Sip.conf : [general] ;context=default allowguest=no allowoverlap=no allowtransfer=yes realm=mydomain bindport=5060 bindaddr=X.X.X.X maxexpiry=1800 minexpiry=60 mohinterpret=default mohsuggest=default language=be useragent=mycorp dtmfmode = rfc2833 alwaysauthreject = yes ;contactdeny=0.0.0.0/0.0.0.0 ;contactpermit=172.16.0.0/255.255.0.0 rtptimeout=60 rtpholdtimeout=300 ;sipdebug = yes ;recordhistory=yes ;dumphistory=yes registertimeout=60 registerattempts=60 rtcachefriends=yes ;rtsavesysname=yes ;rtupdate=yes ;rtautoclear=yes ;ignoreregexpire=yes jbenable = yes jbforce = no allowsubscribe=yes limitonpeer = yes notifyringing=yes notifyhold=yes
then come the registrations... Jonas. On Fri, 2010-03-12 at 11:47 +0530, Prince Singh wrote: > Post your Asterisk's sip.conf > > > On Thu, Mar 11, 2010 at 10:39 PM, jonas kellens > <jonas.kell...@telenet.be> wrote: > > How can I set the prefered codec between 2 calling parties ?? > > My Grandstream supports G729, alaw and gsm... in this order. > The Zoiper softphone has alaw and gsm as codecs... in that > order. > > Although there should be a matching codec found, my > Grandstream can not call the Zoiper softphone. > > CLI shows : > > [Mar 11 17:47:21] WARNING[22367]: channel.c:3340 > ast_channel_make_compatible: No path to translate from > SIP/mygrandstream-09c599e0(2) to SIP/zoiper-09cd57f8(256) > [Mar 11 17:47:21] -- Got SIP response 415 "Unsupported > Media Type" back from 192.168.1.106 (<-- zoiper) > > SIP debug : > > [Mar 11 17:55:57] Peer audio RTP is at port > 192.168.1.101:10110 (<-- the Grandstream) > [Mar 11 17:55:57] Found audio description format PCMA for ID 8 > [Mar 11 17:55:57] Found audio description format GSM for ID 3 > [Mar 11 17:55:57] Found audio description format PCMU for ID 0 > [Mar 11 17:55:57] Found audio description format G729 for ID > 18 > [Mar 11 17:55:57] Found audio description format > telephone-event for ID 101 > [Mar 11 17:55:57] Capabilities: us - 0x10a (gsm|alaw|g729), > peer - audio=0x10e (gsm|ulaw|alaw|g729)/video=0x0 (nothing), > combined - 0x10a (gsm|alaw|g729) > [Mar 11 17:55:57] Non-codec capabilities (dtmf): us - 0x1 > (telephone-event), peer - 0x1 (telephone-event), combined - > 0x1 (telephone-event) > ... > [Mar 11 17:55:57] Audio is at 192.168.1.150 port 11586 (<-- my > Asterisk) > [Mar 11 17:55:57] Adding codec 0x100 (g729) to SDP > [Mar 11 17:55:57] Adding non-codec 0x1 (telephone-event) to > SDP > > This is what Asterisk sends to the Zoiper in the INVITE > (sdp) : > > Content-Type: application/sdp > Content-Length: 263 > v=0 > o=root 3208 3208 IN IP4 192.168.1.150 > s=session > c=IN IP4 192.168.1.150 > t=0 0 > m=audio 11586 RTP/AVP 18 101 > a=rtpmap:18 G729/8000 > > Why isn't Asterisk negotiating with the Zoiper for the > alaw-codec ?? > > The sip-configuration (realtime MySQL) for the Grandstream > is : > > allow : g729;alaw;gsm > > and the Zoiper softphone : > > allow : alaw;gsm;g729 > > > Kind regards, > > Jonas.
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