Codec negotiation in cucm !

Hi all.
I have registered 2 CIPC with cucm 7. Now when i make call and press "i" twice, the codec i see is G.729. In my region i have selected call to be made using G.711. I know thats the highest bit rate actually. So is this due to codec negotiation ? can some one give me any highlights as to how codec negotiation works in cucm ? also if there is any cisco doc ?
thanks

Hi Everyone
Thanks for a wonderful explanation from Aaron and Jaimie ,though has a bit doubt of the same .
My scenario is hq and sb are 2 regions and
hq<--->hq -------->G711
hq<---->sb--------->g729
sb<---->sb---------->g711
and i put them in corresponding device pool .
i have a phone in hq and 1 phone in sb .My sip trunk is in hq device pool.
I have a pstn gateway simulated in which a dial peer is configued to accept incoming sip calls from the trunk.
dial-peer voice 1 voip
incoming-called-number .
codec g711ulaw
now when i call from hq phone call works and 2 ends negotiates g711 codec.
when calling from sb phone call is terminated as codec negotiation fails
then changed scenario of pstn gateway.
dial-peer voice 1 voip
incoming-called-number .
codec g729r8 //default codec
now calls from both hq and sb phone succeeds with negotiating g729 codec.
Now my doubt is why call from hq to pstn phone succeds negotiating g729 codec because
hqphone to siptrunk_hq ---->g711 and pstn phone is specific to use of g729 codec ? why this call has not failed ?
and comapring the fact other call failed
ie sbphone to siptrunk_hq ----->g729 and pstn phone is specific to use g711 codec.
So who negotiates here ? is the source side or is the destination side ?
Any documents related to same will also help .
Thanks,
Harish

Similar Messages

Codec negotiation

Hi all,
In my lab scenario i have left everything to default. No regions defined other then default. Now i have registered 2 ip phones and when i click on their "?" symbol, the codec i saw was G729. Then i created region and assigned codec (or max bandwidth) to G711. Now i made the call it was still G729. Can someone tell me the procedure of this codec negotiation ? like which codec is considered first and so forth ? any reference to document would be appreciated, pls guide me

Are the IP phones on the same cluster? The default codec usually is ciscis g722.
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmsys/a08ipph.html#wp1034008
Regards,
Yosh

Codec Negotiation for Agents

HI
I havae one issue regarding codec negotiation , Customer dials in listen IVR call transfer to Agent 1 (Whose phone is on G711) Agent 1 transfer call back IVR using Blind Transfer Call goes back to IVR & Customer again press 0 to go back to new agent -2 . Agent-2 (Whose codec on Phone is G729) gets ring on his CTIOS client but when I press Answer call goes back to Queue and this happens until agent is forced not ready.
Anyone faced this issue.
Regards
Irfan Tariq

Lets try this way:
Make a call from Agent1 to Agent 2, answer the call and see on phone what codec is being used between two IP Phones.If that works,Lets move on to ICM, enable Display to Screen on CTIOS Server in the registery with regedit from 0 to 1. Something like, My Computer\HEY_LOCAL_MACHINE\SOFTWARE\Cisco Systems, Inc\ICM\icm0\CTIOS1\EMS\CurrentVersion\Library\Processes\ctios
Restart the process ctios server and you should see the agents login in and logout. When the call is dropped, you will see another message on ctios server. Paste that message.
Also which version of CVP you are using? I recall in 8.0.1, there is something like flat G.729 Codec which means use of single CODEC on all call legs with no call legs using different codecs.
Hope this helps.
Cheers

Cisco Jabber Video codec negotiation

Hi,
I am wondering if there is a way to limit video negotiation to a one specific video codec: I need that to troubleshoot one way video call prb. I am using Cisco jabber for telepresence ver 4.6 with VCS ver X7.2.1.
Regards,
YM

I suggest moving this thread to the TelePresence space; most of the people who understand Movi are monitoring that area.

