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  1. Senior Member JeanM's Avatar
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    #1

    Default Setting up my Voice Lab.

    Hi,

    So I've been playing with cme to learn the basics etc, and just picked up another router / switch to set up a lab.

    So far I've gotten the following -

    VoIP phones - 2 7940 phones, 1 7960 phone.
    Switches - two 3550 poe switches
    Routers - two 2811 routers (I also do have a 3725 that I am hoping I can use in between possibly as vpn or some kind of gateway?)

    With the 2811 + 3550 poe I was able to get older version of CME running on the router and registered the phones, got them to call / talk to each other . With the additional 2811/3550 I am hoping to play with TWO cme routers working as "two sites".

    I purchased a CME lab guide, and plan on going through all the scenarios in it for labbing to get my feet wet.

    Please let me know what other components I should look for?
    Last edited by JeanM; 01-22-2014 at 03:00 AM.
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    #2
    That should be all you need for now. One thing you can do is try to link your CME up to a SIP trunking provider and see if you can get calls to work inside and outside of your home. That way you can use translation rules and dial-peers to get really use to how things work. I would skip getting a CUE module as they are expensive and would only recommend them if you are going to run into them often. I would turn the 3725 into a PSTN Router/frame device so you can dial in between the sites. I would say you can use the 3725 as a gatekeeper device as its still tested in the CCNP Voice, but its now gone from CCIE Collaboration.
    Currently Reading

    CUCM SRND 9x/10, UCCX SRND 10x, QOS SRND, SIP Trunking Guide, anything contact center related
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  4. Senior Member JeanM's Avatar
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    #3
    shodown - thank you very much for your reply/input! Glad to hear I am at least going in the right direction with gear selection !

    I don't quite understand, yet, how the SIP trunk would work. I don't even have dial up line at home , would that be required and then I would need to get a module that would let me connect the CME to my home phone line correct?

    I'll also read up on PSTN router/frame as well.

    thank you!
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    #4
    SIP trunk is a full IP connection to the PSTN. You will not even need a analog line. What you are doing in a sense is extending your IP telephone system into the internet. You calls will go over the regular internet to a provider and they will provide the PSTN connection. This is very good for getting rid of PRI's at every location and only having centralized dialing out of your data centers.
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    CUCM SRND 9x/10, UCCX SRND 10x, QOS SRND, SIP Trunking Guide, anything contact center related
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    #5

    Default Might consider FXS/FXO cards plus a NM-4AS

    I use the NM-4AS on a 2600 for frame relay between sites. The FXS/FXO allows you to hook up to you landline phone if you have one.

    Some use the Atlas Adtran to simulate central office FXO PSTN some like PitViper create their own [guy's really a voice wizard].


    I use 1760s with PVDMs because their cheap rather than 2811s and run old CME on them. They work and allow learning the basics.

    If you want to work with Callmanager, Unity voice mail check Ebay. There are some selling VM of Callmanager with Unity and well worth it.
    Last edited by Jollycork; 01-21-2014 at 10:06 PM.
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  7. Senior Member JeanM's Avatar
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    #6
    I do have a couple NM-4T and NM-4A/S modules I can stick in either a 2600 or 3600 / 3725 that I already have. I don't have any FXS/FXO modules/cards at all though. The 2811 I have already has PVDM and CME is up and running, but it's old.

    Now, I did also set up an ESXi hypervisor in a dedicated box with 4 port nic, so I maybe can potentially use it for CUCM/Unity later on.

    Right now, I really just want to concentrate on getting the VoIP call/ing down. I have the basics working now, but it's all internal lab only, it would be nice to also understand how I can make it work outside.... or over PSTN.

