Remote h.323 gateway drop conference calls

Recently I went to H.323 mode on a 1760 remote gateway. This was done to better support caller ID and SRST. This gateway services 6 POTS lines locally and has a T1 point to point to get back to the CCM cluster.
When I set up a MeetMe conference and then try connect a call on one of the POTS lines to the MeetMe, it drops the call.
The same thing happens when trying to directly conference a local extension and a caller on a POTS line.
I can however directly conference a local extension and a caller that can be reached out the T1 point to point line, which would be in the area codes local to the CCM cluster at headquarters.
I can also call in to headquartes and be transfered in to an existing MeetMe conference.
So, it seems that the issue is with the local POTS lins or the gateway. Any help would be greatly appreciated.
TIA - Mike

I'm not using a SW Conference Bridge that I am aware of. Just the default conference and MeetMe feature of CCM 4.1.
I'll look at the G711 in the dial-peers. I know that have it defined globally as the first prefernece. Also, I will look at regions and sees what I can see.
Thanks - Mike

Similar Messages

Dropped conference call issue!

Dear All;
I have cucm 8.5 and h323 GW (2911) with many FXO port , I have used a software conference bridge and I can make internal conference call only ;
my problem is: when I dial out side number via H323 GW the call is esablished, and when I press the conferece softkey button on IP Phone to start another call the call also established successfully , but when i press the conf button again to add the call to the conference , the call became dropped immediatly .
Please Advise me to solve this problem , does I need to create harware conference bridge also.
Thank you.
Rashed.

Dear javalenc;
thank you for replying, but how can i check if i did everything for g711 ?
woulld you please provide me steps to check.
thank you.

Help with setting up Caller-id on FXO in 2911 H.323 gateway.

I have a remote site that has a couple of POTS lines terminating to a FXO on the 2911. This remote site is an H.323 gateway in a CUCM 8.6 cluster. Incoming local calls for that location ring all phones at the location.
What do I need to do to enable Caller-id on these POTS lines terminating to the FXO? I am pretty sure the carrier is sending the caller ID information.
Thanks

Hi
1- Please find the below table as the following link http://www.cisco.com/en/US/products/hw/routers/ps274/products_tech_note09186a00800b53c7.shtml
Caller ID
Requires VIC-2FXO-M1, VIC-2FXO-M2, VIC-4FXO-M1, VIC2-2FXO, VIC2-4FXO, or MRP3-8FXOM1
2- under fxo port
voice-port 0/3/0
caller-id enable
3-If the above configure and still have no caller id , please add the below commannds to the voice-port
caller-id alerting line-reversal
cptone ?              "based on your"
caller-id alerting ring 2    "the default is 1" maximum number of rings to be detected before a call is answered over an FXO voice port.
4-Do debus to make sure all ok
"debug vpm signal "
[0/3/0] get_fxo_caller_id:Caller ID received. Message type=128 length=31 checksum=74
Thank you
please rate all useful information

Can Call Manager support network side Q.931 from H.323 gateways

With CCM 3.1, is it possible to support network side E1 Q.931 on a H.323 gateway. This would allow PABXs to connect to CCM, which then allow CCM to act as a tandem exchange.

The network side support on the H323 gateway
is transparent to the CCM. They are two seperate call legs.
1. pots for E1 PRI and
2. voip call leg to CCM.
The support for network side PRI is on 2600/3600/3700/7700/AS5300,AS5800
and its been available since 12.1(3)T and please use
a more recent image 12.2T train ip plus at the least.

Conference call dropped

We were able to use conference call for 20 senior management in the monthly meeting. In the last meeting, 8 conference calls dropped middle of meeting.
Nothing changes in existing configuration. I have checked in CUCM Server, System > Service Parameter > Call Manager > Clusterwide parameter>
Drop Ad Hoc Conferenece : Never
Advanced Ad Hoc Conference Enable : True
Great appreciated someone advice me above issue.

Does the call actually end when the conference drops or does the audio just stop (one way or no way audio)? If you have Call Detailed Records (CDR) enabled you can check the disconnect reason of the call. If you don't and you have detailed call manager traces you can take a look there to see why the calls dropped if they actually cleared the call.

