| 00:02.32 | dauergast | hi, i want to have music-on-hold while dialing out, my exten looks like the following, but every time the default moh class is being played, whats wrong with it? |
| 00:02.37 | dauergast | exten => _**.,1,Dial(SIP/${EXTEN}@623,45,m(stream-1)KTX) |
| 00:04.15 | dauergast | asterisk V1.6.2.7 |
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| 00:12.54 | nargon | [TK]D-Fender well i would have though you would be able to specify per peer reg timeouts but i guess not |
| 00:13.20 | *** part/#asterisk Get_The_Fish (~Get_The_F@173-14-4-113-Colorado.hfc.comcastbusiness.net) |
| 00:13.52 | nargon | the BV peers are maintinaing the registration now that the reg timeout has been increased to 3600 |
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| 00:17.01 | [TK]D-Fender | nargon: Again, peers have nothing to do with registrations. |
| 00:17.12 | [TK]D-Fender | nargon: You can't be "per peer" when peers have nothing to do with it |
| 00:17.29 | Naikrovek | saisoma: is the background noise other people talking |
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| 00:17.41 | saisoma | Naikrovek, yes, it is |
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| 00:20.05 | Naikrovek | phones are tuned to voice - they attenuate other frequencies and amplify voice. i would say that there's not a lot you can do on the phone. the microphone you use, be it a headset or handset or speaker or whatever, may need to be adjusted |
| 00:21.03 | bcrisp | asterisk is fun |
| 00:21.20 | saisoma | Naikrovek, rgr. i'm not on site, but they say the previous phones didn't have any issues. going to have to see that to believe it. |
| 00:21.21 | trelane | bcrisp, low fat too! |
| 00:21.28 | saisoma | Naikrovek, thanks. i'm out for the night. night all! |
| 00:24.01 | bcrisp | is there anything sweeter than stack overflow exceptions? |
| 00:28.02 | dauergast | no one can help me with this simple configuration issue? Do i have to enable more features to get this working? |
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| 00:53.56 | KavanS | is there a replacement for nvbackground detect for dahdi? |
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| 01:47.18 | ruben23 | hi this a problem as i test my voip carrier on my asterisk---> http://pastebin.com/twR16xea |
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| 02:18.58 | joobie | guys, what's the options to setup asterisk so that MOH continues playback from where it left off - rather than starting at the start of the sound file each time |
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| 02:28.44 | TJNII | Mine does that with no special options. Are you using an external player? |
| 02:31.03 | TJNII | Made using those clips I found of The Brazillian Tropical Orchestra I found difficult, they were only 30 seconds long. |
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| 02:35.31 | joobie | nope, want to use the internal player |
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| 03:19.25 | ChannelZ | I got the jam.. the jam that make you go to work |
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| 03:46.03 | sawgood | What is the correct name for the process where you would see on the LCD of your SIP phone (the number the customer dialed to call you) direct number dialed |
| 03:46.41 | Kyosh | did? |
| 03:46.47 | Kyosh | inbound? |
| 03:47.41 | sawgood | well ... not really a DID number ... |
| 03:47.53 | sawgood | you have caller ID name and number on your LCD |
| 03:48.08 | Kyosh | clid and cnam |
| 03:48.09 | sawgood | right below this would be the telephone number the 'calling in' party dialed to reach you |
| 03:48.22 | sawgood | DNID |
| 03:48.28 | sawgood | something like that |
| 03:48.30 | Kyosh | DNIS |
| 03:48.37 | sawgood | ahw! |
| 03:48.54 | Kyosh | ? |
| 03:48.55 | sawgood | Is this part of Asterisk 1.4 and/or 1.6? |
| 03:48.59 | sawgood | thank you |
| 03:49.02 | sawgood | looking it up now |
| 03:49.16 | Kyosh | the DNIS is the number they dialed to get to you |
| 03:49.18 | Kyosh | same as DID |
| 03:49.47 | sawgood | yes, DNIS is what I am interested in adding to my phones |
| 03:50.08 | Kyosh | http://www.google.com/search?aq=f&sourceid=chrome&ie=UTF-8&q=define:+dnis |
| 03:50.24 | Kyosh | http://www.google.com/search?hl=en&q=define:+DID&btnG=Search |
| 03:51.27 | Kyosh | the DNIS is used to identify the DID that the caller dialed |
| 03:51.33 | Kyosh | asterisk sees these as the DID |
| 03:54.23 | sawgood | Is DNIS and/or DNID something I should be able to support using Asterisk 1.6.2.x |
| 03:54.49 | Kyosh | are you using a PRI? |
| 03:54.55 | sawgood | SIP trunk |
| 03:55.15 | Kyosh | hen you do not need DNIS support since your trunk provider is passing you the DNIS as the DID |
| 03:55.31 | Kyosh | hence, asterisk will see it |
| 03:55.33 | sawgood | I have another non Asterisk IP PBX which does this already ... over a SIP trunk |
| 03:55.50 | Kyosh | and its passed as the DID |
| 03:55.51 | sawgood | Right, but getting that out of the SIP header into a SIP phone is another thing |
| 03:55.58 | Kyosh | DNIS is passed as the DID by the trunk provider |
| 03:56.06 | sawgood | cool ... get that part ... |
| 03:56.29 | sawgood | but ... in a nutshell ... how does one get that information to the SIP phone (does the SIP phone have to specifically support doing this)? |
| 03:56.31 | Kyosh | in asterisk when configuring inbound routes and passing it to extensions, it's easy to modify the message |
| 03:56.46 | Kyosh | using asterisk, asterisk now, piaf, trixbox? |
| 03:56.58 | sawgood | Both Asterisk and AsteriskNOW at times |
| 04:00.52 | Kyosh | lemme see hold on |
| 04:02.02 | Kyosh | http://www.google.com/search?hl=en&q=define:+DID&btnG=Search |
| 04:02.08 | Kyosh | http://www.voip-info.org/wiki/view/Asterisk+func+callerid |
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| 04:06.23 | sawgood | exten => _9.,n,Dial(SIP/+${EXTEN:1}@sipcarrier) |
| 04:06.37 | sawgood | what is the 'purpose' for the ,n, in this statement? |
| 04:06.55 | Naikrovek | it's a priority |
| 04:07.08 | Naikrovek | n means "next" or "in the order listed in the file" |
| 04:07.20 | sawgood | different from using 1,2,3 etc? |
| 04:07.39 | Naikrovek | you used to have to put a number in there, like line numbers in basic |
| 04:07.41 | Naikrovek | yes |
| 04:07.44 | Naikrovek | well no |
| 04:07.45 | Naikrovek | heh |
| 04:07.53 | Naikrovek | yes in that you don't need to type 1 or 2 or 3 |
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| 04:07.59 | Naikrovek | but no in that the effect is the same |
| 04:08.02 | Naikrovek | as 1 2 3 |
| 04:08.04 | sawgood | exten => s,2,Voicemail(u${MACRO_EXTEN}) |
| 04:08.11 | Naikrovek | you must have a 1 for each exten => |
| 04:08.19 | Naikrovek | but after 1 is declared, the rest can be n |
| 04:08.25 | sawgood | what is the difference between using a ,n, and a ,2, in this statement? |
| 04:09.03 | sawgood | Naikrovek: thank you ... for the info |
| 04:09.14 | sawgood | start with a ,1, and then use numbers or ,n, after that |
| 04:09.15 | Naikrovek | none, functionally, but if you're doing a lot of dialplan editing, and you need to insert a line after 5, but your dialplan goes all the way to 55, then you have 50 numbers to change |
| 04:09.18 | Naikrovek | you can't skip numbers |
| 04:09.27 | sawgood | nice! |
| 04:09.31 | Naikrovek | so use n after 1 |
| 04:10.49 | sawgood | exten => s,2,Voicemail(u${MACRO_EXTEN}) |
| 04:10.49 | sawgood | exten => s,3,Hangup |
| 04:10.49 | sawgood | exten => s,102,Voicemail(b${MACRO_EXTEN}) |
| 04:11.05 | sawgood | what did this 'change' from 3 to 102 and not 4 |
| 04:11.29 | sawgood | I meant 'why' use a 102 instead of 4 |
| 04:12.29 | Naikrovek | i don't believe you can use a 102 there |
| 04:13.11 | Naikrovek | i'm no dialplan expert, though |
| 04:15.58 | sawgood | So, say I am able to adjust my dialing plan to include DNID/DNIS ... wouldn't my SIP end point still need to support tihs feature and/or require some adjustment in its config file? |
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| 04:17.33 | roberob | anyone ever have any issue with sip registration and at&t dsL/ |
| 04:17.34 | roberob | ? |
| 04:18.51 | sawgood | sure ... are you not able to register your SIP trunk behind an AT&T DSL box? |
| 04:18.58 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 04:19.31 | roberob | sawgood: exactly |
| 04:19.42 | roberob | i have the modem in bridge mode, with the public IP on the router.. |
| 04:20.05 | roberob | if i set a DMZ and set manual override on my softphone it registers instantly |
| 04:20.25 | roberob | problem with that is, i can only set a dmz for one IP, multiple machines on this network need access, simultaneously |
| 04:21.14 | sawgood | Is your AT&T DSL using a 'modem' or a 'gateway' device? |
| 04:21.26 | roberob | sawgood: i believe a modem |
| 04:21.38 | sawgood | what is the make/model of your AT&T edge device? |
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| 04:21.58 | roberob | its a motorola E141744 |
| 04:22.05 | sawgood | DSL modem then |
| 04:22.08 | roberob | yes |
| 04:22.16 | sawgood | So you only have dynamic IP addressing |
| 04:22.53 | roberob | from AT&T my IP is dynamic yes, then on the router, i disabled dhcp and i'm manually assigning IP's to the local machines, 10.0.0.x |
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| 04:24.05 | roberob | any ideas? |
| 04:24.10 | sawgood | cool .. so your Asterisk box has a 10.x.x.x IP ... but it is in DMZ mode (basically one to one NAT with the router) |
| 04:24.14 | roberob | no no no |
| 04:24.17 | joobie | i have an idea |
| 04:24.24 | roberob | my asterisk box has an external IP, and is colocated |
| 04:24.43 | roberob | not even at this location |
| 04:25.11 | sawgood | So, at the DSL location is only SIP phones? |
| 04:25.16 | roberob | the only time i can get sip registration to occur, is when i set one of the local machines to be the DMZ |
| 04:25.20 | roberob | yes only softphones |
| 04:25.57 | roberob | i'm using a wrt54g router as well |
| 04:25.58 | sawgood | one softphone at a time ... or only the softphone with the DMZ set |
| 04:26.07 | roberob | only the softphone with the DMZ set |
| 04:26.47 | roberob | softphone = xten eyebeam btw |
| 04:26.48 | sawgood | So, you probably could get this same result by using port forwarding to one IP address then |
| 04:26.59 | roberob | sawgood: i would think so.. however, it doesn't work |
| 04:27.08 | sawgood | I've seen this before .... |
| 04:27.14 | roberob | i forward ports 5060, 5061, set manual override on the softphone, also forward that port |
| 04:27.40 | sawgood | generally if you want to have more than one SIP phone (remote) registered being a router in a NAT ... |
| 04:27.59 | sawgood | you'll need to 'adjust' the setting on your Asterisk box to 'keep' the NAT session open longer |
| 04:28.05 | roberob | sawgood: the funny thing is, it worked fine, same router, same machines, same softphones and setups from my house with a cable connection |
| 04:28.31 | roberob | sawgood: ok, i'm open, what setting would that be? |
