMPy | Different levels. See the sample config. |
| 18:35.02 | ryan_turner|MTW | I guess what I mean is, do I need to handle syncing sip.conf on each server, or does DUNDi have some built in method |
| 18:35.10 | ryan_turner|MTW | Alrighty, will do. |
| 18:35.35 | WIMPy | No. They are completely different things. |
| 18:35.57 | WIMPy | But you could take an additional look at dynamic realtime. |
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| 20:28.37 | *** join/#asterisk Katty (~Katty@97-91-121-70.static.stls.mo.charter.com) |
| 20:29.09 | Katty | hello, cupcakes! |
| 20:29.30 | Katty | i've come to be sarcastic! and annoying! |
| 20:29.48 | [TK]D-Fender | Nobody here but us muffins... |
| 20:30.20 | [TK]D-Fender | checkout time, BBL |
| 20:30.25 | Katty | bye fender. |
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| 21:05.33 | dc3 | been at this problem for a few days, dropped a testing PBX with a TDM800P 4 FXO next to the desk and ran my existing FXS DID PSTN line to FXO_1 and experiencing incredible echo |
| 21:05.59 | Katty | you shouldn't drop pbxes. |
| 21:06.01 | Katty | that's not very nice. |
| 21:06.15 | dc3 | echo training and tuning with conventional software means unsuccessful, OSLEC manages it but it sounds overall bad, and random clicks and pops. |
| 21:06.18 | dc3 | ha. |
| 21:06.51 | dc3 | I am wondering about impedance mismatch but beyond building a balan myself to impedance match or xformer couple this, I am at a loss I could be experiencing such bad echo |
| 21:07.08 | Katty | i'm wondering if i should make broccoli or carrots to go with dinner. |
| 21:07.24 | Katty | but my brain is fried at this point of the day. |
| 21:07.28 | WIMPy | Analog is evil! |
| 21:07.34 | Katty | hello, WIMPy! |
| 21:07.50 | WIMPy | Brain with broccoli? |
| 21:07.53 | dc3 | I have read as many tutorials or posts as I can but starting to think I am dealing with something more sinister like the PBX FXS I am tied to is fighting me with its own echo cancellation, etc. I am only 50 feet from the PBX |
| 21:08.12 | Katty | WIMPy: know what else is evil? |
| 21:08.16 | Katty | Nugget: telnet |
| 21:08.22 | dc3 | are these battling bots? |
| 21:08.26 | Katty | Nugget: aww :< |
| 21:08.33 | WIMPy | Katty: SIP |
| 21:08.37 | Katty | YES! SIP is also evil. |
| 21:09.09 | Katty | infobot: WIMPy |
| 21:09.20 | Katty | infobot: how do you not have an entry for WIMPy!? |
| 21:09.55 | Nugget | huggles Katty anyway |
| 21:10.00 | Katty | infobot: WIMPy is <WIMPy> Analog is evil! |
| 21:10.00 | infobot | Katty: okay |
| 21:10.06 | Katty | infobot: WIMPy |
| 21:10.07 | infobot | from memory, wimpy is <WIMPy> Analog is evil! |
| 21:10.18 | Katty | hugskwishes on Nugget |
| 21:10.50 | Katty | Nugget: how're you dear? did you survive biking eleventybillions miles? |
| 21:11.10 | Nugget | crashed two miles from the end, but still managed to finish |
| 21:11.15 | Nugget | bike's sort of banged up though |
| 21:11.25 | Katty | did you break anything? |
| 21:11.55 | Nugget | just some bruises and a little blood, nothing serious |
| 21:12.07 | Katty | well i'm glad you didn't get hurt too bad. |
| 21:12.14 | Katty | do you feel accomplished? |
| 21:22.49 | *** join/#asterisk serkanTan (d983cebb@gateway/web/freenode/ip.217.131.206.187) |
| 21:23.56 | serkanTan | hi, i need a consultant to help me with price. I have a problem about asterisk configuration |
| 21:25.25 | serkanTan | Are there anyone to help me :( |
| 21:25.49 | WIMPy | ~ask |
| 21:25.49 | 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. |
| 21:27.30 | serkanTan | when i create a call file for example test.call to call an extension for example extension 100 its working good |
| 21:28.02 | serkanTan | but when i want to call not local number for example my mobile phone |
| 21:28.31 | serkanTan | there is an error : chan_sip.c:23031 handle_response_invite: Received response: "Forbidden" from '"Unknown" <sip:Unknown@85.105.134.157>;tag=as50de4d1c' |
| 21:29.15 | WIMPy | Well, "Forbidden" seems like a good indication to me. |
| 21:29.36 | serkanTan | Channel: SIP/101 Application: Playback Data: hello-world |
| 21:29.46 | serkanTan | this is my call file for local extensions |
