Third-party SIP Device (Basic) video calling

Hi,
I want video calling with third-part SIP device to register CUCM 8.6. But my sip phones rejecting video calls. (there is no problem, audio calling...)
I'm doing the same process with elastix ip pbx smoothly.
Help me !!

ip phone codec h264 , h263

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RTP Media Stream with "Third Party SIP Device" always through CUCM

Hello,
i have i quite strage problem on one of my customers locations:
we have a cucm 7.1.5(SU4) with the cucm in the datacenter. And we have a small location(branch office) which has a small wan connection to the datacenter (1MBit/s).
In this location we have several Kirk (Polycom) Dect phones which register as "Third Party SIP Device - Basic" on the CUCM.
(The problem is the same if i use the X-Lite SIP Client instead)
When this SIP Phones or the X-Lite Client dials a internal Number of the same location the RTP Media Streams goes directly from the SIP Client to the phone. But if they dial an external number the RTP Stream goes from the SIP Client via the wan connection to the CUCM and back via the wan connection to the 2901 H.323 Gateway (on the same location).
and of course if i now start a big download or upload i'm no longe able to complete the phone call because we have no QOS on the wan connection, because we don't want to make calls over this connection.
When i look at the Sniffer files with Wireshark i see that the CUCM sends his own ip adresse in the SDP Header for the RTP Stream to the SIP Client. And this of course is wrong because the RTP Stream should always reside on the branch office.
i tested this in my lab (CUCM 8.5) and it is the same. i used the "Standard SIP Profile" and a Basic Third Party SIP Device"
The SCCP Phones on the same location which are configured with the same Region, Location, Device Pools, Media Resources and which use the same Gateway for external calls do not have this problem.
In the Gateway configuration "MTP Required" is not activated and i tested it in my lab with some Cisco SIP Phones (9971) and they are also not affacted with this problem.
any ideas?

do you have a SIP trunk to the external devices with MTP required checked ?

Activate a call forward with a Third-party SIP Device or with a analog device

Hi,
In a CUCMv9, how i can activate a call forward (all, busy, no anwser...) with Third-party SIP Device or with a analog device connected to a fxs?
I want to activate a call forward like a Alcatel or Aastra PBX with a code.
For exemple, i pick up the phone, with the code *95 followed by the destination number and hangs up the phone. And use the #95 for désactivate this call forward.
It's possible?
Thanks.

No codes for 3rd party SIP phones, no way to do it. Or for that matter, not even for Cisco Phones, other than CFA.
Anything besides CFA needs to be done via CCMadmin or CCMuser for any kind of phone.
For FXS that's only doable if you're running SCCP
http://www.cisco.com/en/US/partner/docs/ios/voice/fxs/configuration/guide/fxssccpsplmft.html
HTH
java
if this helps, please rate
www.cisco.com/go/pdihelpdesk

CUOM with Third-party SIP Device

Hi everyone,
I have installed CUOM in environment with CUCM, Cisco IPPhones (SCCP), and IPPhones (Third-party SIP Device), I can monitor CUCM and SCCP IPPhones but the Third-party SIP Device can't be monitored with CUOM, Please someone have a solution for this issue
Thanks in advance,

do you have a SIP trunk to the external devices with MTP required checked ?

Video call from Cisco 8945 SCCP to Polycom VVX 1500 Third party Sip Phones between intercluster trunks is not working

Dear All,
We have two cucm Clusters in Different Locations between that clusters i created
Inter-Cluster Trunk (Non-Gatekeeper Controlled) Now all are working fine Between Clusters
audio calls & Video calls between sccp 8945 phones , but iam facing a Problem with third party
Video Phones (Polycom VVX 1500 ) Third Party SIP Phones located in second cluster, From 1 st cluster cisco 8945 Video
phone to 2nd cluster Polycom Video phone all calls are works for voice call only, but no video ,
Please Suggest me Solution.
Thank you,
Sriman

Try setting up a SIP trunk between the two clusters and set a route patten just to the VVX 1500 and check how that goes.
From memory inter-cluster trunks are a H.323 like protocol which might have video inter-op issues with the Polycom device.

