| 00:17.40 | *** join/#asterisk robert_ (~hellspawn@objectx/robert) |
| 00:18.35 | robert_ | so every so often, Asterisk says something about "db_put_sql: Cannot execute statement, the database is locked." or something. What's it mean? |
| 00:18.50 | robert_ | and why is Asterisk keep locking the database? |
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| 00:23.01 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 01:16.13 | mnathani | Whats a good way to prevent certain users from making international calls? |
| 01:17.00 | WIMPy | Don't allow them to do so. |
| 01:19.14 | navaismo | hehe |
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| 01:52.05 | snadge | if someone could help me figure out why my voip connection is no longer working.. i'll love them forever |
| 01:52.06 | RudyValencia | how do we check for SRTP encryption when a call is dialed (to play a message stating that the line is not secure if there is no encryption)? |
| 01:52.34 | snadge | ive moved premises.. and the isp has deleted my old voip account, and created a new one |
| 01:52.57 | snadge | so i've changed all the references i could find of the old username and password.. to the new one |
| 01:53.12 | snadge | yet i still see 603 declined.. after a period of time.. and "all circuits are busy" audio message |
| 01:53.49 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 01:54.05 | snadge | asterisk -r .. and sip show registry.. shows state "Request sent" |
| 01:57.12 | snadge | i see they've changed the ip address of their domain |
| 02:02.41 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.171) |
| 02:03.24 | robert_ | so can I check for _SIP_SRTP_SDES on any particular channel? |
| 02:05.47 | *** join/#asterisk gusto (~gusto@ppp-93-104-86-120.dynamic.mnet-online.de) |
| 02:28.02 | snadge | should i be forwarding some ports through to my asterisk server if its behind a nat? |
| 02:28.45 | snadge | eg.. tcp 5060 and UDP 30000 to 31000 (after editing rtp.conf) |
| 02:29.01 | snadge | or is there some more appropriate way to do that |
| 02:29.41 | snadge | i can configure my voip account with a softphone (thats also behind a nat) and that works.. but if i put those settings into my asterisk server.. it doesn't seem to be able to register |
| 02:47.29 | snadge | ok that worked.. i dont understand why, but i've stopped caring.. lets just hope that exposing ports 5060 tcp/udp and udp 30000 - 31000.. isnt a significant security risk |
| 02:48.38 | snadge | or just unnecessary |
| 02:49.10 | snadge | im not appreciating why i have to forward ports to get it to work.. when a soft phone (also behind nat) just works |
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| 03:06.16 | MrMeek | snadge, STUN maybe. guessing. http://en.wikipedia.org/wiki/STUN |
| 03:10.55 | snadge | asterisk does .. or does not use stun.. or the softphone app im using uses it which is why it works |
| 03:11.12 | snadge | ive dropped tcp port 5060 because apparently thats unnecessary |
| 03:37.36 | RudyValencia | hey is there a way to know if a connection is SRTP-encrypted? |
| 03:37.52 | RudyValencia | (so that we can play a recording that says the line isn't secure if it's not encrypted) |
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| 03:58.07 | sideone | hey all |
| 03:59.24 | sideone | i am working on a project to create a conference room with audio, video, and text. I will most likely develop my own client, would confbridge work for this project? I am trying to decide which platform to build from. |
| 03:59.28 | sideone | thanks. |
| 04:03.23 | *** join/#asterisk FireAndIce (~FireAndIc@219.91.142.244) |
| 04:03.44 | sideone | i know i can send text to a direct caller, but can i send a message to a conf room? |
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| 04:16.20 | lorsungcu | sideone: look at bigbluebutton |
| 04:16.47 | sideone | i was looking into it, but as it uses 12 different sources, i think it may be an overkill for me |
| 04:17.06 | sideone | i think * + conf may do what i need, just unsure if text can be passed between the conf |
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| 04:24.42 | RudyValencia | does Asterisk provide a way to tell if a connection is encrypted (so we can play a warning if there is none)? |
| 05:03.45 | robert_ | Nobody? |
| 05:13.56 | robert_ | Alright then. |
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| 06:19.12 | ChannelZ | RudyValencia: dig through the CHANNEL function, probably something in there |
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| 06:41.58 | robert_ | ChannelZ: We're trying to get TLS and SRTP working with our preferred client, MicroSIP. |
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| 06:45.30 | Rhomber | I can get call parking to work with Dial options kKtT, but i was wondering if there was a default way to allow the function? |
| 06:46.00 | Rhomber | like, i noticed a setting in users.conf.. and in [general] i have everything turned on |
| 06:46.22 | Rhomber | doesn't seem to work though.. i mean, if you park a call.. then collect it.. you can't park/transfer it again |
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| 06:48.27 | cchhat01 | hello |
| 06:48.48 | cchhat01 | I've had my asterisk JIRA submission closed and status changed to "not a bug" |
| 06:48.53 | cchhat01 | by Matt Jordan |
| 06:49.05 | cchhat01 | how do I go about re-opening it, I'm not seeing an option |
| 06:49.58 | *** join/#asterisk Drupl (~Dax@41.222.123.2) |
| 06:51.02 | Drupl | Would anybody be able to help me with troublshooting a SIP trunk, I work for a hotel and need assistance it's gone wonky |
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| 07:30.11 | ChannelZ | what do you mean "a default way to allow the function" ? |
| 07:30.45 | ChannelZ | Like, so you don't have to pass those options in Dial? |
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| 07:47.19 | kaldemar | Rhomber: forget users.conf. see features.conf for parking options. |
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| 07:59.41 | hrolf | sd |
| 07:59.41 | hrolf | f |
| 07:59.41 | hrolf | sd |
| 07:59.41 | hrolf | f |
| 07:59.47 | hrolf | Sorry |
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| 08:05.14 | hrolf | I have set up two asterisk servers and connected them with SIP. I call on Server A and it dials through SIP peer to the other server. Now I want to transfer this same call back to an extension of Server A using the Transfer function. |
| 08:05.48 | hrolf | So I did Transfer(SIP/240/444) |
| 08:06.14 | hrolf | where SIP/240 is the SIP peer extension on Server B |
| 08:06.46 | hrolf | Now the question? Do I have to create a dialplan with extension 444 (the transferred-to number) in Server A? |
| 08:08.04 | hrolf | How do I do such that when anyone from Server B transfers to the number 444 on Server A, an extension on Server A answers the call. |
| 08:08.28 | hrolf | Basically, the call arrives on Server A and the agent is also on Server A, but the IVR is on Server B. |
| 08:08.38 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 08:08.52 | hrolf | Server B has to transfer to the agent sitting in Server A. |
| 08:09.05 | hrolf | Transfer(SIP/240/444) is not working |
| 08:09.08 | hrolf | Any help? |
| 08:11.39 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.171) |
| 08:15.30 | andross | exten => 444,1,Dial(SIP/serverA,30) |
| 08:16.05 | andross | exten => 444,1,Dial(SIP/phone-444,30) |
| 08:16.29 | andross | make sure that the second line is in the incoming context on serverA |
| 08:16.32 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.82.207) |
| 08:17.10 | andross | actually i think my brain isnt working one sec |
| 08:18.19 | hrolf | andross: Is the Transfer function okay? Should it be Transfer(SIP/240/444) or just Transfer(SIP/444)? |
| 08:18.33 | hrolf | I have created the extension on Server A as 444 |
| 08:19.02 | hrolf | and SIP/240 is a SIP extension of type=peer on Server B which connects to Server A |
| 08:19.11 | andross | i have never used the transfer function so im not sure about that |
| 08:22.39 | hrolf | andross: Actually, it worked with Transfer(SIP/444), though I'm astonished how it assumed that 444 is extension on server A and not a SIP extension on Server B. |
| 08:22.44 | hrolf | Anyone care to explain? |
| 08:23.34 | andross | does your context for 444 point to serverB |
| 08:23.37 | andross | in sip.conf |
| 08:24.18 | hrolf | andross: the extension 444 is in my context in Server A, and I'm doing Transfer(SIP/444) from Server B |
| 08:25.40 | hrolf | Call -> Server A -> SIP Peer -> Server B and then on Server B Extension 444 on Server A <- Transfer <- Call on Server B |
| 08:26.00 | ChannelZ | Transfer doesn't transfer to an extension, it transfers to another device/channel. |
| 08:26.43 | hrolf | ChannelZ: In my case, 444 is extension on Server A. |
| 08:26.57 | hrolf | ChannelZ: and Transfer(SIP/444) is working. |
| 08:27.16 | ChannelZ | Do you have a peer called 444? |
| 08:27.37 | hrolf | ChannelZ: Nope. |
| 08:28.22 | hrolf | ChannelZ: On Server A, the SIP peer is 340 with its host=ServerB |
| 08:28.37 | hrolf | ChannelZ: On Server B, the SIP peer is 240 with its host=ServerA |
