Problem Using Android SIP-Client
Hi list!
I use Asterisk 1.8.30.0 on an OpenWRT-Router (I know, not the last
version, but I can’t upgrade). It always runned very well, and it runs very well with our home
phones, too, but now I have problems using the native Android
SIP-Client…
I configured an user for my mobile phone and I can call, but as soon
as the other party answer, I get this error in Log:
[Oct 24 15:32:41] NOTICE[3731]: channel.c:4280 __ast_read: Dropping
incompatible voice frame on SIP/messagenet-0000028e of format gsm
since our native format has changed to 0x8 (alaw)
and I can’t hear anything…
This is the configuration of the user:
[00491771234567]
fullname = 00491771234567
secret = MYVERYSECRET
dahdichan = 1
hassip = yes hasiax = no hash323 = no hasmanager = no callwaiting = no context = default host = dynamic dtmfmode=rfc2833
canreinvite=no sendrpid=pai type=friend
;nat=force_rport,comedia nat=yes qualify=yes qualifyfreq`
;transport=Auto avpf=no force_avp=no icesupport=no encryption=no callgroup=1
pickupgroup=1
dial=SIP/00491771234567
allow = all
Any idea?
The user worked very well with my old mobile phone (Android 4), I
__THINK__ the problem happens since I use my new phone with Android 7…
Thanks Luca Bertoncello
(lucabert@lucabert.de)
5 thoughts on - Problem Using Android SIP-Client
You should try another SIP client, just to check it. (Zoiper or cSipSimple, for example).
Regards, Marcelo H. Terres
IM: mhterres@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres
Luca Bertoncello schrieb:
Hallo again
I tried to call the same number I called before using LTE instead of WLAN, and it worked… Then I tried to call the same number again using my WLAN at home, and it worked again.
So, I must conclude that the problem is somewhere in the WLAN at office… Very curiously I can initiate the SIP-communication, but as soon as the other party answer the connection will be closed…
Since I’m one of the admins at office, I’d like to solve this problem. Can someone give me some advice what can be wrong in our firewall (Sophos)?
Thanks a lot Luca Bertoncello
(lucabert@lucabert.de)
the issue is quiet sure codec based:
[Oct 24 15:32:41] NOTICE[3731]: channel.c:4280 __ast_read: Dropping incompatible voice frame on SIP/messagenet-0000028e of format gsm since our native format has changed to 0x8 (alaw)
shorter:
Dropping incompatible voice frame on SIP/messagenet-0000028e of format gsm since our native format has changed to 0x8 (alaw)
looks like your android phone uses gsm, but only alaw is supported. just try to put gsm as allowed codec.
the sip invite and ok-message would be interesting as well (especially sdp)-
regards, andre
Hi, Is the Sophos a home router or professional one? In many cases what home router does by default needs to be configured manually on professional one. E.G. a home router will allow outgoing sessions and create a return path by default where professional one won’t. Two things I would look for:
1. Look for, and disable, ALG for SIP. The idea of ALG is nice but I
haven’t encountered a device that implements this properly (I’m not a network expert so it doesn’t mean that there isn’t such a router out there).
2. On the Sophos try to statically open the UDP port range of your RTP to outgoing traffic from your phone to your SIP server. Note that outgoing port range is what your SIP server defines as its port range, not your phone. If you get one way voice (remote hears phone) then you are on the right direction. You’ll then need to open the incoming ports too for the ports that your phone is expecting to get its RTP from. KR
Harel
Harel Cohen schrieb:
Of course the professional firewalls (we have two Sophos in Cluster, to manage our two SDSLs)
OK, tomorrow I’ll check it…
Thanks Luca Bertoncello
(lucabert@lucabert.de)