Il 15/01/2014 09.59, Francesco Namuri ha scritto: > 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 > > <------------>
Found the problem, but I'm wondering how it's possible... I've a wrong configuration in a trunk: --------- [dev0x8x4x7x1x] disallow=all username=0x8x4x7x1x type=friend secret=secret qualify=yes port=5060 insecure=port,invite host=siprouter.devtel.it fromuser=0x8x4x7x1x fromdomain=sxpxoxtxr.xextxl.xt context=from-trunk-dxvxtxallow=alaw --------- but this is not the incriminated trunk, it's only one of the trunks of this provider. This wrong configuration makes unusable all the trunks of this provider (only incoming calls), also if other trunk (as in my case) are configured correctly. Another strange behavior is that my other providers works good also with the misconfigured trunk. Doing a dump of a INVITE from others server I get an a= attribute ofr any codec allowed... Maybe is this the problem? Thanks again for all answers... All the best Francesco -- _____________________________________________________________________ -- 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
