(7 votes, average: 4.86 out of 5)
Loading ...Fri 11 Jul 2008
GNS3: How to Register third party SIP phone to Cisco Unified Call Manager Express (CUCME)
Posted by Tariq Ahmad under Cisco Unified Call Manager Express , GNS3 , GNS3 video tutorials , Session Initial Protocol(SIP)[47] Comments
Email This Post
In one of the previous tutorials , i had demostrated ‘Direct Mode’ approach for registering third party SIP phone to Cisco Unified Communication Manager Express (CUCME) . This tutorial will demonstrate ‘true SIP mode’ Or ‘Register Mode’ in which SIP device will actually register to VOIP router/SIP proxy server in order to make and receive calls.
Unlike ephone-dn/ephone which we use for Directory Number/SCCP phone configuration , we will use voice register dn/voice register pool to achieve same purpose for SIP telephony.
I have divided this tutorial into 5 easy steps to configure. I will demonstrate X-lite SIP phone as an example although this tutorial will work perfectly fine with any other standard based SIP device . I have personally tested it with other vendor SIP phones ( 3CX voip client, SJphone …just to name a few) and it works like a charm !
If you want to download supported SIP phones, here are the links:
- Download 3CX VOIP client
- Download SJphone
- Download X-lite Softphone
ENJOY !!!
If you liked this tutorial ,don't hesitate to buy me a Cup of Coffee today !
Loading ...
July 13th, 2008 at 4:53 am
Awesome ! Great tutorial and i just bought you a cup of coffee. Keep you the good work.Thanks
July 14th, 2008 at 6:07 am
great tut, it managed to get it to work, how can i configure that to dial outside numbers
July 14th, 2008 at 8:02 am
@luks, In order to route calls outside (VOIP,PSTN),you have to have your dial-peers setup correctly and you need to allow-connections between SIP and H.323.So,you will need to configure following on your CUCME:
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
This configuration will allow multiple types of VOIP connections.
Hope this helps!
July 15th, 2008 at 2:33 pm
Great tutorial! I managed to get 2 X-Lite phones to register with CME (this was checked using the ‘show voice register all’ command) but I cannot place any calls between the 2 soft phones, I always get the message “The person you are calling is unavailable, please try again”. Is your tutorial enough to place a call between 2 soft phones or is a kind of dialplan also required?
Many thanks,
Ant
July 15th, 2008 at 3:02 pm
Sorry my mistake, forgot to include the ‘allow-connections sip to sip’ command. It now works much better. Sorry for clocking up this page.
July 16th, 2008 at 6:45 am
@Ant, I am glad you got it working.In some cases, ‘redirect ip2ip’ command is also required to redirect SIP phone calls to SIP phone calls globally on a gateway.
Thanks!
July 18th, 2008 at 7:07 am
Hi Admin,
I started preparing CCVP. Pls guide me to setup a lab to practise using stand alone pc.
What are all the things i should download in my pc to setup a lab?
Help me…
Thanks & regards
July 18th, 2008 at 9:29 am
@Sre , Using Vmware and GNS3 , you can setup pretty much more than 70% of all exam requirements provided your PC have enough horse-power to do it.However, there are certain things in Virtual Environment that cannot be done like FXS/FXO testing,transcoding,dsp stuff,CUE simulation etc.For this , you may use online rack sessions to fill in the gaps.
Hope this helps.
July 31st, 2008 at 5:17 am
Hi Guys,
Thx for great tutorial.
I’m still having trouble dialing outside numbers or getting outside numbers to call in.
Keeps hanging up with “service unavailable”; I did the allow-connection bit on CUCME; but I still have a CM6.0 in front of the CUCME, is this my problem?
Regards,
Sven
August 5th, 2008 at 12:24 pm
Sven,your problem seems related to dial-peer configuration.Is your CUCME registered to CCM6 as an MGCP/H.323/SIP GW ?
August 25th, 2008 at 5:42 pm
HI
the first, thanks for your tutorial. It is very good.
I have a problem when I try to usage the X-lite softphone. X-lite can to register with my CME but when I try to outgoing call, in the screen of the sofphone I can read “call failed: Not Acceptable Media”
I can not place call and I can not receive call to my sofphone.
???
Thanks all.
August 26th, 2008 at 3:45 am
Hi Elias,your problem (“not acceptable media”) is related to Codec configuration on your X-lite Softphone. Make sure that you have correct codec’s selected in your X-lite soft phone settings.
September 2nd, 2008 at 4:27 am
Super tutorial. Very straight forward and it works. Well done.
Like everyone else, I have a question. I have a direct FXO port which I can use for dial out and that works. I also have a SIP client account which I use for IP calls. Can CME register with a SIP provider as a client or do I have to get a SIP trunk set up by him?
September 2nd, 2008 at 8:09 am
I am lucky that i found this site.your tutorials are very easy to understand and are extremely informative.Thanks for this invaluable resource.
September 2nd, 2008 at 8:18 am
@ Joe and Alisha , thanks for liking and appreciation.
