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=60 ;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 at lucabert.de)
You should try another SIP client, just to check it. (Zoiper or cSipSimple, for example). Regards, Marcelo H. Terres <mhterres at gmail.com> IM: mhterres at jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On 24 October 2017 at 14:42, Luca Bertoncello <lucabert at lucabert.de> wrote:> 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=60 > ;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 at lucabert.de) > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
Luca Bertoncello <lucabert at lucabert.de> schrieb: Hallo again> 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...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 at lucabert.de)