On 1/15/2014 3:59 AM, Francesco Namuri wrote:
Hello,
I'm having this issue on my pbx, it appears that asterisk is refusing
the codecs that my providers is proposing.
My trunk configuration is:
Pretty simple -
---
username=5x5x7x9x0x3
type=friend
secret=CRcxn7sqwm
qualify=yes
port=5060
insecure=port,invite
host=sip.txtxlxoxp.it
fromuser=5x5x7x9x0x3
fromdomain=sip.txtxlxoxp.it
disallow=all
context=from-trunk
allow=alaw
Here you're disallowing all codecs except alaw.
---
A typical invite from my provider is:
<--- SIP read from UDP:xx.yy.xx.yy:5060 ---> INVITE
sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP xx.yy.xx.yy:5060;branch=z9hG4bKt5sfh7nrvok3d5gqc3ritdv7b7
From: <sip:[email protected];user=phone>;tag=SDdgce901-90915
To: "SIPLineUser SIPLineUser"<sip:[email protected]>
Call-ID: SDdgce901-9cb68ba025684f03a4094ed71e6e04f8-ao92gd1
CSeq: 59458 INVITE
Content-Type: application/sdp
Contact: <sip:[email protected]:5060;user=phone;transport=udp>
User-Agent: Nortel SESM 14.1.0.12
Max-Forwards: 19
Supported:
com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join,x-nortel-sipvc,gin,com.nortelnetworks.im.encryption,replaces,100rel
Remote-Party-ID:
<sip:[email protected];user=phone>;screen=yes;screen-ind=0;party=calling;counter=0;npi=NPI_E164;ton=TON_NATIONAL
P-Asserted-Identity: <sip:[email protected];user=phone>
Allow: UPDATE,REFER
Content-Length: 293
v=0
o=- 0 138163748 IN IP4 xx.yy.xx.yy
s=IMSS
[email protected]
c=IN IP4 xx.yy.xx.yy
t=0 0
m=audio 43718 RTP/AVP 8 18 3 101
a=fmtp:18 annexb=no
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000/1
But....provider will only send GSM or G729.
So either you need to talk your provider into sending alaw or you need
change your allow line to "allow=alaw,gsm".
--
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