| 08:29.02 | ChannelZ | those are terrible peer names but nevermind. Maybe I'm thinking of a different app, looking |
| 08:29.04 | Rhomber | ChannelZ: so it works all the time for a user.. as i said, when any other internal dial operations are performed you can no longer transfer/park |
| 08:29.15 | hrolf | ChannelZ: There is extension 444 in my context on Server A. |
| 08:29.25 | Rhomber | so you transfer to parking lot... then collect parked call.. then # doesn't work any more |
| 08:29.33 | hrolf | ChannelZ: and I'm doing Transfer(SIP/444) on Server B |
| 08:30.03 | andross | on serverB is the default outgoing to serverA |
| 08:30.06 | hrolf | ChannelZ: Now I want to know how does the Transfer(SIP/444) determined that the extension 444 is on ServerA and not locally |
| 08:30.26 | andross | are you matching outgoing on serverB with something line _X. |
| 08:30.29 | ChannelZ | It probably doesn't, what does extension 444 do? Dial(SIP/theotherserver/444) I'd assume |
| 08:30.34 | andross | like |
| 08:30.56 | hrolf | ChannelZ: on Server A, extension 444 does Dial(SIP/100), SIP/100 is local agent sitting on Server A |
| 08:31.10 | hrolf | andross: Nope. |
| 08:31.11 | andross | it makes sense |
| 08:31.11 | andross | Transfer(SIP/240/444) |
| 08:31.15 | andross | 240 is the other server |
| 08:31.19 | andross | so its sending the call there |
| 08:31.34 | hrolf | andross: But Transfer(SIP/240/444) fails |
| 08:32.50 | andross | you may want to call your sip peers by the connecting server IP or domain name |
| 08:32.53 | andross | just for sanity |
| 08:33.01 | *** part/#asterisk deo_ (~deo@122.53.72.218) |
| 08:33.31 | hrolf | You mean I should do Transfer(SIP/240/444) ? |
| 08:33.39 | hrolf | instead of Transfer(SIP/444)? |
| 08:35.09 | ChannelZ | Transfer doesn't use the same syntax as Dial, they're not the same thing. It actually does transfer to an extension. |
| 08:35.48 | kaldemar | what gets sent with Transfer is not a peer, just a number. if you use SIP/100, the Refer-To header has 100@yourserver, which makes the peer act like it dialed 100. |
| 08:36.38 | Rhomber | ok, but i want the # feature to always work |
| 08:36.51 | ChannelZ | sorry Rhomber that was in relation to hrolf |
| 08:36.56 | kaldemar | hrolf: also, don't call [100] in sip.conf an extension, because it is not an extension but a device or a peer rather. extensions are what you have in extensions.conf. |
| 08:36.57 | Rhomber | oh sorry |
| 08:37.11 | *** join/#asterisk taylorbyte2013 (~cyberninj@2001:470:1f05:12cc:1e6f:65ff:fec2:bec7) |
| 08:37.22 | ChannelZ | For your question, all your Dials need appropriate flags if you're using feature code transfers. |
| 08:37.33 | ChannelZ | Aren't you on SIP phones? |
| 08:37.52 | kaldemar | Rhomber: did you look at the options in features.conf? |
| 08:37.56 | Rhomber | will be SIP and mobiles via trunc |
| 08:38.15 | Rhomber | kaldemar: yes, like 50 times |
| 08:38.53 | kaldemar | Rhomber: also note that if you use a dtmfmode for SIP that uses RTP, asterisk must stay on the media path to detect DTMF. |
| 08:38.55 | Rhomber | ChannelZ: well it works, as i said.. using Dial.. but after you collect a parked call you can't use any features |
| 08:38.57 | ChannelZ | Well you can't arbitrarily/globally turn on features that I'm aware of |
| 08:39.13 | Rhomber | dtmf works or i wouldn't be able to park the call in the first place. |
| 08:39.26 | Rhomber | hmm |
| 08:39.44 | kaldemar | Rhomber: that is a different call and no proof of DTMF being detected after it. |
| 08:39.54 | Rhomber | i've read about the 'media path' .. wouldn't Dial/Transfer both be simmilar? |
| 08:40.17 | kaldemar | Rhomber: Dial an Transfer have pretty much nothing in common. |
| 08:40.40 | kaldemar | Rhomber: directmedia parameters in sip.conf are for controlling whether asterisk stays on the media path or not. |
| 08:40.57 | Rhomber | right, sounds like asterisk needs a rewrite |
| 08:41.29 | kaldemar | or your configs need a rewrite. |
| 08:41.54 | ChannelZ | What he's saying is if you use include => parkedcalls and then you transfer someone to a parking lot, when you dial 701 or whatever to pickup the parked call, you then can't re-transfer it anywhere. Hadn't ever thought of that.. because 701 isn't an extension he created to put Dial() options on. Is there a default behavior? |
| 08:41.56 | hrolf | kaldemar: what if I had to transfer the same call to a local extension on Server B? |
| 08:42.34 | Rhomber | directmedia is set to no for all devices |
| 08:42.35 | kaldemar | hrolf: try Transfer(SIP/user@address) |
| 08:43.11 | Rhomber | ChannelZ: exactly |
| 08:43.55 | Rhomber | say i received a call to my mobile.. i'd want to transfer it to the SIP phone incase i lost coverage.. and then say later on.. i wanted to transfer it to a colleague to finish up.. i couldn't |
| 08:44.03 | Rhomber | a perfectly (in my usage) valid scenario |
| 08:44.12 | ChannelZ | I have no idea. It's nothing I've ever played with. It's rare I even use parking, but I don't use feature transfers. |
| 08:45.48 | Rhomber | hmm, fair enough. id like to be able to make my own features too |
| 08:46.37 | Rhomber | but yeah, it would be nice to define features for a user.. which looks to be what's happening in users.conf |
| 08:47.03 | Rhomber | but as i said, i have 'yes' for them all in [general] (or [default].. which ever it is..) |
| 08:48.16 | ChannelZ | yes for what where |
| 08:48.29 | kaldemar | ~users.conf |
| 08:48.30 | infobot | [~users.conf] users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable asterisk config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 08:49.52 | *** join/#asterisk deo_ (~deo@222.127.13.226) |
| 08:49.56 | kaldemar | Rhomber: you don't even need parking for transfers. you just need the transferred calls to go through a Dial that enables transfers. |
| 08:50.12 | ChannelZ | He wants parking for parking. |
| 08:51.29 | ChannelZ | It's then later trying to transfer a call that was picked up from a park that doesn't work, as a feature code, not a SIP transfer (which should work fine because it's unrelated to the Dial transfer options) |
| 08:53.16 | *** join/#asterisk threesome (~threesome@customer-79-127-150-148.net.angelnet.cz) |
| 08:58.28 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-ofhsetopcrvkecvl) |
| 08:58.45 | Rhomber | I imagine transferring after a transfer has a simmilar problem |
| 08:58.55 | *** join/#asterisk Neptu (~Hej@mail.avtech.aero) |
| 08:59.56 | andross | why are you using feature codes anyway |
| 09:00.10 | ChannelZ | Not necessarily, because when you transfer to another extension, that extension is typically doing a Dial to the actual device at which point you can include options |
| 09:00.38 | ChannelZ | He has some people on mobiles apparently |
| 09:00.48 | andross | oh that makes sense |
| 09:01.15 | ChannelZ | It's an unfortunate limitation |
| 09:01.29 | ChannelZ | of being old-tech :) |
| 09:01.50 | andross | well they can repark the call and hand off that way i guess |
| 09:01.59 | andross | ohnvm |
| 09:02.01 | andross | yeah |
| 09:02.02 | andross | thats lame |
| 09:02.32 | ChannelZ | Yeah. If someone picks up a parked call who only has access to feature codes, they're stuck AFAIK |
| 09:03.36 | ChannelZ | Unless you perhaps hotwire it all to hell yourself, by making your own pickup extensions.. that say Dial a Local chan that then does the Pickup? Dunno if that would work |
| 09:04.24 | ChannelZ | in any case it's way the hell past my bedtime |
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| 09:13.53 | andross | i have a bad solution |
| 09:14.12 | andross | rather than parking the call |
| 09:14.30 | andross | well this has to be done from not a cell phone |
| 09:14.59 | andross | um |
| 09:15.16 | andross | you can conference in the cell phone and then put the call on hold |
| 09:16.13 | andross | then when the cell user wants to transfer the call they can hangup and call the extension that initiated the conference call |
| 09:16.16 | andross | and ask them to transfer |
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| 09:28.11 | *** join/#asterisk EmleyMoor (phil@topdeck.tinsleyviaduct.com) |
| 09:29.22 | EmleyMoor | Is there an efficient way to test a variable against a list of 61 regexps, returning true if it matches any one? |
| 09:29.35 | EmleyMoor | (in the dialplan) |
| 09:30.53 | EmleyMoor | (and it will never match more than one) |
| 09:36.50 | *** join/#asterisk bpriddy (~bpriddy@ipv4.host.stabbyspazzout.net) |
| 09:44.08 | Gugge | EmleyMoor: i would think the best way is to rewrite it to one regexp :) |
| 09:44.41 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 09:45.13 | eirirs | yeah, it should be possible |
| 09:45.33 | EmleyMoor | Gugge: It can't be done. The 61 regexps are as simple as it can get without being slack |
| 09:46.02 | EmleyMoor | eirirs: Hmmm... obviously it would be a very complicated regexp |
| 09:46.19 | EmleyMoor | (I am not at all sure how to write such an expression) |
| 09:46.28 | eirirs | EmleyMoor: like in mathematics, you gets a very long line, and can shorten it up alot, you would be surprised |
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| 09:47.56 | EmleyMoor | eirsrs: Most obvious thing is that they all begin "01"... |
