Hi, I have a scenario where outbound to SIP ITSP I’m using early offer (no CUBE, direct SIP trunks on the CUCM).  The IP Phone to the SIP Trunk is configured with Region relationship of a maximum bandwidth of 8k.   Outbound to the ITSP I have the below attributes in the INVITE SDP: b=TIAS:8000 b=AS:8 t=0 0 m=audio 23004 RTP/AVP 113 115 18 101 b=TIAS:8000 a=rtpmap:113 AMR-WB/16000/2 a=fmtp:113 mode-change-capability=2 a=rtpmap:115 AMR-WB/16000 a=fmtp:115 octet-align=1;mode-change-capability=2 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15   The ITSP is replying with a Session 183 with the following SDP: a=sendrecv m=audio 33636 RTP/AVP 115 b=RS:375 b=RR:1125 a=maxptime:40 a=rtpmap:115 AMR-WB/16000 a=fmtp:115 octet-align=1;mode-set=0,1,2;mode-change-period=2;mode-change-capability=2;mode-change-neighbor=1;max-red=0   CUCM then errors in the log stating unable to parse bandwidth elements – which look to be the RS and RR RTCP statements:   96587296.010 |11:33:20.535 |AppInfo  |//SIP/SDPLib/Warning/0x0/ sdp_parse_bandwidth:   Warning: BW Modifier type unsupported (RS) .   96587296.011 |11:33:20.535 |AppInfo  |//SIP/SDPLib/Warning/0x0/ sdp_parse_bandwidth:   Warning: BW Modifier type unsupported (RR) .   CUCM then tries to allocate a transcoder.  96587330.014 |11:33:20.538 |AppInfo  |DET-MediaManager-(897449)::preCheckCapabilities, caps mismatch! Xcoder Reqd. kbps(8) , filtered A[capCount=6 (Cap,ptime)= (98,0) (98,0) (11,20) (12,20) (15,20) (16,20)], B[capCount=0 (Cap,ptime)=] allowMTP=0 numXcoderRequired=1 xcodingSide=2   96587331.000 |11:33:20.538 |SdlSig   |MrmAllocateXcoderResourceReq           |waiting                        |MediaResourceManager(2,100,131,1) |MediaManager(2,100,133,897449)   |2,100,13,15442316.116659^193.113.149.54^* |[R:N-H:0,N:0,L:0,V:0,Z:0,D:0]   CI=38972224 MRGLPkid=13ea34b5-71fb-9408-4333-baeb2c893968 Kpbs=0 RegionA=RE-UK-YABL CapA=10 RegionB=RE-UK-YCEN CapB=1 SuppressFlag=0 Type=1 DeviceCapablity= [0x0] 0 MandatoryCapabilties= [0x0] 0 Count=1   My transcoder has no ability to transcode AMR-WB therefore the call fails.  Why is a transcoder required?? Is it because AMR-WB may use more than 8kbps? Is it because it couldn’t determine the bandwidth required by the ITSP as it was unable to parse some of the SDP content? Or is it because of a packertization mis-match, I see the cap,ptime states 98,0 – which confuses me.   Any thoughts?     ... View more

Hi, I have an issue with a customer who is complaining sometimes their phone invokes a call forward timer after 2 rings rather than 3.  This appears to only happen every couple of weeks, looking at the CUCM logs I can see that the call is offered for exactly 10 seconds prior to forwarding, the issue is obviously we set a call forward timer in seconds rather than number of rings, does anyone have any info on whether the ring cycle can vary depending on how its initiated (for example sometimes my PSTN phone seems to give a quick bust of half a ring prior to returning to its normal ring cycle) The simple solution would be to increase the 10 second timer, however this only happens every couple of weeks and for a single user so would be good to understand the cause. ... View more