Incoming calls issue in Third Party SIP Phone

Hi,
Yesterday I configured my third party sip phone which is yealink in this case on cucm and successfully registered it with cucm, despite of registration i have some calling issue in this phone. I am able to make outbound calls from this phone to any other phone however issue is related to inbound calls.I tried calling its DN from anywhere but call disconnect after sometime. Also didnt get any proper sip session trace in RTMT. Kindly suggest some step to sortout this issue.
Thanks

Dear Manish,
Call normally dicsonnected after 30-40 sec with termination code 102 in session trace. PFB SDI trace with 5030 is Thirdparty sip phone and 5033 is c7945. Looking forward for your suggestion.
CallingPartyNumber=5033
|DialingPartition=
|DialingPattern=5030
|FullyQualifiedCalledPartyNumber=5030
|DialingPatternRegularExpression=(5030)
|DialingWhere=
|PatternType=Enterprise
|PotentialMatches=NoPotentialMatchesExist
|DialingSdlProcessId=(0,0,0)
|PretransformDigitString=5030
|PretransformTagsList=SUBSCRIBER
|PretransformPositionalMatchList=5030
|CollectedDigits=5030
|UnconsumedDigits=
|TagsList=SUBSCRIBER
|PositionalMatchList=5030
|VoiceMailbox=
|VoiceMailCallingSearchSpace=PT-LHR-LOCAL:PT-Local:Unityvmpt:PT-F6-Local:PT-ISL-LOCAL:PT-KHI-LOCAL:PT_Operator_LHR:PT_Operator_KHI:PT_Operator_ISL
|VoiceMailPilotNumber=7103
|RouteBlockFlag=RouteThisPattern
|RouteBlockCause=0
|AlertingName=Syed Ahmer
|UnicodeDisplayName=Syed Ahmer
|DisplayNameLocale=1
|OverlapSendingFlagEnabled=0
12:17:38.028 |//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to 172.16.200.21:[5062]:
[23928282,NET]
INVITE sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/UDP 10.100.200.11:5060;branch=z9hG4bK1ca0cc6e317649
From: "Syed Ahmer" ;tag=8787406~039e2a80-8561-4586-8954-d01ed2aa12c8-246211918
To:
Date: Thu, 30 Jan 2014 07:17:38 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Send-Info: conference, x-cisco-conference
Alert-Info:
Contact:
Remote-Party-ID: "Syed Ahmer" ;party=calling;screen=yes;privacy=off
Max-Forwards: 70
Content-Length: 0
|14,100,50,1.14103336^10.163.14.4^SEP00230432C828
12:17:38.028 |EnvProcessUdpPort - EnvProcessUdpHandler::fireSignal() varId = 0|14,100,50,1.14103336^10.163.14.4^SEP00230432C828
12:17:38.028 |EnvProcessUdpHandler::fireSignal - SEND: index = 0, handler = 0xaf299320|*^*^*
12:17:38.028 |EnvProcessUdpPort::fireSignal - SEND, destination = 172.16.200.21:5062|*^*^*
12:17:38.028 |EnvProcessUdpPort - EnvProcessUdpHandler::send(buff, 850, 172.16.200.21:5062)|*^*^*

CUCM: Third Party SIP Phone "Caller ID" is not displaying for outgoing calls

Hi Team,
we are running CUCM 9.1(2a),
we have integrated Third Party SIP Phone(Avaya 1230 SIP Phone) with CUCM,
Issue: Third Party SIP Phone "Caller ID" is not displaying for outgoing calls, we are able to see only the dailed Number,
When "A" calls to "B", "A" can see only the dailed number of "B" but not the "Caller ID"
Regards
Ananthakumar

Are A and B both Avaya phones?
So it looks like you're not seeing the alerting name/connected name getting updated then? Do you have alerting names configured on the directory numbers? Might need to take a look at the SIP messaging to see if the alerting name/connected name is being sent to the Avaya phones and maybe they just aren't displaying it. Might just be something that needs to be tweaked in the 46xxsettings.txt file.

Paging Third Party SIP Phones connected to CUCM

Current SetUp: CUCM - Cisco 3925, Two MCS 7816 (Call Control Server) and One MCS 7825 (Voice mail server)
We have third party SIP phones configured in auto answer mode. These phones are used to make live announcements.
To Do:
There are approximately 80 phones in the system and the requirement is to select any combination of these phones to make Public announcement (or Paging).
Is there an application that enables us to select any combination of phones on the fly to do paging? How can we select a mp3 file to play on a phone in an auto answer mode?
Any help will be appreciated.
Thanks
Sid

There's nothing built into call manager to do this. You could investigate using the Cisco Unified Application Environment (CUAE) and write a script to do this, or there are some 3rd party applications that might work for you such as Berbee's Informacast.

Add third party SIP Phone to CCM 5

'm not able to register this SIP Phone to the CCM5.0. I have device license that cater all IP Phone models.(LIC-CM-DL-100=)
I got error message " Login Forbidden" "timeout" in the IP Phone.
In the CCM, I got this message in Phone COnfig Window
Registration: Rejected.
Can you explain on how to register this 3rd party IP phone to CCM?
Is it CCM able to support SIP Phone?