| 09:48.23 | eirirs | :) |
| 09:49.08 | eirirs | EmleyMoor: karnaugh map can help alot :D |
| 09:49.53 | *** join/#asterisk Mission-Critical (~MissionCr@unaffiliated/missioncritical) |
| 09:50.45 | v0lZy | u could run it through a shell script. |
| 09:51.14 | v0lZy | or through the dialplan |
| 09:51.45 | v0lZy | define the strings as extensions or something maybe, and have them 'goto' certain point if they match. |
| 09:52.21 | andross | what are you trying to match |
| 09:52.24 | v0lZy | that way asterisk will match them or wont match them just as any other extensions... |
| 09:53.01 | EmleyMoor | andross: first n digits of caller ID match one of 61 known patterns |
| 09:55.35 | v0lZy | eh, just put them down in the dialplan |
| 09:55.36 | EmleyMoor | v0lZy: The need to continue on a "no match" makes that idea complicated |
| 09:55.38 | andross | so international calling? |
| 09:55.46 | v0lZy | and 'goto' . |
| 09:55.51 | EmleyMoor | andross: No, this is inland validation |
| 09:55.52 | v0lZy | EmleyMoor: does it? |
| 09:56.50 | *** join/#asterisk bogrd__ (~asteriskp@115.119.115.26) |
| 09:56.53 | v0lZy | _XXXX123,1,goto(quitifmatch) |
| 09:57.21 | v0lZy | asterisk matches more specific patterns first |
| 09:57.44 | v0lZy | so just put a general pattern below these and anything that doesnt hit those 61 patterns will get matched by that |
| 09:57.51 | EmleyMoor | v0lZy: Yes - but that doesn't help on a "no match" in this case |
| 09:59.24 | kaldemar | EmleyMoor: _X. will handle "no match". |
| 09:59.47 | EmleyMoor | I need "no match" effectively unhandled. |
| 09:59.58 | kaldemar | has a hard time beleiving that 61 different regexp's are needed |
| 10:00.00 | EmleyMoor | (not "dropped", "left to run") |
| 10:00.22 | kaldemar | it the call gets to dialplan, it is handled already. |
| 10:00.28 | EmleyMoor | kaldemar: One would do it, if it weren't for these 40 problem areas. |
| 10:01.10 | v0lZy | EmlyMoor.... u are not getting my point, im saying u convert a no match into a match by saying 'anything that matches the last pattern' |
| 10:01.25 | v0lZy | since the last pattern is less specific then the ones above it, whatever doesnt match the above 61 patterns will match it |
| 10:01.48 | EmleyMoor | This sounds like a two week job... |
| 10:02.02 | v0lZy | ... |
| 10:02.06 | EmleyMoor | I can trap all that do *not* need these in one shot anyway... |
| 10:02.31 | v0lZy | In any kind of logic, you just need to diferentiate 2 states, not matching something is the same as 'matching something else thats non of the above' |
| 10:02.33 | andross | what is inland validation? an incoming CID whitelist |
| 10:03.08 | v0lZy | and if u need to then handle it again, send it off to another context or something |
| 10:03.19 | v0lZy | and use extensions there to match again |
| 10:03.21 | EmleyMoor | andross: Not "white"... a whitelist is easy. This is a "void-and-doesn't-need-listing" detector. |
| 10:05.10 | *** join/#asterisk nunne (~nunne@static-213-115-116-75.sme.bredbandsbolaget.se) |
| 10:06.30 | *** join/#asterisk Rumbles (~jstocker@mail.solutiontelecom.co.uk) |
| 10:07.01 | EmleyMoor | Damn, lost my list - going to have to rebuild it... I think I may have found a way... |
| 10:07.34 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 10:07.50 | andross | are you using astdb |
| 10:08.28 | EmleyMoor | andross: I am, but trying to avoid doing so where obviously not valid. |
| 10:08.43 | v0lZy | EmleyMoor: What exactly do u need to achieve though? |
| 10:09.19 | EmleyMoor | v0lZy: For "these 61 cases", total caller ID length must be 10. |
| 10:09.20 | eirirs | avvoid one of those 61 lovers ringing when wife are present, I guess |
| 10:09.45 | v0lZy | and if its not 10 ? |
| 10:10.10 | EmleyMoor | Not 10 and one of these cases, send to void caller ID handling routine |
| 10:11.21 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 10:11.58 | v0lZy | so why not do it like this |
| 10:13.13 | v0lZy | Set these 61 patterns with pattern,1,GotoIf($[extensionlength =< 10]tag1:tag2) |
| 10:13.50 | v0lZy | (just a sketch) |
| 10:14.01 | EmleyMoor | v0lZy: I'd need to be going to a pattern in the first place, of course. It puts a "dogleg" in... |
| 10:14.42 | v0lZy | EmleyMoor: you can use gosub and pass the pattern as a variable etc... |
| 10:14.48 | v0lZy | erm.. i mean the extension |
| 10:15.49 | v0lZy | just type in 61 patterns the way i wrote above, then define 2 tags where tag1 will be launched if its less then or equal to 10 ... change to = 10 if u need it to be exactly 10, and tag2 will go if the statment is false |
| 10:15.58 | v0lZy | then under those tags, do whatever processing you want. |
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| 10:17.07 | v0lZy | GotoIf($[evaluation condition]?whentrue:whenfalse) |
| 10:17.55 | v0lZy | whentrue, simply continue... with the dialplan, ${EXTEN} will pickup the original number that was called |
| 10:18.44 | v0lZy | whenfalse, hangup or whatever u plan to do |
| 10:20.02 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 10:20.18 | bogrd__ | hello, I am working on asterisk realtime. I need to store the voicemail user details in one table and voicemessages in another table in MySQL. I already have realtime SIP users and extensions working.Can anybody please let me know how to configure this for realtime ? |
| 10:21.54 | *** join/#asterisk areski (~areski@81.184.35.151.dyn.user.ono.com) |
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| 10:23.37 | wdoekes | bogrd__: for starters: recompile asterisk with voicemail odbc support |
| 10:24.10 | bogrd__ | wdoekes: I have done that |
| 10:24.39 | wdoekes | configure extconfig |
| 10:24.59 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 10:25.06 | bogrd__ | the problem is in extconfig.conf, for voicemail family, I can specify odbc,<database>,<tablename> |
| 10:25.17 | bogrd__ | but I have two tables now for voicemail |
| 10:25.34 | wdoekes | user details go in extconfig |
| 10:25.44 | wdoekes | voicemessages goes into voicemail.conf |
| 10:26.27 | bogrd__ | oh ok .. what is the configuration in voicemail.conf to specify voicemessages ? |
| 10:26.31 | wdoekes | ; The default table for ODBC voicemail storage is voicemessages. |
| 10:26.31 | wdoekes | ;odbctable=voicemessages |
| 10:26.42 | wdoekes | the one that says so |
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| 10:27.11 | *** join/#asterisk v0lZy (~Thunderbi@mail.silk-group.net) |
| 10:27.31 | bogrd__ | oh ok ..please check if this works good |
| 10:28.52 | bogrd__ | in extconfig.conf, voicemail => odbc,<dbname>,<voicemailusers> and in voicemail.conf , odbcstorage=<dbname> and odbctable=<voicemessages> |
| 10:30.28 | bogrd__ | wdoekes: this is fine right ? ^^ |
| 10:30.42 | wdoekes | how about |
| 10:30.46 | wdoekes | you check it |
| 10:30.51 | wdoekes | and come back if it doesn't work |
| 10:31.01 | bogrd__ | wdoekes: sure will do it now |
| 10:31.04 | wdoekes | :) |
| 10:31.17 | bogrd__ | thank you |
| 10:40.14 | wizard3001 | Hello. Can some one tell me how to do Inband-DTMF detection on asterisk. Because now my asterisk server it's not detecting it. Sorry for the stupid question |
| 10:44.40 | andross | to set DTMF for a peer in sip.conf ? |
| 10:45.05 | andross | http://www.voip-info.org/wiki/view/Asterisk+sip+dtmfmode |
| 10:45.12 | andross | you can use auto |
| 10:45.37 | andross | although its better to find what works and hardcode it |
| 10:46.03 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.40.105) |
| 10:46.44 | andross | also depends on where your dtmf issue is |
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| 11:03.00 | wizard3001 | What i realy need is when asterisk recives a call whit Inband-DTMF to send it to the other client as RFC2833 or Info |
| 11:05.01 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 11:15.11 | bogrd__ | wdoekes: I am using asterisk 11.2.1 . Since there is no support for realtime in ConfBridge, can I resort to meetme for realtime ? Because the documentation says meetme is deprecated over ConfBridge so wanted to clarify .. |
| 11:16.26 | wdoekes | what do you mean there is no realtime support? support for what? you can open rooms by hand (combine with func_odbc) |
| 11:17.30 | *** join/#asterisk x1user (~chatzilla@212.36.13.6) |
| 11:17.57 | bogrd__ | wdoekes: I ran into this, please check .. http://forums.asterisk.org/viewtopic.php?p=183452 |
| 11:18.25 | wdoekes | you don't need to store the profiles, you can alter them by hand |
| 11:18.40 | wdoekes | Set(CONFBRIDGE(user,admin)=${IF(${l_is_admin}?yes:no)}) |
| 11:19.03 | wdoekes | possibly based off a default profile from confbridge.conf |
| 11:19.09 | x1user | How to catch DTMF tome after Dial() application. I want to record the call if specific DTMF tone is send after the call ? |
| 11:19.16 | wdoekes | s/profile/template/ |
| 11:19.35 | wdoekes | gags infobot |
| 11:20.17 | bogrd__ | If I need something like meetme table like here http://www.voip-info.org/wiki/view/Asterisk+RealTime+MeetMe |
| 11:20.42 | bogrd__ | is this available for confbridge too ? |
| 11:20.53 | wdoekes | did I mention "by hand"? |
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| 11:21.32 | bogrd__ | hmm .. ok |