SPA525G - G722 Codec Negotiation

We are attempting to update phones for using the G722 codec. However, when calls are placed with the "Use remote pref" set to yes, the remote caller will go up to HD whereas the callee stays at g711u. The server is set for g711u, g729a, g722 for preferences ( so that all outbound calls are at a native g711u format ).
canreinvite is set to nonat for all extensions. I suspect that the remote end is seeing that the callee end is set to g722, g711u, g729a and using their preferrred which is g722, however, the callee does not switch over to use this as well.
Not sure if this is a problem with the phone negotiation or the server?
Ideally, we would have all outbound calls not show up as HD on the phone and all internal calls show up with HD. Just want to ensure that users are recieving the experience the phone is reporting.
Any suggestions?

Lets try this way:
Make a call from Agent1 to Agent 2, answer the call and see on phone what codec is being used between two IP Phones.If that works,Lets move on to ICM, enable Display to Screen on CTIOS Server in the registery with regedit from 0 to 1. Something like, My Computer\HEY_LOCAL_MACHINE\SOFTWARE\Cisco Systems, Inc\ICM\icm0\CTIOS1\EMS\CurrentVersion\Library\Processes\ctios
Restart the process ctios server and you should see the agents login in and logout. When the call is dropped, you will see another message on ctios server. Paste that message.
Also which version of CVP you are using? I recall in 8.0.1, there is something like flat G.729 Codec which means use of single CODEC on all call legs with no call legs using different codecs.
Hope this helps.
Cheers

Registering Polycom / Third Party Codecs to CUCM 10.x as SIP Device

Hello,
Has anyone tried registering Polycom or any Third party Codecs as SIP endpoints to CUCM 10.X.
I would like to know of any feature limitations when such registered devices are in conference calls with external parties as well as to the Cisco codecs registered to CUCM 10.x as Room based Endpoints.
Appreciate a helpful response.

It is possible, however I don't know the limitations. Here is a deployment guide from Polycom on how to register their endpoints to CUCM.
Polycom Unified Communications for Cisco Environments

CUCME 8.6 Call not forwarding Voicemail

Hi frieds,
     In our office we are using CUCME 8.6 on Cisco 2951 and unity express 8.5 in ISM module. As per our configuration whenever user is busy or not answering , the call will forward to voicemail. Totally we have 24 PSTN line. So we have an additional gateway 2901. The Issue I’m facing is that, when a PSTN incoming call coming through the second gateway(2901), if the extension is busy or not answering the call is disconnecting instead of forwarding to voicemail.
My 2951 configurations
voice service voip
ip address trusted list
ipv4 172.16.19.80
ipv4 172.16.19.81
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
bind control source-interface GigabitEthernet0/1
bind media source-interface GigabitEthernet0/1
registrar server.
Dial peer we are using for voice mail:
dial-peer voice 99 voip
destination-pattern 1099
session protocol sipv2
session target ipv4:172.16.19.81
dtmf-relay sip-notify
codec g711ulaw
no vad.
2901 Configurations
voice service voip
ip address trusted list
ipv4 172.16.19.80
ipv4 172.16.19.81
ipv4 172.16.19.82
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
dial-peer voice 99 voip
destination-pattern 1099
session protocol sipv2
session target ipv4:172.16.19.81
dtmf-relay sip-notify
codec g711ulaw
no vad
============================
Debug CCSIP Calls
Dec 15 15:23:23.448: //37497/8FD56BBAA9EA/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0xAF40FD8
State of The Call        : STATE_DEAD
TCP Sockets Used         : YES
Calling Number           : 5000
Called Number            : 1099
Source IP Address (Sig ): 172.16.19.80
Destn SIP Req Addr:Port :
Destn SIP Resp Addr:Port :
Destination Name         :
Dec 15 15:23:23.448: //37497/8FD56BBAA9EA/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : No Codec
Negotiated Codec Bytes   : 0
Nego. Codec payload      : 255 (tx), 255 (rx)
Negotiated Dtmf-relay    : 0
Dtmf-relay Payload       : 0 (tx), 0 (rx)
Source IP Address (Media): 172.16.19.80
Source IP Port    (Media): 25364
Destn IP Address (Media): -
Destn IP Port    (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0
Dec 15 15:23:23.448: //37497/8FD56BBAA9EA/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 47
Disconnect Cause (SIP)   : 200
For your reference I here attach a network diagram
What the command which I missed?

