problem of presentation on SIP calls in CUCM-VSC

Similar Questions

• TC7.2.1 problems with SIP calls

  Hi, we had the following problem with sip calls:

  Here is the log:

  Version of the VCS software: Platform X7.2.3 | X8.2.1 Labor

  ______________________________________________________________________________

  Call with TC 7.2.1 does not
  ______________________________________________________________________________

  SIPMSG:
  | INVITE sip: [email protected] / * / SIP/2.0
  Via: SIP/2.0/TCP xx.xxx.60.133:57597; branch = z9hG4bK38f566387dbd719702049ddcb2590ebe.1; rport
  Call ID: 4832bd6a60d9d3d43670c55c573667ec
  CSeq: INVITE 101
  Contact: sip: [email protected] / * /: 57597; gr = urn: uuid:6 c 856033-4cbd - 5 b 15 - bb13 - 0715c4a694cd, ob>
  From: "201101102_C40_ServiceDesk" sip: [email protected] / * />; tag = d25377890df67530
  To: sip: [email protected] / * />
  Max-Forwards: 70
  Directions: sip:xx.xxx.1.51; lr>
  Allow: INVITE, ACK, CANCEL, BYE, update, INFO, OPTIONS, CONSULT, INFORM
  User-Agent: TANDBERG/521 (TC7.2.1.cb31c3d)
  "Proxy-Authorization: Digest ="bb1d830b2d64949caaadfb732cd7ca77414a991699854e8971970a68fe06"nonce, realm ="tplabvcsc01.xyz.com", qop = auth, opaque = 'AwAAAMSaxlh37 + YNQULdXHDdMkXYHVQ1', user name =" ", uri ="sip:xyz.com", answer is"51f225da16a3ac710e86c1b1b5815438", algorithm = MD5, nc = 00000008 cnonce ="21c839c8313e8f7f7477786cb40c5ee6. "
  Supported: replaces, timer, 100rel, gruu, way, way out
  Session time-out: 1800
  Content-Type: application/sdp
  Content-Length: 2842
   
  v = 0
  o = xx.xxx.60.133 IN IP4 6 2 tandberg
  s = -.
  c = IN IP4 xx.xxx.60.133
  b = AS:6000
  t = 0 0
  m = audio 16424 RTP/AVP 107 108 109 110 104 105 9 15 18 8 0 101
  b = TIAS: 128000
  a = rtpmap:107 m4as-LATM/90000
  a = fmtp:107 profile-level-id = 25; object = 23; bitrate = 128000
  a = rtpmap:108 m4as-LATM/90000
  a = fmtp:108 profile-level-id = 24; object = 23; bitrate = 64000
  a = rtpmap:109 m4as-LATM/90000
  a = fmtp:109 profile-level-id = 24; object = 23; bitrate = 56000
  a = rtpmap:110 m4as-LATM/90000
  a = fmtp:110 profile-level-id = 24; object = 23; bitrate = 48000
  a = rtpmap:104 G7221/16000
  a bitrate = 32000 fmtp:104 =
  a = rtpmap:105 G7221/16000
  a = fmtp:105 bitrate = 24000
  a = rtpmap:9 G722/8000
  a = rtpmap:15 G728/8000
  a G729/8000 rtpmap:18 =
  a = annex b fmtp:18 = yes
  a = rtpmap:8 PCMA/8000
  a = rtpmap:0 PCMU/8000
  a rtpmap:101 telephone-event/8000 =
  a = fmtp:101 0-15
  a = sendrecv
  m = video 16426 RTP / AVP 97 126 96 34 31
  b = TIAS: 6000000
  a = rtpmap:97 H264/90000
  a = packetization-mode=0;profile-level-id=428016;max-br=5000;max-mbps=245000;max-fs=9000;max-smbps=245000;max-fps=6000;max-rcmd-nalu-size=3456000 fmtp:97
  a = rtpmap:126 H264/90000
  a = packetization-mode=1;profile-level-id=428016;max-br=5000;max-mbps=245000;max-fs=9000;max-smbps=245000;max-fps=6000;max-rcmd-nalu-size=3456000 fmtp:126
  a = rtpmap:96 H263-1998/90000
  a = fmtp:96 custom = 1280, 768, 3; custom = 1280, 720, 3; custom = 1024, 768, 1; custom = 1024, 576, 2; custom = 800, 600, 1; cif4 = 1; custom = 720, 480, 1; custom = 640, 480, 1; custom = 512, 288, 1; CIF = 1; custom = 352, 240, 1; QCIF = 1; maxbr = 20000
  a = rtpmap:34 H263/90000
  a = fmtp:34 cif4 = 1; CIF = 1; QCIF = 1; maxbr = 20000
  a = rtpmap:31 H261/90000
  a cif fmtp:31 = = 1; QCIF = 1; maxbr = 20000
  a = label: 11
  a = answer: full
  a = content: hand
  a = rtcp-fb: * nack fold
  a = rtcp-fb: * ccm FIR
  a = rtcp-fb: * ccm tmmbr
  a = sendrecv
  m = application 28300 UDP/BFCP
  a = the installer: actpass
  a = confid:1
  a = userid:6
  a = floorid:2 mstrm:12
  a = floorctrl:c - s
  a = connection: new
  m = video 16428 RTP / AVP 97 126 96 34 31
  b = TIAS: 6000000
  a = rtpmap:97 H264/90000
  a = packetization-mode=0;profile-level-id=428016;max-br=5000;max-mbps=108000;max-fs=3840;max-smbps=108000;max-fps=6000;max-rcmd-nalu-size=1474560 fmtp:97
  a = rtpmap:126 H264/90000
  a = packetization-mode=1;profile-level-id=428016;max-br=5000;max-mbps=108000;max-fs=3840;max-smbps=108000;max-fps=6000;max-rcmd-nalu-size=1474560 fmtp:126

