Please confirm.. Whether Cisco CCM support SIP trunking !

Hi,
We r in the process of VoIP implementation nd during that phase came to know that Cisco doesn't support SIP trunking at gateway level.. Can anyone suggest me whether it's true or not??? Cisco says its support SIP then what is the meaning of that?? I m really confused with the twisting of words in different docs...

Thanks to all of you for your reply...When you say SIP trunking is supported then whether I can use SIP at gateway level towards making outward call.. Sorry if I am getting confused myself but when one says SIP trunking is supported what's the mean of that?? If I have a trunk line(PSTN) and a T1 line acting as a gateway, is it possible to design a network only depending upon SIP protocol using CISCO UNIFIED Components and call manager 5.0 ??
Sorry once again if my question sound something ..... but it just to clear my confusion too as I am listening so many stuff from different people when it comes to SIP overall implementation for a Enterprise network using CISCO Unified communication equipments...
With best regards,
Mani

Similar Messages

Cisco IOS Version to Support SIP Trunking on AS5400 Routers

Hi all,
Kindly help .Can anyone tell me which Cisco IOS Version Supports SIP Trunking on AS5400 Routers ?
Regards,
Cliff .

Hi,
Please look at:
http://www.cisco.com/en/US/products/hw/univgate/ps505/products_data_sheet09186a0080091e51.html
http://www.cisco.com/en/US/docs/routers/access/as5350/software/feature/guide/UPfeapu.html
HTH
Jorge Armijo
Please remember to rate helpful responses and identify helpful or correct answers.

Please confirm whether a Lightning to 30-pin Adapter will work with Apple 30-pin Digital AV Adapter?

Please confirm whether a Lightning to 30-pin Adapter is suppose to work with Apple 30-pin Digital AV Adapter?
The Packaging of Lightning to 30-pin Adapter suggests it does.
But I have tried this today and my IPad just stated Unsupported accessory.

Expect Lightning extention cables in black shortly. Since extention need no special circuitry 3rd party's are already working on then.

Please confirm whether its (a) or not?

Send message within an integration process to 8 receivers at the same time, how can I do this?
a) Create a FORK statement with 8 branches
b) Create 8 interfaces
c) Create a loop with 8 interfaces
d) Create 8 branches

Muhammad,
Using the one interface, you can configure multiple receiver steps in your Receiver Determination (Integration Directory)
Taking this further, you also have the option to add conditions to control when a receiver receives a particular message
I hope that this answers your initial question Send message within an integration process to 8 receivers at the same time, how can I do this?
As you have a duplicate thread, I am now confused as to which thread you are following. So I have provided the answer in in each thread
Could you please keep your thread to a specific issue to make it easier for both yourself and fellow Forum users to follow the topic
If you have other questions relating to a different topic then please create a new thread
Regards,
Mike

SIP Trunk to 2811

Hi,
We have Cisco 2811 with CCME and also act as H.323 gateway for branch VOIP traffic. Cisco IP phones are registered in the router and there's a PRI E1 connection to PSTN cloud from the router. One Service provider offers SIP trunk.
Q1. Is it possible to configure SIP trunk in 2811 and enable calls between H.323, SIP & PSTN?
Q2. If possible, what are the pre-requisities ?
* IOS
* Additional Resources
Q3. Highly appriciate if someone can provide sample configuration with simple dial peers to enable voice routing between SIP, H.323 & PSTN cloud?
Thank you
Lakmal

Thanks for the prompt support.
Existing setup: refer attached diagram & router configs
As explained earlier Location-1 is connected with PSTN PRI e1 and 2 mbps data link.
Call flow:
Location 01:
E1 Number series: 038-42911XX
Local Extensions: 33XX
PSTN Access Code: 9 followed by 7 digits for local and 10 digits for national calls
Location 01 can directly dial location02 by using 4-digit extension (62[345]x)
Location 02:
Local Extensions: 62[345]X
PSTN Access C ode: 9 followed by 7 digits for local and 10 digits for national calls. These calls will connect to PSTN cloud through Location 01.
Location 02 can directly dial location-1 by using 4-digit extension (33xx)
In the proposed setup, the SIP trunk colored in red is introduced. Could you please help me to configure the SIP trunk in the 2811 router and route calls through it? For example assume we need to route PSTN calls through SIP trunk using access code 8 which is currently doing by access code 9. Also need to terminate incoming calls from SIP trunk to 3300 extension. Please check whether the current IOS is capable?
If you require any further details, please let me know.
Thank you

UC540 SIP trunk "allowed IP addresses" limitation?

