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| 00:17.43 | josefig | Hello guys, I'm getting this error trying to outbound a call using SIP provider: Ignoring audio media offer because port number is zero this is my trace: http://paste2.org/APpXFMgv please refer to lines 111 and 112 |
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| 00:32.05 | navaismo | josefig, in the sdp exchange th audio is 0 as the warning said, --->m=audio 0 RTP/AVP 18 101 |
| 00:32.25 | josefig | navaismo: what should I change ? |
| 00:32.58 | navaismo | i guess is an client issue, have you tried with another client? |
| 00:33.13 | navaismo | s/an client/a client/ |
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| 00:35.22 | josefig | not at all, the SIP is using g729 and I don't have another SIP Client for Mac and for Android. |
| 00:36.23 | igcewieling | josefig: how many g729 licenses do you have? |
| 00:36.57 | josefig | igcewieling: no one, I'm using a free one. |
| 00:38.16 | igcewieling | there is no free g729 for Asterisk. |
| 00:39.01 | igcewieling | get it working with a supported codec first. |
| 00:39.02 | josefig | I saw the binary for free. |
| 00:39.19 | navaismo | like i saw spartacus for free ;) |
| 00:39.32 | igcewieling | josefig: any such discussion of the illegal g729 codec will get you banned from this channel by Digium. I suggest you do not mention it again |
| 00:39.34 | navaismo | so unsupported |
| 00:39.40 | josefig | where can I buy the license? |
| 00:39.48 | navaismo | digium store |
| 00:39.54 | igcewieling | You can purchase G729 licenses from Digium |
| 00:40.15 | igcewieling | But get it working with a supported codec first, you don't NEED g729. |
| 00:40.46 | josefig | the SIP provider uses g729 |
| 00:41.49 | navaismo | any decent provider at least use GSM & ulaw & g729 |
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| 00:43.49 | josefig | well this not :( |
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| 00:44.54 | navaismo | hmm so try the 15 bucks license |
| 00:45.17 | josefig | 1 license == 1 concurrent call ? |
| 00:45.26 | navaismo | is that a NAt enviroment the rtp ports are opened? and the nat settings configured or all in the lan? |
| 00:48.50 | josefig | is a Nat environment |
| 00:51.25 | igcewieling | g729 licenses are $10/concurrent channel |
| 00:51.56 | igcewieling | josefig: is the asterisk server behind nat, the client behind nat, or both are behind nat? |
| 00:52.26 | josefig | no, only client behid nat |
| 00:52.33 | navaismo | so sad |
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| 00:52.42 | navaismo | you need to open sip and rtp ports on the pbx side |
| 01:03.07 | igcewieling | . |
| 01:03.22 | igcewieling | navaismo: is wrong |
| 01:03.48 | igcewieling | josefig: Unless your NAT box is messing up SIP all you should have to is set nat=yes in sip.conf for it to work. |
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| 01:17.05 | Rahail | Ok I give up ... any one here who can create me dial plan to round robin sip trunk I will pay for it |
| 01:18.59 | Rahail | igcewieling |
| 01:19.16 | Rahail | can you do it please.. i give up i dont think i would figerout any time soon |
| 01:19.22 | igcewieling | I do not do consulting, but you are welcome to ask on the asterisk-biz mailing list |
| 01:20.11 | Rahail | igcewieling do oyou know any one else who can give this quick.. i am sure mailing list will take lot time |
| 01:21.50 | igcewieling | Rahail: with your level of understanding, nothing will be quick. |
| 01:22.04 | igcewieling | the TASK is complex as well |
| 01:22.38 | Rahail | I understand however if any one can make this then i wouldnt have to touch it again |
| 01:22.40 | Rahail | one time deal |
| 01:22.49 | igcewieling | It would take me an hour to understand your system, another hour to write the code, and several hours to test it. And that assumes you understand enough to explain your setup. |
| 01:23.47 | Rahail | they system is like i recive incoming calls and I want them to route to my multi sip trunk I haves in round robin so one trunk dont get all this call |
| 01:24.24 | Rahail | I am reciving call as Iax2 and I need to send this call to SIP |
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| 01:25.35 | Rahail | !pastebin |
| 01:25.41 | Rahail | ~pastebin |
| 01:25.42 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 01:26.03 | Rahail | this how i have it right now |
| 01:26.08 | Rahail | http://pastebin.com/NrBrLKUs @ igcewieling |
| 01:26.17 | igcewieling | delightful, the guy who wrote our international fraud detection software apparently didn't know that some international calls from the usa are not dialed with 011 |
| 01:26.43 | igcewieling | thankfully it "failed closed" instead of "fail open" |
| 01:30.21 | Rahail | igcewieling http://pastebin.com/NrBrLKUs can you see this and see if you can change your midn plz |
| 01:31.40 | igcewieling | *** Rahail has been added to Ignore list. |
| 01:31.45 | igcewieling | much better. |
| 01:32.02 | Rahail | lol |
| 01:34.22 | josefig | lol useful help |
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| 03:09.57 | Rahail | what ma i doing wrong here |
| 03:10.02 | Rahail | its keep going on trunk2 |
| 03:10.04 | Rahail | http://pastebin.com/7s3Mm3aj |
| 03:10.27 | Rahail | i am using some one else example and edit with my need |
| 03:13.18 | WIMPy | Because you're starting at 1 not at 0. |
| 03:15.19 | Rahail | WIMPy which line |
| 03:16.14 | WIMPy | Which line? The whole thing. |
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| 03:16.55 | WIMPy | You have defined a TRY0. That will never be used as it starts at 1. |
| 03:17.54 | Rahail | i change it now it fail |
| 03:18.38 | WIMPy | What did you change? |
| 03:18.50 | Rahail | tries |
| 03:18.57 | Rahail | =0 to 1 |
| 03:19.16 | WIMPy | No. Then you will start at 2. |
| 03:19.21 | WIMPy | Change your globals. |
| 03:19.31 | WIMPy | Start at TRY1. |
| 03:20.29 | Rahail | each trunk have 8 channel limit |
| 03:20.40 | Rahail | i want this go round robin |
| 03:20.41 | *** join/#asterisk nix8n82 (~AndChat27@24.143.11.81) |
| 03:20.56 | Rahail | exten => _X.,n,gotoif($[${GROUP_COUNT(${DIALSTRING})} >= 2]?nextone) ; have we used up the allowed calls on this |
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| 03:21.00 | Rahail | if put 8 |
| 03:21.13 | linocisco | hi all |
| 03:21.33 | WIMPy | Yes, that will change it from 2 to 8 per try. |
| 03:21.49 | Rahail | so if i leave it 1 |
| 03:21.57 | Rahail | then it will go one bye one right |
| 03:22.03 | Rahail | and if i have total 24 calls coming |
| 03:22.19 | linocisco | i want the users to press 91 and wait for dial tone or if CO line is free , if 1st line is busy, * should automatically select next available CO line. how can I configure that? |
| 03:23.21 | WIMPy | Rahail: If you set it to one it will use the second try as long as the first is in use. |
| 03:23.22 | linocisco | i want the users to press 91 and wait for dial tone to check if CO line is free and call desired destination, if 1st line is busy, * should automatically select next available CO line. how can I configure that? and if all lines are busy, it should say all lines are busy and quit |
| 03:23.39 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 03:23.54 | Rahail | letsay 2nd one is also in use and 1st in also in use and 3rd one is also in use |
| 03:24.01 | Rahail | so when 4th call come what will happend |
| 03:24.04 | WIMPy | linocisco: That's what groups are for. |
| 03:24.36 | linocisco | Rahail, WIMPy i m honestly very new. advise me on typical setup |
| 03:24.40 | WIMPy | Rahail: Nothing. |
| 03:25.00 | Rahail | so if i put 1 then i am only limit to 3 calls= 3 trunk |
| 03:25.12 | Rahail | 3 trunks = i have 24 call limit |
| 03:25.16 | WIMPy | linocisco: You put your lines in to groups. Then you just dial that group. |
| 03:25.26 | WIMPy | Rahail: yes |
| 03:25.37 | Rahail | WIMPy yes on which question |
| 03:25.38 | linocisco | WIMPy, is there any sample link? |
| 03:25.55 | WIMPy | linocisco: And in order to get dialtone, you need to use a ! pattern. |
| 03:26.19 | WIMPy | Rahail: If you have 3 tries and set the limit to 1 yu can have a maximum of 3 calls. |
| 03:26.33 | WIMPy | linocisco: The sample chan_dahdi.conf. |
| 03:26.36 | linocisco | WIMPy, that is also i dont know what !pattern is. should I configure it under extensions.conf under which context? |
| 03:26.56 | Rahail | then i should copy paste the try |
| 03:27.17 | Rahail | try4=siptrunk1 |
| 03:27.24 | Rahail | tryp-5siptrunk2 |
| 03:27.26 | WIMPy | linocisco: That's for your extension, yes. Have you read the book? |
| 03:27.26 | Rahail | stuff like that |
| 03:27.47 | WIMPy | Rahail: Yes, you can add as many as you like. |
| 03:27.57 | Rahail | WIMPy thank you so much |
| 03:27.58 | Rahail | :) |
| 03:28.50 | WIMPy | Rahail: If you also have inbound calls on the same account, you have to assigne them to the appropriate group as well to get them counted. |
| 03:29.18 | linocisco | Rahail, WIMPy . Dear bros, based on my questions, u know and understand well. but I dont know what u r saying. I dont really understand where to declare which under which file , under which context |
| 03:30.02 | WIMPy | The contexts are yours. There is no default, so you're the only one who can aswer that question for your dialplan. |
| 03:30.04 | Rahail | WIMPy last quesiton |