    For example, how would a lab CME router, route an internal extension/call over PSTN home phone number to say another land line or another CME lab router?
    Last edited by JeanM; 01-21-2014 at 10:42 PM.
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  8. Senior Member JeanM's Avatar
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    #7
    Quote Originally Posted by shodown View Post
    SIP trunk is a full IP connection to the PSTN. You will not even need a analog line. What you are doing in a sense is extending your IP telephone system into the internet. You calls will go over the regular internet to a provider and they will provide the PSTN connection. This is very good for getting rid of PRI's at every location and only having centralized dialing out of your data centers.
    Nice! How does one go about getting a SIP trunk , is this something I have to contact my ISP provider or look for any firm out there that provides / sells this service to anybody?

    Thanks!
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    #8
    There are plenty of them out there. The one I recommend is flowroute. I use them for our internal network and for several SMB clients.
    Currently Reading

    CUCM SRND 9x/10, UCCX SRND 10x, QOS SRND, SIP Trunking Guide, anything contact center related
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  10. Senior Member JeanM's Avatar
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    #9
    Signed up to to explore this! thanks!

    Now reading this, how to setup CME router with SIP TRUNK.

    http://www.cisco.com/en/US/products/...808f9666.shtml

    Found this , giving it a shot http://blog.alwaysthenetwork.com/tut...runk-with-cme/
    Last edited by JeanM; 01-22-2014 at 04:47 AM.
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    #10
    Quote Originally Posted by JeanM View Post
    I do have a couple NM-4T and NM-4A/S modules I can stick in either a 2600 or 3600 / 3725 that I already have. I don't have any FXS/FXO modules/cards at all though. The 2811 I have already has PVDM and CME is up and running, but it's old.

    Now, I did also set up an ESXi hypervisor in a dedicated box with 4 port nic, so I maybe can potentially use it for CUCM/Unity later on.

    Right now, I really just want to concentrate on getting the VoIP call/ing down. I have the basics working now, but it's all internal lab only, it would be nice to also understand how I can make it work outside.... or over PSTN.

    For example, how would a lab CME router, route an internal extension/call over PSTN home phone number to say another land line or another CME lab router?
    Your NM modules can be used for a WAN link Frame relay or MPLS to multiple CMEs. The different CME books all have the different senarios for connecting different CMEs together. Just a matter of dial-peers. I have a single PSTN landline that I use the FXS/FXO card for dialing out on PSTN.

    I use that landline and FXS/FXO for dialing in to the Callmanager company directory.

    As Shodown mentions, just buy time on a SIP and you've got access to PSTN. you can also buy DIDs [direct inward dial] .
    Last edited by Jollycork; 01-22-2014 at 04:25 AM.
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  12. Senior Member JeanM's Avatar
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    #11
    Okay, trying to get my CME router to register with flowroute for the sip trunk.

    Using this example -

    "sip-ua
    authentication username xxxxx password 7 xxxxxxxxxx realm sip.flowroute.com calling-info pstn-to-sip from number set 1xxx7325736 no remote-party-id registrar dns:sip.flowroute.com expires 3600"

    I replaced the username and password with mine, what do I need to do with the 1xxx7325736 part, is that the Prefix provided by flowroute Such as 12222345* ?
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    #12
    Here is a copy of mine. Replace my username and password with yours. I've also added some dial peers

    sip-ua
    authentication username 00***16* password 7 040C2C36****5E****0E3****3
    registrar dns:sip.flowroute.com expires 3600
    sip-server dns:sip.flowroute.com
    presence enable


    dial-peer voice 1003 voip
    description ** local dialing dial peer **
    translation-profile outgoing outbound
    destination-pattern 9[2-9]..[2-9]......
    session protocol sipv2
    session target sip-server
    voice-class sip dtmf-relay force rtp-nte
    dtmf-relay rtp-nte
    ip qos dscp ef signaling
    no vad
    !
    dial-peer voice 1004 voip
    description ** long distance dialing dial peer **
    translation-profile outgoing outbound
    destination-pattern 91[2-9]..[2-9]......
    session protocol sipv2
    session target sip-server
    voice-class sip dtmf-relay force rtp-nte
    dtmf-relay rtp-nte
    ip qos dscp ef signaling
    no vad
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  14. Senior Member JeanM's Avatar
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    #13
    Quote Originally Posted by shodown View Post
    Here is a copy of mine. Replace my username and password with yours. I've also added some dial peers

