Francesco Namuri
2014-Jan-15 08:59 UTC
[asterisk-users] No compatible codecs, not accepting this offer!
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:5x5x7x9x0x3 at 192.168.1.168:5060 SIP/2.0 Via: SIP/2.0/UDP xx.yy.xx.yy:5060;branch=z9hG4bKt5sfh7nrvok3d5gqc3ritdv7b7 From: <sip:3x8x6x3x3x at 10.39.1.19;user=phone>;tag=SDdgce901-90915 To: "SIPLineUser SIPLineUser"<sip:5x5x7x9x0x3 at sip.txtxlxoxp.it> Call-ID: SDdgce901-9cb68ba025684f03a4094ed71e6e04f8-ao92gd1 CSeq: 59458 INVITE Content-Type: application/sdp Contact: <sip:3x8x6x3x3x at 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:3x8x6x3x3x at 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 at 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=unknown at 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 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:5x5x7x9x0x3 at 192.168.1.168:5060 SIP/2.0 Via: SIP/2.0/UDP xx.yy.xx.yy:5060;branch=z9hG4bKt5sfh7nrvok3d5gqc3ritdv7b7 From: <sip:3x8x6x3x3x at 10.39.1.19;user=phone>;tag=SDdgce901-90915 To: "SIPLineUser SIPLineUser"<sip:5x5x7x9x0x3 at sip.txtxlxoxp.it> Call-ID: SDdgce901-9cb68ba025684f03a4094ed71e6e04f8-ao92gd1 CSeq: 59458 INVITE Content-Type: application/sdp Contact: <sip:3x8x6x3x3x at 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:3x8x6x3x3x at 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 at 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=unknown at 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 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:3x8x6x3x3x at 10.39.1.19;user=phone>;tag=SDdgce901-90915^M To: "SIPLineUser SIPLineUser"<sip:5x5x7x9x0x3 at sip.totalvoip.it>;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
James Sharp
2014-Jan-15 09:09 UTC
[asterisk-users] No compatible codecs, not accepting this offer!
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=alawHere 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 at 192.168.1.168:5060 SIP/2.0 > Via: SIP/2.0/UDP xx.yy.xx.yy:5060;branch=z9hG4bKt5sfh7nrvok3d5gqc3ritdv7b7 > From: <sip:3x8x6x3x3x at 10.39.1.19;user=phone>;tag=SDdgce901-90915 > To: "SIPLineUser SIPLineUser"<sip:5x5x7x9x0x3 at sip.txtxlxoxp.it> > Call-ID: SDdgce901-9cb68ba025684f03a4094ed71e6e04f8-ao92gd1 > CSeq: 59458 INVITE > Content-Type: application/sdp > Contact: <sip:3x8x6x3x3x at 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:3x8x6x3x3x at 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 at 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=unknown at 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/8000/1But....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".
Francesco Namuri
2014-Jan-15 14:34 UTC
[asterisk-users] No compatible codecs, not accepting this offer!
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:5x5x7x9x0x3 at 192.168.1.168:5060 SIP/2.0 > Via: SIP/2.0/UDP xx.yy.xx.yy:5060;branch=z9hG4bKt5sfh7nrvok3d5gqc3ritdv7b7 > From: <sip:3x8x6x3x3x at 10.39.1.19;user=phone>;tag=SDdgce901-90915 > To: "SIPLineUser SIPLineUser"<sip:5x5x7x9x0x3 at sip.txtxlxoxp.it> > Call-ID: SDdgce901-9cb68ba025684f03a4094ed71e6e04f8-ao92gd1 > CSeq: 59458 INVITE > Content-Type: application/sdp > Contact: <sip:3x8x6x3x3x at 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:3x8x6x3x3x at 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 at 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=unknown at 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 > 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:5x5x7x9x0x3 at 192.168.1.168:5060 SIP/2.0 > Via: SIP/2.0/UDP xx.yy.xx.yy:5060;branch=z9hG4bKt5sfh7nrvok3d5gqc3ritdv7b7 > From: <sip:3x8x6x3x3x at 10.39.1.19;user=phone>;tag=SDdgce901-90915 > To: "SIPLineUser SIPLineUser"<sip:5x5x7x9x0x3 at sip.txtxlxoxp.it> > Call-ID: SDdgce901-9cb68ba025684f03a4094ed71e6e04f8-ao92gd1 > CSeq: 59458 INVITE > Content-Type: application/sdp > Contact: <sip:3x8x6x3x3x at 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:3x8x6x3x3x at 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 at 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=unknown at 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 > 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:3x8x6x3x3x at 10.39.1.19;user=phone>;tag=SDdgce901-90915^M > To: "SIPLineUser SIPLineUser"<sip:5x5x7x9x0x3 at sip.totalvoip.it>;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