Check License status on your CUE, I had same issue.. Finally figured out its about license.. sh license status
Sent from Cisco Technical Support iPhone App

Trunking Cisco CUCM 9.0 with UCS 500

In one of our branch offices use Cisco UCS-500 for local IP telephony and PRI. Now we would like to integrate this system to our main offices so that we dial all the 4-digit extensions.
Is it possible to trunk Cisco UCS 500 with CUCM 9.0. ?

Hi
Seems you have codec negotiation problem when you are making calls from UCS to CUCM
. Can you post your UCS config here.
Maybe you have logging buffered enabled on your UCS as a result of which logs are not displayed on screen over telent session.
Regards
Aditya Gupta

Codec problem with one gateway

I am having a very strange problem with one of my gateways. No matter how I set up my regions and device pools I continue to have all calls go through this gateway as G729. This would be fine except that I can't do conference calls because of codec mismatches.
The gateway is a PRI running H323. CM 3.2.2(c) Gateway is a 3640 running 12.2.11(t)
All regions are currently set to G711.
I read something about a command under voice-port that sets the default codec. What I don't understand is, if this is what is causing it to go to G729, why is this the only gateway that ignores the regions.
Any help would be appreciated.

Cisco VoIP gateways support the codec negotiation feature. This feature provides the ability for a Cisco VoIP gateway to connect to other VoIP devices without necessarily knowing which codec is used for a call-setup. Also, this feature allows Cisco VoIP gateways to dynamically adjust to changes on remote devices. As long as the codec used by the remote VoIP device matches the capabilities-list of the Cisco VoIP gateway, the VoIP call is completed.
The following example shows how to configure codec negotiation:
Cisco-router# config t
Cisco-router(config)# voice class codec 1
!--- This sets up class 1 to be assigned to the dial peer.
Cisco-router(config-class)#codec preference 1 g723r63
Cisco-router(config-class)#codec preference 2 g729ar8
Cisco-router(config-class)#codec preference 3 g711ulaw
Cisco-router(config-class)#codec preference 4 g726r32 bytes 240
!--- These commands define the preferred codec list using 1,2,3, and 4 to set the preference.
Cisco-router(config)#dial-peer voice 1 voip
Cisco-router(config-dial-peer)#voice-class codec 1
!--- This assigns voice-class codec 1 to the dial-peer
Cisco-router(config-dial-peer)#destination-pattern 4723155
Cisco-router(config-dial-peer)#session target ipv4:192.168.100.1

Call/video not working between Cisco jabber for Windows and VCS control C40s

Hello,
I've been struggling with no luck how to make a call using Cisco Jabber for Windows 9.6.0 registered to CM 8.6.2 with intercluster ICT to another CM 8.6.2 where we have a VCS Control 7.0.2 via GK H225, and all C40s are registered as H.323.
The VCS has interworking between H323 and SIP, however not sure if there is any problem with that. Assuming it is ok, not sure either if I'm facing any interoperability issue because in my remote site I have C40 (H323 registered at VCS and SIP listening mode) and cisco jabber for windows which is SIP based.
If is not possible, would I be able to change my C40 from H323 to SIP at VCS, or have both H323/SIP registered at VCS? If so, will I need to change as well instead of GK I'll have to establish a SIP Trunk between the CM and VCS?
Another thing I do not believe either I would be able to have one VCS connected with two clusters, right?
I'm just trying to find a solution in case my current topology is not compatible, but feel free if you have any better idea to make it work.
Anyway here is what is happening:
When I make a call from my cisco jabber windows to C40 using alias number. The call is being redirected just fine to the C40 and it rings, however when someoene or the auto answer picks it up, the call dropped right away.
However, if I enabled the MTP in my CSF device, the call gets longer before dropping. I was even able to see my jabber " start video" turns green, before was grayed out all the time and the call dropped faster. I hear a fast busy tone.
I'm able to provide SDI traces, logs, diagnostic sip/h323 calls from VCS in order to know for sure if this is an incompatible issue or something I can workaround.
Let me know if someone of you are interested in read these logs or could point me on the right direction.
Thanks!