| 04:28.45 | roberob | some kind of keep_alive |
| 04:29.16 | sawgood | qualifyfreq=25 |
| 04:29.27 | sawgood | if you are using Asterisk, I believe it would go in sip.conf |
| 04:29.54 | sawgood | The default is 30 or 60 ... and maybe the NAT times out before the next UDP packet |
| 04:29.59 | sawgood | I've faced this a few times |
| 04:30.14 | sawgood | only with remote phones in a NAT envoirnment |
| 04:30.20 | roberob | qualifyfreq or qualifyfreqok ? |
| 04:30.23 | roberob | in asterisk.conf ? |
| 04:30.38 | sawgood | qualifyfreq=25 |
| 04:31.01 | roberob | alright lets give it a shot |
| 04:31.31 | roberob | looks like its going to timeout.. |
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| 04:31.41 | roberob | yeah, pretty sure its a routing issue on this side tbh |
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| 04:36.43 | roberob | any ideas? |
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| 05:29.28 | joobie | guys, how can i setup the queue moh so that it just plays one audio and streams it to everyone on hold? at the moment it seems to play a seperate moh for each caller |
| 05:38.10 | Polis_ttt | joobie: i belive that it shall per default play same stream for same queue, you can simply this by placing two simultanius calls, with 10sec delay, then you will hear the same music on both phones if my thought is correct |
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| 06:02.02 | Corydon76-dig | joobie: it only works that way in mp3 mode, not in file mode |
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| 06:10.41 | joobie | Polis_ttt, I tried that - it doesnt |
| 06:10.48 | joobie | Corydon76-dig, ahh thanks |
| 06:11.26 | joobie | In file mode, is there a way I can randomly select a start point if I have only 1 file? |
| 06:11.44 | joobie | i noticed if i split the file out i can use random, but i just created one big file |
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| 07:08.26 | adyb | hi! I have asterisk with Audio Codes MP-114 FXO, 4 anolog lines. When i want to turn more volume from the FXO using Voice Volume and Input Gain when i start to dial on the line i hear a continous noise and i can't dial any more. Can someone help me with this? Thanks. |
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| 07:34.33 | voipmonk | turn down the volume |
| 07:34.39 | voipmonk | and get clean lines |
| 07:34.42 | voipmonk | to the fxo |
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| 07:46.42 | adyb | voipmonk the volume from asterisk ? |
| 07:46.54 | voipmonk | no |
| 07:47.04 | adyb | from the fxo i tryed and it is ok |
| 07:47.05 | voipmonk | the gain settings on your fxo hrdware |
| 07:47.19 | adyb | but you can bearly hear the men on the phone |
| 07:49.00 | adyb | <PROTECTED> |
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| 08:23.09 | gr0mit | Faustov, ping... |
| 08:23.58 | Faustov | gr0mit: I'm already on the case, once again sorry :( |
| 08:24.20 | gr0mit | Faustov, no prbs thanks! |
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| 08:39.48 | joobie | guys how can i define what hold music context is used for a sip user, when they press 'hold' on the phone? |
| 08:40.25 | joobie | i have a queue setup, which defines hold music for the queue.. but when a caller transfers to a memeber, if the member puts the caller on hold it will go back and play the default context hold music |
| 08:45.25 | *** join/#asterisk _omer (~omer@119.152.47.16) |
| 08:45.28 | _omer | hello |
| 08:46.14 | _omer | Is there a way to secure AGI Script Code? something like compiled AGI Script ? |
| 08:46.25 | *** join/#asterisk odenkos (~odenkos@ip-212-081-019-170.static.nextra.sk) |
| 08:47.00 | WIMPy | _omer: man gcc |
| 08:47.27 | _omer | gcc is a compiler....can I compile AGI Script? |
| 08:47.36 | _omer | :-/ |
| 08:48.09 | WIMPy | Never heard of agi script. |
| 08:48.17 | joobie | werd |
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| 08:48.31 | joobie | _omer, what language is your AGI script in now? |
| 08:48.37 | _omer | Perl |
| 08:49.04 | _omer | but I need a solution to secure my code ... if possible then I can change the language. |
| 08:50.06 | joobie | when you say secure your code.. from what? |
| 08:50.52 | _omer | just like Microsoft doesn't share it's code with the users :) |
| 08:51.25 | _omer | If I deliver AGI to my client, I dont want him to see the code or resell it .. |
| 08:51.57 | joobie | so license the code? |
| 08:52.24 | _omer | that's the next step |
| 08:52.38 | _omer | but to secure the AGI code ... |
| 08:52.41 | joobie | what does your perl script actually do? |
| 08:52.44 | TSM | google obfuscate perl |
| 08:52.57 | _omer | it's a Voip Wholesale billing script ... |
| 08:53.19 | TSM | if he is determined he can reverse it but its time consuming |
| 08:53.20 | joobie | obfuscate can be reversed with deobfuscate.pm |
| 08:53.45 | joobie | TSM, there's a deobfuscate.pm to make it a whole lot easier.. |
| 08:54.22 | TSM | you are still left without meaningfull variable names, but code layout is visible |
| 08:54.29 | TSM | rewrite to PHP then |
| 08:55.14 | TSM | or just rewrite a small vital module in PHP and include it in your PERL though exec, thats poo but depends on how much code you have |
| 08:56.56 | joobie | _omer, your question is best answered in a perl related channel |
| 08:57.23 | joobie | thought TSM's suggestion of offloading modules is a good idea if you cant find a way to encrypt your source |
| 08:57.28 | _omer | I want to hide my code so that the source code is not freely available when my application is distributed. |
| 08:57.35 | _omer | ok joobie |
| 08:57.36 | _omer | thanks |
| 08:57.36 | joobie | I guess you could do something like.. |
| 08:57.48 | joobie | have your module encrypted with some hash |
| 08:58.11 | joobie | and write a small program to sit ontop of it |
| 08:58.12 | _omer | thats a new thing for me ... can you explain a little bit |
| 08:58.16 | joobie | where it justs goes to your website |
| 08:58.18 | joobie | auths the user |
| 08:58.24 | joobie | pulls down the hash and then accesses the module |
| 08:58.28 | TSM | naa thats crud because they still see the code |
| 08:58.51 | joobie | depends how you do it |
| 08:59.36 | joobie | thinks |
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| 09:03.06 | joobie | _omer, can you integrate your code with the web? |
| 09:03.07 | _omer | joobie: if I have AGI Script in Java .. then ? |
| 09:03.14 | joobie | like a web component? |
| 09:03.22 | joobie | there are decompiles for a lot of things man |
| 09:03.24 | joobie | including java |
| 09:03.32 | joobie | you need to look into encryption |
| 09:03.52 | joobie | but even still, once the code is decrypted, then it may still be accessible |
| 09:03.56 | joobie | .. at that point |
| 09:04.10 | joobie | your safest option is to do a module that is part web-based part locally on the box |
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| 09:04.32 | joobie | then you can offload some code on your server, so that it wont work in its entirety if your code on the clients machine was compromised |
| 09:04.35 | TSM | PHP has a good obsfucation |
| 09:04.54 | joobie | any server-side language is |
| 09:04.58 | joobie | as long as it's off that client |
| 09:05.13 | _omer | joobie: but it doesn't mean I have given my client a full standalone application......it must be called has HOSTED solution |
| 09:05.17 | joobie | you can even just use CGI man and keep it all perl basde |
| 09:05.23 | joobie | -basde +based |
| 09:05.53 | joobie | .. and they require an internet connection to their asterisk box |
| 09:05.57 | joobie | which may or may not be possible |
| 09:06.21 | joobie | your problem is most compilers can be decompiled |
| 09:06.53 | _omer | hmmmmm ... |
| 09:06.57 | joobie | even your m$ example above, you can decompile parts of their code |
| 09:07.21 | joobie | ask a perl chan man |
| 09:07.27 | joobie | how you can secure your code |
| 09:07.39 | _omer | according to perl channels...thats not possible .. |
| 09:07.40 | joobie | feed back their response if you don't mind - interested to know waht they suggest |
| 09:07.48 | joobie | what is not possible? |
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| 09:08.05 | _omer | securing perl code |
| 09:08.15 | joobie | ask them if you can encrypt it |
| 09:08.28 | joobie | if there's some model that they can suggest with encryption of the code |
| 09:08.58 | _omer | alright but how about writing a .so function for asterisk ... I think thats gonna be a HUGE work to be done... |
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| 09:16.44 | joobie | _omer, have a read of http://www.blackforest.us/?p=80 |
| 09:16.57 | joobie | it will give you an understand of what an .so file is |
| 09:17.37 | joobie | did a quick google and it looks like there are decompilers that can handle it - such as REC |
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| 09:18.12 | joobie | keep in mind decompilers may not "completely" decompile your app |
| 09:18.19 | joobie | but they will largely |
| 09:19.35 | _omer | sorry I was away .. let me read |
| 09:20.10 | _omer | joobie: how about writing AGI script in C ? |
| 09:20.42 | WIMPy | ... |
| 09:24.07 | joobie | _omer, you are fuked with comments like that. Just stick with perl |
| 09:24.45 | joobie | use obfuscate and be happy that you'll protect your code against the noobs |
| 09:25.48 | joobie | guys how can i define what hold music context is used for a sip user, when they press 'hold' on the phone? |
| 09:25.58 | joobie | i have a queue setup, which defines hold music for the queue.. but when a caller transfers to a memeber, if the member puts the caller on hold it will go back and play the default context hold music |
| 09:26.54 | _omer | joobie: THANKS!!! |
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| 09:49.32 | Gopal | joobie: you can play with musiconhold.conf... |
| 09:57.43 | joobie | Gopal, i setup another context in there |
| 09:57.51 | joobie | how can i get a handset to use that when the users peresses hold |
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| 10:02.50 | hackt0r | I am busy with the Asterisk testsuite, but have problems with it |
| 10:03.17 | hackt0r | most of the tests can't find the 'help console dial' command |
| 10:03.27 | hackt0r | any suggestions? |
| 10:03.54 | hackt0r | do I have to flip some options in the Asterisk Makefile? |
| 10:05.19 | hackt0r | flip=switch |
| 10:06.51 | Gopal | joobie: you mean the physical device? |
| 10:07.00 | kaldemar | hackt0r: use make menuselect to select the built components |
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| 10:23.21 | joobie | Gopal, yes |
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| 10:44.14 | Seb^ | hi! |
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| 11:05.36 | Seb^ | ok |
| 11:05.39 | Seb^ | i can receive calls |
| 11:05.53 | Seb^ | but when i dial out using my zap trunk i get all circuits are busy now |
