Third party IP phone services

Any recommendation for third party IP phone services?
I have found two: www.andtek.com and www.berbee.com.
Thanks!

http://web.csma.biz/apps/xml_xmldir.php is another place. It's a phone directory system released under GPL.
It's pretty good, but you'll want to install it in the root of a website and not under a subdirectory. I had issues with the xml output not sending an xml header and had to edit the source to fix this up.

Similar Messages

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.

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

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?

Third Party IP Phone

HII,
I am working with CCM 5.1 as a service Provider. One of my customer have a third party 3Com 2102. I can't register his phone in my register.I have some configured already that i attached with mail. plz. help me ..and tell me how to configure that phone in my CCM.
Plz as soon as possible give me the suggestion.
thanks & Regards
Madhab

hii all,
I forgot to attached the configuration file which I mentioned in my previous mail.
that's why I am postin this mail with that attachment.
thanks and regards
Madhab

Third party scenario with services

Hi
Is it possible to create PR for services thru Sales order for Third party scenario.
Please suggest.
Thanks

Hi,
as the system creates an MM document from sales order with item category 5 ('S' external -Third-party) for using of materials, you can not enter services into such purchasing items. For handling external services the item categories 9 ('D'), and 1 (B) are used. There is no item category combination for both - join item categories 5 and 9 - available in the standard. Please create a PR in ME51N instead with the concerned order in the accounting data.
Regards,
Edit

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....

Integrate third-party-tool with service desk (SOAMANAGER)

Hello,
to activate the web service for interface between service desk and third-party-tool I have to use transaction SOAMANAGER. My problem is that I can't find any documentation about this transaction belonging to the mentioned interface.
Has anybody experience how to activate and test the web service with SOAMANAGER?
Best regards
Klaus

Hi Santosh Asuthkar,
thank you for your answer.
I've already read the steps in trx. spro and several documentations in sdn.
There was a hint to trx. ictconf, wsconfig, lpconfig, wsadmin. If you call these transactions there
is a message that these trx. are old and trx. SOAMANAGER should be used.
Question is how to activate the webservice "ICT_SERVICE_DESK_API" in SOAMANAGER.
I've never handled with bindings and proxy classes.
Best regards
Klaus

Third party IP phone services

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