| 11:21.51 | wdoekes | Read(pin,...), compare with func_odbc gotten pin |
| 11:22.46 | wdoekes | fetch conference_id, put it in ConfBridge() and start conferencing |
| 11:22.46 | kaldemar | x1user: see automon and automixmon in features.conf |
| 11:24.06 | bogrd__ | wdoekes: ok sure .. thanks |
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| 11:39.14 | danfromuk | Hi, is there a good free opensource speech recognition addon for asterisk? |
| 11:40.16 | danfromuk | Or a reasonably priced addon. |
| 11:42.24 | Tim_Toady | http://zaf.github.com/asterisk-speech-recog/ something 'free as in beer' to experiment, not for production use tho |
| 11:43.10 | _methods | it works pretty well |
| 11:44.04 | Tim_Toady | indeed, but its not an official google service, and there is always the risk of google disabling access to it |
| 11:47.14 | danfromuk | How does it go with non-dictionary words in a production environment of around 10 concurrent recordings? |
| 11:47.28 | danfromuk | non-dictionary like 'Tesco' or 'Microsoft' |
| 11:48.30 | Tim_Toady | 10 concurent recordings wont be much of an issue, the recognision is done remotely, data is send to google |
| 11:48.57 | Tim_Toady | as for non dictionary words... you have to try this |
| 11:49.23 | Tim_Toady | usually works reasonably well, but YMMV |
| 11:49.27 | danfromuk | I need to be able to provide the speech recognition software with a list of about 100 company names, and the caller chooses using speech. I assume google is only able to process well know words and trademarks etc. |
| 11:50.40 | _methods | i just started using it myself so i've only done minor testiing with it |
| 11:51.20 | _methods | there is a sample program that will take in your speech and read it back to you |
| 11:51.52 | _methods | you could probably start with that and test all the company names |
| 11:53.37 | danfromuk | Looks good, but as Tim_Toady said, it may be a problem if google change or close their service. |
| 11:54.25 | _methods | google may close it but they typically open source their stuff after so someone will probably pick up the ball |
| 11:54.38 | _methods | still a gamble though |
| 11:55.38 | Tim_Toady | a small one but a gamle, it actually mimics chrome browser and its speech recognition feature, so if google disables it all current and past chrome versions wont function |
| 11:56.03 | Tim_Toady | not impossible but not very likely either |
| 11:56.09 | _methods | is setting it up in his phone system right now actually |
| 11:56.38 | _methods | i'm having some problems with the interrupt keys |
| 11:56.50 | Rhomber | kinda sounds like google is a spy agency to be honest. lol |
| 11:57.01 | Rhomber | you send your voice data to google and they send you back text? |
| 11:57.05 | Tim_Toady | like we didnt know that already :P |
| 11:57.07 | Rhomber | jesus.. haha |
| 11:57.11 | Rhomber | yeah, exactly |
| 11:57.11 | file | it's a method of building a speech recognition model |
| 11:57.18 | file | crowd sourcing, as it were |
| 11:57.32 | _methods | yes |
| 11:57.34 | Rhomber | i guess, but still |
| 11:57.40 | _methods | emergent intelligence almost |
| 11:58.01 | danfromuk | Rhomber: Apple banned staff from using Android phones because they were worried that google would listen in via speech recognition. |
| 11:58.06 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 11:58.12 | _methods | hahah |
| 11:58.22 | _methods | no wonder their stock has shit the bed |
| 11:58.27 | Rhomber | lol, that sounds reasonable to be honest |
| 11:58.47 | _methods | yeah everything apple does is reasonable |
| 11:58.49 | _methods | hahah |
| 12:00.51 | Rhomber | lol, now that's not true |
| 12:01.17 | _methods | what that apple is a reasonable company or that their stock has gone to shit lol |
| 12:01.27 | _methods | 1 of those statements is true |
| 12:01.59 | Rhomber | lol, i'm not a fan of apple.. so i'll refrain from answering |
| 12:02.05 | Rhomber | also not a fan of google any more |
| 12:02.11 | Rhomber | their 'do no harm' policy is long dead |
| 12:02.21 | _methods | yes |
| 12:02.21 | Rhomber | anyone played with queue member pausing before? |
| 12:02.50 | *** part/#asterisk mjordan (~mjordan@75.76.55.191) |
| 12:03.18 | Rhomber | just wondering whether unpausing happens eventually.. or if it's something i have to implement? |
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| 12:09.26 | EmleyMoor | Is there a simple opposite of a : operator? |
| 12:11.42 | kaldemar | EmleyMoor: do you mean a negation? |
| 12:12.15 | EmleyMoor | Negation of expr1 : regexp - worked around it now anyway |
| 12:12.22 | beefcafe | the voice recognition agi isn't working for me |
| 12:12.25 | beefcafe | AGI Script speech-recog.agi completed, returning 0 |
| 12:15.33 | kaldemar | EmleyMoor: ! |
| 12:17.49 | Rhomber | any suggestions for a VOIP softphone client for windows? |
| 12:17.51 | EmleyMoor | kaldemar: yes - thought of ! the whole thing - just wondered if you could say !: |
| 12:17.56 | Rhomber | the ones i've tried so far are shit |
| 12:18.01 | EmleyMoor | Rhomber: ekiga? |
| 12:18.19 | ghost75 | zoiper |
| 12:18.22 | Rhomber | my friend said it was crap, didn't try it personally.. |
| 12:18.28 | Rhomber | zoiper.. will have a look :) |
| 12:18.43 | Rhomber | Jitsu seemed good at the start, but it's got so many bugs |
| 12:19.47 | Rhomber | ghost75: looks good, thanks ;) |
| 12:20.03 | kaldemar | EmleyMoor: no, you'd need to negate the whole expression. |
| 12:22.08 | _methods | i use microsip |
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| 12:35.02 | Rhomber | zoiper works great :) |
| 12:35.09 | Rhomber | once i got past AVG thinking it was a trojan |
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| 12:44.37 | Rhomber | i set announce-frequency = 20 and periodic-announce-frequency = 50 .. and it plays them both at the same time.. lame |
| 12:45.30 | [TK]D-Fender | 20s for annnounce? May as well be continuous |
| 12:46.14 | Rhomber | just for testing atm |
| 12:46.29 | Rhomber | trying to figure it all out/how things play together |
| 12:48.29 | *** join/#asterisk davlefouAMD (~david@41.225.181.42) |
| 12:49.07 | Rhomber | meh, i just disabled the announce-frequency and set periodic-announce-frequency to 120 |
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| 13:06.57 | Katty | hello my asterisk does not work at all how to fix plz, is urgent thx |
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| 13:17.07 | EmleyMoor | Katty: What does it do? How do you know it doesn't work? |
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| 13:30.54 | leifmadsen | Katty: step 1, unplug all the devies |
| 13:30.57 | leifmadsen | devices* |
| 13:31.05 | leifmadsen | Katty: step 3, profit |
| 13:31.41 | eirirs | you skipped step2 |
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| 13:34.41 | WIMPy | eirirs: Trade secrets |
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| 13:38.02 | zamba | we have two analoge adapters.. one is for connecting phones and the other is for connecting to the pstn.. which is which? :) |
| 13:38.12 | zamba | one has ports named FXO, the other has ports named FXS |
| 13:38.17 | zamba | which goes to the pstn? |
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| 13:43.57 | aakatim | Hi, I'm trying to set a proper diversion header and everything works except the "counter" field. In the IETF document it is written that it should look like this: diversion-counter = "counter" "=" 1*2DIGIT |
| 13:43.57 | aakatim | Please can somebody decode what that means, how to make the counter equal to 4. Thanks |
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| 13:49.10 | zamba | talk about creating a stupid and overly complex gui - AudioCodes |
| 13:54.36 | kaldemar | zamba: phones are connected to FXS, lines to FXO. |
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| 14:03.17 | WIMPy | What's the latest Linux supported by DAHDI? |
| 14:03.23 | zamba | kaldemar: ok |
| 14:03.38 | zamba | can someone please explain to me how you get a audiocodes registered with asterisk? |
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| 14:27.43 | aakatim | does anybody know how to interpret those 2DIGIT field into the Diversion header |
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| 14:32.53 | igcewieling1 | aakatim: example? |
| 14:34.53 | aakatim | igcewieeling1: I'm trying to set a proper diversion header and everything works except the "counter" field. In the IETF document it is written that it should look like this: diversion-counter = "counter" "=" 1*2DIGIT |
| 14:35.11 | aakatim | I make it like that: counter=4 |
| 14:35.34 | aakatim | as it is described in most of the documents, but it is working |
| 14:35.49 | aakatim | but it it not working |
| 14:37.35 | igcewieling1 | aakatim: I never noticed that in the Diversion's we get. I assume it means "How many times was this diverted", but I don't actually know. |
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| 14:40.23 | aakatim | igcewieling1: yes, correct |
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| 14:41.14 | *** join/#asterisk polysics (~Adium@host67-201-dynamic.21-79-r.retail.telecomitalia.it) |
| 14:41.18 | polysics | hi there |