    sip-ua
    authentication username 00***16* password 7 040C2C36****5E****0E3****3
    registrar dns:sip.flowroute.com expires 3600
    sip-server dns:sip.flowroute.com
    presence enable


    dial-peer voice 1003 voip
    description ** local dialing dial peer **
    translation-profile outgoing outbound
    destination-pattern 9[2-9]..[2-9]......
    session protocol sipv2
    session target sip-server
    voice-class sip dtmf-relay force rtp-nte
    dtmf-relay rtp-nte
    ip qos dscp ef signaling
    no vad
    !
    dial-peer voice 1004 voip
    description ** long distance dialing dial peer **
    translation-profile outgoing outbound
    destination-pattern 91[2-9]..[2-9]......
    session protocol sipv2
    session target sip-server
    voice-class sip dtmf-relay force rtp-nte
    dtmf-relay rtp-nte
    ip qos dscp ef signaling
    no vad

    I get the following when I issue #sh sip-ua register status

    Line peer expires(sec) registered
    ================================ ========== ============ ==========
    1111 20002 94 no
    2222 20001 113 no
    3333 20003 113 no
    CMERouter#
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  15. Senior Member JeanM's Avatar
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    #14
    flowroute control panel states "You have no active registrations on this proxy."

    But, maybe SIP trunk IS working.... but it's just showing no next to my local ephone numbers because they are for internal lab local use only right?

    216.115.69.144 is flowroute, 70.167.153.130 is wsip-70-167-153-130.oc.oc.cox.net

    CMERouter#sh sip-ua connections udp detail
    Total active connections : 2
    No. of send failures : 0
    No. of remote closures : 0
    No. of conn. failures : 0
    No. of inactive conn. ageouts : 0


    ---------Printing Detailed Connection Report---------
    Note:
    ** Tuples with no matching socket entry
    - Do 'clear sip <tcp[tls]/udp> conn t ipv4:<addr>:<port>'
    to overcome this error condition
    ++ Tuples with mismatched address/port entry
    - Do 'clear sip <tcp[tls]/udp> conn t ipv4:<addr>:<port> id <connid>'
    to overcome this error condition


    Remote-Agent:70.167.153.130, Connections-Count:1
    Remote-Port Conn-Id Conn-State WriteQ-Size
    =========== ======= =========== ===========
    5060 3 Established 0


    Remote-Agent:216.115.69.144, Connections-Count:1
    Remote-Port Conn-Id Conn-State WriteQ-Size
    =========== ======= =========== ===========
    5060 2 Established 0
    Last edited by JeanM; 01-22-2014 at 06:41 AM.
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  16. Senior Member JeanM's Avatar
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    #15
    CMERouter#sh sip-ua calls
    SIP UAC CALL INFO

    Call 1
    SIP Call ID : 978A3AB8-827E11E3-8047AE76-B5BB168@172.16.2.1
    State of the call : STATE_SENT_INVITE (3)
    Substate of the call : SUBSTATE_NONE (0)
    Calling Number : 2222
    Called Number : 918553569768
    Bit Flags : 0x400018 0x100 0x200
    CC Call ID : 102
    Source IP Address (Sig ): 172.16.2.1
    Destn SIP Req Addr:Port : [70.167.153.130]:5060
    Destn SIP Resp Addr:Port: [70.167.153.130]:5060
    Destination Name : sip.flowroute.com
    Number of Media Streams : 1
    Number of Active Streams: 1
    RTP Fork Object : 0x0
    Media Mode : flow-through
    Media Stream 1
    State of the stream : STREAM_ACTIVE
    Stream Call ID : 102
    Stream Type : voice+dtmf (1)
    Stream Media Addr Type : 1
    Negotiated Codec : No Codec (0 bytes)
    Codec Payload Type : 255 (None)
    Negotiated Dtmf-relay : inband-voice
    Dtmf-relay Payload Type : 0
    Media Source IP Addr:Port: [172.16.2.1]:17636




