SV: [Asterisk-Users] callback script?

[Krystian Filiks]

You can get asterisk source from http://www.asterisk.org/


Arne Morten Johansen wrote:

>It's a sip channel.
>
>http://www.asteriskguru.com/tutorials/unknown_codec_received.html
>
>This might work, but I don't know where to find the source-code of asterisk. I've used the ebuilds in gentoo portage to compile asterisk. And I'm not exactly a linux type of guy, so this is not my field. 
>
>I tried calling the cellphone from a hardware SIP phone and tried pushing some buttons on the cell. Same error appeared and I couldn't hear anything. :(
>
>Any ideas?
>
>Regards,
>Arne Morten Johansen
>
>
>-----Opprinnelig melding-----
>Fra: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] På vegne av Joseph Tanner
>Sendt: 6. februar 2006 12:33
>Til: Asterisk Users Mailing List - Non-Commercial Discussion
>Emne: Re: [Asterisk-Users] callback script?
>
>Sorry for the blank email, here's what I meant to send:
>
>I haven't seen that error before, sorry.  A quick search using google
>turned this up though:
>
>http://www.mail-archive.com/asterisk-users@lists.digium.com/msg08901.html
>
>Not sure if it's relevant in your case.  What is asterisk using to
>dial your remote cellphone?  Is it a sip, iax, or zap channel?  Or are
>you calling out using a cellphone connected to your asterisk server
>(whether by a dock connected to a zap card, or bluetooth)?  I've
>noticed that dtmf is not processed between two cellular phones on a
>cingular account (tried two different motorola phones plus a sony
>ericsson phone).  It may be an issue with other carriers too, to test
>just call from one cellphone to another, press some keys, and see if
>the other side hears any dtmf tones.  If not, then you'll have to find
>another way to do what you're trying to accomplish.
>
>Joseph Tanner
>
>On 2/6/06, Arne Morten Johansen <amj at avio.no> wrote:
>  
>
>>Thanks.
>>
>>I'm able to getting the asterisk calling back to my cellphone. But when I get to the authentication I get this message when I start to dial in my password:
>>
>>NOTICE[5178]: rtp.c:509 ast_rtp_read: Unknown RTP codec 96 received
>>
>>Is this a DTMF failure of some sort?
>>
>>Thanks again.
>>
>>-----Opprinnelig melding-----
>>Fra: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] På vegne av Joseph Tanner
>>Sendt: 4. februar 2006 11:51
>>Til: Asterisk Users Mailing List - Non-Commercial Discussion
>>Emne: Re: [Asterisk-Users] callback script?
>>
>>This is what I use, more or less:
>>http://mundy.org/blog/index.php?p=73 , go down to "Incoming Call
>>Context" (about 1/3 down).  I had to modify it a bit, as I actually
>>need Asterisk to pick up and listen to some DTMF digits before hanging
>>up and calling me back, but it works great for me, and requires no
>>external agi scripts.
>>
>>Joseph Tanner
>>
>>On 2/3/06, bbench at mail.bg <bbench at mail.bg> wrote:
>>    
>>
>>>On Thursday 02 February 2006 11:40, Arne Morten Johansen wrote:
>>>      
>>>
>>>>How do I setup a Callback script?
>>>>
>>>>This script does what I want to do. But how do I set it up?
>>>>
>>>>http://www.junghanns.net/en/callback.html
>>>>
>>>>I see it uses PHP for scriptlanguage. So where do I place it (the .agi)?
>>>>        
>>>>
>>>/var/lib/asterisk/agi-bin
>>>and should be 755
>>>benchev
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