| 00:12.02 | snadge | ok.. if i set name-pres and num-pres to allowed, then it does what im expecting it to.. but i still don't know why ;) |
| 00:12.11 | snadge | poke at something with a stick for long enough.. eventually it will react |
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| 01:13.18 | chris349 | Im having a hard time with caller ID. I want to change the caller ID received over a SIP trunk and add a prefix. If I get John Doe I want to change it to (Office) John Doe but when I try this it gets changed just to (Office) |
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| 01:18.14 | WIMPy | So you obviousely didn't add the original contents. |
| 01:19.02 | chris349 | WIMPy, I did. Actually now that I look at it the callerid is being set to just (Office and missing the final ) |
| 01:19.46 | WIMPy | Smells like a syntax error. |
| 01:20.47 | chris349 | Where can I find the proper syntax? |
| 01:21.00 | WIMPy | ~book |
| 01:21.02 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 01:21.09 | WIMPy | Or you show us what you did. |
| 01:22.00 | WIMPy | Using an editor that counts brackets might help as well. |
| 01:23.23 | chris349 | Currently I have Set(CALLERID(name)="(OFFICE ${OFFICENUM}\) ${CALLERID(name)}") |
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| 01:24.36 | WIMPy | Don't use quotes and don't escape brackets. |
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| 01:25.38 | r3z | Anyone familiar with grandstream ucm6102 devices? |
| 01:26.02 | WIMPy | Grandsteam. They made them. |
| 01:26.17 | chris349 | If I use Set(CALLERID(name)=(OFFICE ${OFFICENUM}) ${CALLERID(name)}) then I still get the callerid as (Office 10 |
| 01:26.18 | r3z | I just bought one and trying to get it configured. I have my sip trunk configured but my outbound rule isnt working for some reason. |
| 01:27.46 | WIMPy | chris349: The dialplan parser can really be a bitch. You could try to use a global variable containing the ")". |
| 01:29.07 | WIMPy | Or expand OFFICENUM to contain the whole prefix. |
| 01:33.14 | chris349 | Well I try Set(PREFIX=(OFFICE ${OFFICENUM}) and Set(CALLERID(num)=${PREFIX}${CALLERID(name)} and still get the same result |
| 01:34.05 | WIMPy | Well, that's the same sa before. |
| 01:34.59 | voipmonk | yes its fun - so set it and check it with a NoOp and try a different method on the next call.. keep at it till you get what you need... |
| 01:35.52 | voipmonk | or I could run the SpoonFeed 2.0 and do it for you. |
| 01:36.53 | WIMPy | Using ) via AGI doesn't seem to cause any issue. |
| 01:37.29 | WIMPy | But as I said, a global variable should do it. |
| 01:37.38 | WIMPy | Ugly, but that sometimes happens. |
| 01:38.53 | WIMPy | And before I get ugly, I take my beauty sleep. |
| 01:42.55 | chris349 | Setting it via AGI doesnt seem to work either |
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| 09:43.11 | pawiecki | Hi, if i dial at 3 numbers at once, and one of them answers - is there any variable, that stores that peers info? |
| 09:45.45 | WIMPy | 'core show application Dial' |
| 09:50.46 | pawiecki | WIMPy: i checked that, but it seems i can not get information which peer answered, only dialstatus |
| 09:54.04 | WIMPy | Oh, incomplete. :-( See DIALEDPEERNAME and DIALEDPEERNUMBER. |
| 09:56.24 | pawiecki | it looks exclusive to version 13 and 14 of * - does it work for version 11? |
| 09:57.06 | pawiecki | oh, no, my bad - it should work |
| 09:58.50 | WIMPy | It definitey does. |
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| 13:14.36 | BLAH_BLAH_BLAH | I was wondering if anyone had seen a way to identifying a call that possibly have been a 3way call. I've read about a 3rd party application that will monitor the call voice-payload and other variables to determine if it may be a 3way call. Is there any free applications of doing this or any scripts to run in asterisk to identify in the same manner? |
| 13:18.15 | WIMPy | BLAH_BLAH_BLAH: No longer available when using SIP. |
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| 13:56.13 | BLAH_BLAH_BLAH | WIMpy: thanks, so it's looking like a third party app to verify the recordings is the only way to go |
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| 14:11.33 | kannan | hello, with respect to Sip trunk between 2 asterisk servers, over TLS with SRTP, and https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial, is TLS v.1.2 supported? the wiki says ..."and we're setting the method to TLSv1." |
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| 16:20.37 | kannan | hi, sprry to repeat.. does asterisk support TLS v.1.2 for secure calling? |
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| 18:35.47 | jrun | would it be possible to use opus within asterisk using pjsip? |
| 18:36.11 | jrun | if yes, would it be possible with a simple allow = opus? |
| 18:36.29 | jrun | or do i need some sort of passthrough for legal reasons? |
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| 19:02.06 | newtonr | jrun, Opus passthrough was added in 12 |
| 19:02.18 | newtonr | allow=opus should do it, but you won't be able to transcode |
| 19:03.28 | jrun | voicemail? |
| 19:06.10 | newtonr | voicemail what? |
| 19:06.24 | newtonr | you'll have to phrase your question in the form of a question :D |
| 19:12.49 | newtonr | kannan, I believe so |
| 19:13.36 | jrun | * won't allow transcoding but would it allow voicemail operations when using opus? |
| 19:19.50 | newtonr | It might allow operations that don't require transcoding like the playback of an opus format file |
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| 19:20.29 | newtonr | but it isn't going to be able to transcode a gsm file to opus or anything like that |
| 19:21.13 | newtonr | I'm not sure to be honest. I'm not sure I've ever tried to use a passthrough only format in voicemail |
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| 22:05.24 | jeffspeff | alright, so random question of the day. can asterisk encrypt data in realtime db as well as voicemails (either in flat file or db) ? |
| 22:11.04 | robmal | Out of the box afaik only md5, so not so much for listening to that voicemail, but what's the point? |
| 22:22.12 | jeffspeff | robmal just trying to secure everything i can think of. |
| 22:22.34 | robmal | Why? |
| 22:23.07 | jeffspeff | box compromised they can't listen to any voicemails containing whatever if they're encrypted. sql compromised, would be difficult to read data if it was encrypted. |
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| 22:24.23 | jeffspeff | HIPAA rules in the US require any patient information to be encrypted during transport and at rest. SIP TLS and SRTP take care of transport, but just seeing what my options are for encryption at rest. |
| 22:24.46 | jeffspeff | the disk is encrypted... that may have to suffice. |
| 22:25.46 | jeffspeff | the rule does say something along the lines of doing what you can that's feasible. |
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| 22:39.52 | WIMPy | Has anyone here ever come across any product that implements J.52? |
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| 23:37.25 | voipmonk | what is it - I can lend another pair of fingers for research |
| 23:38.07 | voipmonk | microwave? |
| 23:39.43 | voipmonk | ITU Rec. J.52 ? |
| 23:44.23 | WIMPy | yes |
| 23:44.49 | voipmonk | need input - |
| 23:44.51 | voipmonk | more input - |