| 03:30.06 | Rahail | what does this line do |
| 03:30.07 | Rahail | exten => _X.,n,gotoif($[${GROUP_COUNT(${DIALSTRING})} >= 8]?nextone) |
| 03:30.52 | WIMPy | That's the limit of simultaneous calls per "TRY". |
| 03:31.07 | WIMPy | thought you already got that |
| 03:32.06 | Rahail | if is limit there |
| 03:32.13 | Rahail | per trunk |
| 03:32.14 | Rahail | ? |
| 03:32.16 | Rahail | limit |
| 03:32.19 | WIMPy | yes |
| 03:32.53 | Rahail | then i am wrong |
| 03:32.54 | Rahail | like this |
| 03:32.55 | Rahail | http://pastebin.com/7YtKwFRq |
| 03:33.04 | Rahail | total 3 trunk i have 24 port |
| 03:34.21 | WIMPy | Just define the 3 accounts and set the limit to 8. |
| 03:34.38 | Rahail | 1st way |
| 03:34.41 | Rahail | or http://pastebin.com/7YtKwFRq |
| 03:34.41 | WIMPy | Now you have 4*8. |
| 03:35.14 | Rahail | where you get 4 from ? |
| 03:35.28 | WIMPy | You used each trunk four times. |
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| 03:36.08 | Rahail | <PROTECTED> |
| 03:36.11 | WIMPy | You can do such a thing if you want to use them round robin instead of prioritizing them. |
| 03:36.16 | Rahail | then it will round robin 2 call each turnk |
| 03:36.48 | WIMPy | Yes. That would make 4*2 changing every 2 calls. |
| 03:37.11 | Rahail | 4*2*3= 24 right |
| 03:37.21 | WIMPy | yes |
| 03:37.27 | Rahail | cool |
| 03:37.28 | Rahail | thank you |
| 03:37.48 | WIMPy | 12 trunks (even if they are the same 4 times over) * 2 calls. |
| 03:38.21 | Rahail | 12 try each try 2 calls = 24 |
| 03:38.23 | Rahail | now i am clear |
| 03:39.16 | linocisco | Rahail, WIMPy . what r u two talking about? i couldnt folow ur brains |
| 03:39.35 | linocisco | Rahail, WIMPy i m still not clear |
| 03:39.48 | WIMPy | Rahails dialplan. |
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| 03:41.21 | Rahail | linocisco last 3 days |
| 03:41.26 | Rahail | i been trying round robin trunk |
| 03:41.28 | Rahail | finaly got it |
| 03:41.31 | Rahail | :) |
| 03:42.15 | linocisco | Rahail, WIMPy , hi bros, finally i got confused coz I dont really understand what u two are talking about as newbie |
| 03:43.24 | Rahail | linocisco i am not even sure how to explain u |
| 03:43.26 | WIMPy | You did realise that it was another conversation? |
| 03:43.35 | Rahail | but all i can tell its working |
| 03:46.08 | linocisco | Rahail, somebody said ChanIsAvail(Zap/2&Zap/1) |
| 03:46.19 | linocisco | what is that? |
| 03:46.28 | WIMPy | Ancient |
| 03:46.47 | WIMPy | zaptel has been replaced by dahdi many years ago. |
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| 03:50.04 | linocisco | WIMPy, i dont understand whatever for my question or setup |
| 03:51.11 | WIMPy | Look at your chan_dahdi.conf. |
| 03:51.19 | WIMPy | You put your channels in to groups there. |
| 03:51.28 | WIMPy | That's all you have to do. |
| 03:51.50 | WIMPy | No need for any manual testing. |
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| 04:59.48 | chesty | is woomera module needed if asterisk is all on one box? |
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| 06:57.21 | Rahail | http://pastebin.com/YrZAfBDB |
| 06:57.25 | Rahail | can some one see it what I am doign wrong |
| 06:57.28 | Rahail | WIMPy |
| 06:57.33 | Rahail | you around please chekc http://pastebin.com/YrZAfBDB |
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| 07:00.30 | kaldemar | Rahail: gotoif($[${DIALSTRING} = ]?donehere) <-- invalid |
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| 07:01.10 | Rahail | wat would be the valid one |
| 07:01.30 | kaldemar | ${TRY${TRIES}} <-- invalid |
| 07:01.53 | kaldemar | Rahail: you must have something on the both sides of = when comparing stuff. |
| 07:02.23 | kaldemar | for example "${DIALSTRING}" = "" or ${DIALSTRING}x = x |
| 07:02.37 | Rahail | kaldemar can you tell me which way it should be |
| 07:02.40 | Rahail | can u give me that please |
| 07:02.54 | kaldemar | oh wait, ${TRY${TRIES}} is not invalid. you have defined those as globals. |
| 07:06.01 | Rahail | what m i missing |
| 07:06.44 | kaldemar | i already told you. which part didn't stick? |
| 07:07.02 | Rahail | which one |
| 07:07.06 | Rahail | can u tell me the valid patern |
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| 07:07.19 | kaldemar | i didn't say anything about patterns. |
| 07:07.29 | kaldemar | < kaldemar> Rahail: you must have something on the both sides of = when comparing stuff. |
| 07:07.39 | kaldemar | < kaldemar> Rahail: gotoif($[${DIALSTRING} = ]?donehere) <-- invalid |
| 07:09.33 | Rahail | kaldemar i am not that exprince so ifyou can tell me how can i fix it |
| 07:09.38 | Rahail | should i remoe the = |
| 07:10.07 | kaldemar | < kaldemar> for example "${DIALSTRING}" = "" or ${DIALSTRING}x = x |
| 07:10.43 | kaldemar | that's how you test if something is empty. isn't that what you want? |
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| 07:12.05 | derjanni | Bonjour everybody! |
| 07:12.23 | derjanni | Is giving every SIP peer a secret sufficient security when my * is publicly available? |
| 07:12.32 | derjanni | just talking about * not about the linux in general |
| 07:12.56 | kaldemar | depends on the rest of your setup. generally no. |
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| 07:14.10 | kaldemar | http://svn.digium.com/svn/asterisk/tags/11.3.0/README-SERIOUSLY.bestpractices.txt |
| 07:14.13 | kaldemar | that's a start. |
| 07:16.22 | derjanni | k thanks |
| 07:25.15 | derjanni | Great tip, thank you so much. |
| 07:25.40 | derjanni | If I set allowguest=no that still allows inbound calls through my SIP provider to get through, right? It just means unauth'd users cant call, right? |
| 07:26.07 | hrolf | I'm not getting any logging on the console/CLI, what might be the problem (Asterisk CLI.) |
| 07:33.25 | wdoekes | derjanni: depends on whether you've configured your sip provider as a user/peer and you match their calls properly |
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| 07:34.08 | wdoekes | you should start by having those calls land in a different context than the "default" context |
| 07:34.21 | wdoekes | if that works, you're set |
| 07:40.39 | derjanni | wdoekes: they provider is fixed that all inbound calls, call 3 local peers |
| 07:41.21 | derjanni | Is that a brute-force attempt? |
| 07:41.30 | derjanni | [Apr 18 07:34:03] NOTICE[3372]: chan_sip.c:22653 handle_request_invite: Failed to authenticate device 2000000<sip:2000000@178.202.xx.xx>;tag=28866f69 |
| 07:41.33 | derjanni | ? |
| 07:42.03 | wdoekes | could be |
| 07:42.29 | derjanni | that extension doesnt exist and the ip is the ext one from the firewall |
| 07:43.02 | derjanni | would it work if I put * on any other than 5060? e.g. 6050 ? |
| 07:43.17 | derjanni | Ive also put my SSH on non-std ports |
| 07:43.39 | derjanni | these brute-forcers are just sucking up my bandwidth - I only have 150Mbps down and 20mbps up |
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| 08:09.44 | hrolf | Is there any guide or tutorial or book which introduces someone who has no previous telephony experience? |
| 08:09.54 | hrolf | from start... |
| 08:12.03 | kaldemar | ~book |
| 08:12.03 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 08:17.21 | hrolf | kaldemar: A quick view at the contents didn't reveal any such thing. |
| 08:17.54 | hrolf | kaldemar: I'm looking more for something like, which explains analog, digital, ISDN, PRI, VoIP etc. |
| 08:19.08 | kaldemar | http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/OutsideConnectivity_id271202.html |
| 08:19.35 | kaldemar | http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/OutsideConnectivity_id291235.html |
| 08:19.42 | kaldemar | http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-OutsideConn-SECT-1.html |
| 08:20.00 | kaldemar | but yes, maybe not the best for telephony 101. |
| 08:20.43 | hrolf | kaldemar: I see it now |
| 08:21.15 | *** join/#asterisk ghost75 (~trechber@dslb-178-002-149-170.pools.arcor-ip.net) |
| 08:21.29 | hrolf | kaldemar: Thanks. If you have any other suggestions that too would be appreciated. Need to get up a new dev up to speed. |
| 08:23.13 | kaldemar | hrolf: nop, sorry- |
| 08:23.27 | hrolf | kaldemar: No problem. |
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| 08:56.29 | help_me_pls | hello everybody... i configured 3 new phones with the ending 7281, 7282 and 7283. when you call 7280 all three ring.... everything is working... but when you call out, people see our main number (7272) instead of 7280. |
| 08:56.54 | help_me_pls | how do i configure asterisk so that calls coming from those 3 phones are shown as coming from 72 80? |
| 08:58.15 | help_me_pls | for example: i call my cellphone from my work phone... it shows 7272.... O.K....... then i call my cell phone from one of the 3 new phones.... it still shows 7272.... not O.K..... i want it to show 72 80 |
| 08:58.19 | kaldemar | you set the caller id to a valid value that your provider allows. |
| 08:58.23 | help_me_pls | i hope you understand what i mean |
| 08:58.46 | help_me_pls | we have a 100 range so i think ishould be able to do that |
| 08:58.54 | help_me_pls | where do i change the caller id? |
| 08:59.20 | kaldemar | in channel configuration file (for example sip.conf) or in dialplan. |
| 08:59.36 | help_me_pls | ok ill try it, thank you |