| 21:29.54 | WIMPy | Check that peers configuration. |
| 21:31.14 | serkanTan | i notice that my peer which use for outgoing calls is rejecting my connection |
| 21:31.21 | serkanTan | ive tried everything |
| 21:31.39 | WIMPy | Do you use it at all? |
| 21:32.34 | serkanTan | ive only one outgoing trunk |
| 21:32.39 | serkanTan | named Fernus |
| 21:33.00 | serkanTan | Channel: SIP/Fernus/0532xxxxxxx |
| 21:33.07 | serkanTan | Application: Playback |
| 21:33.13 | serkanTan | Data: hello-world |
| 21:33.21 | serkanTan | Fernus is my outgoing trunk |
| 21:33.40 | serkanTan | i think there is a something wrong my peer settings |
| 21:34.24 | serkanTan | i can paste my settings if im not disturbing |
| 21:34.31 | serkanTan | really im very bored |
| 21:35.17 | serkanTan | type=peernat=yes |
| 21:35.18 | jameswf | Pastebin |
| 21:35.24 | serkanTan | insecure=very dtmfmode=info&rfc2833 allow=ulaw&alaw disallow=all host=sip.verimor.com.tr username=90312xxxxxxx secret=xxxxxxxxxx sendrpid=yes trustrpid=no context=from-trunk canreinvite=yes |
| 21:36.00 | serkanTan | this is my peer details content |
| 21:36.42 | serkanTan | http://pastebin.com/cV29QFvJ |
| 21:36.58 | WIMPy | Hmm. isn't 'username' incomming only? |
| 21:37.25 | WIMPy | I think that might have to be 'fromuser' instead. |
| 21:37.48 | serkanTan | my USER Details and user context fields are empty |
| 21:38.07 | serkanTan | in peer configuration |
| 21:38.13 | WIMPy | What user details? |
| 21:38.25 | WIMPy | And context if for calls TO Asterisk. |
| 21:38.27 | serkanTan | in freepbx |
| 21:38.32 | WIMPy | is |
| 21:38.37 | serkanTan | when i configure a peer |
| 21:39.06 | serkanTan | there is a user details field empty |
| 21:39.15 | WIMPy | Uh. You should ask there, then. |
| 21:39.20 | WIMPy | ~freepbx |
| 21:39.20 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 21:39.45 | serkanTan | ok |
| 21:39.49 | serkanTan | thank you |
| 21:39.51 | WIMPy | It won;t like it if you change it's config files. |
| 21:40.00 | serkanTan | but freepbx users ca not understand call files |
| 21:40.08 | WIMPy | Or it will just overwrite them on the next opportunity. |
| 21:40.14 | serkanTan | they are only interface users |
| 21:40.17 | jameswf | :-/ |
| 21:41.37 | serkanTan | thank you |
| 21:41.43 | jameswf | writing a whole interface around Asterisk including complex dialplans and scripts while having 0 knowledge of Asterisk |
| 21:44.05 | serkanTan | i think so. Asterisk fans dont li FreePBX, m i right? |
| 21:44.10 | serkanTan | *like |
| 21:46.57 | anonymouz666 | ~gs |
| 21:46.58 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 21:47.12 | anonymouz666 | still valid statement? |
| 21:50.21 | jameswf | serkanTan: You are misinformed. Not supporting it because it is a "specialty thing" is not the same as not liking it. You will find many folks both here and in #FreePBX but you are directed there because it is off topic here. |
| 21:51.10 | jameswf | anonymouz666: I think you will either love or hate grandstream it is a personal preference thing. |
| 21:52.08 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-108-132-18-211.mia.bellsouth.net) |
| 21:52.19 | Mhaddog | evening all |
| 21:53.47 | newtonr | serkanTan, FreePBX is a GUI for configuring Asterisk. If you are a fan of FreePBX, then you are an Asterisk fan. :) |
| 21:54.01 | *** part/#asterisk Mhaddog (~Mhaddog@adsl-108-132-18-211.mia.bellsouth.net) |
| 21:54.21 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-108-132-18-211.mia.bellsouth.net) |
| 21:57.12 | lanning | Hi. Does anyone know if there is some trick to get the Polycom SoundPoint IP 335 (sip.ld v3.3.3) to register correctly? |
| 21:57.22 | lanning | I keep getting "wrong password" |
| 21:57.40 | jameswf | lanning: enter the correct password :) |
| 21:57.45 | lanning | funny |
| 21:58.13 | newtonr | jameswf, double check your user name, probably trying to auth against the wrong peer/endpoint |
| 21:58.17 | newtonr | oops |
| 21:58.19 | lanning | I stripped the password down to just "hello" |
| 21:58.20 | newtonr | lanning, ^ |
| 21:59.00 | lanning | nope, sip debug shows the correct sip user |
| 21:59.24 | lanning | hold on, let me put together a posting |