  
  a = rtpmap:96 H263-1998/90000
  a = fmtp:96 custom = 1280, 768, 3; custom = 1280, 720, 3; custom = 1024, 768, 1; custom = 1024, 576, 2; custom = 800, 600, 1; cif4 = 1; custom = 720, 480, 1; custom = 640, 480, 1; custom = 512, 288, 1; CIF = 1; custom = 352, 240, 1; QCIF = 1; maxbr = 20000
  a = rtpmap:34 H263/90000
  a = fmtp:34 cif4 = 1; CIF = 1; QCIF = 1; maxbr = 20000
  a = rtpmap:31 H261/90000
  a cif fmtp:31 = = 1; QCIF = 1; maxbr = 20000
  a = label: 12
  a = content: slides
  a = rtcp-fb: * nack fold
  a = rtcp-fb: * ccm FIR
  a = rtcp-fb: * ccm tmmbr
  a = sendrecv
  m = 16430 RTP/AVP 100 application
  a = rtpmap:100 H224/4800
  a = sendrecv
  m = application 29789 UDP/UDT/IX
  a = ixmap:0 ping
  a = ixmap:2 xccp
  |
  ________________________________________________________________________________________________

  Work of appeal with TC 7.1.4
  ________________________________________________________________________________________________

  SIPMSG:
  | INVITE sip: [email protected] / * / SIP/2.0
  Via: SIP/2.0/TCP xx.xxx.60.133:37491; branch = z9hG4bKe3dfed840a299516919a79e4a54cd707.1; rport
  Call ID: 480ec504782c0ad894929e882a696e74
  CSeq: INVITE 100
  Contact: sip: [email protected] / * /; gr = urn: uuid:6 c 856033-4cbd - 5 b 15 - bb13 - 0715c4a694cd, ob>
  From: "201101102_C40_ServiceDesk" sip: [email protected] / * />; tag = 3ab87b1c0d8b4d9d
  To: sip: [email protected] / * />
  Max-Forwards: 70
  Directions: sip:xx.xxx.1.51; lr>
  Allow: INVITE, ACK, CANCEL, BYE, update, INFO, OPTIONS, CONSULT, INFORM
  User-Agent: TANDBERG/520 (TC7.1.4.908e4a9)
  Supported: replaces, timer, 100rel, gruu, way, way out
  Session time-out: 1800
  Content-Type: application/sdp
  Content-Length: 2842
   