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| 09:06.34 | help_me_pls | okay... i cant seem to find it.... in sip.conf theres "callerid =" but thats just the name that gets shown for internal calls.... in extensions.conf i dont see it at all |
| 09:07.02 | help_me_pls | where exactly in extensions.conf do i change the outgoing caller id for 3 phones |
| 09:07.20 | help_me_pls | ? |
| 09:08.43 | help_me_pls | okay i think i found something.... |
| 09:09.29 | help_me_pls | http://pastebin.com/66A3bg4u |
| 09:09.36 | help_me_pls | is this it? |
| 09:10.04 | kaldemar | it's not just the name. it's both name and number. usually if you have for example four digit internal numbers which belong to a block of public numbers, you use the short ones in your pbx and add the rest to the caller id number when dialing out. like Set(CALLERID(num)=12345${CALLERID(num)}) |
| 09:10.05 | help_me_pls | how would i have to change that so outgoing calls from those 3 phones show up as 7280 instead of 7272 |
| 09:10.50 | help_me_pls | yes youor right theres ofcourse the number: its liike this : name <282> |
| 09:11.14 | kaldemar | yes, that sets it statically to the 7272 value. what you need is to replace 7272 with ${CALLERID(num)}, assuming you use 4 digits internally. |
| 09:11.34 | help_me_pls | so using the above pastebin and the phone 282.... how should it look |
| 09:12.03 | help_me_pls | we use 3 digits internaly |
| 09:12.10 | kaldemar | Set(CALLERID(number)=0447227${CALLERID(num)}) |
| 09:21.14 | help_me_pls | okay i understand.... maybe not.... please correct me if im wrong... so i add this to extensions.conf: http://pastebin.com/hRa7PTTt and at the top of extensions.conf i add this http://pastebin.com/WzDwaBHa (this is a copy of what already there with 80 added) and the in sip.conf i change the "context" of the 3 users that need the 7280 number to inspectron80 |
| 09:21.52 | help_me_pls | is this correct or am i doing something completely wrong? |
| 09:22.45 | help_me_pls | to be clear: all 3 phones should show up as 72 80 on the outside |
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| 09:31.12 | help_me_pls | does silence mean im wrong? im to scared to just try it x) |
| 09:33.25 | wdoekes | help_me_pls: you have backups, right? |
| 09:33.32 | wdoekes | (or versioning, even better) |
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| 09:34.10 | wdoekes | (1) try it (2) it doesn't work? git stash and reload asterisk config |
| 09:34.42 | help_me_pls | if by backups you mean copying out the config before i change it.... then yes.... otherwise no |
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| 09:37.01 | twanny796 | which is the opensource cdr program |
| 09:39.32 | WIMPy | No. which tells you the path of the program. |
| 09:47.16 | kaldemar | help_me_pls: i wrote Set(CALLERID(number)=0447227${CALLERID(num)}), not Set(CALLERID(number)=0447227280) |
| 09:48.41 | kaldemar | help_me_pls: there's not need to make a separate outgoing-calls-cablecom-inspectron context for each phone. use one that sets the caller id according to the caller. |
| 09:51.21 | help_me_pls | well after a brief shock (everything stopped working) and a core restart now it works |
| 09:51.31 | help_me_pls | at least on the phone i tried |
| 09:51.56 | help_me_pls | im gonna change the other 2 now and well see if it still works |
| 09:51.58 | help_me_pls | ^^ |
| 09:52.26 | help_me_pls | im not making a seperate outgoing-calls-cablecom-inspectron for each phone |
| 09:52.29 | help_me_pls | only 1 |
| 09:53.04 | help_me_pls | and then at the very top i copied the [inspectron] context block thing |
| 09:53.24 | help_me_pls | and named it inspectron80 pointing to the new outgoing-calls-cablecom-inspectron-80 |
| 09:53.53 | help_me_pls | then in SIP.conf i can just change context = inspectron to context = inspectron80 |
| 09:54.01 | help_me_pls | and they show the correct number |
| 09:54.10 | help_me_pls | is this a bad wy to do it? if yes why? |
| 09:54.15 | help_me_pls | *way |
| 09:54.26 | kaldemar | that way you have to make many changes instead of just one. |
| 09:55.03 | kaldemar | if you do it the way i told you, you don't have to touch sip.conf. just change a single line in extensions.conf and that's it. |
| 09:55.58 | help_me_pls | hmm i dont completely understand your method.... could you please ELI5 ^^? |
| 09:56.43 | help_me_pls | and my method doesnt seem to work anyway.... incoming calls to 7280 get hung after 1 second now |
| 09:56.51 | help_me_pls | dangit |
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| 09:58.59 | help_me_pls | okay i changed everything back.... lets see if it still works ^^ |
| 10:00.29 | kaldemar | in the outgoing context you need to change the line that sets the caller id. i've already told you what to put there. |
| 10:07.15 | help_me_pls | im not really comfortable changing that.... and wouldnt that change the outgoing number of ALL phones?? |
| 10:07.25 | help_me_pls | we have ~15 phones |
| 10:07.36 | help_me_pls | all but 3 should show 7272 as outgoing |
| 10:07.47 | help_me_pls | those 3 should show 7280 |
| 10:19.52 | kaldemar | it would change it for all phones. maybe using a few lines like ExecIf($["${CALLERID(num)}" = "272"]?Set(CALLERID(num)=0447227280)) is what you want. or just put 280 in the caller id fields of those phones in sip.conf. |
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| 10:24.29 | *** join/#asterisk sjs205 (~sjs205@host109-156-231-40.range109-156.btcentralplus.com) |
| 10:25.47 | sjs205 | Morning all, by the fact that my sipusers have no voicemail accounts, I'm guessing I need to populate the realtime database automatically... is there anyway I can make it default that all sip users get a voicemail account? |
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| 10:58.44 | sjs205 | Can anyone recommend a TTS engine for use with asterisk? I'm thinking about espeak, but only because I've "played" with this before... |
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| 11:18.04 | igcewieling | sjs205: Cepstral works well |
| 11:18.45 | igcewieling | sjs205: Asterisk is not a PBX. It is a TOOLKIT which allows you to build a PBX. It does not do much "automatically" |
| 11:18.51 | sjs205 | igcewieling, that is proprietry... right? |
| 11:19.08 | igcewieling | sjs205: any TTS which sounds decent will be commercial. |
| 11:19.15 | sjs205 | igcewieling, Yeah, I'm starting to realise that ;) |
| 11:19.59 | sjs205 | igcewieling, sure... I guess that based on my usecase we could always just record the sounds we need... in time... but for now, I don't really care how good it sounds... |
| 11:20.32 | sjs205 | I guess festival has the simpliest integration... so I wil give that a try... |
| 11:20.36 | sjs205 | :D |
| 11:22.47 | igcewieling | We are on year 2 of our Asterisk deployment project. Should be finished in about another year |
| 11:25.37 | *** join/#asterisk taylorbyte2013 (~cyberninj@49.176.99.143) |
| 11:26.18 | igcewieling | only have about 300 more endpoints to migrate over. |
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| 11:30.26 | sjs205 | igcewieling, wow... that is some project! what are you doing, migrating one PBX over to asterisk? |
| 11:30.50 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 11:31.18 | igcewieling | no. Migrating our core SIP server to Asterisk. The endpoints are a mix of Asterisk (about 70 boxes) and Adtran endpoints (most of the rest) |
| 11:31.33 | sjs205 | Why dosn't asterisk support 'ctrl-d' to exit the cli? |
| 11:31.44 | sjs205 | So how many endpoints do you have? |
| 11:32.03 | igcewieling | Last count was around 350 - 400 |
| 11:32.05 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 11:32.16 | sjs205 | igcewieling, and what was your sipserver? |
| 11:32.20 | sjs205 | as in software? |
| 11:32.28 | igcewieling | Nextone/Genband/Voiplogic MSX |
| 11:32.40 | sjs205 | No idea, but it sounds grand... |
| 11:32.47 | sjs205 | So are you a PBX engineer? |
| 11:32.55 | igcewieling | Sort of. |
| 11:33.21 | jkroon | hi guys, when reporting RTCP frames via AMI (busy writing a patch to improve the events being generated), should I output 64-bit NTP timestamps as a seconds.useconds single value (useconds calculated from least-significant NTP word) or as two separate fields (retaining higher accuracy)? |
| 11:33.30 | sjs205 | sweet... three years though, what is that, about one endpoint every 3 days |
| 11:33.35 | igcewieling | my point is deploying Asterisk takes time, even on smaller systems. When you add Realtime it gets more complex |
| 11:34.29 | sjs205 | igcewieling, are you just moving and endpoint and then popping down the pub for 3 days? sounds good... |
| 11:34.33 | igcewieling | no, most of the time was spent on the core AGI scripts and the GUI, we only started migrating endpoints a few months ago. Also I don't work on it full time -- I also do customer setup and support. |
| 11:34.48 | sjs205 | igcewieling, well my system is extemely simple, but that has taken me 3 weeks... |
| 11:35.13 | sjs205 | and even now, I still only have some of the core functionality... |
| 11:35.40 | sjs205 | What I am actually looking for is an automated alarm management system for those in need of care... |
| 11:35.59 | sjs205 | They reaise a trigger in their house, and a call gets made to them and to a carer... |
| 11:35.59 | igcewieling | My boss said yesterday that "Switch Manager (our Asterisk system) makes our old system look like a toy." made me feel good. |
| 11:36.08 | sjs205 | :) |
