How to configure a SIP trunk between Cisco Call Manager 5.x or 6.x or 7.x and OCS 2007 R1 or R2
Posted by Mino on February 20, 2009
How to configure a SIP trunk between Cisco Call Manager 5.x or 6.x and OCS 2007 R1 or R2
Ok you want to ring from MOC to Cisco IP phone and back , hmmm ok then simple we will deal with it as if OCS is an IP PBX with its extensions 3xxx and you need to connect it with Cisco PBX with extensions 7xxx.
To do that we need a SIP trunk and for the SIP trunk to work fine we need to have some specific configuration on that trunk , remember any of these settings if they not configured right then you will not be able to make a stable calling between Cisco and OCS.
First we do the SIP trunk :
Now the SIP trunk which is acting like the bridge between the cisco and the OCS is created , ok then we need now to create a criteria where this trunk is going to be used in. This is where is Pattern comes in where we will say if a Cisco phone set tries to dial extensions starting with 3xxx then you use the trunk which we have just created .
and from the way back from the OCS to Cisco , when the number is sent in the E164 formate with the + , the Cisco will simply ignore all that and will take only the last 4 Digits which are the 7xxx
Now you are ready to make the call and Enjoy the Integration
J
Mino - The UC Guy 
Doug Darrah said
We have MOC outbound dialing through the direct SIP (“SIP trunk”) connection to CUCM working, but have found that DTMF isn’t working (character strings entered into MOC are seemingly ignored.) Any ideas? (In my particular case, my DID is NOT hosted on the CUCM that is processing the MOC call….)
Otherwise, this method works fairly well.
Mino said
Yeah , i have been through this DTMF problem .
try to look for some configurations to disabled RFC2833 DTMF tones on the Cisco.
Ali said
please inform me how can i config Trunk between Tenor ASM400 Quintum vs CUCM 7.7 ? please help me . this is very important for me
redvoz77 said
How would you configure multiple Cisco clusters to the OCS R2 (when trying to setup the Dial in Audio Conference piece of ocs)? Can you use a Gatekeeper and if so would you use h.323..to the mediation server and the SIP to the OCS server????HELP.
Mino said
can you please tell me more about your scenario and what you are trying to achieve , which part of the OCS are you intrested in?
the Dial in Conference or the Responce Group or anything related to the Faxing ?
please tell me more about your environment
Redvoz77 said
I am doing a Dial-In Conference pilot. I have three CUCM clusters and I need to create a SIP connection to the Mediation Server (that is connected to the OCS server). I do not want to make three individual SIP trunks (one per cluster) in Call Manager and send out to the Mediation servver. I would rather if possible make one SIP connection to the mediation server. I was told I could use the Gatekeeper. But I clearly do not understand how I could use the gatekeeper (currently I use prefix zones in my gatekeeper to identify each cluster). If you have any suggestions or information anything would be appreciated. Thanks.
Mino said
Actually you cannot create 3 Sip trunk and they all map to the mediation because simply the mediation is configured only to send to one of them.
what about installation 3 mediations servers under the same pool and connect each mediation server with a sip trunk to one of the Call managers server ?
in the end you will have 3 mediations and 3 Cisco and they all are failoverd so that if a Cisco server is down then the next mediation will use the next Cisco server that is up ???
I dont know actually about gatekeeper but if there is a way to have a single IP from the Cisco side ( sort of cluster or NLB look like thing ) then you will point the mediation to use that cluster IP as the next hope whether on 5060 sip unsecure or 5061 TLS
srndach said
Hi,
how can we integrate MOC R2 with Presence 6.x ?
We can’t control our calls from MOC !
It’s said : “No Phone system connection!”
With OCS work fine … but when we install OCS R2, nto working ?!
Mino said
you might be intrested to check this out
Cisco UC Integration for Microsoft Office Communicator (CUCIMOC)
http://www.cisco.com/cdc_content_elements/flash/voice/uc_10317/index.html
rob said
Thanks for the write up Mino. What if you want users to share the same extension between OCS and CCM? Your scenario has different extensions for each side. I think the term for this is forking. What changes to your post would need to be implemented to accomplish this?
