Setting up my Voice Lab.

JeanM

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?

shodown

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.

JeanM

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!

shodown

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.

Jollycork

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.

JeanM

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?

JeanM

shodown wrote: »

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!

shodown

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.

JeanM

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/sw/voicesw/ps4625/products_configuration_example09186a00808f9666.shtml

Found this , giving it a shot http://blog.alwaysthenetwork.com/tutorials/sip-trunk-with-cme/

Jollycork

JeanM wrote: »

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] .

JeanM

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* ?

shodown

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

JeanM

shodown wrote: »

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#

JeanM

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

JeanM

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

JeanM

Success lol, by accident called someone in India vs my cell.





[TH]Destination[/TH]
[TH]Destination Name[/TH]
[TH]Caller-ID[/TH]
[TH]Start Time[/TH]
[TH]Duration[/TH]
[TH]Cost[/TH]


919167xxxxxx
INDIA - MOBILE - OTHER CARRIERS
2222
2014-1-22 08:36:54
0:00:09
$0.00460000

JeanM

https://support.flowroute.com/entries/22587729-What-dialing-format-should-I-use-when-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.......

Jollycork

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?

Jollycork

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]

JeanM

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.

MAC_Addy

JeanM, thanks for posting all of this! Very valuable information for people getting into Voice.

JeanM

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).