SIP Phones and Asterisk Server

Ia Orana,

We have been running an asterisk box with a bunch of Snom 360 SIP phones without any problem. But, I added an Aastra 6735i SIP phone and I face the current issue for our internal calls :

  • Snom->Snom : OK
  • Aastra->Snom : OK
  • Snom->Aastra : No sound is carried !

Both phone types belong to the same context. I upgraded firmware on both sides and I have been through the SIP debug mode on my asterisk box but did not find anything useful…

Did anyone face this kind of issue already ?

Thanks in advance for your help,

On the Asterisk CLI run the command rtp set debug on… and check the media traffic between those 2 phones. in your sip.conf file set directmedia=no and see if this help with your issue


Thank you for your suggestions but everything lokks fine…