Il 15/01/2014 10.09, James Sharp ha scritto:
> 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:5x5x7x9x0x3@192.168.1.168:5060 SIP/2.0
>> Via: SIP/2.0/UDP
>> xx.yy.xx.yy:5060;branch=z9hG4bKt5sfh7nrvok3d5gqc3ritdv7b7
>> From: <sip:3x8x6x3x3x@10.39.1.19;user=phone>;tag=SDdgce901-90915
>> To: "SIPLineUser SIPLineUser"<sip:5x5x7x9x...@sip.txtxlxoxp.it>
>> Call-ID: SDdgce901-9cb68ba025684f03a4094ed71e6e04f8-ao92gd1
>> CSeq: 59458 INVITE
>> Content-Type: application/sdp
>> Contact: <sip:3x8x6x3...@xx.yy.xx.yy: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:3x8x6x3...@sip.txtxlxoxp.it;user=phone>;screen=yes;screen-ind=0;party=calling;counter=0;npi=NPI_E164;ton=TON_NATIONAL
>> P-Asserted-Identity: <sip:3x8x6x3x3x@10.39.1.19;user=phone>
>> Allow: UPDATE,REFER
>> Content-Length: 293
>>
>> v=0
>> o=- 0 138163748 IN IP4 xx.yy.xx.yy
>> s=IMSS
>> e=unkn...@invalid.net
>> 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"
Hello James,
thanks for your answer, I supposed this too, but my provider answered me
that as
m=audio 43718 RTP/AVP 8 18 3 101
^ ^ ^---- GSM proposal
^ ^------- G729 proposal
^---------- aLaw proposal
And that
a=rtpmap:18 G729/8000 proposed as media conversion
a=rtpmap:3 GSM/8000/1 because the call is made by a mobile
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