Ok,
I have looked at both logs. I have to mentinon though that you didnt
provide the log that shows the h323 setup between cucm and the VCS. This
is most likely because the call originated from a different cucm than
the ones you provided the logs from.
The call would have orginated from the first cucm in the cucm group of
this trunk: Name=RL_TRUNK_VIDEO
The cucm ip will be : 10.252.53.10.
This is the VCS log that confirms where the h323 request originated
from:
pr 10 22:50:29 TWELDVCS01 tvcs: UTCTime="2014-04-11 01:50:29,187"
Module="network.h323" Level="DEBUG": Src-ip="10.252.53.10" Src-
port="54000"
Received RAS PDU:
Having said that here is my analysis of the logs that you sent..
Jabber sent an INVITE to CUCM and advertised all the codecs (audio and
video it can support)..
Observer that Jabber says it doesnt support G729 anexB
21:55:16.576 |//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message
from 10.223.20.73 on port 54677 index 90661 with 2220 bytes:
[862370,NET]
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/TCP 10.223.20.73:54677;branch=z9hG4bK000029d3
From: "4122107" <sip:[email protected]>;tag=00059a3c78000011000070b0
-00000e65
To: <sip:[email protected]>
Call-ID: [email protected]
Max-Forwards: 70
Date: Fri, 11 Apr 2014 01:55:16 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CSF/9.4.1
m=audio 19252 RTP/AVP 0 8 18 105 104 101
c=IN IP4 10.223.20.73
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:105 G7221/16000
a=fmtp:105 bitrate=24000
a=rtpmap:104 G7221/16000
a=fmtp:104 bitrate=32000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 28878 RTP/AVP 97
c=IN IP4 10.223.20.73
++++Now lets observer the capabilites exchange during h245 negotiation
between cucm and VCS++++
Here CUCM advertises its caps to VCS (afterreceiving caps from VCS)
Note that G729A, G729AB, G729 is all advertised..
Apr 10 22:50:31 TWELDVCS01 tvcs: UTCTime="2014-04-11 01:50:31,017"
Module="network.h323" Level="DEBUG": Src-ip="10.252.53.10" Src-
port="45660"
Received H.245 PDU:
value MultimediaSystemControlMessage
::= request : terminalCapabilitySet
capabilityTableEntryNumber 2,
       capability receiveAudioCapability :
g729wAnnexB : 6
       capabilityTableEntryNumber 3,
   capability receiveAudioCapability : g729AnnexAwAnnexB : 6
       capabilityTableEntryNumber 4,
       capability
receiveAudioCapability : g729 : 6
capabilityTableEntryNumber 5,
       capability receiveAudioCapability :
g729AnnexA : 6
++++++
After doing MSD (master slave determination, we move to the OLC phas e..
Here we see that the far end..c40 wants to use G729AB for media++++
Apr 10 22:50:31 TWELDVCS01 tvcs: UTCTime="2014-04-11 01:50:31,783"
Module="network.h323" Level="DEBUG": Src-ip="10.224.114.11" Src-
port="11163"
Received H.245 PDU:
value MultimediaSystemControlMessage
::= request : openLogicalChannel :
   forwardLogicalChannelNumber 1,
forwardLogicalChannelParameters
     dataType audioData :
g729AnnexAwAnnexB : 20,
     multiplexParameters
h2250LogicalChannelParameters :
+++Next VCS sends G729AB as the codec to use to CUCM+++
Apr 10 22:50:31 TWELDVCS01 tvcs: UTCTime="2014-04-11 01:50:31,784"
Module="network.h323" Level="DEBUG": Dst-ip="10.252.53.10" Dst-
port="45660"
Sending H.245 PDU:
value MultimediaSystemControlMessage
::= request : openLogicalChannel :
   forwardLogicalChannelNumber 1,
forwardLogicalChannelParameters
     dataType audioData :
g729AnnexAwAnnexB : 20,
     multiplexParameters
h2250LogicalChannelParameters :
++++The next thing we get is an OLC reject from CUCM and this is where
th call drops++
Apr 10 22:50:31 TWELDVCS01 tvcs: UTCTime="2014-04-11 01:50:31,790"
Module="network.h323" Level="DEBUG": Src-ip="10.252.53.10" Src-
port="45660"
Received H.245 PDU:
value MultimediaSystemControlMessage
::= response : openLogicalChannelReject :
forwardLogicalChannelNumber 1,
   cause dataTypeNotSupported : NULL
Apr 10 22:50:31 TWELDVCS01 tvcs: UTCTime="2014-04-11 01:50:31,790"
Module="network.h323" Level="INFO": Dst-ip="10.224.114.11" Dst-
port="11163"
Detail="Sending H.245 OpenLogicalChannelRejResponse
+++We then receive a call release from cucm with cause code of 47:
resource unavailable++++
Apr 10 22:50:32 TWELDVCS01 tvcs: UTCTime="2014-04-11 01:50:32,365"
Module="network.h323" Level="DEBUG": Src-ip="10.252.53.10" Src-
port="50913"
Received H.225 PDU:
Q931
   Message Type: Release
Complete
   Call reference flag: Message sent from originating side
Call reference value: 0x7b
   Info Element : Cause
     Location: Usr
   Cause Value: Resource unavailable
   Info Element : User User
   Length = 22
Suggestions:
Change the region setting between the ICT trunk to VCS and Jabber to use
G711 and test again.