Hi, I have calls coming into my cluster in E164 format, whilst retaining 7 digit internal extension numbers. I have two clusters setup using a non-Gatekeeper controlled inter cluster trunk, the issue I have is the + not being sent across the ICT, for example: Inbound call to 5006000 from PSTN, displays as +44141123456 5006000 selects transfer and calls 4006000, displays as from 5006000 (consultative transfer) 5006000 completes transfer, the display on 4006000 updates to show the call as from 44141123456 (no leading +) Looking at the logs on the 4006000 cluster the + does not appear, so assume its not supported across the ICT, so my question is how can I re-insert the +, I have tried to do this using the Incoming Calling Party transformation on the Trunk however I can't get this working to update the originally calling party number (44141123456), I can only seem to modify the calling party initiating the call across the ICT (5006000) ... View more

Hi - I am trying to allow E164 dialling and normal 9 prefix dialling for outbound calls.  I currently use 9.@ with route filters to restrict access to specfic numbers.  I therefore want to replicate these to +44.@, the issue is the existing route filters would have the number format with a leading 0 which needs to be removed when changing to E164. Looking at the TAG there is a NATIONAL-ACCESS tag which effectively specifics the leading 0 in GB numbers, so I would think I can simply copy the existing Route Filters but removing the NATIONAL-ACCESS EXISTS part of them.  However even if I remove this, the CUCM still seems to expect to see the leading 0. Any ideas, do has anyone else configured + dialling using Dial Plan for restrictions? ... View more

Hi,  I appreciate there are several discussion along this line already, however I have a H323 configured gateway on a CUCM for PSTN access, with an FXS Card used by Fax Machines which is controlled by MGCP. Reading the exiting posts I believe they describe fallback for when the PSTN access is also MGCP, I hope the below will explain my problem. Below is an extract of the config at present. application global   service alternate Default ! ccm-manager fallback-mgcp ccm-manager redundant-host 10.79.0.11 ccm-manager mgcp no ccm-manager fax protocol cisco ccm-manager music-on-hold ccm-manager config server 10.79.16.18 10.79.16.10 ccm-manager config ! mgcp mgcp call-agent 10.79.16.11 2427 service-type mgcp version 0.1 mgcp modem passthrough voip mode nse mgcp package-capability rtp-package mgcp package-capability sst-package mgcp package-capability pre-package mgcp default-package mt-package no mgcp timer receive-rtcp mgcp sdp simple mgcp fax t38 inhibit ! mgcp profile default ! dial-peer voice 999010 pots service mgcpapp port 0/1/0 ! dial-peer voice 999011 pots service mgcpapp port 0/1/1 ! dial-peer voice 1 pots description *** DEFAULT INCOMING - PSTN *** incoming called-number . direct-inward-dial port 0/0/0:15 ! dial-peer voice 2 voip description *** DEFAULT INCOMING - VOIP *** voice-class codec 1 voice-class h323 1 incoming called-number . ! dial-peer voice 10 pots destination-pattern 9T progress_ind setup enable 3 progress_ind alert enable 8 progress_ind progress enable 8 progress_ind connect enable 8 direct-inward-dial port 0/0/0:15 ! dial-peer voice 100 voip description *** CUCM - PRIMARY *** preference 1 destination-pattern 513.... voice-class codec 1 voice-class h323 1 session target ipv4:10.79.0.11 dtmf-relay h245-alphanumeric fax rate disable ip qos dscp cs5 media no vad ! dial-peer voice 101 voip description *** CUCM - SECONDARY *** preference 2 destination-pattern 513.... voice-class codec 1 voice-class h323 1 session target ipv4:10.79.16.11 dtmf-relay h245-alphanumeric fax rate disable ip qos dscp cs5 media no vad ! To enable the FXS (extension 513100) to recieve calls in SRST I believe I need a new dial-peer along the lines of: dial-peer voice 201 pots description *** FXS 0/1/0 for SRST *** incoming called-number 513100 port 0/1/0 However, if I add this configuration all inbound calls to 513100 even when not in SRST will match dial-peer 201 rather than 100 or 101, will I need to configure 3 new dial-peers? 1st pref IP of CUCM Primary, 2nd pref IP of CUCM Secondary and 3rd pref Port 0/1/0, this will then route the call to the CUCM Subscribers under normal operation but send straight to the FXS port when in SRST? ... View more