Mino said
Hey Robert,
Thanks for your question , to explain this more i have to tell you that there are two stories . the microsoft story for integration and the Cisco story for the integration….of course for each story there is a side where each side wants the benifit for his own technology.
Micrsoft wants to assure that the call processing is happening on the mediation and not on the CUCM so it tells you that you have to deal with OCS as if it is another PBX with another extensions and for this you will have to create a sip trunk.
Cisco on the other hands tells u that you dont have to change ur extensions and we can integrate using CUPS and keep that extension the same , but this will make ur Office communicator just a soft phone for Cisco and u will lose all OCS features like conferencing for example.
Duall forking is the term that everyone uses which means that whenever a call comes to an end point , there is another signle goes to the other end point. in other words still the Microsoft is using diffirent extension than Cisco but when a call comes to cisco phone set it sends another signal to the extension on the OCS which in the end makes the user recieves a call on both end points at the same time. For this you need CUCM 7.x series not less than that and you will be looking for a feature called ” Simmiltanious Ringing ” from Cisco Side to OCS Side and it is only available on CUCM 7.x
There is an easier way now that Cisco is providing without the need for CUPS which is the CUCI MOC ” Coocky Mook ” !!!! funny name .
it is just an add on application for the Communicator client that does this integration without the hard integrations.
Personally I am not a big fan of these kind of integrations with CUPS because it usually doesnt work , I say use a sip trunk , try the similtaniously ringing thing, get the benifit from both products and in the end as an End user all what you are looking for is this :
” when a call comes to my OCS it should ring on my Cisco as well , when a call comes to Cisco it should ring on my OCS as well”
rob said
I am currently running CUCM 6.1 and plan to update to 7.x. I can’t do the update just yet because I also run Cisco ICM and each (of the zillion or so) components will have to be upgraded as well. I also did a CUPS implementation but never rolled it out to the masses. I don’t care for the CUPC; in my opinion it just isn’t ready for enterprise use. I do have the OCS-CUPS integration in place for RCC and it works well enough. I’ve been hearing about the CUCI product and it sounds like it will give me what I’m looking for. I even tried the link to download it from the Cisco site (http://cisco.com/en/US/products/ps10317/tsd_products_support_series_home.html?q=). It tells me that it isn’t available. I’ve pinged my Cisco rep to find out whats going on there. So if I understand you correctly, the steps above will still apply, I just need to implement the similtanious ring feature. Where would I manage this second set of extensions? Employees have DIDs and the last 5 digits of the DID is the CCM extension. The DIDs are listed in the AD telephone field and the CCM extensions are in the AD IP phone. I’ve normalization rules on the OCS side to ensure that numbers pulled from the AD telephone field are displayed as the 5 digit extension. I’m just unsure of where this new set of extensions would be managed.
Mino said
U are walking on the right path Robert , the other extensions has nothing to do with that you have in the Active directory .
you can keep that as is , the OCS extensions are configured from inside the user properties on the OCS server or from the Active directory Communications tab inside the user properties .
you have to enable the Eterprise Voice and configure the Line URI of the user in the format of tel:+1975454xxxx
where the number should be matching ur new OCS extension , it can be in the full DID format and u just change the last 5 digits to match ur new extensions .
by the way they can be imaginary 5 digit extensions range and u configure the Cisco side to replace this number for outgoing calls with the real DID number which the user has got on Cisco.I already did that before
Mino said
For example if your Cisco extensions are 7xxxx and we have a user for example with 76555 then you can make the ocs range with 3xxxx and for this user to be 36555 .
by just changing the first digit and keep the rest as is for the user.
this will make it easier for you on the Cisco side to configure a rule saying whenever you find a call coming from extension 3xxxx just change it to 7xxxx and set this as the caller extension whether internally or to the PSTN
rob said
That makes perfect sense. I can correct the extension with an application dial rule in CCM. One last question, will this configuration break RCC? This field is currently populated with the actual CCM extension and that is passed along to CUPS. How will cups know which extension to control? Is this where the location profile normalization rules come into play?
Mino said
No no no my friend …you will no longer use CUPS
CUPS means that there is no configuration on the OCS and there is no SIP trunk .