UCCX Call not working

Hi,
I have setup a HQ and Branch office. HQ is setup as Customer service location and Branch as Sales location. When PSTN calls Customer Service 800 number, the application runs fine. But when PSTN calls Sales 888 number, the call gets disconnected without any busy tone or anything.
I am using G711ulaw within sites and G.729 in between sites. UCCX and call manager servers are located in the HQ.
When running a reactive debug in Sales application, its triggers and stops at "Accept" step.
Any help is greatly appreciated.
thanks

Hi.
Which region and which codec is the PSTN router negotiating with CUCM?
Do you hear a fast busy tone on pstn phone when calling the application pilot?
On PSTN router:
conf t
logging buffered 4096
logging monitor debug
debug voip ccapi inout
term mon
and during a call from a pstn phone you should see the output of debug command.
Please post it here.
Thanks
Regards
Carlo

Need really urgent help from anyone

Hi ,
We are implementing sip in our business and we tested the dummy numbers on sip it is working fine.
When we test our main numbers it is not working for incoming or outgoing calls. it is giving sip error error 404.
I am using CUCM 8.6.2
Any ideas.
Oct 12 03:52:23.198: //418747/863B1AC6BE1B/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x2BCB7F60
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : 0861036261
Called Number            : 0893640881
Source IP Address (Sig ): 123.102.100.34
Destn SIP Req Addr:Port : 123.102.30.131:5060
Destn SIP Resp Addr:Port : 123.102.30.131:5060
Destination Name         : 123.102.30.131
Oct 12 03:52:23.198: //418747/863B1AC6BE1B/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : g711alaw
Negotiated Codec Bytes   : 160
Nego. Codec payload      : 8 (tx), 8 (rx)
Negotiated Dtmf-relay    : 6
Dtmf-relay Payload       : 101 (tx), 101 (rx)
Source IP Address (Media): 123.102.100.34
Source IP Port    (Media): 26776
Destn IP Address (Media): 58.105.248.129
Destn IP Port    (Media): 55094
Orig Destn IP Address:Port (Media): [ - ]:0
Oct 12 03:52:23.198: //418747/863B1AC6BE1B/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 1
Disconnect Cause (SIP)   : 404
Oct 12 03:52:27.038: //418717/84AF9822BDCD/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x2BC63A50
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : 0478403561
Called Number            : 0893640881
Source IP Address (Sig ): 123.102.100.34
Destn SIP Req Addr:Port : 123.102.30.131:5060
Destn SIP Resp Addr:Port : 123.102.30.131:5060
Destination Name         : 123.102.30.131
Oct 12 03:52:27.038: //418717/84AF9822BDCD/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : g711alaw
Negotiated Codec Bytes   : 160
Nego. Codec payload      : 8 (tx), 8 (rx)
Negotiated Dtmf-relay    : 6
Dtmf-relay Payload       : 101 (tx), 101 (rx)
Source IP Address (Media): 123.102.100.34
Source IP Port    (Media): 18150
Destn IP Address (Media): 58.105.248.129
Destn IP Port    (Media): 44382
Orig Destn IP Address:Port (Media): [ - ]:0
Oct 12 03:52:27.038: //418717/84AF9822BDCD/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 16
Disconnect Cause (SIP)   : 487
Oct 12 03:52:27.042: //418721/84AF9822BDCD/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x2BCBD950
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : 0478403561
Called Number            : 0893640881
Source IP Address (Sig ): 123.102.100.34
Destn SIP Req Addr:Port : 123.102.30.131:5060