| 11:06.03 | Seb^ | details are here: http://forums.digium.com/viewtopic.php?f=1&t=74172&sid=dd0fd869793ee17ee84ac5757b4eee38 |
| 11:06.06 | Seb^ | can anyone help? |
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| 11:06.20 | ms110 | Hi |
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| 11:08.43 | ms110 | is there any application for changing voice while dialing any number, probably more choices (altering pitch, timbre etc, robot voice, old person etc)? |
| 11:13.20 | kaldemar | Seb^: no relevant details there. you need to pastebin a CLI output of a call attempt. and btw, since you're using freepbx, you'll be more likely to find help in #freepbx. |
| 11:14.01 | Seb^ | ah ok |
| 11:14.03 | Seb^ | thanks dude |
| 11:14.25 | Seb^ | the cli output is on that thread btw |
| 11:14.28 | Seb^ | if you read down |
| 11:14.32 | Seb^ | in my 3rd and 4th posts |
| 11:14.33 | Seb^ | :) |
| 11:17.56 | dauergast | i want moh to be played for outgoing calls..but every time the default moh class is being played and not the custom class i defined, what wrong with it? _**.,1,Dial(SIP/${EXTEN}@623,45,KTXm[radio]) |
| 11:20.42 | hackt0r | kaldemar: thnx |
| 11:24.48 | ttwhy | Hi, where do i set the timeout of a incoming capi call? |
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| 11:55.53 | hackt0r | Does this menuselect item: 'pbx_gtkconsole' provide the cli command: 'help console dial'? |
| 11:56.56 | kaldemar | no |
| 11:56.58 | hackt0r | it is unliky, but the only option with console in it |
| 11:57.07 | hackt0r | kaldemar: ok |
| 11:57.43 | kaldemar | hackt0r: compile either chan_oss or chan_alsa and you'll get the console commands |
| 11:57.58 | kaldemar | they are in "Channel Drivers" |
| 11:58.39 | kaldemar | chan_console might also do, but i've never tried it. it depends on portaudio. |
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| 12:07.31 | hackt0r | kaldemar: i use 1.4 |
| 12:08.04 | kaldemar | IIRC 1.4 versions have both chan_oss and chan_alsa |
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| 12:37.43 | DannyZB | is this the channel for VOIP asterisk? |
| 12:38.08 | DannyZB | I'm inexperienced in VOIP programming |
| 12:38.10 | DannyZB | quick question |
| 12:38.30 | DannyZB | how do I sync SIP between two devices in a direct connection? |
| 12:38.43 | DannyZB | I'm still not using an ASTERISK server |
| 12:38.59 | DannyZB | ie . how do I make sure lost packets don't get audio out of sync |
| 12:46.26 | florz | DannyZB: out of sync with what? |
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| 12:58.53 | Dovid | hello ev1 |
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| 13:43.35 | boch | Hi all, im having a problem, i have a DID point to my asterisk and the exten for that DID is configured OK, the fact is sometimes goes perfect but sometimes the call is rejected because exten was not found. never had a problem like this, what may i be doing wrong ? |
| 13:44.11 | Zeeek | boch are you watching the console output when the calls come in? |
| 13:44.22 | boch | Zeeek, yes |
| 13:44.39 | Trixboxer | boch: Is your extension set to qualify=yes ? |
| 13:44.41 | Zeeek | and you compare the exact string you see in the Dial command? |
| 13:45.48 | boch | Zeeek, it is an incoming call, the incoming peer is set to a2billing context and the proper extension is defined there |
| 13:45.56 | boch | Trixboxer, you mean the peer? yes |
| 13:46.08 | kaldemar | exten not found is a dialplan issue |
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| 13:46.33 | Trixboxer | boch: yes... type=friend and qualify=yes and nat=yes (if req) |
| 13:47.11 | boch | Trixboxer, yes, the peer works fine |
| 13:47.36 | Trixboxer | boch: In CLI "sip show peers" should give ok for all |
| 13:47.42 | boch | Trixboxer, it does |
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| 13:48.02 | Trixboxer | boch: So is it working ? |
| 13:48.09 | boch | Trixboxer, i say the calls goes trough, i have to redial 4 or 5 times for matching the exten, without touching anything |
| 13:48.31 | Zeeek | I guess what I meant was, compare the console output for the calls that do and do not work to try to see a difference |
| 13:48.32 | boch | Trixboxer, did you read my question ? |
| 13:49.00 | Trixboxer | boch: "... the call is rejected because exten was not found .." |
| 13:49.14 | Zeeek | boch use the command that shows all the extension (sorry can't remember) |
| 13:49.19 | boch | Trixboxer, i repeated sometimes _twice_ |
| 13:49.32 | boch | Zeeek, it is there, it is well configured |
| 13:49.36 | Zeeek | it may be a corrupted context or sthing |
| 13:50.04 | boch | i dont understand how can works for some calls and for some other not |
| 13:50.06 | Zeeek | boch in that case, it a very rare condition called "Toyota Syndrome" |
| 13:50.17 | Zeeek | Cosmic particles are bombarding your CPU |
| 13:50.30 | boch | yes maybe it is damned |
| 13:50.31 | Trixboxer | Zeeek: I guess you are right ;) |
| 13:50.39 | *** join/#asterisk hugorebelo (~hugo@200-171-132-124.completo.com.br) |
| 13:50.58 | Zeeek | follow these steps to resolve Toyota Syndrome: |
| 13:51.06 | Zeeek | 1) Obtain a large hammer |
| 13:51.25 | Zeeek | 2) collect as many frolg's teeth as you can find |
| 13:51.52 | Zeeek | 3) the hardest part, depending on where you live, will be finding a virgin to sacrifice |
| 13:52.52 | Zeeek | Make the naked virgin kiss the toothless frog and hit your thumb with the hammer in order to remain focused on the CPU issue |
| 13:53.23 | Zeeek | then: extensions reload |
| 13:53.47 | boch | what funny you are.. ho ho ho.. |
| 13:54.01 | Zeeek | we now return you to normal asterisk programming |
| 13:54.45 | Zeeek | what console output preceeds the "extension not found"? |
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| 14:35.07 | wcselby | o/ |
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| 14:37.48 | esaym | howdy, the caller id time sent to all of my phones is 1 hour behind? The time on the machine is set right. How do I adjust how asterisk sees the time? |
| 14:38.40 | [TK]D-Fender | esaym: What phones? |
| 14:39.13 | esaym | [TK]D-Fender: well the time is off in the csv log and the phone. The phone is a cordless connected to a grandstream ata |
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| 14:39.26 | [TK]D-Fender | esaym: Last I hear SIP doesn't pass the time..... |
| 14:39.44 | esaym | hmm |
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| 14:41.57 | esaym | [TK]D-Fender: ok something might be messed up here. Looks like the log is in zulu, |
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| 14:50.58 | propopdan | hello.... first time asterisk administrator here. can I ask newbie questions here? |
| 14:51.41 | Gido-E | ~ask |
| 14:51.42 | 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. |
| 14:51.43 | Baylink_work | Well, I have a few free minutes, so why not. :-) |
| 14:52.07 | Baylink_work | That's not entirely a valid answer to the way he asked the question, Gido-E |
| 14:52.08 | Baylink_work | :-) |
| 14:52.17 | Gido-E | ~ask |
| 14:52.18 | 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. |
| 14:52.55 | KavanS | software echo cancellation - dahdi, asterisk 1.4.28 - pros/cons/thoughts on effectiveness? |
| 14:53.01 | Baylink_work | Ok. propopdan: I rule your inquiry a valid one, regardless of whether Gido-E thinks it falls within the purview of the robot reply. Go ahead: |
| 14:53.16 | [TK]D-Fender | KavanS: What card, how many channels/ |
| 14:53.18 | Baylink_work | KavanS: dahdi vs zaptel? |
| 14:53.19 | Gido-E | I am EVIL! :-) |
| 14:53.24 | *** join/#asterisk Brian_H (~Brian_H@static-173-50-141-22.ptldor.fios.verizon.net) |
| 14:53.26 | Baylink_work | Yeah, we know. :-) |
| 14:53.42 | KavanS | [TK]D-Fender, tdm410p 2 channels, no hardware ec |
| 14:54.00 | [TK]D-Fender | Gido-E: No, you are semi-evil. Quasi-evil. You... are the Diet-Coke of evil. Just 1 calorie. Not quite evil enough.... |
| 14:54.00 | KavanS | Baylink_work, was using zaptel, now I'm using dahdi, just wanted to know what software would work reliably with dahdi |
| 14:54.37 | *** join/#asterisk neurosys (~neurosys@32.163.201.123) |
| 14:54.42 | Baylink_work | Oh, KavanS. I thought you were trying to find out whether dahdi swec was any better or worse than zaptel swec. I run some zaptel swec here, and it works ok. |
| 14:54.44 | [TK]D-Fender | KavanS: YMMV. DAHDI stock EC's may cut it, if not there is HPEC which you are licensed for if your card is under warranty, OSLEC it that fails or you aren't covered |
| 14:55.06 | KavanS | oh, dahdi has ec's already? |
| 14:55.14 | KavanS | I have echo cancellation in the config and it complains I don't have the hardware |
| 14:55.33 | KavanS | I'm assuming you're speaking of software ec - would I find such settings in chan_dahdi.conf ? |
| 14:55.35 | [TK]D-Fender | KavanS: "warns" |
| 14:56.07 | [TK]D-Fender | KavanS: No, that'd be system.conf , chan_dahdi.conf on specifies that EC should be used... not which |
| 14:56.21 | propopdan | Thanks Baylink_work and Gido-E. The first question is like this... I have a user that *sometimes* (I suspect when a caller has a blocked #), she does not get a caller ID, but gets the display SIP@[server].com. When this happens, the user says she is unable to transfer calls, and when she puts the call on hold it drops. I do not know where to even begin to look for a problem. |
| 14:57.18 | wcselby | propopdan - try to recreate, and then pastebin the CLI output of the entire call |
| 14:57.23 | Baylink_work | I would recommend first checking your screenlog to see if you can confirm that's the causative symptom. If you're not already logging enough data, you should turn it up. |
| 14:57.23 | KavanS | [TK]D-Fender, chan_dahdi.c:1780 dahdi_enable_ec: Unable to enable echo cancellation on channel 1 (No such device) |
| 14:57.25 | [TK]D-Fender | propopdan: You start by telling us what the client is using, where this call is coming from, aqnd providing a complete call attempt with SIP DEBUG enabled from CLI |
| 14:57.26 | *** join/#asterisk x303 (~X303@97.102.28.28) |
| 14:57.28 | [TK]D-Fender | ~pb |
| 14:57.29 | infobot | [~pb] 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/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 14:57.30 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 14:57.32 | KavanS | ok, so I must have hardware ec enabled somewhere |
| 14:57.38 | KavanS | I'm not seeing a system.conf however |
| 14:57.48 | [TK]D-Fender | KavanS: "no such device".... sounds like a broken config |
| 14:58.06 | wcselby | KavanS - I got the same error if I had ec turned on in chan_dahdi.conf but nothing deefined in system.conf |
| 14:58.22 | wcselby | KavanS - /etc/dahdi/system.conf |
| 14:58.31 | propopdan | wcselby, Baylink_work and [TK]D-Fender -- thank you for your help. I will collect this info and return. |
| 14:58.37 | Baylink_work | Good man. :-) |
| 14:58.38 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 14:58.43 | KavanS | ahh nice, any recommendations for default settings on this system.conf |
| 14:58.49 | KavanS | i.e. what works for the masses |