Again when CUPS is used then OCS is just a soft phone skin to Cisco , so you can not do diffirent extensions and you will lose most of OCS features whether it is OCS 2007 RTM or R2 because in the end you are not getting any benifit from the OCS features , are are just disabling it and let it be controled by the CUPS in RCC mood.
RCC is totally diffirent from Enterprise Voice.
Take a look on this article and let me know what do you think…. it is such a great article about the Voice scenarios .
http://blogs.technet.com/jkunert/archive/2008/07/30/voice-scenarios-with-ocs-2007.aspx
tsm said
How do I bring in a DID that is terminated on CUCM 7.0 for dial-in Conferencing on OCS 2007-R2.
I already have a sip trunk that is configured for AA. i.e. Another DID translated into AA extension inside.
Mino said
Can you give me more details about your configuration and what exactly are you trying to acquire ?
I have already configured the Dial in Conf on Cisco 4 , 5 and 6.1
tsm said
Thanks for the prompt reply. I would like to enable Dial-in Conferencing for OCS Live Meeting.
Basically, idea is that users can join via dial-in conferencing to the Iive meeting by dialing a DID fro PSTN
.
I am running OCS-R2 STD
I have a sip trunking service for UCUM 7,
The the SIP trunk from Mediation to UCUM being used for AA (ext. 1500) translated into full e164 number. (when someone dials published DID from PSTN, a TP- translation pattern on UCUM matches the dialed number and translates it to ext 1500 (which is Tel URI of AA)
Do I need a another SIP Trunk and new TP on UCUM that will translate the Conferencing dail-in DID? What would this number be translated to in TP?
1. Enabled the Access number (DID) under Forest Properties / Conferencing Attendant Properties
(By default, it creates a SIP:Microsoft.Rtc.Application.xxxxxxxxx. Does it bind the tel URI with this SIP URI, I can’t seem to find this object in AD or OCS Users?)
2. Enabled PSTN dial-conferencing under Global Properties / Meeting / Default Policy
What am I missing in the conf. Thanks for all your help in advance.
Charly said
Hi, in my case Im integrating CCM 5.1 with OCS 2007.
CCM manage all communications from PSTN.
In OCS I have non DID extensions with 5 digits begin 7XXXX. In OCS I used 8XXXX.
I can call from OCS to CCM using E164 format, my users in AD has this formtat
tel:+54115XXX4XXX;ext=8XXXX.
But when a user from CCM call to OCS not work. I was seeing packet sending from CCM with WikeSahrk and view that CCM send as sip:8XXXX@IPaddressofMediation.
I was testing too CCM sent me packt begin 54115XXX4XXX with same sympthoms.
Do you have any ideas?
Mino said
Hi Charly ,
First i need to know if this is OCS R1 or R2 ?
second i would like to ask if you have configured LINE URI inside the user properties from the OCS management consol ? you should define LINE URI for the user with the same extenions you are using which are the 8xxxx series.
however you have mentioned very important information which is that the packets are sent as 8xxxx@mediation IP , which means that they are sent from your CCM without adding E164 paterns to them before sending them to OCS.
If you are using OCS R2 then the OCS will do this translation for you automatically but if you are using R1 then you should configure the Cisco to add to any outbound calls using the sip trunk of 8xxxx to add the E164 paterns.
Charly said
Hi thanks for you response. Im using R1 and sorry Im configuring LINE URI as 8XXXX.
How I say to CCM 5.1 that add E164 parameters? Can I add a “+” or complete number for non DID numbers in CCM?.
How to configure a SIP trunk between Cisco Call Manager 5.x or 6.x or 7.x and OCS 2007 R1 or R2 « msunified.net said
[…] by Ståle Hansen on 28/04/2009 Great post about enabling sip trunk from Cisco to OCS https://theucguy.wordpress.com/2009/02/20/how-to-configure-a-sip-trunk-between-cisco-call-manager-5x-… Possibly related posts: (automatically generated)Microsoft OCS SIP an advantage over […]
Jatinder Mann said
Hi All
I had installed OCS 2007 (A/V, Instant messaging etc etc.. ) working fine .. now i need to Integrate with Cisco Call Manager 6.1 and Exchange 2007 for unified Couminication. Does anyone have step by step configuration notes how i can integrate it and get MY UNIFIED COUMINICATION done.