| 22:01.39 | newtonr | lanning, be sure you have verbose and debug messages going to console and both turned up to 5 in addition to the SIP trace so we can see everything |
| 22:04.31 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 22:05.53 | lanning | http://pastebin.com/NzJviwMf |
| 22:06.06 | lanning | that has everything except for the verbose. |
| 22:08.05 | lanning | actually with verbose set to 5, I don't see anything different from what I posted. |
| 22:09.25 | newtonr | lanning, try changing the type to friend instead of peer |
| 22:10.50 | newtonr | lanning, you can run "logger show channels" to see what debug channels you have going to the console. |
| 22:11.13 | lanning | still get "wrong password" |
| 22:11.35 | *** part/#asterisk volga629 (~bendersky@CPE085b0e07d3f2-CM788df74f6260.cpe.net.cable.rogers.com) |
| 22:11.43 | *** join/#asterisk sethhurst (~sethhurst@in-67-236-228-171.dhcp.embarqhsd.net) |
| 22:12.48 | sethhurst | The asterisk interface is not accessable on the windows vertion is there a way I can use a alternative interface for the windows version? |
| 22:17.41 | newtonr | lanning, grab another output, verify verbose and debug log channels are showing, "logger show channels" to verify that, and "core show settings" to verify your verbosity levels for both verbose and debug log channels |
| 22:19.42 | lanning | ok, hold on |
| 22:20.40 | *** join/#asterisk jpoz (~jpoz@ec2-54-193-11-223.us-west-1.compute.amazonaws.com) |
| 22:22.54 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:28.18 | newtonr | lanning, also, show the output of "sip show peers" and "sip show users" |
| 22:29.22 | *** join/#asterisk ttyS3 (~o@gateway/tor-sasl/omlib) |
| 22:31.11 | *** join/#asterisk mahlon (~mahlon@martini.nu) |
| 22:32.28 | newtonr | sethhurst, what "windows version" are you talking about? |
| 22:41.21 | sethhurst | The windows 32 version of asterisk |
| 22:41.49 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 22:42.17 | *** join/#asterisk moikmellah_ (~moikmella@hephaestus.restechservices.net) |
| 22:42.39 | lanning | Sorry, got called off to support someone... :) here is the new snapshot: http://pastebin.com/z3hsby9Y |
| 22:43.59 | newtonr | sethhurst, No one maintains the windows port since many years ago. You are only going to find problems trying to use that. |
| 22:44.10 | moikmellah_ | Anyone know if there's still a spot I can find a comprehensive Changelog for dahdi-linux? |
| 22:44.38 | moikmellah_ | Seems it's no longer included in the .tgz downloads, since around v2.6.2. |
| 22:44.56 | sethhurst | I did use it on a virtual server it worked because I was using piaf green. |
| 22:45.23 | newtonr | lanning, your 'logger show channels' output shows you don't have the 'verbose' or 'debug' log channels going to the console... |
| 22:46.21 | sethhurst | I'm trying to setup a chat line system is there a way I can have people press 0 and have them sent back to the last menu such as the rooms menu. Then if they want to go back to the main IVR they can press 0 again? |
| 22:48.00 | [TK]D-Fender | sethhurst: Yes, you can do something like that with * |
| 22:48.17 | sethhurst | I can't seam to find a good thing to run the rooms. |
| 22:49.13 | sethhurst | I want it to anounce how many callers are in the room and have star codes for navagating. |
| 22:49.35 | sethhurst | I can give you a number to a line that uses this system. |
| 22:49.45 | sethhurst | That is what I'm looking for. |
| 22:49.47 | [TK]D-Fender | First go learn how *'s 2 conferencing apps work before getting too picky |
| 22:50.03 | [TK]D-Fender | "MeetMe" and "ConfBridge" |
| 22:50.08 | [TK]D-Fender | ~book |
| 22:50.08 | 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 |
| 22:50.10 | [TK]D-Fender | ^^^ |
| 22:50.26 | sethhurst | How do you setup conf bridge? |
| 22:50.57 | sethhurst | I am using a virtual server for a small volume of calls and am running under SSH connection. |
| 22:51.13 | [TK]D-Fender | Go read the book, the sample configs, and try it out |
| 22:51.42 | sethhurst | Ware can I find this? |
| 22:52.14 | [TK]D-Fender | read the link above |
| 22:52.22 | newtonr | :D |
| 22:52.30 | [TK]D-Fender | and go install Asterisk. It comes with sample configs that ARE documentation |