| 14:41.38 | polysics | does anyone know where I can find some precise info on AGI command size limits? |
| 14:41.54 | polysics | there is a 1024 byte number floating around but I can't seem to find a ref to it |
| 14:42.47 | polysics | specifically, I would like to know a size limit for SSML/GRXML grammars when doing TT/ASR |
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| 14:47.19 | igcewieling1 | polysics: easy enough to test with a very long AGI EXEC VERBOSE or something like that. |
| 14:48.29 | polysics | igcewieling1: that will give me empirical evidence which is great, but is that documented anywhere? |
| 14:49.06 | igcewieling1 | polysics: not that I know of. The max length changed in different versions of Asterisk. |
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| 14:49.46 | igcewieling1 | in 1.2 is was something like 1024 chars, but I'm pretty sure in 1.4 and later it was changed to be a dynamic size, your AGI library may have a limit as well. |
| 14:51.18 | polysics | I guess looking in the source won't help much |
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| 14:56.00 | MrMeek | i've got a bunch of binary files in /tmp named like core.ivrAsterisk-2013-03-12T17:14:57-0400 |
| 14:56.08 | MrMeek | can anyone tell me what they are? :) |
| 14:57.56 | MrMeek | core dumps generated when an asterisk exception occurs? |
| 14:58.00 | Katty | HAI MISTER MEEK |
| 14:58.04 | Katty | are you really meek? |
| 14:58.14 | MrMeek | Hehe, good question |
| 14:58.22 | MrMeek | I am often called meek by my friends ;) |
| 14:58.25 | Katty | well... i have my doubts ^_^ |
| 14:58.35 | MrMeek | but i use the alias because it's an anagram for my name |
| 14:58.51 | Katty | 's coo |
| 14:58.57 | Katty | and unique! i likes it. |
| 14:59.02 | Katty | have an upboat from me. |
| 14:59.05 | MrMeek | Mister Meek = Mike Mertes |
| 14:59.05 | MrMeek | hehe |
| 14:59.14 | Katty | you shouldn't share your real name on the internets! |
| 14:59.19 | MrMeek | oh right :P |
| 14:59.27 | MrMeek | i'm sure there's a bunch of filthy people here just waiting to look me up |
| 14:59.28 | Katty | you don't know if i'm a creepy stalker or not! |
| 14:59.35 | MrMeek | lol ;) |
| 15:00.03 | Katty | i hear Qwell is a stalker. |
| 15:00.07 | MrMeek | lol |
| 15:00.12 | MrMeek | katty you a fan of the japanese? ;) |
| 15:00.14 | Katty | Qwell: stalking the coffee shop. |
| 15:00.29 | Katty | MrMeek: wellllllllllll that's a complicated question. |
| 15:00.33 | MrMeek | <Katty> HAI <- |
| 15:00.37 | Katty | MrMeek: i wouldn't say i hate or dislike the Japanese. |
| 15:00.46 | Katty | MrMeek: but i wouldn't say i'm a Japanese FanGirl either. |
| 15:01.25 | Katty | MrMeek: they exist. they do some pretty amazing stuff. their culture is pretty cutesy. |
| 15:01.31 | MrMeek | yeah right! Everyone knows that saying things like, 'hai,' and 'ja matte, ne!' makes you a weeaboo |
| 15:01.32 | MrMeek | lol |
| 15:01.49 | Katty | i'm afeared you've lost me. |
| 15:02.11 | SuperNull | if i want a call to not be cdr accounted.. what is the command ? |
| 15:02.12 | Katty | imma have to google ja matte, ne |
| 15:02.27 | Katty | hi SuperNull |
| 15:02.37 | MrMeek | ja mata* |
| 15:02.42 | MrMeek | need some typo pills |
| 15:03.16 | Katty | TIL how to say see you later in Japanese! |
| 15:03.29 | *** join/#asterisk nricciar (~nricciar@64.207.62.66) |
| 15:03.33 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 15:03.37 | Katty | and here i thought it was just Konnichiwa! |
| 15:03.43 | MrMeek | truth =) |
| 15:03.54 | SuperNull | whats up |
| 15:03.54 | MrMeek | and for me |
| 15:03.55 | MrMeek | TIL TIL |
| 15:03.56 | MrMeek | lol |
| 15:04.00 | MrMeek | had to look that one up :P |
| 15:04.08 | Katty | woot! we're gettin smarter by the minute! |
| 15:04.11 | MrMeek | <3 |
| 15:04.18 | Katty | SuperNull: learnin, n stuff. |
| 15:04.26 | SuperNull | damn thats sexy. |
| 15:04.31 | MrMeek | <- writing ivrs |
| 15:04.47 | Katty | i guess i'm just stalking my ticket list. waiting for updates. |
| 15:04.48 | SuperNull | what ya using to do that MrMeek ? i have yet to play with IVR |
| 15:04.50 | Katty | waiting for something to break. |
| 15:04.55 | SuperNull | heh. |
| 15:05.04 | Katty | the usual. |
| 15:05.06 | SuperNull | kinda similar here.. only getting my cdr stuff in order. |
| 15:05.09 | MrMeek | scripting up some dialplan logic |
| 15:05.12 | MrMeek | nothing extraordinary |
| 15:05.24 | Katty | PRESS 1 FOR MONKEY MUSICS |
| 15:05.34 | MrMeek | lol katty |
| 15:05.36 | SuperNull | press 2 to hear the lion roar |
| 15:05.44 | Katty | Please hold for precisely 13 seconds and you will be transfered to the Cisco Support Black Hole. |
| 15:05.53 | SuperNull | Cxtech or something like that.. a cable maker has options like that |
| 15:06.08 | SuperNull | Cisco ? ;) |
| 15:06.12 | SuperNull | your handsets? |
| 15:06.14 | Katty | I always found it easier to IVRs backwards. |
| 15:06.18 | jmetro | dat cisco call manager default music |
| 15:06.31 | Katty | jmetro: you laugh, but i have an extension with cisco on hold music. |
| 15:06.38 | Katty | jmetro: for when i need to get Work Done |
| 15:06.59 | Katty | jmetro: people will go away when i am on the phone and Busy |
| 15:07.13 | Katty | SuperNull: Cisco makes far more than just phone bits. |
| 15:09.49 | jmetro | Cisco also makes very... interesting CLI's =) |
| 15:13.32 | *** join/#asterisk camerin (hoax@newelite2.bshellz.net) |
| 15:14.09 | MrMeek | Anyone have experience with googletts? |
| 15:14.18 | MrMeek | I'm wondering what the voice quality is like compared to cepstral |
| 15:14.30 | MrMeek | i'm assuming it's somewhere between festival and cepstral |
| 15:17.36 | jmetro | def. better than festival |
| 15:17.47 | jmetro | but i heard they will ban your requests if you do too many |
| 15:17.53 | MrMeek | yeah festival is fugly |
| 15:18.04 | MrMeek | i actually had a client tell me, "I could shit into a handset and it would sound better than this" |
| 15:18.05 | MrMeek | rofl |
| 15:18.25 | MrMeek | jmetro, can cache the prompts to disk for anything that's not dynamic, right? |
| 15:18.39 | jmetro | probably, havent tried. |
| 15:18.43 | MrMeek | nods |
| 15:18.46 | MrMeek | i'll do some more research later |
| 15:18.52 | jmetro | though for static stuff, why not just record yourself or someone |
| 15:23.30 | Qwell | Katty: totally |
| 15:23.40 | Katty | hugs Qwell |
| 15:24.06 | Qwell | Katty: I'm not sure how cheeky I can be, before crossing the sexual harassment line, so I'm just going to say nothing. |
| 15:24.33 | Katty | Qwell: to me, or the coffee shop? |
| 15:24.59 | Qwell | Katty: in re; the coffee shop |
| 15:25.34 | Katty | righto. |
| 15:28.55 | MrMeek | jmetro: there is a few dynamic fields, we don't want to mix live voice with tts |
| 15:30.03 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 15:30.04 | jmetro | MrMeek: Why not? Sounds better than full TTS any day.Id rather hear "To talk to EXTENSION 1 2 7 " than "To ta-lk ta ex-ten-sheeun one, two, seven" |
| 15:30.06 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 15:32.13 | MrMeek | i should say, the client doesn't want to do that :) |
| 15:32.17 | MrMeek | i could care less personally hehe |
| 15:32.27 | MrMeek | i may suggest it, none-the-less |
| 15:32.48 | *** join/#asterisk vlad_starkov (~vlad_star@178.176.148.64) |
| 15:44.08 | *** join/#asterisk pietro (~pietro@78.134.12.196) |
| 15:45.34 | *** join/#asterisk pietro (~pietro@78.134.12.196) |
| 15:45.37 | pietro | hello |
| 15:46.10 | pietro | I'm searching for a tool that can generate graphic SIP call flow from a .pcap file. |
| 15:46.13 | pietro | any suggestion ? |
| 15:46.28 | *** join/#asterisk gonewage (~gonewage@host-72-2-130-205.csinet.net) |
| 15:48.07 | navaismo | wireshark |
| 15:49.34 | slav3_kitten | http://cispapetition.org/ |
| 15:49.43 | slav3_kitten | http://cispapetition.org/ |
| 15:50.09 | *** part/#asterisk gonewage (~gonewage@host-72-2-130-205.csinet.net) |
| 15:50.31 | *** join/#asterisk Rumbles (~jstocker@mail.solutiontelecom.co.uk) |
| 15:50.49 | jmetro | I thought we did the whole CISPA thing already |
| 15:58.30 | _Corey_ | Anyone else noticing Vitelity being down...? |
| 15:58.55 | Qwell | wonders if _Corey_ feels like being a guinea pig. |
| 15:59.44 | _Corey_ | Sadly, I feel like I've crossed that line a long time ago... |
| 16:01.51 | igcewieling1 | my outbound calls from vitelity work |
| 16:02.20 | _Corey_ | Their portal is timing out... seems pretty spotty on inbound |
| 16:06.43 | *** join/#asterisk camerin (hoax@newelite2.bshellz.net) |
| 16:11.38 | *** join/#asterisk caveat- (hoax@gateway/shell/bshellz.net/x-dcvnkscblxheckfq) |
| 16:11.42 | *** join/#asterisk vlad_sta_ (~vlad_star@178.176.148.64) |
| 16:13.17 | *** join/#asterisk ruben231 (~OpenDial@112.198.90.245) |
| 16:28.39 | *** join/#asterisk UnixDev (~UnixDev@unaffiliated/unixdev) |
| 16:33.42 | niklaswe | hello, any Idé ho |
| 16:34.19 | Qwell | What I do with my IDE in the privacy of my office is nobody's business but mine. |
| 16:36.15 | niklaswe | how I should fix this problem.. I using confbrdige with videosupport and it works great, but the problem I got is when more then 2 people are in the same conference, then webcam doesnt work. it says Waiting for remote video.. |