Evelyn said
Hi Mino,
Great posts on CCM UM integration thank you. I have a query; we are setting up E2007UM (VMail only) with CCM 7 – the VM’s side is working but we cannot get CCM to accept unsolicited SIP notify to generate MWI state change on CCM phone. Are you able to offer suggestions on appropriate SIP Trunk settings or are the settings above correct for this configuration? We have same extension on both sides though UM call plan 5 and CCM 4…
JIJO said
Hi,
I have a problem… I am configuring an E1 Trunk line on a cisco 3800 series.
I managed to do the configuration, however when I make a call from the PABX, I cant see the three digits.. extension..e.g. 0711084***, I cant see ***,
Any Ideas?
The exchange is okay and has unmasked….
thanks,
JIJO
KMN said
Ok I could use some help. I have OCS r1 with a mediation server trying to talk to CCM 6.1
When I run wireshark I don’t see ANY traffic between the call manager and the mediation server. To make sure I was actually recording traffic I did a ping and that worked and I saw the ICMP echo request going back and forth. So why can’t my mediation server see the Call manager? I have it configured as you do above (except for the ip address info and specific search groups and such that is unique to the environment.) What gives…
When is the SIP trunk actually formed? Is it a persistent connection? How can I test to see where my problem is.
It would be helpful to show the OCS side in the above configuration for your mediation server.
Thanks for any help you can provide.
~K
Ahmed said
Hi,
I deployed CUCM version 6.1 , and calls between cisco and OCS is fine and inbound from PSTN to OCS also fine, however outbound to PSTN ( national,International ) not working and communicator gave error 404 , do you have suggestion should I make on cisco .
Thanks
Mino said
Hi Ahmed,
this 404 error is usually if u traced it with snooper u will find it ” user not found ” which means that one of two things is happening :
1- either there is no search directory configured on the cisco sip trunk configuration
2- the route patterns created on the cisco side are not properly configured or it might not be able to handle the + sent from the mediation.
please use wirshark on the mediation server and let me know what is the number format which is sent to the cisco side , also check on this blog other cisco articles talking about such problems.
I hope that would help
Ahmed said
Hi,
thanks for you reply , In user telephony option on OCS , I enabled Enterprise Voice with PBX integration and leave SIP URI blank and add in user Line URI +9649983733 ; ext=733.
where 733 is user extension on cisco call manager.
And add in Host Authorization , Routing taps the PSTN Gateway .and do appropriate Normalization Rule . also I did route .*
This from OCS Side , do I miss something from OCS side , or all from Cissco side?
and that time I will check with cisco side.
Thanks
Mino said
as long as you are able to make calls from cisco to OCS and PSTN to OCS then it is not a problem from the OCS side…..it is the way the Cisco trunk is configured to handle the numbers you send from the OCS.
for example if you are dialing from OCS +974 2233445566 where the +974 is the country code then you need to configure the Cisco trunk to remove the first 4 digits and add 9 if required to pick up the line and then dial the rest of the number 2233445566 .
also you might need to strip the + from the OCS so that the number would be sent directly to the cisco without the plus ( 974 2233445566 )
Ahmed said
Hi,
Actually Cissco Sip trunk configure 3 digit significant (OC and Cisco 3 ext ) .and when i call from OC to cisco it going without + .
but Still not to PSTN , and how to configure the Cisco trunk to remove the first 4 digits and add 9 if required to pick up the line and then dial the rest of the number 2233445566 ?
thanks
Ahmed said
Thanks Mino,
I did it by add user Line URI and ext same as cisco .It’s working , thanks at all.
Thanks
Tim said
Any has the step by step configuration of a SIP trunk for sending and receiving calls on UCM 6.1 ?
Eric said
I’m a little new to the integration features. So what can you do with OCS 2007 R2 and Cisco call manager if integrated them and what are the different options to integrate?
Alex P said
-The SIP trunk (Cisco-OCS)works fine with Devices connected in Cisco CM with codec G.711, but the SIP trunk does not work with Devices connected in Cisco CM with codec G.729.
– Between OSC and Cisco, the Video does not work in the SIP trunk.
– Does OCS support G.729 in the SIP Trunk?
– The OCS Mediation Server is able to do trancoding?