Joe, CME can work both as a SIP server(where SIP phones register to it) and as a SIP client (to a VOIP service provider).You don’t really need to setup a SIP-to-SIP trunk for this to work. Infact,i have setup my 2801CME router to a SIP VOIP Service provider that provides DID and all standard VOIP features.You will see a tutorial on it very soon as it has been asked by many! so,Keep tuned in !
Thanks
September 15th, 2008 at 1:25 am
Great site with marvelous tutorials. Ok.
Here my question is whether is it possible to do video telephony through xlite.I tried through all video codec…but unable ?
Again I want to setup a true communication – via video phone, via email , via sms . I will use cisco unity, lotus notes and stonevoice. Is it possible to give a tutiorails on this.
September 15th, 2008 at 7:49 am
Hazra, I am glad you liked tutorials.As far as video telephony goes, yes, its possible to do that but X-lite(as its free) doesn’t offer this feature.Counterpath has another similar product called eyeBeam(licensed) that has video telephony features. Ofcourse, if you have it , you can intergrate it with your CUCME/CUCM. Regarding upcoming tutorials, if you have any specific tutorial request , let me know using contact-us page.
Thanks!
September 16th, 2008 at 3:51 am
Hi admin,
Thanks for your help.
September 16th, 2008 at 4:18 pm
Great tutorial!
Questions. I have an x-lite registered on the CME via SIP, it can place internal and external calls.
1. When calling from another phone (internal extension or external number) to the SIP extension, calls never get transfered to the SIP phone, the call continuosly loop within the CUE’s auto-attendant.
2. When calling another internal extension and the other party doesn’t pick up, the call gets directed to CUE’s login prompt (asking for mailbox username/password) instead of the other party’s voicemail.
Kindly assist on resolving these problems.
September 18th, 2008 at 3:18 am
Hi Admin,
I tryed with eyebeam(license),but again see codec mismatch…..but in gns3 router i have specify all the video codec.What can be the problem.
I can lisen audio but not vedio.Help me please.
September 18th, 2008 at 5:49 am
Hazra, have you specificied right codec in video codec settings in Eyebeam’s configurations? You must specify exact Video codec that you have configured in GNS3 router.
September 18th, 2008 at 9:30 pm
I am not able to found where should I setup video codec setting in Eyebeam.
I have setup all the video codec in GNS3 router.
September 18th, 2008 at 9:51 pm
@Hazra, Right click on your softphone screen(EyeBeam). Then go to Options –> Advanced –> Video Codecs .There you should be able to list/enable/disable video codecs(like H.261,H.263).
Cheerz!
September 19th, 2008 at 1:11 am
ok ok I have setup all these….by default all are set…I configure it one to one manner along with full suport
September 19th, 2008 at 1:11 am
ok ok I have setup all these….by default all are set…I configure it one to one manner along with full suport but unable.
September 19th, 2008 at 1:17 am
Can you show me one video on this…because i am i doing a mistake which i cann’t detect.Through youe video i can do it easily.
September 19th, 2008 at 6:12 am
Thanks for the tutorial!!Good job!! Congratulations!! It is really helpful for me because I am a beginner configuring CUCME (in a recent project).
Anyway I have a pair of questions that I would like somebody expert in CUCME to answer:
1. According to what I have read so far, it seems that the only way to register a SIP phone regardless of being Cisco or non-Cisco is to previously configure in CUCME its MAC address along with user alias and authentication. So everything must be statically configured beforehand and a particular user is statically assigned to a particular device. I find it very rigid and a bit disappointing in my oppinion because in my company we have no fixed desktop and we have an Alcatel VoIP solution. You can log-in in any device you find with your username and password and transform it in your extension automatically (it is like Mobile Extension capability) with no previous confiuration at the PBX. There is no static and permament assignation between device and end user, which I find very flexible and confortable. Is it possible to implement it easily in a CUCME as well?
2. Does anybody know how to register H.323 endpoint in the CUCME? I would like CUCME implement a inner gatekeeper in the same way as the SIP registrar does.
Sorry for this long post and thanks for your attention. It is really appreciated.
September 25th, 2008 at 8:11 am
@hazra, please illustrate your configuration and let me know what exact error you are getting now. If you are still getting “Codec Mismatch” , then you must match Video Codec between CUCM and Eyebeam.If you are past this step and have some other error, give me details and i will reply accordingly.
Thanks
September 25th, 2008 at 8:20 am
Victor, Thanks for your liking.
As far as your first question goes, “Extension Mobility” feature that you are referring to has been in Cisco Call Manager (CCM) 4.x and Cisco Unified Communication Manager versions (CUCM) 5.x/6.x and recently has been ported over onto Cisco Unified Communication Manager Express (CUCME) .You will need Cisco Unified CME 4.2 or a later version for this.
Here is the configuration guide for setting up Extension Mobility in CUCME.
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmemobl.html
September 25th, 2008 at 8:31 am
As for your second question, you can setup and configure CUCME for H.323 endpoints including Tandberg and PolyCom H.323 Video Endpoints.You will need to set this up in H.323 voice-service configuration mode in CUCME.
