You should try another SIP client, just to check it. (Zoiper or cSipSimple, for example).
Regards, Marcelo H. Terres <[email protected]> IM: [email protected] https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On 24 October 2017 at 14:42, Luca Bertoncello <[email protected]> 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 > ([email protected]) > > > -- > _____________________________________________________________________ > -- 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 -- _____________________________________________________________________ -- 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
