Scenario#41 – No Ringback tone from H323 Gateway going to SIP trunk

One of our customer reported an issue with ring back tone when calling their Contact center. I made a test call and observed that as a caller when you call their main Contact center number all you hear is dead silence and then when agent picks up the phone you could hear them talk. There was no ring back tone and no automated messages before an agent picks up the call. The call flow was something like this: PSTN > ISDN30 > H323 Dial-peer > SIP Trunk > 3rd-party Contact Center Equipment First I thought it could be the third-party … Continue reading Scenario#41 – No Ringback tone from H323 Gateway going to SIP trunk

Hair-pinning Or redirecting a call from Gateway

There are times when you would like to forward a call coming into your gateway out to PSTN before it goes to Call manager. Reason could be WAN outage to Call managers or Call manager down situation. This is how you can route out an incoming call out of the gateway. Let suppose the mainline number coming into the gateway is ‘881456’. The following setup will route the call out of the gateway to PSTN. voice translation-rule 10 rule 1 /881456/ /901139886666/ ! voice translation-profile RedirectPSTN translate called 10 ! dial-peer voice 10 pots description Incoming Dial Peer translation-profile incoming … Continue reading Hair-pinning Or redirecting a call from Gateway

Scenario#34 – Cannot make International Calls from US gateway

One of our customer has multiple sites across the globe including sites in UK and US. Their initial design was to send all UK calls from US over a SIP trunk to break out from a UK gateway for LCR (least cost routing). For some reason they stopped using that trunk and configured local dial peers on US gateways for international calls. All sites were working fine as they were going through one route pattern (for international calls), one route list (Local Route Group) except one site. This site was getting an ‘unallocated/unassigned number’ when calling an international number (UK … Continue reading Scenario#34 – Cannot make International Calls from US gateway

Scenario#32 – SIP Calls drop after 75 minutes

I had an interesting case where SIP calls over a SIP trunk were dropping after like 75 minutes. The duration was not confirmed as sometimes it use to drop even before 75 minutes. I looked into few CCSIP debugs (debug ccsip messages) and found that the ‘BYE’ message was actually coming from our end (Call manager/Gateway). Platform information: CCM: System version: 7.1.3.32900-4 CUBE-10#sh ver Cisco IOS Software, 3800 Software (C3825-ADVIPSERVICESK9-M), Version 12.4(24)T4, RELEASE SOFTWARE (fc2) Technical Support: http://www.cisco.com/techsupport Copyright (c) 1986-2010 by Cisco Systems, Inc. Compiled Fri 03-Sep-10 09:15 by prod_rel_team ROM: System Bootstrap, Version 12.4(13r)T, RELEASE SOFTWARE (fc1) CUBE-10 … Continue reading Scenario#32 – SIP Calls drop after 75 minutes

Scenario#31 – Calls over ICT have delayed Voice

I had an interesting case last week where calls over ICT trunk would connect but then either party will not hear each other for 8-10 seconds. The issue was first reported from US users trying to reach UK users over an Inter-cluster trunk (Non-GK). Both clusters had two call managers version 8.0.3. The whole issue started after US call managers were re-located and their IP addresses were changed. The ping response between two clusters was about 145ms which wasn’t too bad. I did a test call to a UK phone by connecting my IP communicator to US (due to time … Continue reading Scenario#31 – Calls over ICT have delayed Voice

Applying QoS for Voice Traffic

QoS is very important for Voice traffic which is delay sensitive. I won’t go into details of QoS over here and will just explain the configuration we normally use on a Voice gateway for QoS. class-map match-any Voice-RTP match ip precedence 5 match ip dscp ef match ip rtp 16384 16383 class-map match-any Voice-Cntl match ip precedence 3 match ip dscp af31 match access-group name voice-signal ! ip access-list extended voice-signal permit tcp any any range 2000 2002 permit udp any any eq 2427 permit tcp any any eq 2428 permit tcp any any eq 1720 permit tcp any any … Continue reading Applying QoS for Voice Traffic

Scenario#29 – Interworking error -0x80FF

I came across this issue with one of  our customer last week where a call was coming in to receptionist at Site ‘A’ but she was not able to transfer the call to someone at Site ‘B’. They had Centralized deployment and all phones on branch sites were registered to Central site. The call was dropping as soon as she would press ‘hold’ to transfer the call. This is what I was getting on Site ‘A’ gateway: 2621-VG01# .Feb  7 09:25:14.805: ISDN Se1/0:15 Q931: RX <- SETUP pd = 8  callref = 0x0002 Sending Complete Bearer Capability i = 0x8090A3 … Continue reading Scenario#29 – Interworking error -0x80FF