Hello! I discovered such a problem and will be greatfull if someone can help me to solve it!
I have some number reacheble via trunk, and the next routing point is new short number in other trunk:
IN-DPLN.Huawei:exten => 5821212,1,Dial(SIP/3570@NCX)
in the incomming trunk only one codec allowed - g711a:
[HuaweiB]
type = friend
…
disallow = all
allow = alaw
so as in outgoing trunk:
[NCX]
type = friend
…
disallow = all
allow = alaw
So i have Incomming INVITE with g729 and 101 for telephone events, and Asterisk do:
Using INVITE request as basis request - SDdg58801-650b3499775e840bee206042cc9c6e72-v3000i1
Found user '0000001’
Found RTP audio format 18
Found RTP audio format 5
Found RTP audio format 101
Peer audio RTP is at port 10.255.102.50:15134
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0x120 (adpcm|g729)/video=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event), combined - 0x0 (nothing)
Looking for 5821212in DLPN_MG+MN (domain 10.255.108.33)
Asterisk in this case simply generate INVITE to out trunk but with alaw. I’m new in Asterisk, but have some expirience in voip, and i think this is a mistake. It musk in this case say - sorry guys, a have no such media you asking by sending 488 or 499 erorr. Instead asterisk replace codec in SDP and call established, but as i should with no rtp from both sides.
Any idias about how to fix it?
Asterisk 1.4.21.2
By the way, if i for examle allow any codecs in the incomming trunk call fail, bun with 6xx reason code - in this case asterisk understand that he has no media to offer to far side, but not on capabilities comparsion level yet when generating second call leg INVITE.
Greatfull for any help.