  v = 0
  o = xx.xxx.60.133 IN IP4 3 2 tandberg
  s = -.
  c = IN IP4 xx.xxx.60.133
  b = AS:6000
  t = 0 0
  m = audio 16394 RTP/AVP 107 108 109 110 104 105 9 15 18 8 0 101
  b = TIAS: 128000
  a = rtpmap:107 m4as-LATM/90000
  a = fmtp:107 profile-level-id = 25; object = 23; bitrate = 128000
  a = rtpmap:108 m4as-LATM/90000
  a = fmtp:108 profile-level-id = 24; object = 23; bitrate = 64000
  a = rtpmap:109 m4as-LATM/90000
  a = fmtp:109 profile-level-id = 24; object = 23; bitrate = 56000
  a = rtpmap:110 m4as-LATM/90000
  a = fmtp:110 profile-level-id = 24; object = 23; bitrate = 48000
  a = rtpmap:104 G7221/16000
  a bitrate = 32000 fmtp:104 =
  a = rtpmap:105 G7221/16000
  a = fmtp:105 bitrate = 24000
  a = rtpmap:9 G722/8000
  a = rtpmap:15 G728/8000
  a G729/8000 rtpmap:18 =
  a = annex b fmtp:18 = yes
  a = rtpmap:8 PCMA/8000
  a = rtpmap:0 PCMU/8000
  a rtpmap:101 telephone-event/8000 =
  a = fmtp:101 0-15
  a = sendrecv
  m = video 16396 RTP / AVP 97 126 96 34 31
  b = TIAS: 6000000
  a = rtpmap:97 H264/90000
  a = packetization-mode=0;profile-level-id=428016;max-br=5000;max-mbps=245000;max-fs=9000;max-smbps=245000;max-fps=6000;max-rcmd-nalu-size=3456000 fmtp:97
  a = rtpmap:126 H264/90000
  a = packetization-mode=1;profile-level-id=428016;max-br=5000;max-mbps=245000;max-fs=9000;max-smbps=245000;max-fps=6000;max-rcmd-nalu-size=3456000 fmtp:126
  a = rtpmap:96 H263-1998/90000
  a = fmtp:96 custom = 1280, 768, 3; custom = 1280, 720, 3; custom = 1024, 768, 1; custom = 1024, 576, 2; custom = 800, 600, 1; cif4 = 1; custom = 720, 480, 1; custom = 640, 480, 1; custom = 512, 288, 1; CIF = 1; custom = 352, 240, 1; QCIF = 1; maxbr = 20000
  a = rtpmap:34 H263/90000
  a = fmtp:34 cif4 = 1; CIF = 1; QCIF = 1; maxbr = 20000
  a = rtpmap:31 H261/90000
  a cif fmtp:31 = = 1; QCIF = 1; maxbr = 20000
  a = label: 11
  a = answer: full
  a = content: hand
  a = rtcp-fb: * nack fold
  a = rtcp-fb: * ccm FIR
  a = rtcp-fb: * ccm tmmbr
  a = sendrecv
  m = application 29489 UDP/BFCP
  a = the installer: actpass
  a = confid:1
  a = userid:3
  a = floorid:2 mstrm:12
  a = floorctrl:c - s
  a = connection: new
  m = video 16398 RTP / AVP 97 126 96 34 31
  b = TIAS: 6000000
  a = rtpmap:97 H264/90000

  
  a = packetization-mode=0;profile-level-id=428016;max-br=5000;max-mbps=108000;max-fs=3840;max-smbps=108000;max-fps=6000;max-rcmd-nalu-size=1474560 fmtp:97
  a = rtpmap:126 H264/90000
  a = packetization-mode=1;profile-level-id=428016;max-br=5000;max-mbps=108000;max-fs=3840;max-smbps=108000;max-fps=6000;max-rcmd-nalu-size=1474560 fmtp:126
  a = rtpmap:96 H263-1998/90000
  a = fmtp:96 custom = 1280, 768, 3; custom = 1280, 720, 3; custom = 1024, 768, 1; custom = 1024, 576, 2; custom = 800, 600, 1; cif4 = 1; custom = 720, 480, 1; custom = 640, 480, 1; custom = 512, 288, 1; CIF = 1; custom = 352, 240, 1; QCIF = 1; maxbr = 20000
  a = rtpmap:34 H263/90000
  a = fmtp:34 cif4 = 1; CIF = 1; QCIF = 1; maxbr = 20000
  a = rtpmap:31 H261/90000
  a cif fmtp:31 = = 1; QCIF = 1; maxbr = 20000
  a = label: 12
  a = content: slides
  a = rtcp-fb: * nack fold
  a = rtcp-fb: * ccm FIR
  a = rtcp-fb: * ccm tmmbr
  a = sendrecv
  m = 16400 RTP/AVP 100 application
  a = rtpmap:100 H224/4800
  a = sendrecv
  m = application 31127 UDP/UDT/IX
  a = ixmap:0 ping
  a = ixmap:2 xccp

  Another problem, indicated in diagnosis on a MX200 device tests, reporting problems of NTP. I tried to choose another NTP server - but this does not solve the problem, and it works with TC7.1.4. In addition to the time and date is wrong on the screen but OK on the touchscreen device.