| 14:58.50 | *** join/#asterisk dauergast (~sag@188-193-136-54-dynip.superkabel.de) |
| 14:58.58 | dauergast | I get "__ast_play_and_record: No audio available on SIP/XXX" on the CLI, on incoming call to voicemail, but there is audio, so whats wrong, can someone help? |
| 14:59.21 | [TK]D-Fender | KavanS: start with mg2 and see how you fare |
| 14:59.51 | KavanS | roger that - will give it a whirl, thanks for the direction on this guys |
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| 15:06.12 | diegomad | is possible run asterisk en other port than default? |
| 15:07.20 | *** join/#asterisk zilla10000 (~keouc@ip70-174-129-179.dc.dc.cox.net) |
| 15:08.44 | qxork | diegomad for sip? |
| 15:08.50 | zilla10000 | how can I disable the CDR report for each extension under cdr-csv - such as 1000.csv 1001.csv |
| 15:08.55 | Zeeek | qxork: 'sup? |
| 15:09.07 | qxork | Sup Zeeek :) |
| 15:09.17 | diegomad | qxork, yes |
| 15:09.31 | Zeeek | We have a great call scheduled for today - as you know qxork |
| 15:09.39 | wcselby | diegomad - yes, change it in sip.conf |
| 15:10.14 | qxork | diegomad: http://tinyurl.com/35urmtk |
| 15:10.27 | *** join/#asterisk HenrikJott (~info@h34n2fls33o954.telia.com) |
| 15:10.29 | Zeeek | http://vuc.me VoIP Users Conference starts in about 35 minutes - "Just a Bunch of Guys like You™" |
| 15:11.02 | qxork | Zeeek: looking forward to it. The weekly #VUC call is why I remember to IRC :) |
| 15:11.33 | Zeeek | 12 Noon: KONNECT a system of phones that connect offices and branches with no pbx. (How TF?) |
| 15:12.05 | *** join/#asterisk Tim_Toady (~moi@188.4.4.16.dsl.dyn.forthnet.gr) |
| 15:12.05 | Zeeek | 1 PM: Dan York, our favorite security guru on his new book "7 Deadliest UC Attacks" |
| 15:13.27 | HenrikJott | Hi all! i´m trying to apply a patch to Asterisk 1.4.21.2, it is supposed to fix some problems with mixmonitor.c, but as i am a rookie to this i can´t seem to apply the patch. I´m using this command: wget 'https://issues.asterisk.org/file_download.php?file_id=25727&type=bug' -O - | patch -p0, and the response i get is "can't find file to patch at input line 5 |
| 15:13.27 | HenrikJott | Perhaps you used the wrong -p or --strip option?". any tips? |
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| 15:14.27 | qxork | HenkikJott... is there a reason you want to stick with .22 and not the more recent version? |
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| 15:18.28 | HenrikJott | qxork: No, not really.. it´s just that i´m using it now in a production environment and it´s works fine if it weren´t for the glitch in app_mixmonitor... but mostly i´m not that familiar with upgrading and patching asterisk... or any other software under debian.. i used apt-get to get this version installed.. |
| 15:18.41 | *** join/#asterisk centoslinux (~centoslin@s0021-0018.dsl.start.no) |
| 15:18.59 | Zeeek | If you're shy about SIP use skype:vuc.me or sip:200901@login.zipdx.com |
| 15:19.03 | Zeeek | see you there |
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| 15:20.49 | HenrikJott | qxork: so no, there is no reason, but i would have no problem with upgrading if it doesn´t cause me any other problems.. |
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| 15:24.08 | Zeeek | qxork: gotta get to makeup in the green room, see you over on #vuc |
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| 15:25.39 | *** join/#asterisk casix (~casix@xenpbxedifici.adamvozip.es) |
| 15:25.42 | casix | hello |
| 15:26.06 | casix | how to use saydigit command with decimal numbers?? |
| 15:27.09 | *** join/#asterisk |amadeus| (~amadeus@75.130.147.152) |
| 15:29.29 | [TK]D-Fender | casix: CUT() |
| 15:30.14 | Tim_Toady | casix use Cut() to seperate itegral and fractorial parts of the number |
| 15:30.22 | Tim_Toady | ah |
| 15:30.28 | casix | [TK]D-Fender: thanks |
| 15:30.33 | casix | Tim_Toady: thanks |
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| 15:40.07 | *** join/#asterisk rocksfrow (~kyle@pool-96-244-85-128.bltmmd.fios.verizon.net) |
| 15:40.39 | rocksfrow | hey i'm about to do a mass calling on my pbx, with like 20 people |
| 15:40.46 | rocksfrow | just curious what to expect... |
| 15:41.08 | *** join/#asterisk jasonjjohnsonjr (~jasonjjoh@asa1.jsmc.org) |
| 15:41.10 | rocksfrow | if they all call at the same time, the PRI should be able to handle them all fine, right? as far as not throwing a busy signal to anybody? |
| 15:41.44 | rocksfrow | i'll know soon, but curious what to expect |
| 15:41.46 | wcselby | as long as you have available channels |
| 15:41.56 | rocksfrow | okay sweet :) |
| 15:42.12 | rocksfrow | i just dont' want to be embarassed when everybody tests and people get busy signals haha |
| 15:42.31 | [TK]D-Fender | PRI is a signalling.. not an implicit number of channels |
| 15:44.33 | Baylink_work | Well, kinda. PRI is in fact implicitly 23+ channels... |
| 15:44.39 | Baylink_work | *ISDN* is not. |
| 15:44.51 | qxork | The VoIP Users Conference is up... :) now connected in wideband. It's the pre-show until 12pm EST http://vuc.me |
| 15:45.10 | Baylink_work | But yes, if you have the traditional ISDN PRI DS-1, all the calls should just come in, if your trunk group is configured properly. |
| 15:45.28 | wcselby | http://www.thinkgeek.com/computing/keyboards-mice/9836/?cpg=froogle |
| 15:45.35 | wcselby | sorry wrong window |
| 15:46.50 | qxork | rocksfrow: assuming your carrier is good, your card should be able to handle it without problem. |
| 15:46.51 | Baylink_work | We're assuming, rocksfrow, that you have one or more DIDs pointed to the trunk group that serves your PRI by your carrier. If so, yes, you should be able to receive more than one simultaneous call to the same DID, and Asterisk should Just Handle It. |
| 15:46.51 | Baylink_work | Assuming that what you *do* with multiple incoming calls to the same DID doesn't block. |
| 15:46.56 | qxork | It's not 100% simultaneous, but like sip... they will be processed sequentially, but extremely quickly |
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| 15:49.15 | [TK]D-Fender | [11:44]<Baylink_work>Well, kinda. PRI is in fact implicitly 23+ channels... <-- tell that to all of my partial-pri clients.... |
| 15:49.42 | Baylink_work | That the DS-0s are not all *active* doesn't mean the DS-1 doesn't *contain* them. |
| 15:49.57 | Baylink_work | But in practice, I've never myself come across a Frac-T PRI. |
| 15:50.12 | [TK]D-Fender | Baylink_work: DS1 is just a carrier for the protocol... keeping in mind you could be dealing with T1, E1, J1, etc... |
| 15:50.30 | Baylink_work | True; I'm Americocentric, generally. |
| 15:50.31 | [TK]D-Fender | Baylink_work: Therefor the medium isn't implicit |
| 15:50.57 | *** join/#asterisk imox1234 (~imox1234@p4FC5C50C.dip0.t-ipconnect.de) |
| 15:51.20 | Baylink_work | But none of this is really germane to the question he was *actually* asking, which was "can I accept calls to the same DID down all the DS-0's I have available on my span simultaneously?" |
| 15:51.24 | [TK]D-Fender | Baylink_work: You're half-baked attempt at assuming the medium wasn't well-done... not rare to see however ;) |
| 15:51.36 | Baylink_work | "Your". :-P |
| 15:51.37 | [TK]D-Fender | suddenly feels like eating raw-meat |
| 15:51.51 | Baylink_work | (IE: neither was yours. :-) |
| 15:52.23 | [TK]D-Fender | Baylink_work: Back off... I'm a former Grammar Ranger and not sufficiently caffeinated.... |
| 15:52.37 | [TK]D-Fender | Baylink_work: And craving red meat... |
| 15:52.48 | [TK]D-Fender | starts gnashing.... |
| 15:52.59 | Baylink_work | Big Band channel on Shoutcast is playing Count Basie's "April In Paris", and /me is jonesing for a film about a black sheriff. |
| 15:53.24 | *** join/#asterisk jmacz (~jmacz@190.248.25.166) |
| 15:54.03 | casix | there is any function like sayalpha but to say comma?? |
| 15:54.30 | *** join/#asterisk hugorebelo (~hugo@200-171-132-124.completo.com.br) |
| 15:54.50 | [TK]D-Fender | casix: That would be a "recording". Perhaps you should look in the sounds folder.. |
| 15:56.10 | casix | ok |
| 15:56.13 | casix | thanks |
| 15:58.54 | Tim_Toady | casix playback(point) or (comma) |
| 15:59.37 | Tim_Toady | im not if this is part of the original sound package or the extra sounds but you better instal them both anyway |
| 15:59.53 | Tim_Toady | s/im not/im not sure |
| 16:00.23 | casix | Tim_Toady: yes, I have found it. thanks |
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| 16:04.27 | rocksfrow | my server didn't even flinch from the 15 calls it got at the same time :) |
| 16:04.52 | rocksfrow | i love PRI |
| 16:06.47 | *** part/#asterisk qxork (~qxork@adsl-70-54-28.gnv.bellsouth.net) |
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| 16:08.01 | Baylink_work | They are nice, aren't they? I have about 35 of them. |
| 16:08.41 | rocksfrow | dayamm |
| 16:09.03 | Baylink_work | Call center. 2 campuses. |
| 16:09.17 | rocksfrow | how many concurrent calls you got going at once? |
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| 16:09.57 | Baylink_work | Well, when we were fully staffed last September, almost 500. We were over-provisioned. |
| 16:10.15 | rocksfrow | damn |
| 16:10.21 | rocksfrow | how many servers |
| 16:10.24 | Baylink_work | We're drawing down as the contracts on some of the spans expire. |
| 16:10.49 | Baylink_work | I have 10 diallers here and 6 at the other campus. Newer machines; 2 sections per instead of only one. |
| 16:11.07 | Baylink_work | 30 bodies vs 46 timeslots, roughly. |
| 16:11.53 | rocksfrow | oh so its all outgoing? |
| 16:11.56 | rocksfrow | mostly*? |
| 16:11.59 | *** join/#asterisk diegomad (~mad@190.146.200.120) |
| 16:12.12 | rocksfrow | some of your vocab confused me :-p |
| 16:13.14 | *** join/#asterisk bcrisp (~bcrisp@70.102.242.138) |
| 16:13.24 | bcrisp | top of the morning |
| 16:13.32 | rocksfrow | i'm curious, with that much volume... how do you route the calls across the multiple PRIs? |
| 16:13.39 | rocksfrow | does the telco do round-robin for you? |
| 16:14.17 | af_ | hi, I am wondering if there are some channel types that exist only for 1.6.x |
| 16:14.38 | bcrisp | has anyone encountered a situation where an IVR doesn't respond immediately to a unique extension (BackGround(soundfile) followed by WaitExten() |
| 16:14.42 | af_ | channel code is changed from 1.4 to 1.6? |
| 16:14.47 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.81) |
| 16:16.16 | wcselby | if i'm still getting some echo with oslec installed, what can I do to tweak? I've got asterisk 1.4.26, dahdi 2.2.0, a digium te420, centos 5.3 64-bit, running 2x quad core processors, with three T1's coming in from our provider and the fourth slot used as a connection to our hylafax server with a brooktrout board in it. |
| 16:16.21 | Baylink_work | rocksfrow: Nearly all of it is outbound; I have one or two trunkgroups used for inbound, and yes, they put multiple trunks in the same TG, and then aim the DIDs at them, and the calls come in wherever they do; I have in the past split them between multiple servers, and just cross-decked the incoming DID via IAX to the server where they're handled. |