| 16:36.40 | niklaswe | but if one off them leave so it only 2 people in the room it starts working again. |
| 16:37.06 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:43.13 | ruben231 | hi guys i have a hosted asterisk server ad when calls are stablish in volume and i switch codec from ulaw to g729 codec immediately the audio quality degrade---using voip |
| 16:45.52 | Qwell | umm, yes. g729 is lower quality than g711. |
| 16:46.28 | jmetro | yep |
| 16:48.04 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 16:48.04 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 16:49.25 | jmetro | http://www.voipforo.com/en/codec/codecs.php looking at "mean opinion score" |
| 16:55.53 | *** join/#asterisk salz212 (~chatzilla@182.185.149.245) |
| 16:57.14 | navaismo | Any hints about terminated calls with warning: Re-invite to non-existing call leg on other UA |
| 16:58.01 | jmetro | oh that sounds fun |
| 16:58.17 | Katty | i need a soda. |
| 16:58.22 | Katty | who else wants a soda. |
| 16:58.26 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 16:58.29 | Katty | on the house. |
| 16:59.39 | *** join/#asterisk bmg505 (~leon@196-209-120-57.dynamic.isadsl.co.za) |
| 17:05.53 | Katty | mmm bubbly cold drink of poison. why do you have to be so bad for me. |
| 17:06.01 | *** join/#asterisk as001 (~uros@82.117.198.142) |
| 17:06.30 | jmetro | its only bad for you if its anything but Dr Pepper or A&W Root beer. |
| 17:06.40 | ruben231 | Qwell: my main problem is this users are on homebase, in the philippines and the asterisk server is in the US..how do i allocate properly to maintained good quality i have like 10TB bandwdith from hosting datacenter |
| 17:07.09 | leifmadsen | unless you control the entire circuit point to point, you can't |
| 17:07.36 | as001 | Can Asterisk 11.2.1 comunicate with Google Chrome with webrtc without webrtc2sip package ? |
| 17:07.54 | Katty | i've been looking for a sip phone for chromebook. |
| 17:07.58 | Katty | never found one. |
| 17:08.05 | Katty | cept the proprietary stuffs. |
| 17:08.08 | leifmadsen | Katty: that uses chromeOS? |
| 17:08.13 | Katty | yes'r |
| 17:08.26 | Katty | the proper chromeOS, not the downloaded lite version. |
| 17:08.46 | leifmadsen | gotcha... hmmm... feel like there are supposed to be sip phones for webrtc |
| 17:08.56 | leifmadsen | which I assume work on chromeOS |
| 17:09.00 | ruben231 | leifmadsen:but any option to make calls mores realiable somehow, coz tehre are many hosted asterisk being offered, how did they make this work somehow |
| 17:09.09 | Katty | leifmadsen: i just browsed through the Google App store. |
| 17:09.14 | leifmadsen | in my case, we have customers setup MPLS networks |
| 17:09.17 | Qwell | ruben231: They have staff that knows what they're doing. |
| 17:09.31 | leifmadsen | unless their network has been assessed to support voip traffic across the internet |
| 17:09.39 | leifmadsen | we also install data centres outside the US |
| 17:09.46 | leifmadsen | and what Qwell said |
| 17:10.00 | Katty | doesn't know enough about APIs to do antyhign custom. |
| 17:10.12 | leifmadsen | Katty: no, I figured stuff would be built already |
| 17:10.27 | leifmadsen | like aren't there examples for webrtc stuff working with asterisk and such? |
| 17:10.34 | leifmadsen | hasn't dug into that area yet |
| 17:10.59 | as001 | I tried with dubango sipml5 browser phone demo I can register to Asterisk box but when I call there are no audio. I failed to install that webrtc2sip package. |
| 17:11.12 | Katty | leifmadsen: i'll go see what i can dig up |
| 17:11.45 | ruben231 | <PROTECTED> |
| 17:12.00 | leifmadsen | I doubt they did |
| 17:12.04 | leifmadsen | MPLS is expensive |
| 17:13.06 | *** join/#asterisk Neptu (~Hej@c213-89-2-159.bredband.comhem.se) |
| 17:13.43 | igcewieling1 | as001: sounds like a standard NAT problem. |
| 17:14.15 | robert_ | so RudyValencia and I added a SIP trunk and when I try making outbound calls, it scrolls on and on about how it can't convert G22 to G29; "Codec mismatch on channel SIP/2000-00000005 setting write format to g729 from g722 native formats (g722)" |
| 17:14.39 | robert_ | er, G72*, G79* |
| 17:14.43 | leifmadsen | robert_: do you have a g729 license and codec installed? |
| 17:14.50 | as001 | igcewieling1: I can see call reach Asterisk Box, I can see Playback application on CLI I also can see RTP packets but no audio. |
| 17:15.18 | navaismo | as001, how is configured your webrtc peer |
| 17:15.36 | leifmadsen | ok, off to do some real work, good luck to those in the know |
| 17:15.37 | *** part/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:16.11 | as001 | I have avpf=yes encryption=yes and nat=force_rport,comedia transport=ws,wss |
| 17:16.29 | igcewieling1 | as001: I'm not hearing anything yet. which would indicate it is not a nat issue. |
| 17:16.48 | igcewieling1 | as001: do you have directmedia=no ? is the Asterisk SERVER behind NAT or just the phone? |
| 17:17.13 | as001 | phone/broswer is behind nat Asterisk is not behind NAT |
| 17:17.22 | igcewieling1 | oh, sorry. rtc. silly me. I can't help with that since it doesn't use SIP. |
| 17:17.26 | as001 | I will put directmedia=no now |
| 17:17.34 | Qwell | igcewieling1: It is using SIP. |
| 17:17.39 | robert_ | no, I don't; I don't want to use G729, however. |
| 17:17.52 | igcewieling1 | Qwell: ah. |
| 17:18.15 | igcewieling1 | robert_: stop allowing g729 |
| 17:18.41 | igcewieling1 | disallow all codecs and ONLY allow the few codecs you want to use, this is just standard practice. |
| 17:19.12 | as001 | same result with directmedia=no. |
| 17:20.23 | navaismo | as001, if you change nat=yes(i know its deprecated) |
| 17:20.33 | igcewieling1 | navaismo: thinking of the nat= bug? |
| 17:20.39 | navaismo | yep |
| 17:21.33 | as001 | same with nat=yes |
| 17:22.32 | navaismo | (stupid question) Did you choose the right media in the browser when it prompt to allow? |
| 17:23.58 | as001 | I choose Allow in browser. Here is pastebin of sip debug http://pastebin.ca/2331838 |
| 17:24.08 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:24.48 | as001 | Do you know what is "Enable RTCWeb Breaker" |
| 17:25.16 | file | you need to enable ICE support |
| 17:26.00 | file | icesupport=yes in rtp.conf, and also icesupport=yes in sip.conf for that specific account |
| 17:26.02 | salz212 | if the external callerid AND the registered extension are same AND having invite=insecure .. external callerID lands on registered user context.instead of its trunk or default context..how to get rid of this. scenario? |
| 17:26.07 | file | as001, that was for you |
| 17:26.23 | as001 | ok I will do it |
| 17:26.44 | as001 | thanks |
| 17:27.06 | robert_ | and we just recently switched to TLS and SRTP; however now Asterisk only ever says that the person at extension [ext. number] is unavailable and 'Asterisk -vvvvvvc' says 'Everyone is busy/congested at this time (1:0/0/1)' |
| 17:27.27 | navaismo | as001, but when it prompt the allow there is no menu to choose hardware? the breaker is for use with the webrtc2sip gateway |
| 17:28.01 | as001 | I will reload browser and tell. |
| 17:28.23 | igcewieling1 | robert_: what about "sip show peers" |
| 17:29.09 | navaismo | as001, in which linux version did you failed to install the webrtc2sip gateway? |
| 17:29.22 | robert_ | hm, that's weird; it says neither of us is using encryption, only I know our clients are. |
| 17:29.35 | robert_ | oh wait |
| 17:30.02 | as001 | I just have sipml wants to use your microphone and I click allow. I put icesupport=yes in rtp and in sip peer config but still nothing. |
| 17:30.30 | as001 | I failed to install it on Ubuntu 12.10 I got error during compiliing |
| 17:30.32 | file | ...I assume you realoded Asterisk |
| 17:30.35 | file | or reloaded |
| 17:30.37 | as001 | yes I did |
| 17:30.43 | file | show me a new SIP debug |
| 17:30.54 | as001 | ok |
| 17:32.18 | as001 | here http://pastebin.ca/2331843 |
| 17:32.29 | robert_ | igcewieling1: ignore the first part about not using encryption, lol; however, it says we are both indeed using TLS. |
| 17:32.44 | igcewieling1 | robert_: what is the status? |
| 17:33.02 | file | better, but I still don't know why RTP packets aren't flowing |
| 17:33.15 | navaismo | as001, any reason to install from source instead package? |
| 17:34.12 | as001 | hmm well I like to intall Asterisk from source so I can control what is turn on and turn off. I see Method not allowed 405 on SIP read from WS... |
| 17:34.53 | robert_ | igcewieling1: status of what? (sorry, I'm slow today.) |
| 17:35.15 | navaismo | as001, i mean the webrt2sip gateway |
| 17:35.15 | robert_ | igcewieling1: oh, sorry. both are OK. |
| 17:35.24 | igcewieling1 | sip show peers has both a Host and a status column |
| 17:35.46 | igcewieling1 | robert_: that is it for my ability to troubleshoot TLS. |
| 17:35.53 | robert_ | yeah; I just realized that. Both are OK. |
| 17:35.54 | as001 | yes I tried to install webrtc2sip from package but I failed beacuse of dependency problems and I tried to compile it but got error. |
| 17:37.48 | as001 | how can I check if my asterisk box is using srtp packages ? I see Sent RTP in cli not SRTP. |