Destn SIP Resp Addr:Port : 123.102.30.131:5060
Destination Name         : 123.102.30.131
Oct 12 03:52:27.042: //418721/84AF9822BDCD/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : No Codec
Negotiated Codec Bytes   : 0
Nego. Codec payload      : 255 (tx), 255 (rx)
Negotiated Dtmf-relay    : 0
Dtmf-relay Payload       : 0 (tx), 0 (rx)
Source IP Address (Media): 123.102.100.34
Source IP Port    (Media): 19128
Destn IP Address (Media): -
Destn IP Port    (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0
Oct 12 03:52:27.042: //418721/84AF9822BDCD/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 16
Disconnect Cause (SIP)   : 487
Oct 12 03:52:27.502: //418722/84F0E7CCBDDA/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x2BC8AFE0

Hi Rohit,
CUCM releasing the call because it's unable to find the called number received i.e. called number does not exist in dial-plan.
Check below points:-
1). Check the CSS on the SIP trunk in CUCM (may be called number is not accessible due to incorrect CSS).
2). Check the translation pattern and it's CSS.
3). If you are not using translation pattern then check for the translation profile in router config. In this case you can check and share your running-config and debug voice dial-peer or debug voice ccapi inout.
Regards,
Nishant Savalia

Cisco DX650 - SX20 compatibility

Hello,
I am testing a DX650 device registered on CUCM 7.1.5 and I have found a problem when this device calls to a SX20 videoconference device.
When I place a call from SX20 to DX650 it works fine but when I call from DX650 to SX20 I only get audio.
If I do the same test with all TANDBERG Edge95, always works fine.
Edge95 and SX20 are registered in the same VCS Control and this problem happen doing the call by H323 or by SIP.
Someone have had the same problem?
Thank you

Thanks for your answer,
The CUCM cluster is connected to VCS by SIP trunk.
I am using TMS 13.2
SX20 is not register en CUCM only is registered in VCS Control cluster as H.323 and SIP endpoint.
The codec negotiated in both cases is H.264
I observed that the call that is established without problems is established in encrypted mode even though the SX20 have encrytion set to off.
Regards

Calls getting dropped

Hi,
My scenario is like this:
IP Phone ---> Call Manager---->h323 gateway--->Avaya -----> Avaya phone.
I have created two device pools, DP-1 and DP-2 and assigned separate regions to these Device Pools. The codec between these regions has been configured as G.729. One hardware transcoder has also been created in the router and assigned to a MRGL which both DP's are using. IP Phone has been assigned to DP-1, and H.323 gateway has been assigned to DP-2. So the calls through this gateway to Avaya will be G.729 codec. Avaya has been configured to make the calls to the H323 gateway as G.729.
Whenever I am making a call to Avaya, I am able to get the ring from Avaya, when the receiver is picked up I am not able to hear anything. This is same when both side makes a call.
The IP reachability is there between all the devices. I have tried making the MTP required checked and unchecked under the gateway configuration, but the same result.
In the debug voip ccapi inout I am getting an error which is attached herewith.
The error code mentioned is 47 which is no resource. I have also checked by changing the codec values on both the ends.
Can anyone tell what could be the problem?
Thanks,
Manu