Testing out a UC540 and I'm using Callcentric. Working great, but the feature under the SIP trunk configuring/Advanced that allows you to set the IP addresses of machines allowed to connect to the UC540 only supports a maximum of 100 entries. Callcentric wants 204.11.192.0/24 or 204.11.192.0-204.11.192-255 unblocked.
The UC540 doesn't seem to support inputting a range in any notation I'm aware of, maybe I'm missing something?
I contacted Callcentric and asked if they could be more specific, LOL, but they said they change things on their end from time to time.
Anyone else figure out a way around this?

Okay, follow-up to the question.
How bad an idea is it to leave this security feature disabled?
I have a bit of experience in this area, I use an Asterisk (FreePBX distro) for my work phone system, and Callcentric is my SIP trunker. Before I limited incoming connections to my FreePBX server to the Callcentric range above, there would be occasions (maybe once every 2-3 weeks) where someone (probably up to no good) found my open port(s) and all my phones would start ringing simultaneously. It lasted for about 20-30 seconds, and then would stop.
My assumption was that someone was looking for security vulnerabilities so they could use my Asterisk box to place their LD (maybe even overseas) calls. They never got anywhere, though, as the Asterisk was fully patched, and my passwords for my handsets were strong.
I wonder if the Cisco UC540 will suffer the same sort of result, or worse, or maybe even not so much.
Obviously I can simply select another trunker. The thing I like about Callcentric are the rates and also the dashboard that allows me to make a lot of adjustments (like forwarding calls to a cell phone number, and the ability to configure faxing reception, too). So if anyone has a suggestion for a supported SIP trunker that supports the same types of features, I'd be open to that, as well.
BTW, I did ask Callcentric one more time, with a link to this topic, whether they can restrict that range to better than 0-255, but they declined. I see their point, just wish there was another simple way around this.

Lync 2013 with SIP trunk with panasonic kx-tde200

Hi
My company has installed a panasonic ip-pbx kx-tde for multiline with 100 number range for telephone service.
Now my company is going to replace multiline by sip trunk . It will still work with Panasonic pbx box just need to reprogramme to be able to connected to the sip proxy which is managed by internet service provider.
For this scenario , would Lync 2013 voice work if I just add PSTN gateway which is the ip of panasonic pbx address to the frontend in topology ? Or I may need a mediation server as a must requirement to make lync voice work?
Thanks
WenFei

Media bypass allow a call to basically skip the mediation server once it's established and go directly from gateway (in this case the PBX) and the endpoint (the telephone handset or Lync client) More information here: http://technet.microsoft.com/en-us/library/gg398719.aspx
By having this (if your PBX supports it) you reduce the load on the mediation servers. Before you go too far down the road also make sure that your PBX supports SIP trunks that are SIP over TCP (as Lync doesn't work with SIP over UDP)
Sort of, the easiest way is to add the .com as an additional SIP domain in Topology builder, you will need to create DNS records for it (both internal and external) and you will need to reissue the certs with additional SANs to support the second domain.
YOu will also need to update all the users to use the new suffix of xxx.com. So it's not a small task.
If this helped you please click "Vote As Helpful" if it answered your question please click "Mark As Answer" | Blog
www.lynced.com.au | Twitter
@imlynced

Is SAP SCM 5.1 and SCM 2007 are same? Please Confirm

Hi,
I have to Install SAP SCM 5.1 Non-Unicode System, but I cant find Installation Media at Marketplace instead I can find SCM 2007. I am able to get SCM 5.1 SPs when I clicked SCM 2007 entry on Marketplace. So please confirm whether these 2 products are same or not?
If yes, Please provide a link/ note that shows SCM 5.1 has renamed to SCM 2007 from SAP.
Thanks.