| 11:36.22 | sjs205 | All my boss ever says to me is "Have you done it yet?" |
| 11:36.23 | sjs205 | ha |
| 11:36.24 | igcewieling | and the old system cost $250k |
| 11:36.27 | sjs205 | ouch! |
| 11:36.47 | sjs205 | Mind you, how big is your team? and how much will the conversion take? |
| 11:36.48 | jkroon | igcewieling, all of my clients (that use half the features we provide) claim that we kick everything else out there (ZA at least) to the curb. So yea, gotta agree with that. |
| 11:36.50 | igcewieling | commercial carrier grade SIP systems are not cheap. |
| 11:37.01 | mjordan | jkroon: you should have a look at oej's pinefrog work, and maybe talk to him about it. He's done a lot of work on the RTCP events |
| 11:37.07 | igcewieling | The "team" is me. |
| 11:37.11 | *** join/#asterisk threesome (~threesome@customer-79-127-150-148.net.angelnet.cz) |
| 11:37.12 | mjordan | as well as RTCP in general :-) |
| 11:37.17 | jkroon | mjordan, where can i locate him? |
| 11:37.23 | jkroon | asterisk-dev? |
| 11:37.28 | mjordan | yup |
| 11:37.29 | sjs205 | So, you for 3 years part time... you gonna bill them $250? ;) |
| 11:37.41 | mjordan | jkroon: I linked your issue in Jira to his |
| 11:37.49 | jkroon | mjordan, thanks. |
| 11:37.51 | mjordan | np |
| 11:39.07 | igcewieling | sjs205: no, but I get bonuses when we meet specific milestones |
| 11:39.42 | sjs205 | A quick question regarding sip... I need one sip endpoint to make a call to my asterisk PBX... from this I need asterisk to call two numbers provided by the sip endpoint, one will be IVR and another moh... when the IVR connection presses a button, I want the IVR and MOH connections to join and speak to each other... how easy would you say that is? |
| 11:40.22 | sjs205 | igcewieling, My boss never says "here's your bonus either" :) |
| 11:43.50 | igcewieling | sjs205: the support contract on the existing system is $10k/yr so we are saving a chunk of money |
| 11:44.21 | igcewieling | the boss didn't like my idea that I could get a percent of that savings each year 8-) |
| 11:44.43 | sjs205 | igcewieling, okay, sure... are you a contractor or employee? |
| 11:44.52 | sjs205 | Damn bosses! |
| 11:44.53 | igcewieling | employee |
| 11:45.07 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 11:45.10 | sjs205 | Employed specifically for this task? |
| 11:45.12 | igcewieling | When I did contracting I made a LOT more money, but I like this job |
| 11:45.40 | igcewieling | no, my "job" is setting up Adtran endpoints, endpoint Asterisk boxes and support. |
| 11:45.47 | sjs205 | Sure, I am thinking about going into contract work... I am on a contract now, but I mean short term contract... |
| 11:46.05 | sjs205 | 3 months here, 6 mnoths there... sounds fun! |
| 11:46.08 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 11:46.16 | igcewieling | Excluding the sales people I like almost everyone i the company. |
| 11:46.25 | igcewieling | The sales people need to be taken out back and beaten. |
| 11:46.29 | sjs205 | But my current contract is two years and I get to work from home all the time... so I'm not sure how I would feel about having to travel! |
| 11:46.47 | igcewieling | I work from home full time. |
| 11:46.52 | sjs205 | igcewieling, I think that is true of sales people the world over! |
| 11:47.03 | coppice | kill them. if you just beat them they'll be back after sick leave |
| 11:47.18 | sjs205 | igcewieling, were you generally working from home when you were contracting too? |
| 11:47.29 | sjs205 | where are you (country)? |
| 11:47.44 | igcewieling | sjs205: 4 weeks home / 2 weeks on-site when I did contracting |
| 11:47.46 | igcewieling | USA |
| 11:48.05 | sjs205 | igcewieling, that wouldn't bee so bad... I could live with that! :D |
| 11:48.44 | sjs205 | I'm in the uk, so I'm really not sure how contract work would compare... I'm also more software/hardware design... |
| 11:54.55 | sjs205 | So, with SIP, is it possible o encapsulate data such as the name of a client and 2 alternative numbers that should be dialed by asterisk? |
| 11:56.51 | igcewieling | no |
| 12:02.01 | sjs205 | igcewieling, OUCH! So I will need an application to connect directly to asterisk using something other than sips... |
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| 12:11.39 | igcewieling | no, Asterisk can dial multiple destinations at the same time. |
| 12:12.10 | igcewieling | sjs205: maybe you should read The Book |
| 12:12.11 | igcewieling | ~book |
| 12:12.12 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
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| 12:13.53 | _zoom_ | did anyone test dtmf on asterisk sipgate ? |
| 12:14.54 | sjs205 | igcewieling, buy that are you saying that I haven't got a hope a realising this functionality? :) |
| 12:15.36 | igcewieling | No, I'm saying that functionality (dialing multiple destinations at once) is such a basic feature of Asterisk that by asking the question you've demonstrated a basic lack of reading the docs |
| 12:16.17 | itgrl | Morning all |
| 12:17.04 | sjs205 | igcewieling, haha, Indeed! |
| 12:17.35 | sjs205 | I've mostly been reading all the stuff about getting the server up and running, and not really looked into actual implimentation detail... |
| 12:17.52 | sjs205 | Actually, I've started with some dialplan today...so that should help! |
| 12:18.23 | itgrl | The book will be a lot of help to you. I still reference it from time to time even though I've been using it for years. |
| 12:18.25 | sjs205 | Is that book the same as the online one? |
| 12:19.01 | sjs205 | yeah, I only found that book yesterday... I wish i round it 2 week ago! |
| 12:19.14 | sjs205 | *found |
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| 12:30.55 | alhemicar | hi, does anyone have experience with patching recent versions of dahdi for HFC-S support? |
| 12:32.46 | alhemicar | what would be best way for this, I would like to be able in the future to update dahdi regularly etc... |
| 12:32.53 | alhemicar | OS is Debian |
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| 12:44.44 | *** join/#asterisk heisemberg (d51beb92@gateway/web/freenode/ip.213.27.235.146) |
| 12:45.51 | heisemberg | Hi everyone. I'm looking for Asterisk-MSCRM Dynamics integration. Anyone can recommend anything? Thank you! |
| 12:46.50 | *** join/#asterisk RadJackson (~RadJackso@fktel.cust.jaguar-network.net) |
| 12:47.22 | [TK]D-Fender | ~book |
| 12:47.23 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 12:47.26 | [TK]D-Fender | heisemberg, ^^^ |
| 12:47.53 | RadJackson | Hi everyone , i'm trying to develop a .call file , which i copy to /var/spool/asterisk/outgoing/ in order to make the call happen, I would like to know if it possible to leave a message on the voicemail directly , without making the phone ring, I appreciate your help! |
| 12:48.27 | [TK]D-Fender | RadJackson, What phone are you talking about? |
| 12:48.43 | [TK]D-Fender | RadJackson, You also never ever "copy" a call file. They need to be MOVED into the spool folder |
| 12:49.31 | RadJackson | the call file contains ; Channel : SIP/XX/NumberToDial ; CallerId : CallingNumber |
| 12:49.48 | RadJackson | so i would like to leave a message in "NumberToDial" voicemail directly |
| 12:49.58 | [TK]D-Fender | RadJackson, When you call that URI ... it rings whatever it is going to ring. You have no control over that |
| 12:50.39 | [TK]D-Fender | RadJackson, Thre is no way for you to magically deposit voicemail to my cell-phone company's voicemail server.... |
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| 12:51.07 | RadJackson | Oh ok , thank you very much |
| 12:54.23 | sjs205 | Does festival have any "nice" voices? |
| 12:54.28 | heisemberg | [TK]D-Fender: There is nothing in the book about crm. |
| 12:54.57 | [TK]D-Fender | heisemberg, The same as for what the definition of "integrate" is |
| 12:55.02 | [TK]D-Fender | heisemberg, It is a vague term |
| 12:55.30 | *** join/#asterisk serafie (~erin@nat/digium/x-yekggxqmljskrkdi) |
| 12:55.46 | [TK]D-Fender | heisemberg, I can integrate Asterisk with my desk. I just install it on a small computer and place it on top of a stack of loos papers and *BAM* now my papers don't fly away any more |
| 12:56.01 | [TK]D-Fender | heisemberg, Asterisk is whatever you make it to be. For me it's a jukebox & coffee-maker |
| 12:56.17 | [TK]D-Fender | heisemberg, So what you mean by "integrate" could mean anything based on your goals. |
| 12:56.44 | [TK]D-Fender | heisemberg, Which could be so unique that there is no book to explicitly say "here's a precise sample" |
| 12:58.04 | igcewieling | heisemberg: Asterisk is not a PBX, you know that right? It is a TOOLKIT which allows you to BUILD a PBX. Think of it as a car which comes in a big box and you have to put it together. |
| 12:58.27 | [TK]D-Fender | More like a LEGO set. The castle is in your mind. |
| 12:59.25 | itgrl | I typically use, it's a blank canvas with all the paints you need, you paint the picture. |
| 12:59.34 | sjs205 | igcewieling, how many times do you find yourself typing that in a day? ;) |
| 13:00.29 | sjs205 | I think of it as a ikea gallows... start to unpack it and you'll be hanging yourself soon enough! |
| 13:00.31 | sjs205 | ;) |
| 13:01.14 | heisemberg | [TK]D-Fender, igcewieling Come on guys, are you kidding me? LOL. Google CRM integration, all have the same features more or less. I was asking for someone that has used (you didn't as I see) any software and can recommend one by its features, support, price... |