Disconnect Cause=47 is indicating issue is media negotiating like Transcoder, MTP & Codec negotiation.
Crosscheck the incoming & outgoing dial-peers for matching codec & voice class codec configurations. saw outbound dial-peer 40 matched but no logs to show matching inbound dial-peer.
CUCM has an option for "Wait for far end H245 TCS" on CUCM which is checked by default. Try to uncheck it and test the behaviour please.
Please collect detailed ccm traces & debug voice ccapi inout, debug h225 asn1 & debug h245 asn1.
what is the connection type from GW to PBX? PRI or IP connection? Please post your GW config also.
Please rate all the useful posts

Confused by basic SIP Trunk configuration.

I've went through a few basic SIP trunk configurations and Youtube videos the last couple days but can't figure out what I'm doing wrong.
I've set up H323 and MGCP no problem, but I can't figure out the SIP trunk set up. I'm guessing there are some concepts I'm not understanding yet.
I've got a CUCM lab set up. A 2851 PSTN Simulator, 2851 H323 Gateway at the Main site with a 9.0 CUCM setup in that site and a Branch site that I'm trying to set up as a SIP trunk to connect two phones.
CUCM is on the 192.168.5.x/24 subnet. 172.16.0.x/24 is the subnet connecting the serial(internet) cable between the two gateways in which I'm trying to establish the trunk between.
The Branch phones are still registering with the CUCM at the main site. The Route Pattern is looking to the Branch Route List which has the SIP Trunk listed. I'm just getting a fast busy when trying to place a call from the branch site to the main site.
The most frustrating thing I'm not understanding, is that the debug ccsip and call debugs on my SIP Branch gateway shows absolutely nothing. I've tried registering the branch phones with the SIP Trunk, but stopped when I figured that shouldn't be necessary.
If someone can make some sense of this, I'd truly appreciate it!