| 16:16.47 | Baylink_work | wcselby: Are all three Ts from the same carier? |
| 16:16.53 | Baylink_work | carrier even? |
| 16:16.56 | wcselby | af_ - I think chan_jingle / chan_gtalk may only work with ast 1.6 |
| 16:17.03 | wcselby | Baylink_work - yes |
| 16:17.24 | wcselby | Baylink_work - 1 inbound, 2 outbound, they all fail-over to each other. |
| 16:17.31 | Baylink_work | Where are you hearing the echo? |
| 16:17.38 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 16:17.48 | wcselby | a couple specific numbers outbound, sometimes randomly on inbound calls. |
| 16:17.52 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 16:18.22 | Baylink_work | Is it near-end or far end on the outbound calls? Cause that sounds like that might be at the other end, if the delay is consonant with the distance to that end. |
| 16:18.25 | Baylink_work | I've had that. |
| 16:18.36 | Baylink_work | far end's echocan fails. |
| 16:18.37 | bmoraca_work | wcselby: at that point, it's likely the originating carrier or one of the interconnected ones |
| 16:18.55 | Baylink_work | Yeah; you'll never get it to 100% |
| 16:19.23 | Baylink_work | My rule of thumb here is that I don't start looking til 1% and I don't start hammering til 3-5%. Course, we move a lot of calls. |
| 16:19.38 | bmoraca_work | wcselby: also, digium's boards suck. if you need perfect sound, buy a TA908e and use that as your CSU. |
| 16:20.08 | rocksfrow | echocan worksk great on my digium card :) |
| 16:20.46 | wcselby | Baylink_work - hmmmmm |
| 16:21.02 | *** join/#asterisk oldhack (~jfincher@cpe-24-27-56-221.austin.res.rr.com) |
| 16:21.17 | wcselby | Baylink_work - the issue is, we've got the te420 using oslec, but there is the hardware echo can module. just not sure if that will improve the situation anymore |
| 16:21.19 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 16:21.32 | wcselby | that wasn't clear |
| 16:21.44 | wcselby | we don't currently have the hw echo can module, but we could pick it up |
| 16:22.03 | wcselby | bmoraca_work - ta908e....will look into that |
| 16:22.12 | Baylink_work | Yes, but I'm not at all certain from your description that your echo problems are local, is my point. |
| 16:22.24 | wcselby | Baylink_work - gotcha |
| 16:22.45 | Baylink_work | Your echocan, unless I *badly* misunderstand what's going on, prevents *the other end* from getting echos from you. |
| 16:23.12 | wcselby | Baylink_work - i guess that's why i'm wondering if there's a way to check other than just "listening" to the call that's generating the echo. it's only one-way echo, my client's users hear the echo, the other caller doesn't. |
| 16:23.14 | bmoraca_work | Baylink_work: depending on how much delay you want to add, it can work both ways |
| 16:23.25 | Baylink_work | Hmmm. |
| 16:23.59 | *** part/#asterisk esaym (~esaym153@cpe-24-174-176-203.satx.res.rr.com) |
| 16:24.04 | Baylink_work | Generally, though, bmoraca_work, that's true, right? Your EC helps the other end; theirs helps you? |
| 16:24.13 | bmoraca_work | wcselby: better echo cancellation can fix that. the HWEC might help. you might also try MG2 |
| 16:24.27 | Baylink_work | Apparently, no, I'm wrong. |
| 16:24.28 | bmoraca_work | Baylink_work: like i said, it can work both ways, depending on how much you want to buffer |
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| 16:25.57 | wcselby | the echo isn't on every call either, it's only on some specific numbers and seemlingly random inbound calls (not sure if it's actually random or if it's just infrequent callers) |
| 16:26.18 | bmoraca_work | could be either |
| 16:26.49 | bmoraca_work | "specific numbers" seems to point to a carrier problem. but, you'll never get a resolution as you treat it that way. |
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| 16:28.00 | wcselby | bmoraca_work - that adtran 908e, that's a PRI-to-SIP device? |
| 16:28.17 | bmoraca_work | wcselby: yes. you'd use it in that capacity. |
| 16:28.40 | bmoraca_work | wcselby: the benefit is that its echo cancellation features are fantastic. it also has some good RTP monitoring capabilities |
| 16:29.15 | wcselby | i'm trying to think how that would work with our current fax setup. i do like the idea of this device though.... |
| 16:29.18 | *** join/#asterisk odenkos (~odenkos@ip-212-081-019-170.static.nextra.sk) |
| 16:29.43 | bmoraca_work | your fax server connects directly to a port on the digium card? |
| 16:30.40 | wcselby | yes, but the calls come into asterisk and then are re-routed back out to the fax server. there are only four channels licensed on the brooktrout board, and then another 20 or iaxmodem channels for additional capacitiy |
| 16:31.04 | bmoraca_work | you could continue using it in that capacity then |
| 16:31.06 | *** join/#asterisk jtodd (vg3ogczm4l@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 16:31.06 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 16:31.15 | bmoraca_work | or move everything to iaxmodem |
| 16:34.19 | *** join/#asterisk Ta^3 (~tacvbo@189.146.181.250) |
| 16:35.37 | wcselby | bmoraca_work - thanks, I'll bring this up with my client |
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| 16:38.56 | bcrisp | this is odd... |
| 16:40.58 | *** join/#asterisk DennisG (DennisG@84.30.136.208) |
| 16:46.48 | bcrisp | when i use Background() to play a message.. waitexten() doesnt register input until about 2 seconds have elapsed |
| 16:47.24 | [TK]D-Fender | bcrisp: pastebin <- |
| 16:47.36 | bcrisp | dialplan? |
| 16:47.47 | [TK]D-Fender | bcrisp: And CLI |
| 16:48.20 | bcrisp | cli isn't showing anything special, no dtmf debug if the user hits an extension very close to the start of the ivr background msg |
| 16:48.23 | bcrisp | one sec |
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| 16:53.27 | bcrisp | http://pastebin.com/Z9gs7xcr |
| 16:53.34 | bcrisp | there is the relevant part of the dialplan |
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| 16:58.41 | bcrisp | im not sure if it has something to do with the TFS provider that we recently changed |
| 16:59.05 | nny | quick q, if my phone has the option "rport enable" and I am trying to get it to work with NAT (the server is not NAT) do i want to enable that option? |
| 16:59.47 | bcrisp | before, if you dial an extension that is unique while the IVR is playing it will immediately recognize it.. now you have to wait for the IVR message to play for a second or 2 before it responds to dtmf input |
| 17:03.16 | bcrisp | if i dial the RTN directly it works as expected, but not when dialing the toll free # |
| 17:04.45 | *** join/#asterisk Z_God (~julius@2001:888:141f:0:221:5dff:fe2a:6806) |
| 17:07.26 | bcrisp | i think tfs is messing something up |
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| 17:11.31 | bmoraca_work | bcrisp: entirely possible. ask them to pass DTMF inband and see if that makes it any better. |
| 17:11.59 | bcrisp | on the phone with em right now, asking them that very question |
| 17:12.19 | bcrisp | in sip.conf i set the dtmfmode to auto for the incoming |
| 17:14.24 | bcrisp | is there a benefit of specifying rfc2833 etc or does auto work just as well? |
| 17:15.36 | *** part/#asterisk ryanlin (~ryanlin@you.better.not.nulroute.us) |
| 17:15.39 | *** join/#asterisk ryanlin (~ryanlin@you.better.not.nulroute.us) |
| 17:15.52 | ryanlin | [TK]D-Fender: are you there? |
| 17:15.57 | [TK]D-Fender | yup |
| 17:16.14 | ryanlin | [TK]D-Fender: i finally know what the context is |
| 17:16.22 | ryanlin | can you help me get this shit to work? |
| 17:19.48 | bcrisp | bmoraca_work: , yep - they ran some kind of diagnostic and it was something not working right on their end, all good now |
| 17:20.34 | [TK]D-Fender | ryanlin: I wasted half an hour waiting for you to provide what should have taken 2 minutes. And never got it. |
| 17:20.40 | bcrisp | i love when you can call customer service and they directly contact technical staff rather than opening a ticket bla bla |
| 17:20.50 | [TK]D-Fender | ryanlin: Ready to actually provide what is requested now? |
| 17:21.01 | bcrisp | negative ghost rider |
| 17:21.05 | ryanlin | [TK]D-Fender: yes |
| 17:21.18 | ryanlin | [TK]D-Fender: apologies |
| 17:25.02 | ryanlin | http://internetworkpro.org/pastebin/6394 |
| 17:26.01 | bcrisp | anyone help me out with a dial plan pattern for international in north america? |
| 17:27.38 | bcrisp | nevermind |
| 17:27.38 | Corydon76-dig | _011NXXXX. |
| 17:28.13 | bcrisp | Corydon76-dig: thanks |
| 17:28.18 | ryanlin | should i paste in the whole extensions.conf? |
| 17:28.26 | ryanlin | there are a lot of irrevelent info in there |
| 17:35.33 | [TK]D-Fender | ryanlin: That isn't your whole extensions.conf, is it? |
| 17:35.46 | [TK]D-Fender | ryanlin: Well your first line has no context... |
| 17:35.59 | [TK]D-Fender | ryanlin: And you aren't showing the failed call with SIP DEBUG.... |
| 17:36.03 | wcselby | anyone else playing pacman on google today? |
| 17:36.41 | ryanlin | oh..one sec..forgot about that |
| 17:39.18 | *** join/#asterisk sgimeno (~chatzilla@83.40.175.14) |
| 17:39.30 | ryanlin | http://internetworkpro.org/pastebin/6395 <===debug output |
| 17:39.48 | ryanlin | can you expalin further on the context for the first line? i am kinda confused |
| 17:40.14 | ryanlin | do i need to define something like this? exten => _XXXX,1,Macro(dialout-callmanager,${EXTEN}) |
| 17:40.47 | [TK]D-Fender | ryanlin: extensions.conf - exten => 3300,1,Macro(corpuser,3300) ;test-phone <- the very first line you list from extensions.conf is not within a context heading <- |
| 17:41.18 | *** join/#asterisk Netgeeks (~chris@173.11.68.155) |
| 17:41.18 | ryanlin | oh..let me take a look at that |
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| 17:45.52 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 17:46.09 | ryanlin | hmm..how do i define context heading in extensions.conf ? |
| 17:46.13 | ryanlin | like [3300] ? |
| 17:46.56 | [TK]D-Fender | RyYes |
| 17:47.01 | [TK]D-Fender | ryanlin: Yes |
| 17:47.21 | ryanlin | oh shit..there you go..thanks |
| 17:47.25 | ryanlin | one second |
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| 17:52.20 | ryanlin | http://internetworkpro.org/pastebin/6396 |
| 17:52.25 | ryanlin | here you go..that's the context for the "corpuser" |
| 17:53.44 | [TK]D-Fender | ryanlin: thats a macro, not a "context" and that isn't even referenced by anything |
| 17:54.09 | [TK]D-Fender | ryanlin: You don't seem to be looking at what peer IS being matched by your call or what context it is then looking in <------------- |
| 17:54.31 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
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| 17:59.46 | ryanlin | how do i check what peer is being matched and what context is being looked into? |
| 18:00.10 | ryanlin | http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Express+Integration |
| 18:00.14 | ryanlin | i was looking at this example here |
| 18:02.20 | [TK]D-Fender | ryanlin: Found peer 'callmanager' for '8578' from 172.30.253.196:62482 |
