[App_rpt-users] SIP Help Needed

[Chris Andrist]

I hate to re-hash this, but setting up IAX for your smart phone will save you lots of time and grief.

The ports are already open and don’t have the routing issues.

Setup Steps.

1. cd /etc/asterisk
2. mkdir custom
3. cd custom
4. nano iax.conf
5. paste this into the file.

[kc7wsu] ; Change this to your desired username for the client. I use callsign.
type=friend
context=smartphone
host=dynamic
auth=md5
secret=Passw0rd1 ; Change to your desired password
disallow=all
allow=ulaw
allow=g726aal2
allow=gsm
codecpriority=host
context=radio-secure
transfer=no
callerid="KC7WSU" ; Change to the callsign you would like to show up.

6. Save and exit file. (CTRL+X), Yes
7. nano extensions.conf
8. paste this into the file. Update 29205 with your node number.

[smartphone]
exten=29205,1,answer()
exten=29205,n,Playback(rpt/node)
exten=29205,n,Playback(digits/2)
exten=29205,n,Playback(digits/9)
exten=29205,n,Playback(digits/2)
exten=29205,n,Playback(digits/0)
exten=29205,n,Playback(digits/5)
exten=29205,n,rpt(29205|P)

9. Save and exit file. (CTRL+X), Yes
10. astres.sh

Regards,

Chris Andrist, KC7WSU

On Feb 12, 2014, at 10:00 AM, Bill Hurlock <bill.hurlock at cpcomms.com> wrote:

> Right after I hit send I saw that in SRC there was a 2nd rtp.conf example that listed what you have sent. I have added this to my RTP.conf file and and added these ports to my router config. I’ll be out of the house and will check later today. Thanks for your help.
>  
> Bill Hurlock
>  
> From: Robert A. Poff WB3AWJ [mailto:wb3awj at comcast.net] 
> Sent: Wednesday, February 12, 2014 11:10 AM
> To: Bill Hurlock
> Cc: app rpt-users
> Subject: Re: [App_rpt-users] SIP Help Needed
>  
> Here's what is in mine verbatim.
> Ah....... I just looked into an ACID install of mine. Didn't see rtp.conf there at all. You may need to create it.
>  
> When I first starting to add extensions, I used the Asterisk book as a guide. Very helpful.
> Here's the online version :http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/index.html
> It covers Asterisk 1.8, but most of the config stuff still applies to 1.4 I find.
>  
> Anyhow, here's what I have in rtp.conf on my node that has SIP phones attached:
> ;
> ; RTP Configuration
> ;
> [general]
> ;
> ; RTP start and RTP end configure start and end addresses
> ; These are the addresses where your system will RECEIVE audio and video stream$
> ; If you have connections across a firewall, make sure that these are open.
> ;
> 
> rtpstart=10000
> rtpend=10999
> 
>  
> And here's a link to a page concerning rtp.conf :http://www.voip-info.org/wiki/view/Asterisk+config+rtp.conf
>  
> Hope this helps.
>  
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