Problems with calls dropping on Android.
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Hello.
I have the following in sip.conf
[sip09]
type=peer
defaultuser=sip09
nat=yes
qualify=no
secret=sip09
host=dynamic
context=outgoing
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=alaw
allow=h263p
deny=0.0.0.0/0.0.0.0
permit2.168.2.2/255.255.255.255
jbenable = yes
jbforce = yes
jbmaxsize = 100
jbresyncthreshold = 200
jbimpl = fixed
transport=tcp
sendrpid=yes
And these settings in Android native client.
Username: sip09
Password: sip09
Server: 192.168.1.10
Username at authentication: sip09
Display name: Same as username
Outgoing proxy: 192.168.1.10
Port: 5060
Transport: TCP
Send keep alive: Always
However, if I make a call FROM android phone, call is dropped after 30
seconds, regardless of answer or not. If I make call TO android phone, it works normally.
No NAT problems inbetween, there is a VPN between the phone and SIP server with full access.
I guess I need to do some trick to have it work with Android. Apparently the packets are received in both ends – else audio wouldn’t work, but guess the stock native SIP client on android ignores certain packets right?
This is an Android 9 phone.
Additionally, I wonder if its possible to change the callerid shown in display when calling out? Like RPID. It works on my desktop phones, if I
enter a short code, the full name and number is shown on display, but on the Android phone, it doesn’t work, only the dialled shortnumber is shown.
Also I wonder if its possible to have asterisk send the remote callerid
(when receiving a call) in such a way it gets stored in call log with full names and such – without having to resort to using phonebook.
SIP debug log:
*CLI> sip set debug ip 192.168.2.2
SIP Debugging Enabled for IP: 192.168.2.2
*CLI> Really destroying SIP dialog
‘6f9956035553ab1b79ca057f5dffe0ac@192.168.2.2’ Method: OPTIONS
Really destroying SIP dialog ‘fc3307059c816094a6c6ce100cf383e5@192.168.2.2’
Method: OPTIONS
<--- SIP read from TCP:192.168.2.2:51729 --->
OPTIONS sip:192.168.1.10 SIP/2.0
Call-ID: e65234cb818a143bc3c167a782b98e96@192.168.2.2
CSeq: 3984 OPTIONS
From: “sip09”
To: “sip09”
Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bK105b3648c13a72f8fbe7ce3049df71aa3130;rport
Max-Forwards: 70
User-Agent: SIPAUA/0.1.001
Content-Length: 0
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