| 18:02.27 | [TK]D-Fender | ryanlin: Looking for 3300 in incoming (domain 172.30.253.198) |
| 18:02.33 | [TK]D-Fender | ryanlin: What do YOU see there? |
| 18:02.52 | [TK]D-Fender | ryanlin: Where is the [incoming] context in yoru extensions.conf? |
| 18:03.48 | KavanS | any suggestions for evaluating the following: if the extension/callerID dialing me is 3 digits in length, do X, if not do Y |
| 18:04.03 | ryanlin | ahhh wtf |
| 18:04.08 | ryanlin | that's what you were referring to |
| 18:04.14 | ryanlin | i see..thx! |
| 18:04.17 | ryanlin | let me go fix that really quick |
| 18:04.28 | bcrisp | can you use conditional statements in Set() ? |
| 18:04.45 | [TK]D-Fender | bcrisp: like? |
| 18:04.56 | bcrisp | id like to set a variable based on a condition |
| 18:05.04 | bcrisp | without spaghetti gotos |
| 18:05.07 | [TK]D-Fender | KavanS: LEN(), GotoIf() |
| 18:05.22 | [TK]D-Fender | bcrisp: IF() |
| 18:05.34 | [TK]D-Fender | bcrisp: ExecIf() |
| 18:05.37 | bcrisp | ok |
| 18:05.39 | bcrisp | ty |
| 18:06.01 | KavanS | [TK]D-Fender, ok, roger that, noted and googling now, thank you good sir! |
| 18:06.18 | [TK]D-Fender | KavanS: NO GOOGLING. |
| 18:06.23 | [TK]D-Fender | KavanS: CLI instructions <- |
| 18:06.32 | bcrisp | google is a liar 80% of the time |
| 18:06.46 | KavanS | ahh, I tend to use "asterisk cmd commandhere" |
| 18:06.51 | KavanS | ok |
| 18:07.50 | ryanlin | hey man |
| 18:07.54 | ryanlin | [TK]D-Fender: YOU did it! |
| 18:08.14 | ryanlin | [TK]D-Fender: it's working!!! thanks a bunch! |
| 18:08.20 | [TK]D-Fender | ryanlin: You're welcome |
| 18:08.24 | ryanlin | ;) |
| 18:08.35 | [TK]D-Fender | ryanlin: You should brush up on your dialplan basics. Chapter 5 is your friend |
| 18:08.36 | [TK]D-Fender | ~book |
| 18:08.37 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 18:13.03 | ryanlin | [TK]D-Fender: will do..thanks :) |
| 18:20.12 | *** join/#asterisk centoslinux (~centoslin@s0021-0018.dsl.start.no) |
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| 18:35.31 | seanjohn | using MYSQL(QUERY resultid SELECT \'pin\' FROM \'users\' WHERE pin=\'${input}\') can we use this with Authenticate() or do I have to create a dialplan just for authentication? |
| 18:36.07 | seanjohn | with Read() |
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| 18:39.36 | surfdue | *PUKE* |
| 18:39.39 | *** part/#asterisk surfdue (~surfdue@unaffiliated/surfdue) |
| 18:39.49 | *** join/#asterisk DelphiWorld (~Miranda@41.200.12.72) |
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| 18:41.50 | Baylink-work | gets the mop and bucket, and mops up the floor of the channel. |
| 18:44.00 | seanjohn | you don't like mysql? |
| 18:44.46 | seanjohn | my main question is, for the file to enter on Authenticate, can i use a variable? |
| 18:47.15 | *** part/#asterisk DelphiWorld (~Miranda@41.200.12.72) |
| 18:50.52 | [TK]D-Fender | seanjohn: File? Pardon? |
| 18:51.24 | seanjohn | currently, i'm using a text file for authenticate with MD5 hashes |
| 18:51.46 | seanjohn | I would like to expand to use more variables stored in a separate database from CDR |
| 18:52.02 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:52.02 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 18:52.26 | seanjohn | i was wondering if I could do authenticate(${result}) where result is a list of all the pins in the database |
| 18:53.08 | [TK]D-Fender | seanjohn: Are you expecting multiple records to match? |
| 18:53.20 | seanjohn | no, just one so I could limit to 1 |
| 18:54.00 | seanjohn | but input would have to be done to limit to 1 so I think I'm stuck with using READ() and then doing comparisons |
| 18:54.09 | [TK]D-Fender | seanjohn: Well so far your query could be taken as "returns a single value" At which point thats the only thing you authenticate as, no? |
| 18:54.29 | [TK]D-Fender | seanjohn: From what you just mentioned, it looks like |
| 18:54.47 | [TK]D-Fender | seanSo yes, you'd be implementing your own equivalent. |
| 18:54.50 | seanjohn | i can't narrow down the pins to just one without the pin being entered BEFORE authenticate |
| 18:54.51 | [TK]D-Fender | seanjohn: So yes, you'd be implementing your own equivalent. |
| 18:54.53 | seanjohn | lol |
| 18:54.58 | [TK]D-Fender | seanjohn: rather minor |
| 18:55.25 | seanjohn | yeah just trying to save code |
| 18:56.21 | seanjohn | some people are so old they can't remember their pin so I have it logging them in automatically by their CID |
| 18:56.43 | seanjohn | for voicemail |
| 18:57.20 | seanjohn | just so others in here might have the same situation, don't make ANY voicemails a blank passcode |
| 18:59.13 | seanjohn | no one calling from outside the system can pass a callerid to the system of 4 digits, otherwise they are played a privacy notice, so its secure to autologin local extensions by the callerid number |
| 18:59.37 | seanjohn | right? |
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| 19:02.25 | seanjohn | agree, fender? i have the NAPA database installed in mysql and each incoming CID number is checked against it for validity. |
| 19:06.07 | [TK]D-Fender | seanjohn: Sure, why not |
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| 19:35.17 | *** join/#asterisk xbp (pbx@irc.wastate.net) |
| 19:35.21 | xbp | good morning |
| 19:42.56 | *** join/#asterisk Schmdnebakerzzzz (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 19:44.45 | seanjohn | language,n,Set(CHANNEL(language=es)) is this correct for spanish? |
| 19:44.55 | seanjohn | good evening, here |
| 19:46.53 | *** join/#asterisk residentgrey (~residentg@72-60-202-186.pools.spcsdns.net) |
| 19:46.53 | sakejub | hello |
| 19:46.54 | residentgrey | hi |
| 19:48.04 | Schmdnebakerzzzz | hello |
| 19:48.37 | sakejub | Does anyone know if there is a way to have Asterisk not put the ANI in the From header for cnam? Asterisk is putting the ANi where the calling name would be. |
| 19:51.16 | *** part/#asterisk glwgoes (~guilherme@200.175.61.250) |
| 19:56.25 | [TK]D-Fender | seanjohn: No. Go re-read the docs on using functions |
| 19:56.55 | devmod | is there a way to set a var when using Dial? |
| 19:58.04 | [TK]D-Fender | devmod: Like...? |
| 19:59.16 | leifmadsen | devmod: SET() function? |
| 19:59.33 | leifmadsen | although likely you want to use something like M() |
| 19:59.56 | *** join/#asterisk flapjacks (~flapjacks@wsip-72-214-208-206.ph.ph.cox.net) |
| 20:02.16 | sakejub | From my carrier I get "From: <sip:8005551212@1.1.1.1>;tag=3482680293-123981" but when it leaves our Asterisk to another provider the From header is populated with From: "8005551212" <sip:8005551212@1.1.1.1>;tag=as423517e6. Just wondering if this is normal for Asterisk to be doing this. |
| 20:02.26 | Schmdnebakerzzzz | freeswitch or asterisk? |
| 20:02.35 | Schmdnebakerzzzz | which is better? |
| 20:03.02 | [TK]D-Fender | Schmdnebakerzzzz: Yes. |
| 20:03.22 | Schmdnebakerzzzz | Yes what |
| 20:03.38 | [TK]D-Fender | Schmdnebakerzzzz: Apples & oranges |
| 20:03.55 | Schmdnebakerzzzz | which do you prefer |
| 20:04.06 | [TK]D-Fender | Schmdnebakerzzzz: Asterisk |
| 20:04.19 | Schmdnebakerzzzz | why |
| 20:04.23 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 20:04.39 | Slugs_ | lol |
| 20:04.48 | [TK]D-Fender | Schmdnebakerzzzz: Because the dialplan is nice and flat |
| 20:05.23 | Schmdnebakerzzzz | i see. |
| 20:05.26 | leifmadsen | Schmdnebakerzzzz: it entirely depends on what you're using it for. Also, you're asking #asterisk what we like. We like Asterisk. |
| 20:05.44 | leifmadsen | If I was using Freeswitch I'd be in #freeswitch |
| 20:05.48 | *** join/#asterisk x303 (~X303@wifi222-144.meruwifi.fit.edu) |
| 20:06.01 | Schmdnebakerzzzz | i'm a freeswitch person but i work with asterisk |
| 20:06.08 | leifmadsen | It's like asking Americans if they like the USA or France |
| 20:06.19 | Schmdnebakerzzzz | i like france |
| 20:06.25 | Schmdnebakerzzzz | and I'm American |
| 20:06.30 | leifmadsen | you prefer France? |
| 20:06.39 | Schmdnebakerzzzz | yes, by far a better country |
| 20:06.43 | leifmadsen | then I'd move there |
| 20:06.51 | Schmdnebakerzzzz | i'd like to, it's not that easy |
| 20:07.12 | leifmadsen | in this case, the difference between asterisk and freeswitch is as easy as which application you download |
| 20:07.16 | leifmadsen | lucky for you! |
| 20:07.52 | Schmdnebakerzzzz | i'm just wondering if you guys have used freeswitch and what you think of it |
| 20:07.58 | leifmadsen | I have no used it |
| 20:08.14 | leifmadsen | Asterisk does everything I need it to do and I don't believe in solving the same problem with multiple solutions |
| 20:08.43 | leifmadsen | there may be more than one way to skin a cat, but the solution that works best for me is the way I skin the cat |
| 20:08.45 | Schmdnebakerzzzz | how many calls run concurrently through your asterisk box |
| 20:08.56 | leifmadsen | that entirely depends on what the calls are doing |
| 20:09.03 | leifmadsen | and what box it is running on |
| 20:09.37 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 20:09.41 | Schmdnebakerzzzz | one thing freeswitch has over asterisk is the CELT codec |
| 20:09.51 | leifmadsen | I don't even know what CELT is |
| 20:09.59 | leifmadsen | I'm google will turn up what it is |
| 20:10.02 | [TK]D-Fender | Schmdnebakerzzzz: How much do you care about it? |
| 20:10.04 | Schmdnebakerzzzz | cd-quality low latency |
| 20:10.10 | Schmdnebakerzzzz | about what |
| 20:10.17 | leifmadsen | I don't eve know what else would use that codec |
| 20:10.19 | [TK]D-Fender | Schmdnebakerzzzz: tHAT CODEC OBVIOUSLY |
| 20:10.36 | Schmdnebakerzzzz | i care, because you can do broadcast media using it |
| 20:10.38 | Slugs_ | doc is better w/ asterisk |
| 20:10.41 | leifmadsen | asterisk also has JACK support, but I don't use asterisk because of it |
| 20:10.45 | Schmdnebakerzzzz | it's extremely powerful when used right |
| 20:10.48 | Slugs_ | doc is better w/ asterisk |
| 20:10.52 | leifmadsen | Schmdnebakerzzzz: then you should use freeswitch for that solution |
| 20:11.02 | Schmdnebakerzzzz | well what good is jack support if it's quality isn't clear |
| 20:11.13 | *** join/#asterisk n3hxs (~HAMming@pool-71-246-12-117.phlapa.fios.verizon.net) |
| 20:11.14 | Schmdnebakerzzzz | i agree, doc is better w/ asterisk |
| 20:11.29 | leifmadsen | Schmdnebakerzzzz: if my end points aren't capable of high quality audio then what does it matter? |
| 20:11.36 | Slugs_ | i use both |
| 20:11.44 | leifmadsen | I don't use my phone system to broadcast high quality audio. I use shoutcast for that. |
| 20:11.45 | Schmdnebakerzzzz | ekiga is a softphone that supports CELT codec |
| 20:12.02 | Schmdnebakerzzzz | right but what if you wanted live callers in your broadcast |
| 20:12.11 | leifmadsen | anyways, my answer is I like Asterisk. I don't need to say anything beyond that. |
| 20:12.16 | Schmdnebakerzzzz | or the ability to broadcast from a 3g connection remotely |
| 20:12.21 | [TK]D-Fender | Schmdnebakerzzzz: Live callers from where? |