| 17:38.14 | robert_ | igcewieling1: right, sorry. I have OK (38 ms) and he has OK (30 ms) |
| 17:38.46 | as001 | it is interesting because I have no open port on 37039 where asterisk is sending RTP packeges. |
| 17:41.14 | robert_ | igcewieling1: It says "Insecure : No" on both of ours. |
| 17:41.27 | igcewieling1 | robert_: I don't know anything more about TLS to help you with. |
| 17:41.40 | robert_ | I see. Thanks anyway. |
| 17:41.46 | igcewieling1 | I was just verifying the peers are reachable, since that seemed like that might be the issue. |
| 17:42.14 | robert_ | yeah, but it says that we're both busy. |
| 17:42.20 | robert_ | which is the weird thing. |
| 17:42.38 | robert_ | sorry, "congested". |
| 17:43.31 | salz212 | HI all... I have this issue need to find a solution : --> if the external callerid AND the registered extension are same AND having invite=insecure .. external callerID lands on registered user context.instead of its trunk or default context..how to get rid of this. scenario? |
| 17:44.56 | *** join/#asterisk DropsOfSerenity (~DropsOfSe@vpn.bctconsulting.com) |
| 17:49.29 | as001 | web socket connection is on port 47225 and RTP is going to 45184 module res_srtp.so is loaded. |
| 17:50.42 | *** join/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
| 18:07.52 | *** join/#asterisk blee (~blee@68.204.1.103) |
| 18:16.21 | *** join/#asterisk threesome (~threesome@ip-94-113-12-74.net.upcbroadband.cz) |
| 18:17.09 | *** join/#asterisk vlad_starkov (~vlad_star@178.176.168.22) |
| 18:17.45 | robert_ | yay! "Unknown or unavailable item requested: 'secure_signaling'" and "Unknown or unavailable item requested: 'secure_media'" |
| 18:22.29 | *** join/#asterisk b3pc (~bepc@108-234-48-184.lightspeed.chrlnc.sbcglobal.net) |
| 18:23.14 | robert_ | I thought setting secure_signaling=1 and secure_media=1 was the "approved" way to enable call security? |
| 18:23.19 | *** join/#asterisk nny (~Scott@mail.hhhealth.com) |
| 18:23.25 | nny | hmm |
| 18:23.58 | nny | if I am working with asterisk manager and nmap -p 5038 host (iptables disabled, port not changed) shouldn't I get an "open" in return ? |
| 18:24.23 | nny | right now having an issue connecting to AMI from another host in network, nmap shows the port closed from that host |
| 18:26.49 | ghost75 | chk manager.conf |
| 18:27.13 | nny | yeah it's set to 5038 |
| 18:27.25 | nny | and I can connect via the localhost |
| 18:27.36 | ghost75 | there is host setting there |
| 18:27.40 | nny | and there is a definition for that host |
| 18:27.46 | nny | that defines it's IP |
| 18:27.55 | nny | but that shouldn't change for nmap |
| 18:28.01 | nny | since it's just checking the port state |
| 18:28.07 | nny | its/it's |
| 18:28.37 | nny | that defines its* IP :S |
| 18:29.17 | ghost75 | permit and bindaddr |
| 18:29.40 | nny | ghost75: it's defined for the host |
| 18:29.46 | nny | (2:27:40 PM) nny: and there is a definition for that host |
| 18:29.50 | ghost75 | how? |
| 18:30.36 | nny | permit = 192.168.1.15/255.255.255.0 |
| 18:30.40 | nny | magic |
| 18:31.01 | ghost75 | bindaddr? |
| 18:31.02 | WIMPy | .15/.0? |
| 18:31.18 | nny | WIMPy: both hosts ont he same subnet |
| 18:31.32 | ghost75 | yes should be /32 |
| 18:31.44 | nny | actually bindaddr may be the issue |
| 18:31.59 | nny | bindaddr = 127.0.0.1 |
| 18:32.05 | ghost75 | wrong |
| 18:32.09 | nny | wrong! |
| 18:32.12 | WIMPy | You don't usually set a host within the network. Some applications don't even allow it. |
| 18:32.13 | nny | thanks ed! |
| 18:33.05 | nny | WIMPy: hmm? |
| 18:33.33 | ghost75 | permit = 192.168.1.15/255.255.255.255 |
| 18:33.44 | nny | ghost75: that would also work, but so would .0 |
| 18:34.12 | nny | but either way that doesn't work |
| 18:34.23 | nny | and since nmap works on a lower OSI layer |
| 18:34.45 | nny | so either way that change is irrelevant, although it is made |
| 18:35.20 | nny | bindaddr = 0.0.0.0 looks to be the right solution, testing |
| 18:36.07 | ghost75 | 0.0.0.0 are all interface |
| 18:36.07 | nny | yup |
| 18:36.38 | nny | so yeah, I was right, bindaddr was the issue. seems fop2 changes the manager.conf file during install, forgot about that |
| 18:37.14 | nny | and permit 192.168.1.15/255.255.255.0 is just all hosts in the 192.168.1.0 network, I was juse bing lazy, either would work |
| 18:37.40 | igcewieling1 | 192.168.1.15/255.255.255.0 is WRONG, but asterisk may fix it up internally |
| 18:38.24 | nny | igcewieling1: yeah manx, hence the lazy part |
| 18:39.12 | nny | fwiw I had no deny definition prior to testing (for testing reasons) |
| 18:39.17 | nny | so I knew it wasn't the cause |
| 18:39.25 | nny | besides |
| 18:39.32 | nny | nmap doesn't give a shit |
| 18:39.50 | nny | but it does care if the service isn't bound to all IPs, so yeah, bindaddr |
| 18:41.17 | navaismo | using openvpn trough iax2 may cause the terminated call with Re-invite to non-existing call leg on other UA.? |
| 18:41.36 | *** join/#asterisk tonyclewis (uid6025@gateway/web/irccloud.com/x-jyxbtnmjihenjelh) |
| 18:47.58 | *** join/#asterisk nephfl (6c09e362@gateway/web/freenode/ip.108.9.227.98) |
| 18:55.46 | navaismo | ?_? |
| 19:02.35 | nny | anyone want a laugh? |
| 19:02.46 | nny | cisco doo series SMB switches have "green ethernet" on my default |
| 19:03.06 | nny | when plugged into cisco SPA phones, this causes hilarious and unpredictable results |
| 19:03.57 | jmetro | green ethernet..cause those 1's and 0's are melting our ice caps |
| 19:04.03 | nny | jmetro: ha indeed |
| 19:04.13 | nny | the ports shut off when the switch thinks they aren't being used |
| 19:04.19 | nny | which would happen in the middle of calls |
| 19:07.31 | *** join/#asterisk dxrt (~dxrt@unaffiliated/dxrt) |
| 19:07.45 | *** join/#asterisk bpriddy (~bpriddy@ipv4.host.stabbyspazzout.net) |
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| 19:21.02 | WIMPy | Ethernet is extremely ungreen indeed. |
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| 19:26.13 | nny | WIMPy: yeah the whole thing was a PIA. Phones going offline during calls, was surprised to track down the cause. Just a stupid default setting, probably to get some kind of tax credit from the gov or some crap |
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| 19:28.42 | jmetro | you bet it was nny |
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| 19:49.50 | nephfl | did you get the pb in the other window? |
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| 20:05.08 | MrMeek | anyone know if System() calls carry much overhead in terms of time to setup the call? I'm wondering if you can save any time by ANDing operations like System(op1 & op2 & op3) |
| 20:07.24 | igcewieling1 | MrMeek: system has as much overhead as AGI, some overhead which will add up with lots of calls, but it isn't major. If you can easily combine commands in a System() call, I recommend you do that. |
| 20:16.56 | MrMeek | thanks igcewieling1 |
| 20:24.10 | robert_ | hm, okay. Asterisk thinks the channel is insecure; do I have to set something somewhere so that it will make the channel secure? (Other than in the dialplan, I mean.) |
| 20:44.40 | navaismo | vpn must be treated as localnet or nat??? |
| 20:47.07 | igcewieling1 | navaismo: are packets natted when going to the other side of the VPN? If not, then it is a localnet |
| 20:47.29 | igcewieling1 | localnet means "these ips do not need nat handling" |
| 20:51.56 | navaismo | right, just trying to figure out from where or what is causing the call drops with warning Re-invite to non-existing call leg on other UA. all day and nothing is clear yet |
| 20:56.40 | igcewieling1 | navaismo: when you figure it out, let me know, we have been having random call drop issues too, though I've not see many of the "Re-invite to non-existing call leg" messages |
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| 21:03.58 | navaismo | i hope i can trace it |
| 21:06.14 | *** join/#asterisk studybot_ (~studybot_@gateway/tor-sasl/studybot/x-68286794) |
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| 21:14.46 | jimw | My 2-line dial plan has errors: http://paste.org/62546 |
| 21:15.17 | jmetro | jimw: what about it? |
| 21:15.30 | jimw | Strangely, I can't find it. |
| 21:15.33 | igcewieling1 | I don't see any errors in that pastebin |
| 21:15.42 | jimw | Asterisk 11 does. |
| 21:15.45 | igcewieling1 | could you be missing the trailing newline |
| 21:16.03 | igcewieling1 | jimw: perhaps you could include the actual error in the pastebin? |
| 21:16.26 | Qwell | JIM isn't a valid config option. |
| 21:16.41 | Qwell | Did you perhaps mean [globals] ? |
| 21:16.43 | jimw | Trailing newline's there. Error message: ==!!== Unknown directive: JIM at line 2 of extensions.conf -- IGNORING!!! |
| 21:17.06 | igcewieling1 | Qwell, as usual, is correct. |
| 21:17.10 | jimw | Ah! Maybe Qwell found it! |
| 21:17.27 | jimw | Now, I must pick up my hair and glue it back on. |
| 21:17.28 | Qwell | That'll be $399.98. |
| 21:18.58 | ChannelZ | and why can't we have Jim as a valid config option? Someone get on this. |
| 21:19.23 | jimw | Apparently, variables have various scopes. global ain't one of 'em. |