Cannot make outside call (H.323 gateway and CUCM 6)

I cannot make outside calls. I am using H.323 gateway configuration and CUCM 6. I have attached configuration file and debug file. I configured H323 gateway and route pattern in CUCM. Please let me know if this is a configuration issue or telecom issue.

Hi 9tysixuae,
I can see that call is hitting the analog ports but from there on it generates error 34 which means circuit not available. You can try the following :
1. Plug Analog phone and verify if the circuit is fine ?
2. Try putting signal ground start on the voice port and see if that makes any difference.
Regards
Aditya Gupta

Conference calls not connecting/dropping

i am having issues using my conference calling feature on my iphone. The second and or third call gets dropped about 30 seconds into the call. Is this morelikely a ATT issue or an iphone issue?

I as well. 2nd caller get's hung up on when trying to conf to the 1st caller.
No options for releasing callers, or holding all but 1 for a private conf, then re-joining, etc.
My Droid does a way better job!
What gives Apple/Verizon - what/when's the fix?

Unable to make conference call + calls dropping with iPhone 6? :(

I have an iPhone 6 and i keep experiencing my phone calls randomly dropping out on me. I would sometimes hear a beeping noise and then a 'call failed' message would appear on the screen. Other times the call would just cut off and the screen would go back to the caller information or the home screen. Second problem is that i cannot create a conference call. Please help!

Hi ,
To make the conference call, click here for information.
The dropped calls, is this the one area?
Do you have any problems with texts?
Thanks.

Drop caller from conference call using Agent Desktop

Is there any way to drop a caller from a conference call using the Agent Desktop like we can do with the ConfList on the physical phone?

No there is not.

Call Forwarding / Displayed Number on Forwarding target with H.323 Gateway

Hi Community,
i´m wondering if there is sort of a simple way to get this working properly.
Scenario:
Germany, variable dial plan, no fixed NANP, and we have ClipNoScreening ;)
We use 0 for getting PSTN-dialing.
We have internal DNs, for example a 6789, 4 Digits. We use external phone number mask on our lines, 123456XXX
Our main number is 0123/456-xxx
When i call outside everything is displayed fine on the called target, +49 123/456789.
When i forward a call on my cellphone, with CFA target of 00111/222333444 (my cellphone example), and an internal colleague from within our office, is calling my office phone, everything is ALSO displayed fine.
Now here comes the BUT:
When someone calls from PSTN on my office-phone, i get displayed on my cellphone the +49 (0) 0xxxxxxxx, which means the caller number PLUS the added 0 from the gateway. Which is completely consequent and correct, since we add them on the gateway, when a call comes in, to be able to just answer directly on the office phone.
The rule on the H323 gateway:
voice translation-profile OUTGOING-VOIP
translate calling 1
translate called 2
voice translation-rule 1
rule 1 /^$.*$/ /0\1/ type subscriber unknown plan any unknown
rule 2 /^$.*$/ /00\1/ type national unknown plan any unknown
rule 3 /^$.*$/ /000\1/ type international unknown plan any unknown
voice translation-rule 2
rule 6 /4560$/ /6600/
rule 9 /^456$...$$/ /6\1/
voice translation-profile OUTGOING-POTS
translate calling 3
translate called 4
voice translation-rule 3
rule 1 /^00049/ /0/ type unknown national
rule 2 /^0/ // type unknown subscriber
rule 3 /^00/ /0/ type unknown national
rule 4 /^000/ /00/ type unknown international
voice translation-rule 4
rule 2 /^00049$.*$$/ /\1/ type unknown national
rule 3 /^000$.*$$/ /\1/ type unknown international
rule 4 /^00$.*$$/ /\1/ type unknown national
rule 5 /^0$.*$$/ /\1/ type unknown subscriber
dial-peer voice 10456 voip
translation-profile outgoing OUTGOING-VOIP
destination-pattern 456.T
progress_ind setup enable 3
modem passthrough nse codec g711ulaw
session target ipv4:<IP-OF-CUCM>
incoming called-number .
voice-class codec 1
voice-class h323 1
dtmf-relay h245-alphanumeric
fax-relay ecm disable
fax rate disable
fax protocol pass-through g711ulaw
no vad
no supplementary-service h225-notify cid-update
dial-peer voice 345000 pots
tone ringback alert-no-PI
translation-profile outgoing OUTGOING-POTS
destination-pattern 0.T
progress_ind alert enable 8
progress_ind progress enable 8
progress_ind connect enable 8
port 0/0/0:15
forward-digits all
In case of forwarding the external call to an external device, like for example a cellphone, this is crap.
Its obvious regarding the debugs all is working as designed ;), because my phone just forwards the full calling number including the added 0, since i put in to forward the originating calling DN.
My question now:
Can i simply correct this behavior somehow, also for international calls which would the 00 get added by the gateway?
Many thanks in advance for some input,
Andreas

