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:
--- 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 --- 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 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sqn: 0 a=cdsc: 1 image udptl t38 <-------------> <--- 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 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sqn: 0 a=cdsc: 1 image udptl t38 <-------------> I noted that in the invite I get the rtpmap attribute only for codec 18, 3 but not for 8, it could be a problem? The refuse is: <--- Reliably Transmitting (NAT) to xx.yy.xx.yy:5060 ---> SIP/2.0 488 Not acceptable here^M Via: SIP/2.0/UDP 77.239.128.7:5060;branch=z9hG4bKt5sfh7nrvok3d5gqc3ritdv7b7;received=77.239.128.7;rport=5060^M From: <sip:[email protected];user=phone>;tag=SDdgce901-90915^M To: "SIPLineUser SIPLineUser"<sip:[email protected]>;tag=as08516b97^M Call-ID: SDdgce901-9cb68ba025684f03a4094ed71e6e04f8-ao92gd1^M CSeq: 59458 INVITE^M Server: FPBX-2.11.0(10.12.3)^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH^M Supported: replaces, timer^M Reason: Q.850;cause=58^M Content-Length: 0^M ^M <------------> Have you any advice on how to troubleshoot it? Thanks in advance All the best, Francesco Namuri -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