| 22:52.49 | sethhurst | okay. |
| 22:53.18 | sethhurst | I am having 10 rooms five are live rooms ware the other five are up to two people. |
| 22:54.55 | [TK]D-Fender | "up to 2 people"? |
| 22:55.23 | [TK]D-Fender | that's a regular "phone call".... not really a "conference" |
| 22:55.25 | sethhurst | Is it possable to have star codes for each of the rooms? |
| 22:55.42 | *** join/#asterisk KOPRajs (c3272c1a@gateway/web/freenode/ip.195.39.44.26) |
| 22:55.47 | sethhurst | Well this is how the system is setup. |
| 22:55.57 | sethhurst | Caller calls in main greating |
| 22:56.06 | sethhurst | gos to main IVR |
| 22:56.19 | sethhurst | option 1 goes to live rooms. |
| 22:56.37 | sethhurst | option 2 goes to private rooms. |
| 22:56.46 | jameswf | hmmmmmm |
| 22:57.09 | lanning | anonymous 1on1 chat |
| 22:57.29 | sethhurst | When caller presses 1 it takes him to live rooms then when caller presses the room number and * they will be put in to the room with the other callers. |
| 22:57.39 | [TK]D-Fender | Like Chatroulette..... without the random dongs ;p |
| 22:57.53 | sethhurst | So when a caller wants to leave they will press *0 to go back. |
| 22:57.56 | [TK]D-Fender | sethhurst: Sure. Go do it |
| 22:58.10 | [TK]D-Fender | sethhurst: Just don't get to picky on the EXIT KEYS necessarily |
| 22:58.34 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
| 22:58.39 | jameswf | those ae the keys that have already been recorded in a sexy voice... |
| 22:58.41 | sethhurst | Sort of like the late night chat line and the 712 432 numbers. |
| 22:59.06 | KOPRajs | hi, I'm running Asterisk 1.8 od Debian behind NAT... my incoming calls are dropped after about 3 minutes with these messages in CLI: |
| 22:59.20 | KOPRajs | [Apr 17 00:48:15] WARNING[38815]: chan_sip.c:20457 handle_response_invite: just did sched_add waitid(2260028) for sip_reinvite_retry for dialog 1c62bba55b870ec97373f3900be9c7b4@91.221.212.167:5060 in handle_response_invite [Apr 17 00:48:22] WARNING[38815]: chan_sip.c:3656 retrans_pkt: Retransmission timeout reached on transmission 1c62bba55b870ec97373f3900be9c7b4@91.221.212.167:5060 for seqno 105 (Critical Request) -- See https://wi |
| 22:59.30 | jameswf | ~pb |
| 22:59.31 | 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. |
| 22:59.42 | *** part/#asterisk Joel (~Joel@unaffiliated/joel) |
| 23:00.59 | KOPRajs | http://pastebin.com/W5bUkf1T |
| 23:01.25 | sethhurst | is it possable to setup conf bridge using a web interface like freepbx? |
| 23:01.51 | sethhurst | I only see a modual for meetme and not conf bridge. |
| 23:01.55 | KOPRajs | any ideas on what am I doing wrong? |
| 23:02.06 | [TK]D-Fender | sethhurst: You want a custom menu setup... this is not what FreePBX is for |
| 23:03.48 | KOPRajs | is this problem with the NAT or is the service provider not handling some re-invites correctly? |
| 23:04.49 | moikmellah_ | KOPRajs: Likely a NAT issue. Try playing with timeout/expiry values in sip.conf. |
| 23:06.16 | *** part/#asterisk moikmellah_ (~moikmella@hephaestus.restechservices.net) |
| 23:09.18 | sethhurst | Freepbx is the web interface witch I can use to run the system like setuping up ivr's and roots. |
| 23:09.25 | KOPRajs | to connect the asterisk to the service provider should I set type peer or friend? |
| 23:09.31 | sethhurst | setting up |
| 23:13.12 | lanning | sethhurst: using a GUI (like FreePBX) will massively limit what you can do. The custom menus and conferences are not supported via a GUI. They need to be hand coded in the configs. Learn extensions.conf |
| 23:14.55 | jameswf | ~book @ sethhurst |
| 23:14.55 | infobot | ACTION smacks @ sethhurst upside the head with a book |
| 23:15.04 | jameswf | that didn't work |
| 23:15.07 | jameswf | or did it |
| 23:15.13 | jameswf | ~book |
| 23:15.13 | 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 |
| 23:15.16 | sethhurst | :D |
| 23:18.03 | *** join/#asterisk jpoz (~jpoz@ec2-54-193-11-223.us-west-1.compute.amazonaws.com) |
| 23:18.56 | *** part/#asterisk sethhurst (~sethhurst@in-67-236-228-171.dhcp.embarqhsd.net) |
| 23:39.04 | lanning | ok, here it is with the debug channel et al. |
| 23:39.07 | lanning | http://pastebin.com/bSFh5bux |