  In addition, an error is reported on the OSD settings.

  Do you have advice?

  Thanks for help.
  |

  So, what's the problem? You just showed us an INVITATION. What is the configuration of your system? What is the reaction to INVITE him? We need complete diagnostic logs. Is this VCS - C registred? Where VCS - C diagnostic logs

• SIP URI B2B calls via CUCM and Highway

  Hi all

  Cannot find out how to configure the SIP URI from SX codecs, recorded calls on CUCM.

  I CUCM and IM & P 10.5.1 a pair of X8.5.2 expressway with two areas of course - for B2B calls, and the external jabber.

  I want to make calls of SX-codecs outside via SIP, by using the URI field, for example, [email protected] / * / or [email protected] / * /. How can I configure that?

  I already have the routing model, which is the SIP Trunk for Exp - C, for H323 calls by ip address. Is it possible to make a routing model to call the SIP URI?

  You can see my deployment in this image.

  fghj.PNG

  

  Best wishes

  Igor

  Hi Igor

  These guides will help you resolve your issues:

  http://www.Cisco.com/c/dam/en/us/TD/docs/voice_ip_comm/Expressway/config_guide/x8-5/Cisco-Expressway-SIP-trunk-to-unified-cm-deployment-guide-CUCM-8-9-10-and-x8-5.PDF

  https://supportforums.Cisco.com/ru/video/12362936

  BR Oleksandr

• VCS or highway: force not encrypted video sip call

  Hello community support.

  Let's say I have a vcs cisco or cisco Expressway server and I want to make a call video sip.

  to facilitate the resolution of problems, I want to have a sip call unencrypted, even if the other side external is capable of encryption (has a _sips SRV record).

  is it possible to do on X8.6.1?

  what I've tried so this withour success: (I tried in this order)

  -> Expressway-Edge: defining the Protocol of emergency (B2B) DNS Zone to the mode of encryption TCP transport and the media to "force unencrypted.

  -> the two highways: "TraversalClient (B2B) SIP" area setting transport encryption mode 'TCP' and the media to "force unencrypted.

  -> CUCM isn't in mixed mode

  ->, but the call is always encrypted Highway to the external client.

  -> When I put "Configuration-> protocols-> SIP-> TLS mode on off the call is aborted and no signal is sent to the other side."

  Based on your previous post:

  "with the outgoing call I mean it's a B2B call of an endpoint is saved to CUCM and then uses the course the way Express series server firewall to connect to a SIP Client on the outside." (external domain) »

  On my understanding, it is a vocation of B2B ARM deployment.

  CUCM controlled environment with ARM in which any calls to or from the highway E, which uses the crossing area of CUCM-Expressway is always encrypted. CUCM (X8.6 Highway) crossing area is automatically configured with the proper settings and cannot be changed which have been set to SIP TLS with TLS check the mode of commissioning and mode of encryption Media Force encryptedvalue.

  Kind regards

  Acevirgil

• SIP calls failing

  Hello

  I've implemented VCS in both sites, Site A and site B and I did the neighbors between Site A and Site B VCS rules. Search and transformations are configured and implemented and applied.

  Calls from Site B to establish a (H323 and SIP) work well, but calls from Site A to Site B work only for H323. Strange things here is localisation tools show the address cannot be resolved and true for this research nevertheless calls SIP of the Site A to Site B do not work when calls is launched of a Cisco endpoint. I already spent a few days and double checked everything but can't break this case, someone knows what's the problem?

  Thanks in advance. Mark

  Hi Mark,

  Since you are able to make SIP calls to internet via MCU and the zones are configured as donotcheckcredential, I think that the problem is with your endpoints rather than VCS. try this:

  -check the DNS configuration on your terminals (EX-series, etc.) and make sure that is correct and the connectivity is here

  -Make sure that the outgoing setting is set to off

  -Make sure the VCS FULL domain name are correct and not poorly spelled and solvable

  s ' there is no proxy between VCS and endpoints server, make sure that this field is empty.

  -I don't think issues of NTP server for SIP calls, make sure however NTP server is configured and works for endpoints

  -one last thing, try saving the end point that is already registered for site A to site B and try to make a call internet SIP, would this work?