– Can the video go across the SIP Trunk ?
Hemal said
Hi Mino,
We have OCS R2 integrated with CCM 7.0. In our case we don’t have phantom numbers for OCS. We use real DID number for both Cisco and OCS. So if a user is OCS user then through Route pattern on CCM it is routed to OCS SIP trunk. If user is Cisco user then it is terminated on Cisco device.
From OCS we transfer everything to Cisco and let Cisco decide how to route a call. So if User 1 is calling User 3 by dialing 5 digit extension and they both are OCS user then call goes from Mediation-> Cisco -> Mediation.
How can I make this as communicator call? I want to dial digits (DID) but still make it as communicator
call.
For Ex:- User 1 is OCS user (30001 as DID)
user 2 is Cisco user (30002 as DID)
user 3 is OCS user (30003 as DID)
When User 1 calls user 3 by dialig 30003 then it shld be communicator call
when user 1 calls user 2 by dialing 30002 then it shld go to Cisco device.
we have tel uri as Tel:+1xxxxx30001;ext=30001
Currently we have rule that anything with 3 and 4 digits add +3xxxx and transfer to Cisco.
I tried deleting this rule and hoping that OCS will know that this is my extension and I am not suppose to transfer to Cisco I can make MOC call, but is failing.
Is this Doable?
Mino said
Hi Hemal,
I will try to give you some examples to what i have in mind and how i test things to make sure they are working fine .
please let me know if the below will help you
let us try somthing together and let me know how it will go :
1- Create rule in the location profile to translate 30001 to +1xxxxx30001 , make this the first rule
2- make sure that the policy and the route will permit anyone to do this dialing
3- Configure the line URI of the user with tel:+1xxxxx30001 only without the extension
4- Configure a second user link URI with tel:+1xxxxx30005 only without the extension
5- from the communicator client of the first user try to dial 30005 only , you should see it automatically translated to +1xxxxx30005 and you should see the communicator client of the second user is ringing . ” this is pure OC to OC call without any mediation or Cisco interfeired ”
From the Cisco side make sure that you configure the range of the OCS DIDs so that it would be automatically routed on the SIP trunk directly to the OCS , where OCS will lookup in the local database and match the number with the line uri of OC user and communicator should ring.
Now From OC to Cisco
1- since the DIDs are the same for OC and Cisco so no need to create a new rule for Cisco extensions
2- when user 1 dials user 2 on Cisco 30002 then the OCS will translate it to +1xxxxx30002 and try to match it with user in the OCS
3- since it wont fine it then it will send it to the next hop which is the Cisco CCM
4- on cisco you should configure route patern whenever it recieves +1xxxxx30002 , it would only accept the first 5 digits from the right and ignore the rest
5- also make sure to configure the address book where the cisco will lookup for the number
6- Cisco address book will find match for 30002 to users 2 Cisco end point and then it will ring the phone end point.
Now From Cisco to OC
1- you should configure Cisco route paterns with the range of the OC extensions to send is as is to the sip trunk with OCS
2- From Cisco phone 30002 User 2 will dial user 1 on his OC 30001 , Cisco will find a rule that the OC extensions range should go to the trunk as is
3- Cisco will send 30001 to mediation and from mediation to FE which will lookup for the translation patern
4- OCS will translate 30001 to +1xxxxx30001 and it will lookup for the local database which will match for user 1 and the communicator will ring
Now From PSTN to OC
1- PSTN user will dial OC user on his DID +1xxxxx30001 , which will be recieved on the Cisco
2- Cisco will be configured to remove the +1xxxxx , and only dial the 30001 so that it would be the same rule created for the internal cisco phone when it tried to dial an OC user from internal ( Phone to OC )
3- Cisco will send 30001 to the mediation through the sip trunk ,and the same matching mechanism will happen just like the one explained and in the end OC will ring.
Now From OC to PSTN
1- OC will dial a PSTN number like 9777666555 , where 9 is the digit for PSTN calls and the PSTN number is 777666555
2- Normalization rule will translate 9777666555 to +1777666555 and sends it to the Cisco
3- Cisco should be configured with route patern to remove the +1 and dial 777666555 acording to the PSTN dial rules on the cisco side.