| 20:13.03 | leifmadsen | if I needed to solve a solution that asterisk couldn't, then I'd find the software that solved the solution I needed solving. It has nothing to do with what I prefer. I prefer Asterisk to solve all the problems that it can, and I'll find something else if I ever need to solve a solution it doesn't. |
| 20:13.03 | Schmdnebakerzzzz | from anywhere, freeswitch transcodes and therefore someone calling from another line can talk to someone using the CELT codec |
| 20:13.11 | leifmadsen | good for it |
| 20:13.18 | leifmadsen | I'm sure that's useful for someone |
| 20:13.24 | [TK]D-Fender | Schmdnebakerzzzz: No... where are your "callers" call's coming from? |
| 20:13.30 | Schmdnebakerzzzz | a phone |
| 20:13.32 | leifmadsen | that's not an decisive reason for why I still wouldn't use asterisk in my solutions |
| 20:13.37 | Schmdnebakerzzzz | any phone |
| 20:13.39 | [TK]D-Fender | Schmdnebakerzzzz: Where/ What kind? |
| 20:13.46 | leifmadsen | well, any phone that supports CELT you mean |
| 20:14.02 | Schmdnebakerzzzz | no, freeswitch transcodes |
| 20:14.04 | [TK]D-Fender | Schmdnebakerzzzz: Well the PSTN uses G.711. So what's the point of CELT? |
| 20:14.13 | *** join/#asterisk jmacz (~jmacz@190.125.31.45) |
| 20:14.24 | [TK]D-Fender | Schmdnebakerzzzz: Transcoding to CELT doesn't make your audio any beter <- |
| 20:14.25 | leifmadsen | my Polycom 501 doesn't support CELT, but my Polycom 335 supports G.722 |
| 20:14.33 | jaytee | Freeswitch is the greatest!!!! ( I get paid five bucks everytime I say this ) :-) |
| 20:14.41 | leifmadsen | jaytee: 50/50! |
| 20:14.42 | [TK]D-Fender | Schmdnebakerzzzz: Start with G.711 and its not going to get any beter |
| 20:14.53 | [TK]D-Fender | better* |
| 20:14.59 | leifmadsen | blueray on my 24" tube TV! |
| 20:15.02 | leifmadsen | looks amazing! |
| 20:15.07 | seanjohn | i disagree, fender :) I love g729 |
| 20:15.17 | leifmadsen | off to pick up my fiancee from work |
| 20:15.18 | seanjohn | sounds BETTER than ulaw |
| 20:15.19 | [TK]D-Fender | seanjohn: ... FAIL |
| 20:15.19 | Naikrovek | you can love it all you like, but you're losing fidelity |
| 20:15.36 | [TK]D-Fender | CRAZY PEOPLE |
| 20:15.40 | Naikrovek | hands seanjohn a pile of hi-fi 8-track |
| 20:15.54 | bmoraca_work | g729 cannot, by definition, sound better than g711 |
| 20:16.04 | Naikrovek | here here |
| 20:16.08 | seanjohn | does to me |
| 20:16.14 | Qwell | get new ears. |
| 20:16.21 | [TK]D-Fender | seanjohn: You aer clearly broken. Time to RMA... |
| 20:16.32 | seanjohn | uhmmm I've played piano for 20 years |
| 20:17.14 | WIMPy | seanjohn: Did you use ear protection then? |
| 20:17.14 | seanjohn | g729 only uses 10 kbits |
| 20:17.14 | Naikrovek | beethoven was deaf and played piano, too |
| 20:17.18 | Naikrovek | you ain't the first |
| 20:17.20 | Qwell | Naikrovek: point |
| 20:17.21 | Naikrovek | :P |
| 20:17.22 | *** part/#asterisk Schmdnebakerzzzz (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 20:17.41 | Naikrovek | kidding |
| 20:17.46 | Naikrovek | i'm sure you're not deaf but come on |
| 20:17.52 | seanjohn | call 8005900213 and see if it sounds crap though |
| 20:17.59 | Naikrovek | it won't sound crap |
| 20:18.04 | Naikrovek | g729 does not sound crappy |
| 20:18.12 | Naikrovek | but it sounds /slightly/ worse than g711 |
| 20:18.18 | Naikrovek | however! |
| 20:18.34 | Naikrovek | if you have some amp or mic or something that warms up g729 more than 711 i can see where you're coming from |
| 20:18.52 | seanjohn | technically i'm using g729A |
| 20:18.53 | Qwell | no. |
| 20:19.00 | bmoraca_work | if you are bandwidth-constricted, then g729 can sound better than g711 |
| 20:19.13 | Naikrovek | yeah that's tru |
| 20:19.14 | bmoraca_work | but that's not because of the codec, necessarily |
| 20:19.14 | Naikrovek | e |
| 20:19.15 | seanjohn | a T1 isn't constricted for voip |
| 20:19.23 | bmoraca_work | yes it is |
| 20:19.27 | bmoraca_work | if you're using g711 |
| 20:19.31 | bmoraca_work | and have more than a few calls |
| 20:19.37 | Naikrovek | 18 |
| 20:19.46 | bmoraca_work | if you're lucky |
| 20:19.48 | seanjohn | i can safely support 50 calls, in a test |
| 20:19.50 | Naikrovek | 18 G711 SIP streams are the max |
| 20:19.53 | Naikrovek | G729 can do WAY more |
| 20:20.04 | Naikrovek | about 49 over SIP, 140 over IAX2 |
| 20:20.27 | Naikrovek | used a calculator. *beam* |
| 20:20.27 | seanjohn | my carrier supports iax2 but I had terrible issues with them |
| 20:20.32 | seanjohn | using iax2 |
| 20:20.45 | *** join/#asterisk Gary_B (~IceChat7@85.211.156.225) |
| 20:20.49 | Qwell | Naikrovek: I don't buy 140 |
| 20:20.51 | bmoraca_work | right, and if you try to stick 10 g711 calls over a T1 with general internet use, you're going to have problems. not necessarily so with g729 and the same number of streams. |
| 20:21.12 | Naikrovek | Qwell: tell your employer that, their G729 page agrees with me |
| 20:21.17 | Qwell | I wouldn't buy more than ~65 or so, over IAX2 |
| 20:21.19 | seanjohn | general internet use? the T1 is only for the phone system |
| 20:21.24 | Qwell | link? |
| 20:22.05 | bmoraca_work | WTF would you buy a T1, make it a data T1, and put phone-only over it? a PRI would be a much better choice |
| 20:22.06 | *** join/#asterisk sysop1 (~gerrym@gw.futuremetals.com) |
| 20:22.10 | Gary_B | hi everyone, anyone know what kind of companies would have high landline to mob call spend apart from call centres |
| 20:22.14 | Naikrovek | Qwell: bah my memory is wrong. you're right. again. shit. |
| 20:22.17 | Qwell | hrm, yeah, you'd have to raise the packetization like crazy |
| 20:22.20 | seanjohn | well, using a high quality headset and playing music over g729, it sounds GREAT |
| 20:22.23 | Naikrovek | Qwell: http://www.digium.com/en/products/g729codec.php |
| 20:22.34 | *** part/#asterisk sysop1 (~gerrym@gw.futuremetals.com) |
| 20:22.35 | seanjohn | musiconhold bumps the speakers lol |
| 20:22.48 | Qwell | under no circumstances will G.729 sound "great" |
| 20:22.55 | Naikrovek | g722 can sound grat |
| 20:22.56 | Naikrovek | great |
| 20:23.03 | Naikrovek | certainly after you've used g729 for a while |
| 20:23.03 | bmoraca_work | wideband |
| 20:23.20 | seanjohn | i have a softphone that only supports ulaw |
| 20:23.25 | bmoraca_work | Naikrovek: i'm sure to seanjohn, g729 sounds better than g722 |
| 20:23.36 | Gary_B | x-lite supports most codecs |
| 20:23.39 | Naikrovek | heh |
| 20:23.51 | seanjohn | 3cx phone |
| 20:23.57 | Naikrovek | bmoraca_work: I burn all my CDs to G729 for highest quality |
| 20:24.11 | seanjohn | g729 is not 320 kbpx |
| 20:24.11 | Naikrovek | hah |
| 20:24.16 | Qwell | seanjohn: you'd really like LPC10 |
| 20:24.19 | Naikrovek | whatthe |
| 20:24.38 | Gary_B | i like the nch express talk softphone but the only thing wrong with it is it doesnt allow you to set codec and doesnt support a lot of them |
| 20:25.02 | [TK]D-Fender | seanjohn: http://en.wikipedia.org/wiki/G.729 - "DTMF tones, Fax transmissions, and high-quality audio cannot be transported reliably with this codec." <- since G711 can... math is looking pretty simple |
| 20:25.02 | bmoraca_work | 3cx is such a piece of shit |
| 20:25.18 | seanjohn | I like 3cx phone the most and for a terminal adapter, I prefer the linksys PAP |
| 20:25.22 | [TK]D-Fender | Qwell: Domo Arigato Mr. Roboto! |
| 20:25.36 | seanjohn | linksys pap uses g729 |
| 20:25.49 | [TK]D-Fender | seanjohn: It CAN use that codec. |
| 20:26.02 | seanjohn | that's what I said |
| 20:26.19 | [TK]D-Fender | seanjohn: I'm sure plenty of others' don't :) |
| 20:26.25 | [TK]D-Fender | loves idle features |
| 20:26.34 | seanjohn | it supports g722 too |
| 20:27.05 | seanjohn | ok, so which would you suggest? g729 or g723 cause those are the only two that my carrier supports, other than Ulaw |
| 20:27.52 | Naikrovek | if bandwidth is your main concern, .. well whatever your concern is i'd use g729 |
| 20:28.03 | [TK]D-Fender | seanjohn: G729 is higher quality than G.723 |
| 20:28.06 | bmoraca_work | to the carrier, g729. just don't pretend it sounds better. |
| 20:28.12 | bmoraca_work | (than ulaw, that is) |
| 20:28.18 | Naikrovek | the quality is much more than sufficient, it's low bandwidth, it works |
| 20:28.19 | Naikrovek | i use it |
| 20:28.33 | bmoraca_work | i use it as well for several customers |
| 20:29.52 | [TK]D-Fender | checkout time, later all |
| 20:29.54 | seanjohn | i have my global channel limit set to 50 |
| 20:30.31 | seanjohn | each customer of mine gets 2 channels |
| 20:31.13 | Gary_B | anyone know markets that use high land 2 mobile calls? |
| 20:31.35 | seanjohn | i only sell 100 channels as the chance of all being on at the same time, in call waiting, is slim |
| 20:32.47 | seanjohn | btw, who tried to call with an unknown number lol |
| 20:33.03 | Baylink-work | In fact, the wideband Poly-specific codec does sound better than 711; which isn't hard. |
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| 20:34.03 | *** part/#asterisk Gary_B (~IceChat7@85.211.156.225) |
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| 20:39.41 | Naikrovek | Baylink-work: you mean the Siren7/14 codec(s) |
| 20:39.59 | Baylink-work | I don't think so; not the name I'm familiar with. |
| 20:40.20 | Naikrovek | what name are you familar with |
| 20:40.38 | Baylink-work | I believe I meant 722 |
| 20:41.01 | Naikrovek | G722 is open. Polycom has a couple of modifications to G722 called Siren7 and Siren14 |
| 20:41.20 | Naikrovek | the sirens are proprietary, but Digium licensed them |
| 20:41.53 | Baylink-work | WP suggests that 722.1 is based on Siren (rather than the other way round) and implies strongly that 722 was not. |
| 20:42.28 | Naikrovek | yeah G722 stands on its own |
| 20:42.47 | Baylink-work | More specifically, .1 and .2 are *not* derived from 722 |
| 20:43.10 | Naikrovek | okay |
| 20:43.13 | Naikrovek | we're not arguing are we |
| 20:43.23 | Baylink-work | I believe we're in violent agreement. |
| 20:43.34 | Naikrovek | you're god damned right we are you MFer |
| 20:43.39 | Naikrovek | OUTSIDE! |
| 20:43.46 | Naikrovek | :P |
| 20:43.58 | Baylink-work | grabs his bouquet of roses and heads for the door. |
| 20:45.20 | wcselby | lol |
| 20:45.30 | wcselby | damnit, naik, i had somethin gto ask you about polycoms today |
| 20:45.34 | wcselby | but I've forgotten what it was |
| 20:45.40 | Naikrovek | in that case i know everything |
| 20:45.44 | Naikrovek | if you ask, that could change |
| 20:45.50 | Naikrovek | what with the quantum rules and all |
| 20:45.55 | wcselby | hmmmm....i'm thinking it may have had to do with time settings on polycoms |
| 20:46.02 | wcselby | not picking up the ntp time |
| 20:46.19 | Naikrovek | i've had that happen before |
| 20:46.33 | Naikrovek | don't remember the specific issues i've had so i can't say what the solutions were |
| 20:46.38 | wcselby | yeah |
| 20:46.42 | wcselby | i've had it before too |
| 20:46.48 | *** join/#asterisk Gary_B (~IceChat7@85.211.156.225) |
| 20:46.53 | Naikrovek | one time it was my NTP server not allowing connections from the phones |
| 20:46.53 | wcselby | i seem to recall the server giving the proper time, eventually |