  Best regards, Ahmad

• SIP call failures

  Hey all,.

  I'm an EX90 out in the field, running TC6.3.  The EX90 is registered via H323 & SIP at our VCS-E that is running X7.2.2.  The user can call our via H323 MCU without problem, but if the same call via SIP of his unit it disconnects immediately and says "No multimedia codecs shared" in the VCS.

  I looked at all the configs on his unit, and they are fine.  I'm not able to reproduce the problem from the other units.

  EX90 joint papers and they have been taken with the 'Extended loggin' enabled.

  Thank you

  Justin Ferello
  Technical Support Specialist
  KBZ, a Cisco authorized dealer
  http://www.KBZ.com
  e/v: [email protected] / * /

  There is no shared multimedia codecs from the MCU - it supports encryption, as endpoint is dealing with media encryption forced - RTP/BIBI, if xConfiguration conference 1 encryption Mode is set to BestEffort, so maybe it's the profile type parameter that requires.

  You will likely see the MCU logs to see what he doesn't like.

  Thank you

  Guy

• Microsoft or windows never contact you by phone about problems, that I get these calls very frustrating!

  Microsoft or windows never contact you by phone about problems, that I get these calls very frustrating!

  Fake phone calls to technical support

  Calls unrequested telephone of this nature are almost always a common scam. Do not let them give any info, do not give access to your PC, not give them all the money and do not go to all the websites that they suggest. One moment.

  Please see:

  http://www.Microsoft.com/en-GB/security/online-privacy/avoid-phone-scams.aspx

  Microsoft issues never not solicited for phone calls of support or security.

  These types of callers can use completely fictitious names or bogusly claim to be of real companies other than Microsoft as well.

• Replacement of VG224 in Cisco Unified Call Manager (CUCM)

  Hello

  I'm about to replace some of the access points virtual (VG224) to one of my sites and have a few concerns:

  1. Copy configs of the old to the new VG224 will be sufficient?
  2. What needs to be done on the Cisco Unified Call Manager (CUCM) to complete the replacement VG224?
  3. Should I change the configs in Cisco Unified Call Manager if I change the IP address of the VG224 (VG224 migration to the new voice VLAN)?

  Thank you

  Yahya,

  Hope it goes well...

  V is standing for VOICE, not virtual. Vg224 is gateway224 of voice. The answers for your quastion will be...

  1 - Yes, it is sufficient.

  2 - CUCM administration page > device > gateway > find. Select your gateway, change the mac address (10 digits) of the old appliance with a new one.

  3. with the help of sccp Protocol, CUCM don't care about ip address, it's care only about mac address. If the answer is, NO.

  hope that could help

  Concerning

• Rogue SIP calls on C40

  Hello

  We have a videoconference system with the public IP address that is registered in the VCS-E received various rogue SIP calls.  All these SIP calls are aliases for figures 3-6 during 32sec each.  The calling address entering is: alias @ public IP address of our VC system. All calls are video calls to 384 k

  To avoid calls, I disabled the SIP option and since then we have not received one of these calls.  However, we need to connect to other devices using SIP.  Is there another way to stop these calls?

  Thanks in advance for your help.

  Amrit

  Make the following settings on your endpoints:

  • xConfiguration SIP ListenPort: Off
  • xConfiguration SIP profile 1 out: on

  See bug CSCue55239 for more details.

  You will also need to take steps to secure your VCS if you have not already, turning off UDP SIP SIP calls stop.  However, last year, we saw these calls come on H323 TCP, the only way to stop calls H323 is either secure endpoint behind your firewall and use a script CPL on the VCS.  See the analysis of sourceh323idcisco-incomingcalls on how to configure a CPL.

  FYI, the forums to search would be my first place to look, or search for bugs.  He is asked everywhere in the forums.

• Problem to receive or make calls to or from different domains

  Hello

  I have control of VCS and VCS Expressway, calls between registered devices including works correctly.

  The problem when I want to call example cisco, do not know what settings should I add to be able to call someone who is not my area, or they can call me.

  There was cisco events where you can enter through telepresence and I could not get the following message is displayed:

  Not registered - Destination not found

  I suspect it's a configuration problem, where, I don't know.  We will need some information to help them to find the problem.