Hi Community,
i´m wondering if there is sort of a simple way to get this working properly.
Scenario:
Germany, variable dial plan, no fixed NANP, and we have ClipNoScreening ;)
We use 0 for getting PSTN-dialing.
We have internal DNs, for example a 6789, 4 Digits. We use external phone number mask on our lines, 123456XXX
Our main number is 0123/456-xxx
When i call outside everything is displayed fine on the called target, +49 123/456789.
When i forward a call on my cellphone, with CFA target of 00111/222333444 (my cellphone example), and an internal colleague from within our office, is calling my office phone, everything is ALSO displayed fine.
Now here comes the BUT:
When someone calls from PSTN on my office-phone, i get displayed on my cellphone the +49 (0) 0xxxxxxxx, which means the caller number PLUS the added 0 from the gateway. Which is completely consequent and correct, since we add them on the gateway, when a call comes in, to be able to just answer directly on the office phone.
The rule on the H323 gateway:
voice translation-profile OUTGOING-VOIP
translate calling 1
translate called 2
voice translation-rule 1
rule 1 /^$.*$/ /0\1/ type subscriber unknown plan any unknown
rule 2 /^$.*$/ /00\1/ type national unknown plan any unknown
rule 3 /^$.*$/ /000\1/ type international unknown plan any unknown
voice translation-rule 2
rule 6 /4560$/ /6600/
rule 9 /^456$...$$/ /6\1/
voice translation-profile OUTGOING-POTS
translate calling 3
translate called 4
voice translation-rule 3
rule 1 /^00049/ /0/ type unknown national
rule 2 /^0/ // type unknown subscriber
rule 3 /^00/ /0/ type unknown national
rule 4 /^000/ /00/ type unknown international
voice translation-rule 4
rule 2 /^00049$.*$$/ /\1/ type unknown national
rule 3 /^000$.*$$/ /\1/ type unknown international
rule 4 /^00$.*$$/ /\1/ type unknown national
rule 5 /^0$.*$$/ /\1/ type unknown subscriber
dial-peer voice 10456 voip
translation-profile outgoing OUTGOING-VOIP
destination-pattern 456.T
progress_ind setup enable 3
modem passthrough nse codec g711ulaw
session target ipv4:<IP-OF-CUCM>
incoming called-number .
voice-class codec 1
voice-class h323 1
dtmf-relay h245-alphanumeric
fax-relay ecm disable
fax rate disable
fax protocol pass-through g711ulaw
no vad
no supplementary-service h225-notify cid-update
dial-peer voice 345000 pots
tone ringback alert-no-PI
translation-profile outgoing OUTGOING-POTS
destination-pattern 0.T
progress_ind alert enable 8
progress_ind progress enable 8
progress_ind connect enable 8
port 0/0/0:15
forward-digits all
In case of forwarding the external call to an external device, like for example a cellphone, this is crap.
Its obvious regarding the debugs all is working as designed ;), because my phone just forwards the full calling number including the added 0, since i put in to forward the originating calling DN.
My question now:
Can i simply correct this behavior somehow, also for international calls which would the 00 get added by the gateway?
Many thanks in advance for some input,
Andreas

Conference Calls dropping off and not getting re-merged

i'm having trouble all of a sudden with conference calling. any ideas?

Can you explain what the problem is and perhaps we can help?

