Hi I have installed asterisk 1.0.9 on my laptop which is running Redhat el3. As it is when i use ulaw / alaw codecs my calls r easily getting thru with good quality, but when i resort to speex i am getting the error message on console : "chan_sip.c:2792 process_sdp: No compatible codecs!" my sip.conf looks like [12] type=friend secret=kk host=dynamic canreinvite=no disallow=all allow=SPEEX context=test_direct dtmfmode=rfc2833 outgoinglimit=1 ;incominglimit=1 [21] type=friend secret=amit host=dynamic canreinvite=no disallow=all allow=SPEEX context=test_direct dtmfmode=rfc2833 outgoinglimit=1 ;incominglimit=1 I am also using a linksys PAP2NA so as to connect two Analogue phones, further i downloaded the latest version of speex for el3 and also the libogg libraries. Further the devel package for speex is also installed. Still when i am making calls i am having problems with asterisk console displaying the above mentioned codec related error message. Regards Hrishikesh shrivastaw India
I do not think that speex is installed by default. run "show translations" in asterisk and see what you get. -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. --- - --- - - - - - - - -- - - - --- - ------ - - --- - - -- - - - -- - - - "hrishikesh shrivastaw" <shrivastaw@gmail.com> wrote in message news:a2c33e3f0512120145j2accc2c2j466ec7bb13779279@mail.gmail.com... Hi I have installed asterisk 1.0.9 on my laptop which is running Redhat el3. As it is when i use ulaw / alaw codecs my calls r easily getting thru with good quality, but when i resort to speex i am getting the error message on console : "chan_sip.c:2792 process_sdp: No compatible codecs!" my sip.conf looks like [12] type=friend secret=kk host=dynamic canreinvite=no disallow=all allow=SPEEX context=test_direct dtmfmode=rfc2833 outgoinglimit=1 ;incominglimit=1 [21] type=friend secret=amit host=dynamic canreinvite=no disallow=all allow=SPEEX context=test_direct dtmfmode=rfc2833 outgoinglimit=1 ;incominglimit=1 I am also using a linksys PAP2NA so as to connect two Analogue phones, further i downloaded the latest version of speex for el3 and also the libogg libraries. Further the devel package for speex is also installed. Still when i am making calls i am having problems with asterisk console displaying the above mentioned codec related error message. Regards Hrishikesh shrivastaw India _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
You need to recompile asterisk after you install speex. On 12/12/05, Steven <asterisk@tescogroup.com> wrote:> > I do not think that speex is installed by default. > run "show translations" in asterisk and see what you get. > > -- > -- > Steven > > May you have the peace and freedom that come from abandoning all hope of > having a better past. > --- - --- - - - - - - - -- - - - --- - ------ > - > - --- - - -- - - - -- - - - > "hrishikesh shrivastaw" <shrivastaw@gmail.com> wrote in message > news:a2c33e3f0512120145j2accc2c2j466ec7bb13779279@mail.gmail.com... > Hi > I have installed asterisk 1.0.9 on my laptop which is running Redhat el3. > > As it is when i use ulaw / alaw codecs my calls r easily getting thru > with good quality, but when i resort to speex i am getting the error > message on console : "chan_sip.c:2792 process_sdp: No compatible > codecs!" > > my sip.conf looks like > > [12] > type=friend > secret=kk > host=dynamic > canreinvite=no > disallow=all > allow=SPEEX > context=test_direct > dtmfmode=rfc2833 > outgoinglimit=1 > ;incominglimit=1 > > > [21] > type=friend > secret=amit > host=dynamic > canreinvite=no > disallow=all > allow=SPEEX > context=test_direct > dtmfmode=rfc2833 > outgoinglimit=1 > ;incominglimit=1 > > I am also using a linksys PAP2NA so as to connect two Analogue phones, > further i downloaded the latest version of speex for el3 and also the > libogg libraries. Further the devel package for speex is also > installed. > > Still when i am making calls i am having problems with asterisk > console displaying the above mentioned codec related error message. > > Regards > > Hrishikesh shrivastaw > India > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051212/2366e053/attachment.htm