    Options-Ping ENABLED:NO ACTIVE:NO
    Number of SIP User Agent Client(UAC) calls: 1


    SIP UAS CALL INFO


    Number of SIP User Agent Server(UAS) calls: 0


    ============================================


    CMERouter#sh sip-ua status
    SIP User Agent Status
    SIP User Agent for UDP : ENABLED
    SIP User Agent for TCP : ENABLED


    SIP User Agent for TLS over TCP : ENABLED
    SIP User Agent bind status(signaling): DISABLED
    SIP User Agent bind status(media): DISABLED
    SIP early-media for 180 responses with SDP: ENABLED
    SIP max-forwards : 70
    SIP DNS SRV version: 2 (rfc 2782)
    NAT Settings for the SIP-UA
    Role in SDP: NONE
    Check media source packets: DISABLED
    Maximum duration for a telephone-event in NOTIFYs: 2000 ms
    SIP support for ISDN SUSPEND/RESUME: ENABLED
    Redirection (3xx) message handling: ENABLED
    Reason Header will override Response/Request Codes: DISABLED
    Out-of-dialog Refer: DISABLED
    Presence support is ENABLED
    protocol mode is ipv4


    SDP application configuration:
    Version line (v=) required
    Owner line (o=) required
    Timespec line (t=) required
    Media supported: audio video image
    Network types supported: IN
    Address types supported: IP4 IP6
    Transport types supported: RTP/AVP udptl
    Last edited by JeanM; 01-22-2014 at 07:29 AM.
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  17. Senior Member JeanM's Avatar
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    #16
    Success lol, by accident called someone in India vs my cell.



    Destination Destination Name Caller-ID Start Time Duration Cost
    919167xxxxxx INDIA - MOBILE - OTHER CARRIERS 2222 2014-1-22 08:36:54 0:00:09 $0.00460000
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  18. Senior Member JeanM's Avatar
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    #17
    https://support.flowroute.com/entrie...sending-calls-

    if I dial 9 + area code + phone number = fast busy
    if I dial 9 + 1 + area code + phone number = worked as international call vs. local
    if I dial area code + 7 digit phone number = fast busy
    if I dial 1 + area code = fast busy as soon as I enter first digit after 1

    After some trial and error, got it to work after changing the destination-pattern to 1916.......
    Last edited by JeanM; 01-22-2014 at 08:55 AM.
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    #18

    Default translation patterns.

    It could be you need to tell CME to strip some digits before you send it out your SIP so SIP knows what to do with the received info.

    The IDD from the US is 011 - "country code" + area code + number.

    India country code is 91 so there's how you got India when you dialed 91....

    note typically dialing 9 gets you the dial tone as the access code, then you dial the number. either the local number prefix and number
    or dial 1+area code+prefix+number.

    So CME doesn't know to drop the 9 access code


    Are your local phones on net or off net?
    Last edited by Jollycork; 01-22-2014 at 11:41 PM.
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    #19
    added note: there's 2 ways with Callmanager. your calls could be off net 9,[2-9] local and off net 9,1[2-9] for SIP for your phone type 794X &796X
    where 9 is dialed for the secondary dial tone

    or 9.[2-9] and 9.1[2-9]
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  21. Senior Member JeanM's Avatar
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    #20
    Thanks! I've been calling same area code numbers (land and cell phone) using 10 digit dialing, and just checking the sip call info. Pretty interesting stuff, I need to configure CME to save the call stuff into syslog or something if possible.
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  22. Happiness is !!!!! MAC_Addy's Avatar
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    #21
    JeanM, thanks for posting all of this! Very valuable information for people getting into Voice.
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  23. Senior Member JeanM's Avatar
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    #22

    Default MAC_Addy

    MAC_Addy - no problem! I got my lab guide, and next thing is to configure my 2nd 2811/3550 combo as "remote" CME, then connect them via vpn tunnel (got that part working over the weekend).
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