MeetingPlace conference call on CCM MoH

Has anyone seen this, or have any ideas? Have a MeetingPlace Enterprise 5.4.0 system and now and then, people on hold on Call Manager (multicast MoH) can here meetingplace conference calls when they are on hold with the CCM MoH playing. The people on the MeetingPlace Conference call do not hear the person on hold or the CCM music.
The Call Manager and Meetingplace are completely seperate servers. The Call Manager version is 4.1(3) SR4D if it helps.
The network is clean, no errors, and can't find where this problem is coming from. It does not happen all the time and doesn't happen on every meetingplace conference call.
I don't know much about Meetingplace, does it use Multicast for its audio conferences? I thought they were kept local to the audio server unit (?).

For the sake of closing out this question... The answer was found deep in a Cisco wiki post.
Suppressing Music-On-Hold
You can configure Cisco Unified Communications Manager to play music for the held party when users put a call on hold. However, if a user places a meeting on hold, the music plays into the meeting until the user retrieves the held call. To prevent this, you can configure Cisco Unified Communications Manager not to play music on hold for calls routed through Cisco Unified MeetingPlace H.323/SIP IP Gateway.
Specifically, make sure you do not assign to the Cisco Unified MeetingPlace H.323/SIP IP Gateway a Media Resource Group List that includes any Media Resource Group that includes Music on Hold servers, as the MeetingPlace IP Gateway can join a multicast Music-On-Hold (MoH) group. Since MeetingPlace is a transmit/receive environment, being put on hold can mean meeting participants listen to MoH, as well meetings in progress to devices configured to participate in the multicast group, resulting unexpected and undesirable privacy breaches.
As an additional workaround/resolution, if the customer is using Cisco Unified Communications Manager (CUCM) Version 6.1 and later, verify the CUCM Service Parameter "Send Multicast MOH in H.245 OLC Message" is set to False.

Remote C2901 MGCP gateway won't register with CUCM

Hi, all,
We have a central call manager, we are setting up a remote office with a C2901 voice gateway that has a T1 PRI, this gateway is being configured as a H.323 gateway, everything worked fine. Now I want to get SRST working such that when WAN goes down, employees can still make outbound calls and take inbound calls. I followed SRST documentations when configuring CUCM and C2901, I would say the documentations are clear and easy to follow -- only thing is there is no adequate explanation of why those commands need to be there and I can not get SRST to work.
1), First, a basic concept question, why do we need to configure "ccm-manager fallback-mgcp"? or any MGCP related configuration on C2901? how does MGCP play the role in SRST? the router is configured as a simple voice gateway, when WAN is intact, we use H.323 protocol, when WAN goes down, we still don't need MGCP ...
2), Assuming that we needed all those SRST related MGCP configuration, I would expect C2901 to register to call manager, but I could not get this device to register as a MGCP gateway, debug mgcp packets shows:
*Sep 5 16:52:47.571: MGCP Packet sent to 10.128.181.5:2427--->
RSIP 201356931 *@voicegw-remoteoffice.local.domain MGCP 0.1
RM: restart
<---
*Sep 5 16:52:47.631: MGCP Packet received from 10.128.181.5:2427--->
500 201356931
<---
I verified the domain name is exactly the same as the name configured on CUCM, I tried multiple time toggling ip domain name on C2901 also, somehow CUCM does not like RSIP message from C2901, I don't have knowledge to get debug information from CUCM side of why CUCM does not like the MGCP message sent by C2901
3), When the communication is blocked from C2901 to central call manager (by using a simple ACL on C2901's default gateway device), I can still make the inbound call to an IP phone, which means, phones were correctly registered to C2901 when communication to call manager is broken, but I can not make outbound call, phones always get quick busy beeps, test call from C2901 itself (by using "csim start") is fine, which means dial-plan and ISDN/PRI is fine.
I believe we are just doing very very basic configuration and this is a common scenario, what could I do wrong?

You dont need the mgcp related commands if you are using H323. The commands that you need are:
application
global
service alternate default
call-manager-fallback (or) telephony-service and the relevant commands under it.
Also make sure that SRST reference has been added to the DP and the phones reset after the change made. This is what tell the phone to register to the SRST gateway when the WAN is down. The IP specified in the SRST reference should match whats configured under call-manager-fallback.

Conference calls all of a sudden do not work??