| 13:01.48 | heisemberg | I mean, life is not white and black ;-) |
| 13:02.00 | heisemberg | *or |
| 13:02.02 | heisemberg | xD |
| 13:02.14 | [TK]D-Fender | heisemberg, It does as much as you make it do. |
| 13:02.37 | *** join/#asterisk darksk1ez (~mhb@fsf/member/darkskiez) |
| 13:02.40 | [TK]D-Fender | heisemberg, You are assuming there is some sort of pre-determined functionality. There really isn't. |
| 13:02.43 | heisemberg | Yeah, you can make Asterisk fly, i know... |
| 13:02.43 | *** join/#asterisk [Outcast] (~anonymous@50-200-130-22-static.hfc.comcastbusiness.net) |
| 13:03.13 | jmetro | Its not great at tapdancing though, it has two left LEGS |
| 13:04.10 | [TK]D-Fender | heisemberg, Asterisk is LEGO as I described. You build everything from scratch unless someone is distributing a ton of their own code as some sort of "package" for some kind of feature you find valuable. |
| 13:04.28 | [TK]D-Fender | heisemberg, * doesn't inherently talk to anything you don't set up in its entirety |
| 13:04.29 | itgrl | heisemberg > When I think of integrating Asterisk with something else, I first think of what do I want this integration to do. Then I develop that integration strategy. |
| 13:06.32 | [TK]D-Fender | heisemberg, Avoid using word like "integration" as though that implies any specific functionality. I could think of dozens of things I could do with the two of them. Not all of them entirely sane. |
| 13:06.35 | heisemberg | itgrl: We won't develop anything. We want a already-done software |
| 13:06.45 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 13:06.45 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 13:07.10 | [TK]D-Fender | heisemberg, Then you should just keep Googling to see if someone has made something that does what you specifically want. |
| 13:07.57 | itgrl | Then Asterisk itself isn't what you are looking for. Since it is just a blank canvas for you to paint your own masterpiece. |
| 13:08.13 | itgrl | You can always hire a consultant integrator to do it for you. |
| 13:08.32 | jmetro | i really hate IT buzzwords.. |
| 13:09.07 | itgrl | heh |
| 13:09.08 | [TK]D-Fender | itgrl, I've never integrated a consultant before :) |
| 13:09.16 | itgrl | lol |
| 13:09.34 | itgrl | me either |
| 13:09.49 | [TK]D-Fender | itgrl, I should probably start spec-ing out exclusively opposite-sex IT firms immediately ;) |
| 13:10.00 | [TK]D-Fender | has awesome plug-ins |
| 13:10.09 | jmetro | its like calling the person at McDonalds a Local Retail Sales and Customer Service Associate |
| 13:10.31 | jmetro | you either got a systems admin, an engineer, or a tech |
| 13:10.57 | itgrl | blushes |
| 13:14.11 | sjs205 | My fav is the vision enhancement engineer... AKA, a window cleaner! |
| 13:14.47 | Katty | hello my asterisk does not work at all how to fix plz?? only srs answers pls, is urgent thx. |
| 13:15.36 | itgrl | ponders what does not work means. |
| 13:15.48 | jmetro | Katty> Pics or it didnt happen |
| 13:15.53 | Katty | hello deary |
| 13:15.56 | Katty | hugs jmetro |
| 13:16.00 | bulkorok | it is somehow reassuring that katty always needs help :-) |
| 13:16.12 | Katty | yes. we can all use a good therapist. |
| 13:16.28 | itgrl | agrees with Katty. |
| 13:16.35 | jmetro | bulkorok: that kind of help...we do not offer. |
| 13:16.37 | Katty | and how is fender bender this morning? |
| 13:16.41 | Katty | is he being nice? |
| 13:16.45 | bulkorok | :-) |
| 13:16.49 | Katty | eyes [TK]D-Fender |
| 13:16.58 | *** join/#asterisk edong23 (~quassel@mptc-dhcp-50-220.mptelco.com) |
| 13:17.15 | bulkorok | this helos me very often: http://www.wickes.co.uk/invt/215726/?source=123_75 |
| 13:17.22 | bulkorok | helos/helps |
| 13:17.31 | carrar | heh |
| 13:17.37 | Katty | carrar: YOU |
| 13:17.39 | Katty | why i outta... |
| 13:17.40 | Katty | just... |
| 13:17.44 | Katty | hug you to bits! |
| 13:17.46 | Katty | hugs carrar |
| 13:17.52 | carrar | w00t!! |
| 13:18.00 | carrar | hugs KattieRoooo back |
| 13:18.08 | [TK]D-Fender | Katty, Mew. |
| 13:18.09 | Katty | carrar: how's stuff n things n life |
| 13:18.30 | carrar | stuff n things n life are ok!! |
| 13:18.43 | Katty | horay! |
| 13:18.44 | carrar | other then the wife is sick |
| 13:18.54 | carrar | so guess not ALL ok |
| 13:18.58 | Katty | [TK]D-Fender: excited for van halen tribute? |
| 13:19.15 | Katty | carrar: well that's crappy :< hope she feels better soon. what germs did she pick up? |
| 13:19.24 | [TK]D-Fender | Katty, RUNNING WITH THE DEVIL!!! |
| 13:19.27 | carrar | some cough and fever thing |
| 13:19.36 | Katty | grooves with [TK]D-Fender |
| 13:19.45 | Katty | carrar: ugg. fever things are the worst. |
| 13:19.58 | [TK]D-Fender | Katty, Friend of mine from WAY back is the singer. He *is* David Lee Roth when he's up on stage.... |
| 13:20.28 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.140) |
| 13:20.44 | youjelly | '-' |
| 13:20.51 | carrar | "WAY back" as in you don't hang around him anymore cause he is so weird? :) |
| 13:20.56 | carrar | heh |
| 13:21.48 | igcewieling | OT: (/msg me if you prefer) can anyone recommend 1-port and 8 or 16 port IP KVMs which were not designed by Satan Himself? |
| 13:22.23 | carrar | Satan does a lot of work lately these days |
| 13:22.35 | [TK]D-Fender | carrar, 15 years |
| 13:23.34 | Katty | [TK]D-Fender: sounds like you will have a blast then! |
| 13:24.54 | *** join/#asterisk bombev (~User@PPPoE-Static-40-132.UnicsBG.Net) |
| 13:25.05 | bombev | Hi |
| 13:25.44 | [TK]D-Fender | Katty, Yar! |
| 13:26.09 | igcewieling | carrar: our 1-port IP KVM always sees key-down events, but often doesn't see keeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeey-up events |
| 13:29.44 | [TK]D-Fender | igcewieling, http://www.quickmeme.com/meme/3tz1ei/ |
| 13:32.54 | carrar | igcewieling: http://www.blackbox.com/Store/Results.aspx/search-VNC%60%60si_product |
| 13:34.35 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-ksjfmurccawuhrco) |
| 13:34.35 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:37.02 | igcewieling | I'm looking for people with experience with specific KVMs. We got into this issue because we just bought some KVM that looked good in the specs. |
| 13:38.23 | jmetro | igcewieling: avocent or nothing |
| 13:38.56 | leifmadsen | synergy! :) |
| 13:39.21 | igcewieling | you have experience with them? |
| 13:39.34 | jmetro | avocents, yes |
| 13:39.42 | igcewieling | the other KVM we have uses active X and steals keystrokes from all other tabs in the browser window. |
| 13:39.49 | Katty | jmetro: i read that as avocados. |
| 13:40.04 | igcewieling | in addition to crashing every 20 mins or so. |
| 13:40.20 | jmetro | Avocado KVM, so you can guacamole across multiple boxes with only one physical taco? |
| 13:40.44 | Katty | sounds good to me! |
| 13:40.47 | Katty | gets the taco sauce. |
| 13:41.35 | igcewieling | leifmadsen: you mean this? http://en.wikipedia.org/wiki/Synergy_(software) without video support I'd not call it a K*V*M |
| 13:41.40 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 13:41.44 | leifmadsen | ya I was trolling |
| 13:41.45 | leifmadsen | it's not a kvm |
| 13:42.01 | igcewieling | leifmadsen: assume I just called you bad names. |
| 13:42.11 | leifmadsen | I always assume that |
| 13:42.20 | Katty | call him a guacamole taco head. |
| 13:42.36 | carrar | who needs video anyhow!! |
| 13:42.51 | igcewieling | leifmadsen: you literally wrote the book on Asterisk, you don't get called bad names unless you REALLY deserve it. |
| 13:42.59 | jmetro | real linux pros dont even need *command lines* |
| 13:46.28 | igcewieling | chan_telepathy to the rescue! |
| 13:47.18 | help_me_pls | so im still stuck on this outgoin caller id bs, and i could really need some help. we have about a dozen phones. all phones but 3 should show the ending 7272 when i call my cellphone (for example) those other 3 should show 7280. this is how it configured right now: http://pastebin.com/RdH60zG3 |
| 13:47.53 | help_me_pls | so i want to make only the phones 281, 282 and 283 show 7280 as the outgoing number, but i failing again and again |
| 13:48.26 | *** join/#asterisk eXcAliBuR (~eXcAliBuR@206.162.174.6) |
| 13:48.29 | [TK]D-Fender | help_me_pls, There is nothing conditional in there at all |
| 13:48.30 | eXcAliBuR | BOO! |
| 13:48.37 | eXcAliBuR | haha scared you |
| 13:48.47 | help_me_pls | yes that just how it is right now |
| 13:48.57 | help_me_pls | right now all phones show as 7272 |
| 13:49.34 | [TK]D-Fender | help_me_pls, so make another pattern to match your cell-phone # |
| 13:50.00 | igcewieling | help_me_pls: I suggest you use setvar=EXTCID=15551212 (or whatever) feature in sip.conf for each of your peers to specify the external callerid for each peer, then Set(CALLERID(num)=${EXTCID}) before your external Dial command |
| 13:50.07 | help_me_pls | all calls that come from those phones should show as 7280... not only those that are going to may cell |
| 13:50.27 | [TK]D-Fender | help_me_pls, http://pastebin.com/Jw869qKL |
| 13:50.37 | [TK]D-Fender | help_me_pls, make MORE extens |
| 13:53.05 | help_me_pls | like this: http://pastebin.com/6gPQ7Jak |
| 13:53.06 | help_me_pls | ??? |
| 13:53.08 | *** join/#asterisk deo (~deo@112.198.90.69) |
| 13:53.48 | bulkorok | then you have to dial 281 or 282 or 283 to change the caller id... |
