This tutorial will demonstrate process of setting up a Session Initiatition Protocol (SIP) trunk on Cisco Unified Express Communication Manager Express (CUCME) to SIP Voice Over IP (VOIP) Service Provider. It will be divided into several parts. This is first part in which you will learn to setup Basic SIP trunking with VOIP service provider. This will be followed by a more Advanced  configuration tutorial which will help you in implementing an IP-based telephony system with CME using SIP trunking for inbound and outbound calls using SIP VOIP dial-peers.

CME SIP Trunk to VOIP Service Provider Scenario

  • Part 1a –> Basic Setup of CME SIP trunk to VOIP Service Provider
  • Part 1b –> Advanced Configuration for both inbound and outbound dial-peers
  • Part 2  –> Setting up  Cisco IP Commuicator (CIPC) for making/receiving calls to/from SIP VOIP SP
  • Part 3  –> Setting up  SIP softphone for making/receiving calls to/from SIP VOIP SP

To Sign-up for a Free SIP VOIP Internet Phone Service  Account ,click here

Here comes the Basic Setup Tutorial:

Download

Enjoy!

Related Posts

PDF    Send article as PDF   

Tags: , , , , , , , , , , , , , , , , , ,

27 Responses to “GNS3:How to Setup CME SIP trunk to VOIP SIP Service Provider – Part 1”

  1. Joe says:

    Amazing Tutorial ! I cannot wait to see other parts rolling in !

    Thumbs up for you !

  2. phlaverdj says:

    thank you

  3. Ak says:

    Hey we have not seen anything new for a while. Are you gona release like 5 videos with in the next month?

  4. admin says:

    @Ak, Rest of videos will be up shortly. I had been intensely busy with couple of VOIP projects so couldn’t get time earlier.

  5. AK says:

    Thanks for the update. I check your site every day. I enjoy all of your videos.

  6. Gopi says:

    Wonderfulll……. very useful us.

    Thanks For the update

  7. Khalid SHiakh says:

    is there any way to save this Tutorial:

    Khalid.

  8. Mauricio says:

    Thank you for the tutorial. Very interesting, but could you show the other configurations in order to make a call from IP Communicator, e.g. to the outside world. I am a little confused about Voice Translation Rules, and I am trying to set up a VoipBuster trunk. My sip-ua is registered, as yours, but I can not make the correct translation rule in combination with the destination-pattern in the dial-peer voip.
    Thank you for any help you could provide.

  9. admin says:

    Thanks for comments.I will put up another tutorial for CIPC too. So, keep checking !

    Cheerz

  10. admin says:

    @Khalid, download links for tutorials will be available soon.

  11. Mauricio says:

    I signed up for Call Centric but when I try to make a test call , I get Busy Tone and the message Unknown Number. No call is possible. I don’t know what is bad in my configuration. I followed all your steps.
    Thanks.

  12. admin says:

    @Mauricio, Are you attempting to call using your IP Communicator ?

  13. gopi says:

    we will wait for your sip based tutorial
    pls spend time for us….

    Thanks in advance.

  14. Chikkis says:

    thanks for nice tutorials. looking at this tutorials and trying some new in it.

    I have asterisk box on vmware. I am trying to make a trunk in between them.

    here few details of the lab.
    1. asterisk box on VMWare ip 192.168.2.14
    2. XP Hostmachine ip address 192.168.2.11
    3. 3725 Router with CME on GNS3 ip 192.168.2.20
    4. insatlled Bind 9.6 working fine.

    However i and trying to add the DNS a record on the DNS server.

    Lets hope i should be able finish this lab.

    Thanks
    Chikki

  15. Mauricio says:

    Yes, I am using IP Communicator. Sip-ua register status is Yes and I try to make a test call to 17771234567 that is a test number that they have. Dial-peer destination pattern is 1777……. but unknown number message I get.
    Do I need another dial peer fo incoming calls??
    Thank you.

  16. Fahad says:

    Its Great, thanks a lot for such a great work hope for more in VOIP,because there is a lot shortage of VOIP lab scenario for practice.

    Kindly do more for us, i am interested in to make CVOICE labs kindly help me in Gns3.

    Which IOS i will upload to do all labs.

    Regards,
    Fahd

  17. Ali says:

    Good tutorial admin. when is your next parts coming. It will really help me for my poject. if you can release them i would be very greatfull

    thanks

  18. Malik says:

    Hi

    I am confused at one point
    You know on Xlite, there are 3 options
    1.Username
    2.Password
    3.Authorization Username

    Your video explains the first 2…what if the 3rd parameter is not same as Username…how can we define that on CME

    Moreover, how to specify the domain proxy

    Waiting for your help

  19. admin says:

    @Chikkis, Hope you have been able to complete your topology by now.

    @Malik, i will cover authorization part in upcoming videos.I have been busy for my CCIE Voice lab preparation and couldn’t get time to complete other parts.

    Hopefully,you guys will see other parts rolling in soon.Thanks

  20. Sara says:

    Keep up the good work! Gr8.Thnx

  21. mac says:

    Hi, excellent tutorial. I have followed the steps and configured my CME router for callcentric. However I get SIP/2.0 403 Incorrect Authentication in the debug log

  22. Rafiq says:

    Great tutorials admin. But couldnot download !

  23. enthus3 says:

    good tutorial, What version of router code was used for this tutorial?

  24. Tariq Ahmad says:

    Thanks enthus3 and others.Here is the info you are looking for :

    IOS Version : 12.4(15)T , Platform :3725/374

  25. Terrance Joesph says:

    Your blog its amazing thx a lot !

  26. [...] information.You can register for SkypeConnect service here. If you want to register for another SIP Voip Service provider & test interoperability with SkypeConnect Service, here’s the [...]

  27. Hazem says:

    thanks brother

Leave a Reply

hide totop