  1. Are you able to communicate with others outside your network, which is not Cisco?
  2. You have a DNS zone configured on your highway to VCS and a rule of research pointing to it?
  3. You are able to see your call attempt in the history of research of the highway?  If so, can you share it with us?

• call of CUCM to the router voice FXS thanks CUBE problem

  Hello

  We had a problem in our test environment:

  - - - - Analog telephone

  When we on the g711 codec, both directions of appeal are ok.

  When we put the codec for g729r8, appeal from analog to IP has no problem. But for reverse, the analog phone can ring but will be disconnected after a few seconds. Before the call is disconnected, we find that the codec is "pre-ietf g729" using the command 'show call active voice brief' on the CUBE.

  The configuration of the CUBE and Router (FXS) voice is attached. Really appreciate your advice.

  Thank you!

  Eric

  It seems that the analog phone only supports pre - itf g729 format...

  Unfortunately this codec format is no longer supported on any ios...

  http://www.Cisco.com/c/en/us/support/docs/voice/h323/14069-codec-complexity.html#pre_ietf

  You can try your luck and see if this pre - itf is available on ios, but I doubt it. You will need to have them on all your dial peers.

   maui-vgw-01(config)#dial-peer voice 100 voip maui-vgw-01(config-dial-peer)#codec g729r8 pre-ietf 

  The pre-ietf in this command option is not supported in Cisco IOS release 12.2 and later.

• can't make an external call with CuCM and H323 gateway

  Hi experts, I have a problem with CuCM and h323 Gateway. I use a 2811 router to make a gateway H323, CuCM server and Switch layer 2 build a VoIP LAB, this my network:

  

  and my router configuration:

  Current configuration: 2335 bytes

  !

  version 12.4

  horodateurs service debug datetime msec

  Log service timestamps datetime msec

  no password encryption service

  !

  router host name

  !

  boot-start-marker

  boot-end-marker

  !

  forest-meter operation of syslog messages

  !

  No aaa new-model

  !

  !

  !

  dot11 syslog

  IP source-route

  !

  !

  IP cef

  DHCP excluded-address 192.168.1.1 IP 192.168.1.10

  !

  IP VOICE dhcp pool

  network 192.168.1.0 255.255.255.0

  option 150 ip 192.168.1.2

  default router 192.168.1.1

  !

  !

  No ipv6 cef

  !

  Authenticated MultiLink bundle-name Panel

  !

  !

  voip phone service

  h323 connections allow h323

  allow connections h323 to SIP

  allow connections sip h323

  allow sip to sip connections

  service additional h450.12

  Fax protocol cisco

  !

  !

  !

  vocal h323 class 1

  H225 timeout tcp establish 3

  !

  !

  !

  translation of the voice-rule 1

  rule 1 / ^ 9.

  !

  !

  voice translation-profile PSTN

  translate 1 called

  !

  !

  voice-card 0

  !

  !

  application

  Service flash aa1: its - cisco.2.0.2.0.tcl

  paramspace English language en

  paramspace English index 1

  Param aa-driver 0839959204

  paramspace location English flash:

  prefix English paramspace

  !

  !

  !

  Archives

  The config log

  hidekeys

  !

  !

  interface FastEthernet0/0

  IP 192.168.1.1 255.255.255.0

  automatic duplex

  automatic speed

  H323-gateway voip interface

  port of link voip H323-gateway 192.168.1.1

  !

  interface FastEthernet0/1

  no ip address

  Shutdown

  automatic duplex

  automatic speed

  !

  IP forward-Protocol ND

  no ip address of the http server

  no ip http secure server

  !

  !

  !

  control plan

  !

  !

  !

  voice-port 0/0/0

  cover of the echo - cancel 32

  connection ERA opx 3100

  Description 08395959204

  activation of the caller ID

  !

  voice-port 1/0/0

  !

  voice-port 0/0/2

  !

  voice-port 0/0/3

  !

  !

  !

  1010 Dial-peer voice pots

  Service aa1

  incoming called number 08395959204

  direct line to inside

  port 0/0/0

  !

  voice pots Dial-peer 2020

  PSTN description

  translation-profile outgoing PSTN

  destination-model 9 t

  Setup progress_ind allow 3

  progress_ind enable progress 8

  port 0/0/0

  Forward-digits all the

  !

  Dial-peer voice 10 voip

  destination-model 1...

  Setup progress_ind allow 3

  progress_ind enable progress 8

  h323 voice-class 1

  session target ipv4:192.168.1.2

  DTMF-relay h245 alphanumeric

  Codec g711ulaw

  No vad

  !

