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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