Hi,   Hoping someone can help with a Jabber MRA problem.  This seems to be affecting mainly IOS (TCT) and Android (BOT) devices and I reckon it's codec related.  The problem is we cannot dial any external landline or cell \ mobile numbers through MRA; this happens when the MRA device is connected to WiFi and 4G\cell.  The Called party device rings once and then the call automatically disconnects.  The call logs report "501 Not Implemented" and "Reason: Q.850;cause=65".   Call route is Jabber MRA device > Expressway-E > Expressway-C > CUCM > SIP Trunk to SIP provider (BT)   Windows CSF devices work fine through MRA using Jabber v11.7.0. Calls to internal numbers from any MRA device are OK (on Wifi and 4G).   CUCM is v11.5.1.13900-52 Expressway Edge and Core are vX8.10.1 Jabber on IOS and Android is latest v11.9.1   I've had a look at the call logs and I can see that the INVITE from the IOS \ Android devices includes an additional audio codec (number "111") compared to the CSF:   INVITE from Expressway to CUCM ========================== v=0 o=tandberg 0 1 IN IP4 xxxxx s=- c=IN IP4 xxxxx b=AS:1024 t=0 0 a=cisco-mari:v1 a=cisco-mari-rate m=audio 55360 RTP/AVP 114 9 104 105 0 8 18 111 101 a=rtpmap:114 opus/48000/2 a=rtpmap:9 G722/8000 a=rtpmap:104 G7221/16000 a=fmtp:104 bitrate=32000 a=rtpmap:105 G7221/16000 a=fmtp:105 bitrate=24000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:111 x-ulpfecuc/8000 a=fmtp:111 max_esel=1420;m=8;max_n=32;FEC_ORDER=FEC_SRTP a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=extmap:14/sendrecv http://protocols.cisco.com/timestamp#100us a=sendrecv a=rtcp:55361 IN IP4 10.50.100.26   INVITE from CUCM to SIP Trunk - includes the same 111 codec: =================================================   v=0 o=CiscoSystemsCCM-SIP 4388142 1 IN IP4 xxxxx s=SIP Call c=IN IP4 xxxxx b=TIAS:8000 b=AS:8 t=0 0 a=cisco-mari:v1 a=cisco-mari-rate m=audio 55360 RTP/AVP 18 114 111 101 a=extmap:14/sendrecv http://protocols.cisco.com/timestamp#100us a=rtpmap:114 opus/48000/2 a=rtpmap:111 X-ULPFECUC/8000 a=fmtp:111  max_esel=1420;m=8;max_n=32;FEC_ORDER=FEC_SRTP a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtcp:55361 IN IP4 10.50.100.26 OFFER from CIP Trunk back to CUCM ================================= We see the SIP Trunk offering back the following.  The "m=8;max_n=32;FEC_ORDER=FEC_SRTP 0" attribute seems completely out of place.   v=0 o=genband 322521088 1508711079 IN IP4 xxxxx s=- c=IN IP4 xxxxx t=0 0 m=audio 47230 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=ptime:20 m=8;max_n=32;FEC_ORDER=FEC_SRTP 0 a=rtpmap:18 0 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15   Question then is is this an issue with the Jabber app, CUCM, Expressway or the SIP provider, or something else completely?   Thanks Lee ... View more

Hi, I'm trying to setup a way to block specific incoming calls on our CUCM 9 platform.  I've got it so that it blocks specific numbers and allows through all other calls providing the caller has calling line ID enabled.  Trouble is it's also blocking all calls from unknown \ withheld numbers.  We're using SIP trunks to connect to the PSTN and the calling line ID for all withheld numbers is sent down the trunk is "9anonymous". RTMT reports Termination Cause Code (1) Unallocated (unassigned) number for the failed withheld calls I've setup the call blocking following these instructions: https://supportforums.cisco.com/document/71966/blocking-calls-based-calling-party-id http://www.netcraftsmen.com/cisco-cucm-blocking-calls-by-calling-party-number-id/ The CSS setting on the Inbound Settings of the SIP trunk uses "CS-SIP-INBOUND" The Partition PA-SIP-INBOUND is in this CSS with two Translation Patterns: ! and <null> (no title) Both have the "Route Next Hop by Calling Party Number" option enabled Both are configured to use CS-SIP-INBOUND-BLOCKED   The CSS CS-SIP-INBOUND-BLOCKED has one Partition PA-SIP-INBOUND-BLOCKED With two Translation Patterns assigned exactly as above but with RNHCPN disabled and using the standard CSS for use for national call access. I've then created Translation Patterns for the numbers to be blocked and assigned them to the PA-SIP-INBOUND-BLOCKED partition   I had though that the withheld numbers would be handled by the <null> Translation Patterns but that doesn't seem to be the case and I can't create a TP called "9anonymous" as a workaround either. Any ideas? Thanks! ... View more