  !

  !

  !

  Line con 0

  line to 0

  line vty 0 4

  Cisco password

  opening of session

  !

  Scheduler allocate 20000 1000

  end

  I have configured the gateway h323 and attach files to router parttern on follow CUCM

  What is the problem and how to solve?

  Thanks for all the help.

  Your DP pointing CUCM is wrong, or your ÉRA OPX, depending on what you need.

  voice-port 0/0/0

  cover of the echo - cancel 32

  connection ERA opx 3100

  Description 08395959204

  activation of the caller ID

  Dial-peer voice 10 voip

  destination-model 1...

  Setup progress_ind allow 3

  progress_ind enable progress 8

  h323 voice-class 1

  session target ipv4:192.168.1.2

  DTMF-relay h245 alphanumeric

  Codec g711ulaw

  No vad

  Your DP is waiting for 1... and you send 3100, they must match.

  You might need to change the significantt numbers and entering CSS, but only you can know

  For outgoing, you are throwing 9 predot, but again, expect it in your RFP, again, change as you need.

  voice pots Dial-peer 2020

  PSTN description

  translation-profile outgoing PSTN

  destination-model 9 t

  Setup progress_ind allow 3

  progress_ind enable progress 8

  port 0/0/0

  Forward-digits all the

  You cannot transfer the numbers all send you also 9 for each call

  Change your configuration you want / need.

  HTH

  Java

  If it helps, please note

  www.Cisco.com/go/pdihelpdesk

• Problem with the microphone in calls

  So when I'm too good to make and receive calls the person on the other end cannot hear me. I can hear them fine but they can't hear anything on my side. The first time it is is past that it could almost be bypassed by pressing the handsfree option then that shut off again. Now, every time I try to make a call or receive they don't heard me not at all. It's on a Z3 Sony android version 5.1.1 any execution help would be fantastic as it is a phone otherwise fantastic!

  Hi @Owellyn

  Please, start your machine in SafeMode as explained here. If the problem is not present in Mode safe, which indicates the origin of the problem is a third-party application. In this case exit safe mode, and then uninstall the third-party applications you don't need and erase data from the cache for applications that you want to keep. Sometimes this is caused by google now 'Google OK detection of any screen' then you can try to turn off. I m at disposal for any question.

• Problems of transfer/answering a call when using the headset with 79XX phone - CSCun26289

  Hello

  Runing on CUCM 8.6.2 and preparing an upgrade to v9, we have recently updated our IP phone firmware.

  We use the 79xx series. We have updated the firmware to * SCCP41.9 - 3-1 SR 4 - 1 s *

  We are now facing a very strange problem that is referenced in the bug below.

  When you use the helmet our users cannot transfer an ongoing call or answer a call when ther is already active.

  There seems to be a fix in 1.0000 ES27

  Does anyone know where this fixed version?

  https://Tools.Cisco.com/bugsearch/bug/CSCun26289

  See you soon,.

  Olivier

  9.4 (2) Firmware was released Friday. There is no release notes for this version, but in tests, it seems that the bug has been squashed.

  GTG

• Dialing SIP calls SIP Trunk (Avaya SM)

  Hello. I work with the telco team to send sip of our endpoints VTC voice calls to an Avaya Session Manager.

  I think I have all the areas and rules of research on the side of VCS in order to transfer the call to the SM.  I am able to confirm with the team of the phone company that the call reached the SM, but they receive the error "cannot determine realm."

  On the side of the VCS, I see the output below.  Is the problem on the VCS or SM side?  I need to create a trust with the SM? THX. - F

  • searchRule (2)
  • • Name: SIP Audio out rule2
  • • Zone (1)
    • • Name: TEST Session Mgr
    • • Type: neighbor
    • • Protocol: SIP
    • • Found: false
    • • Reason: Internal server error (cannot determine the domain)
    • • StartTime: 2015-03-09 13:21:58
    • • Time: 0.05
    • • Access controller (1)
      • • Address: 172.x.x.x:5061
      • • Alias (1)
        • • Type: Url
        • • Origin: unknown
        • • Value: [email protected] / * /

  Another option is to see if you get up with 5060/udp or 5060/tcp, which could make it easier to trace. But yes, I prefer the encryption as well.

  Often you have set a kind of trust for a specific connection, which could include the field src or dst, ip, port,...

  good luck ;-)