| 13:54.29 | bulkorok | try it with _1.,1, for the 7280.... then your phone can dial 0 for the one and 1 for the other callerid |
| 13:55.01 | James87 | help_me_pls: have you considered trying working with freepbx first? |
| 13:55.02 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 13:56.03 | help_me_pls | 3 phones belong to another department.... when you call 7280 only those 3 ring.... but when they call out it shows as 7272 (the number that makes all the others ring) |
| 13:56.21 | *** join/#asterisk FireAndIce (~FireAndIc@182.156.228.157) |
| 14:01.16 | [TK]D-Fender | help_me_pls, http://pastebin.com/XnEmm1VP |
| 14:01.23 | [TK]D-Fender | help_me_pls, Your first wording was poor. |
| 14:01.31 | [TK]D-Fender | help_me_pls, This matches your now clearer description |
| 14:01.32 | sjs205 | Dialplan is lovely! :) |
| 14:01.37 | [TK]D-Fender | help_me_pls, match BY callerid |
| 14:02.15 | [TK]D-Fender | help_me_pls, On the assumption that you set your SIP peers to have the callerid matching those #'s you gave |
| 14:06.16 | help_me_pls | i want those 3 to always show as 7280 when you call out.... i got it working before, but in the process fucked our normal number.... what i did was this: http://pastebin.com/mDSSeNgh silly stuff i do when i dont know what im doing ^^ |
| 14:06.22 | help_me_pls | ill try that |
| 14:07.27 | *** join/#asterisk chris_n (~Chris@koha/developer/chris-n) |
| 14:08.22 | chris_n | is anyone doing any sort of sip or other trunking directly into their local E911 call center? |
| 14:08.29 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 14:08.42 | [TK]D-Fender | help_me_pls, Easiest is jsut use SetVar in your peers as you were already told and use that in your Set() |
| 14:08.48 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.126.47) |
| 14:09.25 | *** join/#asterisk vlad_sta_ (~vlad_star@91.206.59.141) |
| 14:10.22 | *** join/#asterisk k612 (~K610@cred.epid.ucl.ac.be) |
| 14:18.57 | *** join/#asterisk MalMen (~MalMen@2.82.255.245) |
| 14:19.11 | MalMen | there is any api that i can use to make calls from one extencion to one trunk ? |
| 14:19.28 | bombev | Hi I have purchased g729 codec, then I inserted into sip.conf -> disallow=all , allow=ulaw, allow=alaw, allow=gsm, allow=g729 |
| 14:19.55 | bombev | and after that I see 2013-04-18 15:53:25] WARNING[15487] translate.c: No translator path from alaw to unknown |
| 14:19.59 | bombev | [2013-04-16 18:22:03] WARNING[27299] codec_g729a.c: out of G.729 decoder licenses |
| 14:20.14 | bombev | what is the reason |
| 14:20.35 | [TK]D-Fender | MalMen, AMI Originate, Call Files <- Go read up on these |
| 14:21.18 | [TK]D-Fender | MalMen, Next and extensions is a line in your dialplan. It doesn't normally magically cause calls to happen all by itself. A call has to hit the dialplan and then ... ti does whatever it does. |
| 14:21.31 | igcewieling | bombev: did you register your license? |
| 14:21.36 | bombev | yes |
| 14:21.42 | igcewieling | how many licenses? |
| 14:21.48 | bombev | <PROTECTED> |
| 14:21.48 | bombev | 0/0 encoders/decoders of 2 licensed channels are currently in use |
| 14:21.48 | bombev | Licenses Found: |
| 14:22.37 | igcewieling | bombev: pastebin the CLI output of a failed call and don't enable all those extra codecs. |
| 14:25.06 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 14:25.09 | bombev | igcewieling well if I leave only g729 in sip.conf with 2 licensed channels |
| 14:25.24 | bombev | it is no so good because i have 30 extensions |
| 14:25.57 | igcewieling | bombev: you cann't just "failover" when you run out of g729 licenes. calls will simply fail |
| 14:26.33 | bombev | yes I know |
| 14:27.39 | bombev | isn't it possible to use g729 only for one call, and for other calls to use ulaw,alaw? |
| 14:27.43 | igcewieling | decide what codecs you want and only allow those. For example there is seldom any point in allowing both ulaw AND alaw. |
| 14:27.50 | igcewieling | or GSM for that matter |
| 14:29.21 | bombev | well I wanna use g729 for one concurent call and for other calls |
| 14:29.24 | bombev | to use alaw and ulaw |
| 14:30.37 | *** join/#asterisk navaismo (~navaismo@189.191.250.167) |
| 14:31.09 | [TK]D-Fender | <bombev> isn't it possible to use g729 only for one call, and for other calls to use ulaw,alaw? <- no |
| 14:31.09 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:31.39 | [TK]D-Fender | bombev, If the peer's order would cause G.729 to be the combined preferred codec, it will ALWAYS be chosen and when you run out, game over |
| 14:31.52 | *** join/#asterisk kelebeks (~askin@pis-vi.tk) |
| 14:34.23 | bombev | [TK]D-Fender makes sense |
| 14:34.45 | bombev | so I have to purchase a lot of g729 channels |
| 14:34.57 | [TK]D-Fender | bombev, Or better restrict your use |
| 14:35.12 | bombev | and then I should leave only g729 in sip conf |
| 14:35.27 | kaldemar | setting one of the SIP_CODEC variables and keeping count could be worth a try. |
| 14:35.37 | bombev | [TK]D-Fender you mean to remove allow=g729 from sip conf |
| 14:36.35 | igcewieling | kaldemar: the SIP_CODEC variables are a blue bells feature |
| 14:41.01 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:41.03 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:41.03 | [TK]D-Fender | bombev, Look at what peers use it and why |
| 14:41.11 | [TK]D-Fender | bombev, avoid unnecessary transcodes |
| 14:41.12 | help_me_pls | thank you fender that worked perfectly |
| 14:41.27 | [TK]D-Fender | help_me_pls, Ready your dialplan basics. This is right in the pattern.... |
| 14:41.31 | [TK]D-Fender | Read* |
| 14:41.52 | bombev | how shall I avoid transcodes |
| 14:43.11 | igcewieling | bombev: you really can't. for example, the ringing sound you hear has to be transcoded |
| 14:43.40 | kaldemar | igcewieling: "blue bells feature"? |
| 14:43.42 | bombev | so there are two options |
| 14:44.04 | bombev | to remove allow=g729 from sip conf and to buy more g729 channels |
| 14:44.06 | bombev | correct |
| 14:44.38 | igcewieling | kaldemar: a feature which looks SO hot and you are all excited to be using it, then realize once you've sent a lot of time trying to get it working that it simply won't do what you want. |
| 14:45.13 | kaldemar | igcewieling: it won't make you coffee but it lets you change the codec. |
| 14:45.34 | igcewieling | kaldemar: in some situtiaons it will change the codec, in some situations it won't. |
| 14:46.14 | igcewieling | SIP_CODEC_INBOUND is the most problematic |
| 14:46.41 | kaldemar | what exactly is problematic about it? |
| 14:47.35 | *** join/#asterisk taylorbyte2013 (~cyberninj@49.176.99.143) |
| 14:48.01 | igcewieling | the fact the inbound codec is often set in stone before the call gets to the dialplan |
| 14:49.16 | igcewieling | kaldemar: have you actually used the SIP_CODEC_ variables? |
| 14:49.24 | kaldemar | igcewieling: yes. |
| 14:49.41 | igcewieling | your situation must be one where it worked the way you want. |
| 14:49.56 | kaldemar | that's the point... |
| 14:50.30 | bombev | guys |
| 14:50.37 | igcewieling | We were trying to make the outbound codec be the same as the inbound codec. total disaster. |
| 14:50.46 | bombev | am I right about this to remove allow=g729 from sip conf and to buy more g729 channels |
| 14:51.01 | bombev | those are two options |
| 14:51.04 | igcewieling | bombev: buy more g729 channels |
| 14:51.25 | bombev | is it worth it to buy more g729 |
| 14:51.47 | bombev | than using ulaw alaw |
| 14:53.33 | *** join/#asterisk jkroon (~jkroon@kerberos.uls.co.za) |
| 14:54.37 | igcewieling | that is not a technical issue. |
| 14:55.05 | igcewieling | We NEED to support g729 so we bought g729 hardware transcoding cards (720 channels). You may not NEED g729. |
| 14:56.29 | *** part/#asterisk eXcAliBuR (~eXcAliBuR@206.162.174.6) |
| 14:57.00 | bombev | igcewieling how much it colst this g729 hardware transcoding cards (720 channels). |
| 14:57.12 | igcewieling | bombev: lots and lots of money |
| 14:57.16 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.142) |
| 14:57.25 | bombev | ok dont need to know :) |
| 14:57.43 | igcewieling | I don't know. We needed it, so cost was not a major factor. |
| 14:57.47 | [TK]D-Fender | <bombev> am I right about this to remove allow=g729 from sip conf and to buy more g729 channels <- stop saying like it's ALL OR NOTHING |
| 14:57.59 | [TK]D-Fender | bombev, I said RESTRICT it, not ELIMINATE IT |
| 14:58.04 | *** join/#asterisk slidesinger (~slidesing@c-69-141-208-250.hsd1.nj.comcast.net) |
| 14:58.11 | bombev | [TK]D-Fender how to restrict it |
| 14:58.17 | [TK]D-Fender | bombev, Stop having phones use G.729 where * has to transcode. Use it where you HAVE to. |
| 14:58.24 | [TK]D-Fender | bobse-, On EACH peer. |
| 14:58.28 | [TK]D-Fender | bombev, ^ |
| 14:58.39 | vlad_starkov | Question: Is it possible to store peer config (I mean peer without registration) in DB using Asterisk realtime? |
| 14:58.56 | igcewieling | vlad_starkov: yes. We did that for a while. |
| 15:00.30 | bombev | [TK]D-Fender ok |
| 15:00.58 | vlad_starkov | igcewieling: Hi there :-) So I've added peer into realtime db. When I try to make inbound call from outside world to my Asterisk through this peer, the call i declined due to the reason that asterisk doesn't know about this peer and do not process the inbound call at all. |
| 15:01.27 | igcewieling | vlad_starkov: sip show peer xxxx load will force it to load |
