I have installed Asterisk on my dd-wrt router with a connection to my SIP provider upstream and I’m trying to get it working with two UAs.
One UA is connected on the home network over ethernet and in/out to this phone works great.
I am trying to connect to my Asterisk box from my Android phone using sipdroid over 4G and this does not work for making phone calls. The SIP signalling seems to be complete but there is no voice. I have captured the packets using Wireshark and I see the RTP traffic coming into the box but it’s not getting forwarded to Android phone from what I can tell. I seem to have a NAT type of problem even though both my Android phone and Asterisk server is on public networks from what I can tell.
I have tried playing with directmedia=yes/no; directrtpsetup=yes; nat=comedia and nothing seems to work.
Version is “Asterisk 188.8.131.52, Copyright © 1999 - 2012 Digium, Inc. and others.”
Here is my sip.conf
[general] context=incoming bindport=5060 bindaddr=0.0.0.0 recordhistory=yes disallow=all allow=ulaw allow=gsm trustrpid=yes sendrpid=yes dtmfmode=inband relaxdtmf=yes realm=xyz.dyndns.org srvlookup=no allowguest=no directmedia=yes ;directmedia=no ;nat=comedia ;directrtpsetup=yes externip=xyz.dyndns.org ; EC2 instance IP address localnet=192.168.1.1/255.255.255.0 ; local network as seen by .ifconfig eth0. register=><mynumber>:<secret>@<mysip provider>  type=friend context=outbound host=dynamic secret=XXXX disallow=all allow=gsm allow=ilbc outgoinglimit=1 incominglimit=1 ;canreinvite=no
asterisk -rvvvvv output
Verbosity was 1 and is now 5 -- Registered SIP '1001' at 184.108.40.206:52412 > Saved useragent "Sipdroid/3.0 beta/SAMSUNG-SGH-I727" for peer 1001 -- Registered SIP '1001' at 220.127.116.11:45307 -- Unregistered SIP '1001' -- Registered SIP '1001' at 18.104.22.168:53208 == Using SIP RTP CoS mark 5 -- Executing [18003582227@outbound:1] Dial("SIP/1001-00000006", "SIP/18003582227@sipprovider,60,r") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/18003582227@sipprovider -- SIP/sipprovider-00000007 is making progress passing it to SIP/1001-00000006 -- SIP/sipprovider-00000007 answered SIP/1001-00000006 [Jun 10 16:22:49] WARNING: dsp.c:1421 ast_dsp_process: Inband DTMF is not supported on codec gsm. Use RFC2833 == Spawn extension (outbound, 18003582227, 1) exited non-zero on 'SIP/1001-00000006'
Your help is greatly appreciated