| 21:19.57 | ChannelZ | I think it's just name discrimination. |
| 21:21.28 | *** join/#asterisk marksaitis (~marksaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 21:27.04 | jmetro | Variables have scopes, like private and public |
| 21:27.05 | jmetro | and global should be one too |
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| 21:49.06 | *** part/#asterisk jimw (~chatzilla@adsl-74-179-108-63.gnv.bellsouth.net) |
| 22:09.11 | *** join/#asterisk _abc_ (~user@unaffiliated/ccbbaa) |
| 22:09.26 | _abc_ | Hello. I have a daft question. Does anyone check the asterisk releases with a set of phones |
| 22:09.30 | _abc_ | ? |
| 22:09.47 | _abc_ | I tried to upgrade from 1.6.1 to 1.8.20.1 and I could not, all cisco phones which I tried were broken. Using skinny. |
| 22:10.05 | _abc_ | None were recognized/found in despite of several attempts to rewrite the config files |
| 22:10.09 | _abc_ | Has anyone else seen this? |
| 22:10.34 | *** join/#asterisk apb1963_ (~apb1963@174.134.117.244) |
| 22:11.06 | _abc_ | Also skinny phones calling test services which worked before (echo, callback), got stuck and had to be nanually rebooted by power cycling and I saw errors on console concerning 'unexpected packet lengths' and such |
| 22:11.20 | _abc_ | The skinny implementation in 1.6 was crude but functional, that is no longer the case?! |
| 22:11.37 | _abc_ | Please advise? |
| 22:11.57 | _abc_ | Incidentally test units were 7940 and 7960 with skinny/sccp firmware. |
| 22:13.57 | _abc_ | Also, when is this channel more active. I seem to drop in here when it is deathly quiet. North American afternoon times? I am in Europe, I try to stay up late to catch up with such times when I can. |
| 22:14.39 | igcewieling1 | _abc_: was there any mention of Skinny in the UPGRADE-*.txt files? |
| 22:16.06 | _abc_ | Nothing important, a lot more features, and slightly different configs, I edited them and that was okay. |
| 22:16.20 | _abc_ | No config load errors, but none of the registered devices were recognized. |
| 22:16.27 | _abc_ | This is the same hardware used for 1.6 |
| 22:16.40 | _abc_ | I backed out of the 1.8.20.1 upgrade and it all worked again |
| 22:22.05 | mjordan | _abc_: skinny has gone through several major changes, from 1.8 to 10 to 11. The person who best knows all of the changes in skinny is wedhorn |
| 22:22.47 | _abc_ | Is he here sometimes? |
| 22:23.00 | _abc_ | Also does it make sense to move from 1.6 to 10/11 directly? |
| 22:23.19 | _abc_ | I am aiming for low hassle. This pbx is used exclusively offline for testing phones and setups in a lab |
| 22:23.28 | mjordan | _abc_: yes, but more often in #asterisk-dev. He does actively handle issues in Jira. (IIRC, he lives in Australia) |
| 22:23.45 | mjordan | _abc_: as far as 1.6 to a later version: can't answer that. I'm not up on all of the chan_skinny mods |
| 22:23.53 | _abc_ | ok, so aussie timezone |
| 22:24.05 | _abc_ | thanks for the info, very useful. |
| 22:24.14 | mjordan | but there are significant differences between the various versions, as he did some major reorganization of the channel driver in 11 (and is going on further in trunk) |
| 22:24.15 | mjordan | np |
| 22:24.27 | _abc_ | Still, I am curious if new releases are tested. The failure was so obvious it could not have gone through qc imho |
| 22:24.44 | _abc_ | 7940/60 is a very common phone |
| 22:25.29 | _abc_ | I will try to talk to wedhorn about this, after collecting some info. I was so horrified by the bugs I backed out at lightspeed and did not keep many logs. |
| 22:25.31 | mjordan | we don't test it. It's an extended support module. |
| 22:25.36 | _abc_ | Ah. |
| 22:26.15 | _abc_ | Okay, that explains things. Maybe the bit rot got at the source while editing or something. I will look at sources a bit next week and see if I can back port what I need. I ave done such surgery before. |
| 22:26.22 | _abc_ | Anyway, thanks for the info. |
| 22:28.24 | _abc_ | mjordan: what is the recommended version of asterisk to be run now, for new installs? Non alpha stuff, stable? |
| 22:28.45 | _abc_ | My 1.6.1 install was tied to some specific modules and hackery and occurred in circa 2010 |
| 22:29.12 | _abc_ | I might consider upgrading even if I have to edit in my special patches with major changes if it is worth it. |
| 22:29.45 | _abc_ | There are many new important features in the newer skinny implementations which would greatly benefit what this is used for, especially dynamic device provisioning and remote device reset/reload. |
| 22:29.55 | *** join/#asterisk TriJetScud (~TriJetScu@van-app-svr.smb.curriegrad2004.ca) |
| 22:30.54 | jmetro | man our queue has like a 30s - 2m avg wait time...on hold with dell for 15 already.. how do they get away with it |
| 22:31.43 | ChannelZ | And how big is your company? :) |
| 22:32.17 | jmetro | Hm. Depending on how big dell is... maybe 1/1000th the personnell =p |
| 22:35.03 | mjordan | _abc_: both 1.8 and 11 are LTS |
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| 22:36.37 | *** join/#asterisk MrMeek (~meekhime@172-4-223-5.lightspeed.toldoh.sbcglobal.net) |
| 22:37.15 | _abc_ | LTS? I am not familiar with that acronym mjordan? |
| 22:37.47 | ghost75 | dell has also crap audio phone quality |
| 22:37.52 | jmetro | long term sexy |
| 22:38.03 | mjordan | Long Term Support |
| 22:38.13 | _abc_ | ghost75: probably cheaping out on bandwidth and using speex or other mr. roboto gsm compression |
| 22:38.18 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 22:38.29 | _abc_ | Some beancounter must have had an orgasm over that |
| 22:38.38 | jmetro | Dell's phone system also has terrible routing. You can hear the system typing numbers when they go between departments. |
| 22:38.47 | ghost75 | low bandwith because somewhere in india |
| 22:38.48 | _abc_ | I use asterisk in private pbx mode online and the sound quality in ulaw is better than the land line phone |
| 22:38.55 | mjordan | this may also be of interest: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States |
| 22:39.01 | _abc_ | comes from not stealing bits for various things "which will not be noticed" |
| 22:39.07 | _abc_ | mjordan: thanks |
| 22:39.10 | mjordan | np |
| 22:39.38 | _abc_ | I have to say I found voip.org a bit more clueful than asterisk and most web searches seem to point there |
| 22:39.44 | _abc_ | I mean search engine searches |
| 22:39.53 | _abc_ | Maybe I need to review the way I do searches. |
| 22:39.58 | ghost75 | but its a bit outdated |
| 22:40.05 | _abc_ | Yes |
| 22:40.28 | _abc_ | Still, I do not live on the bleeding edge and a detailed example or explanation is better than raw dotted 'release notes' |
| 22:40.44 | mjordan | we've tried to get a bit better with that |
| 22:41.14 | _abc_ | Oh I am sure the problem is on my end. European guy with English not his mother language searching for information on the web. What could possibly go wrong... |
| 22:41.59 | _abc_ | Anyway from personal experience I have to say that prodding Google et al a bit with new pages added to a wiki helps with ranking as long as it is not abused. |
| 22:43.59 | ghost75 | europe where |
| 22:44.03 | _abc_ | Bucharest |
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| 23:01.34 | *** join/#asterisk f0ner00t (~jvandyke@69-170-21-20.static-ip.telepacific.net) |
| 23:01.54 | f0ner00t | Hi is anybody good with 3com switches? |
| 23:03.03 | *** part/#asterisk ThinkGNU- (~ThinkGNU@216.67.183.210) |
| 23:07.56 | pietro | hi, |
| 23:08.26 | pietro | I need to maintain some extra headers in SIP invite added by my UAs |
| 23:08.43 | pietro | Is there a way without using SIPgetheader() and SIPaddheader() |
| 23:08.44 | pietro | ? |
| 23:09.06 | pietro | or SIP_HEADER() |
| 23:17.51 | *** join/#asterisk CunningPike (~CunningPi@d206-116-73-41.bchsia.telus.net) |
| 23:18.58 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 23:20.13 | *** join/#asterisk dimitry7 (~dimitry7@gate.aaamerica.com.mx) |
| 23:21.39 | dimitry7 | Hi Guys, I am configuring two IAX trunks (each one in different cities) I have configured iax.cong and extensions.conf and when i do *CLI> iax2 show registry |
| 23:22.12 | dimitry7 | it says rejected on my links. What can be wrong? I am fairly new to asterisk so I would really appreciate any help |
| 23:23.37 | dimitry7 | This asterisk server was already configured. I am just configuring the trunks. I might give you what I have added for you to have further details of this. |
| 23:25.02 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.28.157) |
| 23:25.21 | *** join/#asterisk DarthExpeditor (~IceChat9@rrcs-71-43-76-226.se.biz.rr.com) |
| 23:32.19 | ruben231 | hi guys any idea how to optimized voip |
| 23:33.53 | igcewieling1 | pietro: what is SIP_HEADER not acceptable? |
| 23:40.06 | jmetro | i keep running out of dell hold music |
| 23:43.51 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-tkoakmyfciuutftw) |
| 23:51.57 | navaismo | ruben231, QoS on the net, use small codecs, and a good network |
| 23:54.56 | RudyValencia | How do I tell if a connection is encrypted (so I can play a warning if there is none)? |
| 23:58.01 | dimitry7 | :( |
| 23:58.07 | dimitry7 | guys... have merci on me |
| 23:58.36 | *** join/#asterisk dxrt (~dxrt@unaffiliated/dxrt) |