Hi,
This is most likely because of the following...
Because third-party SIP phones do not send a MAC address, they must identify themselves by using digest authentication.
The REGISTER message includes the following header:
Authorization: Digest username="swhite",realm="ccmsipline",nonce="GBauADss2qoWr6k9y3hGGVDAqnLfoLk5",uri="sip:172.18.197.224",algorithm=MD5,response="126c0643a4923359ab59d4f53494552e"
The username, swhite, must match an end user that is configured in the End User Configuration window of Cisco Unified CallManager Administration. The administrator configures the SIP third-party phone with the user; for example, swhite, in the Digest User field of Phone Configuration window.
See the following document.
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/5_1_3/ccmcfg/b09sip3p.html
Hope this helps, if so please rate.
Regards,
Dave

Third Party SIP Phone Alerting name

Hello,
We are having cisco ip phones & third party sip phones in our company. both are registered to same cucm 9.X.
Now when cisco phones calls sip Phones. we are able to see the alerting name on cisco phone.
Say, Cisco Phone "Phone-A" calls Third Party Sip phone "Phone-B". So, on cisco ip Phone display we are able to see "Phone-B".
But same when we try from SIP phones to Cisco phones, we are unable to see the Alert name on the Sip Phone. only the Number we can see on the Sip phone.
Any help will be highly appriciated.
Thanks,

Hi Amod.
I'm terribly sorry cause I got what you are asking only now :(
This behaviour depends on client capability.
CUCM always send RPID ( remote-party ID) to the third party SIP client/phone on ring.
If the client is capable to update the calling number into what it receives as RPID, than you'll be able to see calling name.
I've read some release notes of XLITE and other SIP Desk phone and it seems to be not mentioned.
Sorry again
Regards
Carlo

MOH for third party sip phones

Hello ,
I using CUCM version 9.1.1.2000-5 .
Does this version support MOH for third party sip phones ?
Thank you

Hi,
I couldn't found better piece of information which list all cases how MoH is implemented for various terminals and hence thought of testing the same. My observations;
SCCP phone -> Hold -> SIP Phone -> MoH plays to SIP phone
SIP Phone -> Hold -> SCCP phone - > Doesn't play MoH
SIP Phone -> Hold -> SIP phone -> Doesn't play MoH
Please note that I have checked with both Xlite and 3CX, results are same.
I have verified in wireshark also, call manager is not sending RTP packets to held party when call is hold by third party SIP phone.
Checked in CUCM 9.1
Thanks
Vivek

Are CMBE 3000 support third party SIP phones?

Hi everybody!
Are CMBE 3000 support third party SIP phones?

Hi Yang,
The typical IP phone today costs the same or less than an equivalent digital desk phone set. When you factor in the lower overall total cost of ownership (TCO) that results from an IP Communications solution running on a converged IP network for voice, video, and data, an IP based solution can save a company a substantial amount of money in the medium to long term.
Thanks

ABAP source code to connect to third party systems using web service calls?

Hi all,
can any one provide an example ABAP source code to connect to third party systems using web service calls? The base system is CRM.

Do you want to call a web service in a remote system, or do you want to provide a web service?
If you want to call a web service you should create a proxy object via SE80. Open your development package, right click on the tree entry and choose: Create -> Enterprise Service / Web Service -> Proxy Object and provide the needed information (including the WSDL description file). You may then use the proxy object to call the web service (if the connection and everything else works right).
See [http://help.sap.com/saphelp_nw04/helpdata/en/9b/dad1ae3908ee44a5caf57e10918be9/content.htm|http://help.sap.com/saphelp_nw04/helpdata/en/9b/dad1ae3908ee44a5caf57e10918be9/content.htm]

Third party time device

Hi to all,
Can any one share the configuration of how to connect the SAP HR with the Third party time device of swipe in and swipe out of an employee.
You can mail the document to my id of shreyasen1980 at the rate of gmail dot com
Thanks,
Shreyasen

Hi,
There are many third party devices that can be integrated with SAP for capturing time events in IT2011. Timelink is one of them and it is SAP certified partner. Following are the high level steps to be followed considering SAP integration with Timelink.
Setup and assign timelink user with proper profiles
Setup RFC destination for the timelink
Create the logical system for timelink
Define ports for IDOC processing
Define the distribution model for timelink for exchanging inbound and outbound messages
Define the partner profiles
Verify the IDOC list and the communication manually between Timelink and SAP
Best Regards,
Sunny

MOH in third party sip phones

Hello ,
I would like to know if the CUCM support MOH in third party sip phones such as x lite or other ?
Now I can only hear silent .
Thanx

Hi ben Zecharia,
I found your post looking for MoH in 3rd Party SIP Phone and also found another post that said that CUCM 8.x do not support MoH in 3rd Party SIP Phone (check this link).
Hope this helps (you and others).

Third-party SIP Device (Basic) video calling

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