Hello Aditya and thanks for the consideration!
I do have a direct IP connection, but I want to set up a SIP trunk and use it just to know how to do it before I do it in production.
I did end up deleting the phones from CUCM so they can register with the 2851 CME that I'm setting up as a SIP trunk. So it is registering there, and I set the allow connections and bind sip commands.
I am now getting Debugs and calls from the SIP Trunk router going to CUCM, but the error message is No Codec, and I Get the fast busy after the call rings on the CUCM Main Site side. So looks like the negotiation is failing. Here is my CLI for the SIP Trunk now after the changes have been made and phones registered to the SIP Branch site as well as the Debug when I tried to place a call to extension "5000":
Note: I did try to change the codecs in the dial-peers to g729r8 instead of 711 and same fast busy after answering.
==============================================
Branch_SIP#show run
Building configuration...
Current configuration : 3529 bytes
! Last configuration change at 03:15:11 UTC Thu Apr 2 2015
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Branch_SIP
boot-start-marker
boot-end-marker
! card type command needed for slot/vwic-slot 0/2
enable secret 5 $1$hOXF$gvfmWW1ZIQE0mAMVg.u1c/
no aaa new-model
dot11 syslog
ip source-route
ip cef
ip dhcp excluded-address 10.0.10.1 10.0.10.10
ip dhcp excluded-address 10.0.30.1 10.0.30.10
ip dhcp pool Data
network 10.0.10.0 255.255.255.0
default-router 10.0.10.254
option 150 ip 192.168.5.250
dns-server 192.168.5.200
ip dhcp pool Voice
network 10.0.30.0 255.255.255.0
default-router 10.0.30.254
dns-server 192.168.5.200
option 150 ip 172.16.0.1
ip dhcp pool data
option 150 ip 172.16.0.2
no ipv6 cef
multilink bundle-name authenticated
voice service voip
allow-connections sip to sip
sip
bind media source-interface Loopback1
voice-card 0
crypto pki token default removal timeout 0
license udi pid CISCO2851 sn FTX1031A2FM
redundancy
interface Loopback1
ip address 2.2.2.2 255.255.255.255
interface GigabitEthernet0/0
no ip address
duplex auto
speed auto
interface GigabitEthernet0/0.10
encapsulation dot1Q 10
ip address 10.0.10.254 255.255.255.0
interface GigabitEthernet0/0.30
encapsulation dot1Q 30
ip address 10.0.30.254 255.255.255.0
interface GigabitEthernet0/1
no ip address
shutdown
duplex auto
speed auto
interface Serial0/3/0
no ip address
shutdown
clock rate 2000000
interface Serial0/3/1
ip address 172.16.0.1 255.255.255.0
clock rate 250000
interface Internal-Service-Module0/0
no ip address
shutdown
!Application: CUE Running on AIM2
hold-queue 512 out
router eigrp 1
network 0.0.0.0
network 2.2.2.2 0.0.0.0
network 10.0.0.0
network 10.0.10.0 0.0.0.255
network 10.0.30.0 0.0.0.255
network 172.16.0.0
ip forward-protocol nd
no ip http server
no ip http secure-server
ip route 0.0.0.0 0.0.0.0 172.16.0.2
tftp-server flash:term45.default.loads
tftp-server flash:jar45sccp.8-5-3TH1-6.sbn
tftp-server flash:cnu45.8-5-3TH1-6.sbn
tftp-server flash:apps45.8-5-3TH1-6.sbn
tftp-server flash:dsp45.8-5-3TH1-6.sbn
tftp-server flash:cvm45sccp.8-5-3TH1-6.sbn
control-plane
voice-port 0/0/0
voice-port 0/0/1
mgcp profile default
dial-peer voice 1 voip
description **Incoming Call from SIP Trunk**
session protocol sipv2
session target sip-server
codec g711ulaw
dial-peer voice 2 voip
description **Outgoing Call to SIP Trunk**
destination-pattern 5...
session protocol sipv2
session target sip-server
codec g711ulaw
sip-ua
sip-server ipv4:192.168.5.250
telephony-service
codec g711ulaw
max-ephones 24
max-dn 48
ip source-address 172.16.0.1 port 2000
system message SIP Branch Site
cnf-file location flash:
load 7960-7940 P00308010200.bin
max-conferences 8 gain -6
transfer-system full-consult
ephone-dn 1
number 4008
ephone-dn 2
number 4005
ephone 1
device-security-mode none
mac-address 001D.A21A.2065
button 1:1
line con 0
exec-timeout 0 0
line aux 0
line 194
no activation-character
no exec
transport preferred none
transport input all
transport output lat pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
speed 115200
line vty 0 4
password cisco
login
transport input all
line vty 5 15
password cisco
login
transport input all
scheduler allocate 20000 1000
end
Branch_SIP#show debug
TFTP:
TFTP Event debugging is on
CCSIP SPI: SIP Call Statistics tracing is enabled       (filter is OFF)
Branch_SIP#
*Apr 2 03:20:32.351: //25/0C496935804A/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x4B6C5C28
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : 4008
Called Number            : 5005
Source IP Address (Sig ): 172.16.0.1
Destn SIP Req Addr:Port : 192.168.5.250:5060
Destn SIP Resp Addr:Port : 192.168.5.250:5060
Destination Name         : 192.168.5.250
*Apr 2 03:20:32.351: //25/0C496935804A/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : No Codec
Negotiated Codec Bytes   : 0
Nego. Codec payload      : 255 (tx), 255 (rx)
Negotiated Dtmf-relay    : 0
Dtmf-relay Payload       : 0 (tx), 0 (rx)
Source IP Address (Media): 2.2.2.2
Source IP Port    (Media): 19472
Destn IP Address (Media): -
Destn IP Port    (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0
*Apr 2 03:20:32.351: //25/0C496935804A/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 63
Disconnect Cause (SIP)   : 503
Branch_SIP#

Codec negotiation in cucm !

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