Hi Ajay,
SCM 5.1 & 2007 are the same.
Please follow the link
https://websmp110.sap-ag.de/support
under Release & upgrade info > installation & upgrade guides
Regards
R. Senthil Mareeswaran.

I Want Buy Cisco ASA Firewall Supporting SIP

Hello Guys I want to buy cisco ASA Firewall , that support SIP and Session Border Controller (SBC) So please can any one tell me the most power full that support this protocols ,, Than you guys

Hi Vijay,
If can be done but you need any network management software. I personally dont think you can ask your ask to send mails. ASA can trigger alert to a SNMP configured server which will intern send mail to you
HTH,

Please confirm that c3560-ipservicesk9-mz.122-55.SE9 supports xconnect and pseudo-wire

Hi everyone,
Could someone who is actually using this image please confirm that c3560-ipservicesk9-mz.122-55.SE9 supports xconnect and pseudo-wire
Thank you
~B

Hi everyone,
Could someone who is actually using this image please confirm that c3560-ipservicesk9-mz.122-55.SE9 supports xconnect and pseudo-wire
Thank you
~B

Why we dont' see H323 gateway and SIP trunk "registered" in CCM?

in CCM admin, we see the status of H323 gateways and SIP trunks have no "registered to xxxx" status.

Unlike devices with other protocols, there is no registration mechanism for these.

How to create multiple sip trunks between cucm and cisco unified sip proxy

Dear Expert,
Is there a way to create multiple sip trunks between CUCM and Cisco Unified SIP Proxy (CUSP)? How to achieve it without creating multiple IP interfaces on the CUSP module.
CUCM: 8.5.1.10000-9
CUSP: 8.5.2
Thank you,
.wan

Hello Michael,
This SIP trunk is part of UCCE solution, which used between CVP, CUSP, and CUCM.
The requirements:
1) To have different codecs for different type of calls, as the phones are at few countries
2) To pass different number of digits from CUSP to CUCM for different call treatments
.wan

I have Installed OEL5.O. But unable to install Firefox5.o. Please let me know whether OEL5.O Supports Firefox5.0?

I have Installed OEL5.O. But unable to install Firefox5.o. Please let me know whether OEL5.O Supports Firefox5.0?

Hi sandeep2911
Because you are using an unlocked BlackBerry, the device software is limited to what the original provider has released. Let me know who your original provider is and I can see what the latest version of software available is.
Alternatively, you can plug in your BlackBerry to your PC and go to www.blackberry.com/update to install the latest software available.
Thanks
-CptS
Come follow your BlackBerry Technical Team on twitter! @BlackBerryHelp
Be sure to click Kudos! for those who have helped you.Click Solution? for posts that have solved your issue(s)!

Could you confirm whether or not sr_CS is a supported locale for JDK7

Could you confirm whether or not sr_CS is a supported locale for JDK7.
According to Oracle link (http://www.oracle.com/technetwork/java/javase/javase7locales-334809.html) it is not.
We need official confirmation regarding their support of it.

http://en.wikipedia.org/wiki/Leap_year
They have some leap year algorithms on there.

SIP trunking between Microsoft OCS server and Cisco Voice GW router.

Hello All,
I have a client with an existing Microsoft OCS (office communications server) environment with the OCS server in their head office. The OCS clients in the remote Office registers with the OCS server in the head office. The WAN connectivity between the remote office and the Head office is MPLS.I would like to facilitate local call (PSTN) features at the remote site through a newly proposed Voice gateway router.
Can I achieve this by doing a SIP trunk between the OCS server in the head office to the newly proposed voice GW router in the remote office through the existing MPLS link. If yes, Could any one please assist me in this regards or suggest any other best solution to achieve the same.
Thank you in advance,
Mohammed Ameen R

Hi David,
this is a normal behaviour. To CUCM, OCS is a remote destination (just like your mobile phone). When your mobile phone hangs up, the system will put the call on hold for 10 sec.
This is there for the mobile user to go to his desk to pick up the call and continue the conversation (part of single number reach feature)
The best practise will be for the user to ensure that the other party hangs up the call first before he hang up.
Please grade if you think it's useful =)

Please confirm.. Whether Cisco CCM support SIP trunking !

Similar Messages

Maybe you are looking for