Hemal said
Hi Mino,
Sorry for the late reply, yes i was able to achieve this results.
Thanks for your reply.
-Hemal
Hemal said
Hi Mino,
Somewhere above you have mentioned about DTMF issues.
We also have experienced DTMF issues when calling from OCS R2 client. I have experience with Microsoft UM, OCS conferencing. Other users have noticed with other AA systems.
Basically DTMF doesn’t work always from MOC client, they have to either call again or wait till the end of speech. Sometimes you can interrupt speech and its goes fine but not always.
Few of the blogs says disabling “RFC 2833″ but I couldn’t find that setting on SIP trunk going to Cisco call manager 7.0. I probably believe that disabling RFC 2833 is on Cisco device and not on SIP trunk.
Also on SIP trunk we have DTMF set to ” No Preference”.
Any idea how to fix this issue?
Thanks
-Hemal
Janty said
Dear,
i have configured the integration between CUCM 6.1 and the OCS, and i have created 1 route pattern on CUCM that sends the calls to 6014 to the SIP trunk, and the call is getting through to OCS and it is working like a charm, but what is happening is:
if a cisco ip phone with DN 1916 dials 6014, it would ring on the OCS (6014) ip phone, but on the cisco phone it shows that its calling 1916, never the less the call gets through to OCS and it is completed without a problem, i tried from different cisco ip phones and it is the same, whenever im call 6014 (OCS IP Phone) from any cisco phone it shows that im calling the same extension im calling from, its like im calling myself but the call is correctly routed to the OCS and its fully working, please any one has a clue on how to tackle this as its giving me a big headache, thanks.
Lupula said
Hello!
I have CUCM 6.1., 4х the-place number schedule, OCS 2007 R2.
It is possible to carry out calls to two directions having one number on Communicator and phone (1111)?
1. We call from phone. The call goes on phone and on Communicator.
2. We call with коммуникатора. The call goes on phone and on Communicator.
Lupula said
Hello!
I have CUCM 6.1., 4х the-place number schedule, OCS 2007 R2.
It is possible to carry out calls to two directions having one number on Communicator and phone (1111)?
1. We call from phone. The call goes on phone and on Communicator.
2. We call with Communicator. The call goes on phone and on Communicator.
Mino said
you will need CUPS for that , also you will need to work in RCC mood from the OCS side .
this means that all call processing will happen on the Cisco side and the Office communicator will be just a skin or soft phone for Cisco.
you will loose some OCS features for that like conferencing which will be dimmed , also the integration itself is not easy at all
Lupula said
I have customised calls without CUPS with communicator on phone and communicator simultaneously.
I can not customise only from phone on коммуникатор and phone simultaneously (having one number). And with different numbers for phone and communicator all has turned out.
inner_silence said
Hi,
M new to CUCM – OCS integration. I am working on a CUCM – OCS integrated setup and have few questions:
1) Is it mandatory to have mediation server? My topology:
PSTN GW—CUCM—-SIP—Mediation Server—SIP—OCS STD R2. Currently I’ve CUCIMOC setup configured and running. M wondering if I Can’ve SIP directly to OCS without mediation server or is something always mandatory for 3rd party integration irrespective of mode i.e. PC to PC, RCC or Enterprise voice?
2) Can I do audio conference in above topology?
3) Can I do video conference in above topology if I have bridge available at CUCM side?
Thanks in advance,
inner_silence
youssef abdullah said
hey guys
i made an integration between OCS R2 and CUCM 7.0.1
the call flow normally from ip phone to the MOC but from THe MOC to ip phone is not working ….
i m configured the client as enterprise voice with line URi:tel+5000 and the ip phone with ext 2000
what could be the problem of that ??
Mino said
1) Is it mandatory to have mediation server? My topology:
yes it is because Microsoft uses RTAudio codec and it needs the mediation to convert this to G711 for Cisco . also it is a must when creating a sip trunk between OCS and any IP PBX . however in wave 14 of the OCS comming late this year you can have the mediation and the front end on one server.
2) Can I do audio conference in above topology?
you can do audio conference whether on cisco and then invite someone from communicator client , or have a conference call over communicator users and invite Cisco phone set.