| 15:01.39 | igcewieling | however, it should automatically be loaded when the first call happens |
| 15:01.40 | vlad_starkov | vlad_starkov: If I define peer like type=friend and make it register on Asterisk, then it works well for inbound call. |
| 15:02.05 | igcewieling | make SURE when you define your peer in time you populate the port field with 5060 |
| 15:02.13 | igcewieling | s/time/realtime |
| 15:03.36 | *** join/#asterisk edwin_quijada (~macaruchi@231sdl30m59.codetel.net.do) |
| 15:03.47 | navaismo | you miss the last / |
| 15:04.41 | vlad_starkov | igcewieling: "sip show peer xxxx load will force it to load" – what does "xxxx" stand for? |
| 15:04.54 | igcewieling | vlad_starkov: the peer name |
| 15:04.59 | vlad_starkov | igcewieling: I mean what column it related to |
| 15:05.06 | igcewieling | the name field |
| 15:05.11 | vlad_starkov | ok |
| 15:05.46 | kelebeks | hi i have dial plan like this => http://pastebin.com/a597X6ZZ . In this plan, System and Set command wait answering phone. I want to _don't wait_. What could i do? |
| 15:06.28 | vlad_starkov | igcewieling: Yay! It works!) |
| 15:06.40 | vlad_starkov | igcewieling: thanks! |
| 15:06.49 | [TK]D-Fender | kelebeks, Answer first |
| 15:07.03 | edwin_quijada | Hi! I am getting an error from my SIP trunk with codecs http://pastebin.com/M5dspWJn I wanna know if I must install speex as codec to my * |
| 15:07.07 | igcewieling | vlad_starkov: if the load worked then you are missing the port 5060 in the realtime DB |
| 15:08.40 | [TK]D-Fender | edwin_quijada, I see no speex there in your translation table.... |
| 15:08.42 | [TK]D-Fender | edwin_quijada, Of course HAVING to have it is another matter |
| 15:08.52 | [TK]D-Fender | edong23, And you should be ... probably looking at the actual call. |
| 15:08.55 | [TK]D-Fender | edwin_quijada, ^ |
| 15:09.14 | [TK]D-Fender | HATES X-Chat's moronic auto-complete |
| 15:09.19 | vlad_starkov | igcewieling: Yepp. So the rule is: If you have a `friend` with `dynamic` host, then the `port` isn't required. When you use `peer` with static host, then you have to define port explicitly. |
| 15:09.58 | *** join/#asterisk threesome (~threesome@customer-79-127-150-148.net.angelnet.cz) |
| 15:10.04 | igcewieling | vlad_starkov: correct. the port is automatically populated when the peer registers, if the peer doesn't register you have to put the port in yourself |
| 15:10.26 | igcewieling | vlad_starkov: took me several hours to figure that out. Realtime debugging SUCKS |
| 15:10.53 | vlad_starkov | igcewieling: yep. And it rocks |
| 15:10.53 | kelebeks | [TK]D-Fender, i have to call system command for unanswered calls |
| 15:11.09 | igcewieling | vlad_starkov: not in my opinion. |
| 15:12.18 | igcewieling | vlad_starkov: on a single system realtime seems OK, but if you try to cluster asterisk servers using real time then you can run into issues |
| 15:12.34 | edwin_quijada | [TK]D-Fender: Thks, I just wanted to be sure to install speex but when a call incoming there is no problem of codec just when I put a call |
| 15:13.27 | [TK]D-Fender | kelebeks, What "unanswered calls" |
| 15:13.36 | vlad_starkov | igcewieling: I think it needs you to know how to use OpenSIPS/Kamailio + memcached/redis to build robust Asterisk cluster. |
| 15:14.03 | [TK]D-Fender | kelebeks, You are showing a chunk of dialplan that DOES "answer" (priority 3), and also calls an AGI. |
| 15:14.23 | [TK]D-Fender | kelebeks, You are not being thorough in the description of what you are trying to do and when. |
| 15:15.28 | *** join/#asterisk vlad_sta_ (~vlad_star@79.104.6.220) |
| 15:16.12 | *** join/#asterisk vlad_sta_ (~vlad_star@91.206.59.143) |
| 15:17.10 | igcewieling | vlad_starkov: we are doing it with mysql, SRV, and AGIs |
| 15:17.24 | igcewieling | and no registrations |
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| 15:21.36 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 15:21.36 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:21.40 | kelebeks | [TK]D-Fender, sorry. I start call with call file => http://pastebin.com/X0wVA0nT . I parse callfile after the calling and extract asterisk's unique id in call file. So i need to append asterisk's unique id to call file(with system comman). When line is busy, System command not working. I try to fix this. |
| 15:21.44 | vlad_sta_ | igcewieling: all your users have static IP? |
| 15:22.19 | igcewieling | vlad_sta_: correct |
| 15:22.42 | [TK]D-Fender | kelebeks, Your call file is dialing Channel: SIP/0530XXXXXXXXX@sip_sol2 directly. |
| 15:22.56 | [TK]D-Fender | kelebeks, You don't get to do anything unless it connects. |
| 15:23.27 | [TK]D-Fender | kelebeks, instead of dialing that SIP channel directly, use a LOCAL channel to do the actual dial and look at the DIALSTATUS afterwards. |
| 15:24.52 | kelebeks | [TK]D-Fender, i understand, thanks |
| 15:25.04 | vlad_sta_ | igcewieling: I see. I think it depends on your installation. If you have users that often change their IPs your current setup is not convenient. |
| 15:25.32 | igcewieling | vlad_sta_: *nod* but with all static peers my setup is reasonably simple |
| 15:26.39 | edwin_quijada | If I install speex codec works with speex16 codec? |
| 15:27.23 | [TK]D-Fender | edwin_quijada, what do you see in your translation table? |
| 15:27.35 | [TK]D-Fender | edwin_quijada, that might be a clue.... |
| 15:28.53 | SeRi | guys cfom asterisk cli what is the convert command to convert wav to ulaw format? |
| 15:29.30 | vlad_sta_ | igcewieling: u're right |
| 15:29.50 | SeRi | I think is file convert |
| 15:29.53 | SeRi | let me try |
| 15:30.31 | igcewieling | SeRi: that only works if both files are in a format Asterisk supports |
| 15:30.52 | SeRi | damn |
| 15:30.58 | SeRi | yeap I just saw it failed |
| 15:30.59 | SeRi | lol |
| 15:31.18 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 15:35.15 | slav3_kitten | SeRi, what is the capital of Latvia |
| 15:36.17 | [TK]D-Fender | slav3_kitten, the "L" :p |
| 15:36.37 | slav3_kitten | touche |
| 15:42.29 | igcewieling | I believe it is Riga |
| 15:43.07 | igcewieling | I used Riga as the name of my consulting company a long time ago because it sounded exotic |
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| 15:44.55 | SeRi | slav3_kitten: Your house |
| 15:45.49 | slav3_kitten | it is riga igcewieling |
| 15:46.24 | igcewieling | glad I can still remember stuff from 25 years ago |
| 15:49.14 | nricciar | im having an issue with a GotoIf check to see if a call is an internal call or not 2XX, problem is the check seems to be letting calls from 2XX area codes through also.... http://pastebin.com/3BSqJn8k anyone by any chance have a suggestion to fix this? |
| 15:54.31 | Qwell | nricciar: Your patterns will only handle 3 digit extensions. |
| 15:55.32 | Qwell | Is there perhaps more that you aren't showing us? |
| 15:56.51 | nricciar | the main menu sends outside calls to that extension |
| 15:57.01 | nricciar | i can paste the entire dialplan if that would help |
| 15:58.00 | nricciar | http://pastebin.com/FNMBVvzN |
| 15:59.08 | nricciar | everything works fine except for calls coming in from the 2XX area codes |
| 15:59.23 | nricciar | they should not auto answer but they do |
| 16:05.05 | *** join/#asterisk troyt (~troyt@2001:1938:240:2000::3) |
| 16:07.15 | igcewieling | I see nothing in there which matches callerid |
| 16:07.35 | nricciar | line 31 and 32 |
| 16:08.01 | igcewieling | Ah, there it is. that won't work |
| 16:08.25 | igcewieling | you are trying to match ONLY 200 - 299 i.e. where the callerid is EXACTLY 3 digits. |
| 16:08.39 | igcewieling | also > and < tend to not work with quotes |
| 16:09.26 | igcewieling | the easeist thing to do is exten => _2XX/_2NXXNXXXXXX,1,Goto(dial) |
| 16:09.50 | igcewieling | see the ex-girlfriend option in Asterisk. a quick google search will provide details |
| 16:10.11 | *** join/#asterisk vlad_starkov (~vlad_star@79.104.6.220) |
| 16:10.23 | nricciar | hmmm ok... without rewriting this though it sounds like i could just add a 3rd check for callerid length though? |
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| 16:13.15 | *** join/#asterisk TheCompWiz (~TheCompWi@63.214.236.169) |
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| 16:14.16 | TheCompWiz | can anyone tell me if there is a way to "include" voicemail contexts? i.e. in voicemail.conf [default] have a include=>"some-other-voicemail-context" |
| 16:19.17 | igcewieling | nricciar: rewriting? just replace those two lines with my one line |
| 16:19.49 | igcewieling | TheCompWiz: I don't know if voicemail.conf supports include=> but you could put the mailboxes in a separate file and use #include |
| 16:20.27 | nricciar | igcewieling, so i do not need to rewrite the rest of the "exten => _2XX" lines to "exten => _2XX/_2NXXNXXXXXX"? |
| 16:20.55 | igcewieling | nricciar: if you did so they would ONLY match callerids with 2NXXXXXXXX |
| 16:20.57 | TheCompWiz | igcewieling: I tried that... and didn't work. I did discover that you can have duplicate contexts & the contents get merged... i.e. [default] in voicemail.conf and [default] in another conf that is #included in... |
| 16:21.42 | igcewieling | TheCompWiz: don't put the context in the #include'd file |
| 16:22.07 | TheCompWiz | igcewieling: why not? |
| 16:22.07 | nricciar | igcewieling, ahh ok i think i understand now |