We have a Call Manager 5.1 with MGCP controlled gateways. Out of no where our ablility to ad hoc conference call does not work. We connect the calls and no one can hear one another. Any suggestions as to what could be the cause? Any feed back would be great.
Rgds,
Vicky

Here is the Troubleshooting and Debugging VoIP Call Basics.
http://www.cisco.com/en/US/tech/tk1077/technologies_tech_note09186a0080094045.shtml

External callers unable to join conference calls

Scenario: A conference call is set up using the scheduler. Lync users are able to join the conference calls. Users from outside PSTN are unable to join the call. They get to the attendant, and are told they are being joined to the
conference, then get a message "Sorry, I can't seem to connect you to your meeting right now. Goodbye." and the call disconnects.
Looking at call traces, I see the following error messages in the CAAServer log:
Component: CAAServer
Level: TL_ERROR
Flag: TF_DIAG
Function: CaaCall.EndTransfer
Source: caacall.cs(3631)
Local Time: 08/26/2014-18:28:36.161
Sequence# : 0001BE31
CorrelationId : 64624070
ThreadId : 0BA8
ProcessId : 0BB0
CpuId : 0
Original Log Entry :
TL_ERROR(TF_DIAG) [0]0BB0.0BA8::08/26/2014-22:28:36.161.0001be31 (CAAServer,CaaCall.EndTransfer:caacall.cs(3631))[64624070]ConferenceFailureException. ConferenceID=[91127] InstanceID=[b64b9467-db72-41bb-a2b5-673d76777ea4]
Component: CAAServer
Level: TL_ERROR
Flag: TF_DIAG
Function: CaaCall.HandleUnexpectedException
Source: caacallerrorhandling.cs(51)
Local Time: 08/26/2014-18:28:36.226
Sequence# : 0001D265
CorrelationId : 64624070
ThreadId : 0BA8
ProcessId : 0BB0
CpuId : 2
Original Log Entry :
TL_ERROR(TF_DIAG) [2]0BB0.0BA8::08/26/2014-22:28:36.226.0001d265 (CAAServer,CaaCall.HandleUnexpectedException:caacallerrorhandling.cs(51))[64624070][Enter] exception=[Microsoft.Rtc.Collaboration.ConferenceFailureException:The operation failed due to a response
from the server. For more information, examine the properties on the exception and inner exception.
   at Microsoft.Rtc.Signaling.SipAsyncResult`1.ThrowIfFailed()
   at Microsoft.Rtc.Signaling.Helper.EndAsyncOperation[T](Object owner, IAsyncResult result)
   at Microsoft.Rtc.Collaboration.McuSession.EndSendCommandInternal(IAsyncResult result)
   at Microsoft.Rtc.Collaboration.AudioVideo.AudioVideoMcuSession.EndTransfer(IAsyncResult result)
   at Microsoft.LiveServer.Caa.CaaCall.EndTransfer(IAsyncResult asyncResult, Boolean& retry, Exception& caught)
Detected at System.Environment.get_StackTrace()
   at Microsoft.Rtc.Collaboration.ConferenceFailureException..ctor(String message, Exception innerException)
   at Microsoft.Rtc.Collaboration.Conferencing.SendCommandAsyncResult.CreateConferenceFailureException(ConferenceCommandResponse commandResponse, Exception innerException)
   at Microsoft.Rtc.Collaboration.Conferencing.SendCommandAsyncResult.ProcessCccpResponse(SipMessageData messageData, responsetype response, Boolean& isPendingResponse)
   at Microsoft.Rtc.Collaboration.Conferencing.SendCommandAsyncResult.ProcessStatusMessage(SipMessageData statusMessageData, responsetype response)
   at Microsoft.Rtc.Collaboration.Conferencing.StatusMessageReceivedWorkItem.Process()
   at Microsoft.Rtc.Signaling.AsyncWorkitemQueue.ProcessItems()
   at Microsoft.Rtc.Signaling.SerializationQueue`1.ResumeProcessing()
   at Microsoft.Rtc.Signaling.SerializationQueue`1.ResumeProcessingCallback(Object state)
   at Microsoft.Rtc.Signaling.QueueWorkItemState.ExecuteWrappedMethod(WaitCallback method, Object state)