Can I do video conference in above topology if I have bridge available at CUCM side?
you can use any webcam for the video conference of the OCS , however if you want to use Cisco tower phones that includes the screen and video cam then you might need CUPS for this to integrate Cisco Video with OCS and to be used as sip endpoint.
Mino said
first check the communicator logs using a tool like snooper as it will tell you what happened to the packets . you will probably get a 404 reply of user not found.
if this is true then you must check the routing rules on the Cisco partition , as it might not be able to understand the + for example.
youssef abdullah said
this is the error that i get
503 Service Unavailable
Ms-diagnostics: 10503;source=”OCS-MEDIA-01.AHBS.LOC”;reason=”Gateway returned a SIP failure code”;component=”MediationServer”;SipResponseCode=”503″;SipResponseText=”Service Unavailable”;GatewayFqdn=”CUCM-01.ahbs.loc”
and for the + , i removed the + from the request URI … so that the number it suppose to be send without + .
Mino said
How many NICs on the mediation server ?
if 2 NICs then make sure that they are completely in different subnets
if 1 NIC then make sure that you are using 1 single IP only and not multiple and this IP can communicate with both OCS front end and Cisco CUCM .
Mino said
also check this as it might be the same as your problem
http://tech.rundtomrundt.com/2009/04/batteling-mediation-server-or-this-time.html
youssef abdullah said
i have 2 NIC with 2 ip but in the same subnet ..
i will check and tell u what i gonna happen ..
tx in advance
Mino said
Bingo , thats the problem my friend 🙂
disable one of the 2 NICs and configure the OCS to use 1 NIC for everything and make sure that this NIC has got the DNS register configured and that it is updated on the DNS server.
youssef abdullah said
U got the problem man 😀
i used one NIC and all works great
Tx Mino
inner_silence said
I removed the mediation server from my setup and it still works fine. I realized CUCIMOC based setups don’t need Mediation server.
yes, Audio conference works fine in all scenarios.
Video conference with MOC clients and tower phones works fine without CUPS. I’ve Cisco MCU registered to CUCM. Since its CUCIMOC managing the show, that’s why it all works.
Qustion:
1) I’ve set the Users for PC-to-PC mode. When I click on any contact in MOC, I see the work phone (+2000) greyed out, I can’t click on that to dial it, any ideas?
2) I tried using click-to-call via outlook. When I click the contact to call, its a communicator only call. Can’t I dial-hard phone?
Thanks,
inner_silence
AM said
can anybody please help me with the scenario where i have two CUCMs running as completely separate entities on different network. I need to have the phones registered to the first CUCM communicate with the phones registered with second CUCM. I am trying to create a SIP trunk between the two but in total vain.. please help me with this SIP trunk and communication.
thanks
AM
Jamil said
Dear Mino ,
I wanna ask you, how I can do configure the CUCM version 5 to make it possible that any extension in the OCS can call any destination ( mobiles, PSTN , international and local, services number , and the internal extensions on the CUCM )
and one more is there any way to make the same extension from the managers on both the OCS and CUCM.
I have this huge project and am not that expert, any help would be appreciated.
Jamil said
Dear Mino,
What I meant by “nd one more is there any way to make the same extension from the managers on both the OCS and CUCM.”
to have the same extension in a way if some one is calling the extension both the Ip-phone and the OCS extension would ring!
Mino said
Actually with the new media gateways from N.E.T aka Quintum , the UX models has got built in Dual forking feature .
which means that you can connect the PSTN PRI line to the media gateway and then it will send two signals with the same extension to two different IPs which is in this case will be one for Microsoft and the second to CUCM . in the end the two endpoints will have the same extension and both will ring at the same time.
Robin said
Hi Mino,,
We have a CUCM cluster connecting to a mediation server via SIP trunk.My questions is: How do we connect another 3 mediation servers to the CUCM cluster for failover? Can this be done via DNS?
Thanks
Robin
Mahmoud Saber said
Dear ,
We have CCM integrated with OCS R2 for Dial in conferencing only and it was OK , suddenly it stopped working and CISCO said there is a port mismatch between OCS and CCM and they need to chnage the udp medai ports of OCS
1- which OCS ports range they mean ?(Mediation or fromt end)
2- is this will affect other OCS services ?
3- can this be done from CCM ?
regards ,