| 16:22.18 | igcewieling | TheCompWiz: because then you have the problem you have. |
| 16:22.29 | TheCompWiz | igcewieling: I wanted the results I got :) |
| 16:22.47 | TheCompWiz | I didn't expect the contents to get merged. |
| 16:23.04 | ariel_ | TheCompWiz: lunch |
| 16:23.10 | *** join/#asterisk blee (~blee@50-89-200-235.res.bhn.net) |
| 16:23.11 | TheCompWiz | ariel_: good plan. |
| 16:23.11 | igcewieling | TheCompWiz: Great! You were successful at being successful! |
| 16:23.48 | TheCompWiz | igcewieling: sometimes it just helps to verbalize it someplace... then realize a possible solution. |
| 16:24.34 | igcewieling | goes back to swearing at sales people |
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| 17:42.56 | derjanni | Good evening everybody |
| 17:43.34 | derjanni | I keep getting an error that the voicemail does not exist although it shows it as configured: http://susepaste.org/55065424 |
| 17:47.02 | derjanni | lösung: searchcontexts=yes |
| 17:47.54 | derjanni | sry solution ^ |
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| 18:36.11 | [TK]D-Fender | derjanni, proper solution is to actual specify the context in the call to Voicemail like you're supposed to. |
| 18:37.29 | TheCompWiz | <GASP> y'mean... you should RTFM? who does that anymore? |
| 18:38.00 | [TK]D-Fender | Best part is the manual is built-in |
| 18:38.07 | TheCompWiz | if you don't specify the context... isn't it supposed to default to the "default" context? |
| 18:38.10 | [TK]D-Fender | 1 line away |
| 18:38.21 | [TK]D-Fender | TheCompWiz, Yes... also a horrible assumption :) |
| 18:38.49 | TheCompWiz | true.... everyone knows that "default" rarely is the "default". |
| 18:41.31 | derjanni | [TK]D-Fender: thanks! |
| 18:41.38 | derjanni | Any idea which text2speech to use for German? |
| 18:42.58 | *** join/#asterisk edwin_quijada (~macaruchi@190.167.161.58) |
| 18:43.42 | navaismo | maybe cepstral |
| 18:45.20 | derjanni | sounds good. thanks. Ill try mbrola also as its a suggestions from the ubuntu wiki |
| 18:45.44 | TheCompWiz | derjanni: I think festival has some good german text-to-speech stuff.... been a while since I looked... and my German sucks. |
| 18:46.29 | edwin_quijada | I have a problem when I call an extension from my IVR the client doesnt hear anything but really * is calling to extension but there is a silence and the client think the call dropped. I used option r for Dial() but the client keeps doestn hear anything |
| 18:46.31 | derjanni | Festival itself says they have no German |
| 18:46.33 | edwin_quijada | any cluees? |
| 18:48.11 | edwin_quijada | derjanni; How is it the support for spanish? |
| 18:48.24 | navaismo | awful |
| 18:49.46 | navaismo | my top cheapest TTS: GoogleTTS,Cepstral |
| 18:55.08 | derjanni | edwin_quijada: ok, but you clearly notice that is a machine |
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| 19:20.16 | derjanni | <PROTECTED> |
| 19:20.19 | derjanni | [Apr 18 19:19:56] NOTICE[3197]: app_mp3.c:133 timed_read: Poll timed out/errored out with 0 |
| 19:20.22 | derjanni | srry for pasting directly. |
| 19:20.29 | derjanni | whats the issue here with MP3Player? |
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| 19:23.30 | derjanni | the ' were the prob |
| 19:23.31 | derjanni | kewl! |
| 19:23.37 | derjanni | Ill go with Google TTS |
| 19:23.49 | derjanni | precompile them and just upload the mp3z |
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| 19:29.50 | *** join/#asterisk [sr] (~kvirc@pal-213-228-181-48.netvisao.pt) |
| 19:30.11 | [sr] | hi |
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| 20:03.42 | StrangeWill | Does anyone have experience in monitoring line quality? |
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| 20:30.12 | [TK]D-Fender | checkout time, BBIAB |
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| 20:38.10 | jkroon | StrangeWill, as in monitoring of RTCP for SIP? Or JB Stats for IAX/2? Or general MOS for analog? |
| 20:39.38 | igcewieling | jkroon: he said "line" so I can only assume he means POTS lines. |
| 20:41.11 | jkroon | argh ... yea, not much i'm aware of you can do there ... |
| 20:41.17 | jkroon | the noise you get is the noise you transmit, and you have no control over what happens distortion wise once the audio leaves your TDM either ... |
| 20:41.39 | jkroon | at least on ISDN you can have fair confidence that it won't be stuffed completely. |
| 20:41.41 | Kobaz | soooo |
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| 20:42.30 | jkroon | with SIP ... well, you have no guarantees, but at least RTCP gives you some feedback. |
| 20:42.44 | jkroon | with IAX/2, permitting you have jitterbuffers switched on you can also get a reasonable idea. |
| 20:43.20 | igcewieling | jkroon: our RTCP stats always seem to be zero everyting |
| 20:45.37 | jkroon | igcewieling, yea, how are you getting to those? via the QOS variables? |
| 20:45.48 | jkroon | don't trust them. and they don't give you an overall picture either. |
| 20:46.38 | jkroon | use AMI and look for the RTCPReceived and RTCPSent events. Busy as I write this with a patch to extend on those reports too to at least deliver XR events. |
| 20:47.07 | jkroon | and as a bunch of people have also indicated, you should probably incorporate some of the pinefrog work done by oej, so will probably only be ready over the weekend. |
| 20:47.55 | jkroon | he's done some work internally to fix (imho) a lot of the RTCP stuff, including dealing with hold situations, as well as sending RRs when appropriate, including in some bridge scenarios ast wasn't generating rtcp. |
| 20:48.18 | jkroon | as well as (partially) dealing with SDES and generating BYE blocks. |
| 20:49.17 | jkroon | you can also get my patch (will need to merge this with oej's work) on www.uls.co.za/asterisk-11.3.0-rtcp.patch. won't remain at that URL. |
| 20:50.33 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.80) |
| 20:51.53 | jkroon | not to mention getting RTCP to work properly when * is behind NAT. |
| 20:52.12 | jkroon | well, the comments in the code seems to indicate that it should work better than without it :) |
| 20:53.28 | igcewieling | jkroon: dumpchan after the call ends |
| 20:57.49 | jkroon | igcewieling, those never gave me sufficient results, and it only reports on the last rtcp values. |
| 20:58.14 | jkroon | so if the call was crappy mid-way through but the last section just before the last rtcp report was good, you'll think the entire call was good. |
| 20:58.22 | jkroon | part of the pinefrog branch motivation :p |
| 20:58.26 | igcewieling | *nod* We would like to use directmedia so it has not been a high priority fo rus. |
| 20:59.02 | jkroon | most of my clients are behind NAT ... so my chances of using directmedia is somewhere between naught and zero. |
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| 20:59.36 | igcewieling | jkroon: less than 1% of our clients are behind NAT and less than 1% of our clients are on dynamic IPs |
| 20:59.48 | jkroon | lucky you :) |
| 21:00.09 | igcewieling | since we are also their ISP we just assign extra IPs if needed. |
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| 21:00.41 | jkroon | that's convenient. getting there on a portion of my clients, but i'll be slow going to connect the whole lot up. |
| 21:00.50 | deviantlinux | When hand-compiling DAHDI, does anyone know of a way to only compile a specific module, instead of all available cards? I only have 1 card, and one of the modules is erroring out. |
| 21:01.33 | igcewieling | our router is also the SIP device (media gateway) so we assign them a very small block for their firewall, etc |
| 21:02.38 | igcewieling | jkroon: we have a very bizarre agreement with Verizon wholesale which prevents us from using direct media with the VZ TNs. REALLY annoying but to fix the issue we'd have to upgrade to a connection which costs 4x as much. |
| 21:03.53 | jkroon | hehe, sounds similar to our plans. |
| 21:04.22 | jkroon | ok, agreement with Verizon sounds like politics. i hate those, in spite of dealing with it every single day of my live. |
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| 21:58.20 | jmetro | Where can I change the hold music for that magical limbo place in between transferring calls and parking calls? |
| 22:00.49 | [TK]D-Fender | jmetro: My guess .... musiconhold.conf <--------- |
| 22:04.01 | jmetro | hm..i know thats where you configure the default stuff for the normal places, but i didnt see anything magical in there. |
| 22:07.36 | [TK]D-Fender | jmetro: because it isn't magical |
| 22:07.47 | [TK]D-Fender | jmetro: You define your classes and where its gets the MoH from |
| 22:08.42 | jmetro | Yes, though there seem to be two different MOH classes at play when placing a park, the first being when the person presses "park" the second being when the person is actually on park and listening to the MoH |
| 22:08.55 | jmetro | after the parker hangs up |
| 22:10.50 | igcewieling | jmetro: only if your dialplan does it |
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| 22:15.09 | jmetro | hm..what sets the first "magical" one? |
| 22:21.29 | [TK]D-Fender | jmetro: Show us these 2 being used.... |
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| 22:29.03 | igcewieling | jmetro: likely musiconhold.conf I think there are some mohsuggest options in sip.conf as well |
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