Client switched from jabber to hard phones. And when we did the audio quality in recordings is very bad. Very choppy and words getting clipped. We were using 7941s. But switched a couple to 8841s and same thing is happening. Now the odd part is I capture on the switch and stream from the BIB is bad there. But the customer call to agent has no issues and is crystal clear on that same call.  ... View more

Hello,   So at a loss here. We are troubleshooting an audio issue where words are being clipped. Agents are using jabber 12.1.1. It is very intermittent and effecting almost all. Went through normal procedure and don't see anything on network. Took captures at lots of points including at PC. From PC wireshark cap. Do not hear the issue. However can confirm while listening I do  hear the problem.   My real question is from jabber logs.I can see. 19-06-27 15:26:42,467 DEBUG [0x00002744] [rc\framework\ServicesDispatcher.cpp(220)] [services-dispatcher] [CSFUnified::ServicesDispatcher::executeTask] - executed (Calling TelephonyServiceImpl::onConversationMediaStatisticsChanged from TelephonyAdapter call stats timer thread) in [0] milliseconds. Waiting time was [0] milliseconds 2019-06-27 15:26:42,486 DEBUG [0x000014fc] [PME(0) ] [pme] [<Audio>] - Clipped 8 samples. 2019-06-27 15:26:42,500 DEBUG [0x000014fc] [PME(0) ] [pme] [<Audio>] - Clipped 8 samples. 2019-06-27 15:26:42,510 DEBUG [0x000014fc] [PME(0) ] [pme] [<Audio>] - Clipped 8 samples. 2019-06-27 15:26:42,530 DEBUG [0x000014fc] [PME(0) ] [pme] [<Audio>] - Clipped 7 samples.   Anyone familiar with jabber logs. Does that indicate it has a problem with the rtp stream and basically is skipping to next packet?   From the stats on the call however i see no latency or jitter.   2019-06-27 15:26:42,632 DEBUG [0x000010b4] [urce\cpve\src\main\sessionimpl.cpp(1597)] [cpve] [CSF::media::rtp::SessionImpl::logRtpStatsWithLock] - RTP STATS,session_id=116,session_type=audio-main,rx_bytes_recv=37536,rx_pkts_recv=1173,rx_pkts_lost=0,rx_curr_loss=0.00%,rx_cum_loss=0.00%,rx_bitrate=7996,rx_jitter=0,rx_average_jitter=1.89,rx_concealed_seconds=0.000,rx_severely_concealed_seconds=0.000,rx_skew=0,rx_avg_latency=0.00,rx_fec_pkts_recv=1173,rx_fec_pkts_lost=0,rx_fec_curr_loss=0.00%,rx_fec_cum_loss=0.00%,tx_bytes_sent=37024,tx_pkts_sent=1157,tx_pkts_recv=0,tx_pkts_lost=0,tx_curr_loss=0.00%,tx_cum_loss=0.00%,tx_bitrate=7995,tx_jitter=0,tx_round_trip=0,tx_avg_round_trip=0.00,tx_last_rb_ssrc=00000000,tx_last_ext_high_seq=0,tx_received_rb=0,tx_active_sources=1,tx_total_sources=1 20   ... View more

Hello,   Had a question to exactly how this is seizing the session with SIP. With this service on PRI years ago just using dtmf tones you would hit *8 and it would go back to carrier to initiate the take back. But with SIP I don't believe just hitting *8 from a SIP phone the carrier will know what to do with it. This would just send a notify back to SIP provider. So I guess the best way maybe is to setup a script in UCCE/CVP which would play those tones and send a refer back on that same session?   thanks ... View more

Hello,   When logging into CUCM and you are presented with this warning message. Is there any way to monitor for this occurrence? Doesn't appear to be a syslog or RTMT alert. Only if it is in overage for those. I found some OIDs but when I try to walk they do not return anything.   http://www.oidview.com/mibs/9/CISCO-LICENSE-MGMT-MIB.html ... View more

Hello,   Does anyone know a couple queries for CER which will show phones it's learned via switch port? With just mac, DN.   And unallocated phones?   I see the tables via show tech systables. But they don't contain the data I would think. For example. run sql select * from cerunlocatedphone   Only shows about 1/5 of the actual unallocated phones I see from the GUI.   Thanks ... View more

A customer has security concerns with TFTP protocol. Is there any other way to get the firmware to a phone without using TFTP? ... View more