For details and configuration guide, hop on over to this link:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmevideo.html#wp1017496
Thanks!
October 21st, 2008 at 11:20 am
Hi Admin:
Amazing guide. Thanks a lot for doing it.
Only one question:
Instead of dial-peers how do I manage the outgoing calls from sip phones??
Thanks!!!
November 16th, 2008 at 6:37 pm
@Charly, Thanks for liking.
Basically, if you need to route calls outside your CME router, you will require “an Outgoing Call Leg” aka a dial-peer to set this up.This holds true for both SIP/H.323 calls. In case of MGCP, Call Manager (CUCM) routes calls using Route Patterns.
November 17th, 2008 at 4:02 pm
Thanks for the help and the video.
Amazing stuff.
I got a question related to giving users the ability to log into their phones and logging off when they are not on their stations, how can this be done in CCME and is their a CDR or billing software that can extract the details in case their is misuse of Int’l calls.
Thanks!!!!
November 20th, 2008 at 2:02 am
@Emad, Thanks for appreciation.
If you want your users to have the ability to roam around,you can use Extension Mobility feature in CUCME . Also any International/Toll-charged calls can be restricted using COR (Class-Of-Restrictions).You can also configure a dial code to allow access to international calls for only a selected number of people(managers,ceo etc.).CDR can be used in CCM to provide billing details and other stuff.
Hope this helps!
December 18th, 2008 at 10:16 am
Hi,
Im stuck with command: voice registar global.
my IOS is c3725-ipvoice-mz.123-14.T7…
do i need new IOS? which you use?
December 22nd, 2008 at 7:47 pm
Arma, use may want to use “Cisco’s Feature Navigator feature” to verify this.However, upgrade your IOS to 12.4 Main Train versions and you should be good to go.
FYI,The IOS image used in this tutorial is : c3745-adventerprisek9_ivs-mz.124-15.T7.bin
Cheerz
January 9th, 2009 at 5:29 am
I been able to register X-lite phone but no calls could be made. so download SJP Phone and then tried. In this case i did not see any registration on the console. but i was able to make calls.
January 9th, 2009 at 5:53 am
i check the codec also but i am getting following error.
1250 : 55 17420070ms.1 +-1 pid:2 Originate 6666 connecting
dur 00:00:00 tx:0/0 rx:0/0
IP 0.0.0.0:0 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
1256 : 58 17468650ms.1 +-1 pid:2 Originate 6666 connecting
dur 00:00:00 tx:0/0 rx:0/0
IP 0.0.0.0:0 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
January 13th, 2009 at 3:04 am
Hi Admin,
Ur video is great & inftormative..
I have one question?
i have configured that as describled in video but i got error 401 anuthorize
in cisco phone we require load files something like that but in sip which files we needed..
Please guide………………..
Thanks in advance,
Pravin
April 3rd, 2009 at 7:50 am
Thanks it’s really awesome
and very helpful
April 4th, 2009 at 1:22 am
@Chikkis,Nice to know that your problem resolved.
@Pravin,there are different files for each IP phone model that you would need to load.If you are trying to covert your existing SCCP phone images to SIP based images, this URL will be helpful to you.
http://www.cisco.com/application/pdf/paws/5455/handset_to_sip.pdf
Thanks
April 4th, 2009 at 3:39 am
Thankz
May 25th, 2009 at 2:13 am
@all,
i already installed the x-lite sip phone and now i’ve difficult to find the mac address. how can get this mac address ? thanks b4
October 1st, 2009 at 2:34 am
I have CME 2801. I have registered 2 phones to it one is Cisco 7940 ( SCCP) Ext: 101 & other is Snom 320 ( SIP) Ext: 102. Both the phones are registered to CME. I can make a call from 102 t0 101 but if i make a call from 101 to 102 it shows UNKNOWN Number.
October 1st, 2009 at 2:37 am
I have CME 2801. I have registered 2 phones to it one is Cisco 7940 ( SCCP) Ext: 101 & other is Snom 320 ( SIP) Ext: 102. Both the phones are registered to CME. I can make a call from 102 t0 101 but if i make a call from 101 to 102 it shows UNKNOWN Number.
Can somebody help me with this..?
The SIP configuration is given below..
voice register global
mode cme
source-address 192.168.30.1 port 5060
max-dn 144
max-pool 24
create profile sync 0605115427714802
!
voice register dn 10
number 102
name phone2
label Phone2: 102
mwi
!
voice register pool 1
id mac 0004.1324.A8EB
number 1 dn 10
dtmf-relay sip-notify
username cisco password cisco
codec g711ulaw
!
March 4th, 2010 at 6:30 pm
It´s not possible to see the video again? Thanx 4 ur website
April 13th, 2010 at 10:57 am
Where is the tutorial for GNS3: How to Register third party SIP phone to Cisco Unified Call Manager Express (CUCME)?
There is no tutorial link?
Thanks