   at System.Threading.ExecutionContext.Run(ExecutionContext executionContext, ContextCallback callback, Object state)
   at System.Threading._ThreadPoolWaitCallback.PerformWaitCallbackInternal(_ThreadPoolWaitCallback tpWaitCallBack)
   at System.Threading._ThreadPoolWaitCallback.PerformWaitCallback(Object state)] m_conferenceID=[91127] m_conferenceUri=[sip:[email protected];gruu;opaque=app:conf:focus:id:4TPY99KF] InstanceID=[b64b9467-db72-41bb-a2b5-673d76777ea4]
Component: CAAServer
Level: TL_ERROR
Flag: TF_DIAG
Function: CaaCall.HandleUnexpectedException
Source: caacallerrorhandling.cs(51)
Local Time: 08/26/2014-18:28:36.226
Sequence# : 0001D265
CorrelationId : 64624070
ThreadId : 0BA8
ProcessId : 0BB0
CpuId : 2
Original Log Entry :
TL_ERROR(TF_DIAG) [2]0BB0.0BA8::08/26/2014-22:28:36.226.0001d265 (CAAServer,CaaCall.HandleUnexpectedException:caacallerrorhandling.cs(51))[64624070][Enter] exception=[Microsoft.Rtc.Collaboration.ConferenceFailureException:The operation failed due to a response
from the server. For more information, examine the properties on the exception and inner exception.
   at Microsoft.Rtc.Signaling.SipAsyncResult`1.ThrowIfFailed()
   at Microsoft.Rtc.Signaling.Helper.EndAsyncOperation[T](Object owner, IAsyncResult result)
   at Microsoft.Rtc.Collaboration.McuSession.EndSendCommandInternal(IAsyncResult result)
   at Microsoft.Rtc.Collaboration.AudioVideo.AudioVideoMcuSession.EndTransfer(IAsyncResult result)
   at Microsoft.LiveServer.Caa.CaaCall.EndTransfer(IAsyncResult asyncResult, Boolean& retry, Exception& caught)
Detected at System.Environment.get_StackTrace()
   at Microsoft.Rtc.Collaboration.ConferenceFailureException..ctor(String message, Exception innerException)
   at Microsoft.Rtc.Collaboration.Conferencing.SendCommandAsyncResult.CreateConferenceFailureException(ConferenceCommandResponse commandResponse, Exception innerException)
   at Microsoft.Rtc.Collaboration.Conferencing.SendCommandAsyncResult.ProcessCccpResponse(SipMessageData messageData, responsetype response, Boolean& isPendingResponse)
   at Microsoft.Rtc.Collaboration.Conferencing.SendCommandAsyncResult.ProcessStatusMessage(SipMessageData statusMessageData, responsetype response)
   at Microsoft.Rtc.Collaboration.Conferencing.StatusMessageReceivedWorkItem.Process()
   at Microsoft.Rtc.Signaling.AsyncWorkitemQueue.ProcessItems()
   at Microsoft.Rtc.Signaling.SerializationQueue`1.ResumeProcessing()
   at Microsoft.Rtc.Signaling.SerializationQueue`1.ResumeProcessingCallback(Object state)
   at Microsoft.Rtc.Signaling.QueueWorkItemState.ExecuteWrappedMethod(WaitCallback method, Object state)
   at System.Threading.ExecutionContext.Run(ExecutionContext executionContext, ContextCallback callback, Object state)
   at System.Threading._ThreadPoolWaitCallback.PerformWaitCallbackInternal(_ThreadPoolWaitCallback tpWaitCallBack)
   at System.Threading._ThreadPoolWaitCallback.PerformWaitCallback(Object state)] m_conferenceID=[91127] m_conferenceUri=[sip:[email protected];gruu;opaque=app:conf:focus:id:4TPY99KF] InstanceID=[b64b9467-db72-41bb-a2b5-673d76777ea4]
Component: CAAServer
Level: TL_ERROR
Flag: TF_DIAG
Function: CaaCall.TransferCompletedQueued
Source: caacall.cs(3739)
Local Time: 08/26/2014-18:28:36.231
Sequence# : 0001D269
CorrelationId : 64624070
ThreadId : 0BA8
ProcessId : 0BB0
CpuId : 2
Original Log Entry :
