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

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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)|*^*^*

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).

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.

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.

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.

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

CM5 support the third party SIP phone?

Does CCM5 support the third party SIP phone? what manufacturer? all manufacturer?
Does IP phone support the third party SIP server? what manufacturer? all manufacturer?

thanks for ananddiwakar.
I agree your the first answer: CCM5.0 support the third party IP phone, but be some limitations with the features.
But for cisco sip phone supporting standard SIP server, I'm confused that because the cisco SIP phone need download Firmware from SIP server, and for the third party SIP server, is the mechanism of downloading the Firmware from the third party SIP server right?

CUCM Third Party SIP Phone "Time" issue

Hi Team,
we have setup with Avaya 1230 SIP Phone,
and this phone we added to CUCM using "Third Party Basic SIP Phone" option.
Once registered with Call Manager "Date and time" in SIP Phones was showing fine.
we have reset the entire device pool, after that all the Avaya 1230 SIP Phone "Time" is showing +1 hour from the normal time.
How we can reslove this issue.
CUCM Version: 9.1(2a).
SIP Phone Model: Avaya 1230

Thanks for the Suggestion Manish,
I have tried the same But its not working,
In the phone level we have the option to change the "Time Zone", The same we have changed to GMT+5:30 Indian Standard Time.
Any other suggestion....

Third Party SIP Phone Config

Good afternoon all,
I have a client that I am trying to program a Gai-Tronic Titan Phone on thier Cisco phone system. The system has the following:
Cisco 2851 Revision 53.51
IOS Version 12.4(24)T2
CME Version 7.1
Unity Express 7.0.3
Anything I do in CME is by CLI.
After some trial and error, I was able to get the phone to show up in Unity Express. When picking up the reciever, we get nothing but a fast busy signal. I am sure I am missing something somewhere.
Is there anyone out there that has setup one of these phones that might be able to help out?
Here is the programming I have so far for the Titan Phone:
ephone-dn 37 dual-line
number 700 secondary 1234567890
label Titan Phone
description Titan Phone
name Titan
call-forward busy 1000
call-forward noan 1000 timeout 15
hold-alert 30 originator
ephone 70
mac-address 0017.AE01.02AB
username "titan" password ********
Again, any help would be appreciated.
Thanks
Mike

I ended up programming the system based off of your above post. Here is what I added to the conifg....
voice register global
mode cme
source-address 10.170.130.250 port 5060
max-dn 10
max-pool 10
hold-alert
voice register dn 1
number 700
name titan password *******
no-reg
label Titan
voice register pool 1
id mac 0017.AE01.02AB
type 7912
number 1 dn 1
dtmf-relay sip-notify
codec g711ulaw
The problem is, when I do a sh voice register pool 1, it doesn't show any phones registered. Am I missing something? I am not currently on-site. I am going to have someone at the facility check the phone tomorrow morning first thing to see if it's working. If there is something in the code I am missing, please let me know.
Thanks
Mike

3rd Party SIP phone to CUCM via SIP Proxy

Hi all,
This is the scenario i'm currently working on :
3rd party SIP phone <--> Internet <--> SIP Proxy <--> LAN <--> CUCM
The SIP proxy basically terminates everything (REGISTER, INVITE, etc), including the RTP stream.
I can register the 3rd party SIP phone to CUCM and in CUCM and i can see SIP Proxy IP Address as the registered address of the phone.
Calls from the 3rd party SIP phone to internal Cisco or internal 3rd party SIP phone and vice versa work like charm.
The only (fatal) problem is i can only register 1 3rd party SIP phone to CUCM via this SIP proxy.
Since this SIP Proxy always use its internal IP Address and port 5060 (TCP) as its source of registration, CUCM sees multiple registrations for multiple extensions (users) come from a single IP and port, and rejects the second registration request.
It seems that CUCM binds a digest user to an IP address and port, therefore cannot accept multiple registrations from a single IP and port.
Can anyone clarify this? Or is there any way around this?
I'm using CUCM 8.6.2 and CUCM 9.X (both do not work).
Regards,
Christian

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="xxxxxxxxxx",realm="ccmsipline",nonce="GBauADss2qoWr6k9y3hGGVDAqnLfoLk5",uri="sip:172.18.197.224",algorithm=MD5,response="126c0643a4923359ab59d4f53494552e"
The username, xxxxxxxxxxx, 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
Also Try this bug CSCef88775

Third Party SIP & MeetMe numbers

Hello,
Was just wondering if anyone has an elegant way of getting a Third Party SIP phone to initiate a MeetMe?

Well that's not entirely accurate; a Cisco SIP phone can initiate a MeetMe conference.
If you look in the SIP Profile on UCM, I believe there is a setting for this. I've never done it but you'd need to configure the SIP phone to send the xcisco-??? message in the invite probably. Hopefully that'll move you in the right direction at least.

Third party SIP Endpoints in BE6K

Hi
I have a doubt about BE6K, Is this solutions support Third part SIP Endpoints, like grandstream or polycom? And if is true, What license support thjis pohone?
Best regards

Third Party ATA is another story... But a third Party IP phone will work fine.
I have a Polycom conference register with our BE6K.
Just need to find out witch phone model config to push down the GrandStream or the Polycom.
Good Luck

Add third party SIP Phone to CCM 5

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