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- About Yankovskyi
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Member since
04-05-2011
35
Posts
5
Helpful
0
Solutions
Created by
Yankovskyi
in Collaboration, Voice and Video Documents
in Collaboration, Voice and Video Documents
09-05-2014
01:32 AM
09-05-2014
01:32 AM
Good day! We have done this recommendations for Windows 2003 SP2 servers and had a problem. After disable TCP/IP Offload (IPv4 Checksum Offload) we cannot access to RoggerA through RDP.Also we cannot hear port 3389 to RoggerA. After our investigation we found out that the problem was in RDP settings, because we also deleted NIC teaming and RDP was related to this. I
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07-08-2014
05:48 AM
Chintan, I see that INVITE from dialer comes with one codec g711ulaw. Please see below: INVITE sip:0675498693@10.13.81.27 SIP/2.0 Via: SIP/2.0/UDP 10.13.81.3:58800;branch=z9hG4bK-d8754z-57145a04a377e42c-1---d8754z-;rport Max-Forwards: 70 Require: 100rel Contact: <sip:8999909@10.13.81.3:58800> To: <sip:0675498693@10.13.81.27> From: <sip:8999909@10.13.81.3>;tag=23395838 Call-ID: eb13700c-00095a52-50032441-92184f3b CSeq: 1 INVITE Session-Expires: 1800 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, NOTIFY, PRACK, REFER, NOTIFY, OPTIONS --More-- Content-Type: Multipart/mixed;boundary=uniqueBoundary Supported: timer, resource-priority, replaces User-Agent: Cisco-SIPDialer/UCCE8.0 Content-Length: 608 Remote-Party-ID: <sip:@10.13.81.27>;party=calling;screen=no;privacy=off --uniqueBoundary Content-Type: application/sdp Content-Disposition: session;handling=required v=0 o=CiscoSystemsSIP-GW-UserAgent 2884 2524 IN IP4 172.19.155.41 s=SIP Call c=IN IP4 0.0.0.0 t=0 0 m=audio 19994 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 --uniqueBoundary --More-- Content-Type: application/x-cisco-cpa Content-Disposition: signal;handling=optional Events=FT,Asm,AsmT,Sit,Piano CPAMinSilencePeriod=608 CPAAnalysisPeriod=2500 CPAMaxTimeAnalysis=3000 CPAMaxTermToneAnalysis=30000 CPAMinValidSpeechTime=112 Regards, Alex
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07-08-2014
04:39 AM
Hi Chintain, thanks for response. Yes it's right we can create settings on dialpeers, but I don't have any dsp resources and PSTN provider accept only g711alaw. So I am try to find way how to change default SIP dialer codec from g711ulaw to g711alaw. regards, Alex
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07-08-2014
04:13 AM
Hello, There is one question to community: Where we can setup preferable codec in Cisco SIP dialer? It is needed to change g711ulaw which is default SIP Dialer codec to g711alaw. Thanks in advance, Regards, Alex
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Labels:
04-29-2014
08:47 AM
Hi Yuri, Have you found a solution for this case? with standart CUCM functional regards, Alex
... View more
04-08-2014
06:18 AM
Hi! What exactly CUCM version do you have? I have the same problem. regards
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Created by
Yankovskyi
in Contact Center
in Contact Center
02-05-2014
11:31 AM
02-05-2014
11:31 AM
Hi, we have the same problem with russian language. Maybe we should open request to Cisco TAC :-) Sent from Cisco Technical Support Android App
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Created by
Yankovskyi
in Contact Center
in Contact Center
11-18-2013
03:52 AM
11-18-2013
03:52 AM
Hi! Did you install this ES? Aren't there any other bug in this 7.5.10es2 release? We have the same problem with Webview.
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01-31-2013
08:32 AM
Thanks for link. There is very similar situation, but it is one distinction. We use SPANless recording server. IP Phones don't use BIB function. They don't send any RTP to recording server. They don't know anything about recording server. CUCM Publisher sends jtapi information to ZOOM. ZOOM gets RTP from SPAN port.
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01-31-2013
08:03 AM
I have attached screenshot with problems records, you can see this problem status Thank you very much!!! I will ask ZOOM developer.