TL_ERROR(TF_DIAG) [2]0BB0.0BA8::08/26/2014-22:28:36.231.0001d269 (CAAServer,CaaCall.TransferCompletedQueued:caacall.cs(3739))[64624070]Call Transfer Failed. InstanceID=[b64b9467-db72-41bb-a2b5-673d76777ea4]
The unexpected error entry includes this information in the diagnosticUtils section:
Component: CAAServer
Level: TL_INFO
Flag: TF_DIAG
Function: DiagnosticUtils.TryCreateSignalingHeader
Source: diagnosticutils.cs(269)
Local Time: 08/26/2014-18:28:36.231
Sequence# : 0001D268
CorrelationId :
ThreadId : 0BA8
ProcessId : 0BB0
CpuId : 2
Original Log Entry :
TL_INFO(TF_DIAG) [2]0BB0.0BA8::08/26/2014-22:28:36.231.0001d268 (CAAServer,DiagnosticUtils.TryCreateSignalingHeader:diagnosticutils.cs(269))Creating SignalingHeader for InstanceID=[b64b9467-db72-41bb-a2b5-673d76777ea4], header-name=[ms-diagnostics], header-value=[10010;Reason="Gateway
side Media negotiation failed";Source="LYNC-FE.itprocare.com";component="MediationServer";sipresponsetext="Invite with Replaces failed because Gateway side reinvite failed.";DialogID="f94f35b9-7f5d-4bcd-94a8-de0fb2d8c4c6;df1bec7e24;24010080"]
Which matches up with an error in the SIP trace:
TL_INFO(TF_PROTOCOL) [0]0DEC.0A48::08/26/2014-22:28:36.151.0001ac66 (SIPStack,SIPAdminLog::TraceProtocolRecord:SIPAdminLog.cpp(125))$$begin_record
Trace-Correlation-Id: 3329945913
Instance-Id: 00000688
Direction: outgoing
Peer: lync-fe.itprocare.com:51307
Message-Type: response
Start-Line: SIP/2.0 491 Invite with Replaces failed because Gateway side reinvite failed.
From: <sip:[email protected];gruu;opaque=app:conf:audio-video:id:4TPY99KF>;tag=26f350618b;epid=6DF0663499
To: <sip:[email protected];gruu;opaque=srvr:MediationServer:BMFt1DuKo1KYsGtnqeobCwAA;grid=9d179e4c3aeb4fd7aedd36d7853ad98b>;epid=26F55811B8;tag=2e6beef249
CSeq: 5 INVITE
Call-ID: af099053-d8aa-4ca4-9820-936e8522611c
Via: SIP/2.0/TLS 10.160.1.47:51307;branch=z9hG4bKeb2e7ed6;ms-received-port=51307;ms-received-cid=9B00
CONTENT-LENGTH: 0
P-ASSERTED-IDENTITY: <sip:5853307343;[email protected];user=phone>
SERVER: RTCC/4.0.0.0 MediationServer
ms-diagnostics: 10010;source="LYNC-FE.itprocare.com";reason="Gateway side Media negotiation failed";component="MediationServer";SipResponseText="Invite with Replaces failed because Gateway side reinvite failed."
ms-diagnostics-public: 10010;reason="Gateway side Media negotiation failed";component="MediationServer";SipResponseText="Invite with Replaces failed because Gateway side reinvite failed."
ms-endpoint-location-data: NetworkScope;ms-media-location-type=intranet
Message-Body: –
$$end_record
The trunks are configured with REFER off and MediaBypass Off.
We have recently moved to a direct SIP trunk from a vendor on Microsoft's Certified list.
Lync version is 2010, with the latest CUs applied. All other calling appears to be working correctly. The certificate on the servers are using the SHA1 algorithm (I have seen some similar issues discussed if this was not the case.)
At this point, I have reached the end of my immediate troubleshooting skills with this system. Can anyone offer any suggestions as to what might be going on here?
Thanks for any help.
-Tim

Hi,
Please check if the default gateway associated to the Mediation Server is up or not.
Please check if Media traffic on the Gateway be blocked with the issue of wrong encryption. Make sure it is using SRTP (not RTP). If you use RTP, please change it and then test again.
Best Regards,
Eason Huang
Eason Huang
TechNet Community Support

Remote h.323 gateway drop conference calls

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