| 20:47.04 | Naikrovek | other times it was a timezone offset setting |
| 20:47.13 | Naikrovek | but i moved that into DHCP so i know it gets set right |
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| 20:52.12 | *** part/#asterisk Gary_B (~IceChat7@85.211.156.225) |
| 20:53.45 | KavanS | hrm |
| 20:53.46 | KavanS | [May 21 13:49:13] WARNING[20168]: app_dial.c:1296 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) |
| 20:53.49 | KavanS | ^^^ any ideas? |
| 20:54.10 | KavanS | dahdi version 2.2.1, asterisk version 1.4.28 |
| 20:54.43 | KavanS | it *was* working about 30 mins ago |
| 20:54.48 | *** join/#asterisk kerx (~kerx@38.118.129.34) |
| 20:55.07 | l2cache | Anyone have any experience with Hotdesking Polycom phones? |
| 20:56.00 | Naikrovek | yeah |
| 20:56.03 | Naikrovek | a bit |
| 20:56.33 | kerx | Anyone know if it's possible to do a Auto log off like Autopause in queues.conf ? |
| 20:56.53 | *** join/#asterisk sizzers (~sizzers@c-98-209-34-63.hsd1.mi.comcast.net) |
| 20:56.55 | kerx | I'm trying to log off a user or run a System call if they don't answer a call within a time period |
| 20:57.07 | *** join/#asterisk Kyosh (~whoa@pool-96-246-230-27.nycmny.fios.verizon.net) |
| 20:57.07 | kerx | For time tracking purposes |
| 20:59.04 | Naikrovek | you can do that in the dialplan |
| 20:59.06 | Naikrovek | i believe |
| 21:00.01 | Naikrovek | ring the phone. if the user answers, the dialplan will stop there, and when hung up, progress through teh dialplan for that extension will stop |
| 21:00.03 | kerx | Naikrovek, can you give me a hint? I thought I had an idea about it, maybe I was in the right track |
| 21:00.22 | *** join/#asterisk timeshell (~timeshell@gw.lusi.on.ca) |
| 21:00.38 | Naikrovek | if the user does not answer, the phone will ring for however long, then continue through the dialplan. usually this would be to voicemail but you can do other things |
| 21:00.44 | Naikrovek | i'm not a dialplan expert |
| 21:00.49 | kerx | Naikrovek, so your essentially saying in Dialplan for extension 201 (agent201), if user doesn't answer, then run a script to log him off |
| 21:00.52 | Naikrovek | ask [tk]d-fender when he returns |
| 21:01.00 | Naikrovek | yeah i would try that |
| 21:01.21 | Naikrovek | my understanding is that dialplan execution stops when a user answers the phone, and ends entirely (without continuing) when the call is hung up. |
| 21:01.30 | Naikrovek | but when not answered, you can do things after ringing the phone |
| 21:01.36 | Naikrovek | i could be very wrong |
| 21:01.41 | kerx | so for example I have a queue called [salesq] |
| 21:01.44 | Naikrovek | i don't have any decent dialplan experience |
| 21:01.52 | kerx | so in [salesq] for there extensions I can run a macro if the user doesn't answer |
| 21:02.13 | kerx | ok got it |
| 21:02.21 | kerx | nice, i think i was on right track :) |
| 21:02.22 | Naikrovek | after the Dial(SIP/12345) line put something in that logs them out |
| 21:02.33 | Naikrovek | if they answer it'll never get to the logout line |
| 21:02.39 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 21:02.39 | Naikrovek | if they don't answer, it will get there |
| 21:02.46 | Naikrovek | i think |
| 21:02.46 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:02.51 | Naikrovek | ooh there he is |
| 21:03.12 | kerx | Naikrovek, I think you are perfectly correct on this |
| 21:03.18 | kerx | And I think I will give that a shot |
| 21:03.23 | l2cache | Anyone have any experience with Hotdesking Polycom phones? |
| 21:03.29 | kerx | I'm basically trying to build Time Management using Asterisk |
| 21:03.37 | Naikrovek | l2cache: yes, i said so earlier |
| 21:03.40 | kerx | And your route is perfect |
| 21:03.41 | Naikrovek | l2cache: what's your question |
| 21:03.53 | sizzers | I've hotboxed with polycom phones |
| 21:04.04 | l2cache | sizzers, LOL. ty |
| 21:04.10 | *** join/#asterisk odenkos (~odenkos@ip-212-081-019-170.static.nextra.sk) |
| 21:04.20 | sizzers | try reading the tutorial about asking questions l2cache.... |
| 21:05.11 | Naikrovek | l2cache: just ask |
| 21:05.16 | l2cache | Naikrovek, I'm looking to have 2 separate identities on one phone. With only 1 active at one time. And have the user change it from the phone itself. I've done this with Snom, but screw Snom. |
| 21:05.32 | Naikrovek | yes i've done that with polycom, but not with vanilla asterisk |
| 21:05.39 | l2cache | I'm using FreePBX |
| 21:05.54 | Naikrovek | the phone will always show the PHONE extension, but you can log agents in and out and the phone will ring for those agents |
| 21:06.32 | Naikrovek | there are tutorials for this out there |
| 21:06.44 | Naikrovek | and there's a special thing you need to do in freepbx to make it happen |
| 21:06.45 | l2cache | The identity of the phone itself though.Not by using agents |
| 21:07.06 | Naikrovek | you want to change the identity of the phone itself? |
| 21:07.17 | Naikrovek | don't think that's possible. doing that on polycom requires a reboot |
| 21:07.30 | *** join/#asterisk ttwhy (~tekkno@p4FECE5B8.dip.t-dialin.net) |
| 21:07.35 | [TK]D-Fender | Naikrovek: that is a technicality |
| 21:07.40 | ttwhy | Hi, iam getting always after 2 seconds of ringing. "Spawn extension (capi-in, MYMSN, 2) exited non-zero on 'CAPI/ISDN1#02/MYMSN-10'", how do i increase this timeout? i can't find something in the capi.conf or extension.conf |
| 21:07.45 | [TK]D-Fender | Naikrovek: Quite possibly however |
| 21:07.59 | Naikrovek | possible? |
| 21:08.01 | Naikrovek | without reboot |
| 21:08.02 | Naikrovek | really |
| 21:08.03 | Naikrovek | hrm |
| 21:08.03 | l2cache | reboot is fine |
| 21:08.12 | Naikrovek | reboot is fine? easy |
| 21:08.36 | Naikrovek | hell in that case my wimpy perl scripts can do hotdesking as a side-effect |
| 21:08.42 | Naikrovek | if reboots are okay |
| 21:08.58 | l2cache | Well, they need to change this simply from the phone |
| 21:09.00 | l2cache | is the end goal |
| 21:09.12 | l2cache | Snom has a simple way of doing this, from the phone buttons |
| 21:09.37 | Naikrovek | well you could have a phone microbrowser app that you could do that from, then reboot the phone and viola |
| 21:09.46 | Naikrovek | s/viola/voila/ |
| 21:10.19 | Naikrovek | this is one of the things i'd like to see https delivered, dynamically generated phone configs for |
| 21:10.55 | l2cache | any way to put just one identity of the Polycom on DnD? |
| 21:10.58 | l2cache | that would work for me |
| 21:11.02 | Naikrovek | sign into microbrowser app, tell it "yo, i'm at this phone now." then reboot, and the app sends you the right config for your exten |
| 21:11.04 | l2cache | and use the other...switch. etc. |
| 21:11.10 | *** join/#asterisk jmacz (~jmacz@190.125.31.45) |
| 21:11.19 | Naikrovek | l2cache: never done that |
| 21:12.01 | Naikrovek | ponders this application he just imagined, and how to write it |
| 21:12.11 | Naikrovek | think it could be done fairly simply |
| 21:12.23 | Naikrovek | if reboots are not a problem, then this would work for hotdesking |
| 21:12.51 | Naikrovek | the traditional way is to separate devices from users in freepbx and manage them separately |
| 21:13.03 | Naikrovek | but this isn't #freepbx |
| 21:13.09 | Naikrovek | so i won't bore anyone |
| 21:13.20 | Naikrovek | suddenly feels like a bit of a channel veteran |
| 21:13.27 | Naikrovek | i knew something! |
| 21:13.29 | Naikrovek | yay |
| 21:13.42 | Naikrovek | leaves while he's ahead, locking his computer |
| 21:17.38 | KavanS | what would I use in replacement of SetVar ? |
| 21:18.01 | l2cache | KavanS, Set( ) |
| 21:18.12 | KavanS | ok, roger that |
| 21:19.21 | [TK]D-Fender | [17:12]<Naikrovek>the traditional way is to separate devices from users in freepbx and manage them separately <- Except not everything follows.... |
| 21:22.05 | devmod | how can i go around enabling h323 ? |
| 21:22.40 | *** join/#asterisk saghul (~saghul@60.Red-88-8-5.dynamicIP.rima-tde.net) |
| 21:24.50 | [TK]D-Fender | devmod: vi chan_h323.conf |
| 21:28.29 | *** join/#asterisk riddlebox (~james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 21:38.34 | kerx | [TK]D-Fender, Please see PM. |
| 21:38.47 | KavanS | kerx, you just pm'd [TK]D-Fender ? - oh shit. |
| 21:39.00 | kerx | KavanS, heh |
| 21:39.10 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 21:40.01 | kerx | I guess I got the cold shoulder... |
| 21:40.03 | kerx | Ouchie. |
| 21:44.07 | *** join/#asterisk ManxPower (~manxpower@36.sub-75-254-189.myvzw.com) |
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| 22:02.43 | ManxPower | "but the results are excellent, even at relatively low bitrates. 4-6Mbps is more than enough for " I don't know about you, but 4Mbps is not a "low bit rate" in my world. |
| 22:03.14 | *** join/#asterisk smooth_penguin (~smoove@59.95.31.177) |
| 22:03.18 | florz | for a particle accellerator, it certainly is! |
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| 22:17.00 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 22:24.14 | saxa | hey, has something drastically changed in gsm protocol from asterisk 1.4.x to 1.6.2.x ? |
| 22:24.51 | saxa | I had a working 1.4.x install and then I upgraded it on both ends to 1.6.2.7 |
| 22:25.28 | saxa | on 1.4.x with iax2 connected 2 boxes, I had the perfect sound and was hearing the other box perfectly |
| 22:25.45 | saxa | thats not anymore true for 1.6.2.7 install |
| 22:26.17 | saxa | now when I call a voicemail I hear just the first few letters of the word |
| 22:26.20 | *** join/#asterisk prometheanfire (~mthode@70.88.176.253) |
| 22:26.35 | saxa | before I could listen it fully |
| 22:26.40 | prometheanfire | anyone know of an issue with 1.6.2.6 eating up udp ports and not giving them back? |
| 22:26.49 | saxa | any ideas ? |
| 22:27.27 | Kyosh | prome, its hungry? |
| 22:30.16 | TJNII | GRAA UDP PORTS! NOM NOM NOM NOM! |
| 22:30.55 | prometheanfire | OM NOM NOM NOM NOM |
| 22:30.57 | Kyosh | heh |
| 22:31.24 | prometheanfire | ya, it fills up in a day |
| 22:32.40 | prometheanfire | I recompiled it without ooh323 and fax n'stuff |
| 22:33.24 | prometheanfire | it looks like it might be stable now |
| 22:33.32 | prometheanfire | n'stuff |
| 22:33.40 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
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| 23:01.26 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 23:01.58 | ruben23 | hi anyone have idea what this means---> [May 21 22:54:08] WARNING[3362]: chan_sip.c:12370 handle_response_invite: Received response: "Forbidden" from '"1400" <sip:000000000@vp.thinktel.ca>;tag=as1e2092e1' |
| 23:02.37 | ruben23 | i have setup my sip extension 1400 on my asterisk and i can login with it, but when i cal i see that. |
| 23:03.06 | *** join/#asterisk jksM (jks@193.189.93.254) |
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| 23:16.28 | ttwhy | can someone tell me whats the better card at all. A DAHDI HFC-S Card or a chan_capi AVM Fritz Card? I want to receive calls and dont want to use it as a NT device. |
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