... View more
01-31-2013
07:36 AM
sorry, I am saing about different codecs in one calls, is it possible? There is such error for not recorded call in ZOOM. I can attach screenshot if you want. When I didn't see this error, I had thought that a call can negotiate only one codec type for two stream.
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01-31-2013
07:20 AM
Thanks for explanation, I have checked configuration on the GW. There was voice-class codec with three codec: G711ulaw G711alaw and G729. I have left only G711 ulaw. Now I will observe. Chris could you explain more about codecs: Is there possible that call has different codecs for each stream?? I can't imagine how. Can we see what codecs was in past calls in CDR?? Thank
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01-31-2013
06:53 AM
Hi all. There are cluster CUCM 7.1.3. and three AS 5350.(SIP) In our cluster is one default Region with g711 codec. All inbound and outbound calls pass excellent. But it is a problem with recorder server (ZOOM). There are 1-2 calls per day which are not recorded and have status "Different codecs in each stream". We can see this status in ZOOM recorder server. I have checked all settings on CUCM and AS, everywhere is g711 ulaw codec. But ZOOM support insist on settings wutj codecs in CUCM. I can't catch this problem call, because it's very a a little amount of this calls and I don't know when this call will be. My question is next: how can I see what codecs was used in calls? In the past calls? I can't find it in CDR and RTMT. Thanks!!!
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Labels:
Created by
Yankovskyi
in Видео и голосовая связь (Voice and Video)
in Видео и голосовая связь (Voice and Video)
01-22-2013
01:07 AM
01-22-2013
01:07 AM
Всем доборый день, Наконецто выяснилось почему RTMT не хотел собирать логи, точнее он не хотел парсить xml файл в котором находится дата и пути к лог файлам на сервере. Все проблема была в JRE который использует RTMT. Если устанавливать RTMT на новыу Windows (только что проинтсаллирован) то RTMT будет использовать свою java которая идет бандлом и скачивается также с CUCM. Эта java находиться в месте где установлен RTMT в папке jre. После того как поставили более новую java на наш ПК и закоментировали дефолтную папку jre, RTMT начал использовать системную java и все заработало! Всем спасибо за ответы.
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01-18-2013
03:06 AM
For example: when your cell phone is switched off and I try to call you from my cell phone I will hear announcement from Cell provider that your phone is not in the service area.
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Created by
Yankovskyi
in Collaboration, Voice and Video Documents
09-05-2014
09-05-2014
Good day!We have done this recommendations for Windows 2003 SP2 servers
and had a problem.After disa...
07-18-2014
Hi Mike,Сan I guess that 200kb/s will be enough for only audio g711
call?Could you say approximate d...
07-15-2014
Thanks for link Amit.If to be more precise I'm interest in requirements
to bandwith between Internet...
07-14-2014
Hi,Everybody knows how many bandwidth is needed for audio/video calls
for Jabber Guest?Can't find an...
07-08-2014
Chintan, I see that INVITE from dialer comes with one codec g711ulaw.
Please see below:INVITE sip:06...
07-08-2014
Hi Chintain,thanks for response.Yes it's right we can create settings on
dialpeers, but I don't have...
07-08-2014
Hello, There is one question to community:Where we can setup preferable
codec in Cisco SIP dialer?It...
04-29-2014
Hi Yuri, Have you found a solution for this case? with standart CUCM
functional regards,Alex
Created by
Yankovskyi
in Contact Center
02-05-2014
02-05-2014
Hi, we have the same problem with russian language. Maybe we should open
request to Cisco TAC :-)Sen...
11-18-2013
Hi!Did you install this ES? Aren't there any other bug in this 7.5.10es2
release?We have the same pr...
01-31-2013
Thanks for link.There is very similar situation, but it is one
distinction. We use SPANless recordin...
01-31-2013
I have attached screenshot with problems records, you can see this
problem status Thank you very muc...
01-31-2013
sorry, I am saing about different codecs in one calls, is it
possible?There is such error for not re...
Public Statistics
| Date Registered | 04-05-2011 10:48 AM |
| Date Last Visited | 08-18-2017 04:03 AM |
| Total Messages Posted | 35 |
| Total Helpful Votes Received | 5 |
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